February 2010 Archives by author

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Starting: Mon Feb 1 01:38:36 CST 2010
Ending: Sun Feb 28 20:21:23 CST 2010
Messages: 1312

• [asterisk-users] Set CDR userfield for Queues   William Stillwell (Lists)
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   William Stillwell (Lists)
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   William Stillwell (Lists)
• [asterisk-users] IP Phone recommendation   William Stillwell (Lists)
• [asterisk-users] Issue with trying to dial two different servers at the same time.   William Stillwell (Lists)
• [asterisk-users] Qeuee/Agent Question   William Stillwell (Lists)
• [asterisk-users] Re-INVITE on BYE   Lawrence Na (MY-RND at Vyke)
• [asterisk-users] Re-INVITE on BYE   Lawrence Na (MY-RND at Vyke)
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Alan Lord (News)
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Alan Lord (News)
• [asterisk-users] Problem with BLF's   Samael -
• [asterisk-users] Ideasip   David at ULC
• [asterisk-users] Static IP   David at ULC
• [asterisk-users] Static IP   David at ULC
• [asterisk-users] Static IP   David at ULC
• [asterisk-users] Static IP   David at ULC
• [asterisk-users] Static IP   David at ULC
• [asterisk-users] Static IP   David at ULC
• [asterisk-users] Static IP   David at ULC
• [asterisk-users] Static IP   David at ULC
• [asterisk-users] IPKall NOT coming on Asterisk   David at ULC
• [asterisk-users] AMD: HANGUP   David at ULC
• [asterisk-users] AMD: HANGUP   David at ULC
• [asterisk-users] AMI Originate differences between 1.4 and 1.6.1   Ritesh A
• [asterisk-users] AMI Originate differences between 1.4 and 1.6.1   Ritesh A
• [asterisk-users] hi   Ciprian ARSENIE
• [asterisk-users] hi   Ciprian ARSENIE
• [asterisk-users] hi   Ciprian ARSENIE
• [asterisk-users] EAGI delay   Jonathan Addleman
• [asterisk-users] problems with 1.6   Jonathan Addleman
• [asterisk-users] problems with 1.6   Jonathan Addleman
• [asterisk-users] Virtual machine timing (KVM)   Jonathan Addleman
• [asterisk-users] Virtual machine timing (KVM)   Jonathan Addleman
• [asterisk-users] audio glitches in conference   Jonathan Addleman
• [asterisk-users] audio glitches in conference   Jonathan Addleman
• [asterisk-users] audio glitches in conference   Jonathan Addleman
• [asterisk-users] audio glitches in conference   Jonathan Addleman
• [asterisk-users] Moh help needed   Max Alex
• [asterisk-users] Multiple instances of Asterisk on the same host...   Roderick A. Anderson
• [asterisk-users] PAP2   Andres
• [asterisk-users] Issue with trying to dial two different servers at the same time.   Steve Anness
• [asterisk-users] Increasing the dahdi chunk size with Sangoma cards   Lee Archer
• [asterisk-users] Odd error mssage on DAHDI lines   Yves Arikoglu
• [asterisk-users] Asterisk -> SIP-ROUTER -> Internet = no audio   Yves Arikoglu
• [asterisk-users] Asterisk -> SIP-ROUTER -> Internet = no audio   Yves Arikoglu
• [asterisk-users] GSM Gateway   Ron Arts
• [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals   Ron Arts
• [asterisk-users] Robotic sound sometimes   Ron Arts
• [asterisk-users] identify the costumer   Ron Arts
• [asterisk-users] how to remove asterisk from this string X-Asterisk-HangupCauseCode   Mian Asif
• [asterisk-users] Cisco 7940: showing FWD in display.   Michiel van Baak
• [asterisk-users] Important security alert: update your?dialplans now!   Michiel van Baak
• [asterisk-users] Problems with recordings of call using Monitor   David Backeberg
• [asterisk-users] How to run a remote PHP script and still have access to audio stream?   David Backeberg
• [asterisk-users] conferencing without DAHDI   David Backeberg
• [asterisk-users] conferencing without DAHDI   David Backeberg
• [asterisk-users] Know what would be killer?   David Backeberg
• [asterisk-users] agi debug in Asterisk 1.6?   David Backeberg
• [asterisk-users] [asterisk-dev] Maximum call handling capacity on single server   David Backeberg
• [asterisk-users] Virtual machine timing (KVM)   David Backeberg
• [asterisk-users] How to run a remote PHP script and still have access to audio stream?   David Backeberg
• [asterisk-users] How to run a remote PHP script and still have access to audio stream?   David Backeberg
• [asterisk-users] 4 PCIe cards in one asterisk server   David Backeberg
• [asterisk-users] Virtual machine timing (KVM)   David Backeberg
• [asterisk-users] Virtual machine timing (KVM)   David Backeberg
• [asterisk-users] Problem w/ MoH   David Backeberg
• [asterisk-users] subject: 1.4 vs 1.6   David Backeberg
• [asterisk-users] Asterisk RPM's   David Backeberg
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Chris Bagnall
• [asterisk-users] uri tel: instead of sip:accepted ?   Alex Balashov
• [asterisk-users] OpenVPN on phones?   Alex Balashov
• [asterisk-users] OpenVPN on phones?   Alex Balashov
• [asterisk-users] OpenVPN on phones?   Alex Balashov
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Alex Balashov
• [asterisk-users] Get Talk Time   Alex Balashov
• [asterisk-users] IP Phone recommendation   Alex Balashov
• [asterisk-users] Registering of Asterisk against a SIP provider   Daniel Bareiro
• [asterisk-users] Registering of Asterisk against a SIP provider   Daniel Bareiro
• [asterisk-users] Registering of Asterisk against a SIP provider   Daniel Bareiro
• [asterisk-users] Registering of Asterisk against a SIP provider   Daniel Bareiro
• [asterisk-users] Registering of Asterisk against a SIP provider   Daniel Bareiro
• [asterisk-users] Error redirecting an incoming call of a SIP provider to a local extension   Daniel Bareiro
• [asterisk-users] Intel Atom based Asterisk server?   Remco Barendse
• [asterisk-users] directmedia/canreinvite/native bridging question   Jack Bates
• [asterisk-users] High codec translation times on x64   Hristo Benev
• [asterisk-users] CONNECTEDLINE   Magnus Benngård
• [asterisk-users] Fax, T38 and NAT   Magnus Benngård
• [asterisk-users] Fax, T38 and NAT   Magnus Benngård
• [asterisk-users] Fax, T38 and NAT   Magnus Benngård
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Magnus Benngård
• [asterisk-users] Error redirecting an incoming call of a SIP provider to a local extension   Benoit
• [asterisk-users] Dahdi & Congestion status   Benoit
• [asterisk-users] TE410P Spans offline/red after power down/restart   Benoit
• [asterisk-users] Issue when reloading   Pablo Bernasconi
• [asterisk-users] g722 IP Phone   Vineet Bhojnagarwala
• [asterisk-users] 2 Asterisk Boxes, Single Voicemail   Greg Blakely
• [asterisk-users] 2 Asterisk Boxes, Single Voicemail   Greg Blakely
• [asterisk-users] Ongoing calls interface   David de Boer
• [asterisk-users] transcoding with TC400P   Katerina Borin
• [asterisk-users] CDR / billsec / originate / local chan   Sean Brady
• [asterisk-users] 2 Asterisk Boxes, Single Voicemail   Sean Brady
• [asterisk-users] Virtual machine timing (KVM)   Sean Brady
• [asterisk-users] Virtual machine timing (KVM)   Sean Brady
• [asterisk-users] audio glitches in conference   Sean Brady
• [asterisk-users] Help with iax.conf {tesco|freshtel} 1.6   Brian
• [asterisk-users] ? chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received   Brian
• [asterisk-users] Nat Issue - is this Draytek || Asterisk?   Brian
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Brian
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Brian
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Brian
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Brian
• [asterisk-users] How to run a remote PHP script and still have access to audio stream?   Brian
• [asterisk-users] IP Phone recommendation   Brian
• [asterisk-users] Asterisk -> SIP-ROUTER -> Internet = no audio   Brian
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Brian
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Brian
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Brian
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Brian
• [asterisk-users] PAP2   Brian
• [asterisk-users] [SPAM:9.0] extension not found   Brian
• [asterisk-users] [SPAM:9.0] extension not found   Brian
• [asterisk-users] Capture   Brian
• [asterisk-users] call transfer   Brian
• [asterisk-users] Redirect call based on CLI???   Brian
• [asterisk-users] Redirect call based on CLI???   Brian
• [asterisk-users] Redirect call based on CLI???   Brian
• [asterisk-users] Help for MOH - sounding scratchy/static on hold   Jeff Brower
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Jeff Brower
• [asterisk-users] Muted calls occasionally dropping after 30 seconds   Jeff Brower
• [asterisk-users] signal problem   Jeff Brower
• [asterisk-users] audio glitches in conference   Jeff Brower
• [asterisk-users] audio glitches in conference   Jeff Brower
• [asterisk-users] audio glitches in conference   Jeff Brower
• [asterisk-users] High codec translation times on x64   Christopher Brown
• [asterisk-users] Sending a hook flash to a DAHDI channel   Stephen Brown
• [asterisk-users] Sending a hook flash to a DAHDI channel   Stephen Brown
• [asterisk-users] IP Phone recommendation   Robert Broyles
• [asterisk-users] IP Phone recommendation   Pascal Bruno
• [asterisk-users] Asterisk going down   Josiah Bryan
• [asterisk-users] How to run a remote PHP script and still have access to audio stream?   Leo Burd
• [asterisk-users] How to run a remote PHP script and still have access to audio stream?   Leo Burd
• [asterisk-users] How to run a remote PHP script and still have access to audio stream?   Leo Burd
• [asterisk-users] beroNet BN4S0e PCI Express ISDN Card with chan_dahdi   Laurent CARON
• [asterisk-users] Sending back the BYE code gotten on second leg   CDR
• [asterisk-users] Calls per second limit in manager   CDR
• [asterisk-users] OT- Using TR-069   Leonja Cerebro
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   Leonja Cerebro
• [asterisk-users] How can we pickup a call that is not going to a real extension?   Eric Chamberlain
• [asterisk-users] sip realtime md5secret   Carlos Chavez
• [asterisk-users] Aastra 50-limit blf   Carlos Chavez
• [asterisk-users] Callerid problem in 1.6.2.2   Carlos Chavez
• [asterisk-users] 4 PCIe cards in one asterisk server   Carlos Chavez
• [asterisk-users] GSM Gateway   Chris Childress
• [asterisk-users] Beginners Guide to setting up a Call Centre   Peter Childs
• [asterisk-users] SIP Configuration files for Cisco 7905G FW 3-08-12   Soren Christensen
• [asterisk-users] moving a bridged call to a conference room   Dr. Michael J. Chudobiak
• [asterisk-users] Smallest possible Asterisk VM   Frank Church
• [asterisk-users] Smallest possible Asterisk VM   Frank Church
• [asterisk-users] Do the Linksys Sipura series have a known problem with Asterisk?   Frank Church
• [asterisk-users] billsec is set to duration if call is not answered   Frank Church
• [asterisk-users] billsec is set to duration if call is not answered   Frank Church
• [asterisk-users] Asterisk in Active/Active mode   Ricardo Coelho
• [asterisk-users] Asterisk cluster in Active/Active mode   Ricardo Coelho
• [asterisk-users] Astribank problem   Tzafrir Cohen
• [asterisk-users] connect problem unless when verbose   Tzafrir Cohen
• [asterisk-users] Smallest possible Asterisk VM   Tzafrir Cohen
• [asterisk-users] Astribank problem   Tzafrir Cohen
• [asterisk-users] calling into server with cell and originate a call   Tzafrir Cohen
• [asterisk-users] 1.6.2.1: DTMF trouble with PSTN   Tzafrir Cohen
• [asterisk-users] Dial script   Tzafrir Cohen
• [asterisk-users] OpenVPN on phones?   Tzafrir Cohen
• [asterisk-users] Not able to compile asterisk, zaptel, libpri in /usr/src   Tzafrir Cohen
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Tzafrir Cohen
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Tzafrir Cohen
• [asterisk-users] High codec translation times on x64   Tzafrir Cohen
• [asterisk-users] IVR Demo / Create file / Move file / Demo all   Tzafrir Cohen
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Tzafrir Cohen
• [asterisk-users] ways of initiating a call   Tzafrir Cohen
• [asterisk-users] ways of initiating a call   Tzafrir Cohen
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Tzafrir Cohen
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Tzafrir Cohen
• [asterisk-users] forward incomming line to modem   Tzafrir Cohen
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Tzafrir Cohen
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Tzafrir Cohen
• [asterisk-users] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)   Tzafrir Cohen
• [asterisk-users] Security Logging   Tzafrir Cohen
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Tzafrir Cohen
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Tzafrir Cohen
• [asterisk-users] app_dial.c: Unable to create channel oftype 'Zap' (cause 34 - Circuit/channel congestion)   Tzafrir Cohen
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Tzafrir Cohen
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Tzafrir Cohen
• [asterisk-users] [SPAM:9.0] extension not found   Tzafrir Cohen
• [asterisk-users] how to create voicemail   Tzafrir Cohen
• [asterisk-users] Important security alert: update your dialplans now!   Tzafrir Cohen
• [asterisk-users] Important security alert: update your dialplans now!   Tzafrir Cohen
• [asterisk-users] how to have disconnect signals enabled in line   Tzafrir Cohen
• [asterisk-users] how to have disconnect signals enabled in line   Tzafrir Cohen
• [asterisk-users] Important security alert: update your dialplans now!   Tzafrir Cohen
• [asterisk-users] Important security alert: update your dialplans now!   Tzafrir Cohen
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Tzafrir Cohen
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Tzafrir Cohen
• [asterisk-users] splitting sip.conf to two files   Tzafrir Cohen
• [asterisk-users] Sending a hook flash to a DAHDI channel   Tzafrir Cohen
• [asterisk-users] Virtual machine timing (KVM)   Tzafrir Cohen
• [asterisk-users] HFC-S card   Tzafrir Cohen
• [asterisk-users] HFC-S card   Tzafrir Cohen
• [asterisk-users] Running safe_asterisk   Tzafrir Cohen
• [asterisk-users] Multiple instances of Asterisk on the same host...   Tzafrir Cohen
• [asterisk-users] Multiple instances of Asterisk on the same host...   Tzafrir Cohen
• [asterisk-users] HFC-S card   Tzafrir Cohen
• [asterisk-users] IAX devices not registering after upgrade to   Tzafrir Cohen
• [asterisk-users] Problems installing dahdi : kernel sources   Tzafrir Cohen
• [asterisk-users] init.d error when installing trunk   Nic Colledge
• [asterisk-users] Know what would be killer?   Dean Collins
• [asterisk-users] Losing local SIP phones when internet goes down?   Dave Cotton
• [asterisk-users] Losing local SIP phones when internet goes down?   Dave Cotton
• [asterisk-users] Losing local SIP phones when internet goes down?   Dave Cotton
• [asterisk-users] how to allow some extenstions to call outside and some extensions cant call outside   Gergo Csibra
• [asterisk-users] call transfer   Gergo Csibra
• [asterisk-users] Static IP   Gergo Csibra
• [asterisk-users] subject: 1.4 vs 1.6   Gergo Csibra
• [asterisk-users] OpenVPN on phones?   Ken D'Ambrosio
• [asterisk-users] OpenVPN on phones?   Ken D'Ambrosio
• [asterisk-users] OpenVPN/SNOM 820: a review.   Ken D'Ambrosio
• [asterisk-users] BRI vs. PRI?   Ken D'Ambrosio
• [asterisk-users] OpenVPN/SNOM 820: a review.   Ken D'Ambrosio
• [asterisk-users] Not able to receive fax   Deepesh D
• [asterisk-users] Not able to receive fax   Deepesh D
• [asterisk-users] T.38 with reinvite   Deepesh D
• [asterisk-users] "Unexpected message received" when receiving Fax   Deepesh D
• [asterisk-users] wellgate 3804A with frying   Martin D
• [asterisk-users] conferencing without DAHDI   Klaus Darilion
• [asterisk-users] conferencing without DAHDI   Klaus Darilion
• [asterisk-users] conferencing without DAHDI   Klaus Darilion
• [asterisk-users] conferencing without DAHDI   Klaus Darilion
• [asterisk-users] conferencing without DAHDI   Klaus Darilion
• [asterisk-users] SIP RTP ports not released when channel is hung up   Klaus Darilion
• [asterisk-users] SIP RTP ports not released when channel is hung up   Klaus Darilion
• [asterisk-users] SIP RTP ports not released when channel is hung up   Klaus Darilion
• [asterisk-users] Use a BLF for monitoring   Matt Darnell
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Stephen Davies
• [asterisk-users] SIP tunnel   Stephen Davies
• [asterisk-users] Asterisk 1.2.37 + BLF + ParkedCalls + SPA962   Steve Davies
• [asterisk-users] chan_sip.c: Disconnecting call 'SIP/302-b720dd78' for lack of RTP activity in 301 seconds   Steve Davies
• [asterisk-users] Polycom VVX1500 video working yet?   Steve Davies
• [asterisk-users] Polycom VVX1500 video working yet?   Steve Davies
• [asterisk-users] Polycom VVX1500 video working yet?   Steve Davies
• [asterisk-users] Caller ID question   Steve Davies
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Steve Davies
• [asterisk-users] Premicell solutions?   Steve Davies
• [asterisk-users] Pri HDLC aborts and choppy audio when dialling into pri, caused by BIOS option ["CPU enhanced halt" c1e]   Alec Davis
• [asterisk-users] ISDN users: 1.6.x users, I need some testing done please, regarding Overlap Receiving   Alec Davis
• [asterisk-users] Asterisk going down   Danny Dias
• [asterisk-users] Losing local SIP phones when internet goes down?   Danny Dias
• [asterisk-users] Asterisk going down   Danny Dias
• [asterisk-users] Asterisk going dow   Danny Dias
• [asterisk-users] Asterisk going down (Josiah Bryan)   Danny Dias
• [asterisk-users] asterisk-users Digest, Vol 67, Issue 20 Re: Asterisk going down   Danny Dias
• [asterisk-users] chan_sip.c: Disconnecting call 'SIP/302-b720dd78' for lack of RTP activity in 301 seconds   Danny Dias
• [asterisk-users] Asterisk Crashs due to some Sip messages   Danny Dias
• [asterisk-users] Open source or low-budget recommendation for call-center software   Juan David Diaz
• [asterisk-users] Problems in Asterisk Real Time (Urgent help )   Juan David Diaz
• [asterisk-users] add Reason header on hangup   Jim Dickenson
• [asterisk-users] AMI Originate differences between 1.4 and 1.6.1   Jim Dickenson
• [asterisk-users] One-Way Audio after Hold   Mike Diehl
• [asterisk-users] Problem w/ MoH   Mike Diehl
• [asterisk-users] Problem w/ MoH   Mike Diehl
• [asterisk-users] Web operator/softphone with integration features   Carlo Dimaggio
• [asterisk-users] Web operator/softphone with integration features   Carlo Dimaggio
• [asterisk-users] Asterisk 1.6.2 ?   Ben Dinnerville
• [asterisk-users] CDR / billsec / originate / local chan   Ben Dinnerville
• [asterisk-users] Running a script after Dial() ?   Ben Dinnerville
• [asterisk-users] Running a script after Dial() ?   Ben Dinnerville
• [asterisk-users] Running a script after Dial() ?   Ben Dinnerville
• [asterisk-users] CDR / billsec / originate / local chan   Ben Dinnerville
• [asterisk-users] originate, local channel and failure extension   Ben Dinnerville
• [asterisk-users] How to run a remote PHP script and still have access to audio stream?   Ben Dinnerville
• [asterisk-users] sip to dahdi and billsec   Uros Djokic
• [asterisk-users] TOS bits, DSCP, Asterisk & Polycom   Doug
• [asterisk-users] TOS bits, DSCP, Asterisk & Polycom   Doug
• [asterisk-users] Dial script   Michelle Dupuis
• [asterisk-users] Robotic sound sometimes   Michelle Dupuis
• [asterisk-users] Robotic sound sometimes   Michelle Dupuis
• [asterisk-users] Access to header field: event   Michelle Dupuis
• [asterisk-users] Access to header field: event   Michelle Dupuis
• [asterisk-users] Access to header field: event   Michelle Dupuis
• [asterisk-users] Access to header field: event   Michelle Dupuis
• [asterisk-users] directrtp with SIP + H.323   Michelle Dupuis
• [asterisk-users] Which H.323 to use in Ast 1.6   Michelle Dupuis
• [asterisk-users] Which H.323 to use in Ast 1.6   Michelle Dupuis
• [asterisk-users] Which H.323 to use in Ast 1.6   Michelle Dupuis
• [asterisk-users] Codec coversion   Steve Edwards
• [asterisk-users] codec conversion   Steve Edwards
• [asterisk-users] MATH   Steve Edwards
• [asterisk-users] Running a script after Dial() ?   Steve Edwards
• [asterisk-users] Running a script after Dial() ?   Steve Edwards
• [asterisk-users] Running a script after Dial() ?   Steve Edwards
• [asterisk-users] Running a script after Dial() ?   Steve Edwards
• [asterisk-users] Dial script   Steve Edwards
• [asterisk-users] Dial script   Steve Edwards
• [asterisk-users] Dial script   Steve Edwards
• [asterisk-users] Dial script   Steve Edwards
• [asterisk-users] Dial script   Steve Edwards
• [asterisk-users] Asterisk going down   Steve Edwards
• [asterisk-users] syntax   Steve Edwards
• [asterisk-users] IVR Demo / Create file / Move file / Demo all   Steve Edwards
• [asterisk-users] IVR Demo / Create file / Move file / Demo all   Steve Edwards
• [asterisk-users] Security Logging   Steve Edwards
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Steve Edwards
• [asterisk-users] How to avoid AGI script is canceled if callerHangUp   Steve Edwards
• [asterisk-users] Wierdness in AGI file   Steve Edwards
• [asterisk-users] Asterisk Cepstral TTS   Steve Edwards
• [asterisk-users] Asterisk Cepstral TTS   Steve Edwards
• [asterisk-users] Asterisk Cepstral TTS   Steve Edwards
• [asterisk-users] Robotic sound sometimes   Steve Edwards
• [asterisk-users] Important security alert: update your dialplans now!   Steve Edwards
• [asterisk-users] Maximum call handling capacity on single server   Steve Edwards
• [asterisk-users] Asterisk listens on all NICs   Steve Edwards
• [asterisk-users] some newbie questions about gcc   Steve Edwards
• [asterisk-users] Static IP   Steve Edwards
• [asterisk-users] Static IP   Steve Edwards
• [asterisk-users] Static IP   Steve Edwards
• [asterisk-users] Load balance outgoing calls   Steve Edwards
• [asterisk-users] Load balance outgoing calls   Steve Edwards
• [asterisk-users] Load balance outgoing calls   Steve Edwards
• [asterisk-users] : PSTN calls   Steve Edwards
• [asterisk-users] : PSTN calls   Steve Edwards
• [asterisk-users] : PSTN calls   Steve Edwards
• [asterisk-users] : PSTN calls   Steve Edwards
• [asterisk-users] Asterisk AUTHENTICATE Command   Steve Edwards
• [asterisk-users] AUTHENTICATE Command customized prompts - Work around   Steve Edwards
• [asterisk-users] Hardware   Steve Edwards
• [asterisk-users] Can an agent Login to a queue and be paused   Lenz Emilitri
• [asterisk-users] Can an agent Login to a queue and be paused   Lenz Emilitri
• [asterisk-users] Important security alert: update your dialplans now!   Lenz Emilitri
• [asterisk-users] Important security alert: update your dialplans now!   Lenz Emilitri
• [asterisk-users] Important security alert: update your dialplans now!   Lenz Emilitri
• [asterisk-users] Important security alert: update your dialplans now!   Lenz Emilitri
• [asterisk-users] Looping over AstDB   Lenz Emilitri
• [asterisk-users] Dial script   C F
• [asterisk-users] Dial script   C F
• [asterisk-users] Dial script   C F
• [asterisk-users] Dial script   C F
• [asterisk-users] large scale paging   C F
• [asterisk-users] Important security alert: update your dialplans now!   C F
• [asterisk-users] Important security alert: update your dialplans now!   C F
• [asterisk-users] Important security alert: update your dialplans now!   C F
• [asterisk-users] Important security alert: update your dialplans now!   C F
• [asterisk-users] how asterisk knows which context forward the call to?   C F
• [asterisk-users] How to tell if asterisk is handling media or not?   C F
• [asterisk-users] hi   C F
• [asterisk-users] Morse Code   F6HQZ
• [asterisk-users] Conference Calling   Faheem
• [asterisk-users] Zaptel/DAHDI error's on PRI   Sascha Ferley
• [asterisk-users] Realtime extensions   Bruce Ferrell
• [asterisk-users] Problems with SIP realtime   Bruce Ferrell
• [asterisk-users] Dial script   Karl Fife
• [asterisk-users] Dial script   Karl Fife
• [asterisk-users] Dial script   Karl Fife
• [asterisk-users] Website Down ?   Karl Fife
• [asterisk-users] Dial script   Karl Fife
• [asterisk-users] CODECS: Best practice question: Avoid transcode when calling out?   Karl Fife
• [asterisk-users] NVFaxDetect   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Kevin P. Fleming
• [asterisk-users] Asterisk core sounds in English by June Wallack   Kevin P. Fleming
• [asterisk-users] OpenVPN on phones?   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Kevin P. Fleming
• [asterisk-users] Website Down ?   Kevin P. Fleming
• [asterisk-users] Still on spandsp/app_fax and T.38   Kevin P. Fleming
• [asterisk-users] Still on spandsp/app_fax and T.38   Kevin P. Fleming
• [asterisk-users] problems with creating a call   Kevin P. Fleming
• [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals   Kevin P. Fleming
• [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals   Kevin P. Fleming
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Kevin P. Fleming
• [asterisk-users] Still on spandsp/app_fax and T.38   Kevin P. Fleming
• [asterisk-users] Still on spandsp/app_fax and T.38   Kevin P. Fleming
• [asterisk-users] Access to header field: event   Kevin P. Fleming
• [asterisk-users] Access to header field: event   Kevin P. Fleming
• [asterisk-users] Virtual machine timing (KVM)   Kevin P. Fleming
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   Kevin P. Fleming
• [asterisk-users] directrtp with SIP + H.323   Kevin P. Fleming
• [asterisk-users] IAX devices not registering after upgrade to   Kevin P. Fleming
• [asterisk-users] Re-INVITE on BYE   Kevin P. Fleming
• [asterisk-users] Re-INVITE on BYE   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Vinícius Fontes
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Vinícius Fontes
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Vinícius Fontes
• [asterisk-users] Astribank problem   Vinícius Fontes
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Vinícius Fontes
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Vinícius Fontes
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] Losing local SIP phones when internet goes down?   Vinícius Fontes
• [asterisk-users] Losing local SIP phones when internet goes down?   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] Losing local SIP phones when internet goes down?   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] TOS bits, DSCP, Asterisk & Polycom   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] Important security alert: update your?dialplans now!   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] Still on spandsp/app_fax and T.38   Vinícius Fontes
• [asterisk-users] transcoding with TC400P   Vinícius Fontes
• [asterisk-users] MeetMe() and dahdi_dummy on an embedded system   Vinícius Fontes
• [asterisk-users] MeetMe() and dahdi_dummy on an embedded system   Vinícius Fontes
• [asterisk-users] MeetMe() and dahdi_dummy on an embedded system   Vinícius Fontes
• [asterisk-users] MeetMe() and dahdi_dummy on an embedded system   Vinícius Fontes
• [asterisk-users] Astribank problem   Dovey Forman
• [asterisk-users] Astribank problem   Dovey Forman
• [asterisk-users] Asterisk answers inbound call during ringing   Dovey Forman
• [asterisk-users] uri tel: instead of sip:accepted ?   BERGANZ Francois
• [asterisk-users] qsigchannelmapping parameter   Christoph Fuerstaller
• [asterisk-users] NVFaxDetect   Jared Geiger
• [asterisk-users] Virtual machine timing (KVM)   Jared Geiger
• [asterisk-users] string length in dialplan   Jerry Geis
• [asterisk-users] asterisk dahdi fax problem   Peter Gelencser
• [asterisk-users] outgoing callerid problem   Peter Gelencser
• [asterisk-users] Call Pickup with 1.6.2.1 and Snom   RABOUIN Geoffroy
• [asterisk-users] 500 Internal Server Error on Cisco 7940 after INVITE   David Gibbons
• [asterisk-users] OpenVPN on phones?   David Gibbons
• [asterisk-users] Morse Code   David Gibbons
• [asterisk-users] sip to dahdi and billsec   Lyle Giese
• [asterisk-users] Security Logging   Lyle Giese
• [asterisk-users] Security Logging   Lyle Giese
• [asterisk-users] X-Lite won't register   Girard, Jeffrey COL MIL USA
• [asterisk-users] Polycom phone DND state   Jimmy Godbout
• [asterisk-users] Problems with Linksys IP Phone SPA 942   Jimmy Godbout
• [asterisk-users] Polycom VVX1500 video working yet?   Michael Graves
• [asterisk-users] Can an agent Login to a queue and be paused   Robert Grignon
• [asterisk-users] Can an agent Login to a queue and be paused   Robert Grignon
• [asterisk-users] Can an agent Login to a queue and be paused   Robert Grignon
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Robert Grignon
• [asterisk-users] Handling Segmentation Faults / Crashes   Robert Grignon
• [asterisk-users] Asterisk Cepstral TTS   Jeff Grollo
• [asterisk-users] SIP tunnel   Andrew Hakman
• [asterisk-users] Losing local SIP phones when internet goes down?   Dana Harding
• [asterisk-users] Smallest possible Asterisk VM   Darrick Hartman
• [asterisk-users] Problems with recordings of call using Monitor   Peter den Hartog
• [asterisk-users] Problems with recordings of call using Monitor   Peter den Hartog
• [asterisk-users] GSM Gateway   Peter den Hartog
• [asterisk-users] problems with creating a call   Peter den Hartog
• [asterisk-users] problems with creating a call   Peter den Hartog
• [asterisk-users] OpenVPN/SNOM 820: a review.   Paul Hayes
• [asterisk-users] Wrong MOH   Oliver Hehlert
• [asterisk-users] Smallest possible Asterisk VM   Gordon Henderson
• [asterisk-users] Intel Atom based Asterisk server?   Gordon Henderson
• [asterisk-users] GSM Gateway   Gordon Henderson
• [asterisk-users] IP Phone recommendation   Gordon Henderson
• [asterisk-users] IP Phone recommendation   Gordon Henderson
• [asterisk-users] IP Phone recommendation   Gordon Henderson
• [asterisk-users] Multiple instances of Asterisk on the same host...   Gordon Henderson
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Gordon Henderson
• [asterisk-users] Multiple instances of Asterisk on the same host...   Gordon Henderson
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Gordon Henderson
• [asterisk-users] Multiple instances of Asterisk on the same host...   Gordon Henderson
• [asterisk-users] Multiple instances of Asterisk on the same host...   Gordon Henderson
• [asterisk-users] MeetMe() and dahdi_dummy on an embedded system   Gordon Henderson
• [asterisk-users] MeetMe() and dahdi_dummy on an embedded system   Gordon Henderson
• [asterisk-users] How to tell if asterisk is handling media or not?   Gordon Henderson
• [asterisk-users] MeetMe() and dahdi_dummy on an embedded system   Gordon Henderson
• [asterisk-users] Fun with virtual asterisks ...   Gordon Henderson
• [asterisk-users] Fun with virtual asterisks ...   Gordon Henderson
• [asterisk-users] Fun with virtual asterisks ...   Gordon Henderson
• [asterisk-users] Premicell solutions?   Gordon Henderson
• [asterisk-users] Server response time   Gordon Henderson
• [asterisk-users] Premicell solutions?   Gordon Henderson
• [asterisk-users] A2Billing and other prepaid Billing like ASTCC, who is better?   Gavin Henry
• [asterisk-users] Intermittent DAHDI issue with a PRI line causing asterisk to crash!   James Hill
• [asterisk-users] Dial script   Rob Hillis
• [asterisk-users] Important security alert: update your dialplans now!   Rob Hillis
• [asterisk-users] How does holdtime get calculated for queues   Rob Hillis
• [asterisk-users] rawplayer in asterisk 1.0.0   Rob Hillis
• [asterisk-users] Voicemail IMAP storage enhancement   Hoggins!
• [asterisk-users] calling into server with cell and originate a call   Steve Howes
• [asterisk-users] 2 Asterisk Boxes, Single Voicemail   Steve Howes
• [asterisk-users] Not able to compile asterisk, zaptel, libpri in /usr/src   Steve Howes
• [asterisk-users] IP Phone recommendation   Steve Howes
• [asterisk-users] rawplayer in asterisk 1.0.0   Steve Howes
• [asterisk-users] how to remove asterisk from this string X-Asterisk-HangupCauseCode   Steve Howes
• [asterisk-users] Sending back the BYE code gotten on second leg   Steve Howes
• [asterisk-users] Redirect question   Steve Howes
• [asterisk-users] Do i need install Dahdi or libpri ?   Steve Howes
• [asterisk-users] disable comfort noise   Mark Hulber
• [asterisk-users] Losing local SIP phones when internet goes down?   Mark Hulber
• [asterisk-users] Redirect call based on CLI???   Mark Hulber
• [asterisk-users] SIP RTP ports not released when channel is hung up   Marcus Hunger
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Ken Leland III
• [asterisk-users] Smallest possible Asterisk VM   Michael Iedema
• [asterisk-users] mysterious rippled sound with IAX   Giorgio Incantalupo
• [asterisk-users] how asterisk knows which context forward the call to?   Ioan Indreias
• [asterisk-users] Intel Atom based Asterisk server?   Ira
• [asterisk-users] Intel Atom based Asterisk server?   Ira
• [asterisk-users] {top|bottom|interleaved} posting, once again   Ira
• [asterisk-users] IP Phone recommendation   Ira
• [asterisk-users] : PSTN calls   Ira
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   JT
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   JT
• [asterisk-users] Help with Dictate app   Jayesh Jayan
• [asterisk-users] Help with Dictate app   Jayesh Jayan
• [asterisk-users] Help with Dictate app   Jayesh Jayan
• [asterisk-users] Help with Dictate app   Jayesh Jayan
• [asterisk-users] Lua status in asterisk.   Tommy Botten Jensen
• [asterisk-users] Not able to receive fax   Tommy Botten Jensen
• [asterisk-users] ways of initiating a call   Tommy Botten Jensen
• [asterisk-users] IP Phone recommendation   Tommy Botten Jensen
• [asterisk-users] Important security alert: update your dialplans now!   Tommy Botten Jensen
• [asterisk-users] directrtp with SIP + H.323   Tommy Botten Jensen
• [asterisk-users] Calls per second limit in manager   Tommy Botten Jensen
• [asterisk-users] Calls per second limit in manager   Tommy Botten Jensen
• [asterisk-users] Problems in Asterisk Real Time (Urgent help )   Tommy Botten Jensen
• [asterisk-users] identify the costumer   Tommy Botten Jensen
• [asterisk-users] Running a script after Dial() ?   Per Jessen
• [asterisk-users] Running a script after Dial() ?   Per Jessen
• [asterisk-users] Running a script after Dial() ?   Per Jessen
• [asterisk-users] Running a script after Dial() ?   Per Jessen
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Per Jessen
• [asterisk-users] HFC-S card   Per Jessen
• [asterisk-users] HFC-S card   Per Jessen
• [asterisk-users] Running safe_asterisk   Per Jessen
• [asterisk-users] Running safe_asterisk   Per Jessen
• [asterisk-users] Running safe_asterisk   Per Jessen
• [asterisk-users] uri tel: instead of sip:accepted ?   Olle E. Johansson
• [asterisk-users] uri tel: instead of sip:accepted ?   Olle E. Johansson
• [asterisk-users] OpenVPN on phones?   Olle E. Johansson
• [asterisk-users] OpenVPN on phones?   Olle E. Johansson
• [asterisk-users] Losing local SIP phones when internet goes down?   Olle E. Johansson
• [asterisk-users] Losing local SIP phones when internet goes down?   Olle E. Johansson
• [asterisk-users] Losing local SIP phones when internet goes down?   Olle E. Johansson
• [asterisk-users] OpenVPN on phones?   Olle E. Johansson
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Olle E. Johansson
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Olle E. Johansson
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Olle E. Johansson
• [asterisk-users] conferencing without DAHDI   Olle E. Johansson
• [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals   Olle E. Johansson
• [asterisk-users] SIP RTP ports not released when channel is hung up   Olle E. Johansson
• [asterisk-users] SIP RTP ports not released when channel is hung up   Olle E. Johansson
• [asterisk-users] 1.6.x SIP allow incoming calls based on from ip address?   Olle E. Johansson
• [asterisk-users] Important security alert: update your dialplans now!   Olle E. Johansson
• [asterisk-users] Important security alert: update your dialplans now!   Olle E. Johansson
• [asterisk-users] Important security alert: update your dialplans now!   Olle E. Johansson
• [asterisk-users] Important security alert: update your dialplans now!   Olle E. Johansson
• [asterisk-users] Important security alert: update your dialplans now!   Olle E. Johansson
• [asterisk-users] Maximum call handling capacity on single server   Olle E. Johansson
• [asterisk-users] video voicemail   Olle E. Johansson
• [asterisk-users] OT- Using TR-069   Olle E. Johansson
• [asterisk-users] Important security alert: update your dialplans now!   Olle E. Johansson
• [asterisk-users] Empty SIP Packet   Olle E. Johansson
• [asterisk-users] Important security alert: update your dialplans now!   Olle E. Johansson
• [asterisk-users] how to remove asterisk from this string X-Asterisk-HangupCauseCode   Olle E. Johansson
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Olle E. Johansson
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Olle E. Johansson
• [asterisk-users] chan_local and Originate   Olle E. Johansson
• [asterisk-users] chan_local and Originate   Olle E. Johansson
• [asterisk-users] sip.conf - sort order, does it matter   Olle E. Johansson
• [asterisk-users] Access to header field: event   Olle E. Johansson
• [asterisk-users] sip.conf - sort order, does it matter   Olle E. Johansson
• [asterisk-users] sip.conf - sort order, does it matter   Olle E. Johansson
• [asterisk-users] Dial Plan configuration in asterisk   Olle E. Johansson
• [asterisk-users] add Reason header on hangup   Olle E. Johansson
• [asterisk-users] Audio to remote AGI server   Olle E. Johansson
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   Olle E. Johansson
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   Olle E. Johansson
• [asterisk-users] Calls per second limit in manager   Olle E. Johansson
• [asterisk-users] directrtp with SIP + H.323   Olle E. Johansson
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   Olle E. Johansson
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   Olle E. Johansson
• [asterisk-users] PAP2   Tim Johnson
• [asterisk-users] NVFaxDetect   Joseph
• [asterisk-users] NVFaxDetect   Joseph
• [asterisk-users] Audiocodes MP-114 MWI Stutter Tone   Joseph
• [asterisk-users] Losing local SIP phones when internet goes down?   Joseph
• [asterisk-users] Losing local SIP phones when internet goes down?   Joseph
• [asterisk-users] Losing local SIP phones when internet goes down?   Joseph
• [asterisk-users] Domain Authentication - Caller ID Failed to authenticate   Joseph
• [asterisk-users] insecure=invite - not working for different dial plan   Joseph
• [asterisk-users] insecure=invite - not working for different dial plan   Joseph
• [asterisk-users] call is not going to wrong "context"   Joseph
• [asterisk-users] call is not going to wrong "context"   Joseph
• [asterisk-users] insecure=invite - not working for different dial plan   Joseph
• [asterisk-users] sip.conf - sort order, does it matter   Joseph
• [asterisk-users] sending call to correct context   Joseph
• [asterisk-users] how asterisk knows which context forward the call to?   Joseph
• [asterisk-users] how asterisk knows which context forward the call to?   Joseph
• [asterisk-users] how asterisk knows which context forward the call to?   Joseph
• [asterisk-users] how asterisk knows which context forward the call to?   Joseph
• [asterisk-users] sip.conf - sort order, does it matter   Joseph
• [asterisk-users] how asterisk knows which context forward the call to?   Joseph
• [asterisk-users] splitting sip.conf to two files   Joseph
• [asterisk-users] splitting sip.conf to two files   Joseph
• [asterisk-users] splitting sip.conf to two files   Joseph
• [asterisk-users] sip.conf - sort order, does it matter   Joseph
• [asterisk-users] Get Talk Time   Dan Journo
• [asterisk-users] IP Phone recommendation   Dan Journo
• [asterisk-users] Asterisk Redundancy   Dan Journo
• [asterisk-users] Asterisk Redundancy   Dan Journo
• [asterisk-users] Trouble with externalIVR socket connection   Chris Kairalla
• [asterisk-users] Morse Code   Chris Kairalla
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   Kirill 'Big K' Katsnelson
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   Kirill 'Big K' Katsnelson
• [asterisk-users] pickup with gxp2000 does not work..   Oguzhan Kayhan
• [asterisk-users] [OT] Snom M3s   Philipp Kempgen
• [asterisk-users] large scale paging   Philipp Kempgen
• [asterisk-users] {top|bottom|interleaved} posting, once again   Philipp Kempgen
• [asterisk-users] GSM Gateway   Steve Kennedy
• [asterisk-users] Stuck logger rotation   Richard Kenner
• [asterisk-users] Use a BLF for monitoring   Richard Kenner
• [asterisk-users] Use a BLF for monitoring   Richard Kenner
• [asterisk-users] Asterisk core sounds in English by June Wallack   Richard Kenner
• [asterisk-users] IP Phone recommendation   Richard Kenner
• [asterisk-users] Sending "Progress" during dialing   Richard Kenner
• [asterisk-users] Wierdness in AGI file   Richard Kenner
• [asterisk-users] Asterisk Cepstral TTS   Richard Kenner
• [asterisk-users] sip realtime md5secret   Mindaugas Kezys
• [asterisk-users] A2Billing and other prepaid Billing like ASTCC, who is better?   Mindaugas Kezys
• [asterisk-users] Product offerings from DIDforSale   Neha Khandelwal
• [asterisk-users] Use of "603 Declined"   Kristian Kielhofner
• [asterisk-users] large scale paging   Kristian Kielhofner
• [asterisk-users] directrtp with SIP + H.323   Kristian Kielhofner
• [asterisk-users] Dial multiple extensions and know who picks up call   Kyle Kienapfel
• [asterisk-users] uri tel: instead of sip:accepted ?   Kyle Kienapfel
• [asterisk-users] Know what would be killer?   Kyle Kienapfel
• [asterisk-users] CONNECTEDLINE   Kyle Kienapfel
• [asterisk-users] Asterisk going down   Kyle Kienapfel
• [asterisk-users] Asterisk how install speex support   Kyle Kienapfel
• [asterisk-users] SIP tunnel   Kyle Kienapfel
• [asterisk-users] how to allow some extensions to make call outside and some extensions cant call outside   Kyle Kienapfel
• [asterisk-users] how to allow some extensions to make call outside and some extensions cant call outside   Kyle Kienapfel
• [asterisk-users] Important security alert: update your dialplans now!   Kyle Kienapfel
• [asterisk-users] Product offerings from DIDforSale   Kyle Kienapfel
• [asterisk-users] Slightly OT: Has SILK codec gotten anywhere?   Kyle Kienapfel
• [asterisk-users] Redirect call based on CLI???   Kyle Kienapfel
• [asterisk-users] Polycom VVX1500 video working yet?   Jordan Kirby
• [asterisk-users] OpenVPN on phones?   Philipp von Klitzing
• [asterisk-users] large scale paging   Philipp von Klitzing
• [asterisk-users] IP Phone recommendation   Philipp von Klitzing
• [asterisk-users] IP Phone recommendation   Philipp von Klitzing
• [asterisk-users] SIP RTP ports not released when channel is hung up   Philipp von Klitzing
• [asterisk-users] Maximum call handling capacity on single server   Philipp von Klitzing
• [asterisk-users] CODECS: Best practice question: Avoid transcode when calling out?   Philipp von Klitzing
• [asterisk-users] Important security alert: update your dialplans now!   Philipp von Klitzing
• [asterisk-users] Asterisk listens on all NICs   Philipp von Klitzing
• [asterisk-users] Asterisk listens on all NICs   Philipp von Klitzing
• [asterisk-users] Asterisk t38modem Fax gateway evaluation   Philipp von Klitzing
• [asterisk-users] directmedia/canreinvite/native bridging question   Philipp von Klitzing
• [asterisk-users] string length in dialplan   Philipp von Klitzing
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Philipp von Klitzing
• [asterisk-users] IAX devices not registering after upgrade to asterisk 1.4.29   Philipp von Klitzing
• [asterisk-users] SIP provider registration attempts   Philipp von Klitzing
• [asterisk-users] Denying call transfer to certain extensions   Philipp von Klitzing
• [asterisk-users] SIP provider registration attempts   Philipp von Klitzing
• [asterisk-users] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences?   Philipp von Klitzing
• [asterisk-users] Asterisk AUTHENTICATE Command   Matthew A Kolberg
• [asterisk-users] AUTHENTICATE Command customized prompts - Work around   Matthew A Kolberg
• [asterisk-users] Virtual machine timing (KVM)   Bruce Komito
• [asterisk-users] Asterisk & Fax   Stelios Koroneos
• [asterisk-users] : PSTN calls   Aditya Kumar
• [asterisk-users] : PSTN calls   Aditya Kumar
• [asterisk-users] : PSTN calls   Aditya Kumar
• [asterisk-users] : PSTN calls   Aditya Kumar
• [asterisk-users] : PSTN calls   Aditya Kumar
• [asterisk-users] Hardware   Aditya Kumar
• [asterisk-users] sip realtime md5secret   Emre Kurnaz
• [asterisk-users] sip realtime md5secret   Emre Kurnaz
• [asterisk-users] strange asterisk behaviour on XEN   Emre Kurnaz
• [asterisk-users] Asterisk 1.6.0.17 PBX with two interfaces does not routes RTP packets - SIP Conf Problem likely   LATEEF, IRFAN (ATTSI)
• [asterisk-users] Asterisk 1.6.0.17 PBX with two interfaces does notroutes RTP packets - SIP Conf Problem likely   LATEEF, IRFAN (ATTSI)
• [asterisk-users] Premicell solutions?   LATEEF, IRFAN (ATTSI)
• [asterisk-users] Asterisk t38modem Fax gateway evaluation   DLeese at LStelcom.com
• [asterisk-users] [SPAM] - Re: Asterisk t38modem Fax gateway evaluation - Email found in subject   DLeese at LStelcom.com
• [asterisk-users] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences?   DLeese at LStelcom.com
• [asterisk-users] codec conversion   Jeff LaCoursiere
• [asterisk-users] asterisk video support and IPTV   Jeff LaCoursiere
• [asterisk-users] Losing local SIP phones when internet goes down?   Jeff LaCoursiere
• [asterisk-users] Losing local SIP phones when internet goes down?   Jeff LaCoursiere
• [asterisk-users] large scale paging   Jeff LaCoursiere
• [asterisk-users] Dial script   Jeff LaCoursiere
• [asterisk-users] test   Jeff LaCoursiere
• [asterisk-users] IP Phone recommendation   Jeff LaCoursiere
• [asterisk-users] video voicemail   Jeff LaCoursiere
• [asterisk-users] video voicemail   Jeff LaCoursiere
• [asterisk-users] splitting sip.conf to two files   Edwin Lam
• [asterisk-users] PRI Problems with 1.6.0.10   James Lamanna
• [asterisk-users] PRI Problems with 1.6.0.10   James Lamanna
• [asterisk-users] Hung channel problem with 1.4.26.2   James Lamanna
• [asterisk-users] Important security alert: update your dialplans now!   Landy Landy
• [asterisk-users] Asterisk listens on all NICs   Landy Landy
• [asterisk-users] Asterisk listens on all NICs   Landy Landy
• [asterisk-users] Intel Atom based Asterisk server?   Andrew Latham
• [asterisk-users] NVFaxDetect   Mariano Lecuona
• [asterisk-users] Can an agent Login to a queue and be paused   Mariano Lecuona
• [asterisk-users] Can an agent Login to a queue and be paused   Mariano Lecuona
• [asterisk-users] app_dial.c: Unable to create channel oftype 'Zap' (cause 34 - Circuit/channel congestion)   Mariano Lecuona
• [asterisk-users] queue.conf - Set(MONITOR_FILENAME=${})   Mariano Lecuona
• [asterisk-users] queue.conf - Set(MONITOR_FILENAME=${})   Mariano Lecuona
• [asterisk-users] Use a BLF for monitoring   Lee, John (Sydney)
• [asterisk-users] Polycom phone DND state   Lee, John (Sydney)
• [asterisk-users] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)   Lee, John (Sydney)
• [asterisk-users] app_dial.c: Unable to create channel of type'Zap'(cause 34 - Circuit/channel congestion)   Lee, John (Sydney)
• [asterisk-users] app_dial.c: Unable to create channel oftype 'Zap' (cause 34 - Circuit/channel congestion)   Lee, John (Sydney)
• [asterisk-users] Deadlock while using MGCP on Asterisk   Adrien Lemoine
• [asterisk-users] Deadlock while using MGCP on Asterisk   Adrien Lemoine
• [asterisk-users] SPA941 WMI not lighting up when natted   Michael Leonetti
• [asterisk-users] Setting up only one caller at a time   Mike A. Leonetti
• [asterisk-users] Setting up only one caller at a time   Mike A. Leonetti
• [asterisk-users] Virtual machine timing (KVM)   Mike A. Leonetti
• [asterisk-users] Virtual machine timing (KVM)   Mike A. Leonetti
• [asterisk-users] Virtual machine timing (KVM)   Mike A. Leonetti
• [asterisk-users] SPA941 WMI not lighting up when natted   Mike A. Leonetti
• [asterisk-users] ast_cdr_setvar: Attempt to set the 'src' read-only variable!   Tilghman Lesher
• [asterisk-users] Running a script after Dial() ?   Tilghman Lesher
• [asterisk-users] Running a script after Dial() ?   Tilghman Lesher
• [asterisk-users] Losing local SIP phones when internet goes down?   Tilghman Lesher
• [asterisk-users] How to avoid AGI script is canceled if callerHangUp   Tilghman Lesher
• [asterisk-users] issues.asterisk.org   Tilghman Lesher
• [asterisk-users] Important security alert: update your dialplans now!   Tilghman Lesher
• [asterisk-users] video voicemail   Tilghman Lesher
• [asterisk-users] video voicemail   Tilghman Lesher
• [asterisk-users] Important security alert: update your?dialplans now!   Tilghman Lesher
• [asterisk-users] Important security alert: update your?dialplans now!   Tilghman Lesher
• [asterisk-users] Users of the SMS application?   Tilghman Lesher
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Tilghman Lesher
• [asterisk-users] chan_local and Originate   Tilghman Lesher
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Tilghman Lesher
• [asterisk-users] Audio to remote AGI server   Tilghman Lesher
• [asterisk-users] Audio to remote AGI server   Tilghman Lesher
• [asterisk-users] Running safe_asterisk   Tilghman Lesher
• [asterisk-users] Macros, GoSub & StackPop   Tilghman Lesher
• [asterisk-users] subject: 1.4 vs 1.6   Tilghman Lesher
• [asterisk-users] OT: Problems with Linksys IP Phone SPA 942   Tilghman Lesher
• [asterisk-users] curl and ssl certificate   Tilghman Lesher
• [asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available   Tilghman Lesher
• [asterisk-users] Asterisk RPM's   Tilghman Lesher
• [asterisk-users] Asterisk RPM's   Tilghman Lesher
• [asterisk-users] Recording Calls   Luz Lopez
• [asterisk-users] Maximum call handling capacity on single server   Amit Patkar | Avhan Technologies Pvt. Ltd.
• [asterisk-users] Manager Logged off   Anahi Ludueña
• [asterisk-users] E71   Scott L. Lykens
• [asterisk-users] SIP tunnel   Scott L. Lykens
• [asterisk-users] chan_local and Originate   Julian Lyndon-Smith
• [asterisk-users] OpenVPN on phones?   Doug Lytle
• [asterisk-users] OpenVPN on phones?   Doug Lytle
• [asterisk-users] OpenVPN on phones?   Doug Lytle
• [asterisk-users] parked calls   Doug Lytle
• [asterisk-users] [SPAM:9.0] extension not found   Doug Lytle
• [asterisk-users] extension not found   Doug Lytle
• [asterisk-users] issues.asterisk.org   Doug Lytle
• [asterisk-users] OpenVPN/SNOM 820: a review.   Doug Lytle
• [asterisk-users] Help with Dictate app   Doug Lytle
• [asterisk-users] Help with Dictate app   Doug Lytle
• [asterisk-users] asterisk sudden restart - 1.4.18.1   Leif Madsen
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Leif Madsen
• [asterisk-users] Important security alert: update your?dialplans now!   Leif Madsen
• [asterisk-users] string length in dialplan   Leif Madsen
• [asterisk-users] asterisk and mysql connection   Ishfaq Malik
• [asterisk-users] Muted calls occasionally dropping after 30 seconds   Ishfaq Malik
• [asterisk-users] IP Phone recommendation   Ishfaq Malik
• [asterisk-users] OpenVPN/SNOM 820: a review.   Ishfaq Malik
• [asterisk-users] TE410P Spans offline/red after power down/restart   Conor McTernan
• [asterisk-users] TE410P Spans offline/red after power down/restart   Conor McTernan
• [asterisk-users] Line DC   Global Meds
• [asterisk-users] Line DC   Global Meds
• [asterisk-users] Line DC   Global Meds
• [asterisk-users] Line DC   Global Meds
• [asterisk-users] Capture   Global Meds
• [asterisk-users] Capture   Global Meds
• [asterisk-users] Asterisk & Fax   Murray Melvin
• [asterisk-users] Redirect question   Bert Mengerink
• [asterisk-users] Redirect question   Bert Mengerink
• [asterisk-users] Losing local SIP phones when internet goes down?   Anthony Messina
• [asterisk-users] Problems with SIP realtime   Juan Miguel
• [asterisk-users] Polycom phone DND state   Mike
• [asterisk-users] Polycom phone DND state   Mike
• [asterisk-users] Polycom phone DND state   Mike
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Mike
• [asterisk-users] IP Phone recommendation   Sebastian Milioto
• [asterisk-users] IP Phone recommendation   Sebastian Milioto
• [asterisk-users] IP Phone recommendation   Sebastian Milioto
• [asterisk-users] Gotoif Question   Barry Miller
• [asterisk-users] Dial script   Barry Miller
• [asterisk-users] agi debug in Asterisk 1.6?   Barry Miller
• [asterisk-users] string length in dialplan   Barry Miller
• [asterisk-users] Open source or low-budget recommendation for call-center software   Apa Minerala
• [asterisk-users] queue with strategy=linear   Louis-David Mitterrand
• [asterisk-users] rawplayer in asterisk 1.0.0   Arjan Kroon | Mobillion
• [asterisk-users] 4 PCIe cards in one asterisk server   Arjan Kroon | Mobillion
• [asterisk-users] 4 PCIe cards in one asterisk server   Arjan Kroon | Mobillion
• [asterisk-users] Important security alert: update your dialplans now!   Miguel Molina
• [asterisk-users] chan_local and Originate   Miguel Molina
• [asterisk-users] Product offerings from DIDforSale   Miguel Molina
• [asterisk-users] string length in dialplan   Miguel Molina
• [asterisk-users] subject: 1.4 vs 1.6   Miguel Molina
• [asterisk-users] AMD: HANGUP   Miguel Molina
• [asterisk-users] Deadlock while using MGCP on Asterisk   Miguel Molina
• [asterisk-users] syntax   Tom Moore
• [asterisk-users] Set CDR userfield for Queues   Luis Morales
• [asterisk-users] Important security alert: update your dialplans now!   Tony Mountifield
• [asterisk-users] Asterisk RPM's   Tony Mountifield
• [asterisk-users] Asterisk RPM's   Tony Mountifield
• [asterisk-users] Fun with virtual asterisks ...   Tony Mountifield
• [asterisk-users] audio glitches in conference   Marco Mouta
• [asterisk-users] AMI + device status (patch 0016732) + remote control   Marcus Mundt
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Muro, Sam
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Muro, Sam
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Muro, Sam
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Muro, Sam
• [asterisk-users] Important security alert: update your dialplans now!   Steve Murphy
• [asterisk-users] Important security alert: update your dialplans now!   Steve Murphy
• [asterisk-users] Important security alert: update your dialplans now!   Steve Murphy
• [asterisk-users] Virtual machine timing (KVM)   Ian Murray
• [asterisk-users] Virtual machine timing (KVM)   Ian Murray
• [asterisk-users] Losing local SIP phones when internet goes down?   Nikhil Nair
• [asterisk-users] Losing local SIP phones when internet goes down?   Nikhil Nair
• [asterisk-users] Losing local SIP phones when internet goes down?   Nikhil Nair
• [asterisk-users] Losing local SIP phones when internet goes down?   Nikhil Nair
• [asterisk-users] Losing local SIP phones when internet goes down?   Nikhil Nair
• [asterisk-users] Losing local SIP phones when internet goes down?   Nikhil Nair
• [asterisk-users] SS7 and Asterisk   Tim Nelson
• [asterisk-users] IP Phone recommendation   Tim Nelson
• [asterisk-users] Robotic sound sometimes   Tim Nelson
• [asterisk-users] Ongoing calls interface   Håkon Nessjøen
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Håkon Nessjøen
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Håkon Nessjøen
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Håkon Nessjøen
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Håkon Nessjøen
• [asterisk-users] BRI vs. PRI?   Håkon Nessjøen
• [asterisk-users] Unrecognized prilocaldialplan NPI modifier   Håkon Nessjøen
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Håkon Nessjøen
• [asterisk-users] Getting: Can't fix up channel from 5 to 7 because 7 is already in use, and pri_dchannel: Answer requested on channel 0/7 not in use on span 1   Håkon Nessjøen
• [asterisk-users] New patch for app_queue to show all call attempts, even failing ones   Håkon Nessjøen
• [asterisk-users] MATH   Danny Nicholas
• [asterisk-users] MATH   Danny Nicholas
• [asterisk-users] MATH   Danny Nicholas
• [asterisk-users] NVFaxDetect   Danny Nicholas
• [asterisk-users] Two Extensions showing as Busy   Danny Nicholas
• [asterisk-users] Semi-Transfer   Danny Nicholas
• [asterisk-users] Semi-Transfer   Danny Nicholas
• [asterisk-users] Semi-Transfer   Danny Nicholas
• [asterisk-users] # as dial key - chan_dahdi   Danny Nicholas
• [asterisk-users] Running a script after Dial() ?   Danny Nicholas
• [asterisk-users] Gotoif Question   Danny Nicholas
• [asterisk-users] Running a script after Dial() ?   Danny Nicholas
• [asterisk-users] Gotoif Question   Danny Nicholas
• [asterisk-users] Can an agent Login to a queue and be paused   Danny Nicholas
• [asterisk-users] 2 Asterisk Boxes, Single Voicemail   Danny Nicholas
• [asterisk-users] IVR Demo / Create file / Move file / Demo all   Danny Nicholas
• [asterisk-users] IVR Demo / Create file / Move file / Demo all   Danny Nicholas
• [asterisk-users] Strange Problem   Danny Nicholas
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Danny Nicholas
• [asterisk-users] ways of initiating a call   Danny Nicholas
• [asterisk-users] ways of initiating a call   Danny Nicholas
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Danny Nicholas
• [asterisk-users] Optimization of call from server 1 to 2 and thenback to 1   Danny Nicholas
• [asterisk-users] How to avoid AGI script is canceled if callerHangUp   Danny Nicholas
• [asterisk-users] app_dial.c: Unable to create channel of type 'Zap'(cause 34 - Circuit/channel congestion)   Danny Nicholas
• [asterisk-users] Stupid question: Why Cmd Dial and Queue haven'tsame options?   Danny Nicholas
• [asterisk-users] Stupid question: Why Cmd Dial andQueuehaven'tsame options?   Danny Nicholas
• [asterisk-users] call parking   Danny Nicholas
• [asterisk-users] asterisk dahdi fax problem   Danny Nicholas
• [asterisk-users] Setting up only one caller at a time   Danny Nicholas
• [asterisk-users] Volume of Playback() application   Danny Nicholas
• [asterisk-users] splitting sip.conf to two files   Danny Nicholas
• [asterisk-users] Denying call transfer to certain extensions   Danny Nicholas
• [asterisk-users] Caller ID question   Danny Nicholas
• [asterisk-users] Caller ID question   Danny Nicholas
• [asterisk-users] Denying call transfer to certain extensions   Danny Nicholas
• [asterisk-users] Calls per second limit in manager   Danny Nicholas
• [asterisk-users] identify the costumer   Danny Nicholas
• [asterisk-users] X-Lite won't register   Danny Nicholas
• [asterisk-users] Redirect call based on CLI???   Danny Nicholas
• [asterisk-users] hi   Danny Nicholas
• [asterisk-users] Lua status in asterisk.   Matthew Nicholson
• [asterisk-users] Cisco 7940: showing FWD in display.   Oliver Nittka
• [asterisk-users] Cisco 7940: showing FWD in display.   Oliver Nittka
• [asterisk-users] Cisco 7940: showing FWD in display.   Oliver Nittka
• [asterisk-users] Cisco 7940: showing FWD in display.   Oliver Nittka
• [asterisk-users] (no subject)   John Novack
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   John Novack
• [asterisk-users] E71   YC Nyon
• [asterisk-users] OT - MWI, Polycom/kirk and Gigaset handsets   Olivier
• [asterisk-users] PMS (SDMR, ...) support in Asterisk   Olivier
• [asterisk-users] Cisco 7940: showing FWD in display.   Olivier
• [asterisk-users] OT- Using TR-069   Olivier
• [asterisk-users] IP Phone recommendation   Jeffrey Ollie
• [asterisk-users] Gotoif Question   Álvaro Rosendo Olmedo
• [asterisk-users] Gotoif Question   Álvaro Rosendo Olmedo
• [asterisk-users] strange issue with iptables + Asterisk   Ernesto Ongaro
• [asterisk-users] Strange Problem   Alexandru Oniciuc
• [asterisk-users] Empty SIP Packet   Alexandru Oniciuc
• [asterisk-users] CDR duration/billsec   Alexandru Oniciuc
• [asterisk-users] Robotic sound sometimes   Rudi Oosthuizen
• [asterisk-users] IAX devices not registering after upgrade to   Rudi Oosthuizen
• [asterisk-users] Running safe_asterisk   Rudi Oosthuizen
• [asterisk-users] Denying call transfer to certain extensions   Ahmed Ossama
• [asterisk-users] Denying call transfer to certain extensions   Ahmed Ossama
• [asterisk-users] CDR and Queue Reporting windows application looking for Beta testers!   Token PBX
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Jason Parker
• [asterisk-users] Cannot built kmod-dahdi-linux for PAE kvariant from SRPM   Jason Parker
• [asterisk-users] Asterisk RPM's   Jason Parker
• [asterisk-users] 500 Internal Server Error on Cisco 7940 after INVITE   Steven Parker
• [asterisk-users] identify the costumer   Douglas Pasqua
• [asterisk-users] {top|bottom|interleaved} posting, once again   Will Payne
• [asterisk-users] Caller ID question   Will Payne
• [asterisk-users] Caller ID question   Will Payne
• [asterisk-users] IP Phone recommendation   Peder
• [asterisk-users] Robotic sound sometimes   Peder
• [asterisk-users] Robotic sound sometimes   Peder
• [asterisk-users] Not able to compile asterisk, zaptel, libpri in /usr/src   Trevor Peirce
• [asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid   Trevor Peirce
• [asterisk-users] MATH   Thomas Perron
• [asterisk-users] MATH   Thomas Perron
• [asterisk-users] MATH   Thomas Perron
• [asterisk-users] Dial script   Thomas Perron
• [asterisk-users] Dial script   Thomas Perron
• [asterisk-users] Dial script   Thomas Perron
• [asterisk-users] Dial script   Thomas Perron
• [asterisk-users] Dial script   Thomas Perron
• [asterisk-users] syntax   Thomas Perron
• [asterisk-users] syntax   Thomas Perron
• [asterisk-users] IVR Demo / Create file / Move file / Demo all   Thomas Perron
• [asterisk-users] IVR Demo / Create file / Move file / Demo all   Thomas Perron
• [asterisk-users] IVR Demo / Create file / Move file / Demo all   Thomas Perron
• [asterisk-users] Quad Card PRI, Disable Unused Ports or Manage Channels. How?   Pete
• [asterisk-users] GSM Gateway   Peter
• [asterisk-users] GSM Gateway   Peter
• [asterisk-users] IP Phone recommendation   Peter
• [asterisk-users] IP Phone recommendation   Peter
• [asterisk-users] OpenVPN on phones?   Dave Platt
• [asterisk-users] 1.6.1 Voicemail users.conf   Dave Poirier
• [asterisk-users] Avaya with Asterisk   Edwin Quijada
• [asterisk-users] Open source or low-budget recommendation for call-center software   Edwin Quijada
• [asterisk-users] OT: VUC Feb 5th @ 12 Noon Open VPN   Randy R
• [asterisk-users] Losing local SIP phones when internet goes down?   Randy R
• [asterisk-users] Losing local SIP phones when internet goes down?   Randy R
• [asterisk-users] Losing local SIP phones when internet goes down?   Randy R
• [asterisk-users] Losing local SIP phones when internet goes down?   Randy R
• [asterisk-users] VUC Friday Feb 12th: HD Communications Summit   Randy R
• [asterisk-users] Important security alert: update your dialplans now!   Randy R
• [asterisk-users] Feb 19th @12 noon EST: Voxeo's Tropo   Randy R
• [asterisk-users] sip.conf - sort order, does it matter   Randy R
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Randy R
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   Randy R
• [asterisk-users] Denying call transfer to certain extensions   Randy R
• [asterisk-users] Morse Code   Randy R
• [asterisk-users] : PSTN calls   Randy R
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   RESEARCH
• [asterisk-users] BYE message not relayed to caller   Vikram Ragukumar
• [asterisk-users] Sipgate.co.uk on Asterisk 1.6.2.2   Razza
• [asterisk-users] mISDN (HFC-S) and TDM400P - isac xdu no tx_busy   Razza
• [asterisk-users] mISDN (HFC-S) and TDM400P - isac xdu no tx_busy   Razza
• [asterisk-users] HFC-S card   Razza
• [asterisk-users] HFC-S card   Razza
• [asterisk-users] HFC-S card   Razza
• [asterisk-users] HFC-S card   Razza
• [asterisk-users] HFC-S card   Razza
• [asterisk-users] ISDN Options   Razza
• [asterisk-users] agi debug in Asterisk 1.6?   Alejandro Recarey
• [asterisk-users] agi debug in Asterisk 1.6?   Alejandro Recarey
• [asterisk-users] Load balance outgoing calls   Alejandro Recarey
• [asterisk-users] Load balance outgoing calls   Alejandro Recarey
• [asterisk-users] How to tell if asterisk is handling media or not?   Alejandro Recarey
• [asterisk-users] calling into server with cell and originate a call   John Regal
• [asterisk-users] calling into server with cell and originate a call   John Regal
• [asterisk-users] aastra 9480i dtmf ?   John Regal
• [asterisk-users] aastra 9480i dtmf ?   John Regal
• [asterisk-users] DTMF timing - first # keypress not registering   John Regal
• [asterisk-users] hi   John Regal
• [asterisk-users] 1.6.x SIP allow incoming calls based on from ip address?   JR Richardson
• [asterisk-users] 1.6.x SIP allow incoming calls based on from ip address?   JR Richardson
• [asterisk-users] Losing local SIP phones when internet goes down?   Matt Riddell
• [asterisk-users] Losing local SIP phones when internet goes down?   Matt Riddell
• [asterisk-users] strange asterisk behaviour on XEN   Matt Riddell
• [asterisk-users] Realtime extensions   Matt Riddell
• [asterisk-users] Realtime extensions   Matt Riddell
• [asterisk-users] Free iPhone Asterisk Function and Application Reference   Matt Riddell
• [asterisk-users] Calls per second limit in manager   Matt Riddell
• [asterisk-users] Calls per second limit in manager   Matt Riddell
• [asterisk-users] Calls per second limit in manager   Matt Riddell
• [asterisk-users] Calls per second limit in manager   Matt Riddell
• [asterisk-users] RES: Dahdi & Congestion status   Rafael Prado Rocchi
• [asterisk-users] High codec translation times on x64   Ron
• [asterisk-users] dnd sorta working   Ott Rose
• [asterisk-users] Asterisk Redundancy   Chris Rowson
• [asterisk-users] MeetMe() and dahdi_dummy on an embedded system   Shaun Ruffell
• [asterisk-users] ISDN phone not ringing. ISDN PBX not answering?!   René Rössler
• [asterisk-users] SS7 and Asterisk   ABBAS SHAKEEL
• [asterisk-users] SS7 and Asterisk   ABBAS SHAKEEL
• [asterisk-users] Asterisk Cepstral TTS   ABBAS SHAKEEL
• [asterisk-users] Asterisk Cepstral TTS   ABBAS SHAKEEL
• [asterisk-users] Ideasip   SIP
• [asterisk-users] Losing local SIP phones when internet goes down?   Alex Samad
• [asterisk-users] Losing local SIP phones when internet goes down?   Alex Samad
• [asterisk-users] OpenVPN/SNOM 820: a review.   Alex Samad
• [asterisk-users] subject: 1.4 vs 1.6   Juan Sandro
• [asterisk-users] subject: 1.4 vs 1.6   Juan Sandro
• [asterisk-users] Call Pickup with 1.6.2.1 and Snom   Loris Santamaria
• [asterisk-users] HFC-S card   Pedro Santos
• [asterisk-users] HFC-S card   Pedro Santos
• [asterisk-users] HFC-S card   Pedro Santos
• [asterisk-users] HFC-S card   Pedro Santos
• [asterisk-users] Asterisk for productive Calling Card System   Tarek Sawah
• [asterisk-users] SIP RTP ports not released when channel is hung up   Armin Schindler
• [asterisk-users] SIP RTP ports not released when channel is hung up   Armin Schindler
• [asterisk-users] SIP RTP ports not released when channel is hung up   Armin Schindler
• [asterisk-users] SIP RTP ports not released when channel is hung up   Armin Schindler
• [asterisk-users] SIP RTP ports not released when channel is hung up   Armin Schindler
• [asterisk-users] SIP RTP ports not released when channel is hung up   Armin Schindler
• [asterisk-users] Setting up only one caller at a time   Stefan Schmidt
• [asterisk-users] Smallest possible Asterisk VM   Ben Schorr
• [asterisk-users] Not able to compile asterisk, zaptel, libpri in /usr/src   Ben Schorr
• [asterisk-users] extension not found   Ben Schorr
• [asterisk-users] BRI vs. PRI?   Roger Schreiter
• [asterisk-users] Losing local SIP phones when internet goes down?   Warren Selby
• [asterisk-users] Polycom phone DND state   Warren Selby
• [asterisk-users] queue with strategy=linear   Warren Selby
• [asterisk-users] Security Logging   Warren Selby
• [asterisk-users] Security Logging   Warren Selby
• [asterisk-users] Nat Issue - is this Draytek || Asterisk?   Warren Selby
• [asterisk-users] IP Phone recommendation   Warren Selby
• [asterisk-users] IP Phone recommendation   Warren Selby
• [asterisk-users] How does holdtime get calculated for queues   Warren Selby
• [asterisk-users] Important security alert: update your dialplans now!   Warren Selby
• [asterisk-users] Asterisk listens on all NICs   Warren Selby
• [asterisk-users] Important security alert: update your dialplans now!   Warren Selby
• [asterisk-users] Important security alert: update your dialplans now!   Warren Selby
• [asterisk-users] queue.conf - Set(MONITOR_FILENAME=${})   Warren Selby
• [asterisk-users] Registering of Asterisk against a SIP provider   Warren Selby
• [asterisk-users] how asterisk knows which context forward the call to?   Warren Selby
• [asterisk-users] Problems installing dahdi : kernel sources   Warren Selby
• [asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available   Warren Selby
• [asterisk-users] IAX devices not registering after upgrade to asterisk 1.4.29   Vidura Senadeera
• [asterisk-users] IAX devices not registering after upgrade to asterisk   Vidura Senadeera
• [asterisk-users] No RTP from asterisk?   Peter Serwe
• [asterisk-users] No RTP from asterisk?   Peter Serwe
• [asterisk-users] Semi-Transfer   James A. Shigley
• [asterisk-users] Semi-Transfer   James A. Shigley
• [asterisk-users] Semi-Transfer   James A. Shigley
• [asterisk-users] Question   James A. Shigley
• [asterisk-users] Realtime queue strategy issue   Zhang Shukun
• [asterisk-users] Does Playback will answer the call?   Zhang Shukun
• [asterisk-users] Do i need install Dahdi or libpri ?   Zhang Shukun
• [asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available   Zhang Shukun
• [asterisk-users] Do i need install Dahdi or libpri ?   Zhang Shukun
• [asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available   Zhang Shukun
• [asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available   Zhang Shukun
• [asterisk-users] realtime modules not load ?   Zhang Shukun
• [asterisk-users] Realtime extensions   Jared Smith
• [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup   Jared Smith
• [asterisk-users] Dial Plan configuration in asterisk   Chandrakant Solanki
• [asterisk-users] SIP tunnel   Jamie A. Stapleton
• [asterisk-users] Which H.323 to use in Ast 1.6   Jamie A. Stapleton
• [asterisk-users] Which H.323 to use in Ast 1.6   Jamie A. Stapleton
• [asterisk-users] Fax, T38 and NAT   Johann Steinwendtner
• [asterisk-users] Does Playback will answer the call?   Johann Steinwendtner
• [asterisk-users] conferencing without DAHDI   Philippe Sultan
• [asterisk-users] conferencing without DAHDI   Philippe Sultan
• [asterisk-users] conferencing without DAHDI   Philippe Sultan
• [asterisk-users] OT: Problems with Linksys IP Phone SPA 942   Shanon Swafford
• [asterisk-users] Losing local SIP phones when internet goesdown?   Sweet, Larry D
• [asterisk-users] chan_local and Originate   James Northcott / Chief Systems
• [asterisk-users] chan_local and Originate   James Northcott / Chief Systems
• [asterisk-users] Asterisk ignores BYE messages   Szasz Szabolcs
• [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals   Administrator TOOTAI
• [asterisk-users] Registering of Asterisk against a SIP provider   Administrator TOOTAI
• [asterisk-users] Server response time   Administrator TOOTAI
• [asterisk-users] forward call back up same trunk to external cell phone problem   John Taylor
• [asterisk-users] Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 Released   Asterisk Development Team
• [asterisk-users] Asterisk 1.2.39 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.2.40, 1.4.29.1, 1.6.0.24, 1.6.1.16, and 1.6.2.4 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.0.25, 1.6.1.17, and 1.6.2.5 Now Available   Asterisk Development Team
• [asterisk-users] AST-2010-001: T.38 Remote Crash Vulnerability   Asterisk Security Team
• [asterisk-users] AST-2010-002: Dialplan injection vulnerability   Asterisk Security Team
• [asterisk-users] AST-2010-003: Invalid parsing of ACL rules can compromise security   Asterisk Security Team
• [asterisk-users] Aastra 50-limit blf   Andrew Thomas
• [asterisk-users] 1.6.1 Voicemail users.conf   Jonathan Thurman
• [asterisk-users] how asterisk knows which context forward the call to?   John Timms
• [asterisk-users] IP Phone recommendation   Brent Torrenga
• [asterisk-users] IP Kall One-Way Audio   Brent Torrenga
• [asterisk-users] One-Way Audio after Hold   Brent Torrenga
• [asterisk-users] OpenVPN on phones?   Steve Totaro
• [asterisk-users] OpenVPN on phones?   Steve Totaro
• [asterisk-users] OpenVPN on phones?   Steve Totaro
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Steve Totaro
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Steve Totaro
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Steve Totaro
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Steve Totaro
• [asterisk-users] Get Talk Time   Steve Totaro
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Steve Totaro
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   Steve Totaro
• [asterisk-users] T.38 with reinvite   Steve Totaro
• [asterisk-users] Asterisk Redundancy   Steve Totaro
• [asterisk-users] Dial script   Erik de Wild: Tripple-o
• [asterisk-users] {top|bottom|interleaved} posting, once again   Erik de Wild: Tripple-o
• [asterisk-users] Asterisk n-way DTMF detection   Tri Tu
• [asterisk-users] Conference Calling   Tri Tu
• [asterisk-users] Asterisk n-way DTMF detection   Tri Tu
• [asterisk-users] X-Lite won't register   Tri Tu
• [asterisk-users] No RTP from asterisk?   Tri Tu
• [asterisk-users] Compatible IP Phones for Asterisk   Tri Tu
• [asterisk-users] Redirect call based on CLI???   D Tucny
• [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory   Duncan Turnbull
• [asterisk-users] No RTP from asterisk?   Duncan Turnbull
• [asterisk-users] Intel Atom based Asterisk server?   Lyle Underwood
• [asterisk-users] Know what would be killer?   Lyle Underwood
• [asterisk-users] Know what would be killer?   Lyle Underwood
• [asterisk-users] Know what would be killer?   Lyle Underwood
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Steve Underwood
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Steve Underwood
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Steve Underwood
• [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing   Steve Underwood
• [asterisk-users] Still on spandsp/app_fax and T.38   Steve Underwood
• [asterisk-users] Still on spandsp/app_fax and T.38   Steve Underwood
• [asterisk-users] Still on spandsp/app_fax and T.38   Steve Underwood
• [asterisk-users] Still on spandsp/app_fax and T.38   Steve Underwood
• [asterisk-users] Asterisk t38modem Fax gateway evaluation   Steve Underwood
• [asterisk-users] Slightly OT: Has SILK codec gotten anywhere?   Steve Underwood
• [asterisk-users] Codec translation in Asterisk   Asterisk User
• [asterisk-users] Asterisk & Fax   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] Dial Plan configuration in asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] Dial Plan configuration in asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] 4 PCIe cards in one asterisk server   Christian Victor
• [asterisk-users] Do i need install Dahdi or libpri ?   Christian Victor
• [asterisk-users] SIP provider registration attempts   Vieri
• [asterisk-users] SIP provider registration attempts   Vieri
• [asterisk-users] SIP provider registration attempts   Vieri
• [asterisk-users] Server response time   Juan C. Villa
• [asterisk-users] # as dial key - chan_dahdi   Marcus Vinicius
• [asterisk-users] Asterisk RPM's   Jay Vocaire
• [asterisk-users] Asterisk RPM's   Jay Vocaire
• [asterisk-users] Asterisk RPM's   Jay Vocaire
• [asterisk-users] T.38 with reinvite   Kristijan Vrban
• [asterisk-users] billing based on local access number   C. Chad Wallace
• [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals   C. Chad Wallace
• [asterisk-users] Issue with trying to dial two different servers at the same time.   C. Chad Wallace
• [asterisk-users] Asterisk listens on all NICs   C. Chad Wallace
• [asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid   Charles Wang
• [asterisk-users] SIP RTP ports not released when channel is hung up   Karsten Wemheuer
• [asterisk-users] large scale paging   Mark Willis
• [asterisk-users] large scale paging   Mark Willis
• [asterisk-users] large scale paging   Mark Willis
• [asterisk-users] large scale paging   Mark Willis
• [asterisk-users] large scale paging   Mark Willis
• [asterisk-users] Queue problem, ringing agents.   Jeremy Winder
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Thomas Winter
• [asterisk-users] How to avoid AGI script is canceled if caller HangUp   Thomas Winter
• [asterisk-users] OT: Problems with Linksys IP Phone SPA 942   Vahan Yerkanian
• [asterisk-users] some newbie questions about gcc   Givon Zirkind
• [asterisk-users] voip host in israel   Givon Zirkind
• [asterisk-users] security & dtmf   Givon Zirkind
• [asterisk-users] Slightly OT: Has SILK codec gotten anywhere?   Zoa
• [asterisk-users] Web operator/softphone with integration features   Zoa
• [asterisk-users] Two Extensions showing as Busy   --[ UxBoD ]--
• [asterisk-users] Two Extensions showing as Busy   --[ UxBoD ]--
• [asterisk-users] MATH   --[ UxBoD ]--
• [asterisk-users] Intel Atom based Asterisk server?   --[ UxBoD ]--
• [asterisk-users] Intel Atom based Asterisk server?   --[ UxBoD ]--
• [asterisk-users] OpenVPN on phones?   --[ UxBoD ]--
• [asterisk-users] Losing local SIP phones when internet goes down?   --[ UxBoD ]--
• [asterisk-users] Losing local SIP phones when internet goes down?   --[ UxBoD ]--
• [asterisk-users] Losing local SIP phones when internet goes down?   --[ UxBoD ]--
• [asterisk-users] Website Down ?   --[ UxBoD ]--
• [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75   --[ UxBoD ]--
• [asterisk-users] asterisk and mysql connection   --[ UxBoD ]--
• [asterisk-users] Security Logging   --[ UxBoD ]--
• [asterisk-users] Dial Plan configuration in asterisk   --[ UxBoD ]--
• [asterisk-users] OpenVPN/SNOM 820: a review.   --[ UxBoD ]--
• [asterisk-users] OpenVPN/SNOM 820: a review.   --[ UxBoD ]--
• [asterisk-users] OpenVPN/SNOM 820: a review.   --[ UxBoD ]--
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   --[ UxBoD ]--
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   --[ UxBoD ]--
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   --[ UxBoD ]--
• [asterisk-users] [OT] Asterisk 1.6 and DECT Phones   --[ UxBoD ]--
• [asterisk-users] Followme broken   --[ UxBoD ]--
• [asterisk-users] Followme broken   --[ UxBoD ]--
• [asterisk-users] Followme broken   --[ UxBoD ]--
• [asterisk-users] hi   --[ UxBoD ]--
• [asterisk-users] Routing inbound call to correct sip trunk   antselva
• [asterisk-users] Polycom VVX1500 video working yet?   asterisk
• [asterisk-users] Polycom VVX1500 video working yet?   asterisk
• [asterisk-users] transmit_silence_during_record   jonathan augenstine
• [asterisk-users] Volume of Playback() application   Renato bianchini
• [asterisk-users] Problems with Linksys IP Phone SPA 942   Renato bianchini
• [asterisk-users] Cannot built kmod-dahdi-linux for PAE kvariant from SRPM   stephen.hindmarch at bt.com
• [asterisk-users] Cannot built kmod-dahdi-linux for PAE kvariant from SRPM   stephen.hindmarch at bt.com
• [asterisk-users] 500 Internal Server Error on Cisco 7940 after INVITE   ian at comtek.co.uk
• [asterisk-users] 500 Internal Server Error on Cisco 7940 after INVITE   ian at comtek.co.uk
• [asterisk-users] 2 Asterisk Boxes, Single Voicemail   ian at comtek.co.uk
• [asterisk-users] 1.6.2.1: DTMF trouble with PSTN   sean darcy
• [asterisk-users] 1.6.2.1: DTMF trouble with PSTN   sean darcy
• [asterisk-users] 1.6.2.1: DTMF trouble with PSTN   sean darcy
• [asterisk-users] Losing local SIP phones when internet goes down?   sean darcy
• [asterisk-users] E71   sean darcy
• [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals   sean darcy
• [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals   sean darcy
• [asterisk-users] 1.6.2.1: DTMF trouble with PSTN   sean darcy
• [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals   sean darcy
• [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals   sean darcy
• [asterisk-users] Codec coversion   wassim darwich
• [asterisk-users] codec conversion   wassim darwich
• [asterisk-users] Stupid question: Why Cmd Dial and Queue haven't same options?   didier.cuffaut
• [asterisk-users] Stupid question: Why Cmd Dial and Queuehaven'tsame options?   didier.cuffaut
• [asterisk-users] how to allow some extensions to make call outside and some extensions cant call outside   cool dude
• [asterisk-users] how to allow some extensions to make call outside and some extensions cant call outside   cool dude
• [asterisk-users] how to allow some extenstions to call outside and some extensions cant call outside   cool dude
• [asterisk-users] how to allow some extenstions to call outside and some extensions cant call outside   cool dude
• [asterisk-users] parked calls   cool dude
• [asterisk-users] Call Pickup with 1.6.2.1 and Snom   cool dude
• [asterisk-users] problems with 1.6   cool dude
• [asterisk-users] Robotic sound sometimes   cool dude
• [asterisk-users] Robotic sound sometimes   cool dude
• [asterisk-users] Robotic sound sometimes   cool dude
• [asterisk-users] Robotic sound sometimes   cool dude
• [asterisk-users] extension not found   cool dude
• [asterisk-users] how to create voicemail   cool dude
• [asterisk-users] voicemail problem   cool dude
• [asterisk-users] how to have disconnect signals enabled in line   cool dude
• [asterisk-users] signal problem   cool dude
• [asterisk-users] call transfer   cool dude
• [asterisk-users] call parking   cool dude
• [asterisk-users] How to transfer call using function T   cool dude
• [asterisk-users] pickup the call: No target channel found   bilal ghayyad
• [asterisk-users] A2Billing and other prepaid Billing like ASTCC, who is better?   bilal ghayyad
• [asterisk-users] Pickup the call ringing at SIP Phone but was transferred from Zap channel   bilal ghayyad
• [asterisk-users] A2Billing and other prepaid Billing like ASTCC, who is better?   bilal ghayyad
• [asterisk-users] record a user call while playing a background music   huu giang
• [asterisk-users] Optimization of call from server 1 to 2 and then back to 1   mancyborg at gmail.com
• [asterisk-users] Optimization of call from server 1 to 2 and thenback to 1   mancyborg at gmail.com
• [asterisk-users] Encrypted calls between mobile gsm and isdn (asterisk)   mancyborg at gmail.com
• [asterisk-users] Lower kernel version for mISDN   mancyborg at gmail.com
• [asterisk-users] audio glitches in conference   marco.mouta at gmail.com
• [asterisk-users] IP Phone recommendation   mthayeb at gmail.com
• [asterisk-users] Macros, GoSub & StackPop   hugolivude
• [asterisk-users] One way audio with Grandstream HT503   jonas kellens
• [asterisk-users] ast_cdr_setvar: Attempt to set the 'src' read-only variable!   jonas kellens
• [asterisk-users] Realtime extensions   jonas kellens
• [asterisk-users] Realtime extensions   jonas kellens
• [asterisk-users] Realtime extensions   jonas kellens
• [asterisk-users] Realtime extensions   jonas kellens
• [asterisk-users] Problems with SIP realtime   jonas kellens
• [asterisk-users] Problems with SIP realtime   jonas kellens
• [asterisk-users] Problems with SIP realtime   jonas kellens
• [asterisk-users] Problems with SIP realtime   jonas kellens
• [asterisk-users] Problems installing dahdi : kernel sources   jonas kellens
• [asterisk-users] parked calls   hin lee
• [asterisk-users] parked calls   hin lee
• [asterisk-users] Get Talk Time   lemonash
• [asterisk-users] IAX peers one way voice   lore
• [asterisk-users] Problems in Asterisk Real Time (Urgent help )   ahmed magdy
• [asterisk-users] billing based on local access number   umesh maharjan
• [asterisk-users] (no subject)   nasar mahmud
• [asterisk-users] SIP tunnel   wins mallow
• [asterisk-users] directrtp with SIP + H.323   wins mallow
• [asterisk-users] Important security alert: update your dialplans now!   meetmecall
• [asterisk-users] Important security alert: update your dialplans now!   meetmecall
• [asterisk-users] Conference Calling   meetmecall
• [asterisk-users] Asterisk Cepstral TTS   mj
• [asterisk-users] Asterisk Cepstral TTS   mj
• [asterisk-users] Asterisk Cepstral TTS   mj
• [asterisk-users] Asterisk Cepstral TTS   mj
• [asterisk-users] SIP tunnel   mosbah.abdelkader
• [asterisk-users] SIP tunnel   mosbah.abdelkader
• [asterisk-users] SIP tunnel   mosbah.abdelkader
• [asterisk-users] [Fwd: SIP tunnel]   mosbah.abdelkader
• [asterisk-users] SIP tunnel   mosbah.abdelkader
• [asterisk-users] SIP tunnel   mosbah.abdelkader
• [asterisk-users] SIP tunnel   mosbah.abdelkader
• [asterisk-users] SIP tunnel   mosbah.abdelkader
• [asterisk-users] SIP tunnel   mosbah.abdelkader
• [asterisk-users] Asterisk 1.6.2 ?   hadi motamedi
• [asterisk-users] Asterisk 1.6.2 ?   hadi motamedi
• [asterisk-users] Asterisk 1.6.2 ?   hadi motamedi
• [asterisk-users] Asterisk how install speex support   nedo nodo
• [asterisk-users] Not able to compile asterisk, zaptel, libpri in /usr/src   asterisk at opensourcesolution.in
• [asterisk-users] Not able to compile asterisk, zaptel, libpri in /usr/src   asterisk at opensourcesolution.in
• [asterisk-users] test   asterisk at opensourcesolution.in
• [asterisk-users] how to allow some extensions to make call outside and some extensions cant call outside   asterisk at opensourcesolution.in
• [asterisk-users] Use a BLF for monitoring   jon pounder
• [asterisk-users] large scale paging   jon pounder
• [asterisk-users] identify the costumer   jon pounder
• [asterisk-users] connect problem unless when verbose   randall
• [asterisk-users] connect problem unless when verbose   randall
• [asterisk-users] forward incomming line to modem   randall
• [asterisk-users] forward incomming line to modem   randall
• [asterisk-users] Astribank problem   frangky robert
• [asterisk-users] Astribank problem   frangky robert
• [asterisk-users] Help for MOH - sounding scratchy/static on hold   das sandesh
• [asterisk-users] asterisk sudden restart - 1.4.18.1   das sandesh
• [asterisk-users] asterisk sudden restart - 1.4.18.1 - Resolved after upgrade to 1.4.22.1   das sandesh
• [asterisk-users] dropping line (s) for 911   mir shahnawaz
• [asterisk-users] dropping line (s) for 911   mir shahnawaz
• [asterisk-users] ways of initiating a call   tom
• [asterisk-users] ways of initiating a call   tom
• [asterisk-users] ways of initiating a call   tom
• [asterisk-users] nortle BCM450 & SIP-Trunking   tom
• [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?   khalid touati
• [asterisk-users] insecure=invite - not working for different dial plan   uzzi
• [asterisk-users] insecure=invite - not working for different dial plan   uzzi
• [asterisk-users] Call doesn't disconnect in SIP   velusamy velu
• [asterisk-users] call count per peer   voipas
• [asterisk-users] add Reason header on hangup   voipas
• [asterisk-users] curl and ssl certificate   voipas
• [asterisk-users] asterisk and mysql connection   김무성

Last message date: Sun Feb 28 20:21:23 CST 2010
Archived on: Sun Feb 28 20:21:34 CST 2010

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