hybrid ip pbx february 2014
DESCRIPTION
TRANSCRIPT
Hearty Welcome!
ETERNITY as Hybrid IP-PBX
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
Introduction
Soft Phone
VoIP Phone
VoIP Phone
Mobile
Analog Phone
Internet
PSTN
Introduction
Hybrid IP-PBX means PABX which supports IP Extensions and TDM/Analog Extensions
Hybrid IP-PBX can also have different type of trunks like CO, ISDN, GSM, etc. depends on hardware supported by IP-PBX
ETERNITY supports only SIP protocol for VoIP
Such system has capabilities to convert media between IP and TDM
SIP Resources
ETERNITY VARIANT SIP EXTENSIONS SIP TRUNKS VOIP CHANNELS/CARD
ETERNITY PE3SS
ETERNITY PE3SP
ETERNITY PE6SP
ETERNITY GE6S
ETERNITY GE12S
ETERNITY ME10S
50 16 16
50 16 16
50 16 16
500 16 32
500 16 32
1000 32 32
ETERNITY ME16S 1000 32 32
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
LAN Port Configuration
Name can be assigned just for
identification
Hardware Slot & Port Offset
Customization is not possible
MAC Address of LAN Port
Configure IP Address and Subnet Mask for LAN Port
LAN Port doesn’t support DHCP connection
LAN Port Parameters
LAN Port is available in VoIP server card so all SIP extensions in local network with VoIP card can register without using WAN [Internet]
LAN PORT
WAN Port Configuration
MAC Address of WAN Port
Customization is not possible
Enable/Disable MAC Cloning using this flag
Configure Clone MAC Address
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
What is MAC Cloning?
MAC cloning means to configure new MAC address [MAC-2] for the host without changing existing MAC address [MAC-1]
After doing MAC cloning host sends newly configured MAC address [MAC-2] in Ethernet Frames in place of sending
existing MAC address [MAC-1]
How MAC Cloning works ?
ISP Is Authenticating Host With MAC Address
ISP
Cloned MAC:- 01:1d:1a:02:82:34
Fix MAC:- 02:2d:1c:02:32:45
Why MAC Cloning?
Many times ISP tracks MAC address of host installed at customer premise to authenticate him as valid customer to provide Internet service
Due to this reason customer can access Internet only from single Host
Configuration of MAC Cloning
WAN Port Configuration
Select the internet Connection Type here Options: - Static
- PPPoE - DHCP
If the selected internet Connection Type is
‘PPPoE’,program the User ID, Password and PPPoE service
name here
WAN Port Configuration
If “Static” option is selected for DNS Address Assignment, then program the IP address of DNS and Domain
Name here
Select the DNS Address Assignment option here
(Auto/Static). If the selected option is ‘Auto’ then there is no need to
program the DNS address. It will be automatically assigned by the Service Provider/DHCP server
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
What is Dynamic DNS?
Dynamic DNS means assigning a Domain Name to such host whose IP address changes frequently
Due to facility of DDNS that host can always be accessible from WAN by using same Domain Name
Why Dynamic DNS?
ISP
DHCP Connection
What is system’s
present IP?
1st attempt: 116.72.127.98
2nd attempt: 117.89.97.123
3rd attempt : 115.161.181.183
4th attempt: 118.187.24.89
Why Dynamic DNS?
In this case host will not be accessible always using public IP assigned to it by ISP
When ISP gives Internet connection type as PPPoE or DHCP then IP assigned to router at client site may be changed frequently
It can be resolved by using Dynamic DNS
How DDNS works?
DDNS Server is accessible globally, it keeps details of domain name
and global IP of all customers
DDNS Client Which Gives Update To Server
About Global IP Of Router
Matrixcomsec.dyndns.org : 203.88.143.221
Matrixcomsec.dyndns.org : 115.23.143.241
115.23.143.241 203.88.143.221
Internet Cloud
Dynamic DNS Configuration
Enable/Disable Dynamic DNS here. Enabling this option will help the VoIP
card to inform the SIP clients to pass the information of latest IP assigned to the VoIP card by the DHCP or PPPoE Server
Turn ON this option if the internet Connection type is DHCP or PPPoE
and DDNS option is enabled
DDNS option will be useful only if the Internet Connection Type is DHCP or PPPoE
Dynamic DNS Configuration
Program the ‘Password’ provided by Dyndns.org
here, if the DDNS option is enabled
Program the ‘User-ID’
provided by Dyndns.org here, if the
DDNS option is enabled
Program the Host Name provided by Dyndns.org
here, if the DDNS option is enabled
Dynamic DNS Configuration
Number of request send by the VoIP Card to DDNS
Server for the IP update request. Applicable only if
DDNS is enabled
This option helps the VoIP Card to re-establish the
mapping with the DDNS if the IP update request has not been sent in time by
the VoIP Card
Dynamic DNS Configuration
It shows that VoIP card has successfully sent request to DDNS
server to update router’s public IP
detail in database of DDNS server
Dynamic DNS Configuration
It shows that VoIP card failed to send request to DDNS server to update
router’s public IP
Check configuration Gateway and
DNS IP
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
VoIP Server Domain
With this option when user will send SIP messages then VoIP card will listen for SIP message which is
redirected to programmed Domain Name and WAN IP Address
VoIP Server Domain
If client already have fix Domain name purchased from DNS service provider then that DNS can be configured here
That DNS will be assigned to VoIP server card
All SIP users from WAN can register to this DNS assigned to IP server card
Mostly public IP mapped to this Domain remains fixed that make it different from DDNS
VoIP Server Domain
Click on “Advance” to get detailed parameters
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
STUN
Simple Traversal of UDP through NATs
UDP (User Datagram Protocol) is a Network Protocol for Transmission of Data
STUN allows VoIP Card to work behind Asymmetric NAT
STUN Client (VoIP Card) sends a request to STUN Server
STUN
Router
STUN Server
STUN Client
STUN Client requests STUN Server
Server updates with IP address used by router and open port to client
Client uses this information of IP address and free port from the server to ETERNITY NE
STUN will not work if the Router’s NAT Type is ‘Symmetric’
Illustration of STUN
Router with public IP STUN server
SIP server
Invite 203.88.142.119:5063
200 OK
ACK RTP
RTP
STUN
Select this options only if you have not forwarded the SIP & RTP
Listening Port in the Router. If flag is “Enabled” then system will use the SIP & RTP listening Port information
provided by the STUN Server
Program the STUN Server IP Address here
Program the STUN Server port here
STUN Configuration for SIP TRUNK and Extensions
STUN will be effective only when “Source Port IP Address” option is selected as “Use IP Address Fetched using STUN”
Source Port IP Address can be configured in “SIP Extension General Parameters” and in “SIP Trunk Parameters”
STUN Configuration for SIP Extensions
STUN Configuration for SIP TRUNK
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
VLAN (Virtual LAN)
VLAN is good option for big
network to give high data speed
VLAN (Virtual LAN)
Priority can be defined to SIP packets on Layer2 level
Priority can be defined to RTP packets on Layer2 level
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
VoIP Port Parameters - QoS
This field defines the priority Bit for all the SIP message sent
by VoIP card. Range 00-63
This field defines the priority bit for all the RTP message sent by VoIP card.
Range 00-63
Public IP
INTERNET
115.118.161.163
Users can directly access the device over
internet
(Public IP Address)
Router’s Public IP Address
Public IP Address of the NAT Router behind which VoIP card is installed. Program the Router’s IP only if the option of Router’s
Public IP is selected in ‘SIP Trunk Settings’
Router’s Public IP Address for SIP Trunk
Router’s Public IP Address for SIP Extension
VoIP Port Parameters
If ETERNITY detects absence of RTP packets till expiry of this timer then it will disconnect the call
VoIP Port Parameters
This much of channels will not be available for SIP extensions
Following number of physical channels reserved for SIP Trunks
VoIP Port Parameters
Enable this flag, this will make the VoIP card to use ‘100rel’ extension along with all the SIP provisional
messages
100rel and SIP PRACK
SIP PRACK (SIP Provision Acknowledgement) is a method to enable reliability for SIP 1XX messages
The Called Party answer the PRACK by 200OK and PRACK is only for 1XX
messages other than 100 Trying
Generally PRACK message flows from Calling Party to Called Party
100rel and SIP PRACK
To get more reliability on SIP messages
Enabling this flag will make the VoIP card to send the SIP messages over TCP
VoIP Port Parameters
SIP Listening and Source Port for UDP
Range 1025-65535
RTP Listening and Source Port Range 1025-65278
SIP Listening and Source Port for TCP
Range 1025-65535
SIP Listening and Source Port for TLS Range 1025-65535
VoIP Port Parameters
This timer should be less then UDP binding timer in router
(Range 001-999 seconds)
Enable this flag to keep UDP binding refreshing in NAT router “Notify” or “Register” message can
be sent to keep UDP binding alive in router
VoIP Port Parameters
This timer should be less then TCP binding timer in router (Range 0001-
9999 seconds)
Enable this flag to keep TCP binding refreshing in NAT router
VoIP Port Parameters
This is the timer for which system waits for a response from the
called party after sending INVITE message. On expiry of this timer,
system terminates the call
This timer starts on the receipt of the provisional response
receipt from the called party and stops at the final receipt of
response. On this timer’s expiry, system disconnects the
call
This timer is applicable to all request, system will clear transaction after expiry of timer if it will not receive
response for sent request
VoIP Port Parameters
LED2 on VoIP card will show status of SIP trunk defined here
(Range 01-32)
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
What is SIP Extension?
Like any SLT and DKP, ETERNITY can have extensions that can be connected via internet/ LAN
ETERNITY VoIP Card can work as SIP Server to register SIP extensions from LAN, WAN or VPN
SIP Extensions
SIP Extensions Features
Hold Other Extension
Change User Status
Call Budget Toggle Two Calls
Publish/IM CUG
DND (Do Not Disturb)
Dial Operator Transfer Held Call Selective Port Access
Set message on (SLT/DKP)
Self Ring Test
Call Forward Alarm Reminder
Dial Floor Service Group
Room Monitor on Idle DKP Port
Use 3 Parties/ Dial In/Multi Party Conference
Personal/ Global Directory
Group Call Pick – Up
Voice Guided Alarm/ Reminder
Auto Call Back on Busy/Ringing Call
Use Walk – In Feature
Recall to last Caller
Use Keyboard Macro
Selective Call Pick – Up
Dial SA/SE Command
Park Other/SIP Extension
Voice Message Notification
Account Code
CLI Restriction
Forced Answer Feature
Retrieve Parked Call
Use Busy Lamp Field
Emergency Number
Features Other Extension can Use with SIP Extension
Set Call Forward on SIP Extension
Apply Raid on Busy on Busy SIP Extension
Park SIP Extension
Use Walk – In for SIP Extension
CO Call Waiting
Configure SIP Extension in Hotel/Enterprise Installation Wizard
Apply DND Override on SIP Extension
Hold SIP Extension
Retrieve Parked Call
Set Hotline on Extension
DISA
Call Supervision
Apply IR (Interrupt Request), BI (Barge-In)
Transfer Held SIP External Call
Selective port Access
Background Music
Hot Desk
SIP Extensions Settings
Configuring SIP Extensions
Server End Client End
SIP ID
Authentication ID
Authentication Password
SIP ID
Authentication ID
Authentication Password
Registrar Server Address
SIP Extension Settings
Assign VoIP Software Port Number here
Use this flag to enable SIP extension
Configure Name of SIP extension user here, it will be displayed as
caller ID during internal calls (maximum of 18 characters)
If It is “Blank” Then called party will not get name received from INVITE As CLI
SIP Extension Settings
Configure SIP ID using which SIP extension user will register with registrar of VoIP card (it can be up to 6 digits, 0 to 9, * and # are valid
digits)
All extensions can call to SIP extension
user using this number
SIP Extension Settings
VoIP card’s registrar will use this ID to authenticate SIP user (it can be configured
up to 6 digits, 0 to 9 , * and # are valid digits)
It will not be applicable If All
“Authentication” options are disabled
SIP Extension Settings
VoIP card’s registrar will use this password to authenticate SIP user (it can
be configured up to 24 digits, 0 to 9, * and # are valid digits)
Default password:
1234
SIP Extension Settings
SIP extension user can make/receive maximum this number of calls
simultaneously (Range 01-10)
SIP Extension Settings
By enabling these flags you can authenticate SIP users during
these different request messages
SIP Extension Authentication
ETERNITY VoIP card uses MD5 algorithm to authenticate SIP users by using Authentication ID and Password
During specific events ETERNITY VoIP card can authenticate users by asking them to send Authentication Id and Password configured in SIP user device
[SIP Phone]
Types of Authentication
ETERNITY VoIP card can Authenticate SIP user during following SIP messages
REGISTER Request
INVITE Request SUBSCRIBE
Request
Voice Mail subscription
BLF subscription
Presence Subscription
SIP Extension Settings
By enabling this flag you can get notification on call states of al the
phones with the same SIP ID at different locations
SIP Extension Settings
Enable this option to get Voice Mail Notification on
VoIP phone
VoIP phone should support Voice Mail Notification
feature
SIP Extension Settings
To allow this SIP extension user to view the status of the
availability of other SIP enabled terminals, this flag should be
enable
SIP Extension Settings
SIP Extension Settings
BLF Key in SPARSH VP248
LED Glowing RED: User Busy LED Blinking RED: User Ringing
SIP Extension Settings
By enabling this flag VoIP Phone users can publish their availability status
By enabling it VoIP server will ask for
authentication details from SIP users when
receives PUBLISH message
SIP Extension Settings
By enabling this flag VoIP Phone users can see availability status of
other SIP/TDM users
Other VoIP/TDM users should publish
their availability status to use this
feature
SIP Extension Settings
Soft phone user 3304 is subscribing status of 3301
and 3303 (All SIP users registered with
VoIP Server Card)
It is obvious that 3301 and 3303 are publishing their availability status to VoIP
Server Card
SIP Extension Settings
It completely depends on SIP user that which type of availability status it
can Publish or Subscribe
Publish Status of DKP & SLT
DKP and SLT users can also publish their availability status by applying simple commands from their phones
Following is sequence to dial
commands
Off Hook SLT/DKP Phone
Dial 104
Feature Tone---User Password
Enter code [Range 0 to 9]
Publish Status of DKP & SLT
Code Status
0 Absent
1 Present
2 Auto Detect
3 Away
4 On the Phone
5 Do Not Disturb
6 I am on Mobile
7 In Meeting
8 Out for Meal
9 Out of Office
Publish Status on Soft Phone
SIP Extension Settings
Different SIP hardware parameters can be assigned
to different SIP users
Same like SLT and DKP users following Features can be assigned to SIP users also
SIP Extension Settings
Same like SLT and DKP users Call Pick Up group can be assigned to SIP users also
If system is configured to use in hotel mode then SIP extension can
also be configured as “Guest”
SIP Extension General Parameters
This is Name which you have assigned to VoIP server card
Showing Hardware Slot and Port of VoIP card
SIP Extension General Parameters
Select here which IP should be considered as source IP when VoIP card communicates with
SIP users
SIP Extension General Parameters
VoIP card will receive registration request from SIP users only between this timer
interval
Registration Timer Configured in SIP
users must be between this values
SIP Extension General Parameters
Following Private Key is used to encrypt SIP message
[MD5 Authentication] It can be up to any 24 ASCII character
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
SIP Trunk Configuration
VoIP calls can be initiated after suitable programming of SIP Trunks
ETERNITY supports 2 types of SIP trunks: Peer to Peer and Proxy
Peer-to-Peer Calling
203.88.143.218 204.88.142.218
Internet TCP/IP
Making a VoIP call directly to the destination without any intervention of any mediator is called
peer-to-peer calling.
You just need to know the called party’s IP address.
Peer-to-Peer Calling
Peer-to-Peer Calling
Select Peer-To-Peer in the SIP Trunk Mode You can select either
Trunk or Station
If you select Trunk, then it will follow the
Trunk Feature Template as per the
SIP trunk
Configuration: Peer-to-Peer Calling
Peer-to-Peer Calling
If you select Station, then it will follow the direct landing on specified
extension
Calling 205
201 205 207
Proxy Calling
Making VoIP calls through proxy server is called proxy calling
Proxy Server: abc.com
Client 2 SIP ID 402
Client 3 SIP ID 403
401 calling 402
Client 1 SIP ID 401
Requirement for Proxy Calling
Proxy server authenticates the clients for outgoing calls through it
What is required for
authentication?
SIP ID
Authentication ID
Authentication Password
Registrar Server Address
Registrar Server port
Proxy Calling
Program SIP ID here as per given by ITSP Program Registrar Server
address here. To be obtained from ITSP
Program Registrar Server port here. To be obtained from ITSP
(1025-65535)
Proxy Calling
It is the timer after which
request has been sent again
Registration retry if registration request is not acknowledged
User ID & password given by ITSP for authentication
Proxy Calling
Enable outbound proxy from here
Enter outbound proxy sever address here
Enter outbound proxy sever port here
Proxy Calling
Define SIP hardware
template here
Define TFT here if the SIP Trunk entity is “SIP
Client”
SIP Trunk Properties
Define Cost Factor here
Used in Gateway Application (01-64)
Enable RCOC here
SBFT, SAFT are applicable on the SIP Trunk if the SIP Trunk entity is P2P
SIP Trunk Properties
By enabling these flags you can authenticate SIP user during
these different request messages
SIP Trunk Properties
Enable if you want to send CLI on SIP trunk
Enable if you want to accept IC calls without
CLI
SIP Trunk Properties
Define Source port IP address here
Enable/disable Digest Authentication
Enable Symmetric RTP from here
Why Symmetric RTP?
Symmetric RTP can be used in firewalls, debugging and troubleshooting.
Generally it is useful to resolve bidirectional speech problems.
Many firewalls, NATs, RTP implementations don’t work on asymmetric RTP but require symmetric RTP.
Digest Authentication
It is a security feature which is used by VoIP card during peer to peer incoming call
On any incoming SIP call, VoIP card will check the authenticity of the SIP user by Authentication ID and Password
This authentication is done by using Digest Authentication Table
If authentication doesn’t match, VoIP card will reject the incoming SIP call
Digest Authentication
SIP users configured with following User ID and Password will only be allowed to access ETERNITY during
Peer To Peer calling
Digest Authentication
Enable this flag in SIP Trunk
parameters
SIP Trunk Properties
Select default transport for outgoing messages i.e. UDP, TCP
or TLS
UDP v/s TCP v/s TLS
UDP TCP TLS
UDP is connectionless and acknowledgement less protocol (DNS, VOIP)
TCP is connection oriented & provides acknowledgement (WWW, FTP, E-mail)
TLS is connection oriented protocol & provides acknowledgement
Used for time sensitive applications
TCP requires more bandwidth than UDP
TLS requires more bandwidth than TCP
Used for servers that answer small queries from huge number f clients
Used in the applications where secure connection is required & data loss should be less
When secure transportation is to be used
SIP Trunk Properties
Define IC ref. ID & OG ref. ID here for DDI mapping
Program the value of
pause timer here (1-9)
SIP Trunk Properties
Used during Gateway Application
SIP Trunk Properties
This is set as per the requirement of
remote peer
Enable it when SIP trunk is to be generated for an
invite without SDP
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
SIP Hardware Template
Select Preference for Vocoders according to compatibility of SIP users and expected VoIP
quality
Bandwidth requirement for each Vocoder is different
SIP Hardware Template
When G.723 negotiated then selected Bit Rate will be applied to send RTP (5.3 or 6.3 Kbps)
SIP Hardware Template
By enabling this flag Silent RTP packets will not be sent during conversation
Used for efficient usage of available
bandwidth
SIP Hardware Template
TX and RX speech level can be changed from here
SIP Hardware Template
SIP Hardware Template
Select DTMF type from following options
- RTP (RFC 2833) - SIP Info - In band
Same DTMF option must be configured in SIP user device
If RTP(RFC 2833) is
selected then this should be
configured
SIP Hardware Template
By enabling this flag Echo Cancellation will be activated when SIP users are talking to
Stations/Trunks [Analog/Digital]
This parameters can be configured according to strength of Echo required to
be cancelled
Separate options available for analog
and digital interfaces
Jitter Buffer
By considering packet network jitter can be defined as variation of delay in receiving packets
To resolve this problem the mechanism used in VoIP device is called “Jitter Buffer”
VoIP device stores packets according to Jitter Buffer timer supported by it to maintain common delay between successive packets before processing them
for regeneration of voice
SIP Hardware Template
Select “Static” option for network having precise delay but when delay is
not fixed always from network side then select option “Dynamic” Configure Jitter Buffer
timer here
Configure it for Dynamic Jitter Buffer
option
SIP Hardware Template
Select Protocol for FAX over IP. Following are options:
- T.38 (UDPTL) - T.38 (RTP)
- Pass Through
FAX Parameters is customized when protocol is selected as T.38
SIP Hardware Template
FAX Parameters customization when protocol selected as Pass Through
White List IP Address
ETERNITY supports security on Transport Layer
By enabling this security option ETERNITY VoIP card will accept incoming
traffic only from those VoIP devices whose IP address is configured in White List Table
White List IP Address
Configure here, IP Addresses of devices from where you want to receive incoming call traffic on VoIP card
Enable this flag to use IP level security
Static Routing
Some times customer have multiple routers in their network to connect their multiple sites using MPLS (Multiprotocol Label Switching)/ Frame Relay
In same network there can be distinct routers to connect to Internet
In such network scenario to connect VoIP devices at multiple sites using point to point connectivity there is need of some mechanism to route the calls from different
router according to IP address of different destination VoIP devices
Static Routing
Static Routing
Agenda
Introduction
LAN/ WAN Port Configuration
Mac Cloning
Dynamic DNS
VoIP Server Domain
STUN
VLAN
VoIP Port Parameters
SIP Extensions
SIP Trunk
SIP Hardware Template
Matrix Extended Phones
Matrix Extended IP Phone
SIP extensions we registered just previously are called as Open SIP Phones. These phones do not work as a DKP.
Matrix provides its proprietary IP phones to register as an Extended IP Phone which will work as it is as a DKP
Matrix SPARSH VP248
Matrix SPARSH MS
Programming Steps-Eternity
Program VoIP Port No., Name, SIP ID, Authentication ID,
ID Authentication
Password for SIP
Programming Steps-Eternity
In the Location menu Enable Matrix
Extended Phone mode and define MAC address of SPARSH VP248
Phone
Programming Steps-Eternity
Note: User can register Matrix Extended IP phones at three different locations, i.e. a single account can be registered on three IP phones
Programming Steps-Eternity
Configure the Master CPU IP address
Programming Steps-Eternity
This port is to be assigned in the VP
phone where server port is
needed
Matrix Extended IP Phone- SPARSH VP248
Matrix Extended IP Phone- SPARSH VP248
Server port is 80
Master CPU IP Address
Matrix Extended IP Phone- SPARSH VP248
Matrix Extended IP Phone- SPARSH MS
SPARSH MS is a mobile softphone client for android smartphones and iPhones for consistent in-office experience.
You can use Wi-Fi or cellular networks to connect to the system while working from office, home or travelling to any location.
There is a flexibility to reach to office users with direct extension number dialing.
Download Matrix SPARSH MS from play store if you have android device or from apple store if you have iPhone.
Matrix Extended IP Phone- SPARSH MS
Configuring Matrix Extended IP Phone
Matrix Extended IP Phone- SPARSH MS
Video Calling through VoIP
Matrix provides video calling facility also with VoIP calling only.
The phones we use for example
Video Calling with SPARSH M2S
Download Matrix SPARSH M2S from play store if you have android device or from apple store if you have iPhone.
All the settings of SPARSH M2S is similar to SPARSH MS for the server settings at ETERNITY server end
Video Calling with SPARSH M2S
After you install the application of SPARSH M2S; let us suppose you are using an android tablet for video calling
You will have to enter the credentials as per
the server settings (Extended phone type
settings)
Video Calling with SPARSH M2S
User is registered properly
XYZ is calling another SIP
extension 615
Option to start video
call
Option to start audio
call
Option to send IM
Video Calling with SPARSH M2S
Option to make a video call
Here it will be your video
Video Calling with SPARSH M2S
During an audio call, you can
switch it over a video call by selecting this
option
Options: Video Calling with SPARSH M2S
Video Calling with Bria Soft phone
Option to start video
Enter the credentials in the SIP account of
Bria phone
Video Calling with Bria Soft phone
Video Calling with Linphone
Video Calling with Linphone
Video Calling with Linphone
Enter the SIP extension number
whom you want to call
Video calling: Caller’s video
will come