Author: VoIP Info
Asterisk Sound Files International
Posted: September 19, 2005 As of October 2009, it is possible for new languages to be added to Asterisk if you or your company is willing to have the recordings made and then contribute them under the Creative Commons license. They will…
Read More »Asterisk Consultants Italy
Posted: September 19, 2005 Lista dei consulenti Asterisk in Italia, suddivisi per regione e provincia, che forniscono consulenze per l’installazione e l’amministrazione di centralini telefonici basati su Asterisk. E’ possibile avere supporto online in italiano attraverso la rete IRC freenode, canale #asterisk-it…
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 »Asterisk Cisco CallManager Express Integration
Posted: September 18, 2005 The new family of ISR routers provided by Cisco make great media gateways for PSTN connectivity and make a very nice IP enabled key system replacement. Unfortunately, the solution can be pricey and lacks some of the flexibility…
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 »MetroTel
Posted: September 18, 2005 MetroTel jltaylor@metrotel.net Wholesale TDM SIP IAX Flat Rate Products: 4 T1’s – 1,000 DID numbers $500 total monthly charge – local loop includedDID numbers for call centers and calling card companies. Use our numbers to improve call completion.Help…
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…
Read More »Protocol Verification and Testing
Posted: September 17, 2005 Sources for software and services for testing and verifying that a protocol implementation conforms to a standard. IVR Load Testing Tools BreakingPoint Systems Testing Tools Empirix http://www.empirix.com miTester for SIP: An Open Source SIP Testing tool https://mitesterforsip.sourceforge.net/ GL…
Read More »QoS with Linux using PRIO and HTB
Posted: September 17, 2005 I have used the Wondershaper and other QoS scripts for Linux, and they were surely better than nothing, but they were never perfect. The problem is that they all use fair queing schemes where one connection cannot steal…
Read More »Snom Phones
Posted: September 17, 2005 www.snom.com Snom SIP based phones: Snom m9: DECT based cordless solution. Sold as base station with handset/charger station and single handset/charger station. Up to eight handsets can register with the base station and the base station can register…
Read More »UIP200
Posted: September 17, 2005 Uniden UIP200 Sip Phone We have a bunch of UIP200s in our office and they’re mostly fine. We use Asterisk for transfers, so we have no issues with call dropping. Two complaints: The time zone info is wrong…
Read More »Asterisk cmd Dictate
Posted: September 17, 2005 Dictate Virtual Dictation Machine: Asterisk now comes with an application to facilitate dictations, see bug 3893. Description Dictate([<base_dir>[|<filename>]]) Start dictation machine using optional base dir for files. Options dial 1 to toggle record and playback modes dial 0…
Read More »Dial Pulse to Touchtone DTMF Converters
Posted: September 17, 2005 Many VOIP devices with FXS ports do not support traditional rotary dial phones. One possible solution (untested) is a Dial pulse to DTMF (Touchtone) converter between the rotary dial phone and the VOIP FXS port Example: Dial Pulse…
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