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[asterisk-users] Ringback issue


 
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darcy at Vex.Net
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PostPosted: Mon Aug 24, 2015 10:49 pm    Post subject: [asterisk-users] Ringback issue Reply with 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:

[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
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PostPosted: Tue Aug 25, 2015 10:46 am    Post subject: [asterisk-users] Ringback issue Reply with quote

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)

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

--
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jcolp at digium.com
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PostPosted: Tue Aug 25, 2015 12:52 pm    Post subject: [asterisk-users] Ringback issue Reply with quote

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:

<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.

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

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darcy at Vex.Net
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PostPosted: Tue Aug 25, 2015 1:07 pm    Post subject: [asterisk-users] Ringback issue Reply with quote

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

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.

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.

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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