Displaying 2 results from an estimated 2 matches for "333ms".
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133ms
2005 Jan 23
6
Autio cut off at beginning of call
I posted this question a while back, and I'm posting again in hopes that
someone has some ideas. Sorry if you've already seen this.
When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom,
VoicePulse Connect) I often find that after the call is answered the first
few seconds of audio are cut off (i.e. I don't hear the called party). This
usually results in the called
2005 May 11
0
Database of actve calls (as per astguiclient)
...across a number of asterisk servers. The main purpose of this is in
determining where to route calls (e.g. don't send calls to a server with
no free lines) and also for monitoring/recirding calls.
I know that astguiclient does this by telnetting into the * server
management interface ever 333ms and updating a MYSQL database.
Does this place much load on the * server, or the DB server?
Will this sort of model scale to 30 servers, each with 120 Zap channels?
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