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darcy at Vex.Net
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 Posted: Thu Aug 07, 2014 9:13 am Post subject: [asterisk-users] Calls not hanging up |
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This just started after upgrading to 11.11.0. After a call is
completed (both ends hang up) the call still shows as active.
# asterisk -x "core show channels"
Channel Location State Application(Data)
SIP/thinktel-0000000 (None) Up AppDial((Outgoing
Line)) SIP/4164251212-00000 4165555555@LocalSets Up
Dial(SIP/thinktel/4165559999) 2 active channels
1 active call
1 call processed
The 1212 number is mine and is hung up. I even rebooted my ATA to make
sure that it wasn't holding the line. My dialplan is extremely
simple. In fact, I even simplified it from what it was for this
testing. Here it is.
exten => 4164251212,1,Verbose(0, ${CALLERID(all)} Calling ${EXTEN})
same => n,Dial(SIP/4164251212,30)
same => n,VoiceMail(4164251212@LocalSets,u)
same => n,Hangup()
I can post any other log or config excerpts if someone thinks that they
are relevant but all of this was working under 11.10.2.
Thanks.
--
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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asghar144 at gmail.com
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| Posted: Thu Aug 07, 2014 10:18 am Post subject: [asterisk-users] Calls not hanging up |
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Your call is up on VoiceMail you should check dialstatus before sending user to VoiceMail.
On Thu, Aug 7, 2014 at 4:12 PM, D'Arcy J.M. Cain <darcy@vex.net (darcy@vex.net)> wrote:
| Quote: | This just started after upgrading to 11.11.0. After a call is
completed (both ends hang up) the call still shows as active.
# asterisk -x "core show channels"
Channel Location State Application(Data)
SIP/thinktel-0000000 (None) Up AppDial((Outgoing
Line)) SIP/4164251212-00000 [url=tel:4165555555]4165555555[/url]@LocalSets Up
Dial(SIP/thinktel/[url=tel:4165559999]4165559999[/url]) 2 active channels
1 active call
1 call processed
The 1212 number is mine and is hung up. I even rebooted my ATA to make
sure that it wasn't holding the line. My dialplan is extremely
simple. In fact, I even simplified it from what it was for this
testing. Here it is.
exten => [url=tel:4164251212]4164251212[/url],1,Verbose(0, ${CALLERID(all)} Calling ${EXTEN})
same => n,Dial(SIP/[url=tel:4164251212]4164251212[/url],30)
same => n,VoiceMail[url=tel:%284164251212](4164251212[/url]@LocalSets,u)
same => n,Hangup()
I can post any other log or config excerpts if someone thinks that they
are relevant but all of this was working under 11.10.2.
Thanks.
--
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:
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darcy at Vex.Net
Guest
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| Posted: Thu Aug 07, 2014 10:55 am Post subject: [asterisk-users] Calls not hanging up |
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On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad <asghar144@gmail.com> wrote:
| Quote: | Your call is up on VoiceMail you should check dialstatus before
sending user to VoiceMail.
|
so
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
is incorrect now? That page says:
"Unless there is a timeout specified, the Dial application will wait
indefinitely until one of the called channels answers, the user hangs
up, or if all of the called channels are busy or unavailable. Dialplan
executing will continue if no requested channels can be called, or if
the timeout expires. This application will report normal termination if
the originating channel hangs up, or if the call is bridged and either
of the parties in the bridge ends the call."
The second sentence implies that the dialplan will not continue, i.e.
will not go to VM, if the call is answered. The third sentence
reinforces that interpretation. That's certainly what happened in 11.10.
I didn't see anything in the change logs that would suggest such a
drastic change in behaviour.
--
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 --
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darcy at Vex.Net
Guest
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| Posted: Thu Aug 07, 2014 12:15 pm Post subject: [asterisk-users] Calls not hanging up |
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On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad <asghar144@gmail.com> wrote:
| Quote: | Your call is up on VoiceMail you should check dialstatus before
sending user to VoiceMail.
|
I removed the voicemail command from the dialplan and it was exactly
the same behaviour.
--
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 --
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andres at telesip.net
Guest
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| Posted: Thu Aug 07, 2014 1:33 pm Post subject: [asterisk-users] Calls not hanging up |
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On 8/7/14, 12:14 PM, D'Arcy J.M. Cain wrote:
| Quote: | On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad <asghar144@gmail.com> wrote:
| Quote: | Your call is up on VoiceMail you should check dialstatus before
sending user to VoiceMail.
|
I removed the voicemail command from the dialplan and it was exactly
the same behaviour.
|
You have 3 ways to automatically hang up a call.
1) Caller hangs up
2) Callee hangs up
3) Timeout hangs up call
I suggest you capture the SIP messages to see if the hangup messages are
not reaching Asterisk (caller or callee). It is also a good idea to
place a hard limit on calls so they hangup by timeout and not stay there
forever. From the DIAL comand:
L(x[:y[:z]]):
x - Maximum call time, in milliseconds
y - Warning time, in milliseconds
z - Repeat time, in milliseconds
Limit the call to <x> milliseconds. Play a warning when <y> mill
iseconds are left. Repeat the warning every <z> milliseconds until time
expires.
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http://www.cellroute.net
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darcy at Vex.Net
Guest
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| Posted: Thu Aug 07, 2014 10:13 pm Post subject: [asterisk-users] Calls not hanging up |
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On Thu, 7 Aug 2014 10:12:02 -0400
"D'Arcy J.M. Cain" <darcy@Vex.Net> wrote:
| Quote: | This just started after upgrading to 11.11.0. After a call is
completed (both ends hang up) the call still shows as active.
|
New data point - I just reverted to 11.10.2 without a single change to
the configuration and the problem has gone away.
--
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
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universe at truemetal.org
Guest
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| Posted: Thu Aug 07, 2014 10:39 pm Post subject: [asterisk-users] Calls not hanging up |
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Am 08.08.2014 05:13, schrieb D'Arcy J.M. Cain:
| Quote: | New data point - I just reverted to 11.10.2 without a single change to
the configuration and the problem has gone away.
|
Hmm. Could this have to do with session-timers (sip.conf)?
I remember when I went from 1.4 to 10.7 I had to manually mess with the
session-timers because my peers who delivered incoming calls would
always end the call after 30 minutes. But your problem is kind of the
opposite. 
Just a shot in the dark, without knowing much about SIP really, lol.
If you really wanna know, you should fire up tcpdump and see what's
going on there.
--
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