Hi, I'm trying to get rtp media streams to run between endpoints rather than through my * server, and I think I'm getting something wrong. I have an AS5300 speaking both h323 (for a different voip system I run) and sip for *. Dial-peers on the as5300 differentiate inbound from pstn to different chunks of DID numbers between h323 and sip. I'm testing with xlite on a PC.
So here's what I have: Outbound trunks are defined in my extensions.conf that send _9whatever to SIP/pstn_gw/${EXTEN}. In sip.conf I have two friends, one for my xlite softphone, one for pstn_gw: [2085551212] type=friend username=2085551212 secret=1234 host=dynamic canreinvite=yes disallow=all allow=ulaw context=testme mailbox=5551212 callerid="Jeremy Jones" <2085551212> [pstn_gw] type=friend username=pstn_gw disallow=all allow=ulaw context=default canreinvite=yes host=10.0.0.201 I can place a call from the PSTN to 5551212 successfully, and I can place calls from xlite to the PSTN successfully. But in either case I always see two sip channels active on *, and the endpoints (as5300 & xlite) are sending their rtp via *. Here's what I see when I place a call from xlite to: *CLI> -- Executing Prefix("SIP/2085551212-f04d", "9") in new stack -- Prepended prefix, new extension is 93532533 -- Executing Dial("SIP/20825551212-f04d", "SIP/pstn_gw/93532533") in new stack -- Called pstn_gw/93532533 -- SIP/pstn_gw-85a0 is making progress passing it to SIP/2085551212-f04d -- SIP/pstn_gw-85a0 answered SIP/2085551212-f04d -- Attempting native bridge of SIP/2085551212-f04d and SIP/pstn_gw-85a0 *CLI> *CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 10.0.0.201 93532533 02e2e09e167 00103/00651 00000ms 0000ms ULAW 10.0.0.100 2082874602 E0541F6D-81 00102/03763 00000ms 0000ms ULAW 2 active SIP channel(s) *CLI> (I have a Prefix rule for outbound 'cuz this is a system for residential users, and the as5300 has dial-peers that need a 9 prefix...) The output in * is similar for inbound from PSTN to xlite. I can send output from sip debug if that'd help. Thanks, Jeremy Jones Network Nerd WestCom, LLC _______________________________________________ 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