[asterisk-dev] Transfer app and DTMF via SIP info

[Roystan Mendez]

Hi this is Roystan from India,
I read your mail and would like to help u out...
Can I see your zaptel and Zapata files... so that I can get an idea...
Right now I m facing a problem that I m able to received the pstn call but
when I m trying to transfer the call internally (sip users in the same
office) the call gets hang up.
Thanking you
Roystan

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Michael
Konietzny
Sent: Friday, October 06, 2006 5:59 PM
To: Asterisk Developers Mailing List
Subject: [asterisk-dev] Transfer app and DTMF via SIP info

Hello Asterisk Developers,

I'm currently investigating a problem related to the Transfer app and 
DTMF tones via SipInfo.
My setup depends on:

Elmeg IP 290 (snom190)
Wildcard TE400 (E1)

The following dialplan is given:

exten => 555, 1, Transfer(554);

exten => 554, 1,Dial (SIP/tel3, 10, tT);
exten => 554, 2,Dial (Zap/g1/017123123123, 10, tT);
exten => 554, 3,Hangup();

If I dial 555 on my SIP phone it transfers to 554 and connecting me to 
that zap channel.
Arriving there I'm not able to type ANY DTMF tones.

If the Transfer is skipped the DTMF tones are available. I've included 
the SIP debugs
to help you track the problem.

Greetings and many thanks in advance,

Michael Konietzny