Interesting web tool, but the results are completely misleading and wrong on my system that I just tested it on 5-6 times. And this is a connection that I use regularly for VoIP traffic, multiple channels (although just a few) and I am very familiar with the characteristics of the internet paths to various providers and other trunks I am connected to. So ... from at least one datapoint, doesn't seem reliable.<br><br>As far as the QoS issue - there are some reasonable links on the wiki that describe some techniques you can use with the Cisco box and you will also want to investigate if there are issues between you and your provider. Usually though - the issues are at the edge.<br><br>Oh - and if Asterisk isn't running as root, the ToS settings won't happen for the outbound packets. (In case you are using that for your ACLs on Cisco for QoS rules). There is a patch that I have never tried. I find it simpler to just use iptables and create a few simple rules to set the ToS for
outbound packets based on SIP/IAX and RTP ports per your configuration.<br><br>philippe<br><blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"><jdutt@softelinc.com><mailto:afocardi@gmail.com><mailto:asterisk-users@lists.digium.com><mailto:aj@thepaxson5.org><aj@thepaxson5.org><mailto:aj@thepaxson5.org><mailto:afocardi@gmail.com><mailto:asterisk-users@lists.digium.com><mailto:aj@thepaxson5.org><aj@thepaxson5.org><mailto:aj@thepaxson5.org><http:><http:><br>From: "James Hawks" <james.hawks@customerfunding.com><br>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"<br> <asterisk-users@lists.digium.com><br>Date: Thu, 29 Jun 2006 13:39:28 -0700<br>Subject: RE: [Asterisk-Users] need help troubleshooting clipping and<br> garbledVOIP calls<br><br> Sounds like a QoS issue with your DSL provider. If you go to<br>http://www.bandwidth.com/tools/voipTest it might give you some insight.<br><br>James
Hawks<br><br>-----Original Message-----<br>From: asterisk-users-bounces@lists.digium.com<br>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of T. Shaw<br>Sent: Thursday, June 29, 2006 1:27 PM<br>To: asterisk-users@lists.digium.com<br>Subject: [Asterisk-Users] need help troubleshooting clipping and garbledVOIP<br>calls<br><br>Hello all,<br>I have a problem with call quality with my Asterisk setup. I'm doing VOIP <br>only so far, but have a zaptel TDM400P in the box not being used. The <br>problem i'm having is that when calls are placed, connected, and the far-end<br><br>is reporting that they are experiencing clipping, choppy, and garbled voice <br>conversations. So bad that we have to resort to using our cell phones. This <br>entire setup is still being built, but any phone attached is experiencing <br>this. Call volume is almost nil (under 20 total incoming calls a day). This <br>is a small business setup. The server is used exclusively for Asterisk, so <br>it
isn't a fileserver, or anything else.<br><br>The setup is as such:<br><br>ipphone <--->cisco 2900XL switch <----> Cisco 2621 router <---> dsl modem <br><-->DSL <---> VOIPprovider<br><br>I've configured the switch and the router to set priority and qos to <br>prioritize voice traffic above data.<br>Funny thing is, there is not data REALLY hitting the network. I have setup 2<br><br>vlans, data vlan, and voice vlan. There are two work stations on the <br>network, and neither is being used to hit the internet heavily (office is <br>still being setup).<br><br>Any pointers or suggestions anyone have for me as to were to look for this <br>poor quality?<br>It seems only the Far-end (called party), is hearing this and not the <br>calling party.<br><br>I haven't tried switching out the phones because we only have 1 type, and <br>any of the phones i used exhibit these problems. I will try softphones to <br>see if it is truly a "networking" issue or
Phone issue.<br><br>Is anyone using a cisco 2900 switch or router and care to provide config <br>samples of their COS/QOS setup?<br><br>Thanks!<br><br>Terrelle Shaw<br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br><br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:<br>
http://lists.digium.com/mailman/listinfo/asterisk-users<br></http:></http:></mailto:aj@thepaxson5.org></aj@thepaxson5.org></mailto:aj@thepaxson5.org></mailto:asterisk-users@lists.digium.com></mailto:afocardi@gmail.com></mailto:aj@thepaxson5.org></aj@thepaxson5.org></mailto:aj@thepaxson5.org></mailto:asterisk-users@lists.digium.com></mailto:afocardi@gmail.com></jdutt@softelinc.com></blockquote><br><p> 
<hr size=1>Do you Yahoo!?<br>
Get on board. <a href="http://us.rd.yahoo.com/evt=40791/*http://advision.webevents.yahoo.com/handraisers">You're invited</a> to try the new Yahoo! Mail Beta.