|Summary:||ASTERISK-06373: [request] Enable Monitor when the RTP is not going through the Asterisk|
|Date Opened:||2006-02-20 03:36:26.000-0600||Date Closed:||2011-06-07 14:03:16|
When the RTP is going peer-to-peer, bypassing the Asterisk, it is not possible to monitor the call. When the Monitor command is issued, in this scenario, Asterisk produces empty audio files.
Is it possible that when Asterisk receives the Monitor (and RTP is not available to it and canreinvite is set to yes) it performs the SIP reINVITE so that from that point the RTP is going through the Asterisk. This way Asterisk would be capable of performing call monitoring even if RTP was initially bypassing the Asterisk.
|Comments:||By: Tilghman Lesher (tilghman) 2006-02-21 20:09:26.000-0600|
We allow feature requests to remain open on the bugtracker for several days, but the real place these need to go is on the Wiki. If you like, you may also post a bounty for the development of this feature.
You may reopen if/when you have a patch to test.