On 25/03/2008, Gordon Henderson <[EMAIL PROTECTED]> wrote: > On Tue, 25 Mar 2008, Vieri wrote: > > > How can I "force" soft hangup (if that makes sense)? > > > > "show channels" reveals a stale sip channel. It's of > > an analog phone behind a Grandstream ATA which was > > communicating with another SIP softphone. The latter > > crashed. A soft hangup of the softphone seems to have > > worked but it doesn't for the analog/ATA phone. "show > > hints" also shows that it's InUse. But of course it > > isn't and noone can contact this extension since I > > disabled call waiting for it (I also rebooted the ATA > > and forced re-registration of the ATA SIP client). > > > > So how can I "kill" this channel without restarting > > the asterisk daemon? > > Strangely enough the command in the CLI is: > > soft hangup <channel> > > Just type soft hangup then push the TAB key and it will auto complete for > you ...
I think the OP's point is that "soft hangup ..." does not hang up one of the channels successfully. I have seen this before, usually from soft-phones that do not disconnect cleanly, or WiFi phones that lose signal during a call. Using rtptimeout and rtpholdtimeout settings in sip.conf seems to improve matters, as does using more recent versions of asterisk - You do not say what version you are running. Regards, Steve _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
