Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
Does the SPA3102 support TLS or only SRTP?
I don't know, but supporting only SRTP would be ridiculous, since the keys
would then be transmitted in the clear and therefore amenable to interception.
SRTP requires the SIP channel to be encrypted
Hi to All,
Any one please tell me , How to configure soft sip phone to freeswitch with
extension number.
--
If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.
Regards
Venkatesan OV.
The only ATA mentioned on the WIKI that supports TLS/SRTP is the
Grandstream HandyTone 503. But, again according to the wiki, that
doesn't seem to behave to well with TLS ...
On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote:
Mark Campbell-Smith mcampbellsm...@gmail.com
Didn't Michael already answer this? Best read the FS wiki and the
softphone user guide for help with this.
http://wiki.freeswitch.org/wiki/Getting_Started_Guide
http://wiki.freeswitch.org/wiki/Interop_List
On Wed, Nov 25, 2009 at 7:29 PM, ovvenkat ovvenkate...@gmail.com wrote:
Hi to All,
Jeff,
thanks very much for picking this up. You quickly spotted my mistake - I had
the bind_meta_data call in the local extensions but not added it to the
group extension (100).
Appreciate you taking the time to have a look and point out my silly
mistake - all working now,
regards
Dave
I tried removing the codec file extension from uuid_record and
session_record but I'm still unable to record a file in native format for a
bridged call.
record WORKS!, but uuid_record and session_record do not want to record in
native format. do uuid_record and session_record work with native
Hello all,
is there a way how to enable very short recordings (1-3 seconds) in
FreeSWITCH other than editing source code and recompiling?
Thanks for your time!
Best regards,
kokoska.rokoska
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FreeSWITCH-users mailing list
These two options attach media bugs on to the session. Which doesn't
work with native files as far as I know.
/b
On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote:
record WORKS!, but uuid_record and session_record do not want to
record in native format. do uuid_record and session_record
Is this standard recording? or voicemail?
/b
On Nov 25, 2009, at 9:11 AM, kokoska rokoska wrote:
Hello all,
is there a way how to enable very short recordings (1-3 seconds) in
FreeSWITCH other than editing source code and recompiling?
Thanks for your time!
Best regards,
Is this for vm? If so set min-record-len on the profile
-Original Message-
From: kokoska rokoska
Sent: 11/25/2009 3:11:05 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] how to enable short recordings
Hello all,
is there a way how to enable very short recordings
hi is it possibe to enable passive recording in sangoma tdm interface
in feeswich. pls advice
Best Regards
G.Imthiyaz Ahmed
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so is using session_record with .wav my best option for recording bridged
calls?
--matt
On Wed, Nov 25, 2009 at 7:18 AM, Brian West br...@freeswitch.org wrote:
These two options attach media bugs on to the session. Which doesn't
work with native files as far as I know.
/b
On Nov 25,
Thank you very much, Brian, for your interest!
It is standard recording:
action application='record_session' data='/path/file.wav' /
Best regards,
kokoska.rokoska
Brian West napsal(a):
Is this standard recording? or voicemail?
/b
On Nov 25, 2009, at 9:11 AM, kokoska rokoska wrote:
Yes
On Wed, Nov 25, 2009 at 9:44 AM, Matthew Fong mattdf...@gmail.com wrote:
so is using session_record with .wav my best option for recording bridged
calls?
--matt
On Wed, Nov 25, 2009 at 7:18 AM, Brian West br...@freeswitch.org wrote:
These two options attach media bugs on to the
Yes, Brian, I need them :-)
They don't contain speech - instead, they contain few computer
generated tones and I should store them in max quality for later
proccessing (i.e. analysis)...
Best regards,
kokoska.rokoska
Brian West napsal(a):
Really you want to keep 1-3 second files around?
Peter,
I had a similar problem, the way I found to make it work was setting the
mailbox ID in the phone to match the FS domain/hostname.
For instance using a Linksys SPA962, I set the Voice Mail Server in
the extension tab to extens...@domain using FS hostname as the domain.
Regards,
Juliano
Hi,
If I am using proxy_media=true, bypass_media=false, is there anyway of
modifying o= and c= so that it won't show the IP of the far-end B leg?
I am using fs as b2b2a and I want to hide the far-end ip as much as
possible.
I got to hide the IP for invite by modifying the sdp within C code, but
You know FreeSWITCH will proxy media already if you turn off
proxy_media and disable transcoding you'll get the same results and
the IP's will be correct. Proxy media is for one purpose... T.38, it
gains you NOTHING otherwise.
/b
On Nov 25, 2009, at 10:10 AM, Juan Backson wrote:
Hi,
Hi All,
goodmorning to all, i have a scenario, two pjsua clients are connected with
Freeswitch and they are in call and bypass_media=true. i close the
Freeswitch server, still they are in call, again i started the Freeswitch,
and registerd these two endpoints, now how can i end the
hi
is it possibe to enable passive recording in sangoma tdm interface
in feeswich. pls advice
Best Regards
G.Imthiyaz Ahmed
--
Best Regards
G.Imthiyaz Ahmed
PeopleTech systems (P) ltd
http://peopletech.co.in
___
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use the variable RECORD_MIN_SEC
This was added in revision 15271 so if you are below that I recommend
updating to latest trunk.
On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska
kokoska.roko...@post.czwrote:
Yes, Brian, I need them :-)
They don't contain speech - instead, they contain few
What do you mean by passive encoding?
On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed imthiy...@gmail.comwrote:
hi
is it possibe to enable passive recording in sangoma tdm interface
in feeswich. pls advice
Best Regards
G.Imthiyaz Ahmed
--
Best Regards
G.Imthiyaz Ahmed
PeopleTech
What do you mean by passive encoding?
On Wed, Nov 25, 2009 at 11:13 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
What do you mean by passive encoding?
On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed imthiy...@gmail.comwrote:
hi
is it possibe to enable passive recording in
We at Swifcore Technologies, a telephony and server management team, would
like your help in reviewing our latest product. We have created a hosting
platform around the FreeSWITCH engine (for obvious reasons of stability and
extensibility) and would like your feedback so we continue to improve
Thank you very much, Anthony, for your help!
I'm nearly at current trunk (15653) and
action application='set' data='RECORD_MIN_SEC=1'/
works great :-)
Many thanks once more!
Best regards,
kokoska.rokoska
Anthony Minessale napsal(a):
use the variable RECORD_MIN_SEC
This was added in
I mean to tap tx and rx of a PRI line using sangoma tap and record
the call information and actual calls without distrubing the existing
line . freeswitch will work in passive mode like trunk side call
recorder.
Thanks
Imthiyaz
On Wed, Nov 25, 2009 at 10:43 PM, Anthony Minessale
http://code.google.com/p/mod-recpld/
It's out-dated. I originally wrote it to record raw G.729 codec on
passthrough mode. It worked before and then we abandoned that since We felt
G729 cannot deliver good sound particularly on a cross-continent network.
The code is written when I don't know much
And you may also would like to update the wiki as well if the var is not
there.
2009/11/26 kokoska rokoska kokoska.roko...@post.cz
Thank you very much, Anthony, for your help!
I'm nearly at current trunk (15653) and
action application='set' data='RECORD_MIN_SEC=1'/
works great :-)
Many
Try an alias on the sip profile.
Mike
On Nov 24, 2009, at 5:56 PM, Peter P GMX wrote:
Anthony, thanks for the hint,
I receive events like the following
RECV EVENT
Event-Name: MESSAGE_WAITING
Core-UUID: e71632c8-d948-11de-942b-0138c6269e37
FreeSWITCH-Hostname: sip11.mydomain.com
FreeSWITCH will kill the calls when you shut it down, if you intentionally kill
the network without shutting down FreeSWITCH the only thing you can do is
enable session timers or rtp timers in the soft phones to kill the call when
FreeSWITCH dies or when the call is over.
Mike
On Nov 25,
Its a feature we don't have, patches welcome.
Mike
On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote:
Hi members,
I’m controlling freeswitch with the conference module via xmlrpc.
Is it desired that the kick command can only kick users that are connected to
the conference?
Is there
In trunk there is a sofia profile setting to allow dialplan processing of 302
responses. This won't get you back into your same javascript, but you can
probably do something clever from there.
Mike
On Nov 24, 2009, at 5:04 PM, John Platts wrote:
I have considered writing JavaScript code
Yes an alias will be required for every domain you run on the profile
so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
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It's possible it does not. I just added some code to set it on auto-adjust so
it might be there sometimes now. You might need to add some code in mod_sofia
to add it other times. Maybe it makes sense to move that var setting down to
switch_rtp.c. Patches for this would be welcome.
Thanks
You can send the call with secure enabled and if it supports it it will use it.
Mike
On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote:
Hello,
We have a mix of phones that support RTP encryption and those that do not.
I have to support both types in the meanwhile, and would like to
HI,
thanks for your reply, my requirement is i am doing failover stuff with
freeswitch. i dont want cut the calls when freeswitch dies, when failover
happens mean one freeswitch dies we are going to start the second
freeswitch, i dont want close call intiated by the first freeswtich, they
are
Surprisingly, I've found no way to access the HTTP response status code
using mod_spidermonkey_curl. I'd love to see this feature added or discussed
if it already exists and I'm missing it.
--Stephen
On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris m...@jerris.com wrote:
In trunk there is a
this is how i do it from the dialplan:
extension name=ServiceLookup
condition field=destination_number
expression=^(300030)(.*)|^\+(300030)(.*)
action application=set data=bPfx=$1$3/
action application=set data=bNum=$2$4/
action inline=true application=set
It was the first I want to do - update wiki :-)
But someone was much faster (00:14, 30 October 2009 :-)
http://wiki.freeswitch.org/wiki/Variable_record_min_sec
Last time I looked for some hint about the recording (few months ago),
this page (and even the variable) didn't exist...
Best regards,
For that you would need to fully exchange session state into the sip library,
something that is not available in that lib at this time.
On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote:
HI,
thanks for your reply, my requirement is i am doing failover stuff with
freeswitch. i dont want
Awesome, thanks Andrew, I will have to keep an eye out for that patch.
To continue, last night I decided to tackle the business hours and holiday
routing on my FreeSWITCH system. It turned out to not be quite as simple
with the XML dialplan as I thought. After being up until 1am banging my head
I've created a patch to override the value of MIN_TIME in the vmd
modules using a channel variable. In this way, it can be configured on a
call by call basis. The channel variable is name vmd_min_time. I
didn't add the other detection parameters, but doing so would be
straight forward. So, in
Stephen, I think you've jumped into the middle of a thread about sip 302,
not about http.
Anyway, you might want to look at using mod_curl instead of
mod_spidermonkey_curl. mod_curl can give you a json response which you can
then parse easily in javascript or any other language. The json
can you submit your patch to jira as an improvement under the application
modules section please.
On Wed, Nov 25, 2009 at 12:36 PM, Andrew Fritz afritzli...@fritztech.comwrote:
I've created a patch to override the value of MIN_TIME in the vmd modules
using a channel variable. In this way, it
My apologies, and thanks for the info.
--Stephen
On Wed, Nov 25, 2009 at 11:06 AM, Rupa Schomaker r...@rupa.com wrote:
Stephen, I think you've jumped into the middle of a thread about sip 302,
not about http.
Anyway, you might want to look at using mod_curl instead of
?
Is there no chance abort an invitation?
The kick command has no effect until the person I invited with the
dial command is connected.
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can please tell me how can i exchange session state into sip library.
Thanks
srinivas
On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris m...@jerris.com wrote:
For that you would need to fully exchange session state into the sip
library, something that is not available in that lib at this time.
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--
Message: 2
Date: Wed, 25 Nov 2009 12:45:50 -0500
From: Michael Jerris m...@jerris.com
Subject: Re: [Freeswitch-users] Handling
Or if you're dancing with the stars!!
/b
On Nov 25, 2009, at 1:55 PM, Chris Chen wrote:
One suggestion to you, please never consider the GXW4108 for any
business use unless just in LAB. The GXW4108 will work when it is
working,but I can tell you within one year you will be regretting
the person I invited with the
dial command is connected.
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--
Message: 2
Date: Wed
That was a *GREAT* e-mail.
On Wed, Nov 25, 2009 at 2:59 PM, Brian West br...@freeswitch.org wrote:
Or if you're dancing with the stars!!
/b
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
something that is not available in that lib at this time.
Mike
On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote:
can please tell me how can i exchange session state into sip library.
Thanks
srinivas
On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris m...@jerris.com wrote:
For that
?
The kick command has no effect until the person I invited with the
dial command is connected.
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Kill it, sunshine.
/b
On Nov 25, 2009, at 2:40 PM, Chris Chen wrote:
You haven't really put it into production for more than one year.
The issue with GXW4108 is that after some time, say a couple of
months, either all FXO ports not working, or worse, some FXO ports
not working, but after
thanks for your reply mike,
is there any api in freeswitch or any thing else to update lib
programatically from pjsua.
srinivas
On Thu, Nov 26, 2009 at 2:05 AM, Michael Jerris m...@jerris.com wrote:
something that is not available in that lib at this time.
Mike
On Nov 25, 2009, at 2:47 PM,
/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html
--
Message: 2
Date: Wed, 25 Nov 2009 12:45:50 -0500
From: Michael Jerris m...@jerris.com
Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
responsefrom
You can read all about the sip library at
http://sofia-sip.sourceforge.net/refdocs/
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 25-Nov-09, at 3:58 PM, srinivasula reddy wrote:
thanks for your reply mike,
is there any
I can spare you the pain and let you know outright that this sort of
functionality will cost somewhere in the range of 125,000.00 to 150,000.00
to properly implement by assembling a team of consultants including members
of the development team from both FreeSWITCH and Sofia-SIP and even if you
The processor power saved is negligible between PCMU and raw PCM and not
worth the fuss.
If you didn't decode the audio first you would not be able to mix the stream
to produce a single file.
So if we went to the trouble of making native media bugs to be able to do
that you could barely use them
command has no effect until the person I invited with the
dial command is connected.
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On Nov 25, 2009, at 5:18 PM, Adam Ford wrote:
Samuel,
FreeSWITCH has a Skype module that uses Skype client instances to connect to
the Skype network, you can read about it at
http://wiki.freeswitch.org/wiki/Skypiax
As far as an official Skype module for non-Asterisk PBX-es, it looks
:
http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/
288d63a0/attachment-0001.htmlhttp://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/%0A288d63a0/attachment-0001.html
--
Message: 2
Date: Wed, 25 Nov 2009 12:45:50
exactly
On Wed, Nov 25, 2009 at 4:38 PM, Michael Jerris m...@jerris.com wrote:
On Nov 25, 2009, at 5:18 PM, Adam Ford wrote:
Samuel,
FreeSWITCH has a Skype module that uses Skype client instances to connect
to
the Skype network, you can read about it at
I added a patch to do it in more places
On Wed, Nov 25, 2009 at 11:47 AM, Michael Jerris m...@jerris.com wrote:
It's possible it does not. I just added some code to set it on auto-adjust
so it might be there sometimes now. You might need to add some code in
mod_sofia to add it other times.
How do I turn on dialplan processing of 302 responses? I can solve my problem
if I can process 302 responses in my dialplan.
From: m...@jerris.com
Date: Wed, 25 Nov 2009 12:45:50 -0500
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
These guys can on E1, not T1. They are not compatible with FS just yet, but
we are working on it.
Let me know off-list if you are interested.
JM
On Wed, Nov 25, 2009 at 3:29 PM, Imthiyaz Ahmed imthiy...@gmail.com wrote:
I mean to tap tx and rx of a PRI line using sangoma tap and record
the
Yeah, that's why I had to record to two files(readwrite) and need to mix
together by using sox. Do you only try to using PCMU to save CPU power
matt? As Anthony said, the difference can be ignored. And you also need to
take extra effort to make sure transcoding will not happen on a
conversation.
Is there a way of determining if a call-command sent to a session via ESL
has completed? Is there a return event which is always fired? Is there a
identifier I can use to verify that the return event matches my command?
Thanks,
Josh
___
FreeSWITCH-users
from
http://svn.freeswitch.org/svn/freeswitch/trunk/conf/sip_profiles/internal.xml
!-- Handle 302 Redirect in the dialplan --
!--param name=manual-redirect value=true/ --
It appears this never made the wiki, could someone please get it on there.
Thanks
Mike
On Nov 25, 2009, at 6:21 PM, John
Carlos,
Do you have any documentation or scripts for your builds? I'm interested in
having a working automated build and installer build process, and I'm
curious if there's any work you've done that can make my job easier. :)
Thanks,
Josh
On Wed, Nov 4, 2009 at 6:51 AM, Carlos Talbot
There are execute_complete events. I can't recall everything that is in them
but they should always be fired.
Mike
On Nov 25, 2009, at 8:38 PM, Josh Rivers wrote:
Is there a way of determining if a call-command sent to a session via ESL has
completed? Is there a return event which is
I've just checked in the source files for the Inno Setup script I'm using to
build the windows installer(svn 15681). That's about the extent of the
documentation at this point. :)
regards,
Carlos
On Wed, Nov 25, 2009 at 8:10 PM, Josh Rivers j...@radianttiger.com wrote:
Carlos,
Do you have
Are you wanting to provide Lawfull Interecept functionanility for CALEA
Compliance?
http://www.netequalizer.com/caleafaq.php
-E
Gpro.ws
edpimentl [SKype ]
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I need to make faxing easy for some very computer illiterate folk. I am using
an email
service and going to use procmail to print anything incoming automatically but
they cant
get the hang of scanning to an email app, so I am going to buy a Linksys PAP2T
as per the
wiki.
Since the setup will
how about trying Fusionpbx.com ( GUI) -Ram
I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was ready to
release now but decided to do one more release candidate just to be sure. This
should be the last release candidate before the release of version 1.0.
The final release
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