| [..] |
| ASTERISK-13000: [patch] segfault checking messages using mwi events |
| ASTERISK-13001: [patch] multiple voicemail and directory modules |
| ASTERISK-13002: [patch] Response to REGISTER is sent to wrong address when using rport |
| ASTERISK-13003: pulsedial=yes does not work |
| ASTERISK-13004: [patch] Play a sound to caller when picking up a call |
| ASTERISK-13005: [patch] support basic caller ID for snom call pickup |
| ASTERISK-13006: [patch] !! Unknown IE 50 (cs5, Unknown Information Element) & RTCP SR transmission error, rtcp halted |
| ASTERISK-13007: [patch] Hang up during call forward into voicemail crashes Asterisk. |
| ASTERISK-13008: writesql does not work |
| ASTERISK-13009: [patch] Voicemail occasionally blanks out the config file |
| ASTERISK-13010: Add netfilter functionality to asterisk to plumb NAT holes for RTP when running on NATting router |
| ASTERISK-13011: Errors in Manager events |
| ASTERISK-13012: "RTCP SR transmission error, rtcp halted" logged when SIP call put on hold |
| ASTERISK-13013: [patch] Use comma as the parameter separator. |
| ASTERISK-13014: Crash while processing events in manager.c |
| ASTERISK-13015: [patch] [Solaris] ./configure script can not use h323.h and compilation fails |
| ASTERISK-13016: call processing deadlock with dialplan reload and ast_hint_state_changed |
| ASTERISK-13017: The ackcall feature in agent's configurations is misinterpreted |
| ASTERISK-13018: [patch] Small fixes to two scripts in build_tools |
| ASTERISK-13019: TRUNK fails to compile on MacOS |
| ASTERISK-13020: CLI non-responsive |
| ASTERISK-13021: Asterisks threads hangs |
| ASTERISK-13022: GSM sound quality degraded |
| ASTERISK-13023: NAI not set correctly after small configuration changes |
| ASTERISK-13024: [patch] Asterisk crashes if udptl.conf is not available |
| ASTERISK-13025: problem handling race condition - reINVITE before ACK |
| ASTERISK-13026: [patch] bug in console command parser |
| ASTERISK-13027: Unlimited call for limited calls under 1 seconds (L option) |
| ASTERISK-13028: mwi activates for more than one mailbox if they have the same mailbox name but different contexts. |
| ASTERISK-13029: Crash after attended transfer and call park |
| ASTERISK-13030: 'quit' is twice in the Asterisk console history |
| ASTERISK-13031: freezing all channels |
| ASTERISK-13032: Periodic and returned: sip_xmit returned -1: Operation not permitted |
| ASTERISK-13033: [patch] On memory allocation failure return NULL or we will crash. |
| ASTERISK-13034: crash after misdn_set_opt_exec |
| ASTERISK-13035: Show queue and EventQueueStatus showing wrong values |
| ASTERISK-13036: [patch] ERROR[23999]: res_config_ldap.c:1292 update_ldap: Couldn't modify ... Undefined attribute type |
| ASTERISK-13037: The Extension field of the Newexten event shows the actual extension even if that in the dial plan involves a pattern |
| ASTERISK-13038: [patch] 1.6.1 beta2 does not build on PPC |
| ASTERISK-13039: [patch] Fix a memory leak while trying to free a memory that wasn't allocated by ast_alloc_region() |
| ASTERISK-13040: [patch] SIP/TLS enabled - just one call possible - 481 Call/Transaction Does Not Exist |
| ASTERISK-13041: [patch] Reject an incoming call to peer due to call limit with "603 Declined". It`s not correct. |
| ASTERISK-13042: crash after atxfer |
| ASTERISK-13043: Make clean starts configure in menuselect |
| ASTERISK-13044: TDM400P FXO module fails to dialout if alaw=35 is set |
| ASTERISK-13045: Crash in ast_bridge_call() when 'NoCDR' app called inside Local channel |
| ASTERISK-13046: [patch] new eventflag for agiexec-events on ami |
| ASTERISK-13047: AGI Script 'Failed to execute' |
| ASTERISK-13048: [patch] missing DESTDIR for documentation |
| ASTERISK-13049: [patch] Voicemail password does not save properly |
| ASTERISK-13050: [patch] Add CLI command 'manager logout <user> [from <ipaddress>]' |
| ASTERISK-13051: [patch] Can't record early media after sending a "183 Session Progress". |
| ASTERISK-13052: DUNDi queries/lookups from 32-bit to 64-bit machine fails; 64-bit to 32-bit operations OK |
| ASTERISK-13053: default sound files for Core Sound and Music On Hold File Packages not the same format |
| ASTERISK-13054: [patch] AGI command "answer" not really set in answer mode when forkcdr |
| ASTERISK-13055: Changing vm secret produces a deformed users.conf |
| ASTERISK-13056: Join event uses CallerID header for caller ID number, when other events now use CallerIDNum |
| ASTERISK-13057: chan_iax2 flood with 'No private structure for packet' |
| ASTERISK-13058: [patch] Problem with timeout in AGI RECORD FILE |
| ASTERISK-13059: CPU Usage Increases and then Asterisk Crashes |
| ASTERISK-13060: [patch] libtonezone requires -lm |
| ASTERISK-13061: Cannot compile SVN with patch from bug report 11261 (H323Plus) |
| ASTERISK-13062: [patch] Memory leak if the sla_thread is not running |
| ASTERISK-13063: "empty" overlap-dial-in doesn't work |
| ASTERISK-13064: After upgrading from 1.4.21.2 to 1.4.22 unaswered calls aren't correctly saved as CDR |
| ASTERISK-13065: [patch] Non existent option in the sample configuration file followme.conf (typo) |
| ASTERISK-13066: restart gracefully / when convenient doesn't work with the AMI |
| ASTERISK-13067: Voicemail delivered to wrong mailboxes |
| ASTERISK-13068: [patch] Limit connect file and others will not play warnings |
| ASTERISK-13069: After upgrading from 1.4.21.2 to 1.4.22 running MeetMe with D option doesn't ask for conference PIN |
| ASTERISK-13070: [feature request] Asterisk does not respect DNS TTL in iax.conf |
| ASTERISK-13071: switch statement: Empty patterns don't fallback to Default label |
| ASTERISK-13072: Realtime MusicOnHold patch not working on asterisk version 1.4.13 |
| ASTERISK-13073: Pong response not properly terminated. |
| ASTERISK-13074: [patch] Messages not marked as read/unread properly when moved from New to Old folder and back. |
| ASTERISK-13075: Memory Leakage |
| ASTERISK-13076: Recordings with Mixmonitor still ou of sync |
| ASTERISK-13077: Mobile connection is broken after 40+ minutes (asterisk hangs) |
| ASTERISK-13078: My server explodes when runnin astrisk core |
| ASTERISK-13079: app_fax still not building correctly |
| ASTERISK-13080: [patch] Patch diff for new dail plan application |
| ASTERISK-13081: addmin/res_config_mysql - Invalid Database |
| ASTERISK-13082: [patch] fxo modules incorrectly believes channel is answered, if telco reverses line polarity at off hook. |
| ASTERISK-13083: [patch] IAX2 not conforming to standard |
| ASTERISK-13084: [patch] Asterisk core dumps random |
| ASTERISK-13085: Configuration reload overrides channel variable setting of ackcall (autologoff, acceptdtmf and enddtmf) |
| ASTERISK-13086: Can't use G729 with Trunk |
| ASTERISK-13087: External calls can initiate blind transfer using feature keys after call is parked |
| ASTERISK-13088: ldap searchs fails on openldap when you use an additional filter |
| ASTERISK-13089: [patch] Allow for adding message body to the SIP NOTIFY message |
| ASTERISK-13090: Unable to Dial() through FXO port until line has rung once |
| ASTERISK-13091: Allow for adding message body to the SIP NOTIFY message |
| ASTERISK-13092: no dialling tone with AMI command originate & 1.6.0.1 |
| ASTERISK-13093: Calls are not beeing disconnected |
| ASTERISK-13094: jitterbuffer |
| ASTERISK-13095: Didn't call UPDATE or INSERT query to database |
| ASTERISK-13096: [patch] The function ARRAY slows asterisk down |
| ASTERISK-13097: On a configuration reload, cdr_csv ignores options if cdr.conf was unchanged |
| ASTERISK-13098: [patch] Multiple bugs in app_minivm |
| ASTERISK-13099: [patch] update app_readexten to conform to our production version |
| ASTERISK-13100: Asterisk can't watch for more than 5 devices on a single hint |
| ASTERISK-13101: [patch] Add new application MinivmMWI to app_minivm.c |
| ASTERISK-13102: Can't build res_config_mysql from Addons 1.6.1-rc1 |
| ASTERISK-13103: [patch] Skinny switch can still send tones after hangup |
| ASTERISK-13104: IAX2 Channel queue members lose their extension number on queues in 1.6 |
| ASTERISK-13105: on excessive registraton failures: security feature to lockout the IP |
| ASTERISK-13106: Sip Registration for Extensions to allow only private IP address, while Trunks allowed from outside Internet |
| ASTERISK-13107: [patch] Add packet information to debug |
| ASTERISK-13108: Asterisk 1.2.30.2 can be crashed remotely when using realtime for IAX2 users. |
| ASTERISK-13109: Asterisk crashes when Playback/Backround with non existing file. |
| ASTERISK-13110: Asterisk crashes when transfering call multiple times. |
| ASTERISK-13111: SIP Channels Hang - Last Message: Rx BYE - Need Destroy: 2 |
| ASTERISK-13112: SDP replies incorrect - 'a=inactive' - replied to with 'a=sendrecv' |
| ASTERISK-13113: Zaptel/DAHDi freezes E1 PRI channels when recieving fax. |
| ASTERISK-13114: [patch] Average talktime parameter for queues |
| ASTERISK-13115: Blind transfer does not work upgrade to 1.4.23-rc1 |
| ASTERISK-13116: "help iax set jitter" does not return |
| ASTERISK-13117: "help moh (classes show|files show|reload)" does not return |
| ASTERISK-13118: "Failed to open /dev/dahdi/transcode: No such file or directory" after "asterisk -U asterisk -G asterisk -cvv" |
| ASTERISK-13119: AGI Leaves zombies behind it |
| ASTERISK-13120: [patch] Need tab completion of "core set debug X filename.c" |
| ASTERISK-13121: CLI command "help sip set history {on|off}" shows deprectated usage in help text |
| ASTERISK-13122: [branch] gtalk web no incoming or outgoing calls |
| ASTERISK-13123: including same dial plan into 2 different contexts, produces different priorities |
| ASTERISK-13124: Different IMAP settings for realtime and static config setups |
| ASTERISK-13125: setting call duration limit in Dial command makes the call to hangup on answer |
| ASTERISK-13126: meetme plays wrong language announcements |
| ASTERISK-13127: Invalid SDP attributes for boolean T.38 parameters (T38FaxFillBitRemoval, etc.) |
| ASTERISK-13128: [patch] Add 'i' option to app_page (disable call forwarding) |
| ASTERISK-13129: Asterisk 1.4.23 rc1 Blind Transfer to Park fails |
| ASTERISK-13130: no gtalk capable clients to talk to & Asterisk needs to be restarted after adding jabber buddies |
| ASTERISK-13131: incoming calls ignore the user name |
| ASTERISK-13132: [patch] CLI shows agent_calls with verbose set to 0 |
| ASTERISK-13133: [patch] Incoming Gtalk calls fail |
| ASTERISK-13134: No audio on outgoing Gtalk calls if /etc/hosts has Centos default; work-around provided |
| ASTERISK-13135: The caller id is ignored when transferring the call |
| ASTERISK-13136: [patch] struct ucred cred not needed on OpenBSD. Compile error |
| ASTERISK-13137: MWI does not work with SIP using TCP transport |
| ASTERISK-13138: Enabling qualify on a SIP TCP peer brings pain and coredumps |
| ASTERISK-13139: [patch] Janitor, use ARRAY_LEN when possible. |
| ASTERISK-13140: [patch] Make app_rpt compile in dev-mode. |
| ASTERISK-13141: Transfer AMI event missing "TargetChannel" and "TargetUniqueid" |
| ASTERISK-13142: [patch] Ping action response packet doesn't contain ActionID |
| ASTERISK-13143: [patch] Invalid response from ListCategories action if no categories is found |
| ASTERISK-13144: Small delay when using nocallsetup=no |
| ASTERISK-13145: [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML |
| ASTERISK-13146: DTMF via INFO or RFC2833 via SIP for extensions processing in Asterisk 1.6 leads to instant channel hangup |
| ASTERISK-13147: Bug with iax channel (and perhaps IAXmodem) : randomly crashes asterisk |
| ASTERISK-13148: [patch] Wrong usage of sscanf with use of uninitialized variable caused accidental parsing of RTP/SAVP |
| ASTERISK-13149: [patch] Ability to use alaw files with extension ".alw" |
| ASTERISK-13150: [patch] Column count doesn't match value count |
| ASTERISK-13151: [patch] autosupport script and manual changes |
| ASTERISK-13152: [patch] restart gracefully drops cap_net_admin capability |
| ASTERISK-13153: Pickup() can't pickup calls to some SIP devices |
| ASTERISK-13154: [patch] Use of 'uint' instead of 'unsigned int' causes build problems on FreeBSD |
| ASTERISK-13155: bad Hangup Event on channel |
| ASTERISK-13156: Asterisk logs: Max retries exceeded to host on IAX/peer (type = 6, subclass = 11) but these packages can't be found in a tcpdump |
| ASTERISK-13157: [patch] Add static values for db columns on cdr_addon_mysql |
| ASTERISK-13158: [patch] Can't compile Asterisk 1.6.0.2 |
| ASTERISK-13159: [patch] Incorrect jump to extension |
| ASTERISK-13160: Channel "hangs" |
| ASTERISK-13161: 4XX Responses to a BYE request |
| ASTERISK-13162: [patch] DAHDI group dials/members broken with AddQueueMember / queue member add |
| ASTERISK-13163: Asterisk can crash when database schema is changed |
| ASTERISK-13164: [patch] Registrations using a username without domain will fail. |
| ASTERISK-13165: Starting or restarting asterisk causes seg fault and core dump, apparently in ael/pval.c:4833 |
| ASTERISK-13166: Asterisk cuts a piece of the dynamic member dialstring |
| ASTERISK-13167: RTP playout does not match ptime |
| ASTERISK-13168: AEL ¯o("arg") with UTF-8 argument incorrectly compiles into dialplan |
| ASTERISK-13169: UK caller ID on X100P, not working |
| ASTERISK-13170: sip register: reserved character check not RFC 3261 compliant |
| ASTERISK-13171: Action: SIPpeers requires write permissions in manager interface |
| ASTERISK-13172: [patch] error allocationg a manager user |
| ASTERISK-13173: [patch] allow multi-timezone GotoIfTime & ExecIfTime |
| ASTERISK-13174: [patch] Asterisk is using wrong clock frequency in text T140 |
| ASTERISK-13175: mISDN layer stops working |
| ASTERISK-13176: [patch] "pri_find_dchan: No D-channels available!" error on console when using wcb4xxp |
| ASTERISK-13177: [patch] Janitor project to use ARRAY_LEN |
| ASTERISK-13178: Delete a queue from realtime crashes Asterisk |
| ASTERISK-13179: app_queue does not update on realtime update |
| ASTERISK-13180: revision r77858 breaks 'failed'-extension functionality |
| ASTERISK-13181: Memory usage increase when using SUBSCRIBE + vars defined in sip.conf |
| ASTERISK-13182: [patch] apps/app_festival.c does not compile for PPC target |
| ASTERISK-13183: core-sounds-en.txt file is outdated. |
| ASTERISK-13184: [patch] Fix a memory leak while using threadstorage. |
| ASTERISK-13185: [patch] IAXy runs ok for a while, then goes bonkers |
| ASTERISK-13186: misdn trunk meetme user gets dialtone after entering conference |
| ASTERISK-13187: Crash (included coredump) |
| ASTERISK-13188: [patch] Jitterbuffer stops accepting new frames until it is empty after maxjitterbuffer is exceeded. |
| ASTERISK-13189: 'Random' segfault and core dump |
| ASTERISK-13190: trap divide error |
| ASTERISK-13191: Unable to dial out to external numbers using IAX |
| ASTERISK-13192: Asterisk Command line 'Soft Hangup' no longer is recognised |
| ASTERISK-13193: [patch] Attempted improvements to ./doc/tex/qos.tex [patch] |
| ASTERISK-13194: vmail.cgi does not work when using users.conf |
| ASTERISK-13195: [patch] res/res_http_post also need to include libgen.h in FreeBSD. |
| ASTERISK-13196: [patch] mISDN Dial parameter not documented |
| ASTERISK-13197: Asterisk is sending multiple INVITE after 200 OK as been acknowledged |
| ASTERISK-13198: reload of cli_aliases doesn't work |
| ASTERISK-13199: "outboundproxy" in "general" section of sip.conf doesn't work |
| ASTERISK-13200: [patch] chan_dahdi effectively ignores dahdichanname while looking for a configuration file |
| ASTERISK-13201: Deadlock chan_dahdi.c and channel.c |
| ASTERISK-13202: cdr_pgsql does not work in asterisk 1.6.0.x |
| ASTERISK-13203: [patch] chanvar-option for peers in sip.conf |
| ASTERISK-13204: [patch] Astrerisk crashes using the app_queue.c transfer datastores |
| ASTERISK-13205: Looking at this signaling log I found Asterisk is not retransmiting ACK on "603 declined" is this normal ? |
| ASTERISK-13206: [patch] Urgent messages are automatically forwarded as urgent |
| ASTERISK-13207: [patch] Improve documentation for res_monitor |
| ASTERISK-13208: Calls parked fail to return to the correct phone after timeout |
| ASTERISK-13209: App voicemail try to open files in wrong location |
| ASTERISK-13210: [patch] COLP/CONP support in QSIG |
| ASTERISK-13211: too small reponse for dbget through agi |
| ASTERISK-13212: when phone loses connection to asterisk during call, after it can't make any new call |
| ASTERISK-13213: [patch] Improve debugging output for specific context inclusion failure case in pbx_config |
| ASTERISK-13214: Outbound h323 dial via call file transfers control to context/exten/priority before call is answered |
| ASTERISK-13215: app_fax needs additional include to build with spandsp-0.0.6pre3 |
| ASTERISK-13216: Dial hangs up call immediately after answer |
| ASTERISK-13217: [patch] Using Originate as a dialplan application |
| ASTERISK-13218: Extensions configuration is not being sorted correctly |
| ASTERISK-13219: channel.c:2569 ast_indicate_data: Unable to handle indication -1 |
| ASTERISK-13220: [patch] Regression When Playing WAV49 Audio Files |
| ASTERISK-13221: [patch] ast_frdup does not duplicate integer-frame-data |
| ASTERISK-13222: MinivmAccMess() does not set MINIVM_ACCMESS_STATUS |
| ASTERISK-13223: no iax trunking on 1.6.1-beta3 |
| ASTERISK-13224: PlayDTMF is not working |
| ASTERISK-13225: Userfield not updating when the call is answered, if not answered it updates properly |
| ASTERISK-13226: Address out of bounds in queue_log using transfer |
| ASTERISK-13227: Problem with matching incoming calls on sip friend |
| ASTERISK-13228: class with only application is unuseable |
| ASTERISK-13229: [patch] unable to set DAHDI_VMWI to lower level drivers |
| ASTERISK-13230: autologoff does not work |
| ASTERISK-13231: bad callerid on incoming SIP transferred calls |
| ASTERISK-13232: Asterisk stops processing the dialplan after a While() in h extension |
| ASTERISK-13233: libsrtp integration in asterisk |
| ASTERISK-13234: srtp procedure for installation shows some configuration errors |
| ASTERISK-13235: [patch] Simplify main/strings.c code |
| ASTERISK-13236: kill() does not kill all processes |
| ASTERISK-13237: Segfault in chan_sip |
| ASTERISK-13238: Asterisk fails to execute dialplan steps if extensions are stored in mysql database |
| ASTERISK-13239: Dahdi 2.1.0 causes HDLC Bad FCS (8) and Primary D-Channel on span 2 down |
| ASTERISK-13240: [patch] app_followme crashed if followmeid no specified |
| ASTERISK-13241: Incoming SIP invites don't match properly |
| ASTERISK-13242: Asterisk not sending out RTP packets |
| ASTERISK-13243: [patch] Supplying stereo .WAV file causes Asterisk to crash |
| ASTERISK-13244: [patch] SIPAddHeader problems with escaping and quoting |
| ASTERISK-13245: [patch] app_chanspy crashed so-as "chanspy_ds.lock" has random values |
| ASTERISK-13246: [patch] Thread deadlock causes Asterisk to stop routing calls to agents, agents unable to change status |
| ASTERISK-13247: Asterisk 1.6.1-beta4 and 1.6.0.3-rc1 always crash when dialing or receiving a call trough wcb4xxp |
| ASTERISK-13248: realtime queue doesn't update strategy value |
| ASTERISK-13249: asterisk crashes when voicemail app finish to write a message |
| ASTERISK-13250: [patch] wrong phtread_mutex_init |
| ASTERISK-13251: Incorrect processing of "maxuser" parameter in real-time meetme |
| ASTERISK-13252: Asterisk crashes when calling more than a single location |
| ASTERISK-13253: [patch] Multi-host T.38 negotiation |
| ASTERISK-13254: r166257 does not compiles reporting error at chan_iax2.c |
| ASTERISK-13255: asterisk compile error after compiling and installing dahdi |
| ASTERISK-13256: [patch] Macro execution doesn't get to "h" extension |
| ASTERISK-13257: DNSmgr causes port numbers to be set to zero |
| ASTERISK-13258: queues fail to recognize already logged in local channel using \n member |
| ASTERISK-13259: Directory does not work with IMAP Voicemail |
| ASTERISK-13260: Using dtmfmode=info AND canreinvite=yes (both in sip.conf) AND dynamic features (features.conf/Dial() with w or W flags) |
| ASTERISK-13261: Enumerated type and integer comparison do not work as you might expect |
| ASTERISK-13262: [applicationmap] is not detecting for multiple digits |
| ASTERISK-13263: [patch] FreeBSD, OpenBSD and OS/X asterisk compilation fail.. |
| ASTERISK-13264: [patch] CLI shows colorization code incorrectly sometimes |
| ASTERISK-13265: Wrong CallerID on attended transfer |
| ASTERISK-13266: [patch] autosupport script not 100% ready for DAHDI |
| ASTERISK-13267: Subscription not removed when user becomes unsubscribed |
| ASTERISK-13268: Meetme announcing the user name |
| ASTERISK-13269: Asterisk crash when it pickups a DAHDI channel |
| ASTERISK-13270: No CallerID in CDR when inbound calls from Dahdi are routed to Queue |
| ASTERISK-13271: menuselect sets defaults too late |
| ASTERISK-13272: [patch] Function in iLBC conflicts with function in chan_vpb |
| ASTERISK-13273: Agent is shown in use after a SIP Register |
| ASTERISK-13274: [patch] followme should answer the call |
| ASTERISK-13275: Pattern matching for extensions with ranges is broken |
| ASTERISK-13276: Registration uses hwclock and sysclock to retrieve time |
| ASTERISK-13277: [patch] Annoying FSK mwispill is heard on handset on an FXS connected line, No dialtone heard if happens. |
| ASTERISK-13278: res_musiconhold won't compile anymore after ao2 fixes |
| ASTERISK-13279: Unexpected message "One or more of the parameters in the config does not pass our validity checks." |
| ASTERISK-13280: Dialplan variables not set upon blind xfer |
| ASTERISK-13281: crash by ast_dynamic_str_thread_build_va |
| ASTERISK-13282: crash by ast_dynamic_str_thread_build_va |
| ASTERISK-13283: Continuation - Handle BYE instead of CANCEL from callers (issue 0004994) |
| ASTERISK-13284: [patch] Parking from AGI ignores SetMusicOnHold |
| ASTERISK-13285: test |
| ASTERISK-13286: [patch] alsa write EAGAIN is not handled |
| ASTERISK-13287: [patch] Possibility to disable to hold prompt with followme |
| ASTERISK-13288: Asterisk revision 166901 segfault during park using Aastra SPRE function to park extension |
| ASTERISK-13289: [patch] Auto-register realtime contexts |
| ASTERISK-13290: [patch] Set global context to be included in all other contexts |
| ASTERISK-13291: [patch] temporary files should be in temporary directory |
| ASTERISK-13292: reinvite while ringing |
| ASTERISK-13293: [patch] UK (BT) lines produce uncleared red alarm on TDM400P during line tests |
| ASTERISK-13294: Dial() option d is not working |
| ASTERISK-13295: [patch] Make enter/leave sounds on meetme conference customizable |
| ASTERISK-13296: [patch] CDR does not get produced with .call files |
| ASTERISK-13297: dahdi timing doesn't work on sip channel in non-answer mode. |
| ASTERISK-13298: [patch] functions with multiple arguments do not work in AEL dialplan |
| ASTERISK-13299: Asterisk dies once a day |
| ASTERISK-13300: [patch] changing the annoying "no one is answering queue" message level |
| ASTERISK-13301: [patch] Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. |
| ASTERISK-13302: Agent shows "(In use)" and will not receive queue calls while agent is logged in waiting for queue calls (1.4.22) |
| ASTERISK-13303: Asterisk crashes with core dump |
| ASTERISK-13304: Asterisk uses 100% of processor forever when I used the CLI during a load test |
| ASTERISK-13305: [patch] send out the incorrect register request URI to the (fromdomain) outbound proxy |
| ASTERISK-13306: [patch] ast_db_gettree(family, keytree) completely ignores the keytree argument |
| ASTERISK-13307: CLI hang and unresponsive when issuing "show channels" or "core show channels" |
| ASTERISK-13308: chan local show as invalid in app queue |
| ASTERISK-13309: sip show users does not exist |
| ASTERISK-13310: Security Vulnerability |
| ASTERISK-13311: make menuselect exits with install ncurses |
| ASTERISK-13312: On a fresh checkout of 1.4, 'make menuselect' fails |
| ASTERISK-13313: [patch] Setting registration expiry in registration string does not work |
| ASTERISK-13314: Can't make call if an added zero in the IP address of phone |
| ASTERISK-13315: The credentials you supplied were not correct or did not grant access to this resource. |
| ASTERISK-13316: [patch] add option to configure the SIP type in users.conf |
| ASTERISK-13317: segmentation fault in local_queue_frame at chan_local.c:172 |
| ASTERISK-13318: POST files are not truncated |
| ASTERISK-13319: Asterisk 1.6 restarts after completion of incoming fax |
| ASTERISK-13320: upgrading from asterisk-1.6.0.3 to branch/asterisk-1.6.1 breaks qualify for TCP peer |
| ASTERISK-13321: logger.c:531 rotate_file: system() failed for 'gzip -9 /var/log/asterisk/full.2': No child processes |
| ASTERISK-13322: Crash after - Remote peer reported an error, trying to establish the call anyway |
| ASTERISK-13323: [patch] Realtime peers are never qualified after 'sip reload' |
| ASTERISK-13324: [patch] Specifying a host port number in a registration string causes incorrect port in contact header |
| ASTERISK-13325: [patch] Not possible to register to a registrar via a host with different port number |
| ASTERISK-13326: [patch] Asterisk crashes while reading digits in AGI |
| ASTERISK-13327: do_monitor crash |
| ASTERISK-13328: load_modules errors |
| ASTERISK-13329: [patch] Asterisk ALSA interface consumes physical input and output regardless of whether or not they are needed... |
| ASTERISK-13330: [patch] res_phoneprov leaks memory if phoneprov.conf does not exist |
| ASTERISK-13331: Asterisk die after queue transfers |
| ASTERISK-13332: Unable to register SIP device with realtime in 1.6.x |
| ASTERISK-13333: Can't pickup using *8 after updating asterisk to 1.4.23-rc3 |
| ASTERISK-13334: Crash in ast_rtcp_read |
| ASTERISK-13335: iax2 trunked channels not being cleared |
| ASTERISK-13336: Channel not specified - Asterisk-1.6.0.3 |
| ASTERISK-13337: [patch] Asterisk 1.6.0.3-rc1 crashes sometimes |
| ASTERISK-13338: [patch] app_stack weak reference not set up properly for build on Mac OS X |
| ASTERISK-13339: Asterisk Crashes with signal 11 (segmentation fault) at random intervals (but at least 2 times a day) |
| ASTERISK-13340: g option does not work if used from Local channel |
| ASTERISK-13341: [patch] fromuser= doesn't work |
| ASTERISK-13342: Asterisk crashes anytime a call is parked by any method. |
| ASTERISK-13343: Random audio dropouts when jitterbuffer = yes |
| ASTERISK-13344: [patch] app_page causes undefined behavior when paging a page group with more than 128 extensions |
| ASTERISK-13345: [patch] Not possible to disguise display name on calls to trunks even though user can be disguised |
| ASTERISK-13346: [patch] core set verbose cores without arguments |
| ASTERISK-13347: SIP INVITE packets are incorrectly truncated with 1.6.1 svn after approx 1020 characters |
| ASTERISK-13348: [patch] DNS SRV messages inadvertently changed from verbosity >3 to 3 |
| ASTERISK-13349: [patch] Add config option to disable console connect messages |
| ASTERISK-13350: Gtalk/jingle fails with Empathy |
| ASTERISK-13351: [patch] fixes for autoconf 2.63 and ptlib-devel (Fedora 10) |
| ASTERISK-13352: manager.c poll error |
| ASTERISK-13353: crash in comparation with 'nothing' |
| ASTERISK-13354: queue-thankyou should be played only if needed |
| ASTERISK-13355: 1.4.22 crash with Park |
| ASTERISK-13356: [patch] Asterisk exits when trying to load cdr_addon_mysql.so |
| ASTERISK-13357: [patch] Calls are not accepted from an outbound proxy |
| ASTERISK-13358: [patch] Parsing and escaping of characters is broken in some cases (e.g. app_System) |
| ASTERISK-13359: port :0 added to SIP INVITE URI when 'outboundproxy' used in [general] section of sip.conf |
| ASTERISK-13360: [patch] Log and debug messages in ast_rtp_destroy can cause a crash |
| ASTERISK-13361: Crash on Start |
| ASTERISK-13362: handle_request_invite: Call from '101334' to extension 's' rejected because extension not found - FreeBSD sparc64 |
| ASTERISK-13363: Similar to 0013884 but different version - chan_iax2 flood with 'No private structure for packet' |
| ASTERISK-13364: 1.2.31 break authentication of IAX2 registration |
| ASTERISK-13365: [patch] 491-request pending is sent out of dialog |
| ASTERISK-13366: Broken mISDN pickup started after revision 164201 |
| ASTERISK-13367: [patch] h exten getting run at the wrong time |
| ASTERISK-13368: [patch] app_voicemail leaves sockets in close wait. |
| ASTERISK-13369: core dump on pri_schedule_event |
| ASTERISK-13370: No Audio on Call Transfer (Invite not being forwarded to Provider via Asterisk) |
| ASTERISK-13371: SQLGetData returns no data |
| ASTERISK-13372: [patch] SIPRemoveHeader() function to remove previously added headers |
| ASTERISK-13373: Cannot tell which registry item is which when username exceeds 12 characters |
| ASTERISK-13374: [patch] Add new cli "sip show registry xxx" |
| ASTERISK-13375: One way voice after attended transfer |
| ASTERISK-13376: [patch] Incoming calls from registrations and peer matching |
| ASTERISK-13377: Externalivr not sending 'H' event on channel hangup. |
| ASTERISK-13378: crash - mysql_store_result in pbx_realtime |
| ASTERISK-13379: [patch] MWI Subscription for Thomson ST2030S |
| ASTERISK-13380: [regression] Authentication seems to be broken again for SIP NOTIFY requests |
| ASTERISK-13381: [patch] SIP Channel name is not unique |
| ASTERISK-13382: [patch] to specify the location of astdb |
| ASTERISK-13383: app_fax: ReceiveFax fails with more than 1 page. |
| ASTERISK-13384: Asterisk crashes anytime in call queues |
| ASTERISK-13385: UpdateConfig not handling correctly sets of parameters |
| ASTERISK-13386: [patch] Global variables only allow values less than 255 characters |
| ASTERISK-13387: compilation warning for main/editline/history |
| ASTERISK-13388: Variable prefixed with '__' not inherited when call originated via AMI |
| ASTERISK-13389: [patch] Add the immediate=yes option to chan_iax2 |
| ASTERISK-13390: macro compatibility |
| ASTERISK-13391: Do SMS receiving and sending in PDU mode to allow use of localized messages (UCS-2BE for example) |
| ASTERISK-13392: sip show inuser dosplays negative |
| ASTERISK-13393: SIP Remote-Party-ID not fully parsed |
| ASTERISK-13394: Queue timeout default is wrong |
| ASTERISK-13395: gmake of menuconfig fails |
| ASTERISK-13396: Revision 169154 One Touch Park cannot be more than once per call |
| ASTERISK-13397: [patch] Group does not count all channels |
| ASTERISK-13398: [patch] allow storage of vmsecret in users voicemail spool directory |
| ASTERISK-13399: [patch] insufficient stringlength checking in action_userevent |
| ASTERISK-13400: [patch] Pressing only # when app_read is playing multiple prompts does not act as expected |
| ASTERISK-13401: Queue member location is altered |
| ASTERISK-13402: [patch] JSON Manager Event Interface |
| ASTERISK-13403: conference calling crashes Asterisk |
| ASTERISK-13404: Codec negotiation fails on calls from 1.2 -> 1.6, and is sub-optimum on calls from 1.6->1.6 |
| ASTERISK-13405: [patch] Asterisk retransmits the 401 response of failed REGISTER |
| ASTERISK-13406: Crash on failure to execute SQL |
| ASTERISK-13407: [patch] tcptls.c doesn't set correct remote_address |
| ASTERISK-13408: [patch] ooh323 segfault in libc-2.7.so |
| ASTERISK-13409: Users.conf info duplicated on advanced edit |
| ASTERISK-13410: [patch] CCBS/CCNR support for QSIG (libpri & chan_dahdi) |
| ASTERISK-13411: utils/refcounter segfaults due to reference of count1_obj when its NULL |
| ASTERISK-13412: crash in chan_sip function transmit_invite |
| ASTERISK-13413: [patch] Floating point exception crash when saying number between 9999 and 100000 with zh or tw language |
| ASTERISK-13414: Crash if SIP MESSAGE is more than 64 lines |
| ASTERISK-13415: [patch] X-Asterisk-Hangupcause header only in challenged BYEs |
| ASTERISK-13416: SIP on hold problems |
| ASTERISK-13417: chan_dahdi segfaulting (may be related to Bridge() application). |
| ASTERISK-13418: Inband DTMF not working between asterisk servers |
| ASTERISK-13419: [patch] per-mailbox imapfolder option |
| ASTERISK-13420: RTP delayed by 30 seconds when SIP calls is bridged via two LOCAL channel. |
| ASTERISK-13421: [patch] zoneinfo caching causes incorrect time |
| ASTERISK-13422: Access to encryption functions from dial plan |
| ASTERISK-13423: CDR record are posted twice |
| ASTERISK-13424: Test issue |
| ASTERISK-13425: ParkAndAnnounce loses "priority" of the return argument |
| ASTERISK-13426: pbx.c's show_dialplan_helper prints an "\r\n" at the end of every call |
| ASTERISK-13427: CDR not written when Busy() used |
| ASTERISK-13428: ExternalIVR sending 'I' on hangup despite 'i' option not set. |
| ASTERISK-13429: Paging application crashes asterisk |
| ASTERISK-13430: [patch] SIP/realtime problems => 100 % CPU |
| ASTERISK-13431: No voice (ringing tone) after call was diverted |
| ASTERISK-13432: channel.c: ast_indicate_data: Unable to handle indication -1 |
| ASTERISK-13433: pagerbody value is ignored and instead it contains emailbody value |
| ASTERISK-13434: won't answer pstn on TDM400P if MWI |
| ASTERISK-13435: After Dial's L() limit is reached, res_feature's dynamic features don't work |
| ASTERISK-13436: queue show command only show "Not in use" |
| ASTERISK-13437: [patch] Latest cdr_addon_mysql queries borked |
| ASTERISK-13438: CLI Freeze issue |
| ASTERISK-13439: [patch] describe idlecheck in res_odbc.conf.sample more clear |
| ASTERISK-13440: [patch] i18n.testsuite.conf in contribs directory uses old dialplan format an has no Chinese test |
| ASTERISK-13441: [patch] The contact exten field in the sip.conf register string is not parsed |
| ASTERISK-13442: MeetMe conference crashes Asterisk 95% of the time when the last user hangs up/exits the conference. |
| ASTERISK-13443: [patch] Config file layout |
| ASTERISK-13444: [patch] Support for specifying a ring timeout when calling a member interface |
| ASTERISK-13445: RTCP Read too short |
| ASTERISK-13446: Sound overlapping if Read() called again very soon |
| ASTERISK-13447: wrapuptime=0 in 1.6.0.1 and 1.6.0.3 |
| ASTERISK-13448: [patch] reg->username is parsed for each registration refresh rather than once on sip reload |
| ASTERISK-13449: Compile warnings in app_stack on OS/X |
| ASTERISK-13450: [patch] add option to configure locale for date/time string construction |
| ASTERISK-13451: Documentation for configuration of res_ldap |
| ASTERISK-13452: WaitExten doesn't stop background playback |
| ASTERISK-13453: SIP Call Problem, Hangup Message Too Large |
| ASTERISK-13454: set_member_paused ignores return-value from update_realtime_member_field |
| ASTERISK-13455: [patch] contrib/scripts/realtime_pgsql.sql misses uniqueid in queue_member_table |
| ASTERISK-13456: [patch] Calls are not matched to correct peer when using callbackextension parameter |
| ASTERISK-13457: [patch] delete file on hangup in app_record does not make sense |
| ASTERISK-13458: outboundproxy parameter in sip.conf is not atomic |
| ASTERISK-13459: Addons compile error |
| ASTERISK-13460: [patch] Outbound proxy not used for registrations |
| ASTERISK-13461: Friends do not appear in sip show peers cli |
| ASTERISK-13462: [patch] CANCEL gets different via header branch than INVITE |
| ASTERISK-13463: Transfer executes callers channel in wrong context |
| ASTERISK-13464: manager.c - error in action_originate resulting in "Channel not specified" error |
| ASTERISK-13465: [patch] Asterisk does not detect an attended transfer if the 'Replaces=' option is not at the beginning of the query string |
| ASTERISK-13466: [patch] res_odbc.c - ast_odbc_find_table() missing AST_RWLIST_UNLOCK() in error path |
| ASTERISK-13467: crash with asterisk realtime running odbc -> sql 2008 server via freetds when snom 190 reboots cached realtime |
| ASTERISK-13468: [patch] Adding JabberJoin and an argument to JabberSend for groupchats |
| ASTERISK-13469: [patch] RemoveQueueMember could remove realtime members too |
| ASTERISK-13470: [patch] Contact header missing port when non standard is used |
| ASTERISK-13471: [patch] Segfault if you transfer a call into a meetme room |
| ASTERISK-13472: SIP attended transfer cannot re-transfer a caller after a call forward no answer rule returns call to original extension |
| ASTERISK-13473: [patch] lockout after AEL reload |
| ASTERISK-13474: The status of a local channel in state_interface of a queue is wrong the first time is added and lost after a reload |
| ASTERISK-13475: muted doesn't compile on OS/X in dev-mode |
| ASTERISK-13476: Park position announcement broken with blind transfert. |
| ASTERISK-13477: [patch] Putting a comma in an extension dialpattern causes eventual seg fault |
| ASTERISK-13478: [patch] FILTER function is not working correctly (patch attached) |
| ASTERISK-13479: NULL file descriptors causing GUI to eventually stop functioning |
| ASTERISK-13480: [patch] Add audio announce option to app_page.c |
| ASTERISK-13481: [patch] Registration expiry not compatible with some ITSP |
| ASTERISK-13482: Description of registration string in sip.conf.sample is incorrect |
| ASTERISK-13483: Seg Fault when transferring calls from an IAX2 channel |
| ASTERISK-13484: [patch] CDR lost when Got SIP response 484 "Address Incomplete" |
| ASTERISK-13485: Timeout in Read command don't work when maxdigit > 0 |
| ASTERISK-13486: Segmentation Fault on "core stop gracefully" or "core restart gracefully" |
| ASTERISK-13487: [patch] Support setting emailsubject and emailbody per voicemailbox with realtime |
| ASTERISK-13488: Revision 172517 segfault after using A *2 transfer to B and B dial *2 |
| ASTERISK-13489: autopause should not pause interfaces that are busy |
| ASTERISK-13490: MAILBOX_EXISTS crashes Asterisk when called with empty argument |
| ASTERISK-13491: AMI shows utils.c:1198 ast_careful_fwrite: fflush() returned error: Broken pipe |
| ASTERISK-13492: set talker optimization has no OFF function |
| ASTERISK-13493: Modifying wrapup from dialplan or manager - Let a queue member get out of wrapup early |
| ASTERISK-13494: priexclusive parameter ignored if pri = pri_cpe ? |
| ASTERISK-13495: Background leaves files open indefinately |
| ASTERISK-13496: [patch] Unhold fails if first SDP on OK, particularly Cisco CCM 6 |
| ASTERISK-13497: [patch] 1.2.31.1 changes create storm of IAX2 register authentication retries |
| ASTERISK-13498: Sending and receiving a fax between extensions on the same Asterisk machine fails |
| ASTERISK-13499: mpg123 crashes when trying to play stream. |
| ASTERISK-13500: "dialplan show globals" does not show the correct TRUNKMSD setting |
| ASTERISK-13501: [patch] Bridging patches |
| ASTERISK-13502: Asterisk login issue due to, unavailability of handling unicode status messages in agi_handle_presence method of res_jabber.c |
| ASTERISK-13503: DEVICE_STATE is not working for SIP channels after 1.6.0.5 |
| ASTERISK-13504: Addons will not build on 1.6.1 rc-1 or 1.6.0 svn |
| ASTERISK-13505: wrong Via-branch-id in CANCEL and ACK |
| ASTERISK-13506: Menuselect compiles X86 code on armv5tel |
| ASTERISK-13507: Crash - maybe in filestream_destructor |
| ASTERISK-13508: pbx crash |
| ASTERISK-13509: Calls coming in then out do not get recorded in CDR |
| ASTERISK-13510: wrong call-limit count when counteronpeer=yes |
| ASTERISK-13511: [patch] ami fails on high load |
| ASTERISK-13512: dead SIP channels get not cleared though in status SIP_NEEDDESTROY thus are blocking ports |
| ASTERISK-13513: app_playback does not seem to be closing audio files. |
| ASTERISK-13514: [patch] XML Documentation for res_jabber |
| ASTERISK-13515: [patch] Voicemail message recording file is shorter than duration reported in msg????.txt |
| ASTERISK-13516: hold music restarts with each command |
| ASTERISK-13517: ODBC Crash |
| ASTERISK-13518: One audio stream not working after transfer |
| ASTERISK-13519: calling Gosub in macro does not work if the invalid extension is triggered |
| ASTERISK-13520: [patch] 1.6.1 Choppy sound in Dual Xeon 2.8 GHz |
| ASTERISK-13521: state of non existent extensions should not be "Idle" |
| ASTERISK-13522: 1.6.1-rc1 Program terminated with signal 11, Segmentation fault. audiohook.c AST_LIST_TRAVERSE_SAFE_BEGIN |
| ASTERISK-13523: Asterisk app_fax compilation fails in svn 1.6.1 branch |
| ASTERISK-13524: [patch] streamed moh breaks if nobody listen it |
| ASTERISK-13525: Asterisk crash with looped request and pedantic=yes |
| ASTERISK-13526: [patch] If a SIP URI is resolved with SRV records, the port must no be in the Request-URI |
| ASTERISK-13527: [patch] Asterisk must not perform SRV lookups if a port is specified in the URI |
| ASTERISK-13528: Issue with Asterisk 1.6.1-rc1 with Microsoft OCS 2007 |
| ASTERISK-13529: Issue with Initial call setup when Asterisk 1.6.1-rc1 integrated with Microsoft OCS 2007 |
| ASTERISK-13530: Wrong vocalized digits in pt_BR |
| ASTERISK-13531: [patch] Caller ID is missing for h323 calls. |
| ASTERISK-13532: app_chanisavail always set AVAILSTATUS to 0 with option 's' set |
| ASTERISK-13533: [patch] Allow manager command to see if IAX link is trunked or encrypted |
| ASTERISK-13534: One-touch parking from a queue not working |
| ASTERISK-13535: Can't record greetings |
| ASTERISK-13536: Bad branch parameter value in CANCEL request |
| ASTERISK-13537: automon stops working when call retrived from parking lot |
| ASTERISK-13538: [patch] Automatic gain normalization in meetme |
| ASTERISK-13539: [patch] Dahdi does not wait for wink on outbound calls before dialing DTMF with Signalling type = em_w |
| ASTERISK-13540: Asterisk 1.6.1-rc1 compile bomb in apps/app_rpt - undefined AST_PBX_KEEPALIVE at line #13622 |
| ASTERISK-13541: Asterisk crashes in ast_channels_free with "free(): invalid pointer" |
| ASTERISK-13542: crash when setting an incoming call via SIP |
| ASTERISK-13543: [patch] registration query |
| ASTERISK-13544: codec_g729 crashes system |
| ASTERISK-13545: Using the Parking Lot feature randomly causes a segfault |
| ASTERISK-13546: DTMF are collected but nothing happen |
| ASTERISK-13547: cdr_radius duplicate accounting packets. |
| ASTERISK-13548: M() Docs |
| ASTERISK-13549: Random inbound calls do not hear any sound but CLI says file is playing |
| ASTERISK-13550: [patch] chan_sip does not support the maddr attribute in Via headers |
| ASTERISK-13551: Type uninitialized |
| ASTERISK-13552: [patch] chan_sip fails to remove hold when receving a reINVITE without SDP |
| ASTERISK-13553: que don't pass call to agents |
| ASTERISK-13554: app_dial with the g flag doesn't continue on to the next priority |
| ASTERISK-13555: in "_sip_tcp_helper_thread" Buffer is filled with dirty bytes |
| ASTERISK-13556: slow ODBC database completely LOCK chan_sip |
| ASTERISK-13557: [patch] AMI Redirect hangs up channel if it is in Background() |
| ASTERISK-13558: spy-dahdi sound file is missing |
| ASTERISK-13559: Crash after run on FreeBSD 7.1 |
| ASTERISK-13560: Segfault on call termination when attempting to retransmit a packet that should have not been retried due to network issues |
| ASTERISK-13561: Segfault in chan_iax.so |
| ASTERISK-13562: Blind transfer uses the wrong context when blind transffering, combined with 'h' extension. |
| ASTERISK-13563: jitterbuffer delays after a meetme finished |
| ASTERISK-13564: Asterisk plays a continuous tone forever if it never receives a 2833 end packet |
| ASTERISK-13565: ERROR[5003]: channel.c:2043 __ast_read: ast_read() called with no recorded file descriptor. |
| ASTERISK-13566: [patch] func_devstate not updating Custom hints, and not in sync with ASTDB |
| ASTERISK-13567: Asterisk fault with general protection error |
| ASTERISK-13568: MixMonitor stops recording after announcement with Dial option A |
| ASTERISK-13569: [patch] lock during simple call processing |
| ASTERISK-13570: [patch] Incorrect From: header information when CALLERPRES=PRES_PROHIB |
| ASTERISK-13571: Asterisk fault with segfault |
| ASTERISK-13572: Deadlock between (MOH) ast_write on one end and local_hangup on other end (p->lock versus chan->lock_dont_use) |
| ASTERISK-13573: [patch] adding 2 new events for when a channel spy is started or stopped |
| ASTERISK-13574: SIP calls being allowed without valid username/password |
| ASTERISK-13575: [patch] Making outgoing calls DTMF can't be detected |
| ASTERISK-13576: asterisk segfault in AST_LIST_TRAVERSE_SAFE_BEGIN |
| ASTERISK-13577: [patch] Crash in VoiceMailMain if hangup occurs before a valid mailbox number is entered (IMAP only) |
| ASTERISK-13578: Pickup from features |
| ASTERISK-13579: [patch] channels created by app_dial do not inherit the owner's language |
| ASTERISK-13580: CDR not recording in DB even if no error persists |
| ASTERISK-13581: pseudo channel disappears after dahdi restart |
| ASTERISK-13582: Blind transfer to parking from SIP phone crashes Asterisk |
| ASTERISK-13583: "iax2 prune realtime" doesn't prune user, only peer |
| ASTERISK-13584: SIP/tcp calls failing with invalid transport error. |
| ASTERISK-13585: ODBC-based function crashes Asterisk |
| ASTERISK-13586: [patch] FreeBSD: set nonblocking mode on /dev/dahdi/pseudo failed. |
| ASTERISK-13587: crash when calling macro with empty argument |
| ASTERISK-13588: "Channel not specified" in asterisk 1.6.0.5 |
| ASTERISK-13589: CURL() function crashes in /trunk |
| ASTERISK-13590: Audio is not synchronized and the quality of call is bad |
| ASTERISK-13591: [patch] SIPNotify dialplan application |
| ASTERISK-13592: [patch] Add support for loading multiple timing modules |
| ASTERISK-13593: Timing interfaces provided in 1.6.1 and beyond are not documented. |
| ASTERISK-13594: [patch] DTMF blips at end of recordings in __ast_play_and_record() |
| ASTERISK-13595: Asterisk Crashes when making calls to app_voicemail... |
| ASTERISK-13596: alwaysauthreject option in sip.conf should default to yes |
| ASTERISK-13597: IMAP segfault under load or multiple vmail users |
| ASTERISK-13598: [patch] Enforce password strengths |
| ASTERISK-13599: [patch] SIP Realtime not reading database for changes to realtime peers after initial registration |
| ASTERISK-13600: [patch] IMAP crash multiple callers / callers hangup at beep |
| ASTERISK-13601: [patch] Wrong order of Channeltype and Uniqueid sent to manager from chan_sip.c |
| ASTERISK-13602: String operator ':' error with UTF8 code |
| ASTERISK-13603: Crash when a agent is connected with a customer |
| ASTERISK-13604: [patch] Addition of DAHDI application to call libpri functions for MWI via ISDN/CISC. |
| ASTERISK-13605: [patch] cleanup bridging from shared variables |
| ASTERISK-13606: [patch] Allow app_dial to provide 'tone while ringing' like 'option m' which will provide 'music on hold while dialling' |
| ASTERISK-13607: Resolve remaining issues left over from 'kill-the-user' |
| ASTERISK-13608: QUEUE_VARIABLES function doesn't work as it should |
| ASTERISK-13609: [patch] Usage of IMAP mailboxes still cause asterisk to crash, even after 0013653 committed patch |
| ASTERISK-13610: [patch] users.conf (and other .conf files) have incorrect whitespacing |
| ASTERISK-13611: SIP REINVITE broken in 1.6 (was working in 1.4.13) |
| ASTERISK-13612: [patch] Compilation issues due to zone_tones update |
| ASTERISK-13613: [patch] Do not define sysinfo_show command if we do not have sysctl/sysinfo |
| ASTERISK-13614: [patch] SIP Forking Feature |
| ASTERISK-13615: [patch]Timeout settings in features.conf don't work as intended |
| ASTERISK-13616: Can not make install Asterisk |
| ASTERISK-13617: [patch] signals.h - syntax error before '*' token |
| ASTERISK-13618: Asterisk stop responding after unsuccesfull registration |
| ASTERISK-13619: Rtp socket are not closed after Hangup |
| ASTERISK-13620: Subscriptions to hints on another server only allows 1 watcher at a time |
| ASTERISK-13621: [patch] Add a couple of manager commands to list skinny devices and lines |
| ASTERISK-13622: sip channels stay open |
| ASTERISK-13623: iax2 show peers show encryption on all peers which is not true |
| ASTERISK-13624: Lost SMS |
| ASTERISK-13625: DIALEDTIME and TIMEOUT(absolute) in attended transfer doesn't work as expected |
| ASTERISK-13626: Queue members associated with ZOMBIE channels (breaks wrapuptime and queue_log). |
| ASTERISK-13627: chan_dahdi loses dialtone after some time after reboot |
| ASTERISK-13628: [patch] asterisk crashes during attended transfer |
| ASTERISK-13629: Debian Asterisk 1.4.21.2~dfsg-3 + issue with voicemail permissions |
| ASTERISK-13630: MWI subscriptions does not works if there is no HINT for extension |
| ASTERISK-13631: iLBC transcoding times are always zero |
| ASTERISK-13632: H extension doesn't execute inside Macros in 1.4.23.1 |
| ASTERISK-13633: Chan Alsa reports that the audio device is busy when using Pulseaudio |
| ASTERISK-13634: problems building asterisk 1.6.0.6 |
| ASTERISK-13635: [patch] After a caller is processed by app_queue the queue_log logs the hangup as TRANSFER |
| ASTERISK-13636: M() ignored in Dial |
| ASTERISK-13637: [patch] astcanary does not exit when asterisk dies |
| ASTERISK-13638: Missing one way audio on one phone after the bridge |
| ASTERISK-13639: #exec command needs minor documentation in extensions.conf |
| ASTERISK-13640: [patch] crash during AGI call (regression, it works under 1.6.0.1) |
| ASTERISK-13641: [patch] #exec lines causing failure of parsing of extensions.conf |
| ASTERISK-13642: [patch]AEL parser broken in 1.4 branch |
| ASTERISK-13643: Paging app does not page last extension in list |
| ASTERISK-13644: [patch] MEETME_RECORDINGFILE is not read by MeetMe() |
| ASTERISK-13645: [patch] Patch to improve NAT handling for Polycoms behind proxy |
| ASTERISK-13646: [patch] Resending CANCEL doesn't stop |
| ASTERISK-13647: Can't build db_dump185 using main/db1-ast/Makefile as db_dump.c is missing |
| ASTERISK-13648: [bulid] app_fax do not compile with spandsp-0.0.6pre4 |
| ASTERISK-13649: Rtp socket are not closed after Hangup |
| ASTERISK-13650: [patch] meetme - conference stays active even if marked user exits |
| ASTERISK-13651: 1.4.23.1 Segmentation Fault in channel.c |
| ASTERISK-13652: [patch] # for fastforward goes beyond end of message |
| ASTERISK-13653: When i park a call after the slot announcement the call is not hangup |
| ASTERISK-13654: core restart now crashed asterisk |
| ASTERISK-13655: [patch] app_playtones segfaults asterisk when invalid tone is specified. |
| ASTERISK-13656: [patch] Queue setting to disregard penalty on few queue members |
| ASTERISK-13657: [patch] Ability to use DUNDi channel variables when using dynamic weights |
| ASTERISK-13658: When using SMDI Asterisk crashes after message is left |
| ASTERISK-13659: [patch] safe_asterisk can get multiple instances if killproc escalates to SIGKILL in service asterisk restart |
| ASTERISK-13660: Segmentation fault caused by sqlite3_log |
| ASTERISK-13661: [patch] Option C in Queue() app crashes Asterisk |
| ASTERISK-13662: Asterisk segfaults when parking call |
| ASTERISK-13663: Wrong pathname for SIREN sounds in TRUNK configurations |
| ASTERISK-13664: [patch] fix channelvariables documentation |
| ASTERISK-13665: Directed call pickup does not work using subscription based pickup |
| ASTERISK-13666: AMD hangs, leaving orphaned channel |
| ASTERISK-13667: AGI GOSUB command causes crash if no Return in routine. |
| ASTERISK-13668: one way voice on incoming isdn calls |
| ASTERISK-13669: [patch] 'autodomain' doesn't work |
| ASTERISK-13670: [patch] Audiohook volume does not honor the write adjustment when both is specified |
| ASTERISK-13671: BYE to 408 Request Timeout |
| ASTERISK-13672: ENUMLOOKUP - broken regex. |
| ASTERISK-13673: [patch] FXO channels "hookstate" incorrect on startup |
| ASTERISK-13674: [patch] Generic Speech API Engine - Sphinx |
| ASTERISK-13675: say_alpha() is not working |
| ASTERISK-13676: [patch] unloading chan_iax2.so segfaults asterisk |
| ASTERISK-13677: playback and set language issue |
| ASTERISK-13678: Getting SSL cipher error with Asterisk-1.6.1-rc1 version |
| ASTERISK-13679: [patch] Asterisk does not stop retransmission |
| ASTERISK-13680: [patch] meetme doesn't play conf-has left prompts |
| ASTERISK-13681: chan_iax2.c:7410 socket_process: Packet Decrypt Failed! |
| ASTERISK-13682: [patch] updatecdr='yes' in agents.conf is not working |
| ASTERISK-13683: urgent help....need urgent help on say string task.. |
| ASTERISK-13684: [patch] export the SIP peer username of the transferer |
| ASTERISK-13685: [patch] UserEvent Duplicate Previous Information |
| ASTERISK-13686: [patch] Add option s to chanspy |
| ASTERISK-13687: [patch] Added new event class bridge |
| ASTERISK-13688: [patch] Provide timestamp with pong response |
| ASTERISK-13689: greetings can not be retrieved from IMAP |
| ASTERISK-13690: asterisk crashes when voicemail app finish to write a message |
| ASTERISK-13691: searchcontexts=yes causes voicemail boxes to be setup wrong |
| ASTERISK-13692: [patch] "make config " errors, zero length byte file create /etc/init.d.asterisk |
| ASTERISK-13693: [patch] Duplication of code for dial_exec_options 'option k' and 'option K' |
| ASTERISK-13694: [patch] Support additonal FXS module alerting methods before FSK spill |
| ASTERISK-13695: reopen issue 10984 |
| ASTERISK-13696: HFC-s card cannot initiate outgoing calls |
| ASTERISK-13697: chan_iax2.c: Packet Decrypt Failed! encrypted IAX2 during packet loss causes hangup and end of call |
| ASTERISK-13698: Goto(s-${DIALSTATUS},1) for status ANSWER incorrect |
| ASTERISK-13699: Set(ARRAY(var1,var2)=1,2) not working |
| ASTERISK-13700: Console flooded with warnings |
| ASTERISK-13701: [patch] SIP Attended Transfer fails |
| ASTERISK-13702: Create New feature for callcenter solution |
| ASTERISK-13703: [patch] func_odbc's OPT_ESCAPECOMMA's is undone by second ast_app_separate_args call when using Set(ARRAY...) |
| ASTERISK-13704: exten => ANSWER not found when extenpatternmatchnew=yes |
| ASTERISK-13705: [patch] sip channel freezed in ChanSpy() app |
| ASTERISK-13706: Asterisk crashes when Dial() to a sip channel terminates |
| ASTERISK-13707: app_fax not compiling |
| ASTERISK-13708: [patch] extend / rewrite volume function and audiohook |
| ASTERISK-13709: Crashing Asterisk, make simple |
| ASTERISK-13710: [patch] Race condition between bridge and channel masquerading |
| ASTERISK-13711: 1.4.23.1 DeadAGI() in "h" extension |
| ASTERISK-13712: Random deadlocks leading to lockup on 1.4.23.1 |
| ASTERISK-13713: SIP dial ignores destination port |
| ASTERISK-13714: Asterisk says: "No DAHDI Transcoders found.", but dahdi_transcode module reports 1 transcoder |
| ASTERISK-13715: "SIP/2.0 404 Not found" when attended transferring a private number |
| ASTERISK-13716: [patch] Add support in AEL for macro return values and direct assignment of them to variables and functions. |
| ASTERISK-13717: [patch] Ghost calls with queues and spa942 and 922 |
| ASTERISK-13718: IAX2 doesn't issue HANGUP when using DIAL ring |
| ASTERISK-13719: "ael reload" crashes Asterisk every time |
| ASTERISK-13720: [patch] Changes to manager interface for registering event hooks |
| ASTERISK-13721: memory leak in "strings.c" |
| ASTERISK-13722: Bogus values in chan_dahdi.conf are not signalled in console |
| ASTERISK-13723: Asterisk CPU usage 100% (deadlock?) when using Queue() with Local channels |
| ASTERISK-13724: [patch] Patch: seg fault in chan_local when other->generator is used without checking that other is not null |
| ASTERISK-13725: Attended transfer is not working correctly |
| ASTERISK-13726: Segfaults if queue name is wrong |
| ASTERISK-13727: r236981 Attended transfer fails to complete if A calls B, B uses Asterisk atxfer to C, and C picks up before B hangs up |
| ASTERISK-13728: [patch] Asterisk should transform SIP 503 code to SIP 500 |
| ASTERISK-13729: LOCALSTATEDIR default doesn't include PREFIX as documented by ./configure --help |
| ASTERISK-13730: The default sip.conf file contains bogus CLI command tip |
| ASTERISK-13731: Incorrect argument parsing in RetryDial causes asterisk to crash |
| ASTERISK-13732: Asterisk becomes unresponsive and seems to be locked waiting on a mutex |
| ASTERISK-13733: SRTP fail before a complete Answer is done |
| ASTERISK-13734: leavewhenempty=yes doesn't work as expected, operates in reverse. |
| ASTERISK-13735: [patch] early-dial SIP 484 "incomplete address" |
| ASTERISK-13736: Never reply 503, use 500 instead (don't break DNS SRV failover) |
| ASTERISK-13737: [patch] Wrong text for HELP DAHDI |
| ASTERISK-13738: [patch] unfreed memory in Local channel |
| ASTERISK-13739: [patch] Callee cannot use dynamic features |
| ASTERISK-13740: Asterisk crashes when trying to leave a message via advanced mailbox options |
| ASTERISK-13741: [patch] MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port (5060) |
| ASTERISK-13742: [patch] Lost info in CDR when transfering call via AMI's Redirect |
| ASTERISK-13743: MOH Realtime crash |
| ASTERISK-13744: [patch] SetVar from Cli or AMI not showing value on CDR Record |
| ASTERISK-13745: [patch] Adding two variables for dynamic features (applicationmap) |
| ASTERISK-13746: Asterisk crash investigation |
| ASTERISK-13747: Hangup causecode is improper in case ring/dial timeout specified. |
| ASTERISK-13748: waitExten(|d) does not play any dialtone |
| ASTERISK-13749: 1.6.1.0-rc2: dmsmgr can't resolve addresses |
| ASTERISK-13750: register: '/' in username not supported |
| ASTERISK-13751: [patch] pbx_config writes a warning but 'dontwarn = yes' is activated in asterisk.conf |
| ASTERISK-13752: FILE function reads 1 character less than specified in length |
| ASTERISK-13753: [patch] NET-SNMP configure test fails; libraries passed in LDFLAGS instead of LIBS |
| ASTERISK-13754: Incorrect calling of free() at alloc_queue() in app_queue.c |
| ASTERISK-13755: [patch] Missing mute facility |
| ASTERISK-13756: [patch] Dial application with the 'n' option not removing introductions |
| ASTERISK-13757: crash and core dump |
| ASTERISK-13758: unfreed memory in try_calling |
| ASTERISK-13759: AUDIOHOOK_INHERIT |
| ASTERISK-13760: [patch] Race condition in ast_db_get() |
| ASTERISK-13761: Hangup cause 18 (no user response) is an end user condition and should not be used for protocol timeouts |
| ASTERISK-13762: Hangup cause 20 (subscriber absent), clearly an end user condition, is being used for unregistered trunks |
| ASTERISK-13763: When using IMAP voicemail storage, you cannot retrieve messages by logging into VoicemailMain() |
| ASTERISK-13764: Regression: #13867 Reject an incoming call to peer due to call limit with "603 Declined". It`s not correct. |
| ASTERISK-13765: Asterisk does not say the digits of the parked call position. |
| ASTERISK-13766: Crash in operation |
| ASTERISK-13767: Asterisk Crashes when typing 'remove extension' and using the tab key in CLI |
| ASTERISK-13768: music on hold outputs no sound on incoming calls (using sip) but it works when the call is put hold |
| ASTERISK-13769: [patch] ENUMQUERY does not differentiate non-existant domain vs. no DNS records |
| ASTERISK-13770: [patch] ISDN-Transfer causes backcall attempt of attendent phone |
| ASTERISK-13771: Codec negotiation issue when codecs not defined in [general] section of sip.conf |
| ASTERISK-13772: No ringback tone (from Cisco to Asterisk Extension) |
| ASTERISK-13773: [patch] Add Hangupcause to manager action Hangup |
| ASTERISK-13774: reload in console overwrites priindication=outofband setting |
| ASTERISK-13775: [patch] PrivacyManager generates a core dump when testing |
| ASTERISK-13776: race condition in res/timing_* interfaces |
| ASTERISK-13777: Long Page() arguments crash Asterisk |
| ASTERISK-13778: call that is from sap3000 to mobile causes kernel panic |
| ASTERISK-13779: CallerID lost some digit |
| ASTERISK-13780: [patch] missing DESTDIR in addons install-xmldoc |
| ASTERISK-13781: [patch] debugging CID matching |
| ASTERISK-13782: sip call setup bug |
| ASTERISK-13783: Using | symbol instead of ','s in call file causes segfault when using RetryDial |
| ASTERISK-13784: [patch] Deadlock when manipulating module_list over AMI and CLI |
| ASTERISK-13785: [patch] A clean svn checkout needs bison to compile but configure does not check for it |
| ASTERISK-13786: No audio from Gtalk to Asterisk |
| ASTERISK-13787: CID matching is wrong |
| ASTERISK-13788: [patch] chan_local generates MoH instead of just passing HOLD/UNHOLD further |
| ASTERISK-13789: error/warnings for difficult if() statement in AEL |
| ASTERISK-13790: directrpsetup=yes does not work when canreinvite=n |
| ASTERISK-13791: [patch] Backport floating timeout in Read() |
| ASTERISK-13792: Asterisk randomly crashes while playing moh |
| ASTERISK-13793: Compilation Errors on Mac OS X 1.5.6 - Asterisk 1.6.2.0-beta1 ( also confirmed on trunk ) |
| ASTERISK-13794: [patch] Simplify h323 Make process |
| ASTERISK-13795: Crash using func_odbc |
| ASTERISK-13796: [patch] chan_iax2 reports endless if a peer cannot be registered (>100 logs/sec) |
| ASTERISK-13797: [patch] relax badshell tilde test |
| ASTERISK-13798: [patch] when userid is set in the user profile, the userid used by the remote device is lost |
| ASTERISK-13799: make install can't install sounds file on Mac OS X 10.5 |
| ASTERISK-13800: Asterisk just crash |
| ASTERISK-13801: Unable to get the hook status as offhook for line connected modules on TDM 400P card |
| ASTERISK-13802: [patch] dahdiras.c will not compile |
| ASTERISK-13803: Asterisk doesn't add Route headers in NOTIFY when the SUBSCRIBE came from a proxy |
| ASTERISK-13804: Chan Local Fails to bridge from originate |
| ASTERISK-13805: Conditional compilation of a diagnostic message needs an L modifier to %d for a 64 bit integer |
| ASTERISK-13806: [patch] lock or crash after changing sip 'transport' |
| ASTERISK-13807: [patch] global mohinterpret setting is ignored |
| ASTERISK-13808: [patch] Added ability to specify different SIP and media addresses |
| ASTERISK-13809: [patch] Fix runlevels in Debian rc files |
| ASTERISK-13810: Unable to dial '0' to talk to reach an operator while in voice mail |
| ASTERISK-13811: Asterisk using 100% of cpu on 64 bit ubuntu |
| ASTERISK-13812: AUDIOHOOK_INHERIT crash after sip attended transfer |
| ASTERISK-13813: Detection of call pickup code in chan_dahdi should have higher priority than dialplan matches |
| ASTERISK-13814: [patch] message "you have no messages" garbled |
| ASTERISK-13815: [patch] Cannot flavour (flavor) version number because make_version_c looks in wrong directory |
| ASTERISK-13816: Address device state performance issues in 1.6.1 |
| ASTERISK-13817: [patch] Voicemail(ARGS) is limtted to 1024 characters, large 'blast' groups are silently left off |
| ASTERISK-13818: [patch] autoplay option for VoiceMailMain generates warning about invalid value |
| ASTERISK-13819: Timed out parked calls always return to originating extension |
| ASTERISK-13820: Asterisk make failed: poll-compat.h:97: error: redefinition of ‘struct pollfd’ |
| ASTERISK-13821: Is it possible to increase the number of groups above 64 for pickup etc |
| ASTERISK-13822: ForkCDR creates 3 CDR's |
| ASTERISK-13823: Asterisk crashes on incoming call that is hung up before answered. |
| ASTERISK-13824: segfault following httpd_helper_thread -> generic_http_callback -> ast_str_append |
| ASTERISK-13825: Coredump of asterisk in ramdom time |
| ASTERISK-13826: [patch] SQL Error makes res_odbc reconnect to odbc dsn |
| ASTERISK-13827: [patch] Add support for relative path to digits,letters and phonetics files in say.c |
| ASTERISK-13828: Asterisk allowed access by anonymous SIP user |
| ASTERISK-13829: Chan_Sip route to an incorrect DID |
| ASTERISK-13830: Crashes when executing a Stored Procedure against Sybase ASE 1.5.03 |
| ASTERISK-13831: [patch] Realtime bad Reconstruct of field 'fullcontact' after restart |
| ASTERISK-13832: mixmonitor crash |
| ASTERISK-13833: [patch] Realtime - incorrect set peer ip-address and port after restart |
| ASTERISK-13834: [patch] 'h' extension not being executed, if caller hangs up before call is answered |
| ASTERISK-13835: app_followme doesn't initialize targs |
| ASTERISK-13836: [patch] Asterisk commands "moh reload" or "reload res_musiconhold.so" causes MOH not to work properly |
| ASTERISK-13837: [patch] chan_alsa.so fails to load with no reason given in log |
| ASTERISK-13838: [patch] 1.6.1: unidirectional PCM if (FXS?) hardware DTMF detection enabled |
| ASTERISK-13839: Asterisk exits randomly when on Originate command |
| ASTERISK-13840: [patch] On Dial with Macro, re-INVITE to the caller happens upon callee answer, and not when Macro |
| ASTERISK-13841: Asterisk crashes randomly, with Exchange 2007 |
| ASTERISK-13842: [patch] Typo in app_ices description. |
| ASTERISK-13843: [patch] Typo in app_ices description. |
| ASTERISK-13844: [patch] Typo in app_ices description. |
| ASTERISK-13845: TLS Client Hello handshake sent within SSLv2 header and not TLS header |
| ASTERISK-13846: [patch] Improvements/fixes for app_fax |
| ASTERISK-13847: Need ability to select TLS version in outgoing messages |
| ASTERISK-13848: [branch] Asterisk Calendar Integration |
| ASTERISK-13849: Crash in SQL routine from call in ast_odbc_prepare_and_execute |
| ASTERISK-13850: Hold/Unhold does Link/Unlink/Link |
| ASTERISK-13851: Compile errors |
| ASTERISK-13852: [patch] Added ability to perform SRV lookups for AGI URIs |
| ASTERISK-13853: [Help] no voice after call get connected in g729 codec |
| ASTERISK-13854: No RTP ports remaining. Can't setup media stream for this call. |
| ASTERISK-13855: Can't forward voicemail when using IMAP storage |
| ASTERISK-13856: Asterisk abort (signal 6) in local_pvt_destroy at chan_local.c:159 |
| ASTERISK-13857: Initializing Samsung GT-C6620 |
| ASTERISK-13858: chan_dahdi.c: Asked to delete sched id -1212150824??? |
| ASTERISK-13859: [patch] segfault in ast_cdr_start() at cdr.c |
| ASTERISK-13860: crash after native bridging |
| ASTERISK-13861: [invalid] 'make distclean' leaves channels/h323 unclean |
| ASTERISK-13862: Asterisk crash; stems from chan_agent |
| ASTERISK-13863: Not possible to make two transfer in the same call |
| ASTERISK-13864: [patch] add possibility to read |
| ASTERISK-13865: SIP channels never close when using tcp or tls |
| ASTERISK-13866: chan_h323 build fails with gcc 3 |
| ASTERISK-13867: Asterisk crashed |
| ASTERISK-13868: Asterisk shouldn't send dialog NOTIFY <state>early</state> until it receives a 1XX-non100 response from the callee |
| ASTERISK-13869: unfreed memory in ast_frdup() in frame.c |
| ASTERISK-13870: Asterisk crashes when empty member in queues.conf |
| ASTERISK-13871: [patch] Make. Parallel build on multi CPU host |
| ASTERISK-13872: Adaptive jitter buffer and Zoiper DTMF problem |
| ASTERISK-13873: Getting the wrong peer when INVITE |
| ASTERISK-13874: 183 response although progressinband=never when routed to PRI |
| ASTERISK-13875: app_forkcdr - mispelled description and sometimes sentences does not make sens |
| ASTERISK-13876: Cause code for INVITE timeout doesn't match intent of code and semi-randomly chooses between DIALSTATUS of BUSY and CONGESTION |
| ASTERISK-13877: Crash with DUNDi |
| ASTERISK-13878: Users defined in users.conf can not use feature codes defined in features.conf |
| ASTERISK-13879: SendFax and T38 issues |
| ASTERISK-13880: res_pktccops does not compile on Darwin (MacOS 10.5) |
| ASTERISK-13881: Attended feature transfers are broken in latest 1.6.0 SVN |
| ASTERISK-13882: Asterisk crashes when received faxes use IAX2 channels. |
| ASTERISK-13883: [patch] channel-specific hangupcauses |
| ASTERISK-13884: DeadAGI at h extension |
| ASTERISK-13885: ast_read() used with incorrect ast_waitfor() in app_fax.c |
| ASTERISK-13886: Asterisk Crashes when conference bridge feature "user join/leave" is enabled |
| ASTERISK-13887: [patch] DTMF Appears to be broken from certain sources on asterisk 1.4.24 - double digit. |
| ASTERISK-13888: [patch] "setvar" configuration option for mgcp |
| ASTERISK-13889: Chan_sip core dump on program execution |
| ASTERISK-13890: inotify header file prevents Asterisk from compiling |
| ASTERISK-13891: Ver 1.6.0.8 Seg Fault |
| ASTERISK-13892: [patch] Allow app_mp3 to play m3u playlist file |
| ASTERISK-13893: [patch] http Queue Actions Crash Asterisk |
| ASTERISK-13894: barev |
| ASTERISK-13895: FOR ALL VERSIONS: Add for all events, always a context param |
| ASTERISK-13896: early media playback doesn't work |
| ASTERISK-13897: Asterisk generates Ring instead of Coloring Ring Back Tone (Early Media). |
| ASTERISK-13898: Mixmonitor/Monitor stops when automated dial out call files used with local channel |
| ASTERISK-13899: Chan_mobile problem with initilizing L6 motorola phone |
| ASTERISK-13900: Issue starting Asterisk with LinuxHA |
| ASTERISK-13901: [patch] memory leak in some part of code |
| ASTERISK-13902: Asterisk crashes on blind transfer to park extension |
| ASTERISK-13903: Random Coredumps with IAX2 load |
| ASTERISK-13904: [patch] Add rtsavesysname to chan_iax |
| ASTERISK-13905: [patch] Asterisk cannot process calls if it cannot reach the voicemail database |
| ASTERISK-13906: Asterisk crashes on directed Pickup. |
| ASTERISK-13907: Outbound Invite has the from field and Remote-Party-ID bad |
| ASTERISK-13908: [patch] Typo on format wav and wav_gsm ... must read frequency instead of freqency |
| ASTERISK-13909: 1 in 3 incoming zap PRI calls do no hear audio (are not bridged) when call is answered with agi script |
| ASTERISK-13910: Asterisk call file has CDR always set to NO ANSWER |
| ASTERISK-13911: asterisk does not play warning file when have SIP-SIP Packet2Packet bridging |
| ASTERISK-13912: [patch] undefined symbols - modules can't be loaded |
| ASTERISK-13913: Truncation problem with AMI ActionID |
| ASTERISK-13914: [patch] func_odbc not returning expected results |
| ASTERISK-13915: [patch] SendFax function not working as expected on > 1.6.0.7 |
| ASTERISK-13916: One Way audio on incoming calls from SIP provider trunk |
| ASTERISK-13917: [patch] prevent a segfault when use of RetryDial is incorrect |
| ASTERISK-13918: Missing line in CHANGES |
| ASTERISK-13919: Warning message refers to non-existant file |
| ASTERISK-13920: Initializing Sagem my850 carat |
| ASTERISK-13921: So much silence inserted in recorded sound files by Manager Monitor |
| ASTERISK-13922: [patch] Regular segfault with chan_unistim |
| ASTERISK-13923: [patch] lock is not released on channel masquerade |
| ASTERISK-13924: Asterisk Realtime - res_odbc failing MySQL connection on rtupdate=yes |
| ASTERISK-13925: [patch] filter for manager events |
| ASTERISK-13926: [patch] Park does not say digits back to caller |
| ASTERISK-13927: [patch] LSB header for init scripts |
| ASTERISK-13928: 183 Session Progress Stops Ringback |
| ASTERISK-13929: Background application executed in post-Dial Application Macro terminates call |
| ASTERISK-13930: 100% cpu problem with channel hangup |
| ASTERISK-13931: IAX2 failed registration notices are spamming the CLI until /var/log/asterisk/messages file fills hard drive 100% |
| ASTERISK-13932: I have 100% CPU usage with a few calls |
| ASTERISK-13933: Motorola EM330 |
| ASTERISK-13934: Asterisk build 1.6.1.0-rc4 reloads on SIP Transfer notice |
| ASTERISK-13935: [patch] Add support in Asterisk 1.4.24 for Bri cards (Digium B410P - wcb4xxp) |
| ASTERISK-13936: [patch] New dialplan application that tell Asterisk to not generate/send Manager Event for the current call |
| ASTERISK-13937: [patch] Missing \r\n in response to JabberSend manager action |
| ASTERISK-13938: pri_resolve_span assumes span's channels have consecutive numbers |
| ASTERISK-13939: No disconnecting when the call is hangup up (cellphone <> cellphone) |
| ASTERISK-13940: SMS support for Motorola phones |
| ASTERISK-13941: [patch] New feature for incrementing and decrementing channel variables |
| ASTERISK-13942: [patch] smsq uses '|' rather than ',' for options in call file |
| ASTERISK-13943: Parking extension number is not overriden in custom parking lots |
| ASTERISK-13944: [patch] rtupdate=no not working |
| ASTERISK-13945: chan_sip deadlock |
| ASTERISK-13946: Redundant AEL Noops |
| ASTERISK-13947: odbc show says that limit is 232, and blocks when exceeding that amount |
| ASTERISK-13948: Thread-specific vm_state tracking issue if a voicemail is left immediately after a restart. |
| ASTERISK-13949: [patch] no audio: chan_mobile with windows mobile 6.1 needs HEADSET_AGW, not HANDSFREE_AGW |
| ASTERISK-13950: [SRTP branch] RTP Read error: Success: Hanging up. If res_timing_pthread.so is loaded |
| ASTERISK-13951: [patch] Using regcontext=something and reload from Asterisk CLI with realtime peers , Asterisk destroys content on sipregistrati |
| ASTERISK-13952: [patch] Asterisk crashes when extenpatternmatchnew=yes |
| ASTERISK-13953: SMS support for Motorola phones (unexpected AT message) |
| ASTERISK-13954: expressions are not working |
| ASTERISK-13955: crash |
| ASTERISK-13956: [patch] Add environment variables to #exec for config files, add new AST_BUILD dialplan function |
| ASTERISK-13957: SMS support for Motorola phones (unexpected second AT message) |
| ASTERISK-13958: Troubleshooting in fax extension (analog port through SPA8000) |
| ASTERISK-13959: [patch] RTP timeout problem |
| ASTERISK-13960: Registration looks only at IP not port |
| ASTERISK-13961: [patch] Log message storms prevention. |
| ASTERISK-13962: [patch] Incorrectly configure (autoconf) when using the --with-something=directory construct with non standard directories |
| ASTERISK-13963: [1.6.0.9] exten => i ignored |
| ASTERISK-13964: .call file issue.. call is always listed as 'down'.. even when up? |
| ASTERISK-13965: [patch] hangup when the spied-on channel hangups |
| ASTERISK-13966: voicemail umask / permissions bug |
| ASTERISK-13967: limitonpeers=yes vs call-limit=1 didn't work on asterisk 1.4.24 |
| ASTERISK-13968: DB Reading Crashes Asterisk |
| ASTERISK-13969: SIP video phone requires numberofmediastreams > 4 (chan_sip.c) |
| ASTERISK-13970: SIP video phone requires numberofmediastreams > 4 (chan_sip.c) |
| ASTERISK-13971: Asterisk 1.4.24.1 deadlock |
| ASTERISK-13972: RTP ports dont get closed with SIP over TCP |
| ASTERISK-13973: Segmentation fault after almost exactly ~ 500 finished SIP over TCP calls. |
| ASTERISK-13974: Cannot make outbound call through analog trunk |
| ASTERISK-13975: ast_closestream leaves file open indefinitely |
| ASTERISK-13976: res_fax fails on fax receive |
| ASTERISK-13977: Unanswered transfers return to transferee's vm |
| ASTERISK-13978: Core dump on high load , high dialplan MYSQL usage |
| ASTERISK-13979: Multiple "unreachable" and "reachable" messages for each SIP registrations |
| ASTERISK-13980: [patch] Create option to require audio before reinvite |
| ASTERISK-13981: sip unregister CLI command doesn't send register invite with expire=0 to remote client. |
| ASTERISK-13982: [patch] blank lines in _sip_tcp_helper_thread caused by sip dummy packages |
| ASTERISK-13983: [patch] Detect pthread_rwlock_timedwrlock() before usage |
| ASTERISK-13984: [patch] asterisk-1.6.0.9-x86_64 segfaults when leaving a voicemail internally to another extension |
| ASTERISK-13985: SVN Trunk fails to link on Mac OS X |
| ASTERISK-13986: Asterisk auto dial out will execute immediatetly no matter the receiver reponse for the call or not |
| ASTERISK-13987: Problem in iLBC Source Fetch Script on FreeBSD |
| ASTERISK-13988: Playback Produces Staggered Audio |
| ASTERISK-13989: Option to Allow Escape Sequences in Logs |
| ASTERISK-13990: Using '@' to specify a context in AEL will cause parse errors |
| ASTERISK-13991: New function to escape any character in a string |
| ASTERISK-13992: SIP INVITE contains external IP causing 404 Not Found on the Cisco 9740 |
| ASTERISK-13993: Hints are messed up on phones after a sip reload |
| ASTERISK-13994: Abort in free() in local_hangup, possibly related to failure to provide ringback indication |
| ASTERISK-13995: chan_h323 crashes Asterisk in version 1.6.2 |
| ASTERISK-13996: Asterisk crashes with using 'Flash Hook' - *0 |
| ASTERISK-13997: [patch] 'V' Command sets one variable only. |
| ASTERISK-13998: #0013548 breaks dialplans that need to retain control when calling an invalid number as a subroutine |
| ASTERISK-13999: [patch] Can't delete temporary greeting when using IMAP storage |