[Asterisk-Users] Shutting down Asterisk when not in RTP Stream

[Douglas Garstang]

Nope. If I did, then the phones wouldn't reinvite.

-----Original Message-----
From: Diyanat Ali [mailto:diyanat at hotmail.com]
Sent: Thursday, December 15, 2005 11:14 AM
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] Shutting down Asterisk when not in RTP
Stream


Do you have 't' or 'T' in the Dial Application?

Diyanat


>From: "Douglas Garstang" <dgarstang at oneeighty.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>
>To: "Asterisk Users Mailing List - Non-Commercial 
>Discussion"<asterisk-users at lists.digium.com>
>Subject: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream
>Date: Thu, 15 Dec 2005 10:38:22 -0700
>MIME-Version: 1.0
>
>I'm very confused about something.
>
>I have two phones that have reinvited and have an RTP session open. I 
>confirmed this by running ngrep on the Asterisk box. Asterisk still shows 
>the calls on the console.
>
>*CLI> sip show channels
>Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last 
>Message
>192.168.10.125   a00090201   45dfabad1bd  00103/00000  ulaw  No       Tx: 
>ACK
>192.168.10.4     a00090101   ca3279d8-3e  00102/00001  ulaw  No       Tx: 
>ACK
>
>When I shut asterisk down, the call terminates. I don't understand that. If 
>Asterisk isn't in the RTP path, how can shutting it down terminate an 
>active call?
>
>Don't know if it's relevant, but the 192.168.10.4 is an OpenSER box.
>
>Thanks.
>Doug.
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