|
Look at canreinvite= in the sip.conf. If you ‘remove’ Asterisk from
the stream them you are using Asterisk more like a Proxy and less like a PBX.
If this is the case and you want to support ‘tons’ of users look at
something like SER. Asterisk is not a Sip proxy but rather a PBX and Media
transcodeing gateway From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bijan In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or
may go through Asterisk's media bridge. Currently with my settings, I notice that all rtp’s
are passing through my asterisk. How could I achieve that they go directly from
phone to phone? I assume this way, my machine will have less load and
therefore could handle more calls. regards Bijan Karimi |
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