Recently in VoIP Category

Microsoft just published the latest and greatest version of their OCS 2007 R2 documentation. I asked the official Microsoft OCS MVP mailing list, "Is there anything new about the OCS 2007 R2 docs contained on this page or is this just a centralized repository of all the Word docs and the single .chm help file?" I added, "Perhaps this is the first "official" 1.0 release of ALL the documentation?"

All you know-it-all techies out here are probably thinking, "Documentation? We don't need no stinkin' documentation!"

Yeah, in most cases I'm right there with ya, but not something as complicated as OCS 2007 R2. I often tell users who come to me repeatedly with the same IT question to go RTFM. So I suppose if I tell users to 'Read The "Friggin" Manual', I should read the manual as well. But I digress...

In any event, Jeff Schertz from Pointbridge, a consulting firm (including OCS) commented, "I hope that is really not the case. IMO if there is going to be three different formats of the same content (online, .CHM, .DOC) then they really need to be consistent across all three. I'd assume the .CHM will fall behind and require periodic updates, online content should be updated in parallel to downloadable docs. Forum questions regarding 'confusing OCS documentation' is a daily occurrence.

My sentiments exactly! I should mention that the .chm (Compiled HTML Help file) is a nice quick searchable index of the help content, but updates often lag behind other forms of documentation.

Elan Shudnow from CDW (I buy most of TMC's computer equipment @CDW by the way) also responded when he wrote, "This is one of the issues I saw with R1.. I saw a ton of inconsistencies with information on Technet and the Word files. Now there's Word files, CHM, and Technet with probably additional inconsistencies because now there's 3 places for information instead of 2 like with R1."

Microsoft's Patricia (Trish) Anderson responded, "In OCS R2, we are now single sourcing all docs. The chm were updated 5 hours after the online update for this refresh and will generally follow that same model. All three deliverables have been refreshed with the latest info and are inline with each other."

A few people, including myself all reacted to the good news of consistent and updated documentation. Getting Office Communications Server 2007 R2 deployed is a fairly complicated task so good documentation is crucial for OCS resellers & installers, as well as customers' IT departments that install and maintain OCS deployments. We don't need no stinkin documentation!

Download the docs

Some interesting news from D2 Technologies about them showcasing their mCUE™ converged communications client with embedded VoIP for Android at OESF Japan. I should point out that D2's mCUE mobile convergence software solution combines a communications user interface with the company's vPort MP VoIP software platform and is targeted towards OEMs and service providers to help deliver integrated Fixed Mobile Convergence (FMC) and Unified Communications (UC) functionality.

At OESF they will demo how mCUE can enable VoIP, video chat and other IP communications capabilities in stationary Android-based embedded equipment and consumer electronics devices. With mCUE, these devices can offer premium multi-service unified communications capabilities and deliver simultaneous interoperability with any communication service provider, Instant Messaging (IM) community or social networking platform. All popular communication modes are converged to a single communications user interface (UI), including circuit switched voice (PSTN or cellular), VoIP, Instant Messaging (IM), SMS and video chat.

Full release after the jump...

Last week, I wrote how Microsoft is making inroads in the enterprise with their Office Communications Server 2007 R2 platform and how they are looking to achieve five 9s of reliability. Well, one other critical feature needed for an enterprise phone system is decent call recording. Unfortunately, there does not seem to be a lot of options for call recording on OCS 2007. One of the problems is that not all calls go through a PBX.

For instance, you can use the Microsoft Communicator client to call a co-worker who is also using Communicator. The call is a peer-to-peer SIP session that doesn't go through a PBX, so the PBX can't leverage it's call recording capabilities to record the call.

So what are your options?

1) You can go the "cheap" route and download the FREE open-source Wireshark packet sniffer program. Wireshark can decode VoIP packets, including the RTP audio stream and save it as an .au file (you can convert to .WAV if you need to). All you need to do next is "mirror" a port on your switch that sees all the VoIP traffic and then hookup a PC to monitor and record the traffic.

2) Another similar option is Cain and Abel. Cain & Abel was designed as a "network administrator security tool" to pinpoint security holes. I put that it quotes, since it can also be used as a hacker's tool.  Essentially, this "security" software product can record your SIP-based VoIP applications. Cain's sniffer can now extract audio conversations based on SIP/RTP protocols and save them into WAV files. The following codecs are supported: G711 uLaw, G711 aLaw, GSM, MS-GSM, ADPMC, DVI, LPC, L16, G729, Speex, iLBC.

3) You can go the "professional" route and install a 3rd party SIP-recording platform that monitors ALL SIP VoIP traffic.

One good one I recommend is SIP Print. It's a 1U appliance that captures SIP VoIP traffic and has a web interface to access recordings. In theory, it should work with OCS but I haven't tried it. I like the SIP Print appliance and am finishing up a full-fledged review which I hope to publish this week.

4) You can try http://live-pa.com/ which is launching this summer and is specifically design for OCS. To record, all you have to do is "invite" the Live-PA contact. It starts recording immediately when invited into a conversation on any Office Communicator (OC) enabled computer or communications device. It's a hosted offering, without the need for on-site software and hardware recording equipment. According to Live-PA, "Just think of Live-PA as another OC contact, but, in this case, one that you hire and use whenever the need arises. To hire the services of Live-PA, sign up, and simply add the Live-PA.com Manager to your list of OC contacts. That's it - no software or hardware to install and maintain!" The Live-PA software resides on its own secure servers, and is accessed via the internet whenever you use Live-PA.com Manager. They give you unlimited storage capacity. Nice!

5) Telrex has a call recording offering for OCS available today. According to their website, CallRex™ software enables businesses to implement Microsoft Unified Communications while maintaining and extending their call recording and monitoring business processes. CallRex further extends the value of Office Communicator 2007 by making recorded calls accessible at the desktop. For example, Microsoft Unified Communications users can link recorded calls from the CallRex solution to customer records in Microsoft Dynamics CRM. In addition, call recording and monitoring features can be integrated with business applications via the CallRex API™.

Update: I should mention that although many of the above solutions can capture the SIP traffic - including the RTP stream - you might have problems playing the audio back since Microsoft uses their packet loss concealment RTAudio adaptive codec within Communicator. Not sure if Microsoft Media Player can play this codec or not. Hmmm, might have to do some packet captures and see if this works!

The Future of Recording in OCS...
It's certainly possible that a future release of OCS could feature a decent call recording feature built-in that is 100% software. It would be nice if you could simply click a record button from within Communicator and it records the call as a .wav file directly to your PC. Microsoft no doubt is working on a future OCS release that is 100% software without the need for a PBX, so they'll need a 100% software-based recording solution as well. Maybe it won't be a fully-featured call recorder that puts recordings on a centralized server for call center managers, admins, etc. with fully reporting capabilities -- but certainly OCS's Communicator client should act at least as a "personal" recorder.

I should mention that while Microsoft OCS 2007 R2 did not quite herald the death of the IP-PBX, most experts, myself included, predict a future release of OCS that doesn't require a PBX. Who knows - maybe OCS 2010, OCS 2011, or a later release will be PBX-free? Probably sooners rather that later, that day is coming.

A 100% Microsoft UC solution without the need for a PBX/IP-PBX at all could be a game changer. Of course, the current version, Microsoft OCS 2007 R2, does have some limited support for SIP IP phones, so you could throw out your existing PBX today if you wanted to. Any future release of OCS will have to support SIP phones from popular SIP phone players such as Aastra, Polycom, and snom. Also, most businesses aren't ready to toss desktop hard phones for a 100% software-based softphone solution, i.e. Microsoft Communicator. Additionally, any future OCS release will have to include all the advanced call center functionality you get from Nortel, Avaya, Mitel, or even some low-cost Asterisk-based PBXs, if they plan on completely eliminating the need for a PBX. A 100% software-based IP-PBX with unified communications capabilities, advanced call center functionality, and call recording would certainly be a compelling choice for many businesses.

Google Voice just added SIP connectivity through Gizmo5 which basically enables FREE inbound and outbound calling! With the Gizmo5-to-Google Voice connectivity not only can you can connect any SIP device (softphone, IP phone), but you can even use regular telephones for free calls in the entire United States. Google Voice already offers DID numbers in nearly every area code, which means businesses, especially SMBs can take advantage of this without resorting to some obscure out-of-state area code.

As you already know, Google Voice already gives you FREE outbound calling in the U.S., but the missing piece of the puzzle is free INBOUND calling. Well, Gizmo5's beta service called Gizmo Voice is the final piece to the puzzle. Gizmo Voice lets you take full advantage of the messaging and calling services of Google Voice combined with Gizmo5's support for any SIP device. Thus, in addition to the free inbound and outbound calling, you also can take advantage of Google Voice's free voicemail and free voicemail transcription.

With Google Voice + Gizmo Voice you can make and receive U.S. calls without any monthly or per minute fees. This is a game changer! SIP termination providers surely aren't going to be happy about this deal. How can they compete with free?

In today's increasing cost-conscious economy, SMBs are looking for feature-rich IP-PBXs at the lowest cost. Many SMBs are willing to sacrifice some advanced telephony features to just get the basics, including call transfer, three-way conferencing, auto-attendant, and voicemail. Advanced features such as call queues or call recording are nice features to have, but many SMBs aren't willing to pay for higher-end IP-PBXs with this functionality. Fortunately, Grandstream's GXE5024 and GXE5028 products not only have the "basics", they also have some advanced functionality such as call queues -- and at a reasonable price of just $899 for the GXE5024 and $1399 for the GXE5028. Grandstream sent me a GXE5024 for a test drive review.

First, the only difference between the GXE5024 and the GXE5028 is the number of analog PSTN FXO ports - 4 ports vs. 8 ports. They both also sport two FXS ports for connecting analog phones, fax machines, etc. The 5028 has 2 more conference bridges, allows bigger capacity for voicemail/faxmail and in the future, videomail, supports more concurrent accesses to IVR voice menus/voicemail/faxmail, supports more registered extensions and concurrent calls (in an upcoming official release soon). What's nice about the GXE502x is that SMBs don't have to get rid of their existing local analog lines if they don't want to. They can slowly add one or two VoIP SIP trunks into the GXE502x system and migrate from analog trunks to SIP trunks over time.


Hooking up the GXE5024 was a snap. What's nice is that it sports a Power over Ethernet (PoE) port on the LAN interface, so you can skip the use of the included AC adapter if you want. I logged into the default IP address 192.168.10.1 for the GXE5024. It sports a quick set-up wizard which is pre-configured with basic call settings that enable you to quickly configure your GXE.

I tried both the wizard and non-wizard method and both are pretty straightforward. From the web interface, I clicked the Auto Provision link, which kicks off auto-detection of any Grandstream phones I connect to the same LAN segment. I simply had to select a "Starting extension" (701) and "Ending extension" (703) to set my extension range. For testing purposes I had a couple of Grandstream GXP2000 phones and a GXV3000 video phone. I booted all the Grandstream phones and they immediately discovered the Grandstream GXE5024 and were autoassigned extensions 701-703.

Next, I figured I'd try some third party phones to see if the auto-provision worked with them as well. I attempted an Aastra 57i and a Polycom IP-650 phone, but neither seemed to auto-provision. The phones did discover the TFTP boot server (Option 66 via DHCP) on the GXE5024, but they weren't assign an extension or any SIP settings. Although the auto-provisioning didn't work, I was able to manually add the phones with no trouble.

After configuring some extensions, the next step was adding some trunks. The GXE5024 supports 4 analog lines, so I configured it to use a 4-port Teltone analog simulator very easily. In addition, I was able to assign extensions 790 and 791 to the two analog FXS ports, which can be used for analog phones, credit card machines, or fax machines. I also configured a SIP trunk using one of the promotional "trial" SIP trunk providers built into the Grandstream.


                                                              Auto-provisioning web tool

For unified messaging, it supports a voicemail-to-email feature with the ability to set the proper SMTP settings for proper email routing. Additionally, it also supports fax-to-email and can also automatically detect fax tones and route it to a user's fax mailbox. This is done per extension, so users can simply logon to their web portal using their username and password and retrieve their voicemail and faxes. Obviously, with a web interface, remote teleworkers can access voicemail and faxes quite easily. Your personal web portal also lets you manage individual phone/call settings.

The GXE502X supports 2 (GXE5024) or 4 (GXE5028) password protected conference bridges that allow up to 12 (GXE5024) or 20 (GXE5028) simultaneous participants from PSTN trunks, SIP trunks or Internal Extensions.
You can simply dial the conference bridge extension to join, or even invite other participants by entering in their extension from the web interface. Administrators can also mute/unmute conference participants from the web interface as well as kick them out.

I liked that I was able to record the auto-attendant greeting from the web admin simply by choosing my extension to ring, but the same is not true for users. Unfortunately they can't ring their desk phone from their personal web portal to record their personal greeting. Instead, users have to record in a specified format - 8kHz,16 bits, mono and manually upload it via the web tool. Of course, they can also dial into the voicemail system and record their greeting. It's just a nice feature to be able to do it via the web rather than navigating the voicemail system.

Important to most small and medium businesses is support for hunt/ring groups. The GXE502x series supports parallel (simultaneous) ringing, as well as serial or round-robin ringing. Adding various auto-attendant menus such as business hours, after hours, and holidays was pretty straightforward. It lets you select days of the week as well as specify exception dates, i.e. 12/25 for Christmas, 1/1 for New Years Day, etc. Overall, defining call routing rules and auto-attendants was pretty easy to do.

One of the most powerful features of the GXE502x is the ability to define call queues so calls are answered in the order they were received and assigned to agents with the best skills. Having advanced queues and skills-based routing in such a low-cost IP-PBX is unheard of, so I tip my cap to Grandstream.

Some of the features of the call queues include:

• Priority: Lets you set the priority of the call queue from the drop down list.
• Queue Status Update Frequency: This determines how often callers will be updated on the status of the queue via an uploaded update message.
• Other Announcements: This determines the frequency in which any other announcements that the user has added will be played tcallers in the queue.
• Maximum Caller Wait Time: This field lets you set the maximum amount of time that callers will wait within the queue before being forwarded to voicemail.
• Maximum Queued Callers: This field allow users to set how many callers can be within the queue simultaneously.
• Group Email Address for Voicemail Delivery: Enter the email address where all voicemail for the queue/group will be delivered.
• Automatic Call Distribution: This setting lets users configure, enable and disable skill-based call routing. If skill-based routing is enabled, users can configure the call to be routed to the least skilled or most skilled agent first. The skill level for each agent can be configured on the agent page.
  Ring Mode to Agents of Same Skill Level: This feature is similar to that of hunt/ring groups as users can configure the mode in which agents of the same skill level will ring.
  Serial - Agents ring one at a time based on availability.
  Parallel - All agents ring simultaneously.
  Circular - A different agent will ring first each time a caller enters the queue.
  Least Busy - The least busy agent will ring first.
  Serial Ring Attempts Per Member: If the serial ring mode is selected, you may select the number of ring attempts to each agent before forwarding the call to the next agent.
  Serial Ring Interval: If the serial ring mode is selected, users may set the ring time interval (in seconds) between call queue members.

It also supports "Agent Call Wrap-Up Time" which allows users to specify the amount of wrap-up time an agent will have before receiving another call (time between two calls).

One really cool feature is that the GXE will allow admins to capture all the packet traffic coming in and out of either the LAN or WAN Ethernet interface of the GXE. This is very helpful to debug certain configuration issues and do SIP troubleshooting.

The GXE502x supports peering with other IP-PBX systems. While this may not be critical for SMBs, as companies do grow, they can deploy multiple GXE502x devices if they so desire and have a unified extension dialing plan that routes calls over IP. Further, larger organizations might deploy the GXE502x at branch office locations and peer back to their main corporate headquarters' IP-PBX.

I asked Khris Kendrick, Sr. Director Business Development for Grandstream some questions about the GXE502x. I asked, "Since this product was launched, how has this been performing in your channel?" Khris responded, "The product has been in the market for over a year now and it has receive great accreditation from our users and distributors, that said we will be offering more power and features to the IP-PBX in the near future."

"What are the top couple of features customers like the most in the GXE?" He responded, "The main features are the multiple Auto Attendant capability, One-Button-Service Provider Provisioning, ACD and UC features, our fax to email is definitely a fav."

Lastly, I asked, "What key advantages does the Grandstream GXE product line have over some of your IP-PBX competitors within the SMB space that often cause customers to choose your product?" Khris answered, "One-Button provisioning, Mulitmedia capability(video surveillance) support, Hybrid FX0/FXS and embedded fax server."

Feature Specifications:

Feature Specifications	GXE5024	GXE5028
FXO Ports	4 FXO	8 FXO
FXS Ports	2	2
Ethernet Ports	1 x WAN, 1 x LAN (10/100Mbps, integrated PoE)	1 x WAN, 1 x LAN (10/100Mbps, integrated PoE)
PSTN Life Line Ports	2 PSTN fail-over life lines	2 PSTN fail-over life lines
Peripheral Ports	USB, Audio In, Audio Out	USB, Audio In, Audio Out
Conference Rooms	2	4
Unified Message Storage	75 hours of voicemail, 5000 fax pages, 2 hours of video mail	150 hours of voicemail, 10000 fax pages, 4 hours of video mail
Registered Extensions	100	100
Voice Codecs	G.711, G.723, G.729 A/B/E, G.726, iLBC, T.38 fax relay	G.711, G.723, G.729 A/B/E, G.726, iLBC, T.38 fax relay
Video Codecs	H.264, H.263/H.263+	H.264, H.263/H.263+
Communication/Security Protocols	TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DDNS, DHCP, NTP, TFTP, TELNET, HTTP/HTTPS, PPoE, SIP(RFC3261),STUN, SRTP, TLS/SIP	TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DDNS, DHCP, NTP, TFTP, TELNET, HTTP/HTTPS, PPoE, SIP(RFC3261),STUN, SRTP, TLS/SIP
Compliance	FCC : Part 68 & 15B; CE: EN55022, EN55024, TBR21, EN60950, C-Tick: AS/NZX CISPR22, CIS PR24 A-Tick :  AS-ACIF S002, AS/NZS60950 UL (power supply)	FCC : Part 68 & 15B; CE: EN55022, EN55024, TBR21, EN60950, C-Tick: AS/NZX CISPR22, CIS PR24 A-Tick :  AS-ACIF S002, AS/NZS60950 UL (power supply)
Universal Power Supply	Input: 100-240V, 50-60Hz Output: 12 Vdc, 1.25Amp	Input: 100-240V, 50-60Hz Output: 12 Vdc, 1.25Amp
Configuration & Management	HTTP/HTTPS, TELNET, Syslog, TR-069 (pending)	HTTP/HTTPS, TELNET, Syslog, TR-069 (pending)

Overall, I was very impressed with the feature-set of the GXE5024. It's very small and lightweight partly due to its use of Flash memory and not a hard drive. I still find it amazing that such a lightweight PBX can pack such a powerful punch. After all, the PBXs of old weren't called "big iron" for nothing! The GXE502x appliance is a excellent all-in-one communication solution for small-to-medium sized businesses sporting a plethora of features at an attractive price-point, so I would not hesitate to recommend it.

Ratings Score Installation Documentation Features Usability Performance Value Overall

Website: www.grandstream.com
Price: $899 : GXE5024; $1399 : GXE5028

Pros:

• Firmware upgrades are free for the life of the product
• Very low cost IP-PBX appliance
• Includes voicemail-to-email and fax-to-email
• Integrated fax server
• Excellent analog support - both FXS and FXO
• SIP trunks and SIP-based peering supported
• Two conference bridges built-in
• PoE-enabled LAN port
• Web admin tool
• Supports auto-fax tone detection
• Busy Lamp Field (BLF) and Message Waiting Indication (MWI) support

Cons:

• Web admin tool could be more intuitive and have links to more Web-based help & tutorials
• No support for auto-provisioning of 3rd party phones
• Relies on external PoE switch for IP phones Would be nice if it had a 4 port PoE switch built it, especially for SMBs looking for a "unified" phone platform. Less costs and one less support vendor to deal with.
• No call recording

Shhh don't tell Rich Tehrani, but I think I got my email-to-blog
script to work with iPhone .mov videos. Once he finds out, he'll be posting tons of iPhone videos on his new iPhone 3GS and putting me to shame! Just testing it here first.
That's my daughter Hannah sleeping in the swing. If I slept with my
head to the side like her I'd be making a chiropractic appointment!



click Play button above to play video

Microsoft Office Communicator Mobile is a versatile Java-based unified communications client for Microsoft Office Communications Server 2007 R2. Communicator Mobile runs on Microsoft Windows Mobile 6.0 or higher which includes Pocket PC and smartphone devices.

It also supports:

• Motorola Razr V3xx
• Nokia S40 series: Nokia 3120 Classic, Nokia 3600 slide, Nokia 5220/5310/5610 XpressMusic, Nokia 6212 classic, Nokia 6300i, Nokia 6301, Nokia 6500 classic, Nokia 6500 slide, Nokia 6600 fold, Nokia 6600 slide, Nokia 7210/7310/7510/7610 Super Nova, Nokia 7900 Prism, Nokia 8800 Arte.
• Nokia S60 series: Nokia E 51/63/66/71, Nokia N95

This month they released an important hotfix update for Communicator Mobile 2007 R2.

What's cool about Communicator Mobile 2007 R2 is that it enables users running the app on their mobile phone to make work calls using their corporate phone system leveraging "Single Number Reach" functionality. This allows you to use a single telephone number on your business card. With Single Number Reach, your desk phone and mobile phone will ring when an incoming call arrives. Importantly, outbound calling on your mobile device also gives the same caller identity regardless of whether you use a desk phone or a mobile phone. Another key advantage in Communicator Mobile R2 is that you can simultaneously sign in to more than one application at the same time and have multiple options for communication. In order to keep presence information up to date, the presence indicator now shows "Mobile" as an option for your availability, enabling people to easily keep track of your status. Apparently they've done some major overhaul to the code from the prior version since Microsoft said they optimized the performance and battery life is now improved by 350 percent.

Issues that this month's hotfix package fixes

• Provides home screen support for new home screens in Windows Mobile 6.5+ phones.
• Provides integration within the phone dialer for Windows Mobile 6.5+ phones.
• Enables Communicator Mobile 2007 R2 to recognize when the phone is roaming and by default prevents Communicator Mobile 2007 R2 from signing in to roaming networks.
• Provides additional support for joining conference calls from a Windows Mobile appointment. To do this, press Menu, and then press Join Conference.
• Lets users log on by using a user name in the user@example.com format, in addition to the domain\user format.
• Enables the functionality by which callbacks are now automatically accepted when the user uses the Call via Work option.
• Resolves the problem in which the Microsoft Installer (.msi) installation fails on a Windows XP Service Pack 3 (SP3)-based computer. In this situation, users should install Communicator Mobile 2007 R2 by using a (.cab) installation.
• Fixes the problem in which AT&T FUZE devices that are set for a High-Speed Downlink Packet Access (HSDPA) connection cannot handle voice and data at the same time. In this situation, calls that use the Call via Work option fail unless the device is reverted to 3rd Generation (3G) by disabling HSDPA.

I came across a Microsoft page that lists "OCS R2 XMPP Gateway" with a General Availability date of 9/28/09. Very interesting. This would enable for example Google Talk (XMPP) users to instant message (IM) OCS users. Jabber XCP has a "SIP/SIMPLE gateway"  as well, but I would assume a Microsoft OCS 2007 R2 XMPP gateway might feature tighter integration.

For instance, it might automatically sync the XMPP users with the OCS 2007 contact list, so any users added to the XMPP server are automatically added to the OCS 2007 contact database, which then gets pushed out to all the OCS 2007 Communicator clients. No need to manually add an XMPP contact manually one-at-a-time. Of course, that's just speculation, so we'll have to wait and see what features this XMPP gateway has come 9/28/09!

Wired is reporting rumors about Apple preparing a new version of the iPod touch that includes a camera and microphone, which when combined with Skype for the iPod touch would negate the need for a home phone line. VoIP on an iPod touch? That's just heresy! Essentially, the iPod touch becomes an iPhone without the need for an AT&T contract.

The article then points out you can add a portable Verizon MiFi 3G wireless access point, which shares your 3G connection using WiFi to finally bring the iPhone experience to Verizon's 3G network. -- and without AT&T's locked-in contract obligations. Of course, you'll need a Skype account with SkypeOut minutes plus a SkypeIn number, which aren't exactly "free". Or you can use another SIP provider using a SIP softphone for the iPod touch. Both fring and Nimbuzz offer SIP capabilities built-in, and there are other apps as well.

With yesterday's news about the new Nimbuzz for iPhone app and the launch of a Nimbuzz Mac client, I contacted Tobias Kemper, Head of Communications for Nimbuzz and asked him a question about DTMF/touch-tone support in any pending release.

Any plans for DTMF support?

Noticed when I tested it by calling my Skype for SIP account matched up to my corporate PBX that I couldn't reach my extension since there is no dialpad once the call is initiated.

I know touchtones are tricky when sent over an IP connection, which is what RFC 2833 is for. However, RFC 2833 is probably impossible for you guys to support.

However, I've found if you simulate the exact frequency tones and send that as a long tone - say 0.5s long, 99% of the time it gets transmitted just fine. Can often get away with ¼ of a second as well. Thoughts?

He wrote back, "I will send you the official news in a few mins - WITH more stats and data! DTMF is a great next step for us but I cannot speak to it yet."

Ok, so he's not admitting to DMTF support in the future, but he isn't ruling it out either. That's good news, since I really like Nimbuzz. It features SIP capabilities, so I can register it with a SIP-based PBX. Though oddly enough, I had a minor issue with the Nimbuzz for iPhone app with case-sensitivity. I have some test SIP accounts with the format SOFTPHONE01, SOFTPHONE02, etc. all in upper-case. The Asterisk-based trixbox PBX is case-sensitive, so it is expecting the username in uppercase. Well, although I was able to enter in SOFTPHONE01 into Nimbuzz in all upper-case, it wouldn't register. So I SSH'ed into the trixbox server and ran "asterisk - r" to see what was going on. I attempted to register again, and immediately saw the problem in the Asterisk console. Nimbuzz converted my SIP username to all lowercase. Thus, I had to add a new SIP account that was in all lowercase. Looks like they need to update their back-end database to support mixed-case for the SIP username. It's pretty common to use a mix of upper & lowercase letters to help secure the SIP account.

In any event, Tobias also told me, "We developed a Mac client (download) in combination with a new iPhone version (download) because we got massive traction since releasing the iPhone client and have a huge number of active users. (official press release below)  We did it Apple style, focused on slick UI and usability! "

He added, "(Here are) a few stats on why we made the decision to deliver something specific for the Apple fans out there:"

• In just 9 months since launch, consistent Top 10 ranking on all iTunes AppStores in Europe, Latin America & the Middle East, including regular No. 1 What's Hot positions in influential iPhone markets such as the UK
• Achieved massive penetration and popularity.  For example, 1 in 5 iPhone users in France and in the Netherlands are using Nimbuzz!
• Average 848% growth in downloads of iPhone & iPod touch apps in key European markets since the iPhone OS 3.0 update
• One of just 5% of Apps to boast more than 100,000 active users, with an active user base of more than 41% (AdMob Metrics)
• Massive nearly one million newly registered users per month growth across all Nimbuzz clients

With the new Nimbuzz Mac you can connect and interact with friends from the most popular instant messaging communities and social networks like, Facebook, AIM, MySpace, Google Talk (Orkut), Yahoo! Messenger, Windows Live Messenger (MSN), ICQ and many more, right from your (i)Mac or MacBook (Pro).  It also lets you call your buddies on Nimbuzz, Windows Live Messenger (MSN), Yahoo! Messenger and Google Talk.

The iPhone and iPod touch upgrade lets you share photos, music and videos. The beauty of it is that the files are stored online so you have access to your files from your Mac, iPhone or iPod touch! That also saves device memory.

Push Notification for iPhone and iPod Touch - with home screen alerts of incoming calls or chats, effectively keeping you available to the rest of the world even though the app itself has been closed.

We also introduced another much requested feature: location sharing.  Now you can share and retrieve the location of your Nimbuzz buddies on the go. This makes it easier to find your friends and set up face to face meetings.

Here is a demo video to check out before trying : )