June 2010 Archives by author

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Starting: Tue Jun 1 00:30:23 CDT 2010
Ending: Wed Jun 30 23:54:38 CDT 2010
Messages: 1385

• [asterisk-users] How to pass variable back and forth from dialplan to php file?   William Stillwell (Lists)
• [asterisk-users] Dialplan for conference   William Stillwell (Lists)
• [asterisk-users] SIP Delay with remote stations?   William Stillwell (Lists)
• [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?   William Stillwell (Lists)
• [asterisk-users] PSTN call hunting   Gopalakrishnan A.N
• [asterisk-users] Press twice *   Alyed
• [asterisk-users] UDPTL T38 via NAT   Miguel Amez
• [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT   Miguel Amez
• [asterisk-users] Silence suppression and internal timing   Benny Amorsen
• [asterisk-users] Using Local in queues a good idea? (or at least not a very bad idea?)   Benny Amorsen
• [asterisk-users] Time variables in system application   Roderick A. Anderson
• [asterisk-users] OT: Free DID/SIP accounts   Roderick A. Anderson
• [asterisk-users] OT: Free DID/SIP accounts   Roderick A. Anderson
• [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?   Roderick A. Anderson
• [asterisk-users] Voicemail ODBC   Andraž
• [asterisk-users] Asterisk ODBC   Andraž
• [asterisk-users] Voicemail ODBC   Andraž
• [asterisk-users] Voicemail ODBC   Andraž
• [asterisk-users] Voicemail ODBC   Andraž
• [asterisk-users] Voicemail ODBC   Andraž
• [asterisk-users] Voicemail ODBC   Andraž
• [asterisk-users] Problem with GROUP()   Andres
• [asterisk-users] Problem with GROUP()   Andres
• [asterisk-users] Why asterisk down when inet server down?   Andres
• [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)   Andres
• [asterisk-users] G729 license key registration   Kiss András
• [asterisk-users] G729 license key registration   Kiss András
• [asterisk-users] CDR in case of CallForwarding   srinivas Antarvedi
• [asterisk-users] Qwest PRIs   Voip Asterisk
• [asterisk-users] Qwest PRIs   Voip Asterisk
• [asterisk-users] Qwest PRIs   Voip Asterisk
• [asterisk-users] Qwest PRIs   Voip Asterisk
• [asterisk-users] Qwest PRIs   Voip Asterisk
• [asterisk-users] Qwest PRIs   Voip Asterisk
• [asterisk-users] Qwest PRIs   Voip Asterisk
• [asterisk-users] Delay in IVR   David Backeberg
• [asterisk-users] Meetmee user introduction disabled   David Backeberg
• [asterisk-users] problem with inserting records into cdr   David Backeberg
• [asterisk-users] AGI library for C/C++   David Backeberg
• [asterisk-users] ring no answer / RONA versus HangUp   David Backeberg
• [asterisk-users] ring no answer / RONA versus HangUp   David Backeberg
• [asterisk-users] when to use e1/t1 card?   David Backeberg
• [asterisk-users] Big time system   David Backeberg
• [asterisk-users] Big time system   David Backeberg
• [asterisk-users] Where can I find a minimal set of empty configuration files (SIP only)?   David Backeberg
• [asterisk-users] Small PC to build and run Asterisk   Chris Bagnall
• [asterisk-users] Small PC to build and run Asterisk   Chris Bagnall
• [asterisk-users] Blind transfer feature   Chris Bagnall
• [asterisk-users] OT: Bandwidth calculations   Chris Bagnall
• [asterisk-users] Polycom firmware: split vs. combined   John Balogh
• [asterisk-users] ZA16E and FXO-200 modules with asterisk   Szalai Balázs
• [asterisk-users] Why asterisk down when inet server down?   Theo Band
• [asterisk-users] Asterisk distribution for a Call Center   Gerardo Barajas
• [asterisk-users] About Sangoma cards and Asterisk integration with other PBX   Daniel Bareiro
• [asterisk-users] About Sangoma cards and Asterisk integration with other PBX   Daniel Bareiro
• [asterisk-users] no sound between extensions   Gary Baribault
• [asterisk-users] no sound between extensions   Gary Baribault
• [asterisk-users] no sound between extensions   Gary Baribault
• [asterisk-users] no sound between extensions   Gary Baribault
• [asterisk-users] no sound between extensions   Gary Baribault
• [asterisk-users] no sound between extensions   Gary Baribault
• [asterisk-users] no sound between extensions   Gary Baribault
• [asterisk-users] Muti Asterisk   Paul Belanger
• [asterisk-users] Create Conference and exit myself   Paul Belanger
• [asterisk-users] joining 2 conferences together   Paul Belanger
• [asterisk-users] joining 2 conferences together   Paul Belanger
• [asterisk-users] help with sip 401 unauthorized   Paul Belanger
• [asterisk-users] 50 mantis issues marked 'Ready for Testing'   Paul Belanger
• [asterisk-users] Dialplan for conference   Paul Belanger
• [asterisk-users] Is there a default dial plan that is not in extention.conf?   Paul Belanger
• [asterisk-users] Detecting hook flash in asterisk   Paul Belanger
• [asterisk-users] Up-to-date list of Asterisk appliances?   Paul Belanger
• [asterisk-users] unwanted entries created in dialplan from users.conf, how can I get rid of it?   Paul Belanger
• [asterisk-users] [voice mail] Estimating file size?   Paul Belanger
• [asterisk-users] Problem attended transfer with ilbc   Paul Belanger
• [asterisk-users] Pickup a ringing Queue member   Paul Belanger
• [asterisk-users] sip server   Paul Belanger
• [asterisk-users] How to Add IP address to SIP Domain   Paul Belanger
• [asterisk-users] What‘s the best operating system suggest for Asterisk 1.6.2.9   Paul Belanger
• [asterisk-users] Voiceprompts i.e. voicemail and conferencing in multiple codecs   Paul Belanger
• [asterisk-users] SIP Delay with remote stations?   Paul Belanger
• [asterisk-users] Strange Asterisk/SIP call forwarding behavior   Paul Belanger
• [asterisk-users] Friday at 1PM: SIPVicious has a new tool: svcrash   Niccolò Belli
• [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?   Niccolò Belli
• [asterisk-users] 87.230.80.186 - Trying to register   Dovid Bender
• [asterisk-users] RE How to break pri DID to multiple SIP Trunks   Dovid Bender
• [asterisk-users] Dial options not working   Dovid Bender
• [asterisk-users] Can't call my extension   Dovid Bender
• [asterisk-users] How to stop intruder from registering sip?   Benoit
• [asterisk-users] Suddenly "HDLC Bad FCS (8)" errors on ISDN-PRI, changed nothing   Gareth Blades
• [asterisk-users] Getting ANI on UK BT ISDN - Is SS& required?   Gareth Blades
• [asterisk-users] Is this failed Asterisk setup typical?   Gareth Blades
• [asterisk-users] HDLC Bad FCS (8) on Primary D-channel   Gareth Blades
• [asterisk-users] Asterisk reject SIP INTITE from different source ports   Gareth Blades
• [asterisk-users] Asterisk reject SIP INTITE from different source ports   Gareth Blades
• [asterisk-users] DAHDI PRI error message   Gareth Blades
• [asterisk-users] DAHDI PRI error message   Gareth Blades
• [asterisk-users] OT: Bandwidth calculations   Gareth Blades
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gareth Blades
• [asterisk-users] Handling DTMF for number 4   Gareth Blades
• [asterisk-users] Handling DTMF for number 4   Gareth Blades
• [asterisk-users] What‘s the best operating system suggest for Asterisk 1.6.2.9   Gareth Blades
• [asterisk-users] Is Centos 64 bit or 32 bit better?   Gareth Blades
• [asterisk-users] peer IP address in CDR   Gareth Blades
• [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?   Gareth Blades
• [asterisk-users] peer IP address in CDR   Gareth Blades
• [asterisk-users] Find a way to block brute force attacks.   Gareth Blades
• [asterisk-users] Echo problem in VoIP-calls   Gareth Blades
• [asterisk-users] Echo problem in VoIP-calls   Gareth Blades
• [asterisk-users] Echo problem in VoIP-calls   Gareth Blades
• [asterisk-users] Adding Congestion to CDR logs   Gareth Blades
• [asterisk-users] Adding Congestion to CDR logs   Gareth Blades
• [asterisk-users] Echo problem in VoIP-calls   Gareth Blades
• [asterisk-users] Echo problem in VoIP-calls   Gareth Blades
• [asterisk-users] Echo problem in VoIP-calls   Gareth Blades
• [asterisk-users] Echo problem in VoIP-calls   Gareth Blades
• [asterisk-users] Asterisk SIP/IAX peers can't connect after Firewall change?   Chris Brentano
• [asterisk-users] Asterisk SIP/IAX peers can't connect after Firewall change?   Chris Brentano
• [asterisk-users] Asterisk SIP/IAX peers can't connect after Firewall change?   Chris Brentano
• [asterisk-users] UDPTL T38 via NAT   Remco Bressers
• [asterisk-users] UDPTL T38 via NAT   Remco Bressers
• [asterisk-users] UDPTL T38 via NAT   Remco Bressers
• [asterisk-users] UDPTL T38 via NAT   Remco Bressers
• [asterisk-users] G729 license key registration   Remco Bressers
• [asterisk-users] peer IP address in CDR   Remco Bressers
• [asterisk-users] peer IP address in CDR   Remco Bressers
• [asterisk-users] peer IP address in CDR   Remco Bressers
• [asterisk-users] peer IP address in CDR   Remco Bressers
• [asterisk-users] Running SIP on non-standard ports   Stephen Brown
• [asterisk-users] Definite app_jack trouble - unsolvable   Russell Bryant
• [asterisk-users] sip server   C.Savinovich
• [asterisk-users] Limit total length of calls to a specifig SIP peer   Laurent CARON
• [asterisk-users] Limit total length of calls to a specifig SIP peer   Laurent CARON
• [asterisk-users] Limit total length of calls to a specifig SIP peer   Laurent CARON
• [asterisk-users] How do I access the Dialstatus numeric code received?   CDR
• [asterisk-users] How do I access the Dialstatus numeric code received?   CDR
• [asterisk-users] Ring + Music on Hold in the same call   Matteo Campana
• [asterisk-users] Ring + Music on Hold in the same call   Matteo Campana
• [asterisk-users] Anyone can share their config file for Cisco phone please?   Emanuele Carbone
• [asterisk-users] dahdi modules installed wrong location   Steve Casto
• [asterisk-users] dahdi modules installed wrong location   Steve Casto
• [asterisk-users] Reloading queue members (realtime DB)   Aksel Celasun
• [asterisk-users] Call queues - issues, can't make it work.   Aksel Celasun
• [asterisk-users] Call queues - issues, can't make it work.   Aksel Celasun
• [asterisk-users] error message in CLI regarding SET Timeout   Aksel Celasun
• [asterisk-users] Automatic attendant - Error in CLI.   Aksel Celasun
• [asterisk-users] Automatic attendant - Error in CLI.   Aksel Celasun
• [asterisk-users] Automatic attendant - Error in CLI.   Aksel Celasun
• [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'   Aksel Celasun
• [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'   Aksel Celasun
• [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'   Aksel Celasun
• [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'   Aksel Celasun
• [asterisk-users] features.conf - parkedcalls - transfer   Aksel Celasun
• [asterisk-users] features.conf - parkedcalls - transfer   Aksel Celasun
• [asterisk-users] features.conf - parkedcalls - transfer   Aksel Celasun
• [asterisk-users] Unregister and register SIP phones by using num pad on phones?   Aksel Celasun
• [asterisk-users] Asterisk 1.6 and multiple parking   Aksel Celasun
• [asterisk-users] realtime queues "membername" problem   Jean Chassoul
• [asterisk-users] realtime queues "membername" problem   Jean Chassoul
• [asterisk-users] realtime queues "membername" problem   Jean Chassoul
• [asterisk-users] memory leak   Julien Chavanton
• [asterisk-users] 1.6 issues   Carlos Chavez
• [asterisk-users] Eyebeam hangs when you dial an unavailable number   Carlos Chavez
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   Carlos Chavez
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   Carlos Chavez
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   Carlos Chavez
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   Carlos Chavez
• [asterisk-users] Problem with TE411P and DAHDI   Carlos Chavez
• [asterisk-users] What‘s the best operating system suggest for Asterisk 1.6.2.9   Carlos Chavez
• [asterisk-users] Dial with MOH   Khaled W. Chehab
• [asterisk-users] Assign dhadi channel to several groups   Adolphe Cher-aime
• [asterisk-users] Assign dhadi channel to several groups   Adolphe Cher-aime
• [asterisk-users] Minimum modules required to run VoIP and CDR   Frank Church
• [asterisk-users] Minimum modules required to run VoIP and CDR   Frank Church
• [asterisk-users] Asterisk and gtalk part 2   Julien Claassen
• [asterisk-users] Asterisk and gtalk part 2   Julien Claassen
• [asterisk-users] Voicemail bug(?) with Asterisk 1.6.2.8-rc1   Julien Claassen
• [asterisk-users] Definite app_jack trouble - unsolvable   Julien Claassen
• [asterisk-users] Voicemail bug(?) with Asterisk 1.6.2.8-rc1   Julien Claassen
• [asterisk-users] Voicemail bug(?) with Asterisk 1.6.2.8-rc1   Julien Claassen
• [asterisk-users] Definite app_jack trouble - unsolvable   Julien Claassen
• [asterisk-users] Definite app_jack trouble - unsolvable   Julien Claassen
• [asterisk-users] Persuing the gtalk issue - not only jack-related   Julien Claassen
• [asterisk-users] Persuing the gtalk issue - not only jack-related   Julien Claassen
• [asterisk-users] Persuing the gtalk issue - not only jack-related   Julien Claassen
• [asterisk-users] Persuing the gtalk issue - not only jack-related   Julien Claassen
• [asterisk-users] Persuing the gtalk issue - not only jack-related   Julien Claassen
• [asterisk-users] originating a sip call from the CLI   Julien Claassen
• [asterisk-users] originating a sip call from the CLI   Julien Claassen
• [asterisk-users] Still sipping frustration - only getting state ACK   Julien Claassen
• [asterisk-users] Still sipping frustration - only getting state ACK   Julien Claassen
• [asterisk-users] Still sipping frustration - only getting state ACK   Julien Claassen
• [asterisk-users] Still sipping frustration - only getting state ACK   Julien Claassen
• [asterisk-users] Still sipping frustration - only getting state ACK   Julien Claassen
• [asterisk-users] Still sipping frustration - only getting state ACK   Julien Claassen
• [asterisk-users] Still sipping frustration - only getting state ACK   Julien Claassen
• [asterisk-users] Still sipping frustration - only getting state ACK   Julien Claassen
• [asterisk-users] Still no(isy) app_jack in the box   Julien Claassen
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   Seann Clark
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   Seann Clark
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   Seann Clark
• [asterisk-users] Asterisk SIP realtime and realtime DB tools   Seann Clark
• [asterisk-users] Local channel usage   Harel Cohen
• [asterisk-users] Local channel usage   Harel Cohen
• [asterisk-users] DAHDI volume   Tzafrir Cohen
• [asterisk-users] Assign dhadi channel to several groups   Tzafrir Cohen
• [asterisk-users] How to play Floating point numbers?   Tzafrir Cohen
• [asterisk-users] understand which asterisk thread is consuming CPU   Tzafrir Cohen
• [asterisk-users] tuning software echo cancellation   Tzafrir Cohen
• [asterisk-users] ISDN -> SIP   Tzafrir Cohen
• [asterisk-users] chan_dahdi compilation with embedded   Tzafrir Cohen
• [asterisk-users] MeetMe   Tzafrir Cohen
• [asterisk-users] MeetMe   Tzafrir Cohen
• [asterisk-users] How to stop intruder from registering sip?   Tzafrir Cohen
• [asterisk-users] How to stop intruder from registering sip?   Tzafrir Cohen
• [asterisk-users] Problem with dahdi and with freepbx   Tzafrir Cohen
• [asterisk-users] Small PC to build and run Asterisk   Tzafrir Cohen
• [asterisk-users] Asterisk + Dahdi does not work with BRI NT mode   Tzafrir Cohen
• [asterisk-users] Check if variable contains strings   Tzafrir Cohen
• [asterisk-users] Automatic attendant - Error in CLI.   Tzafrir Cohen
• [asterisk-users] dahdi modules installed wrong location   Tzafrir Cohen
• [asterisk-users] Using SetVar with System() is it possible?   Tzafrir Cohen
• [asterisk-users] asterisk appache issue   Tzafrir Cohen
• [asterisk-users] Compiling H323   Tzafrir Cohen
• [asterisk-users] when to use e1/t1 card?   Tzafrir Cohen
• [asterisk-users] DAHDI: Inbound BRI call, DDI not presented   Tzafrir Cohen
• [asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver   Tzafrir Cohen
• [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?   Tzafrir Cohen
• [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?   Tzafrir Cohen
• [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?   Tzafrir Cohen
• [asterisk-users] PRI span problem - no D channel   Tzafrir Cohen
• [asterisk-users] Asterisk + E1 card   Tzafrir Cohen
• [asterisk-users] "Hidden" memory leak   Tzafrir Cohen
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Tzafrir Cohen
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Tzafrir Cohen
• [asterisk-users] Is there a default dial plan that is not in extention.conf?   Tzafrir Cohen
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Tzafrir Cohen
• [asterisk-users] Is there a default dial plan that is not in extention.conf?   Tzafrir Cohen
• [asterisk-users] [voice mail] Estimating file size?   Tzafrir Cohen
• [asterisk-users] Use one group for ISN truncs   Tzafrir Cohen
• [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?   Tzafrir Cohen
• [asterisk-users] calls dropped after 20 seconds in a non NAT situation   Bart Coninckx
• [asterisk-users] Voicemail : mail attachment to multiple mail-addresses   Rob Coward
• [asterisk-users] Unable to pickup an extension, tryi   Rob Coward
• [asterisk-users] Unable to pickup an extension, tryi   Rob Coward
• [asterisk-users] realtime queues "membername" problem   Rob Coward
• [asterisk-users] ISDN -> SIP   Gergo Csibra
• [asterisk-users] Asterisk no audio on calls problem.   Albert Culleton
• [asterisk-users] Asterisk no audio on calls problem.   Albert Culleton
• [asterisk-users] Is this failed Asterisk setup typical?   CunningPike
• [asterisk-users] Update the LCD with the callee's name after dialing   CunningPike
• [asterisk-users] Polycom firmware: split vs. combined   Ken D'Ambrosio
• [asterisk-users] Generate cdr on Hangup   Deepesh D
• [asterisk-users] Dialplan for conference   Deepesh D
• [asterisk-users] Detecting hook flash in asterisk   Deepesh D
• [asterisk-users] OT: Windows TAPI command-line driver   Klaus Darilion
• [asterisk-users] OT: Windows TAPI command-line driver   Klaus Darilion
• [asterisk-users] OT - How to query vcard-like data for CTI app   Klaus Darilion
• [asterisk-users] Friday at 1PM: SIPVicious has a new tool: svcrash   Klaus Darilion
• [asterisk-users] Update the LCD with the callee's name after dialing   Matt Darnell
• [asterisk-users] Update the LCD with the callee's name after dialing   Matt Darnell
• [asterisk-users] Codec negotiation   Steve Davies
• [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel   Alec Davis
• [asterisk-users] how to run deadagi script after "status: expired"   Necati Demir
• [asterisk-users] how to get call duration   Necati Demir
• [asterisk-users] run script after completed   Necati Demir
• [asterisk-users] how to get call duration   Necati Demir
• [asterisk-users] when to use e1/t1 card?   Necati Demir
• [asterisk-users] when to use e1/t1 card?   Necati Demir
• [asterisk-users] when to use e1/t1 card?   Necati Demir
• [asterisk-users] Asterisk 1.6 (and 1.4) DTMF problems using RFC2833   Mark Deneen
• [asterisk-users] Long shot... Order Logix   Matt Desbiens
• [asterisk-users] Asterisk on Ubuntu   Danny Dias
• [asterisk-users] Asterisk + E1 card   Juan David Diaz
• [asterisk-users] SIP message problems - retransmit and lost messages   Jim Dickenson
• [asterisk-users] get Asterisk version from within dialplan   Jim Dickenson
• [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'   Jim Dickenson
• [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'   Jim Dickenson
• [asterisk-users] queue command in asterisk 1.4 with macro-argument   Jim Dickenson
• [asterisk-users] queue command in asterisk 1.4 with macro-argument   Jim Dickenson
• [asterisk-users] Want to retrieve the value of contact header   Jim Dickenson
• [asterisk-users] ISDN -> SIP   Stefan Dreyer
• [asterisk-users] ISDN -> SIP   Stefan Dreyer
• [asterisk-users] IAXmodem in dialplan   Michelle Dupuis
• [asterisk-users] SIP Witch   Michelle Dupuis
• [asterisk-users] Dual Atom mobo - call capacity   Michelle Dupuis
• [asterisk-users] Compiling H323   Michelle Dupuis
• [asterisk-users] Compiling H323   Michelle Dupuis
• [asterisk-users] Compiling H323   Michelle Dupuis
• [asterisk-users] Update to chan_ooh323 wrapper   Michelle Dupuis
• [asterisk-users] how to get call duration   Steve Edwards
• [asterisk-users] Lookup ${EXTEN} in database, update context/route if found... AGI?   Steve Edwards
• [asterisk-users] how to get call duration   Steve Edwards
• [asterisk-users] <UsingWaitorPlaybackinhextension at gmail.com>,   Steve Edwards
• [asterisk-users] Lookup ${EXTEN} in database, update context/route if found... AGI?   Steve Edwards
• [asterisk-users] Press twice *   Steve Edwards
• [asterisk-users] Controlling calls   Steve Edwards
• [asterisk-users] Re : Controlling calls   Steve Edwards
• [asterisk-users] Controlling calls   Steve Edwards
• [asterisk-users] Re : Controlling calls   Steve Edwards
• [asterisk-users] Controlling calls   Steve Edwards
• [asterisk-users] Re : Controlling calls   Steve Edwards
• [asterisk-users] Re : Re : Controlling calls   Steve Edwards
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   Steve Edwards
• [asterisk-users] Re : Re : Re : Controlling calls   Steve Edwards
• [asterisk-users] Limit total length of calls to a specifig SIP peer   Steve Edwards
• [asterisk-users] own Caller ID   Steve Edwards
• [asterisk-users] PSTN-IVR call   Steve Edwards
• [asterisk-users] Out of Office   Steve Edwards
• [asterisk-users] Dual Atom mobo - call capacity   Steve Edwards
• [asterisk-users] WARNING message when play   Steve Edwards
• [asterisk-users] contacting   Steve Edwards
• [asterisk-users] OT: Free DID/SIP accounts   Steve Edwards
• [asterisk-users] How to stop intruder from registering sip?   Steve Edwards
• [asterisk-users] How to stop intruder from registering sip?   Steve Edwards
• [asterisk-users] contacting   Steve Edwards
• [asterisk-users] PSTN call hunting   Steve Edwards
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   Steve Edwards
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   Steve Edwards
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   Steve Edwards
• [asterisk-users] Qwest PRIs   Steve Edwards
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   Steve Edwards
• [asterisk-users] can't seem to register, status unmonitored   Steve Edwards
• [asterisk-users] Asterisk hangs up for some calls   Steve Edwards
• [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'   Steve Edwards
• [asterisk-users] numbers   Steve Edwards
• [asterisk-users] Fwd: can't seem to register, status unmonitored   Steve Edwards
• [asterisk-users] read data from file system and put in a variable   Steve Edwards
• [asterisk-users] a2billing for residential voip usage   Steve Edwards
• [asterisk-users] calling machine over sip   Steve Edwards
• [asterisk-users] asterisk issue   Steve Edwards
• [asterisk-users] asterisk issue   Steve Edwards
• [asterisk-users] Why asterisk down when inet server down?   Steve Edwards
• [asterisk-users] Why asterisk down when inet server down?   Steve Edwards
• [asterisk-users] asterisk appache issue   Steve Edwards
• [asterisk-users] Using SetVar with System() is it possible?   Steve Edwards
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Steve Edwards
• [asterisk-users] when to use e1/t1 card?   Steve Edwards
• [asterisk-users] when to use e1/t1 card?   Steve Edwards
• [asterisk-users] Xorocom Missing files...where to get it? astribank_upgrade   Steve Edwards
• [asterisk-users] one for your filters   Steve Edwards
• [asterisk-users] one for your filters   Steve Edwards
• [asterisk-users] one for your filters   Steve Edwards
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Steve Edwards
• [asterisk-users] Local channel usage   Steve Edwards
• [asterisk-users] How to stop intruder from registering sip?   Steve Edwards
• [asterisk-users] Pbx_lua vs. calling lua thru AGI?   Steve Edwards
• [asterisk-users] call file question   Steve Edwards
• [asterisk-users] Error - Failed to extend from xxx to xxx   Stuart Elvish
• [asterisk-users] Big time system   Stuart Elvish
• [asterisk-users] Error - Failed to extend from xxx to xxx   Stuart Elvish
• [asterisk-users] Error - Failed to extend from xxx to xxx   Stuart Elvish
• [asterisk-users] Call file structure and syntax   Mike Ely
• [asterisk-users] Call file structure and syntax   Mike Ely
• [asterisk-users] Call file structure and syntax   Mike Ely
• [asterisk-users] Call file structure and syntax   Mike Ely
• [asterisk-users] Asterisk distribution for a Call Center   Lenz Emilitri
• [asterisk-users] Small PC to build and run Asterisk   John Ervin
• [asterisk-users] Qwest PRIs   C F
• [asterisk-users] Qwest PRIs   C F
• [asterisk-users] Qwest PRIs   C F
• [asterisk-users] append CID label   C F
• [asterisk-users] CID   C F
• [asterisk-users] How to play Floating point numbers?   Faheem
• [asterisk-users] How to play Floating point numbers?   Faheem
• [asterisk-users] OT: Free DID/SIP accounts   Don Fanning
• [asterisk-users] Asterisk-Addons 1.6.0.6 and 1.6.1.4 Now Available   Jonathan Feally
• [asterisk-users] 1.6 issues   Sascha Ferley
• [asterisk-users] Problem with GROUP()   Helius Ferreira
• [asterisk-users] Truth in advertising   Karl Fife
• [asterisk-users] Why asterisk down when inet server down?   Cary Fitch
• [asterisk-users] Big time system   Cary Fitch
• [asterisk-users] Big time system   Cary Fitch
• [asterisk-users] Big time system   Cary Fitch
• [asterisk-users] <UsingWaitorPlaybackinhextension at gmail.com>,   Kevin P. Fleming
• [asterisk-users] <UsingWaitorPlaybackinhextension at gmail.com>,   Kevin P. Fleming
• [asterisk-users] <UsingWaitorPlaybackinhextension at gmail.com>,   Kevin P. Fleming
• [asterisk-users] 1.6 issues   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.2.7 + app_fax + OpenBSD 4.7 minor issue   Kevin P. Fleming
• [asterisk-users] Internal timing bad for Fax?   Kevin P. Fleming
• [asterisk-users] Big time system   Kevin P. Fleming
• [asterisk-users] tuning software echo cancellation   Vinícius Fontes
• [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)   Joe Freeman
• [asterisk-users] Big time system   Joe Freeman
• [asterisk-users] IAXmodem in dialplan   Dave Fullerton
• [asterisk-users] Problem with Hylafax   Dave Fullerton
• [asterisk-users] timeout problem with basic conf   Jose Flores Galicia
• [asterisk-users] one for your filters   Jian Gao
• [asterisk-users] asterisk-users Digest, Vol 71, Issue 36   Dharmesh Garg
• [asterisk-users] problem with inserting records into cdr   Tiago Geada
• [asterisk-users] Asterisk on Ubuntu   Tiago Geada
• [asterisk-users] own Caller ID   Tiago Geada
• [asterisk-users] Local channel usage   Tiago Geada
• [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?   Tiago Geada
• [asterisk-users] SMS in landline   Tiago Geada
• [asterisk-users] NO ANSWER before playback or background function?   Tiago Geada
• [asterisk-users] realtime queues "membername" problem   Tiago Geada
• [asterisk-users] Asterisk distribution for a Call Center   Tiago Geada
• [asterisk-users] What‘s the best operating system suggest for Asterisk 1.6.2.9   Tiago Geada
• [asterisk-users] read data from file system and put in a variable   Jerry Geis
• [asterisk-users] read data from file system and put in a variable   Jerry Geis
• [asterisk-users] question on nortel sip connection   Jerry Geis
• [asterisk-users] help with sip 401 unauthorized   Jerry Geis
• [asterisk-users] dialplan reload 1.4.33   Jerry Geis
• [asterisk-users] dialplan reload 1.4.33   Jerry Geis
• [asterisk-users] sip add header   Jerry Geis
• [asterisk-users] Still sipping frustration - only getting state ACK   Lyle Giese
• [asterisk-users] Is this failed Asterisk setup typical?   Gilles
• [asterisk-users] DAHDI volume   Gilles
• [asterisk-users] DAHDI volume   Gilles
• [asterisk-users] DAHDI volume   Gilles
• [asterisk-users] DAHDI volume   Gilles
• [asterisk-users] Weaknesses of Asterisk still there?   Gilles
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Gilles
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Gilles
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Gilles
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Gilles
• [asterisk-users] Is this failed Asterisk setup typical?   Gilles
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Gilles
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Gilles
• [asterisk-users] Up-to-date list of Asterisk appliances?   Gilles
• [asterisk-users] [voice mail] Estimating file size?   Gilles
• [asterisk-users] Where can I find a minimal set of empty configuration files (SIP only)?   Gilles
• [asterisk-users] Up-to-date list of Asterisk appliances?   Gilles
• [asterisk-users] Pbx_lua vs. calling lua thru AGI?   Gilles
• [asterisk-users] [voice mail] Estimating file size?   Gilles
• [asterisk-users] problem with inserting records into cdr   GlenM
• [asterisk-users] problem with inserting records into cdr   GlenM
• [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)   Jimmy Godbout
• [asterisk-users] a2billing for residential voip usage   Jimmy Godbout
• [asterisk-users] Polycom firmware: split vs. combined   Jimmy Godbout
• [asterisk-users] Is there a default dial plan that is not in extention.conf?   Eyal Goltzman
• [asterisk-users] Is there a default dial plan that is not in extention.conf?   Eyal Goltzman
• [asterisk-users] unwanted entries created in dialplan from users.conf, how can I get rid of it?   Eyal Goltzman
• [asterisk-users] Where can I find a minimal set of empty configuration files (SIP only)?   Eyal Goltzman
• [asterisk-users] Configure Voicemail for Large Systems   Jonathan González
• [asterisk-users] Configure Voicemail for Large Systems   Jonathan González
• [asterisk-users] Voicemail vm-intro played even when temp greeting is setup   Jonathan González
• [asterisk-users] Extract user part from SIP URI   Jonathan González
• [asterisk-users] Voicemail vm-intro played even when temp greetingis setup   Jonathan González
• [asterisk-users] other codecs   Michael Graves
• [asterisk-users] Small PC to build and run Asterisk   Michael Graves
• [asterisk-users] SIP: match_auth_username=yes doesn't seem to work   Kenny Gryp
• [asterisk-users] warning : sip_xmit   Kenny Gryp
• [asterisk-users] Hangup Detection Problem In Turkey   Mehmet GÜLER
• [asterisk-users] Need USA DIDs   Hall, Rick
• [asterisk-users] early media issue from phone co.   Trevor Hammonds
• [asterisk-users] NMI received for unknown reason   Wade Hampton
• [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'   Faisal Hanif
• [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'   Faisal Hanif
• [asterisk-users] a2billing for residential voip usage   Faisal Hanif
• [asterisk-users] Asterisk reject SIP INTITE from different source ports   Faisal Hanif
• [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'   Faisal Hanif
• [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'   Faisal Hanif
• [asterisk-users] restricting sip users to a certain useragent   Faisal Hanif
• [asterisk-users] peer IP address in CDR   Faisal Hanif
• [asterisk-users] TDD/TTY Support   Karl Harris
• [asterisk-users] Configure WAN Phone   Nicholas Hart
• [asterisk-users] Can't call my extension   Nicholas Hart
• [asterisk-users] CDR in case of CallForwarding   Vardan Harutyunyan
• [asterisk-users] Asterisk Call routing problem   Vardan Harutyunyan
• [asterisk-users] Re : Asterisk Call routing problem   Vardan Harutyunyan
• [asterisk-users] a2billing for residential voip usage   Vardan Harutyunyan
• [asterisk-users] a2billing for residential voip usage   Vardan Harutyunyan
• [asterisk-users] a2billing for residential voip usage   Vardan Harutyunyan
• [asterisk-users] a2billing for residential voip usage   Vardan Harutyunyan
• [asterisk-users] Dual Atom mobo - call capacity   Paul Hayes
• [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)   Gordon Henderson
• [asterisk-users] tuning software echo cancellation   Gordon Henderson
• [asterisk-users] tuning software echo cancellation   Gordon Henderson
• [asterisk-users] Dual Atom mobo - call capacity   Gordon Henderson
• [asterisk-users] Small PC to build and run Asterisk   Gordon Henderson
• [asterisk-users] Why asterisk down when inet server down?   Gordon Henderson
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Gordon Henderson
• [asterisk-users] one for your filters   Gordon Henderson
• [asterisk-users] one for your filters   Gordon Henderson
• [asterisk-users] one for your filters   Gordon Henderson
• [asterisk-users] [voice mail] Estimating file size?   Gordon Henderson
• [asterisk-users] transfering active call to user's voicemail   Gordon Henderson
• [asterisk-users] G729 license key registration   Gordon Henderson
• [asterisk-users] DTMF detection issues   David Matías Hernández
• [asterisk-users] Can sip clients connect with each other directly (RTP session) ?   Rob Hillis
• [asterisk-users] Asterisk 1.6.2.9 Now Available   Hoggins!
• [asterisk-users] asterisk log problem   Dean Hoover
• [asterisk-users] one for your filters   Dean Hoover
• [asterisk-users] Voicemail : mail attachment to multiple mail-addresses   Steve Howes
• [asterisk-users] how to get call duration   Steve Howes
• [asterisk-users] reloading realtime sip peers   Steve Howes
• [asterisk-users] Out of Office   Steve Howes
• [asterisk-users] Slightly OT: Cisco SPA525G and network errors   Steve Howes
• [asterisk-users] one for your filters   Steve Howes
• [asterisk-users] one for your filters   Steve Howes
• [asterisk-users] Very strange registration problem   Steve Howes
• [asterisk-users] sip add header   Steve Howes
• [asterisk-users] Pickup a ringing Queue member   Steve Howes
• [asterisk-users] Echo problem in VoIP-calls   Steve Howes
• [asterisk-users] Voicemail : mail attachment to multiple mail-addresses   Brian C. Huffman
• [asterisk-users] How to Add IP address to SIP Domain   DHAVAL INDRODIYA
• [asterisk-users] Error compiling DAHDI...   Alejandro Imass
• [asterisk-users] Very strange registration problem   Giorgio Incantalupo
• [asterisk-users] sip_xmit: sip_xmit returned -1: Operation not permitted   Giorgio Incantalupo
• [asterisk-users] AMD   Tetra Informatica
• [asterisk-users] RES: AMD   Tetra Informatica
• [asterisk-users] Asterisk on Ubuntu   Niles Ingalls
• [asterisk-users] Still sipping frustration - only getting state ACK   Ira
• [asterisk-users] Still sipping frustration - only getting state ACK   Ira
• [asterisk-users] Still sipping frustration - only getting state ACK   Ira
• [asterisk-users] tuning software echo cancellation   Ira
• [asterisk-users] How to stop intruder from registering sip?   Ira
• [asterisk-users] features.conf - parkedcalls - transfer   Ira
• [asterisk-users] Deleting extension makes it usable?   J
• [asterisk-users] Deleting extension makes it usable?   J
• [asterisk-users] Definite app_jack trouble - unsolvable   Motiejus Jakštys
• [asterisk-users] Persuing the gtalk issue - not only jack-related   Motiejus Jakštys
• [asterisk-users] Create dialplan restrictions based on the IP Address of the SIP Client?   Motiejus Jakštys
• [asterisk-users] Still sipping frustration - only getting state ACK   Motiejus Jakštys
• [asterisk-users] Muti Asterisk   Motiejus Jakštys
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Motiejus Jakštys
• [asterisk-users] Issues running Asterisk + Iaxmodem + Hylafax on same machine   Andrew Joakimsen
• [asterisk-users] Dual Atom mobo - call capacity   Stephen Brown Jr
• [asterisk-users] Vestec Tech Support   Kashif Kahn
• [asterisk-users] Support from Vestec   Kashif Kahn
• [asterisk-users] Voicemail : mail attachment to multiple mail-addresses   Jonas Kellens
• [asterisk-users] Voicemail : mail attachment to multiple mail-addresses   Jonas Kellens
• [asterisk-users] Voicemail : mail attachment to multiple mail-addresses   Jonas Kellens
• [asterisk-users] Voicemail : mail attachment to multiple mail-addresses   Jonas Kellens
• [asterisk-users] Problem with GROUP()   Jonas Kellens
• [asterisk-users] Problem with GROUP()   Jonas Kellens
• [asterisk-users] Can one adjust the voicemail-menu when using VoiceMailMain() ?   Jonas Kellens
• [asterisk-users] Can one adjust the voicemail-menu when using VoiceMailMain() ?   Jonas Kellens
• [asterisk-users] reloading realtime sip peers   Jonas Kellens
• [asterisk-users] reloading realtime sip peers   Jonas Kellens
• [asterisk-users] reloading realtime sip peers   Jonas Kellens
• [asterisk-users] Can one adjust the voicemail-menu when using VoiceMailMain() ?   Jonas Kellens
• [asterisk-users] warning : sip_xmit   Jonas Kellens
• [asterisk-users] logging stopped suddenly   Jonas Kellens
• [asterisk-users] logging stopped suddenly   Jonas Kellens
• [asterisk-users] Unable to pickup an extension, trying everything   Jonas Kellens
• [asterisk-users] Unable to pickup an extension, trying everything   Jonas Kellens
• [asterisk-users] Unable to pickup an extension, trying everything   Jonas Kellens
• [asterisk-users] Unable to pickup an extension, trying everything   Jonas Kellens
• [asterisk-users] Unable to pickup an extension, trying everything   Jonas Kellens
• [asterisk-users] Unable to pickup an extension, tryi   Jonas Kellens
• [asterisk-users] Unable to pickup an extension, tryi   Jonas Kellens
• [asterisk-users] Unable to pickup an extension   Jonas Kellens
• [asterisk-users] Call hangs up after exactly 1 minute   Jonas Kellens
• [asterisk-users] Unable to pickup an extension   Jonas Kellens
• [asterisk-users] Check if variable contains strings   Jonas Kellens
• [asterisk-users] Check if variable contains strings   Jonas Kellens
• [asterisk-users] Check if variable contains strings   Jonas Kellens
• [asterisk-users] Unable to pickup an extension   Jonas Kellens
• [asterisk-users] Unable to pickup an extension   Jonas Kellens
• [asterisk-users] Unable to pickup an extension   Jonas Kellens
• [asterisk-users] Unable to pickup an extension   Jonas Kellens
• [asterisk-users] Very strange registration problem   Jonas Kellens
• [asterisk-users] Very strange registration problem   Jonas Kellens
• [asterisk-users] Very strange registration problem   Jonas Kellens
• [asterisk-users] sip_xmit: sip_xmit returned -1: Operation not permitted   Jonas Kellens
• [asterisk-users] Pickup a ringing Queue member   Jonas Kellens
• [asterisk-users] Pickup a ringing Queue member   Jonas Kellens
• [asterisk-users] Problem with GoToIfTime   Jonas Kellens
• [asterisk-users] Codec negotiation   Jonas Kellens
• [asterisk-users] Echo problem in VoIP-calls   Jonas Kellens
• [asterisk-users] Echo problem in VoIP-calls   Jonas Kellens
• [asterisk-users] Echo problem in VoIP-calls   Jonas Kellens
• [asterisk-users] Echo problem in VoIP-calls   Jonas Kellens
• [asterisk-users] Echo problem in VoIP-calls   Jonas Kellens
• [asterisk-users] Echo problem in VoIP-calls   Jonas Kellens
• [asterisk-users] Echo problem in VoIP-calls   Jonas Kellens
• [asterisk-users] queue command in asterisk 1.4 with macro-argument   Jonas Kellens
• [asterisk-users] Echo problem in VoIP-calls   Jonas Kellens
• [asterisk-users] queue command in asterisk 1.4 with macro-argument   Jonas Kellens
• [asterisk-users] queue command in asterisk 1.4with macro-argument   Jonas Kellens
• [asterisk-users] queue command inasterisk 1.4with macro-argument   Jonas Kellens
• [asterisk-users] Out of Office   Don Kelly
• [asterisk-users] Out of Office   Don Kelly
• [asterisk-users] Is this failed Asterisk setup typical?   Richard Kenner
• [asterisk-users] Wierd error when compiling 1.6.2 branch from SVN   Richard Kenner
• [asterisk-users] dsp.c: digit_state.current_len   Richard Kenner
• [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel   Richard Kenner
• [asterisk-users] Issues with Vestec ASR   Richard Kenner
• [asterisk-users] Issues with Vestec ASR   Richard Kenner
• [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel   Richard Kenner
• [asterisk-users] Issues with Vestec ASR   Richard Kenner
• [asterisk-users] CID name in Facility message for Q.SIG   Richard Kenner
• [asterisk-users] Support from Vestec   Richard Kenner
• [asterisk-users] Skype for SIP   Thomas Kenyon
• [asterisk-users] Codec negotiation   Mindaugas Kezys
• [asterisk-users] peer IP address in CDR   Mindaugas Kezys
• [asterisk-users] Persuing the gtalk issue - not only jack-related   Kyle Kienapfel
• [asterisk-users] Codec G.129 A vs A/B   Kyle Kienapfel
• [asterisk-users] OrderlyStats SE 1.6.2l now available.   Matt King
• [asterisk-users] Non-native codecs - MELPe?   Kirin, Carol (IS)
• [asterisk-users] parking on ast 1.6.2.8   Jeremy Kister
• [asterisk-users] <UsingWaitorPlaybackinhextension at gmail.com>,   Philipp von Klitzing
• [asterisk-users] <UsingWaitorPlaybackinhextension at gmail.com>,   Philipp von Klitzing
• [asterisk-users] How to play Floating point numbers?   Philipp von Klitzing
• [asterisk-users] ISDN -> SIP   Philipp von Klitzing
• [asterisk-users] no ring back 180 with SDP   Philipp von Klitzing
• [asterisk-users] ISDN -> SIP   Philipp von Klitzing
• [asterisk-users] no ring back 180 with SDP   Philipp von Klitzing
• [asterisk-users] bug with Moh on MeetMe ?   Philipp von Klitzing
• [asterisk-users] How to disable day light saving on Snom 360 phones?   Philipp von Klitzing
• [asterisk-users] Unable to pickup an extension, trying everything   Philipp von Klitzing
• [asterisk-users] Skype for Asterisk - what processors/platforms does it run on?   Philipp von Klitzing
• [asterisk-users] Unable to pickup an extension, trying everything   Philipp von Klitzing
• [asterisk-users] Unable to pickup an extension, tryi   Philipp von Klitzing
• [asterisk-users] Unable to pickup an extension   Philipp von Klitzing
• [asterisk-users] Unable to pickup an extension   Philipp von Klitzing
• [asterisk-users] Unable to pickup an extension   Philipp von Klitzing
• [asterisk-users] Why asterisk down when inet server down?   Philipp von Klitzing
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Philipp von Klitzing
• [asterisk-users] NO ANSWER before playback or background function?   Philipp von Klitzing
• [asterisk-users] Unregister and register SIP phones by using num pad on phones?   Philipp von Klitzing
• [asterisk-users] Local channel usage   Philipp von Klitzing
• [asterisk-users] Call file structure and syntax   Philipp von Klitzing
• [asterisk-users] cmd Authenticate   Philipp von Klitzing
• [asterisk-users] peer IP address in CDR   Philipp von Klitzing
• [asterisk-users] Codec negotiation   Philipp von Klitzing
• [asterisk-users] Codec negotiation   Philipp von Klitzing
• [asterisk-users] peer IP address in CDR   Philipp von Klitzing
• [asterisk-users] Minimum modules required to run VoIP and CDR   Philipp von Klitzing
• [asterisk-users] Echo problem in VoIP-calls   Philipp von Klitzing
• [asterisk-users] MeetMe problem   Daniel Knoll
• [asterisk-users] MeetMe problem   Daniel Knoll
• [asterisk-users] bug with Moh on MeetMe ?   Daniel Knoll
• [asterisk-users] debug message: Internal timing is disabled   Daniel Knoll
• [asterisk-users] dahdi span   Daniel Knoll
• [asterisk-users] load balance meetme   Daniel Knoll
• [asterisk-users] dahdi span   Daniel Knoll
• [asterisk-users] joining 2 conferences together   Daniel Knoll
• [asterisk-users] Migrating from key system to asterisk   Brian Kolaci
• [asterisk-users] transfering active call to user's voicemail   Rustam Kovhaev
• [asterisk-users] Fwd: Read and set the UUI in asterisk   velusamy Krishnan
• [asterisk-users] Read and set the UUI in asterisk   velusamy Krishnan
• [asterisk-users] AMI Queue information about incoming call's channel before link   Alexandr Krylovskiy
• [asterisk-users] (no subject)   Dmitry Kupchinetsky
• [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)   Jeff LaCoursiere
• [asterisk-users] tuning software echo cancellation   Jeff LaCoursiere
• [asterisk-users] tuning software echo cancellation   Jeff LaCoursiere
• [asterisk-users] tuning software echo cancellation   Jeff LaCoursiere
• [asterisk-users] Dual Atom mobo - call capacity   Jeff LaCoursiere
• [asterisk-users] Small PC to build and run Asterisk   Jeff LaCoursiere
• [asterisk-users] Small PC to build and run Asterisk   Jeff LaCoursiere
• [asterisk-users] Sangoma - how to show channels in use?   Jeff LaCoursiere
• [asterisk-users] Sangoma - how to show channels in use?   Jeff LaCoursiere
• [asterisk-users] one for your filters   Jeff LaCoursiere
• [asterisk-users] one for your filters   Jeff LaCoursiere
• [asterisk-users] one for your filters   Jeff LaCoursiere
• [asterisk-users] one for your filters   Jeff LaCoursiere
• [asterisk-users] call file question   Jeff LaCoursiere
• [asterisk-users] LumenVox *.gram reload   Tony LaMear
• [asterisk-users] early media issue from phone co.   Edwin Lam
• [asterisk-users] Small VoIP company looking for Asterisk Scalability and Maintenance Engineer   James Lamanna
• [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)   James Lamanna
• [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)   James Lamanna
• [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)   James Lamanna
• [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)   James Lamanna
• [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)   James Lamanna
• [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)   James Lamanna
• [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)   James Lamanna
• [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)   James Lamanna
• [asterisk-users] a2billing for residential voip usage   Landy Landy
• [asterisk-users] a2billing for residential voip usage   Landy Landy
• [asterisk-users] a2billing for residential voip usage   Landy Landy
• [asterisk-users] a2billing for residential voip usage   Landy Landy
• [asterisk-users] a2billing for residential voip usage   Landy Landy
• [asterisk-users] a2billing for residential voip usage   Landy Landy
• [asterisk-users] Find a way to block brute force attacks.   Rodrigo Lang
• [asterisk-users] Find a way to block brute force attacks.   Rodrigo Lang
• [asterisk-users] Find a way to block brute force attacks.   Rodrigo Lang
• [asterisk-users] Return agi script.   Rodrigo Lang
• [asterisk-users] Return agi script.   Rodrigo Lang
• [asterisk-users] Return agi script.   Rodrigo Lang
• [asterisk-users] Can one adjust the voicemail-menu when using VoiceMailMain() ?   Glenn O Larsen
• [asterisk-users] Can one adjust the voicemail-menu when using VoiceMailMain() ?   Glenn O Larsen
• [asterisk-users] Warning spamming for any unsynchronized ISDN port with dahdi-2.3.0.1   Alex Villací­s Lasso
• [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel   Andrew Latham
• [asterisk-users] Asterisk distribution for a Call Center   Andrew Latham
• [asterisk-users] Asterisk distribution for a Call Center   Andrew Latham
• [asterisk-users] one for your filters   Andrew Latham
• [asterisk-users] OT: Bandwidth calculations   Andrew Latham
• [asterisk-users] OT: Bandwidth calculations   Andrew Latham
• [asterisk-users] Update the LCD with the callee's name after dialing   Andrew Latham
• [asterisk-users] Find a way to block brute force attacks.   Andrew Latham
• [asterisk-users] Is there a default dial plan that is not in extention.conf?   Geraint Lee
• [asterisk-users] Fwd: Read and set the UUI in asterisk   Tilghman Lesher
• [asterisk-users] Wierd error when compiling 1.6.2 branch from SVN   Tilghman Lesher
• [asterisk-users] problem with port 5090 registration   Tilghman Lesher
• [asterisk-users] problem with port 5090 registration   Tilghman Lesher
• [asterisk-users] reloading realtime sip peers   Tilghman Lesher
• [asterisk-users] [compat] section in asterisk.conf : compatibility with pipe delimiter   Tilghman Lesher
• [asterisk-users] no ring back 180 with SDP   Tilghman Lesher
• [asterisk-users] How to stop intruder from registering sip?   Tilghman Lesher
• [asterisk-users] How to stop intruder from registering sip?   Tilghman Lesher
• [asterisk-users] ring no answer / RONA versus HangUp   Tilghman Lesher
• [asterisk-users] Check if variable contains strings   Tilghman Lesher
• [asterisk-users] Asterisk SIP/IAX peers can't connect after Firewall change?   Tilghman Lesher
• [asterisk-users] What's diff between ${CDR(start)} , ${CDR(answer)} , ${CDR(calldate)}   Tilghman Lesher
• [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'   Tilghman Lesher
• [asterisk-users] Voicemail ODBC   Tilghman Lesher
• [asterisk-users] dahdi span   Tilghman Lesher
• [asterisk-users] Voicemail ODBC   Tilghman Lesher
• [asterisk-users] Voicemail ODBC   Tilghman Lesher
• [asterisk-users] How do I access the Dialstatus numeric code received?   Tilghman Lesher
• [asterisk-users] How do I access the Dialstatus numeric code received?   Tilghman Lesher
• [asterisk-users] "Hidden" memory leak   Tilghman Lesher
• [asterisk-users] Error - Failed to extend from xxx to xxx   Tilghman Lesher
• [asterisk-users] Error - Failed to extend from xxx to xxx   Tilghman Lesher
• [asterisk-users] How to stop intruder from registering sip?   Tilghman Lesher
• [asterisk-users] Return agi script.   Tilghman Lesher
• [asterisk-users] Detecting hook flash in asterisk   Ye Liu
• [asterisk-users] Press twice *   Anahi Ludueña
• [asterisk-users] Call ended after 31 seconds   Anahi Ludueña
• [asterisk-users] Call ended after 31 seconds   Anahi Ludueña
• [asterisk-users] Music on Hold problema   Anahi Ludueña
• [asterisk-users] Music on Hold problema   Anahi Ludueña
• [asterisk-users] Music on Hold problema   Anahi Ludueña
• [asterisk-users] Music on Hold problema   Anahi Ludueña
• [asterisk-users] FW: Music on Hold problema   Anahi Ludueña
• [asterisk-users] FW: Music on Hold problema   Anahi Ludueña
• [asterisk-users] Dial options not working   Anahi Ludueña
• [asterisk-users] Dial options not working   Anahi Ludueña
• [asterisk-users] Dial options not working   Anahi Ludueña
• [asterisk-users] Dial options not working   Anahi Ludueña
• [asterisk-users] Problem with extensions in IVR and queues   Anahi Ludueña
• [asterisk-users] Problem with extensions in IVR and queues   Anahi Ludueña
• [asterisk-users] Problem with extensions in IVR and queues   Anahi Ludueña
• [asterisk-users] Problem with extensions in IVR and queues   Anahi Ludueña
• [asterisk-users] applicationmap and ChannelRedirect   Per-Henrik Lundblom
• [asterisk-users] applicationmap and ChannelRedirect   Per-Henrik Lundblom
• [asterisk-users] applicationmap and ChannelRedirect   Per-Henrik Lundblom
• [asterisk-users] Lookup ${EXTEN} in database, update context/route if found... AGI?   Doug Lytle
• [asterisk-users] 1.6 issues   Doug Lytle
• [asterisk-users] Qwest PRIs   Doug Lytle
• [asterisk-users] Qwest PRIs   Doug Lytle
• [asterisk-users] Issues running Asterisk + Iaxmodem + Hylafax on same machine   Doug Lytle
• [asterisk-users] Qwest PRIs   Doug Lytle
• [asterisk-users] Asterisk + E1 card   Doug Lytle
• [asterisk-users] Automatic attendant - Error in CLI.   Doug Lytle
• [asterisk-users] What is the voicemail "u option"   Doug Lytle
• [asterisk-users] Is there a default dial plan that is not in extention.conf?   Doug Lytle
• [asterisk-users] cmd Authenticate   Doug Lytle
• [asterisk-users] Never seen Problem !!!   G M
• [asterisk-users] Never seen Problem !!!   G M
• [asterisk-users] Asterisk 1.6.2.8 Now Available   Leif Madsen
• [asterisk-users] Persuing the gtalk issue - not only jack-related   Leif Madsen
• [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel   Leif Madsen
• [asterisk-users] Ring + Music on Hold in the same call   Leif Madsen
• [asterisk-users] Using SetVar with System() is it possible?   Leif Madsen
• [asterisk-users] Asterisk 1.6 + Jabber crashes   Leif Madsen
• [asterisk-users] What‘s the best operating system suggest for Asterisk 1.6.2.9   Leif Madsen
• [asterisk-users] Minimum modules required to run VoIP and CDR   Leif Madsen
• [asterisk-users] Problem with iax2/rsa registration   Andre Magalhaes
• [asterisk-users] HElP me I am a beginner   Ishfaq Malik
• [asterisk-users] HElP me I am a beginner   Ishfaq Malik
• [asterisk-users] Deleting extension makes it usable?   Ishfaq Malik
• [asterisk-users] Group call limit   Ishfaq Malik
• [asterisk-users] logging stopped suddenly   Ishfaq Malik
• [asterisk-users] Hint priority in RealTime   Ishfaq Malik
• [asterisk-users] How to get asterisk to playback personal greetings using grandstream gxp-2000   Ishfaq Malik
• [asterisk-users] Blind transfer feature   Adrian Marsh
• [asterisk-users] How to stop intruder from registering sip?   Martin
• [asterisk-users] How to stop intruder from registering sip?   Martin
• [asterisk-users] How to stop intruder from registering sip?   Martin
• [asterisk-users] UDPTL T38 via NAT   Martin
• [asterisk-users] How to kick/mute using ConfBridge application   Bruce McAlister
• [asterisk-users] 'NO ANSWER' with cdr duration of 0   Josh McAllister
• [asterisk-users] Asterisk 1.6 + Jabber crashes   Michael
• [asterisk-users] Reloading queue members (realtime DB)   Mike
• [asterisk-users] OT: Windows TAPI command-line driver   Mike
• [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)   Mike
• [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)   Mike
• [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)   Mike
• [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)   Mike
• [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)   Mike
• [asterisk-users] Limit total length of calls to a specifig SIP peer   Mike
• [asterisk-users] Priority between calls in different queues   Mike
• [asterisk-users] Call ended after 31 seconds   Mike
• [asterisk-users] Multiple parking lots - 1.6   Mike
• [asterisk-users] Slightly OT: Cisco SPA525G and network errors   Mike
• [asterisk-users] OT: Physical SIP phone with inbuilt VPN support   Mike
• [asterisk-users] PRI span problem - no D channel   Mike
• [asterisk-users] PRI span problem - no D channel   Mike
• [asterisk-users] PRI span problem - no D channel   Mike
• [asterisk-users] PRI span problem - no D channel   Mike
• [asterisk-users] PRI span problem - no D channel   Mike
• [asterisk-users] Call drops on group paging asterisk - 1.4.22.1   Mike
• [asterisk-users] Meetme delay - normal?   Mike
• [asterisk-users] Asterisk 1.6 and multiple parking   Mike
• [asterisk-users] Asterisk 1.6 and multiple parking   Mike
• [asterisk-users] Asterisk 1.6 and multiple parking   Mike
• [asterisk-users] Asterisk 1.6 and multiple parking   Mike
• [asterisk-users] Asterisk 1.6 and multiple parking   Mike
• [asterisk-users] How to get asterisk to playback personal greetings using grandstream gxp-2000   Eddie Mikell
• [asterisk-users] What is the voicemail "u option"   Eddie Mikell
• [asterisk-users] OT: Cisco ATA 186   Sebastian Milioto
• [asterisk-users] Time variables in system application   Barry Miller
• [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'   Barry Miller
• [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'   Barry Miller
• [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'   Barry Miller
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Barry Miller
• [asterisk-users] T.38 Peer Negotiation Fails   Chris Miller
• [asterisk-users] CallWaiting   Akshay Mishra
• [asterisk-users] How to work Asterisk with Video Conference   Hiren Mistry
• [asterisk-users] Use one ring-group for ISN truncs   Arjan Kroon | Mobillion
• [asterisk-users] Use one group for ISN truncs   Arjan Kroon | Mobillion
• [asterisk-users] Carrier needs more call examples   Adam Moffett
• [asterisk-users] DAHDI volume   Miguel Molina
• [asterisk-users] when to use e1/t1 card?   Miguel Molina
• [asterisk-users] "Hidden" memory leak   Miguel Molina
• [asterisk-users] "Hidden" memory leak   Miguel Molina
• [asterisk-users] MeetMe   Mickael Monsieur
• [asterisk-users] contacting   Mickael Monsieur
• [asterisk-users] contacting   Mickael Monsieur
• [asterisk-users] MeetMe   Mickael Monsieur
• [asterisk-users] bug with Moh on MeetMe ?   Mickael Monsieur
• [asterisk-users] I look ARI (Asterisk Recording Interface)   Mickael Monsieur
• [asterisk-users] I look ARI (Asterisk Recording Interface)   Mickael Monsieur
• [asterisk-users] ISP down internal phones become unavailable   Lacy Moore
• [asterisk-users] Asterisk on Ubuntu   Luis Morales
• [asterisk-users] Is Centos 64 bit or 32 bit better?   Luis Morales
• [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?   Luis Morales
• [asterisk-users] Asterisk distribution for a Call Center   Luciano Moreira
• [asterisk-users] How to find a single call in logs   Douglas Mortensen
• [asterisk-users] How to tell if a dropped call is my fault   Douglas Mortensen
• [asterisk-users] Endless loop with asterisk directory   Douglas Mortensen
• [asterisk-users] Using 5th gen TE420 with Asterisk 1.2?   Tony Mountifield
• [asterisk-users] Create Conference and exit myself   RSCL Mumbai
• [asterisk-users] Need USA DIDs   RSCL Mumbai
• [asterisk-users] Need USA DIDs   RSCL Mumbai
• [asterisk-users] Corba interface   Muro, Sam
• [asterisk-users] Corba interface   Muro, Sam
• [asterisk-users] Deleting extension makes it usable?   Steve Murphy
• [asterisk-users] HElP me I am a beginner   Tim Nelson
• [asterisk-users] HElP me I am a beginner   Tim Nelson
• [asterisk-users] Lookup ${EXTEN} in database, update context/route if found... AGI?   Tim Nelson
• [asterisk-users] Why asterisk down when inet server down?   Tim Nelson
• [asterisk-users] when to use e1/t1 card?   Tim Nelson
• [asterisk-users] when to use e1/t1 card?   Tim Nelson
• [asterisk-users] Using Local in queues a good idea? (or at least not a very bad idea?)   Håkon Nessjøen
• [asterisk-users] Voicemail : mail attachment to multiplemail-addresses   Danny Nicholas
• [asterisk-users] no sound between extensions   Danny Nicholas
• [asterisk-users] Voicemail bug(?) with Asterisk 1.6.2.8-rc1   Danny Nicholas
• [asterisk-users] DAHDI volume   Danny Nicholas
• [asterisk-users] run script after completed   Danny Nicholas
• [asterisk-users] Press twice *   Danny Nicholas
• [asterisk-users] How to play Floating point numbers?   Danny Nicholas
• [asterisk-users] Issues with Vestec ASR   Danny Nicholas
• [asterisk-users] Limit total length of calls to a specifig SIP peer   Danny Nicholas
• [asterisk-users] LumenVox *.gram reload   Danny Nicholas
• [asterisk-users] Issues with Vestec ASR   Danny Nicholas
• [asterisk-users] PSTN-IVR call   Danny Nicholas
• [asterisk-users] Out of Office   Danny Nicholas
• [asterisk-users] Ring + Music on Hold in the same call   Danny Nicholas
• [asterisk-users] Ring + Music on Hold in the same call   Danny Nicholas
• [asterisk-users] Out of Office   Danny Nicholas
• [asterisk-users] Out of Office   Danny Nicholas
• [asterisk-users] Small PC to build and run Asterisk   Danny Nicholas
• [asterisk-users] Voicemail vm-intro played even when temp greetingis setup   Danny Nicholas
• [asterisk-users] TDD/TTY Support   Danny Nicholas
• [asterisk-users] read data from file system and put in avariable   Danny Nicholas
• [asterisk-users] Check if variable contains strings   Danny Nicholas
• [asterisk-users] IVR extension dialing error   Danny Nicholas
• [asterisk-users] Check if variable contains strings   Danny Nicholas
• [asterisk-users] IVR extension dialing error   Danny Nicholas
• [asterisk-users] Unable to pickup an extension   Danny Nicholas
• [asterisk-users] IVR extension dialing error   Danny Nicholas
• [asterisk-users] Unable to pickup an extension   Danny Nicholas
• [asterisk-users] Music on Hold problema   Danny Nicholas
• [asterisk-users] asterisk issue   Danny Nicholas
• [asterisk-users] IVR extension dialing error   Danny Nicholas
• [asterisk-users] Music on Hold problema   Danny Nicholas
• [asterisk-users] Why asterisk down when inet server down?   Danny Nicholas
• [asterisk-users] asterisk issue   Danny Nicholas
• [asterisk-users] using call file   Danny Nicholas
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Danny Nicholas
• [asterisk-users] What is the voicemail "u option"   Danny Nicholas
• [asterisk-users] storing DTMF inputs   Danny Nicholas
• [asterisk-users] Sangoma - how to show channels in use?   Danny Nicholas
• [asterisk-users] Call file structure and syntax   Danny Nicholas
• [asterisk-users] Running SIP on non-standard ports   Danny Nicholas
• [asterisk-users] joining 2 conferences together   Danny Nicholas
• [asterisk-users] Asterisk + E1 card   Danny Nicholas
• [asterisk-users] cmd Authenticate   Danny Nicholas
• [asterisk-users] cmd Authenticate   Danny Nicholas
• [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?   Danny Nicholas
• [asterisk-users] Dial options not working   Danny Nicholas
• [asterisk-users] Echo problem in VoIP-calls   Danny Nicholas
• [asterisk-users] queue command in asterisk 1.4 withmacro-argument   Danny Nicholas
• [asterisk-users] Problem with extensions in IVR and queues   Danny Nicholas
• [asterisk-users] queue command in asterisk 1.4with macro-argument   Danny Nicholas
• [asterisk-users] Asterisk 1.6 and multiple parking   Danny Nicholas
• [asterisk-users] queue command inasterisk 1.4with macro-argument   Danny Nicholas
• [asterisk-users] Return agi script.   Danny Nicholas
• [asterisk-users] Problem with extensions in IVR and queues   Danny Nicholas
• [asterisk-users] Return agi script.   Danny Nicholas
• [asterisk-users] Asterisk reject SIP INTITE from different source ports   Deepika Nijhawan
• [asterisk-users] Asterisk reject SIP INTITE from different   Deepika Nijhawan
• [asterisk-users] Asterisk reject SIP INTITE from different source ports   Deepika Nijhawan
• [asterisk-users] Asterisk reject SIP INTITE from different source ports   Deepika Nijhawan
• [asterisk-users] Asterisk reject SIP INTITE from different source ports   Deepika Nijhawan
• [asterisk-users] CDRs not getting generated on Free PBX   Deepika Nijhawan
• [asterisk-users] CDRs not getting generated on Free PBX   Deepika Nijhawan
• [asterisk-users] CDRs not getting generated on Free PBX   Deepika Nijhawan
• [asterisk-users] About Sangoma cards and Asterisk integration with other PBX   John Novack
• [asterisk-users] Is this failed Asterisk setup typical?   John Novack
• [asterisk-users] one for your filters   John Novack
• [asterisk-users] What‘s the best operating system suggest for Asterisk 1.6.2.9   John Novack
• [asterisk-users] Codec G.129 A vs A/B   Alejandro Cabrera Obed
• [asterisk-users] Asterisk + E1 card   Alejandro Cabrera Obed
• [asterisk-users] IVR extension dialing error   Alejandro Cabrera Obed
• [asterisk-users] IVR extension dialing error   Alejandro Cabrera Obed
• [asterisk-users] IVR extension dialing error   Alejandro Cabrera Obed
• [asterisk-users] IVR extension dialing error   Alejandro Cabrera Obed
• [asterisk-users] Asterisk distribution for a Call Center   Alejandro Cabrera Obed
• [asterisk-users] Asterisk distribution for a Call Center   Alejandro Cabrera Obed
• [asterisk-users] Asterisk + E1 card   Alejandro Cabrera Obed
• [asterisk-users] IVR extension dialing error   Alejandro Cabrera Obed
• [asterisk-users] Handling DTMF for number 4   Alejandro Cabrera Obed
• [asterisk-users] Handling DTMF for number 4   Alejandro Cabrera Obed
• [asterisk-users] OT - Astmanproxy download broken ?   Olivier
• [asterisk-users] OT - Astmanproxy download broken ?   Olivier
• [asterisk-users] Pri show span and PtMP mode   Olivier
• [asterisk-users] HDLC Bad FCS (8) on Primary D-channel   Olivier
• [asterisk-users] HDLC Bad FCS (8) on Primary D-channel   Olivier
• [asterisk-users] HDLC Bad FCS (8) on Primary D-channel   Olivier
• [asterisk-users] Asterisk + Dahdi does not work with BRI NT mode   Olivier
• [asterisk-users] OT - Explain RDNIS   Olivier
• [asterisk-users] How do I access the Dialstatus numeric code received?   Olivier
• [asterisk-users] Generate cdr on Hangup   Olivier
• [asterisk-users] Create Conference and exit myself   Olivier
• [asterisk-users] features.conf - parkedcalls - transfer   Olivier
• [asterisk-users] Group call limit   Alexandru Oniciuc
• [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?   Elliot Otchet
• [asterisk-users] H323 Trunk Problem calling from Asterisk to Avaya PBX   Shina Owolabi
• [asterisk-users] Is this failed Asterisk setup typical?   Peder
• [asterisk-users] Unable to pickup an extension, trying everything   Peder
• [asterisk-users] VAD and cRTP, any thing else?   Peder
• [asterisk-users] Problem with GoToIfTime   Peder
• [asterisk-users] Carrier needs more call examples   Peder
• [asterisk-users] MeetMe problem   Thomas Perron
• [asterisk-users] append CID label   Thomas Perron
• [asterisk-users] append CID label   Thomas Perron
• [asterisk-users] Unable to set callerid for incoming skype calls   Matteo Piazza
• [asterisk-users] How to stop intruder from registering sip?   Dave Platt
• [asterisk-users] How to stop intruder from registering sip?   Dave Platt
• [asterisk-users] one for your filters   Dave Platt
• [asterisk-users] one for your filters   Dave Platt
• [asterisk-users] How to stop intruder from registering sip?   Fred Posner
• [asterisk-users] How to stop intruder from registering sip?   Fred Posner
• [asterisk-users] Corba interface   Steve Prior
• [asterisk-users] Problem with dahdi and with freepbx   Claudio Prono
• [asterisk-users] sipconnect 1.0   James Puckett
• [asterisk-users] Asterisk 1.6.2.7 + app_fax + OpenBSD 4.7 minor issue   Noah Pugsley
• [asterisk-users] Queue with PopUP screen for customer   Edwin Quijada
• [asterisk-users] own Caller ID   Edwin Quijada
• [asterisk-users] [AGI] What scripting language for embedded hardware?   Edwin Quijada
• [asterisk-users] Asterisk distribution for a Call Center   Edwin Quijada
• [asterisk-users] Find a way to block brute force attacks.   Mark R
• [asterisk-users] HElP me I am a beginner   Randy R
• [asterisk-users] Asterisk 1.6.2.7 + app_fax + OpenBSD 4.7 minor issue   Randy R
• [asterisk-users] Small PC to build and run Asterisk   Randy R
• [asterisk-users] Small PC to build and run Asterisk   Randy R
• [asterisk-users] Small PC to build and run Asterisk   Randy R
• [asterisk-users] Skype for Asterisk - what processors/platforms does it run on?   Randy R
• [asterisk-users] Skype for SIP   Randy R
• [asterisk-users] Skype for SIP   Randy R
• [asterisk-users] Small PC to build and run Asterisk   Randy R
• [asterisk-users] Small PC to build and run Asterisk   Randy R
• [asterisk-users] Friday June 18th at 12 Noon EDT: Session Border Controllers, 1PM Bria iPhone SIP app   Randy R
• [asterisk-users] Friday at 1PM: SIPVicious has a new tool: svcrash   Randy R
• [asterisk-users] OT: Bandwidth calculations   Randy R
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Randy R
• [asterisk-users] OT: Bandwidth calculations   Randy R
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Randy R
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Randy R
• [asterisk-users] Cutting the CallerID(RDNIS)   Chris Ramirez
• [asterisk-users] Cutting the CallerID(RDNIS)   Chris Ramirez
• [asterisk-users] Cutting the CallerID(RDNIS)   Chris Ramirez
• [asterisk-users] cmd Authenticate   Coco Richard
• [asterisk-users] cmd Authenticate   Coco Richard
• [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT   JR Richardson
• [asterisk-users] Caller id, sip header from problem   Alexandre Rodrigues
• [asterisk-users] I look ARI (Asterisk Recording Interface)   Carlos Rojas
• [asterisk-users] Asterisk AMI   Ron
• [asterisk-users] AMD   John Rose
• [asterisk-users] SIP Witch   Matthew J. Roth
• [asterisk-users] SIP Witch   Matthew J. Roth
• [asterisk-users] Problem with TE411P and DAHDI   Shaun Ruffell
• [asterisk-users] Asterisk on Ubuntu   ABBAS SHAKEEL
• [asterisk-users] Small PC to build and run Asterisk   SIP
• [asterisk-users] Small PC to build and run Asterisk   SIP
• [asterisk-users] Need USA DIDs   SIP
• [asterisk-users] Small PC to build and run Asterisk   Saflis ST
• [asterisk-users] Create dialplan restrictions based on the IP Address of the SIP Client?   Raimund Sacherer
• [asterisk-users] Problem with Hylafax   Samantha
• [asterisk-users] RE How to break pri DID to multiple SIP Trunks   Samantha
• [asterisk-users] Delay in IVR   Sasa
• [asterisk-users] Delay in IVR   Sasa
• [asterisk-users] Delay in IVR   Sasa
• [asterisk-users] Delay in IVR   Sasa
• [asterisk-users] Loud Noise when trying to call through PSTN.   Arun Sasidhar
• [asterisk-users] How to stop intruder from registering sip?   Tarek Sawah
• [asterisk-users] Call queues - issues, can't make it work.   Tarek Sawah
• [asterisk-users] calling peer from server   Tarek Sawah
• [asterisk-users] asterisk issue   Tarek Sawah
• [asterisk-users] Asterisk distribution for a Call Center   Tarek Sawah
• [asterisk-users] Need USA DIDs   Tarek Sawah
• [asterisk-users] help with sip 401 unauthorized   Tarek Sawah
• [asterisk-users] one for your filters   Tarek Sawah
• [asterisk-users] Need USA DIDs   Tarek Sawah
• [asterisk-users] Big time system   Tarek Sawah
• [asterisk-users] restricting sip users to a certain useragent   Tarek Sawah
• [asterisk-users] Hot to configure trunk in asterisk with a2billing.   Tarek Sawah
• [asterisk-users] restricting sip users to a certain useragent   Tarek Sawah
• [asterisk-users] SIP Delay with remote stations?   Tarek Sawah
• [asterisk-users] Voicemail bug(?) with Asterisk 1.6.2.8-rc1   Stefan Schmidt
• [asterisk-users] Call ended after 31 seconds   Stefan Schmidt
• [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)   Stefan Schmidt
• [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)   Stefan Schmidt
• [asterisk-users] Problem attended transfer with ilbc   Stefan Schmidt
• [asterisk-users] Read and set the UUI in asterisk   Warren Selby
• [asterisk-users] Error compiling DAHDI...   Warren Selby
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   Warren Selby
• [asterisk-users] How to stop intruder from registering sip?   Warren Selby
• [asterisk-users] WARNING message when play   Warren Selby
• [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'   Warren Selby
• [asterisk-users] features.conf - parkedcalls - transfer   Warren Selby
• [asterisk-users] Polycom firmware: split vs. combined   Warren Selby
• [asterisk-users] Anyone can share their config file for Cisco phone please?   Warren Selby
• [asterisk-users] Minimum modules required to run VoIP and CDR   Warren Selby
• [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?   Hugo Serrano
• [asterisk-users] app_swift v2.0 released   Darren Sessions
• [asterisk-users] Relaunch of the Kansas City Asterisk User Group   Kyle Sexton
• [asterisk-users] What's diff between ${CDR(start)} , ${CDR(answer)} , ${CDR(calldate)}   Zhang Shukun
• [asterisk-users] What's diff between ${CDR(start)} , ${CDR(answer)} , ${CDR(calldate)}   Zhang Shukun
• [asterisk-users] NO ANSWER before playback or background function?   Zhang Shukun
• [asterisk-users] NO ANSWER before playback or background function?   Zhang Shukun
• [asterisk-users] What‘s the best operating system suggest for Asterisk 1.6.2.9   Zhang Shukun
• [asterisk-users] Is Centos 64 bit or 32 bit better?   Zhang Shukun
• [asterisk-users] ISP down internal phones become unavailable   Mr Shunz
• [asterisk-users] Priority between calls in different queues   Hapworth Slim
• [asterisk-users] Priority between calls in different queues   Hapworth Slim
• [asterisk-users] Priority between calls in different queues   Hapworth Slim
• [asterisk-users] Voicemail : mail attachment to multiple mail-addresses   Jared Smith
• [asterisk-users] DAHDI volume   Jared Smith
• [asterisk-users] Still sipping frustration - only getting state ACK   Jared Smith
• [asterisk-users] Can sip clients connect with each other directly (RTP session) ?   Kamonwat Sookkara
• [asterisk-users] UDPTL T38 via NAT   Johann Steinwendtner
• [asterisk-users] UDPTL T38 via NAT   Johann Steinwendtner
• [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?   A J Stiles
• [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?   A J Stiles
• [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?   A J Stiles
• [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?   A J Stiles
• [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?   A J Stiles
• [asterisk-users] PRI span problem - no D channel   A J Stiles
• [asterisk-users] restricting sip users to a certain useragent   A J Stiles
• [asterisk-users] DAHDI PRI error message   Scott Stingel
• [asterisk-users] DAHDI PRI error message   Scott Stingel
• [asterisk-users] DAHDI: Inbound BRI call, DDI not presented   Scott Stingel
• [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?   Scott Stingel
• [asterisk-users] DAHDI: Inbound BRI call, DDI not presented   Scott Stingel
• [asterisk-users] FIXED: DAHDI: Inbound BRI call, DDI not presented   Scott Stingel
• [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?   Scott Stingel
• [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?   Scott Stingel
• [asterisk-users] one for your filters   Administrator TOOTAI
• [asterisk-users] Asterisk distribution for a Call Center   Carlo Taguinod
• [asterisk-users] Delay in IVR   Kingsley Tart
• [asterisk-users] Problem attended transfer with ilbc   John Taylor
• [asterisk-users] Asterisk 1.6.2.8 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.32 Now Available   Asterisk Development Team
• [asterisk-users] libpri 1.4.11.1 Now Available   Asterisk Development Team
• [asterisk-users] libpri 1.4.11.2 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk-Addons 1.6.0.6 and 1.6.1.4 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.33 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.9 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.33.1 Released   Asterisk Development Team
• [asterisk-users] libpri 1.4.11.3 Now Available   Asterisk Development Team
• [asterisk-users] Announcement before absolute timeout / how to terminate a meetme conf?   Vidensus Techsupport
• [asterisk-users] SPA8000 outbound CID problem   Mark G. Thomas
• [asterisk-users] Polycom firmware: split vs. combined   Jonathan Thurman
• [asterisk-users] Fwd: [INSTALL #RKZ-745226]: Digium Support Survey, Partial Faxes   Steve Totaro
• [asterisk-users] Strange problem with zap channel.   Tim Uckun
• [asterisk-users] Codec G.129 A vs A/B   Steve Underwood
• [asterisk-users] TDD/TTY Support   Steve Underwood
• [asterisk-users] TDD/TTY Support   Steve Underwood
• [asterisk-users] Is Centos 64 bit or 32 bit better?   Steve Underwood
• [asterisk-users] get Asterisk version from within dialplan   Vieri
• [asterisk-users] AGI library for C/C++   Vieri
• [asterisk-users] Unavailable issue with SIP realtime and app_queue (*-1.4)   Kristijan Vrban
• [asterisk-users] Internal timing bad for Fax?   Kristijan Vrban
• [asterisk-users] How to stop intruder from registering sip?   Ryan Wagoner
• [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)   Ryan Wagoner
• [asterisk-users] ISP down internal phones become unavailable   Ryan Wagoner
• [asterisk-users] ISP down internal phones become unavailable   Ryan Wagoner
• [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)   Ryan Wagoner
• [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)   Ryan Wagoner
• [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)   Ryan Wagoner
• [asterisk-users] Codec negotiation   Ryan Wagoner
• [asterisk-users] Codec negotiation   Ryan Wagoner
• [asterisk-users] Update the LCD with the callee's name after dialing   Ryan Wagoner
• [asterisk-users] HElP me I am a beginner   Myles Wakeham
• [asterisk-users] Strange Asterisk/SIP call forwarding behavior   Myles Wakeham
• [asterisk-users] Can one adjust the voicemail-menu when using VoiceMailMain() ?   C. Chad Wallace
• [asterisk-users] sip add header   C. Chad Wallace
• [asterisk-users] question on nortel sip connection   Watkins, Bradley
• [asterisk-users] Find a way to block brute force attacks.   Kenny Watson
• [asterisk-users] Voiceprompts i.e. voicemail and conferencing in multiple codecs   Kenny Watson
• [asterisk-users] Adding Congestion to CDR logs   Kenny Watson
• [asterisk-users] Adding Congestion to CDR logs   Kenny Watson
• [asterisk-users] Adding Congestion to CDR logs   Kenny Watson
• [asterisk-users] Dial options not working   Kenny Watson
• [asterisk-users] HDLC Bad FCS (8) on Primary D-channel   Karsten Wemheuer
• [asterisk-users] HDLC Bad FCS (8) on Primary D-channel   Karsten Wemheuer
• [asterisk-users] Asterisk reject SIP INTITE from different sourceports   David White
• [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT   Ben Winslow
• [asterisk-users] other codecs   Hans Witvliet
• [asterisk-users] own Caller ID   Hans Witvliet
• [asterisk-users] Small PC to build and run Asterisk   Hans Witvliet
• [asterisk-users] Why asterisk down when inet server down?   Hans Witvliet
• [asterisk-users] Why asterisk down when inet server down?   Hans Witvliet
• [asterisk-users] Why asterisk down when inet server down?   Hans Witvliet
• [asterisk-users] Why asterisk down when inet server down?   Hans Witvliet
• [asterisk-users] DAHDI volume   Greg Woods
• [asterisk-users] DAHDI volume   Greg Woods
• [asterisk-users] SkypeKit   Jay R. Worthington
• [asterisk-users] "Hidden" memory leak   CHEN XUEQIN
• [asterisk-users] a2billing for residential voip usage   Vahan Yerkanian
• [asterisk-users] Controlling calls   Adil Zaaraoui
• [asterisk-users] Re : Controlling calls   Adil Zaaraoui
• [asterisk-users] Re : Controlling calls   Adil Zaaraoui
• [asterisk-users] Controlling calls   Adil Zaaraoui
• [asterisk-users] Re : Controlling calls   Adil Zaaraoui
• [asterisk-users] Controlling calls   Adil Zaaraoui
• [asterisk-users] Re : Controlling calls   Adil Zaaraoui
• [asterisk-users] Re : Re : Controlling calls   Adil Zaaraoui
• [asterisk-users] Re : Re : Re : Controlling calls   Adil Zaaraoui
• [asterisk-users] Re : Re : Re : Re : Controlling calls   Adil Zaaraoui
• [asterisk-users] Asterisk Call routing problem   Adil Zaaraoui
• [asterisk-users] Re : Asterisk Call routing problem   Adil Zaaraoui
• [asterisk-users] Re : Re : Asterisk Call routing problem   Adil Zaaraoui
• [asterisk-users] Asterisk hangs up for some calls   Adil Zaaraoui
• [asterisk-users] Re : Asterisk hangs up for some calls   Adil Zaaraoui
• [asterisk-users] Peer status unreachable/problem in IAX packets   Adil Zaaraoui
• [asterisk-users] no sound between extensions   Zeeshan Zakaria
• [asterisk-users] no sound between extensions   Zeeshan Zakaria
• [asterisk-users] no sound between extensions   Zeeshan Zakaria
• [asterisk-users] HElP me I am a beginner   Zeeshan Zakaria
• [asterisk-users] HElP me I am a beginner   Zeeshan Zakaria
• [asterisk-users] HElP me I am a beginner   Zeeshan Zakaria
• [asterisk-users] how to get call duration   Zeeshan Zakaria
• [asterisk-users] Is this failed Asterisk setup typical?   Zeeshan Zakaria
• [asterisk-users] how to get call duration   Zeeshan Zakaria
• [asterisk-users] <UsingWaitorPlaybackinhextension at gmail.com>,   Zeeshan Zakaria
• [asterisk-users] Create dialplan restrictions based on the IP Address of the SIP Client?   Zeeshan Zakaria
• [asterisk-users] how to get call duration   Zeeshan Zakaria
• [asterisk-users] <UsingWaitorPlaybackinhextension at gmail.com>,   Zeeshan Zakaria
• [asterisk-users] <UsingWaitorPlaybackinhextension at gmail.com>,   Zeeshan Zakaria
• [asterisk-users] Deleting extension makes it usable?   Zeeshan Zakaria
• [asterisk-users] reloading realtime sip peers   Zeeshan Zakaria
• [asterisk-users] reloading realtime sip peers   Zeeshan Zakaria
• [asterisk-users] Deleting extension makes it usable?   Zeeshan Zakaria
• [asterisk-users] reloading realtime sip peers   Zeeshan Zakaria
• [asterisk-users] own Caller ID   Zeeshan Zakaria
• [asterisk-users] Out of Office   Zeeshan Zakaria
• [asterisk-users] Out of Office   Zeeshan Zakaria
• [asterisk-users] Out of Office   Zeeshan Zakaria
• [asterisk-users] Out of Office   Zeeshan Zakaria
• [asterisk-users] Out of Office   Zeeshan Zakaria
• [asterisk-users] Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?   Zeeshan Zakaria
• [asterisk-users] asterisk log problem   Zeeshan Zakaria
• [asterisk-users] How to disable day light saving on Snom 360 phones?   Zeeshan Zakaria
• [asterisk-users] How to disable day light saving on Snom 360 phones?   Zeeshan Zakaria
• [asterisk-users] Configure Voicemail for Large Systems   Zeeshan Zakaria
• [asterisk-users] Configure Voicemail for Large Systems   Zeeshan Zakaria
• [asterisk-users] can't seem to register, status unmonitored   Zeeshan Zakaria
• [asterisk-users] Cutting the CallerID(RDNIS)   Zeeshan Zakaria
• [asterisk-users] Cutting the CallerID(RDNIS)   Zeeshan Zakaria
• [asterisk-users] Cutting the CallerID(RDNIS)   Zeeshan Zakaria
• [asterisk-users] read data from file system and put in a variable   Zeeshan Zakaria
• [asterisk-users] IVR extension dialing error   Zeeshan Zakaria
• [asterisk-users] Music on Hold problema   Zeeshan Zakaria
• [asterisk-users] Why asterisk down when inet server down?   Zeeshan Zakaria
• [asterisk-users] Why asterisk down when inet server down?   Zeeshan Zakaria
• [asterisk-users] How to disable day light saving on Snom 360 phones?   Zeeshan Zakaria
• [asterisk-users] Voicemail ODBC   Zeeshan Zakaria
• [asterisk-users] applicationmap and ChannelRedirect   Zeeshan Zakaria
• [asterisk-users] How to find a single call in logs   Zeeshan Zakaria
• [asterisk-users] Create Conference and exit myself   Zeeshan Zakaria
• [asterisk-users] Generate cdr on Hangup   Zeeshan Zakaria
• [asterisk-users] Local channel usage   Zeeshan Zakaria
• [asterisk-users] when to use e1/t1 card?   Zeeshan Zakaria
• [asterisk-users] Local channel usage   Zeeshan Zakaria
• [asterisk-users] Call file structure and syntax   Zeeshan Zakaria
• [asterisk-users] FW: Music on Hold problema   Zeeshan Zakaria
• [asterisk-users] Asterisk + E1 card   Zeeshan Zakaria
• [asterisk-users] Asterisk + E1 card   Zeeshan Zakaria
• [asterisk-users] Very strange registration problem   Zeeshan Zakaria
• [asterisk-users] Astersik can not detect DTMF key   Zeeshan Zakaria
• [asterisk-users] Very strange registration problem   Zeeshan Zakaria
• [asterisk-users] Dialplan for conference   Zeeshan Zakaria
• [asterisk-users] Dialplan for conference   Zeeshan Zakaria
• [asterisk-users] Astersik can not detect DTMF key   Zeeshan Zakaria
• [asterisk-users] restricting sip users to a certain useragent   Zeeshan Zakaria
• [asterisk-users] peer IP address in CDR   Zeeshan Zakaria
• [asterisk-users] cmd Authenticate   Zeeshan Zakaria
• [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?   Zeeshan Zakaria
• [asterisk-users] peer IP address in CDR   Zeeshan Zakaria
• [asterisk-users] peer IP address in CDR   Zeeshan Zakaria
• [asterisk-users] Find a way to block brute force attacks.   Zeeshan Zakaria
• [asterisk-users] Asterisk 1.6.2.8 Now Available   --[ UxBoD ]--
• [asterisk-users] Dual Atom mobo - call capacity   --[ UxBoD ]--
• [asterisk-users] Small PC to build and run Asterisk   --[ UxBoD ]--
• [asterisk-users] OT: Physical SIP phone with inbuilt VPN support   --[ UxBoD ]--
• [asterisk-users] Free SIP/DID provider   ayodele abejide
• [asterisk-users] Out of Office   doug at accessgate.net
• [asterisk-users] Out of Office   doug at accessgate.net
• [asterisk-users] Out of Office   doug at accessgate.net
• [asterisk-users] Want to retrieve the value of contact header   kamrun nahar bina
• [asterisk-users] No 2nd invite from asterisk after challenging original invite   jobrien at broadsmart.com
• [asterisk-users] problem with port 5090 registration   bruce bruce
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   bruce bruce
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   bruce bruce
• [asterisk-users] problem with port 5090 registration   bruce bruce
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   bruce bruce
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   bruce bruce
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   bruce bruce
• [asterisk-users] Deleting extension makes it usable?   bruce bruce
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   bruce bruce
• [asterisk-users] Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?   bruce bruce
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   bruce bruce
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   bruce bruce
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   bruce bruce
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   bruce bruce
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   bruce bruce
• [asterisk-users] Small PC for Asterisk appliance to support 2 x Sangoma A200 (2 x PCIe standard cards)   bruce bruce
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   bruce bruce
• [asterisk-users] How to pass variable back and forth from dialplan to php file?   bruce bruce
• [asterisk-users] asterisk issue   bruce bruce
• [asterisk-users] Using SetVar with System() is it possible?   bruce bruce
• [asterisk-users] Using SetVar with System() is it possible?   bruce bruce
• [asterisk-users] Deleting some of the CDR data - How to do it safely?   bruce bruce
• [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?   bruce bruce
• [asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver   bruce bruce
• [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?   bruce bruce
• [asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver   bruce bruce
• [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?   bruce bruce
• [asterisk-users] Xorocom Missing files...where to get it? astribank_upgrade   bruce bruce
• [asterisk-users] Xorocom Missing files...where to get it? astribank_upgrade   bruce bruce
• [asterisk-users] I look ARI (Asterisk Recording Interface)   bruce bruce
• [asterisk-users] A lot of : doing dnsmgr_lookup for - Asterisk installed from YUM   bruce bruce
• [asterisk-users] What are the guts of AsteriskNOW and how it compares to other popular flavors available?   bruce bruce
• [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?   bruce bruce
• [asterisk-users] Anyone can share their config file for Cisco phone please?   bruce bruce
• [asterisk-users] Anyone can share their config file for Cisco phone please?   bruce bruce
• [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel   covici at ccs.covici.com
• [asterisk-users] problem with inserting records into cdr   covici at ccs.covici.com
• [asterisk-users] problem with inserting records into cdr   covici at ccs.covici.com
• [asterisk-users] problem with inserting records into cdr   covici at ccs.covici.com
• [asterisk-users] Asterisk T.38 Gateway code testing   marek cervenka
• [asterisk-users] UDPTL T38 via NAT   marek cervenka
• [asterisk-users] UDPTL T38 via NAT   marek cervenka
• [asterisk-users] Switchboad like application   voip crazy
• [asterisk-users] sip server   mohamed daif
• [asterisk-users] sip server   mohamed daif
• [asterisk-users] joining 2 conferences together   daniel at danielknoll.de
• [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel   sean darcy
• [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel   sean darcy
• [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel   sean darcy
• [asterisk-users] How to stop intruder from registering sip?   sean darcy
• [asterisk-users] How to stop intruder from registering sip?   sean darcy
• [asterisk-users] How to stop intruder from registering sip?   sean darcy
• [asterisk-users] How to stop intruder from registering sip?   sean darcy
• [asterisk-users] How to stop intruder from registering sip?   sean darcy
• [asterisk-users] How to stop intruder from registering sip?   sean darcy
• [asterisk-users] Why asterisk down when inet server down?   sean darcy
• [asterisk-users] Why asterisk down when inet server down?   sean darcy
• [asterisk-users] Why asterisk down when inet server down?   sean darcy
• [asterisk-users] Why asterisk down when inet server down?   sean darcy
• [asterisk-users] Why asterisk down when inet server down?   sean darcy
• [asterisk-users] Why asterisk down when inet server down?   sean darcy
• [asterisk-users] Why asterisk down when inet server down?   sean darcy
• [asterisk-users] Why asterisk down when inet server down?   sean darcy
• [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk   dotnetdub
• [asterisk-users] How to stop intruder from registering sip?   dotnetdub
• [asterisk-users] Slightly OT: Cisco SPA525G and network errors   dotnetdub
• [asterisk-users] CDRs not getting generated on Free PBX   dotnetdub
• [asterisk-users] How to tell if a dropped call is my fault   dotnetdub
• [asterisk-users] SPA8000 outbound CID problem   dotnetdub
• [asterisk-users] Configure WAN Phone   dotnetdub
• [asterisk-users] Echo problem in VoIP-calls   dotnetdub
• [asterisk-users] asterisk issue   salaheddine elharit
• [asterisk-users] asterisk issue   salaheddine elharit
• [asterisk-users] asterisk issue   salaheddine elharit
• [asterisk-users] asterisk issue   salaheddine elharit
• [asterisk-users] timeout problem with basic conf   iscario at free.fr
• [asterisk-users] Muti Asterisk   michel freiha
• [asterisk-users] Muti Asterisk   michel freiha
• [asterisk-users] no ring back 180 with SDP   dave george
• [asterisk-users] no ring back 180 with SDP   dave george
• [asterisk-users] no ring back 180 with SDP   dave george
• [asterisk-users] no ring back 180 with SDP   dave george
• [asterisk-users] VAD and cRTP, any thing else?   bilal ghayyad
• [asterisk-users] Asterisk distribution for a Call Center   adriano ghezzi
• [asterisk-users] Astersik can not detect DTMF key   huu giang
• [asterisk-users] Astersik can not detect DTMF key   huu giang
• [asterisk-users] Astersik can not detect DTMF key   huu giang
• [asterisk-users] BerkeleyTIP Join June Global Free SW HW Culture Mtgs via VOIP or in Berkeley   giovanni_re
• [asterisk-users] Hot to configure trunk in asterisk with a2billing.   gokulakrishnan
• [asterisk-users] Problem in establish call from a2billing users.   gokulakrishnan
• [asterisk-users] no sound between extensions   taimur hasan
• [asterisk-users] HElP me I am a beginner   taimur hasan
• [asterisk-users] HElP me I am a beginner   taimur hasan
• [asterisk-users] own Caller ID   taimur hasan
• [asterisk-users] How do you hangup a call without terminating your session?   hugolivude
• [asterisk-users] Press twice *   hugolivude
• [asterisk-users] How do you hangup a call without terminating your session?   hugolivude
• [asterisk-users] How do you hangup a call without terminating your session?   hugolivude
• [asterisk-users] How do you hangup a call without terminating your session?   hugolivude
• [asterisk-users] How do you hangup a call without terminating your session?   hugolivude
• [asterisk-users] How do you hangup a call without terminating your session?   hugolivude
• [asterisk-users] How do you hangup a call without terminating your session?   hugolivude
• [asterisk-users] How to use one single IP as origination   joea at j4computers.com
• [asterisk-users] numbers   mattias jonsson
• [asterisk-users] How to configure key sequence in features.conf.   louis liu
• [asterisk-users] Asterisk +Dahdi does not work with BRI NT   liuxin
• [asterisk-users] Asterisk + Dahdi does not work with BRI NT mode   liuxin
• [asterisk-users] Asterisk + Dahdi does not work with BRI NT mode   liuxin
• [asterisk-users] Asterisk + Dahdi does not work with BRI NT mode   liuxin
• [asterisk-users] PRI span problem - no D channel   liuxin
• [asterisk-users] PRI span problem - no D channel   liuxin
• [asterisk-users] PRI span problem - no D channel   liuxin
• [asterisk-users] Video not working with PortSIP SDK   list mail
• [asterisk-users] Delay in IVR   mike mosier
• [asterisk-users] Codec negotiation   mike mosier
• [asterisk-users] Dual Atom mobo - call capacity   mgraves at mstvp.com
• [asterisk-users] 1.6 how to use groupcount and counteronpeer in queues to avoid ringinuse   nik600
• [asterisk-users] [compat] section in asterisk.conf : compatibility with pipe delimiter   nik600
• [asterisk-users] understand which asterisk thread is consuming CPU   nik600
• [asterisk-users] call dialing   niksinghania
• [asterisk-users] asterisk appache issue   pankaj pandey
• [asterisk-users] asterisk appache issue   pankaj pandey
• [asterisk-users] a2billing for residential voip usage   ram
• [asterisk-users] chan_dahdi compilation with embedded   garge rama
• [asterisk-users] asterisk sip trunk configure   garge rama
• [asterisk-users] asterisk log problem   das sandesh
• [asterisk-users] asterisk log problem   das sandesh
• [asterisk-users] Call drops on group paging asterisk - 1.4.22.1   das sandesh
• [asterisk-users] Call drops on group paging asterisk - 1.4.22.1   das sandesh
• [asterisk-users] HElP me I am a beginner   prashant shrestha
• [asterisk-users] PSTN-IVR call   nikhil singhania
• [asterisk-users] asterisk registration   nikhil singhania
• [asterisk-users] Fwd: asterisk registration   nikhil singhania
• [asterisk-users] calling peer from server   nikhil singhania
• [asterisk-users] calling peer from server   nikhil singhania
• [asterisk-users] can't seem to register, status unmonitored   nikhil singhania
• [asterisk-users] can't seem to register, status unmonitored   nikhil singhania
• [asterisk-users] Fwd: can't seem to register, status unmonitored   nikhil singhania
• [asterisk-users] writing echo in inbound file   nikhil singhania
• [asterisk-users] can't seem to register, status unmonitored   nikhil singhania
• [asterisk-users] calling machine over sip   nikhil singhania
• [asterisk-users] device or sound card busy   nikhil singhania
• [asterisk-users] playing file when using call file in /var/spool/asterisk/outgoing in asterisk   nikhil singhania
• [asterisk-users] using call file   nikhil singhania
• [asterisk-users] storing DTMF inputs   nikhil singhania
• [asterisk-users] WARNING message when play   equis software
• [asterisk-users] WARNING message when play   equis software
• [asterisk-users] WARNING message when play   equis software
• [asterisk-users] WARNING message when play   equis software
• [asterisk-users] Time variables in system application   khalid touati
• [asterisk-users] Time variables in system application   khalid touati
• [asterisk-users] Time variables in system application   khalid touati

Last message date: Wed Jun 30 23:54:38 CDT 2010
Archived on: Wed Jun 30 23:54:47 CDT 2010

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