I’m at a loss… I’ve tried basically every thing I can find in this community and multiple Guides, gpt, but nothing I try is working. I’ve managed to get my DID to when called hit my FreePBX server, but nothing I’ve tried for the setup completes the call. All I need to happen is for it to answer and go straight to a voicemail box. I don’t need interaction. I just want a greeting to be played and then directed to an extensions vm.
The error I’m getting is [2025-05-07 17:38:14] NOTICE[2147301] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘sip:[email protected];isup-oli=62’ failed for ‘54.172.60.2:5060’ (callid: [email protected]) - No matching endpoint found
My SIP Server in PJSIP Trunk settings is the same as the inbound from.
Given the redaction, it is not possible to verify whether
is the same as ec2-54-172-60-2.compute-1.amazonaws.com, which is where the request is coming from. If it isn’t, you may need a match/permit setting for the actual valid source address ranges.
When a hostname is used, the behavior depends on whether srv_lookups is enabled and/or a source port is provided. If srv_lookups is enabled and a source port is not provided, Asterisk will perform an SRV lookup on the provided hostname, adding all of the A and AAAA records that are resolved.
If the SRV lookup fails, srv_lookups is disabled, or a source port is specified when the hostname is configured, Asterisk will resolve the hostname and add all A and AAAA records that are resolved.
I actively use FQDNs in the match= setting and it resolves the SRV records associated with it.
OK @aimsopp I’ll take one more stab at it since it’s not the hostname. Check Asterisk SIP Settings - chan_pjsip - and ensure the Endpoint Identifier Order starts with ip
Well it still might be the hostname, just for another reason. Effectively by declaring the hostname along with all the IPs, it is most likely creating duplicate records for the same IP. That could cause it to not play nice.
@aimsopp Go into the system, get into the Asterisk console and do pjsip show identify <name of trunk> and see if the IPs are being duplicated.
Under some circumstances, Asterisk uses the trunk name as a username for matching. While yours shouldn’t be one of them, don’t push your luck, choose a simple trunk name such as Twilio1.
So I did this and its saying “Unable to find object”.
Under Connectivity > Trunks I have a pjsip trunk setup and the Trunk name assigned in there is what I tried to perform the search to. I tried case sensitivity to see if that made a difference but did not… so I’m assuming the FreePBX isn’t creating the trunk in the database?
OK do, pjsip show endpoint <name of trunk> and see what that outputs. There should be an Identify section at the top showing what is in use. If that’s missing, then that’s the problem.
Not sure what changed other then the restart of Asterisk… its now technically answering the call and sending me to a vm box but its hanging up (i didn’t as the log says).. Either way I really appreciate all the help and guidance you guys have given me! I’m going to keep plugging along to try to get over the finish line on this.
