hello, I am having issue with incoming calls not working on a chan_sip trunk. However, I can make outgoing calls.
My inbound route destination is to an extension I created.
I also changed the port setting of sip to 5060 and I am still unable to receive calls
What can be the issue?
Does the trunk require registration? If so, does it show as registered? What, if anything, appears in the Asterisk log on an attempted incoming call?
If not using registration, is the provider’s portal set to send calls to the correct IP address and port? Is your hardware router/firewall set to pass the incoming INVITE to the PBX? Is FreePBX firewall set to trust the address(es) from which the provider can send calls?
Why are you using chan_sip at all?
The sip provider only support chan_sip. I disabled both firewall
What does this mean? In my experience, providers don’t support any SIP channel technology driver in Asterisk as the sample configurations they give out are generally badly designed. Typically they copy from others, and somewhere back in history, lots of unnecessary options get added, in some cases compromising security. Options included are often ones that were deprecated about a decade ago (although usually because they were renamed, rather than removed).
There is only one somewhat valid reason for not supporting chan_pjsip, and that is because they insist on TEL: URI support, but that means they will not work with a minimal compliant SIP implementation.
I took your advice to register the trunk using PJSIP. The trunk is registered and available. However, when I try to call, I see this error and I also can not receive calls
Did you create a incoming call route?
You need to setup an incoming call route from FreePBX.
In it you need to give it a destination (for you it will very probably be an extension you created)
I have an inbound route with destination to an extension, I also have an outbound route too.
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