Hi Michael,

Per this example [1] it seems "Record(en/custom-menu.g722)" would record what 
you want, but "slin" format might be more efficient transcoding to other 
formats.

Lonnie

[1] 
https://docs.asterisk.org/Deployment/Basic-PBX-Functionality/Auto-attendant-and-IVR-Menus/Record-Application/



> On Jun 15, 2025, at 7:11 PM, Michael Knill 
> <michael.kn...@ipcsolutions.com.au> wrote:
> 
> PS just thought I would reply to this post.
>  It appears that the Asterisk Record application does not natively record at 
> G.722 (see below):
>  CLI> core show channel SIP/1410-0000029b
> -- General --
>            Name: SIP/1410-0000029b
>            Type: SIP
>        UniqueID: 39990-IPCBuild-CM1-1749719640.3701
>        LinkedID: 39990-IPCBuild-CM1-1749719640.3701
>       Caller ID: 1410
> Caller ID Name: (N/A)
> Connected Line ID: *480
> Connected Line ID Name: AH Announcement
> Eff. Connected Line ID: *480
> Eff. Connected Line ID Name: AH Announcement
>     DNID Digits: *7480
>        Language: en_AI
>           State: Up (6)
>   NativeFormats: (g722)
>     WriteFormat: slin
>      ReadFormat: slin
> WriteTranscode: Yes (slin@8000)->(g722@16000)
>   ReadTranscode: Yes (g722@16000)->(slin@8000)
> Time to Hangup: 0
>    Elapsed Time: 0h0m8s
>       Bridge ID: (Not bridged)
> --   PBX   --
>         Context: subFeature-Recordannounce
>       Extension: start
>        Priority: 3
>      Call Group: 0
>    Pickup Group: 0
>     Application: Record
>            Data: /mnt/kd/monitor/announce*480:g722,30,120,k
> Call Identifer: [C-000002b2]
>       Variables:
> RECORDED_FILE=/mnt/kd/monitor/announce*480
> GOSUB_RETVAL=
> conname_dblookup=AH Announcement
> DB_RESULT=AH Announcement
> name_dblookup=
> ARGC=1
> ARG1=*480
> SIPCALLID=0_3708201006@172.30.253.14
> SIPDOMAIN=172.30.253.1
> SIPURI=sip:1410@172.30.253.14:5060
> -- Streams --
> Name: audio-0
>     Type: audio
>     State: sendrecv
>     Group: -1
>     Formats: (g722)
>     Metadata:
>  Rather annoying that I need to record everything in slin!
>  Regards
> Michael Knill
>   From: Michael Knill <michael.kn...@ipcsolutions.com.au>
> Date: Sunday, 8 June 2025 at 10:14 am
> To: AstLinux Users Mailing List <astlinux-users@lists.sourceforge.net>
> Subject: Re: [Astlinux-users] Use of HD codecs for the Asterisk Record 
> application
> Hi All
>  Thanks for the reply. No problems with DTMF and current Asterisk sounds. 
> They are all working fine. PS I am setting SIP_CODEC=alaw when dialling 
> externally so there is no transcoding
>  The problem is when I am recording messages for announcements etc. They are 
> written as {filename}.g722 and can be listened to fine as G.722 but just 
> don’t sound as good as I would have expected for a G.722 recording.
> After doing further tests however, it does actually sound better than when 
> recording in wav format but just not as good as the inbuilt prompts in G.722 
> format.
>  A bit of background for this, our next release will include integration into 
> Azure’s Speech API which is pretty fantastic now. Im going to create my 
> custom messages all now with TTS and I will likely rerecord all the inbuilt 
> prompts as well. Incredible flexibility. Will also be doing voicemail 
> transcription which is working great.
>  Im extremely thankful for the amazing toolbox we have.
>  Regards
> Michael Knill
>   From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>
> Date: Sunday, 8 June 2025 at 12:36 am
> To: AstLinux Users Mailing List <astlinux-users@lists.sourceforge.net>
> Subject: Re: [Astlinux-users] Use of HD codecs for the Asterisk Record 
> application
> For my internal phones, for many years...
> --
> dtmfmode=rfc2833
> disallow=all
> allow=g722
> allow=ulaw
> --
> 
> $ upgrade-asterisk-sounds show
> Installed: asterisk-moh-opsound-g722-2.03, asterisk-moh-opsound-ulaw-2.03, 
> asterisk-core-sounds-en-g722-1.6.1, asterisk-core-sounds-en-ulaw-1.6.1, 
> asterisk-extra-sounds-en-g722-1.5.2, asterisk-extra-sounds-en-ulaw-1.5.2
> 
> 
> Lonnie
> 
> 
> > On Jun 7, 2025, at 8:55 AM, Michael Keuter <li...@mksolutions.info> wrote:
> > 
> > Just a short info: 
> > I had sometimes problems with DTMF tones when using G.722. You might need 
> > to tweak the DTMF type.
> > 
> >> Am 07.06.2025 um 08:52 schrieb Michael Knill 
> >> <michael.kn...@ipcsolutions.com.au>:
> >> 
> >> Hi Group
> >> Not sure if someone has come across this before. I have decided to use 
> >> G.722 for internal calls and alaw for external calls and all seems fine so 
> >> far. All the internal sounds are clear and crisp at G.722
> >> The issue that I have come across is using the Record application which I 
> >> want to record native G.722. It appears to be working fine e.g. it stores 
> >> the file as {filename}.g722 and it plays fine however it certainly does 
> >> not sound like a G.722 recorded call.
> >> Could this be because there is no format_g722.so included?
> >> Do I have any other options?
> >> Im getting to the stage where I don’t really care!
> >> Regards
> >> Michael Knill
> >> Managing Director
> >> D: +61 2 6189 1360
> >> P: +61 2 6140 4656
> >> E: michael.kn...@ipcsolutions.com.au
> >> W: ipcsolutions.com.au
> >>  <image001.png>Smarter Business Communications
> > 
> > 
> > Michael
> > 
> > https://mksolutions.info
> > 
> > 
> > 
> > 
> > 
> > 
> > _______________________________________________
> > Astlinux-users mailing list
> > Astlinux-users@lists.sourceforge.net
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> > 
> > Donations to support AstLinux are graciously accepted via PayPal to 
> > pay...@krisk.org.
> 
> 
> 
> 
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