Author: VoIP Info
Asterisk .NET
Posted: October 22, 2005 Note: This project has been re-homed (original maintainer has stopped working on the project). The new home for Asterisk.NET is http://asternet.codeplex.com/ and has been renamed to AsterNET. Latest version: http://sourceforge.net/projects/asterisk-dotnet 2008-06-19 Fix a bugs and add Asterisk 1.6…
Read More »Pennytel
Posted: October 22, 2005 Pennytel provides a variety of VoIP services and calling cards. Their voip is primarily via pre-paid fixed dollar amounts. Its popular amongst Australian VoIP users for their cheap international and mobile phone rates (relatively speaking!). Their rates for…
Read More »Asterisk-IM – Jive Software Integration
Posted: October 20, 2005 Asterisk-IM Integration component to Asterisk for Jive’s Jabber/XMPP server The homepage for the plug-in component Plug-in Readme Asterisk-IM compatible clients Asterisk-IM is a server-side module so it will work with any XMPP client application to provide integrated presence…
Read More »Sipura Syslogging SIP INVITE incoming
Posted: October 20, 2005 Incoming Completed Call Here’s an example of an incoming call from 1-866-510-6559 to the SPA. Assumptions SIP Port: 5062 (SPA SIP Port) SIP Proxy Server: sip.voip-info.org SPA IP address: 192.168.0.2 User ID: spa1 Display Name: Good Body …
Read More »Sipura Syslogging SIP INVITE outgoing complete
Posted: October 20, 2005 Complete Outgoing Call Here’s an example of the SIP messages between an SPA calling to phone number 1-800-555-1212 through an Outbound SIP Proxy. Assumptions SIP Port: 5062 (SPA SIP Port) SIP Proxy Server: sip.voip-info.org SPA IP address: 192.168.0.2…
Read More »Sipura Syslogging SIP NOTIFY MWI
Posted: October 20, 2005 Message Waiting Indicator Here is a syslog entry capturing the SIP message exchange from a SIP Proxy to an SPA updating information to the SPA about possible Message Waiting Indications. Assumptions: SIP Port: 5062 (SPA SIP Port) SIP…
Read More »Sipura Syslogging SIP INVITE outgoing cancel
Posted: October 20, 2005 Outgoing SIP INVITE with CANCELation Assumptions SIP Port: 5062 (SPA SIP Port) SIP Proxy Server: sip.voip-info.org SPA IP address: 192.168.0.2 User ID: spa1 Display Name: Good Body Message Flow Explanation SPA attempting to make a call…
Read More »Sipura Syslogging Interpretation
Posted: October 20, 2005 Explanation of the Syslogging Info Coming from the SPA Here are some general explanation for what to look for in the syslog messages generated by an SPA. SIP REGISTER to a SIP Proxy that requires authentication. A SIP…
Read More »Sipura Syslogging SIP REGISTER
Posted: October 20, 2005 Registration Assumptions SPA behind a Full Cone NAT SIP Port: 5062 (SPA SIP Port) SIP Proxy Server: sip.voip-info.org SPA Private IP address: 192.168.0.2 SPA Public IP address: 172.181.0.254 User ID: spa1 Display Name: Good Body Line 1…
Read More »Sipura Syslogging
Posted: October 20, 2005 Configuring the Sipura Phone Adapter for Syslogging Support The SPA has the capability to log execution and debugging information to a remote syslog server. The syslog server must understand the BSD syslog protocol RFC3164 and use UDP as…
Read More »Voice over IP Directory Services
Posted: October 20, 2005 It is now a well-known fact that VOIP is the next best thing since the Internet. In the next couple of years, the majority of voice communications across the world would be based on VOIP. Home users and…
Read More »SRV implementations
Posted: October 19, 2005 Here you find some test results of SRV based failover. Please extend this page with your test results: bad clients: – eyeBeam 1.1 3004w stamp 16863 uses first SRV, no failover, ignores ICMP (retransmissions) – eyeBeam 1.1 3010n…
Read More »Asterisk hardware Gumstix
Posted: October 18, 2005 At Astricon 2005 in Anaheim, CA I (Kristian Kielhofner) introduced the world’s smallest VoIP enabled PBX. Here are the facts: Gumstix 400xm – 16mb flash, 64mb RAM, 400mhz xscale Root filesytem (with Asterisk) as low as 7mb with…
Read More »Aastra 480i Dial Plan Config
Posted: October 18, 2005 This is the updated dial plan information for firmware version 1.3 A dial plan describes the number and pattern of digits that a user dials to reach a particular telephone number. The SIP local dial plan syntax is…
Read More »NetAppel
Posted: October 18, 2005 NetAppel.fr According to some comments on this wiki they’re connected with VoipBuster but they have different FREE countries you can call with their Application. The idea is like Skype where you have your buddies you double click on…
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