<p>I can&#39;t comment on the windows comparability, but it&#39;s a one time fee and is only required when you need to transcode (including media bugs for recording, TTS, voicemail, etc.). If you don&#39;t do that on a channel then that license isn&#39;t used. </p>

<div class="gmail_quote">On Apr 10, 2012 2:02 PM, &quot;Malay Thakershi&quot; &lt;<a href="mailto:mthakershi@gmail.com">mthakershi@gmail.com</a>&gt; wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Are these licenses supported on windows? If not, how can I use G729 on it?<div><br></div><div>One license is 1 encoder/decoder per channel. So if I make 30 concurrent calls, do I need 30 licenses?</div><div><br></div><div>

Is the $10 fee per month or one-time?</div><div><br></div><div>If I can&#39;t use G729 at all, what could be my options to save bandwidth + get acceptable audio quality?</div><div><br></div><div>Thanks for responses.</div>

<div><br></div><div>Malay<br><br><div class="gmail_quote">On Tue, Apr 10, 2012 at 12:34 AM, Peter Olsson <span dir="ltr">&lt;<a href="mailto:peter.olsson@visionutveckling.se" target="_blank">peter.olsson@visionutveckling.se</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">TTS needs a real license for G729 - please check here how to purchase it: <a href="http://www.freeswitch.org/node/235" target="_blank">http://www.freeswitch.org/node/235</a><br>


<br>
The free G729 modules is for passthrough only, if you generate audio on the FS instance, it will require encoder/decoder licenses.<br>
<br>
/Peter<br>
<br>
________________________________<br>
Från: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] för Malay Thakershi [<a href="mailto:mthakershi@gmail.com" target="_blank">mthakershi@gmail.com</a>]<br>


Skickat: den 10 april 2012 07:22<br>
Till: FreeSWITCH Users Help<br>
Ämne: Re: [Freeswitch-users] Possible to disable core codecs?<br>
<div><div><br>
Upgraded FS to latest version before coming back for help.<br>
<br>
Now, I get error saying:<br>
[ERR] mod_g729.c:102 This codec is only usable in passthrough mode!<br>
[ERR] switch_core_io.c:1081 Codec G.729 encoder error!<br>
<br>
I think what is happening is, it agrees to use G.729 codec but as soon as TTS is opened, there is an error. I read in documentation that mod_g729 is for free codec.<br>
<br>
Please help.<br>
<br>
Here is the debug trace of the call:<br>
----------------------------------<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5572 Remote SDP:<br>
v=0<br>
o=root 29993 29993 IN IP4 IP1<br>
s=session<br>
c=IN IP4 IP1<br>
t=0 0<br>
m=audio 15582 RTP/AVP 0 8 3 18 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[G72<br>
21:115:32000:20:48000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[G72<br>
21:115:32000:20:48000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[G722<br>
1:115:32000:20:48000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G72<br>
21:115:32000:20:48000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:2<br>
0:0]/[G7221:115:32000:20:48000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[AMR<br>
:96:8000:20:12200]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[AMR<br>
:96:8000:20:12200]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[AMR:<br>
96:8000:20:12200]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[AMR<br>
:96:8000:20:12200]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:2<br>
0:0]/[AMR:96:8000:20:12200]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPE<br>
EX:99:32000:20:44000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[SPE<br>
EX:99:32000:20:44000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[SPEE<br>
X:99:32000:20:44000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[SPE<br>
EX:99:32000:20:44000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:2<br>
0:0]/[SPEEX:99:32000:20:44000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[G72<br>
9:18:8000:20:8000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[G72<br>
9:18:8000:20:8000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[G729<br>
:18:8000:20:8000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G72<br>
9:18:8000:20:8000]<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:3006 Set Codec sofia/sipinterface_1/Phone2@x.24<br>
1.99.201 G729/8000 20 ms 160 samples 8000 bits<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] switch_core_codec.c:111 sofia/sipinterface_1/Phone2@x.241.99<br>
.201 Original read codec set to G729:18<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@<br>
IP1) Running State Change CS_NEW<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:380 (sofia/sipinterface_1/Phone2@<br>
IP1) State NEW<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5786 (sofia/sipinterface_1/Phone2@IP1) Stat<br>
e Change CS_NEW -&gt; CS_INIT<br>
<br>
2012-04-10 00:13:32.940267 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278<br>
29132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@<br>
IP1) Running State Change CS_INIT<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401 (sofia/sipinterface_1/Phone2@<br>
IP1) State INIT<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:85 sofia/sipinterface_1/Phone2@IP1 SOFI<br>
A INIT<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:125 (sofia/sipinterface_1/Phone2@IP1) S<br>
tate Change CS_INIT -&gt; CS_ROUTING<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278<br>
29132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401 (sofia/sipinterface_1/Phone2@<br>
IP1) State INIT going to sleep<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@<br>
IP1) Running State Change CS_ROUTING<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_channel.c:1886 (sofia/sipinterface_1/Phone2@x.241.99.<br>
201) Callstate Change DOWN -&gt; RINGING<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@<br>
IP1) State ROUTING<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148 sofia/sipinterface_1/Phone2@IP1 SOF<br>
IA ROUTING<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104 sofia/sipinterface_1/Phone2@6<br>
6.241.99.201 Standard ROUTING<br>
<br>
2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing +1Phone2 &lt;Phone2&gt;-&gt;Phone1 in context inbound<br>
Dialplan: sofia/sipinterface_1/Phone2@IP1 parsing [inbound-&gt;vitel-inbound] continue=fa<br>
lse<br>
Dialplan: sofia/sipinterface_1/Phone2@IP1 Regex (PASS) [vitel-inbound] destination_num<br>
ber<a href="tel:%288774542559" value="+18774542559" target="_blank">(8774542559</a>) =~ // break=on-false<br>
Dialplan: sofia/sipinterface_1/Phone2@IP1 Action transfer(1056 XML default)<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154 (sofia/sipinterface_1/Phone2@<br>
IP1) State Change CS_ROUTING -&gt; CS_EXECUTE<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278<br>
29132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@<br>
IP1) State ROUTING going to sleep<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@<br>
IP1) Running State Change CS_EXECUTE<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/Phone2@<br>
IP1) State EXECUTE<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241 sofia/sipinterface_1/Phone2@IP1 SOF<br>
IA EXECUTE<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192 sofia/sipinterface_1/Phone2@6<br>
6.241.99.201 Standard EXECUTE<br>
EXECUTE sofia/sipinterface_1/Phone2@IP1 transfer(1056 XML default)<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_ivr.c:1711 (sofia/sipinterface_1/Phone2@IP1)<br>
 State Change CS_EXECUTE -&gt; CS_ROUTING<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278<br>
29132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:731 Send signal sofia/sipinterface_1/972782<br>
9132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:32.960773 [NOTICE] switch_ivr.c:1717 Transfer sofia/sipinterface_1/Phone2@x.24<br>
1.99.201 to XML[1056@default]<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/Phone2@<br>
IP1) State EXECUTE going to sleep<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@<br>
IP1) Running State Change CS_ROUTING<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@<br>
IP1) State ROUTING<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148 sofia/sipinterface_1/Phone2@IP1 SOF<br>
IA ROUTING<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104 sofia/sipinterface_1/Phone2@6<br>
6.241.99.201 Standard ROUTING<br>
<br>
2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing +1Phone2 &lt;Phone2&gt;-&gt;1056<br>
in context default<br>
Dialplan: sofia/sipinterface_1/Phone2@IP1 parsing [default-&gt;CHPhoneAsmtDev] continue=f<br>
alse<br>
Dialplan: sofia/sipinterface_1/Phone2@IP1 Regex (PASS) [CHPhoneAsmtDev] destination_nu<br>
mber(1056) =~ /^105\d$/ break=on-false<br>
Dialplan: sofia/sipinterface_1/Phone2@IP1 Action sleep(1000)<br>
Dialplan: sofia/sipinterface_1/Phone2@IP1 Action managed(clsAsmtApp)<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154 (sofia/sipinterface_1/Phone2@<br>
IP1) State Change CS_ROUTING -&gt; CS_EXECUTE<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278<br>
29132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@<br>
IP1) State ROUTING going to sleep<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@<br>
IP1) Running State Change CS_EXECUTE<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/Phone2@<br>
IP1) State EXECUTE<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241 sofia/sipinterface_1/Phone2@IP1 SOF<br>
IA EXECUTE<br>
<br>
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192 sofia/sipinterface_1/Phone2@6<br>
6.241.99.201 Standard EXECUTE<br>
EXECUTE sofia/sipinterface_1/Phone2@IP1 sleep(1000)<br>
EXECUTE sofia/sipinterface_1/Phone2@IP1 managed(clsAsmtApp)<br>
<br>
2012-04-10 00:13:33.980239 [DEBUG] switch_cpp.cpp:1227 FreeSWITCH.Managed: attempting to run applica<br>
tion &#39;clsAsmtApp&#39;.<br>
<br>
2012-04-10 00:13:34.180422 [DEBUG] switch_cpp.cpp:1172 CoreSession::seHangupHook, hangup_func: 00000<br>
000<br>
<br>
2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:1227 Not an outbound call.<br>
<br>
2012-04-10 00:13:34.780969 [INFO] switch_cpp.cpp:1227 caller_id_number: Phone2<br>
<br>
2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3258 AUDIO RTP [sofia/sipinterface_1/Phone2@x.2<br>
41.99.201] 10.25.20.202 port 16542 -&gt; IP1 port 15582 codec: 18 ms: 20<br>
<br>
2012-04-10 00:13:34.780969 [DEBUG] switch_rtp.c:1669 Not using a timer<br>
<br>
2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3522 Set 2833 dtmf send payload to 101<br>
<br>
2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3528 Set 2833 dtmf receive payload to 101<br>
<br>
2012-04-10 00:13:34.780969 [DEBUG] mod_sofia.c:754 Local SDP sofia/sipinterface_1/Phone2@x.241.<br>
99.201:<br>
v=0<br>
o=FreeSWITCH 1334018272 1334018273 IN IP4 64.22.232.56<br>
s=FreeSWITCH<br>
c=IN IP4 64.22.232.56<br>
t=0 0<br>
m=audio 16542 RTP/AVP 18 101<br>
a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
<br>
2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:731 Send signal sofia/sipinterface_1/972782<br>
9132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:34.780969 [DEBUG] switch_channel.c:3244 (sofia/sipinterface_1/Phone2@x.241.99.<br>
201) Callstate Change RINGING -&gt; ACTIVE<br>
<br>
2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:599 Channel [sofia/sipinterface_1/Phone2@x.2<br>
41.99.201] has been answered<br>
<br>
2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782<br>
9132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:34.780969 [DEBUG] sofia.c:5561 Channel sofia/sipinterface_1/Phone2@x.241.99.20<br>
1 entering state [completed][200]<br>
<br>
2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782<br>
9132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782<br>
9132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782<br>
9132@IP1 [BREAK]<br>
<br>
2012-04-10 00:13:35.420577 [INFO] switch_cpp.cpp:1227 OK. Connected to customer DB.<br>
<br>
2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS flite<br>
<br>
2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec Activated<br>
<br>
2012-04-10 00:13:35.460613 [DEBUG] switch_ivr_play_say.c:2160 Speaking text: &lt;break strength=&#39;medium<br>
&#39;/&gt;Hello.&lt;break strength=&#39;medium&#39;/&gt;Welcome<br>
<br>
2012-04-10 00:13:35.481120 [DEBUG] sofia.c:5561 Channel sofia/sipinterface_1/Phone2@x.241.99.20<br>
1 entering state [ready][200]<br>
<br>
2012-04-10 00:13:35.481120 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!<br>
<br>
2012-04-10 00:13:35.481120 [ERR] switch_core_io.c:1081 Codec G.729 encoder error!<br>
<br>
2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2354 done speaking text<br>
<br>
2012-04-10 00:13:35.500650 [NOTICE] switch_cpp.cpp:1227 ab127cc9-a729-4ea8-b4e0-6863f3ab243f-Inside<br>
clsAsmtApp.Run (args &#39;&#39;); HookState is CS_EXECUTE.<br>
<br>
2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS flite<br>
<br>
2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec Activated<br>
<br>
2012-04-10 00:13:35.520180 [DEBUG] switch_ivr_play_say.c:2160 Speaking text: &lt;break strength=&#39;medium<br>
&#39;/&gt;We must verify your identity.<br>
<br>
2012-04-10 00:13:35.540686 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!<br>
<br>
2012-04-10 00:13:35.540686 [ERR] switch_core_io.c:1081 Codec G.729 encoder error!<br>
<br>
2012-04-10 00:13:35.540686 [DEBUG] switch_ivr_play_say.c:2354 done speaking text<br>
<br>
2012-04-10 00:13:35.560216 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16@8000hz 1 channels<br>
20ms<br>
<br>
2012-04-10 00:13:35.560216 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!<br>
<br>
2012-04-10 00:13:35.560216 [ERR] switch_core_io.c:1081 Codec G.729 encoder error!<br>
----------------------<br>
<br>
</div></div>On Thu, Apr 5, 2012 at 1:58 PM, Avi Marcus &lt;<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</a>&lt;mailto:<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</a>&gt;&gt; wrote:<br>

Try absolute_codec_string&lt;<a href="http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string" target="_blank">http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string</a>&gt;<br>
<div>-Avi<br>
<br>
_________________________________________________________________________<br>
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</div>!DSPAM:4f83c26e32762080149657!<br>
<div><div><br>
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</div></div></blockquote></div><br></div>
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<br></blockquote></div>