May 2013 Archives by date

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Starting: Wed May 1 01:09:12 CDT 2013
Ending: Fri May 31 20:32:50 CDT 2013
Messages: 426

• [asterisk-users] Gateway?   jg
• [asterisk-users] asterisk-users Digest, Vol 105, Issue 39   bipin singh
• [asterisk-users] SMS Scenario   bilal ghayyad
• [asterisk-users] SMS Scenario   A J Stiles
• [asterisk-users] multiple provider for incoming   Matt Hamilton
• [asterisk-users] multiple provider for incoming   Eric Wieling
• [asterisk-users] Call "stuck" in queue   Mitch Claborn
• [asterisk-users] Call "stuck" in queue   Mitch Claborn
• [asterisk-users] Queues with different technologies for members, like Khomp Driver   Daniel Varella
• [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000   Dale Noll
• [asterisk-users] debug strategy for one-way audio calls   Marie Fischer
• [asterisk-users] debug strategy for one-way audio calls   Johan Wilfer
• [asterisk-users] looking for a way to do appointment reminders   Lenz Emilitri
• [asterisk-users] Building Asterisk 11.4.0-rc1 with PJSIP 2.1   James Mortensen
• [asterisk-users] Playing a sound file during a call   Carlos Alvarez
• [asterisk-users] Playing a sound file during a call   Kevin Larsen
• [asterisk-users] Playing a sound file during a call   Carlos Alvarez
• [asterisk-users] Playing a sound file during a call   Kevin Larsen
• [asterisk-users] Playing a sound file during a call   Carlos Alvarez
• [asterisk-users] Playing a sound file during a call   Carlos Alvarez
• [asterisk-users] looking for a way to do appointment reminders   Brandon Coale
• [asterisk-users] Playing a sound file during a call   Richard Mudgett
• [asterisk-users] Playing a sound file during a call   Michael L. Young
• [asterisk-users] Playing a sound file during a call   Carlos Alvarez
• [asterisk-users] Building Asterisk 11.4.0-rc1 with PJSIP 2.1   Yves A.
• [asterisk-users] VoIP Incoming Issue   Gopalakrishnan N
• [asterisk-users] debug strategy for one-way audio calls   Gopalakrishnan N
• [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000   Danilo Dionisi
• [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000   Danilo Dionisi
• [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000   Dale Noll
• [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000   Richard Mudgett
• [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000   Danilo Dionisi
• [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000   Dale Noll
• [asterisk-users] changing ringtones to a group of phones   Dr. Michael J. Chudobiak
• [asterisk-users] changing ringtones to a group of phones   jg
• [asterisk-users] changing ringtones to a group of phones   Dr. Michael J. Chudobiak
• [asterisk-users] Digium D70 visual voicemail - won't play   Dr. Michael J. Chudobiak
• [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000   Johann Steinwendtner
• [asterisk-users] Cisco 9971 help   Patrick Lidstone
• [asterisk-users] Cisco 9971 help   Stoyan Marinov
• [asterisk-users] AMI help needed   Pat Collins
• [asterisk-users] debug strategy for one-way audio calls   Olivier
• [asterisk-users] Cisco 9971 help   Patrick Lidstone
• [asterisk-users] My new Polycom 450's can't xfer to 4-digit extension   Mike Diehl
• [asterisk-users] 11.4.-rc1: new segfault in iksemel ??   sean darcy
• [asterisk-users] BLF and asterisk Queue   Alec Davis
• [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)   Sandeep Raju
• [asterisk-users] GotoIf DIALSTATUS - not working   Joseph
• [asterisk-users] Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"   Sandeep Raju
• [asterisk-users] GotoIf DIALSTATUS - not working   Alec Davis
• [asterisk-users] Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"   Alec Davis
• [asterisk-users] Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"   Sandeep Raju
• [asterisk-users] Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"   Sandeep Raju
• [asterisk-users] Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"   Alec Davis
• [asterisk-users] debug strategy for one-way audio calls   Marie Fischer
• [asterisk-users] debug strategy for one-way audio calls   Olivier
• [asterisk-users] My new Polycom 450's can't xfer to 4-digit extension   Doug Lytle
• [asterisk-users] GotoIf DIALSTATUS - not working   Joseph
• [asterisk-users] Testing 911 call   Joseph
• [asterisk-users] Cisco 9971 help   Patrick Lidstone
• [asterisk-users] Cisco 9971 help   Patrick Lidstone
• [asterisk-users] Testing 911 call   Mark Engelhardt
• [asterisk-users] Testing 911 call   Dale Noll
• [asterisk-users] Testing 911 call   James Miller
• [asterisk-users] Joining an astablished call   neo haux
• [asterisk-users] Joining an astablished call   Ian Pilcher
• [asterisk-users] Joining an astablished call   Alec Davis
• [asterisk-users] Load Balancing   Olivier
• [asterisk-users] OT - Differences between Aastra 6730i and 6750i series   Olivier
• [asterisk-users] OT - Question on Aastra 6735i - Was: Sip phone displaying caller name while on call   Olivier
• [asterisk-users] MRCPSynth() change voice   Grant Bagdasarian
• [asterisk-users] Testing 911 call   David Wessell
• [asterisk-users] OT - Differences between Aastra 6730i and 6750i series   Rusty Newton
• [asterisk-users] My new Polycom 450's can't xfer to 4-digit extension   Dave Fullerton
• [asterisk-users] Joining an astablished call   John Novack
• [asterisk-users] Joining an astablished call   Jacob.E.Miles at L-3Com.com
• [asterisk-users] Installing on an OpenVZ instance   James Wystead
• [asterisk-users] Installing on an OpenVZ instance   Johan Wilfer
• [asterisk-users] What is bootstrap.sh for ? Possible bug in 11.3.0 ?   Olivier
• [asterisk-users] What is bootstrap.sh for ? Possible bug in 11.3.0 ?   Matthew Jordan
• [asterisk-users] chan_alsa and confbridge   Chris Gentle
• [asterisk-users] НА: asterisk-users Digest, Vol 105, Issue 40   virus.chel at mail.ru
• [asterisk-users] (no subject)   virus.chel at mail.ru
• [asterisk-users] Get Channel Variables in AMI Event NewExten   Faheem
• [asterisk-users] НА: asterisk-users Digest, Vol 105, Issue 40   Hans Witvliet
• [asterisk-users] What is bootstrap.sh for ? Possible bug in 11.3.0 ?   Olivier
• [asterisk-users] passing '302 moved temporarily' back to the SIP provider   Johann Steinwendtner
• [asterisk-users] chan_alsa and confbridge   Chris Gentle
• [asterisk-users] AMI help needed   Faheem
• [asterisk-users] What is bootstrap.sh for ? Possible bug in 11.3.0 ?   Jason Parker
• [asterisk-users] What is bootstrap.sh for ? Possible bug in 11.3.0 ?   Olivier
• [asterisk-users] Asterisk and hylafax: how to debug ...   Sebastian Niehaus
• [asterisk-users] Asterisk and hylafax: how to debug ...   Sebastian Niehaus
• [asterisk-users] Obtaining high voice quality in dahdi channel   bilal ghayyad
• [asterisk-users] Asterisk and hylafax: how to debug ...   James Cloos
• [asterisk-users] passing '302 moved temporarily' back to the SIP provider   Satish Barot
• [asterisk-users] Obtaining high voice quality in dahdi channel   jg
• [asterisk-users] Confbridge Dynamic video_mode   Rizwan Hisham
• [asterisk-users] hwo to stok variable wiith menu   Salaheddine Elharit
• [asterisk-users] Transfer cmd via AsyncAGI   Dan Cropp
• [asterisk-users] No early media on 302 redirect via two servers   Carlos Alvarez
• [asterisk-users] passing '302 moved temporarily' back to the SIP provider   Carlos Alvarez
• [asterisk-users] passing '302 moved temporarily' back to the SIP provider   Satish Barot
• [asterisk-users] passing '302 moved temporarily' back to the SIP provider   Satish Barot
• [asterisk-users] question about CDR   Salaheddine Elharit
• [asterisk-users] question about CDR   Ishfaq Malik
• [asterisk-users] question about CDR   Salaheddine Elharit
• [asterisk-users] question about CDR   Asghar Mohammad
• [asterisk-users] Asterisk and hylafax: how to debug ...   Karsten Wemheuer
• [asterisk-users] Get Channel Variables in AMI Event NewExten   Dan Cropp
• [asterisk-users] Elastix vs vicidial   Maaz Bin Mahmood
• [asterisk-users] Get Channel Variables in AMI Event NewExten   Matthew Jordan
• [asterisk-users] chanstats console errors   asterisk-02 at jeremykister.com
• [asterisk-users] monitoring Asterisk 1.8   motty cruz
• [asterisk-users] Planned maintenance for community services on May 11, 2013   Asterisk Development Team
• [asterisk-users] monitoring Asterisk 1.8   Carlos Alvarez
• [asterisk-users] monitoring Asterisk 1.8   motty cruz
• [asterisk-users] monitoring Asterisk 1.8   Carlos Rojas
• [asterisk-users] monitoring Asterisk 1.8   Carlos Alvarez
• [asterisk-users] monitoring Asterisk 1.8   Carlos Rojas
• [asterisk-users] monitoring Asterisk 1.8   motty cruz
• [asterisk-users] monitoring Asterisk 1.8   Carlos Alvarez
• [asterisk-users] monitoring Asterisk 1.8   Carlos Rojas
• [asterisk-users] DID providers   Jeff LaCoursiere
• [asterisk-users] monitoring Asterisk 1.8   Bruce Reeves
• [asterisk-users] monitoring Asterisk 1.8   Jai Rangi
• [asterisk-users] Elastix vs vicidial   brahim abidar
• [asterisk-users] qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`   Brian LaVallee
• [asterisk-users] qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`   Jeremy Kister
• [asterisk-users] qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`   Nathan Anderson
• [asterisk-users] Thanks! qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`   Brian LaVallee
• [asterisk-users] question about CDR   Salaheddine Elharit
• [asterisk-users] Get Channel Variables in AMI Event NewExten   Muhammad Faheem
• [asterisk-users] monitoring Asterisk 1.8   Michel Verbraak
• [asterisk-users] monitoring Asterisk 1.8   Luis Morales
• [asterisk-users] Voicemail send to e-mail   Bory's Rouliane Kouassi
• [asterisk-users] Asterisk 12 and OPUS Codec   James Mortensen
• [asterisk-users] Asterisk 12 and OPUS Codec   Paul Belanger
• [asterisk-users] Job Posting   JR Richardson
• [asterisk-users] ISP trunk session ID?   Sergej Petrovsky
• [asterisk-users] ISP trunk session ID?   Asghar Mohammad
• [asterisk-users] Tier 1 Service Providers (AT&T, Level 3)   Nick Khamis
• [asterisk-users] 11.4: no incoming gv/xmpp   sean darcy
• [asterisk-users] ISP trunk session ID?   Nick Khamis
• [asterisk-users] ISP trunk session ID?   Asghar Mohammad
• [asterisk-users] dahdi driver not getting install   Harish Mandowara
• [asterisk-users] dahdi driver not getting install   Asghar Mohammad
• [asterisk-users] ISP trunk session ID?   Sergej Petrovsky
• [asterisk-users] dahdi driver not getting install   Alec Davis
• [asterisk-users] dahdi driver not getting install   Asghar Mohammad
• [asterisk-users] AMI Originate issue   Muhammad Faheem
• [asterisk-users] dahdi driver not getting install   Andrew Colin
• [asterisk-users] dahdi driver not getting install   Asghar Mohammad
• [asterisk-users] Which channels are required for FAX, GTALK and Jaber   bilal ghayyad
• [asterisk-users] AT&T uverse Motorolga nvg510   David Wessell
• [asterisk-users] dahdi driver not getting install   Andrew Colin
• [asterisk-users] HD Voice -- connecting Asterisk into HD Voice compatible mobile phone   Brandon B.
• [asterisk-users] AMI Originate issue   Matthew Jordan
• [asterisk-users] time zone setting in asterisk   Joseph
• [asterisk-users] time zone setting in asterisk   Gregory Malsack
• [asterisk-users] Integrate Astreisk with SIP interface   luke devon
• [asterisk-users] time zone setting in asterisk   Joseph
• [asterisk-users] time zone setting in asterisk   Gregory Malsack
• [asterisk-users] Integrate Astreisk with SIP interface   longst
• [asterisk-users] Integrate Astreisk with SIP interface   Asghar Mohammad
• [asterisk-users] time zone setting in asterisk   Asghar Mohammad
• [asterisk-users] ISP trunk session ID [SOLVED]   Sergej Petrovsky
• [asterisk-users] time zone setting in asterisk   Joseph
• [asterisk-users] time zone setting in asterisk   Asghar Mohammad
• [asterisk-users] Integrate Astreisk with SIP interface   Steve Edwards
• [asterisk-users] Sangoma Wanpipe Driver   Yves A.
• [asterisk-users] Sangoma Wanpipe Driver   Duncan Turnbull
• [asterisk-users] Sangoma Wanpipe Driver   Yves A.
• [asterisk-users] dahdi driver not getting install   Salaheddine Elharit
• [asterisk-users] Sangoma Wanpipe Driver   Asghar Mohammad
• [asterisk-users] Sangoma Wanpipe Driver   Yves A.
• [asterisk-users] dahdi driver not getting install   Patrick Lists
• [asterisk-users] [SOLVED] time zone setting in asterisk   Joseph
• [asterisk-users] amiDebugger - might make your life easier if you program through the AMI   Lenz Emilitri
• [asterisk-users] Tier 1 Service Providers (AT&T, Level 3)   Nick Khamis
• [asterisk-users] Upgrade from 1.0.x to AsteriskNOW 3.0   Andre Goree
• [asterisk-users] amiDebugger - might make your life easier if you program through the AMI   Yves A.
• [asterisk-users] Upgrade from 1.0.x to AsteriskNOW 3.0   Dennis Dryden
• [asterisk-users] Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial   Daniel-Constantin Mierla
• [asterisk-users] Tier 1 Service Providers (AT&T, Level 3)   Nick Khamis
• [asterisk-users] Upgrade from 1.0.x to AsteriskNOW 3.0   Andre Goree
• [asterisk-users] Upgrade from 1.0.x to AsteriskNOW 3.0   Kevin Larsen
• [asterisk-users] Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial   Matthew Jordan
• [asterisk-users] Call Diversion Override   Dominik George
• [asterisk-users] Monitoring SIP trunk status on call by call basis   Ishfaq Malik
• [asterisk-users] dial and bridge   Lenz Emilitri
• [asterisk-users] Monitoring SIP trunk status on call by call basis   Asghar Mohammad
• [asterisk-users] dial and bridge   Mitul Limbani
• [asterisk-users] dial and bridge   Warren Selby
• [asterisk-users] dial and bridge   Mitul Limbani
• [asterisk-users] Monitoring SIP trunk status on call by call basis   Chris Bagnall
• [asterisk-users] chanstats console errors   Jeremy Kister
• [asterisk-users] mfcr2 channel state IDLE 0x00 and call trace log file not ended ??   Leonardo Rivanera
• [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database   Lobna Hegazy
• [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database   Gertjan Baarda
• [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database   Lobna Hegazy
• [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database   Asghar Mohammad
• [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database   Gertjan Baarda
• [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database   Lobna Hegazy
• [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database   Brian LaVallee
• [asterisk-users] Sangoma Wanpipe Driver   Yves A.
• [asterisk-users] dial and bridge   Lenz Emilitri
• [asterisk-users] dial and bridge   Lenz Emilitri
• [asterisk-users] dial and bridge   Mitul Limbani
• [asterisk-users] Initial REGISTER Request: Contains Credentials before 401   Brian LaVallee
• [asterisk-users] dial and bridge   Lenz Emilitri
• [asterisk-users] dial and bridge   Ioan Indreias
• [asterisk-users] dial and bridge   Dan Cropp
• [asterisk-users] dial and bridge   Ioan Indreias
• [asterisk-users] 3. mfcr2 channel state IDLE 0x00 and call trace log file not ended ?? (Leonardo Rivanera)   Mc GRATH Ricardo
• [asterisk-users] How to allow AMI access to Originate yet deny Application: System   Alex Villací­s Lasso
• [asterisk-users] Initial REGISTER Request: Contains Credentials before 401   Matthew J. Roth
• [asterisk-users] Cut offs on outgoing SIP calls   Daniel - Asterisk
• [asterisk-users] Cut offs on outgoing SIP calls   Asghar Mohammad
• [asterisk-users] Cut offs on outgoing SIP calls   Gertjan Baarda
• [asterisk-users] Polycom and forwarding.   Ken D'Ambrosio
• [asterisk-users] Cut offs on outgoing SIP calls   Daniel - Asterisk
• [asterisk-users] Cut offs on outgoing SIP calls   Daniel - Asterisk
• [asterisk-users] Cut offs on outgoing SIP calls   Asghar Mohammad
• [asterisk-users] Polycom and forwarding.   Richard Mudgett
• [asterisk-users] Cut offs on outgoing SIP calls   Daniel - Asterisk
• [asterisk-users] Cut offs on outgoing SIP calls   Asghar Mohammad
• [asterisk-users] Polycom and forwarding.   Carlos Alvarez
• [asterisk-users] SetCallerPres questions   Adam Moffett
• [asterisk-users] Initial REGISTER Request: Contains Credentials before 401: KDDI Japan   Brian LaVallee
• [asterisk-users] Integrate Astreisk with SIP interface   luke devon
• [asterisk-users] Asterisk Web Meetme module not loading   Rohit Mahajan
• [asterisk-users] Call Transfer question   Muhammad Faheem
• [asterisk-users] Call Transfer question   qasimakhan at gmail.com
• [asterisk-users] AstriCon 2013 (our 10th AstriCon) needs YOU!   David Duffett
• [asterisk-users] 11.4: motif can only handle one channel at a time?   sean darcy
• [asterisk-users] 11.4: motif can only handle one channel at a time?   Joshua Colp
• [asterisk-users] 11.4: motif can only handle one channel at a time?   sean darcy
• [asterisk-users] dial and bridge   Lenz Emilitri
• [asterisk-users] Asterisk High-availability/failover solutions   Andre Goree
• [asterisk-users] Asterisk Web Meetme module not loading   Dan Austin
• [asterisk-users] asterisk-users Digest, Vol 106, Issue 23   Nicholas Hart
• [asterisk-users] Planned maintenance for community services on May 16, 2013   Asterisk Development Team
• [asterisk-users] Initial REGISTER Request: Contains Credentials before 401: KDDI Japan   Matthew J. Roth
• [asterisk-users] wanpipe and digium, oslec and hardware echo canceller   bilal ghayyad
• [asterisk-users] Temporarily features (transfer) off during Read   Dmitriy Serov
• [asterisk-users] wanpipe and digium, oslec and hardware echo canceller   Angelo Delphini
• [asterisk-users] Auto dialer scripts and software   cjwstudios
• [asterisk-users] Asterisk 1.8.22.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 11.4.0 Now Available   Asterisk Development Team
• [asterisk-users] Auto dialer scripts and software   Carlos Chavez
• [asterisk-users] wanpipe and digium, oslec and hardware echo canceller   jg
• [asterisk-users] SetCallerPres questions   Maximilian Grobecker
• [asterisk-users] Asterisk 1.8-cert and AGC   Maximilian Grobecker
• [asterisk-users] Performance Asterisk large installation on Vmware/Xen   Rafael dos Santos Saraiva
• [asterisk-users] Asterisk 1.8 vici and the fax, SMS, gtalk, Jaber channels   bilal ghayyad
• [asterisk-users] Performance Asterisk large installation on Vmware/Xen   Mitul Limbani
• [asterisk-users] Performance Asterisk large installation on Vmware/Xen   Chris Bagnall
• [asterisk-users] Performance Asterisk large installation on Vmware/Xen   James Cloos
• [asterisk-users] Performance Asterisk large installation on Vmware/Xen   Stelios Koroneos
• [asterisk-users] Performance Asterisk large installation on Vmware/Xen   Mike
• [asterisk-users] Cut offs on outgoing SIP calls   Philipp von Klitzing
• [asterisk-users] Asterisk 1.8-cert and AGC   Matthew Jordan
• [asterisk-users] Question   CDR
• [asterisk-users] Loopback question   CDR
• [asterisk-users] Secure Calling   Felix Vazquez
• [asterisk-users] Performance Asterisk large installation on Vmware/Xen   Angelo Delphini
• [asterisk-users] Passcode   Felix Vazquez
• [asterisk-users] Question   Joshua Colp
• [asterisk-users] Loopback question   Leandro Dardini
• [asterisk-users] Secure Calling   Leandro Dardini
• [asterisk-users] Passcode   Leandro Dardini
• [asterisk-users] Passcode   sbasurto at soft-gator.com
• [asterisk-users] How to allow AMI access to Originate yet deny Application: System   Alex Villací­s Lasso
• [asterisk-users] Performance Asterisk large installation on Vmware/Xen   Hans Witvliet
• [asterisk-users] Stress testing Asterisk   Tommy Cooper
• [asterisk-users] 11.4: motif can only handle one channel at a time?   sean darcy
• [asterisk-users] 11.4: motif can only handle one channel at a time?   Richard Mudgett
• [asterisk-users] Planned maintenance for community services on May 23, 2013   Asterisk Development Team
• [asterisk-users] Asterisk Log rotate not working   Ahmed Munir
• [asterisk-users] Asterisk Log rotate not working   Chris Bagnall
• [asterisk-users] Asterisk Log rotate not working   Jason Parker
• [asterisk-users] Failed to authenticate device "Ext 110"   asterisk users
• [asterisk-users] Asterisk Log rotate not working   Tzafrir Cohen
• [asterisk-users] Failed to authenticate device "Ext 110"   Matthew J. Roth
• [asterisk-users] Failed to authenticate device "Ext 110"   asterisk users
• [asterisk-users] Asterisk Log rotate not working   Ahmed Munir
• [asterisk-users] Asterisk Log rotate not working   Jim Lucas
• [asterisk-users] Automatic Speech Recognition and Text To Speech using iSpeech   Lefteris Zafiris
• [asterisk-users] Auto dialer scripts and software   A J Stiles
• [asterisk-users] Stress testing Asterisk   Marie Fischer
• [asterisk-users] Auto dialer scripts and software   Chris Bagnall
• [asterisk-users] Auto dialer scripts and software   Don Kelly
• [asterisk-users] Fw: Stress testing Asterisk   Tommy Cooper
• [asterisk-users] Stress testing Asterisk   Mitul Limbani
• [asterisk-users] Changes to the community service maintenance notifications   Asterisk Development Team
• [asterisk-users] Fw: Stress testing Asterisk   Tommy Cooper
• [asterisk-users] Fw: Stress testing Asterisk   Robert-GMAIL
• [asterisk-users] Error 488 Not Acceptable Here   Andrew Colin
• [asterisk-users] Auto dialer scripts and software   Steve Edwards
• [asterisk-users] Fw: Stress testing Asterisk   Paul Belanger
• [asterisk-users] Fw: Stress testing Asterisk   Matias Banchoff
• [asterisk-users] Failed to authenticate device "Ext 110"   Matthew J. Roth
• [asterisk-users] Asterisk Log rotate not working   Ahmed Munir
• [asterisk-users] Stress testing Asterisk   Marie Fischer
• [asterisk-users] Diversion vs. P-Asserted-Id vs. Remote-Party-Id vs. P-Charge-Info vs. From Fields   Positively Optimistic
• [asterisk-users] Asterisk Log rotate not working   Tzafrir Cohen
• [asterisk-users] Asterisk Log rotate not working   Dmitry
• [asterisk-users] Jabber   bilal ghayyad
• [asterisk-users] Integration with skype   bilal ghayyad
• [asterisk-users] Integration with skype   A J Stiles
• [asterisk-users] Jabber   Steven Howes
• [asterisk-users] Asterisk and hylafax: how to debug ...   Sebastian Niehaus
• [asterisk-users] Integration with skype   Marie Fischer
• [asterisk-users] Seeking for TTS engine supporting Hebrew   Andrey Utkin
• [asterisk-users] Integration with skype   Markus
• [asterisk-users] GotoIf function   Gopalakrishnan N
• [asterisk-users] Integration with skype   Richard Kenner
• [asterisk-users] Asterisk on Solaris   Nick Khamis
• [asterisk-users] Error 488 Not Acceptable Here   Maximilian Grobecker
• [asterisk-users] Auto dialer scripts and software   cjwstudios
• [asterisk-users] Auto dialer scripts and software   Ron Wheeler
• [asterisk-users] Integration with skype   Markus
• [asterisk-users] GotoIf function   Gopalakrishnan N
• [asterisk-users] Error 488 Not Acceptable Here   Gopalakrishnan N
• [asterisk-users] Error 488 Not Acceptable Here   Karsten Wemheuer
• [asterisk-users] Registration timed out - for created users   luke devon
• [asterisk-users] Registration timed out - for created users   Asghar Mohammad
• [asterisk-users] Pri-Debug-Log / Is Early Media supported by provider?   Thorsten Göllner
• [asterisk-users] asterisk-gui-2.1.0-rc1   luke devon
• [asterisk-users] asterisk-gui-2.1.0-rc1   aristidis tsitras
• [asterisk-users] Asterisk 11 dtmf not recognised   Gopalakrishnan N
• [asterisk-users] Asterisk 11 dtmf not recognised   Doug Lytle
• [asterisk-users] Asterisk 11 dtmf not recognised   Gopalakrishnan N
• [asterisk-users] asterisk-gui-2.1.0-rc1   asterisk asterisk
• [asterisk-users] Asterisk on Solaris   Nick Khamis
• [asterisk-users] asterisk-gui-2.1.0-rc1   luke devon
• [asterisk-users] Asterisk 11 dtmf not recognised   Gopalakrishnan N
• [asterisk-users] Asterisk 11 dtmf not recognised   Gopalakrishnan N
• [asterisk-users] asterisk-gui-2.1.0-rc1   Alec Davis
• [asterisk-users] Asterisk on Solaris   Doug Lytle
• [asterisk-users] Asterisk on Solaris   Nick Khamis
• [asterisk-users] Asterisk 1.8 wrong Def. Username   Administrator TOOTAI
• [asterisk-users] asterisk-gui-2.1.0-rc1   luke devon
• [asterisk-users] Asterisk 11 dtmf not recognised   Gopalakrishnan N
• [asterisk-users] Integration with skype   me at tdiehl.org
• [asterisk-users] asterisk-gui-2.1.0-rc1   aristidis tsitras
• [asterisk-users] asterisk-gui-2.1.0-rc1   Matthew Jordan
• [asterisk-users] G.729 codec in pass-thru mode   Kamlesh Kumar
• [asterisk-users] Not able to build the chan_sip.c module   upendra
• [asterisk-users] Not able to build the chan_sip.c module   qasimakhan at gmail.com
• [asterisk-users] Not able to build the chan_sip.c module   upendra
• [asterisk-users] Not able to build the chan_sip.c module   Alec Davis
• [asterisk-users] Not able to build the chan_sip.c module   upendra
• [asterisk-users] G.729 codec in pass-thru mode   Eric Wieling
• [asterisk-users] RED on DAHDI channel   Mitch Claborn
• [asterisk-users] ChanIsAvail function is breaking the round robin strategy   stefano scotti
• [asterisk-users] RED on DAHDI channel   Duncan Turnbull
• [asterisk-users] DTMF recognized after call establishment   Gopalakrishnan N
• [asterisk-users] Not able to build the chan_sip.c module   upendra
• [asterisk-users] Not able to build the chan_sip.c module   Mitul Limbani
• [asterisk-users] Not able to build the chan_sip.c module   upendra
• [asterisk-users] Not able to build the chan_sip.c module   Mitul Limbani
• [asterisk-users] Not able to build the chan_sip.c module   Tzafrir Cohen
• [asterisk-users] Not able to build the chan_sip.c module   upendra
• [asterisk-users] Not able to build the chan_sip.c module   Asghar Mohammad
• [asterisk-users] DTMF recognized after call establishment   Asghar Mohammad
• [asterisk-users] G.729 codec in pass-thru mode   Kamlesh Kumar
• [asterisk-users] DTMF recognized after call establishment   Gopalakrishnan N
• [asterisk-users] DTMF recognized after call establishment   Asghar Mohammad
• [asterisk-users] Asterisk 11 dtmf not recognised   asterisk users
• [asterisk-users] RED on DAHDI channel   Shaun Ruffell
• [asterisk-users] Not able to build the chan_sip.c module   Alex Villací­s Lasso
• [asterisk-users] Asterisk - Soundcard - Recording?   Tim Nelson
• [asterisk-users] Asterisk - Soundcard - Recording?   Ron Wheeler
• [asterisk-users] Asterisk - Soundcard - Recording?   Tim Nelson
• [asterisk-users] RED on DAHDI channel   Mitch Claborn
• [asterisk-users] RED on DAHDI channel   Mitch Claborn
• [asterisk-users] Asterisk - Soundcard - Recording?   Ron Wheeler
• [asterisk-users] Asterisk - Soundcard - Recording?   Ron Wheeler
• [asterisk-users] Initial cut off audio   jg
• [asterisk-users] Asterisk - Soundcard - Recording?   Tim Nelson
• [asterisk-users] RED on DAHDI channel   Shaun Ruffell
• [asterisk-users] Asterisk - Soundcard - Recording?   Ron Wheeler
• [asterisk-users] Asterisk - Soundcard - Recording?   Jeff LaCoursiere
• [asterisk-users] Asterisk - Soundcard - Recording?   Jitesh Gala
• [asterisk-users] RED on DAHDI channel   john rodgers
• [asterisk-users] RED on DAHDI channel   Mitch Claborn
• [asterisk-users] G.729 codec in pass-thru mode   Matthew J. Roth
• [asterisk-users] DTMF recognized after call establishment   Gopalakrishnan N
• [asterisk-users] Asterisk - Soundcard - Recording?   A J Stiles
• [asterisk-users] G.729 codec in pass-thru mode   Kamlesh Kumar
• [asterisk-users] Asterisk - Soundcard - Recording?   Tim Nelson
• [asterisk-users] Asterisk - Soundcard - Recording?   Tim Nelson
• [asterisk-users] Asterisk - Soundcard - Recording?   Tim Nelson
• [asterisk-users] G.729 codec in pass-thru mode   Matthew J. Roth
• [asterisk-users] IM through Asterisk   Sergej Petrovsky
• [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.7.0-rc1 Now Available   Asterisk Development Team
• [asterisk-users] Executing a dynamic sequence of applications   Grant Bagdasarian
• [asterisk-users] Executing a dynamic sequence of applications   Yves A.
• [asterisk-users] Executing a dynamic sequence of applications   Matthew Jordan
• [asterisk-users] Queue Periodic Announce not working...   Gopalakrishnan N
• [asterisk-users] Executing a dynamic sequence of applications   Grant Bagdasarian
• [asterisk-users] Queue Periodic Announce not working...   Doug Lytle
• [asterisk-users] Queue Periodic Announce not working...   Gopalakrishnan N
• [asterisk-users] how to launch a URl when dialing a number   Salaheddine Elharit
• [asterisk-users] asterisk-users Digest, Vol 106, Issue 41   Andre Gronwald
• [asterisk-users] how to launch a URl when dialing a number   Justin Killen
• [asterisk-users] WebRTC softphone for Asterisk - any suggestion?   Lenz Emilitri
• [asterisk-users] G.729 codec in pass-thru mode   Kamlesh Kumar
• [asterisk-users] G.729 codec in pass-thru mode   Matthew J. Roth
• [asterisk-users] how to launch a URl when dialing a number   Salaheddine Elharit
• [asterisk-users] how to launch a URl when dialing a number   Alex Villací­s Lasso
• [asterisk-users] how to launch a URl when dialing a number   Salaheddine Elharit
• [asterisk-users] how to launch a URl when dialing a number   A J Stiles
• [asterisk-users] how to launch a URl when dialing a number   Warren Selby
• [asterisk-users] TE410P PCI card in 1U rackount server   Andre Goree
• [asterisk-users] TE410P PCI card in 1U rackount server   jg
• [asterisk-users] TE410P PCI card in 1U rackount server   Andre Goree
• [asterisk-users] WebRTC softphone for Asterisk - any suggestion?   Adnan
• [asterisk-users] Help me understand these log messages   Chris Gentle
• [asterisk-users] Help me understand these log messages   Yves A.
• [asterisk-users] Help me understand these log messages   Alec Davis
• [asterisk-users] Help me understand these log messages   Chris Gentle
• [asterisk-users] How to know the conflict in the dependencies?   bilal ghayyad
• [asterisk-users] How to know the conflict in the dependencies?   Jean-Denis Girard

Last message date: Fri May 31 20:32:50 CDT 2013
Archived on: Fri May 31 20:36:46 CDT 2013

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