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rstralbergOffline



Joined: Feb 05, 2008
Posts: 1
Location: Microbit 2.0 AB, Kalix, Sweden
Status: Offline
Posted: Feb 05, 2008 - 01:28 PM Reply with quote Back to top
Hi.

This is a C-code problem. I am writing some special code for
handling SIP/MESSAGES and handling of DTMF tone on ZAP.

Asterisk version:
SVN--r872M

Scenario:
    1. SIP/MESSAGE comes in

    2. Message content parsed in channels/chan_sip.c reveive_message(...)

    3. Depending on the message content I want to :
      a) If the Zap-channels has not been called
      before (SIP/INVITE), call a number on a
      Zap channel or group. This has been solved
      by using ast_request_and_dial(...)


      b) Collect DTMF digits from the dialied Zap channel,
      The called ZAP reciever will send some
      DTFM digits when it answers the call.


I have no problems with the above scenario, when the caller (SIP-sender)
makes an INVITE and then sends the MESSAGE. In that case we have a
bridged connection between ZAP and SIP channels until either
of them disconnects.

My way of hooking into the DTMF-tone detection on the ZAP-side is to
check the DTMF tome coming in on channels/chan_sip.c sip_senddigit_end(...)

My problem:
Sometimes the caller (the SIP sender) doesn't start a dialog
(ie doesn't send any INVITE). It just sends a SIP/MESSAGE.
In this case I have to dial the ZAP number from my code.
I'm using ast_request_and_dial(..) and gets a connection ok.
But when the called Zap receiver sends DTMF I never gets them
as my "dtmf-parser-hook" is in channels/chan_sip.c sip_senddigit_end(...), and I don't have any bridged SIP-channel in this case.


Question:
How do I get the DTMF-tones from a called ZAP-channel without have
any bridged SIP-channels?


Many thanks
Roland Strålberg

<snip>No self-promo please, thanks Wink</snip>
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