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/66.226.76.70-0000d0b0", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0b0'
== Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-0000d0b1", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0b1'
== Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-0000d0b2", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0b2'
== Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-0000d0b3", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0b3'
== Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-0000d0b4", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0b4'
== Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-0000d0b5", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0b5'
== Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-0000d0b6", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0b6'
== Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-0000d0b7", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0b7'
== Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-0000d0b8", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0b8'
== Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-0000d0b9", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0b9'
== Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-0000d0ba", "41") in new stack
== Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-0000d0ba'
On Tue, Feb 14, 2017 at 6:03 AM, Joshua Colp <jc...@digium.com> 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: <sip:2126555763@66.226.76.70:5060>;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
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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