Dear all,
For implementing a IVR I planned to use event outbound socket.For that I am
in the process of analyzing the /libs/esl/perl/server2.pl code.In that they
created a object for ESL::ESLconnection package then they called some of the
methods like getInfo,sendRecv ...etc using that object.
Hi Michael ,
thanks a lot for support on this.
As to the main problem of your DLL not working, can you send me the full
source code, or all the logging output from loading it? Try managedreload
my.dll to reload the DLL and see how it is registering them. It should
output something like
Hi Łukasz,
Would you please send me the DLL offlist and I'll figure it out?
The new session you create is the b-leg. The parameter it takes in
originate is the a-leg. So you'd do:
var session = new ManagedSession();
session.Originate(context.Session, sofia/default/1000,10);
Hi,
Is there a way to execute more than 1 commands in the execute_on_answer
variable? I want to execute both a python script AND the sched_hangup
application.
--
---
Apostolos Pantsiopoulos
Kinetix Tele.com R D
email: r...@kinetix.gr
can't you just have your python script execute the sched_hangup command, and
then finish the remainder of the python script?
On Wed, Jul 29, 2009 at 12:15 AM, Apostolos Pantsiopoulos
r...@kinetix.grwrote:
Hi,
Is there a way to execute more than 1 commands in the execute_on_answer
variable? I
is there a way to have the freeswitch_http.log, log what command is being
executed across the webapi? For most requests the log looks like
127.0.0.1:17093 - freeswi...@127.0.0.1 - [29/Jul/2009:00:58:29 +] POST
200 422
but it would be useful to know more precisely what is being executed across
Yes, I could do that but the logic of the python script is
not dialplan oriented and I hate mixing different things. I just
found a workaround though. I am executing the python script on the
b leg and the sched_hangup on the a leg (with the help of nolocal:)
Thanx for the help anyway.
Matthew
According to http://wiki.freeswitch.org/wiki/Sofia.conf.xml, if the
remote end say 404, it can be reached. So i guess the problem is that
the remote number 300 can't be found on the otherside? Or could it be
that the PBX is reachable but there's a problem of communication between
FS and the
Hi All
Does Freeswitch support SIP-T / RFC 3372 ?
Regards,
Maxim Tsvetov
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On Suse Linux (Pick a version, 10.3 and 11.1 confirmed) you need to
rebuild perl entirely, otherwise FS will crash and burn upon a perl
script being called from the dial plan.
Best Regards,
Karl J. Vesterling
k...@ken-ton.com
202-461-3231 x0
On Jul 24, 2009, at 12:03 PM, Michael Jerris
Hello freeswitchers!
I thought you're be pleased to know: FS beated Prosody S on subjective test
on lay users.
After 3 users testing the two conferences and reporting far better quality
on FS, we decided to make a more rigorous test.
Then we made a blind test by asking 4 lay users to dial two urls
Hey,Having a problem, first time setting up a FreeSWITCH phone system (or
any networked phone system) and seem to be in over my head.
We cannot call out or call in (though it does go to the voice mail) or call
inside the network (that did work for a time, but then trying to make
everything else
On Wed, Jul 29, 2009 at 1:05 AM, julien jgonza...@sqli.com wrote:
According to http://wiki.freeswitch.org/wiki/Sofia.conf.xml, if the
remote end say 404, it can be reached. So i guess the problem is that
the remote number 300 can't be found on the otherside? Or could it be
that the PBX is
On Wed, Jul 29, 2009 at 12:59 AM, Apostolos Pantsiopoulos
r...@kinetix.grwrote:
Yes, I could do that but the logic of the python script is
not dialplan oriented and I hate mixing different things. I just
found a workaround though. I am executing the python script on the
b leg and the
This is great news. Before we start sending out commercials that say 4 out
of 5 Brazilians prefer FreeSWITCH to Prosody I would very much like know
more about the tests:
What kind of computers were the end users working?
What codecs were being used by the x-lite clients when connecting to FS?
I have a freeswitch server located behind 1:1 nat that I have been
using to test things with. Today I upgraded to the latest trunk and
began having troubles with my private ip being exposed in the sdp
instead of the value set with ext-rtp-ip in the sip_profile. Needless
to say all of my
I'm wondering if the default configs got messed up somehow. We need more
information.
Please do two things:
Pastebin your Local_Extension dialplan entry
Turn on debug level output (type console loglevel 7), make another call,
then pastebin the output
See this page for more tips on collecting
You need to set the local-network-acl on the profile too please refer
to the default configs.
/b
On Jul 29, 2009, at 11:30 AM, Dale wrote:
I have a freeswitch server located behind 1:1 nat that I have been
using to test things with. Today I upgraded to the latest trunk and
began having
Its because he's doing FS and one softphone on the same PC.
/b
On Jul 29, 2009, at 11:36 AM, Michael Collins wrote:
I'm wondering if the default configs got messed up somehow. We need
more information.
Please do two things:
Pastebin your Local_Extension dialplan entry
Turn on debug level
I have the same issue with my freeswitch server running on the Amazon EC2
instance, I set up the ext-rtp-ip address with the public routable IP
address
but the private IP address showing up in the SDP. When I checked my home
router's internet sessions the rtp sessions were built using Amazon EC2
Hi Fernando,
Greetings from Rio..!
It'd be interesting to understand more about these results - roughly
speaking, two conferencing systems using the same codecs, etc., ought to
perform pretty much identically, particularly with just a few callers.
I'd be interested to see a network packet dump
Hmm, just tried that by adding the
param name=local-network-acl value=localnet.auto/
to the profile. I also verified that the acl was working properly.
I just tried defaulting back to the stock sample configs and setting
the ext-rtp-ip to my public ip and I still see the private in the sdp
On Wed, Jul 29, 2009 at 9:40 AM, Brian West br...@freeswitch.org wrote:
Its because he's doing FS and one softphone on the same PC.
Naughty naughty!
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Are you dialing the echo test from a public IP or a private IP? btw
you need to set the ext-sip-ip too.
/b
On Jul 29, 2009, at 12:25 PM, Dale wrote:
Hmm, just tried that by adding the
param name=local-network-acl value=localnet.auto/
to the profile. I also verified that the acl was
Are you dialing the echo test from a public IP or a private IP? btw
Its a client natted behind a public ip. I don't have a pure public IP
to test from at the moment.
you need to set the ext-sip-ip too
Yep, got that set too.
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What do you mean? Is the client behind NAT? if so what client? and
where is the IP in the SDP? the one from the client or the one from
FreeSWITCH and what is the network layout of the scenario because that
is critical to understanding what the heck is going on here.
/b
On Jul 29, 2009,
The box is currently setup with the default config(and password) if
you want to try registering to it and placing a call.
FS (172.16.50.10) - NAT (Public IP 207.136.252.50) - INTERNET - NAT
(Public IP 64.9.127.104) - Zoiper(Soft Phone) (10.10.10.147)
Freeswitch offers SDP
v=0
o=FreeSWITCH
Are you on SVN trunk? I can tell you 100% that this has to be a
config error. Are you talking over the right profile?
/b
On Jul 29, 2009, at 1:10 PM, Dale wrote:
The box is currently setup with the default config(and password) if
you want to try registering to it and placing a call.
FS
I'm running FreeSWITCH Version 1.0.trunk (14412)
Okay, I'll keep hunting around.
-Dale
On Jul 29, 2009, at 2:21 PM, Brian West wrote:
Are you on SVN trunk? I can tell you 100% that this has to be a
config error. Are you talking over the right profile?
/b
On Jul 29, 2009, at 1:10 PM,
The error comes from a condition where the remote side of the socket has
disconnected before it can get the socket details to add to the log.
I added a patch to svn to increase the max conns to 10 from 16 that's
about all I can do
you may want to consider using ESL for some of the remote
Find me on IRC and let me in the box please.
/b
On Jul 29, 2009, at 1:32 PM, Dale wrote:
I'm running FreeSWITCH Version 1.0.trunk (14412)
Okay, I'll keep hunting around.
-Dale
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Hi,
Is there a way to terminate the C leg when using att_xfer if the C leg
ends up being a voicemail?
Thanks in advance
Adnan
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Still stumped on trying to write a streamlined pocketsphinx speech
recognition. I went through SpeechTools.jm and I think I pulled out
the necessary pieces, but the speech event is not being fired. I get
dtmf events if I push a key, but not speech. I must be missing a
piece, but I have
SpeechTools.jm is rather simple if you look at ps_pizza.js that uses
it.. you'll get a better feel for how that works. Have you looked at
the pizza demo javascript?
/b
On Jul 29, 2009, at 3:28 PM, Greg Thoen wrote:
Still stumped on trying to write a streamlined pocketsphinx speech
Hi. Yes, I have had that running, no problem.
I agree that SpeechTools as a class is easy to use, but I am trying to
write a smaller version that I can integrate more with what I am
doing; so I am trying to pull out pieces of it. In the example I
gave, I am concerned that I can't even get
http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts
/b
On Jul 29, 2009, at 4:05 PM, Greg Thoen wrote:
Hi. Yes, I have had that running, no problem.
I agree that SpeechTools as a class is easy to use, but I am trying
to write a smaller version that I can integrate more with what
On Mon, Jul 20, 2009 at 3:06 PM, Kristian
Kielhofnerkristian.kielhof...@gmail.com wrote:
I don't know of any new issues with Sonus as used by the carriers I
deal with but YMMV.
Wow. I just finished reading 3 long pages of super drama over at
Yahoo! Finance :)
So, it sounds like the FS team
Given the choice... I would avoid Sonus! But thats my personal opinion.
/b
On Jul 29, 2009, at 7:19 PM, Gabriel Gunderson wrote:
Wow. I just finished reading 3 long pages of super drama over at
Yahoo! Finance :)
So, it sounds like the FS team has made provisions to work with Sonus'
On Wed, Jul 29, 2009 at 6:30 PM, Brian Westbr...@freeswitch.org wrote:
Given the choice... I would avoid Sonus! But thats my personal opinion.
I don't get to make the decision, just support it :)
Gabe
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On Wed, Jul 29, 2009 at 8:19 PM, Gabriel Gundersong...@gundy.org wrote:
On Mon, Jul 20, 2009 at 3:06 PM, Kristian
Kielhofnerkristian.kielhof...@gmail.com wrote:
I don't know of any new issues with Sonus as used by the carriers I
deal with but YMMV.
Wow. I just finished reading 3 long pages
David Knell wrote:
Hi Fernando,
Greetings from Rio..!
It'd be interesting to understand more about these results - roughly
speaking, two conferencing systems using the same codecs, etc., ought to
perform pretty much identically, particularly with just a few callers.
I'd be interested to
Yeah, I have a softphone on the FS computer for tests, guessing this
is bad?Sorry,
complete noob at FS =P
On Thu, Jul 30, 2009 at 3:28 AM, Michael Collins m...@freeswitch.org wrote:
On Wed, Jul 29, 2009 at 9:40 AM, Brian West br...@freeswitch.org wrote:
Its because he's doing FS and one
On Wed, Jul 29, 2009 at 6:48 PM, Kristian
Kielhofnerkristian.kielhof...@gmail.com wrote:
On Wed, Jul 29, 2009 at 8:19 PM, Gabriel Gundersong...@gundy.org wrote:
On Mon, Jul 20, 2009 at 3:06 PM, Kristian
Kielhofnerkristian.kielhof...@gmail.com wrote:
I don't know of any new issues with Sonus
Its usually not a supported scenario!
/b
On Jul 29, 2009, at 9:45 PM, Drew Hopcroft wrote:
Yeah, I have a softphone on the FS computer for tests, guessing this
is bad?
Sorry, complete noob at FS =P
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SELL SELL SELL!!! :P
/b
On Jul 29, 2009, at 9:44 PM, Gabriel Gunderson wrote:
Dude. Don't. I now own SONS stock and I don't want to make waves ;)
Gabe
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thanks!
On Thu, Jul 30, 2009 at 12:52 PM, Brian West br...@freeswitch.org wrote:
Its usually not a supported scenario!
/b
On Jul 29, 2009, at 9:45 PM, Drew Hopcroft wrote:
Yeah, I have a softphone on the FS computer for tests, guessing this
is bad?
Sorry, complete noob at FS =P
Have you got my questions.?
Using ESL connection object some of the function or subroutines or methods
has bee called. Is there any way to find out all the function names and its
functionality.
Could you please help me?
-- Forwarded message --
From: Thangappan.M
HAHA found you on twitter! :P tweet tweet!
/b
On Jul 29, 2009, at 9:44 PM, Gabriel Gunderson wrote:
Dude. Don't. I now own SONS stock and I don't want to make waves ;)
Gabe
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In the header files?
On Jul 29, 2009, at 11:11 PM, Thangappan.M thangappan...@gmail.com
wrote:
Have you got my questions.?
Using ESL connection object some of the function or subroutines or
methods has bee called. Is there any way to find out all the
function names and its
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