A2billing limits calltime as per available balance . This is definitely you a2billing configuration problem . Post some cli output it will show parameters of dial command .
On 05/08/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > 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 > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-biz >
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