Unfortunately the only log messages regarding that channel are the "joined"
and the "left" for both legs.
VERBOSE[18771][C-066c] bridge_channel.c: Channel
SIP/201-boxoffice-0f66 joined 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
VERBOSE[18779][C-066c]
Hi,
OK, then it looks like the client transferred the call anywhere else.
Do you see an entry in your log that refers to the bridge ID
00bd58c3-3bce-4f1b-9d79-11eb96f37260 ?
If there was a transfer, the call *may* have been bridged with the transfer
destination. Also, the destination might be
No, there is no Music On Hold starting and the bad thing is the call
duration reported by asterisk was just few seconds while the call duration
reported by the provider was few thousand seconds, the max allowed. So they
will be able to terminate the call on the asterisk side and have it run on
the
Maybe the client just put the call on hold.
So the call technically has not ended AND the client does not need to
send or handle any RTP data.
Is there any mention of "music on hold" for this channel?
Greetings
Max
- Nachricht von Leandro Dardini -
Datum:
The best is to get a PCAP so you can see exactly what is going on. Look
into voipmonitor.org or homersip to capture all of your traffic. There are
many ways that people commit fraud. If you are thinking about transfers
what they do is the call the fraudulent number then they send a 302 to
another
I am banging my head over a simple asterisk trick I was seeing on one
asterisk server.
An extension dials an international premium number, the called number
answers, then the extension hangups, but the call continue to run on the
international number side, generating an high profit for the