Hello,
I am trying to do test call below diagram:
SIP phone/Soft
phone------SIP----->Asterisk---H323---->GnuGK----H323----->Verizon.
I made several test calls, but always fail. Verizon technical team tells me
below. Below parameter affect call completion?
Any suggestion?
Please change the service info transfer capability to Speech" or "3.1 kHz
Audio" rather than "Unrestricted Digital Information" in User service info
octet 3.
On the service info transfer rate please change this to Multirate ( 64kbit/s
base rate) and if possible please change the transfer mode to circuit mode in
User service info octet 4.
If you can make the above changes and make a few more tests we can go from
there. I think we may need to look at the forward and reverse logic channel
timers but lets try the above changes first.
Balgaa
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