June 2014 Archives by date

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Starting: Sun Jun 1 05:28:42 CDT 2014
Ending: Mon Jun 30 15:30:28 CDT 2014
Messages: 259

• [asterisk-users] s4 built in sip client and 481 call/transation does not exist error   Torbjorn Jansson
• [asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card   Scott L. Lykens
• [asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card   jg
• [asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card   Scott L. Lykens
• [asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card   jg
• [asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card   Scott L. Lykens
• [asterisk-users] H323 Transfer   Uni Work
• [asterisk-users] Voicemail Realtime issue "Failed to obtain database object for"   Vadim Lungu
• [asterisk-users] second connected PBX not showing Caller ID   Kevin Larsen
• [asterisk-users] Get last dialed number in a context?   Stefan Gofferje
• [asterisk-users] Get last dialed number in a context?   Israel Gottlieb
• [asterisk-users] Get last dialed number in a context?   Patrick Laimbock
• [asterisk-users] Get last dialed number in a context?   jg
• [asterisk-users] Get last dialed number in a context?   jg
• [asterisk-users] Get last dialed number in a context?   Eric Wieling
• [asterisk-users] Get last dialed number in a context?   jg
• [asterisk-users] Get last dialed number in a context?   Stefan Gofferje
• [asterisk-users] Get last dialed number in a context?   Stefan Gofferje
• [asterisk-users] Get last dialed number in a context?   Stefan Gofferje
• [asterisk-users] Get last dialed number in a context?   Carlos Chavez
• [asterisk-users] Get last dialed number in a context?   Stefan Gofferje
• [asterisk-users] issue installing voicemail imap support: imap_tk module missing   Bart Remmerie
• [asterisk-users] H323 Transfer Problem   Uni Work
• [asterisk-users] Renegotiate SIP audio codec after call is up   Matteo Campana
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   Mojtaba
• [asterisk-users] Renegotiate SIP audio codec after call is up   Eric Wieling
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   John Novack
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   Richard Mudgett
• [asterisk-users] Renegotiate SIP audio codec after call is up   Matteo Campana
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   A J Stiles
• [asterisk-users] Sip Channel Ring Detect   chawki hammoud
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   Mojtaba
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   Mojtaba
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   Shaun Ruffell
• [asterisk-users] Change time between DTMF   CDR
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   A J Stiles
• [asterisk-users] Voicemail Realtime   Vadim Lungu
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   Mojtaba
• [asterisk-users] Change time between DTMF   Mojtaba
• [asterisk-users] Outgoing call queues   [Digital^Dude] ®
• [asterisk-users] Using asterisk as voicemail for cisco call manager   bilal ghayyad
• [asterisk-users] Using macros in extensions.lua?   Dennis Guse
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   A J Stiles
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   Mojtaba
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   John Novack SCII
• [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)   Shaun Ruffell
• [asterisk-users] Problem reload queue dynamical members   Eduardo Leones
• [asterisk-users] Problem reload queue dynamical members   Josh Metzger
• [asterisk-users] Problem reload queue dynamical members   Eduardo Leones
• [asterisk-users] Using macros in extensions.lua?   George Joseph
• [asterisk-users] Shorten time between DTMF   CDR
• [asterisk-users] Shorten time between DTMF   Eric Wieling
• [asterisk-users] asterisk-users Digest, Vol 119, Issue 7   CDR
• [asterisk-users] asterisk-users Digest, Vol 119, Issue 7   Eric Wieling
• [asterisk-users] asterisk-users Digest, Vol 119, Issue 7   Mojtaba
• [asterisk-users] iPhone TLS reg problem: FILE * open failed   Patrick Laimbock
• [asterisk-users] iPhone TLS reg problem: FILE * open failed   Mehroz Ashraf
• [asterisk-users] iPhone TLS reg problem: FILE * open failed   Patrick Laimbock
• [asterisk-users] SIP Softphone   Mark Robinson
• [asterisk-users] SIP Softphone   Carlos Rojas
• [asterisk-users] SIP Softphone   binary
• [asterisk-users] SIP Softphone   Mark Robinson
• [asterisk-users] SIP Softphone   William Hetherington
• [asterisk-users] Inter-Digit delay when dialing out   CDR
• [asterisk-users] SIP Softphone   Giles Coochey
• [asterisk-users] SIP Softphone   Patrick Laimbock
• [asterisk-users] Using macros in extensions.lua?   Dennis Guse
• [asterisk-users] Asterisk realtime peer registration   orteipam at tiscali.it
• [asterisk-users] Asterisk realtime peer registration   Ishfaq Malik
• [asterisk-users] Mixing res_mysql and res_odbc   Ishfaq Malik
• [asterisk-users] R: Asterisk realtime peer registration   orteipam at tiscali.it
• [asterisk-users] R: Mixing res_mysql and res_odbc   orteipam at tiscali.it
• [asterisk-users] R: Mixing res_mysql and res_odbc   Ishfaq Malik
• [asterisk-users] CDR custom variable on second call leg - via originate or .call file   Tiago Geada
• [asterisk-users] CDR custom variable on second call leg - via originate or .call file   Mikael Fredin
• [asterisk-users] CDR custom variable on second call leg - via originate or .call file   Eric Wieling
• [asterisk-users] SIP call control via RTCP   Jan Gaida
• [asterisk-users] SSL/TLS weakness impact on Asterisk authentication   Michelle Dupuis
• [asterisk-users] SSL/TLS weakness impact on Asterisk authentication   Matthew Jordan
• [asterisk-users] Fwd: Asterisk 12.1.1 on Ubuntu 12.04 crashing   Gervasio Marchand Cassataro
• [asterisk-users] SSL/TLS weakness impact on Asterisk authentication   Patrick Laimbock
• [asterisk-users] WSS over Asterisk   Steve Ng
• [asterisk-users] CDR custom variable on second call leg - via originate or .call file   Tiago Geada
• [asterisk-users] issue installing voicemail imap support: imap_tk module missing   Tzafrir Cohen
• [asterisk-users] Hold   Kelly Opal
• [asterisk-users] Hold   jonathan white
• [asterisk-users] Hold   Matthew Jordan
• [asterisk-users] CDR custom variable on second call leg - via originate or .call file   Matthew Jordan
• [asterisk-users] WSS over Asterisk   Matthew Jordan
• [asterisk-users] WSS over Asterisk   William Hetherington
• [asterisk-users] Asterisk 12 AMI Hold Event   David Huebner
• [asterisk-users] WSS over Asterisk   Matthew Jordan
• [asterisk-users] Asterisk 12 AMI Hold Event   Richard Mudgett
• [asterisk-users] Asterisk 12 AMI Hold Event   Eric Wieling
• [asterisk-users] WSS over Asterisk   Miguel Molina
• [asterisk-users] WSS over Asterisk   Steve Ng
• [asterisk-users] WSS over Asterisk   Marco Signorini
• [asterisk-users] issue installing voicemail imap support: imap_tk module missing   Bart Remmerie
• [asterisk-users] issue installing voicemail imap support: imap_tk module missing   Tzafrir Cohen
• [asterisk-users] WSS over Asterisk   Richard Kenner
• [asterisk-users] Asterisk 1.8.15-cert6, 1.8.28.1, 11.6-cert3, 11.10.1, 12.3.1 Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe Framework   Asterisk Security Team
• [asterisk-users] AST-2014-006: Asterisk Manager User Unauthorized Shell Access   Asterisk Security Team
• [asterisk-users] AST-2014-007: Exhaustion of Allowed Concurrent HTTP Connections   Asterisk Security Team
• [asterisk-users] AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions   Asterisk Security Team
• [asterisk-users] dahdi-linux 2.6.2 failing to compile with linux 3.13   David Cunningham
• [asterisk-users] Asterisk 1.8.15-cert7, 1.8.28.2, 11.6-cert4, 11.10.2, 12.3.2 Now Available (Security/Regression Release)   Asterisk Development Team
• [asterisk-users] Need to spoof the callerid using the AMI Originate   Dan Cropp
• [asterisk-users] dahdi-linux 2.6.2 failing to compile with linux 3.13   Shaun Ruffell
• [asterisk-users] pull a call from a queue   Adam Moffett
• [asterisk-users] dahdi-linux 2.6.2 failing to compile with linux 3.13   David Cunningham
• [asterisk-users] pull a call from a queue   Lenz Emilitri
• [asterisk-users] pull a call from a queue   Ioan Indreias
• [asterisk-users] pull a call from a queue   Sid El Mokhtari
• [asterisk-users] Explicit Call Transfer(ECT)   babak
• [asterisk-users] Explicit Call Transfer(ECT)   babak
• [asterisk-users] CDR custom variable on second call leg - via originate or .call file   Tiago Geada
• [asterisk-users] OT - How to configure Apache2 server to receive Polycom log files ?   Olivier
• [asterisk-users] Explicit Call Transfer(ECT)   Richard Mudgett
• [asterisk-users] libpri 1.4.15 Now Available   Asterisk Development Team
• [asterisk-users] libss7 2.0.0 Now Available   Asterisk Development Team
• [asterisk-users] quickstart   thufir
• [asterisk-users] quickstart   Rainer Piper
• [asterisk-users] quickstart   binary dreamer
• [asterisk-users] quickstart   binary dreamer
• [asterisk-users] OT - How to configure Apache2 server to receive Polycom log files ?   Stepan Hradsky
• [asterisk-users] Request for spandsp paid support   Ramesh Hegde
• [asterisk-users] Request for spandsp paid support   Rusty Newton
• [asterisk-users] quickstart   Rusty Newton
• [asterisk-users] quickstart   Thufir
• [asterisk-users] quickstart   Thufir
• [asterisk-users] DTMF transmitting letter A   Markus
• [asterisk-users] DTMF transmitting letter A   Eric Wieling
• [asterisk-users] quickstart   thufir
• [asterisk-users] quickstart   thufir
• [asterisk-users] quickstart   Rainer Piper
• [asterisk-users] PJSIP question   CDR
• [asterisk-users] PJSIP question   Matthew Jordan
• [asterisk-users] France DID num2sip setup   Sean Darcy
• [asterisk-users] Making sense of SDP for debugging of missing audio in SIP trunk   Alex Villací­s Lasso
• [asterisk-users] Asterisk and LDAP   Linus Lüssing
• [asterisk-users] Start/Stop recording in confbridge   Shishir Pokharel
• [asterisk-users] SugarAsterisk vs. ________   thufir
• [asterisk-users] SugarAsterisk vs. ________   thufir
• [asterisk-users] BLF (hints) in Asterisk 12 PJSIP how to?   pepesz
• [asterisk-users] SugarAsterisk vs. ________   A J Stiles
• [asterisk-users] SugarAsterisk vs. ________   thufir
• [asterisk-users] Asterisk and LDAP   Patrick Laimbock
• [asterisk-users] Asterisk and LDAP   Linus Lüssing
• [asterisk-users] Asterisk and LDAP   Patrick Laimbock
• [asterisk-users] BLF (hints) in Asterisk 12 PJSIP how to?   pepesz
• [asterisk-users] BLF (hints) in Asterisk 12 PJSIP how to?   Joshua Colp
• [asterisk-users] Making sense of SDP for debugging of missing audio in SIP trunk   Alex Villací­s Lasso
• [asterisk-users] Making sense of SDP for debugging of missing audio in SIP trunk   Andres
• [asterisk-users] BLF (hints) in Asterisk 12 PJSIP how to?   pepesz
• [asterisk-users] BLF (hints) in Asterisk 12 PJSIP how to?   Joshua Colp
• [asterisk-users] BLF (hints) in Asterisk 12 PJSIP how to?   pepesz
• [asterisk-users] Popup URL for outgoing calls.   Inventions
• [asterisk-users] Asterisk and LDAP   Linus Lüssing
• [asterisk-users] T1 Card RED ALARM   arun kumar
• [asterisk-users] T1 Card RED ALARM   Josh Metzger
• [asterisk-users] T1 Card RED ALARM   arun kumar
• [asterisk-users] Redfone FoneBridge2 Quad T1/E1 Alternative   tirveni yadav
• [asterisk-users] Redfone FoneBridge2 Quad T1/E1 Alternative   Mitul Limbani
• [asterisk-users] T1 Card RED ALARM   Steve Totaro
• [asterisk-users] share mailbox Asterisk 1.8.22   motty cruz
• [asterisk-users] T1 Card RED ALARM   Mc GRATH Ricardo
• [asterisk-users] share mailbox Asterisk 1.8.22   Kevin Larsen
• [asterisk-users] share mailbox Asterisk 1.8.22   motty cruz
• [asterisk-users] share mailbox Asterisk 1.8.22   Kevin Larsen
• [asterisk-users] share mailbox Asterisk 1.8.22   motty cruz
• [asterisk-users] share mailbox Asterisk 1.8.22   Kevin Larsen
• [asterisk-users] T1 Card RED ALARM   arun kumar
• [asterisk-users] T1 Card RED ALARM   arun kumar
• [asterisk-users] Multiple Servers: Multiple Peers: call-limit   Brian LaVallee
• [asterisk-users] T1 Card RED ALARM   Brian LaVallee
• [asterisk-users] Multiple Servers: Multiple Peers: call-limit   Gregory Malsack
• [asterisk-users] OT - How to configure Apache2 server to receive Polycom log files ?   Olivier
• [asterisk-users] Play announcement only once in a Call Queue after 10 seconds   Jonas Kellens
• [asterisk-users] Multiple Servers: Multiple Peers: call-limit   Eric Wieling
• [asterisk-users] OPTIONS Request without username <-> Forbidden   Rafael Visser
• [asterisk-users] RPM updates   Tzafrir Cohen
• [asterisk-users] Echo Cancellation when calling from softphone to mobile.   Anurag Rana
• [asterisk-users] Echo Cancellation when calling from softphone to mobile.   Mitul Limbani
• [asterisk-users] Echo Cancellation when calling from softphone to mobile.   Anurag Rana
• [asterisk-users] Echo Cancellation when calling from softphone to mobile.   Eric Wieling
• [asterisk-users] Asterisk 12 and chan_local   CDR
• [asterisk-users] Asterisk 12 and chan_local   Shaun Ruffell
• [asterisk-users] Play announcement only once in a Call Queue after 10 seconds   Brian LaVallee
• [asterisk-users] PJSIP Dial via IP fails   CDR
• [asterisk-users] PJSIP Dial via IP fails   Joshua Colp
• [asterisk-users] CLID Presentation & Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info   Positively Optimistic
• [asterisk-users] CLID Presentation & Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info   Nyamul Hassan
• [asterisk-users] CLID Presentation & Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info   Eric Wieling
• [asterisk-users] CLID Presentation & Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info   Positively Optimistic
• [asterisk-users] CLID Presentation & Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info   Eric Wieling
• [asterisk-users] CLID Presentation & Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info   Positively Optimistic
• [asterisk-users] Changing recorded file storage directory.   Anurag Rana
• [asterisk-users] Executing an AGI python script in Asterisk after call is bridged.   Anurag Rana
• [asterisk-users] PJSIP question   CDR
• [asterisk-users] PJSIP question   Chad Mothersell
• [asterisk-users] PJSIP Include not working   CDR
• [asterisk-users] Executing an AGI python script in Asterisk after call is bridged.   Rafael Visser
• [asterisk-users] Executing an AGI python script in Asterisk after call is bridged.   Anurag Rana
• [asterisk-users] Executing an AGI python script in Asterisk after call is bridged.   Rafael Visser
• [asterisk-users] Originate with Caller ID Name   Dan Cropp
• [asterisk-users] CLID Presentation & Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info   Brian LaVallee
• [asterisk-users] Changing recorded file storage directory.   Ishfaq Malik
• [asterisk-users] Early media recognition   David Pinedo
• [asterisk-users] Attack on Sip server.   Anurag Rana
• [asterisk-users] Attack on Sip server.   arun kumar
• [asterisk-users] Attack on Sip server.   Prakash N
• [asterisk-users] Attack on Sip server.   Anurag Rana
• [asterisk-users] Attack on Sip server.   Anurag Rana
• [asterisk-users] Attack on Sip server.   Steven Howes
• [asterisk-users] Attack on Sip server.   Eric Wieling
• [asterisk-users] Attack on Sip server.   Markus Weiler
• [asterisk-users] Attack on Sip server.   Michelle Dupuis
• [asterisk-users] AGI script VERBOSE cmd   Bryant Zimmerman
• [asterisk-users] Attack on Sip server.   Ron Wheeler
• [asterisk-users] CLID Presentation & Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info   Positively Optimistic
• [asterisk-users] AGI script VERBOSE cmd   Rafael Visser
• [asterisk-users] AGI script VERBOSE cmd   Bryant Zimmerman
• [asterisk-users] Attack on Sip server.   Prakash N
• [asterisk-users] Originate with Caller ID Name   Dan Cropp
• [asterisk-users] Attack on Sip server.   andrew Colin
• [asterisk-users] Attack on Sip server.   Jai Rangi
• [asterisk-users] Attack on Sip server.   Mitul Limbani
• [asterisk-users] Attack on Sip server.   Prakash N
• [asterisk-users] Attack on Sip server.   Anurag Rana
• [asterisk-users] Attack on Sip server.   Anurag Rana
• [asterisk-users] Attack on Sip server.   Mitul Limbani
• [asterisk-users] Attack on Sip server.   Anurag Rana
• [asterisk-users] Attack on Sip server.   Steve Edwards
• [asterisk-users] CDR custom variable on second call leg - via originate or .call file   Tiago Geada
• [asterisk-users] PJSIP Include not working   Rusty Newton
• [asterisk-users] Changing recorded file storage directory.   Rusty Newton
• [asterisk-users] OPTIONS Request without username <-> Forbidden   Rusty Newton
• [asterisk-users] Popup URL for outgoing calls.   Rusty Newton
• [asterisk-users] How to execute an AGI script for each call.   Anurag Rana
• [asterisk-users] CDR custom variable on second call leg - via originate or .call file   Richard Mudgett
• [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.9.2-rc1 Now Available   Asterisk Development Team
• [asterisk-users] Popup URL for outgoing calls.   Ikka Vertika Tirtawidjaja
• [asterisk-users] Popup URL for outgoing calls.   Prakash N
• [asterisk-users] Popup URL for outgoing calls.   Carlos Rojas
• [asterisk-users] PJSIP endpoint max-calls limit missing   CDR
• [asterisk-users] PJSIP endpoint max-calls limit missing   George Joseph
• [asterisk-users] 200 OK however still rinnging   Nick Cameo
• [asterisk-users] 200 OK however still rinnging   Nick Cameo
• [asterisk-users] Passing parameters to voiceglue.conf   Raj Roy Ghandhi
• [asterisk-users] Attack on Sip server.   Andres
• [asterisk-users] Best approach in asterisk configuration   sylvain GOTRI
• [asterisk-users] Asterisk 11.10.2 update breaks snom TLS transport   Heiko Wundram
• [asterisk-users] Best approach in asterisk configuration   A J Stiles
• [asterisk-users] Sippeers realtime with minimum table   Rafael dos Santos Saraiva
• [asterisk-users] Fwd: Regarding packet2packet bridging   Sameer Rathod
• [asterisk-users] Asterisk 11.10.2 update breaks snom TLS transport   Heiko Wundram
• [asterisk-users] recording in mp3   andrew Colin
• [asterisk-users] recording in mp3   Scott Griepentrog

Last message date: Mon Jun 30 15:30:28 CDT 2014
Archived on: Mon Jun 30 15:30:29 CDT 2014

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