So a caller can record its call with DTMF command.
This is working very well with codecs like ulaw. However when I change it for opus I can make nice calls but I can’t record anymore; I get this error:
[Mar 5 09:42:58] WARNING[9977][C-00000061]: translate.c:485 ast_translator_build_path: No translator path: (starting codec is not valid)
[Mar 5 09:42:58] WARNING[9977][C-00000061]: file.c:246 ast_writestream: Unable to translate to format wav, source format opus
[Mar 5 09:42:58] WARNING[9977][C-00000061]: channel.c:4075 __ast_read: Failed to write data to channel monitor read stream
So I tried to change my record extension/format like that but it changed nothing, it still does not work:
exten => _6XXX,1,Set(TOUCH_MONITOR_FORMAT=opus)
same => n,Dial(SIP/${EXTEN},20,wWxX)
same => n,Hangup()
With the command “core show file formats” I don’t see any opus format. Do I have to install anything more or Asterisk does not support opus records ? So then what would be the best codec alternative ?
Note the –disable-opus. Strange no ?
I still don’t see opus after the “core show file formats” command, and I still have the same error while trying to record.
Did the module download and install? Do you see it in “module show”? If you manually go into “make menuselect” is it selected? You may not have the required dependency to download and install it (it’s a binary module). That’s why I mentioned going into menuselect instead, to confirm.
As for PJSIP - we don’t use their media stack and disable vast parts of it so it’s not strange at all.
In “module show” I can see “res_format_attr_opus.so” but no “codec_opus”.
And yes in make menuselect it is selected.
Did I miss the “install_prereq” step ?
Shall I install opus from sources http://opus-codec.org/downloads/ ?
And the module is in /usr/lib/asterisk/modules? Does it appear in “module show”? Is there an error at startup if not? Does a translation path appear in “core show translation”?