Hi,
My setup is quite complicated.
I have to Asterisk server linked via IAX.
My Sip phones are connected to one and go out (PSTN)
via the IAX trunk and the other server is connected to
a Quintum CMS via H323.
Phone---(SIP)---Asterisk1---(IAX)---Asterisk2--(H323)---CMS-->
PSTN
All work fine but when a call someone who is busy I
didn't hear the corresponding tone and asterisk2 go to
timeout and after asterisk1.
In my log the call is well routed through the two
asterisk server.
I don't know what is the problem so if you have any
idea ?
Thanks in advance !
Anthony
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