Sekiur My Thoughts

In just under a minute I migrated a couple of GrandCentral account to Google Voice and I am very exited to see a transcript of a voicemail show up in my Inbox.

I will definitely miss the GrandCentral interface as its much more intuitive than the new Google Voice GUI.

A limitation currently in place on both platforms is the capability to have 2 different accounts ring one same number. I particularly like this to have a personal and a business number both ring my cell and landlines. The workaround for the moment is leaving an account with GrandCentral and on one Google Voice. Lets see how long that lasts.!

One thing that I have seen more and more recently is my GrandCentral dropping calls on me. Maybe its Google’s way of getting users migrated.

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Last week I deployed a PBX in a Flash system using SIPConnect from CBeyond. It was so successful that I will start using PIAF in lieu of Trixbox from now on for all future deployments of this type and will replace my home PBX to take advantage of Skype and Google Voice integration.

In this case I used the Aastra 53i (English edition) VoIP phones which when connected to the network, retrieved an IP from the DHCP server, contacted the PBX using mDNSResponse, checked and downloaded the most recent firmware available on the PBX, and downloaded the default configuration which prompts for a user to login. After login in the phone created a config file on the PBX for future restarts.

These Aastra phones come in 2 editions (The English/American edition and the European edition). The power supply for the European edition has different connectors and the display had symbols instead of words. Apart from that they appeared to be identical but getting the European edition to automatically connect to the PBX and configure itself was very painful, having to reset the phone to factory defaults and erase the local configuration multiple times and finally having to define on the phone the TFTP server (PBX) IP address for it to download the configuration.

Two thumbs up for the PBX in a Flash (PIAF) developers who have done a superb job with this distribution holding up the ideals of the original Asterisk@home open source project.

Their documentation was almost flawless although it was difficult trying to find the most recent version of instructions as they are all layed out in bits and pieces across a blog. In pursuit of a perfect install I narrowed down the install to running the iso install, going through the online download and compilation of asterisk and running the update/fix scripts. Now before upgrading/installing any module or OS updates, I downloaded and installed the files necessary to deploy the Aastra phones which is also done by a script and then I proceeded to install/update the software via the FreePBX module admin and finally the OS updates.

Below is the trunk configuration for connecting via SIPConnect to CBeyond from PBX in a Flash:

Outbound caller ID: 5551231234
Never overrride caller ID: checked
Maximum Channels: 6

Outbound Settings

trunk name=cbeyond

allow=ulaw&alaw&gsm&ilbc&g726&adpcm
context=from-trunk
disallow=all
dtmfmode=auto
fromdomain=sipconnect.dal0.cbeyond.net
host=sipconnect.dal0.cbeyond.net
insecure=very
outboundproxy=sip-proxy.dal0.cbeyond.net
qualify=250
secret=[secret-password]
type=peer
username=5551231234

Regitration String: 5551231234:secret-password@cbeyond/5551231234

Note: Notice there is no inbound settings required. DID incoming configuration will determine were each channel from the trunk will ring.

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