On Fri, 2005-12-02 at 14:22, [EMAIL PROTECTED] wrote: > Help! I've encountered some problems with Asterisk that Im unable to solve. > We have been running Asterisk version 1.0.9 for many months using a few local > network connected Cisco 7960 phones as SIP clients. All our phones are > currently internal so there is no NAT involved. We were not having any > problems until last week when some strange issues started to crop up. I > started experiencing calls that I initially believed were being dropped, but > discovered that only one side of the conversation had dropped. The other > party could hear me but I couldn't hear them. This seems to happen more often > on longer calls but is not consistent. I am also seeing issues where > incoming or local extension calls that are hung up by the originator before > being answered will continue to ring the SIP phone. At the time the errors > occur, the Asterisk console displays a variety of "...retrans_pkt: Maximum > retries exceeded on call.." messages. I scoured the forums for an answer, > found many refere nce > s to these errors, tried every suggested fix that I could find, but none > have resolved these problems. After working on the problem for several days, > I finally built a new box and installed Asterisk 1.2 on it. Using this new > 1.2 box I no longer see the "Maximum retries exceeded on call" warnings on > the console but still experience the strange behavior. Unfortunately, the > errors occur randomly so I am unable to reproduce the error on demand. I > turned on SIP debugging and set console logging to debug and captured an > instance of the problem with the hang up not being recognized. The details > are below: > > I dial in from my cell phone. My Cisco phone begins to ring. I then hang up > my cell phone. Asterisk acknowledges the hang up, but the Cisco phone > continues to ring. After a minute or so, or if I pickup the phone, Asterisk > display the following message "That's odd... Got a response on a call we > dont know about. Cseq 102 Cmd SIP/2.0" I've included a copy of the console > output when this occurs that shows both the SIP message and the Asterisk > debug output.
Odds are you have local network congestion -- Dropped packets or delayed packets. Try moving your phone and asterisk server to an isolated network switch - no other traffic (certainly no computers) - then test. If the problems go away, then update your virus scanners and check your computers. Good Luck Jon Carnes _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
