April 2013 Archives by date

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Starting: Mon Apr 1 11:01:50 CDT 2013
Ending: Tue Apr 30 23:33:22 CDT 2013
Messages: 355

• [asterisk-users] FreePBX, Asterisk and Twinkle - Testing a new setup   James B. Byrne
• [asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition   Dmitriy Serov
• [asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition   Paul Belanger
• [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled   Daniel Pocock
• [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled   Joshua Colp
• [asterisk-users] Getting DIALSTATUS via agi   Mike Diehl
• [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled   Daniel Pocock
• [asterisk-users] CLI flood : requested media update control 26   Jonas Kellens
• [asterisk-users] CLI flood : requested media update control 26   A J Stiles
• [asterisk-users] CLI flood : requested media update control 26   Jonas Kellens
• [asterisk-users] CLI flood : requested media update control 26   A J Stiles
• [asterisk-users] CLI flood : requested media update control 26   Jonas Kellens
• [asterisk-users] CLI flood : requested media update control 26   Matthew Jordan
• [asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition   Dmitriy Serov
• [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support   Marshall Henderson
• [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive   Duane Larson
• [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support   Tzafrir Cohen
• [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support   Marshall Henderson
• [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support   Patrick Lists
• [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled   Joshua Colp
• [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support   Marshall Henderson
• [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support   Patrick Lists
• [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive   Duane Larson
• [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive   Duane Larson
• [asterisk-users] ring group failure with "ExtensionState: 4"   asterisk at newkirk.us
• [asterisk-users] Fundemental changes to CDR within single asterisk family   Ishfaq Malik
• [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive   Jim Lucas
• [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive   Duane Larson
• [asterisk-users] fax - sound/tone - dealing with SPAM   Joseph
• [asterisk-users] fax - sound/tone - dealing with SPAM   Mitul Limbani
• [asterisk-users] fax - sound/tone - dealing with SPAM   Joseph
• [asterisk-users] fax - sound/tone - dealing with SPAM   Doug Lytle
• [asterisk-users] fax - sound/tone - dealing with SPAM   Joseph
• [asterisk-users] fax - sound/tone - dealing with SPAM   Doug Lytle
• [asterisk-users] fax - sound/tone - dealing with SPAM   Steve Edwards
• [asterisk-users] fax - sound/tone - dealing with SPAM   Patrick Lists
• [asterisk-users] fax - sound/tone - dealing with SPAM   Joseph
• [asterisk-users] fax - sound/tone - dealing with SPAM   Doug Lytle
• [asterisk-users] ring group failure with "ExtensionState: 4"   C Goodwin
• [asterisk-users] blacklist/V* - using wildcard   Joseph
• [asterisk-users] blacklist/V* - using wildcard   Joseph
• [asterisk-users] blacklist/V* - using wildcard   Doug Lytle
• [asterisk-users] sip registration   Thomas Perron
• [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive   Duane Larson
• [asterisk-users] blacklist/V* - using wildcard   Joseph
• [asterisk-users] sip registration   Steve Edwards
• [asterisk-users] blacklist/V* - using wildcard   Doug Lytle
• [asterisk-users] sip registration   Thomas Perron
• [asterisk-users] sip registration   Steve Edwards
• [asterisk-users] Broadvoice/MWI?   Nick B
• [asterisk-users] [Asterisk 1.6] Mysql cdr addon doen't write full channel infomation when disposition is Failed   Trung Nguyen Dac
• [asterisk-users] extensions.conf / test DID   Thomas Perron
• [asterisk-users] extensions.conf / test DID   A J Stiles
• [asterisk-users] extensions.conf / test DID   Jacob.E.Miles at L-3Com.com
• [asterisk-users] extensions.conf / test DID   Satish Barot
• [asterisk-users] extensions.conf / test DID   Doug Lytle
• [asterisk-users] extensions.conf / test DID   Steve Edwards
• [asterisk-users] OT - How to simulate public IPs for lab testing   Olivier
• [asterisk-users] OT - How to simulate public IPs for lab testing   Johan Wilfer
• [asterisk-users] OT - How to simulate public IPs for lab testing   Olivier
• [asterisk-users] dahdi "strange state" error   Greg Woods
• [asterisk-users] CDR unanswered setting   Shanavaz E A
• [asterisk-users] CDR unanswered setting   Marie Fischer
• [asterisk-users] Feature request: What about a new DB_IFEXISTS function ?   Olivier
• [asterisk-users] Cisco SPA 5xx/3xx/9xx phones don't respond to SIPAddHeader(Call-Info: answer-after=0)   adamk at 3a.hu
• [asterisk-users] Looking for a reporter for SQLite3 with Lighttpd and PHP   Daniel - Asterisk
• [asterisk-users] Looking for a reporter for SQLite3 with Lighttpd and PHP   jon pounder
• [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee   Nick Khamis
• [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee   Joshua Colp
• [asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!   Nick Khamis
• [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee   Nick Khamis
• [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee   Joshua Colp
• [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee   Nick Khamis
• [asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!   Paul Belanger
• [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee   Joshua Colp
• [asterisk-users] Looking for a reporter for SQLite3 with Lighttpd and PHP   Daniel - Asterisk
• [asterisk-users] Connect to an outbound channel and dial a phone number??   Thomas Perron
• [asterisk-users] Connect to an outbound channel and dial a phone number??   Marie Fischer
• [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee   Nick Khamis
• [asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!   Nick Khamis
• [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee   Nathan Anderson
• [asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!   Marie Fischer
• [asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!   Nick Khamis
• [asterisk-users] realtime peer w/ callbackextension does not register after 'sip reload'   Marie Fischer
• [asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!   Steve Edwards
• [asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!   Nick Khamis
• [asterisk-users] my "blacklist" is not working   Joseph
• [asterisk-users] my "blacklist" is not working   Joseph
• [asterisk-users] my "blacklist" is not working   Joseph
• [asterisk-users] External call control for Asterisk   Simon Green
• [asterisk-users] External call control for Asterisk   Steve Edwards
• [asterisk-users] External call control for Asterisk   Simon Green
• [asterisk-users] Feature request: What about a new DB_IFEXISTS function ?   Satish Barot
• [asterisk-users] Feature request: What about a new DB_IFEXISTS function ?   Olivier
• [asterisk-users] OT - How to simulate public IPs for lab testing   Johan Wilfer
• [asterisk-users] my "blacklist" is not working   Matthew Jordan
• [asterisk-users] my "blacklist" is not working   Joseph
• [asterisk-users] Logging SIP connection status for review   Carlos Alvarez
• [asterisk-users] Logging SIP connection status for review   Steve Edwards
• [asterisk-users] Logging SIP connection status for review   Carlos Alvarez
• [asterisk-users] Logging SIP connection status for review   Ron Wheeler
• [asterisk-users] ACD problem   Tommy Cooper
• [asterisk-users] ACD problem   Paul Belanger
• [asterisk-users] ACD problem   Salman Zafar
• [asterisk-users] AMI Reload action, returning generated errors?   John Kiniston
• [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?   Daniel - Asterisk
• [asterisk-users] ACD problem & outbound calls   Tommy Cooper
• [asterisk-users] Follow Me CID   Nyamul Hassan
• [asterisk-users] Setting a CDR field from using feature codes...   Carlos Chavez
• [asterisk-users] Logging SIP connection status for review   Duncan Turnbull
• [asterisk-users] AMI Reload action, returning generated errors?   Trung Nguyen Dac
• [asterisk-users] ACD problem   Bharat Lalcheta
• [asterisk-users] ACD problem   Lenz Emilitri
• [asterisk-users] Logging SIP connection status for review   Ishfaq Malik
• [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes   Thorsten Göllner
• [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asteriskcrashes   Alec Davis
• [asterisk-users] PRI DEBUG   Yves A.
• [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes   Yves A.
• [asterisk-users] "Dropping call because extensions '200', 's' and 'i' doesn't exists"   s m
• [asterisk-users] "Dropping call because extensions '200', 's' and 'i' doesn't exists"   A J Stiles
• [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes   Thorsten Göllner
• [asterisk-users] "Dropping call because extensions '200', 's' and 'i' doesn't exists"   s m
• [asterisk-users] "Dropping call because extensions '200', 's' and 'i' doesn't exists"   Asghar Mohammad
• [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asteriskcrashes   Richard Mudgett
• [asterisk-users] Setting a CDR field from using feature codes...   Asghar Mohammad
• [asterisk-users] Voicemail Prepend not working properly on 1.8.18   James Lamanna
• [asterisk-users] Setting a CDR field from using feature codes...   Carlos Chavez
• [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?   Asghar Mohammad
• [asterisk-users] PRI DEBUG   Richard Mudgett
• [asterisk-users] Setting a CDR field from using feature codes...   Asghar Mohammad
• [asterisk-users] Setting a CDR field from using feature codes...   Carlos Chavez
• [asterisk-users] Setting a CDR field from using feature codes...   Asghar Mohammad
• [asterisk-users] Setting a CDR field from using feature codes...   Carlos Chavez
• [asterisk-users] Setting a CDR field from using feature codes...   Asghar Mohammad
• [asterisk-users] PRI DEBUG   Yves A.
• [asterisk-users] Progress() on outgoing calls   Jonas Kellens
• [asterisk-users] Network based transcoding   Nick Khamis
• [asterisk-users] Network based transcoding   Eric Wieling
• [asterisk-users] Network based transcoding   jg
• [asterisk-users] Network based transcoding   Nick Khamis
• [asterisk-users] Network based transcoding   Chris Bagnall
• [asterisk-users] Network based transcoding   jg
• [asterisk-users] (no subject)   Thomas Perron
• [asterisk-users] (no subject)   A J Stiles
• [asterisk-users] Network based transcoding   Nick Khamis
• [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?   Daniel - Asterisk
• [asterisk-users] Biggest Fake Conference in Computer Science   georgepeter at hushmail.com
• [asterisk-users] Polycom Soundpoint IP 330 provisioning   Daniel - Asterisk
• [asterisk-users] Polycom Soundpoint IP 330 provisioning   Kevin Larsen
• [asterisk-users] Polycom Soundpoint IP 330 provisioning   Daniel - Asterisk
• [asterisk-users] Polycom Soundpoint IP 330 provisioning   Daniel - Asterisk
• [asterisk-users] Polycom Soundpoint IP 330 provisioning   Dave Fullerton
• [asterisk-users] Progress() on outgoing calls   Rusty Newton
• [asterisk-users] Network based transcoding   Nick Khamis
• [asterisk-users] Polycom Soundpoint IP 330 provisioning   Daniel - Asterisk
• [asterisk-users] "Dropping call because extensions '200', 's' and 'i' doesn't exists"   s m
• [asterisk-users] "Dropping call because extensions '200', 's' and 'i' doesn't exists"   Asghar Mohammad
• [asterisk-users] "Dropping call because extensions '200', 's' and 'i' doesn't exists"   Gertjan Baarda
• [asterisk-users] jrasoeu at gmail.com wants to follow you. Accept?   jrasoeu at gmail.com
• [asterisk-users] Dial multiple device cancellation   Santi Anton
• [asterisk-users] Dial multiple device cancellation   jg
• [asterisk-users] Dial multiple device cancellation   Satish Barot
• [asterisk-users] Dial multiple device cancellation   Olivier
• [asterisk-users] Dial multiple device cancellation   Santi Anton
• [asterisk-users] Asterisk SIP TCP   Zohair Raza
• [asterisk-users] Asterisk SIP TCP   Mehroz Ashraf
• [asterisk-users] Asterisk SIP TCP   Mark Henry
• [asterisk-users] Asterisk SIP TCP   Mark Henry
• [asterisk-users] Traffic Crossover   Geoffrey Yeoh
• [asterisk-users] Traffic Crossover   Eric Wieling
• [asterisk-users] Asterisk SIP TCP   Zohair Raza
• [asterisk-users] Asterisk SIP TCP   Bharat Lalcheta
• [asterisk-users] Asterisk SIP TCP   Zohair Raza
• [asterisk-users] Asterisk SIP TCP   Bharat Lalcheta
• [asterisk-users] Asterisk SIP TCP   Zohair Raza
• [asterisk-users] "Dropping call because extensions '200', 's' and 'i' doesn't exists"   s m
• [asterisk-users] Access postgresql directly from dialplan?   Sebastian Arcus
• [asterisk-users] Access postgresql directly from dialplan?   Gertjan Baarda
• [asterisk-users] erro compiling dahdi   Jonas Kellens
• [asterisk-users] erro compiling dahdi   Shaun Ruffell
• [asterisk-users] erro compiling dahdi   Russ Meyerriecks
• [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.   Markus
• [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.   Matthew J. Roth
• [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.   Joshua Colp
• [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.   Matthew J. Roth
• [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.   Markus
• [asterisk-users] Transfer only, no outbound calling   Todd Routhier
• [asterisk-users] Transfer only, no outbound calling   Nathan Anderson
• [asterisk-users] Access postgresql directly from dialplan?   Sebastian Arcus
• [asterisk-users] Phpagi action based on outbound call user response   Rahul R
• [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.   Joshua Colp
• [asterisk-users] failed to extend from 512 to 676 on cli   Kamlesh Kumar
• [asterisk-users] Caller ID is not persisted when using Channel Redirect   Jacob.E.Miles at L-3Com.com
• [asterisk-users] core console debug on single file   Gabriel Ortiz Lour
• [asterisk-users] core console debug on single file   Richard Mudgett
• [asterisk-users] Users.conf vs Sip.conf   Bryan Anderson
• [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger   bilal ghayyad
• [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.   Matthew J. Roth
• [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger   isrlgb at gmail.com
• [asterisk-users] How to show caller number ?   neo haux
• [asterisk-users] How to show caller number ?   Steve Edwards
• [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.   Matthew J. Roth
• [asterisk-users] Transfer only, no outbound calling   Todd Routhier
• [asterisk-users] CLI flood : requested media update control 26   Jonas Kellens
• [asterisk-users] ODBC dialplan looping problem   Pat Collins
• [asterisk-users] ODBC dialplan looping problem   Doug Lytle
• [asterisk-users] ODBC dialplan looping problem   Bharat Lalcheta
• [asterisk-users] ODBC dialplan looping problem   Pat Collins
• [asterisk-users] ODBC dialplan looping problem   Pat Collins
• [asterisk-users] ODBC dialplan looping problem   Bharat Lalcheta
• [asterisk-users] ODBC dialplan looping problem   jg
• [asterisk-users] ODBC dialplan looping problem   Pat Collins
• [asterisk-users] ODBC dialplan looping problem   Dale Noll
• [asterisk-users] ODBC dialplan looping problem   Matthew Jordan
• [asterisk-users] Dynamic realtime + queues   Tommy Cooper
• [asterisk-users] Dynamic realtime + queues   Tommy Cooper
• [asterisk-users] Dynamic realtime + queues   Leandro Dardini
• [asterisk-users] Dynamic realtime + queues   Tommy Cooper
• [asterisk-users] Dynamic realtime + queues   Leandro Dardini
• [asterisk-users] Dynamic realtime + queues   Tommy Cooper
• [asterisk-users] Dynamic realtime + queues   Jose Flores Galicia
• [asterisk-users] Fw: Dynamic realtime + queues   Tommy Cooper
• [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger   Lenz Emilitri
• [asterisk-users] Phpagi action based on outbound call user response   Lenz Emilitri
• [asterisk-users] 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger   kingman chui
• [asterisk-users] External call control for Asterisk   Lenz Emilitri
• [asterisk-users] 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger   Lenz Emilitri
• [asterisk-users] To enhance the voice quality of the SIP trunk   bilal ghayyad
• [asterisk-users] ODBC dialplan looping problem   Satish Barot
• [asterisk-users] ODBC dialplan looping problem   Satish Barot
• [asterisk-users] E911 Voip Trunking   Chris Nighswonger
• [asterisk-users] Sip phone displaying caller name while on call   Olivier
• [asterisk-users] External call control for Asterisk   Asghar Mohammad
• [asterisk-users] set google voice callerid as Unknown/Unavailable ?   sean darcy
• [asterisk-users] E911 Voip Trunking   Terry Brummell
• [asterisk-users] E911 Voip Trunking   Chris Nighswonger
• [asterisk-users] Dynamic realtime + queues.conf Unresolved   Tommy Cooper
• [asterisk-users] Dynamic realtime + queues.conf Unresolved   Nathan Anderson
• [asterisk-users] E911 Voip Trunking   Warren Selby
• [asterisk-users] E911 Voip Trunking   Nathan Anderson
• [asterisk-users] E911 Voip Trunking   Chris Nighswonger
• [asterisk-users] CDR Question   Nyamul Hassan
• [asterisk-users] CDR Question   jg
• [asterisk-users] Strange problem with Asterisk 1.8.9.3   Dereck D
• [asterisk-users] Strange problem with Asterisk 1.8.9.3   dotnetdub
• [asterisk-users] Strange problem with Asterisk 1.8.9.3   Dereck D
• [asterisk-users] Strange problem with Asterisk 1.8.9.3   Dereck D
• [asterisk-users] CDR Question   Nyamul Hassan
• [asterisk-users] CDR Question   jg
• [asterisk-users] Strange problem with Asterisk 1.8.9.3   dotnetdub
• [asterisk-users] h323-sip: one way connection   s m
• [asterisk-users] Installing Asterisk on Virtual Machine   Sandeep Raju
• [asterisk-users] Installing Asterisk on Virtual Machine   Doug Lytle
• [asterisk-users] Installing Asterisk on Virtual Machine   Sandeep Raju
• [asterisk-users] Installing Asterisk on Virtual Machine   Doug Lytle
• [asterisk-users] Installing Asterisk on Virtual Machine   Tzafrir Cohen
• [asterisk-users] Device states   Tommy Cooper
• [asterisk-users] H.264 high profile support   Jeff Morriss
• [asterisk-users] h323-sip: one way connection   Asghar Mohammad
• [asterisk-users] Planned maintenance for community services on April 22, 2013   Asterisk Development Team
• [asterisk-users] Asterisk Tech Job Posting Dallas Texas   JR Richardson
• [asterisk-users] /dev/dahdi/pseudo leaking   Valter Nogueira
• [asterisk-users] Installing Asterisk on Virtual Machine   Sandeep Raju
• [asterisk-users] Installing Asterisk on Virtual Machine   Hans Witvliet
• [asterisk-users] h323-sip: one way connection   s m
• [asterisk-users] Installing Asterisk on Virtual Machine   Sandeep Raju
• [asterisk-users] Installing Asterisk on Virtual Machine   Sandeep Raju
• [asterisk-users] Installing Asterisk on Virtual Machine   Sandeep Raju
• [asterisk-users] Jitter Buffer in asterisk 1.8.11.0   Muhammad Yousuf
• [asterisk-users] Dialplan reload not reloading everything   Brandon Mackie
• [asterisk-users] asterisk music on hold recommendations   Frederic Van Espen
• [asterisk-users] Installing Asterisk on Virtual Machine   Tzafrir Cohen
• [asterisk-users] Installing Asterisk on Virtual Machine   Sandeep Raju
• [asterisk-users] Installing Asterisk on Virtual Machine   Sandeep Raju
• [asterisk-users] asterisk music on hold recommendations   Tzafrir Cohen
• [asterisk-users] asterisk music on hold recommendations   Frederic Van Espen
• [asterisk-users] Installing Asterisk on Virtual Machine   mailinglist at turnopen.com
• [asterisk-users] cdr report   aristidis tsitras
• [asterisk-users] cdr report   Thorsten Göllner
• [asterisk-users] cdr report   aristidis tsitras
• [asterisk-users] cdr report   A J Stiles
• [asterisk-users] cdr report   Ron Wheeler
• [asterisk-users] cdr report   aristidis tsitras
• [asterisk-users] cdr report   Ron Wheeler
• [asterisk-users] h323-sip: one way connection   Asghar Mohammad
• [asterisk-users] cdr report   Stelios Koroneos
• [asterisk-users] Dialplan reload not reloading everything   Rusty Newton
• [asterisk-users] Asterisk 11.4.0-rc1 refuses to use the TURN server   James Mortensen
• [asterisk-users] h323-sip: one way connection   s m
• [asterisk-users] Fundemental changes to CDR within single asterisk family   Steve Davies
• [asterisk-users] Fundemental changes to CDR within single asterisk family   Ishfaq Malik
• [asterisk-users] h323-sip: one way connection   Asghar Mohammad
• [asterisk-users] CDR Question   Nyamul Hassan
• [asterisk-users] Jitter Buffer in asterisk 1.8.11.0   qasimakhan at gmail.com
• [asterisk-users] Asterisk 1.8 and 11   bilal ghayyad
• [asterisk-users] Sip and the media path   David Wessell
• [asterisk-users] Asterisk 1.8 and 11   qasimakhan at gmail.com
• [asterisk-users] Sip and the media path   Kevin Larsen
• [asterisk-users] Asterisk Calendar integration suggestions   john at millican.us
• [asterisk-users] Asterisk Calendar integration suggestions   Steve Totaro
• [asterisk-users] Load Balancing   Olivier
• [asterisk-users] Sip and the media path   David Wessell
• [asterisk-users] Sip and the media path   Kevin Larsen
• [asterisk-users] Load Balancing   acheraime at gmail.com
• [asterisk-users] h323-sip: one way connection   s m
• [asterisk-users] h323-sip: one way connection   Asghar Mohammad
• [asterisk-users] Asterisk Calendar integration suggestions   Hans Witvliet
• [asterisk-users] Users appending # sign when dialing an extension from automated greeting   Vernon Polinkichov
• [asterisk-users] looking for a way to do appointment reminders   Brandon Coale
• [asterisk-users] looking for a way to do appointment reminders   Yves A.
• [asterisk-users] looking for a way to do appointment reminders   jg
• [asterisk-users] h323-sip: one way connection   s m
• [asterisk-users] looking for a way to do appointment reminders   Hans Witvliet
• [asterisk-users] looking for a way to do appointment reminders   jg
• [asterisk-users] looking for a way to do appointment reminders   Chris Bagnall
• [asterisk-users] looking for a way to do appointment reminders   Chris Bagnall
• [asterisk-users] looking for a way to do appointment reminders   jg
• [asterisk-users] looking for a way to do appointment reminders   Chris Bagnall
• [asterisk-users] looking for a way to do appointment reminders   jg
• [asterisk-users] looking for a way to do appointment reminders   Ron Wheeler
• [asterisk-users] h323-sip: one way connection   Asghar Mohammad
• [asterisk-users] caller_id vs cid_number   Nick Khamis
• [asterisk-users] glibc detected crash   Kelly Opal
• [asterisk-users] Radius Based Accounting for Asterisk   basteon
• [asterisk-users] looking for a way to do appointment reminders   David stahl
• [asterisk-users] Sip and the media path   qasimakhan at gmail.com
• [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones   Carlos Alvarez
• [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones   Nathan Anderson
• [asterisk-users] Radius Based Accounting for Asterisk   Nathan Anderson
• [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones   Patrick Lists
• [asterisk-users] Asterisk 11.3.0 - Mask for new file not correct   Ludovic Boué
• [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones   Carlos Alvarez
• [asterisk-users] Asterisk 11.3.0 - Mask for new file not correct   David M. Lee
• [asterisk-users] Gateway?   James Wystead
• [asterisk-users] Gateway?   jg
• [asterisk-users] Asterisk 11.3.0 - Mask for new file not correct   Ludovic Boué
• [asterisk-users] Gateway?   A J Stiles
• [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones   Bob Kyeyune
• [asterisk-users] Gateway?   Eric Wieling
• [asterisk-users] hello!   Rahul Pachauri
• [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000   Danilo Dionisi
• [asterisk-users] Gateway?   James Wystead
• [asterisk-users] Gateway?   Don Kelly
• [asterisk-users] Gateway?   Asghar Mohammad
• [asterisk-users] asterisk 1.4 and SMS module   bilal ghayyad
• [asterisk-users] multiple provider for incoming   Matt Hamilton
• [asterisk-users] multiple provider for incoming   David Wessell
• [asterisk-users] multiple provider for incoming   Warren Selby
• [asterisk-users] multiple provider for incoming   David Wessell
• [asterisk-users] multiple provider for incoming   Matt Hamilton
• [asterisk-users] multiple provider for incoming   Don Kelly
• [asterisk-users] multiple provider for incoming   Matt Hamilton
• [asterisk-users] multiple provider for incoming   Don Kelly
• [asterisk-users] multiple provider for incoming   Carlos Alvarez
• [asterisk-users] multiple provider for incoming   Gregory Malsack

Last message date: Tue Apr 30 23:33:22 CDT 2013
Archived on: Tue Apr 30 23:36:52 CDT 2013

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