I am having same trouble , but I dont have any Zaps just SIP and h323 is very strange some calls can be active very long time but sometimes it just get cutted off.
may be the silence suppresion I will look forweard in that regards Humberto On Wed, 26 Jan 2005 14:28:39 -0500, Noah Miller <[EMAIL PROTECTED]> wrote: > > I got a problem with asterisk 1.0.2 - it drops the calls, > > both sip<-->sip, and zap<-->sip. > > > > The conntions can stay for seconds to several minuttes, > > and then the connection just cut off. > > > > I can't see anything in the logfiles. (or dont know what > > to look at.) > > > > It drops several connections at a time, but not all. > > > > Where to start looking ?? > > > > /HHA > > First, make sure you are running asterisk with enough verbosity. Try: > > asterisk -vvvvvvrgc > > That should tell you at least something that happens when the connection is > cut. > > Second, more information - what SIP devices are you using? What Zap hardware > and PSTN connections? (This sounds strangely like the common problem with > X-lite, the Xten softphone. If you are using X-lite, make sure to turn off > silence suppression.) If you are using another SIP device - what is it? > > Last, can we see the pertinent sections of your config files > (extensions.conf, sip.conf, zapata.conf, zaptel.conf)? > > Thanks, > Noah > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
