Author: VoIP Info
les.net
Posted: September 13, 2005 LES.NET (1996) INC. Toll-Free 1-888-VOIP-DID Winnipeg, Manitoba, Canada LES.NET Toll Free SIP LES.NET USA DID – West Coast LES.NET USA DID – Level3 LES.NET Canadian DID Contact Information: Toll-Free: 1-866-944-0009 In Winnipeg: 1-204-944-0009 Web Site: https://les.net Email: voip…
Read More »LES.NET Toll Free SIP
Posted: September 13, 2005 les.net LES.NET Toll-Free SIP Toll-Free 1-888-VOIP-DID Canada / USA-50 TollFree Numbers that ring to your Asterisk Box or SIP device. IAX Available as well. Choose from our Toll-Free Number Pool We have a large number of 800,888,877,666 numbers…
Read More »Asterisk sound files
Posted: September 13, 2005 The Asterisk CVS distribution includes a wide range of sound files, listed below and at Digium. In addition, there’s an additional set of sound files, managed by John Todd. News April 14th 2015: High Quality voice Prompts in…
Read More »Asterisk config sip.conf
Posted: September 13, 2005 Configuration file for Asterisk SIP channels, for both inbound and outbound calls. Starting with Asterisk v1.2.0: The global option “port” in 1.0.X that is used to set which port to bind to has been changed to “bindport” to…
Read More »How To Debug and Troubleshoot VOIP
Posted: September 13, 2005 (SIP, MGCP, H.323, RTP, Skinny etc.) One of the primary techniques is to view what is actually getting sent and received by VOIP devices. There are several ways to do this: Monitor Ethernet Traffic Debugging displays from a…
Read More »NAT and VOIP
Posted: September 13, 2005 What is NAT? NAT (Network Address Translation) is a technology most commonly used by firewalls and routers to allow multiple devices on a LAN with ‘private’ IP addresses to share a single public IP address. A private IP…
Read More »Asterisk tips 911
Posted: September 13, 2005 Priority for emergency calls This extension for 911 tries to dial out. If no channel is available on the Zap interface, one call is terminated, we wait one second and then dial out on the line. exten =>…
Read More »VoIP User Groups USA
Posted: September 12, 2005 Our mission is to promote the use of Asterisk PBX and related technologies. We meet at least once a month in a virtual environment (shared desktop for presentations and VoIP for all memebers). Membership is free regardless of…
Read More »Asterisk FAQ
Posted: September 12, 2005 Here are answers to common Asterisk questions from the asterisk-users mailing list. Please keep the answers short, referencing other pages for longer texts. General questions How do I search the Asterisk mailing list archives? See: Asterisk Mailing Lists…
Read More »Cisco Phone Headsets
Posted: September 12, 2005 Some Cisco phones don’t use standard headset connections Download VoIP Headset & Phone Compatibility guide Comments from the Asterisk Mailing list (and private email) for 7960 I use a Plantronics Supra H51 plugged straight into the headset port,…
Read More »zaptel-x86_64
Posted: September 12, 2005 This issue is only relevant to very old versions of Zaptel: fixed in 1.2 . I asked Digium sales support about compiling Zaptel on an AMD64 / x86_64 / SMP Opteron system prior to buying a card from…
Read More »Zaptel
Posted: September 12, 2005 Short for “Zapata Telephony” The Zaptel project has been renamed ‘DAHDI‘ as of May 19th 2008. Page Contents Zaptel Hardware Software See also Zaptel Zaptel refers to Jim Dixon’s open computer telephony hardware driver API. Zaptel drivers were…
Read More »Asterisk RealTime Extensions
Posted: September 12, 2005 extconfig.conf Setup Add the following line, swapping your own personal values if you wish: extensions => mysql,asterisk,extensions_table You can change mysql to odbc if you want to use odbc. You can change asterisk to be the name of…
Read More »HT-496
Posted: September 12, 2005 Grandstream HandyTone 496 Product Features Two FXS (RJ11) ports Two Ethernet (RJ45) ports Supports SIP 2.0(RFC 3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP (both client and server), NTP, PPPoE, STUN, TFTP, etc. Built-in router, NAT, Gateway and…
Read More »Unlimitel
Posted: September 11, 2005 THE INFORMATION BELOW IS OUTDATED. AS OF MARCH 1st 2018: 1.5cents/minute and a staggering $7.95/mth for a DID. Unlimitel provides a wholesale VoIP service. There is no fuss with this service. When you subscribe, you get an email…
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