I have no interest in the penis-measurement competition firing up here, but I'll say that we have 100% abandoned IAX from all of our systems due to a myriad of issues. These days it offers no real advantages in our opinion.
On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro <stot...@asteriskhelpdesk.com> wrote: > 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. > > $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. > > You have to have thick skin on these lists, the more technical, the more you > better have done your homework or get flamed. > > 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. > > Digium and Digium FanBoys will argue that IAX2 is the best thing since > sliced bread. > > 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. > > 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. > > 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. > > 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. > > To answer your question, maybe you can spend time and get it to work > correctly, I have no idea, but why? > > Why not just use SIP and be done with it. > > Also realize that the dated posts have replies that are ridiculous like > VoicePulse is probably laying people off right now as we speak. > > 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. > > 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. > > My standard now is Vyatta with NTOP, Asterisk, Webmin installed. I only use > SIP and use OpenVPN. > > 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. > > It just works. I like things that just work. > > Thanks, > Steve Totaro > > On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas <da...@debsinc.com> wrote: >> >> 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? >> >> >> >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro >> Sent: Tuesday, February 28, 2012 3:59 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds >> great >> >> >> >> OOOOPSS >> >> >> >> http://bit.ly/ywiwzt >> >> On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro >> <stot...@asteriskhelpdesk.com> wrote: >> >> Google or click this link http://bit.ly/ywiwzteve " Steve Totaro IAX" and >> then stop wasting your time, go with SIP even if you need to create VPN >> tunnel(s). >> >> >> >> Forget IAX2 and save yourself time you will never get back. >> >> >> >> IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS, >> prime contractors to ITSPs around the world. >> >> >> >> Thanks for IAX2 Digium! >> >> >> >> Thanks, >> >> Steve Totaro >> >> >> >> On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford <ttelford.gro...@gmail.com> >> wrote: >> >> 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?) >> >> >> >> On 2012-02-28 21:12:48 +0000, Noah Engelberth said: >> >> 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. >> >> >> -----Original Message----- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy Telford >> Sent: Tuesday, February 28, 2012 4:08 PM >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great >> >> On my Asterisk system, I'm using a provider that provides both IAX2 and >> SIP connectivity. >> >> Personally, I'd prefer to use IAX2, and that's what my account is setup to >> use. However, I'm having a problem: >> >> With IAX2: >> - Incoming Voice from my Provider -> Asterisk = Sounds great >> - Outgoing Voice from Asterisk -> my Provider = Sounds terrible >> >> 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). >> >> On the other hand: >> With SIP: >> - Incoming Voice from my Provider -> Asterisk = Sounds great >> - Outgoing Voice from Asterisk -> my Provider = Sounds great >> >> 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). >> >> The server for my provider is identical in either case. So I figure it's >> one of a few things: >> - misconfiguration >> - My ISP (Comcast) is throttling or giving a low priority to IAX, but not >> SIP >> - If there's something I can do here, I'd like to know, but I doubt >> it. >> - a problem with my provider >> - In which I'll contact them. >> >> For the first case - misconfiguration, I'd appreciate some input. My >> iax.conf is fairly straightforward: >> [general] >> bandwidth=low >> jitterbuffer=yes >> forcejitterbuffer=no >> encryption = yes >> autokill=yes >> maxcallnumbers=12 >> maxcallnumbers_nonvalidated=4 >> >> [guest] >> type=user >> context=default >> callerid="Guest IAX User" >> >> [myprovider] >> type=friend >> >> usernamesecretcontext=somecontext >> >> >> host=provider_server >> qualify=1000 >> disallow=all >> allow=g729 >> allow=ulaw >> auth=md5,rsa >> requirecalltoken=yes >> trunk=yes >> >> Firewall: >> 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). >> >> I have configured my firewall to forward incoming connections on port >> 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). >> >> Outgoing connections are fairly typical for a NAT setup - anything can go >> out. >> >> Any other ideas before I give up on using IAX? >> Thanks >> -- >> Troy Telford >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> >> The message does not contain any threats >> >> AVG for MS Exchange Server (2012.0.1913 - 2114/4837) >> >> >> >> -- >> Troy Telford >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 > > > > -- > _____________________________________________________________________ > -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _____________________________________________________________________ -- 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