[Asterisk-Users] timing and dropped calls

Sathya Weerasooriya sathyaw at sbcglobal.net
Mon Nov 8 11:11:08 MST 2004


Hi,

I have a * server which does only SIP to H323 completely in IP domain, there
is no digium h/w in it. In your experience, for this type of application, is
it required to have a timing source to prevent the calls being dropped.

Cheers

SW

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