[Freeswitch-users] RECORD_STEREO and record_sample_rate aren't working with conference

[Bote Man]

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. 

 

The typical profile in the Vanilla config files contains these lines:

 

  <gateways>

    <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>

  </gateways> 

 

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:

 

<gateway name="asterisk"> 

 

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.

 

Since your Asterisk SIP profile works, just copy that, set the owner and permissions appropriately, and edit the contents to point to your 2nd FreeSWITCH box IP and port number and whatever other parameters need to change.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of David P
Sent: Friday, 9 October, 2020 16:44
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] RECORD_STEREO and record_sample_rate aren't working with conference

 

Sure, I filed https://github.com/signalwire/freeswitch/issues/895

 

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.

 

In the 1st FS we need to change the sofia gateway path from this:

 

<action application="conference_set_auto_outcall"
        data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p']sofia/gateway/asterisk/${destination_number}"/>

 

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.

 

Is that right?

 

From: Brian West <brian at freeswitch.com <mailto:brian at freeswitch.com> >

Yah that's probably true, you should file issues on github please.

 

On Thu, Oct 8, 2020 at 5:59 PM David P <davidswalkabout at gmail.com <mailto:davidswalkabout at gmail.com> > wrote:

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:

 

<action application="set" data="RECORD_STEREO=true"/>
<action application="set" data="record_sample_rate=44100"/>

Instead:

1) The mp4s have the same audio in both channels, contrary to https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO

 

2) The sampling rate is 8kHz, contrary to https://freeswitch.org/confluence/display/FREESWITCH/record_sample_rate

FWIW:

a) We noticed that mp4s from a year ago when we were using FS 1.8 also have the same audio on both channels.

 

b) We also have this, and I don't recall why and can't find a description in confluence:
<action application="set" data="record_concat_video=true"/>

 

Is it that these are known not to work with conferences, or is there some setting we might have overlooked?

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