On Sun, 11 Oct 2009 02:11:47 +0200 Ivan Stepaniuk <[email protected]> wrote:
> [email protected] wrote: > > > > On the LAN side I can see the INVITE and OKAY messages which end > > with a CANCEL, apparently initiated by the Asterisk gateway. > > > > On the WAN side I can see that my Asterisk gateway is repeatedly > > sending OKAY messages in response to the INVITE from my ITSP. I > > assume the trouble is that these messages are either not getting > > back to my provider or something is blocking the confirmation from > > them. This more or less confirms what was seen in the sip debug > > trace as well. > Post that SIP message from the CLI (sip debug), try adding > "externip=XXX.XXX.XXX.XXX" (your external/public IP address) to your > sip.conf global section, asterisk may be including it's private > address in the OKAY sent to your provider. > > Here's the last message in sip debug before it gives up: ... Retransmitting #6 (no NAT) to 66.51.127.173:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bKee03.6a63373.0;received=66.51.127.173 Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bKpae7KcSeB2Bte Record-Route: <sip:66.51.127.173;lr;ftag=9Z5N4eayXp3Qm> From: "2508864577" <sip:[email protected]>;tag=9Z5N4eayXp3Qm To: <sip:[email protected]>;tag=as32af6364 Call-ID: b734bd26-2fae-122d-91a5-653b331e358a CSeq: 121440337 INVITE User-Agent: Asterisk PBX 1.6.0.15 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 262 v=0 o=root 992672626 992672626 IN IP4 96.50.76.138 s=Asterisk PBX 1.6.0.15 c=IN IP4 96.50.76.138 t=0 0 m=audio 15550 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical packet (see doc/sip-retransmit.txt). Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE) ... 66.51.127.173 is my provider's SIP server 66.51.127.163 is my provider's RTP server I even check DNS to make sure both forward and reverse records jive. Externip was a good suggestion, and worth a try, though because I'm registering with my provider and using dynamic=yes, wouldn't they just reply to that anyway, especially given that the registration works fine? Anyway, after adding externip=<my-external-ip> to [general] and doing a sip reload in the console the problem remains... GM -- Greg Maruszeczka _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
