As usual NV (Nerd Vittles) has done it again. This time with Trixbox 1.2.3 the open source IPPBX. The newbies guide includes setting up of trixbox 1.2.3 (hey Vittles where is the Trixbox 2.0? just kidding) add FreePBX and a plethora of NV goodies. At the end you will have a fantastic VOIP IP Telephony solution, IPPBX based on TrixBox which is in turn based on ever popular Asterisk IPPBX.
Here are some goodies from NV;
The Game Plan.
Upgrading TrixBox from a Prior Version of Asterisk@Home.
Loading CentOS/4 and TrixBox 1.2.3
Securing Your Passwords.
Securing and Activating A2Billing
Securing SugarCRM Contact Management.
If It Ain't Broke, Don't Fix It!
Upgrading TrixBox to Support MailCall.
Updating freePBX to the Latest Release.
Activating Bluetooth Support. HAH!
Activating Apache HTTPS Support.
IP Configuration for Asterisk.
Designing Your PBX System.
Initial Setup of freePBX.
Configuring freePBX Trunks.
Well so far I have reached only the middle of the article. I guess you head over to NV and digest the whole article.
He has a tip for windows Trixbox setup! Look for this info in the article.
HOT TIP: For a turnkey version of TrixBox 1.2.3 that runs on your Windows desktop and includes the entire setup we'll be discussing as well as an out-of-the-box setup for 10 extensions and two VoIP providers.
Well done NV, I will have mine without the wine, I am already drunk with info!.
Tuesday, October 31, 2006
As usual NV (Nerd Vittles) has done it again. This time with Trixbox 1.2.3 the open source IPPBX. The newbies guide includes setting up of trixbox 1.2.3 (hey Vittles where is the Trixbox 2.0? just kidding) add FreePBX and a plethora of NV goodies. At the end you will have a fantastic VOIP IP Telephony solution, IPPBX based on TrixBox which is in turn based on ever popular Asterisk IPPBX.
Monday, October 30, 2006
The book, "Asterisk The Future Of Telephony", Keeping up with the free as in beer philosophy, O'Reilly (released under the Creative Commons license) and available for download, read and use it in your Asterisk VOIP IP Telephony projects. I have dealt with Asterisk long before this book was written and I wish I had something like this handy at hand during all those experiments in VOIP. This book represents the work of Jim Van Meggelen, Jared Smith, and Leif Madsen over the past two years.
If you are even a bit interested in VOIP IP Telephony, I think you should add this to your arsenal of resources. Although it is written for Asterisk IPPBX, it covers most of the daya to day VOIP, IPPBX, IP Telephony, setup, NAT traversal, VOIP Codecs and VOIP protocol setups etc. Remember Asterisk IPPBX encompasses more than 100 years (VOIP) experiences due it's Open Source nature.
From the Preface of the book;
This is a book for anyone who is new to Asterisk™.
Asterisk is an open source, converged telephony platform, which is designed primarily to run on Linux. Asterisk combines over 100 years of telephony knowledge into a robust suite of tightly integrated telecommunications applications. The power of Asterisk lies in its customizable nature, complemented by unmatched standards compliance. No other PBX can be deployed in so many creative ways. Applications such as voicemail, hosted conferencing, call queuing and agents, music on hold, and call parking are all standard features built right into the software. Moreover, Asterisk can integrate with other business technologies in ways that closed, proprietary PBXs can scarcely dream of. Asterisk can appear quite daunting and complex to a new user, which is why documentation is so important to its growth. Documentation lowers the barrier to entry and helps people contemplate the possibilities. Produced with the generous support of O’Reilly Media, Asterisk: The Future of Telephony was inspired by the work started by the Asterisk Documentation Project. We have come a long way, and this book is the realization of a desire to deliver documentation which introduces the most fundamental elements of Asterisk-the things someone new to Asterisk needs to know. It is the first volume in what we are certain will become a huge library of knowledge relating to Asterisk.
This book was written for, and by, the Asterisk community.
Asterisk Documentation Project (One good resource for Asterisk, Please Join)
O'Reilly Asterisk (you can get updates errata etc from here)
To further interest you to check before you download here is the
TABLE of CONTENTS
1. A Telephony Revolution
VoIP: Bridging the Gap Between Traditional Telephony and Network Telephony
Massive Change Requires Flexible Technology
Asterisk: The Hacker's PBX
Asterisk: The Professional's PBX
The Asterisk Community
The Business Case
2. Preparing a System for Asterisk
Server Hardware Selection
Types of Phone
3. Installing Asterisk
What Packages Do I Need?
Obtaining the Source Code
Installing Additional Prompts
Updating Your Source Code
Common Compiling Issues
Loading Zaptel Modules
Directories Used by Asterisk
4. Initial Configuration of Asterisk
What Do I Really Need?
Working with Interface Configuration Files
FXO and FXS Channels
Configuring an FXO Channel
Configuring an FXS Channel
Configuring Inbound IAX Connections
Configuring Outbound IAX Connections
5. Dialplan Basics
A Simple Dialplan
Adding Logic to the Dialplan
6. More Dialplan Concepts
Expressions and Variable Manipulation
Using the Asterisk Database (AstDB)
Handy Asterisk Features
7. Understanding Telephony
The Digital Circuit-Switched Telephone Network
8. Protocols for VoIP
The Need for VoIP Protocols
Quality of Service
Asterisk and VoIP
9. The Asterisk Gateway Interface (AGI)
Fundamentals of AGI Communication
Writing AGI Scripts in Perl
Creating AGI Scripts in PHP
Writing AGI Scripts in Python
Debugging in AGI
10. Asterisk for the Über-Geek
Call Detail Recording
Customizing System Prompts
11. Asterisk: The Future of Telephony
The Problems with Traditional Telephony
The Promise of Open Source Telephony
The Future of Asterisk
A. VoIP Channels
B. Application Reference
C. AGI Reference
D. Configuration Files
E. Asterisk Command-Line Interface Reference
Posted by ravenII at 10/30/2006 08:44:00 AM
Saturday, October 28, 2006
I was surfing the web searching for information on Skype technology and if it affects users as popular beliefs on the net and else where. Specially in the case of Supernodes and relaying nodes, mentioned in the article;
VOIP IP Telephony: How to be or not to be a skype supernode?
I did find some very good information. One of the studies (the best in my view) were done by Saikat Guha of Cornell University and Neil Daswani, Ravi Jain of Google. The study was conducted over more than four months extending from September 1, 2005 to January 14, 2006. Also one factor that they state is that, Experiments on this data were done in a black-box manner, i.e., without knowing the internals or specifics of the Skype system or messages, as Skype encrypts all user traffic and signaling traffic payloads. The results indicate that although the structure of the Skype system appears to be similar to other P2P systems, particularly KaZaA, there are several significant differences in traffic. The number of active clients shows diurnal and work-week behavior, correlating with normal working hours regardless of geography. The population of supernodes in the system tends to be relatively stable; thus node churn, a significant concern in other systems, seems less problematic in Skype. The typical bandwidth load on a supernode is relatively low, even if the supernode is relaying VoIP traffic.
So is Skype good or bad? well you have to take all the facts and decide for your self.
Despite its popularity, little is known about Skype's encrypted protocols and proprietary network. Garfinkel (His site has a wealth of information but the related article was found on another site Perhaps why the original authors just cited the article, I have uploaded the article to iptelephony at google groups.), concludes that Skype is related to KaZaA; both the companies were founded by the same individuals, there is an overlap of technical staff, and that much of the technology in Skype was originally developed for KaZaA. Network packet level analysis of KaZaA and of Skype (and again the article was uploaded to iptelephony at google groups) support this claim by uncovering striking similarities in their connection setup, and their use of a ``supernode''-based hierarchical peer-to-peer network.
Supernode-based peer-to-peer networks organize participants into two layers: supernodes, and ordinary nodes. Such networks have been the subject of recent research. Typically, supernodes maintain an overlay network among themselves, while ordinary nodes pick one (or a small number of) supernodes to associate with; supernodes also function as ordinary nodes and are elected from amongst them based on some criteria. Ordinary nodes issue queries through the supernode(s) they are associated with.
Here are the two experiments;
Basic operation. We conducted an initial experiment to examine the basic operation and design of the Skype network in some more detail. We ran two Skype clients (version 126.96.36.199 for Linux) on separate hosts, and observed the destination and source IP addresses for packets sent and received in response to various application-level tasks. We observed that in Skype, ordinary nodes send control traffic including availability information, instant messages, and requests for VoIP and file-transfer sessions over the supernode peer-to-peer network. If the VoIP or file-transfer request is accepted, the Skype clients establish a direct connection between each other. To examine this further, we repeated the experiment for a single client behind a NAT2, and both clients behind different NATs. We observed that if one client is behind a NAT, Skype uses connection reversal whereby the node behind the NAT initiates the TCP/UDP media session regardless of which end requested the VoIP or file-transfer session. If both clients are behind NATs, Skype uses STUN-like NAT traversal to establish the direct connection. In the event that the direct connection fails, Skype falls back to a TURN-like approach where the media session is relayed by a publicly reachable supernode. This latter approach is invoked when NAT traversal fails, or a firewall blocks some Skype packets. Thus the overall mechanism that Skype employs to service VoIP and file transfer requests is quite robust to NAT and firewall reachability limitations.
Promotion to supernode. We next investigated how nodes are promoted to supernodes. In an experiment we conducted, we ran several Skype nodes in various environments and waited two weeks for them to become supernodes. A Skype node behind a saturated network uplink, and one behind a NAT, did not become supernodes, while a fresh install on a public host with a 10 Mbps connection to the Internet joined the supernode network within minutes. Consequently, it appears that Skype supernodes are chosen from nodes that have plenty of spare bandwidth, and are publicly reachable. This approach clearly favors the overall availability of the system. We did not test additional criteria such as a history of long session times, or low processing load as suggested in this article, p2p supernode. As we show later, the population of supernodes selected by Skype, apparently based on reachability and spare bandwidth, tends to be relatively stable. Skype, therefore, represents an interesting point in the P2P design-space where heterogeneity is leveraged to control churn, not just cope with it.
Well the article seems to be getting too long. I will have to make a second article to finish the rest. It will be interesting as you will be able to see all Skype info in visual format, i.e. graphs. I am sure it will be interesting.
Posted by ravenII at 10/28/2006 08:24:00 PM
Friday, October 27, 2006
The Rimax Mystic is a portable media player that handles MP3, MP4, and a handful of other audio and video files. It's got a 1.5-inch OLED screen, and an FM radio with 20 presets. So where is VOIP or IP Telephony? Don't worry, you are covered, Just connect it's USB to your PC, and you can make all your VOIP calls over this device.
Only problem is that you have stay tethered to your computer, it is not portable when making VOIP Calls.
Hope it will adopt the Mobile VOIP described here.
The tiny 1.5-inch screen, will not be your killer video player, but the small size means it'll probably fit comfortably in the hand when used as a phone, and of course as a portable music player.
Here are the Key features and benefits
512MB, 1GB or 2GB internal memory.
1.5” OLED Screen with 65K colors.
Light, slim player with elegant design.
Folder search system allowing you to enter up to 99 songs in to folders for easier and faster searches.
Doubles up as a VOIP Phone.
OSD menu in 14 languages.
Compatible Audio formats: MP1, MP2, MP3, WMA, WMV, WAV, ACT, WAV.
Compatible Video format: AMV.
Compatible Text format: TXT.
Compatible Picture formats: JPG, JPEG.
FM radio with space for up to 20 preset stations.
Recording from built-in microphone as a voice recorder or direct from radio.
Up to 8 hours of battery duration in play mode.
Polymer battery recharged via USB port.
Dimensions: 70mm x 41mm x 7mm.
Extra colored covers included
- Mystic 512MB - White and Orange
- Mystic 1GB & 2GB - Green, White and Orange
It will cost you around $100 for the 512MB version.
Posted by ravenII at 10/27/2006 09:07:00 PM
Thanks to EQO Communications, whose EQO Mobile application which supports RIM's BlackBerry handhelds and Windows Mobile-powered devices. Now you can try you VOIP calls from those devices. What is the feeling if you can catch another Skype user for chit chat, on your Blackberry!
EQO Mobile is an application that supports instant messaging via AIM, Google Talk, ICQ, Jabber, MSN, Skype, and Yahoo!, and the EQO Mobile application is available as a free download.
From EQO's press release,
"We're rapidly evolving toward a world of ubiquitous connectivity in which we will be connected to each other all the time, everywhere we go," said Jill Aldort, Wireless/Mobile senior analyst for Yankee Group. "One of the most important elements in achieving this vision is the ability to deliver an easy to use, communications service that can reside on a broad range of handsets."
You can find the Press release here
You also can download EQO Mobile from the same site.
Posted by ravenII at 10/27/2006 06:34:00 PM
Thursday, October 26, 2006
University of North Carolina's ITS department is making headways in providing location services for VOIP and Mobile VOIP services within the campus.
ITS has asked assistance from Intrado, Inc. in Longmont, Colo., for deployment of next generation 911 services
“911 for VoIP is a hot topic for VoIP carriers like Vonage right now,” said Patrick Brooks, lead developer for Telecom R&D’s Real Time Communications (RTC) project. “But our added component of mobility will likely put UNC ahead of the carriers.”
(Brooks is a member of the National Emergency Number Association VoIP Location Working Group, which is defining the architecture of the next generation national emergency services platform in the United States.)
Read more about the E911 services and more at UNC ITS site.
Posted by ravenII at 10/26/2006 10:46:00 AM
Wednesday, October 25, 2006
Fonality today announced Trixbox 2.0, the newest version of its easy to use open source telephony and application platform. Fonality acquired Trixbox, the open source IPPBX a few weeks ago. The last version of Trixbox was 1.2 before it was acquired by Fonality.
The Trixbox 2.0, under Fonality, which can be installed in less than an hour, provides the trixbox community with increased reliability and many new features including a point-and-click, web-based graphical user interface (GUI) that dramatically increases ease of use, streamlines installation and virtually eliminates the need for the command line. trixbox 2.0 expands the traditional open source LAMP (Linux, Apache, MySQL and PHP/PERL) stack to LAAMPS by providing Asterisk telephony and SugarCRM customer relationship management. The software also includes FreePBX, FOP and HUDlite. The beta version of trixbox 2.0 is available for immediate download from www.trixbox.org and the final version will be available within 30 days.
Now again, LAAMPS, the extended LAMP, stands for Linux, Apache, Asterisk, MySQL, PHP, SugerCRM should attract more Open source followers, Now which includes large and small business'.
MySQL, part of the LAAMPS stack in Trixbox 2.0, brings powerful database capabilities to the trixbox community. "As part of the Trixbox stack, MySQL provides a high-speed database foundation that improves uptime and enables easy integration with other business applications," said Marten Mickos, CEO of MYSQL AB. "MySQL database software is already being used by hundreds of thousands of trixbox users. We look forward to working closely with Fonality to grow and expand the thriving trixbox community."
I had my doubts when Fonality took Trixbox over, but now I am relieved to see that Fonality is actually helping Trixbox.
Welcome Trixbox 2.0!
Posted by ravenII at 10/25/2006 07:55:00 PM
Tuesday, October 24, 2006
A man holds a mobile phone during a presentation of the hybrid Unik telephone which offers, from a single number, standard fixed and mobile connections as well as VoiP using either WiFi or GSM networks at a news conference in Paris September 25, 2006. REUTERS/Benoit Tessier (FRANCE)
Posted by ravenII at 10/24/2006 10:08:00 AM
According to a news report by Business wire, VOXBONE, A worldwide Virtual Number provider, will support Asterisk IAX (IAX™ (Inter-Asterisk Exchange)) trunking. IAX is the native communication protocol for Asterisk. It is used to connect between Asterisk IPPBX server (server to server) and Server to Clients that support IAX.
According to Rodrigue Ullens, co-founder of Voxbone, "By adding support for IAX, we are responding to strong demand from customers. IAX is completely plug and play, and optimized for Asterisk platforms. It also offers integrated security and consumes less bandwidth while offering the same high quality as SIP. Customers now have a choice of two popular standards from which they can choose."
Voxbone leases international VoIP virtual phone numbers and worldwide origination services via VoIP to organizations in North and South America, Europe and Asia/Pacific regions.
Voxbone is the leading VoIP carrier providing centralized access to DID and toll-free numbers around the world. Headquartered in Brussels Belgium, it has expanded its VoIP backbone from Europe to America and recently to Asia. The Voxbone network covers more than 4,000 cities in roughly 50 countries and is expanding its coverage. The switchless architecture of the Voxbone network enables customers to realize the benefits of IP communications by rapidly deploying new services with local presence and simultaneously reducing costs.
ASTERISK, the Open source IPPBX
Voxbone, Virtual number Provider
Posted by ravenII at 10/24/2006 09:51:00 AM
In conjunction with VON Europe, OpenSER, the GPL SIP server, has announced the first OpenSER Summit, which will take place in Berlin, on the 8th November.
The registration is open and you are invited to attend. Also The participants registered via OpenSER will have free access to VON Europe summit and exhibition ;).
Here is the agenda for 8TH;
Wednesday, November 8, 2006
09:00-11:15 Morning session
9:00 - 10:00
1. Summit Opening
2. Agenda Overview
3. OpenSER History
Daniel-Constantin Mierla - co-founder and board member
10:00 - 11:15
1. "SIP Application Servers & weSIP for OpenSER"
Gines Gomez - VozTelecom Chief Innovation Officer
2. "High availability environments with geographic redundancy"
Benjamin Wolf - Basis AudioNet Project Director
3. "GetTru - bring VoIP to your mobile"
James Body - Truphone Director-Networks
4. "VoIPUser - community driven VoIP services"
Dean Elwood - VoIPUser co-founder
You can register for the event by following this link!
Posted by ravenII at 10/24/2006 01:46:00 AM
If you ever craved to have a presence support in OpenSER, you are in luck. OpenSer this week released a newly designed presence server in the development trunk.
It was released after a short period of beta testing and is part of the development trunk.
This is supposed to work with most of the SIP clients with presence extensions. But you should be aware that it will be some time before the code will include all SIMPLE extensions, for now OpenSER has the most common features of the presence extensions: presence info, watcher info and XCAP.
OpenSER would like you to test the code to help in further development and bug hunting. OpenSER has opened a Wiki for you to enter your findings and to provide feedback.
OpenSER the SIP router, for VOIP IP telephony applications, promises to give you one more feature soon, which will allow to publish presence info in behalf of different resources (e.g., user location).
Go get your presence.
These are the Wiki links;
Posted by ravenII at 10/24/2006 12:36:00 AM
Monday, October 23, 2006
A new player in the VOIP IP Telephony arena, released it's product today, EasyVoxBox is an open source, community based, business grade IP PBX phone system. It is based on Asterisk and FreePBX.
May be the Trixbox getting sponsored by a commercial vendor lead to the birth of this product. According to the news release by the EasyVoxBox, they have been preparing this system for a while. But I see that this product will fill the gap, if Trixbox will be tricked out by anyone.
According to the product information, the project is hosted at Sourceforge, under the name easyvoxbox.
The Version 0.001 has following base packages in the released ISO;
Asterisk Recording Interface
Flash Operator Panel
MeetMe Control Center
I am happy to see another product based on my favorite Linux distro, CentOS.
Welcome to the VOIP world, EasyVoxBox!
Posted by ravenII at 10/23/2006 08:13:00 PM
Sunday, October 22, 2006
The “ PacketSmart for Asterisk ” solution has three distinct functions:
* Network assessment for VoIP Ip Telephony
* VoIP troubleshooting
* Ongoing VoIP SLA monitoring
During the deployment planning phase, Asterisk VARs can use the micro-appliances to simulate live VoIP traffic on the SMB network, to identify and fix problems before VoIP deployment. Since the micro-appliances can be connected in-line in a network, they can also provide visibility into the ongoing data patterns in the customer network. After Asterisk deployment, the Packet Island software agent that is left installed on the Asterisk PBX continues to collect detailed quality metrics that can be used by the SMB and VAR for SLA monitoring and to isolate and troubleshoot transient VoIP quality issues. For multi-site deployments with multiple Asterisk PBXs, the software agents can even be employed to do periodic network assessments to ensure that the inter-Asterisk connectivity continues to support good VoIP quality. With its specific focus on Asterisk, and a SaaS-based delivery model, PacketSmart for Asterisk is a simple, functionally rich, and cost effective solution for managing Asterisk-based VoIP deployments.
Read more about “PacketSmart for Asterisk” and other products at Packet Island
Posted by ravenII at 10/22/2006 10:01:00 PM
Skype the Internet Phone service seems to have done more than letting people talk free or at lower cost.
Skype seems to have brought many a couples together and ended up in marriages, Alana Semuels reports. Mark Passerby and Salwa Al-Saban were separated by the Atlantic Ocean, a time difference of seven hours and vast cultural contrasts. He lived in Lansing, Mich., she in Egypt's capital, Cairo.They say they fell in love over Skype, a service that allows users to call each other free over the Internet. In November 2005, a month after they first "clicked" online, they married.
The software routes phone calls over the Internet, substituting voice for instant messages. Web mail services such as Google, MSN and Yahoo also allow customers to make Net phone calls.
Since it was founded in 2003, Skype has added voice mail and video communication. The service says it has more than 100 million users. At first, Skype was used mostly by people who already knew each other: spouses on business trips, camp friends and college students. Then specialized dating Web sites discovered Skype, and its role as a matchmaker started growing.
On most online dating sites, singles send messages to one another, but "it takes an awful long time for them to find out if they're compatible," said David Finlay, the co-owner of SomeoneNew.com, a 14,000-member dating site. Finlay says that by using Skype, people on his Web site are able to determine whether they're compatible after one or two phone calls.
Skype has spawned love connections between Belgians and Japanese, Germans and Israelis, Americans and Egyptians and even a Guatemalan nail technician and a Canadian.
But not all are that happy;
Some psychologists say a relationship created and sustained by Net phone can be incomplete. Net phone contact is "simultaneously allowing people to become more intimate and yet have less patience with real life and real-time human fumbles and foibles," said Linda Young, a psychologist at Seattle University who has counseled many students who have sustained or developed relationships over Skype.
I thank all these users that make love happen over the phone, and keep those super nodes up, so we all can make skype calls. May be after getting married or breaking up, they can remove skype!
Posted by ravenII at 10/22/2006 12:15:00 PM
SKYPE VOIP telephony, got boost from VOIP Phone Buddy for Skype which adds automatic telephone dialing to almost any windows application, including Microsoft CRM, Microsoft Access, Microsoft Outlook, Goldmine, ACT, other CRM and Accounts packages. How do you do that? Pretty easy once you have got VOIP Phone Buddy for Skype installed. Simply move the cursor to a phone number and press the Hot Key to activate "VOIP Phone Buddy for Skype" phone dialer. The VOIP Phone Buddy for Skype and Jajah dialer then looks at the phone number, where the cursor was previously, and formats the international country code automatically. If the phone number does not contain the international country code then the user can select the code from their own favorite countries in the Phone Buddy. The system also provides redial and Quick dial (favorite phone numbers) facilities too.
What did you say about Jajah? Yes the misnamed application, "VOIP Phone Buddy for Skype" also supports Jajah VOIP IP Telephony application.
How much does this cost? It is $19.00 per user and if you want try, you can get a 30 day trial.
Alright, I want this where do I get this from? Hold on. Yes go to Hmm..
If you search for it you will find it in many a shareware sites. When I checked for the home page of the application, http://voipphonebuddy.lllii.com/, you just get the latest version of the application to down load. But is a redirected down load from drh.digitalriver.com. So I cant find the author. Anyway the application seems to be free of viruses, but I need to check for Malware rootkits etc.
Posted by ravenII at 10/22/2006 11:11:00 AM
Friday, October 20, 2006
Notice to all Asterisk users:
A security Advisory has been issued on Asterisk open source PBX, IPPBX. Please fix as soon as possible, any application or servers using Asterisk like TRIXBOX, should fix this overflow. I am running Asterisk 1.4-beta2 and not affected.
Asterisk - chan_skinny Remote Unauthenticated Heap Overflow
All 1.2-branch releases prior to and including 188.8.131.52
All 1.0-branch releases prior to and including 1.0.12
All 1.4-branch beta releases (1.4.0-beta1, 1.4.0-beta2)
== Overview ==
Asterisk is "The Opensource PBX", a popular software telephony server.
The Asterisk Skinny channel driver for Cisco SCCP phones chan_skinny.so)
incorrectly validates a length value in the packet header. An integer
wrap-around leads to heap overwrite, and arbitrary remote code execution
== Details ==
The function 'static int get_input(struct skinnysession *s)' in
chan_skinny.c incorrectly validates a user supplied length in the packet
header. In the code below, four bytes of data are read from the socket,
cast to a signed integer, and assigned to dlen. If dlen is between -1
and -8 then (dlen + 8) will integer wrap to be greater than zero, but
less than sizeof(s->inbuf) for the purposes of this comparison.
Next, dlen + 4 is passed to read() as the maximum number of bytes to
write to s->inbuf+4. Read() takes an unsigned value, so dlen is
interpreted as a very large number. For example, a value of -6 is
interpreted as 0xfffffffa bytes. This instructs read() to write beyond
the allocated 1000 byte length of the buffer s->inbuf.
== Solutions ==
- Disable the chan_skinny module if it is not required.
- Firewall port 2000/tcp from untrusted networks.
- Install the vendor supplied upgrades:
1.0-branch: Upgrade to 1.0.12 or later
1.2-branch: Upgrade to 1.2.13 or later
== Credit ==
Discovered and advised to Digium 17th October, 2006 by Adam Boileau of
Posted by ravenII at 10/20/2006 10:51:00 PM
It has been a few months now since Netopia released Timbuktu that solved a long standing problem for Timbuktu, NAT traversal. Timbuktu is a remote desktop software which was initially developed for Apple Macintosh but now run on both platforms, MAC and windows. Earlier versions, required users to have a routable IP address in order to connect. If you were behind a NAT or a Firewall, you were out of luck.
But good folks at Netopia saw the chance in Skype, Skype connects two or more users, without any problems, where ever they are as long as they are connected to the internet. It reverse firewalls and NAT routers without any hindrance. So Netopia decided to take a ride on Skypes back.
It is a good achievement for Timbaktu and now users can talk to each other via Skype while Timbuktu connects desktops.
I have not used Timbuktu in a long while, I think more than 10 years, so I cannot tell you how it operates now. But it does not matter, the point I am bringing out is, if Timbuktu can take a ride on Skype's back, what else will?
According to Netopia,
"Skype Integration - Timbuktu Pro features integration with Skype's popular Internet Telephony software to enhance Timbuktu Pro's connectivity options.
* From the familiar Timbuktu Pro interface, search for and connect to other Timbuktu Pro and Skype users through your Skype contacts list
* Timbuktu Pro leverages SkypeÂs API to automatically navigate through routers, firewalls and NAT devices".
Italsos states that, "Timbuktu Pro 8.6 works with Skype software. This product uses the Skype API but is not endorsed or certified by Skype. Skype is a trademark of Skype Technologies S.A. in Luxembourg and other countries."
So anyone could, many have already done similar to Netopia, develop an application thatleveragess Skype's connection profiles.
The Skype API is available for anyone to download and I would suggest, that users take good care as to what runs through your Skype connection. It could be a good application as well as bad applications such as rootkits, malware etc.
This is not only a fault of Skype, any P2P software could bring you the same risk. But the shear number of users of skype, (many malware and viruses were written for windows also the same reason, shear numbers), will attract those bad people. Users must take care of their PC's or MAC's, using appropriate firewall and virus protection software, and good old common sense.
I did a search on google to find if any one has noted this danger, but all I found in first few pages were phrases and encouragement, to both Netopia and Skype. So there goes the common sense, so rely on the other two.
Posted by ravenII at 10/20/2006 12:23:00 AM
Tuesday, October 17, 2006
I have known SNOM for ages, I even beta tested their IP PBX a few years ago. Now the offshoot of SNOM, 4S newcom, has released or about to release a Mac Mini based IP PBX, iBlueÂ®, That comes to you in a iPOD shuffle .
4S newcom focuses on delivering carrier-grade software platforms to VoIP providers, ISPs and enterprises. 4S newcom delivers system, components and know-how along the full VoIP value chain: carrier-grade class-4 "dial tone systems", (on-premise) IP PBX, hosted PBX as well as IP Centrex, based on SNOM technology.
If you are to visit VON Europe, which will be hosted in Berlin between November 6th and 8th, you will be able to see touch and if you got the bucks buy one.
According to Dr. Harry Behrens, Managing Director of 4S newcom, "We have put our complete IP PBX on it [the iPod Shuffle, It is so compact that even on the smallest iPod Shuffle (512 MB) enough room is left for 4 full hours of music." Hope it will not come with 4 hours of German music. Or if it does, it might help to expelthose thse sales calls.
The iBlueÂ® entry level system consists of a Mac mini, the iPod Shuffle with the 4S IP PBX licensed for up to 250 users and 30 parallel calls, as well as five snom300 VoIP phones. It will be priced at 2,999.00 Euro or 3762.00 Dollars. According to 4S newcom online sales will commence on November 6th 2006 - just in time for VON Europe, which will be hosted in Berlin between November 6th and 8th.
But hold it, Andy at VOIP Watch, says;
"This is similar to a Voxilla offering based on Communigate Pro. The interesting point of distinction is how much less expensive Voxilla's system is. So you don't get the iPOD...."
So if you already have an iPod or similar, check out his post for Voxilla link.
Links: 4s Newcom and iBlue, SNOM, VON EUROPE, VOIP Watch
Posted by ravenII at 10/17/2006 08:39:00 PM
Saturday, October 14, 2006
This post needed to be updated for a log time and I really need to do a new post about it. A comment by an Anonymous reader prompt me to search further and this is what I found;
*Skype now include the ability to disable supernode status and is described on Skype site instructing Universities how to disable supernode status from clients, http://www.skype.com/security/universities/
As we have mentioned before, if the computer is behind a NAT / Firewall or a combination, it is said to disable the supernode capabilities from the Skype client. It is also said to disable the supernode capabilities if the Skype is behind an HTTP or SOCKS5 proxy.
But the best and surefire way to disable it is a simple registry hack;
[HKEY_LOCAL_MACHINE\SOFTWARE\Policies\Skype\Phone]Open your regedit.exe and find the key "HKEY_LOCAL_MACHINE\SOFTWARE\Policies\Skype\Phone" and create the DWORD, "DisableSupernode"=dword:00000001.
If you are not comfortable with hacking your registry, please cut and paste the following text into your notepad and sabe it as nosupernode.reg and double click on the saved file. It will update your registry. You will need to reboot the computer for the registry key change to take effect.
Windows Registry Editor Version 5.00Now you are sure that you will not be a supernode, no matter which network you are connected to. If you really want to be a supernode, change the DWORD to 00000000.
All complaints and concerns about Skype revolves around a PC using Skype becoming a super node. I have written about removing skype many a times because it seems hard for regular PC user.
I have been looking for information about operating Skype and yet not be a supernode. Not much information is available from skype itself and I cannot test any other way. Because Skype EULA specifically warns you against modifying or changing software and you must read it completely if you are a user of Skype. Also 4.1 explains the usage of your computer resources for skype network.
But there are other ways that you can still use Skype and not be a super node, while not violating Skype EULA. May be Network admins should adopt educating users or Skype should give an option of a user not becoming a supernode if he or she wishes to. One caveat in this method is that you have to trust the user to follow the procedures.
I found this information on Fermilab, High-Energy Physics, the science of matter, space and time research lab belonging to the US government.
The page describing Skype usage on the site states;
"Skype (www.skype.com) is a free P2P (peer to peer) application that provides free voice-over-IP communication over the Internet. Use of Skype on systems attached to the Laboratory network is not prohibited per se. However, Skype calls are routed across the Skype network through other Skype systems, called SuperNodes. Any system running the Skype application runs the risk of surreptitiously being elevated to the status of SuperNode. Computers with a fast connection to Internet (such as at Fermilab), combined with high speed CPU, are most likely to become SuperNodes. A SuperNode can generate a considerable amount of traffic by opening a large number of concurrent connections for off-site systems, even after the local user has discontinued using the Skype application on his system. Generally, the user needs to reboot his system to stop the SuperNode call-routing activity."
The article further states how you can limit skype and prevent it from becoming a supernode without having to resort to remove skype type action!
"Based on extended testing by the Fermilab Networking Section, a procedure has been developed to configure systems to run the Skype application, without becoming a Supernode. The testing has been conducted on a Windows XP system, however the general approach and configuration settings should be applicable to other platforms & operating systems. These procedures work with the current release of Skype. There is no guarantee these procedures will function properly in future releases."
1. For Windows XP Skype version 184.108.40.206 or greater should be installed. For Linux or MacOS the most current version of Skype should be used.
2. The user should configure his system such that Skype is not loaded at startup.
3. A software firewall must be installed and enabled.
* Windows XP firewall or Zone Alarm are examples of suitable firewalls for XP
4. Exceptions for Skype must be disabled or deleted in the firewall setup.
* Windows XP Firewall: Settings>Control Panel>Windows Firewall>Exceptions
Detailed Configuration Instructions are located below under Instructions for Skype Configuration.
This configuration should prevent the local system from becoming a SuperNode.
One can also use Zonelab's Zonealarm to perform the same firewall configuration. Just prevent Skype from becoming a server.
Explanation for #2
Explanation for #4
To address the problems with trixbox 1.2 and Kernel 42, Trixbox the formerly Asteris@home, has released the Trixbox 1.2.2 with degraded (version only :) )kernel 34. This eliminates the various problems reported and resolved ;) by the users. Read the story in the following article;
VOIP IP Telephony: TRIXBOX 1.2 problems resolved, by giving up some.
This update has a new yum configuration file that will keep yum from updating the kernel. If you want to update the kernel you can do it manually.
The version of Asterisk 220.127.116.11 that is included with the 1.2.1 trixbox patch has a lot of the Asterisk patch files from Digium and other sources included. This may have caused some of the instability and FreePBX reload issues that were reported.
The Asterisk 18.104.22.168 that is included with 1.2.2 has only 4 patches.
SpanDSP : for fax recognition
NV fax detect : allows faxes to be detected on Sip channels and during the playing of audio prompts
HUDlite : a patch that creates some extra Manager messages used by HUDlite to monitor the status of calls
Digium SpeechAPI : used for speech recognition
Posted by ravenII at 10/14/2006 05:18:00 PM
Tuesday, October 10, 2006
IMC - new module enhancing OpenSER with Instant Messaging Conferencing capability was released by OpenSER. The main goal is to overcome the limitations of the SIP clients with IM support, the new module uses IRC-like conferencing style, with embedded commands in body of messages. It gives the possibility to create public or private rooms, to invite other users in a room.
Module information could be found following this link.
Also in the news is a new Management Interface (MI) - was also released by openSER, to optimize the management of OpenSER internals at runtime.
Checking one more bullet points or in the roadmap, the new management interface has arrived in the OpenSER scene. It is going to remove the code duplicity existing now for FIFO and Unix sockets implementations. The design was done with extensibility and scalability in mind and to open the way for new transports for management commands, like HTTP, (XML)RPC or SOAP.
More details: http://openser.org/pipermail/users/2006-October/007110.html
Posted by ravenII at 10/10/2006 09:17:00 AM
VONEUROPE, VON Autumn slated to start on 6th November and end on 8th, a two day event, has a slew of exhibitions and speaches, as ususal to any VON event. But I am writing about an invitation to OPENSER Summit 2006, held within the VON Europe.
OpenSER Summit is presented by OpenSER, a one of the well known Open Source VOIP IP telephony solutions in the market today.
According to Daniel Mierla, Main focus for this event is to reveal the status of OpenSER project -- achievements and roadmap (board members and main developers of the project will be there), discuss businesses cases of OpenSER, network people for business opportunities and cooperations (many VOIP IP Telephony executives will attend the event).
The agenda for the OpenSER Summit is;
Tuesday, November 7, 2006
15:45 - 17:00 OpenSER - a world of diversity in SIP services
-- Birds of a Feather session in exhibition area
* Hosted services
* ENUM and carrier grade
* scalability and distribution
* 3/4G and fixed mobile convergence
* peering with heterogeneous networks
Wednesday, November 8, 2006
09:00-11:15 Morning session
* Welcome and introduction of panelists
* OpenSER Project overview
* Industry perspectives - invited executives
* Visit of exhibition
15:00-18:00 Afternoon session
* Industry perspectives - invited executives
* Roadmap of OpenSER
* Open discussion
So if you happened to be in Berlin in November 6-8 or will be attending VON Europe, Please take a note of OpenSER.
Please register to the event and exhibitions by filling up the form here.
Posted by ravenII at 10/10/2006 08:52:00 AM
Monday, October 09, 2006
The UAEÂs ban on voice over internet protocol (VoIP) websites and related services is slowing down business growth, according to respondents surveyed at itp.net.
Of the 1492 users that responded, more than three quarters (just over 77%) answered ÂYes, very muchÂ, with a further 12.1% confirming, ÂTo a degree, yesÂ.
Less than 11% of respondents thought the VoIP ban was having no effect on torganizationationÂs growth.
May be Jajah is the solution to UAE problem.
VOIP IP Telephony: Disruption in VOIP IP Telephony Field By Jajah! Or is it?
Perhaps Jajah could terminate calls to and from UAE to the nearest Gateway. Since last leg is over regular phone lines, UAE authorities have no control over it. Just a thought
Posted by ravenII at 10/09/2006 11:15:00 AM
Sunday, October 08, 2006
Splinter.net has a gateway for googletalk client. This is one of the many emerging gtalk gateways. I checked out the service and the rates are better some areas and not so good for some areas. You can check the rates table if you are interested in calling landline (POTS) or a cell phone from your googletalk client.
It is pretty simple to setup, you just add firstname.lastname@example.org to your contact list on GoogleTalk, and Spinternet act as a middle man and enable you to reach regular telephones (landline and cell phones).
Then you You merely send the contact a message like "call 12127773456" and we connect you. You can also check the cost of the call (per minute) by sending a message COST xxxyyyzzzz - Displays the cost of calling xxxyyyzzzz for one minute.
If you are interested in testing the waters before using the service, once you setup your googletalk to use the splinternet service They will accept your invitation and give you a small credit so you can make a few calls without giving them any payment details.
And if you are happy with the service, you can use paypal to pay for the service.
Posted by ravenII at 10/08/2006 10:15:00 AM
Thursday, October 05, 2006
I have been a fan of disruptive technologies, Be it in the form of VOIP or Grid Technology, for that matter any thing that makes our lives easier to manage. I wrote earlier about disruptive technologies;
1.VOIP IP Telephony: Disruptive technologies disrupted? May be! No way!!
2. VOIP IP Telephony: What is the driving force behind VOIP?
related to VOIP IP Telephony.
Here comes the new disruptive technology in our field. JAJAH, ring a phone, I meant a bell?
If not you need to get Jajah to ring your phones.
The main part of Jajah, is that they have moved the total call processing to a web front end, (by the way I like the simple interface!). Yes, if you want to call someone, you type in the phone numbers, yours and receivers, on a web form and click dial! Well your phone rings and you talk. How simple is that compared to setting up a gateway, finding IP configurations, firewall holes, FXO and FXS ports, to mention a few items that we go through setting up a VOIP call. Of course it is different if you are using a service like skype, Gizmoproject etc.
Jajah operates just like Skype, JAJAH is based on a P2P (peer-to-peer) network system pure peer-to-peer network does not have the notion of clients or servers, but only equal peer nodes that simultaneously function as both "clients" and "servers" to the other nodes on the network. This model of network arrangement differs from the client-server model where communication is usually to and from a central server.
But there is a difference from skype, That's the reason that this might be a real skype killer! Here are the stuff that Jajah edges out Skype; It supports other VOIP protocols, including SIP, IAX2 etc, in addition to JAJAH´s uses its own proprietary protocol, supporting firewall traversal, NAT detection and bandwidth saving peer-to-peer technology. Again unlike Skype, it's not using somebody's else's infrastructure—i.e. users'—to build its network. It has 200 switching engines in 85 countries, most of them leased or managed computers running Jajah's software.
When you place a call with Jajah, your phone rings once, you pick it up and hear a recorded message saying Jajah will now connect your call. Then the phone rings at the other end. It's a reasonably intuitive, but unfamiliar to most VOIP users. Connection quality according to most users has been good. Again unlike Skype, Jajah doesn't utilize wide band technology, so voice quality is never better than on a regular phone call. But it's at least adequate, usually comparable to normal PSTN calls.
Calls are free when registered Jajah members call each other—if they're both in Zone 1 (which includes the U.S., Canada, China, Singapore, and Hong Kong) or Zone 2 (which includes Australia, most of Europe, and places in Asia and South America). Other calls are charged at rates ranging from 7.5 to 65 cents per minute (with calls to mobile phones in exotic locations being far and away the most costly connections).
Jajah originally launched a conventional VoIP soft phone service last July (2005) using the codec Mattes' team had developed. The service was championed in the blogosphere and had phenomenal early success—720,000 users signed up by September of that year.
The other thing about Jajah. As mentioned, users don't need a broadband connection, since calls are carried the "last mile" over the PSTN, not over the Internet.
But the question is will it really fly?
I think it will!
Posted by ravenII at 10/05/2006 10:54:00 AM
Wednesday, October 04, 2006
You can also read the news release and discussion going on at trixbox!, There might come a fork of trixbox already.!
MySQL's Brian "Krow" Aker is reporting that Trixbox, a asterisk Linux distribution, has been acquired by Fonality.
Brian thinks it is a cool transaction.
Brians other thought (which caught my eyes right away) are;
"Fonality just moved into first place for the telephony Linux vertical stack, and this is a trillion dollar market, one in which Microsoft doesn't have the technology to respond with.
Today the online world for Telephony is like the world for networks at the beginning of the 80's. Google, Skype, and the rest are creating their own lakes hoping to become the ocean. The same could be said of Compuserve, AOL, and Prodigy around 1993. Telephony on the internet is not about being locked into any single provider, it is about being able to talk to anyone, anywhere."
Read more of his thoughts at his blog.
Brian himself got the information from VoIP & Gadgets Blog on TMCnet. A blog I frequent a lot to get voip IP Telephony dose of the day. But I missed this one and to come through /. --> Brian's Blog --> VoIP & Gadgets Blog. Anyway both are good reads. Read VoIP & Gadgets Blog to get the facts first and then Brian's to digest and wash it down.
Posted by ravenII at 10/04/2006 09:05:00 PM
Tuesday, October 03, 2006
Gumstix have been a favorite gadget for me for some time. My first experience was with their waysmall 200. I am still busy with it as, there are many a things you could do with these miniature computers. My first successful project was a out door Wireless Antenna /Gateway. Only connection I had was a 20M Ethernet cable from my office to roof top. Anyway that is a subject for another article.
Today we will focus on the new netstix 200xm-cf.(pictured right).
The netstix 200xm-cf computer has received FCC Class A certification for business use. See Part 15, Subpart B, of the Federal Register (CFR 47, Parts 0-19). At 35mm x 103mm [1-3/8 x 4-1/8"], the netstix 200xm-cf comes with 64MB of Ram and 16MB of flash memory, runs at 200MHz and starts at $165 USD each for orders of 1,000 units or more.
Let me give you a little taste before I go on;
PID Uid VmSize Stat Command
1 root 368 S init
2 root SWN [ksoftirqd/0]
3 root SW< [events/0]
4 root SW< [khelper]
9 root SW< [kthread]
20 root SW< [kblockd/0]
36 root SW [pdflush]
37 root SW [pdflush]
39 root SW< [aio/0]
38 root SW [kswapd0]
49 root SW [mtdblockd]
66 root SWN [jffs2_gcd_mtd2]
213 root SW [pccardd]
225 root 448 S /sbin/cardmgr
269 root 400 S udhcpc -b -i eth0
304 root 1020 S /usr/sbin/sshd
336 root 348 S /sbin/getty -L ttyS0 115200 vt100
337 root 360 S /sbin/syslogd -n -m 0
338 root 336 S /sbin/klogd -n
441 root 5368 S /usr/sbin/asterisk -p
442 root 5368 S /usr/sbin/asterisk -p
444 root 5368 S /usr/sbin/asterisk -p
445 root 5368 S /usr/sbin/asterisk -p
446 root 5368 S /usr/sbin/asterisk -p
447 root 5368 S /usr/sbin/asterisk -p
448 root 5368 S /usr/sbin/asterisk -p
449 root 5368 S /usr/sbin/asterisk -p
450 root 5368 S /usr/sbin/asterisk -p
451 root 5368 S /usr/sbin/asterisk -p
452 root 5368 S /usr/sbin/asterisk -p
453 root 5368 S /usr/sbin/asterisk -p
454 root 5368 S /usr/sbin/asterisk -p
455 root 5368 S /usr/sbin/asterisk -p
456 root 5368 S /usr/sbin/asterisk -p
457 root 5368 S /usr/sbin/asterisk -p
507 root 1220 S /usr/sbin/sshd: root@pts/0
509 root 524 S -sh
557 root 368 R ps auwx
Yes this is the "ps auwx" command from the gumstix. Yes I arrived at this via World's Smallest VoIP PBX page on the AstLinux site.
But the information was too little;
"The Gumstix is a very flexible line of Xscale based SBC's (single board computers) manufactured by the nice guys over at Gumstix. Work is currently underway to fully integrate this effort with AstLinux. The model I used was the 400xm with the following hardware specs:
- 400mhz Intel XScale Procesor
- 64mb RAM
- 16mb flash
- netCF daughterboard (one CF socket and one 10/100 ethernet)".
But I went to Kristian Kielhofner's site via the link provided.
That's where I found all the gems I needed. But now I will have to spend sometime getting these stuff in to my gumstix or a new one.
But voipstix will be a boon for all the road warriors who want to carry voip on a stix.
Posted by ravenII at 10/03/2006 01:40:00 AM
Monday, October 02, 2006
I don't know about you but for me this is going to be a problem! I prefer mouse that is a mouse. Imagine you have to browse while you are on a call! Anyway if you carry a skype phone and a mouse when you travel this might work and at least on a notebook you can use the built in mouse, thumb track.
Released as ASG 142 Skype Mouse phone, the mouse uses a 800 DPI optical sensor and connects it to users USB port. It looks like a mouse but functions like a phone as well as mouse ensuring high voice quality according to the various news and the product site.
Features ASG 142 Skype Mouse phone:
1.A green color 1.3 inch LCD display screen
2.Displays Skype Contact lists and their status
3.Build-in microphone and speaker
4.Separate sound chip to enhance Softphone VoIP call quality
5.Single USB Cable to handle both mouse and phone
6.Supports hands-free speaker phone
7.Dot matrix caller-ID and Volume Control IM
8.Auto detection between Mouse and Phone
9.Uses USB 1.1 USB interface without extra power or sound card
10.Light weight (100 grams)
I am sure the newest phone will make special appeal to the millions of Skype users.
Posted by ravenII at 10/02/2006 10:41:00 AM
Sunday, October 01, 2006
I was at news site today reading about VOIP/IM by Carl Weinschenk. He brings out ideas and thoughts brought about by an article on Light Reading by Mark Sullivan, Google: Resistance Is Futile...
Both the articles tells us how diverse is the VOIP/IM sphere right now is. Just like the early early early email. When email from one client could not go to another email client nor read it even if it was received.
Google is going about this in another angle, with google talk.
For Internet-based VOIP, Google uses XMPP, based on Jabber protocol and setting it as the standard. ÂWe said we were going to use an open standard from day one,Â Google Talk product manager Mike Jazayeri says. Other IM clients such as Apple's iChat and Cerulean Studios' Trillian also are underpinned by the XMPP standard.
Google Talk can now jabber away with users of such services as EarthLink Inc. , Gizmo Project , NetEase.com Inc. Chikka Asia Inc. , and MediaRing Ltd. , as well as with numerous other Jabber-based clients homegrown by ISPs, universities, and corporations.
When email federation happened, usage of email exploded. Usage of wireless text messaging also grew exponentially after the wireless service providers adopted the common SMS standard enabling, for example, a Cingular Wireless LLC customer to send messages to a Sprint Wireless account.
Read both the article to get a good idea of where VOIP/IM is heading!
Google is planning to go places with googletalk too! see the this googletalk blog post;
Now anyone can Talk
Google Talk is now open to everyone! Until now, users needed a Gmail account to use Google Talk. Now, anyone can use the service by creating a Google Account.
Posted by ravenII at 10/01/2006 05:53:00 PM
Matrix Telecom has released Setu ATA2LL, IP based product that interfaces the VoIP network to traditional telephony interfaces and vice-versa.
Setu ATA2LL, a SIP-based analog telephone adaptor (ATA) with 2 FXS and 1 FXO lifeline ports. It interfaces legacy telephone devices to IP-based networks and is specially designed for SOHO users to offer them advantages of low-tariff Internet telephony for long distance and international calls.
Setu ATA2LL can propagate the call released on the FXS in the form of CPC signal. The device senses this signal and frees the FXS port. An FXS port can be programmed for any of the three CLIP protocols - DTMF, FSK ITU-T V.23 and FSK Bellcore 202. all of are traditional wired telephony protocols.
The product provides a list of programmable numbers or part numbers with a SIP account, one from any of the VOIP IP Telephony providers that supports SIP protocol. The FXO port connects to a regular telephone line. This port can be used to dial emergency numbers and during unavailability of the Internet or power failures.
Call arriving from a SIP account can be routed to either one or both FXS ports. Matrix Setu ATA2LL supports PPPoE client and hence can be used with DSL connection with fixed IP as well as PPPOE accounts. If you have multiple SIP accounts, dynamic allocation of SIP account is also possible using dial plan.
2 FXS Ports with 2 SIP Accounts ATA
1 Lifeline FXO Port
Codec G.711, G.723.1, G.729A/B
Fax - Pass-through and T.38 Real-Time
NAT and STUN
Echo Cancellation: G.168 up to 32ms
Posted by ravenII at 10/01/2006 04:07:00 PM