Recently in Asterisk Category

ibm-smartcube.jpgIn a fascinating deal, IBM and Digium announced today that they are teaming up to offer Asterisk for Smart Cube, a customized version of Asterisk Business Edition. IBM's Smart Cube is very similar to Microsoft Small Business Server (SBS), a pre-packaged bundle of various IT & business applications - except in this case Smart Cube is based on Linux not Windows.

Businesses using Smart Cube can be up and running with a complete IT solution to run their business, including the operating system, integrated middleware, database, security and back-office functionality such as file, print, backup and recovery, and more. Extending the IBM Smart Cube to IP telephony and unified communications is a natural extension of the Smart Cube.

Asterisk for Smart Cube has administration capabilities built right into the Smart Cube Smart Desk GUI, which is perfect for the SMB. Asterisk initially ran on rPath but now uses the very popular CentOS distribution. IBM on the other hand is very partial to SuSe Linux. thus one of the technical challenges IBM and Digium worked on was getting Asterisk Business Edition to run on the SuSE Linux platform. Additionally they worked on seamlessly integrating it into the Smart Desk GUI.

How this affects Digium's own home-grown Switchvox SMB offering remains to be seen. But Digium gaining access to IBM's huge distribution and reseller channel is great news for Digium.

Via internetnews.com
astricon.jpgThe 10th Anniversary of the AstriCon 2009 conference is next week in Arizona (October 13-15) and it is shaping up to be a great show. For one, AstriCon is sporting 30% more companies in the Expo Hall than last year's conference. The AstriCon organizers have even thrown in some cool freebies to add to your Asterisk arsenal.  All attendees receive three popular Asterisk licenses, including: Skype for Asterisk, Fax for Asterisk, and G.729 for Asterisk.

AstriCon 2009 will be held at the Renaissance Glendale Resort and Spa near Phoenix, Arizona. You can register for the conference at www.astricon.net.


Polycom VVX 1500 Video Phone Quick Demo

September 29, 2009 11:30 AM | 0 Comments
Check out the quick video recording I made of two Polycom VVX 1500 IP video phones making a test video call. I used an iPhone 3GS to capture the video of the test call between the two phones. The iPhone's video recording quality isn't too shabby, but doesn't truly give you an idea of the quality of the Polycom's VVX 1500 touchscreen. (Note: The video phone was tilted far back, so it was aimed directly at the fluorescent lighting in the ceiling. So you see some whiteout when the VVX 1500 camera gets blinded by the ceiling lighting.)

Having a touchscreen IP videophone is pretty cool and I'm enjoying testing it so far. This is just a quickie video demo. I hope to do a full review of these phones very soon. Till then, enjoy the Youtube video below:

AstriCon Asterisk Conference Soon

September 18, 2009 10:01 AM | 1 Comment
astricon.jpgAfter ITEXPO's resounding success in Los Angeles (over 6,000 attendees), we can definitively say VoIP hasn't been as badly affected as other industry sectors within the U.S. economy. In just about 3 weeks, we can confirm this is true with the big AstriCon event held at the Renaissance Glendale Hotel & Spa in Glendale, Arizona. TMC's Internet Telephony Magazine is a media sponsor for the event. Companies participating include Aastra, Adhersion, Digium, PIKA, Polycom, Sangoma, Xorcom, and more

This marks AstriCon's sixth year as the official conference for Asterisk, the world's leading open source PBX. According to the event organizers, "AstriCon's mission is to expand awareness and knowledge of Asterisk over the course of a three-day conference and exhibition. AstriCon includes a wealth of information for every Asterisk user, whether you are getting started or have already discovered the power of Asterisk."

I for one am a little sad I won't be going. Too much stuff to do back at the home office. TMC is growing like gangbusters and we are very close to moving into a state-of-the-art facility. Now imagine you are in charge of moving TMC's entire data center to this new facility with minimal downtime. It's enough to keep any CTO awake at night. It's not happening for a couple of months, but this will require some massive planning by me and my team.
itexpo09.gif I tested Siphon, a SIP-based VoIP application for the iPhone, in California at ITEXPO. Interestingly, Siphon worked perfectly in California over AT&T's 3G data network. Yes, you read that right - VoIP over 3G! I couldn't contain my giddiness when I realized I could now register my iPhone with TMC's Asterisk-based IP-PBX and make/receive calls. I've tried Siphon a few times in the past and it never worked over 3G - only WiFi. I thought perhaps AT&T was now easy their restrictions and allowing it. (silly me)

However, once back in Connecticut I tested it again and it didn't work. Apparently, in some parts of the country AT&T is blocking port 5060, the default SIP port. I did some port testing on my iPhone and indeed AT&T is blocking outbound port 5060. While I was in Los Angeles I was able to use Siphone to make & receive VoIP calls over the 3G data connection through my corporate Asterisk-based PBX. I was able to receive calls to my TMC extension as well. Guess it was good while it lasted...

Now, Siphon does let you change the local SIP port from the default 5060. In theory, the Siphon application can be modified to use a different outbound port and then you could setup some port forwarding rules on your firewall, i.e. map the 'always open' port 80 (web) on your firewall to forward to port 5060 when connecting to your SIP-based IP-PBX's IP address. Or if you IP-PBX is already using port 80, there are plenty of other outbound ports that AT&T doesn't block.

Apple has rejected and blocked Siphon from the App Store. Interestingly, Apple allows other SIP clients (WiFi-only) to be downloaded from the App Store, including iPico, fring, iSip (supports push notifications of calls), Acrobits Softphone, WeePhone SIP, and Nimbuzz. What's interesting about the Siphon app is the whole saga the developer had to go through with Apple when submitting this SIP application to the App Store. It wasn't pretty...

The short story is that even when Siphon didn't support VoIP over 3G a few versions ago, Apple still rejected the app providing a lame excuse. Then after several attempts, Siphon went "underground" and provided their SIP app to Cydia, the primary jailbroken app store - with full VoIP over 3G functionality. If you can't beat em', screw em'! That's why a lot of apps have gone to the Cydia App Store to get around Apple's ridiculous restrictions.

Check out this screenshot of my iPhone showing how you can enable Siphon over EDGE/3G:

siphon-iphone-sip-settings.jpg

Unfortunately, like I said earlier, AT&T is blocking outbound port 5060 in some parts of the country, so simply enabling Siphon over Edge/3G by itself won't work if they block it. Apparently, the AT&T cellular network in Los Angeles, California works though. If anyone else has gotten Siphon to work over the AT&T 3G network, post a comment - or even if it didn't work. Would be a good gauge of how widespread they allow/disallow this.

The day is coming when the carriers will have to allow VoIP over 3G. Look at what VoIP, and especially Vonage did to the traditional landline industry. We went from paying long distance minutes by the minute to an UNLIMITED plan with UNLIMITED minutes for a flat rate. The mobile industry will soon have to follow suit.

In fact, the first wireless carrier that lets me register my cell phone to my SIP-based IP-PBX over a 3G data connection will become my new wireless service provider and have my business. I'm sure millions of others feel the same. Heck, charge me a few cents for terminating or originating my SIP-based calls. I'd pay for the ability to use my corporate identity (CallerID) when making business calls on my personal cell phone. Or just count SIP calls as 1.5x or 2x per minute of usage towards my current monthly plan's bucket of minutes. Of course, the carriers would have to detect when a SIP call originates or terminates, which is a technical challenge. They'd have to do packet inspection on a mass scale to support this.

Still, there has to be an appropriate revenue-generating business model for the wireless carriers that will allow their customers to use SIP over 3G. Make it $5/month extra or something. Vonage took the traditional landline providers by surprise, causing the defection of millions of users. So if the wireless carriers wait too long, some new wireless carrier is going to come along and do the same by offering VoIP/SIP over 3G. You mark my words...
Thumbnail image for aastra-57i.jpgAt ITEXPO, Aastra announced G.722 wideband audio codec support (HD audio) in their new 67xxi firmware version 2.5.0 or later. It's available as a free download from Aastra with no strings attached. No need to upgrade your 67xxi (formerly 57XXi) phone to a newer model to get HD audio. I spoke with Aastra at ITEXPO about this free upgrade and got a demo as well. First, it's important to note that the speaker and the microphone built into the existing 67xxi don't have the full frequency response for full HD audio. However, there is still a noticeable improvement in audio quality, especially in the low-end bass side. For 100% HD audio from the microphone to the speaker to the full-range frequency response you can purchase their newer phones which feature upgraded hardware components.

But for the thousands of Aastra phones out there - including one I use as my primary desk phone - you can simply upgrade to the new firmware and immediately see a performance improvement. I also mentioned to Aastra some issues with the speakerphone switching to half-duplex mode when there is a lot of ambient noise or if the remote caller is talking too loud. The remote caller who is speaking can't hear you (half-duplex) no matter how loud you yell. Well, apparently this new firmware does some tweaks and solves that issue as well.

Aastra is calling their wideband audio feature Hi-Q. Let's face it, Polycom has done a good job positioning themselves as having the best sounding IP phones, especially with their HD Voice product line. Aastra's new Hi-Q offering now allows them to compete with other HD phones including not only Polycom, but also Cisco, Snom and others. Aastra Hi-Q wideband audio will be supported on the 6757i CT, 6757i, 6755i, 6753i, 6751i, 6731i and 6730i.

I did a demo with Aastra on the ITEXPO show floor with Hi-Q turned off and then on and I noticed the difference right away. Can't wait to get back to Connecticut and upgrade my Aastra 6757i CT phone! They also demo'ed some cool new DECT 6.0 phones and new WiFi phones that are just now coming to the United States. (They are currently available in Europe)

According to Aastra, "Aastra's Hi-Q audio technology is a software based acoustic optimization, backwards compatible with existing 67xxi series SIP phones, delivering a more life-like conversation and richer user experience via an industry standard G.722 wideband codec."

Aastra 67xxi firmware with Hi-Q wideband audio support can be downloaded here.

Skype for Asterisk Launches

September 1, 2009 11:10 AM | 1 Comment
skype-for-asterisk.pngAt TMC's ITEXPO, Digium and Skype announced the official launch of Skype for Asterisk, which was launched as a closed beta back in September 2008. Well, now anyone can now download Skype for Asterisk and make & receive low-cost calls leveraging Skype.

According to Digium, "Now businesses can take advantage of Skype's low-cost calling to landlines and mobile phones and free calling to more than 400 million registered Skype users around the world. Skype for Asterisk allows businesses to access the world's largest community of people communicating over the Internet, natively encrypts all voice calls and lets companies manage their Skype user accounts via Skype's Web-based Business Control Panel. Businesses already using an Asterisk-based phone system can add Skype as another complementary form of communications by downloading Skype for Asterisk, without additional costly hardware. Skype users can benefit from the advanced call features of Asterisk, including call transfer, interactive voice response, automated call distribution, flexible call-routing and many more."

"Digium has been using Skype for Asterisk for the past few months while the product has been in development," said Danny Windham, CEO of Digium. "We created Skype accounts such as Digium Sales and Digium Support--a convention I suspect many companies will quickly adopt. Now, our customers all over the world can call us for free using Skype and our Asterisk PBX processes the inbound call just like it would a normal call. This is going to save Digium and our customers a lot of money."

DATUS Corporation is a Digium Select Partner with nearly four decades of experience designing and implementing communications networks in Germany. The company has nearly completed an Asterisk installation at 2,100 sites for LVM Versicherungen, a major insurance firm, and also works with Digium to design features for Asterisk that are of particular interest to European businesses. "Adding Skype for Asterisk to the DATUS indali OBX, our IP-PBX, will offer our customers inexpensive and secure international calling that, for instance, could be used for toll free customer services," said Jonny Kueppers, vice president of sales and marketing at DATUS. "We believe that the price and cost savings will be welcome with today's budgets."

"The combination of Skype and Asterisk gives those companies that have relied on Skype the advanced call management capabilities of Asterisk, while Asterisk users get free calling to more than 400 million registered Skype users and low Skype rates when calling landlines and mobiles," said Stefan Öberg, vice president and general manager of Skype for Business. "We believe the product will bring together two of the largest groups of users that value flexibility and cost savings in their PBX systems."

Foehn Ltd, Digium's U.K. Solutions Partner, has been designing and implementing Asterisk-based solutions for more than five years. The company's technical director, James Passingham, commented: "With Skype for Asterisk, we can offer our clients even more freedom in business communications. The ability to unlock the lower call costs of Skype provides a huge savings opportunity, especially for those with offices and customers around the globe."

Skype for Asterisk Features
Skype for Asterisk, which is compatible with the free and open source Asterisk versions 1.4, 1.6 and AsteriskNOW™, as well as the commercially licensed Asterisk Business Edition™, is unique in the market today. It is the only solution that integrates directly with Skype, enables multiple concurrent Skype calls from a single Skype account, and supports both G.711 and G.729a calling.
  • Make Skype-to-Skype calls.
  • Receive calls with online numbers (SkypeIn).
  • Make world-wide PSTN calls to landline and mobile phones (SkypeOut).
  • Make and receive multiple concurrent Skype calls from the same Skype account.
  • DTMF support for incoming and outgoing calls.
  • Read Skype profile fields from incoming calls.
  • Set and retrieve online status.
  • Set privacy settings.
  • Handle incoming Skype calls using Asterisk applications such as voicemail, ACD, MeetMe conferencing, etc.
  • Simultaneous access from both Asterisk and the Skype desktop client.
  • Trunk calls between Asterisk servers over Skype.
  • Supports G.711 and G.729 (included) codecs.
sip-print-voip-recording.jpgSIP Print is announcing today the general availability of a new, enterprise-class call recording platform for mid-market enterprises. The new SIP Print SME platform offers support for up to 200 seats per location, along with RAID hot-swappable drive bays, dual hot-swappable power supplies, and a Core 2 Quad Series processor.  Today's announcement is being issued in conjunction with TMC's ITEXPO Conference in Los Angeles.
 
According to SIP Print, SIP Print SME is a new, more powerful appliance designed for the needs of small and mid-size enterprises, or any organization with the requirement to record up to 200 seats per location.
 
"We introduced our highly affordable SMB product one year ago to meet the needs of small business with a need to record calls for training, QA, or compliance purposes, but simply couldn't justify the expense or hassle of the legacy recording systems on the market," said Jonathan Fuld, CTO for SIP Print.  "Since that time we've seen tremendous demand for a similar, but more powerful system in the mid-market enterprise arena.  We're pleased to introduce SIP Print SME as the ideal solution for mid-sized enterprises with the need for a system that is easy to install, easy to use and maintain, and easy to afford."
 
SIP Print SME is a 1U appliance and is certified as compatible and interoperable with many of today's leading IP PBX systems, including: Allworx, Aastralink, ADTRAN, Altigen, Avaya Distributed Office, Cisco, Epygi, Fonality, Grandstream, Mitel, NEC 8100, NEC 8300, Nortel, ShoreTel, SIPfoundry, Toshiba, Zultys, 3Com, and more. As configured, SIP Print SME is capable of recording and storing the equivalent of one handset, 24x7 for 15 years.

Check out my recent review (last month) of their previous SIP Print appliance which I gave extremely high marks.

Asterisk Training Courses at ITEXPO

August 17, 2009 10:22 AM | 0 Comments
itexpo09.gifCan you believe ITEXPO is just two weeks away? It's also almost September. Where did the Summer go?

ITEXPO, the #1 VoIP conference in the U.S., has several educational tracks you might be interested in checking out. Of particular interest to me are the two separate Asterisk and the Switchvox training courses. As Asterisk's popularity continues to grow, so does its development and complexity. Last year's Asterisk isn't the same as this year's, so it's never too late for a refresher or to learn about the newest features.

I've only seen demos of Switchvox and haven't actually put Switchvox through the full test-drive ringer, so I might want to check out the Switchvox training course just to see what's new and what's different from regular Asterisk. Also can't hurt to learn how to use and manage it since I'd like to review it at some point.

You can check out and register for one or both courses here.

SIP Print VoIP Appliance Review

August 12, 2009 10:18 AM | 1 Comment
sip-print-voip-recording.jpg
Call recording on VoIP phone systems is critical because it can ensure good customer service, provide employee training, and offer liability protection. SIP Print offers a SIP-compliant VoIP recording appliance at a price point that is much lower than legacy analog or digital E1/T1 recording systems. Part of the reason is you don't have to pay for expensive Dialogic or NMS telephony hardware to interface with analog or digital trunks. Instead, SIP Print's appliances sit on the network and simply record the SIP VoIP traffic. Obviously, the cost economies of packet capture are much lower than using telephony cards. SIP Print's appliance offers true system-level call recording for many of today's leading VoIP phone systems, including Allworx 6x & Allworx 24x, Altigen, Avaya S8XXX systems (when used with Microsoft OCS), Asterisk / trixbox variants, Cisco, Microsoft OCS and UM, Mitel 3300, NEC, Nortel SCS, Shoretel,  SIPXecs, Vertical, and Zultys MX-250. Because it supports SIP at the communications system level, SIP Print's platform can also record advanced functions like voicemail or "follow-me" calls forwarded to mobile phones or off-premise phone numbers.

SIP Print sent me one of their VoIP call recording 1U rackmountable appliances for review. There are two models available - SMB and SME (higher-end). The SIP Print Enterprise (SME) is a higher-end system that has dual redundant power supplies and supports RAID. SME also has alarms for an unplugged or bad power supply, and it has an alarm for a bad or missing disk drive. SIP Print sent me their SMB model to test.

sipprint-architecture.jpg
                                      SIP Print Architecture

Installation:
Installation was a breeze. Not like the old days when I tested call recording systems from NICE Systems which required tapping into the T1 line or analog ports. All I had to do was configure port mirroring on a Gigabit switch that connects to our Asterisk-based phone system. I "mirrored" all the traffic going to/from the IP-PBX and send it to the SIP Print appliance. This way, the appliance "sees" all of the SIP traffic, including the RTP audio packets. Next, I added an additional IP address to my network card (192.168.2.50) since the appliance uses the 192.168.2.x network. I then launched my browser and went to http://192.168.2.253/, the default IP address of the appliance.

I logged in with the master username and password. The first thing I did was change the network setting to match our own. Then I added the SIP gateway (IP-PBX) and a bunch of IP addresses used by our SIP endpoints (IP phones). You can have up to 3 gateways/SIP proxies/IP-PBXs, while the SIP endpoints are licensed, so it depends how many licenses you purchase. Our configuration was licensed for 100 SIP endpoints. It was a bit of a pain to add each SIP endpoint's IP address one at a time. I wished the device had an auto-discovery mode which would automatically listen for SIP REGISTER messages, which all SIP endpoints send (usually) every 5 minutes. Would make adding the entire list of IP phones an automatic process that takes only 5 minutes. Thus, I had to copy/paste one-at-a-time each phone's IP address that belonged to our SIP phones into each of the form fields in the "100 SIP End Points" section.

sipprint-configuration.jpg
  SIP Configuration Screen for adding SIP gateways & SIP Endpoints.

Although adding the IP phones was a bit tedious, it wasn't too bad. After the IP phones were added I made some test calls. I also added TMC's IP phones so there were also some production IP phones in use that would allow me to record regular business calls. I went into the web interface and was able to view all the call recordings. I noticed it was displaying the MAC address of the IP phones instead of the extension number or the person's name. This is by design. I then created "aliases" which map the MAC address to the person's full name (with no spaces). I could have used extension numbers, but then searching for calls would be harder, since it's much easier to remember someone's name than their extension number. In any event, after creating the aliases, here's what the call recording screen looks like:

sip-print-call-recordings.jpg
                                                 Call recordings

The speaker icon under the 'Play' heading allows you to quickly play any recording using your favorite .wav player. Since the results are paginated, you can go page-to-page by clicking Next or Previous. Additionally, the web interface sports some powerful search capabilities. I was able to search by username (alias), phone number, and date range. The alias and phone number support partial matching, so instead of typing 'thomasjones' you could just type 'thom' and it'll match. The search was blazingly fast, which is crucial for any call recording system. As soon as I clicked search, less than a second later, it would return the results.

My initial impression of the web GUI was that it was sort of ugly and simplistic. But after using the system, I appreciated its simple GUI and the ability to just get things done. In fact, another speedy feature is the ability to sort any of the column headings, including alias, date, and phone number. Once again, sorting is nearly instantaneous. Thus, finding a specific recorded call couldn't be easier both for users and managers that have access to other employee's call recordings.

Speaking of managers, I should point out that granting permissions on SIP Print was very easy to do. You simply grant access rights to the specific user by checking boxes next to the names (aliases) of users. This will then allow the user to view recordings for these 'checked' aliases/users as seen here:
sip-print-assign-permissions.jpg
                                                Assign Aliases to User

One minor nuisance is that you cannot reset a user's password, even if you are the 'master' administrator account. Often users forget their password, which requires a password reset. Instead, as a workaround, I simply deleted the username, re-added the user and then had to re-add the aliases they had access rights to.

Feature Specifications:
SYSTEM
- Web Based GUI
- Remote Administration
- Call Playback on Standard Media Players
- Search by User Name
- Extension & Name Lookup
- Caller ID
- Search by Area Code & Prefix
- Fast Forward & Rewind
- Time and Date stamping
- Email-ready Call File Formats
- Multiple Manager Access
- Remote Access
- Easy Archive & Audit Trail
- Column Sort (on the fly)

COMPATIBLE WITH MOST IP-PBX'S
- SIP
- SIP Hybrid
- MGCP

CALL CAPTURE
- Trunk-side Analog CO, T1/PRI & SIP Trunking
- Extension-to-Extension
- Follow-Me Calls
- User- and Extension-specific
- SIP Registration
- Captures SIP & RTP traffic

PURE SYSTEM-LEVEL SIP VOICE RECORDING
- Records one handset 24x7 for 2.5 years
- Records all specified user calls on the local network
- No logger patches
- One appliance records all of your specified calls
- No IP-PBX or handset integration

PRODUCT SPECIFICATIONS SOFTWARE
- Number of Users Supported: 15-200
- Operating System: Proprietary O/S
- Administrators: 1
- Managers: 9
- Total Concurrent Calls: 150 MEDIA
- WAV (.wav) media file

MEDIA PLAYERS
- Web Browsers Supported: MS Internet Explorer, Mozilla Firefox, Apple Safari
- Media Players Supported: MS Windows Media Player, Adobe Media Player, Apple Quicktime

APPLIANCE
- CPU Xeon 5100/5000 Dual Core Series Processor
- Dual Port Ethernet/Fast Ethernet/Gigabit
- Form Factor: 1U rack-mountable chassis
- Dimensions: 17.2"W x 1.7"H x 14.5"D
- Front Panel Power Button
- Reset Button
- Power LED
- Hard Drive activity LED
- Network activity LEDs
- System overheat LED
- 110-250 Volt Auto Adjusting Power Supply

It's worth mentioning that updates are free in the appliance. I inquired about simple RAID mirroring and at the time they said that there is no mirroring in this version because it adds too much to the cost. However, the SME edition does now have RAID 5 support. They do offer some additional backup protection. You can archive the recordings to a SMB shared folder across the network as seen here:
sip-print-archiving.jpg

While the screenshot above shows a "delete" option, interestingly you cannot delete individual call recordings from the main call recording screen. I asked SIP Print about this, and they told me that allowing users to delete their individual calls caused more headache for managers, CEOs, and executives looking to keep a sharp eye on their employees interactions with customers. They cited examples where an employee claimed one thing and the customer another, but since the recording was deleted, so there was no way of determining who is at fault. Truth be known, the customer is NOT always right. In fact, they also discussed lawyer and financial offices that require 100% recording of all calls for liability purposes. A client could for instance claim they said they only wanted "100 shares of Company XYZ" but the stock broker purchased 1,000 shares. By examining the call recording, it can be determined who is right. Thus, SIP Print has taken the approach of blocking the deletion of individual call recordings, but they do allow administrators to archive recordings and then delete them en masse.

My next thought was "What if you don't archive anything? How much recording storage would you get before running out of space?" SIP Print told me that if you take the phone off hook you would get 5 years of recording. Impressive! The 'Enterprise' version will get you 15 years of recording. Thus, archiving doesn't have to be done on a regular basis if you don't want to.

When I asked for further details on why they designed their platform to record 100% of calls, SIP Print's CTO Jonathan Fuld said, "When we set out to design SIP Print, we said, we want to make a device that records all calls all the time. We did that to differentiate our selves from a feature standpoint, because all IP PBX's and most IP Phones have a "push to record" feature already incorporated in their software/firmware/hardware." He added, "The xlite softphone allows you to click on a button to record to your desktop. Zultys allows you to record calls on an as needed basis or record all calls - but there is a limit here as with Allworx and other IP PBX's. Zultys can only record so many calls before reaching its limits. Allworx eats up voicemail space."

Jonathan continued, "In an IP Telephony world, the IP PBX is a SIP proxy - a gateway - that incorporates voicemail. As you know the ip phone is now the phone and the pbx - it does it all, except recording. The network is the telephone - peer to peer. The peer to peer model distinctly changes the control of the phone call from the pbx to the phone. No more can the PBX offer "barge-in", "whisper", "monitoring" without distinct overlays of the SIP Protocol. And that is fine. It means the phone must interact according to the TDM model (for now) - control most call functions are via the PBX, because no-one has yet offered "barge-in", "whisper", "monitoring" in the peer to peer model."

He went on to explain, "On the other hand, recording all calls becomes quite simple in the peer to peer model. Place SIP Print in the middle of the network stream of traffic, capture, record, obtain metadata (quasi-cdr), display and play - all quite distinct processor intensive functions which would grind most IP PBX's and IP Phones to a halt. When you and I make a phone call, it is just one conversation. If I press a couple of buttons on my phone, to record the call, the "device" (whatever it is) records just that one conversation. SIP Print records all the conversations."

The prior version I was testing didn't have a web-based reboot for administrators. Thus, I had to press the reset button on the unit and it would do a graceful shutdown (though current calls won't be saved). Fortunately, they just added a web-based reboot option in v1.3+, so administrators don't have to walk to unit to reset it.

Room for Improvement
I would like the phone numbers to convert to hyperlinks that point to a reverse lookup web page. That would be quite convenient to know if employees are making business calls or calling their spouses 10 times per day!

One other suggestion is the ability to "share" the call recording with your customers using the web interface. While you can email the recording, often times email servers have a size limit on attachments (i.e 5 MB), so you can't use email. Of course, SIP Print would have to allow some sort of "guest" account that lets customers authenticate onto the appliance to retrieve the recording via their browser. This could open a security Pandora's box if the "guest" account lists every "shared" recording. One way around that would be to allow each SIP Print user to have a secondary username & password, which logs customers in with limited access to only the recordings that each SIP Print user shares. Of course, if the SIP Print user forgets to "unshare" the recording that could be an issue as well, since future customers that logon can see previously shared recordings. Perhaps limit it to 1 shared recording and have it auto-expire (unshare) after 5 hours? This gives enough time for the customer to retrieve the recording and then auto unshare it.

Pricing:

SIP Print SMB Solution

Description

Users

Model

MSRP

100001

SIP Print SMB base server with 20 users

20

SPB20

$5,745

100125

SIP Print SMB 25 user expansion key

25

SPUE25

$3,976

 

SIP Print SME Solution

Description

Users

Model

MSRP

100301

SIP Print SME base server w/50 users

50

SPEB50

$13,098

100325

SIP Print SME 25 user expansion key

25

SPEUE25

$3,976

The initial purchase includes the license of 20 users for SMB, 50 users for SME.

Ratings Score
Installation
Documentation
Features
Usability
Performance
Value
Overall
Conclusion:
The latest version adds support for handset to ITSP bypassing the IP-PBX (including remote phones), which is a nice feature to have. I really like the SIP Print appliance for several reasons. It's very affordable, it's easy to setup and maintain, finding call recordings is a breeze, and it works with just about any SIP-based platform, including Microsoft Office Communications Server 2007 R2. The SIP Print appliance does exactly what it advertises - records calls - and it does it very well, so I highly recommend this product to any organization looking to add call recording capabilities to their SIP-based communications infrastructure.


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  • Chipmunk: It's a great app! Though it is not integrated with read more
  • christmas stocking fillers: I hope it ends as soon as possible. I do read more
  • Nikki Brown: Interesting. Is there a side-by-side comparison with Google voice anywhere? read more
  • External Hard Drive: Hi, The Skype's legal battle is in progress and now read more

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