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 -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 _______________________________________________ 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