On 1/18/06, Graham Todd <[EMAIL PROTECTED]> wrote: > - Are there any online resources folks could refer me to re: network/bandwidth > usage in different "typical" asterisk deployments? The ORA Asterisk book has a > chapter on hardware but how best to estimate bandwidth requirements?
Try this: http://www.asteriskguru.com/tools/bandwidth_calculator.php > - Is there a way in the Asterisk configuration to limit connections to a > conference or "reserve" resources to ensure that a conference can happen? I > can think of complicated ways of doing this at the OS level but ... You want to look into the GROUP() and GROUP_COUNT() functions. This will allow you to control the number of connections that you accept to the server (this is in the dialplan, so you can control the number of connections to a particular part of the dialplan). > - Can anyone point me to any howtos for ALTq/pf and firewalling/bandwidth > management with asterisk? (m0n0wall was mentioned on the list a while back). I > guess this is easy to figure out but if people have already done this perhaps > it's documented? This doesn't really seem like an Asterisk issue, but rather a straight up firewall issue. For Asterisk, here are some things to keep in mind: SIP -- signalling port is 5060. RTP -- this is what carries the media for the SIP protocol. By default it is ports 10,000 through 20,000, but is configurable in rtp.conf. Each "call" requires 4 ports. 2 ports for one direction of audio, and 2 ports for the other direction. IAX2 -- port 4569 handles both the signalling and the media. I really like m0n0wall, its pretty straight forward, but you will probably also want to get informed about iptables. > - I'm trying to use AGI to "script" things in Asterisk - but I can't quite > grok how to do simple things like kill off instances of mpg123 after a > conference ends (it gets started for music on hold and never dies) ... e.g. > shouldn't something like this work? How about simply not using mpg123 at all for music on hold? I like using the rawplayer script which is located in the ./contrib/utils/ directory of your Asterisk source. More information is located in the README.rawplayer in the same directory. The other method is to use the native music on hold which uses the format drivers to play the files. All you have to do is convert your music to ulaw, gsm, G.729, ilbc, or whatever codecs you plan on accepting on your PBX. Then Asterisk just streams the frames for you. Hope that helps! -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk
