asterisk

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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)
Jan Engelhardt's avatar Jan Engelhardt (jengelh) accepted request 871603 from Diederik de Groot's avatar Diederik de Groot (chan-sccp-b) (revision 65)
- 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

Zombie Ryushu's avatar

the chan_mobile.so module seems not to be packaged. Can this be fixed?


Jan Engelhardt's avatar

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?

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