[asterisk-users] Everyone is busy/congested: IP Trunk

[Vivek Shrivastava]

yeah i found openvpn helpful in NAT cases.

-Vivek


On 11/6/07, Baji Panchumarti <baji.panchumarti at gmail.com> wrote:
>
> after a copious loss of follicles :-), I finally got outbound working.
>
> Basically the channel statement in the call file needs to have the
> number to be called. For eg., in  test.call  format the statement
> as follows :
>
>    Channel: SIP/3012345678@<your-sip-provider>
>
> And there is no need for a DIAL statement in extensions.conf
> unless you need to dial an additional number / extension.
>
> Then in sip.conf you need a para that matches <your-sip-provider>
> with the relevant auth info.
>
> These two wiki pages, they were very helpful in figuring out a
> solution to the problem :
>
> http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out
>
>
> http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message
>
> hth,
>
> -baji.
>
> --
>
> On Oct 30, 2007 8:43 AM, Gabriel Natale  wrote:
>
> > I have the same problem.
> >
> > I trying with more 4 SIP providers, the account is registering, receive
> > inboud calls, but can`t make outbound calls for "congestion".
> >
> > Can be the out call id the problem?
> >
> > Thanks
> > Gabriel
>
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