asterisk users - Dec 2005 - Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem

I have the follwoing setup:
Asterisk  SVN-tag-1.2.1-r7367
6 Polycom 500 Sip version 1.5.x
4 Sipura SPA3000 (not sure what build) (FXO port)
All on flat single network, no NAT, and no gateways to reach each other.
Sometimes (happens around 3 times a day, but sometimes far more
often), while on the phone to an outside caller (on the PSTN using the
FXO on the spa3k), the call dissconects from the polycom and goes thru
the incoming extension for the sipura. In other words, astrisk at
least as far as I can see from what gets executed in the DP (and maybe
spa3k) sees this as if the follwoing has happened: 1. The polycom user
hungup, 2. A new call came in on the spa3k.
The follwoing is part of the log that I think might help:
Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from
channel: SIP/201-8ba1
Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels
SIP/201-8ba1 and SIP/804-fd83

SIP/201 is the Polycom, while SIP/804 is the spa3k.

If I'm losing a frame, is there a way to configure asterisk not to
drop the channel? Or is this something the Polycom/Sipura are doing?

FYI, asterisk is running on a VIA/MPIA platform.

Thank You

> I have the follwoing setup:
> Asterisk  SVN-tag-1.2.1-r7367
> 6 Polycom 500 Sip version 1.5.x
> 4 Sipura SPA3000 (not sure what build) (FXO port)
> All on flat single network, no NAT, and no gateways to reach each other.
> Sometimes (happens around 3 times a day, but sometimes far more
> often), while on the phone to an outside caller (on the PSTN using the
> FXO on the spa3k), the call dissconects from the polycom and goes thru
> the incoming extension for the sipura. In other words, astrisk at
> least as far as I can see from what gets executed in the DP (and maybe
> spa3k) sees this as if the follwoing has happened: 1. The polycom user
> hungup, 2. A new call came in on the spa3k.
> The follwoing is part of the log that I think might help:
> Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from
> channel: SIP/201-8ba1
> Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels
> SIP/201-8ba1 and SIP/804-fd83
> 
> SIP/201 is the Polycom, while SIP/804 is the spa3k.
> 
> If I'm losing a frame, is there a way to configure asterisk not to
> drop the channel? Or is this something the Polycom/Sipura are doing?
> 
> FYI, asterisk is running on a VIA/MPIA platform.
Pure guess is that something happened (unknown what) and the error messages
posted above are the result of that, and not the root cause. Finding the
root cause may require you to implement the "syslog server" and
"debug
server" options in the spa3k, and compare those log entries to what *
records for log messages during a failure.

Implementing the log functions on the spa3k "does" require a reboot.
Their
log messages are rather cryptic, but looking at keywords and timestamps
might identify which box(es) are involved with the dropped calls.