[bbbeavis]

OK, PNphpBB2 (powers these forums) is irritating me. It keeps losing my edits of my posts. Anyway, I just realized you manually edited sip_addtional.conf. I've had corruption problems before doing that as it is supposed to be wriiten to by freePBX only. Delete the trunk and reinstall it only this time ditch g729.

[rizsher]

Zzoom,

I've only been with Asterisk a week now, so don't really know the implications of this, but on the 1.7 GHz Centrino laptop with 256 MB of RAM I have TrixBox installed, memory usage used to be solid at 96%, calls used to be find most of the time, i.e if they'd started OK, they'd Stay OK, however, randomely, and specially incoming calls, would be totally crap.

I upgraded the laptop ram to 512, usage is down to aroudn 43%, and over the last 2 days, I haven't encountered the bad sound quality yet.

It appears your motherboard has a built in VideoCard which seems to share your System Ram. Seems like a 256 MB system, check with other's here, but adding more memory may fix your issue.

[bbbeavis]

Asterisk will run on a 200mhz Linksys WRT54G router with 16 meg of ram and 4 meg of flash as long as your not transcoding. I run it solidly on an old AMD Duron 700 with 256 meg of ram with a whole bunch of transoding going on. If you use Munin which was added to Trixbox you will see the allocation for memory contains a lot of cache. A couple of simultaneous conversations is no issue for this machine. I could run a small office off of it though I would probably not go any further with it unless I took steps to eliminate transcoding (codec-to-codec and protocol-to-protocol). I'm also using IVR and Music On Hold. Currently, I have taken no steps to lighten the processor load or memory demands (if anything, mine is a case study of just the opposite). If you run everything (clients and trunks) on SIP and you use one codec that does not highly compress you are putting very very low stress on the machine with multiple simultaneous conversations. Unless you have a memory leak from a corrupted install. Once again, check Munin. Also test your bandwidth.

[mberlant]

Echoing bbbeavis, I run my production Asterisk (64 Registered peers, plus ad hoc relationships like ENUM and IPKall) on a Pentium 200 MMX with 96MB RAM. It has no problem serving a dozen simultaneous conversations (my high water mark), IVR, MoH, transcoding (one client has only enough bandwidth for GSM and his service providers require G.711), voice mail and MeetMe conferencing.

I also have Asterisk running on Linksys WRT routers and can confirm that it can handle quite a lot of SIP traffic if you don't ask it to actually process any voice (transcoding, voice mail, IVR, etc.).

Now, zz000mm, back at the top of this thread a couple of us advised you to disable G.729 in your Asterisk, yet there it still is in your latest sip.conf excerpt. What's the story?