[asterisk-users] CALLS NOT HANGING UP THROUGH AGI

[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: <sip:2126555763 at 66.226.76.70:5060>;tag=5H54caUKre8gc To: <
> sip:12023300643 at 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