September 2006 Archives by author

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Starting: Fri Sep 1 00:03:26 MST 2006
Ending: Sat Sep 30 20:54:35 MST 2006
Messages: 3621

• [asterisk-users] Any Hardphone with VPNClient embedded?   Francesco Peeters (Asterisk)
• [asterisk-users] Asterisk behind Sonicwall firewall   Francesco Peeters (Asterisk)
• [asterisk-users] Good Book on Asterisk   Leonardo Kamache (Gmail)
• [asterisk-users] Asterisk mysql cdr   Matt Riddell (IT)
• [asterisk-users] Using Thunderbird (mail client) to call Contacts from Address Book   Matt Riddell (IT)
• [asterisk-users] Asterisk not sending RTP   Matt Riddell (IT)
• [asterisk-users] UK (BT) Problem with TDM 400P   Matt Riddell (IT)
• [asterisk-users] Asterisk not sending RTP   Matt Riddell (IT)
• [asterisk-users] UK (BT) Problem with TDM 400P   Matt Riddell (IT)
• [asterisk-users] app_conference not working for me   Matt Riddell (IT)
• [asterisk-users] A couple more interviews with Digium staff   Matt Riddell (IT)
• [asterisk-users] Codec Thread   Matt Riddell (IT)
• [asterisk-users] Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne cheaper than existing one.   Matt Riddell (IT)
• [asterisk-users] Adding custom fields (more than one) to CDR DB   Matt Riddell (IT)
• [asterisk-users] Response to KP Flemming...   Matt Riddell (IT)
• [asterisk-users] Capacity for transcode G711 to G729   Matt Riddell (IT)
• [asterisk-users] Capacity for transcode G711 to G729   Matt Riddell (IT)
• [asterisk-users] Asterisk Outgoing Spool Failed   Matt Riddell (IT)
• [asterisk-users] svn trunk or branches ???   Matt Riddell (IT)
• [asterisk-users] Capacity for transcode G711 to G729   Matt Riddell (IT)
• [asterisk-users] Asterisk Outgoing Spool Failed with ViciDial (MattF?)   Matt Riddell (IT)
• [asterisk-users] Asterisk crashing when monitoring SIP device with Chanspy   Matt Riddell (IT)
• [asterisk-users] Building Zaptel 1.2.9 with Octasic   Matt Riddell (IT)
• [asterisk-users] OT -- echo cancellation of an audio file   Matt Riddell (IT)
• [asterisk-users] Copyright issues with libcurl and OpenSSL   Matt Riddell (IT)
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   Matt Riddell (IT)
• [asterisk-users] Copyright issues with libcurl and OpenSSL   Matt Riddell (IT)
• [asterisk-users] Flexible Wrap Up Time for Queue   Matt Riddell (IT)
• [asterisk-users] (Off-topic) Voip number "tracert"   Matt Riddell (IT)
• [asterisk-users] Sphinx2   Matt Riddell (IT)
• [asterisk-users] Sphinx2   Matt Riddell (IT)
• [asterisk-users] How to download asterisk 1.3 development version   Matt Riddell (IT)
• [asterisk-users] "Too many files..." error - best way to fix?   Matt Riddell (IT)
• [asterisk-users] non-technical, dealing with users giving feedback   Matt Riddell (IT)
• [asterisk-users] non-technical, dealing with users giving feedback   Matt Riddell (IT)
• [asterisk-users] Sphinx2   Matt Riddell (IT)
• [asterisk-users] Sphinx2   Matt Riddell (IT)
• [asterisk-users] Asterisk 1.2.12.1 and Zaptel 1.2.9.1 Released   Matt Riddell (IT)
• [asterisk-users] Reliability of the newer IAXy's   Matt Riddell (IT)
• [asterisk-users] SHSU asterisk installation?   Matt Riddell (IT)
• [asterisk-users] RTCP and RTP packetization in 1.4   Matt Riddell (IT)
• [asterisk-users] RTCP and RTP packetization in 1.4   Matt Riddell (IT)
• [asterisk-users] Asterisk 1.4 Beta Uploaded!!!   Matt Riddell (IT)
• [asterisk-users] Digium G.729 codec binaries updated for Asterisk 1.4 beta   Matt Riddell (IT)
• [asterisk-users] Digium G.729 codec binaries updated for Asterisk 1.4 beta   Matt Riddell (IT)
• [asterisk-users] How to test TE405P T1   Andy Chung (Power-All)
• [asterisk-users] How to test TE405P T1   Andy Chung (Power-All)
• [asterisk-users] Grandstream and H.264 !   Sergio (Red)
• [asterisk-users] No more linux/compiler.h in Fedora Core 6.   William F. Acker WB2FLW +1-303-722-7209
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• [asterisk-users] Asterisk as a B2BUA and/or a FXS-SIP gateway?   AK 4asterisk
• [asterisk-users] When does Scalability requests Asterisk   AK 4asterisk
• [asterisk-users] When does Scalability requests Asterisk   AK 4asterisk
• [asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?   Raphaël Jacquot
• [asterisk-users] is there anyone working with 5ESS?   Raphaël Jacquot
• [asterisk-users] is there anyone working with 5ESS?   Raphaël Jacquot
• [asterisk-users] Format_MP3, Streaming, File Formats, MOH   Raphaël Jacquot
• [asterisk-users] Re: Dual core   Raphaël Jacquot
• [asterisk-users] How can I stop lost DNS from killing Asterisk?   Raphaël Jacquot
• [asterisk-users] University dumps CISCO VoIP for Asterisk   Raphaël Jacquot
• [asterisk-users] señalizacion te110p, signaling te110p   Raphaël Jacquot
• [asterisk-users] WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long]   Raphaël Jacquot
• [asterisk-users] Re: max number of devices in hint   Raphaël Jacquot
• [asterisk-users] TE406P not working on Intel D101Ggc motherboard.   Álvaro Palma
• [asterisk-users] RTT in rtcp debug   André Abrantes
• [asterisk-users] IVR not able to Play the Balance.. need some helphere   Héctor Maldonado
• [asterisk-users] Little help for a newbie configuring a TDM13B - ztcfg fails on channel 4   Iván Vega R.
• [asterisk-users] Re: Little help for a newbie configuring a TDM13B - ztcfg fails on channel 4   Iván Vega R.
• [asterisk-users] No such device -> TDM13B   Iván Vega R.
• [asterisk-users] Stupid question about FXS/FXO   Iván Vega R.
• [asterisk-users] Re: No such device -> TDM13B   Iván Vega R.
• [asterisk-users] Re: No such device -> TDM13B   Iván Vega R.
• [asterisk-users] Stupid question about FXS/FXO   Iván Vega R.
• [asterisk-users] Little help for a newbie configuring a TDM13B - ztcfg fails on channel 4   Iván Vega R.
• [asterisk-users] Trouble connecting to my telco with fonebridge   Leif Kristian Hetlesæther
• [asterisk-users] External calls from Asterisk over a Siemens(legacy) RDSI PBX   Llorenç Suau
• [Asterisk-Users] External calls from Asteris over a legacy Siemens BusinessPhone 250 PBX   Llorenç Suau
• [Asterisk-users] Calls between IAX2 Clients don't work correctly   Llorenç Suau
• [asterisk-users] How to install HUDLite Server   Nicolás Gudiño
• [asterisk-users] FOP Installation help   Nicolás Gudiño
• [asterisk-users] How to install HUDLite Server   Nicolás Gudiño
• [asterisk-users] Periodic announcements & MySQL Realtime   Sébastien Mortier
• [asterisk-users] pickup call little complicated   Miloš Kocbek
• [asterisk-users] pickup call little complicated   Miloš Kocbek
• RE: [asterisk-users] Re: ASTTAPI   Guido Hecken
• [asterisk-users] mobile refusing call   René Enskat [Teamware GmbH]
• [asterisk-users] WG: mobile refusing call   René Enskat [Teamware GmbH]
• [asterisk-users] native sounds   Ed Nuñez
• [asterisk-users] Format_MP3, Streaming, File Formats, MOH   Frédéric Marti
• [asterisk-users] Format_MP3, Streaming, File Formats, MOH   Frédéric Marti
• [asterisk-users] Problems with app_directed_pickup   Jon Schøpzinsky
• SV: [asterisk-users] txfax reliability on TDM cards   Jon Schøpzinsky
• [asterisk-users] Section '12345678' lacks type   Lennart Utgård
• [asterisk-users] ATA with wireless client   Francisco Pérez Botella
• [asterisk-users] No answer time   Dominik Kiełb
• [asterisk-users] Re: Any way to go from factory reset 7970 to SIP without Call Manager?   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Tomislav Parčina
• [asterisk-users] Re: How to use Grandstream GX-2000 phones for paging   Tomislav Parčina
• [asterisk-users] Cisco 7970 directories and services xml   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 directories and services xml   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 directories and services xml   Tomislav Parčina
• [asterisk-users] Re: MSSQL connection   Tomislav Parčina
• [asterisk-users] Re: Take 3 -- Trying to get SIP firmware on a 7970G   Tomislav Parčina
• [asterisk-users] Re: Features.. phone vs. asterisk?   Tomislav Parčina
• [asterisk-users] Queue - persistent members   Tomislav Parčina
• [asterisk-users] Re: PRI: sometimes Asterisk drop calls   Tomislav Parčina
• [asterisk-users] Queue - static members   Tomislav Parčina
• [asterisk-users] Re: PRI: sometimes Asterisk drop calls   Tomislav Parčina
• [asterisk-users] RE: voicemailmain errors on CLI   Tomislav Parčina
• [asterisk-users] Re: Queue - static members   Tomislav Parčina
• [asterisk-users] Re: PRI: sometimes Asterisk drop calls   Tomislav Parčina
• [asterisk-users] Re: Maximum retries exceeded on transmission   Tomislav Parčina
• [asterisk-users] Re: BLF across asterisk trunks   Tomislav Parčina
• [asterisk-users] RE: Asterisk 1.4 Docs   Tomislav Parčina
• [asterisk-users] AOC - advice of charge   Tomislav Parčina
• [asterisk-users] Re: Cisco 79xx and vlan   Tomislav Parčina
• [asterisk-users] Re: Can you explain why multiple registration is an important (missing) feature ?   Tomislav Parčina
• [asterisk-users] RE: Asterisk 1.4 Docs   Tomislav Parčina
• [asterisk-users] Re: Can you explain why multiple registration is an important (missing) feature ?   Tomislav Parčina
• [asterisk-users] Queue - Agent language   Tomislav Parčina
• [asterisk-users] Re: Log out an Agent on RNA   Tomislav Parčina
• [asterisk-users] Playtones   Tomislav Parčina
• [asterisk-users] Re: unable to change the emailbody for email notification   Tomislav Parčina
• [asterisk-users] Re: Playtones   Tomislav Parčina
• [asterisk-users] Re: Playtones   Tomislav Parčina
• [asterisk-users] Re: mpg123   Tomislav Parčina
• [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?   Tomislav Parčina
• [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?   Tomislav Parčina
• [asterisk-users] Dual core   Tomislav Parčina
• [asterisk-users] E1 - PCI-Express   Tomislav Parčina
• [asterisk-users] RE: Dual core   Tomislav Parčina
• [asterisk-users] Re: Dual core   Tomislav Parčina
• [asterisk-users] Re: Dual core   Tomislav Parčina
• [asterisk-users] Re: Very high ping times from 7960 phones   Tomislav Parčina
• [asterisk-users] Cisco 7970 - DTMF   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 - DTMF   Tomislav Parčina
• [asterisk-users] RE: Dual core   Tomislav Parčina
• [asterisk-users] Re: Dual core   Tomislav Parčina
• [asterisk-users] Re: asterisk to cell phone network   Tomislav Parčina
• [asterisk-users] Voip Buster - CID   Tomislav Parčina
• [asterisk-users] Re: max number of devices in hint   Tomislav Parčina
• [asterisk-users] Re: Advice of charge   Tomislav Parčina
• [asterisk-users] Re: max number of devices in hint   Tomislav Parčina
• [asterisk-users] Re: Voip Buster - CID   Tomislav Parčina
• [asterisk-users] Re: ASTTAPI   Tomislav Parčina
• [asterisk-users] Re: RPID   Tomislav Parčina
• [asterisk-users] Asterisk <=> E1 <=> Alcatel OXO   Tomislav Parčina
• [asterisk-users] running ooh323 on asterisk-1.14beta2   Илья Шипицин
• [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's   Álvaro Palma
• [asterisk-users] g729 problem   Benoît Mérouze
• [asterisk-users] Getting no Audio with G729   Benoît Mérouze
• [asterisk-users] Asterisk mysql cdr   Abdul
• [asterisk-users] Silence Call {very very urgent plz}   Abdul
• [asterisk-users] Realtime; User not registering {Urgent}   Abdul
• [asterisk-users] Re: notransfer local channel on redirect   Juan Pablo Abuyeres
• [asterisk-users] Virtualise asterisk on Xen   Mail Lists Account
• [asterisk-users] Answer Machine detection   Mark Ackroyd
• [asterisk-users] Sipura SPA3000   Rich Adamson
• [asterisk-users] Sipura SPA3000   Rich Adamson
• [asterisk-users] Sipura SPA3000   Rich Adamson
• [asterisk-users] Sipura SPA3000   Rich Adamson
• [asterisk-users] UK (BT) Problem with TDM 400P   Rich Adamson
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Rich Adamson
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Rich Adamson
• [asterisk-users] ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server   Rich Adamson
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Rich Adamson
• [asterisk-users] Re: Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute   Rich Adamson
• [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)   Rich Adamson
• [asterisk-users] Zaptel-1.2.9 compile error   Rich Adamson
• [asterisk-users] Zaptel-1.2.9 compile error   Rich Adamson
• [asterisk-users] Polycom related question   Rich Adamson
• [asterisk-users] Voip providers and sip origination and termination?   Rich Adamson
• [asterisk-users] Zaptel-1.2.9 compile error   Rich Adamson
• [asterisk-users] Voip providers and sip origination and termination?   Rich Adamson
• [asterisk-users] Polycom related question   Rich Adamson
• [asterisk-users] using residential voip for business?   Rich Adamson
• [asterisk-users] using residential voip for business?   Rich Adamson
• [asterisk-users] using residential voip for business?   Rich Adamson
• [asterisk-users] Context   Rich Adamson
• [asterisk-users] question...   Rich Adamson
• [asterisk-users] question...   Rich Adamson
• [asterisk-users] PRI: sometimes Asterisk drop calls   Rich Adamson
• [asterisk-users] PRI: sometimes Asterisk drop calls   Rich Adamson
• [asterisk-users] Switch Experiences   Rich Adamson
• [asterisk-users] 9 becomes 99 ? And other strangeness   Rich Adamson
• [asterisk-users] 9 becomes 99 ? And other strangeness   Rich Adamson
• [asterisk-users] Cisco Distinctive ring using alert-info   Rich Adamson
• [asterisk-users] Cisco Distinctive ring using alert-info   Rich Adamson
• [asterisk-users] Why not g726-32?   Rich Adamson
• [asterisk-users] Reliability of the newer IAXy's   Rich Adamson
• [asterisk-users] Why not g726-32?   Rich Adamson
• [asterisk-users] Cisco Distinctive ring using alert-info   Rich Adamson
• [asterisk-users] Asterisk Design Question   Rich Adamson
• [asterisk-users] SPA941 -> Asterisk -> Voip provider -> PSTN -> ShoreTel garble   Rich Adamson
• [asterisk-users] Setting QOS settings in asterisk and/or CentOS?   Rich Adamson
• [asterisk-users] Setting QOS settings in asterisk and/or CentOS?   Rich Adamson
• [asterisk-users] Re: Very high ping times from 7960 phones   Rich Adamson
• [asterisk-users] Cisco 7970 - DTMF   Rich Adamson
• [asterisk-users] voicemail greeting   Rich Adamson
• [asterisk-users] TDM2400P vs Sangoma A200   Rich Adamson
• [asterisk-users] Re: Setting QOS settings in asterisk and/or CentOS?   Rich Adamson
• [asterisk-users] How can I stop lost DNS from killing Asterisk?   Rich Adamson
• [asterisk-users] How can I stop lost DNS from killing Asterisk?   Rich Adamson
• [asterisk-users] Line Pickup Problem   Rich Adamson
• [asterisk-users] Asterisk behind Sonicwall firewall   Rich Adamson
• [asterisk-users] Asterisk behind Sonicwall firewall   Rich Adamson
• [asterisk-users] T1 timing errors (Frame Slips) on Nortel 61C to TE110P   Rich Adamson
• [asterisk-users] Right way to prevent analog channel from answering the phone?   Rich Adamson
• [asterisk-users] E1 crossover system   Christoph Adomeit
• [asterisk-users] Asterisk 1.2 and SATA drives   Gabriel Afana
• [asterisk-users] Asterisk Hangups on PRI Interface   Vicente Aguilar
• [asterisk-users] Asterisk Hangups on PRI Interface   Vicente Aguilar
• [asterisk-users] Is asterisk's mgcp support(NAS) Network access server package   Ibrar Ahmed
• [asterisk-users] Articulation Palm client and Asterisk   Jorge Alayon
• [asterisk-users] Re: Cisco 7960G SIP firmware 8.4   Nathan Alberti
• [asterisk-users] Polycom new firmware and bootrom   Nathan Alberti
• [asterisk-users] Asterisk with cisco 7935   Nathan Alberti
• [asterisk-users] Display message on voip phone...hint?   Ale
• [asterisk-users] What does 'trunk' mean in outgoing and incoming?   Larry Alkoff
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Larry Alkoff
• [asterisk-users] Please help route incoming PSTN calls to Asterisk   Larry Alkoff
• [asterisk-users] What really happens between Asterisk and an SPA-3000?   Larry Alkoff
• [asterisk-users] Re: [A*UG] What does 'trunk' mean in outgoing and incoming?   Larry Alkoff
• [asterisk-users] Asterisk with Addpac 2120   Pablo Almido
• [asterisk-users] Calling to PSTN newbie question   Pablo Almido
• [asterisk-users] Setting QOS settings in asterisk and/or CentOS?   Bob Amen
• [asterisk-users] Reading the raw E1 channels ?   Azher Amin
• [asterisk-users] Two phones, same number   Benny Amorsen
• [asterisk-users] Re: Two phones, same number   Benny Amorsen
• [asterisk-users] Re: Two phones, same number   Benny Amorsen
• [asterisk-users] Re: Running Multiple Instances of Asterisk   Benny Amorsen
• [asterisk-users] asterisk and PowerEdge 1950   Ryan Amos
• [asterisk-users] Asterisk with cisco 7935   Ryan Amos
• [asterisk-users] SER with multiple asterisk deployment   Zac Amsler
• [asterisk-users] CURL   Colin Anderson
• [asterisk-users] CURL   Colin Anderson
• [asterisk-users] Display message on voip phone...hint?   Colin Anderson
• [asterisk-users] OT: Anybody remember this from last Dec?   Colin Anderson
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Colin Anderson
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Colin Anderson
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Colin Anderson
• [asterisk-users] PRI Outbound CallerID Question   Colin Anderson
• [asterisk-users] speaker phone echo   Colin Anderson
• [asterisk-users] 64 analog phones   Colin Anderson
• [asterisk-users] 64 analog phones   Colin Anderson
• [asterisk-users] WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long]   Colin Anderson
• [asterisk-users] Forcing Transcode   Colin Anderson
• [Asterisk-Users] Off-hooking Snom hanset doesn't answer incom ing call   Colin Anderson
• [asterisk-users] t1-pri or sip trunk?   Colin Anderson
• [asterisk-users] 480i phone: Is there a trick to registering with * ??   Colin Anderson
• [asterisk-users] FW: 480i phone: Is there a trick to registering with * ?? <--forg ot my sip.conf   Colin Anderson
• [asterisk-users] 480i phone: Is there a trick to registering with *??   Colin Anderson
• [asterisk-users] 480i phone: Is there a trick to registering with *??   Colin Anderson
• [asterisk-users] Semi-OT: SIP or IAX provider in the Boston area?   Erik Anderson
• [asterisk-users] Re: Semi-OT: SIP or IAX provider in the Boston area?   Erik Anderson
• [asterisk-users] SPA-942 Sound Quality   Andres
• [asterisk-users] Roundrobin not working on PRI   Andres
• [asterisk-users] Roundrobin not working on PRI   Andres
• [asterisk-users] Call Processing Slow 11 seconds   Andres
• [asterisk-users] RE: 1.4 Beta 2 Config Problem   Andres
• [asterisk-users] RE: 1.4 Beta 2 Config Problem   Andres
• [asterisk-users] Forcing Transcode   Andres
• [asterisk-users] Polycom MyStat   James Andrewartha
• [asterisk-users] polycom expansion module   James Andrewartha
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Cory Andrews
• [asterisk-users] Any Hardphone with VPNClient embedded?   Cory Andrews
• [asterisk-users] Reliability of the newer IAXy's   Cory Andrews
• [asterisk-users] Asterisk / Audiocodes annoying issue - Seeking Suggestions   Cory Andrews
• [asterisk-users] Cisco 7960 part numbers ...   Cory Andrews
• [asterisk-users] Cisco 7960 part numbers ...   Cory Andrews
• [asterisk-users] Linksys SPA400   Cory Andrews
• [asterisk-users] Asterisk with cisco 7935   Cory Andrews
• [asterisk-users] IAX phones?   Cory Andrews
• [asterisk-users] Zaptel-1.2.9 compile error   Samy Antoun
• [asterisk-users] Zaptel-1.2.9 compile error   Samy Antoun
• [asterisk-users] Zaptel-1.2.9 compile error   Samy Antoun
• [Asterisk-Users] T1 echo canceller   Michael Araba
• [Asterisk-Users] T1 echo canceller   Michael Araba
• [asterisk-users] Using Asterisk with IVR connected with legacy pbxvia rs-232   kritikus Araklidas
• [asterisk-users] Sip configuration using mysql   Arkaitz
• [asterisk-users] Sip configuration using mysql   Arkaitz
• [asterisk-users] Sip configuration using mysql   Arkaitz
• [asterisk-users] Re: Re: FW: Peter Dicks Chairman ofSportingbet PLCisarrested at JFK!!   Julio Arruda
• [asterisk-users] Unknown RTP codec 100 received   Thomas Artner
• [asterisk-users] 64 analog phones   Thomas Artner
• [asterisk-users] When does Scalability requests Asterisk   Jay R. Ashworth
• [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out   Jay R. Ashworth
• [asterisk-users] DTMF Detection Problems with certain phones incoming zap channels   Jay R. Ashworth
• [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out   Jay R. Ashworth
• [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out   Jay R. Ashworth
• [asterisk-users] Pri Event 6 and 8   Jay R. Ashworth
• [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out   Jay R. Ashworth
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Jay R. Ashworth
• [asterisk-users] University dumps CISCO VoIP for Asterisk   Jay R. Ashworth
• [asterisk-users] Incoming calls, identify   Jay R. Ashworth
• [asterisk-users] Audiocodes 124, Mediatrix 1124, or Citel field reports?   Jay R. Ashworth
• [asterisk-users] grandstream gxp 2000 does not display names when calling out   Jay R. Ashworth
• [asterisk-users] TDM2400P   Jay R. Ashworth
• [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Jay R. Ashworth
• [asterisk-users] TDM2400P   Jay R. Ashworth
• [asterisk-users] TDM2400P   Jay R. Ashworth
• [asterisk-users] Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???   Jay R. Ashworth
• [asterisk-users] Audiocodes 124, Mediatrix 1124, or Citel field reports?   Jay R. Ashworth
• [asterisk-users] Leased line interconnect   Jay R. Ashworth
• [asterisk-users] Leased line interconnect   Jay R. Ashworth
• [asterisk-users] Forwarding to a caller-pays mobile   Jay R. Ashworth
• [asterisk-users] Trixbox Documentation   Jay R. Ashworth
• [asterisk-users] Forwarding to a caller-pays mobile   Jay R. Ashworth
• [asterisk-users] Trixbox Documentation   Jay R. Ashworth
• [asterisk-users] Trixbox Documentation   Jay R. Ashworth
• [asterisk-users] Trixbox Documentation   Jay R. Ashworth
• [asterisk-users] Missing sound in spanish from 1.4 beta2   Jay R. Ashworth
• [asterisk-users] spandsp (foip)   Jay R. Ashworth
• [hylafax-users] [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Jay R. Ashworth
• [asterisk-users] spandsp (foip)   Jay R. Ashworth
• [asterisk-users] Need a recommended T38 FOIP solution   Jay R. Ashworth
• [asterisk-users] Missing sound in spanish from 1.4 beta2   Jay R. Ashworth
• [asterisk-users] spandsp (foip)   Jay R. Ashworth
• [asterisk-users] Missing sound in spanish from 1.4 beta2   Jay R. Ashworth
• [asterisk-users] RE: Dual core   Jay R. Ashworth
• [asterisk-users] Re: Very high ping times from 7960 phones   Jay R. Ashworth
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Jay R. Ashworth
• [asterisk-users] Running Multiple Instances of Asterisk   Jay R. Ashworth
• [asterisk-users] TDM2400P vs Sangoma A200   Jay R. Ashworth
• [asterisk-users] Running Multiple Instances of Asterisk   Jay R. Ashworth
• [asterisk-users] PRI Outbound CallerID Question   Jay R. Ashworth
• [asterisk-users] TE406P not working on Intel D101Ggc motherboard.   Jay R. Ashworth
• [asterisk-users] PRI Outbound CallerID Question   Jay R. Ashworth
• [asterisk-users] PRI Outbound CallerID Question   Jay R. Ashworth
• [asterisk-users] I doubt it...   Jay R. Ashworth
• [asterisk-users] PRI Outbound CallerID Question   Jay R. Ashworth
• [asterisk-users] PRI Outbound CallerID Question   Jay R. Ashworth
• [asterisk-users] I doubt it...   Jay R. Ashworth
• [asterisk-users] I doubt it...   Jay R. Ashworth
• [asterisk-users] 64 analog phones   Jay R. Ashworth
• [asterisk-users] I doubt it...   Jay R. Ashworth
• [asterisk-users] 64 analog phones   Jay R. Ashworth
• [asterisk-users] 64 analog phones   Jay R. Ashworth
• [asterisk-users] 64 analog phones   Jay R. Ashworth
• [asterisk-users] Polycom 501 One-way Audio   Jay R. Ashworth
• [asterisk-users] Building the Perfect Box   Jay R. Ashworth
• [asterisk-users] Re: Voip Buster - CID   Jay R. Ashworth
• [asterisk-users] Re: Voip Buster - CID   Jay R. Ashworth
• [asterisk-users] pstn failback   Jay R. Ashworth
• [asterisk-users] Building the Perfect Box   Jay R. Ashworth
• [asterisk-users] Fiber Outage: LA->Washington   Jay R. Ashworth
• [asterisk-users] Dial-9 (was Extension Numbering)   Jay R. Ashworth
• [asterisk-users] Building the Perfect Box   Jay R. Ashworth
• [asterisk-users] Dial-9 (was Extension Numbering)   Jay R. Ashworth
• [asterisk-users] Re: Any way to go from factory reset 7970 to SIP without Call Manager?   Lacy Moore - Aspendora
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Lacy Moore - Aspendora
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Lacy Moore - Aspendora
• [asterisk-users] BLF across asterisk trunks   Lacy Moore - Aspendora
• [asterisk-users] Cisco Distinctive ring using alert-info   Lacy Moore - Aspendora
• [asterisk-users] Cisco 7940 Problem (Mess)   Lacy Moore - Aspendora
• [asterisk-users] How to Dial a number with Sangoma PRI card?   Lacy Moore - Aspendora
• [asterisk-users] pickup call little complicated   Lacy Moore - Aspendora
• [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out   Lacy Moore - Aspendora
• [asterisk-users] Polycom 2.0.1 Software   Lacy Moore - Aspendora
• [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?   Lacy Moore - Aspendora
• [asterisk-users] Call/Voicemail Screening   Lacy Moore - Aspendora
• [asterisk-users] trixbox t38 pass through   Lacy Moore - Aspendora
• [asterisk-users] trixbox t38 pass through   Lacy Moore - Aspendora
• [asterisk-users] "does /var/run/asterisk.ctl exist?" -- but Asterisk *is* running.   Lacy Moore - Aspendora
• [asterisk-users] Line Pickup Problem   Lacy Moore - Aspendora
• [asterisk-users] voicemail greeting   Lacy Moore - Aspendora
• [asterisk-users] voicemail greeting   Lacy Moore - Aspendora
• [asterisk-users] Set hint status from dialplan?   Lacy Moore - Aspendora
• [asterisk-users] Asterisk with cisco 7935   Lacy Moore - Aspendora
• [asterisk-users] Re: Set hint status from dialplan?   Lacy Moore - Aspendora
• [asterisk-users] Re: Set hint status from dialplan?   Lacy Moore - Aspendora
• [asterisk-users] PRI Outbound CallerID Question   Lacy Moore - Aspendora
• [asterisk-users] max number of devices in hint   Lacy Moore - Aspendora
• [asterisk-users] I doubt it...   Lacy Moore - Aspendora
• [asterisk-users] How to change pager notification message   Lacy Moore - Aspendora
• [asterisk-users] How to change pager notification message   Lacy Moore - Aspendora
• [asterisk-users] Asterisk behind Sonicwall firewall   Lacy Moore - Aspendora
• [asterisk-users] Re: max number of devices in hint   Lacy Moore - Aspendora
• [asterisk-users] Re: max number of devices in hint   Lacy Moore - Aspendora
• [asterisk-users] 64 analog phones   Lacy Moore - Aspendora
• [asterisk-users] 7940 vs. 7941   Lacy Moore - Aspendora
• [asterisk-users] Re: max number of devices in hint   Lacy Moore - Aspendora
• [asterisk-users] pstn failback   Lacy Moore - Aspendora
• [asterisk-users] pstn failback   Lacy Moore - Aspendora
• [asterisk-users] Extension Numbering   Lacy Moore - Aspendora
• [asterisk-users] Re: max number of devices in hint   Lacy Moore - Aspendora
• [asterisk-users] Polycom phone references needed   AstATN
• [asterisk-users] Polycom phone help needed   AstATN
• [asterisk-users] Customize host in INVITE's Contact header?   Yoann Aubineau
• [asterisk-users] Asterisk AGI question   Yoann Aubineau
• [asterisk-users] Asteisk plays music on hold starting from random point   Yoann Aubineau
• [asterisk-users] Asterisk & MSN ?   Yoann Aubineau
• [asterisk-users] ASTERISK HIGH AVAILABILITY   Giedrius Augys
• [asterisk-users] Asterisk Fax Server   Giedrius Augys
• [asterisk-users] Problem with "Background" DTMF detection with A200D   Alvin Austin
• [asterisk-users] 0005162: RTP Packetization : Few questions   Dan Austin
• [asterisk-users] 0005162: RTP Packetization : Few questions   Dan Austin
• [asterisk-users] 0005162: RTP Packetization : Few questions   Dan Austin
• [asterisk-users] Application of Asterisk Packetization Patch   Dan Austin
• [asterisk-users] Cisco CAll Manger and H323   Dan Austin
• [asterisk-users] Cisco CAll Manger and H323   Dan Austin
• [asterisk-users] Cisco CAll Manger and H323   Dan Austin
• [asterisk-users] SNMP with 1.2.11 stable   Azfhasterisk
• [asterisk-users] problems with Polycom 500 boot up   Azfhasterisk
• [asterisk-users] Polycom 501 One-way Audio   Azfhasterisk
• [asterisk-users] Cisco MWI   Michiel van Baak
• [asterisk-users] Using option 'r' in queue doesn't announce frequeny etc.   Michiel van Baak
• [asterisk-users] How to notify an ACD agent before he/she picks up   Michiel van Baak
• [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2   Michiel van Baak
• [asterisk-users] grandstream gxp 2000 does not display names when calling out   Michiel van Baak
• [asterisk-users] Channel kept busy when creating ssh tunnel via AGI   Michiel van Baak
• [asterisk-users] enumlookup - deprecated working - but appreciated one duznt :-(   Michiel van Baak
• [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s   Michiel van Baak
• [asterisk-users] Forwarding   Michiel van Baak
• [asterisk-users] Forwarding to a caller-pays mobile   Michiel van Baak
• [asterisk-users] Segmentation fault on Asterisk startup: res_config_mysql.so problem?   Michiel van Baak
• [asterisk-users] Re: Very high ping times from 7960 phones   Michiel van Baak
• [asterisk-users] Re: Very high ping times from 7960 phones   Michiel van Baak
• [asterisk-users] asterisk to cell phone network   Michiel van Baak
• [asterisk-users] Very high ping times from 7960 phones   Michiel van Baak
• [asterisk-users] Re: asterisk to cell phone network   Michiel van Baak
• [asterisk-users] Re: asterisk to cell phone network   Michiel van Baak
• [asterisk-users] Context default & incoming ENUM   Michiel van Baak
• [asterisk-users] Are you using app_meetme or app_conference   Michiel van Baak
• [asterisk-users] Multiple asterisk same GUI   Michiel van Baak
• [asterisk-users] Extension Numbering   Michiel van Baak
• [asterisk-users] Zero length queue   Wes Baehr
• [asterisk-users] Re: Two phones, same number   Wes Baehr
• [asterisk-users] Via Epia platforms and asterisk   Chris Bagnall
• [asterisk-users] sound file length   Time Bandit
• [asterisk-users] Modem calls   Time Bandit
• [asterisk-users] Variable that gives the SIP channel   Time Bandit
• [asterisk-users] Cisco 7940 Problem (Mess)   Time Bandit
• [asterisk-users] Right way to prevent analog channel from answering the phone?   Time Bandit
• [asterisk-users] IAX phones?   Time Bandit
• [asterisk-users] Re: [asterisk-dev] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Remco Barendse
• [asterisk-users] Zaptel-1.2.9 compile error   Remco Barendse
• [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s   Remco Barendse
• [asterisk-users] Kirk IP600 V3 DECT Wireless server   Remco Barendse
• [asterisk-users] mISDN versus ZapHFC with BRIstuff   Remco Barendse
• [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s   Remco Barendse
• [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s   Remco Barendse
• [asterisk-users] fxotune failure! "Could not fill input buffer - got -1 bytes, expected 4000 bytes"   Jonathan Barratt
• [asterisk-users] fxotune failure! "Could not fill input buffer - got -1 bytes, expected 4000 bytes"   Jonathan Barratt
• [asterisk-users] Dropped Calls on TDM400p   Jonathan Barratt
• [asterisk-users] Dropped Calls on TDM400p   Jonathan Barratt
• [asterisk-users] Dropped Calls on TDM400p   Jonathan Barratt
• [asterisk-users] AEL2 patch for Asterisk 1.2.12.1   Barzilai
• [asterisk-users] Line Pickup Problem   Mohsen Basirat
• [asterisk-users] INSTALL_PREFIX=   Forrest Beck
• [asterisk-users] sip.conf for talking to other Asterisk machines   Forrest Beck
• [asterisk-users] PRI Backup   Forrest Beck
• [asterisk-users] Help with Tieing Outbound calls to Zap Channels   Forrest Beck
• [asterisk-users] ATA with wireless client   Forrest Beck
• [asterisk-users] SIP Gateway   Forrest Beck
• [asterisk-users] SIP Gateway   Forrest Beck
• [asterisk-users] Re: SIP Gateway   Forrest Beck
• [asterisk-users] Trixbox Documentation   J. Duffy Beischel
• [asterisk-users] Trixbox Documentation   J. Duffy Beischel
• [asterisk-users] problem with trying to use two extensions for different announcements   J. Duffy Beischel
• [asterisk-users] macro-dialout-trunk without agi or manager   Nathan Bell
• [asterisk-users] dialout-trunk vs. dial group   Nathan Bell
• [asterisk-users] T1 incoming connects, but no sound   Nathan Bell
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Nic Bellamy
• H.264 basic backport (was Re: [asterisk-users] Grandstream and H.264 !)   Nic Bellamy
• [asterisk-users] What I always get asked in SME * deployments   Dovid Bender
• [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC isarrested at JFK!!   Dovid Bender
• [asterisk-users] Re: notransfer local channel on redirect   Christian Benke
• [asterisk-users] notransfer local channel on redirect   Benko
• [asterisk-users] Re: notransfer local channel on redirect   Benko
• [asterisk-users] All circuits are busy now???   BerkHolz, Steven
• [asterisk-users] Setting QOS settings in asterisk and/or CentOS?   BerkHolz, Steven
• [asterisk-users] hint status from dialplan?   BerkHolz, Steven
• [asterisk-users] dialplan applications   Robert Bielik
• [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry   Matt Birmingham
• [asterisk-users] Polycom Soundpoint Key Remap   Matt Birmingham
• [asterisk-users] What I always get asked in SME * deployments   Eric Bishop
• [asterisk-users] Leased line interconnect   Eric Bishop
• [asterisk-users] Leased line interconnect   Eric Bishop
• [asterisk-users] How can I stop lost DNS from killing Asterisk?   Eric Bishop
• [asterisk-users] Cisco GW & CID Name   Steve Blair
• [asterisk-users] Queue - static members   Kevin Bockman
• [asterisk-users] Getting Music On Hold working in * 1.2.12.1 with Fedora?   Kevin Bockman
• [asterisk-users] asterisk - alcatel   Nicolas Bocquet
• [asterisk-users] SIPP problem   Greg Boehnlein
• [asterisk-users] Digium at Ohio Linuxfest   Greg Boehnlein
• [asterisk-users] Remote tone access   Tony Di Bona
• [asterisk-users] How to download asterisk 1.3 development version   Boneym
• [asterisk-users] Any Hardphone with VPNClient embedded?   Leo Ann Boon
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   Leo Ann Boon
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   Leo Ann Boon
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   Leo Ann Boon
• [asterisk-users] Uninstalling Trixbox   Leo Ann Boon
• [asterisk-users] Running Multiple Instances of Asterisk   Leo Ann Boon
• [asterisk-users] E1 crossover system   Leo Ann Boon
• [asterisk-users] Sip answer one side , ring other side   Leo Ann Boon
• [asterisk-users] Realtime; User not registering {Urgent}   Leo Ann Boon
• [asterisk-users] File structure question   Peter Bowyer
• [asterisk-users] File structure question   Peter Bowyer
• [asterisk-users] File structure question   Peter Bowyer
• [asterisk-users] Probelm with incoming calls to my DID-Please help me   Crazy Boy
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Crazy Boy
• [asterisk-users] Probelm with incoming calls to my DID-Please help me   Crazy Boy
• [asterisk-users] Incoming call problem-calling part is busy(IPKall)   Crazy Boy
• [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)   Crazy Boy
• [asterisk-users] Accounts registered, but call is not going   Crazy Boy
• [asterisk-users] How to configure Fritz ISDN2 card with Trixbox?   Crazy Boy
• [asterisk-users] Re: Calling Card and Billing   Crazy Boy
• [asterisk-users] HFC isdn card and bristuff 0.2.0 rc8n   Crazy Boy
• [asterisk-users] Re: chan_zap.so stopped working after upgrading CentOS   Crazy Boy
• [asterisk-users] A Strange doubt and problem   Crazy Boy
• [asterisk-users] SPA941 -> Asterisk -> Voip provider -> PSTN -> ShoreTel garble   Cliff Brake
• [asterisk-users] SPA941 -> Asterisk -> Voip provider -> PSTN -> ShoreTel garble   Cliff Brake
• [asterisk-users] INVITE re-try interval   David Brazier
• [asterisk-users] Internal message being heard on pstn line   Ryder Brook
• [asterisk-users] modifying the INVITE headers   Paco Brufal
• [asterisk-users] modifying the INVITE headers   Paco Brufal
• [asterisk-users] 64 analog phones   Paco Brufal
• [asterisk-users] quadbri + tdm400p + modem-fax   Paco Brufal
• [asterisk-users] quadbri + tdm400p + modem-fax   Paco Brufal
• [asterisk-users] Ericsson MD110   Patricio A. Bruna
• [asterisk-users] Dell hardware ...   Alan Bunch
• [asterisk-users] Xorcom Astribank   Nick Burch
• [asterisk-users] Integrating Asterisk with LDAP Realtime   Nick Burch
• [asterisk-users] Re: [asterisk‑users] Integrating Asterisk with LDAP Realtime   Nick Burch
• [asterisk-users] When does Scalability requests Asterisk to Use SER ?   Ryan Burke
• [asterisk-users] Cisco ATA escaping # into %23?   Scott Call
• [asterisk-users] Correct settings for UK (BT) FXO   Brian Candler
• [asterisk-users] 9 becomes 99 ? And other strangeness   Brian Candler
• [asterisk-users] Getting 'i' functionality on internal extensions   Brian Candler
• [asterisk-users] 9 becomes 99 ? And other strangeness   Brian Candler
• [asterisk-users] 9 becomes 99 ? And other strangeness   Brian Candler
• [asterisk-users] Getting 'i' functionality on internal extensions   Brian Candler
• [asterisk-users] 9 becomes 99 ? And other strangeness   Brian Candler
• [asterisk-users] 9 becomes 99 ? And other strangeness   Brian Candler
• [asterisk-users] Getting 'i' functionality on internal extensions   Brian Candler
• [asterisk-users] How to download asterisk 1.3 development version   Brian Candler
• [asterisk-users] 9 becomes 99 ? And other strangeness   Brian Candler
• [asterisk-users] Cisco 7940 Problem (Mess)   Brian Candler
• [asterisk-users] Alcatel OXO Sip   Brian Candler
• [asterisk-users] Alcatel OXO Sip   Brian Candler
• [asterisk-users] DSL router with integrated SIP proxy?   Brian Candler
• [asterisk-users] ATA with wireless client   Brian Candler
• [asterisk-users] extensions.conf strangeness   Brian Candler
• [asterisk-users] extensions.conf strangeness   Brian Candler
• [asterisk-users] Difference between "SIP Server" and "Outbound Proxy"   Brian Candler
• [asterisk-users] extensions.conf strangeness   Brian Candler
• [asterisk-users] Digium GUI?   Brian Capouch
• [asterisk-users] (no subject)   Brian Capouch
• [asterisk-users] RE: 1.4 Beta 2 Config Problem   Brian Capouch
• [asterisk-users] Re: Very high ping times from 7960 phones   Brian Capouch
• [asterisk-users] SER with multiple asterisk deployment   Brian Capouch
• [asterisk-users] New ptlib dependency-requirement in SVN-trunk?   Brian Capouch
• [asterisk-users] Asterisk Voicemail with Sonus?   Chris Carey
• [asterisk-users] One server SUBSCRIBE for information on multiple voicemail boxes   Chris Carey
• [asterisk-users] Asterisk and peers behind nat with port forward, how to proxy?   Marcus Carlson
• [asterisk-users] 911 versus 9.911   Strom Carlson
• [asterisk-users] How to play a sound on a periodic basis during a call?   Dallas Carter
• [asterisk-users] asterisk t.38 fax failed   Ricardo Carvalho
• [asterisk-users] asterisk t.38 fax failed   Ricardo Carvalho
• [asterisk-users] distinguishing users by their domain   Ricardo Carvalho
• [asterisk-users] distinguishing users by their domain   Ricardo Carvalho
• [asterisk-users] Can Asterisk bind on multiple ports?   Ricardo Carvalho
• [asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right   Ricardo Carvalho
• [asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right   Ricardo Carvalho
• [asterisk-users] Grandstream Budgetone phones don't show   Ricardo Carvalho
• [asterisk-users] Grandstream Budgetone phones don't show   Ricardo Carvalho
• [asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right   Ricardo Carvalho
• [asterisk-users] Thomson 2030   Ricardo Carvalho
• [asterisk-users] Re: Can you explain why multiple registration is an important (missing) feature ?   Ricardo Carvalho
• [asterisk-users] Voicemail adjustments   Ricardo Carvalho
• [asterisk-users] Does Asterisk 1.4 going to support realtime ex-girlfriend logic?   Ricardo Carvalho
• [asterisk-users] Getting 'i' functionality on internal extensions   Anthony Cennami
• [asterisk-users] urgently requires help regarding MWI   Anthony Cennami
• [asterisk-users] Problem with # locking up call   Anthony Cennami
• [asterisk-users] Problem with # locking up call   Anthony Cennami
• [asterisk-users] Problem with # locking up call   Anthony Cennami
• [asterisk-users] Call is dead after featuredigittimeout   Anthony Cennami
• [asterisk-users] Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.209   Anthony Cennami
• [asterisk-users] TDM2400P vs Sangoma A200   Anthony Cennami
• [asterisk-users] Does a "HST Saphir III ML PCI" work with Asterisk?   Patrick Cervicek
• [asterisk-users] Asterisk Server Down   Patrick Cervicek
• [asterisk-users] Wrong outgoing port   Patrick Cervicek
• [asterisk-users] Cisco 7960 part numbers ...   Cesc
• [asterisk-users] asterisk on 2.4 kernel ... scheduler problem?   Cesc
• [asterisk-users] asterisk on 2.4 kernel ... scheduler problem?   Cesc
• [asterisk-users] Format_MP3, Streaming, File Formats, MOH   Robert Chadwell
• [asterisk-users] Asteisk plays music on hold starting from randompoint   Robert Chadwell
• [asterisk-users] Format_MP3, Streaming, File Formats, MOH   Robert Chadwell
• [asterisk-users] Format_MP3, Streaming, File Formats, MOH   Robert Chadwell
• [asterisk-users] Asteisk plays music on hold startingfrom randompoint   Robert Chadwell
• [asterisk-users] Format_MP3, Streaming, File Formats, MOH   Robert Chadwell
• [asterisk-users] UK (BT) Problem with TDM 400P   Nick Chalk
• [asterisk-users] real time billing system   Chapeti
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Carlos Chavez
• [asterisk-users] Grandstream and H.264 !   Carlos Chavez
• [asterisk-users] Linksys SPA400   Carlos Chavez
• [asterisk-users] Asterisk and Panasonic D500   Carlos Chavez
• [asterisk-users] Dynamic DNS asterisk server?   Carlos Chavez
• [asterisk-users] 64 analog phones   Carlos Chavez
• [asterisk-users] Missing sound in spanish from 1.4 beta2   Carlos Chavez
• [asterisk-users] Missing sound in spanish from 1.4 beta2   Carlos Chavez
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Carlos Chavez
• [asterisk-users] Context   Khaled Chehab
• [asterisk-users] Callback   Khaled Chehab
• [asterisk-users] Callback   Khaled Chehab
• [asterisk-users] Call back   Khaled Chehab
• [asterisk-users] Call back   Khaled Chehab
• [asterisk-users] TDM04B Installation Problem   Ian Chilton
• [asterisk-users] Sipura SPA3000   Bob Chiodini
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Bob Chiodini
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Bob Chiodini
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Bob Chiodini
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Bob Chiodini
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Bob Chiodini
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Bob Chiodini
• [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?   Bob Chiodini
• [asterisk-users] Asterisk and NAT ?   Bob Chiodini
• [asterisk-users] DID not getting passed?   Bob Chiodini
• [asterisk-users] Re: Asterisk hangs up after 10-15 minutes when SIPPhone is on mute   Dr. Michael J. Chudobiak
• [asterisk-users] Dropped call question - "Maximum retries exceeded on transmission"   Dr. Michael J. Chudobiak
• [asterisk-users] Asterisk 1.2 & snom 360 MWI   Dr. Michael J. Chudobiak
• [asterisk-users] Asterisk behind Sonicwall firewall   Dr. Michael J. Chudobiak
• [asterisk-users] phpagi syntax and SendDTMF   Frank Church
• [asterisk-users] uConnect Voip device   Frank Church
• [asterisk-users] Re: uConnect Voip device   Frank Church
• [asterisk-users] Re: uConnect Voip device   Frank Church
• [asterisk-users] How to send DTMF down a channel   Frank Church
• [asterisk-users] How to send DTMF down a channel   Frank Church
• [asterisk-users] How to send DTMF down a channel   Frank Church
• [asterisk-users] How to send DTMF down a channel   Frank Church
• [asterisk-users] Please help with a telular mod. SX5e   Jorge Cisneros
• [asterisk-users] Cannot hear the other side of the phone call   Dennis P. Clark
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Mike Clark
• [asterisk-users] disconnect code in featuremap doesn't work on unanswered calls   Stanley Cline
• [asterisk-users] Looped message playback   Earle Clubb
• [asterisk-users] Looped message playback   Earle Clubb
• [asterisk-users] Looped message playback   Earle Clubb
• [asterisk-users] Looped message playback   Earle Clubb
• [asterisk-users] Looped message playback   Earle Clubb
• [asterisk-users] Looped message playback   Earle Clubb
• [asterisk-users] Looped message playback   Earle Clubb
• [asterisk-users] g729 failover when out of licenses   Henry J. Cobb
• [asterisk-users] IAX phone recommandation   richard Coco
• [asterisk-users] unable to change the emailbody for email notification   richard Coco
• [asterisk-users] Asterisk Bussiness Edition and Realtime.   Aquiles Cohen
• [asterisk-users] Asterisk speaks Italian!   Tzafrir Cohen
• [asterisk-users] Asterisk speaks Italian!   Tzafrir Cohen
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Tzafrir Cohen
• [asterisk-users] "best" BRI card ?   Tzafrir Cohen
• [asterisk-users] Asterisk speaks Italian!   Tzafrir Cohen
• [asterisk-users] Help compiling asterisk-addons on Debian?   Tzafrir Cohen
• [asterisk-users] Asterisk server crashes after two years   Tzafrir Cohen
• [asterisk-users] Problems compil 1.2.11   Tzafrir Cohen
• [asterisk-users] Problems compil 1.2.11   Tzafrir Cohen
• [asterisk-users] File structure question   Tzafrir Cohen
• [asterisk-users] UK (BT) Problem with TDM 400P   Tzafrir Cohen
• [asterisk-users] Unable to open pseudo channel for timing... Sound may be choppy.   Tzafrir Cohen
• [asterisk-users] SIPP problem   Tzafrir Cohen
• [asterisk-users] Zaptel-1.2.8 compile problem   Tzafrir Cohen
• [asterisk-users] FAX handling   Tzafrir Cohen
• [asterisk-users] missing pri connect (wwomera to pri)   Tzafrir Cohen
• [asterisk-users] FAX handling   Tzafrir Cohen
• [asterisk-users] Catch an event   Tzafrir Cohen
• [asterisk-users] bristuff compile problems with kernel 2.6.17.11   Tzafrir Cohen
• [asterisk-users] Re: bristuff compile problems with kernel 2.6.17.11   Tzafrir Cohen
• [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?   Tzafrir Cohen
• [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990   Tzafrir Cohen
• [asterisk-users] Digits are played in english in french voicemail   Tzafrir Cohen
• [asterisk-users] Little help for a newbie configuring a TDM13B - ztcfg fails on channel 4   Tzafrir Cohen
• [asterisk-users] su - postgres -bash-3.00$   Tzafrir Cohen
• [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)   Tzafrir Cohen
• [asterisk-users] Re: RE : Re: [asterisk-dev] Forwarding sip requests from none local domains   Tzafrir Cohen
• [asterisk-users] Setting system time via Asterisk   Tzafrir Cohen
• product annocements [was: Re: [asterisk-users] Finally a VoIP GSM Gateway single port..]   Tzafrir Cohen
• [asterisk-users] call waiting   Tzafrir Cohen
• [asterisk-users] Building Zaptel 1.2.9 with Octasic   Tzafrir Cohen
• [asterisk-users] [asterisk-dev] 491 request pending   Tzafrir Cohen
• [asterisk-users] two safe_asterisk processes on the same PBX???   Tzafrir Cohen
• [asterisk-users] voicemail access thru apache on another server   Tzafrir Cohen
• [asterisk-users] two safe_asterisk processes on the same PBX???   Tzafrir Cohen
• [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic   Tzafrir Cohen
• [asterisk-users] ZT_SPANCONFIG failed on span 1: No such device or address (6)   Tzafrir Cohen
• [asterisk-users] read variable from shell script   Tzafrir Cohen
• [asterisk-users] system cmd   Tzafrir Cohen
• [asterisk-users] Asterisk Server Down   Tzafrir Cohen
• [asterisk-users] Asterisk Server Down   Tzafrir Cohen
• [asterisk-users] Xorcom Astribank   Tzafrir Cohen
• [asterisk-users] A1200+fxo, anyone using this?   Tzafrir Cohen
• [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic   Tzafrir Cohen
• [asterisk-users] Xorcom Astribank   Tzafrir Cohen
• [asterisk-users] Digium GUI?   Tzafrir Cohen
• [asterisk-users] Enabling Second Processor Trashes Audio Quality   Tzafrir Cohen
• [asterisk-users] Digium GUI?   Tzafrir Cohen
• [asterisk-users] Uninstalling Trixbox   Tzafrir Cohen
• [asterisk-users] mpg123   Tzafrir Cohen
• [asterisk-users] Uninstalling Trixbox   Tzafrir Cohen
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Tzafrir Cohen
• [asterisk-users] two safe_asterisk processes on the same PBX???   Tzafrir Cohen
• [asterisk-users] Asteisk plays music on hold starting from random point   Tzafrir Cohen
• [asterisk-users] Asteisk plays music on hold starting from random point   Tzafrir Cohen
• [asterisk-users] CURL   Tzafrir Cohen
• [asterisk-users] CURL   Tzafrir Cohen
• [asterisk-users] TDM2400 wired description and skiping frames   Tzafrir Cohen
• [asterisk-users] multiple zaptel cards   Tzafrir Cohen
• [asterisk-users] Fax Detection on outbound call   Tzafrir Cohen
• [asterisk-users] Iax Netstat Output   Tzafrir Cohen
• [asterisk-users] Iax Netstat Output   Tzafrir Cohen
• [asterisk-users] alternatives to mpg123: format_mp3, rawplayer or madplay?   Tzafrir Cohen
• [asterisk-users] dialout-trunk vs. dial group   Tzafrir Cohen
• [asterisk-users] Problem with zaptel 1.4b2 and X101P Wildcard   Tzafrir Cohen
• [asterisk-users] Problem with zaptel 1.4b2 and X101P Wildcard   Tzafrir Cohen
• [asterisk-users] ztcfg / X100P question   Tzafrir Cohen
• [asterisk-users] ztcfg / X100P question   Tzafrir Cohen
• [asterisk-users] Running Multiple Instances of Asterisk   Tzafrir Cohen
• [asterisk-users] Anybody have the opvx1200.c driver?   Tzafrir Cohen
• [asterisk-users] asterisk on 2.4 kernel ... scheduler problem?   Tzafrir Cohen
• [asterisk-users] WAS: 64 analog phones NOW: Selection criteria and recipie for a good Asterisk install [long]   Tzafrir Cohen
• [asterisk-users] Replacing mpg123 with madplay under Solaris?   Tzafrir Cohen
• [asterisk-users] How do I reset a password?   Tzafrir Cohen
• [asterisk-users] Dial-9 (was Extension Numbering)   Tzafrir Cohen
• [asterisk-users] Asterisk speaks Italian!   Dean Collins
• [asterisk-users] Asterisk speaks Italian!   Dean Collins
• [asterisk-users] Nokia N80   Dean Collins
• [asterisk-users] faktortel   Dean Collins
• [asterisk-users] netmask   Dean Collins
• [asterisk-users] netmask   Dean Collins
• [asterisk-users] Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute   Dean Collins
• [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC is arrested at JFK!!   Dean Collins
• [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC is arrested at JFK!!   Dean Collins
• [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC isarrested at JFK!!   Dean Collins
• [asterisk-users] Re: FW: Peter Dicks Chairman of SportingbetPLCisarrested at JFK!!   Dean Collins
• [asterisk-users] FW: Peter Dicks Chairman ofSportingbet PLC is arrested at JFK!!   Dean Collins
• [asterisk-users] RE: Peter Dicks Chairman ofSportingbet PLC is arrested at JFK!!   Dean Collins
• [asterisk-users] Re: FW: Peter Dicks Chairman ofSportingbet PLCisarrested at JFK!!   Dean Collins
• [BULK] Re: [asterisk-users] Prompts recording for Asterisk   Dean Collins
• [asterisk-users] RE: [asterisk-biz] Come see us at VON   Dean Collins
• [asterisk-users] "Ping" a phone   Dean Collins
• [asterisk-users] RE: [Asterisk-video] VXIasterisk is available !   Dean Collins
• [asterisk-users] problems with Polycom 500 boot up   Dean Collins
• [asterisk-users] problems with Polycom 500 boot up   Dean Collins
• [asterisk-users] problems with Polycom 500 boot up   Dean Collins
• [asterisk-users] Polycom 650 Question   Dean Collins
• [asterisk-users] IAX or SIP termination provider that reaches6421xxxxxxx?   Dean Collins
• [asterisk-users] Forwarding   Dean Collins
• [asterisk-users] Trixbox Documentation   Dean Collins
• [asterisk-users] lumenvox speech recognition   Dean Collins
• [asterisk-users] Any Suggestions for Election Polling Application?   Dean Collins
• [asterisk-users] Tracking the source of a disconnect?   Jamin W. Collins
• [asterisk-users] Re: Tracking the source of a disconnect?   Jamin W. Collins
• [asterisk-users] Re: Tracking the source of a disconnect?   Jamin W. Collins
• [asterisk-users] Problems Unpacking tarball For Asterisk Application   Jamin W. Collins
• [asterisk-users] Tracking the source of a disconnect?   Jamin W. Collins
• [asterisk-users] Tracking the source of a disconnect?   Jamin W. Collins
• [asterisk-users] Tracking the source of a disconnect?   Jamin W. Collins
• [asterisk-users] Tracking the source of a disconnect? - SOLVED   Jamin W. Collins
• [asterisk-users] Tracking the source of a disconnect? - SOLVED   Jamin W. Collins
• [asterisk-users] VMware and Digium TDM400P card   Jamin W. Collins
• [asterisk-users] Sipura SPA3000   Kevin Collins
• [asterisk-users] Asterisk speaks Italian!   Michael Collins
• [asterisk-users] Re: Detect PBX vs Network message   Michael Collins
• [asterisk-users] Priority "n"   Michael Collins
• [asterisk-users] Chanspy crashing the server, again   Joshua Colp
• [asterisk-users] Polycom 501 display bug   William M Conlon
• [asterisk-users] OT -- echo cancellation of an audio file   William M Conlon
• [asterisk-users] g729 failover when out of licenses   Tod Detre (CampusEAI Consortium)
• [asterisk-users] Variable that gives the SIP channel   Andre Courchesne - Consultant
• [asterisk-users] MOH problem   Andre Courchesne - Consultant
• [asterisk-users] PRI channel becomming unavailable   Andre Courchesne - Consultant
• [asterisk-users] You are not the next caller   Sean Cook
• [asterisk-users] Voip providers and sip origination and termination?   Christopher Corn
• [asterisk-users] Voip providers and sip origination and termination?   Christopher Corn
• [asterisk-users] can someone recommend a voip provider that...   Christopher Corn
• [asterisk-users] Voip providers and sip origination and termination?   Christopher Corn
• [asterisk-users] can someone recommend a voip provider that...   Christopher Corn
• [asterisk-users] using residential voip for business?   Christopher Corn
• [asterisk-users] using residential voip for business?   Christopher Corn
• [asterisk-users] using residential voip for business?   Christopher Corn
• [asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right   Christopher Corn
• [asterisk-users] experience with axvoice.com?   Christopher Corn
• [asterisk-users] question...   Christopher Corn
• [asterisk-users] question...   Christopher Corn
• [asterisk-users] question...   Christopher Corn
• [asterisk-users] question...   Christopher Corn
• [asterisk-users] DID not getting passed?   Christopher Corn
• [asterisk-users] sip origination and termination   Christopher Corn
• [asterisk-users] call waiting   Christopher Corn
• [asterisk-users] call waiting   Christopher Corn
• [asterisk-users] call waiting   Christopher Corn
• [asterisk-users] success   Christopher Corn
• [asterisk-users] voxee, callerid and trixbox   Christopher Corn
• [asterisk-users] grandstream gxp 2000 does not display names when calling out   Christopher Corn
• [asterisk-users] grandstream gxp 2000 does not display names when calling out   Christopher Corn
• [asterisk-users] grandstream gxp 2000 does not display names when calling out   Christopher Corn
• [asterisk-users] grandstream gxp 2000 does not display names whencalling out   Christopher Corn
• [asterisk-users] grandstream gxp 2000 does not display names when calling out   Christopher Corn
• [asterisk-users] Configuring Codecs   Christopher Corn
• [asterisk-users] fax over ip   Christopher Corn
• [asterisk-users] e911   Christopher Corn
• [asterisk-users] spandsp (foip)   Christopher Corn
• [asterisk-users] spandsp (foip)   Christopher Corn
• [asterisk-users] Need a recommended T38 FOIP solution   Christopher Corn
• [asterisk-users] Need a recommended T38 FOIP solution   Christopher Corn
• [asterisk-users] spandsp (foip)   Christopher Corn
• [asterisk-users] can someone recommened a reliable, cheap t38 origination/termination provider   Christopher Corn
• [asterisk-users] trixbox t38 pass through   Christopher Corn
• [asterisk-users] speaker phone echo   Christopher Corn
• [asterisk-users] blf aastra 9133i working but can't pickup calls   Dave Cotton
• [asterisk-users] Whcih phones are better for mass deployment   Dave Cotton
• [asterisk-users] RE : Re: [asterisk-dev] Forwarding sip requests from none localdomains   Dave Cotton
• [asterisk-users] Changes in extensions.conf handling between 1.2 & 1.4   Dave Cotton
• [asterisk-users] Aastra 9133i and Atcom AT-320 - Comments please   Dave Cotton
• [asterisk-users] HINT problems with SVN-trunk-r43322   Dave Cotton
• [asterisk-users] 480i phone: Is there a trick to registering with *??   Dave Cotton
• [asterisk-users] 480i phone: Is there a trick to registering with *??   Dave Cotton
• [asterisk-users] Integrating Asterisk with LDAP Realtime   Nick Couchman
• [asterisk-users] Re: [asterisk‑users] Integrating Asterisk with LDAP Realtime   Nick Couchman
• [asterisk-users] Integrating Asterisk with LDAP Realtime   Nick Couchman
• [asterisk-users] Integrating Asterisk with LDAP Realtime   Nick Couchman