Hi guys,
I’ve a big problem here and nobody seems to be able to fix it.
A is a PSTN Caller
B is an asterisk extension on a IP Phone
C is an asterisk extension on a IP Phone
A call B and B blind/attended transfer to C.
Call is transferred but the call between A and C has a lot of audio holes and echo.
What the problem could be? Some info:
- Asterisk is running on a very powerful machine (no memory or CPU problem for sure)
- We have made a loopback test on each PRI port succesfully passed
- We use asterisk 1.2 branch release 44110M (please don’t tell me to upgrade: is not there the issue)
- libpri and zaptel are up and running
- *2 and #1 are understood by Asterisk and the transfer process works
We have spent on buying a 4 CPU server with 2 GB RAM, we have spent on buying a Digiunm 410p card: it would be very disappointing to drop the project for asterisk behaviour.
Please Help 