Displaying 20 results from an estimated 3000 matches similar to: "OpenSky: Digium Skype gateway?"
2009 Apr 02
1
Friday April 3rd Gizmo, OpenSky, Skype for Asterisk, SIP for Skype - where are they?
Hi All,
At the usual time, 12 Noon ET on Friday April 3rd, we expect Michael
Robertson to join the discussion to filed questions about OpenSky and
Gizmo5. I have been testing all of these Skype to X methods except SIP
for Skype since I have no word from them. I can tell you that we've
had good results with bith Skype for Asterisk and OpenSky.
In fact, I am currently accepting calls to my
2009 Feb 17
0
Questions about OpenSky - Asterisk to Skype Gateway
>> On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote:
>>
>> > Hi there,
>> >
>> > is gizmo the first user of the Digium Skype solution, or do they use a
>> > different approach/product - any clue?
>> >
>> > http://www.gizmo5.com/pc/opensky/
>> >
>> > Philipp
OpenSky is no related to any product from Digium.
2009 Mar 24
2
Ebay's SIP for Skype
> Anyone connected up to it yet?
>
> http://www.skypeforsip.com/
This service is vaporware. It's just surveyware at this point with no actual
service. An alternative is OpenSky which is a launched service which does
SIP to Skype and Skype to SIP so you can answer and make all your Skype
calls from any SIP aware device. There's a comparison chart at:
http://sipforskype.com and
2009 Feb 15
1
Gizmo SIP / Skype gateway
Anyone got any thoughts on this and how it compares to the chan_skype
that's due soon ?
"OpenSky is a free service provided by Gizmo5 which allows *any* mobile
phone, web browser or IP aware phone network (SIP, asterisk, etc) to
communicate with Skype users. OpenSky supports sending text messages and
voice calls."
http://www.gizmo5.com/pc/opensky/
Julian
2009 Mar 25
1
More on SIP for Skype
Daniel wrote:
For us, opensky can be OK for individual users, not for allowing
Asterisk users to call Skype users. Why? Simply that when you buy the 20
USD connection to Skype and don't want your calls to be cutted after 5
mn, you have to use the Gizmo Skype aliases system which is in your
account. Not really helpful if you want to connect transparently your
users to Skype! They better had to
2009 Feb 17
0
Optimizing this script for calling Skype users from Asterisk
I have written this configuration script which uses OpenSky to make Skype
calls directly from Asterisk devices using my companies SIP to Skype
gateway. Users can dial skype_anyskypeusername or manually add names or
extensions which can get mapped to the correct dialing sequence. The right
sequence is username at opensky.gizmo5.com but that gets mapped to sipphone
address so I set that up to map
2009 Mar 25
1
Skype TO SIP (Was SIP to Skype)
From: "Guillermo Salas M." <gsalas at manta.telconet.net>
> http://www.gizmo5.com/opensky Free calls are available up to 5
> minutes. If you need longer calls there's a commercial service you can
> purchase.
> Can be used to receive calls from skype?
Yes it can. For example anyone who calls me now on Skype at michaelGizmo5 it
will ring the IP phone connected to
2009 Mar 27
0
SIP for Skype Solutions: Hosted v Non-hosted
2009/3/27 Marco Sambo <derwidtel at gmail.com>
> I have to try Skip2PBX, integrated into my Asterisk machine, but it seem
> more invasive than Gizmo5 opensky. Doesn't it?
Gizmo5.com/opensky is a hosted solution SIP to Skype solution meaning
there's no software to install on your system. In minutes the system can be
working for your Asterisk box. This is like using
2009 Mar 19
0
Can I tell if a call picked up on PSTN extension... for example?
Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-)
My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL).
Calls come in and are
2009 Jul 31
0
Friday July 31 @ 12 Noon EDT: Talkshoe former CEO Dave Nelsen, Skype for Asterisk open beta, Gizmo Voice+Google Voice
Hi,
Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a
lot of experience in the telecommunications space and he joins us
today to chat about its current state, conferencing and whatever else
comes to mind. So we have a meta conference aout conferencing, it
won't be the first time :)
You probably saw John Todd's message on one of the lists: Skype for
Asterisk is in open
2009 Jul 31
0
Friday July 31st at 12 Noon EDT: Dave Nelsen, Skype for Asterisk beta opens, Gizmo Voice + Google Voice = free SIP calls
Hi,
Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a
lot of experience in the telecommunications space and he joins us
today to chat about its current state, conferencing and whatever else
comes to mind. So we have a meta conference aout conferencing, it
won't be the first time :)
You probably saw John Todd's message on one of the lists: Skype for
Asterisk is in open
2009 Aug 05
1
Gizmo Dial Out No CALLED PARTY AUDIO??
Hi all,
I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a
while and it works fine .... I just added CALL OUT ... I have no problem
with call setup ... the called party hears me ... but I can't hear them ....
again if the call comes INTO the server both sides work fine.
Just looking for some tips at where I should be looking .... firewall port
forwarding ....
2009 Aug 05
0
Asterisk with gizmo5 and google voice only takes one call at a time.
my problem is this. I have google forward the call to gizmo5. I have this line in my sip file :
register => user:password at proxy01.sipphone.com
I believe this lines connects asterisk with gizmo5 so when it gets a call from Google, asterisk will answer it?
At the end of my sip file i have this
[Calls-From-Gizmo-Network]
type=user
context=demo
disallow=all
allow=ulaw
allow=ilbc
allow=gsm
2006 Mar 15
2
Help with Gizmo from outside firewall
I've beaten myself bloody dealing with this one... No luck so far. In
summary, incoming calls from Gizmo establish, but neither get nor send
sound. Outbound calls to Gizmo work fine (well a bit choppy but work)
My thought is that the SIP connection is being made fine, but the RTP
is getting stopped / blocked / misdone somewhere.
Here is the thing:
Asterisk 2.5 on Linux
(No hardware
2009 Jul 19
1
Re: skype
I have been using GiZMO for about 6 months now - love it!. You do have to pay for some services, but its nice to have the knowledge that you are not being listened to or your private info is not given out. The only reason that Skype is worth having is because Opera uses it. Now, are you Opera's B-tch?
2006 Nov 05
1
skype and SIP hardware for linux
I'm looking at the <http://support.a-link.com/phonemate/IPU1.htm> phone
because it works with Skype (from Linux), but can do SIP, too.
Not necessarily asterisk related, but possibly. My networking situation
might require IAX if I'm running Linux and want to use SIP, I'm not
certain (Skype works fine). Putting that unknown aside for the moment, how
does this phone work under
2007 Mar 22
1
Gizmo project answers every call - can I use it in hunt group?
Hi,
I've set up a Gizmo Project account for access on my Nokia E61 because
they work through NAT. Trouble is If I include my gizmo account in an
asterisk hunt group and I'm not connected (phone is off / outside
wireless coverage) the gizmo project always answers. Either the call
goes to voice mail or if I turn voicemail off the call gets answered
by a recording saying I'm not
2006 Oct 06
3
regexten & regcontext broken for SIP?
Hi ho,
is there anyone out here that is making use of the regcontext and
regexten settings in sip.conf? I've tried this on two Asterisk boxes
(1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1
being created upon SIP client registration, "show dialplan xxx" reveals
no change.
And yes, I have also read and checked bug 7144; if I go down that route
and no
2006 Nov 13
1
Dial : Executing context/priority after bridge?
Hi,
I am using Asterisk to set up a reminder-like system, with asterisk
auto-dialing a user via SIP and playing a reminder file when the user picks
the phone. I use Gizmo service for SIP and I'm able to call through it.
However, when asterisk dials a number, Gizmo first answers then tries
bridging 2 channels. Right after answer Asterisk starts playing the
reminder.
It obviously results in
2006 Dec 13
1
Phone routing - curious what others are doing?
I just went through an exercise of writing a Perl script called from my
Asterisk dialplan to look at a list of area codes and exchanges to
determine which ones are local (no or little cost) under my current
Verizon plan. I route calls outside of my local limits to Gizmo. It works
fine but when I called Verizon to change (lower) my service it was a
bewildering spider web of rates structures just in