IRC log for #asterisk-doc on 20090326

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00:45.45Juggieleifmadsen, it would be possible using the macro on dial i think.
00:56.04leifmadsenit's ok -- I solved it a different way... a more manual process, but saves me a ton of time.
00:59.37leifmadsenAnd i'm not sure that would really work. What I'm trying to do is determine, after a SIP transfer, whether the initial call (that had call recording enabled) should continue to record the channel. I just enabled a feature code in features.conf and told them to press ** to stop recording on calls they don't want recorded.
01:00.41leifmadsensince there is no dialplan being executed at the point the attended transfer is completed, there is really no good way have asterisk determine when to stop that call
01:00.49leifmadsenI think it's a feature that is lacking
01:01.27leifmadsenideally I could do a 'setvar' in sip.conf, and then tell asterisk to check that variable just prior to a bridge, and to do a StopMonitor() on the channels being bridged together
01:01.29leifmadsen(or something like that)
01:21.43leifmadsenjsmith: ping>?
03:11.25Juggieleifmadsen: ping
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11:39.34leifmadsenJuggie: pong
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20:12.59leifmadsenhmmmmmmmmm.... am I just forgetting a way of doing this from the dialplan, or is using the AMI the only way of doing this;  I want to call multiple people and put them into a meetme conference (all of them, not just the first to answer)
20:21.11leifmadsennevermind... Originate() to the rescue!

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