I have two trunks to the same machine (x.x.x.2), one is type=friend, other is type=peer. Asterisk seems to choose which trunk to use by the order by which they are set out in sip.conf. When a incoming call comes into Asterisk, it always uses the last trunk. My understanding was that a peer trunk can't receive incoming calls. Does Asterisk ignore the type when dealing with incoming calls from the same host/machine ?
I want all incoming calls to use the back-trunk only. When I change the order around from what it looks like below it works perfectly. I've been told that order of things appearing in sip.conf should not matter. --Shaun sip.conf: [back-trunk] type = friend username = 8880006111 secret = vvvvvv host = x.x.x.2 dtmfmode = rfc2833 nat = no canreinvite = no insecure = port,invite qualify = no disallow = all allow = ulaw allow = alaw allow = g729 context = shared-back-trunk-incoming [back-trunk-ulaw] type = peer username = 8880006113 secret = vvvvvv host = x.x.x.2 dtmfmode = rfc2833 nat = no canreinvite = no insecure = port,invite qualify = no disallow = all allow = ulaw context = shared-back-trunk-ulaw-incoming Asterisk CLI: Aug 10 12:17:15 DEBUG[21756]: chan_sip.c:7242 check_user_full: Setting NAT on RTP to 0 Aug 10 12:17:15 DEBUG[21756]: chan_sip.c:10497 handle_request_invite: Checking SIP call limits for device 8880006113 Aug 10 12:17:15 DEBUG[21756]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Match Found _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
