similar to: FW: Alec Saunders post about Mashable Telco's

Interesting read in Alec Saunders blog today. http://saunderslog.com/2007/03/01/mashable-telcos/ Thought it may interest some people on this list. As food for thought, why it is that ITSP's haven't come up with more 'interesting' voice applications? When asterisk first became available I thought it was the beginning of seeing really neat applications, think Verzion's

Interesting to read this email; http://mashable.com/2009/09/10/google-voice-gmail/ I remember when I started the bounty to get this function built into asterisk..... no one ever delivered ion the solution so it was disbanded.....Come on guys we are being left behind. Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New

Greetings Everyone! The version 1.0 of the PhoneCALL Administrative Manual has been released. It is more of an outline of the features and interface, and we'll be adding lots of more detailed information in the manual over the next few days/weeks. Of course, we'd love to get your input on the manual and areas we need to clarify or even some new sections in the manual that would help

Has anyone every heard of Phonecall? : www.vecsector.com/phonecall/ Feedback? Is there something as good as it or better ? Recommendations? -- ================================= Joshua Abbott, Support Technician http://www.successfulhosting.com/ Direct Line: PENDING Phone: (866) 494-5096 x1207 E-Fax: (419) 858-3241 Alt E-Fax: (801) 217-1123 jabbott@SuccessfulHosting.com

Hello all! It's come time where I need to add another programmer to our team. You should have at least 3 years of "work" experience with PHP/MySQL. Please send me your resume and a few code samples if you can. If you can only work part-time or full-time, please include that in your response. Along with your salary requirements. You'll be working with PhoneCALL, so be sure to

Hello All! Just a notice that our PHP/Smarty-based GPL version of PhoneCALL version 2.6.1 has been released, and is the current stable release. http://www.vecsector.com/phonecall We're always looking for feedback/testers to help us enhance it and make it even easier for everyone to use. The current version is designed around the advanced Asterisk user, and we are working on a more

hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another

Just saw this article http://news.yahoo.com/s/mashable/20101103/tc_mashable/google_helps_autotune_your_web_server (/rant - stupid seo links, lame /rant off) it mentions it is set up for linux distros, notably centos. Anyone use this yet? Affect the auto yum update at all?

Hi all, I need to use asterisk to record all phonecall I have test using mixmonitor to record a call. Now I need to set the configure file to let asterisk auto record all calls. I have searched many document but still can not succeed. My version is 1.8beta and I prefer using mixmonitor. Regards!

Hi, I tried to use Monitor(wav,filename) function in dialplan to record Meetme conference. When I monitored on IAX2 or SIP channels in that conference It recorded all audio (in and out) but when I monitored on ZAP channels I could hear only IN audio and piece of OUT audio (announcement get pin and than nothing). Anyone knows why this so happens??? I have asterisk 1.0.7 (debian package)

Hey there, Just wanted to drop a line and let people know that I'll be heading to San Francisco for O'Reilly's Etel. If you are interested in attending, there are some free passes floating around. If anyone is interested in getting together for a beer, let me know! Info on the conference can be found here: http://conferences.oreillynet.com/etel2006/ I'm looking forward to an

am getting the following.... [root at linux11 ~]# glxinfo | grep direct Error: unable to open display (null) [root at linux11 ~]# uname -a SMP Thu May 7 11:51:15 EDT 2009 i686 i686 i386 GNU/Linux [root at linux11 ~]# cat /etc/redhat-release CentOS release 5.3 (Final) on the otherhand when i run wine application i get the following too err:wgl:X11DRV_DescribePixelFormat No libGL on this box -

JonY wrote: > On 2/5/2012 13:30, Erik de Castro Lopo wrote: > > Dave Yeo wrote: > > > >> --- > >> configure.ac | 1 + > > > > Both applied (first with a minor tweak). Thanks. > > > > Erik > > Unfortunately, this steps on my patch. Updated patch attached. Cygwin > really shouldn't be calling win32 API. > > On the

https://bugzilla.mindrot.org/show_bug.cgi?id=1953 TJ Saunders <tj at castaglia.org> changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |tj at castaglia.org --- Comment #6 from TJ Saunders <tj at castaglia.org> --- Instead of using custom SFTP

Hello, I am trying to use postfix/dovecot as mail server to be the imap storage for my voicemail system.For that I installed postfix and dovecot and trying to follow the instructions in this post http://etel.wiki.oreilly.com/wiki/index.php?title=Storing_Voicemail_on_an_IMAP_server&printable=yes I should add a master user for Asterisk to your IMAP server that has access to all user's

Hi. I'm trying to configurate Quintum AX to work with Asterisk SIP egister/Proxy server and my problem is that only the first user account get logged in and only that user is able to make call correctly. It seems to be a problem with authorization - I have noticed no "Proxy-Authorization" information in SIP INVITE, ACK, CANCEL messages. I have also noticed that when I remove

http://etel.wiki.oreilly.com/wiki/index.php/Main_Page -- ==================================================== J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' "Wise men talk because they have something to say; fools, because they have to say something." -- Plato -------------- next part -------------- A

I have some problems with busy detection and SIP. When I'm making a phonecall (out or internal) and someone else is calling me, the phone (Snom200) is ringing and leaving the first caller (no difference if I call someone or if someone calling me) in the background waiting. It doesn?t hang up the first call but the second one is overriding the first. Is there anyone that has experienced the

Hi, I tried to use Monitor(wav,filename) function in dialplan to record Meetme conference. When I monitored on IAX2 or SIP channels in that conference It recorded all audio (in and out) but when I monitored on ZAP channels I could hear only IN audio and piece of OUT audio (announcement get pin and than nothing). Anyone knows why this so happens??? I have asterisk 1.0.7 (debian package) and

Here is a mod to X86 that allows GAS to be used with MS Visual C++. I introduces a 'forWindows' variable like 'forCygwin' in th X86SharedAsmPrinter class. This may prompt thurther normalization, on the otherhand it may not :) Aaron -------------- next part -------------- An HTML attachment was scrubbed... URL: