<div dir="ltr">Does freeswitch generate his own ringback if I dont execute ringback?<br>I just straight bridge the call after it hits dialplan.<br><br>What can to fix this behavior? Switch to g711, or passthrough g729?</div><div class="gmail_extra"><br><div class="gmail_quote">2016-07-13 16:19 GMT+03:00 Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Its probably not even what you think it is, Its probably trying to generate ringback and failing.</div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">On Wed, Jul 13, 2016 at 7:35 AM, Vladislav Ivanov <span dir="ltr"><<a href="mailto:deforceczt@gmail.com" target="_blank">deforceczt@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div dir="ltr">Hello,<br><br>I have an issue with codec negotiation, seems to me that freeswitch is trying to transcode g729 to g729 annexb=no for some reason, even when I have different codecs available?<br><br>I'm bridging 2 calls, with profile settings:<br><div> <param name="inbound-late-negotiation" value="true"/></div><div> <param name="disable-transcoding" value="true"/></div><div>and dialplan:</div><div> <action application="set" data="inherit_codec=true"/></div><div><br></div><div>with global codec:<br><div> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729,G723"/></div><div><br></div>here is invite and 183 progress message:<br><br><div>Session Initiation Protocol (INVITE)</div><div> Request-Line: INVITE sip: SIP/2.0</div><div> Message Body</div><div> Session Description Protocol</div><div> Media Description, name and address (m): audio 20268 RTP/AVP 18 8 0 101 13</div><div> Media Attribute (a): rtpmap:18 G729/8000</div><div> Media Attribute (a): rtpmap:8 PCMA/8000</div><div> Media Attribute (a): rtpmap:0 PCMU/8000</div><div> Media Attribute (a): rtpmap:101 telephone-event/8000</div><div> Media Attribute (a): fmtp:101 0-16</div><div> Media Attribute (a): rtpmap:13 CN/8000</div><div> Media Attribute (a): ptime:20</div><div> </div><div> </div><div>Session Initiation Protocol (183)</div><div> Status-Line: SIP/2.0 183 Session Progress</div><div> Message Body</div><div> Session Description Protocol</div><div> Media Description, name and address (m): audio 59852 RTP/AVP 18 0 8 101</div><div> Media Attribute (a): rtpmap:18 G729/8000</div><div> Media Attribute (a): fmtp:18 annexb=no</div><div> Media Attribute (a): rtpmap:0 PCMU/8000</div><div> Media Attribute (a): rtpmap:8 PCMA/8000</div><div> Media Attribute (a): rtpmap:101 telephone-event/8000</div><div> Media Attribute (a): fmtp:101 0-15</div><div> Media Attribute (a): sendrecv</div><div> Media Attribute (a): maxptime:20</div><div> </div><div>After that I get error message:</div><div><br></div><div>2016-07-13 11:29:11.136011 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!</div><div>2016-07-13 11:29:11.136011 [ERR] switch_core_io.c:1434 Codec G.729 encoder error!</div></div><div><br></div><div>and freeswitch sends CANCEL to B-side leg.</div><div><br></div><div>Maybe I have missed my settings, but if codecs "mismatch" shouldn't freeswitch choose g711 a/u and not try to "transcode" the codec?</div><div><br></div><div>Best Regards,<br>Vladislav</div></div>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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