August 2010 Archives by author

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Starting: Sun Aug 1 01:25:31 CDT 2010
Ending: Tue Aug 31 22:15:36 CDT 2010
Messages: 1138

• [asterisk-users] IAX softphone   Alan Lord (News)
• [asterisk-users] FYI: Seen the 2600Hz announcement?   Alan Lord (News)
• [asterisk-users] install asterisk   Alan Lord (News)
• [asterisk-users] asterisk + openBTS   Alan Lord (News)
• [asterisk-users] asterisk + openBTS   Alan Lord (News)
• [asterisk-users] Opensource Speech recognition for Asterisk   Jason Aarons (US)
• [asterisk-users] What do you use for Invoicing?   adamk at 3a.hu
• [asterisk-users] fail2ban does not work for my asterisk installation   Lonnie Abelbeck
• [asterisk-users] IAX softphone   Ronaldo Zacarias Afonso
• [asterisk-users] Dahdi issue on sangoma A200   Max Alex
• [asterisk-users] Dahdi issue on sangoma A200   Max Alex
• [asterisk-users] install asterisk   Benny Amorsen
• [asterisk-users] CDR on Transfer...   Benny Amorsen
• [asterisk-users] CDR on Transfer...   Benny Amorsen
• [asterisk-users] Asterisk 1.6 without DAHDI   Roderick A. Anderson
• [asterisk-users] Asterisk 1.6 without DAHDI   Roderick A. Anderson
• [asterisk-users] Using a 1.4 config with 1.6   Roderick A. Anderson
• [asterisk-users] Using a 1.4 config with 1.6   Roderick A. Anderson
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Roderick A. Anderson
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Roderick A. Anderson
• [asterisk-users] FreeTDS (Microsoft MsSQL 2008) and CDR   Andraž
• [asterisk-users] ODBC Voicemail storage   Andraž
• [asterisk-users] ODBC Voicemail storage   Andraž
• [asterisk-users] CDR on Transfer...   Andraž
• [asterisk-users] double DTMF digits   Andres
• [asterisk-users] TELUS British Columbia PRI Settings   Andres
• [asterisk-users] TELUS British Columbia PRI Settings   Andres
• [asterisk-users] TELUS British Columbia PRI Settings   Andres
• [asterisk-users] TELUS British Columbia PRI Settings   Andres
• [asterisk-users] Use of AGISIGHUP   Lee Archer
• [asterisk-users] Use of AGISIGHUP   Lee Archer
• [asterisk-users] Asterisk Crashes - Segmentation Fault   Dan Austin
• [asterisk-users] Asterisk Crashes - Segmentation Fault   Dan Austin
• [asterisk-users] Asterisk and RAID   David Backeberg
• [asterisk-users] MeetMe VS. Conference   David Backeberg
• [asterisk-users] How to set up Asterisk to deliver a trunk sip connection?   David Backeberg
• [asterisk-users] asterisk on Vmware   David Backeberg
• [asterisk-users] How to set up Asterisk to deliver a trunk sip connection?   David Backeberg
• [asterisk-users] 4 Port FXO interface   David Backeberg
• [asterisk-users] Asterisk on AMD   David Backeberg
• [asterisk-users] Polycom 331 freezes connecting to FreePBX   David Backeberg
• [asterisk-users] Opensource Speech recognition for Asterisk   David Backeberg
• [asterisk-users] asterisk + cisco 3825 with ISDN   David Backeberg
• [asterisk-users] No Mailbox Subscription in SIP Users Suddenly   Jayson Baker
• [asterisk-users] Missing Mailboxes on SIP   Jayson Baker
• [asterisk-users] shrinkcallerid   Jayson Baker
• [asterisk-users] Asterisk Crashed - But why?   Jayson Baker
• [asterisk-users] Asterisk Crashed - But why?   Jayson Baker
• [asterisk-users] MP3Player audio format   Andy Beak
• [asterisk-users] .call files with application/data are not generating correct CDR   Andy Beak
• [asterisk-users] # -key not to be 'transfer'   Paul Belanger
• [asterisk-users] Asterisk and TV media server   Paul Belanger
• [asterisk-users] FYI: Seen the 2600Hz announcement?   Paul Belanger
• [asterisk-users] outboundproxy timeout or qualify   Paul Belanger
• [asterisk-users] Using a 1.4 config with 1.6   Paul Belanger
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   Paul Belanger
• [asterisk-users] codec_g729.so not work!   Paul Belanger
• [asterisk-users] Opensource Speech recognition for Asterisk   Paul Belanger
• [asterisk-users] How to do barging using asterisk server.   Paul Belanger
• [asterisk-users] Make a transfer for external line.   Paul Belanger
• [asterisk-users] How to prevent soft hangup from being necessary ?   Paul Belanger
• [asterisk-users] Opensource Speech recognition for Asterisk   Paul Belanger
• [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3   Paul Belanger
• [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3   Paul Belanger
• [asterisk-users] All phones ringing when temporary loss of Internet   Paul Belanger
• [asterisk-users] DAHDI not detecting caller hangup   Paul Belanger
• [asterisk-users] DAHDI not detecting caller hangup   Paul Belanger
• [asterisk-users] Asterisk voicemail server - gsm notifications   Paul Belanger
• [asterisk-users] Make a transfer for external line.   Paul Belanger
• [asterisk-users] DAHDI not detecting caller hangup   Paul Belanger
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Paul Belanger
• [asterisk-users] Asterisk 1.6.1 Won't Play Default ULAW Files   Paul Belanger
• [asterisk-users] 1.6 and asterisk gui   Paul Belanger
• [asterisk-users] music on hold in blind transfer   Paul Belanger
• [asterisk-users] TELUS British Columbia PRI Settings   Paul Belanger
• [asterisk-users] Could MeetMe invite someone to the conference?   Paul Belanger
• [asterisk-users] SIP Debug Messages   Paul Belanger
• [asterisk-users] help with dialplan   Paul Belanger
• [asterisk-users] help with dialplan   Paul Belanger
• [asterisk-users] help with dialplan   Paul Belanger
• [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6   Paul Belanger
• [asterisk-users] asterisk core dump   Paul Belanger
• [asterisk-users] help with dialplan   Paul Belanger
• [asterisk-users] Yes it is a dimensioning question! Atom CPU   Paul Belanger
• [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6   Paul Belanger
• [asterisk-users] AstriCon approaches: Innovation Awards, your attendance wanted!   Alex Bell
• [asterisk-users] help with dialplan   Alex Bell
• [asterisk-users] Enhancing snmp mib   Benoit
• [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question   Trevor Benson
• [asterisk-users] asterisk compatible cards?   Jeremy Betts
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   Jeremy Betts
• [asterisk-users] Asterisk and RAID   Gareth Blades
• [asterisk-users] AMI Command   Gareth Blades
• [asterisk-users] AMI Command   Gareth Blades
• [asterisk-users] asterisk on Vmware   Gareth Blades
• [asterisk-users] asterisk on Vmware   Gareth Blades
• [asterisk-users] realtime sip peers : musiconhold class   Gareth Blades
• [asterisk-users] realtime sip peers : musiconhold class   Gareth Blades
• [asterisk-users] 4 Port FXO interface   Gareth Blades
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Gareth Blades
• [asterisk-users] Quick Question - Jabra Headset and Aastra 53i - Where is the speaker/headset enable setting on Aastra UI?   Gareth Blades
• [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question   Steven C. Blair
• [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question   Steven C. Blair
• [asterisk-users] install asterisk   Albert Bonomo
• [asterisk-users] installing with yum   Albert Bonomo
• [asterisk-users] installing with yum   Albert Bonomo
• [asterisk-users] installing with yum   Albert Bonomo
• [asterisk-users] install asterisk   Albert Bonomo
• [asterisk-users] [SIP/H.264] Codec negotiation problem ?   Nicolas Bourbaki
• [asterisk-users] Convert wav-file to alaw-file   Sean Bright
• [asterisk-users] Codec Conversion   Jeff Brower
• [asterisk-users] Codec Conversion   Jeff Brower
• [asterisk-users] Codec Conversion   Jeff Brower
• [asterisk-users] Codec Conversion   Jeff Brower
• [asterisk-users] Codec Conversion   Jeff Brower
• [asterisk-users] Opensource Speech recognition for Asterisk   Jeff Brower
• [asterisk-users] What do you use for Invoicing?   Nick Brown
• [asterisk-users] PRI errors no D channel   Nick Brown
• [asterisk-users] 4 Port FXO interface   Krishna Sumanth Chava
• [asterisk-users] Using SIP to dial extension that will give anoutside line   Carlos Chavez
• [asterisk-users] sip.conf register in realtime DB   Carlos Chavez
• [asterisk-users] Executing system commands through Manager API   Carlos Chavez
• [asterisk-users] CDR on Transfer...   Carlos Chavez
• [asterisk-users] Asterisk with Blockhosts   Carlos Chavez
• [asterisk-users] How does deny/permit work in sip.conf?   Frank Church
• [asterisk-users] How does deny/permit work in sip.conf?   Frank Church
• [asterisk-users] How does deny/permit work in sip.conf?   Frank Church
• [asterisk-users] Should external hosts be able to register on Asterisk behind a firewall?   Frank Church
• [asterisk-users] Opensource Speech recognition for Asterisk   Seann Clark
• [asterisk-users] mapping of disconnect reasons   Harel Cohen
• [asterisk-users] mapping of disconnect reasons   Harel Cohen
• [asterisk-users] mapping of disconnect reasons   Harel Cohen
• [asterisk-users] FYI: Seen the 2600Hz announcement?   Tzafrir Cohen
• [asterisk-users] Asterisk on Ben NanoNote?   Tzafrir Cohen
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   Tzafrir Cohen
• [asterisk-users] DAHDI config file system.conf   Tzafrir Cohen
• [asterisk-users] Asterisk on AMD   Tzafrir Cohen
• [asterisk-users] IXJ Quicknet PhoneJack issues   Tzafrir Cohen
• [asterisk-users] install asterisk   Tzafrir Cohen
• [asterisk-users] install asterisk   Tzafrir Cohen
• [asterisk-users] Removing `chan_dahdi.conf`   Tzafrir Cohen
• [asterisk-users] colored CLI with reattach   Tzafrir Cohen
• [asterisk-users] OT - How to blacklist a driver in /etc/modprobe.d without reboot   Tzafrir Cohen
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Tzafrir Cohen
• [asterisk-users] TELUS British Columbia PRI Settings   Tzafrir Cohen
• [asterisk-users] asterisk-users Digest, Vol 73, Issue 63   Tzafrir Cohen
• [asterisk-users] FYI: Seen the 2600Hz announcement?   Dean Collins
• [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?   Rob Coward
• [asterisk-users] Asterisk and RAID   Gergo Csibra
• [asterisk-users] dynamic MeetMe, min. digits   Xavier D.
• [asterisk-users] speciality of SIPp and SER(Sip Express Router)   Klaus Darilion
• [asterisk-users] Asterisk on Ben NanoNote?   Dave
• [asterisk-users] How to debug this specific issue?   Steve Davies
• [asterisk-users] How to debug this specific issue?   Steve Davies
• [asterisk-users] How to debug this specific issue?   Steve Davies
• [asterisk-users] How to debug this specific issue?   Steve Davies
• [asterisk-users] How to debug this specific issue?   Steve Davies
• [asterisk-users] $250 Asterisk app install bounty   Troy Davis
• [asterisk-users] Compiling snmp_res.so into AsteriskNow install   Tyler Davis
• [asterisk-users] Dahdi install gone wrong   Doug Dawson
• [asterisk-users] Dahdi install gone wrong   Doug Dawson
• [asterisk-users] Removing `chan_dahdi.conf`   Randall Degges
• [asterisk-users] Asterisk 1.6.1 Won't Play Default ULAW Files   Randall Degges
• [asterisk-users] Asterisk 1.6.1 Won't Play Default ULAW Files   Randall Degges
• [asterisk-users] Asterisk 1.6.1 Won't Play Default ULAW Files   Randall Degges
• [asterisk-users] double DTMF digits   Matt Desbiens
• [asterisk-users] Maximum Wait Time queue option   Danny Dias
• [asterisk-users] How to track a call result originated from originate AMI command   Jim Dickenson
• [asterisk-users] Playback during call   Jim Dickenson
• [asterisk-users] Early media and IAX2   Russ Dill
• [asterisk-users] real-time queue problems   Cristian Dimache
• [asterisk-users] Opensource Speech recognition for Asterisk   Tim Dobson
• [asterisk-users] Asterisk on AMD   Doug
• [asterisk-users] Make a transfer for external line.   Gustavo Duarte
• [asterisk-users] Make a transfer for external line.   Gustavo Duarte
• [asterisk-users] Make a transfer for external line.   Gustavo Duarte
• [asterisk-users] Make a transfer for external line.   Gustavo Duarte
• [asterisk-users] Fwd: Re: Make a transfer for external line.   Gustavo Duarte
• [asterisk-users] Use of Storage Area Network with Asterisk   Michelle Dupuis
• [asterisk-users] NVidia component out   Michelle Dupuis
• [asterisk-users] Pickup parcked call from Aastra 9480i ct cordless   Michelle Dupuis
• [asterisk-users] Caller ID issue   Steve Edwards
• [asterisk-users] How to reuse mysql connection between AGI's   Steve Edwards
• [asterisk-users] check channels   Steve Edwards
• [asterisk-users] Connecting two calls with Originate   Steve Edwards
• [asterisk-users] 'System' application in asterisk   Steve Edwards
• [asterisk-users] 'System' application in asterisk   Steve Edwards
• [asterisk-users] channel variables in AGI   Steve Edwards
• [asterisk-users] OT: UK PPP certification -- what is it?   Steve Edwards
• [asterisk-users] OT: UK PPP certification -- what is it?   Steve Edwards
• [asterisk-users] Create File Directory   Steve Edwards
• [asterisk-users] sending sms from Asterisk server   Steve Edwards
• [asterisk-users] Codec choice   Steve Edwards
• [asterisk-users] Executing system commands through Manager API   Steve Edwards
• [asterisk-users] MySQL Connect problem...   Steve Edwards
• [asterisk-users] setting variable for a DID number   Steve Edwards
• [asterisk-users] sending sms from Asterisk server   Steve Edwards
• [asterisk-users] setting variable for a DID number   Steve Edwards
• [asterisk-users] setting variable for a DID number   Steve Edwards
• [asterisk-users] Codec choice   Steve Edwards
• [asterisk-users] Codec choice   Steve Edwards
• [asterisk-users] channel variables in AGI   Steve Edwards
• [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):   Steve Edwards
• [asterisk-users] help for a new user   Steve Edwards
• [asterisk-users] help for a new user   Steve Edwards
• [asterisk-users] Use of AGISIGHUP   Steve Edwards
• [asterisk-users] Use of AGISIGHUP   Steve Edwards
• [asterisk-users] Asterisk Crashed - But why?   Steve Edwards
• [asterisk-users] Play a number of files to a caller   Steve Edwards
• [asterisk-users] Play a number of files to a caller   Steve Edwards
• [asterisk-users] Running System() after call completion, not in 'h'?   Steve Edwards
• [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?   Lenz Emilitri
• [asterisk-users] How to track a call result originated from originate AMI command   Lenz Emilitri
• [asterisk-users] How to track a call result originated from originate AMI command   Lenz Emilitri
• [asterisk-users] Asterisk routing to SoftSwitch   Jose P. Espinal
• [asterisk-users] RTP stream not passing through router with port forwarding   C F
• [asterisk-users] Caller ID issue   C F
• [asterisk-users] FYI: Seen the 2600Hz announcement?   C F
• [asterisk-users] Correct Caller-ID   C F
• [asterisk-users] Correct Caller-ID   C F
• [asterisk-users] How to do voice barge in using asterisk server   C F
• [asterisk-users] How to reuse mysql connection between AGI's   Faheem
• [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?   Faheem
• [asterisk-users] redirect based on incoming number   Barry Fawthrop
• [asterisk-users] Cisco 9971   Sascha Ferley
• [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6   Alex Ferrara
• [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6   Alex Ferrara
• [asterisk-users] How does deny/permit work in sip.conf?   Bruce Ferrell
• [asterisk-users] How does deny/permit work in sip.conf?   Bruce Ferrell
• [asterisk-users] How does deny/permit work in sip.conf?   Bruce Ferrell
• [asterisk-users] Looking for MIB description   Bruce Ferrell
• [asterisk-users] Migrating 1.4 to 1.6.2   Bruce Ferrell
• [asterisk-users] Youmail RDNIS   Karl Fife
• [asterisk-users] setting variable for a DID number   Karl Fife
• [asterisk-users] # -key not to be 'transfer'   Felipe Figueiredo
• [asterisk-users] COnfig File question   Felipe Figueiredo
• [asterisk-users] check channels   Felipe Figueiredo
• [asterisk-users] check channels   Felipe Figueiredo
• [asterisk-users] check channels   Felipe Figueiredo
• [asterisk-users] Skype   Felipe Figueiredo
• [asterisk-users] Any Free software that can connect to an Asterisk Server and Do video Conferencing?   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6 without DAHDI   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6 without DAHDI   Kevin P. Fleming
• [asterisk-users] Asterisk on AMD   Kevin P. Fleming
• [asterisk-users] Asterisk on AMD   Kevin P. Fleming
• [asterisk-users] Asterisk on AMD   Kevin P. Fleming
• [asterisk-users] IAX2 - Separate Signaling and Media?   Kevin P. Fleming
• [asterisk-users] Why does Digium not respect their own development guidelines?   Kevin P. Fleming
• [asterisk-users] Play a number of files to a caller   Kevin P. Fleming
• [asterisk-users] Fax/Modem, Asterisk, Channel Banks   Dave Fullerton
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   Jian Gao
• [asterisk-users] realtime sip peers : musiconhold class   Tiago Geada
• [asterisk-users] CDR variables   Tiago Geada
• [asterisk-users] CDR variables   Tiago Geada
• [asterisk-users] sending sms from Asterisk server   Tiago Geada
• [asterisk-users] DAHDI config file system.conf   Jerry Geis
• [asterisk-users] PRI errors no D channel   Jerry Geis
• [asterisk-users] Any Free software that can connect to an Asterisk Server and Do video Conferencing?   Siju George
• [asterisk-users] Transfer + speed dial button problem?   Gerard
• [asterisk-users] redirect based on incoming number   Lyle Giese
• [asterisk-users] sip probe syntax   Lyle Giese
• [asterisk-users] Asterisk on Ben NanoNote?   Gilles
• [asterisk-users] Asterisk on Ben NanoNote?   Gilles
• [asterisk-users] Codec Conversion   Michael Graves
• [asterisk-users] Codec Conversion   Michael Graves
• [asterisk-users] Calling Line Identity - any ideas   Paddy Grice
• [asterisk-users] Calling Line Identity - any ideas   Paddy Grice
• [asterisk-users] Calling Line Identity - any ideas   Paddy Grice
• [asterisk-users] Calling Line Identity - any ideas   Paddy Grice
• [asterisk-users] Calling Line Identity - any ideas   Paddy Grice
• [asterisk-users] Prepay Limited Calls.   Faisal Hanif
• [asterisk-users] check channels   Faisal Hanif
• [asterisk-users] speciality of SIPp and SER(Sip Express Router)   Faisal Hanif
• [asterisk-users] speciality of SIPp and SER(Sip Express Router)   Faisal Hanif
• [asterisk-users] asterisk compatible cards?   Faisal Hanif
• [asterisk-users] How to determine which party hangup the call? cause of Hang-up needed.‏   Faisal Hanif
• [asterisk-users] How to determine which party hangup the call? cause of Hang-up needed.‏   Faisal Hanif
• [asterisk-users] How to determine which party hangup the call? cause of Hang-up needed.‏   Faisal Hanif
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   Faisal Hanif
• [asterisk-users] Playing with sipvicious ..   Dana Harding
• [asterisk-users] Need USA DIDs   Nyamul Hassan
• [asterisk-users] Dial option 'r' not working anymore?   Vlasis Hatzistavrou
• [asterisk-users] Asterisk Hardwares   Paul Hayes
• [asterisk-users] Playing with sipvicious ..   Paul Hayes
• [asterisk-users] asterisk compatible cards?   Gordon Henderson
• [asterisk-users] IAX softphone   Gordon Henderson
• [asterisk-users] What do you use for Invoicing?   Gordon Henderson
• [asterisk-users] Asterisk and RAID   Gordon Henderson
• [asterisk-users] asterisk compatible cards?   Gordon Henderson
• [asterisk-users] Asterisk on AMD   Gordon Henderson
• [asterisk-users] Asterisk with Motorola Canopy   Gordon Henderson
• [asterisk-users] Playing with sipvicious ..   Gordon Henderson
• [asterisk-users] Playing with sipvicious ..   Gordon Henderson
• [asterisk-users] Opensource Speech recognition for Asterisk   Gordon Henderson
• [asterisk-users] DAHDI not detecting caller hangup   Gordon Henderson
• [asterisk-users] Attempted SIP connection by foreign host. Help!   Gordon Henderson
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Gordon Henderson
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Gordon Henderson
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Gordon Henderson
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Gordon Henderson
• [asterisk-users] DAHDI not detecting caller hangup   Gordon Henderson
• [asterisk-users] Protect yourself   Gordon Henderson
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   Gordon Henderson
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   Gordon Henderson
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   Gordon Henderson
• [asterisk-users] Yes it is a dimensioning question! Atom CPU   Gordon Henderson
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   Gordon Henderson
• [asterisk-users] Duplicate channel variables after transfer   Alex Hermann
• [asterisk-users] asterisk on Vmware   Rob Hillis
• [asterisk-users] sending sms from Asterisk server   Johann Hoehn
• [asterisk-users] ASterisk CDR file Master.csv   Dean Hoover
• [asterisk-users] install asterisk   Steve Howes
• [asterisk-users] Codec choice   Steve Howes
• [asterisk-users] Yes it is a dimensioning question! Atom CPU   Steve Howes
• [asterisk-users] Yes it is a dimensioning question! Atom CPU   Steve Howes
• [asterisk-users] Yes it is a dimensioning question! Atom CPU   Steve Howes
• [asterisk-users] What do you use for Invoicing?   Brian C. Huffman
• [asterisk-users] FAX Options   Alejandro Imass
• [asterisk-users] FAX Options   Alejandro Imass
• [asterisk-users] FAX Options   Alejandro Imass
• [asterisk-users] IXJ Quicknet PhoneJack issues   Infra
• [asterisk-users] IXJ Quicknet PhoneJack issues   Infra
• [asterisk-users] IXJ issues on 1.4.35   Infra
• [asterisk-users] IXJ issues on 1.4.35   Infra
• [asterisk-users] IXJ issues on 1.4.35   Infra
• [asterisk-users] outbound SIP trunk hunting (or any fxo for that matter)   Infra
• [asterisk-users] IAX softphone   Nasir Iqbal
• [asterisk-users] What do you use for Invoicing?   Nasir Iqbal
• [asterisk-users] What do you use for Invoicing?   Nasir Iqbal
• [asterisk-users] How to reuse mysql connection between AGI's   Nasir Iqbal
• [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?   Nasir Iqbal
• [asterisk-users] [Asterisk-Users] How do I install speex for   Nasir Iqbal
• [asterisk-users] AMD setup in Astersik   Nasir Iqbal
• [asterisk-users] Monitor asterisk   Nasir Iqbal
• [asterisk-users] How to track a call result originated from originate AMI command   Nasir Iqbal
• [asterisk-users] Monitor asterisk   Nasir Iqbal
• [asterisk-users] Prepay Limited Calls.   Nasir Iqbal
• [asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?   Nasir Iqbal
• [asterisk-users] MySQL Connect problem...   Nasir Iqbal
• [asterisk-users] realtime sip peers : musiconhold class   Nasir Iqbal
• [asterisk-users] Calling Line Identity - any ideas   Nasir Iqbal
• [asterisk-users] Calling Line Identity - any ideas   Nasir Iqbal
• [asterisk-users] Calling Line Identity - any ideas   Nasir Iqbal
• [asterisk-users] Calling Line Identity - any ideas   Nasir Iqbal
• [asterisk-users] Calling Line Identity - any ideas   Nasir Iqbal
• [asterisk-users] dial_exec_full problems with TDM400 - getting critical.   Ira
• [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3   Ira
• [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3   Ira
• [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3   Ira
• [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3   Ira
• [asterisk-users] How to record a file and play some other file at the same time   Motiejus Jakštys
• [asterisk-users] AMI Command   Motiejus Jakštys
• [asterisk-users] Identify remote prompts: Partial audio matching?   Motiejus Jakštys
• [asterisk-users] How to track a call result originated from originate AMI command   Motiejus Jakštys
• [asterisk-users] realtime sip peers : musiconhold class   Motiejus Jakštys
• [asterisk-users] outbound SIP trunk hunting (or any fxo for that matter)   Motiejus Jakštys
• [asterisk-users] Asterisk Crashes - Segmentation Fault   Manmohan Singh Jandu
• [asterisk-users] Asterisk Crashes - Segmentation Fault   Manmohan Singh Jandu
• [asterisk-users] Asterisk Crashes - Segmentation Fault   Manmohan Singh Jandu
• [asterisk-users] RTP stream not passing through router with port forwarding   Nasir Javaid
• [asterisk-users] asterisk-users Digest, Vol 73, Issue 5   Nasir Javaid
• [asterisk-users] Can ChanIsAvail return status from sip uri using router ip   Nasir Javaid
• [asterisk-users] sending sms from Asterisk server   Per Jessen
• [asterisk-users] Youmail RDNIS   Andrew Joakimsen
• [asterisk-users] Why does Digium not respect their own development guidelines?   Andrew Joakimsen
• [asterisk-users] asterisk & distributed device state => res_jabber Versus res_ais   Olle E. Johansson
• [asterisk-users] asterisk & distributed device state => res_jabber Versus res_ais   Olle E. Johansson
• [asterisk-users] Connecting two calls with Originate   Kathryn Jones
• [asterisk-users] Connecting two calls with Originate   Kathryn Jones
• [asterisk-users] Connecting two calls with Originate   Kathryn Jones
• [asterisk-users] WaitExten() always times out   Kathryn Jones
• [asterisk-users] WaitExten() always times out   Kathryn Jones
• [asterisk-users] WaitExten() always times out   Kathryn Jones
• [asterisk-users] WaitExten() always times out   Kathryn Jones
• [asterisk-users] WaitExten() always times out   Kathryn Jones
• [asterisk-users] WaitExten() always times out   Kathryn Jones
• [asterisk-users] WaitExten() always times out   Kathryn Jones
• [asterisk-users] WaitExten() always times out   Kathryn Jones
• [asterisk-users] Realtime Context   Dan Journo
• [asterisk-users] Create File Directory   Dan Journo
• [asterisk-users] Create File Directory   Dan Journo
• [asterisk-users] Create File Directory   Dan Journo
• [asterisk-users] Include and Realtime   Dan Journo
• [asterisk-users] Include and Realtime   Dan Journo
• [asterisk-users] Include and Realtime   Dan Journo
• [asterisk-users] Include and Realtime   Dan Journo
• [asterisk-users] Include and Realtime   Dan Journo
• [asterisk-users] CDR Help   Dan Journo
• [asterisk-users] CDR Help   Dan Journo
• [asterisk-users] CDR Help   Dan Journo
• [asterisk-users] Call Forwarding   Dan Journo
• [asterisk-users] Call Forwarding   Dan Journo
• [asterisk-users] Pickup parcked call from Aastra 9480i ctcordless   Dan Journo
• [asterisk-users] COnfig File question   Ujjval Karihaloo
• [asterisk-users] COnfig File question   Ujjval Karihaloo
• [asterisk-users] rolling over Master.csv CDR File   Ujjval Karihaloo
• [asterisk-users] Call-limit field   Ujjval Karihaloo
• [asterisk-users] ASterisk CDR file Master.csv   Ujjval Karihaloo
• [asterisk-users] ASterisk CDR file Master.csv   Ujjval Karihaloo
• [asterisk-users] Monitor asterisk   Kevin Keane
• [asterisk-users] asterisk on Vmware   Kevin Keane
• [asterisk-users] # -key not to be 'transfer'   Jonas Kellens
• [asterisk-users] # -key not to be 'transfer'   Jonas Kellens
• [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)   Jonas Kellens
• [asterisk-users] sip.conf register in realtime DB   Jonas Kellens
• [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)   Jonas Kellens
• [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)   Jonas Kellens
• [asterisk-users] sip.conf register in realtime DB   Jonas Kellens
• [asterisk-users] sip.conf register in realtime DB   Jonas Kellens
• [asterisk-users] sip.conf register in realtime DB   Jonas Kellens
• [asterisk-users] sip.conf register in realtime DB   Jonas Kellens
• [asterisk-users] Call agent when queue is empty and there is a voicemail left   Jonas Kellens
• [asterisk-users] VM Extension : asterisk   Jonas Kellens
• [asterisk-users] VM Extension : asterisk   Jonas Kellens
• [asterisk-users] Call agent when queue is empty and there is a voicemail left   Jonas Kellens
• [asterisk-users] Recording the conversation with MixMonitor() ends when the call is transfered   Jonas Kellens
• [asterisk-users] realtime sip peers : musiconhold class   Jonas Kellens
• [asterisk-users] realtime sip peers : musiconhold class   Jonas Kellens
• [asterisk-users] realtime sip peers : musiconhold class   Jonas Kellens
• [asterisk-users] realtime sip peers : musiconhold class   Jonas Kellens
• [asterisk-users] realtime sip peers : musiconhold class   Jonas Kellens
• [asterisk-users] realtime sip peers : musiconhold class   Jonas Kellens
• [asterisk-users] realtime sip peers : musiconhold class   Jonas Kellens
• [asterisk-users] Convert wav-file to alaw-file   Jonas Kellens
• [asterisk-users] Add & play moh-files without reload   Jonas Kellens
• [asterisk-users] Add & play moh-files without reload   Jonas Kellens
• [asterisk-users] realtime sip peers : musiconhold class   Jonas Kellens
• [asterisk-users] Recording the conversation with MixMonitor() ends when the call is transfered   Jonas Kellens
• [asterisk-users] MusicOnHold class working for internal calls, not for external   Jonas Kellens
• [asterisk-users] only part of dialplan available   Jonas Kellens
• [asterisk-users] Asterisk does not translate from wav to alaw   Jonas Kellens
• [asterisk-users] only part of dialplan available   Jonas Kellens
• [asterisk-users] What do you use for Invoicing?   Don Kelly
• [asterisk-users] Asterisk, HylaFax and Cardiff   Don Kelly
• [asterisk-users] Asterisk, HylaFax and Cardiff   Don Kelly
• [asterisk-users] Asterisk, HylaFax and Cardiff   Don Kelly
• [asterisk-users] Femtocell to VoIP?   Steve Kennedy
• [asterisk-users] Femtocell to VoIP?   Steve Kennedy
• [asterisk-users] OT: UK PPP certification -- what is it?   Steve Kennedy
• [asterisk-users] sip probe syntax   Matt Kershnar
• [asterisk-users] SIP Status: 401 Unauthorized (0 bindings)   Kyle Kienapfel
• [asterisk-users] How can i switch to samba server omitting sshfs   Kyle Kienapfel
• [asterisk-users] Asterisk and TV media server   Kyle Kienapfel
• [asterisk-users] asterisknow   Kyle Kienapfel
• [asterisk-users] fail2ban does not work for my asterisk installation   Kyle Kienapfel
• [asterisk-users] Asterisk on AMD   Kyle Kienapfel
• [asterisk-users] evil disconnect of call with cisco 1760   Jeremy Kister
• [asterisk-users] Announcing Adhearsion 0.8.5   Ben Klang
• [asterisk-users] Opensource Speech recognition for Asterisk   Bob Kleiner
• [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)   Philipp von Klitzing
• [asterisk-users] mapping of disconnect reasons   Philipp von Klitzing
• [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)   Philipp von Klitzing
• [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)   Philipp von Klitzing
• [asterisk-users] outboundproxy timeout or qualify   Philipp von Klitzing
• [asterisk-users] mapping of disconnect reasons   Philipp von Klitzing
• [asterisk-users] Identify remote prompts: Partial audio matching?   Philipp von Klitzing
• [asterisk-users] Identify remote prompts: Partial audio matching?   Philipp von Klitzing
• [asterisk-users] Can ChanIsAvail return status from sip uri using router ip   Philipp von Klitzing
• [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)   Philipp von Klitzing
• [asterisk-users] Dial option 'r' not working anymore?   Philipp von Klitzing
• [asterisk-users] Re: How to determine which party hangup the call? cause of Hang-up needed.‏   Philipp von Klitzing
• [asterisk-users] Re: How to determine which party hangup the call? cause of Hang-up needed.‏   Philipp von Klitzing
• [asterisk-users] Asterisk on AMD   Philipp von Klitzing
• [asterisk-users] parkcall: How to remove announcement.   Philipp von Klitzing
• [asterisk-users] Asterisk 1.8.0-beta4 Now Available   Philipp von Klitzing
• [asterisk-users] outbound SIP trunk hunting (or any fxo for that matter)   Philipp von Klitzing
• [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6   Philipp von Klitzing
• [asterisk-users] alaw.h in app_meetme.c   Daniel Knoll
• [asterisk-users] iax stresstest client   Daniel Knoll
• [asterisk-users] asterisk on Vmware   Philipp Kolmann
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Bruce Komito
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Bruce Komito
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Bruce Komito
• [asterisk-users] What do you use for Invoicing?   Jeff LaCoursiere
• [asterisk-users] What do you use for Invoicing?   Jeff LaCoursiere
• [asterisk-users] Opensource Speech recognition for Asterisk   Jeff LaCoursiere
• [asterisk-users] Opensource Speech recognition for Asterisk   Jeff LaCoursiere
• [asterisk-users] Asterisk, HylaFax and Cardiff   Jeff LaCoursiere
• [asterisk-users] Polling DND status of a Linksys SPA9xx/5xx phone?   James Lamanna
• [asterisk-users] chinaroby fxo card - never heard of them   Landy Landy
• [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):   Rodrigo Lang
• [asterisk-users] Garbled messages - format_wav_gsm.c:414 wav_read: Short read (60) (No such file or directory)!   Bobby Larson
• [asterisk-users] rolling over Master.csv CDR File   Andrew Latham
• [asterisk-users] 1.6 and asterisk gui   Andrew Latham
• [asterisk-users] Yes it is a dimensioning question! Atom CPU   Andrew Latham
• [asterisk-users] Yes it is a dimensioning question! Atom CPU   Andrew Latham
• [asterisk-users] installing with yum   Geraint Lee
• [asterisk-users] MySQL Connect problem...   Geraint Lee
• [asterisk-users] MySQL Connect problem...   Geraint Lee
• [asterisk-users] MySQL Connect problem...   Geraint Lee
• [asterisk-users] Codec choice   Geraint Lee
• [asterisk-users] CDR on Transfer...   Geraint Lee
• [asterisk-users] asterisk-users Digest, Vol 73, Issue 24   Cédric Lemarchand
• [asterisk-users] asterisk-users Digest, Vol 73, Issue 58   Jonathan Leong
• [asterisk-users] asterisk-users Digest, Vol 73, Issue 58   Jonathan Leong
• [asterisk-users] Exporting Blacklist database   Tilghman Lesher
• [asterisk-users] mapping of disconnect reasons   Tilghman Lesher
• [asterisk-users] Dial() M parameter in 1.6.2.11-rc2   Tilghman Lesher
• [asterisk-users] AMI Command   Tilghman Lesher
• [asterisk-users] 'System' application in asterisk   Tilghman Lesher
• [asterisk-users] Correct Caller-ID   Tilghman Lesher
• [asterisk-users] DEBUG: Cannot find variable 'XXX' ??   Tilghman Lesher
• [asterisk-users] billsec exceeds duration on some calls   Tilghman Lesher
• [asterisk-users] billsec exceeds duration on some calls   Tilghman Lesher
• [asterisk-users] colored CLI with reattach   Tilghman Lesher
• [asterisk-users] IXJ Quicknet PhoneJack issues   Tilghman Lesher
• [asterisk-users] WaitExten() always times out   Tilghman Lesher
• [asterisk-users] Opensource Speech recognition for Asterisk   Tilghman Lesher
• [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3   Tilghman Lesher
• [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3   Tilghman Lesher
• [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3   Tilghman Lesher
• [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3   Tilghman Lesher
• [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3   Tilghman Lesher
• [asterisk-users] can't build resODBC on SUSE 11.3   Tilghman Lesher
• [asterisk-users] How to debug this specific issue?   Tilghman Lesher
• [asterisk-users] ODBC Voicemail storage   Tilghman Lesher
• [asterisk-users] How to debug this specific issue?   Tilghman Lesher
• [asterisk-users] Timecondition fallthrough on 2nd GSM Modem, First modem and ZAP's are all fine   Tilghman Lesher
• [asterisk-users] only part of dialplan available   Tilghman Lesher
• [asterisk-users] Asterisk does not translate from wav to alaw   Tilghman Lesher
• [asterisk-users] Asterisk does not translate from wav to alaw   Tilghman Lesher
• [asterisk-users] Why does Digium not respect their own development guidelines?   Tilghman Lesher
• [asterisk-users] Play a number of files to a caller   Tilghman Lesher
• [asterisk-users] asterisk core dump   Tilghman Lesher
• [asterisk-users] asterisk-users Digest, Vol 73, Issue 63   David Cook (Asterisk List)
• [asterisk-users] 4 Port FXO interface   Eric Merkel (Mail Lists)
• [asterisk-users] Playback during call   Gabriel Ortiz Lour
• [asterisk-users] Playback during call   Gabriel Ortiz Lour
• [asterisk-users] 'System' application in asterisk   Julian Lyndon-Smith
• [asterisk-users] Mobile answer machine cut off   Julian Lyndon-Smith
• [asterisk-users] Mobile answer machine cut off   Julian Lyndon-Smith
• [asterisk-users] NVidia component out   Julian Lyndon-Smith
• [asterisk-users] Mobile answer machine cut off   Julian Lyndon-Smith
• [asterisk-users] Play a number of files to a caller   Julian Lyndon-Smith
• [asterisk-users] Play a number of files to a caller   Julian Lyndon-Smith
• [asterisk-users] Play a number of files to a caller   Julian Lyndon-Smith
• [asterisk-users] Play a number of files to a caller   Julian Lyndon-Smith
• [asterisk-users] Exporting Blacklist database   Doug Lytle
• [asterisk-users] Fax/Modem, Asterisk, Channel Banks   Doug Lytle
• [asterisk-users] PRI errors no D channel   Doug Lytle
• [asterisk-users] PRI errors no D channel   Doug Lytle
• [asterisk-users] install asterisk   Doug Lytle
• [asterisk-users] installing with yum   Doug Lytle
• [asterisk-users] Opensource Speech recognition for Asterisk   Doug Lytle
• [asterisk-users] Asterisk, HylaFax and Cardiff   Doug Lytle
• [asterisk-users] Asterisk, HylaFax and Cardiff   Doug Lytle
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Doug Lytle
• [asterisk-users] dynamic MeetMe, min. digits   Doug Lytle
• [asterisk-users] dynamic MeetMe, min. digits   Doug Lytle
• [asterisk-users] What do you use for Invoicing?   Leif Madsen
• [asterisk-users] asterisk on Vmware   Leif Madsen
• [asterisk-users] asterisk & distributed device state => res_jabber Versus res_ais   Leif Madsen
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Leif Madsen
• [asterisk-users] rolling over Master.csv CDR File   Ishfaq Malik
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   Ishfaq Malik
• [asterisk-users] Realtime Context   Ishfaq Malik
• [asterisk-users] Realtime Context   Ishfaq Malik
• [asterisk-users] Realtime Context   Ishfaq Malik
• [asterisk-users] Realtime Context   Ishfaq Malik
• [asterisk-users] Executing system commands through Manager API   Ishfaq Malik
• [asterisk-users] Attempted SIP connection by foreign host. Help!   Ishfaq Malik
• [asterisk-users] Include and Realtime   Ishfaq Malik
• [asterisk-users] Include and Realtime   Ishfaq Malik
• [asterisk-users] CDR Help   Ishfaq Malik
• [asterisk-users] MusicOnHold class working for internal calls, not for external   Ishfaq Malik
• [asterisk-users] CDR Help   Ishfaq Malik
• [asterisk-users] Fax/Modem, Asterisk, Channel Banks   Joel Maslak
• [asterisk-users] Dial() M parameter in 1.6.2.11-rc2   Joel Maslak
• [asterisk-users] Use of Storage Area Network with Asterisk   Joel Maslak
• [asterisk-users] Femtocell to VoIP?   Matt
• [asterisk-users] Femtocell to VoIP?   Matt
• [asterisk-users] Femtocell to VoIP?   Matt
• [asterisk-users] Correct Caller-ID   Matt
• [asterisk-users] Correct Caller-ID   Matt
• [asterisk-users] Correct Caller-ID   Matt
• [asterisk-users] Asterisk voicemail server - gsm notifications   Matt
• [asterisk-users] AMR Codec   Matt
• [asterisk-users] MySQL Connect problem...   Sherwood McGowan
• [asterisk-users] Calling Line Identity - any ideas   Sherwood McGowan
• [asterisk-users] Directory routing to wrong extension if dial tones are pressed too quick.   Sherwood McGowan
• [asterisk-users] Calling Line Identity - any ideas   Sherwood McGowan
• [asterisk-users] Codec choice   Sherwood McGowan
• [asterisk-users] Codec choice   Sherwood McGowan
• [asterisk-users] Calling Line Identity - any ideas   Sherwood McGowan
• [asterisk-users] MySQL Connect problem...   Sherwood McGowan
• [asterisk-users] Executing system commands through Manager API   Sherwood McGowan
• [asterisk-users] codec_g729.so not work!   Sherwood McGowan
• [asterisk-users] Calling Line Identity - any ideas   Sherwood McGowan
• [asterisk-users] MySQL Connect problem...   Sherwood McGowan
• [asterisk-users] setting variable for a DID number   Sherwood McGowan
• [asterisk-users] Calling Line Identity - any ideas   Sherwood McGowan
• [asterisk-users] setting variable for a DID number   Sherwood McGowan
• [asterisk-users] setting variable for a DID number   Sherwood McGowan
• [asterisk-users] Codec choice   Sherwood McGowan
• [asterisk-users] Calling Line Identity - any ideas   Sherwood McGowan
• [asterisk-users] Codec choice   Sherwood McGowan
• [asterisk-users] How to prevent soft hangup from being necessary ?   Sherwood McGowan
• [asterisk-users] EMail on Missed Call   Sherwood McGowan
• [asterisk-users] Asterisk and OCS2007 R2   Lyle McKarns
• [asterisk-users] Asterisk on AMD   Lyle McKarns
• [asterisk-users] Asterisk on AMD   Lyle McKarns
• [asterisk-users] Asterisk on AMD   Lyle McKarns
• [asterisk-users] Asterisk on AMD   Lyle McKarns
• [asterisk-users] Asterisk not working with Festival   Davinder Kumar Meen
• [asterisk-users] Asterisk not working with Festival   Davinder Kumar Meen
• [asterisk-users] Asterisk not working with Festival   Davinder Kumar Meen
• [asterisk-users] Asterisk not working with Festival   Davinder Kumar Meen
• [asterisk-users] channel variables in AGI   Anthony Messina
• [asterisk-users] Click2call from an OpenOffice document   Anthony Messina
• [asterisk-users] channel variables in AGI   Anthony Messina
• [asterisk-users] Detecting Called party Ring indication (and act on it)   Juan Miguel
• [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?   Mike
• [asterisk-users] Directory routing to wrong extension if dial tones are pressed too quick.   Eddie Mikell
• [asterisk-users] Asterisk on Ben NanoNote?   Barry Miller
• [asterisk-users] Web-meetme   Flavio Miranda
• [asterisk-users] No CDR with originate from manager and then an redirect to a dial from manager   Arjan Kroon | Mobillion
• [asterisk-users] Codec Conversion   Miguel Molina
• [asterisk-users] Codec Conversion   Miguel Molina
• [asterisk-users] Codec Conversion   Miguel Molina
• [asterisk-users] WaitExten() always times out   Miguel Molina
• [asterisk-users] Recording the conversation with MixMonitor() ends when the call is transfered   Miguel Molina
• [asterisk-users] WaitExten() always times out   Miguel Molina
• [asterisk-users] WaitExten() always times out   Miguel Molina
• [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3   Miguel Molina
• [asterisk-users] dial_exec_full problems with TDM400   Jason Morgan
• [asterisk-users] dial_exec_full problems with TDM400   Jason Morgan
• [asterisk-users] dial_exec_full problems with TDM400 - getting critical.   Jason Morgan
• [asterisk-users] TELUS British Columbia PRI Settings   Charles Moye
• [asterisk-users] TELUS British Columbia PRI Settings   Charles Moye
• [asterisk-users] how to place a call on hold and play music on hold using agi   Janu Mukherjee
• [asterisk-users] How can i switch to samba server omitting sshfs   Janu Mukherjee
• [asterisk-users] how to place a call on hold and play music on hold using agi   Janu Mukherjee
• [asterisk-users] How to record a file and play some other file at the same time   Janu Mukherjee
• [asterisk-users] How to do barging using asterisk server.   Janu Mukherjee
• [asterisk-users] How to do voice barge in using asterisk server   Janu Mukherjee
• [asterisk-users] CDR: MySQL query   RSCL Mumbai
• [asterisk-users] CDR: MySQL query   RSCL Mumbai
• [asterisk-users] Asterisk does not translate from wav to alaw   Steve Murphy
• [asterisk-users] Asterisk does not translate from wav to alaw   Steve Murphy
• [asterisk-users] help with dialplan   Steve Murphy
• [asterisk-users] help with dialplan   Steve Murphy
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   Nikhil Nair
• [asterisk-users] FAX Options   Tim Nelson
• [asterisk-users] Fax/Modem, Asterisk, Channel Banks   Tim Nelson
• [asterisk-users] Codec Conversion   Tim Nelson
• [asterisk-users] Codec Conversion   Tim Nelson
• [asterisk-users] billsec exceeds duration on some calls   Tim Nelson
• [asterisk-users] codec_g729.so not work!   Tim Nelson
• [asterisk-users] IAX2 - Separate Signaling and Media?   Tim Nelson
• [asterisk-users] IAX2 - Separate Signaling and Media?   Tim Nelson
• [asterisk-users] 1.6 and asterisk gui   Tim Nelson
• [asterisk-users] ASterisk CDR file Master.csv   Tim Nelson
• [asterisk-users] Asterisk Crashed - But why?   Tim Nelson
• [asterisk-users] Web-meetme   Tim Nelson
• [asterisk-users] Running System() after call completion, not in 'h'?   Tim Nelson
• [asterisk-users] Running System() after call completion, not in 'h'?   Tim Nelson
• [asterisk-users] how to place a call on hold and play music on holdusing agi   Danny Nicholas
• [asterisk-users] Stupid Macro question   Danny Nicholas
• [asterisk-users] FAX Options   Danny Nicholas
• [asterisk-users] FAX Options   Danny Nicholas
• [asterisk-users] Garbled messages - format_wav_gsm.c:414 wav_read:Short read (60) (No such file or directory)!   Danny Nicholas
• [asterisk-users] Dial() M parameter in 1.6.2.11-rc2   Danny Nicholas
• [asterisk-users] Dial() M parameter in 1.6.2.11-rc2   Danny Nicholas
• [asterisk-users] Using SIP to dial extension that will give anoutside line   Danny Nicholas
• [asterisk-users] How to record a file and play some other file atthe same time   Danny Nicholas
• [asterisk-users] Tweaking AMD in Asterisk   Danny Nicholas
• [asterisk-users] Asterisk and RAID   Danny Nicholas
• [asterisk-users] Tweaking AMD in Asterisk   Danny Nicholas
• [asterisk-users] Identify remote prompts: Partial audio matching?   Danny Nicholas
• [asterisk-users] AMI Command   Danny Nicholas
• [asterisk-users] COnfig File question   Danny Nicholas
• [asterisk-users] Using a 1.4 config with 1.6   Danny Nicholas
• [asterisk-users] Asterisk 1.4 and TE420P   Danny Nicholas
• [asterisk-users] Reinstalling Asterisk due to hardware changes   Danny Nicholas
• [asterisk-users] Scilence problem on running call   Danny Nicholas
• [asterisk-users] SIP response 500 "Server Internal Error"   Danny Nicholas
• [asterisk-users] Connecting two calls with Originate   Danny Nicholas
• [asterisk-users] Connecting two calls with Originate   Danny Nicholas
• [asterisk-users] 'System' application in asterisk   Danny Nicholas
• [asterisk-users] check channels   Danny Nicholas
• [asterisk-users] check channels   Danny Nicholas
• [asterisk-users] Correct Caller-ID   Danny Nicholas
• [asterisk-users] Correct Caller-ID   Danny Nicholas
• [asterisk-users] Correct Caller-ID   Danny Nicholas
• [asterisk-users] Asterisk not working with Festival   Danny Nicholas
• [asterisk-users] VM Extension : asterisk   Danny Nicholas
• [asterisk-users] VM Extension : asterisk   Danny Nicholas
• [asterisk-users] Problems with meetme in 1.4.26   Danny Nicholas
• [asterisk-users] Problems with meetme in 1.4.26   Danny Nicholas
• [asterisk-users] Callback script anyone   Danny Nicholas
• [asterisk-users] Callback script anyone   Danny Nicholas
• [asterisk-users] install asterisk   Danny Nicholas
• [asterisk-users] installing with yum   Danny Nicholas
• [asterisk-users] colored CLI with reattach   Danny Nicholas
• [asterisk-users] parkcall: How to remove announcement.   Danny Nicholas
• [asterisk-users] parkcall: How to remove announcement.   Danny Nicholas
• [asterisk-users] parkcall: How to remove announcement.   Danny Nicholas
• [asterisk-users] Polycom 331 freezes connecting to FreePBX   Danny Nicholas
• [asterisk-users] Convert wav-file to alaw-file   Danny Nicholas
• [asterisk-users] Add & play moh-files without reload   Danny Nicholas
• [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for allinbound trunks   Danny Nicholas
• [asterisk-users] WaitExten() always times out   Danny Nicholas
• [asterisk-users] WaitExten() always times out   Danny Nicholas
• [asterisk-users] WaitExten() always times out   Danny Nicholas
• [asterisk-users] CDR variables   Danny Nicholas
• [asterisk-users] CDR variables   Danny Nicholas
• [asterisk-users] setting variable for a DID number   Danny Nicholas
• [asterisk-users] WaitExten() always times out   Danny Nicholas
• [asterisk-users] How to do voice barge in using asterisk server   Danny Nicholas
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Danny Nicholas
• [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):   Danny Nicholas
• [asterisk-users] Use of AGISIGHUP   Danny Nicholas
• [asterisk-users] Use of AGISIGHUP   Danny Nicholas
• [asterisk-users] ASterisk CDR file Master.csv   Danny Nicholas
• [asterisk-users] help with dialplan   Danny Nicholas
• [asterisk-users] help with dialplan   Danny Nicholas
• [asterisk-users] Pickup parcked call from Aastra 9480i ct cordless   Danny Nicholas
• [asterisk-users] Pickup parcked call from Aastra 9480ictcordless   Danny Nicholas
• [asterisk-users] No audio on call forward after upgrade fromAsterisk 1.4 to 1.6   Danny Nicholas
• [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?   Deepika Nijhawan
• [asterisk-users] [Asterisk-Users] How do I install speex for   Deepika Nijhawan
• [asterisk-users] Codec choice   Deepika Nijhawan
• [asterisk-users] Codec choice   Deepika Nijhawan
• [asterisk-users] Codec choice   Deepika Nijhawan
• [asterisk-users] Codec choice   Deepika Nijhawan
• [asterisk-users] chinaroby fxo card - never heard of them   John Novack
• [asterisk-users] Correct Caller-ID   John Novack
• [asterisk-users] Correct Caller-ID   John Novack
• [asterisk-users] install asterisk   John Novack
• [asterisk-users] dial_exec_full problems with TDM400 - getting critical.   John Novack
• [asterisk-users] DAHDI not detecting caller hangup   John Novack
• [asterisk-users] DAHDI compile warning   John Novack
• [asterisk-users] Asterisk and RAID   Alejandro Cabrera Obed
• [asterisk-users] Which Python binding to use ?   Olivier
• [asterisk-users] CDR report   Olivier
• [asterisk-users] Click2call from an OpenOffice document   Olivier
• [asterisk-users] Click2call from an OpenOffice document   Olivier
• [asterisk-users] How to prevent soft hangup from being necessary ?   Olivier
• [asterisk-users] OT - How to blacklist a driver in /etc/modprobe.d without reboot   Olivier
• [asterisk-users] OT - How to blacklist a driver in /etc/modprobe.d without reboot [SOLVED]   Olivier
• [asterisk-users] R: asterisk compatible cards?   Alexandru Oniciuc
• [asterisk-users] op_div: non-numeric argument   Positively Optimistic
• [asterisk-users] Loop Detection / SIP   Positively Optimistic
• [asterisk-users] SIP Debug Messages   Positively Optimistic
• [asterisk-users] Callback script anyone   J. Oquendo
• [asterisk-users] Callback script anyone   J. Oquendo
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   J. Oquendo
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   J. Oquendo
• [asterisk-users] Connecting two calls with Originate   Elliot Otchet
• [asterisk-users] Play a number of files to a caller   Elliot Otchet
• [asterisk-users] asterisk + openBTS   Tim Panton
• [asterisk-users] can't build resODBC on SUSE 11.3   Tim Panton
• [asterisk-users] asterisk + openBTS   Tim Panton
• [asterisk-users] can't build resODBC on SUSE 11.3   Tim Panton
• [asterisk-users] IAX2 - Separate Signaling and Media?   Tim Panton
• [asterisk-users] Caller ID issue   Peder
• [asterisk-users] Voicemail prompts fuzzy and quiet   Peder
• [asterisk-users] asterisk compatible cards?   Daniel Petre
• [asterisk-users] asterisk compatible cards?   Daniel Petre
• [asterisk-users] Whither app_nv_faxdetect   Philip Prindeville
• [asterisk-users] CDR report   Dario Quiroz
• [asterisk-users] fail2ban does not work for my asterisk installation   Randy R
• [asterisk-users] FYI: Seen the 2600Hz announcement?   Randy R
• [asterisk-users] Allison Smith Hilarity   Randy R
• [asterisk-users] VUC Friday 13th: Skype, then Video   Randy R
• [asterisk-users] asterisk + openBTS   Randy R
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   Randy R
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   Randy R
• [asterisk-users] sending sms from Asterisk server   Anton Raharja
• [asterisk-users] channel stay up when extension unreachable   Anton Raharja
• [asterisk-users] channel stay up when extension unreachable   Anton Raharja
• [asterisk-users] real-time queue problems   Alvaro Ramirez
• [asterisk-users] Asterisk + SMS + PRI   Ramu
• [asterisk-users] OT: UK PPP certification -- what is it?   Faris Raouf
• [asterisk-users] Dahdi install gone wrong   Todd Reese
• [asterisk-users] Dahdi install gone wrong   Todd Reese
• [asterisk-users] Dahdi install gone wrong   Todd Reese
• [asterisk-users] GXP-2000 transfer & hold problem   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] help with dialplan   Todd Reese
• [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?   Matt Riddell
• [asterisk-users] Good script to make appointment?   Matt Riddell
• [asterisk-users] How does deny/permit work in sip.conf?   Matt Riddell
• [asterisk-users] Monitor asterisk   Matt Riddell
• [asterisk-users] Convert wav-file to alaw-file   Matt Riddell
• [asterisk-users] Realtime Context   Matt Riddell
• [asterisk-users] IAX2 - Separate Signaling and Media?   Matt Riddell
• [asterisk-users] How to debug this specific issue?   Matt Riddell
• [asterisk-users] AMD message   Matt Riddell
• [asterisk-users] billsec exceeds duration on some calls   Matt Riddell
• [asterisk-users] Mobile answer machine cut off   Matt Riddell
• [asterisk-users] AMD message   Matt Riddell
• [asterisk-users] How to debug this specific issue?   Matt Riddell
• [asterisk-users] Logging the CID from the Privacy Manager   Matt Riddell
• [asterisk-users] Mobile answer machine cut off   Matt Riddell
• [asterisk-users] asterisk-1.8 problem with one-way audio with no nat   Matt Riddell
• [asterisk-users] Early media and IAX2   Matt Riddell
• [asterisk-users] Digest Username/auth name mismatch   Matt Riddell
• [asterisk-users] parkcall: How to remove announcement.   Alexandre Rodrigues
• [asterisk-users] parkcall: How to remove announcement.   Alexandre Rodrigues
• [asterisk-users] parkcall: How to remove announcement.   Alexandre Rodrigues
• [asterisk-users] AMI Command   Ron
• [asterisk-users] AMI Command   Ron
• [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932   Ron
• [asterisk-users] asterisk + cisco 3825 with ISDN   Ron
• [asterisk-users] asterisk + cisco 3825 with ISDN   Ron
• [asterisk-users] Pickup parcked call from Aastra 9480i ct cordless   Nicolas Ross
• [asterisk-users] Pickup parcked call from Aastra 9480i ctcordless   Nicolas Ross
• [asterisk-users] Pickup parcked call from Aastra9480i ctcordless   Nicolas Ross
• [asterisk-users] installing with yum   Matthew J. Roth
• [asterisk-users] colored CLI with reattach   Matthew J. Roth
• [asterisk-users] Set outgoing number in filename of the recordings   Rushikesh
• [asterisk-users] AMD setup in Astersik   Rushikesh
• [asterisk-users] Transfer to non registered extension creates call hangup   Rushikesh
• [asterisk-users] double DTMF digits   M S
• [asterisk-users] double DTMF digits   M S
• [asterisk-users] Prepay Limited Calls.   Catalin S.
• [asterisk-users] billsec exceeds duration on some calls   SIP
• [asterisk-users] Timecondition fallthrough on 2nd GSM Modem, First modem and ZAP's are all fine   Raimund Sacherer
• [asterisk-users] outboundproxy timeout or qualify   Abeed Saleh
• [asterisk-users] outboundproxy timeout or qualify   Abeed Saleh
• [asterisk-users] outboundproxy timeout or qualify   Abeed Saleh
• [asterisk-users] how to place a call on hold and play music on hold using agi   Abeed Saleh
• [asterisk-users] Asterisk 1.6.2 FastAGI Hangup Problem   Abeed Saleh
• [asterisk-users] BRI line issue on third call   Paulo Santos
• [asterisk-users] How to debug this specific issue?   Stefan Schmidt
• [asterisk-users] How to debug this specific issue?   Stefan Schmidt
• [asterisk-users] Call Forwarding   Stefan Schmidt
• [asterisk-users] # -key not to be 'transfer'   Mark Scholten
• [asterisk-users] FAX Options   Mark Scholten
• [asterisk-users] Asterisk (1.8-beta2) and SIP IPv4/IPv6 dual-stack possibilities   Wouter Schoot
• [asterisk-users] Polycom 331 freezes connecting to FreePBX   Ben Schorr
• [asterisk-users] Polycom 331 freezes connecting to FreePBX   Ben Schorr
• [asterisk-users] asterisk compatible cards?   Seann
• [asterisk-users] Add & play moh-files without reload   Seann
• [asterisk-users] Caller ID issue   Warren Selby
• [asterisk-users] Asterisk 1.6 and PrivacyManager with SIP   Warren Selby
• [asterisk-users] Using SIP to dial extension that will give anoutside line   Warren Selby
• [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used   Warren Selby
• [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used   Warren Selby
• [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used   Warren Selby
• [asterisk-users] COnfig File question   Warren Selby
• [asterisk-users] Asterisk 1.4 and TE420P   Warren Selby
• [asterisk-users] Asterisk 1.4 and TE420P   Warren Selby
• [asterisk-users] Security - What inbound variables can attackers populate or use when calling?   Warren Selby
• [asterisk-users] op_div: non-numeric argument   Warren Selby
• [asterisk-users] can't build resODBC on SUSE 11.3   Warren Selby
• [asterisk-users] Maximum Wait Time queue option   Warren Selby
• [asterisk-users] Monitor asterisk   Shazaum
• [asterisk-users] Monitor asterisk   Shazaum
• [asterisk-users] asterisk + openBTS   Shazaum
• [asterisk-users] Opensource Speech recognition for Asterisk   Nickolay V. Shmyrev
• [asterisk-users] Opensource Speech recognition for Asterisk   Nickolay V. Shmyrev
• [asterisk-users] asterisk compatible cards?   M Shokuie
• [asterisk-users] Asterisk routing to SoftSwitch   Pratik Shrestha
• [asterisk-users] Asterisk routing to SoftSwitch   Pratik Shrestha
• [asterisk-users] MeetMe VS. Conference   Zhang Shukun
• [asterisk-users] MeetMe will record automaticlly even without 'r' option??   Zhang Shukun
• [asterisk-users] How to Record with Konference when it has no record option?   Zhang Shukun
• [asterisk-users] codec_g729.so not work!   Zhang Shukun
• [asterisk-users] Could MeetMe invite someone to the conference?   Zhang Shukun
• [asterisk-users] How to Billing for MeetMe Conference?   Zhang Shukun
• [asterisk-users] Could MeetMe invite someone to the conference?   Zhang Shukun
• [asterisk-users] Wifi + SIP + Asterisk   Narendra Sisodiya
• [asterisk-users] Tweaking AMD in Asterisk   Aurimas Skirgaila
• [asterisk-users] Tweaking AMD in Asterisk   Aurimas Skirgaila
• [asterisk-users] AMD message   Aurimas Skirgaila
• [asterisk-users] Caller ID issue   Cassius Smith
• [asterisk-users] Caller ID issue   Cassius Smith
• [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932   Cassius Smith
• [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932   Cassius Smith
• [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932   Cassius Smith
• [asterisk-users] Caller ID issue   Cassius Smith
• [asterisk-users] Dahdi install gone wrong   Cassius Smith
• [asterisk-users] colored CLI with reattach   Eric Smith
• [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?   Chandrakant Solanki
• [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?   Chandrakant Solanki
• [asterisk-users] Polycom 331 freezes connecting to FreePBX   Connor Spiess
• [asterisk-users] Monitor asterisk   Jamie A. Stapleton
• [asterisk-users] PRI D-channel bouncing   Andrew Stewart
• [asterisk-users] PRI D-channel bouncing   Andrew Stewart
• [asterisk-users] How can i switch to samba server omitting sshfs   A J Stiles
• [asterisk-users] Asterisk on Ben NanoNote?   A J Stiles
• [asterisk-users] billsec exceeds duration on some calls   A J Stiles
• [asterisk-users] dial_exec_full problems with TDM400   A J Stiles
• [asterisk-users] billsec exceeds duration on some calls   A J Stiles
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   A J Stiles
• [asterisk-users] Dahdi install gone wrong   A J Stiles
• [asterisk-users] asterisk-1.8.0-beta4 - compile error   Václav Strachoň
• [asterisk-users] sending sms from Asterisk server   Administrator TOOTAI
• [asterisk-users] PBX Status-like module for AsteriskNow?   Frank Tarczynski
• [asterisk-users] Problem routing incoming from-pstn calls using different contexts   Frank Tarczynski
• [asterisk-users] 1.6 and asterisk gui   Christian Tardif
• [asterisk-users] 1.6 and asterisk gui   Christian Tardif
• [asterisk-users] Asterisk 1.4.35 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.11 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.0-beta3 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.0-beta4 Now Available   Asterisk Development Team
• [asterisk-users] Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.   Asterisk Development Team
• [asterisk-users] Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.   Asterisk Development Team
• [asterisk-users] 1.6 and asterisk gui   Terry
• [asterisk-users] 1.6 and asterisk gui   Terry
• [asterisk-users] help for a new user   Terry
• [asterisk-users] help for a new user   Terry
• [asterisk-users] Asterisk 1.4 and TE420P   James Texter
• [asterisk-users] Dial() M parameter in 1.6.2.11-rc2   Mark G. Thomas
• [asterisk-users] Dial() M parameter in 1.6.2.11-rc2   Mark G. Thomas
• [asterisk-users] Include and Realtime   Jonathan Thurman
• [asterisk-users] Looking for MIB description   Jonathan Thurman
• [asterisk-users] Asterisk and TV media server   Tino
• [asterisk-users] Asterisk and TV media server   Tino
• [asterisk-users] can't write to queues_additional.conf   Tino
• [asterisk-users] Tweaking AMD in Asterisk   Tino
• [asterisk-users] Tweaking AMD in Asterisk   Tino
• [asterisk-users] Tweaking AMD in Asterisk   Tino
• [asterisk-users] AMD setup in Astersik   Tino
• [asterisk-users] 'System' application in asterisk   Tino
• [asterisk-users] 'System' application in asterisk   Tino
• [asterisk-users] 'System' application in asterisk   Tino
• [asterisk-users] 'System' application in asterisk   Tino
• [asterisk-users] asterisk on Vmware   Tino
• [asterisk-users] asterisk on Vmware   Tino
• [asterisk-users] channel variables in AGI   Tino
• [asterisk-users] Asterisk Hardwares   Tino
• [asterisk-users] sending sms from Asterisk server   Tino
• [asterisk-users] sending sms from Asterisk server   Tino
• [asterisk-users] AMD message   Tino
• [asterisk-users] setting variable for a DID number   Tino
• [asterisk-users] setting variable for a DID number   Tino
• [asterisk-users] AMD message   Tino
• [asterisk-users] sms - your suggestions   Tino
• [asterisk-users] music on hold in blind transfer   Tino
• [asterisk-users] queue agent and blind transfer   Tino
• [asterisk-users] Maximum Wait Time queue option   Tino
• [asterisk-users] Maximum Wait Time queue option   Tino
• [asterisk-users] AstriCon approaches: Innovation Awards, your attendance wanted!   John Todd
• [asterisk-users] Femtocell to VoIP?   Steve Totaro
• [asterisk-users] What do you use for Invoicing?   Steve Totaro
• [asterisk-users] asterisk + openBTS   Steve Totaro
• [asterisk-users] Aastra 6739i Support   Joshua Tressler
• [asterisk-users] asterisk + openBTS   Thomas Tsou
• [asterisk-users] FYI: Seen the 2600Hz announcement?   Duncan Turnbull
• [asterisk-users] AMD setup in Astersik   Duncan Turnbull
• [asterisk-users] Call agent when queue is empty and there is a voicemail left   Duncan Turnbull
• [asterisk-users] Opensource Speech recognition for Asterisk   Duncan Turnbull
• [asterisk-users] .call files with application/data are not generating correct CDR   Duncan Turnbull
• [asterisk-users] Codec Conversion   Steve Underwood
• [asterisk-users] Codec Conversion   Steve Underwood
• [asterisk-users] Codec Conversion   Steve Underwood
• [asterisk-users] asterisk compatible cards?   Steve Underwood
• [asterisk-users] asterisk compatible cards?   Steve Underwood
• [asterisk-users] How to set up Asterisk to deliver a trunk sip connection?   Kent Varmedal
• [asterisk-users] How to set up Asterisk to deliver a trunk sip connection?   Kent Varmedal
• [asterisk-users] Opensource Speech recognition for Asterisk   Vieri
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   Kristijan Vrban
• [asterisk-users] asterisk-1.8.0-beta4 - compile error   Kristijan Vrban
• [asterisk-users] Exporting Blacklist database   Myles Wakeham
• [asterisk-users] Asterisk on Ben NanoNote?   C. Chad Wallace
• [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):   Watkins, Bradley
• [asterisk-users] Click2call from an OpenOffice document   Doddle WebPhone
• [asterisk-users] Attempted SIP connection by foreign host. Help!   Shaun Wingrin
• [asterisk-users] Asterisk 1.6 and PrivacyManager with SIP   Jaap Winius
• [asterisk-users] Asterisk 1.6 and PrivacyManager with SIP   Jaap Winius
• [asterisk-users] Logging the CID from the Privacy Manager   Jaap Winius
• [asterisk-users] Monitor asterisk   Hans Witvliet
• [asterisk-users] AMR Codec   Hans Witvliet
• [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used   Joe Wood
• [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used   Joe Wood
• [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used   Joe Wood
• [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for all inbound trunks   Joe Wood
• [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for allinbound trunks   Joe Wood
• [asterisk-users] Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround() but no audio is heard from the phone   Joe Wood
• [asterisk-users] Asterisk 1.6 Displaying BackGround() in call trace but no audio is heard from caller   Joe Wood
• [asterisk-users] Push to talk over cellular   Jay R. Worthington
• [asterisk-users] dynamic MeetMe, min. digits   Xavier
• [asterisk-users] PRI D-channel bouncing   Michael L. Young
• [asterisk-users] T.38 fax between ATA's and Asterisk and Cisco PGW 2200   P Z
• [asterisk-users] T.38 fax between ATA's and Asterisk and Cisco PGW 2200   P Z
• [asterisk-users] RTP stream not passing through router with port forwarding   Zeeshan Zakaria
• [asterisk-users] What do you use for Invoicing?   Zeeshan Zakaria
• [asterisk-users] What do you use for Invoicing?   Zeeshan Zakaria
• [asterisk-users] Dial() M parameter in 1.6.2.11-rc2   Zeeshan Zakaria
• [asterisk-users] Dial() M parameter in 1.6.2.11-rc2   Zeeshan Zakaria
• [asterisk-users] Realtime Context   Zeeshan Zakaria
• [asterisk-users] Realtime Context   Zeeshan Zakaria
• [asterisk-users] Realtime Context   Zeeshan Zakaria
• [asterisk-users] Realtime Context   Zeeshan Zakaria
• [asterisk-users] Realtime Context   Zeeshan Zakaria
• [asterisk-users] Monitor asterisk   Zeeshan Zakaria
• [asterisk-users] Realtime Context   Zeeshan Zakaria
• [asterisk-users] Opensource Speech recognition for Asterisk   Zeeshan Zakaria
• [asterisk-users] Opensource Speech recognition for Asterisk   Zeeshan Zakaria
• [asterisk-users] Codec choice   Zeeshan Zakaria
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Zeeshan Zakaria
• [asterisk-users] Include and Realtime   Zeeshan Zakaria
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Zeeshan Zakaria
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Zeeshan Zakaria
• [asterisk-users] Include and Realtime   Zeeshan Zakaria
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Zeeshan Zakaria
• [asterisk-users] Looking for MIB description   Zeeshan Zakaria
• [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex: Unhandled char(s):   Zeeshan Zakaria
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Zeeshan Zakaria
• [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):   Zeeshan Zakaria
• [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):   Zeeshan Zakaria
• [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?   Zeeshan Zakaria
• [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):   Zeeshan Zakaria
• [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):   Zeeshan Zakaria
• [asterisk-users] STUN   Redouane Zerargui
• [asterisk-users] Asterisk DTMF RFC2833 issues   Bryant Zimmerman
• [asterisk-users] Protect yourself   Bryant Zimmerman
• [asterisk-users] Migrating 1.4 to 1.6.2   Bryant Zimmerman
• [asterisk-users] only part of dialplan available   Bryant Zimmerman
• [asterisk-users] Asterisk does not translate from wav to alaw   Bryant Zimmerman
• [asterisk-users] help with dialplan   Bryant Zimmerman
• [asterisk-users] help with dialplan   Bryant Zimmerman
• [asterisk-users] Monitor asterisk   Richard Zulu
• [asterisk-users] Monitor asterisk   Richard Zulu
• [asterisk-users] Monitor asterisk   Richard Zulu
• [asterisk-users] OT: Grandstream GXV3140   --[ UxBoD ]--
• [asterisk-users] OT: Grandstream GXV3140   --[ UxBoD ]--
• [asterisk-users] EMail on Missed Call   --[ UxBoD ]--
• [asterisk-users] DAHDI not detecting caller hangup   --[ UxBoD ]--
• [asterisk-users] All phones ringing when temporary loss of Internet   --[ UxBoD ]--
• [asterisk-users] DAHDI not detecting caller hangup   --[ UxBoD ]--
• [asterisk-users] DAHDI not detecting caller hangup   --[ UxBoD ]--
• [asterisk-users] All phones ringing when temporary loss of Internet   --[ UxBoD ]--
• [asterisk-users] fail2ban does not work for my asterisk installation   mosbah abdelkader
• [asterisk-users] fail2ban does not work for my asterisk installation   mosbah abdelkader
• [asterisk-users] fail2ban does not work for my asterisk installation   mosbah abdelkader
• [asterisk-users] Queue to queue transfer error   toqeer ali
• [asterisk-users] Stupid Macro question   unserossi at aol.com
• [asterisk-users] Stupid Macro question   unserossi at aol.com
• [asterisk-users] Fwd: Stupid Macro question   unserossi at aol.com
• [asterisk-users] Dial() M parameter in 1.6.2.11-rc2   unserossi at aol.com
• [asterisk-users] Dial() M parameter in 1.6.2.11-rc2   unserossi at aol.com
• [asterisk-users] callerid between 2 asterisk servers   unserossi at aol.com
• [asterisk-users] Asterisk and OCS2007 R2   unserossi at aol.com
• [asterisk-users] Asterisk and OCS2007 R2   unserossi at aol.com
• [asterisk-users] Asterisk and OCS2007 R2   unserossi at aol.com
• [asterisk-users] sip.conf register in realtime DB   unserossi at aol.com
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   unserossi at aol.com
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   unserossi at aol.com
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   unserossi at aol.com
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   unserossi at aol.com
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   unserossi at aol.com
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   unserossi at aol.com
• [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3   unserossi at aol.com
• [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3   unserossi at aol.com
• [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3   unserossi at aol.com
• [asterisk-users] Fwd: problem with mssql and Asterisk 1.8.0 beta3   unserossi at aol.com
• [asterisk-users] SIP response 500 "Server Internal Error"   asterisk asterisk
• [asterisk-users] SIP response 500 "Server Internal Error"   asterisk asterisk
• [asterisk-users] SIP response 500 "Server Internal Error"   asterisk asterisk
• [asterisk-users] 603 error   asterisk asterisk
• [asterisk-users] Fwd: 603 error   asterisk asterisk
• [asterisk-users] sending sms from Asterisk server   asterisk asterisk
• [asterisk-users] chinaroby fxo card - never heard of them   asteriskguru asteriskguru
• [asterisk-users] Dahdi issue on sangoma A200   asteriskguru asteriskguru
• [asterisk-users] speciality of SIPp and SER(Sip Express Router)   kamrun nahar bina
• [asterisk-users] speciality of SIPp and SER(Sip Express Router)   kamrun nahar bina
• [asterisk-users] speciality of SIPp and SER(Sip Express Router)   kamrun nahar bina
• [asterisk-users] What do you use for Invoicing?   bruce bruce
• [asterisk-users] What do you use for Invoicing?   bruce bruce
• [asterisk-users] What do you use for Invoicing?   bruce bruce
• [asterisk-users] What do you use for Invoicing?   bruce bruce
• [asterisk-users] What do you use for Invoicing?   bruce bruce
• [asterisk-users] What do you use for Invoicing?   bruce bruce
• [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?   bruce bruce
• [asterisk-users] How to determine which party hangup the call? cause of Hang-up needed.‏   bruce bruce
• [asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?   bruce bruce
• [asterisk-users] How to determine which party hangup the call? cause of Hang-up needed.‏   bruce bruce
• [asterisk-users] Asterisk with Motorola Canopy   bruce bruce
• [asterisk-users] Pfsense and IAX2 - What is the proper firewall NAT setup?   bruce bruce
• [asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?   bruce bruce
• [asterisk-users] Opensource Speech recognition for Asterisk   bruce bruce
• [asterisk-users] Opensource Speech recognition for Asterisk   bruce bruce
• [asterisk-users] Opensource Speech recognition for Asterisk   bruce bruce
• [asterisk-users] Quick Question - Jabra Headset and Aastra 53i - Where is the speaker/headset enable setting on Aastra UI?   bruce bruce
• [asterisk-users] Quick Question - Jabra Headset and Aastra 53i - Where is the speaker/headset enable setting on Aastra UI?   bruce bruce
• [asterisk-users] OrderlyStats or QueueMetrics   bruce bruce
• [asterisk-users] Problems with meetme in 1.4.26   covici at ccs.covici.com
• [asterisk-users] asterisk-1.8 problem with one-way audio with no nat   covici at ccs.covici.com
• [asterisk-users] How to read span debug ?   chetan
• [asterisk-users] DEBUG: Cannot find variable 'XXX' ??   sean darcy
• [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon   sean darcy
• [asterisk-users] FYI: Seen the 2600Hz announcement?   dotnetdub
• [asterisk-users] ConfBridge   iscario at free.fr
• [asterisk-users] Codec Conversion   michel freiha
• [asterisk-users] Codec Conversion   michel freiha
• [asterisk-users] Codec Conversion   michel freiha
• [asterisk-users] SIP Status: 401 Unauthorized (0 bindings)   ast guy
• [asterisk-users] callerid between 2 asterisk servers   jwexler at mail.usa.com
• [asterisk-users] callerid between 2 asterisk servers   jwexler at mail.usa.com
• [asterisk-users] Security - What inbound variables can attackers populate or use when calling?   jwexler at mail.usa.com
• [asterisk-users] How does deny/permit work in sip.conf?   jwexler at mail.usa.com
• [asterisk-users] Security - What inbound variables can attackers populate or use when calling?   jwexler at mail.usa.com
• [asterisk-users] Security - What inbound variables can attackers populate or use when calling?   jwexler at mail.usa.com
• [asterisk-users] asterisknow   mattias
• [asterisk-users] Identify remote prompts: Partial audiomatching?   mattias
• [asterisk-users] Timing on Asterisk   colin mcdermott
• [asterisk-users] Femtocell to VoIP?   info at meetmecall.nl
• [asterisk-users] Security - What inbound variables can attackers populate or use when calling?   mike mosier
• [asterisk-users] Asterisk dialup connection?   hadi motamedi
• [asterisk-users] asterisk core dump   jordan pan
• [asterisk-users] 3g call support for ISDN line   pankaj pandey
• [asterisk-users] Asterisk codec config issue with ALAW   garge rama
• [asterisk-users] Tones of dtmf during call   das sandesh
• [asterisk-users] Yes it is a dimensioning question! Atom CPU   jmillican at sentinelcommunications.com
• [asterisk-users] asterisk + openBTS   equis software
• [asterisk-users] asterisk + openBTS   equis software
• [asterisk-users] asterisk + openBTS   equis software
• [asterisk-users] asterisk + openBTS   equis software
• [asterisk-users] Using SIP to dial extension that will give an outside line   Jeremy.Hellstrom at synovate.com
• [asterisk-users] Using SIP to dial extension that will give anoutside line   Jeremy.Hellstrom at synovate.com
• [asterisk-users] Reinstalling Asterisk due to hardware changes   Jeremy.Hellstrom at synovate.com
• [asterisk-users] TELUS British Columbia PRI Settings   Jeremy.Hellstrom at synovate.com
• [asterisk-users] TELUS British Columbia PRI Settings   Jeremy.Hellstrom at synovate.com
• [asterisk-users] TELUS British Columbia PRI Settings   Jeremy.Hellstrom at synovate.com
• [asterisk-users] TELUS British Columbia PRI Settings   Jeremy.Hellstrom at synovate.com
• [asterisk-users] TELUS British Columbia PRI Settings   Jeremy.Hellstrom at synovate.com
• [asterisk-users] TELUS British Columbia PRI Settings   Jeremy.Hellstrom at synovate.com
• [asterisk-users] Scilence problem on running call   kishorej at techroutes.com
• [asterisk-users] only part of dialplan available   kishorej at techroutes.com
• [asterisk-users] only part of dialplan available   kishorej at techroutes.com
• [asterisk-users] Digest Username/auth name mismatch   kawanobe tomohito
• [asterisk-users] How to track a call result originated from originate AMI command   thiyagu venkatesan
• [asterisk-users] Play a number of files to a caller   Ondrej Škopek
• [asterisk-users] Play a number of files to a caller   Ondrej Škopek
• [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6   Ondrej Škopek

Last message date: Tue Aug 31 22:15:36 CDT 2010
Archived on: Tue Aug 31 22:15:47 CDT 2010

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