Outbound calls mysteriously going to dialtone?

FreePBX Providers
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I’ve changed nothing - some calls work - some don’t

with this one I’ve specifically used a prefix (322) which sends calls to the PSTN instead of using Sipgate but the problem occurs with and without the prefix

>----Trace----<

Any ideas??

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35294	[2022-10-04 17:34:26] VERBOSE[11606][C-0000d4c8] app_dial.c: Called SIP/1-pstn/07621280034	
35295	[2022-10-04 17:34:26] VERBOSE[11606][C-0000d4c8] app_dial.c: SIP/1-pstn-00000fe3 is ringing	
35296	[2022-10-04 17:34:26] VERBOSE[11606][C-0000d4c8] app_dial.c: SIP/1-pstn-00000fe3 answered PJSIP/22-0000daae

This shows proper behavior by Asterisk. I suspect that the PSTN gateway somehow did not relay the call properly. Make/model? If it has a syslog feature, that may show what went wrong. It also may be useful to listen to the analog line to hear whether dialtone was present, DTMF was properly sent, etc.

BTW, when you paste a trace, we prefer you to use pastebin.freepbx.org . But if you must use pastebin or some other service, please ensure that expiration is set to Never, so future readers of the thread can follow along.

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