I think I can confidently say, after most of a day and reading the following....
https://stackoverflow.com/questions/66768885/why-doesnt-asterisk-17-catch-hangup-request-from-pjsip-client-solved https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+AGICommand_set+autohangup https://community.freepbx.org/t/inbound-calls-dont-hang-up/53612 https://community.freepbx.org/t/pjsip-problem-channel-not-closing/65311/7 https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_ENDPOINT https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_pjsip ... that Asterisk doesn't like mixing autoHangup with AGI, and there appears to be no way of the ts-agi library I'm using knowing that it has autoHungup, so it can't close the AGI connection which seems to release Asterisk to hangup properly. I had thought that the AGIEXITONHANGUP variable might help, but it appears to do nothing, although I'm unsure if I'm setting it correctly as: Here it says the flag is "1" https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables And here it says the flag is "yes" https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Application_AGI That said, I wish I could use ARI not AGI but without the current media offset available on ARI, I need AGI :) So for now, the workaround is to forget about setting autohangup, and just hangup the caller manually at the point they don't over-ride the timeout. Thanks! On Wed, 26 May 2021 at 18:01, Joshua C. Colp <jc...@sangoma.com> wrote: > > On Wed, May 26, 2021 at 1:58 PM Jonathan H <lardconce...@gmail.com> wrote: >> >> I have also tried configuring pjsip wizard like this. >> >> endpoint/rtp_timeout=5 >> >> And I see this shortly after the "hangup" command has been sent, so >> that part is working: >> >> [May 26 17:36:37] NOTICE[1276]: res_pjsip_sdp_rtp.c:150 >> rtp_check_timeout: Disconnecting channel >> 'PJSIP/fromvoipfone-206-0000000b' for lack of audio RTP activity in 5 >> seconds >> >> But, again, it doesn't disconnect. The line stays open. And yes, my >> fallthrough after agi is >> >> same => n, Hangup() >> >> Also, apparently I now have a load of channels, which won't even hangup with >> >> channel request hangup all >> >> Requested Hangup on channel 'PJSIP/fromvoipfone-206-0000000b' >> Requested Hangup on channel 'PJSIP/fromvoipfone-206-0000000a' >> Requested Hangup on channel 'PJSIP/fromvoipfone-206-00000009' >> >> ...and wait.. and then... >> >> Channel Location State Application(Data) >> PJSIP/fromvoipfone-2 s@test:2 Up AGI(agi://localhost:3456) >> PJSIP/fromvoipfone-2 s@test:2 Up AGI(agi://localhost:3456) >> PJSIP/fromvoipfone-2 s@test:2 Up AGI(agi://localhost:3456) >> 3 active channels >> 3 active calls >> >> So they just won't die. >> >> Asterisk 18.4.0 - worth filing a bug? > > > Is your AGI closing the connection or are you expecting Asterisk to drop it? > (I'm not that familiar with FastAGI or AGI these days, just wondering what > happens if you drop the connection) > > -- > Joshua C. Colp > Asterisk Technical 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 -- _____________________________________________________________________ -- 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
