[asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

Leandro Dardini ldardini at gmail.com
Thu Sep 15 11:06:14 CDT 2016

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 premium number
company and a big loss for the asterisk owner.

I think some sort of "transfer" takes place, but I can't identify how they
do it and most important, how to prevent it.

Here the relevant logs:

[2016-09-08 21:00:25] VERBOSE[18771][C-0000066c] pbx.c: Executing
[0021628990XXX at dialoutbound:595] Dial("SIP/201-boxoffice-00000f66",
"SIP/0021628990XXX at SBC002_VirginMedia,60,T") in new stack
[2016-09-08 21:00:25] VERBOSE[18771][C-0000066c] app_dial.c: Called
SIP/0021628990XXX at SBC002_VirginMedia
[2016-09-08 21:00:27] VERBOSE[18771][C-0000066c] app_dial.c:
SIP/SBC002_VirginMedia-00000f67 answered SIP/201-boxoffice-00000f66
[2016-09-08 21:00:27] VERBOSE[18771][C-0000066c] bridge_channel.c: Channel
SIP/201-boxoffice-00000f66 joined 'simple_bridge' basic-bridge
[2016-09-08 21:00:27] VERBOSE[18779][C-0000066c] bridge_channel.c: Channel
SIP/SBC002_VirginMedia-00000f67 joined 'simple_bridge' basic-bridge
[2016-09-08 21:00:28] VERBOSE[18771][C-0000066c] bridge_channel.c: Channel
SIP/201-boxoffice-00000f66 left 'simple_bridge' basic-bridge
[2016-09-08 21:00:28] VERBOSE[18779][C-0000066c] bridge_channel.c: Channel
SIP/SBC002_VirginMedia-00000f67 left 'simple_bridge' basic-bridge

Any idea?

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