Re: [asterisk-users] Asterisk 18.0.0 Now Available
On 21.10.20 at 12:49 Joshua C. Colp wrote: > On Wed, Oct 21, 2020 at 7:46 AM Michael Maier wrote: > >> Hello! >> >> On 20.10.20 at 14:00 Asterisk Development Team wrote: >>> The Asterisk Development Team would like to announce the release of >> Asterisk 18.0.0. >>> This release is available for immediate download at >>> https://downloads.asterisk.org/pub/telephony/asterisk >> >> I just tested the new codec negotiation feature and unfortunately wasn't >> able to get it working as expected. I tried several configurations - but >> none has been working - the result >> has always been the same. >> > > This is expected right now. Foundational aspects were put in, but there is > still work to be done for PJSIP which will land in a future release. Oh - thanks for the information - I missed this :-(. How do I know if this feature is finally enabled? Will it be in asterisk 18 - or will it come in some later major version? > The > complexity of it and the investigation of how things work, interactions, > etc took considerably longer than expected. If there's specific scenarios > that you'd like to ensure are met you can reach out on the asterisk-dev > mailing list and George Joseph will add them to the list if not already > present. Well, I think the scenario I have should be a very easy and basic scenario. I already discussed it in the past. Therefore I think it's not necessary to add it again. Thanks Michael -- _ -- 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 18.0.0 Now Available
Hello! On 20.10.20 at 14:00 Asterisk Development Team wrote: > The Asterisk Development Team would like to announce the release of Asterisk > 18.0.0. > This release is available for immediate download at > https://downloads.asterisk.org/pub/telephony/asterisk I just tested the new codec negotiation feature and unfortunately wasn't able to get it working as expected. I tried several configurations - but none has been working - the result has always been the same. Use case: Alice calls Bob - sends INVITE G722 / alaw / ulaw Configured in Asterisk for this device: G722 / alaw / ulaw / gsm A: codec_prefs_incoming_offer = prefer: configured, operation: intersect, keep: all, transcode: prevent Bob: Configured in Asterisk for this device: alaw / ulaw B: codec_prefs_outgoing_offer = prefer: configured or pending, operation: intersect, keep: first or all, transcode: prevent Asterisk sends INVITE to Bobalaw / ulaw Asterisk receives OK from Bob alaw B: codec_prefs_incoming_answer = prefer: configured or pending, operation: intersect, keep: first or all, transcode: prevent Asterisk sends OK to Alice G722 / alaw / ulaw A: codec_prefs_outgoing_answer = prefer: pending, operation: intersect, keep: first or all, transcode: prevent => I would have expected alaw to be sent to A - but G722 / alaw / ulaw is sent and transcoding is active! What did I do wrong? Could you please add the correct configuration you expect to get the expected result alaw? Thanks Kind regards Michael -- _ -- 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 18.0.0 Now Available
On Wed, Oct 21, 2020 at 8:49 AM Michael Maier wrote: > On 21.10.20 at 12:49 Joshua C. Colp wrote: > > On Wed, Oct 21, 2020 at 7:46 AM Michael Maier > wrote: > > > >> Hello! > >> > >> On 20.10.20 at 14:00 Asterisk Development Team wrote: > >>> The Asterisk Development Team would like to announce the release of > >> Asterisk 18.0.0. > >>> This release is available for immediate download at > >>> https://downloads.asterisk.org/pub/telephony/asterisk > >> > >> I just tested the new codec negotiation feature and unfortunately wasn't > >> able to get it working as expected. I tried several configurations - but > >> none has been working - the result > >> has always been the same. > >> > > > > This is expected right now. Foundational aspects were put in, but there > is > > still work to be done for PJSIP which will land in a future release. > > Oh - thanks for the information - I missed this :-(. How do I know if this > feature is finally enabled? Will it be in asterisk 18 - or will it come in > some later major version? > > It is planned to land in a future Asterisk 18 release, it will be stated in the changelog/release notes and we will likely do a blog post about it as well. -- 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
Re: [asterisk-users] Asterisk 18.0.0 Now Available
On Wed, Oct 21, 2020 at 7:46 AM Michael Maier wrote: > Hello! > > On 20.10.20 at 14:00 Asterisk Development Team wrote: > > The Asterisk Development Team would like to announce the release of > Asterisk 18.0.0. > > This release is available for immediate download at > > https://downloads.asterisk.org/pub/telephony/asterisk > > I just tested the new codec negotiation feature and unfortunately wasn't > able to get it working as expected. I tried several configurations - but > none has been working - the result > has always been the same. > This is expected right now. Foundational aspects were put in, but there is still work to be done for PJSIP which will land in a future release. The complexity of it and the investigation of how things work, interactions, etc took considerably longer than expected. If there's specific scenarios that you'd like to ensure are met you can reach out on the asterisk-dev mailing list and George Joseph will add them to the list if not already present. Even just from AstriDevCon there were some things that individuals brought up. -- 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
[asterisk-users] Asterisk 18.0.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Security bugs fixed in this release: --- * ASTERISK-28589 - chan_sip: Depending on configuration an INVITE can alter Addr of a peer (Reported by Andrey V. T.) * ASTERISK-28580 - Bypass SYSTEM write permission in manager action allows system commands execution (Reported by Eliel Sardañons) * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash (Reported by Alexei Gradinari) New Features made in this release: --- * ASTERISK-6863 - [patch] allow Asterisk to set high ToS bits as non-root on Linux (Reported by Matt Addison) * ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything (Reported by candrews) * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add ability to match on source port (Reported by Sean Bright) * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl) * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) * ASTERISK-28533 - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) * ASTERISK-17808 - [patch] Unregister a realtime moh class (Reported by Byron Clark) * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) Bugs fixed in this release: --- * ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16 (Reported by Ross Beer) * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-29043 - app_queue: Leave empty sometimes not recorded as abandoned (Reported by Kfir Itzhak) * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-29034 - Lastpause of realtime members is reseting (Reported by Evandro César Arruda) * ASTERISK-27273 - app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command (Reported by Leandro Dardini) * ASTERISK-29033 - res_pjsip_session: Aggressively terminates session on failed re-INVITE (Reported by Joshua C. Colp) * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have appended RTP string to each message block. (Reported by Thomas Johnson) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis) * ASTERISK-29021 - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions (Reported by cmaj) * ASTERISK-28927 - Asterisk crash in music on hold (Reported by David Cunningham) * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address) (Reported by Michael Neuhauser) * ASTERISK-28995 - res_pjsip_registrar: Expires on statically configured contacts is not correct (Reported by tootai) * ASTERISK-28987 - BridgeCreated ARI event shows wrong video_mode info (Reported by sungtae kim) * ASTERISK-28978 - acl: named_acl rule misconfiguration results in segfault on reading rule from realtime (Reported by Andrew Yager) * ASTERISK-28975 - res_http_websocket: Text payload data doesn't necessary include trailing zero (Reported by Nickolay V. Shmyrev) * ASTERISK-28951 - Inconsistent behaviour queues.conf when there is (not) a [general] section (Reported by Walter Doekes) * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) * ASTERISK-28930 - ./configure --without-ssl build failure (Report
