PBX to Teams

PBX ext. calls Teams user…
Teams user tries to transfer to fellow Teams user…
Call fails and falls back to Teams user that tried to transfer…

Is this Teams or my PBX?

How are you interfaced with Microsoft Teams?

AA Call trace would be useful
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs

None I figured the calls would route out of PBX to Teams without issues…

So the Teams user is just connected to the PSTN (whether through Microsoft directly or some other means) and you are calling a PSTN number from your PBX?

/var/log/asterisk/full:[2022-06-06 17:45:55] VERBOSE[6878][C-0000293e] pbx.c: Executing [[email protected]:11] ExecIf("PJSIP/Mediant1000-00008d5c", "0?Set(sipheader=<http://127.0.0.1>unset)") in new stack
/var/log/asterisk/full:[2022-06-06 17:45:55] VERBOSE[6878][C-0000293e] pbx.c: Executing [[email protected]:12] ExecIf("PJSIP/Mediant1000-00008d5c", "0?SIPAddHeader(Alert-Info:unset)") in new stack
/var/log/asterisk/full:[2022-06-06 17:45:55] VERBOSE[6878][C-0000293e] pbx.c: Executing [[email protected]:13] ExecIf("PJSIP/Mediant1000-00008d5c", "0?Set(PJSIP_HEADER(add,Alert-Info)=unset)") in new stack
/var/log/asterisk/full:[2022-06-06 17:45:55] VERBOSE[6878][C-0000293e] pbx.c: Executing [[email protected]:14] EndWhile("PJSIP/Mediant1000-00008d5c", "") in new stack
/var/log/asterisk/full:[2022-06-06 17:45:55] VERBOSE[6878][C-0000293e] pbx.c: Executing [[email protected]:6] While("PJSIP/Mediant1000-00008d5c", "0") in new stack
/var/log/asterisk/full:[2022-06-06 17:45:55] VERBOSE[6878][C-0000293e] pbx.c: Executing [[email protected]:15] Return("PJSIP/Mediant1000-00008d5c", "") in new stack
/var/log/asterisk/full:[2022-06-06 17:45:55] VERBOSE[6878][C-0000293e] app_stack.c: Spawn extension (from-pstn, 2768, 1) exited non-zero on 'PJSIP/Mediant1000-00008d5c'
/var/log/asterisk/full:[2022-06-06 17:45:55] VERBOSE[6878][C-0000293e] app_stack.c: PJSIP/Mediant1000-00008d5c Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
/var/log/asterisk/full:[2022-06-06 17:45:55] VERBOSE[6878][C-0000293e] app_dial.c: Called PJSIP/[email protected]
/var/log/asterisk/full:[2022-06-06 17:45:56] VERBOSE[6878][C-0000293e] app_dial.c: PJSIP/Mediant1000-00008d5c is ringing
/var/log/asterisk/full:[2022-06-06 17:45:57] VERBOSE[6878][C-0000293e] app_dial.c: PJSIP/Mediant1000-00008d5c is ringing
/var/log/asterisk/full:[2022-06-06 17:45:58] VERBOSE[6878][C-0000293e] app_dial.c: PJSIP/Mediant1000-00008d5c is ringing
/var/log/asterisk/full:[2022-06-06 17:45:59] VERBOSE[6878][C-0000293e] app_dial.c: PJSIP/Mediant1000-00008d5c is ringing
/var/log/asterisk/full:[2022-06-06 17:45:59] VERBOSE[6878][C-0000293e] app_dial.c: PJSIP/Mediant1000-00008d5c is ringing
/var/log/asterisk/full:[2022-06-06 17:46:02] VERBOSE[6878][C-0000293e] app_dial.c: PJSIP/Mediant1000-00008d5c is making progress passing it to SIP/2770-0000001a
/var/log/asterisk/full:[2022-06-06 17:46:02] VERBOSE[6878][C-0000293e] app_dial.c: PJSIP/Mediant1000-00008d5c answered SIP/2770-0000001a
/var/log/asterisk/full:[2022-06-06 17:46:02] VERBOSE[6878][C-0000293e] app_stack.c: PJSIP/Mediant1000-00008d5c Internal Gosub(sub-send-obroute-email,s,1(12189362768,2768,1,1654537554,BHTEST,2770)) start
/var/log/asterisk/full:[2022-06-06 17:46:02] VERBOSE[6878][C-0000293e] pbx.c: Executing [[email protected]:1] GotoIf("PJSIP/Mediant1000-00008d5c", "0?sendEmail") in new stack
/var/log/asterisk/full:[2022-06-06 17:46:02] VERBOSE[6878][C-0000293e] pbx.c: Executing [[email protected]:2] NoOp("PJSIP/Mediant1000-00008d5c", "email notifications disabled..exiting.") in new stack
/var/log/asterisk/full:[2022-06-06 17:46:02] VERBOSE[6878][C-0000293e] pbx.c: Executing [[email protected]:3] Return("PJSIP/Mediant1000-00008d5c", "") in new stack
/var/log/asterisk/full:[2022-06-06 17:46:02] VERBOSE[6878][C-0000293e] app_stack.c: Spawn extension (from-pstn, , 1) exited non-zero on 'PJSIP/Mediant1000-00008d5c'
/var/log/asterisk/full:[2022-06-06 17:46:02] VERBOSE[6878][C-0000293e] app_stack.c: PJSIP/Mediant1000-00008d5c Internal Gosub(sub-send-obroute-email,s,1(12189362768,2768,1,1654537554,BHTEST,2770)) complete GOSUB_

The transfer rings about 5 times then fails

The log you shared, shows, that the Mediant1000 rang for 7 seconds and was answered, but was immediately hung up. I assume that this is because of a codec mismatch.

In the Asterisk console, please enable pjsip debugging, reproduce the issue, upload the output to pastebin.freepbx.org and share the link.

asterisk -rvvvvvv
pjsip set logger on

That didn’t answer Bill’s question! Assuming he is right, Asterisk has no knowledge that it is dealing with Teams, and this is likely to be a Teams/PSTN issue.

Sorry guys level one tech here… but from what I can understand of FreePBX I don’t see any teams integrating here

Hello @boudreau1996,

I think that your problem is in the Audiocodes device (Mediant 1000). This is the gateway that connects you to the Teams cloud. You will need to check the Audiocodes log.

Do you have a problem with regular calls (without a transfer from the Teams user)?

Thank you,

Daniel Friedman
Trixton LTD.

No problem on the transfer through the pbx just on teams side trying to transfer pbx to another teams user

Hello @boudreau1996,

I asked if you have problems with regular calls between the PBX extensions and the Teams users.

Can you make calls from the Teams cloud and the PBX extensions with no problems? If yes, you have a problem on the Audiocodes side.

Thank you,

Daniel Friedman
Trixton LTD.

https://pastebin.freepbx.org/view/93da88e1

Here you go. Help me please lol

All you can see from the log is that ext 2770 called the number on the Mediant gateway and then the Mediant hung up. You need to examine this on the SBC.

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