Asterisk settings Net2phone
Posted: August 2, 2005 This is from a working config used to call out to net2phone using their SIP serverThe only trick is they restrict access to specific user agents, so you must falsify Asterisks user agent string. sip.conf[general]useragent = X-Lite release…
Read More »Asterisk PSTN interface debugging
Posted: August 2, 2005 When interfacing between Asterisk and the PSTN many problems can arise, you can debug your PSTN interface. Echo Signal Level ztmonitor <channel number> -v gives you a visual representation of the sound strengths and makes it easy to…
Read More »Local Number Portability
Posted: August 2, 2005 What is LNP? What is LNP (Local Number Portability) or Number Portability Answer: LNP or Number portability Porting a DID number means that you can leave your current carrier, and move your number to another carrier, that has…
Read More »ITU G.726
Posted: August 2, 2005 G.726 is an ITU standard codec. This codec uses the Adaptive Differential Pulse Code Modulation (ADPCM) scheme. Like G.711, G.726 has its roots in the PSTN network. It is primarily used for international trunks to save bandwidth. Where…
Read More »Asterisk Bounty Outlook Integration
Posted: August 2, 2005 Contributions Darren Wright: $2000 USD Jason Sjobeck: 25USD … I know … I know … paltry … but I need to get some more buy-in then I’ll up it. Chris Blackburn: $500 USD your name here $0 your…
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 »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 »
Phone System 