On Tue, 2011-08-02 at 10:58 +0100, Ishfaq Malik wrote: > Hi > > I'm using asterisk 1.8.3.2 (with a couple of patches) > > I have the following scenario... > > SIP call comes in and gets answered by extension A (MixMonitor is > executed as part of this inbound dial plan of the number being called) > > Extension A puts call on hold and calls extension B > > Extension A then does an attended transfer of incoming call to extension > B > > I'm finding that the recording only lasts up to the point that the > transfer is made. > > Is this the correct behaviour? Is there any way I could make this > inbound call into a single continuous recording? > > Thanks in advance > > Ish
Here's part of the log for this procedure [2011-08-02 13:47:13] VERBOSE[6475] rtp_engine.c: -- Locally bridging SIP/A-00000049 and SIP/B-0000004a [2011-08-02 13:47:20] VERBOSE[6475] rtp_engine.c: -- Locally bridging SIP/inbound-00000047 and SIP/B-0000004a [2011-08-02 13:47:20] VERBOSE[6463] pbx.c: == Spawn extension (inbound, s, 4) exited non-zero on 'SIP/A-00000049<ZOMBIE>' [2011-08-02 13:47:20] VERBOSE[6464] app_mixmonitor.c: == MixMonitor close filestream [2011-08-02 13:47:26] VERBOSE[6475] app_macro.c: == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/inbound-00000047' in macro 'stdexten' [2011-08-02 13:47:26] VERBOSE[6475] pbx.c: == Spawn extension (local, B, 1) exited non-zero on 'SIP/inbound-00000047' [2011-08-02 13:47:26] VERBOSE[6464] app_mixmonitor.c: == End MixMonitor Recording SIP/inbound-00000047 Obviously, I've obscured some of the more sensitive details in there The thing to notice here though is that MixMonitor closes the filestream when I hit the transfer button but actually Ends the recording 6 seconds later when the whole call was ended. This seems like inconsistent behaviour and more like an unintentional consequence of changes rather than intended behaviour, i.e. why would you close the filestream yet not end the recording? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
