FreePBX will not forward calls externally

[2020-09-21 18:00:41] NOTICE[7577][C-0000000a] chan_sip.c: Failed to authenticate on INVITE to '<sip:[email protected]>;tag=as4907944d'

This may be two problems:

First, Telnyx requires destination numbers and caller IDs to start with the country code (with or without a leading +). So, you should be calling 17865780969. If you want users to be able to dial both with and without the initial 1, your Outbound Route should prepend a 1 when a ten digit domestic number is called.

If that doesn’t fix your problem, the authentication failure may be separate. Why did you set up a chan_sip trunk for Telnyx (pjsip is much simpler)? I suspect that without constraining fromuser, chan_sip is sending out the wrong user name on outgoing. Try adding
authuser=xxxxxx
to your PEER Details.
(Replace xxxxxx with the Telnyx username that you have in your Register String.)

Ty. I have the 1 prepend. I set up as Chan_sip because that’s what they sent me as guide. They also sent me the pjsip, which I also did as they suggested but, I got the same result.

In Settings -> Asterisk Logfile Settings -> Log Files, confirm that for ‘full’, Debug, Error, Notice, Verbose and Warning are turned on.

At the Asterisk command prompt, type
sip set debug on
or
pjsip set logger on
according to trunk type then make a test call and paste the resulting Asterisk log.

Here you go: <script src=“https://pastebin.com/embed_js/QfwzeVvt”></script>

The most recent log has nothing useful.

Please confirm that the logfile settings include Debug, Error, Notice, Verbose and Warning.
Also confirm that when you issued
sip set debug on
you got confirmation
SIP Debugging enabled
and you then made your test call.

The expected log would be several hundred lines.

BTW, please paste it at https://pastebin.freepbx.org and post the link here. Whatever you are doing at pastebin.com is making it hard to read.

https://pastebin.freepbx.org/view/6f8c34da

Here’s the latest:

<script src=“https://pastebin.com/embed_js/99JVDv9i”></script>

Your latest log starts after the PBX decided that the call failed. Please note the time that you make the test call and paste everything from that time onward.

Here you go. I set it to send to Misc Destinations:

<script src=“https://pastebin.com/embed_js/6URWjHzn”></script>

[2020-09-21 20:48:12] WARNING[722] res_pjsip_outbound_registration.c: No response received from ‘sip:sip.telnyx.com:5160’ on registration attempt to ‘sip:[email protected]:5160’, retrying in ‘60’

Server Port should be 5060, not 5160.

O.K. I’ll try that, but I also have to restart Asterisk…

Dude, you’re a genius… Can I leave it as 5060, or is it unsecured?.. Thank you very much…

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