| [..] |
| ASTERISK-11000: Fake ringback and huge sound delay |
| ASTERISK-11001: [patch] allow multiple skip_ms in waitstream_core() |
| ASTERISK-11002: Asterisk Realtime not resoving host names |
| ASTERISK-11003: [patch] Chash after Monitor w/o options |
| ASTERISK-11004: [patch] free already allocated datastore on error. |
| ASTERISK-11005: Maximum retries exceeded on transmission |
| ASTERISK-11006: Asterisk Restarted after some hours |
| ASTERISK-11007: [patch] On error free memory used in ast_el_strtoarr() and complete the /*TODO*/ |
| ASTERISK-11008: [patch] Don't query without connection and fix of debug messages |
| ASTERISK-11009: unanswered=no (cdr.conf) has no effect. |
| ASTERISK-11010: Asterisk 1.4.15 uses 200% of CPU randomly and crashes the machine. |
| ASTERISK-11011: Asterisk TRUNK version - problem with NAT, ignoring localnet, .. |
| ASTERISK-11012: [patch] Re-enable one-touch parking on timeout park recall |
| ASTERISK-11013: [patch] Enable one-touch parking for Queue Members |
| ASTERISK-11014: [patch] DYNAMIC_FEATURES variable looked up in wrong channel during feature interpretation |
| ASTERISK-11015: NetBSD Build Needs RPATH set in 1.2.25 |
| ASTERISK-11016: About SRTP connection error |
| ASTERISK-11017: Asterisk crash when call hung up |
| ASTERISK-11018: Asterisk doesn't build with new Speex 1.2beta3 |
| ASTERISK-11019: [patch] Please increase MAX_SPANS ("Channel ___ does not lie on a span I know of") |
| ASTERISK-11020: One recording file empty for calls in queue with callback agents |
| ASTERISK-11021: [patch] answer before Say* apps |
| ASTERISK-11022: sip_poke_noanswer; peer in now unreachable |
| ASTERISK-11023: [patch] Macros names that deleted after 1.2 still in source code |
| ASTERISK-11024: [patch] Resource generating white noise (small step to CNG) |
| ASTERISK-11025: [patch] chan_sip can cause 100% CPU usage on illegal listening IP |
| ASTERISK-11026: With "pedantic=yes" Asterisk shouldn't match To tag if the dialog is not established |
| ASTERISK-11027: [patch] fix for "possible deadlock" warning |
| ASTERISK-11028: ast_streamfile expects filename argument to have extension removed |
| ASTERISK-11029: [patch] allow resetting group parameters |
| ASTERISK-11030: SIP Redirect causes Asked to transmit frame type 64, while native formats is 0x4 (ulaw) |
| ASTERISK-11031: Issue #10690 breaks compatibility with FreePBX and other types of configurations. |
| ASTERISK-11032: [patch] Check the value of the 'penalty' parameter for the AMI action 'QueuePenalty' |
| ASTERISK-11033: chan_sip updates route set on re-invites, which is not allowed |
| ASTERISK-11034: Asterisk causes the whole system to hang |
| ASTERISK-11035: [patch] Change ast_verbose to ast_verb |
| ASTERISK-11036: FAILURE status report |
| ASTERISK-11037: [patch] Properly set caller id |
| ASTERISK-11038: Prodding channel... causes 100% utilization |
| ASTERISK-11039: sip show channels do not show used codec but hex value |
| ASTERISK-11040: [patch] ChannelRedirect causes the channel executing it to terminate if any form of error occurs. |
| ASTERISK-11041: Compile/Make error |
| ASTERISK-11042: "No audio" on incoming bluetooth call to voicemail, call dropped |
| ASTERISK-11043: Asterisk segfault coincides with the "locking" when attempting to receiving inbound calls |
| ASTERISK-11044: Asterisk segfault coincides with the "locking" when attempting to place outbound calls |
| ASTERISK-11045: [patch] Tweet, tweet |
| ASTERISK-11046: ENUM LOOKUP IS NOT WORKING AGAIN |
| ASTERISK-11047: ENUM LOOKUP IS NOT WORKING AGAIN |
| ASTERISK-11048: "rtptimeout" doesn't terminate channel if RTP is lost during "Echo()" |
| ASTERISK-11049: Compile-Error the the Mashine sparc (tested) or hppa (also tested) - debian |
| ASTERISK-11050: [PATCH] Removing the need for the second app in ExecIf |
| ASTERISK-11051: SIP Attended Tranfer - No Music on hold or Ringing |
| ASTERISK-11052: [patch] chan_mobile.c has no mutexes for data accessed by different threads |
| ASTERISK-11053: [patch] Add config option to force sms to be disabled |
| ASTERISK-11054: Bluetooth client volume should be set upon connection, per spec |
| ASTERISK-11055: Add support for Bluetooth call waiting |
| ASTERISK-11056: leavewhenempty and joinempty only function until an agent has logged in. |
| ASTERISK-11057: [patch] replace ast_log(LOG_DEBUG) with ast_debug() |
| ASTERISK-11058: segfault in devicestate.c |
| ASTERISK-11059: [patch] app_queue.so depends on res_monitor.so but has missing entry in menuselect (causes symbol freeze) |
| ASTERISK-11060: [patch] Music on hold / MOH will not reload or unload |
| ASTERISK-11061: [patch] prevent a segfault when loading |
| ASTERISK-11062: attended transfer let user hear air if transferer hangup. |
| ASTERISK-11063: [patch] Add a new function to check if a extension[,priority] exits in a context |
| ASTERISK-11064: Test |
| ASTERISK-11065: New test |
| ASTERISK-11066: Deadlock? when briging calls on PRI device after sometime |
| ASTERISK-11067: [branch] Allow disconnect feature before a call is bridged |
| ASTERISK-11068: Almost no codec translations present after compile of trunk version of Asterisk |
| ASTERISK-11069: Missed include in channels/chan_h323.c |
| ASTERISK-11070: chan_skinny doesn't compile on FreeBSD out of the box |
| ASTERISK-11071: [patch] Allow AGI script writer to set username/password for the outgoing SIP call leg |
| ASTERISK-11072: Determining SIP session codec from the AGI script |
| ASTERISK-11073: Monitor and One Touch Recording call for "soxmix" |
| ASTERISK-11074: [patch] zap logic group limit |
| ASTERISK-11075: Initial OPTIONS not sent to "peer" context (IP matching) but to the general one |
| ASTERISK-11076: Allow for the conditional logging of CDRs based on the call disposition |
| ASTERISK-11077: Asterisk dies on chan_zap reload |
| ASTERISK-11078: Ring requested on channel |
| ASTERISK-11079: Parametrized DEFAULT_FREQ_OK (Qualification: How often to check for the host to be up) |
| ASTERISK-11080: rasterisk, selecting socket to connect via command line |
| ASTERISK-11081: [patch] syntax fix for both brazilian and european portuguese (reported for SayUnixTime and VoiceMailMain) |
| ASTERISK-11082: Voicemail cuts off at 60 seconds regardless of config settings |
| ASTERISK-11083: crashes with "*** glibc detected *** corrupted double-linked list" with connections to asterisk manager |
| ASTERISK-11084: AST-2007-027 change causes seg fault |
| ASTERISK-11085: Queue and Transfer using Local channel |
| ASTERISK-11086: Queue and Transfer using Local channel |
| ASTERISK-11087: [patch] Optionally swap the first and second CLI arguments for a more "human" interface |
| ASTERISK-11088: 93668 introduces segfault into chan_iax |
| ASTERISK-11089: POTS->VoIP calling defficiency |
| ASTERISK-11090: Deadlock in pbx/sip introduced by fix for bug 0011323 |
| ASTERISK-11091: CDR dates not in local time when stored |
| ASTERISK-11092: Repeatedly calling Page with a LOCAL channel crashes asterisk |
| ASTERISK-11093: Unable to Create Outgoing SIP Channel |
| ASTERISK-11094: [patch] T38 support for SendFax/ReceiveFax |
| ASTERISK-11095: [patch] Email notification for forwarded voicemail doesn't set VM_DUR properly |
| ASTERISK-11096: Forwarded voicemail with prepended message replaces source voicemail |
| ASTERISK-11097: AST-2007-027 caused a bug with outbound dialing using the SIPPEERS table |
| ASTERISK-11098: version.h does not get updated with incremental builds |
| ASTERISK-11099: Sip channels lock |
| ASTERISK-11100: 1.4.16.2 application Park() recalls to caller instead of callee |
| ASTERISK-11101: SQL Insert Support For func_odbc |
| ASTERISK-11102: ChanIsAvail shouldn't generate warnings |
| ASTERISK-11103: Store queue_log in RT |
| ASTERISK-11104: iax2 show peers are showing duplicate lines |
| ASTERISK-11105: Big Latency of about 1-2 Secs. |
| ASTERISK-11106: [patch?] SIP t38state is not properly set |
| ASTERISK-11107: rev 94660 breaks realtime_peer |
| ASTERISK-11108: Get "No such channel" when querring variable |
| ASTERISK-11109: HOLD PRI-ZAP CHANNEL |
| ASTERISK-11110: [patch] multiple issues with autoservice |
| ASTERISK-11111: many retries to lock channels when using attended transfer |
| ASTERISK-11112: Also Header on BYE will Crash Asterisk |
| ASTERISK-11113: Can't make two channels compatible if they have common video codec |
| ASTERISK-11114: Directed Call Pickup does not work from IVR or DIDs. |
| ASTERISK-11115: [patch] Improved ODBC error results in cdr_odbc |
| ASTERISK-11116: Asterisk is crashing when using chanspy + MixMonitor |
| ASTERISK-11117: use additional mysql fields based on CDR variables |
| ASTERISK-11118: Allow for the conditional logging of CDRs based on call disposition |
| ASTERISK-11119: [patch] res_jabber appears to not be working correctly in /trunk |
| ASTERISK-11120: [patch] Unable to specify the socket file to use from the command line and manual page update |
| ASTERISK-11121: [patch] pbx_dundi is not setting EID neither on FreeBSD or OSX (null mac address checking) |
| ASTERISK-11122: 'an-error-has-occured' should be 'an-error-has-occurred' |
| ASTERISK-11123: Extra increment of req->headers in add_header function |
| ASTERISK-11124: [patch] correct action_ping() and action_events(), to set correct header in responce. |
| ASTERISK-11125: [patch] Additional AMD logging and added maximum word length option. |
| ASTERISK-11126: Get the parent on a new channel event |
| ASTERISK-11127: I can't dial out because ACCOUNTCODE not be set |
| ASTERISK-11128: Chanspy crashing asterisk |
| ASTERISK-11129: Asterisk gives segfault |
| ASTERISK-11130: in a type-9 NAT environment, asterisk 1.4.15 and 1.4.16.2 fail to playback/background/datetime/etc |
| ASTERISK-11131: app_followme broken in /trunk |
| ASTERISK-11132: [patch] Avoid compile warning message (uninitialized variable) |
| ASTERISK-11133: v1.4.16.2 - Function STAT() returns 1 if file exists, blank if not found, should return zero |
| ASTERISK-11134: Asterisk random crash during load. |
| ASTERISK-11135: [patch] Proper Hebrew support for Asterisk |
| ASTERISK-11136: [patch] prevent a segfault |
| ASTERISK-11137: tonezone=au causes incorrect tones to be dialed. |
| ASTERISK-11138: IMAP support causes Asterisk to hang once in a while |
| ASTERISK-11139: [patch] Doc update: Substitute the pipe with the comma |
| ASTERISK-11140: show channels concise fails |
| ASTERISK-11141: [patch] AST_NONSTANDARD_APP_ARGS with a , as seperator should be AST_STANDARD_APP_ARGS |
| ASTERISK-11142: [patch] Asterisk misses cast to char * on some execl calls |
| ASTERISK-11143: hangupcause is always 0 if call made via call file fails |
| ASTERISK-11144: Using static realtime configuration with postgres crashes Asterisk on startup |
| ASTERISK-11145: IAX crashes |
| ASTERISK-11146: chan_mobile cell phone will not connect via bt. |
| ASTERISK-11147: Voicemail(103@other,u) does not allow voicemail retrieval when x103 is in [other] in voicemail.conf |
| ASTERISK-11148: [patch] information stored before unlock, but not used |
| ASTERISK-11149: On high call volumne chan_zap is not working properly |
| ASTERISK-11150: [patch] Notification email should use the voicemail's metadata |
| ASTERISK-11151: REGRESSION: broken callerid |
| ASTERISK-11152: [patch] live_ast: script to run asterisk without installing |
| ASTERISK-11153: module unload <tab> doesnt work |
| ASTERISK-11154: [patch] ast_feature_request_and_dial from res_features, used on atxfer, is not setting the channel language |
| ASTERISK-11155: I can crash it at will |
| ASTERISK-11156: How do I troublesheeo this crash |
| ASTERISK-11157: segfault with the AMI |
| ASTERISK-11158: Crash in app_meetme when invoking the page function |
| ASTERISK-11159: [patch] Allow Sipura handsets (SPA942) to use Asterisk SLA via implementation of Broadsoft Sip extensions |
| ASTERISK-11160: When I start my sipp simulator, it crashes in a minute |
| ASTERISK-11161: [patch] use signalling from zaptel with signalling = auto |
| ASTERISK-11162: [patch] add the possibility to know if a meetme conference is locked before joining it. |
| ASTERISK-11163: app_system() sets SYSTEMSTATUS randomly not dependant on the return result of the linux system call |
| ASTERISK-11164: Asterisk failed after speex upgrade |
| ASTERISK-11165: Asterisk crashes when connecting the 10th simultaneous ODBC pooled connections |
| ASTERISK-11166: Set(CDR(accountcode)=value) does not work after ForkCDR |
| ASTERISK-11167: Asterisk hanging whole system on certain configuration |
| ASTERISK-11168: Low trunk frequency + jitter buffer = broken audio, weird netstats |
| ASTERISK-11169: Segmentation fault / Memory corruption with intensive AMI |
| ASTERISK-11170: [patch] fix manager's ModuleLoad/ModuleCheck reporting double description, ModuleCheck reported like registered twice |
| ASTERISK-11171: Pickup of calls in queue not possible |
| ASTERISK-11172: Cannot hangup the line (using astribank) - chan_zap.c:1625 zt_set_hook: zt hook failed: Device or resource busy |
| ASTERISK-11173: Misbehaviur of chan_zap in comparision to any regular PBX or IP telephone |
| ASTERISK-11174: Error on utils/extconf.c when buiding aelparse |
| ASTERISK-11175: [patch] Improve CLI output |
| ASTERISK-11176: [patch] Extended Help Dialogs in Voicemail |
| ASTERISK-11177: [patch] code cleanup: remove not maintained pbx_kdeconsole and pbx_gtkconsole from tree |
| ASTERISK-11178: Can't use func_realtime from realtime dialplan |
| ASTERISK-11179: MeetMe doesn't work with Monitor and Zap channels |
| ASTERISK-11180: RTPs not sent to the correct IP |
| ASTERISK-11181: core show channels becomes unresponsive - precursor to deadlock |
| ASTERISK-11182: [info] core show locks |
| ASTERISK-11183: Originating calls using Asterisk manager - reinvites receive 491 Request Pending rather than immediate response |
| ASTERISK-11184: Wrong matching of "type=friend" (but username mismatches!!) |
| ASTERISK-11185: soft hangup is not working |
| ASTERISK-11186: Attended transfer completes but keeps the zap channel with music on hold |
| ASTERISK-11187: Register statement works differently using the plain text file or the Asterisk Realtime Architecture |
| ASTERISK-11188: [patch] Have `help foo` report that "foo" isn't valid, if it isn't. |
| ASTERISK-11189: [patch] GotoIfTime currently only handles the 'true' case. This patch adds proper Goto behavior for the 'false' case as well. |
| ASTERISK-11190: Asterisk 1.4.17 does not playback parking digits to caller |
| ASTERISK-11191: When using ForkCDR Asterisk does not set the duration and billsec correcty |
| ASTERISK-11192: [patch] core show translation listing unknown non-voice codecs |
| ASTERISK-11193: caller side of sip hints not updated |
| ASTERISK-11194: GROUP_COUNT called in macro cause crash |
| ASTERISK-11195: codec_resample crashes asterisk on loading ("Bus error") |
| ASTERISK-11196: Binds a local variable, and uses it once its out of scope |
| ASTERISK-11197: Sending 503 after 100 trying and BYE message |
| ASTERISK-11198: Calling Macro after hangup from withing GoSub executes only first priority of macro. |
| ASTERISK-11199: in certain scenarios, asterisk can send rtp in an unsupported payload type to an endpoint |
| ASTERISK-11200: Multiple deadlock / show channels freeze on Queues 1.4.16/17 |
| ASTERISK-11201: The issue 11726 was closed, but I am witness to the problem |
| ASTERISK-11202: func_devstate not updating Custom hints |
| ASTERISK-11203: Error when saving a message to IMAP: Message contains bare newlines |
| ASTERISK-11204: User is unable to leave a message for themself |
| ASTERISK-11205: sip hung channels and UDP ports |
| ASTERISK-11206: Cancel sending not conform to SIP RFC 3261 |
| ASTERISK-11207: ChanSPY beep continually when there are not active calls. |
| ASTERISK-11208: Not possible to pass a NULL user or Password into SQLConnect |
| ASTERISK-11209: DTMF problem on 1.4.17 |
| ASTERISK-11210: Recorded files get deleted before mixing if the call was blind transfered |
| ASTERISK-11211: Please merge the ToS/libcap patch to 1.4 branch |
| ASTERISK-11212: [patch] agi RECORD FILE incorrect result code on hangup |
| ASTERISK-11213: [patch] multiple bugs in Directory application |
| ASTERISK-11214: All ODBC calls do not timeout |
| ASTERISK-11215: My asterisk crashes randomly with very low volume |
| ASTERISK-11216: [patch] res_config_curl |
| ASTERISK-11217: app_amd doesnt works properly |
| ASTERISK-11218: [patch] AMI challenge/response authentication uses user supplied secret to calculate hash |
| ASTERISK-11219: Asterisk ignores SDP when multiple Content-Type headers are present |
| ASTERISK-11220: misdn reload break with glibc detected |
| ASTERISK-11221: the latest trunk breaks codec_h323.so, asterisk crashes |
| ASTERISK-11222: app_channelredirect relies on ast_parseable_goto which fails to redirect channels |
| ASTERISK-11223: app_controlplayback: Remove the default skip keys * and # |
| ASTERISK-11224: Asterisk crash when make call from SIP endpoint to H323 using chan_h323 |
| ASTERISK-11225: Asterisk stops responding to sip traffic if DNS doesn't respond during SIP registration |
| ASTERISK-11226: AMI bug - call track lost - when using queues |
| ASTERISK-11227: Hangup request recieved on q931 but asterisk does not hangup the channel |
| ASTERISK-11228: [patch] small warning added when using Set() or MSet() and variable names with spaces |
| ASTERISK-11229: VoiceMailMain with res_adsi.so unloaded crashes asterisk with no error logged |
| ASTERISK-11230: Attempt To Add T38 Gateway Support As A Codec Translator |
| ASTERISK-11231: Hangup request recieved on q931 but asterisk does not hangup the channel |
| ASTERISK-11232: [patch] Small addition to documentation and fixed misprint |
| ASTERISK-11233: MixMonitor doesn't work right with SIP and FLASH on FXS channels |
| ASTERISK-11234: Fake ring tone |
| ASTERISK-11235: [patch] Add log of version updates |
| ASTERISK-11236: [patch] agi SET VARIABLE does not allow the substitution of functions/variables. |
| ASTERISK-11237: Need to play back the call parking announcement to the phone that parked the call - not via a callback |
| ASTERISK-11238: IAX2 with RealTime |
| ASTERISK-11239: Spy'ing on channel crashes Asterisk |
| ASTERISK-11240: [patch] Wrong value in XML HTTP manager response |
| ASTERISK-11241: Interrupt swift text |
| ASTERISK-11242: Interrupt swift text |
| ASTERISK-11243: CALLERID is not logged correctly into cdrs |
| ASTERISK-11244: Asterisk segfaults randomly on INVITE from phone to external number |
| ASTERISK-11245: [authenticaion] in sip.conf: A malicius "Contact" header in REGISTER can get free calls through SIP provider |
| ASTERISK-11246: unknown media description format |
| ASTERISK-11247: DTMF received through rfc2833 having double digits, missing digits, wrong information |
| ASTERISK-11248: Asterisk crashes due to non-atomic check on chan_iax.c:schedule_delivery |
| ASTERISK-11249: [patch] Add file to execute cli commands in on startup |
| ASTERISK-11250: chanspy - crashes Asterisk |
| ASTERISK-11251: Via header in responses might get truncated |
| ASTERISK-11252: [patch] documentation enhancement |
| ASTERISK-11253: [patch] sending_complete on overlap span not handled correctly |
| ASTERISK-11254: In "Static Realtime" app_meetme does a complete SQL query for each "MeetMe()" execution |
| ASTERISK-11255: Asterisk blocks all activity after SIP call attempt, hints related |
| ASTERISK-11256: Build fix for app_voicemail w/IMAP |
| ASTERISK-11257: Wrong behaviour using "d" and "D" options in MeetMe(confno,d/D,pin) |
| ASTERISK-11258: iax2 deadlock? |
| ASTERISK-11259: Crash on iax2 channel hangup |
| ASTERISK-11260: Asterisk Crashes while trying to destroy something |
| ASTERISK-11261: Existing Channels are reported as Not Existing |
| ASTERISK-11262: One way audio problem |
| ASTERISK-11263: [patch] not all greetings copied to database |
| ASTERISK-11264: [patch] refactoring of fax tone detection in DSP |
| ASTERISK-11265: [patch] app_rtpstream: Application to Page Multicast capable receivers (e.g. Snom, Linksys, Cisco, Barix devices) |
| ASTERISK-11266: [patch] Idle check for res_odbc |
| ASTERISK-11267: [patch] Fix for distinctive ring detection on TDM400P |
| ASTERISK-11268: asterisk 1.6-beta1 destroys cisco 7960 (sip firmware 7.4) outbound calls after 20sec due to no response to 200 OK |
| ASTERISK-11269: mobile to asterisk audio stability strongly depends on asterisk to mobile audio activity |
| ASTERISK-11270: Make File- Warning out of order |
| ASTERISK-11271: LDAP RealTime driver loaded. Segmentation fault |
| ASTERISK-11272: A good samples for Spanish-Spain voices. |
| ASTERISK-11273: Voicemail password is not reset in memory when externpass option is used |
| ASTERISK-11274: chan_mobile unable to connect to Motorola L7e: rfcomm_connect: connect() failed (111) |
| ASTERISK-11275: Crash. bridged->chan is null |
| ASTERISK-11276: [patch] build error on main/tcptls.c when no ssl support is used |
| ASTERISK-11277: Hold button works in debug mode only |
| ASTERISK-11278: receive 491 after reinvite handeling |
| ASTERISK-11279: Add B-Leg Call-ID to A-Leg as a channel variable |
| ASTERISK-11280: [branch] [sound] Ability to mark a voicemail message as URGENT. |
| ASTERISK-11281: Deadlock in chan_zap between zt_request and do_monitor |
| ASTERISK-11282: Incorrect SIP Notification after failed dialing attempt |
| ASTERISK-11283: Asterisk core dump |
| ASTERISK-11284: RTP gets passed on without early media session |
| ASTERISK-11285: [patch] Placeholder for format_aiff |
| ASTERISK-11286: Add addtional commands to app_externalivr.c and some code restructuring |
| ASTERISK-11287: Call transfer with 4 and more participants |
| ASTERISK-11288: [patch] Add TCP socket support to app_externalivr.c |
| ASTERISK-11289: [patch] Log that format does not exists when it is no true |
| ASTERISK-11290: [patch] configure script correct applications |
| ASTERISK-11291: Mantis issue with SVN revision no > 100000 |
| ASTERISK-11292: [patch] Sounds for fallback searched not in the right place |
| ASTERISK-11293: [patch] Compile fixes for solaris |
| ASTERISK-11294: [patch] SIP connected to AGI over chan_local problem |
| ASTERISK-11295: compile - make fails if directory path contains spaces |
| ASTERISK-11296: Marked conference mode not working properly! |
| ASTERISK-11297: Create random audio failure in Asterisk |
| ASTERISK-11298: [patch] [sound] Ability to send to multiple recipients. |
| ASTERISK-11299: SIGSEGV ast_read from ast_channel_bridge/..generic_bridge, chan->tech==0x2 |
| ASTERISK-11300: Crash when reload and dialout at the same time |
| ASTERISK-11301: [patch] prevent a segfault when loading |
| ASTERISK-11302: Asterisk core dump |
| ASTERISK-11303: Asterisk core dump |
| ASTERISK-11304: Moved Temporarily Contact Transport information not used in next invite |
| ASTERISK-11305: Asterisk Core dump while chan spy |
| ASTERISK-11306: Small memory leak in cdr.c |
| ASTERISK-11307: configure does not find floor, pow, rint, sqrt in tgmath.h |
| ASTERISK-11308: Incorrect dialog matching and requests on blind transfer |
| ASTERISK-11309: Missing CDR's for Transfers |
| ASTERISK-11310: Minor "appearance" improvements |
| ASTERISK-11311: G729 license with two ethernet card |
| ASTERISK-11312: RTCP Read too short |
| ASTERISK-11313: bindaddr problems |
| ASTERISK-11314: The ring option in queue doesn't work if no audio has been played previously on the channel |
| ASTERISK-11315: Asterisk crashed while trying ot hang up an IAX call |
| ASTERISK-11316: memory corruption on freebsd sparc64 (identical case 10300 exists) |
| ASTERISK-11317: Trunk requires externip=ip:port if bindport!=5060 |
| ASTERISK-11318: Random Segmentation Fault (crash) |
| ASTERISK-11319: [patch] misdn_cfg_get return uninitialized array if no default value |
| ASTERISK-11320: "auth" doesn't work inside a [peer] |
| ASTERISK-11321: Possible deadlock on realtime queues. |
| ASTERISK-11322: [asterisk-dev] [patch] Choosing common codec and select "free" codec for transcoding |
| ASTERISK-11323: [PATCH] Device state is incorrect on incoming call on FXO channel |
| ASTERISK-11324: Contributing access-granted and access-denied |
| ASTERISK-11325: test |
| ASTERISK-11326: pvt test |
| ASTERISK-11327: one more time! |
| ASTERISK-11328: pub test |
| ASTERISK-11329: Problem connecting Asterisk to UMC-1000 using GR-303 |
| ASTERISK-11330: WARNING: Freeing unused memory at (nil), in ast_yyfree of ast_expr2f.c, line 3089 |
| ASTERISK-11331: AST_FLAG_MOH not cleared during attended transfers |
| ASTERISK-11332: [patch] minimal API to control SIP T38 from application |
| ASTERISK-11333: [patch] VoicemailUsersList AMI action does not always send VoicemailUserEntryComplete event |
| ASTERISK-11334: Default (sample setting) of channel group selection will cause glare on analog circuits |
| ASTERISK-11335: Crash. Can't get more info |
| ASTERISK-11336: Crash in chanspy |
| ASTERISK-11337: segfault, ast_slinfactory_read(), connected with DTMF sending? |
| ASTERISK-11338: SIP username -> defaultuser confuses realtime system(s) |
| ASTERISK-11339: MPEG4 video capabilities integrated in videocaps |
| ASTERISK-11340: [patch] Fix some more solaris build issues |
| ASTERISK-11341: function REALTIME() broken |
| ASTERISK-11342: [patch] new REALTIME_STORE() and REALTIME_DESTROY() functions |
| ASTERISK-11343: Crash. I think in AUDIOHOOCKS |
| ASTERISK-11344: [patch] realtime Queue members seem not to work using res_config_curl |
| ASTERISK-11345: Asterisk hangs due to lost network connections |
| ASTERISK-11346: 100% CPU if zaptel timing fails |
| ASTERISK-11347: G729 decoders not freed after IAX2 call |
| ASTERISK-11348: [patch] added line numbers to warnings and errors |
| ASTERISK-11349: sip tcp and tls transport doesn't work, locks asterisk |
| ASTERISK-11350: [patch] Optimized update_realtime_member_field function |
| ASTERISK-11351: Missed protection from incorrect dial string in parse_dial_string |
| ASTERISK-11352: DIAL option 'T' still has a bad behaviour |
| ASTERISK-11353: [patch] Change argc != xx to argc != e->args [apps] |
| ASTERISK-11354: Call-bridged Macro feature request |
| ASTERISK-11355: bridging chan_h323 and chan_sip |
| ASTERISK-11356: Incoming Call with E&M signalling is not working properly. (AsteriskNow 1.4.9.) |
| ASTERISK-11357: Wrong error file shown if includes used |
| ASTERISK-11358: [patch] deprecate some obsolete MOH constructs |
| ASTERISK-11359: Recording unavailable voicemail message returns error on MSSQL when busy and name does not |
| ASTERISK-11360: Show channels |
| ASTERISK-11361: Can't compile with dev-mode |
| ASTERISK-11362: [patch] CoreSettings and CoreStatus are missing the trailing "\r\n" delimiter |
| ASTERISK-11363: [patch] VoiceMail d([context]) option does not set extension, priority |
| ASTERISK-11364: [patch] DTMF digits duplicated |
| ASTERISK-11365: [patch] SPRINTF function is broken |
| ASTERISK-11366: segfault in codec_zap line 150 |
| ASTERISK-11367: Redirect through AMI not working |
| ASTERISK-11368: The asterisk service crashes twice a day |
| ASTERISK-11369: Asterisk don't get the BYE packet from callee |
| ASTERISK-11370: Not getting answers from get_data in a call-file call |
| ASTERISK-11371: Asterisk not playing busy after media bridge |
| ASTERISK-11372: dtmf buffer not cleaned when hangup |
| ASTERISK-11373: AbsoluteTimeOut fails to hang up when loosing signal from an inbound connection |
| ASTERISK-11374: park position announcement no longer working |
| ASTERISK-11375: chan_zap pri_dchannel mutex Invalid argument, then segfault |
| ASTERISK-11376: chan_iax2 does not send full video frames when it is supposed to |
| ASTERISK-11377: DUNDi lookup crashes asterisk |
| ASTERISK-11378: Remote Console Freezes on TAB completion |
| ASTERISK-11379: Asterisk 1.6-beta2 on Mac Intel extrange CLI operation |
| ASTERISK-11380: The dialplan executes wrong, it skips a priority |
| ASTERISK-11381: sip reload should not unregister tcp/tls peers |
| ASTERISK-11382: [patch] Fix last few verbose msg to use new macro |
| ASTERISK-11383: Compilation of app_voicemail with ODBC breaks |
| ASTERISK-11384: RTCP interferes with Cisco 7940 jitter buffer |
| ASTERISK-11385: [patch] replace option_verbose ast_verbose calls with ast_verb |
| ASTERISK-11386: [patch] change cdr_odbc to use SQLExecDirect() instead of SQLPrepare() then SQLExecute(). |
| ASTERISK-11387: Compilation failed with malloc debug |
| ASTERISK-11388: AelParse dump separates command parameters with pipes, that are not accepted by 1.6 |
| ASTERISK-11389: Invalid interpretation of INVITE SIP frame |
| ASTERISK-11390: The "contact" section of the invite is wrong and many gateways reject the invite |
| ASTERISK-11391: This crash happenned this morning |
| ASTERISK-11392: [patch] Feature to write variables to existing channels other than your own (func chanvar) |
| ASTERISK-11393: DUNDi Does Not Follow Naming Conventions |
| ASTERISK-11394: MixMonitor - Out Of Sync Audio With Zap Channels |
| ASTERISK-11395: DUNDi Lookups and Queries Fail |
| ASTERISK-11396: incorrect callerID during attended transfer |
| ASTERISK-11397: Ability to use Polycom's server based DND |
| ASTERISK-11398: [patch] log what digits feature tries to match |
| ASTERISK-11399: Bug or Miss-Configuration?? |
| ASTERISK-11400: dbsecret is not considered a "secret" |
| ASTERISK-11401: segfault on module reload chan_console.so |
| ASTERISK-11402: [patch] When the caller hangs up - transfer the called party to the specified context and extension provided by this option. |
| ASTERISK-11403: Fast RFC2833 dialing results in asterisk generating overlapping |
| ASTERISK-11404: rejected due to usage limit of 1 |
| ASTERISK-11405: Dialplan functions are executed before conditions are evaluated |
| ASTERISK-11406: ast_print_group misuses strncat |
| ASTERISK-11407: add address to some PeerStatus events, and add some PeerStatus Rejected events to chan_sip.c |
| ASTERISK-11408: Call from '' to extension '7104' rejected because extension not found |
| ASTERISK-11409: [patch] Simple file recording buffer - great performance gain |
| ASTERISK-11410: res_snmp.so hangs asterisk at startup |
| ASTERISK-11411: Compile Fail when enable Module Embedding |
| ASTERISK-11412: [patch] replace ast_verbose with ast_debug(1, for debugging |
| ASTERISK-11413: [patch] DSP cleanup phase 2 |
| ASTERISK-11414: [patch] Changes to configuration files to make setting up virtual hosting more obvious |
| ASTERISK-11415: Usage of Goto() in an included context can cause unexpected behaviour |
| ASTERISK-11416: ifaddrs.h not available on all linux distributions => asterisk fails to compile |
| ASTERISK-11417: tls transport often causes asterisk to lock |
| ASTERISK-11418: The L(XXX) option does not work when the call is bridged. |
| ASTERISK-11419: external lines connected with message !! Got Busy in Connected State !?! |
| ASTERISK-11420: Asterisk Core dump |
| ASTERISK-11421: Asterisk-Addons 1.4.5 still wont build with codec negotiation patch applied |
| ASTERISK-11422: CID name from sip.conf when no CID name is supplied |
| ASTERISK-11423: SLA failing in 1.6rcbeta2 with repeated restarting of the meetme conference |
| ASTERISK-11424: Asterisk crash when I park a call |
| ASTERISK-11425: 'sip show registry' shows wrong registration |
| ASTERISK-11426: [branch] Report that channels are controlled by AGI |
| ASTERISK-11427: Can not register 2 different SIP accounts with provider |
| ASTERISK-11428: asterisk giving core dump every 10 minute-using latest tarball |
| ASTERISK-11429: Crash with AMI originates |
| ASTERISK-11430: asterisk giving core dump on my production server very frequently |
| ASTERISK-11431: Problem adding ${CDR(foo)} while using .call file |
| ASTERISK-11432: IMAP Compile-time warnings of redeclared variables |
| ASTERISK-11433: [patch] possible memory leak while allocating PRI channels |
| ASTERISK-11434: Callers on Hold in SLA that hang up do not change status in SLA |
| ASTERISK-11435: Zap procedding is coming twice |
| ASTERISK-11436: [BSD Portability] ooSocketGetInterfaceList can't get interface list right |
| ASTERISK-11437: asterisk crashes when trying to make a call from SIP endpoint to h323 endpoint registered to gnugk |
| ASTERISK-11438: Introducing en extension pattern matching variables similar to X or Z, but also recognizes the * and/or # |
| ASTERISK-11439: T.38 Re-Invite header replaced by audio header when SIP asks for authorization |
| ASTERISK-11440: [patch] Fix Doxygen errors |
| ASTERISK-11441: [patch] Partial doxygen update to app_queue |
| ASTERISK-11442: Asterisk-Addons 1.4.5 wont build when codec negotiation patch applied to asterisk |
| ASTERISK-11443: memory leak |
| ASTERISK-11444: snom DTMF duration detected wrong |
| ASTERISK-11445: invalid strcpy for conf2ael |
| ASTERISK-11446: [patch] invalid strcpy for conf2ael |
| ASTERISK-11447: [patch] configure doesn't detect required libraries |
| ASTERISK-11448: zoiper DTMF stop working suddenly |
| ASTERISK-11449: [patch] SIP INVITES authorization from multiple IP addresses |
| ASTERISK-11450: [patch] chan_sip fails to set contact, via, and sdp headers correctly with outboundproxy set |
| ASTERISK-11451: Asterisk 1.4 AMI Originate does not sufficiently set CDR(accountcode) on the first call leg |
| ASTERISK-11452: After update gets havy noise when Session Progress or Ringing state |
| ASTERISK-11453: Asterisk Frequent Crash |
| ASTERISK-11454: asterisk 1.2.18 crash |
| ASTERISK-11455: Asterisk 1.4.17 T.38 doesn't work |
| ASTERISK-11456: Remove verbose from say numbers in hebrew |
| ASTERISK-11457: SIGABORT in ast_channel_trylock in chan_local local_queue_frame when channel disappears underneath the code |
| ASTERISK-11458: SIP with canreinvite=yes through multiple Asterisk instances fails |
| ASTERISK-11459: loading chan_gtalk.so before res_jabber.so is segfaulting * |
| ASTERISK-11460: Loading chan_mobile.so results in undefined symbol hci_get_route and immediate asterisk stop |
| ASTERISK-11461: asterisk-ADDONS-1.4.5 not compile: 'struct ast_filestream' has no member named '_private' |
| ASTERISK-11462: Asterisk crashes on dial_exec_full |
| ASTERISK-11463: Randomly asterisk crash using MixMonitor (with ou without Chanspy) |
| ASTERISK-11464: Incorrect description of echo can status |
| ASTERISK-11465: [patch] Update to 'core show settings' text |
| ASTERISK-11466: crash when trying to record unkonwn format |
| ASTERISK-11467: Compile fails with "structure has no member named `tm_gmtoff'" |
| ASTERISK-11468: dont`t correctly work ast_device_state In app_queue when ring entry to the queue |
| ASTERISK-11469: sending or receiving faxes thru ISDN will cause the system to slow down |
| ASTERISK-11470: added Expiry value column to the sip show subscriptions commands |
| ASTERISK-11471: Asterisk 1.6-beta3 does not follow sip redirect using sip/tcp |
| ASTERISK-11472: Forwarding a message using file-based storage causes asterisk to crash. |
| ASTERISK-11473: [patch] Add the ability to forward a message with comment using IMAP storage. |
| ASTERISK-11474: asterisk 1.6.0-beta3 fails to cross compile |
| ASTERISK-11475: 1.6 Beta3 Authenticate won't play the agent-pass.gsm, insists on ulaw |
| ASTERISK-11476: Crash when receiving IAX2 call (peer_hash_cb) |
| ASTERISK-11477: Autocreate IMAP folders that don't exist |
| ASTERISK-11478: [patch] Print warning if module cannot be located to unload |
| ASTERISK-11479: 1.6beta3 - Does not record with monitor() |
| ASTERISK-11480: crash. 1-3 times in day |
| ASTERISK-11481: [patch] RFC 3372 SIP-T receive implementation |
| ASTERISK-11482: Compile fails with "`u_int32_t' undeclared" |
| ASTERISK-11483: Asterisk crash when Paging |
| ASTERISK-11484: Caller ID Inter-operability issue |
| ASTERISK-11485: crash in local_hangup() when p->owner disappears in another thread |
| ASTERISK-11486: Segfault in release_chan when calling own number from the inside |
| ASTERISK-11487: add some unsupported event in astman |
| ASTERISK-11488: [patch] menuselect - Remove timeout after ESC key is pressed |
| ASTERISK-11489: Multiple deadlock / show channels freeze on Queues 1.4.16/17 |
| ASTERISK-11490: setting the channel language first read fails 1.6.0-beta3 |
| ASTERISK-11491: Disable dead channel notice if AGI is called in the 'h' extension. |
| ASTERISK-11492: Mixmonitor crashing asterisk randomly |
| ASTERISK-11493: Cannot setup an empty database password |
| ASTERISK-11494: Leftover modules from 1.4 will crash 1.6 .. |
| ASTERISK-11495: [patch] Compile fails with `IPTOS_MINCOST' undeclared |
| ASTERISK-11496: [patch] debug trace that won't help in __sip_ack() |
| ASTERISK-11497: chan_iax2 seems to ignore the bindport parameter and uses 4569 anyway on 1.6.0-beta4 |
| ASTERISK-11498: Zap channel can't hangup |
| ASTERISK-11499: AJAM error when sending commands |
| ASTERISK-11500: Incorrect DTMF tone emitted - cannot dial calls on Zap channel |
| ASTERISK-11501: [patch] deprecate "stripmsd" in zapata.conf |
| ASTERISK-11502: [patch] chan_sip in pedantic mode : error in tag checking |
| ASTERISK-11503: Crash on SQL Query |
| ASTERISK-11504: Two Asterisk crashes |
| ASTERISK-11505: [patch] extconfig.conf does not allow #exec |
| ASTERISK-11506: Messages silently deleted when user goes over maxmsg in Old folder |
| ASTERISK-11507: Gtalk call fails unless restarts (Maybe?! reopening issue 10707) |
| ASTERISK-11508: Configure Script does not properly test for net-snmp. |
| ASTERISK-11509: configure complains during check for openh323 |
| ASTERISK-11510: func_odbc does not use readhandle for readsql |
| ASTERISK-11511: pri stops receiving answering calls |
| ASTERISK-11512: [patch] chan_zap fails to close file descriptors in case of an error |
| ASTERISK-11513: Hard coded pipes in automon |
| ASTERISK-11514: Asterisk sometimes crash when queue member answered to call |
| ASTERISK-11515: Do not process the UDPTL proxy under two sip channel (like in 1.4.18.x) |
| ASTERISK-11516: Segmentation Fault when Calling app Voicemail after Blind Transfer |
| ASTERISK-11517: when calling chanspy(,q) deadlocks occurs |
| ASTERISK-11518: tz setting in general context is ignored |
| ASTERISK-11519: [patch] T38 passthru broken in trunk |
| ASTERISK-11520: [patch] meaningful variable names in chan_zap |
| ASTERISK-11521: Deadlock if feature in use while reload attempted |
| ASTERISK-11522: Build fails on PPC64 |
| ASTERISK-11523: Dial plan pattern matching not working from odbc tables |
| ASTERISK-11524: Build astcanary fails because of missing mode in open call |
| ASTERISK-11525: [patch] chan_vpb doesn't compile against latest stable release of Voicetronix drivers (4.2.24) |
| ASTERISK-11526: problem with guest account in gtalk |
| ASTERISK-11527: unload res_smdi.so crash asterisk (from rev. 104119) |
| ASTERISK-11528: sip queue members state becomes invalid if after asterisk restart |
| ASTERISK-11529: Random crashes in different places |
| ASTERISK-11530: [patch] [sound] Ability to enforce voicemail minimum password lengths |
| ASTERISK-11531: Asterisk sends 491 Pending for a new INVITE |
| ASTERISK-11532: Asterisk crashes on chan_sip |
| ASTERISK-11533: Listening to Allison voicemail prompt on SIP phone causes [pop] sounds |
| ASTERISK-11534: [patch] Ability to execute external application to verify voicemail password |
| ASTERISK-11535: [patch] chan_vpb segfaults if no cards found |
| ASTERISK-11536: Asterisk Segmentation Fault - High fence violation in ast_frdup of frame.c |
| ASTERISK-11537: Segmentation fault in chan_sip.c |
| ASTERISK-11538: pickup launched with null option will crash asterisk |
| ASTERISK-11539: SIP channel hung due to CANCEL ReliableXmit (ReTX) |
| ASTERISK-11540: curl does not seem to work when cross-compiling |
| ASTERISK-11541: AGI + sounds dir + language |
| ASTERISK-11542: IAX2 extension when dialed takes 15 seconds until ring tone, handset never rings. |
| ASTERISK-11543: asterisk not playing files after upgrade from beta4 to svn |
| ASTERISK-11544: [patch] Ability to set device state via CLI |
| ASTERISK-11545: variable data truncated when attempting password changes using odbc backed |
| ASTERISK-11546: Crash in app_voicemail - unable to open file |
| ASTERISK-11547: segfault when the call hangup after have been spied |
| ASTERISK-11548: CCM sends multiple INVITEs which causes MusicOnHold to execute on a channel |
| ASTERISK-11549: No ringback toward SIP trunk on inbound SLA call (Again) |
| ASTERISK-11550: asterisk-addons 1.6: make install cross compiling issue |
| ASTERISK-11551: [patch] realtime_multi_ldap and pattern matching |
| ASTERISK-11552: [patch] asterisk crash at reload chan_misdn.so |
| ASTERISK-11553: [patch] Wrong type for astConfigCallsProcessed and astNumChannels |
| ASTERISK-11554: MoH file playback is broken |
| ASTERISK-11555: Asterisk Crash |
| ASTERISK-11556: menuselect - CORE-SOUNDS-EN-GSM cannot be replaced |
| ASTERISK-11557: Channels gets blocked after 1 hour of calling |
| ASTERISK-11558: [patch] unnecessary complex expression in last build_user() modification |
| ASTERISK-11559: Zap restart fails |
| ASTERISK-11560: Why Asterisk replies with "486 Busy Here" if it receives a "603 Decline"? |
| ASTERISK-11561: [patch] Sending DTMF when receiving the PROGRESS status |
| ASTERISK-11562: More Solaris 10 building issues |
| ASTERISK-11563: Not recognizing media files on Solaris 10 |
| ASTERISK-11564: Using state_interface and Local channels allows several simultenous calls to be sent to agent |
| ASTERISK-11565: Sometimes (but often) WAV files recorded by voicemail are corrupted |
| ASTERISK-11566: Unable To Start Three-Way Call |
| ASTERISK-11567: Transcoded G.722 calls unintelligible |
| ASTERISK-11568: Asterisk crashed today twice |
| ASTERISK-11569: Playback does not work now when sending to other directory and playing .wav files |
| ASTERISK-11570: [patch] add more bandwidth settings to the SDP |
| ASTERISK-11571: During page no audio can be heard from Polycom or Aastra phones, Snom phones are OK |
| ASTERISK-11572: [patch] Reload of persistent members (if enabled) on module reload |
| ASTERISK-11573: menuselect - review and merge let-there-be-newt branch |
| ASTERISK-11574: [patch] Attempted to delete nonexistent schedule entry |
| ASTERISK-11575: Crash with no reason |
| ASTERISK-11576: Asterisk stops responding after quick hangup between asterisk boxes |
| ASTERISK-11577: Agent status in queues |
| ASTERISK-11578: Asterisk Crash |
| ASTERISK-11579: Ability to change folders while listening to messages (available but not advertised) |
| ASTERISK-11580: BackgroundDetect on 1.6.0 svn crashes asterisk |
| ASTERISK-11581: RTP timestamp skewed |
| ASTERISK-11582: [patch] Added analysis time to BackgroundDetect |
| ASTERISK-11583: [patch] var_metric documentation for when using the old Realtime Static mode |
| ASTERISK-11584: No sound during bringe ZAP<->SIP |
| ASTERISK-11585: WARNING Messege |
| ASTERISK-11586: race condition between sip hangup and "core show channel xxx" results in crash |
| ASTERISK-11587: Global Register vs Global Authentication error |
| ASTERISK-11588: New AMI atxfer limited to digits only and ignores context and priority; looks like just wrapper on PlatDTMF |
| ASTERISK-11589: Monitor with blind transfer deletes files returns bad paths. |
| ASTERISK-11590: [patch] channel alarm set when a channel is opened |
| ASTERISK-11591: res_agi.c can still tell you to use DeadAGI for hung up channels |
| ASTERISK-11592: sipsock_read using unsafe structure |
| ASTERISK-11593: Asterisk segfaults when 'md5secret = userPassword' in res_ldap.conf |
| ASTERISK-11594: Distorted playback of G.722 prompts |
| ASTERISK-11595: Zap restart, deadlock message |
| ASTERISK-11596: MYSQL Query using wrong delimiters and not returning results |
| ASTERISK-11597: Warning message on ast_waitfordigit_full |
| ASTERISK-11598: sip debug can't be stopped |
| ASTERISK-11599: SIP channel isn't closed when using TLS transport |
| ASTERISK-11600: [patch] asciidoc documentation |
| ASTERISK-11601: redirect through AMI not working reliable in 1.4 |
| ASTERISK-11602: app_fax doesn't compile |
| ASTERISK-11603: Timout of getting messages. |
| ASTERISK-11604: Crash asterisk 1.4.19-rc1 |
| ASTERISK-11605: asterisk random crashes |
| ASTERISK-11606: calls to multi_ldap function return with zero results (Pattern match/Queuemember broken) |
| ASTERISK-11607: chan_iax2.c expiring registrations prematurely on an Asterisk that has been running a long time |
| ASTERISK-11608: [patch] chan_zap call progress does not connect if there is talking |
| ASTERISK-11609: ' -fno-strict-overflow' unknown by gcc |
| ASTERISK-11610: Comment on AST_MASQ_NOSTREAM in park_call_full misleading; announcement *can* go to parked channel. |
| ASTERISK-11611: asterisk-1.6.0-beta5 + asterisk-addons-1.6.0-beta2 |
| ASTERISK-11612: Asterisk stop accepting invites or registers |
| ASTERISK-11613: Spelling Mistake |
| ASTERISK-11614: app_page devicestate AST_DEVICE_UNKNOWN |
| ASTERISK-11615: app_page devicestate AST_DEVICE_UNKNOWN |
| ASTERISK-11616: Updated documentation for storing ODBC voicemail in PostgreSQL |
| ASTERISK-11617: Crash in ChanSpy / AST_LIST_TRAVERSE_SAFE_BEGIN |
| ASTERISK-11618: Too many open files, asterisk stops responding |
| ASTERISK-11619: [patch] Update to current API, changed in rev.107791 |
| ASTERISK-11620: [patch] On modules load loader.c prin many dots |
| ASTERISK-11621: System crash at format_wav.c:375 |
| ASTERISK-11622: PARKINGEXTEN misbehaves when it is not a canonical form number and can double park or destroy the 700 entry |
| ASTERISK-11623: Parking timeout fails to return to console and. probably any VoIP device |
| ASTERISK-11624: The tab provided suggestions for the moh command are wrong |
| ASTERISK-11625: Deadlock/Locked channels |
| ASTERISK-11626: [patch] Make Menuselect bombs on a clean checkout or make dist-clean if configure not run. |
| ASTERISK-11627: Make Asterisk-trunk handle 'prefix' in httpd.conf as 1.4 does. |
| ASTERISK-11628: configure script flaw on Solaris 10 (and maybe others) |
| ASTERISK-11629: Incorrect /etc/init.d/asterisk |
| ASTERISK-11630: Registration with SIP-provider never succeeds |
| ASTERISK-11631: running make does not identify cdr_tds correctly |
| ASTERISK-11632: Failed to cross compile on arm |
| ASTERISK-11633: EXTEN gets printf-transformed when written to the logs |
| ASTERISK-11634: [patch] Fix potential crash in manager |
| ASTERISK-11635: Asterisk 1.4.18 crash when peers deregister while in a queue. |
| ASTERISK-11636: Calls drop due to unanswered INVITES |
| ASTERISK-11637: [patch] janitor project for getvar_helper in app_queue |
| ASTERISK-11638: if the LDAP port is not 389, it's not possible to connect a LDAP server |
| ASTERISK-11639: Possible crash if chan is null |
| ASTERISK-11640: The LDAP version is not taken into account |
| ASTERISK-11641: [patch] RFC2833 DTMF relaying error when using p2p bridging |
| ASTERISK-11642: [patch] Added event based MWI to chan_skinny |
| ASTERISK-11643: [patch] Asterisk returns 482 Loop Detected upon receiving re-invite |
| ASTERISK-11644: application map not working for called party |
| ASTERISK-11645: MixMonitor - Out Of Sync Audio With Zap Channels |
| ASTERISK-11646: Authentication for the SIP client through LDAP |
| ASTERISK-11647: Double writelock in ast_wrlock_contexts() |
| ASTERISK-11648: res_features.c don't return calls to the [park-call] "t" extension. when the extension is busy and the parktime occurs |
| ASTERISK-11649: Voicemail is non-functional in head |
| ASTERISK-11650: [Patch] Add support for contexts in polled mailboxes |
| ASTERISK-11651: Use ast_verb instead of ast_debug for channel driver debugging |
| ASTERISK-11652: Crash shortly after launching |
| ASTERISK-11653: Crash in chanspy_ds_destroy (__ast_pthread_mutex_unlock) |
| ASTERISK-11654: main/dial.c case AST_CONTROL_SRCUPDATE missing break |
| ASTERISK-11655: [patch] Add Aastra supported events to sip_notify.conf |
| ASTERISK-11656: Instead of just going to the failed extension, add a couple more for OutgoingSpoolFailed |
| ASTERISK-11657: Crash on initial REGISTER on asterisk start |
| ASTERISK-11658: Deadlock on 1.4 pre-rc3, seems to be related either to chanspy, queue or devicestate |
| ASTERISK-11659: which source for ldap authentication for the SIP user |
| ASTERISK-11660: [patch] A couple of trivial patches for res_config_ldap |
| ASTERISK-11661: [patch] Improve LDAP server configuration for res_config_ldap |
| ASTERISK-11662: [patch] Support for ipddr, defaultuser and regserver in asterisk LDAP schema |
| ASTERISK-11663: Update default res_ldap.conf |
| ASTERISK-11664: include block not loaded on asterisk start |
| ASTERISK-11665: Update phones 'Placed Call' log when using a local analog channel |
| ASTERISK-11666: segfaults when calling ber_bvecfree() |
| ASTERISK-11667: Core dump during normal usage of queues |
| ASTERISK-11668: Deadlock after "core show channels" or "queues show" |
| ASTERISK-11669: Core Dump when Polycom Phone attempts to pick up configuration |
| ASTERISK-11670: [patch] Support for RFC2833 DTMF for dumb SIP proxies |
| ASTERISK-11671: Deleting old voicemail |
| ASTERISK-11672: Choppy voicemail recording for SIP calls via TELES softswitch |
| ASTERISK-11673: [patch] Add option to return all available channels on app_chanisavail |
| ASTERISK-11674: All SIP users InUse counters reset to zero(even if they are on call) during sip reload |
| ASTERISK-11675: chan_isavail segfaulting asterisk after sip_hangup (introduced after #8556 changes) |
| ASTERISK-11676: Read() does not work using 1.6b4 / 1.6b5 |
| ASTERISK-11677: Deadlock in chan_sip |
| ASTERISK-11678: G.726 codec may function incorrectly because of ilog2() asm routine |
| ASTERISK-11679: [patch] [sound] New feature for voicemail to wrap back to last or forward to first message |
| ASTERISK-11680: asterisk crashes (core) when starting with menuselect DO_CRASH |
| ASTERISK-11681: Deadlock in update_status &q->lock |
| ASTERISK-11682: announce-frequency waits to finish the full message before connecting |
| ASTERISK-11683: Call forwarding scenario with Playback() causes dispositions to be reversed |
| ASTERISK-11684: Using state interface and local channels with 12127v3.patch |
| ASTERISK-11685: Periodic app_page pthread_mutex_lock segfault |
| ASTERISK-11686: ChanSpy with spygroup set crashes in strlen(), not consistently |
| ASTERISK-11687: Voicemail |
| ASTERISK-11688: SIP caller hanging up before answer does not stop Dial |
| ASTERISK-11689: problem with autoservice |
| ASTERISK-11690: asterisk terminates with SIGABRT |
| ASTERISK-11691: Asterisk sometimes crash when I use realtime queue member for callcenter |
| ASTERISK-11692: [branch] Deadlock after Originate from AMI to Agent |
| ASTERISK-11693: Makefile optimization for mips architecture |
| ASTERISK-11694: segfault when using DETECT_DEADLOCKS flag |
| ASTERISK-11695: Deprecated SIP-Client session crashes Asterisk in startup |
| ASTERISK-11696: Receiving RTP PDUs crashes Asterisk core |
| ASTERISK-11697: 1.4.18.1 segfaults |
| ASTERISK-11698: [patch] Add support for setting individual contexts in users.conf |
| ASTERISK-11699: hanguponpolarityswitch ignored because I am not using answeronpolarityswitch |
| ASTERISK-11700: Asterisk 1.6.0-beta6 crashes on Nessus scanning |
| ASTERISK-11701: [patch] Add Server: instead of User-Agent: header in Asterisk generated SIP responses |
| ASTERISK-11702: [patch] Remove Event: header in Asterisk generated REGISTER requests |
| ASTERISK-11703: [patch] Putting an ast_verbose call under a debug condition check |
| ASTERISK-11704: [patch] Putting an ast_verbose call under a debug condition check |
| ASTERISK-11705: [patch] Asterisk sends SIP-over-TCP INVITE to wrong port number |
| ASTERISK-11706: Core dump from svn trunk build - not sure why |
| ASTERISK-11707: Started to crash every 2-3 hours |
| ASTERISK-11708: [patch] Permit 'not null' fields to always be set |
| ASTERISK-11709: [patch] immediate=yes treats s extension differently than others: doesn't return cause code 1 if doesn't exist |
| ASTERISK-11710: No audio or audio one way |
| ASTERISK-11711: Remote Asterisk Does not accept Digits pressed. |
| ASTERISK-11712: [patch] Cannot compile asterisk with DONT_OPTIMIZE on a 2.4 kernel and an old version of Slackware |
| ASTERISK-11713: Can not choose res_snmp in "make menuconfig" because dependency (net-snmp package) not recognized. |
| ASTERISK-11714: Problems with http manager interface |
| ASTERISK-11715: [PATCH] Fix DST issue with RealTime scheduling feature |
| ASTERISK-11716: [patch] Migrating stdexten and stdPrivacyexten in extensions.conf from Macro to Gosub |
| ASTERISK-11717: Oneway Audio with trunk/asterisk-1.6.beta6 even getting ringback tone before answer |
| ASTERISK-11718: Crash in chan_sip |
| ASTERISK-11719: MixMonitor with 'b' option record the 'Ringing' and audio is out of sync |
| ASTERISK-11720: Aastra BLF support not working in 1.4.19rc3 |
| ASTERISK-11721: The new pattern matching does not work |
| ASTERISK-11722: [patch] Can't pickup if pickupcode include # |
| ASTERISK-11723: [patch] tun0 and no default route cause crash |
| ASTERISK-11724: [bounty] speed control and volume control |
| ASTERISK-11725: ${EXTEN} variable is corrupted within switch() |
| ASTERISK-11726: digit extension pattern matches also dialed string, that looks like pattern itself |
| ASTERISK-11727: Advanced Options Menu |
| ASTERISK-11728: Allow user to hear greetings |
| ASTERISK-11729: Option to disable "forward to another user" |
| ASTERISK-11730: Deadlock in chan_local |
| ASTERISK-11731: high SIP call volume locks Asterisk 1.4.19rc3 |
| ASTERISK-11732: * segfault when not using DONT_OPTIMIZE flag |
| ASTERISK-11733: Asterisk crashes if MeetMe(..., i) is called and "/var/spool/asterisk/meetme" does not exist |
| ASTERISK-11734: [patch] DNS SRV lookups causing re-registration problems |
| ASTERISK-11735: unanswered calls lose the information for userfield and accountcode |
| ASTERISK-11736: CDR_ODBC don't log the first call when it needs to reconnect |
| ASTERISK-11737: VoicemailMain Hangs, asterisk goes to 100% cpu |
| ASTERISK-11738: Sip channel is not deleted while agent logoff with active call. |
| ASTERISK-11739: Originate call is already answered even if destination channel is still ringing |
| ASTERISK-11740: [patch] Taiwanese / Chinese in say.c |
| ASTERISK-11741: valgrind options change |
| ASTERISK-11742: Unable to record Speex |
| ASTERISK-11743: SIP reinvite record-route problem after hangup |
| ASTERISK-11744: Asterisk gives up on registration after receiving 408 Timeout response once |
| ASTERISK-11745: [patch] chan_skinny call control problems with 7921 |
| ASTERISK-11746: callgroup and pickupgroup don't work for 7921G phone |
| ASTERISK-11747: [patch] code documentation |
| ASTERISK-11748: IAX2 out of threads - spinning 100% when frame delivery to * is slow |
| ASTERISK-11749: [patch] Global wrapuptime for shared members across queues |
| ASTERISK-11750: [patch] improved proxy challenge support by propagating user credentials |
| ASTERISK-11751: Detect DTMF mode of SIP provider whenever is changed |
| ASTERISK-11752: Warning while compiling asterisk 1.6-beta7 |
| ASTERISK-11753: DSP_FEATURE_DTMF_DETECT issues in asterisk 1.6 and asterisk-addons 1.6 |
| ASTERISK-11754: chan_iax2.c:6699 socket_read: Out of idle IAX2 threads for I/O, pausing! |
| ASTERISK-11755: [patch] chan_mgcp NCS PacketCable patch |
| ASTERISK-11756: [patch] There is no voice channel establishing between GoogleTalk and asterisk |
| ASTERISK-11757: cdr_addon_mysql doesnt log the date/time with since 1.6 beta's |
| ASTERISK-11758: Speed dials don't work on 7921G Phone |
| ASTERISK-11759: "switch" keyword erases ${EXTEN} variable |
| ASTERISK-11760: more code documentation |
| ASTERISK-11761: make asterisk-addons-1.6.0-beta2 issue |
| ASTERISK-11762: [patch] call-limit is not an error, it should be logged as just a warning (it's a VERY TINY change) |
| ASTERISK-11763: null pointer in chan_skinny when 'regcontext' used |
| ASTERISK-11764: [patch] Voicemail accepts an invalid mailbox as valid when using realtime |
| ASTERISK-11765: Neither Flite nor Cepstral TTS works with Asterisk 1.6 |
| ASTERISK-11766: Optional timeout parameter is not optional |
| ASTERISK-11767: the Authenticate app does not authenticate my second call |
| ASTERISK-11768: [patch] Permit dialplan to control if more digits should be entered |
| ASTERISK-11769: BLF's/Hints not working properly on some phones due an incorrect NOTIFY being sent. |
| ASTERISK-11770: Generated DTMF (from features) broken with some SIP providers |
| ASTERISK-11771: [patch] Race condition in manager interface. |
| ASTERISK-11772: core dump in in find_user |
| ASTERISK-11773: iLBC compile error |
| ASTERISK-11774: Asterisk crashes after timeout / redirect / hangup when directly parking a call via AMI interface |
| ASTERISK-11775: [sound] Hello, good morning/afternoon/evening promts |
| ASTERISK-11776: chan_zap deferred dialling, bad-condition causes no WARN/DEBUG output |
| ASTERISK-11777: Asterisk crashes everytime i try to dial a realtime peer. e.g.DIAL(SIP/peer/number,60,tTwW) |
| ASTERISK-11778: Asterisk crash and receive lot of 'Bad request protocol' error |
| ASTERISK-11779: [patch] downloaded iLBC code yields warning on compilation; halts dev-mode builds |
| ASTERISK-11780: Cannot keep authentication over http -> manager |
| ASTERISK-11781: Crash within a few seconds of starting |
| ASTERISK-11782: Codec negotiation failure |
| ASTERISK-11783: CDR written with incorrect uniqueid |
| ASTERISK-11784: Asterisk is rejecting the translation of codec, we're using G279 Digium Licensed |
| ASTERISK-11785: [patch] AsyncAGI: AGI action with invalid channel closes AMI connection |
| ASTERISK-11786: Dial(<channel>|<time>|m) terminates MOH before call is answered on early "Call <> left from hold>" |
| ASTERISK-11787: chan_sip.c: realtime_peer function make crash on version 1.4.19 and 1.6.0Beta7.1 |
| ASTERISK-11788: astgenkey creates world-readable private keys |
| ASTERISK-11789: STRPTIME does not respect time dayligh saving time |
| ASTERISK-11790: Asterisk 1.4.19 console (asterisk -r) - Segmentation fault |
| ASTERISK-11791: Asterisk becomes unresponsive after locking 1.4.19 |
| ASTERISK-11792: [patch] channel groups dont work |
| ASTERISK-11793: lost packets when using Polycom 550 and ulaw codec to outbound sip also using ulaw, if change polycom to alaw no lost packets |
| ASTERISK-11794: Changes in #0012115 break Voicemail multi-language capability |
| ASTERISK-11795: [patch] T.38 support for chan_h323 with t.38 - t30 conversion |
| ASTERISK-11796: [patch] Allow called parties to continue after the caller has hung up |
| ASTERISK-11797: [patch] Add dialtone detection to chan_zap FXO devices |
| ASTERISK-11798: Background() application ignores the channel language |
| ASTERISK-11799: constants to implement call features not exported |
| ASTERISK-11800: AUTHREQ gets ignored / ACKed |
| ASTERISK-11801: parkandannounce_exec uses sizeof wrongly on array of strings to announce |
| ASTERISK-11802: AddQueueMember and RemoveQueueMember randomly lock up asterisk. |
| ASTERISK-11803: [patch] CLI wrapper for the asterisk manager written in perl |
| ASTERISK-11804: [patch] some fixes to astcli |
| ASTERISK-11805: route information is lost for incoming calls before sending BYE |
| ASTERISK-11806: asterisk handles 488 responses after initial invite incorrectly |
| ASTERISK-11807: SIP channel hungs with a message |
| ASTERISK-11808: Can´t trap SIGHUP signal in eagi script |
| ASTERISK-11809: realtime queue crash with |
| ASTERISK-11810: [patch] Add a flag to VoiceMail to force write to 'default' context |
| ASTERISK-11811: [patch] : support for new Nortel i2050 softphone |
| ASTERISK-11812: variables utilized with Dial command and parameter L, appear to be neglected when utilizing a chan_local combo |
| ASTERISK-11813: NOTICE[3754]: chan_iax2.c:6691 socket_read: Out of idle IAX2 threads for I/O, pausing! |
| ASTERISK-11814: [patch] Added functionality to astcli |
| ASTERISK-11815: [patch] Restrict what is printed during -rx output |
| ASTERISK-11816: I always get a warning like below |
| ASTERISK-11817: [patch] Russian language problem with languageprefix=yes |
| ASTERISK-11818: chan_sip added a ';' if SIP_URI_OPTIONS was "" |
| ASTERISK-11819: Asterisk crash I don´t know why |
| ASTERISK-11820: Spelling of "existent" |
| ASTERISK-11821: asterisk thinks that my ericsson T39 is headset |
| ASTERISK-11822: Asterisk deadlocks with tc400b board. |
| ASTERISK-11823: T.38 after transfer call |
| ASTERISK-11824: sip.conf.sample mixes nonlocal and local in description of domain= |
| ASTERISK-11825: Asterisk replies with "No compatible codecs" for t.38 Re-INVITE |
| ASTERISK-11826: chan_h323 doesn't respect rtp packetization settings |
| ASTERISK-11827: gratuitous, incorrect database lookups when loading a realtime SIP peer/user |
| ASTERISK-11828: Realtime Dial Crash |
| ASTERISK-11829: PrivacyManager not detecting anonymous SIP calls |
| ASTERISK-11830: configs/res_ldap.conf.sample is out of date |
| ASTERISK-11831: Bug in ldapsearch result check |
| ASTERISK-11832: error when setting user-defined variable in the dialplan |
| ASTERISK-11833: Single duplex channels in 1.4.19 |
| ASTERISK-11834: [patch] astcli tab completion |
| ASTERISK-11835: AGISTATUS is not set to the correct value |
| ASTERISK-11836: app_meetme.c : compilation failed in dev-mode. |
| ASTERISK-11837: SLA not working in 1.6.0. Beta7.1 |
| ASTERISK-11838: Gtalk channel allocation fails after large number of buddy |
| ASTERISK-11839: Jabber gtalk not connecting anymore |
| ASTERISK-11840: Handle wrong at offer/answer in sdp in media description(m=) |
| ASTERISK-11841: channel_find_locked: Avoided initial deadlock for... |
| ASTERISK-11842: Run generators from zap when available instead of depending on received audio |
| ASTERISK-11843: Asterisk negotiates only T.38 when answering even if the other end offers audio |
| ASTERISK-11844: [patch] Asterisk throws warnings about not being able to update the user's LDAP record when it's actually successful |
| ASTERISK-11845: Asterisk does not match trunk. |
| ASTERISK-11846: DTMF needed to enter long distance PIN codes not recognized by telco, DTMF duration appears to be 0 ms |
| ASTERISK-11847: All mysql related addons can't be installed on FEDORA CORE 8 x86_64 |
| ASTERISK-11848: pri loop TestClient/TestServer fails: server SEND DTMF 8 |
| ASTERISK-11849: Corrupted sound or call recording via IAX and GSM |
| ASTERISK-11850: [patch] Change remaining DEBUG logs to ast_debug in channel.c |
| ASTERISK-11851: Autofill does not work from general section for realtime queues |
| ASTERISK-11852: asterisk doesn't play MOH files randomly when "random=yes" is specified |
| ASTERISK-11853: Local channel didn't get answered from answered child SIP channel |
| ASTERISK-11854: Soft Hangup un called channel of Dial() returns NO ANSWER |
| ASTERISK-11855: app_voicemail doesn't correctly handle multiple names assigned to a mailbox |
| ASTERISK-11856: It happens when the calls reach more than 100 calls |
| ASTERISK-11857: join empty categories are not the same as leavewhenempty ones, joinempty probably wrong |
| ASTERISK-11858: autosupport script additions |
| ASTERISK-11859: [patch] British Telecom line testing causes segfault on chan_zap.c |
| ASTERISK-11860: [patch] SS7 support for chan_zap is missing clean-up code |
| ASTERISK-11861: configuration option toneduration not working with zaptel 1.4.10 |
| ASTERISK-11862: Building chan_vpb in 1.4 fails with libvpb-dev 4.2.26-1 |
| ASTERISK-11863: [patch] chan_zap incorrectly passes information between sections in users.conf in 1.4 |
| ASTERISK-11864: Unistim call to VoiceMailMain application fails |
| ASTERISK-11865: app_chanspy group broken in 1.4.19 |
| ASTERISK-11866: [patch] Value of AGISTATUS is set to SUCCESS even though the agi program could not be executed |
| ASTERISK-11867: My asterisk crashes randomly with very low volume |
| ASTERISK-11868: Unable to receive caller id from Coral SDBX PBX to Asterisk |
| ASTERISK-11869: probleme Sip et ldap |
| ASTERISK-11870: AGISTATUS is set to SUCCESS although the agi program returns not 0 |
| ASTERISK-11871: Can't add 0 in front of function result within if() block of AEL |
| ASTERISK-11872: regcontext |
| ASTERISK-11873: Channel deadlock in ast_autoservice_stop / pbx_find_extension |
| ASTERISK-11874: SetCallerPres() does not consume the same format as ${CALLINGPRES} channel variable returns |
| ASTERISK-11875: [patch] loopback switch does not allow modification of extension |
| ASTERISK-11876: [patch] Registration when using fromuser@fromdomain:pass@host is incomplete |
| ASTERISK-11877: [patch] message about number of new and old messages not properly conjugated in Russian |
| ASTERISK-11878: AMI Agents command tries to access null pointer to cid.cid_num for agent user channel |
| ASTERISK-11879: Deadlock on reload asterisk |
| ASTERISK-11880: IC_NEW + IC_ACK recreates reflective amplification DoS |
| ASTERISK-11881: IAX2 dial string behavior patch |
| ASTERISK-11882: Missing CDRs |
| ASTERISK-11883: [patch] Answer preferred codec only in SIP response |
| ASTERISK-11884: Segfault when using realtime queues and 'strategy' is not NULL |
| ASTERISK-11885: Asterisk crashes with no explanation |
| ASTERISK-11886: placing an outbound SLA call on hold crashes system |
| ASTERISK-11887: Queue language is not inherited or available to answering interface |
| ASTERISK-11888: Queue variables are not created correctly when setXXXvar is defined for the queue. |
| ASTERISK-11889: "502 Bad Gateway" is translated as "NO ANSWER" instead of "FAILED", why? |
| ASTERISK-11890: State problems on queues with Direct and Local Agents |
| ASTERISK-11891: The Variables section of an AMI AgentConnect (etc.) event is truncated by two characters. |
| ASTERISK-11892: asterisk locks after p2p sip channel bridge |
| ASTERISK-11893: Moh not start |
| ASTERISK-11894: [patch] ChanSpy and ExtenSpy applications don't accept a colon-delimited list of groups |
| ASTERISK-11895: [patch] ChanSpy doesn't correctly move to the next channel when * is pressed |
| ASTERISK-11896: [patch] Channelredirect hangs until hangup |
| ASTERISK-11897: Macro Interferes with GoSub Stack |
| ASTERISK-11898: autoframing only works if SDP contains rtpmap |
| ASTERISK-11899: SayNumber() hangs up channel when asked to say a number >= 1,000,000,000. |
| ASTERISK-11900: It happens when the amount of calls isw over 200 |
| ASTERISK-11901: Changing port number on SIP trunk is not reflected in Asterisk |
| ASTERISK-11902: Jitterbuffer used with Queues causes large audio delay |
| ASTERISK-11903: Reception of RFC2833 (DTMF) when dtmfmode=info stops RTP transmission |
| ASTERISK-11904: Behavior difference between commands JACK() and Set(Manipulate()=on). |
| ASTERISK-11905: handle_response_peerpoke floods asterisk cli with notices, cli crashes |
| ASTERISK-11906: [patch] MFC/R2 support for chan_zap |
| ASTERISK-11907: Astmanproxy crashes on free |
| ASTERISK-11908: transfer number of caller to callee when doing attended transfer |
| ASTERISK-11909: Deadlock during processing of IAM |
| ASTERISK-11910: RTP is not reinvited back to Asterisk when a native bridge is broken by AMI redirect on both channels |
| ASTERISK-11911: I get hundreds of this messager per minute, and it makes the computer unresponsive |
| ASTERISK-11912: IAX Peers are in state UNREACHABLE |
| ASTERISK-11913: [patch] comments on sample files refers to unexistent README file |
| ASTERISK-11914: Internal Calls cut after 20 seconds |
| ASTERISK-11915: Version 114606 becomes "mute" after a few thousdand calls |
| ASTERISK-11916: Current SVN version fails to compile |
| ASTERISK-11917: Asterisk crash randomly while doing transfer |
| ASTERISK-11918: The channel SIP fails to close the call properly and the 'h' extensions never executes |
| ASTERISK-11919: res_ldap (or chan_sip?) produces SegFault |
| ASTERISK-11920: [patch] Add CLI command for "sip qualify peer PEERNAME" |
| ASTERISK-11921: [patch] safe_asterisk can be started multiple times. |
| ASTERISK-11922: [patch] Possible race condition in pbx_lua call to ast_channel_datastore_find() |
| ASTERISK-11923: [patch] vmail.cgi problem with forwarding without selecting a message |
| ASTERISK-11924: linux/armel calls the OSARCH "linux-gnueabi" |
| ASTERISK-11925: Reloading asterisk causes a deadlock |
| ASTERISK-11926: [patch] team/seanbright/res-jabber-astobj2 patches |
| ASTERISK-11927: app_voicemail fails to load with monolithic asterisk build: "res_adsi.so" not found |
| ASTERISK-11928: Macro STORE passed too many arguments when using ODBC Storage |
| ASTERISK-11929: dringXrange defaults to 0, not 10 |
| ASTERISK-11930: Bad name when calling between two extensions |
| ASTERISK-11931: Chanspy crash when using spygroup |
| ASTERISK-11932: deadlock using hardware transcoding card |
| ASTERISK-11933: [patch] Janitor for getvar_helper threadsafe |
| ASTERISK-11934: REGISTER "deadlock" between SPA's and Asterisk/Non-SPA Interoperability (again) |
| ASTERISK-11935: I get the error below |
| ASTERISK-11936: [patch] Congestion feature request |
| ASTERISK-11937: Line name issue on Cisco 7940 and 7960 phones |
| ASTERISK-11938: Non freed channel then kernel panic |
| ASTERISK-11939: SIP channel protocol illegally reverses direction when ringing channel AMI redirected (to parked channel) |
| ASTERISK-11940: App Queue extension/context copying |
| ASTERISK-11941: [SIP/TCP] received 302 Moved Temporarily via TCP, but invite sent via UDP |
| ASTERISK-11942: Queue handling problems with IAX/possibly others |
| ASTERISK-11943: asterisk crashes when a sip peers in realtime tries registration with provider |
| ASTERISK-11944: Segmentation Fault in autoservice_run |
| ASTERISK-11945: Crash is sip_destroy |
| ASTERISK-11946: [patch] Realtime queue callers |
| ASTERISK-11947: 100% CPU when connecting console to SIP FXS |
| ASTERISK-11948: Crash in chan_iax2 |
| ASTERISK-11949: [patch] Problems with NOTIFY due to Asterisk sending wrong CALL-ID and duplicate sip: tag in header of NOTIFY |
| ASTERISK-11950: after agentdump, abandon message no longer sent to queue_log |
| ASTERISK-11951: [patch] Manager SIPnotify command |
| ASTERISK-11952: switch => Realtime/context@failmy doesn't accept variables in context |
| ASTERISK-11953: SendText() ignores headers added via SIPAddHeader() |
| ASTERISK-11954: Loss of audio after 2 seconds of P2P RTP bridging |
| ASTERISK-11955: Big latency (up to 3 sec) when call waiting enabled |
| ASTERISK-11956: no call indication for modern SE phones |
| ASTERISK-11957: [branch] Receiving Text from res_jabber |
| ASTERISK-11958: Changing of RTP SSRC between early-session and actual call session |
| ASTERISK-11959: Incoming REMOTE_HOLD on Zap is always passed to the bridged channel |
| ASTERISK-11960: [patch] Asterisk crashes when retrieving password from LDAP |
| ASTERISK-11961: Agent cannot unlock a channel after transfer calls |
| ASTERISK-11962: warning message when loading dialplan if using #exec |
| ASTERISK-11963: IAX forcejitterbuffer + maxjitterbuffer |
| ASTERISK-11964: PgSQL Search on SIP table for SIP Channel |
| ASTERISK-11965: Asterisk MUST add VIA "received" parameter always and send replies there even if "nat=no" |
| ASTERISK-11966: Voicemail flag g() not working |
| ASTERISK-11967: [patch] Don't work -dev-mode and appropriate fix for app_fax |
| ASTERISK-11968: Fails to load if sock not specified in config |
| ASTERISK-11969: Asterisk crashes when a call comes in from a Mediatrix 2102 |
| ASTERISK-11970: [patch] Add missing description of 'B' option |
| ASTERISK-11971: Possible deadlock between chan_sip and app_meetme |
| ASTERISK-11972: MYSQL() from dialplan appears to "eat" the first field retrieved from a MySQL Table |
| ASTERISK-11973: Lock errors on the CLI |
| ASTERISK-11974: Add ctl-d to exit |
| ASTERISK-11975: During a blind transfer BLINDTRANSFER is set incorrectly |
| ASTERISK-11976: Segmentation faults, looks to be related to issue 9520 (expire_register) |
| ASTERISK-11977: After patch 0012402 to output add symbol ^@ |
| ASTERISK-11978: crash on call transfer |
| ASTERISK-11979: [patch] Add AgentRingNoAnswer event to app_queue |
| ASTERISK-11980: DTMF issues in 1.4.19 with missing digits |
| ASTERISK-11981: [patch] get_ilbc_source.sh script "read" command needs an argument |
| ASTERISK-11982: [patch] Add manager users from users.conf to the users LIST |
| ASTERISK-11983: Asterisk lock |
| ASTERISK-11984: duplicate function calls to get_header() in find_call() |
| ASTERISK-11985: New LDAP Schema update |
| ASTERISK-11986: With no basedn set, a basedn of "asterisk" is not used like indicated |
| ASTERISK-11987: If binding as anonymous, no need to save password or use a default |
| ASTERISK-11988: [patch] Will not bind anonymously |
| ASTERISK-11989: SIP PUBLISH implementation |
| ASTERISK-11990: Asterisk locks up and cannot process incoming SIP messages on 1.4.20-rc1 |
| ASTERISK-11991: * not returning to 'a' extension |
| ASTERISK-11992: * not returning to 'a' extension |
| ASTERISK-11993: SIP INVITE msg without "From" field crashes asterisk 1.2.28 if pedantic=yes |
| ASTERISK-11994: Always transcoding G729 so slin |
| ASTERISK-11995: potential segfault if tmpchan is NULL |
| ASTERISK-11996: Make a Call from Gtalk to specific SIP user through Asterisk |
| ASTERISK-11997: contrib/initd/rc.debian.asterisk non working since rc2 patch |
| ASTERISK-11998: protocol or version not honoured. |
| ASTERISK-11999: DTMF transmission is randomly ignored during SIP-SIP calls |