similar to: opus from git : install questions

On 27 January 2018 at 09:27, Ludovic Gasc <gmludo at gmail.com> wrote: > Hi Jonathan, > > If you put the cursor on the line XXX, you will see what are the > dependencies are missing 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

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

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

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

So as y'all know, with your help I managed to get Opus installed at last. Yay! With excitement, I wrote my dialplan, dialled in, and.... [Jan 28 21:30:11] ERROR[29977][C-0000001d]: format_ogg_opus.c:95 ogg_opus_rewrite: Cannot write OGG/Opus streams. Sorry :( [Jan 28 21:30:11] WARNING[29977][C-0000001d]: file.c:468 fn_wrapper: Unable to rewrite format ogg_opus Any idea where I'm going

Hi all, in witch way can I put in a log file the asterisk console? I have tried with some settings in file logger.conf but the log not contain the same debug that I can see with "asterisk -rvvv". I need it in debugging purpose for tracking some bug. Thanks Enrico. -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:

1.6.2: -- Executing [s at incoming-pstn-line:4] VoiceMail("DAHDI/4-1", "100 at default,u") in new stack -- <DAHDI/4-1> Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]:

I've configured asterisk 1.4 to use imap storage for voice-mail and while I'm happy with it generally speaking it really seem to hammer the IMAP server. It appear, from the IMAP server logs that it's polling the imap server every *second* for mailbox updates for the users' voice-mail folders. Is it really necessary to do this once a second? Is this tunable anywhere? Thanx, b.

Hi, I have just installed the fresh svn version of asterisk and when I run it I get the following errors: [Mar 4 14:19:27] WARNING[24527]: loader.c:728 load_modules: No 'modules.conf' found, no modules will be loaded. [Mar 4 14:19:27] NOTICE[24527]: manager.c:2681 init_manager: Unable to open management configuration manager.conf. Call management disabled. [Mar 4 14:19:27] NOTICE[24527]:

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

The Asterisk Development Team has announced the release of Asterisk 14.0.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.0.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

I am trying to launch Asterisk on a different directory with the parameter 'C asterisk -vvvvvvvvvvvvvvvvvvgc -C /etc/asterisk1/asterisk.conf Parsing '/etc/asterisk1/extconfig.conf': Found Resetting translation matrix UUID system initiated Parsing /etc/asterisk1/asterisk.conf == Parsing '/etc/asterisk1/asterisk.conf': Found Not changing threadpool size since new size 0 is

On Mon, Oct 5, 2020 at 9:59 PM Marvin Scholz <epirat07 at gmail.com> wrote: > > Hi, CC'd the mailing list. > > It seems you are not up to date, git master already has a > --without-openssl option. Without this patch, I get /bin/bash ../libtool --tag=CC --mode=link mips-openwrt-linux-musl-gcc -Wall -ffast-math -fsigned-char

hi, any plans update patch for 11.12.0? |https://github.com/meetecho/asterisk-opus https://github.com/netaskd/asterisk-opus/ | patching file build_tools/menuselect-deps.in patching file channels/chan_sip.c Hunk #1 succeeded at 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

The Asterisk Development Team would like to announce the release of Asterisk 16.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hello, I am trying to get a repeatable build setup for asterisk. Part of doing so involves using the -C option to specify the master config file. The problem is that asterisk reads the config file location that i specify, however it still tries to read two other config files, namely: * /etc/asterisk/extconfig.conf, and * /etc/asterisk/logger.conf I have specified in my config file that the

<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

Hello On a uClinux-based appliance, "ps aux" shows multiple Asterisk processes: 380 root 11990 S asterisk -f 381 root 11990 S asterisk -f 383 root 11990 S asterisk -f 384 root 11990 S asterisk -f 385 root 11990 S asterisk -f 386 root 11990 S asterisk -f 387 root 11990 S asterisk -f 388 root 11990 S asterisk -f

On Wed, Oct 7, 2020 at 3:43 AM Marvin Scholz <epirat07 at gmail.com> wrote: > > Hi, please provide the contents of the config.log > > I can not reproduce your issue on my machine, --without-openssl > works fine here. > > Oh and next time please mention that its about libshout right > away :) https://gist.github.com/neheb/a26714035978d07a822985477524cd45 > > On 7

Hi, I have set up asterisk to run as non root, and allow admin users to log in to the server as asterisk, which gives them privileges to edit configs in the asterisk home directory. As for connecting to the console with 'asterisk -r' - this by default does not work as asterisk is owned stored in /usr/sbin/asterisk I am reading that the best way to solve this is to use 'visudo' -