Did you try ./wanrouter start before starting FreeSWITCH ?
- Original Message -
From: dalech...@yahoo.com
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, December 12, 2008 2:51 AM
Subject: [Freeswitch-users] config help: openzap and T1 A102u
I am stuck trying to bring up
Hi,
I'm working on a simple dialer, and I have the following problem: the
audio file starts playing before the user answeres the phone (while it's
ringing). It only works when I introduce a delay, but that doesn't seem
right.
For instance in the asterisk context referred in the call files, I
Hi,
I see that mod_cdr is marked as being non-functional on the wiki. I'm working
on a dialer and I need a way to log information about calls.
What module should I use?
Thanks,
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On Saturday 13 December 2008 01:10:29 Alexandru Nedelcu wrote:
Hi,
I see that mod_cdr is marked as being non-functional on the wiki. I'm
working on a dialer and I need a way to log information about calls.
What module should I use?
This was answered on IRC and a note added to the mod_cdr
I'm happy to see that you can add consumers to queues using the fifo.conf.xml
configuration file. I have made some tests and I hope it may lead to a more
universal way of setting up queues for small organisations than the one I have
described in the wiki, and which includes (too) many
How are you originating calls? You probably need to add
{ignore_early_media=true}. This tells FreeSWITCH not to return from
origination when early media (progress/ringing) was received (I think
anyway)...
See http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media
There is a sample
Hello,
First, I would like to apologize for a mistake I have made: by adding the
following line in the profile
param name=enter-sound value=path/to/file.wav /
the enter sound is played.
I am sorry for this. I did not hear it because in the case I have been
analyzing the members of the
FreeSWITCHers,
I would like to thank everyone that donated. Enough was raised to
cover the sound order. ;)
Thanks,
Brian West
FreeSWITCH.org
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Hello,
I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04
server in a non-root environment.
We experienced a timer problem which led to this FS console error
does anyone have a sample of the
config file for mod_xml_ldap? and know where to put it?
Vinicius Kobashi escreveu:
i did it... i still had some
problems with sasl, but i managed to fix them.
now the module is up and running but i still dunno where to put the
mod_xml_ldap
if you open a jira issue on it we can probably add your patch and/or the
config option.
the users-list is a tough place to manage TDM issues.
On Fri, Dec 12, 2008 at 9:01 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
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Hello,
I try to establish
the entries are standard originate strings so all of the {} variables apply.
On Fri, Dec 12, 2008 at 7:30 AM, Jon Bruel j...@consiglia.dk wrote:
I'm happy to see that you can add consumers to queues using the
fifo.conf.xml configuration file. I have made some tests and I hope it may
lead to
No,
there is currently no way.
On Fri, Dec 12, 2008 at 8:26 AM, Carole O. carole.oliv...@enst.fr wrote:
Hello,
First, I would like to apologize for a mistake I have made: by adding the
following line in the profile
param name=enter-sound value=path/to/file.wav /
the enter sound is
I have a call scenario that involves transferring the call and dropping out
of the SIP/RTP stream. I need to accept the SIP call, play a prompt, and
retrieve a pin code. After a database lookup, I need to transfer the call
to another FS server and drop out of the SIP path. I have done this with
You can use deflect to accomplish this.. it will do a refer to the
other FS box.
/b
On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote:
I have a call scenario that involves transferring the call and
dropping out of the SIP/RTP stream. I need to accept the SIP call,
play a prompt,
Thank you, that is exactly what I need.
On Fri, Dec 12, 2008 at 9:14 AM, Brian West br...@freeswitch.org wrote:
You can use deflect to accomplish this.. it will do a refer to the
other FS box.
/b
On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote:
I have a call scenario that
In Asterisk I was able to set a custom CDR field by doing something
like:
Set(CDR(userfield)=${SOMETHING})
I need to set a custom field in FreeSwitch, and preferably I want to
have control over its value from Javascript.
Can someone tell me how? :)
Thanks,
--
Alexandru Nedelcu
Software
Yes, I'm familiar with that since i invented that feature for Asterisk =D
In FreeSWITCH, All variables are already available from the cdr
just set regular channel variables.
for xml cdr they are all there right away
for csv cdr you can reference any channel variable in your template.
On
Are you using CSV or XML? The reason I ask is because I personally use
XML and I find that having lots of information (even too much) is
better than not enough. The only drawback to XML that I find is that
you have to know how to parse it properly. :) The level of detail in
the XML CDRs is
Jason,
If I understand correctly software other than PA can lock up the sound
card so that PA doesn't see it. That might explain why PA reports
number of devices = 0. Could you check to see if possibly something
else has control of your sound card, perhaps ALSA? Turn off anything
that might use
On Fri, 2008-12-12 at 13:18 -0600, Anthony Minessale wrote:
Yes, I'm familiar with that since i invented that feature for Asterisk
=D
In FreeSWITCH, All variables are already available from the cdr
just set regular channel variables.
for xml cdr they are all there right away
for csv
Thanks Michael,
I'm going to use XML, since I don't really know what variables I want.
Another problem with CSV is that many people parse them with regular
expressions and scripts break when you add a new column.
On Fri, 2008-12-12 at 11:50 -0800, Michael Collins wrote:
Are you using CSV or
On Sat, 2008-12-13 at 01:26 +1300, Hadley Rich wrote:
This was answered on IRC and a note added to the mod_cdr wiki page.
Thanks Hadley,
I'm a total newbie to FreeSwitch and voip in general, sorry for my
persistence :) I'll try writing an article about my setup this weekend.
What I think would be neat is to have a perl script to parse the XML
cdr and spit out a graphic of the call path... now that would be neat.
/b
On Dec 12, 2008, at 2:14 PM, Alexandru Nedelcu wrote:
Thanks Michael,
I'm going to use XML, since I don't really know what variables I want.
On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu a...@sinapticode.ro wrote:
Thanks Michael,
I'm going to use XML, since I don't really know what variables I want.
Another problem with CSV is that many people parse them with regular
expressions and scripts break when you add a new column.
Hi there,
you have to use the default ALSA audio device to share it, and to
have it automatically format and rate converted.
the default ALSA device is not the default portaudio device (not in
the portaudio version used currently by FS).
You have to find out what device id it has under
Are there any good examples floating around of XML parsers for this to dump
to MySQL?
On Fri, Dec 12, 2008 at 2:22 PM, Michael Collins m...@freeswitch.org wrote:
On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu a...@sinapticode.ro
wrote:
Thanks Michael,
I'm going to use XML, since I
On Thu, 2008-12-11 at 09:55 -0500, Raymond Chandler wrote:
i think i answered all of this for you on irc yesterday
Yes you did, thanks for your help.
I'm a total newbie, but the good news is that I'm almost finished with
my setup. FS is great :)
use the bridge dialplan app to dial by
Sorry, the previous one was sent by mistake.
This one is complete:
Hi there,
you have to use the default ALSA audio device to share it, and to
have it automatically format and rate converted.
the default ALSA device is not the default portaudio device (not in
the portaudio version used
On Fri, Dec 12, 2008 at 12:21 PM, Brian West br...@freeswitch.org wrote:
What I think would be neat is to have a perl script to parse the XML
cdr and spit out a graphic of the call path... now that would be neat.
/b
I think that is a great idea. I was kicking that around as an add-on
feature
I don't know about good examples. I just hacked together a perl
script to extract the very specific elements for my application. If
anyone out there has a sample XML-to-db parser that would be very
welcomed...
-MC
On Fri, Dec 12, 2008 at 12:28 PM, Shelby Ramsey sicfsl...@gmail.com wrote:
Are
Hi,
I'm trying to integrate Cepstral TTS I read in the wiki that I need
Ceptral's SDK to compile the mod_ceptral, can somebody tell me where can I
get the trial version of this SDK?.
Thanks.
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If you're on linux you need to go download and install any voice. If
you're on windows I have to forward your request to Cepstral to get
the SDK for windows.
/b
On Dec 12, 2008, at 3:47 PM, Pedro . wrote:
Hi,
I'm trying to integrate Cepstral TTS I read in the wiki that I need
On Fri, Dec 12, 2008 at 09:30:16PM +0100, Giovanni Maruzzelli wrote:
But in this specific case, no device at all was found.
So, maybe portaudio was not commpiled with ALSA support (do you have
the ALSA development library installed?).
Yes, and in any case the version of PortAudio which is
Is there a way to schedule a certain DTMF tone to be played into a
bridge (both a and b legs) after a scheduled number of seconds into the
call?
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sched_api (hint uuid_send_dtmf)
API CALL [sched_api()] output:
-ERR Invalid syntax. USAGE: [...@]time group_name command_string
/b
On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:
Is there a way to schedule a certain DTMF tone to be played into a
bridge (both a and b legs) after a
This is interesting...
I wrote the following test.
#includestdio.h
#include portaudio.h
int main(int argc, char **argv)
{
Pa_Initialize();
printf(Number of devices: %d\n,Pa_GetDeviceCount());
Pa_Terminate();
}
then I compiled and executed it:
gcc -o pa_test pa_test.c -lportaudio
With apologies for the noise on the list, I just realized that FreeSWITCH is
building its own version of PortAudio.
I can confirm that Alsa is being detected and support for it included.
So, there must be some difference between the version of PortAudio that comes
with FreeSWITCH, and the
Thanks again Anthony !
You fixed the issue with DTMF i had reported :
http://jira.freeswitch.org/browse/FSCORE-251
Chris Danielson added to Wiki a nice page collecting these issues with
Sonus :
http://wiki.freeswitch.org/wiki/RTP_Issues
Cheers,
El mié, 10-12-2008 a las 03:10 +0100, Angel
The problem is now solved.
It turned out to be permissions: the freeswitch user wasn't added to the audio
group in /etc/group, hence didn't have permission to interrogate the audio
devices.
Perhaps a future version of the Debian package could address this, or at least
it should be noted
Jason,
Thanks for troubleshooting this! At the very least I will add a note
to the PA section of the wiki.
-MC
On Fri, Dec 12, 2008 at 6:01 PM, Jason White ja...@jasonjgw.net wrote:
The problem is now solved.
It turned out to be permissions: the freeswitch user wasn't added to the audio
Not much written in the wiki on this. Also searched the list and not
much on either sched_api or uuid_send_dtmf.
So from an xml dialplan, can sched_api as an application?
Is there any way to have the time offset reference the point at which
the call started ? ie. When the called party answers?
Frank,
I'm sure this is possible. Please give me a little bit to look into
this. I'm going to see if I can lab it up and give you a sample
dialplan. Also, thanks for the heads up on the wiki not having this
information. I will put that on my not-so-short wiki todo list.
Thanks,
MC
On Fri, Dec
Frank,
I found a simple way to handle this scenario. I decided just to create
a small Lua script that would do the job. It's committed in latest
trunk. Look in src/scripts/contrib/mcollins for uuid_send_dtmf.lua. It
has comments on how to call it, including a sample dp call.
The way I would use
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