VoIP Blog - Page 750
NRenum
Posted: September 28, 2006 ENUM tree created by a Terena effort to be used in the countries where the Golden Tree (e164.arpa) is not (yet) available. Official site: www.nrenum.net Instead of delivering the countries delegations to the national telecommunications regulator, the delegation…
Read More »Port forwarding
Posted: September 28, 2006 Port Forwarding is configured on the router (machine making nat) and will forward connections originated from external clients to internal machine running asterisk. If using Linux/Netfilter, run, on router: SET SIP SERVER INTERNAL IP (LAN) SIP_SRV_LAN_IP=192.168.0.123 FORWARDING…
Read More »CNAM
Posted: September 28, 2006 CNAM is an acronym which stands for Caller ID Name. When phone calls are made, there are usually two user-facing identifiable pieces of information: a phone number and a Caller ID Name (usually a 15-character string). CNAM can…
Read More »Asterisk Google Talk
Posted: September 28, 2006 Starting with Asterisk 1.4, you can connect Asterisk and Google Talk, however as of now (Asterisk 1.4.4) you should consider this feature to be in beta phase, there are still quite a number of problems and glitches around….
Read More »VoIP Calling Card Platform Providers
Posted: September 25, 2006 This page is for all of the different Calling Card Solution Providers out there that have Turn-Key Solutions to offer. Please post info about your solution with links to the Data Sheets or White Papers as well as…
Read More »ZIDial
Posted: September 25, 2006 ZIDial is an application (module) for Low Cost Routing. The advantages of this application are: multiple redials on each route in the case of call failure (busy, cancelled, no answer, chan unavailable). All options from the Dial command…
Read More »Asterisk cmd Backticks
Posted: September 22, 2006 Synopsis: Execute a shell command and save the result as a variable. Description of application: Backticks(<VARNAME>|<command>) <VARNAME> – The name of the variable to be set. <command> – The shell command to execute; be sure to provide the…
Read More »Noa Sistemas
Posted: September 20, 2006 Noa Sistemas y Comunicaciones S.L. Desde Noa Sistemas ofrecemos a la PYME soluciones de negocio y de comunicaciones basadas en el uso de software de dominio pàºblico. Nuestro objetivo es acercar la Telefonàa IP a la empresa y…
Read More »RAKETU
Posted: September 18, 2006 Raketu is a proprietary protocol VOIP system built using PURE Peer-to-peer (P2P) techniques, in combination with SIP. NO SUPERNODES! Unlike other PC dependent software, Raketu users have choice to either install application on computers or use RakWeb to…
Read More »OpenSER 1.1.x and SerWEB 0.9.4
Posted: September 18, 2006 OpenSER 1.1.x and SerWEB 0.9.4 To use SerWEB 0.9.4 with OpenSER 1.1.0 you have to apply the patch attached below. Get the following patch (serweb-0.9.4-openser-1.1.0.patch): diff -urN serweb-0.9.4.orig/config/config_data_layer.php serweb-0.9.4/config/config_data_layer.php — serweb-0.9.4.orig/config/config_data_layer.php 2005-06-28 12:32:24.000000000 +0300 +++ serweb-0.9.4/config/config_data_layer.php 2006-09-18…
Read More »Pacific Consultants
Posted: September 18, 2006 We are providing PAKISTAN TOLL FREE DID’S: No Hidden Charges and DID FORWADED TO ANY PSTN or MOBILE Number We are offering suitable packages to serve your calling needs. In addition to this we offer fringe benefits to…
Read More »TOUBATEL
Posted: September 14, 2006 Services : – End user VOIP services (all major services like conference, Click to Dial …) – IP Centrex – Internationals DID (geol number, tollfree, paid number …) – CallShop solution – Wholesale service offering SIP/IAX/E1 * termination…
Read More »Asterisk bounty fix stability issue with app_AMD
Posted: September 14, 2006 500$ bounty for fixing a crash problem in Asterisk when using app_AMD. See bug ID 7885 although we believe there are two separate issues on this bug ID: segfault when zap channels are full and additional calls are…
Read More »FreeSwitch Dialplan Directory
Posted: September 7, 2006 FreeSwitch is an open-source telephony application capable of being a class 4 or class 5 soft-switch as well as a PBX. Freeswitch runs on Linux, OSX, Windows NT/XP/CE, *BSD and many other platforms. It is licensed under the…
Read More »FreeSwitch Dialplan
Posted: September 7, 2006 FreeSwitch is an open-source telephony application capable of being a class 4 or class 5 soft-switch as well as a PBX. Freeswitch runs on Linux, OSX, Windows NT/XP/CE, *BSD and many other platforms. It is licensed under the…
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