[Prev Page][Next Page]
- Re: Asterisk Log rotate not working
- Re: Asterisk Log rotate not working
- Re: Failed to authenticate device "Ext 110"
- Re: Failed to authenticate device "Ext 110"
- Re: Asterisk Log rotate not working
- Failed to authenticate device "Ext 110"
- Re: Asterisk Log rotate not working
- Re: Asterisk Log rotate not working
- Asterisk Log rotate not working
- Planned maintenance for community services on May 23, 2013
- From: Asterisk Development Team
- Re: 11.4: motif can only handle one channel at a time?
- Re: 11.4: motif can only handle one channel at a time?
- Stress testing Asterisk
- Re: Performance Asterisk large installation on Vmware/Xen
- Re: How to allow AMI access to Originate yet deny Application: System
- From: Alex Villacís Lasso
- Re: Passcode
- Re: Secure Calling
- Re: Passcode
- Re: Performance Asterisk large installation on Vmware/Xen
- Re: Loopback question
- Re: Question
- Passcode
- Secure Calling
- Loopback question
- Question
- Re: Asterisk 1.8-cert and AGC
- Re: Cut offs on outgoing SIP calls
- From: Philipp von Klitzing
- Re: Performance Asterisk large installation on Vmware/Xen
- Re: Performance Asterisk large installation on Vmware/Xen
- Re: Performance Asterisk large installation on Vmware/Xen
- Re: Performance Asterisk large installation on Vmware/Xen
- Re: Performance Asterisk large installation on Vmware/Xen
- Asterisk 1.8 vici and the fax, SMS, gtalk, Jaber channels
- Performance Asterisk large installation on Vmware/Xen
- From: Rafael dos Santos Saraiva
- Asterisk 1.8-cert and AGC
- From: Maximilian Grobecker
- Re: SetCallerPres questions
- From: Maximilian Grobecker
- Re: wanpipe and digium, oslec and hardware echo canceller
- Re: Auto dialer scripts and software
- Asterisk 11.4.0 Now Available
- From: Asterisk Development Team
- Asterisk 1.8.22.0 Now Available
- From: Asterisk Development Team
- Auto dialer scripts and software
- Re: wanpipe and digium, oslec and hardware echo canceller
- Temporarily features (transfer) off during Read
- wanpipe and digium, oslec and hardware echo canceller
- Re: Initial REGISTER Request: Contains Credentials before 401: KDDI Japan
- Planned maintenance for community services on May 16, 2013
- From: Asterisk Development Team
- Re: asterisk-users Digest, Vol 106, Issue 23
- Re: Asterisk Web Meetme module not loading
- Asterisk High-availability/failover solutions
- Re: dial and bridge
- Re: 11.4: motif can only handle one channel at a time?
- Re: 11.4: motif can only handle one channel at a time?
- 11.4: motif can only handle one channel at a time?
- AstriCon 2013 (our 10th AstriCon) needs YOU!
- Re: Call Transfer question
- From: qasimakhan@xxxxxxxxx
- Call Transfer question
- Re: Asterisk Web Meetme module not loading
- Re: Integrate Astreisk with SIP interface
- Initial REGISTER Request: Contains Credentials before 401: KDDI Japan
- SetCallerPres questions
- Re: Polycom and forwarding.
- Re: Cut offs on outgoing SIP calls
- Re: Cut offs on outgoing SIP calls
- Re: Polycom and forwarding.
- Re: Cut offs on outgoing SIP calls
- Re: Cut offs on outgoing SIP calls
- Re: Cut offs on outgoing SIP calls
- Polycom and forwarding.
- Re: Cut offs on outgoing SIP calls
- Re: Cut offs on outgoing SIP calls
- Cut offs on outgoing SIP calls
- Re: Initial REGISTER Request: Contains Credentials before 401
- How to allow AMI access to Originate yet deny Application: System
- From: Alex Villacís Lasso
- Re: 3. mfcr2 channel state IDLE 0x00 and call trace log file not ended ?? (Leonardo Rivanera)
- Re: dial and bridge
- Re: dial and bridge
- Re: dial and bridge
- Re: dial and bridge
- Initial REGISTER Request: Contains Credentials before 401
- Re: dial and bridge
- Re: dial and bridge
- Re: dial and bridge
- Re: Sangoma Wanpipe Driver
- Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database
- Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database
- Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database
- Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database
- Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database
- Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database
- Using PHPMyAdmin to remotely access Asterisk MySQL Database
- mfcr2 channel state IDLE 0x00 and call trace log file not ended ??
- Re: chanstats console errors
- Re: Monitoring SIP trunk status on call by call basis
- Re: dial and bridge
- Re: dial and bridge
- Re: dial and bridge
- Re: Monitoring SIP trunk status on call by call basis
- dial and bridge
- Monitoring SIP trunk status on call by call basis
- Call Diversion Override
- Re: Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial
- Re: Upgrade from 1.0.x to AsteriskNOW 3.0
- Re: Upgrade from 1.0.x to AsteriskNOW 3.0
- Re: Tier 1 Service Providers (AT&T, Level 3)
- Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial
- From: Daniel-Constantin Mierla
- Re: Upgrade from 1.0.x to AsteriskNOW 3.0
- Re: amiDebugger - might make your life easier if you program through the AMI
- Upgrade from 1.0.x to AsteriskNOW 3.0
- Re: Tier 1 Service Providers (AT&T, Level 3)
- amiDebugger - might make your life easier if you program through the AMI
- Re: [SOLVED] time zone setting in asterisk
- Re: dahdi driver not getting install
- Re: Sangoma Wanpipe Driver
- Re: Sangoma Wanpipe Driver
- Re: dahdi driver not getting install
- From: Salaheddine Elharit
- Re: Sangoma Wanpipe Driver
- Re: Sangoma Wanpipe Driver
- Sangoma Wanpipe Driver
- Re: Integrate Astreisk with SIP interface
- Re: time zone setting in asterisk
- Re: time zone setting in asterisk
- Re: ISP trunk session ID [SOLVED]
- Re: time zone setting in asterisk
- Re: Integrate Astreisk with SIP interface
- Re: Integrate Astreisk with SIP interface
- Re: time zone setting in asterisk
- Re: time zone setting in asterisk
- Integrate Astreisk with SIP interface
- Re: time zone setting in asterisk
- time zone setting in asterisk
- Re: AMI Originate issue
- HD Voice -- connecting Asterisk into HD Voice compatible mobile phone
- Re: dahdi driver not getting install
- AT&T uverse Motorolga nvg510
- Which channels are required for FAX, GTALK and Jaber
- Re: dahdi driver not getting install
- Re: dahdi driver not getting install
- AMI Originate issue
- Re: dahdi driver not getting install
- Re: dahdi driver not getting install
- Re: ISP trunk session ID?
- Re: dahdi driver not getting install
- dahdi driver not getting install
- Re: ISP trunk session ID?
- Re: ISP trunk session ID?
- 11.4: no incoming gv/xmpp
- Tier 1 Service Providers (AT&T, Level 3)
- Re: ISP trunk session ID?
- ISP trunk session ID?
- Re: Asterisk 12 and OPUS Codec
- Asterisk 12 and OPUS Codec
- Voicemail send to e-mail
- From: Bory's Rouliane Kouassi
- Re: monitoring Asterisk 1.8
- Re: monitoring Asterisk 1.8
- Re: Get Channel Variables in AMI Event NewExten
- Re: question about CDR
- From: Salaheddine Elharit
- Thanks! qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`
- Re: qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`
- Re: qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`
- qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`
- Re: Elastix vs vicidial
- Re: monitoring Asterisk 1.8
- Re: monitoring Asterisk 1.8
- DID providers
- Re: monitoring Asterisk 1.8
- Re: monitoring Asterisk 1.8
- Re: monitoring Asterisk 1.8
- Re: monitoring Asterisk 1.8
- Re: monitoring Asterisk 1.8
- Re: monitoring Asterisk 1.8
- Re: monitoring Asterisk 1.8
- Re: monitoring Asterisk 1.8
- Planned maintenance for community services on May 11, 2013
- From: Asterisk Development Team
- monitoring Asterisk 1.8
- chanstats console errors
- Re: Get Channel Variables in AMI Event NewExten
- Elastix vs vicidial
- Re: Get Channel Variables in AMI Event NewExten
- Re: Asterisk and hylafax: how to debug ...
- Re: question about CDR
- Re: question about CDR
- From: Salaheddine Elharit
- Re: question about CDR
- question about CDR
- From: Salaheddine Elharit
- Re: passing '302 moved temporarily' back to the SIP provider
- Re: passing '302 moved temporarily' back to the SIP provider
- Re: passing '302 moved temporarily' back to the SIP provider
- No early media on 302 redirect via two servers
- Transfer cmd via AsyncAGI
- Re: hwo to stok variable wiith menu
- From: Salaheddine Elharit
- Confbridge Dynamic video_mode
- Re: Obtaining high voice quality in dahdi channel
- Re: passing '302 moved temporarily' back to the SIP provider
- Re: Asterisk and hylafax: how to debug ...
- Obtaining high voice quality in dahdi channel
- Re: Asterisk and hylafax: how to debug ...
- Asterisk and hylafax: how to debug ...
- Re: What is bootstrap.sh for ? Possible bug in 11.3.0 ?
- Re: What is bootstrap.sh for ? Possible bug in 11.3.0 ?
- Re: AMI help needed
- Re: chan_alsa and confbridge
- passing '302 moved temporarily' back to the SIP provider
- From: Johann Steinwendtner
- Re: What is bootstrap.sh for ? Possible bug in 11.3.0 ?
- Re: НА: asterisk-users Digest, Vol 105, Issue 40
- Get Channel Variables in AMI Event NewExten
- (no subject)
- НА: asterisk-users Digest, Vol 105, Issue 40
- chan_alsa and confbridge
- Re: What is bootstrap.sh for ? Possible bug in 11.3.0 ?
- What is bootstrap.sh for ? Possible bug in 11.3.0 ?
- Re: Installing on an OpenVZ instance
- Installing on an OpenVZ instance
- Re: Joining an astablished call
- Re: Joining an astablished call
- Re: My new Polycom 450's can't xfer to 4-digit extension
- Re: OT - Differences between Aastra 6730i and 6750i series
- Re: Testing 911 call
- MRCPSynth() change voice
- OT - Question on Aastra 6735i - Was: Sip phone displaying caller name while on call
- OT - Differences between Aastra 6730i and 6750i series
- Re: Load Balancing
- Re: Joining an astablished call
- Re: Joining an astablished call
- Joining an astablished call
- Re: Testing 911 call
- Re: Testing 911 call
- Re: Testing 911 call
- Re: Cisco 9971 help
- Re: Cisco 9971 help
- Testing 911 call
- Re: GotoIf DIALSTATUS - not working
- Re: My new Polycom 450's can't xfer to 4-digit extension
- Re: debug strategy for one-way audio calls
- Re: debug strategy for one-way audio calls
- Re: Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"
- Re: Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"
- Re: Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"
- Re: Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"
- Re: GotoIf DIALSTATUS - not working
- Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"
- GotoIf DIALSTATUS - not working
- Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
- Re: BLF and asterisk Queue
- 11.4.-rc1: new segfault in iksemel ??
- My new Polycom 450's can't xfer to 4-digit extension
- Re: Cisco 9971 help
- Re: debug strategy for one-way audio calls
- AMI help needed
- Re: Cisco 9971 help
- Cisco 9971 help
- Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000
- From: Johann Steinwendtner
- Digium D70 visual voicemail - won't play
- From: Dr. Michael J. Chudobiak
- Re: changing ringtones to a group of phones
- From: Dr. Michael J. Chudobiak
- Re: changing ringtones to a group of phones
- changing ringtones to a group of phones
- From: Dr. Michael J. Chudobiak
- Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000
- Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000
- Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000
- Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000
- Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000
- Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000
- Re: debug strategy for one-way audio calls
- VoIP Incoming Issue
- Re: Building Asterisk 11.4.0-rc1 with PJSIP 2.1
- Re: Playing a sound file during a call
- Re: Playing a sound file during a call
- Re: Playing a sound file during a call
- Re: looking for a way to do appointment reminders
- Re: Playing a sound file during a call
- Re: Playing a sound file during a call
- Re: Playing a sound file during a call
- Re: Playing a sound file during a call
- Re: Playing a sound file during a call
- Playing a sound file during a call
- Building Asterisk 11.4.0-rc1 with PJSIP 2.1
- Re: looking for a way to do appointment reminders
- Re: debug strategy for one-way audio calls
- debug strategy for one-way audio calls
- Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000
- Queues with different technologies for members, like Khomp Driver
- Re: Call "stuck" in queue
- Call "stuck" in queue
- Re: multiple provider for incoming
- Re: multiple provider for incoming
- Re: SMS Scenario
- SMS Scenario
- Re: asterisk-users Digest, Vol 105, Issue 39
- Re: Gateway?
- Re: multiple provider for incoming
- Re: multiple provider for incoming
- Re: multiple provider for incoming
- Re: multiple provider for incoming
- Re: multiple provider for incoming
- Re: multiple provider for incoming
- Re: multiple provider for incoming
- Re: multiple provider for incoming
- Re: multiple provider for incoming
- multiple provider for incoming
- asterisk 1.4 and SMS module
- Re: Gateway?
- Re: Gateway?
- Re: Gateway?
- Asterisk QSIG doesnt send the calling name to Nortel CS1000
- Re: Gateway?
- Re: Can't register to Asterisk 1.6 with old Aastra phones
- Re: Gateway?
- Re: Asterisk 11.3.0 - Mask for new file not correct
- Re: Gateway?
- Gateway?
- Re: Asterisk 11.3.0 - Mask for new file not correct
- Re: Can't register to Asterisk 1.6 with old Aastra phones
- Asterisk 11.3.0 - Mask for new file not correct
- Re: Can't register to Asterisk 1.6 with old Aastra phones
- Re: Radius Based Accounting for Asterisk
- Re: Can't register to Asterisk 1.6 with old Aastra phones
- Can't register to Asterisk 1.6 with old Aastra phones
- Re: Sip and the media path
- From: qasimakhan@xxxxxxxxx
- Re: looking for a way to do appointment reminders
- Re: Radius Based Accounting for Asterisk
- glibc detected crash
- caller_id vs cid_number
- Re: h323-sip: one way connection
- Re: looking for a way to do appointment reminders
- Re: looking for a way to do appointment reminders
- Re: looking for a way to do appointment reminders
- Re: looking for a way to do appointment reminders
- Re: looking for a way to do appointment reminders
- Re: looking for a way to do appointment reminders
- Re: looking for a way to do appointment reminders
- Re: looking for a way to do appointment reminders
- Re: h323-sip: one way connection
- Re: looking for a way to do appointment reminders
- Re: looking for a way to do appointment reminders
- looking for a way to do appointment reminders
- Users appending # sign when dialing an extension from automated greeting
- Re: Asterisk Calendar integration suggestions
- Re: h323-sip: one way connection
- Re: h323-sip: one way connection
- Re: Load Balancing
- Re: Sip and the media path
- Re: Sip and the media path
- Load Balancing
- Re: Asterisk Calendar integration suggestions
- Asterisk Calendar integration suggestions
- Re: Sip and the media path
- Re: Asterisk 1.8 and 11
- From: qasimakhan@xxxxxxxxx
- Sip and the media path
- Asterisk 1.8 and 11
- Re: Jitter Buffer in asterisk 1.8.11.0
- From: qasimakhan@xxxxxxxxx
- Re: CDR Question
- Re: h323-sip: one way connection
- Re: Fundemental changes to CDR within single asterisk family
- Re: Fundemental changes to CDR within single asterisk family
- Re: h323-sip: one way connection
- Asterisk 11.4.0-rc1 refuses to use the TURN server
- Re: Dialplan reload not reloading everything
- Re: cdr report
- Re: h323-sip: one way connection
- Re: cdr report
- Re: cdr report
- Re: cdr report
- Re: cdr report
- Re: cdr report
- Re: cdr report
- cdr report
- Re: Installing Asterisk on Virtual Machine
- Re: asterisk music on hold recommendations
- Re: asterisk music on hold recommendations
- Re: Installing Asterisk on Virtual Machine
- Re: Installing Asterisk on Virtual Machine
- Re: Installing Asterisk on Virtual Machine
- asterisk music on hold recommendations
- Dialplan reload not reloading everything
- Jitter Buffer in asterisk 1.8.11.0
- Re: Installing Asterisk on Virtual Machine
- Re: Installing Asterisk on Virtual Machine
- Re: Installing Asterisk on Virtual Machine
- Re: h323-sip: one way connection
- Re: Installing Asterisk on Virtual Machine
- Re: Installing Asterisk on Virtual Machine
- /dev/dahdi/pseudo leaking
- Asterisk Tech Job Posting Dallas Texas
- Planned maintenance for community services on April 22, 2013
- From: Asterisk Development Team
- Re: h323-sip: one way connection
- H.264 high profile support
- Device states
- Re: Installing Asterisk on Virtual Machine
- Re: Installing Asterisk on Virtual Machine
- Re: Installing Asterisk on Virtual Machine
- Re: Installing Asterisk on Virtual Machine
- Installing Asterisk on Virtual Machine
- h323-sip: one way connection
- Re: Strange problem with Asterisk 1.8.9.3
- Re: CDR Question
- Re: CDR Question
- Re: Strange problem with Asterisk 1.8.9.3
- Re: Strange problem with Asterisk 1.8.9.3
- Re: Strange problem with Asterisk 1.8.9.3
- Strange problem with Asterisk 1.8.9.3
- Re: CDR Question
- CDR Question
- Re: E911 Voip Trunking
- Re: E911 Voip Trunking
- Re: E911 Voip Trunking
- Re: Dynamic realtime + queues.conf Unresolved
- Dynamic realtime + queues.conf Unresolved
- Re: E911 Voip Trunking
- Re: E911 Voip Trunking
- set google voice callerid as Unknown/Unavailable ?
- Re: External call control for Asterisk
- Sip phone displaying caller name while on call
- E911 Voip Trunking
- Re: ODBC dialplan looping problem
- Re: ODBC dialplan looping problem
- To enhance the voice quality of the SIP trunk
- Re: 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
- Re: External call control for Asterisk
- 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
- Re: Phpagi action based on outbound call user response
- Re: Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
- Fw: Dynamic realtime + queues
- Re: Dynamic realtime + queues
- From: Jose Flores Galicia
- Re: Dynamic realtime + queues
- Re: Dynamic realtime + queues
- Dynamic realtime + queues
- Re: Dynamic realtime + queues
- Dynamic realtime + queues
- Dynamic realtime + queues
- Re: ODBC dialplan looping problem
- Re: ODBC dialplan looping problem
- Re: ODBC dialplan looping problem
- Re: ODBC dialplan looping problem
- Re: ODBC dialplan looping problem
- Re: ODBC dialplan looping problem
- Re: ODBC dialplan looping problem
- Re: ODBC dialplan looping problem
- Re: ODBC dialplan looping problem
- ODBC dialplan looping problem
- Re: CLI flood : requested media update control 26
- Re: Transfer only, no outbound calling
- Re: On SIP INVITE answering to IP:port found in Contact: header.
- Re: How to show caller number ?
- How to show caller number ?
- Re: Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
- Re: On SIP INVITE answering to IP:port found in Contact: header.
- Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
- Users.conf vs Sip.conf
- Re: core console debug on single file
- core console debug on single file
- Caller ID is not persisted when using Channel Redirect
- failed to extend from 512 to 676 on cli
- Re: On SIP INVITE answering to IP:port found in Contact: header.
- Phpagi action based on outbound call user response
- Re: Access postgresql directly from dialplan?
- Re: Transfer only, no outbound calling
- Transfer only, no outbound calling
- Re: On SIP INVITE answering to IP:port found in Contact: header.
- Re: On SIP INVITE answering to IP:port found in Contact: header.
- Re: On SIP INVITE answering to IP:port found in Contact: header.
- Re: On SIP INVITE answering to IP:port found in Contact: header.
- On SIP INVITE answering to IP:port found in Contact: header.
- Re: erro compiling dahdi
- Re: erro compiling dahdi
- erro compiling dahdi
- Re: Access postgresql directly from dialplan?
- Access postgresql directly from dialplan?
- Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists"
- Re: Asterisk SIP TCP
- Re: Asterisk SIP TCP
- Re: Asterisk SIP TCP
- Re: Asterisk SIP TCP
- Re: Asterisk SIP TCP
- Re: Traffic Crossover
- Traffic Crossover
- Re: Asterisk SIP TCP
- Re: Asterisk SIP TCP
- Re: Asterisk SIP TCP
- Asterisk SIP TCP
- Re: Dial multiple device cancellation
- Re: Dial multiple device cancellation
- Re: Dial multiple device cancellation
- Re: Dial multiple device cancellation
- Dial multiple device cancellation
- Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists"
- Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists"
- Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists"
- Re: Polycom Soundpoint IP 330 provisioning
- Re: Network based transcoding
- Re: Progress() on outgoing calls
- Re: Polycom Soundpoint IP 330 provisioning
- Re: Polycom Soundpoint IP 330 provisioning
- Re: Polycom Soundpoint IP 330 provisioning
- Re: Polycom Soundpoint IP 330 provisioning
- Polycom Soundpoint IP 330 provisioning
- Re: Is there a php script to analyse and show call detail reports from Asterisk CDR?
- Re: Network based transcoding
- Re: (no subject)
- (no subject)
- Re: Network based transcoding
- Re: Network based transcoding
- Re: Network based transcoding
- Re: Network based transcoding
- Re: Network based transcoding
- Network based transcoding
- Progress() on outgoing calls
- Re: PRI DEBUG
- Re: Setting a CDR field from using feature codes...
- Re: Setting a CDR field from using feature codes...
- Re: Setting a CDR field from using feature codes...
- Re: Setting a CDR field from using feature codes...
- Re: Setting a CDR field from using feature codes...
- Re: PRI DEBUG
- Re: Is there a php script to analyse and show call detail reports from Asterisk CDR?
- Re: Setting a CDR field from using feature codes...
- Voicemail Prepend not working properly on 1.8.18
- Re: Setting a CDR field from using feature codes...
- Re: Asterisk 11.2.1 / dahdi destroy channel / asteriskcrashes
- Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists"
- Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists"
- Re: Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes
- Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists"
- "Dropping call because extensions '200', 's' and 'i' doesn't exists"
- Re: Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes
- PRI DEBUG
- Re: Asterisk 11.2.1 / dahdi destroy channel / asteriskcrashes
- Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes
- Re: Logging SIP connection status for review
- Re: ACD problem
- Re: ACD problem
- Re: AMI Reload action, returning generated errors?
- Re: Logging SIP connection status for review
- Setting a CDR field from using feature codes...
- Follow Me CID
- Re: Is there a php script to analyse and show call detail reports from Asterisk CDR?
- Re: ACD problem & outbound calls
- AMI Reload action, returning generated errors?
- Re: ACD problem
- Re: ACD problem
- ACD problem
- Re: Logging SIP connection status for review
- Re: Logging SIP connection status for review
- Re: Logging SIP connection status for review
- Logging SIP connection status for review
- Re: my "blacklist" is not working
- Re: my "blacklist" is not working
- Re: OT - How to simulate public IPs for lab testing
- Re: Feature request: What about a new DB_IFEXISTS function ?
- Re: Feature request: What about a new DB_IFEXISTS function ?
- Re: External call control for Asterisk
- Re: External call control for Asterisk
- External call control for Asterisk
- Re: my "blacklist" is not working
- Re: my "blacklist" is not working
- Re: Asterisk Peaking and 91 Calls And not a Dime More!
- my "blacklist" is not working
- Re: Asterisk Peaking and 91 Calls And not a Dime More!
- realtime peer w/ callbackextension does not register after 'sip reload'
- Re: Asterisk Peaking and 91 Calls And not a Dime More!
- Re: Asterisk Peaking and 91 Calls And not a Dime More!
- Re: [OpenSIPS-Users] 404 When BYE initiated by external callee
- Re: Asterisk Peaking and 91 Calls And not a Dime More!
- Re: [OpenSIPS-Users] 404 When BYE initiated by external callee
- Re: Connect to an outbound channel and dial a phone number??
- Connect to an outbound channel and dial a phone number??
- Re: Looking for a reporter for SQLite3 with Lighttpd and PHP
- Re: [OpenSIPS-Users] 404 When BYE initiated by external callee
- Re: Asterisk Peaking and 91 Calls And not a Dime More!
- Re: [OpenSIPS-Users] 404 When BYE initiated by external callee
- Re: [OpenSIPS-Users] 404 When BYE initiated by external callee
- Re: [OpenSIPS-Users] 404 When BYE initiated by external callee
- Asterisk Peaking and 91 Calls And not a Dime More!
- Re: Looking for a reporter for SQLite3 with Lighttpd and PHP
- Re: [OpenSIPS-Users] 404 When BYE initiated by external callee
- Re: [OpenSIPS-Users] 404 When BYE initiated by external callee
- Re: Looking for a reporter for SQLite3 with Lighttpd and PHP
- Re: Cisco SPA 5xx/3xx/9xx phones don't respond to SIPAddHeader(Call-Info: answer-after=0)
- Feature request: What about a new DB_IFEXISTS function ?
- Re: CDR unanswered setting
- CDR unanswered setting
- dahdi "strange state" error
- Re: OT - How to simulate public IPs for lab testing
- Re: OT - How to simulate public IPs for lab testing
- OT - How to simulate public IPs for lab testing
- Re: extensions.conf / test DID
- Re: extensions.conf / test DID
- Re: extensions.conf / test DID
- Re: extensions.conf / test DID
- Re: extensions.conf / test DID
- extensions.conf / test DID
- [Asterisk 1.6] Mysql cdr addon doen't write full channel infomation when disposition is Failed
- Broadvoice/MWI?
- Re: sip registration
- Re: sip registration
- Re: blacklist/V* - using wildcard
- Re: sip registration
- Re: blacklist/V* - using wildcard
- Re: Asterisk SIP deadlocks - update_provisional_keepalive
- sip registration
- Re: blacklist/V* - using wildcard
- Re: blacklist/V* - using wildcard
- blacklist/V* - using wildcard
- Re: ring group failure with "ExtensionState: 4"
- Re: fax - sound/tone - dealing with SPAM
- Re: fax - sound/tone - dealing with SPAM
- Re: fax - sound/tone - dealing with SPAM
- Re: fax - sound/tone - dealing with SPAM
- Re: fax - sound/tone - dealing with SPAM
- Re: fax - sound/tone - dealing with SPAM
- Re: fax - sound/tone - dealing with SPAM
- Re: fax - sound/tone - dealing with SPAM
- Re: fax - sound/tone - dealing with SPAM
- fax - sound/tone - dealing with SPAM
- Re: Asterisk SIP deadlocks - update_provisional_keepalive
- Re: Asterisk SIP deadlocks - update_provisional_keepalive
- Re: Fundemental changes to CDR within single asterisk family
- ring group failure with "ExtensionState: 4"
- Re: Asterisk SIP deadlocks - update_provisional_keepalive
- Re: Asterisk SIP deadlocks - update_provisional_keepalive
- Re: TigerJet 320G Chip / TDM400 Chipset / DAHDI Support
- Re: TigerJet 320G Chip / TDM400 Chipset / DAHDI Support
- Re: [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
- Re: TigerJet 320G Chip / TDM400 Chipset / DAHDI Support
- Re: TigerJet 320G Chip / TDM400 Chipset / DAHDI Support
- Re: TigerJet 320G Chip / TDM400 Chipset / DAHDI Support
- Asterisk SIP deadlocks - update_provisional_keepalive
- TigerJet 320G Chip / TDM400 Chipset / DAHDI Support
- Re: Feature request: Need to INVITE to peer with other domain without peer domain addition
- Re: CLI flood : requested media update control 26
- Re: CLI flood : requested media update control 26
- Re: CLI flood : requested media update control 26
- Re: CLI flood : requested media update control 26
- Re: CLI flood : requested media update control 26
- CLI flood : requested media update control 26
- Re: [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
- Getting DIALSTATUS via agi
- Re: [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
- Re: [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
- Re: Feature request: Need to INVITE to peer with other domain without peer domain addition
- Re: Feature request: Need to INVITE to peer with other domain without peer domain addition
- FreePBX, Asterisk and Twinkle - Testing a new setup
- Re: [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
- Re: [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
- Re: asterisk-users Digest, Vol 104, Issue 53
- Re: Feature request: Need to INVITE to peer with other domain without peer domain addition
- SRTP woes
- Feature request: Need to INVITE to peer with other domain without peer domain addition
- Re: ISDN- E1 PRI module in network side signaling
- Re: ISDN- E1 PRI module in network side signaling
- Re: ISDN- E1 PRI module in network side signaling
- ISDN- E1 PRI module in network side signaling
- Re: IPv6
- IPv6
- Re: Pattern matching repeating digits
- Re: Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
- Re: Asterisk 11 -CDR values changed in hangup handler not saved ?
- Re: Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
- From: Julian Lyndon-Smith
- Getting Unknown Error while configuring Asterisk with Linux HA
- Re: Asterisk 11 -CDR values changed in hangup handler not saved ?
- Re: To queue or not to queue...
- Re: Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
- Re: "sip set debug on" output to file only (not to console)
- Re: "sip set debug on" output to file only (not to console)
- Re: Asterisk 11 -CDR values changed in hangup handler not saved ?
- Re: "sip set debug on" output to file only (not to console)
- Re: "sip set debug on" output to file only (not to console)
- Re: "sip set debug on" output to file only (not to console)
- "sip set debug on" output to file only (not to console)