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I am trying to use a the SIP tapi from 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 |
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