[Freeswitch-users] setInputCallback not working with Javascript?

2009-03-24 Thread mszlazak
I'm getting in build 12653M: 

[ERR] notify.js:130 mod_spidermonkey()? TypeError: session.setInputCallback is 
not a function

The wiki says this function should work in Javascript.

http://wiki.freeswitch.org/wiki/CoreSession_Constructor#session:setInputCallback

Also, has there been changes to session.collectInput with type=event? I get 
dtmf type events with my callback function but can't seem to get type=event 
with speech events.

Mark.
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[Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error

2009-03-24 Thread Gilles
Brian West  We changed this to autodetect it means you have the libs 
but no devel headers.. either disable it or install the dev headers.

Thanks guys, that was it: The stock SLED 10 that I have on this MSI 
host did have unixODBC.rpm but not its unixODBC-devel.rpm counterpart.

I re-read the Download  Installation Guide page, which doesn't say 
that Freeswitch will compile for ODBC instead of SQLite if it detects 
unixODBC... but doesn't check that its counterpart unixODBC-devel is 
also installed :-/

I suggest that either the config/make script be updated to check for 
this, or the wiki be edited to tell users to check for both packages 
(if N.A. : yast install unixODBC-devel OR rpm -UVH 
unixODBC-devel.rpm) or configure Freeswitch to use SQLite instead of 
ODBC (good enough to get newbies started).

Cheers,


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Re: [Freeswitch-users] Sip for Skype - g.729 requirement

2009-03-24 Thread David Sugar
They require one use g.729, which is patent encumbered as well as rather
computationally intensive.

Dan wrote:
 You probably already saw this but
 
 http://www.skypeforsip.com/
 
 Skype is supporting sip for business users.
 
 
 
 
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begin:vcard
fn:David Sugar
n:Sugar;David
org:GNU Telephony
email;internet:dy...@gnutelephony.org
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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Saeed Ahmed
Did you started FS with -nc option?
with this option you can connect to FS using ./fs_cli
OR 
use screen!

  _  

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Will
Smith
Sent: Tuesday, March 24, 2009 1:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Console Window

 


Hi,

I closed the console window that ran FS without shutting down FS. I had to
restart the server to get back to the console window. I just wonder if there
is a way to bring that window up without restart server. In the file
/log/freeswitch.pid, I found a number, is that a seesion id?

 

Thank you.

 

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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Will Smith
No, I started FS with this:
/usr/local/freeswitch/bin/freeswitch . What can I do in this case ?
 
thank you for help

--- On Tue, 3/24/09, Saeed Ahmed saeedahmad1...@gmail.com wrote:

From: Saeed Ahmed saeedahmad1...@gmail.com
Subject: Re: [Freeswitch-users] Console Window
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, March 24, 2009, 6:25 AM









Did you started FS with –nc option?
with this option you can connect to FS using ./fs_cli
OR 
use screen!




From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Will Smith
Sent: Tuesday, March 24, 2009 1:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Console Window
 





Hi,

I closed the console window that ran FS without shutting down FS. I had to 
restart the server to get back to the console window. I just wonder if there is 
a way to bring that window up without restart server. In the file 
/log/freeswitch.pid, I found a number, is that a seesion id?

 

Thank you.
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[Freeswitch-users] sip cancel request fails

2009-03-24 Thread Steven Ward
A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while
the call is still ringing does not work.

Why is this request resulting in a 481?

I appreciate the help - I'm still just starting to learn SIP  FS.  The
CANCEL request and 481 response appear as follows on my FS console:


recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
   
   CANCEL 
sip:70...@b-pbx-lab-1.mynet.netsip%3a70...@b-pbx-lab-1.mynet.netSIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
   From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
;tag=as7f6965ea
   To: sip:70...@b-lab-1.mynet.net sip%3a70...@b-lab-1.mynet.net
   Call-ID: 237598fd102b739a03b4a4047bf69...@10.1.21.44
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0

   
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
   
   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
   From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
;tag=as7f6965ea
   To: sip:70...@b-lab-1.mynet.net sip%3a70...@b-lab-1.mynet.net
;tag=71m745HKHKyjc
   Call-ID: 237598fd102b739a03b4a4047bf69...@10.1.21.44
   CSeq: 103 CANCEL
   Content-Length: 0
   --



Thanks.  - SW
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[Freeswitch-users] sip cancel request fails

2009-03-24 Thread Steven Ward
A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while
the call is still ringing does not work.

Why is this request resulting in a 481?

I appreciate the help - I'm still just starting to learn SIP  FS.  The
CANCEL request and 481 response appear as follows on my FS console:


recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
   
   CANCEL 
sip:70...@b-pbx-lab-1.mynet.netsip%3a70...@b-pbx-lab-1.mynet.netSIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
   From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
;tag=as7f6965ea
   To: sip:70...@b-lab-1.mynet.net sip%3a70...@b-lab-1.mynet.net
   Call-ID: 237598fd102b739a03b4a4047bf69...@10.1.21.44
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0

   
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
   
   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
   From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
;tag=as7f6965ea
   To: sip:70...@b-lab-1.mynet.net sip%3a70...@b-lab-1.mynet.net
;tag=71m745HKHKyjc
   Call-ID: 237598fd102b739a03b4a4047bf69...@10.1.21.44
   CSeq: 103 CANCEL
   Content-Length: 0
   --



Thanks.  - SW
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Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Steven Ward
Here it is:

freeswi...@b-pbx-lab-1 recv 517 bytes from udp/[10.1.21.44]:5060 at
13:53:07.644865:
   
   OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
   From: Unknown sip:unkn...@10.1.21.44 sip%3aunkn...@10.1.21.44
;tag=as5adee8f4
   To: sip:b-pbx-lab-1.mynet.net
   Contact: sip:unkn...@10.1.21.44 sip%3aunkn...@10.1.21.44
   Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
   CSeq: 102 OPTIONS
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:07 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Length: 0
   
send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:
   
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
   From: Unknown sip:unkn...@10.1.21.44 sip%3aunkn...@10.1.21.44
;tag=as5adee8f4
   To: sip:b-pbx-lab-1.mynet.net;tag=DytraHp3K84aD
   Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
   CSeq: 102 OPTIONS
   Contact: sip:10.1.21.45
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:
   
   INVITE 
sip:70...@b-pbx-lab-1.mynet.netsip%3a70...@b-pbx-lab-1.mynet.netSIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
;tag=as4863e49a
   To: sip:70...@b-pbx-lab-1.mynet.net sip%3a70...@b-pbx-lab-1.mynet.net
   Contact: sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:11 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 258
   v=0
   o=root 4756 4756 IN IP4 10.1.21.44
   s=session
   c=IN IP4 10.1.21.44
   t=0 0
   m=audio 17956 RTP/AVP 0 8 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv
   
send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:
   
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
;tag=as4863e49a
   To: sip:70...@b-pbx-lab-1.mynet.net sip%3a70...@b-pbx-lab-1.mynet.net
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Content-Length: 0
   
send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:
   
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
;tag=as4863e49a
   To: sip:70...@b-pbx-lab-1.mynet.net sip%3a70...@b-pbx-lab-1.mynet.net
;tag=e7KHcc76gHUXr
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm=10.1.21.44,
nonce=1d23f0ec-187b-11de-8c60-ad87768304bc, algorithm=MD5, qop=auth
   Content-Length: 0
   
recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103:
   
   ACK sip:70...@b-pbx-lab-1.mynet.net
sip%3a70...@b-pbx-lab-1.mynet.netSIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
;tag=as4863e49a
   To: sip:70...@b-pbx-lab-1.mynet.net 

Re: [Freeswitch-users] Console Window

2009-03-24 Thread Diego Viola
I put /usr/local/freeswitch/bin in my path, like this.

export PATH=/usr/local/freeswitch/bin:$PATH on my ~/.bashrc, you can
put it on your /etc/profile if you want it to be global.

Then I just do `freeswitch -nc' when I need to start it, `fs_cli' to
connect, and `fsctl shutdown asap' to shut it down, or `freeswitch
-stop'.

Hope that helps.

Diego

2009/3/24 Will Smith willbel...@yahoo.com:
 Thank you all for your help.
 I use : /usr/local/freeswitch/bin/fs_cli  to open a FS instance. But then
 cannot use shutdown command. Then I used 'fsctl shutdown asap'  (directed
 by Szymon). It works perfectly.

 Again, thank you. Have a great day you all.

 Will

 --- On Tue, 3/24/09, Leon de Rooij l...@scarlet-internet.nl wrote:

 From: Leon de Rooij l...@scarlet-internet.nl
 Subject: Re: [Freeswitch-users] Console Window
 To: freeswitch-users@lists.freeswitch.org
 Date: Tuesday, March 24, 2009, 6:47 AM

 Just make sure you have mod_event socket loaded in
 conf/autoload_modules/modules.conf.xml :
     load module=mod_event_socket/
 And have it configured in conf/autoload_modules/event_socket.conf.xml
 Then you can use bin/fs_cli to connect to running FS instance.
 I don't think you can reconnect to a process of which you have disconnected
 the terminal (without using screen)..
 regards,
 Leon
 On Mar 24, 2009, at 2:35 PM, Will Smith wrote:

 No, I started FS with this:
 /usr/local/freeswitch/bin/freeswitch . What can I do in this case ?

 thank you for help

 --- On Tue, 3/24/09, Saeed Ahmed saeedahmad1...@gmail.com wrote:

 From: Saeed Ahmed saeedahmad1...@gmail.com
 Subject: Re: [Freeswitch-users] Console Window
 To: freeswitch-us...@lists.freeswitch.org
 Date: Tuesday, March 24, 2009, 6:25 AM

 Did you started FS with –nc option?
 with this option you can connect to FS using ./fs_cli
 OR
 use screen!

 

 From: freeswitch-users-boun...@lists.freeswitch.org
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] on Behalf Of Will
 Smith
 Sent: Tuesday, March 24, 2009 1:52 PM
 To: freeswitch-us...@lists.freeswitch.org
 Subject: [Freeswitch-users] Console Window



 Hi,

 I closed the console window that ran FS without shutting down FS. I had to
 restart the server to get back to the console window. I just wonder if there
 is a way to bring that window up without restart server. In the file
 /log/freeswitch.pid, I found a number, is that a seesion id?



 Thank you.



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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Szymon Olko
Will Smith pisze:
 Thank you all for your help.
 I use : /usr/local/freeswitch/bin/fs_cli  to open a FS instance. But
 then cannot use shutdown command. Then I used 'fsctl shutdown asap' 
 (directed by Szymon). It works perfectly.
  
 Again, thank you. Have a great day you all.
  
Remeber there are other flags to shutdown freeswitch

freeswi...@vertux fsctl
API CALL [fsctl()] output:
-USAGE: [send_sighup|hupall|pause|resume|shutdown
[cancel|elegant|asap|restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration
 [num]|loglevel [level]]

Look for descriptions in wiki.

 Will
 
 --- On *Tue, 3/24/09, Leon de Rooij /l...@scarlet-internet.nl/* wrote:
 
 From: Leon de Rooij l...@scarlet-internet.nl
 Subject: Re: [Freeswitch-users] Console Window
 To: freeswitch-users@lists.freeswitch.org
 Date: Tuesday, March 24, 2009, 6:47 AM
 
 Just make sure you have mod_event socket loaded in
 conf/autoload_modules/modules.conf.xml :
 
 load module=mod_event_socket/
 
 And have it configured in conf/autoload_modules/event_socket.conf.xml
 
 Then you can use bin/fs_cli to connect to running FS instance.
 
 I don't think you can reconnect to a process of which you have
 disconnected the terminal (without using screen)..
 
 regards,
 
 Leon 
 
 On Mar 24, 2009, at 2:35 PM, Will Smith wrote:
 
 No, I started FS with this:
 /usr/local/freeswitch/bin/freeswitch . What can I do in this case ?
  
 thank you for help

 --- On *Tue, 3/24/09, Saeed Ahmed /saeedahmad1...@gmail.com
 mailto:saeedahmad1...@gmail.com/* wrote:

 From: Saeed Ahmed saeedahmad1...@gmail.com
 mailto:saeedahmad1...@gmail.com
 Subject: Re: [Freeswitch-users] Console Window
 To: freeswitch-users@lists.freeswitch.org
 mailto:freeswitch-users@lists.freeswitch.org
 Date: Tuesday, March 24, 2009, 6:25 AM

 Did you started FS with –nc option?
 with this option you can connect to FS using ./fs_cli
 OR 
 use /screen/!

 
 

 *From:* freeswitch-users-boun...@lists.freeswitch.org
 mailto:freeswitch-users-boun...@lists.freeswitch.org
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] *On
 Behalf Of *Will Smith
 *Sent:* Tuesday, March 24, 2009 1:52 PM
 *To:* freeswitch-users@lists.freeswitch.org
 mailto:freeswitch-users@lists.freeswitch.org
 *Subject:* [Freeswitch-users] Console Window

  

 Hi,

 I closed the console window that ran FS without shutting down
 FS. I had to restart the server to get back to the console
 window. I just wonder if there is a way to bring that window
 up without restart server. In the file /log/freeswitch.pid, I
 found a number, is that a seesion id?

  

 Thank you.

  

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 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Even André Fiskvik

Please see documentation at:
http://wiki.freeswitch.org/wiki/Fs_cli
and
http://wiki.freeswitch.org/wiki/Mod_commands


Best regards,
Even André

On 24. mars. 2009, at 15.23, Will Smith wrote:


Thank you all for your help.
I use : /usr/local/freeswitch/bin/fs_cli  to open a FS instance. But  
then cannot use shutdown command. Then I used 'fsctl shutdown  
asap'  (directed by Szymon). It works perfectly.


Again, thank you. Have a great day you all.

Will

--- On Tue, 3/24/09, Leon de Rooij l...@scarlet-internet.nl wrote:
From: Leon de Rooij l...@scarlet-internet.nl
Subject: Re: [Freeswitch-users] Console Window
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, March 24, 2009, 6:47 AM

Just make sure you have mod_event socket loaded in conf/ 
autoload_modules/modules.conf.xml :


load module=mod_event_socket/

And have it configured in conf/autoload_modules/event_socket.conf.xml

Then you can use bin/fs_cli to connect to running FS instance.

I don't think you can reconnect to a process of which you have  
disconnected the terminal (without using screen)..


regards,

Leon

On Mar 24, 2009, at 2:35 PM, Will Smith wrote:


No, I started FS with this:
/usr/local/freeswitch/bin/freeswitch . What can I do in this case ?

thank you for help

--- On Tue, 3/24/09, Saeed Ahmed saeedahmad1...@gmail.com wrote:
From: Saeed Ahmed saeedahmad1...@gmail.com
Subject: Re: [Freeswitch-users] Console Window
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, March 24, 2009, 6:25 AM

Did you started FS with –nc option?
with this option you can connect to FS using ./fs_cli
OR
use screen!

From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org 
] On Behalf Of Will Smith

Sent: Tuesday, March 24, 2009 1:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Console Window


Hi,

I closed the console window that ran FS without shutting down FS. I  
had to restart the server to get back to the console window. I just  
wonder if there is a way to bring that window up without restart  
server. In the file /log/freeswitch.pid, I found a number, is that  
a seesion id?




Thank you.


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Re: [Freeswitch-users] Sip for Skype - g.729 requirement

2009-03-24 Thread Derrick Edwards
 It seems to me that they are just offering SIP trunking like
voicepulse. Very interested to see what there rates would be like since
they have a large infrastructure. 



On Tue, 2009-03-24 at 08:54 -0400, David Sugar wrote:
 They require one use g.729, which is patent encumbered as well as rather
 computationally intensive.
 
 Dan wrote:
  You probably already saw this but
  
  http://www.skypeforsip.com/
  
  Skype is supporting sip for business users.
  
  
  
  
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Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Michael Jerris
I note that its missing the to tag from the 180 sent 5 seconds earlier  
(I think thats okay) but the via branch tag is also different, which  
seems wrong.  Can anyone else chime in, I can't recall the dialog  
matching rules of early dialog like this.


Mike

On Mar 24, 2009, at 9:57 AM, Steven Ward wrote:


Here it is:

freeswi...@b-pbx-lab-1 recv 517 bytes from udp/[10.1.21.44]:5060 at  
13:53:07.644865:



   OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
   From: Unknown sip:unkn...@10.1.21.44;tag=as5adee8f4
   To: sip:b-pbx-lab-1.mynet.net
   Contact: sip:unkn...@10.1.21.44
   Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
   CSeq: 102 OPTIONS
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:07 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Length: 0



send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:



   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
   From: Unknown sip:unkn...@10.1.21.44;tag=as5adee8f4
   To: sip:b-pbx-lab-1.mynet.net;tag=DytraHp3K84aD
   Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
   CSeq: 102 OPTIONS
   Contact: sip:10.1.21.45
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH

   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include- 
session-description, presence.winfo, message-summary, refer

   Content-Length: 0



recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:



   INVITE sip:70...@b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: Steve sip:70...@10.1.21.44;tag=as4863e49a
   To: sip:70...@b-pbx-lab-1.mynet.net
   Contact: sip:70...@10.1.21.44
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:11 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 258
   v=0
   o=root 4756 4756 IN IP4 10.1.21.44
   s=session
   c=IN IP4 10.1.21.44
   t=0 0
   m=audio 17956 RTP/AVP 0 8 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv



send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:



   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: Steve sip:70...@10.1.21.44;tag=as4863e49a
   To: sip:70...@b-pbx-lab-1.mynet.net
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Content-Length: 0



send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:



   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: Steve sip:70...@10.1.21.44;tag=as4863e49a
   To: sip:70...@b-pbx-lab-1.mynet.net;tag=e7KHcc76gHUXr
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH

   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include- 
session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm=10.1.21.44,  
nonce=1d23f0ec-187b-11de-8c60-ad87768304bc, algorithm=MD5,  
qop=auth

   Content-Length: 0



recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103:



   ACK sip:70...@b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: Steve sip:70...@10.1.21.44;tag=as4863e49a
   To: 

[Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user

2009-03-24 Thread Leon de Rooij
Hi,

I'm trying to get some cli commands working in combination with xml- 
curl.

Endpoints are parsed properly for SIP registrations and invites, but  
when I use the CLI command user_exists it returns false, while I do  
return an  endpoint (same syntax as for a sofia_reg_parse_auth event)  
on the webserver.

Should switch_xml_locate_user event receive a different syntax ?

freeswi...@internal user_exists accountcode gigaset toyos.nl
false

XML returned from webserver:

?xml version=1.0 encoding=UTF-8?
document type=freeswitch/xml
section name=directory
   domain name=toyos.nl
 user id=gigaset mailbox=
   params
 param name=a1-hash  
value=a211336e29c4f3756e6af343ce6da27b/
 param name=email-addr value=t...@test.com/
 param name=vm-delete-file value=true/
 param name=vm-attach-file value=true/
   /params
   variables
 variable name=accountcode value=gigaset/
 variable name=caller_id_name value=gigaset/
 variable name=caller_id_number value=1000/
 variable name=email_addr value=t...@test.com/
 variable name=privacy value=no/
 variable name=process_cdr value=true/
 variable name=sip_secure_media value=false/
 variable name=country_code value=31/
 variable name=area_code value=320/
   /variables
 /user
   /domain
/section
/document

thanks  regards,

Leon


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Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Steven Ward
Mike,

Thanks for taking the time to look at this - I appreciate it.  I'll go ahead
and test it out on the current svn trunk.

- Steve

2009/3/24 Michael Jerris m...@jerris.com

 This appears to be a bug in FreeSWITCH.  Can you please test this on
 current svn trunk and if it is still a problem, please report this as a bug
 to http://jira.freeswitch.org.
 MIke

  On Mar 24, 2009, at 10:54 AM, Michael Jerris wrote:

  I note that its missing the to tag from the 180 sent 5 seconds earlier (I
 think thats okay) but the via branch tag is also different, which seems
 wrong.  Can anyone else chime in, I can't recall the dialog matching rules
 of early dialog like this.
 Mike

  On Mar 24, 2009, at 9:57 AM, Steven Ward wrote:

  Here it is:

 freeswi...@b-pbx-lab-1 recv 517 bytes from udp/[10.1.21.44]:5060 at
 13:53:07.644865:

OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
From: Unknown sip:unkn...@10.1.21.44 sip%3aunkn...@10.1.21.44
 ;tag=as5adee8f4
To: sip:b-pbx-lab-1.mynet.net
Contact: sip:unkn...@10.1.21.44 sip%3aunkn...@10.1.21.44
Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 24 Mar 2009 13:53:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

 send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
From: Unknown sip:unkn...@10.1.21.44 sip%3aunkn...@10.1.21.44
 ;tag=as5adee8f4
To: sip:b-pbx-lab-1.mynet.net;tag=DytraHp3K84aD
Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
CSeq: 102 OPTIONS
Contact: sip:10.1.21.45
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
 NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
 include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

 recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:

INVITE 
 sip:70...@b-pbx-lab-1.mynet.netsip%3a70...@b-pbx-lab-1.mynet.netSIP/2.0
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
 ;tag=as4863e49a
To: sip:70...@b-pbx-lab-1.mynet.netsip%3a70...@b-pbx-lab-1.mynet.net
 
Contact: sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 24 Mar 2009 13:53:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 4756 4756 IN IP4 10.1.21.44
s=session
c=IN IP4 10.1.21.44
t=0 0
m=audio 17956 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

 send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
 ;tag=as4863e49a
To: sip:70...@b-pbx-lab-1.mynet.netsip%3a70...@b-pbx-lab-1.mynet.net
 
Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Content-Length: 0

 send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
From: Steve sip:70...@10.1.21.44 sip%3a70...@10.1.21.44
 ;tag=as4863e49a
To: sip:70...@b-pbx-lab-1.mynet.netsip%3a70...@b-pbx-lab-1.mynet.net
 ;tag=e7KHcc76gHUXr
Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
 NOTIFY, REFER, 

Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Andy Spitzer
Woof!

Appears to be a recently fixed * bug:

0014431: Bad branch parameter value in CANCEL request
http://bugs.digium.com/view.php?id=14431

--Woof!




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[Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread dujinfang
Hi all,

In about a year of playing with FreeSWITCH, I really like it. Not only  
the software is great but also the community. As people keep asking me  
where can find some documentation in Chinese, I told them if they want  
to play in deep they need to know English better. However, the fact  
is, even one can read English without problem, he still prefer find  
some information in his native language first. So, I'd like to start a  
Chinese community hoping FreeSWITCH can get more popular  in China. As  
we saw big telecom companies merged in the last year, I bet VoIP  
technology and application will go more and more faster in the near  
future...

While FS official site has plenty of documentation, obviously we don't  
want to translate word by word.  Then where should we start from? A  
forum? There was a big argue between a forum and mailing list in the  
last few weeks, and finally an English forum and an Italian one is  
out.  While I can find server in China, the major pain is that running  
any kind of BBS in China mainland need to get some kind of permissions  
by the government first.

Any idea, suggestion? Anyone want to help or cooperate about this?

And, if I can get this runing, can I use the FreeSWITCH logo and 
www.freeswitch.org.cn 
  ?

Best regards,
Seven.

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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Michael Collins
 While FS official site has plenty of documentation, obviously we don't
 want to translate word by word.  Then where should we start from? A
 forum?

I would start by finding as many people as possible who are literate
in both English and Chinese and who are willing to help out. Once you
have your group assembled you will have a better idea of what your
goals should be. More people helping will make things easier to
accomplish.

There was a big argue between a forum and mailing list in the
 last few weeks, and finally an English forum and an Italian one is
 out.  While I can find server in China, the major pain is that running
 any kind of BBS in China mainland need to get some kind of permissions
 by the government first.

 Any idea, suggestion? Anyone want to help or cooperate about this?

I am looking into creating a multi-language wiki at
wiki.freeswitch.org. Anyone with experience in setting up multiple
languages with MediaWiki software please contact me. So far we have
people willing to help create documentation in French, Spanish,
Portugese, Russian, and now Chinese. I think we have some Italians out
there as well! (ciao bella)

Please email me off list if you are in a position to assist with the
wiki and languages other than English.


 And, if I can get this runing, can I use the FreeSWITCH logo and 
 www.freeswitch.org.cn
  ?

Please direct questions about the FreeSWITCH logo and domain names to
consult...@freeswitch.org. FreeSWITCH and the logo are trademarks so
it would be best to get permission from Anthony before doing anything.

-MC

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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Mitul Limbani
Hello,

I can provide you with the hosting on our box in US let me know.

Regards,
Mitul Limbani,
Founder  CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/


On 24-Mar-09, at 23:00, dujinfang dujinf...@gmail.com wrote:

 Hi all,

 In about a year of playing with FreeSWITCH, I really like it. Not only
 the software is great but also the community. As people keep asking me
 where can find some documentation in Chinese, I told them if they want
 to play in deep they need to know English better. However, the fact
 is, even one can read English without problem, he still prefer find
 some information in his native language first. So, I'd like to start a
 Chinese community hoping FreeSWITCH can get more popular  in China. As
 we saw big telecom companies merged in the last year, I bet VoIP
 technology and application will go more and more faster in the near
 future...

 While FS official site has plenty of documentation, obviously we don't
 want to translate word by word.  Then where should we start from? A
 forum? There was a big argue between a forum and mailing list in the
 last few weeks, and finally an English forum and an Italian one is
 out.  While I can find server in China, the major pain is that running
 any kind of BBS in China mainland need to get some kind of permissions
 by the government first.

 Any idea, suggestion? Anyone want to help or cooperate about this?

 And, if I can get this runing, can I use the FreeSWITCH logo and 
 www.freeswitch.org.cn
  ?

 Best regards,
 Seven.

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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Brian West
How about we all work together and work on the FreeSWITCH.org  
infrastructure instead of spreading the resources thinner and thinner  
till nobody is doing really much of anything.


We need people to step up and help out with the website, wiki, jira,  
irc, testing and various other things that currently are spread  
thin.  ;)  Please!!!  I'm not asking everyone to step up and code in  
C...


/b

On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote:


Hello,

I can provide you with the hosting on our box in US let me know.

Regards,
Mitul Limbani,
Founder  CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/



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Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Steven Ward
Ah ha!

Thanks for finding that.

I updated my * server and I'm all set.

Many thanks for all the feedback and help.

2009/3/24 Andy Spitzer w...@nortel.com

 Woof!

 Appears to be a recently fixed * bug:

 0014431: Bad branch parameter value in CANCEL request
 http://bugs.digium.com/view.php?id=14431

 --Woof!




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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Mitul Limbani

Brian,

I can help with website, wiki and testing, tell me what's next step  
forward.


Regards,
Mitul Limbani,
Founder  CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/


On 25-Mar-09, at 0:12, Brian West br...@freeswitch.org wrote:

How about we all work together and work on the FreeSWITCH.org  
infrastructure instead of spreading the resources thinner and  
thinner till nobody is doing really much of anything.


We need people to step up and help out with the website, wiki, jira,  
irc, testing and various other things that currently are spread  
thin.  ;)  Please!!!  I'm not asking everyone to step up and code in  
C...


/b

On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote:


Hello,

I can provide you with the hosting on our box in US let me know.

Regards,
Mitul Limbani,
Founder  CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/



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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Michael Collins
2009/3/24 Mitul Limbani mi...@enterux.com:
 Brian,
 I can help with website, wiki and testing, tell me what's next step forward.


Mitul, do you have any experience with MediaWiki?
-MC

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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread dujinfang

On Mar 25, 2009, at 2:03 AM, Michael Collins wrote:

 While FS official site has plenty of documentation, obviously we  
 don't
 want to translate word by word.  Then where should we start from? A
 forum?

 I would start by finding as many people as possible who are literate
 in both English and Chinese and who are willing to help out. Once you
 have your group assembled you will have a better idea of what your
 goals should be. More people helping will make things easier to
 accomplish.

Ppl not only literate in both English and Chinse, but also need to has  
good knowledge of VoIP and FreeSWITCH.


 There was a big argue between a forum and mailing list in the
 last few weeks, and finally an English forum and an Italian one is
 out.  While I can find server in China, the major pain is that  
 running
 any kind of BBS in China mainland need to get some kind of  
 permissions
 by the government first.

 Any idea, suggestion? Anyone want to help or cooperate about this?

 I am looking into creating a multi-language wiki at
 wiki.freeswitch.org. Anyone with experience in setting up multiple
 languages with MediaWiki software please contact me. So far we have
 people willing to help create documentation in French, Spanish,
 Portugese, Russian, and now Chinese. I think we have some Italians out
 there as well! (ciao bella)

 Please email me off list if you are in a position to assist with the
 wiki and languages other than English.


How's the plan of a multi-language wiki?  I think mediaWiKi do support  
multi-language. So, maybe we can start from wiki.freeswitch.org/zh_CN/  
or some structure else. While it is possible to follow the change log  
of the EN wiki to change Chinese version accordingly, it would be
challenge to make sure the consistency. And regardlessly what if  
Chinese version change first?

I'd like to help with this. But I cannot translate all hundreds of  
pages and make it consistency by my self. But I'm pretty sure that we  
would get a group of ppl to do this. So the start maybe either  
translate some key pages and make the consistency, or translate as  
much pages ...

Email me off list on how to start this. :)


 And, if I can get this runing, can I use the FreeSWITCH logo and 
 www.freeswitch.org.cn
  ?

 Please direct questions about the FreeSWITCH logo and domain names to
 consult...@freeswitch.org. FreeSWITCH and the logo are trademarks so
 it would be best to get permission from Anthony before doing anything.

 -MC

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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread dujinfang


On Mar 25, 2009, at 2:42 AM, Brian West wrote:

How about we all work together and work on the FreeSWITCH.org  
infrastructure instead of spreading the resources thinner and  
thinner till nobody is doing really much of anything.




I agree with this and would like to work with the wiki.freeswitch.org  
on the EN and CN language part. However, I don't think we can maintain  
a Chinese version of IRC, seems still we need a Chinese BBS.



We need people to step up and help out with the website, wiki, jira,  
irc, testing and various other things that currently are spread  
thin.  ;)  Please!!!  I'm not asking everyone to step up and code in  
C...




Take me in :)


/b

On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote:


Hello,

I can provide you with the hosting on our box in US let me know.

Regards,
Mitul Limbani,
Founder  CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/



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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Brian West

On Mar 24, 2009, at 7:47 PM, dujinfang wrote:


 On Mar 25, 2009, at 2:42 AM, Brian West wrote:
 How about we all work together and work on the FreeSWITCH.org  
 infrastructure instead of spreading the resources thinner and  
 thinner till nobody is doing really much of anything.


 I agree with this and would like to work with the  
 wiki.freeswitch.org on the EN and CN language part. However, I don't  
 think we can maintain a Chinese version of IRC, seems still we need  
 a Chinese BBS.

Someone tried to do a spanish IRC channel it doesn't have anyone in it  
but me most of the time... hrm.. I'm not trying to discount  
anything here but it seems language specific anything seems to not  
take off.

We need more cheerleaders btw cluecon is coming up remember to  
register early! ;)


 We need people to step up and help out with the website, wiki,  
 jira, irc, testing and various other things that currently are  
 spread thin.  ;)  Please!!!  I'm not asking everyone to step up and  
 code in C...


 Take me in :)

What do you want to be involved in?  #freeswitch-web is the correct  
channel to be in if you want to help out on IRC.  ;)



 /b


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[Freeswitch-users] ANNOUNCEMENT: ClueCon 2009 - Register Today!

2009-03-24 Thread Michael Collins
The FreeSWITCH Team is pleased to announce that all may register for
ClueCon 2009 immediately!

ClueCon is the Telephony Developers Conference by developers, for
developers. This year's event will be held at the beautiful Wyndham
Hotel in Chicago, August 4-6. Special room rates have been secured and
you will enjoy staying at this luxurious hotel. The conference
attendee price is only $499 per person, but hurry! This early bird
special price is valid only until June 30th.

The schedule is still being finalized, however you can expect to hear
from notable names in the world of telephony. You will also have
opportunities to spend one-on-one time with developers, vendors, and
other attendees. One special highlight for this year is the great
MacBook Pro giveaway! On Thursday at the end of the conference a 15
MacBook Pro will be given away to one lucky attendee. This laptop will
be laser engraved with the logos of all of the sponsors for ClueCon
2009 and will be used as the presenters' laptop for the entire
conference.

Start making your travel plans right away. Visit
http://www.cluecon.com and sign up today. We look forward to seeing
you in Chicago this August!

Michael S Collins
877-742-CLUE

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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Mitul Limbani
MC,

I don't have much exp with mediawiki but have some with drupal.

Regards,
Mitul Limbani,
Founder  CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/


On 25-Mar-09, at 0:35, Michael Collins m...@freeswitch.org wrote:

 2009/3/24 Mitul Limbani mi...@enterux.com:
 Brian,
 I can help with website, wiki and testing, tell me what's next step  
 forward.


 Mitul, do you have any experience with MediaWiki?
 -MC

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