On Fri, Jul 08, 2011 at 10:58:06AM -0400, Shawn L wrote: > I have a situation where I have an Asterisk box which receives 8 > analog lines from a > Mitel PBX and then drives 8 cordless SIP phones in a 1-to-1 mapping (a > call coming in > on port 1 of the digium FXO board is delivered to SIP phone 1, an > outgoing call on SIP > phone 2 goes out FXO line 2, etc. > > This works fine normally, but every once in a while (no set time, or > pattern that I can > see -- It may be caused by the wifi sip phone going out of range of an > access point and > not coming back into range fast enough) the FXO port does not hangup > after the call is > terminated and just sits in an in-use state. Since it's a 1-to-1 > mapping, the SIP phone > associated with the in-use line now produces a fast busy when you > attempt to make a > call because it cannot get an outbound line. > > Is there a way to detect that there is no longer really an active call > happening and force a > hangup or reset the channel? It'd be great if this could happen > automatically. Or as a > temporary fix , is there a way to setup and extension that the SIP > phone could dial which > would clear any active calls associated with it? Right now if this > happens, I need to login > to the Asterisk CLI and issue a hangup command. If I don't, the > channel appears to be > in-use forever.
look for 'busydetect' in chan_dahdi.conf . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users