> Call comes in from remote SIP, authorised, does the following and dies > > Any idea why.. > > I have ports 5060 and 16384 to 16482 open > > Do I need any others? > > What am I missing > > Remote user is using X-lite for windows.. > > -- Executing Dial("SIP/2004-944c", "SIP/2001|20") in new stack > -- Called 2001 > -- SIP/2001-4f3b is ringing > -- SIP/2001-4f3b answered SIP/2004-944c > -- Attempting native bridge of SIP/2004-944c and SIP/2001-4f3b > Jul 5 19:04:17 WARNING[-224801872]: chan_sip.c:497 retrans_pkt: Maximum > retries exceeded on call > [EMAIL PROTECTED] for seqno 24922 > (Response)
It is dying because the audio stream (rtp packets) aren't getting through. Not sure why you picked rtp ports 16384-16482; each sip phone vendor picks there own set of port ranges, and the Xlite product use to use ports in the 8000 range (haven't checked lately). Read the stuff on the wiki relative to NAT parameters (for *) and you should be able to get it to work. _______________________________________________ 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