Im not sure about it, cause I never had chance to play with PSTN
numbers properly. There's something about DID's on page 58 of the
document.
On 3/3/06, roger leszczynski <rogerles(a)gmail.com> wrote:
i found it, thanks, now the question is, say i have
100 concurrent calls,
and i want to update the configuration with a new DID, how can i avoid
restarting SER to make these changes take places....has anyone made a
similar module to reload in asterisk? Or how can i go about using a
database to implement routing instead?
On 3/2/06, Andrey Kouprianov <andrey.kouprianov(a)gmail.com > wrote:
would "getting started" document
from
www.onsip.org help? they have a
good explanation on how to configure SER with RTPProxy...
On 3/2/06, roger leszczynski <rogerles(a)gmail.com> wrote:
> Anyone use the rtp proxy module yet? I am looking to proxy rtp of my
> carrier who sends out calls to me, so if anyone has example configurations
> that work i would appreciate it!
>
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