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<br>Look at this sip debug its saying something related Retransmitting #1 (no NAT) to 0.0.29.200:5060:<br><br><------------><br> -- Executing [7624@from-sip:1] Macro("SIP/7527-000000c2", "stdexten,7624,SIP/7624") in new stack<br> -- Executing [s@macro-stdexten:1] Dial("SIP/7527-000000c2", "SIP/7624&IAX2/7624,20,t") in new stack<br> == Using SIP RTP CoS mark 5<br>[Apr 8 12:20:53] WARNING[15194]: acl.c:698 ast_ouraddrfor: Cannot connect<br>Audio is at 5060<br>Adding codec 0x4 (ulaw) to SDP<br>Adding codec 0x2 (gsm) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>Reliably Transmitting (no NAT) to 0.0.29.200:5060:<br>INVITE sip:7624 SIP/2.0<br>Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2<br>Max-Forwards: 70<br>From: "Cambridge Guest" <sip:7527@172.30.1.47>;tag=as6f6822ba<br>To: <sip:7624><br>Contact: <sip:7527@172.30.1.47:5060><br>Call-ID: 0ca7784d38d29be168f8f85711c43e4f@172.30.1.47:5060<br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 1.8.3.2<br>Date: Fri, 08 Apr 2011 19:20:53 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>Content-Length: 257<br><br>v=0<br>o=root 1407056235 1407056235 IN IP4 172.30.1.47<br>s=Asterisk PBX 1.8.3.2<br>c=IN IP4 172.30.1.47<br>t=0 0<br>m=audio 16720 RTP/AVP 0 3 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<br><br>---<br>[Apr 8 12:20:53] WARNING[15194]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2ef3f00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument<br> -- Called 7624<br> -- Called 7624<br>Retransmitting #1 (no NAT) to 0.0.29.200:5060:<br>INVITE sip:7624 SIP/2.0<br>Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2<br>Max-Forwards: 70<br>From: "Cambridge Guest" <sip:7527@172.30.1.47>;tag=as6f6822ba<br>To: <sip:7624><br>Contact: <sip:7527@172.30.1.47:5060><br>Call-ID: 0ca7784d38d29be168f8f85711c43e4f@172.30.1.47:5060<br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 1.8.3.2<br>Date: Fri, 08 Apr 2011 19:20:53 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>Content-Length: 257<br><br><br><br><br>> Date: Fri, 8 Apr 2011 11:12:59 -0400<br>> From: pabelanger@digium.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] IAX2/0.0.29.199<br>> <br>> On 11-04-08 10:48 AM, satish patel wrote:<br>> ><br>> > Where this revers IP comes from ?<br>> ><br>> > == Using SIP RTP CoS mark 5<br>> > -- Executing [7623@from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack<br>> > -- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack<br>> > -- Hungup 'IAX2/0.0.29.199:4569-5255'<br>> > -- Executing [s@macro-stdexten:2] NoOp("SIP/7527-0000006b", "IAX2/0.0.29.199:4569-5255") in new stack<br>> > -- Executing [s@macro-stdexten:3] NoOp("SIP/7527-0000006b", "0&0") in new stack<br>> > -- Auto fallthrough, channel 'SIP/7527-0000006b' status is 'UNKNOWN'<br>> ><br>> Asterisk 1.8? Are you using realtime? Looks to be an issue with <br>> netsock2.c.<br>> <br>> -- <br>> Paul Belanger<br>> Digium, Inc. | Software Developer<br>> twitter: pabelanger | IRC: pabelanger (Freenode)<br>> Check us out at: http://digium.com & http://asterisk.org<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br> </body>
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