Trust me, I don't WANT you to look at my code, it's butt-ugly! lol, lust kidding... -- but at http://www.astsee.com/ you can download the source code to my AstSee project -- it may provide some insight into what needs to be (or CAN be) gleaned from asterisk. I struggled with all this a year or more ago and manged to get it to work fairly as expected. A problem you might notice is that I believe I mistakenly update my internal arrays before asterisk's manager interface has sent a complete packet... argh.... But you can see me dealing with NewState, NewExten, NewChannel, etc etc and what I do with them :)
Moj Devraj Mukherjee wrote: > Thanks all :) > > Appreciate it. > > On Feb 1, 2008 12:04 PM, Ex Vito <[EMAIL PROTECTED]> wrote: > >> I've struggled with this recently. In short: >> >> >> - Observed behaviour is expected as of asterisk 1.2 and later, >> as previously described by Mojo >> >> - If you want to get the caller id for the channel calling (dialling) >> into that channel for that specific Newstate: Ringing event, you >> can use the 'o' flag to the Dial command; in this case you'll get >> old asterisk 1.0 behaviour -- do you really want to depend on >> such an old behaviour ? well I decided I didn't... >> >> - Otherwise, you'll need to track other events (IIRC, at least, Dial, >> AgentCalled, Newstate, etc) in the AMI so as to know who is calling >> who at a given instant >> >> - BEWARE: if memory serves me right (search the list archives in the >> Nov/Dec >> timeframe), the behaviour is not 100% homogeneous for different channel >> types SIP, ZAP, mISDN, IAX, etc. What this means for a simple Dial() from >> one channel to the other is that a) at times you get the Dial >> event first then the >> Newstate: Ringing event; and that b) with other/different >> orig/dest channel types >> you'll get the events in the reverse order... Nothing much but: i) >> you'll have to >> track them either way and ii) it reveals that the AMI events >> aren't 100% clean!!! >> >> :/ >> -- >> exvito >> >> >> On Feb 1, 2008 12:08 AM, Mojo with Horan & Company, LLC >> <[EMAIL PROTECTED]> wrote: >> >>> The snippet is asterisk telling you "I'm just letting you know that the >>> correct caller id for Channel: SIP/103-098500d8 is CallerID: 103" >>> >>> This is absolutely correct, it's just not a piece of information you >>> expected to be receiving at that point. >>> >>> You probably also received a packet like that with the following: >>> Channel: SIP/101-xxxxxxxx >>> CallerID: 101 >>> telling you, again, the caller id for only that channel. >>> >>> Moj >>> >>> >> _______________________________________________ >> -- 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 >> >> > > > > _______________________________________________ -- 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