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SPA-2100

The SPA2102 replaces the SPA-2100, and now includes dual 10/100Mbps Ethernet, and support for two concurrent G.729 calls. The SPA-2100 with firmware > 2.0.4 does support two concurrent G.729 calls too. The Sipura SPA-2100 is a combination NAT…

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Asterisk phone Mitel 5220

Mitel 5220 / 5215 phones in SIP mode – General notes The Mitel 5220 Phone is a full-featured, standards-based business telephone. The original 5220 only supported Mitel’s Proprietary MiNet protocol, but the later Dual Mode versions can be…

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Asterisk tips MythTV integration

The open source PVR application suite named MythTV supports on-screen caller ID notifications. The original source of the caller ID information was from an attached modem, but the application structure allows any source to broadcast the caller ID…

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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…

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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…

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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…

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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…

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Asterisk sound files

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…

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Asterisk config sip.conf

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…

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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…

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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…

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Asterisk tips 911

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 =>…

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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…

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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…

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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,…

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