On 17/07/10 2:52 PM, Mark Phillips wrote: > Yup, it's not looking good at all. > > In the pure SIP domain I can route 50 calls concurrently (haven't tried > any more than that) and it was solid for over 2 hours after which I > stopped the test. > > In the T1 domain using a Digium TE110P I can't get any more than 10 > calls from it. After that the T1 takes a dump with all sorts of HDLC errors.
I'd start with a much lower number of calls per second. 5 is pretty high even for a normal machine. You need to start at a level where things work and progress to somewhere that they don't :) In order to confirm you test scenario is correct, you'll need to have it running in a stable configuration at say 0.001 calls per second and a maximum of 2 total calls. Once you're sure this runs stably then increase it. If this doesn't run successfully I'd relook at the test scenario. Is the TE110P card one of the recent voicebus cards? The latest DAHDI drivers allow for changing the interrupt rate on the fly which may help too. -- Cheers, Matt Riddell _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) ------------------------------------------------------------------------------ This SF.net email is sponsored by Sprint What will you do first with EVO, the first 4G phone? Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first _______________________________________________ Astlinux-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [email protected].
