asterisk
No description set
-
9
derived packages
- Download package
-
Checkout Package
osc -A https://api.opensuse.org checkout network:telephony/asterisk && cd $_
- Create Badge
Refresh
Refresh
Source Files
Filename | Size | Changed |
---|---|---|
asterisk-18.2.0.tar.gz | 0027895005 26.6 MB | |
asterisk-18.2.0.tar.gz.asc | 0000000836 836 Bytes | |
asterisk-cflags.diff | 0000000946 946 Bytes | |
asterisk-configure-paths.diff | 0000000987 987 Bytes | |
asterisk-init.diff | 0000000967 967 Bytes | |
asterisk-rundir.diff | 0000000876 876 Bytes | |
asterisk.changes | 0000056077 54.8 KB | |
asterisk.init | 0000001107 1.08 KB | |
asterisk.keyring | 0000015559 15.2 KB | |
asterisk.service | 0000000301 301 Bytes | |
asterisk.spec | 0000021417 20.9 KB | |
jansson-2.12.tar.bz2 | 0000404669 395 KB | |
pjproject-2.10.tar.bz2 | 0007339188 7 MB |
Revision 65 (latest revision is 83)
- update to 18.2.0: * Security - [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains * Bug - [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event - [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy - [ASTERISK-29155] - app_queue: Deadlock between queues container and individual queues - [ASTERISK-29161] - Incorrect setup of recall channels - [ASTERISK-29168] - Asterisk crashes during call transfer - [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable - [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses - [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses - [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped - [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send - [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the Transfer result - [ASTERISK-29210] - res_pjsip: Crash when examining transport - [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts - [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are accepted. - [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled. - [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. - [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. - [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are accepted. - [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled. - [ASTERISK-29209] - Debug messages printed by scope trace might be missing newlines - [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels spuriously - [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend - [ASTERISK-29188] - null media causing the Asterisk crash - [ASTERISK-29173] - Media cache URL requests allow infinite redirects - [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without entries - [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS response - [ASTERISK-29191] - tel: URI in Diversion header causes crash - [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered - [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable - [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription - [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description - [ASTERISK-29191] - tel: URI in Diversion header causes crash - [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set * Improvement - [ASTERISK-29118] - VoiceMail() should have an option to play greetings as Early Media - [ASTERISK-28549] - Two repeated 183 - [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other newer linux versions - [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in /tmp - [ASTERISK-28549] - Two repeated 183
Comments 2
the chan_mobile.so module seems not to be packaged. Can this be fixed?
It looks like you already fixed this in home:zombie_ryushu. Where is the difficulty in using the
osc sr home:zombie_ryushu/asterisk network:telephony/asterisk
command?