[Asterisk-Users] SIP tapi

[Joash Herbrink]

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/>
.

 

This works fine from all kinds of applications which support TAPI, like
outlook and Dialer Pro.

 

However when making tapi controlled calls, the signaling to and from
PSTN seems to fail.

I have used the digium hardware ISDN PRI boards, but also a SIP gateway.

 

Both result in a audio message from asterisk saying that the number is
unavailable.

 

But, what I need is to have the original PSTN status transferred to the
SIP phone( xten eyebeam in this case) so I can see whether the end point
was just busy, or that the number dialed was just plain wrong.

 

Any help would be very very much appreciated.

 

Joash

 

Maanlander 14a/b                 m: +31 6 53 80 28 20 
3824 MP Amersfoort              e: joash.herbrink at kahuna.nl 
t: +31 33 4500370 ext 1006   URL: www.kahuna.nl
<file:///\\www.kahuna.nl\>  
f: +31 33 4500371 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051128/55a5bb1d/attachment.htm