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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>You create as many SIP profiles or gateways as you need. You can name the files anything you wish because the important bits are contained inside the files, typically near the top. <o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>The typical profile in the Vanilla config files contains these lines:<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> <span style='font-family:Consolas'><gateways><o:p></o:p></span></p><p class=MsoNormal><span style='font-family:Consolas'> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/><o:p></o:p></span></p><p class=MsoNormal><span style='font-family:Consolas'> </gateways></span> <o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>which tells FS to scan the “internal” subdirectory for all XML files. Inside one of those would be your Asterisk file that contains a line like:<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> <span style='font-family:Consolas'><gateway name="asterisk"> <o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>which appears in your current dial string. That string in the gateway name= is what tells FS which gateway to use when you specify it in the dial string.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Since your Asterisk SIP profile works, just copy that, set the owner and permissions appropriately, and edit the contents to point to your 2<sup>nd</sup> FreeSWITCH box IP and port number and whatever other parameters need to change.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='font-size:10.0pt;color:#7F7F7F'> --- <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;color:#7F7F7F'>John Boteler <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;color:#7F7F7F'>BnC Group U.S.A. <o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><b><o:p> </o:p></b></p><p class=MsoNormal><b><o:p> </o:p></b></p><p class=MsoNormal><b>From:</b> FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> <b>On Behalf Of </b>David P<br><b>Sent:</b> Friday, 9 October, 2020 16:44<br><b>To:</b> FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br><b>Subject:</b> [Freeswitch-users] RECORD_STEREO and record_sample_rate aren't working with conference<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>Sure, I filed <a href="https://github.com/signalwire/freeswitch/issues/895">https://github.com/signalwire/freeswitch/issues/895</a><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>In the meantime, we're urgently looking for a workaround, and we're considering introducing a 2nd FS between our current FS and the peer destination (Asterisk). We'd remove transcoding from the conference config of the 1st FS.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal>In the 1st FS we need to change the sofia gateway path from this:<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><action application="conference_set_auto_outcall"<br> data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p']sofia/gateway/asterisk/${destination_number}"/><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>and I'm guessing the path should be " sofia/gateway/freeswitch/" (with a corresponding sip_profiles/internal/freeswitch.xml) But googling "sofia gateway freeswitch" doesn't reveal any promising matches.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Is that right?<o:p></o:p></p><div><div><p class=MsoNormal><o:p> </o:p></p></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><p class=MsoNormal style='margin-bottom:12.0pt'>From: Brian West <<a href="mailto:brian@freeswitch.com" target="_blank">brian@freeswitch.com</a>><o:p></o:p></p><div><p class=MsoNormal>Yah that's probably true, you should file issues on github please.<o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>On Thu, Oct 8, 2020 at 5:59 PM David P <<a href="mailto:davidswalkabout@gmail.com" target="_blank">davidswalkabout@gmail.com</a>> wrote:<o:p></o:p></p></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><p class=MsoNormal>We're using FSv10.5 on Debian 10 with verto/Opus (i.e. no codec explicitly configured), and we're using a conference. We've tried these settings but they're having no effect on the recorded mp4s:<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><action application="set" data="RECORD_STEREO=true"/><br><action application="set" data="record_sample_rate=44100"/><br><br>Instead:<o:p></o:p></p></div><div><p class=MsoNormal>1) The mp4s have the same audio in both channels, contrary to <a href="https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO</a><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>2) The sampling rate is 8kHz, contrary to <a href="https://freeswitch.org/confluence/display/FREESWITCH/record_sample_rate" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/record_sample_rate</a><br><br>FWIW:<o:p></o:p></p></div><div><p class=MsoNormal>a) We noticed that mp4s from a year ago when we were using FS 1.8 also have the same audio on both channels.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>b) We also have this, and I don't recall why and can't find a description in confluence:<br><action application="set" data="record_concat_video=true"/><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Is it that these are known not to work with conferences, or is there some setting we might have overlooked?<o:p></o:p></p></div></div></blockquote></div></blockquote></div></div></div></div></body></html>