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

[Rushowr]

IIRC, you'll want to look at 'hint' extensions, and possibly subscriptions
to get status updates


  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mir
Sent: Monday, August 28, 2006 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RE: [asterisk-dev] Phone status


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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