<div dir="ltr">Solved on IRC - mod_spandsp was failing to load because of a build error where some libraries were missing. GSM wasn't available as a result, so FS couldn't initiate media once the call rang / was answered.<div>
<br></div><div><a href="http://xkcd.com/979/">http://xkcd.com/979/</a><br>
<div class="gmail_extra"><br><br><div class="gmail_quote">On 17 October 2013 16:22, Aseem Anand <span dir="ltr"><<a href="mailto:anandaseem89@gmail.com" target="_blank">anandaseem89@gmail.com</a>></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, "Danny Iland" <<a href="mailto:iland@cs.ucsb.edu" target="_blank">iland@cs.ucsb.edu</a>> 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'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 'audio codec compare' 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 -> 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="32776"||| </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>
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