| [..] |
| ASTERISK-00001: SIP re-invites failing with certain proxies |
| ASTERISK-00002: Libpri crashes asterisk |
| ASTERISK-00003: Spooled Calls with Application/Data combination |
| ASTERISK-00004: System stops processing calls about one per day. Built from 07/18/03 CVS |
| ASTERISK-00005: [patch] DTMF CLIP not supported by Asterisk (DE,SW,NL) |
| ASTERISK-00006: ztdummy causes IAX2 trunks not to work.. |
| ASTERISK-00007: chan_sip.c:1278: too few arguments to function `ast_rtp_new' |
| ASTERISK-00008: DMTF *,# are not detected via sip info, they're got as '1' |
| ASTERISK-00009: Native 64k zaptel channel briding |
| ASTERISK-00010: DTMF inconsistency somewhere between rtp.c and channel.c? |
| ASTERISK-00011: AGI channel_status failure |
| ASTERISK-00012: app_voicemail2 became a bit silent, lately |
| ASTERISK-00013: AGI Record File crashes Asterisk |
| ASTERISK-00014: asterisk leaves zombie mpg123 |
| ASTERISK-00015: app_directory does not handle greetings in VoiceMail2 directory structure |
| ASTERISK-00016: [patch] Update MOH to deal with more than just MP3s |
| ASTERISK-00017: Using variables in extension-designations |
| ASTERISK-00018: app_queue fails to respond to dialed extension while at head of queue |
| ASTERISK-00019: Festival speech at double speed |
| ASTERISK-00020: Asterisk doesn't start if it can't connect to MySQL for cdr_mysql |
| ASTERISK-00021: [request] SIP Session Timer Support |
| ASTERISK-00022: Registration-requests originated from chan_sip don't use the source-IP specified as "bindaddr". |
| ASTERISK-00023: [request] External Auth and Change Password apps for Voicemail(2) |
| ASTERISK-00024: Speex support incomplete for IAX2 |
| ASTERISK-00025: Variable Substitution would be nice |
| ASTERISK-00026: Voicemail redirection to the same mailbox is not considered as a voicemessage in your mailbox |
| ASTERISK-00027: Make attended Call transfer to work with ATA186 too with a simple update of the call transfer function. |
| ASTERISK-00028: Voice message redirection to another extension change the message file format when attached to the mail |
| ASTERISK-00029: [request] Better handling for multiple queues with common members |
| ASTERISK-00030: app_queue Dynamic member add, also adding unique call ID |
| ASTERISK-00031: ADSI function keys kill numeric keys in Voicemail2 |
| ASTERISK-00032: Calls into Asterisk to SIP phone get dropped when put on hold |
| ASTERISK-00033: Call placed via IAX2 to busy Zap channel, is instantly hung up. |
| ASTERISK-00034: Asterisk AGI leaves zombies |
| ASTERISK-00035: agentcallbacklogin does not use the optional context to find a ext |
| ASTERISK-00036: Audio is not passed to an agent who is logged in using the AgentCallbackLogin |
| ASTERISK-00037: app_queue Dynamic member add, agent logged do not survive a reload |
| ASTERISK-00038: app_queue returns circuit-busy when using fewest call routing |
| ASTERISK-00039: Asterisk sends the wrong secret |
| ASTERISK-00040: chan_iax does not initialize all phtread_mutexs it uses |
| ASTERISK-00041: app_queue alternate queue routing not working correctly |
| ASTERISK-00042: Several minor improvements to the agent channel and application. |
| ASTERISK-00043: g.729 licenses do not release when used in Voicemail |
| ASTERISK-00044: [patch] hide blocked caller IDs |
| ASTERISK-00045: cdr_mysql and MySQL socket |
| ASTERISK-00046: Using AgentCallbackLogin with optional @context does not log agent out with just '#' |
| ASTERISK-00047: Add these notes to configs/voicemail.conf.sample |
| ASTERISK-00048: [patch] Set caller ID blocking from dialplan |
| ASTERISK-00049: Manager interface issue |
| ASTERISK-00050: Out-of-service or stuck B channels on PRIs not handled correctly |
| ASTERISK-00051: Apparent bug in chan_sip.c register timer routines causes crash |
| ASTERISK-00052: parked calls on timeout fail redial if macros used |
| ASTERISK-00053: Debugging output fix for Calling Party number presentation |
| ASTERISK-00054: soft hangups on SIP channels seem to crash asterisk |
| ASTERISK-00055: When you call from X100 to Phone/phone0 |
| ASTERISK-00056: chan_zap won't compile without libpri. |
| ASTERISK-00057: "501 Not Implemented" replies cause unclean and unexpected SIP channel closure |
| ASTERISK-00058: moh for an agents on a ZAP channel uses the musiconhold from zapata.conf NOT the musiconhold from agent.conf |
| ASTERISK-00059: cosmetic changes |
| ASTERISK-00060: [patch] comment out unused variables |
| ASTERISK-00061: [patch] fix un-initialized variables |
| ASTERISK-00062: [patch] code fix-ups and clean-ups |
| ASTERISK-00063: Festival speech speed double in recent CVS again |
| ASTERISK-00064: Crashes when AgentCallbackLogin type agent is called and answers the call (before # is pressed) |
| ASTERISK-00065: queue app seems to crash regularly |
| ASTERISK-00066: Links for Private Asterisk Users Public Pages |
| ASTERISK-00067: [request] Make * Codes Into Applications (DND,*69,*8# etc) |
| ASTERISK-00068: do_monitor logs strange warning spuriously |
| ASTERISK-00069: Festival eats DTMF |
| ASTERISK-00070: [tracking] VoiceMail2 always seems to download scripts to the Aastra PT390. |
| ASTERISK-00071: NoAudio passed with calls terminating on app_queue via IAX/IAX2 Bridged to a Zap channel |
| ASTERISK-00072: Voicemail doesn't properly handle >100 messages |
| ASTERISK-00073: SIP blind xfer to parking extension |
| ASTERISK-00074: "database get" returns extra characters |
| ASTERISK-00075: Voicemail2 App doesn't properly record audio |
| ASTERISK-00076: Strip quotes from callerid being sent to phones. |
| ASTERISK-00077: stderr |
| ASTERISK-00078: IAX2/speex, problems during the first several seconds of a call |
| ASTERISK-00079: chan_sip Failed to grab lock, trying again... |
| ASTERISK-00080: Add indication definitions for Norway |
| ASTERISK-00081: Add indications for Norway |
| ASTERISK-00082: New calls to strdupa have been added |
| ASTERISK-00083: wcfxs driver crashes when compiled with gcc295 |
| ASTERISK-00084: Loading wct1xxp module when wcfxs is configured gives an error, but then works |
| ASTERISK-00085: Monitor application doesn't synchronize audio legs in file |
| ASTERISK-00086: iax.conf to SIP call bug |
| ASTERISK-00087: Pseudo timer interface w/o hardware interfaces |
| ASTERISK-00088: Show codec names instead of numbers |
| ASTERISK-00089: Manager Events Don't Seem Right |
| ASTERISK-00090: poor speech quality when bridging two channels on Zap interface via pbx_spool |
| ASTERISK-00091: X-Lite DTMF irregularities in rtp.c/channel.c |
| ASTERISK-00092: Setup cvs commit mails to a mailinglist |
| ASTERISK-00093: [patch] PBX Regex Matcher enhanced |
| ASTERISK-00094: [patch] Outgoing Limit (similar to h.323 outgoinglimit) |
| ASTERISK-00095: [patch] Fixes and enhancements for app_enumlookup |
| ASTERISK-00096: [patch] Stack corruption in srv.c |
| ASTERISK-00097: Chan_sip gets stuck in a loop when congestion is indicated. |
| ASTERISK-00098: "Virtual Extensions" for teleworkers and hot desks.. |
| ASTERISK-00099: Callback-agents don't pass caller-ID. |
| ASTERISK-00100: [patch] Cisco-like NAT trick for outbound SIP connections |
| ASTERISK-00101: festival 1.4.3 patch |
| ASTERISK-00102: CallerID appends character in h323 chan driver |
| ASTERISK-00103: Feature Request: Console Timestamp |
| ASTERISK-00104: VoiceMail2 doesn't play vm-password after you enter mailbox number |
| ASTERISK-00105: Agents can hear caller, but caller can't hear agent SIP |
| ASTERISK-00106: Adding an option for double # transfers |
| ASTERISK-00107: Format-names are not used or used incorrectly in some places. |
| ASTERISK-00108: ast_waitfor(transferer, ms) seems not to wait any more |
| ASTERISK-00109: [patch] say.c digit speaking cannot deal with * and # |
| ASTERISK-00110: App Monitor should pad outbound audio with silence. |
| ASTERISK-00111: [patch] rtp.c complains about G723.1 frames |
| ASTERISK-00112: Typing *8# on keypad blocks Asterisk. |
| ASTERISK-00113: SIP rfc2833 DTMF eaten before reaching SIP extension. |
| ASTERISK-00114: Joining two connections to an asterisk application and then hanging up leaves a channel open forever |
| ASTERISK-00115: Manager: Logoff should work without Authentication |
| ASTERISK-00116: res_monitor crashes when parking a monitored call |
| ASTERISK-00117: Manager: Implement Unique ID for Actions |
| ASTERISK-00118: [request] Manager: Open Ended Actions Create |
| ASTERISK-00119: [patch] Manager Protocol Ambiguous For Long Responses [Post 1.0] |
| ASTERISK-00120: cdr_mysql connecting to specified socket doesn't work |
| ASTERISK-00121: Manager: Events for T1 Alarms |
| ASTERISK-00122: B channels restart on PRI |
| ASTERISK-00123: chan_h323 fails to link. |
| ASTERISK-00124: RingAall strategy, using mix of agentlogin, agentcallbacklogin does not ring first avaiable after retry time expires |
| ASTERISK-00125: add the option to prevent native bridge - canreinvite |
| ASTERISK-00126: TDMXX does not correctly pass DTMF after Dial Command |
| ASTERISK-00127: Asterisk doesn't respond to 407 for BYE |
| ASTERISK-00128: Asterisk doesn't respond to 407 for BYE |
| ASTERISK-00129: SPEEX / ILBC not working (Xten) |
| ASTERISK-00130: [request] SIMPLE support |
| ASTERISK-00131: PRI channels die under load |
| ASTERISK-00132: Hangup Not Properly Detected / Strange Line State |
| ASTERISK-00133: wct1xxp + wcfxs + ZAPATA_NET creates crashes/instability |
| ASTERISK-00134: Flush CDR without closing it |
| ASTERISK-00135: Asterisk crashes when plaing file in own application |
| ASTERISK-00136: Asterisk crashes when checking for callerid in own application |
| ASTERISK-00137: MeetMeCount patch to allow count to be assigned to variable |
| ASTERISK-00138: More problems when using a bindaddr in chan_sip on multi-homed hosts. |
| ASTERISK-00139: Change pthread_ functions in VPB channel driver |
| ASTERISK-00140: [request] VAD (voice activity detection + comfort noise) support |
| ASTERISK-00141: Build against some installs of MySQL fails |
| ASTERISK-00142: Playing 'pbx-transfer' does not stream audio to agent line, after using a '#' transfer |
| ASTERISK-00143: qualify = y crashes Grandstream phones |
| ASTERISK-00144: Agent Groups and Strategy leastrecent |
| ASTERISK-00145: app_voicemail2 doesn't compile when mysql support enabled |
| ASTERISK-00146: [request] Support other Speex bitrates |
| ASTERISK-00147: Manager Event Join |
| ASTERISK-00148: Another callerid "" |
| ASTERISK-00149: mem leak chan_zap |
| ASTERISK-00150: Update app_directory to support MySQL (like vmail2) and Last and/or First option switches |
| ASTERISK-00151: Format shown as number, not name for authenticated iax2 calls |
| ASTERISK-00152: Mediatrix 1204 confused by its From: being used as the To: on BYE |
| ASTERISK-00153: [patch] Enhanced voicemail features |
| ASTERISK-00154: multiple SIP registrations |
| ASTERISK-00155: Support native SIP transfer feature |
| ASTERISK-00156: persistent SIP registration |
| ASTERISK-00157: ENUM Lookups and 482 loopbacks |
| ASTERISK-00158: Be a bit smarter about version updates |
| ASTERISK-00159: Be smarter about locating MySQL for app_voicemail2 |
| ASTERISK-00160: AgentCallBackLogin is logged in but doesn't answer is never retried |
| ASTERISK-00161: chan_h323 fails to compile in current CVS. |
| ASTERISK-00162: An app_setcidnum would come in handy |
| ASTERISK-00163: sip-h323 call coredumps latest cvs version |
| ASTERISK-00164: app_voicemail2 load/unload fix |
| ASTERISK-00165: [patch] Voicemailmain2 causes localtime_r() to return inconsistent results |
| ASTERISK-00166: agentcallbacklogin creates memory creep |
| ASTERISK-00167: ackcall on agentlogin does not work on a masquerade, with ':' groups |
| ASTERISK-00168: DTMF tones on remote IVRs are to short to be detected |
| ASTERISK-00169: Minor issue in "channels/h323/Makefile". |
| ASTERISK-00170: Two people tried to transfer each other to the parking extension at the same time |
| ASTERISK-00171: chan_zap fails to build on latest CVS |
| ASTERISK-00172: chan_zap fails to build on latest CVS |
| ASTERISK-00173: [patch] Handling of Q.931 "sending complete" messages |
| ASTERISK-00174: SIP From: header not RFC compliant |
| ASTERISK-00175: pbx_outgoing should take multiple channels |
| ASTERISK-00176: WARNING[5126]: File chan_sip.c, Line 2216 (__transmit_response): Unable to determine sequence number from '' |
| ASTERISK-00177: Creeping memory |
| ASTERISK-00178: Proposed feature: app_input |
| ASTERISK-00179: AgentLogin, Hard hangup on a call from the queue B4 doing a *, freezes the Zap channel the agent is logged in on |
| ASTERISK-00180: [patch] asterisk -rx "database show" produces short results |
| ASTERISK-00181: Queue grabs call when no agents |
| ASTERISK-00182: Call in queue with agents configured but no answer asterisk 'dies' |
| ASTERISK-00183: TCP/TLS support for SIP |
| ASTERISK-00184: Digit exit to a context fails if client in the Q, entered the Q from a call spool |
| ASTERISK-00185: chan_sip keeps sockets open after trying a REGISTER to a non-responsive peer |
| ASTERISK-00186: app_dial go on in context (next prio after connection) patch |
| ASTERISK-00187: cdr update patch |
| ASTERISK-00188: [patch] new app "Dialtone" Patch for res_indications.c |
| ASTERISK-00189: [patch] module for posting cdr through manager interface |
| ASTERISK-00190: Asterisk Memory Leak? |
| ASTERISK-00191: chan_sip doesn't forward outgoing caller-ID numbers beginning with "**". |
| ASTERISK-00192: [patch] QueueTimeout Patch |
| ASTERISK-00193: [patch] Add a "ParkedCalls" action to astman |
| ASTERISK-00194: [patch] using disa with a sip client causes seg fault |
| ASTERISK-00195: G729 (and other codecs?) cannot be specified on a per-peer basis |
| ASTERISK-00196: PATCH: makefile fix for ASTMODULESDIR |
| ASTERISK-00197: [patch] Add DESTDIR support to makefiles |
| ASTERISK-00198: Fails to send Busy or Congestion to PRI |
| ASTERISK-00199: iax,iax2 and h323 support in ENUM |
| ASTERISK-00200: DTMF and early audio via chan_sip blocks Asterisk if set to inband |
| ASTERISK-00201: DTMF not passed via chan_sip with early audio |
| ASTERISK-00202: Inbound IAX2 calls block asterisk and/or crash it. |
| ASTERISK-00203: 2 agent Zap channels are bridged when ackcall=always, using 2 different : groups |
| ASTERISK-00204: sip session timer support/chan_sip continues to stream data even after SIP clients have terminated or crashed |
| ASTERISK-00205: [patch] Add support for H.323 NAPTR records to EnumLookup |
| ASTERISK-00206: T-Mobile (Rochester NY market) roaming on Microcell Hamilton/Toronto is unable to successfully call TE410P-terminated PRI |
| ASTERISK-00207: app_voicemail2 doesn't build quite right with MySQL support |
| ASTERISK-00208: The extensions continue giving tone, but not dialing at all. |
| ASTERISK-00209: [patch] Added MOH to meetme with single party waiting |
| ASTERISK-00210: Forwarded voicemail messages don't trigger an "Event: MessageWaiting" in Manager |
| ASTERISK-00211: [patch] Announce callers position in queue every x seconds / service level reporting / queue timeouts / hold time estimation |
| ASTERISK-00212: Asterisk not following username in URI of Contact header |
| ASTERISK-00213: New IAX peers are not qualified until complete Asterisk restart |
| ASTERISK-00214: New Pending agents are created with last config values read from the agent.conf |
| ASTERISK-00215: Fixes a nit with exiting from remote console mode |
| ASTERISK-00216: [request] SIP signalling on TCP and UDP |
| ASTERISK-00217: SIP Queue members on otherside of IAX trunks |
| ASTERISK-00218: [PATCH]Transfers for chan_h323 |
| ASTERISK-00219: Feature Request: initiate an event/script when a hangup occurs |
| ASTERISK-00220: Agentlogin misinterprets '#' transfer key as '#' acknowledged when call from Q is bridged |
| ASTERISK-00221: Penalty is does respected with Pending agents using ':' groups |
| ASTERISK-00222: Application "VarLength" |
| ASTERISK-00223: Remote console does not show entire version string |
| ASTERISK-00224: ACD terminology |
| ASTERISK-00225: blacklists |
| ASTERISK-00226: [patch] Add relative substring length to SubString |
| ASTERISK-00227: SetMusicOnHold does not appear to change Zap channel MOH outbound |
| ASTERISK-00228: Regular expressions matching against the * character seem to fail |
| ASTERISK-00229: [patch] Move common DNS resolver code to separate module |
| ASTERISK-00230: Agentlogin does not respect digittimeout when call from Q is masquerade bridged doing a '#' transfer to ext with > 2 digits |
| ASTERISK-00231: Correct most compile warnings |
| ASTERISK-00232: [patch] Built-in application to reset CDR data |
| ASTERISK-00233: [patch] CGI/Perl Queue Viewer Script |
| ASTERISK-00234: [patch] SayNumber-s in spanish |
| ASTERISK-00235: printf should be ast_log in pbx.c |
| ASTERISK-00236: chan_local does not correctly sandbox variables |
| ASTERISK-00237: CDR not logging.. |
| ASTERISK-00238: WARNING[5126]: File chan_sip.c, Line 1442 (sip_alloc): Unable to create RTP session: Too many open files |
| ASTERISK-00239: Fix some logic errors and compile warnings |
| ASTERISK-00240: New zaptel cvs screws fxs card |
| ASTERISK-00241: [patch] Add patch to zap channels to hide caller ID |
| ASTERISK-00242: changing password in Voicemail2 crashes * |
| ASTERISK-00243: Not compiling on 2.4.20-20.9 kernel (RH 9) |
| ASTERISK-00244: [patch] compile warning removal in app_queue.c |
| ASTERISK-00245: WaitMusicOnHold does not timeout ???? |
| ASTERISK-00246: [patch] # out of VoiceMailMain2 "Password?" Prompt |
| ASTERISK-00247: [patch] Change ZapBarge into ZapScan to allow monitoring of active channels |
| ASTERISK-00248: [patch] put bridge option in [general] and userbyalias option |
| ASTERISK-00249: [patch] Fix bug in pbx.c for "show application ..." too-small buffers |
| ASTERISK-00250: Pointer not checked before FREEing |
| ASTERISK-00251: [patch] wav49 format isn't compatible with Windows Media Player |
| ASTERISK-00252: [patch] error in agi record from functionality |
| ASTERISK-00253: [patch] To check the language of the sound file being played |
| ASTERISK-00254: chan skinny makes asterisk exit if it can't find skinny.conf |
| ASTERISK-00255: [patch] NAT via H.323 |
| ASTERISK-00256: [patch] Modulated ringtones (for Aus) |
| ASTERISK-00257: [patch] Asterisk doesn't respond to 407 for REFER |
| ASTERISK-00258: chan_skinny doesn't re-read config on reload |
| ASTERISK-00259: socket is not closed after stop now |
| ASTERISK-00260: Asterisk crashes on phoning a Cisco 30VIP |
| ASTERISK-00261: [patch] Random data on port 2000 will SEGV asterisk |
| ASTERISK-00262: Log files build until 2GB and Asterisk fails to run |
| ASTERISK-00263: Latest CVS checkout for zaptel breaks WCFXS |
| ASTERISK-00264: [patch] app_voicemail says "goodbye" twice after exceeding the max. amount of login-attempts. |
| ASTERISK-00265: [patch] fix a few compile warnings. |
| ASTERISK-00266: [patch] Add SAY TIME to AGI |
| ASTERISK-00267: Hang Up not detected |
| ASTERISK-00268: [patch] Fix a variety of compile warnings on RedHat |
| ASTERISK-00269: Fixes some minor things in the G.723 code |
| ASTERISK-00270: [patch] Be better about finding mpg123 |
| ASTERISK-00271: Suggestion: Standardize parameter passing and config file parsing |
| ASTERISK-00272: [patch] Request App: SayTime |
| ASTERISK-00273: # from MGCP during Ringtone causes Asterisk to stop responding |
| ASTERISK-00274: [patch] AGI Zombies on systems with NPTL |
| ASTERISK-00275: [patch] Move SIGCHLD handler to prevent more zombies |
| ASTERISK-00276: Dial returns 1 on success when using chan_local |
| ASTERISK-00277: Update zapata.conf.sample with more info |
| ASTERISK-00278: * sends image data to chan_capi |
| ASTERISK-00279: chan_h323 dies if the gatekeeper isn't available |
| ASTERISK-00280: asterisk-announce mailing list archives inaccessible |
| ASTERISK-00281: [patch] Add notes for LEN variable type in README.variables |
| ASTERISK-00282: Use documented method of including all symbols in parent |
| ASTERISK-00283: Action: Originate is broken, latest cvs |
| ASTERISK-00284: [request] Saving agent / queue-state more permanently. |
| ASTERISK-00285: [patch] use md5-hashes as secret in addition to plain-text passwords |
| ASTERISK-00286: [patch] problem with SERVICE ACKNOWLEDGE IE? |
| ASTERISK-00287: app_voicemailcheck? |
| ASTERISK-00288: Module use count never decreases |
| ASTERISK-00289: voicemail2 segfaults on digit playback |
| ASTERISK-00290: D-channel failure does not result in call rejection |
| ASTERISK-00291: UNIQUEID variable not set correctly |
| ASTERISK-00292: TE410P: Double/missed interrupt detected |
| ASTERISK-00293: [iaxtel] entry in iax.conf must be the last entry |
| ASTERISK-00294: No sound on PSTN --> */PRI connection |
| ASTERISK-00295: Unused files |
| ASTERISK-00296: After today's change in CVS, chan_h323 no longer loads. |
| ASTERISK-00297: [patch] Portuguese in say.c |
| ASTERISK-00298: [patch + redesign request] Non-english support in app_voicemail |
| ASTERISK-00299: Latest CVS causes D channel handling to go south with TE410P |
| ASTERISK-00300: Adding more licenses to G.729 cause repeating loop of warning messages |
| ASTERISK-00301: "context" directive ignored in modem.conf for i4l |
| ASTERISK-00302: leastrecent and fewestcalls methods keep on trying on one member channel, even that it is not answering, |
| ASTERISK-00303: Init script for systems using /etc/init.d is brutal |
| ASTERISK-00304: [patch] Let 'show codec n' display all relevant codecs |
| ASTERISK-00305: The attached voicemail2 box crashes asterisk |
| ASTERISK-00306: iLBC audio quality degraded within the last week CVS |
| ASTERISK-00307: Directory application doesn't allow FXO hangup until prompt is done |
| ASTERISK-00308: Pointing moh to an existing but empty directory causes fd leak |
| ASTERISK-00309: [patch] Startup script |
| ASTERISK-00310: Xlite SIP codec problems |
| ASTERISK-00311: [patch] Changed KINCLUDES definition in Zaptel Makefile |
| ASTERISK-00312: [patch] Problems building libpri with late versions of GCC |
| ASTERISK-00313: [PATCH]Add Add/RemoveQueueMember to manager interface |
| ASTERISK-00314: S100U gets stuck in an unusable state |
| ASTERISK-00315: incominglimit=1 doesn't work |
| ASTERISK-00316: Zap interfaces ring on restart |
| ASTERISK-00317: Make app_festival less chatty |
| ASTERISK-00318: Improve build slightly |
| ASTERISK-00319: Remove yet more warnings |
| ASTERISK-00320: There's an extraneous newline in app_directory's documentation |
| ASTERISK-00321: [patch] New minor issues |
| ASTERISK-00322: [patch] accept locks file descriptor |
| ASTERISK-00323: sample.call (/var/spool/asterisk/outgoing) doesn't record dialled number in MySQL |
| ASTERISK-00324: Queues dial the extension but the phone is not ringing. |
| ASTERISK-00325: Calls Droping |
| ASTERISK-00326: outgoing limit in chan_sip not working as described |
| ASTERISK-00327: No audio with CVS 09/28/03 |
| ASTERISK-00328: Recent CVS code breaks with Grandstream Budgetel Phones |
| ASTERISK-00329: pbx_wilcalu uses printf where ast_log should be used |
| ASTERISK-00330: setlanguage function doesnt work |
| ASTERISK-00331: Can't install cdr_mysql.so with make target |
| ASTERISK-00332: cvs complains about files in stdtime |
| ASTERISK-00333: [PATCH] Fix mutexes in chan_sip |
| ASTERISK-00334: DoS attack: SIP "BYE" messages create channels, fill file descriptors |
| ASTERISK-00335: res_adsi.c has description of "Call Parking Resource" |
| ASTERISK-00336: [patch] Added dtmf debug CLI command |
| ASTERISK-00337: [patch] postgreSQL backend for app_voicemail2.c |
| ASTERISK-00338: Asterisk fails to re-open logfiles after rotation |
| ASTERISK-00339: [patch] make sip show channel display channels that start with a channel ID |
| ASTERISK-00340: [patch] retrieve_voicemail_from_mysql.pl |
| ASTERISK-00341: [PATCH] Voicemail: start numbering of email messages from 1 instead of 0 |
| ASTERISK-00342: 'logger reload' CLI command |
| ASTERISK-00343: * can't decode DTMF properly with Feature Group D trunks on incoming calls |
| ASTERISK-00344: chan H.323 can't close channel if call wasn't answered |
| ASTERISK-00345: app_mp3 doesn't use zaptel for clocking |
| ASTERISK-00346: [request] Delayed delete of voicemail |
| ASTERISK-00347: [patch] SRV lookups broken |
| ASTERISK-00348: [patch] Various ENUM fixes |
| ASTERISK-00349: SIP address fields too small. |
| ASTERISK-00350: Cisco 7960 and DTMF |
| ASTERISK-00351: my diff for Freebsd 5.1 gcc 3.2 |
| ASTERISK-00352: IAX with different codecs on endpoints |
| ASTERISK-00353: Memory Leak? |
| ASTERISK-00354: app_festival causes channel errors w/IAX |
| ASTERISK-00355: Digium cards lock up when using dialing out over sip |
| ASTERISK-00356: [request] Add outbound SIP Proxy |
| ASTERISK-00357: Rhino Channel Bank Appears To Break DTMF |
| ASTERISK-00358: Spaces after extension number cause error for Dial |
| ASTERISK-00359: SIP URIs are inappropriately shortened |
| ASTERISK-00360: [patch] app_cut |
| ASTERISK-00361: Caller ID does not work on Wildcard X101P FXO card |
| ASTERISK-00362: Transfers on IAX lose context |
| ASTERISK-00363: [PATCH] add announce option to app_dial |
| ASTERISK-00364: Compile errors (warning only on some compilers) |
| ASTERISK-00365: SIP doesn't honor "anti-exgirlfriend" extension |
| ASTERISK-00366: Outgoing queue context ignored |
| ASTERISK-00367: astman not refreshing |
| ASTERISK-00368: [request] vm symbolic link |
| ASTERISK-00369: [patch] User defined ring cadences |
| ASTERISK-00370: Calls on hold in "call waiting" state are dropped if user hangs up on new caller |
| ASTERISK-00371: new patch file for freebsd. |
| ASTERISK-00372: [patch] adjusting Makefile for cdr_pgsql makes Debian users happy |
| ASTERISK-00373: Reorder includes and add ast_ prefix to strdupa |
| ASTERISK-00374: System compiler does not look in /usr/local |
| ASTERISK-00375: Really clean up editline, too |
| ASTERISK-00376: MusicOnHold stops when no RTP is going from remote party |
| ASTERISK-00377: After unloading ztdummy module access to /proc/interrupts will be crashed |
| ASTERISK-00378: [addition] Postgresql schema |
| ASTERISK-00379: H323 core using the specified versions of openh323 and pwlib |
| ASTERISK-00380: [request] Documentation for Algorithm Details for Echo Cancellers |
| ASTERISK-00381: Voicemail2 cuts off characters in emailbody= setting |
| ASTERISK-00382: Voicemail2 email timezones not appropriately set |
| ASTERISK-00383: [request] Channel silence detection and hangup |
| ASTERISK-00384: IAX2- PUBLIC vs. NAT doenst like qualify = yes in iax.conf |
| ASTERISK-00385: [patch] expire old voicemail messages |
| ASTERISK-00386: [patch] Build Asterisk on FreeBSD |
| ASTERISK-00387: [patch?] README.channels |
| ASTERISK-00388: Unable to create thread from manager using originate |
| ASTERISK-00389: asterisk.pid reports incorrect pid when asterisk is forked. |
| ASTERISK-00390: Wrong error message - iax.conf instead of sip.conf |
| ASTERISK-00391: [patch] Implement system command through the manager interface |
| ASTERISK-00392: [patch] AM/PM wrong at midnight |
| ASTERISK-00393: [patch] ComboBox hostname in Gastman |
| ASTERISK-00394: [patch] manager and secretary in the directory |
| ASTERISK-00395: [patch] ast_cli() limited to 4096 characters |
| ASTERISK-00396: [patch] rxgain/txgain ingnored for outgoing calls |
| ASTERISK-00397: Calls from IAX to Zap PRI have audio quality problems |
| ASTERISK-00398: Gotoif condition is not as expected |
| ASTERISK-00399: SayUnixTime() reports 12am as 12pm |
| ASTERISK-00400: [patch] gastman segfaults |
| ASTERISK-00401: [patch] addmailbox2 script for creating VM2 mailboxes |
| ASTERISK-00402: [patch] CLI command 'help <command>' always produces help for !<command> |
| ASTERISK-00403: tonezones.h - for Austria / Germany |
| ASTERISK-00404: SetGlobalVar seems to create a local variable instead of changing the global |
| ASTERISK-00405: [patch] incominglimit fix for call waiting |
| ASTERISK-00406: Segmentationfault caused by "Originate" through manager interface |
| ASTERISK-00407: Add CRT / SecureCRT native terminal type |
| ASTERISK-00408: Setting default language for SIP channels does not work |
| ASTERISK-00409: app_voicemail2 says "SQL init" during startup, even when no problem |
| ASTERISK-00410: [patch] Allow mp3 streams to be played for MOH |
| ASTERISK-00411: callerid field in iax.conf effects inbound callerid and not outbound callerid |
| ASTERISK-00412: Route lookup, better soundcard integration |
| ASTERISK-00413: parentheses cannot be used within application arguments |
| ASTERISK-00414: Some documentation to app_dial announce option |
| ASTERISK-00415: [patch] extensions.conf IAXTel pattern error |
| ASTERISK-00416: [patch] Hangup Cause |
| ASTERISK-00417: Inbound IAXTEL calls do not work |
| ASTERISK-00418: [patch] h.323 with G.729 for openh323 12.0 |
| ASTERISK-00419: * crashes once in a week at this function: ast_waitfor |
| ASTERISK-00420: SIP Call Seq Error (SIP/2.0 481 Invalid CSeq Number) |
| ASTERISK-00421: ztdummy device won't load - unresolved symbol zt_.... |
| ASTERISK-00422: phone still ringing after picking up netmeeting |
| ASTERISK-00423: save dialplan erases [globals] definitions |
| ASTERISK-00424: Show application dial doesn't document announce option |
| ASTERISK-00425: app_cut.c re-introduces strdupa -- again! |
| ASTERISK-00426: No warnings given when include => non-existant-context |
| ASTERISK-00427: More minor build issues |
| ASTERISK-00428: chan_agent should have the same capabilities as the channel agent logged in on |
| ASTERISK-00429: [request] Call monitoring/trapping from console |
| ASTERISK-00430: New status command for number of calls / group |
| ASTERISK-00431: Need a link on Digium web site to external Asterisk resources |
| ASTERISK-00432: [patch] SendDTMF lacks error handling, undocumented |
| ASTERISK-00433: dies on multiple ringback tones |
| ASTERISK-00434: [patch] Wrong handling of 407 response retransmissions |
| ASTERISK-00435: TE410p requiring a complete power down on the server. |
| ASTERISK-00436: chan_skinny: DTMF doesn't relay properly with 7910 |
| ASTERISK-00437: [patch] Minor cleanup and better MacOS X support |
| ASTERISK-00438: [PATCH] cannot build mysql-vm-routines into voicemail2 |
| ASTERISK-00439: [patch] Added User fields to the CDR |
| ASTERISK-00440: cdr_pgsql.c needs more debugging output |
| ASTERISK-00441: [patch] Forward of the accountcode over IAX2 |
| ASTERISK-00442: PGSQL (and MySQL ?) may not log every CDR |
| ASTERISK-00443: make install does not copy hours.gsm to sounds/digits directory |
| ASTERISK-00444: [patch] Postgresql CDR Config sample |
| ASTERISK-00445: make depend does no update stdtime/.depend |
| ASTERISK-00446: [patch] If mpg123 is mpg321, refuse to load MusicOnHold |
| ASTERISK-00447: SIP_CODEC variable no longer sets codecs as expected |
| ASTERISK-00448: Onhook messaging (VMWI) doesn't work on TDM400P FXS modules |
| ASTERISK-00449: [patch] More sophisticated expression parsing |
| ASTERISK-00450: [patch] WAV format updates |
| ASTERISK-00451: Incorrect child handling |
| ASTERISK-00452: callerid= in iax.conf does not function when incoming call has blank callerid |
| ASTERISK-00453: [patch] directory compatible with Voicemail2 data stored in mysql |
| ASTERISK-00454: [patch] Unload modules at shutdown/restart |
| ASTERISK-00455: [patch] an ${ACCOUNTCODE} variable to asterisk |
| ASTERISK-00456: [patch] format_g723.c |
| ASTERISK-00457: Documentation for app_meetme |
| ASTERISK-00458: -DBUSYDETECT_TONEONLY AND -DBUSYDETECT_COMPARE_TONE_AND_SILENCE |
| ASTERISK-00459: ADSIProg fails |
| ASTERISK-00460: [patch] Campon feature |
| ASTERISK-00461: Unresolved symbols in zaptel.o |
| ASTERISK-00462: [patch] verbose level |
| ASTERISK-00463: implicit declaration of function `echo_can_traintap' |
| ASTERISK-00464: No ringing tone when using H323 |
| ASTERISK-00465: reload from "remote console" will crash asterisk |
| ASTERISK-00466: OOPS in Zaptel driver from latest CVS |
| ASTERISK-00467: [request] Endpoints Bracket option needs way of adding brackets around endpoints that define IP address after @ sign for MGCP |
| ASTERISK-00468: [patch] Configure sendmail command in voicemail2 |
| ASTERISK-00469: DateTime order is wrong |
| ASTERISK-00470: RTP problems with Cisco IP Phones (SIP) |
| ASTERISK-00471: Problems with threeway calling when both destinations are PSTN |
| ASTERISK-00472: [patch] OpenH323 bug on logical channel opening? |
| ASTERISK-00473: [patch] Passing g723.1 packets can result in infinite loop |
| ASTERISK-00474: voicemail 2 forwarding issue |
| ASTERISK-00475: [patch] New description of application cut |
| ASTERISK-00476: codec 64 wrongly identified: IAX2 connection refused |
| ASTERISK-00477: [patch] cdr_pgsql is not careful enough with CDR data |
| ASTERISK-00478: zonedata for NZ |
| ASTERISK-00479: [patch] SIP forwarding |
| ASTERISK-00480: Order of entries in iax.conf causes failures |
| ASTERISK-00481: Dial option H doesn't acutally hang up if the call has been answered |
| ASTERISK-00482: [patch] app_sayunixtime.c needs to answer the channel if it hasn't already |
| ASTERISK-00483: [patch] When tranfering call to park or other extension, person you transfer gets prompt if you type in invalid extension |
| ASTERISK-00484: [patch] Add more useful info to mgcp.conf sample |
| ASTERISK-00485: [patch] app_voicemail2 exit priority+101 when no mailbox is found |
| ASTERISK-00486: DTMF outbound on i4l |
| ASTERISK-00487: [patch] for extension selection based on distinctive ring for inbound calls via FXO |
| ASTERISK-00488: app_readdigits.c ReadDigits(max) -- read some digits, terminated by # |
| ASTERISK-00489: [request] Limit calls going over an interface. |
| ASTERISK-00490: [patch] Handling of a 302 Redirect should be configurable |
| ASTERISK-00491: ExtraChannel in transfer causes crash |
| ASTERISK-00492: Makefile doesn't do uname -m on FreeBSD |
| ASTERISK-00493: when transferring a sip client to another sip client the transferring client gets the voicemailbox of the transferred client |
| ASTERISK-00494: [patch] ast_callerid_parse() typo |
| ASTERISK-00495: Memory leak probably in chan_local |
| ASTERISK-00496: [patch] memory leak in cli |
| ASTERISK-00497: H323 documentation has an inconsistency which causes confusion |
| ASTERISK-00498: cdr_mysql.conf not found error |
| ASTERISK-00499: Documentation required on new Round-Robin Zap features |
| ASTERISK-00500: [patch] add support for sending calls to different contexts based on the distinctive ring |
| ASTERISK-00501: http://bugs.digium.com crashes firebird 0.6.1 for FreeBSD |
| ASTERISK-00502: [patch] There are compile warnings from wct4xxp.c |
| ASTERISK-00503: Incompatible with uClibc |
| ASTERISK-00504: ZAP BUSY DETECT |
| ASTERISK-00505: [patch] New application app_campon |
| ASTERISK-00506: Asterisk generates incorrect 200 response to REGISTER |
| ASTERISK-00507: Transfer doesn't work after setting someone on hold |
| ASTERISK-00508: echotraining=yes in zapata.conf makes dtmf to be unreliable on x100p |
| ASTERISK-00509: [patch] Minor fix for indications for Norway |
| ASTERISK-00510: [patch] Minor fix for indications for Norway |
| ASTERISK-00511: VM2 plays "goodbye" file twice in a row on failed login attempt |
| ASTERISK-00512: invalid/incorrect tone definitions in chan_vpb.c |
| ASTERISK-00513: T extension never gets called and timeout doesn't seem to timeout on time |
| ASTERISK-00514: "NO ANSWER" in CDR |
| ASTERISK-00515: Asterisk will block if you have register statments in sip.conf and you reload config. |
| ASTERISK-00516: * should report and not die on SIGXFSZ |
| ASTERISK-00517: Pressing 8 during voicemail login before msg play blocks * |
| ASTERISK-00518: SIP Transfers - REFER & Replaces |
| ASTERISK-00519: app_voicemail2 beeps twice when invent_message is used. |
| ASTERISK-00520: app_voicemail2 continues to play instructions when recording greetings if # pressed. |
| ASTERISK-00521: ${SIPDOMAIN} needs explanation of what it does. |
| ASTERISK-00522: pcm.c / chan_alsa.c -- asterisk crashes with: Assertion val < pcm->buffer_size failed |
| ASTERISK-00523: not Registered SIP ua reported as "busy" |
| ASTERISK-00524: ast_hangup should assing NULL to freed chan |
| ASTERISK-00525: vm2 -> vm rename fix |
| ASTERISK-00526: INSTALL_PREFIX typo in zaptel Makefile |
| ASTERISK-00527: iax2 channel is sometimes not freed. |
| ASTERISK-00528: [patch] Fix apps Makefile to include correct mysql lib path when linking |
| ASTERISK-00529: [request] [tracking] gsm codec won't build on PPC - error in Makefile |
| ASTERISK-00530: [request] AGI TDD support needs delay between receive and send |
| ASTERISK-00531: TDM400P has noise on line |
| ASTERISK-00532: Events in dialplan |
| ASTERISK-00533: Call park request gets reorder tone on sipura SPA-2000 SIP phone adapter |
| ASTERISK-00534: [patch] Be sure to unregister both apps in app_voicemail.c |
| ASTERISK-00535: [patch] voicemail emails have mime header error |
| ASTERISK-00536: Incoming calls can't be picked up by SIP and Zap unless they are forked to both. |
| ASTERISK-00537: Voicemail2 Email Content-Type |
| ASTERISK-00538: [patch] Color the application help |
| ASTERISK-00539: Onhook messaging (VMWI) doesn't work on TDM400P FXS modules |
| ASTERISK-00540: [patch] chan_vpb missing BUSY_AUST detection |
| ASTERISK-00541: [patch] Voicemail with PostgreSQL backend: NULL passwords |
| ASTERISK-00542: Seg. fault upon reload |
| ASTERISK-00543: [patch] enhanced music on hold |
| ASTERISK-00544: [patch] several improvements on app_queue/chan_agent |
| ASTERISK-00545: compilation error on Linux 2.4.9 |
| ASTERISK-00546: compilation error on Linux 2.4.9 |
| ASTERISK-00547: [patch] VoiceMail called without an arg doesn't ask for mailbox |
| ASTERISK-00548: [patch] [gnophone-0.2.4] failed tout compile with gcc 3.3.2 |
| ASTERISK-00549: [http://bugs.digium.com] cannot access /doc/documentation.html |
| ASTERISK-00550: Call Transfer & Voicemail issue |
| ASTERISK-00551: [patch] Added option 'i' to voicemail to play only instructions, skipping greeting/intro message |
| ASTERISK-00552: Re-Invite failure |
| ASTERISK-00553: Public IP * forgets to check on registered dynamic IP *: Unreachable! |
| ASTERISK-00554: Hangup not detected when calling other Zaptel PRI |
| ASTERISK-00555: [patch] Allow SIP/peer/extension as well as SIP/extension@peer |
| ASTERISK-00556: Asterisk Crash |
| ASTERISK-00557: [patch] FastStart and PSTN intercepted announcements don't work when call is originated by h323 driver |
| ASTERISK-00558: Asterisk crashes on phoning a Cisco 7910 |
| ASTERISK-00559: [patch] for app_directory to support PostgreSQL voicemail |
| ASTERISK-00560: Swissvoice IP10 via MGCP duplicates digits |
| ASTERISK-00561: [patch] Named Arguments In Apps |
| ASTERISK-00562: realtime static ignores manager.conf |
| ASTERISK-00563: [patch] Manager Interface Protocol |
| ASTERISK-00564: seg fault when calling chan_local from q |
| ASTERISK-00565: [patch] New Sip Debug output formating |
| ASTERISK-00566: permit / deny lines do not work in chan_sip.c |
| ASTERISK-00567: [patch] libiax2 in blocking mode bug |
| ASTERISK-00568: [request] zttool should show UNCONFIGURED instead of OK on unconfigured spans. |
| ASTERISK-00569: [patch] manager originate with variable and accountcode setting |
| ASTERISK-00570: Codec negotiation problem |
| ASTERISK-00571: safe_asterisk does not properly restart asterisk after a crash |
| ASTERISK-00572: [patch] cdr_unixodbc |
| ASTERISK-00573: enum.c misparses E2U+IAX2 |
| ASTERISK-00574: [patch] chan_sip doesn't reply with the "Contact:" filled in correctly in all cases. |
| ASTERISK-00575: asterisk sometimes hangs when using asterisk -r. |
| ASTERISK-00576: [patch] allows FLASH button to hangup if not in 3-way call or conference |
| ASTERISK-00577: [patch] Remove a little stack abuse from format_jpeg.c |
| ASTERISK-00578: chan_local loses variables on spooled calls |
| ASTERISK-00579: Deadlock in chan_agent |
| ASTERISK-00580: [patch] odbc-vm-routines.h |
| ASTERISK-00581: [patch] /working/ syslog support ;) |
| ASTERISK-00582: "Contact" patch from bug #580 breaks SIP hold |
| ASTERISK-00583: Memory leaks in channel drivers (so many to say exactly) |
| ASTERISK-00584: [request] More common "builtin" functionality handling |
| ASTERISK-00585: Enum regexp replacements error |
| ASTERISK-00586: Missing digit sounds for 24h format (voicemail2) |
| ASTERISK-00587: Unexpected freqency 16000 (Unable to playback() gsm prompt) |
| ASTERISK-00588: voicemail terminates when attempting to play a 0 byte gsm message |
| ASTERISK-00589: Deadlock, presumable because of ast_debug() |
| ASTERISK-00590: [patch] Spelling error in asterisk.c |
| ASTERISK-00591: When all operators are busy app_queue doesnt switch to priority +101 |
| ASTERISK-00592: Memory not released after off hook/on hook transition by mgcp endpoint. |
| ASTERISK-00593: [patch] Logger patch to send VERBOSE msg to log files |
| ASTERISK-00594: callerID generated by asterisk don't work on telephons released for France. |
| ASTERISK-00595: incominglimit increments when sip phone not registered |
| ASTERISK-00596: [patch] app_directory unixodbc support added. |
| ASTERISK-00597: [patch] Add caller ID reporting to voicemail system |
| ASTERISK-00598: IAX phones and CPU usage problem |
| ASTERISK-00599: [patch] Fix for tranlator misspelling |
| ASTERISK-00600: When using Dial() to contact a remote * via IAX, context specififed must have exten, not include |
| ASTERISK-00601: [patch] ${TIMESTAMP} 20031130-195940 |
| ASTERISK-00602: [patch] app_agi should trim end stream when using FILE RECORD |
| ASTERISK-00603: [patch] cdr_pgsql fails to compile on debian |
| ASTERISK-00604: Spool calls setvar doesn't set variables in both ends |
| ASTERISK-00605: All MGCP phones turn to constant busy - restart required |
| ASTERISK-00606: [patch] Make agi-test.agi a little less funky |
| ASTERISK-00607: [patch] GNU autoconf for libpri |
| ASTERISK-00608: IAX2 hold on asterisk->asterisk |
| ASTERISK-00609: Asterisk hangs and uses up all the cpu time appr. once a week |
| ASTERISK-00610: Increasing AST_MAX_EXTENSION makes app_dial segfault |
| ASTERISK-00611: Call Transfer & Voicemail issue |
| ASTERISK-00612: [patch] Spooler Has Many Bugs |
| ASTERISK-00613: add extension seems to be broken in latest CVS |
| ASTERISK-00614: Destroyed gateway addresses when using host=dynamic in MGCP |
| ASTERISK-00615: [patch] IAX2 native bridge - two trivial (though critical) fixes |
| ASTERISK-00616: [patch] Language forwarding on IAX2 does not happen |
| ASTERISK-00617: asterisk -rx "logger rotate" causes * to lock up |
| ASTERISK-00618: [workaround] Context: ????? not working for Action: Originate |
| ASTERISK-00619: Voicemail with postgres backend called with no context puts msgs in wrong place |
| ASTERISK-00620: [patch] crash from select() |
| ASTERISK-00621: Here is Allison saying just 'Welcome.' |
| ASTERISK-00622: [tracking] Sip to Sip call quality with latent networks |
| ASTERISK-00623: [patch] Can't transfer calls from H.323 phone |
| ASTERISK-00624: app_sayunixtime.c reports bogus out of memory error |
| ASTERISK-00625: [patch] terminology d4 framing also refered to as sf or superframe |
| ASTERISK-00626: kernel 2.6.0 support request. |
| ASTERISK-00627: [request] Barge and Whisper the Agent [from admin view] |
| ASTERISK-00628: [patch] Change of help text for meetme |
| ASTERISK-00629: [patch] Fixed typo, minor changes |
| ASTERISK-00630: [patch] Typo fix in "show application transfer" |
| ASTERISK-00631: [patch] app_hasnewvoicemail uses AST_SPOOL_DIR instead of ast_config_AST_SPOOL_DIR |
| ASTERISK-00632: [patch] app_voicemail.c should include libpq-fe.h not postgresql/libpq-fe.h |
| ASTERISK-00633: [patch] cdr_odbc doesn't properly escape some strings. |
| ASTERISK-00634: [patch] Improve ENUM handling for national numbers (patch) |
| ASTERISK-00635: DNIS routing ignores "i"nvalid extension in dialplan |
| ASTERISK-00636: Add extension "e" for error handling |
| ASTERISK-00637: [patch] advanced directory context handling/remove deprecated app_voicemail compat |
| ASTERISK-00638: [patch] Problem saying time |
| ASTERISK-00639: Sometimes 0-size audio files are left around |
| ASTERISK-00640: dialplan needs to be completely configurable from extensions.conf |
| ASTERISK-00641: [request] allowing gsm codec should not override allow= ordering |
| ASTERISK-00642: DateTime doesn't support language |
| ASTERISK-00643: DTMF 1 not detected on E&M circuit with Feature B signalling |
| ASTERISK-00644: [patch] ast_rtp_read doesn't retry on improperly returned recvfrom |
| ASTERISK-00645: CDR "dst" field is not updating properly |
| ASTERISK-00646: [request] implement SIP "PUBLISH" and "SUBSCRIBE" |
| ASTERISK-00647: chan_sip hangs/deadlocks on retransmits |
| ASTERISK-00648: no user@ in diaplan, * chooses random username |
| ASTERISK-00649: [patch] Remove FXS check on app_flash |
| ASTERISK-00650: [patch] Hold Time Functionality |
| ASTERISK-00651: Repetitive RELOADs make * unstable |
| ASTERISK-00652: Patch: Added Makefile config option for HDLC interface APIs. |
| ASTERISK-00653: App monitor 2 files |
| ASTERISK-00654: SDP is badly negotiated for telephone events |
| ASTERISK-00655: Rejecting call doesn't mark the channel as free |
| ASTERISK-00656: [patch] Post from ML to fix mp3 coreing during ast_moh_destroy |
| ASTERISK-00657: [patch] Reinvites are not treated as new transactions |
| ASTERISK-00658: [patch] APP_AGI will not accept multiple arguments (fix included) |
| ASTERISK-00659: [request] add date/timestamp to show channels command |
| ASTERISK-00660: using soft hangup zap emits "Use STOP NOW to shutdown Asterisk" |
| ASTERISK-00661: [request] Improve MWI to make it more immediate |
| ASTERISK-00662: [patch] Remove another generated file on "make clean" |
| ASTERISK-00663: CDR unique id |
| ASTERISK-00664: [patch] PBX loglevel fixups |
| ASTERISK-00665: [patch] mpg123 leaves core dump when stopped |
| ASTERISK-00666: [PATCH] makefile add -I/usr/local/include for ODBC on FreeBSD |
| ASTERISK-00667: [patch] Qualify and dynamic IAX peers sometimes stop working |
| ASTERISK-00668: [patch] sip debug confusion |
| ASTERISK-00669: [request] Add SIP support for RFC3264 |
| ASTERISK-00670: zaptel driver hard-locks kernel |
| ASTERISK-00671: [patch] allow commandline args to addmailbox script |
| ASTERISK-00672: AgentCallBackLogin gets stuck in logged in state. |
| ASTERISK-00673: [workaround] first ringback when forwarded to a SIP extension is jerky |
| ASTERISK-00674: TE410P only receives audio |
| ASTERISK-00675: [workaround] Using switch => with IAX2 causes block |
| ASTERISK-00676: [patch] Error When forwarding voicemail messages with vmail.cgi |
| ASTERISK-00677: wav49 recoding doesn't playback |
| ASTERISK-00678: [request] Voicemail Advanced Options |
| ASTERISK-00679: [request] Record and Send Voicemail |
| ASTERISK-00680: [patch] language is not supported in datetime(). |
| ASTERISK-00681: [patch] hangup during initiation of 3-way call |
| ASTERISK-00682: [patch] Video negotiation error, with fix |
| ASTERISK-00683: Unable to bridge CAPI calls |
| ASTERISK-00684: [patch] Add 'show voicemail users' CLI command to app_voicemail |
| ASTERISK-00685: [patch] mgcp.conf.sample in cvs has bad line wrap |
| ASTERISK-00686: ztmonitor no longer shows VU meter with -v option |
| ASTERISK-00687: [patch] MGCP interoperability with Cisco IADs |
| ASTERISK-00688: [request] AgentCallbackLogin Shell Script |
| ASTERISK-00689: AGI: User hangup causes asterisk crash |
| ASTERISK-00690: Fax detected, but no fax extension |
| ASTERISK-00691: Zaptel answering on a call proceeding |
| ASTERISK-00692: app_agi leaves open pipes, * crashes depending on system's ulimit |
| ASTERISK-00693: iax2 trunking not working for dynamic hosts |
| ASTERISK-00694: masqurade whacks availability of variables set in spool file |
| ASTERISK-00695: Change Directory Application to allow two parameters |
| ASTERISK-00696: New DSP routines cause SIP fax problems |
| ASTERISK-00697: Segfault in h323 |
| ASTERISK-00698: [patch] res_monitor fails if no file name given |
| ASTERISK-00699: * dumps core when loading chan_iax if udp/5036 is already open |
| ASTERISK-00700: [patch] AMD64 support and some misc fixes |
| ASTERISK-00701: TE410p Stop generrating interrupts |
| ASTERISK-00702: TE410p/Zaptel/chan_zap Stops passing dialed digits |
| ASTERISK-00703: [patch] app_festival needs patch for big endian systems |
| ASTERISK-00704: [patch] Fix format of 'zap show channels' |
| ASTERISK-00705: [patch] Cancel voicemail |
| ASTERISK-00706: [patch] Custom User-Agent |
| ASTERISK-00707: The diff for festival-1.4.3 has a bug |
| ASTERISK-00708: Crash after "stop now" comand |
| ASTERISK-00709: Random crashes when SIP protocol is used |
| ASTERISK-00710: Random crashes (openh323 bug?) |
| ASTERISK-00711: Crash: uptime 10 hours, 40 simultaneously calls |
| ASTERISK-00712: Crash in app_agi |
| ASTERISK-00713: Crash: uptime 14 hours, 50 simultaneously calls |
| ASTERISK-00714: Crash after 50 simultaneously calls |
| ASTERISK-00715: Crash (no additional info) |
| ASTERISK-00716: Crash |
| ASTERISK-00717: AES code won't build on ppc |
| ASTERISK-00718: [TDM400P] Ring signal frequency not compatible with some devices |
| ASTERISK-00719: [workaround] Asterisk crash after a few consecutive reload |
| ASTERISK-00720: asterisk crash at start up if chan_zap.o failed to load |
| ASTERISK-00721: [patch] cdr_odbc not built with default unixODBC install |
| ASTERISK-00722: SIP over VPN having local network = remote network |
| ASTERISK-00723: make process creates .h files too late |
| ASTERISK-00724: [patch] Sip 30X-reply support |
| ASTERISK-00725: [patch] app_festival fails to respect usecache config setting |
| ASTERISK-00726: [patch] parameters to Contact: header stripped off, breaks SNOM phone support |
| ASTERISK-00727: [patch] [VM] Allow custom ADSI FDN and SEC codes in voicemail.conf |
| ASTERISK-00728: wrapuptime ignored with AgentCallBackLogin |
| ASTERISK-00729: pgsql includes not found on debian sid |
| ASTERISK-00730: [patch] Better checking to avoid chan_zap segfault |
| ASTERISK-00731: CVS sounds config error |
| ASTERISK-00732: T400P not passing more than 3 digits on T1 E&M trunks |
| ASTERISK-00733: Grandstream SIP phones drop digits, set to SIP DTMF Mode INFO |
| ASTERISK-00734: Fax extensions broken all calls dialout get redirected to fax extension |
| ASTERISK-00735: [request] pbx_builtin_getvar_helper for UNIQUEID |
| ASTERISK-00736: sip calls hang up when dtmfmode=info or rfc with asterisk phone |
| ASTERISK-00737: [patch] Allow dynamic conferences to be joined by channels without dynamic flag |
| ASTERISK-00738: [patch] Allow ENUM support to be left out when building Asterisk |
| ASTERISK-00739: [patch] New command PlayInterruptibleTones - like PlayTones only interrupted by digits |
| ASTERISK-00740: [patch] Generation of tones that have one tone amplitude modulated by another |
| ASTERISK-00741: [patch] Adjust indications.conf.sample for correct modulated indications for Australia (needs patch in bug 746) |
| ASTERISK-00742: [patch] indications.conf.sample entry for South Africa |
| ASTERISK-00743: Lose all audio on TE410P |
| ASTERISK-00744: [patch] Make the sample macro usage in extensions.conf actually work |
| ASTERISK-00745: [patch] Change zaptel /proc entries into something machine parsable |
| ASTERISK-00746: [patch] preliminary upgrade to app_dial.c for Privacy |
| ASTERISK-00747: [request] Voicemail doesn't properly handle >99 messages |
| ASTERISK-00748: Segfault with more that 125 sip calls. |
| ASTERISK-00749: permit/deny not parsing mask |
| ASTERISK-00750: [patch] copy permit/deny data |
| ASTERISK-00751: Netgear Router not capable of forwarding some UDP-packets |
| ASTERISK-00752: New aes* source files in CVS have DOS lineendings - older GCCs can't compile them |
| ASTERISK-00753: [patch] New sip channel - chan_sip2.c |
| ASTERISK-00754: [patch] Using Announce feature of Dial app may have unexpected side affects! |
| ASTERISK-00755: [patch] AgentCallBackLogout |
| ASTERISK-00756: GotoIfTime fails to execute context/ext/pri |
| ASTERISK-00757: [request] MeetMe musiconhold disable |
| ASTERISK-00758: DND Kills out Sip to Sip calls |
| ASTERISK-00759: Calls fail if IP Phone responds too slowly (Circuit-busy / Auto-Congested) |
| ASTERISK-00760: gastman failed to compile die to db api mis-match |
| ASTERISK-00761: omitting username in sip.conf don't let to call a cisco phone until it registers again |
| ASTERISK-00762: [patch] The "SystemCID" app -- submitting for Announcements and other uses. |
| ASTERISK-00763: mac address for TDMoE is not parsed correctly in ztd-eth.c |
| ASTERISK-00764: Show Memory Allocations hangs asterisk |
| ASTERISK-00765: Fax Not working |
| ASTERISK-00766: [patch] make originate return faster from the manager |
| ASTERISK-00767: [patch] Play a tone seconds before absolutetimeout |
| ASTERISK-00768: asterisk cli output looses synch with "-cn" |
| ASTERISK-00769: dialing own number with cisco 12SP+/VIP30 causes segfault |
| ASTERISK-00770: Software SIP phone can not register when computer has no gateway |
| ASTERISK-00771: [patch] Contact: header problems |
| ASTERISK-00772: [patch] Siemens optiPoint 400 support |
| ASTERISK-00773: [patch] Add #include to extensions.conf.sample |
| ASTERISK-00774: sip-h323 call coredumps CVS-01/11/04 |
| ASTERISK-00775: [request] permit/deny for hostnames and ranges... |
| ASTERISK-00776: 127th Conncurent SIP<->SIP call causes segfault |
| ASTERISK-00777: [patch] new help text for musiconhold app |
| ASTERISK-00778: [request] [tracking] - No sound files should be hard coded. |
| ASTERISK-00779: propagate the new userfield to cdr_pgsql |
| ASTERISK-00780: [patch] show application setcdruserfield |
| ASTERISK-00781: [patch] show application meetme |
| ASTERISK-00782: [patch] possible SIP buffer overflow |
| ASTERISK-00783: [request + patch] Dial - Announce - Stop MOH after playing |
| ASTERISK-00784: cdr-logging reports disposition=ANSWERED even if phone is not answered at destination |
| ASTERISK-00785: asterisk segfaults when doing outgoing or incoming call |
| ASTERISK-00786: [patch] app_random.c |
| ASTERISK-00787: [request + patch] Spell() and SpellPhonetic() |
| ASTERISK-00788: [patch] Update for HARDWARE file |
| ASTERISK-00789: Unload and load chan_zap.so make asterisk crash |
| ASTERISK-00790: unload chan_sip.so and press tab (or try any command) make * crash |
| ASTERISK-00791: Unload chan_sip.so and load chan_sip.so back again, make stop command unavailable |
| ASTERISK-00792: Unloading chan_sip.so doesn't unregister sip protocol and unregister sipdebug |
| ASTERISK-00793: unload chan_iaxX type command make crash, also if load back, it crash |
| ASTERISK-00794: [patch] tautological truth test in logger.c |
| ASTERISK-00795: [patch] Give deprecated chan_iax separate config file |
| ASTERISK-00796: compile * term.c make warning |
| ASTERISK-00797: logger reload cores asterisk wne using syslog support |
| ASTERISK-00798: show voicemail users for EXISTING CONTEXT show all context instead of the one speficied |
| ASTERISK-00799: [patch] logger.c to work before logchain is initialized |
| ASTERISK-00800: [patch] Misc Bugs (Look at note for more bugs) |
| ASTERISK-00801: BKW |
| ASTERISK-00802: [patch] ast_verbose() calls /always/ get logged to syslog |
| ASTERISK-00803: Valgrind complains about uninitialised values on call |
| ASTERISK-00804: [security] CLI command init keys should not echo passcode |
| ASTERISK-00805: zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
| ASTERISK-00806: Asterisk crash when call from SIP to Skinny |
| ASTERISK-00807: [patch] chan_sip typo |
| ASTERISK-00808: Mailbox full soundfile and sounds.txt diff |
| ASTERISK-00809: [design request] voicemail directory listing |
| ASTERISK-00810: [patch] New sounds and application diffs for VM and MeetMe |
| ASTERISK-00811: [patch] Default to use old DSP routines |
| ASTERISK-00812: [patch] Changes to README |
| ASTERISK-00813: [patch] app_SayUnixTime does not support saying seconds |
| ASTERISK-00814: [patch] Fix 'show voicemail users for <context>' |
| ASTERISK-00815: [request] call robot |
| ASTERISK-00816: [patch] CVSID Macro to support embedding CVS Version information in source files |
| ASTERISK-00817: [patch] Time based includes are no longer working |
| ASTERISK-00818: [patch] Don't return -1 for successful conf in meetme |
| ASTERISK-00819: make webvmail fails |
| ASTERISK-00820: [patch] OLD_DSP_ROUTINES redefined error |
| ASTERISK-00821: [zaptel] zconfig.h contains a little typo |
| ASTERISK-00822: Linux locks hard when deselecting loopback (on SMP system) |
| ASTERISK-00823: Asterisk + chan_sip + chan_capi leaks pipes |
| ASTERISK-00824: "Restart Now" issued through manager interface sends * console into convulsions |
| ASTERISK-00825: [workaround] ztdummy and musc on hold |
| ASTERISK-00826: [patch] make install should install sound files w/o execute perms |
| ASTERISK-00827: [patch] update contrib/scripts/postgres_cdr.sql to match change userfield addtion |
| ASTERISK-00828: [tracking] e1000 network card causes kernel panic when used with TE410P and possibly other Zaptel devices. |
| ASTERISK-00829: [patch] configure.ac and build.h.in |
| ASTERISK-00830: [patch] ast_manager_register doesn't check if action is already registered |
| ASTERISK-00831: [patch] astmm.c contains a small typo |
| ASTERISK-00832: [patch] IAX and IAX2 both register the same manager action |
| ASTERISK-00833: [patch] add a vasprintf replacement to astmm |
| ASTERISK-00834: [patch] Minor nitpicks to formating and verbose. |
| ASTERISK-00835: [patch] "Value" column name in app_unixodbc is a reserved word on MySQL |
| ASTERISK-00836: [patch] assume a remote console if the program name is rasterisk |
| ASTERISK-00837: [patch] Update sample meetme.conf for pins |
| ASTERISK-00838: [patch] parse ENUM NAPTR-records which use "!...!...!x" format |
| ASTERISK-00839: TE410p stops functioning in current cvs. |
| ASTERISK-00840: [patch] README: Clarify hardware needs, link Wiki, mention chan_capi |
| ASTERISK-00841: [bounty] Zaptel drivers under BSD |
| ASTERISK-00842: [bounty] Win32-based Asterisk call manager client |
| ASTERISK-00843: [bounty] incoming/outgoing limit |
| ASTERISK-00844: [patch] sip debug [peername] |
| ASTERISK-00845: [patch] you can unload chan_zap.so and reload from cli without segfault |
| ASTERISK-00846: Ever increasing module use count |
| ASTERISK-00847: [patch] Tone Italy |
| ASTERISK-00848: [patch] basic app_cepstral |
| ASTERISK-00849: RTP does not get bound to a designated IP in a multi IP linux box |
| ASTERISK-00850: rfc3389 Filling up log |
| ASTERISK-00851: Another patch for gsm Makefile and PPC |
| ASTERISK-00852: [patch] minor output nitpicks |
| ASTERISK-00853: Using G.729 codec causes asterisk to crash unless in Debug mode |
| ASTERISK-00854: [request]: documentation for undocumented features: |
| ASTERISK-00855: Asterisk threads hang |
| ASTERISK-00856: SERVICE_ACKNOWLEDGE message building problem |
| ASTERISK-00857: X100P and TDM doesn't work well with modem above 24000 |
| ASTERISK-00858: stop now flood when doing a asterisk restart |
| ASTERISK-00859: [request] asterisk as non root user... |
| ASTERISK-00860: voicemail hangs up when saving to a Cust{1,2,3,4,5} mailbox |
| ASTERISK-00861: [patch] app_festival tweaks for answer and lost sound |
| ASTERISK-00862: [patch] Documentation for the local channel |
| ASTERISK-00863: [patch] Manager.txt update |
| ASTERISK-00864: [patch] README.cdr update |
| ASTERISK-00865: [patch] Add "Lookup" command for full channel ID to manager interface |
| ASTERISK-00866: /modules/cdr_pgsql.so: undefined |
| ASTERISK-00867: [patch] Print warning on unknown option in sip.conf |
| ASTERISK-00868: after a stop gracefully using remote console, things hang |
| ASTERISK-00869: [patch] Updated zaptelrtc for 2.6.X kernels |
| ASTERISK-00870: [patch] add example and description of busycount param in zapata.conf |
| ASTERISK-00871: context error |
| ASTERISK-00872: [patch] Present "ringing" instead of music on hold when folks enter the queue (used for autoattendant) |
| ASTERISK-00873: [patch] SIP 300 Redirection Implementation |
| ASTERISK-00874: [patch] CLI sip reload command |
| ASTERISK-00875: 3way calling using flash fails with Swissvoice ip10 |
| ASTERISK-00876: [patch] patch to add peers and make faststart, h245tunneling configurable |
| ASTERISK-00877: app_festival, possible to exceed string length in code MD5Hex |
| ASTERISK-00878: app_festival connects to festival server via socket even if cache is used |
| ASTERISK-00879: [patch] doublehash patch not working under 0.7.1 |
| ASTERISK-00880: [request] turn off DTMF-muting in meetme |
| ASTERISK-00881: kphone 4 crashes asterisk with subscribe message |
| ASTERISK-00882: Sending prepended voicemail will crash * |
| ASTERISK-00883: SIP mis-directed REGISTER statements |
| ASTERISK-00884: zaptel ignoring Q931 release code 17 (busy) |
| ASTERISK-00885: cdr_pgsql hangs up asterisk if postgresql dbms goes down |
| ASTERISK-00886: deadlock undiagnosed |
| ASTERISK-00887: * crashes when flashhook is received in voicemail. |
| ASTERISK-00888: [patch] ast_readstring returns different value depending on the action |
| ASTERISK-00889: [patch] Fix for * on PPC not logging integer fields through cdr_odbc |
| ASTERISK-00890: [patch] IAX2 on-the-fly codec choosing and inbound codec fix |
| ASTERISK-00891: [patch] comment type |
| ASTERISK-00892: [patch] Fix manager CLI command output |
| ASTERISK-00893: [patch] Diff to chan_modem, three new features, dtmf inband |
| ASTERISK-00894: [patch] add userfield in mysql cdr |
| ASTERISK-00895: [patch] safe_asterisk with init script does not restart asterisk on crash |
| ASTERISK-00896: [patch] Add CLI command 'cdr mysql status' |
| ASTERISK-00897: [audit-request] audit chan_sip.c for bugs and buffer overflows. |
| ASTERISK-00898: another typo |
| ASTERISK-00899: Calls still goes thru when out of G729 licenses |
| ASTERISK-00900: case for SIG_R2 not handled in zt_call( ) in chan_zap.c |
| ASTERISK-00901: case for SIG_R2 not handled in zt_call( ) in chan_zap.c |
| ASTERISK-00902: [patch] New ztmonitor "visual" output. |
| ASTERISK-00903: Directory doesn't interoperate with postgresql voicemail interface |
| ASTERISK-00904: [request] SRV support requested for chan_iax2 |
| ASTERISK-00905: [patch] chan_sip hangs channel on blind transfer in an app_* |
| ASTERISK-00906: [patch] Enable only SIP_MYSQL_FRIENDS, not IAX2 |
| ASTERISK-00907: Dump of table structure for iaxfriends and sipfriends |
| ASTERISK-00908: [patch] wrong description for "logger rotate" after "show applications" |
| ASTERISK-00909: [request] Adding qualify support for temp peers, like mysql ones |
| ASTERISK-00910: [patch] add new mode of class to musiconhold |
| ASTERISK-00911: [patch] Send ACK on CANCEL |
| ASTERISK-00912: [patch] reg files for x-pro and x-lite and ilbc/speex to work |
| ASTERISK-00913: When host=dynamic and mailbox=## in config chan_sip::__sip_xmit gives error |
| ASTERISK-00914: "difference is XXXX ms" dropped SIP connection |
| ASTERISK-00915: deadlock for unknown reason |
| ASTERISK-00916: [patch] Add SIPFROM variable |
| ASTERISK-00917: [patch] Fixups for libpri Makefile |
| ASTERISK-00918: Change 'iax' CLI commands to 'iax1', make 'iax' equal to 'iax2' |
| ASTERISK-00919: [patch] remove nested comment in cvsid.h |
| ASTERISK-00920: Asterisk ignores codecs list from phone |
| ASTERISK-00921: [request] 'w' does not show up in "show application dial" |
| ASTERISK-00922: [post-1.0][patch] Add option to app_dial for variable communication |
| ASTERISK-00923: Fax Detected Notice on calls to outside lines |
| ASTERISK-00924: DTMF not being passed when IVR does not send "Connect" status |
| ASTERISK-00925: [patch] Various minor fixes |
| ASTERISK-00926: [patch] double double fix fix |
| ASTERISK-00927: [patch] Fix Makefile for SIP and USE_MYSQL_FRIENDS |
| ASTERISK-00928: [patch] several files do not compile with strict gcc flags |
| ASTERISK-00929: [patch] chan_sip fails into setting the ipaddr of the mysql peer (sometimes) |
| ASTERISK-00930: Sip Retransmit and problem placing receiving calls |
| ASTERISK-00931: [patch] Dial to IAX2 doesn't let you specify a port |
| ASTERISK-00932: [patch] Prioritizing of queues |
| ASTERISK-00933: [patch] Fix for warnings in mandrake startup script |
| ASTERISK-00934: [patch] app_voicemail context fixed and vmail.cgi suid fix |
| ASTERISK-00935: [patch] logger.conf.sample |
| ASTERISK-00936: [patch] show application record update help text |
| ASTERISK-00937: [patch] Show application agi - new text |
| ASTERISK-00938: libpri(q931.c) responds on CALLPROCEEDING with STATUS(wrong message) |
| ASTERISK-00939: Q921 N(S) behaves weird causing resending on E100P |
| ASTERISK-00940: PRI Cause Code 17 is not making Dial jump to n+101 |
| ASTERISK-00941: [patch] pass channel structure to cdr routines |
| ASTERISK-00942: [patch] Latest MS WAV format not supported |
| ASTERISK-00943: extensions.conf sample wildcard error |
| ASTERISK-00944: PRI Cause: No user responding (18) not handled in chan_zap.c |
| ASTERISK-00945: [patch] fix AST_LIST_INSERT_TAIL bug |
| ASTERISK-00946: [patch] cdr_pgsql logging of arbitrary fields and channel variables |
| ASTERISK-00947: Unhandled DTMF Tones |
| ASTERISK-00948: Half of AGI broken when using VERBOSE |
| ASTERISK-00949: [request] Add leave_only_sound and enter_only_sound options to meetme |
| ASTERISK-00950: SIP channel calling unregistred hosts at 0.0.0.0 |
| ASTERISK-00951: [request] Get variable in app_agi.c |
| ASTERISK-00952: Asterisk calls unregistered SIP user |
| ASTERISK-00953: memory leak |
| ASTERISK-00954: [patch] data calls through * with EuroISDN E1 |
| ASTERISK-00955: [patch] chan_vpb needs method for indicating pause and flash hook |
| ASTERISK-00956: [patch] New soundfiles |
| ASTERISK-00957: Kernel panic when running meetme with 30 legs |
| ASTERISK-00958: cannot record both sides of conversaion in meetme |
| ASTERISK-00959: [tracking] load modules in order listed in modules.conf |
| ASTERISK-00960: CLI feature include context - doesn't |
| ASTERISK-00961: several crashes with chan_h323 |
| ASTERISK-00962: Patch to add "uninstall" to Makefile |
| ASTERISK-00963: [patch] correct includes for NetBSD-current |
| ASTERISK-00964: [patch] make acl.c build correctly on FreeBSD and NetBSD |
| ASTERISK-00965: chan_aopen, chan_bestdata, etc. should not be built by default |
| ASTERISK-00966: [patch] Uninstall patch for Makefile |
| ASTERISK-00967: [request] No way to automagically log out non-answering synamic agents |
| ASTERISK-00968: [patch] No cdr logging for calls placed via /var/spool/asterisk/outgoing |
| ASTERISK-00969: segmentation fault with speex |
| ASTERISK-00970: [patch] Start/stop messages on cli 'sip reload' |
| ASTERISK-00971: [request + patch] comedian mail leaves admin menu before user is done |
| ASTERISK-00972: [request] comedian mail duplicates recorded message on prepending voice mail |
| ASTERISK-00973: [request] Voice mail broadcast list |
| ASTERISK-00974: kernel panic crash freeze caused either by manager or meetme |
| ASTERISK-00975: [patch] Manager Redirect with two parties sometimes gets second party hungup |
| ASTERISK-00976: segfault or deadlock when doing a few reloads |
| ASTERISK-00977: emailbody in voicemail.conf broken |
| ASTERISK-00978: [patch] Use correct compiler for building kernel modules |
| ASTERISK-00979: [request + files] Weather/Automation/APRS words and phrases for Allison |
| ASTERISK-00980: Apparent deadlock behaviour Zap Chan three way call after park via IAX2 |
| ASTERISK-00981: [patch] JitterBuffer should not be disabled by default |
| ASTERISK-00982: Tab completion displays incorrect completions |
| ASTERISK-00983: [patch] nicer codec debug messages |
| ASTERISK-00984: [patch] Allow enable/disable jitter buffer at run time |
| ASTERISK-00985: [request] app_disa should prompt for password, and confirm it was accepted before dialtone. |
| ASTERISK-00986: Cisco 12SP+ can not send voice, only recieve |
| ASTERISK-00987: [patch] - design suggestion: Codec names unification |
| ASTERISK-00988: [patch] A wrong path is used when installing samples |
| ASTERISK-00989: wct4xxp driver doesn't 'detect' E1 |
| ASTERISK-00990: #include does not work for changing voicemail passwords |
| ASTERISK-00991: SIP REFER message does not end with the required blank line |
| ASTERISK-00992: linux Kernels now using CONFIG_SMP instead of __SMP__ in Makefiles |
| ASTERISK-00993: chan_SIP chan_ZAP require res_musiconhold.so to load |
| ASTERISK-00994: [patch] Incorrect implementation of function ast_waitfordigit_full |
| ASTERISK-00995: Pressing flash to transfer in h323 phone crashes Asterisk |
| ASTERISK-00996: [patch] compil on OpenBSD 3.4 |
| ASTERISK-00997: can't hear ringing or busy signal when calling from within a meetme conference |
| ASTERISK-00998: exten => i does not work |
| ASTERISK-00999: [patch] ast_monitor_stop tmp[255] isn't large enough |