Super Skype Gateway
The Best Solution to Take the advantages of Skype Unlimited Calls to Landlines Skype Unlimited Calls to Landlines from Skype.com Callfree Gateway 1600 Professional – CG1600PRO – Save Up to 160,000 Minutes per Month One of the best…
Read More »AT-510
Infineon Chipset Inside AT510 series IP phone is an internet based voice network phone terminal supporting power supply through Ethernet. AT510 series IP phone adopts multiple voice control protocols and voice compression codec to directly convert analog voice…
Read More »Toll Free Termination Providers
Toll-free termination are calls destined to 8YY destinations. These providers allow you to terminate toll-free calls from the US and Canada for free in some cases. If you have a large volume of calls to toll-free numbers, some…
Read More »Perl Add/Modify/Delete user script
i have created AddUser and EditUser scripts in perl for adding/modify/view/delete sip/voicemail entry in config files. Requirements: Asterisk::config CPAN module Features and working add/edit sip entry add/edit voicemail entry if required add/remove hint in extensions.conf if required. when…
Read More »Use a second internet connection for SIP
Situation: Asterisk is running on a box which is internet router for a LAN and has two internet connections. Assume that the default gateway sits on the first internet connection. Goal: To use the first internet connection for…
Read More »Asterisk tips Call Back
Callback script I made, Callback works when called channel is busy, if SIP user is busy it will return user unavailable and send the control to s-NOANSWER. How it works If called channel or phone is busy, it…
Read More »Asterisk app_h324m compatibility
app_h324m compatibility This page collects compatiblity information about app_h324m/libh324m with mobile phones. Please add your test results in the following syntax: Mobile Phone Mobile Phone Firmware Version Revision of h324m Results H324M–>h324m_gw Results h324m_call–>H324M Notes Nokia 6630…
Read More »Pakistan DIDs ,Web development
Dear Customers: DIDs for just 7$ without any setup charges and no hidden charges. S0lution4voip is very delighted to announce that they have now also started s0lution4dids. We are offering DIDs of Pakistan at very low cost as…
Read More »AU-120F
USB Phone :AU-120F Open SDK USB phone with flash memory AU120F is a USB phone with 32M-1G flash memory made by ATCOM Company. With the flash memory and open SDK, the customer can put their softphone into it….
Read More »VMukti Call Center 1.1
VMukti 1.1 Open Source Call Center Software web-based 2.0, multi-tenant, distributed, multi-lingual, inbound, outbound Video enabled World’s first open source Enterprise P2P Call center software More world firsts: – Combination of power of Web, P2P, & Telephony. –…
Read More »IP04
4 FXO/FXS port IP PBX The IP04 is a complete Asterisk Appliance with four FXO or FXS module. It is an embedded open source Linux system with built-in SIP/IAX2 proxy server and NAT functions. It provides a solid,…
Read More »AX-1600P
4 ports FXO/FXS card AX1600P Asterisk card is the telephony PCI card that supports up to sixteen FXO and FXS ports. Using AX1600P analogue card, open source Asterisk PBX and stand alone PC, users can create their SOHO…
Read More »AG-188N
AG-188N One Port ATA AG-188N series voice gateway is an Internet based one port voice gateway. AG-188 series adapts multi voice control protocols and voice compression codec to directly convert analog voice into IP packet for internet transport,…
Read More »AG-110
AG-110 Gateway Most cost effective Mini ATA AG-110 is an Internet-based one FXS port VoIP gateway. It adapts SIP protocol and multiple voice compression codecs to directly convert analogue voice into IP packet for internet transport, thus effectively…
Read More »Polycom Trixbox
Loading the latest firmware on a Polycom can be difficult. Trixbox makes this easy by providing RPMs with the latest Polycom 3.0 firmware. Once you have the firmware installed on your Trixbox you can scan the network for…
Read More »