VoIP Blog - Page 788
Analog Telephone Adapters
Posted: September 21, 2005 Important notice to product manufacturers, resellers and other people adding to this page If you want to add your company’s products, please read the Posting Guidelines for Promoting Products and Services The lists shall be in alphabetical order;…
Read More »Asterisk Cisco CallManager Voicemail Integration
Posted: September 21, 2005 This is my first wiki entry and needs to be tidied up a bit. I’ll do that when I get around to it, my main priority for now was getting this information online as it seems to be…
Read More »Asterisk tips Wake-Up Call PHP
Posted: September 21, 2005 Set a Wake Up call Features: Schedule or delete a single wake-up call No means to edit an entry No CRON job or daemon needed, scheduling works by .call file timestamp Ubuntu 13.10 – After upgrading to Ubuntu…
Read More »IXC
Posted: September 20, 2005 IXC offers VoIP solution, designed to handle contemporary VoIP carrier’s or service provider requirements. Portfolio of the company constitutes Billing Retail, Billing Wholesale, IXC Softswitch and IXC Autotester. * Main development directions Softswitch, VoIP solution with build-in web-based…
Read More »Monitor setup sample
Posted: September 20, 2005 The following only works with CVS head as of 2004-06-03. It allows a user to call out by dialing 8, then the number. Any calls dialed out using 8 are recorded. The “m” option on the 2nd set…
Read More »STUN
Posted: September 20, 2005 STUN stands for Session Traversal Utilities used for NAT (Network Address Translation). It is a server-client protocol for assisting devices behind a NAT firewall or router with their packet routing. RFC 5389 redefines the term STUN as ‘Session…
Read More »QueueMetrics
Posted: September 20, 2005 QueueMetrics is an inbound and outbound call center analysis, monitoring and reporting software application. The current version is 17.06 as of Jun 21, 2017. QueueMetrics is available as an on-premise or hosted solution. It offers over 150 quantitative…
Read More »ShadyDial
Posted: September 20, 2005 http://shadydial.sourceforge.net/ Predictive Dialer for Asterisk The link above at sourceforge is quite old, but does get you a feel for the product. The database backend is still PostgreSQL and it uses stock Asterisk parts, along with some freely…
Read More »GizmoProject
Posted: September 19, 2005 Gizmo Project is a closed-source software phone developed by SIPphone. It is a VoIP phone, made specifically for the SIPPhone network. It complies with Open Standards, directly peers with several other VOIP providers, and will directly dial sip…
Read More »Low Bandwidth VOIP
Posted: September 19, 2005 Although voice quality may suffer, VOIP can be used over a dial-up Internet or other low bandwidth connection. Some users have reported being able to use the Vonage service over dial-up, although Vonage doesn’t officially support this. Hints…
Read More »Asterisk cmd Random
Posted: September 18, 2005 This application is deprecated. Use ‘GotoIf($[${RAND(1,100)} > <number>]?<label>)’ instead. Synopsis Random goto Description Random(probability:label) Based upon a probability of 1-100 (percentage likeliness), goto the label specified. A probability of 1 will take the branch about 1% of the…
Read More »Protocol Stacks and Development Services
Posted: September 18, 2005 Protocol Stacks and VOIP software Data Connection http://www.dataconnection.com Encore Software Ltd. ncoretech Flextronics Software Systems (formerly: Hughes Software Systems) http://www.hssworld.com HCL Technologies Ltd http://www.hcltech.com/voip M5T http://www.m5t.com Advanced SIP Toolkits with security (TLS, SRTP, MIKEY) and NAT traversal (STUN,…
Read More »Open Source Billing Systems
Posted: September 18, 2005 This page lists Open Source Billing systems. BillRun Cloud An open-source, scalable billing solution designed for big data, handles highly-intricate billing, supporting any data structure or VoIP switch technology. Enabling intuitive, flexible configuration of all your billing needs,…
Read More »800 Number
Posted: September 18, 2005 Toll-Free or “800” Numbers are programmed by a Responsible Organization: RESPORG. Toll-Free Number and Service Providers (alpha sequence) 1Pipe Telecom | OnePipe is a licensed carrier providing Hosted PBX and SIP Trunking in over 60 countries. We have regional…
Read More »Asterisk Modem channels
Posted: September 17, 2005 Modem channels The “Modem” channel is factually only of use for ISDN cards operated with the i4l (isdn4linux) driver. Note that there is an alternative method to use ISDN hardware within Asterisk that employs the Asterisk CAPI Channels…
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