Do you have busydetect=yes and/or callprogress= in zapata.conf? If so set them to no.
On Mon, 2004-02-02 at 11:10, Steve Foy wrote: > Hi, > > On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: > > Steve, > > > > this really is a FAQ. You need add to EACH (!) sip user something like > > > > disallow=all > > allow=ulaw > > allow=alaw > > allow=gsm > > I do have that in my sip.conf. I am using ulaw. > > Calls from the SIP phones through Asterisk and out one of my X100P cards are > working 95% of the time and also, incoming calls through the X100P cards to > the SIP phones are the same. > > The only problem is that every once in a while, without any odd circustances > that I can see, the call just drops and the remote user is gone. > > The box running asterisk isn't under heavy load, so I can't see why this is > happening. > > I am not using g.729 or 723, just plain old ulaw, which I have got enabled in > sip.conf > > Cheers, > Steve -- Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the "Asterisk Resource Pages". BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 _______________________________________________ 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