are upgrading from previous versions and having trouble, you may
want to
mv /usr/local/freeswitch /usr/local/freeswitch.bak and then do make
current vm-sync from the build root.
On Wed, Aug 27, 2008 at 3:56 AM, Woody Dickson [EMAIL PROTECTED]wrote:
Hi,
When typing in load mod_voicemail , I am
Hi,
I tried to run ./configure --enable-core-odbc-support but got the
following error during make:
Compiling src/switch_pcm.c ...
Compiling libs/libteletone/src/libteletone_detect.c ...
Compiling libs/libteletone/src/libteletone_generate.c ...
Compiling src/switch_odbc.c ...
In file included
Hi Brian,
Thanks for your help. It works now, but I have another problem. The MOH
that is being heard by the other side does not sound right. It sounded like
the bit rate is not right or something. Freeswitch did send out Audio but
xlite can't properly translate. Any idea what this may be
no
need to run a 16k hold music on an 8k channel.
/b
On Oct 20, 2008, at 9:15 PM, Woody Dickson wrote:
There are three rates specified ( 8000, 16000, and 32000), so how
do I select which one to use?
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Hi,
I would like to know if the following scenario is possible with Freeswitch:
A connection is established between two endpoints ( soft phones). On one
end, someone presses *1, a BEEP that indicates the beginning of recording is
heard, and then Freeswitch starts to record. I can get recording
Hi,
Thank you very much for your prompt response. I actually did try to put the
read app in a separate extension, but what happened was that when *1 was
pressed, Freeswitch would play MOH and both both soft phone couldn't hear
each other anymore. After MOH is played, Freeswitch just continues
Hi Anthony,
Thanks for your reply. It solves the problem. Just one more thing. Is it
possible to have Freeswitch not to play moh when *1 is pressed? In the
following log, the moh is always played after *1 is pressed and before the
execution of the new extension:
2008-10-26 18:21:22 [DEBUG]
Hi,
I tried to avoid Freeswitch from playing moh when the meta_app is executed
by setting hold_music=silence:
condition field=destination_number expression=^(.*)$
action application=set data=hold_music=silence /
action application=set data=call_timeout=120 /
action application=set
Hi,
I am trying to write an event listener that can record the time when a FIFO
consumer rejoins the queue after the caller hangs up. Tracing through all
the event traffic, I notice that there is FIFO-Action= consumer_start,
consumer_stop, and consumer_pop, but there isn't one that indicates
Hi,
I have my dialplan set as follows:
action application=set data=playback_terminators=# /
action application=phrase data=vm_count,4:new:9:old /
But the playback terminators does not work when the phrase is being played.
Is the the wrong way of specifying the terminator key for phrase?
Thanks,
Hi
I am using Openser as the sip proxy in front of freeswitch. When using
Record-Route, Freeswitch hangs of every call after 30 s.
277.32.22.33:5060 is the public ip of openser and 192.168.1.101:5800 is
freeswitch's external profile port. Both openser and freeswitch are within
the same box.
Hi,
I am sorry again for sending another email to the group again. I am working
on embedding libfreeswitch to provide better monitoring. The first thing I
attempt to do is to run the sample code provided in the website:
#include switch.h
int main(int argc, char **argv)
{
Hi,
I am just having a dumb question and hoping someone can help me. I am
trying to run a c program with libfreeswitch embedded so I can use some
external mechanism to keep track of freeswitch, but I am having problem
while compiling:
[EMAIL PROTECTED] fs]# gcc switchnode.c
Hi,
Is it possible to change the directory where freeswitch looks for .lua
scripts?
I would like to place the lua scripts in the shared drive so multiple
freeswitch can refer to it.
Thanks,
Woody
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Hi,
In my deployment scenario, I plan to have two redundant freeswitch
servers running on two different boxes. Two key features I am
leveraging on freeswitch are voicemail and call recording and
playback., and as a result of that, a shared storage for playback of
the recorded wav files is
Hi,
I tried to configure opensips as sip proxy and sip registrars and
freeswitch as B2BUA. Everything works until I start to connect sip
clients that are behind ADSL.
Both freeswitch and opensips are on public IP and I am using external
profile as well.
Does anyone have experience in setting
Hi
My external.xml is just the default configuration:
param name=debug value=0/
param name=sip-trace value=no/
param name=rfc2833-pt value=101/
param name=sip-port value=$${external_sip_port}/
param name=dialplan value=XML/
param name=context value=public/
param
Hi,
Is it possible to dynamically add entries to an ACL without having to
go through the xml file? Can it be done via command line or api?
Thanks,
Woody
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Hello
In the wiki, it is suggested that more than one profile should be used
if libsofia is the bottleneck. When using multiple profiles to handle
incoming call and each profile having an unique port, what is the best
way to redirect and distribute incoming traffic? Is there any mod in
Hello,
I am getting a strange problem in my dialplan.
After doing SET, I want to use it in the next condition field. But then
the value is not being set properly.
Could someone please tell me what is wrong?
Thanks,
Woody
Here is the dialplan:
context name=conf-execution
extension
Hi,
I would like to use freeswitch as a gateway for sending and receiving short
message.
Does Freeswitch have the capability to send and recevie SIP MESSAGE?
How can I set it up? I can't find any document on how to use Freeswitch for
text message.
Thanks,
Woody
Hi,
I want to implement a module where freeSWITCH would try to bridge to an
extension and if the bridging operation fails, my module can use the hangup
code to determine the next cause of action.
With switch_caller_extension_add_application(session, extension, bridge,
Am 5-Aug-09 um 7:20 PM schrieb Woody Dickson:
Hi,
The problem is that I need freeswitch to continue executing the code after
switch_status_t channel_receive_message even when it gets error SIP code
from the destination. Is that possible?
I know if I set up another action after my module
Hello,
I find hangup_hook, but I would like to define different actions for
different hangup codes. Is there anyway to do that?
Woody
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...@freeswitch.orgwrote:
On Sun, Aug 16, 2009 at 4:24 AM, Woody Dickson woodydick...@gmail.comwrote:
Hello,
I find hangup_hook, but I would like to define different actions for
different hangup codes. Is there anyway to do that?
I can think of at least two ways you could do this: one
Hi,
I am running 1.0.4 right now using latest trunk.
After a high traffic session, I do show channels, I would find a bunch of
CS_HIBERNATE channels that don't get removed after all the traffic is
gone.
Does anyone know what is the case of thoes CS_HIBERNATE'd channels? How can
I set a
:54 AM, Woody Dickson wrote:
Hi,
I am running 1.0.4 right now using latest trunk.
After a high traffic session, I do show channels, I would find a
bunch of CS_HIBERNATE channels that don't get removed after all
the traffic is gone.
Does anyone know what is the case of thoes
wish, you can also hop on #freeswitch / irc.freenode.net and
have someone look into it.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 21-Aug-09, at 5:54 AM, Woody Dickson wrote:
Hi,
I am running 1.0.4 right now
Hello,
I read something that talks about using memcache for mod_limit before. Is
it something that is available now?
If I have multiple instances of freeswitch that need to share the same limit
status, it there any existing solution?
If no existing solution is available, what is the best way
Hi,
I would like to set up freeswitch to automatically expire a user
registration if either NOTIFY or REGISTER is not received within certain
time frame.
Does anyone know how to do that?
Thanks,
Woody
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Hi,
I am having a strange problem here. sofia status shows that the user is
registered, but sofia_contact says the user is not registered.
Does anyone know why this is happening?
freeswi...@localhost.localdomain sofia status profile internal reg 180004
API CALL [sofia(status profile internal
On Fri, Sep 11, 2009 at 10:06 AM, Woody Dickson woodydick...@gmail.comwrote:
Hi,
I am having a strange problem here. sofia status shows that the user is
registered, but sofia_contact says the user is not registered.
Does anyone know why this is happening?
freeswi...@localhost.localdomain sofia
Thanks Brian, you are correct. My problem is solved.
Thank you so much.
On Sun, Sep 13, 2009 at 9:20 AM, Brian West br...@freeswitch.org wrote:
Also I'm going to suspect you have removed the domain aliases from the
profile. If you have then you can't just do sofia_contact
u...@domain...
Hi,
While trying to record some sounds with the voicemail app, I keep getting
message saying my record is below the minimal length even I was actually
still speaking.
Is it not detecting my voice? How can I configure it so that freeswitch's
vm app can detect my speech?
Thanks,
woody
Hi,
I tried to performance test freeswitch with media proxy thur fs. With 400
cps, I start to see 2000 channels remaining in Freeswitch, and then no read
codec error starts to pop up. With only 1875 channels, how come freeswitch
is complaining about no read codec? Also, I am using media_proxy
Hi,
Is is possible to override any of the setting specified in the conference
profile?
What I want to do is to have a default profile, and be able to modify
certain fields if necessary in the dialplan.
Alternatively, I would prefer to have a dynamic profile setting for the
conference to obtain
Hi,
Is this just me who is having this problem? I can't compile the latest
freeswitch source code and here is the error:
checking for gcc option to accept ANSI C... none needed
checking for style of include used by make... GNU
checking dependency style of gcc... gcc3
checking whether gcc and cc
Hi,
Is there anyway of using curl without having to setup a standalone http
service? Is it possible to generate curl xml using scripts?
woody
On Mon, Oct 5, 2009 at 11:05 PM, Michael Jerris m...@jerris.com wrote:
On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
Is is possible
Hi,
I am trying to setup a Digium TDM400P following the instruction on the
wiki.
It is a 1 fxo and 1 fxs card, so I tried
loadzone=in
defaultzone=in
fxsks=2
fxoks=1
and
loadzone=in
defaultzone=in
fxsks=1
fxoks=2
None works. Does anyone know how it should be configured?
Here is what I get by
at 3:40 AM, Woody Dickson woodydick...@gmail.comwrote:
Hi,
I am trying to setup a Digium TDM400P following the instruction on the
wiki.
It is a 1 fxo and 1 fxs card, so I tried
loadzone=in
defaultzone=in
fxsks=2
fxoks=1
and
loadzone=in
defaultzone=in
fxsks=1
fxoks=2
None works
Hi,
Is there any API to tell freeswitch to send a SIP OPTION message to check
the availability of a SIP client?
thanks,
woody
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Hi,
I am having problem trying to use mod_xml_odbc using freeswitch-1.0.5pre.
Here is the error I am getting:
2009-11-15 00:17:23.571293 [INFO] mod_xml_odbc.c:647 XML ODBC module
loading...
2009-11-15 00:17:23.571354 [NOTICE] mod_xml_odbc.c:563 Binding XML Search
Function [directory]
2009-11-15
Hi,
Is there anyway to detect when a channel is park in a way that is similar to
hangup-hook or answer-hook? I would like to detect that inside a custom
mod, without using the event mechanism?
woody
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