Greetings, I have a custom built click to dial system that integrates with our Intranet (Windows2003/IIS6) and MSSQL 2000 DB. It uses a mix of JavaScript, PHP, Apache, and Asterisk dial logic to accomplish its tasks. However, I have hit a snag and am unable to determine where to troubleshoot any more as everything I see looks normal.
My users are experiencing dropped calls while using our Asterisk system. Most of the problems are with our "click to dial" system, but as that is how most of the calls are made, that makes sense. I have received a few complaints of incoming or manually dialed calls being dropped, but unfortunately I do not have any debug/trace information for those calls. I have not heard of any instances of internal calls being dropped. I have a Sangoma A104d QUAD T1/E1 AFT card with 2 PRIs connected (port 1 and 2). The phones are mostly Polycom 330s with SIP version 3.0.0.0258. The server is a Dell 2950 with 4 gigs of RAM. I have almost 170 SIP peers with less than 25 external active calls at any one time. All of the phones and Asterisk are on the same LAN connected using 4 Cisco 3650s (most phones are powered via POE) and 2 Dell PowerConnect 5224s. I have tried searching for "asterisk debug dropped calls" and various similar terms and was unable to find a scenario close to what I am experiencing. The system was recently upgraded from this: Asterisk 1.4.11 wanpipe-2.3.4-13 libpri-1.4.1 zaptel-1.4.5.1 to this: Asterisk 1.4.20 WANPIPE Release: 3.2.5 libpri-1.4.4 zaptel-1.4.10.1 The upgrade was primarily to try and alleviate the "dropped call" problem, however, it appears to have no affect on the issue. Here are the various configs: http://pastebin.ca/1056735 - zaptel.conf http://pastebin.ca/1056739 - zapata.conf http://pastebin.ca/1056733 - asterisk config for the click to dial http://pastebin.ca/1056753 - how the click to dial call is originated http://pastebin.ca/1056731 - output from verbose level 3 or 4 as well as pri intense debug span 1, the only slight change I made was to obscure part of the phone number dialed When listening to the recording (as a result of line 7 in http://pastebin.ca/1056733) I hear the "Please wait while I connect the call", and then it rings once and the recording stops. This final pastebin is a SIP conversation (via wireshark) for a different call that was reported as "dropped", however, I don't have the additional logging information (and I don't have the packet information for the dropped call in the above output), but I'd bet it would be a similar exchange: http://pastebin.ca/1056748 The problem is not tied to a specific channel on the PRIs as dropped calls have been experienced on various individual channels. Sometimes the call drops almost immediately after connection, other times several minutes into the conversation. >From my understanding of the SIP message and the PRI debug, it looks like the phone is sending a BYE message, and of course Asterisk is hanging up the call and clearing the channel on the PRI. I can only think of 2 possible locations for the problem: 1) The phones think they need to send a BYE before they actually do 2) The PRI/telco side of Asterisk (unlikely based on the SIP packet) Am I correct in understanding the PRI output and SIP message? If so, what can I check on the phone side of things? Or is there a 3rd (or more) place where the problem could be? Mike McLain ----------------------------------------------------------------------------- The information contained in this e-mail message is confidential and/or privileged and is intended only for the use of the individual or entity named above. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. If you are not the intended recipient you are hereby notified that any unauthorized disclosure, copying, distributing or taking any action in reliance of contents of this information is strictly prohibited. ----------------------------------------------------------------------------- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
