[asterisk-users] [External] Asterisk 18.12.0 question

[Joshua C. Colp]

On Fri, May 13, 2022 at 2:43 PM Dan Cropp <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 and www.asterisk.org
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