Nope. If I did, then the phones wouldn't reinvite. -----Original Message----- From: Diyanat Ali [mailto:[EMAIL PROTECTED] Sent: Thursday, December 15, 2005 11:14 AM To: [email protected] 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" <[EMAIL PROTECTED]> >Reply-To: Asterisk Users Mailing List - Non-Commercial >Discussion<[email protected]> >To: "Asterisk Users Mailing List - Non-Commercial >Discussion"<[email protected]> >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. >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
