And the dude arrives talking about penis...... On Tue, Feb 28, 2012 at 6:07 PM, Carlos Alvarez <car...@televolve.com>wrote:
> 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 >
-- _____________________________________________________________________ -- 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