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SIP 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.

Come to ITEXPO, the premiere IP communications event and you might win
this.

Here's the ITEXPO sign just as you enter the L.A. Convention Center.

Pre-registration is looking good so should be a well attended show.

Last week I spoke with Vonage's Michael Tempora, Senior VP Product & Program Management about the launch of Vonage World and Visual Voicemail. The big news was "Vonage World" with unlimited calling across the world to over 60 countries, which covers 2/3 of the world's population (4.5 billion) - all for $24.99/month.

As an aside I also asked him if Vonage was working on an iPhone app. He said it was in the works but couldn't comment further. I was going to blog about it back on the 19th, but got busy preparing for ITEXPO. Speaking of which, I'm blogging this using Virgin America's GoGo wireless. Too bad it isn't free. Was $12.95 which was nearly the same price ($15) I had to pay to check a single bag. Whatever happened to the first bag is free? Anyway...

Sources have revealed that Vonage has submittd their app to Apple. Having a Vonage iPhone app combined with Vonage's World's global unlimited $24.99 plan could be a killer app that gives Vonage a shot in the arm. Imagine a SIP client on the iPhone that lets you call 60 countries - 2/3 of the world - as much as you want for just $24.99/month. Combine the large user base of iPhone users and the millions of Vonage users and what you could have is the most largely deployed WiFi VoIP phone on the planet! Forget about buying those standalone WiFi SIP or Skype phones which is another device you have to carry.

But it gets better... Vonage's current SIP client called Vonage Companion has unique technology which enables it to "share" the same SIP credentials with your ATA device connected to your home cordless phone system. Thus, you can accept an incoming call on your PC using Vonage Companion or answer the call using your regular analog phone connected to your SIP-compliant ATA. I discussed this single identity to multiple SIP devices with Vonage last year.

In a mid-2008 interview I had with Mary Grikas, Vonage's Executive Director of Device Development I asked, "What sort of technical challenges did you have having the same CallerID and the same phone number? Obviously, you are leveraging SIP and the ATA that sits at the home residence logs on with one set of SIP credentials and Vonage Companion logs on with different SIP credentials, but it's mapped on your back-end to the same phone number."

Mary responded, "That's really a great question because we actually had to put almost a whole new infrastructure in place with a lot of new equipment. And we actually have proxies that are just dedicated to Companion to handle those calls. We do differentiate between the home TA DID credential and we do have flag for Companion. That way we know, as you said on the back-end it's all mapped and then we know where the call originates. We know if a call originates from Companion versus originating from the TA." Mary continued, "That was a lot of work for our call processing team. They had to do a lot of work configuring that system and all the redundancy and mapping. It was a pretty big effort, but it's something we were committed to doing because we had such an overwhelming request from our customers to implement a feature like that.

So what does this mean for the iPhone & the Vonage app?

Well, since their single SIP identity technology is done on the back-end, you will have a Vonage iPhone app that does the same thing as Vonage Companion. Thus, you will be able to receive calls to your home phone directly on your iPhone using VoIP or of course you can answer the call on your home phone.. No more remembering to setup your home phone to call forward to your cell phone when traveling! Further, your outbound calls can use your home phone number's CallerID and use your unlimited bucket of VoIP minutes.

The real question will be if Apple allows the Vonage iPhone app to work over 3G. Don't hold your breath. They haven't allowed a single "official" VoIP app to work over 3G. Actually, I just remembered siphon for the iPhone allows you to turn on VoIP over 3G & Edge in the Settings screen in the app. How did that get past the Apple censors? Though, I only got siphone to work 1 time over 3G. Must be AT&T blocking port 5060 on their network.
Update: I just tried siphon in California at ITEXPO and then back here in Connecticut and siphon worked both times over AT&T's 3G data network. Maybe was just a hiccup that one time I was trying it and siphon didn't work. I can now make & receive VoIP calls over the 3G data connection through my corporate PBX. I don't get it though -- why is Apple allowing a 100% standard SIP client to run on the iPhone while limiting other VoIP clients (such as Skype - or even other SIP clients such as fring or Nimbuzz) to just WiFi?

Features I expect in the Vonage iPhone app:

• Voicemail playback
• Voicemail push notification
• Send to Voicemail
• Block caller

I'll see if I can get Vonage to commit to a tentative date. So stay tuned for the launch of the Vonage iPhone app.

Vonage today has announced some significant value add to their core $24.99/month unlimited plan. Vonage's Michael Tempora, Senior VP Product & Program Management told me they are announcing two new services. First, they are announcing "Vonage World" with unlimited calling across the world to over 60 countries, which covers 2/3 of the world's population (4.5 billion). The existing $24.99 plan included U.S., Canada, and five European countries. The new Vonage World adds an additional 60 countries to the plan. One of those countries is Iraq, so now spouses, friends, and family can contact their loves ones serving in Iraq.

Michael said, "That's really 60 whole countries, unlike some competitors who offer specific destinations or cities within a country. When we say it's a country that is included for unlimited we mean every place within the borders of that country. These countries represent nearly two-thirds of the world's population."

The second feature is called Vonage Visual Mail featuring unlimited speech-to-text transcription. It gives Vonage users the ability to read their voice messages. via email or SMS depending on the user's preference. In addition to email and SMS, Visual Voicemail will also be accessible via Vonage's online web portal. Michael explained, "Vonage has been offering Visual Voicemail for over a year on pay-per-user basis. When we launch this, we will be the only U.S. home phone service provider to include unlimited visual voicemail in their base plan at the flat rate of $24.99". I asked Michael which speech recognition engine they use and he said Nuance, which is well known for their very good accuracy.

I inquired, "So customers will wake up tomorrow and have this as their base package?" Michael responded, "This is available to new customers. They'll get the [Vonage] World Plan and the unlimited Visual Voicemail. For existing customers all they have to do is call and ask to migrate to the new plan. So we're sure they want it. We'll move them over with no change fees or any other issues involved."

I asked, "What is the main reason for adding so many more countries at the same $24.99 monthly plan?" Michael responded, "There's really an unmet need in the marketplace today. If you look at calling statistics in the U.S., since 2000 domestic long distance calling has fallen 6% annually. Over that same period, outbound calling from the U.S. to international destinations has increased 15% annually. With very little changing in pricing or pricing models in that industry. We think these customers have been underserved in terms of value. We did this to meet the needs of those customers who want to call internationally and not worry about calling to the store for a new calling card, or dial an access number or pay a high per minute rate from traditional carriers."

I asked, "With this announcement, how will you now compare with Comcast, traditional single-play providers such as Packet8, as far as the number of countries and price?" Michael responded, "When we launch this no one will match the breadth of countries or the unlimited calling that we will provide. No one is close to this. Nobody will reach the number of people throughout the world - 4.5 billion people - that we will reach with this offer. We think that this will be the greatest value in flat-rate calling that's offered in the U.S."

Complete Chart of Available Countries/Territories

 

Andorra	Cyprus	India**	Monaco	Singapore**
Argentina	Czech Republic	Iraq	Netherlands	Slovakia
Australia	Denmark	Ireland	New Zealand	Slovenia
Austria	Dominican Republic	Israel	Norway	South Africa
Bahamas**	Estonia	Italy	Peru	South Korea
Bahrain	France	Japan	Poland	Taiwan
Belgium	Finland	Jordan	Portugal	Thailand**
Brazil	Georgia	Kenya	Puerto Rico**	Turkey
Brunei**	Germany	Latvia	Romania	United Kingdom
Bulgaria	Greece	Luxembourg	Russia	United States**
Canada**	Guadeloupe	Macau**	Spain	U.S. Virgin Islands**
Chile	Guam**	Macedonia	Sweden	Venezuela
China**	Hong Kong**	Malaysia**	Switzerland	Zambia
Colombia	Hungary	Malta	Saipan**

** Includes calls to mobile phones

Presence is a valuable tool used in a variety of consumer and business applications, including Microsoft OCS 2007, Skype, AOL's AIM, Windows Messenger, Yahoo! Messenger, and more. Businesses which used to ban IM applications (block it on the firewall, Group Policy, etc.) are starting to see the productivity value in IM and have implemented IM solutions in the enterprise. With the convergence of voice, video, and chat in applications such as Microsoft's Office Communicator, rich presence functionality enables your fellow co-workers to know if you are on the phone, in Do Not Disturb mode, away, working at home, on vacation, etc.

Knowing your co-workers presence certainly improves how, when and if you interact with them. However, not every employee wants their fellow co-workers tracking when they sign-in or are away. Two hour Friday lunches anyone? Executives, CEOs, and managers may not care what employees below them think, and they never want them to see when they are or aren't there. This is where the "Appear Offline" presence state comes in handy. Most IM applications allow this, including Skype, MSN Messenger, and others. However, when it comes to business IM applications such as Office Communicator, you may not want to allow employees to "Appear offline" or at least you'll want to restrict the privilege to just executives and managers.

By default Office Communicator does not support "Appear offline" since the whole point of Microsoft's unified communications strategy is to "share" information to improve communications in the enterprise. However, you can easily add it by doing the following:
Just be careful about enabling the "Appear offline" presence state in any corporate IM solution. If everyone enables this feature, then you defeat the whole purpose of presence. However, with Group Policy and corporate procedures in place, certainly executives, managers, etc. will find this privacy feature useful. As for the rest of the minions, you must share your presence so we know you've been on a 2-hour lunch break. Or at least you'll feel somewhat guilty knowing your presence status hasn't changed to Online in a couple hours.

Can 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.

Are you running 64-bit Windows and can't get the OCS 2007 R2 Outlook add-in to work? Bummed you can't use the add-in to schedule meetings? As a workaround, those running Office 2010 (64-bit version) on a 64-bit machine have to manually paste the OCS conferencing number and ID into their Calendar, which is a bit tedious. You also miss out on the nice calendar presence information.

Well, Microsoft is still working on the 64-bit version of the Outlook add-in, which will be available when Office 2010 RTMs, however they just released 'Office Communications Server 2007 R2 Web Scheduler', which you can use to schedule meetings in the meantime until they release the 64-bit Outlook add-in. So for you bleeding edge folks running Office 2010 64-bit and OCS 2007 R2 should find this useful.

According to Microsoft, Web Scheduler is a 64-bit tool for Microsoft Office Communications Server 2007 R2 that provides a Web-based alternative to the add-in for the Microsoft Outlook messaging and collaboration client for the purpose of scheduling a meeting using OCS 2007 R2. It also provides a browser-based conference management experience that includes operations such as:

• Scheduling a new Live Meeting conference or conference call.
• Viewing and modifying details of an existing conference.
• Listing all existing user schedules of a Microsoft Office conference.
• Deleting an existing conference.
• Sending an e-mail invitation to conference participants by using a configured SMTP mail server.
• Joining an existing conference.

Download it here.

Microsoft released a new UC How To training application leveraging Microsoft Silverlight™ 2 to help users and administrators learn how to use Office Communications Server 2007 R2. The Microsoft Unified Communications "How-To" training tool provides step-by-step instructions for common UC tasks, such as find and add contacts, create groups in contact list, initiate a voice or video call, create a chat room, search chat history, and much more. It also goes into "etiquette" in instant messaging and etiquette in managing your presence status. I really liked this aspect of this training tool, since it is often difficult for IT/MIS folks to document etiquette procedures and get users to actually listen. So having a respected third-party such as Microsoft offer up their etiquette rules certainly will get users to pay more attention, which will result in better productivity.


Using the tool is very easy. You simply pick one of the main colored categories to the left and then drill down to the section you want to read. Interestingly, you can customize the How-To application based on the UC features you've installed in your organization. For example, if you have installed all UC features except Communicator Mobile and Communicator Group Chat, you can modify the XML file so that those features and topics do not appear in the interface. The How-To training tool is delivered as both a desktop application and a Web application. The desktop client must have a Web browser with the Silverlight 2 client installed.

Additionally, Microsoft also just launched Unified Communications Adoption and Training Kit 2007 R2. The Unified Communications Adoption and Training Kit for 2007 R2 provides guidance and resources for IT Pros, Helpdesk, and Trainers to speed adoption and usage of Unified Communications technologies in the enterprise. The kit includes Planning Checklists, Awareness materials, including Poster, Door Hangers, and E-mail samples, and User Education Materials such as Quick Reference Cards, Flash Cards, and links to Web-based Training. I played around with it and it also includes a mildy funny intro video. Definitely a useful kit, especially the Quick Reference Cards that you can print and give out to users.


Lastly, there is an online tutorial with six modules of multimedia training for the OCS 2007 R2 platform, including:

Module 1: Intro to UC and Enterprise Voice
Module 2: Communicator Instant Messaging and Presence
Module 3: Enterprise Voice-Soft phone Experience
Module 4: Desk Phone Experience
Module 5: Communicator - Audio and Video
Module 6: Microsoft Office Live Meeting & Microsoft® RoundTable™

Check it out: http://www.microsoft.com/communicationsserver/enduser/tutorials/

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.


                                      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.


  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:


                                                 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:

                                                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:


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.