<div>Thanks a lot Anthony !!!</div>
<div>&nbsp;</div>
<div>It helped when I loaded mod_native_file before mod_voipcodecs.</div>
<div>&nbsp;</div>
<div>Many thanks again,</div>
<div>&nbsp;</div>
<div>Miroslav</div>
<div><br><br>&nbsp;</div>
<div class="gmail_quote">On Fri, Jun 6, 2008 at 3:53 PM, Anthony Minessale &lt;<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>&gt; wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">try reversing the order that you load mod_voipcodecs and mod_sndfile in your modules.conf.xml<br>iirc there is some symbol collision between the 2 that we need to resolve somehow. 
<div>
<div></div>
<div class="Wj3C7c"><br><br>
<div class="gmail_quote">On Fri, Jun 6, 2008 at 1:29 PM, Miroslav Mostic &lt;<a href="mailto:nuvovoip@gmail.com" target="_blank">nuvovoip@gmail.com</a>&gt; wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<div>It is not&nbsp;inbound codec issue.&nbsp;When&nbsp;I enable only GSM codec&nbsp;in my x-lite I can see&nbsp;that GSM is&nbsp;negotiated inbound codec.&nbsp;Even in that case&nbsp;recording in gsm&nbsp;is&nbsp;not working. However&nbsp;recording in uncompressed PCM wav format&nbsp;is working properly. </div>

<div>I have the&nbsp;same&nbsp;behaviour with&nbsp;rc3 and some snapshot release after that, but for the sake of this test I have&nbsp;default Freeswitch 1.0 configuration on the separate box.</div>
<div>&nbsp;</div>
<div>Could it be that compressed gsm encoded recordings are&nbsp;not supported anymore?&nbsp;</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div><font color="#888888">
<div>Miroslav&nbsp;&nbsp;<br><br></div></font>
<div>
<div></div>
<div>
<div class="gmail_quote">On Tue, Jun 3, 2008 at 1:49 PM, Miroslav Mostic &lt;<a href="mailto:nuvovoip@gmail.com" target="_blank">nuvovoip@gmail.com</a>&gt; wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<div>&nbsp;</div>
<div>G711 u-Law. I&nbsp;was&nbsp;calling from&nbsp;aastra&nbsp;9133i and X-lite v3.0 and I have default codec configuration of Freeswitch 1.0. After installation of Freeswitch 1.0 the only thing that are&nbsp;changed&nbsp;from default config are sip port 5090 on internal sip profile and additional&nbsp;extensions 9991 and 9992 in default.xml dialplan.&nbsp;In both scenarios I have the same result. </div>

<div>&nbsp;</div>
<div>Console output when I dial from&nbsp;aastra 9133i :</div>
<div>&nbsp;</div>
<div>2008-06-03 12:52:18 [DEBUG] sofia.c:1695 sofia_handle_sip_i_state() Remote SDP:<br>v=0<br>o=MxSIP 0 1966242209 IN IP4 <a href="http://172.22.2.102/" target="_blank">172.22.2.102</a><br>s=SIP Call<br>c=IN IP4 <a href="http://172.22.2.102/" target="_blank">172.22.2.102</a><br>
t=0 0<br>m=audio 3000 RTP/AVP 0 8 18 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=ptime:30<br>a=silenceSupp:on - - - -</div>
<div>2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[G722:9:8000]<br>2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000]<br>
2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMA:8:8000]<br>2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[GSM:3:8000]<br>
2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2137 sofia_glue_negotiate_sdp() Substituting codec <a href="mailto:PCMU@30i@8000h" target="_blank">PCMU@30i@8000h</a><br>2008-06-03 12:52:18 [DEBUG] sofia_glue.c:1392 sofia_glue_tech_set_codec() Set Codec <a href="mailto:sofia/internal/1006@10.47.40.56:5090" target="_blank">sofia/internal/1006@10.47.40.56:5090</a> PCMU/8000 30 ms 240 samples<br>
</div>
<div>Console output when I dial from&nbsp;X-Lite Version 3.0:</div>
<div>&nbsp;</div>
<div>2008-06-03 12:57:03 [DEBUG] sofia.c:1695 sofia_handle_sip_i_state() Remote SDP:<br>v=0<br>o=- 4 2 IN IP4 <a href="http://172.22.2.141/" target="_blank">172.22.2.141</a><br>s=CounterPath X-Lite 3.0<br>c=IN IP4 <a href="http://172.22.2.141/" target="_blank">172.22.2.141</a><br>
t=0 0<br>m=audio 51672 RTP/AVP 0 8 3 101<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=alt:1 1 : uY3hci3U 9llPveYf <a href="http://172.22.2.141/" target="_blank">172.22.2.141</a> 51672</div>
<div>2008-06-03 12:57:03 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() <a href="mailto:sofia/internal/1004@10.47.40.56:5090" target="_blank">sofia/internal/1004@10.47.40.56:5090</a> Running State Change CS_NEW<br>
2008-06-03 12:57:03 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[G722:9:8000]<br>2008-06-03 12:57:03 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000]<br>
2008-06-03 12:57:03 [DEBUG] sofia_glue.c:1392 sofia_glue_tech_set_codec() Set Codec <a href="mailto:sofia/internal/1004@10.47.40.56:5090" target="_blank">sofia/internal/1004@10.47.40.56:5090</a> PCMU/8000 20 ms 160 samples</div>

<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div><font color="#888888">
<div>Miroslav<br></div></font>
<div>
<div></div>
<div>
<div class="gmail_quote">On Mon, Jun 2, 2008 at 8:52 PM, Brian West &lt;<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>&gt; wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<div><font face="Calibri, Verdana, Helvetica, Arial"><span style="FONT-SIZE: 11pt">What codec are you using on the inbound side?<br><br>/b 
<div>
<div></div>
<div><br><br><br><br>On 6/2/08 4:27 PM, &quot;Miroslav Mostic&quot; &lt;<a href="http://nuvovoip@gmail.com" target="_blank">nuvovoip@gmail.com</a>&gt; wrote:<br><br></div></div></span></font>
<blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span style="FONT-SIZE: 11pt">
<div>
<div></div>
<div>Hi everyone!<br>&nbsp;<br>I am probably missing something very basic, but I have problems with simple file record and playback in gsm file format.<br>&nbsp;<br>I have just installed 1.0 release system and I changed only default dialplan with following two extensions:<br>
&nbsp;<br>&lt;extension name=&quot;gsm_record&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;condition field=&quot;destination_number&quot; expression=&quot;^9991$&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;answer&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;record&quot; data=&quot;/tmp/rec.gsm&quot;/&gt;<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;/condition&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&lt;/extension&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&lt;extension name=&quot;gsm_playback&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;condition field=&quot;destination_number&quot; expression=&quot;^9992$&quot;&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;answer&quot;/&gt;<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;action application=&quot;playback&quot; data=&quot;/tmp/rec.gsm&quot;/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;/condition&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&lt;/extension&gt;<br>&nbsp;<br>When I place call to 9991extension I see INFO message about opening /tmp/rec.gsm, no error messages and when I hangup I can see that file /tmp/rec.gsm exists. However when I try to play this file using 9992 extension the only thing I can hear is noise. <br>
&nbsp;<br>When I just change name to /tmp/rec.wav in both extensions everything is working fine. <br>&nbsp;<br>What am I missing?<br>&nbsp;<br>Many regards,<br>&nbsp;<br>Miroslav<br><br></div></div>
<hr align="center" width="95%" size="3">
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<br></blockquote></div><br><br clear="all"><br></div></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
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