"Number not answering" but only a specific number?

oxyde_v2 · August 16, 2022, 7:33am

Hello everyone ! My probleme is, when i try outgoing call, it work fine.
But, one specific number don’t work, and FreePBX tell me the number is not answering.

FreePBX 15.0.17.24

-- Executing [s@func-apply-sipheaders:9] ExecIf("PJSIP/ORANGE-OBS1-00005382", "0?Set(sipheader=<http://127.0.0.1>;info=unset)") in new stack
-- Executing [s@func-apply-sipheaders:10] ExecIf("PJSIP/ORANGE-OBS1-00005382", "0?Set(sipheader=<http://127.0.0.1>unset)") in new stack
-- Executing [s@func-apply-sipheaders:11] ExecIf("PJSIP/ORANGE-OBS1-00005382", "0?SIPAddHeader(Alert-Info:unset)") in new stack
-- Executing [s@func-apply-sipheaders:12] ExecIf("PJSIP/ORANGE-OBS1-00005382", "0?Set(PJSIP_HEADER(add,Alert-Info)=unset)") in new stack
-- Executing [s@func-apply-sipheaders:13] EndWhile("PJSIP/ORANGE-OBS1-00005382", "") in new stack
-- Executing [s@func-apply-sipheaders:5] While("PJSIP/ORANGE-OBS1-00005382", "0") in new stack
-- Executing [s@func-apply-sipheaders:14] Return("PJSIP/ORANGE-OBS1-00005382", "") in new stack

== Spawn extension (from-pstn, 0438022400, 1) exited non-zero on ‘PJSIP/ORANGE-OBS1-00005382’
– PJSIP/ORANGE-OBS1-00005382 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
– Called PJSIP/0438022400@ORANGE-OBS1
– PJSIP/ORANGE-OBS1-00005382 is ringing
– PJSIP/ORANGE-OBS1-00005382 is ringing
– No one is available to answer at this time (1:0/0/0)
– Executing [s@macro-dialout-trunk:35] NoOp(“PJSIP/160-00005381”, “Dial failed for some reason with DIALSTATUS = NOANSWER and HANGUPCAUSE = 19”) in new stack
– Executing [s@macro-dialout-trunk:36] GotoIf(“PJSIP/160-00005381”, “0?continue,1:s-NOANSWER,1”) in new stack
– Goto (macro-dialout-trunk,s-NOANSWER,1)
– Executing [s-NOANSWER@macro-dialout-trunk:1] NoOp(“PJSIP/160-00005381”, “Dial failed due to trunk reporting NOANSWER - giving up”) in new stack
– Executing [s-NOANSWER@macro-dialout-trunk:2] Progress(“PJSIP/160-00005381”, “”) in new stack
– Executing [s-NOANSWER@macro-dialout-trunk:3] Playback(“PJSIP/160-00005381”, “number-not-answering,noanswer”) in new stack
> 0x3433d40 – Strict RTP learning after remote address set to: 192.168.1.120:5016
– <PJSIP/160-00005381> Playing ‘number-not-answering.ulaw’ (language ‘fr’)
> 0x3433d40 – Strict RTP switching to RTP target address 192.168.1.120:5016 as source
– Executing [s-NOANSWER@macro-dialout-trunk:4] Congestion(“PJSIP/160-00005381”, “20”) in new stack
== Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on ‘PJSIP/160-00005381’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 0438022400, 13) exited non-zero on ‘PJSIP/160-00005381’
– Executing [h@from-internal:1] Macro(“PJSIP/160-00005381”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/160-00005381”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/160-00005381”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“PJSIP/160-00005381”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘PJSIP/160-00005381’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/160-00005381’
– PJSIP/160-00005381 Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“PJSIP/160-00005381”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/160-00005381”, “HANGUP CAUSE: 34”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/160-00005381”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“PJSIP/160-00005381”, “MASTER CHANNEL: 1660633773.64353 = 1660633773.64353”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/160-00005381”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“PJSIP/160-00005381”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“PJSIP/160-00005381”, “agi://127.0.0.1/sangomacrm.agi”) in new stack
– <PJSIP/160-00005381>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“PJSIP/160-00005381”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/160-00005381’
– PJSIP/160-00005381 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
3ci*CLI>

david55 · August 16, 2022, 10:32am

Please use the full log, not a screen scrape, and please post logs to pastebin.freepbx.org, or failing that, mark them up as pre-formatted text.

The full log contains timestamps which would confirm the length of time between Dial and failure, however there is little doubt in my mind that the number was successfully called but did not answer, and this is a straightforward no answer case,.

oxyde_v2 · August 16, 2022, 10:53am

I can join the number with my personnal phone

david55 · August 16, 2022, 11:44am

The call is shown as ringing, indicating that the provider is claiming the call has reached the destination. Having the timing would tell whether Asterisk was timing it out, and having the “pjsip set logger on” output would confirm that, and if the provider was reporting no answer, might given more of their reasons.

oxyde_v2 · August 16, 2022, 11:58am

Ok thanks, the pastebin for the pjsip logger
https://pastebin.freepbx.org/view/b5e217a7

david55 · August 16, 2022, 12:28pm

It’s rather a small extract, and you still seem to be screen scraping, but the provider has said that the callee is temporarily unavailable:

https://pastebin.freepbx.org/view/b5e217a7#L25

Which is defined here:

https://www.rfc-editor.org/rfc/rfc3261.html#section-21.4.18

They have provided no reason header to make things clearer.

Note that, as there seems to be more than one transaction captured here, I’m assuming the 480 response is what triggered your no answer report.

Cause 19 is No answer from user (user alerted)

oxyde_v2 · August 16, 2022, 12:46pm

Thanks, so if i understand, this is my provider and not my FreePBX ?

david55 · August 16, 2022, 12:47pm

Assuming the right number is in the INVITE, which you didn’t include, it is your provider that is claiming they are not answering.

oxyde_v2 · August 16, 2022, 1:01pm

NoAnswering - FreePBX Pastebin maybe this pastbin is better ?

mahbell · August 21, 2022, 2:19am

Line 212. SIP/2.0 407 Proxy Authentication Required - 4009

Probably need to talk to your provider to get a good understanding of what’s missing/how to configure.

Stewart1 · August 21, 2022, 2:52am

Line 203 shows received=10.201.111.210, presumably the LAN address of the PBX. That’s not possible so it appears that your hardware router/firewall has a SIP ALG enabled. Turn that off and retest. If you still have trouble, paste a new log from /var/log/asterisk/full (the console log lacks timestamps), covering the entire call, as much as 1000 lines.

system · September 21, 2022, 2:52am

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