Re: [asterisk-users] Asterisk/Skype
Can anyone make it more clear please
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: kche...@xplorium.com
MSN ID :khalidche...@hotmail.com
Web Site: http://www.xplorium.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, February 25, 2011 11:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk/Skype
AFAIK, the issue here is not Skype or Gtalk. The Asterisk client isn't
really designed to easily transport messages during the call or otherwise.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Friday, February 25, 2011 3:14 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk/Skype
I am assuming that goes the same for Gtalk chat messages too?
Or has nobody played with that?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson
Sent: Friday, February 25, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk/Skype
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
The dialplan application is only for receiving chat messages during an
actual call. If you want to receive messages from outside of a call, you
will have to use the manager interface and look for SkyeChatMessage events.
image001.png--
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[asterisk-users] Asterisk/Skype
i installed skype for asterisk i can send and recieve calls normaly how can i receive messages from another skype user i Succeed to send only using for example: exten = 2233,1,SkypeChatSend(fSkypeBcp,User,message text) how to receive messages using this code SKYPE_CHAT_RECEIVE(account,from,timeout),and where and how I should add this code in extensions.conf my chan_Skype.conf [Account] secret=XX context=from-pstn exten= Account disallow=all allow=g729 allow=alaw allow=slin allow=ulaw auth_policy=accept buddy_presence=yes direction=both ;auth_policy=ignore buddy_autoadd=true ;buddy_presence=no mohinterpret=default ;mohsuggest=none Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com image001.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Skype
Maybe something like this? [skype_chat_receieve] Exten = account,user,1,do something here? What do you see in the CLI on the incoming txt message? I just figured out how to handle a different google talk account today [google-in] Exten = us...@gmail.com,1,Dial(SIP/100) Exten = us...@gmail.com,1,Dial(SIP/101) Exten = us...@gmail.com,1,Dial(SIP/102) It doesn't matter the context in gtalk or jingle ,.. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Friday, February 25, 2011 2:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Asterisk/Skype i installed skype for asterisk i can send and recieve calls normaly how can i receive messages from another skype user i Succeed to send only using for example: exten = 2233,1,SkypeChatSend(fSkypeBcp,User,message text) how to receive messages using this code SKYPE_CHAT_RECEIVE(account,from,timeout),and where and how I should add this code in extensions.conf my chan_Skype.conf [Account] secret=XX context=from-pstn exten= Account disallow=all allow=g729 allow=alaw allow=slin allow=ulaw auth_policy=accept buddy_presence=yes direction=both ;auth_policy=ignore buddy_autoadd=true ;buddy_presence=no mohinterpret=default ;mohsuggest=none Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com image001.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Skype
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
any idea
regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: kche...@xplorium.com
MSN ID :khalidche...@hotmail.com
Web Site: http://www.xplorium.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Friday, February 25, 2011 9:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk/Skype
Maybe something like this?
[skype_chat_receieve]
Exten = account,user,1,do something here?
What do you see in the CLI on the incoming txt message?
I just figured out how to handle a different google talk account today
[google-in]
Exten = us...@gmail.com,1,Dial(SIP/100)
Exten = us...@gmail.com,1,Dial(SIP/101)
Exten = us...@gmail.com,1,Dial(SIP/102)
It doesn't matter the context in gtalk or jingle ,..
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Friday, February 25, 2011 2:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Asterisk/Skype
i installed skype for asterisk
i can send and recieve calls normaly
how can i receive messages from another skype user
i Succeed to send only
using for example: exten = 2233,1,SkypeChatSend(fSkypeBcp,User,message
text)
how to receive messages using this code
SKYPE_CHAT_RECEIVE(account,from,timeout),and where and how I should
add this code in extensions.conf
my chan_Skype.conf
[Account]
secret=XX
context=from-pstn
exten= Account
disallow=all
allow=g729
allow=alaw
allow=slin
allow=ulaw
auth_policy=accept
buddy_presence=yes
direction=both
;auth_policy=ignore
buddy_autoadd=true
;buddy_presence=no
mohinterpret=default
;mohsuggest=none
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: kche...@xplorium.com
MSN ID :khalidche...@hotmail.com
Web Site: http://www.xplorium.com
image001.png--
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Re: [asterisk-users] Asterisk/Skype
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
The dialplan application is only for receiving chat messages during an actual
call. If you want to receive messages from outside of a call, you will have to
use the manager interface and look for SkyeChatMessage events.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Asterisk/Skype
Can you please send me a how to please or a simple lines?
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: mailto:kche...@xplorium.com kche...@xplorium.com
MSN ID :khalidche...@hotmail.com
Web Site: http://www.xplorium.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson
Sent: Friday, February 25, 2011 10:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk/Skype
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
The dialplan application is only for receiving chat messages during an
actual call. If you want to receive messages from outside of a call, you
will have to use the manager interface and look for SkyeChatMessage events.
image001.png--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] Asterisk/Skype
I am assuming that goes the same for Gtalk chat messages too?
Or has nobody played with that?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson
Sent: Friday, February 25, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk/Skype
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
The dialplan application is only for receiving chat messages during an
actual call. If you want to receive messages from outside of a call, you
will have to use the manager interface and look for SkyeChatMessage events.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Skype
AFAIK, the issue here is not Skype or Gtalk. The Asterisk client isn't
really designed to easily transport messages during the call or otherwise.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Friday, February 25, 2011 3:14 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk/Skype
I am assuming that goes the same for Gtalk chat messages too?
Or has nobody played with that?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson
Sent: Friday, February 25, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk/Skype
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
The dialplan application is only for receiving chat messages during an
actual call. If you want to receive messages from outside of a call, you
will have to use the manager interface and look for SkyeChatMessage events.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] Asterisk/Skype
Can you please send me a how to please or a simple lines? Regards Please see the README file that came with skypeforaterisk. Search for SkypeChatMessage. As far as AMI tutorial, please see Asterisk: The Definitive Guide chapter 20 (and consider ordering a copy). http://ofps.oreilly.com/titles/9780596517342/ch20.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + Skype deployment
Just FYI Really, nothing to do with me... http://www.thevarguy.com/2009/10/01/systems-integrator-dials-skype-for-asterisk/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Skype update
Enjoyed the podcast :) Does anyone have any idea what the pricing structure will be for this? are we talking $10/channel? $100/channel? Does this log into the Skype network as multiple users? One global user for the business as a whole? Do I have to have 1 user login per inbound channel? What I am hoping to be able to do with this is allow for 10-15 simultaneous inbound from Skype calls, no interest at first for receiving nor making PSTN calls via Skype. Casey Boone Tim Panton wrote: On 23 Feb 2009, at 15:13, Dean Collins wrote: Asterisk/Skype update available here - http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/ …. It’s definitely an update that updates absolutely nothing J, more news at 11 :P John Todd and I discussed this at some length on the VoIP user conference on friday (I'm on a jittery hotel wifi so a bit garbled.) http://recordings.talkshoe.com/TC-22622/TS-198841.mp3 Also briefly covered in my blog on ecomm : http://tinyurl.com/b60-ecomm Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Skype update
Enjoyed the podcast :) Does anyone have any idea what the pricing structure will be for this? are we talking $10/channel? $100/channel? Does this log into the Skype network as multiple users? One global user for the business as a whole? Do I have to have 1 user login per inbound channel? What I am hoping to be able to do with this is allow for 10-15 simultaneous inbound from Skype calls, no interest at first for receiving nor making PSTN calls via Skype. Casey Boone Tim Panton wrote: On 23 Feb 2009, at 15:13, Dean Collins wrote: Asterisk/Skype update available here - http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/ …. It’s definitely an update that updates absolutely nothing J, more news at 11 :P John Todd and I discussed this at some length on the VoIP user conference on friday (I'm on a jittery hotel wifi so a bit garbled.) http://recordings.talkshoe.com/TC-22622/TS-198841.mp3 Also briefly covered in my blog on ecomm : http://tinyurl.com/b60-ecomm Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Skype update
On 23 Feb 2009, at 15:13, Dean Collins wrote: Asterisk/Skype update available here - http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/ …. It’s definitely an update that updates absolutely nothing J, more news at 11 :P John Todd and I discussed this at some length on the VoIP user conference on friday (I'm on a jittery hotel wifi so a bit garbled.) http://recordings.talkshoe.com/TC-22622/TS-198841.mp3 Also briefly covered in my blog on ecomm : http://tinyurl.com/b60-ecomm Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk/Skype update
Asterisk/Skype update available here - http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/ It's definitely an update that updates absolutely nothing :-), more news at 11 :P Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Tue, 20 Dec 2005, AR Tarzi wrote: could you please tell how it interfaces with Asterisk? Could I receive calls into Asterisk? send calls out? I've just downloaded it and am searching (unsuccessfully) for these on Gizmo's site/software. Gizmo isn't just a soft phone. Like Skype, its a service. Unlike Skype, though, the service is open to the rest of the SIP world. So - to call your Asterisk system from Gizmo, simply tell Gizmo to dial [EMAIL PROTECTED] To call Gizmo from Asterisk, simply tell it to dial SIP/[EMAIL PROTECTED] Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Mon, Dec 19, 2005 at 02:56:30PM -0800, Kerry Garrison wrote: Yes you can send and receive calls via Asterisk. http://voipspeak.net/index.php?/content/view/19/28/ Which demostrates how to connect to sipphone.com . This is very simple, indeed. But what about text chats with gizmo-project users? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
http://www.gizmoproject.com/ from the website, it quite looks like skype - no network setup, IM integration, you may call POTS phones by paying the company who did it. not very useful, in the end, if the purpouse is asterisk-skype interoperability - I doubt that every one of the millions of skype users out there will uninstall it and tell their contacts to switch to gizmo. like it or not, skype is here to stay. just my $0.02 l. On Mon, 19 Dec 2005 23:46:20 +0100, AR Tarzi [EMAIL PROTECTED] wrote: could you please tell how it interfaces with Asterisk? Could I receive calls into Asterisk? send calls out? I've just downloaded it and am searching (unsuccessfully) for these on Gizmo's site/software. - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, December 19, 2005 23:23 Subject: RE: [Asterisk-Users] Asterisk - Skype anywhere/anyhow? Everyone should simply uninstall Skype and switch to the Gizmo project because it interfaces quite nicely with Asterisk. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Tue, Dec 20, 2005 at 12:31:01PM +0100, Lenz wrote: http://www.gizmoproject.com/ from the website, it quite looks like skype - no network setup, IM integration, you may call POTS phones by paying the company who did it. not very useful, in the end, if the purpouse is asterisk-skype interoperability Huh? 1. http://sipphone.com/numbers/ : there is inter-operability with other VoIP networks. 2. At least for voice calls, you can register your asterisk to sipphone.com: http://sipphone.com/gettingstarted/existing.html#1 (Rapid users: ast-cmd add-trunk supports the type sipphone). - I doubt that every one of the millions of skype users out there will uninstall it and tell their contacts to switch to gizmo. like it or not, skype is here to stay. And this is bad for us. With Gizmo we can talk. With google talk we have stand a chance of talking. But we're blocked from Skype. So the future of Skype is a future of disconnected networks rather than interconnected networks. In that sense it is bad. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Tue, Dec 20, 2005 at 01:41:30PM +0200, Tzafrir Cohen wrote: ... And this is bad for us. With Gizmo we can talk. With google talk we have stand a chance of talking. But we're blocked from Skype. since you cite it, what compatibility is there with google talk ? any pointer to descriptions of the protocols used ? thanks luigi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Tue, Dec 20, 2005 at 03:50:27AM -0800, Luigi Rizzo wrote: On Tue, Dec 20, 2005 at 01:41:30PM +0200, Tzafrir Cohen wrote: ... And this is bad for us. With Gizmo we can talk. With google talk we have stand a chance of talking. But we're blocked from Skype. since you cite it, what compatibility is there with google talk ? any pointer to descriptions of the protocols used ? Google have released a set of libraries for GoogleTalk, libjingle available from their site, with example code ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Tue, Dec 20, 2005 at 03:50:27AM -0800, Luigi Rizzo wrote: On Tue, Dec 20, 2005 at 01:41:30PM +0200, Tzafrir Cohen wrote: ... And this is bad for us. With Gizmo we can talk. With google talk we have stand a chance of talking. But we're blocked from Skype. since you cite it, what compatibility is there with google talk ? any pointer to descriptions of the protocols used ? Google talk is jabber based and they intend to support SIP... Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Tuesday, 20 December 2005 06:06 Mark Hulber wrote: The paper is definitely interesting and I commend them for their effort but it doesn't represent a complete understanding of the Skype protocol to the extent that an Asterisk server could speak the Skype protocol. They say that much of the Skype protocol is encrypted and needs to be inferred to this point from the types and locations of messages that are being sent. So despite Skype's popularity they basically have their whole product locked down. It is greatly complex, and it also has a number of stealth elements that do nasty things with accepted norms of network etiquette. The bottom line is: Skype *is* evil, and the Asterisk folks, for the most part, have on the white hats of Open Source. IMO we should steer 1000 miles clear of it. Yah, yah, everyone uses Skype. Well everyone uses Micro$oft, too. That doesn't mean Asterisk should get into bed with them. B. http://www.rsdevs.com/psgw_sip.shtml -- Regards, Hilton Travis Phone: +61 (0)7 3344 3889 (Brisbane, Australia) Phone: +61 (0)419 792 394 Manager, Quark IT http://www.quarkit.com.au Quark Group http://quarkgroup.com.au/ Microsoft Small Business Specialists http://www.threatcode.com/ -- its now time to shame poor coders into writing code that is acceptable for use on today's networks War doesn't determine who is right. War determines who is left. This document and any attachments are for the intended recipient only. It may contain confidential, privileged or copyright material which must not be disclosed or distributed. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
Interesting that Skype cant prevent itself becoming a super-node unlike KaZaa. Wonder what that does to capped ADSL lines in South Africa...RobOn 12/19/05, Paul Hewlett [EMAIL PROTECTED] wrote: On Monday 19 December 2005 14:23, Francesco Peeters (Asterisk) wrote: On Mon, December 19, 2005 11:33, Evert Meulie said: Hi all! I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act as a gateway, but what I'd really like is a for example an Asterisk module that can route calls to Skype, perhaps the same principle as IAX2? I'm assuming more people are interested in this, but... does it exist already? There is no such thing yes, and as Skype is closed source, it'll have to wait until someone reverse-engineers it... (Sniffing the protocol will be hard, as it is - supposedly - encrypted)2 guys (Schulzrinne and Baset] of Columbia University have done it. See www1.cs.columbia.edu/~library/TR-repository/reports/reports-2004/cucs-039-04.pdf--Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.zaTel: +27 21 852 8812Cel: +27 84 420 9282Fax: +27 86 672 0563 --___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - Skype anywhere/anyhow?
Hi all! I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act as a gateway, but what I'd really like is a for example an Asterisk module that can route calls to Skype, perhaps the same principle as IAX2? I'm assuming more people are interested in this, but... does it exist already? Regards, Evert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Mon, December 19, 2005 11:33, Evert Meulie said: Hi all! I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act as a gateway, but what I'd really like is a for example an Asterisk module that can route calls to Skype, perhaps the same principle as IAX2? I'm assuming more people are interested in this, but... does it exist already? There is no such thing yes, and as Skype is closed source, it'll have to wait until someone reverse-engineers it... (Sniffing the protocol will be hard, as it is - supposedly - encrypted) I'd love to connect my (*) to Skype as well, but I do not see it happening soon! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Monday 19 December 2005 14:23, Francesco Peeters (Asterisk) wrote: On Mon, December 19, 2005 11:33, Evert Meulie said: Hi all! I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act as a gateway, but what I'd really like is a for example an Asterisk module that can route calls to Skype, perhaps the same principle as IAX2? I'm assuming more people are interested in this, but... does it exist already? There is no such thing yes, and as Skype is closed source, it'll have to wait until someone reverse-engineers it... (Sniffing the protocol will be hard, as it is - supposedly - encrypted) 2 guys (Schulzrinne and Baset] of Columbia University have done it. See www1.cs.columbia.edu/~library/TR-repository/reports/reports-2004/cucs-039-04.pdf -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
The paper is definitely interesting and I commend them for their effort but it doesn't represent a complete understanding of the Skype protocol to the extent that an Asterisk server could speak the Skype protocol. They say that much of the Skype protocol is encrypted and needs to be inferred to this point from the types and locations of messages that are being sent. MARK. Paul Hewlett wrote: On Monday 19 December 2005 14:23, Francesco Peeters (Asterisk) wrote: On Mon, December 19, 2005 11:33, Evert Meulie said: Hi all! I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act as a gateway, but what I'd really like is a for example an Asterisk module that can route calls to Skype, perhaps the same principle as IAX2? I'm assuming more people are interested in this, but... does it exist already? There is no such thing yes, and as Skype is closed source, it'll have to wait until someone reverse-engineers it... (Sniffing the protocol will be hard, as it is - supposedly - encrypted) 2 guys (Schulzrinne and Baset] of Columbia University have done it. See www1.cs.columbia.edu/~library/TR-repository/reports/reports-2004/cucs-039-04.pdf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
Mark Hulber wrote: The paper is definitely interesting and I commend them for their effort but it doesn't represent a complete understanding of the Skype protocol to the extent that an Asterisk server could speak the Skype protocol. They say that much of the Skype protocol is encrypted and needs to be inferred to this point from the types and locations of messages that are being sent. So despite Skype's popularity they basically have their whole product locked down. It is greatly complex, and it also has a number of stealth elements that do nasty things with accepted norms of network etiquette. The bottom line is: Skype *is* evil, and the Asterisk folks, for the most part, have on the white hats of Open Source. IMO we should steer 1000 miles clear of it. Yah, yah, everyone uses Skype. Well everyone uses Micro$oft, too. That doesn't mean Asterisk should get into bed with them. B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
Everyone should simply uninstall Skype and switch to the Gizmo project because it interfaces quite nicely with Asterisk. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Monday, December 19, 2005 12:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow? Mark Hulber wrote: The paper is definitely interesting and I commend them for their effort but it doesn't represent a complete understanding of the Skype protocol to the extent that an Asterisk server could speak the Skype protocol. They say that much of the Skype protocol is encrypted and needs to be inferred to this point from the types and locations of messages that are being sent. So despite Skype's popularity they basically have their whole product locked down. It is greatly complex, and it also has a number of stealth elements that do nasty things with accepted norms of network etiquette. The bottom line is: Skype *is* evil, and the Asterisk folks, for the most part, have on the white hats of Open Source. IMO we should steer 1000 miles clear of it. Yah, yah, everyone uses Skype. Well everyone uses Micro$oft, too. That doesn't mean Asterisk should get into bed with them. B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
could you please tell how it interfaces with Asterisk? Could I receive calls into Asterisk? send calls out? I've just downloaded it and am searching (unsuccessfully) for these on Gizmo's site/software. - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, December 19, 2005 23:23 Subject: RE: [Asterisk-Users] Asterisk - Skype anywhere/anyhow? Everyone should simply uninstall Skype and switch to the Gizmo project because it interfaces quite nicely with Asterisk. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
Yes you can send and receive calls via Asterisk. http://voipspeak.net/index.php?/content/view/19/28/ Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AR Tarzi Sent: Monday, December 19, 2005 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow? could you please tell how it interfaces with Asterisk? Could I receive calls into Asterisk? send calls out? I've just downloaded it and am searching (unsuccessfully) for these on Gizmo's site/software. - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, December 19, 2005 23:23 Subject: RE: [Asterisk-Users] Asterisk - Skype anywhere/anyhow? Everyone should simply uninstall Skype and switch to the Gizmo project because it interfaces quite nicely with Asterisk. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Mon, Dec 19, 2005 at 02:56:30PM -0800, Kerry Garrison wrote: Yes you can send and receive calls via Asterisk. http://voipspeak.net/index.php?/content/view/19/28/ so let me understand. One nice feature of skype is the excellent (for the user; i understand the sysadmin may see this as a nightmare) ability to circumvent firewalls. Do you know how gizmo works in this respect ? is it plain sip/stun/proxy or does it use any smarter (and maybe documented) trick ? cheers luigi Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
I don't know exactly how it works, but since it appears to just be SIP, I would have to assume a STUN setup. I haven't bothered to sit there and watch the packets go by to see what its doing under the hood. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luigi Rizzo Sent: Monday, December 19, 2005 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow? On Mon, Dec 19, 2005 at 02:56:30PM -0800, Kerry Garrison wrote: Yes you can send and receive calls via Asterisk. http://voipspeak.net/index.php?/content/view/19/28/ so let me understand. One nice feature of skype is the excellent (for the user; i understand the sysadmin may see this as a nightmare) ability to circumvent firewalls. Do you know how gizmo works in this respect ? is it plain sip/stun/proxy or does it use any smarter (and maybe documented) trick ? cheers luigi Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On 14:56, Mon 19 Dec 05, Kerry Garrison wrote: Yes you can send and receive calls via Asterisk. http://voipspeak.net/index.php?/content/view/19/28/ Is there any change you can provide the sip.conf and extensions.conf stuff this generates? I'm not an amp user, nor do I want to use it just to see what it generates for the sip.conf The extensions.conf stuff is optional, I can manage that. Thnx -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Mon, Dec 19, 2005 at 03:18:18PM -0800, Kerry Garrison wrote: I don't know exactly how it works, but since it appears to just be SIP, I would have to assume a STUN setup. I haven't bothered to sit there and watch the packets go by to see what its doing under the hood. thanks - luigi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
I sincerely believe that it's completely non-sense to make a channel for Skype. Skype is a *proprietary* protocol. If they(ebay) don't like the idea of someone messing around their network, they will change the protocol specification, launching a new version, for example, and *all* the work and time spent on this will just going to sink. Probably it is better to loose time with something else. Isamar On Mon, 19 Dec 2005, Luigi Rizzo wrote: On Mon, Dec 19, 2005 at 03:18:18PM -0800, Kerry Garrison wrote: I don't know exactly how it works, but since it appears to just be SIP, I would have to assume a STUN setup. I haven't bothered to sit there and watch the packets go by to see what its doing under the hood. thanks - luigi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Tue, Dec 20, 2005 at 09:54:50AM +0900, [EMAIL PROTECTED] wrote: I sincerely believe that it's completely non-sense to make a channel for Skype. Skype is a *proprietary* protocol. If they(ebay) don't like the idea of someone messing around their network, they will change the protocol specification, launching a new version, for example, and *all* the work and time spent on this will just going to sink. Probably it is better to loose time with something else. I agree. Perhaps put the time into making new cool features so that Skype folks can look at some SIP client and say 'wow - I want that too, let's switch from Skype' ;) The world would be a better place without Skype, without proprietary standards.. Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
I agree. Perhaps put the time into making new cool features so that Skype folks can look at some SIP client and say 'wow - I want that too, let's switch from Skype' ;) The world would be a better place without Skype, without proprietary standards.. Let me disagree with that. The advantage of skype is that it's making millions of people worldwide familiar with VoIP technology and showing them that it can work. IMHO you can use Skype to your advantage when selling Asterisk based VoIP systems. It works like skype, except it's just like your good old telephone! Now it's true that it /would/ be better if Skype was open, but it isn't. Still, I believe the world /is/ a better place with Skype. Without it, VoIP awareness would be much lower and the technology would struggle even more to emerge. Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk = SKYPE
Hi Any solution for connecting Asterisk to Skype without using fsx/fxo hardware ? Best Regards HB Norway ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk = SKYPE
Why and how would you use hardware to integrate with skype? I hope you don't mean making actual phone calls, even then, there would be easier ways to do it. A much better solution, not involvong phone calls, would be to use the Skype API, with D-Bus. I don't know if its available publicly yet, but regardless you can go to the skype forums and request it as a beta tester. Regards, Shidan On Mon, 07 Feb 2005 16:45:53 +0100, HBK [EMAIL PROTECTED] wrote: Hi Any solution for connecting Asterisk to Skype without using fsx/fxo hardware ? Best Regards HB Norway ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
