[Asterisk-Users] Zap to SIP transfer problem

[Eric Wieling]

Maybe you are using inband DTMF with a compressed codec.  DTMF on a call
with any codec other than ulaw or alaw MUST use OOB DTMF like RFC2833 or
INFO.

On Fri, 2004-03-05 at 20:39, Barton Hodges wrote:
> I'm having a problem with transferring a call that comes in a Zap
> channel and is connected with a SIP channel (on a GS HT-286).
> 
> The call is answered automatically, then the user enters an extension.
> Dial() is called with both T and t flags.  When the bridge is made
> between the channels, the caller on the Zap channel can hit '#' to
> transfer, but the caller on the SIP channel cannot.  No messages
> whatsoever are displayed on the console when the SIP user hits any
> keys.  What am I missing?
> 
> 
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-- 
For Asterisk PBX related documentation go to
http://www.digium.com/index.php?menu=documentation and look at the
"Unofficial Links" section also see
http://www.voip-info.org/wiki-Asterisk also see my site at
http://www.fnords.org/~eric/asterisk/

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