How to configure Airtel sip trunk in asterisk-vicidial-Freepbx

Airtel SIP Trunk configuration in Asterisk

Topic: How to configure Airtel sip trunk in asterisk-vicidial-Freepbx

  Overview: Airtel Sip Trunk

    Airtel SIP trunk is an advanced voice connectivity solution via network, it replaces traditional multiple fixed PSTN with a single Physical line that support 1000 plus calls simultaneous calls.

Airtel SIP Trunk reduces the cost of multiple lines, as well hardware requirement for PRI Trunks.

  Network Connectivity:

Airtel SIP trunk is provided to customer via dedicated SBC gateway and router, for which you need a additional ethernet port on your asterisk server or you need to setup your LAN in the same subnet range provide by airtel. ref below Pic for better understanding.

    eth1 is used to connect to AIRTEL network, and eth0 is used to connect to customer LAN network were ip-phones , agents pc will be connected.

  STEP 1: Configure the AIRTEL network IP to eth1 

        Assign the IP provided by airtel to one of the NIC in you server, for centos based server you may use below commands

ifconfig eth1 10.232.130.172/30 

OR edit the ifcfg-eth1 file and manually enter the ip, OR if you have GUI manager configure manually.

vi /etc/sysconfig/network-scripts/ifcfg-eth1

then enter

IPADDR=10.232.131.172

PREFIX=30

ONBOOT=YES

*** Note: 10.232.131.172  is an example,  you need to enter the airtel ip provided to you.

  STEP  2: Configuring Route in Linux to reach Airtel Network.

This step is required if the AIRTEL SBC IP and your IP is in different subnet then you need a static route to reach the SBC IP.

for eg : my SBC ip is 10.232.130.170

Edit the route-eth1 for manual entry for static route. Note this is for centos based server

 vi /etc/sysconfig/network-scripts/route-eth1

and add below line

10.232.130.0/24 via 10.232.131.171

service network restart

Linux  command to set a static route to SIP proxy ip and media ip

ip route add 10.232.130.0/24 via 10.232.131.171 dev eth1

Command to check the routes

ip route show 

or

route -n

  Step 3: Add static HOST entry 

Airtel SIP Trunks only accepts traffic with header ims.airtel.in,
You need to enter a static host entry for ims.airtel.in with the SBC IP.

go to hosts file and add the host entry

vi /etc/hosts

and the line which is in last line below.

# Do not remove the following line, or various programs

# that require network functionality will fail.

127.0.0.1               localhost.localdomain localhost

::1             localhost6.localdomain6 localhost6

127.0.0.1         go.goautodial.org go

10.232.130.170  ims.airtel.in

save and exit

  Step 4:  SIP Carrier settings.

For Vicidial - goautodial  you can use the admin utility to configure below settings.

For Freepbx you can use the GUI trunk configuration

For Plain Asterisk enter the below details in vi /etc/asterisk/sip.conf

Asterisk Registration String

register => +91441231234:PASSWORD:+91441231234@ims.airtel.in@ims.airtel.in/+91441231234

Airtel SIP Peer settings

[airtelsip]

disallow=all

allow=all

type=friend

dtmfmode=rfc2833

qualify=yes

nat=force_rport,comedia

insecure=invite,port

host=ims.airtel.in

username=+914441231234@ims.airtel.in

secret=PASSWORD

fromdomain=ims.airtel.in

defaultexpirey=120

canreinvite=no

context=trunkinbound      ; change this according to your inbound context

maxexpiry=600

progressinband=yes

  STEP 5: Dialplan entry to Dialout.

For Vicidial /goautodial use the below dialplan

exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,n,SipAddHeader(P-Preferred-Identity: <sip:+914441231234@ims.airtel.in>)
exten => _9X.,n,Progress()
exten => _9X.,n,Dial(SIP/${EXTEN:1}@airtelsip,,tTo)
exten => _9X.,n,Hangup()

For Plain asterisk or freepbx 

exten => _9X.,1,SipAddHeader(P-Preferred-Identity: <sip:+914441231234@ims.airtel.in>)

exten => _9X.,n,Progress()

exten => _9X.,n,Dial(SIP/${EXTEN:1}@airtelsip)

exten => _9X.,n,Hangup()

  Conclusion:

   Hope this article is useful , for professional support reach me at skype:striker24x7