Yes I believe you would be far better recording in wav/alaw/ulaw e.g. @8K natively and if you could be bothered, use Asterisk ‘file convert’ to g722 if you don’t want to transcode when listening to the announcement internally. I use Set(__SIP_CODEC=alaw) to force this in the dialplan.
If you want HD announcements e.g. if building Public Address applications which sound rubbish at 8K, then you basically need to create an audio file separately and use sox to convert to wav16 and then ‘file convert’ if you want any other formats. This has been an interesting learning experience for me. Regards Michael Knill From: Lonnie Abelbeck <li...@lonnie.abelbeck.com> Date: Tuesday, 17 June 2025 at 12:20 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 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. > > > > > _______________________________________________ > 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. > _______________________________________________ > 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. _______________________________________________ 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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