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 .<br><br><div><span class="gmail_quote">On 05/08/07,
<b class="gmail_sendername"><a href="mailto:nigel.dennis@sympatico.ca">nigel.dennis@sympatico.ca</a></b> <<a href="mailto:nigel.dennis@sympatico.ca">nigel.dennis@sympatico.ca</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi Members,<br> I am setting up termination from another Sip provider<br>through my Asterisk box from 1 customer and find a few snags in the set up.I<br>have the configuration set up where I can control the billing through
<br>A2billing.For the SIP config that I give to the customer I include the<br>Username:xxxxxxxxx Password :xxxxxxxxxx Codec G711(ulaw) Protocol:Sip, Sip<br>gateway: xx.xxx.xxx.xxx Dial Plan:164NXXXXXX Dtmfmode :rfc2833.<br>
<br>The user name and password is generated by A2billing, so now I can setup a<br>rate card and add funds to the customer's account.This part works fine, but<br>after 1 minute and 14 seconds the call that the customer places through my
<br>server hangs up and plays a "bye" message. On the other hand when I place a<br>direct call through this trunk from my server it works fine. What am I doing<br>wrong for the interconnection ?.<br><br>Yours Truly,
<br><br>Nigel Dennis<br><br><br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br><br>asterisk-biz mailing list
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