[Freeswitch-users] Methods in the ESL connection
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. So here is my question, where can I get all the methods of ESL connection and its explanation.This methods will be useful for me to implement my own IVR. -- Regards, Thangappan.M ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_managed users?
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 Registering API FullName with Aliases fullname,
shortname.
I'm just using the Demo.cs example, I compile it to dll undef VC#, not
mono, maybe that is the difference?
The output from the log is just as you stated Registering API
FullName with Aliases fullname, shortname. The difference between
loading dll and csx is that, when loading csx all api and app classes
are listed as registered, while with dll nothing is listed..
Anyway, I have another question regarding usage of the CoreSession and
ManagedSession object.
Basically in my script I want to start new session and originate a call.
So what I do is
ManageSession session = new ManagedSession();
session.Originate(session,sofia/default/1000,10);
And it works but I have some doubts. First thing is, why the first
param of the Originate method is the CoreSession object? Can't it just
use 'this'? Or is there more to it?
Second thing is the third parameter - timeout in seconds. Can't the
call be started in non-blocking mode? I can start in a different
thread of course, is that the intended behavior?
Thanks for help,
Łukasz
2009/7/28 Michael Giagnocavo m...@giagnocavo.net:
Hello Lukasz,
Thanks for testing mod_managed. I apologize for the problems you've
encountered, and I'll try to sort them out for you.
A few things first:
- Scripting support: This is made to allow true scripts, as invoked
as an EXE - similar to the Lua and spidermonkey support. So, without a
Main(), it won't compile as an EXE. If you aren't using it as a script, then
an empty Main method will work fine.
- Entry points must be public for Mono. I'll update the demo code to
make sure that Main is public. This is a bug in Mono's lightweight code
generation -- it won't skip the JIT access checks.
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 Registering API FullName with Aliases fullname,
shortname.
Thanks,
Michael
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lukasz
Zwierko
Sent: Tuesday, July 28, 2009 1:26 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_managed users?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I've just tried new mod_managed under Win32 and I get a weird behavior.
I try the example below:
public class DemoScript : IApiPlugin
{
public void Execute(ApiContext context)
{
context.Stream.Write(string.Format(DemoScripts executed with
args '{0}' and event type {1}.,
context.Arguments, context.Event == null ? none :
context.Event.GetEventType()));
}
public void ExecuteBackground(ApiBackgroundContext context)
{
Log.WriteLine(LogLevel.Notice, DemoScripts on a background
thread #({0}), with args '{1}'.,
System.Threading.Thread.CurrentThread.ManagedThreadId,
context.Arguments);
}
}
It's just like the ApiDemo from Demo.cs
So When I copy DemoScript.csx to managed dir the console log is:
freeswi...@zwierko-laptop Loading
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx from
domain DemoScripts.csx_3
2009-07-28 20:57:16.71 [INFO] switch_cpp.cpp:1130 Compiling
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx
2009-07-28 20:57:16.97 [ERR] switch_cpp.cpp:1130 There were 1 errors
compiling
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx.
2009-07-28 20:57:16.97 [ERR] switch_cpp.cpp:1130 CS5001: Program
'c:\Users\Zwierko\AppData\Local\Temp\fc8jnlir.exe' does not contain a
static 'Main' method
suitable for an entry point
Adding
public static void Main()
{
}
solves the issue. Is this how it's supposed to work?
Another strange thing is that when I compile this class to DLL (release)
it does not work at all...
freeswi...@zwierko-laptop Loading
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\FSScripts.dll from
domain FSScripts.dll_5
freeswi...@zwierko-laptop
freeswi...@zwierko-laptop managed DemoScript 111
API CALL [managed(DemoScript 111)] output:
2009-07-28 21:13:03.542000 [ERR] switch_cpp.cpp:1130 API plugin
DemoScript not found.
2009-07-28 21:13:03.542000 [ERR] mod_managed.cpp:393 Execute failed for
DemoScript 111 (unknown module or exception).
And another issue with scripts. I use script code as example:
public class ScriptDemo
{
static void Main()
{
switch
Re: [Freeswitch-users] mod_managed users?
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);
As to non-blocking, I'm quite sure it's possible, but I don't recall
offhand which functions. This should be the same as in any other language for
FreeSWITCH -- these functions are just passthrough from the FS C++ API.
-Michael
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lukasz
Zwierko
Sent: Wednesday, July 29, 2009 12:13 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_managed users?
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 Registering API FullName with Aliases fullname,
shortname.
I'm just using the Demo.cs example, I compile it to dll undef VC#, not
mono, maybe that is the difference?
The output from the log is just as you stated Registering API
FullName with Aliases fullname, shortname. The difference between
loading dll and csx is that, when loading csx all api and app classes
are listed as registered, while with dll nothing is listed..
Anyway, I have another question regarding usage of the CoreSession and
ManagedSession object.
Basically in my script I want to start new session and originate a call.
So what I do is
ManageSession session = new ManagedSession();
session.Originate(session,sofia/default/1000,10);
And it works but I have some doubts. First thing is, why the first
param of the Originate method is the CoreSession object? Can't it just
use 'this'? Or is there more to it?
Second thing is the third parameter - timeout in seconds. Can't the
call be started in non-blocking mode? I can start in a different
thread of course, is that the intended behavior?
Thanks for help,
Łukasz
2009/7/28 Michael Giagnocavo m...@giagnocavo.net:
Hello Lukasz,
Thanks for testing mod_managed. I apologize for the problems you've
encountered, and I'll try to sort them out for you.
A few things first:
- Scripting support: This is made to allow true scripts, as invoked
as an EXE - similar to the Lua and spidermonkey support. So, without a
Main(), it won't compile as an EXE. If you aren't using it as a script, then
an empty Main method will work fine.
- Entry points must be public for Mono. I'll update the demo code to
make sure that Main is public. This is a bug in Mono's lightweight code
generation -- it won't skip the JIT access checks.
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 Registering API FullName with Aliases fullname,
shortname.
Thanks,
Michael
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lukasz
Zwierko
Sent: Tuesday, July 28, 2009 1:26 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_managed users?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I've just tried new mod_managed under Win32 and I get a weird behavior.
I try the example below:
public class DemoScript : IApiPlugin
{
public void Execute(ApiContext context)
{
context.Stream.Write(string.Format(DemoScripts executed with
args '{0}' and event type {1}.,
context.Arguments, context.Event == null ? none :
context.Event.GetEventType()));
}
public void ExecuteBackground(ApiBackgroundContext context)
{
Log.WriteLine(LogLevel.Notice, DemoScripts on a background
thread #({0}), with args '{1}'.,
System.Threading.Thread.CurrentThread.ManagedThreadId,
context.Arguments);
}
}
It's just like the ApiDemo from Demo.cs
So When I copy DemoScript.csx to managed dir the console log is:
freeswi...@zwierko-laptop Loading
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx from
domain DemoScripts.csx_3
2009-07-28 20:57:16.71 [INFO] switch_cpp.cpp:1130 Compiling
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx
2009-07-28 20:57:16.97 [ERR] switch_cpp.cpp:1130 There were 1 errors
compiling
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx.
2009-07-28 20:57:16.97 [ERR] switch_cpp.cpp:1130 CS5001: Program
'c:\Users\Zwierko\AppData\Local\Temp\fc8jnlir.exe' does not contain a
static 'Main' method
suitable for an entry
[Freeswitch-users] multiple commands in execute_on_answer
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 --- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] multiple commands in execute_on_answer
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 want to execute both a python script AND the sched_hangup application. -- --- Apostolos Pantsiopoulos Kinetix Tele.com R D email: r...@kinetix.gr --- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] freeswitch_http.log - format
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 the webapi. Also, I'm seeing this error popup in my freeswitch_http.log ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - freeswi...@127.0.0.1 - [29/Jul/2009:00:32:42 +] POST 200 17 ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - freeswi...@127.0.0.1 - [29/Jul/2009:00:32:42 +] POST 200 17 ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - freeswi...@127.0.0.1 - [29/Jul/2009:00:32:42 +] POST 200 17 and I think it's related to some problems I'm having with my program. A few other users http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg09225.html and http://jira.freeswitch.org/browse/MDXMLINT-28 noted similar items appearing in their logs, but I could not find a definitive solution. Does anyone have any solutions to this error? When this error appears in my freeswitch_http.log, all webapi commands seem to block, and than rapidly get executed all at once, whenever the block is released. I'm using 14163. The errors appear to happen only once a certain load level is reached, so I'm having trouble reproducing it consistently. Could this be caused by an xmlrpc request closing the socket connection before FreeSWITCH has a chance to respond? Can anyone recommend any better ways for me to diagnose this issue? Thanks. I'm using the XMLRPC with Ruby on Rails (REE) and Passenger. --matt ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] multiple commands in execute_on_answer
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 Fong wrote: 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.gr mailto:r...@kinetix.gr wrote: 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 mailto:r...@kinetix.gr --- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto:FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R D email: r...@kinetix.gr --- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem.
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 PBX? Brian West a écrit : The remote end said 404 /b On Jul 28, 2009, at 10:00 AM, julien wrote: 2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel sofia/external/300 entering state [terminated][404] ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] SIP-T / RFC 3372 support
Hi All Does Freeswitch support SIP-T / RFC 3372 ? Regards, Maxim Tsvetov ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem in mod_perl
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 wrote: It is not built by default because it requires manual intervention to make sure you have a proper threadsafe perl and all its dev libs installed first. We work hard to make sure all default modules build out of the box with minimal external dependencies. Also, this module still does not work 100% on some platforms (solaris?) Mike On Jul 24, 2009, at 11:36 AM, Shawn Boyle wrote: Did you also uncomment the line: languages/mod_perl in modules.conf when you compiled FS? I believe it's commented out by default. [Something I personally disagree with...but I would bear Larry Wall's children if I could manage it physiologically.] -Shawn ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org PGP.sig Description: This is a digitally signed message part ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] FS beats Aculab Prosody S on subjective test on lay users for conference quality
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 for 10 minutes from their x-lite and report the results. These users are non-techie and never heard about FS at all. The first url was Aculab Prosody S test app, then, a second call to FS on default conf (extension 3000). All users preferred FS. Some users were almost enthusiastic after entering FS conf. Some even started to blame ProsodyS after hearing FS. The average report told 50% more satisfaction on FS. Questions on report were simple ones, with results from 1(worse) to 5(best) : 1-Overall opinion(1-crap; 5-awesome); 2-Delay (1-more delay, 5-no delay at all); 3-Audio quality (1-noisy, choppy, etc; 5-cristal clear); Machines used: Prosody S: Windows Vista Business, Intel Core 2 Duo E7400 @ 2.8 GHz, 4Gb de RAM FreeSwitch: CentOS 5.4, Intel(R) Pentium(R) 4 CPU 2.40GHz, 1Gb RAM Congratulations! You made a great product! -- Fernando Gregianin Testa Voice Technology Ltda +55 11 35882166 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Help setting up FreeSWITCH
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 work it stopped). No idea what to do and hoping for some advice? The output I am getting is: (broken it up a little) sofia status API CALL [sofia(status)] output: Name Type Data State = internal profile sip:mod_so...@10.10.0.254:5060 RUNNING (0) external profile sip:mod_so...@10.10.0.254:5080 RUNNING (0) example.com gateway sip:joeu...@example.comsip%3ajoeu...@example.com NOREG iinetphone.iinet.net.au gateway sip:03x...@sip.vic.iinet.net.au sip%3a03x...@sip.vic.iinet.net.au REGED 10.10.0.254 alias internal ALIASED default alias internal ALIASED nat alias external ALIASED outbound alias external ALIASED = 2 profiles 4 aliases sofia status profile internal API CALL [sofia(status profile internal)] output: = Nameinternal Domain Name N/A DBName sofia_reg_internal Pres Hosts DialplanXML Context public Challenge Realm auto_from RTP-IP 10.10.0.254 SIP-IP 10.10.0.254 URL sip:mod_so...@10.10.0.254:5060 BIND-URLsip:mod_so...@10.10.0.254:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN3 FAILED-CALLS-IN 1 CALLS-OUT 0 FAILED-CALLS-OUT0 Registrations: = Call-ID:ZjRkMmIyZjYwZGI0NjMzYWEzZTRhY2M0Nzk0N2JjYTE. User: 1...@10.10.0.254 Contact:SERV ER sip:1...@10.10.0.254:32022 ;rinstance=9f1b2d2e32fdd2dc;fs_nat=yes;fs_path=sip%3A1001%4010 .10.0.254%3A32022%3Brinstance%3D9f1b2d2e32fdd2dc Agent: X-Lite Beta release 4.0 Beta stamp 54292 Status: Registered(UDP-NAT)(unknown) EXP(2009-07-29 14:54:10) Host: EIN-Server IP: 10.10.0.254 Port: 32022 Auth-User: 1001 Auth-Realm: 10.10.0.254 Call-ID:OWE3ZDc2ZWYwZGU4M2Y2ZGQ1YzQxYTMxODBiMDAwMDY. User: 1...@10.10.0.254 Contact:Drew sip:1...@10.10.1.4:47894 ;rinstance=45362e614cd99099;fs_nat=yes;fs_path=sip%3A1015%4010.10.1 .4%3A47894%3Brinstance%3D45362e614cd99099 Agent: X-Lite release 1103d stamp 53117 Status: Registered(UDP-NAT)(unknown) EXP(2009-07-29 13:04:13) Host: EIN-Server IP: 10.10.1.4 Port: 47894 Auth-User: 1015 Auth-Realm: 10.10.0.254 = Calls Inside the LAN 2009-07-29 12:54:17.426580 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@10. 10.0.254 [f6e84bb4-3037-d943-98f7-974b5a9569d7] 2009-07-29 12:54:17.440580 [INFO] mod_dialplan_xml.c:252 Processing Drew-1001 in context public 2009-07-29 12:54:17.441580 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/1...@10.10.0.254 to xml[1...@default] 2009-07-29 12:54:17.442580 [INFO] mod_dialplan_xml.c:252 Processing Drew-1001 in context default 2009-07-29 12:54:17.445580 [WARNING] switch_ivr.c:2046 can't find user [defa...@] 2009-07-29 12:54:18.566580 [INFO] switch_ivr_async.c:1793 Bound B-Leg: 1 execute_extension::dx XML features 2009-07-29 12:54:18.566580 [INFO] switch_ivr_async.c:1793 Bound B-Leg: 2 record_session::C:\Program Files (x86)\Fre eSWITCH/recordings/1015.2009-07-29-12-54-18.wav 2009-07-29 12:54:18.567580 [INFO] switch_ivr_async.c:1793 Bound B-Leg: 3 execute_extension::cf XML features 2009-07-29 12:54:19.549580 [WARNING] mod_dptools.c:2363 Can't find user [1...@] 2009-07-29 12:54:19.549580 [ERR] switch_ivr_originate.c:1494 Cannot create outgoing channel of type [user] cause: [ SUBSCRIBER_ABSENT] 2009-07-29
Re: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem.
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 reachable but there's a problem of communication between FS and the PBX? A 404 in SIP is just like a 404 when web surfing: the target server can't find whatever it is that you're looking for. In other words, your FS server made contact with the server at the far end, told it what endpoint you're looking for, and the server there said, I can't find this endpoint, sorry. You'll need to see what's happening at the far end, like whether there really is a 300 or not. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] multiple commands in execute_on_answer
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 sched_hangup on the a leg (with the help of nolocal:) Thanx for the help anyway. You can also do something like this: action application=set data=execute_on_answer=execute_extension HANDLE_ANSWER/ And then define your answer handler extension: extension name=Handle Answered Calls condition field=destination_number expression=^HANDLE_ANSWER$/ action application=python data=/path/to/my/python/script.py/ action application=sched_hangup data=+300 TIMED_OUT/ action application=log data=INFO You can do all sorts of stuff/ !-- If this extension was called with 'execute_extension' then it will go back to the calling extension when done... -- /condition /extension Just a thought... -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS beats Aculab Prosody S on subjective test on lay users for conference quality
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? What codecs were being used by the x-lite clients when connecting to Prosody? Were there any other factors that could possibly have affected the call quality? The reason I ask all of these is that I would love to put this account into the testimonials page on the wiki and make it available to other techie guys who are trying to make their business cases for using FreeSWITCH (or FOSS in general) to the corporate executives. Thanks for the great report! -Michael On Wed, Jul 29, 2009 at 7:53 AM, Fernando Testa te...@voicetechnology.com.br wrote: 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 for 10 minutes from their x-lite and report the results. These users are non-techie and never heard about FS at all. The first url was Aculab Prosody S test app, then, a second call to FS on default conf (extension 3000). All users preferred FS. Some users were almost enthusiastic after entering FS conf. Some even started to blame ProsodyS after hearing FS. The average report told 50% more satisfaction on FS. Questions on report were simple ones, with results from 1(worse) to 5(best) : 1-Overall opinion(1-crap; 5-awesome); 2-Delay (1-more delay, 5-no delay at all); 3-Audio quality (1-noisy, choppy, etc; 5-cristal clear); Machines used: Prosody S: Windows Vista Business, Intel Core 2 Duo E7400 @ 2.8 GHz, 4Gb de RAM FreeSwitch: CentOS 5.4, Intel(R) Pentium(R) 4 CPU 2.40GHz, 1Gb RAM Congratulations! You made a great product! -- Fernando Gregianin Testa Voice Technology Ltda +55 11 35882166 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] ext-rtp-ip Troubles
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 audio is broken now. :) I'm working on getting the details together for a jira but figured I would see if anyone else has noticed anything like this. -Dale ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help setting up FreeSWITCH
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 information and reporting it: http://wiki.freeswitch.org/wiki/Reporting_Bugs -MC On Tue, Jul 28, 2009 at 10:55 PM, Drew Hopcroft hopcroft.d...@gmail.comwrote: 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 work it stopped). No idea what to do and hoping for some advice? The output I am getting is: (broken it up a little) sofia status API CALL [sofia(status)] output: Name Type Data State = internal profile sip:mod_so...@10.10.0.254:5060 RUNNING (0) external profile sip:mod_so...@10.10.0.254:5080 RUNNING (0) example.com gateway sip:joeu...@example.comsip%3ajoeu...@example.com NOREG iinetphone.iinet.net.au gateway sip:03x...@sip.vic.iinet.net.ausip%3a03x...@sip.vic.iinet.net.au REGED 10.10.0.254 alias internal ALIASED default alias internal ALIASED nat alias external ALIASED outbound alias external ALIASED = 2 profiles 4 aliases sofia status profile internal API CALL [sofia(status profile internal)] output: = Nameinternal Domain Name N/A DBName sofia_reg_internal Pres Hosts DialplanXML Context public Challenge Realm auto_from RTP-IP 10.10.0.254 SIP-IP 10.10.0.254 URL sip:mod_so...@10.10.0.254:5060 BIND-URLsip:mod_so...@10.10.0.254:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN3 FAILED-CALLS-IN 1 CALLS-OUT 0 FAILED-CALLS-OUT0 Registrations: = Call-ID:ZjRkMmIyZjYwZGI0NjMzYWEzZTRhY2M0Nzk0N2JjYTE. User: 1...@10.10.0.254 Contact:SERV ER sip:1...@10.10.0.254:32022 ;rinstance=9f1b2d2e32fdd2dc;fs_nat=yes;fs_path=sip%3A1001%4010 .10.0.254%3A32022%3Brinstance%3D9f1b2d2e32fdd2dc Agent: X-Lite Beta release 4.0 Beta stamp 54292 Status: Registered(UDP-NAT)(unknown) EXP(2009-07-29 14:54:10) Host: EIN-Server IP: 10.10.0.254 Port: 32022 Auth-User: 1001 Auth-Realm: 10.10.0.254 Call-ID:OWE3ZDc2ZWYwZGU4M2Y2ZGQ1YzQxYTMxODBiMDAwMDY. User: 1...@10.10.0.254 Contact:Drew sip:1...@10.10.1.4:47894 ;rinstance=45362e614cd99099;fs_nat=yes;fs_path=sip%3A1015%4010.10.1 .4%3A47894%3Brinstance%3D45362e614cd99099 Agent: X-Lite release 1103d stamp 53117 Status: Registered(UDP-NAT)(unknown) EXP(2009-07-29 13:04:13) Host: EIN-Server IP: 10.10.1.4 Port: 47894 Auth-User: 1015 Auth-Realm: 10.10.0.254 = Calls Inside the LAN 2009-07-29 12:54:17.426580 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@10. 10.0.254 [f6e84bb4-3037-d943-98f7-974b5a9569d7] 2009-07-29 12:54:17.440580 [INFO] mod_dialplan_xml.c:252 Processing Drew-1001 in context public 2009-07-29 12:54:17.441580 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/1...@10.10.0.254 to xml[1...@default] 2009-07-29 12:54:17.442580 [INFO] mod_dialplan_xml.c:252 Processing Drew-1001 in context default 2009-07-29 12:54:17.445580 [WARNING] switch_ivr.c:2046 can't find user [defa...@] 2009-07-29
Re: [Freeswitch-users] ext-rtp-ip Troubles
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 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 audio is broken now. :) I'm working on getting the details together for a jira but figured I would see if anyone else has noticed anything like this. -Dale ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help setting up FreeSWITCH
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 output (type console loglevel 7), make another call, then pastebin the output See this page for more tips on collecting information and reporting it: http://wiki.freeswitch.org/wiki/Reporting_Bugs -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ext-rtp-ip Troubles
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 private IP address. Chris On Wed, Jul 29, 2009 at 12:30 PM, Dale fdh...@gmail.com 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 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 audio is broken now. :) I'm working on getting the details together for a jira but figured I would see if anyone else has noticed anything like this. -Dale ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS beats Aculab Prosody S on subjective test on lay users for conference quality
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 of a conference on each of the machines, if you're able to make one available. Cheers -- Dave 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 for 10 minutes from their x-lite and report the results. These users are non-techie and never heard about FS at all. The first url was Aculab Prosody S test app, then, a second call to FS on default conf (extension 3000). All users preferred FS. Some users were almost enthusiastic after entering FS conf. Some even started to blame ProsodyS after hearing FS. The average report told 50% more satisfaction on FS. Questions on report were simple ones, with results from 1(worse) to 5(best) : 1-Overall opinion(1-crap; 5-awesome); 2-Delay (1-more delay, 5-no delay at all); 3-Audio quality (1-noisy, choppy, etc; 5-cristal clear); Machines used: Prosody S: Windows Vista Business, Intel Core 2 Duo E7400 @ 2.8 GHz, 4Gb de RAM FreeSwitch: CentOS 5.4, Intel(R) Pentium(R) 4 CPU 2.40GHz, 1Gb RAM Congratulations! You made a great product! -- Fernando Gregianin Testa Voice Technology Ltda +55 11 35882166 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ext-rtp-ip Troubles
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 when I dial the echo test. -Dale On Jul 29, 2009, at 12:39 PM, Brian West wrote: 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 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 audio is broken now. :) I'm working on getting the details together for a jira but figured I would see if anyone else has noticed anything like this. -Dale ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help setting up FreeSWITCH
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! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ext-rtp-ip Troubles
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 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 when I dial the echo test. -Dale ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ext-rtp-ip Troubles
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. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ext-rtp-ip Troubles
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, at 12:52 PM, Dale wrote: 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. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ext-rtp-ip Troubles
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 1248869138 1248869139 IN IP4 172.16.50.10 s=FreeSWITCH c=IN IP4 172.16.50.10 t=0 0 m=audio 20054 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Zoiper offers SDP v=0 o=Z 0 0 IN IP4 64.9.127.104 s=Z c=IN IP4 64.9.127.104 t=0 0 m=audio 16000 RTP/AVP 110 98 0 8 3 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 On Jul 29, 2009, at 1:55 PM, Brian West wrote: 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, at 12:52 PM, Dale wrote: 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. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ext-rtp-ip Troubles
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 (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 1248869138 1248869139 IN IP4 172.16.50.10 s=FreeSWITCH c=IN IP4 172.16.50.10 t=0 0 m=audio 20054 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Zoiper offers SDP v=0 o=Z 0 0 IN IP4 64.9.127.104 s=Z c=IN IP4 64.9.127.104 t=0 0 m=audio 16000 RTP/AVP 110 98 0 8 3 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ext-rtp-ip Troubles
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, 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 (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 1248869138 1248869139 IN IP4 172.16.50.10 s=FreeSWITCH c=IN IP4 172.16.50.10 t=0 0 m=audio 20054 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Zoiper offers SDP v=0 o=Z 0 0 IN IP4 64.9.127.104 s=Z c=IN IP4 64.9.127.104 t=0 0 m=audio 16000 RTP/AVP 110 98 0 8 3 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch_http.log - format
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 control needs as it may scale better since we wrote all of that code and we only did our best to hack up this xmlrpc mod. On Wed, Jul 29, 2009 at 2:42 AM, Matthew Fong mattdf...@gmail.com wrote: 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 the webapi. Also, I'm seeing this error popup in my freeswitch_http.log ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - freeswi...@127.0.0.1 - [29/Jul/2009:00:32:42 +] POST 200 17 ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - freeswi...@127.0.0.1 - [29/Jul/2009:00:32:42 +] POST 200 17 ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - freeswi...@127.0.0.1 - [29/Jul/2009:00:32:42 +] POST 200 17 and I think it's related to some problems I'm having with my program. A few other users http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg09225.html and http://jira.freeswitch.org/browse/MDXMLINT-28 noted similar items appearing in their logs, but I could not find a definitive solution. Does anyone have any solutions to this error? When this error appears in my freeswitch_http.log, all webapi commands seem to block, and than rapidly get executed all at once, whenever the block is released. I'm using 14163. The errors appear to happen only once a certain load level is reached, so I'm having trouble reproducing it consistently. Could this be caused by an xmlrpc request closing the socket connection before FreeSWITCH has a chance to respond? Can anyone recommend any better ways for me to diagnose this issue? Thanks. I'm using the XMLRPC with Ruby on Rails (REE) and Passenger. --matt ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ext-rtp-ip Troubles
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Canceling att_xfer?
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Stumped on speech recognition
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 stared at SpeechTools.jm until I can't see straight.
Here is my code:
function onInput(s, type, inputEvent, _this) {
console_log(debug, EVENT: + type + \n);
}
session.answer();
var blankobj = new Object();
var rv;
var hit;
var dup;
var grammar_name=pizza_yesno;
var collected_index=0;
var req=1;
var obj_path=result;
var halt=true;
var min_score=20;
var confirm_score=10;
console_log('INFO', Setting grammar\n);
session.execute(detect_speech, nogrammar + grammar_name);
session.execute(detect_speech, pocketsphinx + grammar_name);
session.execute(detect_speech, resume);
console_log('INFO', entering while loop\n);
session.streamFile( audiofile.wav, onInput, blankobj) ;
while(session.ready() collected_index req) {
var x;
if (!rv) {
rv = session.collectInput(onInput, blankobj, 500);
if (rv !rv[0]) {
rv = false;
}
}
if (!rv) {
session.execute(detect_speech, resume);
rv = session.collectInput(onInput, blankobj, 5000);
}
hit = false;
if (rv) {
var items = rv;
rv = undefined;
for (y = 0; y items.length; y++) {
console_log(debug, +items[y]+\n);
}
}
if (!rv) {
rv = session.collectInput(onInput, blankobj, 1000);
}
}
And here is some of the output from the console:
2009-07-29 16:10:09 [INFO] stest-examp.js:1 console_log() Setting
grammar
EXECUTE sofia/internal/+...@199.173.94.88:5060 detect_speech(nogrammar
pizza_yesno)
EXECUTE sofia/internal/+...@199.173.94.88:5060
detect_speech(pocketsphinx pizza_yesno)
EXECUTE sofia/internal/+...@199.173.94.88:5060 detect_speech(resume)
2009-07-29 16:10:09 [INFO] stest-examp.js:1 console_log() entering
while loop
2009-07-29 16:10:09 [DEBUG] switch_ivr_play_say.c:1084
switch_ivr_play_file() Codec Activated l...@8000hz 1 channels 20ms
2009-07-29 16:10:09 [DEBUG] switch_core_io.c:649
switch_core_session_write_frame() sofia/internal/+...@199.173.94.88:5060
receive message [TRANSCODING_NECESSARY]
2009-07-29 16:10:09 [DEBUG] sofia.c:2979 sofia_handle_sip_i_state()
Channel sofia/internal/+...@199.173.94.88:5060 entering state [ready][200]
2009-07-29 16:10:12 [DEBUG] switch_ivr_play_say.c:1379
switch_ivr_play_file() done playing file
EXECUTE sofia/internal/+...@199.173.94.88:5060 detect_speech(resume)
2009-07-29 16:10:16 [DEBUG] switch_rtp.c:1876
switch_rtp_dequeue_dtmf() RTP RECV DTMF 5:2440
2009-07-29 16:10:16 [DEBUG] stest-examp.js:45 console_log() EVENT:dtmf
EXECUTE sofia/internal/+...@199.173.94.88:5060 detect_speech(resume)
EXECUTE sofia/internal/+...@199.173.94.88:5060 detect_speech(resume)
2009-07-29 16:10:27 [DEBUG] switch_rtp.c:1876
switch_rtp_dequeue_dtmf() RTP RECV DTMF 5:2320
2009-07-29 16:10:27 [DEBUG] stest-examp.js:45 console_log() EVENT:dtmf
Greg
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Re: [Freeswitch-users] Stumped on speech recognition
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 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 stared at SpeechTools.jm until I can't see straight. Here is my code: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Stumped on speech recognition
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 my onInput to see the event. I feel I must be missing some step in my code. -- Greg Thoen, Vice President CGI Communications, Inc. 1-585-427-0020 x260 On Jul 29, 2009, at 4:34 PM, Brian West wrote: 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 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 stared at SpeechTools.jm until I can't see straight. Here is my code: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Stumped on speech recognition
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 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 my onInput to see the event. I feel I must be missing some step in my code. -- Greg Thoen, Vice President CGI Communications, Inc. 1-585-427-0020 x260 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sonus - what's the latest?
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 has made provisions to work with Sonus' borken RTP *and* if there are still issues, most of them can be overcome by explicitly setting some variables in the FS config. Does that sound like an accurate summary of the situation? Thanks, Gabe ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sonus - what's the latest?
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' borken RTP *and* if there are still issues, most of them can be overcome by explicitly setting some variables in the FS config. Does that sound like an accurate summary of the situation? Thanks, Gabe ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sonus - what's the latest?
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sonus - what's the latest?
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 of super drama over at Yahoo! Finance :) Ha, I can't believe you, um, wasted your time with that! How are those guys doing? We should drop in and see! So, it sounds like the FS team has made provisions to work with Sonus' borken RTP *and* if there are still issues, most of them can be overcome by explicitly setting some variables in the FS config. Does that sound like an accurate summary of the situation? That's basically it... -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS beats Aculab Prosody S on subjective test on lay users for conference quality
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 see a network packet dump of a conference on each of the machines, if you're able to make one available. High quality conferencing is a difficult task, and still a research topic. No two conferencing systems perform alike. The interesting thing about this and other reports is that the conferencing in Freeswitch is not very clever right now, yet people are already saying it beats various other offerings, including long time commercial offerings. Steve ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help setting up FreeSWITCH
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 softphone on the same PC. Naughty naughty! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sonus - what's the latest?
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 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 :) Ha, I can't believe you, um, wasted your time with that! How are those guys doing? We should drop in and see! Dude. Don't. I now own SONS stock and I don't want to make waves ;) Gabe ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help setting up FreeSWITCH
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sonus - what's the latest?
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help setting up FreeSWITCH
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Fwd: Methods in the ESL connection
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 thangappan...@gmail.com Date: Wed, Jul 29, 2009 at 11:41 AM Subject: Methods in the ESL connection To: freeswitch-users freeswitch-users@lists.freeswitch.org 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. So here is my question, where can I get all the methods of ESL connection and its explanation.This methods will be useful for me to implement my own IVR. -- Regards, Thangappan.M -- Regards, Thangappan.M ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sonus - what's the latest?
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Fwd: Methods in the ESL connection
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 functionality. Could you please help me? -- Forwarded message -- From: Thangappan.M thangappan...@gmail.com Date: Wed, Jul 29, 2009 at 11:41 AM Subject: Methods in the ESL connection To: freeswitch-users freeswitch-users@lists.freeswitch.org 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. So here is my question, where can I get all the methods of ESL connection and its explanation.This methods will be useful for me to implement my own IVR. -- Regards, Thangappan.M -- Regards, Thangappan.M ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
