HI, I got a one way audio when an ip phone dial to another ip phone in the same network. What I find is TCP & UDP run different legs. Below is my configuration.
asterisk (192.168.1.10) ipphone-A (192.168.1.111) ipphone-B (192.168.1.101) router (192.168.1.1) external IP (116.48.138.83) When A makes call to B, signal from A to router goes in the internal network. Then B pickup the call and I find that B will use external IP to reach the router. The signal from B finally can't reach to A. Below is a flow and you can see it involves using external IP. Is it related to the setting? Where and how to set it to make it work? U 192.168.1.10:5060 -> 192.168.1.101:5060 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0. Via: SIP/2.0/UDP 116.48.138.83:5060;branch=z9hG4bK612c1103;rport. From: "111" <sip:[EMAIL PROTECTED]>;tag=as0a0b2a95. To: <sip:[EMAIL PROTECTED]:5060>. Contact: <sip:[EMAIL PROTECTED]>. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: PBX. Max-Forwards: 70. Date: Fri, 13 Jun 2008 17:20:14 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 21200 21200 IN IP4 116.48.138.83. s=session. c=IN IP4 116.48.138.83. t=0 0. m=audio 19770 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users