This question may have been answered 100 times already but I am new to the list. Sorry.
I have an x100p as my main PSTN (Canada). Everything works fine but the x100p takes a very long time to hangup on calls. It takes up to 30 secs, before the card will receive a new call. What is really frustrating is that a caller who hanges up during a "please leave message" prompt, still generates a message, even if they hang up without leaving a message. Is there a way improving this? Thanks for your help. Martin -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Chan Sent: Sunday, August 01, 2004 3:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk scalability? Hi, Scott Thanks for your information. I have worse luck in load testing with asterisk. I have tried both SIP and H323 inbound calls and terminating on PSTN PRIs. I am using a single Xeon 2.8G chip and 512M Ram and in both cases, once it gets more than a T1, call quality starts to degrade with choppiness, and Asterisk becomes very unstable and resets itself like every 5-15 minutes. Can you let me know more about your tests, like which version of Asterisk are you using for the test, and which version of H323 and your computer configuration please, thanks a million TC -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Saturday, July 31, 2004 2:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk scalability? Hi Roy- I've done a lot of load testing with asterisk and TE410P's. My guess, with no transcoding, is that you might be able to handle 8 E1's max on the PSTN side absolute max (ie: 2 TE410P's). This assumes you have a fast processor. If you're using T1's, scale these numbers up accordingly, as there are fewer channels per span. If this answer is lower than you might expect, consider that every byte of data has to pass through the processor. The 410's are capable of bus-mastering, and so are an improvement over the T400P's, but still I think you run into horsepower limitations. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Saturday, July 31, 2004 8:25 AM To: Asterisk Users Subject: [Asterisk-Users] Asterisk scalability? Hi I plan to setup an asterisk box to function as a SIP gateway forwarding lots of calls to/from a backend of several other asterisk boxes, each with a TE410 card for PSTN connectivity. It will only gateway the calls into the PSTN gateways. No transcoding is planned - only plain ALAW. How many concurrent calls would you think this can handle? I'm asked to plan a system that can handle >1000 concurrent calls... thanks for any input regards roy _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
