I tried to read dtmfs via event socket and came across this thread.
What do you mean by:
If you're using event socket you have really no reason to use the read
application.
Is there another chance to do this?
Currently I do the following and this isn't successful:
I send the following message
Hello,
I receive the following message during CS_INIT
*Failed to load library libceplang_de.so due to:
/opt/swift/lib/libceplang_de.so: undefined symbol: cst_rx_int*
Later however, FS at least tries to speak:
2008-10-28 23:40:05 [NOTICE] mod_dptools.c:605 answer_function() Channel
Hello Michael,
No, I startet with a 5.1 installation.
Cepstral works on the command line
opt/swift/bin/swift -o hello.wav 'Hallo Peter'
And the voice is registered:
[EMAIL PROTECTED]:/opt/swift/bin# ./swift --voices
Swift command-line synthesis program
Version 5.1.0 of July 2008
Copyright (c)
PROTECTED] On Behalf Of Peter P GMX
Sent: Tuesday, October 28, 2008 4:16 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cepstral 5.1 no sound
Hello Michael,
No, I startet with a 5.1 installation.
Cepstral works on the command line
opt/swift/bin/swift -o
... sorry I couldn't be of further
assistance.
-MC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:freeswitch-
[EMAIL PROTECTED] On Behalf Of Peter P GMX
Sent: Tuesday, October 28, 2008 4:36 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users
OK, I got it. The replay vulnerability only happens when key exchange is
done via unencrypted SIP. I understand that with TLS the Invite message
cannot be replayed as it cannot be seen in clear text.
Brian West schrieb:
Its called TLS...
/b
On Oct 21, 2008, at 4:30 PM, Peter P GMX wrote
Thanks for your support for the vm-passwords.
The most important part for us however is having hashed passwords for
external gateway definitions (we have a lot) and securing pins for
conferences.
Do we have a chance to add this also?
In our environment DTMF is of course transported via SRTP so
, Peter P GMX wrote:
Thanks,
I got it for the directory password (a1-hash).
But what about the voicemail-password and the passwords stored for
external gateways?
Best regards
Peter
___
Freeswitch-users mailing list
Freeswitch-users
I've seen in the XCML files that passwords and credentials e.g. for
directory entries are always stored in clear text. Is there a way to use
encrypted passwords?
Bet regaerds
Peter
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
, 2008 at 6:25 AM, Peter P GMX [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hello,
is there any way to determine whether the outbound leg is ringing?
For example when (in SIP terms) the called UA is sending back a
Ringing
message via SIP, can I somehow recognize
I have another 3 Questions. I know I had already 2 before within the
last 15 minutes, but I need to qualify whether we can build this special
app with freeswitch or not.
The questions are:
1.)
When I do a uuid_playback I want to be able to stop this playback #1
immediately. My Idea is that I may
Hello,
is it possible to play special ring tones or a wav file before answering
the call? I do not really believe so, but maybe there is a chance?
Best regards
Peter
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
My Freeswitch on an 64bit Ubuntu server crashed with a core dump and
there are no infos in the log. Sometimes it crashes while loading the
codecs, sometimes it finishes loading and then crashes.
After all prerequisites were finished it compiled and installed well.
But program execution crashes:
to the dir
where you build FS and do the following:
# cd libs/libedit
# make clean
# sh configure.gnu
# make
once that is done
# cd ../..
# make clean
# make install
On Wed, Oct 8, 2008 at 9:40 AM, Peter P GMX [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
My Freeswitch
Hello,
I want Freeswitch do do the following thing:
* after a call is setup between UA1 and UA2, then if UA2 presses
5 the call is diverted from UA1 to UA3
Is this possible? If yes, how can I do this?
Best regards
Peter
___
:
^(234)$
$1 will = '231'
-MC
-Original Message-
From: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:freeswitch- mailto:freeswitch-
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]] On Behalf Of Peter P GMX
Sent: Friday, October 03, 2008 4:14
I have TLS working with Snom phones. I could not get TLS working with
ZoiperBiz, as the current 2.06 Linux version of 31-Jul simply didn't
communicate to Port 5061.
Atractel has - as a reply to the submitted problems - announced a new
Zoiper release 2.17 for Windows, but It still isn't available
Today I had a strange behaviour:
I am routing calls to an Asterisk PBX. It has a very sophisticated
dynamic least cost router built in - so I use it to terminate mobile and
international calls.
For a first test I created a dialplan which checks for ^231$ in the
dialled number and then routes the
Thanks for the hint. With the following dial-string in the directory it
works:
param name=dial-string value=[EMAIL
PROTECTED],transfer_fallback_extension=${dialed_user}}${sofia_contact([EMAIL
PROTECTED])}/
Best regards
Peter
Michael Jerris schrieb:
On Sep 25, 2008, at 4:04 PM, Peter P
parameter in this case:
user/[EMAIL PROTECTED])
Where does user come from? Is it an implicit internal context?
May I use another context in that case when the phone is registered in
the internal context?
Best regards
Peter
Peter P GMX schrieb:
Hello,
I have setup Freeswitch with xml_curl and provide
Hello Michael,
thanks for the hint, but how shall a dial-string param look like? I
looked up the internet but could not find an example.
Can you provide an example?
Best regards
Peter
Michael Jerris schrieb:
On Sep 25, 2008, at 7:19 AM, Peter P GMX wrote:
I figured out (via ngrep
Hello,
I have setup Freeswitch with xml_curl and provide configs almost
identical to the local xml files. I can call external gateways, however
when I call a local phone, I cannot connect and it drops me to the VM of
the phone.
2008-09-24 23:30:55 [DEBUG] switch_core_state_machine.c:140
schrieb:
Peter P GMX wrote:
According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
set up gateways dynamically, but I do net get it to work:
I always get : Invalid profile
My assumptions for a right xml answer back to FS are as follows, but I
think at least one
Nobody has an idea anbout this?
Peter P GMX schrieb:
According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
set up gateways dynamically, but I do net get it to work:
I always get : Invalid profile
My assumptions for a right xml answer back to FS are as follows, but I
think
Hello Raymond,
I had a look at your code. You are submitting a complete profile for
sofia.conf, right?
Best regards
Peter
Raymond Chandler schrieb:
Peter P GMX wrote:
According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
set up gateways dynamically, but I do net get
Hello Michael,
yes, mod_sofia is loaded. I would not be able to route calls to external
gateways defined under sofia.conf in the xml file if it was not loaded
Best regards Peter
Michael Jerris schrieb:
On Sep 16, 2008, at 4:02 PM, Peter P GMX wrote:
Hello,
as explained everything
fs_curl[3450]: [key_value] = 'limit.conf'
Sep 16 15:56:20 lmdt fs_curl[3448]: [key_value] = 'dingaling.conf'
Sep 16 15:56:20 lmdt fs_curl[3648]: [key_value] = 'sofia.conf'
Sep 16 15:56:22 lmdt fs_curl[3450]: [key_value] = 'iax.conf'
-Ray
Peter P GMX wrote:
Hello
:20 lmdt fs_curl[3648]: [key_value] = 'sofia.conf'
Sep 16 15:56:22 lmdt fs_curl[3450]: [key_value] = 'iax.conf'
-Ray
Peter P GMX wrote:
Hello,
as explained everything works fine, except that I do not get any
sofia.conf requests.
For section=configuration I receive
Hello,
I have done it the following way:
xml_curl.conf.xml:
configuration name=xml_curl.conf description=cURL XML Gateway
bindings
binding name=example
param name=gateway-url
value=http://192.168.0.35:3000/xml_curls/directory;
bindings=configuration|dialplan|directory/
/binding
/bindings
According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
set up gateways dynamically, but I do net get it to work:
I always get : Invalid profile
My assumptions for a right xml answer back to FS are as follows, but I
think at least one of it is false:
1. ) I start with
document
Hello,
as explained everything works fine, except that I do not get any
sofia.conf requests.
For section=configuration I receive requests for
key_value=post_load_modules.conf
key_value=event_socket.conf
key_value=acl.conf
key_value=post_load_switch.conf
key_value=switch.conf
at
Concerning TLS and SRTP on S60 see
http://mosh.nokia.com/common/download/4452B13D5F854A8DE040050A45306C1B/original/Developing_3rd_party_VoIP_clients_on_S60_platform_v1_0_en.pdf
But I a not sure whether they use SDES or Mikey for key exchange.
Brian West schrieb:
Eric,
I wasn't aware that
Brian West schrieb:
Open a Jira on this. SO we can track it at jira.freeswitch.org i'm
not sure what is required but it should work like it is. The
transport=tls won't work on gateways because you set the transport in
the gateway config.
/b
On Aug 29, 2008, at 4:38 PM, Peter P GMX wrote
Hello,
did anyone manage to get a TLS and SRTP connection working between 2
Freeswitch servers?
For my understanding Freeswitch should just behave like a normal UA. So
TLS and SRTP should also be possible, when routing calls between 2 FS
servers, hein?
Maybe someone may also post a sample
all the nat settings.
/b
On Aug 6, 2008, at 7:47 AM, Peter P GMX wrote:
Hello,
I have successfully set up Snom phones for TLS and SRTP so the whole
communication should be encrypted.
However I see OPTIONS SIP requests in clear text coming from
Freeswitch.
Is there a chance
Hello Darren,
I've tried the GUI and it looks fine. However I would like to use TLS in
Freeswitch. Is there a way to use TLS with this GUI?
Best regards
Peter
Darren Schreiber schrieb:
Hi folks,
Nice to see that interest in this project found it's way here...
The FreeSwitch GUI
Has anybody successfulkly tested a Softphone that works with TLS/SRTP
and Freeswitch?
- I tried Minisip (I think it works with MIKEY) - no success
- I tried Zoiper Biz - it did not like to connect via TLS (any hints?)
- more ??
Best regards
Peter
___
and AES_CM_128_HMAC_SHA1_80. I highly recommend you only enable one
cypher suite...
/b
On Aug 3, 2008, at 2:14 PM, Peter P GMX wrote:
I got TLS working right now. It turned out that the modified start/
stop
script for freeswitch which I had from the Ubuntu package caused that
problem
()
Creating agent for internal
2008-08-03 18:57:32 [*ERR] sofia.c:552 sofia_profile_thread_run() Error
Creating SIP UA for profile: internal*
Best regards
Peter
Brian West schrieb:
Did you turn tls on the profile on?
/b
Sent from my iPhone
On Aug 3, 2008, at 6:44 AM, Peter P GMX [EMAIL
the call is setup correctly and then it hangs up.
Did I miss something?
Best regards
Peter
Brian West schrieb:
And you have everything in conf/ssl right?
/b
On Aug 3, 2008, at 12:01 PM, Peter P GMX wrote:
Hello Brian,
Yes it's turned on:
!-- TLS: disabled by default, set to true
-an but there are no processes listening
on port 5061.
Also gentls_cert with real values didn't work.
Without TLS enabled it works well on port 5060.
Any hints how to continue?
Brian West schrieb:
This should get you started.
http://wiki.freeswitch.org/wiki/Tls
/b
On Jul 23, 2008, at 1:29 PM, Peter P
Hello,
has anybody managed to setup TLS? When I change tls to true in
internal.xml, then freeswitch doens't listen on any ports (5060 5061).
I use freeswitch 1.0.0-0ubuntu1~ppa4
Is there any tutorial available (could not find it while googling)? I
would like to set it up with Snom phones
Hello,
did anybody get Twinkle with ZRTPworking?
I tried this with 2 Twinkle clients with both zrtp enabled.
Twinkle sends a=zrtp in the sdp message but this dows not arrive at the
other Twinkle instance after the message passed Freeswitch.
Any help is welcome.
Just to get it: I just run a zrtp daemon and then FS can handle it?
Where can I get zrtp daemon?
best regards
Peter
Brian West schrieb:
Or you can run the zrtp daemon on the linux box and it works also
right to FS.
/b
On Jul 17, 2008, at 4:23 PM, Michael Jerris wrote:
it should in
Hello,
I have the standard Ubuntu installation for Hardy and freeswitch mostly
works fine so far.
Now I test the xmlrpc functionality in order to try the new Telegraph
project which is based on freeswitch now (
http://code.google.com/p/telegraph
Thanks. Now it works!
Brian West schrieb:
Edit conf/autoload_config/modules.conf.xml and uncomment the line so
the module will load.
/b
On Jul 17, 2008, at 5:29 PM, Peter P GMX wrote:
What must I do in order to start xmlrpc for freeswitch?
Brian West
sip:[EMAIL PROTECTED
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