April 2011 Archives by date

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Starting: Fri Apr 1 01:18:11 CDT 2011
Ending: Sat Apr 30 22:31:56 CDT 2011
Messages: 1145

• [asterisk-users] ** to disconnect and make a new call   DHAVAL INDRODIYA
• [asterisk-users] OneAPI / ParlayX   Necati Demir
• [asterisk-users] Hold problem with Queue   Elensarde
• [asterisk-users] The SIP channel driver - I'm giving up.   Olle E. Johansson
• [asterisk-users] The SIP channel driver - I'm giving up.   Tony Mountifield
• [asterisk-users] ** to disconnect and make a new call   Abid Saleem
• [asterisk-users] ** to disconnect and make a new call   DHAVAL INDRODIYA
• [asterisk-users] Incoming SRTP call not working with Bria iPhone Edition   Alexis de BRUYN
• [asterisk-users] Fax   Khaled W. Chehab
• [asterisk-users] Double ## Feature to make another call during current call   Atif Razzaq
• [asterisk-users] Best Scripting Language   Gopalakrishnan A.N
• [asterisk-users] Best Scripting Language   mahesh katta
• [asterisk-users] Best Scripting Language   Gopalakrishnan A.N
• [asterisk-users] Hold problem with Queue   Satish Patel
• [asterisk-users] Best Scripting Language   Satish Patel
• [asterisk-users] Best Scripting Language   Roger Burton West
• [asterisk-users] Vinicius Bossle Fagundes enviou-lhe o seguinte abaixo-assinado.   Petição Pública
• [asterisk-users] Fax   David Backeberg
• [asterisk-users] Best Scripting Language   Khaled W. Chehab
• [asterisk-users] BRI detection   mahesh katta
• [asterisk-users] Hold problem with Queue   Elensarde
• [asterisk-users] Best Scripting Language   Satish Patel
• [asterisk-users] Best Scripting Language   Danny Nicholas
• [asterisk-users] Best Scripting Language   Mark Deneen
• [asterisk-users] Hold problem with Queue   Satish Patel
• [asterisk-users] Best Scripting Language   Gopalakrishnan A.N
• [asterisk-users] Can gtalk.conf work with multiple GoogleVoice numbers?   Chris Gentle
• [asterisk-users] Digium launches flying phone-phone   Tim Panton
• [asterisk-users] Best Scripting Language   Tim Panton
• [asterisk-users] call parking issues in asterisk 1.6.2.16.2   vip killa
• [asterisk-users] call parking issues in asterisk 1.6.2.16.2   Mike
• [asterisk-users] Best Scripting Language   Gopalakrishnan A.N
• [asterisk-users] Android tablet voip?   Dean Collins
• [asterisk-users] Best Scripting Language   Tim Panton
• [asterisk-users] Android tablet voip?   jon pounder
• [asterisk-users] Best Scripting Language   Goke M Aruna
• [asterisk-users] Best Scripting Language   Danny Nicholas
• [asterisk-users] codec_dahdi find_transcoders: Failed to open /dev/dahdi/transcode   satish patel
• [asterisk-users] codec_dahdi find_transcoders: Failed to open /dev/dahdi/transcode   Shaun Ruffell
• [asterisk-users] codec_dahdi find_transcoders: Failed to open /dev/dahdi/transcode   satish patel
• [asterisk-users] Polycom 501 alternate   satish patel
• [asterisk-users] Polycom 501 alternate   Warren Selby
• [asterisk-users] Polycom 501 alternate   Satish Patel
• [asterisk-users] Problem getting TDM400P clone card to go off-hook and dial   Stewart Loving-Gibbard
• [asterisk-users] automixmon output file location and exec command options   Bruce McAlister
• [asterisk-users] Registration from '"000000" x 1000   Jonas Kellens
• [asterisk-users] Registration from '"000000" x 1000   Steve Davies
• [asterisk-users] tarnsfer automatically   salaheddine elharit
• [asterisk-users] Problem getting TDM400P clone card to go off-hook and dial   Stewart Loving-Gibbard
• [asterisk-users] Registration from '"000000" x 1000   Jonas Kellens
• [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?   Rafael Bermúdez
• [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?   John Kiniston
• [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?   Rafael Bermúdez
• [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?   Steve Edwards
• [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?   Steve Edwards
• [asterisk-users] Registration from '"000000" x 1000   Terry Brummell
• [asterisk-users] Best Scripting Language   Hans Witvliet
• [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?   Rafael Bermúdez
• [asterisk-users] Top posting - there is no rule.   Alec Davis
• [asterisk-users] Top posting - there is no rule.   Warren Selby
• [asterisk-users] Top posting - there is no rule.   Alec Davis
• [asterisk-users] BRI detection   Tzafrir Cohen
• [asterisk-users] [DIGIUM FAX] HANGUP problem   cyril paris
• [asterisk-users] The SIP channel driver - I'm giving up.   Tzafrir Cohen
• [asterisk-users] hello   ALAEDDINE abbech
• [asterisk-users] Asterisk 1.6 => No sound/voice when i redirect the call   Olivier CALVANO
• [asterisk-users] Asterisk 1.6 => No sound/voice when i redirect the call   Mark Murawski
• [asterisk-users] Asterisk 1.6 => No sound/voice when i redirect the call   Olivier CALVANO
• [asterisk-users] From 1.4 to 1.8: stdexten issue   Mathieu Chouquet-Stringer
• [asterisk-users] Asterisk 1.6 => No sound/voice when i redirect the call   Mark Murawski
• [asterisk-users] From 1.4 to 1.8: stdexten issue   Benny Amorsen
• [asterisk-users] hello   Bradley D. Thornton
• [asterisk-users] hello   Doug Lytle
• [asterisk-users] Best Scripting Language   Thorsten Göllner
• [asterisk-users] Hold problem with Queue   Bertrand Miquel
• [asterisk-users] SIP channel able to add codecs once up and running?   Steve Davies
• [asterisk-users] Finding out asterisk settings from console   Ishfaq Malik
• [asterisk-users] BRI detection   Andrew Thomas
• [asterisk-users] CDR fields not being written from "h" extension after "Dial" command completes.   Arjan Kroon | Mobillion
• [asterisk-users] Asterisk crashes on high IO load   Maximilian Grobecker
• [asterisk-users] Load Asterisk Module with parameters?   Meckel
• [asterisk-users] CDR fields not being written from "h" extension after "Dial" command completes.   Ishfaq Malik
• [asterisk-users] Load Asterisk Module with parameters?   Kevin P. Fleming
• [asterisk-users] CDR fields not being written from "h" extension after "Dial" command completes.   Arjan Kroon | Mobillion
• [asterisk-users] call forwarding   salaheddine elharit
• [asterisk-users] Asterisk crashes on high IO load   Thorsten Göllner
• [asterisk-users] Dialplan matching   Asterisk User
• [asterisk-users] Load Asterisk Module with parameters?   Ulrich Meckel
• [asterisk-users] how to check if the call is using t38 except in the sip packets   Israel Gottlieb
• [asterisk-users] Dialplan matching   Steve Murphy
• [asterisk-users] SIP register and contact header   Jonas Kellens
• [asterisk-users] Asterisk crashes on high IO load   Maximilian Grobecker
• [asterisk-users] Can gtalk.conf work with multiple GoogleVoice numbers?   Jamie A. Stapleton
• [asterisk-users] Asterisk crashes on high IO load   Nic Colledge
• [asterisk-users] Can gtalk.conf work with multiple GoogleVoice numbers?   Chris Gentle
• [asterisk-users] call forwarding   Rizwan Hisham
• [asterisk-users] Dialplan matching   A J Stiles
• [asterisk-users] call-limit bypass   Rizwan Hisham
• [asterisk-users] DAHDI-Linux 2.4.1.1 Released   Asterisk Development Team
• [asterisk-users] MeetMe headache   D. Rick Anderson
• [asterisk-users] From 1.4 to 1.8: stdexten issue   Mathieu Chouquet-Stringer
• [asterisk-users] Dialplan matching   Tilghman Lesher
• [asterisk-users] Asterisk 1.6 => No sound/voice when i redirect the call   Olivier CALVANO
• [asterisk-users] call-limit bypass   Bryant Zimmerman
• [asterisk-users] dialplan is not finding my number asterisk 1.8.3   Jerry Geis
• [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit   satish patel
• [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit   Fellipe Paes
• [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit   Mark Deneen
• [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit   satish patel
• [asterisk-users] SIP register and contact header   Andreas Sikkema
• [asterisk-users] Read VoiceMail direct   satish patel
• [asterisk-users] Read VoiceMail direct   bakko
• [asterisk-users] Read VoiceMail direct   satish patel
• [asterisk-users] CDR fields not being written from "h" extension after "Dial" command completes.   Tilghman Lesher
• [asterisk-users] dialplan is not finding my number asterisk 1.8.3   Warren Selby
• [asterisk-users] dialplan is not finding my number asterisk 1.8.3   Paul Belanger
• [asterisk-users] Number Conversion   Flavio Miranda
• [asterisk-users] Number Conversion   Matt Riddell
• [asterisk-users] Number Conversion   Flavio Miranda
• [asterisk-users] dialplan is not finding my number asterisk 1.8.3   Mark Deneen
• [asterisk-users] dialplan is not finding my number asterisk1.8.3   bayardo.sanchez at gmail.com
• [asterisk-users] Issues with Digum Repos / AsteriskNOW & Bad Packages   Andrew Joakimsen
• [asterisk-users] how to check if the call is using t38 except in the sip packets   Larry Moore
• [asterisk-users] CDR fields not being written from "h" extension after "Dial" command completes.   Arjan Kroon | Mobillion
• [asterisk-users] Asterisk => Request a Code   Olivier CALVANO
• [asterisk-users] CDR fields not being written from "h" extension after "Dial" command completes.   Arjan Kroon | Mobillion
• [asterisk-users] Asterisk crashes on high IO load   Thorsten Göllner
• [asterisk-users] CDR fields not being written from "h" extension after "Dial" command completes.   Arjan Kroon | Mobillion
• [asterisk-users] Asterisk 1.8 and new the command: exten => _X., 4, Wait, 2   bilal ghayyad
• [asterisk-users] Asterisk 1.8 and new the command: exten => _X., 4, Wait, 2   isrlgb at gmail.com
• [asterisk-users] Asterisk => Request a Code   Robles Román, José Miguel
• [asterisk-users] Asterisk 1.8 and new the command: exten => _X., 4, Wait, 2   isrlgb at gmail.com
• [asterisk-users] call-limit bypass   Rizwan Hisham
• [asterisk-users] dialplan is not finding my number asterisk 1.8.3   Jerry Geis
• [asterisk-users] Asterisk crashes on high IO load   Maximilian Grobecker
• [asterisk-users] dialplan is not finding my number asterisk 1.8.3   Steve Murphy
• [asterisk-users] dialplan is not finding my number asterisk 1.8.3   Jerry Geis
• [asterisk-users] dialplan is not finding my number asterisk 1.8.3   Jerry Geis
• [asterisk-users] Iptables configuration to handle brute force registrations?   Gilles
• [asterisk-users] Vestec for Asterisk   Lee Archer
• [asterisk-users] Vestec for Asterisk   Danny Nicholas
• [asterisk-users] Vestec for Asterisk   Stelios Koroneos
• [asterisk-users] Iptables configuration to handle brute force registrations?   Steve Edwards
• [asterisk-users] allpage issu on asterisk 1.8.3.x   satish patel
• [asterisk-users] dialplan is not finding my number asterisk 1.8.3   Steve Murphy
• [asterisk-users] allpage issu on asterisk 1.8.3.x   satish patel
• [asterisk-users] agi voicemail callback   vip killa
• [asterisk-users] call forwarding   salaheddine elharit
• [asterisk-users] agi create mailbox   vip killa
• [asterisk-users] agi voicemail callback   Doug Lytle
• [asterisk-users] agi voicemail callback   Steve Edwards
• [asterisk-users] agi create mailbox   Steve Edwards
• [asterisk-users] asterisk hints   satish patel
• [asterisk-users] asterisk hints   Danny Nicholas
• [asterisk-users] asterisk hints   satish patel
• [asterisk-users] asterisk hints   Danny Nicholas
• [asterisk-users] Iptables configuration to handle brute, force registrations?   Bill Michaelson
• [asterisk-users] asterisk hints   satish patel
• [asterisk-users] Iptables configuration to handle brute, force registrations?   Sherwood McGowan
• [asterisk-users] dahdi and linux-2.6.38   covici at ccs.covici.com
• [asterisk-users] dahdi and linux-2.6.38   Danny Nicholas
• [asterisk-users] dahdi and linux-2.6.38   Shaun Ruffell
• [asterisk-users] dahdi and linux-2.6.38   Shaun Ruffell
• [asterisk-users] dahdi and linux-2.6.38   Shaun Ruffell
• [asterisk-users] dahdi and linux-2.6.38   Danny Nicholas
• [asterisk-users] Iptables configuration to handle brute, force registrations?   Steve Edwards
• [asterisk-users] Iptables configuration to handle brute, force registrations?   Sherwood McGowan
• [asterisk-users] dahdi and linux-2.6.38   covici at ccs.covici.com
• [asterisk-users] dahdi and linux-2.6.38   Danny Nicholas
• [asterisk-users] dahdi and linux-2.6.38   covici at ccs.covici.com
• [asterisk-users] dahdi and linux-2.6.38   Shaun Ruffell
• [asterisk-users] Iptables configuration to handle brute, force registrations?   Steve Edwards
• [asterisk-users] dahdi and linux-2.6.38   Danny Nicholas
• [asterisk-users] asterisk-users Digest, Vol 81, Issue 12   Bill Michaelson
• [asterisk-users] dahdi and linux-2.6.38   Shaun Ruffell
• [asterisk-users] dahdi and linux-2.6.38   covici at ccs.covici.com
• [asterisk-users] dahdi and linux-2.6.38   Kevin P. Fleming
• [asterisk-users] dahdi and linux-2.6.38   Danny Nicholas
• [asterisk-users] spa8000 t38 faxing   isrlgb at gmail.com
• [asterisk-users] minmessage / maxsilence in voicemail.conf   Hose
• [asterisk-users] IAS trunk error AES encryption disabled. Install OpenSSL.   satish patel
• [asterisk-users] Iptables configuration to handle brute, force registrations?   Warren Selby
• [asterisk-users] Iptables configuration to handle brute, force registrations?   Paul Dugas
• [asterisk-users] Iptables configuration to handle brute, force registrations?   Danny Nicholas
• [asterisk-users] Asterisk 1.8.3   Bryant Zimmerman
• [asterisk-users] Asterisk crashes on high IO load   Matt Riddell
• [asterisk-users] Asterisk 1.8.3   Tilghman Lesher
• [asterisk-users] Iptables configuration to handle brute, force registrations?   Sherwood McGowan
• [asterisk-users] asterisk-users Digest, Vol 81, Issue 12   Sherwood McGowan
• [asterisk-users] agi voicemail callback   Thorsten Göllner
• [asterisk-users] Call duration problem or maybe calls not hanging up problem   Ira
• [asterisk-users] Asterisk 1.8.3   Ishfaq Malik
• [asterisk-users] Asterisk 1.8.3   Michel Verbraak
• [asterisk-users] Iptables configuration to handle brute force registrations?   Gordon Henderson
• [asterisk-users] MeetMe headache   DHAVAL INDRODIYA
• [asterisk-users] Call recording - methodology   Silver Thorne
• [asterisk-users] Call recording - methodology   Steven Howes
• [asterisk-users] Call recording - methodology   Sherwood McGowan
• [asterisk-users] Call recording - methodology   Dan Journo
• [asterisk-users] agi voicemail callback   vip killa
• [asterisk-users] agi voicemail callback   DHAVAL INDRODIYA
• [asterisk-users] Read VoiceMail direct   DHAVAL INDRODIYA
• [asterisk-users] spa8000 t38 faxing   Larry Moore
• [asterisk-users] Fax   Bert Van Kets
• [asterisk-users] agi voicemail callback   Steve Edwards
• [asterisk-users] Call recording - methodology   Danny Nicholas
• [asterisk-users] asterisk hints   Stefan Schmidt
• [asterisk-users] BRI Configuration help me   mahesh katta
• [asterisk-users] agi voicemail callback   vip killa
• [asterisk-users] agi voicemail callback   Danny Nicholas
• [asterisk-users] agi voicemail callback   vip killa
• [asterisk-users] agi voicemail callback   Danny Nicholas
• [asterisk-users] Question About Codecs   Jon Farmer
• [asterisk-users] asterisk hints   satish patel
• [asterisk-users] asterisk hints   Danny Nicholas
• [asterisk-users] asterisk hints   satish patel
• [asterisk-users] asterisk hints   Danny Nicholas
• [asterisk-users] BRI Configuration help me   Tzafrir Cohen
• [asterisk-users] agi voicemail callback   vip killa
• [asterisk-users] agi voicemail callback   Danny Nicholas
• [asterisk-users] voicemail call back loop   vip killa
• [asterisk-users] asterisk hints   Olivier
• [asterisk-users] asterisk hints   Danny Nicholas
• [asterisk-users] voicemail call back loop   Steven Howes
• [asterisk-users] voicemail call back loop   vip killa
• [asterisk-users] Problems with woomera (ISDN BRI) and playback app: Dropping incompatible voice frame   Niccolò Belli
• [asterisk-users] asterisk hints   satish patel
• [asterisk-users] MWI not working on most ATAs in Asterisk 1.6.2.17   maillist at lightspeed.ca
• [asterisk-users] voicemail call back loop   Steve Edwards
• [asterisk-users] voicemail call back loop   vip killa
• [asterisk-users] IAS trunk error AES encryption disabled. Install OpenSSL.   satish patel
• [asterisk-users] voicemail call back loop   Danny Nicholas
• [asterisk-users] IAS trunk error AES encryption disabled. Install OpenSSL.   Warren Selby
• [asterisk-users] realtime mysql for 1.8   Hans Witvliet
• [asterisk-users] IAS trunk error AES encryption disabled. Install OpenSSL.   satish patel
• [asterisk-users] IAS trunk error AES encryption disabled. Install OpenSSL.   satish patel
• [asterisk-users] [SOLVED] IAX trunk error AES encryption disabled. Install OpenSSL.   satish patel
• [asterisk-users] Best Scripting Language   Mohammad Khan
• [asterisk-users] Options for DS3 to SIP   Kyle Sexton
• [asterisk-users] realtime mysql for 1.8   Paul Belanger
• [asterisk-users] realtime mysql for 1.8   Jonathan Thurman
• [asterisk-users] asterisk meetme invalid extension   satish patel
• [asterisk-users] asterisk meetme invalid extension   Danny Nicholas
• [asterisk-users] Asterisk 1.8.3   Edwin Lam
• [asterisk-users] asterisk meetme invalid extension   Steve Edwards
• [asterisk-users] asterisk meetme invalid extension   satish patel
• [asterisk-users] asterisk meetme invalid extension   Danny Nicholas
• [asterisk-users] realtime mysql for 1.8   Hans Witvliet
• [asterisk-users] Asterisk 1.8.3   Bryant Zimmerman
• [asterisk-users] Call duration problem or maybe calls not hanging up problem   Ira
• [asterisk-users] realtime mysql for 1.8   Jonathan Thurman
• [asterisk-users] Asterisk 1.8.3   Edwin Lam
• [asterisk-users] Fax   Pezhman Lali
• [asterisk-users] Iptables configuration to handle brute force registrations?   Pezhman Lali
• [asterisk-users] agi create mailbox   Pezhman Lali
• [asterisk-users] BRI Configuration help me   mahesh katta
• [asterisk-users] Asterisk Avaya SIP Trunking One Way Audio   Shariq Khan
• [asterisk-users] BRI Configuration help me   Tzafrir Cohen
• [asterisk-users] asterisk SIP MESSAGE method support   Deka, Rajib IN MAA SL
• [asterisk-users] Iptables configuration to handle brute force registrations?   Gilles
• [asterisk-users] BRI Configuration help me   mahesh katta
• [asterisk-users] Iptables configuration to handle brute, force registrations?   Gilles
• [asterisk-users] Compiling asterisk using NDK build   Nikhil
• [asterisk-users] asterisk SIP MESSAGE method support   Olivier
• [asterisk-users] BRI Configuration help me   Tzafrir Cohen
• [asterisk-users] Compiling asterisk using NDK build   Tzafrir Cohen
• [asterisk-users] BRI Configuration help me   mahesh katta
• [asterisk-users] Trunk form asterisk1 to asterisk2 fails   GiGi
• [asterisk-users] Asterisk 1.8.3   Satish Patel
• [asterisk-users] Asterisk 1.8.3   Bryant Zimmerman
• [asterisk-users] BRI Configuration help me   mahesh katta
• [asterisk-users] Asterisk 1.8.3   Satish Patel
• [asterisk-users] Asterisk 1.8.3   Ishfaq Malik
• [asterisk-users] Asterisk 1.8.3   Satish Patel
• [asterisk-users] Asterisk 1.8.3   Bryant Zimmerman
• [asterisk-users] Asterisk 1.8.3   Paul Belanger
• [asterisk-users] Asterisk 1.8.3   Satish Patel
• [asterisk-users] asterisk SIP MESSAGE method support   Deka, Rajib IN MAA SL
• [asterisk-users] AgentCallbackLogin slow in Asterisk 1.4   Eduardo Leones
• [asterisk-users] Asterisk 1.8.3   Chris Owen
• [asterisk-users] asterisk SIP MESSAGE method support   Steven Howes
• [asterisk-users] Asterisk 1.8.3   --[ UxBoD ]--
• [asterisk-users] asterisk SIP MESSAGE method support   Deka, Rajib IN MAA SL
• [asterisk-users] asterisk SIP MESSAGE method support   Paul Belanger
• [asterisk-users] Asterisk 1.8.x Skips DTMF Digits on a First DAHDI Initiated Call   Vladimir Mikhelson
• [asterisk-users] Asterisk 1.8.3   Olivier
• [asterisk-users] Asterisk 1.8.3   Danny Nicholas
• [asterisk-users] Asterisk 1.8.3   Olivier
• [asterisk-users] Asterisk 1.8.3   Paul Belanger
• [asterisk-users] No ringback even though progressinband=yes is set   Douglas Mortensen
• [asterisk-users] Asterisk 1.8.3   Leif Madsen
• [asterisk-users] MOH on DAHDI PRI Channels   Shariq Khan
• [asterisk-users] MOH on DAHDI PRI Channels   Danny Nicholas
• [asterisk-users] MOH on DAHDI PRI Channels   Don Kelly
• [asterisk-users] MOH on DAHDI PRI Channels   Shariq Khan
• [asterisk-users] No ringback even though progressinband=yes is set   Steve Davies
• [asterisk-users] chan_dahdi unknown dependency problem   SebA
• [asterisk-users] chan_dahdi unknown dependency problem   SebA
• [asterisk-users] chan_dahdi unknown dependency problem   SebA
• [asterisk-users] asterisk hints   Tilghman Lesher
• [asterisk-users] Question About Codecs   Tilghman Lesher
• [asterisk-users] realtime mysql for 1.8   Tilghman Lesher
• [asterisk-users] Call duration problem or maybe calls not hanging up problem   Sherwood McGowan
• [asterisk-users] No ringback even though progressinband=yes is set   Sherwood McGowan
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] asterisk login to voicemail   Danny Nicholas
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] asterisk login to voicemail   Danny Nicholas
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] asterisk login to voicemail   Danny Nicholas
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] asterisk login to voicemail   Danny Nicholas
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit   satish patel
• [asterisk-users] asterisk login to voicemail   Danny Nicholas
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] asterisk login to voicemail   Danny Nicholas
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit   Paul Dugas
• [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit   satish patel
• [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit   isrlgb at gmail.com
• [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit   satish patel
• [asterisk-users] Any way to temporarily disable a registered SIP PEER in Asterisk?   Bruce B
• [asterisk-users] Any way to temporarily disable a registered SIPPEER in Asterisk?   Danny Nicholas
• [asterisk-users] Occasional call from "asterisk"   Brian Henning
• [asterisk-users] No ringback even though progressinband=yes is set   Douglas Mortensen
• [asterisk-users] No ringback even though progressinband=yes is set   Douglas Mortensen
• [asterisk-users] No ringback even though progressinband=yes is set   Sherwood McGowan
• [asterisk-users] asterisk login to voicemail   Dan Journo
• [asterisk-users] Occasional call from "asterisk"   Cary Fitch
• [asterisk-users] Occasional call from "asterisk"   Warren Selby
• [asterisk-users] Occasional call from "asterisk"   Bruce B
• [asterisk-users] Asterisk Avaya SIP Trunking One Way Audio   Lyle Giese
• [asterisk-users] Asterisk Avaya SIP Trunking One Way Audio   Skyler
• [asterisk-users] MOH not working   virendra bhati
• [asterisk-users] 488 error in T38 Gatewaying in Asterisk 1.8 with patch 13405   Robert Thomas
• [asterisk-users] CRC Zaptel.conf   salaheddine elharit
• [asterisk-users] asterisk-users Digest, Vol 81, Issue 21   Deka, Rajib IN MAA SL
• [asterisk-users] Variable inheritance with dialplan command Originate   Naomi Rosenberg
• [asterisk-users] Asterisk Avaya SIP Trunking One Way Audio   Shariq Khan
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] asterisk login to voicemail   Dan Journo
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] asterisk login to voicemail   Dan Journo
• [asterisk-users] User registration failure bug ?   Axelle
• [asterisk-users] Maniuplate callerID based off of callerID   Louis Carreiro
• [asterisk-users] MOH not working   Warren Selby
• [asterisk-users] asterisk login to voicemail   Satish Patel
• [asterisk-users] Maniuplate callerID based off of callerID   Michel Verbraak
• [asterisk-users] IAX2/0.0.29.199   satish patel
• [asterisk-users] Any PHP Ming + for Asterisk guides, tutorial, how-to anywhere?   Bruce B
• [asterisk-users] IAX2/0.0.29.199   Paul Belanger
• [asterisk-users] Maniuplate callerID based off of callerID   Louis Carreiro
• [asterisk-users] IAX2/0.0.29.199   satish patel
• [asterisk-users] Asterisk 1.8.3   Bryant Zimmerman
• [asterisk-users] Variable inheritance with dialplan command Originate   Sherwood McGowan
• [asterisk-users] IAX2/0.0.29.199   satish patel
• [asterisk-users] Variable inheritance with dialplan command Originate   Naomi Rosenberg
• [asterisk-users] CRC Zaptel.conf   Shaun Ruffell
• [asterisk-users] Variable inheritance with dialplan command Originate   Jim Dickenson
• [asterisk-users] Variable inheritance with dialplan command Originate   Sherwood McGowan
• [asterisk-users] Variable inheritance with dialplan command Originate   Sherwood McGowan
• [asterisk-users] IAX2/0.0.29.199   satish patel
• [asterisk-users] asterisk login to voicemail   vip killa
• [asterisk-users] asterisk login to voicemail   satish patel
• [asterisk-users] IAX2/0.0.29.199   Paul Belanger
• [asterisk-users] Call recording - methodology   Silver Thorne
• [asterisk-users] No ringback even though progressinband=yes is set   Steve Davies
• [asterisk-users] IAX2/0.0.29.199   Paul Belanger
• [asterisk-users] IAX2/0.0.29.199   satish patel
• [asterisk-users] IAX2/0.0.29.199   satish patel
• [asterisk-users] IAX2/0.0.29.199   satish patel
• [asterisk-users] IAX2/0.0.29.199   satish patel
• [asterisk-users] send voicemail to multiple emails   vip killa
• [asterisk-users] send voicemail to multiple emails   Danny Nicholas
• [asterisk-users] send voicemail to multiple emails   Sherwood McGowan
• [asterisk-users] send voicemail to multiple emails   vip killa
• [asterisk-users] send voicemail to multiple emails   Danny Nicholas
• [asterisk-users] send voicemail to multiple emails   Danny Nicholas
• [asterisk-users] send voicemail to multiple emails   Sherwood McGowan
• [asterisk-users] send voicemail to multiple emails   Sherwood McGowan
• [asterisk-users] Documentation for Asterisk AMI Events?   Bruce B
• [asterisk-users] Documentation for Asterisk AMI Events?   Paul Belanger
• [asterisk-users] Call Recording using MixMonitor - close, but would like some more words of wisdom.   Silver Thorne
• [asterisk-users] Call Recording using MixMonitor - close, but would like some more words of wisdom.   Warren Selby
• [asterisk-users] send voicemail to multiple emails   Warren Selby
• [asterisk-users] MOH not working   virendra bhati
• [asterisk-users] asterisk-users Digest, Vol 81, Issue 27   darin iv
• [asterisk-users] Documentation for Asterisk AMI Events?   Bruce B
• [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?   Bruce B
• [asterisk-users] Asterisk FOP   Flavio Miranda
• [asterisk-users] asterisk-users Digest, Vol 81, Issue 27   Steve Edwards
• [asterisk-users] Asterisk FOP   Doug Lytle
• [asterisk-users] Asterisk FOP   Flavio Miranda
• [asterisk-users] Asterisk FOP   Doug Lytle
• [asterisk-users] IAX2/0.0.29.199   Satish Patel
• [asterisk-users] Call Recording using MixMonitor - close, but would like some more words of wisdom.   Dan Journo
• [asterisk-users] Call recording - methodology   Dan Journo
• [asterisk-users] Asterisk FOP   Flavio Miranda
• [asterisk-users] send voicemail to multiple emails   vip killa
• [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]   Tzafrir Cohen
• [asterisk-users] IAX2/0.0.29.199   Tzafrir Cohen
• [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI doesn't work   Frank Tarczynski
• [asterisk-users] Call recording - methodology   Silver Thorne
• [asterisk-users] Call recording - methodology   Dan Journo
• [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI doesn't work   asterisk asterisk
• [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI doesn't work   bakko
• [asterisk-users] IAX2/0.0.29.199   Paul Belanger
• [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI doesn't work   Shaun Ruffell
• [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI doesn't work   Shaun Ruffell
• [asterisk-users] Call recording - methodology   Silver Thorne
• [asterisk-users] Asterisk as a Condo door opener/intercom   Bruce B
• [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI doesn't work   me at tdiehl.org
• [asterisk-users] IAX2/0.0.29.199   Satish Patel
• [asterisk-users] Call recording - methodology   Dan Journo
• [asterisk-users] Call recording - methodology   Silver Thorne
• [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI doesn't work   Shaun Ruffell
• [asterisk-users] Call recording - methodology   Dan Journo
• [asterisk-users] Call recording - methodology   Silver Thorne
• [asterisk-users] changing port 5060 to 5061   darin iv
• [asterisk-users] Variable stripping/removing part of string   magnus.b at inputinterior.se
• [asterisk-users] Variable stripping/removing part of string   Tilghman Lesher
• [asterisk-users] Variable stripping/removing part of string   magnus.b at inputinterior.se
• [asterisk-users] changing port 5060 to 5061   Steve Edwards
• [asterisk-users] MWI not working on most ATAs in Asterisk 1.6.2.17   Benny Amorsen
• [asterisk-users] Variable stripping/removing part of string   Tilghman Lesher
• [asterisk-users] Variable stripping/removing part of string   magnus.b at inputinterior.se
• [asterisk-users] Variable stripping/removing part of string   Tilghman Lesher
• [asterisk-users] send voicemail to multiple emails   A J Stiles
• [asterisk-users] Variable stripping/removing part of string   magnus.b at inputinterior.se
• [asterisk-users] changing port 5060 to 5061   darin iv
• [asterisk-users] Fwd: changing port 5060 to 5061   darin iv
• [asterisk-users] changing port 5060 to 5061   Steven Howes
• [asterisk-users] Unable to negotiate codec with iax   Arjan Kroon | Mobillion
• [asterisk-users] Variable stripping/removing part of string   Jeroen Eeuwes
• [asterisk-users] Variable inheritance with dialplan command Originate   Naomi Rosenberg
• [asterisk-users] update CDR fields after Queue   shayne.alone at gmail.com
• [asterisk-users] send voicemail to multiple emails   Andrew Thomas
• [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?   Pezhman Lali
• [asterisk-users] Variable stripping/removing part of string   magnus.b at inputinterior.se
• [asterisk-users] Asterisk MOH not working with Elastix asterisk 1.6.2.18   virendra bhati
• [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]   Steve Totaro
• [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]   Tzafrir Cohen
• [asterisk-users] send voicemail to multiple emails   vip killa
• [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]   Satish Patel
• [asterisk-users] Asterisk-Asterisk E1 connection   Alejandro Cabrera Obed
• [asterisk-users] Occasional call from "asterisk"   Brian Henning
• [asterisk-users] Asterisk-Asterisk E1 connection   satish patel
• [asterisk-users] Asterisk-Asterisk E1 connection   Joel Maslak
• [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]   satish patel
• [asterisk-users] Problem with E1 (ISDN) + DTMF   Luiz Gustavo Chiaretto
• [asterisk-users] Occasional call from "asterisk"   Danny Nicholas
• [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]   Tzafrir Cohen
• [asterisk-users] Occasional call from "asterisk"   Bruce B
• [asterisk-users] Require dialplan   mahesh katta
• [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?   Bruce B
• [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]   satish patel
• [asterisk-users] Asterisk codec negotiation and canreinvite=no   Effie Mouzeli
• [asterisk-users] voicemail odbc "Length is ....."   vip killa
• [asterisk-users] voicemail odbc "Length is ....."   Steven Howes
• [asterisk-users] voicemail odbc "Length is ....."   vip killa
• [asterisk-users] Asterisk kernel CONFIG_HZ=1000   satish patel
• [asterisk-users] voicemail odbc "Length is ....."   Danny Nicholas
• [asterisk-users] Asterisk kernel CONFIG_HZ=1000   Danny Nicholas
• [asterisk-users] voicemail odbc "Length is ....."   vip killa
• [asterisk-users] Asterisk kernel CONFIG_HZ=1000   satish patel
• [asterisk-users] Asterisk kernel CONFIG_HZ=1000   Danny Nicholas
• [asterisk-users] Asterisk kernel CONFIG_HZ=1000   Tzafrir Cohen
• [asterisk-users] Asterisk MOH not working with Elastix asterisk 1.6.2.18   Warren Selby
• [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?   Pezhman Lali
• [asterisk-users] send voicemail to multiple emails   Sherwood McGowan
• [asterisk-users] Asterisk MOH not working with Elastix asterisk 1.6.2.18   virendra bhati
• [asterisk-users] Asterisk MOH not working with Elastix asterisk1.6.2.18   Danny Nicholas
• [asterisk-users] Variable inheritance with dialplan command Originate   Sherwood McGowan
• [asterisk-users] Occasional call from "asterisk"   Chris Gentle
• [asterisk-users] Asterisk kernel CONFIG_HZ=1000   satish patel
• [asterisk-users] Asterisk as a Condo door opener/intercom   C F
• [asterisk-users] Unable to negotiate codec with iax   Paul Belanger
• [asterisk-users] send voicemail to multiple emails   vip killa
• [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?   Bruce B
• [asterisk-users] send voicemail to multiple emails   Sherwood McGowan
• [asterisk-users] Asterisk codec negotiation and canreinvite=no   Paul Belanger
• [asterisk-users] send voicemail to multiple emails   vip killa
• [asterisk-users] send voicemail to multiple emails   vip killa
• [asterisk-users] send voicemail to multiple emails   Sherwood McGowan
• [asterisk-users] Asterisk MOH not working with Elastix asterisk1.6.2.18   virendra bhati
• [asterisk-users] How to know the SIP status   virendra bhati
• [asterisk-users] "Wait for leader" allows crosstalk between participants   Alessandro Spagna
• [asterisk-users] How to know the SIP status   Sherwood McGowan
• [asterisk-users] Asterisk MOH not working with Elastix asterisk1.6.2.18   Warren Selby
• [asterisk-users] Voicemail to email issue   satish patel
• [asterisk-users] [SOLVED] Voicemail to email issue   satish patel
• [asterisk-users] Voicemail to email issue   Tiago Geada
• [asterisk-users] DAHDI-Linux 2.4.1.2 Released   Asterisk Development Team
• [asterisk-users] Asterisk as a Condo door opener/intercom   Don Kelly
• [asterisk-users] Features.conf - Blind Transfer   Neeraj Chand
• [asterisk-users] Variable stripping/removing part of string   Chad Wallace
• [asterisk-users] Poor call quality – line drop, chopping sound, like robotic voice, Both party could not hear caller voice   Mohd Aiman Rosli
• [asterisk-users] Templates   José Pablo Méndez Soto
• [asterisk-users] Variable stripping/removing part of string   magnus.b at inputinterior.se
• [asterisk-users] Templates   Tzafrir Cohen
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• [asterisk-users] Asterisk codec negotiation and canreinvite=no   Effie Mouzeli
• [asterisk-users] send voicemail to multiple emails   Andrew Thomas
• [asterisk-users] [OT] Yealink IP Phones   gincantalupo
• [asterisk-users] Variable inheritance with dialplan command Originate   Naomi Rosenberg
• [asterisk-users] send voicemail to multiple emails   vip killa
• [asterisk-users] [OT] Yealink IP Phones   bakko
• [asterisk-users] Queue(): How to know Estimated wait time for caller in advance   Asterisk Man
• [asterisk-users] send voicemail to multiple emails   Steve Edwards
• [asterisk-users] Queue(): How to know Estimated wait time forcaller in advance   Danny Nicholas
• [asterisk-users] Authentication failure   Adrian Marsh
• [asterisk-users] CEL Logging to MySQL - Please Test   Jonathan Penny
• [asterisk-users] Poor call quality – line drop, chopping sound, like robotic voice, Both party could not hear caller voice   satish patel
• [asterisk-users] From CDR to CEL   Olivier
• [asterisk-users] Variable inheritance with dialplan command Originate   Sherwood McGowan
• [asterisk-users] CEL Logging to MySQL - Please Test   Sherwood McGowan
• [asterisk-users] Asterisk codec negotiation and canreinvite=no   Ira
• [asterisk-users] Templates   Tilghman Lesher
• [asterisk-users] send voicemail to multiple emails   Steve Edwards
• [asterisk-users] send voicemail to multiple emails   vip killa
• [asterisk-users] Basic queue question   satish patel
• [asterisk-users] Basic queue question   Lyle McKarns
• [asterisk-users] Basic queue question   federico cabiddu
• [asterisk-users] Asterisk as a Condo door opener/intercom   C F
• [asterisk-users] Basic queue question   satish patel
• [asterisk-users] send voicemail to multiple emails   Steve Edwards
• [asterisk-users] send voicemail to multiple emails   Jim Dickenson
• [asterisk-users] send voicemail to multiple emails   Warren Selby
• [asterisk-users] send voicemail to multiple emails   Tilghman Lesher
• [asterisk-users] [OT] Yealink IP Phones   Larry Moore
• [asterisk-users] Templates   José Pablo Méndez Soto
• [asterisk-users] Problem with Swift app and escape digits   zac wolfe
• [asterisk-users] Debugging DTMF Detection   Antônio Theóphilo
• [asterisk-users] Aastra 480i & Asterisk 1.8.3.2: No musiconhold   Anthony Messina
• [asterisk-users] Aastra 480i & Asterisk 1.8.3.2: No musiconhold   Ira
• [asterisk-users] Basic queue question   federico cabiddu
• [asterisk-users] Realtime SIP & peer status   Jonas Kellens
• [asterisk-users] 0018818: [patch] Crashing when using local channels and realtime on asterisk 1.8.3-rc2   Ishfaq Malik
• [asterisk-users] Realtime SIP & peer status   Michel Verbraak
• [asterisk-users] send voicemail to multiple emails   Andrew Thomas
• [asterisk-users] Realtime SIP & peer status   Ishfaq Malik
• [asterisk-users] Poor call quality - line drop, chopping sound, like robotic voice, Both party could not hear caller voice   Andrew Thomas
• [asterisk-users] [OT] Yealink Phones   Russell Brown
• [asterisk-users] Realtime SIP & peer status   Jonas Kellens
• [asterisk-users] Realtime SIP & peer status   Andrew Thomas
• [asterisk-users] Realtime SIP & peer status   Ishfaq Malik
• [asterisk-users] Realtime SIP & peer status   Jonas Kellens
• [asterisk-users] How to know extensions status ???   virendra bhati
• [asterisk-users] Realtime SIP & peer status   Andrew Thomas
• [asterisk-users] Realtime SIP & peer status   Ishfaq Malik
• [asterisk-users] Realtime SIP & peer status   Jonas Kellens
• [asterisk-users] Realtime SIP & peer status   Ishfaq Malik
• [asterisk-users] Realtime SIP & peer status   Ishfaq Malik
• [asterisk-users] Realtime SIP & peer status   Andrew Thomas
• [asterisk-users] Realtime SIP & peer status   Andrew Thomas
• [asterisk-users] Realtime SIP & peer status   Jonas Kellens
• [asterisk-users] [OT] Yealink Phones   Andrew Thomas
• [asterisk-users] Variable inheritance with dialplan command Originate   Naomi Rosenberg
• [asterisk-users] send voicemail to multiple emails   M Hulber
• [asterisk-users] accessing currents calls from outside asterisk   Albert
• [asterisk-users] [OT] Yealink Phones   --[ UxBoD ]--
• [asterisk-users] Fwd: Re: Asterisk as a Condo door opener/intercom   David - asterisk list
• [asterisk-users] send voicemail to multiple emails   vip killa
• [asterisk-users] AGI and forking   A J Stiles
• [asterisk-users] How to know extensions status ???   Danny Nicholas
• [asterisk-users] AGI and forking   Thorsten Göllner
• [asterisk-users] AGI and forking   Steve Edwards
• [asterisk-users] Problems With DAHDI on Ubuntu   Shawn L
• [asterisk-users] Realtime SIP & peer status   Steve Edwards
• [asterisk-users] Problems With DAHDI on Ubuntu   Matt Riddell
• [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card   Kaushal Shriyan
• [asterisk-users] T38 fax detection using g729   Niccolò Belli
• [asterisk-users] Problems With DAHDI on Ubuntu   Shaun Ruffell
• [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card   satish patel
• [asterisk-users] Problems With DAHDI on Ubuntu   satish patel
• [asterisk-users] Asterisk thread limit   satish patel
• [asterisk-users] Asterisk thread limit   Andrew Latham
• [asterisk-users] Asterisk thread limit   satish patel
• [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card   Kaushal Shriyan
• [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card   Jim Dickenson
• [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card   Tim Nelson
• [asterisk-users] Asterisk Tech Tips: Cookin' with Asterisk   Steve Sokol
• [asterisk-users] Fw: SIP Trunk send DID or DNIS information   A S
• [asterisk-users] T38 fax detection using g729   Larry Moore
• [asterisk-users] [OT] Yealink Phones   Russell Brown
• [asterisk-users] AGI and forking   Tilghman Lesher
• [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card   Kaushal Shriyan
• [asterisk-users] Asterisk as a Condo door opener/intercom   Andreas Sikkema
• [asterisk-users] Realtime SIP & peer status   Rob Coward
• [asterisk-users] Safe to upgrade to Centos 5.6 now ???   Jian Gao
• [asterisk-users] Safe to upgrade to Centos 5.6 now ???   Shaun Ruffell
• [asterisk-users] Applet based softphone for Asterisk   ABBAS SHAKEEL
• [asterisk-users] Safe to upgrade to Centos 5.6 now ???   Jian Gao
• [asterisk-users] Safe to upgrade to Centos 5.6 now ???   Danny Nicholas
• [asterisk-users] Asterisk as a Condo door opener/intercom   Kyle Kienapfel
• [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card   Satish Patel
• [asterisk-users] Safe to upgrade to Centos 5.6 now ???   Satish Patel
• [asterisk-users] T38 fax detection using g729   Niccolò Belli
• [asterisk-users] Asterisk port 5000 open   Robert Thomas
• [asterisk-users] Asterisk port 5000 open   David White
• [asterisk-users] Processing sip messages   Nasir Iqbal
• [asterisk-users] Processing sip messages   Godson Gera
• [asterisk-users] How to know extensions status ???   virendra bhati
• [asterisk-users] Asterisk Tech Tips: Cookin' with Asterisk   virendra bhati
• [asterisk-users] Fwd: Re: Asterisk as a Condo door opener/intercom   Marco Signorini
• [asterisk-users] AGI and forking   A J Stiles
• [asterisk-users] Safe to upgrade to Centos 5.6 now ???   Vahan Yerkanian
• [asterisk-users] [OT] Yealink Phones   Andrew Thomas
• [asterisk-users] T38 fax detection using g729   Larry Moore
• [asterisk-users] T38 fax detection using g729   Niccolò Belli
• [asterisk-users] Safe to upgrade to Centos 5.6 now ???   me at tdiehl.org
• [asterisk-users] T38 fax detection using g729   Larry Moore
• [asterisk-users] setting sip headers when using call files   Israel Gottlieb
• [asterisk-users] setting sip headers when using call files   Israel Gottlieb
• [asterisk-users] setting sip headers when using call files   Satish Patel
• [asterisk-users] T38 fax detection using g729   Niccolò Belli
• [asterisk-users] Microsoft Lync server and Asterisk access   Jim Dickenson
• [asterisk-users] Asterisk modifies from header   Niraj Roy
• [asterisk-users] Microsoft Lync server and Asterisk access   BryantZ at zktech.com
• [asterisk-users] Existing Asterisk 1.8 upgrade with new release   satish patel
• [asterisk-users] T38 fax detection using g729   Larry Moore
• [asterisk-users] Followme() and variables   Jared Mauch
• [asterisk-users] Realtime SIP & peer status   Jonas Kellens
• [asterisk-users] sip error logging   Jeremy Kister
• [asterisk-users] sip error logging   Jeremy Kister
• [asterisk-users] Existing Asterisk 1.8 upgrade with new release   Kirill Katsnelson
• [asterisk-users] Would a job posting be ok for this list?   Kirill Katsnelson
• [asterisk-users] Realtime SIP & peer status   Adolphe Cher-aime
• [asterisk-users] Realtime SIP & peer status   Jonas Kellens
• [asterisk-users] How to get back park call   virendra bhati
• [asterisk-users] [OT] 802.11x roaming   Niccolò Belli
• [asterisk-users] Friday April 15 at 12 Noon EDT   randulo
• [asterisk-users] Friday April 15 at 12 Noon EDT   virendra bhati
• [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)   Vlasis Hatzistavrou
• [asterisk-users] Friday April 15 at 12 Noon EDT   randulo
• [asterisk-users] If voice mail not found dialplan   Satish Patel
• [asterisk-users] Friday April 15 at 12 Noon EDT   Satish Patel
• [asterisk-users] Friday April 15 at 12 Noon EDT   randulo
• [asterisk-users] If voice mail not found dialplan   Doug Lytle
• [asterisk-users] Good by asterisk 1.4? Please not.   Kristijan Vrban
• [asterisk-users] Good by asterisk 1.4? Please not.   Matt Riddell
• [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)   Jim Dickenson
• [asterisk-users] Good by asterisk 1.4? Please not.   Satish Patel
• [asterisk-users] Require dialplan   Matt Florell
• [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)   Vlasis Hatzistavrou
• [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)   Jim Dickenson
• [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)   Steve Davies
• [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)   Vlasis Hatzistavrou
• [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)   Vlasis Hatzistavrou
• [asterisk-users] If voice mail not found dialplan   satish patel
• [asterisk-users] sangoma card rx/tx gain level   satish patel
• [asterisk-users] Hot to make call parking to Mult tenant   virendra bhati
• [asterisk-users] Reach PSTN from another Asterisk   Alejandro Cabrera Obed
• [asterisk-users] accessing currents calls from outside asterisk   Pezhman Lali
• [asterisk-users] Reach PSTN from another Asterisk   Jim Dickenson
• [asterisk-users] Good by asterisk 1.4? Please not.   Julian Lyndon-Smith
• [asterisk-users] 1.8.4-rc2: ReceiveFAX fails   sean darcy
• [asterisk-users] 1.8.4-rc2: ReceiveFAX fails   Shaun Ruffell
• [asterisk-users] 1.8.4-rc2: ReceiveFAX fails   Ryan Wagoner
• [asterisk-users] sangoma card rx/tx gain level   cb
• [asterisk-users] Duplicate cdr records with channel local   Adolphe Cher-Aime
• [asterisk-users] any experience with cisco media gw with fax???   Oguzhan Kayhan
• [asterisk-users] 1.8.4-rc2: ReceiveFAX fails   Steve Underwood
• [asterisk-users] any experience with cisco media gw with fax???   Larry Moore
• [asterisk-users] Google Voice receiving call problem   Leandro Dardini
• [asterisk-users] write system command output into a variable   Daniel Knoll
• [asterisk-users] any experience with cisco media gw with fax???   Jared Mauch
• [asterisk-users] Safe to upgrade to Centos 5.6 now ???   Vahan Yerkanian
• [asterisk-users] "chan_sip.c: No such host:" but I can resolve it from command line ?   rob.r374 at gmail.com
• [asterisk-users] 1.8.4-rc2: ReceiveFAX fails   Ryan Wagoner
• [asterisk-users] 1.8.4-rc2: ReceiveFAX fails   Steve Underwood
• [asterisk-users] Google Voice receiving call problem   William Stillwell
• [asterisk-users] any experience with cisco media gw with fax???   Steve Underwood
• [asterisk-users] "chan_sip.c: No such host:" but I can resolve it from command line ?   Mark Deneen
• [asterisk-users] PADLOCK asterisk 1.6   Flavio Miranda
• [asterisk-users] any experience with cisco media gw with fax???   Jared Mauch
• [asterisk-users] any experience with cisco media gw with fax???   Oguzhan Kayhan
• [asterisk-users] Fwd: write system command output into a variable   Daniel Knoll
• [asterisk-users] any experience with cisco media gw with fax???   Jared Mauch
• [asterisk-users] CDR & ARI Question   John Jolly
• [asterisk-users] sip error logging   Jeremy Kister
• [asterisk-users] Jabber / facebook chat?   Stefan Gofferje
• [asterisk-users] Jabber / GTalk / hints   Stefan Gofferje
• [asterisk-users] sip error logging   bayardo.sanchez at gmail.com
• [asterisk-users] Jabber / facebook chat?   bayardo.sanchez at gmail.com
• [asterisk-users] Jabber / GTalk / hints   bayardo.sanchez at gmail.com
• [asterisk-users] bayardo.sanchez probably doesnt know he is autoresponding to lists   Jeremy Kister
• [asterisk-users] sip error logging   Sherwood McGowan
• [asterisk-users] sip error logging   Jeremy Kister
• [asterisk-users] sip error logging   Sherwood McGowan
• [asterisk-users] Asterisk 1.8.3: Started but no SIP talking   bilal ghayyad
• [asterisk-users] Asterisk 1.8.3: Started but no SIP talking   Warren Selby
• [asterisk-users] [OT, Job] Senior Software Engineer for exciting, high-growth startup   Kirill Katsnelson
• [asterisk-users] accessing currents calls from outside asterisk   virendra bhati
• [asterisk-users] Asterisk, virendra bhati has invited you to open a Gmail account   virendra bhati
• [asterisk-users] accessing currents calls from outside asterisk   virendra bhati
• [asterisk-users] CDR & ARI Question   Rizwan Hisham
• [asterisk-users] Registrations stops after 403 FORBIDDEN   Jonas Kellens
• [asterisk-users] Call Center Reporting   bilal ghayyad
• [asterisk-users] No Internet, no asterisk   Niccolò Belli
• [asterisk-users] R: No Internet, no asterisk   Alexandru Oniciuc
• [asterisk-users] No Internet, no asterisk   A J Stiles
• [asterisk-users] No Internet, no asterisk   Захаров Антон
• [asterisk-users] Call Center Reporting   Steven Howes
• [asterisk-users] No Internet, no asterisk   Dave Cotton
• [asterisk-users] Call Center Reporting   Sherwood McGowan
• [asterisk-users] Call Center Reporting   Sherwood McGowan
• [asterisk-users] Asterisk, virendra bhati has invited you to open a Gmail account   Sherwood McGowan
• [asterisk-users] Call Center Reporting   A J Stiles
• [asterisk-users] Softphone IAX   Eduardo Leones
• [asterisk-users] Call Center Reporting   Sherwood McGowan
• [asterisk-users] Asterisk unresponsive   Jonas Kellens
• [asterisk-users] Asterisk unresponsive   Terry Brummell
• [asterisk-users] Asterisk unresponsive   Jonas Kellens
• [asterisk-users] Registrations stops after 403 FORBIDDEN   Warren Selby
• [asterisk-users] Registrations stops after 403 FORBIDDEN   Jonas Kellens
• [asterisk-users] Meetme Time Limit?   Bryant Zimmerman
• [asterisk-users] Asterisk unresponsive   C. Savinovich
• [asterisk-users] canreinvite yes or no for PBX   satish patel
• [asterisk-users] Asterisk unresponsive   Paul Belanger
• [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit   satish patel
• [asterisk-users] A101DE Sangoma Card in AsteriskNow 1.7.1   Kaushal Shriyan
• [asterisk-users] core show channels consise in asterisk 1.8.3   Jerry Geis
• [asterisk-users] core show channels consise in asterisk 1.8.3   Leif Madsen
• [asterisk-users] core show channels consise in asterisk 1.8.3   Andrew Latham
• [asterisk-users] Safe to upgrade to Centos 5.6 now ???   Vahan Yerkanian
• [asterisk-users] core show channels consise in asterisk 1.8.3   Paul Belanger
• [asterisk-users] A101DE Sangoma Card in AsteriskNow 1.7.1   Kaushal Shriyan
• [asterisk-users] Asterisk unresponsive   Jonas Kellens
• [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND   Deka, Rajib IN MAA SL
• [asterisk-users] ConfBridge and AGI   Deka, Rajib IN MAA SL
• [asterisk-users] R: No Internet, no asterisk   Niccolò Belli
• [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND   Tony Mountifield
• [asterisk-users] R: No Internet, no asterisk   A J Stiles
• [asterisk-users] R: No Internet, no asterisk   Захаров Антон
• [asterisk-users] R: R: No Internet, no asterisk   Alexandru Oniciuc
• [asterisk-users] ConfBridge and AGI   DHAVAL INDRODIYA
• [asterisk-users] Good by asterisk 1.4? Please not.   Kristijan Vrban
• [asterisk-users] R: No Internet, no asterisk   Mark Deneen
• [asterisk-users] R: No Internet, no asterisk   Eric Wieling
• [asterisk-users] chan_dahdi under bristuff 0.4.0-RC11 not building under Linux 10.04.2 LTS   Mosiuoa Tsietsi
• [asterisk-users] Good by asterisk 1.4? Please not.   Paul Belanger
• [asterisk-users] Good by asterisk 1.4? Please not.   Julian Lyndon-Smith
• [asterisk-users] Good by asterisk 1.4? Please not.   William Stillwell
• [asterisk-users] chan_mobile: Dropping incompatible voice frame   Stefan Gofferje
• [asterisk-users] Good by asterisk 1.4? Please not.   Julian Lyndon-Smith
• [asterisk-users] RTP and Signalling Dropping   Jon Farmer
• [asterisk-users] sterisk+SS7 Error: chan_dahdi.c: Unable to start PBX on DAHDI/288-1   Camilo Echeverry
• [asterisk-users] How to know how many calls are into hold by asterisk command   virendra bhati
• [asterisk-users] How to know how many calls are into hold byasterisk command   Danny Nicholas
• [asterisk-users] Call Center Reporting   Bruce B
• [asterisk-users] [SOLVED] Asterisk thread limit   satish patel
• [asterisk-users] Linux Based Billing and CDR   Anton VG
• [asterisk-users] T38 fax detection using g729   Kevin P. Fleming
• [asterisk-users] Softphone IAX   Matt Riddell
• [asterisk-users] R: No Internet, no asterisk   Matt Riddell
• [asterisk-users] IP Address Management / Open Source / IPAM   Thomas Perron
• [asterisk-users] Configure IVR(Inbound and Outbound)   Kaushal Shriyan
• [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND   Deka, Rajib IN MAA SL
• [asterisk-users] How to know how many calls are into hold byasterisk command   virendra bhati
• [asterisk-users] Configure IVR(Inbound and Outbound)   Gopalakrishnan A.N
• [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND   Tony Mountifield
• [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND   DHAVAL INDRODIYA
• [asterisk-users] chan_dahdi under bristuff 0.4.0-RC11 not building under Linux 10.04.2 LTS   Tzafrir Cohen
• [asterisk-users] Configure IVR(Inbound and Outbound)   Gopalakrishnan A.N
• [asterisk-users] Configure IVR(Inbound and Outbound)   Kaushal Shriyan
• [asterisk-users] No voice in MeetMe for SIP with   Deka, Rajib IN MAA SL
• [asterisk-users] T38 fax detection using g729   Niccolò Belli
• [asterisk-users] No voice in MeetMe for SIP with   DHAVAL INDRODIYA
• [asterisk-users] R: No Internet, no asterisk   Ron Arts
• [asterisk-users] No voice in MeetMe for SIP with   Tony Mountifield
• [asterisk-users] Configure IVR(Inbound and Outbound)   Gopalakrishnan A.N
• [asterisk-users] T38 fax detection using g729   Kevin P. Fleming
• [asterisk-users] 1.8.x sip error 0.0.27.191:5060 returned -1: Invalid argument   satish patel
• [asterisk-users] dtmf payload type problem during faxing..   Oguzhan Kayhan
• [asterisk-users] How to know how many calls are into hold byasterisk command   Danny Nicholas
• [asterisk-users] dtmf payload type problem during faxing..   Kevin P. Fleming
• [asterisk-users] issue with installtion asterisk   salaheddine elharit
• [asterisk-users] Call files or AMI originate for mass outbound call   Adolphe Cher-Aime
• [asterisk-users] issue with installtion asterisk   Jose P. Espinal
• [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)   John Alexis
• [asterisk-users] issue with installtion asterisk   salaheddine elharit
• [asterisk-users] asterisk log - "=======" extension not found?   ERIC HERRON
• [asterisk-users] asterisk log - "=======" extension not found?   Paul Belanger
• [asterisk-users] issue with installtion asterisk   Jose P. Espinal
• [asterisk-users] asterisk log - "=======" extension not found?   ERIC HERRON
• [asterisk-users] asterisk log - "=======" extension not found?   Danny Nicholas
• [asterisk-users] allowguest=yes, how?   Paul van der Vlis
• [asterisk-users] allowguest=yes, how?   Danny Nicholas
• [asterisk-users] VoiceMail to text mail   satish patel
• [asterisk-users] asterisk log - "=======" extension not found?   Paul Belanger
• [asterisk-users] Call files or AMI originate for mass outbound call   Paul Belanger
• [asterisk-users] VoiceMail to text mail   Steve Edwards
• [asterisk-users] VoiceMail to text mail   satish patel
• [asterisk-users] Call files or AMI originate for mass outbound call   Adolphe Cher-Aime
• [asterisk-users] VoiceMail to text mail   Mark Deneen
• [asterisk-users] py-Asterisk or pyst?   Ye Liu
• [asterisk-users] py-Asterisk or pyst?   Danny Nicholas
• [asterisk-users] Call files or AMI originate for mass outbound call   Paul Belanger
• [asterisk-users] py-Asterisk or pyst?   Paul Belanger
• [asterisk-users] Call files or AMI originate for mass outbound call   Adolphe Cher-Aime
• [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND   Deka, Rajib IN MAA SL
• [asterisk-users] How to know how many calls are into hold byasterisk command   virendra bhati
• [asterisk-users] allowguest=yes, how?   Paul van der Vlis
• [asterisk-users] issue with installtion asterisk   salaheddine elharit
• [asterisk-users] Meetme Time Limit?   DHAVAL INDRODIYA
• [asterisk-users] Voicemail forward issue   virendra bhati
• [asterisk-users] Nat=yes   Alexandru Oniciuc
• [asterisk-users] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()   RAJNIKANT VANZA
• [asterisk-users] Meetme Time Limit?   Sherwood McGowan
• [asterisk-users] Asterisk Export Fax from Wave file   Khaled W. Chehab
• [asterisk-users] VoiceMail to text mail   Sherwood McGowan
• [asterisk-users] Asterisk Export Fax from Wave file   A J Stiles
• [asterisk-users] Asterisk Export Fax from Wave file   Steven Howes
• [asterisk-users] allowguest=yes, how?   Danny Nicholas
• [asterisk-users] Asterisk Export Fax from Wave file   Steve Underwood
• [asterisk-users] Transcode ulaw/g722 problem   Eric Wieling
• [asterisk-users] Transcode ulaw/g722 problem   Danny Nicholas
• [asterisk-users] Transcode ulaw/g722 problem   virendra bhati
• [asterisk-users] IAX2 codec selection and video   Steve Davies
• [asterisk-users] missed call notification   satish patel
• [asterisk-users] missed call notification   Sherwood McGowan
• [asterisk-users] missed call notification   satish patel
• [asterisk-users] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()   Mark Deneen
• [asterisk-users] Nationalprefix chan_dahdi option   Eric Wieling
• [asterisk-users] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()   Steve Edwards
• [asterisk-users] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()   Mark Deneen
• [asterisk-users] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()   Steve Edwards
• [asterisk-users] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()   Mark Deneen
• [asterisk-users] AST-2011-005: File Descriptor Resource Exhaustion   Asterisk Security Team
• [asterisk-users] AST-2011-006: Asterisk Manager User Shell Access   Asterisk Security Team
• [asterisk-users] Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 Now Available (Security Releases)   Asterisk Development Team
• [asterisk-users] Help Required---Problem in Installation without dahdi   Muhammad Ali
• [asterisk-users] Flite issue   virendra bhati
• [asterisk-users] Flite issue   Lefteris Zafiris
• [asterisk-users] T38 fax printer Windows client for asterisk 1.8   Alec Davis
• [asterisk-users] Flite issue   virendra bhati
• [asterisk-users] Flite issue   Lefteris Zafiris
• [asterisk-users] ZRTP SDK Source   Nasir Iqbal
• [asterisk-users] Flite issue   virendra bhati
• [asterisk-users] Flite issue   Satish Patel
• [asterisk-users] Flite issue   Doug Lytle
• [asterisk-users] question on register and dnsmgr_lookup   Jerry Geis
• [asterisk-users] Flite issue   Lefteris Zafiris
• [asterisk-users] missed call notification   satish patel
• [asterisk-users] missed call notification   Sherwood McGowan
• [asterisk-users] Cannot call to my server with SIP   Paul van der Vlis
• [asterisk-users] Cannot call to my server with SIP   Mark Deneen
• [asterisk-users] Cannot call to my server with SIP   Eric Wieling
• [asterisk-users] Flite issue   virendra bhati
• [asterisk-users] (no subject)   Dmitry Kupchinetsky
• [asterisk-users] WARNING T.30 ECM carrier not found   satish patel
• [asterisk-users] Multi tenant Parking issue   virendra bhati
• [asterisk-users] nyc area pbx rfp 4000 extensions   A_ Navone
• [asterisk-users] Huawei K3765 + Internet + SMS + Telephone   Michelle Konzack
• [asterisk-users] Cannot call to my server with SIP   Paul van der Vlis
• [asterisk-users] Cannot call to my server with SIP   Paul van der Vlis
• [asterisk-users] AST-2011-006: Asterisk Manager User Shell Access   Tzafrir Cohen
• [asterisk-users] Cannot call to my server with SIP   Steve Edwards
• [asterisk-users] Cannot call to my server with SIP   Jamie A. Stapleton
• [asterisk-users] Huawei K3765 + Internet + SMS + Telephone   asterisk asterisk
• [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)   John Alexis
• [asterisk-users] Warm Transfer in Asterisk   Jeff Johnson
• [asterisk-users] Warm Transfer in Asterisk   Ryan Wagoner
• [asterisk-users] Cannot call to my server with SIP   Paul van der Vlis
• [asterisk-users] Warm Transfer in Asterisk   Jeff Johnson
• [asterisk-users] Cannot call to my server with SIP   Paul van der Vlis
• [asterisk-users] DTMF not being sent ( RFC2833 )   David
• [asterisk-users] call files   Tiago Geada
• [asterisk-users] call files   Tiago Geada
• [asterisk-users] call files   Sherwood McGowan
• [asterisk-users] ARA table definitions (1.8.*)   Jason Rogers
• [asterisk-users] ARA table definitions (1.8.*)   Hans Witvliet
• [asterisk-users] Huawei K3765 + Internet + SMS + Telephone   Michelle Konzack
• [asterisk-users] Cannot call to my server with SIP   Eric Wieling
• [asterisk-users] Nat=yes   Pezhman Lali
• [asterisk-users] IAX2 codec selection and video   Pezhman Lali
• [asterisk-users] Nat=yes   Steve Totaro
• [asterisk-users] Nat=yes   Muhammad Ali
• [asterisk-users] Nat=yes   Steve Totaro
• [asterisk-users] Nat=yes   Muhammad Ali
• [asterisk-users] call files   Tiago Geada
• [asterisk-users] DTMF not being sent ( RFC2833 )   David
• [asterisk-users] Best modem for chan_datacard   Dovid Bender
• [asterisk-users] DTMF not being sent ( RFC2833 )   David
• [asterisk-users] DTMF incorrectly sent ( RFC2833 or SIPInfo )   David
• [asterisk-users] Realtime and priority labels   Bruce Ferrell
• [asterisk-users] Repost: Jabber / facebook chat?   Stefan Gofferje
• [asterisk-users] Repost: Jabber / GTalk / hints   Stefan Gofferje
• [asterisk-users] (no subject)   Abid Saleem
• [asterisk-users] Flite issue   virendra bhati
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   David Vossel
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   Richard Kenner
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   David Backeberg
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   David Vossel
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   Andrew Latham
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   C. Savinovich
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   David Vossel
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   David Backeberg
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   David Vossel
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   David Vossel
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   Paul Belanger
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   Andrew Latham
• [asterisk-users] new confbridge   Jerry Geis
• [asterisk-users] new confbridge   David Vossel
• [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)   Camilo Echeverry
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   Richard Kenner
• [asterisk-users] FILTER function and multiple ranges?   Brian J. Murrell
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   Shaun Ruffell
• [asterisk-users] Channel Hunting Origination   Abid Saleem
• [asterisk-users] Cannot call to my server with SIP   Jamie A. Stapleton
• [asterisk-users] Registration problems - Vitelity   scott at evtmedia.com
• [asterisk-users] Transfer beep w/ Polycom phone   Mike Diehl
• [asterisk-users] PAP2T auto answer?   Mike Diehl
• [asterisk-users] Transfer beep w/ Polycom phone   Danny Nicholas
• [asterisk-users] Realtime and priority labels   Edwin Lam
• [asterisk-users] DTMF not being sent ( RFC2833 )   Edwin Lam
• [asterisk-users] AST-2011-006: Asterisk Manager User Shell Access   Matt Riddell
• [asterisk-users] PAP2T auto answer?   Terry Brummell
• [asterisk-users] PAP2T auto answer?   Matt Riddell
• [asterisk-users] DTMF not being sent ( RFC2833 )   Jim Dickenson
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   Jeremy Kister
• [asterisk-users] R: Nat=yes   Alexandru Oniciuc
• [asterisk-users] How does wrandom strategy works with Queue?   Asterisk Man
• [asterisk-users] How does wrandom strategy works with Queue?   Jeroen Eeuwes
• [asterisk-users] How does wrandom strategy works with Queue?   Asterisk Man
• [asterisk-users] Orginate not working well with PSTN lines   Ashik Ali
• [asterisk-users] Orginate not working well with PSTN lines   Jim Dickenson
• [asterisk-users] Overlap dialing with MFC/R2   Vinícius Fontes
• [asterisk-users] Password to be ecrypted?   bilal ghayyad
• [asterisk-users] Password to be ecrypted?   Danny Nicholas
• [asterisk-users] play audio file to destination SIP channel on attended call transfer   Deka, Rajib IN MAA SL
• [asterisk-users] Password to be ecrypted?   A J Stiles
• [asterisk-users] siren sound   María Esperanza Ballestero Campillo
• [asterisk-users] Asterisk 1.4.40.2 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.41 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.18 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.40.2 Now Available   Warren Selby
• [asterisk-users] Asterisk 1.4.40.2 Now Available   Kevin P. Fleming
• [asterisk-users] Seattle WA Asterisk Users' Group   C.J. Adams-Collier
• [asterisk-users] Retaining original caller id   Jan Bakuwel
• [asterisk-users] How does wrandom strategy works with Queue?   Jeroen Eeuwes
• [asterisk-users] Has anybody been able to install CDR-Stats all the way through?   José Pablo Méndez Soto
• [asterisk-users] RTP keepalive doesn't work   Alok
• [asterisk-users] PAP2T auto answer?   C F
• [asterisk-users] Orginate not working well with PSTN lines   Ashik Ali
• [asterisk-users] Orginate not working well with PSTN lines   Gilles
• [asterisk-users] Orginate not working well with PSTN lines   Ashik Ali
• [asterisk-users] how to know status of asterisk from php   virendra bhati
• [asterisk-users] Konference module issue   virendra bhati
• [asterisk-users] Orginate not working well with PSTN lines   Eric Wieling
• [asterisk-users] Digium WCTDM24XXP DTMF CallerID   Antonio Modesto
• [asterisk-users] AGI WAIT FOR DIGIT - key press BEFORE command   David
• [asterisk-users] Digium WCTDM24XXP DTMF CallerID   Shaun Ruffell
• [asterisk-users] how to know status of asterisk from php   Juan David Diaz
• [asterisk-users] how to know status of asterisk from php   nik600
• [asterisk-users] h323 with NAT   Danny Nicholas
• [asterisk-users] h323 with NAT   Jose P. Espinal
• [asterisk-users] Asterisk, SIP & Firewalls   Myles Wakeham
• [asterisk-users] asterisk practices   vip killa
• [asterisk-users] h323 with NAT   Danny Nicholas
• [asterisk-users] asterisk practices   David
• [asterisk-users] Asterisk, SIP & Firewalls   Stelios Koroneos
• [asterisk-users] Asterisk, SIP & Firewalls   Ryan Wagoner
• [asterisk-users] DHCP / DNS   Thomas Perron
• [asterisk-users] DHCP / DNS   Andrew Latham
• [asterisk-users] DHCP / DNS   Eric Wieling
• [asterisk-users] h323 with NAT   Danny Nicholas
• [asterisk-users] h323 with NAT   Jose P. Espinal
• [asterisk-users] Echocancellation OSLEC vs MG2 ?   satish patel
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Olle E. Johansson
• [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)   John Alexis
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Danny Nicholas
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Tim Nelson
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Andrew Latham
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Olle E. Johansson
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Administrator TOOTAI
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Gordon Henderson
• [asterisk-users] PAP2T auto answer?   Mike Diehl
• [asterisk-users] Echocancellation OSLEC vs MG2 ?   Anthony Messina
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Matteo Piazza
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Michael L. Young
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Kevin Keane
• [asterisk-users] how to know status of asterisk from php   virendra bhati
• [asterisk-users] how to know status of asterisk from php   Tiago Geada
• [asterisk-users] Echocancellation OSLEC vs MG2 ?   Tiago Geada
• [asterisk-users] how to know status of asterisk from php   virendra bhati
• [asterisk-users] how to know status of asterisk from php   Matt Riddell
• [asterisk-users] how to know status of asterisk from php   virendra bhati
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Danny Nicholas
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Bryant Zimmerman
• [asterisk-users] Echocancellation OSLEC vs MG2 ?   satish patel
• [asterisk-users] Echocancellation OSLEC vs MG2 ?   satish patel
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Russell Bryant
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Chris Owen
• [asterisk-users] [RESOLVED] Echocancellation OSLEC vs MG2 ?   satish patel
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Russell Bryant
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   Bruce B
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Chris Owen
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   David
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   Danny Nicholas
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   Stefan Gofferje
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Russell Bryant
• [asterisk-users] how to know status of asterisk from php   Steve Edwards
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   David
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   Tony Mountifield
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   satish patel
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Administrator TOOTAI
• [asterisk-users] odbc error - server is gone   Rizwan Hisham
• [asterisk-users] how to know status of asterisk from php   Juan David Diaz
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ira
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   satish patel
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Andrew Latham
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Paul Belanger
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   satish patel
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   Bruce B
• [asterisk-users] odbc error - server is gone   Sherwood McGowan
• [asterisk-users] Best modem for chan_datacard   Tiago Geada
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Leif Madsen
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   Andreas Sikkema
• [asterisk-users] anybody out there sucessfully using gnugk?   Danny Nicholas
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Administrator TOOTAI
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ira
• [asterisk-users] anybody out there sucessfully using gnugk?   Andreas Sikkema
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Leif Madsen
• [asterisk-users] how to know status of asterisk from php   Matt Riddell
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Administrator TOOTAI
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Danny Nicholas
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   Matt Riddell
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   Matt Riddell
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Matt Riddell
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Alec Davis
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Paul Belanger
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ira
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Russell Bryant
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Russell Bryant
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Matt Riddell
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Russell Bryant
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ira
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Paul Belanger
• [asterisk-users] Orginate not working well with PSTN lines   Gilles
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Alec Davis
• [asterisk-users] asterisk practices   Jan Bakuwel
• [asterisk-users] Asterisk 1.6.2.18 Now Available   Jan Bakuwel
• [asterisk-users] Asterisk 1.6.2.18 Now Available   Matt Riddell
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Leif Madsen
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ernie Dunbar
• [asterisk-users] Asterisk 1.6.2.18 Now Available   Jan Bakuwel
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Leif Madsen
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Matt Riddell
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ira
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Alec Davis
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Matt Riddell
• [asterisk-users] Friday on VUC: Jabber/XMPP   randulo
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Matt Riddell
• [asterisk-users] Asterisk 1.6.2.18 Now Available   Matt Riddell
• [asterisk-users] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()   RAJNIKANT VANZA
• [asterisk-users] Multiple Asterisk   RAJNIKANT VANZA
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Olle E. Johansson
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Olle E. Johansson
• [asterisk-users] Discussion: 1.8 quality issues   Olle E. Johansson
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Administrator TOOTAI
• [asterisk-users] Hardware Server Configuration/8 or 4 port PRI Card   Kaushal Shriyan
• [asterisk-users] Multiple Asterisk   Henrique Fernandes
• [asterisk-users] odbc error - server is gone   Rizwan Hisham
• [asterisk-users] Multiple Asterisk   Satish Patel
• [asterisk-users] Multiple Asterisk   Henrique Fernandes
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Leif Madsen
• [asterisk-users] Multiple Asterisk   Satish Patel
• [asterisk-users] (no subject)   Muhammad Usman
• [asterisk-users] Multiple Asterisk   Henrique Fernandes
• [asterisk-users] Multiple Asterisk   satish patel
• [asterisk-users] Multiple Asterisk   vip killa
• [asterisk-users] Multiple Asterisk   Henrique Fernandes
• [asterisk-users] Discussion: 1.8 quality issues   Russell Bryant
• [asterisk-users] Multiple Asterisk   satish patel
• [asterisk-users] SIP bad request   Mike
• [asterisk-users] SIP bad request   Захаров Антон
• [asterisk-users] Local channel scenario flushes CDR before dialplan end   Grigoriy Puzankin
• [asterisk-users] SIP bad request   Mike
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   satish patel
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Mark Deneen
• [asterisk-users] odbc error - server is gone   Warren Selby
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Richard Zheng
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Danny Nicholas
• [asterisk-users] HA Asterisk   Kaushal Shriyan
• [asterisk-users] HA Asterisk   Michelle Dupuis
• [asterisk-users] Orginate not working well with PSTN lines   Ashik Ali
• [asterisk-users] HA Asterisk   RAJNIKANT VANZA
• [asterisk-users] HA Asterisk   Alex Balashov
• [asterisk-users] HA Asterisk   Kaushal Shriyan
• [asterisk-users] Best modem for chan_datacard   asterisk asterisk
• [asterisk-users] odbc error - server is gone   Pezhman Lali
• [asterisk-users] SIP bad request   Pezhman Lali
• [asterisk-users] AGI WAIT FOR DIGIT - key press BEFORE command   Pezhman Lali
• [asterisk-users] HA Asterisk   Satish Patel
• [asterisk-users] HA Asterisk   Adolphe Cher-aime
• [asterisk-users] HA Asterisk   Adrian Serafini
• [asterisk-users] HA Asterisk   Michelle Dupuis
• [asterisk-users] dial multiple extensions   Roy Kidder
• [asterisk-users] dial multiple extensions   Satish Patel
• [asterisk-users] HA Asterisk   Satish Patel
• [asterisk-users] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend   Alec Taylor
• [asterisk-users] HA Asterisk   Michelle Dupuis
• [asterisk-users] HA Asterisk   Kaushal Shriyan
• [asterisk-users] HA Asterisk   Michelle Dupuis
• [asterisk-users] HA Asterisk   Kaushal Shriyan

Last message date: Sat Apr 30 22:31:56 CDT 2011
Archived on: Mon Aug 29 15:37:25 CDT 2011

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