Business Internet Phone Service Blog

SIP based communication services are changing the world. If you haven't experienced the revolution yet you will, here's why.

1)It allows you to elegantly juggle how you want to respond to incoming calls. On the phone with an important customer but see a vital call from an employee about to go to sleep in Germany? With a SIP based solution you can re-direct the incoming call to an enterprise IM response and save it from ever going to voice mail. Now your closing the deal and ensuring the product will be available seamlessly.

2)It's anywhere you are. Many people are still tied to a physical phone location. Are you still changing your voice-mail while you are traveling to direct customers to your cell phone? With SIP you will never send your important customer looking for pen and paper again--your number will find you wherever you are, on whatever phone you input.

3)It does everything a million dollar PBX solution used to do. Want to dial shorter interoffice extensions? Want a fancy attendant menu when people call your business? SIP can do it, digitally at a fraction of the costs. Do you have more than one employee filling a similar role? Use the hunt group function to ring each of their lines in succession instead of sending a valuable customer to voicemail.

4)Data, data, data. How long does it take for incoming calls to be picked up by the sales department? Who answers the most incoming sales calls? Which customers are calling the service department the most. Because SIP solutions are all digital data is readily available. Great managers have great metrics; which new ones can you create with SIP at the core of our communications?

If better, faster, cheaper, more transparent, slicker; and downright cooler sound sexy to you check out SIP.

Being your own boss is great, but every business, even one run out of your home, has to turn a profit. The shorter the time between writing your business plan to being in the black, the better. Most of that will depend on how successful you are at landing clients or making sales. But you can help things along by keeping your expenses down.

The best place to start is with the things you don't need. Yes, the trappings of business can be cool and make you feel like you're a real business, but if you're just starting out it might be best to save the fleet of company trucks until a bit further down the road. If you can run your business out of the spare bedroom, for instance, that might make more sense than renting downtown office space. Even if you do need to get out of the house, look for cheap alternatives to high rents. When short story writer Ray Bradbury discovered he couldn't get any work done at home, he started working in his local public library.

Staying flexible in your business is a lot like staying flexible on the road. You need options that won't tie you down, and can grow with your business. If you're going to be doing business over the phone, you'll want a PBX eventually. The last thing you're going to want is telephone company technicians crawling through your home installing the thing, coming back to repair it, and then making periodic visits to install upgrades. Getting an online PBX gives you all the benefits of the real thing, without the equipment taking over a corner of your in-home office. It also means you can adjust how your calls are routed and check your call stats from the library, coffee shop, or wherever you are best able to work, even if it's not where your PBX is.

A flexible PBX also allows you to take advantage of competitive pricing that might otherwise be beyond your reach. With virtual PBX, just because you live in Chicago doesn't mean your employees need to. This means you're not limited to the local labor pool, and can seek out people who have key skills all across the country. Even if your team is spread across the continent, your phone system will route calls to everyone as if you were all in the same building, even if none of you are actually inside a building.

Finally, one of these experts should probably be an accountant, and you'll probably want them to be at least familiar with the laws of your state. A good accountant can help you keep track of exactly where your money is going, and help you keep track of data you'll need to maximize your tax savings. Money is the lifeblood of your business, and if you don't know what it's doing, you can't speak intelligently about how healthy your business actually is. It might seem a large expense at first, but as soon as you can't keep it all straight in your head, you'll need someone on top of the money before it gets away from you.

Photo credits: borman818, Lachlan Hardy.

Have you ever wanted to give someone in the office the chance to answer the phone before the call routes to your auto-attendant? Thanks to a conversation with Cheryl during a new account set up, I think we’ve come up with a great solution. The old methods were kind of unwieldy, using two lines on your phone or having a user’s no answer destination become the auto attendant thus robbing them of their own voicemail box like the rest of your employees.

What Cheryl and I did was create a group called Receptionist group and set the members to be the one user: Receptionist. Then, under settings, we made the failover of the group the company’s auto-attendant after ten seconds. Finally, under Resources, we set the destination of the main DID to be that newly created Receptionist group.

So the call flow now ran like this:
Caller calls in, the call is handed off to the Receptionist group which has one member: the Receptionist. If they’re available to answer, they do so and are able to provide direct one ring assistance to that caller. If they were not available after ten seconds, the call would then ring to the Welcome auto-attendant providing indirect one ring assistance to that caller.

Being on the road is no longer an excuse for being out of the office. Keeping on top of things on the road or in the air requires tools that keep you informed, organized, and in touch. Last time, we mentioned online CRM and noted that among its many strengths is your ability to pull up customer information anywhere you have internet access. With the 'net, you can take the rest of your office with you, no matter where you go. Here are two more tools to help you do that:

Voice Over IP (VOIP): Voice Over IP is basically just using your internet connection to transmit audio in real time, just like using a telephone. It has a number of key benefits over standard telephone and cellular services. The bulk of these stem from using a virtual PBX to manage your calls. This allows you to, among other things, change hunt groups and call routing at the drop of a hat through a web page, have a single number ring at multiple locations simultaneously, and track all your phone traffic through automated reports. It also allows you to treat locations that are geographically scattered across the contiguous 48 states as if they were all at the same location. A client who calls the main office in San Francisco can talk to an agent in Chicago or Seattle just as easily as they would someone across the hall. Having VOIP means you're always in touch, and your clients can always reach you.

Google Docs: Google Docs do for your online documents what online CRM does for your client resources. With your standard array of documentation software, such as spreadsheets and word processing, Google Docs allows you to save and share your work online. This means everyone in a scattered organization can access it, and it also means that you can pull it up on your laptop. So when the client calls over your VOIP service about your last billing report while you're in an airport halfway across the country, the numbers you need are a mouse click away.

Photo Credits: JoF, jimmyharris.

Having the right tools is vital to making the most of your business. It seems today that everyone has a new gadget or toy that can save you time and make you money. The real trick is figuring out which ones are right for your business. Here is a short list of recent innovations that really do make sense for many small and medium sized businesses.

Twitter - We'll start with what has probably become the most hyped new tool out there today. On paper, Twitter makes no sense. Mini-blogs of 140 characters seem silly. In practice, however, Twitter has become a potent builder of communities. Dell has announced that they've moved $1 million in merchandise through their Twitter groups. Domino's Pizza turned to Twitter when they needed to reassure customers after a disastrous YouTube video showed employees engaging in very unhygienic hijinks in the kitchen.

Twitter is easy to use, though it's suggested you get a helper application like TweetDeck to keep track of communications sent and received. The most obvious use of Twitter is alerting your fans and customers of new promotions and initiatives. Twitter communication can, however, work both ways. If you find a trendsetter your customers respect, following this person's tweets can keep you in the know about what your customers are interested in and what they think of you. It’s even better if you can become that trendsetter.

LinkedIn - This one hasn't gotten quite the buzz that Twitter has, but it may be even more useful from a business perspective. Keeping track of those old friends from college, contacts from previous ventures and jobs, and clients is a lot easier with this utility. Sometimes the difference between a big contract and the one that got away is who you know. With LinkedIn, it's a lot easier to find those old contacts who might be in a good position to help you today.

Google - According to a global Nielson Consumer Report published this past November, over 85% of the world's web surfers are shopping online. This was a precipitous increase from 40% in 2006. There's every reason to expect these numbers to grow. This means your customers are almost certainly using Google to find products and services just like the ones you sell. Actively engaging Google, either through the purchase of ads or the techniques of search engine optimization (SEO), only makes sense.

Hosted PBX - Your phones are the lifeline of your business. Your customers and clients need to reach you no matter where you are. A hosted PBX with a web interface to manage hunt groups makes it very easy for you to adjust who those calls reach inside your organization and adjust it as needed, without having to wait for a specialist to make time for it. In addition, a hosted PBX can help you track your calls, so you know just who is calling and why. Finally, a hosted PBX gives your small or medium-sized business the look of a big company, increasing customer confidence and willingness to buy.

Online CRM – Customer Relationship Management software helps you stay on top of everything you're doing with your clients, from prospecting to customer service. By organizing and standardizing your interactions with your clients, no matter who they speak with or what they want, they interface with every part of your organization as if it were a seamless whole. The most popular currently is salesforce.com. Because it's online, it travels with you everywhere you go. Even when you're in Boston, your office is in Chicago, and your client is in San Francisco, you have everything you need at your fingertips to complete whatever transaction comes up.

All of these technologies and tools are available for you today via the Internet. At the pace of change today, it can be difficult to keep up with all the new options available, but the health of your business requires you to make the most of every edge you can get. That means taking the time to learn the full range of your options, and which best fit your business model, is well spent.

Image credits: Appfrica, dannysullivan

Super techy post today.

We ran down a problem where a Grandstream phone had two users (Lisa and Michele) registered. Both users were in the same call group so the Grandstream phone received two calls that were nearly identical. The difference was the two calls designated two different call branches and were to different usernames, but both reflected the same call-ID (because they ARE the same call, just separate branches.)

After some work, we discovered that the Grandstream phone rang the line of the first SIP INVITE it received, but 'answered' the last INVITE. Following through the trail of INVITEs and CANCELs below, you can see where the confusion starts. We have submitted this information to Grandstream for their consideration - we are asking that their devices honor the "branch" tag.

For those SIP tech's out there, enjoy our analysis of a call to two users on the same SIP UA (all phone numbers, domains and IP addresses have been modified):

Two users in group Lisa and Michele

Lisa is on 192.168.1.107:5060 on a Grandstream - this phone receives Branch .2.
Michele is also on 192.168.1.107 but on port 5064 on a Grandstream - this phone receives Branch .5.

Customer reports that this call was picked up by the "Lisa" line.

**Invite to Lisa on Grandstream (GS) - Branch .2. of call ID 133...3a4c**
U 2009/05/08 09:40:18.325507 66.227.1xx.2xx:5060 -> 24.33.2xx.3xx:5060
INVITE sip:lisa@192.168.1.107:5060;transport=udp;aor=lisa%40xxuserdomainxx.onsip.com SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKd146.d2c78147.2.
Via: SIP/2.0/UDP 66.227.100.215:5060;received=66.227.100.215;branch=z9hG4bK1a92f9f1;rport=5060.
From: "0000000000" ;tag=as2bbf1438.
To: .
Contact: .
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
Date: Fri, 08 May 2009 13:40:18 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Content-Type: application/sdp.
Content-Length: 381.
P-Gateway: 66.227.100.215.
.
v=0.
o=root 26678 26678 IN IP4 66.227.100.215.
s=session.
c=IN IP4 66.227.100.215.
t=0 0.
m=audio 19870 RTP/AVP 0 8 3 111 18 4 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:111 G726-32/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:4 G723/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.


**Invite to Michele on GS - Branch .5. of call ID 133...3a4c**
U 2009/05/08 09:40:18.325571 66.227.1xx.2xx:5060 -> 24.33.2xx.3xx:5064
INVITE sip:michele@192.168.1.107:5064;transport=udp;aor=michele%40xxuserdomainxx.onsip.com SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKd146.d2c78147.5.
Via: SIP/2.0/UDP 66.227.100.215:5060;received=66.227.100.215;branch=z9hG4bK1a92f9f1;rport=5060.
From: "0000000000" ;tag=as2bbf1438.
To: .
Contact: .
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
Date: Fri, 08 May 2009 13:40:18 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Content-Type: application/sdp.
Content-Length: 381.
P-Gateway: 66.227.100.215.
.
v=0.
o=root 26678 26678 IN IP4 66.227.100.215.
s=session.
c=IN IP4 66.227.100.215.
t=0 0.
m=audio 19870 RTP/AVP 0 8 3 111 18 4 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:111 G726-32/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:4 G723/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.

**Receive "Trying" from Lisa on GS - Branch .2.**
U 2009/05/08 09:40:18.404981 24.33.2xx.3xx:5060 -> 66.227.1xx.2xx:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKd146.d2c78147.2.
Via: SIP/2.0/UDP 66.227.100.215:5060;received=66.227.100.215;branch=z9hG4bK1a92f9f1;rport=5060.
From: "0000000000" ;tag=as2bbf1438.
To: .
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
CSeq: 102 INVITE.
User-Agent: Grandstream GXP2000 1.1.6.46.
Content-Length: 0.
.

**"RINGING" from Lisa on GS - Branch .2.**
U 2009/05/08 09:40:18.416313 24.33.2xx.3xx:5060 -> 66.227.1xx.2xx:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKd146.d2c78147.2.
Via: SIP/2.0/UDP 66.227.100.215:5060;received=66.227.100.215;branch=z9hG4bK1a92f9f1;rport=5060.
Record-Route: .
From: "0000000000" ;tag=as2bbf1438.
To: ;tag=b71cb81c3beef471.
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
CSeq: 102 INVITE.
User-Agent: Grandstream GXP2000 1.1.6.46.
Contact: .
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE.
Content-Length: 0.
.


** "RINGING" from Michele on GS - .5.  We did not receive "Trying" from the .5. branch, but that's not an issue.  However, customer reports that only one 'line' is ringing on the phone itself: the 'Lisa' line.**
U 2009/05/08 09:40:18.479063 24.33.2xx.3xx:5064 -> 66.227.1xx.2xx:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKd146.d2c78147.5.
Via: SIP/2.0/UDP 66.227.100.215:5060;received=66.227.100.215;branch=z9hG4bK1a92f9f1;rport=5060.
Record-Route: .
From: "0000000000" ;tag=as2bbf1438.
To: ;tag=b71cb81c3beef471.
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
CSeq: 102 INVITE.
User-Agent: Grandstream GXP2000 1.1.6.46.
Contact: .
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE.
Content-Length: 0.
.

**Status:  At this point we have received provisional responses on both branch .2. and branch .5., but the customer states that only one 'line' on the phone is ringing.

**OK from Lisa on Grandstream - NOTE:  The ok is from the user 'Lisa' BUT it's in response to the .5. branch.  This is a problem.  The .5. branch was the user 'Michele's branch.  Theory - the first invite received is the line that rings, but when answered, the phone answers the last invite received.**
U 2009/05/08 09:40:27.712967 24.33.2xx.3xx:5060 -> 66.227.1xx.2xx:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKd146.d2c78147.5.
Via: SIP/2.0/UDP 66.227.100.215:5060;received=66.227.100.215;branch=z9hG4bK1a92f9f1;rport=5060.
Record-Route: .
From: "0000000000" ;tag=as2bbf1438.
To: ;tag=b71cb81c3beef471.
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
CSeq: 102 INVITE.
User-Agent: Grandstream GXP2000 1.1.6.46.
Contact: .
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE.
Content-Type: application/sdp.
Supported: replaces, timer.
Content-Length: 213.
.
v=0.
o=lisa 8000 8000 IN IP4 192.168.1.107.
s=SIP Call.
c=IN IP4 192.168.1.107.
t=0 0.
m=audio 5056 RTP/AVP 0 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.

**Sending "CANCEL" to Lisa at the .2. branch**
U 2009/05/08 09:40:27.713409 66.227.1xx.2xx:5060 -> 24.33.2xx.3xx:5060
CANCEL sip:lisa@192.168.1.107:5060;transport=udp;aor=lisa%40xxuserdomainxx.onsip.com SIP/2.0.
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKd146.d2c78147.2.
From: "0000000000" ;tag=as2bbf1438.
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
To: .
CSeq: 102 CANCEL.
Max-Forwards: 70.
User-Agent: OpenSIPS (1.5.1-notls (x86_64/linux)).
Content-Length: 0.
.



**OK back from Lisa on GS regarding CANCEL - Branch .2.**
U 2009/05/08 09:40:27.784852 24.33.2xx.3xx:5060 -> 66.227.1xx.2xx:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKd146.d2c78147.2.
From: "0000000000" ;tag=as2bbf1438.
To: ;tag=b71cb81c3beef471.
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
CSeq: 102 CANCEL.
User-Agent: Grandstream GXP2000 1.1.6.46.
Supported: replaces, timer.
Content-Length: 0.
.

**Lisa user on GS saying Request Cancelled - however this shows branch .5. is being cancelled - the Grandstream previously gave an OK to the .5. branch in the OK above, now it's cancelling the call.  Basically it's confused between the branches.  The customer experience is that the call now drops.**
U 2009/05/08 09:40:27.793189 24.33.2xx.3xx:5060 -> 66.227.1xx.2xx:5060
SIP/2.0 487 Request Cancelled.
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKd146.d2c78147.5.
Via: SIP/2.0/UDP 66.227.100.215:5060;received=66.227.100.215;branch=z9hG4bK1a92f9f1;rport=5060.
Record-Route: .
From: "0000000000" ;tag=as2bbf1438.
To: ;tag=b71cb81c3beef471.
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
CSeq: 102 INVITE.
User-Agent: Grandstream GXP2000 1.1.6.46.
Content-Length: 0.
.


**Junction Networks ACK's the 'Request Cancelled" on the .5. branch in the above packet.**
U 2009/05/08 09:40:27.793432 66.227.1xx.2xx:5060 -> 24.33.2xx.3xx:5064
ACK sip:michele@192.168.1.107:5064;transport=udp;aor=michele%40xxuserdomainxx.onsip.com SIP/2.0.
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKd146.d2c78147.5.
From: "0000000000" ;tag=as2bbf1438.
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
To: ;tag=b71cb81c3beef471.
CSeq: 102 ACK.
Max-Forwards: 70.
User-Agent: OpenSIPS (1.5.1-notls (x86_64/linux)).
Content-Length: 0.
.


**Asterisk PBX sending "BYE" to user Lisa on GS for call - now that all other branches were cancelled, the call is now branch .0.**
U 2009/05/08 09:41:18.413797 66.227.1xx.2xx:5060 -> 24.33.2xx.3xx:5060
BYE sip:lisa@24.33.2xx.3xx:5060;transport=udp SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKe146.a562be32.0.
Via: SIP/2.0/UDP 66.227.100.215:5060;received=66.227.100.215;branch=z9hG4bK3f1ae941;rport=5060.
From: "0000000000" ;tag=as2bbf1438.
To: ;tag=b71cb81c3beef471.
Contact: .
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
CSeq: 103 BYE.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
Content-Length: 0.
.

**Lisa on GS complaining that there is no such call to send the BYE to - that phone had already cancelled the call**
U 2009/05/08 09:41:18.480821 24.33.2xx.3xx:5060 -> 66.227.1xx.2xx:5060
SIP/2.0 481 No Such Call.
Via: SIP/2.0/UDP 66.227.1xx.2xx;branch=z9hG4bKe146.a562be32.0.
Via: SIP/2.0/UDP 66.227.100.215:5060;received=66.227.100.215;branch=z9hG4bK3f1ae941;rport=5060.
Record-Route: .
From: "0000000000" ;tag=as2bbf1438.
To: ;tag=b71cb81c3beef471.
Call-ID: 1330352b536b8d7a74970bcf4df83a4c@jnctn.net.
CSeq: 103 BYE.
User-Agent: Grandstream GXP2000 1.1.6.46.
Content-Length: 0.
.

Last month Nadeem Unuth posed the question: "are touch screens better for VoIP?" The central argument was that VoIP is closer to natural language and freer from the restriction of digits. Totally agreed. VoIP lends itself to a techie and glorious level of openness, with apps like fring®. VoIP is making smart phones even smarter. Why does anyone need to be tethered to desks or computers anymore?
Touch screens are here to stay simply because they are an appealing and useful innovation - but touch screens for the sake of it can be expensive and pointless for SMBs.

POI terminals cannot replace the value of being greeted by a real person, but they can provide basic information to people in search of help. The hospitality industry has widely adopted touch screens and for good reason. But I wonder, does the waiter or waitress at my local diner really need to save the extra ten seconds by entering an order on a networked tablet table-side? If his or her short-hand is no good, than maybe it will save about 15 seconds... When employing touch screen technology, companies need to get over the cool factor of being an early adopter and think of the business benefits. The business benefits of VoIP are clear, and they will still be there when touch screen adoption grows and interfaces improve.
I actually do agree with Nadeem about the possibilities of VoIP and touch technology, it just makes sense. Why not take that extra step and combine VoIP with touch screen technology?

I can easily picture a scenario where the waiter or waitress also need to answer calls for the Manager who had to step out. The manager can set up a hunt group from his/her hand-held tablet and feel sure that they won't miss that call for a reservation from Mayor Bloomberg's people or perhaps... Beyonce?

Top 5 most inappropriate uses for 911, in no particular order.

5. On a dare. If someone dares you to call 911, wait and check to see if you are no longer in junior high. If not, then find new friends - aren't you a little old for pranking?

4. Reporting fake crimes or exaggerating smaller ones. Your cat Fluffy is stuck in a tree! While this is definitely not an ideal situation, try a can of tuna or some treats before declaring an emergency. Fluffy might not be stuck at all, he/she might just be a little lazy, in a bad mood or stubborn.

3. Calling to ask the 911 operator or police officer on a date. While he or she is probably a great catch, there are probably better methods to asking someone on a date.

2. Testing your new cell phone. This is apparently very common during the winter holidays when everyone gets phones as gifts. Try testing with 411 instead!

1. Reporting any and all grievances related to fast food, whether it is with mcdonald's and the lack of nuggets or burger king's lack of lemonade. Sometimes, you order one thing at a restaurant and then get another.

I guess life is not always fair...

The Junction Networks lab is pleased to announce that we have reviewed the Stromberg Carlson Candlestick and have added it to our list of certified phones. Our configuration guide can be found here.

All around, this phone is pleasurable to use and has a competitive price to other mid-range phones. It has a very authentic ringtone that makes it ideal for VIPs. Unfortunately, the phone is not able to store speed dials or integrate with contact lists as other SIP phones can do, but the retro clicking dialing experience makes manually dialing the number a small sacrifice. Additionally, the rotary dial does not have letters printed on it, which may be a small setback for some.

The major problem with the Stromberg Carlson Candlestick is that it cannot pass DTMF tones, which provides some difficulty for dialing conference bridges and auto-attendants. All other features that we would expect of a fully-fledged enterprise phone operate appropriately. We worked around the DTMF issue by a liberal use of the transfer functionality. There are also currently no published future plans for the Candlestick to support high definition telephony.

Sound quality of the Candlestick is surprisingly good if the user follows the phone's usage guidelines and surpasses the quality of some other SIP phones that we've reviewed. For proper usage, the user should maintain a 3" distance from the transmitter (the "stick") and rest the elbow of the arm holding up the earpiece on a desk to resist fatigue. Deviation from the 3" distance can result in the user sounding faint to the other party, but the quality is still better than a call on the cellular network.

It is difficult to imagine a more durable phone than those produced by Stromberg Carlson. The craftsmanship far surpasses any other phone or PBX that we've seen in our labs. The etching of the company name on the back of the transmitter shows a real dedication to quality by the manufacturer that was readily apparent. Our inner geek also appreciates Stromberg Carlson's insistence on using bakelite, the world's first synthetic plastic.

Stromberg Carlson really ups the voice hardware game and provides a true challenge to other manufacturers. We look forward to seeing what results from this competition.

Insecure Extension Passwords on Asterisk (VoIP PBXs)

Junction Networks has become aware of four separate hack attempts against our PSTN Gateway customers over the last few days. Three of these customers were Asterisk customers and one was another SIP-based VoIP PBX. After communicating with our customers, it appears that the hack has nothing to do with any sort of Asterisk vulnerability, but with insecure passwords set for extensions. This blog post captures the issue well. Blocking the offending IP addresses at the router level does not help as they will just continue the attack from another address.

The best solution is to create secure passwords for your extensions. The passwords that come with sip.conf must not be used:

;[polycom]
;type=friend             ; Friends place calls and receive calls
;context=from-sip        ; Context for incoming calls from this user
;secret=blahpoly
;host=dynamic            ; This peer register with us
;dtmfmode=rfc2833        ; Choices are inband, rfc2833, or info
;username=polly          ; Username to use in INVITE until peer registers
                         ; Normally you do NOT need to set this parameter
;disallow=all
;allow=ulaw              ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no       ; Polycom phones don't work properly with "never"

Instead of secret=blahpoly, we would recommend that the password be at least 12 characters. Here are some good sites for password generation:

PC Tools

GRC

Cut and paste the secure password into sip.conf and into the phone. Use a different password for each extension.

Additionally, we would recommend the above strong random passwords in conjunction with limiting the IP addresses extensions can connect from to particular networks. There is some documentation on how to do this in your sip.conf here: http://www.voip-info.org/wiki/view/Asterisk+sip+permit-deny-mask

If all of your phones are on the LAN, and your LAN is 192.168.0.0/24 the input would be:

;Deny every address except for the LAN.
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/255.255.255.0

From the asterisk-security mailing list, Olle Johansson, the maintainer of the Asterisk SIP module had this to say...

[asterisk-security] Person Trying to Register on my Asterisk multiple times
Johansson Olle E oej at edvina.net 
Fri Jan 23 15:51:46 CST 2009

...

Attacks are never fun. Use the ACL (permit/deny) in sip.conf to block this IP or range of IPs at least. 

Or use IPtables. There are a lot of IPtables scripts to prevent this kind of attacks if you look at the solutions for the very common SSH attacks that keep testing multiple usernames. Maybe someone on the list has a version for SIP attempts over TCP and/or UDP?

It's always good to have a bit less obvious peer names than the ones they test. Don't use usernames or extension numbers. Make sure you separate the namespaces. Kevin usually suggest Ethernet MAC addresses, which are harder to guess, but still relates to something even though they do have a well-known pattern.

Finally, it's important to make sure you have good passwords. There's no reason to have simple passwords in something you only install in software in devices or applications. There's no user who has to learn to remember the MD5 auth secrets.

That's my 10 cents. Please, list, fill in and correct me when wrong!
/O