November 2007 Archives by date

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Starting: Thu Nov 1 00:44:20 CDT 2007
Ending: Fri Nov 30 19:35:13 CST 2007
Messages: 1756

• [asterisk-bugs] [Asterisk 0010997]: Asterisk 1.4.13 segfaults at least once daily   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009511]: Q931 inband information not interpreted as alerting signal   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011088]: IMAP client fails with: IMAP Error: Unable to create selectable TCP socket (1024 >= 1024)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011133]: app_voicemail.c's imapfolder option is not documented   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010604]: Codec options in gtalk.conf not respected   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011137]: [patch] Fix for clean project in devmode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011085]: Hint stuck on hold after attended transfer with notifyhold=yes in sip.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011138]: [patch] use ast_free() instead of free().   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010740]: [patch] Move deleted messages to a Deleted Folder   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011137]: [patch] Fix for clean project in devmode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011138]: [patch] use ast_free() instead of free().   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011138]: [patch] use ast_free() instead of free().   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010890]: When parking lot ring back times out, error is generated, line is hung up and timeout extension isn't reached.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011138]: [patch] use ast_free() instead of free().   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0009684]: Adding callgroup / pickupgroup settings for a user   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010931]: dialplan reload shoud also reload extensions.ael   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011088]: IMAP client fails with: IMAP Error: Unable to create selectable TCP socket (1024 >= 1024)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011017]: [patch] zap restart fails to generate channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010787]: Impossible to make optional macro arguments in 1.4   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011017]: [patch] zap restart fails to generate channels   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011127]: TE212P+VPM450M: VPM450M can't inizialize   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011139]: Request for ROSE IE causes connection to drop.   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011139]: Request for ROSE IE causes connection to drop.   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011127]: TE212P+VPM450M: VPM450M can't inizialize   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011139]: Request for ROSE IE causes connection to drop.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011088]: IMAP client fails with: IMAP Error: Unable to create selectable TCP socket (1024 >= 1024)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011048]: music on hold ends before new user enters   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011048]: music on hold ends before new user enters   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010605]: 'Unknown' member status in app_queue   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011127]: TE212P+VPM450M: VPM450M can't inizialize   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010936]: Crash in ast_queue_frame   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011084]: asterisk segfault   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011048]: music on hold ends before new user enters   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010936]: Crash in ast_queue_frame   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008786]: [patch] Properly handle tempates on config read/write   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010743]: Separate RTP pool for remote vs. local connections   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0011122]: astman.js variables are static, not upgrade with parameters passed into ./configure   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0011122]: astman.js variables are static, not upgrade with parameters passed into ./configure   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010783]: 0010754: [patch] Finish reading extension after user pressed # is not ok!   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010164]: Incorrect notify handling when hints contain multiple devices   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010783]: 0010754: [patch] Finish reading extension after user pressed # is not ok!   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010783]: 0010754: [patch] Finish reading extension after user pressed # is not ok!   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010743]: Separate RTP pool for remote vs. local connections   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010743]: Separate RTP pool for remote vs. local connections   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010813]: Should be able to dynamically link against libc-client for IMAP storage   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009972]: [branch] res_jabber over OpenSSL   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009972]: [branch] res_jabber over OpenSSL   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010783]: 0010754: [patch] Finish reading extension after user pressed # is not ok!   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009912]: * is looping and CPU goes at 100%   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010783]: 0010754: [patch] Finish reading extension after user pressed # is not ok!   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010783]: 0010754: [patch] Finish reading extension after user pressed # is not ok!   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009685]: chan_oss & chan_alsa in conference with zap channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009414]: [patch] app_rtsp - playback RTSP media resources   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010813]: Should be able to dynamically link against libc-client for IMAP storage   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008786]: [patch] Properly handle tempates on config read/write   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008786]: [patch] Properly handle tempates on config read/write   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010813]: Should be able to dynamically link against libc-client for IMAP storage   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0009684]: Adding callgroup / pickupgroup settings for a user   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011140]: [patch] pbx_lua.so: a lua pbx switch that allows dialplans written in pure lua   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011143]: Solaris build issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011085]: Hint stuck on hold after attended transfer with notifyhold=yes in sip.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010946]: chan_sip can generate several outstanding requests, but ignores responses   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011091]: autopark strange behaviour if canreinvite=update, nonat   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011091]: autopark strange behaviour if canreinvite=update, nonat   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010991]: [PATCH] Add some functional to SayPosition in App_Queue   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011143]: Solaris build issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010956]: 1.4.13 lockups   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011143]: Solaris build issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011146]: 'h' extension is broken in trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011143]: Solaris build issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011144]: [patch] Fix small memory leak in config file proccessing when there are included files   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011140]: [patch] pbx_lua.so: a lua pbx switch that allows dialplans written in pure lua   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010956]: 1.4.13 lockups   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011140]: [patch] pbx_lua.so: a lua pbx switch that allows dialplans written in pure lua   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010915]: Problems when doing an attended and unattended transfer with Thomson Phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009567]: Notify sent to a non-existent call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010647]: SIP Reinvite behaviour does not work as expected with certain dial() options   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008677]: Asterisk does not reinvite peer for G.711 after T.38 negotiated failed with a "488" Event   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009402]: T.38 passthrough fails if caller offers T.38 in the initial INVITE   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008736]: asterisk reinvites to G.711 after a T.38 negotiation - fax fails depending on ATA config   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009759]: T.38 Fax Outbound (revision of bug 9356)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011140]: [patch] pbx_lua.so: a lua pbx switch that allows dialplans written in pure lua   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010417]: T.38 with devices behind NAT does not work in all circumstances   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008786]: [patch] Properly handle tempates on config read/write   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008786]: [patch] Properly handle tempates on config read/write   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011066]: Asterisk SIP Connections to systems that support t38 fax detection may fail   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011147]: SayDigits seems to be broken in current trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011147]: SayDigits seems to be broken in current trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011147]: SayDigits seems to be broken in current trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011147]: SayDigits seems to be broken in current trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011147]: SayDigits seems to be broken in current trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011147]: SayDigits seems to be broken in current trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011147]: SayDigits seems to be broken in current trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011147]: SayDigits seems to be broken in current trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008786]: [patch] Properly handle tempates on config read/write   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008786]: [patch] Properly handle tempates on config read/write   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008786]: [patch] Properly handle tempates on config read/write   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009843]: [patch] /etc/init.d/asterisk is not "Linux Standard Base" compatible   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010937]: app_mixmonitor crash in ast_channel_spy_read_frame   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011151]: IMAP: Mailbox does not exist   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011152]: Over quota errors ignored   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011152]: Over quota errors ignored   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011151]: IMAP: Mailbox does not exist   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011071]: No audio on calls into queues with non-persistent agents   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011149]: Asterisk 1.4.13 Produces mutex errors and too many files open error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011149]: Asterisk 1.4.13 Produces mutex errors and too many files open error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011151]: IMAP: Mailbox does not exist   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011150]: issue 0011146 has not been fixed in trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011071]: No audio on calls into queues with non-persistent agents   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010809]: Asterisk segfaults after an attended transfer to a queue using "Eyebeam" softphone.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011150]: issue 0011146 has not been fixed in trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010532]: Some information from INVITE of Peer A not passed to INVITE of Peer B   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011150]: issue 0011146 has not been fixed in trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011150]: issue 0011146 has not been fixed in trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011034]: Segfault in app_voicemail, inside c-client   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011150]: issue 0011146 has not been fixed in trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011150]: issue 0011146 has not been fixed in trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010894]: pulsedial=no setting is runtime changed in chan_zap.c so you can't simply disable pulse dialling   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005747]: * Sends 403 Unauthorized upon reception of INFO method from a Nortel MCS 5200 sip proxy   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011152]: Over quota errors ignored   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011150]: issue 0011146 has not been fixed in trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011151]: IMAP: Mailbox does not exist   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011150]: issue 0011146 has not been fixed in trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010917]: Stop gracefully complains in _ast_pthread_mutex_unlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010702]: subscribecontext ignored   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008864]: [patch] new channel driver : chan_unistim for Nortel Unistim IP phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011151]: IMAP: Mailbox does not exist   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011151]: IMAP: Mailbox does not exist   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010926]: double call to ast_frame_free   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011148]: I need the uniqueid to show un core show channels concise   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010790]: Cannot add channel to group if using AMI Originate   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010790]: Cannot add channel to group if using AMI Originate   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010790]: Cannot add channel to group if using AMI Originate   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008864]: [patch] new channel driver : chan_unistim for Nortel Unistim IP phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008864]: [patch] new channel driver : chan_unistim for Nortel Unistim IP phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010763]: 'make asterisk.pdf' produces an unfound .sty error.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010751]: Duplicated and meaningless CDR Records   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010751]: Duplicated and meaningless CDR Records   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011071]: No audio on calls into queues with non-persistent agents   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005768]: [branch][post 1.4] LDAP Realtime driver   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010504]: Patches to build V1.4.10.1 under Solaris 10 X86   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011071]: No audio on calls into queues with non-persistent agents   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009306]: waitforsilence still timesout inappropriately   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011071]: No audio on calls into queues with non-persistent agents   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010689]: Callback Application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010689]: Callback Application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008864]: [patch] new channel driver : chan_unistim for Nortel Unistim IP phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010690]: sip peer with missing close bracket causes all subsequent sip peers not to load   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010815]: [patch] SendFAX/ReceiveFAX   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011149]: Asterisk 1.4.13 Produces mutex errors and too many files open error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010890]: When parking lot ring back times out, error is generated, line is hung up and timeout extension isn't reached.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010890]: When parking lot ring back times out, error is generated, line is hung up and timeout extension isn't reached.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0004903]: [patch] SIP over TCP project   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011155]: callback failed on atxfer from members of queue   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011155]: callback failed on atxfer from members of queue   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010926]: double call to ast_frame_free   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009256]: [patch] fix totalAnalysisTime to handle periods of no channel activity   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011149]: Asterisk 1.4.13 Produces mutex errors and too many files open error   noreply at bugs.digium.com
• [asterisk-bugs] [LibSS7 0011156]: added the generic address through to extension.conf like the charge number   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009815]: [patch] major fix for SIP video codec negociations   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011155]: callback failed on atxfer from members of queue   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011070]: cdr_sqlite3_custom backend error when logging to db   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011070]: cdr_sqlite3_custom backend error when logging to db   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009256]: [patch] fix totalAnalysisTime to handle periods of no channel activity   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011158]: [patch] chan_unistm.c Memory leak & Deadlock & Crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010956]: 1.4.13 lockups   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011160]: CDR RECORD CAN'T BE ADDED (REGRESSION)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011157]: Asterisk does not send a provisional response at every minute   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011157]: Asterisk does not send a provisional response at every minute   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010946]: chan_sip can generate several outstanding requests, but ignores responses   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009924]: Responses to Manager Commands Should Be Called 'Responses' and not 'Events'   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010354]: Add Basic Support For RFC 4662 (Subscribe to lists)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008587]: [patch] Caller Id and Message Waiting Indicator problems   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010993]: asterisk 1.2.24 doesn't log TRANSFER in dinamic agent login   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011071]: No audio on calls into queues with non-persistent agents   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010809]: Asterisk segfaults after an attended transfer to a queue using "Eyebeam" softphone.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011160]: CDR RECORD CAN'T BE ADDED (REGRESSION)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011160]: CDR RECORD CAN'T BE ADDED (REGRESSION)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011157]: Asterisk does not send a provisional response at every minute   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011157]: Asterisk does not send a provisional response at every minute   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011159]: pbx_builtin_setvar_helper crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011159]: pbx_builtin_setvar_helper crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010993]: Doesn't log TRANSFER in dynamic agent login   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011158]: [patch] chan_unistm.c Memory leak & Deadlock & Crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011157]: Asterisk does not send a provisional response at every minute   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009815]: [patch] major fix for SIP video codec negociations   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011157]: Asterisk does not send a provisional response at every minute   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011158]: [patch] chan_unistm.c Memory leak & Deadlock & Crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011157]: Asterisk does not send a provisional response at every minute   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010956]: 1.4.13 lockups   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011158]: [patch] chan_unistm.c Memory leak & Deadlock & Crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010926]: double call to ast_frame_free   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010915]: Problems when doing an attended and unattended transfer with Thomson Phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010956]: 1.4.13 lockups   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010185]: [patch]: Added automixmonitor feature / ability to use mixmonitor to record queue conversations on demand.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010645]: Not a time watchdog to BYE request   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009924]: Responses to Manager Commands Should Be Called 'Responses' and not 'Events'   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010809]: Asterisk segfaults after an attended transfer to a queue using "Eyebeam" softphone.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010543]: Some characters (such as #) are not escaped in SIP headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009660]: [patch]Asterisk can't establish dialtone after brief hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010543]: Some characters (such as #) are not escaped in SIP headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011143]: Solaris build issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011143]: Solaris build issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011101]: Greater than 256 New messages crashes Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011034]: Segfault in app_voicemail, inside c-client   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010818]: Hinting not working reliably   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011034]: Segfault in app_voicemail, inside c-client   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011163]: [patch] free config structure while exiting on error.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011164]: incoming audio lag   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009660]: [patch]Asterisk can't establish dialtone after brief hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011071]: No audio on calls into queues with non-persistent agents   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011085]: Hint stuck on hold after attended transfer with notifyhold=yes in sip.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011085]: Hint stuck on hold after attended transfer with notifyhold=yes in sip.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010474]: hint is hanging when remote party ends call on hold (re: 0010399   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009567]: Notify sent to a non-existent call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010946]: chan_sip can generate several outstanding requests, but ignores responses   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010936]: Crash in ast_queue_frame   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010595]: crash while send call via sip to another asterisk server into meetme   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010368]: app_dial segfaults asterisk while trying to bridge channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011084]: asterisk segfault   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010840]: segmentation faults on installation with 3000 calls/day.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010040]: random crashes in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011070]: cdr_sqlite3_custom backend error when logging to db   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009567]: Notify sent to a non-existent call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010543]: Some characters (such as #) are not escaped in SIP headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010543]: Some characters (such as #) are not escaped in SIP headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011154]: Asterisk does not retry a call on receving a 3XX response   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010828]: 302 Handling   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011154]: Asterisk does not retry a call on receving a 3XX response   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011070]: cdr_sqlite3_custom backend error when logging to db   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009066]: Cannot make compatible if video codecs do not match and audio codecs require transcoding   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009815]: [patch] major fix for SIP video codec negociations   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010532]: Some information from INVITE of Peer A not passed to INVITE of Peer B   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010532]: Some information from INVITE of Peer A not passed to INVITE of Peer B   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010532]: Some information from INVITE of Peer A not passed to INVITE of Peer B   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010991]: [PATCH] Add some functional to SayPosition in App_Queue   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010727]: "Presence" subscription causes Internal Server Error on Polycom 600 (re-open bug #5164)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011035]: Record on sip trunk does not maintian voice codec data rate   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010727]: "Presence" subscription causes Internal Server Error on Polycom 600 (re-open bug #5164)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010331]: [patch] PCMA/16000 and PCMU/16000 support (hd telephony)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0004821]: [branch] IPv6 support in chan_[sip, iax2]   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011159]: pbx_builtin_setvar_helper crash   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010593]: Zaptel crashes kernel - zt_init_tone_state   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010815]: [patch] SendFAX/ReceiveFAX   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009660]: [patch]Asterisk can't establish dialtone after brief hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009660]: [patch]Asterisk can't establish dialtone after brief hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010743]: Separate RTP pool for remote vs. local connections   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011106]: Sent time on voicemail notification emails incorrect after upgrade to 1.4.13   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010989]: Add fractional timeouts and default timeout variable values to Read   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010990]: Detection of scrambled memory (noisy) circuits of TDM400 POTS after nearby lightening strike   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010314]: [patch] Merged in support for high resolution timers in kernel >=2.6.22   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0007403]: [patch] allow SIP Spiral to work instead of causing a '482 Loop Detected' condition   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010532]: Some information from INVITE of Peer A not passed to INVITE of Peer B   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011145]: [patch] Improved support of QoS in channels using RTP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010532]: Some information from INVITE of Peer A not passed to INVITE of Peer B   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010532]: Some information from INVITE of Peer A not passed to INVITE of Peer B   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010677]: [post 1.4] SIP change at r77616 (rizzo) causes all outbound calls to fail authentication with 403 Forbidden   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011110]: progressinband=yes send 180 and 183 together   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009209]: race condition in sip hangup with reinvited media   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009305]: [patch] REINVITE before 200ok causes a call to be ended   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010313]: Chan SIP and ACL Source Based Routing   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010355]: RTP Stream with wrong Timestamp after 200 ok when 183 session in progress   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011154]: Asterisk does not retry a call on receving a 3XX response   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011154]: Asterisk does not retry a call on receving a 3XX response   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011168]: Activate general jitterbuffer for Unistim   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011143]: Solaris build issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011154]: Asterisk does not retry a call on receving a 3XX response   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011169]: unable to perform attended transfer of incoming call from mISDN   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010677]: [post 1.4] SIP change at r77616 (rizzo) causes all outbound calls to fail authentication with 403 Forbidden   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011167]: [patch] Substitute the pipe with the comma on the applications documentation.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010881]: Asterisk 1.4 console repeatedly showing *CLI> on MacOSX Tiger 10.4.10   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011166]: [patch] Fixed not probable memory leak but possible on chan_zap.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010743]: Separate RTP pool for remote vs. local connections   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008786]: [patch] Properly handle tempates on config read/write   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008786]: [patch] Properly handle tempates on config read/write   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011006]: hints display 'unreachable' peers still as 'idle'   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010989]: Add fractional timeouts and default timeout variable values to Read   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010989]: Add fractional timeouts and default timeout variable values to Read   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010790]: Cannot add channel to group if using AMI Originate   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011100]: deadlock in iax   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011034]: Segfault in app_voicemail, inside c-client   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011143]: Solaris build issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009972]: [branch] res_jabber over OpenSSL   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009972]: [branch] res_jabber over OpenSSL   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011116]: [patch] *BSD mutex lock issue   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011143]: Solaris build issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011143]: Solaris build issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009972]: [branch] res_jabber over OpenSSL   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011082]: Asterisk 1.4.13 stock segfault on pthread_mutex_lock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011035]: Record on sip trunk does not maintian voice codec data rate   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010835]: Add context field to app_meetme.so application for realtime mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011096]: [patch] Modify the return values on load_module().   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011035]: Record on sip trunk does not maintian voice codec data rate   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011035]: Record on sip trunk does not maintian voice codec data rate   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011173]: MoH classes init failed   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011173]: MoH classes init failed   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011173]: MoH classes init failed   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011082]: Asterisk 1.4.13 stock segfault on pthread_mutex_lock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011096]: [patch] Modify the return values on load_module().   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011007]: Circular call distribution no longer works   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011172]: Allow direct RTP in "Dial" with "tT" options if phones are "canreinvite=yes" and "dtmfmode=info"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010993]: Doesn't log TRANSFER in dynamic agent login   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010593]: Zaptel crashes kernel - zt_init_tone_state   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010593]: Zaptel crashes kernel - zt_init_tone_state   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010363]: ExternalIVR changes not playing audio   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010363]: ExternalIVR changes not playing audio   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009327]: Manager Dropping Events Under Moderate Call Load   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009327]: Manager Dropping Events Under Moderate Call Load   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0007904]: Transfer capability is inherited by a channel after being transfered via atxfer   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011035]: Record on sip trunk does not maintian voice codec data rate   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011175]: WaitExten hangs up channel when first digit is entered   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011035]: Record on sip trunk does not maintian voice codec data rate   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008030]: [patch] addition to support timeout and warning into MeetMe   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010519]: add support for wideband speex (Openwengo variant)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010455]: Qualify intervals >1000ms create needless double OPTIONS transmissions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011175]: WaitExten hangs up channel when first digit is entered   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010926]: double call to ast_frame_free   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010519]: add support for wideband speex (Openwengo variant)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011176]: "reload" command causes crash in Mac OSX Leopard 10.5   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008580]: [patch] Limit on simultaneous calls for queue members   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011176]: "reload" command causes crash in Mac OSX Leopard 10.5   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011107]: [patch] to better debug main DSP busydetect function   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011107]: [patch] to better debug main DSP busydetect function   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011176]: "reload" command causes crash in Mac OSX Leopard 10.5   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011078]: Patch CLI app_meetme.c - Provides "meetme concise"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011133]: app_voicemail.c's imapfolder option is not documented   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010727]: "Presence" subscription causes Internal Server Error on Polycom 600 (re-open bug #5164)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010519]: add support for wideband speex (Openwengo variant)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010730]: [patch] see channel in agi debug   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011174]: Crash related to ast_module_user_remove   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005424]: [patch] SIP peer authentication on an external database (RADIUS - LDAP)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010451]: Sending caller ID to ZAP Extension fails if sendcalleridafter=0 or sendcalleridafter=1   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010451]: Sending caller ID to ZAP Extension fails if sendcalleridafter=0 or sendcalleridafter=1   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011133]: app_voicemail.c's imapfolder option is not documented   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011133]: app_voicemail.c's imapfolder option is not documented   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009310]: [patch] Only apply externip on SIP, not on media streams   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011133]: app_voicemail.c's imapfolder option is not documented   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011174]: Crash related to ast_module_user_remove   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010891]: [patch] Add support for setting log levels on remote console   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010803]: [patch] Allow ParkedCall to pickup the first parked call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010024]: Adding ZRTP security protocol support for asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011088]: IMAP client fails with: IMAP Error: Unable to create selectable TCP socket (1024 >= 1024)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011107]: [patch] to better debug main DSP busydetect function   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011128]: hint does now work with the calling SIP channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010775]: Asterisk suddenly slows down, and eats 100% cpu   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011184]: menuselect is broken in revision 220   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011116]: [patch] *BSD mutex lock issue   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011183]: Segfault on Action: Command / Command: core show channels concise   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011183]: Segfault on Action: Command / Command: core show channels concise   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011180]: call limits not work as expected (limitonpeer, busy-level)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010896]: 2 features, ring expiry and periodic announce firstplay   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011184]: menuselect is broken in revision 220   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011185]: call routing based on caller-id fails to match   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011186]: Crash in chan_sip   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008925]: [post-1.4] CLI command audit   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011185]: call routing based on caller-id fails to match   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008925]: [post-1.4] CLI command audit   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010700]: [patch] Asterisk case sensitive problem.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010775]: Asterisk suddenly slows down, and eats 100% cpu   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011187]: Time sent header is incorrect when sending voicemails via email   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009506]: problem with CDR record when using AUTOMON fetaure or res_monitor on outgoing calls (phone->*->telco)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011179]: "core show channels verbose" small simple bug   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011070]: cdr_sqlite3_custom backend error when logging to db   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010156]: 'udevinfo: command not found' on a devfs-ed Debian Sarge   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011174]: Crash related to ast_module_user_remove   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010531]: [branch] bug in time-zone with daylight saving time   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011188]: CPU load spikes every 10 seconds   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010578]: Asterisk crashes on reload of pbx_config.so   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011100]: deadlock in iax   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011174]: Crash related to ast_module_user_remove   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010681]: Handling of escaped characters (#, etc...)   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011188]: CPU load spikes every 10 seconds   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011151]: IMAP: Mailbox does not exist   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011152]: Over quota errors ignored   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010957]: chan_vpb sample configuration file messy, not well documented, and missing a few options   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011179]: "core show channels verbose" small simple bug   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011179]: "core show channels verbose" small simple bug   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011174]: Crash related to ast_module_user_remove   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011174]: Crash related to ast_module_user_remove   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010742]: Random replacement of channel name with other text in queue log entries   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011185]: call routing based on caller-id fails to match   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010742]: Random replacement of channel name with other text in queue log entries   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011174]: Crash related to ast_module_user_remove   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010946]: chan_sip can generate several outstanding requests, but ignores responses   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010915]: Problems when doing an attended and unattended transfer with Thomson Phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010164]: Incorrect notify handling when hints contain multiple devices   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011058]: Asterisk doesn't recognize a "408 Request Timeout"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011055]: [patch] Register to the SIP server with domain name   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010836]: Patch for an inteligent parkedcalls.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010756]: NOTIFY contains invalid To header   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010727]: "Presence" subscription causes Internal Server Error on Polycom 600 (re-open bug #5164)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011170]: Add option to ResetCDR allowing users to re-enable CDR (only)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011154]: Asterisk does not retry a call on receving a 3XX response   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010956]: 1.4.13 lockups   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011179]: "core show channels verbose" small simple bug   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010956]: 1.4.13 lockups   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009660]: [patch]Asterisk can't establish dialtone after brief hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005768]: [branch][post 1.4] LDAP Realtime driver   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009815]: [patch] major fix for SIP video codec negociations   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009815]: [patch] major fix for SIP video codec negociations   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011154]: Asterisk does not retry a call on receving a 3XX response   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009815]: [patch] major fix for SIP video codec negociations   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011189]: chan_zap causing reset on E1 and eventually crashed asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010742]: Random replacement of channel name with other text in queue log entries   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009506]: problem with CDR record when using AUTOMON fetaure or res_monitor on outgoing calls (phone->*->telco)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011189]: chan_zap causing reset on E1 and eventually crashed asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011190]: DTMF minimal duration and Recommendation Q.24   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011190]: DTMF minimal duration and Recommendation Q.24   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011190]: DTMF minimal duration and Recommendation Q.24   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010196]: my asterisk comes down in flames randomly, it appears to be related to chanspy   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011190]: DTMF minimal duration and Recommendation Q.24   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011190]: DTMF minimal duration and Recommendation Q.24   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011190]: DTMF minimal duration and Recommendation Q.24   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011057]: Incorrect handling of international settings   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010956]: 1.4.13 lockups   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010956]: 1.4.13 lockups   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010578]: Asterisk crashes on reload of pbx_config.so   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011139]: Request for ROSE IE causes connection to drop.   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011139]: Request for ROSE IE causes connection to drop.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011190]: DTMF minimal duration and Recommendation Q.24   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010742]: Random replacement of channel name with other text in queue log entries   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011192]: MIX monitor file name and Agent Name that pickup call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0011191]: unable to set extension used for marked users in a meetme room conference   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011194]: ztmonitor issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011058]: Asterisk doesn't recognize a "408 Request Timeout"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011195]: [patch] Avoid asterisk WARNINGS while using the configs files in trunk configs directory   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011197]: ff and rew buttons for control stream file do not work.   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011182]: Http Admin timeout   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011182]: Http Admin timeout   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011198]: chan_h323.c needs to be fixed to match trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011198]: chan_h323.c needs to be fixed to match trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011199]: dependencies are not properly set   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011182]: Http Admin timeout   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010934]: Option r with early files   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011201]: Problem when using NAT and Subscriptions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009650]: No ringing heard on unattanded transfer   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011199]: dependencies are not properly set   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011202]: [patch] Add check_hangup() method to pbx_lua   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011203]: minor typo bug in ast_say_date_with_format_fr()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011203]: minor typo bug in ast_say_date_with_format_fr()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011204]: when IMAP storage is enabled, a duplicate "regular" email is sent when no email account is specified   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011204]: when IMAP storage is enabled, a duplicate "regular" email is sent when no email account is specified   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011200]: Problem when using NAT and Subscriptions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011205]: Issues while cross compiling asterisk-addons   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011206]: Crush if DO_CRUSH   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011194]: ztmonitor issues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009650]: No ringing heard on unattanded transfer   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011207]: DEBUG_THREADLOCALS: lock in main/threadstorage.c must be untracked?   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009650]: No ringing heard on unattanded transfer   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011207]: DEBUG_THREADLOCALS: lock in main/threadstorage.c must be untracked?   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009520]: realtime prune (and others) may segfault asterisk (timing issue)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011153]: ztdummy causing audio failure on Playback or Background   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011206]: Crush if DO_CRUSH   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011208]: [patch] Some deadlocks while loading config   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011153]: ztdummy causing audio failure on Playback or Background   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011209]: [patch] Change free() to ast_free()   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011153]: ztdummy causing audio failure on Playback or Background   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011209]: [patch] Change free() to ast_free()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011210]: SIP reload for large config halts SIP Processing   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010578]: Asterisk crashes on reload of pbx_config.so   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0007063]: [branch][post 1.4] Allow caller to dial 1-9 while leaving voicemail   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010845]: [patch] preventing parallel logins from the same line or by the same agent   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0004825]: [patch][post 1.4] New codec negotiation algorithm   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008925]: [post-1.4] CLI command audit   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008925]: [post-1.4] CLI command audit   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008925]: [post-1.4] CLI command audit   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008925]: [post-1.4] CLI command audit   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011141]: Cannot build modules using the latest Zaptel 1.4 with Linux kernel 2.6.24-rc1+   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010278]: tor2ee fails to build   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010278]: tor2ee fails to build   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011213]: [patch] Trivial: Use ast_free() insted of free()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011189]: chan_zap causing reset on E1 and eventually crashed asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011189]: chan_zap causing reset on E1 and eventually crashed asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011189]: chan_zap causing reset on E1 and eventually crashed asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010677]: [post 1.4] SIP change at r77616 (rizzo) causes all outbound calls to fail authentication with 403 Forbidden   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010278]: tor2ee fails to build   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011214]: Restart of MOH doesn't work   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011215]: extensions.conf looks for a [default] context, even if not defined in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011215]: extensions.conf looks for a [default] context, even if not defined in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011215]: extensions.conf looks for a [default] context, even if not defined in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011189]: chan_zap causing reset on E1 and eventually crashed asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011209]: [patch] Change free() to ast_free()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011215]: extensions.conf looks for a [default] context, even if not defined in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011215]: extensions.conf looks for a [default] context, even if not defined in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011215]: extensions.conf looks for a [default] context, even if not defined in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009325]: After some time Sip show peers cut 1 symbol (first)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011215]: extensions.conf looks for a [default] context, even if not defined in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011215]: extensions.conf looks for a [default] context, even if not defined in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011128]: hint does now work with the calling SIP channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011058]: Asterisk doesn't recognize a "408 Request Timeout"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011128]: hint does now work with the calling SIP channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011189]: chan_zap causing reset on E1 and eventually crashed asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011218]: trunk compilation error on FreeBSD (chan_unistim.c, hashtest)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011189]: chan_zap causing reset on E1 and eventually crashed asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011218]: trunk compilation error on FreeBSD (chan_unistim.c, hashtest)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011216]: The BLINDTRANSFER variable is not populated when call transfered via SIP 302   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011128]: hint does now work with the calling SIP channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011200]: Problem when using NAT and Subscriptions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011212]: [patch] expose zap DND mode to the dialplan   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011193]: IAX truncking works on one side only   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011128]: hint does now work with the calling SIP channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010826]: ChannelRedirect non-working   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010845]: [patch] preventing parallel logins from the same line or by the same agent   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011215]: extensions.conf looks for a [default] context, even if not defined in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010763]: 'make asterisk.pdf' produces an unfound .sty error.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009305]: [patch] REINVITE before 200ok causes a call to be ended   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010196]: my asterisk comes down in flames randomly, it appears to be related to chanspy   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011210]: SIP reload for large config halts SIP Processing   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009448]: "request sent" hang on sip show regitry...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011210]: SIP reload for large config halts SIP Processing   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010947]: Content-Type: multipart/mixed not recognised correctly (SIP-T on not working)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009066]: Cannot make compatible if video codecs do not match and audio codecs require transcoding   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011210]: SIP reload for large config halts SIP Processing   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010220]: HTTP manager delivery invalid XML   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010220]: HTTP manager delivery invalid XML   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010486]: [patch] Handle libcurl errors   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010781]: [patch] Doesn't lock config files when writing   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0007495]: [patch][post-1.4] AOC-D ("Advice of Charge - During call") passthrough   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011221]: [patch] Error while linking with DEBUG_THREADLOCALS enabled   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011223]: [patch] Create doxygen for hashtabs   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011225]: URI direct dialing to target domain : call rejected if source extension exist in the destination dialplan.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011222]: [patch] Doxygen fixes for various files   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010826]: ChannelRedirect non-working   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011227]: crash at __ast_pthread_mutex_lock &pkt->owner->lock can not access   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008925]: [post-1.4] CLI command audit   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011024]: Calls Looping back into call groups cause confusion or overload.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011229]: find lock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011229]: find lock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010961]: [patch] Add HTTP Basic Authentication Scheme (rfc2617) for manager web interface.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011149]: Asterisk 1.4.13 Produces mutex errors and too many files open error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011149]: Asterisk 1.4.13 Produces mutex errors and too many files open error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010196]: my asterisk comes down in flames randomly, it appears to be related to chanspy   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011139]: Request for ROSE IE causes connection to drop.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010881]: Asterisk 1.4 console repeatedly showing *CLI> on MacOSX Tiger 10.4.10   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011232]: Asterisk core not multitreading sip-to-sip calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011210]: SIP reload for large config halts SIP Processing   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010196]: my asterisk comes down in flames randomly, it appears to be related to chanspy   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010185]: [patch]: Added automixmonitor feature / ability to use mixmonitor to record queue conversations on demand.   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011234]: AsteriskNOW File Editor Strips required characters from lines   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010455]: Qualify intervals >1000ms create needless double OPTIONS transmissions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010993]: Doesn't log TRANSFER in dynamic agent login   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011236]: [patch] WaitForNoise added to accomplish WaitForSilence   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011238]: Compilation of chan_iax fails on Fedora 8   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010690]: sip peer with missing close bracket causes all subsequent sip peers not to load   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011232]: Asterisk core not multitreading sip-to-sip calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011237]: Parking -- execution does continue at next priority when requested parking extension is in use.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008925]: [post-1.4] CLI command audit   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010891]: [patch] Add support for setting log levels on remote console   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009896]: [patch] Authentication support for SIP NOTIFY requests   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011100]: deadlock in iax   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011226]: [patch] ast_cdr_free: CDR on channel 'SIP/02571-09174500' not posted   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011224]: [patch] add 'concise' option to sip show channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010891]: [patch] Add support for setting log levels on remote console   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010690]: sip peer with missing close bracket causes all subsequent sip peers not to load   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011236]: [patch] WaitForNoise added to accomplish WaitForSilence   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011234]: AsteriskNOW File Editor Strips required characters from lines   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011234]: AsteriskNOW File Editor Strips required characters from lines   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011234]: AsteriskNOW File Editor Strips required characters from lines   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011234]: AsteriskNOW File Editor Strips required characters from lines   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011234]: AsteriskNOW File Editor Strips required characters from lines   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010690]: sip peer with missing close bracket causes all subsequent sip peers not to load   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011152]: Over quota errors ignored   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011234]: AsteriskNOW File Editor Strips required characters from lines   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011227]: crash at __ast_pthread_mutex_lock &pkt->owner->lock can not access   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011227]: crash at __ast_pthread_mutex_lock &pkt->owner->lock can not access   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011217]: DLCX verb not recgonized   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011217]: DLCX verb not recgonized   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009260]: [Patch] SMDI features   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009055]: [patch] extend SMDI support to chan_sip   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011228]: While working on T38 terminal support for asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011240]: SRV lookups broken in SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009260]: [Patch] SMDI features   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011240]: SRV lookups broken in SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009896]: [patch] Authentication support for SIP NOTIFY requests   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011217]: DLCX verb not recgonized   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011049]: implementation of application/dtmf for SIP INFO   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011231]: [patch] Many retransmits when chan_sip generates multiple outstanding requests   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011232]: Asterisk core not multitreading sip-to-sip calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011049]: implementation of application/dtmf for SIP INFO   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011049]: implementation of application/dtmf for SIP INFO   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011049]: implementation of application/dtmf for SIP INFO   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011217]: DLCX verb not recgonized   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011232]: Asterisk core not multitreading sip-to-sip calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011232]: Asterisk core not multitreading sip-to-sip calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011230]: Asterisk MUST NOT update Route-Set during in-dialog messages   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011242]: Error while launching /etc/init.d/asterisk start in non-bash shell   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009449]: "Dial(mISDN/g:te/${EXTEN})" don't set up L2Link but "Dial(mISDN/1/${EXTEN}" works   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009260]: [Patch] SMDI features   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010196]: my asterisk comes down in flames randomly, it appears to be related to chanspy   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011244]: ast_variable_new() typo   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011242]: Error while launching /etc/init.d/asterisk start in non-bash shell   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011242]: Error while launching /etc/init.d/asterisk start in non-bash shell   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011224]: [patch] add 'concise' option to sip show channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011232]: Asterisk core not multitreading sip-to-sip calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011205]: Issues while cross compiling asterisk-addons   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011245]: Asterisk unable to handle Multple Authorization Headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011205]: Issues while cross compiling asterisk-addons   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011245]: Asterisk unable to handle Multple Authorization Headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011248]: [patch] Add CallerIDNum: and CallerIDName: to Event: Hangup output   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011205]: Issues while cross compiling asterisk-addons   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011205]: Issues while cross compiling asterisk-addons   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011205]: Issues while cross compiling asterisk-addons   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011253]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011245]: Asterisk unable to handle Multple Authorization Headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011245]: Asterisk unable to handle Multple Authorization Headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011252]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011252]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011253]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011253]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011253]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011253]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011100]: deadlock in iax   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011253]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011253]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011253]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011232]: Asterisk core not multitreading sip-to-sip calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011232]: Asterisk core not multitreading sip-to-sip calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011232]: Asterisk core not multitreading sip-to-sip calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011049]: implementation of application/dtmf for SIP INFO   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011255]: crash ast_queue_hangup (chan=0x0)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011245]: Asterisk unable to handle Multple Authorization Headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011224]: [patch] add 'concise' option to sip show channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011230]: Asterisk MUST NOT update Route-Set during in-dialog messages   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011239]: [patch] cleanup in t38 structure initialization   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011239]: [patch] cleanup in t38 structure initialization   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011228]: While working on T38 terminal support for asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011228]: While working on T38 terminal support for asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011228]: While working on T38 terminal support for asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011210]: SIP reload for large config halts SIP Processing   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011210]: SIP reload for large config halts SIP Processing   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009896]: [patch] Authentication support for SIP NOTIFY requests   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011228]: While working on T38 terminal support for asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010947]: Content-Type: multipart/mixed not recognised correctly (SIP-T on not working)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010947]: Content-Type: multipart/mixed not recognised correctly (SIP-T on not working)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011228]: While working on T38 terminal support for asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009305]: [patch] REINVITE before 200ok causes a call to be ended   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011228]: While working on T38 terminal support for asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011211]: [patch] Allow dialplan to set prefix for SIP Call ID   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011058]: Asterisk doesn't recognize a "408 Request Timeout"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011058]: Asterisk doesn't recognize a "408 Request Timeout"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011058]: Asterisk doesn't recognize a "408 Request Timeout"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010069]: SIP compile warnings, cashes crash on invite received   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010058]: Asterisk rfc2833 DTMF fails with SIP service providers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009431]: Modify connection: Response 491 not handled according to RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010481]: SIP with canreinvite=yes through multiple Asterisk instances fails   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009431]: Modify connection: Response 491 not handled according to RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010677]: [post 1.4] SIP change at r77616 (rizzo) causes all outbound calls to fail authentication with 403 Forbidden   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008677]: Asterisk does not reinvite peer for G.711 after T.38 negotiated failed with a "488" Event   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011063]: Peers become unreachable when 'sip set debug' is enabled and reachable again when 'sip set debug off'   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011025]: Unavailable peer not show in BLF of Thomson and Grandstream phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011228]: While working on T38 terminal support for asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010058]: Asterisk rfc2833 DTMF fails with SIP service providers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011228]: While working on T38 terminal support for asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011253]: Blowup after one-two hours with Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011232]: Asterisk core not multitreading sip-to-sip calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011256]: [patch] Deadlock in find_session(unsigned long ident)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011224]: [patch] add 'concise' option to sip show channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011224]: [patch] add 'concise' option to sip show channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011228]: bugs in udptl.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010937]: app_mixmonitor crash in ast_channel_spy_read_frame   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009896]: [patch] Authentication support for SIP NOTIFY requests   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009896]: [patch] Authentication support for SIP NOTIFY requests   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011257]: sip stops workking because of non rtp ports available   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011256]: [patch] Deadlock in find_session(unsigned long ident)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011260]: chan_h323 with H323Plus   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011257]: sip stops workking because of non rtp ports available   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011256]: [patch] Deadlock in find_session(unsigned long ident)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011257]: sip stops workking because of non rtp ports available   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009896]: [patch] Authentication support for SIP NOTIFY requests   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011255]: crash ast_queue_hangup (chan=0x0)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011255]: crash ast_queue_hangup (chan=0x0)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010297]: Unload/load support for chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010659]: CDRs are not merged for not answered calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010297]: Unload/load support for chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011251]: Parking into already taken parking slot leaves stalls channels and partially blocks asterisk (fix to issue 11237 is bad)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011263]: [patch] HOLD notification for Polycom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011263]: [patch] HOLD notification for Polycom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011254]: Typo in UPGRADE.txt   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011096]: [patch] Modify the return values on load_module().   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011096]: [patch] Modify the return values on load_module().   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011088]: IMAP client fails with: IMAP Error: Unable to create selectable TCP socket (1024 >= 1024)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011265]: [patch] Reuse code in hashtab.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011251]: Parking into already taken parking slot leaves stalls channels and partially blocks asterisk (fix to issue 11237 is bad)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009046]: [patch] Add confirmation of forwarded-to user via name or extension when forwarding a voicemail to another mailbox   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011266]: When invalid IP address is specified chan_iax2 crashes.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011204]: when IMAP storage is enabled, a duplicate "regular" email is sent when no email account is specified   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011266]: When invalid IP address is specified chan_iax2 crashes.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011258]: [patch] missing locks while calling astman_send_error()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011266]: When invalid IP address is specified chan_iax2 crashes.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011262]: I can take it down in 2 minutes using a simulator.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011262]: I can take it down in 2 minutes using a simulator.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011258]: [patch] missing locks while calling astman_send_error()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011266]: When invalid IP address is specified chan_iax2 crashes.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011262]: I can take it down in 2 minutes using a simulator.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011262]: I can take it down in 2 minutes using a simulator.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011268]: [patch] Context tracing for channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011266]: When invalid IP address is specified chan_iax2 crashes.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011266]: When invalid IP address is specified chan_iax2 crashes.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009896]: [patch] Authentication support for SIP NOTIFY requests   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011088]: IMAP client fails with: IMAP Error: Unable to create selectable TCP socket (1024 >= 1024)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011266]: When invalid IP address is specified chan_iax2 crashes.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011266]: When invalid IP address is specified chan_iax2 crashes.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011096]: [patch] Modify the return values on load_module().   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011145]: [patch] Improved support of QoS in channels using RTP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011025]: Unavailable peer not show in BLF of Thomson and Grandstream phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011266]: When invalid IP address is specified chan_iax2 crashes.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010659]: CDRs are not merged for not answered calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011266]: When invalid IP address is specified chan_iax2 crashes.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011229]: find lock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011262]: I can take it down in 2 minutes using a simulator.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011229]: find lock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011229]: find lock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011169]: unable to perform attended transfer of incoming call from mISDN   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011269]: chan_misdn don't compile in trunk because lock renamed   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009449]: "Dial(mISDN/g:te/${EXTEN})" don't set up L2Link but "Dial(mISDN/1/${EXTEN}" works   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011142]: [PATCH] Better handling of temporary channel name. Fixes FOP.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010631]: One way sound on calls between mISDN and SIP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011142]: [PATCH] Better handling of temporary channel name. Fixes FOP.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010944]: mISDN don't make hangup and crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010944]: mISDN don't make hangup and crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010807]: Incorrectly sized IAX2 frames sent from mISDN with b410p   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011270]: [patch] ast_tvdiff_us   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011169]: unable to perform attended transfer of incoming call from mISDN   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011083]: [patch] Empty voicemail message file causes call into voicemail to disconnect   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011262]: I can take it down in 2 minutes using a simulator.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009046]: [patch] Add confirmation of forwarded-to user via name or extension when forwarding a voicemail to another mailbox   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011271]: [patch] QUEUE_MEMBER_COUNT doesn't load realtime queue members   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011271]: [patch] QUEUE_MEMBER_COUNT doesn't load realtime queue members   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011271]: [patch] QUEUE_MEMBER_COUNT doesn't load realtime queue members   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011272]: service providers does not display   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011083]: [patch] Empty voicemail message file causes call into voicemail to disconnect   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009046]: [patch] Add confirmation of forwarded-to user via name or extension when forwarding a voicemail to another mailbox   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009046]: [patch] Add confirmation of forwarded-to user via name or extension when forwarding a voicemail to another mailbox   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011273]: possible memory leak in asp_dsp_process   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010923]: crash in ast_var_name on SIP hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011274]: Zaptel: No Audio After First DTMF   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011272]: service providers does not display   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011272]: service providers does not display   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011272]: service providers does not display   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011270]: [patch] ast_tvdiff_us   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011275]: On certain deadlocks, running core show locks segfaults asterisk and yields no lock info   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009939]: Transfer implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011275]: On certain deadlocks, running core show locks segfaults asterisk and yields no lock info   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011275]: On certain deadlocks, running core show locks segfaults asterisk and yields no lock info   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010953]: SIP deadlocks unexpectedly at random intervals, trigger unknown   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011262]: I can take it down in 2 minutes using a simulator.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010185]: [patch]: Added automixmonitor feature / ability to use mixmonitor to record queue conversations on demand.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011262]: I can take it down in 2 minutes using a simulator.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009896]: [patch] Authentication support for SIP NOTIFY requests   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011278]: this patch makes transcoding smarter,   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010540]: timeout value should accept floating point numbers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010479]: Registration/Un-Registration Problem with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011279]: [patch] trunk: rwlock tracking support (tracking and untracking static rwlock)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011279]: [patch] trunk: rwlock tracking support (tracking and untracking static rwlock)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011277]: [patch] sysinfo dialplan function   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011277]: [patch] sysinfo dialplan function   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011270]: [patch] ast_tvdiff_us   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011270]: [patch] ast_tvdiff_us   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011165]: Deadlock in channel.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011270]: [patch] ast_tvdiff_us   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011139]: Request for ROSE IE causes connection to drop.   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0010929]: CallingName Facility IE not transmited to Meridian Option/11 PBX   noreply at bugs.digium.com
• [asterisk-bugs] [LibSS7 0011156]: added the generic address through to extension.conf like the charge number   noreply at bugs.digium.com
• [asterisk-bugs] [LibSS7 0011156]: added the generic address through to extension.conf like the charge number   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011145]: [patch] Improved support of QoS in channels using RTP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011268]: [patch] Context tracing for channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011281]: [patch] Check if config file changed before reloading the configuration   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009077]: Playback(<file>|noanswer) and in-band info are not working with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009077]: Playback(<file>|noanswer) and in-band info are not working with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011145]: [patch] Improved support of QoS in channels using RTP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011268]: [patch] Context tracing for channels   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011176]: "reload" command causes crash in Mac OSX Leopard 10.5   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010202]: idle console disconnects   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0007787]: [patch] DTMF Caller-ID support.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009077]: Playback(<file>|noanswer) and in-band info are not working with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011145]: [patch] Improved support of QoS in channels using RTP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011280]: [patch] add some constants to chan_zap   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011282]: [patch] New Application to take control of applications executed on a channel from the AMI or Console   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011280]: [patch] add some constants to chan_zap   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011274]: Zaptel: No Audio After First DTMF   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011262]: Asterisk crash on non-responsive gateway   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011263]: [patch] HOLD notification for Polycom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011263]: [patch] HOLD notification for Polycom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011283]: [patch] Blackfin optimizations in arith.h   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011283]: [patch] Blackfin optimizations in arith.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009015]: make install assumes udev rules file exists   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011283]: [patch] Blackfin optimizations in arith.h   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011276]: UDEV zaptel.rules & CentOS 5   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011283]: [patch] Blackfin optimizations in arith.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009015]: make install assumes udev rules file exists   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011276]: UDEV zaptel.rules & CentOS 5   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009015]: make install assumes udev rules file exists   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010393]: [patch] per-channel alarms in Zaptel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011285]: [patch] Asterisk segfaults while doing a 'module reload'.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011176]: "reload" command causes crash in Mac OSX Leopard 10.5   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011286]: [patch] usage of unitialized parameters in Monitor   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011285]: [patch] Asterisk segfaults while doing a 'module reload'.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011285]: [patch] Asterisk segfaults while doing a 'module reload'.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011285]: [patch] Asterisk segfaults while doing a 'module reload'.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011263]: [patch] HOLD notification for Polycom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011287]: Failed compile pbx_ael on solaris   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011287]: Failed compile pbx_ael on solaris   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011263]: [patch] HOLD notification for Polycom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011283]: [patch] Blackfin optimizations in arith.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011287]: Failed compile pbx_ael on solaris   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011145]: [patch] Improved support of QoS in channels using RTP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011263]: [patch] HOLD notification for Polycom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011283]: [patch] Blackfin optimizations in arith.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011284]: Agent transfering cal via SIP transfer gets logged out   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011141]: Cannot build modules using the latest Zaptel 1.4 with Linux kernel 2.6.24-rc1+   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009225]: sip doesnt bind to all   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011287]: Failed compile pbx_ael on solaris   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009225]: sip doesnt bind to all   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011287]: Failed compile pbx_ael on solaris   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009225]: sip doesnt bind to all   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011180]: call limits not work as expected (limitonpeer, busy-level)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011180]: call limits not work as expected (limitonpeer, busy-level)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011180]: call limits not work as expected (limitonpeer, busy-level)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011283]: [patch] Blackfin optimizations in arith.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011293]: commit 8938 should be reverted   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011293]: commit 8938 should be reverted   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011291]: Latest MySQL CDR crash on start   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011289]: Crash with IAX2 incoming call: Bad magic number   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011284]: Agent transfering cal via SIP transfer gets logged out   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011288]: [patch] Convert warnings to debug when using say.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011285]: [patch] Asterisk segfaults while doing a 'module reload'.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011286]: [patch] usage of unitialized parameters in Monitor   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011279]: [patch] trunk: rwlock tracking support (tracking and untracking static rwlock)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011236]: [patch] WaitForNoise added to accomplish WaitForSilence   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011246]: Answering local channel does not remove it from call path   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011265]: [patch] Reuse code in hashtab.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011169]: unable to perform attended transfer of incoming call from mISDN   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011286]: [patch] usage of unitialized parameters in Monitor   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010196]: my asterisk comes down in flames randomly, it appears to be related to chanspy   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011169]: unable to perform attended transfer of incoming call from mISDN   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011257]: sip stops workking because of non rtp ports available   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0011191]: unable to set extension used for marked users in a meetme room conference   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0011191]: unable to set extension used for marked users in a meetme room conference   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0011249]: Voicemail users id's aren't recognised   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0011249]: Voicemail users id's aren't recognised   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011276]: UDEV zaptel.rules & CentOS 5   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010934]: Option r with early files   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011246]: Answering local channel does not remove it from call path   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011284]: Agent transfering cal via SIP transfer gets logged out   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011211]: [patch] Allow dialplan to set prefix for SIP Call ID   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011211]: [patch] Allow dialplan to set prefix for SIP Call ID   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011154]: Asterisk does not retry a call on receving a 3XX response   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011154]: Asterisk does not retry a call on receving a 3XX response   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011292]: Unlock not locked.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010976]: NAT settings ignored on calls recieved to [general]   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011295]: [patch] Trivial: When logging, use features.conf insted of parking.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010937]: app_mixmonitor crash in ast_channel_spy_read_frame   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011257]: sip stops workking because of non rtp ports available   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009225]: sip doesnt bind to all   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009896]: [patch] Authentication support for SIP NOTIFY requests   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011262]: Asterisk crash on non-responsive gateway   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011277]: [patch] sysinfo dialplan function   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011297]: network.h needs compiler.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011297]: network.h needs compiler.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0011122]: astman.js variables are static, not upgrade with parameters passed into ./configure   noreply at bugs.digium.com
• [asterisk-bugs] Possible SIP bug, needs confirmation/testing   Robert Dyck
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009649]: BYE calls too fast when connected to a mediatrix box makes asterisk acts weird   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009649]: BYE calls too fast when connected to a mediatrix box makes asterisk acts weird   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009503]: [patch] separate sections in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011284]: Agent transfering cal via SIP transfer gets logged out   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011301]: utils.h needs stdarg.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011301]: utils.h needs stdarg.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011299]: frame.h needs stdint.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011300]: utils.h needs string.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011299]: frame.h needs stdint.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011297]: network.h needs compiler.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011298]: abstract_jb.h needs stdio.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011300]: utils.h needs string.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011300]: utils.h needs string.h   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009503]: [patch] separate sections in zapata.conf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011296]: astobj2.h needs size_t   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011180]: call limits not work as expected (limitonpeer, busy-level)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009649]: BYE calls too fast when connected to a mediatrix box makes asterisk acts weird   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011180]: call limits not work as expected (limitonpeer, busy-level)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011180]: call limits not work as expected (limitonpeer, busy-level)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009649]: BYE calls too fast when connected to a mediatrix box makes asterisk acts weird   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011066]: Asterisk SIP Connections to systems that support t38 fax detection may fail   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009431]: Modify connection: Response 491 not handled according to RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009431]: Modify connection: Response 491 not handled according to RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009431]: Modify connection: Response 491 not handled according to RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009431]: Modify connection: Response 491 not handled according to RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010677]: [post 1.4] SIP change at r77616 (rizzo) causes all outbound calls to fail authentication with 403 Forbidden   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011157]: Asterisk does not send a provisional response at every minute   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010481]: SIP with canreinvite=yes through multiple Asterisk instances fails   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010937]: app_mixmonitor crash in ast_channel_spy_read_frame   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011308]: Picking the wrong extension   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011308]: Picking the wrong extension   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk-GUI 0011122]: astman.js variables are static, not upgrade with parameters passed into ./configure   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011004]: Implement READSTATUS   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010691]: WARNING[29872]: translate.c:163 framein: no samples for g729tolin   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010691]: WARNING[29872]: translate.c:163 framein: no samples for g729tolin   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011070]: cdr_sqlite3_custom backend error when logging to db   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011217]: DLCX verb not recgonized   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008993]: missing audio for sayduration parameter   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011313]: [patch] Solaris build with editline   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011315]: [patch] Solaris fixes to compile   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009650]: No ringing heard on unattanded transfer   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011308]: Picking the wrong extension   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011308]: Picking the wrong extension   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011311]: h323 does not compile in latest Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011316]: [patch] Add a check to the 'core show translation' function and fixed a text usage error.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011314]: [patch] Prevent an asterisk crash if we do a 'module unload app_dial.so'   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011317]: Fix for buil unistim in dev_mode (array subscript is above array bounds)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011291]: Latest MySQL CDR crash on start   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011083]: [patch] Empty voicemail message file causes call into voicemail to disconnect   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011083]: [patch] Empty voicemail message file causes call into voicemail to disconnect   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011292]: Unlock not locked.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010691]: WARNING[29872]: translate.c:163 framein: no samples for g729tolin   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011312]: [patch] CLI command 'sip show history' parameter is a <call-id> not a <channel>   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011319]: Unable to forward voice frame   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011319]: Unable to forward voice frame   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011312]: [patch] CLI command 'sip show history' parameter is a <call-id> not a <channel>   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009431]: Modify connection: Response 491 not handled according to RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011162]: Added variable "agi_asteriskthreadid" to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011169]: unable to perform attended transfer of incoming call from mISDN   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011107]: [patch] to better debug main DSP busydetect function   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011091]: autopark strange behaviour if canreinvite=update, nonat   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011091]: autopark strange behaviour if canreinvite=update, nonat   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011091]: autopark strange behaviour if canreinvite=update, nonat   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011091]: autopark strange behaviour if canreinvite=update, nonat   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011169]: unable to perform attended transfer of incoming call from mISDN   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010908]: Fix Round Robin Routing Allow 1 Up Port   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011083]: [patch] Empty voicemail message file causes call into voicemail to disconnect   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011172]: Allow direct RTP in "Dial" with "tT" options if phones are "canreinvite=yes" and "dtmfmode=info"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011172]: Allow direct RTP in "Dial" with "tT" options if phones are "canreinvite=yes" and "dtmfmode=info"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011169]: unable to perform attended transfer of incoming call from mISDN   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011321]: CDR(billsec) return 0 in some scenarios   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010937]: app_mixmonitor crash in ast_channel_spy_read_frame   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011320]: [patch] ooh323 does not compile with latest trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011312]: [patch] CLI command 'sip show history' parameter is a <call-id> not a <channel>   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011311]: h323 does not compile in latest Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011316]: [patch] Add a check to the 'core show translation' function and fixed a text usage error.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011324]: Errors in queues-with-callback-members.txt   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylimit' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylimit' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011306]: [patch] Include action: getcategory for the GUI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011306]: [patch] Include action: getcategory for the GUI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010099]: [branch] Event Based CDR system -- CEL (channel Event Logging)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylimit' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylimit' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011306]: [patch] Include action: getcategory for the GUI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylimit' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylimit' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylimit' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylimit' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylevel' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylevel' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylevel' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylevel' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylevel' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011307]: use 'busylevel' consistently   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011328]: chan_local does not propogate cause codes   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011329]: AEL macro argument variables aren't properly quoted when set   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011330]: No clean way to get a list of active channels in the dialplan   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011330]: No clean way to get a list of active channels in the dialplan   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011293]: commit 8938 should be reverted   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011141]: Cannot build modules using the latest Zaptel 1.4 with Linux kernel 2.6.24-rc1+   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011333]: [patch] Trivial: Missing ast_module_user_remove() in res_features.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011306]: [patch] Include action: getcategory for the GUI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011306]: [patch] Include action: getcategory for the GUI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011321]: CDR(billsec) return 0 in some scenarios   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011306]: [patch] Include action: getcategory for the GUI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011326]: asterisk modifying the in-dialogue route set which is a violation of RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011264]: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011335]: Hook-Flash behaviour results in echo-cancellation disabled on channel during call.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0007274]: [patch] record spooling support in cdr_addon_mysql.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011330]: [patch] CHANNELS dialplan function, get channel list in the dialplan   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011317]: Fix for buil unistim in dev_mode (array subscript is above array bounds)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011337]: AGI manager events   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011330]: [patch] CHANNELS dialplan function, get channel list in the dialplan   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011337]: AGI manager events   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011004]: Implement READSTATUS   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011338]: "languageprefix=yes" doesn't work   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011330]: [patch] CHANNELS dialplan function, get channel list in the dialplan   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011338]: "languageprefix=yes" doesn't work   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011331]: [patch] remove obsolete code from dsp.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011339]: No audio midcall   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011338]: "languageprefix=yes" doesn't work   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011325]: Issue in recordthread   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011338]: "languageprefix=yes" doesn't work   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011338]: "languageprefix=yes" doesn't work   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011338]: "languageprefix=yes" doesn't work   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011095]: asterisk releases progressively breaking queues   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010220]: HTTP manager delivery invalid XML   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011338]: "languageprefix=yes" doesn't work   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011338]: "languageprefix=yes" doesn't work   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011093]: CDR Created incorrectly on Transfer of outgoing call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011176]: "reload" command causes crash in Mac OSX Leopard 10.5   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011285]: [patch] Asterisk segfaults while doing a 'module reload'.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010815]: [patch] SendFAX/ReceiveFAX   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011290]: Asterisk crashed on reloading   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011344]: [patch] Remove privacy.conf and minor code cleanup.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011219]: [patch] Avoid including not needed header files or already included.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011291]: Latest MySQL CDR crash on start   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010815]: [patch] SendFAX/ReceiveFAX   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010815]: [patch] SendFAX/ReceiveFAX   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011332]: revision 89461 Problems with loader.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: [patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011294]: [patch] find_context_locked() must return with conlock held   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010815]: [patch] SendFAX/ReceiveFAX   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009506]: problem with CDR record when using AUTOMON fetaure or res_monitor on outgoing calls (phone->*->telco)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010815]: [patch] SendFAX/ReceiveFAX   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011349]: [patch] Deprecate SIPPEER() and move functionality into CHANNEL()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011349]: [patch] Deprecate SIPPEER() and move functionality into CHANNEL()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011349]: [patch] Deprecate SIPPEER() and move functionality into CHANNEL()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010815]: [patch] SendFAX/ReceiveFAX   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010220]: HTTP manager delivery invalid XML   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010220]: HTTP manager delivery invalid XML   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010220]: HTTP manager delivery invalid XML   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010220]: HTTP manager delivery invalid XML   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010220]: HTTP manager delivery invalid XML   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010220]: HTTP manager delivery invalid XML   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011353]: Problem with Like query on MS SQL   noreply at bugs.digium.com
• [asterisk-bugs] Strange problem with Asterisk-Zap-libSS7   Dmitri Sologoubenko
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011353]: Problem with Like query on MS SQL   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010659]: CDRs are not merged for not answered calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011275]: On certain deadlocks, running core show locks segfaults asterisk and yields no lock info   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011204]: when IMAP storage is enabled, a duplicate "regular" email is sent when no email account is specified   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011282]: [patch] New Application to take control of applications executed on a channel from the AMI or Console   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010220]: HTTP manager delivery invalid XML   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011083]: [patch] Empty voicemail message file causes call into voicemail to disconnect   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011359]: setting voicemail greeting when using IMAP backend causes zero byte messages to be left   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011275]: On certain deadlocks, running core show locks segfaults asterisk and yields no lock info   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011275]: On certain deadlocks, running core show locks segfaults asterisk and yields no lock info   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011279]: [patch] trunk: rwlock tracking support (tracking and untracking static rwlock)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011275]: On certain deadlocks, running core show locks segfaults asterisk and yields no lock info   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010953]: SIP deadlocks unexpectedly at random intervals, trigger unknown   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011083]: [patch] Empty voicemail message file causes call into voicemail to disconnect   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011357]: filestream is not closed before executing commands   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011362]: [patch] Error compiling chan_h323 with DEBUG_THREADS   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011360]: backport ast_debug to 1.4   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011083]: [patch] Empty voicemail message file causes call into voicemail to disconnect   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011083]: [patch] Empty voicemail message file causes call into voicemail to disconnect   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011361]: serving multiple Realms with one Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011357]: filestream is not closed before executing commands   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011362]: [patch] Error compiling chan_h323 with DEBUG_THREADS   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011362]: [patch] Error compiling chan_h323 with DEBUG_THREADS   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010450]: Turkey country support for ZAPTEL, hangups not recognized, callerid not working   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011364]: ESCAPE clause in first parameter not escaped properly   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011364]: ESCAPE clause in first parameter not escaped properly   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011363]: Picking the wrong extension with _21x.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011363]: Picking the wrong extension with _21x.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011363]: Picking the wrong extension with _21x.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011363]: Picking the wrong extension with _21x.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011365]: [patch] Add 'voicemail reload' CLI command   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011326]: asterisk modifying the in-dialogue route set which is a violation of RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011364]: ESCAPE clause in first parameter not escaped properly   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011365]: [patch] Add 'voicemail reload' CLI command   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010217]: [patch] proper user information layer 1 handling   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011368]: chan_mobile does not recognize dtmf together with Authenticate or DISA   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011365]: [patch] Add 'voicemail reload' CLI command   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011365]: [patch] Add 'voicemail reload' CLI command   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011365]: [patch] Add 'voicemail reload' CLI command   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011369]: Asterisk crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011365]: [patch] Add 'voicemail reload' CLI command   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011357]: filestream is not closed before executing commands   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011357]: filestream is not closed before executing commands   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005014]: [patch] implement call pickup on Snom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005014]: [patch] implement call pickup on Snom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011373]: crash at manager.c pointer error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010807]: Incorrectly sized IAX2 frames sent from mISDN with b410p   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010487]: IMAP integration with MS Exchange crashes Asterisk-1.4.10.1   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010487]: IMAP integration with MS Exchange crashes Asterisk-1.4.10.1   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011372]: Dial option L(limit in ms) is not working   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011361]: serving multiple Realms with one Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005424]: [patch] SIP peer authentication on an external database (RADIUS - LDAP)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011369]: Asterisk crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011372]: Dial option L(limit in ms) is not working   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011361]: serving multiple Realms with one Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010217]: [patch] proper user information layer 1 handling   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011318]: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011374]: Asterisk crash when executing SQLPREPARE statememt at cdr_odbc.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011370]: Add Extension without information produced NO LOGIN ANYMORE   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011375]: app_controlplayback used with option o (restart playback at position) craches Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011375]: app_controlplayback used with option o (restart playback at position) craches Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011368]: chan_mobile does not recognize dtmf together with Authenticate or DISA   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011367]: [patch] Avoid asterisk crash when unloading module app_meetme   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011366]: `sip show channels' does not display properly the codec in use for audio and video calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011348]: RTP session ID is negative half the time on x86_64   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011355]: [patch] Missing sched_context_destroy() in ast_channel_alloc() error condition.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011356]: [patch] Promote more reuse in hashtab.c   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011346]: VPM450: Failed to initialize   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011332]: revision 89461 Problems with loader.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011341]: MTX_PROFILE broken again ael_main.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011365]: [patch] Add 'voicemail reload' CLI command   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011339]: No audio midcall   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011378]: [patch] Update documentation that both 'num' and 'number' are valid for use in CALLERID()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011371]: People with a 2 letter surname cannot be looked up from the directory   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011332]: revision 89461 Problems with loader.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011349]: [patch] Deprecate SIPPEER() and move functionality into CHANNEL()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011243]: Distortion in Playback of .gsm files over non-GSM channel   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011306]: [patch] Include action: getcategory for the GUI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011357]: filestream is not closed before executing commands   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011365]: [patch] Add 'voicemail reload' CLI command   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011347]: [patch] dialplan remove extension make coredump if asterisk compiled with MALLOC_DEBUG options   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010500]: [patch] teach chan_iax2 to offer the calling channel's codec first, like chan_sip does it   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009077]: Playback(<file>|noanswer) and in-band info are not working with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011350]: Crash on calls AST_LIST_REMOVE_HEAD Macro   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011350]: Crash on calls AST_LIST_REMOVE_HEAD Macro   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011373]: crash at manager.c pointer error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011373]: crash at manager.c pointer error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011373]: crash at manager.c pointer error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011373]: crash at manager.c pointer error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011329]: AEL macro argument variables aren't properly quoted when set   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011333]: [patch] Trivial: Missing ast_module_user_remove() in res_features.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010500]: [patch] teach chan_iax2 to offer the calling channel's codec first, like chan_sip does it   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010500]: [patch] teach chan_iax2 to offer the calling channel's codec first, like chan_sip does it   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011380]: having same voicemail pin as voicemail password should force user to change their password, but it does not   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011381]: [patch] Update examples to use CALLERID(num) instead of CALLERID(number)   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011217]: DLCX verb not recgonized   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011369]: Asterisk crash   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010659]: CDRs are not merged for not answered calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011379]: 1.4.14 breaks cdr posting   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011334]: core show locks crush   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011275]: On certain deadlocks, running core show locks segfaults asterisk and yields no lock info   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011334]: core show locks crush   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011093]: CDR Created incorrectly on Transfer of outgoing call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009077]: Playback(<file>|noanswer) and in-band info are not working with chan_skinny   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009650]: No ringing heard on unattanded transfer   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011314]: [patch] Prevent an asterisk crash if we do a 'module unload app_dial.so'   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011314]: [patch] Prevent an asterisk crash if we do a 'module unload app_dial.so'   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011285]: [patch] Asterisk segfaults while doing a 'module reload'.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011196]: RealTime MusicOnHold   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011382]: Dial option G does not handle labels under some conditions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011373]: crash at manager.c pointer error   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011291]: Latest MySQL CDR crash on start   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011291]: Latest MySQL CDR crash on start   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010659]: CDRs are not merged for not answered calls   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011304]: problems with counting call limits   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011304]: problems with counting call limits   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005215]: [patch] Provisional responses to INVITE are ignored if a request has been sent in an early dialog   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009838]: Bye authorization working only one way.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009838]: Bye authorization working only one way.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011272]: service providers does not display   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: [patch] Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011376]: [patch] Codec negotiation results in asterisk sending unsupported codec   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011377]: added a return variable to app_controlplayback that reports the key used to stop playback.   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011346]: VPM450: Failed to initialize   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011384]: instalation problam   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011385]: Option eventwhencalled seems not working properly.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011361]: serving multiple Realms with one Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005014]: [patch] implement call pickup on Snom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011388]: voicemail directory is not created until user is left a message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005014]: [patch] implement call pickup on Snom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005014]: [patch] implement call pickup on Snom phones   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0004825]: [patch][post 1.4] New codec negotiation algorithm   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011382]: Dial option G does not handle labels under some conditions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011382]: Dial option G does not handle labels under some conditions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011326]: asterisk modifying the in-dialogue route set which is a violation of RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011382]: Dial option G does not handle labels under some conditions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011382]: Dial option G does not handle labels under some conditions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011391]: caller_chan_id/callee_chan_id does not contain correct info when call is xfered   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011391]: caller_chan_id/callee_chan_id does not contain correct info when call is xfered   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011385]: Option eventwhencalled seems not working properly.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011385]: Option eventwhencalled seems not working properly.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011329]: AEL macro argument variables aren't properly quoted when set   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011329]: AEL macro argument variables aren't properly quoted when set   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0006113]: [branch] Enhance parking to allow multiple parking lots   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011388]: voicemail directory is not created until user is left a message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Gastman 0011352]: Wrong argument in strncpy() causes problems with ztdynamic clients   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011226]: [patch] ast_cdr_free: CDR on channel 'SIP/02571-09174500' not posted   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011385]: Option eventwhencalled seems not working properly.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011310]: Impliment CallForwardAll   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011387]: Loading of res_config_pgsql will crash on dbhost/dbsock combination   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011387]: Loading of res_config_pgsql will crash on dbhost/dbsock combination   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011139]: Request for ROSE IE causes connection to drop.   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0011250]: Q.931 on PRI connection to analog PBX causes hangup when receiving a call   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011188]: CPU load spikes every 10 seconds   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011188]: CPU load spikes every 10 seconds   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011274]: Zaptel: No Audio After First DTMF   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011274]: Zaptel: No Audio After First DTMF   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010941]: zaptel digital (Data) calls require overlapdial=no   noreply at bugs.digium.com
• [asterisk-bugs] [LibPRI 0010376]: [patch] Set SONAME of libpri   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011291]: Latest MySQL CDR crash on start   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011394]: [patch] Add missing includes in SVN Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011394]: [patch] Add missing includes in SVN Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011394]: [patch] Add missing includes in SVN Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011394]: [patch] Add missing includes in SVN Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011387]: Loading of res_config_pgsql will crash on dbhost/dbsock combination   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011161]: New application app_pickupextn.c   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0010941]: zaptel digital (Data) calls require overlapdial=no   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011385]: Option eventwhencalled seems not working properly.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011326]: asterisk modifying the in-dialogue route set which is a violation of RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011230]: Asterisk MUST NOT update Route-Set during in-dialog messages   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011326]: asterisk modifying the in-dialogue route set which is a violation of RFC3261   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0008290]: [patch] zap hookstate is never set offhook if wctdm module is loaded with an active fxo line   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011397]: Dont play video on console dial   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011359]: setting voicemail greeting when using IMAP backend causes zero byte messages to be left   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011396]: Few unfortunate misprint in log messages   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011392]: Using Net-SNMP (RPM) to compile Asterisk with SNMP on CentOS 4 not possible.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011395]: [patch] hide CLI commands starting with '_'   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010037]: SIP Reinvite Packets incorrect Sequence causes no audio when more than 1 softswitch in callpath.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010530]: DTMFs passing only in part between two asterisk machines with an IAX2 connection   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010530]: DTMFs passing only in part between two asterisk machines with an IAX2 connection   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011011]: No ring tone is heard when calling a channel after the calling channel has been answered   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010052]: NOTIFY race condition when state changes happen very fast   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011398]: [patch] Send Asterisk version to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011398]: [patch] Send Asterisk version to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011393]: [patch] Trivial: Replace free() with ast_free() to match up with ast_calloc() / ast_malloc()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011361]: serving multiple Realms with one Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011400]: Resetting the SEQ number back to 0 without sending a new INVITE SSRC   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011400]: Resetting the SEQ number back to 0 without sending a new INVITE SSRC   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011400]: Resetting the SEQ number back to 0 without sending a new INVITE SSRC   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011400]: Resetting the SEQ number back to 0 without sending a new INVITE SSRC   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011400]: Resetting the SEQ number back to 0 without sending a new INVITE SSRC   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011400]: Resetting the SEQ number back to 0 without sending a new INVITE SSRC   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011400]: Resetting the SEQ number back to 0 without sending a new INVITE SSRC   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005215]: [patch] Provisional responses to INVITE are ignored if a request has been sent in an early dialog   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011181]: Manager API hangs on "Command: show channels"   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011402]: Interdigit timeout is half time of the defined time   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011401]: Variable identifier not compatible with C++   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011349]: [patch] Deprecate SIPPEER()/IAXPEER() and move functionality into CHANNEL()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011349]: [patch] Deprecate SIPPEER()/IAXPEER() and move functionality into CHANNEL()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011386]: Asterisk 1.4.14 and load average.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011403]: [patch] Remove old CLI style.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011385]: Option eventwhencalled seems not working properly.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010659]: CDRs are not merged for not answered calls   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0011274]: Zaptel: No Audio After First DTMF   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0005768]: [branch][post 1.4] LDAP Realtime driver   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011080]: SIP channel stops processing calls, but no apparent deadlock   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009492]: [patch] Asterisk sends wrong CSEQ in CANCEL if using INFO dtmf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009492]: [patch] Asterisk sends wrong CSEQ in CANCEL if using INFO dtmf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0009492]: [patch] Asterisk sends wrong CSEQ in CANCEL if using INFO dtmf   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011275]: On certain deadlocks, running core show locks segfaults asterisk and yields no lock info   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011407]: IAX crashing Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011408]: Buffer overflow when maxmsg not used for IMAP storage users   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011406]: New voicemail count incorrect when using IMAP storage and delete=yes flag   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011385]: Option eventwhencalled seems not working properly.   noreply at bugs.digium.com
• [asterisk-bugs] [Gastman 0011352]: Wrong argument in strncpy() causes problems with ztdynamic clients   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011386]: Asterisk 1.4.14 and load average.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011405]: [patch] free the returned data after a ast_load_realtime()   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011368]: chan_mobile does not recognize dtmf together with Authenticate or DISA   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011329]: AEL macro argument variables aren't properly quoted when set   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010540]: timeout value should accept floating point numbers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011408]: Buffer overflow when maxmsg not used for IMAP storage users   noreply at bugs.digium.com
• [asterisk-bugs] [AsteriskNOW 0011234]: AsteriskNOW File Editor Strips required characters from lines   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011383]: zaptel handel leak in meetme conference   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011383]: zaptel handel leak in meetme conference   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011383]: zaptel handel leak in meetme conference   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011399]: Crash with signal 6 on Channel Hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011340]: Allow channel unique IDs when searching for channels by name   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] ChanIsAvail always returns 0   Bruce Goldstein
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011399]: Crash with signal 6 on Channel Hangup   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011397]: Dont play video on console dial   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011413]: Strange freazing of the Manager and Asterisk Consol   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011413]: Strange freazing of the Manager and Asterisk Consol   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011394]: [patch] Add missing includes in SVN Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011397]: Dont play video on console dial   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0008290]: [patch] zap hookstate is never set offhook if wctdm module is loaded with an active fxo line   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011398]: [patch] Send Asterisk version to AGI   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011123]: [patch] Implement asterisk CLI permissions.   noreply at bugs.digium.com
• [asterisk-bugs] [Zaptel 0008290]: [patch] zap hookstate is never set offhook if wctdm module is loaded with an active fxo line   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011401]: Variable identifier not compatible with C++   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010185]: [patch]: Added automixmonitor feature / ability to use mixmonitor to record queue conversations on demand.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010530]: DTMFs passing only in part between two asterisk machines with an IAX2 connection   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011408]: Buffer overflow when maxmsg not used for IMAP storage users   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011417]: chanisavail always returns the same status   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011418]: DTMF buffer not purged if the dialing timeout   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011417]: chanisavail always returns the same status   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011394]: [patch] Add missing includes in SVN Trunk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011411]: [patch] Default penalty for member added using QueueAdd   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011420]: Calls barf and could crash on onelegged transfer   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011419]: Saving to IMAP folder (other than INBOX) not working with Cyrus IMAP   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011415]: Better quota handling with IMAP storage   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011402]: Interdigit timeout is half time of the defined time   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011388]: voicemail directory is not created until user is left a message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011245]: Asterisk unable to handle Multple Authorization Headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011402]: Interdigit timeout is half time of the defined time   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011423]: MySQL addons break latest asterisk 1.4.15   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011423]: MySQL addons break latest asterisk 1.4.15   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011423]: MySQL addons break latest asterisk 1.4.15   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011241]: Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011424]: Asterisk 1.4.15 breaks format_mp3.c build   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011413]: Strange freazing of the Manager and Asterisk Consol   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011292]: Unlock not locked.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011420]: Calls barf and could crash on onelegged transfer   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011292]: Unlock not locked.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011424]: Asterisk 1.4.15 breaks format_mp3.c build   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011423]: MySQL addons break latest asterisk 1.4.15   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011426]: Random seg faults...   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011420]: Calls barf and could crash on onelegged transfer   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011292]: Unlock not locked.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011220]: Email notification of voicemail segfaults Asterisk   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011417]: chanisavail always returns the same status   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011428]: incorrectly recognized codecs capabilities of windows mobile 6 device   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011427]: Asterisk crashes when using ODBC connected to Sybase   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011427]: Asterisk crashes when using ODBC connected to Sybase   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011428]: incorrectly recognized codecs capabilities of windows mobile 6 device   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011245]: Asterisk unable to handle Multple Authorization Headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011245]: Asterisk unable to handle Multple Authorization Headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011413]: Strange freazing of the Manager and Asterisk Consol   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011386]: Asterisk 1.4.14 and load average.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011245]: Asterisk unable to handle Multple Authorization Headers   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010185]: [patch]: Added automixmonitor feature / ability to use mixmonitor to record queue conversations on demand.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010185]: [patch]: Added automixmonitor feature / ability to use mixmonitor to record queue conversations on demand.   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0010530]: DTMFs passing only in part between two asterisk machines with an IAX2 connection   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011429]: Sent RTP video packets have a timestamp based on a 8000 Hz clock instead of 90000 Hz when mark bit is on   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011429]: Sent RTP video packets have a timestamp based on a 8000 Hz clock instead of 90000 Hz when mark bit is on   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011429]: Sent RTP video packets have a timestamp based on a 8000 Hz clock instead of 90000 Hz when mark bit is on   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011429]: Sent RTP video packets have a timestamp based on a 8000 Hz clock instead of 90000 Hz when mark bit is on   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011382]: Dial option G does not handle labels under some conditions   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011429]: Sent RTP video packets have a timestamp based on a 8000 Hz clock instead of 90000 Hz when mark bit is on   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011429]: Sent RTP video packets have a timestamp based on a 8000 Hz clock instead of 90000 Hz when mark bit is on   noreply at bugs.digium.com
• [asterisk-bugs] [Asterisk 0011281]: [patch] Check if config file changed before reloading the configuration   noreply at bugs.digium.com

Last message date: Fri Nov 30 19:35:13 CST 2007
Archived on: Fri Nov 30 19:36:25 CST 2007

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