Description: How to configure Tata Sip PRI Trunk in vicidial ,asterisk ,goatuodail,
Overview about TATA SIP trunk
The TATA SIP trunk will need a need a dedicated lan connectivity to your dialler,
so in dialer or pbx you need two Ethernet Port to connect to TATA network and a Dedicated network for phone registration and agent logins, as per the below pic.
eth1 is used to connect to TATA network, and eth0 is used to connect to customer lan network were ipphones , agents pc will be connected.
Once the above setup is ready follow the below steps, if any support ping me in skype: striker24x7
STEP 1: Configure the TATA network ip to eth1 of dialer/pbx
Either you can use the below command to configure the tata ip in eth1
ifconfig eth1 10.0.70.18/30
OR edit the ifcfg-eth1 file by using below command
vi /etc/sysconfig/network-scripts/ifcfg-eth1
then enter
IPADDR=10.0.70.18
PREFIX=30
ONBOOT=YES
*** Note : the ip address 10.0.70.18 might be differ for each customer.
STEP 2: Configure the route in linux to reach tata network
as SIP trunk network is on different subnet than the customer IP, a static route needed
to reach the TATA SBC .
Goto
vi /etc/sysconfig/network-scripts/route-eth1
and add below line
10.0.70.2/32 via 10.0.70.71
Once file save restart the network by typing
service network restart
then type route -n to make sure route is added
STEP 3: Asterisk sip settings.
Goto vi /etc/asteris/sip.conf and make below changes
defaultexpiry=600
progressinband=yes
STEP 4: Sip Carrier settings
For vicidial/goautodial you can use the admin utility- Carrier settings
for plain asterisk enter the below details in sip.conf.
register => 66810000:1234:66810000@10.0.70.2/66810000
[tatasip]
type=friend
disallow=all
allow=alaw
allow=ulaw
allow=g729
host=10.0.70.2 ;this is tata SBC ip
dtmfmode=rfc2833
nat=no
canreinvite=no
context=tata
insecure=invite,port
STEP 5: Dialplan to dialout via tata trunk
For vicidial / goautodial you can use the ADMIN-Carrier- Dialplan entry
For asterisk users you need to enter in extensions.conf under default context
For vicidial/goautodial Dialers dialplan
exten => _7X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _7X.,n,SipAddHeader(P-Preferred-Identity: <sip:66810000@10.0.70.18>)
exten => _7X.,n,Progress()
exten => _7X.,n,Dial(SIP/0${EXTEN:1}@tatasip)
exten => _7X.,n,Hangup()
For asterisk/Freepbx Dialers dialplan
once above entry done, do a asterisk reload by typing asterisk -rx "reload"
STEP 6 : Dialplan to Receive inbound calls form tatasip trunk
Enter the below dialplan after the last line of extensions.conf (vi /etc/asterisk/extensions.conf)
For Vicidial/goautodial use the below dialplan in extensions.conf
[tata]
exten => _X.,1,Goto(s,1)
exten => s,1,Noop(Let us look deeper into the soul of the invite)
exten => s,n,Set(pseudodid=${SIP_HEADER(To)})
exten => s,n,Set(pseudodid=${CUT(pseudodid,@,1)})
exten => s,n,Set(pseudodid=${CUT(pseudodid,:,2)})
exten => s,n,Goto(trunkinbound,${pseudodid},1)
Then in Vicidial GUI create DID's under INBOUND tab with your respective Tata DID no
for me its 66810000
For People using Freepbx/elastix/ or plain asterisk who use from-pstn as inbound context use the below dialplan in extensions.conf
[tata]
exten => _X.,1,Goto(s,1)
exten => s,1,Noop(Let us look deeper into the soul of the invite)
exten => s,n,Set(pseudodid=${SIP_HEADER(To)})
exten => s,n,Set(pseudodid=${CUT(pseudodid,@,1)})
exten => s,n,Set(pseudodid=${CUT(pseudodid,:,2)})
exten => s,n,Goto(from-pstn,${pseudodid},1)
How to configure TATA sip trunk in vicidial
STEP 5: Dialplan to dialout via tata trunk
For vicidial / goautodial you can use the ADMIN-Carrier- Dialplan entry
For asterisk users you need to enter in extensions.conf under default context
For vicidial/goautodial Dialers dialplan
exten => _7X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _7X.,n,SipAddHeader(P-Preferred-Identity: <sip:66810000@10.0.70.18>)
exten => _7X.,n,Progress()
exten => _7X.,n,Dial(SIP/0${EXTEN:1}@tatasip)
exten => _7X.,n,Hangup()
For asterisk/Freepbx Dialers dialplan
exten => _7X.,1,SipAddHeader(P-Preferred-Identity: <sip:66810000@10.0.70.18>)
exten => _7X.,n,Progress()
exten => _7X.,n,Dial(SIP/0${EXTEN:1}@tatasip)
exten => _7X.,n,Hangup()
exten => _7X.,n,Progress()
exten => _7X.,n,Dial(SIP/0${EXTEN:1}@tatasip)
exten => _7X.,n,Hangup()
once above entry done, do a asterisk reload by typing asterisk -rx "reload"
STEP 6 : Dialplan to Receive inbound calls form tatasip trunk
Enter the below dialplan after the last line of extensions.conf (vi /etc/asterisk/extensions.conf)
For Vicidial/goautodial use the below dialplan in extensions.conf
[tata]
exten => _X.,1,Goto(s,1)
exten => s,1,Noop(Let us look deeper into the soul of the invite)
exten => s,n,Set(pseudodid=${SIP_HEADER(To)})
exten => s,n,Set(pseudodid=${CUT(pseudodid,@,1)})
exten => s,n,Set(pseudodid=${CUT(pseudodid,:,2)})
exten => s,n,Goto(trunkinbound,${pseudodid},1)
Then in Vicidial GUI create DID's under INBOUND tab with your respective Tata DID no
for me its 66810000
For People using Freepbx/elastix/ or plain asterisk who use from-pstn as inbound context use the below dialplan in extensions.conf
[tata]
exten => _X.,1,Goto(s,1)
exten => s,1,Noop(Let us look deeper into the soul of the invite)
exten => s,n,Set(pseudodid=${SIP_HEADER(To)})
exten => s,n,Set(pseudodid=${CUT(pseudodid,@,1)})
exten => s,n,Set(pseudodid=${CUT(pseudodid,:,2)})
exten => s,n,Goto(from-pstn,${pseudodid},1)