sip trunk 2 ip-pbx user guide asterisk - clocoinc.com · 5 2．purchase/settings in web portal for...

SIP Trunk 2 IP-PBX User Guide （Asterisk）

2

Index

1. SIP Trunk 2 Overview　　　……………………………………………………… 3

2. Purchase/Settings in Web Portal ……………………………… 5

3. Configuration Example of your IP-PBX ……………………………… 12

4. Technical Data ……………………………… 24

SIP Trunk 2 is a next genera2on IP phone service that connects to PBX making an external line call which is compa2ble to Asterisk, Aspire X IP-‐PBX. ＜SIP Trunk 2 FEATURE HIGHLIGHTS＞ ■ Compa2ble to Asterisk, Aspire X PBX. ■ Op2ons for “ Authen2ca2on Method” are:

•  Password Authen2ca2on •  Authen2ca2on with IP Address •  Authen2ca2on using both IP Address and Password.

■ CPS （Call Per Second) has been significantly improved from normal SIP trunk. *Our Cloud PBX Recording Op2on is currently not supported by SIP trunk 2 (If you need the recording op2on, please Contact us) ===== Verified IP-‐PBX ===== ・Asterisk 　Asterisk PBX/1.4.x 　Asterisk PBX 1.6.x 　Asterisk PBX 1.8.x 　Asterisk PBX 11 　Asterisk PBX 12 ・Aspire X 　IP3WW-‐32VOIPDB-‐A1 　version: 05.01 *IP-‐PBX versions not listed above are not fully supported by SIP trunk 2. ======================== ※Please permit on your firewall incoming network traffic from our VoIP server IP addresses with 5060, 10000~20000 UDP ports. Our Server IP address list 221.243.8.194 101.110.51.82

１．SIP Trunk 2 Overview

3

Ext. 200 Ext. 201

4

１．SIP Trunk 2 Overview

To:<sip:[email protected]>

Recipient number is set “To header” and “Alert-‐Into” in SIP messages for Incoming call. See sec2on 4 ”Technical Data" for more details.

From: <sip:[email protected]>

Caller ID must be set “From header” for outgoing call. See sec2on 4 ”Technical Data" for more details.

Image 1. Configura2on Diagram of Incoming/Outgoing Calls

xxx.xxx.xxx.xxx SIP Trunk 2

Your IP-PBX

DID: 0312123434 DID: 0312345678

0000.0000.0000.0000

*In case of Japanese toll free numbers such as prefix 0120, 0800 and 0570, you should set its background number showing in Phone Number List of the web portal. ex.) A number enclosed in parentheses is its background number. 0120****** [03******]

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２．Purchase/Settings in Web Portal

For purchasing SIP Trunk 2, access the UI of our IP-PBX. Buy additional SIP trunk channel for 2 or more simultaneous external calls. ＜SIP Trunk 2 Purchase Screen>

①　Select “Purchase” at the top menu and choose ”Purchase Unique” in Circle Management Page ②　Select quantity of SIP trunk 2 ③　Click “Add to Cart” to proceed for your purchase

③

①

②

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Purchase phone number here *At least one phone number will be needed for external phone calls through SIP Trunk ＜Phone Number Purchase Screen>

①　Select “Purchase” at the top menu and choose ”Purchase Phone Number” in Circle Management Page ②　On the Purchase Phone Number page, find your desired phone number by clicking “Search” button. Add to cart and select “Your Cart” to proceed.

①

②

7


＜SIP Trunk 2 List＞

①　Select “SIP Trunk List” to open all your SIP trunk account ②　Select the icon under “Detail” for detailed settings of SIP Trunk 　(See next page) ③ Your unique is used as client user ID of your user PBX end

①② ③

0000123456

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＜SIP Trunk 2 Detailed Selngs ・ Password Authen2ca2on＞

①　Login server name of SIP Trunk 2 ②　Unique is used as client user ID of your user PBX end. ③　Item “Name” is where you can name/rename your SIP Trunk account. ④　Select your desired authen2ca2on method from 　 “Password Authen2ca2on” or “Authen2ca2on with IP Address” or 　 “Authen2ca2on using both IP Address and Password” ⑤　Enter your terminal password is used as client user password of your PBX end. ⑥　Set mul2ple call count. It’s 1 by default. Purchase “Addi2onal 1 channel for 　 SIP Trunk 2” if you need more than 2 concurrent calls.

xxx.xxx.xxx.xxx ①② ③ ④ ⑤ ⑥

0000123456

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＜SIP Trunk 2 Detailed Selngs ・ Authen2ca2on with IP Address＞

①　Login server name of SIP Trunk 2 ②　Unique is used as client user ID of your user PBX end. ③　Item “Name” is where you can name/rename your SIP Trunk account. ④　Select your desired authentication method from “Password Authentication” or “Authentication with IP Address” or “Authentication using both IP Address and Password” ⑤　Enter a public IP address / a port number of your IP-PBX *You can add multiple IP addresses/ports from “+Insert” button. ⑥　Your IP-PBX will receive incoming call if ticked. *If unticked it will work only for outgoing calls. ⑦　Set multiple call count. It’s 1 by default. Purchase “Additional 1 channel for SIP Trunk 2” if you need more than 2 concurrent calls.

xxx.xxx.xxx.xxx ①② ③ ④

⑤ ⑥

⑦

0000123456

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＜SIP Trunk 2 Detailed Settings ・ Authentication using both IP Address and Password>

①　Login server name of SIP Trunk 2 ②　Unique is used as client user ID of your user PBX end. ③　Item “Name” is where you can name/rename your SIP Trunk account. ④　Select your desired authentication method from “Password Authentication” or “Authentication with IP Address” or “Authentication using both IP Address and Password” ⑤　Enter your terminal password is used as client user password of your PBX end. ⑥　Enter a public IP address of your IP-PBX. ⑦　Set multiple call count. It’s 1 by default. Purchase “Additional 1 channel for SIP Trunk 2” if you need more than 2 concurrent calls.

①② ③ ④ ⑤ ⑥ ⑦

xxx.xxx.xxx.xxx

0000123456

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Select phone number(s) you desire to assign to SIP Trunk 2 ＜Phone Number List>

①　Click “Phone Number List” to open your Phone Number List. ②　Select SIP Trunk 2 unique for phone number(s) you desire to assign for it

②

①

〔0000123456〕

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３．Configuration Example of your IP-PBX

3.1. Configura4on Example in Asterisk [Account Example] Unique: 0000123456 Password: password DIDs: 0312345678 , 0312123434 Extensions: 200, 201 Login Server: xxx.xxx.xxx.xxx ※login the web portal to confirm your login server. [SeMngs Example] Incoming call for 0312345678 is to be arrived at Ext. 200. Incoming call for 0312123434 is to be arrived at Ext. 201. Outgoing call from a phone with Ext. 200 is to be called with CallerID: 0312345678 Outgoing call from a phone with Ext. 201 is to be called with CallerID: 0312123434 ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; sip.conf (for either password or IP address with password authen2ca2on) ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp register => 0000123456:password@siptr [siptr] type=friend username=0000123456 secret=password context=inbound canreinvite=no host=xxx.xxx.xxx.xxx insecure=port,invite disallow=all allow=ulaw qualify=yes nat=yes ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 ;<see also next page for the rest seMngs of sip.conf>

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; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; sip.conf (for either password or IP address with password authen2ca2on) ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [200] type=friend username=200 secret=200pass host=dynamic context=outbound-‐1 [201] type=friend username=201 secret=201pass host=dynamic context=outbound-‐2 ;<see also next page for sip.conf for IP address authen4ca4on>

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; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; sip.conf (for IP address authen2ca2on) ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp [siptr] type=friend context=inbound canreinvite=no host=xxx.xxx.xxx.xxx insecure=port,invite disallow=all allow=ulaw qualify=yes nat=yes ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 [peer1] type=friend context=inbound host=221.243.8.194 nat=yes [peer2] type=friend context=inbound host=101.110.51.82 nat=yes [200] type=friend username=200 secret=200pass host=dynamic context=outbound-‐1 [201] type=friend username=201 secret=201pass host=dynamic context=outbound-‐2

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; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; extensions.conf ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] writeprotect=no priorityjumping=yes [inbound] exten => 0312345678,1, Dial(SIP/200,120,t) exten => 0312345678,2,Conges2on exten => 0312345678,102,Busy exten => 0312123434,1, Dial(SIP/201,120,t) exten => 0312123434,2,Conges2on exten => 0312123434,102,Busy [outbound-‐1] exten => _0., 1,Set(CALLERID(num)= 0312345678 exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Conges2on exten => _0.,104,Busy exten => _1., 1,Set(CALLERID(num)= 0312345678 exten => _1., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _1., 3,Conges2on exten => _1.,104,Busy ;prefix 1xx is for special (external) phone numbers such as 117, 177 and so on. exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Conges2on exten => _ XXX, 102,Busy ; XXX represents 3 digit-‐extensions. Please adjust digit number as yours. ;<see also next page for the rest seMngs of extensions.conf>

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[outbound-‐2] exten => _0., 1,Set(CALLERID(num)= 0312123434) exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Conges2on exten => _0.,104,Busy exten => _1., 1,Set(CALLERID(num)= 0312123434) exten => _1., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _1., 3,Conges2on exten => _1.,104,Busy ;prefix 1xx is for special (external) phone numbers such as 117, 177 and so on. exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Conges2on exten => _ XXX, 102,Busy ; XXX represents 3 digit-‐extensions. Please adjust digit number as yours.

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Group １: Max multiple count 　2 Extensions 　201 ～ 202 Phone Numbers 　03-1234-5678

Group ２: Max multiple count 　3 Extensions 　301 ～ 302 Phone Numbers 　03-1212-3434

3.2. Configura4on Example to limit mul4ple call count for each extension group in Asterisk. [SeMngs Example] Set max mul2ple call count (for external calls) as 2 for Group 1 Set max mul2ple call count (for external calls) as 3 for Group 2 ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; sip.conf (for either password or IP address with password authen2ca2on) ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 context=extd port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp register=>0000123456:[email protected]/0000123456 [0000123456] type=friend username=0000123456 secret=password host=xxx.xxx.xxx.xxx insecure=port,invite context=inbound qualify=yes nat=yes ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 ;<see also next page for the rest seMngs of sip.conf>

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; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; sip.conf (for either password or IP address with password authen2ca2on) ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; Group 1 [201] type=friend context=group1_outbound username=201 secret=password host=dynamic [202] type=friend context=group1_outbound username=202 secret=password host=dynamic ; Group 2 [301] type=friend context=group2_outbound username=301 secret=password host=dynamic [302] type=friend context=group2_outbound username=302 secret=password host=dynamic ;<see also next page for sip.conf for IP address authen4ca4on>

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３．Configuration Example of your IP-PBX ;-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ;sip.conf (IP address authen4ca4on) ;-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 context=extd port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp [siptr] type=friend context=inbound canreinvite=no host= xxx.xxx.xxx.xxx insecure=port,invite disallow=all allow=ulaw qualify=yes nat=yes ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 [peer1] type=friend context=inbound host=221.243.8.194 nat=yes [peer2] type=friend context=inbound host=101.110.51.82 nat=yes ;<see also next page for the rest seMngs of sip.conf>

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;-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ;sip.conf (IP address authen4ca4on) ;-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; Group 1 [201] type=friend context=group1_outbound username=201 secret=password host=dynamic [202] type=friend context=group1_outbound username=202 secret=password host=dynamic ; Group 2 [301] type=friend context=group2_outbound username=301 secret=password host=dynamic [302] type=friend context=group2_outbound username=302 secret=password host=dynamic

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<extensions.conf Example in your Asterisk> ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; extensions.conf ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] writeprotect=no priorityjumping=yes ; Group 1 [inbound] exten => 0312345678,1,NoOp(EXTEN: ${EXTEN}) exten => 0312345678,2,Set(GROUP(CALLS)=GROUP1) exten => 0312345678,3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => 0312345678,4,Set(MAXCALLS=2) exten => 0312345678,5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => 0312345678,6,Dial(SIP/201&SIP/202,120) exten => 0312345678,7,Conges2on exten => 0312345678,106,Busy ; Group 2 exten => 0312123434,1,NoOp(EXTEN: ${EXTEN}) exten => 0312123434,2,Set(GROUP(CALLS)=GROUP2) exten => 0312123434,3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)}) exten => 0312123434,4,Set(MAXCALLS=3) exten => 0312123434,5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => 0312123434,6,Dial(SIP/301&SIP/302,120) exten => 0312123434,7,Conges2on exten => 0312123434,106,Busy ;<see also next page for the rest seMngs of extensions.conf>

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<extensions.conf Example in your Asterisk> ; Group 1 [group1_outbound] exten => _0., 1,Set(CALLERID(num)=0312345678) exten => _0., 2,Set(CALLERID(name)=GROUP1) exten => _0., 3,Set(GROUP(CALLS)=GROUP1) exten => _0., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => _0., 5,Set(MAXCALLS=2) exten => _0., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => _0., 7,Dial(SIP/${EXTEN}@0000123456,120) exten => _0., 8,Conges2on exten => _0.,106,Busy exten => _1., 1,Set(CALLERID(num)=0312345678) exten => _1., 2,Set(CALLERID(name)=GROUP1) exten => _1., 3,Set(GROUP(CALLS)=GROUP1) exten => _1., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => _1., 5,Set(MAXCALLS=2) exten => _1., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => _1., 7,Dial(SIP/${EXTEN}@0000123456,120) exten => _1., 8,Conges2on exten => _0.,106,Busy exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Conges2on exten => _ XXX, 102,Busy ; Group 2 [group2_outbound] exten => _0., 1,Set(CALLERID(num)= 0312123434) exten => _0., 2,Set(CALLERID(name)=GROUP2) exten => _0., 3,Set(GROUP(CALLS)=GROUP2) exten => _0., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)}) exten => _0., 5,Set(MAXCALLS=3) exten => _0., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => _0., 7,Dial(SIP/${EXTEN}@0000123456,120) exten => _0., 8,Conges2on exten => _0.,106,Busy exten => _1., 1,Set(CALLERID(num)= 0312123434) exten => _1., 2,Set(CALLERID(name)=GROUP2) exten => _1., 3,Set(GROUP(CALLS)=GROUP2) exten => _1., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)}) exten => _1., 5,Set(MAXCALLS=3) exten => _1., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => _1., 7,Dial(SIP/${EXTEN}@0000123456,120) exten => _1., 8,Conges2on exten => _1.,106,Busy exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Conges2on exten => _ XXX, 102,Busy

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４．Technical Data

4.1. SIP REGISTER message： ■ Sending REGISTER message Is required to register your ID, IP address and port number for authen2ca2on.

figure 4.1 SIP flow for REGISTER

※Sending REGISTER message is NOT required in case your authen4ca4on method is “Authen4ca4on with IP Address”

REGISTER From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]> Call-ID: [email protected]

your IP-PBX

000.000.000.000 SIP Trunk 2

xxx.xxx.xxx.xxx

1 100 Trying From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]> Call-ID: [email protected]

2 401 Unauthorized From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]>;tag=as245298a3 Call-ID: [email protected]

3 REGISTER(with credential information) From: <sip: [email protected]>;tag=as2031f6e2 To: <sip: [email protected]> Call-ID: [email protected]

4 SIP/2.0 100 Trying From: <sip: [email protected]>;tag=as2031f6e2 To: <sip: [email protected]> Call-ID: [email protected]

5 200 OK From: <sip: [email protected]>;tag=as2031f6e2 To: <sip: [email protected]>;tag=as245298a3 Call-ID: [email protected]

6

Your ID (SIP Trunk 2 unique number

IP address of SIP Trunk 2

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4.1.1 PBX → GUEST REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;rport From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]> Call-‐ID: [email protected] CSeq: 1749 REGISTER User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Expires: 120 Contact: <sip: [email protected]> Event: registra2on Content-‐Length: 0 4.1.2 GUEST →　PBX SIP/2.0 100 Trying Via:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;received=000.000.000.000;rport=5060 From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]> Call-‐ID: [email protected] CSeq: 1749 REGISTER User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip: [email protected]> Content-‐Length: 0 4.1.3 GUEST → PBX SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;received=000.000.000.000;rport=5060 From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]>;tag=as245298a3 Call-‐ID: [email protected] CSeq: 1749 REGISTER User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-‐Authen2cate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx", nonce="3deff552" Content-‐Length: 0

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4.1.4 PBX → GUEST REGISTER sip: xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1db71efa;rport From: <sip: [email protected] >;tag=as2031f6e2 To: <sip: [email protected] > Call-‐ID: [email protected] CSeq: 1750 REGISTER User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Authoriza2on: Digest username="0000123456", realm=" xxx.xxx.xxx.xxx ", algorithm=MD5, uri="sip: xxx.xxx.xxx.xxx", nonce="3deff552", response="bace343abbe8362868dba84e58d7e056", opaque="" Expires: 120 Contact: <sip: [email protected]> Event: registra2on Content-‐Length: 0 4.1.5 GUEST → PBX SIP/2.0 100 Trying Via:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1db71efa;received=000.000.000.000;rport=5060 From: <sip: [email protected] >;tag=as2031f6e2 To: <sip: [email protected] > Call-‐ID: [email protected] CSeq: 1750 REGISTER User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip: [email protected] > Content-‐Length: 0 4.1.6 GUEST → PBX SIP/2.0 200 OK Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1db71efa;received=000.000.000.000;rport=5060 From: <sip: [email protected] >;tag=as2031f6e2 To: <sip: [email protected] >;tag=as245298a3 Call-‐ID: [email protected] CSeq: 1750 REGISTER User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: <sip: [email protected]>;expires=120 Date: Mon, 05 Jul 2010 04:20:13 GMT Content-‐Length: 0


4.2. SIP INVITE message of outgoing call from your IP-‐PBX through SIP Trunk 2 SIP From header should be : From: “Phone Display name”<sip:CallerID@SIP Trunk 2 IP address or FQDN＞

INVITE From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Call-ID: [email protected]

407 Proxy Authentication Required From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65 Call-ID: [email protected]

ACK From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65 Call-ID: [email protected]

INVITE(with credential information) From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Call-ID: [email protected]

100 Trying From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Call-ID: [email protected]

180 Ringing From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-ID: [email protected]

183 Session Progress From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-ID: [email protected]

200 OK From: "aiueo PBX" <[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-ID: [email protected]

ACK From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-ID: [email protected]

BYE From: <sip:[email protected]>;tag=as54380085 To: "aiueo PBX" <[email protected]>;tag=as5dd4eaee Call-ID: [email protected]

200 OK From: <sip:[email protected]>;tag=as54380085 To: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee Call-ID: [email protected]

SIP Trunk 2 xxx.xxx.xxx.xxx

your IP-PBX 000.000.000.000

Phone Display Name CallerID

IP address of SIP Trunk 2 server

starting a call

Terminating a call

1

2

3

4

5

6

7

8

9

10

11

Receiver Phone

Number

27


4.2.1 PBX → GUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;rport From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Date: Fri, 02 Jul 2010 03:05:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-‐Type: applica2on/sdp Content-‐Length: 267 v=0 o=root 22702 22702 IN IP4 000.000.000.000 s=session c=IN IP4 000.000.000.000 t=0 0 m=audio 18572 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-‐event/8000 a=fmtp:101 0-‐16 a=silenceSupp:off -‐ -‐ -‐ -‐ 4.2.2　GUEST → PBX SIP/2.0 407 Proxy Authen2ca2on Required Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65 Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-‐Authen2cate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx ", nonce="23a44cfd" Content-‐Length: 0

28


4.2.3 PBX → GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;rport From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65 Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0 4.2.4 PBX → GUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;rport From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Proxy-‐Authoriza2on: Digest username=" 0000123456 ", realm="xxx.xxx.xxx.xxx ", algorithm=MD5, uri="sip:[email protected]", nonce="23a44cfd", response="cc6c5a668cbd435dee31c767981ff710", opaque="" Date: Fri, 02 Jul 2010 03:05:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-‐Type: applica2on/sdp Content-‐Length: 267 v=0 o=root 22702 22703 IN IP4 000.000.000.000 s=session c=IN IP4 000.000.000.000 t=0 0 m=audio 18572 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-‐event/8000 a=fmtp:101 0-‐16 a=silenceSupp:off -‐ -‐ -‐ -‐

29


4.2.5 GUEST → PBX SIP/2.0 100 Trying Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Length: 0 4.2.6. GUEST → PBX SIP/2.0 180 Ringing Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Length: 0

30


4.2.7 GUEST → PBX SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Type: applica2on/sdp Content-‐Length: 242 v=0 o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18922 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-‐event/8000 a=fmtp:101 0-‐16 a=silenceSupp:off -‐ -‐ -‐ -‐ a=p2me:20 a=sendrecv

31


4.2.8 GUEST → PBX SIP/2.0 200 OK Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Type: applica2on/sdp Content-‐Length: 242 v=0 o=root 4414 4415 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18922 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-‐event/8000 a=fmtp:101 0-‐16 a=silenceSupp:off -‐ -‐ -‐ -‐ a=p2me:20 a=sendrecv 4.2.9 PBX → GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK6c101c7f;rport From: " aiueo PBX " <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0

32


4.2.10 GUEST → PBX BYE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK166bf514;rport From: <sip:[email protected]>;tag=as54380085 To: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee Call-‐ID: [email protected] CSeq: 102 BYE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0 4.2.11. PBX → GUEST SIP/2.0 200 OK Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK166bf514;received=xxx.xxx.xxx.xxx;rport=5060 From: <sip:[email protected]>;tag=as54380085 To: " aiueo PBX " <sip:[email protected]>;tag=as5dd4eaee Call-‐ID: [email protected] CSeq: 102 BYE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[email protected]> Content-‐Length: 0 X-‐Asterisk-‐HangupCause: Normal Clearing

33


4.3. SIP Busy message while outgoing call in case receiver is on another call Busy message sent by SIP Trunk 2 when receiver is currently on another call,

figure 4.3 SIP flow including Busy message while outgoing call


your IP-PBX 000.000.000.000 CallerID


1

2

3

4

5

6

7

INVITE From:　"aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Call-ID: [email protected]

407 Proxy Authentication Required From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as291aca90 Call-ID: [email protected]

ACK From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as291aca90 Call-ID: [email protected]

INVITE(with authentication information) From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Call-ID: [email protected]

100 Trying From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Call-ID: [email protected]

SIP/2.0 486 Busy Here From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as715c3c5e Call-ID: [email protected]

ACK From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as715c3c5e Call-ID: [email protected]

34


4.3.1　　PBX → GUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;rport From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Date: Tue, 06 Jul 2010 10:09:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-‐Type: applica2on/sdp Content-‐Length: 267 v=0 o=root 22702 22702 IN IP4 000.000.000.000 s=session c=IN IP4 000.000.000.000 t=0 0 m=audio 14646 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-‐event/8000 a=fmtp:101 0-‐16 a=silenceSupp:off -‐ -‐ -‐ -‐ 4.3.2　　GUEST→ PBX SIP/2.0 407 Proxy Authen2ca2on Required Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;received=000.000.000.000;rport=5060 From: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as291aca90 Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-‐Authen2cate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx ", nonce="15a6e863" Content-‐Length: 0

35


4.3.3　　PBX → GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;rport From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected] >;tag=as291aca90 Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0 4.3.4　　PBX→GUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;rport From: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Proxy-‐Authoriza2on: Digest username="0000123456", realm="xxx.xxx.xxx.xxx ", algorithm=MD5, uri="sip:[email protected] ", nonce="15a6e863", response="54ebd3bdb5bab4b621f55�d3ffe5e0b", opaque="" Date: Tue, 06 Jul 2010 10:09:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-‐Type: applica2on/sdp Content-‐Length: 267 v=0 o=root 22702 22703 IN IP4 000.000.000.000 s=session c=IN IP4 000.000.000.000 t=0 0 m=audio 14646 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-‐event/8000 a=fmtp:101 0-‐16 a=silenceSupp:off -‐ -‐ -‐ -‐

36


4.3.5　　GUEST→ PBX SIP/2.0 100 Trying Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;received=000.000.000.000;rport=5060 From: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Length: 0 4.3.6. GUEST → PBX SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;received=000.000.000.000;rport=5060 From: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as715c3c5e Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Contact: <sip:[email protected]> Content-‐Length: 0 4.3.7 PBX → GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;rport From: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as715c3c5e Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0

37


4.4. SIP INVITE message of incoming call from SIP Trunk 2 to your IP-‐PBX SIP To header will be : To: <sip:Recipient Phone Number@Your IP PBX IP address＞ *SIP Trunk 2 sets the same recipient phone number to Alert-‐info header as well.

figure 4.4 SIP INVITE flow (incoming)


your IP-PBX 000.000.000.000

IP address of your IP-PBX

1

2

3

4

5

6

CallerID

INVITE From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]> Call-ID: [email protected]

100 Trying From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]> Call-ID: [email protected]

200 OK From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]>;tag=as577af7ce Call-ID: [email protected]

ACK From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]>;tag=as577af7ce Call-ID: [email protected]

BYE From: <sip:[email protected]>;tag=as577af7ce To: “ 080AAAAXXXX " <sip:[email protected]>;tag=as1dddca7a Call-ID: [email protected]

200 OK From: <sip:[email protected]>;tag=as577af7ce To: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a Call-ID: [email protected]

Recipient


Starting a call

Terminating a call

38


4.4.1　　 GUEST→PBX INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK546a1def;rport From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip: 0312345678 @000.000.000.000> Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Date: Fri, 02 Jul 2010 05:41:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-‐Asterisk-‐Guest-‐Tag: 00008 X-‐Asterisk-‐Guest-‐Uniqueid: 1278049293.36 Alert-‐info: 0312345678 Content-‐Type: applica2on/sdp Content-‐Length: 242 v=0 o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 15224 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-‐event/8000 a=fmtp:101 0-‐16 a=silenceSupp:off -‐ -‐ -‐ -‐ a=p2me:20 a=sendrecv 4.4.2. GUEST←PBX SIP/2.0 100 Trying Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK546a1def;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <sip: 080AAAAXXXX @xxx.xxx.xxx.xxx>;tag=as1dddca7a To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[email protected]> Content-‐Length: 0

39


4.4.3. GUEST ←PBX SIP/2.0 200 OK Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK546a1def;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]>;tag=as577af7ce Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[email protected]> Content-‐Type: applica2on/sdp Content-‐Length: 220 v=0 o=root 22702 22702 IN IP4 000.000.000.000 s=session c=IN IP4 000.000.000.000 t=0 0 m=audio 18182 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-‐event/8000 a=fmtp:101 0-‐16 a=silenceSupp:off -‐ -‐ -‐ -‐ 4.4.4 GUEST →PBX ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3afc8626;rport From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]>;tag=as577af7ce Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0

40


4.4.5. GUEST ←PBX BYE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK5b3130a7;rport From: <sip:[email protected]>;tag=as577af7ce To: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a Call-‐ID: [email protected] CSeq: 102 BYE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0 4.4.6. GUEST →PBX SIP/2.0 200 OK Via:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK5b3130a7;received=000.000.000.000;rport=5060 From: <sip:[email protected]>;tag=as577af7ce To: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a Call-‐ID: [email protected] CSeq: 102 BYE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Length: 0

41


4.5. SIP Busy message while incoming call in case receiver is on another call Busy message sent by SIP Trunk 2 when receiver is currently on another call,

figure 4.5 SIP flow including Busy message while incoming call


your IP-PBX 000.000.000.000

IP address of SIP Trunk 2

server

1

2

3

4

CallerID

INVITE From: "080AAAAXXXX" <sip:080AAAAXXXX"@xxx.xxx.xxx.xxx>;tag=as0f1a5f0c To: <sip:[email protected]> Call-ID: [email protected]

100 Trying From: "080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-ID: [email protected]

486 Busy Here From: "080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-ID: [email protected]

ACK From: " 080AAAAXXXX" " <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-ID: [email protected]

Recipient IP address of your IP-PBX

42


4.5.1　　 GUEST → PBX INVITE sip:[email protected] SIP/2.0 Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7�7b8;rport From:" 080AAAAXXXX"<sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Contact: <sip: [email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Date: Fri, 09 Jul 2010 02:27:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-‐Asterisk-‐Guest-‐Tag: 00024 X-‐Asterisk-‐Guest-‐Uniqueid: 1278642466.508 Alert-‐info: 0312345678 Content-‐Type: applica2on/sdp Content-‐Length: 242 v=0 o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 10408 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-‐event/8000 a=fmtp:101 0-‐16 a=silenceSupp:off -‐ -‐ -‐ -‐ a=p2me:20 a=sendrecv 4.5.2　　PBX → GUEST SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7�7b8;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[email protected]> Content-‐Length: 0

43


4.5.3.　　PBX → GUEST SIP/2.0 486 Busy Here Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7�7b8;received=xxx.xxx.xxx.xxx;rport=5060 From: " 080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE Contact: <sip:[email protected]> Content-‐Length: 0 4.5.4.　　GUEST→ PBX Transmilng (NAT) to GUEST ACK sip: [email protected] SIP/2.0 Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7�7b8;rport From:"080AAAAXXXX"<sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0

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