Re: [asterisk-users] Fw: Stress testing Asterisk
I believe there are options for rtp / audio.. Look at pcap play and rtp echo... Transcoding would be another beast - if you are allowing it Sent from my iPhone 5 On May 22, 2013, at 10:02 AM, Tommy Cooper tomcoope...@yahoo.com wrote: From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP only initiates and eventually terminates the call, once that the call is connected, SIP and therefore Asterisk are no longer involved. Once the call is connected it is assigned to a trapsport layer protocol such as RTP. RTP is the actual protocol that delivers the voice call between endpoints. I believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system. - Forwarded Message - From: Mitul Limbani mi...@enterux.in To: Tommy Cooper tomcoope...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 3:23 PM Subject: Re: [asterisk-users] Stress testing Asterisk I have a question here. How can we test the quality of voice upon increasing the call load? Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario? Mitul On Wednesday, May 22, 2013, Tommy Cooper wrote: Thank you for your help I finally solved this issue. Is it possible that my setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core using 3.5 GHz, and 1Gb of RAM? - Forwarded Message - From: Marie Fischer ma...@vtl.ee To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 1:16 PM Subject: Re: [asterisk-users] Stress testing Asterisk On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote: Hi, I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. Do you have a peer and extension configured for SIPP in your Asterisk configuration? You also needat least the -s extension_to_dial option on your sipp command line. http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has some simple instructions which should get you started. If the calls still fail, Asterisk console output would be helpful. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif
Might also want to check the google hasnt detected an unusual login and is asking for the ip to be accepted. Log in to gmail with that account and check Sent from my iPhone 5 On Feb 5, 2013, at 4:31 PM, Joshua Colp jc...@digium.com wrote: Josue Freitas wrote: Thank you! What about the XMPP traffic? Even when I place calls using GV there's no XMPP traffic on 5222. Do I really need to have the XMPP port (5222) open in the firewall? Asterisk acts as an XMPP client. It establishes an outgoing connection to port 5222 of the Google Talk XMPP server. No incoming connections occur. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
Sometimes just the act of collecting performance data degrades the quality Sent from my iPhone 5 On Jan 6, 2013, at 6:00 AM, XBrian bobo...@yahoo.co.uk wrote: Thanks What would you use to measure jitter / packetloss in real time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon SIP trunking Field Trial
Good luck! Finding the right person at VZ has always been a beef of mine Sent from my iPhone 5 On Jan 5, 2013, at 11:12 AM, Logan Bibby lo...@keobi.com wrote: Does anyone have a good contact for their sales? I've attempted calling their Enterprise sales a few times and was just spun around in circles. Having a sales rep I can just call would be awesome. - Logan On Fri, Jan 4, 2013 at 1:36 PM, Michael L. Young myo...@acsacc.com wrote: - Original Message - From: Matthew J. Roth mr...@imminc.com At least Verizon maintains a consistent customer experience. ; ) Overall, we've found the service to be reliable and stable, but when there are problems or changes needed you're dealing with Verizon and the w...h...e...e...l...s..t...u...r...n..s...l...o...w...l...y. Haha... that is funny... it is sooo true. Well, you are right. Once it is working, it is usually pretty stable. Just a pain in the butt when things are not working. Hopefully we can get through the Field Trial and that is all I have to worry about for a while. Thanks Matthew for all the encouragement as I go down this temporary (I hope) unpleasant path. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Logan Logan Bibby, CEO Keobi Communications Tuscaloosa, Alabama -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
Asterisk sip show peers lists the qualify value in ms (milliseconds). Please read up on this and the setting for it in sip.conf config file Sent from my iPhone 5 On Jan 5, 2013, at 5:30 AM, XBrian bobo...@yahoo.co.uk wrote: Joachim, thanks for the reply - delay you can somewhat estimate prior to the call (with qualify for example) Pls be explicit. How do I use qualify to measure delay - The jitter / packetloss you can only figure out when the call is already up for a while. what would you use to measure jitter / packetloss in real time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
