And the dude arrives talking about penis......<br><br><div class="gmail_quote">On Tue, Feb 28, 2012 at 6:07 PM, Carlos Alvarez <span dir="ltr"><<a href="mailto:carlos@televolve.com">carlos@televolve.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I have no interest in the penis-measurement competition firing up<br>
here, but I'll say that we have 100% abandoned IAX from all of our<br>
systems due to a myriad of issues. These days it offers no real<br>
advantages in our opinion.<br>
<br>
<br>
On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro<br>
<div class="HOEnZb"><div class="h5"><<a href="mailto:stotaro@asteriskhelpdesk.com">stotaro@asteriskhelpdesk.com</a>> wrote:<br>
> People around here either hate me or love me. I post experience and am<br>
> accused of bragging or whatever. As a reader, I want to know who is giving<br>
> me advice and what it is based on.<br>
><br>
> $40k/wk of long distance through VoicePulse. I have the invoices, that is<br>
> high usage, others attack me for posting information like this, I think I<br>
> know why but I don't care.<br>
><br>
> You have to have thick skin on these lists, the more technical, the more you<br>
> better have done your homework or get flamed.<br>
><br>
> It is from years of experience, not outsmarting anyone. It took me months<br>
> to figure out that it just doesn't work well and as you can see, all of the<br>
> posts are very dated. Nobody outsmarted anyone, just pure experience and<br>
> experience of MANY other people that use Asterisk. Many did not wish to<br>
> make waves and emailed me directly that they either came to the same<br>
> conclusion or that they switched due to my suggesting and had no more<br>
> problems.<br>
><br>
> Digium and Digium FanBoys will argue that IAX2 is the best thing since<br>
> sliced bread.<br>
><br>
> Digium will ALWAYS tow the party line. It was either Flemming or Lesher<br>
> that actually posted that it was in an official release so it couldn't have<br>
> bugs. That was the end of listening to Digium about IAX2. That statement<br>
> was archived with my reply of how ridiculous the statement was. It is all<br>
> on the mailing list.<br>
><br>
> The compensation thing is very true, people drink the cool-aide about IAX2<br>
> and it sounds great. Then it turns out that they go to production, and<br>
> audio sucks, customers are complaining. It becomes a huge problem obviously<br>
> to an ITSP or any call center.<br>
><br>
> As I said, my experience is dated, but I have been one of the most prolific<br>
> people in the Asterisk community, I spoke at Astricon in 2007 on Large Call<br>
> Center Track and was the #1 poster for the year, a year or two ago. I<br>
> predate most of Digium Staff.<br>
><br>
> I do this stuff in the real world, over VSAT or whatever connectivity you<br>
> can think of, my experience is real, not a developer in the world of code.<br>
><br>
> To answer your question, maybe you can spend time and get it to work<br>
> correctly, I have no idea, but why?<br>
><br>
> Why not just use SIP and be done with it.<br>
><br>
> Also realize that the dated posts have replies that are ridiculous like<br>
> VoicePulse is probably laying people off right now as we speak.<br>
><br>
> If a challenge drives you and you have tons of time to possibly never figure<br>
> it out and go to SIP, then by all means, do it.<br>
><br>
> If you want it to just work, use OpenVPN to get your single port, don't<br>
> believe the Digium party line and replies about using OpenSER or whatever it<br>
> is called now. I get past the firewall and NAT issues with OpenVPN.<br>
><br>
> My standard now is Vyatta with NTOP, Asterisk, Webmin installed. I only use<br>
> SIP and use OpenVPN.<br>
><br>
> I build Asterisk from source and menuconfig, I remove all that is not<br>
> needed, including IAX2. I do download all the sound files in different<br>
> languages and codecs.<br>
><br>
> It just works. I like things that just work.<br>
><br>
> Thanks,<br>
> Steve Totaro<br>
><br>
> On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas <<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>> wrote:<br>
>><br>
>> Ok Steve, obviously you’ve outsmarted at least this poster. On the one<br>
>> hand, IAX2 has purchased things for you (won’t go as far as saying it bought<br>
>> your Mercedes), but on the other hand it is being dropped by providers as we<br>
>> speak. So are you saying it can be a good thing if you have the time and<br>
>> skill level to pursue it, but beginners should leave it alone?<br>
>><br>
>><br>
>><br>
>> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Steve Totaro<br>
>> Sent: Tuesday, February 28, 2012 3:59 PM<br>
>> To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
>> Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds<br>
>> great<br>
>><br>
>><br>
>><br>
>> OOOOPSS<br>
>><br>
>><br>
>><br>
>> <a href="http://bit.ly/ywiwzt" target="_blank">http://bit.ly/ywiwzt</a><br>
>><br>
>> On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro<br>
>> <<a href="mailto:stotaro@asteriskhelpdesk.com">stotaro@asteriskhelpdesk.com</a>> wrote:<br>
>><br>
>> Google or click this link <a href="http://bit.ly/ywiwzteve" target="_blank">http://bit.ly/ywiwzteve</a> " Steve Totaro IAX" and<br>
>> then stop wasting your time, go with SIP even if you need to create VPN<br>
>> tunnel(s).<br>
>><br>
>><br>
>><br>
>> Forget IAX2 and save yourself time you will never get back.<br>
>><br>
>><br>
>><br>
>> IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS,<br>
>> prime contractors to ITSPs around the world.<br>
>><br>
>><br>
>><br>
>> Thanks for IAX2 Digium!<br>
>><br>
>><br>
>><br>
>> Thanks,<br>
>><br>
>> Steve Totaro<br>
>><br>
>><br>
>><br>
>> On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford <<a href="mailto:ttelford.groups@gmail.com">ttelford.groups@gmail.com</a>><br>
>> wrote:<br>
>><br>
>> I've tried turning jitterbuffer off - doesn't make a difference. (And why<br>
>> should it? The Jitterbuffer only applies to incoming calls, doesn't it?)<br>
>><br>
>><br>
>><br>
>> On 2012-02-28 21:12:48 +0000, Noah Engelberth said:<br>
>><br>
>> I'd try turning off the jitterbuffer and see if that makes things better.<br>
>> I just traced a similar call quality issue transferring calls incoming<br>
>> DAHDI on one * box to another * box, and turning off the jitterbuffer on the<br>
>> side that "couldn't hear" (in my case, the * box with the DAHDI lines, as<br>
>> the DAHDI callers couldn't hear the remote callers) fixed the call quality<br>
>> issue.<br>
>><br>
>><br>
>> -----Original Message-----<br>
>> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">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">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<br>
>> SIP connectivity.<br>
>><br>
>> Personally, I'd prefer to use IAX2, and that's what my account is setup to<br>
>> 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<br>
>> (sometimes impossible) to understand. It doesn't matter what codec I use (at<br>
>> 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<br>
>> prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only<br>
>> 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<br>
>> one of a few things:<br>
>> - misconfiguration<br>
>> - My ISP (Comcast) is throttling or giving a low priority to IAX, but not<br>
>> SIP<br>
>> - If there's something I can do here, I'd like to know, but I doubt<br>
>> 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<br>
>> 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<br>
>><br>
>> usernamesecretcontext=somecontext<br>
>><br>
>><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<br>
>> noticed that my own connection to my provider has always been sufficient to<br>
>> allow connection tracking to "just work" - and incoming calls are accepted<br>
>> without problems, and voice travels in both directions (albeit not so well<br>
>> 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<br>
>> (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<br>
>> 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>
>> The message does not contain any threats<br>
>><br>
>> AVG for MS Exchange Server (2012.0.1913 - 2114/4837)<br>
>><br>
>><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><br>
>><br>
>><br>
>><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>
><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>
<br>
<br>
--<br>
</div></div><span class="HOEnZb"><font color="#888888">Carlos Alvarez<br>
TelEvolve<br>
<a href="tel:602-889-3003" value="+16028893003">602-889-3003</a><br>
</font></span><div class="HOEnZb"><div class="h5"><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>
</div></div></blockquote></div><br>