Author: VoIP Info
Fonize
Posted: July 5, 2010 We are exclusively dedicated to provide state of the art solutions for your company to increase online sales and customer satisfaction. We provide a button/widget that allows visitors to call your company directly from their computer (using VoIP)…
Read More »Asterisk ss7 loopback
Posted: July 1, 2010 chan_ss7 configuration for loopback useful for– stress testing– hardware testing voip*CLI> ss7 statuslinkset idle busy initiating resetting total incoming total outgoingpeer1 167 18 0 0 0 18linkset idle busy initiating resetting total incoming total outgoingpeer2 167 18 0…
Read More »DTH Free Call Rating System
Posted: June 27, 2010 This free, hosted service was designed primarily for companies who need to rate and tax their call records but may not be able to justify implementation of a full VoIP Billing and Customer Management system. Using our service…
Read More »ICTInvoice an open source Elastix module for invoice managment
Posted: June 25, 2010 Introduction ICTInovice is an Elastix module that enhances capabilities of Elastix billing and empower Elastix admin to automatically generate and email invoices to users of system on monthly basis based on call details records applying rates already set…
Read More »Contaque – Predictive Dialer
Posted: June 18, 2010 A predictive dialer is an automatic dialing system that can dial many telephone numbers at a time.Many call centers are operated by using a predictive dialer to improve the total productivity.Predictive Dialing is an ideal dialing pattern to…
Read More »Goldfish.ie
Posted: June 17, 2010 Goldfish Fax Email Business VoIP is a provider of IT communications services for small and growing businesses. All of our services are provided using our hardware and hosting services & the services are manageable either over the phone…
Read More »Asterisk RTCP
Posted: June 8, 2010 Before patch 10590 (which was applied to Asterisk 1.6) the RTCP data presented in Asterisk 1.4 were basically useless: For example packet loss values could be 10x higher than the number of total received packets, and only one…
Read More »FreeSentral
Posted: June 3, 2010 Official website FreeSentral home page FreeSentral is the free IP PBX based on the telephony engine Yate. Page Contents PBX Features User-Friendly Features LiveCD Resources Download Support It consists of a Linux distribution, an IP PBX and a…
Read More »Asterisk4arab
Posted: June 1, 2010 The Revolution arrived the Arabic World, Opening the first Asterisk forum in Arabic language. Discussing and explaining all the needs off Arabic user to use Asterisk. We will try to localized Asterisk to Arabic language, like Arabic Voice…
Read More »OCN
Posted: May 31, 2010 OCN (Operating Company Number) is a 4 character ID for North American phone companies. OCN numbers are assigned by NECA and used to identify companies in other telecommunications resources such as the LERG. Examples of OCN Numbers: 0001…
Read More »NECA
Posted: May 31, 2010 NECA (National Exchange Carrier Association, Inc.) is a USA membership organization of over 1,350 telephone companies and exchange carriers created in 1984. manages the access charges phone companies pay each other when they exchange phone traffic management of…
Read More »One way Voice with Linux SIP conntrack
Posted: May 20, 2010 One way voice problem is seen on device running behind NAT, if call is initiated from device to Soft phone which is not running on SIP Server m/c.( Softphone and SIP server, running on diffrent m/c)Following is Setup…
Read More »CDRCat
Posted: May 19, 2010 SVK Software Corp _CDRCat_: CDR Processing Software for Gateways & IP PBX’s CDRCat produces customer statements, invoices, statistical & performance reports by simply reading and parsing any CDR file produced by VoIP Gateways, PBX Servers, Switches etc. CDRCat…
Read More »ConferenceCat
Posted: May 19, 2010 SVK Software Corp: Carrier/Enterprise Voice Conferencing Solution for SIP Servers ConferenceCat Voice Conferencing Server is a Multi Featured; Multi Language Voice Conference Server (Conference Bridge) that interoperates with your PBX/Gateway using the Session Initiation (SIP) Protocol. ConferenceCat also…
Read More »Digium Free Fax For Asterisk and sendEmail
Posted: May 18, 2010 Description This is a working example of a digium free fax for asterisk and fax to email implementation for asterisk . Packages used • Digium Fax for Asterisk (receive faxes) • sendEmail (sending emails) – http://caspian.dotconf.net/menu/Software/SendEmail/ Overview Create…
Read More »
Phone System 