[asterisk-users] Debug dropped calls

[asterisk]

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.