=loglevel value=debug/
your relevant sip profile:
param name=sip-trace value=yes/
T.
On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller klaus.tel...@gmx.net
wrote:
Hi,
Say i have an inbound VoIP/SIP call that hits my FS box. Is it possible
to
to extract information about
command on FS console to get information on what
profile etc. are available as well as their status.
*sofia status*
For more info consult Wiki page at,
http://wiki.freeswitch.org/wiki/Sofia
Thank you.
On Sun, Sep 20, 2009 at 4:44 PM, Klaus Teller klaus.tel...@gmx.net
wrote:
Hi T
Hi,
Say i have an inbound VoIP/SIP call that hits my FS box. Is it possible to to
extract information about the intermediate hops that the call or the signaling
went through? If so, what information can i get?
Thanks,
Gregoire.
--
Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla
Hi,
Anybody with experience in passing caller ID number in junction network? I need
some help.
I have the following origination string:
{origination_caller_id_number=1514xx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/1514xx
Unfortunately, with this, junctionetwork won't pass
On Aug 19, 2009, at 11:39 AM, Klaus Teller wrote:
Hi,
Anybody with experience in passing caller ID number in junction
network? I need some help.
I have the following origination string:
{origination_caller_id_number
=1514xx,ignore_early_media=true}sofia/gateway/sip.jnctn.net
Hi,
I'm playing with Freeswitch and Les.NET right now. It strikes me that the
caller id as passed to javascript doesn't contain the country code.
Anyone knows where teh issue lies?
Thanks,
Klaus.
--
Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 -
sicherer,
Hi,
I tend to believe that we already had this working. Here is my origination
string:
{effective_caller_id_name=Paul
Gascogne,effective_caller_id_number=16478343812}sofia/gateway/sip.gafachi.com/164783486421
The caller number is not being passed to the destination. Is there something
i'm
: [Freeswitch-users] Gafachi no passing caller number
You need to escape the spaces with \s in the caller id name.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 13-Jul-09, at 10:34 PM, Klaus Teller wrote:
Hi,
I
On 13-Jul-09, at 10:43 PM, Klaus Teller wrote:
It doesn't seem to work though. I tried removing the space
completely as well as removing the caller name parameter.
Original-Nachricht
Datum: Mon, 13 Jul 2009 22:36:37 -0400
Von: Mathieu Rene mrene_li...@avgs.ca
Hi Team,
I'm still in need of a way to reject a call without answering it. I very much
appreciate your help.
Klaus.
--
GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss
für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02
On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote:
Hi Team,
I'm still in need of a way to reject a call without answering it. I
very much appreciate your help.
Klaus.
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
--
GMX FreeDSL
responding w/ 486 Busy if you know the call doesn't need to fail somewhere
else...
From: Klaus Teller klaus.tel...@gmx.net
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Thu, 11 Jun 2009 18:21:30 +0200
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users
Hi,
I'm still looking for a way to reject a call without answering. I've tried
various things without solution.
From the socket interface i tried:
SendMsg 9015430e-82cf-418c-bf4c-f3ac6e85caf2
call-command: execute
execute-app-name: respond
execute-app-arg: 503
From Javascript, i tried each
Hi,
Freeswitch is taking quiet some time to start. Is is normal these days? it
didn't used to be the case few months ago. Is there anything i can turn off to
start faster?
Thanks,
Klaus.
--
GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT!
Jetzt freischalten unter
Hi,
Going through the socket api, how can i reject a call without having to answer
it first?
I tried sending a hangup command with cause set either to NO_ANSWER or
NORMAL_CLEARING. In both cases, Freeswitch does create another socket to
deliver the very same call.
More precisely, when a call
Datum: Fri, 5 Jun 2009 22:45:29 -0500
Von: Brian West br...@freeswitch.org
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] How to reject a call without answering
Try the respond app.
/b
On Jun 5, 2009, at 10:34 PM, Klaus Teller wrote:
Hi,
Going through
, 2009, at 11:04 PM, Klaus Teller wrote:
Hi,
My settings does not allow me to test the following right now. So
I'm wondering if somebody knowledgeable could help me answer the
following question.
I do know that if i call Freeswitch, i can use Javascript to read
DTMF even
Hi,
My settings does not allow me to test the following right now. So I'm wondering
if somebody knowledgeable could help me answer the following question.
I do know that if i call Freeswitch, i can use Javascript to read DTMF even
without answering the call. My question is can i do this even
Hi,
Can somebody tell me how to achieve the same behavuior as session.waitForAnswer
via the socket interface?
That is, when i call a device, i want to block until the call is completely
answered (not just early media).
Thanks,
Klaus.
--
Sensationsangebot verlängert: GMX FreeDSL -
: [Freeswitch-users] waitForAnswer on the Socket Interface
Klaus,
What is your dialstring? If you ignore_early_media=true then I believe
it will have the same net effect, but it would be good to know exactly
what you're hoping to accomplish.
-MC
On Thu, Jan 15, 2009 at 10:55 AM, Klaus Teller
Hi,
Need your help on this. I have the following Javascript statement:
session.execute(bridge,sofia/gateway/sip.gafachi.com/someNumber) in a file
called gafachiDialout.js
Then, i have the following extension in default.xml:
extension name=6337
condition field=destination_number
anthony.miness...@gmail.com
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] No Sound Heared
is it only a problem in js
what if you call the bridge app in the dialplan?
On Thu, Jan 15, 2009 at 3:20 PM, Klaus Teller klaus.tel...@gmx.net
wrote:
Hi,
Need your help
Hi,
I'm unable to detect whether a call was answered or not. Using a custom Java
API to connect to Freeswitch Socket Interface, i observed that:
1) session.originate(sofia/internal/1003%192.168.50.94) does block until the
callee picks up while
2)
This is a very much awaited tool. Thanks to you guys. Sounds like 2009 will be
a very exciting year in the community.
Klaus.
Original-Nachricht
Datum: Mon, 29 Dec 2008 17:19:44 -0800
Von: Michael Collins m...@freeswitch.org
An: freeswitch-users@lists.freeswitch.org,
Hi All,
Thanks for your feedback. I must be doing something fundamentally wrong.
Inbound socket is working without problems. But the exact things that i do on
inbound socket, i'm not able to replcate them on outbound socket.
The global picture: I have on Xlite registered at extension 1002 and
2. You don't need to send the UUID in after the sendmsg - FS already
knows which call you're controlling.
Bingo! That was it.
Thanks,
Klaus.
--
Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL
für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a
Hi Folks,
so far i could understand how to bridge calls with Javascript. I'm trying to do
the same with Java via the Socket Interface. My first trials weren't
successful. maybe you can help me understand what is goin on.
What i want to do is to bridge an existing leg (Unique-ID is known) to a
Were is PhoneTest located? What is its the package and what is the containing
jar file?
The jar file you need to specify is not the freeswitch.jar but yours.
Freeswitch will find freeswitch.jar if it's in the default location.
Cheers,
Klaus.
Original-Nachricht
Datum: Mon,
to keep FS reading audio to trigger the outbound dtmf.
On Wed, Nov 5, 2008 at 3:37 PM, Klaus Teller
[EMAIL PROTECTED]wrote:
I'm updating now and will try the flush dtmf. I yet have a hard time
understanding how this is related to sending DTMF (instead of reading
DTMF).
This because
Hi Folks,
It's me again with a DTMF issue. Here is what's going on. I have a remote IVR
and i'm writing some code to communicate with it, exercise it, and test it. The
remote IVR is playing DTMF as inband.
Right now, Freeswitch can detect all DTMF digits from the IVR. The interaction
between
: [Freeswitch-users] Inband DTMF Problem
gather a console log at debug level and a trace of the event_socket
traffic.
On Wed, Nov 5, 2008 at 9:08 AM, Klaus Teller [EMAIL PROTECTED] wrote:
Hi Folks,
It's me again with a DTMF issue. Here is what's going on. I have a
remote
IVR and i'm writing
application you can execute before playing the file
and the uuid_flush_dtmf api command you can send to an arbitrary uuid
maybe try those
On Wed, Nov 5, 2008 at 9:59 AM, Klaus Teller [EMAIL PROTECTED] wrote:
HI,
I'm attaching the console log and event-socket traffic. In the latter
: [Freeswitch-users] Inband DTMF Problem
Extract the RTP audio into wav/au files in wireshark and lets look at
them.
/b
On Nov 5, 2008, at 2:30 PM, Klaus Teller wrote:
Hi Anthony,
Let me add some few facts.
1) I am sending DTMF using RFC2833 and the IVR is using Inband
2
PM, Klaus Teller [EMAIL PROTECTED] wrote:
OK, we made some progress on this issue. It appears that if we introduce
50ms delay between the reading (not playing) of the DTMF and the playing
of
the file (or the sending of DTMF), the problem doesn't occur. And if we
place many such calls
Hi Folks,
Just to let you know that we are working on a library for connecting to the
Freeswitch via the socket interface. We plan to release it under LGPL as soon
as it's somewhat robust.
And for those of you who are in the USA, please don't forget to go vote.
Thanks,
Klaus.
--
Feel free
Hi,
I have Freeswitch running on a CentOS 5 box. From this box, i can resolve all
domain names without problem. Yet Freeswitch is not able to originate to
addresses when the domain name is specified. When i use the IP address
everything is fine.
Any idea how to handle this domain name
: Michael Jerris [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Domain Resolution Problem
On Nov 3, 2008, at 3:39 PM, Klaus Teller wrote:
Hi,
I have Freeswitch running on a CentOS 5 box. From this box, i can
resolve all domain names
?
if you add the port to your srv records nobody would even know.
On Tue, Oct 28, 2008 at 3:02 PM, Klaus Teller [EMAIL PROTECTED]
wrote:
Hi Folks,
I need some additional help with this issue. I already had some from
Brian
i'm but still not able to move forward.
I want a non
of the box have something like this in place already
iirc.
On Wed, Oct 29, 2008 at 8:38 AM, Klaus Teller [EMAIL PROTECTED]
wrote:
Sorry if i gave the impression i'm tried to avoid something. There is
nothing i'm trying to avoid, i'm just ignorant.
So how can i translate your
Hi Folks,
I need some additional help with this issue. I already had some from Brian i'm
but still not able to move forward.
I want a non-registered device to be able to call extension 56900 in my
Freeswitch in such a way that i can manage the call using the socket interface.
I believe the
Hi,
I'm calling a registered soft phone (ext. 1003) via the event socket interface.
That is, on one side i have some Java code connecting to the Freeswitch event
socket interface and placing calls and on the other hand i have the soft phone
registered to Freeswitch and awaiting for calls.
] DTMF Star Event Inconsistent
you should be looking for the DTMF event and not reacting to any others
Event-Name: DTMF
any other ones are not necessarily related to what you want.
On Mon, Oct 27, 2008 at 8:49 AM, Klaus Teller [EMAIL PROTECTED]
wrote:
Hi,
I'm calling
at 11:36 AM, Klaus Teller [EMAIL PROTECTED]
wrote:
I do indeed look for the Event-Name attribute. But since for a single
DTMF
digit two events are received from Freeswitch (with Event-Name: DTMF) ,
i
need to differentiate them somehow such that one is processed and the
other
ignored
27, 2008 at 2:08 PM, Klaus Teller [EMAIL PROTECTED]
wrote:
Thanks. I am not bridging any call. Calls are originated via the socket
interface to the extension 1003. And for the same call, all digits
except
star will produce two events while star will produce one event sometimes
and
two
code.
can you update to trunk before testing anymore?
On Mon, Oct 27, 2008 at 1:32 PM, Klaus Teller [EMAIL PROTECTED]
wrote:
Interesting, thanks for pointing that. I would have thought that all
events
related to a call would have the same Unique-ID. Now I'm even more
confused
On Oct 27, 2008, at 3:13 PM, Klaus Teller wrote:
As far as i can tell, there is one single channel. Call is
initiated via the socket interface to the extension 1003 and parked.
Or does parking generate a second channel?
I'm using Xlite to listen on 1003 and for sending DTMF digits
you to execute commands where you
intentionally want to block until you get the answer and in cases where
you
don't want that you can use the bgapi command to launch the command in a
thread and have it return instantly.
On Wed, Oct 22, 2008 at 8:42 AM, Klaus Teller [EMAIL PROTECTED
Hi,
I have a pretty silly question here, so please bear with me. When reading
events through a socket (socket event interface), it is said that i should
either wait for 2CR or if the content length was specified, i must continue
reading until i got the exact number of bytes from the input
Hi,
How do i configure my Freeswitch to accept SIP calls from peers/devices not
registered with it?
Thanks,
Klaus.
--
GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen!
Jetzt dabei sein: http://www.shortview.de/[EMAIL PROTECTED]
auth?
/b
On Oct 10, 2008, at 1:48 PM, Klaus Teller wrote:
Hi,
How do i configure my Freeswitch to accept SIP calls from peers/
devices not registered with it?
Thanks,
Klaus.
--
___
Freeswitch-users mailing list
public and the stuff you want auth on. Then make sure you turn auth-
calls to false on your profile.
/b
On Oct 10, 2008, at 1:57 PM, Klaus Teller wrote:
Anonynous would be enough for me.
Klaus.
--
GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen
?
/b
On Oct 10, 2008, at 3:37 PM, Klaus Teller wrote:
OK, i moved it to the sofia profile and it works just fine. Next
step in my authorization problem. I call in with a softphone and
want my call to be transferred via event socket to a remote
application.
Yet, Freeswitch
PM, Klaus Teller wrote:
Well, i want to manage calls remotely using the socket interface (as
described here http://wiki.freeswitch.org/wiki/
Event_socket_outbound). The calls i want to manage come from non-
registered devices.
Does that make sense?
Klaus.
--
Pt! Schon
Hi,
I have a voice application running in a voice browser and sending DTMF tones.
On the other hand, I have a self-written Java tool running through Freeswitch
to load-test the voice application. To interact, both applications essential do
the following: sending DTMFs, reading DTMS, playing
Hi,
Please help:
1) How do i send DTMF via the socket events interface?
2) It seems playing audios (playback) is non-blocking. Is this the intended
behavior? How can i make it blocking? Alternately, how can i know when the
audio is completed?
Thanks,
Klaus.
--
Pt! Schon vom neuen GMX
Hi,
I'm trying to figure out how to use the command API via XML-RPC. I want to do
the two following things one after the other: 1) place a call, and 2) play an
audio on the newly created call.
The Java code for placing a call (copied from wiki) works fine and is following:
Hi Folks,
Do you by any chance know if the Digium Lumenvox Starter Kit
(store.digium.com/productview.php?product_code=8ASTLUMSTART) works with
Freeswitch?
Apparently i'm not clever enough to find out where to buy a license (or even
register as customer) directly from the Lumenvox site.
-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] LumenVox Speech Starter Kit
Yes it will work. I would also recommend looking at PocketSphinx.
http://wiki.freeswitch.org/wiki/Mod_pocketsphinx
/b
On Sep 14, 2008, at 11:24 AM, Klaus Teller wrote:
Hi Folks,
Do you by any
On Sep 14, 2008, at 11:59 AM, Klaus Teller wrote:
Hi Brian,
Thanks for your reply. I found the following performance result on
the Sphinx website: for a vocabulary of 5000 words, the word error
rate is 7.3 %. Which is excellent for my purpose. My question is,
does it apply
@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] DTMF Reading and Playing
Klaus Teller wrote:
I tried the following but for unknown reason, the caller is not getting
anything:
JavaSession s = new JavaSession(uuid);
s.answer();
s.streamFile(/usr/local/freeswitch/sounds
Datum: Thu, 11 Sep 2008 20:58:17 +0200
Von: Klaus Teller [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] DTMF Reading and Playing
Thanks Anthony. I was wondering if it would make sense to make this
blocking in the same way other methods are (e.g
Folks,
I got two questions:
1) How do i configure Freeswitch to detect inband DTMF?
2) How do i play DTMF to the remote caller?
Appreciate your answers.
Klaus.
--
GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen!
Jetzt dabei sein: http://www.shortview.de/[EMAIL PROTECTED]
Hi,
I'm trying to convert recorded conversations from wav to mp3. I'm using ffmpeg
for this purpose and i'm aiming to do this wthin Javascript.
Yet, I'm encountering a problem. The conversion fails with the error message
copied below. I must note that the same command works perfectly when
Here is the command:
session.execute(system, ffmpeg -i +file+.wav+ -ab 32kb +file+.mp3);
I just tried with the full path and had the same result:
session.execute(system, /usr/bin/ffmpeg -i +file+.wav+ -ab 32kb
+file+.mp3);
Klaus.
--
GMX startet ShortView.de. Hier findest Du Leute mit
]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Gafachi Again
you can set the variables on the originate line with {} , see
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
second example.
Mike
On Jul 30, 2008, at 1:24 AM, Klaus Teller
Hi,
i have a javscript script that was working with Freeswitch 1.0. The only issue
i had is that outgoing dialing via Gafachi was not transmitting the caller ID.
Then I upgraded to the current trunk version (FreeSwitch Version 1.0.trunk
(9205)). But now I'm not able to place calls anymore.
);
new_session1.setCallerData(effective_caller_id_name, 'Klaus Teller');
But this doesn't seem to help.
Any idea?
Thanks again.
klaus.
Original-Nachricht
Datum: Tue, 29 Jul 2008 23:40:14 -0400
Von: Michael Jerris [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
I just did the following:
new_session1 = new Session();
new_session1.originate(new_session2,
'sofia/gateway/sip.gafachi.com/1'+callee_number, 15);
new_session1.setVariable(effective_caller_id_number, caller_number);
new_session1.setVariable(effective_caller_id_name, 'Klaus Teller');
As a result
Hi Folks,
Great jobs. Congratulations. I have a question related to the ASR and TTS
aspects of Freeswitch. I know Freeswitchsupport OpenMRCP. So, are these ASR and
TTS additions somehow integrated through OpenMRCP?
Klaus.
--
GMX Kostenlose Spiele: Einfach online spielen und Spaß haben mit
I guess i missed the discussion on the ASR integration. Can somebody refresh my
memory as to why we went for PocketSphinx instead of Sphinx itsself?
Thanks,
Klaus.
Original-Nachricht
Datum: Thu, 24 Jul 2008 17:33:47 -0500
Von: Brian West [EMAIL PROTECTED]
An:
Not so fast. I need to take a look at what that all involves. Plus, it's a long
time i wrote C/C++ code. Well, i never really produced useful C/C++ code. So,
give me some few days.
Klaus.
Original-Nachricht
Datum: Thu, 24 Jul 2008 17:43:45 -0500
Von: Brian West [EMAIL
@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Gafachi Origination
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Gafachi
/b
On Jul 18, 2008, at 7:22 PM, Klaus Teller wrote:
Hi Folks,
anybody there to help me setup Gafachi for origination?
here is what they propose
@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Gafachi Origination
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Gafachi
/b
On Jul 18, 2008, at 7:22 PM, Klaus Teller wrote:
Hi Folks,
anybody there to help me setup Gafachi for origination?
here is what they propose
in the public.xml file (aka the public context)
/b
On Jul 18, 2008, at 7:33 PM, Klaus Teller wrote:
Thanks Brian,
unless i'm missing something, that link only contains instructions
about termination. My question is about origination (inbound calls).
Klaus.
Original
-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Gafachi Origination
What does sofia status show?
/b
On Jul 18, 2008, at 8:00 PM, Klaus Teller wrote:
freeswitch doesn't even show in the logs that it is being contacted.
And Gafachi doesn't show either that Freeswitch
? That huge negative number seems wrong.
/b
On Jul 18, 2008, at 8:22 PM, Klaus Teller wrote:
sip.gafachi.com gateway
sip:a6XXXremoved@sip.gafachi.comFAILED (retry:
-1216430345s)
___
Freeswitch-users mailing list
Hi Folks,
Questions regading Java (not Javascript):
1) How do I originate a call? I tried somethis as following without success:
JavaSession session2= new JavaSession();
session2.originate(null,sofia/gateway/did.voip.les.net/14163442000);
The error messgae I get is that session2 is not
Thanks for the feedback. But where do I get a GUID/UUID from if I want to
create a brand new session?
Do I need to create one myself?
As the sample code on the wiki shows, if a FS session already exists, i can use
the JavaSession constructor and pass it the UUID. But I tend to beleive that to
.
session.execute(brodge, sofia/default/[EMAIL PROTECTED]);
On Thu, Jun 26, 2008 at 9:06 AM, Klaus Teller [EMAIL PROTECTED]
wrote:
Thanks for the feedback. But where do I get a GUID/UUID from if I want
to
create a brand new session?
Do I need to create one myself?
As the sample
();
On Thu, Jun 26, 2008 at 5:49 PM, Klaus Teller [EMAIL PROTECTED]
wrote:
Party A is indeed created with the sessionUuid created by FS. And yes,
the
bridging that you propose works just fine.
session = new JavaSession(sessionUuid);
session.execute(bridge,
sofia/gateway
of broadcast (see
uuid_broadcast api command) and mod_tone_stream which lets you use
teletone just like you are playing a file.
Mike
On Jun 22, 2008, at 10:49 PM, Klaus Teller wrote:
Hi Folks,
I was was wondering if it's possible to play a teletone in a non-
blocking way
) *Easiest* set the ringback variable on the A leg to the teletone script
of choice or path to an audio file so when you dial B the A will hear it.
On Sun, Jun 22, 2008 at 9:49 PM, Klaus Teller [EMAIL PROTECTED]
wrote:
Hi Folks,
I was was wondering if it's possible to play a teletone
Hi Folks,
I was was wondering if it's possible to play a teletone in a non-blocking way.
What I want to achieve is following:
1) Party A calls to my FS box,
2) A Javascript script picks up the call
3) The script generates a teletone to party A.
4) Then the script starts some processing while
Hi Brian,
I created a LES profile located at sip_profiles/external/les.xml with following
content:
include
gateway name=did.voip.les.net
param name=username value=1490236124/
param name=realm value=did.voip.les.net/
param name=password value=mash.n2rown4/
/gateway
/include
My
Hi Folks,
I have 30$ bounty for the following task. I want a FS profile to support DIDWW.
I intend to use Les.net for outbound calls and DIDWW for incoming. Right now,
the outbound part seems to be OK. But I'm having difficulties creating a
profile (and the related diaplan) for receiving
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