Digium is spreading its Asterisk-based phone system love further with a new deal with Synnex, a company with a network of 15,000 IT resellers. Synnex has a dedicated telephony group to provide unified communications solutions to resellers that service the small-to-medium sized business.

The partnership will get Digium's Switchvox business phone systems to resellers throughout the U.S. and Canada. The deal will allow Synnex resellers to offer Switchvox business phone systems, Asterisk Business Edition software, and digital and analog telephony cards. Digium's open source Asterisk software-based systems will provide an interesting alternative offering for the resellers' catalogs.

For more:
- read the release

Related articles
IBM and Digium add Asterisk VoIP calling to Smart Cube
Digium launches 'app store' for Asterisk
Asterisk downloads up 50% in 2008

SIPit is the main interoperability event for all things SIP. It’s organized by the SIP Forum and creates good feedback to the IETF. Asterisk has been participating in SIPit during many years and in many variants  - videocaps, Marc Blanchet’s IPv6 branch and the standard Digium releases. All these tests has lead to a large amount of improvements for Asterisk and have helped us to build a network with other developers in the business, a network which helps when we have bugs that involve interoperability with these devices or servers. SIPit has proven important for the success of Asterisk, and thus it is also important for  everyone in the Asterisk community. 

Now, when we are working on the next  long-term release (1.8) we really need to test again and make sure that we interoperate properly. New stuff, like Terry’s SRTP branch, my RTCP work and the call completion and caller ID update work needs serious testing. We need feedback to be able to fix the issues with the TCP and TLS support. (more…)

© Edvina AB, Sollentuna, Sweden 2010 VoIP-Forum. All Rights Reserved.

With thousands of new downloads per day, millions of deployments, and a community of more than 65,000 members, the acceptance and growth of Asterisk has spawned an ecosystem spanning more than 170 countries. AstriCon gives all members of the Asterisk community—from telephony enthusiasts to people betting their businesses on the booming appeal of Asterisk communications—a forum to learn about the technology in depth, to discuss its newest uses and to meet potential collaborators.

Mark Spencer, Digium’s CTO and creator of Asterisk, said: “AstriCon 2009 was a total success as Asterisk has moved into mainstream use in phone systems used by organizations of all sizes. The ability to get together with so many Asterisk users to exchange ideas is always invaluable to Digium and we continue to be grateful for our community’s strong support. Looking ahead to AstriCon 2010, in addition to technical sessions, we expect to focus on use of Asterisk in commerce, in the cloud, by government agencies and larger enterprises, call centers, and more.”

Digium is once again pleased to be partnering with Technology Marketing Corporation (TMC) to promote the event to a broader audience. TMC has helped support other Digium events, including Digium|Asterisk World, with training sessions, video production, attendee registration and exhibit management. Companies interested in sponsoring AstriCon and participating on the EXPO floor should contact Joe Fabiano at TMC: +1 (203) 852-6800 ext. 132.

Registration for AstriCon 2010 is open now on the official event site: http://www.astricon.net.

Early bird rates are available until August 1, 2010.

I am working with two very different implementation projects this winter. One is about using Asterisk in large call center environments. Another is building a SIP network for 15.000 phones in a university.

Asterisk for large call centers - full control

The call center platform is focused on a lot of PBX functionality. Every agent stays connected to a conference session, where customers are connected and disconnected, recording is enabled and disabled, dtmf is used to control various services and both AGI and AMI (the manager interface) are both being used heavily to integrate with a controlling application. Asterisk is in full control of each and every call, all the time, but the application controls the flow of the calls. The dialplan is very small for a large-scale Asterisk installation.

Large scale SIP networks - Asterisk on the edge, providing services

For the university, scaling is important. It’s a plain SIP network, with two computer centers in different buildings. All accounts are managed by LDAP, there’s no account defined locally in the VoIP platform. SIP proxys (Kamailio) rule this network and DNS is used for failover. Many services like call transfer, three-party conference calls and call forwarding will be handled by the phones. Asterisk serves as gateways to the old world, Nortel systems, and feature servers for IVR, voicemail and switchboard. No single server is in control of any call.

The power of open standards and open source

Two completely different designs, both made possible by Open Source and Open Standards. Both of them needs to scale. Both of them needs failover, redundancy and stability. And in both cases, we’re replacing expensive legacy telecom equipment with new platforms that will cost less to operate, that has a higher degree of interoperability and much more functionality than the previous solutions. Open telephony wins.

© Edvina AB, Sollentuna, Sweden 2010 VoIP-Forum. All Rights Reserved.

Hans Petter Selasky alerted the Asterisk developer community about a potential harmful pattern in Asterisk dialplans on February 9th.  His example is as follows:

[from_sip]

exten => _X.,1,Dial(SIP/${EXTEN}@testsip)

He writes: “And if ${EXTEN} = “000@testsip&SIP/333” what turns out to happen then is similar to SQL injection  ”He is exactly right. Many VoIP protocols, including IAX2 and SIP, has a very large allowed character set in the dialed extension, a character set that allows characters that are used as separators to the dial() and the queue() applications, as well as within the dialstring that these applications send to the channel drivers in Asterisk. A user can change the dial options and dial something we should not be able to dial in your system. This article describes the issue in more detail and gives you some help on how to avoid this causing trouble in your Asterisk server. (more…)

© Edvina AB, Sollentuna, Sweden 2010 VoIP-Forum. All Rights Reserved.

The Asterisk Development Team has announced the release of Asterisk PBX 1.2.39. This release is available for immediate download at

http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 1.2.38 was created, but not released, to resolve two regression fixes caused by security updates. Prior to the release of Asterisk 1.2.38, one additional regression fix has been resolved, causing the release of Asterisk 1.2.39.

 * Fixes regression caused by randomized call numbers.  (Closes issue #15997) Reported by exarv. Patched by dvossel.

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.2.39

Thank you for your continued support of Asterisk!

On my wishlist for 1.8 long-term-support release the #1 item is a new media negotiation framework for Asterisk.

The history - something to learn from

If we had integrated John Martin’s videocaps in time for release of 1.4 we now would have enjoyed four releases of an Asterisk with much better video support, instead of the broken support we have today. Integration of that code was denied because of plans of something greater and bigger, which we have discussed for years at many Astridevcons. Because of us denying this and other code,  outside-of-tree code has been around that fixes a lot of issues and the original Asterisk still lacks proper video support.There’s also a few codec negotation patches that has been around and maintained for years, one of the major ones being Maxim Sobolev’s patch.

What are the issues we need to solve?

There are a few main issues here:

• Codecs: Codecs should not be handled as a bit, being turned on and off. Both audio and video codecs have attributes that we need to take care of properly
• Call setup: When answering, the properties of the answer needs to be relayed to the calling channel, not just a control frame that says “somebody answered”.
• Media streams: We should be prepared for multiple media streams, including multiple media streams of the same format

The new solution should be extensible, it should be easy to add both new codecs and properties.The solution has to be configurable. The way you want Asterisk to set up a bridge between two channels varies much. Some people prefer Asterisk doing transcoding some people want Asterisk to stay out of as much trouble as possible and just set up whatever is most simple. Other users just want to standardize all call legs to one type of phone, but have a different policy for other connections. There’s no one-solution-fits-all.

Asterisk needs a new media negotiation framework for continued success

We did write up a few documents on this a year ago at Astridevcon in Phoenix. I would really like to see some work on this. My personal feeling is that this is very important for the continued success of Asterisk in the marketplace. The amount of long-lived patches in this area that has been maintained for years shows that we have to do something (and that we sadly have ignored customer demand). The arrival of new codecs and new solutions, like video conferences with multiple audio and video channels, with text channels and possibly other types of channels (lika a binary channel for MSRP and digital ISDN) - all this tells us that we have to get there.

Everyone needs to contribute, not only Digium

I haven’t seen Digium invest in this project during the years we have discussed this. The Digium team has fixed the codec list, it is no longer a limited bitmap. Building a new media negotiation framework not a simple project and it’s not something any customer alone would fund, it would propably allocate too much resources from the team. (my personal guesses) I don’t think the community as a whole can expect or demand that Digium funds every needed change themselves, especially not in these times of financial worry. Everyone in the eco system will have to contribute to make Asterisk a better product - and I’m not only talking money here.Maybe we have to apply for funding somewhere else. It requires much more knowledge of Asterisk and experience than what I think we can get through Google summer-of-code, and it’s more urgent. Let’s discuss this and see how we can make this happen. Read the docs, start an open discussion and hopefully we can get not all of this, but at least the core of it, inside the next long term release of Asterisk, 1.8. It will require a lot of job from all of us in the dev team, as well as the community. Testing, feeding input, building a stabile architecture. I am sure we can get it done.

• The Astridevcon documentation is at: http://astridevcon.pbworks.com/Media+Negotiation It includes a lot of thoughts going down to code structures.
• My presentation that was a base for the discussion is athttp://svnview.digium.com/svn/asterisk/team/group/astridevcon2008/NewMediaArchitecture.pdf

The floor is open. What do you think?/Olle

© Edvina AB, Sollentuna, Sweden 2010 VoIP-Forum. All Rights Reserved.

Let’s change everything - and cause no damage for the end users!

The current version of the Internet is up for a big overhaul. We have to change the whole infrastructure it runs on, the famous IP protocol. A lot of work needs to be done and it affects everyone that works with the infrastructure. The result of all the hard work? Everything will work exactly as before. Nothing gained for the end-user experience. That’s why very few are paying attention to IPv6 - it is very hard to tell the persons in charge of IT projects what the benefit is compared with upgrading all PCs to Windows 7 or installing that new spam filter for e-mail.

Internet telephony needs IPv6 peer-to-peer addressing

It is easy to explain the need for a unified address space using telephony as an example. The fact that all companies and homes use a private address space that can’t be reached from the outside doesn’t matter when it comes to the old-fashioned Internet applications. The web browser contacts a server on the Internet. The e-mail client contacts an e-mail server on the Internet. The IM/Presence application contacts a server on the Internet. Nothing needs to reach in. Until you start using peer-2-peer applications. And telephony is a very common p2p application.

• -”You know that you have a broadband router that use one IP address from the Internet, assigned to you by the provider?
• “- “Yes”
• - “Do you also know that the broadband router let’s all your devices on the inside share this address by setting up a private address range?”
• - “Yes”
• - “If you add an IP phone on the inside - do you want to be able to receive calls?”
• - “Yes”
• - “How do you think I can call your phone  directly, if we don’t share the address space?”
• - “I don’t know.”

With IPv6, true p2p Internet telephony will become possible. When we get rid of the need for NAT, network address translation, we can finally separate access and policy. With a unified address plan, every device on the net has the possibility of reaching every other device. Policys might prevent that and we implement the policy in firewall software in the systems or in dedicated systems.Currently, in order for a phone to work on the inside of a NAT, most implementations connect to a server on the Internet. In order for an incoming call to get through, the phone or the server keeps sending empty messages. As long as these messages are sent - occupying unneeded bandwidth and resources in the network - the NAT believes there’s a communication session going on and let the messages in.The NAT itself has no policy, it just checks if there’s a client-initiated session going on or not. As long as the NAT believes there’s a session, it will forward packets from the outside to the inside device. When you have an incoming call, the server can forward an alert to the phone and the phone will start ringing. The same setup is used if your organization has a PBX system on the inside and use a SIP trunk provider on the Internet.This solution is uses by a range of applications and are not unique in any way for IP telephony.  IPv6 will make it easier to enable true Internet telephony and other p2p applications, as long as your firewalls let it happen. (more…)

© Edvina AB, Sollentuna, Sweden 2010 VoIP-Forum. All Rights Reserved.

The website will serve as a hub for the Asterisk open-source VoIP community including a place for users to review applications and phones. Developers can get exposure for their Asterisk-based applications and users can get the purchase info right from the site.

At the moment it looks like AsteriskExchange is not actually doing any selling itself, but instead directs users to vendor sites who sell the products. The site has a number of tabs including a 'most popular' tab. Currently, the Bria softphone app is at the top of the list, but with no reviews or star ratings yet, I am not sure how they are determining the popularity. It will be interesting in the future to see what apps end up the top of the list when more users access the site.

Unlike other app stores, AsteriskExchange also includes some hardware that you can learn about, review, and connect to sellers to get your equipement. Clicking on the Snom 360 deskphone will bring to an info page and an offsite link to 'Buy Now.'

For more:
- read this article from Connected Planet

Ever since Apple adapted their wildly successfull iTunes music store to sell apps for the iPhone, technology vendors everywhere have been trying to find opportunities to replicate the innovation. Now Digium is launching AsteriskExchange--while not quite a true app store--it's a marketplace and reviews site for Asterisk's open source community.

The website will serve as a hub for the Asterisk open-source VoIP community including a place for users to review applications and phones. Developers can get exposure for their Asterisk-based applications and users can get the purchase info right from the site. At the moment it looks like AsteriskExchange is not actually doing any selling itself, but instead directs users to vendor sites who sell the products. The site has a number of tabs including a 'most popular' tab. Currently, the Bria softphone app is at the top of the list, but with no reviews or star ratings yet, I am not sure how they are determining the popularity. It will be interesting in the future to see what apps end up the top of the list when more users access the site.

Unlike other app stores, AsteriskExchange also includes some hardware that you can learn about, review, and connect to sellers to get your equipement. Clicking on the Snom 360 deskphone will bring to an info page and an offsite link to 'Buy Now.'

For more:
- read this article from Connected Planet

Related articles
IBM and Digium add Asterisk VoIP calling to Smart Cube
Dialogic, Digium team on media gateway interoperability
Digium targets Latin America, South Africa with sales partnership

I’ve created a test branch for patches hidden in several Asterisk development branches - all based on Asterisk 1.4

• RTCP improvements from pinefrog-1.4
• “Sip show chanstats” cli command
• The branch pinequality-* giving you the manager “sipchannel” event to check QoS

This branch is now open for testing and I need feedback. Among the improvements you’ll find:

• Manager QoS events during a call and after a call
• Improved RTCP - it now works for p2p bridge in RTP, which means that we will get RTCP stats for many, many more sip calls
• RTCP over NAT improvements - if Asterisk is behind NAT, we will now kick-start RTCP from the remote end by sending a first “emtpy” RTCP packet to open a NAT port.
• QoS reports to realtime storage after each call - one report per call leg (The amount of data and the names will change)

The reason that I store  QoS data in realtime, is that the CDR is usually gone or frozen at the time that we freeze the RTP channels and get the last QoS data. The QoS reports can’t thus be included in CDR, you have to merge it in automatically later in your database.There’s still a lot to do, but please test it so I get some sort of feedback.For testing, don’t forget to run the “rtcp debug” cli command so you can see what’sgoing on in the RTCP channel.

FAQ

• Yes, this work will be ported to trunk and hopefully merged soon.
• No, we don’t support RTCP XR or MOS in this work
• No, I have no reason or funding to adapt it to 1.6.x at this point.
• No, the RTPAUDIOQOS channel variable is not changed. You will get more data than before - for many more calls.

This work is funded to 20% by companies in the community. If you want to cover the80% that’s still not funded, please contact me by e-mail: oej@edvina.net.
URL:  http://svn.asterisk.org/svn/asterisk/team/oej/pinefrog-deluxe-rtcp-test/

© Edvina AB, Sollentuna, Sweden 2010 VoIP-Forum. All Rights Reserved.

During the last week, I’ve been diving down into the RTCP protocol and Asterisk’s implementation of it. What is RTCP? In short, it’s the way to understand what’s going with your SIP calls on the network.

VoIP relies on the IP network for media transmission

Voice over IP protcols, like SIP, needs to send the media in digital form across the network. The core IP protocols allow for packets to get lost, actually there’s nothing stopping a busy router from just deleting packets it doesn’t have time to deal with. On top of the IP protocol there are two main transport protocols, TCP and UDP. TCP tries to have control of message delivery, so if a router drops a packet, TCP will resend it until it gets confirmation from the other end that the message is received. UDP is like our royal mail service, you have no idea of what happened to the message. If you’re lucky, it will reach the destination.

Interactive realtime conversations are different from downloads

Protocols like HTTP and SMTP for web and email can use TCP and rely on retransmissions, since there is no realtime hurry. A few retransmissions won’t bother you much, but missing data in an e-mail message or web page might be very irritating. For media, especially interactive media, the situation is very different. Retransmissions will not help, since we propably already played the message in your speakers. We can’t reinsert part of a second worth of audio in the media stream after you’ve already enjoyed listening to it. For interactive media, like a phone call, we’re also in a hurry. We can’t wait for missing packets to arrive, since if we delay media too much, the call will break down. You will hear this and if it gets too hard, switch to James Bond walkie-talkie mode. “over”, “over, roger that”, “over and out”…Media for most of the standard VoIP protocols use RTP, the real time protocol. RTP is a way to send media over an IP network. Each message has a time stamp that tells the receiver where the payload fits into the media stream. If a packet is lost, the receiver will discover a glitch in the time stamps. Some receivers insert some noice to prevent you from discovering the issue. Of course, if there are too many packets lost you will notice. 20 milli-seconds here and there will propably go un-noticed in most cases.

Introducing RTCP - the bi-directional reporting system

RTP has a companion protocol called RTCP, the Real Time Control Protocol. This protocol is an out-of-band communication channel between the sender and the receiver. In a phone call, both ends are of course both receiving and sending.  RTCP is used to send reports to the other end, saying “I have sent xxx packets since we started and received yyy packets”. Both ends can then compare data and calculate the packet loss. The reports also include time stamps, so that the round trip time, the time for a packet to travel between the devices, can be measured. There are many pieces of data, and also a standard for extended reports that will deliver a bit more data.If a device, like an Asterisk server, collects this data, we can measure not only the quality of a particular phone call. We can gather data and get hints about issues with SIP trunks to a particular provider, about users on weak WiFi networks in hotels and a lot of other situations. We could potentially deliver this data in real time and alert users when the link is really bad, midcall.There are also advanced codecs that are adaptive to the situation and could use this data in real time. If the network shows signs of problems, the codec can try to change the flow of data - size of packets sent, rate of packets or other properties, like error correction and packet loss concealment.

Project Pinefrog - improvements to Asterisk’s RTCP support

Asterisk today has a very simple implementation of RTCP reports that isn’t very useful, but still a very good starting point. I’ve been working to make it a bit more useful by sending reports over manager, storing quality data in a database and also trying to improve the NAT support for RTCP. I’ve been testing a large number of phones to see how they have implemented RTCP and how Asterisk handles the received data. Hopefully, the turnout will be a large improvement and help us all in getting better and managed quality for our Internet telephony.This work is sponsored by a few companies in the Asterisk community who answered my earlier call for sponsorships.  I am always happy when things work out and Asterisk users step forward and contribute to the process of creating a better version of Asterisk. Being able to get quality data about the calls is a huge improvement for all of us that use the Internet as a transport for our telephone calls. As always your feedback is as welcome as the RTCP feedback on our SIP calls!

© Edvina AB, Sollentuna, Sweden 2010 VoIP-Forum. All Rights Reserved.

Ars Technica offers a detailed guide to VoIP using Asterisk. It starts with a primer on what VoIP is all about and then digs down into how to get things done with Asterisk. Article

A named ACL is an Access Control List that can be manipulated after configuration and live in it’s own name space. The NACL module manage a list of NACL objects that can be used by other modules, like channel drivers, manager and dialplan apps.

Several SIP devices can share the same access control list and there will be one for the whole SIP channel. An external application that reads the security events in 1.8 can manipulate the NACLs in real time through AMI and block/unblock devices. There’s also an API so that Asterisk modules can modify NACLs internally. Applications can be added, so that NACLs can be manipulated through the dialplan. Call in, identify yourself and add yourself to an NACL for the next call…

Amongst the future ideas are NACLs that can be set by referring to a DNS name and use the DNSmgr to stay up to date with DNS. That requires some changes to the ACL.c api that will happen in the trunk version only.

I have also been playing with the idea of having a callback so that an app will know when a NACL is matched or some sort of counters to measure activity per time period and trigger alarms. Kamailio has one implementation of something like this in the pike module.

A lot of security-related ideas for Asterisk has been based on named ACLs, so I thought that was a starting point and a good holiday hack The code is in the deluxpine branches for your testing!

Feedback and comments are, as always, welcome./olle

© Edvina AB, Sollentuna, Sweden 2010 VoIP-Forum. All Rights Reserved.

Yesterday, Russell Bryant finally made up his mind and confirmed on the asterisk-dev mailing list that the next release of Asterisk will be 1.8, which will also be a Long Term Support (LTS) release. This also means that the 1.4 is now officially classed as a LTS release too.

What is a long term release, LTS?

Long Term Releases are releases that are going to be supported for four years. Standard releases, like 1.6.x, are going to be supported for one year, with one additional year of security fixes. This means that the support for 1.6.0 will cease in october 2010. There’s a new release schedule on the Asterisk.org web site that explains the details.

Open Source Projects have to be easy to understand and use

I feel that this is a very good solution for the whole Asterisk community and that we all will benefit from it. I have personally not been happy with the 1.6.x release schedule, which has been very misunderstood and has confused a large group of users. Hopefully, we can now continue with a release schedule that the world understands and that makes sense for everyone. While I understand the need for releasing quicker than we’ve done in the past, the detail about the naming, the actual release numbers (1.6.0, 1.6.1 etc) was very hard to explain to people. With years of experience of doing Asterisk and VoIP training, I have a lot of respect of the need of being able to easily explain things, from configuration details to release schedule…

Time to focus on defining Asterisk 1.8

Now we, the Asterisk community, need to focus quickly on the new release and plan what’s going in there. If you have code for new features lying around (as I have tons of in various branches of my svn repository), now is the proper time to step forward, contribute it to the bug tracker and get it evaluated, discussed and maybe finally included in Asterisk. Whatever goes into 1.8 at release time, will be what we will have for production use for  a long time.

Please dedicate time for testing during Q1 and Q2 2010!

We also ask you to dedicate time during next year to help the Asterisk project with testing. You don’t have to be a developer to test - and we need tests of everything from documentation to configuration and technichal issues. We don’t have all of the equipment you have, we don’t have your dialplans, we don’t have all the applications you integrate Asterisk with. If Asterisk is important to your organization, please make sure that you dedicate time during the first half of 2010 to do regular testing of the new release betas and release candidates. We do need your help to make Asterisk 1.8 a good release, worthy to replace the 1.4 as a new LTS release.If you’re a member of a Linux or Asterisk group, please help in organizing Asterisk 1.8 test-partys. If you need help with ideas, please contact our community liason, John Todd. Meeting other Asterisk users, testing stuff together is one of the best ways to expand your knowledge of Asterisk. Sharing ideas and how-to’s in real time while setting up test labs and scenarious is really, really fun.

Asterisk 1.8 will make a difference

Asterisk has added a lot of new features and internal scalability and stability since 1.4. The 1.6.x releases are to me test releases to show and run practical tests with all of these changes. The core has changed, the API’s has changed and the internal PBX is practically new. We’ve proven scalability to over 10.000 calls on one server. We’ve proven interoperability with many, many products out there. We’ve changed the way we do development and we’ve moved Asterisk into the world of non-PSTN wideband audio. Of course, there’s a lot of more things we can do, but if we consider all of the changes since 1.4, Asterisk 1.8 LTS will be a really cool telephony toolkit.

© Edvina AB, Sollentuna, Sweden 2009 VoIP-Forum. All Rights Reserved.

The Asterisk Development Team has announced the release of Asterisk PBX 1.6.1.12.

This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.1.12 resolved several issues reported by the community, and would have not been possible without your participation. Thank you!

 * Fix multiple issues with musiconhold, which led to classes not getting
   destroyed properly.
   (closes issues #16279, #16207), reported by: parisioa, dcabot, patched by:
   tilghman, tested by: parisioa, tilghman

 * Fix compatibility with valgrind 3.3 and older.
   (noticed in issue #16388), reported by: parisioa, patched by: atis, tested
   by: atis, parisioa

 * Prevent double closing of FDs by EIVR
   (closes issue #16305), reported by: diLLec, patched, tested by: thedavidfactor


 * Send ack (response/message) after receiving manager action userevent
   (closes issue #16264), reported, patched by: dimas

 * Make manager response to "Action: events" finish with empty line
   (closes issue #16275), reported, patched by: vnovy

This release also contains significant improvements to T.38 support. Anyone who has tried T.38 faxing in the past should try again as most problems should now be resolved.

A summary of changes in this release can be found in the release summary:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.1.12-summary.txt

For a full list of changes in this releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.12

Thank you for your continued support of Asterisk!

Dialogic, a provider of media gateways that provide PBX integration between applications on SIP-based media servers and the installed base of TDM and hybrid IP-PBX systems, has just signed an interoperability partnership with Digium.

With this interoperability partnership, Dialogic's 1000 Media Gateway Series and the Dialogic 2000 Media Gateway Series are now certified for compatibility with Asterisk open-source telephony. The gateways will connect with Asterisk through a SIP standards-based interface. The solution means that when there is no direct SIP to SIP integration for an Asterisk system and the existing PBX, Dialogic's gateways can make them work together by providing the necessary signalling and media translation.

For more:
- read this post

Related articles
International contact center provider adds Asterisk to the mix
IBM and Digium add Asterisk VoIP calling to Smart Cube
Skype for Asterisk now available
Digium launches support services for Asterisk

At the first Astricon I was very happy to see Marc Blanchet as one of the attendees. I knew he was one of the IPv6 gurus and wanted someone to show some interest in Asterisk and IPv6.Well, he did not only get interested in it, but started coding on it. The results have been available for quite some time at http://www.asteriskv6.org/ and Marc has tested it at several SIPits for interoperability.

This patch is very large and affects large areas of Asterisk. In order to support IPv6, we need to update the way we interact with sockets, with DNS, with URI’s. The SIP channel needs to handle multiple UDP as well as TCP sockets in both protocols. The ACL’s we use for all VoIP protocols and manager needs support for IPv6. And much more.

Marc hasn’t been able to spend time to keep it up to date with the everchanging trunk. I feel we need to move this forward and try to divide the large patch into smaller pieces that can be reviewed separately by the developer team and  be merged gradually. First, Marcs branch needs a serious overhaul to get up to date with trunk.

In order to work on this, Marc and I needs funding .I have a few interested parties, but need more interested parties that can commit to funding during the first half of 2010 for this project.  It’s not a small task, the current estimate is at least one month’s work for each of us for updating, cutting it up, merging, going through the review process, testing and finalizing with new tests at SIPit or a similar event.

If your organization is interested, please let me know off list and we’ll discuss from there. My e-mail is as always oej@edvina.net. Please don’t hesitate to mail me with any questions you might have about this project.

/Olle

© Edvina AB, Sollentuna, Sweden 2009 VoIP-Forum. All Rights Reserved.

Xorcom, an Asterisk based IP PBX provider, and Broadvox, a VoIP solutions provider, announced that their Asterisk-based IP PBX and SIP trunking services respectively have been certified as interoperable.

Xorcom's devices allow communication using both VoIP and legacy telephony while Broadvox's SIP trunking enables the use of bandwidth for voice and data apps while lowering costs. By using SIP they can provide unlimited local calling and reduce the cost of long-distance and international calls. 

For more:
- read this story

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Customers buy the Asterisk application from IBM’s Smart Market, and rely on IBM for support. Digium support staff is on call to IBM for tier 2 support, but customers only have to deal with IBM.

The PBX software is sold in two sizes, for 20 concurrent calls ($2,000.00 USD) and for 40 concurrent calls ($4,000 USD).

Smart Cube is sold as a hardware platform with a base set of applications on it and a second set of applications available for customers to buy via Smart Market to address their specific business needs.

The Asterisk software can be configure and managed via IBM’s Smart Desk management dashboard, which gives a common look to management of all the Smart Market applications.

Click Here to Read the Complete Article

Digium is the creator, sponsor and driving force behind Asterisk, the most widely used open source telephony software. The company’s product lines include a wide range of software and hardware that enable businesses to implement turnkey unified communications solutions or to design their own Voice over IP (VoIP) systems. Software developers, resellers and telecom professionals choose Digium’s products because only Digium delivers the technical superiority, security and flexibility associated with Asterisk.

Incendonet empowers organizations to add advanced speech capabilities as a seamless enhancement to their IP-PBX and other IT network investments at a price point never before possible for enterprise speech recognition technology. SpeechBridge is the perfect complement to core enterprise applications, and with its integrated speech-attendant, email and calendaring applications customers will begin realizing a return on their investment from day one. With an emphasis on ease of deployment, self-configuring technology allows for SpeechBridge to be provisioned and deployed in under an hour. Now all the benefits of automated speech recognition technology are no longer reserved for large enterprises and call centers.

“We are excited to make SpeechBridge available as a plug and play enhancement to the millions of servers running Asterisk,” said Tim Kruse, VP of sales and business development at Incendonet. “We look forward to working with Digium to extend the capabilities of their customers’ Asterisk deployments with our turnkey speech recognition solutions.”

Bill Miller, vice president of product management at Digium, said: “Incendonet’s SpeechBridge is a SIP-based, enterprise-grade, complete speech solution that we believe will be of substantial interest to businesses using Asterisk. It’s fast to deploy, easy to use and gives cost-effective Asterisk deployments the feel of much more expensive systems.”

Source: Digium & Incendonet

AstriCon 2009 will be held October 13-15, 2009, at the Renaissance Glendale Resort and Spa near Phoenix, Arizona. Business, technical, carrier and advanced Asterisk tracks offer detailed information for a variety of attendees. Registration for the conference is open at www.astricon.net.

“Polycom’s commitment to telephony innovation combined with its expansive line of desktop, conference and wireless VoIP phones has made them a core Digium partner over the years,” said Leslie Conway, vice president of marketing at Digium. “AstriCon attendees—developers, integrators and carriers alike—will benefit from hearing about the latest developments from one of the industry’s most influential players.”

“AstriCon is the place to learn about new and better ways to serve the unified communication needs of today’s businesses,” said Jim Kruger, vice president of marketing for Polycom Voice Communications Solutions. “At this year’s event, we’ll demonstrate new capabilities and applications, that combined with our VoIP endpoints, will give the Asterisk community the ability to deliver an exceptional customer experience.”

In addition to Polycom, other sponsors of AstriCon include Aastra, AG Projects, AMTELCO, Freeside, Infradapt, Loquendo, LumenVox, OpenVox, OrecX, PIKA Technologies, Presence Technology, Sangoma, ScanSource, snom, Vestec and Xorcom.

Source: Business Wire

The Asterisk Development Team has announced several Asterisk-Addons releases, including Asterisk-Addons 1.4.9, 1.6.0.3, and 1.6.1.1. These releases are available for immediate download at:

http://downloads.asterisk.org/pub/telephony/asterisk/

These releases are an incremental release after some community reported issues were resolved, primarily in the MySQL and chan_mobile realms.

* Using chan_local with chan_mobile (issue #15299, affects all 1.6.x versions)

* Don't reset a reconnect time unless a reconnect really occurred (issue #15375, affects all versions)

For a full list of changes in these releases, please see the ChangeLogs:

http://svn.asterisk.org/svn/asterisk-addons/tags/1.4.9/ChangeLog
http://svn.asterisk.org/svn/asterisk-addons/tags/1.6.0.3/ChangeLog
http://svn.asterisk.org/svn/asterisk-addons/tags/1.6.1.1/ChangeLog

Thank you for your continued support of Asterisk!

Digium is the creator, sponsor and driving force behind Asterisk, the most widely used open source telephony software. The company’s product lines include a wide range of hardware and software to enable businesses to implement turnkey solutions or to design their own VoIP systems.

Digium provides the award-winning Switchvox family of turnkey IP PBXs for small and medium enterprises. In addition, for custom implementations, Digium offers both commercially licensed and open source versions of Asterisk as premier development frameworks for application builders looking to leverage the power of Asterisk to create custom telephony solutions.

Since its inception in 1976, AMTELCO has been a leader in call center and computer telephony innovations. These innovations led to the XDS CTI Boards Division in 1980. AMTELCO has provided XDS E&M interface solutions for connection to special PBXs, radio dispatch and wireless communication devices in the police, military, aircraft, call center and healthcare markets. By providing reliable, easy-to-implement solutions, AMTELCO’s global XDS customer base continues to grow.

Jim Becker, AMTELCO vice president and director of the XDS division, stated, “This new partnership between AMTELCO and Digium provides developers with new opportunities when interfacing to radio interface devices, as well as more modern wireless dispatch platforms.”

“This new partnership provides market expansion opportunities for Asterisk developers in new and emerging wireless applications,” said Bill Miller, vice president of product management at Digium. “We welcome AMTELCO to the Digium ecosystem.”

Source: BusinessWire

Digium, developer of the Asterisk open source telephony platform, announced a partnership with DOW Networks to increase Digium's business in Latin America and South Africa. DOW Networks offers international enhanced VoIP services to businesses, and the company has offices in the U.S., Jamaica, Costa Rica and South Africa. DOW already offers Asterisk-based IP PBXs and Digium doesn't have a dedicated Latin American sales operation, so the teaming makes sense.

"As a leading, international VoIP service provider, DOW Networks represents a valuable path to fast growing markets like South Africa and Latin America," said Bill Miller, Digium VP of product management. "We are pleased to expand our already strong relationship with DOW Networks." 

For more:
- see the TMC.net article here 

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Digium moves forward

Available since April, Switchvox SMB 4.0 includes the new Switchvox Extend API. This new toolset lets developers integrate Switchvox with their business applications using an XML API, IVR management tools and event notifications. Utilities such as Fire Dialer, the click-to-call extension for Firefox, or the Switchvox Outlook Plugin are examples of the applications that can be created using the Switchvox Extend API. The newly released API is currently in beta.

Switchvox Developer Central is a website for developers to connect with one another to share ideas and solve problems. It includes a wiki containing all documentation for the Switchvox Extend API, a forum for ongoing discussion, a blog for the Digium engineering team to post news to the community, and tools to simplify development and testing. Digium’s new developer crossroads, http://developers.digium.com, lets users choose their development platform or path—Asterisk.org if they want to contribute directly to the open source software or Switchvox Developer Central if they want to integrate with Switchvox using the Switchvox Extend API.

“The Extend API was one of the most important new capabilities released in Switchvox SMB 4.0 and we want to provide documentation for it in a living format,” said Joshua Stephens, general manager of Digium’s San Diego operations, where Switchvox is developed. “An administrator or reseller of a Switchvox system can integrate their phone system with a custom web application that’s completely tailored to their business or an employee’s job function.

If they have the skills to create the web application, integrating with Switchvox will be easy because they can use whatever programming language they’re comfortable with, so there’s virtually no learning curve or specialized knowledge required. If they’ve worked with any web-based API before, this is going to look really familiar, so they should be able to ramp up quickly.”

Source: Schwartz Communications & Digium Inc.

In order to more closely align the services that Digium provides to the Asterisk open source community with the Asterisk project itself, we've recently renamed many of the servers that provide these services.

Effective immediately:

1) http://bugs.digium.com has moved to https://issues.asterisk.org

 There are no content or functional changes (except for the new site being SSL/TLS enabled), only a renaming of the site. The old URLs will continue to operate indefinitely, automatically redirecting the user to the new site.

2) http://reviewboard.digium.com has moved to https://reviewboard.asterisk.org

 There are no content or functional changes (except for the new site being SSL/TLS enabled), only a renaming of the site. The old URLs will continue to operate indefinitely, automatically redirecting the user to
the new site.

3) http://svn.digium.com has moved to http://svn.asterisk.org

 There are no content or functional changes, and the old URLs will continue to operate indefinitely, *without* redirects, as Subversion does not handle redirects in a transparent fashion and we don't want to break users' existing checkouts.

4) http://downloads.digium.com has partially moved to http://downloads.asterisk.org

 The open source Asterisk project content has moved to the new site, which contains *only* open source content. The Digium commercial products present on downloads.digium.com will continue to be hosted
there. URLs to open source content that used to be present on downloads.digium.com will automatically redirect to downloads.asterisk.org.

Hopefully these changes have been made in as transparent a fashion as possible, and you won't experience any problems. If you do, please don't hesitate to post on the asterisk-users mailing list and we'll try to get the problem addressed as quickly as possible.

Thanks for using Asterisk!

The Asterisk Development Team is pleased to announce the fifth release candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc5 is available for immediate download at http://downloads.digium.com/pub/asterisk/

This release fixes a couple of issues with realtime music on hold that could cause Asterisk to crash, and an issue that caused hungup channels to stay up, leading to 100% CPU usage. Additionally, several minor issues and edge case scenarios have been resolved.

For a full list of changes in this release candidate, please see the ChangeLog:

http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc5/ChangeLog

Issues found in this release candidate can be reported at http://bugs.digium.com

Thank you for your continued support of Asterisk!

Editor's Note:  It's that time again.  Time to get ready for Astricon 2009, the largest official Asterisk Conference for this wonderful open-source PBX. 

Digium®, Inc., the Asterisk® Company, today released details about the sixth annual AstriCon Open Source Telephony Conference and Exhibition. The event brings together open source and telephony developers, systems integrators, entrepreneurs and Digium partners to discuss Asterisk, the most widely used open source telephony platform for creating custom communication solutions. Digium invites those who would like to speak at AstriCon to submit information for consideration by June 1, 2009, at www.astricon.net.

The event will take place from October 13-15, 2009, at the Renaissance Glendale Hotel and Spa near Phoenix, Arizona. Registration is now open and early bird rates are available until July 1, 2009.

AstriCon 2008 proved to be the open source telephony event of the year, attracting over 600 Asterisk enthusiasts, a record number of attendees, for three days of in-depth discussions. This year’s convergence of users, developers, resellers, entrepreneurs and other fans of open source technology will continue the celebration of one of the most influential open source projects with educational sessions devoted to the developing Asterisk ecosystem, trends in Asterisk use, the latest applications and a broad range of technical topics.

Click Here for More Information

Digium offers Asterisk for free download, but until now users had to rely on the open source community or other vendors for help.

The company says customers asked for the service, claiming that their CIOs were interested in the cost savings Asterisk could offer, but leery of lack of support.

The Level 1 service provides coverage for a single PBX server, two support cases per year, discounts on both training and attendance at the Asterisk conference, and response time of 48 hours. The top-tier Level 4 service includes coverage of 10 servers, unlimited support cases, an hour of consultation time and a response time of four hours on calls.

Customers can upgrade their Level 3 and 4 services to add coverage for more servers for US$495 and $395 each, respectively. Level 1 through 3 customers can buy additional support cases for $295 each. The services come without SLAs.

The service is available now.

Source: Computer World

In a surprising move, Digium in partnership with Edvina today released a new channel driver for Asterisk, chan_tweet. The driver connects seamlessly to several microblogging platforms, including Twitter, Facebook, Laconi.ca/Identi.ca and GSM text/SMS. The main feature of this new module is to add audio and video capabilities to microblogging, making the popular microblogging networks a new platform for VoIP and IP realtime communication.

- “I have seen that the microblogging solutions building on the social network infrastructure have had enourmous unexploited capabilities”, says Mill Biller at Digium, “I’ve used it for a long time both personally and for the company and we realized early that by adding IAX2 support, we could now take these platforms one giant leap forward by adding realtime multimedia. I can now spend evenings chit-chatting in audio and HD-resolution video with all my audience around the world instead of sending short text messages. It’s truly awsome!”

Digium contracted Edvina in Sweden, a well-known company in the Asterisk community and long-term Digium business partner, to build this solution. Edvina has many years of experience in building large-scale IAX2 networks, as well as doing development on the IAX2 support in Asterisk.

- “IAX2 recently was published in an IETF RFC and we’re pushing it heavily in all VoIP forums.” says Olle Johansson of Edvina, “We’re hoping that the IAX2FORUM will get a lot of new members that are willing to adopt this technology for their intranets, microblogging services and VoIP infrastructures. In the coming month, we will present more information about new partners with more than 100K users that are going to switch from old technologies, like Hype, SIP and H.323. All of these protocols failed, either because they where proprietary or simply became too complex. SIP currently has more than 5.000 pages of documents describing all the features of the protocol and there’s no single implementation of all of these to test with. Considering the protocol being over 10 years old, this is a sad story.”- “We’ve done our best to fix the Asterisk SIP channel support for customers, but the customer base has been shrinking as more and more converted their networks to IAX2 and now, there’s simply no one interested in us doing that work. We’ve stated over and over again that the SIP channel in Asterisk is broken and no one can prove us right or wrong, because the protocol is just too complex.”

The Microblogmedia platform

The Microblogmedia(TM) platform, developed by Digium and Edvina, let’s users use any microblogging network to set up multimedia sessions. By compressing an IAX2 call setup event in the microblog message, web browsers and clients will connect automatically peer-2-peer if possible, or through the MicroBlogMediaRelay network that supports seamless NAT and firewall traversal by using automatic IPv6 tunnels.Asterisk 1.6.3, released later this month, will support this feature in the IAX2, H.323 and maybe in the old SIP channel (that is now marked deprecated). There is work on adding this feature to ISDN calls, by using messages in the D-channel for tunneling the IAX2 call setup messages. Digium’s VoxSwitch will support this feature in the next release, planned for q3 2009.

Ending the Hype project

In the same press release, Sock Stevens, product manager at Digium finally acknowledged that the Hype channel driver that was launched at Astricon 2008 will not be released after all.

- “We found only one partner to test interoperability with, and that’s not enough to make sure the channel driver being compatible with the protocol. And the protocol wasn’t published in any RFC at all, or any other document. So we finally gave up. We’re now dedicating resources for the new chan_tweet project and enhancing presence support in our IAX2 solution. With the installed base of IAX2 and the new MicroBlogMedia platform, this will be an even more impressive solution, reaching millions of IAX2 users in the enterprise as well as public sector and homes.”

Technichal factoids

• chan_tweet is the result of the project labelled “Codename orangepeel” amongst the development team and builds on the new “Pinemango” architecture. This is the first channel driver not connecting directly to the Asterisk core, but to the Pinemango API over Adversion, the Ruby framework developed by Phil Jaysip.
• The MicroBlogMediaRelay IAX2 platform is an open distributed network that builds on IPv6 and a facebook application, thus using the enormous bandwidth provided for free by the Facebook(TM) platform
• chan_tweet will be released with the core module in Open Source, but with a license exception for plugin developers to add proprietary modules, like the Wireless Village plugin provided by the 3GPP project and the Unistim Microblog Solution by Nertol Networks.

For more information, please do not contact Digium sales.

To be released: 2009-04-01

© Edvina AB, Sollentuna, Sweden 2009 VoIP-Forum. All Rights Reserved.

Coming 10 years after Asterisk's launch, Digium announced the general availability of annual support subscriptions for open source Asterisk. The new service options are expected to give organizations warm-fuzzies when working with the IP telephony platform and include technical support, hardware replacements and substantial discounts on training programs.

"More and more requests from larger and larger companies" were the ultimate deciding factor in rolling out for-pay support, said Digium spokesperson Steve Sokol, with businesses wanting a "throat to choke" if issues came up in implementing and/or working with the software.

Four categories of support services have been defined at this point in time. Level 1, the basic SMB-style package, starts at $595 for a year and includes two trouble tickets with support available during normal business hours; it also has discounts on AstriCon and training classes thrown in. Level 2 provides phone access on a 24x7 basis and 5 trouble tickets/problems for $1995. 

Moving into the enterprise space, Level 3 prices at $3395 and has 10 trouble tickets and covers up to five servers. At Level 4 - - enterprises can cover up to 10 servers and make an unlimited number of trouble tickets for $7995. Level 4 also throws in a free chunk of time for consultative support to set aside time with an engineer to discuss such things as an upcoming rollout of new services or upgrade of software. Level 4 is currently the highest tier of service defined, but Sokol indicated there is another one for developers under development.

Support extends to "any version of Asterisk you want," said Sokol, including binary and the open source code itself. Most of the time, users have one of the latest release versions, with binary distribution problems easier to solve than potentially modified code.

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Skype for SIP is a very different animal. This service provides VOIP trunk support for existing SIP based PBX systems, which may include Asterisk. Unlike SFA where calls may be place to any Skype user, SFS calls may only be terminated to PSTN end points.

So what does this all mean to the Voice/Telco 2.0 marketplace. Overall Skype is beginning to leverage their extensive VOIP network to compete in the VOIP origination and termination marketplaces. Both of these services would enable a SIP based PBX user to utilize Skype as their transport vendor. For example, a traditional SIP PBX customers would directly use SFS for call termination and would provision Skype in numbers to provide origination.

Click Here to Continue Reading

EasyRun, a provider of multimedia contact center solutions to firms such as Coca Cola, the Dallas Cowboys, Pizza Hut and Viacom, is announcing a "PBX agnostic" contact center solution bundled with a full-blown Asterisk IP telephony infrastructure. The EPICAcce (Pronounced "EPICAce") solution is the company's first move into offering its own branded solution to the enterprise space; its contact center offerings have been sold by the likes of 3Com and ShoreTel.

With EpicAcce, EasyRun is coming out from behind the curtain of its OEM heritage to provide an affordable, open standards-based product for SMBs up to 300 people.  The use of Asterisk and other open source/open standard pieces is proving to be both attractive and relatively secure.

"It's a good time to release a new product," said Mike Rose, EasyRun VP of marketing. "Now people are willing to take more risks, they're willing to work with Asterisk." Not that there's anything wrong with the open source IP telephony platform going on 10 years old. "There are a lot of people offering Asterisk solutions today," Rose said.

EpicAcce uses a 2U off-the-shelf Asterisk IP PBX appliance from Xorcom and puts its call center software on top of the full PBX. A basic configuration starts at around $5,000 and goes up as customers add more seat licenses and additional functionality, such as voice recording and cradle-to-grave reporting.

Customers can choose to use the Asterisk IP PBX as their primary phone system or simply drop the box in beside their existing legacy or IP PBX solution for a contact center solution, as the "secret sauce" for EpicAcce is its ability to easily talk to other PBXes. If customers need IP phones, EasyRun is currently supporting snom's line of IP handsets and is working on qualifying other handset manufacturers.

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FEATURE: Microsoft sets its sights on the UC call center market ...

Introduction of Skype for SIP is likely to rattle a lot of pure-SIP providers, but it is Skype's march to ubiquity across the consumer and business fields that bears watching.

To borrow a cliché, Skype is the Apple of the VoIP world.  It talks a good game about playing well with others when it suits its business goals, has a lot of proprietary/closed box technology that it doesn't want people looking at too closely, and makes a lot of PR noise relative to the rest of the unwashed masses.

Apple has made several runs at the business world with mixed results. I expect the same with Skype's efforts. Like Apple, Skype has to build a business story and its story is shaping up to be 1) Having more than 400 million (and counting) registered users and 2) Use us, save money. Like Apple, Skype is in the process of tuning a targeted message for businesses; warm fuzzy colors on the Skype for Business site are out, toned down professional ones are in.

Will Skype end up replacing the need for a standard SIP trunk? Not today, not tomorrow.  First, Skype has to get businesses into the idea that being able to allow customers to call in via Skype is a good thing.  It also has to qualify a bunch of third-parties as "Skype Certified" and that won't be something that happens overnight.

On the other side of the coin, Skype has a good brand and near-instant name recognition, two attributes which a lot of business VoIP providers would love to have. There's also the power of Skype's global reach - one that could prove enticing to many small businesses that thrive on international projects.

Finally, Skype has another attribute that tends to get unrecognized among all of its press releases: steady, methodical persistence. In this, it is more like Microsoft than Apple.  Skype for Asterisk was the warm-up round, Skype for SIP is the first hand using real money, and there are a lot of hands still to play over the months ahead.  Skype has already hinted it has a bigger/better UC card to play down the road and I wouldn't be surprised for it to put down a bigger business video card down by the end of the year; certified interoperability with Polycom and TANDBERG might be an interesting card to be played.

- Doug

Editor's Note:  Here is a follow-up Asterisk help article from my good friend & VoIP Engineer Matt Birkland at VoiceIP Solutions.

A while back I wrote an article for Asterisk VoIP News about routing loops occurring from misconfigured DNS(or lack there of). My article was about routing loops caused by Asterisk attempting to send mail outside the network when the mail server destination is internal.

The goals of this post:

- explain Asterisk voicemail to email routing loops

- LAB: Install Postfix and configure ’smarthost’ for mail relay

Here is an excerpt from my previous post that explains the nature of the problem:

Click Here for the Technical Details

Digium claims to have logged “tens of thousands of hours of Skype-to-Asterisk communication”. They’ve also learned a lot about “the art of connecting Asterisk to with the Skype global network”. The post then goes on to provide more details of the forthcoming offering:

The SFA product will be the only solution that integrates Asterisk directly with Skype. This is not a “proxy” solution and the call quality will be superior to anything else on the market. Customers will have the ability to make, receive and transfer Skype calls from within Asterisk phone systems, using existing hardware and existing Asterisk configurations: Skype calls become just another Asterisk call.

Some of the features that will be supported in the market release are:

1. SkypeIn: Receive calls from the public telephone network using standard phone numbers

2. SkypeOut: Make calls to landline and mobile numbers at incredibly low rates

3. Standard phone features: Incoming/outgoing digits (DTMF), Caller ID

4. Smart call routing based on called Skype Name, Caller ID, country of the caller, language they have chosen in their Skype client and etc.

5. Retrieve Skype credit balance information

6. Store and call PSTN and Skype contacts

7. Retrieve and set Skype user presence information

8. Support for G.711 and G.729 voice codecs

9. Each Skype channel license includes a Digium G.729 codec license

 

Click Here to Continue Reading

The Asterisk.org development team is proud to announce the release of Asterisk 1.6.0.6. This release is available for download from http://downloads.digium.com/.

This release is a significant bug fix update for the 1.6.0 release series.

In addition, this release is recommended for all users of the Asterisk GUI. Two issues with the manager interface have been resolved. The first being with the manager interface improperly handling async. requests from the GUI (see issue #14364). It resulted in manager session file descriptors being improperly destroyed and overwritten. The other being an issue with the Originate action that would cause issues with the GUI. They have both been resolved in this release.

The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help!

* Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the
initiator attribute contains uppercase characters.
- Closes issue #13984. Submitted and tested by: jcovert

* Fix odd "thank you" sound playing behavior in app_queue.c
- Closes issue #14227. Reported and tested by: caspy

* Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2.
- Closes issue #14419. Reported and patched by: klaus3000

* Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage.
- Closes issue #13905. Reported and patched by: jaroth

* Fix devicestate problems for "always-on" agent channels.
- Closes issue #14173. Reported by: nathan. Tested by: nathan, aramirez

For a full list of changes, see the ChangeLog:

http://svn.digium.com/svn-view/asterisk/tags/1.6.0.6/ChangeLog?view=co

Thank you for your support of Asterisk!

As a last important change, we now support the Oreka recording solution as an audio storage medium. This will be loved by those of you running the largest Asterisk deployments, who will be able to factor out call recordings from the main Asterisk server.

There is also a long list of new features and bugs fixed that should make this release interesting for most of you.  This release works with the same activation keys you used on your current version of QueueMetrics.  As always, we value your feedback on how to improve QueueMetrics to make it a better fit to your needs.

Enjoy the update

New features:
- #498: QueueMetrics is now able to track lost outbound calls and will show outbound calls correctly as they are being placed in the Realtime page.
- #586: Qloaderd tool lets you do partial upload of older queue_log data
- #632: It is now possible to use Oreka for audio recordings.
- #605: Now officially using the PolyVox translator's portal
- #612: QM supports AMI version 1.1 for Asterisk 1.6
- #633: New XML-RPC call to export audio files for a given call.
- #575: Broadcast messages are now shown on the Agent's page.
- #576: AGAW information (if present) will now be displayed on the Agent's page
- #577: URLs can be launched automatically from the Agent's page
- #582: Agents can have a visibility key - useful to fully partition QM for multi-tenant systems
- #619: On the Agent's page it is now possible to do logins to all queues at once using ADDMEMBER. Useful for Asterisk 1.6.
- #620: Assisted outbound dialing on the Agent's page
- #621: Custom buttons that can lanch extenal URLs or trigger dialplan functions added to the Agent's page.
- #600: All dialplan hooks of QM now receive the login of the user performing the call - useful for auditing

Click Here for Information

While the company hasn't sent out an official press release, Digium has its web page up for AstriCon 2009. Previous attendees should be pleased at the venue it has chosen, while new attendees will likely care more about hearing the latest and greatest in Asterisk news and knowledge than partying in the Phoenix 'burbs.

AstriCon 2009 will be held Oct. 13-15, 2009, at the Renaissance Glendale Hotel, in Glendale, Ariz. It will be the third straight year the event will be held in the Phoenix, Ariz., area and the second year at the Glendale Hotel. Football junkies can walk next door to look at the Phoenix Cardinals stadium.

However we doubt most attendees will care about the stadium and will be more interested in interacting with their peers in the open source community, catching up with the Asterisk programming team, and finding out if Mark Spencer is still dating Paris Hilton.

And, hey, if you hear anything about a Digium announcement at the show, we'd appreciate any insider tips on Big News you can give us prior to the start of the session; Digium likes to make key announcements at its developer conferences. Two years ago, Digium announced the acquisition of Switchvox.  Last year, it was the Skype/Digium partnership, so we tend to think Digium wants to keep its streak going.

For more:
- Registration doesn't start until March 2, but the AstriCon website is here.

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The company behind SIPphone has deployed a way to gateway between its service and Skype. The OpenSky service is a neat trick, but not novel by any stretch of the imagination.

OpenSky offers free web-based or Gizmo5 client calls up to five minutes. Longer calls are going to require you to buy some annual credits, starting at $20 with a two-hour per call cap and up to 10 Skype names; calls to mobile phones will require some Gizmo CallOut credits. SIP calls can also be forwarded to Skype addresses through Gizmo5.

Interestingly, Gizmo5 head Michael Robertson cites businesses as being the segment most likely to use the service. He also told Om Malik that an Asterisk-based Trixbox solution can't call into the Skype network - something that isn't necessarily true, since Digium announced a Skype channel manager wayyy back at AstriCon this fall. There's also a Java-based SIP to Skype (SipToSis) gateway available for download floating around on the ‘net, so we're not talking unique code or concept here, plus gobs of mobile VoIP players such as TruPhone offer their own gateways to the private-land of Skype-ness.

Digium was/is planning to make a little money by selling licenses for running multiple channels (phone calls) of Skype on Asterisk and (more importantly) bringing organization and call management to Skype users in a business environment, so we're not sure how much extra mileage and $20 bills Gizmo5 will get pushing a relatively unsupported lash-up.

On OpenSky's webpage, there's also this kvetching footnote:

*OpenSky is NOT a service endorsed by Skype/Ebay but they should because it means even more Skype usage.

Didn't Robertson learn how to play nice with people after MP3.com?

For more:
- Om gets the Jedi Mind Trick treatment from Robertson, then recants.  Post.

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Digium has rolled out a seriously hot upgrade to Switchvox in Miami for ITEXPO. The new version includes integrated faxing, video calling support, and XMPP open source IM/chat/presence, along with a bunch of other goodies including introductory API support, IMAP integration for voicemail, a new Windows desktop client, IP phone autoconfig for snom and Polycom phones, and BRI support.

"This release, is jam-packed with big things," said Digium's Tristan Degenhardt. "We can attribute to the fact we have more developers and we're working on the same stuff...fax is finally going in, finally, at long last. We have wonderful wonderful fax integration, video calling support with H.263 and H.263, and we also have chat built in. I would say those were the three things we were lacking, and now they are in."

Fax is added through a third-party technology Digium has licensed, and it is the only part of Switchvox 4.0 that has a licensing cost associated with it. The new release will ship with one bundled T.38 license, with a "sub $50" charge added for each additional concurrent fax call supported.  The fax support includes full auto-detection, so the same number can be used for both voice and inbound faxes with support for both SIP and analog calls.

Video calling support is pretty straight forward, with native support for H.263 and H.264 codecs previously supported within Asterisk; the release is now a "rock solid offering that can compete with big systems," said Degenhardt. Users should be aware that Switchvox won't support transcoding due to the processing requirements, so video calls should be made with the same codecs on both ends.

Chat is supported in a couple of fashions and also adds additional functionality through the use of presence. There's a chat client added into the switchboard feature, and since the chat server uses XMPP, third-party clients such as Jabber, Trillian and Google Talk can also be used. Chat traffic can/is typically kept on the local network by using the Switchvox XMPP server, rather than going outside of the business. Presence information is now provided for both phone and chat, so an in-house Switchvox server or a group of Switchvox peered servers can provide availability information for anyone logged in.

Asked about what to look forward to in future releases, Degenhardt pointed to the Switchvox Extend API. "The API with 4.0 is just in its infancy," she said.  Currently, functionality for creating new extensions and access call logs, reports and extension lists is available, but that's just the beginning. "We would like to expose everything through the GUI through Extend so you can build your own applications."  Third-party developers will ultimately be able to easily reach in and build their own applications for different vertical markets.

Other riches supported in the new release include new IVR functions for building customized applications, IMAP integration for unified messaging, organized phone books, call queue improvements, HD voice support via G.722 and the aforementioned auto-provisioning for snom and Polycom phones and BRI support.

Switchvox SMB 4.0 is available free of charge for customers with a current SMB software subscription. New customers can get Switchvox SMB at the same price, beginning at $3,390 for a 10-user system including hardware, software, a one-year subscription and warranty.

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Digium announced 50 percent growth in Asterisk downloads in 2008 Wednesday, as the company reported 1.5 million downloads of its popular open-source software, up from 1 million downloads in 2007. Digium claims the increase is testament to its reliable low-cost telephony solution. The company also reports increased uptake of the product in September through November, when companies began scaling back budgets in anticipation of a recession.

Digium has had quite a year in 2008, as it integrated the Switchvox acquisition, announced a partnership that will allow Skype users to dial out from within any Asterisk phone system, and delivered its 4 millionth port. Adoption of Asterisk could continue to increase as companies roll out reduced 2009 budgets and search for low cost, high-quality solutions to make up the difference.

For more:
- see the Digium press release here

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• Digium?s Switchvox AA300 ? The award-winning AA300 appliance is a flexible, easy-to-use IP PBX that gives companies with up to 150 users a turnkey business telephone solution based on Asterisk software. Its release completed the initial planned build-out of the Switchvox appliance family, along with the AA60 appliance and the AA350 appliance. After reviewing the Switchvox AA300, Matthew Nickasch, a blogger from Network World, wrote, ?In my opinion, if there?s an appliance or SMB IP-PBX to beat, this is the one.?
• Digium Exceptional Satisfaction Program ? A bold new guarantee program ensuring the quality of Digium?s hardware and software products and underscoring the reliability of open source technology. The Digium Exceptional Satisfaction Program is the most comprehensive product guarantee program in open source telephony today.
• Four Millionth Port ? Digium announced delivery of the four millionth port for connecting telephony systems based on Asterisk to communication networks. The number demonstrates the rapid rise in popularity of Asterisk for managing voice communications and integrating voice with corporate data.
• Sponsored Industry Events ? 2008 marked the second Digium|Asterisk World, an event to advance open source IP telephony communications in business environments. Additionally, Digium hosted the fifth-annual AstriCon, the industry?s largest, most informative Asterisk community event, which boasted expanded track sessions and high-caliber industry keynote speakers. The event recognized the winners of the annual Digium ?Innovation Awards,? which honor businesses using Asterisk in new and exciting ways.
• AsteriskNOW 1.5 ? Digium released a new version of the award-winning AsteriskNOW software appliance that includes the popular FreePBX Web-based Asterisk management interface and simplifies the process of installing, operating and managing an Asterisk-based telephony system.
• Industry Awards ? Digium?s Switchvox appliance family continued to collect honors for product excellence, including Unified Communications? ?Product of the Year? and ?Best of Show? and ?Best of Open Source? at TMC?s INTERNET TELEPHONY Conference and EXPO West 2008. Asterisk won a ?2008 Technology of the Year? award from InfoWorld and a ?Product Leadership Award? from SearchNetworking.com. eWEEK named Digium Founder and CTO Mark Spencer to its ?100 Most Influential People in IT? list and FierceVoIP named Spencer one of 10 ?VoIP Leaders.? As a company, Digium was recognized on Linux Magazine?s ?2008 Top 20 Companies to Watch,? by VoIP-News as a ?Top 20 VoIP Influencer,? and by FierceVoIP as one of the ?Fierce 15? VoIP companies of 2008.

On Friday, the FBI issued a warning about an Asterisk vulnerability being exploited for vishing purposes by criminals. No details were provided, however, leaving businesses to guess and/or rush to upgrade to the latest version.

Posted on December 5 by the Internet Crime Complaint Center (IC3), the Intelligence Note says the FBI has information concerning a new technique to conduct vishing attacks in Asterisk. Without describing the vulnerability or which versions of Asterisk could be at risk, the note warns that it can be exploited by cyber criminals (not to be confused with bank robbers and other ordinary criminals) to use an Asterisk system with an autodialer to make thousands of vishing phone calls within an hour.

The warning implores businesses using Asterisk to upgrade their software to a version that has the vulnerability fixed. We would presume that would mean the latest version, but without details, the G-men really aren't helping.

US-CERT, the national repository of exploits, most recently lists a report for the Asterisk IAX2 channel driver on April 23, 2008, with an update on November 15. But the vulnerability is cited to have caused a denial-of-service attack - not large scale mass-dialing attacks.

Digium thinks the FBI might be referring to a vulnerability found in Asterisk 1.4.18 and other branches reported by MuSecurity on March 18. If properly exploited, the vulnerability would allow an attacker to take over the account of one individual and make thousands of calls in an hour.  A Digium spokesperson notes that the flaw affects older versions of Asterisk but not the last version, 1.6.

We hope in the future the FBI coordinates a bit better with US-CERT and/or affected vendors.

For more:
- FBI issues vague warning about hacked VoIP systems. Article.

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BSDTalk 166 - Listen to the podcast: MP3 | OGG

For those interested in Asterisk on FreeBSD with a lot of preconfiguring already done and a lot of extras, try AskoziaPBX.

As for the purpose of the investment, Philippe said it was mostly due to Bandwidth.com's desire to grow the market and help build the FreePBX community. The idea is that the more IP-PBXs out there, the more SIP trunks, and hence more revenue for Bandwidth.com. I have some further thoughts on this, but I'm pretty busy today and wanted to share the news.

Click Here to Continue Reading 

Bandwidth.com is playing parent to the open-source project FreePBX, hiring the project's developer as its Open Source Community Developer and apparently committing "significant resources and effort" to expand the scope of the project.

An informal announcement of Bandwidth.com's commitment to FreePBX came through chief developer Philippe Lindheimer's blog at www.freepbx.org on November 14 in a post entitled "A Bright Future for FreePBX." Lindheimer said he had "joined forces" with Bandwidth.Com as its Open Source Community Director and indicated both he and Bandwidth.com would work to expand the scope of FreePBX and to assure it remains "open and strong."

Lindheimer cited Bandwidth.com's efforts in purchasing the FreePBX trademark and its efforts with FreeSwitch as areas where the company has been helpful to the open source community. Since Bandwidth.com sells VoIP and data services - not software - there's no fundamental conflict of interest. Instead, FreePBX likely would act in a complementary role in promoting the Bandwidth.com services such as SIP trunking.

This is the latest coup for FreePBX; the web-based GUI provides pre-programmed functionality and ease of use on top of Asterisk, including features such as "follow me," ring groups with calls confirmation, music on hold, unlimited conferencing, and paging and intercom functionality for many SIP phones. Digium incorporated FreePBX into its compilation of the AsteriskNow 1.5 turnkey release in October.

For more:
- FreePBX blog posting praising Bandwidth.com investment.

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I’ve worked with Asterisk many years. I started in 2002 when I was working with a service provider here in Sweden, then co-founded Astricon, started the Asterisk Bootcamp trainings and the dCAP. Many years of working with Asterisk, but almost always in combination with a SIP proxy (mainly SER/OpenSER) and in carrier networks.During the last weeks I have assisted a Swedish company installing an office PBX. This was a new experience for me. Of course, I’ve installed Asterisk for my own use and in the trainings, but this time it was a customer with very well-specified requirements. And I enjoyed every moment.

 Asterisk works very well in these enviroments. It’s almost as if it was built as a PBX. Right. Of course. Sorry. Asterisk is built for this. Exactly this. It’s just that in my work, I’ve used Asterisk as a PSTN gateway, conference server, voicemail server, billing server, session border controller, queue server, IVR server and much more. In those cases, we send a lot of traffic through Asterisk and push it to it’s limits. In the Office PBX market, Asterisk has more than enough power and shines. The flexibility is enormous and the things we can do with just a few lines of code is marvellous.

Working with all kinds of issues in the large scale environments, it’s always important to remember what Asterisk is built for and how well it fits that market. I had a lot of fun configuring this PBX, discovering new parts of Asterisk and trying to solve the challenges from the customer. Asterisk really stood up to this challenge and came out as a shining new powerful sports car, replacing the old PBX.

Of course, I came up with a few ideas that would make this easier. I reported them on Asteriskideas.org - go there and check and report your ideas too!   

© Edvina AB, Sollentuna, Sweden 2008 VoIP-Forum. All Rights Reserved.

Asterisk 1.4 not only adds features to your PBX, it also adds enhanced voice quality for VoIP. The new and improved jitterbuffer implementation covers all RTP-based VoIP channels. Previoiusly, only the IAX2 channel driver had a jitter buffer implementation. (more…)

© Edvina AB, Sollentuna, Sweden 2008 VoIP-Forum. All Rights Reserved.

The Asterisk project has a very strict policy in regards to backwards compatibility. Unless we can’t find another solution, we’re not allowed to remove a function between releases. A configuration for Asterisk 2.4 should work in the next release. In order to be able to change functionality we warn users in one release and then remove the functionality in the coming release. So a configuration in 3.0 works in 3.2 but maybe not in 3.4.

This article tries to provide help with known problems with upgrading. Read on to learn how to avoid the traps! (more…)

© Edvina AB, Sollentuna, Sweden 2008 VoIP-Forum. All Rights Reserved.

Asterisk 1.4 delivers many new features. In regards to call state subscriptions, there are many news for you. Call state subscriptions are what makes the lamps blink on your phone when your collegue’s phone rings. In 1.4, you can make it blink based on activity in parking lots and meetme conferences as well. Read on! (more…)

© Edvina AB, Sollentuna, Sweden 2008 VoIP-Forum. All Rights Reserved.

Asterisk 1.4 introduces a new level of Jabber integration, developed by Matthew O’Gorman at Digium. The Asterisk Open Source PBX integrates with Jabber/XMPP in many ways. (more…)

© Edvina AB, Sollentuna, Sweden 2008 VoIP-Forum. All Rights Reserved.

For a long time, we have needed a platform for managing feature requests - things that the community or developers would like to see in Asterisk. We used to have a “feature request” category in the bug tracker, but there was no good way to handle them in the bug tracker and they where in the way for the work done by developers in the tracker. They ended up getting closed, only to be reachable by searching closed bug reports. Not a very good solution for brainstorms and good ideas.

The new site is basically a blog with comments and voting capability. You register on the site to be able to file a feature request. Other people may then add comments or vote for requests.

Hopefully, this will be a repository of ideas and a good discussion platform. Things will be stored and accessible. As usual, filing a feature request is not a guarantee that anything will happen. You still need to make sure developer resources are put to it somehow.

Please also remember that it’s not a support forum. You can’t get help in the idea repository. There are already mailing lists and forums in place for that.

Let’s try this out for a while and see if it’s a good tool that works for us. Register for an account on www.asteriskideas.org today!

Thanks for any feedback!

/Olle

© Edvina AB, Sollentuna, Sweden 2008 VoIP-Forum. All Rights Reserved.

Happy New Year, Asterisk community!

During 2007 we accomplished a lot. We polished Asterisk 1.4 to a new state of readiness for production. We moved Asterisk 1.2 into no-maintenance mode, something I think we might have to reconsider after a discussion on the Asterisk-users mailing list. And we did not release anything new. Which is a good thing.

Why is that a good thing? Well, in an Open Source project you can choose between many different release strategies, depending on the software. Asterisk is a PBX. A PBX is in most cases something you don’t upgrade unless there’s a need to. We see that on the slow uptake everytime we release a new version of Asterisk. One year after the release of Asterisk 1.4, most of the installed base seems to run Asterisk 1.2. And they’re happy with it.

The problem is getting new features out there. We have a policy of not introducing new features to a released version of the software. That means we’re forcing people to upgrade to get new features - and new bugs. Would it be possible to create a new module interface so we can release various modules independently of the core? I don’t know, but that would create more complexity at the same time as it gives us a bit more flexibility for upgrades. At this point, 1.6 modules will not run in an 1.2 or 1.4 environment.

Anyway, I just wanted to write a note to say Happy New 2008! During this year, we hope to release a new version of Asterisk. During next year, you might be interested to put it into production. We developers just have to realize that it takes an awfully long time from idea to implementation in real life in the Open Source PBX market.

© Edvina AB, Sollentuna, Sweden 2008 VoIP-Forum. All Rights Reserved.

During the last week, I’ve been discussion upgrading Asterisk on the asterisk-user’s mailing list. I asked the community on what the problems was with upgrading to 1.4, asking for success stories as well as reasons not to upgrade. I’ve learned a lot from the feedback.As a developer, I spend too much time in the bug tracker, working with particular bugs, so I often wonder howon earth anyone can use this buggy platform for anything business-like…With that background, it really feelsgood to get reports on people successfully using our software and meetAsterisk users who just love the product and handle tons of callsevery hour with it.And as a developer, everything is of course more simple and you live inthe future, moving forward to new features, new functions all the timebasedon customer requirements or feature requests in the mailing list or thebug tracker…

Summary of the feedback

Now over to a summary of the feedback. I’m not going deeper intobugs reported, those will be handled separately.

DON’T TOUCH MY ASTERISK PBX

For a lot of users there’s simply no reason toupgrade a PBX everytime we release a new Asterisk.Existing installations that work should not be touched unless there’s a very good reason to, like a new feature that makes business sense.Just upgrading for the cause of upgrading is a feature of the non-open software industry that gets a lot of revenue from upgrades.We developers has to accept that people appreciate our work, but decide not to upgrade every installation at every release. We might have to reconsider our support policy in the Asterisk.org project, where wedevelopers abandoned 1.2 this summer. We might need another team that runs 1.2 support in the bug tracker.

MAKE UPGRADING EASIER

Another issue is to make the upgrade much smoother. We can’t anticipate that people upgrade from 1.0 to 1.2 to 1.4 and readall the docs for every release. They can jump from 0.8 to 1.4. Or 1.0 to the future release of 1.6. We need to assist that and haven’t made a good effort in doing so.But even for upgrades from 1.2 to 1.4, we need to be more clear about changes that are required, especially for 1.2installations that already was upgraded from 1.0 and still use the 1.0 configuration syntax. They are going to havea broken configuration in 1.4 and this is the first time that happens in Asterisk.We need to make clear that Asterisk admins need to go through the log files in 1.2 and check all deprecation warnings. These needsto be fixed before even testing 1.4.

USE ASTERISK 1.4 FOR NEW INSTALLATIONS, PLEASE

My personal goal would be to get the community to start using 1.4 for all new installations. We need to produce informationto help this upgrade path. It’s not about upgrading systems, since we’re talking about new installations. It’s about upgrading the Asterisk admins and installers - human beings.

The success stories reported to me personally and on the list indicates that 1.4 is indeed ready for production and it’s a great product.

With that, I’m now changing my focus from SIP invite states, RTP sessions and video formats to Christmas ham purchasing, baking Christmas bread (julvört) and decorating the Christmas tree. Of course, you understand that there’s an Asterisk asterisk on top of all those trees, right?After Christmas, I’m running the new Asterisk SIP Masterclass together with Daniel Mierla here in Stockholm. He’s one of the core OpenSER developers and it’s going to be a great class. I’m sure we will locate a set of new interesting bugs in svn trunk during that week. I’m really looking forward to that training. (Hint: We still have a few open seats… )Greetings from a dark and cold place in Sweden, without a decentamount of snow…Have a wonderful, merry and cheerful Christmas!/Olle

© Edvina AB, Sollentuna, Sweden 2008 VoIP-Forum. All Rights Reserved.