[Freeswitch-users] mod_portaudio send 3 rtppacket/60msinsteadof1 packet/20ms
Anthony Minessale
anthony.minessale at gmail.com
Wed May 7 06:18:42 PDT 2008
Have you tried setting the codec-ms in the portaudio.conf.xml to 60 or 120
ms?
Maybe the soundcard is not able to do 20ms intervals and portaudio is doing
the least common multiple and chopping it up for us.
I think what's happening is the timer in the module is set to the interval
from the config file (20ms) and during every 60ms period there is no audio
until the last ms. so in each 60 ms:
20ms (timeout..... flush buffer)
20ms (timeout..... flush buffer)
20ms (get 60ms worth of audio at once [3 20ms packets] but we have already
read 2 filler frames from the timeouts)
So now we have read 5 packets instead of 3 and erased some of our buffer
because of perceived timeouts.
The code is using the assumption that if the device will obey the chosen
frame size and sample rate requests down to the interval.
If you find and edit conf/autoload_configs/portaudio.conf.xml
look for this:
<param name="codec-ms" value="20"/>
and change 20 to 60
Setting this to 60 will change the frame size of all the packets from 320 to
960 and set the timer to clock at an interval of 60ms
Since the card seems to be able to reliably produce 3 20ms packets every
60ms it should also be able to produce 1 60ms packet.
FreeSWITCH should then buffer the audio and still deliver it over SIP at
20ms if you want but you can opt to set the codec PCMU at 60i to disable
buffering if you are in a reliable network.
The same should be true for setting the codec-ms to 120
On Wed, May 7, 2008 at 3:27 AM, Sluschny, Thomas <
Thomas.Sluschny at siemens.com> wrote:
> Hi Anthony,
>
> i also tested your patch with no success.
> As i already described below, the problem with all 60ms 3 packets comes
> from the soundcard.
> The hardware delivers its samples all 60 ms.
> Our problem is (like Csaba said) that we read out the buffer after 60ms, 3
> times, each with samples for 20ms, AND WITH NO DELAY!
> So we get: 60ms wait and 3 RTP packets within <1ms to send, and after that
> we already wait 60 ms for the next samples.
>
> In my patch i wait appr.20 ms if last method call was no longer than 4ms
> ago,
> but i think we can do better with switch_core_timer_check() method, but i
> don't know exactly how.
>
> You are absolutly right with your demand for a better timing resolution
> under Windows,
> but this 60ms mystery is caused by the soundcard.
>
> Thomas
>
> ------------------------------
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Zelei
> Csaba
> *Gesendet:* Dienstag, 6. Mai 2008 20:22
>
> *An:* freeswitch-users at lists.freeswitch.org
> *Betreff:* Re: [Freeswitch-users] mod_portaudio send 3
> rtppacket/60msinsteadof1 packet/20ms
>
> Yes, but it didnt help with portaudio (just together with the patch)
> What else do you need to test it with? Please tell me, and i will do it
> for you tomorrow.
> I also found a test code for timeBeginPeriod on this site:
> http://www.geisswerks.com/ryan/FAQS/timing.html and experienced that
> sleep(1) is usually 2 ms long (sometimes 4)
>
>
> 2008/5/6 Anthony Minessale <anthony.minessale at gmail.com>:
>
> > Did you try the trunk with no pathc since my latest email r8269? I
> > added code that makes the windows timer accurate to 1ms precision. If the
> > windows timer must be made accurate or many other things in FS will not work
> > right.
> >
> >
> >
> > On Tue, May 6, 2008 at 10:59 AM, Csaba Zelei <csaba.zelei at gmail.com>
> > wrote:
> >
> > > Hi Anthony,
> > >
> > > I tried your suggestions (the PCMU at 30i 60i and omitting line 158 in
> > > pablio.c) but none solve the problem.
> > > I also tried the latest trunk but its the same.
> > > I reached the best performance with the latest trunk and the patch
> > > Thomas wrote.(~20ms rtp packet delta, rarely 100+ms)
> > > The only problem with it thats FS ate up 95% cpu but I am not sure if
> > > it has to do anything with mod_portaudio or the windows timer.
> > >
> > > If I am correct we have 2 problem here.
> > > The first is windows timer isnt very accurate.
> > > The second is that mod_portaudio dont have any delay when it can
> > > return audio data immadiately .
> > > Let me explain:
> > >
> > > 1. call for channel_read_frame
> > > 2. waiting for audio data (~60ms)
> > > 3. got 60ms audio data
> > > 4. get 20ms (remaining 40ms) audio data, return (and send rtp frame)
> > > 5. call for channel_read_frame
> > > 6. get 20ms, return(and send rtp frame) within ~1ms
> > > 7. call for channel_read_frame
> > > 8. get 20ms, return(and send rtp frame) within ~1ms
> > > go back to 1 and start again
> > >
> > > That's what I tried to solve with my patch, but because of the windows
> > > timer it failed.
> > > I hope I could help, and dont misunderstand something.
> > >
> > > Csaba
> > >
> > > Anthony Minessale írta:
> > >
> > > I added a small patch to turn up the resolution of Sleep to 1ms.
> > > Can you see if that helps?
> > >
> > > On Tue, May 6, 2008 at 8:38 AM, Sluschny, Thomas <
> > > Thomas.Sluschny at siemens.com> wrote:
> > >
> > > > i already tried to implement such a routine, but as i remember i
> > > > had problems to compile,
> > > > anyway this may be help you:
> > > >
> > > > // sleep with smaller tick time, wait in µs
> > > > void betterSleep(long wait)
> > > > {
> > > > LARGE_INTEGER lElapse;
> > > > BOOL succ;
> > > > HANDLE timerHandle = 0;
> > > > DWORD dwWaitResult;
> > > >
> > > > __try {
> > > >
> > > > // return on 0
> > > > if (wait < 1)
> > > > return;
> > > >
> > > > // create the timer
> > > > timerHandle = (HANDLE) CreateWaitableTimerA(NULL, FALSE, NULL);
> > > >
> > > > // set the timer
> > > > lElapse.QuadPart = 2500LL - (wait * 10000LL);
> > > > succ = SetWaitableTimer(timerHandle, &lElapse, 0, NULL, NULL,
> > > > FALSE);
> > > >
> > > > if (!succ) {
> > > > printf( "set timer not successful!\n" );
> > > > }
> > > >
> > > > dwWaitResult = WaitForSingleObject(
> > > > timerHandle,
> > > > wait // fall back timeout with 15ms granularity
> > > > );
> > > >
> > > > switch (dwWaitResult)
> > > > {
> > > > // The thread got mutex ownership.
> > > > case WAIT_OBJECT_0:
> > > > //printf( "WAIT_OBJECT_0!\n" );
> > > > break;
> > > >
> > > > // Cannot get mutex ownership due to time-out.
> > > > case WAIT_TIMEOUT:
> > > > printf( "WAIT_TIMEOUT!\n" );
> > > > break;
> > > >
> > > > // Got ownership of the abandoned mutex object.
> > > > case WAIT_ABANDONED:
> > > > printf( "WAIT_ABANDONED!\n" );
> > > > break;
> > > > }
> > > > }
> > > > __finally {
> > > > if (timerHandle) CloseHandle(timerHandle);
> > > > }
> > > > }
> > > >
> > > > It may be hard work but: Its cool to be cross platform - you reach
> > > > all people around
> > > > we count on you ;)
> > > >
> > > > Thomas
> > > >
> > > > ------------------------------
> > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> > > > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
> > > > Minessale
> > > > *Gesendet:* Dienstag, 6. Mai 2008 15:08
> > > > *An:* freeswitch-users at lists.freeswitch.org
> > > > *Betreff:* Re: [Freeswitch-users] mod_portaudio send 3 rtp
> > > > packet/60msinsteadof1 packet/20ms
> > > >
> > > > if switch_yield for windows is not working properly maybe if we fix
> > > > that the code will work as planned.
> > > >
> > > > switch_timer_check tells you if the timer has ticked or not and it's
> > > > based on a single timer thread that also has the necessity to do 1ms sleeps
> > > > to broadcast the time changes to the rest of the system. so let's try using
> > > > performance counters to implement proper switch_yield and see if the code
> > > > begins to work as planned. I'll try to come up with a patch because there
> > > > are several places including the RTP code where semi-accurate 1ms sleeps are
> > > > absolutely necessary.
> > > >
> > > > This is one of the many joys of being cross platform. =D
> > > >
> > > >
> > > >
> > > >
> > > > On Tue, May 6, 2008 at 7:37 AM, Sluschny, Thomas <
> > > > Thomas.Sluschny at siemens.com> wrote:
> > > >
> > > > > Anthony seams to be right.
> > > > > I tested around a little bit and i see:
> > > > > - the main problem is the soundcard (-driver), it gives 480 frames
> > > > > all 60 ms
> > > > > - if i chose 160 (256) frames ringbuffer size i get only 160
> > > > > frames all 60 ms, and it sounds really croppy (of course, there are not
> > > > > enough samples ...)
> > > > > - i get really good results if i store the last timestamp when
> > > > > ReadAudioStream() returns, and if current timestamp is less than
> > > > > 20ms from last i wait 20 ms,
> > > > > so the packet difference went from 60-0-0-60-0-0 ms to
> > > > > 20-20-20-20-20-20
> > > > > - i use HighPerformanceCounter to measure the time in WinXP
> > > > >
> > > > > I thought we could use switch_core_timer_check() to do that
> > > > > timestamp work, but it doesn't work as expected,
> > > > > seams we have to synchronize to the 60ms clock from soundcard, at
> > > > > least at the beginning, which means the delay increase.
> > > > >
> > > > > Thomas
> > > > >
> > > > > PS: i attach a patch for better understanding
> > > > >
> > > > > ------------------------------
> > > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> > > > > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
> > > > > Minessale
> > > > > *Gesendet:* Dienstag, 6. Mai 2008 00:38
> > > > > *An:* freeswitch-users at lists.freeswitch.org
> > > > > *Betreff:* Re: [Freeswitch-users] mod_portaudio send 3 rtp
> > > > > packet/60ms insteadof1 packet/20ms
> > > > >
> > > > > This is not normal sleep it's microsecond sleep. Its done
> > > > > because we are doing nonblocking read on the ringbuffer that's tied to the
> > > > > hardware. Since this is voip, we must drop audio frames when they are late
> > > > > and in order to do that we must have a high resolution loop. This is only a
> > > > > problem when the audio device is not sending audio at the interval we asked
> > > > > it to. Some cheap hardware cannot reliably deliver audio at 20ms intervals
> > > > > which is why i suggested higher value intervals in the config. The request
> > > > > to remove the sleep is to confirm the proposition that sleep 1ms was really
> > > > > taking 15ms.
> > > > >
> > > > > I appreciate the suggestion and I understand you are not trying to
> > > > > be a smart ass.
> > > > >
> > > > >
> > > > >
> > > > > On Mon, May 5, 2008 at 4:35 PM, Michael Jerris <mike at jerris.com>
> > > > > wrote:
> > > > >
> > > > > > The place this is trickiest is when you are in a loop where you
> > > > > > want
> > > > > > to wait for audio, AND do something every x ms or so. You can't
> > > > > > do a
> > > > > > blocking read, and a read with timeout tends to be quite
> > > > > > expensive.
> > > > > > There are some ways around this, but sometimes its just the most
> > > > > > efficient, even if not preferred method.
> > > > > >
> > > > > > Mike
> > > > > >
> > > > > >
> > > > > > On May 5, 2008, at 5:26 PM, ?ukasz Zwierko wrote:
> > > > > >
> > > > > > > Hi,
> > > > > > >
> > > > > > > Just a thought here: using calls like delay(), sleep() etc.
> > > > > > for very
> > > > > > > short amount of time (like in this case a couple of
> > > > > > miliseconds)
> > > > > > > should in my opinion be really discouraged. It is often a
> > > > > > symptom of
> > > > > > > bad programming even in an embedded enviroment, not mentioning
> > > > > > > platforms like Windows or Linux where you can't really tell
> > > > > > what
> > > > > > > priority does your task have, and will it not be starved for a
> > > > > > long
> > > > > > > time by other tasks. I should be avoided whenever possible.
> > > > > > > Don't want to sound like a smart ass here, but isn't there any
> > > > > > other
> > > > > > > way? From what I understood you wait until some amount of
> > > > > > voice
> > > > > > > samples is collected? If that's the case than perhaps you can
> > > > > > measure
> > > > > > > an amount of data collected not the time... If these are PCM
> > > > > > samples
> > > > > > > than the correlation is straightforward.
> > > > > > > Again, sorry if I'm being a smart ass here but I've seen some
> > > > > > really
> > > > > > > bad code with sleep() calls and such like, and I can tell you
> > > > > > that it
> > > > > > > only worked fine in specific conditions, and had a tendency to
> > > > > > work
> > > > > > > very poorly when for example CPU was under heavy load.
> > > > > > >
> > > > > > > Luaksz
> > > > > > >
> > > > > > > 2008/5/5 Anthony Minessale <anthony.minessale at gmail.com>:
> > > > > > >> did you try setting the ptime on the rtp to 30 or 60ms
> > > > > > >>
> > > > > > >> when you choose a codec in your sip settings on FS in
> > > > > > vars.xml,
> > > > > > >> instead of
> > > > > > >> PCMU try PCMU at 30i or PCMU at 60i
> > > > > > >> it may be that the other side is doing 30 or 60 ms and not
> > > > > > telling
> > > > > > >> us.
> > > > > > >>
> > > > > > >> also in pablio.c in the portaudio_mod directory in
> > > > > > ReadAudioStream
> > > > > > >> func,
> > > > > > >> there is a sleep 1 ms too
> > > > > > >> if the windows is really sleeping a lot longer than that, try
> > > > > > >> omitting line
> > > > > > >> 158.
> > > > > > >> This probably will consume the whole cpu but if it fixes your
> > > > > > >> problem it
> > > > > > >> will support the theory that the sleep on windows in
> > > > > > inaccurate.
> > > > > > >>
> > > > > > >>
> > > > > > >>
> > > > > > >>
> > > > > > >>
> > > > > > >> On Mon, May 5, 2008 at 11:38 AM, Csaba Zelei
> > > > > > >> <csaba.zelei at gmail.com> wrote:
> > > > > > >>>
> > > > > > >>>
> > > > > > >>>
> > > > > > >>>
> > > > > > >>> With a little hack I can make mod_portaudio to send rtp
> > > > > > packets with
> > > > > > >> ~24ms, ~16ms delay alternately on linux. This result in a
> > > > > > constant
> > > > > > >> 4ms
> > > > > > >> jitter but its better than the original. (diff attached)
> > > > > > >>> However on windows the delta between rtp packets is 15-32 ms
> > > > > > >>> randomly,
> > > > > > >> with occasionally high 70-100ms delta.
> > > > > > >>> I also tried to tweak the windows timer without success.
> > > > > > >>> Does anybody has any idea how to make windows xp more
> > > > > > accurate?
> > > > > > >>>
> > > > > > >>> Sluschny, Thomas wrote:
> > > > > > >>>
> > > > > > >>>
> > > > > > >>>
> > > > > > >>>
> > > > > > >>> as you can see here:
> > > > > > >>>
> > > > > > >>>
> > > > > > >>> http://jira.freeswitch.org/browse/MODENDP-40
> > > > > > >>>
> > > > > > >>> i have this problem all the time (the error mentioned in
> > > > > > this
> > > > > > >>> issue was
> > > > > > >> only related with this).
> > > > > > >>>
> > > > > > >>> It has to do with windows handle sleep() method, you has say
> > > > > > >>> sleep(1) for
> > > > > > >> 1ms but on my
> > > > > > >>>
> > > > > > >>> machine it waits 15ms (it depends on your hardware, other
> > > > > > PCs behave
> > > > > > >> different!). So i tested around with high performance
> > > > > > counters.
> > > > > > >>>
> > > > > > >>> For now i ignore that problem an set jitterbuffers on other
> > > > > > device
> > > > > > >>> big
> > > > > > >> enough.
> > > > > > >>>
> > > > > > >>> Thomas
> > > > > > >>>
> > > > > > >>> ________________________________
> > > > > > >> Von: freeswitch-users-bounces at lists.freeswitch.org
> > > > > > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im
> > > > > > Auftrag
> > > > > > >> von Zelei
> > > > > > >> Csaba
> > > > > > >>> Gesendet: Donnerstag, 24. April 2008 19:05
> > > > > > >>> An: freeswitch-users at lists.freeswitch.org
> > > > > > >>> Betreff: [Freeswitch-users] mod_portaudio send 3 rtp
> > > > > > packet/60ms
> > > > > > >>> instead
> > > > > > >> of1 packet/20ms
> > > > > > >>>
> > > > > > >>> Dear all,
> > > > > > >>>
> > > > > > >>> I tried to use FS in client mode, starting calls with
> > > > > > >>> mod_portaudio to our
> > > > > > >> providers gateway ( a Cirpack softswitch )
> > > > > > >>> I experienced that there is 2-3 sec delay in the call, its
> > > > > > choppy
> > > > > > >>> and
> > > > > > >> robot like.
> > > > > > >>> I tested it with a softphone, and an ip phone and everything
> > > > > > was
> > > > > > >>> fine. I
> > > > > > >> traced back the problem to mod_portaudio sending 3 rtp packet
> > > > > > in 60ms
> > > > > > >> instead of 1 packet/20ms.
> > > > > > >>>
> > > > > > >>> Here is an rtp statistic from a call: (see
> > > > > > >> http://pastebin.freeswitch.org/4307 for the complete list and
> > > > > > sip
> > > > > > >> trace)
> > > > > > >>>
> > > > > > >>> Packet Sequence Delta (ms)
> > > > > > >>> 42 26138 0.00
> > > > > > >>> 43 26139 0.02
> > > > > > >>> 46 26140 45.69
> > > > > > >>> 47 26141 0.02
> > > > > > >>> 48 26142 2.96
> > > > > > >>> 52 26143 56.31
> > > > > > >>> 53 26144 5.75
> > > > > > >>> 54 26145 0.02
> > > > > > >>> 58 26146 51.99
> > > > > > >>> 59 26147 0.03
> > > > > > >>> 60 26148 2.96
> > > > > > >>> 63 26149 42.95
> > > > > > >>> 65 26150 17.06
> > > > > > >>> 66 26151 0.02
> > > > > > >>> 67 26152 2.90
> > > > > > >>> 71 26153 56.99
> > > > > > >>> 72 26154 0.03
> > > > > > >>> 73 26155 0.02
> > > > > > >>>
> > > > > > >>> Did anyone else experience similar problems?
> > > > > > >>> Is this the desired behaviour, because portaudio get data in
> > > > > > 60ms
> > > > > > >>> interval
> > > > > > >> or can I set it to 20ms somehow?
> > > > > > >>>
> > > > > > >>> Thanks,
> > > > > > >>>
> > > > > > >>> Csaba Zelei
> > > > > > >>>
> > > > > > >>>
> > > > > > >>> ________________________________
> > > > > > >>
> > > > > > >>> _______________________________________________
> > > > > > >>> Freeswitch-users mailing list
> > > > > > >>> Freeswitch-users at lists.freeswitch.org
> > > > > > >>>
> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > >>> UNSUBSCRIBE:
> > > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > > >>> http://www.freeswitch.org
> > > > > > >>>
> > > > > > >>>
> > > > > > >>>
> > > > > > >>> Index: mod_portaudio.c
> > > > > > >>>
> > > > > > ===================================================================
> > > > > > >>> --- mod_portaudio.c (revision 8260)
> > > > > > >>> +++ mod_portaudio.c (working copy)
> > > > > > >>> @@ -121,6 +121,7 @@
> > > > > > >>> int ring_interval;
> > > > > > >>> GFLAGS flags;
> > > > > > >>> switch_timer_t timer;
> > > > > > >>> + switch_timer_t sync_timer;
> > > > > > >>> } globals;
> > > > > > >>>
> > > > > > >>>
> > > > > > >>> @@ -282,7 +283,15 @@
> > > > > > >>> }
> > > > > > >>>
> > > > > > >>> switch_set_flag_locked(tech_pvt, TFLAG_IO);
> > > > > > >>> +
> > > > > > >>> + /* Start Synchronization Timer */
> > > > > > >>> + //Is it ok to always use 20ms? What about the 160
> > > > > > sample????
> > > > > > >>> + if (
> > > > > > >> switch_core_timer_init(&globals.sync_timer,"soft",
> > > > > > >> 20,160,switch_core_session_get_pool(session))
> > > > > > >> != SWITCH_STATUS_SUCCESS)
> > > > > > >>> + {
> > > > > > >>> + switch_log_printf(SWITCH_CHANNEL_LOG,
> > > > > > >>> SWITCH_LOG_DEBUG,
> > > > > > >> "Sync Timer failed!!\n");
> > > > > > >>> + }
> > > > > > >>>
> > > > > > >>> +
> > > > > > >>> /* Move Channel's State Machine to RING */
> > > > > > >>> switch_channel_set_state(channel, CS_RING);
> > > > > > >>>
> > > > > > >>> @@ -412,6 +421,8 @@
> > > > > > >>> }
> > > > > > >>>
> > > > > > >>> switch_log_printf(SWITCH_CHANNEL_LOG,
> > > > > > SWITCH_LOG_DEBUG, "%s
> > > > > > >>> CHANNEL
> > > > > > >> HANGUP\n",
> > > > > > >>
> > > > > > switch_channel_get_name(switch_core_session_get_channel(session)));
> > > > > > >>> + /* Destroy timer */
> > > > > > >>> + switch_core_timer_destroy(&globals.sync_timer);
> > > > > > >>>
> > > > > > >>> return SWITCH_STATUS_SUCCESS;
> > > > > > >>> }
> > > > > > >>> @@ -542,12 +553,17 @@
> > > > > > >>> switch_mutex_lock(globals.device_lock);
> > > > > > >>>
> > > > > > >>> get_samples:
> > > > > > >>> -
> > > > > > >>> +
> > > > > > >>> if ((samples = ReadAudioStream(globals.audio_stream,
> > > > > > >> globals.read_frame.data,
> > > > > > >>>
> > > > > > >> globals.read_codec.implementation->samples_per_frame,
> > > > > > >>>
> > > > > > >> &globals.timer)) == 0) {
> > > > > > >>> +
> > > > > > >>> + //switch_log_printf(SWITCH_CHANNEL_LOG,
> > > > > > SWITCH_LOG_DEBUG,
> > > > > > >>> "No data
> > > > > > >> reset timer\n");
> > > > > > >>> + switch_core_timer_sync(&globals.sync_timer);
> > > > > > >>> switch_yield(1000);
> > > > > > >>> +
> > > > > > >>> goto get_samples;
> > > > > > >>> +
> > > > > > >>> } else {
> > > > > > >>> globals.read_frame.datalen = samples * 2;
> > > > > > >>> globals.read_frame.samples = samples;
> > > > > > >>> @@ -562,7 +578,9 @@
> > > > > > >>> status = SWITCH_STATUS_SUCCESS;
> > > > > > >>> }
> > > > > > >>> switch_mutex_unlock(globals.device_lock);
> > > > > > >>> -
> > > > > > >>> +
> > > > > > >>> + switch_core_timer_next(&globals.sync_timer);
> > > > > > >>> +
> > > > > > >>> return status;
> > > > > > >>>
> > > > > > >>> }
> > > > > > >>>
> > > > > > >>>
> > > > > > >>> _______________________________________________
> > > > > > >>> Freeswitch-users mailing list
> > > > > > >>> Freeswitch-users at lists.freeswitch.org
> > > > > > >>>
> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > >>> UNSUBSCRIBE:
> > > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > > >>> http://www.freeswitch.org
> > > > > > >>>
> > > > > > >>>
> > > > > > >>
> > > > > > >>
> > > > > > >>
> > > > > > >> --
> > > > > > >> Anthony Minessale II
> > > > > > >>
> > > > > > >> FreeSWITCH http://www.freeswitch.org/
> > > > > > >> ClueCon http://www.cluecon.com/
> > > > > > >>
> > > > > > >> AIM: anthm
> > > > > > >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> > > > > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > > > > > >> IRC: irc.freenode.net #freeswitch
> > > > > > >>
> > > > > > >> FreeSWITCH Developer Conference
> > > > > > >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> > > > > > >> iax:guest at conference.freeswitch.org/888
> > > > > > >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > > > > > >> pstn:213-799-1400
> > > > > > >> _______________________________________________
> > > > > > >> Freeswitch-users mailing list
> > > > > > >> Freeswitch-users at lists.freeswitch.org
> > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > >> UNSUBSCRIBE:
> > > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > > >> http://www.freeswitch.org
> > > > > > >>
> > > > > > >>
> > > > > > >
> > > > > > > _______________________________________________
> > > > > > > Freeswitch-users mailing list
> > > > > > > Freeswitch-users at lists.freeswitch.org
> > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > > UNSUBSCRIBE:
> > > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > > > http://www.freeswitch.org
> > > > > >
> > > > > >
> > > > > > _______________________________________________
> > > > > > Freeswitch-users mailing list
> > > > > > Freeswitch-users at lists.freeswitch.org
> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > UNSUBSCRIBE:
> > > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > > http://www.freeswitch.org
> > > > > >
> > > > >
> > > > >
> > > > >
> > > > > --
> > > > > Anthony Minessale II
> > > > >
> > > > > FreeSWITCH http://www.freeswitch.org/
> > > > > ClueCon http://www.cluecon.com/
> > > > >
> > > > > AIM: anthm
> > > > > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > > > > IRC: irc.freenode.net #freeswitch
> > > > >
> > > > > FreeSWITCH Developer Conference
> > > > > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> > > > > iax:guest at conference.freeswitch.org/888
> > > > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > > > > pstn:213-799-1400
> > > > >
> > > > > _______________________________________________
> > > > > Freeswitch-users mailing list
> > > > > Freeswitch-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > http://www.freeswitch.org
> > > > >
> > > > >
> > > >
> > > >
> > > > --
> > > > Anthony Minessale II
> > > >
> > > > FreeSWITCH http://www.freeswitch.org/
> > > > ClueCon http://www.cluecon.com/
> > > >
> > > > AIM: anthm
> > > > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > > > IRC: irc.freenode.net #freeswitch
> > > >
> > > > FreeSWITCH Developer Conference
> > > > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> > > > iax:guest at conference.freeswitch.org/888
> > > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > > > pstn:213-799-1400
> > > >
> > > > _______________________________________________
> > > > Freeswitch-users mailing list
> > > > Freeswitch-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > >
> > > >
> > >
> > >
> > > --
> > > Anthony Minessale II
> > >
> > > FreeSWITCH http://www.freeswitch.org/
> > > ClueCon http://www.cluecon.com/
> > >
> > > AIM: anthm
> > > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > > IRC: irc.freenode.net #freeswitch
> > >
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> > > iax:guest at conference.freeswitch.org/888
> > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > > pstn:213-799-1400
> > >
> > > ------------------------------
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
> > >
> > >
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing list
> > > Freeswitch-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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