Iqbal
Tracy R Reed wrote:
On Tue, Dec 07, 2004 at 08:44:50PM -0800, Gonzalo Gasca Meza spake thusly:
I have just setup Asterisk, but the problem is that all RTP stream pass through Asterisk, I mean all call setup and voice stream pass trough Asterisk once the call is established. Is there a way that call setup is established, the RTP stream pass just between the SIP endpoints.
Yes. For this you should use SER:
www.iptel.org/ser
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