[asterisk-biz] Calls Hang up after 1 min 14 sec : SIP INTERCONNECT

Jaswinder Singh vicky.r at gmail.com
Sun Aug 5 08:50:27 CDT 2007


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, nigel.dennis at sympatico.ca <nigel.dennis at sympatico.ca> 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
>
>
>
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