VoIPtel Blog

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)