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:
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

</snip>