Hello. I think i've seen this problem, it was generated by a missing ACK on 200 OK. If that is the case try modifying session timer parameters in sip.conf so a missing ACK will not lead to call termination.
Peter Ishfaq Malik wrote: > Hi > > I'm having an issue with just one of the phones (snom300) attached to > our asterisk server (1.4.17 using RealTime) > Sometimes (not consistently), any outbound call cust off at 20 seconds > exactly and I see the following in my asterisk console > [Aug 6 10:37:24] WARNING[1679]: chan_sip.c:1946 retrans_pkt: Maximum > retries exceeded on transmission 3c3251e0edaf-4jnjbmy9uupi for seqno 2 > (Critical Response) > [Aug 6 10:37:24] WARNING[1679]: chan_sip.c:1970 retrans_pkt: Hanging up > call 3c3251e0edaf-4jnjbmy9uupi - no reply to our critical packet. > > There as another SIP phone plugged into the same router and that has no > issues at all and inbound calls are not affected either. > The codecs order on the phone match up to those set on the server > (g729;alaw;ulaw;;). > > There are about 50 other phones attached to the server and none of the > others have this issue. Well actually, one did but that person got a new > handset (they were previously using a very old and rubbish Grandstream) > and the problem immediately stopped. > > Has anyone experienced anything like this before? > > _______________________________________________ -- 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
