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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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