The release of Asterisk 17.6.0 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:
Bugs fixed in this release:
-----------------------------------
| chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 | (Reported by Joseph Ades) | |
| res_pjsip: Apply outbound proxy to static contacts on AOR | (Reported by Joshua C. Colp) | |
| ./configure --without-ssl build failure | (Reported by Jaco Kroon) | |
| chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2 | (Reported by Jared Smith) | |
| chan_sip: chan_sip does not process 400 response to an INVITE. | (Reported by Frederic LE FOLL) | |
| res_corosync: causes asterisk crash in huge distributed environment. | (Reported by Università di Bologna - CESIA VoIP) | |
| "setvar" doesn't work properly in dahdi-channels.conf | (Reported by Marin Odrljin) | |
| StreamEcho() only returns 1 active stream | (Reported by Bill Kervaski) | |
| res_pjsip_session: Preserve stream label | (Reported by Joshua C. Colp) | |
| res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching | (Reported by Joshua C. Colp) | |
| Queue wrapuptime sometimes not respected (based on stale lastcall time) | (Reported by Walter Doekes) | |
| Stale code in app_queue to check untouched channel | (Reported by Walter Doekes) | |
| Stale comment in app_queue about ring_entry exception | (Reported by Walter Doekes) | |
| core_unreal / core_local: Add support for multistream and re-negotiation | (Reported by Joshua C. Colp) | |
| ARI channel create doesn't referencing the channel_id parameter | (Reported by sungtae kim) | |
| res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC | (Reported by Joshua C. Colp) | |
| bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation | (Reported by Joshua C. Colp) | |
| T.38 Segfaults in chan_pjsip_queryoption | (Reported by Yury Kirsanov) | |
| /channels/create doesn't get any parameters from the body | (Reported by sungtae kim) | |
| res_pjsip: crash when dialing non-sip uri | (Reported by Walter Doekes) | |
| res_fax: Double frame free when gateway in use with off-nominal format usage | (Reported by Gregory Massel) | |
| pjproject_bundled: Honor --without-pjproject. | (Reported by Alexander Traud) | |
| res_pjsip_logger writing too big packets | (Reported by nappsoft) | |
| bridge show all causes crash | (Reported by sungtae kim) | |
| Wrong return value check for fwrite when writing to pcap file | (Reported by nappsoft) |
Improvements made in this release:
-----------------------------------
| res_pjsip: Added option for disable rport parameter set | (Reported by sungtae kim) | |
| Continue reading string when ping received by websocket | (Reported by Nickolay V. Shmyrev) | |
| AMI SendText - add Content-Type parameter | (Reported by Kevin Harwell) | |
| res_http_websocket: Add masking to websocket client | (Reported by Moises Silva) | |
| Upgrade Asterisk to bundled pjproject 2.10 | (Reported by Kevin Harwell) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
Thank you for your continued support of Asterisk!
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