Are the RTP timers applicable with canreinvite=yes ? > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Olle E Johansson > Sent: 22 February 2007 10:49 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Channels hanging when SIP phone > gets resetduring call > > > 21 feb 2007 kl. 12.54 skrev Steve Langstaff: > > > Hi All. > > > > This is on Asterisk 1.2.13 > > > > I place a call between 2 SIP phones (with canreinvite=yes, > > qualify=yes). > > > > I reset the phones (so they don't have time to say BYE). > > > > Asterisk seems to think that the call is still ongoing. > This persists > > until I do a 'restart now'. > Check the RTP timers in sip.conf. They will hangup the call > if there's no audio. > > /O > _______________________________________________ > --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 --
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