Re: Placing a call through a gatway: Cisco Analog 2811

Thanks for the response Elliot. Do you have any recommendations for Call
Progress Detection code that we may be able to use in our product?

Regards,
Paul

-----Original Message-----
From: Discussion list for Java Telephony API
[mailto:[email protected]] On Behalf Of Elliot Otchet
Sent: Saturday, October 15, 2005 10:53 AM
To: [email protected]
Subject: Re: [JTAPI] Placing a call through a gatway: Cisco Analog 2811

Paul,

You are in need of something some call call progress detection which is
common on digital pri type lines, but not on analog ones.  It usually
takes DSP rsources to listen and analyze the call proogress indicators
(e.g. Ringing, busy, answered by live person, answered by
voicemail/answering machine).  I hit this problem on a Cisco vg 200
years ago and had to write my own call progress algorithms.  While it
takes some time to do, you actually can mitigate this.  The nice part is
for call progress, you're almost always comparing audio to known
patterns.

Sorry there is not a better answer for that analog gear.  If you can
change the hardware out, there are alternatives.

Regards,

Elliot

-----Original Message-----
    From: "Paul Yago" <[email protected]>
    Sent: 10/14/05 8:16:08 PM
    To: "[email protected]" <[email protected]>
    Subject: [JTAPI] Placing a call through a gatway: Cisco Analog 2811

    We have a Jtapi application which is placing calls out through a
2811
    Analog gateway, through a company DID, to a phone on the PSTN. When
the
    call is established then our application begins sending RTP data.

    When we make a call, the CallCtlConnEstablishedEv is sent to the
    callChangedEvent() method, and then is processed by our application,
    indicating that it's time to send the RTP.

    The CallCtlConnEstablishedEv should be sent when the caller picks up
the
    phone; and this is how it works from within our Cisco Network.
However
    whenever we call out using the Analog 2811, the
CallCtlConnEstablishedEv
    is sent as soon as the call reaches the gateway. This is a problem
    because it's the only indication that we need to start sending RTP.
    Therefore we start sending and the beginning of our audio is always
    truncated whenever the callee picks up from a PSTN phone. (This is
not a
    problem when we go to the PSTN through a 2611 T1 gateway).

    Unless I know better, there is no event generated whenever the
far-end
    phone picks up.  Is there some event that I'm not looking for,
whether
    it be a terminal, address, or provider event?

    Or maybe, there's a configuration change I could make with my 2811
    although I'm not a gateway expert. It's using MGCP to interface with
the
    Call Manager. I've got to believe that the Gateway is smart enough
to
    know when the far end has picked up, why can't it relay this
information
    to the CallManager?

    Any help is much appreciated.

    Thanks,
    Paul

 
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