My last problem was nicely solved through this mailing list so
hopefully this new problem will have the same happy outcome.
My situation is that I have many extensions. Here is a sample:
[client-phone](!)
type=friend
host=dynamic
secret=XXXXXXXXXX
dtmfmode=auto
disallow=all
allow=ulaw
allow=gsm
allow=g723
allow=ilbc
subscribemwi=no
[4165555555](client-phone)
secret=xxxxxxxxxxxxxxxxxxxxxxxxxx
callerid=D'Arcy <4165555555>
mailbox=4165555555 at VoiceMail
context=LocalSets
I can send calls to this extension with this:
exten => 1,Verbose(0,${CALLERID(all)} Calling ${EXTEN})
same => n,Dial(SIP/4165555555,30)
same => n,VoiceMail(4165555555 at VoiceMail,u)
same => n,Hangup()
Up to this point everything works as expected. I call in and I hear a
ringback until the extension is picked up.
Now I add a virtual PBX to the mix.
[pbx-17842]
exten => s,1,Verbose(0,${CALLERID(all)} Calling PBX 17842)
same => n,Answer
same => n,Wait(2)
same => n(announce),Background($SOUNDS/pbx-17842/announce)
same => n,WaitExten()
same => n,DigitTimeout,5
same => n,ResponseTimeout,10
same => n,Goto(s,announce)
exten => i,1,Verbose(0,${CALLERID(all)} dialed invalid extension
${EXTEN}) same => n,Playback(invalid)
same => n,Goto(s,announce)
exten => 200,1,Verbose(0,${CALLERID(all)} Calling PBX darcy)
same => n,GoTo(LocalSets,4165555555,1)
Finally I add an extension to go to that context:
exten => 4165556666,1,GoTo(pbx-17842,s,1)
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 "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.
So, when I call 4165555555 it works fine but if I call it through the
virtual PBX it fails. I tried various combinations of "Ringing" and
'r' options and "prematuremedia=no" and
"progressinband=yes" but
nothing seems to help. Can someone suggest a line of enquiry?
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net
On Mon, 24 Aug 2015 23:48:50 -0400 "D'Arcy J.M. Cain" <darcy at Vex.Net> wrote:> exten => 200,1,Verbose(0,${CALLERID(all)} Calling PBX darcy) > same => n,GoTo(LocalSets,4165555555,1)I tried changing the above to; same => n,Dial(SIP/4165555555) and same => n,Dial(SIP/4165555555,,r) Same problem. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net VoIP: sip:darcy at Vex.Net
D'Arcy J.M. Cain wrote:> My last problem was nicely solved through this mailing list so > hopefully this new problem will have the same happy outcome. > > My situation is that I have many extensions. Here is a sample:<snip>> > [pbx-17842] > exten => s,1,Verbose(0,${CALLERID(all)} Calling PBX 17842) > same => n,Answer > same => n,Wait(2) > same => n(announce),Background($SOUNDS/pbx-17842/announce) > same => n,WaitExten() > same => n,DigitTimeout,5 > same => n,ResponseTimeout,10 > same => n,Goto(s,announce) > > exten => i,1,Verbose(0,${CALLERID(all)} dialed invalid extension > ${EXTEN}) same => n,Playback(invalid) > same => n,Goto(s,announce) > > exten => 200,1,Verbose(0,${CALLERID(all)} Calling PBX darcy) > same => n,GoTo(LocalSets,4165555555,1) > > Finally I add an extension to go to that context: > > exten => 4165556666,1,GoTo(pbx-17842,s,1) > > 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 "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.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. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
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 dialIn 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
On Mon, 24 Aug 2015 23:48:50 -0400 "D'Arcy J.M. Cain" <darcy at Vex.Net> 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 "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.Found it. Somehow I wound up with an empty indications.conf file so Asterisk did not know how to play the ringing sound. Found the default file and everything is good again. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net VoIP: sip:darcy at Vex.Net