Re: [Astlinux-users] PJSIP Best Practice?

[Lonnie A. <li...@lo...>]

Hi Dan,

Sorry for the lack of replies, but many of us are still using chan_sip.

There may be a command more useful than "pjsip show channels"
--
pbx*CLI> pjsip show 
aor                    aors                   auth                   auths                  channel                
channels               channelstats           contact                contacts               endpoint               
endpoints              history                identifiers            identifies             identify               
qualify                registration           registrations          scheduled_tasks        settings               
subscription           subscriptions          transport              transports             unidentified_requests  
--

Lonnie



> On Jan 18, 2022, at 8:19 AM, Dan Ryson <da...@ry...> wrote:
> 
> AstLinux Team,
> 
> I'm delighted to see the ongoing progress of AstLinux.  Thank you!
> 
> Although Asterisk 13SE is still working fine for me, the Pre-Release recommendation (highlighted below) prompted me to start experimenting with Asterisk 16 and pjsip/pjproject on a new droplet PBX, which is now 48 hours old.  It's working well for the most part and I'm making slow and steady progress.  
> 
> To force myself to learn pjsip, I defeated the chan_sip module.  Accordingly, the AstLinux Status page now shows "No such command 'sip show registry'" and "No such command 'sip show peers'.  As a possible work around, I updated the Active Channels Command to "pjsip show channels" and repurposed the Show DAHDI Command to "pjsip show endpoints" - since DAHDI isn't being used.  This provides useful data and works fine.  However, the raw CLI output isn't particularly pretty.   
> 
> Since I'm surely not the first person to head down this path, I have a feeling that I'm missing something obvious and should ask the pros.  I'm hoping to hear your thoughts and advice for showing pjsip status.  Is there a better recommended practice?
> 
> Thanks,
> 
> Dan
> 
> On Thu, Jan 13, 2022 at 02:20 PM, Lonnie Abelbeck <li...@lo...> wrote:
> Announcing AstLinux Pre-Release: astlinux-1.4-5333-94c1eb
> 
> Key new features:
> 
> -- Asterisk 18.x is now supported, along with Asterisk 16.x and Asterisk 13.x built --without-pjproject
> 
> -- Previous ast13-firmware-1.x is no longer being updated, ast13-firmware-1.x users should either switch to ast16-firmware-1.x (recommended)
>    or use ast13se-firmware-1.x if chan_pjsip is not used in your dialplan.
> 
> ** The AstLinux Team is regularly upgrading packages containing security and bug fixes as well as adding new features of our own.
> 
> -- Linux Kernel 4.19.224 (version bump), security and bug fixes
> 
> -- OpenSSL, version bump to 1.1.1m, security fixes: none
> 
> -- WireGuard VPN, module 1.0.20211208 (version bump), tools 1.0.20210914 (no change)
> 
> -- libcurl (curl) version bump to 7.81.0
> 
> -- LibreTLS, version bump to 3.4.2
> 
> -- msmtp, version bump to 1.8.19, 'msmtpd' security fix
> 
> -- nano, version 2.7.5, fix issue where not saving a file could still copy the file to /mnt/asturw/
> 
> -- Network tab, Non-ACME Self-Signed HTTPS Certificate, use 2048 key length.
> 
> -- Asterisk 13.38.3 ('13se' no change)
>    Last Asterisk 13.x "Legacy" version, built --without-pjproject
> 
> -- Asterisk 16.23.0 (version bump) and 18.9.0 (new version)
> 
> -- Complete Pre-Release ChangeLog:
> https://s3.amazonaws.com/beta.astlinux-project/astlinux-changelog/ChangeLog.txt
> 
> The "AstLinux Pre-Release ChangeLog" and "Pre-Release Repository URL" entries can be found under the "Development" tab of the AstLinux Project web site ...
> 
> AstLinux Project -> Development
> https://www.astlinux-project.org/dev.html
> 
> AstLinux Team
> 
> _______________________________________________
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> 
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