VoIP Blog - Page 791
Asterisk RealTime IAX
iax.conf Setup You can keep any iax users in the flatfile AND use RealTime. How cool is that? Extconfig.conf Setup Add the following line, swapping your own personal values if you wish: iaxusers => mysql,asterisk,iax_buddies iaxpeers => mysql,asterisk,iax_buddies…
Read More »unlimitel.ca
Unlimitel Unlimitel offers 911 emergency services for all their customers and is now compliant with CRTC ruling on VoIP 911 calls. MichaelRichardson got their wholesale DID service to connect to my * pbX. The quality is good, and…
Read More »les.net
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
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 »How To Debug and Troubleshoot VOIP
(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
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 »VoIP User Groups USA
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
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
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
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
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 »HT-496
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
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…
Read More »Grandstream HandyTone 286
Lines/SIP Accounts 1 lines/1 SIP account Protocol Support SIP 2.0 (RFC 3261) TCP/UDP/IP RTP/RTCP HTTP ARP/RARP ICMP DNS DHCP NTP TFTP PPPoE protocols Feature Keys 1 button LAN Interface RJ-45 10 Mbps Device Management Web interface or via…
Read More »Asterisk Developer Guidelines
Disclaimer From the bug tracking page: Patch format The following are not official… These are place-holder suggestions! -JP It is important that patches be easily applied and maintained so that people can test your code, and it can…
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