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GIII User Guide Version 12.1.0 software
Revision 1.19
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PROPRIETARY INFORMATION The information contained in this document is the property of Aculab plc and may be the subject of patents pending or granted, and must not be copied or disclosed without prior written permission. It should not be used for commercial purposes without prior agreement in writing. All trademarks recognised and acknowledged. Aculab plc endeavours to ensure that the information in this document is correct and fairly stated but does not accept liability for any error or omission. The development of Aculab’s products and services is continuous and published information may not be up to date. It is important to check the current position with Aculab plc. Copyright © Aculab plc. 2002-2019 all rights reserved.
Document Revision Rev Date By Detail 1.0 11.10.17 ac First issue of GIII User Guide 1.1 21.11.17 ebj Changed to corporate fonts and reformatted 1.2 22.11.17 ac Updates 1.3 23.11.17 ebj Reformatted 1.4 23.11.17 ac Remove G.729 license requirements
1.5 28.11.17 ac Add more error pages to appendix. Update Clock Sources image. Add roles usage example.
1.6 01.12.17 ac Add cluster status page. Add warnings about certain clustering operations.
1.7 07.12.17 ac Update Alarm Mapping page. Minor updates. 1.8 10.01.18 ac Update Web services help transcript. 1.9 25.01.18 ac Update card statuses showing firmware. 1.10 1.02.18 ac Update SS7 Advanced configuration images.
1.11 06.04.18 ac Add cluster start dependency information. Update images to R740.
1.12 10.04.18 ac Amend cluster start dependency information. 1.13 13.04.18 ac Stop web services help example being a hyperlink
1.14 09.04.19 ac Change default IP address information. Update copyright date.
1.15 18.04.19 ac RFC3355 LDAP Authentication
1.16 30.04.19 ac Add information regarding LDAP admin user access, and IPv6 server addresses.
1.17 28.05.19 ac/ebj Change default IP address information for GIII Admin interface.
1.18 29.05.19 ac/ebj Update for 12.1 SIGTRAN (RFC3356). 1.19 20.11.19 ebj Updated SS7 Figures
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CONTENTS
1 Introduction ........................................................................................................................... 19
2 System overview ................................................................................................................... 20 2.1 Chassis installation guides ....................................................................................................... 20 2.2 Signalling protocols supported ................................................................................................ 20 2.3 GIII web interface ....................................................................................................................... 20 2.4 Activation key mechanism ........................................................................................................ 20 2.5 Principles of operation .............................................................................................................. 21 2.6 GIII IP telephony principles ....................................................................................................... 21 2.7 2U carrier grade trunk numbering ............................................................................................ 22
2.7.1 Dual trunk breakout adapters ........................................................................................................ 22 2.7.2 Trunk numbering............................................................................................................................ 22
2.8 System configuration................................................................................................................. 23
3 System boot-up ..................................................................................................................... 25 3.1 Overview. .................................................................................................................................... 25 3.2 Initial hardware checks. ............................................................................................................. 25 3.3 Basic fault finding ...................................................................................................................... 26
4 Initial Configuration .............................................................................................................. 27 4.1 Cluster the device ...................................................................................................................... 27
4.1.1 Configure Cluster ........................................................................................................................... 27 4.1.2 Joining a Cluster ............................................................................................................................ 27
4.2 Create a new role ....................................................................................................................... 28 4.2.1 Role configuration order ................................................................................................................ 29
4.3 IP telephony network configuration ......................................................................................... 29 4.4 Media IP Configuration .............................................................................................................. 30
5 GIII Role Configuration ......................................................................................................... 31 5.1 Role Configuration options ....................................................................................................... 31 5.2 Role Management....................................................................................................................... 32
5.2.1 Create a new Role ......................................................................................................................... 33 5.2.1.1 Create a new role from an existing role ............................................................................. 34 5.2.1.2 Create a new role for a GroomerII backup ......................................................................... 34
5.2.2 Apply Role ..................................................................................................................................... 38 5.2.2.1 Immediately applied Role changes .................................................................................... 40
5.2.3 Delete Role .................................................................................................................................... 41 5.2.4 Revert Role .................................................................................................................................... 42 5.2.5 Display Revision ............................................................................................................................ 43 5.2.6 Comments ..................................................................................................................................... 43
5.3 General configuration ................................................................................................................ 44 5.3.1 Clock Sources................................................................................................................................ 44 5.3.2 Call Setup Timers .......................................................................................................................... 45 5.3.3 Alarm mapping............................................................................................................................... 46 5.3.4 Advanced options .......................................................................................................................... 49
5.4 Trunks configuration ................................................................................................................. 50 5.4.1 SIP Configuration........................................................................................................................... 51
5.4.1.1 SIP Trunk Setup ................................................................................................................. 51 5.4.1.2 SIP - Call Handling ............................................................................................................. 53 5.4.1.3 SIP - Default codecs .......................................................................................................... 54 5.4.1.4 SIP - Authentication ........................................................................................................... 55 5.4.1.5 SIP - gateways ................................................................................................................... 56 5.4.1.6 Protocol Settings ................................................................................................................ 62 5.4.1.7 SIP Proxy ........................................................................................................................... 66
5.4.2 TDM Trunks ................................................................................................................................... 67 5.4.2.1 Trunk Setup tab.................................................................................................................. 67
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5.4.2.2 Edit an option setting .......................................................................................................... 70 5.4.2.3 Protocol configuration for SS7 ............................................................................................ 71 5.4.2.4 Call Handling ...................................................................................................................... 76
5.5 SS7 Configuration ...................................................................................................................... 79 5.5.1 Add pointcode tab .......................................................................................................................... 80
5.5.1.1 SS7 configuration error messages ..................................................................................... 83 5.5.2 Advanced configuration tab ............................................................................................................ 83 5.5.3 Interworking tab .............................................................................................................................. 85
5.5.3.1 International tab .................................................................................................................. 85 5.5.3.2 ANSI tab options ................................................................................................................. 86 5.5.3.3 UK ISUP tab options ........................................................................................................... 87
5.5.4 Point code tab ................................................................................................................................ 88 5.6 Routing configuration ................................................................................................................ 89
5.6.1 Groups configuration ...................................................................................................................... 89 5.6.1.1 Group Configuration ........................................................................................................... 90
5.6.2 Routes configuration ...................................................................................................................... 93 5.6.2.1 Route Setup tab .................................................................................................................. 94 5.6.2.2 Routes – Q.931/ISUP ....................................................................................................... 101 5.6.2.3 Routes – CAS ................................................................................................................... 106 5.6.2.4 Routes – DPNSS .............................................................................................................. 108 5.6.2.5 Routes Incoming and Outgoing SIP calls ......................................................................... 109 5.6.2.6 Routes – General tab ....................................................................................................... 115
5.6.3 Cause Maps configuration ........................................................................................................... 118 5.7 SIP TLS certificates .................................................................................................................. 120 5.8 Call Trace .................................................................................................................................. 121 5.9 Trunk Blocking ......................................................................................................................... 121
5.9.1 Trunk blocking for ITU-T, China and UK ISUP ............................................................................. 123 5.9.2 Trunk blocking for ANSI SS7 and non-SS7 TDM trunks .............................................................. 124 5.9.3 Trunk blocking for IP telephony trunks ......................................................................................... 124
5.10 SS7 signalling links .................................................................................................................. 125 5.11 SNMP reporting ........................................................................................................................ 125
5.11.1 Ports tab options .......................................................................................................................... 127 5.11.2 System tab options ....................................................................................................................... 128 5.11.3 Chassis tab options ...................................................................................................................... 129 5.11.4 Call Monitoring tab options ........................................................................................................... 131
6 GIII Maintenance.................................................................................................................. 133 6.1 Node Maintenance Mode ......................................................................................................... 133 6.2 Trunk Download/Reset ............................................................................................................. 134 6.3 SS7 Circuit Reset ...................................................................................................................... 134 6.4 SS7 Continuity checking ......................................................................................................... 134 6.5 IP diagnostics ........................................................................................................................... 136
6.5.1 Ping .............................................................................................................................................. 137 6.5.2 Traceroute .................................................................................................................................... 137
6.6 Trace Files ................................................................................................................................. 137 6.6.1 Common behaviour ...................................................................................................................... 138
6.6.1.1 View .................................................................................................................................. 138 6.6.1.2 Download .......................................................................................................................... 139
6.6.2 Call Trace Files ............................................................................................................................ 140 6.6.3 SS7 Trace Files ............................................................................................................................ 141 6.6.4 SIP Trace Files ............................................................................................................................. 141 6.6.5 Packet Capture Trace Files .......................................................................................................... 142
6.6.5.1 Host Address(es) .............................................................................................................. 143 6.6.5.2 Ports ................................................................................................................................. 143 6.6.5.3 Interface ............................................................................................................................ 144
6.6.6 Web System ................................................................................................................................. 144 6.7 Technical Support .................................................................................................................... 145
7 GIII Status monitors ............................................................................................................ 147 7.1 System Summary ..................................................................................................................... 147
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7.1.1 GIII back panel image .................................................................................................................. 149 7.1.1.1 TDM Trunk Status ............................................................................................................ 149 7.1.1.2 IP interface status ............................................................................................................ 149 7.1.1.3 PSU status ....................................................................................................................... 149 7.1.1.4 System LED ..................................................................................................................... 149
7.1.2 Node status.................................................................................................................................. 150 7.1.3 Role status ................................................................................................................................... 150 7.1.4 Admin Interface status ................................................................................................................. 150 7.1.5 Traffic Interface status ................................................................................................................. 150 7.1.6 Heartbeats status......................................................................................................................... 150 7.1.7 System status .............................................................................................................................. 150 7.1.8 Alarms status ............................................................................................................................... 150
7.2 Call Traffic ................................................................................................................................. 153 7.3 TDM Trunks .............................................................................................................................. 155 7.4 CAS Trunks ............................................................................................................................... 156 7.5 Ethernet Trunks ........................................................................................................................ 157 7.6 Call Trace .................................................................................................................................. 157 7.7 SS7 ............................................................................................................................................ 158
7.7.1 MTP3 Signalling........................................................................................................................... 158 7.7.2 ISUP Bearers ............................................................................................................................... 158 7.7.3 MTP3 Routes ............................................................................................................................... 159 7.7.4 MTP3 Destinations ...................................................................................................................... 160 7.7.5 M3UA Connections ...................................................................................................................... 161 7.7.6 M3UA Connection Stats .............................................................................................................. 161 7.7.7 M3UA Addresses ......................................................................................................................... 162 7.7.8 M3UA Routing Key Stats ............................................................................................................. 162
7.8 SIP Gateways ............................................................................................................................ 163 7.9 NTP ............................................................................................................................................ 164 7.10 Temperatures and Fans ........................................................................................................... 164 7.11 Essential Processes................................................................................................................. 165 7.12 System card status .................................................................................................................. 165 7.13 Cluster Status ........................................................................................................................... 166
8 Node Administration........................................................................................................... 167 8.1 Configure Admin IP Address .................................................................................................. 167 8.2 Web Site TLS Certificate .......................................................................................................... 167
8.2.1 Installing new certificate............................................................................................................... 167 8.3 Disk Management ..................................................................................................................... 168
8.3.1 Replacing a defective drive.......................................................................................................... 168 8.3.2 SMART failures and alarms ......................................................................................................... 170
8.4 Card Setup ................................................................................................................................ 171 8.5 Node LED Control .................................................................................................................... 172 8.6 Rename a Node ........................................................................................................................ 172 8.7 Software Upgrades................................................................................................................... 173
8.7.1 System card updates ................................................................................................................... 173 8.7.2 How to upgrade node software .................................................................................................... 174
8.8 Software Activation key ........................................................................................................... 176 8.8.1 Viewing Activation key ................................................................................................................. 176 8.8.2 Changing or adding an Activation key ......................................................................................... 176
8.9 Software Licenses .................................................................................................................... 176 8.10 Documentation ......................................................................................................................... 177
9 Cluster Administration ....................................................................................................... 178 9.1 Administer Cluster ................................................................................................................... 178
9.1.1 Start a cluster............................................................................................................................... 178 9.1.2 Join a Cluster ............................................................................................................................... 179 9.1.3 Decluster a Node ......................................................................................................................... 180
9.1.3.1 Decluster a node that is currently unavailable ................................................................. 180 9.1.4 Change Cluster version ............................................................................................................... 181
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9.2 Administration of User Accounts............................................................................................ 181 9.2.1 Changing a User Password ......................................................................................................... 181 9.2.2 Create a new user account .......................................................................................................... 182 9.2.3 Modify user account ..................................................................................................................... 183 9.2.4 Delete user account ..................................................................................................................... 183
9.3 Administer CDRs (Call data record) ........................................................................................ 184 9.3.1 Generating a CDR report ............................................................................................................. 189
9.4 Configure NTP .......................................................................................................................... 190 9.4.1 Add NTP server ............................................................................................................................ 191 9.4.2 Delete NTP server ........................................................................................................................ 191
9.5 Configure SNMP Agent ............................................................................................................ 191 9.5.1 General ........................................................................................................................................ 192 9.5.2 Users ............................................................................................................................................ 192 9.5.3 Communities ................................................................................................................................ 193 9.5.4 Hosts ............................................................................................................................................ 194
9.5.4.1 SNMPv1 and SNMPv2c Traps ......................................................................................... 195 9.5.4.2 SNMPv3 Traps ................................................................................................................. 195
9.5.5 Access .......................................................................................................................................... 196 9.5.6 Configuring SNMP on GIII ............................................................................................................ 196
9.5.6.1 Configuration for SNMPv1/SNMPv2c ............................................................................... 196 9.5.6.2 Configuration for SNMPv3 ................................................................................................ 197
9.6 Configure Syslog ...................................................................................................................... 197 9.6.1 Add Syslog Server ....................................................................................................................... 197 9.6.2 Delete Syslog Server ................................................................................................................... 198
9.7 Configure LDAP ........................................................................................................................ 198 9.7.1 LDAP ............................................................................................................................................ 198
9.7.1.1 Server ............................................................................................................................... 199 9.7.1.2 Search .............................................................................................................................. 200 9.7.1.3 Login Page ....................................................................................................................... 201
9.7.2 Test LDAP .................................................................................................................................... 202 9.8 Backup & Restore ..................................................................................................................... 202
9.8.1 Backup ......................................................................................................................................... 202 9.8.1.1 Role Backup ..................................................................................................................... 202 9.8.1.2 System Backup ................................................................................................................. 203
9.8.2 Restore ......................................................................................................................................... 204 9.8.2.1 Role Restore ..................................................................................................................... 204 9.8.2.2 System Restore ................................................................................................................ 204
9.9 Restart & Shutdown ................................................................................................................. 205 9.9.1 Restart .......................................................................................................................................... 205 9.9.2 Shutdown ..................................................................................................................................... 205
10 Call Transfer ........................................................................................................................ 206 10.1 Call transfer mapping in GIII .................................................................................................... 206 10.2 ETS 300 Call Transfer to SS7 Call Redirection Mapping....................................................... 206
10.2.1 Overview ...................................................................................................................................... 206 10.2.2 Configuration ................................................................................................................................ 206 10.2.3 CDR generation ........................................................................................................................... 207
10.3 CAMA to SIP Call Transfer Mapping ....................................................................................... 207 10.4 SIP to CAMA Call Transfer Mapping ....................................................................................... 208
11 SIP telephony ...................................................................................................................... 209 11.1 IPv4 and IPv6 telephony .......................................................................................................... 209
11.1.1 Incoming call handling .................................................................................................................. 209 11.1.2 Outgoing call handling .................................................................................................................. 209 11.1.3 Call retry and recovery ................................................................................................................. 209
11.2 SIP security ............................................................................................................................... 209 11.2.1 Configuration ................................................................................................................................ 209 11.2.2 TLS ............................................................................................................................................... 210 11.2.3 SRTP ............................................................................................................................................ 210
11.3 SIP call recovery ....................................................................................................................... 211
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11.4 SIP call hold .............................................................................................................................. 212 11.5 SIP call redirection ................................................................................................................... 212 11.6 Functional Number mapping ................................................................................................... 213 11.7 ISDN subaddress mapping ...................................................................................................... 213 11.8 Interworking with TDM protocols ........................................................................................... 213
11.8.1 Interworking between SIP and SS7 ............................................................................................. 213 11.8.2 Interworking between SIP and ETS 300 ...................................................................................... 214 11.8.3 Interworking between SIP and CAMA ......................................................................................... 214
11.9 Additional IETF RFC support .................................................................................................. 214
12 SS7 telephony ..................................................................................................................... 215 12.1 SS7 stack file ............................................................................................................................ 215
12.1.1 Example configuration files .......................................................................................................... 218 12.2 SS7 trunk configuration ........................................................................................................... 221 12.3 Continuity checking ................................................................................................................. 221
13 CAMA telephony ................................................................................................................. 223 13.1 Configuration ............................................................................................................................ 223
13.1.1 Signalling firmware configuration ................................................................................................. 223 13.1.2 GIII application configuration ....................................................................................................... 223
13.1.2.1 Trunk configuration .......................................................................................................... 223 13.1.2.2 Route configuration ......................................................................................................... 224
13.2 Interworking with CAMA .......................................................................................................... 224 13.2.1 CAMA to SIP Interworking ........................................................................................................... 224 13.2.2 SIP to CAMA Interworking ........................................................................................................... 224 13.2.3 Interworking between CAMA and TDM ....................................................................................... 225
13.3 Disabling incoming CAMA call clearing ................................................................................. 225
14 Interworking supplementary services............................................................................... 226 14.1 Name identification supplementary services (CNIP/CONP) ................................................. 226
14.1.1 Q.SIG ........................................................................................................................................... 226 14.1.2 SIP 226 14.1.3 Q.SIG to Q.SIG interworking ....................................................................................................... 226 14.1.4 Interworking between Q.SIG and SIP .......................................................................................... 226 14.1.5 SIP to SIP interworking ................................................................................................................ 227
14.2 User to user signalling (UUS) .................................................................................................. 227 14.2.1 ETS 300 ....................................................................................................................................... 227 14.2.2 SS7 .............................................................................................................................................. 228 14.2.3 SIP 228 14.2.4 ETS 300 to ETS 300 interworking ............................................................................................... 228 14.2.5 SS7 to SS7 interworking .............................................................................................................. 229 14.2.6 Interworking between ETS 300 and SS7 ..................................................................................... 229 14.2.7 Interworking between ETS 300 and SIP ...................................................................................... 229 14.2.8 Interworking between SS7 and SIP ............................................................................................. 230 14.2.9 SIP to SIP interworking ................................................................................................................ 231
14.3 Multi level precedence and pre-emption (MLPP) ................................................................... 231 14.3.1 ETS 300 and Q.SIG ..................................................................................................................... 231 14.3.2 SIP 231 14.3.3 ETS 300/Q.SIG to ETS 300/Q.SIG interworking ......................................................................... 231 14.3.4 ETS 300/Q.SIG to SIP interworking ............................................................................................. 232 14.3.5 SIP to ETS 300/Q.SIG interworking ............................................................................................. 233 14.3.6 SIP to SIP interworking ................................................................................................................ 233 14.3.7 Interworking to protocols without MLPP support ......................................................................... 234 14.3.8 Call diversion and re-routing ........................................................................................................ 234 14.3.9 Call recovery ................................................................................................................................ 234
14.4 Additional calling party number ............................................................................................. 234 14.5 Connected party number (COLP/COLR) ................................................................................ 234
14.5.1 Q.931 ........................................................................................................................................... 235 14.5.2 SS7 .............................................................................................................................................. 235 14.5.3 SIP 235
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14.5.4 TDM interworking ......................................................................................................................... 235 14.5.5 TDM to SIP interworking .............................................................................................................. 235 14.5.6 SIP to TDM interworking .............................................................................................................. 236 14.5.7 SIP interworking ........................................................................................................................... 236
15 TTY/RTT mapping ............................................................................................................... 237 15.1 Configuring a TTY system ....................................................................................................... 237 15.2 Operating limits ........................................................................................................................ 238
15.2.1 Call capacity ................................................................................................................................. 238 15.2.2 Audio codec selection .................................................................................................................. 238 15.2.3 Secure SIP ................................................................................................................................... 238
15.3 RTT stream negotiation ........................................................................................................... 238 15.4 T.140 redundancy ..................................................................................................................... 239 15.5 Voice and hearing carry over .................................................................................................. 239 15.6 SIP call retry .............................................................................................................................. 239 15.7 SIP call recovery ....................................................................................................................... 239
16 Web Services ....................................................................................................................... 240
17 Advanced fault finding and maintenance ......................................................................... 241 17.1 Dongle not present ................................................................................................................... 241 17.2 Invalid activation key ............................................................................................................... 241 17.3 Status errors ............................................................................................................................. 241
17.3.1 Openin failed ................................................................................................................................ 241 17.3.2 Card clock stopped (TDM ports only) ........................................................................................... 241 17.3.3 NO SIGNAL DETECTED (TDM ports only) ................................................................................. 241 17.3.4 DISCONNECTED (IP ports only) ................................................................................................. 241 17.3.5 REMOTE ALARM INDICATION (TDM ports only) ....................................................................... 241 17.3.6 ALARM INDICATION SIGNAL (TDM ports only) ......................................................................... 242
17.4 Obtaining protocol trace from GIII (TDM ports only) ............................................................. 242 17.5 Using CDRs to monitor system performance ........................................................................ 242 17.6 Monitoring a single call through the GIII ................................................................................ 242 17.7 Common set-up problems ....................................................................................................... 243
17.7.1 System clock ................................................................................................................................ 243 17.7.2 Dedicated timeslots ...................................................................................................................... 243
17.8 Aculab technical support ......................................................................................................... 243
Appendix A: Web Site Concepts ............................................................................................. 244 A.1 Menus ........................................................................................................................................ 244 A.2 Tabs ........................................................................................................................................... 244 A.3 Accordions ................................................................................................................................ 244 A.4 Checkboxes .............................................................................................................................. 245 A.5 Hints .......................................................................................................................................... 245 A.6 Drop down selects .................................................................................................................... 245 A.7 Spinners .................................................................................................................................... 245 A.8 Sliders ....................................................................................................................................... 246 A.9 Date selector ............................................................................................................................. 246 A.10 Drag and drop boxes ................................................................................................................ 246 A.11 Submit and Cancel ................................................................................................................... 248
Appendix B: SNMP facility ...................................................................................................... 249 B.1 SNMP alarms ............................................................................................................................. 249 B.2 SNMP Requests ........................................................................................................................ 252
Appendix C: TDM signalling firmware configuration options .............................................. 254 C.1 -cA/B and X/Y bits configuration (DPNSS only) ..................................................................... 254 C.2 –cBBY Backbusy control (CAS only) ..................................................................................... 254 C.3 -cCA connect acknowledgement (ETS300 only) .................................................................... 254 C.4 –cCICnnnn circuit identification codes (ISUP only) ............................................................ 254
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C.5 -cCn number of CLI digits (CAS only) .................................................................................... 255 C.6 –cDn number of DDI digits (CAS only) ................................................................................... 255 C.7 -cDPCnnnnn signalling point code (ISUP only) ..................................................................... 255 C.8 -cEn Call Charging Switch (ETS300 only) ............................................................................. 255 C.9 –cEX Primary Rate Call Charging (ETS300 only) .................................................................. 256 C.10 -cFD Diversion (QSIG, ETS300, AT&T T1, and NI-2) .............................................................. 257 C.11 -cFF facility (QSIG, ETS300, and NI2) ..................................................................................... 257 C.12 -cFP MLPP activation (ETS300 only) ...................................................................................... 257 C.13 -cFR Raw Data (ETS300 and QSIG) ........................................................................................ 257 C.14 -cFU user to user (QSIG and ETS300) .................................................................................... 257 C.15 -cIMP75 ...................................................................................................................................... 257 C.16 –cME .......................................................................................................................................... 257 C.17 -cNA1 release link trunk (NI-2 only) ........................................................................................ 257 C.18 -cNCRC disable CRC4 (ISUP only) ......................................................................................... 257 C.19 –cNE network end configuration (DASS, ETS300, AT&T, and NI2)...................................... 258 C.20 -cOPCnnnnn[,i] (ISUP only) ..................................................................................................... 258 C.21 -cQM/S-A/B master/slave priority (QSIG only) ....................................................................... 258 C.22 -cRn Default clearing cause (all versions) ............................................................................. 258 C.23 -cSLCnn signalling link code (ISUP only) .............................................................................. 259 C.24 -cSO disable service message (AT&T and NI2) ..................................................................... 259 C.25 -cSP stop call proceeding (ETS300 only) ............................................................................... 259 C.26 -cSU stop setup acknowledge (ETS300 only) ........................................................................ 259 C.27 –cSW configuration (ETS300 only) ......................................................................................... 259 C.28 -cTS[I]nn (ISUP only) ............................................................................................................. 259 C.29 -cSPEEDnnk (ISUP only) ...................................................................................................... 259 C.30 -cDAUUS (ETS 300 only) ......................................................................................................... 259
Appendix D: Interpreting GIII trace file................................................................................... 260 D.1 Alarms ....................................................................................................................................... 260 D.2 Layer 1 ....................................................................................................................................... 260 D.3 Data Link ................................................................................................................................... 260 D.4 Call Control ............................................................................................................................... 261 D.5 Routing ...................................................................................................................................... 262 D.6 Switching .................................................................................................................................. 262 D.7 RTP/RTT .................................................................................................................................... 262 D.8 Protocol ..................................................................................................................................... 263
Appendix E: SIP custom headers ........................................................................................... 264 E.1 SS7 to SIP mapping headers ................................................................................................... 264 E.2 SIP call recovery headers ........................................................................................................ 268 E.3 Advice of charge headers ........................................................................................................ 268 E.4 Call information headers ......................................................................................................... 269 E.5 Call control and interworking headers ................................................................................... 270
Appendix F: Adding, removing, or replacing Prosody T cards ........................................... 271 F.1 Adding or removing a Prosody T card ................................................................................... 271 F.2 Replacing a Prosody T card .................................................................................................... 271
Appendix G: System Console ................................................................................................. 272 G.1 Setting IP Addresses ............................................................................................................... 272 G.2 System Console errors ............................................................................................................ 273
Appendix H: Website Errors ................................................................................................... 274
Appendix I: An example of a role workflow ........................................................................... 276
Appendix J: Web Services ...................................................................................................... 285
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Appendix K: LDAP Authentication examples ........................................................................ 296 K.1 Active Directory ........................................................................................................................ 296 K.2 Fedora 389DS ............................................................................................................................ 298 K.3 ApacheDS.................................................................................................................................. 300
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TABLE OF FIGURES
Figure 2-1 Software Activation Key ............................................................................................................ 20
Figure 2-2 Trunk Numbering ...................................................................................................................... 22
Figure 2-3 Configure Admin IP Address .................................................................................................... 23
Figure 3-1 GIII System Summary .............................................................................................................. 25
Figure 3-2 TDM Trunks status with error ................................................................................................... 25
Figure 3-3 TDM Trunks status working ...................................................................................................... 25
Figure 4-1 Start a cluster ........................................................................................................................... 27
Figure 4-2 Join a cluster ............................................................................................................................ 27
Figure 4-3 Create a new role ..................................................................................................................... 28
Figure 4-4 Apply Role and create revision ................................................................................................. 29
Figure 4-5 SIP trunk configuration ............................................................................................................. 30
Figure 5-1 Role Configuration Menu .......................................................................................................... 31
Figure 5-2 Role Workflow .......................................................................................................................... 32
Figure 5-3 Creating a new role .................................................................................................................. 33
Figure 5-4 Creating a new role based on an existing role .......................................................................... 34
Figure 5-5 GroomerII - Backup and Restore menu option ......................................................................... 34
Figure 5-6 GroomerII - Backup menu option.............................................................................................. 35
Figure 5-7 GroomerII - Create Backup dialog ............................................................................................ 35
Figure 5-8 GroomerII - Backup creation progress dialog ........................................................................... 35
Figure 5-9 GroomerII - Backup complete dialog ........................................................................................ 35
Figure 5-10 GroomerII - Create zip of backup ........................................................................................... 36
Figure 5-11 Import GroomerII Backup ....................................................................................................... 36
Figure 5-12 Select GroomerII configuration ............................................................................................... 37
Figure 5-13 Role created from GroomerII configuration ............................................................................ 37
Figure 5-14 Apply role (not currently assigned to a role) ........................................................................... 38
Figure 5-15 Apply role (currently assigned to a node) ............................................................................... 38
Figure 5-16 Apply role (currently assigned to a role, remove selected) ..................................................... 39
Figure 5-17 Remove Role confirmation dialog ........................................................................................... 39
Figure 5-18 Apply Role warnings ............................................................................................................... 40
Figure 5-19 Apply Role confirmation dialog ............................................................................................... 40
Figure 5-20 Immediately applied role configuration icon ............................................................................ 40
Figure 5-21 Deleting a role ........................................................................................................................ 41
Figure 5-22 Deleting a role revision ........................................................................................................... 42
Figure 5-23 Delete Role confirmation dialog .............................................................................................. 42
Figure 5-24 Revert role configuration to a particular revision .................................................................... 42
Figure 5-25 Displaying a role revision ........................................................................................................ 43
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Figure 5-26 Adding comments to a role ..................................................................................................... 43
Figure 5-27 General - Clocks tab ............................................................................................................... 44
Figure 5-28 General – Call setup timers .................................................................................................... 45
Figure 5-29 General - Call setup timers ..................................................................................................... 45
Figure 5-30 System configuration alarm mapping tab option ..................................................................... 47
Figure 5-31 Alarm map problem ................................................................................................................ 47
Figure 5-32 Alarm map configuration page with SIP .................................................................................. 47
Figure 5-33 Alarm map configuration with TDM trunks .............................................................................. 48
Figure 5-34 Advanced options ................................................................................................................... 49
Figure 5-35 Trunk configuration setup tab option (SIP example) ............................................................... 50
Figure 5-36 Trunk configuration setup tab option (TDM example) ............................................................. 51
Figure 5-37 SIP Trunk Setup tab ............................................................................................................... 51
Figure 5-38 SIP trunk with static routes ..................................................................................................... 52
Figure 5-39 SIP Call Handling .................................................................................................................... 53
Figure 5-40 SIP Default Codecs Tab ......................................................................................................... 54
Figure 5-41 Codec VAD settings ................................................................................................................ 54
Figure 5-42 SIP Authentication .................................................................................................................. 55
Figure 5-43 A SIP Authentication entry ...................................................................................................... 56
Figure 5-44 SIP Gateways ......................................................................................................................... 57
Figure 5-45 SIP Gateways Routes ............................................................................................................. 59
Figure 5-46 Gateway Route view and configuration .................................................................................. 59
Figure 5-47 viewing SIP gateway routes .................................................................................................... 60
Figure 5-48 Gateway Group - Gateways available .................................................................................... 60
Figure 5-49 Gateway Group - drag gateway into group ............................................................................. 60
Figure 5-50 Gateway Group - Enable gateway .......................................................................................... 60
Figure 5-51 Gateway Retries ..................................................................................................................... 61
Figure 5-52 SIP Protocol Settings .............................................................................................................. 62
Figure 5-53 SIP Protocol Settings - Miscellaneous Parameters ................................................................. 64
Figure 5-54 SIP Proxy ................................................................................................................................ 66
Figure 5-55 SIP Proxy Registration ............................................................................................................ 67
Figure 5-56 Trunk Setup - No protocol assigned ....................................................................................... 68
Figure 5-57 TDM protocol selection ........................................................................................................... 68
Figure 5-58 Available protocol options ....................................................................................................... 69
Figure 5-59 Adding protocol option ............................................................................................................ 69
Figure 5-60 Editing protocol option ............................................................................................................ 70
Figure 5-61 Additional switches string ........................................................................................................ 70
Figure 5-62 Protocol option tool tip ............................................................................................................ 71
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Figure 5-63 Trunk configuration ................................................................................................................. 71
Figure 5-64 Initial page for SS7 T1 protocol trunk ..................................................................................... 72
Figure 5-65 SS7 Signalling option ............................................................................................................. 72
Figure 5-66 Configuring SS7 Signalling Link ............................................................................................. 73
Figure 5-67 Configured SS7 Signalling Link .............................................................................................. 74
Figure 5-68 SS7 ISUP Bearer Group option .............................................................................................. 74
Figure 5-69 SS7 Bearer Group Configuration ............................................................................................ 75
Figure 5-70 Additional switches string ....................................................................................................... 76
Figure 5-71 SS7 Call Handling Options ..................................................................................................... 76
Figure 5-72 SS7 page with existing pointcodes ......................................................................................... 79
Figure 5-73 Add pointcode tab ................................................................................................................... 80
Figure 5-74 M3UA Routing Keys ................................................................................................................ 81
Figure 5-75 M3UA Connections .................................................................................................................. 81
Figure 5-76 Invalid point code for variant ................................................................................................... 83
Figure 5-77 System message showing invalid point code error ................................................................. 83
Figure 5-78 Indication of an invalid point code........................................................................................... 83
Figure 5-79 System message showing duplicate point code error ............................................................ 83
Figure 5-80 Initial empty SS7 configuration ............................................................................................... 83
Figure 5-81 viewing and editing the SS7 Advanced configuration ............................................................. 84
Figure 5-82 System configuration Protocol Settings - SS7, International tab option .................................. 85
Figure 5-83 SS7 ANSI Interworking ........................................................................................................... 86
Figure 5-84 SS7 UK ISUP ......................................................................................................................... 87
Figure 5-85 Point code deletion confirmation dialog .................................................................................. 88
Figure 5-86 Routing Configuration - Groups .............................................................................................. 89
Figure 5-87 Groups - create a new group .................................................................................................. 90
Figure 5-88 Groups configuration .............................................................................................................. 90
Figure 5-89 SIP Routing controls added when SIP Trunk added............................................................... 91
Figure 5-90 TDM trunk added to Trunks List ............................................................................................. 92
Figure 5-91 TDM trunk selection and timeslot configuration ...................................................................... 92
Figure 5-92 Routes Configuration .............................................................................................................. 93
Figure 5-93 Routing configuration setup tab option ................................................................................... 94
Figure 5-94 Routing - Common Q.931/ISUP tab ..................................................................................... 101
Figure 5-95 Routing configuration Q.931/ISUP – Q.931 tab option ......................................................... 103
Figure 5-96 Routing configuration Q.931/ISUP – ISUP tab option........................................................... 104
Figure 5-97 Routing configuration CAS tab option ................................................................................... 106
Figure 5-98 Routing configuration DPNSS tab option .............................................................................. 108
Figure 5-99 Routing - Incoming SIP calls tab........................................................................................... 109
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Figure 5-100 Routing - Outgoing SIP calls tab ......................................................................................... 109
Figure 5-101 Incoming SIP calls configuration for a route........................................................................ 110
Figure 5-102 Routing Incoming SIP calls tab with codecs ....................................................................... 113
Figure 5-103 Routing - Outgoing SIP calls tab with codecs ..................................................................... 113
Figure 5-104 Routing - SIP Codec configuration ...................................................................................... 114
Figure 5-105 Codec VAD settings ............................................................................................................ 114
Figure 5-106 Routing configuration general tab option ............................................................................ 115
Figure 5-107 PCM conversion selection .................................................................................................. 116
Figure 5-108 Routes - re-route causes .................................................................................................... 116
Figure 5-109 Cause Map list .................................................................................................................... 118
Figure 5-110 Cause mapping configuration ............................................................................................. 119
Figure 5-111 SIP TLS Certificates page ................................................................................................... 120
Figure 5-112 Default Call Trace settings .................................................................................................. 121
Figure 5-111 Trunk Blocking .................................................................................................................... 122
Figure 5-114 Trunk blocking for ITU-T, China, and UK ISUP................................................................... 123
Figure 5-115 Trunk blocking page............................................................................................................ 124
Figure 5-116 SS7 Signalling Links ........................................................................................................... 125
Figure 5-117 SNMP Reporting ................................................................................................................. 126
Figure 5-118 SNMP Reporting – Ports tab ............................................................................................... 127
Figure 5-119 SNMP reporting – System tab ............................................................................................ 128
Figure 5-118 SNMP reporting – Chassis tab ............................................................................................ 129
Figure 5-121 SNMP Reporting – Call Monitoring ..................................................................................... 131
Figure 6-1 System Status showing node/role maintenance mode ........................................................... 133
Figure 6-2 Node Maintenance Mode page ............................................................................................... 133
Figure 6-3 Node in an active state ........................................................................................................... 133
Figure 6-4 Trunk Download/Reset page .................................................................................................. 134
Figure 6-5 Circuit Reset Page .................................................................................................................. 134
Figure 6-6 SS7 Continuity Checking ........................................................................................................ 134
Figure 6-7 Selecting timeslots to check .................................................................................................... 135
Figure 6-8 Status of continuity checks ..................................................................................................... 135
Figure 6-9 SS7 Trunk ............................................................................................................................... 136
Figure 6-10 Non SS7 Trunk ..................................................................................................................... 136
Figure 6-11 IP diagnostics page .............................................................................................................. 136
Figure 6-12 Ping diagnostic results .......................................................................................................... 137
Figure 6-13 Traceroute diagnostic results ................................................................................................ 137
Figure 6-14 Trace Files page ................................................................................................................... 138
Figure 6-15 Trace file view ....................................................................................................................... 138
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Figure 6-16 Paged contents selection ..................................................................................................... 138
Figure 6-17 Search text tool ..................................................................................................................... 139
Figure 6-18 Search highlighting ............................................................................................................... 139
Figure 6-19 Download files ...................................................................................................................... 139
Figure 6-20 Constructing file progress ..................................................................................................... 140
Figure 6-21 Call trace files ....................................................................................................................... 140
Figure 6-22 Node selection ...................................................................................................................... 140
Figure 6-23 SS7 trace files ...................................................................................................................... 141
Figure 6-24 SIP Trace page ..................................................................................................................... 141
Figure 6-25 SIP Trace page when SIP trace is running ........................................................................... 142
Figure 6-26 Packet capture page ............................................................................................................. 143
Figure 6-27 Packet capture host address tool tip ..................................................................................... 143
Figure 6-28 Packet capture port tool tip ................................................................................................... 143
Figure 6-29 Web System log file page ..................................................................................................... 144
Figure 6-30 Technical Support page ........................................................................................................ 145
Figure 6-31 Snapshot in progress ............................................................................................................ 145
Figure 6-32 Snapshot created ................................................................................................................. 146
Figure 7-1 System Summary ................................................................................................................... 147
Figure 7-2 Node with no active role ......................................................................................................... 147
Figure 7-3 Node with a role starting ......................................................................................................... 148
Figure 7-4 Node with an active role ......................................................................................................... 148
Figure 7-5 Node with a failed role, or is unavailable ................................................................................ 148
Figure 7-6 GIII back panel image ............................................................................................................. 149
Figure 7-7 Trunk status ............................................................................................................................ 149
Figure 7-8 The IP interfaces graphic showing status ............................................................................... 149
Figure 7-9 The fan statuses graphic ........................................................................................................ 149
Figure 7-10 System LED status graphic .................................................................................................. 149
Figure 7-11 Card alarm ............................................................................................................................ 150
Figure 7-12 Cores alarm .......................................................................................................................... 150
Figure 7-13 Fans alarm ............................................................................................................................ 151
Figure 7-14 Activation Key alarm ............................................................................................................. 151
Figure 7-15 NTP alarm ............................................................................................................................ 151
Figure 7-16 Essential Processes alarm ................................................................................................... 151
Figure 7-17 RAID alarm ........................................................................................................................... 151
Figure 7-18 SMART alarm ....................................................................................................................... 152
Figure 7-19 Temperature alarm ............................................................................................................... 152
Figure 7-20 Upgrade Cluster version alarm .............................................................................................. 152
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Figure 7-21 Cluster problem alarm ............................................................................................................ 152
7-22 Call Traffic ........................................................................................................................................ 153
7-23 Call Monitor Alert thresholds ............................................................................................................ 154
Figure 7-24 TDM Trunks page ................................................................................................................. 155
Figure 7-25 GIII status monitor CAS tab option ....................................................................................... 156
Figure 7-26 Ethernet Trunks page ........................................................................................................... 157
Figure 7-27 Call Trace page .................................................................................................................... 157
Figure 7-28 SS7 Signalling tab ................................................................................................................ 158
Figure 7-29 GIII status monitor SS7 ISUP Bearers .................................................................................. 158
Figure 7-30 GIII status monitor SS7 MTP3 Routes .................................................................................. 159
Figure 7-31 GIII status monitor SS7 MTP3 Destinations ......................................................................... 160
7-32 GIII status monitor SS7 M3UA Connections ..................................................................................... 161
7-33 GIII status monitor SS7 M3UA Connection Stats .............................................................................. 161
7-34 GIII status monitor SS7 M3UA Connections ..................................................................................... 162
7-35 GIII status monitor SS7 M3UA Routing Key Stats ............................................................................ 162
Figure 7-32 GIII Status Monitor SIP Gateways page ............................................................................... 163
Figure 7-33 NTP status ............................................................................................................................ 164
Figure 7-34 Temperatures and Fans........................................................................................................ 164
Figure 7-35 Fans status ........................................................................................................................... 164
Figure 7-36 Essential processes page ..................................................................................................... 165
Figure 7-37 System card status page ...................................................................................................... 165
Figure 7-38 – Card status showing firmware update required................................................................... 165
Figure 7-39 Cluster Status page ............................................................................................................... 166
Figure 8-1 Admin IP configuration page ................................................................................................... 167
Figure 8-2 Chrome browser warning ........................................................................................................ 167
Figure 8-3 Web site TLS certificate page ................................................................................................. 167
Figure 8-4 Certificate error ....................................................................................................................... 168
Figure 8-5 Disk Management ................................................................................................................... 168
Figure 8-6 The RAID alarm on the status page ........................................................................................ 169
Figure 8-7 Drive showing faulty status ..................................................................................................... 169
Figure 8-8 Enable replacement drive ....................................................................................................... 169
Figure 8-9 RAID rebuilding ....................................................................................................................... 170
Figure 8-10 System Summary showing SMART failures alarms .............................................................. 170
Figure 8-11 Node with disks exhibiting SMART failures........................................................................... 170
Figure 8-12 SMART failure log ................................................................................................................. 171
Figure 8-13 Deleting SMART failure log ................................................................................................... 171
Figure 8-14 Card Setup page ................................................................................................................... 171
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Figure 8-15 Node LED control page ........................................................................................................ 172
Figure 8-16 Changing the Node LED status for a node ........................................................................... 172
Figure 8-17 Rename Node ...................................................................................................................... 172
Figure 8-18 Example of node name with serial number ........................................................................... 173
Figure 8-19 Software Upgrades ............................................................................................................... 173
Figure 8-20 – System card requires an update ......................................................................................... 173
Figure 8-21 – System card update dialog ................................................................................................. 174
Figure 8-22 Choose alternative software version ..................................................................................... 174
Figure 8-23 Choose software version dialog............................................................................................ 174
Figure 8-24 Software upgrade upload ..................................................................................................... 175
Figure 8-25 Activation key page .............................................................................................................. 176
Figure 8-26 Software licenses page ........................................................................................................ 176
Figure 8-27 GIII Documentation page ...................................................................................................... 177
Figure 9-1 Cluster administration ............................................................................................................. 178
Figure 9-2 Start a cluster ......................................................................................................................... 178
Figure 9-3 Node already clustered ........................................................................................................... 179
Figure 9-4 Join a Cluster .......................................................................................................................... 179
Figure 9-5 Decluster a Node .................................................................................................................... 180
Figure 9-6 Change cluster version page .................................................................................................. 181
Figure 9-7 Change password button ........................................................................................................ 181
Figure 9-8 Change user password ........................................................................................................... 182
Figure 9-9 New user account setup ......................................................................................................... 183
Figure 9-10 Viewing current user accounts.............................................................................................. 183
Figure 9-11 Modifying a user account ...................................................................................................... 183
Figure 9-12 Deleting a user account ........................................................................................................ 183
Figure 9-13 CDR Query settings .............................................................................................................. 184
Figure 9-14 CDR Query Files .................................................................................................................. 190
Figure 9-15 Configure NTP ...................................................................................................................... 190
Figure 9-16 NTP status page ................................................................................................................... 191
Figure 9-17 NTP not synchronized alarm ................................................................................................ 191
Figure 9-18 Delete NTP server ................................................................................................................ 191
Figure 9-19 SNMP agent configuration page ........................................................................................... 192
Figure 9-20 viewing SNMP agent users................................................................................................... 193
Figure 9-21 SNMP agent user configuration page ................................................................................... 193
Figure 9-22 viewing SNMP agent communities ....................................................................................... 193
Figure 9-23 SNMP agent community configuration page ........................................................................ 194
Figure 9-24 Deleting a community by dragging to trash .......................................................................... 194
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Figure 9-25 viewing SNMP agent hosts ................................................................................................... 194
Figure 9-26 SNMP agent host configuration page ................................................................................... 195
Figure 9-27 configuring a SNMP agent host for SNMPv3 ........................................................................ 195
Figure 9-28 viewing SNMP agent SNMPv3 access users........................................................................ 196
Figure 9-29 Configuring SNMP agent SNMPv3 user access ................................................................... 196
Figure 9-30 Configure syslog ................................................................................................................... 197
Figure 9-31 Add Syslog server ................................................................................................................. 197
Figure 9-32 Delete Syslog Server ............................................................................................................ 198
Figure 9-33 Server Configuration ............................................................................................................. 199
Figure 9-34 User and group search configuration .................................................................................... 200
Figure 9-35 Admin and monitor group mapping ....................................................................................... 201
Figure 9-36 Login page configuration....................................................................................................... 201
Figure 9-37 LDAP Testing ........................................................................................................................ 202
Figure 9-38 Example of testing LDAP login ............................................................................................. 202
Figure 9-39 Role backup .......................................................................................................................... 203
Figure 9-35 System backup ..................................................................................................................... 203
Figure 9-36 Role restore .......................................................................................................................... 204
Figure 9-37 System restore ...................................................................................................................... 204
Figure 9-38 Restart & Shutdown page ..................................................................................................... 205
Figure 9-39 Selecting a node to be restarted ........................................................................................... 205
Figure 9-40 Selecting a node to be shutdown .......................................................................................... 205
Figure 10-1 Wait for Transfer Routing Configuration ............................................................................... 206
Figure 11-1 User defined ISDN subaddress extension ............................................................................ 213
Figure 12-1 Single SS7 link option ........................................................................................................... 219
Figure 12-2 Two SS7 links option ............................................................................................................ 219
Figure 12-3 Single SS7 link looped back option ....................................................................................... 220
Figure 13-1 CAMA trunk configuration settings ........................................................................................ 224
Figure 13-2 CAMA trunk configured for called party only call clearing ..................................................... 225
Figure 15-1 Selecting TTY/RTT operation ............................................................................................... 237
Figure 15-2 GIII engine log showing TTY/RTT mapping active................................................................ 237
Figure 15-3 Warning about secure SIP and RTT ..................................................................................... 237
Figure 15-4 Warning about secure SIP and RTT in Route ....................................................................... 237
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1 Introduction The sections System overview, System boot-up, and Initial Configuration introduce the basic steps required to setup and configure a GIII from initial deployment, or after de-clustering a node. The sections GIII Role Configuration, GIII Maintenance, GIII Status monitors, Node Administration, and Cluster Administration cover each menu available on the GIII web interface, and each sub section is in the order of the menu items available. The sections Call Transfer, SIP telephony, SS7 telephony, CAMA telephony, Interworking supplementary services, and TTY/RTT mapping provide detailed information regarding various telephony operations of the GIII. The Web Services section covers the web services provided by the GIII. The Advanced fault finding and maintenance section covers fault finding and maintenance of the GIII. The Appendices contain a number of informational items which are mentioned in the document.
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2 System overview
2.1 Chassis installation guides This user guide describes the GIII application software packages, and all aspects of system configuration. For a description of the hardware please refer to the appropriate installation guide for your chassis. The installation guides available are:
Aculab GIII gateway, 2U carrier grade chassis (R730) Installation guide. The installation guides can be downloaded from the Aculab website at http://www.aculab.com, or may be downloaded from your GIII’s website via the Node Administration->Documentation menu.
2.2 Signalling protocols supported GIII supports an extensive list of CCS, CAS, and IP Telephony protocols. In some instances, Aculab have developed country or user specific variations of standard protocols. Visit the Aculab web site at http://www.aculab.com, or contact Aculab support for further details. A full set of protocol firmware is shipped with GIII.
2.3 GIII web interface The GIII is configured using a web interface. The Appendix A: Web Site Concepts covers the different mechanisms used.
2.4 Activation key mechanism An activation key is required to activate a role on a GIII node, and enable it to process calls. To update the software activation key, select Software Activation Key, under the Node Administration menu. The activation key is a 12-character string in the form 58C-3F8-YKQ-G9R, and will be pre-installed when your GIII is delivered. Its purpose is to allow a particular revision of software to run on a specific GIII chassis. The activation key required will change whenever the major or minor software version number changes. For example, versions 12.1.0 and 12.1.2 will use the same activation key, whilst versions 12.1.0 and 12.2.0 will require different activation keys, as will versions 12.0.0 and 13.0.0.
Figure 2-1 Software Activation Key
The GIII will not allow a role to be active if no valid activation key is found for the current software version. The GIII will not allow an alternate software version to be used until a valid activation key is found for that software version. Once entered, the system will remember the key.
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2.5 Principles of operation The primary function of GIII is to receive an incoming call, make an outgoing call in response and then connect the voice paths between the two call legs. Each TDM trunk in the system can be loaded with any TDM protocol from the Aculab portfolio, with the trunk being automatically configured for E1 or T1 operation depending upon the protocol in use. IP telephony ports are automatically enabled for SIP telephony. When configured and active, the GIII will automatically listen for incoming calls on every available channel in the system. When an incoming call is detected the Aculab call control software passes the call information up to the GIII application software. This call information is used to search the routing table and select a trunk (timeslot) on which to place the outgoing call. Call control events such as alerting, connect and disconnect are mapped between the calls by the GIII application software, and when the far end answers the call the voice paths are connected, using the H.100 bus for cross-card routes. GIII supports any-to-any protocol conversions. Calls can be routed in either direction between any two trunks, with no restrictions on the type of protocol conversions that can be achieved.
2.6 GIII IP telephony principles The principle of operation of GIII for routing of traffic from one protocol to another is the same for IP Telephony as it is for all other protocols on GIII, the only difference for IP Telephony is that the transmission medium for IP Telephony calls into and out of GIII is via an Ethernet digital connection rather than the usual E1/T1 digital trunk. An E1/T1 trunk is a point to point connection with known end addresses. As the end points are known, data sent out on an E1/T1 trunk (timeslot) does not need to contain originating and destination address information. An ethernet digital connection however does not use a specific timeslot for a call. Call data may take many different routes to reach the destination address. Call data is transmitted over an ethernet network in packets using the appropriate IP protocol. Each IP packet contains details of the originating and destination IP addresses, allowing the packet to be routed through the network to the correct destination address. To achieve this, GIII has to translate standard PSTN call address details into IP address formats that are recognised by the IP protocols, this can be achieved in a number of ways: Route all calls to a specific IP address (gateway). Route calls to specific IP addresses via a SIP proxy server.
NOTE GIII currently supports IP call addressing in the format DDI@IP address, for example, [email protected] or 123456@fdac::2998:5000.
Encoding When interworking between IP and TDM protocols, by default GIII will detect the type of connection that is being switched and will use the appropriate encoding.
Pseudo timeslots IP telephony has no concept of timeslots; however GIII assigns a pseudo timeslot to each call for consistency, and to assist with call traffic management.
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Codec negotiation The system can be configured with a default list of available codecs, listed in preference order. Each route can be set to use the default list, or can have its own list defined with its own preference order. GIII uses these lists when negotiating call setup with the remote endpoint.
SIP transport types SIP is able to use both UDP and TCP for call control signalling. This is implemented in the following way: • All messages originating from the far end (either an incoming or outgoing call) will
use the same protocol as the incoming message for the response. • All messages originating with GIII and sent to an incoming call will use the
transport protocol that the far end used to send the incoming INVITE. • All messages originating with GIII and sent to an outgoing call will use the
transport protocol configured for the route used.
2.7 2U carrier grade trunk numbering Each GIII trunk is assigned a unique number. These numbers are assigned sequentially beginning at one, with the trunks on card 1 being assigned their numbers first, followed by the trunks on card 2, and so on.
2.7.1 Dual trunk breakout adapters Each physical trunk on a Prosody T card is cabled in a fashion that allows two trunks to be connected. Your system will be delivered with a set of dual trunk adapters allowing the trunks to be broken out. The chassis installation guide contains a detailed description of the adapters.
2.7.2 Trunk numbering The following image shows the trunk numbering for typical chassis configurations when viewed from the rear; this is the image that is shown on the GIII system summary page.
Figure 2-2 Trunk Numbering
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2.8 System configuration If you did not arrange for your system to be configured by the Aculab Professional Services team prior to delivery, you will need to carry out the following steps to produce a basic configuration. This should be done before attaching any E1/T1 cables to your system. Power on the system. The GIII start-up process will run automatically and the configuration used by Aculab to test the system will be loaded. 1. The GIII will ship with an admin IP address that was used to test the system, which
will be 192.168.1.100/24. The iDRAC interface will be preconfigured with 192.168.1.101/24 as the address. It is possible to change the GIII admin IP address from the console which can be accessed over the iDRAC interface [Appendix H: System Console]. If the GIII admin IP address is changed then the GIII will be rebooted.
The Admin IP addressing is configured from the Configure Admin IP Address page which is located under the Node Administration menu. This page will only allow the IP address information to be changed if there is no active role on the GIII. The GIII is required to reboot when the Admin IP address information is changed.
Figure 2-3 Configure Admin IP Address
2. Login to the web interface using username: admin, and password: password. It is suggested that the admin password is changed from the default.
3. Create or join a cluster
NOTE If the GIII is not going to be used in a cluster, i.e. it will neither join an existing cluster, nor have another GIII join it to form a cluster then Step 5 can be skipped.
NOTE If the GIII is going to join an existing cluster, then it will be configured with the cluster-wide settings, and will have access to the existing roles in a cluster.
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CAUTION If a GIII device which is part of a cluster is going to be switched off for more than a week then it is imperative that the node is actually declustered before the long term power off, and then re-joined to the cluster when required. No configuration will have been lost as the existing cluster will maintain the cluster and role data.
4. Configure NTP
NOTE If the GIII is not going to be used in a cluster, then NTP is not essential, but is recommended. If the GIII is going to be used in a cluster then NTP is essential for the correct operation of the cluster.
5. Create a role. This can be a new role, a new role based on an existing role, a restored role, or a role imported from a GroomerII backup.
6. Configure the SIP trunk. 7. Then:
In the Role Configuration – Trunks page, verify that the TDM trunk settings are appropriate to the protocol in use. In the Role Configuration - Routing page configure the groups and routes that you require to route your calls. In the Role Configuration – General - Clocks page select your system clock sources. If your GIII will be processing IP telephony calls, use the SIP tab items in Role Configuration – Trunks page to:
• Select the default codecs to be offered.
• Configure how SIP destination addresses will be selected. 8. Then:
Go to Role Configuration – Role Management, to apply the role to the node. Select Create revision with current settings and Select created revision. Select the System Summary page from the Status Monitors menu. The node that you have applied the role to will show Node Maintenance mode, Enabled. Click Enabled, uncheck Maintenance mode, and click submit. If you go back to the System Summary page, you will see the node will be starting the role. The section for this node will be yellow whilst it is starting (or is in maintenance mode), and will be green once it is ready to receive calls.
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3 System boot-up
3.1 Overview. The web interface will be accessible using HTTPS. The default username is admin, the default password is password.
3.2 Initial hardware checks. Once a GIII has booted, and started its applications, then log in to the web interface. Select the Status Monitors menu item System Summary. There is an Alarms column which will display any alarms the system has.
Figure 3-1 GIII System Summary
The Traffic interface for the GIII should show at least one of I/F 1 or I/F 2 in a working state. If the GIII has an active role then the TDM Trunks page under the Status Monitors menu will display the status for the trunks. A trunk should show No Alarms Detected. If any other status is present then either there is no circuit attached (No Signal Detected), or a fault is present on the network. Faults should be resolved before call passing is attempted.
Figure 3-2 TDM Trunks status with error
Figure 3-3 TDM Trunks status working
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3.3 Basic fault finding The following list provides examples of possible reasons for not receiving expected event messages.
Expected message Possible cause of failure Layer 1 ready (alarms page)
Check cabling / restart GIII or switch.
Layer 2 established Check if you are using the Net or User ends of the protocol.
Incoming call Log protocol. If you do not see a large RX string, the switch is not sending the call.
Called number If there is no number shown you may not be able to route. If you are using CAS, check the digits are in the voice path.
Routing route 0-1 If no routing statement appears you may have a bad Layer 1 state on your outgoing port, or a suitable route could not be found. Check that the incoming group is included in the expected routing table entry.
Outgoing ringing If you receive a Remote disconnect or call failed message at this point, check that the remote switch is set for incoming calls, or log protocol trace for details. If no message is seen then the call has passed through GIII but is waiting for an acknowledgement that a connection is being attempted.
Outgoing connected Incoming connected
The call has not yet switched through and is therefore not yet successful.
Remote disconnect If you do not receive this message when attempting to clear down the call, use protocol trace to check that the remote switch is sending the disconnect/release.
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4 Initial Configuration The GIII will require some of the following for initial configuration: 1. Cluster the device 2. Create a role 3. Activate a role
4.1 Cluster the device The GIII may be either a stand-alone device, a cluster of one device, or may be joined to an existing cluster. If GIII is used in stand-alone mode, i.e. no other GIII will be joining it to form a cluster, you will not be required to cluster the device. Cluster administration is performed via the Administer Cluster menu item, located in the Cluster Administration menu.
4.1.1 Configure Cluster The Cluster Name tab on the Administer Cluster page is used to configure the cluster name and password. This is only required if the GIII will have another GIII joining it to form a cluster.
Figure 4-1 Start a cluster
4.1.2 Joining a Cluster The Join a Cluster radio option on the Cluster Name page is used to join a cluster. The page below will be presented.
Figure 4-2 Join a cluster
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To join a cluster you will need to know: 1. The cluster name. 2. The cluster password. 3. The IP address of any node in that cluster. Enter this information and press the submit button. If the fields entered are accepted then the device will restart and on startup will join the configured cluster. This may take some time.
CAUTION If a GIII device which is part of a cluster is going to be switched off for more than a week then it is imperative that the node is actually declustered before the long term power off, and then re-joined to the cluster when required. No configuration will have been lost as the existing cluster will maintain the cluster and role data.
4.2 Create a new role A role contains a collection of configuration information used for a node. An individual role can only be assigned to a single GIII device. To create a role, select the Role Management menu item in the Role Configuration menu. The page below will be presented.
Figure 4-3 Create a new role
There are a number of ways to create a role:
• restore a role (backup and restore)
• create a brand new role
• create a role based upon an existing role
• create a role from a GroomerII backup You are required to give a role a name, and select the number of trunks for this role.
NOTE You may not change the number of trunks in a role once it is configured. You may copy a role though, and change the number of trunks at that point. To copy a role you are required to create a new role based upon an existing role.
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Once you have created a role, the following page will be displayed.
Figure 4-4 Apply Role and create revision
A role must be applied to a node in the cluster for its configuration to be used, but before doing this, configuration of the role settings should be done.
4.2.1 Role configuration order Some configuration parameters are dependent on others, for example, Routes depends on Groups which in turn depend on Trunks. To avoid problems, a configuration should be built in the following order: 1. Trunks. 2. Routing - Groups. 3. Routing – Routes. Followed by the remaining configuration options, as is required.
CAUTION Changes to Trunks may change previous settings in Groups, Routes and Tone Generation.
4.3 IP telephony network configuration Before GIII can be used to pass traffic, the IP telephony resources must be configured with settings from your own network. This step must be carried out even if your system will not be making IP telephony calls. GIII supports IP telephony using both IPv4 and/or IPv6: • If IPv4 telephony is to be used the SIP trunk of the role must be configured with
IPv4 addresses. • If IPv6 telephony is to be used the SIP trunk of the role must be configured with
IPv6 addresses.
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NOTE GIII must be configured with static IP addresses. The use of DHCP to obtain dynamic IP addresses is not supported. The use of automatically generated IPv6 addresses is not supported. Contact your network administrator to obtain the correct addresses and settings to be used.
4.4 Media IP Configuration The IP addressing for SIP traffic is configured for a particular role. This is accessed on the Trunks menu item from the Role Configuration menu. The page will display a number of tabs, the first is always SIP, and the remainder are for each trunk present in the role. The SIP tab will present the following page.
Figure 4-5 SIP trunk configuration
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the running configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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5 GIII Role Configuration Configuration of GIII is performed through the web interface. The main configuration settings are stored in a role, which is applied to the node. The available roles can be accessed and modified through the Role Configuration menu.
Figure 5-1 Role Configuration Menu
5.1 Role Configuration options The Role Configuration menu options are: Role Management – Create, apply, remove or delete roles from a node. General – Parameters for Clock Sources, Call setup timers, Alarm Mapping, and various advanced configuration options. Trunks – Configure trunk firmware settings. SS7 – SS7 specific configuration settings. Routing – Add, Update or Delete Groups, Routes and Cause Maps. SIP TLS Certificates – Specify certificates to use when making secure SIP calls. Call Trace – Enable and disable tracing options on a per trunk basis. Trunk Blocking – Block traffic from individual timeslots or whole trunks. SS7 Signalling Links – Activate and Deactivate SS7 signalling links. SNMP – Enable SNMP and configure thresholds at which traps are triggered.
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5.2 Role Management A role contains a collection of configuration information used for a node. An individual role can only be assigned to a single node in a cluster. A role will have a revision, which is a saved version of a role. A new role cannot be assigned to a cluster node until a revision has been created. A role revision will not be used by a node in a cluster until it is actually applied to a node.
Figure 5-2 Role Workflow
Once a role has been applied to a node, any further edits to the role configuration will not be applied to the node (unless they are immediate) until a new revision is created, and this revision is selected. The role workflow in Figure 5-2 shows the edit role, create revision, and select revision steps that would be used to put new changes for a role onto the running configuration. These three steps can be repeated each time changes are required.
NOTE The Appendices contain an example of a typical role workflow.
The configuration pages that are role based will have accordion segments for each role in the Cluster, and the accordion banner will show the Current state, and the Selected revision. A new role would show:
If a role has not been changed since its revision was created then the accordion would show something similar to below:
If a role has been changed since a revision was created then you would see:
If you have reverted to an earlier revision of a role and made changes then you would see:
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5.2.1 Create a new Role To create a new role, select Role Management in the Role Configuration menu, and select the Create a new Role accordion.
Figure 5-3 Creating a new role
New Role name – The name of the role to create
Number of TDM trunks – Select 8, 16, 24, or 32
Default TDM trunk type – Select T1 or E1
Based on – Select Brand new, Existing Role, or GroomerII Backup.
Depending on what is selected from Based on, additional items will be displayed. For Based on and Existing Role:
For Based on and GroomerII Backup:
You are required to give a role a name, and select the number of trunks for this role.
NOTE You may not change the number of trunks in a role once it is configured. You may copy a role though by creating a new role based on an existing one, and change the number of trunks at that point.
Once you have created a role, the manage role accordion for this role will be displayed. A new role cannot be assigned to a node in a cluster until a revision has been created; this is done using the Apply Role tab. A role must be applied to a node in the cluster for its configuration to be used.
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5.2.1.1 Create a new role from an existing role
Figure 5-4 Creating a new role based on an existing role
A new role can be created based upon an existing role. To do this, select Existing Role from the Based on selection, and select the particular version and role required from the Role to copy selection. The number of trunks can be changed (this is how you change the number of trunks in a role – create a new role based on a role and change the number of trunks).
5.2.1.2 Create a new role for a GroomerII backup When migrating from an existing GroomerII system to a GIII platform, it is possible to automatically import a GroomerII configuration file (*.cfg) contained in a compressed GroomerII backup, to a GIII role. The steps to do this are: 1. Create a GroomerII backup from your current system, by going to Start – All
Programs – Aculab GroomerII – Utilities – Backup and Restore.
Figure 5-5 GroomerII - Backup and Restore menu option
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2. Go to Operations - Backup…
Figure 5-6 GroomerII - Backup menu option
3. Enter a file path for where the backup should be stored, make sure Backup Log Files is unticked, and click OK.
Figure 5-7 GroomerII - Create Backup dialog
4. A progress dialog will be displayed while the backup is created.
Figure 5-8 GroomerII - Backup creation progress dialog
5. Once the backup is complete, you will be notified, and should click OK.
Figure 5-9 GroomerII - Backup complete dialog
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6. Go to the directory where the backup has been created, right-click on the backup folder, and select Send to – Compressed (zipped) folder. A zip will be created, which will be called <backup folder name>.zip. In this example, G5979.zip.
Figure 5-10 GroomerII - Create zip of backup
7. Log in to your GIII, and go to Role Configuration – Role Management, and select the Create a new Role accordion.
8. Enter a New Role Name, select the Number of TDM trunk on the node, and the Default TDM trunk type. Select GroomerII Backup, from the Based on dropdown, click Choose File, browse to the location of the GroomerII backup, click Open, and click the Submit button.
Figure 5-11 Import GroomerII Backup
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9. Select the GroomerII configuration on which the new role should be based, and click Continue.
Figure 5-12 Select GroomerII configuration
Figure 5-13 Role created from GroomerII configuration
Any issues will be reported in the text box. Pressing Continue will present the Manage Role accordion.
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5.2.2 Apply Role From the Manage Role accordion, the Apply Role tab can be selected. This allows a role to be applied to a Node in the Cluster, and/or a revision to be created from current settings.
Figure 5-14 Apply role (not currently assigned to a role)
Node to apply Role to – This control is only displayed if the role is not assigned to a node. This will allow a node to be select if it is not already assigned to a role. If a node is already assigned to a role then the node will be listed in the drop-down, but will not be selectable. Create revision with current settings – If ticked then a revision will be created from the current configuration Select created revision – This is the revision that is used by a node when the role is assigned to it. You can change the selected revision to any existing revision, or create one from the current settings, and use that. Before a new role can be used a revision needs to be created.
Figure 5-15 Apply role (currently assigned to a node)
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Figure 5-16 Apply role (currently assigned to a role, remove selected)
Remove Role from Node – If ticked then all other controls are no longer displayed, and a warning message is displayed as the change could cause active calls to be lost. Pressing the Submit button will present a further confirmation dialog with the warning repeated.
Figure 5-17 Remove Role confirmation dialog
When applying a new revision of a role, for a role that is already active, warning messages may be displayed informing whether the role will be restarted, and trunks/timeslots blocked, all depending upon what particular changes have been made.
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Figure 5-18 Apply Role warnings
If the Submit button is pressed, a further confirmation dialog will be shown, with the warning repeated.
Figure 5-19 Apply Role confirmation dialog
5.2.2.1 Immediately applied Role changes
Figure 5-20 Immediately applied role configuration icon
There are a number of role settings which do not require creating a revision for, as their settings are applied immediately. The pages which have this behavior will have the icon shown in Figure above if the role is active on a node.
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The pages which have immediately applied changes are:
• Call Trace
• Trunk Blocking
• SS7 Signalling Links
• SNMP
Call Trace Changes made to the Call Trace page are immediately applied to the running configuration. This allows you to alter tracing settings on a node, without having to re-apply the role.
Trunk Blocking The trunk blocking page, under Role Configuration, shows the trunk types, and timeslots of the 'selected' revision. Any changes to the blocking will be immediately applied the node using the role (if any). When changing the selected revision, the current blocking is applied to the new revision where possible. Certain changes will cause the blocking to be modified, e.g. changing the firmware type, such as ETS to DASS, will result in all timeslots being blocked.
SS7 Signalling Links The SS7 Signalling Links page, under Role Configuration, shows the current state of each signalling link for a role. Any changes to the signalling links will be immediately applied to the node using the role (if any) when the Submit button is pressed.
SNMP The SNMP page, under Role Configuration, shows the current settings for SNMP reporting for a role. Any changes to the settings on a tab are immediately applied to the node using the role (if any) when the Submit button is pressed.
NOTE If another tab in SNMP is selected, any changes made will be discarded unless the Submit button has been pressed.
5.2.3 Delete Role
Figure 5-21 Deleting a role
Delete entire role – If ticked then all other controls are no longer displayed. Selecting the Submit button will completely delete the role. Revision to delete – You can select a particular revision to delete by selecting it and selecting Submit. You cannot delete the currently selected revision, it will be shown in list, but not selectable.
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The chosen revision will be deleted, and will not affect any existing revisions. This may result in a non contiguous list of revisions.
Figure 5-22 Deleting a role revision
If the Submit button is pressed then there will be a confirmation dialog shown.
Figure 5-23 Delete Role confirmation dialog
5.2.4 Revert Role
Figure 5-24 Revert role configuration to a particular revision
A role revision is a saved snapshot of the role’s configuration at a particular point in time. You may revert the current state of a role to a previous revision, to allow you to edit a previous configuration. To apply this edited revision to a node, you must go to the Apply Role tab, select Create revision with current settings and click submit. This will create a revision using the next available revision number, and apply this role revision to the node.
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5.2.5 Display Revision
Figure 5-25 Displaying a role revision
You can make the role configuration pages display saved revisions by selecting a revision on the Display revision tab of the role management page. This only applies for the current session. If you log out, and then log in, you will revert to the selected role.
NOTE When a particular revision is displayed, changes can't be saved, unless you choose to display Current settings.
5.2.6 Comments
Figure 5-26 Adding comments to a role
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5.3 General configuration The GIII Role Configuration - General page is used to configure system wide resources, and settings that will be applied to all calls passing through the system. This page will show tabs for:
• Clock sources
• Call setup timers
• Alarm mapping
• Advanced Options
5.3.1 Clock Sources
Figure 5-27 General - Clocks tab
The Clock Sources page is used to specify which trunks GIII will use as the system clock source. The system clock source is used to drive the H.100 bus, which in turn provides clocking to all other ports in the system. Preferred Clock Source List – This specifies which trunks may drive the system clock, and the order in which they should be used. GIII will use the first available trunk in the list to drive the clock, and should this trunk become unavailable the next available trunk will be selected. Should a previously failed trunk become available GIII will restore this trunk as the clock source. If this list is empty then the GIII is implicitly using the system locally generated clock source.
NOTE Network connections should come before PBX and user connections in the Preferred Clock Source List.
Trunks not in the Preferred Clock Source List – Contains a list of the trunks available but not selected as a fallback clock. For the pair of boxes, use the drag and drop to move a trunk from one list to the other. To add a clock source to the Preferred Clock Source List, drag a trunk from the Trunks not in the Preferred Clock Source List. List into the Preferred Clock Source List.
To remove a clock source drag it back into the Trunks not in the Preferred Clock Source List.
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The order of the clock sources can be changed by dragging a trunk and dropping into a new position.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.3.2 Call Setup Timers The Call setup timers tab is used to configure timeouts that will be applied to all calls passing through the system.
Figure 5-28 General – Call setup timers
If the checkboxes are not selected, the configuration options are hidden.
Figure 5-29 General - Call setup timers
The selected checkboxes for incomplete dialling timeout and maximum ringing period timer will show the configuration options below. Incomplete Dialling timeout – This is used to prevent an unfinished call from using up a system resource. When the value is one or more, then a timer is started when an incoming call is detected. If an outgoing call has not been made and reached a ringing state when the timer reaches the Incomplete dialling timeout value, the call is disconnected with the specified Cause. If ringing is detected within the timeout period, then the Incomplete dialling timeout is cancelled. Maximum ringing period timer – This is similar to the incomplete dialling timer but starts at the ringing stage and goes on until connect is received or the specified Maximum ringing period is reached. If the Maximum ringing period is reached before the outgoing leg is connected, then the call is cleared with the selected Cause.
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CAUTION If a Maximum Ringing Period is used, the timer should be set to a high value, at least 180 seconds (3 minutes) is suggested. Lower values may cause the call to drop prematurely.
NOTE The timeout release will not work on most CAS protocols as they do not allow the call to be cleared by the B party.
Cause values The operation of the Cause type control group is identical for both the Incomplete Dialling and Maximum Ringing timers.
Default – When selected will use the generic clearing cause of NUMBER UNOBTAINABLE for incomplete dialling, or NO ANSWER for Maximum Ringing. Raw – This option allows a protocol specific release code to be used. For example, 17 would be entered to clear an incoming EuroISDN call with User Busy. Generic – This option allows the clearing cause to be selected from a list of generic causes. This allows an appropriate protocol specific cause to be automatically selected when clearing the incoming call leg. For example selecting NUMBER BUSY will use the correct busy release cause for the incoming protocol.
Cause location The Cause location field applies to SS7 and Q.931 calls only, and is ignored by all other protocols. This control allows the location field in the Q.850 cause parameter to be set when the timer clears a call. A location of User will be applied when Default is selected.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.3.3 Alarm mapping GIII can transmit a layer 1 alarm in response to a received layer 1 alarm, for example to warn backward switches that it is no longer able to forward their calls. The Alarm Mapping page is used to define how an incoming alarm on one trunk should be used to generate outgoing alarms on other ports.
NOTE Alarms cannot be generated on IP telephony trunks.
To add a new alarm map, drag the Drag to add an alarm map into the Alarm Map List. The new map will be added in the position it is dropped, and the configuration page for this new alarm map will be shown. An alarm map can also be created by copying an existing entry in the list by selecting the Clone button on a particular alarm map.
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To update an existing map, select the Edit button on that alarm map entry, this will show the configuration page for this entry. To remove an alarm map, drag the alarm map entry into the Trash Can. The entry, or entries will not actually be deleted until the Submit button is selected.
Figure 5-30 System configuration alarm mapping tab option
Figure 5-31 Alarm map problem
If an item has a red exclamation mark then it is indicating a problem with the entry. If the pointer is hovered over the exclamation mark then a tooltip will show information regarding the problem.
Figure 5-32 Alarm map configuration page with SIP
If there is just the SIP trunk selected in the Trunks receiving alarm list then there will only be three Incoming alarms that can be configured.
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Figure 5-33 Alarm map configuration with TDM trunks
When an incoming TDM trunk is checked then the Remote and Multiframe alarms are also shown. Name – An identification string for user reference only. Incoming – Specifies the incoming trunks to be checked for received alarms. The trunks that are checked are the trunks that the alarm map will check. If only one trunk is selected from the list, then only this trunk is checked. However, if two or more trunks are selected, then all the selected trunks must be in an alarm condition for the outbound alarm to be generated. If you wish to monitor Trunk 0 OR Trunk 1 rather than Trunk 0 AND Trunk 1, then create an alarm map for each trunk. Outgoing – Lists the trunks on which the mapped alarms should be generated, any number of trunks can be checked. The Incoming alarm and Outgoing alarm sections show the alarms are mapped. Only a single outgoing alarm can actually be generated on a trunk at any one time, and this will be the most severe alarm.
Alarm precedence For the purpose of alarm mapping, when an alarm is received on a trunk, all of the less severe alarms are also considered to be present on that port. The following shows the comparative severity of each alarm, where NOS is the most severe: No signal (NOS)
Loss of synchronisation (LOS)
Alarm indicator signal (AIS), IP link failure (Ports A and B both Disconnected)
Remote alarm indication (RAI), Multiframe alarm (MFA)
Only one alarm can be generated on a trunk at any time. Should the mapping scheme require multiple outgoing alarms to be generated; only the most severe alarm will be applied.
CAUTION If your PBX has alternative routing, generating alarms to a PBX may force it to re-route calls. Sending Layer 1 alarms to the PTT is highly inadvisable, as this may put the line out of service requiring you to request that the line is returned to service.
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NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.3.4 Advanced options The Advanced options tab contains several configuration parameters, which are used to either enable or disable specific functions. The functions are useful in certain specific circumstances or modes of operation. The Advanced options tab is located in the Role Configuration – General menu item.
Figure 5-34 Advanced options
These parameters should only be selected when the operator completely understands what is required, and the effect the parameter will have. Use Network Congestion Cause – When GIII cannot route a call, the call is released with a user busy cause. When this option is selected, the release cause is changed to Network Congestion. Map CAS Busy Port to Port – This option is used when converting between 2 different forms of CAS, for example, a CAS busy (abcd=1111) is mapped to the same timeslot on the other side. Please contact Aculab support for advice before using this function. Support Charge Mapping – This option allows charging information to pass through the GIII, however this is restricted to a limited number of protocols. Please contact Aculab support for further information. Number Unobtainable Tone for Route Fail – Generates number unobtainable tone instead of busy tone when a route failure occurs. Busy Tone for Incomplete Dialling Timeout – Generates BUSY tone instead of NU tone when timeout occurs. ETS/1TR6 to DPNSS Charge Mapping – This option enables the conversion of German meter pulses to meter pulses used on DPNSS.
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Disable AOC after Initial Disconnect – This option is used to ensure that AOC pulses/messages are stopped as soon as either of the parties clears down. If this is left unchecked, AOC pulses/messages will only stop when the calling party disconnects.
NOTE There is usually a delay between a called party clearing and a disconnect message being received, this is due to most called party switches having a delay to allow for called party reconnect, (moving to another phone on the same analogue connection).
Tone Mapping – When converting between CAS and ISDN you often get release causes in the ISDN signalling that cannot be mapped to an appropriate tone, this option causes any call reject except Number Unobtainable and User Busy to generate a selected tone. Tone when call fails – Select whether a Busy or Number unobtainable tone should be generated on call failure. Select None, to not generate a tone.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.4 Trunks configuration There are two types of trunks: SIP – Handles inbound and outbound VoIP calls on a node TDM – Handles inbound outbound ISDN calls on a node To edit the setting for a trunk, go to Role Configuration – Trunks, and select the SIP or the TDM trunk tab required.
Figure 5-35 Trunk configuration setup tab option (SIP example)
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Figure 5-36 Trunk configuration setup tab option (TDM example)
5.4.1 SIP Configuration The SIP tab can be found on the Trunks menu item on the Role Configuration menu.
5.4.1.1 SIP Trunk Setup The Trunk Setup tab is used to configure the IP address settings for traffic interface. This interface is used to handle both inbound and outbound SIP and RTP traffic.
Figure 5-37 SIP Trunk Setup tab
Trunk name – specifies the user preferred name for the SIP trunk. This name will be shown either in place of, or as well as the SIP trunk in other parts of the website. IPv4 address – specifies the IPv4 address. This is purely the IP address without netmask or CIDR IPv4 netmask – specifies the IPv4 netmask, i.e. 255.255.255.0
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IPv4 default gateway – specifies the IPv4 default gateway for the traffic interface, and is optional. IPv6 address (optional) – specifies the IPv6 address IPv6 prefix (optional) – specifies the IPv6 prefix, this is required if the IPv6 address has a value IPv6 default gateway (optional) – specifies the IPv6 default gateway, and is optional Nameservers – This is not currently used in GIII on the traffic interface Search Domains – This is not currently used in GIII on the traffic interface
Figure 5-38 SIP trunk with static routes
The New Route button, if pressed, will present a new line for configuration of a static route. Network address – The network destination, an IPv4 or IPv6 address Netmask or prefix – The IPv4 netmask, or IPv6 prefix Gateway – The IPv4 or IPv6 gateway address
NOTE The IP addresses configured on the SIP trunk are also used by SS7, if the node is part of a dual MTP3 pair.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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5.4.1.2 SIP - Call Handling The SIP – Call Handling tab contains advanced settings which affect how SIP calls being routed through GIII are handled.
Figure 5-39 SIP Call Handling
Incoming minimum digits – specifies how many digits must be collected before any attempt is made to route the call. A call will not be routed under any circumstances until this number of digits has been collected. This parameter can be set to zero if no minimum digit check is required. Use AOC mapping – When checked any charging messages received from the outgoing leg of an end-to-end call (regardless of the trunk on which the outgoing call is placed) will be mapped through to the incoming leg.
NOTE AOC mapping will only be successful when the protocols used for both the incoming and outgoing legs of the call support charge messages. Contact Aculab support for further information on the charge mapping capabilities of the protocols you wish to use.
NOTE AOC mapping is not supported by CAMA ports.
Routing Failed Cause – If an incoming call on the selected trunk cannot be routed, it is cleared by GIII. The default clearing cause is user busy. When user busy is not appropriate, subject to the protocol being used, the cause may be manually set to another cause value. Disconnect Calls on layer 1 Alarm – Checking the Disconnect Calls on layer 1 Alarm field causes GIII to clear calls when a layer 1 alarm is received on the selected port. An extra field will be display for configuration: Wait (Secs) before Disconnecting. Wait (Secs) before Disconnecting – The number of seconds to wait before disconnecting.
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5.4.1.3 SIP - Default codecs The SIP - Default Codecs tab allows the default codec list used by IP telephony calls to be configured. This list will be offered by all IP telephony call legs that do not have an explicit codec list configured in their route.
Figure 5-40 SIP Default Codecs Tab
The codecs to be used can be dragged from the codecs not selected for use box into the Currently selected codec order box. Once the codec is dropped into the currently selected box, any available configuration controls will be presented on the codec item. To add a codec, drag the required codec from the codecs not selected for use list to the Currently selected codec order list. To remove a codec, drag the required codec from the Currently selected codec order list into the codecs not selected for use list. To change a codec priority, drag the codec in the Currently selected codec order list and move it to the preferred position. Priority order is top down.
NOTE The G.726 and Clear Channel codecs are not available when making or receiving secure SIP calls.
The VAD (voice activity detection) settings applied to the transmitted RTP stream for the selected codec may be configured using the VAD control on the codec.
Figure 5-41 Codec VAD settings
VAD – The options available depend upon the codec. When a G.711 or G.726 codec is selected the options available are:
Disabled – No silence suppression will be applied. The received audio stream will be packetised and transmitted across RTP
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Muting (DTX) – Discontinuous transmission. The received audio stream will be analysed, and no packets will be sent across RTP when silence is detected. Comfort Noise – This will offer a comfort noise codec (RFC 3551, payload 13) to the remote endpoint. If this is accepted any silence detected will be replaced by comfort noise packets in the RTP stream. If the remote endpoint does not accept the comfort noise codec then DTX will be used.
When using a G.711 or G.726 codec, if the remote endpoint offers a comfort noise codec this will be accepted by GIII regardless of the VAD settings, and any comfort noise packets received decoded into the audio stream. When a G.729A codec is selected the options are for Annex B, and the options available are:
Annex B – When selected the G.729AB codec will be offered to the remote endpoint. If this is accepted the voice activity detector may transmit both 'A' and 'B' frames on the RTP stream, otherwise only 'A' frames will be transmitted. Disabled – When selected the voice activity detector will generate only 'A' frames on the RTP stream.
When using a G.729A codec, any 'B' frames received from the remote endpoint will be accepted and decoded regardless of the setting in this control.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.4.1.4 SIP - Authentication SIP authentication is configured in the Authentication tab on the SIP tab for the Trunks page, which is found in the Role Configuration menu.
Figure 5-42 SIP Authentication
This page allows SIP credential information to be configured, which will be used when making calls to authenticating SIP proxy servers and endpoints. The page shows each authentication record configured, and has the following columns:
Authentication Domain – This is the SIP realm for which the authentication record is to be used, and is the string that will be sent by the remote endpoint to identify itself. Username – The username that GIII will return when challenged by that domain. Password – The password for the above user. This will always be displayed as stars.
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New Authentication Entry – Click this button and a new row will be created allowing a new set of authentication credentials.
Figure 5-43 A SIP Authentication entry
Delete – Click the Delete button on the right of an entry to delete it. The entry will be removed from the page. Edit – To edit an entry simply click in the entry field you wish to change and modify. Click the Submit button to apply the changes, or Cancel to discard the changes.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.4.1.5 SIP - gateways The SIP Gateways tab is used to manage the routing of outgoing SIP calls to gateways. You will need to configure the settings in this tab if you wish to direct any outgoing SIP calls to a gateway. The procedure for gateway routing is:
• A pool of the individual gateways to which GIII can direct outgoing SIP calls is created.
• Gateway routes are built, each containing one or more gateways from the pool. A gateway may only be used once within an individual route, but can be used in multiple routes.
Each gateway route contains one or more gateway groups that are placed in fallback order, with each group holding one or more gateways.
When GIII selects the gateway for an outgoing call the search begins at the first group in the gateway route, with the groups being searched in list order until an available gateway is found. Within a group the gateways are searched in round-robin fashion, with each group maintaining it’s own round-robin pointer. This allows the benefits of fallback and round-robin gateway selection to be combined.
To implement a purely fallback strategy place each of your gateways into a group of its own. To implement a purely round-robin strategy, place all of your gateways into a single group.
• In Role Configuration – Routing – Groups, configure your trunk group(s) to use gateway routing as described in the Group Configuration section. Multiple trunk groups can use the same gateway route.
Use the controls on the Gateways, Gateway Routes and Gateway Retries tabs to configure gateway routing.
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NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
Gateways Use the controls on the Gateways page to create a pool of SIP gateways to which outgoing SIP calls can be directed. Up to 100 gateways can be configured.
Figure 5-44 SIP Gateways
The Gateways tab shows a list of configured gateways, and contains the following columns:
Gateway Name – the name of the gateway. If not configured then the gateway will be identified by the IP address Gateway Address – the IP address of the gateway. Port – the IP port on the gateway to which SIP calls will be directed. For clarity, if the default port number of 5060 is to be used there will be no entry in this column. Capacity – the call capacity of the gateway. For clarity, if the gateway has an unlimited capacity then the value 0 is used. Monitor – indicates whether GIII is monitoring the gateway for availability. This can have selected either Off, UDP, or TCP. Enable – indicates that the gateway is currently eligible for selection when routing outgoing calls.
Use the Insert New Gateway button to add a new gateway to the list. A Gateway can also be copied using the Clone button associated with an existing Gateway entry. Entries are removed from the gateway list using the Delete button associated with the Gateway. Gateway Address – the IP address of the gateway. All calls made to the gateway will be directed to this address. Port – the IP port on the gateway to which calls will be directed. The default port number is 5060.
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Capacity – the call capacity of the gateway. Leave this field blank to allow unlimited calls to be made to the gateway. When a call capacity has been applied, GIII will only make outgoing calls to a gateway when the total number of calls in progress to that gateway (incoming and outgoing) is less than its capacity. Call capacity is not checked when receiving incoming calls from the gateway - these will always be accepted. Enable – un-tick this box to prevent GIII routing new calls to the gateway. This control can used to stop calls being routed to a gateway that is temporarily unavailable, for example when the gateway is undergoing maintenance. Calls already in progress to and from the gateway will not be affected. Unticking this box will stop all gateway routes from directing calls to the gateway. Monitoring – change this from Off, to either UDP or TCP if you want GIII to monitor the availability of the gateway to accept calls. When availability monitoring has been enabled, GIII will send a SIP OPTIONS method to the gateway at one minute intervals. If the gateway replies with a 2xx response then GIII will continue to direct outgoing calls to the gateway. If the gateway replies with any other response, with the exception of 1xx which is ignored, then no further calls will be directed to the gateway until a subsequent OPTIONS method has received a 2xx response. Setting Monitoring to UDP or TCP determines the transport protocol that should be used when sending a SIP OPTIONS method to the selected gateway.
NOTE The UDP or TCP setting applies to the SIP OPTIONS method only, and does not influence the transport that will be used for SIP call control signalling.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
Gateway Routes Use the controls on the Gateway Routes page to create the gateway routes that will be used to select the destination IP address for each call. Up to 40 gateway routes can be created. The workflow used for entries on the Gateways page is quite complex. Due to the amount of information that can be stored in each gateway route, the entries can be shown ‘collapsed’, and may look like they have no configuration, but this may not be true. By default the first Gateway Route will be shown fully expanded on entry to this page, but it can still be collapsed.
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Figure 5-45 SIP Gateways Routes
The Gateway Routes are listed in rows. Use the Insert New Route button to add a new route to the list. Entries are removed from the route list using the Del button. A gateway route has a name which defaults to ‘Gateway Route X’, with X being a number. This can be changed by editing the name. A gateway route is composed of Gateway Groups. The groups in a route are in fallback order. To change the order an entry can be dragged and dropped into a different position. A gateway group is composed of gateways. A gateway route entry has a Select button which if used will expose gateway groups in the route, and the button will change to show Selected >. A gateway group also has a Del and Select button. The Del button will delete the group if selected. The Select button will expose the gateways configured and available, and the button will change to Selected >. Figure 5-39 Gateway Route view and configuration shows a route with both a group and gateways exposed.
Figure 5-46 Gateway Route view and configuration
Gateway Groups Gateway groups can be added to a route by selecting the Insert new Group button. An entry will be added with a default name of ‘Gateway Group X’, where X is a number. The name can be changed by editing the name. There are two buttons shown with a gateway group item, Del and Select. The Del button can be used to delete a group from the route. The Select button will expose the gateway configuration, showing any gateways configured, and any gateways available to be added to the group.
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Figure 5-47 viewing SIP gateway routes
Initially a group will not contain any gateways, and will show any gateways that are available. The Gateways available list will show any gateways not currently in use by the route being configured.
Figure 5-48 Gateway Group - Gateways available
To add a gateway into the group, it is dragged from the Gateways available list, and dropped into the Gateways in this Group list. The shadow background of the area to drag to and from is shown.
Figure 5-49 Gateway Group - drag gateway into group
If a gateway is successfully dragged and dropped into a group it will be added to the group and show an Enabled checkbox.
Figure 5-50 Gateway Group - Enable gateway
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Enabled – untick this checkbox to prevent the currently selected gateway route from directing calls to the selected gateway. Calls already in progress to and from the gateway will not be affected. This control is intended for use during setup and fault finding. Use the Enable checkbox in the Gateways tab if you wish to prevent all routes from directing calls to a gateway.
To remove a gateway from a group, it is dragged from the Gateways in this Group list and dropped into the Gateways available list.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
Gateway Retries Use the controls on the Gateway Retries page to specify what action should be taken when an outgoing call to a gateway fails. The settings are applied globally to all outgoing SIP gateway calls.
Figure 5-51 Gateway Retries
Call Retries – If this is set to None, then no action is taken, but if set to a value other than None then any outgoing calls rejected with a response code from the Retry Responses list will be retried using the next available gateway. The value specifies the maximum number of times a call may be retried before being rejected back to the caller. If a call cannot be connected after the specified number of retries, then it will be rejected using the last response code returned. If the list of available gateways is exhausted before the maximum number of retries, then the call will be rejected with routing failed. Retry Responses – These are only used if Call retries is not set to None. If a call to a gateway is rejected with one of the response codes in the list then the call will be retried. Response codes are added using the Insert new button. The values can be edited in place. To delete a retry response, click the Del button.
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Wildcards may be used, and the following are examples of how response codes can be defined:
604 – The specific response code 604,
48x – Any response code between 480 and 489,
3xx – Any response code between 300 and 399.
Unavailable Responses – If a call to a gateway returns one of the responses in this list, and the gateway has availability monitoring enabled, then that gateway will be marked as unavailable until such time as it has returned a 2xx response in reply to a SIP OPTIONS method. Use the Insert New button as described above to add, modify and remove list entries. Call Recovery Delay – This is used to select the time that must elapse without receiving an RTP packet from the remote end before call recovery is invoked. If set to No recovery then no call recovery is used. An explanation of SIP call recovery is covered in its own section of the document.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.4.1.6 Protocol Settings
Figure 5-52 SIP Protocol Settings
The controls on the Protocol Settings tab are used to configure settings that are specific to SIP calls.
Aculab Custom SIP Headers These custom SIP headers are Aculab specific, and allow certain call information to be passed between the TDM and SIP legs of a call., Call Information – When enabled an Aculab-Call-Information header will be presented to the remote endpoint as described in the SIP custom headers Appendix. Functional Number – Enable this control to allow production and processing of an Aculab-Functional-Number header. See Function Number mapping section for an explanation.
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SS7-Specific Custom Headers These headers only apply to calls interworking between SIP and SS7. The Aculab SS7 custom headers are fully described in the SIP Custom headers Appendix.
SS7 Calling Party Category – When enabled and an Aculab-SS7-Calling-Party-Category header is presented by an incoming SIP call, the information contained will be mapped to an outgoing SS7 call. The calling party category information presented by an incoming SS7 call will be used to generate an Aculab-SS7-Calling-Party-Category header in the outgoing SIP call. When disabled no mapping is carried out in either direction.
NOTE When interworking from SIP to SS7, any Aculab-SS7-Calling-Party-Category header present will be overridden by an explicit setting in the routing table.
SS7 Carrier Identification – When enabled and an Aculab-SS7-Carrier-Identification header is presented by an incoming SIP call, the information contained will be mapped to an outgoing ANSI SS7 call. Any carrier identification information presented by an incoming ANSI SS7 call will be used to generate an Aculab-SS7-Carrier-Identification header in the outgoing SIP call. When disabled no mapping is carried out in either direction. SS7 Carrier Selection Information – When enabled and an Aculab-SS7-Carrier-Selection-Information header is presented by an incoming SIP call, the information contained will be mapped to an outgoing ANSI SS7 call. Any carrier selection information presented by an incoming ANSI SS7 call will be used to generate an Aculab-SS7-Carrier-Selection-Information header in the outgoing SIP call. When disabled no mapping is carried out in either direction. SS7 Charge Number – When enabled and an Aculab-SS7-Charge-Number header is presented by an incoming SIP call, the information contained will be mapped to an outgoing ANSI SS7 call. Any charge number information presented by an incoming ANSI SS7 call will be used to generate an Aculab-SS7-Charge-Number header in the outgoing SIP call. When disabled no mapping is carried out in either direction. SS7 Jurisdiction Information – When enabled and an Aculab-SS7-Jurisdiction-Information header is presented by an incoming SIP call, the information contained will be mapped to an outgoing ANSI SS7 call. Any jurisdiction information presented by an incoming ANSI SS7 call will be used to generate an Aculab-SS7-Jurisdiction-Information header in the outgoing SIP call. When disabled no mapping is carried out in either direction. SS7 Originating Line Information – When enabled and an Aculab-SS7-Origination-Line-Information header is presented by an incoming SIP call, the information contained will be mapped to an outgoing ANSI SS7 call. Any originating line information presented by an incoming ANSI SS7 call will be used to generate an Aculab-SS7-Originating-Line-Information header in the outgoing SIP call. When disabled no mapping is carried out in either direction.
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SS7 Transit Network Selection – When enabled and an Aculab-SS7-Transit-Network-Selection header is presented by an incoming SIP call, the information contained will be mapped to an outgoing ANSI SS7 call. Any transit network selection information presented by an incoming SS7 call will be used to generate an Aculab-SS7-Transit-Network-Selection header in the outgoing SIP call. When disabled no mapping is carried out in either direction.
Other Headers P-Asserted-Identity use Errata 3744 – Use IETF Errata RFC 3744 when populating P-Asserted-Identity Header
Figure 5-53 SIP Protocol Settings - Miscellaneous Parameters
Outgoing Diversion Limit – Use this control to specify how many times an outgoing SIP call may be diverted before being cleared with an appropriate number changed cause. Setting the control to zero will allow unlimited diversions. History-Info Headers – The options in this dropdown are used to manage the production of SIP History-Info headers. History-Info headers are described in IETF RFC 4244 – Request History Information.
Disabled – Disable the inclusion of history-info headers are included in outgoing SIP call legs, or backwards on incoming SIP call legs. Enabled – Enable the production of History-Info headers. When enabled History-Info headers will be sent in all outgoing SIP call legs, and backwards to any incoming SIP call legs that have presented a Supported header containing the histinfo tag. Enabled with Privacy – Place a Privacy header containing the history tag into any SIP request/response containing History-Info headers.
User-to-User Text – The text in this field will be appended to all User-to-User headers included in both incoming and outgoing SIP call legs, using a content tag. If this field is blank then no content tag will be appended to the header. User-to-User headers are described in IETF document draft-ietf-cuss-sip-uui-09. Differentiated services format – Sets the type of service field within the IP header of real time protocol (RTP) transmitted packets.
Code Point – allows the setting to be selected from a list of commonly used named constants, whose values are based upon the information provided in IETF RFC 4595. Raw (hex) – the setting is entered as a hexadecimal value, allowing settings for which no DSCP constant is provided to be used.
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DiffServ value for SIP – the Code Point or Raw (hex) value which is sent in the type of service field.
Real Time Text These settings define how the SIP RTT stream will be negotiated by systems configured for TTY operation. These settings have no effect on systems configured for standard operation.
Use Real Time Text (RTT) –Specify how media negotiations initiated by GIII will be carried out. There are three settings:
Never – Use this option if RTT support is not required. Always – Use this option if GIII should always request an RTT stream. On Request – Use this option if GIII should not request an RTT stream until a TTY device has been detected on the TDM side.
RTT Redundancy Level – Use this control to specify how many levels of T.140 redundancy will be applied to the RTT stream.
Trunk Group Parameters These settings are used to add trunk group parameters to the Contact header in a SIP INVITE. Trunk group parameters are described in IETF RFC 4904 – Representing Trunk Groups in tel/sip URIs.
Trunk Group Contact Name – The contents of this field will be added as the user part of the sip URI in the contact header. Trunk Context for Contact Name – The contents of this field will be used as the trunk-context parameter in the Contact header.
Timer T1 – Allow adjustment of the T1 SIP timer. Most of the timers in SIP scale with timer T1 which by default is 500ms. It is best to look at RFC 3261 for reasons why it might be necessary to change this. Raw Cause on Timeout – Set the Raw cause code returned for an outgoing SIP call, if the INVITE does not receive a 200 OK
NOTE The tgrp parameter in the Contact header will be set to the index of the group on which the incoming call leg arrived.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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5.4.1.7 SIP Proxy GIII can forward outgoing SIP calls to a proxy server for onward routing, and maintain alias registrations with the proxy server. If you wish to route any of your outgoing calls to a SIP proxy server you must configure the settings on this page.
Figure 5-54 SIP Proxy
When directing a call to a SIP proxy server, GIII uses the address format: sip:[email protected]
1234567890 being the equivalent PSTN DDI number and 192.168.1.106 the proxy server IP address. Subject to how the proxy server is configured, it will either respond with a redirect message to another proxy server or endpoint, or it will pass on your request to another proxy or endpoint. This process will continue until the two call endpoints are identified at which time the call can proceed. Enable SIP Proxy Server routing – tick this box if you want to route outgoing calls via a SIP proxy server. Other controls on this page will only be displayed when this box is ticked. IP Address – the IP address of the SIP Proxy Server you wish to use. This can be either an IPv4 or IPv6 address. Port – the IP port that your SIP Proxy Server is using. The default is port 5060. Transport – the data format to be used to communicate with your SIP Proxy Server. The options available are:
UDP – user datagram protocol format. Connectionless feature of the IP protocol, the exchange of datagrams without acknowledgements or guaranteed delivery. TCP – transmission control protocol format. End to end IP protocol, the exchange of sequential data between host processes.
Registrations – This control group allows a list of contact addresses to be registered with a Proxy Server. GIII maintains these registrations from the point at which the role is active, until they are no longer required, for example, the role configuration is changed and applied, or the role is removed from the device. Registration list entries are added by selecting the Insert New button. They can be modified in place.
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Entries are removed from the registration list using the Del button on the entry.
Figure 5-55 SIP Proxy Registration
The Public Address or Alias must be added in the form sip:Alias@IPaddress or Alias@IPaddress or Alias. In the case of just the username part, the proxy address is used. The scheme may be specified, otherwise it defaults to ‘sip:’. The Contact must be added in the form sip:Contact@IPaddress or Contact@IPaddress. The IPaddress would normally be the IPv4 or IPv6 address of the SIP trunk, but it does not have to be. The scheme may be specified, otherwise it defaults to ‘sip:’.
NOTE To maintain flexibility, IP addresses are not automatically appended.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.4.2 TDM Trunks 5.4.2.1 Trunk Setup tab
The Trunk Setup tab option is used to define the parameters associated with a trunk. This includes: Trunk Name – A user defined identification string, which may be up to 63 characters in length. This name will appear within the status application when referencing the trunk. For ease of use it is recommended that you include the trunk number in this name i.e. DASS (5) etc.
NOTE There are a number of fields throughout the GIII applications that are not of sufficient width to display the entire 63 characters, and the name will be truncated when displayed in such fields. It is recommended that all names are uniquely identified within the first 30 characters.
Selecting the Trunks menu item from the Role Configuration menu will present a page with tabs for each TDM trunk (plus one for SIP). This configuration is for a specific role. Initially a TDM trunk configuration tab within a will show a page as below, with no trunk protocol configured.
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Figure 5-56 Trunk Setup - No protocol assigned
There is a drop down for TDM Protocol which allows a specific protocol to be assigned to this TDM trunk.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
Protocol selection
Figure 5-57 TDM protocol selection
The TDM Protocol drop down is used to select and configure the signalling protocol to be loaded onto a TDM port. The behaviour of the protocol firmware can be modified by the addition of configuration switches. The more commonly used configuration switches can be applied from the Available options box, by drag and dropping options into the Current options for this TDM protocol box.
NOTE This list is not guaranteed to contain all the switches that may be applied to a protocol.
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Figure 5-58 Available protocol options
There is a link provided to View protocol details, which will open another window with the Aculab description of the protocol and details about options available. Each configuration option also shows the option switch in brackets.
Adding an option Dragging an option from the Available options box to the Current options for this TDM protocol box, will present the ability to adjust any available options for the selected protocol option. For example, dragging across the Call Proceed to Alert Timer (-s20) option will result in the following settings being displayed.
Figure 5-59 Adding protocol option
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5.4.2.2 Edit an option setting The Edit button within a particular option can be used to change the setting of the option. Shown below is an example for editing the Call Proceed to Alert Timer settings.
Figure 5-60 Editing protocol option
Remove an option To remove an option, simply drag it back into the Available options box.
Additional switches string Those switches that are not in the Available options box can be added using the Enter additional switches string button, and will be displayed alongside the Enter additional switches string button. Pressing the Enter additional switches string button will display a page to enter or edit the additional switches string, as shown below.
Figure 5-61 Additional switches string
Hovering over a protocol firmware option will provide a more detailed description of the functionality it provides. Further details on some protocol options can be found in Appendix D.
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Figure 5-62 Protocol option tool tip
5.4.2.3 Protocol configuration for SS7 If a Trunk is configured to use an SS7 TDM Protocol then the Trunk Setup tab will allow Signalling link and ISUP Bearer configuration.
Figure 5-63 Trunk configuration
NOTE Please refer to the SS7 telephony section for details on how to configure SS7 links by editing the SS7 stack file ss7.cfg
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SS7 Signalling link setup Subject to any system constraints, for example, excluding TS 0 on an E1 port, use the E1_Signalling or T1_Signalling option to define which timeslots on the trunk are to be used as SS7 signalling links. The initial page for a SS7 T1 protocol trunk is shown below.
Figure 5-64 Initial page for SS7 T1 protocol trunk
To use the Signalling link option, drag the T1 Signalling Link of E1 Signalling Link option into the Current options for this TDM Protocol box as shown below.
Figure 5-65 SS7 Signalling option
To assign a timeslot as a signalling link click the Edit button on the E\T1 Signalling Link option and select the appropriate Timeslot, set the appropriate Originating Point Code (OPC), Destination Point Code (DPC), and Signalling Link Code (SLC) parameters as described below, and then select Submit.
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Figure 5-66 Configuring SS7 Signalling Link
Timeslot – Used to select the signalling timeslot to which the settings are to be applied. If the T1 control is checked only timeslot in the range 0-23 will be presented, otherwise timeslots in the range 1-31 will be available.
NOTE The OPC, DPC and SLC settings are derived from information supplied by your service provider.
Originating Point Code – This is point code used to identify GIII. Destination Point Code – This will usually be the point code of the SP or STP to which GIII is directly connected. Exceptionally, this may be the identity of a distant SP or STP.
NOTE Point codes will be in the range 1-16383 when using the ITU-T or UK ISUP variants, 1-16777215 when using the ANSI or CHINA variants.
Signalling Link Code – SLC is a unique numeric Signalling Link Code between zero and fifteen (0 - 15), allowing you to define up to sixteen SS7 signalling timeslots per set of point codes.
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Figure 5-67 Configured SS7 Signalling Link
ISUP Bearer Group Option To add an ISUP Bearer Group use the T1_Bearer_Group or E1_Bearer_Group option by dragging the option into the Current options for this TDM Protocol box.
Figure 5-68 SS7 ISUP Bearer Group option
The ISUP Bearer Group option is used to define which timeslots you wish to use as ISUP bearers. Each timeslot must be assigned an Originating Point Code (OPC), Destination Point Code (DPC) and CIC before it can be used as a bearer.
NOTE The OPC, DPC, CIC Base, CIC Map and Circuit Map settings are derived from information supplied by your service provider.
Originating Point Code (OPC) – This is the point code used to identify GIII. Destination Point Code (DPC) – The point code of the destination SP or STP. This may be the adjacent node, or another distant node on your network via onward SS7 links.
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NOTE Point codes will be in the range 1-16383 when using the ITU-T or UK ISUP variants, 1-16777215 when using the ANSI or CHINA variants.
First assigned Circuit Identification Code – This is the number assigned to the first timeslot allocated a Circuit Identification Code (CIC) on a trunk. The CIC number range used on one trunk to a given DPC must not overlap with the number range assigned to other trunks to the same DPC. For example, assuming that you are assigning CIC to all timeslots (excluding TS0) to multiple E1 trunks to a single DPC, the CIC Base would typically be 1 for the first trunk, 33 on the second trunk and so on in steps of 32. Individual CIC values will be assigned sequentially to each timeslot selected in the Assign CIC checkboxes. Assign CIC – This has a checkbox for each available timeslot, if a checkbox is checked then a CIC will be assigned to it.
Figure 5-69 SS7 Bearer Group Configuration
NOTE Multiple Assign CIC checkboxes can be cleared or checked by holding the selection and dragging across the checkboxes.
CAUTION
! You must press the Submit button to save the options configuration in the Trunk tab. The red exclamation mark is hinting at this. Pressing another Trunk tab will lose the currently configured settings.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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Adding additional switches Those switches that are not in the Available options box can be added using the Enter additional switches string button, and will be displayed alongside the Enter additional switches string button. Pressing the Enter additional switches string button will display a page to enter or edit the additional switches string, as shown below.
Figure 5-70 Additional switches string
5.4.2.4 Call Handling The TDM – Call Handling tab contains advanced settings which affect how SS7 calls being routed through GIII are handled.
Figure 5-71 SS7 Call Handling Options
Incoming Minimum digits – specifies how many incoming digits must be collected before any attempt is made to route the call. A call will not be routed under any circumstances until this number of digits has been collected. This parameter can be set to zero if no minimum digit check is required. After the minimum required digits have been received, the call will be routed on the first occurrence of any of the following:
Incoming Maximum digits – when enabled (set to a non-zero value), the call will be routed immediately the specified number of digits has been received. Incoming Digit timeout – when enabled (set to non-zero), the call will be routed immediately the timer expires. Route on EOS – if enabled, the call will be routed immediately the end of send indicator (as defined in the Setup screen) is detected.
Use AOC mapping – When checked any charging messages received from the outgoing leg of an end-to-end call (regardless of the trunk on which the outgoing call is placed) will be mapped through to the incoming leg.
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NOTE AOC mapping will only be successful when the protocols used for both the incoming and outgoing legs of the call support charge messages. Contact Aculab support for further information on the charge mapping capabilities of the protocols you wish to use.
NOTE AOC mapping is not supported by CAMA ports.
Call Clearing – Use the controls in this group to configure the behaviour during call clearing.
Bidirectional – This is the default setting and allows either the called or calling party to clear the call. PSTN Emulation – When an outbound call is made to a busy extension, the call is often rejected with a cause default value. Under normal conditions, this is passed back to the originating trunk causing the call to disconnect. Once a call is disconnected, it is not possible to send any tones back down the line. When checked, as the PSTN Emulation requires any call to be cleared forward, it effectively blocks the reject message and holds the voice path open until the originating party clears. PSTN emulation will be applied to all incoming calls received on this trunk. It will only be possible to clear the call forward, irrespective of which trunk the outgoing call is on. Outgoing calls on this trunk will only have PSTN emulation applied if the corresponding incoming port has PSTN Emulation applied. Therefore, PSTN emulation may be applied to some outgoing calls, whilst it may not be applied to others. PSTN Emulation is applicable to TDM calls only. Called Party – This setting applies to CAMA trunks only, and is described in the CAMA telephony section. When applied to non-CAMA trunks bidirectional call clearing will be used.
Routing Failed Cause – If an incoming call on the selected trunk cannot be routed, it is cleared by GIII. The default clearing cause is user busy. When user busy is not appropriate, subject to the protocol being used, the cause may be manually set to another cause value. If this field is blank, then the default clearing cause will be used. For example, Euro ISDN uses clearing cause 42 to indicate Switching Equipment Congestion. To use this cause instead of the default User Busy, enter 42 into the Routing Failed Cause field. Routing Failed Location – this field applies to SS7 and Q.931 calls only, and is ignored by all other protocols. The control allows the location field in the cause parameter to be set. A location of User will be applied when Default is selected.
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The options available for the Routing Failed Location are: Default User Private Network Serving Local User Public Network Serving Local User Transit Network Public Network Serving Remote User Private Network Serving Remote User International Network Network Beyond interworking point
Send Progress/Send Proceeding after the last digit – These controls are used to trigger the sending of a progress or proceeding message if no digits have been received for # seconds. Both messages have a progress element indicating that in-band tones are available. Sending progress/proceeding messages in this way can prevent incoming calls from timing out when interworking with very slow far end networks, or can be used to prompt switches to open the voice paths.
NOTE Some protocols, for example Euro ISDN, send a proceeding message automatically and will not allow a second one to be sent. In such instances use a progress message if you wish to open up the voice path.
Call Progress Location –This is used to configure how a call progress message received from an outgoing call leg will be mapped to the incoming call leg. When left at the default setting of Transparent the location field in a call progress message will be mapped between calls unchanged. In the case of, for example, mapping a network trunk to a network trunk, or a user trunk to a user trunk, it may be necessary to override the default and manually redefine the originator location part of the information. For example, redefining the originating location as a remote user trunk instead of a network trunk. The options available for the progress information element are:
Transparent User Private Network serving Local User Public Network serving Local User Public Network serving Remote User Private Network serving Remote User International Network Network beyond interworking point Transit Network
NOTE Not all values are supported by every protocol. Refer to the appropriate protocol specification for guidance.
SS7 Continuity Check – This control applies only to SS7 ports, and will be ignored by all other protocols. See the SS7 telephony Continuity checking section for an explanation of SS7 continuity checks.
Type – select the type of continuity check that is to be carried out by this port. The same type will be used for both inbound and outbound continuity checks. The available options are:
4-wire (Loopback) 2-wire (Transponder)
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Disconnect Calls on Layer 1 Alarm – This checkbox will, when set, display configuration elements related to alarm handling, and causes GIII to clear calls when a layer 1 alarm is received on the selected trunk.
Wait (Secs) before Disconnecting – The length of time that the trunk alarm must be continually present before calls will be disconnected.
Alarms that will be Ignored/Alarms that will cause Disconnect – This pair of boxes support drag and drop of items between them. As default, disconnect will apply to all of the alarms shown in the Alarms that will cause Disconnect box. On some occasions it may be preferable to ignore certain alarms and allow calls to proceed, for example when receiving a Remote Alarm Indication (RAI) that is caused by alarm mapping rather than by a fault. To ignore an alarm drag it into the Alarms that will be Ignored box.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.5 SS7 Configuration The SS7 configuration page has a three tabs by default: Add pointcode, Advanced Configuration, Interworking. It will also have a tab for each pointcode that has been created.
Figure 5-72 SS7 page with existing pointcodes
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5.5.1 Add pointcode tab
Figure 5-73 Add pointcode tab
The SS7 telephony section covers the use of the Advanced configuration to create more complicated SS7 configurations. The more basic configurations can be created using the Add pointcode tab, and it may not be necessary to use the Advanced configuration tab at all.
NOTE The M3UA Routing Keys and M3UA Connections panel tabs will only be shown once the cluster version has been changed to 2 (or more).
The various fields that can be configured are: Local pointcode – This is the SS7 local point code. The ITU variant must use a decimal (14 bits) number, 24 bit variants can use either decimal or the 8-8-8 format. ISUP variant – This can be either ITU, ITU UK ISUP, China, ANSI, or ANSI 2005. Network indicator – This can be either 0 – International, 1 – Spare (for international use only), 2 – National network, or 3 – Reserved for National use.
MTP3 Signalling: Signalling method – This can be either Local only, Dual master, Dual slave, or Remote only. Selecting Dual master, or Dual slave, will change the fields available, adding Dual peer address, Dual connect SLC, and Dual listen SLC. Selecting Remote only will remove the Support remote only nodes, and add new fields: Remote signalling host, and Additional remote signalling host. Dual peer address – The address. Dual connect SLC – This can be either Default for mode, or a value of 0-15. Dual listen SLC - can be either Default for mode, or a value of 0-15. Remote signalling host – The remote signalling host.
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Additional remote signalling host – An additional remote signalling host. Support remote only nodes – A checkbox to enable or disable supporting remote nodes only. This will not be displayed for the Remote only signalling method. Default route priority – This can be None, or a number between 0 and 127. STP point codes – A button, which if pressed will display an extra field to enter a point code. There can be a maximum of 6 STP point codes entered.
Figure 5-74 M3UA Routing Keys
M3UA Routing Keys: Routing Keys – Routing keys are added by pressing the Add key button. Context – This is a 32 bit unsigned integer. This defines the routing context for this routing key. It is also used as the Local-RK-Identifier when sending registration requests. Address groups – The address groups can be added by pressing the Add group button. This will add a new group of Remote PCs. To see each individual address group press the Select button. The address group can be deleted by pressing the Del button. Allow large messages – Defines if the remote end can support a broadband message length of 4096 SIF octets. remotePC – The signalling point code in the remote network, or at the remote end to which traffic is to be sent. There can be multiple remote point codes specified. To add a remote point code press the Add PC button and a new row will be added. Each row has a Del button which will delete the row if pressed.
Figure 5-75 M3UA Connections
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M3UA Connections: Connections – Connections can be added by pressing the Add connection button. This will add a new connection row. To see individual connection settings press the select button. A connection can be deleted by pressing the Del button. Default Port – The default port to be used for any connections not specifying a port. If this is left blank then the standard M3UA port number of 2905 will be assumed. Connection type – This can select Client, IPSP Client, or IPSP Server. The configurable items will change depending upon the type selected. Server addresses (Client) – To add a host group press the Add host group button. Each host entry can be of the form hostname[,hostname_1...] , where the hostname is either a numeric IP address, or a hostname. A port number can also be specified to overide the Port/Default Port settings on each addresse, i.e. 10.1.1.1:42905. To use a port number with IPv6, use an interface id of zero before the port number, i.e. fdac::1%0:42905. IPSP Server addresses (IPSP Client) – To add a host group press the Add host group button. The host group format is the same as described for the Client Server addresses. Allowed IPSP Client addresses (IPSP Server) – To add a host group press the Add host group button. The host group format is the same as described for the Client Server addresses. Port – The SCTP (or TCP) port number to use for this connection. This is the remote port for an outward connection (Client), and the local port for a listener (Server). Used routing key contexts – A list of routing key contexts used by this connection. The routing key contexts can be moved between the Used and Available drag and drop boxes. Available routing key contexts – A list of routing key contexts available to be used by this connection. The routing key contexts can be moved between the Used and Available drag and drop boxes.
NOTE The remote point codes do not need to be specified for SS7, as the SS7 driver can determine these for itself.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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5.5.1.1 SS7 configuration error messages The point code and variant used will dictate what is allowed to be configured. For ITU and ITU UK ISUP 2001, only decimal 14 bit numbers may be entered. For non ITU based variants, the 24 bit number may be entered in decimal, or in 8-8-8 format. Duplicate point codes are not allowed.
Figure 5-76 Invalid point code for variant
Figure 5-77 System message showing invalid point code error
Duplicate point codes would produce the following error:
Figure 5-78 Indication of an invalid point code
Figure 5-79 System message showing duplicate point code error
5.5.2 Advanced configuration tab
Figure 5-80 Initial empty SS7 configuration
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The use of the Advanced configuration to create more complicated SS7 configurations is detailed in its own section of the document. The more basic configurations can be created using the Add pointcode tab. If configuration is entered using the Add pointcode tab, then the configuration will appear in this text box.
Figure 5-81 viewing and editing the SS7 Advanced configuration
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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5.5.3 Interworking tab The Role Configuration - SS7 – Interworking page provides access to configuration settings that are specific to SS7 calls.
5.5.3.1 International tab
Figure 5-82 System configuration Protocol Settings - SS7, International tab
option Country Code – This is the international dialling code for the country in which GIII is located. Currently this field is only used when processing UK ISUP calls, and UK ISUP users must set this to 44.
International Dialling Prefix – This is the international dialling prefix used by the country in which GIII is located. Currently this field is used when processing UK ISUP calls only, and UK ISUP users must set this to 00.
International Interworking – Check this control if GIII is connected to an SS7 switch using a different country code. This control is currently unused and should be left at its default setting.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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5.5.3.2 ANSI tab options
Figure 5-83 SS7 ANSI Interworking
The controls on the ANSI page (illustrated above) are used to configure settings that will be used by calls placed on ports running the ANSI SS7 variant. SIP Interworking – Use this control to select the standards to be applied when interworking between ANSI SS7 and SIP. The available options are:
T1.769-2004 – Interworking will be carried out as described in American National Standard T1.769-2004.
NENA i3 – Interworking will be carried out as described in the (American) National Emergency Number Association stage 3 specifications. These specifications are based upon T1.769-2004, but have a number of alternative or additional mappings.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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5.5.3.3 UK ISUP tab options
Figure 5-84 SS7 UK ISUP
The controls on the UK ISUP page are used to configure settings that will be used by calls placed on SS7 ports running the UK ISUP variant. Partial CLI – These controls are used to configure the settings of the Partial Calling Line Identity (PCLI) parameter presented in an outgoing Initial Address message (IAM). The PCLI parameter will be included in the IAM if there is no Calling Party Number parameter. Your network operator will assign the values entered into the fields in this group.
Type of Switch – This value will be used in the Type of Switch field in the PCLI parameter, and should be a numeric value in the range 0-99. PNO Identity – This value will be used in the PNO Identity field in the PCLI parameter, and should be a numeric value in the range 0-999. Switch Number – This value will be used in the Switch Number field in the PCLI parameter, and should be a numeric value in the range 0-999. Partial CLI Octets – These fields allow octets 5-9 in the PCLI parameter to be configured, one octet per field. The values are entered into the fields in hexadecimal format and should be in the range 00 to FF.
Priority Numbers – The controls in this group are for use when interworking from SIP to UK ISUP. If the incoming SIP call does not present a P-Asserted-Identity header containing a calling party’s category, then the Priority Numbers table will be searched using the DDI presented by the SIP call. If the incoming DDI can be matched against one of the entries in the table then the Calling Party’s Category parameter in the outgoing Initial Address message (IAM) will be set to ‘calling subscriber with priority’, otherwise the Calling Party’s Category will be set to ‘ordinary calling subscriber’.
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NOTE The Calling Party’s Category determined by priority number matching can be overridden by the settings in the Routing Configuration screen.
Priority number list entries are added using the New Number button. The entries can be edited in place. The following wildcards are available to allow a range of priority numbers to be matched.
? – may appear anywhere within the number and will allow a single digit in the range 0-9 to be matched. % - may be used to terminate the number, and will allow zero or more trailing digits in the range 0-9 to be matched.
Entries are removed from the priority numbers list using the Del button.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.5.4 Point code tab The point code tabs are shown for existing configuration. A point code may be deleted by pressing the delete button. This will present a confirmation dialog.
Figure 5-85 Point code deletion confirmation dialog
NOTE The settings are saved to the current role configuration by pressing the Confirm delete button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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5.6 Routing configuration The Routing page has three tabs: Groups, Routes, and Cause Maps. It is recommended to configure the Groups before Routes, as the Routes use Groups in their configuration.
5.6.1 Groups configuration Groups are used as the basic elements for routing of a call. They define a number of resources (trunks and timeslots) being used for a common purpose. For example, if any one of a number of Trunks can be used to make an outgoing call, then all the Trunks can be put into one Group. The Groups configuration page is a tab in the Routing menu item under the Role Configuration menu.
Figure 5-86 Routing Configuration - Groups
To add a new group drag the Drag to add a group button into the Groups List box. This will display a page to configure a new group. If entries already exist then you can copy an existing group by clicking the Clone button on the group you wish to copy. A group can be modified by clicking the Edit button. To remove a group from the Groups list, you drag the group item into the Trash Can and press the Submit button.
CAUTION Deleting a group will cause any route using that group to be changed to <None> for its Group setting.
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Figure 5-87 Groups - create a new group
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.6.1.1 Group Configuration A Group can contain more than one Trunk. A Trunk can belong to more than one Group. A Group definition can contain one or more subsets of the available timeslots on a Port. Individual trunks can be moved between groups or new trunks can be added to groups without affecting any of the routing definitions.
Figure 5-88 Groups configuration
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Name – A unique identification string for user reference only. Clear list – A button which if pressed will remove all trunks from the Trunks List Add all – A button which if pressed will add all the trunks in the role to the Trunks List. The list can then have trunks removed to achieve desired trunk list. Hunt strategy – Specifies how the outgoing timeslot will be selected from the resources defined for the Group. The options are:
Lowest Available – places the call on the available timeslot that is closest to the beginning of the Group. Highest Available – places the call on the available timeslot that is closest to the end of the Group. Follow Input – places the call from the incoming group onto the corresponding port/timeslot in the outgoing port. This hunt strategy can only be used where the incoming and outgoing groups are of identical composition (i.e. contain the same number and type of ports, in the same order, with all bearer timeslots in identical positions). If the corresponding outgoing timeslot is being used, routing will fail. Random – selects a timeslot at random from those in the Group that are not currently carrying calls. Cyclic – starts with the first timeslot in the Group, and thereafter selects the next available timeslot following the last one used. When the end of the group is reached the process starts again at the beginning of the Group. One to One – is similar to Follow Input but only requires the incoming and outgoing groups to have an identical number of bearer timeslots. Corresponding timeslots are determined by counting from the beginning of the group, which allows channel mapping between E1s and T1s. If the corresponding outgoing timeslot is being used, routing will fail.
Load Balancing – is used to balance calls across all ports in a group by looking for the trunk that is currently carrying the fewest calls.
NOTE IP telephony has no concept of timeslots, When configuring a group for outgoing IP telephony calls the Load Balancing hunt strategy should always be selected.
NOTE The hunt strategy only has an effect if the Group is on the outgoing side of a route. The equipment sending the call controls the incoming timeslot.
Figure 5-89 SIP Routing controls added when SIP Trunk added
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SIP Routing – These controls will only be shown if the group contains at least one IP telephony port, and are used to specify how the IP address to which an outgoing SIP call is directed will be selected:
Use Proxy – The outgoing SIP call will be directed to the currently configured SIP proxy server for onward routing. Configuring a SIP proxy server is covered in the SIP trunk section of the document. Use Gateways – The outgoing SIP call will be directed to one of the gateways in the selected gateway route. Configuring SIP gateway routing is covered in the SIP trunk section of the document.
Gateway Route – Select the gateway route to be used by this group from the drop down list. This control will only be enabled when Gateway is selected.
NOTE SIP routing only has an effect if the Group is on the outgoing side of a route.
Trunks List – Contains a list of the trunks in the group. To remove a trunk from the group, drag the trunk item into the Trash Can. The Clear list button will remove all trunks from the group.
Figure 5-90 TDM trunk added to Trunks List
To add a trunk to the group, drag the Drag to add a trunk control into the Trunks List box.
NOTE When selecting a timeslot for an outgoing call, trunks are searched in the order in which they are listed in the Trunks List box.
For each trunk in the group, you must specify the range of timeslots to be used. IP telephony has no concept of timeslots, and the timeslot controls will not be displayed if an IP telephony trunk is selected.
Figure 5-91 TDM trunk selection and timeslot configuration
Trunk Select the trunk to use using the Trunk control. Slots
First Set the low (first) timeslot for the selected port. Last Set the high (last) timeslot for the selected port.
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NOTE For simplicity framing and signalling timeslots should be included in the timeslot range, as they will be automatically excluded from selection during operation.
Minimum Digits Timer – The purpose of the Minimum Digits Timer is to route a call if it arrives with insufficient dialled digits. If the number of digits specified in the Incoming Minimum digits field on the Call Handling tab for the Trunk in the Trunks page have not arrived within a specified time then the outgoing call will be redirected to a default destination.
Timeout secs – The time to wait for the minimum number of digits to be dialled. Default DA – The destination to which the call will be redirected.
The minimum digit timer can be disabled by specifying a timeout of zero seconds.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.6.2 Routes configuration Routing definitions are used to specify how GIII will select the trunk and timeslot to be used for an outgoing call, and to configure settings that are applied on a call-by-call basis. Routing is configured using the Routes tab in the Routing menu item, which is found in the Role Configuration menu.
Figure 5-92 Routes Configuration
The Routes page displays a list of configured routes for the role selected. As can be seen in Figure above Routes Configuration, the page is displaying one route, Route66, which has been configured for the role Bank. To add a new route, drag the Drag to add a route control into the Routes List box, this will then show a route configuration page.
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A new route can be copied from an existing one by selecting the Clone button on the desired route to copy. To edit a route select the Edit button on the route. To delete a route drag the route item from the Routes List box into the Trash Can. To change the position of a route within the Routes list, drag and drop the route in the new position.
NOTE The order of the routes is the order that a call searches until it finds a route that meets the required parameters.
The configuration page for a route has a number of tabs at the top, which when selected will display related configuration items.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.6.2.1 Route Setup tab
Figure 5-93 Routing configuration setup tab option
Name – An identification string used for reference only.
Incoming Call fields The Incoming Call fields specify the criteria an incoming call must meet in order to use the route. Destination address – The incoming dialled number must match this field. Wildcards may be used to allow the route to handle a range of numbers.
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NOTE The destination address presented by an incoming SIP call may contain a leading ‘+’ and/or embedded ‘–‘ characters. When parsed from the SIP headers such characters are discarded, will not be presented to the routing mechanism, and should not be included in the number matching mask. For example to match a dialled number presented as +44-1908-273800 use the mask 441908273800.
Destination nature of address – This allows the called party nature of address indicator in the incoming call to be matched. The options available along with their corresponding protocol values are:
All Subscriber (1) Unknown (2) National (3) International (4) Network Specific (5) Network Routing (National) (6) Network Routing (Network Specific) (7) Network Routing (Called Directory) (8) Subscriber, Operator Requested (113) National, Operator Requested (114) International, Operator Requested (115) No Number, Operator Requested (116) No Number, Cut Through (117) 950+ Call (118) Test Line Test Code (119) UK Specific Address (126)
Select All to allow any nature of address setting to be matched.
NOTE Nature of address is only supported by the SS7 protocol, and some variants may not support all available settings.
Originating address – The incoming calling number must match this field. Wildcards may be used to allow the route to handle a range of numbers.
NOTE The originating address presented by an incoming SIP call may contain a leading ‘+’ and/or embedded ‘–‘ characters. When parsed from the SIP headers such characters are discarded, will not be presented to the routing mechanism, and should not be included in the number matching mask. For example to match a calling number presented as +44-1908-273800 use the mask 441908273800.
Alow SIP From Header as CLI – This setting applies to incoming SIP calls only. GIII expects an incoming SIP call to present the calling party number in a P-Asserted-Identity header. If the SIP call does not present a P-Asserted-Identity header, the call is considered not to have a calling party number. Ticking this checkbox will allow GIII to extract the calling party number from the From header if no P-Asserted-Identity header is presented.
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Originating nature of address – This allows the calling party nature of address indicator in the incoming call to be matched. The options available along with their corresponding protocol values are:
All Subscriber (1) Unknown (2) National (3) International (4) Non-Unique Subscriber (113) Non-Unique National (115) Non-Unique International (116) Test Line Test Code (119)
Select All to allow any nature of address setting to be matched.
NOTE Nature of address is only supported by the SS7 protocol, and some variants may not support all available settings.
CLI presentation – This allows the calling party number presentation indication in the incoming call to be matched. The options available are:
All Allowed Restricted Not Available Reserved
Calling party number presentation indication is supported by the SS7 and Q.931 protocols only, and this control will be ignored by all other protocols. Select All to allow any calling party number presentation indication to be matched.
Destination sub-address – The incoming destination subaddress must match this field. Wildcards may be used to allow the route to handle a range of numbers. Originating sub-address – The incoming originating subaddress must match this field. Wildcards may be used to allow the route to handle a range of numbers.
NOTE Subaddresses are supported by SIP, SS7 and Q.931 protocols. Only subaddresses with the Type of subaddress set to User specified and the subaddress information coded using IA5 digits can be pattern matched. Any form of subaddress will be matched against the % wildcard.
NOTE If you wish to match any number to the above fields use the % wildcard. Leaving a field blank will only allow calls that have no associated number to use the route.
Group – The timeslot on which the incoming call arrived must be part of the selected group. The drop down list contains all of the groups defined in the Groups Configuration screen.
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Service indicator code (SIC) – The Service indicator code or Bearer code. This field is used to define what type of calls may use this route. For example, when set to Speech the route will only accept voice calls. The default is to accept all call types. This feature is commonly used to identify calls that need to avoid compressed lines, such as 64k data calls. The options available are:
All Speech 64K Data 3.1Khz Fax
NOTE Some protocols may not support all Service Indicator Codes.
Outgoing Call fields The Outgoing Call fields specify the parameters that will be used when making an outgoing call on the route. Destination address – is the dialled number to which the outgoing call will be directed. This can be either a fixed number or can be built from the incoming call parameters using number translation features below. Destination nature of address – this sets the called party nature of address indicator in the outgoing call. The options available along with their corresponding protocol values are:
Transparent Subscriber (1) Unknown (2) National (3) International (4) Network Specific (5) Network Routing (National) (6) Network Routing (Network Specific) (7) Network Routing (Called Directory) (8) Subscriber, Operator Requested (113) National, Operator Requested (114) International, Operator Requested (115) No Number, Operator Requested (116) No Number, Cut Through (117) 950+ Call (118) Test Line Test Code (119) UK Specific Address (126)
Select Transparent to map the incoming call type through to the outgoing call. If the incoming call type does not support the nature of address indicator then a default setting of International will be used.
NOTE Nature of address is only supported by the SS7 protocol, and some variants may not support all settings.
Originating address – The calling party number which will be sent when making the outgoing call. This can be either a fixed number or can be built from the incoming call parameters using number translation features below.
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Originating nature of address – this sets the calling party nature of address indicator in the outgoing call. The options available along with their corresponding protocol values are:
Transparent Subscriber (1) Unknown (2) National (3) International (4) Non-Unique Subscriber (113) Non-Unique National (115) Non-Unique International (116) Test Line Test Code (119)
Select Transparent to map the incoming call type through to the outgoing call. If the incoming call type does not support the nature of address indicator then a default setting of International will be used.
CLI presentation – sets the calling party number presentation indication in the outgoing call. The options available are:
Transparent Allowed Restricted Not Available Reserved
Calling party number presentation indication is supported by the SS7 and Q.931, and the setting will be ignored by all other protocols. Select Transparent to map the incoming value through to the outgoing call. If the incoming call type does not support calling party number presentation indication then a default setting of Allowed will be used.
Destination sub-address – is the destination subaddress that will be included in the outgoing call. This can be either a fixed number or can be built from the incoming call parameters using number translation features below. Originating sub-address – is the originating subaddress that will be included in the outgoing call. This can be either a fixed number or can be built from the incoming call parameters using number translation features below.
NOTE Subaddresses are supported by SIP, SS7 and Q.931 protocols. Only subaddresses with the Type of subaddress set to User specified and the subaddress information coded using IA5 digits can be translated, and the outgoing subaddresses will use this format. Any form of subaddress can be passed through transparently by using the % wildcard in both the Source and Destination field.
Group – the timeslot on which the outgoing call will be made will be selected from this group. The timeslot to be used will be selected according to the hunt strategy defined for the group. If the group has no timeslot available then an alternative route will be sought. The drop down list contains all of the groups defined in the Groups Configuration screen. The Destination Group field contains the following system entries.
<Wait for Transfer> This is used when mapping ETS 300 call transfer requests to SS7 call redirection.
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<Test Call> When the destination group is set to <Test Call> no outgoing call is made. Instead the user busy tone will be played to the caller until the call is cleared forward (note that some protocols may clear the call automatically after a period of time). If a Tone Generation scheme containing a busy tone has been configured in the General page, Advanced Options, Tone when call fails, then this tone will be used. Otherwise the UK busy tone will be played for to incoming E1 and IP calls, whilst the US busy tone will be used if the incoming call is a T1 call.
Service indicator code (SIC) – The Service indicator code. This sets the call type of the outgoing call. The options available are:
Transparent Speech 64K Data 3.1Khz Fax Custom
Select Transparent to map the incoming call type through to the outgoing call. If the incoming call type is not supported by the outgoing protocol then an appropriate alternative will be used. The Custom option is only available with Q.931 and ISUP outgoing calls and requires the Custom bearer octets field in the Common Q.931/ISUP tab to be configured.
NOTE Some protocols may not support all Service Indicator Codes.
Wildcards When matching an incoming call against the routing entries, the following wildcard characters can be used in the Dest Addr, Orig Addr, DSubAddr and OSubAddr fields to allow one routing entry to handle a range of numbers. ? - Single character
% - Any other sequence of characters string (wildcard)
For example, 81% will match any number beginning with 81, whereas 8??2 will match any 4 digit number beginning with 8 and ending in 2.
Number translation The following characters are used for translating the Dest Addr, Orig Addr, DSubAddr and OSubAddr from the incoming call to the outgoing call. ? Use next character from incoming string x Delete next character from incoming string % Use remainder of incoming string, any other incoming characters are sent
overlap ; Delete remainder of incoming string
E.g. If the incoming called number is 8120 and the outgoing Destination address is 123x% the actual outgoing number will be 123120.
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You can transfer information from the incoming Originating address field to the outgoing destination address field by putting [OA]% in the outgoing Destination address field. You can also combine information from incoming Originating address and Destination address by putting [OA]????[DA]% in the outgoing field. In our example, following the Originating address with four ?’s will take the first 4 digits of the Originating address then add the whole Destination address value.
NOTE Placing % in the outgoing field signifies the end of the translation, for example, you cannot put xx??%52 in the outgoing field because the 52 will be ignored.
NOTE The [DA] and [OA] tokens are not supported by the Destination sub-address and Originating sub-address fields.
Customer specific options Two customer specific options have been created, these are [RN], redirecting number and [OCN], original called number. They may be used in the same way as the [AO] and [DA] options detailed earlier.
CAUTION If you use these values but the incoming call does not include them, the call will fail to route.
Tokens The [EOS], end of send token can be used to append the end of send indicator specified in the Trunk Call Handling page to the outgoing call at the point when the call is made. The [TO n]x, timeout token can be used to append a terminating sequence to the outgoing call at a specified time after the call has been made. n is the time interval specified in one-tenth second increments. x is the terminating sequence, which can be either [EOS], which will cause the end of send indicator for the outgoing trunk to be used, or it can be a literal sequence of digits. For example [TO 50][EOS] will cause the end of send indicator to be sent 5 seconds after the outgoing call is made, whilst [TO 30]*9# would cause *9# to be sent three seconds after making the call.
The [IN_PORT] token can be used in both the outgoing Destination address and outgoing Originating address fields to insert the incoming trunk number into the string as a 2 digit field with leading zeroes, for example 00, 05, 32. The [IN_TS] token can be used in both the outgoing Destination address and outgoing Originating address fields to insert the incoming timeslot number into the string as a 3 digit field with leading zeroes, for example 000, 055, 234. The [CALL_ID] token can be used in both the outgoing Destination address and outgoing Originating address fields to insert the 10 digit call ID into the string. This token is provided to assist with debugging and fault finding.
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NOTE These fields are case sensitive, for example, [eos] will not be recognized but [EOS] will.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.6.2.2 Routes – Q.931/ISUP The Common Q.931/ISUP, Q.931, and ISUP tabs allow outgoing call parameters that are specific to Q.931 and ISUP calls to be configured. Parameters that are common to both protocols appear on the Common Q.931/ISUP tab, whilst those that are specific to an individual protocol appear on the Q.931 or ISUP tab as appropriate. These parameters should be left at their default settings unless you are familiar with the protocol in use. When a parameter is set to Transparent and the incoming protocol supports that parameter then its value will be passed unchanged from the incoming to the outgoing call. If the incoming protocol does not support the parameter then a suitable default value will be applied to the outgoing call.
NOTE Some protocols may not support all of the values available.
Common Q.931/ISUP tab
Figure 5-94 Routing - Common Q.931/ISUP tab
Custom Bearer Octets – This field is used when a greater level of control over bearer capability settings is required than that which can be achieved using the outgoing call service indicator code field on the Setup page. For example when V.110 rate adaptation is required. Up to 15 bytes of information can be entered. The information must be entered as a series of hexadecimal numbers, for example 1F 36 CA 01 05 04 1D, and must be in the form of a Q.931 Bearer Capability information element.
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When the outgoing protocol is Q.931 the contents of this field will be copied unmodified into a Bearer Capability information element. When the outgoing protocol is ISUP the contents of this field will be used to build a Transmission Medium Requirement parameter. The Outgoing call Service indicator code field on the Setup tab must be set to Custom if you wish to use this feature. If the Outgoing Call Service indicator field field is not set to Custom this field will be ignored.
NOTE This parameter is used to set the bearer capabilities only. It cannot be used to modify the speech path data.
NOTE This field can also be used to set the Service Indicator Code (SIC) octets in an outgoing DASS2 or DPNSS call. The first byte must be set to 1 or 2 to indicate how many SIC octets are to be set. The second byte will be written unmodified into SIC octet 1, and the third byte (if specified) into SIC octet 2. Again the Destination service indicator code field on the Setup page must be set to Custom.
Numbering Plan – The Originating field sets the calling party numbering plan identification, and the Destination field sets the called party numbering plan identification. In addition to Transparent the following options are available:
Unknown ISDN (CCITT E.164/E.163) Data (CCITT X.121) Telex (CCITT F.69 National Standard) Private
Screening – This field sets the calling party number screening indicator. In addition to Transparent the following options are available:
Not screened Verified and passed Verified and failed Network Provided
Connected Number Screening – This field sets the connected party number screening indicator sent to the incoming side. In addition to Transparent the following options are available:
Not screened Verified and passed Verified and failed Network Provided
This setting specify how connected party number information should be sent to an incoming Q.931 or SS7 call using this route. Currently only the screening indicator can be modified, and all other information is passed through unmodified or using suitable default values.
NOTE If the outgoing call does not present a connected party number at the connect stage, no connected party number information will be sent to the incoming call.
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NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
Q.931 tab
Figure 5-95 Routing configuration Q.931/ISUP – Q.931 tab option
Numbering Type – The Originating field sets the calling party type of number, and the Destination field sets the called party type of number. In addition to Transparent the following options are available:
Unknown International National Network specific Subscriber Number Abbreviated Number
Multi Level Precedence and Pre-emption (MLPP) – Use the controls in this group to specify how the Service Domain Scheme should be mapped between an incoming and outgoing call. There is further information on MLPP in its own section of the document.
Service Domain Scheme – Use this field to specify how the Service Domain Scheme parameter will be set in the outgoing call leg. When Transparent is selected the setting presented by the incoming call is passed through unchanged. If no Service Domain Scheme is presented by the incoming call a default setting will be applied. When Custom is selected any setting presented by the incoming call will be replaced with the content of the Custom Domain Octets control. Custom Domain Octets – Use this control to specify the five octet Service Domain Scheme parameter to be used in the outgoing call leg, using hexadecimal form. This control will only be displayed when Service Domain Scheme is set to Custom.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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ISUP tab
Figure 5-96 Routing configuration Q.931/ISUP – ISUP tab option
Calling Party Category – This field sets the calling party category. In addition to Transparent the following options are available:
Custom French operator English operator German operator Russian operator Spanish operator Ordinary calling subscriber Calling subscriber with priority Data call Test call Payphone
When Custom is selected a Custom CPC value field is visible, and a decimal value, between 1 and 255, can be selected. This value will be written directly to the Calling Party’s category parameter of the outgoing Initial Address Message. Override with Incoming Value – This field will only be enabled when Calling Party Category is set to a value other than Transparent.
When Calling Party Category is set to Transparent a calling party category presented by the incoming call leg will be used in the outgoing call, otherwise a default setting will be used. When Calling Party Category is set to a value other than Transparent, then the selected value will be used in the outgoing call leg regardless of whether a value has been presented by the incoming call. Checking Override with Incoming Value will modify this behaviour:
If a calling party category is presented by the incoming call then this will be used in the outgoing call. If the incoming call does not present a calling party category then the selected value will be used in the outgoing call.
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NOTE A calling party category will be presented by an incoming SS7 call, or an incoming SIP call that includes an Aculab-SS7-Calling-Party-Category custom header.
Echo Control Indication – The controls in this group are used to set the Echo control device indicator (ECDI) bit in both the incoming and/or outgoing SS7 call legs.
Incoming Call – This control is used to configure how the ECDI bit in the Backward call indicators returned to an incoming SS7 call will be set. In addition to Transparent the following options are available.
IECD not included The ECDI bit will be set to incoming echo control device not included (0).
IECD included The ECDI bit will be set to incoming echo control device included (1).
IECD not included if EC applied
If echo cancellation has been applied to the outgoing call leg the ECDI bit will be set to incoming echo control device not included (0), otherwise incoming echo control device included (1) will be used.
IECD included if EC applied
If echo cancellation has been applied to the outgoing call leg the ECDI bit will be set to incoming echo control device included (1), otherwise incoming echo control device not included (0) will be used.
NOTE The IECD not included if EC applied and IECD included if EC applied options are for use when making outgoing SIP calls only. Other protocols do not support echo cancellation.
Outgoing Call – This control is used to configure how the ECDI bit in the Nature of connection indicators sent in an incoming SS7 call will be set. In addition to Transparent the following options are available.
OECD not included The ECDI bit will be set to outgoing echo control device not included (0).
OECD included The ECDI bit will be set to outgoing echo control device included (1).
Charge Indicator – This field sets the Charge indicator bits in the Backward call indicators parameter sent to an incoming SS7 call leg. In addition to Transparent the following options are available:
No indication No charge Charge
Outgoing Continuity Checks – When this box is checked, a continuity check will be carried out before the outgoing SS7 call is made. Interworking Indicator – This field sets the Interworking indicator bit in the Forward call indicators parameter sent by the outgoing SS7 call leg. In addition to Transparent the following options are available:
No interworking encountered Interworking encountered
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ISDN User Part – This field sets the ISDN user part indicator bit in the Forward call indicators parameter sent by the outgoing SS7 call leg. In addition to Transparent the following options are available:
ISDN user part not used all the way ISDN user part used all the way
ISDN User Part Preference – This field sets the ISDN user part preference indicator bits in the Forward call indicators parameter sent by the outgoing SS7 call leg. In addition to Transparent the following options are available:
ISDN user part preferred ISDN user part not required ISDN user part required
ISDN Access – This field sets the ISDN access indicator bit in the Forward call indicators parameter sent by the outgoing SS7 call leg. In addition to Transparent the following options are available:
Originating access non-ISDN Originating access ISDN
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.6.2.3 Routes – CAS The CAS page contains features intended for use when interworking with or between CAS protocols. Although the features are designed for use with CAS protocols, many are implemented in a protocol independent fashion and can be applied even when both call legs are ISDN.
Figure 5-97 Routing configuration CAS tab option
Bidirectional voice path after Ringing – When interworking with CAS protocols, by default only the backwards voice path will be opened when the call reaches the ringing state. Selecting this option will open a bi-directional voice path at this stage. Selecting this option will have no effect if tone generation has been configured to provide a ring tone on this route, or the Voice Path Timer field below has been configured to open up a backwards voice path.
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NOTE This feature is not supported if the incoming call is using the SS5 protocol.
Map Progress to Alerting – Some ISDN protocols use call progress indication to open up voice paths and allow in-band tones to be passed through. This option provides a way to achieve this when interworking between an outgoing protocol that supports call progress indication, and an incoming protocol which does not. When selected, any call progress messages received on the outgoing side which indicate in-band tones may be available are translated into ringing on the incoming side. If both the incoming and outgoing protocols support call progress indication, then call progress will be mapped through.
NOTE Care should be taken when using this option to ensure that in does not cause a calling side switch to generate a local ringing tone in response to a busy or unobtainable announcement.
Set charge flag on R2 – By default an accepted call will be charged. Check the Set Charge Flag on R2 option should you wish to indicate that the incoming call should not be charged, for example, when calling a Freephone number. This option is only supported by R2 protocols. CAMA to SIP call transfer mapping – When enabled a hookflash call transfer request from the outgoing CAMA call leg will be mapped to the incoming SIP call leg using RFC 2833 messaging. There is further information on this in the Call Transfer section of the document. This control applies to CAMA ports only. Enabling this feature for other CAS protocols, or when the incoming call leg is not a SIP call will have no effect. Voice Path Timer – Voice path switching in GIII is based on the protocols used on both ports. This is normally sufficient and copes with most situations encountered, however there may be a time where the default switching needs to overridden. With some CAS protocols, it is not possible to know the state of the outbound call, so it may be necessary to switch the voice path and/or send a message to the incoming call to indicate a state change. Delay after last digit – Using this field to send a switch command and/or call control message a specified time after the last digit was received. Entering 0 will disable the feature. Message to send – Selecting this option will send a call control message to the incoming call when the timer expires:
None – Disabled. Alerting – will send the appropriate call control message for the incoming protocol to indicate that the far end is ringing.
Progress – will send an ISDN call progress message to the incoming call indicating that in-band tones are available.
NOTE Only selected ISDN protocols support call progress messages.
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Voice path switching – Selecting this option will connect the voice paths when the timer expires:
None – Disabled. Backward – will switch only the backward voice path between the incoming and outgoing calls. This will allow in-band announcements to reach the caller, whilst still allowing the Aculab call control software to pass dialled digits between the calls.
Bidirectional – will switch both the forward and backward voice paths between the incoming and outgoing calls.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.6.2.4 Routes – DPNSS
Figure 5-98 Routing configuration DPNSS tab option
Calling Line Category – is used to indicate where the call originated. It is often used to differentiate between calls on a private network and calls that originated in the public network in order to change the ring cadence used by the PBX. The options available are:
Transparent Ordinary DASS2 PSTN Decadic SSMF5 Operator Network
Unless you are familiar with DPNSS protocols, you should leave this at the default setting of Transparent.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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5.6.2.5 Routes Incoming and Outgoing SIP calls The Incoming and Outgoing SIP call tabs allow the media and codec settings for SIP calls using this route to be configured. The codecs configuration will only be displayed when Use default codecs is not enabled.
Figure 5-99 Routing - Incoming SIP calls tab
Figure 5-100 Routing - Outgoing SIP calls tab
Echo cancellation – The controls in this group are used to configure how echo cancellation will be applied.
Type – Use this control to specify whether echo cancellation should be applied. The options available are:
Off – disables the echo canceller. On – enables the echo canceller. On with NLP (Muting) – enables the echo canceller with non-linear processing to suppress the signal whenever echo or background noise are dominant. This mode of operation is suitable for large-scale conferencing and/or IVR applications, where it is important that the signal is completely muted whenever the caller is not speaking.
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On with NLP (CNG) – enables the echo canceller with non-linear processing to replace the signal with comfort noise whenever echo might otherwise be audible. This mode of operation is most appropriate to IP gateway applications where background noise is desirable to maintain the illusion of continuity.
NOTE If the Clear Channel codec is selected during codec negotiation the setting in this field will be ignored and echo cancellation will not be applied.
Figure 5-101 Incoming SIP calls configuration for a route
Apply echo cancellation – This control is only displayed when Type is set to On, On with NLP (Muting) or On with NLP (CNG), and is used to specify the conditions under which echo cancellation will be applied. The options available are:
Always Echo cancellation will always be applied.
The following settings should be used only when interworking between SIP and SS7. When interworking with non-SS7 protocols using these settings will result in echo cancellation not being applied.
Backward ECDI = 0 (incoming page only)
Echo cancellation will only be applied when the echo control device indicator bit in the Backward call indicators returned by the outgoing SS7 call is set to incoming echo control device not included (0).
Backward ECDI = 1 (incoming page only)
Echo cancellation will only be applied when the echo control device indicator bit in the Backward call indicators returned by the outgoing SS7 call is set to incoming echo control device included (1).
Forward ECDI = 0 (outgoing page only)
Echo cancellation will only be applied when the echo control device indicator bit in the Nature of connection indicators presented by the incoming SS7 call is set to incoming echo control device not included (0).
Forward ECDI = 1 (outgoing page only)
Echo cancellation will only be applied when the echo control device indicator bit in the Nature of connection indicators presented by the incoming SS7 call is set to incoming echo control device included (1).
Echo cancellation span – This control is only available when Type is set to On, On with NLP (Muting) or On with NLP (CNG), and is used to select the length of time for which the echo canceller will monitor the return voice path looking for the forward signal.
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NOTE The echo span may be set to 40ms, 70ms or 104ms. The 70ms and 104ms settings should not be used unnecessarily, as the additional processor requirement may cause echo to be missed during periods of heavy call load.
Differentiated service format – Sets the format in which to specify the Type of Service setting. The available types are:
Code Point – allows the setting to be selected from a list of commonly used named constants. Raw (hex) –the setting is entered as a hexadecimal value, allowing settings for which no DSCP constant is provided to be used.
DiffServe value for RTP – Sets the type of service field within the IP header of real time protocol (RTP) packets transmitted.
The controls available are dependent upon the setting in the Differentiated service format group, as described above.
Code Point – the drop down list contains a list of named constants whose values are based upon the information provided in IETF RFC 4595. The following constants are available, the numbers alongside show their hexadecimal equivalents: AF11 (0A) AF12 (0C) AF13 (0E) AF21 (12) AF22 (14) AF23 (16) AF31 (1A) AF32 (1C) AF33 (1E) AF41 (22) AF42 (24) AF43 (26) CS0 (00) CS1 (08) CS2 (10) CS3 (18) CS4 (20) CS5 (28) CS6 (30) EF (2E)
Raw (hex) – allows the type of service setting to be specified as an 8-bit value in hexadecimal form.
DTMF digit detection – This control specifies whether DTMF digits should be removed from the audio stream and sent out-of-band using RFC 2833 frames. The following settings are available.
Disabled – DTMF digits will not be removed from the audio stream, and no out-of-band frames will be sent. Immediate Detection – An out-of-band tone will be generated as soon as the appropriate tone has been detected. 40ms Detection – An out-of-band tone will not be generated until the appropriate tone has been detected continuously for at least 40ms.
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In most applications Immediate Detection will be the appropriate setting. The 40ms Detection setting is designed for use in environments where high levels of background noise could cause spurious digits to be detected. Report loss of received RTP after – Setting this control to a value other than Disabled will cause a groomerReceivedRTPStoppedTrap SNMP alarm to be generated if no RTP packets are received from the remote endpoint for the specified period. The groomerReceivedRTPStoppedTrap SNMP alarm is detailed in the SNMP facility Appendix.
NOTE The use of DTX by the remote endpoint may cause silence on the line to be interpreted as loss of RTP.
NOTE SNMP reporting must be enabled for SNMP traps to be produced.
Use 183 responses – This option is available for incoming calls only. Enabling this control will cause a 183 message to be sent instead of a 180 message at the ringing stage. This control is provided to overcome interoperability issues. Unless its use is essential, it is recommended that it be left at the default value. Send reliable 18x responses – This option is available for incoming calls only. Enabling this control causes 180 series responses to be sent reliably instead of unreliably. Unexpected behaviour may be experienced if this option is enabled when the caller does not support the protocol extension required for reliable provisional responses. Request early media – This option is available for incoming calls only. This option is primarily intended for use when interworking with protocols such as CAS, which do not themselves support call progress. Progress information from the outgoing leg of a call will automatically be used to request an early media path from an incoming SIP call if appropriate. Enabling this option will cause an early media request to be made at the ringing stage regardless of what progress information may have been received from the outgoing call. Transport – This option is available for outgoing calls only. Sets the IP transport protocol used by SIP to send call control messaging. The default is UDP. Permitted values are:
UDP – User Datagram Protocol. TCP – Transmission Control Protocol. TLS – This option is available for outgoing calls only. When this control is checked all SIP calls made using this route will be secure. More information can be found in the SIP Telephony section of the document.
NOTE This control will only be available when the restricted availability variant of GIII software is in use.
SIP to CAMA call transfer mapping – This option is available for outgoing calls only. When enabled, an RFC 2833 hookflash call transfer request from the outgoing SIP call leg will be mapped to the incoming CAMA call leg. More information can be found in the Call Transfer section of the document.
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This control applies when interworking from CAMA to SIP only. Enabling this feature when the incoming call leg is not a CAMA call will have no effect.
Figure 5-102 Routing Incoming SIP calls tab with codecs
Figure 5-103 Routing - Outgoing SIP calls tab with codecs
Use default codecs – Check this box if you want the route to offer the system default codec list defined in the Trunks – SIP – Default Codecs tab. Other codec configuration controls will be hidden when this option is selected.
NOTE When Use default codecs is selected the codecs offered by this route will change whenever the default codec list is modified.
Uncheck the box if you want the route to offer a different set of codecs to the default list, or if you do not want the set of codecs offered by the route to change when the default list is modified.
Codecs configuration The codecs to be used can be dragged from the codecs not selected for use box into the Currently selected codec order box. Once the codec is dropped into the currently selected box, any available configuration controls will be presented on the codec item.
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Figure 5-104 Routing - SIP Codec configuration
To add a codec, drag the required codec from the codecs not selected for use Codecs list to the Currently selected codecs order list. To remove a codec, drag the required codec from the Currently selected codecs order list into the codecs not selected for use Codecs list. To change a codec priority, drag the codec in the Currently selected codecs order list and move it to the preferred position. Priority order is top down.
NOTE The G.726 and Clear Channel codecs are not available when making or receiving secure SIP calls.
The VAD (voice activity detection) settings applied to the transmitted RTP stream for the selected codec may be configured using the VAD control on the codec.
Figure 5-105 Codec VAD settings
VAD – The options available depend upon the codec. When a G.711 or G.726 codec is selected the options available are:
Disabled – No silence suppression will be applied. The received audio stream will be packetised and transmitted across RTP Muting (DTX) – Discontinuous transmission. The received audio stream will be analysed, and no packets will be sent across RTP when silence is detected. Comfort Noise – This will offer a comfort noise codec (RFC 3551, payload 13) to the remote endpoint. If this is accepted any silence detected will be replaced by comfort noise packets in the RTP stream. If the remote endpoint does not accept the comfort noise codec then DTX will be used.
When using a G.711 or G.726 codec, if the remote endpoint offers a comfort noise codec this will be accepted by GIII regardless of the VAD settings, and any comfort noise packets received decoded into the audio stream.
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When a G.723.1 codec is selected the options are for silence compression and the options available are:
Enabled – When selected Annex A silence compression will be applied to the transmitted RTP stream. Disabled – When selected no Annex A frames will be transmitted in the RTP stream.
When using a G.723.1 codec, any Annex A frames received from the remote endpoint will be accepted and decoded regardless of the setting in this control. When a G.729A codec is selected the options are for Annex B, and the options available are:
Enabled – When selected the G.729AB codec will be offered to the remote endpoint. If this is accepted the voice activity detector may transmit both 'A' and 'B' frames on the RTP stream, otherwise only 'A' frames will be transmitted. Disabled – When selected the voice activity detector will generate only 'A' frames on the RTP stream.
When using a G.729A codec, any 'B' frames received from the remote endpoint will be accepted and decoded regardless of the setting in this control.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.6.2.6 Routes – General tab
Figure 5-106 Routing configuration general tab option
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Cause map – Used to select between default mapping of clear causes, and a user defined mapping list. The user defined options for Cause Map are created using the Cause Mapping option covered in the Cause mapping section of the document. Tone Generation – Used to select which call progress tone set will be played to the incoming leg of calls using this route. Select None if no call progress tones are to be generated. Incoming/Outgoing Immediate Release – Under normal conditions, GIII does not complete the release of the call until both sides have cleared. This option is used when you have a requirement to clear incoming or outgoing calls as soon a clear request is received. When these fields are checked, GIII will not wait for the other part of the call to clear first. PCM Conversion – The controls in this group configure how PCM conversion is applied when routing calls between ports.
Automatic – The default mode of operation is for GIII to detect the type of ports in use (E1/T1/IP) and apply any PCM conversion required automatically. In some instances, for example when E1 links are being used to carry µ-law speech, it may be necessary to disable such conversion. Checking this box will cause PCM conversion to be carried out automatically. Unchecking the box allows the voice path to pass between incoming and outgoing call legs unmodified.
Figure 5-107 PCM conversion selection
None/RTP translation – When selected the RTP translation controls will be displayed, these controls are hidden when Automatic is selected. Use the A-Law and µ-Law radio buttons to select the translation to be applied when passing speech between a TDM trunk and the RTP stream on an IP telephony trunk. Incoming RTP will be translated into the selected encoding, whilst outgoing RTP will be translated using the appropriate algorithm for the encoding selected.
NOTE It is not possible to apply different translations to the incoming and outgoing RTP streams.
Call Re-routing – The controls in this group allow GIII to search for an alternative route should an outgoing call using the selected route be rejected. If set to Disabled then the Re-routing causes controls will not be displayed.
NOTE Call re-routing is not supported in any of the following circumstances: • The incoming call leg is an ETS 300 call requesting explicit UUS activation. • The incoming call leg is using the SS5 protocol. • The outgoing call leg is using any CAS protocol.
Figure 5-108 Routes - re-route causes
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Re-route on match – If selected then the Re-routing causes box will be displayed. This setting will allow calls using this route to search for an alternative route should they fail to complete with a cause listed in the Re-routing causes box. The causes can be added by selecting the New cause button. A cause can be deleted by selecting the Del button associated with the cause. The x wildcard may be used in trailing positions to allow a range of causes to be matched against a single entry. Re-route unless match – selected then the Re-routing causes box will be displayed. It may not be appropriate to re-route calls under all circumstances (for example when the call returns user busy). This setting will prevent calls from being re-routed when they fail with a cause in the list. The causes can be added by selecting the New cause button. A cause can be deleted by selecting the Del button associated with the cause. The x wildcard may be used in trailing positions to allow a range of causes to be matched against a single entry.
Re-route on SS7 Continuity Check Failure – If this box is checked, SS7 calls using this route that fail an outgoing continuity check will be re-routed. Override Global maximum ring period timer – The control in this group allow the system level Maximum Ring Timer to be overridden with a route specific setting.
CAUTION If a Maximum Ring Period is used, the timer should be set to a high value, at least 180 seconds (3 minutes) is suggested. Lower values may cause the call to drop prematurely.
NOTE The timeout release will not work on most CAS protocols as they do not allow the call to be cleared by the B party.
Maximum ringing period – The time the outgoing call will be allowed to remain in the ringing state before being cleared. Setting this field to zero will disable the ringing timer. Cause – Use the controls in this group to specify the cause to be used when clearing the incoming call leg on timer expiry.
Default – When selected the generic clearing cause of NO ANSWER will be used. Generic – This option allows the clearing cause to be selected from a list of generic causes. This allows an appropriate protocol specific cause to be automatically selected when clearing the incoming call leg. For example, selecting NUMBER BUSY will use the correct busy release cause for the incoming protocol. Raw – This option allows a protocol specific release code to be used. For example, 17 would be entered to clear an incoming EuroISDN call with User Busy.
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Cause location – The Cause location field applies to SS7 and Q.931 calls only, and is ignored by all other protocols. This control allows the location field in the Q.850 cause parameter to be set when the timer clears a call. A location of User will be applied when Default is selected.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.6.3 Cause Maps configuration
Figure 5-109 Cause Map list
By default GIII automatically maps the call clearing cause presented by the outgoing protocol to the equivalent one for the incoming protocol. Cause mapping allows an alternative mapping to be defined. This is generally not required unless a device such as a PBX acts on a special release code. The Cause Maps tab is found under Role Configuration – Routing.
NOTE All Cause Map Sets defined here will be added to the Cause Map pull down options list of the Routes - General tab.
Each cause map contains one or more individual mapping translations. Any codes that are not defined in the mapping are passed through using standard mapping. To add a new cause map, drag the Drag to add a cause map button into the Cause Map List box, and drop. The new cause map will be added in the position dropped. The cause map will be assigned a default name, which can be changed in the Name field. To update an existing cause map, select the Edit button Cause Map List box for the cause map, and make the required changes. To remove a cause map drag it from the list to the Trash Can.
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Figure 5-110 Cause mapping configuration
Map entries – Individual mappings are added to the selected cause map using the controls in this group:
Add entry – Use this button to add a new individual mapping. The new entry will be assigned a default name and added to the bottom of the Map entries list. Delete – Use this button to remove the selected entry from the Map entries list. Mapping Name – Use this field to modify the default name assigned to the mapping. Type – Translations are in the order of received to transmitted and can be either:
Generic to Generic – the generic cause received from the outgoing call leg will be mapped and passed to the incoming leg using a generic cause. Generic to Raw – the generic cause received from the outgoing call leg will be mapped and passed to the incoming leg using a raw cause. Raw to Generic – the raw cause received from the outgoing call leg will be mapped and passed to the incoming leg using a generic cause. Raw to Raw – the raw cause received from the outgoing call leg will be mapped and passed to the incoming leg using a raw cause.
Incoming cause – This is the clearing cause presented by the outgoing call leg. Outgoing cause – This is the clearing cause passed to the incoming call leg. Location – This field is only used when the incoming call leg is using the SS7 protocol, and will be used to set the location field in the release message sent to the incoming call.
Once you are satisfied with the mapping of all the cause translations for your mapping set, either repeat the process to create new mapping sets, select Cancel to ignore any updates or Submit to accept any updates and return to the cause mapping page.
Cause values The Incoming cause and Outgoing cause controls will change to reflect the mapping type selected. When mapping to or from a Raw cause, the decimal value for the release code defined by the protocol being used should be entered into the field. For example, if the protocol is EuroISDN then 17 might be entered to represent User Busy. When mapping to or from a Generic cause, a drop down list of generic causes will be presented. This can be used to select a cause without needing to know the exact value for that protocol. For example, selecting NUMBER BUSY will set the correct busy release code for the protocol in use.
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NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.7 SIP TLS certificates The TLS Certificates page is located under the Role Configuration menu, and is where the certificate files that will be used when making secure SIP calls are specified.
Figure 5-111 SIP TLS Certificates page
Remove Certificates – Remove all certificates from this node Private Key and Site Certificate – This file, which is often referred to as the ‘Server Certificate’, contains the private key and site certificate that GIII will present to remote endpoints during the authentication process.
Private key and site certificate to upload – Click Choose File to select the certificate file to upload, and click open. Private Key Password – If the private key is password protected, enter the password here. If the private key is not password protected then this field must be left blank.
Trusted Certificates – This file contains the identities of those Certificate Authorities that GIII will accept when conducting endpoint authentication.
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Trusted Certificate file to upload – Click Choose File to select the certificate file to upload, and click open.
NOTE Both the private key and site certificate must be contained in the same file. GIII does not support the use of separate files for each.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.8 Call Trace The Call Trace page is located in the Role Configuration menu. It configures which trunks generate what call trace data, and where this data is sent.
Figure 5-112 Default Call Trace settings
NOTE The appearance of the page will change according to the number of trunks in a role. Also if the role is active on a node then an immediate icon will be displayed to show that any changes made are immediately applied to the active role when submit is pressed.
5.9 Trunk Blocking Trunk blocking allows individual timeslots on the trunk to be removed from service so that maintenance operations can be performed. The current blocking status of all trunks is re-applied on role startup. If a blocked trunk is reloaded, its blocking state will be restored following completion of the reload. GIII will not route outgoing calls to blocked timeslots. For SS7, T1 Robbed Bit, AT&T, NI2 and DMS100 this applies equally to both timeslots blocked locally, and timeslots blocked at the far end.
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When the GIII application is closed down normally, hardware blocking will be applied to any ITU-T, China, and UK ISUP trunks that are not already blocked. Any shutdown applied blocking will be removed when GIII is restarted. To apply or remove trunk blocking, select Trunk Blocking from the Role Configuration menu.
Figure 5-113 Trunk Blocking
NOTE The appearance of the page will change according to the type of trunks configured for the role, and the configuration of those trunks. Also if the role is active on a node then an immediate icon will be displayed to show that any changes made are immediately applied to the active role when submit is pressed.
The trunk blocking page has a column of selectable buttons on the left side, each of which is a trunk for the role being configured. The button’s state can be toggled by selecting it, and multiple buttons can be selected. The purpose of this is to allow set or clear the blocked state of timeslots on multiple trunks at the same time. At the bottom of the page there are some further buttons, and their function is described below.
Set – This will check all the timeslots for the selected trunks. Clear – This will clear the timeslots for the selected trunks. Select All Trunks – This button will select or de-select all the trunk buttons.
There are two columns on the right-hand side, Block Trunk, and Clear Calls. A checkbox will only be shown in a column if it is an allowed operation for that particular trunk.
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5.9.1 Trunk blocking for ITU-T, China and UK ISUP
Figure 5-114 Trunk blocking for ITU-T, China, and UK ISUP
To block all timeslots on trunk, select the trunk, so it is highlighted in the Click to select Trunks column, and click set. This will select all timeslots for blocking. To apply the hardware blocking to all calls, click the submit button. Clear calls – When selected, all calls in progress on the trunk will be cleared down, and no new calls will be allowed. When un-checked, no new calls will be allowed, and blocking will only occur when all calls have gracefully cleared on the trunk.
CAUTION Applying a hardware block for SS7 causes all calls on the trunk to be immediately cleared down, and prevents any further calls from being made on the trunk until the hardware block has been removed.
Timeslots to Block – This group of checkboxes allow maintenance blocking to be applied to individual bearer timeslots, and only those checkboxes that correspond to bearer timeslots will be displayed. When checked, maintenance blocking will be applied to the timeslot. Any call currently in progress on the timeslot will be allowed to complete, but no new calls will be accepted. Un-checking a box will remove the maintenance blocking and allow calls to be made on the timeslot once again. The Set and Clear buttons will set or clear all timeslot checkboxes for the selected trunks. As SS7 blocking operations need to communicate with the far end of a link, they are posted for asynchronous execution. When an SS7 blocking operation fails because the far end is unreachable, it is re-posted and executed again. To determine when an SS7 blocking operation has taken effect, the layer 2 state of the trunk can be monitored using the Data Link filter. The layer 2 reports contain a (hexadecimal) bitmap of the current layer 2 state of the trunk. When a timeslot has been successfully blocked, its layer 2 state will be reported as 0.
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5.9.2 Trunk blocking for ANSI SS7 and non-SS7 TDM trunks
Figure 5-115 Trunk blocking page
Block Timeslot – Only the checkboxes representing the bearer timeslots on the trunk will be enabled. Check a box to place the corresponding timeslot out of service, and uncheck the box to place it back in service. Once out of service, GIII will not allow either incoming or outgoing calls to be made on the timeslot. When the protocol on the trunk is T1RB (Robbed Bit), ‘Back Busy’ will be used to take timeslots in and out of service. This signals the availability of the timeslots to the remote end. When the protocol is one of AT&T, NI2 or DMS100, SERVICE messages will be used to block or unblock the timeslots. These messages signal the state of the timeslot to the remote end. When the protocol on the trunk is ANSI SS7, CIC blocking will be used to block and unblock the timeslots. Clear Calls – When checked, if a timeslot on this trunk that is to be placed out of service is currently carrying a call, that call will be cleared down. When unchecked, any calls currently in progress on blocked timeslots will be allowed to complete, but no new calls will be accepted.
5.9.3 Trunk blocking for IP telephony trunks Block Trunk – IP telephony has no concept of timeslots, and so the entire trunk must be placed out of service. Check the box to place the trunk out of service, and uncheck the box to place it back in service. Once out of service, GIII will not allow either incoming or outgoing calls to be made on the trunk. Clear Calls – When checked, if the trunk is to be placed out of service and is currently carrying calls, those calls will be cleared down. When unchecked, any calls currently in progress on the trunk will be allowed to complete, but no new calls will be accepted. Blocking operations are reported to the GIII Status Monitor using the Data Link events filter.
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5.10 SS7 signalling links The SS7 Signalling Links page is used to activate and deactivate SS7 signalling links. Select the Role Configuration menu and the SS7 Signalling Links page.
Figure 5-116 SS7 Signalling Links
The page shows each SS7 signalling link that has been configured on a trunk. The links may be activated or deactivated by selecting the appropriate radio button.
NOTE The settings will not be applied until the Submit button is selected.
NOTE The settings are applied immediately and are stored against the complete role, i.e. not against an individual revision.
The GIII will maintain signalling link activation/deactivation settings across system shutdown in the following way:
• When the active role is removed from a GIII, or a GIII is restarted, all deactivated signalling links are activated as part of the shutdown procedure (all SS7 bearer channels are hardware blocked at this point to prevent call delivery).
• When the GIII is restarted, or a role is active on a GIII, any signalling links that were deactivated prior to running the shutdown procedure will be deactivated again as part of the startup procedure (the configured blocking status of all bearer channels is re-applied at this point to allow call delivery).
TODO: A warning will be given if the last signalling link to a destination is about to be deactivated, and you will be required to confirm the operation.
5.11 SNMP reporting Select SNMP from the Role Configuration menu, and select the role for which you wish to view or change the SNMP reporting information. This page specifies which system faults will be reported using SNMP alarms.
NOTE The SNMP agent information must be configured and SNMP enabled before SNMP reporting would actually be able to send the alarms. To configure the SNMP agent on a node go to Cluster Administration – Configure SNMP Agent
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The SNMP appendix details SNMP traps and tables.
Figure 5-117 SNMP Reporting
Enable SNMP Reporting – Check this option to enable the SNMP reporting functions. Trunks – use this tab to configure the trunk specific traps:
groomerPortIPStatusTrap groomerTDMRxL1AlarmTrap groomerTDMTxL1AlarmTrap groomerTDMLayer2Trap groomerTDMRxCASMultiframeAlarmTrap groomerTDMClockSlipsTrap groomerCallRoutingFailedTrap
System – use this tab to configure the operating system performance traps: groomerCPUUsageTrap groomerMemoryUsageTrap groomerAvailableDiskSpaceTrap
Chassis – use this tab to configure the chassis hardware monitoring traps: groomerPSUStateTrap groomerDiskStateTrap
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Call Monitoring – use this tab to configure the call life cycle monitoring trap: groomerCallState2Trap
NOTE You must press the Submit button to save the changes for a particular tab. Pressing another tab will lose the changes made.
NOTE When you submit changes for a role, and the role is active on a node, these changes will not be used until you create a role revision with these changes and make that revision active.
5.11.1 Ports tab options
Figure 5-118 SNMP Reporting – Ports tab
This dialog is used to configure trunk specific traps. The dialog options available are: Ports – A list of all the IP telephony and E1/T1 trunks in the system. To disable an SNMP trap for a given trunk, select the required trunk from this list then check the appropriate options as detailed below. Disable Layer 1 Received Alarms – Suppress the groomerTDMRxL1AlarmTrap (TDM trunks) or groomerPortIPStatusTrap (IP telephony trunks) for the selected trunk.
Disable Layer 1 Transmitted Alarms – Suppress the groomerTDMTxL1AlarmTrap for the selected trunk. This option will be disabled if an IP telephony trunk is selected. Disable Layer 2 Alarms – Suppress the groomerTDMLayer2Trap for the selected trunk. This option will be disabled if an IP telephony or CAS trunk is selected. Disable CAS Multiframe Alarm – Suppress the groomerTDMRxCASMultiframeAlarmTrap for the selected trunk. This option will be disabled if a non-CAS trunk is selected. Clock Slips – Use the controls in this group to configure how clock slip alarms will be reported. This control group will be disabled if an IP telephony trunk is selected.
Disable Alarms – Suppress the groomerTDMClockSlipsTrap for the selected trunk.
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Alarm at x slips/second – The number of slips per second at, or above which a slip alarm is generated. Clear at x slips/second – The number of slips per second at, or below which a slip alarm is cleared.
Disable Routing Failed Alarms – Prevents the groomerCallRoutingFailedTrap from being generated when GIII fails to route an incoming call arriving on this trunk.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.11.2 System tab options
Figure 5-119 SNMP reporting – System tab
This dialog is used to configure system performance alerts relating to CPU usage, memory usage, and disk space availability. CPU – The controls in the CPU group are used to manage CPU load alarms:
Raise Alarm at x % Load – A groomerCPUUsageTrap alarm is generated whenever CPU load reaches or exceeds the value entered in this field. Clear Alarm at x % Load – A groomerCPUUsageTrap alarm is cleared whenever CPU usage falls to or below the value entered in this field.
NOTE CPU load is averaged across several readings to prevent transient spikes from unnecessarily raising or clearing alarms.
Memory Usage – The controls in the Memory Usage group are used to manage memory usage alarms:
Raise Alarm at x Mb – A groomerMemoryUsageTrap alarm is generated whenever memory usage reaches, or exceeds, the value entered in this field.
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Clear Alarm at x Mb – A groomerMemoryUsageTrap alarm is cleared whenever memory usage falls to or below the value entered in this field.
Disk Space – The controls in the Disk Space group are used to manage disk space usage alarms:
Raise Alarm at x Mb Free – A groomerAvailableDiskSpaceTrap alarm is generated whenever free disk space falls to or below the value entered in this field. Clear Alarm at x Mb Free – A groomerAvailableDiskSpaceTrap alarm is cleared whenever free disk space rises to or above the value entered in this field.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
5.11.3 Chassis tab options
Figure 5-120 SNMP reporting – Chassis tab
The controls on this page allow the production of individual chassis monitoring alarms to be disabled. These controls are provided to allow suppression of spurious alarms generated by known faulty equipment. It is recommended that at all other times the alarms remain enabled. GIII chassis can monitor a variety of system components, although monitoring capabilities vary between individual chassis models. Only those controls applicable to your chassis will be enabled. Where a chassis monitors several components of the same type (for example system fans), you should refer to the installation guide for your chassis to identify specific components.
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NOTE Only those controls that are applicable to the chassis on which the GIII is running will be enabled.
Disks – The controls in the Disks group are used to suppress the system disk alarms. Disable RAID Alarms – Use this control to disable the groomerDiskStateTrap alarm for RAID. Disable HDD1 Alarm – Use this control to disable the groomerDiskStateTrap alarm for HDD1. Disable HDD2 Alarm – Use this control to disable the groomerDiskStateTrap alarm for HDD2.
PSU – The controls in the PSU group represent each of the power supplies fitted in the chassis.
Disable PSU # Alarms – Check the appropriate Disable PSU # Alarms option to suppress groomerPSUStatusTrap alarms relating to that PSU.
Fans – The controls in the Fan group represent each of the system fans fitted in the chassis.
Disable Fan # Alarms – Check the appropriate Disable Fan # Alarms option to suppress groomerSystemFanTrap alarms relating to that fan.
Temperatures – The controls in the Temperature group represent each of the internal temperature sensors fitted in the chassis.
Disable T# Alarms – Check the appropriate Disable T# Alarms option to suppress groomerChassisTemperatureTrap alarms relating to that temperature sensor.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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5.11.4 Call Monitoring tab options
Figure 5-121 SNMP Reporting – Call Monitoring
Call Monitoring provides a mechanism to monitor the life cycle of individual calls and report any faults encountered through SNMP. Faults are reported whenever a call has remained in a state for longer than the period configured by the user by raising a groomerCallState2Trap alarm. Any alarms generated will be automatically cleared when the call moves to another state. Ports – A list of all the IP telephony and E1/T1 ports in the system. To configure the Call Monitoring alarm for a given port, select the required trunk from this list then modify the appropriate options as detailed below. Openin Timeout – This is a GIII internal process, and represents the time allowed between both legs of a call in progress completing the release procedure, and the timeslot carrying the incoming leg being ready to receive a new incoming call. A timeslot is allowed to remain in the ‘listen for incoming’ state indefinitely. This alarm, when generated, indicates that GIII is unable to accept incoming calls on the particular timeslot. Setup Timeout – The number of seconds allowed after detecting an incoming call before GIII reaches the ringing or connected states. If the ringing or connected state has not been reached within this time an alarm will be generated. Ringing – Use the controls in this group to manage how alarms will be generated during the ringing stage of the call:
Enable Ringing – Checking this control will cause an alarm to be generated if GIII remains at the ringing stage beyond the specified period. When unchecked the call will be allowed to ring for an indefinite period. Ringing Timeout – The number of minutes an incoming call is allowed to ring before an alarm is generated. This control will only be enabled when Enable Ringing is checked.
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NOTE Some CAS protocols do not support a ringing stage. In such cases this timer should be set to a sufficient length to allow the outgoing call to be answered.
Connect – Use the controls in this group to manage how alarms will be generated during the connected stage of a call:
Enable Connected – Checking this control will cause an alarm to be generated if the call remains in the connected state beyond the specified period. When unchecked the call will be allowed to remain connected for an indefinite period. Connect Timeout – The number of minutes a call stays connected before an alarm is generated. This control will only be enabled when Enable Connected is checked.
Release Timeout – The number of minutes allowed between receiving a call disconnect and both call legs having completed the release procedure. If the release procedure has not been completed within the specified time then an alarm will be generated.
NOTE The settings are saved to the current role configuration by pressing the submit button. If the role is assigned to a node then the configuration will not be changed on the node until a new role revision is created, and that revision is selected.
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6 GIII Maintenance
6.1 Node Maintenance Mode If a role is changed on a node, or a new role applied to a node then it will initially be in maintenance mode, which means that all ports and trunks are blocked. To change the maintenance mode, navigate to the Node Maintenance Mode page from the Maintenance menu. The System Summary page also shows the Node maintenance mode state, and a link which can be clicked to navigate to the Node Maintenance Mode page.
Figure 6-1 System Status showing node/role maintenance mode
Figure 6-2 Node Maintenance Mode page
If you navigate back to the System Summary page, you will see the node is now active, and able to take calls.
Figure 6-3 Node in an active state
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6.2 Trunk Download/Reset
Figure 6-4 Trunk Download/Reset page
This page can be accessed from the Maintenance menu. It will allow a trunk to have its firmware reloaded.
6.3 SS7 Circuit Reset The SS7 Circuit Reset page is used to reset individual bearer channels on a trunk loaded with SS7 firmware. Select the Maintenance menu, and the SS7 Circuit Reset page.
Figure 6-5 Circuit Reset Page
Select the tab for the trunk which contains the bearer channels that require a reset. Select the timeslots of the bearers you wish to reset, and click Reset Circuits.
6.4 SS7 Continuity checking Go to Maintenance – SS7 Continuity Checking, to get to the SS7 Continuity Checking page. This page is used conduct manual continuity checks on SS7 bearer circuits. Select SS7 Continuity Checking from the Status Monitors menu.
Figure 6-6 SS7 Continuity Checking
To perform the continuity checks, select the timeslots required, and select the Start button.
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Figure 6-7 Selecting timeslots to check
Figure 6-8 Status of continuity checks
The State row reports the current state of the circuit which maybe one of the following: Idle (I) – The circuit is idle. Recheck (R) – The circuit is performing periodic continuity rechecks. Blocked (B) – The circuit is blocked. Active (A) – A call is present on the circuit. Testing (T) – A manual continuity check is in progress. Stopping (S) – A manual continuity check is stopping.
The Result row reports the last manual check result of the circuit which maybe empty if no check has been carried out, or one of the following:
Pass (P) – The check passed. Fail (F) – The check failed. Remote release (R) – The remote end terminated the check with the clearing cause n. No Resources (N) – GIII has insufficient resources to run the check. Circuit Blocked (B) – Circuit was blocked whilst the check was in progress. Circuit Reset (r) – Circuit was reset whilst the check was in progress. Circuit Active (A) – An incoming call arrived whilst the check was in progress. Stopped (S) – The operator has manually stopped the check, or GIII stopped the check for an unspecified reason. Not tested (-) – There has been no continuity test performed Unknown result (?) – Unknown result
To carry out manual continuity checks, select the tab associated with the trunk on which to carry out the continuity checks. Check the timeslots on which you want to run the continuity checks, and click the Start button.
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It is only possible to start manual checks on circuits that are in the Idle or Recheck state. Starting manual checks on a group of circuits will only be permitted if all selected circuits are in the appropriate state.
Figure 6-9 SS7 Trunk
Figure 6-10 Non SS7 Trunk
NOTE GIII will not place the circuit into the recheck state when a manual continuity check fails. GIII will take the circuit out of the recheck state when a manual continuity check passes.
6.5 IP diagnostics
Figure 6-11 IP diagnostics page
The IP diagnostics page provides some basic methods to test connectivity from GIII node.
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6.5.1 Ping To use the ping diagnostic, select the interface required: Traffic, or Admin, enter the IP address or hostname required, select the type of IP address: IPv4 or IPv6, and press the Ping button.
Figure 6-12 Ping diagnostic results
6.5.2 Traceroute To use the traceroute diagnostic, select the interface required: Traffic, or Admin, enter the IP address or hostname required, select the type of IP address: IPv4 or IPv6, then press the Traceroute button.
Figure 6-13 Traceroute diagnostic results
6.6 Trace Files Go to Maintenance – Trace Files, to:
• Download generated trace files
• Enable generation of trace from various elements within GIII There are four tabs available:
• Call
• SS7
• SIP
• Packet Capture
• Web System These pages share common behaviour for the Download, View and Delete buttons, as described below.
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6.6.1 Common behaviour 6.6.1.1 View
Figure 6-14 Trace Files page
The View button will be enabled when a trace file is selected. Pressing the button will display the file contents.
Figure 6-15 Trace file view
The contents of a file being view are paged such that a large file will only display a section of the file. The following controls are provided to move through, or search a file’s contents.
Figure 6-16 Paged contents selection
Display the start of the file
Display previous page
The current page of the total pages.
Display next page
Display last page of the file
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Figure 6-17 Search text tool
Search backwards for the text
The text to search for
Search forwards for the text The matching text will be highlighted in red.
Figure 6-18 Search highlighting
The back button will return to the previous trace file page.
6.6.1.2 Download The Download button will not be usable until one or more log files are selected. Multiple files can be selected by holding the ctrl key, and selecting multiple files. When multiple files are selected, they will be downloaded as a zip file. Compression will be used if the Use compression checkbox is selected.
Figure 6-19 Download files
If a download contains multiple files, then a zip file will be created. Due to the possibility of the file being large, it can take some time, and progress information may be displayed:
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Figure 6-20 Constructing file progress
Once the file has been constructed it will be downloaded by the browser.
6.6.2 Call Trace Files The Call tab will show the call trace files page.
Figure 6-21 Call trace files
The log files are displayed for the selected node. You can view the log file for a particular node by selecting it from the dropdown.
Figure 6-22 Node selection
The Refresh button will refresh the file list for the selected node.
NOTE Call Trace files are rotated every hour, and up to a month of trace files is kept. A trace file that is currently being written to can be deleted, and will be recreated the following minute.
NOTE It is recommended to use compression to download large trace files, as they are text, and will reduce in size.
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6.6.3 SS7 Trace Files The SS7 tab will show the SS7 trace files page. No files will be displayed, if they have yet to be generated.
Figure 6-23 SS7 trace files
The Generate button will create a SS7 log file and dump the SS7 ring buffer contents to the file for analysis. The log file contains the node G number and a date and timestamp. To decode the variant information select the SS7 trace file, and press the Decode button. This will create a new file with ‘_<variant>’ appended to the name. The Refresh button may need to be pressed to show the new file. The Decode Variant select control determines which variant the decode operation will use.
NOTE It is recommended that housekeeping is done to delete log files no longer needed.
6.6.4 SIP Trace Files The SIP tab will show the SIP trace file page.
Figure 6-24 SIP Trace page
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NOTE SIP trace files can be large. The GIII will protect against a SIP trace left running by continuously rotating the current file across 5 files of 1GB in size.
The Start button can only be selected if no SIP trace is already running. Pressing the Start button will start SIP trace on the selected node. It will show that SIP trace is running as below:
Figure 6-25 SIP Trace page when SIP trace is running
The Stop button will finish the SIP trace. The SIP trace file can be downloaded or viewed by selecting the file and pressing the appropriate button.
NOTE It is recommended to use compression to download large SIP trace files, as they are text and will reduce in size.
NOTE It is recommended that housekeeping is done to delete log files no longer needed.
6.6.5 Packet Capture Trace Files This page allows a packet capture to be performed on a certain interface.
NOTE A packet capture will be rotated across 5 files with a maximum size of 1GB per capture file. Use the Host Address(es) and Port(s) fields to restrict the capture if possible.
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Figure 6-26 Packet capture page
The files can be downloaded or deleted by selecting the files required for the operation and pressing the appropriate button.
NOTE It is recommended that housekeeping is done to delete log files no longer needed.
6.6.5.1 Host Address(es) If this field is empty then no filtering on host addressing will be performed.
Figure 6-27 Packet capture host address tool tip
Hover over the Host Address(es) text to see the tooltip. This field accepts a single host (ipv4|ipv6|hostname), a CIDR <ipaddr>/<bits>. This can be a single item, or a comma separated list of these. i.e. 10.0.0.1 10.0.0.0/24 10.0.1.1, 10.1.1.95/29, 10.1.2.3
6.6.5.2 Ports If this field is empty then no port filtering will be performed.
Figure 6-28 Packet capture port tool tip
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Hover over the Port(s) text to see the tooltip. As show the field accepts a single TCP/UDP port (1-65535), a range 5060-5061, or a comma separated list of these. i.e. 5060 5060-5061 53, 5060-5061, 443
6.6.5.3 Interface The interface to capture can be selected from the Interface drop down. It can be Traffic, or Admin, and the IP address shown will depend upon which Node is selected.
6.6.6 Web System
Figure 6-29 Web System log file page
There is a log file kept by the web site containing the system messages which have been shown in the System Message dialog. This may be useful if you miss a message, or wish to look at what has been changed on a GIII.
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6.7 Technical Support
Figure 6-30 Technical Support page
Aculab support may request a diagnostics snapshot file from the GIII to investigate an issue. This is done using the Technical Support page. Due to the possibility of very large files being included in the snapshot, it is possible to reduce the number of files using the Include additional files controls. The period over which to collect files can be specified from the last hour, day, week, 2 weeks, 4 weeks, or all. Also checkboxes allow certain types of logs to be included or excluded. Aculab support may specify what to use here.
Figure 6-31 Snapshot in progress
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Figure 6-32 Snapshot created
Once the file has been created, it may be downloaded by pressing the Download button.
NOTE Only one diagnostics snapshot may be created on a node at any time. Creating a new diagnostics snapshot will replace any previously created one.
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7 GIII Status monitors The status monitors can be accessed from the Status Monitors menu. They various pages provide status information about different aspects of GIII.
7.1 System Summary Go to Status Monitors – System Summary, to get to the System Summary page. This page presents a high level overview of the health of each GIII node in a cluster. From this page you are able to see the following information for each Node:
• Installed software version
• Installed role
• Admin and Traffic interface status and IP information
• Power supply status
• RAID disk status
• TDM trunk status
• System alarms
Figure 7-1 System Summary
The System Summary page will change colour for each node, depending upon the status of that node.
Figure 7-2 Node with no active role
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Figure 7-3 Node with a role starting
Figure 7-4 Node with an active role
Figure 7-5 Node with a failed role, or is unavailable
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7.1.1 GIII back panel image
Figure 7-6 GIII back panel image
The GIII back panel image shows various statuses.
7.1.1.1 TDM Trunk Status
Figure 7-7 Trunk status
The status of each trunk can be seen. The trunk number is shown, and the colour will depict the status, green is No Alarms Detected and layer 2 up, orange is No Alarms Detected and layer 2 down, and red is for Alarms Detected. Navigating to the Status Monitors -> TDM Trunks page will show more information.
7.1.1.2 IP interface status
Figure 7-8 The IP interfaces graphic showing status
The status of the IP interfaces is displayed, with green for working, and red for not.
7.1.1.3 PSU status
Figure 7-9 The fan statuses graphic
The statuses of the PSUs are shown as green for present/working, and red for not present/working.
7.1.1.4 System LED
Figure 7-10 System LED status graphic
The system LED will be shown to toggle on and off if the Node LED has been configured to on.
TDM Trunk Status
System LED IP Interface Status PSU Status
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7.1.2 Node status This shows the availability, call processing engine status, and the software version. The availability will be either Available, or Unavailable. The call processing engine status can be one of: Idle, Starting, Running, Restarting, Stopping, or Failed.
7.1.3 Role status The role status simply shows the role that is applied to the node, or [none] if no role is applied.
7.1.4 Admin Interface status This will show the IPv4/6 addressing configured for the admin interface on this node, a small CPU histogram chart, and the status of each admin interface.
7.1.5 Traffic Interface status This will show the IPv4/6 addressing configured for the traffic interface on this node, a small CPU histogram chart, and the status of each traffic interface.
7.1.6 Heartbeats status The heartbeats status shows the last time a status heartbeat was received. This is mainly useful in a cluster to see that nodes are still in contact, and there is not an issue.
7.1.7 System status The system status displays the time the device has been up for, and 3 small graphs to show the percentage of disk IO for both disks, and the percentage memory usage on the device.
7.1.8 Alarms status There are various alarms that can be shown in the Alarm status section for a node. The icons will show a tool tip if hovered over and possibly a link that can be clicked to load the relevant page to show more information.
Figure 7-11 Card alarm
The card alarm indicates that there is a problem with one or more of the Prosody-T traffic cards.
Figure 7-12 Cores alarm
The cores alarm indicates that one of the software processes has had a crash and a core file has been generated. The tool tip will load the technical support page, where a diagnostics snapshot package can be created, the Core dumps option needs to be checked. This diagnostics snapshot should be sent to Aculab Support for analysis. The creation of the diagnostics snapshot will clear the Cores alarm on the status page.
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Figure 7-13 Fans alarm
The Fans alarm indicates there is a problem detected with an alarm. The tool tip will load the Temperatures and Fans page, and the Fans tab will show the status of each fan.
Figure 7-14 Activation Key alarm
The Activation Key alarm indicates a problem with the activation key, either it has not yet been configured, or it is invalid for the GIII software version.
Figure 7-15 NTP alarm
The NTP alarm indicates a problem with NTP. NTP is an essential feature of GIII as it is used to synchronize database state between nodes in a cluster. For a standalone device the database synchronization is not an issue, but log files will no longer have timestamps which are correct.
Figure 7-16 Essential Processes alarm
The essential processes alarm indicates that one or more of the GIII software processes are in an unexpected state. The tool tip will provide a link to the Essential Processes page for more information.
Figure 7-17 RAID alarm
The RAID alarm indicates a problem with the RAID status. A tool tip will provide a link to the Disk Management page for more information.
NOTE If a disk is rebuilding as part of a disk replacement, then this alarm will be shown on the System Summary page until the activity is complete. The disk rebuild can take over a day to complete as the rate is throttled to ensure no interference with call traffic.
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Figure 7-18 SMART alarm
The SMART alarm indicates a problem with the SMART status on one, or both, disks. A tool tip will provide a link to the Disk Management page for more information. Depending upon what is being reported, it may be an early indicator of a failing disk.
Figure 7-19 Temperature alarm
The Temperature alarm indicates there is a problem with one or more temperatures of components of the GIII system. The tool tip provides a link to the Temperatures and Fans page which will display more information about what components are causing this alarm.
Figure 7-20 Upgrade Cluster version alarm
The upgrade cluster version alarm indicates that the device is running a cluster version which is less than the supported version of the software. The tool tip provides a link to the Administer Cluster, Change Cluster Version page. This will allow the cluster version to be set to the latest supported, but only if all nodes in the cluster have been upgraded to run a software version which supports this cluster version.
Figure 7-21 Cluster problem alarm
The cluster problem alarm indicates that there is something potentially wrong with the cluster. The tool tip provides a link to the Status Monitors - Cluster Status page. This alarm would be seen if a node in the cluster is down, rebooting, or leaving. It provides an indication that the cluster is not stable, and would not allow certain clustering operations to be performed, i.e. you cannot join a new node if the cluster does not have all nodes with an Up – Normal status.
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7.2 Call Traffic Go to Status Monitors – Call Traffic, to get to the Call Traffic page. Each of the trunks present in the assigned role for the node will be displayed, and the status of each of its bearer timeslots shown.
7-22 Call Traffic
The Key button shows the symbols that are used to represent timeslot states. The following symbols are used to represent timeslot states:
(Black box with grey background) The timeslot is not a bearer timeslot, but is an HDLC or signalling timeslot.
(Black box with turquoise background) The timeslot is an SS7 signalling link that is currently in service.
(Black box with turquoise background and black cross) The timeslot is an SS7 signalling link that is currently out of service.
(Black box with black cross) The timeslot is currently out of service, and unavailable for calls. Usually this will indicate that the timeslot has been placed out of service in the Trunk Blocking page. In the case of SS7 timeslots, it can also indicate that the timeslot has had CIC blocking applied by the far end.
(Black box with white background) The timeslot is currently idle.
(Black box with green circle) The timeslot is carrying a call in the setup phase.
(Red box with green circle) The timeslot is carrying a call that has remained in the setup phase beyond the alert threshold.
(Black box with green background) The timeslot is carrying a connected call.
(Red box with green background) The timeslot is carrying a call that has remained in the connected state beyond the alert threshold.
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(Black box with grey circle) The timeslot is carrying a call that reached the connected state and is now in the release phase.
(Red box with grey circle) The timeslot is carrying a call that reached the connected state and has remained in the release phase beyond the alert threshold.
(Black box with red circle) The timeslot is carrying a call that did not reach the connected state and is now in the release phase.
(Red box with red circle) The timeslot is carrying a call that did not reach the connected state and has remained in the release phase beyond the alert threshold.
(Black box with white background and black T) The timeslot is carrying a test call.
(Black box with white background and black R) The timeslot is an SS7 bearer channel that is currently in recheck mode because it failed an in-call continuity check.
(Red box with white background and black R) The timeslot is an SS7 bearer channel that is currently in recheck mode because it failed an in-call continuity check, and has remained in recheck mode for more than 30 minutes. The Alert Thresholds button brings up the Alert Thresholds page, which allows you to set threshold timers for visual warnings when a timeslot stays in the same state for a prolonged period, as shown if Figure 7-18.
Alerts It is possible to display a visual alert on a call timeslot, when the call in progress has remained in the same state for a prolonged period. The thresholds at which visual alerts are displayed can be configured using the Alerts Threshold page. Select the Alerts Threshold button on the Call Traffic page from the Status Monitors menu. The configured thresholds apply to all calls in the system.
7-23 Call Monitor Alert thresholds
Setup Threshold – This is the period, after which calls that are still in the setup phase will be highlighted. The default period is 180 seconds.
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Connected Threshold – This is the period, after which calls in the connected state will be highlighted. The default period is 30 minutes. Clearing Threshold – This is the period, after which calls that are still in the release phase will be highlighted. The default period is 180 seconds. Submit – This will apply the changes to both calls currently in the system, and all future calls. The settings will be applied cluster-wide Cancel – This will discard any changes made.
7.3 TDM Trunks Go to Status Monitors – TDM Trunks, to get to the TDM Trunks page. This page displays the current state of each of the trunks present in the role assigned to a node.
Figure 7-24 TDM Trunks page
Trunk – the identifier for the trunk in the role Layer 1 – the overall layer 1 status for the trunk. Possible values include:
No Signal Detected – indicates a cable fault or a failure of the far end equipment. A loop back on that trunk can be used to prove the card; status No Alarms detected proves the cabling is OK. No Alarms Detected – Indicates everything is good. Layer 2 – The layer 2 status can be up or down. Slips - are a count of the slips received on a trunk. Use the Clear button to zero this count and monitor how long it takes to build up. Slips will occur when the trunk is in an alarm state. When the trunks show ready and are showing slips, the primary clock source should be checked. BV (bipolar violations) – low level hardware monitors and have little meaning outside Aculab. NOS (No Signal Detected) – this will be set to 0x00 when a signal detected is detected, and 0xff when no signal is detected AIS (Alarm Indication Signal) – indicates the trunk is receiving an Alarm Indication Signal from the far end equipment LOS (Loss of synchronisation) – indicates the trunk is receiving a Loss of Synchronisation indication from the far end equipment
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RRA (Received Remote Alarm) – indicates the trunk is receiving a Remote Alarm Indication from the far end equipment TRA (Transmit Remote Alarm) – indicates the trunk is transmitting a Remote Alarm Indication to the far end equipment RMA (Received Multiframe Alarm) – indicates the trunk is receiving a CAS Multiframe Alarm from the far end equipment TMA (Transmit Multiframe Alarm) – indicates the trunk is transmitting a CAS Multiframe Alarm to the far end equipment USR – (User state) – 0xf1 is operational, 0xf3 is network side, and 0xf7 is user side FW Ver. – the version number of the protocol firmware running on the trunk Signal Ver. – the version number of the signalling protocol being used on the trunk
7.4 CAS Trunks Select Status Monitors – CAS Trunks to monitor the transmitted and received ABCD bits on a CAS port. This is useful for checking back-busied and/or seized lines.
Figure 7-25 GIII status monitor CAS tab option
Select the require trunk tab from the list provided.
NOTE Only CAS ports will be available in the list. Trunks not running CAS firmware will detail that no data is available, i.e.:
The Tx column shows the transmitted ABCD bits, the Rx shows the received ABCD bits. With E1 CAS, timeslots 0 & 16 are used as timing and system data channels and not for voice or data calls/traffic. Therefore TS 00 and TS 16 will show a different state from the other timeslots (channels).
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7.5 Ethernet Trunks Go to Status Monitors – Ethernet Trunks, to get to the Ethernet Trunks page.
Figure 7-26 Ethernet Trunks page
7.6 Call Trace
Figure 7-27 Call Trace page
The Call Trace page can be used to see any call trace data, for each node in the cluster that is configured to be sent to the UI. The Role Call Trace configuration settings [TODO: ref] determine what is sent to this page. It will only display data if the role is active. The Pause button will halt the update of this page when pressed. The Clear button will clear the current data.
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7.7 SS7 Go to Status Monitors – SS7, to get to the SS7 page, which showing information on the state of the SS7 signalling links and bearers configured on each node.
7.7.1 MTP3 Signalling This page provides continuous monitoring of the SS7 MTP3 signalling links in the system.
Figure 7-28 SS7 Signalling tab
The page contains an entry for each signalling link that has been configured for use on a trunk. The Variant field shows which variant has been configured in the OPC configuration in SS7 Configuration.
NOTE The resultant configuration can be seen in the [MTP3] section of the SS7 stack file for that signalling link; this can be viewed from the Advanced configuration tab in the SS7 Configuration page.
The State column reports the current state of the link, the normal state being MTP3 and MTP2 Available. When a link is not in the normal state, selecting it will list the reasons in the Diagnostics column.
NOTE Any issues reported in the Diagnostics column will prevent traffic from being passed. However, some issues are transient in nature and will only be present whilst a link is being brought into service.
7.7.2 ISUP Bearers The Bearers tab provides continuous monitoring of the SS7 ISUP bearer channels in the system.
Figure 7-29 GIII status monitor SS7 ISUP Bearers
The tab show an entry for each bearer channel configured for a trunk. Buttons are provided to Sort by Pointcode or Sort by Trunk and Timeslot.
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The Variant field shows which variant has been configured in the OPC configuration in SS7 Configuration.
NOTE The resultant configuration can be seen in the [MTP3] section of the SS7 stack file for that signalling link; this can be viewed from the Advanced configuration tab in the SS7 Configuration page.
The Bearer State column reports the current state of the bearer, the normal states being:
Idle Incoming Outgoing Connected Released Free
When a bearer is not in the normal state (this includes any of the above states preceded by Fail/), it will list the reasons in the Diagnostics column.
NOTE Any issues reported in the Diagnostics column will prevent calls from being passed. However, some issues are transient in nature and will only be present whilst a bearer is being reset or returned to service.
7.7.3 MTP3 Routes The MTP3 Routes tab provides continuous monitoring of the SS7 routes that are used by the system.
Figure 7-30 GIII status monitor SS7 MTP3 Routes
The tab contains entries for each destination that has been configured for a trunk. Two types of route may be reported:
• Direct routes, where a direct connection exists between GIII and the destination. For such a route the AdjacentPC and RemotePC fields will be identical, whilst the Priority field will always be set to 0.
• Indirect routes, where the path to the destination passes through a Signalling Transfer Point (STP). For such a route the values in the AdjacentPC and RemotePC fields will differ, whilst the Priority field will indicate the priority assigned to the route OPC configuration. The Variant field shows which variant has been configured in the OPC configuration in SS7 Configuration.
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NOTE The resultant config can be seen in the [MTP3] section of the SS7 stack file for that signalling link; this can be viewed from the Advanced configuration tab in the SS7 Configuration page.
The State column reports whether the route is currently available for use, the normal state being Working for direct routes, and Working or Standby for indirect routes. A route that shows a state of Restricted/Working or Restricted/Standby indicates that a Transfer Restricted (TFR) messages have been received for the route, although the route is still able to carry traffic. When a route is not in the normal state, it will list the reasons in the Diagnostics column.
NOTE Any issues reported in the Diagnostics column will prevent traffic from being passed. However, some issues are transient in nature and will only be present whilst a route is being brought into service.
7.7.4 MTP3 Destinations The MTP3 Destinations tab provides continuous monitoring of the SS7 destinations that are used by the system.
Figure 7-31 GIII status monitor SS7 MTP3 Destinations
The tab contains an entry for each destination that has been configured for a trunk. The Variant field shows which variant has been configured in the OPC configuration in SS7 Configuration.
NOTE The resultant config can be seen in the [MTP3] section of the SS7 stack file for that signalling link; this can be viewed from the Advanced configuration tab in the SS7 Configuration page.
The State column reports whether the destination can currently be reached, the normal state being Accessible. A destination that remains accessible will show a state of Restricted if it can only be reached using restricted routes (i.e. Transfer Restricted (TFR) messages have been received for all routes to the destination). When a destination is not in the normal state, selecting it will list the reasons in the Diagnostics column.
NOTE Any issues reported in the Diagnostics column will prevent the destination from being reached. However, some issues are transient in nature and will only be present whilst a restart is in progress.
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7.7.5 M3UA Connections The M3UA Connections tab provides continuous monitoring of the SS7 M3UA connections in the system.
7-32 GIII status monitor SS7 M3UA Connections
The tab contains an entry for each connection that has been configured for the Client and IPSPClient, and will show entries for any inbound connections for an IPSP Server. The Variant field shows which variant has been configured in the OPC configuration in SS7 Configuration.
NOTE The resultant configuration can be seen in the [M3UA] section of the SS7 stack file for that signalling link; this can be viewed from the Advanced configuration tab in the SS7 Configuration page.
The Routing Context State column reports one of the states: Disconnected, Idle, Registering, Registered, Activating, Activated, Active, Inactivating, Deregistering.
7.7.6 M3UA Connection Stats The M3UA Connection Stats tab provides continuous monitoring of the SS7 M3UA connections in the system.
7-33 GIII status monitor SS7 M3UA Connection Stats
The tab contains an entry for each connection that has been configured for the Client and IPSPClient, and will show entries for any inbound connections for an IPSP Server.
NOTE The resultant configuration can be seen in the [M3UA] section of the SS7 stack file for that signalling link; this can be viewed from the Advanced configuration tab in the SS7 Configuration page.
The Connection State column reports one of the states: Disconnected, Idle, Registering, Registered, Activating, Activated, Active, Inactivating, Deregistering.
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7.7.7 M3UA Addresses The M3UA Addresses tab provides continuous monitoring of the SS7 M3UA addresses in the system.
7-34 GIII status monitor SS7 M3UA Connections
The tab contains an entry for each connection that has been configured for the Client and IPSPClient, and will show entries for any inbound connections for an IPSP Server. The Variant field shows which variant has been configured in the OPC configuration in SS7 Configuration.
NOTE The resultant configuration can be seen in the [M3UA] section of the SS7 stack file for that signalling link; this can be viewed from the Advanced configuration tab in the SS7 Configuration page.
The Routing Context State column reports one of the states: Disconnected, Idle, Registering, Registered, Activating, Activated, Active, Inactivating, Deregistering.
7.7.8 M3UA Routing Key Stats The M3UA Routing Key Stats tab provides continuous monitoring of the SS7 M3UA routing keys in the system.
7-35 GIII status monitor SS7 M3UA Routing Key Stats
The tab contains an entry for each connection that has been configured for the Client and IPSPClient, and will show entries for any inbound connections for an IPSP Server. The Variant field shows which variant has been configured in the OPC configuration in SS7 Configuration.
NOTE The resultant configuration can be seen in the [M3UA] section of the SS7 stack file for that signalling link; this can be viewed from the Advanced configuration tab in the SS7 Configuration page.
The ASP State column reports one of the states: Idle, Listening, Need to connect, Connect retry, Data transfer, Waiting for ASPSM up ack, Waiting for ASPSM down ack, Active .
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7.8 SIP Gateways Go to Status Monitors – SIP Gateways, to get to the SP Gateways page. This page displays the status of each of the SIP gateways to which outgoing calls can be directed.
Figure 7-36 GIII Status Monitor SIP Gateways page
There is one entry in the list for each of the gateways configured in the Role Configuration – Trunks - SIP – Gateways tab and only those gateways configured in that page are reported on this screen. The Address column will list IPv4 addresses ahead of IPv6 addresses, with each address group sorted in network order. The Status field shows the current status of the gateway, and will be one of the following:
Discovering – the gateway is waiting to be polled for the first time. These entries will be orange. Available – the latest poll reported that the gateway was able to accept SIP calls. When monitoring of the gateway is not enabled this status will be shown. These entries will be green if monitored, otherwise will have no colour. Unavailable(nnn) – the latest poll reported that the gateway was unable to accept SIP calls, with nnn indicating the SIP OPTIONS response. These entries will be red. Unavailable(Call) – an outgoing call to the gateway has been rejected with one of the responses in the Unavailable Responses list. These entries will be red. Unavailable(Timeout) – the gateway is not responding and is unreachable. These entries will be red. Unavailable(Server) – an unexpected error occurred when determining the gateway status. These entries will be red.
The Current Calls field shows the number of calls in progress to and from the gateway. The Monitor field will show Yes if this gateway monitoring enabled, and will be blank otherwise. The Disabled field will show Yes if the Disable Selection checkbox in the Role Configuration – SIP – Gateways page is checked, and will be blank otherwise. This field does not report the setting of the Disable Selection checkbox on the Role Configuration – SIP – Gateway Routes page.
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7.9 NTP This page will show the NTP status for each node in the cluster. Each node is displayed in an accordion.
Figure 7-37 NTP status
7.10 Temperatures and Fans
Figure 7-38 Temperatures and Fans
This temperatures tab will show the temperatures for each card in the GIII as well as the system temperatures.
Figure 7-39 Fans status
The fans tab will display the fan speeds for the system fans.
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7.11 Essential Processes
Figure 7-40 Essential processes page
This page will show the statuses for the essential processes of the GIII. If a problem is detected with one of these processes then an alarm is shown on the System Summary page.
NOTE The list of processes will be different, depending upon whether a role is active or not.
7.12 System card status
Figure 7-41 System card status page
This page will simply show the status of the Prosody T cards in the system. Any problems will be shown highlighted in red. This page also shows the firmware statuses, and if a card firmware requires updating then the System card status alarm will be shown on the system summary page. The firmware can be upgraded from the Node Administration – Software Upgrades page.
Figure 7-42 – Card status showing firmware update required
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7.13 Cluster Status
Figure 7-43 Cluster Status page
The cluster status page shows each nodes view of the cluster status. Networking issues could cause one nodes cluster status to be different than another. The Status is shown as “<state> - <status>”, where state is either Up or Down, and status is Joining, Normal, or Leaving. The cluster status is required to have all nodes as ‘Up – Normal’ to allow a node to join the cluster. When removing a node from the cluster, where the node is no longer in contact with the cluster (for instance a failed or repurposed device), the cluster would show the state as ‘Down – Leaving’ whilst it is being administratively removed. During this period the cluster would not allow a new node to join until it is removed, and would no longer appear in the Nodes listed.
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8 Node Administration
8.1 Configure Admin IP Address The Admin IP addressing is configured from the Configure Admin IP Address page which is located under the Node Administration menu. This page will only allow the IP address information to be changed if there is no active role on the GIII. The GIII is required to reboot when the Admin IP address information is changed.
Figure 8-1 Admin IP configuration page
8.2 Web Site TLS Certificate The Web Site TLS Certificate page is located in the Node Administration menu. A number of web browsers will not offer to save credentials, and will display warnings, if a web site is not deemed secure.
Figure 8-2 Chrome browser warning
To stop the warnings, and allow credentials to be saved for the GIII website, you will need to install a certificate that is trusted in your domain. Please consult your IT department to achieve this.
8.2.1 Installing new certificate
Figure 8-3 Web site TLS certificate page
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To install a new certificate you press the Choose File button which will open your system file browser, and select the required certificate. The certificate is expected to be in PEM format. If the private key is present and requires a password, then this must be set in the Private key passphrase field. Pressing Submit will install this certificate for web site and web services usage. There are two accordions displayed which can be expanded to show the certificate and its information. If the certificate is not acceptable then a warning message will be shown in the system messages
Figure 8-4 Certificate error
8.3 Disk Management The Disk Management page is located in the Node Administration menu. A GIII node has two disks configured in a software RAID 1 configuration.
Figure 8-5 Disk Management
When functioning correctly, the Disk Management page should show both disks reporting state as In use. The Remote Disk buttons allow a drive to be removed from the software RAID.
8.3.1 Replacing a defective drive GIII will generate SNMP traps for disk problems, namely degraded RAID, and disk SMART errors. An alarm will also be shown on the System Summary page if an error is detected.
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Figure 8-6 The RAID alarm on the status page
Figure 8-7 Drive showing faulty status
The Disk Management page will show Faulty for a drive that is defective. If the Remove Disk button is pressed then the drive will be removed from the RAID.
CAUTION It is recommended that a system backup is performed before replacing a drive.
The defective drive can now be removed and a new drive added.
NOTE The replacement drive must be the same size or greater that the existing disk. Aculab support can provide details of a suitable replacement disk.
Figure 8-8 Enable replacement drive
The Enable Disk button will, once pressed, add the new drive into the RAID, and will start rebuilding the RAID.
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Figure 8-9 RAID rebuilding
The Disk Management page will show the progress of the RAID rebuilding, it will refresh the information automatically.
NOTE The RAID rebuild will take some time. The system should operate as normal with this being done in the background.
8.3.2 SMART failures and alarms The GIII disk SMART statuses are checked periodically, and if a potential problem is detected then a SMART alarm will be shown on the System Summary page.
Figure 8-10 System Summary showing SMART failures alarms
Figure 8-11 Node with disks exhibiting SMART failures
To examine the disk failure information, press the View button, and this will open a dialog containing the failure log.
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Figure 8-12 SMART failure log
If the SMART failures are not considered to be of concern, then the alarm can be removed from the System Summary page by pressing the Clear button. The SMART failure log will remain accessible unless the Delete button is pressed, which will show a confirmation dialog, and pressing Continue will result in removing this failure log file.
Figure 8-13 Deleting SMART failure log
CAUTION SMART disk failures may be an early sign of a pending disk problem, and if the SMART failures are deemed serious enough, replacing the disk as soon as possible may be the best thing to do. Interpreting the SMART failure logs is outside the remit of this document.
8.4 Card Setup
Figure 8-14 Card Setup page
This page allows the number of cards used by the GIII to be changed. It does not require the actual physical cards to be changed, but if more cards are configured than are physically present in the GIII, then the role will take longer to startup as it will be timing out on waiting for a non-existent card to come into service.
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The Adding, removing, or replacing Prosody T cards Appendix has details on how to replace or change the number of Prosody T cards in a GIII.
8.5 Node LED Control
Figure 8-15 Node LED control page
Once you have selected the node in the cluster, a System ID LED control is shown which allows the system LED to be turned on or off.
Figure 8-16 Changing the Node LED status for a node
The System summary page will also show the LED toggling state when enabled.
8.6 Rename a Node The individual nodes will have serial numbers of the form G7XXX, where XXX is a number. The serial number is shown by default to identify a node. An optional name can also be provided for a node, which will appear with the serial number. To rename a node, go to Node Administration – Rename Node. Select the node you wish to rename from the Select the Node to rename dropdown. Enter the New name, and click the Submit button.
Figure 8-17 Rename Node
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Figure 8-18 Example of node name with serial number
8.7 Software Upgrades Go to Node Administration – Software Upgrades to perform upgrade of the software on a GIII node. GIII has a Current software version, which is the software currently installed on the node. It can also have an Alternative software version, which is a version of software which can be used to upgrade/downgrade the current software version. It is possible to replace the alternative software version on a GIII node, prior to software upgrade.
Figure 8-19 Software Upgrades
8.7.1 System card updates
Figure 8-20 – System card requires an update
The Software Upgrades page will also show if any of the system cards require updating. The System Summary page will show a System Card alarm if any cards require updating, and the card firmware status is displayed on the Status Monitors – System Card Status page. If the Update system cards button is pressed then a warning dialog will be shown.
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Figure 8-21 – System card update dialog
NOTE The GIII will be restarted if the system cards are updated.
8.7.2 How to upgrade node software Here are the steps to follow when performing the upgrade of software version:
1. Go to Node Administration – Software Upgrades, and the Replace alternative software tab. 2. Click Choose File next to file to upload on the Replace alternative software tab.
Figure 8-22 Choose alternative software version
3. Browse to the directory where the update file is located, select the file, and click open.
Figure 8-23 Choose software version dialog
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4. The file will be uploaded to the GIII node.
Figure 8-24 Software upgrade upload
5. Once the software upload is complete, go to the Alternative software tab, and click the Use this version button.
NOTE The Alternative software tab may display that the system cards need to be updated.
NOTE If the cluster version has changed then additional steps are required. All devices in a cluster must first be upgraded, and then the Change Cluster Version tab on the Administer Cluster page will allow the cluster version to be changed. The software release notes will inform if this is necessary.
CAUTION It is recommended that a system backup is performed before performing a software upgrade.
CAUTION The GIII node will be restarted during the upgrade process. It is therefore recommended to put the node in to maintenance mode, and wait for all call traffic to clear, before performing a software upgrade.
6. Once the software upload is complete, go to the Alternative software tab, and click the Use this version button.
NOTE If, for any reason, a rollback to the previously install software version is required, this can be achieved by going to the Alternative Software tab, and clicking Use this version.
NOTE Once an upgrade has been completed, check the release notes to ensure that there are no remedial actions to take.
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8.8 Software Activation key 8.8.1 Viewing Activation key
Selecting Node Administration – Software Activation Key, to change the activation key. You will be presented with the following page.
Figure 8-25 Activation key page
Enter a valid activation key for the software version being used, for example, TKE-QLY-RRK-8LY as shown here. You will continue to be prompted until a valid activation key is entered.
8.8.2 Changing or adding an Activation key If you have already entered the activation key for your system, you can change it by going to the Node Administration – Software Activation Key menu option. Enter a valid activation key for the software version being used, for example, RQE-WHG-CFQ-LVX as shown here. You will continue to be prompted until a valid activation key is entered. A valid reason for changing the activation key may be due to a change of major revision to the GIII software.
8.9 Software Licenses
Figure 8-26 Software licenses page
This page provides information about software licenses for the GIII product.
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8.10 Documentation
Figure 8-27 GIII Documentation page
This page provides links to download the GIII user guide and hardware installation guide.
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9 Cluster Administration
9.1 Administer Cluster The Administer Cluster page is located in the Cluster Administration menu. A new GIII device, or a GIII device which has been de-clustered may be used for traffic without requiring starting a cluster, or joining to an existing cluster, it is a stand-alone cluster at this point, and will be internally be allocated a name of Cluster<serial>, where <serial> is the serial number of the GIII, i.e. if the GIII serial was G6666, then the GIII will show Cluster6666.
Figure 9-1 Cluster administration
9.1.1 Start a cluster You may only start a cluster if the node is not already a cluster. To start a cluster, enter both a name and a password for the cluster and select the Submit button.
Figure 9-2 Start a cluster
NOTE Starting a cluster does not require the GIII to reboot, and no settings that have currently been configured will be lost.
Once the cluster Cluster name and Cluster password have been submitted, then you can join this cluster using these credentials.
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Figure 9-3 Node already clustered
9.1.2 Join a Cluster To join a cluster, the Join a Cluster radio item must be selected, this will then show a page where you can enter the name, password, and IP address of a cluster. The IP address can be any of the existing nodes in this cluster.
Figure 9-4 Join a Cluster
NOTE You may only join a cluster if the cluster is stable, i.e. all nodes are in an ‘Up – Normal’ state. An alarm will be shown on the System Summary page if the cluster is not stable. The Cluster status can be viewed by looking at the System Monitors – Cluster Status page.
CAUTION The GIII will be restarted if you join a cluster. All settings apart from the Admin interface settings will be replaced with the cluster configuration.
CAUTION If a GIII device which is part of a cluster is going to be switched off for more than a week then it is imperative that the node is actually declustered before the long term power off, and then re-joined to the cluster when required. No configuration will have been lost as the existing cluster will maintain the cluster and role data.
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9.1.3 Decluster a Node To remove a node from a cluster, go to Cluster Administration – Administer Cluster, and select the Decluster a Node tab. Select the node you wish to decluster from the Select the Node to Decluster dropdown, and click the Submit button.
Figure 9-5 Decluster a Node
CAUTION As messages warn, be sure you select the correct node to decluster before pressing submit button. The node will be restarted if you decluster it. All cluster related data will be deleted.
9.1.3.1 Decluster a node that is currently unavailable Whilst it is recommended that nodes are declustered whilst on and active in the cluster, it is possible to remove a node from a cluster that is currently unavailable. This may be due to the unit being powered off, or a unit has failed and has been replaced or reinstalled.
NOTE Whilst the node is being removed from the cluster you will not be able to perform certain cluster operations as the cluster is not stable. For instance, you may only join a cluster if the cluster is stable, i.e. all nodes are in an ‘Up – Normal’ state. An alarm will be shown on the System Summary page if the cluster is not stable. The Cluster status can be viewed by looking at the System Monitors – Cluster Status page.
CAUTION If you decluster a node that is unavailable, you should be very sure that it will not come back, i.e. if it is powered off, you decluster this node whilst it is unavailable, then power the node on again, this could produce many undesirable configuration changes. This is considered misuse of the cluster and plenty of warnings are presented to advise you.
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9.1.4 Change Cluster version
Figure 9-6 Change cluster version page
The cluster version allows all nodes in a cluster to understand configuration and status for this particular version. If a software upgrade has increased the cluster version then all nodes in a cluster must be upgraded to be running the same software version before this it may be changed. An alarm will be displayed on the System summary page to indicate that a node can change its cluster version.
NOTE The page shows warnings that all the saved role revisions will be lost, and only the selected revision settings of each role will be available.
9.2 Administration of User Accounts The default account is the user admin. The default password for this account is password.
9.2.1 Changing a User Password There are two ways to change a user’s password. You can change the password of the user whom is logged in using the Change Password button located on the right side of the menu.
Figure 9-7 Change password button
Alternatively you can go to Cluster Administration – User Accounts, and the Modify user account tab. From here, when logged in as a user with Administrator rights, you can select an account from the Account dropdown, tick Change password, and enter the new password in the New password and Confirm password boxes, then click submit.
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Figure 9-8 Change user password
From the Modify user account tab, it is also possible to enable/disable a user account if required.
NOTE The admin account may not be disabled or deleted.
NOTE Only an Administrator level account can disable, delete, or change the password for a user other than themselves.
9.2.2 Create a new user account To create new user account, go to Cluster Administration – User Accounts, and select the Add user account tab. Enter the Account username, along with the account password, in the Password and Confirm password boxes. You also need to select the account role from the Account role dropdown. The options are: Monitor – Allow the user to only gain access to the Status Monitors menu, to look at the heath of the gateway Administrator – Allow the user full administrative right over the configuration of the Node Press the submit button, to create the account.
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Figure 9-9 New user account setup
Figure 9-10 Viewing current user accounts
9.2.3 Modify user account
Figure 9-11 Modifying a user account
The Modify user account tab allows a user to be enabled, or disabled, and/or the password to be changed.
9.2.4 Delete user account
Figure 9-12 Deleting a user account
The Delete user account page will allow an administrator to be able to delete user accounts.
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9.3 Administer CDRs (Call data record) Call Data Records (CDRs) are generated and logged by GIII for each call that has completed. The Administer CDRs page allows you to query the logged CDRs to create report files, which can then be downloaded from the node. To query CDRs, select Administer CDRs from the Cluster Administration menu.
Figure 9-13 CDR Query settings
On the CDR Query tab you can define the output format for the CDR report. The previously used query will be saved cluster-wide. Once a CDR report has been generated, it can be downloaded via the CDR Query Files tab.
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CDR Field options The fields in the following table can be used as one of the query fields when generating a CDR report.
Field Description Call direction A two character field that indicates the type of end to
end call made. The first character represents the incoming call leg and the second character the outgoing call leg, for example TI. The following characters are used: T – The call leg was a TDM call. I – The call leg was an IP telephony call. - – No call Leg was made (outgoing position only).
Incoming port number The port number on which the incoming call leg was received. For IP telephony calls, which receive all incoming calls on the host port, this will be the port selected to carry the media.
Incoming timeslot The number of the timeslot on which the incoming call leg was received. For IP telephony calls this will be a value assigned by the GIII application, and will be the same as that used in the call control trace.
Incoming signalling type This is a string returned by the Aculab call control software to indicate the protocol used to carry the incoming call leg.
Incoming IP DSP stream This field is used for fault finding and has no meaning outside of Aculab.
Incoming IP DSP timeslot This field is used for fault finding and has no meaning outside of Aculab.
Incoming codec type If an incoming IP telephony call reaches the connected state, this field will contain the codec that was initially negotiated with the remote endpoint, otherwise the field will be empty.
Outgoing port number The port number on which the outgoing leg of the call was placed. For IP telephony calls this will be the port selected to carry the media.
Outgoing timeslot The number of the timeslot on which the outgoing call leg was placed. For IP telephony calls this will be a value assigned by the GIII application, and will be the same as that used in the call control trace.
Outgoing signalling type This is a string returned by the Aculab call control software to indicate the protocol used to carry the outgoing call leg.
Outgoing IP DSP stream This field is used for fault finding and has no meaning outside of Aculab.
Outgoing IP DSP timeslot This field is used for fault finding and has no meaning outside of Aculab.
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Field Description Outgoing codec type If an outgoing IP telephony call reaches the connected
state, this field will contain the codec that was initially negotiated with the remote endpoint, otherwise the field will be empty.
Call start date The date on which the incoming call leg was received in the form dd/mm/yy.
Call start time The time at which the incoming call leg was received in the form hh:mm:ss. A 24 hour clock is used.
Call answer time The time at which the outgoing call leg was answered in the form hh:mm:ss. A 24 hour clock is used.
Call end time The time at which the call was terminated in the form hh:mm:ss. A 24 hour clock is used.
Call duration The duration for the which the call was in the connected state in the form hh:mm:ss. This is the difference between Call answer time and Call end time.
Call ID This is the numeric part of the call ID (ID:) used to identify the call in the GIII call control trace, for example 0000012345.
Incoming destination address This is the called party number presented by the incoming call leg. If the incoming call leg is a SIP call this will be the user part of the Request-URI.
Incoming IP network destination This field is used only when the incoming call leg is an IP telephony call and will be the host part of the SIP Request-URI, typically the IP address of the GIII host port. If the incoming call leg is a TDM call then this field will be empty.
Incoming originating address This is the calling party number presented by the incoming call leg. If the incoming call leg is a SIP call this will be the user part of the URI in the P-Asserted-Identity, if present, or otherwise the From header.
Incoming IP network origin This field is used only when the incoming call leg is an IP telephony call and will be the host part of the URI in the P-Asserted-Identity, if present, or otherwise the From header. If the incoming call leg is a TDM call then this field will be empty.
Outgoing destination address This is the called party number sent in the outgoing call leg. If the outgoing call leg is a SIP call this will be the user part of the Request-URI.
Outgoing IP network destination This field is used only when the outgoing call leg is an IP telephony call, and will be the IP address of the endpoint to which GIII has directed the call. If the incoming call leg is a TDM call then this field will be empty.
Outgoing originating address This is the calling party number sent in the outgoing call leg. If the outgoing call leg is a SIP call this will be the user part of the URI in the P-Asserted-Identity, if present, or otherwise the From header.
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Field Description Outgoing IP network origin This field is used only when the outgoing call leg is an
IP telephony call. It is the value used in the call control signalling to indicate the endpoint from which the call originated, and will always be the GIII host IP address. If the incoming call leg is a TDM call then this field will be empty.
Incoming service indicator code This indicates the type of line requested by the incoming call, and will be one of 3.1KHz, Speech, 7KHz, FAX or DATA64K. Protocols that cannot present this information (CAS and SIP) will always report Speech. Line types that are not recognised by GIII will be reported as Other. This field is directly related to the SIC control for the incoming call in the Routing Configuration screen.
Raw call termination cause code If the remote end disconnects the outgoing call leg before the connected state is reached, then this field will contain the clearing cause returned by the remote end. Raw clearing causes are decimal values that are specific to the signalling protocol in use. If the outgoing call leg is disconnected after reaching the connected state then this field will contain zero.
Generic call termination cause code
This field is used for fault finding and has no meaning outside of Aculab.
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Field Description Call termination reason This is a single character field that indicates why the
call was terminated, and will be one of the following: Y - A successful call was terminated normally. R - Routing failed. O - The Aculab call control software rejected the
request to make an outgoing call. E - The outgoing call was rejected with user busy. U - The outgoing call was rejected with number
unobtainable. N - The outgoing call was rejected for an unspecified
reason. H - The caller hung up before the call reached the
ringing stage. I - Terminated by the incomplete dialling timer (see
Call Setup Timers section) M - Terminated by the maximum ringing timer (see
Call Setup Timers section) X - A ringing call was terminated before being
answered. L - Terminated by the disconnect calls on layer 1
alarm feature (see the Call Handling section for TDM trunks).
S - The call was terminated to carry out a firmware download, or because the system is closing down.
B - The call was disconnected when a timeslot carrying one of the call legs was blocked (see Trunk Blocking section).
T - The call was disconnected because the timeslot carrying it was reset (see the SS7 Circuit Reset section).
C - The call was terminated because the service and/or media type requested is not supported.
P - The call was terminated due to out of procedure action by the remote endpoint.
Call redirection address Incoming enquiry call port Incoming enquiry call timeslot
When interworking from SS7 to ETS 300, GIII supports the mapping of an ETS 300 call transfer request to an SS7 call redirection. Should such a request be processed, Call redirection address will contain the dialled number to which the incoming SS7 call has been redirected. The Incoming enquiry call port and Incoming enquiry call timeslot will contain the port and timeslot numbers used by the outgoing ETS 300 call to make the call transfer request. If no call transfer request has been received these fields will be blank. ETS 300 call transfer to SS7 call diversion mapping is described in the Call Transfer section.
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Field Description Recovered call This is a single character field. Y indicates that one or
more SIP recovery calls have been made and successfully connected, N indicates that no SIP recovery calls have been connected. See the SIP call recovery section for more information.
Total charging units This is the total number of Advice of Charge (AOC) units mapped from the outgoing call leg to the incoming call leg. AOC units are specific to the signalling protocols in use. AOC mapping is described in Call Handling part of the Trunks configuration section.
Re-routing count This is the number of times a rejected outgoing call was re-routed before either a successful outgoing call was made, or the call was rejected back to the calling party. See the Call re-routing section for more information.
Connected address This is the connected party number mapped from the outgoing to the incoming call leg at the connected stage.
CDR call duration Call duration will always be correct. GIII uses UTC only.
9.3.1 Generating a CDR report To generate a CDR report, go to the CDR Query tab on the Administer CDRs page. Select the CDR query fields you want to include in your report, by dragging them from CDR fields available to CDR query fields. Select the other required query options, and click submit. Time range – The time range to search within for the query Start (dd/mm/yy hh:mm:ss) – The start date and time for the CDR query End (dd/mm/yy hh:mm:ss) – The end date and time for the CDR query Search on – Find records within the time range based on the call start time or call end time
Node – Select whether to run the query on one specific node, or all nodes in cluster Originating address – Find records with the specified incoming originating address. Match all, if blank. Destination address – Find records with the specified incoming destination address. Match all, if blank. Raw cause code – Find records with the specified Raw cause code. Generic cause code – Match records with the specified Generic cause code. Termination reason – Find records that match the specified termination reason. First record – Specify the first record to include in the report, skipping any previous records. Max records – Specify the maximum number of records to include in the report. To download the CDR report, go to the CDR Query Files tab on the Administer CDRs page.
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Figure 9-14 CDR Query Files
9.4 Configure NTP NTP is required by GIII. It is not only used for log files, but also is a key component to clustering. Due to this requirement, NTP is enabled by default with 4 servers pre-configured. Administration of the NTP servers can be done by going to Cluster Administration – Configure NTP.
NOTE Due to GIII’s requirement for NTP, If all NTP servers are deleted from the list, the 4 default servers will be re-instated.
Figure 9-15 Configure NTP
You can check whether a GIII node is synchronized with NTP by going to Status Monitors – NTP.
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Figure 9-16 NTP status page
If NTP is not synchronized then an alarm will be show on the Status Monitors – System Summary page.
Figure 9-17 NTP not synchronized alarm
9.4.1 Add NTP server To add a new NTP server, go to Cluster Administration – Configure NTP, and click the New server button. Enter the FQDN or IP address of the NTP server, and click the submit button.
9.4.2 Delete NTP server To delete an NTP server, go to Cluster Administration – Configure NTP, click the Delete button next to the NTP server you wish to delete, and click the Submit button to commit the deletion.
Figure 9-18 Delete NTP server
9.5 Configure SNMP Agent To configure the SNMP Agent, go to Cluster Configuration – Configure SNMP Agent. This page provides a number of tabs which are used to configure SNMP settings within GIII. The agent supports SNMPv3.
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9.5.1 General The General tab on the Configure SNMP page allows agent information to be added.
Figure 9-19 SNMP agent configuration page
The General tab allows the MIB-2 System group items to be configured: System location, System contact, System name, System description, and System services. The SNMPv3 EngineID, if configured, will show the Full EngineID. The V1 Trap Address is only used for SNMPv1 traps. If configured, the value will be inserted into SNMPv1 traps. This option is useful mainly when the agent is visible from outside world by specific address only (e.g. because of network address translation or firewall). The Send authentication trap checkbox, if checked, will generate authentication failure traps. There are two download button items: Aculab MIB, and GIII MIB, which if clicked will download the relevant MIB file. These can then be incorporated into the SNMP client.
9.5.2 Users The SNMP users are configured from the Users tab. These are used for SNMPv3 only, and are required to be configured before Access items are configured. The page will list any existing users. To add a user, drag the Drag for new User item into the box below, and a page will be presented to enter details of a new user.
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Figure 9-20 viewing SNMP agent users
Figure 9-21 SNMP agent user configuration page
The UserPrivProtocol and UserPrivPassword are optional. Name The name of the user. AuthProtocol MD5 or SHA from drop down select. AuthProtocolPassword The authentication password for the user, minimum 8 characters. UserPrivProtocol DES or AES from drop down select. UserPrivPassword The privacy password for user.
9.5.3 Communities The Communities tab will list the configured SNMP communities. These are used for SNMPv1 and SNMPv2c.
Figure 9-22 viewing SNMP agent communities
To add a new community, drag the Drag for new Community item into the box below and release, this will then display a page to configure a new SNMP community.
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Figure 9-23 SNMP agent community configuration page
An SNMP community requires a Name and an Access level, the Source is optional. Name The name of the SNMP community Access ro or rw, selected from drop down Source The allowed hostname, IP address, or IP network
To delete a community, drag the listed item over the trash can symbol, and once the lid show removed, release the community item into the trash.
NOTE The Submit button must be pressed to actually delete the item.
Figure 9-24 Deleting a community by dragging to trash
9.5.4 Hosts The Hosts tab will list the hosts configured to send SNMP traps to.
Figure 9-25 viewing SNMP agent hosts
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9.5.4.1 SNMPv1 and SNMPv2c Traps
Figure 9-26 SNMP agent host configuration page
Host The hostname or IP address of the trap destination Community The SNMP community, a selectable item from the configured
communities Version The SNMP version of the Trap notification. NOTE: selecting
SNMPv3 will remove the Community field and add a User Access field.
9.5.4.2 SNMPv3 Traps To configure an SNMPv3 Trap you will need to have first added User Access items using the Access tab.
Figure 9-27 configuring a SNMP agent host for SNMPv3
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9.5.5 Access The Access tab list configured SNMPv3 User Access items.
Figure 9-28 viewing SNMP agent SNMPv3 access users
Figure 9-29 Configuring SNMP agent SNMPv3 user access
User A user selected from configured users Access ro or rw SecurityLevel selected from noauth, auth, or priv
9.5.6 Configuring SNMP on GIII Depending upon whether SNMPv1/SNMPv2c, or SNMPv3 is being configured requires different configuration items in the SNMP Agent.
9.5.6.1 Configuration for SNMPv1/SNMPv2c • From the General tab configure V1 Trap Address and Send authentication trap as
required
• From the Communities tab add any required communities
• From the Hosts tab, add any trap destinations, selecting the Community to be used.
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9.5.6.2 Configuration for SNMPv3 • From the General tab configure EngineID if required
• From the Users tab add any required users, selecting the required AuthProtocol and UserAuthProtocolPassword, and optionally the UserPrivProtocol and UserPrivPassword
• From the Access tab, add user access mappings.
• From the Hosts tab, add any trap destinations, selecting Version as SNMPv3 to be able to configure the user.
9.6 Configure Syslog GIII can be configured to send logging message to one or more syslog servers. The syslog server settings are cluster-wide, meaning the same settings will be provided to all nodes within a cluster. To configure syslog settings, go to Cluster Administration – Configure Syslog.
Figure 9-30 Configure syslog
9.6.1 Add Syslog Server To add a syslog server, select the Add button and a page will be displayed allowing the client information to be added.
Figure 9-31 Add Syslog server
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Enter the following information, and click the Submit button: Server - The IP address Port - The Port number to use Protocol - UDP, TCP, or RELP, selected from drop down list.
NOTE If RELP is selected, then the port number is changed to the default RELP port number of 2514.
9.6.2 Delete Syslog Server To delete a Syslog server, select the server, and click the Delete button. Click Submit to confirm deletion.
Figure 9-32 Delete Syslog Server
9.7 Configure LDAP It is possible to configure GIII to use an LDAP server to authenticate and authorize user login to the GIII. To configure LDAP, go to Cluster Administration – Configure LDAP. The page will show two tabs; LDAP and Test LDAP. Examples of different LDAP server configurations can be seen in Appendix K:LDAP Authentication examples.
NOTE A user login using LDAP that is authorized as an Admin will have the same access rights as the local Admin users, apart from viewing or editing local user accounts.
9.7.1 LDAP The LDAP tab is split into four panels; Server, Search, Group Mapping, and Login Page.
NOTE There is a small amount of validation done on some of the configuration items, but in general any errors on input would be picked up when using the Test LDAP tab, i.e. a bad search filter or DN.
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9.7.1.1 Server
Figure 9-33 Server Configuration
Enable Enable or disable LDAP Authentication. Use DN for group membership If not using memberOf functionality, then
when searching for group membership this will determine whether the user’s DN is used to determine membership, otherwise the Group name attribute is used.
Bind type Select which type of bind is used. Anonymous means no user or password, Unauthenticated means just user name, and Authenticated will use both the user name and password
Server There are three servers that can be configured. There is no guarantee to which one is used first. The server format is entered as <server>:<port>. For IPv6 use [<server>]:<port>.
Encryption The type of encryption to use. If TLS or StartTLS is used and not certificate is uploaded the GIII will use an internal certificate for the encrypted connection.
Certificate A certificate can be uploaded to be used for TLS/StartTLS connections.
Validate Certificate This determines whether the GIII ldap client will validate the certificate when connecting.
Timeout The timeout. Bind DN The user DN used to bind to the server and perform searches
for user and group information. Bind password The password used for Authenticated bind. Base DN The base DN to start searches from.
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9.7.1.2 Search The search panel provides configuration of items used to build search filters for searching for the user’s DN, and the user’s group membership.
Figure 9-34 User and group search configuration
LDAP Filter The basic filter to use for searching for users, additional items
are added to this filter depending upon configuration. Login attribute The attribute used to search for the user login name User’s group membership attribute If memberOf like functionality is
available then this attribute will be used to find the user’s membership from the user’s account information. Active Directory supports memberOf by default, and other LDAP servers may have the ability to provide this as an add-on.
Group membership attribute The attribute that describes a user as a member of a group
Group name attribute If the group does not use the distinguished name for the user membership, then configure this item with the attribute used, i.e. 389DS uses the common name (CN) by default.
Group filter The filter used to search for groups Search scope This will limit the scope of search. Chase referrals If referrals are present in searches, then this can be enabled to
chase for the actual information. Use Active Directory nested group membership Active directory supports
query on nested group membership using: LDAP_MATCHING_RULE_IN_CHAIN: '1.2.840.113556.1.4.1941'. This is used in a filter, i.e. (&(member:1.2.840.113556.1.4.1941:={user-DN}))
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Group Mappings Once a user login has passed the authentication phase, i.e. it can bind to the LDAP server using its name and password, and then the authorization phase will get a list of groups that the user is a member of, and will compare these to a list of groups that represent either admin or monitor level in the GIII.
Figure 9-35 Admin and monitor group mapping
9.7.1.3 Login Page
Figure 9-36 Login page configuration
The login page panel allows you to configure information to be displayed to the user at the login page. You can enter HTML in the login page HTML area, or a URI in the login page URI.
NOTE The login page URI may not work due to CORS restrictions.
If the login page HTML contains configuration, then this will be used regardless of the login page URI configuration. The Preview button will display the configured information in a dialog box.
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9.7.2 Test LDAP
Figure 9-37 LDAP Testing
Once the LDAP configuration is completed you can test these settings from the Test LDAP tab.
Figure 9-38 Example of testing LDAP login
9.8 Backup & Restore 9.8.1 Backup
GIII allows either an individual role, or the system configuration, to be backed up. This is also possible using the web services API.
9.8.1.1 Role Backup To backup a role, go to Cluster Administration – Backup & Restore, and select the Role tab. Select the role required for backup, and press the Submit button. This will download a role backup file to your browser system. A password can be optionally added to the file, by filling in the File password option. No password will be set, if the File password is left blank.
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Figure 9-39 Role backup
9.8.1.2 System Backup To backup the system configuration, go to Cluster Administration – Backup & Restore, and select the System tab, and press the Submit button. This will download a system backup file to your browser system. A password can be optionally added to the file, by filling in the File password option. No password will be set, if the File password is left blank.
Figure 9-40 System backup
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9.8.2 Restore GIII allows either a role, or the system configuration, to be restored. This is also possible using the web services API.
9.8.2.1 Role Restore To restore a role, go to Cluster Administration – Backup & Restore, select the Restore accordion, and select the Role tab. Click Choose File, browse to your role backup file, and click open. Enter the File password if required, and set the Destination Role name, if you wish the role name to be different from the name set against the file. When you press the submit button, the role will be restored to the node.
Figure 9-41 Role restore
9.8.2.2 System Restore To perform a system restore, go to Cluster Administration – Backup & Restore, select the Restore accordion, and select the System tab. Click Choose File, browse to your role backup file, and click open. Enter the File password if required, and press the submit button. The system backup will be restored to the node, and the node will be restarted.
Figure 9-42 System restore
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9.9 Restart & Shutdown
Figure 9-43 Restart & Shutdown page
9.9.1 Restart
Figure 9-44 Selecting a node to be restarted
To restart a node in a cluster, select the Restart tab, select the node and press the Submit button.
9.9.2 Shutdown
Figure 9-45 Selecting a node to be shutdown
To shutdown a node in a cluster, select the Shutdown tab, select the node and press the Submit button.
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10 Call Transfer
10.1 Call transfer mapping in GIII GIII is able to map call transfer requests from the called side through to the calling side. At present only the mapping of ETS 300 call transfer requests to SS7 call redirection requests is supported.
10.2 ETS 300 Call Transfer to SS7 Call Redirection Mapping 10.2.1 Overview
When performing protocol conversions in the direction SS7 to ETS 300, GIII will automatically map an explicit call transfer request received from the ETS 300 side into a call redirection request on the SS7 side. If an ETS 300 call transfer request is received when the incoming call leg is not SS7, then the request will be rejected.
10.2.2 Configuration
NOTE The –cFF firmware switch must be applied to all ETS 300 ports that will be involved in call transfer operations. This is the Activate Facility Information Support option.
The called party will initiate a transfer by making a new enquiry call into GIII, and this call must be made on a separate timeslot from the original call. A separate route is configured to receive this call, where the destination group is set to the system entry of <Wait for Transfer>, as illustrated below.
Figure 10-1 Wait for Transfer Routing Configuration
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When a call is received on this route, no outgoing call will be made. The call will be progressed to the ringing stage, and will remain in this state until cleared down. Whilst in this state the call will respond to a Link ID request, which the original called party should return to the original outgoing ETS 300 call in a call transfer execute request. At this point GIII will:
• Clear down the original ETS 300 call leg in an appropriate fashion,
• Clear down the ETS 300 enquiry call in an appropriate fashion,
• Redirect the SS7 call by releasing it with a fixed cause value of 23. The same incoming group can be used, in separate routes, to receive both call transfer requests and normal incoming calls as long as the calls can be differentiated during route selection, for example by using a Destination Address mask.
10.2.3 CDR generation The following CDR query fields may be used to report the call transfer mapping Call redirection address – this is the dialled number to which the original call has been redirected. Enquiry call port – this is the number of the port on which the enquiry call requesting the transfer was received. Enquiry call timeslot – this is the timeslot on the above port used by the enquiry call.
10.3 CAMA to SIP Call Transfer Mapping NOTE
The –s26 firmware switch must be applied to all CAMA ports that will generate hookflash call transfer requests. The recommended setting is –s26,30.
When performing protocol conversions in the direction SIP to CAMA, GIII can be configured to map a hookflash call transfer request from the CAMA side to the SIP side. The following procedure is invoked if a hookflash is detected on the CAMA side whilst the call is in the connected state:
• The hookflash is mapped to the SIP side using the RFC 2833 event 16 (Flash).
• The US dial tone is played to the CAMA side.
• When the first DTMF dialled digit is detected on the CAMA side, the dial tone will be discontinued. This digit and all subsequent DTMF dialled digits will be mapped to the SIP side using RFC 2833 digits.
CAMA to SIP call transfer mapping is configured via the CAS tab on a route in Role Configuration – Routing – Routes tab, and must be enabled separately for each route on which mapping is required. The configuration procedure is:
• On the CAS tab select the CAMA to SIP Call Transfer Mapping checkbox.
• On the Incoming SIP calls tab on a route under Role Configuration – Routing – Routes page, enable RFC 2833 event generation by setting the DTMF digit detection control to either Immediate Detection or 40ms.
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10.4 SIP to CAMA Call Transfer Mapping NOTE
The –s25 firmware switch must be applied to all CAMA ports to which hookflash call transfer requests will be mapped. The recommended setting is –s25,15.
When performing protocol conversions in the direction CAMA to SIP, GIII can be configured to map an RFC 2833 hookflash (event 16) call transfer request from the SIP side to the CAMA side. The following procedure is invoked if a hookflash is detected on the SIP side whilst the call is in the connected state:
• The hookflash is mapped to the CAMA side.
• The US dial tone is played to the SIP side.
• When the first DTMF dialled digit is detected on the SIP side, the dial tone will be discontinued. This digit and all subsequent DTMF dialled digits will be mapped to the CAMA side in the normal fashion.
SIP to CAMA call transfer mapping is configured via the CAS tab on a route in Role Configuration – Routing – Routes tab, and must be enabled separately for each route on which mapping is required. The configuration procedure is:
• On the CAS tab select the CAMA to SIP Call Transfer Mapping.
• On the Outgoing SIP calls tab on a route under Role Configuration – Routing – Routes page, enable RFC 2833 event generation by setting the DTMF Digit Detection control to either Immediate Detection or 40ms.
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11 SIP telephony
11.1 IPv4 and IPv6 telephony SIP telephony supports the use of IPv4 and/or IPv6 addresses, with both IPv4 calls and IPv6 calls being handled at the same time. GIII must be configured for IPv4 and/or IPv6 telephony as described in the Admin IP configuration section. The call handling procedures applied by systems configured to use both IPv4 and IPv6 are described in the following sections.
11.1.1 Incoming call handling The following procedure will be applied when handling incoming SIP calls:
• When the incoming INVITE is accompanied by an SDP, GIII will select a media stream that matches the signalling address whenever possible. For example an IPv4 media stream will be selected for a call signalled using IPv4. If this is not possible then any suitable media stream will be chosen which may result in, for example, a call signalled using IPv4 selecting an IPv6 media stream.
• When the incoming INVITE does not present an SDP, GIII will initiate media negotiation at an appropriate point:
o If the incoming call was signalled using an IPv4 address GIII will present an IPv4 media offer.
o If the incoming call was signalled using an IPv6 address GIII will present an IPv6 media offer.
11.1.2 Outgoing call handling When making an outgoing call GIII will always match the media offer to the signalling type:
• When directing a call to an IPv4 endpoint, the INVITE will be accompanied by an SDP offering IPv4 media only.
• When directing a call to an IPv6 endpoint, the INVITE will be accompanied by an SDP offering IPv6 media only.
It is not possible to direct a call to an IPv4 endpoint and offer IPv6 media, or vice versa.
11.1.3 Call retry and recovery When a SIP call is retried or recovered GIII may direct the replacement call to a different type of endpoint (for example, the original call might have been directed to an IPv4 endpoint whilst the replacement call is directed to an IPv6 endpoint). In this situation a new SDP will be created following the rules in the previous section.
11.2 SIP security GIII is able to make secure SIP calls using Transport Layer Security (TLS) for call control signalling and Secure RTP (SRTP) for the media stream.
11.2.1 Configuration GIII will not be able to make or receive secure SIP calls until the appropriate RSA certificate files have been loaded.
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NOTE It is the responsibility of the customer to provide the appropriate certificate files.
Once the appropriate certificate files have been loaded both secure and non-secure incoming calls will be handled automatically. Outgoing calls must be explicitly configured to use SIP security on a route-by-route basis. The G.726 codec cannot be used with secure SIP calls.
11.2.2 TLS TLS negotiation will be conducted on port 5061. Incoming secure calls may be identified using either the sips: prefix in the To: header (for example To: <sips:[email protected]>), or by the transport=tls parameter in the Contact: header (for example Contact: <sip:[email protected];transport=tls>).
If the TLS layer refuses an incoming SIP call this will NOT be reported by the GIII application.
11.2.3 SRTP Support for Secure RTP is in accordance with IETF RFC 4568.
Media transport type Incoming calls using the RTP/SAVP transport type will be negotiated in accordance with IETF RFC 4568.
NOTE In some instance incoming calls may arrive that request SRTP but use the RTP/AVP transport type. Such calls are outside of specification. GIII will attempt to negotiate a secure media stream on such occasions, although success cannot be guaranteed.
All outgoing secure calls will use the RTP/SAVP transport type.
Crypto suites and key parameters The following Crypto-Suites are supported, and will be offered in the order listed
AES_CM_128_HMAC_SHA1_80
AES_CM_128_HMAC_SHA1_32
F8_128_HMAC_SHA1_80
Key parameters Each crypto suite offered will use a different key, and new keys will be generated for each call. GIII does not support the use of multiple keys, and only a single inline key is permitted. GIII does not support the key lifetime parameter. It will not be present in a media offer, and will be ignored if received in a media proposal. GIII does not support the master key identifier (MKI) parameter. It will not be present in a media offer, and will be ignored if received in a media proposal.
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Session parameters GIII will not include any session parameters in a media offer. Session parameters included in an offer from a remote endpoint will be processed in the following manner:
KDR – This parameter is not supported by GIII and will be ignored.
UNENCRYPTED_SRTCP – GIII does not support RTCP processing. If this parameter is offered it will be included in the media answer.
UNENCRYPTED_RTP – GIII will use unencrypted RTP.
UNAUTHENTICATED_RTP – GIII will use unauthenticated RTP.
FEC_ORDER – This parameter is not supported by GIII and will be ignored.
FEC_KEY – This parameter is not supported by GIII and will be ignored.
WSH – This parameter is not supported by GIII and will be ignored.
11.3 SIP call recovery SIP call recovery is a customer specified feature. Its purpose is to identify when an outgoing SIP call leg has failed and re-establish the call automatically via an alternative gateway. This feature is independent of the protocol used by the incoming call leg. Call recovery is only available when outgoing SIP calls are being routed to a gateway. It is configured on a system wide basis, and once enabled all new outgoing SIP calls will be monitored.
NOTE Call recovery can only be used when a gateway list containing two or more gateways is in operation.
Call failure is detected by monitoring the RTP stream received from the remote endpoint, and monitoring begins when the outgoing SIP call reaches connected state. Call recovery cannot be used with a remote endpoint that is performing discontinuous transmission (DTX).
NOTE When a SIP call is placed on hold by the remote endpoint, RTP monitoring is suspended. RTP monitoring is resumed when the call is taken off hold.
When call failure is detected the failed SIP call leg is released and a replacement call made. The call attributes (for example called party number, calling party number) of the failed call will be used when making the recovery call. The following gateways will be excluded from the search for an alternative destination:
• The gateway to which the failed call was directed,
• Any gateway that is currently marked as unavailable,
• Any gateway whose call capacity has been reached.
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Failed call retry will be applied to the recovery call if enabled. Once the recovery call has reached the connected state it may be recovered a second and subsequent time. Call recovery is reported using the Call Control filter, and will be reported using the Call ID of the failed call leg. A SIP custom header is used to identify a recovery call. The SIP Custom headers Appendix contains more information.
11.4 SIP call hold When a SIP call is in the connected state GIII will acknowledge a call hold request from a remote endpoint and discontinue the transmitted RTP stream. The received RTP stream will continue to be processed to accommodate features such as music on hold. To place a call on hold the remote endpoint should send a REINVITE with an SDP whose media description contains the sendonly or inactive attribute. To remove the call from hold the remote endpoint should send a further REINVITE with an SDP whose media description contains the sendrecv or recvonly attribute. Whilst a call is on-hold, any REINVITE received that is not an off-hold instruction will be processed in the normal way. GIII is resilient to changes in the session ID of a call whilst it is on-hold. A call will not be removed from hold unless the session ID of the off-hold instruction matches that of the on-hold instruction.
NOTE GIII can only place a call on hold in response to a request from the remote endpoint. GIII cannot make a call hold request.
11.5 SIP call redirection GIII will redirect a call when an outgoing INVITE is cleared with a 300, 301, 302 or 305 response before reaching the connected stage. The INVITE response must present a Contact header, as described in RFC 3261 section 20.10, with the URI containing both a DDI and an IP Address, for example Contact: <sip:[email protected]>.
If the IP address in the Contact header matches the GIII traffic IP address, then the DDI will be extracted and the call will be passed back through the routing table to establish the new destination. All other routing parameters will be taken from the original incoming call. This allows an outgoing SIP call to be redirected to the TDM network. If the IP address in the Contact header does not match the GIII traffic IP address, then a new SIP call will be made to the destination specified. All other call parameters will be those used for the original outgoing call. If GIII is configured to use a SIP proxy server for outgoing calls then the new call will be directed to that SIP proxy server for onward routing. The 300 – Multiple Choices response may present multiple Contact headers to facilitate analysis and selection. No such analysis and selection will be carried out, and the first Contact header encountered in the response will be used.
The requirement that local directories should be updated on receipt of a 301 – Moved Permanently response is not supported.
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All other 3xx responses (including specifically 380 – Alternative Service) will be treated as rejected calls and will not be redirected.
11.6 Functional Number mapping GIII supports the mapping of a Functional Number between an ITU-T SS7 call leg and a SIP call leg by means of the Aculab-Functional-Number header.
If an ITU-T SS7 call presents a Functional Number in the User-to-user information parameter of the incoming IAM, the number will be placed unmodified into an Aculab-Functional-Number header in the outgoing SIP INVITE.
If a SIP call presents an Aculab-Functional-Number header in the incoming SIP INVITE, the number will be placed unmodified into a User-to-user information parameter in the outgoing ITU-T SS7 IAM. The presentation of a functional number in the User-to-user information parameter of an SS7 call is described in section 5.1 of ETSI TS 102 610 V1.1.0 (2008-01), whilst the format of the Aculab-Functional-Number header is described in the SIP custom headers Appendix. To enable Functional Number mapping refer to Protocol Settings in the SIP trunk section of this document.
11.7 ISDN subaddress mapping GIII supports mapping of called, calling and connected party ISDN subaddresses using the isub URI parameter. The isub parameter is described in IETF RFC 3966, and is extended by the addition of the isub-encoding parameter described in IETF RFC 4715. RFC 4715 does not support the user specified subaddress type, and a private extension of has been implemented to accommodate these. When transporting a user defined subaddress the isub-encoding parameter is set to acu-user. The first character in the isub parameter must be 0 or 1 to indicate the setting of the odd/even indicator bit in octet 3, with the remaining characters being hexadecimal digit pairs containing the settings of octet 4 onwards. This is illustrated in the following diagram using the calling party subaddress as an example. The called and connected party subaddresses are coded in an identical way.
Figure 11-1 User defined ISDN subaddress extension
11.8 Interworking with TDM protocols 11.8.1 Interworking between SIP and SS7
Interworking between SIP and SS7 is carried out as described in the following specifications.
Bits Octet 8 7 6 5 4 3 2 1
1 0 1 1 0 1 1 0 1 2 0 0 0 0 0 1 0 0 3 1 0 1 0 1 0 0 0 4 0 0 0 1 0 0 1 0 5 0 0 1 1 1 0 1 0 6 0 1 0 0 1 0 1 1
isub=1123A4B;isub-encoding=acu-user
Calling party subaddress IE
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SS7 variant Interworking specification ANSI (1995 and 2005 variants)
ANSI T1.769-2004. The NENA i3 modification may be applied using the configuration option in the Protocol Settings screen, the SIP configuration section refers.
ITU-T ITU-T Recommendation Q.1912.5. China ITU-T Recommendation Q.1912.5. UK ISUP NICC Document ND1017:2006/07
11.8.2 Interworking between SIP and ETS 300 Interworking between SIP and ETS 300 is based upon ETSI TS 183 036. This includes:
• Mapping of the calling party number to/from the P-Asserted-Identity header, including support for the Privacy header.
• Mapping of the additional calling party number (when present).
• Mapping of clearing causes to/from the Reason header.
11.8.3 Interworking between SIP and CAMA Interworking between SIP and CAMA is described in the CAMA telephony section of this document.
11.9 Additional IETF RFC support GIII supports the production and/or interpretation of optional headers specified in the following IETF RFCs.
RFC 3891 The Session Initiation Protocol (SIP) “Replaces” Header. RFC 4244 Request History Information, see the SIP Configuration section for
further information. RFC 4412 SIP Resource Priority, see the Interworking supplementary services
section of this document for further information. RFC 4715 The Integrated Services Digital Network (ISDN) Subaddress Encoding
Type for tel URI. RFC 4715 has been extended by Aculab, see the ISDN subaddress mapping part in the SIP telephony section of this document for details. RFC 4715 carries implied support for the isub parameter described in RFC 3966.
RFC 4904 Representing Trunk Groups in tel/sip URIs, see the SIP Configuration section for further information.
When interworking with ANSI SS7, GIII also supports the P-Charge-Info header as described in IETF Internet Draft draft-york-sipping-p-charge-info-08. Information is automatically mapped between the P-Charge-Info header and the SS7 Charge Number parameter. GIII also uses the Remote-Party-ID header described in IETF Internet Draft draft-ietf-sip-privacy-04 to carry name identification information. See the Name identification supplementary services (CNIP/CONP) part of the Interworking supplementary section of this document for further information. GIII supports the transfer of user-to-user information with the User-to-User header specified in IETF Internet Draft draft-ietf-cuss-sip-uui-09. See the User to user signalling (UUS) part of the Interworking supplementary section of this document for further information.
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12 SS7 telephony If you are using SS7 on your GIII, you will need to configure:
• The SS7 links using SS7 page from the Role Configuration menu. Adding point code configurations will add configuration to the SS7 stack file, which can be viewed (and edited if necessary) using the Advanced Configuration tab.
• The SS7 link protocol firmware timeslot usage. This is done using the Trunks page from the Role Configuration menu, and selecting the appropriate trunk.
NOTE The SS7 configuration on GIII using the options does not offer, or require, the destination point codes to be specified. These can be determined by the SS7 driver.
NOTE The SS7 stack file can be modified, as described below, using the Advanced Configuration tab on the Role Configuration - SS7 page.
Making SS7 protocol selections for SS7 Signalling and ISUP Bearer timeslot usage is covered in the Trunks configuration section of this document.
12.1 SS7 stack file The SS7 stack file is created from the GIII Role Configuration – SS7 page. It may not be necessary to edit the stack file directly, but if required, it can be viewed/edited from the Advanced Configuration tab in Role Configuration – SS7 page.
Many of the entries in the SS7 stack file replace firmware switches that were previously present, and as such the stack file can become very complicated. In most cases however, only a very simple stack file is required. This section shows the stack files that would be required for the most popular GIII SS7 configurations. Should you need to write a stack file for a different configuration, or need to use additional parameters, contact Aculab support for assistance. A stack file contains one [SP] section for each unique OPC/DPC pair
[SP] LocalPC=aaaaa NI=x [ISUP] variant=vvv [EndISUP] [MTP3] variant=vvv [DESTINATION] RemotePC = bbbbb [EndDESTINATION] [EndMTP3] [EndSP]
There are two mandatory and two optional parameters.
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Mandatory LocalPC=aaaaa specifies the point code used by GIII for the particular
signalling link. RemotePC=bbbbb specifies the point code of the remote SP or STP to which
GIII is physically connected.
Optional NI=x This is the SS7 network indicator, and will be in the range
0 to 3. The value to be used in this parameter must be obtained from you service provider, and in most instances will be either:
0 - International network
2 - National network
If this parameter is omitted the default of NI=0 will be applied.
variant=vvv This is the protocol variant used by ISUP and MTP3, and must be specified separately in each section. The combination of values currently supported is shown in the following table.
[ISUP] section [MTP3] section ITU_UKISUP_2001 ITU
ITU ITU
CHINA CHINA
ANSI ANSI
ANSI_2005 ANSI
If this parameter is omitted the default of variant=ITU will be applied.
Unsupported parameters GIII uses Aculab’s SS7 stack, and does not support use of the following SS7 stack file parameters. default_cpc=n This parameter specifies the value that will be used in
the Calling party’s category parameter if no setting is provided by application program. The GIII application always sets the Calling party’s category and so using this parameter will have no effect.
auto_inf=y This setting instructs the SS7 driver to respond to an INR with an INF. The GIII application will respond to all INR messages. The behaviour if this parameter is included is undefined.
report_non_q931_cause=n This setting instructs the SS7 driver to convert any non ITU-T standardised causes to ITU-T cause 127 (Interworking, unspecified). This will prevent GIII from mapping and reporting non ITU-T standardised causes correctly.
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continuity_defer_event=y This setting prevents the GIII application from being notified of in-call continuity checks until they have been completed. If this parameter is included continuity checks will not be reported to the call control log.
apply_continuity_loop=y This setting instructs the SS7 driver to apply a voice path loop to incoming continuity checks. This may override the switching carried out by the GIII application and cause the continuity check to fail.
continuity_check_output_value This setting is associated with the use of apply_continuity_loop. The behaviour should this parameter be set to a value other than none is undefined.
ccr_application=n This setting instructs the SS7 driver not to inform the GIII application when an inbound CCR is received. If the parameter is included, the GIII application will be unable to process inbound CCR messages.
ccr_auto_lpa=y This setting is only applicable to ANSI SS7, and instructs the SS7 driver to automatically send an LPA upon receipt of a CCR. The GIII application will send an LPA in response to a CCR. The behaviour if this parameter is included is undefined.
System X interworking Some versions of System X exhibit problems if large numbers of ISUP circuit related housekeeping messages are received simultaneously. GIII can be configured to throttle delivery of such messages by using the system_x_throttle_rate and system_x_throttle_limit parameters in the [ISUP] section of the stack.
When throttling is enabled the delivery rate of the following messages will be restricted
UPT User part test BLO Blocking UBL Unblocking CGB Circuit group blocking CGU Circuit group unblocking RSC Reset circuit GRS Circuit group reset
Throttling is implemented by accumulating credits into a buffer at the rate of one per millisecond. The maximum value of this buffer (which is also the initial value) is configured using the system_x_throttle_limit parameter. A message credit value is configured using the system_x_throttle_rate parameter. Each time a throttled message is dispatched the credit buffer is decremented by the message credit value. The credit buffer is checked continuously and a message dispatched whenever its value is zero or greater. Sending a message when the credit buffer has a value of zero will cause the buffer to be decremented to a negative value, and no further messages will be sent until the buffer has been incremented back to zero.
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The following example will send five messages immediately upon startup (one each at 200, 150, 100, 50 and zero), or after a period of idle, and thereafter one message every 50 milliseconds.
[SP] ... [ISUP]
variant=ITU_UKISUP_2001 system_x_throttle_rate=50 system_x_throttle_limit=200
[EndISUP] ...
[EndSP]
Using the settings system_x_throttle_rate=50 and system_x_throttle_limit=0 will avoid a burst of messages at startup/idle, and will dispatch each message at 50ms intervals. Removing the system_x_throttle_rate and system_x_throttle_limit parameters from a stack will disable throttling.
Disabling group messages Some SS7 switches can be configured to refuse group reset and/or blocking messages. GIII can be configured to send a series of individual circuit messages in place of a group message in such circumstances by using the send_group_reset_messages and send_group_block_messages parameters. The following example illustrates how to use these parameters.
[SP] ... [ISUP]
send_group_reset_messages=never [DESTINATION]
send_group_block_messages=never RemotePC=7070
[EndDESTINATION] [EndISUP] ...
[EndSP]
Placing the parameter in the [ISUP] section will cause it to be applied to all destination point codes in that section, whether implicitly or explicitly declared. Placing the parameter inside a [DESTINATION] section will apply the setting to that destination point code only. The send_group_reset_messages parameter will replace a GRS with a series of RSC messages. The send_group_block_messages parameter will replace a CGB with a series of BLO, and a CGU with a series of UBL. Hardware blocking (ITU-T/UK ISUP/China) and blocking with immediate release (ANSI) must be signalled using a CGB, even when only a single circuit is being blocked. The procedure applied here is for the CGB to be replaced by an appropriate series of BLO, and on any circuit carrying an active call the BLO will be preceded by an RSC.
12.1.1 Example configuration files In this section you will find some example SS7 stack configuration files for standard setups. If you require assistance creating a configuration file for your particular setup, please contact Aculab support.
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Example 1 GIII uses one link (one or more trunks) to connect to a Signalling Point (SP) or Signalling Transfer Point (STP)
Figure 12-1 Single SS7 link option
The stack file would be [SP] LocalPC=2065 NI=0 [ISUP] variant=ITU_UKISUP_2001 [EndISUP] [MTP3] variant=ITU [DESTINATION] RemotePC = 7734 [EndDESTINATION] [EndMTP3] [EndSP]
Example 2 GIII uses two links (each with one or more trunks) to connect to a two different Signalling (Transfer) Points, using a different OPC’s on each link. The system is configured as a signalling gateway.
Figure 12-2 Two SS7 links option
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The stack file would be [SP] LocalPC=986985 NI=0 [ISUP] variant=CHINA [EndISUP] [MTP3] variant=CHINA [DESTINATION] RemotePC = 7368816 [EndDESTINATION] [EndMTP3] [EndSP] [SP] LocalPC=7244 NI=2 [ISUP] variant=ITU [EndISUP] [MTP3] variant=ITU [DESTINATION] RemotePC = 5090 [EndDESTINATION] [EndMTP3] [EndSP]
Example 3 GIII uses a single link (one or more trunks) in a loopback configuration for testing purposes.
Figure 12-3 Single SS7 link looped back option
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The stack file would be [SP] LocalPC=4242 NI=2 [ISUP] variant=ANSI [EndISUP] [MTP3] variant=ANSI [DESTINATION] RemotePC = 1077 [EndDESTINATION] [EndMTP3] [EndSP] [SP] LocalPC=1077 NI=2 [ISUP] variant=ANSI [EndISUP] [MTP3] variant=ANSI [DESTINATION] RemotePC = 4242 [EndDESTINATION] [EndMTP3] [EndSP]
12.2 SS7 trunk configuration It is possible to configure non-provisioned SS7 trunks that carry neither signalling links nor bearer channels. To create a non-provisioned trunk configure a trunk as SS7 E1/T1 Bearers and/or Signalling and create a revision with these settings in the role, and select this revision. These trunks can subsequently be partly, or fully provisioned, by adding signalling links and/or bearer channels and selecting a created revision of these changes.
NOTE When configuring SS7 ports, care must be taken to ensure that the bearer channels at each end of the trunk are identically configured, as mismatching configurations will prevent the circuit group reset (GRS) from completing. GIII cannot detect this condition, which may result in incorrect reporting by the GIII Call Traffic page and/or attempts to place calls on bearer channels that are not in service.
12.3 Continuity checking GIII supports inbound and outbound continuity checks when using the ITU-T and ANSI variants. Both 2-wire and 4-wire continuity checks are supported. The type of continuity check to be used by each trunk must be selected in the Role Configuration - Trunks page. The default action is to use a 4-wire (Loopback) continuity check. The tones used for continuity checking depend on the variant of SS7 being used by a circuit. The table below shows the tones used by GIII.
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Variant and continuity check type
Outbound IAM or CCR Inbound IAM or CCR
ITU-T/4-wire (Loopback)
Tone to send: 2000Hz Tone to detect: 2000Hz
Voice path loopback applied.
ITU-T/2-wire (Transponder)
Tone to send: 2000Hz Tone to detect: 1780Hz
Tone to detect: 2000Hz Tone to send: 1780Hz
ANSI/4-wire (Loopback)
Tone to send: 2010Hz Tone to detect: 2010Hz
Voice path loopback applied.
ANSI/2-wire (Transponder)
Tone to send 2010Hz Tone to detect: 1780Hz
Tone to detect: 2010Hz Tone to send: 1780Hz
Inbound continuity checks Inbound continuity checks, whether requested by an IAM or a CCR, are handled automatically without the need for additional configuration.
Outbound in-call continuity checks GIII may be configured to perform outgoing continuity checks for all calls made on a route, and to re-route the call should the continuity check fail. If a call is routed to a circuit that fails an outgoing continuity check, the circuit will be placed into the recheck state. GIII will automatically perform periodic continuity checks for circuits that are in the recheck state until a pass result is achieved or until the circuit is taken out of the recheck state for another reason. Whilst in the recheck state it will not be possible to route calls to the circuit. Circuits remain in the recheck state indefinitely until one of the following occurs:
• GIII performs a periodic continuity check that passes.
• The operator uses the Continuity Check dialog to perform a continuity check test call that passes.
• The circuit is reset.
• An incoming IAM is received.
• GIII is restarted or the SS7 firmware is reloaded.
Continuity check test calls Outbound continuity check test calls (CCR) can be made on individual circuits using the SS7 Continuity Checking.
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13 CAMA telephony CAMA is a pseudonym used to describe the MF and Enhanced MF signalling specified by the National Emergency Number Association (NENA) for use in an E911 environment. There are four distinct variants of CAMA, which are described in the following specifications:
NENA 03-502 (April 2003) Trunking for Private Switch 9-1-1 Service NENA 03-002 (January 2007) Enhanced MF Signalling, E9-1-1 Tandem to
PSAP Each specification describes both single number (calling party number (ANI) only) and dual number (called party and calling party number (ANI)) signalling variants.
13.1 Configuration 13.1.1 Signalling firmware configuration
CAMA has been implemented as a variant of the Aculab T1 Robbed Bit protocol. Each CAMA trunk should be configured with the CAMA : USA T1 NENA Robbed-bit protocol, and the appropriate configuration switch from the table below applied in addition to the mandatory switches (-s59,1 –s96,1 –s99,224). With the exception of the –cMD switch, described in the CAMA to SIP Call Transfer mapping section of this document, no other firmware switches should be applied.
Firmware switch
CAMA variant
-s60,1 CAMA 2003 single number (calling party number (ANI) only). Only single numbers can be signalled when this switch is applied.
-s60,2 CAMA 2003 dual number. Only dual numbers can be signalled when this switch is applied.
-s60,3 CAMA 2007 single number (calling party number (ANI) only). CAMA 2007 dual number. The signalling arrangements for CAMA 2007 differentiate between the single and dual number variants. This switch will allow both variants to be used on the same trunk simultaneously.
13.1.2 GIII application configuration 13.1.2.1 Trunk configuration
When an incoming call is received, the GIII application software collects the dialled digits using the ST/STP digits as an indication that dialling is complete. To ensure that the call is only routed on completion of dialling, the Call Handling controls on the particular trunk tab in the Trunks page must always be configured as follows:
Incoming Minimum digits – This must be set to zero. Incoming Maximum digits – This must be set to zero. Incoming Digit timeout – This must be set to zero. Route on EOS – This must be enabled.
The following diagram illustrates these settings.
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Figure 13-1 CAMA trunk configuration settings
13.1.2.2 Route configuration When processing an incoming CAMA call, the KP and ST/STP digits will automatically be removed from the called and/or calling party numbers. Unlike some other Aculab CAS protocols, there is no need to accommodate these digits in the Destination address and Originating address masks in the routing table. The information digits (I or II) are not automatically removed from the dialled number, and must be removed using the routing table. See the Routing configuration section of this document for an explanation of number translation. When making an outgoing CAMA call, the KP and SP/STP digits will automatically be added to the called and/or calling party numbers. Unlike some other Aculab CAS protocols, there is no need to add these digits in the Destination address and Originating address masks in the routing table.
13.2 Interworking with CAMA 13.2.1 CAMA to SIP Interworking
All interworking from CAMA to SIP will be carried out in accordance with NENA 08-003 – Detailed Functional and Interface Specification for the NENA i3 Solution – Stage 3, where GIII will perform the role of a Legacy Network Gateway. The following behaviour is drawn to the attention of users. When the incoming CAMA call does not present a called party number, the destination IP address alone will be used in the SIP Request-URI and To headers. If a called party number is required in the SIP INVITE, it should be inserted using the GIII routing table. The incoming calling party number, after translation by the GIII routing table, will be presented in a P-Charge-Info header, with the From header containing only the IP address of the GIII host port.
13.2.2 SIP to CAMA Interworking NENA 03-008 does not specify procedures for interworking from SIP to CAMA, and general GIII procedures for interworking from SIP to non-SS7 TDM calls will be applied. Specifically the called party number used in the outgoing CAMA call will be taken from either the P-Asserted-Identity or From header in the SIP INVITE (see Routing configuration section for an explanation of which header will be used). Any P-Charge-Info header presented by the SIP INVITE will be ignored.
A call transfer request received from the CAMA side can be mapped to the SIP side as described in the CAMA to SIP Call Transfer Mapping section.
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13.2.3 Interworking between CAMA and TDM All interworking between CAMA and other TDM protocols will be carried out using general GIII procedures. When interworking from TDM to CAMA it should be noted that the Aculab implementation of CAMA supports en-bloc dialling only, overlapped dialling is not supported. All dialled digits from the incoming TDM call must be collected before the call is routed. See the Routes part of the Routing configuration section for an explanation of collecting dialled digits before routing a call.
13.3 Disabling incoming CAMA call clearing In some situations it is desirable to prevent the calling party from clearing a call, and this can be achieved when interworking from CAMA to SIP. This feature is available once the call has reached the connected state. When enabled and the calling party enters the on-hook condition, the outgoing SIP call leg is maintained and the idle pattern inserted onto the forward voice path. Should the calling party return to the off-hook condition, bidirectional speech between the two parties is restored. The feature can be invoked repeatedly on the same call, which will only be cleared when the outgoing SIP call leg signals disconnect. This feature requires the –cMD configuration switch to be applied when downloading the signalling firmware to a CAMA port, select Called Party in the Call Clearing dropdown on the Call Handling page of the Trunk.
Figure 13-2 CAMA trunk configured for called party only call clearing
When a CAMA trunk is configured for called party clearing and the outgoing call leg is not a SIP call, bidirectional clearing will be applied and either party will be allowed to clear the call.
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14 Interworking supplementary services GIII supports interworking of a number of supplementary services between protocols. The following sections describe the ISDN and/or SIP support available for such services.
NOTE This section describes those supplementary services where support has been implemented in only a limited number of protocols. Where a supplementary service is implemented by all supported protocols, it is documented more generally throughout this guide. Calling party number is an example of such a service.
14.1 Name identification supplementary services (CNIP/CONP) The calling name identification presentation (CNIP) and connected name identification presentation (CONP) services can be interworked between Q.SIG and SIP.
14.1.1 Q.SIG The CNIP and CONP services are implemented for the Q.SIG protocol as described in ECMA-164.
NOTE The name identification supplementary service must be enabled by applying the –cFNP configuration switch when downloading the signalling firmware to a Q.SIG port.
14.1.2 SIP CNIP/CONP information is transported in SIP using the Remote-Party-ID header. The Remote-Party-ID header is described in IETF Internet Draft draft-ietf-sip-privacy-04, and has the form: Remote-Party-ID: “John Smith” <[email protected]>;party=calling; id-type=subscriber;privacy=name;screen=no
14.1.3 Q.SIG to Q.SIG interworking Interworking CNIP/CONP information between two Q.SIG call legs is achieved by copying the calling/connected name component unmodified between the incoming and outgoing call legs.
14.1.4 Interworking between Q.SIG and SIP The mapping of CNIP/CONP information between Q.SIG messages and SIP requests/responses is illustrated below.
SETUP + callingName
↔ INVITE Remote-Party-ID: party=calling
ALERTING + calledName
↔ 180 Ringing Remote-Party-ID: party=called
CONNECT + connectedName
↔ 200 Ok Remote-Party-ID: party=called
DISCONNECT + busyName
↔ 3xx/4xx/5xx/6xx Remote-Party-ID: party=called
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Mapping is only performed when an inbound CNIP/CONP component or Remote-Party-ID header of the correct type is received, otherwise the information is discarded. The name information is mapped unmodified between the display parameter in the Remote-Party-ID header and the NameData field in the calling/connected name component. When sending an outgoing INVITE the URI in the Remote-Party-ID header will be the same as that in the Request-URI. When sending a backward SIP request/response the URI will be the same as that received in the forward From header.
The privacy parameter in the Remote-Party-ID header will be set according to the following table when mapping from Q.SIG to SIP.
Q.SIG name presentation privacy parameter namePresentationRestrictedSimple namePresentationRestrictedExtended namePresentationRestrictedNull
privacy=name
namePresentationAllowedSimple namePresentationAllowedExtended
No privacy parameter included.
Name presentation will be set according to the following table when mapping from SIP to Q.SIG.
privacy parameter Q.SIG name presentation privacy=name privacy=full
namePresentationRestrictedSimple
All other values, or no privacy parameter.
namePresentationAllowedSimple
GIII does not process the URI, id-type and screen parameters in an incoming Remote-Party-ID header.
GIII will not include the id-type and screen parameters in an outgoing Remote-Party-ID header.
14.1.5 SIP to SIP interworking A Remote-Party-ID header presented in an INVITE request/response will be passed unmodified between incoming and outgoing call legs.
14.2 User to user signalling (UUS) GIII is able to map user-to-user service requests (UUS) and user-to-user information (UUI) between the ETS 300, SS7 and SIP protocols.
14.2.1 ETS 300 The user-to-user signalling service 1 (UUS1), described in ETS 300 286-1, is supported by the ETS 300 protocol as described in the sections below.
NOTE The UUS supplementary service must be enabled by applying both the –cFU and –cDAUUS configuration switches when downloading the signalling firmware to an ETS 300 port.
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If an incoming ETS 300 call requests UUS2 and/or UUS3 activation and either request is for required activation then the incoming call will be disconnected with clearing cause #50 (Requested facility not subscribed), otherwise the first appropriate backward message will carry a rejection response and the call will be allowed to proceed.
NOTE Some legacy ETS 300 equipment imposes a limit of 35 octets on the length of a User-user information element. By default GIII supports the current 131 octet standard. Applying the –s8,1 switch when downloading the ETS 300 signalling firmware will truncate all outgoing User-user information elements to 35 octets. No truncation is carried out on incoming User-user information elements.
NOTE Incoming ETS 300 calls that have requested explicit UUS1 activation cannot be re-routed. Call re-routing is described in the General part of the Routing configuration section.
14.2.2 SS7 The user-to-user signalling service 1 (UUS1), described in Q.737.1, is supported by the SS7 protocol as described in the sections below.
NOTE GIII does not support the user-to-user signalling service when using the ANSI variants of SS7.
If an incoming SS7 call requests UUS2 and/or UUS3 activation and either request is for required activation then the incoming call will be disconnected with clearing cause #50 (Requested facility not subscribed), otherwise the first appropriate backward message will carry a rejection response and the call will be allowed to proceed.
NOTE Incoming SS7 calls that have requested explicit UUS1 activation cannot be re-routed. Call re-routing is described in the General part of the Routing configuration section.
14.2.3 SIP GIII supports the transfer of user-to-user information in the setup and disconnect phases of a SIP call. This is implemented using the User-to-User header described in IETF Internet Draft draft-ietf-cuss-sip-uui-09. An example of the User-to-User header is:
User-to-User: 001234ABCD;encoding=hex;purpose=foo;content=bar
Including the User-to-User header as a parameter in a redirection URI is not supported.
14.2.4 ETS 300 to ETS 300 interworking The UUIE is mapped unmodified between the incoming and outgoing call legs with the following rules applied.
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If the incoming SETUP contains a UUIE then this will be mapped unmodified to the outgoing call leg, otherwise a UUIE with an empty User Information parameter will be inserted into the outgoing call leg to invoke an implicit UUS1 service. The mapping of UUI between PROGRESS messages is supported. However, if an explicit UUS request has been made and a UUIE is received in a backward PROGRESS message ahead of the explicit UUS response, then that UUIE will be discarded.
14.2.5 SS7 to SS7 interworking The User-to-user information parameter is mapped unmodified between the incoming and outgoing call legs.
14.2.6 Interworking between ETS 300 and SS7 Interworking between SS7 and ETS 300 is based upon the procedures described in ITU-T Recommendation Q.699.
14.2.7 Interworking between ETS 300 and SIP Interworking between ETS 300 and SIP is based upon the procedures described in ETSI TS 183 036. The mapping of UUI between ETS 300 messages and SIP requests/responses during call setup is illustrated below.
SETUP ↔ INVITE
PROGRESS ↔ 183 Session Progress
ALERTING ↔ 180 Ringing
CONNECT ↔ 200 Ok
DISCONNECT ↔ 3xx/4xx/5xx/6xx
The mapping of UUI between ETS 300 messages and SIP requests/responses during normal call clearing initiated by the ETS 300 side is illustrated below.
DISCONNECT → BYE
RELEASE ← 200 Ok
The mapping of UUI between ETS 300 messages and SIP requests/responses during normal call clearing initiated by the SIP side is illustrated below.
DISCONNECT ← BYE
RELEASE → 200 Ok
Mapping of UUI to and from RELEASE COMPLETE messages is not supported.
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The following table describes how GIII uses the parameters in the User-to-User header.
Parameter Application UUI data This is a concatenation of the protocol discriminator and UUI data,
in hexadecimal form. The first two digits are the protocol discriminator and the remaining digits are the UUI data.
encoding When present in an inbound header this parameter must be in the form encoding=hex, otherwise the header will be discarded. If the parameter is not present in an inbound header encoding=hex will be assumed. This parameter will always be included in outbound headers in the form encoding=hex.
purpose When present in an inbound header this parameter is discarded. It will not be included in an outbound header.
content When present in an inbound header this parameter is discarded. This parameter may optionally be included in outbound headers, see the Protocol settings part of the SIP trunk configuration section for details.
When interworking from ETS 300 to SIP, and the incoming call leg requests explicit UUS1 activation, GIII will automatically send an accepted response in the first appropriate backward message.
14.2.8 Interworking between SS7 and SIP There are no formal procedures specified for interworking the UUS service between SS7 and SIP. The procedures adopted are similar to those applied when interworking between ETS 300 and SIP. The mapping of UUI between SS7 messages and SIP requests/responses during call setup is illustrated below.
IAM ↔ INVITE
ACM/PRG1 ↔ 183 Session Progress
ACM/PRG2 ↔ 180 Ringing
ANM/CON ↔ 200 Ok
REL ↔ 3xx/4xx/5xx/6xx
1. When there is no subscriber free/alerting indication. 2. When subscriber free/alerting is indicated.
The mapping of UUI between SS7 messages and SIP requests/responses during normal call clearing is illustrated below.
REL ↔ BYE
Mapping of UUI to and from a 200 Ok at this stage is not supported.
The table in the previous section describes how GIII uses the parameters in the User-to-User header.
When interworking from SS7 to SIP, and the incoming call leg requests UUS1 activation, GIII will automatically send an accepted response in the first appropriate backward message.
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When interworking between SS7 and SIP with Functional Number mapping enabled:
• If the incoming IAM includes a User-to-user information parameter containing a Functional Number, both Aculab-Functional-Number and User-to-User headers will be included in the outgoing INVITE.
• If the incoming INVITE contains both Aculab-Functional-Number and User-to-User headers, the Aculab-Functional-Number header will be mapped into the User-to-user information parameter in the outgoing IAM and the User-to-User header discarded.
• See the Functional Number mapping section for a description of Functional Numbers.
14.2.9 SIP to SIP interworking A User-to-User header presented in INVITE requests and responses, or BYE requests and responses will be passed unmodified between incoming and outgoing call legs.
14.3 Multi level precedence and pre-emption (MLPP) GIII is able to map MLPP service requests between the ETS 300, Q.SIG and SIP protocols. MLPP service requests are mapped transparently, and neither management nor enforcement of is performed.
14.3.1 ETS 300 and Q.SIG The MLPP supplementary service for ETS 300 and Q.SIG is based upon ITU-T Recommendation Q.955. The Look-ahead For Busy (LFB) query described in Q.955 is not supported, and such queries are discarded.
14.3.2 SIP MLPP service requests are mapped in SIP calls using the Resource-Priority header described in IETF RFC 4412. An example of the Resource-Priority header is:
Resource-Priority: q735.4
The single parameter is composed of a namespace and priority value in the form namespace.priority. In the above example q735 is the namespace and 4 is the priority. GIII will only process a Resource-Priority header presented in an INVITE. If a Resource-Priority header is presented in any other request, or any request response, it will be discarded. GIII will only insert a Resource-Priority header into an outbound INVITE.
The resource-priority option tag is not supported.
14.3.3 ETS 300/Q.SIG to ETS 300/Q.SIG interworking The MLPP request will be mapped unmodified between the incoming and outgoing SETUP messages, with the exception that the MLPP Service Domain for the outgoing call leg may be modified as described in the Q.931/ISUP part of the Routing configuration section. The MLPP request response will be mapped unmodified between the ALERTING message received from the outgoing call leg and the ALERTING message sent to the incoming call leg.
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NOTE Calls requesting MLPP that bypass the ringing stage (for example by moving directly to the connected stage) will be allowed to proceed, although the requested MLPP support is unlikely to be enforced.
If the outgoing call is disconnected with a clearing cause other than #8 before the ringing stage has been reached, the clearing cause along with any MLPP request response in the DISCONNECT and RELEASE messages will be mapped unmodified to the incoming call leg. If either call leg receives a DISCONNECT message with clearing cause #8 (Preemption) then:
• If the MLPPCallpreemption component in the inbound DISCONNECT message is set to circuitReservedForReuse, the timeslot carrying that call will be reserved for 12 seconds to receive the incoming preempting call.
• A DISCONNECT message will be sent to the opposite call leg using clearing cause #8, with the MLPPCallpreemption component set to circuitNotReservedForReuse.
• The MLPP information in the subsequent RELEASE message will be mapped unmodified between the two call legs.
NOTE The incoming pre-empting call will be treated as a new call and routed in the normal fashion.
14.3.4 ETS 300/Q.SIG to SIP interworking When interworking from ETS 300/Q.SIG to SIP, a Resource-Priority header with a q735 namespace will be included in the outgoing INVITE, with the priority determined by the Q.955 precedence level in the MLPP request. The following table illustrates the mapping between the precedence levels defined in Q.955 and the priority values in the q735 namespace defined in RFC 4412.
Q.955 precedence RFC 4412 priority flashOverride (0) q735.0
flash (1) q735.1
immediate (2) q735.2
priority (3) q735.3
routine (4) q735.4
When the SIP call reaches the ringing stage, an ALERTING message will be sent to the incoming call leg with an MLPP request response of successCalledUserMLPPSubscriber. If the SIP call leg is cleared before the ringing stage is reached and the response includes a Reason header specifying a Q.850 cause of #8 (Preemption) or #46 (Precedence call blocked), a DISCONNECT message will be sent to the incoming call using that Q.850 clearing cause and an MLPP request response as shown in the table below.
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Q.850 cause MLPP request response 8 failureCaseB 46 failureCaseA
If the SIP call leg is cleared after the ringing stage is reached, and the INVITE response or BYE includes a Reason header specifying a Q.850 cause of #8, a DISCONNECT message will be sent to the incoming call with clearing cause #8 and an MLPP request response of circuitNotReservedForReuse. When the incoming call leg is cleared at any stage with clearing cause #8 a Reason header will be included in the CANCEL/BYE specifying Q.850 cause #8, and if the MLPPCallpreemption component in the DISCONNECT message is set to circuitReservedForReuse the timeslot will be reserved for 12 seconds to receive the preempting call.
NOTE The incoming pre-empting call will be treated as a new call and routed in the normal fashion.
In all other circumstances standard call clearing will be used.
14.3.5 SIP to ETS 300/Q.SIG interworking The Resource-Priority header presented by the SIP INVITE must contain a q735 namespace, otherwise the header will be discarded and MLPP will not be requested. The outgoing call will be made with an MLPPCallRequest component initialized as follows:
• The MLPP service domain will be set according to the GIII routing table (see Q.931/ISUP part of Routing configuration section).
• The precedence level will be set using the table in the section above. If the outgoing call leg is disconnected with clearing cause #46 (Precedence call blocked), the INVITE response sent to the incoming SIP call will include a Reason header specifying Q.850 cause #46. If the outgoing call leg is disconnected with clearing cause #8 (Preemption) the INVITE response/BYE sent to the incoming SIP call will include a Reason header specifying Q.850 cause #8, and if the MLPPCallpreemption component in the DISCONNECT message is set to circuitReservedForReuse the timeslot will be reserved for 12 seconds to receive the preempting call.
NOTE The incoming pre-empting call will be treated as a new call and routed in the normal fashion.
When the SIP call leg is cleared and the CANCEL/BYE includes a Reason header specifying Q.850 cause #8, a DISCONNECT will be sent to the outgoing call leg with clearing cause #8 and an MLPPCallpreemption component set to circuitNotReservedForReuse. In all other circumstances standard call clearing will be used.
14.3.6 SIP to SIP interworking A Resource-Priority header presented in an INVITE request will be passed unmodified between incoming and outgoing call legs.
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14.3.7 Interworking to protocols without MLPP support If an incoming ETS 300/Q.SIG call containing an MLPP request is routed to a protocol that does not have MLPP support, GIII will return a request response of successCalledUserNotMLPPSubscriber in the ALERTING message, and the call will be allowed to proceed. SIP has no mechanism to indicate that a called user is not an MLPP subscriber. If an incoming SIP INVITE containing a Resource-Priority header is routed to a protocol that does not have MLPP support, then the Resource-Priority header will be discarded and the call will be allowed to proceed.
14.3.8 Call diversion and re-routing If an incoming MLPP call is re-routed or diverted and the new outgoing call is made using a protocol with MLPP support, an MLPP request will be included in the new outgoing call. If the re-routing/diversion takes place before the ringing stage has been reached, then all MLPP responses will be mapped to the incoming call in the appropriate manner. However, if the call re-routing/diversion takes place after the ringing stage has been reached (i.e. an MLPP response has already been sent to the incoming call leg), the call will proceed on the basis of the MLPP response already sent to the incoming call leg.
14.3.9 Call recovery When an outgoing SIP call is recovered and the original INVITE included a Resource-Priority header, an identical Resource-Priority header will be included in the recovery call leg.
14.4 Additional calling party number Additional calling party number is supported by, and interworked between, the SS7 and Q.931 protocols. Q.931 protocols present an additional calling party number by including two Calling Party Number information elements in the SETUP message. The first Calling Party Number information element is interpreted as the calling party number, with the second interpreted as the additional calling party number.
NOTE The ETS 300 protocol allows the calling party number and additional calling party number to be reversed in the SETUP message by applying the -s7,1 firmware download switch.
GIII supports additional calling party number interworking between SIP and SS7/ETS 300, as described in the specifications listed in the Interworking with TDM protocols section.
14.5 Connected party number (COLP/COLR) GIII supports connected party number interworking between the Q.931, SS7 and SIP protocols. The connected party information presented by the outgoing call at the connected stage is passed through to the incoming call.
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14.5.1 Q.931 GIII supports connected party number when using the ETS 300, Q.SIG and AT&T protocols. The connected party number is presented using a Connected number parameter in the CONNECT message as described in ITU-T Recommendation Q.951.
14.5.2 SS7 GIII supports connected party number when using the ITU-T, China and UK ISUP variants. The connected party number is presented using a Connected number parameter in the ANM/CON message as described in the protocol specification for the variant in use.
14.5.3 SIP Connected party number presentation (COLP) GIII uses a P-Asserted-Identity header in the 200 OK response to present the connected party number. The P-Asserted-Identity header is described in IETF RFC 3325, and has the form:
P-Asserted-Identity: “Bob” <sip:[email protected]>
In the above example +441908273800 will be used as the connected party number.
Connected party number restriction (COLR) The 200 OK may use a Privacy header to indicate connected party number presentation. The Privacy header is described in IETF RFC 3323 and has the form:
Privacy: id;header;user
14.5.4 TDM interworking Interworking of connected party number between TDM calls is based upon standard procedures.
14.5.5 TDM to SIP interworking The following table describes how a Q.931 Connected number parameter will be formed from the inbound P-Asserted-Identity and Privacy headers.
Q.931 parameter Setting Type of number If the connected number digits are preceded by a + then
this will be set to international number otherwise it will be set to national number.
Numbering plan identification ISDN/Telephony numbering plan (CCITT Rec. E.164/E.163). Presentation indicator If a Privacy header containing any of the tags id, header
or user is received this will be set to presentation restricted otherwise it will be set to presentation allowed.
Screening indicator This will be set to the value configured in the Q.931/ISUP page of the Routing Configuration screen (see Routing configuration section). If Transparent is selected network provided will be used.
Number digits The connected number digits are transferred unmodified, with the exception of a leading +, which will be discarded.
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The following table describes how an SS7 Connected number parameter will be formed from the inbound P-Asserted-Identity and Privacy headers.
SS7 parameter Setting Nature of address indicator If the connected number digits are preceded by a + then
this will be set to international number otherwise it will be set to national (significant) number.
Numbering plan indicator ISDN (Telephony) numbering plan (ITU-T recommendation E.164). Address presentation restricted indicator
If a Privacy header containing any of the tags id, header or user is received this will be set to presentation restricted otherwise it will be set to presentation allowed.
Screening indicator This will be set to the value configured in the Q.931/ISUP page of the Routing Configuration screen (see Routing configuration section). If Transparent is selected network provided will be used.
Address signal The connected number digits are transferred unmodified, with the exception of a leading +, which will be discarded.
A connected subaddress may be included in the P-Asserted-Identity header, and this will be mapped as described in the ISDN subaddress mapping section.
14.5.6 SIP to TDM interworking The information in the inbound Connected number parameter will be used to present a P-Asserted-Identity header in the following form:
P-Asserted-Identity: <sip:[email protected]>
The Number digits/Address signal will be copied unmodified into the URI user part. If the Type of number/Nature of address indicator is set to international number then the user part will be prefixed with a +.
The URI host part will contain the IP address of the GIII host port. If the Presentation indicator/Address presentation restricted indicator is set to a value other than presentation allowed, then a Privacy header will be added to the 200 OK as shown in the following table.
TDM protocol Privacy header ETS 300 Privacy: id;header;user
All others Privacy: id
If a connected subaddress is presented by the TDM call leg, this will be mapped into the P-Asserted-Identity header as described in the ISDN subaddress mapping section.
14.5.7 SIP interworking When interworking between SIP calls, any P-Asserted-Identity and Privacy headers presented in the 200 OK response from the outgoing call leg will be passed through unmodified to the incoming call leg.
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15 TTY/RTT mapping When interworking between TDM and SIP calls, GIII can be configured to convert Baudot tones generated by a TTY device to, and from, a real time text (RTT) stream. The frequencies and codes used by Baudot tones are described in ANSI TIA/EIA-825 (2000). The RTT stream is described in IETF RFC 4103.
15.1 Configuring a TTY system The GIII must be configured to perform TTY/RTT mapping. This is achieved by changing the Role Configuration – Trunks - SIP Protocol Settings – Use Real Time Text (RTT) setting, and then selecting the created revision from these changes in the role.
Figure 15-1 Selecting TTY/RTT operation
If the system is configured for TTY operation, the message System configured for TTY/RTT mapping will be printed in the GIII Engine log during role startup. This can be viewed from the Maintenance – Trace Files – Call – GXXXX_engine.log file (where XXXX is the G number of the GIII).
Figure 15-2 GIII engine log showing TTY/RTT mapping active
When RTT is selected as either Always, or On Request, there will be an exclamation showing a warning that if secure SIP is configured, the calls will instead use SIP over UDP.
Figure 15-3 Warning about secure SIP and RTT
The Routing – Route – Outgoing SIP Calls page will also show a warning about configuring for secure SIP and RTT.
Figure 15-4 Warning about secure SIP and RTT in Route
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15.2 Operating limits The GIII configured for RTT/TTY loads alternative and additional firmwares. These firmwares consume extra resources, and so place a number of limitations on system operation. Where necessary, GIII will override the system configuration to enforce these limitations.
NOTE For clarity it is recommended that you reconfigure your system when switching to TTY operation, rather than rely on GIII to override your configuration settings.
15.2.1 Call capacity The TTY functionality in GIII is designed for use in T1 environments, and so each Prosody T card is limited to handling 192 (eight trunk) SIP calls, rather than the 240 available on a standard system. Any surplus TDM channels (for example where one or more trunks on the card is configured with an E1 protocol) remain available for TDM interworking.
NOTE Although the TTY functionality is designed for use in T1 environments, GIII is able to detect Baudot tones on both E1 and T1 trunks.
15.2.2 Audio codec selection Only the use of G.711 audio codecs is supported. GIII will remove all codecs other than G.711 from the codec list selected by the route prior to making a media offer. If the codec list does not contain the G.711 codec, then G.711 with voice activity detection disabled will be offered. Only the G.711 codec will be selected from an offer made by the remote endpoint. If the remote endpoint does not include a G.711 codec in its media offer then the offer will be declined.
15.2.3 Secure SIP Secure SIP telephony (TLS and SRTP) is not available on TTY systems. GIII will not use TLS signalling nor will it make media offers requesting secure RTP. Offers from remote endpoints requesting secure SIP will be declined.
15.3 RTT stream negotiation When making a media offer, GIII can be configured to request an RTT stream in one of two ways:
• Always include an RTT stream in the media offer.
• Request an RTT stream only when Baudot tones have been detected on the TDM side. If media negotiation has already been completed prior to this a SIP reINVITE will be used.
RTT stream negotiation is configured using the Use Real Time Text (RTT) selection in the Protocol Settings tab in the Role Configuration – Trunks page (see SIP Configuration section). GIII will always accept a media offer from the remote endpoint requesting an RTT stream.
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15.4 T.140 redundancy The RTT stream can be configured to use up to ten levels of T.140 redundancy. The redundancy level is configured using the RTT Redundancy Level selection in the Protocol Settings tab in the Role Configuration – Trunks page (SIP Configuration).
15.5 Voice and hearing carry over GIII supports voice carry over (VCO) and hearing carry over (HCO):
• Any Baudot tones detected on the inbound TDM audio path are removed and transported across the outbound RTT stream. All other inbound audio is passed unmodified across the outbound RTP stream.
• All text characters received on the inbound RTT stream are converted to Baudot tones and played on the outbound TDM audio path. All audio received on the inbound RTP stream is passed unmodified across the outbound TDM audio path.
15.6 SIP call retry If an outgoing SIP call that offered an RTT stream is retried, then an identical RTT stream will be offered in the retry call.
15.7 SIP call recovery If an outgoing SIP call has negotiated an RTT stream prior to call recovery being invoked, then an identical RTT stream will be offered in the recovery call.
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16 Web Services There are a number of web services available on GIII. There is a help web service to get information on these. If using curl the following would get the web services help:
curl -k --user admin:password https://10.202.104.148/ws/help? A transcript of the output of this command is in the Appendices.
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17 Advanced fault finding and maintenance
17.1 Dongle not present Check that the dongle is connected to the internal USB port.
17.2 Invalid activation key Should the GIII fail to validate its Activation Key, the website will display an alarm on the system summary page, and a message will be periodically shown. If the software you are using is the version pre-installed when your GIII was delivered, then the activation key will be found on a label applied to the service tag of the system. If the software you are using is an upgrade provided to you on a USB stick, then the activation key will be found on the USB stick. If the software you are using is an upgrade received by email, the email will have included the activation key. If you have lost your Activation Key, contact Aculab Support, quoting the serial number of your GIII chassis (G7xxx), and they will be able to tell you what the activation key is. They will not give out activation keys for any version of software other than the one you are authorised to use.
17.3 Status errors 17.3.1 Openin failed
Openin Failed can be due to configuration problems:
• Firmware download problems, can be checked by going to the Status Monitors TDM Trunks page and checking for a valid firmware signature.
17.3.2 Card clock stopped (TDM ports only) Card Clock Stopped, seen on the Status Monitors TDM Trunks page, indicates a card fault. A reboot may clear the problem.
17.3.3 NO SIGNAL DETECTED (TDM ports only) No Signal Detected seen on the Status Monitors TDM Trunks page indicates a cable fault or a failure of the far end equipment. A loop back on that port can be used to prove the card, status No Alarms proves the cabling is OK.
17.3.4 DISCONNECTED (IP ports only) Disconnected on an IP port indicates a cable fault or a failure of the Ethernet switch to which the port is connected. If both IP ports on a Prosody X card show Disconnected, the card will be unable to send or receive calls. When both the IP ports on a Prosody X card are cabled and working correctly, the status should read Active and Connected.
17.3.5 REMOTE ALARM INDICATION (TDM ports only) Remote Alarm Indication seen on the Status Monitors TDM Trunks page indicates that the connected equipment is sending an alarm towards GIII because it is not seeing a signal from GIII or is receiving a major alarm from GIII. The alarm may be present while the equipment is initialising and should disappear when the switch has settled.
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17.3.6 ALARM INDICATION SIGNAL (TDM ports only) Alarm Indication Signal is the highest level of alarm possible, the connected equipment should be restarted and the configuration checked.
17.4 Obtaining protocol trace from GIII (TDM ports only) The GIII Status Monitors Call Trace page can display the raw protocol transmitted and received on an ISDN trunk. This can also be sent to a file using the Role Configuration Call Trace page. From this trace file, all information about a call is displayed allowing the source of a problem to be quickly traced. On the Role Configuration Call Trace page: Select the trunks to be monitored and destination of the trace e.g. UI and Disk. The Disk log files are accessed from the Maintenance - Trace Files page, via the Call tab.
17.5 Using CDRs to monitor system performance If GIII is running under a very high load, the protocol trace may be missing from the trace files. The best way to monitor system performance is use the CDR query information saved to a file; this can show where calls are routing and if they are failing. The GIII will continuously store CDR information, with up to a month of data.
NOTE Any single CDR query is limited to 30000 records. This would mean having to do multiple requests with an incrementing start record to get a large set of results. This may best be achieved using the web service. See the Web Services section for more information.
17.6 Monitoring a single call through the GIII Using Call Control, Routing and Protocol trace watch the progress of a call through the system. The example below shows a successful call through the system without protocol trace enabled.
NOTE Depending on protocol and system configuration the trace may look slightly different. See the Appendices for a detailed description of trace file contents.
12:09:59 P:08 C:01 ID:0000000241 Incoming Call (0x5040802), opened ts 1 12:09:59 P:08 C:01 ID:0000000241 Called Num : 0123 12:09:59 P:08 C:01 ID:0000000241 Calling Num : 3210 12:09:59 P:08 C:01 ID:0000000241 Routing "Route 8 - SIP" P:01 C:-- 12:09:59 P:08 C:01 ID:0000000241 Opened outgoing (0x05260802) on P:01 (SIP) channel 00 to <sip:[email protected];transport=tcp> 12:09:59 P:01 C:00 ID:0000000241 SIP:Outgoing call from channel 00 on st:48, ts:00 12:09:59 P:08 C:01 ID:0000000241 Send progress 12:10:03 P:01 C:00 ID:0000000241 SIP:Outgoing Connected 12:10:03 P:08 C:01 ID:0000000241 Incoming Call Accepted 12:10:03 P:08 C:01 ID:0000000241 Incoming Connected 12:10:08 P:01 C:00 ID:0000000241 SIP:Outgoing Call Gone Idle 12:10:08 P:08 C:01 ID:0000000241 Incoming Call Disconnected 12:10:08 P:08 C:01 ID:0000000302 Openin from previous call ID:0000000241 12:10:08 P:08 C:01 ID:0000000302 waiting for incoming
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17.7 Common set-up problems 17.7.1 System clock
The most common set-up problem concerns the configuration of the system clock; this is the source of the clock within the system. This should be set to clock from a network source. If a long distance CAS trunk is connected, you may have to clock from the CAS link to stop calls being dropped.
17.7.2 Dedicated timeslots Some CAS switches and even some PRI ISDN (US local lines) have dedicated incoming and outgoing time slots; this should be considered when constructing the role.
17.8 Aculab technical support Aculab offer technical support to customers with a valid technical support agreement. Should you not have a technical support agreement, please contact your Aculab account manager.
NOTE Aculab can only support a GIII purchased directly from Aculab. In all other cases please contact your supplier.
To assist Aculab in resolving you problem as quickly as possible, the following information should accompany any problem report.
• The GIII serial number (G7XXX) of the system experiencing the problem.
• A full description of the problem you are experiencing.
• A diagnostics snapshot of the system produced using the Technical Support page.
• A trace file produced by the GIII Status Monitor showing both a good and bad call, with the following event filters enabled on both incoming and outgoing ports. See the Interpreting GIII trace file Appendix for a description of trace logging.
o Call Control o Routing o Switching o RTP/RTT o Protocol
The diagnostics snapshots can be very large, but depending upon what kind of problem is occurring, selecting the smallest timeframe to include additional files will help reduce the size. Advice from Aculab Support may be needed in what to supply.
Contact details UK Tel +44 (0)1908 273805 Office Hours 9am-5pm GMT US Tel +1(781) 352 3551 Office Hours 9am-5pm Eastern Time Email: [email protected] For 24-7 support please check the contact details in your SLA.
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Appendix A: Web Site Concepts GIII is configured though a web interface to which a user logs in. There are a number of concepts used throughout the web interface, and their usage is described in this section.
A.1 Menus The web interface is menu based. Hovering over a menu item will expose further sub menu items. Selecting the menu item will open the relevant page.
Menus
A.2 Tabs The tab when selected will display a different page. There can be multiple tab lists within a page.
Tabs
A.3 Accordions The accordion will allow a single item to be displayed from a set of items. The items are layered horizontally, and selecting the horizontal bar of the accordion will expand it, and hide the previously show item.
Accordions
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A.4 Checkboxes There are a few differing types of checkboxes show below. The item is selected when ticked.
Checkbox examples
A.5 Hints There are many items that have hints, and these can be exposed by hovering the mouse pointer over an item.
Hint
A.6 Drop down selects Drop down selectable items will have a downward triangle. When the button is pressed it will display a number of items, of which one can be selected.
Dropdown
A.7 Spinners A spinner will have both an up and down triangle offering ability to increase or decrease a value respectively.
Spinner
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A.8 Sliders A slider will allow a value to be changed by sliding the control left, or right for a horizontal slider, and up or down for a vertical slider.
Slider
A.9 Date selector If a date is to be input, as on the Administer CDR page, then selecting the input will also present a date selector.
Date selector
A.10 Drag and drop boxes There are many places where items can be moved between adjacent boxes. This is done by selecting and dragging an item from its current box into the other box, and then releasing the item.
Drag and drop boxes
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Drag to create There are variants of this where a new item can be created by dragging a button into a box
Empty drag to create box
Dragging the button into the Alarm Map List box will present a new item configuration page as shown below.
Drag to create configuration example
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Drag to Trash can for delete If a trash can is shown on a page, normally where an item is dragged to create, then items may be dragged from the box into the trash can to delete them.
Trash can example
When the item is dragged to the trash can the lid will be show to be removed when the item can be released into the trash can. This is shown below.
Trash can usage
NOTE The items will not be deleted until a submit button is pressed.
A.11 Submit and Cancel Most pages will present a submit and cancel button. These may sometimes only be displayed once a configuration item is changed. These should normally also show a red exclamation mark to draw attention to the fact that something has been changed. There may also be a tooltip provided when hovering over the exclamation mark.
Submit and cancel buttons
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Appendix B: SNMP facility Simple network management protocol (SNMP) is used for the exchange of management information between host systems on IP networks. The SNMP agent must be configured to enable traps to be sent to the correct destination and to allow requests to be processed. Refer to the SNMP agent section of the document for information regarding how to configure. The Aculab-GIII.mib MIB file contains the complete specification of the Aculab SNMP traps and objects. Refer to the SNMP reporting section for information on enabling, and configuring, the Aculab SNMP sub-agent.
B.1 SNMP alarms The following application specific traps are supported:
groomerRunningTrap Issued at GIII start-up, and thereafter whenever the operational state of GIII changes. To prevent spurious alarms being raised, the service will not issue an initial ‘stopped’ trap until three minutes after system start-up.
groomerSNMPReportingTrap At start-up the service reports whether SNMP is enabled. If during operation the SNMP reporting state changes, the new state will be reported. If SNMP reporting is disabled, it will not be possible to get the running state of GIII, the response will be reported as unknown.
groomerOtherProcessStateTrap There is a built-in list of processes, external to the Call Processing Engine, that the system monitors. GIII will periodically check the number of instances running for each process. A trap is issued if the number of instances is outside the expected range. If the number of instances returns to being within the configured range then a trap will be issued. When the system returns to the all processes running normally state then an additional ‘all processes running’ trap will be issued.
groomerInternalProcessStateTrap GIII periodically checks that its internal processes and threads are running. If any are stopped, a trap will be issued. If a process/thread restarts after stopping then a trap will be issued. When the system returns to the all processes running state then an additional ‘all processes running’ trap will be issued.
groomerTDMRxL1AlarmTrap Issued once for each trunk at start-up, and whenever the link state changes. This trap can be suppressed on a trunk-by-trunk basis.
groomerTDMRxCASMultiframeAlarmTrap Only CAS trunks will produce this trap. It is issued once for each trunk at start-up, and whenever the multiframes alarm state changes. This trap can be suppressed on a trunk-by-trunk basis.
groomerTDMClockSlipsTrap Only TDM trunks issue this trap. The slip threshold as an average of slips per second over the previous minute is configurable. This trap can be suppressed on a trunk-by-trunk basis.
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groomerTDMTxL1AlarmTrap Issued once for each TDM trunk at start-up, and whenever the transmitted alarm state changes
groomerTDMLayer2Trap On TDM Ports, with the exception of CAS and SS7, which has no concept of layer 2. The trap is issued once for each trunk at start-up, and whenever the layer 2 state changes. This trap can be suppressed on a trunk-by-trunk basis.
groomerTrunkOperationalStatusTrap Issued once for each TDM port at start-up, and whenever the state of the trunk changes.
groomerConfigLoadTrap Issued every time a configuration file is loaded, including at start-up. Reload information is retrieved using the system status object.
groomerFirmwareReloadTrap An information trap issued whenever a port is reloaded (from inside or outside GIII). No status is reported, a failure of the operation will be obtained through other SNMP objects.
groomerTrunkBlockingStateChangeTrap Issued for each trunk at start-up, and whenever the service state is changed.
groomerCallRoutingFailedTrap For information only, and will be raised whenever GIII cannot route an incoming call. This trap can be suppressed on a port-by-port basis.
groomerCallStateTrap Indicates a fault whenever a call has remained in one of its progress states for longer than the configured time interval. The exception to this will be when the call is in the idle state (waiting to receive an incoming call) - calls are allowed to remain in this state indefinitely.
groomerChassisTemperatureTrap This alarm is issued at system startup to report the current state of the internal chassis temperature. Thereafter it will be raised whenever the internal chassis temperature exceeds the recommended maximum for the chassis, or returns to normal. If a system has multiple temperature sensors separate alarms are used for each.
NOTE Not all chassis models support internal temperature monitoring.
groomerSystemFanTrap System fans are monitored in one of two ways. Some chassis use simple state detection (rotating or stopped), whilst others report the actual speed of rotation. In the latter case GIII will raise alarms should the speed of a fan fall to a level where it is reasonable to assume that the fan is about to fail. This alarm is issued at system startup to report the current state of the system fans. Thereafter it will be raised whenever a fan moves between the working and failed states. If a system has multiple fans fitted separate alarms are used for each.
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NOTE Not all chassis models support system fan monitoring.
groomerPSUStatusTrap Power supply units are monitored in one of two ways. Some chassis use simple state detection (working or failed), whilst others report the actual voltage output by the PSU. In the latter case GIII will raise alarms should the output of a PSU move outside an acceptable range. This alarm is issued at system startup to report the current state of the PSUs. Thereafter it will be raised whenever a fan moves between the working and failed states. If a system has multiple PSUs fitted separate alarms are used for each.
NOTE Not all chassis models support PSU monitoring.
groomerCPUUsageTrap Warns when the CPU load constantly exceeds a configured threshold. To overcome the effects of applications such as PC Anywhere, which run at a low priority but tend to use up all excess CPU capacity, a list of processes to exclude is included in the configuration. The trap is issued when the CPU load exceeds the upper threshold, and when it returns below the lower threshold.
groomerMemoryUsageTrap Raised to warn when system memory usage has risen above a configured upper threshold, and when it falls below the lower threshold.
groomerAvailableDiskSpaceTrap Raised to warn when the available disk space has fallen below a configurable threshold, and when the condition clears.
groomerClockSourceUpdatedTrap Raised at start-up, and whenever the clock source used by the H.100 bus changes.
groomerCardStatusTrap Issued once at start-up for each card in the system. For E1/T1 Digital Access Cards the trap is issued when the Kernel application reaches the system started state. For Prosody X cards, the trap is issued once the card is detected. The system waits for three minutes after startup to detect a Prosody X card before issuing a not found alarm. If a card that was not detected at start-up is subsequently found, a further trap is issued.
groomerCardModuleStatusTrap This alarm is raised to report the operational state of a Prosody T card.
groomerSystemInitialisedTrap This trap is for information, and is raised at start-up when GIII initialisation is completed and the system is ready to pass calls.
groomerReceivedRTPStoppedTrap Raised when the RTP traffic from a remote endpoint stops unexpectedly. If this alarm has been issued to warn of an RTP fault it will be issued again to clear the fault when either RTP traffic is detected once more, or the associated call is cleared.
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groomerSS7SignallingLinkStateTrap This alarm will be raised at system startup for each SS7 signalling link in the system, and thereafter whenever the signalling link state changes between available (able to pass traffic) and unavailable (unable to pass traffic).
groomerContinuityCheckStateTrap This alarm will be raised when an SS7 bearer channel fails an in-call continuity check. The alarm is cleared when the bearer channel next passes a continuity check, receives an incoming call or is reset. Note that GIII does not persist continuity check status across shutdown.
groomerCardTemperatureTrap This alarm will be raised if the status of the card temperatures changes.
groomerNodeRoleTrap This alarm will be raised whenever the role assigned to a GIII node changes.
groomerHostNICStateTrap This alarm will be raised whenever the state of a host NIC changes.
groomerNTPSynchronizedTrap This alarm will be raised whenever the GIII NTP synchronization changes, including at startup.
groomerCDRWriteTrap This alarm will be raised whenever the GIII detects that CDR records are now being written correctly (including at start-up), or are no longer being written correctly.
groomerCDRStorageTrap This alarm will be raised whenever the GIII detects that is now sufficient available CDR storage space (including at startup), or that there is no longer sufficient storage.
groomerDiskStateTrap This alarm will be raised whenever the state of a GIII disk changes.
B.2 SNMP Requests The following elements from the Aculab-GIII MIB allow GIII information to be presented as tables:
groomerTrunkTable This table collects alarm information, counters and the current status for GIII trunks.
groomerCallTable This table contains details of the current state of each call in the system.
groomerCardTable This table contains details of each Prosody T card that GIII expects to be fitted in the system, including the temperatures of the 4 sensors on the card. Also included is an entry for Prosody S.
groomerProcessStatusTable This table lists the state of any internal threads the GIII has started.
groomerSystemProcessTable This table lists the state of system processes being monitored by the GIII.
groomerPSUTable This table lists the current state of each power supply unit in the system.
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NOTE Not all chassis models support PSU monitoring.
groomerVoltageTable This table lists the current state of the chassis power rails.
NOTE Not all chassis models support power rail monitoring.
groomerChassisTemperatureTable This table lists the current state of each chassis temperature sensor.
NOTE Not all chassis models support chassis temperature monitoring.
groomerSystemFanTable This table lists the current state of each system fan.
NOTE Not all chassis models support system fan monitoring.
groomerCLEITable Retrieves the CLEI codes of the GIII chassis and all installed cards.
groomerSS7SignallingLinkStatusTable This table lists the current state of all SS7 signalling links in the system.
groomerContinuityCheckStatusTable This table lists the current continuity check status of all SS7 bearer channels in the system.
groomerHostNICTable This table lists the current state of each host NIC.
groomerDiskTable This table lists the state of each disk in GIII. This comprises of the physical disks and the raid disks. Requesting the following elements from the Aculab-GIII MIB using a ‘walk’ allows other, non-table based, information to be presented:
groomerServiceStatus This collects information on the status of the GIII restricted to the version, whether the GIII is running and whether SNMP reporting is currently enabled.
groomerKernelStatus This collects information on the current configuration of the GIII (if running).
groomerSystemResources This collects information on memory usage, CPU usage and free disk space on the GIII system.
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Appendix C: TDM signalling firmware configuration options This section describes signalling firmware configuration options that are additional to those listed in the firmware release notes, and the protocols with which each may be used.
C.1 -cA/B and X/Y bits configuration (DPNSS only) Default setting: A/X
This option allows the dpnss A/B X/Y bits to be configured for layers 2 and 3 to enable DPNSS to DPNSS configuration
-cAX -cAY -cBX -cBY
Incorrect setting of the A/B bits will result in an inactive DPNSS layer 2.
C.2 -cBBY Backbusy control (CAS only) Instructs the CAS driver (and signalling system) to use ‘backbusy’:
-cBBY configure all timeslots all ports
-cBBY, xxxxxxxx configure all timeslots indicated in the mask xxxxxxxx
The mask is an 8 digit hexadecimal representation of a 32bit value where each bit indicates the busy mode. A bit set to 1 will enable backbusy. A bit reset to 0 will disable backbusy. Bit 0 of the 32bit value controls timeslot 0 and bit 31 controls timeslot 31, for example:
1110 0110 0000 0010 0000 0000 0100 0000 will enable backbusy on timeslots: 31, 30, 29, 26, 25, 17, 6 and would be represented by the value: E6020040 and would therefore be configured by: -cBBY, E6020040
NOTE On E1 systems timeslots 0 and 16 are reserved and will be ignored.
C.3 -cCA connect acknowledgement (ETS300 only) Some equipment require a CONNECT_ACKNOWLEDGE from the user end of the protocol after receipt of a CONNECT message. This configuration switch provides for this behaviour.
C.4 -cCICnnnn circuit identification codes (ISUP only) -cCICnnnn -cCICnnnn,aaaaaaaa -cCICnnnn,aaaaaaaa,bbbbbbbb
Where nnnn is a decimal number of 1 to 4 digits that will be the CIC assigned to the first bearer timeslot, with subsequent CICs allocated sequentially (including any timeslot used for signalling, although this CIC will never be used). The OPC and DPC assigned to the ISUP circuits will be the most recent values preceding the –cSLC parameter itself. If specified, aaaaaaaa is a CIC MAP, which allows the user to define which timeslots have CICs assigned to them. It is specified as a 32bit number expressed in hexadecimal, where bit zero corresponds to timeslot zero etc. If not specified, a value of fffffffe is assumed.
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If specified, bbbbbbbb is a Circuit MAP, which allows the user to define which timeslots have may be used for ISUP call control. It is specified as a 32bit number expressed in hexadecimal, where bit zero corresponds to timeslot zero etc. If not specified, a value of fffffffe is assumed Any timeslots which are in use for signalling are automatically excluded from ISUP call setup, regardless of whether this is explicitly shown in the circuit map. If –cCICnnnn is not specified, ISUP will not be available for call setup on the timeslots of that port.
C.5 -cCn number of CLI digits (CAS only) It is necessary for the device driver and signalling system firmware to know the number CLI digits supported by the trunk. The driver configuration switch available for this purpose is –cCn, where n is a one or two digit decimal value of the number of digits required. The -cCn option specifies the number of CLI digits expected by the system, for example, -cC4 instructs the driver/signalling system to expect 4 CLI digits.
It is important that this initial configuration is correct or unexpected behaviour may result.
C.6 -cDn number of DDI digits (CAS only) It is necessary for the device driver and signalling system firmware to know the number of DDI digits supported by the trunk. The driver configuration switch available for this purpose is –cDn, where n is a one or two digit decimal value of the number of digits required. The -cDn option specifies the number of DDI digits expected by the signalling system, for example, -cD3 instructs the driver/signalling system to expect 3 DDI digits.
It is important that this initial configuration is correct or unexpected behaviour may result.
C.7 -cDPCnnnnn signalling point code (ISUP only) -cDPCnnnnn
Where nnnnn is a decimal number equating to the signalling pointcode of the adjacent network component to which the link connects. The network component may be a Signalling Transfer Point, or, if using fully associated signalling, a Signalling End Point. nnnnn must coincide with the pointcode of an STP (AdjacentPC), or of an SP (RemotePC) in the stack configuration file as described in the SS7 installation and administration guide. This parameter may be repeated to specify that signalling link(s) and ISUP bearer timeslots have different DPC values. See –cSLC and –cCIC.
C.8 -cEn Call Charging Switch (ETS300 only) This switch allows the application to send or receive call charging information depending on the country or exchange being used at the time. For further information on the call_put_charge() and call_get_charge() functions it is advisable to consult the 'Receiving Call Charge Information' and 'Sending Call Charge Information' sections in the Call Control Driver API Guide. The value transmitted will be one of two categories of charging for EuroISDN.
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1. UNITS - One method is to transmit the value as UNITS. When charging uses the UNITS format a value representing the number of units accrued up to that point during the call will be transmitted. This value is initially one and is incremented by one each time the call_put_charge() function is used. The driver does not require any parameters in the call to call_put_charge() by the application.
2. CURRENCY - The second method is CURRENCY. When this type of information is used by the protocol the transmitted value is the accumulated cost of the call up to that point. This information is passed to the driver in the "charge" field of put_charge_xparms. The format of this string is an ASCII representation of the numeric value. Example : To send the value of 100 the three ASCII characters necessary to encode 100 as a string would be used. Printing this string to the screen should show '100' (without quotes).
Codes For Different Countries (n) n=1 Switzerland/CURRENCY Charging will use Facility information elements based on ETS300 182. A charging string in put_charge_xparms() is required.
n=2 Germany/UNITS Charging will use Facility information elements based on ETS300 182. The charging string in put_charge_xparms() is ignored.
n=3 Holland/UNITS Charging will use Display information elements based on a country Specific specification. The charging string in put_charge_xparms() should be left blank. n=4 Switzerland/CURRENCY Charging will use Display information elements based on a country Specific specification. A charging string in put_charge_xparms()is required.
n=5 Austria/CURRENCY Charging will use Facility information elements based on ETS300 182. A charging string in put_charge_xparms() is required.
n=6 Norway/CURRENCY Charging will use Facility information elements based on ETS300 182. A charging string in put_charge_xparms() is required.
n=7 Sweden/CURRENCY Charging will use Facility information elements based on ETS300 182. A charging string in put_charge_xparms() is required.
n=99 Disable Charging
C.9 -cEX Primary Rate Call Charging (ETS300 only) This configuration switch is for use with ETS300 Advice of Charging where there may well be some national/network dependent differences in the way the charging information is presented. X is a one or two digit decimal country code:
0 is the default 1 for Switzerland (type I) 2 for Germany 3 for Holland 4 for Switzerland (type II old style) 5 for Austria 6 for Norway 7 for Sweden 99 for disable charging
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Consult the release notes supplied with the latest revision of firmware for further details.
C.10 -cFD Diversion (QSIG, ETS300, AT&T T1, and NI-2) This configuration switch activates the diversion feature in the driver. If the switch for this service is not set, any attempt to use this service through the API will fail.
Example: If Diversion and Facility features are required for QSIG on port zero then the switches can be supplied as follows using fwdspldr.
Fwdspldr 12345 0 qsig_sup.upr -cFF –cFD
C.11 -cFF facility (QSIG, ETS300, and NI2) This configuration switch activates the facility feature in the driver. If the switch for this service is not set, any attempt to use this service through the API will fail.
Example: If Diversion and Facility features are required for QSIG on port zero then the switches can be supplied as follows using fwdspldr.
Fwdspldr 12345 0 qsig_sup.upr -cFF –cFD
C.12 -cFP MLPP activation (ETS300 only) Activates Multi-Level Precedence and Pre-emption (MLPP Q.955) in the firmware and driver.
C.13 -cFR Raw Data (ETS300 and QSIG) Activates firmware to use Raw Data information. See the Call Control Driver API guide.
C.14 -cFU user to user (QSIG and ETS300) This configuration switch activates the user-to-user feature in the driver. User to User : -cFU
User to User : -cFUN (no negotiation)
If the switch for this service is not set, any attempt to use this service through the API will fail.
C.15 -cIMP75 Sets line impedance to 75 ohms (default is 120 ohms).
C.16 -cME If an outbound DASS/DPNSS call is made with the sending complete flag set, then an ISRMC is used, otherwise an ISRMI will be sent. If the -cMC switch is applied an ISRMC will be used for all outbound calls, regardless of whether the sending complete flag is set.
C.17 -cNA1 release link trunk (NI-2 only) Enables RLT (release link trunk) call transfer with NI-1 signalling. Without the switch NI-2 signalling is used for call transfers.
C.18 -cNCRC disable CRC4 (ISUP only) This instructs the firmware not to use CRC4 framing at L1.
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C.19 -cNE network end configuration (DASS, ETS300, AT&T, and NI2) Default setting: User end
Network end configuration, sets up the device driver for network end working -cNE configure port for network end
This switch applies to signalling systems with the exception of CAS, DPNSS, QSIG and ISUP.
NOTE For correct network end operation, you will require the network end firmware for that signalling system.
C.20 -cOPCnnnnn[,i] (ISUP only) Where nnnnn is a decimal number equating to the signalling pointcode, and optional instance, of your application. This must coincide with the pointcode (LocalPC) of an SS7 signalling point in the stack configuration file, as described in the SS7 installation and administration guide.
C.21 -cQM/S-A/B master/slave priority (QSIG only) This configuration switch enables the QSIG Master and Slave bits, M for master and S for slave, and also the ‘priority if call clash’ bits A/B. Valid options are:
-cQSB
-cQMA
-cQSA
-cQMB
One end must be set to be master and the other end must be set to be slave. Incorrect setting of these bits will result in an inactive QSIG layer 2. The default configuration for QSIG is –cQSB.
C.22 -cRn Default clearing cause (all versions) Default setting: BUSY
This option allows the default clearing cause to be configured, where n is a one or two digit hexadecimal value of the required clearing cause and must be a correct value supported by the signalling system.
-cR3E configure ‘Call Rejected’ on all ports
-cR3En0 configure ‘Call Rejected’ on port 0 only
-cR3En1 configure ‘Call Rejected’ on port 1 only
-cR3En2 configure ‘Call Rejected’ on port 2 only
-cR3En3 configure ‘Call Rejected’ on port 3 only
The above example shows the CALL_REJECTED cause for 1TR6.
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C.23 -cSLCnn signalling link code (ISUP only) Where nn is a decimal number equating to the MTP3 Signalling Link Code for the signalling link. The signalling link is created when the -cSLCnn parameter is processed, the parameters for the link must appear in this option. If -cSLCnn is specified more than once, multiple signalling links will be created. The OPC and DPC assigned to the signalling link will be the most recent values preceding the –cSLC parameter itself.
C.24 -cSO disable service message (AT&T and NI2) This configuration switch disables the transmission of the ‘service message’ by the protocol stack. The service message brings the timeslot into service and may cause problems with some manufacturers equipment.
C.25 -cSP stop call proceeding (ETS300 only) -cSP switch stops a CALL_PROCEEDING message from being sent automatically after a SETUP message has been received. It is useful if extra information is required to be included in the message by the use of the call_proceeding function.
- cSP configure for all ports
C.26 -cSU stop setup acknowledge (ETS300 only) Stops a SETUP_ACKNOWLEDGE message from being sent automatically after a SETUP message has been received. It is useful if extra information is required to be included in the message by the use of the call_setup_ack function.
C.27 -cSW configuration (ETS300 only) This configuration swtch modifies the driver for use in Sweden where a call_proceeding message is required instead of a setup_acknowledge message.
C.28 -cTS[I]nn (ISUP only) Where nn is a decimal number equating to the timeslot that will be used for the signalling link on the E1 line connected to the network port. For E1 cards that support the function, if I is specified, then the signalling link will face inwards (that is towards the switch matrix and H.110 bus), it can then be switched out on a different E1 using the Aculab switch API.
C.29 -cSPEEDnnk (ISUP only) Where nn is a multiple of 8 less than, or equal to, 64. This sets the speed of the MTP2 signalling links. If specified once it will apply to all signalling links on the trunk, otherwise the last setting prior to the –cSLC parameter is used. If not specified the default is 64k unless –cAMI is specified.
C.30 -cDAUUS (ETS 300 only) The call driver will automatically respond to UU service requests by either accepting them (if the –cFU switch is set) or refusing them (if the –cFU switch is omitted). The automatic UU service response occurs when the application fails to supply the call driver with a UU service response before the ringing/connected state is reached. This switch will disable the automatic response.
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Appendix D: Interpreting GIII trace file GIII event reports are too extensive in number and dynamic in content for an exhaustive reference to be produced. This purpose of this section is purely to illustrate the types of event report that can be expected and give an insight into their meaning. With the exception of a CDR, all trace entries begin with a timestamp, port and channel number, and call ID, for example 16:26:48 P:02 C:001 ID:0002141615.
The timestamp is the time at which the record was generated. The port and channel numbers are those being used by the call that generated the report. If the call is an IP telephony call the channel number is an arbitrary value assigned by GIII for reporting purposes only. The call ID is a unique identifier assigned to each incoming call by GIII. The call ID is used to identify all events generated by a particular incoming call and its corresponding outgoing call leg.
D.1 Alarms The alarms filter reports changes in transmitted alarms on a TDM port, the Alarm mapping part of the General section refers. 10:07:32 P:04 C:--- ID:---------- GENERATING AIS ON PORT : 04 FROM NOS ON PORT : 05 10:23:11 P:04 C:--- ID:---------- CLEARED AIS ON PORT : 04 FROM NOS ON PORT : 05
The report is generated by the port transmitting the alarm. It shows whether the alarm is being generated or cleared, which alarm is being generated, for example AIS ON PORT : 04, and the reason it is being generated, for example NOS ON PORT : 05.
D.2 Layer 1 The layer 1 filter reports the received alarm state of a telephony port at two second intervals. 12:14:02 P:04 C:--- ID:---------- LAYER 1 ESTABLISHED 12:14:02 P:05 C:--- ID:---------- NO SIGNAL DETECTED 12:14:02 P:05 C:--- ID:---------- LOSS OF SYNCHRONISATION 12:14:02 P:05 C:--- ID:---------- SLIPS : 02720
Where multiple alarms are present on a port, each is reported separately.
D.3 Data Link The data link filter reports the layer 2 status of a TDM port at two second intervals. 16:26:48 P:02 C:--- ID:---------- LAYER 2 ESTABLISHED – 0XFFFEFFFE 16:26:48 P:03 C:--- ID:---------- LAYER 2 DOWN
The hexadecimal bitmask indicates which bearer timeslots on the port are available for calls. SS7 CIC blocking operations are also reported using the layer 2 filter. 16:30:50 P:22 C:--- ID:---------- MAINTENANCE BLOCKING USING 0X000000FE FROM 0x000000FE 16:30:50 P:22 C:--- ID:---------- MAINTENANCE UNBLOCKING USING 0XFFFFFF00 FROM 0x000000FE
The first bitmask shows which channels were blocked/unblocked. The second bitmask shows the settings from which this bitmask was derived (these are the settings in the Port Blocking dialog, the Trunk Blocking section refers). In all of the above bitmasks, bit 0 represents channel 0 and bit 31 represents channel 31.
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D.4 Call Control Call control trace will report the significant events between setup and completion of an end-to-end call. The examples shown below are produced by a Euro ISDN to SIP call.
NOTE Call control events will not always appear in the strict order shown below, and may be interleaved with events from other calls or filters.
NOTE Events from IP telephony ports are prefixed with SIP:.
NOTE When tracing an end-to-end call, logging must be enabled on the ports carrying both the incoming and outgoing leg of the call.
The first events to appear show that an incoming call has been detected on port 2 channel 1, and report it’s called and calling party numbers. An incoming SIP call will also report the SIP URI from which these numbers were parsed. The hexadecimal number and OPENED TS 1 have no meaning outside Aculab. 16:26:48 P:02 C:001 ID:0002141615 INCOMING CALL (0X3b7002), OPENED TS 1 16:26:48 P:02 C:001 ID:0002141615 CALLED NUM : 234501 16:26:48 P:02 C:001 ID:0002141615 CALLING NUM : 543201
The incoming call will generate an event to report that an outgoing call has been made. It shows the port and channel (TDM calls only) on which the call has been placed, along with the called number/network address. The hexadecimal number has no meaning outside Aculab. 16:26:48 P:02 C:001 ID:0002141615 OPENED OUTGOING (0X00239802) ON P:00 (SIP) TO <SIP:[email protected]>
The outgoing call leg will generate events to show that the call has been established, and that alerting has been received from the called party. The ST: and TS: values are present only on IP telephony calls and have no meaning outside Aculab. 16:26:48 P:00 C:000 ID:0002141615 SIP:OUTGOING CALL ON ST:64, TS:00 16:26:48 P:00 C:000 ID:0002141615 SIP:OUTGOING RINGING
This event is generated by the incoming call and shows that alerting has been mapped through and the calling party is now waiting for the called party to answer. 16:26:48 P:02 C:001 ID:0002141615 WAIT FOR ACCEPT
Only IP telephony calls will generate this event, which reports the codec selected for the RTP stream along with the RTP endpoint addresses. 16:26:50 P:00 C:000 ID:0002141615 ESTABLISHED G.711 RTP CONNECTION (LOCAL 192.168.16.229:16642 REMOTE 192.168.16.248:3084)
The outgoing call leg generates this event to show that the calling party has answered and the call leg between GIII and the called party is now fully connected. 16:26:50 P:00 C:000 ID:0002141615 SIP:OUTGOING CONNECTED
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These events show that GIII has answered the incoming call, which has now become fully connected. 16:26:50 P:02 C:001 ID:0002141615 INCOMING CALL ACCEPTED 16:26:50 P:02 C:001 ID:0002141615 INCOMING CONNECTED
These events are generated by the outgoing call to show the called party has disconnected the call, and the RTP resources have therefore been relinquished. 16:27:00 P:00 C:000 ID:0002141615 DISCONNECTING G.711 RTP CONNECTION (LOCAL 192.168.16.229:16642 REMOTE 192.168.16.248:3084) 16:27:00 P:00 C:000 ID:0002141615 SIP:OUTGOING CALL GONE IDLE 16:27:00 P:00 C:000 ID:0002141615 SIP:RTP_RELEASE(OUT)
These events show that GIII has disconnected the incoming leg of the call in response to the called party hangup. 16:27:00 P:02 C:001 ID:0002141615 INCOMING CALL DISCONNECTED 16:27:00 P:02 C:001 ID:0002141615 INCOMING CALL HAS GONE IDLE
These two events appear at the end of each call to show that GIII has prepared the incoming channel to receive a new call. 16:27:00 P:02 C:001 ID:0002141645 OPENIN FROM PREVIOUS CALL ID:0002141615 16:27:00 P:02 C:001 ID:0002141645 WAITING FOR INCOMING
D.5 Routing The following is an example of routing trace. 16:26:48 P:02 C:001 ID:0002141615 ROUTING "ROUTE TDM - SIP" P:00 C:--
The report is generated by the incoming leg of the call and identifies the route against which it was matched, along with the port and channel selected for the outgoing call leg. No channel is specified when an IP telephony port is selected for the outgoing call.
D.6 Switching The following is a typical example of switching trace. 16:26:48 P:02 C:001 ID:0002141615 ON CARD BI ITOO : SW:00 IST:33 ITS:01 OST:64 OTS:00 16:27:00 P:02 C:001 ID:0002141615 SWITCH DISABLE : SW:00 OST:33 OTS:01
Switching trace identifies the point at which voice paths switching takes place, and also the resources used on the Aculab telephony card. This latter information has no meaning outside of Aculab.
D.7 RTP/RTT Events of this type will only be seen when at least one leg of the call is an IP telephony call. They report the various stages of resource allocation and configuration for the RTP and RTT streams. The following are examples of such events. 16:26:48 P:00 C:000 ID:0002141615 RTP (OUT) ALLOCATED
16:26:48 P:00 C:000 ID:0002141615 RTT (OUT) ALLOCATED
16:26:49 P:00 C:000 ID:0002141615 RTP (OUT) ECHO CANCELLATION DISABLED
16:26:50 P:00 C:000 ID:0002141615 ESTABLISHED G.711 RTP CONNECTION (LOCAL 192.168.16.229:16642 REMOTE 192.168.16.248:3084)
16:26:50 P:00 C:000 ID:0002141615 ESTABLISHED RTT CONNECTION (LOCAL 192.168.16.229:16412 REMOTE 192.168.16.248:16422)
16:27:00 P:00 C:000 ID:0002141615 DISCONNECTING G.711 RTP CONNECTION (LOCAL 192.168.16.229:16642 REMOTE 192.168.16.248:3084)
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D.8 Protocol Protocol trace contains the raw protocol messages sent and received by GIII in hexadecimal form. The following is an example of protocol trace. 16:26:49 P:02 C:-- RX: 02 01 00 00 08 02 00 2F 05 A1 04 03 80 90 A3 18 03 A9 83 81 6C 08 00 80 32 33 34 35 30 31 70 07 80 32 33 34 35 30 31 16:26:49 P:02 C:-- TX: 00 01 00 02 08 02 80 2F 02 18 03 A9 83 81
Protocol trace is identified only by the port from which it was retrieved. The channel number is usually encoded in the trace itself. Messages received from the remote end are identified by RX:, whilst messages transmitted by GIII are identified with TX:.
The format of protocol trace is specific to the protocol in use, and its interpretation is outside the scope of this document.
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Appendix E: SIP custom headers E.1 SS7 to SIP mapping headers
This section describes the SIP custom headers that are provided to allow SS7 parameters to be presented in a SIP INVITE.
Aculab-SS7-Calling-Party-Category When this header is presented in the INVITE of an incoming SIP call:
• If the outgoing call leg is ANSI SS7 the content will be used to populate the Calling Party’s Category parameter in the outgoing Initial Address Message, subject to the rules of the route being used.
• If the outgoing call leg is SIP then a copy of this header will be presented unmodified in the outgoing INVITE.
This header will be included in the INVITE presented by an outgoing SIP call if the incoming call is:
• An ANSI SS7 call. The content of the Calling party’s category parameter from the incoming Initial Address Message will be used to populate the header.
• A SIP call that has presented an Aculab-SS7-Calling-Party-Category header in the incoming INVITE. The incoming header will be copied unmodified into the outgoing call.
The following is an example of the header, along with a description of its parameters. Aculab-SS7-Calling-Party-Category: Category=224
Field Description Category This is the Calling party’s category parameter defined in ANSI T1.113-
1995, para 3.8. The value is expressed in decimal form.
Aculab-SS7-Carrier-Identification When this header is presented in the INVITE of an incoming SIP call:
• If the outgoing call leg is ANSI SS7 the content will be used to populate a Carrier identification parameter in the outgoing Initial Address Message.
• If the outgoing call leg is SIP then a copy of this header will be presented unmodified in the outgoing INVITE.
This header will be included in the INVITE presented by an outgoing SIP call if the incoming call is:
• An ANSI SS7 call that contains a Carrier identification parameter in the incoming Initial Address Message. The contents of this parameter will be used to populate the header.
• A SIP call that has presented an Aculab-SS7-Carrier-Identification header in the incoming INVITE. The incoming header will be copied unmodified into the outgoing call.
The following is an example of the header, along with a description of its parameters. Aculab-SS7-Carrier-Identification: ID=2;Plan=1;Digits=e34
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Field Description ID This is the Type of network identification subfield from the Carrier identification
parameter defined in ANSI T1.113-1995, para 3.8A. The value is expressed in decimal form.
Plan This is the Network identification plan subfield from the Carrier identification parameter defined in ANSI T1.113-1995, para 3.8A. The value is expressed in decimal form.
Digits This is a hexadecimal representation of the Digits subfield from the Carrier identification parameter defined in ANSI T1.113-1995, para 3.8A. The correlation between the digits in the above example and their positions in Figure 9A/T1.113.3 of the specification is:
e Digit 1 3 Digit 2 4 Digit 3
Aculab-SS7-Carrier-Selection-Information When this header is presented in the INVITE of an incoming SIP call:
• If the outgoing call leg is ANSI SS7 the content will be used to populate a Carrier selection information parameter in the outgoing Initial Address Message.
• If the outgoing call leg is SIP then a copy of this header will be presented unmodified in the outgoing INVITE.
This header will be included in the INVITE presented by an outgoing SIP call if the incoming call is:
• An ANSI SS7 call that contains a Carrier selection information parameter in the incoming Initial Address Message. The contents of this parameter will be used to populate the header.
• A SIP call that has presented an Aculab-SS7-Carrier-Selection-Information header in the incoming INVITE. The incoming header will be copied unmodified into the outgoing call.
The following is an example of the header, along with a description of its parameters Aculab-SS7-Carrier-Selection-Information: Sel=2
Field Description Sel This is the Carrier selection information parameter defined in ANSI T1.113-
1995, para 3.8B. The value is expressed in decimal form.
Aculab-SS7-Charge-Number When this header is presented in the INVITE of an incoming SIP call:
• If the outgoing call leg is ANSI SS7 the content will be used to populate a Charge number parameter in the outgoing Initial Address Message.
• If the outgoing call leg is SIP then a copy of this header will be presented unmodified in the outgoing INVITE.
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This header will be included in the INVITE presented by an outgoing SIP call if the incoming call is:
• A SIP call that has presented an Aculab-SS7-Charge-Number header in the incoming INVITE. The incoming header will be copied unmodified into the outgoing call.
The following is an example of the header, along with a description of its parameters. Aculab-SS7-Charge-Number: NOA=3;Plan=1;Digits=1234bc7
Field Description NOA This is the Nature of address indicator subfield from the Charge number
parameter defined in ANSI T1.113-1995, para 3.10. The value is expressed in decimal form.
Plan This is the Numbering plan indicator subfield from the Charge number parameter defined in ANSI T1.113-1995, para 3.10. The value is expressed in decimal form.
Digits This is a hexadecimal representation of the Address signal subfield from the Charge number parameter defined in ANSI T1.113-1995, para 3.10. The correlation between the digits in the above example and their positions in Figure 10B/T1.113.3 of the specification is:
1 1st Address signal 2 2nd Address signal 3 3rd Address signal 4 4th Address signal b 5th Address signal c 6th Address signal 7 7th Address signal
Aculab-SS7-Jurisdiction-Information When this header is presented in the INVITE of an incoming SIP call:
• If the outgoing call leg is ANSI SS7 the content will be used to populate a Jurisdiction information parameter in the outgoing Initial Address Message.
• If the outgoing call leg is SIP then a copy of this header will be presented unmodified in the outgoing INVITE.
This header will be included in the INVITE presented by an outgoing SIP call if the incoming call is:
• An ANSI SS7 call that contains a Jurisdiction information parameter in the incoming Initial Address Message. The contents of this parameter will be used to populate the header.
• A SIP call that has presented an Aculab-SS7-Jurisdiction-Information header in the incoming INVITE. The incoming header will be copied unmodified into the outgoing call.
The following is an example of the header, along with a description of its parameters. Aculab-SS7-Jurisdiction-Information: Address=2075d4
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Field Description Address This is a hexadecimal representation of the Address signal subfield from
the Jurisdiction information parameter defined in ANSI T1.113-1995, para 3.23A. The correlation between the digits in the above example and their positions in Figure 21A/T1.113.3 of the specification is:
2 1st Address signal 0 2nd Address signal 7 3rd Address signal 5 4th Address signal d 5th Address signal 4 6th Address signal
Aculab-SS7-Originating-Line-Information When this header is presented in the INVITE of an incoming SIP call:
• If the outgoing call leg is ANSI SS7 the content will be used to populate an Originating line information parameter in the outgoing Initial Address Message.
• If the outgoing call leg is SIP then a copy of this header will be presented unmodified in the outgoing INVITE.
This header will be included in the INVITE presented by an outgoing SIP call if the incoming call is:
• An ANSI SS7 call that contains an Originating line information parameter in the incoming Initial Address Message. The contents of this parameter will be used to populate the header.
• A SIP call that has presented an Aculab-SS7-Originating-Line-Information header in the incoming INVITE. The incoming header will be copied unmodified into the outgoing call.
The following is an example of the header, along with a description of its parameters. Aculab-SS7-Originating-Line-Information: II=34
Field Description II This is the Originating line information parameter defined in ANSI T1.113-
1995, para 3.26A. The value is expressed in decimal form.
Aculab-SS7-Transit-Network-Selection When this header is presented in the INVITE of an incoming SIP call:
• If the outgoing call leg is ANSI SS7 the content will be used to populate a Transit network selection parameter in the outgoing Initial Address Message.
• If the outgoing call leg is SIP then a copy of this header will be presented unmodified in the outgoing INVITE.
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This header will be included in the INVITE presented by an outgoing SIP call if the incoming call is:
• An ANSI SS7 call that contains a Transit network selection parameter in the incoming Initial Address Message. The contents of this parameter will be used to populate the header.
• A SIP call that has presented an Aculab-SS7-Transit-Network-Selection header in the incoming INVITE. The incoming header will be copied unmodified into the outgoing call.
The following is an example of the header, along with a description of its parameters. Aculab-SS7-Transit-Network-Selection: ID=2;Plan=2;Digits=56f8;Circuit=0
Field Description ID This is the Type of network identification subfield from the Transit network
selection parameter defined in ANSI T1.113-1995, para 3.31C. The value is expressed in decimal form.
Plan This is the Network identification plan subfield from the Transit network selection parameter defined in ANSI T1.113-1995, para 3.31C. The value is expressed in decimal form.
Digits This is a hexadecimal representation of the Digits subfield from the Transit network selection parameter defined in ANSI T1.113-1995, para 3.31C. The correlation between the digits in the above example and their positions in Figure 27D/T1.113.3 of the specification is:
5 Digit 1 6 Digit 2 f Digit 3 8 Digit 4 Circuit This is the Circuit code subfield from the Transit network selection parameter
defined in ANSI T1.113-1995, para 3.31C. The value is expressed in decimal form.
E.2 SIP call recovery headers This section describes the SIP custom headers that are provided to support SIP call recovery. See the SIP call recovery section for more information.
Aculab-Call-Recovery This header is presented in the INVITE of an outgoing SIP call when that call is a recovery call. The purpose of this header is to allow a remote system to identify a recovery call, and the failed call that it is replacing. This header has a single parameter, which is the SIP Call-ID: header presented by the failed SIP call that is being replaced. At present the maximum length of this field is 51 characters. The following is an example of the header. Aculab-Call-Recovery: [email protected]
E.3 Advice of charge headers This section describes the SIP custom headers that are provided to allow advice of charge information to be passed to and from a SIP call.
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Aculab-Charge This header is presented in an INFO message when the end-to-end call is in the connected state. An INFO message can only contain one Aculab-Charge header. Where multiple headers are presented the behaviour is undefined. When interworking from Finnish ISUP to SIP, if the incoming Finnish ISUP port has AOC Mapping enabled then any Aculab-Charge header presented by the outgoing call will be mapped into Finish ISUP Metering pulse message and forwarded to the incoming side. When interworking from SIP to Finnish ISUP, if the incoming SIP port has AOC Mapping enabled then a Metering pulse message presented by the outgoing call will be mapped into an Aculab-Charge header and forwarded to the SIP side. The following is an example of the header. Aculab-Charge: Type=sfs-meter;Pulses=15;Tariff=1a
Field Description Type This is a string that indicates how the header will be decoded, and
must always be set to sfs-meter. Pulses This is the decimal representation of the value retrieved from, or to
be used in, the Number of metering pulses parameter defined in SFS 5779:1996, para 3.62 and SFS 5869:2001 para 3.101. Valid values for pulses are 1 to 15 inclusive. The behaviour for values outside of this range is undefined.
Tariff This is a hexadecimal representation of the data retrieved from, or to be used in the Tariff Type parameter defined in SFS 5779:1996, para 3.63 and SFS 5869:2001 para 3.100. This is an optional field, and if not specified a value of 0 (tariff type not indicated) will be used.
E.4 Call information headers This section describes the SIP custom headers that are provided to allow fault finding information to be presented to remote systems. When making an outgoing call GIII will present this header in the SIP INVITE.
When receiving an incoming call GIII will present this header in each of the following backward messages:
183 Session Progress 180 Ringing 200 OK BYE
The following is an example of the header. Aculab-Call-Information: Call-ID=0000001234;Trunk=05;Channel=015
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Field Description Call-ID This is the ID: against which the call will be recorded in the GIII call
control trace. Trunk When the header is presented by an outgoing SIP call, this is the
number of the GIII trunk carrying the incoming call leg. When the header is returned to an incoming SIP call, this is the number of the GIII trunk selected to carry the outgoing call leg. When the header is presented in a BYE returned to the incoming call leg this will contain the value –1 if the call was rejected before a trunk was selected for the outgoing leg (for example the call could not be routed).
Channel When the header is presented by an outgoing SIP call, this is the number of the channel within the above trunk carrying the incoming call leg.
When the header is returned to an incoming SIP call, this is the number of the channel within the above trunk selected to carry the outgoing call leg.
When the header is presented in a BYE returned to the incoming call leg this will contain the value –1 if the call was rejected before a trunk and/or channel for the outgoing leg was selected (for example the call could not be routed).
E.5 Call control and interworking headers This section describes the SIP custom headers that are provided to allow miscellaneous call control parameters to be passed between SIP endpoints.
Aculab-Functional-Number This header is used to pass a Functional Number parameter between GIII and a remote endpoint. See the Function Number mapping section for an explanation. GIII will present the header in an outgoing SIP INVITE, and will process the header when received in an incoming SIP INVITE. The header is not supported in any other SIP message. The following is an example of the header: Aculab-Functional-Number: 01908273800
The Functional Number may be between 1 and 31 characters in length, and may include any combination of the hexadecimal digits 0 to E.
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Appendix F: Adding, removing, or replacing Prosody T cards F.1 Adding or removing a Prosody T card
Changing the number of Prosody T cards
The Node Administration – Card Setup page allows the number of Prosody T cards in the system to be changed. If you are adding or removing cards, then please ensure that the cards are contiguous starting from card 1 position, and installation is done according to the instructions in MANXXXX Aculab GIII gateway 2U carrier grade chassis (R730) Installation guide. This document is available from the Aculab website, or from the GIII website in Node Administration – Documentation. The order in which the activities are carried out is not important, i.e. change the number of Prosody T cards, then change the settings via the card setup page, or do this in reverse.
NOTE If the number of cards configured is changed first, and the number is more than the physical cards currently present, then after rebooting the role will take longer than normal to become active due to waiting for a non-existent card, and timing out.
F.2 Replacing a Prosody T card The GIII system does not require any changes to its configuration to support a replaced Prosody T card. The card must be replaced according to instructions in MANXXXX Aculab GIII gateway 2U carrier grade chassis (R730) Installation guide. This document is available from the Aculab website, or from the GIII website in Node Administration – Documentation.
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Appendix G: System Console The system console can be accessed via the iDRAC, or by connecting a monitor to the vga port.
Console home screen showing GIII information
Using the system console, the GIII can be powered off, restarted, or have its admin interface configurations settings changed.
G.1 Setting IP Addresses
Console screen for configuring the Admin interface settings
The admin IP address information can be configured from the Set IP screen.
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G.2 System Console errors The system console may display warning text while starting up, but if a serious problem is seen then a red box would be displayed with the problem.
System Console showing missing dongle error
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Appendix H: Website Errors
The image above is the page seen if there has been a problem detected. If more information is available then it will be shown. Examples can be seen below:
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The image above is seen if for some reason a unit was asked to join a cluster, and on
restarting it cannot contact the cluster to join. As the message describes, the unit must be restarted, and the join can be attempted again.
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Appendix I: An example of a role workflow Whilst the document describes in detail the individual parts of using and configuring a role, this appendix item will show these parts combined to give an example of a role workflow. Create a new role, based upon an imported role, but change the number of trunks to be 32.
Go to the Cluster Administration – Backup & Restore page. Press the Choose File button to select the backed up role to restore, but restore as a role named tmp. Press the submit button.
Go to the Role Configuration – Role Management page and you will see the new tmp role as an accordion.
Select Create a new Role accordion if not already selected. Add a new role, but based upon the tmp role just restored. Change the number of trunks to 32. Press the Submit button.
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The new role will be displayed.
Now before making any further changes create an initial revision as a known starting point, and add some information into the Comments.
Press the Create revision with current settings checkbox, and press Submit.
Press the Comments tab, and add some information, and press Submit.
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The restored role tmp is no longer needed, so delete it completely.
Go to the Role Configuration – Role Management page. Select the tmp accordion, and press the Delete Role tab.
Press the Delete entire Role checkbox, and press Submit. A confirmation dialog will be presented, press Confirm.
Modify trunk 31 to be ETS300 User-end, and trunk 32 to be ETS300 Network-end.
Go to the Role Configuration – Trunks page and select the TDM 31 tab.
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Change the TDM Protocol to be ETS300 User-end by pressing and selecting the required protocol, and press Submit.
Do the same for TDM 32, but select ETS300 Network-end protocol, remembering to press Submit. Make a new revision from the ETS300 changes, and remove the active role from this device, and make Piccadilly the active role.
Go to the Role Configuration – Role Management page. Select the Piccadilly accordion, and press the Apply Role tab (if not already selected). Press the Create revision with current setting checkbox and press submit. Also (optionally) add a new comment to the role about the change made.
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Now to remove the active role (if any) select the role accordion which is showing active:
Press the Remove Role from Node checkbox and press submit. A confirmation dialog will be presented. Press the Confirm button.
Press the Piccadilly role accordion, and select the current node from the Node to apply Role to select options, and then press Submit.
The node will be in Maintenance mode when a role is first applied, so go to the Maintenance – Node Maintenance Mode page and press the Maintenance mode checkbox to uncheck, then press Submit.
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Now view the System Summary page by going to Status Monitors – System Summary.
The Node should be green if the role is running and not in maintenance mode. Now want to see what configuration was being used for TDM 31 and TDM 32 in revision 1. Go to the Role Configuration – Role Management page and select the Piccadilly role accordion if not already selected. Press the Display revision tab. Select revision 1 from the Revision to display selection and press Submit.
The accordion will display that it is showing revision 1, but it is using selected revision 2.
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Go to the Role Configuration – Trunks page, and press the TDM 31 and TDM 32 tabs to view what configuration was being used in revision 1.
Now mistakenly configure the TDM 31 protocol to be DMS 100 User-end, and press Submit. There will be a system message explaining that you cannot write the configuration data to a saved revision. This is because you are showing revision 1, and cannot change its settings.
Now we want to restore the settings we were using in revision 1, so we can make different changes based on those settings. First we need to change to be displaying the current settings so that we can modify them. Go to the Role Configuration – Role Management page and select the Piccadilly role accordion if not already selected. Press the Display revision tab. Select Current settings from the Revision to display selection and press Submit.
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Go to the Role Configuration – Role Management page and select the Piccadilly accordion if not already selected. Press the Revert Role tab and select revision 1 from the Revision to revert to select items, and press Submit.
Now the current state is the same as revision 1. Any modifications made from this point will be made based on revision 1 configuration. This means that creating the next revision will not have any of the changes made in revision 2. Configure TDM 29 as AT&T User-end, and TDM 30 as AT&T Network-end. Make these settings active. Go to the Role Configuration – Trunks page and select the TDM 29 and TDM 30 tabs, changing the protocol as required, remembering to press Submit before changing tabs. Now go to the Role Configuration – Role Management page and press the Create revision with current settings checkbox so that it is checked. The Select created revision is automatically checked, which is what we want in this case.
There is a warning message displayed which says that applying this change may cause calls to be lost on trunks 29, 30, 31, and 32. This is because we were running ETS 300 on 31/32, and now will not be running any firmware on these, and we are changing 29/30 to be AT&T. Pressing Submit will display a further confirmation dialog. Press Confirm to make the changes active.
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Go to the Status Monitors – TDM Trunks page and once the firmware is active the display should update to show the changes on trunks 29-32.
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Appendix J: Web Services To access the web services help information you would access, using a tool like curl or wget; https://<giii-ip-address>/ws/help with the username/password of a user configured on the GIII device. NOTE: The default user/password is admin/password. i.e. curl -k --user admin:password https://172.21.158.1/ws/help Below is a transcript of the web services help: GIII Web Services Help
======================
This document describes web services that use secure HTTP to transfer
data to/from the GIII. There are a number of methods/tools that can be
used to run web service commands. This document focuses on the use of
the 'curl' tool (https://en.wikipedia.org/wiki/curl).
Command line syntax:
<Text inside angle brackets> - Placeholder for which users must supply
a value.
[Text inside square brackets] - Optional items.
Text outside brackets - Text that must be entered as shown.
Prefixing ampersands with backslash:
When running curl commands, most shells will require ampersand characters
to be prefixed with backslash characters. For example, the curl command to
backup a role and specify an optional password is defined as:
curl -k --user <username>:<password> https://<admin_ip>/ws/roles?role=<backup_role_name>[&password=<zip_password>] -o <zip_filename>
If running this command and specifying a password, it is likely "&password"
will need to be prefixed with an backslash thus the command required may
resemble:
curl -k --user alice:passwd https://10.10.10.10/ws/roles?role=my_role\&password=my_zip_passwd -o my_zip_file.zip
Backup and restore - Roles
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==========================
Commands:
To list roles:
curl -k --user <username>:<password> https://<admin_ip>/ws/roles?list
Returns a JSON object containing "status" and "result" data.
"status" with non-200 value
Indicates the web service failed and "result" describes the failure.
"status: 200"
Indicates that "result" contains a list of role names
To backup a role:
curl -k --user <username>:<password> https://<admin_ip>/ws/roles?role=<backup_role_name>[&password=<zip_password>] -o <zip_filename>
On success, the role specified (backup_role_name) will be zipped up and
saved to the location specified (zip_filename)
To restore a role:
curl -k --user <username>:<password> -F file=@<zip_filename> https://<admin_ip>/ws/roles?[password=<zip_password>]
curl -k --user <username>:<password> -T <zip_filename> https://<admin_ip>/ws/roles?[password=<zip_password]
Returns a JSON object containing "status" and "result" data.
"status" with non-200 value
Indicates the web service failed and "result" describes the failure.
"status: 200"
Indicates the web service was successful and "result" contains
the name of the restored role.
To restore a role to a different name:
curl -k --user <username>:<password> -F file=@<zip_filename> https://<admin_ip>/ws/roles?role=<restore_role_name>[&password=<zip_password>]
curl -k --user <username>:<password> -T <zip_filename> https://<admin_ip>/ws/roles?role=<restore_role_name>[&password=<zip_password>]
Returns a JSON object containing "status" and "result" data.
"status" with non-200 value
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Indicates the web service failed and "result" describes the failure.
"status: 200"
Indicates the web service was successful and "result" contains
the name of the restored role.
Where:
username = Login username
password = Login password
admin_ip = Admin IP address
zip_filename = Filename of zip file containing role backup data
zip_password = Password of the zip file containing role backup data
backup_role_name = Name of the role to backup
restore_role_name = Name given to the role once it has been restored
Backup and restore - System
===========================
Introduction:
A restore_preview predicts GIII changes that will occur if a specified
backup file is restored using restore_apply_start. A restore_preview
returns a description of the prediction and a checksum value based on
the prediction. The checksum may be included as a restore_apply_start
parameter to ensure GIII has not been changed since being previewed.
If restore_apply_start is called and a checksum is specified, the
backup file will only be restored if there have been no GIII changes
since obtaining the checksum. If restore_apply_start is called with
the checksum parameter omitted, the backup file will be restored
regardless of outcome.
Commands:
To backup a system (type=backup):
curl -k --user <username>:<password> https://<admin_ip>/ws/system_backup_and_restore?type=backup[&password=<zip_password>] -o <zip_filename>
On success, a system backup file will be zipped up and
saved to the location specified (zip_filename)
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To preview a system restore (type=restore_preview):
curl -k --user <username>:<password> -F file=@<zip_filename> https://<admin_ip>/ws/system_backup_and_restore?type=restore_preview[&password=<zip_password>]
curl -k --user <username>:<password> -T <zip_filename> https://<admin_ip>/ws/system_backup_and_restore?type=restore_preview[&password=<zip_password>]
Returns a JSON object containing "status" and "result" data.
"status" with non-200 value
Indicates the web service failed and "result" describes the failure.
"status: 200"
Indicates that "result" contains a JSON object (dictionary)
containing the following:
"checksum" : string
The string is a checksum that may be included as a
restore_apply_start parameter to ensure GIII has not been changed
since being previewed
"warnings" : string
The string is a verbose description of the predicted changes
that will occur if the file is restored.
To start applying a system restore (type=restore_apply_start):
curl -k --user <username>:<password> -F file=@<zip_filename> https://<admin_ip>/ws/system_backup_and_restore?type=restore_apply_start[&password=<zip_password>][&checksum=<checksum>]
curl -k --user <username>:<password> -T <zip_filename> https://<admin_ip>/ws/system_backup_and_restore?type=restore_apply_start[&password=<zip_password>][&checksum=<checksum>]
Returns a JSON object containing "status" and "result" data.
"status" with non-200 value
Indicates the web service failed and "result" describes the failure.
"status: 200"
Indicates the system restore has been started and "result" contains a
uid_str.
To get the status of a system restore (type=restore_apply_status):
curl -k --user <username>:<password> https://<admin_ip>/ws/system_backup_and_restore?type=restore_apply_status&restore_uid=<uid_str>
Returns a JSON object containing "status" and "result" data.
"status" with non-200 value
Indicates the web service failed and "result" describes the failure.
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"status: 200"
Indicates that "result" contains a JSON object (dictionary)
containing one or more of the following:
"result" : "success" | "failure" | "uid_unknown" | "in_progress"
"success" - Indicates the system restore has completed
successfully, and "info" may contain additional
information
"failure" - Indicates that the system restore has failed,
and "info" will contain additional information
"uid_unknown" - Indicates the restore_uid was invalid or expired
"in_progress" - Indicates the system restore is running and the
result is not currently available
"info" : string
The string gives further information, and will be empty if there is
none. Present only if "result" is "success" or "failure".
Where:
username = Login username
password = Login password
admin_ip = Admin IP address
zip_filename = Filename of zip file containing system backup data
zip_password = Password of the zip file containing system backup data
checksum = Checksum obtained by performing a restore_preview
uid_str = A string returned after a successful call with
type=restore_apply_start. The uid_str can be used to
get the status of a system restore.
Administer CDRs
===============
Introduction:
CDR searches may take time to complete. For this reason, searching CDRs
is a 2 stage process. The 1st stage is to start the search using
cdr_search_start. The 2nd stage is to repeatedly check for the results
using cdr_search_get_results. While the search is running,
cdr_search_get_results will return an "in_progress" result. When the
search has finished, cdr_search_get_results will return a "success"
result together with a list of CDR records found.
When a search is started (cdr_search_start), users are given a search_uid
that exists for the lifetime of the search. The search_uid is used to
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stop a running search (cdr_search_kill) or obtain results for that search
(cdr_search_get_results). The lifetime of a search_uid ends when:
- The search_uid is used to kill a running search (cdr_search_kill)
- The search_uid is used to get the results of a completed search
(cdr_search_get_results with a "success" result).
- The search_uid is left to expire (To ensure the search_uid does not
expire, cdr_search_get_results should be called at least once every
30 seconds).
Commands:
To start a CDR search:
curl -k --user <username>:<password> -d <search_parms> https://<admin_ip>/ws/cdr_search_start
Returns a JSON object containing "status" and "result" data.
"status" with non-200 value
Indicates the web service failed and "result" describes the failure.
"status: 200"
Indicates the CDR search has been started and "result" contains a
search_uid.
To kill a running CDR search:
curl -k --user <username>:<password> -d search_uid=<uid_str> https://<admin_ip>/ws/cdr_search_kill
Returns a JSON object containing "status" and "result" data.
"status" with non-200 value
Indicates the web service failed and "result" describes the failure.
"status: 200"
Indicates that "result" contains a JSON object (dictionary)
containing one or more of the following:
"result" : "success" | "uid_unknown" | "partial_success"
"success" - Indicates the search has been stopped
"uid_unknown" - Indicates the search_uid was invalid
"partial_success" - Indicates it was not possible to stop
the search on all nodes. This can happen
should network communication fail during
a CDR search on clustered nodes.
"qual_info" : string
The string gives qualifying information which may be empty if
there is none. Present only if "result" is "success".
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To get the CDR search results:
curl -k --user <username>:<password> https://<admin_ip>/ws/cdr_search_get_results?search_uid=<uid_str>
Returns a JSON object containing "status" and "result" data.
"status" with non-200 value
Indicates the web service failed and "result" describes the failure.
"status: 200"
Indicates that "result" contains a JSON object (dictionary)
containing one or more of the following:
"result" : "success" | "uid_unknown" | "in_progress"
"success" - Indicates "record_list" contains a list
of CDR records
"uid_unknown" - Indicates the search_uid was invalid
"in_progress" - Indicates the search is running and the
results are not currently available
"record_list" : list
The list of CDR records found during a search. Present only
if "result" is "success"
"hit_max_records" : True | False
True if the number of CDR records found on any node has reached
the maximum, else False. Present only if "result" is "success".
"qual_info" : string
The string gives qualifying information, and will be empty if
there is none. Present only if "result" is "success".
Where:
username = Login username
password = Login password
admin_ip = Admin IP address
search_parms = A combination of ampersand separated "Search Parameters"
(As described in the section below).
uid_str = A string returned after a successful call to cdr_search_start.
The uid_str can be used to kill an running CDR search
(By calling cdr_search_kill) or to retrieve results from
a CDR search (By calling cdr_search_get_results)
Search Parameters:
field=<cdr_query_val>
This is a compulsory parameter - The cdr_query_val can be any of the
values listed in the "CDR Query Field Table" below. Multiple field
parameters may be specified in the the search_parms. For example:
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to specify both the call_id and call_duration values, the search_parms
may resemble:
field=call_id&field=call_duration&start_date=23/08/17
start_date=<start_date_str>
This is a compulsory parameter - The format of start_date_str is:
yyyy-mm-dd
end_date=<end_date_str>
This is a optional parameter - The format of end_date_str is:
yyyy-mm-dd
If this parameter is omitted, the search will be performed from
state_date to the current date.
start_time=<start_time_str>
This is a optional parameter - The format of start_time_str is:
hh:mm:ss
If this parameter is omitted, start_time_str will be set to:
00:00:00
end_time=<end_time_str>
This is a optional parameter - The format of end_time_str is:
hh:mm:ss
If this parameter is omitted, end_time_str will be set to:
23:59:59
search_on_time=<search_on_val>
This is a optional parameter - search_on_val may be set to one of
following values: "call_start" or "call_end".
If set to call_start, the search will only return CDRs details for
calls that started within the specified date and time window.
If set to call_end, the search will only return CDRs details for
calls that terminated within the specified date and time window.
If this parameter is omitted, search_on_val will be set to:
call_start
in_orig_addr=<in_orig_str>
This is a optional parameter - Search for CDRs matching a
particular incoming originating address. The format of in_orig_str
is a string of numeric digits (For TDM calls) or a username
(For SIP calls).
If this parameter is omitted, CDRs with any incoming originating
address will be searched for.
in_dest_addr=<in_dest_str>
This is a optional parameter - Search for CDRs matching a
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particular incoming destination address. The format of in_dest_str
is a string of numeric digits (For TDM calls) or a username
(For SIP calls).
If this parameter is omitted, CDRs with any incoming destination
address will be searched for.
call_termination_reason=<termination_reason_val>
This is a optional parameter - Used to search
for calls that were terminated with a particular reason.
termination_reason_val is set to one of the values (A single
character) listed in the "CDR Call Termination Reason Table".
If this parameter is omitted, CDRs terminated for any reason will
be searched for.
raw_cause=<raw_cause_str>
This is a optional parameter - Used to search for calls
that were terminated with a particular raw clearing cause.
The format of raw_cause_str is a string of numeric digits.
If this parameter is omitted, CDRs terminated for any raw
clearing cause will be searched for.
generic_cause=<gen_cause_str>
This is a optional parameter - Used to search for calls
that were terminated with a particular generic clearing cause.
The format of gen_cause_str is a string of numeric digits.
If this parameter is omitted, CDRs terminated for any generic
clearing cause will be searched for.
max_records=<max_records_str>
This is a optional parameter - The maximum number of CDR records
that will be returned by each node from a search. The format of
max_records_str is a string of numeric digits from 1 to 30000.
If this parameter is omitted, max_records_str is set to 30000.
first_record=<first_record_str>
This is a optional parameter - The first CDR record that will be
returned by each node from a search. The format of
first_record_str is a string of numeric digits, 0 or greater.
If this parameter is omitted, first_record_str is set to 0.
node=<node_str>
This is a optional parameter - Used to specify a particular node
on which CDRs should only be searched for. The format of node_str
is the G number of node (I.e. node=G6982).
If this parameter is omitted, CDRs across all clustered nodes will
be searched for.
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CDR Query Field Table:
Value Description
----- -----------
call_answer Call answer timestamp
call_answer_date Call answer date
call_answer_time Call answer time
call_direction Call direction
call_duration Call duration
call_end Call end timestamp
call_end_date Call end date
call_end_time Call end time
call_id Call ID
call_redirection_address Call redirection address
call_start Call start timestamp
call_start_date Call start date
call_start_time Call start time
call_termination_reason Call termination reason
connected_address Connected address
generic_call_termination_cause_code Generic call termination cause code
incoming_codec_type Incoming codec type
incoming_destination_address Incoming destination address
incoming_enquiry_call_timeslot Enquiry call timeslot
incoming_enquiry_call_trunk Enquiry call port number
incoming_ip_dsp_stream Incoming IP DSP stream
incoming_ip_dsp_timeslot Incoming IP DSP timeslot
incoming_ip_network_destination Incoming IP network destination
incoming_ip_network_origin Incoming IP network origin
incoming_originating_address Incoming originating address
incoming_service_indicator_code Incoming service indicator code
incoming_signalling_type Incoming signalling type
incoming_timeslot Incoming timeslot
incoming_trunk_number Incoming port number
outgoing_codec_type Outgoing codec type
outgoing_destination_address Outgoing destination address
outgoing_ip_dsp_stream Outgoing IP DSP stream
outgoing_ip_dsp_timeslot Outgoing IP DSP timeslot
outgoing_ip_network_destination Outgoing IP network destination
outgoing_ip_network_origin Outgoing IP network origin
outgoing_originating_address Outgoing originating address
outgoing_signalling_type Outgoing signalling type
outgoing_timeslot Outgoing timeslot
outgoing_trunk_number Outgoing port number
raw_call_termination_cause_code Raw call termination cause code
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recovered_call Recovered call
rerouting_count Re-routing count
CDR Call Termination Reason Table:
Value Description
----- -----------
C The call was terminated because the service and/or media type requested is not supported
B The call was disconnected when a timeslot carrying one of the call legs was blocked
E The outgoing call was rejected with user busy
I Terminated by the incomplete dialling timer
H The caller hung up before the call reached the ringing stage
M Terminated by the maximum ringing timer
L Terminated by the disconnect calls on layer 1 alarm feature
O The Aculab call control software rejected the request to make an outgoing call
N The outgoing call was rejected for an unspecified reason
P The call was terminated due to out of procedure action by the remote endpoint
S The call was terminated to carry out a firmware download, or because the system is closing down
R Routing failed
U The outgoing call was rejected with number unobtainable
T The call was disconnected because the timeslot carrying it was reset
Y A successful call was terminated normally
X A ringing call was terminated before being answered
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Appendix K: LDAP Authentication examples Below are some very basic examples of configuration used with different LDAP servers. These examples may not use encrypted connections which would not be advisable in a production environment.
K.1 Active Directory
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If you wanted to login using the email address, then change the Login attribute to userPrincipalName.
There must be at least one group mapping present to be able to log in a user, otherwise the authorization phase will fail. The group should have the users required to log in to the GIII added as members.
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K.2 Fedora 389DS
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The 389ds LDAP server used here does not have the ‘memberOf’ functionality installed, and has a schema which gives us information like:
# Managers, Groups, 389ds.lab
dn: cn=Managers,ou=Groups,dc=389ds,dc=lab
uniqueMember: cn=Directory Manager
uniqueMember: cn=admin
uniqueMember: cn=john
ou: groups
description: People who can manage entries
cn: Managers
objectClass: top
objectClass: groupOfUniqueNames
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K.3 ApacheDS
Note that the Use DN for group membership is checked as the schema is using the DN:
# HMS Bounty, crews, groups, sevenSeas
dn: cn=HMS Bounty,ou=crews,ou=groups,o=sevenSeas
uniquemember: cn=William Bligh,ou=people,o=sevenSeas
uniquemember: cn=Fletcher Christian,ou=people,o=sevenSeas
uniquemember: cn=John Fryer,ou=people,o=sevenSeas
cn: HMS Bounty
objectclass: top
objectclass: groupOfUniqueNames
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