Hi Members,
I am setting up termination from another Sip provider
through my Asterisk box from 1 customer and find a few snags in the set up.I
have the configuration set up where I can control the billing through
A2billing.For the SIP config that I give to the customer I include the
Username:xxxxxxxxx Password :xxxxxxxxxx Codec G711(ulaw) Protocol:Sip, Sip
gateway: xx.xxx.xxx.xxx Dial Plan:164NXXXXXX Dtmfmode :rfc2833.
The user name and password is generated by A2billing, so now I can setup a
rate card and add funds to the customer's account.This part works fine, but
after 1 minute and 14 seconds the call that the customer places through my
server hangs up and plays a "bye" message. On the other hand when I place a
direct call through this trunk from my server it works fine. What am I doing
wrong for the interconnection ?.
Yours Truly,
Nigel Dennis
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