[asterisk-users] Compiling asterisk makes Systemd timeout when starting the service
I am banging my head. Stock asterisk on Ubuntu 22.04 (Jammy) installs and works fine, but I want to update the source code. I use this configure line ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib/x86_64-linux-gnu --with-unixodbc=$(odbc_config --include-prefix)/ --with-pjproject-bundled --disable-dev-mode --with-user=asterisk And it works, but the when I do systemctl start asterisk it hangs , on my command line because systemd times out waiting for a confirmation that the service started, and start it id did. But a few seconds later systemd kills asterisk and again tries to start the service. systemctl status asterisk * asterisk.service - Asterisk PBX Loaded: loaded (/lib/systemd/system/asterisk.service; enabled; vendor preset: enabled) Drop-In: /run/systemd/system/service.d `-zzz-lxc-service.conf Active: activating (start) since Sat 2023-04-29 03:46:30 UTC; 1min 25s ago Docs: man:asterisk(8) Main PID: 93000 (asterisk) Tasks: 67 CPU: 1.597s CGroup: /system.slice/asterisk.service |-93000 /usr/sbin/asterisk -g -f -p -U asterisk `-93001 astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 93000 Apr 29 03:46:30 asterisk systemd[1]: Starting Asterisk PBX... The machine log Apr 29 03:48:06 asterisk systemd[1]: asterisk.service: Main process exited, code=exited, status=203/EXEC Apr 29 03:48:06 asterisk systemd[1]: asterisk.service: Failed with result 'exit-code'. Apr 29 03:48:06 asterisk systemd[1]: Failed to start Asterisk PBX. Apr 29 03:48:07 asterisk systemd[1]: asterisk.service: Scheduled restart job, restart counter is at 12. Apr 29 03:48:07 asterisk systemd[1]: Stopped Asterisk PBX. Apr 29 03:48:07 asterisk systemd[1]: asterisk.service: Start request repeated too quickly. Apr 29 03:48:07 asterisk systemd[1]: asterisk.service: Failed with result 'exit-code'. Apr 29 03:48:07 asterisk systemd[1]: Failed to start Asterisk P -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk issue reporting is now live on GitHub
All Asterisk issues should now be reported at https://github.com/asterisk/asterisk/issues The previous issue system at https://issues.asterisk.org remains in read-only mode for reference but will eventually be replaced with a searchable archive. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.
On Fri, Apr 28, 2023 at 10:43 AM Benoît Panizzon wrote: > Hi List > > Asterisk 16.28.0 in use. > > PJSIP in use > Two endpoints > Both using IPv6 > > One Endpoint on UDP, the other via TLS. > > Both with: > > t38_udptl=yes > ;fax_detect=yes > ;fax_detect_timeout=30 > rtp_ipv6=yes > > Both sides are T.38 capable and detect fax tone so no need for fax > detection on asterisk. > > Voice calls between the two work fine. > > But on a Fax call, I see this situation: > > A <=> Asterisk <=> B > > A: INVITE + Audio SDP => Asterisk => (same SDP) => B > > B: 200 OK + Audio SDP => Asterisk => (same SDP) => A > > * B Detects Fax-Tone! > > B: Re-Invite + UDPTL => Asterisk => (same SDP) => A > > A: 200 OK + UDPTL => Asterisk => 488 => B > > I tweakted the udptl setting in various ways, but I am unable to figure > out, why Asterisk is sending a 488 to B, after it first happily > forwarded the SDP to A and got confirmation from A it was happy to > accept that DSP. > > You could enable core debug and see if there's any insight, otherwise you'd have to actually provide the full traces. Asterisk also doesn't forward SDPs between sides so they're not the same SDP. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.
Hi List Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice calls between the two work fine. But on a Fax call, I see this situation: A <=> Asterisk <=> B A: INVITE + Audio SDP => Asterisk => (same SDP) => B B: 200 OK + Audio SDP => Asterisk => (same SDP) => A * B Detects Fax-Tone! B: Re-Invite + UDPTL => Asterisk => (same SDP) => A A: 200 OK + UDPTL => Asterisk => 488 => B I tweakted the udptl setting in various ways, but I am unable to figure out, why Asterisk is sending a 488 to B, after it first happily forwarded the SDP to A and got confirmation from A it was happy to accept that DSP. Any hint? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
