[asterisk-users] Ringback issue

[D'Arcy J.M. Cain]

On Tue, 25 Aug 2015 14:51:56 -0300
Joshua Colp <jcolp at digium.com> wrote:
> > When I dial 4165556666 I get no ringback which I can sort of live
> > with since it gets answered pretty quickly but then when I dial

In fact I added a "Ringing" and a "Wait(3)" so there is at least one
ringback now.

> > "200" it transfers me correctly to the 4165555555 extension but
> > there is no ringback there either and that is a problem because
> > caller think that the phone has gone dead.

This still fails though.

> What is the complete console output and do you have an
> indications.conf configuration file?
> 
> I ask because in this scenario Asterisk would be generating the
> ringback itself as audio.

Here is the sanitized output.  4165551111 is the external caller,
4165552222 is the internal extension attached to PBX ext. 212 and
4165553333 is the cell that also gets called.

    -- Executing [6473512047 at LocalSets:1] Goto("SIP/4165551111-0000001b", "pbx-17842,s,1") in new stack
    -- Goto (pbx-17842,s,1)
    -- Executing [s at pbx-17842:1] Verbose("SIP/4165551111-0000001b", "0,"Caller" <4165551111> Calling PBX 17842") in new stack 
"Caller" <4165551111> Calling PBX 17842 
    -- Executing [s at pbx-17842:2] Ringing("SIP/4165551111-0000001b", "") in new stack
[Aug 25 14:01:31] WARNING[-1][C-0000000e]: channel.c:4674 ast_indicate_data: Unable to handle indication 3 for 'SIP/4165551111-0000001b'
    -- Executing [s at pbx-17842:3] Wait("SIP/4165551111-0000001b", "3") in new stack
    -- Executing [s at pbx-17842:4] BackGround("SIP/4165551111-0000001b", "/usr/local/var/sounds/pbx-17842/announce") in new stack
       > 0x7f7fef34f000 -- Probation passed - setting RTP source address to 207.35.13.14:16432
    -- <SIP/4165551111-0000001b> Playing '/usr/local/var/sounds/pbx-17842/announce.gsm' (language 'en')
       > 0x7f7fef34f000 -- Probation passed - setting RTP source address to 207.35.13.14:16432
[Aug 25 14:01:36] DTMF[-1][C-0000000e]: channel.c:4214 __ast_read: DTMF begin '2' received on SIP/4165551111-0000001b
[Aug 25 14:01:36] DTMF[-1][C-0000000e]: channel.c:4218 __ast_read: DTMF begin ignored '2' on SIP/4165551111-0000001b
[Aug 25 14:01:36] DTMF[-1][C-0000000e]: channel.c:4128 __ast_read: DTMF end '2' received on SIP/4165551111-0000001b, duration 180 ms
[Aug 25 14:01:36] DTMF[-1][C-0000000e]: channel.c:4198 __ast_read: DTMF end passthrough '2' on SIP/4165551111-0000001b
[Aug 25 14:01:36] DTMF[-1][C-0000000e]: channel.c:4214 __ast_read: DTMF begin '1' received on SIP/4165551111-0000001b
[Aug 25 14:01:36] DTMF[-1][C-0000000e]: channel.c:4218 __ast_read: DTMF begin ignored '1' on SIP/4165551111-0000001b
[Aug 25 14:01:37] DTMF[-1][C-0000000e]: channel.c:4128 __ast_read: DTMF end '1' received on SIP/4165551111-0000001b, duration 180 ms
[Aug 25 14:01:37] DTMF[-1][C-0000000e]: channel.c:4198 __ast_read: DTMF end passthrough '1' on SIP/4165551111-0000001b
[Aug 25 14:01:37] DTMF[-1][C-0000000e]: channel.c:4214 __ast_read: DTMF begin '2' received on SIP/4165551111-0000001b
[Aug 25 14:01:37] DTMF[-1][C-0000000e]: channel.c:4218 __ast_read: DTMF begin ignored '2' on SIP/4165551111-0000001b
[Aug 25 14:01:37] DTMF[-1][C-0000000e]: channel.c:4128 __ast_read: DTMF end '2' received on SIP/4165551111-0000001b, duration 160 ms
[Aug 25 14:01:37] DTMF[-1][C-0000000e]: channel.c:4198 __ast_read: DTMF end passthrough '2' on SIP/4165551111-0000001b
  == CDR updated on SIP/4165551111-0000001b
    -- Executing [212 at pbx-17842:1] Verbose("SIP/4165551111-0000001b", "0,"Caller" <4165551111> Calling PBX extension 4165552222") in new stack 
"Caller" <4165551111> Calling PBX extension 4165552222
    -- Executing [212 at pbx-17842:2] Ringing("SIP/4165551111-0000001b", "") in new stack
    -- Executing [212 at pbx-17842:3] Goto("SIP/4165551111-0000001b", "LocalSets,4165552222,1") in new stack
    -- Goto (LocalSets,4165552222,1)
    -- Executing [4165552222 at LocalSets:1] Verbose("SIP/4165551111-0000001b", "0,Entering extension 4165552222") in new stack
Entering extension 4165552222
    -- Executing [4165552222 at LocalSets:2] Ringing("SIP/4165551111-0000001b", "") in new stack
    -- Executing [4165552222 at LocalSets:3] GotoIf("SIP/4165551111-0000001b", "0?DialDesk") in new stack
    -- Executing [4165552222 at LocalSets:4] GotoIf("SIP/4165551111-0000001b", "0?DialDesk") in new stack
    -- Executing [4165552222 at LocalSets:5] Verbose("SIP/4165551111-0000001b", "0,"Caller" <4165551111> Calling "4165552222" and cell "4165553333"") in new stack 
"Caller" <4165551111> Calling "4165552222" and cell "4165553333"
    -- Executing [4165552222 at LocalSets:6] Dial("SIP/4165551111-0000001b", "SIP/4165552222&SIP/thinktel/4165553333,30,r") in new stack
    -- Called SIP/4165552222
    -- Called SIP/thinktel/4165553333
    -- SIP/4165552222-0000001c connected line has changed. Saving it until answer for SIP/4165551111-0000001b
    -- SIP/4165552222-0000001c is ringing
    -- SIP/thinktel-0000001d is making progress passing it to SIP/4165551111-0000001b
       > 0x7f7ff077d000 -- Probation passed - setting RTP source address to 206.80.250.102:26014
  == Spawn extension (LocalSets, 4165552222, 6) exited non-zero on
'SIP/4165551111-0000001b'


-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net