Hello FreePBX community,
I am new to this forum and need some help with setting up outbound dialing on my free pbx
I recently installed FreePBX (FreePBX 16.0.21 with Asterisk 18) and I am trying to connect it to a SIP peer via a SIP trunk, I have two extensions 7000 and 7001 ( softphones) which can register and talk to each other but when I try to dial out to PSTN from either of these extensions, I get a 503 Service Unavailable from the PBX.
Please point me to an example of how to set up the trunk and dial out to PSTN.
NOTE: I don’t have to register my trunk with the SIP peer at least from the SIP provider’s settings as this peer is a B2BUA and doesn’t require trunk registration.
Have you set up an Outbound Route (Connectivity → Outbound Route)?
That’s where the extension knows which Trunk to use.
Being a back to back user agent is not a reason that you wouldn’t require registration (similarly authentication) for a trunk. If that were really the case, given that FreePBX is most definitely a back to back user agent, there would be no point in having the Registration: Receive (Authentication: Inbound/Both) options in the trunk pjsip settings.
Conversely a SIP proxy doesn’t have to use them.
It may well be the case that you need neither, but it is not because the peer is a B2BUA.
503 is a catch all error, and it is likely that the speculation about the failure to link the trunk to a route is correct, and it is coming from FreePBX, but if it is coming from the provider, it could be many things, although malformed numbers, and unacceptable caller IDs seem common.
It is generally easiest to debug if you provide the full log for the failed call, and would also help to tell us what the peer is, and who is providing service through it.
(Technically you register with neither a user agent nor a proxy, but that is probably unnecessary detail here.)
If you go to Reports → Asterisk Info → Channels does it show that the trunk is online?
Thank! you, I had the Outbound route set to Intra Company and I guess that was the problem which led to call not going out of the PBX. It is fixed now and the PBX now invites the peers
For some reason this ari Rest interface isn’t working but when I scroll down, I do see the trunk endpoint as available.
The issue seems to be fixed for now, I see the INVITE going out of the PBX.
Do I need to set anything else via the FreePBX CLI, I mean for the extensions in the pjsip.conf files or extensions.conf files? Just curious if there are any other settings that I am missing apart from the Trunk, Outbound/Inbound Routes and the extension creation.
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