Hi Rex,
You need to allow your acl in internal.xml like the one,
param name=apply-inbound-acl value=internal-network/
Change the internal-network according to your configuration you allowed in
acl.conf.xml.
I have tested with audiocode with PRI line its working fine.
On Mon, Jun 8, 2009 at
...@gmail.com wrote:
That was pretty much a repeat of the same explanation.
I am still not sure what you mean?
What is the call and what is the extension and what is not hanging up?
On Thu, May 28, 2009 at 2:03 AM, Gopalakrishnan A.N sai...@gmail.comwrote:
Let me explain in different way
our abstraction layer).
On May 28, 2009, at 10:14 AM, Gopalakrishnan A.N wrote:
Hi,
I saw the apache portable runtime is included in freeswitch. So
far I understand that using APR will give good performance. Am I
correct? or it has been used for some other scenarios like even
socket
, the extension has to
hangup automatically rite? Thats not happening.
On Wed, May 27, 2009 at 6:31 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
I am not sure what you mean at this point.
On Wed, May 27, 2009 at 5:53 AM, Gopalakrishnan A.N sai...@gmail.comwrote:
Hi Anthony,
thanks
Hi,
I saw the apache portable runtime is included in freeswitch. So far I
understand that using APR will give good performance. Am I correct? or it
has been used for some other scenarios like even socket or dialplan?
--
Thank you with regards,
Gopal,
trying to make FS into a dialer app using
JS.
for every sessionX you create in js with the new Session constructor
sessionX.setAutoHangup(0);
This allows the channels to remain alive outside the script once they are
hungup/transferred etc.
On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N
constructor
sessionX.setAutoHangup(0);
This allows the channels to remain alive outside the script once they are
hungup/transferred etc.
On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N sai...@gmail.comwrote:
Hi,
I had some discussion with the IRC regarding the uuid_transfer gets
hang-up
Hi,
I had some discussion with the IRC regarding the uuid_transfer gets
hang-up where the call is originated via javascript thru event socket. I was
suggested to install latest SVN trunk. I did that again the same issue, the
log is attached with here http://pastebin.freeswitch.org/9103
My call
Yes, you can connect freeswitch with another media gateway like audiocode or
any softswitch.
you can find here to connect with audiocode
http://wiki.freeswitch.org/index.php?title=Configuring_AudioCodes_MP-114/118printable=yes
this is for analog audiocode, you can also connect with same setting
Hi Rex,
Please find the attached file for the PHP script. This script has been
executed in FS 1.0.2. put these two scripts in htdocs directory. access the
http://localhost/sample2.php so that two text box will appear. you can able
to give the extension number and mobile number to dial. Try this
Hi,
I have installed Freeswitch 1.0.3. I am using event socket with
Javascript. When I try to dial the script with below command, the call is
not going thru it seems to be idle. and segmentation fault core dump error,
(freeswitch hangs).[?]
new_session = new Session.originate(session,
Hi,
I am trying to execute the following script, its working fine for call
origination, but cant able to get the status for dialed numbers, able to get
only the last dialed number not for both the numbers. The script as follows,
Javascript
var array = [2];
array[0]=39841799874;
Hi,
I am using event socket. I am trying to dial a outbound number in
Javascript (api jsrun script arguments) and need to bridge the call
outside the javascript. Can we do this as a batch processing. When I run the
script I tried to play a voice file, so that the session answer the call and
a
How can I capture Q931 packets by separating the D channel and B channel?
On Mon, Jan 19, 2009 at 7:23 PM, Michal Bielicki
michal.bieli...@voiceworks.pl wrote:
It is both :)
Helmut Kuper schrieb:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Michael,
I'm currently on my way to
:
action application=set
data=continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,486,503/
Regards,
Ognjen
On Mon, Jan 5, 2009 at 8:53 AM, Gopalakrishnan A.N sai...@gmail.comwrote:
Hi,
Is there any possibilities that Freeswitch may detect the SIP response
code from the IP media gateway
Hi,
Is there any possibilities that Freeswitch may detect the SIP response
code from the IP media gateway.
--
Thank you with regards,
Gopal,
___
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Freeswitch-users@lists.freeswitch.org
Does freeswitch support VXML? Is there any separate module for this.
--
Thank you with regards,
Gopal,
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Hi,
I have configured the freeswitch, we are dialing through event socket,
if i dial a call per day say around 200 to 300 calls, at the end of the day
the sessions are not ending up in the freeswitch, i can able to see in the
console till all the calls were hanged up, I am using .NET crm.
Hi Micheal,
Is it anything like i am violating the laws? please let me know.
On Fri, Dec 5, 2008 at 8:11 PM, Michael Jerris m...@jerris.com wrote:
On Dec 5, 2008, at 6:23 AM, Gopalakrishnan A.N wrote:
Hi Micheal,
Thanks for the reply! cant I try with tone detect?
Like dial
Hi,
I would like to have predictive dialing. In asterisk we used manager api
and for outbound we use originate. The originate command will dial a number
where asterisk answer the call and then we predict the answering machine
with the silence file. Inspite of that human voice is detected and
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