Can I force reinvite on SIP calls which I forward?


My test system is an old Ultra-5 with only a 270MHz UltraSparcII CPU and it doesn’t seem to be handling Asterisk very well.

The system seems to cope with one or two calls outbound from my LAN, and the IVR can handle incoming calls via SIP, but when I try to forward an incoming WAN SIP call to an outbound WAN IAX call I get dropped calls. (The messages file shows SIP unreachable without 3000-4000ms pings when the call is dropped, and then drops back down to about 1400ms when it reconnects)

I’ve read about SIP re-invites and I’m wondering if I connect to both inbound and outbound ITSP using SIP can I offload the bandwidth to the ITSP’s after I’ve connected the channels?

Or does a re-invite require the inbound ITSP to know my username password at the outbound ITSP, making this impossible?

Can it be done another way? or should I go out and buy another PC?


Just as a note, the most CPU intensive part of asterisk is Transcoding. So if your SIP & IAX connections are using different codecs (say from ULAW to GSM), that will really chew up your CPU Usage. So I’d check to make sure that you aren’t transcoding. If you are, you might be able to squeeze another call or two out of that old box.

Just a though, as for SIP re-invites, I got nada.

Thanks. However, as an * newbie I’m not sure which log file tells me what codecs were used. I’ve tried /var/opt/asterisk/log/messages and the cdr logs too. Or do I have to trace a call?

Anyone know if its possible to use reinvites to get 2 different ITSPs to connect their media using my accounts but not using my bandwidth?