sipura/linksys
look in ebay.
On Sun, Nov 22, 2009 at 5:35 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
HI All,
Has anyone got some recommendations on which ATA to buy that supports
TLS and SRTP?
Thanks!
--
Itamar Reis Peixoto
e-mail/msn/google talk/sip:
Do LInksys devices support TLS and SRTP that FS supports? 3102 at
least doesn't according to this post
http://osdir.com/ml/telephony.freeswitch.user/2008-08/msg00904.html
On Sun, Nov 22, 2009 at 7:20 PM, Itamar Reis Peixoto
ita...@ispbrasil.com.br wrote:
sipura/linksys
look in ebay.
On
it's support SRTP
On Sun, Nov 22, 2009 at 7:21 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Do LInksys devices support TLS and SRTP that FS supports? 3102 at
least doesn't according to this post
--
Itamar Reis Peixoto
e-mail/msn/google talk/sip:
Are either global or regular channel variable mutable during a call?
Or can they only be set before and after?
Any clarification would help, since the existing wiki doesn't make it clear.
Lon
___
FreeSWITCH-users mailing list
Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah:
I need help as I cannot receive calls through VOIPUSER. This is a learning
setup Attached are my conf files. What is wrong with them ? When I dial from
a landline I get a continuous beep.
Attached are my gateway and the conf file to
it is better to enhance mod_fax with t.38 support... we have done sometihng
and it is close to be work...
T.
On Sat, Nov 21, 2009 at 2:17 AM, Michael Jerris m...@jerris.com wrote:
I think a better approach here is to use spandsp. We already have some
groundwork done for this. If you are
For only sending and receiving that's true.
But my customer wants 2 things:
- Using HylaFAX as fax server, as there are a lot of client apps and other tools
- Connecting real fax machines using a Linksys/Cisco SPA2102 (as this is
certified by their SIP/ISDN gateway vendor)
So I could really
After the help of a couple of people from this list, I now have FreeSWITCH
running - yeah! I have installed X-Lite on a couple of computers and they dial
each other, play music on hold, etc. I have not yet connected to the outside
world.
I purchased an IP-0010 phone off eBay ($20 including
Hi Michael
Thanks
I had set it to send incoming calls to extension 1001. This is in the
file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory.
The contents are :
extension name=inbound-*userna...@sip.voipuser.org]
condition field=destination_number expression=08444846450
Not sure where to start with this one, the outgoing leg to our sip
provider sounds perfectly fine but with the our M3's the incoming leg
is super choppy. Using twinkle on my laptop yields good results in
both directions so there must be an issue with just the snoms, their
firmware was very old but
We discussed build integration related issues a few months ago with Mike and
seemed to find a solution which would work for both UniMRCP and FreeSWITCH
source trees.
Now I've just got a chance to look into this a bit closer trying to further
complete VS2008 build integration in FreeSWITCH. So
Jira is the best, otherwise just mail me the patch and I'll take a
look. Also, I just synced lib up to current trunk. Can you take a
look at my last patch to the module to make it build please.
Mike
On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan achalo...@yahoo.com wrote:
We discussed build
I'm trying to conserve processor power by recording in native file format,
PCMU in my case. It works great with the following line
session:execute(record,
/tmp/my_recording...session:getVariable(read_codec));
however it fails to work with
session:execute(record_session,
Mike,
Jira is the best, otherwise just mail me the patch and I'll take a look.
I've uploaded the patch against svn trunk to
http://jira.freeswitch.org/browse/MODUNIMRCP-6
it's made for win32 debug only yet.
Can you take a look at my last patch to the module to make it build please.
I see. I've
I now created a file inbox.PCMA and get the following:
* inbox.PCMA is played
* the recorded voive mail file is not played (FS does not even try
to do that)
* then I hear
o to listen to the recording press 1
o to save the recording press 2
o ...
Mike,
upgrade UniMRCP to http://code.google.com/p/unimrcp/source/detail?r=1297
and remove that #if from mod_unimrcp. API is backward compatible now
src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c
===
---
On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman rob4manh...@gmail.com wrote:
Hi Sam,
Take a look at mod_xml_curl. Pretty sure it'll do everything you're looking
for.
Looking at that diagram it seems like mod_xml_curl makes a call for
every SIP connection. That seems like overkill. Is there a
did you try without any .wav or .PCMU?
2009/11/23 Matthew Fong mattdf...@gmail.com
I'm trying to conserve processor power by recording in native file format,
PCMU in my case. It works great with the following line
session:execute(record,
/tmp/my_recording...session:getVariable(read_codec));
Yes, the latest trunk works.
Thank you!
Gaurav
--- On Fri, 11/20/09, Michael Collins m...@freeswitch.org wrote:
From: Michael Collins m...@freeswitch.org
Subject: Re: [Freeswitch-users] Broadvoice 32 transcoding support?
To: freeswitch-users@lists.freeswitch.org
Date: Friday, November 20,
Hi,
I am using the Freeswitch 1.0.4pre7. Great application, but I encountered a
problem wich I can not solve since I am very new to it.
Two things are happening.
1) The mod_cdr_csv.c (line 122 do_rotate()) does not always respond to sighup
signal to rotate the cdr-csv files. Some times it
Hi Arsen,
I would be happy to help with the FS integration if you want - please do put
your patch in a Jira.
Jeff
Date: Sun, 22 Nov 2009 10:09:41 -0800
From: ml-node+4047148-1118239...@n2.nabble.com
To: jl...@frontiernet.net
Subject: Re: [Freeswitch-users] need help !! Problem with
I am waiting only for DTMF events. That's why I am setting freeswitch
variable for knowing whether the playback has done.
My question is why this freeswitch variable is not setting properly when I
play back more than 10 files using playback_delimiter option?.
When I play back lesser than ten
On Nov 22, 2009, at 11:51 PM, 大泥人 qinglan_z...@hotmail.com
wrote:
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
23 matches
Mail list logo
