Voice Over IP (VoIP)

PhonefromHere.com Widget

VoIP Users Conference - Sat, 09/04/2010 - 06:50

No iframe support

Categories: Voice Over IP (VoIP)

HackingVoIP Hints

VoIP Users Conference - Sat, 09/04/2010 - 06:50

Clues for Jan 15th Hacking VoIP

#HackingVoip For an attack on VoIP to be possible, only one side of the conversation needs to be using VoIP

#HackingVoip The use of cleartext protocols, the lack of proper authentication, and the complexity of deploying strong end-to-end security

#HackingVoip Listening to a voicemail system using insecure VoIP phones allows any person on the local segment to listen as well

#HackingVoip Insecure wireless access points and insecure VoIP technology can allow [anyone] to listen to your phones calls

#HackingVoip Organizations limit the spread of sensitive user information across their data networks. Voice networks using IP should, too

#HackingVoip IAX is the one protocol that does both session setup and media transfer

#HackingVoip Once the session is set up using SIP or H.323, the call is sent to the media protocol, which is RTP

#HackingVoip SIP is designed similar to HTTP, where methods like REGISTER, INVITE, FORWARD, LOOKUP, and BYE are used to set up call

#HackingVoip H.323 uses a collection of subprotocols, such as H.225, H.245, H.450, H.239, and H.460, to perform the session setup

#HackingVoip IAX does not use RTP for media transfer because the support is built into the protocol itself

#HackingVoip Usually digital phones are in business environments, analog in home environments. Neither are VoIP hard phones.

#HackingVoip SIP hard/soft phones are usually know as User Agents, while H.323 hard/soft phones are usually referred to as endpoints

#HackingVoip The authentication process in most VoIP deployment occurs at the session layer (SIP, H323, IAX).

#HackingVoip The most common default authentication for SIP is Digest authentication.

#HackingVoip When two phones are calling each other, they authenticate not to each other but to intermediate support servers.

#HackingVoip MAC addresses are sometimes used to authorize certain devices on VoIP networks.

#HackingVoip Encrypting VoIP traffic in both segments is often required. Authentication in SIP (signaling), audio in media (RTP)

#HackingVoip SIP usually listens on TCP or UDP port5060, but it can be configured to any port desired.

#HackingVoip Network port scanners can be used to enumerate SIP User Agents, Registrars, Proxy servers, and other SIP-enabled systems.

#HackingVoip ability to spoof a legitimate gatekeeper, Registrar, Proxy server, or other VoIP authentication entity can be quite harmful

#HackingVoip An attacker can monitor the network simply force a reboot by performing a DoS attack on the endpoint

#HackingVoip Infrastructure immune to users sniffing on the network or security attacks on TFTP, DNS, and DHCP is desperately needed

Categories: Voice Over IP (VoIP)

Skype Introduces 10-Way Video Calling

Asterisk VoIP News - Fri, 09/03/2010 - 16:44

Skype — apparently pleased with its five-way beta group video-calling functionality — has just released a new version of Skype 5.0 for Windows that doubles group support. It now allows for up to 10 video callers.Skype 5.0 beta two is... Dal http://www.asteriskvoipnews.com/

Categories: Voice Over IP (VoIP)

HD Voice with Andy Abramson

VoIP Users Conference - Thu, 09/02/2010 - 19:00

“High Definition Voice” on phone conversations is something you may not notice until you have experienced either very bad or very good quality. Andy Abramson of VoIP Watch will be joining our discussion of HD rollouts by [a carrier whose name begins with O] and the ramifications and challenges as he expressed them in his recent article.

Categories: Voice Over IP (VoIP)

Crestron offers new MTX-3 wireless touchpanel remote

Asterisk VoIP News - Thu, 09/02/2010 - 18:38

The MTX-3 offers seamless interaction with AV and environmental systems, providing true feedback of all settings, and displaying metadata information for all digital media.  Crestron’s infiNET EX wireless technology provides reliable two-way communications throughout a residence or commercial structure utilising... Dal http://www.asteriskvoipnews.com/

Categories: Voice Over IP (VoIP)

Cisco making a play for Skype?

Asterisk VoIP News - Thu, 09/02/2010 - 18:36

 Cisco is reportedly looking to buy Skype before the Internet phone provider goes public.  The blog TechCrunch posted over the weekend that Cisco made an offer for Skype before it completed its IPO process. The site attributed the unconfirmed... Dal http://www.asteriskvoipnews.com/

Categories: Voice Over IP (VoIP)

VUC After Hours

VoIP Users Conference - Thu, 09/02/2010 - 18:00

Actually, it was before hours, but who’s counting? Talk about Apple TV, Cisco wanting to buy Skype, Skype for Business (was SIP for Skype), bluetooth headset woes and much more.

SIP for Skype is now called “Skype Connect”. We’re wondering about the cost of the service. At one point I heard there would be a base charge for Skype Manager and then a charge for each user, plus a per channel charge. If this is the case, I can only say “Get a Life, Skype!”

Categories: Voice Over IP (VoIP)

Freeswitch Today

VoIP Users Conference - Fri, 08/27/2010 - 06:00

Some intense examination of the requirement to tell people you are recording a call with the regulars while waiting for FS bridge to successfully UNmute it self. Note that ZipDX implemented the warning when you connect to the bridge if recording is on. Some reminders about *6 to toggle mute state.

The origin of “Look to your skies for a warning! Keep watching the skies!” Naw, I cut that part but I wanted to add the link in this post, because I didn’t get more info from FS about what to put here. Great movie, “The Thing from Another World”, featuring James Arness (“Gunsmoke”) as “The Thing”. One of the best science fiction pictures from the fifties, and one that helped define the genre.

From yet another world, (the Freeswitch bridge), Pacman sounds, and then belched forth the actual meat-space aural personna of the like of Brian West, Michael Collins, Anthony Minessale, Darren Schreiber and a few less-stellar voices…

From their site at Freeswitch.org: “Created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.

FreeSWITCH was originally designed and implemented by Anthony Minessale with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX.  The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis.”

Categories: Voice Over IP (VoIP)

Freeswitch After Hours

VoIP Users Conference - Thu, 08/26/2010 - 21:00

The final part of the Freeswitch sessions. The ever-efficient Karl Fife wrings yet more tidbits of useful information from whoever was left on the Freeswitch bridge. This would have been in the fourth or fifth hour.

Freeswitch book is here. The official title of the book is FreeSWITCH 1.0.6, however don’t let the number fool you. The information applies to both FreeSWITCH 1.0 and 1.2 branches.

Dave Michels‘ son Dustin plays a mean guitar. Here’s  Dustin Michels video.

Categories: Voice Over IP (VoIP)

Taiwan government authorities astounded at Intel WiMAX move

Asterisk VoIP News - Thu, 08/26/2010 - 18:26

Intel’s announcement regarding its support on the WiMAX technology became very popular as the company’s decision affected the concerns of many people. According to the Taipei Times (one of the leading newspapers in Taiwan), a government official said that the... Dal http://www.asteriskvoipnews.com/

Categories: Voice Over IP (VoIP)

Russell Bryant Q & A (and more)

VoIP Users Conference - Thu, 08/19/2010 - 18:00

With everything up in the air, we weren’t sure they’d be a session, but I arrived back in time, so I should be there to talk about whatever anyone brings up.

By a bit of good luck, Russell Bryant stopped by and shared a lot of interesting details and answered question about what’s new with Asterisk. Russell is Engineering Manager for the Open Source Software team.

We later got into talking about the 3G used, unlocked and jailbroken iPhone I bought in the USA and broght back to Europe without knowing how it would work. I was saying that in California, the iPhone showed my location to withjin feet without data or wifi connection, as long as the map was already loaded (using wifi at home). That brought us to GPS and how it works. Here in France, the location it finds is much less accurate.

Categories: Voice Over IP (VoIP)

Going Nowhere Fast

VoIP Users Conference - Thu, 08/19/2010 - 18:00

Extended discussion of GPS and how it works. Triangulation or strangulation?

Categories: Voice Over IP (VoIP)

Chris Veazey of Blink Mind On SIP Video Calling

VoIP Users Conference - Thu, 08/12/2010 - 19:00

This coming Friday, August 13th will be yet another Voip Users Conference double-header. Starting at our usual 12 Noon EDT we have an overview of the Freetalk Connect SMB PBX featuring Skype integration.

Then immediately following, at 1pm EDT, we have Chris Veazey, VP Engineering of Blink Mind, to discuss the current state of the industry with regard to multi-media phones and SIP-based video calling.

Blink Mind has agreed to provide a video conference bridge (MCU) as part of their appearance. Further, as Blink Mind are a Polycom partner, Polycom has graciously provided a pair of Polycom VVX-1500 Business Media Phones on loan. One VVX is at Randy’s location in California, and the other in my home office.

As we aim to explore the capabilities of the VVX-1500 and similar desk phones, the video portion of the Blink Mind call with be limited to CIF (352 x 240) resolution using H.264 compression.

The Blink Mind bridge will allow compatible connections from a variety of clients. If you have access to a Polycom VVX-1500 or Grandstream GXV series phone you will be able to join the video bridge. Also, many larger video conference systems will drop down to CIF resolution, so they too can join the bridge.

If you don’t have access to such hardware you may be able to use a compatible soft phone. Rudimentary testing with the Mirial soft phone last Friday proved successful. Mirial is available for Windows and OSX. Mirial is a commercial product, but a 30 day trail download is available.

Since we have a relatively low-resolution video stream this time we’ll be issuing two kinds of connect details for the Blink Mind bridge. Yes, there will be two different SIP URIs for access to the Blink Mind bridge:

• Some will be able to join in a fashion that allows them to participate in the video stream.
• Others will join the bridge using a non-participatory SIP URI that gives them a view of the video stream but does not add their video to the MCU feed.

In this manner we will avoid having a 16+ way split where everyone looks like a tiny animated smudge on-screen.

If you have a suitable hardware based means of joining the video conference please let me know. While I hate to be arbitrary, there are a limited number of participatory seats available and preference will be given to those who have suitable hardware.

The details of the non-participatory connection and the web stream will be announced to the VUC mailing list early in the week before the call. We’ll also announce a SIP URI for anyone who needs to test that their end-point is capable of joining the Blink Mind video bridge.

Of course, we will join the ZipDX wideband audio bridge to the Blink Mind video bridge. A recording of the video stream will also be available after the fact, along with the regular podcast.

If you don’t have a webcam or suitable soft phone you can still view the video stream. Blink Mind will be providing a Flash-based streaming feed of the call. Since the stream is post-processed it’s a few second delayed from the bridge, but it includes the audio and video. This is a “view only” feed so you will not be able to participate in the call if this is your only means of connection.

If you are joined to the web stream and the ZipDX bridge we’d asked you to be very careful. The audio will be out of sync between these two bridges and it could become very confused for everyone if you allow the delayed audio into the ZipDX conference.

Of course, we will join the ZipDX wideband audio bridge to the Blink Mind video bridge. A recording of the video stream will also be available after the fact, along with the regular podcast

Watch the VUC mailing list over the coming few days. As more details get locked down things will be announced there.

Update: In an earlier version of this post I referred to Chris Veazey as CTO of Blink Mind, which was in error. In fact we will be joined by three people from Blink Mind:

• Joe Baird, CEO
• Nathan Stratton, CTO
• Chris Veazey, VP Engineering

Categories: Voice Over IP (VoIP)

Skype to Business with FREETALK Connect

VoIP Users Conference - Thu, 08/12/2010 - 18:00

Freetalk Connect wants to put Skype on every phone in your office.
Designed for offices with between two and 50 users, the FREETALK Connect offers  unified communications functionality, including Find Me, Follow Me; a unified voice mailbox; automated attendant and auto call distribution.

Intelligent routing capabilities:  incoming Skype calls, as well as calls over SIP, the PSTN and IAX2, can be routed by the FREETALK Connect to any local or remote Skype user, SIP endpoint, analog or mobile phone.

Set up is simplified, enabling small business users that are not tech savvy to use it quickly and simply, without formal training. Supported telephones are plugged into the company’s network and the device auto-detects and configures them. An on-screen wizard then guides the customer through a few business-related questions that help configure the company’s communication system and enables Skype calling from every supported desktop phone in the office. Adding users and administering the system after installation is just as easy.

Categories: Voice Over IP (VoIP)

FCC's Closed-Door Net Neutrality Meetings Break Down

Asterisk VoIP News - Mon, 08/09/2010 - 17:37

The Federal Communications Commission (FCC) has called off its closed-door meetings with various tech companies after they failed to reach common ground on net neutrality. "We have called off this round of stakeholder discussions. It has been productive on several... Dal http://www.asteriskvoipnews.com/

Categories: Voice Over IP (VoIP)

5 Ways Small Businesses Can Create Happier Customers

VoIP News - Mon, 08/09/2010 - 11:55

On-Demand Webinar >     Watch Now!SPONSOR: AvayaWHAT:Hear about what steps are necessary to boost customer loyalty to make a positive impact on your business without letting other im...

Categories: Voice Over IP (VoIP)

New Allison Smith Collection of Funny Prompts

VoIP Users Conference - Sun, 08/08/2010 - 14:16

Allison Smith (@VoiceGal) pinged me about the collection she recorded for Joey Lindstrom for use with Asterisk pbx. The full set, which Mr. Lindstrom has generously decided to place in the public domain is available for dowload free here:
http://vuc.li/FunnyAllison.

Joey wrote to say this was his way to give something back to the community, and he’s asking Digium to put these prompts in the Asterisk distribution should they wish to do so.

Categories: Voice Over IP (VoIP)

ClueCon and 2600hz Project

VoIP Users Conference - Thu, 08/05/2010 - 19:00

2600hz is home to a collection of open-source telephony software that enables the use of the FreeSWITCH, Asterisk and YATE switching libraries. Initially built around the blue.box project, we aim to provide a collection of software to power your GUI, your cloud-based telephony switch and/or your monitoring and maintenance tool set.

Video testing – Next week we’ll be doing a real video conference, watch the site for more info

ClueCon - No one wished to comment. Whatever.

Categories: Voice Over IP (VoIP)

Meet The 2600hz Project, The New Sound of Open Source Telephony

Asterisk VoIP News - Tue, 08/03/2010 - 18:11

I really miss my Bluebox and the days then that would work.  Gotta love the $5 rRdbox from hallmark or the infamous Blotto Box that was no joke at all.  We salute you Captain Crunch!  Gigaom - Some of the... Dal http://www.asteriskvoipnews.com/

Categories: Voice Over IP (VoIP)

OpenVBX from Twilio

VoIP Users Conference - Thu, 07/29/2010 - 19:00

OpenVBX is a web-based open source phone system for business.
With a Twilio account and a web server with PHP 5.2+ and MySQL 5, you can build your own “hosted pbx”.

Build your own custom phone applets with just a little bit of PHP. Rebrand and resell OpenVBX to your customers.

Give every user their own phone number and personal conference line. Dial whole departments, share voicemail messages with the team. OpenVBX is for companies and collaboration.

Adam Ballai, lead engineer on the OpenVBX project and Twilio CEO Jeff Lawson are our guests.

Categories: Voice Over IP (VoIP)