search for: seshu

Seshu, Whats with people who work at Morgan Stanley? You on the one hand bash the opensource software, and then on the other hand a few weeks later, ask for assistance from the open source community for assistance??????????????????????????????????????????? Today, May 11, you request assistance in ins...

...them 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...

...ch 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 know a Mirror Site for the latest build of Asterisk? Google is not much help. Seshu -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.

...la! has 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.

Abel, In have the same issue when I have burned the image to an 800MB CF Disk. All it displays is GRUB CLI in a continuous stream. Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of abel Sent: Monday, June 06, 2005 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Live! CF I downloaded teh...

...tup Grandstream 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...

...making 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 1...

...rors: 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 running postgresql) in all the files involved in the application, so that it works. Seshu ------------------------------------------------------------------- Warning: pg_pconnect(): Unable to connect to PostgreSQL server: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? . in /var/www/html/areskicc/lib...

...isk-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-users-bounc...

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.

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

...tion problem due to H245 Tunnel, check the h323 Config embeded at the end. Comment the offending line as under: ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; -----Original Message----- From: Tola Ogunsan [mailto:tolaniye@hotmail.com] Sent: Wednesday, May 25, 2005 1:03 PM To: Kanuri, Seshu (Company IT) Subject: RE: oh323 problems Thanks a million Seshu, it worked like a champ. Thanks >From: "Kanuri, Seshu (Company IT)" <Seshu.Kanuri@morganstanley.com> >To: "Tola Ogunsan" <tolaniye@hotmail.com> >Subject: RE: oh323 problems >Date: Tue, 24...

...em? VAR: agi_request: init.sh ;( Init.sh is sent from PBXware) VAR: agi_channel: IAX2/101@101/2 <mailto:IAX2/101@101/2> VAR: agi_language: en VAR: agi_type: IAX2 VAR: agi_uniqueid: asterisk-28947-1110463619.0 VAR: agi_callerid: Seshu Kanuri <101> VAR: agi_dnid: 56 ; Actual number dialed was 51 VAR: agi_rdnis: unknown VAR: agi_context: default VAR: agi_extension: 56 VAR: agi_priority: 1 VAR: agi_enhanced: 0.0 VAR: agi_accountcode: Detected...

We have replaced the simple contact management system in Asterisk@Home with SugarCRM a full CRM system. This might seem like over kill for a home PBX but Sugar has the best contact management we have seen. With click to dial functionality and the ability to import data from other contact managers it’s a great fit for Asterisk@Home. We have also added new version of the usual Asterisk software AMP

...guration and even if you add this manually in the sip_additional.conf, AMP will overwrite(remove) this line from the sip_additional.conf To make Polycom phone work with Asterisk, add the sip.conf entry into the main sip.conf, using progressinband=no option. Polycom phone will immediately connect. Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wiley Siler Sent: Monday, June 06, 2005 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] How to make Polycom phon...

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

...r connectivity 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

Folks! Netweb Group, Inc. fully supports connectivity to any Asterisk PBX systems you have and can provide A-Z termination with immediate effect. Any volume is good enough for us, even an amount as small 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

Hi, I appreciate it if someone knows what is available for SIP web phones out there. I am interested in putting a soft phone on a website that registers with Asterisk using SIP. Then, when someone uses it, it directly calls into an asterisk call queue.. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone, I am developing an online SMDR / call log system for asterisk. This is going to take the form of an executable with embedded sql and webserver, pdf generation, excel generation, graphs. Actually, we have been selling this for a while now with great success and now I am starting work on the integration with Asterisk. Its a windows executbale and the executable is just about 1MB.