January 2016 Archives by thread

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Starting: Mon Jan 4 08:55:40 CST 2016
Ending: Sun Jan 31 13:27:18 CST 2016
Messages: 178

• [asterisk-users] Forwarding call if extension busy   Luca Bertoncello
  • [asterisk-users] Forwarding call if extension busy   Kevin Larsen
    • [asterisk-users] Forwarding call if extension busy   Luca Bertoncello
    • [asterisk-users] Forwarding call if extension busy   Luca Bertoncello
      • [asterisk-users] Forwarding call if extension busy   Kevin Larsen
  • [asterisk-users] Forwarding call if extension busy   jg
• [asterisk-users] Asterisk Behind Firewall   IPN Comm
  • [asterisk-users] Asterisk Behind Firewall   Ron Wheeler
    • [asterisk-users] Asterisk Behind Firewall   IPN Comm
  • [asterisk-users] Asterisk Behind Firewall   Madushan Geethanga
• [asterisk-users] Detected alarm on channel 3: Red Alarm   Vitor Mazuco
  • [asterisk-users] Detected alarm on channel 3: Red Alarm   Ryan, Travis
    • [asterisk-users] Detected alarm on channel 3: Red Alarm   Vitor Mazuco
      • [asterisk-users] Detected alarm on channel 3: Red Alarm   Ryan, Travis
      • [asterisk-users] Detected alarm on channel 3: Red Alarm   Vitor Mazuco
    • [asterisk-users] Detected alarm on channel 3: Red Alarm   Shaun Ruffell
      • [asterisk-users] Detected alarm on channel 3: Red Alarm   Vitor Mazuco
• [asterisk-users] placing calls with linphone.org SIP account   Yves
  • [asterisk-users] placing calls with linphone.org SIP account   Frank
• [asterisk-users] Getting Asterisk to use the SIP Path header   Peter Baines
  • [asterisk-users] Getting Asterisk to use the SIP Path header   Peter Baines
• [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11   Juergen Sauer
  • [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11   Frank
    • [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11   Juergen Sauer
      • [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11   Scott Griepentrog
      • [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11   Juergen Sauer
      • [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11   Frank
• [asterisk-users] No joy with my first AGI Python script   D'Arcy J.M. Cain
  • [asterisk-users] No joy with my first AGI Python script   Steve Edwards
    • [asterisk-users] No joy with my first AGI Python script   D'Arcy J.M. Cain
      • [asterisk-users] No joy with my first AGI Python script   D'Arcy J.M. Cain
• [asterisk-users] Virtual domain redirects   D'Arcy J.M. Cain
  • [asterisk-users] Virtual domain redirects   D'Arcy J.M. Cain
• [asterisk-users] ST2030 replacement   Sil
  • [asterisk-users] ST2030 replacement   Frank
    • [asterisk-users] ST2030 replacement   Markos Vakondios
      • [asterisk-users] ST2030 replacement   Sil
    • [asterisk-users] ST2030 replacement   Sil
  • [asterisk-users] ST2030 replacement   Glenn Geller (VDOPh)
    • [asterisk-users] ST2030 replacement   Sil
• [asterisk-users] The To header was truncated in call... Whats this means?   Vitor Mazuco
  • [asterisk-users] The To header was truncated in call... Whats this means?   Yves
• [asterisk-users] SNMP order of channel types   Tony Mountifield
• [asterisk-users] Call Recording   Ian Harding
  • [asterisk-users] Call Recording   Steve Edwards
    • [asterisk-users] Call Recording   Ian Harding
• [asterisk-users] res_pjsip/pjsip_configuration.c: Unable to create ast_sip_contact_status for contact   Dmitriy Serov
  • [asterisk-users] res_pjsip/pjsip_configuration.c: Unable to create ast_sip_contact_status for contact   George Joseph
• [asterisk-users] cdr_odbc: Error in ExecDirect: -1   Vitor Mazuco
  • [asterisk-users] cdr_odbc: Error in ExecDirect: -1   Patrick Laimbock
• [asterisk-users] "pjsip show endpoints" returns "No Objects Found" in 13.7.0-rc2   Matthew Murphy
  • [asterisk-users] "pjsip show endpoints" returns "No Objects Found" in 13.7.0-rc2   Joshua Colp
    • [asterisk-users] "pjsip show endpoints" returns "No Objects Found" in 13.7.0-rc2   Matthew Murphy
• [asterisk-users] asterisk-users Digest, Vol 138, Issue 8   waqas.mehmood90
  • [asterisk-users] asterisk-users Digest, Vol 138, Issue 8   A J Stiles
• [asterisk-users] PJSIP Returning 421 Extension Required   Trey Hilyard
  • [asterisk-users] PJSIP Returning 421 Extension Required   Matthew Jordan
• [asterisk-users] warble or clicking sound with 11.20.0 with Console/dsp   Jerry Geis
• [asterisk-users] X-RTP-Stat SIP header   Marcelo Terres
  • [asterisk-users] X-RTP-Stat SIP header   Joshua Colp
• [asterisk-users] Digium board considerations   Gary Kuznitz
• [asterisk-users] how to flush user input before READ()   Ethy H. Brito
  • [asterisk-users] how to flush user input before READ()   Ethy H. Brito
    • [asterisk-users] how to flush user input before READ()   Steve Edwards
      • [asterisk-users] how to flush user input before READ()   Ethy H. Brito
      • [asterisk-users] how to flush user input before READ()   Ethy H. Brito
      • [asterisk-users] how to flush user input before READ()   Steve Edwards
      • [asterisk-users] how to flush user input before READ()   D'Arcy J.M. Cain
      • [asterisk-users] how to flush user input before READ()   Steve Edwards
• [asterisk-users] Help me please i am facing much trouble   waqas.mehmood90
  • [asterisk-users] Help me please i am facing much trouble   Steve Edwards
• [asterisk-users] Asterisk 11.21.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 13.7.0 Now Available   Asterisk Development Team
• [asterisk-users] asterisk-users Digest, Vol 138, Issue 10   waqas.mehmood90
• [asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".   Jonathan H
  • [asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".   Joshua Colp
    • [asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".   Jonathan H
• [asterisk-users] How to get PJSIP SIP messages in a log file and not in console ?   Olivier
  • [asterisk-users] How to get PJSIP SIP messages in a log file and not in console ?   Olivier
    • [asterisk-users] How to get PJSIP SIP messages in a log file and not in console ?   Olivier
• [asterisk-users] best practices - ari reconnect   Marek Červenka
• [asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)   Matthew Murphy
  • [asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)   A J Stiles
    • [asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)   Matthew Murphy
• [asterisk-users] Statsd Dialplan Application   Annus Fictus
  • [asterisk-users] Statsd Dialplan Application   Kevin Harwell
    • [asterisk-users] Statsd Dialplan Application   Annus Fictus
• [asterisk-users] PJSIP TLS sometimes RTP, sometimes no RTP   Chirag Desai
  • [asterisk-users] PJSIP TLS sometimes RTP, sometimes no RTP   George Joseph
  • [asterisk-users] PJSIP TLS sometimes RTP, sometimes no RTP   Chirag Desai
• [asterisk-users] 488 Not acceptable here   bilal ghayyad
  • [asterisk-users] 488 Not acceptable here   A J Stiles
    • [asterisk-users] 488 Not acceptable here   bilal ghayyad
• [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome   Alex Villací­s Lasso
  • [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome   Alex Villací­s Lasso
    • [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome   Alex Villací­s Lasso
• [asterisk-users] Queue logfile txt format in mySQL needed   Thomas
  • [asterisk-users] Queue logfile txt format in mySQL needed   Kevin Larsen
    • [asterisk-users] Queue logfile txt format in mySQL needed   Thomas
• [asterisk-users] is there some blocking in 11.21.0   Jerry Geis
  • [asterisk-users] is there some blocking in 11.21.0   A J Stiles
  • [asterisk-users] is there some blocking in 11.21.0   Jerry Geis
    • [asterisk-users] is there some blocking in 11.21.0   Tony Mountifield
    • [asterisk-users] is there some blocking in 11.21.0   Jerry Geis
      • [asterisk-users] is there some blocking in 11.21.0   Tony Mountifield
      • [asterisk-users] is there some blocking in 11.21.0   Jerry Geis
      • [asterisk-users] is there some blocking in 11.21.0   Tony Mountifield
      • [asterisk-users] is there some blocking in 11.21.0   A J Stiles
• [asterisk-users] NAME/USERNAME conflict   kazabe
  • [asterisk-users] NAME/USERNAME conflict   Ethy H. Brito
    • [asterisk-users] NAME/USERNAME conflict   kazabe
      • [asterisk-users] NAME/USERNAME conflict   Ethy H. Brito
• [asterisk-users] Mixing PJSIP realtime and flat files   David Cunningham
  • [asterisk-users] Mixing PJSIP realtime and flat files   Joshua Colp
    • [asterisk-users] Mixing PJSIP realtime and flat files   David Cunningham
• [asterisk-users] set framing on dynamic interface DAHDI   Rafael dos Santos Saraiva
  • [asterisk-users] set framing on dynamic interface DAHDI   Vinicius Fontes
• [asterisk-users] t.38 fax over IAX2?   Benoit Panizzon
  • [asterisk-users] t.38 fax over IAX2?   Doug Lytle
• [asterisk-users] Asterisk 13.7.0 failed to start - PJSIP 2.4.5   Administrator TOOTAI
• [asterisk-users] asterisk-users Digest, Vol 138, Issue 19   waqas.mehmood90
  • [asterisk-users] asterisk-users Digest, Vol 138, Issue 19   Steve Edwards
  • [asterisk-users] asterisk-users Digest, Vol 138, Issue 19   A J Stiles
• [asterisk-users] PJSIP NAT traversal.   Bryant Zimmerman
• [asterisk-users] GSM Gateway behind SIP ATA?   Belal
  • [asterisk-users] GSM Gateway behind SIP ATA?   John Kiniston
• [asterisk-users] PJSIP Stun/ICE   Bryant Zimmerman
  • [asterisk-users] PJSIP Stun/ICE   Joshua Colp
    • [asterisk-users] PJSIP Stun/ICE   Bryant Zimmerman
      • [asterisk-users] PJSIP Stun/ICE   Joshua Colp
      • [asterisk-users] PJSIP Stun/ICE   Bryant Zimmerman
      • [asterisk-users] PJSIP Stun/ICE   Joshua Colp
      • [asterisk-users] PJSIP Stun/ICE   Daniel Heckl
      • [asterisk-users] PJSIP Stun/ICE   Bryant Zimmerman
      • [asterisk-users] PJSIP Stun/ICE   Daniel Heckl
      • [asterisk-users] PJSIP Stun/ICE   James Cloos
      • [asterisk-users] PJSIP Stun/ICE   Joshua Colp
      • [asterisk-users] PJSIP Stun/ICE   Francis Mendoza
      • [asterisk-users] PJSIP Stun/ICE   Bryant Zimmerman
      • [asterisk-users] PJSIP Stun/ICE   Joshua Colp
      • [asterisk-users] PJSIP Stun/ICE   James Cloos
      • [asterisk-users] PJSIP Stun/ICE   George Joseph
      • [asterisk-users] PJSIP Stun/ICE   John Kiniston
      • [asterisk-users] PJSIP Stun/ICE   A J Stiles
      • [asterisk-users] PJSIP Stun/ICE   James Cloos
      • [asterisk-users] PJSIP Stun/ICE   George Joseph
      • [asterisk-users] PJSIP Stun/ICE   Bryant Zimmerman
      • [asterisk-users] PJSIP Stun/ICE   George Joseph
• [asterisk-users] PJSIP - Realtime - Transports module?   Bryant Zimmerman
• [asterisk-users] asterisk 13 mixmonitor - random missing syllables   Marek Červenka
  • [asterisk-users] asterisk 13 mixmonitor - random missing syllables   A J Stiles
    • [asterisk-users] asterisk 13 mixmonitor - random missing syllables   Marek Červenka
      • [asterisk-users] asterisk 13 mixmonitor - random missing syllables   A J Stiles
      • [asterisk-users] asterisk 13 mixmonitor - random missing syllables   Marek Červenka
      • [asterisk-users] asterisk 13 mixmonitor - random missing syllables   Brian ::
      • [asterisk-users] asterisk 13 mixmonitor - random missing syllables   Marek Červenka
      • [asterisk-users] asterisk 13 mixmonitor - random missing syllables   Brian ::
• [asterisk-users] Asterisk 13.7.0 losing database connection   Matthew Murphy
• [asterisk-users] Asterisk 13.7.0 AutoMixMonitor   John Kiniston
• [asterisk-users] Cisco BLF c7975 notifications not working with asterisk realtime   Neeraj Chand
• [asterisk-users] tlsverifyclient=yes option not working   shubham verma
• [asterisk-users] Caller ID Sent in PAI header.   Aziz TestAccount
  • [asterisk-users] Caller ID Sent in PAI header.   Laurent Schweizer
    • [asterisk-users] Caller ID Sent in PAI header.   Aziz TestAccount
• [asterisk-users] Resource List Subscriptions/BLF List and Aastra phones   Olivier
• [asterisk-users] 11.21.0 : echo woes : can't install canceller   sean darcy
  • [asterisk-users] 11.21.0 : echo woes : can't install canceller   sean darcy
• [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API   Sonny Rajagopalan
  • [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API   George Joseph
  • [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API   Bryant Zimmerman
    • [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API   Matthew Jordan
• [asterisk-users] 11.21.0 : echo woes : can't install canceller (sean darcy)   Mc GRATH Ricardo
  • [asterisk-users] 11.21.0 : echo woes : can't install canceller (sean darcy)   sean darcy
  • [asterisk-users] 11.21.0 : echo woes : can't installcanceller (sean darcy)   Dave Platt
• [asterisk-users] PJSIP RTP Timeout - Calls not ending   John Roth
  • [asterisk-users] PJSIP RTP Timeout - Calls not ending   Richard Mudgett
• [asterisk-users] Android native SIP client and 183 (Session Progress) call Declined   Dmitriy Serov
• [asterisk-users] 11.21.0 : echo woes : can't install canceller (sean)   Mc GRATH Ricardo

Last message date: Sun Jan 31 13:27:18 CST 2016
Archived on: Sun Jan 31 13:25:10 CST 2016

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