On 09/24/2010 06:06 AM, Roger E. Rustad, Jr. wrote:
Allow me to explain...
I currently have one Google Voice phone number (+1 949 xxx xxxx) that
routes to several other phone numbers (Skype In +1 949 area code, +1
951 iPhone, +1 949 home, etc).
Does google voice host your voicemail if you don't answer? Or does it go
to one of the back ends?
In countries that I do significant amount of business in, I will pick
up a number for relatively cheap on Skype (small cost considering it
lets the NOC guys easily call me on their cell phones without picking
up a charge).
Right. So now this is a direct dial to Skype vs going through google voice.
With your own SIP server you could pickup a local DID (using something
like http://www.inphonex.com/availability/) and have it routed to your
VOIP server. Combined with presence it would pick the best device to
reach you on. Presumably you would do this before departure and
provisioning would be completed by the time you are wheels down.
A few problems, however...
--Google Voice does not have inbound numbers for non-US country. So,
I use Skype In to order a +55 11 xxxx-yyyy number here in Brazil (same
as Mexico, +52 11 xxxx yyyy), and then have it forward to various
phones when I am away using Skype's control panel.
Using local SIM cards I presume?
--While in non-US countries, Google Voice does not always let me
respond to a texts (unless I log on the web). When I try to do so, it
says that there is not enough money on my Google Voice account (even
though there is like $100 USD credit). (Presumably because it looks
like a weird number without the proper the IDD, NDD, etc digits?)
Hmmm. Interesting. Several of the SIP apps let you text message. So that
might be a way around it. Have you tried google voice mobile app?
--Calls to my Skype In Brazilian phone number do *not* always transfer
properly to my iPhone number (yet calls to Google Voice transfer okay
most of the time).
Hmmm. Interesting. So the call flow would be <caller> -> skype in ->
skype out -> <roger>?
--Calls to the US sometimes (from my iPhone) sometimes require the GSM
standard dialing (e.g. +1 949 xxx xxxx), othertimes it doesn't take
the +1. I cannot figure out why this is the case. I suspect it is
because of weird issues between bouncing around the local carriers
(Oi, Claro, TIM, etc).
Probably. I imagine dial plan recipes exist to handle those cases, or at
the very least automatically retry a few different ways until it works.
--Countries that I am working more and more with do not have Skype In
phone numbers (e.g. Kenya).
For this I suppose you could just have your PBX route the call to a
legacy PSTN number on a local SIM. That could get expensive though.
--Google Voice does not allow inbound faxes (I have to use MyFax for that)
Yes faxing is always a bit complicated. I haven't found any good
solutions yet but am looking.
--The Skype In number for Brazil (+55 11 xxxx yyyy) seems to only
forward to my Brazilian cell phone on (Samsung Galaxy on Claro's
network, +55 11 xxxx yyyy). I can't figure out why it won't just call
my iPhone with AT&T.
carrier blocking?
Ideally, I need...
--One US number to ring all my phones (even my foreign cell phones).
This should be doable with freeswitch and a few different trunk providers.
--Ring all my cell phones (including American one) when I get a call
from the local number I set up in that country
Yeah. Trunk the local SIP DID to your PBX and have it hit your devices.
--Receive texts to all my phones.
This might be a bit difficult. The SIP apps on Android do have texting
abilities. I haven't played too much with them yet.
--Be able to turn up a local phone number very quickly (e.g. African
country that Skype In does not service).
Usually provisioning happens near instantly.
--Faxes sent to my one phone number magically get routed to my fax.
A real fax machine? Or a virtual fax machine that outputs to e-mail?
I'm willing to roll my own solution here, as I see all of the hosted
solutions as only doing part of what I need....
Yeah you will almost certainly have to.
Any suggestions?
Yeah. Setup freeswitch and terminate a couple trunks on it. See how the
latency is. Then build out from there.
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