On Wed, 14 Oct 2009 11:51:02 +0200 Ivan Stepaniuk <i...@albafotonica.com> wrote:
> listm...@websage.ca wrote: > > On Sat, 10 Oct 2009 18:02:04 -0700 > > listm...@websage.ca wrote: > > > >> On Sun, 11 Oct 2009 02:11:47 +0200 > >> Ivan Stepaniuk <i...@albafotonica.com> wrote: > >> > >>> listm...@websage.ca 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:2508864...@66.51.127.163>;tag=9Z5N4eayXp3Qm To: > >> <sip:12504129...@66.51.127.173>;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:12504129...@96.50.76.138> > >> 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 '4e8ef1b977bf0e062212334634080...@192.168.11.1' 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. > Ivan, Thanks for the sympathetic words. After studying the configuration until I was ready to scream and testing everything I could think of I came to the conclusion that it was not likely my very simple setup that was to blame. I opened a ticket with my ITSP and although they were initially quite eager to help, we ultimately couldn't sort it out. I figure they think it's all my fault <shrug>. I moved to another provider with nearly the same configuration and interestingly enough, the problem disappeared. Wish I knew why it failed in the first place but on the other hand I'm happy that I at least have a working phone system once again and can get back to my regular job. Thanks again for your interest in trying to help. 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