We’re going to take a little time off for Spring Break and leave you with a terrific new tutorial from our good friend, Tom King. But first, despite pitching Google Voice as one of Asterisk’s Top 10 Tricks as recently as last October, Digium® apparently has had a change of heart. Our frustration with Asterisk® and Digium over the tepid support for Google Voice™ continues to build with the discovery that the latest (several) releases of Asterisk 10 break Google Voice connectivity entirely. The default Asterisk 10 install in PBX in a Flash™ continues to work just fine. The Digium response can be summed up in two words: “Oh Well.” They’re apparently too busy doing Amazing New Things™ to worry about keeping your one-month-old PBX functioning reliably. So… we’ve pretty much given up on Digium’s attitude toward Google Voice ever changing. It’s simply not a priority for them which, of course, is their prerogative. But it also means everyone needs to start considering other alternatives if Google Voice reliability matters to you.
So today we start down a new path for our users and readers as well as the rest of the VoIP community. We hope to have a FreeSwitch® announcement soon to reliably handle Google Voice and Skype for Asterisk-based servers. These two functions have worked flawlessly with FreeSwitch since Anthony Minessale and Brian West first released them a couple years ago. In the meantime, reliability of Google Voice in Asterisk continues its downward spiral with almost monthly nightmares. The latest debacle is a month old today. Happy Birthday!
There’s another alternative as well. Sherman Scholten at OBiHai tells us they are poised to release the OBi202 with all the usual OBi110 goodies plus T.38 real-time faxing over IP plus support for PPPOE, VLANs, and up to 4 SIP or Google Voice trunks. Add a firewall with DRDOS attack protection and VPN pass-through plus some amazing PBX-like functionality for management of collaborative calling, and you really couldn’t ask for much more in a product which will retail for under $100. OBiHai has been kind enough to send us a complimentary unit, and we’ll have a full review for you soon.
In the meantime, we have a short term answer for anyone that depends upon Google Voice to perform tasks (such as making phone calls) where reliability matters. It’s the under $50 OBi110. You’ll find a link to buy one while supporting Nerd Vittles in the right column. And today we’ll show you how to set it up to use with Asterisk and PBX in a Flash™ so that Google Voice calls flow into and out of your server reliably and transparently without worrying about who may have “improved” things while you were sleeping.
PIAF2 Preliminaries. If you’re currently using PBX in a Flash 2 for your Google Voice needs, then the first thing you need to do is remove any Google Voice trunks you’ve activated using the Google Voice module in FreePBX. Once you’ve done that, you’ll also want to disable the jabber and gtalk modules in Asterisk. This has no impact upon the separate gvoice command line utility which will continue to work fine with the speech-to-text apps that we’ve released over the last month. The Google Voice for Python project is well supported and (fortunately) is separate and apart from the Asterisk project. We’ve also documented on the PIAF Forums how to keep gvoice running reliably on your server.
To disable Google Voice in Asterisk, log into your server as root and edit modules.conf in /etc/asterisk. Change the two lines in the [modules] context for these two modules by changing the word load to noload. Then save your changes and restart Asterisk: amportal restart.
noload => res_jabber.so
noload => chan_gtalk.so
Step2. Once you have your OBi110 in hand, the rest of the process to get it handling inbound and outbound Google Voice calls for Asterisk is simple as long as you don’t skip any steps. Just download Tom King’s new tutorial and follow along. You’ll be up and running in under 15 minutes with a reliable, independent alternative for Google Voice calling with Asterisk. Enjoy!
Originally published: Friday, March 16, 2012
Well, we’re just a few folks shy of 5,000 followers on Google+. See the right column for today’s tally under Google Goodies. That’s less than 10% of our weekly Nerd Vittles fan club. So what are you waiting for? We can’t promise you one of these but, if you become #5000 to put us in your Google+ circles, we do want to hear from you! Please include your mailing address.
Need help with Asterisk? Visit the NEW PBX in a Flash Forum.
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New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
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