similar to: IAX: No outgoing audio for 10 seconds

Hi, We all know what the principal advantage of IAX is, doing it all on a single port, right? But now and again I hear complaints about it. What specific griefs have you had with IAX and has it stopped you from using it entirely? Under what conditions have you had problems? I have used SIP and IAX for about three years now. We don't do a lot of traffic, but I haven't really seen a

Hi, I'd like to invite all asterisk users to answer two questions on this form: http://food4wine.ning.com/poll 1) What version do you use in production (1.2, 1.4 or both) 2) and what distro(s) It'll just take a second and the results are public and live (link on the page above)

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TODAY, Friday January 4th at 12 Noon EST, 11 AM Central, 9AM Pacific, Mountain figure it out, 17:00 UTC Mark joins us to talk about IAX, the appliance, what's new in the asterisk worldwide communities and answer any questions you may have. Why not take this opportunity to ask questions or make comments? This conference is the largest *live* online meeting of asterisk users in the world. Each

Hi, Quick reminder that the conference is happening today at 12:30 PM EDT. I'd like to talk more about updating to 1.4. I now have a test box running asterisk 1.4.5, CentOS 5 and Lumenvox speech rec software. Seems to be fine except for some double NAT issues that could be router specific. Byran Johns from Shelton-Johns is our guest to share some of his extensive experience. More about him

FRIDAY September 7th at 12:30 PM EDT http://www.asteriskusersconference.org for more information on how to listen, talk, or both :) This week, ENUM is the main subject, although our friends at e164.org haven't been able to talk to us as planned. Come on by and share what you know about ENUM or ask questions. Also, during Astricon, we are hoping people will call us with reports, either live

Hi, Tomorrow, Friday Nov 16th, 2007 at 12:30 PM, we'll be exploring a simple, well-commented example of an AGI script for asterisk. I have absolutely nothing against GUI, but if you want to unleash the real power of asterisk, you'll need to get into AGI (or pay someone else to do it). Because asterisk solutions are mostly limited to your imagination (and a hired hand may or may not have

Hi, Quoting Digium's blog post (blogs.digium.com): "The market talked, we listened, and today we?re ready to take calls, answer questions and do whatever needs to be done to make Asterisk work for our customers. It is now safe to step into the Open." John Todd and Steve Sokol will be online live at 12 Noon EDT (9AM Pacific, 10 Mountain, 11 Central, 5PM UK, and 6PM in Western

Doing a little research before Friday's Voip Users Conference call with Dan Behringer. Are any of the newer Polycom wideband codecs implemented in v1.6? Specifically, G.722.1 or G.722.2? Thanks, Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at mstvp.onsip.com skype mjgraves

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On the Python Tutor mailing list Kent Johnson uses a script to find the top posters for the year. If this or something like it has been posted, sorry for the noise; 2008 ==== Steve Totaro 796 Tzafrir Cohen 749 Tilghman Lesher 496 Alex Balashov 354 Olivier 334 Philipp Kempgen 251 Gordon Henderson 242 Atis Lezdins 239 Jay R. Ashworth 230 Doug Lytle 207

I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set "notransfer=no" in the iax.conf file at the office so that the office system can

Hello For those SOHO customers (ie. at most, a couple of POTS/ISDN connections and simultaneous SIP calls) who'd rather not use a big, noisy PC to run Asterisk, I'd like to offer an alternative that has the following features: - not old hardware sold on eBay, ie. it must be up-to-date hardware sold by a company currently in business - compact, silent - has room for a 2.5" hard-disk,

I have found out that the reason why my call transfers are not working when using the IAX protocol is because Asterisk is performing a native bridge. If I force the user of one of the clients to use a different codec so that Asterisk is unable to do a native transfer then it works. How can I disable native bridge for IAX calls? I know for SIP you can put 'canreinvite=no' but this does

Hi, I am going to be on the road for the next few days and with the variable delay on the mailing list, I am posting this now, 4 days before the conference. If you haven't yet listened or participated, please consider doing it. We have a great kernel of people at all levels of expertise and ideas and questions can be kicked around immediately (well, there's a few milliseconds lag). This

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

Hi all, I have spent some time searching, but I haven't found a way to prevent * from concatenating two frames into one IAX packet. I have a situation where I make an IAX GSM call to *, which transcodes to an iLBC SIP call. Every second voice packet the IAX client receives contains 2x 20ms frames, the other containing only one. I presume this is related to the mismatch of 20ms GSM vs

Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively.

Happy New Year in advance by a few ticks for the northern hemisphere. Here's the first topic and guest for 2009: In any voice path there are several potential sources of quality problems, ranging from echo to voice dropouts and everything in between. With VoIP systems the potential for quality problems increases dramatically, often times making it very difficult to identify the source of

Hello There's a lot of information on VoIP at www.voip-info.org ... but there's also a lot of outdated information there as well :-/ Since SIP is a pain to use when NAT is involved, especially when both the Asterisk server and the remote phones are behind NAT... I'd like to try IAX to see how it works and if it solves the issue. I'd like to start with a softphone (Windows