I have a problem with calls not hanging up if for some reason the physical phone dies or gets unplugged I can demonstrate this in practice by making a call from a handset, then unplugging the handset from the power. The call remains active and asterisk never seems to disconnect it. More annoyingly when power is re-applied the handset comes back to life, won't receive incoming calls (because asterisk thinks it's busy), but likewise the handset itself doesn't think it's in a call so it can't retrieve the call or do a proper hangup. I have no NAT in place and the handsets are all set to register/login and "qualify=yes" set (which I had hoped would sort this...) The handsets are SNOM 360s but I don't think this is directly relevant. Asterisk is setup to use FreePBX dialplan (but again don't think this is relevant?) Can someone please suggest a way to ensure that the calls get hungup - we had a 9 hour call earlier before someone noticed.... It's rare, but the consequences are potentially quite dire. Cheers Ed W
Ed W wrote:> I have a problem with calls not hanging up if for some reason the > physical phone dies or gets unpluggedHave you tried the RTP timeout settings in sip.conf? ;--------------------------- RTP timers ---------------------------------------------------- ; These timers are currently used for both audio and video streams. The RTP timeouts ; are only applied to the audio channel. ; The settings are settable in the global section as well as per device ; ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity ; on the audio channel ; when we're not on hold. This is to be able to hangup ; a call in the case of a phone disappearing from the net, ; like a powerloss or grandma tripping over a cable. ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity ; on the audio channel ; when we're on hold (must be > rtptimeout) ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open ; (default is off - zero) Best regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk -> http://www.das-asterisk-buch.de
Philipp Kempgen wrote:> Ed W wrote: > > >> I have a problem with calls not hanging up if for some reason the >> physical phone dies or gets unplugged >> > > Have you tried the RTP timeout settings in sip.conf? >Sounds exactly like what I need! Thanks Is there no default set then?? Cheers Ed W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070118/f455be66/attachment.htm