Author: VoIP Info
OptiPoint 600 SIP with Asterisk
Posted: September 10, 2005 OptiPoint 600 SIP with Asterisk The Siemens OptiPoint 600 SIP works quite well with Asterisk. Creating and using a custom logo Updating firmware / Switching from HFA to SIP Sample configuration for use with Asterisk Multi-line operation Distictive…
Read More »SS7
Posted: September 10, 2005 Common Channel Signaling System No. 7 (aka SS7 or C7) is a global standard for telecommunications defined by the International Telecommunication Union (ITU) Telecommunication Standardization Sector (ITU-T). The standard defines the procedures and protocol by which network elements…
Read More »How to use G.729 codec in Asterisk and license info
Posted: September 10, 2005 Open source Many people use Asterisk because they want an open source solution. They are frustrated when they find that the G.729 module from Digium (the Asterisk creator) is not open source. In fact, it is technically possible…
Read More »Asterisk G.729 pass-thru
Posted: September 10, 2005 How to use the G.729 codec in pass-thru mode in Asterisk “Pass-thru” means that if you were, for example, using two phones which both have inbuilt support for g729 codec. You can let them talk to each other…
Read More »Asterisk codecs
Posted: September 10, 2005 Asterisk Codecs Asterisk supports the following narrow-band and wideband (HD audio) codecs: G.711 ulaw (as used in US) G.711 alaw (as used in Europe) G.722 – 16 kHz wideband codec; passthrough, playback and recording in Asterisk 1.4; full…
Read More »zaptelBRI ISDN Interface cards
Posted: September 10, 2005 junghanns.net makes a quadBRI and octoBRI card for interface with ISDN phones or ISDN lines providing a PSTN connectivity gateway with Asterisk. They also have a single and double E1. (PRI) Currently, they make a quad and octo-port…
Read More »SS7 Protocol Converters and Gateways
Posted: September 10, 2005 Convert SS7 connections to other connection types (e.g. PRI) to make interconnecting SS7 networks and non-SS7 equipment possible. Products and Services SS7 TDM E1 ISDN R2 T1 Signaling/SIGTRAN gateways converters. www.acecomnet.com [email protected] Cosmos SS7-SIP Gateway Turn-key, high performance…
Read More »PSTN
Posted: September 10, 2005 PSTN Definition from Webopedia “Short for Public Switched Telephone Network, which refers to the international telephone system based on copper wires carrying analog voice data. This is in contrast to newer telephone networks base on digital technologies, such…
Read More »cisco 79xx
Posted: September 10, 2005 Cisco’s 7900 Series IP Phones Index: Configuring Cisco 79xx phones with Asterisk (applies for any SIP utilisation) Asterisk phone cisco 79×1 xml configuration files for SIP Configuring your 7920 or other Skinny phones with Asterisk via chan_skinny, included…
Read More »Vonage
Posted: September 9, 2005 Vonage is the biggest and most visible provider of SIP based VoIP to PSTN gateway services. Resources Vonage VoIP Forums Skype vs Vonage Vonage for iPhone Vonage for BlackBerry There is also a lot of discussion of Vonage…
Read More »Atcom AT-320
Posted: September 9, 2005 Atcom AT-320 This is a cheap consumer grade phone based on the PA168S chip. Works good in both the office and at home. Pros it’s cheap. About $79 for a dual port unit. Nice display Good status display…
Read More »Asterisk at Home Examples
Posted: September 9, 2005 After finding a significant lack of information on how to actually configure Asterisk@Home, I thought I would just jot down some notes on how to set up a couple of services. Setting up a Softphone on your PC…
Read More »Kiax
Posted: September 9, 2005 Kiax started in early 2004 as a small program mainly aimed to provide a simple user interface for making VoIP calls with Asterisk PBX (an open source VoIP PBX). Its first versions showed the existing need for a…
Read More »Asterisk “i” invalid extension
Posted: September 9, 2005 Each Asterisk contexts in extensions.conf may have an ‘i’ extension that is used for invalid extensions, i e, non-existing extensions dialled by clients. Not Automatic You would think that Asterisk would automatically jump to the ‘i’ extension if…
Read More »Asterisk sip dtmfmode
Posted: September 9, 2005 DTMFmode for SIP client configuration Syntax dtmfmode=inband Choices are inband, rfc2833, info or auto inband: This send tones as inband audio within the voice stream. The device that you press the key on generates the DTMF tones. –…
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