Hey,
I've done some searching for this and never really found a concrete
answer. Is there a specific reason or solution why just in the middle
of a call Asterisk will drop it and I'll get dial tone again? Anyways,
this is the output from /var/log/asterisk/full at the time of disconnection:
May 13 08:37:13 DEBUG[5379]: Stopping retransmission on
'660440d07b1155645cabd0ce681609d9@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@10.125.1.220'
Any ideas/solutions would be greatly appreciated.
