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...a couple of weeks ago, April 20, this about both the AreskiCC platform, and the Asterisk ASTCC application: http://lists.digium.com/pipermail/asterisk-users/2005-April/102710.html >[Asterisk-Users] Which free calling card app most suited forcommercial use? Kanuri, Seshu (Company IT) Seshu.Kanuri at morganstanley.com Wed Apr 20 08:46:23 CDT 2005 -----Original Message----- >Subject: RE: [Asterisk-Users] >Which free calling card app most suited for commercial >use?...

...fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk. Seshu Kanuri "G Rao" <grkalaga@dlink.co.in> Add to Address Book To: "Seshu Kanuri" <seshukanuri@yahoo.com> CC: mandarpise@yahoo.com Subject: Re: STUN server settings Date: Tue, 20 Jul 2004 20:30:26 +0530 Dear Mr. Seshu Kanuri, Thanks for your mail. D'Link Phones...

...mall as $1.00 a day will do for us. We will provide connectivity from our Softswitch IP 216.162.116.46. If anyone is interested, add this to your Asterisk IPBX and then email me for setting up a test account. My email address is voip@netwebgroup.com Thanks and have a great holiday weekend Asha Kanuri Netweb Group, Inc. http://www.netwebgroup.com

...s been developed for the masses. It's licensed under the GNU/GPL license, easy to install and administer and reliable. Joomla! doesn't even require the user or administrator of the system to know HTML to operate it once it's up and running. http://www.joomla.org/ -------------- Seshu Kanuri -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.

...andstream to connect to my Asterisk Server. All the digits 0-9 are accepting dtmf. But When I try to send the call by Pressing # Key, nothing happens. Does anyone has a standard configuration for Asterisk and Grandstream as a PDF file or something to see. How do you send the connect signal? Seshu Kanuri -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Foster Sent: Thursday, July 29, 2004 9:13 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] BugetTone Bug Showstopper, On Wed, 28 Jul 2004 2...

...this part of the ongoing effort on finding a free billing solution. Once again, this is not my code and I dont want to steal any credit from anyone in this community. I am going to make a tarball of all the .PHP files and the My SQL DB and send to those who responded to my posting today. Seshu Kanuri Data Architect & Modeler Citibank NA 1 Court Square Long Island City New York, NY 11002 Ph: 718-248-3074 Fx: 413-812-3152 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Areski Sent: Wednesday, July 14, 2004...

I am trying to download the latest release of Asterisk from: ftp://ftp.digium.com/pub/asterisk/ The link provided by Digium is incorrect for the Asterisk Tarball as there is no such file at ftp://ftp.digium.com/pub/asterisk/asterisk-1.0.7.tar.gz However the links for the Asterisk-Addons and other Tarballs is OK ftp://ftp.digium.com/pub/asterisk/asterisk/asterisk-addons-1.0.7.tar.gz Does anyone

...a: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di FaberK Inviato: gioved? 13 ottobre 2005 17.56 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] PA168S/AT320P Right now, but nothing changed. 2005/10/13, Kanuri, Seshu (Company IT) <Seshu.Kanuri@morganstanley.com>: > have you configured the STUN server on the phone to any one of the > available stun servers like stun.xten.net? > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-user...

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI

Why is no one suggesting any solution here for this problem, which has been lingering for a while. Are we not supporting H.323 on Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of ruixun wu Sent: Thursday, July 22, 2004 4:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] still can't

...ectivity to Asterisk. Recommendation: -------------- 1)Asterisk community should work on passthrough connectivity for g723(Currently it does not work, though I read that it is there) I will post results of tests with our own Brand of SIP Phone "eezeephone *Ship" in the next post. Seshu Kanuri eezeePhone.com

Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated

...codecs. Best is to post the SIP.CONF entries here to see what is missing. By the where did you get the G723 and G729 from? If you have compiled them on your own, did you statically link the libraries? Or just copied the .SO files from another dude's Asterisk box? Post all the details Seshu Kanuri -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.

If anyone in the list has a working version of the chan_oh323.so file for Fedora Core 2 and Redhat, can he email the same to the list as attachment. This will reduce the pain for many of the users who are trying to compile the same from the libraries, which never seemed to work. Seshu Kanuri -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Humberto Aicardi Sent: Wednesday, January 05, 2005 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] chan_oh3...

I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux on this router and included a bunch of Linux tools, one of which is Bandwidth Management. The QoS aspect of this is supposed to be much more granular than the previous solution (Wonder Shaper). I have not been able to find any suggestions for how to impliment QoS (Bandwidth Management) using the web interface of Alchemy.

Eric Wrote: ----------- The trick is not to use options you don't understand. "show application dial" will show you what the t and T options are for. Most people use the transfer feature of their phone, rather than using the T/t hack on the Dial line. Sounds like you are using CVS-HEAD and so will have to configure stuff in /etc/asterisk/features.conf. /Snip/ Eric, Thanks for

I have followed the Idiots' guide for installation, but still could not make it work. When I try to login at the web page coming from /var/www/html/areski , I get the following errors: Can some body give me some hints where and what to check for this error?. I am looking for info on the changes we have to make for 1) the database name 2) user name 3) password 4)connection name (server

I am wondering if anyone has a working install of oh323 on fedora Core2. An replies would be appreciated as we need this urgently. Seshu Kanuri -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of steve@nexusuk.org Sent: Thursday, July 22, 2004 6:12 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] error while compiling asterisk-oh323 -----BE...

...kup up the line ...etc. etc. This phone would soon be available in India(by next week or so, first branded asr "Netweb Phone") and USA in the next couple of months as "NetwebPhone" and could be priced around $65/-(tentative) See the communication from D'Link below. Seshu Kanuri ----- Original Message ----- From: Mandar Pise (Netweb India Ltd) To: Abhijit M ( Voip Dept.) Sent: Monday, July 19, 2004 1:51 PM Subject: Re: D-link ( DPH-80 ),Sip Firmware Upgrade . Dear Mr. Abhijit, As per our teleconversation with you, I am forwarding you the sip server logs attached al...

Good day all I've tried my iax conf and I'm struggling.So I want to know If someone else got this working and if they can pleas send my their configs I have to asterisk server,in different tows,both offices connected wit a direct line so both servers are on the same network running SIP.Each town got different extension register to each sever.Town A=100+ town B=200+ How do I get town A