Author: VoIP Info
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 »Budgetone
Posted: August 2, 2005 Grandstream SIP Phones: BudgeTone and HandyTone Grandstream makes several models of the Budgetone phone and Handytone ATA All of them support SIP VOIP. Grandstream lists several models of phones. The original group is the BT-101, BT-102 and the…
Read More »Asterisk sip.conf, peer definition: canreinvite option
Posted: August 2, 2005 Versions Migration from Asterisk 1.2 to 1.4: The “canreinvite” option has changed. canreinvite=yes used to disable re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat to disable re-invites when NAT=yes. This is propably what you…
Read More »SCCP
Posted: August 1, 2005 SS7 SCCP (Signalling Connection Control Part) see: SCCP Tutorial SCCP offers enhancements to MTP level 3 SS7 – ITU-SCCP protocol SS7 Asterisk and Cisco SCCP (Cisco Skinny Client Control Protocol) Proprietary protocol used between Cisco Call Manager and…
Read More »Asterisk cmd Rpt
Posted: August 1, 2005 Rpt() Synopsis Rpt(NODE) -OR- Rpt(Options) Options are as follows: Not specifying an option and specifying a node puts it in normal endpoint mode (where source IP and nodename are verified). Rannounce-string[|timeout[|timeout-destination]] – Amateur Radio Reverse Autopatch. The caller…
Read More »Asterisk Security
Posted: August 1, 2005 Page Contents Integrated Security SecAst Fail2Ban Perimeter Security Hardware Firewall IPtables Port Knock PBX Configuration Strong Passwords Default Context sip.conf configuration iax.conf Dialplan Custom modification to chan_sip.c Logs, CDR References Articles See also If you are looking to…
Read More »Firefly
Posted: August 1, 2005 Firefly Softphone Firefly is Freshtel’s IAX-based softphone for Windows, which works with Freshtel’s VoIP service. It is based on Virbiage‘s Cubix softphone platform. Firefly is known to work under Wine on Linux, although changes to font settings and/or…
Read More »Asterisk RPM
Posted: August 1, 2005 Package candidates for Fedora Extras Special Interest group for VoIP packages under Fedora Extras, including asterisk https://fedoraproject.org/wiki/Extras/SIGs/VoIP Asterisk RPMS for Fedora Core 5/6 and RHEL4 (with openh323 and zaptel xen kernel modules – not tested) Also available at…
Read More »Asterisk-mirrors
Posted: August 1, 2005 Mirrors for downloading Asterisk tarballs Americas http://redhat.theplanet.com/asterisk/, U.S., Dallas, Texas (unofficial mirror, but very fast!) – Tim left TP, moving mirror to other servers! Europe Australia SVN You can also use that page for instructions to download Asterisk…
Read More »SJphone
Posted: August 1, 2005 SJphone is a softphone that allows you to speak over the Internet using any desktops, notebooks, PDAs, stand-alone IP phones, and even any traditional landline or mobile phones. It supports both SIP and H.323 industry open standards and…
Read More »Asterisk phone sjphone
Posted: August 1, 2005 How to configure Asterisk for the SJphone Asterisk configuration In this sample configuration, 192.168.0.1 represents the Asterisk server and 192.168.0.2 is the client running SJPhone. Configure Asterisk to accept registration and inbound calls in sip.conf like this: [mysjphone]…
Read More »Asterisk installation tips
Posted: August 1, 2005 Asterisk install guide Before You Install Consult the planning and dimensioning checklist if you are interested running a larger PBX system, and you are concerned about hardware and software capabilities. Operating systems: Asterisk runs best on Linux systems,…
Read More »Asterisk indications Germany
Posted: August 1, 2005 Asterisk indications for Germany Insert this into indications.conf and set the Asterisk cmd SetLanguage to de. [de] http //www.teltone.com/prodmanuals/TLE Telephone Line Emulator, Rev M.pdf, Page 54 http //www.hettronic.de/hettronic/computer/hardware/isdn/ta2ab/ description = Germany ringcadance = 1000,4000 ; W hlton dial…
Read More »ITU G.711
Posted: August 1, 2005 G.711 is a high bit rate (64 Kbps) ITU standard codec. It is the native language of the modern digital telephone network. Although formally standardised in 1988, the G.711 PCM codec is the granddaddy of digital telephony. Invented…
Read More »AreskiCC CallingCard Application The idiots guide
Posted: July 31, 2005 IMPORTANT : The new version of AreskiCC have changed his name -> “A2Billing” Please for a followup of the project go to the appropriate pages: WIKI : https://www.voip-info.org/a2billing/ The idiots guide to installing AreskiCC CallingCard Application This guide…
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