My Little Howto for Polycom IP500 Ringtones

First I scoured the internet for as many cool ringtones I could think of
that my users would get a kick out of and play with on their lunch-hours.

Note that I'm not sure of copyright on all of these, so use them at your
own risk.

Ringtone files must match the following criteria:
- Bit Rate 64kbps
- Audio Sample Size 8bit
- Channels 1 (mono)
- Audio Sample Rate 8 kHz
- Audio Format CCIT u-Law
- Less than 300k in size

To find all this information out in windows, select the file and right click, go to
properties, then view the general properties for the file size, and the summary
for the audio information.

The audio files can be converted using a combination of Audacity and Windows Sound Recorder

I went into the sip.cfg on my ftp server, and found the following lines

sampled_audio saf.1="SoundPointIPWelcome.wav" saf.2=""
saf.3="" saf.4="" saf.5=""
saf.6="" saf.7="" saf.8=""
saf.9="" saf.10="" saf.11=""
saf.12="" saf.13="" saf.14=""
saf.15="" saf.16="" saf.17=""
saf.18="" saf.19="" saf.20=""
saf.21="" saf.22="" saf.23="" saf.24=""

and changed them to add my ringtone selection.

sampled_audio saf.1="SoundPointIPWelcome.wav" saf.2="macgyver_theme_song.wav" saf.3="dr_who_theme.wav" saf.4="twilight_zone_short.wav" saf.5="inspector_gadget_theme.wav" saf.6="the_ateam_theme.wav" saf.7="benny_hill_theme.wav" saf.8="james_bond_theme.wav" saf.9="mission_impossible_theme.wav" saf.10="airwolf_theme.wav" saf.11="night_court_theme.wav"
saf.12="knight_rider_theme.wav" saf.13="twilight_zone_full.wav" saf.14="" saf.15="" saf.16="" saf.17="" saf.18="" saf.19="" saf.20="" saf.21="" saf.22="" saf.23="" saf.24=""

then i copied all the wav files to the home ftp directory, and
made sure they were owned by the polycom user

# mv /tmp/*.wav /home/ip500
# chown ip500:users /home/ip500/*.wav

Reboot your phone and you should see them in your list of ringtones!

To download my sample ringtone files:
Polycom_Ring_Tones.tgz
Polycom_Ring_Tones.zip

PGP Coming to an IP call near you…

Looks like Phil Zimmermann has gone and developed a method to make PGP Encrypted calls over the "hostile internet". This sounds really promising, and anyone looking for a company to invest in, might want to talk to Phil since he's looking for some venture capatalists.

Read the article here: at the CNET site
By Joris Evers

Phil Zimmermann hopes that his secure Net phone-calling efforts will be as successful as his Pretty Good Privacy e-mail encryption program.

Zimmermann has developed a prototype of an Internet telephony application that encrypts calls to prevent eavesdropping. He plans to unveil the prototype on Thursday at the Black Hat Briefings security industry conference in Las Vegas.

"I am revealing this now because I want to help shape the direction of secure VoIP," Zimmermann said in an interview. VoIP stands for Voice over Internet Protocol, the technology used to enable people to make phone calls using an Internet connection.

VoIP is increasingly popular because it is cheaper than traditional phone service or, in some cases, free. Organizations can run their own VoIP service using products from vendors such as Cisco Systems. For consumers, companies including Packet8 and Vonage offer an actual phone that plugs into a broadband connection, while others such as Skype sell software that runs on a PC. Most popular instant messaging applications also have VoIP capabilities.

Security of VoIP systems is getting more attention in general. Cisco Systems identified several vulnerabilities in its products earlier this month. The flaws could lead to denial-of-service attacks on Cisco IP telephony networks, which are used by businesses.

Within the next two years, 97 percent of new phone systems installed in North America will be VoIP-based or will use a combination of traditional and VoIP technology, according to research firm Gartner. Cisco claims to have sold some 5 million VoIP phones to customers throughout the world.

It is already possible to encrypt VoIP data. However, today's technology uses the public key infrastructure coding system, which secures the exchange of data by providing each party with digital certificates that validate their authenticity. Setting up and managing PKI can be laborious. Zimmermann's system does not use PKI.

Zimmermann hopes to start a business that will sell products based on the encryption technology. It could also be licensed to other companies for use in their Internet telephony lineup. "I will have my own products, and there will be agreements with other companies to use it in their products as well," he said.

The security expert said while his prototype can be used to make calls, it still has some problems to be ironed out and is not close to being a finished product. "It is not mature enough," he said. "The crypto is real solid, but the VoIP client has some bugs." Zimmermann said. The application doesn't have an official name yet.

The VoIP client is based on the open-source Shtoom VoIP phone client. Zimmermann said he added cryptography to it.

This is not the first time that Zimmermann has worked on putting protections on Internet telephony. Almost 10 years ago, he launched PGPfone, a little ahead of its time. "The Internet was not ready then," he said.

Digium Announces Second Generation Firmware for T1/E1 Cards

Digium Announces Second Generation Firmware for T1/E1 Cards. New firmware increases quality, performance, and reliability

Digium Inc., the creator of Asterisk and pioneer of open source telephony, recently announced the availability of its second generation firmware for Digium.

The new firmware solution relies less on the server's CPU for the operation of the Digium card, therefore reducing CPU overhead and improving performance and increasing channel processing. For example, a dual-processor, 3-GHz 800FSB Intel XEON server with 1MB L2 cache, and a Digium 4-port T1/E1 card, can now convert 120 SIP channels with G.729 compression to the PSTN without Digium's echo cancellation module and 150 channels with G.729 compression with the Digium echo cancellation module.

"This development is significant for Digium's Asterisk customers," said Mark Spencer, president of Digium and creator of Asterisk. "There is always room for improvement in technology and we are dedicated to providing our customers with solutions that offer higher quality and reliability than our competitors. Our second generation firmware results in up to a 67% performance increase over previous benchmarks."

Additional improvements in the firmware include hardware TDM channel alignment (instead of software channel alignment) resulting in greater voice integrity and reliability, master clock source distribution for synchronized timing across multiple cards, assuring synchronization of clocks and increasing reliability and quality of data transmission, zero-latency TDM direct hardware-level cross-connect, support of new echo cancellation module resulting in increased performance, hardware DTMF detection with echo cancellation module and field upgradeable firmware for future updates.

Support and Availability

The Second Generation Firmware for Digium quad-span and dual-span T1/E1 cards is now available. For more information, please contact sales@digium.com or call 1-256-428-6262. All Digium products are warranted and include installation and troubleshooting support.

Improvements in the firmware include:

- TDM channel alignment now done in hardware (instead of software), for greater voice integrity and reliability
- Master clock source distribution for synchronized timing across multiple cards, assuring synchronization of clocks and increasing reliability and quality of data transmission
- Zero-latency TDM direct hardware-level cross-connect
- Supports new echo-cancellation module which further increases performance
- DTMF detection can be done in hardware with echo cancellation module
- Field-upgradeable firmware for future updates

The Firmware improvements affected the following Digium cards:

- TE411P—Quad-Span PCI T1/E1 termination with on-board echo cancellation for 3.3-volt PCI slots
- TE406P—Quad-Span PCI T1/E1 termination with on-board echo cancellation for 5-volt PCI slots
- TE410P—Quad-Span PCI T1/E1 termination for 3.3-volt PCI slots
- TE405P—Quad-Span PCI T1/E1 termination for 5-volt PCI slots
- TE210P—Dual-Span PCI T1/E1 termination for 3.3-volt PCI slots
- TE205P—Dual-Span PCI T1/E1 termination for 5-volt slots

For more information, please contact sales@digium.com, or +1-256-428-6262.

Asterisk version 1.2 :: What’s new?

In response to a large number of questions on the mailing list I've
decided to publish a presentation I have been running in the Asterisk
bootcamp - our one-week training class.

This presentation covers many, but does not claim to cover ALL, new
features of Asterisk version 1.2. I hope it will wet your appetite to
help us test the new code.

The presentation is available here:
* http://www.astricon.net/asterisk1-2/

For information on how you can help with Asterisk 1.2, see here:
* http://www.voip-forum.com/?p=176&more=1

If I've forgotten a new feature that you think is important, I'm
grateful for all input.

/Olle

Critical Patch for those using the L option on Dial()

http://bugs.digium.com/view.php?id=4760
>
> If you use the L() option on dial and say the latest CVS-HEAD in the
> past month you're potentially getting screwed out of a lot of money.
>
> We originally wrote the L() option for dial and it worked great till
> someone came along and hijacked the timer for something else thus
> causing the L option to fail/reset the timer to zero thus causing it
> to never timeout if someone were to say press a DTMF digit.
>
> So if you use this please test this and report back to the bug ASAP.
>
> Thanks,
> Brian West

Important changes in CVS-Head Asterisk

I have just committed some changes to CVS HEAD that make the effort to eliminate 'priority jumping' applications sooner vs. later...

Basically, there is now a global option, settable in extensions.conf, to disable all priority jumping. The only application that has been updated to respect this option is app_dial, but I will update the "janitor project" list to reflect what needs to be done. The 'j' option to Dial() now has the exact opposite behavior it had before: if priority jumping has been disabled globally, the 'j' option will cause that instance of Dial() to do jumping. This will be the model for all applications to be converted over to.

Anyone who does not change the global setting in extensions.conf (it defaults to 'on' in the source code) will not experience any change in behavior unless they were using 'j' to suppress jumping in Dial(). In that case, you will need to global suppress jumping, and then enable it in any Dial() calls where you need it. As more applications get converted to use this new option, you may need to add more 'j' options in different parts of your dialplan.

New users of CVS HEAD (anyone who uses the 'make samples' starter configuration file) will not have priority jumping on by default, since it is explicitly turned off in the sample configuration file. We will try to get the remaining apps converted over as quickly as possible, so that their behavior will be consistent.

- Kevin P Flemming

Asterisk 1.2 Release Plans Updated

As previously mentioned on the lists by Olle Johannson, we are actively trying to get Asterisk in shape for a 1.2 release within the next 60 days. To accomplish this, we need a few things to happen:

1) A feature freeze - This will occur at the end of this month, with no new feature submissions accepted after July 31st. Any _pending_ feature patches in Mantis that have passed architecture review and functionality testing before August 1st can be accepted into 1.2, if they make it through the remainder of the review processes and are able to be merged before August 15th.

2) Progress on open bugs - There are a number of bugs open in Mantis that are waiting for the poster to provide additional information, test results, call traces, etc. We would much prefer to not release 1.2 with suspected problems already identified, but we cannot solve them without adequate input from you. If you have an open bug and are not in a position to continue providing assistance in solving it, please post a message to the mailing lists asking for volunteers to help replicate the problem so it can get resolved.

3) Testing - We need a _lot_ of help testing. If you have not previously tested CVS HEAD, please download it, read the UPGRADE.txt file and install it on one or more systems to play around with. Please do _not_ put it into a production environment unless you are willing to accept the consequences of that action. If you do find a bug or other issue, when you open a bug in Mantis, please try to provide _all_ the configuration information, call traces, etc. that the bug guidelines request, so that we don't waste 3-4 days just going back and forth requesting more information from you. If possible, join the #asterisk or #asterisk-dev IRC channel to find out exactly what debugging information will be required and how to produce it, if you don't already have that knowledge.

4) Release Candidates - I will produce the first release candidate on August 20th, with followup versions produced every week until we deem the release ready for public consumption. I expect it will require at least three -RC releases for us to get things in shape, so that means that 1.2 itself may be ready by September 15th.

We are very thankful for the community's help and support, and we want Asterisk 1.2 to be as important a release as 1.0 itself was. The number of new features, performance improvements, bug fixes and interoperability enhancements in CVS HEAD is astonishing, and a very large percentage of them came directly from community contributions. We hope that all of the 'non-developers' in the community will be able to help us 'shake out' the bugs and problems remaining in the code, so we can be assured of the most stable 1.2 release possible :-)

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