Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

2017-02-13 Thread Anas Moiz 
Ok, I also tried to hangup directly through dialplan, it doesn't work.

  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b0", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b0'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b1", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b1'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b2", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b2'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b3", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b3'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b4", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b4'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b5", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b5'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b6", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b6'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b7", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b7'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b8", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b8'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b9", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b9'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0ba", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0ba'



On Tue, Feb 14, 2017 at 6:03 AM, Joshua Colp  wrote:

> On Mon, Feb 13, 2017, at 08:58 PM, Anas Moiz wrote:
> > Yes Joshua, Its SIP and but the problem is I have tried everything but it
> > doesn't seem to work.
> >
> > In the SIP Trace I can see that I am sending 503 Service Unavailable as a
> > response.
> >
> > You can check the SIP trace attached below:
> >
> > 162.243.107.173:5060 -> 66.226.76.70:5060
> > SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP
> > 66.226.76.70:5060;branch=
> > z9hG4bK643.e44ea565.0;received=66.226.76.70;rport=5060 Via: SIP/2.0/UDP
> > 74.117.36.136;received=74.117.36.136;rport=5060;branch=
> z9hG4bKHBe9cmy3QX2Se
> > From: ;tag=5H54caUKre8gc To: <
> > sip:12023300643@162.243.107.173:5060>;tag=as61c328a0 Call-ID:
> > 15-8824754a-f58560c9-335bcd48-45558f71 CSeq: 103180201 INVITE Server:
> > user_Anas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> > NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length:
> > 0
>
> You would need to determine what will stop the remote server from
> sending you the call again. Once you do that and can provide what it is
> then we can figure out how to get Asterisk to do that. As it is the
> problem isn't Asterisk, it is what is sending you the call.
>
> --
> 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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> org/
>
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

2017-02-13 Thread Joshua Colp 
On Mon, Feb 13, 2017, at 08:58 PM, Anas Moiz wrote:
> Yes Joshua, Its SIP and but the problem is I have tried everything but it
> doesn't seem to work.
> 
> In the SIP Trace I can see that I am sending 503 Service Unavailable as a
> response.
> 
> You can check the SIP trace attached below:
> 
> 162.243.107.173:5060 -> 66.226.76.70:5060
> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP
> 66.226.76.70:5060;branch=
> z9hG4bK643.e44ea565.0;received=66.226.76.70;rport=5060 Via: SIP/2.0/UDP
> 74.117.36.136;received=74.117.36.136;rport=5060;branch=z9hG4bKHBe9cmy3QX2Se
> From: ;tag=5H54caUKre8gc To: <
> sip:12023300643@162.243.107.173:5060>;tag=as61c328a0 Call-ID:
> 15-8824754a-f58560c9-335bcd48-45558f71 CSeq: 103180201 INVITE Server:
> user_Anas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length:
> 0

You would need to determine what will stop the remote server from
sending you the call again. Once you do that and can provide what it is
then we can figure out how to get Asterisk to do that. As it is the
problem isn't Asterisk, it is what is sending you the call.

-- 
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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Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

2017-02-13 Thread Anas Moiz 
Yes Joshua, Its SIP and but the problem is I have tried everything but it
doesn't seem to work.

In the SIP Trace I can see that I am sending 503 Service Unavailable as a
response.

You can check the SIP trace attached below:

162.243.107.173:5060 -> 66.226.76.70:5060
SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 66.226.76.70:5060;branch=
z9hG4bK643.e44ea565.0;received=66.226.76.70;rport=5060 Via: SIP/2.0/UDP
74.117.36.136;received=74.117.36.136;rport=5060;branch=z9hG4bKHBe9cmy3QX2Se
From: ;tag=5H54caUKre8gc To: <
sip:12023300643@162.243.107.173:5060>;tag=as61c328a0 Call-ID:
15-8824754a-f58560c9-335bcd48-45558f71 CSeq: 103180201 INVITE Server:
user_Anas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0
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Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

2017-02-13 Thread Joshua Colp 
On Mon, Feb 13, 2017, at 05:46 PM, Anas Moiz wrote:
> Hi Everyone,
> 
> I am dealing with a problem for now and its really annoying.
> 
> I want to hangup calls from AGI but it seems that my AGI is not rejecting
> the calls properly.
> 
> {
> $agi->verbose("number-not-in-service");
> $agi->exec("Congestion","1");
> $agi->hangup();
> exit;
> }
> 
> with the above logic, all of my calls should be rejected and should be
> disconnected instantaneously.
> 
> But this doesn't seem to be happening, in asterisk CLI I can see
> that AGI is executing multiple times.
> 
> Can anyone tell what I am doing wrong?

Is this SIP? If so what may be happening is that the system sending you
calls may not consider a 503 Service Unavailable (which Congestion will
send) to be a final response which terminates the call and thus send you
a call again, and again, and again, in hopes that you'll accept it.
Since that is behavior of the system sending you calls you would need to
determine if there is any SIP response which will provide the behavior
you need and then find the appropriate cause code to cause it to be
sent.

-- 
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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[asterisk-users] CALLS NOT HANGING UP THROUGH AGI

2017-02-13 Thread Anas Moiz 
Hi Everyone,

I am dealing with a problem for now and its really annoying.

I want to hangup calls from AGI but it seems that my AGI is not rejecting
the calls properly.

{
$agi->verbose("number-not-in-service");
$agi->exec("Congestion","1");
$agi->hangup();
exit;
}

with the above logic, all of my calls should be rejected and should be
disconnected instantaneously.

But this doesn't seem to be happening, in asterisk CLI I can see
that AGI is executing multiple times.

Can anyone tell what I am doing wrong?

Thanks.
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[asterisk-users] Calls not hanging up

2014-08-07 Thread D'Arcy J.M. Cain 
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-000 (None)   Up  AppDial((Outgoing
Line)) SIP/4164251212-0 416555@LocalSets Up
Dial(SIP/thinktel/416555) 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:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] Calls not hanging up

2014-08-07 Thread Asghar Mohammad 
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 da...@vex.net wrote:

 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-000 (None)   Up  AppDial((Outgoing
 Line)) SIP/4164251212-0 416555@LocalSets Up
 Dial(SIP/thinktel/416555) 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:da...@vex.net
 VoIP: sip:da...@vex.net

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Re: [asterisk-users] Calls not hanging up

2014-08-07 Thread D'Arcy J.M. Cain 
On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad asghar...@gmail.com wrote:
 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:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] Calls not hanging up

2014-08-07 Thread D'Arcy J.M. Cain 
On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad asghar...@gmail.com wrote:
 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:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] Calls not hanging up

2014-08-07 Thread Andres 

On 8/7/14, 12:14 PM, D'Arcy J.M. Cain wrote:

On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad asghar...@gmail.com wrote:

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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Re: [asterisk-users] Calls not hanging up

2014-08-07 Thread D'Arcy J.M. Cain 
On Thu, 7 Aug 2014 10:12:02 -0400
D'Arcy J.M. Cain da...@vex.net wrote:
 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:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] Calls not hanging up

2014-08-07 Thread Markus 

Am 08.08.2014 05:13, schrieb D'Arcy J.M. Cain:

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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[Asterisk-Users] Calls not hanging up.

2004-03-02 Thread Darren Wiebe 
I really do not think this is an asterisk problem, but I was not sure 
were else to look for support.  I am using an asterisk setup to phone 
out church services to people via the meetme app.  The way I have it 
setup, you phone in, give the system your telephone number and it calls 
you back via nufone or voicepulse over IAX2.  The complaint I'm getting 
from a few people is that when they hang up their phones, they still 
cannot get dialtone for a while.  Two people said last night that even 
20 seconds after they hung up their phones, when they picked up again, 
they still did not have a dial tone.  I'm not sure when it came back.  
For most people it works fine.  Any suggestions?  I don't think it is 
their phones because it worked fine for both of them other times.  Then 
again, I don't know what it could be besides their phones.

Darren Wiebe
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