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.

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

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.

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.

KATHMANDU: An investigation team dispatched from Metropolitan Police Crime Division raided a well-equipped underground call centre in Ravibhawan and arrested a racketeer for his involvement in flouting telecommunication laws recently.

The accused Anwar Hussein (24), who hails from Kolkata-24 Porguna, India, and his two accomplices were found to be operating VoIP (Voice over Internet Protocol) from a rented three-storey building, illegally blocking Nepal Telecom’s gateway, thus inflicting a loss of millions. While, the other accused, Bharat Lal Shrestha (27), hailing from Chaugada-3, Nuwakot, and another named Raju are at large. Raju’s details were still sketchy.

What is VOIP Gateway?

Call by-passers make use of VoIP GSM Gateway to divert international rings from the official gateway. The call is then transferred to the telecom subscribers through a GSM SIM card. The ISD then displays a personal caller ID on the receiver’s gadget. VoIP has always been a headache for telecom service providers in the country. As per the Telecommunications Act, 2053, any person convicted of posing threat to telecommunications systems and service can face a fine equal to the principal amount of loss caused or sentence up to five years in prison or both.

VoIP uses broadband Internet for routing phone calls, unlike conventional switching and fibre-optic alternatives.

Source:  Himalayan Times

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.

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.

Maybe there is something to this idea that, in a recession, Voice over IP service is an affordable alternative to traditional telephone service.

In the first half of 2009, VoIP services brought in nearly $21 billion in revenue, with both residential and business services looking healthy and poised for even more growth for the second half of the year, according to a report by market research firm Infonetics Research.

Residential voice services still brings in the majority of revenue, with the number of subscribers growing 14 percent from the end of 2008 through the first half of ‘09. On the business side, the research firm said it expected IP Centrex and hosted unified communications service revenue to grow 26 percent year-over-year.

But the current sweet spot, at least in North America, is small businesses with fewer than 100 employees. In the first half of the year, roughly two-thirds of all IP Centrex seats sold went to small businesses.

Click Here to Continue Reading

Editor's Note:  WOW OMG YAY THANK YOU AT&T - Do I really need to say more.  This is a landmark decision and kudos have to go Ma Bell for finally giving your loyal customers what they want.  With the direction the trend is going, people want data in their pocket and they want to be able to do what ever they want if they are paying such a premium for the iPhone.  Hope everyone is happy, because I am, time to get my Skype number ready.

AT&T today announced it has taken the steps necessary so that Apple can enable VoIP applications on iPhone to run on AT&T’s wireless network. Previously, VoIP applications on iPhone were enabled for Wi-Fi connectivity. For some time, AT&T has offered a variety of other wireless devices that enable VoIP applications on 3G, 2G and Wi-Fi networks. AT&T this afternoon informed Apple Inc. and the FCC of its decision.

In late summer, AT&T said it was taking a fresh look at VoIP capabilities on iPhone for use on AT&T’s 3G network, consistent with its regular review of device features and capabilities to ensure attractive options for consumers.

“iPhone is an innovative device that dramatically changed the game in wireless when it was introduced just two years ago,” said Ralph de la Vega, president and CEO, AT&T Mobility & Consumer Markets. “Today’s decision was made after evaluating our customers’ expectations and use of the device compared to dozens of others we offer.”

AT&T allows customers to download or launch on their wireless devices a multitude of compatible applications directly from any lawful Internet website. Additionally, because AT&T uses GSM technology, the most pervasive and open wireless technology platform in the world, we support customers using any GSM phone that works on AT&T's frequencies.

Source: BusinessWire

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

Discussing Verizon’s second-quarter results, which included a 7.2 per cent fall in net income to $3.2bn, John Killian, chief financial officer, said: “Clearly the broader economic issues are affecting the business.

“Although we are taking steps to mitigate the negative impacts of the economy in the short term, we also need to more ­significantly reduce the wireline cost structure over the next 12 to 18 months.”

Operating revenues in Verizon’s global enterprise business segment, which mainly serves big companies, fell 6.7 per cent to $3.7bn as customers reacted to the downturn. Wholesale revenues fell 7.5 per cent to $2.4bn.

Revenues in Verizon’s fixed-line business fell 5.2 per cent to $11.5bn in spite of growth in fibre optic-based video services and broadband services. The number of fixed lines served by Verizon fell by a further 630,000, or 12.3 per cent, to 19.7m, mostly reflecting wireless substitution.

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ChannelWeb first reported on the issue in May, when Microsoft layed off much of the group behind Response Point, and wouldn't commit to future development of a 2.0 release.

This week, ChannelWeb reports that Response Point Program Manager John Frederickson told a town hall meeting with VoIP resellers that the company doesn't currently plan to release future versions of Response Point but will continue to maintain the product and evaluate specific feature requests.

Hard to know exactly what that means, but if I were looking for a VoIP PBX for my small business, I'd be looking for a little more reassurance about Response Point's future. Otherwise, I'd keep looking elsewhere.

According to ChannelWeb, Response Point is "a full-fledged IP PBX system designed for organizations with up to 50 employees." It boasts an "affordable price tag and robust feature set, which includes SIP trunking and click-to-call functionality via Outlook."

See Microsoft's Response Point site.

Source: Bmighyl

Editor's Note:  I want to bring attention to a free book offer "Creating RFP's for IP Telephony Systems" that is being presented by VoiceIP Solutions.  The book covers the process that goes into creating a RFP when getting bids for a new phone system / PBX.  This is ideal for the IT Manager or Telecommunication System Director that has been tasked with getting competitive bids from various vendors.  If you use this book and it does come in handy for your PBX project, please send me a note and let me know.  Enjoy.

Here is what Amazon.com states the book covers:

· What are RFPs and RFQs
· Why use and RFP for IP Telephony Systems
· What are the Key RFP Objectives and Processes
· How to Identify Company Communication Requirements
· Who is involved in the Creation of an RFP
· The Typical Steps in Creating an RFP Document
· How to Issue and Manage RFPs
· Evaluating RFP Responses
· RFP Communication between Issuer and Responder
· Outline Template for a typical RFP 

 Goto the following link and fill out the information form and a book will be send out within 7-10 business days (While supplies last):

VoIP & IP PBX RFPs Book Offer - VoiceIP Solutions

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

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!

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.

During 2008, I worked with my Portuguese business partner Wavecom to build a large installation of Asterisk and OpenSER in Portugal. The network is now running 300 servers, 250 of them running Asterisk, FreePBX and OpenSER. It’s handling 200 connections to legacy PBXs of all kinds and over 10.000 phone numbers. During 2009 we’ve developed a failover solution, to make sure that every FreePBX server always stays up, regardless of the state of the hardware.

We’ve migrated 34 Universities to use Open Realtime Communication with Open Source. They are now migrating their user base from old PBX phones to modern VoIP phones.

Want to learn more?
Come to Amoocom in Germany, May 4-5, 2009 and listen to my talk!   And if you need help with similar large-size projects, contact me on info@edvina.net!

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

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

AT&T Inc Mobility announced on Monday it will provide FEMA workers with RIM BlackBerry mobile phones and laptop wireless devices as part of a $50 million solution contract. The smartphones will be powered by AT&T's nationwide EDGE network while its LaptopConnect will run on a similar Wi-Fi network.

The FEMA contract is good for one-year with an option which could be extended up to four years. "A secure wireless communications infrastructure is critical for first responders and their emergency management operations," said Don Herring, senior vice president, AT&T Government Solutions.

AT&T will become FEMA's primary wireless carrier to deliver secure wireless data and voice communications. FEMA workers will use the new mobile services to utilize e-mail, data applications, and Push To Talk voice communications while in the field. Workers will use the BlackBerry smartphones and wireless laptop cards to connect to AT&T's LaptopConnect service.

"With the proven reliability, bandwidth and speed of our network, AT&T Government Solutions can help FEMA employees access the information they need for real-time decision making in the field while ensuring they can collect, access and transmit this information in a secure environment at all times, regardless of location," Herring said.

Click Here to Continue Reading

I was among those lucky enough to have an account with GrandCentral before their acquisition by Google in 2007.  Actually, I have a couple, but have mostly used one as my primary business number for the past two years.  As a web developer, being able to check my voicemail and manage my calls just a click away from my inbox has done wonders for my productivity.  It also makes screening the numerous daily calls I get about outsourcing my projects much easier to deal with

But what’s more fun than a new(ish) Google product?  Complaining about all the features it doesn’t have!

Click Here for the Feature List

"The significance of this patent is far reaching," said Donovan Jones, President and CEO of CounterPath. "Architecturally we made some technical choices that make handover faster. Also, because we leveraged standards-based protocols like SIP, our technology works with virtually any network architecture and technology without requiring proprietary access points or clients. This approach works in both pre-IMS and IMS networks."

Mobile operators who integrate CounterPath's Network Convergence Gateway (NCG) into their offerings will be able to leverage the Internet as an extension to their network reach, giving their end users access from any Internet-connected location. Enterprise customers can utilize the NCG to access their PBX from the mobile network regardless of whether they are connected via IP or GSM/CDMA networks, enabling their workforce to use their Wi-Fi-enabled mobile phones virtually anywhere in the world without having to pay expensive roaming fees.

Typically, mobile operators only receive a small percentage of the roaming revenue when their customers roam in foreign networks but if the call takes place over Wi-Fi they receive all the revenue. In addition to significantly improving their roaming revenue while reducing costs, mobile operators will also be able to pass on savings to their subscribers, thereby increasing their competitive advantage in the marketplace.

This patent is the latest in CounterPath technology to enable seamless, robust and cost-effective IP-based communications. CounterPath has 26 patents issued or in progress.

Source: PR Web

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

Editor Note:   I found this interesting.  According to this source, a satellite communications satellite collided in orbit with a non-functioning Russian satellite.  Ouch.

February 10th , Iridium lost an operational satellite. According to information shared with the company by various U.S. government organizations that monitor satellites and other space objects (such as debris), it appears that the satellite loss is the result of a collision with a non-operational Russian satellite. Although this event has relatively limited impact on Iridium’s service, the company is taking immediate action to address the loss. The Iridium constellation is healthy, and this event is not the result of a failure on the part of Iridium or its technology.

Click Here for More

Online Gadget retailer Firebox is selling what it calls the first 3D webcam int he world and it is available for only £49.95; the Minoru even comes with five pairs of 3D glasses (ed : when will they have stylish 3D glasses?)

The webcam looks like a single legged red-aced alien with arms protruding. Minoru works the trick by combining two USB webcams altogether and adjusting the red and blue levels to create a near perfect 3D video stream.

Obviously, the trick works best if the other party also has a similar webcam. Minoru is compatible with existing messaging programs like MSN or Skype and you can always upload pictures of your antics on Youtube.

Don't try to convert a 2D movie into a 3D one using this technique though or you could be in for a rather nasty headache. Minoru can still be used as a 2D webcam if you or your friends do not have the glasses.

Source: IT Pro Portal

According to Nortel:

“Consolidating our voice infrastructure to VoIP will provide network efficiencies that serve to increase productivity while driving costs lower,” said Steve Bandrowczak, chief information officer for Nortel. “In the initial phase of deployment alone, we anticipate significant operational savings with a return on investment in a little over a year. The endgame is that it allows us to focus on our business, which is serving our customers.”

Source: VoIP-News 

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 

Jajah has launched Jajah Direct in the Czech Republic, Hungary, Poland, and Slovakia. The new service will allow its users to make low-cost VoIP calls without a PC or special phone or headset. From TMCnet:

This solution from JAJAH has successfully triggered what is considered a revolution in global calling. In fact, 10 million people in America, England and other parts of Europe are already using it to save on their phone calls. JAJAH offers the ability to combine the savings of an Internet call with the convenience of being able to use a normal phone. Traditionally, VoIP calls were only available with an Internet connection. Now, all consumers in Slovakia, Hungary, Poland and Czech Republic can make low cost, high quality VoIP calls with the JAJAH Direct solution.

T-Mobile is close to launching their T-Mobile@Home VoIP service, which is expected to compete with AT&T, Verizon, and Vonage. The new service will start at $10/month for unlimited nationwide calling. From the NY Times:

T-Mobile@Home transmits calls from the phone handset through a $50 T-Mobile router to the Internet, where voice-over-Internet-protocol technology is used to complete the call. Customers are allowed to keep their existing home phone number. At the same time, said Chief Executive Robert Dotson, they can get the same services they get from mobile phones, such as personalized ring tones.

TwitterFone, a voice-to-text-message service for Twitter launched just earlier today. From their press release: “Twitterfone voice-enables Twitter, a text message rebroadcast service and the hottest social networking service at the moment. With Twitterfone, people can dictate text messages via their mobile to be sent out to everyone on their Twitter social network.”

Yep, that’s right… 10 million now for JAJAH. And in other news… JAJAH has also just announced that they have opened their platform to third parties with their new JAJAH Managed Services initiative. Yahoo! is the initiative’s first customer. More from Alec Saunders…

Garret Smith writes some thoughtful words about WiFi phones in his blog:

Infonetics Research released a report that showed WiFi ip phone sales increase 60% in 2007, with 682,000 units were sold worldwide. The report cited ?increased vendor support? as the primary reason for the growth. 

[…]

While some people seem to like WiFi phones, they aren?t for the faint at heart, especially if your aren?t technically savvy. My advice, if you want to go wireless, is to pickup a DECT based solution. A little more expensive, but it works?for everyone. 

I still haven’t met a WiFi SIP phone I liked and used more than a couple of days…  

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

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.

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.