Airtel sip trunk configuration in asterisk vicidial freepbx
Step by step guide to configure the Airtel SIP trunk in asterisk based dialers like vicidial, goautodial,Freepbx,elastix,issabel. If you have purchased the Airtel VOIP trunk which supports SIP protocol and want to configure the same in your asterisk PBX then this Tutorial is for you. In the article I have provide SIP settings required to configure the Airtel VOIP in asterisk and vicidial.
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.
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-eth1then enterIPADDR=10.232.131.172PREFIX=30ONBOOT=YES
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.
ip route add 10.232.130.0/24 via 10.232.131.171 dev eth1
ip route showorroute -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.
vi /etc/hosts
# 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 localhost6127.0.0.1 go.goautodial.org go10.232.130.170 ims.airtel.in
Step 4: SIP Carrier settings.
register => +918060006000@ims.airtel.in:Convate#1:+918060006000@ims.airtel.in@10.232.139.146/+918060006000
[airtelsip]disallow=allallow=alltype=frienddtmfmode=rfc2833qualify=yesnat=force_rport,comediainsecure=invite,porthost=ims.airtel.inusername=+918060006000@ims.airtel.insecret=PASSWORDfromdomain=ims.airtel.indefaultexpirey=120canreinvite=nocontext=trunkinbound ; change this according to your inbound contextmaxexpiry=600progressinband=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/${EXTEN:1}@airtelsip,,tTo)
exten => _9X.,n,Hangup()
For Plain asterisk or freepbx
;exten => _9X.,1,SipAddHeader(P-Preferred-Identity: <sip:+914441231234@ims.airtel.in>)exten => _9X.,1,Progress()exten => _9X.,n,Dial(SIP/${EXTEN:1}@airtelsip)exten => _9X.,n,Hangup()
Conclusion:
Hope this article is useful in configuring your airtel sip trunk in asterisk vicidial freepbx , for professional support reach me at skype:striker24x7
Please share the JIO sip trunk configuration
Jio sip also same method.
if you are facing issue , reach me i will configure jio
BSNL sip
BSNL SIP is similar to JIO SIP trunk , check my blog for jio sip trunk configuration
Can your share your blog
require two trunk same SBC on single server
what is the password for the sip gateway for airtel..?
you need to ask airtel ,password is unique might be 1234 or 0000
Please provide contact number for airtel
Please provide the phone number that we can contact with. I searched for the contacts, no help.
sorry i dont any number
check your email they might given the details
they dont have the functionality it seems, i checked with the customer care.
Hey! Can you pleaseeee help me configure airtel SIP? I just got a connection with them. I want to make calls on PC and android using SIP. I can't buy a landline.
I use windows. I have all the details. I can't ping ims.airtel.in and lots of settings on router page have changed, I can't understand properly. Thank you so much!
hi
this tutorial is for configuring PRI sip trunk.
hope you are using the fiber home broadband with voice.
i have no idea in configuring that.
Hey! My previous message has disappeared.
I'm using airtel xstrem fiber with voice.
If I can ping to ims.airtel.in, I can use any SIP client to make calls.
For that, I have to create a route with IP address, I don't know how.
Hope you can help me. I'll reward you.
Thank you!
What will be the password for airtel SIP?
ask your provider they will provide you.
mostly its 0000 or 1234.
After I have done this, how would I test the setup?
in asterisk cli check the sip peer status
sip show peer
sip show registry
finally make calls
how to handle maximum concurrent calls
if you are using vicidial then you can set MAX trunk under server settings.
if plain asterisk you can set call-limit=
to limit max calls via this trunk
Hi i am following above steps but getting registration error. can u suuprt
are you using airtel PRI sip trunk?
reach me on skype for support ; Skypeid: striker24x7
Hi, was anyone of you able to make Airtel SIP work with PJSIP Channel Driver on Asterisk? I did the settings but the call disconnects in 30 seconds (due to callsetup failure).
I do not face any such issues with chan_sip.so channel driver.
How to a setup host on switchvox
register => +918060006000@ims.airtel.in:Convate#1:+918060006000@ims.airtel.in@10.232.139.146/+918060006000
From where did you get ip address 10.232.139.146
did not changing in vicidial after done changes in vici Dialer