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@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@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@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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