Hi,
I'm using the default Asterisk function Monitor to record calls, but i have
some issue's with this, the problem is when a call is finished, it never mix
in & out together, bellow you can see my call configuration:
exten => _8.,1,Monitor(wav,${EXTEN},m)
exten => _8.,n,Dial(SIP/${EXTEN:1...@${exten:1})
(the 8 prefix is due to testing of the system)
The reason you see the ex...@exten is because of OpenSips, it's connected to
Asterisk, and some of my users i would like to record are behind opensips
and reachable by dialing <ext>@<domain> but in sip.conf i defined the host,
that's why i'm using ex...@exten.
Even on a normal Asterisk machine, i have issue's with recording, i'm using
Asterisk 1.6.2.
Anybody got any tips on this?
Thanks,
Peter
--
Groet // Kind regards,
Peter den Hartog
Sent from Amsterdam, NH, Netherlands
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users