<div>People around here either hate me or love me. I post experience and am accused of bragging or whatever. As a reader, I want to know who is giving me advice and what it is based on.</div><div><br></div><div>$40k/wk of long distance through VoicePulse. I have the invoices, that is high usage, others attack me for posting information like this, I think I know why but I don't care.</div>
<div><br></div><div>You have to have thick skin on these lists, the more technical, the more you better have done your homework or get flamed. </div><div><br></div>It is from years of experience, not outsmarting anyone. It took me months to figure out that it just doesn't work well and as you can see, all of the posts are very dated. Nobody outsmarted anyone, just pure experience and experience of MANY other people that use Asterisk. Many did not wish to make waves and emailed me directly that they either came to the same conclusion or that they switched due to my suggesting and had no more problems.<div>
<br></div><div>Digium and Digium FanBoys will argue that IAX2 is the best thing since sliced bread. <br><div><div><div><div><br></div><div>Digium will ALWAYS tow the party line. It was either Flemming or Lesher that actually posted that it was in an official release so it couldn't have bugs. That was the end of listening to Digium about IAX2. That statement was archived with my reply of how ridiculous the statement was. It is all on the mailing list.</div>
<div><br></div><div>The compensation thing is very true, people drink the cool-aide about IAX2 and it sounds great. Then it turns out that they go to production, and audio sucks, customers are complaining. It becomes a huge problem obviously to an ITSP or any call center. </div>
<div><br></div><div>As I said, my experience is dated, but I have been one of the most prolific people in the Asterisk community, I spoke at Astricon in 2007 on Large Call Center Track and was the #1 poster for the year, a year or two ago. I predate most of Digium Staff.</div>
<div><br></div><div>I do this stuff in the real world, over VSAT or whatever connectivity you can think of, my experience is real, not a developer in the world of code.</div><div><br></div><div>To answer your question, maybe you can spend time and get it to work correctly, I have no idea, but why?</div>
<div><br></div><div>Why not just use SIP and be done with it.</div><div><br></div><div>Also realize that the dated posts have replies that are ridiculous like VoicePulse is probably laying people off right now as we speak.</div>
<div><br></div><div>If a challenge drives you and you have tons of time to possibly never figure it out and go to SIP, then by all means, do it.</div><div><br></div><div>If you want it to just work, use OpenVPN to get your single port, don't believe the Digium party line and replies about using OpenSER or whatever it is called now. I get past the firewall and NAT issues with OpenVPN.</div>
<div><br></div><div>My standard now is Vyatta with NTOP, Asterisk, Webmin installed. I only use SIP and use OpenVPN.</div><div><br></div><div>I build Asterisk from source and menuconfig, I remove all that is not needed, including IAX2. I do download all the sound files in different languages and codecs.</div>
<div><br></div><div>It just works. I like things that just work.</div><div><br></div><div>Thanks,</div><div>Steve Totaro</div><div><br><div class="gmail_quote">On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang="EN-US" link="blue" vlink="purple"><div><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?<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></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""> <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>] <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<u></u><u></u></span></p>
<div><div class="h5"><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">OOOOPSS<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt"><a href="http://bit.ly/ywiwzt" target="_blank">http://bit.ly/ywiwzt</a><u></u><u></u></p>
<div><p class="MsoNormal">On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro <<a href="mailto:stotaro@asteriskhelpdesk.com" target="_blank">stotaro@asteriskhelpdesk.com</a>> wrote:<u></u><u></u></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).<u></u><u></u></p>
<div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Forget IAX2 and save yourself time you will never get back.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></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.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Thanks for IAX2 Digium!<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Thanks,<u></u><u></u></p></div><div><p class="MsoNormal">Steve Totaro<u></u><u></u></p><div><div><p class="MsoNormal" style="margin-bottom:12.0pt">
<u></u> <u></u></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:<u></u><u></u></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?)<u></u><u></u></p><div><div><p class="MsoNormal" style="margin-bottom:12.0pt">
<br><br>On 2012-02-28 21:12:48 +0000, Noah Engelberth said:<u></u><u></u></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<u></u><u></u></p></div></div><p class="MsoNormal">usernamesecretcontext=somecontext<u></u><u></u></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>
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<br><u></u><u></u></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)<u></u><u></u></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><u></u><u></u></p></div></div></div><p class="MsoNormal"><u></u> <u></u></p></div>
</div></div></div><p class="MsoNormal"><u></u> <u></u></p></div></div></div></div><br>--<br>
_____________________________________________________________________<br>
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New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
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