similar to: VUC Friday: Twilio OpenVBX

Hi, We're pleased have a 25-year telephony veteran with us tomorrow, Aswath Rao. Aswath maintains that "Trapezoidal VoIP is Evil". Join us and ask questions, make comments, argue about geeky details... and maybe win a Gigaset S675IP SIP/DECT g722-capable phone with an additional handset. Those of us who have these phones like them a lot. All dial in info is here: http://VUC.me -

Hi all, You're welcome as always to join the talk on the VoIP Users Conference, VUC for short. VUC began as the "Asterisk Users Conference" but for obvious reasons, we changed the name in the first year, although Digium was our sponsor for three years. We still have plenty of you who are asterisk users and developers on this our fifth year. Come on by, listen, talk ro text on IRC.

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

Hi, Got some great news a few days ago from Sandro Gauci (@SandroGauci) and we'll be talking about this with him this Friday at 1PM. SIPVicious, the free security tools for SIP scanning, now include a new tool: svcrash. It is aimed at helping system administrators stop bandwidth consuming scans making use of svwar and svcrack. Here is the announcement on SIPViscious blog:

Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING, TWILIO)). It does not work and NO error message in CLI. I have also tried http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I first emailed this group, but that does not seem to work either. Here is my log: [Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call from

Hello, I am running Asterisk 13.6.0 in an AWS instance, and I set it up with Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the calls actually "reach" the PBX, but for some reason, they are not caught by any of my extensions context. Here's what I observe when I test this from a non-PBX connected E164 number (a landline), say 555-666-1212. My Twilio number is

On this weeks VUC call we will collectively be our own guests. That is, we'd like to know what was the big issue that impacted YOU in 2010? All opinions welcome. Here are a few things to get you thinking in advance: - Apple's Antenna-gate - Asterisk 1.8 Launches - Amazon EC2 as a DOS platform - Cisco launched UMI video conference device - More HDVoice capable phones - Skype Outage - VoIP

http://www.techcrunch.com/2009/12/30/twilio-raises-3-7-million-for-power ful-telephony-api/ wow really? Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091230/4829ae70/attachment.htm

That sounds like the ideal app for me too. Fring requires we register with Fring and give them user id/password pair. In our case it did not work until we put a public IP on our Asterisk. I just bought WeePhone and I'll give it a try on the iPhone. Cheers, Enrique -----Mensaje original----- De: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] En

To celebrate three years of the VoIP Users Conference, we're doing a 24-hour VoIP conference call today. Details are at http://voipathon.org IRC: #vuc on Freenode.net SIP: voipathon at vuc.onsip.com - Enter 22622# and your PIN# if you have no PIN you can listen using 1# iNum - +883 51007 039 9924 PSTN: +1 724 444 7444 again, 22622#1# or PIN# if you have one. Those of you in the

*Disclosure: I am a Developer Evangelist at Twilio.* Hey everyone, I just published a blog post on how to use Webhooks and Concerns to integrate Twilio into a Rails 4 application. I hope some of you might find this useful: https://www.twilio.com/blog/2014/02/twilio-on-rails-integrating-twilio-with-your-rails-4-app.html I've also published the full source for this tutorial on Github:

Hi all, What is Bufferbloat? http://gettys.wordpress.com/bufferbloat-faq/ Maybe this kind of discussion will bring out the John Todds of this world, I can only hope and dream: Bufferbloat: http://www.voipusersconference.org/2011/bufferbloat/ Call in and talk to Jim Gettys, who co-developed X Window System and was a part of HTTP/1.1 - this is someone we'll all be proud to meet, and you can

Hi, Few subjects cause as many arguments as "which SIP client works best?" on IRC #asterisk, voip forums, and probably the -users mailing list. I have tried most of the SIP clients available in the last 5 years, both with Asterisk and other platforms such as OnSIP.com, IConnectHere.com, ZipDX.com and the venerable old FWD (in the days when that almost worked). Speaking of ZipDX, we

Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got

This should be interesting, a double header Friday at 12 Noon EDT, session 2 at 1PM EDT. 1) Pascal Dor?, Media5corp. Pascal will talk about what they've been up to in the year since his last visit. Thanks to the Asterisk mailing list and VoIP community, their Media5fone was able to fix its g722 implementation. I like their product a lot and used it extensively on my old iPod Touch to make and

Today, sessions 320-321 of the VoIP Users Conference will take place at the usual time, 12 Noon Eastern [ http://vuc.me/next for local times ] We'll be talking to Sangoma's Frederic Dickey about NetBorder 4.0. You can download or watch his accompanying slide presentation here: http://vuc.li/Sangoma-2011 Next, Pascal Dor?, mVoIP chief over at Media5 Corporation will talk about the

Hi, Today at 12 Non EDT, Dan York will be with us to talk about the recent on and off moments of Google Voice SIP URI calling. Like Skype + Asterisk (or any SIP), Google Voice and SIP compose the other shoe waiting to drop. We're following this with interest. So GV turned on SIP URI and then a few days later, turned it off. Why? Did the geeks (like us) jump on this too quickly or too heavily?

Hi, We had a last-minute cancellation from Vivox for today's conference. It happens that someone suggested a guest idea, Howler Technologies CTO Jay Fenton, who agreed to join the call from the road. Anything you want to know about transcoding to and from g729 is out topic for the first hour. My pal David Duffet knows this technology well and has kindly signed in to help guide us through this

Hi all, Friday at 12 Noon EDT, we'll be talking to Emil Ivov of Jitsi.org (formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz) about Jabber, something the Asterisk community is becoming more interested in by the day. Join us to learn more about Jabber and SIP or to share your knowledge and experience. As always, the VUC discussion includes people from very diverse backgrounds, so

Hello, This Friday March 4th is the 307th VoIP Users Conference. In a few weeks, we'll be starting our 5th year of this weekly live event that began life as the Asterisk Users Conference. We'd love to have you join on on our call with this week's guest OnSIP.com by connecting via SIP:200901 at login.zipdx.com Skype:vuc.me PSTN: +1 567 252 2286 iNum: +883 5100 123 94882 Tex backchat