[Asterisk-Dev] spike / asterisk hang?

[Adam Goryachev]

On Fri, 2005-07-15 at 10:54 -0600, Matt Hess wrote:
> We have a steadily growing dial plan for our users on a sparc netra t105..
> I have noticed that audio going through the asterisk (stable) system 
> sometimes becomes choppy (cuts out and restores after a few seconds) 
> when new calls are being handled.. on the system running a vmstat 1 I 
> watch the traps sys counter jumps up really high from the normal run stats..
> 
> An example dialplan entry for a user is as simple as:
> exten => 201,1,Dial(SIP/201,,t)
> (we've got a few hundred of these but note that we only have around 15 
> calls active at any given time)
> because of the transfer requirement media goes through the asterisk server..
> we are using sip on both side of the asterisk server ulaw codec is 
> preferred at the endpoints.

Have you tried simplifying your dialplan? ie, using patterns pointing a
group of extensions to a macro instead of defining each entry
individually?

Have you tried realtime, storing the dialplan outside of asterisk,
perhaps on a different machine?

If it is truly the size of the dialplan making the difference, then I'd
suggest you should take a look and see if there is any code that
searches the dialplan for the appropriate path to take which could be
optimised... This would help regardless of the size of the CPU/etc... 

Just my 0.02c worth :)

Regards,
Adam