[email protected] wrote: > On Sat, 10 Oct 2009 18:02:04 -0700 > [email protected] wrote: > >> 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... >> > > > [Bumping this in the hope that someone might have some new insight or > suggestions since I posted this on a holiday weekend (in my part of > the world anyway)...]
I was staring at the SIP transcript and I don't see anything wrong, I'm out of suggestions, except that you could analyze and compare the packets when your phone is connected directly (if it's physically possible). I hope someone throws some light over this. -- Iván Stepaniuk Alba Fotónica S.L. _______________________________________________ -- 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
