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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Ok Steve, obviously you’ve outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won’t go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Steve Totaro<br><b>Sent:</b> Tuesday, February 28, 2012 3:59 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>OOOOPSS<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'><a href="http://bit.ly/ywiwzt">http://bit.ly/ywiwzt</a><o:p></o:p></p><div><p class=MsoNormal>On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro <<a href="mailto:stotaro@asteriskhelpdesk.com">stotaro@asteriskhelpdesk.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Google or click this link <a href="http://bit.ly/ywiwzteve" target="_blank">http://bit.ly/ywiwzteve</a> " Steve Totaro IAX" and then stop wasting your time, go with SIP even if you need to create VPN tunnel(s).<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Forget IAX2 and save yourself time you will never get back.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS, prime contractors to ITSPs around the world.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Thanks for IAX2 Digium!<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Thanks,<o:p></o:p></p></div><div><p class=MsoNormal>Steve Totaro<o:p></o:p></p><div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford <<a href="mailto:ttelford.groups@gmail.com" target="_blank">ttelford.groups@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>I've tried turning jitterbuffer off - doesn't make a difference. (And why should it? The Jitterbuffer only applies to incoming calls, doesn't it?)<o:p></o:p></p><div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br><br>On 2012-02-28 21:12:48 +0000, Noah Engelberth said:<o:p></o:p></p></div></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><p class=MsoNormal>I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call quality issue transferring calls incoming DAHDI on one * box to another * box, and turning off the jitterbuffer on the side that "couldn't hear" (in my case, the * box with the DAHDI lines, as the DAHDI callers couldn't hear the remote callers) fixed the call quality issue.<br><br><br>-----Original Message-----<br>From: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Troy Telford<br>Sent: Tuesday, February 28, 2012 4:08 PM<br>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great<br><br>On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity.<br><br>Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem:<br><br>With IAX2:<br>- Incoming Voice from my Provider -> Asterisk = Sounds great<br>- Outgoing Voice from Asterisk -> my Provider = Sounds terrible<br><br>By "terrible," I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw).<br><br>On the other hand:<br>With SIP:<br>- Incoming Voice from my Provider -> Asterisk = Sounds great<br>- Outgoing Voice from Asterisk -> my Provider = Sounds great<br><br>The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system).<br><br>The server for my provider is identical in either case. So I figure it's one of a few things:<br>- misconfiguration<br>- My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP<br> - If there's something I can do here, I'd like to know, but I doubt it.<br>- a problem with my provider<br> - In which I'll contact them.<br><br>For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward:<br>[general]<br>bandwidth=low<br>jitterbuffer=yes<br>forcejitterbuffer=no<br>encryption = yes<br>autokill=yes<br>maxcallnumbers=12<br>maxcallnumbers_nonvalidated=4<br><br>[guest]<br>type=user<br>context=default<br>callerid="Guest IAX User"<br><br>[myprovider]<br>type=friend<o:p></o:p></p></div></div><p class=MsoNormal>usernamesecretcontext=somecontext<o:p></o:p></p><div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br>host=provider_server<br>qualify=1000<br>disallow=all<br>allow=g729<br>allow=ulaw<br>auth=md5,rsa<br>requirecalltoken=yes<br>trunk=yes<br><br>Firewall:<br>Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to "just work" - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing).<br><br>I have configured my firewall to forward incoming connections on port<br>4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic).<br><br>Outgoing connections are fairly typical for a NAT setup - anything can go out.<br><br>Any other ideas before I give up on using IAX?<br>Thanks<br>--<br>Troy Telford<br><br><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br><o:p></o:p></p></div></div><p class=MsoNormal>The message does not contain any threats<br><br>AVG for MS Exchange Server (2012.0.1913 - 2114/4837)<o:p></o:p></p></blockquote><div><div><p class=MsoNormal><br><br>-- <br>Troy Telford<br><br><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div></div></div><p class=MsoNormal><o:p> </o:p></p></div></div></div></div><p class=MsoNormal><o:p> </o:p></p></div></body></html>