Quick round-up of things to bolt on to your Asterisk PBX.
35 Free Asterisk Applications
Quick round-up of things to bolt on to your Asterisk PBX.
Quick round-up of things to bolt on to your Asterisk PBX.
After waiting a couple days after applying, I received an eMail invitation for Google Voice. This service, formerly called GrandCentral before it was acquired by Google, allows you have a phone number which is independent of your actual phones. It allows you to create a phone number in an area code of your choice. Calls from this number are forwarded to any or all of your phone lines (home, work, or cell).
Using Google Voice, I’ll finally be able to ditch MCI long-distance service on my work landline. Calls within the U.S. and Canada are free. The way this works is that you enter the calling number from your list of contacts on the web site, and designate which phone (work, home, mobile) that you want to talk from. Google Voice will then ring your phone. Once you pick up you’ll immediately hear a dial-tone as it attempts to ring the called number.
Calls from the U.S. to Germany landlines and Hong-Kong are 2 cents per minute. Some African countries (Gambia) are 33 cents per minute. Rates for to mobile numbers can be much higher; German calls are 18 cents when the recipient is a mobile phone.
Google Voice includes a full-fledged voicemail service. There is a text to speech service which attempts to render messages into text, and then display them in an eMail-like list. I tried a similar service that was provided with a standard TrixBox installation.
The standard greeting isn’t anything like the graceful “Alison” provided with TrixBox and Asterisk installs; instead it something like, um, “yenta-in-a-hurry”. I’ll have to redo my own voice greeting shortly.
You can send SMS messages to mobile phones by typing the message into a web page. Great for those of us who don’t text.
You can screen all unknown callers (i.e those with out an entry in the contact list which include their caller ID), or you can screen blocked callers.
Notifications of new voicemails can be sent to an eMail address, and/or mobile phone via text message.
Of course some of the usual things are also included, such as the ability to listen to voicemail from your phone, variable greeting by caller, the ability to forward voicemail. You can also record calls and store them online and create conference calls
So, Google Voice has definitely passed the Five Minute Test…. and it looks quite promising.
Especially during a recession, it amazes me the utter contempt company call centers show toward their customers. I spend a lot of time on calls with technical support people, and it remains as irritating as ever to get to them. Once I get a live person, however, I can usually calm down.
I really hate hearing that “This call may be monitored or recorded for training or quality control purposes”. Especially right at the outset of a call. Maybe for my broker… (what broker?) when giving financial instructions. For all the “training” that is going on sitting in phone tree hell doesn’t seem to be getting any easier. And quality control purposes? It makes me uneasy that I’m being recorded at all.
On hold, I really hate hearing every 15 seconds that “We appreciate your patience, and thank you for waiting during this brief delay”. and “We know that you are very busy, we appreciate your call”. Or worse, blabbing on about the web site, or the new product, or alternate ways to contact us, or whatever. These constant interruptions makes it impossible to concentrate on other work while waiting. What ever happened to playing Vivaldi and not inserting commercial messages?
While there are “secret” phone numbers floating around the internet for various services, I can’t imagine a company would want these internal numbers published; they would get spammed quickly.
So, I’m holding for Fairpoint right now, and have had 5 dumb “Thank you for holding messages” in the previous minute.
“Thank you for holding, your call will be answered in just a moment”
“We know your time is important, and appreciate your patience while on hold”
“Every effort is being made to ensure that your wait is as short as possible… Thank you”
“Thank you for holding, someone will be right with you.”
“Your call is very important to us. Thank you for waiting and bearing with us during this brief delay”.
And then the cycle starts again. All this accompanied by ear-splitting muzak (tacky fake FM-synthesized saxophones.)
I haven’t focused on VoIP for awhile… but others do.
FierceVoIP announced that Logitech has bought out SightSpeed. Sightspeed was (is) one of my favorite videoconferencing applications, and it will be very interesting to see what becomes of the product.
VoipInsider reports that Polycom has updated the firmware for Soundpoint phones.
VoicePulse has announced a fail-over option for their accounts. They’ve completely redesigned their web site with a new interface, that looks really classy. This is one VoIP provider who appears to be here to stay. Hooray!
Windows for Devices reports that Motorola will discontinue development of Symbian and MotoMax phones, and concentrate on Windows Mobile, and Google Android. The site emphasizes hardware running embedded versions of Windows, there is a companion site for the Linux crowd at Linux For Devices.
The wiki documentation for Ekiga has a nice discussion of how to deal with routers when using SIP and H.323.
In addition to Trixbox there is a non-derivative project called Freeswitch which is not based directly on the Asterisk open source code. From the introductory web page:
FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow.
We support various communication technologies such as SIP, H.323, IAX2 and GoogleTalk making it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.
FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.
FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 16 or 32 kilohertz and can bridge channels of different rates.
FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.
Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipX, The Asterisk Open Source PBX and Call Weaver.
I was intrigued to see that VoIP Supply, the folks that sold me my Trixbox and my Polycom SIP phones are now offering SIP trunking and data services.
Don’t know how this stacks up against suppliers like VoicePulse. For one thing the pricing model is slightly different, with VoIP Supply looking for a minimum $25.00 per month, but with unlimited local and long-distance calling in the lower 48 states. VoicePulse, at least the version for Asterisk/Trixbox, was on a pre-paid model but charges 2 cents or so per minute.
What about the quality of these calls though? Maybe I’m just cranky, but I’ve had literally dozens of calls from vendors in the past year that clearly were low-quality VoIP calls. I would be appalled if my own calls to my clients and prospects sounded like many of these calls.
In one case, I was (supposedly) working with a sophisticated and highly-paid consultant who was using either Vonage or the Comcast VoIP. The guy couldn’t get out of his own way…I just couldn’t understand him, over multiple calls. How are we supposed to conduct business this way? And, where is the savings per month, at $25.00 or $125 or even $1025 per month that the person is supposedly saving, when as a result a client drops this person, after originally looking forward to a multi-thousand dollar contract? False economy.
Bottom Line: The landline isn’t dead yet. Use VoiP for long-distance calls to friends and family, and non-critical overseas calls. If there is any question, during a VoIP call, have a back-up landline available.
And if you have contracted out any functions to a call center (perish the thought…my local newspaper has done this to verify authorship of letters to the editor), be sure you get yourself on the receiving end of such calls to assess the quality. Nothing turns off customers and prospects more quickly then struggling with foreign-based tech support, heavily accented, with stupid calling scripts, and bad sound quality.
The Trixbox Wiki has a number of digestible pages of advice on how to successfully deploy a VoIP application. Here are recommendations for remote sites.
Formula for the best remote telecommuter Experience
- Use T1 internet access at the main location, not DSL or Cable.It’s worth the additional expense in order to ensure good, steady performance at your main location.
- If your routers and/or firewalls support QoS features, activate them. Give priority to the SIP and RTP protocols. Consider replacing equipment that lacks VoIP-aware QoS features. See Also: How do I use QoS on my network?
- Consider using one of our Suggested Routers with QoS on both ends of your connection.
- If your QoS solution allows you to limit total bandwidth, set the limit to slightly less than the line speed of your internet connection. Use a DSL line speed test to determine where you should set your limits. Setting it about 5-10 Kb below your maximum speed will keep the packet buffers from filling up on your DSL/Cable modem. This will yield better overall performance.
- Consider having two internet connections… one for your existing data application, and one for your VOIP phone and trixbox Pro servers. You can use this approach in your main location, as well as your remote locations. If you use this approach, you may not need any QoS capable equipment.
- If possible, connect your main office and your remote office using the same internet provider. Usually performance on the same provider’s network is superior to the performance when traffic needs to traverse multiple internet backbone networks.
- If possible, remove NAT devices between the trixbox Pro system, and the remote telecommuters.
- If you must use a NAT configuration, consider using a “DMZ Host/Server” configuration rather than port forwarding. This uses less CPU power in the router/firewall and yields optimal performance.
- At the main location, the setting will forward all unknown packets to your trixbox Pro server.
- At the remote locations, the setting will forward all unknown incoming packets to the IP Phone.
- Reserve the phone’s IP address in DHCP or give the phone a static IP Address on your private network in the remote location so the IP Address does not change. If you use a static IP Address, pick one outside of your dynamic DHCP IP Address range.
- For mission critical remote employees, consider using a fractional T1 internet service at the remote office instead of a Cable/DSL connection.
One deficiency is that the article doesn’t really encompass the whole picture necessary for putting in VoIP. For example, most installers would consider using conventional PSTN phone lines or a T-1 connection for multiple lines, rather than attempt to use public IP connections for their “production” phone trunks.