I think there would be more compatibility across devices & networks with
a SIP based client than an IAX. Asterisk supports both, although I not
sure of the benefits of using one versus the other.
I previously suggested looking into LinPhone since it is a GTK solution
that is the basis for PhoneGaim & several other solutions. I'm not sure
I like the idea of Mozilla's Zap if it requires all the overhead of
running Mozilla.
Right now I use the Gizmo Project server (without the Gizmo client) for
handling of all my SIP calls. I use GrandCentral to get a local POTS
number (and a number local to my parents), voicemail I can check from
the internet or any phone, and forward calls to both my cellphone & my
SIP phone for free. If someone has a better setup that doesn't require
a land-line & still allows POTS calls, I'm happy to hear it.
I don't see running my own Asterisk server being that much more
beneficial to my needs. My dream is to have a freerunner that would
default to SIP when attached to a wifi network and switch to the GSM
network when wifi is not available.
Meadhbh S. Hamrick wrote:
Hey Kyle...
Yes.. this is a basic "convergence" function. I worked at Divitas
Networks last year trying to make this happen on WinMoblie and Symbian
phones. Sadly, I was unsuccessful in getting them to work on a Linux
Solution, so I had to do all my Linux based VoIP experimentation on my
own time, though I've been working exclusively in the SIP/SRTP
solution space.
There's a lot of flexibility in this kind of solution.
For one, you don't really need a POTS phone number connected to your
VoIP system, you can use it as a complete net-bearer system using IAX
or SIP.
You could then add an IPCSP (IP Communications Service Provider) to
link the POTS phone number to your asterisk box. This is sort of what
Skype does with Skype-In / Skype-Out, but their network is closed and
you have to wait for them to port their client to various platforms :-(.
And if you wanted to have a lot of fun, you could even roam between
VoIP and cell. This is essentially what the carriers do with UMA, and
it's what Divitas did, though uptake on their product has been pretty
slow.
Several folks are in this game.... I've been looking at Mozilla's ZAP
project recently, and "under the hood" on the client I can't imagine
rewriting a media framework given that gStreamer seems to work quite
nicely.
-cheers!
-M.
On Feb 21, 2008, at 9:33 AM, Kyle Bassett wrote:
Hey guys,
I've been contemplating writing an IAX client for OM which would be
capable of the following:
Prerequisites:
-user has dedicated VoIP phone number routed to an Asterisk server[1]
---OR a compatible VoIP provider that supports fallback[2] calling
-user has smartphone+OM with some form of internet access (wifi/bt
internet/ethernet)
---in addition to regular GSM/CDMA service on the smartphone
-[optional] user has regular GSM/CDMA cell phone+service
Usage situation:
The user exchanges the VoIP number with all contacts. When someone
attempts to contact the user, via dialing the VoIP number, the
asterisk server answers the call and checks to see if the user is
available over VoIP. If the user's smartphone is on and connected to
the internet, the OM IAX client should connect to the asterisk server
automatically (depending on the user's settings, etc.) If the phone
is available over VoIP, asterisk attempts to ring the user over VoIP
for a specified time. If the user does not answer or a connection
problem persisted, then the asterisk server can forward the call to
the user's regular (OM or third party) cell line. Asterisk is very
flexible and many permutations of this example can be accomplished,
ie. calling all the numbers at once, and forwarding the call to the
first to pickup.
There are many benefits to this system:
--User has complete control over the call routing and voicemail system
--User can prevent the usage of regular cell airtime by using VoIP as
much as possible
--User can give one phone number to all contacts and have asterisk
decide how to handle the call (routing not just to the cell phones,
but to home lines, etc.)
--During the debugging process with OM+GTA0x, users can carry both
phones and still use just one number throughout the day
----Call comes in->asterisk tries OM[VoIP]-> tries OM[GSM] -> tries
regular third party cell phone (can also ring all numbers at once)
[1] asterisk.org
[2] fallback calling is a service that allows VoIP users to enter a
number (regular landline/cell) as a fallback in case the VoIP call
cannot be established
I have tried to remain as general as possible, that way this post
won't become outdated with specifics to any specific hardware. In
reality, we want OM on as many phones as possible. ;-)
Please provide any feedback or ideas!
-Kyle
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Meadhbh S. Hamrick (It's pronounced "Maeve")
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