The best type of error possible, intermittent.
We have PSTN numbers being switched to SIP then forwarded to our Asterisk
server which sits inside our LAN
Every once and a while (maybe 1 out of every 20 calls) goes like this:
-- Executing Answer("SIP/213.199.36.50-0818e3e8", "") in new stack
-- Executing Ringing("SIP/213.199.36.50-0818e3e8", "") in new stack
-- Executing Dial("SIP/213.199.36.50-0818e3e8", "ZAP/g1/:8213") in new
stack
-- Called g1/:8213
-- Zap/1-1 answered SIP/213.199.36.50-0818e3e8
-- Hungup 'Zap/1-1'
== Spawn extension (from-gv-uk, 441252580625, 3) exited non-zero on
'SIP/213.199.36.50-0818e3e8'
Looks normal right? During this whole exchange, neither side can hear the
other. Not even a ringing sound.
The above looks no different than the successful calls.
Has anyone seen this type of behavior before?
Thanks!
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
