[asterisk-users] [External] [External] Asterisk 18.12.0 question

[Dan Cropp]

Thanks Joshua.

I didn’t describe that very well.

When I first noticed the res_http_transport_websocket wasn’t loading on that box, I compared the modules folder on both boxes.  My thought was I forgot some module that was required.

I noticed I forgot to include these files, so I added them to the package.  Rolled back the VM and re-installed.  Didn’t make a difference whether they were present or not.
/usr/lib/asterisk/modules/codec_g729a.*
/usr/lib/asterisk/modules/codec_silk.*
/usr/lib/asterisk/modules/codec_siren14.*
/usr/lib/asterisk/modules/codec_siren7.*
/usr/lib/asterisk/modules/format_ogg_opus.so

Comparing the menuselect-tree between the two versions, only changes I see are
func_evalexten
res_aeap
res_speech_aeap
and four test_aeap_... added to the TEST_FRAMEWORK.

Would it make sense for me to modify my bash script to disable those settings, compile, and try installing?  Bash script configures the menuselect options and compiles asterisk.
Seems like that would be a better apples to apples comparison.  Eliminating the new features.

From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Joshua C. Colp
Sent: Friday, May 13, 2022 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Subject: Re: [External] [asterisk-users] [External] Asterisk 18.12.0 question

On Fri, May 13, 2022 at 2:43 PM Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote:
Hi Joshua,

Thank you for helping me diagnose this.

Interesting that they are the exact same between versions.
File sizes are slightly different between the two when I compile them.  18.12.0 would have any new default configurations that were not part of 18.11.2, such as aeap.  Some codecs also seem to be defaulted now.  Not sure if that makes a difference.  After this e-mail, I will try disabling some of the new additions.  Maybe that will resolve things.
18.11.2 408136
18.12.0 411856

The source files are different, but the resulting binaries can differ between versions if headers/other things change. You'd also need to specify what "Some codecs also seem to be defaulted now.", unless things are deprecated or binary then the default is to have them enabled.


Here is output as it loads the res_pjsip_transport_websocket.so being loaded on 18.11.2…

Loading app_stack.so.
  == AGI Command 'gosub' registered
  == Registered application 'StackPop'
  == Registered application 'Return'
  == Registered application 'GosubIf'
  == Registered application 'Gosub'
  == Registered custom function 'LOCAL'
  == Registered custom function 'LOCAL_PEEK'
  == Registered custom function 'STACK_PEEK'
  == app_stack.so => (Dialplan subroutines (Gosub, Return, etc))
Loading res_pjsip_path.so.
  == res_pjsip_path.so => (PJSIP Path Header Support)
Loading res_pjsip_transport_websocket.so.
  == WebSocket registered sub-protocol 'sip'
  == res_pjsip_transport_websocket.so => (PJSIP WebSocket Transport Support)
Loading res_stasis_recording.so.
  == res_stasis_recording.so => (Stasis application recording support)
Loading res_pjsip_nat.so.
  == res_pjsip_nat.so => (PJSIP NAT Support)
Loading res_pjsip_diversion.so.
  == res_pjsip_diversion.so => (PJSIP Add Diversion Header Support)

Here is what logs show on 18.12.0….

Loading app_stack.so.
  == AGI Command 'gosub' registered
  == Registered application 'StackPop'
  == Registered application 'Return'
  == Registered application 'GosubIf'
  == Registered application 'Gosub'
  == Registered custom function 'LOCAL'
  == Registered custom function 'LOCAL_PEEK'
  == Registered custom function 'STACK_PEEK'
  == app_stack.so => (Dialplan subroutines (Gosub, Return, etc))
Loading res_pjsip_path.so.
  == res_pjsip_path.so => (PJSIP Path Header Support)
Loading res_pjsip_transport_websocket.so.
Loading res_stasis_recording.so.
  == res_stasis_recording.so => (Stasis application recording support)
Loading res_pjsip_nat.so.
  == res_pjsip_nat.so => (PJSIP NAT Support)
Loading res_pjsip_diversion.so.
  == res_pjsip_diversion.so => (PJSIP Add Diversion Header Support)
…
[May 13 12:34:08] WARNING[1400]: loader.c:2393 load_modules: Some non-required modules failed to load.
[May 13 12:34:08] WARNING[1400]: loader.c:2487 load_modules: Module 'chan_sip' has been loaded but was deprecated in Asterisk version 17 and will be removed in Asterisk version 21.
[May 13 12:34:08] ERROR[1400]: loader.c:2508 load_modules: Error loading module 'app_queue.so', missing dependency: res_monitor
[May 13 12:34:08] ERROR[1400]: loader.c:2508 load_modules: res_pjsip_transport_websocket declined to load.
[May 13 12:34:08] WARNING[1407]: chan_sip.c:35470 deprecation_notice: chan_sip has no official maintainer and is deprecated.  Migration to
[May 13 12:34:08] WARNING[1407]: chan_sip.c:35471 deprecation_notice: chan_pjsip is recommended.  See guides at the Asterisk Wiki:
[May 13 12:34:08] WARNING[1407]: chan_sip.c:35472 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
[May 13 12:34:08] WARNING[1407]: chan_sip.c:35473 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip

Because some of our customers have refused to migrate to PJSIP, we still compile support for chan_sip.
However, we make sure to disable the chan_sip web support in the configuration files.

The two reasons, according to the source code, that it would decline is if either it failed to register with the core res_pjsip support, or if the "sip" websocket protocol was already registered. The failure to register with the core res_pjsip support would output an error message, so more likely for some reason that the "sip" websocket protocol was already registered.

--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org>
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