ASTERISK-30064: pbx: iax2 switch causes crash due to deadlock and assertion: Reporter: N A (InterLinked) Labels: Date Opened: 05:15:28: Date Closed: 17:42:48: ... Attempting to use "switch" with IAX2 causes a crash. I am marking this major considering there is a major problem with the dialplan itself. {noformat} [2022-05 ...
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The Asterisk Development Team has announced the second beta of Asterisk 14.0.0. This beta is available for immediate download. Skip to content. ... [ASTERISK-26238] – res_pjsip: Empty global default_from_user causes crash [ASTERISK-26239] – res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname [ASTERISK ...
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Our favorite question is "wait – but why?" Asterisk is by and for the kind of people who aren't satisfied until they've tested their assumptions and chased every claim back to its source. (Our team is pedantic, and proud of it). We're constantly …
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Adding a hangup handler in the h extension or during a hangup handler execution is undefined behaviour. As always, hangup handlers, like the h extension, need to execute quickly because …
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( 0) Asterisk_Crash.rtf ( 1) BT_-_3-22-06.txt ( 2) convert_mp3_to_gsm.sh: Description: I don't know exactly what this debug output means, but shortly after these errors (attached in file) were seen in the messages log, asterisk crashed. OS is CentOS 4.2 Final, and the only addon we used was format_mp3. SIP peers, voicemail, and extensions (not ...
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ASTERISK-24336: PJSIP timer_min_se value under 90 causes crash: Reporter: Leon Rowland (zogot) Labels: Date Opened: 07:43:44: Date Closed: ... The crash showed the following message when I attempted to call between 101 and 102 When changing the value to 91 the call happened correctly. Asterisk Ready.
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Environment: Attachments: ( 0) backtrace_spychan_crash.txt ( 1) bt_&_bt_full_output.txt Description: Version 1.12.1, fresh install. Never experienced on old install, 1.2.3 When chanspy is hung up whilst spying on a PROVIDER channel (IAX2) it causes immediate crash.
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One undeniable sign that you've got a crush is when you start finding reasons—no matter how small or subtle—to initiate physical contact. It could be a light touch on their arm …
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Asterisk Value ISDN Cause codes (Q.850 & Q.931 unless specified) MFC/R2 SIP/PJSIP Motif; AST_CAUSE_NOT_DEFINED: Cause not defined: OR2_CAUSE_UNSPECIFIED
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The hangup cause AST_CAUSE_NOT_DEFINED is not actually a Q.931 cause code, and is used to capture hangup causes that do not map cleanly to a Q.931 cause code. IAX2, ISDN, and …
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[ASTERISK-23103] – Crash in ast_format_cmp, in ao2_find [ASTERISK-23104] – Specifying the SetVar AMI without a Channel cause Asterisk to crash [ASTERISK-23134] – res_rtp_asterisk port selection cannot handle selinux port restrictions [ASTERISK-23135] – Crash – segfault in ast_channel_hangupcause_set – probably introduced in 11.7.0
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ASTERISK-17176: [patch] WARNING message for each IAX peer with a qualifyfreqnotok setting Revision: 355999 Reporter: jcovert Testers: Sean Bright Coders: John ASTERISK-17339: [patch] memory leak with IAX in 1.8 Revision: 350222 Reporter: tzafrir Coders: Alex ASTERISK-17610: Crash on chan_iax2.so Revision: 330593 Reporter: mgrobecker Coders ...
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Understanding the Asterisk server issues, SIP trunk challenges, and specific troubleshooting techniques for the latest versions, like Asterisk 21, will enable you to resolve …
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ASTERISK-29191: tel: URI in Diversion header causes crash: Reporter: Mikhail Ivanov ([email protected]) Labels: Date Opened: 08:23:02.000-0600: Date Closed: 16:49:08.000-0600: Priority: Major: ... As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need ...
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( 1) ASTERISK-24368-13-root.diff ( 2) ASTERISK-24368-bt.txt ( 3) pjsip.conf: Description: When some subscriptions are persisted to the AstDB, there appear to be code paths where the re-construction of the subscription is rejected due to an invalid transport. This causes an endless cycle of crashes on startup, preventing Asterisk from ever starting.
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asterisk does not support tel-uri without reason or cause in History-Info receiving such a header will result in asterisk crashing. This patch will check if a tel URI is received and sets the reason to unknown instead of crashing: Comments: By: Asterisk Team (asteriskteam) 02:33:58.749-0600 Thanks for creating a report!
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[ASTERISK-28888] – res_corosync: causes asterisk crash in huge distributed environment. (Reported by Università di Bologna – CESIA VoIP) [ASTERISK-28954] – StreamEcho() only returns 1 active stream (Reported by Bill Kervaski) [ASTERISK-28955] – "setvar" doesn't work properly in dahdi-channels.conf (Reported by Marin Odrljin)
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[ASTERISK-24741] – dtls_handler causes Asterisk to crash [ASTERISK-24742] – Fix ast_odbc_find_table function in res_odbc [ASTERISK-24748] – res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur [ASTERISK-24751] – Integer values in json payload to ARI cause asterisk to crash
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Environment: Debian 10: Attachments: ( 0) core-asterisk-running-T23-24-08Z-full.txt ( 1) deadlock.txt ( 2) hangup_quickly.txt Description: Something seems to be very wrong with the bridging logic for handing video calls. When making a direct call between two endpoints, including going through local channels, all seems well.
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[ASTERISK-23220] – STACK_PEEK function with no arguments causes crash/core dump [ ASTERISK-23231 ] – Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load [ ASTERISK-23232 ] – LocalBridge AMI Event LocalOptimization value is opposite to what's expected
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Asterisk occasionally crashes while adding channel to bridge with ARI. I'm not sure if this is because of ARI or not, and also not sure if the real problem is adding channel to …
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Setting a port number on the host declaration, such as host=1.2.3.4:123 causing asterisk to crash. Confirmed to crash every time the peer is loaded with 1.8.7.1 using realtime and 1.8.7.2 with sip.conf.
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If you have been using an asterisk solution for some time, you probably know that asterisk may crash (yes, it happens!). And when it crashes (and if you have configured it) asterisk will write a coredump file.. Having been using asterisk in our solution for quite some time (since version 1.4 of asterisk till 13.17.2 in our last LTS Version XiVO Polaris) and having deployed it …
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ASTERISK-28506 asterisk crashes at random frequent intervals: is related to: ASTERISK-28209 res_pjsip_t38: Passthrough causes crash when re-invite collision: is related to: ASTERISK-28473 res_pjsip_t38: Crash on Asterisk 16.4: is related to: ASTERISK-28517 Asterisk segfault in t38_interpret_parameters at res_pjsip_t38.c:457
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[ASTERISK-23605] – res_http_websocket: Race condition in shutting down websocket causes crash [ASTERISK-23616] – Big memory leak in logger.c [ASTERISK-23620] – Code path in app_stack fails to unlock list [ASTERISK-23639] – PJSIP Realtime: Alembic migration needed in order to widen some string columns
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The problem is that when heavily used the asterisk containers start to crash, kubernetes restarts the container and its on again but this issue is constant, asterisk log says …
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Summary: ASTERISK-01327: Using SIPDtmfMode(inband) when calling voicemail causes crash: Reporter: sdolloff (sdolloff) Labels: Date Opened: 18:16:39.000-0600
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Asterisk is a term that refers to the use of a symbol (*) as a replacement for letters or words in order to censor or obscure them. It is often used in written communication, such as text …
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Contribute to liyingliang2022/fr development by creating an account on GitHub.
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[ASTERISK-25371] – Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts to generate hangup event (Reported by Abhay Gupta) [ASTERISK-28435] – cdr_pgsql: Unix socket doesn't work (Reported by Dmitry Svyatogorov) [ASTERISK-27981] – res_fax: Fax session leak with fax gatewaying (Reported by pasandev) [ASTERISK-28427] –
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For a given channel, there are two sources of hangup cause information: Note that in some cases, the hangup causes returned may not be reflected in . For example, if a Dial to a SIP UA is …
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ASTERISK-25960: The config_hook unit test causes Asterisk to crash if run a second time Reported by: George Joseph. Corey Farrell -- test_config: Fix failure and segfault when config_hook is run twice. Category: Third-Party/pjproject
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The Asterisk Development Team would like to announce the release of Asterisk 18.13.0. This release is available for immediate download at https://downloads.asterisk. org/pub/telephony/asterisk The release of Asterisk 18.13.0 resolves several issues reported by the
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[ASTERISK-24336] – PJSIP timer_min_se value under 90 causes crash [ASTERISK-24438] – res_pjsip_multihomed.so blocks Asterisk reload when DNS settings invalid [ASTERISK-24444] – PBX: Crash when generating extension for pattern matching hint [ASTERISK-24447] – Bridge DTMF hooks: Audio doesn't pass when waiting for more matching digits.
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I have to relaunch asterisk manually. (see additional information) 4. Also since Mac OSX 10.4, using the console command "asterisk restart now" would also shutdown asterisk, and would not restart. Need to start asterisk manually again. 5. When I use "asterisk -cvvv", I will get color syntax highlighted in the console, very nice!
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I suggest you use the first sentence that contains the word: "I would advise him [Kavanaugh] against letting Senate Republicans ram his nomination through in a fashion that will forever …
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