January 2011 Archives by author

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Starting: Fri Dec 31 22:34:12 CST 2010
Ending: Mon Jan 31 21:09:05 CST 2011
Messages: 1108

• [asterisk-users] AstLinux 0.7.5 released   Darrick Hartman (lists)
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• [asterisk-users] asterisk-users Digest, Vol 78, Issue 66   Chris Cooper 325
• [asterisk-users] sip dos question   adamk at 3a.hu
• [asterisk-users] sip dos question   adamk at 3a.hu
• [asterisk-users] How to check a number online or offline   enkillar 87
• [asterisk-users] Why are 4 ports used for a single call?   Gary Allen
• [asterisk-users] Music on Hold not working?   Gary Allen
• [asterisk-users] Music on Hold not working?   Gary Allen
• [asterisk-users] Music on Hold not working?   Gary Allen
• [asterisk-users] Music on Hold not working?   Gary Allen
• [asterisk-users] Music on Hold not working?   Gary Allen
• [asterisk-users] ReceiveFAX issue.   Benny Amorsen
• [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0   Benny Amorsen
• [asterisk-users] Using asterisk and icecast for live audio streaming.   Goke M Aruna
• [asterisk-users] Recommended Windows client to display CID?   Asterisk
• [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?   Bruce B
• [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?   Bruce B
• [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?   Bruce B
• [asterisk-users] Asterisk Outlook integration   Bruce B
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Bruce B
• [asterisk-users] Asterisk Outlook integration   Bruce B
• [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?   Bruce B
• [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?   Bruce B
• [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?   Bruce B
• [asterisk-users] Do I need a sip proxy?   Bruce B
• [asterisk-users] OpenVPN + SIP configuration?   Bruce B
• [asterisk-users] Do I need a sip proxy?   Bruce B
• [asterisk-users] OpenVPN + SIP configuration?   Bruce B
• [asterisk-users] Paid or Free software that would do pop-up from Outlook 2007 via Asterisk AMI   Bruce B
• [asterisk-users] OpenVPN + SIP configuration?   Bruce B
• [asterisk-users] OpenVPN + SIP configuration?   Bruce B
• [asterisk-users] CallerID and URL pop up for windows...   Bruce B
• [asterisk-users] CallerID and URL pop up for windows...   Bruce B
• [asterisk-users] Why are 4 ports used for a single call?   Bruce B
• [asterisk-users] Why are 4 ports used for a single call?   Bruce B
• [asterisk-users] Why are 4 ports used for a single call?   Bruce B
• [asterisk-users] Why are 4 ports used for a single call?   Bruce B
• [asterisk-users] Why are 4 ports used for a single call?   Bruce B
• [asterisk-users] Why are 4 ports used for a single call?   Bruce B
• [asterisk-users] Why are 4 ports used for a single call?   Bruce B
• [asterisk-users] Tools to Monitor Asterisk Servers and VMs   Bruce B
• [asterisk-users] Why are 4 ports used for a single call?   Bruce B
• [asterisk-users] Top Posting   Bruce B
• [asterisk-users] Top Posting   Bruce B
• [asterisk-users] Bruce B   Bruce B
• [asterisk-users] Do I need a sip proxy?   Bruce B
• [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?   Bruce B
• [asterisk-users] SIP channel status - Why is it different when calling an internal extension rather than an outside line over SIP?   Bruce B
• [asterisk-users] Can a duration limit be specified in spool call file?   Bruce B
• [asterisk-users] Using the Telco Call Transfer Features.   Jeff B
• [asterisk-users] Using the Telco Call Transfer Features.   Jeff B
• [asterisk-users] Info on using LDAP with Asterisk?   Jeff B
• [asterisk-users] Info on using LDAP with Asterisk?   Jeff B
• [asterisk-users] Crossover cable for E1 ?   Sébastien BERGER
• [asterisk-users] Weird phone behavior after recent CentOS 5 update   David Backeberg
• [asterisk-users] ReceiveFax   David Backeberg
• [asterisk-users] Asterisk to asterisk t.38   David Backeberg
• [asterisk-users] ReceiveFAX issue.   David Backeberg
• [asterisk-users] U-verse DTMF tuning for Zaptel   David Backeberg
• [asterisk-users] U-verse DTMF tuning for Zaptel   David Backeberg
• [asterisk-users] ReceiveFAX issue.   David Backeberg
• [asterisk-users] ReceiveFAX issue.   David Backeberg
• [asterisk-users] ReceiveFAX issue.   David Backeberg
• [asterisk-users] Spectralink 8002   Jonathan C. Bailey
• [asterisk-users] Spectralink 8002   Jonathan C. Bailey
• [asterisk-users] Spectralink 8002   Jonathan C. Bailey
• [asterisk-users] Spectralink 8002   Jonathan C. Bailey
• [asterisk-users] How to disable srtp in asterisk 1.8.2.3?   Miguel Baptista
• [asterisk-users] Go from CALLINGout to just CALLING   Bob Beers
• [asterisk-users] Mailing list question   Bob Beers
• [asterisk-users] Determine When Call Is Picked Up In Queue   Joseph Begumisa
• [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?   Paul Belanger
• [asterisk-users] Call Back on Busy   Paul Belanger
• [asterisk-users] Top Posting   Paul Belanger
• [asterisk-users] Basic Sip.conf and extensions.conf   Paul Belanger
• [asterisk-users] res_fax_digium.so crashing   Paul Belanger
• [asterisk-users] res_fax_digium.so crashing   Paul Belanger
• [asterisk-users] Top Posting   Paul Belanger
• [asterisk-users] Top Posting   Paul Belanger
• [asterisk-users] Asterisk extension not found problem...   Paul Belanger
• [asterisk-users] FUNC_ODBC and ARRAY   Paul Belanger
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Paul Belanger
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Paul Belanger
• [asterisk-users] Return variables from func_odbc calls?   Paul Belanger
• [asterisk-users] Return variables from func_odbc calls?   Paul Belanger
• [asterisk-users] feedback mechanism   Paul Belanger
• [asterisk-users] Lots of warnings: SUBSCRIBE failure: no Accept header: pvt   Paul Belanger
• [asterisk-users] Regarding error in Asterisk dail plan:   Paul Belanger
• [asterisk-users] Multi-Tenant   Paul Belanger
• [asterisk-users] Disabling Music On Hold   Paul Belanger
• [asterisk-users] Issue with Asterisk not hanging up second leg when first leg hangs up   Dovid Bender
• [asterisk-users] Issue with Asterisk not hanging up second leg when first leg hangs up   Dovid Bender
• [asterisk-users] Queue announce parameter problem   Eduardo Lobo Blanco
• [asterisk-users] Cross Queue Priorities   Nick Brown
• [asterisk-users] Disabling Music On Hold   Urs Buob
• [asterisk-users] Disabling Music On Hold   Urs Buob
• [asterisk-users] Disabling Music On Hold   Urs Buob
• [asterisk-users] Senddtmf inside a macro   CDR
• [asterisk-users] Callback form to place on site for customers. Recomendation to achieve this.   JP CR
• [asterisk-users] Queue periodic announce...   Carlos Chavez
• [asterisk-users] CallerID and URL pop up for windows...   Carlos Chavez
• [asterisk-users] Asterisk stops responding   Carlos Chavez
• [asterisk-users] SIP Originate on 1.8.1.1   Carlos Chavez
• [asterisk-users] IAX between 1.6 and 1.8 has bad voice quality   Carlos Chavez
• [asterisk-users] Asterisk extension not found problem...   Carlos Chavez
• [asterisk-users] Asterisk stops responding   Carlos Chavez
• [asterisk-users] Asterisk stops responding   Carlos Chavez
• [asterisk-users] Asterisk stops responding   Carlos Chavez
• [asterisk-users] A1200P comments?   Carlos Chavez
• [asterisk-users] Call queues on load-balanced asterisks   Pan B. Christensen
• [asterisk-users] Call queues on load-balanced asterisks   Pan B. Christensen
• [asterisk-users] Do I need a sip proxy?   Pan B. Christensen
• [asterisk-users] Call queues on load-balanced asterisks   Pan B. Christensen
• [asterisk-users] Do I need a sip proxy?   Pan B. Christensen
• [asterisk-users] Unable to insert cdr-data into mysql-DB   Enrico Cicconi
• [asterisk-users] How to update sound files?   Enrico Cicconi
• [asterisk-users] Callback when available   Harel Cohen
• [asterisk-users] system lockup when going into conference   Tzafrir Cohen
• [asterisk-users] Problem with chan_dahdi and conferencing   Tzafrir Cohen
• [asterisk-users] Problem with chan_dahdi and conferencing   Tzafrir Cohen
• [asterisk-users] Problem with chan_dahdi and conferencing   Tzafrir Cohen
• [asterisk-users] Music on Hold not working?   Tzafrir Cohen
• [asterisk-users] Problem with chan_dahdi and conferencing   Tzafrir Cohen
• [asterisk-users] Music on Hold not working?   Tzafrir Cohen
• [asterisk-users] Top Posting   Tzafrir Cohen
• [asterisk-users] Top Posting   Tzafrir Cohen
• [asterisk-users] Top Posting   Tzafrir Cohen
• [asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny   Tzafrir Cohen
• [asterisk-users] Recommended Windows client to display CID?   Tzafrir Cohen
• [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)   Nic Colledge
• [asterisk-users] progressinband, how much extra CPU load?   David Cunningham
• [asterisk-users] Introducing easySysAdmin - automated security and telecom fraud protection   David Cunningham
• [asterisk-users] Error compiling Dahdi: invalid use of undefined type struct module   David Cunningham
• [asterisk-users] Error compiling Dahdi: invalid use of undefined type struct module   David Cunningham
• [asterisk-users] Hi, agent intro-speech for outside caller   DSR
• [asterisk-users] queue_log in MySQL database   Steve Davies
• [asterisk-users] VoIP PoE phones for restaurant   Mark Deneen
• [asterisk-users] OpenVPN + SIP configuration?   Mark Deneen
• [asterisk-users] OpenVPN + SIP configuration?   Mark Deneen
• [asterisk-users] Top Posting   Mark Deneen
• [asterisk-users] Top Posting   Mark Deneen
• [asterisk-users] Polycom 500 / MWI   Mark Deneen
• [asterisk-users] Asterisk stops responding   Mark Deneen
• [asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny   Mark Deneen
• [asterisk-users] Benefit of PRI vs SIP trunk calls   Jim Dickenson
• [asterisk-users] Benefit of PRI vs SIP trunk calls   Jim Dickenson
• [asterisk-users] How to check a number online or offline   Jim Dickenson
• [asterisk-users] Call Back on Busy   Jim Dickenson
• [asterisk-users] How to check a number online or offline   Jim Dickenson
• [asterisk-users] Determine When Call Is Picked Up In Queue   Jim Dickenson
• [asterisk-users] CDR Questions   Mike Diehl
• [asterisk-users] waitforsilence changed after upgrade to 1.6   Mike Diehl
• [asterisk-users] A1200P comments?   Mike Diehl
• [asterisk-users] Calling Directory app from AGI   Mike Diehl
• [asterisk-users] Calling Directory app from AGI   Mike Diehl
• [asterisk-users] [OT] Streaming video on variable bandwidth connection?   Tim Dobson
• [asterisk-users] Top Posting   Lesly Dorval
• [asterisk-users] Lots of warnings: SUBSCRIBE failure: no Accept header: pvt   Doug
• [asterisk-users] Lots of warnings: SUBSCRIBE failure: no Accept header: pvt   Doug
• [asterisk-users] Lots of warnings: SUBSCRIBE failure: no Accept header: pvt   Doug
• [asterisk-users] No RTP Engine problem in 1.8.2   Paradise Dove
• [asterisk-users] Really wacky problem with internal extensions.   Ernie Dunbar
• [asterisk-users] Occasional robotic sound while call in progress   Michelle Dupuis
• [asterisk-users] Max call duration   Michelle Dupuis
• [asterisk-users] Dialplan to bridge 2 legs?   Michelle Dupuis
• [asterisk-users] Dialplan to bridge 2 legs?   Michelle Dupuis
• [asterisk-users] Dialplan to bridge 2 legs?   Michelle Dupuis
• [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)   Steve Edwards
• [asterisk-users] Base memory usage   Steve Edwards
• [asterisk-users] incoming   Steve Edwards
• [asterisk-users] incoming   Steve Edwards
• [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?   Steve Edwards
• [asterisk-users] Problems with ZAP Channels   Steve Edwards
• [asterisk-users] SetVar Warning   Steve Edwards
• [asterisk-users] Call hung up?   Steve Edwards
• [asterisk-users] Call hung up?   Steve Edwards
• [asterisk-users] Selecing the E1 cards for the call center   Steve Edwards
• [asterisk-users] how to read mp3   Steve Edwards
• [asterisk-users] Calling rules   Steve Edwards
• [asterisk-users] Calling rules   Steve Edwards
• [asterisk-users] Calling rules   Steve Edwards
• [asterisk-users] Asterisk extension not found problem...   Steve Edwards
• [asterisk-users] Asterisk extension not found problem...   Steve Edwards
• [asterisk-users] Asterisk extension not found problem...   Steve Edwards
• [asterisk-users] Asterisk extension not found problem...   Steve Edwards
• [asterisk-users] Asterisk extension not found problem...   Steve Edwards
• [asterisk-users] Asterisk extension not found problem...   Steve Edwards
• [asterisk-users] Hi, agent intro-speech for outside caller   Steve Edwards
• [asterisk-users] Dialplan to bridge 2 legs?   Steve Edwards
• [asterisk-users] Dialplan to bridge 2 legs?   Steve Edwards
• [asterisk-users] U-verse DTMF tuning for Zaptel   Steve Edwards
• [asterisk-users] U-verse DTMF tuning for Zaptel   Steve Edwards
• [asterisk-users] U-verse DTMF tuning for Zaptel   Steve Edwards
• [asterisk-users] U-verse DTMF tuning for Zaptel   Steve Edwards
• [asterisk-users] U-verse DTMF tuning for Zaptel   Steve Edwards
• [asterisk-users] Regarding error in Asterisk dail plan:   Steve Edwards
• [asterisk-users] Regarding error in Asterisk dail plan:   Steve Edwards
• [asterisk-users] Calling Directory app from AGI   Steve Edwards
• [asterisk-users] Newbie Question...   Steve Edwards
• [asterisk-users] cannot answer incoming calls   Jeroen Eeuwes
• [asterisk-users] Basic Sip.conf and extensions.conf   Jeroen Eeuwes
• [asterisk-users] context problem   Jeroen Eeuwes
• [asterisk-users] How to reject an incoming call using AMI ?   Lenz Emilitri
• [asterisk-users] chan_h323 and menuselect dependencies problem   Jose P. Espinal
• [asterisk-users] Asterisk SlackBuilds for Slackware Linux   Jose P. Espinal
• [asterisk-users] intermittent problem on 1.4   Jose P. Espinal
• [asterisk-users] context problem   Jose P. Espinal
• [asterisk-users] Benefit of PRI vs SIP trunk calls   C F
• [asterisk-users] Top Posting   C F
• [asterisk-users] end a call after a specific time period   C F
• [asterisk-users] Unknow "T" callerid   C F
• [asterisk-users] end a call after a specific time period   C F
• [asterisk-users] Base memory usage   Robert Fantini
• [asterisk-users] Multiple Registrations   Jon Farmer
• [asterisk-users] Asterisk 1.8.2.3 Now Available   Helius Ferreira
• [asterisk-users] Unable to insert cdr-data into mysql-DB   Bruce Ferrell
• [asterisk-users] Realtime SIP, multiple AX servers question   Bryan Field-Elliot
• [asterisk-users] Realtime SIP, multiple AX servers question   Bryan Field-Elliot
• [asterisk-users] Realtime SIP, multiple AX servers question   Bryan Field-Elliot
• [asterisk-users] Top Posting   Cary Fitch
• [asterisk-users] Top Posting   Cary Fitch
• [asterisk-users] Top Posting   Cary Fitch
• [asterisk-users] Top Posting   Cary Fitch
• [asterisk-users] Calling rules   Vitor Carlos Flausino
• [asterisk-users] Calling rules   Vitor Carlos Flausino
• [asterisk-users] Calling rules   Vitor Carlos Flausino
• [asterisk-users] Calling rules   Vitor Carlos Flausino
• [asterisk-users] Calling rules   Vitor Carlos Flausino
• [asterisk-users] Calling rules   Vitor Carlos Flausino
• [asterisk-users] Calling rules   Vitor Carlos Flausino
• [asterisk-users] Unable to receive calls (inbound)   Vitor Carlos Flausino
• [asterisk-users] Where are stored the CDR's?   Vitor Carlos Flausino
• [asterisk-users] Where are stored the CDR's?   Vitor Carlos Flausino
• [asterisk-users] Inbound routes   Vitor Carlos Flausino
• [asterisk-users] Inbound routes   Vitor Carlos Flausino
• [asterisk-users] Channel in an unkown state   Vitor Carlos Flausino
• [asterisk-users] Clarification on DAHDI Fax Detection   Kevin P. Fleming
• [asterisk-users] Clarification on DAHDI Fax Detection   Kevin P. Fleming
• [asterisk-users] Clarification on DAHDI Fax Detection   Kevin P. Fleming
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Kevin P. Fleming
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Kevin P. Fleming
• [asterisk-users] Realtime SIP, multiple AX servers question   Kevin P. Fleming
• [asterisk-users] Too Few Fax Detections   Kevin P. Fleming
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Kevin P. Fleming
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Kevin P. Fleming
• [asterisk-users] asterisk fax problem   Kevin P. Fleming
• [asterisk-users] How to reject an incoming call using AMI ?   Kevin P. Fleming
• [asterisk-users] Failed SIP registration kicks registered device off?   Kevin P. Fleming
• [asterisk-users] Unable to get Fax t38 working with IrisTel trunk   Kevin P. Fleming
• [asterisk-users] Can I know if a call is transffered to asterisk   Kevin P. Fleming
• [asterisk-users] res_fax_digium.so crashing   Kevin P. Fleming
• [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] ReceiveFax   Kevin P. Fleming
• [asterisk-users] Mailing list question   Kevin P. Fleming
• [asterisk-users] Introducing easySysAdmin - automated security and telecom fraud protection   Kevin P. Fleming
• [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver   Kevin P. Fleming
• [asterisk-users] Crossover cable for E1 ?   Kevin P. Fleming
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Kevin P. Fleming
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] ReceiveFAX issue.   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] Asterisk 1.8.2.3 Now Available   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] SIP, IAX2 and ISDN ISUP data   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] RTP keepalive doesn't work   Kevin P. Fleming
• [asterisk-users] RTP keepalive doesn't work   Kevin P. Fleming
• [asterisk-users] Disabling Music On Hold   Kevin P. Fleming
• [asterisk-users] Lots of warnings: SUBSCRIBE failure: no Accept header: pvt   Kevin P. Fleming
• [asterisk-users] Disabling Music On Hold   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] res_fax   Kevin P. Fleming
• [asterisk-users] Queues with ringinuse=yes   Vinícius Fontes
• [asterisk-users] Unknow "T" callerid   Jose Flores Galicia
• [asterisk-users] Unknow "T" callerid   Jose Flores Galicia
• [asterisk-users] chan_sip bug? (Asterisk 1.4)   Jian Gao
• [asterisk-users] chan_sip bug? (Asterisk 1.4)   Jian Gao
• [asterisk-users] Call parking question   Chris Gentle
• [asterisk-users] Crossover cable for E1 ?   Godson Gera
• [asterisk-users] end a call after a specific time period   Godson Gera
• [asterisk-users] Base memory usage   Gilles
• [asterisk-users] Log and forward calls to cellphone?   Gilles
• [asterisk-users] Base memory usage   Gilles
• [asterisk-users] Base memory usage   Gilles
• [asterisk-users] Log and forward calls to cellphone?   Gilles
• [asterisk-users] Log and forward calls to cellphone?   Gilles
• [asterisk-users] Log and forward calls to cellphone?   Gilles
• [asterisk-users] Log and forward calls to cellphone?   Gilles
• [asterisk-users] OpenVPN + SIP configuration?   Gilles
• [asterisk-users] OpenVPN + SIP configuration?   Gilles
• [asterisk-users] OpenVPN + SIP configuration?   Gilles
• [asterisk-users] OpenVPN + SIP configuration?   Gilles
• [asterisk-users] OpenVPN + SIP configuration?   Gilles
• [asterisk-users] OpenVPN + SIP configuration?   Gilles
• [asterisk-users] OpenVPN + SIP configuration?   Gilles
• [asterisk-users] CallerID and URL pop up for windows...   Gilles
• [asterisk-users] CallerID and URL pop up for windows...   Gilles
• [asterisk-users] Why are 4 ports used for a single call?   Gilles
• [asterisk-users] Recommended Windows client to display CID?   Gilles
• [asterisk-users] Recommended Windows client to display CID?   Gilles
• [asterisk-users] Recommended Windows client to display CID?   Gilles
• [asterisk-users] Recommended Windows client to display CID?   Gilles
• [asterisk-users] Recommended Windows client to display CID?   Gilles
• [asterisk-users] Recommended Windows client to display CID?   Gilles
• [asterisk-users] Recommended Windows client to display CID?   Gilles
• [asterisk-users] Recommended Windows client to display CID?   Gilles
• [asterisk-users] Recommended Windows client to display CID?   Gilles
• [asterisk-users] Recommended Windows client to display CID?   Gilles
• [asterisk-users] Reducing number of Asterisk processes?   Gilles
• [asterisk-users] Reducing number of Asterisk processes?   Gilles
• [asterisk-users] Reducing number of Asterisk processes?   Gilles
• [asterisk-users] Reducing number of Asterisk processes?   Gilles
• [asterisk-users] Reducing number of Asterisk processes?   Gilles
• [asterisk-users] Reducing number of Asterisk processes?   Gilles
• [asterisk-users] Post Dial Delay + Playtones   Morgan Gilroy
• [asterisk-users] 'Bad authorization' error with Asterisk 1.8   Arie Goldfeld
• [asterisk-users] 'Bad authorization' error with Asterisk 1.8   Arie Goldfeld
• [asterisk-users] OT: VoIP Users Conf Feb 4 with LifeSize   Michael Graves
• [asterisk-users] VoIP PoE phones for restaurant   Andy Graybeal
• [asterisk-users] VoIP PoE phones for restaurant   Andy Graybeal
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Andy Graybeal
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Andy Graybeal
• [asterisk-users] VoIP PoE phones for restaurant   Andy Graybeal
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Andy Graybeal
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Andy Graybeal
• [asterisk-users] [tech] VoIP PoE phones for restaurant (kitchen)   Andy Graybeal
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Andy Graybeal
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Andy Graybeal
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Andy Graybeal
• [asterisk-users] Newbie Question...   Piotr Górski
• [asterisk-users] New Dahdi error   Thorsten Göllner
• [asterisk-users] DTMF not being heard correctly by far end conference system   Thorsten Göllner
• [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'   Thorsten Göllner
• [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'   Thorsten Göllner
• [asterisk-users] Sangoma A104d / overlapdial=yes / dial with audio one-way issue   Thorsten Göllner
• [asterisk-users] agi dial termination cause ?   Thorsten Göllner
• [asterisk-users] Where are stored the CDR's?   Thorsten Göllner
• [asterisk-users] Where are stored the CDR's?   Thorsten Göllner
• [asterisk-users] Asterisk stops responding   Thorsten Göllner
• [asterisk-users] Variable HANGUPCAUSE always empty with DAHDI   Thorsten Göllner
• [asterisk-users] incoming   Rick Hall
• [asterisk-users] Question About Conferencing Capabilities   Siobhan Hamilton
• [asterisk-users] Question About Conferencing Capabilities   Siobhan Hamilton
• [asterisk-users] Question About Conferencing Capabilities   Siobhan Hamilton
• [asterisk-users] changed datadir   Nicholas Hart
• [asterisk-users] changed datadir   Nicholas Hart
• [asterisk-users] logging   Nicholas Hart
• [asterisk-users] Asterisk fail over. From IP rewrite issues   Peter den Hartog
• [asterisk-users] OpenVPN + SIP configuration?   Gordon Henderson
• [asterisk-users] 5-7 second delay in connecting outgoing FXO calls   Gordon Henderson
• [asterisk-users] Reducing number of Asterisk processes?   Gordon Henderson
• [asterisk-users] app_fax watchdog timeout   Eric Hiller
• [asterisk-users] How to check a number online or offline   Phuong Hoang
• [asterisk-users] How to check a number online or offline   Phuong Hoang
• [asterisk-users] How to check a number online or offline   Phuong Hoang
• [asterisk-users] How to check a number online or offline   Phuong Hoang
• [asterisk-users] How to reject an incoming call using AMI ?   Phuong Hoang
• [asterisk-users] How to check a number online or offline   Phuong Hoang
• [asterisk-users] How to check a number online or offline   Phuong Hoang
• [asterisk-users] How to check a number online or offline   Phuong Hoang
• [asterisk-users] How to check a number online or offline   Phuong Hoang
• [asterisk-users] How to check a number online or offline   Phuong Hoang
• [asterisk-users] How to reject an incoming call using AMI ?   Phuong Hoang
• [asterisk-users] How to reject an incoming call using AMI ?   Phuong Hoang
• [asterisk-users] How to check a number online or offline   Steve Howes
• [asterisk-users] How to check a number online or offline   Steve Howes
• [asterisk-users] Continuously core dumping of 1.8 on SLES   Steve Howes
• [asterisk-users] Mailing list question 2   Steve Howes
• [asterisk-users] Asterisk stops responding   Steve Howes
• [asterisk-users] Unable to insert cdr-data into mysql-DB   Steve Howes
• [asterisk-users] faxter   Steve Howes
• [asterisk-users] Polycom 500 / MWI   Brian C. Huffman
• [asterisk-users] Losing registration - ast 1.4.39 and innomedia 6328-2Re   Brian C. Huffman
• [asterisk-users] Question About Conferencing Capabilities   DHAVAL INDRODIYA
• [asterisk-users] Call queues on load-balanced asterisks   DHAVAL INDRODIYA
• [asterisk-users] How to check a number online or offline   DHAVAL INDRODIYA
• [asterisk-users] How to check a number online or offline   DHAVAL INDRODIYA
• [asterisk-users] How to check a number online or offline   DHAVAL INDRODIYA
• [asterisk-users] Asterisk and Kamailio integration on cloud EC2 amazon no voice.   DHAVAL INDRODIYA
• [asterisk-users] Can a duration limit be specified in spool call file?   DHAVAL INDRODIYA
• [asterisk-users] Asterisk Outlook integration   Giorgio Incantalupo
• [asterisk-users] Ongoing problem with 1.8   Ira
• [asterisk-users] Ongoing problem with 1.8   Ira
• [asterisk-users] Ongoing problem with 1.8   Ira
• [asterisk-users] Polycom 500 / MWI   Kai-Uwe Jensen
• [asterisk-users] Realtime SIP, multiple AX servers question   Olle E. Johansson
• [asterisk-users] No more ISDN in Malaysia Telekom???   Arstan Jusupov
• [asterisk-users] No more ISDN in Malaysia Telekom???   Arstan Jusupov
• [asterisk-users] 1.8 MIBs   Kirill Katsnelson
• [asterisk-users] 1.8 MIBs   Kirill Katsnelson
• [asterisk-users] asterisk fax problem   Oguzhan Kayhan
• [asterisk-users] asterisk fax problem   Oguzhan Kayhan
• [asterisk-users] asterisk fax problem   Oğuzhan Kayhan
• [asterisk-users] Go from CALLINGout to just CALLING   Jonas Kellens
• [asterisk-users] Add Privacy: id to SIP-invite   Jonas Kellens
• [asterisk-users] DTMF-troubles with Snom   Jonas Kellens
• [asterisk-users] dtmf-troubles with Snom   Jonas Kellens
• [asterisk-users] DTMF-troubles with Snom   Jonas Kellens
• [asterisk-users] Show voicemail in GUI   Jonas Kellens
• [asterisk-users] Fail2Ban & CSF   Jonas Kellens
• [asterisk-users] queue_log in MySQL database   Jonas Kellens
• [asterisk-users] queue_log in MySQL database   Jonas Kellens
• [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'   Jonas Kellens
• [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'   Jonas Kellens
• [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'   Jonas Kellens
• [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'   Jonas Kellens
• [asterisk-users] audiohook.c: Write factory 0x153cf678 was pretty quick last time, waiting for them   Jonas Kellens
• [asterisk-users] context problem   Jonas Kellens
• [asterisk-users] context problem   Jonas Kellens
• [asterisk-users] context problem   Jonas Kellens
• [asterisk-users] context problem   Jonas Kellens
• [asterisk-users] context problem   Jonas Kellens
• [asterisk-users] Unable to insert cdr-data into mysql-DB   Jonas Kellens
• [asterisk-users] Pickup local/.... not working   Jonas Kellens
• [asterisk-users] Pickup local/.... not working   Jonas Kellens
• [asterisk-users] Pickup local/.... not working   Jonas Kellens
• [asterisk-users] Top Posting   Don Kelly
• [asterisk-users] Top Posting   Don Kelly
• [asterisk-users] Top Posting   Don Kelly
• [asterisk-users] Top Posting   Don Kelly
• [asterisk-users] Top Posting   Don Kelly
• [asterisk-users] Top Posting   Don Kelly
• [asterisk-users] Sendind e-mail with Hylafax   Don Kelly
• [asterisk-users] Top Posting   Don Kelly
• [asterisk-users] Top Posting   Don Kelly
• [asterisk-users] Top Posting   Don Kelly
• [asterisk-users] res_fax   Don Kelly
• [asterisk-users] Mailing list question   Don Kelly
• [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 forT1s?   Don Kelly
• [asterisk-users] Top Posting   Richard Kenner
• [asterisk-users] extconfig, realtime, and SIP   Richard Kenner
• [asterisk-users] Top Posting   William Kenworthy
• [asterisk-users] Bruce B   Kyle Kienapfel
• [asterisk-users] sip dos question   Kyle Kienapfel
• [asterisk-users] Queue - agent auto-answer   Jordan Kirby
• [asterisk-users] Base memory usage   Jeremy Kister
• [asterisk-users] res_fax_digium.so crashing   Jeremy Kister
• [asterisk-users] res_fax_digium.so crashing   Jeremy Kister
• [asterisk-users] res_fax_digium.so crashing   Jeremy Kister
• [asterisk-users] Asterisk 1.8.2.3 Now Available   Jeremy Kister
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Stelios Koroneos
• [asterisk-users] SetVar Warning   Gary Kuznitz
• [asterisk-users] Call hung up?   Gary Kuznitz
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Jeff LaCoursiere
• [asterisk-users] Asterisk replying to wrong port for NOTIFY messages   Jeff LaCoursiere
• [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver   Jeff LaCoursiere
• [asterisk-users] Help determining SpanDSP version   Mr. James W. Laferriere
• [asterisk-users] SendFAX dialplan example   Pezhman Lali
• [asterisk-users] Anybody ever see this before?   Pezhman Lali
• [asterisk-users] Reducing number of Asterisk processes?   Pezhman Lali
• [asterisk-users] Reducing number of Asterisk processes?   Pezhman Lali
• [asterisk-users] faxter   Pezhman Lali
• [asterisk-users] faxter   Pezhman Lali
• [asterisk-users] SILK codec   Edwin Lam
• [asterisk-users] Asterisk replying to wrong port for NOTIFY messages   James Lamanna
• [asterisk-users] Asterisk replying to wrong port for NOTIFY messages   James Lamanna
• [asterisk-users] Failed SIP registration kicks registered device off?   James Lamanna
• [asterisk-users] queue_log in MySQL database   James Lamanna
• [asterisk-users] DTMF not being heard correctly by far end conference system   James Lamanna
• [asterisk-users] feedback mechanism   Lito Lampitoc
• [asterisk-users] How to reject an incoming call using AMI ?   Rodrigo Lang
• [asterisk-users] Forward voicemail not working   Duane Larson
• [asterisk-users] Forward voicemail not working   Duane Larson
• [asterisk-users] Forward voicemail not working   Duane Larson
• [asterisk-users] Mail list Woes?   Andrew Latham
• [asterisk-users] sendrpid does not work!   Andrew Latham
• [asterisk-users] sendrpid does not work!   Andrew Latham
• [asterisk-users] OpenVPN + SIP configuration?   Andrew Latham
• [asterisk-users] Asterisk hardware server   Andrew Latham
• [asterisk-users] Asterisk hardware server   Andrew Latham
• [asterisk-users] Asterisk timezone issue   Andrew Latham
• [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'   Andrew Latham
• [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'   Andrew Latham
• [asterisk-users] Top Posting   Andrew Latham
• [asterisk-users] Max call duration   Andrew Latham
• [asterisk-users] Info on using LDAP with Asterisk?   Andrew Latham
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Andrew Latham
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Andrew Latham
• [asterisk-users] No more ISDN in Malaysia Telekom???   Lee, John (Sydney)
• [asterisk-users] No more ISDN in Malaysia Telekom???   Lee, John (Sydney)
• [asterisk-users] SIP, IAX2 and ISDN ISUP data   Phil Lello
• [asterisk-users] Sound quality issue   Cédric Lemarchand
• [asterisk-users] Sound quality issue   Cédric Lemarchand
• [asterisk-users] Realtime SIP, multiple AX servers question   Tilghman Lesher
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Tilghman Lesher
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Tilghman Lesher
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Tilghman Lesher
• [asterisk-users] Realtime SIP, multiple AX servers question   Tilghman Lesher
• [asterisk-users] environment variable + res_mysql.conf   Tilghman Lesher
• [asterisk-users] Why are 4 ports used for a single call?   Tilghman Lesher
• [asterisk-users] Top Posting   Tilghman Lesher
• [asterisk-users] Top Posting   Tilghman Lesher
• [asterisk-users] Ongoing problem with 1.8   Tilghman Lesher
• [asterisk-users] Ongoing problem with 1.8   Tilghman Lesher
• [asterisk-users] FUNC_ODBC and ARRAY   Tilghman Lesher
• [asterisk-users] Crossover cable for E1 ?   Tilghman Lesher
• [asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny   Tilghman Lesher
• [asterisk-users] Return variables from func_odbc calls?   Tilghman Lesher
• [asterisk-users] Return variables from func_odbc calls?   Tilghman Lesher
• [asterisk-users] CDR issue - Problem logging CDR(userfield) in Master.csv   Tilghman Lesher
• [asterisk-users] Can a duration limit be specified in spool call file?   Tilghman Lesher
• [asterisk-users] Can a duration limit be specified in spool call file?   Tilghman Lesher
• [asterisk-users] Reducing number of Asterisk processes?   Tilghman Lesher
• [asterisk-users] Reducing number of Asterisk processes?   Tilghman Lesher
• [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0   Tilghman Lesher
• [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0   Tilghman Lesher
• [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver   Marc Leurent
• [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver   Marc Leurent
• [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver   Marc Leurent
• [asterisk-users] Call queues on load-balanced asterisks   Thomas Liu
• [asterisk-users] Failed SIP registration kicks registered device off?   Ye Liu
• [asterisk-users] Phone multi-registration   Ye Liu
• [asterisk-users] Do not disturbe   Doug Lytle
• [asterisk-users] Do not disturbe   Doug Lytle
• [asterisk-users] Benefit of PRI vs SIP trunk calls   Doug Lytle
• [asterisk-users] Using the Telco Call Transfer Features.   Doug Lytle
• [asterisk-users] SetVar Warning   Doug Lytle
• [asterisk-users] Music on Hold not working?   Doug Lytle
• [asterisk-users] Music on Hold not working?   Doug Lytle
• [asterisk-users] Polycom 500 / MWI   Doug Lytle
• [asterisk-users] sendrpid does not work!   Mickael MONSIEUR
• [asterisk-users] sendrpid does not work!   Mickael MONSIEUR
• [asterisk-users] MOH and parking   Leif Madsen
• [asterisk-users] Info on using LDAP with Asterisk?   Leif Madsen
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Leif Madsen
• [asterisk-users] Recommended Windows client to display CID?   Leif Madsen
• [asterisk-users] Asterisk 1.8.2.3 Now Available   Leif Madsen
• [asterisk-users] 'Bad authorization' error with Asterisk 1.8   Andre Magalhaes
• [asterisk-users] Blind Transfer not working - 1.4.38   Ishfaq Malik
• [asterisk-users] Blind Transfer not working - 1.4.38   Ishfaq Malik
• [asterisk-users] Show voicemail in GUI   Ishfaq Malik
• [asterisk-users] Blind Transfer not working - 1.4.38   Ishfaq Malik
• [asterisk-users] Asterisk 1.8.2 and digium yum repositories   Ishfaq Malik
• [asterisk-users] Asterisk 1.8.2 and digium yum repositories   Ishfaq Malik
• [asterisk-users] extconfig, realtime, and SIP   Ishfaq Malik
• [asterisk-users] Call forwrading but call transfer back   Asterisk Man
• [asterisk-users] Member penalty and Queue strategies   Asterisk Man
• [asterisk-users] Unable to get Fax t38 working with IrisTel trunk   Karim Mardhani
• [asterisk-users] Benefit of PRI vs SIP trunk calls   Joel Maslak
• [asterisk-users] Top Posting   Joel Maslak
• [asterisk-users] res_fax_digium.so crashing   Joel Maslak
• [asterisk-users] Recommended Windows client to display CID?   Joel Maslak
• [asterisk-users] Recommended Windows client to display CID?   Joel Maslak
• [asterisk-users] FUNC_ODBC and ARRAY   Sherwood McGowan
• [asterisk-users] FUNC_ODBC and ARRAY   Sherwood McGowan
• [asterisk-users] FUNC_ODBC and ARRAY   Sherwood McGowan
• [asterisk-users] end a call after a specific time period   Sherwood McGowan
• [asterisk-users] end a call after a specific time period   Sherwood McGowan
• [asterisk-users] Return variables from func_odbc calls?   Sherwood McGowan
• [asterisk-users] Return variables from func_odbc calls?   Sherwood McGowan
• [asterisk-users] Return variables from func_odbc calls?   Sherwood McGowan
• [asterisk-users] Return variables from func_odbc calls?   Sherwood McGowan
• [asterisk-users] Return variables from func_odbc calls?   Sherwood McGowan
• [asterisk-users] Pickup local/.... not working   Sherwood McGowan
• [asterisk-users] Recommended Windows client to display CID?   Sherwood McGowan
• [asterisk-users] Callback when available   Sherwood McGowan
• [asterisk-users] Queue - agent auto-answer   Sherwood McGowan
• [asterisk-users] Queue - agent auto-answer   Sherwood McGowan
• [asterisk-users] Queue - agent auto-answer   Sherwood McGowan
• [asterisk-users] Multi-Tenant   Sherwood McGowan
• [asterisk-users] Can a duration limit be specified in spool call file?   Sherwood McGowan
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Mike
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Mike
• [asterisk-users] Queue periodic announce...   Mike
• [asterisk-users] Blind Transfer not working - 1.4.38   Mike
• [asterisk-users] Ghost ringing   Mike
• [asterisk-users] Ghost ringing   Mike
• [asterisk-users] Ghost ringing   Mike
• [asterisk-users] Queue - agent auto-answer   Mike
• [asterisk-users] Queue - agent auto-answer   Mike
• [asterisk-users] Queue - agent auto-answer   Mike
• [asterisk-users] Fax stopped working when upgrading to 1.8.2   Vladimir Mikhelson
• [asterisk-users] Fax stopped working when upgrading to 1.8.2   Vladimir Mikhelson
• [asterisk-users] 1.8 MIBs   Barry Miller
• [asterisk-users] Music on Hold not working?   James Miller
• [asterisk-users] Music on Hold not working?   James Miller
• [asterisk-users] Music on Hold not working?   James Miller
• [asterisk-users] Top Posting   James Miller
• [asterisk-users] Top Posting   James Miller
• [asterisk-users] Music on Hold not working?   James M Miller
• [asterisk-users] Top Posting   James M Miller
• [asterisk-users] Do not disturbe   Flavio Miranda
• [asterisk-users] Do not disturbe   Flavio Miranda
• [asterisk-users] WARNING T.30 ECM carrier not found   Flavio Miranda
• [asterisk-users] WARNING T.30 ECM carrier not found   Flavio Miranda
• [asterisk-users] Sendind e-mail with Hylafax   Flavio Miranda
• [asterisk-users] ReceiveFax   Flavio Miranda
• [asterisk-users] ReceiveFax   Flavio Miranda
• [asterisk-users] Channel name changed in asterisk 1.8   Arjan Kroon | Mobillion
• [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'   Arjan Kroon | Mobillion
• [asterisk-users] Caching CALLERID(dnid)   Arjan Kroon | Mobillion
• [asterisk-users] Problems with ZAP Channels   Antonio Modesto
• [asterisk-users] Do not disturbe   Miguel Molina
• [asterisk-users] TE420 issue: card 0 span N: isr2=XX isr3=Y   Tony Mountifield
• [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?   Tony Mountifield
• [asterisk-users] Spectralink 8002   MrHanMan
• [asterisk-users] Spectralink 8002   MrHanMan
• [asterisk-users] Spectralink 8002   MrHanMan
• [asterisk-users] Top Posting   MrHanMan
• [asterisk-users] Asterisk 1.8 and Cisco 7920   MrHanMan
• [asterisk-users] Polarity Reverseal....with analog line   Mark Murawski
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Mark Murawski
• [asterisk-users] Polycom Blf / Directed Pickup   Mark Murawski
• [asterisk-users] Polycom Blf / Directed Pickup   Mark Murawski
• [asterisk-users] Polycom Blf / Directed Pickup   Mark Murawski
• [asterisk-users] Top Posting   Mark Murawski
• [asterisk-users] Top Posting   Mark Murawski
• [asterisk-users] Top Posting   Mark Murawski
• [asterisk-users] Top Posting   Mark Murawski
• [asterisk-users] Dialplan to bridge 2 legs?   Mark Murawski
• [asterisk-users] Calls Transfers   Elliot Murdock
• [asterisk-users] Calls Transfers   Elliot Murdock
• [asterisk-users] T.38 Digium Fax Driver Success on Fail   Elliot Murdock
• [asterisk-users] Regarding error in Asterisk dail plan:   Steve Murphy
• [asterisk-users] Voicemail hangs up   Alan Murrell
• [asterisk-users] Bruce B   Tim Nelson
• [asterisk-users] Asterisk to asterisk t.38   Amit Nepal
• [asterisk-users] Asterisk to asterisk t.38   Amit Nepal
• [asterisk-users] Asterisk to asterisk t.38   Amit Nepal
• [asterisk-users] Asterisk to asterisk t.38   Amit Nepal
• [asterisk-users] Callback form to place on site for customers. Recomendation to achieve this.   Danny Nicholas
• [asterisk-users] Call forwrading but call transfer back   Danny Nicholas
• [asterisk-users] Calls Transfers   Danny Nicholas
• [asterisk-users] Call queues on load-balanced asterisks   Danny Nicholas
• [asterisk-users] Show voicemail in GUI   Danny Nicholas
• [asterisk-users] queue_log in MySQL database   Danny Nicholas
• [asterisk-users] queue_log in MySQL database   Danny Nicholas
• [asterisk-users] CallerID and URL pop up for windows...   Danny Nicholas
• [asterisk-users] CallerID and URL pop up for windows...   Danny Nicholas
• [asterisk-users] 5-7 second delay in connecting outgoing FXO calls   Danny Nicholas
• [asterisk-users] Why are 4 ports used for a single call?   Danny Nicholas
• [asterisk-users] how to read mp3   Danny Nicholas
• [asterisk-users] Top Posting   Danny Nicholas
• [asterisk-users] how to read mp3   Danny Nicholas
• [asterisk-users] Top Posting   Danny Nicholas
• [asterisk-users] Calling rules   Danny Nicholas
• [asterisk-users] Calling rules   Danny Nicholas
• [asterisk-users] 1.8.2: dahdi-2.4: calls dropping   Danny Nicholas
• [asterisk-users] agi dial termination cause ?   Danny Nicholas
• [asterisk-users] Top Posting   Danny Nicholas
• [asterisk-users] Top Posting   Danny Nicholas
• [asterisk-users] Hi, agent intro-speech for outside caller   Danny Nicholas
• [asterisk-users] context problem   Danny Nicholas
• [asterisk-users] OT - TTS in spanish   Danny Nicholas
• [asterisk-users] ReceiveFax   Danny Nicholas
• [asterisk-users] context problem   Danny Nicholas
• [asterisk-users] Mailing list question   Danny Nicholas
• [asterisk-users] U-verse DTMF tuning for Zaptel   Danny Nicholas
• [asterisk-users] regarding quit, exit and stop now in asterisk   Danny Nicholas
• [asterisk-users] Help determining SpanDSP version   Danny Nicholas
• [asterisk-users] ReceiveFAX issue.   Danny Nicholas
• [asterisk-users] Pickup local/.... not working   Danny Nicholas
• [asterisk-users] Pickup local/.... not working   Danny Nicholas
• [asterisk-users] How to update sound files?   Danny Nicholas
• [asterisk-users] OPTIONS Packet is retransmitting continuously   Nikhil
• [asterisk-users] Why Local Channels are creating   Nikhil
• [asterisk-users] OpenVPN + SIP configuration?   Dale Noll
• [asterisk-users] Inbound routes   Dale Noll
• [asterisk-users] Call Back on Busy   John Novack
• [asterisk-users] Selecing the E1 cards for the call center   John Novack
• [asterisk-users] Bruce B   John Novack
• [asterisk-users] Top Posting   John Novack
• [asterisk-users] Top Posting   John Novack
• [asterisk-users] A1200P comments?   Valer Nur
• [asterisk-users] How to detect line tone?   Massimo Nuvoli
• [asterisk-users] app_calendar and SSL   Muhammad Nuzaihan
• [asterisk-users] OT - Contact center - How to Gmail-like label to incoming email   Olivier
• [asterisk-users] How to reject an incoming call using AMI ?   Olivier
• [asterisk-users] How to reject an incoming call using AMI ?   Olivier
• [asterisk-users] How to reject an incoming call using AMI ?   Olivier
• [asterisk-users] How to reject an incoming call using AMI ?   Olivier
• [asterisk-users] OT - TTS in spanish   Olivier
• [asterisk-users] Phone multi-registration   Olivier
• [asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny   Olivier
• [asterisk-users] Caching CALLERID(dnid)   Olivier
• [asterisk-users] Top Posting   Chris Owen
• [asterisk-users] Sound quality issue   Sevana Oy
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Tim Panton
• [asterisk-users] Crossover cable for E1 ?   Tim Panton
• [asterisk-users] Top Posting   Jason Parker
• [asterisk-users] Asterisk 1.8.2 and digium yum repositories   Jason Parker
• [asterisk-users] res_fax   Jason Parker
• [asterisk-users] Problems with ZAP Channels   Satish Patel
• [asterisk-users] Unable to insert cdr-data into mysql-DB   Marius Pedersen
• [asterisk-users] incoming   Thomas Perron
• [asterisk-users] incoming   Thomas Perron
• [asterisk-users] Basic Sip.conf and extensions.conf   Thomas Perron
• [asterisk-users] Basic Sip.conf and extensions.conf   Thomas Perron
• [asterisk-users] Basic Sip.conf and extensions.conf   Thomas Perron
• [asterisk-users] console debugging   Thomas Perron
• [asterisk-users] console debugging   Thomas Perron
• [asterisk-users] Recommended Windows client to display CID?   Magnus Persson
• [asterisk-users] mobile integration   Wolfgang Pichler
• [asterisk-users] Anyone have Festival application working?   Ian Pilcher
• [asterisk-users] Make ConfBridge hang up on last participant   Ian Pilcher
• [asterisk-users] U-verse DTMF tuning for Zaptel   Ian Pilcher
• [asterisk-users] problems inserting dahdi modules using Debian Leni   Dave Platt
• [asterisk-users] context problem   Dave Platt
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Dave Platt
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Dave Platt
• [asterisk-users] Top Posting   Fred Posner
• [asterisk-users] Top Posting   Fred Posner
• [asterisk-users] Top Posting   Fred Posner
• [asterisk-users] Polarity Reverseal....with analog line   Edwin Quijada
• [asterisk-users] Polarity Reverseal....with analog line   Edwin Quijada
• [asterisk-users] Issue with Red Alarm with DAhDi   Edwin Quijada
• [asterisk-users] Issue with Red Alarm with DAhDi   Edwin Quijada
• [asterisk-users] Callback form to place on site for customers. Recomendation to achieve this.   Randy R
• [asterisk-users] iNum at 12 Noon EST Friday   Randy R
• [asterisk-users] Bufferbloat! Friday on VUC @ 12 Noon EST   Randy R
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Asterisk on Debian Lenny with timerfd   RR
• [asterisk-users] Top Posting   Anton Raharja
• [asterisk-users] Asterisk Scenary   José Luis Hernández Ramos
• [asterisk-users] Multi-Tenant   Amardeep Rana
• [asterisk-users] Recommended Windows client to display CID?   Amardeep Rana
• [asterisk-users] Basic Sip.conf and extensions.conf   Zuhair Raza
• [asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny   JR Richardson
• [asterisk-users] Top Posting   Matt Riddell
• [asterisk-users] B410P: DAHDI BRI PTMP HDLC Abort (6) on Primary D-channel of span 1 (TEI Errors)   Matt Riddell
• [asterisk-users] Internode weirdness   Da Rock
• [asterisk-users] Internode weirdness   Da Rock
• [asterisk-users] Internode weirdness   Da Rock
• [asterisk-users] Internode weirdness   Da Rock
• [asterisk-users] Asterisk SIP with no RTP audio port (was Internode weirdness)   Da Rock
• [asterisk-users] Asterisk SIP with no RTP audio port (was Internode weirdness)   Da Rock
• [asterisk-users] SIP RTP streams   Da Rock
• [asterisk-users] Call Back on Busy   Ron
• [asterisk-users] Call Back on Busy   Ron
• [asterisk-users] slow response to INVITE   Ron
• [asterisk-users] digim tdm2400p fxo fake answer supervision problem.   Shaun Ruffell
• [asterisk-users] problems inserting dahdi modules using Debian Leni   Shaun Ruffell
• [asterisk-users] problems inserting dahdi modules using Debian Leni   Shaun Ruffell
• [asterisk-users] problems inserting dahdi modules using Debian Leni   Shaun Ruffell
• [asterisk-users] problems inserting dahdi modules using Debian Leni   Shaun Ruffell
• [asterisk-users] New Dahdi error   Shaun Ruffell
• [asterisk-users] New Dahdi error   Shaun Ruffell
• [asterisk-users] Fix Fake Answer Supervision In asterisk1.6   Shaun Ruffell
• [asterisk-users] Issue with Red Alarm with DAhDi   Shaun Ruffell
• [asterisk-users] Issue with Red Alarm with DAhDi   Shaun Ruffell
• [asterisk-users] Top Posting   Shaun Ruffell
• [asterisk-users] 1.8.2: dahdi-2.4: calls dropping   Shaun Ruffell
• [asterisk-users] 1.8.2: dahdi-2.4: calls dropping   Shaun Ruffell
• [asterisk-users] Error compiling Dahdi: invalid use of undefined type struct module   Shaun Ruffell
• [asterisk-users] Clarification on DAHDI Fax Detection   Thomas Rymes
• [asterisk-users] MOH problems (asterisk 1.4.38)   Thomas Rymes
• [asterisk-users] Clarification on DAHDI Fax Detection   Thomas Rymes
• [asterisk-users] Too Few Fax Detections   Thomas Rymes
• [asterisk-users] Clarification on DAHDI Fax Detection   Tom Rymes
• [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?   Tom Rymes
• [asterisk-users] Clarification on DAHDI Fax Detection   Tom Rymes
• [asterisk-users] VoIP PoE phones for restaurant   Tom Rymes
• [asterisk-users] MOH problems (asterisk 1.4.38)   Tom Rymes
• [asterisk-users] DAHDI and dialdebounce   Tom Rymes
• [asterisk-users] Replacing digital pri card   Tom Rymes
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Tom Rymes
• [asterisk-users] Too Few Fax Detections   Tom Rymes
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   Tom Rymes
• [asterisk-users] Benefit of PRI vs SIP trunk calls   Tom Rymes
• [asterisk-users] Mail list Woes?   Tom Rymes
• [asterisk-users] queue_log in MySQL database   Tom Rymes
• [asterisk-users] queue_log in MySQL database   Tom Rymes
• [asterisk-users] Selecing the E1 cards for the call center   Tom Rymes
• [asterisk-users] Why are 4 ports used for a single call?   Tom Rymes
• [asterisk-users] Why are 4 ports used for a single call?   Tom Rymes
• [asterisk-users] Why are 4 ports used for a single call?   Tom Rymes
• [asterisk-users] Why are 4 ports used for a single call?   Tom Rymes
• [asterisk-users] Top Posting   Tom Rymes
• [asterisk-users] Asterisk stops responding   Tom Rymes
• [asterisk-users] Top Posting   Tom Rymes
• [asterisk-users] Selecting the E1 cards for the call   Tom Rymes
• [asterisk-users] Top Posting   Tom Rymes
• [asterisk-users] res_fax_digium.so crashing   Tom Rymes
• [asterisk-users] Calling rules   Tom Rymes
• [asterisk-users] res_fax   Tom Rymes
• [asterisk-users] Top Posting   Tom Rymes
• [asterisk-users] Calling rules   Tom Rymes
• [asterisk-users] Hi, agent intro-speech for outside caller   Tom Rymes
• [asterisk-users] context problem   Tom Rymes
• [asterisk-users] Internode weirdness   Tom Rymes
• [asterisk-users] Asterisk to asterisk t.38   Tom Rymes
• [asterisk-users] Asterisk to asterisk t.38   Tom Rymes
• [asterisk-users] res_fax   Tom Rymes
• [asterisk-users] Inbound routes   Tom Rymes
• [asterisk-users] res_fax   Tom Rymes
• [asterisk-users] Help determining SpanDSP version   Tom Rymes
• [asterisk-users] Help determining SpanDSP version   Tom Rymes
• [asterisk-users] Recommended Windows client to display CID?   Tom Rymes
• [asterisk-users] res_fax   Tom Rymes
• [asterisk-users] res_fax   Tom Rymes
• [asterisk-users] faxter   Tom Rymes
• [asterisk-users] save the calls with asterisk   Tom Rymes
• [asterisk-users] save the calls with asterisk   Tom Rymes
• [asterisk-users] end a call after a specific time period   ABBAS SHAKEEL
• [asterisk-users] end a call after a specific time period   ABBAS SHAKEEL
• [asterisk-users] end a call after a specific time period   ABBAS SHAKEEL
• [asterisk-users] Force Dahdi modules to load   Pablo Schuhwerk
• [asterisk-users] Log and forward calls to cellphone?   Sebastian
• [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?   Sebastian
• [asterisk-users] load balance with 2 wan connections   Sebastian
• [asterisk-users] load balance with 2 wan connections   Sebastian
• [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?   Sebastian
• [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?   Sebastian
• [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)   Sebastian
• [asterisk-users] Log and forward calls to cellphone?   Sebastian
• [asterisk-users] Log and forward calls to cellphone?   Sebastian
• [asterisk-users] Log and forward calls to cellphone?   Sebastian
• [asterisk-users] Log and forward calls to cellphone?   Sebastian
• [asterisk-users] OpenVPN + SIP configuration?   Sebastian
• [asterisk-users] Music on Hold not working?   Warren Selby
• [asterisk-users] Music on Hold not working?   Warren Selby
• [asterisk-users] Phone multi-registration   Warren Selby
• [asterisk-users] Asterisk 1.8.2.3 Now Available   Warren Selby
• [asterisk-users] How to update sound files?   Warren Selby
• [asterisk-users] console debugging   Warren Selby
• [asterisk-users] Asterisk stops responding   Justin Sherrill
• [asterisk-users] Do I need a sip proxy?   Andreas Sikkema
• [asterisk-users] Sound quality issue   Andreas Sikkema
• [asterisk-users] Occasional robotic sound while call in progress   Andreas Sikkema
• [asterisk-users] Problem registering two (and more) sip trunks   Luis Silva
• [asterisk-users] Cisco IP Phones and AVAYA IP Phones: How to configure in Asterisk   Cassius Smith
• [asterisk-users] TDM410 and DSL   Cassius Smith
• [asterisk-users] environment variable + res_mysql.conf   Chandrakant Solanki
• [asterisk-users] Log and forward calls to cellphone?   A J Stiles
• [asterisk-users] how to read mp3   A J Stiles
• [asterisk-users] Top Posting   A J Stiles
• [asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny   A J Stiles
• [asterisk-users] Unable to receive calls (inbound)   A J Stiles
• [asterisk-users] Crossover cable for E1 ?   A J Stiles
• [asterisk-users] Crossover cable for E1 ?   A J Stiles
• [asterisk-users] Unable to insert cdr-data into mysql-DB   A J Stiles
• [asterisk-users] Recommended Windows client to display CID?   A J Stiles
• [asterisk-users] Recommended Windows client to display CID?   A J Stiles
• [asterisk-users] AGI->Macro w/Agruments   William Stillwell
• [asterisk-users] Mail list Woes?   William Stillwell
• [asterisk-users] AGI->Macro w/Agruments   William Stillwell
• [asterisk-users] ReceiveFax   William Stillwell
• [asterisk-users] Accessing a 'user' variable via. dialplan.   William Stillwell
• [asterisk-users] spandsp download   William Stillwell
• [asterisk-users] FUNC_ODBC and ARRAY   William Stillwell
• [asterisk-users] Anybody ever see this before?   William Stillwell
• [asterisk-users] Reducing number of Asterisk processes?   William Stillwell
• [asterisk-users] Hi, agent intro-speech for outside caller   Bart Swedrowski
• [asterisk-users] Recommended Windows client to display CID?   TDF
• [asterisk-users] Log and forward calls to cellphone?   Administrator TOOTAI
• [asterisk-users] Log and forward calls to cellphone?   Administrator TOOTAI
• [asterisk-users] Log and forward calls to cellphone?   Administrator TOOTAI
• [asterisk-users] Log and forward calls to cellphone?   Administrator TOOTAI
• [asterisk-users] Sound quality issue   Administrator TOOTAI
• [asterisk-users] 5-7 second delay in connecting outgoing FXO calls   Frank Tarczynski
• [asterisk-users] Outgoing FXO calls have no audio with callprogress=no   Frank Tarczynski
• [asterisk-users] Outgoing FXO calls have no audio with callprogress=no   Frank Tarczynski
• [asterisk-users] cannot answer incoming calls   John Taylor
• [asterisk-users] intermittent problem on 1.4   John Taylor
• [asterisk-users] Asterisk 1.4.39 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.16 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.3 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.2 Now Available (not 1.8.3)   Asterisk Development Team
• [asterisk-users] Asterisk Security Releases: AST-2011-001   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.2.2 Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.2.3 Now Available   Asterisk Development Team
• [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver   Asterisk Security Team
• [asterisk-users] MOH problems (asterisk 1.4.38)   Earl Terwilliger
• [asterisk-users] MOH problems (asterisk 1.4.38)   Earl Terwilliger
• [asterisk-users] MOH problems (asterisk 1.4.38)   Earl Terwilliger
• [asterisk-users] Top Posting   Andrew Thomas
• [asterisk-users] Sound quality issue   Andrew Thomas
• [asterisk-users] Top Posting   Andrew Thomas
• [asterisk-users] Top Posting   Andrew Thomas
• [asterisk-users] Top Posting   Andrew Thomas
• [asterisk-users] Accessing a 'user' variable via. dialplan.   Andrew Thomas
• [asterisk-users] Accessing a 'user' variable via. dialplan.   Andrew Thomas
• [asterisk-users] context problem   Andrew Thomas
• [asterisk-users] Mailing list question   Andrew Thomas
• [asterisk-users] Mailing list question   Andrew Thomas
• [asterisk-users] Mailing list question   Andrew Thomas
• [asterisk-users] Mailing list question 2   Andrew Thomas
• [asterisk-users] Mailing list question   Andrew Thomas
• [asterisk-users] Mailing list question 2   Andrew Thomas
• [asterisk-users] Mailing list question   Andrew Thomas
• [asterisk-users] MOH and parking   Andrew Thomas
• [asterisk-users] Unable to insert cdr-data into mysql-DB   Andrew Thomas
• [asterisk-users] MOH and parking   Andrew Thomas
• [asterisk-users] Polycom Blf / Directed Pickup   Sebastien Thomas
• [asterisk-users] Polycom Blf / Directed Pickup   Sebastien Thomas
• [asterisk-users] Fwd: Announce: telepathy-ring 2.1.1   Steve Totaro
• [asterisk-users] T.38 Digium Fax Driver Success on Fail   Steve Totaro
• [asterisk-users] res_fax_digium.so crashing   Steve Totaro
• [asterisk-users] Queues, priorities and (miscalculated) holdtimes   Daniel Tryba
• [asterisk-users] Force different codecs on call base   Daniel Tryba
• [asterisk-users] CDR issue - Problem logging CDR(userfield) in Master.csv   Athanasia Tsertou
• [asterisk-users] RTCP packets when on hold   Ryan Tucker
• [asterisk-users] RTCP packets when on hold   Ryan Tucker
• [asterisk-users] RTP keepalive doesn't work   Ryan Tucker
• [asterisk-users] A1200P comments?   Ryan Tucker
• [asterisk-users] RTP keepalive doesn't work   Ryan Tucker
• [asterisk-users] RTP keepalive doesn't work   Ryan Tucker
• [asterisk-users] DTMF not being heard correctly by far end conference system   Duncan Turnbull
• [asterisk-users] DTMF not being heard correctly by far end conference system   Duncan Turnbull
• [asterisk-users] Clarification on DAHDI Fax Detection   Steve Underwood
• [asterisk-users] Clarification on DAHDI Fax Detection   Steve Underwood
• [asterisk-users] Clarification on DAHDI Fax Detection   Steve Underwood
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Steve Underwood
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Steve Underwood
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Steve Underwood
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Steve Underwood
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   Steve Underwood
• [asterisk-users] asterisk fax problem   Steve Underwood
• [asterisk-users] res_fax_digium.so crashing   Steve Underwood
• [asterisk-users] ReceiveFax   Steve Underwood
• [asterisk-users] res_fax   Steve Underwood
• [asterisk-users] res_fax   Steve Underwood
• [asterisk-users] res_fax   Steve Underwood
• [asterisk-users] spandsp download   Steve Underwood
• [asterisk-users] Polycom Blf / Directed Pickup   Gord Urquhart
• [asterisk-users] digim tdm2400p fxo fake answer supervision problem.   Muhammad Usman
• [asterisk-users] Fix Fake Answer Supervision In asterisk1.6   Muhammad Usman
• [asterisk-users] chan_sip.c: Failed to parse contact info   Nick Ustinov
• [asterisk-users] Top Posting   Vince Vielhaber
• [asterisk-users] ReceiveFAX issue.   Kristijan Vrban
• [asterisk-users] Weird phone behavior after recent CentOS 5 update   Myles Wakeham
• [asterisk-users] Weird phone behavior after recent CentOS 5 update   Myles Wakeham
• [asterisk-users] Hint for queues   Philipp Walker
• [asterisk-users] Forward voicemail not working   Chad Wallace
• [asterisk-users] Occasional robotic sound while call in progress   Chad Wallace
• [asterisk-users] Top Posting   Chad Wallace
• [asterisk-users] chan_sip bug? (Asterisk 1.4)   Chad Wallace
• [asterisk-users] Replacing digital pri card   Matt Watson
• [asterisk-users] incoming   Roger Burton West
• [asterisk-users] Callback form to place on site for customers. Recomendation to achieve this.   Roger Burton West
• [asterisk-users] Failed SIP registration kicks registered device off?   Roger Burton West
• [asterisk-users] Asterisk on Debian Lenny with timerfd   Roger Burton West
• [asterisk-users] Why are 4 ports used for a single call?   David White
• [asterisk-users] load balance with 2 wan connections   Hans Witvliet
• [asterisk-users] OpenVPN + SIP configuration?   Hans Witvliet
• [asterisk-users] Continuously core dumping of 1.8 on SLES   Hans Witvliet
• [asterisk-users] SIP client floods port 5060 and gets blocked   Julian Yap
• [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?   Zeeshan Zakaria
• [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?   Zeeshan Zakaria
• [asterisk-users] DIALSTATUS on CANCEL   Bryant Zimmerman
• [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)   Bryant Zimmerman
• [asterisk-users] Add Privacy: id to SIP-invite   Bryant Zimmerman
• [asterisk-users] DTMF-troubles with Snom   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] ReceiveFax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] Asterisk to asterisk t.38   Bryant Zimmerman
• [asterisk-users] spandsp download   Bryant Zimmerman
• [asterisk-users] ReceiveFAX issue.   Bryant Zimmerman
• [asterisk-users] ReceiveFAX issue.   Bryant Zimmerman
• [asterisk-users] ReceiveFAX issue.   Bryant Zimmerman
• [asterisk-users] ReceiveFAX issue.   Bryant Zimmerman
• [asterisk-users] Regarding error in Asterisk dail plan:   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] res_fax   Bryant Zimmerman
• [asterisk-users] Voicemail Forwarding   --[ UxBoD ]--
• [asterisk-users] app_calendar and SSL   --[ UxBoD ]--
• [asterisk-users] app_calendar and SSL   --[ UxBoD ]--
• [asterisk-users] Asterisk extension not found problem...   abhinav anand
• [asterisk-users] Asterisk extension not found problem...   abhinav anand
• [asterisk-users] Asterisk extension not found problem...   abhinav anand
• [asterisk-users] Asterisk extension not found problem...   abhinav anand
• [asterisk-users] Asterisk extension not found problem...   abhinav anand
• [asterisk-users] MOH problems (asterisk 1.4.38)   bakko
• [asterisk-users] Unable to insert cdr-data into mysql-DB   bakko
• [asterisk-users] SendFAX dialplan example   bakko
• [asterisk-users] DNS A queries   tkawanob at bb.softbank.co.jp
• [asterisk-users] problems inserting dahdi modules using Debian Leni   covici at ccs.covici.com
• [asterisk-users] problems inserting dahdi modules using Debian Leni   covici at ccs.covici.com
• [asterisk-users] problems inserting dahdi modules using Debian Leni   covici at ccs.covici.com
• [asterisk-users] problems inserting dahdi modules using Debian Leni   covici at ccs.covici.com
• [asterisk-users] problems inserting dahdi modules using Debian Leni   covici at ccs.covici.com
• [asterisk-users] problems inserting dahdi modules using Debian Leni   covici at ccs.covici.com
• [asterisk-users] problems inserting dahdi modules using Debian Leni   covici at ccs.covici.com
• [asterisk-users] system lockup when going into conference   covici at ccs.covici.com
• [asterisk-users] system lockup when going into conference   covici at ccs.covici.com
• [asterisk-users] Problem with chan_dahdi and conferencing   covici at ccs.covici.com
• [asterisk-users] Problem with chan_dahdi and conferencing   covici at ccs.covici.com
• [asterisk-users] Problem with chan_dahdi and conferencing   covici at ccs.covici.com
• [asterisk-users] Problem with chan_dahdi and conferencing   covici at ccs.covici.com
• [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0   covici at ccs.covici.com
• [asterisk-users] VoIP PoE phones for restaurant   cjwstudios
• [asterisk-users] New Dahdi error   cjwstudios
• [asterisk-users] New Dahdi error   cjwstudios
• [asterisk-users] using google for vm transcripts   sean darcy
• [asterisk-users] 1.8.2: dahdi-2.4: calls dropping   sean darcy
• [asterisk-users] 1.8.2: dahdi-2.4: calls dropping   sean darcy
• [asterisk-users] 1.8.2: dahdi-2.4: calls dropping   sean darcy
• [asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny   dotnetdub
• [asterisk-users] Polycom SoundPoint IP 650 freezes on boot after adding just one custom ringtone   Marco Lechner - FOSSGIS e.V.
• [asterisk-users] how to read mp3   salaheddine elharit
• [asterisk-users] how to read mp3   salaheddine elharit
• [asterisk-users] how to read mp3   salaheddine elharit
• [asterisk-users] save the calls with asterisk   salaheddine elharit
• [asterisk-users] 5-7 second delay in connecting outgoing FXO calls   ftarz
• [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)   bilal ghayyad
• [asterisk-users] Cisco IP Phones and AVAYA IP Phones: How to configure in Asterisk   bilal ghayyad
• [asterisk-users] Selecing the E1 cards for the call center   bilal ghayyad
• [asterisk-users] Selecting the E1 cards for the call   bilal ghayyad
• [asterisk-users] Forward voicemail not working   duane.larson at gmail.com
• [asterisk-users] agi dial termination cause ?   mancyborg at gmail.com
• [asterisk-users] agi dial termination cause ?   mancyborg at gmail.com
• [asterisk-users] Mail list Woes?   jonp at inline.net
• [asterisk-users] Fax stopped working when upgrading to 1.8.2   magnus.b at inputinterior.se
• [asterisk-users] Fax stopped working when upgrading to 1.8.2   magnus.b at inputinterior.se
• [asterisk-users] Fax stopped working when upgrading to 1.8.2   magnus.b at inputinterior.se
• [asterisk-users] SendFAX dialplan example   magnus.b at inputinterior.se
• [asterisk-users] Ghost ringing   jfratantoni at iswan.net
• [asterisk-users] Ghost ringing   jfratantoni at iswan.net
• [asterisk-users] Ghost ringing   jfratantoni at iswan.net
• [asterisk-users] regarding quit, exit and stop now in asterisk   viswavardhanreddy karna
• [asterisk-users] regarding quit, exit and stop now in asterisk   viswavardhanreddy karna
• [asterisk-users] Regarding error in Asterisk dail plan:   viswavardhanreddy karna
• [asterisk-users] Regarding error in Asterisk dail plan:   viswavardhanreddy karna
• [asterisk-users] Regarding error in Asterisk dail plan:   viswavardhanreddy karna
• [asterisk-users] list of errorswhile registering client at asterisk with sipp   viswavardhanreddy karna
• [asterisk-users] Regarding error in asterisk or SIPp   viswavardhanreddy karna
• [asterisk-users] Fwd: regarding error in SIPp   viswavardhanreddy karna
• [asterisk-users] regarding error in asterisk   viswavardhanreddy karna
• [asterisk-users] Top Posting   markus_weiler at mailworks.org
• [asterisk-users] VoIP PoE phones for restaurant (kitchen)   mgraves at mstvp.com
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   mgraves at mstvp.com
• [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?   mgraves at mstvp.com
• [asterisk-users] Can I know if a call is transffered to asterisk   ishagh ouldbah
• [asterisk-users] Asterisk+h324m gateway issue   pankaj pandey
• [asterisk-users] Call forwrading but call transfer back   satish patel
• [asterisk-users] Asterisk hardware server   satish patel
• [asterisk-users] Asterisk hardware server   satish patel
• [asterisk-users] Asterisk timezone issue   satish patel
• [asterisk-users] Asterisk timezone issue   satish patel
• [asterisk-users] Queue_log with Splunk   satish patel
• [asterisk-users] Top Posting   jon pounder
• [asterisk-users] Mailing list question   jon pounder
• [asterisk-users] Recommended Windows client to display CID?   jon pounder
• [asterisk-users] Friday Jan 14th @ 12 Noon EST: Humbug   randulo
• [asterisk-users] Top Posting   randulo
• [asterisk-users] Top Posting   randulo
• [asterisk-users] Top Posting   randulo
• [asterisk-users] Grandstream GXE2504A codec disable option   amit salunkhe
• [asterisk-users] Returned mail: see transcript for details   tzafrir.cohen at xorcom.com
• [asterisk-users] res_fax   BryantZ at zktech.com
• [asterisk-users] How to update sound files?   Сикорский Сергей
• [asterisk-users] How to update sound files?   Сикорский Сергей
• [asterisk-users] How to update sound files?   Сикорский Сергей
• [asterisk-users] Asterisk 1.8.2 - TLS, user certificate   Gilles 李乐

Last message date: Mon Jan 31 21:09:05 CST 2011
Archived on: Mon Apr 11 11:29:20 CDT 2011

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