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.
