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 reference 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.
Let me know if any more information would be of use and thanks in advance! The Cisco phone is on IP 192.168.2.203 The Asterisk switch is on IP 192.168.2.30 <-- SIP read from 192.168.2.203:50237: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 192.168.2.30:5060;branch=z9hG4bK3dd277f1;rport From: "JOHN DOE " <sip:[EMAIL PROTECTED]>;tag=as78389007 To: <sip:[EMAIL PROTECTED]:5060>;tag=001380df7eee002b0c2db83c-5ecedbb5 Call-ID: [EMAIL PROTECTED] Date: Fri, 02 Dec 2005 17:04:49 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060> Content-Length: 0 Dec 2 09:04:37 VERBOSE[3842] logger.c: --- (10 headers 0 lines)Dec 2 09:04:37 VERBOSE[3842] logger.c: --- (10 headers 0 lines)--- Dec 2 09:04:37 DEBUG[3842] chan_sip.c: That's odd... Got a response on a call we dont know about. Cseq 102 Cmd SIP/2.0 _______________________________________________ --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
