VoIPtel Blog

This was definetely not what I was planning to write about! We have had a lot of things happening lately, but they all have to wait in order to make room for this article that I found through Twitter!

Just when you thought it couldn’t get any worse in the world of software patents, a reliable source sent me this response from Steve Jobs about a patent pool that’s forming and aiming to nail the open source codecs projects. It’s enough to make the weak at heart go weak in the knees and stop working on open source projects.

Here is Steve Jobs’ response to the letter from Hugo Roy:

From: Steve Jobs
To: Hugo Roy
Subject: Re: Open letter to Steve Jobs: Thoughts on Flash
Date: Fri, 30 Apr 2010 06:21:17 -0700
All video codecs are covered by patents. A patent pool is being assembled to go after Theora and other “open source” codecs now. Unfortunately, just because something is open source, it doesn’t mean or guarantee that it doesn’t infringe on others patents. An open standard is different from being royalty free or open source.

Sent from my iPad

You can read the story here!

Original letter sent to Steve Jobs

We are only two weeks away from the AstriEurop Exhibit in Paris (April 14-16) and I though it would be a good idea to give you a preview of some products that we are planning to bring along ;o)

The RP08 is finally here! A versatile SOHO/Residential Router and PBX appliance
PBX: Open Source (Asterisk)
High performance OSLEC (Open Source Line Echo Canceller)
Configurable IVR menus
Voice Mail,Voicemail to Email
Call forward, call waiting, call transfer
Call conference
Call queues
SIP/IAX2 trunking
8 analog ports (FXO/FXS)
Call Detail Records
Access via: SSH, HTTP
30 concurrent calls

Router: Linux-based (kernel 2.6.32)
Automatic MDI/MDI-X crossover on all ports
Full 100Mbps supported on all ports
802.1Q tag-based VLAN (16 VLANs, full range VID)
VLAN ID and 802.1P tag/untag option per port
QoS/CoS per port 802.1p and DiffServ-based
802.1D spanning tree protocol support
Extensive MIB counter management support
802.11b/g/n WiFi support with optional EnGenius EMP9602 Card

VoIP X
It is some time since I gave you the first preview of our upcoming VoIP X Appliance, but even though it has been ready for release since December 2009 we decided to wait until the AstriEurop Exhibit before we made it available. This has given us ample time to test it properly with various Blades manufactured by Positron in  Canada, and we will now start working on software for the Server Blade that will turn it into a powerful and easily upgradeable TCS
(Total Communications System) or “UC on steroids.” ;o)

Atcom’s IP-4G, IP-2G4A and IP-2G4B
I got a call from Andres of EuropeSIP some time ago telling me that they were preparing an Exhibit in Madrid, Spain. He was asking for news about the latest Atcom PBX’s, so I decided to reroute my own IP-2G4A and get it shipped directly to them from the factory. As a result I still haven’t been able to test it, but hopefully it will arrive from our Partner in Spain within the next few days. In the mean time, have a look at these photos of the IP-4G:

The PIKA WARP appliance is available with different levels of enablement depending on your business needs.

As a development platform you have a lot of flexibility to customize and add your own telephony application. With the FreePBX image already loaded, PIKA has  done all the integration for you so you have a customizable PBX solution to offer your customers.

Review the profiles below to see which best describes your business.

Developer; Highly technical
Comfortable to write in C+ or other
Comfortable with a Linux environment
Contribute to open source software
Possible embedded experience
Comfortable with PADS (PIKA Application Development Environment)
Comfortable with forum-type support
Expectations for time to market is typically measured in months
Business model based on value add of developed software plus support contracts

VAR/System Integrator/Value-added SIP Provider
Integrate (hardware/software) systems for end customers
Purchase and resell complete systems
Some technical capability and knowledge of Asterisk but not C+ code writing
Possible capability and knowledge of Web development and PHP code
Expect and require technical assistance from vendor/manufacturer
Expectation for time to market is typically measured in days/weeks/months
Business model based on margin, installation and service contracts
Technical capability is at the configuration and programming level

PIKA WARP the Appliance for Asterisk® is ideal for developers looking for a small, low cost computer replacement to deploy Asterisk based applications in the Small Office/Home Office (SOHO) and Small/Medium Enterprises (SME) markets. Unlike the other PBX’s that we are normally working with it is 100%  i386 compatible. What this actually mean is that you can basically run the same applications on this cute little box as what you can do on a full size PC!

Completely customizable, it is compatible with VOIP phones as well as analog sets. Unlike your typical computer or appliance, PIKA has covered all your customer’s traditional telephony requirements. Music on Hold (MOH) and Paging can be cumbersome to add to a data centric solution as is power failure transfer (PFT), but all are included in the PIKA appliance. The configuration of the appliance is modular and can include up to 9 ports of a combination of FXO/FXS/BRI plus VoIP stations and trunks. And Yes; the GSM module is currently in BETA testing and expected to be available within the very near future! The appliance is designed to address businesses with up to 75 IP endpoints (trunks and stations) and 45 simultaneous calls.

For anybody who has used FreePBX this is a dream come through! Our box came with the Asterisk GUI, but upgrading to FreePBX was really a breeze! Copy the firmware to a USB memory stick, turn off the appliance, insert the USB stick in one of the USB ports and restart the PBX. About 20 minutes later it was running FreePBX and was ready do download the latest updates as well as new features over the internet. And it works exactly the same way as what you are used to from trixbox and FreePBX.

In addition to the standard features that you find on all the PBX’s in this roundup you have a built-in FXS and the capability of adding two modules. You first have to brake off the cover plate made of plastic after removing the top cover which s held in place by two screws on the side panels. The modules has a nice touch that I would like to mention. The FXS module has four ports and one FXO while the FXO has four ports and one FXS. This comes in addition to the FXS port integrated as standard on the PBX.

Another nice feature is the Sound In/Out and the display window on the front telling you the units IP address when you apply pressure to the right side of the display.

The sound quality is very good, and the small fan won’t disturb you even if you keep it a few meters away from you desk (or bed). Although the modules are a bit on the pricey side compared to similar modules from Atcom it is still a real bargain specially considering how easy it is to get it up and running as well as maintaining it. Try it, I am sure you are going to love it!

UPDATE:  Two new GSM modules was released a few days ago, one with capacity of 2 SIM cards and the other able to hold 4. Both are available from our shop!

Next: Meet Positron Telecommunications; Embedded Appliance as well as complete PBX on a PCI card!

Have you ever been looking for a VoIP provider that support IAX2? Or a provider that will have complete configurations for PBX’s manufactured by Atcom, Positron, PIKA WARP or Tinyworx in their firmware? With better rates for “Pay-as-you-go” services than most companies offer for a monthly fee?

We are currently testing the services of what looks like a very promising wholesale provider. I am expecting the test period will be finished within the next few weeks after which we’ll sign up and invite you all to join!

The Basics

You connect any kind of device; server, switch, or software that supports one of the 2 protocols we offer (SIP, IAX2) and at least one of the codec’s offered to our server. You can authenticate by IP address or dynamic registration, which is supported by almost every piece of VoIP software and hardware on the globe. You can configure multiple devices with different usernames by using our sub account section. When you place calls, we terminate them for you. You can also order DID numbers. When people call these numbers, we send them to you via VoIP.

Before we make our services publicly available we will need a number of people to test it out from different parts of the world. If you would like to have one of those test accounts please send a mail with some informations about where you live, what VoIP service you use today and why we should give you one of our test accounts. Please send the email to
post@voiptel-pro.com, write “TEST ACCOUNT” in the subject field.

For some time now we have been eagerly waiting for a new line of Atcom PBX’s that can handle one or more GSM modules. The wait is finally over, the first few units has already come of the assembly line and will be available through our web shop in the very near future. There are currently 3 models available, all based on the same motherboard and steel chassis; IP-4G (4 GSM), IP-2G4A (2 GSM 4 Analogue) and IP-2G4B (2 GSM 4 BRI). Another model with 4 Analogue and 4 BRI (IP-4A4B) will be available in the near future.

Prototype PBX

GSM module

BRI module

UPDATE: Just got a call from Ruben, he has shared his thoughts about these PBX’s here. As always, it’s well worth reading!

And for once I am not talking about the size around my waist! The last couple of months I have been working on a new concept, a network of Partners all over the world. As a small company you meet all kinds of problems such as high prices, costly freight, none or very limited influence on the products that you are marketing. While the large corporations have the resources to develop fancy equipment carrying their name and logo the small companies is left with the rest. But guess what, we are about to change that! With the support of a lean organization handling both ordering as well as shipment to the various Partners we are able to cut cost and grow influence. We still need more members in order to have a real impact, at the moment we have Partners in Norway, the Netherlands, United Kingdom, France and USA (east coast). And another four companies is going to join within a week or so. I believe there is strength in numbers, and we will use our strength to the benefit of our customers.

UPDATE: We have decided to participate at AstriEurop together with our  French Partner Analytel. For details please click here.

For inquiries about our global network of Retailers and Partners please go here.

2009 was a tough year, every time I turned on the TV or sat down with a newspaper there was new disasters, bankruptcy’s, rising unemployment rates, the works. Most companies scaled back, research and development of new products was put on hold. But not every company followed the trend and put their heads in the sand hoping the international crisis would go away. I know of at least three companies that did the opposite, that invested time and money in the future of communication the Asterisk Way. In some ways their products are very similar, yet they are so different that they are the champions of different categories altogether. I got a sample of each to play with, and I’ll try to share my experience.

First one out is my old friends at Atcom Technology, the manufacturers of David Rowe’s open source IP-04 PBX. Who could have known that David’s basement project would become so popular that it changed the entire price structure for large (and obsolete) PBX manufacturers and finally put the PBX within reach of even the smallest companies. From an average price of $5000 or more you can now have the same features packed in the size of an ATA for $135! The IP-04 is no longer alone, it has grown to an entire family of PBX’s with no end in sight. And they are GREEN, with an average consumption of less than 5W it can even been run from solar power! The only negative sides of this incredible device is the fact that it has taken us way too long to make the firmware stable enough for professional use in addition to the fact that the person given the task of setting it up should have some experience with Asterisk. Or get access to direct assistance through an SE Support Contract.  But for the seasoned professional it is a dream, you can even develop your own version of the firmware.

As I mentioned earlier Atcom has developed an entire family of PBX’s sharing the same basic design and hardware with the following specifications:

Interface
1 or 2 RJ45 port(s)
1 x Power port
1 x RS232 port
1 to 8  RJ11 port (FXS/FXO interchangeable)  or 4 RJ45 (IP-BRI)
1 to 4 FXO/FXS module slot (Except IP-BRI)
1 slot for SD FLASH  (not IP-01 and IP-02)

Hardware
CPU: 400MHz Blackfin BF532 Chip  (IP-BRI use a 600Mhz BF537 chip)
NAND flash 256 M
SDRAM 64M

System
Open Source uClinux with Asterisk firmware, VoIPtel CE by default  ( Optional: VoIPtel SE, BAPs or Astfin)

Size
100 * 100 * 28mm    IP-01 and IP-02
225 * 120 * 30mm    IP-04, IP-08 and IP-BRI

The graphical user interface (GUI) is a modified version of AsteriskNOW GUI v2, so users of AsteriskNOW will feel right at home. It is unfortunately very common that open source projects often has either none or very poor user documentation, but this has been addressed in a rather unique way. After logging in you will be presented with a similar screen to the one below, and in the upper right corner there is a question-mark (?). Click on it and an Administrators Manual created on FLASH paper will be downloaded and displayed separately from an on-line server. This way of presenting the Manual has several benefits; it is easy to update it for the developers, it is always available in the latest version and it is very environmentally sound.

Assisted by the Manual it’s not too difficult to get the PBX online, but I miss a Quick-start guide outlining how to get it operational and in what order the steps should be performed. I better get my act together and write it.

There is currently no auto-provisioning system for VoIP phones in this firmware (will come soon), so I have to configure the phones manually. I have five phones available in my lab; one analogue Siemens Gigaset, two Atcom phones (AT-610P and AT-620P) and two VoIPtel Pro phones (Executive and SOHO). The analogue phone is plugged into the FXS port while the four other phones takes 3-5 minutes each to configure through their GUI.

The sound quality is very good, but there is a difference between the analogue Siemens phone compared to the VoIP phones. The VoIP phones has a clear edge when it comes to voice quality, and they are not experiencing any problems with echo on the line. The Siemens phone has a very good voice quality considering that it is an analogue device, but I have experienced problems with echo a few times. Unfortunately this is not the kind of problems that can be solved by the software developers, it has to be tuned by the owner of the PBX. But this is not a serious problem, for the large majority the OSLEC does a very good job.

I recommend this line of PBX’s for people and integrators with very limited budgets and who has the required skills or available time to learn. Even though it has a very good GUI accompanied by tool-tips and an on-line Administrators Manual it still require some degree of experience and skills.

The prices for the IPxx start at US$135.00 for an IP-01 on special offer to the IP-BRI clocking in at $500.00 For more info about prices and availability please go here.

We just received information from Atcom that the latest batch of their AX-400P cards has a problem that might cause a short circuit. This is an excerpt from the letter:

Today, we found a serious issue with the AX-400P new design we shipped you last time, the issue relates to potential short circuit in the screw on the bottom of the board.

When the short circuit happen, the board may be damaged, you can see there is a broken black line in the circuit of the PCB. The card will still be able to use in this case, but there is potential danger since the circuit exposures.

If you have purchased this type of card recently you should contact your supplier and ask how to find out if your card is affected as well as their policy relating to this problem.

Go here for original story.

It has been a long wait for the majority of us, but finally it looks like Skype has decided to shower us with new features! May it just keep coming, we’ll never get enough when it comes to new ways of integrating communication platforms.

Some important news for Skype for SIP beta “open” users I thought I’d share. Just last week Skype announced going from closed “beta” to “open” to the public, and now today they’re adding some cool new Skype for SIP features.

Once the upgrades are complete there will be cool new features, such as direct inward dialing (DID) calling over the Skype network to a specific IP-PBX extension. Similarly, if someone calls your Skype business account name, i.e. tkeatingtmc, Skype will prefix the extension number into the SIP header so that the call will be routed directly to that person’s extension. Sweet!

The upgrade will also add IP authentication, which is critical to businesses that do not support registrations. According to Skype, “For SIP enabled PBX’s and SIP gateways that do not support registrations, you can now change your authentication settings in your SIP profile to a fixed IP address/port. Based on your IP address we will work out the best settings for your SIP enabled PBX when using Skype for SIP.”

Today, Skype’s Kevin Holmes Senior Product manager, Skype for Business announced the upgrades, outage, etc. Check it out:

We are please to announce three new features for Skype for SIP open beta. We will start upgrading from today and expect this to last one week. These new features are designed to enhance the way you can integrate Skype for SIP with your SIP enabled PBX.

Upgrades are due to start at 1pm GMT on Wednesday 9th Dec until Wed 16th Dec 2009.

Skype for SIP new features include:

Inbound from Online numbers
Inbound calls will now include the Online number, which means you can now set up rules in your SIP enabled PBX to automatically route incoming calls from your Online numbers to extensions and other resources such as auto attendant or IVR systems.

Extension tagging for Business Account Skype names attached to your SIP profile. You will be able to add a Business account Skype name’s target extension, so when Skype users call your Business account Skype name, Skype for SIP will prefix the extension number in the SIP message for calls from Skype users to your SIP enabled PBX.

IP authentication. For SIP enabled PBX’s and SIP gateways that do not support registrations, you can now change your authentication settings in your SIP profile to a fixed IP address/port. Based on your IP address we will work out the best settings for your SIP enabled PBX when using Skype for SIP.

We will start upgrading Skype for SIP open beta from Wednesday 9th Dec 2009 starting at 1pm GMT, the upgrade is likely to last one week.

We will be working with our Skype for SIP certified PBX vendors to update their configuration guides for setting up Skype for SIP.

Important notes: This upgrade will not affect:

* SIP Registrations
* Outbound calls to landlines and mobile numbers

Recommendations to avoid inbound call failures
We recommend that you add an additional route as soon as possible for inbound calls from online numbers so that your SIP enabled PBX can accept and route based on:

• SIP user name (current configuration)
• Online numbers (international format)

Technical details
Please note the following changes to Inbound Invite messages from Skype: (no other features or use cases are affected)

Inbound calls from Online numbers to your SIP profile
We have changed the SIP format which now includes the Online number that was dialled via the PSTN or Skype user in the To header. With this SIP message change, you can now assign up to 99 Online numbers to your SIP profile and be able to distinguish which online was called.

An example of the change in SIP headers:

R-URI: <SIP_username> 99051000000000@pbx_ip – taken from your SIP profile
To: sip:<onlinenumber>@sip.skype.com
From: sip:<callers_cli>@sip.skype.com

The remainder of the SIP format message remains unchanged.

Extension tagging for calls to Business Account Skype names
Inbound calls from Skype users to your SIP profile (from the 10th Dec 2009) You can now add a target DDI or extension number to Business Account Skype names that are attached to your SIP profile. The reason we have done this to enable SIP enabled PBX’s to route inbound calls from Skype to PBX extensions or resources such as an Auto attendant or IVR. You will find a numeric text box for this feature in the BCP –> SIP profile –> calling tab next to the Business Account Skype name. Up to 11 characters are allowed in this field.

Extension numbers will be shown in the Invites request URI for example:

R-URI: <extension_number>@pbx_ip
To: <SIP username> 99051000000000@sip.skype.com
From: caller_id@sip.skype.com

Further information can be found from Wed 10th Dec in the Skype Business support section at the following address:

http://forum.skype.com/index.php?showtopic=486951

http://forum.skype.com/index.php?showforum=252

We hope you enjoy the new features in Skype for SIP open beta
Skype product management

Just got the news, it’s finally OPEN! After months of testing  Skype for Sip is finally available to everybody with a business account! Does it really work? Is it worth the wait? And the price? We’ll definitely find out, we’ll try to get it configured and operational as soon as possible! ;o)

In the meantime I found some interesting info  on this blog:

Skype for SIP connects your company’s phone switch to Skype. SkypeIn and Skype-to-Skype calls come in to your phone system, outbound calls can go over SkypeOut. VoIP people call a connection between your phone system and a phone company a “trunk.” Some people call Skype for SIP “Skype trunking.” SFS is a limited add-on. No emergency dialing: You still need a regular phone service to dial police, fire, ambulance. No phone number portability. You need a Skype Online Number and you don?t get to use an existing number. Service levels aren?t regulated by local or government authorities or guaranteed by Skype. SFS isn?t free. US$ 6.95 per month for each channel, one call at a time per channel. You have to rent an Online Skype phone number for your business. You pay for SkypeOut at published rates. At the moment, the Skype Global Rate is 2.1¢/minute in more than 36 countries. You’ll pay more for mobiles in most places. Unlike SkypeOut for consumers, Skype doesn’t allow or offer flat-rate calling plans. Calls coming in to your phone from the Skype ecosystem are free. Many smart phone systems let you write rules for routing outbound calls. You might choose SkypeOut for international calls or if you haven’t the buying power to negotiate discounts with your phone company. Skype is building a distribution channel. They’ve partnered with PBX makers like ShoreTel, Cisco, and SIPfoundry. Together they have thousands of value added resellers (VARs) who serve local businesses. Those resellers will be eligible to earn affiliate referral commissions from Skype, although a separate program for VARs is not in place. Skype is talking with more PBX makers to make adding a Skype channel a built-in menu option. Skype for SIP is an indirect sales effort. SFS partners with PBX makers, their VARs, to reach IT and telecom departments responsible for configuring telephone systems and buying telephone services. So Skype gets to know your Phone Guy. This gives Skype a beachhead in your company, a relationship to sell more Skype products, and a champion for Skype technology.

Phil Wolff at Skype Journal talked Monday with Skype’s Matthew Jordan about the latest update. Here are the details.

And for those who want to try it out you’ll find more info and links here.

Have fun! ;o)

UPDATE: We made an attempt to make it work a few hours ago, and the first attempt disconnected all our trunks and SIP phones! Since this was on our production PBX (yes, we like to live dangerously) we removed it rather quickly. Our second attempt was on a test PBX at a different location (we like to learn the hard way), no crashes and it registered both ordinary SIP and Skype for SIP trunks. And that was all it did, no calls was able to get through. Unfortunately we were running out of time, but we will continue our little experiment as soon as we have a few minutes available. ;o)

Just found this info, could it be this simple???
The clues are in the documentation; SkypeIn and SkypeOut use G.729 for nearly all calls, so handling calls via those paths requires a G.729 transcoder on the system if the target of the call will not also be using G.729. This is why the Skype For Asterisk license includes licenses for Digium’s G.729 software transcoder as well.

It works! At least when we call out, just have to get a SkypeIn phone number for inbound calls and then we are home free (I hope)! ;o)