SIP Blog - Page 4
SIP Express Router
Posted: September 22, 2005 SER – SIP Express Router Project Website: www.iptel.org/ser For a different version see: OpenSER On Nov 04, 2008, SIP Express Router (SER) and Kamailio (OpenSER) started the SIP Router Project SIP Express Router (ser) is a high-performance, configurable,…
Read More »SIP Video Phones
Posted: September 20, 2005 Hardware Phones: TANDBERG E20 – SIP VOIP and Video Phone Grandstream GXV3000 – Cheap H.264 SIP Videophone Grandstream GXV3240 IP Multimedia Phone Grandstream GXV3275 SIP Videophone – Voice and/or video H.264/H.263/H.263+ and embedded Skype, browsing etc too. fanvil…
Read More »Sipura Technology Inc.
Posted: September 3, 2005 Sipura was purchased by Linksys and the Sipura product line has been rebranded to Linksys Update Jan 2009: Linksys Voice products (what was Sipura) are now going to be rebranded as Cisco Makers of some of the most…
Read More »SIP mwi
Posted: August 25, 2005 Message Waiting notifications in SIP are done with the SIP method notify method. This is supported by: Aastra phones 480i, 9133i, 9112i, 480i CT Cordless Asterisk: The Open Source PBX Atcom phones AT-320 AudioCodes MediaPack FXS Gateways -draft-ietf-sipping-mwi-04.txt…
Read More »Sipura 3000
Posted: August 20, 2005 The SPA-3000 has been discontinued by Linksys, it has been replaced by the SPA-3102 (Same features but with added router functionality). The Sipura SPA-3000 has one FXS port and one FXO port. Both ports are fully controllable via…
Read More »SIPURA SPA-2000
Posted: August 20, 2005 Page Contents Sipura SPA-2000 devices Limited CPU means limited compression To upgrade the firmware via tftp Distinctive rings Custom Ring Cadences FXS impedence settings Documentation Notes See Also Sipura SPA-2000 devices Sipura Phone Adapters (SPA-2000) connect standard telephones…
Read More »SIP user agent identification
Posted: August 13, 2005 SIP user agent identification strings Every SIP user agent (phone, software, device) identifies itself with a string. The syntax of this string is not defined, but a common practise is: <vendor> <product> <firmware version> List user agent strings…
Read More »SIPphone
Posted: August 2, 2005 Website: SIPphone.com Service launched in 2004 by the founder of Lindows and MP3.com. Offers many unique and interesting features including a do-it-yourself conference bridge, and lots of diagnostic tools. These people seem to be on a mission to…
Read More »SIP SS7 gateways
Posted: July 26, 2005 SIP <-> SS7 Gateways SS7 (or C7) is the main signalling for PSTN interconnections. How do we interface between a SS7 network and a SIP network? SIP to SS7 Gateway from Terratel The signaling converter allows to combine…
Read More »SIP Info DTMF
Posted: June 17, 2005 The SIP INFO method can be used by SIP network elements to transmit digits out-of-band as telephone-events in a reliable manner independent of the media stream. The SIP INFO method contains a message body in addition to the…
Read More »sipX
Posted: June 2, 2005 sipX – The SIP PBX for Linux Original project: sipXecs Original Website: http://www.sipfoundry.org This project has also been forked and continues with the original developers under the sipXcom name. New project: http://www.sipxcom.org Wiki: http://wiki.sipxcom.org User Forums: https://groups.google.com/forum/#!forum/sipxcom-users See…
Read More »IAX versus SIP
Posted: June 1, 2005 Date: Mon, 5 Jul 2004 18:59:52 -0500 (CDT) From: Mark Spencer <[email protected]> Let me summarize some differences between SIP and IAX, and it might help you make a decision about what is best for you. 1) IAX is…
Read More »sipX Solution Summary
Posted: May 13, 2005 sipX – Easy to use, scalable, and feature rich open source unified communications and IP PBX solution for Linux SIPfoundry info: Original Website: http://www.sipfoundry.org Github: https://github.com/SIPfoundry sipXcom info: Website: http://www.sipxcom.org sipXcom Wiki: http://wiki.sipxcom.org Github: https://github.com/sipXcom sipX is a…
Read More »Asterisk sip tos
Posted: May 11, 2005 SIP.conf: [GENERAL] section: tos= Asterisk 1.2 Set to value or lowdelay throughput reliability mincost none Alternatively you can directly set the bit values: tos=0x18 ; where the tos bits are set individually and in combination For example tos=0x10…
Read More »SIP tools
Posted: May 10, 2005 Various SIP tools Sip debugging and testing SIPsak: The swiss army knife PlaySIP: Recording a SIP media stream VoIP Recorder: Monitor and Recording SIP media streams. http://www.pcbest.net SIP Server tools Load balancer: SIP Server load balancer SIP development…
Read More »
Phone System 