Need help to use Jio SIP trunk

Hi Members,
Need help to setup Freepbx for JIO sip trunk in India.
My setup is exactly same as this post here - how-to-setup-reliance-jio-trunk-sip-provider-in-india/60765

I did all the things mentioned on the post but not able to make call though my peer trunk (JIO)status show ‘ONLINE’ .

I think I doing something wrong Dial pattern
@and_keller @amjithlal can you please help or give me any example for Indian number dial pattern.

If the trunk status is showing online then you’ve got the trunk setup. can you provide what you have setup for the Outbound Route that’s supposed to use the trunk? What error messages are you getting in the asterisk CLI.

Do you have any inbound routes setup at all?

Thank you @dobrosavljevic for response,
Below is the my setup:

Outbound route setup

What does your Dial Patterns tab look like? What’s the error you get when trying to dial out?

error - ‘The number you are dialing is not in service’.
please find the logs during the call.
I am trying a 10 digit mobile number

I am not sure that I’ve ever seen asterisk throw the INVALIDNMBR error or how it gets there. Maybe another regular here has and could assist.

If the FreePBX dialplan logging mirrors that of the underlying SIP signaling and it states “Address Incomplete” that would mean the provider doesn’t like the dialed number and considers it incomplete. A SIP trace using “pjsip set logger on” would confirm that, and also what was actually dialed.

Hey Joshua, not sure if you noticed but asterisk is showing s-INVALIDNMBR before that address incomplete line. I think there is a problem with how it’s building the number that it’s dialing but I have no idea how you get there as I’ve never had to troubleshoot that particular problem.

Right, because the FreePBX logic jumped there based on its own conditions and logic in the dialplan. It’s all just guesses because the provided output is incomplete and doesn’t show the complete call from start to finish. Only after it has failed.

@ranjit can you try a call but provide the full output from start to finish? It should show the full number that you are trying to dial at the start.

Thank you @jcolp & @dobrosavljevic , you are right. I was able to fix the issue by correcting the dial pattern in OUTbound rule.I had add my required prefix.

Currently my outgoing and incoming call are working BUT voice in one side , from my softphone voice is not delivering to the actual phone number I am calling.
Any headsup ?

Make sure you have the correct external and internal addresses set in Settings → Asterisk SIP settings as well as the appropriate RTP ports forwarded to the PBX from the external firewall.

My question would be that since this is a SIP provider out of India and the PBX is in India, how would North American Number Plan patterns work? Calling the US or Canada from India would be an International call, not a local/in-country call.

Issue is resolved by proper Natting some additional IP which my freepbx was trying to reach and which was not provided by provider.
Found this looking at wan interface packet capture.
Thank you so much @dobrosavljevic and @jcolp for your help.

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