Message: 9
Date: Sun, 30 Nov 2008 18:42:30 -0800 (PST)
From: Marc Orenberg [EMAIL PROTECTED]
Subject: [Freeswitch-users] Problem importing modules in mod_python
To: freeswitch-users@lists.freeswitch.org
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
In the
Hi,
*
**It is possible to dial outbound through console dialing. Yes means me
How ?**
Without using the softphone how can i dial outbound from freeswitch
console itself. *
*
I want to Know without using any softphone for calling.
It is possible in asterisk. we can dial from console
Hello Baskar,
in FS it is possible to call from console using the endpoint mod_portaudio.
Please have a look at
http://wiki.freeswitch.org/wiki/Freeswitch_softphone , it is *NOT REAL
SOFTPHONE* it is FS used *LIKE* a softphone. Exactly as in Asterisk
with chan_alsa or chan_oss.
Sincerely,
Hello Maxim,
can you reach another internal device except the GSM one in order to see
whether it's GSM codec specific?
However I can see that you're using local IPs (10.x.x.x) so I expect
that they are natted. This often causes one way audio when the external
rtp-ip is not set. Please try to set
hi simon,
i am not sure, if i understood your problem right, but if you do not
want leg a to hang up after leg b (the originated call) hangs up, set
park_after_bridge=true when you make the originate.
as far as i know, hangup_after_bridge=false is only for the inbound
and helps nothing with the
Hi Simon,
You can get the A leg uuid and B leg uuid seperately and can hangup
whichever the leg you need...:)
--
Thank you with regards,
Gopal,
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Hi,
I'm having problems using TLS to receive calls.
I'm using a Nokia N95 to test TLS against freeswitch. I can register my
client against freeswitch and make outbound calls to the test numbers (e.g.
).
I can also make calls to other users registered over UDP.
However if I try to make a
Hy!
You were right about the contact in 183, its port 5060 in there.
I've tried turning of 100rel, it seemed to work with calls, but caused some
problems with others things, so I would really appreciate if there is another
option.
Btw, I have mentioned that, that I had gateway
All the variables that I set show up only in the a-leg CDR.
How can I set a variable that can be used during the b-leg CDR generation?
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Did you add
action application=export data=sip_secure_media=true/
into youy dialplan before bridging that call. How is your internal.conf,
is TLS enabled there?
Best regards
Peter
matrim schrieb:
Hi,
I'm having problems using TLS to receive calls.
I'm using a Nokia N95 to test TLS against
Please tell that to everyone out there in the REAL world. It was my
understanding that sips: was the one that went away in favor of
transport= which is what everyone uses.
/b
On Dec 1, 2008, at 7:09 AM, matrim wrote:
According to RFC3261 the use of the transport=tls parameter isn't
sip_secure_media only activates SRTP.
/b
On Dec 1, 2008, at 9:47 AM, Peter P GMX wrote:
Did you add
action application=export data=sip_secure_media=true/
into youy dialplan before bridging that call. How is your
internal.conf,
is TLS enabled there?
Best regards
Peter
Hi Peter,
Thanks for your response.
When I use PCMU two-way audio works fine.
When I make outgoing calls from a Freeswitch extension (using GSM) and then
out to a gateway using PCMU everything works fine.
When I receive calls from the same gateway, the end point behind the gateway
can't hear
if you enable nat mode on the registrations it will lock the ip:port
make an acl that matches the ip of the client and add the param
apply-nat-acl with the name of the acl you created to your sofia profile
then all calls from that ip will be known to be nat and the port locking
code will
if you use the export instead of set app then they will get set on both
legs.
otherwise vars you only want set on b leg you can add to the dial string
{foo=bar,test=true}sofia/default/[EMAIL PROTECTED]
On Mon, Dec 1, 2008 at 8:01 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
All the variables
404 not found means the extension you are dialing is not found, not the
sound file the extension is playing.
press f8 and try again and the debug log will help you figure it out.
On Mon, Dec 1, 2008 at 12:09 AM, Faisal Maqsoodi
[EMAIL PROTECTED]wrote:
I tried to play a sound file using the
I'm not sure about current implementations that servers are using. I'm used
to using sip over UDP and TCP but this is my first time testing SIP over
TLS. So I'm just going by what's in the specification and what's implemented
on the devices I'm trying to test against, which are Nokia S60 devices
probably pstn side has acknowledged our gsm then sent ulaw anyway and we
think its gsm.
most likely there are multiple codecs in the accept packet from the gateway
and they expect us to figure out what codec to use based on the first packet
we get from them rather than just accepting one codec in
we no longer use global name space in our shared objects which seems to have
a side effect on modules who in turn try to load it's own shared objects
because they too inherit the non-global namespace param.
you can either add an attribute to the modues.conf to ask it to load with
global name
Try changing the module definition to use global symbols the same way
we did in mod_lua, see if that resolves the issue.
Mike
On Dec 1, 2008, at 3:31 AM, Traun Leyden wrote:
Message: 9
Date: Sun, 30 Nov 2008 18:42:30 -0800 (PST)
From: Marc Orenberg [EMAIL PROTECTED]
Subject:
yes,
mod_shout will broadcast calls as MP3 that you can listen to in
itunes/winamp live.
On Fri, Nov 28, 2008 at 8:51 AM, Dennis [EMAIL PROTECTED] wrote:
so i would have to make a call with a phone to a specific dialplan? if
so, this would not be, what i whished (although it is nice to have
the guy who made mod_openmrcp has stopped development and is now making a
new library called
unimrcp it will take some time to create a new module and remove the now
unsupported openmrcp.
On Fri, Nov 28, 2008 at 12:15 PM, [EMAIL PROTECTED] wrote:
I'm getting the following errors when trying
Thanks Dennis,
That did exactly what I needed.
Cheers!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dennis
Sent: December 1, 2008 3:47 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Leg A terminated by Leg B on a
Hi Anthony,
Oh! OK.
So is this module totally broken.
I say this because I can't seem to get it to work at all with the example in
that Mod_openmrcp wiki page but I thought it might because I'm not be using the
right Cepstral software (freetrial download versus the paided for SDK) or that
Greetings,
Can somebody tell me, if it is possible to use duoBRI card
(http://www.junghanns.net/en/duobri_express_produkt.html) from
Junghanns.net together with Freeswitch?
I've found that this card has Zaptel drivers, and Freeswitch has
mod_openzap. On the other side, I saw somewhere in wiki
I would not say it is totally broken, it is known to work in quite a
few places, but we are unlikely to be doing any new fixes in it.
Mike
On Dec 1, 2008, at 1:19 PM, [EMAIL PROTECTED] wrote:
Hi Anthony,
Oh! OK.
So is this module totally broken.
I say this because I can't seem to get it
The bri support is still in development, basic calls on ptmp bri do
appear to work, although I am not sure with what hardware.
Mike
On Dec 1, 2008, at 10:26 AM, Sergey Kirillov wrote:
Greetings,
Can somebody tell me, if it is possible to use duoBRI card
MikeJ, if openMRCP isn't totally broken then would you mind helping me get the
example in Mod_openMRCP working or something like it since I don't know what
the heck I'm doing wrong.
I can meet you now over at the IRC channel for Freeswitch users if you like.
Thanks.
-Original
mod_openmrcp was a contribution to the community by a 3rd party individual.
As i have clearly stated in 2 previous emails, the man has decided to
discontinue the openmrcp project.
So now we are left with the remains of the module and discontinued code.
This was not our decision it was his.
Since
Brian,
Will setting progress_timeout = 8 and originate_timeout = 30 help me out
in this situation without using pre_answer?
Basically I'd like to timeout the INVITE to the phone in 8 seconds if it
doesn't respond to the INVITE (phone is not on the network) and send the
call to voicemail, but if
FYI,
I've updated the wiki to reflect the current status of OpenMRCP with a link
to the new UniMRCP project. Hopefully enough people who want MRCP in FS will
support UniMRCP...
-MC
On Mon, Dec 1, 2008 at 11:17 AM, Anthony Minessale
[EMAIL PROTECTED] wrote:
mod_openmrcp was a contribution to
Yes, setting the var to the full path works. Sorry, should have taken
the full path in the wiki more seriously.
MP3s are played only once, 8kHz WAVs work perfectly.
Cheers,
Jan
Can you try putting the full path to the file? Also what does the
console output look like?
/b
On Nov 30, 2008,
Hi Faisal,
the path is either an absolute path or a path relative to the
directory in the sound_prefix var in vars.xml.
So this
action application=playback
data=/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav/
works fine on my box. You sure this one doesn't work for you?
Does bridging a call from FS to Voxeo's Prophecy server require openMRCP? If
not then the other issue I might have is a database look up that is part of the
dialogue that maybe need as the person response to prompts from the asr. It's
possible to run a php script for the database stuff
On 12/1/08, Thomas Troy [EMAIL PROTECTED] wrote:
..snip..
Out of interest do you have any links to anywhere this is discussed in terms
of general sip implementations?
Uh oh, here we go again...
http://www.iana.org/assignments/sip-parameters
http://tools.ietf.org/html/rfc3969
*Hi Giovanni Maruzzelli*,
To list the available devices i have given this command *pa devlist*
*output:*
[EMAIL PROTECTED] pa devlist
2008-12-02 11:27:34 [CONSOLE] switch_console.c:255 switch_console_process()
Unknown Command: pa
But when i check in my system *hwconf *there is auido drives
Just to follow up.
Moshe Yudkowsky has an article on Routing calls from FreeSwitch to Prophecy:?
http://www.prophecy2006.com/node/145
My problem is that Freeswitch and Prophecy need to be on the same machine BUT
both need to bind to port 5060 so I'm getting errors from one or the other
Actually i copied the following text in a new text file and saved it as
test1.xml file in /conf/dialplan/extensions, where 9_enum.xml and
00_pizza_demo.xml exist, but it didnt worked.
extension name=wavs
condition field=destination_number expression=^2009$
action
Hi Baskar,
you have to compile and enable the module mod_portaudio.
Please edit the modules.conf in the main directory of the FS sources,
and remove the # before mod_portaudio. Also, after compilation and
installation (make install), in the directory
/usr/local/freeswitch/conf/autoload/ edit the
I need to barge in again and add to my last post with this email from Voxeo
support. Here is their response to the port binding conflict and it brings up a
possible problem if FreeSwitch will be looking for Prophecy at that port? I
assumed it would if I set up the extension right but now I
On Tuesday 02 December 2008 19:44:21 Faisal Maqsoodi wrote:
Actually i copied the following text in a new text file and saved it as
test1.xml file in /conf/dialplan/extensions, where 9_enum.xml and
00_pizza_demo.xml exist, but it didnt worked. extension name=wavs
condition
Sir thank you very much. It really works.
Faisal
--- On Mon, 12/1/08, Hadley Rich [EMAIL PROTECTED] wrote:
From: Hadley Rich [EMAIL PROTECTED]
Subject: Re: [Freeswitch-users] How to specify Path for sound files
To:
*Hi,
I have updated all the above events you told .It's working fine but when i
call extension 1002 from freeswitch console, call is connected to extension
1002, but FS is aborted but call is established in1002. what shall i do.
what was the error.
Full freeswitch get cut.*
*output:*
[EMAIL
Does the core dump always happen in this call scenario? If so, can you
get a back trace? Put it on pastebin. That will hopefully help narrow
down the issue.
-MC
Sent from my iPhone
On Dec 1, 2008, at 11:27 PM, Baskar [EMAIL PROTECTED] wrote:
Hi,
I have updated all the above events you
Baskar,
that is bizarre.
Seems there is a problem with mod_sofia, the module that manages SIP
connection to the SIP client at 1002 extension.
Maybe someone else on the list can be of more help.
Sincerely,
Giovanni Maruzzelli
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