Author: VoIP Info
Digium Wildcard TE405P
Posted: July 25, 2006 The TE405P is the next generation of Digium hardware that improves performance and scalability through bus mastering architecture. The TE405P supports both E1 and T1 environments and is selectable on a per-card or per-port basis. This feature enables…
Read More »Digium Wildcard TE212P
Posted: July 25, 2006 The TE212P offers an on-board DSP-based echo cancelation module. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis. The TE212P is a bundling of our leading TE210P product and our new…
Read More »Digium Wildcard TE210P
Posted: July 25, 2006 The TE210P is the next generation of Digium hardware that improves performance and scalability through bus mastering architecture. The TE210P supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis. This feature enables…
Read More »Digium Wildcard TE207P
Posted: July 25, 2006 The TE207P offers an on-board DSP-based echo cancelation module. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis. The TE207P is a bundling of our leading TE205P product and our new…
Read More »Digium Wildcard TE205P
Posted: July 25, 2006 The TE205P is the next generation of Digium hardware that improves performance and scalability through bus mastering architecture. The TE205P supports both E1 and T1/J1 environments and is selectable on a per-card or per-port basis. This feature enables…
Read More »Digium Wildcard TE110P
Posted: July 24, 2006 The Wildcard TE110P brings a high-performance, cost-effective, and flexible single span togglable T1/PRI or E1/PRI interface to the Digium line-up of telephony interface devices. The TE110P is a single span, selectable T1 (24-channel) or E1 (32-channel) card that…
Read More »Text-to-Speech (TTS)
Posted: July 21, 2006 What Is Text-to-Speech, or TTS? From Wikipedia, TTS (Text To Speech) is defined as follows: “Speech synthesis (Text to speech) is the artificial production of human speech. A computer system used for this purpose is called a speech…
Read More »app_zapscan.so Modification
Posted: July 20, 2006 Description of the function requested: Currently zapscan.so can be compiled to ignore a group, but not individual channels. The ideal situation would be to have a new config file added in /etc/asterisk, zapscan.conf perhaps, where a list of…
Read More »GXP-2000 XML Idle Screen
Posted: July 19, 2006 General Info This page will serve as a repository to help explain the functionality of custom idle screens, as well as document bugs and identify workarounds where possible. Page Contents Firmware Notes and Bugs Firmware 1.1.1.14 Bugs Firmware…
Read More »F3000
Posted: July 19, 2006 UTstarcom F3000 Wireless IP Phone Color LCD Screen (65000k) 802.11b/g Compatible Protocols: SIP Product Description The UTstarcom F3000 is a clam shell style wireless IP phone that combines the power of Voice Over IP technology with the…
Read More »Intraway Corporation
Posted: July 19, 2006 Website: http://www.intraway.com/ Provisioning, Management and Monitoring of network elements, including DOCSIS Cable Modems, PacketCable and SIP based Telephony, different types of SoftSwitches, large WiFi Hotspots networks and digital and analog Set-Top Boxes from different vendors. Intraway offers products…
Read More »GXP-2000 XML Phonebook
Posted: July 18, 2006 Page Contents Known XML Phonebook Issues BETA 1.1.1.14 Beta 1.1.1.9 Beta 1.1.1.7 XML Phonebook Manual Generate phonebook from Outlook Dynamic XML Phonebook from MySQL table MySQL table create script PHP script Alternate PHP Script that has additional field…
Read More »SIP/IAX Services for Asterisk
Posted: July 17, 2006 This page lists providers of SIP/IAX services for Asterisk. Page Contents Argentina Australia Austria Belgium Bulgaria Canada Chile China Colombia Czech Republic Denmark France Germany Hong Kong Iceland Italy Ireland Japan Mexico Mongolia Netherlands New Zealand Poland Portugal…
Read More »Asterisk sip allowguest
Posted: July 15, 2006 sip.conf: allowguest This setting determines if anonymous callers are permitted to place calls to Asterisk. This defaults to ‘yes’. You will need to either set the appropriate default context in the [general] section of sip.conf, or you can…
Read More »SPA-921
Posted: July 14, 2006 Linksys SPA921 Low cost SIP phone costing usually $119.95 Availability: Now Available Page Contents Features Reviews Support Quick compare Linksys/Sipura Devices (to the SPA941) Behind NAT configuration (for home use) Bugs, Tweaks, Feature Requests Where to Buy: See…
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