November 2007 Archives by thread
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