> asterisk itself still loads the full set of modules...  I 
> find I only need 31 modules loaded for my setups as opposed 
> to the 140+ asterisk will load using a full auto-load..

Yes, I should look at this, though since Astlinux is my first Asterisk
experience, I am underconfident in this. I shall take a tentative look.

> it may also be that the Burst 
> network traffic is causing some of the SIP packets to be lost 
> going to your handsets..

How can one mitigate against this?

> If you are sending out an "all sets page".. you may want to 
> write a script that puts just a bit of a delay in to allow 
> the network to handle the traffic...

I would consider this, but it's probably beyond me. I have no clue where to
start...

> or perhaps go to sip 
> TCP instead of UDP if your version of asterisk and the 
> handsets will support it.. this would give a lot better 
> chance of the packets making it...

I am using whatever the default Astlinux install uses. Does 1.4.x handle SIP
over TCP? Does TCP by definition increase latency?

> The other thing I do is use only ulaw for handsets connected 
> to the LAN..
> and all my voice files are also in ulaw..

I've tried to ensure that this is the case, but I find working with multiple
codecs a bit of a black art. We are alaw over here in the UK, though I guess
I could use ulaw in Asterisk, and let the ISDN hardware handle that
conversion. I use GSM across the IAX trunks between offices, since they only
have max 0.8Mbps, so bandwidth is an issue.

I leave ulaw in there to try to avoid unnecessary transcoding, but I am
rather taking it on trust that this is working. Also, where does Asterisk
native sound format (is it .sln?) come into this consideration?

Thanks for all the suggestions - hopefully I can tighten things up a bit
over the next few months.

Tom


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