<div dir="ltr">:)</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Oct 17, 2013 at 10:15 PM, Steven Ayre <span dir="ltr">&lt;<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Solved on IRC - mod_spandsp was failing to load because of a build error where some libraries were missing. GSM wasn&#39;t available as a result, so FS couldn&#39;t initiate media once the call rang / was answered.<div>



<br></div><div><a href="http://xkcd.com/979/" target="_blank">http://xkcd.com/979/</a><div><div class="h5"><br> 
<div class="gmail_extra"><br><br><div class="gmail_quote">On 17 October 2013 16:22, Aseem Anand <span dir="ltr">&lt;<a href="mailto:anandaseem89@gmail.com" target="_blank">anandaseem89@gmail.com</a>&gt;</span> wrote:<br>



<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div class="gmail_quote"><div><div>On Oct 16, 2013 11:47 PM, &quot;Danny Iland&quot; &lt;<a href="mailto:iland@cs.ucsb.edu" target="_blank">iland@cs.ucsb.edu</a>&gt; wrote:<br type="attribution"></div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">




<div><div>
<div dir="ltr"><div><div>I&#39;m using Freeswitch as a PBX for OpenBTS, to route calls from mobile phones to VoIP providers.</div><div><br></div><div>When bridging a call from a cellular handset to a mod_skypopen interface, I get the following error as soon as the Skype call is answered. The handset then hangs up. It appears the problem lies in the &#39;audio codec compare&#39; phase.</div>






<div><br></div><div>2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal sofia/internal/IMSI[me]@[ip] [BREAK]</div><div>2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype101/echo123] has been answered</div>






<div>2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 (skypopen/skype101/echo123) Callstate Change RINGING -&gt; ACTIVE</div><div>2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232         [543dc3c|22dd4bf] [DEBUG_SKYPE  1232 ][skype101       ][UP,INPROGRS] MSG_ID=41</div>






<div>2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479         [543dc3c|22dd4bf] [DEBUG_SKYPE  2479 ][skype101       ][UP,INPROGRS] outbound_channel_answered!</div><div>2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172    [543dc3c|22dd4bf] [DEBUG_SKYPE  1172 ][skype101       ][UP,INPROGRS] ACCEPTED here you send me 32776</div>






<div>2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177    [543dc3c|22dd4bf] [DEBUG_SKYPE  1177 ][skype101       ][UP,INPROGRS] 4 SO_RCVBUF is 5120, size is 4</div><div>2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181    [543dc3c|22dd4bf] [DEBUG_SKYPE  1181 ][skype101       ][UP,INPROGRS] 4 SO_SNDBUF is 5120, size is 4</div>






<div>2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207     [543dc3c|22dd4bf] [DEBUG_SKYPE  207  ][skype101       ][UP,INPROGRS] READING: |||ALTER CALL 32 SET_INPUT PORT=&quot;32776&quot;||| </div><div>2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000]</div>






<div>2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000]</div><div>2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO</div>






<div>2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/IMSI[me]@[ip] [BREAK]</div><div>2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]</div>






</div><div><br></div><div>Any advice on forcing a transcode or otherwise having the Audio Codec Compare result in a successful bridge is appreciated.</div><div><br></div><div>Thanks,</div><div>Danny Iland</div><div><br></div>






<div><br></div><div><br></div></div>
<br></div></div><div>_________________________________________________________________________<br>
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<br></div></blockquote></div>
<br>_________________________________________________________________________<br>
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<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
<a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
<br>
Official FreeSWITCH Sites<br>
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<br></blockquote></div><br></div></div></div></div></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
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<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
<a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Sincerely,<br><br>Giovanni Maruzzelli<br>Cell : +39-347-2665618<br>
</div>