This release of PIKA Connect for Asterisk, available in November, allows Asterisk-based applications to use Skype to receive incoming and/or make outgoing calls, provides access to the calling Skype ID profile information (caller ID), and has touch tone (DTMF) detection capabilities.  Skype clients running on Windows based PCs are connected to the channel driver via PIKA’s AllOnHost™ (host-based) voice processing technology.  Skype clients can be distributed across an unlimited number of Windows PCs to achieve the density requirements the voice application may require. 

“Given the ever increasing use of Asterisk in business communication solutions and the rapid adoption of Skype as a business-to-business communication tool it seemed natural to do a mashup of these two very popular technologies,” said David Clarke, Business Development Manager at PIKA Technologies. “Our second generation PIKA Connect for Asterisk channel driver makes it a seamless exercise to Skype-enable your Asterisk based solution.”

PIKA has already implemented this technology in their own corporate PBX system, allowing customers to contact them directly via Skype. Using Skype, customers are able to call PIKA from anywhere in the world, free of charge, using any computer with Skype installed.  It is as simple as a click of a button on the PIKA website: www.pikatechnologies.com   The Skype call in turn is carried over the Skype network and terminated directly on PIKA’s Asterisk based auto attendant.  No extra phone lines or ports on the PBX are required.  Once answered, callers can navigate the auto attendant in the normal fashion using the Skype dial pad (touch tone digits).  All internal extensions as well as speed dials for sales and support are accepted. 

PIKA is currently enlisting Beta candidates for the second generation PIKA Connect for Asterisk.  If you are an Asterisk user or developer interested in running a Beta trial, please contact David Clarke at david.clarke@pikatech.com or skype davidclarkepika

In addition, we are not certain of the exact origin of the code, and so we are concerned that the package of source code that was posted may contain code from third parties that is not licensed for redistribution, or not licensed under the terms that the posting suggested would apply to it.

We have therefore removed the links to the package from our mailing list archives.  We recognize the importance of the integrity of these archives, but we do not wish to facilitate violation of anyone's copyrights or license agreements.

--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
 

The 0.3-m1 milestone release focuses on ease of use and provides the new org.asteriskjava.live package that takes care of the lowlevel action and event handling of the Manager API and offers an intuitive API for Java developers. Asterisk-Java has been updated to take advantage of the new features of Java 5.0 and therfore requires a Java Virtual Machine of at least version 1.5.0.

Asterisk-Java is used in several commercial environments and by the following Open Source projects:

    * Asterisk-JTAPI
      JTAPI implementation for Asterisk.
      http://asterisk-jtapi.sf.net/
    * Asterisk-IM
      A plugin for the Jive Messenger XMPP (jabber) server. It provides integrated presence between your IM client and phone, notification of incoming calls by IM and originate calls from supported IM clients.
      http://www.jivesoftware.org/asterisk-im/
    * Asterisk Desktop Manager (ADM)
      A desktop application that will allow for automatic on-call volume reduction, one click dial from clipboard, integrated phonebook and more.
      http://adm.hamnett.org/

Asterisk-Java is available under Apache 2.0 license at:
http://asteriskjava.org
 

Joshua Colp joined our development team a few months ago. Josh (file on IRC/Mantis) has been working on Asterisk development for quite some time and had contributed many features and bug fixes as a volunteer community member, along with being very active on the IRC channels and issue tracker.

Steve Murphy joined our development team at the beginning of June. Steve (murf on IRC/Mantis) had rewritten Asterisk's expression parser and the AEL language parser as a volunteer community member, along with various other bug fixes and improvements.

Jason Parker joined our development team at the beginning of this week. Jason (qwell on IRC/Mantis) has been maintaining the chan_skinny driver for Cisco SCCP phones as well acting as a bug marshal and fixing various bugs in Asterisk for the past year or more.

Russell Bryant has been a Digium part-time employee and an active Asterisk maintainer since before I got involved with Asterisk :-) His contributions are innumerable, and he has worked far more than the 'ten to twenty hours per week' he claims to have available outside of his school work! Russell (russellb on IRC/Mantis) will be joining us full time in Huntsville after the winter semester is complete, when he expects to graduate.

Please join me in welcoming all these new members of our development team; they are helping to make Asterisk (and our other software products) better every day and will enable us to accelerate our products into the future.

--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
 

Onvoy Corporate
300 S. Highway 169, Suite 700
Minneapolis, MN 55426
http://maps.google.com/maps?q=Onvoy,+300+Highway+169+S,+55426

Meetings are held monthly on the second Saturday of each month, excluding July and December. The Agenda is posted online
http://www.voip-info.org/wiki/index.php?page=Twin+Cities+Asterisk+User+Group+Agenda

This month we'll be discussing telephone set options for Asterisk VoIP systems.

Please bring your telephone sets in to share with other members. I'm hoping we all get a change to touch and feel most of the telephones the market has to offer.

Don't miss this meeting!
DOOR PRIZES: I know we're going to be giving away atleast one Polycom telephone set. These baby's are a wonderful addition to your Asterisk hardware collection.

If you're having a problem with Asterisk, bring your questions to a meeting for free help. We love helping new users!

Come to a meeting to meet other asterisk users, see asterisk solutions, win a door prize, eat food, or for the good company, to look for work, if your looking for employees, to go out for a drive, to get out of your house, whatever, JUST COME TO THE MEETING!

New visitors can help themselves to FREE FXO Interface cards (So you can connect your phone line, and have a timing source for meetme and IAX protocols). Some members have been known to swap hardware at the meetings.  Have extra VoIP gear, looking for VoIP gear? There's plenty of hardware to see. Have you been to a meeting recently?

Please come and share your own ideas and learn from others. As always,we'll have food.

We are always looking for help with meeting topics. If you feel like taking the lead, please do and simply let me know if you need anything.

Meeting starts at 11:30am and parking is available everywhere. Meetings run about 2 hours.

Look forward to seeing you there.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group%20TwinCities%20Minnesota%20USA
http://www.tcaug.net/

If you have a product or service you'd like to introduce to our members, send a private message to ejo1(at)soundchoicecomm.com and we'll see if we can't get you listed as next month's sponsor.

PS: Eric Osterberg will have a dozen or so Polycom IP500 series telephone sets at a onetime special price only available at this meeting. So bring your checkbooks. This will be a blowout sale on used but refurbished hardware. He has 30 sets available, so contact me if you'd like to have more at the meeting. (New handsets, new cords, new keycaps, lastest software from an authorized polycom dealer)
 

Currently, it has been used (making and receiving phone calls) and  developed in an ITU SS7 environment, but I have a good chunk of the code included which is required for ANSI support as well.  I think I'm going to get an ANSI link in a few weeks, so hopefully I'll have that  tested and working relatively soon.

It supports MTP2, MTP3, and ISUP.  After I get these layers fleshed  out, I'm planning on starting on SCCP and the layers above that with the eventual goal of database-lookup and SMS support.

To test, you must have a T1/E1 card as well as an SS7 link.  You also need to have zaptel installed on your system.

Here are the instructions for checking it out of subversion and getting it working:

`svn co http://svn.digium.com/svn/libss7/trunk libss7`
`cd libss7`
`make install`

Right now, the changes to chan_zap are implemented in a special developer branch of asterisk. These are the instructions to check it out
`svn co http://svn.digium.com/svn/asterisk/team/mattf/asterisk-ss7
asterisk-ss7`
`cd asterisk-ss7`

If you haven't compiled trunk yet, you may have to run `make` a few times so that the configure script runs and sets things up properly.  It should find libss7, and compile chan_zap with support for it.  The link is brought up automatically when Asterisk starts.

Configuration in zaptel.conf is similar to that of a PRI.  Your signalling channel will be set as a "dchan" and the bearer channels are set as "bchan".  For information about setting up zapata.conf, see the sample zapata.conf in the configs/zapata.conf.sample in the asterisk-ss7 branch.  There also is a libss7 project section on Mantis now for any bugs that you might encounter.

Matthew Fredrickson
 

Asterisk is an open source, freely available application that allows organizations to access all of the features of a typical telephony PBX, including voicemail services, call conferencing, interactive voice response, call queuing, three-way calling and caller ID services.

"Users of Voice over Internet Protocol (VoIP) systems must be mindful not only of denial-of-service vulnerabilities in their VoIP PBX implementations, such as the vulnerability discovered in Asterisk, but underlying VoIP protocol weaknesses that may leave organizations open to vishing, a new security threat which uses VoIP to steal user information, and spam over the VoIP network," said Chris Rouland, chief technology officer of Internet Security Systems. "By leveraging preemptive protection from Internet Security Systems, organizations can avoid the potential loss of productivity and the business ramifications caused by these VoIP flaws as well as the underlying operating systems vulnerabilities that VoIP platforms run on."

ISS X-Force has discovered a denial of service vulnerability in the IAX2, which is used by Asterisk PBX to exchange Voice over Internet Protocol (VoIP) and call content. The vulnerability is apparent if an attacker floods the phone service with call requests, thereby preventing the phone service from handling new telephone calls.

ISS X-Force discovered a second vulnerability that allows an attacker to leverage accounts without passwords on an Asterisk PBX network to flood another network with large amounts of traffic. The volume of traffic can saturate the victim's Internet connection and cause complete denial of Internet service to the victim. Additionally, victims of the attack may experience reduced quality of service.

Asterisk has already released a patch to address the denial of service vulnerability. Asterisk users are urged to upgrade as soon as they can practically do so, or ensure that they do not expose IAX2 services to the public if it is not necessary. Asterisk users are strongly advised to ensure that no accounts are configured without passwords. For more details visit www.asterisk.org.

ISS has provided customers with preemptive protection for these flaws through its Proventia security platform. ISS' preemptive technology is based on the research and discoveries of its X-Force research and development team. By protecting against vulnerabilities rather than known exploits, ISS' Virtual Patch(R) technology keeps organizations ahead of Internet threats until they are able to obtain, test and apply patches from affected vendors.

The ISS X-Force advisory on this vulnerability can be found at: http://xforce.iss.net/xforce/alerts/id/228 and http://xforce.iss.net/xforce/alerts/id/229.

We are getting ready to release our Call Control Gateway application which allows for both remote phone control and PC to phone integration between LCS and an Asterisk PBX. The gateway is scheduled to be released in the beginning of Nov. Currently we are looking for Beta Testers that are interested in this solution. More information on the product, along with the Beta Application can be found on our website at:

http://www.m-networks.net/uccg/


William Mandra
M-Networks

We need to determine a set of tests to run. It is my intention to simulate a typical enterprise/provider environment by having discreet components of the test operation - sip proxy (ser), sbc, gateway, soft-switch.

We will send various different types of common and uncommon traffic patterns to the test system, so we can measure and report on the performance of Asterisk on each component. I would also like to put Asterisk's SS7 implementation to some serious testing, but very honestly I have limited SS7 skills at this time. The Abacus also supports H.323 and SCCP, so we should also dedicate some time to these protocols as well.

Lets discuss,

Jeremy McNamara

The INRIA office is outside of Paris, in Rocquencourt. There will be buses going to and
from downtown Paris, so we're staying in Paris to get proper inspiration :-)

My suggestion is that we start 10 AM on Monday, nov 20th, and continue to 3 PM (15:00) on Wednesday, November 22nd. On Tuesday, we're eating dinner together in Paris (sponsors welcome :-) )

This is going to be a very practical meeting with interoperability tests between various devices, SIP debugging and coding. We need to figure out a way to add proper handling of video attributes in Asterisk and maybe look at additional features I know that you are working on out there:

- Video on hold (streaming)
- Video prompts for IVRs
- T.140 text in addition to video
- Integration with 3G video
- Video conferencing
- Additional topics (codename-pinapple etc)

Let's investigate these areas together, trying to find solutions that we can work forward on integration with other Open Source products or just experience in connections to commercial products.

There will be a very limited amount of seats. You will have to cover your own costs for travelling and eating, but there will be no registration fee (thanks to INRIA)

**** Register by mailing info@edvina.net today.

We have no seats for people who wants to "listen in" or "meet people" 
- this is a working meeting only. I hope you understand this. You do not need to  be an
Asterisk guru or developer, but still have knowledge about SIP and Video so that you can contribute.

My travel agency will handle hotel reservations, so we can stay in the same place. They can also assist you with booking flights, if needed. Emnet, my admin, will take care of the details.

A big thank you to INRIA for hosting this meeting. And a big thank you to other french contributors that offered to help with meeting rooms.

Cheers,
/Olle & Philippe

Links:

INRIA's website :
http://www.inria.fr/index.en.html

Access INRIA Rocquencourt research unit :
http://www-rocq.inria.fr/en/inria-rocq/moyens_acces/index.htm

 

As additional features, chan_celliax can manage (on Linux) voicemodems that use the ALSA modem driver (but with voicemodems it can't interact with Skype), and has got CLI commands: console, dial and hangup similar to the ones in chan_oss, useful for testing without hardware.

chan_celliax runs on Asterisk 1.2 on Linux and Windows (with cygwin and a little modification to the compilation of Asterisk to avoid the calls to sigkill). At date it runs on Asterisk 1.4 only in Linux
(because of the different compilation procedure of Asterisk 1.4, that does not build on cygwin). Hopefully it will run on FreeBSD.

Together with chan_celliax is distributed the Celliax Developer's LiveCD, with a working installation of Asterisk, chan_celliax, and configuration utilities based on Knoppix. Celliax Developer's LiveCD comes complete with all the developer's tools needed to recompile from svn and remaster
the LiveCD itself.  The LiveCD contains also the cygwin installer and the tgz with the asterisk-celliax stuff to be untarred in a basic cygwin installation.

The lucky ones of you guys who are at Astricon Developer's Summit will find some cables and CDs in the Code Zone.

More info on the site www.celliax.org, with forums, downloads, svn, trac, etc.

Please, report bug and issues to www.celliax.org/trac , is pre-beta software ;-)

Happy hacking,

Giovanni Maruzzelli
 

If you want to listen into the Developers Summit meeting at Astricon you can call 972-961-7666 or IAX2/conference@switch-2.nufone.net/4569.

Jeremy McNamara
 

Now, within the bowels of Asterisk, each channel records the context name, the extension name, and the priority number of the next priority to execute. The first thing Asterisk does is lookup the context, then looks up the extension in that context, and then searches for the priority number to execute. These three lookups get done for every line of code. Why? Millions of reasons: with goto's, etc, the context/exten/priority is just plugged into the channel, and the engine will then fetch that. Saving these refs by name makes it so the dialplan can be reloaded mid-stream, and for the most part survive without a hitch.

The important thing to remember is that all 3 of these objects are stored in linear linked lists.

Now, the fun part. The lookups for the extensions and priorities get done quite often. So, I did studies of the 3 individual lookups that are done to find an arbitrary priority to execute. I am currently using threaded red-black binary trees instead of the simple linked-lists for contexts and priorities. It's not LGPL code, so I can't use them in asterisk, but until I get LGPL'd code, or our own hashtabs, I'm using them for an experiment.

The extension patterns are matched in a linear fashion, with no early cutoffs; that last extension in the list might yield the best match. To speed this up, I take a different route. I take the first character of each pattern, and I form a tree, where each level down is another, subsequent character set from the patterns. Patterns that start with similar character sequences share a path down the tree. The maximum depth of the tree coincides with the longest pattern in the set. To match, I follow a path down thru the tree until the final leaf is reached. It is possible that one sequence of characters can match more than one path down the tree. In this case, all possible paths are followed, and the metrics followed along the way determine the 'best' match. This algorithm is fairly simple, and much faster-- when you have a lot of patterns. Let's see, it's about O(5*(averagePatternLength)). You form this tree after the context is created and populated. It is stored in the ast_context structure.

Threaded red-black binary trees are a cousin of the AVL trees, binary trees that implement algorithms to keep them optimally balanced at all times. They have log(n) search times, which is pretty good compared to linked lists, which will have n/2 search times, where n is the number of items in the structure. So, if your set has 1000 items, a linear search will find the result, by looking at 500 items, on the average. With RB trees, you'd look thru at less than 9 or 10 items on the average. Much nicer!

Friends,

The focus among the core developer team now is: FIXING BUGS! We need to focus on fixing all bug reports for 1.2 and 1.4 to get 1.4 out. Please have some patience with us if we don't respond directly to proposals for new things, review your cool patches for future addition or comment on your proposal to make Asterisk 200% more effective by moving one line of code. The focus has to be on the bug tracker and the issues we have in the current beta.

We need all the assistance we can get in locating reporting bugs, testing patches, creating patch proposals. The more the community activates itself, the faster we can release a good and production-ready 1.4.

Thanks for your help and understanding,

/Olle

svn checkout: http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui

Using this interface will require Asterisk 1.4.

A few days ago i asked about certificate negotiation in asterisk.

On a related topic, i have been thinking for a while on how to provide TLS support for HTTP, AMI and more services within asterisk, keeping in mind the current state of affairs, the feedback i received on the above request, and with the goal of minimizing changes to the current code base.  I think I am at a stage where i have no more ideas, so attached you find:

- a description of the current situation with respect to the implementation of TLS support for HTTP and AMI;
- a workplan on how to add TLS support to AMI,
- general considerations on how to provide TLS TCP sockets.

www.chanskype.com

This release has better startup handling and fully supports DTMF - either inbound or outbound your Asterisk box.

Best regards,

Paulo Mannheimer
The ChanSkype team