How to configure JIO sip trunk in asterisk
JIO SIP trunk configuration in Asterisk PBX
TOPIC : How to configure JIO sip trunk in asterisk
About JIO SIP trunk :
JIO sip trunk details:
Once you have purchased the new jio sip trunk, you will be provided with below details
Step 1: Network Connection
As show in picture above ,connect your Asterisk server to the JIO network and assign the IP address from the subnet provided by jio either to a second ethernet or existing ethernet port(based on your network design).
The network configuration is not covered here , it might defer between OS type you are using, but concept is same.
Step 2: JIO SIP Proxy Static route
SIP proxy ip :100.64.216.4Media IP :100.64.216.4
ServerIP : 100.65.161.116
gateway ip :100.65.161.113
interface : Eth1
ip route add 100.64.216.4 via 100.65.161.113 dev eth1ip route add 100.64.216.5 via 100.65.161.113 dev eth1
ip route showorroute -n
Step 3: JIO SIP peer configuration sip.conf
Step 4: asterisk dialplan for jio sip
Note: The key factor to dial out via JIO , you need to set proper caller-id while dialing out.Use the below dialplan , alter according to your requirement.
For vanilla Asterisk :
exten => _X.,1,Set(CALLERID(num)=+914412345678);+914412345678 is my pilot no, you have to enter your pilot no or any number from the DID range
exten => _X.,2,Dial(SIP/jiosiptrunk/${EXTEN})
exten => _X.,3,Hangup
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)exten => _X.,2,Set(CALLERID(num)=+914412345678)exten => _X.,3,Dial(SIP/jiosiptrunk/${EXTEN},,tTo)exten => _X.,4,Hangup
If you want individual CID for each user make sure each user extension is configured with outbound callerid with there DID number.
For Freepbx, you need create outbound route , with necessary Dial pattern and make sure to set the ROUTE CID (either pilot or DID numbers)
Conclusion:
Hope the above tutorial is helpful to configure the JIO siptrunk in your sip server.
For professional support reach me via skype: striker24x7
Youtube Channel : https://www.youtube.com/c/striker24x7
Hi i have configured the JIO sip line as per the above but the auto calls are not landing on agent page. What shall i need to enable on routing extensions 8368.
have you set the callerid in dialplan or campaing?
can you post your cli log and dialplan used
Hi i have tried both.
what dialplan you are using, can you post the cli log
sir please help me i have problem in goautodial
let me know your problem in detail
Dear Striker
I just configured jio sip trunk but no voice is coming calls are maturing fine.
1st NIC we have jio sip trunk and on 2nd NIC we have internet with static ip
can you suggest me .....thanks in advance
add a static route for the jip media ip
Hi,
With Jio SIP trunk, both incoming/outgoing call are working.
Only problem is incoming call disconnects are 5-6 seconds. Even both parties can hear each other. What could be the reason. Please suggest
Hi,
Sorry i could not reply you back. Anyway i have sorted out the voice issue by disabling NAT under Asterisk SIP Settings.
But is there any other work around for this. Actually i want to keep NAT enable for calling from outside as my PBX is running behind firewall.
disable nat to specific trunk
Hi,
Incoming call on my JIO SIP trunk is disconnecting after 6 second. Please help.
Please find below cli logs:
== Spawn extension (from-internal, 901, 1) exited non-zero on 'SIP/901-00000005'
-- SIP/901-00000005 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called SIP/901
-- Connected line update to SIP/JIO-00000004 prevented.
-- SIP/901-00000005 is ringing
-- Connected line update to SIP/JIO-00000004 prevented.
-- SIP/901-00000005 answered SIP/JIO-00000004
-- Channel SIP/901-00000005 joined 'simple_bridge' basic-bridge <17d1cde9-8d8a-40b3-88a6-04cda4571390>
-- Channel SIP/JIO-00000004 joined 'simple_bridge' basic-bridge <17d1cde9-8d8a-40b3-88a6-04cda4571390>
[2021-07-17 11:40:39] NOTICE[2581]: chan_sip.c:28825 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 901
[2021-07-17 11:40:41] WARNING[2581]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 14175160…
How to write registry for jio sip trunk in sip.conf please tell
How to write registry for jio trunk in sip.conf
jio sip is on ip based..
if they have provided username secret then add them in sip.conf
as register => above your actual sip peer configuration
Really Helpful,
Thanks
Really useful post. It's working for me
hi
i am also facing the same issue, here outbound calls are working fine and when we come for inbound facing the issue
can i have cli log ,
Hello sir
Incoming calls are disconnected automatically after 6 second
im still trying to figure out on how to get the jip sip trunk from jio as they didnt provided any information or contact number to reach them :(