Intercom no longer working on 12.x beta

Howdy all,

Having an intercom issue. Dialing *80(ext) not working any longer, the call simply disconnects to fast busy. Testing on 2 different brands of phones produces the same results from both ends.

Can you post some logs of the failed call?

I cut out the top part of this, seemed to be irrelevant.

== Using SIP RTP CoS mark 5
– Executing [*[email protected]:23] GotoIf(“SIP/113-0000002f”, “0?end”) in new stack
– Executing [*[email protected]:24] GotoIf(“SIP/113-0000002f”, “0?godial”) in new stack
– Executing [*[email protected]:25] Set(“SIP/113-0000002f”, “CONNECTEDLINE(name,i)=Francis”) in new stack
– Executing [*[email protected]:26] Set(“SIP/113-0000002f”, “CONNECTEDLINE(num)=117”) in new stack
– Executing [*[email protected]:27] Dial(“SIP/113-0000002f”, “SIP/117,5,IA(beep)b(autoanswer^s^1(Ring Answer,;answer-after=0))”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– SIP/117-00000031 Internal Gosub(autoanswer,s,1(Ring Answer,;answer-after=0)) start
– Executing [[email protected]:1] GosubIf(“SIP/117-00000031”, “1?addheader,1(Alert-Info,Ring Answer)”) in new stack
– Executing [[email protected]:1] SIPAddHeader(“SIP/117-00000031”, “Alert-Info: Ring Answer”) in new stack
– Executing [[email protected]:2] Set(“SIP/117-00000031”, “PJSIP_HEADER(add,Alert-Info)=Ring Answer”) in new stack
[2014-09-11 10:03:48] ERROR[23350][C-0000002c]: pbx.c:4390 ast_func_write: Function PJSIP_HEADER not registered
– Executing [[email protected]:3] Return(“SIP/117-00000031”, “”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/117-00000031”, “1?addheader,1(Call-Info,;answer-after=0)”) in new stack
– Executing [[email protected]:1] SIPAddHeader(“SIP/117-00000031”, “Call-Info: ;answer-after=0”) in new stack
– Executing [[email protected]:2] Set(“SIP/117-00000031”, “PJSIP_HEADER(add,Call-Info)=;answer-after=0”) in new stack
[2014-09-11 10:03:48] ERROR[23350][C-0000002c]: pbx.c:4390 ast_func_write: Function PJSIP_HEADER not registered
– Executing [[email protected]:3] Return(“SIP/117-00000031”, “”) in new stack
– Executing [[email protected]:3] Return(“SIP/117-00000031”, “”) in new stack
== Spawn extension (from-internal, *80117, 1) exited non-zero on ‘SIP/117-00000031’
– SIP/117-00000031 Internal Gosub(autoanswer,s,1(Ring Answer,;answer-after=0)) complete GOSUB_RETVAL=
– Called SIP/117
– Connected line update to SIP/113-0000002f prevented.
== Extension Changed 117[ext-local] new state Ringing for Notify User 114
== Extension Changed auto_hint_117[from-internal] new state Ringing for Notify User 117
== Extension Changed auto_hint_117[from-internal] new state Ringing for Notify User 129
== Extension Changed auto_hint_117[from-internal] new state Ringing for Notify User 113
== Extension Changed auto_hint_117[from-internal] new state Ringing for Notify User 106
– SIP/117-00000031 is ringing
== Extension Changed 117[ext-local] new state Ringing for Notify User 114
== Extension Changed auto_hint_117[from-internal] new state Ringing for Notify User 117
== Extension Changed auto_hint_117[from-internal] new state Ringing for Notify User 129
== Extension Changed auto_hint_117[from-internal] new state Ringing for Notify User 113
== Extension Changed auto_hint_117[from-internal] new state Ringing for Notify User 106
– Nobody picked up in 5000 ms
– Executing [*[email protected]:28] ExecIf(“SIP/113-0000002f”, “?Return()”) in new stack
– Executing [*[email protected]:29] Busy(“SIP/113-0000002f”, “20”) in new stack
[2014-09-11 10:03:53] WARNING[23350][C-0000002c]: channel.c:4816 ast_prod: Prodding channel ‘SIP/113-0000002f’ failed
== Spawn extension (ext-intercom, *80117, 29) exited non-zero on ‘SIP/113-0000002f’
== Extension Changed 117[ext-local] new state Idle for Notify User 114
== Extension Changed auto_hint_117[from-internal] new state Idle for Notify User 117
== Extension Changed auto_hint_117[from-internal] new state Idle for Notify User 129
== Extension Changed auto_hint_117[from-internal] new state Idle for Notify User 113
== Extension Changed auto_hint_117[from-internal] new state Idle for Notify User 106
== Extension Changed auto_hint_113[from-internal] new state Idle for Notify User 129
== Extension Changed auto_hint_113[from-internal] new state Idle for Notify User 117
== Extension Changed auto_hint_113[from-internal] new state Idle for Notify User 113
== Extension Changed auto_hint_113[from-internal] new state Idle for Notify User 106

[2014-09-11 10:03:53] WARNING[23350][C-0000002c]: channel.c:4816 ast_prod: Prodding channel ‘SIP/113-0000002f’ failed
== Spawn extension (ext-intercom, *80117, 29) exited non-zero on ‘SIP/113-0000002f’

What version of Asterisk are you using? If Asterisk 12 its broke right now as we have no way to get the user agent of the device in Asterisk 12

Never should’ve taken the upgrade challenge.

So downgrade to asterisk 11, you can still use freepbx 12. This is simple in the FreePBX distro.

For all of us noobs out there it is this easy:

http://wiki.freepbx.org/display/FD/Changing+Major+Asterisk+Versions+on+the+Fly

Downgrade complete, no change. Fast busy.

Give the output of the CLI or use FreePBX 2.11

Ugh… since downgrading, I no longer have DPMA and now I have to re-register my DPMA module, which is not cooperating. Never should’ve upgraded. Thanks for your time.