[Asterisk-Users] Dropped Calls between Sip and Zaptel

[Andrew Elchuk]

Hi,

I am having trouble with dropped calls in Asterisk.  I've done a bunch 
of searching but all I could find was setting busydetect and 
callprogress to yes in zapata.conf to help combat the problem, but I did 
this to no avail.  The following is the output from 
/var/log/asterisk/full at the time the call was dropped on me.

May 13 08:37:13 DEBUG[5379]: Stopping retransmission on 
'660440d07b1155645cabd0ce681609d9 at 10.125.1.245' of Request 102: Found
May 13 08:37:16 DEBUG[8480]: Didn't get a frame from channel: 
SIP/cronus-116-78ed
May 13 08:37:16 DEBUG[8480]: Bridge stops bridging channels Zap/1-1 and 
SIP/cronus-116-78ed
May 13 08:37:16 DEBUG[8480]: update_user_counter(cronus-116) - decrement 
outUse counter
May 13 08:37:16 DEBUG[8480]: Exiting with DIALSTATUS=ANSWER.
May 13 08:37:16 VERBOSE[8480]:   == Spawn extension 
(macro-netvoice-stdexten, s, 302) exited non-zero on 'Zap/1-1' in macro 
'netvoice-stdexten
May 13 08:37:16 VERBOSE[8480]:   == Spawn extension (main-menu, 116, 1) 
exited non-zero on 'Zap/1-1'
May 13 08:37:16 DEBUG[8480]: Hangup: channel: 1 index = 0, normal = 21, 
callwait = -1, thirdcall = -1
May 13 08:37:16 DEBUG[8480]: disabled echo cancellation on channel 1
May 13 08:37:16 DEBUG[8480]: Set option TDD MODE, value: OFF(0) on Zap/1-1
May 13 08:37:16 DEBUG[8480]: Updated conferencing on 1, with 0 
conference users
May 13 08:37:16 VERBOSE[8480]:     -- Hungup 'Zap/1-1'
May 13 08:37:17 DEBUG[5379]: Auto destroying call 
'0c1a1511cf06369e467f66c9bd69a571 at 10.125.1.220'

Does anyone know any other solutions to this problem?  Faulty hardware 
perhaps?  By the way I have a TDM400P with 4 FXO ports on it and have 
asterisk running on Debian 3.0.

Thanks