[asterisk-users] RE: [asterisk-dev] Phone status

[Mir]

Your are right, I dont have to invent the wheel again, and I'm getting
cleverer by looking at other peoples code.

But this does not solve my problems, I have worked in the PABX business as a
software developer for about 8 years, and coming to * is not all that easy.

For instance, * does not give you very good information of the state of
extensions (like we are used to in the "old-fashioned" PABX business), or
maybe I'm not good at finding the information.

I'm trying to port an existing Windows application to *, its a dialer, used
to dial and se information about received calls.

I know how to dial new calls, by using ORIGINATE on the AMI.
I can receive some status information via the AMI, but consider this
example:

I receive a call, which I accept. I get an event from the AMI, that the call
is now in the UP state.
I receive another call, I get en event from the AMI, that the new call is in
the RINGING state.

So far, so good.

I now answer the other call (for instance by the line button on my phone).
Both calls are now in the UP state, who am I talking to?

This, and many other questions, are currently making me even more thin
haired than normal :-)


Michael


2006/8/25, C F <shmaltz at gmail.com>:
>
> So how about inventing a car? The auto industry is much more profitable.
>
> The point; there is no point in reinventing the wheel, why are you
> writing this from scratch?
>
> On 8/24/06, Mir <michael.sysdba at gmail.com> wrote:
> >
> > What do you mean?
> >
> > I'm not looking for someone elses work, I'm developing an application
> from
> > scratch.
> >
> > Michael
> >
> >
> > 2006/8/24, Andrew Kirch <AKirch at allthingsit.com>:
> > >
> >
> >
> >
> >
> >
> > Umm… Flash operator panel?
> >
> >
> >
> > Andrew
> >
> >
> >
> >  ________________________________
> >
> >
> > From: asterisk-dev-bounces at lists.digium.com
> > [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of
> > Mir
> > Sent: Thursday, August 24, 2006 2:18 PM
> > To: asterisk-users at lists.digium.com; asterisk-dev at lists.digium.com
> >  Subject: [asterisk-dev] Phone status
> >
> >
> >
> >
> >
> > Hi
> >
> >
> >
> >
> >
> > I'm working on a project, where I need the status of every telephone on
> the
> > system. (Idle,ringing,busy)
> >
> >
> > If a phone is busy, I also need to know the callerid of the other end.
> >
> >
> >
> >
> >
> > I have made a deamon, which query Asterisk every second for active
> calls,
> > this works by issuing a "Status" to the manager-interface, and
> processing
> > the return data and then put the result into a MySQL table.
> >
> >
> >
> >
> >
> > The clients will query the MySQL table every second for the state of
> their
> > phone, if there are no records with their numbers in it, they are
> considered
> > idle.
> >
> >
> >
> >
> >
> > This works fine for calls from one SIP-phone to the other, this is for
> > instance what it look like when extension 310 is connected to extension
> 311:
> >
> >
> >
> >
> >
> > Event: Status
> > Privilege: Call
> > Channel: SIP/310-08697fb8
> > CallerID: 310
> > CallerIDName: <unknown>
> > Account:
> > State: Up
> > Link: SIP/311-0868fd98
> > Uniqueid: 1156442804.74
> >
> >
> > Event: Status
> > Privilege: Call
> > Channel: SIP/311-0868fd98
> > CallerID: 311
> > CallerIDName: Snom
> > Account:
> > State: Up
> > Context: macro-vm
> > Extension: s
> > Priority: 5
> > Seconds: 13
> > Link: SIP/310-08697fb8
> > Uniqueid: 1156442804.73
> >
> > That is pretty easy to decode.
> >
> > However when an external call is made to a SIP-phone, the result is
> > different, this is a call from another Asterisk via an IAX trunk:
> >
> > Event: Status
> > Privilege: Call
> > Channel: SIP/311-08695698
> > CallerID: 35254390
> > CallerIDName: <unknown>
> > Account:
> > State: Up
> > Link: IAX2/MR-1
> > Uniqueid: 1156442974.76
> >
> >
> > Event: Status
> > Privilege: Call
> > Channel: IAX2/MR-1
> > CallerID: 35436121
> > CallerIDName: <unknown>
> > Account:
> > State: Up
> > Context: macro-vm
> > Extension: s
> > Priority: 5
> > Seconds: 9
> > Link: SIP/311-08695698
> > Uniqueid: 1156442974.75
> >
> > The actual callerid of the caller is 3536121, 35254390 is the called
> number.
> >
> > How do I get the information, that 35436121 is connected to 311?
> >
> > Am I doing it in a stupid way, I'm aware that the Manager can give me
> > realtime events, but I'm under the impression, that it is not very
> stable in
> > a high traffic environment?
> >
> > Any help or good ideas would be appriceated.
> >
> > Michael
> >
> >
> >
> >
> >
> >
> >
> >
> >
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