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darcy at Vex.Net Guest
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Posted: Mon Aug 24, 2015 10:49 pm Post subject: [asterisk-users] Ringback issue |
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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@VoiceMail
context=LocalSets
I can send calls to this extension with this:
exten => 1,Verbose(0,${CALLERID(all)} Calling ${EXTEN})
same => n,Dial(SIP/4165555555,30)
same => n,VoiceMail(4165555555@VoiceMail,u)
same => n,Hangup()
Up to this point everything works as expected. I call in and I hear a
ringback until the extension is picked up.
Now I add a virtual PBX to the mix.
[pbx-17842]
exten => s,1,Verbose(0,${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@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
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darcy at Vex.Net Guest
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Posted: Tue Aug 25, 2015 10:46 am Post subject: [asterisk-users] Ringback issue |
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On Mon, 24 Aug 2015 23:48:50 -0400
"D'Arcy J.M. Cain" <darcy@Vex.Net> wrote:
Quote: | exten => 200,1,Verbose(0,${CALLERID(all)} Calling PBX darcy)
same => n,GoTo(LocalSets,4165555555,1)
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I tried changing the above to;
same => n,Dial(SIP/4165555555)
and
same => n,Dial(SIP/4165555555,,r)
Same problem.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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jcolp at digium.com Guest
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Posted: Tue Aug 25, 2015 12:52 pm Post subject: [asterisk-users] Ringback issue |
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D'Arcy J.M. Cain wrote:
Quote: | My last problem was nicely solved through this mailing list so
hopefully this new problem will have the same happy outcome.
My situation is that I have many extensions. Here is a sample:
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<snip>
Quote: |
[pbx-17842]
exten => s,1,Verbose(0,${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.
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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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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darcy at Vex.Net Guest
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Posted: Tue Aug 25, 2015 1:07 pm Post subject: [asterisk-users] Ringback issue |
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On Tue, 25 Aug 2015 14:51:56 -0300
Joshua Colp <jcolp@digium.com> wrote:
Quote: | Quote: | 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
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In fact I added a "Ringing" and a "Wait(3)" so there is at least one
ringback now.
Quote: | Quote: | "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.
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This still fails though.
Quote: | 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.
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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
Quote: | 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')
Quote: | 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
Quote: | 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@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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