From jsun at junsun.net  Sat Feb  2 07:37:10 2013
From: jsun at junsun.net (Jun Sun)
Date: Fri, 01 Feb 2013 20:37:10 -0800
Subject: [Freeswitch-dev] [patch] build with the recent binutils (DSO link
	change)
Message-ID: <510C97F6.5080807@junsun.net>


When I tried to build against the latest 64bit amazon ec2 instance, I 
got the following error:

----
...
checking for tgetent in -lncurses... no
checking for tgetent in -lcurses... no
configure: error: libtermcap, libcurses or libncurses are required!
----

A little bit research shows there was a ld behavior change recently. See 
the link below.

http://fedoraproject.org/wiki/UnderstandingDSOLinkChange

The attached patch fixes the problem. I *believe* the patch should be 
safe to older systems, but someone would have check it out.

Cheers.

Jun
-------------- next part --------------
diff -Nru freeswitch-1.2.5.3/libs/libedit/configure.orig freeswitch-1.2.5.3/libs/libedit/configure
--- freeswitch-1.2.5.3/libs/libedit/configure.orig	2012-12-07 15:22:41.000000000 +0000
+++ freeswitch-1.2.5.3/libs/libedit/configure	2013-02-02 03:58:05.043776662 +0000
@@ -15200,7 +15200,7 @@
   $as_echo_n "(cached) " >&6
 else
   ac_check_lib_save_LIBS=$LIBS
-LIBS="-lncurses  $LIBS"
+LIBS="-lncurses -ltinfo $LIBS"
 cat confdefs.h - <<_ACEOF >conftest.$ac_ext
 /* end confdefs.h.  */
 
@@ -15235,7 +15235,7 @@
 #define HAVE_LIBNCURSES 1
 _ACEOF
 
-  LIBS="-lncurses $LIBS"
+  LIBS="-lncurses -ltinfo $LIBS"
 
 else
   as_fn_error $? "libtermcap, libcurses or libncurses are required!" "$LINENO" 5
diff -Nru freeswitch-1.2.5.3/libs/esl/Makefile.orig freeswitch-1.2.5.3/libs/esl/Makefile
--- freeswitch-1.2.5.3/libs/esl/Makefile.orig	2012-12-07 15:21:14.000000000 +0000
+++ freeswitch-1.2.5.3/libs/esl/Makefile	2013-02-02 04:15:22.816032763 +0000
@@ -7,7 +7,7 @@
 CFLAGS=$(BASE_FLAGS) $(PICKY)
 CXXFLAGS=$(BASE_FLAGS)
 MYLIB=libesl.a
-LIBS=-lncurses -lesl -lpthread -lm
+LIBS=-lncurses -ltinfo -lesl -lpthread -lm
 LDFLAGS=-L.
 OBJS=src/esl.o src/esl_event.o src/esl_threadmutex.o src/esl_config.o src/esl_json.o src/esl_buffer.o
 SRC=src/esl.c src/esl_json.c src/esl_event.c src/esl_threadmutex.c src/esl_config.c src/esl_oop.cpp src/esl_json.c src/esl_buffer.c
diff -Nru freeswitch-1.2.5.3/configure.orig freeswitch-1.2.5.3/configure
--- freeswitch-1.2.5.3/configure.orig	2012-12-07 15:24:03.000000000 +0000
+++ freeswitch-1.2.5.3/configure	2013-02-02 03:57:07.287330057 +0000
@@ -21763,7 +21763,7 @@
   $as_echo_n "(cached) " >&6
 else
   ac_check_lib_save_LIBS=$LIBS
-LIBS="-lncurses  $LIBS"
+LIBS="-lncurses -ltinfo $LIBS"
 cat confdefs.h - <<_ACEOF >conftest.$ac_ext
 /* end confdefs.h.  */
 
@@ -21798,7 +21798,7 @@
 #define HAVE_LIBNCURSES 1
 _ACEOF
 
-  LIBS="-lncurses $LIBS"
+  LIBS="-lncurses -ltinfo $LIBS"
 
 else
   { $as_echo "$as_me:${as_lineno-$LINENO}: checking for tgetent in -lcurses" >&5

From krice at freeswitch.org  Sat Feb  2 18:46:50 2013
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 02 Feb 2013 09:46:50 -0600
Subject: [Freeswitch-dev] [patch] build with the recent binutils (DSO
 link change)
In-Reply-To: <510C97F6.5080807@junsun.net>
Message-ID: <CD32910A.6B196%krice@freeswitch.org>

Bugs and Patches to go to http://jira.freeswitch.org if you want them to be
considered for inclusion

K


On 2/1/13 10:37 PM, "Jun Sun" <jsun at junsun.net> wrote:

> 
> When I tried to build against the latest 64bit amazon ec2 instance, I
> got the following error:
> 
> ----
> ...
> checking for tgetent in -lncurses... no
> checking for tgetent in -lcurses... no
> configure: error: libtermcap, libcurses or libncurses are required!
> ----
> 
> A little bit research shows there was a ld behavior change recently. See
> the link below.
> 
> http://fedoraproject.org/wiki/UnderstandingDSOLinkChange
> 
> The attached patch fixes the problem. I *believe* the patch should be
> safe to older systems, but someone would have check it out.
> 
> Cheers.
> 
> Jun
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From fdelawarde at wirelessmundi.com  Mon Feb  4 13:54:07 2013
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Mon, 04 Feb 2013 11:54:07 +0100
Subject: [Freeswitch-dev] DTMF duration unit
Message-ID: <1359975247.8347.78.camel@luna.madrid.commsmundi.com>

Hi all,

I don't clearly get what unit we should use in dtmf-duration parameters
(core min-dtmf-duration / max-dtmf-duration and sofia dtmf-duration).

The variable name and some parts of the code/wiki imply it's in
milliseconds (for example in sofia.c, we can see a line saying
profile->dtmf_duration = 100;).

However, the default values in switch_types.h suggest the values should
be based on timestamp like in RTP. Something like:

minimum = 400 @ 8kHz => 50ms
default = 2000 @ 8kHz => 250ms
maximum = 192000 @ 8kHz => 24s


If the later is the right one, should I change values when using WB
codecs that are NOT sampling at 8kHz?

Also, is the "unit" consistent across all endpoint modules?


Thanks,
Fran?ois.



From dxj19831029 at gmail.com  Tue Feb  5 13:23:28 2013
From: dxj19831029 at gmail.com (Xijing Dai)
Date: Tue, 5 Feb 2013 18:23:28 +0800
Subject: [Freeswitch-dev] question about CNG on late packages
Message-ID: <CAMsCbX9jux9bn-V3GZp7CnseUczXTXP_z6_VsCJSNL=PVzdeNQ@mail.gmail.com>

Hey all,


Inside code

switch_rtp.c file:

   static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t
*payload_type, switch_frame_flag_t *flags, switch_io_flag_t io_flags)
                 ................
                 if (check || (bytes && !switch_test_flag(rtp_session,
SWITCH_RTP_FLAG_USE_TIMER))) {
if (!bytes && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER)) { /*
We're late! We're Late! */
if (!switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK) && status ==
SWITCH_STATUS_BREAK) {
switch_cond_next();
continue;
}

return_cng_frame();  <------- this lane to return cng frame if late.
}
}


Could someone explain why we want to return cng frame when package is late?
this is caused for me to hear very bad noise audio.

Can I disable return CNG frame?


Cheers
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From steveayre at gmail.com  Tue Feb  5 13:32:49 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Feb 2013 10:32:49 +0000
Subject: [Freeswitch-dev] question about CNG on late packages
In-Reply-To: <CAMsCbX9jux9bn-V3GZp7CnseUczXTXP_z6_VsCJSNL=PVzdeNQ@mail.gmail.com>
References: <CAMsCbX9jux9bn-V3GZp7CnseUczXTXP_z6_VsCJSNL=PVzdeNQ@mail.gmail.com>
Message-ID: <CAFiqYun1gDVe637o5zKv3giGHAw281essMhHKj0NC9MhUOLqDA@mail.gmail.com>

Because if a RTP packet is due and hasn't arrived yet you need something to
play. Audio isn't like TCP where you dropped packets can be retransmitted,
if it hasn't arrived by the time you want to play it it's too late and you
have to play silence to handle it.

Your audio problems are more likely due to packet loss or jitter which is
why the RTP packet is overdue, not the CNG itself.

It jitter's the problem enabling the jitterbuffer might help (at the
expense of increased lag). If loss is the problem there's no way to handle
that except as silence for the missing interval, you need to fix whatever
is causing the loss.

-Steve


On 5 February 2013 10:23, Xijing Dai <dxj19831029 at gmail.com> wrote:

> Hey all,
>
>
> Inside code
>
> switch_rtp.c file:
>
>    static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t
> *payload_type, switch_frame_flag_t *flags, switch_io_flag_t io_flags)
>                  ................
>                  if (check || (bytes && !switch_test_flag(rtp_session,
> SWITCH_RTP_FLAG_USE_TIMER))) {
> if (!bytes && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER)) { /*
> We're late! We're Late! */
>  if (!switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK) && status ==
> SWITCH_STATUS_BREAK) {
> switch_cond_next();
>  continue;
> }
>
> return_cng_frame();  <------- this lane to return cng frame if late.
>  }
> }
>
>
> Could someone explain why we want to return cng frame when package is
> late? this is caused for me to hear very bad noise audio.
>
> Can I disable return CNG frame?
>
>
> Cheers
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
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From fdelawarde at wirelessmundi.com  Tue Feb  5 13:47:37 2013
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Feb 2013 11:47:37 +0100
Subject: [Freeswitch-dev] [Freeswitch-users] DTMF duration unit
In-Reply-To: <1359975247.8347.78.camel@luna.madrid.commsmundi.com>
References: <1359975247.8347.78.camel@luna.madrid.commsmundi.com>
Message-ID: <1360061257.8347.197.camel@luna.madrid.commsmundi.com>

Reply to my own questions:

On Mon, 2013-02-04 at 11:54 +0100, Fran?ois Delawarde wrote: 
> I don't clearly get what unit we should use in dtmf-duration parameters
> (core min-dtmf-duration / max-dtmf-duration and sofia dtmf-duration).


"*dtmf-duration" variables should be specified in timestamp units, as in
RFC-2833.



> Also, is the "unit" consistent across all endpoint modules?

For SIP at least, the behavior seems a bit inconsistent:


* RFC-2833, FS always specifies a timestamp rate 8000Hz (the default) in
rtpmap (telephone-event/8000), so the default values are:

min-dtmf-duration=400 => 50ms
default-dtmf-duration=2000 => 250ms
max-dtmf-duration=192000 => 24s


* SIP INFO, FS feeds the same values into the "Duration=" header of SIP
INFO messages, but these are interpreted as milliseconds, so when using
SIP INFO, the default values end up being:

min-dtmf-duration=400 => 400ms
default-dtmf-duration=2000 => 2s
max-dtmf-duration=192000 => 192s


* INBAND, teletone probably uses milliseconds as well, did not check.



So as long as you are using RFC-2833, everything should be okay. I don't
know of any device requiring SIP INFO anyway, do you?

I can open a jira if a dev confirms the behavior is not intended.

Regards,
Fran?ois.



From anthony.minessale at gmail.com  Tue Feb  5 18:35:02 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Feb 2013 09:35:02 -0600
Subject: [Freeswitch-dev] question about CNG on late packages
In-Reply-To: <CAFiqYun1gDVe637o5zKv3giGHAw281essMhHKj0NC9MhUOLqDA@mail.gmail.com>
References: <CAMsCbX9jux9bn-V3GZp7CnseUczXTXP_z6_VsCJSNL=PVzdeNQ@mail.gmail.com>
	<CAFiqYun1gDVe637o5zKv3giGHAw281essMhHKj0NC9MhUOLqDA@mail.gmail.com>
Message-ID: <CAKbxfG8MW6yY1JA8YDvtRoViF3HhCDZD1Q1GHUn9f06Q=J3_Mw@mail.gmail.com>

Don't negotiate CNG in the SDP and it won't use it or disable it with
suppress_cng=true in your channel or global vars.



On Tue, Feb 5, 2013 at 4:32 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Because if a RTP packet is due and hasn't arrived yet you need something
> to play. Audio isn't like TCP where you dropped packets can be
> retransmitted, if it hasn't arrived by the time you want to play it it's
> too late and you have to play silence to handle it.
>
> Your audio problems are more likely due to packet loss or jitter which is
> why the RTP packet is overdue, not the CNG itself.
>
> It jitter's the problem enabling the jitterbuffer might help (at the
> expense of increased lag). If loss is the problem there's no way to handle
> that except as silence for the missing interval, you need to fix whatever
> is causing the loss.
>
> -Steve
>
>
> On 5 February 2013 10:23, Xijing Dai <dxj19831029 at gmail.com> wrote:
>
>> Hey all,
>>
>>
>> Inside code
>>
>> switch_rtp.c file:
>>
>>    static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t
>> *payload_type, switch_frame_flag_t *flags, switch_io_flag_t io_flags)
>>                  ................
>>                  if (check || (bytes && !switch_test_flag(rtp_session,
>> SWITCH_RTP_FLAG_USE_TIMER))) {
>> if (!bytes && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER)) { /*
>> We're late! We're Late! */
>>  if (!switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK) && status
>> == SWITCH_STATUS_BREAK) {
>> switch_cond_next();
>>  continue;
>> }
>>
>> return_cng_frame();  <------- this lane to return cng frame if late.
>>  }
>> }
>>
>>
>> Could someone explain why we want to return cng frame when package is
>> late? this is caused for me to hear very bad noise audio.
>>
>> Can I disable return CNG frame?
>>
>>
>> Cheers
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From dxj19831029 at gmail.com  Tue Feb  5 19:32:09 2013
From: dxj19831029 at gmail.com (Xijing Dai)
Date: Wed, 6 Feb 2013 00:32:09 +0800
Subject: [Freeswitch-dev] question about CNG on late packages
In-Reply-To: <CAKbxfG8MW6yY1JA8YDvtRoViF3HhCDZD1Q1GHUn9f06Q=J3_Mw@mail.gmail.com>
References: <CAMsCbX9jux9bn-V3GZp7CnseUczXTXP_z6_VsCJSNL=PVzdeNQ@mail.gmail.com>
	<CAFiqYun1gDVe637o5zKv3giGHAw281essMhHKj0NC9MhUOLqDA@mail.gmail.com>
	<CAKbxfG8MW6yY1JA8YDvtRoViF3HhCDZD1Q1GHUn9f06Q=J3_Mw@mail.gmail.com>
Message-ID: <CAMsCbX8k5DScricvdZJ-FXmZJ-zWRz=52B12YUcCdeG3jX09Ew@mail.gmail.com>

I see.


I did set it suppress_cng=true, and it did not help. No difference at all.

This happens inside LAN.

When talking about the jitter buffer, we want to enable it on client, and
disable it inside freeswitch.
Therefore, in this case, should freeswitch do nothing?


Cheers




On Tue, Feb 5, 2013 at 11:35 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Don't negotiate CNG in the SDP and it won't use it or disable it with
> suppress_cng=true in your channel or global vars.
>
>
>
> On Tue, Feb 5, 2013 at 4:32 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Because if a RTP packet is due and hasn't arrived yet you need something
>> to play. Audio isn't like TCP where you dropped packets can be
>> retransmitted, if it hasn't arrived by the time you want to play it it's
>> too late and you have to play silence to handle it.
>>
>> Your audio problems are more likely due to packet loss or jitter which is
>> why the RTP packet is overdue, not the CNG itself.
>>
>> It jitter's the problem enabling the jitterbuffer might help (at the
>> expense of increased lag). If loss is the problem there's no way to handle
>> that except as silence for the missing interval, you need to fix whatever
>> is causing the loss.
>>
>> -Steve
>>
>>
>> On 5 February 2013 10:23, Xijing Dai <dxj19831029 at gmail.com> wrote:
>>
>>> Hey all,
>>>
>>>
>>> Inside code
>>>
>>> switch_rtp.c file:
>>>
>>>    static int rtp_common_read(switch_rtp_t *rtp_session,
>>> switch_payload_t *payload_type, switch_frame_flag_t *flags,
>>> switch_io_flag_t io_flags)
>>>                  ................
>>>                  if (check || (bytes && !switch_test_flag(rtp_session,
>>> SWITCH_RTP_FLAG_USE_TIMER))) {
>>> if (!bytes && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER)) { /*
>>> We're late! We're Late! */
>>>  if (!switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK) && status
>>> == SWITCH_STATUS_BREAK) {
>>> switch_cond_next();
>>>  continue;
>>> }
>>>
>>> return_cng_frame();  <------- this lane to return cng frame if late.
>>>  }
>>> }
>>>
>>>
>>> Could someone explain why we want to return cng frame when package is
>>> late? this is caused for me to hear very bad noise audio.
>>>
>>> Can I disable return CNG frame?
>>>
>>>
>>> Cheers
>>>
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
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From dxj19831029 at gmail.com  Tue Feb  5 19:48:20 2013
From: dxj19831029 at gmail.com (Xijing Dai)
Date: Wed, 6 Feb 2013 00:48:20 +0800
Subject: [Freeswitch-dev] question about CNG on late packages
In-Reply-To: <CAMsCbX8k5DScricvdZJ-FXmZJ-zWRz=52B12YUcCdeG3jX09Ew@mail.gmail.com>
References: <CAMsCbX9jux9bn-V3GZp7CnseUczXTXP_z6_VsCJSNL=PVzdeNQ@mail.gmail.com>
	<CAFiqYun1gDVe637o5zKv3giGHAw281essMhHKj0NC9MhUOLqDA@mail.gmail.com>
	<CAKbxfG8MW6yY1JA8YDvtRoViF3HhCDZD1Q1GHUn9f06Q=J3_Mw@mail.gmail.com>
	<CAMsCbX8k5DScricvdZJ-FXmZJ-zWRz=52B12YUcCdeG3jX09Ew@mail.gmail.com>
Message-ID: <CAMsCbX9HTD_Sjro=jRHvo-RD7DgcPUgfhkr_-L5ywCX-vWht0Q@mail.gmail.com>

by the way, what is the latency in ms to determine if a packet is late?


Our ptime is 10ms, and the sender always sends packets in 8-12ms.

And I did use wireshark to capture packets on freeswitch server, no lost
packet is found before reaching freeswitch.

When our ptime is 20ms, it gets better, but still has CN.

When we are using proxy/proxy media, every thing works great.

when sample rate is 16000, the quality of call is worst.  8000 gets better.


Any other suggestions?

On Wed, Feb 6, 2013 at 12:32 AM, Xijing Dai <dxj19831029 at gmail.com> wrote:

> I see.
>
>
> I did set it suppress_cng=true, and it did not help. No difference at all.
>
> This happens inside LAN.
>
> When talking about the jitter buffer, we want to enable it on client, and
> disable it inside freeswitch.
> Therefore, in this case, should freeswitch do nothing?
>
>
> Cheers
>
>
>
>
> On Tue, Feb 5, 2013 at 11:35 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Don't negotiate CNG in the SDP and it won't use it or disable it with
>> suppress_cng=true in your channel or global vars.
>>
>>
>>
>> On Tue, Feb 5, 2013 at 4:32 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Because if a RTP packet is due and hasn't arrived yet you need something
>>> to play. Audio isn't like TCP where you dropped packets can be
>>> retransmitted, if it hasn't arrived by the time you want to play it it's
>>> too late and you have to play silence to handle it.
>>>
>>> Your audio problems are more likely due to packet loss or jitter which
>>> is why the RTP packet is overdue, not the CNG itself.
>>>
>>> It jitter's the problem enabling the jitterbuffer might help (at the
>>> expense of increased lag). If loss is the problem there's no way to handle
>>> that except as silence for the missing interval, you need to fix whatever
>>> is causing the loss.
>>>
>>> -Steve
>>>
>>>
>>> On 5 February 2013 10:23, Xijing Dai <dxj19831029 at gmail.com> wrote:
>>>
>>>> Hey all,
>>>>
>>>>
>>>> Inside code
>>>>
>>>> switch_rtp.c file:
>>>>
>>>>    static int rtp_common_read(switch_rtp_t *rtp_session,
>>>> switch_payload_t *payload_type, switch_frame_flag_t *flags,
>>>> switch_io_flag_t io_flags)
>>>>                  ................
>>>>                  if (check || (bytes && !switch_test_flag(rtp_session,
>>>> SWITCH_RTP_FLAG_USE_TIMER))) {
>>>> if (!bytes && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER))
>>>> { /* We're late! We're Late! */
>>>>  if (!switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK) && status
>>>> == SWITCH_STATUS_BREAK) {
>>>> switch_cond_next();
>>>>  continue;
>>>> }
>>>>
>>>> return_cng_frame();  <------- this lane to return cng frame if late.
>>>>  }
>>>> }
>>>>
>>>>
>>>> Could someone explain why we want to return cng frame when package is
>>>> late? this is caused for me to hear very bad noise audio.
>>>>
>>>> Can I disable return CNG frame?
>>>>
>>>>
>>>> Cheers
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
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From enst.bupt at gmail.com  Tue Feb  5 19:52:42 2013
From: enst.bupt at gmail.com (Bing LI)
Date: Tue, 5 Feb 2013 11:52:42 -0500
Subject: [Freeswitch-dev] question about CNG on late packages
In-Reply-To: <CAMsCbX8k5DScricvdZJ-FXmZJ-zWRz=52B12YUcCdeG3jX09Ew@mail.gmail.com>
References: <CAMsCbX9jux9bn-V3GZp7CnseUczXTXP_z6_VsCJSNL=PVzdeNQ@mail.gmail.com>
	<CAFiqYun1gDVe637o5zKv3giGHAw281essMhHKj0NC9MhUOLqDA@mail.gmail.com>
	<CAKbxfG8MW6yY1JA8YDvtRoViF3HhCDZD1Q1GHUn9f06Q=J3_Mw@mail.gmail.com>
	<CAMsCbX8k5DScricvdZJ-FXmZJ-zWRz=52B12YUcCdeG3jX09Ew@mail.gmail.com>
Message-ID: <CAP9o=2=wT4aVyfcVrFMBOaJkoJ=1ahACfrLTzfKMiuSc7Hku1A@mail.gmail.com>

In my experience suppress_cng did work but only prevented sending CNG
packet.
freeswitch would ignore the late packets because it thought the cng packet
had been sent out. it makes a hole in the seq number in B-leg.

So later I modified the switch_rtp.c to change this mechanism.



On Tue, Feb 5, 2013 at 11:32 AM, Xijing Dai <dxj19831029 at gmail.com> wrote:

> I see.
>
>
> I did set it suppress_cng=true, and it did not help. No difference at all.
>
> This happens inside LAN.
>
> When talking about the jitter buffer, we want to enable it on client, and
> disable it inside freeswitch.
> Therefore, in this case, should freeswitch do nothing?
>
>
> Cheers
>
>
>
>
> On Tue, Feb 5, 2013 at 11:35 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Don't negotiate CNG in the SDP and it won't use it or disable it with
>> suppress_cng=true in your channel or global vars.
>>
>>
>>
>> On Tue, Feb 5, 2013 at 4:32 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Because if a RTP packet is due and hasn't arrived yet you need something
>>> to play. Audio isn't like TCP where you dropped packets can be
>>> retransmitted, if it hasn't arrived by the time you want to play it it's
>>> too late and you have to play silence to handle it.
>>>
>>> Your audio problems are more likely due to packet loss or jitter which
>>> is why the RTP packet is overdue, not the CNG itself.
>>>
>>> It jitter's the problem enabling the jitterbuffer might help (at the
>>> expense of increased lag). If loss is the problem there's no way to handle
>>> that except as silence for the missing interval, you need to fix whatever
>>> is causing the loss.
>>>
>>> -Steve
>>>
>>>
>>> On 5 February 2013 10:23, Xijing Dai <dxj19831029 at gmail.com> wrote:
>>>
>>>> Hey all,
>>>>
>>>>
>>>> Inside code
>>>>
>>>> switch_rtp.c file:
>>>>
>>>>    static int rtp_common_read(switch_rtp_t *rtp_session,
>>>> switch_payload_t *payload_type, switch_frame_flag_t *flags,
>>>> switch_io_flag_t io_flags)
>>>>                  ................
>>>>                  if (check || (bytes && !switch_test_flag(rtp_session,
>>>> SWITCH_RTP_FLAG_USE_TIMER))) {
>>>> if (!bytes && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER))
>>>> { /* We're late! We're Late! */
>>>>  if (!switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK) && status
>>>> == SWITCH_STATUS_BREAK) {
>>>> switch_cond_next();
>>>>  continue;
>>>> }
>>>>
>>>> return_cng_frame();  <------- this lane to return cng frame if late.
>>>>  }
>>>> }
>>>>
>>>>
>>>> Could someone explain why we want to return cng frame when package is
>>>> late? this is caused for me to hear very bad noise audio.
>>>>
>>>> Can I disable return CNG frame?
>>>>
>>>>
>>>> Cheers
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
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From POlsson at enghouse.com  Tue Feb  5 20:12:00 2013
From: POlsson at enghouse.com (Peter Olsson)
Date: Tue, 5 Feb 2013 17:12:00 +0000
Subject: [Freeswitch-dev] question about CNG on late packages
In-Reply-To: <CAMsCbX9HTD_Sjro=jRHvo-RD7DgcPUgfhkr_-L5ywCX-vWht0Q@mail.gmail.com>
References: <CAMsCbX9jux9bn-V3GZp7CnseUczXTXP_z6_VsCJSNL=PVzdeNQ@mail.gmail.com>
	<CAFiqYun1gDVe637o5zKv3giGHAw281essMhHKj0NC9MhUOLqDA@mail.gmail.com>
	<CAKbxfG8MW6yY1JA8YDvtRoViF3HhCDZD1Q1GHUn9f06Q=J3_Mw@mail.gmail.com>
	<CAMsCbX8k5DScricvdZJ-FXmZJ-zWRz=52B12YUcCdeG3jX09Ew@mail.gmail.com>,
	<CAMsCbX9HTD_Sjro=jRHvo-RD7DgcPUgfhkr_-L5ywCX-vWht0Q@mail.gmail.com>
Message-ID: <1FFF97C269757C458224B7C895F35F15217D13@cantor.std.visionutv.se>

There are lots of probable causes to this. First of all, the timing in your server might be bad. What does timer_test return? Is this a real machine, or is it a virtual one?

I agree with the others saying that the CNG itself is probably not a problem. However, I know a minor changes was made some months back (in head) that made sure not to cause holes after a CNG frame was handled (I know, since I was the one reporting the issue - and it was fixed by Tony). But this problem didn't really cause bad audio, it made the audio disappear for a few seconds.

First of all, make sure you are using a current version, and not something from way back. If the issue still exist, and the timing on the machine seems ok, and you are not on a virtual server, please add logs and pcaps to a Jira issue. You might also need to enable some extra RTP debugging features in the RTP stack to get all logs needed.

/Peter

________________________________
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f?r Xijing Dai [dxj19831029 at gmail.com]
Skickat: den 5 februari 2013 17:48
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] question about CNG on late packages

by the way, what is the latency in ms to determine if a packet is late?


Our ptime is 10ms, and the sender always sends packets in 8-12ms.

And I did use wireshark to capture packets on freeswitch server, no lost packet is found before reaching freeswitch.

When our ptime is 20ms, it gets better, but still has CN.

When we are using proxy/proxy media, every thing works great.

when sample rate is 16000, the quality of call is worst.  8000 gets better.


Any other suggestions?

On Wed, Feb 6, 2013 at 12:32 AM, Xijing Dai <dxj19831029 at gmail.com<mailto:dxj19831029 at gmail.com>> wrote:
I see.


I did set it suppress_cng=true, and it did not help. No difference at all.

This happens inside LAN.

When talking about the jitter buffer, we want to enable it on client, and disable it inside freeswitch.
Therefore, in this case, should freeswitch do nothing?


Cheers




On Tue, Feb 5, 2013 at 11:35 PM, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
Don't negotiate CNG in the SDP and it won't use it or disable it with suppress_cng=true in your channel or global vars.



On Tue, Feb 5, 2013 at 4:32 AM, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
Because if a RTP packet is due and hasn't arrived yet you need something to play. Audio isn't like TCP where you dropped packets can be retransmitted, if it hasn't arrived by the time you want to play it it's too late and you have to play silence to handle it.

Your audio problems are more likely due to packet loss or jitter which is why the RTP packet is overdue, not the CNG itself.

It jitter's the problem enabling the jitterbuffer might help (at the expense of increased lag). If loss is the problem there's no way to handle that except as silence for the missing interval, you need to fix whatever is causing the loss.

-Steve


On 5 February 2013 10:23, Xijing Dai <dxj19831029 at gmail.com<mailto:dxj19831029 at gmail.com>> wrote:
Hey all,


Inside code

switch_rtp.c file:

   static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t *payload_type, switch_frame_flag_t *flags, switch_io_flag_t io_flags)
                 ................
                 if (check || (bytes && !switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER))) {
if (!bytes && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER)) { /* We're late! We're Late! */
if (!switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK) && status == SWITCH_STATUS_BREAK) {
switch_cond_next();
continue;
}

return_cng_frame();  <------- this lane to return cng frame if late.
}
}


Could someone explain why we want to return cng frame when package is late? this is caused for me to hear very bad noise audio.

Can I disable return CNG frame?


Cheers





_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org<mailto: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



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org<mailto: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<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900<tel:%2B19193869900>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org<mailto: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



!DSPAM:511134c132767372939513!


From jm at mayfirst.org  Mon Feb 11 17:31:58 2013
From: jm at mayfirst.org (Jamie McClelland)
Date: Mon, 11 Feb 2013 09:31:58 -0500
Subject: [Freeswitch-dev] simultaneous interpretation conference system
Message-ID: <20130211143158.GI4143@mayfirst.org>

Hi freeswitch devs,

This is my first post to the list so I want to start by thanking you for
the great work you have done on Freeswitch. And, sorry for the long
post! I have specific questions at the bottom...

I'm looking for advice on how to build a simultaneous intrepration
conference call system. 

The way we'd like it to work:

The system is required to handle two languages, but the ability to
handle more than two languages would be an added bonus. For the purposes
of this description, I'm giving the example of spanish and english. The
conference call would have spanish-only, english-only, and bi-lingual
speakers. In addition, one or more interpreters would be on the call.

Bi-lingual speakers would participate in the call like it was a normal
conference call. They would hear speakers in the original english or
spanish, and would have the option of speaking either english or spanish
based on their preferences.

The english-only and spanish-only speakers would press 1 to hear a live
intrepretation of the call. When spanish is spoken on the call, an
english intrepretaton would be heard. When English is spoken, a spanish
interpretation would be heard. This would only affect what they hear -
they could still speak into the main conference. Participants would
press 0 to return to hearing the main conference call. 

The interpreters would press 2 to enter interpretation mode. During this
mode their voice is not heard on the main conference, instead it's only
heard by people who have pressed 1 to hear the interpretation. When
english is spoken, the intrepret to spanish and vice-versa. They could
press 0 to return to the normal conference mode.

Ideas on how to implement:

First - has anyone implemented anything like this already?

Based on the wiki docs, I think the conference relate function would be
ideally suited for ensuring people who press 1 only heard the audio from
people who have press 2 and for reversing the effect when 0 is pressed.
Any ideas on a better way?

The next problem is how to capture the digits pressed by conference call
participants. I *think* this could be done by running a process that
listens for freeswitch events. However, I'm not sure how much
information is sent with the event. It would need to pass the conference
id, the user id, and the digit they pressed. Is that information sent by
the event handler? Or, is there a more elegant way to capture
key-presses by conference participants?

Thanks in advance for any light you can shed on this topic,

jamie




-- 
Jamie McClelland

May First/People Link
Growing networks to build a just world
http://www.mayfirst.org
https://support.mayfirst.org

OpenPGP Key: http://current.workingdirectory.net/pages/identity/ 
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From msc at freeswitch.org  Tue Feb 12 04:05:31 2013
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Feb 2013 17:05:31 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxW3-dgJOUsP85NgMcEroCDTkupsEOg06=BumasgEOfyVw@mail.gmail.com>

Hello all!

News and notes are back after a brief hiatus last week. We (the FreeSWITCH
team) were in Milwaukee last week and we appreciated being fed by the
community. Thank you! We love this community.

It seems the big news <http://freeswitch.org/node/435> last week came
courtesy of our friend Kristian Kielhofner <http://blog.krisk.org/>. As
reported in this blog
post<http://blog.krisk.org/2013/02/packets-of-death.html>,
Kristian ran into a rather unusual set of circumstances that resulted in
Packets of Death for some Intel NICs. In addition to being
Slashdotted<http://tech.slashdot.org/story/13/02/06/2024251/intel-gigabit-nic-packet-of-death>,
Kristian's research ended up being featured in a Wired.com
story<http://www.wired.com/wiredenterprise/2013/02/packet-of-death/>.
Don't forget to read Kristian's update
post<http://blog.krisk.org/2013/02/packets-of-death-update.html> that
includes information about Intel's response to the whole situation.

This week things will start to return to normal with our weekly conference
call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13>. We are going
to spend a few weeks talking about various FreeSWITCH GUIs, starting with
the CudaTel <>. For the past four years or so we've
been building the CudaTel Communication Server and we'd like to show off
some of the cool things it can do.

We have two other news items. The first one has to do with
mod_ha_cluster<http://wiki.freeswitch.org/wiki/Mod_ha_cluster>,
which our very own Eliot Gable is building. Financial support for this
open-source module will be done through FreeSWITCH
Solutions<http://www.freeswitchsolutions.com/>.
An interesting discussion can be found in this email
thread<http://lists.freeswitch.org/pipermail/freeswitch-users/2013-February/092095.html>.
The other item comes from Ken Rice who has some updates on Jira and ZRTP.
Check out his mailing list
post<http://lists.freeswitch.org/pipermail/freeswitch-users/2013-February/092155.html>for
more information.

Have a great week and we'll talk to you on Wednesday.

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From richard.screene at netdev.co.uk  Tue Feb 12 14:03:46 2013
From: richard.screene at netdev.co.uk (Richard Screene)
Date: Tue, 12 Feb 2013 11:03:46 +0000
Subject: [Freeswitch-dev] Dingaling does not retain payload IDs
Message-ID: <F54F6F57-D6FB-4188-80EC-9AAD07B4A0DB@netdev.co.uk>

Hello,

When FreeSWITCH receives a Jingle session-initiate message containing a Opus codec with a payload type of 111, it will respond with a session-accept.  But the session-accept will assign the payload type for Opus to 116.  For example:

2013-02-12 10:44:52.543318 [INFO] libdingaling.c:1747 RECV:
...
  <jin:jingle xmlns:jin="urn:xmpp:jingle:1" action="session-initiate" initiator="Richard at 127-0-0-1-852145229.xmpps.drum-collab" sid="3441545841">
    <content creator="initiator" name="audio">
      <description xmlns="urn:xmpp:jingle:apps:rtp:1" profile="RTP/SAVPF" media="audio">
        <payload-type id="103" name="ISAC" clockrate="16000"></payload-type>
        <payload-type id="104" name="ISAC" clockrate="32000"></payload-type>
        <payload-type id="111" name="opus" clockrate="48000"></payload-type>
        <payload-type id="0" name="PCMU" clockrate="8000"></payload-type>
?

2013-02-12 10:44:52.543318 [NOTICE] libdingaling.c:1749 SEND:
...
  <jin:jingle xmlns:jin="urn:xmpp:jingle:1" action="session-accept" sid="3441545841" initiator="Richard at 127-0-0-1-852145229.xmpps.drum-collab">
    <jin:content name="audio" creator="initiator">
      <rtp:description xmlns:rtp="urn:xmpp:jingle:apps:rtp:1" media="audio" ssrc="2585956684">
        <rtp:payload-type id="116" name="opus" clockrate="48000" ptime="48000" maxptime="48000">
          <rtp:parameter name="bitrate" value="32000"></rtp:parameter>
        </rtp:payload-type>
...


From my reading of XEP-0167 (Section 5) it would appear that FreeSWITCH should retain the payload ID that was present in the session-initiate and use it in the session-accept.

Is there a config flag to control this behaviour?  Is it intentional behaviour? Or is it a bug?

I could hack the payload type in the Opus module but we have another endpoint that insists on using a completely different payload type ID.

Many thanks,
  Richard

From msc at freeswitch.org  Wed Feb 13 03:20:34 2013
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Feb 2013 16:20:34 -0800
Subject: [Freeswitch-dev] simultaneous interpretation conference system
In-Reply-To: <20130211143158.GI4143@mayfirst.org>
References: <20130211143158.GI4143@mayfirst.org>
Message-ID: <CAKzWOxUfKFRW4ZzN0oVBoaN0_TLsg28=z6WRwFVHPir74t2Jiw@mail.gmail.com>

Hello Jamie,

I think you're on the right track with the 'relate' setting for members.
That will allow you to control who hears whom. I'm not aware of anyone who
has done anything quite like this but I'm sure that if you can keep track
of which 'mode' each conference member puts himself in then you could have
a program to control everything.

If you are looking at a DIY project then come join us in #freeswitch on
irc.freenode.net and we'll talk in realtime. There are a few telephone
geeks in there. :)

If you'd like professional assistance you can always email
support at freeswitch.org and they can help you with scope, pricing, etc.

Thanks,
Michael

On Mon, Feb 11, 2013 at 6:31 AM, Jamie McClelland <jm at mayfirst.org> wrote:

> Hi freeswitch devs,
>
> This is my first post to the list so I want to start by thanking you for
> the great work you have done on Freeswitch. And, sorry for the long
> post! I have specific questions at the bottom...
>
> I'm looking for advice on how to build a simultaneous intrepration
> conference call system.
>
> The way we'd like it to work:
>
> The system is required to handle two languages, but the ability to
> handle more than two languages would be an added bonus. For the purposes
> of this description, I'm giving the example of spanish and english. The
> conference call would have spanish-only, english-only, and bi-lingual
> speakers. In addition, one or more interpreters would be on the call.
>
> Bi-lingual speakers would participate in the call like it was a normal
> conference call. They would hear speakers in the original english or
> spanish, and would have the option of speaking either english or spanish
> based on their preferences.
>
> The english-only and spanish-only speakers would press 1 to hear a live
> intrepretation of the call. When spanish is spoken on the call, an
> english intrepretaton would be heard. When English is spoken, a spanish
> interpretation would be heard. This would only affect what they hear -
> they could still speak into the main conference. Participants would
> press 0 to return to hearing the main conference call.
>
> The interpreters would press 2 to enter interpretation mode. During this
> mode their voice is not heard on the main conference, instead it's only
> heard by people who have pressed 1 to hear the interpretation. When
> english is spoken, the intrepret to spanish and vice-versa. They could
> press 0 to return to the normal conference mode.
>
> Ideas on how to implement:
>
> First - has anyone implemented anything like this already?
>
> Based on the wiki docs, I think the conference relate function would be
> ideally suited for ensuring people who press 1 only heard the audio from
> people who have press 2 and for reversing the effect when 0 is pressed.
> Any ideas on a better way?
>
> The next problem is how to capture the digits pressed by conference call
> participants. I *think* this could be done by running a process that
> listens for freeswitch events. However, I'm not sure how much
> information is sent with the event. It would need to pass the conference
> id, the user id, and the digit they pressed. Is that information sent by
> the event handler? Or, is there a more elegant way to capture
> key-presses by conference participants?
>
> Thanks in advance for any light you can shed on this topic,
>
> jamie
>
>
>
>
> --
> Jamie McClelland
>
> May First/People Link
> Growing networks to build a just world
> http://www.mayfirst.org
> https://support.mayfirst.org
>
> OpenPGP Key: http://current.workingdirectory.net/pages/identity/
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Feb 13 20:14:53 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Feb 2013 09:14:53 -0800
Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Today
Message-ID: <CAKzWOxV-4EPuTVmd-VwJzbpU-YP3fprhtjtyyejrWBCu=SVCMg@mail.gmail.com>

Hello folks,

Today's conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13

I would like to talk about the interesting HA thread from the mailing list.
Depending on the interest level there we will talk about that for the
conference topic. I really want to know how people feel about this. Links
to the ML threads are on the agenda page under "Items Needing Discussion."

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Feb 13 20:24:30 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Feb 2013 09:24:30 -0800
Subject: [Freeswitch-dev] simultaneous interpretation conference system
In-Reply-To: <CAKzWOxUfKFRW4ZzN0oVBoaN0_TLsg28=z6WRwFVHPir74t2Jiw@mail.gmail.com>
References: <20130211143158.GI4143@mayfirst.org>
	<CAKzWOxUfKFRW4ZzN0oVBoaN0_TLsg28=z6WRwFVHPir74t2Jiw@mail.gmail.com>
Message-ID: <CAKzWOxUgj82VEyY1QbiDqEEQn8-JEmF-msi19DeLF3aHeb6Jrg@mail.gmail.com>

Oh, I forgot to mention: regarding DTMFs, you'll need to modify the
conference profile not to use whatever keys you want the callers to use.
Just remove them from the profile. (See
conf/autoload_configs/conference.conf.xml where there's all sorts of key
bindings like 0=mute/unmute)

You could listen for DTMF events, but most likely you'll need to set
verbose_events to true so that you get all the event headers, not just the
pre-selected ones. I like to use fs_cli for tinkering with events so that I
can see what's going on. Check out the /events and /filter commands. Also
see chapter 4 of the FreeSWITCH Cookbook.

-MC

On Tue, Feb 12, 2013 at 4:20 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hello Jamie,
>
> I think you're on the right track with the 'relate' setting for members.
> That will allow you to control who hears whom. I'm not aware of anyone who
> has done anything quite like this but I'm sure that if you can keep track
> of which 'mode' each conference member puts himself in then you could have
> a program to control everything.
>
> If you are looking at a DIY project then come join us in #freeswitch on
> irc.freenode.net and we'll talk in realtime. There are a few telephone
> geeks in there. :)
>
> If you'd like professional assistance you can always email
> support at freeswitch.org and they can help you with scope, pricing, etc.
>
> Thanks,
> Michael
>
> On Mon, Feb 11, 2013 at 6:31 AM, Jamie McClelland <jm at mayfirst.org> wrote:
>
>> Hi freeswitch devs,
>>
>> This is my first post to the list so I want to start by thanking you for
>> the great work you have done on Freeswitch. And, sorry for the long
>> post! I have specific questions at the bottom...
>>
>> I'm looking for advice on how to build a simultaneous intrepration
>> conference call system.
>>
>> The way we'd like it to work:
>>
>> The system is required to handle two languages, but the ability to
>> handle more than two languages would be an added bonus. For the purposes
>> of this description, I'm giving the example of spanish and english. The
>> conference call would have spanish-only, english-only, and bi-lingual
>> speakers. In addition, one or more interpreters would be on the call.
>>
>> Bi-lingual speakers would participate in the call like it was a normal
>> conference call. They would hear speakers in the original english or
>> spanish, and would have the option of speaking either english or spanish
>> based on their preferences.
>>
>> The english-only and spanish-only speakers would press 1 to hear a live
>> intrepretation of the call. When spanish is spoken on the call, an
>> english intrepretaton would be heard. When English is spoken, a spanish
>> interpretation would be heard. This would only affect what they hear -
>> they could still speak into the main conference. Participants would
>> press 0 to return to hearing the main conference call.
>>
>> The interpreters would press 2 to enter interpretation mode. During this
>> mode their voice is not heard on the main conference, instead it's only
>> heard by people who have pressed 1 to hear the interpretation. When
>> english is spoken, the intrepret to spanish and vice-versa. They could
>> press 0 to return to the normal conference mode.
>>
>> Ideas on how to implement:
>>
>> First - has anyone implemented anything like this already?
>>
>> Based on the wiki docs, I think the conference relate function would be
>> ideally suited for ensuring people who press 1 only heard the audio from
>> people who have press 2 and for reversing the effect when 0 is pressed.
>> Any ideas on a better way?
>>
>> The next problem is how to capture the digits pressed by conference call
>> participants. I *think* this could be done by running a process that
>> listens for freeswitch events. However, I'm not sure how much
>> information is sent with the event. It would need to pass the conference
>> id, the user id, and the digit they pressed. Is that information sent by
>> the event handler? Or, is there a more elegant way to capture
>> key-presses by conference participants?
>>
>> Thanks in advance for any light you can shed on this topic,
>>
>> jamie
>>
>>
>>
>>
>> --
>> Jamie McClelland
>>
>> May First/People Link
>> Growing networks to build a just world
>> http://www.mayfirst.org
>> https://support.mayfirst.org
>>
>> OpenPGP Key: http://current.workingdirectory.net/pages/identity/
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From eduardonunesp at gmail.com  Thu Feb 14 14:17:55 2013
From: eduardonunesp at gmail.com (Eduardo Nunes Pereira)
Date: Thu, 14 Feb 2013 09:17:55 -0200
Subject: [Freeswitch-dev]  Event engine like FreeSWITCH
Message-ID: <CAAXv2Z+S0eeqqE+bSP+=yzSz2E-MZqtG=52+HVJKxsxQRAJayg@mail.gmail.com>

Hi, folks this topic it's a little bit off-topic, i just want to know if
there is an event engine written in C/C++ like FS's event socket system,
that you can register your application and fire and listen internal events
binding with app our by network. When i was programming for iOS i seen
something similar with NSNotificationCenter.

-- 
Eduardo Nunes Pereira
skype: eduardonunesp
msn:eduardonunesp
http://about.me/eduardonunesp
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From steveayre at gmail.com  Thu Feb 14 15:09:22 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Feb 2013 12:09:22 +0000
Subject: [Freeswitch-dev] Event engine like FreeSWITCH
In-Reply-To: <CAAXv2Z+S0eeqqE+bSP+=yzSz2E-MZqtG=52+HVJKxsxQRAJayg@mail.gmail.com>
References: <CAAXv2Z+S0eeqqE+bSP+=yzSz2E-MZqtG=52+HVJKxsxQRAJayg@mail.gmail.com>
Message-ID: <CAFiqYukN=gAaHvHgCV9Un+BLiDNQGv95LBKaFKmLCe+Rf=_TjQ@mail.gmail.com>

Try looking at ZMQ, AMQP or similar.

-Steve



On 14 February 2013 11:17, Eduardo Nunes Pereira <eduardonunesp at gmail.com>wrote:

> Hi, folks this topic it's a little bit off-topic, i just want to know if
> there is an event engine written in C/C++ like FS's event socket system,
> that you can register your application and fire and listen internal events
> binding with app our by network. When i was programming for iOS i seen
> something similar with NSNotificationCenter.
>
> --
> Eduardo Nunes Pereira
> skype: eduardonunesp
> msn:eduardonunesp
> http://about.me/eduardonunesp
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
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From eduardonunesp at gmail.com  Thu Feb 14 17:16:30 2013
From: eduardonunesp at gmail.com (Eduardo Nunes Pereira)
Date: Thu, 14 Feb 2013 12:16:30 -0200
Subject: [Freeswitch-dev] Event engine like FreeSWITCH
In-Reply-To: <CAFiqYukN=gAaHvHgCV9Un+BLiDNQGv95LBKaFKmLCe+Rf=_TjQ@mail.gmail.com>
References: <CAAXv2Z+S0eeqqE+bSP+=yzSz2E-MZqtG=52+HVJKxsxQRAJayg@mail.gmail.com>
	<CAFiqYukN=gAaHvHgCV9Un+BLiDNQGv95LBKaFKmLCe+Rf=_TjQ@mail.gmail.com>
Message-ID: <CAAXv2Z+GEoxvpEn2ZjsHciUfRiM5jT=M5kXt-+E6o+HOj=k_qw@mail.gmail.com>

Very very cool ! i will learn more about 0MQ. Thanks

On Thu, Feb 14, 2013 at 10:09 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Try looking at ZMQ, AMQP or similar.
>
> -Steve
>
>
>
> On 14 February 2013 11:17, Eduardo Nunes Pereira <eduardonunesp at gmail.com>wrote:
>
>> Hi, folks this topic it's a little bit off-topic, i just want to know if
>> there is an event engine written in C/C++ like FS's event socket system,
>> that you can register your application and fire and listen internal events
>> binding with app our by network. When i was programming for iOS i seen
>> something similar with NSNotificationCenter.
>>
>> --
>> Eduardo Nunes Pereira
>> skype: eduardonunesp
>> msn:eduardonunesp
>> http://about.me/eduardonunesp
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Eduardo Nunes Pereira
skype: eduardonunesp
msn:eduardonunesp
http://about.me/eduardonunesp
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From jm at mayfirst.org  Thu Feb 14 17:39:27 2013
From: jm at mayfirst.org (Jamie McClelland)
Date: Thu, 14 Feb 2013 09:39:27 -0500
Subject: [Freeswitch-dev] simultaneous interpretation conference system
In-Reply-To: <CAKzWOxUgj82VEyY1QbiDqEEQn8-JEmF-msi19DeLF3aHeb6Jrg@mail.gmail.com>
References: <20130211143158.GI4143@mayfirst.org>
	<CAKzWOxUfKFRW4ZzN0oVBoaN0_TLsg28=z6WRwFVHPir74t2Jiw@mail.gmail.com>
	<CAKzWOxUgj82VEyY1QbiDqEEQn8-JEmF-msi19DeLF3aHeb6Jrg@mail.gmail.com>
Message-ID: <20130214143927.GJ4401@mayfirst.org>

On Wed Feb 13, Michael Collins wrote:
> Oh, I forgot to mention: regarding DTMFs, you'll need to modify the
> conference profile not to use whatever keys you want the callers to use.
> Just remove them from the profile. (See
> conf/autoload_configs/conference.conf.xml where there's all sorts of key
> bindings like 0=mute/unmute)
> 
> You could listen for DTMF events, but most likely you'll need to set
> verbose_events to true so that you get all the event headers, not just the
> pre-selected ones. I like to use fs_cli for tinkering with events so that I
> can see what's going on. Check out the /events and /filter commands. Also
> see chapter 4 of the FreeSWITCH Cookbook.

Thanks so much for the support. I'll give it a try and report back
my progress.

jamie



-- 
Jamie McClelland

May First/People Link
Growing networks to build a just world
http://www.mayfirst.org
https://support.mayfirst.org

OpenPGP Key: http://current.workingdirectory.net/pages/identity/ 
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From msc at freeswitch.org  Fri Feb 15 01:41:57 2013
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Feb 2013 14:41:57 -0800
Subject: [Freeswitch-dev] FreeSWITCH HA BoF Meetings
Message-ID: <CAKzWOxXX11di5YwYzB0KajaxHgMh8G3VzE6to-ftDpCh_eNu1Q@mail.gmail.com>

Hello!

Yesterday we had a nice discussion
<http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13>about all things HA.
Eliot Gable gave us a lot of very good information about the state of
mod_ha_development and all the work that he's done over the past 4+ years
in the world of HA and FreeSWITCH.

It's now time to let those who have a vested interest in HA to get together
with Eliot and continuing the discussion. To that end we propose a
FreeSWITCH HA BoF (Birds of a Feather) conference. Eliot says that he can
be available after 8PM EST. I would like to propose that the meetup be
Tuesday evenings at 8:00PM EST in the main FreeSWITCH public conference.

If you would like to meetup then please email me off-list and CC Eliot (
egable+freeswitch at gmail.com) so that we can keep track of who is most
interested in this topic.

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From dxj19831029 at gmail.com  Sat Feb 16 14:51:39 2013
From: dxj19831029 at gmail.com (Xijing Dai)
Date: Sat, 16 Feb 2013 19:51:39 +0800
Subject: [Freeswitch-dev] question about CNG on late packages
In-Reply-To: <CAP9o=2=wT4aVyfcVrFMBOaJkoJ=1ahACfrLTzfKMiuSc7Hku1A@mail.gmail.com>
References: <CAMsCbX9jux9bn-V3GZp7CnseUczXTXP_z6_VsCJSNL=PVzdeNQ@mail.gmail.com>
	<CAFiqYun1gDVe637o5zKv3giGHAw281essMhHKj0NC9MhUOLqDA@mail.gmail.com>
	<CAKbxfG8MW6yY1JA8YDvtRoViF3HhCDZD1Q1GHUn9f06Q=J3_Mw@mail.gmail.com>
	<CAMsCbX8k5DScricvdZJ-FXmZJ-zWRz=52B12YUcCdeG3jX09Ew@mail.gmail.com>
	<CAP9o=2=wT4aVyfcVrFMBOaJkoJ=1ahACfrLTzfKMiuSc7Hku1A@mail.gmail.com>
Message-ID: <CAMsCbX95owD+UC9gkFYLtzDRaX7NQoWO-BFZR8SpfHpgV8N3RA@mail.gmail.com>

How to let freeswitch send late packets? I tried in following way, but the
call is missing ACK in freeswitch end when the client sent 200 OK back.

Any suggestions?

The changed code is: (I simply go to sleep 1ms and then try to read again)


                if (check || (bytes && !switch_test_flag(rtp_session,
SWITCH_RTP_FLAG_USE_TIMER))) {
if (!bytes && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER)) { /*
We're late! We're Late! */
if (!switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK) && status ==
SWITCH_STATUS_BREAK) {
switch_cond_next();
continue;
}

//return_cng_frame();  <------- comment this lane and directly go to sleep
and read again
                                goto do_continue;
}
}

On Wed, Feb 6, 2013 at 12:52 AM, Bing LI <enst.bupt at gmail.com> wrote:

> In my experience suppress_cng did work but only prevented sending CNG
> packet.
> freeswitch would ignore the late packets because it thought the cng packet
> had been sent out. it makes a hole in the seq number in B-leg.
>
> So later I modified the switch_rtp.c to change this mechanism.
>
>
>
> On Tue, Feb 5, 2013 at 11:32 AM, Xijing Dai <dxj19831029 at gmail.com> wrote:
>
>> I see.
>>
>>
>> I did set it suppress_cng=true, and it did not help. No difference at all.
>>
>> This happens inside LAN.
>>
>> When talking about the jitter buffer, we want to enable it on client, and
>> disable it inside freeswitch.
>> Therefore, in this case, should freeswitch do nothing?
>>
>>
>> Cheers
>>
>>
>>
>>
>> On Tue, Feb 5, 2013 at 11:35 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Don't negotiate CNG in the SDP and it won't use it or disable it with
>>> suppress_cng=true in your channel or global vars.
>>>
>>>
>>>
>>> On Tue, Feb 5, 2013 at 4:32 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>>> Because if a RTP packet is due and hasn't arrived yet you need
>>>> something to play. Audio isn't like TCP where you dropped packets can be
>>>> retransmitted, if it hasn't arrived by the time you want to play it it's
>>>> too late and you have to play silence to handle it.
>>>>
>>>> Your audio problems are more likely due to packet loss or jitter which
>>>> is why the RTP packet is overdue, not the CNG itself.
>>>>
>>>> It jitter's the problem enabling the jitterbuffer might help (at the
>>>> expense of increased lag). If loss is the problem there's no way to handle
>>>> that except as silence for the missing interval, you need to fix whatever
>>>> is causing the loss.
>>>>
>>>> -Steve
>>>>
>>>>
>>>> On 5 February 2013 10:23, Xijing Dai <dxj19831029 at gmail.com> wrote:
>>>>
>>>>> Hey all,
>>>>>
>>>>>
>>>>> Inside code
>>>>>
>>>>> switch_rtp.c file:
>>>>>
>>>>>    static int rtp_common_read(switch_rtp_t *rtp_session,
>>>>> switch_payload_t *payload_type, switch_frame_flag_t *flags,
>>>>> switch_io_flag_t io_flags)
>>>>>                  ................
>>>>>                  if (check || (bytes && !switch_test_flag(rtp_session,
>>>>> SWITCH_RTP_FLAG_USE_TIMER))) {
>>>>> if (!bytes && switch_test_flag(rtp_session,
>>>>> SWITCH_RTP_FLAG_USE_TIMER)) { /* We're late! We're Late! */
>>>>>  if (!switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK) &&
>>>>> status == SWITCH_STATUS_BREAK) {
>>>>> switch_cond_next();
>>>>>  continue;
>>>>> }
>>>>>
>>>>> return_cng_frame();  <------- this lane to return cng frame if late.
>>>>>  }
>>>>> }
>>>>>
>>>>>
>>>>> Could someone explain why we want to return cng frame when package is
>>>>> late? this is caused for me to hear very bad noise audio.
>>>>>
>>>>> Can I disable return CNG frame?
>>>>>
>>>>>
>>>>> Cheers
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
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From shaheryarkh at gmail.com  Mon Feb 18 04:55:22 2013
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Mon, 18 Feb 2013 02:55:22 +0100
Subject: [Freeswitch-dev] Change Voicemail IVR Menu
Message-ID: <CAFZQphxKLrxeWFrqS-jPFfdU7_kQRKamJN1Wdk+knARAXVN-Nw@mail.gmail.com>

Hi,

I am trying to change menu order of FS voicemail module. I am using
voicemail_ivr with xml_curl, when a voicemail user checks voicemail, i
check if user has his own greeting set, then play normal voicemail menu,
otherwise instead of playing message count and related options, i simply
want to play voicemail preferences menu, in fact only voicemail record
greeting menu. I have tried many different possibilities including many
changes to voicemail_ivr.xml  and voicemail_ivr.conf.xml but so far none
worked. FS Wiki on voicemail_ivr module seems outdated. Till now i am only
able to play standard voicemail menu through voicemail_ivr module.

Do I have to modify native code of these modules (mod_voicemail and
mod_voicemail_ivr) to achieve the goal? Any suggestions / guidelines?

P.S. Note that sample files (voicemail_ivr.conf.xml and voicemail_ivr.xml)
do not work as is, i had to modify many typos and tag attributes to get it
working same as standard mod_voicemail works. I will update wiki once this
problem is solved.

Thank you.


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From ec.amazumdar at tatapowersed.com  Mon Feb 18 09:48:00 2013
From: ec.amazumdar at tatapowersed.com (Anindo Mazumdar)
Date: Mon, 18 Feb 2013 12:18:00 +0530
Subject: [Freeswitch-dev] Integration of FreeSWITCH and Open IMS
Message-ID: <201302180648.r1I6m0YM027327@blr.tatapowersed.com>

Hello all,

I am working on FreeSWITCH and Open IMS server. Just wanted to know whether their is a way by which I can integrate both.  I have both of them installed as a standalone application on my machine.
-- 
With Regards,

Anindo Mazumdar,

From msc at freeswitch.org  Mon Feb 18 21:14:35 2013
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Feb 2013 10:14:35 -0800
Subject: [Freeswitch-dev] Integration of FreeSWITCH and Open IMS
In-Reply-To: <201302180648.r1I6m0YM027327@blr.tatapowersed.com>
References: <201302180648.r1I6m0YM027327@blr.tatapowersed.com>
Message-ID: <CAKzWOxX6A6-ZHv3MzFwyv7+qbEidBs3NwNT5w4P17ua_qd2Q2Q@mail.gmail.com>

I suspect you can have them both running but you'll need to have your
FreeSWITCH SIP profile use a different port than the one used by Open IMS.
I don't know anything about Open IMS so I can't tell you if it has any
other limitations that would disallow FreeSWITCH from being on the same
host.

-MC

On Sun, Feb 17, 2013 at 10:48 PM, Anindo Mazumdar <
ec.amazumdar at tatapowersed.com> wrote:

> Hello all,
>
> I am working on FreeSWITCH and Open IMS server. Just wanted to know
> whether their is a way by which I can integrate both.  I have both of them
> installed as a standalone application on my machine.
> --
> With Regards,
>
> Anindo Mazumdar,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From manieq at wp.eu  Mon Feb 18 22:27:37 2013
From: manieq at wp.eu (Mariusz Czulada)
Date: Mon, 18 Feb 2013 20:27:37 +0100
Subject: [Freeswitch-dev] Odp:  Integration of FreeSWITCH and Open IMS
Message-ID: <512280a94b9169.32105691@wp.pl>

Hi Anindo,

We have FSW servers integrated with an IMS platform (ALU is the vendor). They play the role of MRF (media announcements and ivr/conferences), but not as strictly as in 3GPP docs (no MRFC/MRFP splitting). We have only Mr interface towards S-CSCF and Mb to MGWs. No Mr, no Rf. Is this what you plan to do to, or you want to build a full AS (like BroadWorks) in the IMS application layer?

Regards,
Mariusz

Dnia Poniedzia?ek, 18 Lutego 2013 07:48 Anindo Mazumdar <ec.amazumdar at tatapowersed.com> napisa?(a) 
> Hello all,
> 
> I am working on FreeSWITCH and Open IMS server. Just wanted to know whether their is a way by which I can integrate both.  I have both of them installed as a standalone application on my machine.
> -- 
> With Regards,
> 
> Anindo Mazumdar,





From marketing at cluecon.com  Mon Feb 18 22:38:53 2013
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 18 Feb 2013 11:38:53 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxVsm9XcVi-xH9hy4FHj-+m7t0hpvj4_RTuPp7Kq-f4mnw@mail.gmail.com>

Greetings all!

Last week was rather interesting. Initially we had planned on doing a
CudaTel <> demonstration on the weekly conference
call. However, interest in Eliot Gable's
mod_ha_cluster<http://wiki.freeswitch.org/wiki/Mod_ha_cluster>and the
accompanying
conversation<http://lists.freeswitch.org/pipermail/freeswitch-users/2013-February/092187.html>was
particularly intense. That being the case, on last week's conference
call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13> we spent most
of the time talking about HA in general and how we could build a FreeSWITCH
HA system. We also invited everyone who is interested in the subject to
call in to the public FreeSWITCH conference at 8PM EST on Tuesday evening
(Feb 19) for the first HA conference call. (Eliot won't join until about
8:15PM.) If you have a vested interest in HA for FreeSWITCH then please
join the conference call.

For this week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_20>we will ask
one of the participants on the Tuesday night call to give us a
brief overview of the HA discussion. After that I will be doing a
demonstration of the CudaTel to show off what the FreeSWITCH team has been
working so hard to develop these past few years. We hope you enjoy it!

For the DIY crowd you may enjoy this site <http://switchpi.org/> that Ken
Rice set up. It's dedicated to doing cool things with FreeSWITCH and
the Raspberry
Pi <http://www.raspberrypi.org/>. Check it out!

In ClueCon <http://www.cluecon.com>2013 news we are getting things all set.
In the coming days we will be making announcements about new sponsors,
registration and hotel information, and a call for speakers for this year's
event. Don't forget to plan for a full 3 days, August 6th - 8th. Also, we
tend to do a lot of fun stuff on the Monday before ClueCon, so it's a good
idea to arrive on or before Monday August 7.

Hope you have a great week!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From richard.screene at netdev.co.uk  Mon Feb 18 22:39:46 2013
From: richard.screene at netdev.co.uk (Richard Screene)
Date: Mon, 18 Feb 2013 19:39:46 +0000
Subject: [Freeswitch-dev] Rtcp-mux and mod_dingaling
Message-ID: <lljuvnr6t44iak24qu67uvbb.1361216386147@email.android.com>

Hello,

Does mod_dingaling support rtcp-mux?  If not can anyone suggest the best way for me to add this functionality?  I'm only really interested in receiving the RTP and RTCP on the same port (I don't need to have FreeSWITCH send them on the same port).

Many thanks,
  Richatd

From cbeeton at avaya.com  Wed Feb 20 18:10:39 2013
From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn))
Date: Wed, 20 Feb 2013 15:10:39 +0000
Subject: [Freeswitch-dev] reload logfile.conf on reloadxml or sighup?
Message-ID: <88D19A5E9ADC2D4988914F6773EB521F38560544@AZ-US1EXMB04.global.avaya.com>

I would like to be able to trigger Freeswitch to reload the logfile.conf file (to pick up changes in log level on the fly).  I know this can be done by changing levels on the console, but since my application uses configuration files to persist the changes, I would prefer to keep all configuration data in the files.

The "reloadxml"  command triggers FS to reload some but not all of the config files.  Could it be enhanced to reload logfile.conf?  Or would it make sense to reload the logfile.conf file on receipt of sig_hup?  Or have I missed another way to accomplish this?  If there's any interest, I will create a jira and offer a patch.

Thanks,
Carolyn
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From steveayre at gmail.com  Wed Feb 20 18:18:39 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Feb 2013 15:18:39 +0000
Subject: [Freeswitch-dev] reload logfile.conf on reloadxml or sighup?
In-Reply-To: <88D19A5E9ADC2D4988914F6773EB521F38560544@AZ-US1EXMB04.global.avaya.com>
References: <88D19A5E9ADC2D4988914F6773EB521F38560544@AZ-US1EXMB04.global.avaya.com>
Message-ID: <CAFiqYunfCzRTJH-LSVoGwAg9+k=XWg=gK0=z65jAZ-kq0+g-KA@mail.gmail.com>

'reloadxml' then 'reload mod_logfile'

-Steve



On 20 February 2013 15:10, Beeton, Carolyn (Carolyn) <cbeeton at avaya.com> wrote:
> I would like to be able to trigger Freeswitch to reload the logfile.conf
> file (to pick up changes in log level on the fly).  I know this can be done
> by changing levels on the console, but since my application uses
> configuration files to persist the changes, I would prefer to keep all
> configuration data in the files.
>
>
>
> The ?reloadxml?  command triggers FS to reload some but not all of the
> config files.  Could it be enhanced to reload logfile.conf?  Or would it
> make sense to reload the logfile.conf file on receipt of sig_hup?  Or have I
> missed another way to accomplish this?  If there?s any interest, I will
> create a jira and offer a patch.
>
>
>
> Thanks,
>
> Carolyn
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>


From msc at freeswitch.org  Wed Feb 20 19:58:28 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Feb 2013 08:58:28 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call
Message-ID: <CAKzWOxUd+3pbwu0GX1f=Dgq18YYtAPgyQmo0KAa8GyQjnDHEqA@mail.gmail.com>

Hey folks, today's conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_20

We will be doing a few news updates, including a recap of last night's
"FreeSWITCH HA" conference call. Then I will be doing a tour of the CudaTel
Communications Server.

Talk to you at 1PM EST/10AM PST

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From Kevin.Snow at ooma.com  Wed Feb 20 20:29:23 2013
From: Kevin.Snow at ooma.com (Kevin Snow)
Date: Wed, 20 Feb 2013 17:29:23 +0000
Subject: [Freeswitch-dev] uuid_record Question
Message-ID: <372DF5B90E1994439B2280570FEEDAA33C31326A@mbx025-e1-nj-4.exch025.domain.local>


I'm using the uuid_record function to record the read side of a sofia session. Used in conjunction with switch_ivr_play_file() and an echo server I can place a call, play a sound file and record the echo. Easy enough and works with a problem of occasional little dropouts.

I recorded my daughter saying "Ooma" and duplicated it a bunch of times as a test sound file. This makes a nice repetitive sample to look at. I've included an image of the recorded audio (from echo) as displayed in Audacity (hopefully this makes it through). In the 5th and 7th "Ooma" there is a dropout. When I listen to the live echo it sounds fine, but listening to the recording I can hear the ticks. This does not happen every time, but enough to be a bother.

Has run into this? Ideas on how to stop it?

Thanks

Kevin


[cid:96A0D21F-8B75-475F-98F1-710EF4EE7231]
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From alex at jajah.com  Thu Feb 21 13:15:11 2013
From: alex at jajah.com (Alex Massover)
Date: Thu, 21 Feb 2013 12:15:11 +0200
Subject: [Freeswitch-dev] on_init hook and outgoing channels
Message-ID: <569384504C492C4580E88B5D54DFEAEA33DA3DF2EE@jjex01.jajah.dublin>

Hello,

We have services that manage calls via ESL and want to write a custom authorization and billing module.

The idea is to catch any call received by FreeSwitch before service gets control on ESL for incoming call and on_init hook works for us.

But for outgoing channels (created by 'route' or 'originate' commands over ESL) on_init hook doesn't work. Well the channel hits the hook but already after outgoing INVITE being sent, although channel state goes from CS_NEW->CS_INIT->CS_ROUTING correctly. And I want to authorize the channel before it goes out.

Is that a normal behavior or I completely miss something? Is there other way to block an outgoing channel after it's created but before it goes out from a custom module?

--
Best Regards,
Alex Massover

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From jm at mayfirst.org  Fri Feb 22 02:09:49 2013
From: jm at mayfirst.org (Jamie McClelland)
Date: Thu, 21 Feb 2013 18:09:49 -0500
Subject: [Freeswitch-dev] simultaneous interpretation conference system
In-Reply-To: <CAKzWOxUgj82VEyY1QbiDqEEQn8-JEmF-msi19DeLF3aHeb6Jrg@mail.gmail.com>
References: <20130211143158.GI4143@mayfirst.org>
	<CAKzWOxUfKFRW4ZzN0oVBoaN0_TLsg28=z6WRwFVHPir74t2Jiw@mail.gmail.com>
	<CAKzWOxUgj82VEyY1QbiDqEEQn8-JEmF-msi19DeLF3aHeb6Jrg@mail.gmail.com>
Message-ID: <20130221230949.GC14334@mayfirst.org>

Hi everyone,

Thanks to the foundational work of the freeswitch devs and community,
I've managed to write my application in about 150 lines of code.

It's still in early alpha testing phase, but if you are interested in
seeing the code, it's available here:

git://git.mayfirst.org/mfpl/mexcla

I wanted to also thank Alexandre Fiori - because it wouldn't have been
so easy and pleasant without https://github.com/fiorix/eventsocket.

jamie

On Wed Feb 13, Michael Collins wrote:
> Oh, I forgot to mention: regarding DTMFs, you'll need to modify the
> conference profile not to use whatever keys you want the callers to use.
> Just remove them from the profile. (See
> conf/autoload_configs/conference.conf.xml where there's all sorts of key
> bindings like 0=mute/unmute)
> 
> You could listen for DTMF events, but most likely you'll need to set
> verbose_events to true so that you get all the event headers, not just the
> pre-selected ones. I like to use fs_cli for tinkering with events so that I
> can see what's going on. Check out the /events and /filter commands. Also
> see chapter 4 of the FreeSWITCH Cookbook.

> 
> -MC
> 
> On Tue, Feb 12, 2013 at 4:20 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
> > Hello Jamie,
> >
> > I think you're on the right track with the 'relate' setting for members.
> > That will allow you to control who hears whom. I'm not aware of anyone who
> > has done anything quite like this but I'm sure that if you can keep track
> > of which 'mode' each conference member puts himself in then you could have
> > a program to control everything.
> >
> > If you are looking at a DIY project then come join us in #freeswitch on
> > irc.freenode.net and we'll talk in realtime. There are a few telephone
> > geeks in there. :)
> >
> > If you'd like professional assistance you can always email
> > support at freeswitch.org and they can help you with scope, pricing, etc.
> >
> > Thanks,
> > Michael
> >
> > On Mon, Feb 11, 2013 at 6:31 AM, Jamie McClelland <jm at mayfirst.org> wrote:
> >
> >> Hi freeswitch devs,
> >>
> >> This is my first post to the list so I want to start by thanking you for
> >> the great work you have done on Freeswitch. And, sorry for the long
> >> post! I have specific questions at the bottom...
> >>
> >> I'm looking for advice on how to build a simultaneous intrepration
> >> conference call system.
> >>
> >> The way we'd like it to work:
> >>
> >> The system is required to handle two languages, but the ability to
> >> handle more than two languages would be an added bonus. For the purposes
> >> of this description, I'm giving the example of spanish and english. The
> >> conference call would have spanish-only, english-only, and bi-lingual
> >> speakers. In addition, one or more interpreters would be on the call.
> >>
> >> Bi-lingual speakers would participate in the call like it was a normal
> >> conference call. They would hear speakers in the original english or
> >> spanish, and would have the option of speaking either english or spanish
> >> based on their preferences.
> >>
> >> The english-only and spanish-only speakers would press 1 to hear a live
> >> intrepretation of the call. When spanish is spoken on the call, an
> >> english intrepretaton would be heard. When English is spoken, a spanish
> >> interpretation would be heard. This would only affect what they hear -
> >> they could still speak into the main conference. Participants would
> >> press 0 to return to hearing the main conference call.
> >>
> >> The interpreters would press 2 to enter interpretation mode. During this
> >> mode their voice is not heard on the main conference, instead it's only
> >> heard by people who have pressed 1 to hear the interpretation. When
> >> english is spoken, the intrepret to spanish and vice-versa. They could
> >> press 0 to return to the normal conference mode.
> >>
> >> Ideas on how to implement:
> >>
> >> First - has anyone implemented anything like this already?
> >>
> >> Based on the wiki docs, I think the conference relate function would be
> >> ideally suited for ensuring people who press 1 only heard the audio from
> >> people who have press 2 and for reversing the effect when 0 is pressed.
> >> Any ideas on a better way?
> >>
> >> The next problem is how to capture the digits pressed by conference call
> >> participants. I *think* this could be done by running a process that
> >> listens for freeswitch events. However, I'm not sure how much
> >> information is sent with the event. It would need to pass the conference
> >> id, the user id, and the digit they pressed. Is that information sent by
> >> the event handler? Or, is there a more elegant way to capture
> >> key-presses by conference participants?
> >>
> >> Thanks in advance for any light you can shed on this topic,
> >>
> >> jamie
> >>
> >>
> >>
> >>
> >> --
> >> Jamie McClelland
> >>
> >> May First/People Link
> >> Growing networks to build a just world
> >> http://www.mayfirst.org
> >> https://support.mayfirst.org
> >>
> >> OpenPGP Key: http://current.workingdirectory.net/pages/identity/
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >>
> >>
> >
> >
> > --
> > Michael S Collins
> > Twitter: @mercutioviz
> > http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> >
> >
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org

> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


-- 
Jamie McClelland

May First/People Link
Growing networks to build a just world
http://www.mayfirst.org
https://support.mayfirst.org

OpenPGP Key: http://current.workingdirectory.net/pages/identity/ 
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From cbeeton at avaya.com  Fri Feb 22 18:37:19 2013
From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn))
Date: Fri, 22 Feb 2013 15:37:19 +0000
Subject: [Freeswitch-dev] reload logfile.conf on reloadxml or sighup?
In-Reply-To: <CAFiqYunfCzRTJH-LSVoGwAg9+k=XWg=gK0=z65jAZ-kq0+g-KA@mail.gmail.com>
References: <88D19A5E9ADC2D4988914F6773EB521F38560544@AZ-US1EXMB04.global.avaya.com>
	<CAFiqYunfCzRTJH-LSVoGwAg9+k=XWg=gK0=z65jAZ-kq0+g-KA@mail.gmail.com>
Message-ID: <88D19A5E9ADC2D4988914F6773EB521F38561927@AZ-US1EXMB04.global.avaya.com>

Thanks - that's what I was looking for.  I was a bit surprised to find that 'reload mod_logfile' calls switch_xml_reload itself, so there is no need to call reloadxml explicitly.

Carolyn

> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-
> dev-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
> Sent: Wednesday, February 20, 2013 10:19 AM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] reload logfile.conf on reloadxml or
> sighup?
> 
> 'reloadxml' then 'reload mod_logfile'
> 
> -Steve
> 
> 
> 
> On 20 February 2013 15:10, Beeton, Carolyn (Carolyn)
> <cbeeton at avaya.com> wrote:
> > I would like to be able to trigger Freeswitch to reload the
> logfile.conf
> > file (to pick up changes in log level on the fly).  I know this can
> be done
> > by changing levels on the console, but since my application uses
> > configuration files to persist the changes, I would prefer to keep
> all
> > configuration data in the files.
> >
> >
> >
> > The "reloadxml"  command triggers FS to reload some but not all of
> the
> > config files.  Could it be enhanced to reload logfile.conf?  Or would
> it
> > make sense to reload the logfile.conf file on receipt of sig_hup?  Or
> have I
> > missed another way to accomplish this?  If there's any interest, I
> will
> > create a jira and offer a patch.
> >
> >
> >
> > Thanks,
> >
> > Carolyn
> >
> >
> >
> _______________________________________________________________________
> __
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
> 
> _______________________________________________________________________
> __
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


From msc at freeswitch.org  Fri Feb 22 21:26:40 2013
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Feb 2013 10:26:40 -0800
Subject: [Freeswitch-dev] reload logfile.conf on reloadxml or sighup?
In-Reply-To: <88D19A5E9ADC2D4988914F6773EB521F38561927@AZ-US1EXMB04.global.avaya.com>
References: <88D19A5E9ADC2D4988914F6773EB521F38560544@AZ-US1EXMB04.global.avaya.com>
	<CAFiqYunfCzRTJH-LSVoGwAg9+k=XWg=gK0=z65jAZ-kq0+g-KA@mail.gmail.com>
	<88D19A5E9ADC2D4988914F6773EB521F38561927@AZ-US1EXMB04.global.avaya.com>
Message-ID: <CAKzWOxX+Rgc228n+qfDSeTJJdPEo1LOrgwqnT3LrngT0Ys7_3g@mail.gmail.com>

There was a time that reload mod_xxx did not call switch_xml_reload but we
realized that in almost every case of reloading a module it was preceded by
a reloadxml command so it seemed logical to roll in the reloadxml for the
sake of convenience.
-MC

On Fri, Feb 22, 2013 at 7:37 AM, Beeton, Carolyn (Carolyn) <
cbeeton at avaya.com> wrote:

> Thanks - that's what I was looking for.  I was a bit surprised to find
> that 'reload mod_logfile' calls switch_xml_reload itself, so there is no
> need to call reloadxml explicitly.
>
> Carolyn
>
> > -----Original Message-----
> > From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-
> > dev-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
> > Sent: Wednesday, February 20, 2013 10:19 AM
> > To: freeswitch-dev at lists.freeswitch.org
> > Subject: Re: [Freeswitch-dev] reload logfile.conf on reloadxml or
> > sighup?
> >
> > 'reloadxml' then 'reload mod_logfile'
> >
> > -Steve
> >
> >
> >
> > On 20 February 2013 15:10, Beeton, Carolyn (Carolyn)
> > <cbeeton at avaya.com> wrote:
> > > I would like to be able to trigger Freeswitch to reload the
> > logfile.conf
> > > file (to pick up changes in log level on the fly).  I know this can
> > be done
> > > by changing levels on the console, but since my application uses
> > > configuration files to persist the changes, I would prefer to keep
> > all
> > > configuration data in the files.
> > >
> > >
> > >
> > > The "reloadxml"  command triggers FS to reload some but not all of
> > the
> > > config files.  Could it be enhanced to reload logfile.conf?  Or would
> > it
> > > make sense to reload the logfile.conf file on receipt of sig_hup?  Or
> > have I
> > > missed another way to accomplish this?  If there's any interest, I
> > will
> > > create a jira and offer a patch.
> > >
> > >
> > >
> > > Thanks,
> > >
> > > Carolyn
> > >
> > >
> > >
> > _______________________________________________________________________
> > __
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.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
> > >
> >
> > _______________________________________________________________________
> > __
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From ec.amazumdar at tatapowersed.com  Sat Feb 23 14:41:37 2013
From: ec.amazumdar at tatapowersed.com (Anindo Mazumdar)
Date: Sat, 23 Feb 2013 17:11:37 +0530
Subject: [Freeswitch-dev] Error while registering client in FreeSWITCH
Message-ID: <201302231141.r1NBfbg8022158@blr.tatapowersed.com>

Hello All,

I have configured FreeSWITCH on my system and wanted to test it by registering client to it. The client in my case is Grandstream GXP1405. Once I have configured the client,I am  trying to register it with FreeSWITCH. I am getting the error "Destination Unreachable(Port Unreachable)".I am attaching the wireshark screenshot of the same. Can anyone guide me how to rectify the same.
-- 
With Regards,

Anindo Mazumdar,

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From steveayre at gmail.com  Sat Feb 23 15:57:29 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 23 Feb 2013 12:57:29 +0000
Subject: [Freeswitch-dev] Error while registering client in FreeSWITCH
In-Reply-To: <201302231141.r1NBfbg8022158@blr.tatapowersed.com>
References: <201302231141.r1NBfbg8022158@blr.tatapowersed.com>
Message-ID: <CAFiqYu=12A8gx1YRwhn_x-REK9jzbabAuDiLOB8aTsSmV3nz+w@mail.gmail.com>

Check the packets that the ICMP packet is a reply to.

Verify the IP and port are correct - that packet usually will mean either
nothing is listening on that IP:port or that it has been blocked by a
firewall.

Also is the ICMP being sent FS->Client or Client->FS? That is, is it the
REGISTER or 200 OK packet that is failing?

-Steve



On 23 February 2013 11:41, Anindo Mazumdar <ec.amazumdar at tatapowersed.com>wrote:

> Hello All,
>
> I have configured FreeSWITCH on my system and wanted to test it by
> registering client to it. The client in my case is Grandstream GXP1405.
> Once I have configured the client,I am  trying to register it with
> FreeSWITCH. I am getting the error "Destination Unreachable(Port
> Unreachable)".I am attaching the wireshark screenshot of the same. Can
> anyone guide me how to rectify the same.
> --
> With Regards,
>
> Anindo Mazumdar,
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
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From alex at jajah.com  Sun Feb 24 10:12:57 2013
From: alex at jajah.com (Alex Massover)
Date: Sun, 24 Feb 2013 09:12:57 +0200
Subject: [Freeswitch-dev] on_init hook and outgoing channels
In-Reply-To: <569384504C492C4580E88B5D54DFEAEA33DA3DF2EE@jjex01.jajah.dublin>
References: <569384504C492C4580E88B5D54DFEAEA33DA3DF2EE@jjex01.jajah.dublin>
Message-ID: <569384504C492C4580E88B5D54DFEAEA33DA3DF858@jjex01.jajah.dublin>

Hi,

After some research it looks like mod_sofia sends outgoing INVITE also from on_init hook and endpoint's hooks always happen before other modules hooks (core's behavior).

I see one way to take control over channel before it sends INVITE is to change states in mod_sofia. Could someone point me to more elegant solution, that doesn't require changing mod_sofia?
I read on wiki that it's possible to add custom states, can it help somehow?

BR, Alex.

From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Alex Massover
Sent: Thursday, February 21, 2013 12:15 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] on_init hook and outgoing channels

Hello,

We have services that manage calls via ESL and want to write a custom authorization and billing module.

The idea is to catch any call received by FreeSwitch before service gets control on ESL for incoming call and on_init hook works for us.

But for outgoing channels (created by 'route' or 'originate' commands over ESL) on_init hook doesn't work. Well the channel hits the hook but already after outgoing INVITE being sent, although channel state goes from CS_NEW->CS_INIT->CS_ROUTING correctly. And I want to authorize the channel before it goes out.

Is that a normal behavior or I completely miss something? Is there other way to block an outgoing channel after it's created but before it goes out from a custom module?

--
Best Regards,
Alex Massover

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From anthony.minessale at gmail.com  Sun Feb 24 16:24:04 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 24 Feb 2013 07:24:04 -0600
Subject: [Freeswitch-dev] on_init hook and outgoing channels
In-Reply-To: <569384504C492C4580E88B5D54DFEAEA33DA3DF858@jjex01.jajah.dublin>
References: <569384504C492C4580E88B5D54DFEAEA33DA3DF2EE@jjex01.jajah.dublin>
	<569384504C492C4580E88B5D54DFEAEA33DA3DF858@jjex01.jajah.dublin>
Message-ID: <CAKbxfG-j3N8GZXKEwkNnGXnX1AV6YXgoTk+4nbdEUOUMQ6s7SQ@mail.gmail.com>

There probably is not a way.  File a Jira as a feature Req for it....
On Feb 24, 2013 1:43 AM, "Alex Massover" <alex at jajah.com> wrote:

> Hi,****
>
> ** **
>
> After some research it looks like mod_sofia sends outgoing INVITE also
> from on_init hook and endpoint's hooks always happen before other modules
> hooks (core's behavior).****
>
> ** **
>
> I see one way to take control over channel before it sends INVITE is to
> change states in mod_sofia. Could someone point me to more elegant
> solution, that doesn't require changing mod_sofia?****
>
> I read on wiki that it's possible to add custom states, can it help
> somehow?****
>
> ** **
>
> BR, Alex.****
>
> ** **
>
> *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Alex Massover
> *Sent:* Thursday, February 21, 2013 12:15 PM
> *To:* freeswitch-dev at lists.freeswitch.org
> *Subject:* [Freeswitch-dev] on_init hook and outgoing channels****
>
> ** **
>
> Hello,****
>
> ** **
>
> We have services that manage calls via ESL and want to write a custom
> authorization and billing module.****
>
> ** **
>
> The idea is to catch any call received by FreeSwitch before service gets
> control on ESL for incoming call and on_init hook works for us.****
>
> ** **
>
> But for outgoing channels (created by 'route' or 'originate' commands over
> ESL) on_init hook doesn't work. Well the channel hits the hook but already
> after outgoing INVITE being sent, although channel state goes from
> CS_NEW->CS_INIT->CS_ROUTING correctly. And I want to authorize the channel
> before it goes out.****
>
> ** **
>
> Is that a normal behavior or I completely miss something? Is there other
> way to block an outgoing channel after it's created but before it goes out
> from a custom module?****
>
> ** **
>
> --****
>
> Best Regards,****
>
> Alex Massover****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
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From lconroy at insensate.co.uk  Mon Feb 25 02:35:30 2013
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Sun, 24 Feb 2013 23:35:30 +0000
Subject: [Freeswitch-dev] Q: anyone built fS (head) on Mountain Lion?
Message-ID: <E4EC0AD2-2C6E-4419-AE55-86BB59D28F1B@insensate.co.uk>

Hi there,
 quick question: as per subject, has anyone succeeded in building fS on Mac OS X mountain lion with portaudio.
At this end it seems to barf trying to make universal binaries (i.e., with ppc code).

That isn't going to fly as ppc64 is no longer supported with Mountain Lion's SDK, AFAICT.

Yes, on a clean update to ML I did have to pull in and install (m4), autoconf and automake and libtool first.

all the best from a puzzled
  Lawrence



From dujinfang at gmail.com  Mon Feb 25 04:12:34 2013
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 25 Feb 2013 09:12:34 +0800
Subject: [Freeswitch-dev] Q: anyone built fS (head) on Mountain Lion?
In-Reply-To: <E4EC0AD2-2C6E-4419-AE55-86BB59D28F1B@insensate.co.uk>
References: <E4EC0AD2-2C6E-4419-AE55-86BB59D28F1B@insensate.co.uk>
Message-ID: <2D6C0B365FBC4F658D83C38CED47E0C8@gmail.com>

you could manually edit the Makefile and remove the -arch ppc -arch i386 etc.

however, it still have problems to build and there's some open lira's you could search.

there's a portaudio lib you can install with brew, it build cleanly and I had successfully linked to it in FS without problem but it crashes we you load or use it.

Comment on jira and perhaps some bounty might help.

Seven.

On Monday, February 25, 2013 at 7:35 AM, Lawrence Conroy wrote: 
> Hi there,
> quick question: as per subject, has anyone succeeded in building fS on Mac OS X mountain lion with portaudio.
> At this end it seems to barf trying to make universal binaries (i.e., with ppc code).
> 
> That isn't going to fly as ppc64 is no longer supported with Mountain Lion's SDK, AFAICT.
> 
> Yes, on a clean update to ML I did have to pull in and install (m4), autoconf and automake and libtool first.
> 
> all the best from a puzzled
> Lawrence
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org (mailto: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 marketing at cluecon.com  Mon Feb 25 21:29:26 2013
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 25 Feb 2013 10:29:26 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxXNW6dCJKRxdNMMMuJB4swACpGoMqFDLENx80mhK-G3qQ@mail.gmail.com>

Greetings!

First item in the news today is a happy report from long-time FreeSWITCH
user Henry Gavin. Henry runs a company in the U.K. called
SureVoIP<http://www.surevoip.co.uk>.
He is pleased to report that "thanks to FreeSWITCH and FusionPBX" his
company is once again a finalist for the annual ITSPA
awards<http://www.surevoip.co.uk/blog/2013/02/25/surevoip-and-the-surevoip-api-finalists-for-the-2013-itspa-awards/>.
Congrats to Henry for leveraging FreeSWITCH in a successful business
endeavor.

Another annual event is the Google Summer of
Code<http://www.google-melange.com/gsoc/homepage/google/gsoc2013>(GSoC).
FreeSWITCH will once again apply as a mentoring organization.
Please start thinking of project ideas that we can include in our
organization's application. Applications will be submitted starting March
18 and no later than March 29. Ken Rice and I will be coordinating this
process. Stay tuned for more details.

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_20>we did a
nice tour of the
CudaTel <> Communication Server. In the coming weeks
we will have more presentations for GUI front-ends that community members
have built. On this week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_27>we will have
Ken Rice give us an update on his new project:
SwitchPi <http://switchpi.org>. If you like DIY projects then you'll
appreciate what Ken has done with integrating the Raspberry Pi with
FreeSWITCH and some other items to create something new. We look forward to
seeing it in action.

Don't forget about the FreeSWITCH HA discussion on Tuesday evening at 8PM
EST. Last week's discussion was very fruitful. Eliot Gable gave us all a
lot of information about the different approaches that he can take for
building mod_ha_cluster <http://wiki.freeswitch.org/wiki/Mod_ha_cluster>.
We look forward to his report on the potential of using
OpenMPI<http://www.open-mpi.org/>.
For those who can't make it to the HA discussion please join weekly
conference call on Wednesday where we will have a brief recap of the HA
call.

In ClueCon <http://www.cluecon.com>news we have uploaded two new videos:

* What's new in sipXecs 4.6 <http://youtu.be/mz14UqXRQDU> - Douglas Hubler
* Challenges and Opportunities in Open Source
VoIP<http://youtu.be/QnDYE-DfjgU>- Travis Cross

Stay tuned for more ClueCon 2012 videos and ClueCon 2013 announcements.

Have a great week and we look forward to talking to you on Wednesday.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From huw.selley at netdev.co.uk  Tue Feb 26 12:28:06 2013
From: huw.selley at netdev.co.uk (Huw Selley)
Date: Tue, 26 Feb 2013 09:28:06 +0000
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKzWOxXNW6dCJKRxdNMMMuJB4swACpGoMqFDLENx80mhK-G3qQ@mail.gmail.com>
References: <CAKzWOxXNW6dCJKRxdNMMMuJB4swACpGoMqFDLENx80mhK-G3qQ@mail.gmail.com>
Message-ID: <EC67EBD4-8DA6-44C0-BA42-BC5F0B548114@netdev.co.uk>


On 25 Feb 2013, at 18:29, Michael Collins <marketing at cluecon.com> wrote:

> Greetings!
> 
> First item in the news today is a happy report from long-time FreeSWITCH user Henry Gavin. Henry runs a company in the U.K.  called SureVoIP. He is pleased to report that "thanks to FreeSWITCH and FusionPBX" his company is once again a finalist for the annual ITSPA awards. Congrats to Henry for leveraging FreeSWITCH in a successful business endeavor.

Don't you mean Gavin Henry? ;)

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From msc at freeswitch.org  Wed Feb 27 19:50:26 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Feb 2013 08:50:26 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxVPP1KHG3RTDiWkdvB6MFjtaAvqQ+kUsZDENQenkVUXeQ@mail.gmail.com>

Hello all. Today's conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_27

We will have an update on the mod_ha_cluster discussion and then Ken Rice
will be showing off his SwitchPi project.

Talk to you soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Feb 28 21:52:31 2013
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Feb 2013 10:52:31 -0800
Subject: [Freeswitch-dev] Friday Free For All Topic: GSoC 2013
Message-ID: <CAKzWOxW_L0QrDS1OR8bwAsnkNf1u54-wpsZ+k+p1rw5QN1+8dw@mail.gmail.com>

Hey all!

We thought that maybe a good idea for a discussion topic for tomorrow's
Friday Free For All (F3A) would be potential projects that FreeSWITCH/OSTAG
could sponsor as a mentoring organization. Please bring your fresh ideas
and we'll talk about them!

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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