On Tue, 25 Aug 2015 14:51:56 -0300
Joshua Colp <[email protected]> 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@LocalSets:1] Goto("SIP/4165551111-0000001b",
"pbx-17842,s,1") in new stack
-- Goto (pbx-17842,s,1)
-- Executing [s@pbx-17842:1] Verbose("SIP/4165551111-0000001b", "0,"Caller"
<4165551111> Calling PBX 17842") in new stack
"Caller" <4165551111> Calling PBX 17842
-- Executing [s@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@pbx-17842:3] Wait("SIP/4165551111-0000001b", "3") in new
stack
-- Executing [s@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@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@pbx-17842:2] Ringing("SIP/4165551111-0000001b", "") in
new stack
-- Executing [212@pbx-17842:3] Goto("SIP/4165551111-0000001b",
"LocalSets,4165552222,1") in new stack
-- Goto (LocalSets,4165552222,1)
-- Executing [4165552222@LocalSets:1] Verbose("SIP/4165551111-0000001b",
"0,Entering extension 4165552222") in new stack
Entering extension 4165552222
-- Executing [4165552222@LocalSets:2] Ringing("SIP/4165551111-0000001b",
"") in new stack
-- Executing [4165552222@LocalSets:3] GotoIf("SIP/4165551111-0000001b",
"0?DialDesk") in new stack
-- Executing [4165552222@LocalSets:4] GotoIf("SIP/4165551111-0000001b",
"0?DialDesk") in new stack
-- Executing [4165552222@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@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:[email protected]
VoIP: sip:[email protected]
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