November 2007 Archives by thread

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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