On Thursday 01 December 2011, Hans Witvliet wrote:
On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote:
On Thursday 01 December 2011, gincantalupo wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
1. Migrate your Skype users over
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On 12/01/2011 08:30 AM, gincantalupo wrote:
any idea about how to replace Skype For Asterisk?
Replace with what?
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Thursday 01 December 2011, gincantalupo wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
1. Migrate your Skype users over to a better product which supports proper
open standards.
2. Write to your elected representatives asking that they order
On Thu, Dec 1, 2011 at 8:30 AM, gincantalupo
gincantal...@fgasoftware.com wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
We are going through this right now and have chosen to Pay The Man
via per channel subscription to Skype Connect.
Watch the fun
On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote:
On Thursday 01 December 2011, gincantalupo wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
1. Migrate your Skype users over to a better product which supports proper
open standards.
Hi Alex,
replace with anything which could make Asterisk connect to Skype
network, make and receive calls, etc...the usual stuff.
Giorgio
On 12/01/2011 02:40 PM, Alex Balashov wrote:
On 12/01/2011 08:30 AM, gincantalupo wrote:
any idea about how to replace Skype For Asterisk?
Replace
Dear Abdul Basit,
http://nerdvittles.com/index.php?p=784 works, I tested it few months back
and it works. Cant say if its still working or not.
On Wed, Nov 16, 2011 at 5:01 PM, Abdul Basit basit.e...@gmail.com wrote:
Any has Skype For Asterisk (SFA) license.
Any has Skype For Asterisk (SFA) license.
http://www.digium.com/en/products/software/skypeforasterisk.php
PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for
Asterisk will be supported for two more years, until July 26, 2013.
I want to test this thing. Any Idea. any free
Yes, Skype was a good thing. R.I.P
On Wed, Nov 16, 2011 at 5:01 PM, Abdul Basit basit.e...@gmail.com wrote:
Any has Skype For Asterisk (SFA) license.
http://www.digium.com/en/products/software/skypeforasterisk.php
PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for
I can tell you that siptosis is workable but the support has been dropped
recently as well.
It is a great program and especially the paid version with trunk builder
i.e. you can have multiple skype instances
On Wed, Nov 16, 2011 at 8:01 PM, Abdul Basit basit.e...@gmail.com wrote:
Any has Skype
On Wednesday 16 November 2011, Abdul Basit wrote:
Any has Skype For Asterisk (SFA) license.
http://www.digium.com/en/products/software/skypeforasterisk.php
PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for
Asterisk will be supported for two more years, until July
On Wed, 16 Nov 2011, A J Stiles wrote:
You would be better off persuading Skype users to transition to something else.
Skype is the absolute antithesis of the whole point of telephony, which is to
connect people together. This includes, implicitly, the ability for
subscribers on one
Sent: Wednesday, November 16, 2011 11:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Skype For Asterisk (SFA)
On Wed, 16 Nov 2011, A J Stiles wrote:
You would be better off persuading Skype users to transition to something
else.
Skype
On 11/16/2011 10:44 AM, Gordon Henderson wrote:
The other thing - LAN to LAN calls STAY ON THE LAN! So I can Skype my
wife next door and it doesn't use up any of my own broadband bandwidth
wheras if I use a hosted SIP service, calls go out come back in again.
Skype also seems to be able to run
On Wed, Nov 16, 2011 at 8:56 AM, Kevin P. Fleming kpflem...@digium.comwrote:
As SIP endpoints (servers, phones, etc.) get upgraded to support the ICE
NAT traversal mechanism, this will start happening for regular SIP calls as
well. This *should* already happen with the Blink softphone, for
On Wednesday 16 November 2011, Gordon Henderson wrote:
On Wed, 16 Nov 2011, A J Stiles wrote:
You would be better off persuading Skype users to transition to something
else.
Sadly, my experience in the SOHO environment is that Skype wins.
[stuff deleted]
And now I'm seeing some of my
On 07/11/2011 09:48 PM, d tbsky wrote:
1. SFA can not be registered after 26 July. so I want to prepare a
backup machine for our server. I read in the document that I can
re-register my SFA once. so I want to make sure if I can re-register with
my backup server now, and in the same time my
On 07/12/2011 08:26 AM, Kevin P. Fleming wrote:
It is unknown whether it will continue to be usable after that period;
Skype has the ability to disable SFA from accessing the Skype network
if they feel that is what they want to do. Since it won't get any
updates between now and then, it is
On Tuesday 12 Jul 2011, d tbsky wrote:
hi:
I am a SFA (skype for asterisk) user. I had ask Digium questions
about SFA usage in the future. but they seem too busy to reply. so I
tried at this list. I hope there are SFA users or Digium people can
solve my confusion.
Poor you!
To my mind,
hi:
thanks for all the information. I don't use skype and I ban skype
at our network. but there are some people who use skype and want us to
use skype to contact them. SFA is my saver because our users can use
their phone to talk with skype users and no need to install any skype
software.
I
hi:
I am a SFA (skype for asterisk) user. I had ask Digium questions
about SFA usage in the future. but they seem too busy to reply. so I
tried at this list. I hope there are SFA users or Digium people can
solve my confusion.
1. SFA can not be registered after 26 July. so I want to prepare a
Will you consider alternatives such as siptosis? The uncertainties are really
there for SFA
CK Lee
On 12 Jul, 2011, at 10:48 AM, d tbsky tbs...@gmail.com wrote:
hi:
I am a SFA (skype for asterisk) user. I had ask Digium questions
about SFA usage in the future. but they seem too busy to
On Wednesday 25 May 2011, randulo wrote:
On Tue, May 24, 2011 at 10:50 PM, Matt Darnell mattdarn...@gmail.com
wrote:
We expect that users of Skype for Asterisk will be able to continue
using their Asterisk systems on the Skype network until at least July
26, 2013. Skype may extend
On Wed, May 25, 2011 at 10:53 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
Bgeh. Serves 'em right for using that POC! Who honestly *hadn't* seen this
coming since the day Skype was first released?
Tim Panton, who's beenworking with SfA since it came out, posted this
article today:
On Tue, May 24, 2011 at 10:50 PM, Matt Darnell mattdarn...@gmail.com wrote:
We expect that users of Skype for Asterisk will be able to continue
using their Asterisk systems on the Skype network until at least July
26, 2013. Skype may extend this at their discretion.
It's widely believed
On 07/18/2010 11:56 AM, Vieri wrote:
I still don't see why one should pay for a channel when using a PBX but not
when using a client such as Skype. OK, I know that the Skype network is
proprietary and I have to accept whatever they say.
Usage of the standard Skype client is not free; it
On 07/18/2010 12:18 PM, Steve Kennedy wrote:
On Sun, Jul 18, 2010 at 09:56:30AM -0700, Vieri wrote:
As I said above, once you have purchased your SIP channel
you can make
free calls to your PBX using the special number but it's
only INBOUND
AFAIK.
[lots snipped]
With Skype's just
I have been using SiSky Enterprise Edition to integrate Skype with asterisk.
You can even call saved skype users from your asterisk system, by creating
speed dials in SiSky. Unfortunately it is not a free product but it is very
reasonable.
Thank you,
Brad Finberg
- Original Message
--- On Mon, 7/19/10, Kevin P. Fleming kpflem...@digium.com wrote:
Usage of the standard Skype client is not free; it
involves acting as
part of the peer-to-peer Skype network
The Skype
business solutions (including Skype For Asterisk) don't
participate in
the peer-to-peer network
Any
Hi,
I'm trying to integrate Skype and Asterisk but I'm only interested in these 2
things:
1) allow any Asterisk SIP extension to call any Skype user. I do not need to
call landlines via Skype.
2) allow Internet Skype users to call my Asterisk PBX Skype user and route
the call to a specific
not supported in the Skype for SIP docs.
2) allow Internet Skype users to call my Asterisk PBX Skype user and
route the call to a specific Asterisk SIP extension.
Here is how it goes from my experience with Skype: each SIP channel
will cost you about $5 a month, regardless if you have
this is _explicitly_ not supported in the Skype for
SIP docs.
2) allow Internet Skype users to call my Asterisk
PBX Skype user and route the call to a specific Asterisk
SIP extension.
Here is how it goes from my experience with Skype: each SIP
channel
will cost you about $5 a month, regardless if you
On Sun, Jul 18, 2010 at 09:56:30AM -0700, Vieri wrote:
As I said above, once you have purchased your SIP channel
you can make
free calls to your PBX using the special number but it's
only INBOUND
AFAIK.
[lots snipped]
With Skype's just released SkypeKit it should be possible to build
Hi!
I understand that SfA is a binary module? There are processors it will not
work on, correct? Are there limits as to operating system or distros?
Requires kernel 2.6, does not work on 2.4 - as I had to find out the hard
way (this is not documented - so embedded people: be aware!).
Philipp
I understand that SfA is a binary module? There are processors it will
not work on, correct? Are there limits as to operating system or
distros?
tia,
/r
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Can Skype for Asterisk process instant messages from Skype users?
I'm wondering if they can be forwarded via email or SMS.
TIA and regards to all
Enrique Mora
Context M.I.S. SL
em...@context.es
Skype: context-m.i.s.
[cid:image001.jpg@01CAF2BF.0F497C70]
inline: image001.jpg--
Is there something strange about using regular expressions in the context
to which incoming Skype calls go?
If I set up accounts, foobar1, foobar2, etc, it doesn't seem to work to
have:
exten = _foobarX,1,...
should it?
--
On Wed, 30 Dec 2009 11:43:59 -0800, vijay.go...@alliance-infotech.com
wrote:
case 2: This skype account (rexesbposolutions) has been assigned with a
online virtual number (00 44 20 ). If somebody dial this number
from their landline/cellphone, call is transfered to Asterisk queue but
Dear,
there is a problem in codec translation..so change the ulaw codec to
g729. .if problem persist then u must have same codex on asterisk server and
clients (skype)...
On Mon, Jan 4, 2010 at 11:24 AM, Tim Panton t...@westhawk.co.uk wrote:
On 30 Dec 2009, at 19:43,
On 30 Dec 2009, at 19:43, vijay.go...@alliance-infotech.com wrote:
Hi Sir,
We have integrated Skype with Asterisk (skype user id:- rexesbposolutions).
Each call which is coming to skype account is getting transfered to Asterisk
Queue. It has following two cases:
case 1: When we call
Hi Sir,
We have integrated Skype with Asterisk (skype user id:-
rexesbposolutions). Each call which is coming to skype account is
getting transfered to Asterisk Queue. It has following two cases:
case 1: When we call from normal skype account to skype account
(rexesbposolutions), everything is
Hi list,
To make outgoing calls by skype i would like to have our crm app create
callfiles like we do for normal calls.
If i read the instructions it says this :
---quote---
The syntax for making an outgoing call using Skype for Asterisk is as
follows:
Dial(Skype/[originator@]destination)
On 2/09/09 7:45 PM, Remco Barendse wrote:
So i create a callfile that looks like this:
---
Channel: SIP/228
MaxRetries: 0
Dial(Skype/asterisk...@somebodyonskype)
Priority: 1
Callerid: Somebodyonskypesomebodyonskype
You're combining technologies there :)
You can do:
Channel
Context
On Wed, 2 Sep 2009, Matt Riddell wrote:
On 2/09/09 7:45 PM, Remco Barendse wrote:
So i create a callfile that looks like this:
---
Channel: SIP/228
MaxRetries: 0
Dial(Skype/asterisk...@somebodyonskype)
Priority: 1
Callerid: Somebodyonskypesomebodyonskype
You're combining technologies
On Wed, 19 Aug 2009, Terry Wilson wrote:
I haven't seen (or heard of) it happening. Please post a bug report
on http://betareports.digium.com/mantis/ with a backtrace from one of
the core dumps along with the relevant information about your setup,
dialplan, chan_skype.conf, etc. If there
On Wed, 19 Aug 2009, Terry Wilson wrote:
Have you posted a bug describing the issues you are having at
http://betareports.digium.com/mantis/
yet? I would love to have the opportunity to actually fix any bugs
that people find. :-)
I installed the 1.0 release of Skype for Asterisk and last
On Aug 20, 2009, at 3:41 AM, Remco Barendse wrote:
I never used Skype myself but i installed it to try and i noticed
that i
got added by lots of strange skype users (spam bots?), i guess some of
those were trying some funny stuff on my skype for asterisk account. I
want to use Skype for
Nope - but you are also running on an unsupported version of asterisk,
so I am not surprised. From the readme:
===[ Installation Overview ]===
It is required that the proper version of Asterisk is installed prior to
installing Skype For Asterisk.
Oops sorry, the Asterisk version should read 1.4.26.1
On Wed, 19 Aug 2009, Julian Lyndon-Smith wrote:
Nope - but you are also running on an unsupported version of asterisk,
so I am not surprised. From the readme:
===[ Installation Overview ]===
Julian Lyndon-Smith wrote:
Nope - but you are also running on an unsupported version of asterisk,
so I am not surprised. From the readme:
===[ Installation Overview
]===
It is required that the proper version of Asterisk is installed prior
a few days ago slashdot (sorry i havent the link now) wrote about skype has
a very huge problem whit a licence in a core codec, and if they dont get an
aregment whit the codec owner they will close the doors...
David
2009/8/19 Thomas Kenyon dig...@sanguinarius.co.uk
Julian Lyndon-Smith wrote:
I wonder if that was not a codec specific issue, but rather the matter
of their license to the p2p technology provided by JoltID? Since Skype
has recently dveloped their own codec (SILK) they could easily drop
support for any codec that they previously licensed from outside. I
think that the
Michael Graves wrote:
I wonder if that was not a codec specific issue, but rather the matter
of their license to the p2p technology provided by JoltID? Since Skype
has recently dveloped their own codec (SILK) they could easily drop
support for any codec that they previously licensed from
Have you posted a bug describing the issues you are having at
http://betareports.digium.com/mantis/
yet? I would love to have the opportunity to actually fix any bugs
that people find. :-)
I installed the 1.0 release of Skype for Asterisk and last night on my
production box running
On Mon, 17 Aug 2009, Pascal Bruno wrote:
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
http://store.digium.com/productview.php?product_code=1SFA0001
Yes, pretty pricey indeed especially considering that you can buy Skype
ATA
On Tuesday, August 18, 2009, Remco Barendse wrote:
But then again, who needs Skype for business purposes anyways, i
don't think there is a huge market for it.
Me ... at least in theory! Our cellphones have built-in Skype, so a
Skype gateway should give me call forwarding and diversion to our
On Tue, 18 Aug 2009, Geoff Lane wrote:
On Tuesday, August 18, 2009, Remco Barendse wrote:
But then again, who needs Skype for business purposes anyways, i
don't think there is a huge market for it.
Me ... at least in theory! Our cellphones have built-in Skype, so a
Skype gateway should
On Tuesday, August 18, 2009, Gordon Henderson wrote:
I was under the impression that Three (who I guess you're using)
placed a regular call over their network then Skyped it at their
HQ - rather than have the Skype client actually reside in the
handset.. (And I'm suspecting their 3G
Good luck with the N95... my experiences of the N95 and SIP haven't been
great... the phone likes to restart... regularly. Nokia may well have fixed
these glitches by now though. Getting it configured was a bit of a mission
too... and as expected the battery life shoots down when it's enabled...
On Tue, 18 Aug 2009, Geraint Lee wrote:
Good luck with the N95... my experiences of the N95 and SIP haven't been
great... the phone likes to restart... regularly. Nokia may well have fixed
these glitches by now though. Getting it configured was a bit of a mission
too... and as expected the
Geoff Lane wrote:
On Tuesday, August 18, 2009, Gordon Henderson wrote:
I was under the impression that Three (who I guess you're using)
placed a regular call over their network then Skyped it at their
HQ - rather than have the Skype client actually reside in the
handset.. (And I'm
I would have happily bought 20 channels at $10/channel, but at most will
be buying only a single channel now :\
Pascal Bruno wrote:
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
Pricing is a very legitimate way to minimise support effort. It winnows
down the market size to a point where the company offering the goods
can sustain the projected per user support issues.
You can always drop the price later on when you have a better handle on
the per user support issue.
Casey Boone wrote:
I would have happily bought 20 channels at $10/channel, but at most will
be buying only a single channel now :\
That does sound a bit pricey, although it it's as stable as the latest
beta, I wont be buying it at all.
___
--
Michael Graves wrote:
Pricing is a very legitimate way to minimise support effort. It winnows
down the market size to a point where the company offering the goods
can sustain the projected per user support issues.
You can always drop the price later on when you have a better handle on
the
Lol but he has a good point and makes a lot of sense. Never thought about
that strategy...
On Tue, Aug 18, 2009 at 12:16 PM, Thomas Kenyon
dig...@sanguinarius.co.ukwrote:
Michael Graves wrote:
Pricing is a very legitimate way to minimise support effort. It winnows
down the market size to
I just want to also remind people that Skype for SIP is also to be released
shortly. When I last talked to Skype they said it would be out in late
July. So I assume if you wait another few more weeks the entire issue will
be moot. No $60/channel fee, just the free SIP platform for people using
Snipped from:
http://store.digium.com/productview.php?product_code=1SFA0001
Supports G.711 and G.729 (included) codec's
Do I understand correctly that my SFA-to-Skype calls would be throttled down
to 8khz even if not traversing the PSTN?
My biggest draw to SFA is 440+ million subscribers in a
Karl Fife wrote:
To some degree, I (and I'm sure other wideband disciples), feel somewhat
like the only guy on the block with a fax machine. Per Metcalfe's law it
seemed that Skype (as a easily accessible namespace supporting wideband)
could have been a shot in the arm for 8khz telephony.
That does sound a bit pricey, although it it's as stable as the latest
beta, I wont be buying it at all.
Have you posted a bug describing the issues you are having at
http://betareports.digium.com/mantis/
yet? I would love to have the opportunity to actually fix any bugs
that people find.
- Original Message -
From: Kevin P. Fleming kpflem...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 18, 2009 5:03 PM
Subject: Re: [asterisk-users] Skype for Asterisk -- Codec support
Karl Fife wrote
I just want to also remind people that Skype for SIP is also to be
released shortly. When I last talked to Skype they said it would be
out in late July. So I assume if you wait another few more weeks
the entire issue will be moot. No $60/channel fee, just the free
SIP platform for
Karl Fife wrote:
Any idea what timeframe?
Not that I can disclose, no. Sorry.
Can I assume that SFA licenses now will be valid for future releases?
Yes. It is quite unlikely (although not impossible, of course, like any
piece of software) that existing SFA licenses would not be valid for
On Tue, 18 Aug 2009, Terry Wilson wrote:
That does sound a bit pricey, although it it's as stable as the latest
beta, I wont be buying it at all.
Have you posted a bug describing the issues you are having at
http://betareports.digium.com/mantis/
yet? I would love to have the opportunity
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
http://store.digium.com/productview.php?product_code=1SFA0001
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon
I am using the beta and its pretty good for remote access for clients
It would help if they had some discount structure for volume
Cheers Duncan
Pascal Bruno wrote:
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
On Thu, Jul 30, 2009 at 8:50 PM, John Todd jt...@digium.com wrote:
I know many of you have been waiting for this for a while, so I'll
keep this short: The Skype for Asterisk Public Beta is now available
on the Digium store.
We are pleased to announce the open beta of Skype For Asterisk is
Hi,
This is the error message I get. Any idea where I may find some further
debug logs?
[Aug 3 08:01:23] ERROR[23831] chan_skype.c: Unable to start Skype For
Asterisk library.
Thanks,
Emrah
Tim Panton wrote:
I don't know then. My understanding is that the message is caused by
the wrong
Pascal Bruno wrote:
Well I think thats what the problem was, I dont have it named as eth0.
So if your NIC is not labeled eth0 you cannot use skypeforasterisk???
Why cant it just scan you nic handles? Can someone point me to where I
can change the NIC name in the source file or
Pascal Bruno wrote:
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
Any help would be appreciated
It uses the same licensing scheme as the G.729 licenses (so as soon as
you need to upgrade the machine,
Thomas Kenyon wrote:
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
chef*CLI skype show users
Skype Users
[2009-08-02
Hi Thomas,
I am experiencing the same problem, with the same error messages.
Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686
Regards,
Emrah
Thomas Kenyon wrote:
Thomas Kenyon wrote:
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0
I reported an issue on Mantis (#14).
Waiting for an update.
http://betareports.digium.com/mantis/view.php?id=14
Emrah
Emrah wrote:
Hi Thomas,
I am experiencing the same problem, with the same error messages.
Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686
Regards,
Emrah
Thomas
So what do you think I can do to register my license? I am running
Asterisk 1.6.10 on CentOS 5.
Sent from my iPod
On Aug 2, 2009, at 3:49 AM, Thomas Kenyon dig...@sanguinarius.co.uk
wrote:
Pascal Bruno wrote:
Unfortunately for me, I cannot register my license. Kept saying:
Could not
On Sun, Aug 2, 2009 at 8:24 AM, Pascal Brunotipas...@gmail.com wrote:
So what do you think I can do to register my license? I am running
Asterisk 1.6.10 on CentOS 5.
Could not generate Host-ID.
Make sure that you have eth0 enabled.
The MAC is used in the scheme to register and it looks like
On Sun, Aug 2, 2009 at 12:13 PM, randulo spamsucks2...@gmail.com wrote:
On Sun, Aug 2, 2009 at 8:24 AM, Pascal Brunotipas...@gmail.com wrote:
So what do you think I can do to register my license? I am running
Asterisk 1.6.10 on CentOS 5.
Could not generate Host-ID.
Make sure that you
Well I think thats what the problem was, I dont have it named as eth0. So
if your NIC is not labeled eth0 you cannot use skypeforasterisk??? Why cant
it just scan you nic handles? Can someone point me to where I can change
the NIC name in the source file or something???
On Sun, Aug 2, 2009
I had that too, I cured it by kill -9 'ing the skypeforasterisk
process that was left over from
the previous version of the beta.
Hope that helps.
Tim.
On 2 Aug 2009, at 11:20, Emrah wrote:
I reported an issue on Mantis (#14).
Waiting for an update.
Hi Tim,
I don't have any skypeforasterisk process running. I tried to killall
-9 asterisk but it did not solve my issue.
Any other suggestions?
Thanks for your help,
Emrah
Tim Panton wrote:
I had that too, I cured it by kill -9 'ing the skypeforasterisk
process that was left over from
the
I don't know then. My understanding is that the message is caused by
the wrong skypeforasterisk process running.
- did you (ever) run it as a different user ?
If it is a test box, you could try a full reboot.
Tim.
On 2 Aug 2009, at 19:35, Emrah wrote:
Hi Tim,
I don't have any
Nice job. It worked right away for me with my 10 channel trial license.
Asterisk 1.4.26
I'm already building a dtmf access menu to bridge to my SIP world :-)
As much I hate Skype for being a closed system, it would make the ultimate
remote Asterisk extension as Skype drills through so many
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
Any help would be appreciated
On Sat, Aug 1, 2009 at 9:01 PM, Tom Browning ttbrown...@gmail.com wrote:
Nice job. It worked right away for me with my 10
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
On Sat, Aug 1, 2009 at 9:01 PM, Tom Browning ttbrown...@gmail.com wrote:
Nice job. It worked right away for me with my 10 channel trial license.
Asterisk 1.4.26
I have problems with it...
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1153 skype_cp_handler:
Found license 'XX' providing 1 concurrent calls
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1000 display_host: Skype
For Asterisk Host-ID: X
[Jul 30 14:34:17] NOTICE[30529]:
I know many of you have been waiting for this for a while, so I'll
keep this short: The Skype for Asterisk Public Beta is now available
on the Digium store.
We are pleased to announce the open beta of Skype For Asterisk is
ready to begin and we look forward to you participation. To obtain
I have problems with it...
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1153 skype_cp_handler: Found license
'XX' providing 1 concurrent calls
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1000 display_host: Skype For Asterisk
Host-ID: X
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1320
The first time is always free :)
On Thu, Jul 30, 2009 at 1:50 PM, John Todd jt...@digium.com wrote:
I know many of you have been waiting for this for a while, so I'll
keep this short: The Skype for Asterisk Public Beta is now available
on the Digium store.
We are pleased to announce the
On 7/30/09, Steve Totaro stot...@asteriskhelpdesk.com wrote:
The first time is always free :)
On Thu, Jul 30, 2009 at 1:50 PM, John Todd jt...@digium.com wrote:
I know many of you have been waiting for this for a while, so I'll
keep this short: The Skype for Asterisk Public Beta is now
On 27 Jun 2009, at 10:06, Olivier wrote:
Hi,
As many remember, almost one year this Skype for Asterisk extension
program was announced.
Has anyone tried it ?
Is there any available pricelist ?
I've just had a talk on Skype for Asterisk accepted at Astricon (www.astricon.net
), so
if
1 - 100 of 117 matches
Mail list logo
