configure Tata Sip Trunk with Asterisk

2 Answer(s)

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

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

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

Answered on January 10, 2021.

This is optional setting:

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()

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)

on January 10, 2021.
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Configure network interfaces for sip trunk

First: Interface LAN was configured by installer itself, then it add mask /24 that must be removed.

iface LAN inet static
    address 192.168.1.247/24
    gateway 192.168.1.254
    # dns-* options are implemented by the resolvconf package, if installed
    dns-nameservers 192.168.1.254 

allow-hotplug SIP
iface SIP inet static
    address 172.xxx.xxx.xxx
    netmask 255.255.255.252

Was modified to:

iface LAN inet static
    address 192.168.1.247
    netmask 255.255.255.0
    gateway 192.168.1.254
    # dns-* options are implemented by the resolvconf package, if installed
    dns-nameservers 192.168.1.254 

allow-hotplug SIP
iface SIP inet static
    address 172.xxx.xxx.IP
    netmask 255.255.255.252

Then at /etc/rc.local must add two static routes

  1. This route sends all trafic to LAN through LAN interface
    route add 192.168.1.0 gw 192.168.1.254
    
  2. This route sends all traffic to my sip server through SIP interface
    route add [SIP SERVR IP] gw 172.xxx.xxx.GW
    

This all done the work.

Note:

172.xxx.xxx.IP is the Ip assigned by telecom provider

172.xxx.xxx.GW is the gateway assigned by telecom provider

[SIP SERVR IP] is the Sip server ip assigned by telecom provider.

Answered on January 10, 2021.
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