search for: codec_opus

Before I got an log a ticket, can I just check I'm not doing anything wrong? In 15.2, to install Opus: 1) run `make menuselect` 2) Highlight "Codec Translators" and press enter. 3) Scroll down to "codec_opus" in the section labeled "External" 4) Press enter to select the codec if it is not already selected. ... at this point, I see XXX codec_opus and a message to download codec opus from Digium, using this link: http://downloads.digium.com/pub/telephony/codec_opus/README (note: same...

...sing to enable the option. > In this case, it's certainly curl that is missing on your system. Ah, OK! No, it wasn't curl that was missing, but I think the way it was phrased confused me. It says: "Download the Opus codec from Digium. See http://downloads.digium.com/pub/telephony/codec_opus/README. Depends on: xmlstarlet(E), bash(E), res_format_attr_opus(M) Support Level: external" However, while I managed to apt install xmlstarlet, I already had bash, and I can't find res_format_attr_opus. Any ideas? Many thanks, Jonathan > Regards. > > -- > Ludovic Gasc (GML...

I tried to download Digium’s Opus Codec via the following link, but the server is unavailable: http://downloads.digium.com/pub/telephony/codec_opus/ It took me a while to figure this out, because initially I tried downloading via selecting the Opus codec in make menuselect and realizing that it isn’t there after make install step. Can someone from Digium/Sangoma please confirm? FLORIAN FLOIMAIR Software Development - IMS Commend Internat...

...Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11) * ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett) * ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard...

...provements made in this release: ----------------------------------- * ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan) * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by...

On 13.9.0 https://github.com/traud/asterisk-opus The README: Alternatively, you can use the Makefile of this repository to create just the shared libraries of the modules. That way, you do not have to (re-) make your whole Asterisk. The Makefile generates: codecs/codec_opus_open_source.so formats/format_ogg_opus_open_source.so formats/format_vp8.so res/res_format_attr_opus.so Without any of the patches the asterisk build generates: codec_opus.so format_ogg_opus.so res_format_attr_opus.so Questions: Should the *_opus_open_source.so be in modules with the *opus.so...

..._musl/usr/lib -L/home/build/build_dir/staging_dir/toolchain-mips_24kc_gcc-8.4.0_musl/lib -znow -zrelro -o libshout.la -rpath /usr/lib shout.lo util.lo queue.lo connection.lo proto_http.lo proto_xaudiocast.lo proto_icy.lo proto_roaraudio.lo format_ogg.lo format_webm.lo format_mp3.lo codec_vorbis.lo codec_opus.lo tls.lo common/net/libicenet.la common/timing/libicetiming.la common/avl/libiceavl.la common/httpp/libicehttpp.la common/thread/libicethread.la -L/home/build/build_dir/staging_dir/target-mips_24kc_musl/usr/lib -L/home/build/build_dir/staging_dir/target-mips_24kc_musl/lib -L/home/build/build_dir...

Hello, I am trying to configure new opus codec in asterisk 14, but unable to find any examples of codecs.conf settings for this codec. All I am trying to do - setup peer with using opus in narrow band mode (8kHz sampling rate). Does anybody know how to configure chan_opus? -- Regards, Igor Goncharovsky Unistim Dev: http://unistim.igorg.ru -------------- next part -------------- An HTML

...support to Asterisk TLS support (Reported by Maciej Szmigiero) * ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan) * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by...

...aging_dir/toolchain-mips_24kc_gcc-8.4.0_musl/lib > > -znow -zrelro -o libshout.la -rpath /usr/lib shout.lo util.lo > > queue.lo connection.lo proto_http.lo proto_xaudiocast.lo proto_icy.lo > > proto_roaraudio.lo format_ogg.lo format_webm.lo format_mp3.lo > > codec_vorbis.lo codec_opus.lo tls.lo common/net/libicenet.la > > common/timing/libicetiming.la common/avl/libiceavl.la > > common/httpp/libicehttpp.la common/thread/libicethread.la > > -L/home/build/build_dir/staging_dir/target-mips_24kc_musl/usr/lib > > -L/home/build/build_dir/staging_dir/target-mi...

Hey! I just upgraded our machines from Ubuntu focal to jammy. A separate package asterisk-opus does not exist any more. https://launchpad.net/ubuntu/+source/asterisk-opus/+changelog It looks like this is now included in the default packages. Required modules are loaded: *CLI> module show like opus Module Description Use Count Status

...t 7659 (offset -98 lines). Hunk #2 succeeded at 11011 (offset -34 lines). Hunk #3 succeeded at 11050 (offset -34 lines). Hunk #4 succeeded at 11117 with fuzz 1 (offset -34 lines). Hunk #5 FAILED at 12722. 1 out of 6 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej patching file codecs/codec_opus.c patching file codecs/ex_opus.h patching file configure.ac Hunk #2 succeeded at 2150 (offset 31 lines). patching file formats/format_vp8.c patching file include/asterisk/format.h patching file main/channel.c patching file main/format.c Hunk #6 succeeded at 1098 (offset 12 lines). patching file mai...

...erisk The release of Asterisk 14.0.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Improvements made in this release: ----------------------------------- * ASTERISK-26409 - codec_opus: Update Asterisk to support the translation codec. (Reported by Kevin Harwell) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.1 Thank you for your continued support of Asterisk!

<div>Hello,</div><div> </div><div>I'm looking to implement an internet radio server that would serve up something similar to <a href="http://dir.xiph.org/by_format/Opus">http://dir.xiph.org/by_format/Opus</a> using the C libraries for working with opus.  I tried looking at the Icecast source code, with no luck - it doesn't seem to depend

...call is not answered after a queue (Reported by Sebastian Gutierrez) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by Sebastian Gutierrez) * ASTERISK-26217 - [patch] Codec 2 Mode 2400 (Reported by Alexander Traud) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan) * ASTERISK-26418 - res_rtp_asterisk: Speed up...

...ue: add variable to know if the call is not answered after a queue (Reported by scgm11) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) * ASTERISK-26217 - [patch] Codec 2 Mode 2400 (Reported by Alexander Traud) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan) * ASTERISK-26418 - res_rtp_asterisk: Speed up...

...e/build/build_dir/staging_dir/toolchain-mips_24kc_gcc-8.4.0_musl/lib > -znow -zrelro -o libshout.la -rpath /usr/lib shout.lo util.lo > queue.lo connection.lo proto_http.lo proto_xaudiocast.lo proto_icy.lo > proto_roaraudio.lo format_ogg.lo format_webm.lo format_mp3.lo > codec_vorbis.lo codec_opus.lo tls.lo common/net/libicenet.la > common/timing/libicetiming.la common/avl/libiceavl.la > common/httpp/libicehttpp.la common/thread/libicethread.la > -L/home/build/build_dir/staging_dir/target-mips_24kc_musl/usr/lib > -L/home/build/build_dir/staging_dir/target-mips_24kc_musl/lib &gt...

...(Reported by George Joseph) * ASTERISK-28238 - PJSIP realtime. getcontext not working with DUNDI (Reported by Ray) * ASTERISK-28173 - Deadlock in chan_sip handling subscribe request during res_parking reload (Reported by Giuseppe Sucameli) * ASTERISK-28263 - codec_opus: errors setting max_playback_rate and bitrate to "sdp" (Reported by Gianluca Merlo) * ASTERISK-28250 - build: Cross-compilation fails for target arm-linux-gnueabihf (Reported by Jean Aunis - Prescom) * ASTERISK-28156 - Race condition involving session->media...

...context for stasis apps (Reported by George Joseph) * ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will not compile (Reported by David Wilcox) * ASTERISK-28238 - PJSIP realtime. getcontext not working with DUNDI (Reported by Ray) * ASTERISK-28263 - codec_opus: errors setting max_playback_rate and bitrate to "sdp" (Reported by Gianluca Merlo) * ASTERISK-28250 - build: Cross-compilation fails for target arm-linux-gnueabihf (Reported by Jean Aunis - Prescom) * ASTERISK-28257 - res_http_websocket: PING / PONG opcodes...

...ages (Reported by Jonathan Rose) * ASTERISK-26653 - pjproject_bundled doesn't verify already downloaded tarballs (Reported by George Joseph) * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed, setting up new calls (Reported by Walter Doekes) * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp line (Reported by J??rgen H) * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all aors (Reported by George Joseph) * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains &quot...