From scaram at hotmail.de  Fri Apr  2 05:04:11 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Fri, 2 Apr 2010 14:04:11 +0200
Subject: [Freeswitch-dev] hooked functions are not called while using
 one single session [SOLVED]
In-Reply-To: <191c3a031003261747q76feb4d5p7e44d80988010c64@mail.gmail.com>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>,
	<SNT133-w324431DD601582973573FEA0270@phx.gbl>,
	<191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>,
	<SNT133-w55A5A011A53DC01B04C610A0230@phx.gbl>,
	<191c3a031003260931k2f10e495q1ad1e0b7e76eaf19@mail.gmail.com>,
	<SNT133-w377811CC94108A22C47B3AA0230@phx.gbl>,
	<191c3a031003261037r27e53b58qfd3a46051f730370@mail.gmail.com>,
	<SNT133-w36DF971B966B882126F3B8A0230@phx.gbl>,
	<191c3a031003261107y794010b4u3d434ba856e3e61e@mail.gmail.com>,
	<SNT133-w202C90975089CB4AE4C743A0220@phx.gbl>,
	<191c3a031003261747q76feb4d5p7e44d80988010c64@mail.gmail.com>
Message-ID: <SNT133-w63C112A42FC6E93B477DA8A01C0@phx.gbl>


I solved it. My implementation was not the problem but my softphone had acoustic echo cancellation (AEC) setting enabled which caused the problem (even I use an headset).  I didn't know this feature before so it was hard to find out..

Date: Fri, 26 Mar 2010 19:47:01 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session


The code you are looking at for playback and record has also changed the format of the channel to signed linear before writing, you are tyring to mimic that same thing without setting the read codec.

you should be treating the input as unsigned char and writing the full frame->datalen into your file.


you also may want to not open and close the file every 20ms since that can't be good on the OS.

I hate to break it to you, but if you look around at all the code in FS, it's safe to assume that the code to read and write media is well tested, the more you start understanding that you need to learn more instead of thinking something is wrong with the code, the sooner you will find what you are looking for.


You might want to try irc in #freeswitch-dev or #freeswitch on irc.freenode.net for a little more help since most people don't check this list for someone asking to learn how to program audio code with FS.







On Fri, Mar 26, 2010 at 7:26 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:






Function switch_ivr_session_echo is doing the same thing as I do in my test, and it seems that something doesn't work correctly.
If you just edit switch_ivr_session_echo and add 3 lines of code for logging audio data into a file, you can listen to the audio gaps. 


To reproduce the gaps it's the best to read a text for about 60 seconds, then you will hear it.

For the test I edited the dialplan:

 <extension name="testit22">
      <condition field="destination_number" expression="47112">

         <action application="answer"/>
         <action application="echo"/>
      </condition>
  </extension>
    
.. and the code..

SWITCH_DECLARE(void) switch_ivr_session_echo(switch_core_session_t *session, switch_input_args_t *args)

{
        FILE *tmp = NULL;
...
...
        status = switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);
        if (!SWITCH_READ_ACCEPTABLE(status)) {

            break;
        }
        /*******************************************************************************************************************/
        /*      TEST CODE FOR WRITING AUDIO INTO FILE , IF RECORD 1 MINUTE AND LISTEN TO THE FILE YOU WILL GET AUDIO GAPS. */

                tmp = fopen("C:\\files\\recorded_echo", "ab");
                fwrite((short *)(read_frame)->data, 2, ((read_frame)->datalen)/2 ,tmp);
                fclose(tmp);

        /*******************************************************************************************************************/
...
...
}

Furthermore, If you remove the line "switch_core_session_write_frame(session, read_frame, SWITCH_IO_FLAG_NONE, 0);" in function switch_ivr_session_echo, echo will not be sent back, but then spoken audio will be recorded correctly. This is what I meant with possible cross effect.


Date: Fri, 26 Mar 2010 13:07:14 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org

Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you put 1 frame in write for each read with the data size i told you about in the last email


On Fri, Mar 26, 2010 at 12:55 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:







yes, switch_core_session_read_frame will be timed right. This is working for me, but my troubles are in switch_core_session_write_frame.
Is this function timed write too when I put data into this function?



Date: Fri, 26 Mar 2010 12:37:04 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org


Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

You can use the data you have,

yes the the read will already be timed right 


when you get a frame that has flag SFF_CNG you know that there was not a real frame to read but you should still write
the size id determined from getting the read_impl and using read_impl->encoded_bytes_per_frame






On Fri, Mar 26, 2010 at 12:23 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:








ok,  which is the best mechanism to controll the amount of data. Can you recommend a timer? I use PCMA codec with 20 msec intervall.


Date: Fri, 26 Mar 2010 11:31:43 -0500

From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session




you can only write back the exact number of bytes you negotiated per interval.


On Fri, Mar 26, 2010 at 7:27 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:









Now I call switch_core_session_read_frame and switch_core_session_write_frame in the same foreground loop. Then I write audio into a file and as a result I see that there are many audio gaps. 

There seems to be a cross effect by using switch_core_session_write_frame and switch_core_session_read_frame together.




If I only use switch_core_session_read_frame in the loop, there are no gaps and everything sounds perfect!

What could be the problem? 



while (1)
{
    // READ
    tstatus = switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);




    if (tstatus==SWITCH_STATUS_SUCCESS)
    {        // write audio in a testfile
            fwrite((short *)(read_frame)->data, 2, ((read_frame)->datalen)/2 ,tmp);
            switch_size_t ts_rf=read_frame->timestamp;




    }

    // WRITE
    vector<unsigned char> vucIn;
    if (getAudio(&vucAudio)) // get next audio and send it back
    {
        memcpy( write_frame.data, &vucIn[0], vucIn.size());
        write_frame.datalen=vucIn.size();




        tstatus = switch_core_session_write_frame(session, &write_frame, SWITCH_IO_FLAG_NONE, 0);
    }
    //Sleep(1);//?
}

Date: Mon, 22 Mar 2010 18:34:46 -0500

From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session





you don't use the hook functions you use a frame pointer 

switch_frame_t *read_frame;

switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);

when i said to look at the stream file example i meant in the function itself to see how 





to read and write audio from a channel in a foreground loop.



On Mon, Mar 22, 2010 at 6:23 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:










I have an selfmade software which is controlled by a telephone call (ISDN/CAPI). The user calls with his phone, gets 
connected and can speak commands which are detected by an internal speech recognizer. Furthermore the user can accept 





or hangup incoming calls just by his voice command. It is also possible to dial out. 

Now I am adding VOIP functionality to my system and decided to use FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP interface for my software. The module must be able to:






1. recognize an incoming call which is the users control session (single session)
   I do this in statehandler function channel_on_init.
2. pick up another incoming call and intercept with my users control session. 





3. Get the audio data from voip session (user control session) and send it to my application (by own tcp-socket)
    My idea is to use channel_write_frame to receive audio.
3. Send audio data from my application to voip user control session so the user can hear audio.





    My idea is to use channel_read_frame.

explanation for using hooks:
----------------------------
The endpoint interface callback functions where never called in my module, even I tried to adapt the way it is done 





in mod_portaudio. So my workaround was to set hook functions channel_read_frame/channel_write_frame to handle audio. 

My problems:
------------
The hook functions are working fine if I have 2 sessions (put together by switch_ivr_intercept_session), but if I only have the single user control session no hooked function is called. In this case the only way to achieve that hooks are called, is to play audio with switch_ivr_play_file. But then, audio samples got lost sometimes.






Maybe there is a better way to solve my requirements? 




Date: Mon, 22 Mar 2010 11:55:42 -0500
From: anthony.minessale at gmail.com





To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

Why do you even need the hooks?





What exactly are you trying to do?
Maybe you should explain it.


On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:











I am now using the application interface and play a file after answering the call. The call is not connected to a another session, it's just a single session.

SWITCH_STANDARD_APP(mod_my_function)
{
     /* play audio file */






    switch_ivr_play_file(session, NULL, "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav", NULL);
}

Now the hooks for channel_write_frame and channel_read_frame are called! I write the audio frames into a file for testing if everything is correct. The data written in channel_write_frame seems to be perfect, but audio written in channel_read_frame only seems to be good at first sight. If I record 20 seconds of audio and listen to the audiofile I hear that 2 or 3 times short parts are missing. 







static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
  // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into a file






  fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t **frame, switch_io_flag_t flags, int stream_id)

{
    // Spoken audio samples from my microphone are written, but sometimes samples are missing
    fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}

Is it possible that I don't get all audio samples by hooked functions? why audio recorded in channel_write_frame is brilliant and in channel_read_frame not?












Date: Tue, 16 Mar 2010 13:59:28 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org






Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you probably want to use the application interface and a while loop while reading a frame.

see most of the functions in switch_ivr_play_say








On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:






The module I am writing should be an endpoint VOIP-interface for another program. The module should answer a call and transport incoming audio into a 3rd party application. Furthermore the 3rd party application sends audio data back to the caller. This is why I hooked channel_read_frame and channel_write_frame. 








How can I trigger the channel to do something that requires reading or writing audio? Does it make a difference to use the endpoint interface instead of the hooks? 


Date: Tue, 16 Mar 2010 08:34:18 -0500







From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session








What exactly is your goal?

those hooks will probably only be called if you send the channel to do something that 
requires reading and writing audio.


On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:













Hello,
I am writing my own freeswitch module and have implemented 2 hooks for reading and writing audio (I did not implement the endpoint interface).
My problem is that the channel_write_frame and channel_read_frame callback-functions are only called if I make an intercept with another session (switch_ivr_intercept_session), but my usecase has only one single Session for getting and putting audio data in my module. 









How can I achieve this? 
scaram

/* on channel init I answer the session and set hooks */
static switch_status_t channel_on_init(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);








    switch_channel_answer(channel);

    switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
    switch_set_flag(tech_pvt, TFLAG_IO);

    switch_core_event_hook_add_write_frame(session, &channel_write_frame);








    switch_core_event_hook_add_read_frame(session, &channel_read_frame);
}

static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)








{
    /* is only called when I intercept 2 Sessions */
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
    /* is only called when I intercept 2 Sessions */








}


 		 	   		  
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From wasim at convergence.pk  Mon Apr  5 03:38:54 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Mon, 5 Apr 2010 15:38:54 +0500
Subject: [Freeswitch-dev] bicc
Message-ID: <l2qb8ad2a5b1004050338w916fb64dkf471b66b185241b@mail.gmail.com>

folks:

has anyone given thought to mod_bicc?
how much effort is required?

-- 
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peace be upon you ...
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From msc at freeswitch.org  Wed Apr  7 08:22:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Apr 2010 08:22:27 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call - Moises Silva From
	Sangoma Talks FreeTDM!
Message-ID: <x2y87f2f3b91004070822l898a4f5cqaf593693ee932050@mail.gmail.com>

Hello everyone!

Moises (moy) is going to be speaking today about FreeTDM. Please join the
conference call at noon Eastern, 9am Pacific time:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_07

Moises will start his presentation at about 12:15/9:15. Be sure to bring
your questions about TDM and Sangoma!

-Michael
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From douglas at hubler.us  Tue Apr  6 13:15:53 2010
From: douglas at hubler.us (Douglas Hubler)
Date: Tue, 6 Apr 2010 16:15:53 -0400
Subject: [Freeswitch-dev] fedora 12 compiler error and workaround
Message-ID: <j2w5d36de591004061315r634aa13dwde9532b0d5e5a716@mail.gmail.com>

Here is the complier error (sorry for pasting it in here, but you'll
see why it's important in a second...)

Creating mod_amr.so...
/usr/lib/gcc/i686-redhat-linux/4.4.3/../../../crt1.o: In function `_start':
(.text+0x18): undefined reference to `main'
collect2: ld returned 1 exit status
quiet_libtool: link: gcc -I/home/dhubler/work/freeswitch/src/include
-I/home/dhubler/work/freeswitch/src/include
-I/home/dhubler/work/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
-ggdb -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement
-D_GNU_SOURCE -DAMR_PASSTHROUGH -o .libs/mod_amr.so -Wl,-x
.libs/mod_amr.o  -lm
/home/dhubler/work/freeswitch/.libs/libfreeswitch.so
-L/home/dhubler/work/freeswitch/libs/apr-util/xml/expat/lib
/home/dhubler/work/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/home/dhubler/work/freeswitch/libs/apr/.libs/libapr-1.a -lpq -luuid
-lrt -lcrypt -lpthread -lssl -lcrypto -ldl -lz -lncurses -Wl,-rpath
-Wl,/usr/local/freeswitch/lib -Wl,-rpath -Wl,/usr/local/freeswitch/mod


For some reason "-shared" is not on the link line

Don't ask me how i figured this out, but If I change in file build/modmake.rules
   SOLINK = -shared -Xlinker -x
to
   SOLINK = --shared -Xlinker -x

Everything compiles and freeswitch runs.  Something in the gnu tool
chain wants to strip out any "-foo" parameter to gcc but adding an
extra "-" it allows it through.

So i didn't find the problem, but i did find a weird workaround.


From jasonchu at avaya.com  Wed Apr  7 10:54:53 2010
From: jasonchu at avaya.com (CHU, XINGJUN (XINGJUN))
Date: Wed, 7 Apr 2010 13:54:53 -0400
Subject: [Freeswitch-dev] An issue wtih attended call transfer
In-Reply-To: <x2y87f2f3b91004070822l898a4f5cqaf593693ee932050@mail.gmail.com>
References: <x2y87f2f3b91004070822l898a4f5cqaf593693ee932050@mail.gmail.com>
Message-ID: <47AB18AC0F23934383F2BBA7EE3D8D7421F727703D@DC-US1MBEX4.global.avaya.com>

Hi,

I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to  control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc.

Now I got a problem when A calls B then attended transfer B to freeswitch,
The prompt B hear is not from the beginning, it actually hears  the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer.  I understand that's because nobody tell freeswitch to stop and start from the beginning.  I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning.

Any suggestions are greatly appreciated.

Thanks
Xingjun

From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Wednesday, April 07, 2010 11:22 AM
To: freeswitch-users at lists.freeswitch.org; freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] FreeSWITCH Conference Call - Moises Silva From Sangoma Talks FreeTDM!

Hello everyone!

Moises (moy) is going to be speaking today about FreeTDM. Please join the conference call at noon Eastern, 9am Pacific time:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_07

Moises will start his presentation at about 12:15/9:15. Be sure to bring your questions about TDM and Sangoma!

-Michael
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From vipkilla at gmail.com  Thu Apr  8 09:58:42 2010
From: vipkilla at gmail.com (vip killa)
Date: Thu, 8 Apr 2010 12:58:42 -0400
Subject: [Freeswitch-dev] streaming with mod_shout
Message-ID: <j2n957f61371004080958u10c68015x61beea58d47086d9@mail.gmail.com>

I'm trying to figure out how to stream to a shoutcast server. There is no
examples in the WIKI
I've been able to stream to an icecast server using the command
"conference name record shout://username:password at host/stream.mp3"

my shoutcast server does not require a username, so what would the command
be for an actual shoutcast server?
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From msc at freeswitch.org  Thu Apr  8 11:06:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 11:06:26 -0700
Subject: [Freeswitch-dev] Another IRC day in #freeswitch!
Message-ID: <g2s87f2f3b91004081106q88161761w198af77b76ace035@mail.gmail.com>

Hey everyone, come join us in #freeswitch on irc.freenode.net! We are going
for 250 users. (No sockpuppets, please.) If you don't have an IRC client
then visit http://www.freeswitch.org and click the Join Us On IRC link to
use the Web IRC applet.

See you in channel!
-MC (IRC: mercutioviz)
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From msc at freeswitch.org  Thu Apr  8 11:19:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 11:19:37 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
Message-ID: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>

Hi folks,

Just a quick FYI, there is g.729 license info now on the wiki:
http://wiki.freeswitch.org/wiki/Mod_com_g729

Thanks to Steve Ayre for starting this page. If you have g.729 knowledge to
add to the page please do so. Also, the INSTALL.txt file has been updated
with a bit more detail:
http://files.freeswitch.org/g729/INSTALL.txt

Happy transcoding!

-Michael
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From msc at freeswitch.org  Thu Apr  8 14:07:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Apr 2010 14:07:26 -0700
Subject: [Freeswitch-dev] UPDATE: G.729 Codec Licensing *IS* Available
Message-ID: <r2k87f2f3b91004081407j30c860f4t92164067b38fd858@mail.gmail.com>

Greetings all,

The FreeSWITCH team would like to let everyone know that we do indeed sell
g.729 licenses for $10 each. Use this link to initiate a purchase:
http://www.freeswitch.org/node/235

Note: licenses are available only for Linux-based systems at this time.
Please stay tuned for updates.

The INSTALL.txt file has very detailed instructions:
http://files.freeswitch.org/g729/INSTALL.txt

Keep in mind that a single license includes one encoder and one decoder,
that is, it can transcode both directions of a single phone call. If you
have any other questions please email us here or join us in #freeswitch on
irc.freenode.net.
-Michael
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From steveayre at gmail.com  Fri Apr  9 08:23:11 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 9 Apr 2010 16:23:11 +0100
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
Message-ID: <s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>

No problem Michael,

I was also going to rewrite the Mod_g729 page since it's out of date
now, but it's locked to prevent editing at the moment.

-Steve



On 8 April 2010 19:19, Michael Collins <msc at freeswitch.org> wrote:
> Hi folks,
>
> Just a quick FYI, there is g.729 license info now on the wiki:
> http://wiki.freeswitch.org/wiki/Mod_com_g729
>
> Thanks to Steve Ayre for starting this page. If you have g.729 knowledge to
> add to the page please do so. Also, the INSTALL.txt file has been updated
> with a bit more detail:
> http://files.freeswitch.org/g729/INSTALL.txt
>
> Happy transcoding!
>
> -Michael
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Apr  9 09:43:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 09:43:10 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
Message-ID: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>

On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:

> No problem Michael,
>
> I was also going to rewrite the Mod_g729 page since it's out of date
> now, but it's locked to prevent editing at the moment.
>
> -Steve
>
Thanks for the heads up. I'll check it out.
-MC
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From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From fowlerp at avaya.com  Mon Apr 12 11:58:30 2010
From: fowlerp at avaya.com (Fowler, Peter (Peter))
Date: Mon, 12 Apr 2010 14:58:30 -0400
Subject: [Freeswitch-dev] mod_conference: no events fire if member energy
	level is zero
Message-ID: <47AB18AC0F23934383F2BBA7EE3D8D7421F72FA553@DC-US1MBEX4.global.avaya.com>


If the conference participant sets his/her energy level threshold to zero (or the conference energy-level is set to zero), then
The floor_change event will never fire if the particpant becomes the loudest speaker. This is due to the line

If ((switch_test_flag(member, MFLAG_CAN_SPEAK) || switch _test_flag(member, MFLAG_MUTE_DETECT)) && energy _level) {
                                                                                                                                                                     ^^^^^^^^^^^^
I folks agree I'll enter a Jira ..

Peter


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From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
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From brian at freeswitch.org  Mon Apr 12 12:44:22 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Apr 2010 14:44:22 -0500
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
	<h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>
Message-ID: <0CE2BE15-B8A9-4AA4-95D0-AE750272B860@freeswitch.org>

NOTE freeswitch_licence_server must be started as root before FreeSWITCH if you're running as a non-root user.

/b

On Apr 9, 2010, at 12:59 PM, Michael Collins wrote:

> 
> FYI, I've update the page:
> http://wiki.freeswitch.org/wiki/Mod_g729
> 
> -MC 

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From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
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From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
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From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
-------------- next part --------------
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From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100409/9f0d1071/attachment-0006.html 

From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100409/9f0d1071/attachment-0007.html 

From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100409/9f0d1071/attachment-0013.html 

From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
-------------- next part --------------
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From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
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From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100409/9f0d1071/attachment-0018.html 

From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100409/9f0d1071/attachment-0019.html 

From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20100409/9f0d1071/attachment-0020.html 

From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
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From msc at freeswitch.org  Fri Apr  9 10:59:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Apr 2010 10:59:53 -0700
Subject: [Freeswitch-dev] G.729 Info On Wiki
In-Reply-To: <t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
References: <v2w87f2f3b91004081119h637e66eep9354ac8e35fe0b90@mail.gmail.com>
	<s2reafa772f1004090823ocbc292bta907f89a2dd19ca1@mail.gmail.com>
	<t2s87f2f3b91004090943x65c5f9cer812c35112f15ea6@mail.gmail.com>
Message-ID: <h2q87f2f3b91004091059t1d322b0y74d957d2043dfc80@mail.gmail.com>

On Fri, Apr 9, 2010 at 9:43 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 9, 2010 at 8:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No problem Michael,
>>
>> I was also going to rewrite the Mod_g729 page since it's out of date
>> now, but it's locked to prevent editing at the moment.
>>
>> -Steve
>>
> Thanks for the heads up. I'll check it out.
> -MC
>
>
FYI, I've update the page:
http://wiki.freeswitch.org/wiki/Mod_g729

-MC
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From paulo at esensetec.com.br  Sat Apr 10 16:42:42 2010
From: paulo at esensetec.com.br (Paulo Emerique)
Date: Sat, 10 Apr 2010 23:42:42 +0000 (UTC)
Subject: [Freeswitch-dev] Trying to send SIP MESSAGE to device
Message-ID: <979218.1061270942962603.JavaMail.root@zimbra>

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now.

We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform.

I'm really enjoying FS, congratulations!


Now for the actual question:

The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like:

MESSAGE sip: 1003 at 192.168.1.126:5060;transport=udp SIP/2.0
From: <SIP 1000 at 192.168.1.100>;tag=5031
To:<SIP: 1003 at 192.168.1.126>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008 at 192.168.1.100
CSeq: 3 MESSAGE
Contact: <sip: 1000 at 192.168.1.100>
Max-Forwards: 16
Date: Tue, 08 Apr 2010 06:36:37 GMT
User-Agent: Paulo
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 17
96913511
hi paulo


GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways:

1) using mod event sockets: I telnet to server and after auth, issue:
sendevent SEND_MESSAGE
profile: sipinterface_1
content-length: 18
content-type: text/plain
user: 1003
host: 192.168.1.126

96913511
Hello Baby

and returns:
Content-Type: command/reply
Reply-Text: +OK accepted

I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going.

then I tried

2)  chat sip|1030 at 192.168.1.132|1003 at 192.168.1.126|Test message
which returns:
Error! Message Not Sent
2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp]
from [1030 at 192.168.1.132]
to [1003 at 192.168.1.126]
Test message
Invalid Profile 192.168.1.126

So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows:
2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3

To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg:
2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4


Thank for your help,

Paulo Cesar Emerique


From brian at freeswitch.org  Tue Apr 13 06:20:51 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Apr 2010 08:20:51 -0500
Subject: [Freeswitch-dev] mail queue
Message-ID: <EB492B54-F3C9-41D1-AFA0-F9DE59F4A87A@freeswitch.org>

I dumped the mail queue this morning in hopes it would clear up anyone getting duplicates.  The log file reached 2 gigs and the MTA decided to just exit since the OS wouldn't let it open the logfile for append over the weekend.  Which caused this problem in the first place.

Sorry for any trouble.

Thanks,
Brian



From marketing at cluecon.com  Tue Apr 13 16:24:24 2010
From: marketing at cluecon.com (Michael Collins)
Date: Tue, 13 Apr 2010 16:24:24 -0700
Subject: [Freeswitch-dev] ClueCon MMX - Update
Message-ID: <u2w87f2f3b91004131624j2f6eb63frc516afb4e402ee91@mail.gmail.com>

Greetings,

We would like to let everyone know that ClueCon MMX preparations are moving
forward. New sponsors have signed up and we are starting to fill up our
speaker slots. Stay tuned for more information.

In the meantime, if you have not already signed up for ClueCon then please
call 877.742.CLUE (2583) and get registered right away!

Looking forward to seeing you in Chicago,
The ClueCon Team
877.742.CLUE
http://www.cluecon.com
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From babak.freeswitch at gmail.com  Tue Apr 13 22:15:02 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Wed, 14 Apr 2010 09:45:02 +0430
Subject: [Freeswitch-dev] ticketing system
Message-ID: <j2s1fdf92721004132215n646ea685l47ed2d1bee58280f@mail.gmail.com>

Hi
I need to implement a system in which customers call support operators and
if there is no idle operator they are placed in a queue and all
conversations are recorded and assigned a number so later customers can hear
their conversations. Can anyone help me in choosing right modules to use? I
have to add I am using sip to rcv calls and currently I'm using .Net to do
this
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From msc at freeswitch.org  Tue Apr 13 23:33:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Apr 2010 23:33:44 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call
Message-ID: <q2g87f2f3b91004132333r3d6a463cx94b4f6bafcc65ce@mail.gmail.com>

Hello all,

Our agenda for tomorrow is extremely light:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_14

No one was available for a formal discussion so I thought maybe we could
have an open discussion instead. If you have any agenda items please add
them.

Thanks!

-MC
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From vectorno at googlemail.com  Thu Apr 15 07:55:57 2010
From: vectorno at googlemail.com (vectorno vectorno)
Date: Thu, 15 Apr 2010 15:55:57 +0100
Subject: [Freeswitch-dev] how to setup lua as a daemon for event socket
	outbound
Message-ID: <j2id46d677f1004150755j2aa6c738l79e25df2f69a253e@mail.gmail.com>

How do I setup lua as a daemon so that it can be conencted to through
tcp socket?


From cshore at csolve.net  Wed Apr 14 10:23:40 2010
From: cshore at csolve.net (Daniel Dickinson)
Date: Wed, 14 Apr 2010 13:23:40 -0400
Subject: [Freeswitch-dev] Using system libraries
Message-ID: <20100414132340.04ff0cc9@daniloth.fionavar.dd>

Hi,

I'm working on an embedded OpenWRT system, and am now the package
maintainer for FreeSWITCH on OpenWRT.  I have FS successfully building
and packaged (and installing and working) on brcm63xx, and building on
the other OpenWRT architectures as well.  With 1.0.6 the build doesn't
require any patches, although it does require some finagling (as is to
be expected)

However, during the process of getting this to work for 1.0.6 I noticed
that FS doesn't use system libraries, but instead uses it's own copies
of packages it downloads, configures, and compiles and statically
links.  For OpenWRT this is problematic on two fronts.  The first is
that some packages require special build options to cross-compile
properly, and the second is on embedded systems have statically copies
of packages that are also on the system, uses up valuable storage and
RAM.

Would it be possible for the build system to support the *option* of
using system versions of the packages instead of static-linked versions?

Regards,

Daniel

-- 
And that's my crabbing done for the day.  Got it out of the way early, 
now I have the rest of the afternoon to sniff fragrant tea-roses or 
strangle cute bunnies or something.   -- Michael Devore
GnuPG Key Fingerprint 86 F5 81 A5 D4 2E 1F 1C      http://gnupg.org
The C Shore (Daniel Dickinson's Website) http://cshore.is-a-geek.com


From cshore at csolve.net  Wed Apr 14 10:45:27 2010
From: cshore at csolve.net (Daniel Dickinson)
Date: Wed, 14 Apr 2010 13:45:27 -0400
Subject: [Freeswitch-dev] I'll be working on the system libraries
Message-ID: <20100414134527.2b720596@daniloth.fionavar.dd>

Ok, just to let you know it looks like I have the go ahead to create
configuration switches (i.e. options) to do this, and I'll will be
working on it (it won't be quick though, because I'm trying to get some
more pressing work done first).

Regards,

Daniel

-- 
And that's my crabbing done for the day.  Got it out of the way early, 
now I have the rest of the afternoon to sniff fragrant tea-roses or 
strangle cute bunnies or something.   -- Michael Devore
GnuPG Key Fingerprint 86 F5 81 A5 D4 2E 1F 1C      http://gnupg.org
The C Shore (Daniel Dickinson's Website) http://cshore.is-a-geek.com
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From anthony.minessale at gmail.com  Thu Apr 15 08:10:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Apr 2010 10:10:43 -0500
Subject: [Freeswitch-dev] Using system libraries
In-Reply-To: <20100414132340.04ff0cc9@daniloth.fionavar.dd>
References: <20100414132340.04ff0cc9@daniloth.fionavar.dd>
Message-ID: <l2t191c3a031004150810x5e2bbb16t7bd87d1246da89d6@mail.gmail.com>

Yes perhaps, but someone needs to provide the patch to make it work
*exactly* 100% the same as it does now by default and configure options to
specify the system packages, but some of them, sqlite3 and apr for example
will cause crashes if you use the system one because we have made local mods
to them.

Maybe you should uninstall the system copy if you don't need it and want to
save space.
Statically linking them only uses the part of the library we need anyway and
is not the same size cost
as using a shared version.

If there are special build options to the packages you can specify them to
the FS build system to pass them on.  Clearly you already have a package so
you managed to do something right.

Also keep in mind that the needs of us as the maintainers of the code and
the users at large outweigh the needs one one use case.

As a heads up we will never endorse using system libs with our application
unless at least 5 years have passed with a specific version of a specific
library that we can prove is not an issue.



On Wed, Apr 14, 2010 at 12:23 PM, Daniel Dickinson <cshore at csolve.net>wrote:

> Hi,
>
> I'm working on an embedded OpenWRT system, and am now the package
> maintainer for FreeSWITCH on OpenWRT.  I have FS successfully building
> and packaged (and installing and working) on brcm63xx, and building on
> the other OpenWRT architectures as well.  With 1.0.6 the build doesn't
> require any patches, although it does require some finagling (as is to
> be expected)
>
> However, during the process of getting this to work for 1.0.6 I noticed
> that FS doesn't use system libraries, but instead uses it's own copies
> of packages it downloads, configures, and compiles and statically
> links.  For OpenWRT this is problematic on two fronts.  The first is
> that some packages require special build options to cross-compile
> properly, and the second is on embedded systems have statically copies
> of packages that are also on the system, uses up valuable storage and
> RAM.
>
> Would it be possible for the build system to support the *option* of
> using system versions of the packages instead of static-linked versions?
>
> Regards,
>
> Daniel
>
> --
> And that's my crabbing done for the day.  Got it out of the way early,
> now I have the rest of the afternoon to sniff fragrant tea-roses or
> strangle cute bunnies or something.   -- Michael Devore
> GnuPG Key Fingerprint 86 F5 81 A5 D4 2E 1F 1C      http://gnupg.org
> The C Shore (Daniel Dickinson's Website) http://cshore.is-a-geek.com
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> 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_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Apr 15 09:04:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Apr 2010 11:04:30 -0500
Subject: [Freeswitch-dev] how to setup lua as a daemon for event socket
	outbound
In-Reply-To: <j2id46d677f1004150755j2aa6c738l79e25df2f69a253e@mail.gmail.com>
References: <j2id46d677f1004150755j2aa6c738l79e25df2f69a253e@mail.gmail.com>
Message-ID: <p2u191c3a031004150904w5cd0a4ase751fd57490d19c2@mail.gmail.com>

You could use ivrd to run as a forking server and run the lua script in a
separate process otherwise ask the lua group if they have a module for tcp
sockets and forking because I don't know.



On Thu, Apr 15, 2010 at 9:55 AM, vectorno vectorno
<vectorno at googlemail.com>wrote:

> How do I setup lua as a daemon so that it can be conencted to through
> tcp socket?
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> 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_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From vectorno at googlemail.com  Thu Apr 15 12:16:12 2010
From: vectorno at googlemail.com (vectorno vectorno)
Date: Thu, 15 Apr 2010 20:16:12 +0100
Subject: [Freeswitch-dev] how to setup lua as a daemon for event socket
	outbound
Message-ID: <j2vd46d677f1004151216xa91ecc8bka2b3eb317ce1c401@mail.gmail.com>

IIs ivrd availible for windows


From msc at freeswitch.org  Thu Apr 15 14:55:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Apr 2010 14:55:31 -0700
Subject: [Freeswitch-dev] how to setup lua as a daemon for event socket
	outbound
In-Reply-To: <j2vd46d677f1004151216xa91ecc8bka2b3eb317ce1c401@mail.gmail.com>
References: <j2vd46d677f1004151216xa91ecc8bka2b3eb317ce1c401@mail.gmail.com>
Message-ID: <x2p87f2f3b91004151455m2635e9ectf7ce127f8fc7be9b@mail.gmail.com>

On Thu, Apr 15, 2010 at 12:16 PM, vectorno vectorno <vectorno at googlemail.com
> wrote:

> IIs ivrd availible for windows
>
It doesn't look like ivrd builds in windows. I'll ask Jeff Lenk to check it
out.
-MC
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From vectorno at googlemail.com  Thu Apr 15 16:33:53 2010
From: vectorno at googlemail.com (vectorno vectorno)
Date: Fri, 16 Apr 2010 00:33:53 +0100
Subject: [Freeswitch-dev] how to setup lua as a daemon for event socket
	outbound
Message-ID: <h2ud46d677f1004151633na159e876vae4b372346409ef4@mail.gmail.com>

Thanks for that I just wanted to quickly run the demos while I was
logged in to window.  Anyway I have now compiled a fresh install on
Linux and have manged get up and running.

The reason why I initialy asked about lua to control  outbound sockets
is because it is claimed that it is very fast however I am new too
llua and for now  I am sticking to php but how much is there a
difference is speed and Is there a more optimal way to use php for
sockets connections?

II have setup socket connection to ivrd calling php with sucesscull
execution of the script. There is one issue I have come across i am
getting a error in doing  the following command

echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: playback\n\n";
echo "execute-app-arg:
tone_stream://%(400,200,400,450);%(400,2200,400,450);loops=2\n\n";

IIt throws up the warning: switch_core_file.c:176 Sample rate doesn't
match. Any ideas?






On Thu, Apr 15, 2010 at 12:16 PM, vectorno vectorno <vectorno at googlemail.com
> wrote:

> IIs ivrd availible for windows
>
It doesn't look like ivrd builds in windows. I'll ask Jeff Lenk to check it
out.
-MC


From mike at jerris.com  Thu Apr 15 17:20:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Apr 2010 20:20:20 -0400
Subject: [Freeswitch-dev] how to setup lua as a daemon for event socket
	outbound
In-Reply-To: <x2p87f2f3b91004151455m2635e9ectf7ce127f8fc7be9b@mail.gmail.com>
References: <j2vd46d677f1004151216xa91ecc8bka2b3eb317ce1c401@mail.gmail.com>
	<x2p87f2f3b91004151455m2635e9ectf7ce127f8fc7be9b@mail.gmail.com>
Message-ID: <3F656B07-487A-4493-B15A-41580414433E@jerris.com>

ivrd uses fork, we will have to come up with something else for windows to do this.

Mike

On Apr 15, 2010, at 5:55 PM, Michael Collins wrote:

> 
> 
> On Thu, Apr 15, 2010 at 12:16 PM, vectorno vectorno <vectorno at googlemail.com> wrote:
> IIs ivrd availible for windows
> It doesn't look like ivrd builds in windows. I'll ask Jeff Lenk to check it out.
> -MC
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Thu Apr 15 17:45:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Apr 2010 19:45:13 -0500
Subject: [Freeswitch-dev] how to setup lua as a daemon for event socket
	outbound
In-Reply-To: <3F656B07-487A-4493-B15A-41580414433E@jerris.com>
References: <j2vd46d677f1004151216xa91ecc8bka2b3eb317ce1c401@mail.gmail.com>
	<x2p87f2f3b91004151455m2635e9ectf7ce127f8fc7be9b@mail.gmail.com>
	<3F656B07-487A-4493-B15A-41580414433E@jerris.com>
Message-ID: <r2s191c3a031004151745lcb039ba2se16a50be2dd8d18e@mail.gmail.com>

sample rate doesn't match just means you are playing a file that is not the
same sample rate as the channel.


On Thu, Apr 15, 2010 at 7:20 PM, Michael Jerris <mike at jerris.com> wrote:

> ivrd uses fork, we will have to come up with something else for windows to
> do this.
>
> Mike
>
> On Apr 15, 2010, at 5:55 PM, Michael Collins wrote:
>
>
>
> On Thu, Apr 15, 2010 at 12:16 PM, vectorno vectorno <
> vectorno at googlemail.com> wrote:
>
>> IIs ivrd availible for windows
>>
> It doesn't look like ivrd builds in windows. I'll ask Jeff Lenk to check it
> out.
> -MC
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> 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_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Fri Apr 16 19:26:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Apr 2010 19:26:30 -0700
Subject: [Freeswitch-dev] Wiki info: fs_ivrd
Message-ID: <u2j87f2f3b91004161926neaed410cza34a1315fa202921@mail.gmail.com>

Gang,

I just finished some awesome documentation on using fs_ivrd and the Perl
ESL::IVR module:

http://wiki.freeswitch.org/wiki/Ivrd

If anyone else is using fs_ivrd and has some examples then by all means
please add to this page. Also, feel free to check the page's accuracy.

Thanks!
-MC
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From msc at freeswitch.org  Mon Apr 19 16:14:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Apr 2010 16:14:18 -0700
Subject: [Freeswitch-dev] FreeSWITCH Connector For Vestec Speech Recognition
	Engine
Message-ID: <w2y87f2f3b91004191614h9a66f6e0jdedb9e493d109742@mail.gmail.com>

Good news! FreeSWITCH and Vestec have teamed up to deliver a FreeSWITCH
connector for their speech recognition engine:

http://www.freeswitch.org/node/252

Kudos to everyone who made this possible. Please go buy some speech
recognition licenses and start building some great new applications!

-MC
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From msc at freeswitch.org  Wed Apr 21 08:07:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Apr 2010 08:07:49 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call
Message-ID: <v2p87f2f3b91004210807j2fa1fedteeedbd920a4cc334@mail.gmail.com>

Hello all. Here is a reminder that the FreeSWITCH community conference call
will be starting soon. The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_21

Please feel free to add your agenda items.
-MC
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From devel at thom.fr.eu.org  Wed Apr 21 13:33:29 2010
From: devel at thom.fr.eu.org (devel at thom.fr.eu.org)
Date: Wed, 21 Apr 2010 22:33:29 +0200
Subject: [Freeswitch-dev] Problem with sending DTMF on FXS port bridged to
	an FXO port
Message-ID: <004101cae191$e81ad2e0$b85078a0$@fr.eu.org>

Hello,

 

I?m having trouble with this configuration with sangoma A400 card.

What happens is this : the call is established (bridged from FXS to FXO) to
an IVR, then the caller press some key.

At that moment, the sangoma card detects the DTMF on the FXS channel, and
queue the DTMF to be sent of the FXO channel. As a result, to the remote
IVR, the DTMF is received twice (once inband, and a second time when
generated from the FXO).

I sometimes get an even worse scenario where the DTMF is detected by the
card on both channels, and an endless loop start as the DTMF gets detected
by the other channel when generated and so on and so forth.

 

I tried to stop DTMF generation by adding the following lines to the
dialplan before the bridge

<action application="stop_dtmf" data=""/>

<action application="stop_dtmf_generate" data=""/>

<action application="bridge" data="openzap/2/2/$1"/>

 

But this did not help.

 

I could kindof solve the endless loop by modifying zap_io.c in
zap_channel_set_state, when the new state is dialing (which means the FXO
channel is an outbound one, and I therefore don?t need DTMF detection) I do
a zap_clear_flag(zchan, ZAP_CHANNEL_DTMF_DETECT); which prevent further DTMF
detection on that channel.

 

I don?t know where to carry on now, as just preventing DTMF generation on
outbound FXO channel would prevent DTMF generation where A leg is sofia or
any other.

 

Fran?ois

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From cstomi.levlist at gmail.com  Thu Apr 22 08:33:12 2010
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Thu, 22 Apr 2010 17:33:12 +0200
Subject: [Freeswitch-dev] windows json build
In-Reply-To: <66E9C220-86E3-4CCF-AD50-83715B5E6AD1@jerris.com>
References: <E1NtiY8-0005qO-65@svn.freeswitch.org>	<4BA8A822.4020401@gmail.com>
	<4BA8E867.1020702@gmail.com>
	<66E9C220-86E3-4CCF-AD50-83715B5E6AD1@jerris.com>
Message-ID: <4BD06C38.9020203@gmail.com>

Hello,

I changed every boolean to json_boolean, and it compiles fine for me:
(http://jira.freeswitch.org/browse/MODEVENT-63)

I added mod_curl and mod_json_cdr to the 2008 express solution.

Regards,
Tamas


Michael Jerris wrote:
> The new tarball will be up shortly, there was one more change.
>
> Mike
>
> On Mar 23, 2010, at 12:12 PM, Tamas Cseke wrote:
>
>   
>> Hello,
>>
>> Responding (for others who may be intrested)
>> for json build a patched json tarball is needed.
>>
>> Here is the modification
>> http://pastebin.freeswitch.org/12524
>>
>> After this libjson build works.
>> mod_curl still used json header files from 0.8 version.
>> And I got compile errors
>> http://pastebin.freeswitch.org/12525
>>
>> Jlenk is working on mod_curl build...it's coming soon...
>> Thanks the help!
>>
>> Tamas
>>
>>
>>
>> Tamas Cseke wrote:
>>     
>>> Hello,
>>>
>>> I've seen the json upgrade (0.8 ->0.9) has been done for windows as well 
>>> in r17064.
>>>
>>> If it would work, I'd like to fix the windows build for mod_json_cdr
>>> (http://jira.freeswitch.org/browse/MODEVENT-62)
>>>
>>> I was trying with vc++ 2008 express, but I can't compile jsonlib.
>>>
>>> json_util.c(62) : error C2065: 'O_RDONLY' : undeclared identifier
>>> json_util.c(97) : error C2065: 'O_WRONLY' : undeclared identifier
>>> json_util.c(97) : error C2065: 'O_TRUNC' : undeclared identifier
>>> json_util.c(97) : error C2065: 'O_CREAT' : undeclared identifier
>>>
>>> Is it works for you? does mod_curl build works on windows?
>>> or the jsonlib doesn't support windows at all (it seems to me) ?
>>>
>>> Thanks in advance,
>>> Tamas
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>   



From rudek at lyth.de  Thu Apr 22 10:31:36 2010
From: rudek at lyth.de (Sebastian Rudek)
Date: Thu, 22 Apr 2010 19:31:36 +0200
Subject: [Freeswitch-dev] Fwd: thousands of memory leaks when using own
	module and mfc dll
Message-ID: <4BD087F8.2080504@lyth.de>

Dear Freeswitch developers,

i have a problem with FreeSwitch (Version 1.0.4) and my own module,
which loads a MFC based DLL -- where obviously memory leaks are
produced. let me try to explain what i have.
1) First of all, let's discuss the case where everything works fine in
freeswitch. My FreeSwitch module (which here loads no further DLL) is
loaded and unloaded correctly when a call is accepted and later on hung up.
2) Then i load an "empty" DLL, without MFC and all that stuff, which i
created, within my FreeSwitch module. I do this when accepting a call.
when hanging up, I FreeLibrary(), quit FreeSwitch and everything is
fine. No memory leaks, no errors, just everything as it ought to be.
3) But then! Then i take my module and load a DLL which uses MFC. and
then, when after unloading the DLL on hanging up, I exit the FreeSwitch
application and get that many memory leak reports in the debug out of
visual studio 2008... that leads me to my question: Are there any know
issues with FreeSwitch and MFC DLLs ?

I tried to simulate the problem with creating an Application which
simply loads the Non-MFC DLL (see number 2 above). No memory leaks.
Also, no memory leaks are detected when loading the MFC-DLL from 3) (see
above). but Visual Studio detects one (!) 4-byte-memory-leak when
loading the MFC DLL inside the Non-MFC-Dll (like the FreeSwitch module
ought to do it).

Back to the question, if there is an issue with FreeSwitch and MFC-DLLs.
The object dump, which Visual Studio gives to me, says that leaks are
detected at "VARCHAR(255)", "shma", "TeleTone" or "loopback.auto". I do
not know where that comes from. Maybe, you have an idea which you could
share with me to solve the problem, on which i have spent really much
time already...
I just attach the out as text file.

Thank you so much in advance

Sebastian

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From mike at jerris.com  Thu Apr 22 12:25:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 22 Apr 2010 15:25:29 -0400
Subject: [Freeswitch-dev] Fwd: thousands of memory leaks when using own
	module and mfc dll
In-Reply-To: <4BD087F8.2080504@lyth.de>
References: <4BD087F8.2080504@lyth.de>
Message-ID: <DF6F072E-3286-40AC-874C-7CC4AF6D634E@jerris.com>

Please try our latest release version 1.0.6

Mike

On Apr 22, 2010, at 1:31 PM, Sebastian Rudek wrote:

> Dear Freeswitch developers,
> 
> i have a problem with FreeSwitch (Version 1.0.4) and my own module,



From vipkilla at gmail.com  Thu Apr 22 12:52:57 2010
From: vipkilla at gmail.com (vip killa)
Date: Thu, 22 Apr 2010 15:52:57 -0400
Subject: [Freeswitch-dev] recommended sip phone or adapter for outside LAN
Message-ID: <p2t957f61371004221252x58e4b071h8b103d0f467d639c@mail.gmail.com>

I have a freeswitch server running at one location. I want to setup a sip
phone at another location (outside the local freeswitch network). can
someone recommend a sip phone or adapter that is not too expensive and will
easily connect/work with the freeswitch server at location one?
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From mike at jerris.com  Thu Apr 22 13:21:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 22 Apr 2010 16:21:25 -0400
Subject: [Freeswitch-dev] recommended sip phone or adapter for outside
	LAN
In-Reply-To: <p2t957f61371004221252x58e4b071h8b103d0f467d639c@mail.gmail.com>
References: <p2t957f61371004221252x58e4b071h8b103d0f467d639c@mail.gmail.com>
Message-ID: <DBF5DD49-6596-4580-B8A1-318ECCE99C89@jerris.com>

This question is more appropriate for the users list.

Mike

On Apr 22, 2010, at 3:52 PM, vip killa wrote:

> I have a freeswitch server running at one location. I want to setup a sip phone at another location (outside the local freeswitch network). can someone recommend a sip phone or adapter that is not too expensive and will easily connect/work with the freeswitch server at location one?
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From vetali100 at gmail.com  Fri Apr 23 03:43:32 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 23 Apr 2010 13:43:32 +0300
Subject: [Freeswitch-dev]  Create applications in FreeSWITCH
Message-ID: <h2ma858e7691004230343u2b609e13n4cbd56b81559359b@mail.gmail.com>

Hi dear group,

I would like to start writing plugins (applications) for FreeSWITCH that
will be called from dialplan:

<action application="my_cool_application"...>

Could you please provide me some steps following which I will be able to
create "helloworld" C/C++ application that will be called from dialplan?

Or some link if this is already described somewhere... I tried to google but
did not find that.

Then, if I will create something useful, I will contribute it to the
community with pleasure... :-)

Thank you very much!

Vitalie
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From mrene_lists at avgs.ca  Fri Apr 23 05:29:19 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 23 Apr 2010 08:29:19 -0400
Subject: [Freeswitch-dev] Create applications in FreeSWITCH
In-Reply-To: <h2ma858e7691004230343u2b609e13n4cbd56b81559359b@mail.gmail.com>
References: <h2ma858e7691004230343u2b609e13n4cbd56b81559359b@mail.gmail.com>
Message-ID: <1EE3EB32-71F5-47EF-AE80-C5C5DBED0E34@avgs.ca>

Hi

Look at mod_skel in src/mod/applications

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-23, at 6:43 AM, Vitalii Colosov wrote:

> Hi dear group,
> 
> I would like to start writing plugins (applications) for FreeSWITCH that will be called from dialplan:
> 
> <action application="my_cool_application"...>
> 
> Could you please provide me some steps following which I will be able to create "helloworld" C/C++ application that will be called from dialplan?
> 
> Or some link if this is already described somewhere... I tried to google but did not find that.
> 
> Then, if I will create something useful, I will contribute it to the community with pleasure... :-)
> 
> Thank you very much!
> 
> Vitalie
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From rudek at lyth.de  Fri Apr 23 06:36:24 2010
From: rudek at lyth.de (Sebastian Rudek)
Date: Fri, 23 Apr 2010 15:36:24 +0200
Subject: [Freeswitch-dev] Fwd: thousands of memory leaks when using own
 module and mfc dll
In-Reply-To: <DF6F072E-3286-40AC-874C-7CC4AF6D634E@jerris.com>
References: <4BD087F8.2080504@lyth.de>
	<DF6F072E-3286-40AC-874C-7CC4AF6D634E@jerris.com>
Message-ID: <4BD1A258.6090902@lyth.de>

Thank you very much for replying. But, unfortunately, an update to the 
newest version did not help.


Am 22.04.2010 21:25, schrieb Michael Jerris:
> Please try our latest release version 1.0.6
>
> Mike
>
> On Apr 22, 2010, at 1:31 PM, Sebastian Rudek wrote:
>
>    
>> Dear Freeswitch developers,
>>
>> i have a problem with FreeSwitch (Version 1.0.4) and my own module,
>>      
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>    


-- 
Sebastian Rudek, Softwareentwickler


Lyncker&  Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany

Fon +49 611/89038960
Fax +49 611/9406125


Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 4023897051
USt-IdNr.: DE255806399

Gesch?ftsf?hrer:
Filip Lyncker,
Armin Theis



From brian at freeswitch.org  Fri Apr 23 06:41:27 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 23 Apr 2010 08:41:27 -0500
Subject: [Freeswitch-dev] Fwd: thousands of memory leaks when using own
	module and mfc dll
In-Reply-To: <4BD1A258.6090902@lyth.de>
References: <4BD087F8.2080504@lyth.de>
	<DF6F072E-3286-40AC-874C-7CC4AF6D634E@jerris.com>
	<4BD1A258.6090902@lyth.de>
Message-ID: <3EFCF661-A475-4D7F-AD88-CD3EE78228A1@freeswitch.org>

I don't see any info in that leak report that points out where any leaks are... its just a bunch of jibberish.

/b

On Apr 23, 2010, at 8:36 AM, Sebastian Rudek wrote:

> Thank you very much for replying. But, unfortunately, an update to the 
> newest version did not help.



From vetali100 at gmail.com  Fri Apr 23 07:05:23 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 23 Apr 2010 17:05:23 +0300
Subject: [Freeswitch-dev] Create applications in FreeSWITCH
In-Reply-To: <1EE3EB32-71F5-47EF-AE80-C5C5DBED0E34@avgs.ca>
References: <h2ma858e7691004230343u2b609e13n4cbd56b81559359b@mail.gmail.com>
	<1EE3EB32-71F5-47EF-AE80-C5C5DBED0E34@avgs.ca>
Message-ID: <g2la858e7691004230705h65a8131ds81b3836f015fa817@mail.gmail.com>

Thanks Mathieu,
This really helped!
I was able to get Hello World string during the module load :-)

Moving to the next step, how can I define an application, let say "skel_app"
inside this module?
In order to be able to execute it from dialplan.

I thought the following does this: SWITCH_ADD_API(api_interface, "skel",
"Skel API", skel_function, "syntax");
But when I executed "skel" from dialplan, I got error: "Invalid application
skel".

Could you hint me please?

Thank you very much!

Vitalie



2010/4/23 Mathieu Rene <mrene_lists at avgs.ca>

> Hi
>
> Look at mod_skel in src/mod/applications
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-04-23, at 6:43 AM, Vitalii Colosov wrote:
>
> > Hi dear group,
> >
> > I would like to start writing plugins (applications) for FreeSWITCH that
> will be called from dialplan:
> >
> > <action application="my_cool_application"...>
> >
> > Could you please provide me some steps following which I will be able to
> create "helloworld" C/C++ application that will be called from dialplan?
> >
> > Or some link if this is already described somewhere... I tried to google
> but did not find that.
> >
> > Then, if I will create something useful, I will contribute it to the
> community with pleasure... :-)
> >
> > Thank you very much!
> >
> > Vitalie
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From vetali100 at gmail.com  Fri Apr 23 07:19:21 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 23 Apr 2010 17:19:21 +0300
Subject: [Freeswitch-dev] Create applications in FreeSWITCH
In-Reply-To: <g2la858e7691004230705h65a8131ds81b3836f015fa817@mail.gmail.com>
References: <h2ma858e7691004230343u2b609e13n4cbd56b81559359b@mail.gmail.com>
	<1EE3EB32-71F5-47EF-AE80-C5C5DBED0E34@avgs.ca>
	<g2la858e7691004230705h65a8131ds81b3836f015fa817@mail.gmail.com>
Message-ID: <o2ia858e7691004230719hd5491240sfc2cbf92d6573abc@mail.gmail.com>

OK I think I found, need to use: SWITCH_ADD_APP

Thank you very much!!!


2010/4/23 Vitalii Colosov <vetali100 at gmail.com>

> Thanks Mathieu,
> This really helped!
> I was able to get Hello World string during the module load :-)
>
> Moving to the next step, how can I define an application, let say
> "skel_app" inside this module?
> In order to be able to execute it from dialplan.
>
> I thought the following does this: SWITCH_ADD_API(api_interface, "skel",
> "Skel API", skel_function, "syntax");
> But when I executed "skel" from dialplan, I got error: "Invalid application
> skel".
>
> Could you hint me please?
>
> Thank you very much!
>
> Vitalie
>
>
>
> 2010/4/23 Mathieu Rene <mrene_lists at avgs.ca>
>
> Hi
>>
>> Look at mod_skel in src/mod/applications
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 2010-04-23, at 6:43 AM, Vitalii Colosov wrote:
>>
>> > Hi dear group,
>> >
>> > I would like to start writing plugins (applications) for FreeSWITCH that
>> will be called from dialplan:
>> >
>> > <action application="my_cool_application"...>
>> >
>> > Could you please provide me some steps following which I will be able to
>> create "helloworld" C/C++ application that will be called from dialplan?
>> >
>> > Or some link if this is already described somewhere... I tried to google
>> but did not find that.
>> >
>> > Then, if I will create something useful, I will contribute it to the
>> community with pleasure... :-)
>> >
>> > Thank you very much!
>> >
>> > Vitalie
>> > _______________________________________________
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
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From fdelawarde at wirelessmundi.com  Fri Apr 23 07:25:55 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 23 Apr 2010 16:25:55 +0200
Subject: [Freeswitch-dev] Create applications in FreeSWITCH
In-Reply-To: <1EE3EB32-71F5-47EF-AE80-C5C5DBED0E34@avgs.ca>
References: <h2ma858e7691004230343u2b609e13n4cbd56b81559359b@mail.gmail.com>
	<1EE3EB32-71F5-47EF-AE80-C5C5DBED0E34@avgs.ca>
Message-ID: <1272032755.12512.613.camel@luna.tc.commsmundi.com>

And please share your cool applications!

Fran?ois.


On Fri, 2010-04-23 at 08:29 -0400, Mathieu Rene wrote:
> Hi
> 
> Look at mod_skel in src/mod/applications
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2010-04-23, at 6:43 AM, Vitalii Colosov wrote:
> 
> > Hi dear group,
> > 
> > I would like to start writing plugins (applications) for FreeSWITCH that will be called from dialplan:
> > 
> > <action application="my_cool_application"...>
> > 
> > Could you please provide me some steps following which I will be able to create "helloworld" C/C++ application that will be called from dialplan?
> > 
> > Or some link if this is already described somewhere... I tried to google but did not find that.
> > 
> > Then, if I will create something useful, I will contribute it to the community with pleasure... :-)
> > 
> > Thank you very much!
> > 
> > Vitalie
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From msc at freeswitch.org  Fri Apr 23 07:58:11 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 23 Apr 2010 07:58:11 -0700
Subject: [Freeswitch-dev] Create applications in FreeSWITCH
In-Reply-To: <o2ia858e7691004230719hd5491240sfc2cbf92d6573abc@mail.gmail.com>
References: <h2ma858e7691004230343u2b609e13n4cbd56b81559359b@mail.gmail.com>
	<1EE3EB32-71F5-47EF-AE80-C5C5DBED0E34@avgs.ca>
	<g2la858e7691004230705h65a8131ds81b3836f015fa817@mail.gmail.com>
	<o2ia858e7691004230719hd5491240sfc2cbf92d6573abc@mail.gmail.com>
Message-ID: <F0C2CA59-B2A2-44DE-8095-BF7DA30B6406@freeswitch.org>

> SWITCH_ADD_APP is indeed for dialplan apps. SWITCH_ADD_API is for  
> adding API commands that are executed from the CLI
-MC

Sent from my iPhone

On Apr 23, 2010, at 7:19 AM, Vitalii Colosov <vetali100 at gmail.com>  
wrote:

> OK I think I found, need to use: SWITCH_ADD_APP
>
> Thank you very much!!!
>
>
> 2010/4/23 Vitalii Colosov <vetali100 at gmail.com>
> Thanks Mathieu,
> This really helped!
> I was able to get Hello World string during the module load :-)
>
> Moving to the next step, how can I define an application, let say  
> "skel_app" inside this module?
> In order to be able to execute it from dialplan.
>
> I thought the following does this: SWITCH_ADD_API(api_interface,  
> "skel", "Skel API", skel_function, "syntax");
> But when I executed "skel" from dialplan, I got error: "Invalid  
> application skel".
>
> Could you hint me please?
>
> Thank you very much!
>
> Vitalie
>
>
>
> 2010/4/23 Mathieu Rene <mrene_lists at avgs.ca>
>
> Hi
>
> Look at mod_skel in src/mod/applications
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-04-23, at 6:43 AM, Vitalii Colosov wrote:
>
> > Hi dear group,
> >
> > I would like to start writing plugins (applications) for  
> FreeSWITCH that will be called from dialplan:
> >
> > <action application="my_cool_application"...>
> >
> > Could you please provide me some steps following which I will be  
> able to create "helloworld" C/C++ application that will be called  
> from dialplan?
> >
> > Or some link if this is already described somewhere... I tried to  
> google but did not find that.
> >
> > Then, if I will create something useful, I will contribute it to  
> the community with pleasure... :-)
> >
> > Thank you very much!
> >
> > Vitalie
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> dev
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Fri Apr 23 08:07:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Apr 2010 10:07:17 -0500
Subject: [Freeswitch-dev] Fwd: thousands of memory leaks when using own
	module and mfc dll
In-Reply-To: <4BD1A258.6090902@lyth.de>
References: <4BD087F8.2080504@lyth.de>
	<DF6F072E-3286-40AC-874C-7CC4AF6D634E@jerris.com>
	<4BD1A258.6090902@lyth.de>
Message-ID: <o2k191c3a031004230807v8f5694bfm9bce0abe3912f309@mail.gmail.com>

Are you loading and unloading the dll each call?
you probably only have to load it once when the module is loaded into FS.
Maybe it's caching the old copy of the subsequent loads.
Clearly this is not a FS problem anyway as your report has already proved.


On Fri, Apr 23, 2010 at 8:36 AM, Sebastian Rudek <rudek at lyth.de> wrote:

> Thank you very much for replying. But, unfortunately, an update to the
> newest version did not help.
>
>
> Am 22.04.2010 21:25, schrieb Michael Jerris:
> > Please try our latest release version 1.0.6
> >
> > Mike
> >
> > On Apr 22, 2010, at 1:31 PM, Sebastian Rudek wrote:
> >
> >
> >> Dear Freeswitch developers,
> >>
> >> i have a problem with FreeSwitch (Version 1.0.4) and my own module,
> >>
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> >
>
>
> --
> Sebastian Rudek, Softwareentwickler
>
>
> Lyncker&  Theis GmbH
> Wilhelmstr. 16
> 65185 Wiesbaden
> Germany
>
> Fon +49 611/89038960
> Fax +49 611/9406125
>
>
> Handelsregister: HRB 23156 Amtsgericht Wiesbaden
> Steuernummer: 4023897051
> USt-IdNr.: DE255806399
>
> Gesch?ftsf?hrer:
> Filip Lyncker,
> Armin Theis
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> 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_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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pstn:+19193869900
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From steveayre at gmail.com  Fri Apr 23 10:15:07 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Apr 2010 18:15:07 +0100
Subject: [Freeswitch-dev] Create applications in FreeSWITCH
In-Reply-To: <o2ia858e7691004230719hd5491240sfc2cbf92d6573abc@mail.gmail.com>
References: <h2ma858e7691004230343u2b609e13n4cbd56b81559359b@mail.gmail.com>
	<1EE3EB32-71F5-47EF-AE80-C5C5DBED0E34@avgs.ca>
	<g2la858e7691004230705h65a8131ds81b3836f015fa817@mail.gmail.com>
	<o2ia858e7691004230719hd5491240sfc2cbf92d6573abc@mail.gmail.com>
Message-ID: <o2xeafa772f1004231015qe0dc1498y6feedbe848044f57@mail.gmail.com>

If you're looking for help on writing new applications then of course
everything under src/mod/applications/*/ provide good examples.

There's also some documentation online of the APIs available to you
within FreeSWITCH:
http://docs.freeswitch.org/

In particular I recommend you look at the following sections
Modules -> Brought To You By APR
Modules -> Core Library

Core Library will tell you the APIs for interacting with
sessions/channels (calls) and provide you with useful interfaces for
doing may useful things like DB access, hash tables, file access etc,
while APR gives you a platform independant way of doing things like
like memory allocation, network access, threading etc. Using them will
save you much of the work and keeps the program style nicely
consistent across FreeSWITCH. Done right your module should also work
on any platform FS runs on without any changes needed (unless you app
depends on something that really is platform-specific such as loading
a DLL file on Windows).

-Steve



On 23 April 2010 15:19, Vitalii Colosov <vetali100 at gmail.com> wrote:
> OK I think I found, need to use:?SWITCH_ADD_APP
> Thank you very much!!!
>
> 2010/4/23 Vitalii Colosov <vetali100 at gmail.com>
>>
>> Thanks Mathieu,
>> This really helped!
>> I was able to get Hello World string during the module load :-)
>> Moving to the next step, how can I define an application, let say
>> "skel_app" inside this module?
>> In order to be able to execute it from dialplan.
>> I thought the following does this:?SWITCH_ADD_API(api_interface, "skel",
>> "Skel API", skel_function, "syntax");
>> But when I executed "skel" from dialplan, I got error: "Invalid
>> application skel".
>> Could you hint me please?
>> Thank you very much!
>> Vitalie
>>
>>
>> 2010/4/23 Mathieu Rene <mrene_lists at avgs.ca>
>>>
>>> Hi
>>>
>>> Look at mod_skel in src/mod/applications
>>>
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>>
>>> On 2010-04-23, at 6:43 AM, Vitalii Colosov wrote:
>>>
>>> > Hi dear group,
>>> >
>>> > I would like to start writing plugins (applications) for FreeSWITCH
>>> > that will be called from dialplan:
>>> >
>>> > <action application="my_cool_application"...>
>>> >
>>> > Could you please provide me some steps following which I will be able
>>> > to create "helloworld" C/C++ application that will be called from dialplan?
>>> >
>>> > Or some link if this is already described somewhere... I tried to
>>> > google but did not find that.
>>> >
>>> > Then, if I will create something useful, I will contribute it to the
>>> > community with pleasure... :-)
>>> >
>>> > Thank you very much!
>>> >
>>> > Vitalie
>>> > _______________________________________________
>>> > FreeSWITCH-dev mailing list
>>> > FreeSWITCH-dev at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From vetali100 at gmail.com  Fri Apr 23 11:54:06 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 23 Apr 2010 21:54:06 +0300
Subject: [Freeswitch-dev] Create applications in FreeSWITCH
In-Reply-To: <o2xeafa772f1004231015qe0dc1498y6feedbe848044f57@mail.gmail.com>
References: <h2ma858e7691004230343u2b609e13n4cbd56b81559359b@mail.gmail.com>
	<1EE3EB32-71F5-47EF-AE80-C5C5DBED0E34@avgs.ca>
	<g2la858e7691004230705h65a8131ds81b3836f015fa817@mail.gmail.com>
	<o2ia858e7691004230719hd5491240sfc2cbf92d6573abc@mail.gmail.com>
	<o2xeafa772f1004231015qe0dc1498y6feedbe848044f57@mail.gmail.com>
Message-ID: <p2ya858e7691004231154ydd4570c1g33e5c36260b51d32@mail.gmail.com>

Thanks for the advice, Steven!

I would like to thank all the responders, I managed to create an application
that uses core ODBC and is accessing MySQL table to get the needed
information.

It took less then 3 hours...

Taking into consideration that I did not have any idea from the beginning,
looks like the learning curve here is just perfectly optimized for the
newbies... :-)

FreeSWITCH really rocks!

Will keep learning.

Regards,
Vitalie

2010/4/23 Steven Ayre <steveayre at gmail.com>

> If you're looking for help on writing new applications then of course
> everything under src/mod/applications/*/ provide good examples.
>
> There's also some documentation online of the APIs available to you
> within FreeSWITCH:
> http://docs.freeswitch.org/
>
> In particular I recommend you look at the following sections
> Modules -> Brought To You By APR
> Modules -> Core Library
>
> Core Library will tell you the APIs for interacting with
> sessions/channels (calls) and provide you with useful interfaces for
> doing may useful things like DB access, hash tables, file access etc,
> while APR gives you a platform independant way of doing things like
> like memory allocation, network access, threading etc. Using them will
> save you much of the work and keeps the program style nicely
> consistent across FreeSWITCH. Done right your module should also work
> on any platform FS runs on without any changes needed (unless you app
> depends on something that really is platform-specific such as loading
> a DLL file on Windows).
>
> -Steve
>
>
>
> On 23 April 2010 15:19, Vitalii Colosov <vetali100 at gmail.com> wrote:
> > OK I think I found, need to use: SWITCH_ADD_APP
> > Thank you very much!!!
> >
> > 2010/4/23 Vitalii Colosov <vetali100 at gmail.com>
> >>
> >> Thanks Mathieu,
> >> This really helped!
> >> I was able to get Hello World string during the module load :-)
> >> Moving to the next step, how can I define an application, let say
> >> "skel_app" inside this module?
> >> In order to be able to execute it from dialplan.
> >> I thought the following does this: SWITCH_ADD_API(api_interface, "skel",
> >> "Skel API", skel_function, "syntax");
> >> But when I executed "skel" from dialplan, I got error: "Invalid
> >> application skel".
> >> Could you hint me please?
> >> Thank you very much!
> >> Vitalie
> >>
> >>
> >> 2010/4/23 Mathieu Rene <mrene_lists at avgs.ca>
> >>>
> >>> Hi
> >>>
> >>> Look at mod_skel in src/mod/applications
> >>>
> >>> Mathieu Rene
> >>> Avant-Garde Solutions Inc
> >>> Office: + 1 (514) 664-1044 x100
> >>> Cell: +1 (514) 664-1044 x200
> >>> mrene at avgs.ca
> >>>
> >>>
> >>>
> >>>
> >>> On 2010-04-23, at 6:43 AM, Vitalii Colosov wrote:
> >>>
> >>> > Hi dear group,
> >>> >
> >>> > I would like to start writing plugins (applications) for FreeSWITCH
> >>> > that will be called from dialplan:
> >>> >
> >>> > <action application="my_cool_application"...>
> >>> >
> >>> > Could you please provide me some steps following which I will be able
> >>> > to create "helloworld" C/C++ application that will be called from
> dialplan?
> >>> >
> >>> > Or some link if this is already described somewhere... I tried to
> >>> > google but did not find that.
> >>> >
> >>> > Then, if I will create something useful, I will contribute it to the
> >>> > community with pleasure... :-)
> >>> >
> >>> > Thank you very much!
> >>> >
> >>> > Vitalie
> >>> > _______________________________________________
> >>> > FreeSWITCH-dev mailing list
> >>> > FreeSWITCH-dev at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >>> > http://www.freeswitch.org
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-dev mailing list
> >>> FreeSWITCH-dev at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Apr 23 12:36:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 23 Apr 2010 12:36:55 -0700
Subject: [Freeswitch-dev] Create applications in FreeSWITCH
In-Reply-To: <p2ya858e7691004231154ydd4570c1g33e5c36260b51d32@mail.gmail.com>
References: <h2ma858e7691004230343u2b609e13n4cbd56b81559359b@mail.gmail.com>
	<1EE3EB32-71F5-47EF-AE80-C5C5DBED0E34@avgs.ca>
	<g2la858e7691004230705h65a8131ds81b3836f015fa817@mail.gmail.com>
	<o2ia858e7691004230719hd5491240sfc2cbf92d6573abc@mail.gmail.com>
	<o2xeafa772f1004231015qe0dc1498y6feedbe848044f57@mail.gmail.com>
	<p2ya858e7691004231154ydd4570c1g33e5c36260b51d32@mail.gmail.com>
Message-ID: <z2q87f2f3b91004231236j68f93013o4759d0e4a0cc1536@mail.gmail.com>

On Fri, Apr 23, 2010 at 11:54 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Thanks for the advice, Steven!
>
> I would like to thank all the responders, I managed to create an
> application that uses core ODBC and is accessing MySQL table to get the
> needed information.
>
This community is awesome! Welcome to the club...


> It took less then 3 hours...
>
Nice!


>
> Taking into consideration that I did not have any idea from the beginning,
> looks like the learning curve here is just perfectly optimized for the
> newbies... :-)
>
I think we need to frame this quote. :) In reality there's no way you can
build an app in 3 hours and call yourself a newbie. Still, you've probably
seen that the code is self-documenting and elegant, which means you can copy
code and ideas from other files and learn a lot about how everything works.


>
FreeSWITCH really rocks!
>
Agreed!


>
> Will keep learning.
>
Welcome to the community!

-MC
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From d at d-man.org  Sat Apr 24 12:41:50 2010
From: d at d-man.org (Darren Schreiber)
Date: Sat, 24 Apr 2010 12:41:50 -0700
Subject: [Freeswitch-dev] LAST REMINDER: FreeSWITCH Users Group & Install
 Fest is tomorrow in San Francisco!
Message-ID: <8A034A3098ED3C4990F7D9DE40F5585F17DA22F84E@EXVMBX020-3.exch020.serverdata.net>

Hi everyone,
                A final reminder that we?ll be doing a FreeSWITCH users group meet-up and install-fest tomorrow in San Francisco at Borders Books. It will be at 4pm. If you plan to attend please R.S.V.P. via the MeetUp website so I have an accurate headcount before the meeting.

                Look forward to seeing you Bay Area folks tomorrow!

http://www.meetup.com/fsusers/calendar/13009468/ for details, location & RSVP ability.

Sincerely,
                Darren Schreiber

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From vetali100 at gmail.com  Sun Apr 25 04:02:51 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 25 Apr 2010 14:02:51 +0300
Subject: [Freeswitch-dev] switch_odbc.c connect "max_tries" variable
Message-ID: <g2oa858e7691004250402s10347834s27382703d1965831@mail.gmail.com>

Hi,

I am using core ODBC for MySQL connect.

Trying to implement failover using 2 servers - main and backup.
If main server is not responding, it should try to query backup server.

The following is used in my module:

if (switch_odbc_handle_callback_exec(*globals.master_odbc*, sql,
my_func_callback, &pdata, NULL) != SWITCH_ODBC_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "(***)Error running
query on master database: [%s]\n", sql);

switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "(***)Trying backup
database: [%s]\n", globals.backup_odbc);


if (switch_odbc_handle_callback_exec(*globals.backup_odbc*,
sql, my_func_callback, &pdata, NULL) != SWITCH_ODBC_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "(***)Error running
query on master and backup database: [%s]\n", sql);
 }
 }
...



Right now it tries to reconnect to main server 120 times before it gives up
and goes to backup server.

I found the following in the switch_odbc.c:
int max_tries = 120;


Is there is a way to disable this check, so if it will not be able to
connect to master, it will immediate go to backup server?
I know I can change the source, but this will be already "customized core" -
don't want to go in this direction.


Or maybe there is a better way to handle database fail over in my module?


Thank you,
Vitalie
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From msc at freeswitch.org  Tue Apr 27 14:40:33 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Apr 2010 14:40:33 -0700
Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call April 28
Message-ID: <l2n87f2f3b91004271440x4ed4d068wb5d7d2e9abf57fc8@mail.gmail.com>

Hello all!

I've posted the agenda for tomorrow's call:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_28

Please get in there and add your items. It's been a bit hectic for me of
late so I appreciate any and all contributors to the agenda. If you have a
topic to discuss or a question to ask then definitely put it on the agenda
so that we can be ready for it.

Don't forget, we've pushed the meeting back one hour, so it starts at 1PM
Eastern, 10AM Pacific (1700 UTC)

See you all tomorrow!
-Michael
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From msc at freeswitch.org  Wed Apr 28 10:00:24 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Apr 2010 10:00:24 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <t2v87f2f3b91004281000n2ee96a1ep51dd9e86ff12f5b0@mail.gmail.com>

Come join us!

http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_28

-MC
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From errotan at elder.hu  Thu Apr 22 12:11:35 2010
From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Thu, 22 Apr 2010 19:11:35 -0000
Subject: [Freeswitch-dev] Fwd: thousands of memory leaks when using own
	module and mfc dll
In-Reply-To: <4BD087F8.2080504@lyth.de>
References: <4BD087F8.2080504@lyth.de>
Message-ID: <201004222110.33977.errotan@elder.hu>

2010. ?prilis 22. 19.31.36 d?tummal Sebastian Rudek az al?bbiakat ?rta:
> Dear Freeswitch developers,
> 
> i have a problem with FreeSwitch (Version 1.0.4) and my own module,
> which loads a MFC based DLL -- where obviously memory leaks are
> produced. let me try to explain what i have.
> 1) First of all, let's discuss the case where everything works fine in
> freeswitch. My FreeSwitch module (which here loads no further DLL) is
> loaded and unloaded correctly when a call is accepted and later on hung up.
> 2) Then i load an "empty" DLL, without MFC and all that stuff, which i
> created, within my FreeSwitch module. I do this when accepting a call.
> when hanging up, I FreeLibrary(), quit FreeSwitch and everything is
> fine. No memory leaks, no errors, just everything as it ought to be.
> 3) But then! Then i take my module and load a DLL which uses MFC. and
> then, when after unloading the DLL on hanging up, I exit the FreeSwitch
> application and get that many memory leak reports in the debug out of
> visual studio 2008... that leads me to my question: Are there any know
> issues with FreeSwitch and MFC DLLs ?
> 
> I tried to simulate the problem with creating an Application which
> simply loads the Non-MFC DLL (see number 2 above). No memory leaks.
> Also, no memory leaks are detected when loading the MFC-DLL from 3) (see
> above). but Visual Studio detects one (!) 4-byte-memory-leak when
> loading the MFC DLL inside the Non-MFC-Dll (like the FreeSwitch module
> ought to do it).
> 
> Back to the question, if there is an issue with FreeSwitch and MFC-DLLs.
> The object dump, which Visual Studio gives to me, says that leaks are
> detected at "VARCHAR(255)", "shma", "TeleTone" or "loopback.auto". I do
> not know where that comes from. Maybe, you have an idea which you could
> share with me to solve the problem, on which i have spent really much
> time already...
> I just attach the out as text file.
> 
> Thank you so much in advance
> 
> Sebastian

You should try the latest git version, 1.0.4 is very old:

http://wiki.freeswitch.org/wiki/Download_FreeSWITCH



From rajkiran.talusani at gmail.com  Thu Apr 22 22:29:33 2010
From: rajkiran.talusani at gmail.com (Raj Kiran Talusani)
Date: Fri, 23 Apr 2010 05:29:33 -0000
Subject: [Freeswitch-dev] sending custom SDP in 200 OK
Message-ID: <u2waf030d981004222228tcd36a6f7t6c7dbc29581b4c87@mail.gmail.com>

I have written a application called mod_custom. In this module i would like
to set a user-defined RTP port and IP adress and then send the 200 OK
(answer the call). But i see that, before my application is invoked, 183
message is sent with SDP. And i didnot find a way to either disable sending
183 and just send 180 OR modify the SDP to be sent in 200 OK.

please help.

Rajk
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From mwhite at thesummit-grp.com  Wed Apr 28 15:33:54 2010
From: mwhite at thesummit-grp.com (Matt White)
Date: Wed, 28 Apr 2010 18:33:54 -0400
Subject: [Freeswitch-dev] Build fails on Suse
Message-ID: <4BD87F92020000F000017D82@firewall.thesummit-grp.com>


I'm using the opensuse build service to create rpm's for Suse. 

The build fails with the following error below.  I think its due to the gcc 4.3 used on SLE11 as I can't replicate it in older versions 


I: Expression compares a char* pointer with a string literal. 
   Usually a strcmp() was intended by the programmer 
E: freeswitch stringcompare strings/apr_snprintf.c:1261 

Any thoughts?  I'm using the nightly snapshot.   

-M 
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From khovayko at gmail.com  Fri Apr 30 20:37:24 2010
From: khovayko at gmail.com (Oleg Khovayko)
Date: Fri, 30 Apr 2010 23:37:24 -0400
Subject: [Freeswitch-dev] portaudio help
Message-ID: <4BDBA1F4.3020709@gmail.com>

Dear Developers,

Can you help me with portaudio module?


On the my machine (FreeBSD OS),
I have active sound card and driver:

{{{

dmesg | grep pcm
pcm0: <ESS 18xx DSP> on sbc0
pcm0: [GIANT-LOCKED]
pcm0: [ITHREAD]

}}}

Also, when I send some random file to device, I hear noise:

cat mime.types > /dev/dsp



In the freeswitch config, I have enabled load module portaudio.

Devlist shows device OK:



freeswitch at internal> pa devlist xml
<xml>
<devices>
<device id="0" name="/dev/dsp" inputs="2" outputs="2" />
</devices>
<bindings>
<ring device="0" />
<input device="0" />
<output device="0" />
</bindings>
</xml>



*************************************************************************
But, when I try to use "pa" commands, I see message "FAIL:Device Error!"
*************************************************************************


freeswitch at internal> pa call 5000
FAIL:Device Error!

freeswitch at internal> 2010-04-30 18:37:42.325070 [NOTICE] 
switch_channel.c:642 New Channel portaudio/ 
[97a42efd-a854-df11-b1f2-005004c3cb7e]
2010-04-30 18:37:42.873149 [ERR] mod_portaudio.c:1394 Error opening 
audio device retrying
2010-04-30 18:37:44.384778 [ERR] mod_portaudio.c:1403 Can't open audio 
device
2010-04-30 18:37:44.384778 [NOTICE] mod_portaudio.c:1784 Close Channel 
portaudio/ [CS_NEW]

freeswitch at internal> pa call 1001
FAIL:Device Error!

freeswitch at internal> 2010-04-30 18:38:00.223324 [NOTICE] 
switch_channel.c:642 New Channel portaudio/ 
[5456da07-a954-df11-b1f2-005004c3cb7e]
2010-04-30 18:38:00.754025 [ERR] mod_portaudio.c:1394 Error opening 
audio device retrying
2010-04-30 18:38:02.264923 [ERR] mod_portaudio.c:1403 Can't open audio 
device
2010-04-30 18:38:02.264923 [NOTICE] mod_portaudio.c:1784 Close Channel 
portaudio/ [CS_NEW]
2010-04-30 18:38:15.264125 [NOTICE] sofia_reg.c:340 Registering 
callwithus.com

freeswitch at internal> pa dump
-ERR no reply

freeswitch at internal> 2010-04-30 18:51:09.345048 [INFO] 
mod_portaudio.c:1163 PortAudio version number = 1899
PortAudio version text = 'PortAudio V19-devel (built Dec  4 2009)'
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1187 Number of devices = 1
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1194 
--------------------------------------------------------------------------------
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1196 Device #0 
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1201 **Default 
Input2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1212 **Default 
Output2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1223 ** | 
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1226 Name: /dev/dsp
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1227 Host: OSS | 
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1228 inputs: 2 | 
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1229 outputs: 2 | 
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1231 Default rate: 
44100.00
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1233 Default input 
latency: 0.012 | 2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1234 
Default output latency: 0.012
2010-04-30 18:51:09.345048 [INFO] mod_portaudio.c:1251 half-duplex 16 
bit 2 channel input rates:
         8000.00, 9600.00, 11025.00, 12000.00, 16000.00, 22050.00, 24000.00
         32000.00, 44100.00, 48000.00, 88200.00, 96000.00, 192000.00
2010-04-30 18:51:09.365448 [INFO] mod_portaudio.c:1256 half-duplex 16 
bit 2 channel output rates:
         8000.00, 9600.00, 11025.00, 12000.00, 16000.00, 22050.00, 24000.00
         32000.00, 44100.00, 48000.00, 88200.00, 96000.00, 192000.00
2010-04-30 18:51:09.365448 [INFO] mod_portaudio.c:1262 full-duplex 16 
bit 2 channel input, 2 channel output rates:
         8000.00, 9600.00, 11025.00, 12000.00, 16000.00, 22050.00, 24000.00
         32000.00, 44100.00, 48000.00, 88200.00, 96000.00, 192000.00
2010-04-30 18:51:09.637693 [INFO] mod_portaudio.c:1269 
--------------------------------------------------------------------------------


freeswitch at internal> /exit







olegh# ls /dev/ds*
/dev/dsp0.0
olegh#