Author: VoIP Info
Asterisk settings for Bluesip
Posted: November 5, 2007 Bluesip.net provides one free Germany DID. To set up with your asterisk box add the following settings to /etc/asterisk/sip.conf : ; Bluesip uses bluesip/username as login, the number 498971234567890 is the DIDregister => bluesip/username:[email protected]/498971234567890 [bluesip]disallow=allallow=alawallow=ulawtype=friendusername=bluesip/usernamefromuser=usernamesecret=passwordhost=bluesip.netcontext=from-pstnfromdomain=bluesip.netauth=md5,plaintextqualify=yescanreinvite=nonat=noinsecure=very If you are…
Read More »Asterisk settings for Eutelia
Posted: November 5, 2007 EuteliaVoip accounts offers you free geographic numbers (Italian, English, Romanian). Each account can have up to 15 numbers. To configure your account with your asterisk box you need the following settings: Edit /etc/asterisk/sip.conf and add the following lines:…
Read More »Asterisk CTI Open Source
Posted: November 2, 2007 CTI for Asterisk I have seen a few and they are all commercial. For this reason I have decided to initiate a project to develop a client-server solution that meets a series of requirements CTI base. Go to…
Read More »AGX Extra Addons for Asterisk
Posted: October 31, 2007 Due to Asterisk project dual license some usefull code cannot be submitted into the official repository. I’ve ported some GPL code from version 1.2 to 1.4 for production use. This includes: enchanced app_rxfax (app_rxfax) enchanced app_txfax (app_txfax) device…
Read More »Siemens Gigaset S675IP
Posted: October 24, 2007 Siemens Gigaset S675 & S685IP Summary A VoIP DECT Phone based on the Siemens Chagall Platform. The Siemens Gigaset S675IP/S685IP is a successor to the Siemens Gigaset S450IP. It adds the following features: Page Contents Firmware Update…
Read More »xia308
Posted: October 21, 2007 The product of XIA308s has many advantages :first ,support protocals h..323 , SIP and MGCP. In the most area of Middle East , the risk of voip block can be reduced if you use h.323. Second, we will…
Read More »Nakul Malik Consultancy Services
Posted: October 19, 2007 NMCS is a full service Network and Telecommunications consulting, design and training company specializing in a variety of Cutting-edge and Legacy technologies. Our Mission: Our mission is to successfully integrate the latest network and telecom solutions into our…
Read More »MySQL Macros for Asterisk RealTime
Posted: October 16, 2007 If you are using Asterisk Realtime, here are some macros that will help with performing different functions in the dialplan. Page All Phones This will Page all extensions in the mysql database that begin with a specify number….
Read More »Asterisk RealTime H323
Posted: October 14, 2007 Asterisk RealTime H323 (chan_h323) The realtime code was merged into trunk more then a year ago but still no documentation on that. So I decided to fill this vacuum. Asterisk realtime for chan_h323 is configured in same ways…
Read More »Link2voip.com
Posted: October 14, 2007 IAX not supported. SIP only. Taken from https://www.link2voip.com/login/help.php [link2voip-sw1] context=from-trunk ; change me type=friend host=sip.ca1.link2voip.com username=usernamehere secret=passwordhere canreinvite=no ; if using a nat, do not change insecure=port,invite ; do NOT remove this qualify=5000 ; do NOT remove this…
Read More »SIPVicious
Posted: October 11, 2007 What is SIPVicious tool suite? Consists of 4 tools: svmap: an active scanner to identify SIP devices on the network svwar: scans SIP PBX servers for existing extensions svcrack: an online password cracker against SIP PBX servers svreport:…
Read More »How to Doorbell or Doorphone
Posted: October 10, 2007 I was frustrated when trying to create a doorbell for my office to simulate the Toshiba Strata digital doorphone we had. I replaced the system with a custom Asterisk box with a TDM800 card (8 fxo). Since our…
Read More »Visualware
Posted: October 9, 2007 Visualware is a leading provider of VoIP testing software, providing tools to simulate VoIP traffic and measure the level of VoIP quality and number of VoIP lines supported by a network connection. The MCS MyVoIP tools enable businesses…
Read More »Asterisk Flite
Posted: October 4, 2007 Install Flite and asterisk-flite (in RHEL, CentOS and similar) Flite (festival-lite) is a small, fast run-time synthesis engine developed at CMU and primarily designed for small embedded machines and/or large servers. Flite is designed as an alternative synthesis…
Read More »Automatic Call Distributor
Posted: October 4, 2007 Automatic Call Distribution or ACD, is a tool commonly used in the telephony industry. ACD systems are commonly found in any office that handles a large volume of inbound calls. The primary purpose of an Automatic Call Distributor…
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