Hi Sándor, > I can hear the beep of Home Assistant, and after I say something, it happily > announces, that he couldn't understand that :)
Excellent! Great to hear. BTW, if needed there are OPUS CODEC custom config settings in Asterisk by editing /etc/asterisk/codecs.conf and at the end of the file "OPUS Examples", commented-out by default. As for attribution, no personal credit is needed, but mentioning the AstLinux Project would be fine. Be sure to mention the Asterisk OPUS CODEC "Usage Tracking" as some may be uncomfortable with that. For the record, I have no experience with Home Assistant, but I found this "Support for other codecs in VOIP integration" [1] which describes your issue here. Even if Home Assistant supported ulaw/alaw in the future, having an Asterisk PBX managing the voice calls could be generally useful. The AstLinux Project provides a small footprint solution for those willing to do some CLI asterisk configuration. Lonnie [1] https://community.home-assistant.io/t/support-for-other-codecs-in-voip-integration/568580 > On Jun 28, 2023, at 11:49 PM, Sándor Balázs <baluso...@hotmail.com> wrote: > > Hi Lonnie, > > Thank you for your quick, and very accurate response! > This was everything I needed. I can hear the beep of Home Assistant, and > after I say something, it happily announces, that he couldn't understand that > :) > (this is most likely a voice recognition issue, I didn't finished configuring > that yet) > So thank you very much again! > > I'm planning to create a tutorial about how to get Home Assistant working > with older cisco phones for voice control. > Can I use your description for this purpose? If so, then do you have a > contact information you would like to be used in the attribution? > > Sándor > ________________________________________ > Feladó: Lonnie Abelbeck <li...@lonnie.abelbeck.com> > Elküldve: 2023. június 28., szerda 22:27 > Címzett: AstLinux Users Mailing List > Tárgy: Re: [Astlinux-users] Opus codec for home assitant > > Hi Sándor, > > Thanks for giving AstLinux a spin. > > First, answer your Status page question. Asterisk supports either the older > chan_sip or newer chan_pjsip SIP drivers. Asterisk 18's default config only > loads chan_pjsip and not chan_sip, so the 'sip show registry' and 'sip show > peers' CLI commands are not supported by chan_pjsip. The web interface Prefs > tab can be used to uncheck "Show SIP Trunk Registrations" and "Show SIP Peer > Status" if you decide to not use chan_sip. > > AstLinux does not include the OPUS CODEC as part of the standard build. Two > reasons... > > 1) While it may seem the OPUS CODEC is free to use [1], patent issues may > still exist. Perform proper due diligence. > > 2) Asterisk/Digium/Sangoma have built in a "Usage Tracking" feature [2]. > Both codec_opus.so and format_ogg_opus.so modules are linked with libcurl.so > to provide the "Usage Tracking" feature. > > > OK, with that out of the way, I took your challenge to add the OPUS CODEC to > AstLinux. I used Asterisk 18 as you did. Proceeded to install in a VM. > =-=-= > > == First, in order to write to /usr/lib/asterisk/modules/ you must set an > advanced configuration option. Using the web interface: > Network tab -> Advanced Configuration -> User System Variables: { Edit User > Variables } > > add the line... > > ASTERISK_RW_MODULES_DIR="yes" > > click { Save Changes } followed by clicking { Reload/Restart } [ Apply > user.conf variables ] - x Confirm > > == Using the AstLinux CLI, restart asterisk > > pbx ~ # service asterisk stop > Stopping Asterisk... > pbx ~ # service asterisk init > Starting Asterisk... > > == Now download and add the codec_opus modules from Digium. > > pbx ~ # mkdir /mnt/kd/opus > > pbx ~ # cd /mnt/kd/opus > > pbx opus # curl -O > https://downloads.digium.com/pub/telephony/codec_opus/asterisk-18.0/x86-64/codec_opus-18.0_1.3.0-x86_64.tar.gz > > pbx opus # tar xzvf codec_opus-18.0_1.3.0-x86_64.tar.gz > > pbx opus # cd codec_opus-18.0_1.3.0-x86_64 > > pbx codec_opus-18.0_1.3.0-x86_64 # cp *_opus.so /usr/lib/asterisk/modules/ > > == The codec_opus_config-en_US.xml file needs to be copied (AstLinux specific > location) > > pbx codec_opus-18.0_1.3.0-x86_64 # cp codec_opus_config-en_US.xml > /stat/var/lib/asterisk/documentation/thirdparty/ > > == As a quick sanity check, use the AstLinux "show-union" command, it should > look like... > > pbx codec_opus-18.0_1.3.0-x86_64 # show-union > /mnt/asturw/usr/lib/asterisk/modules/codec_opus.so > /mnt/asturw/usr/lib/asterisk/modules/format_ogg_opus.so > /mnt/asturw/etc/shadow- > /mnt/asturw/etc/passwd > /mnt/asturw/etc/passwd- > /mnt/asturw/etc/shadow > /mnt/asturw/stat/var/lib/asterisk/documentation/thirdparty/codec_opus_config-en_US.xml > /mnt/asturw/stat/var/www/admin/.htpasswd > > == Finally, restart asterisk to use the new modules > > pbx codec_opus-18.0_1.3.0-x86_64 # service asterisk stop > Stopping Asterisk... > pbx codec_opus-18.0_1.3.0-x86_64 # service asterisk init > Starting Asterisk... > > =-=-= > > I hope this gets you started. > > Be aware that there is some Asterisk knowledge required to perform the CODEC > translation task you desire. > > Lonnie > > > [1] https://opus-codec.org/license/ > > [2] Opus Software Codec for Asterisk README: "Usage Tracking" The codec_opus > module will periodically attempt to send usage statistics to an Asterisk > community server. The statistics are sent at most every 24 hours. > > > > >> On Jun 28, 2023, at 9:39 AM, Sándor Balázs <baluso...@hotmail.com> wrote: >> >> I have some older cisco phones with SIP and alaw/ulaw support. And I want to >> connect to home assistant. >> The direct IP call thing failed for some reason and not knowing what the >> reason might be, I turned to asterisk. I didn't want to install linux for >> this... but of course this thing is linux only... >> So I happily found this project, and got my VM working in a few minutes. >> astlinux-1.5.0 x86_64 - Asterisk 18.16.0 >> >> I want to note here, that on the Status page these messages appear instead >> of the content of the div... >> SIP Trunk Registrations: No such command 'sip show registry' (type 'core >> show help sip show' for other possible commands) >> SIP Peer Status: No such command 'sip show peers' (type 'core show help sip >> show' for other possible commands) >> >> So after some experimenting I noticed, that microsip can only communicate >> with home assistant only if the opus codec is enabled. >> so home assitant supposedly uses opus. My phones do not support opus, and >> there is no way I can get CISCO to create a firmware that does. >> >> So of course I googled it. And asterisk can translate between codecs. I >> found a maillist thread that said, that version 13 is the first version >> containing OPUS. I use version 18 so it's not a problem. Itried to enable it >> in astlinux, but after a few attempts I came to the conclusion, that I might >> miss the "codec_opus.so" in "/usr/lib/asterisk/modules/". >> Firstly it said, that it is read only... so I remounted it as read-write but >> then it complains about not enought space, as it is 100% utilised... >> >> So my question is: >> Where can I download the version that contains support for opus? >> If for some reason there isn't any, then how can I manually add it without >> recompiling the whole thing? (I do not want to use linux after all) > _______________________________________________ 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.
