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Last modified: 16-06-2020 |
Free for six users or less
Axon SIP Server (free, for Windows)
http://brekeke.com (free and/or commercial; Java-based)
http://www.ntwizards.net/2004/09/07/asterisk_for_windows
"Pour la passerelle PSTN, je te deconseille fortement l'utilisation d'une ligne analogique, la qualite est acceptable, surtout avec une TDM400P, mais tout ce qui est informations sur l'appel est vraiment tres loin d'etre fiable (CalledID, Call Progress etc.)."
2/ Arf, evite la x100p et desactive l'annulateur d'echo de la Freebox qui est une merde (prefix annulateur: 3699). Active l'annulateur d'echo au niveau d'asterisk.
4/ Si tu veut utiliser un tel/fax, utilise une TDM400p avec 1FXS vers ton fax. Evite la conversion analogique/digital/analogique."
"I've had good luck with the Linksys SPA-841 phone and the Grandstream Budgetone 101. The latter is at the low end of the curve and it does take some careful research to find which version of the firmware is stable. There are quite a few more expensive > $100 phones which are well out of my budget. Poke around www.voip-info.org tons of information on sip phones."
"Quand a la qualite des telephones, cela depends de votre utilsation. Perso, je ne jure plus que par les CISCO 7940 et 7960, la qualite sonore est EXCELENTE, ils ont toutes les fonctions que l'on attend d'un telephone d'entreprise et ils sont blinde. C'est le genre de telephone ideale pour faire des conferences : tout confort a tous les niveaux."
"SIP Registrar and Proxy Server" ? "There are several SIP implementations that are OSS, but they are primarily what are known as "call proxies" instead of more full-featured PBX applications. This means that they function only to connect two endpoints together, and are basically just large, fast, directory servers. Examples of SIP Proxies are ser and Vocal. There are also other open source PBX projects like Bayonne and OpenPBX, which have slightly different feature sets than Asterisk."
http://www.nch.com.au/skypetosip/index.html
http://www.nch.com.au/soundtap/index.html
http://www.nchsoftware.com/screen/index.html
http://www.nch.com.au/recordpad/index.html
http://www.nch.com.au/conference/index.html
http://www.nch.com.au/oi/index.html
http://www.nch.com.au/ivm/index.html
http://www.nch.com.au/talk/index.html
http://www.nch.com.au/pbx/index.html
Overview of SIP and Networking Issues
http://nch.invisionzone.com/index.php?showtopic=737
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SIP Firewalls
http://www.intertex.se/index2.asp?iMenuID=279&iItemID=223
Enabling NATs and Firewalls With SIP
http://www.intertex.se/index2.asp?iMenuID=283&iItemID=227
Frequently Asked Questions (FAQ)
http://www.sipswitch.net/default.asp?pg=5
SIP, Nat Traversal and Firewall Issues
http://www.tmcnet.com/sip/0506/sip-feature-article-nat-0506.htm
Enabling NATs And Firewalls With SIP
http://www.tmcnet.com/it/0503/0503sip.htm
STUN over symmetric NAT, how come it works for me?
http://www.voipuser.org/forum_topic_4339.html
NAT Traversal for Multimedia over IP Services - White Paper
http://www.newport-networks.com/whitepapers/nat-traversal1.html
STUN
SIP, la révolution Internet dans la téléphonie
http://www.figer.com/Publications/Sip.htm
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Detection du Hangup : TDM400P
http://www.asterisk-france.net/forum/archive/index.php/t-1755.html
http://www.voip-info.org/tiki-index.php?page=AstWind
http://www.freephonie.org/doku/tutoriel:asterisk
http://www.voipsolutions.be/product_info.php/cPath/57_58/products_id/113/tdm01b-:-1-fxo-port.html
http://www.widevoip.com/shop/product_info.php?products_id=122