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
• [asterisk-users] Verizon ISDN service in NY & Hunt Groups   Bernie Courtney
• [asterisk-users] Any way to go from factory reset 7970 to SIP without Call Manager?   Matthew Crocker
• [asterisk-users] 911 versus 9.911   Matthew Crocker
• [Asterisk-Users] T1 echo canceller   Matthew Crocker
• [asterisk-users] Asterisk -> Tekelec T6000 (Vocaldata, voiss)   Matthew Crocker
• [asterisk-users] Asterisk -> Tekelec T6000 (Vocaldata, voiss)   Matthew Crocker
• [asterisk-users] SIPP problem   Diego Quintana Cruz
• [asterisk-users] SIPP problem   Diego Quintana Cruz
• [asterisk-users] SIPP problem   Diego Quintana Cruz
• [asterisk-users] INX (internationalnumber.com) Outgoing problem   Daniel Cyt
• [asterisk-users] (Off-topic) Voip number "tracert"   Daniel Cyt
• [asterisk-users] Polycom SoundPoint 2.0.1 SIP firmware?   Ken D'Ambrosio
• [asterisk-users] "does /var/run/asterisk.ctl exist?" -- but Asterisk *is* running.   Ken D'Ambrosio
• [asterisk-users] IAX phones?   Ken D'Ambrosio
• [asterisk-users] voicemailmain errors on CLI   Sergio R. D'Ippolito
• [asterisk-users] Login user   Sergio R. D'Ippolito
• [asterisk-users] RPID   DANIEL, AARON MATTHEW
• [asterisk-users] Fax with asterisk?   DRi at b-w-computer.de
• [asterisk-users] problem installing func_odbc on asterisk 1.2 ...   Vince Daneff
• [asterisk-users] Cisco 7960 won't download dialplan.xml   Aaron Daniel
• [asterisk-users] Cisco MWI   Aaron Daniel
• [asterisk-users] Response to KP Flemming...   Aaron Daniel
• [asterisk-users] SHSU asterisk installation?   Aaron Daniel
• [asterisk-users] Fedora   Aaron Daniel
• [asterisk-users] University dumps CISCO VoIP for Asterisk   Aaron Daniel
• [asterisk-users] University dumps CISCO VoIP for Asterisk   Aaron Daniel
• [asterisk-users] RPID   Aaron Daniel
• [asterisk-users] sangoma A104d echo canceller and fax   Klaus Darilion
• [Asterisk-Users] Sirrix BRI errors   Klaus Darilion
• [asterisk-users] Xorcom Astribank   Klaus Darilion
• [asterisk-users] fast SIP failover (outgoing sIP requests) with 1.2   Klaus Darilion
• [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2   Klaus Darilion
• [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2   Klaus Darilion
• [asterisk-users] asterisk / chan_capi problems   Klaus Darilion
• [asterisk-users] asterisk / chan_capi problems   Klaus Darilion
• [asterisk-users] asterisk / chan_capi problems   Klaus Darilion
• [asterisk-users] PRI Problems   Klaus Darilion
• [asterisk-users] progress problems from SIP to PRI   Klaus Darilion
• [asterisk-users] progress problems from SIP to PRI   Klaus Darilion
• [asterisk-users] core dump with 1.2.7.1 and chan-capi-cm 0.6.5   Klaus Darilion
• [asterisk-users] Cannot make outbound calls to some numbers with PRI line from legacy PBX thru Asterisk with Sangoma   Klaus Darilion
• [asterisk-users] New ptlib dependency-requirement in SVN-trunk?   Klaus Darilion
• [asterisk-users] Sirrix to Legacy PBX   Klaus Darilion
• [Asterisk-Users] Off-hooking Snom hanset doesn't answer incoming call   Klaus Darilion
• [asterisk-users] PRI: sometimes Asterisk drop calls   Steve Davies
• [asterisk-users] PRI: sometimes Asterisk drop calls   Steve Davies
• [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)   Steve Davies
• [asterisk-users] WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner.   Steve Davies
• [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)   Steve Davies
• [asterisk-users] WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner.   Steve Davies
• [Asterisk-Users] Suggestion for directed pickup in bristuffed 1.2 Asterisk   Steve Davies
• [asterisk-users] PRI: sometimes Asterisk drop calls   Steve Davies
• [asterisk-users] PRI: sometimes Asterisk drop calls   Steve Davies
• [asterisk-users] Re: PRI: sometimes Asterisk drop calls   Steve Davies
• [asterisk-users] rxfax, spandsp and lack of ecm   Steve Davies
• [asterisk-users] rxfax, spandsp and lack of ecm   Steve Davies
• [asterisk-users] rxfax, spandsp and lack of ecm   Steve Davies
• [asterisk-users] progress problems from SIP to PRI   Steve Davies
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Steve Davies
• [asterisk-users] ASTTAPI   Steve Davies
• [asterisk-users] g729 and polycoms problem   Delca
• [asterisk-users] g729 and polycoms problem   Delca
• [asterisk-users] g729 and polycoms problem   Delca
• [asterisk-users] g729 and polycoms problem   Delca
• [asterisk-users] Dual asterisk, CallerID(name|number) problem   Delca
• [asterisk-users] How does SIP work?   Vincent Delporte
• [asterisk-users] VMware and Digium TDM400P card   Joe Dennick
• [asterisk-users] Uninstalling Trixbox   Mike Dent
• [asterisk-users] Uninstalling Trixbox   Mike Dent
• [asterisk-users] Snom MWI not turning off when message picked up.   Mike Dent
• [asterisk-users] Re: Snom MWI not turning off when message picked up.   Mike Dent
• [asterisk-users] asterisk to cell phone network   Mike Dent
• [asterisk-users] UK Colocation services   Mike Dent
• [asterisk-users] Dynamic DNS asterisk server?   Austin Denyer
• [asterisk-users] Looking to hire somebody to setup a SER load balancer   Deon
• [asterisk-users] Asterisk and peers behind nat with port forward, how to proxy?   Raul Dias
• [asterisk-users] Asterisk and peers behind nat with port forward, how to proxy?   Raul Dias
• [asterisk-users] REQUERIMIENTOS TE110P Y PANASONIC TDA620   DiegoF
• [asterisk-users] PBX TDA620 AND TE110P   DiegoF
• [asterisk-users] señalizacion te110p, signaling te110p   DiegoF
• [asterisk-users] Rewriting CID number w/o changing CDR src field   Mike Diehl
• [asterisk-users] How to change pager notification message   Mike Diehl
• [asterisk-users] How to change pager notification message   Mike Diehl
• [asterisk-users] sip peer question   Dijkstra, Roelof
• [asterisk-users] Digits are played in english in french voicemail   Dijkstra, Roelof
• [asterisk-users] Zork & Asterisk; zoip 0.2.0 released   Simon P. Ditner
• [asterisk-users] DTMF Tone Not Passing Help   Nitesh Divecha
• [asterisk-users] trixbox t38 pass through   Don
• [asterisk-users] Polycom new firmware and bootrom   Chris Dos
• [asterisk-users] Polycom new firmware and bootrom   Chris Dos
• [asterisk-users] Polycom new firmware and bootrom   Chris Dos
• [asterisk-users] Setting QOS settings in asterisk and/or CentOS?   Redouane Doumer
• [asterisk-users] Polycom 501 One-way Audio   Redouane Doumer
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Cristian Draghici
• [asterisk-users] Any Suggestions for Election Polling Application?   Gerald Drouillard
• [asterisk-users] Polycom 501 One-way Audio   Paul Dugas
• [asterisk-users] Polycom 501 One-way Audio   Paul Dugas
• [asterisk-users] Polycom 501 One-way Audio   Paul Dugas
• [asterisk-users] Queue failover and wrap time   Michelle Dupuis
• [asterisk-users] Linking Asterisk with PBX through E1   Marlon Dutra
• [asterisk-users] Forward recorded voicemail message to more than one extension using sendvoicemail=yes   Jay Dutt
• [asterisk-users] Call forwarding   Vladimir Dvorak
• [asterisk-users] Aastra 9133i and Atcom AT-320 - Comments please   James Dyer
• [asterisk-users] Asterisk and SIP Redirect message   Johansson Olle E
• [asterisk-users] Setting system time via Asterisk   Gary Eck
• [asterisk-users] Contacts for Chan_gsm_bt maintainer?   Ed
• [asterisk-users] te110p and te205p behavioural differences   Mark Edwards
• [asterisk-users] Intel 945G and Digium TE110P compatibility issue   Mark Edwards
• [asterisk-users] Fax Detection on outbound call   Mark Edwards
• [asterisk-users] TDM2400P   Mark Edwards
• [asterisk-users] Fax Detection on outbound call   Mark Edwards
• [asterisk-users] Fax Detection on outbound call   Mark Edwards
• [asterisk-users] app_conference not working for me   Steve Edwards
• [asterisk-users] Asterisk -> Tekelec T6000 (Vocaldata, voiss)   Steve Edwards
• [asterisk-users] Hrmm.. OK, what am I missing? sendmail: Cannot open mail:25   Bernhard Egger
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] Blind transfer 3/4 digits   Nick Ellson
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] Asterisk calling through FWD?   Nick Ellson
• [asterisk-users] Asterisk calling through FWD?   Nick Ellson
• [asterisk-users] Asterisk calling through FWD?   Nick Ellson
• [asterisk-users] Asterisk calling through FWD?   Nick Ellson
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] music onhold choppy music problems   Nick Ellson
• [asterisk-users] Hrmm.. OK, what am I missing? sendmail: Cannot open mail:25   Nick Ellson
• [asterisk-users] Hrmm.. OK, what am I missing? sendmail: Cannot open mail:25   Nick Ellson
• [asterisk-users] Features.. phone vs. asterisk?   Nick Ellson
• [asterisk-users] saved.gsm -> Voicemail greeting ??   Nick Ellson
• [asterisk-users] saved.gsm -> Voicemail greeting ??   Nick Ellson
• [asterisk-users] saved.gsm -> Voicemail greeting ??   Nick Ellson
• [asterisk-users] saved.gsm -> Voicemail greeting ??   Nick Ellson
• [asterisk-users] saved.gsm -> Voicemail greeting ??   Nick Ellson
• [asterisk-users] saved.gsm -> Voicemail greeting ??   Nick Ellson
• [asterisk-users] A1200+fxo, anyone using this?   Nick Ellson
• [asterisk-users] A1200+fxo, anyone using this?   Nick Ellson
• [asterisk-users] Forwarding   Nick Ellson
• [asterisk-users] Dual core   Nick Ellson
• [asterisk-users] Anybody have the opvx1200.c driver?   Nick Ellson
• (GOT IT) Re: [asterisk-users] Anybody have the opvx1200.c driver?   Nick Ellson
• [asterisk-users] Right way to prevent analog channel from answering the phone?   Nick Ellson
• [asterisk-users] Right way to prevent analog channel from answering the phone?   Nick Ellson
• [asterisk-users] Right way to prevent analog channel from answering the phone?   Nick Ellson
• [asterisk-users] X100M location in circuit requirement?   Nick Ellson
• [asterisk-users] Asterisk lockup and boot error after one day?   Nick Ellson
• [asterisk-users] Running Multiple Instances of Asterisk   Alex Epshteyn
• [asterisk-users] Running Multiple Instances of Asterisk   Alex Epshteyn
• [asterisk-users] Play wav file during conversation   Eric
• [asterisk-users] Codec Thread   Erik
• [asterisk-users] Re: Mediatrix 1204 trix   Erik
• [asterisk-users] Re: Mediatrix 1204 trix   Erik
• [asterisk-users] importance of crc4 in zaptel.conf?   Erik
• [asterisk-users] transcoding error?   Damon Estep
• [asterisk-users] fast SIP failover (outgoing sIP requests) with 1.2   Damon Estep
• [asterisk-users] Polycom default handset volume   Damon Estep
• [asterisk-users] Asterisk Outgoing Spool Failed with ViciDial (MattF?)   Andrea Cristofanini -- [Gedam Europe]
• [asterisk-users] Connecting Motorola VT2442 Device   David Evennou
• [asterisk-users] Polycom Firmware   Forum Expansive
• [asterisk-users] problems with Polycom 500 boot up   Forum Expansive
• [asterisk-users] How to send correct Caller ID on PRI   C F
• [asterisk-users] Hrmm.. OK, what am I missing? sendmail: Cannot open mail:25   C F
• [asterisk-users] Rack for Asterisk with TDM2400 Digium board   C F
• [asterisk-users] Copyright issues with libcurl and OpenSSL   C F
• [asterisk-users] Mediatrix 1204   C F
• [asterisk-users] Re: Mediatrix 1204 trix   C F
• [asterisk-users] is chanisavail command reliable?   C F
• [asterisk-users] Re: Mediatrix 1204 trix   C F
• [asterisk-users] Re: Mediatrix 1204 trix   C F
• [asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe   C F
• [asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe   C F
• [asterisk-users] Running Multiple Instances of Asterisk   C F
• [asterisk-users] include "context"   C F
• [asterisk-users] Set hint status from dialplan?   C F
• [asterisk-users] PRI Outbound CallerID Question   C F
• [asterisk-users] Included context   C F
• [asterisk-users] Re: Set hint status from dialplan?   C F
• [asterisk-users] Termination   C F
• [asterisk-users] Set hint status from dialplan?   C F
• [asterisk-users] Blind transfer 3/4 digits   Fabio
• [asterisk-users] Digium GUI?   Don Fanning
• [asterisk-users] Yet another processor question   Mark Farver
• [asterisk-users] T1 incoming connects, but no sound   Mark Farver
• [asterisk-users] Cisco 7960 Double Natted   Barry Fawthrop
• [asterisk-users] Cisco 7960 Double Natted   Barry Fawthrop
• [asterisk-users] Asterisk behind Sonicwall firewall   Barry Fawthrop
• [asterisk-users] Asterisk behind Sonicwall firewall   Barry Fawthrop
• [asterisk-users] Asterisk behind Sonicwall firewall   Barry Fawthrop
• [asterisk-users] Asterisk behind Sonicwall firewall   Barry Fawthrop
• [asterisk-users] Fax with asterisk?   Matthias Fechner
• [asterisk-users] Replacing mpg123 with madplay under Solaris?   Matthias Fechner
• [asterisk-users] How to notify an ACD agent before he/she picks up   Manrique Feoli
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Ferguson, Michael
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Ferguson, Michael
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Ferguson, Michael
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Ferguson, Michael
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Ferguson, Michael
• [asterisk-users] University dumps CISCO VoIP for Asterisk   Ferguson, Michael
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Bruce Ferrell
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Bruce Ferrell
• [asterisk-users] Astmanproxy authentication problems   Leonardo Gomes Figueira
• [asterisk-users] Display message on voip phone...hint?   Sven Fischer
• [asterisk-users] Caller ID has extra digits to strip   Bart Fisher
• [asterisk-users] Caller ID has extra digits to strip   Bart Fisher
• [asterisk-users] Caller ID has extra digits to strip   Bart Fisher
• [asterisk-users] I'm I wrong - No 3-way calling for Single line sets?   Bart Fisher
• [asterisk-users] 2 CPU's, Only 1 taking IRQ's   Bart Fisher
• [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's   Bart Fisher
• [asterisk-users] QuadBRI and Zyxel Wifi phone stop working together after 3 calls   Frederik Fix
• [asterisk-users] QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls   Frederik Fix
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Kevin P. Fleming
• [asterisk-users] Digium G.729 codec binaries updated   Kevin P. Fleming
• [asterisk-users] Digium's response to posting of G.729 and G.723 source code   Kevin P. Fleming
• [asterisk-users] Digium G.729 codec now available for Solaris/SPARC   Kevin P. Fleming
• [asterisk-users] libpri current extracts as beta1   Kevin P. Fleming
• [asterisk-users] Digium G.729 codec binaries updated for Asterisk 1.4 beta   Kevin P. Fleming
• [asterisk-users] Asterisk vicidial question   Matt Florell
• [asterisk-users] Answer Machine detection   Matt Florell
• [asterisk-users] Asterisk Outgoing Spool Failed with ViciDial (MattF?)   Matt Florell
• [asterisk-users] New astGUIclient VICIDIAL Release: 2.0.1   Matt Florell
• [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy   Matt Florell
• [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy   Matt Florell
• [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy   Matt Florell
• [asterisk-users] Polycom SoundPoint 2.0.1 SIP firmware?   Matt Florell
• [asterisk-users] Dual core   Matt Florell
• [asterisk-users] Re: Dual core   Matt Florell
• [asterisk-users] Re: Dual core   Matt Florell
• [asterisk-users] Re: Dual core   Matt Florell
• [asterisk-users] Any Suggestions for Election Polling Application?   Matt Florell
• [asterisk-users] automatic dialer with number pool and auto dialing   Matt Florell
• [asterisk-users] Polycom Firmware   Forum
• [asterisk-users] problems with Polycom 500 boot up   Forum
• [asterisk-users] Polycom 500 power supply   Forum
• [asterisk-users] problems with Polycom 500 boot up   Forum
• [asterisk-users] problems with Polycom 500 boot up   Forum
• [asterisk-users] problems with Polycom 500 boot up   Forum
• [asterisk-users] problems with Polycom 500 boot up   Forum
• [asterisk-users] problems with Polycom 500 boot up   Forum
• [asterisk-users] set global variable   Jan Fousek
• [asterisk-users] set global variable   Jan Fousek
• [asterisk-users] ISDN / Multiplink PPP (ZapRAS)   Brent Franks
• [asterisk-users] question...   Brent Franks
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   Brent Franks
• [asterisk-users] Rack for Asterisk with TDM2400 Digium board   Matthew Fredrickson
• [asterisk-users] TDM2400P   Matthew Fredrickson
• [asterisk-users] Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???   Matthew Fredrickson
• [asterisk-users] Problems configuring Polycom 301   Jim Freeze
• [asterisk-users] attended transfer unreliable   Stefan Friedrich
• [asterisk-users] [Slightly OT] Grandstream configurator tool   Dave Fullerton
• [asterisk-users] [Slightly OT] Grandstream configurator tool   Dave Fullerton
• [asterisk-users] Re: 911 versus 9.911   Dave Fullerton
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Dave Fullerton
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Dave Fullerton
• [asterisk-users] Call parking and RTP traffic   Dave Fullerton
• [asterisk-users] What don't I get about SIP?   Dave Fullerton
• [asterisk-users] Comments on new system plan.   Dave Fullerton
• [asterisk-users] TDM2400P vs Sangoma A200   Dave Fullerton
• [asterisk-users] TDM2400P vs Sangoma A200   Dave Fullerton
• [asterisk-users] bristuff compile problems with kernel 2.6.17.11   Arik Raffael Funke
• [asterisk-users] Re: bristuff compile problems with kernel 2.6.17.11   Arik Raffael Funke
• [asterisk-users] Re: bristuff compile problems with kernel 2.6.17.11   Arik Raffael Funke
• [asterisk-users] I am not getting 302 redirects...   Arik Raffael Funke
• [asterisk-users] Virtualise asterisk on Xen   Arik Raffael Funke
• [asterisk-users] Problems with call parking   Arik Raffael Funke
• [asterisk-users] Dealing with FINAREA redirects   Arik Raffael Funke
• [asterisk-users] mISDN problem: no version for "capi_cmd2str" found   Arik Raffael Funke
• [asterisk-users] chan_isdn / chan_sip problems   Arik Raffael Funke
• [asterisk-users] Call Processing Slow 11 seconds   G.Jacobsen
• [asterisk-users] Call Processing Slow 11 seconds   G.Jacobsen
• [asterisk-users] Handling Disconnection Causes   Rafael J. Risco G.V.
• [asterisk-users] Re: [Users] Integrating the Openser for VoiceMail and PBX with Asterisk, For Account   Rafael J. Risco G.V.
• [asterisk-users] Blind transfer 3/4 digits   David Gagnon
• [asterisk-users] Blind transfer 3/4 digits   David Gagnon
• [asterisk-users] Asterisk 1.2.11 and # key   David Gagnon
• [asterisk-users] Blind transfer 3/4 digits   David Gagnon
• [asterisk-users] Asterisk 1.2.11 and # key   David Gagnon
• [asterisk-users] Call center reports   David Gagnon
• [asterisk-users] Asterisk 1.2.11 and # key   David Gagnon
• [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute   David Gagnon
• [asterisk-users] Polycom new firmware and bootrom   David Gagnon
• [asterisk-users] Polycom HD Voice - 16 Khz - Asterisk support ?   David Gagnon
• [asterisk-users] Polycom IP430 sound level too low?   David Gagnon
• [asterisk-users] Dial and Timeout   David Gagnon
• [asterisk-users] How to make Polycom 501 go off hook whenpressingany digits   David Gagnon
• [asterisk-users] Problem with # locking up call   David Gagnon
• [asterisk-users] Polycom default handset volume   David Gagnon
• [asterisk-users] TDM2400P   David Gagnon
• [asterisk-users] DSL router with integrated SIP proxy?   David Gagnon
• [asterisk-users] TDM2400P vs Sangoma A200   David Gagnon
• [asterisk-users] Budgetones - multiple phones losing IP address during day   Brandon Galbraith
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Brandon Galbraith
• [asterisk-users] Polycom new firmware and bootrom   Brandon Galbraith
• [asterisk-users] Asterisk 1.2 and SATA drives   Brandon Galbraith
• [asterisk-users] Re: FW: Peter Dicks Chairman ofSportingbet PLC isarrested at JFK!!   Brandon Galbraith
• [asterisk-users] University switches to Asterisk   Brandon Galbraith
• [Asterisk-Users] Can you explain why multiple registration is an important (missing) feature ?   Brandon Galbraith
• [asterisk-users] Enabling Second Processor Trashes Audio Quality   Brandon Galbraith
• [asterisk-users] Forwarding   Brandon Galbraith
• [asterisk-users] Comments on new system plan.   Brandon Galbraith
• [asterisk-users] Forwarding to a caller-pays mobile   Brandon Galbraith
• [asterisk-users] Building the Perfect Box   Brandon Galbraith
• [asterisk-users] TDM2400P vs Sangoma A200   Sampath Ganji
• [asterisk-users] Using Asterisk with IVR connected with legacy pbx via rs-232   Paulo Garcia
• [asterisk-users] Using Asterisk with IVR connected with legacy pbxvia rs-232   Paulo Garcia
• [asterisk-users] Problems with outgoing calls   Roy Gardner
• [asterisk-users] includes in realtime ??   Douglas Garstang
• [asterisk-users] New polycom firmware / presence   Douglas Garstang
• [asterisk-users] Voicemail Delete Bug?   Douglas Garstang
• [asterisk-users] Polycom new firmware and bootrom   Douglas Garstang
• [asterisk-users] Polycom new firmware and bootrom   Douglas Garstang
• [asterisk-users] Polycom new firmware and bootrom   Douglas Garstang
• [asterisk-users] Polycom Soundpoint Key Remap   Douglas Garstang
• [asterisk-users] Polycom Soundpoint Key Remap   Douglas Garstang
• [asterisk-users] Polycom MyStat   Douglas Garstang
• [asterisk-users] Polycom Soundpoint Key Remap   Douglas Garstang
• [asterisk-users] Polycom MyStat   Douglas Garstang
• [asterisk-users] BLF across asterisk trunks   Douglas Garstang
• [asterisk-users] Asterisk 1.4 Docs   Douglas Garstang
• [asterisk-users] Sphinx2   Douglas Garstang
• [asterisk-users] Sphinx2   Douglas Garstang
• [asterisk-users] Sphinx2   Douglas Garstang
• [asterisk-users] Sphinx2   Douglas Garstang
• [asterisk-users] Polycom Expansion Module   Douglas Garstang
• [asterisk-users] Polycom SoundPoint 2.0.1 SIP firmware?   Douglas Garstang
• [asterisk-users] Polycom Expansion Module   Douglas Garstang
• [asterisk-users] Polycom programmable buttons   Douglas Garstang
• [asterisk-users] Digium GUI?   Douglas Garstang
• [asterisk-users] sip.conf for talking to other Asterisk machines   Douglas Garstang
• [asterisk-users] Digium GUI?   Douglas Garstang
• [asterisk-users] Polycom default handset volume   Douglas Garstang
• [asterisk-users] Digium GUI?   Douglas Garstang
• [asterisk-users] University dumps CISCO VoIP for Asterisk   Douglas Garstang
• [asterisk-users] Polycom 2.0.1 Software   Douglas Garstang
• [asterisk-users] Polycom 2.0.1 Software   Douglas Garstang
• [asterisk-users] Running Multiple Instances of Asterisk   Douglas Garstang
• [asterisk-users] DUNDi Servers   Douglas Garstang
• [asterisk-users] Asterisk 1.4 autoconf and /etc/asterisk directory   Douglas Garstang
• [asterisk-users] Running Multiple Instances of Asterisk   Douglas Garstang
• [asterisk-users] Running Multiple Instances of Asterisk   Douglas Garstang
• [asterisk-users] Running Multiple Instances of Asterisk   Douglas Garstang
• [asterisk-users] Shared Line Appearances in 1.4   Douglas Garstang
• [asterisk-users] MOH in 1.4 - Still Broken?   Douglas Garstang
• [asterisk-users] MOH in 1.4 - Still Broken?   Douglas Garstang
• [asterisk-users] MOH in 1.4 - Still Broken?   Douglas Garstang
• [asterisk-users] Asterisk 1.4 mohsuggest   Douglas Garstang
• [asterisk-users] Asterisk 1.4 mohsuggest   Douglas Garstang
• [asterisk-users] SER with multiple asterisk deployment   Douglas Garstang
• [asterisk-users] SER with multiple asterisk deployment   Douglas Garstang
• [asterisk-users] Polycom Queues, Login, Logout etc   Douglas Garstang
• [asterisk-users] Queue AddQueueMember()   Douglas Garstang
• [asterisk-users] Polycom Queues, Login, Logout etc   Douglas Garstang
• [asterisk-users] DUNDi Docs   Douglas Garstang
• [asterisk-users] A Step back with AddQueueMember() ?   Douglas Garstang
• [asterisk-users] Queue AddQueueMember()   Douglas Garstang
• [asterisk-users] Alcatel OXO Sip   Christian Gatti
• [asterisk-users] Alcatel OXO Sip   Christian Gatti
• [asterisk-users] Alcatel OXO Sip   Christian Gatti
• [asterisk-users] how to transfer a caller out of a queue ?   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] how to transfer a caller out of a queue ?   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] how to transfer a caller out of a queue ?   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] switching from IAX to SIP   Jerry Geis
• [asterisk-users] sip and iax over the internet (asterisk to asterisk) drop outs normal???   Jerry Geis
• [asterisk-users] consitent half channel loss after 6 minutes   Jerry Geis
• [asterisk-users] audio drop out half channel   Jerry Geis
• [asterisk-users] asterisk server to server using sip question   Jerry Geis
• [asterisk-users] asterisk server to server using sip question   Jerry Geis
• [asterisk-users] where download app_txfax?   Jerry Geis
• [asterisk-users] app_txfax segv fault   Jerry Geis
• [asterisk-users] tx_fax over sip to TDM card   Jerry Geis
• [asterisk-users] CURL   Jerry Geis
• [asterisk-users] experience with phones locking up uniden and cisco   Jerry Geis
• [asterisk-users] txfax question   Jerry Geis
• [asterisk-users] txfax reliability on TDM cards   Jerry Geis
• [asterisk-users] 2 CPU's, Only 1 taking IRQ's   Csibra Gergo
• [asterisk-users] Cisco PIX firewall and nat=yes   Bill Gibbs
• [asterisk-users] Cisco PIX firewall and nat=yes   Bill Gibbs
• [asterisk-users] Receive Fax with rxfax on asterisk with debian   Bill Gibbs
• [asterisk-users] Third Lane PBX Manger Multi-Tenant   Bill Gibbs
• [asterisk-users] QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls   Bill Gibbs
• [asterisk-users] saved.gsm -> Voicemail greeting ??   Bill Gibbs
• [asterisk-users] saved.gsm -> Voicemail greeting ??   Bill Gibbs
• [asterisk-users] saved.gsm -> Voicemail greeting ??   Bill Gibbs
• [asterisk-users] sip.conf for talking to other Asterisk machines   Bill Gibbs
• [asterisk-users] Looped message playback   Bill Gibbs
• [asterisk-users] Looped message playback   Bill Gibbs
• [asterisk-users] Dual core   Bill Gibbs
• [asterisk-users] 64 analog phones   Bill Gibbs
• [asterisk-users] Polycom 501 One-way Audio   Bill Gibbs
• [asterisk-users] Forcing Transcode   Bill Gibbs
• [asterisk-users] Very high ping times from 7960 phones   Steve Glaus
• [asterisk-users] Re: Very high ping times from 7960 phones   Steve Glaus
• [asterisk-users] Very high ping times from 7960 phones   Steve Glaus
• [asterisk-users] Very high ping times from 7960 phones   Steve Glaus
• [asterisk-users] pstn failback   Steve Glaus
• [asterisk-users] Replacing mpg123 with madplay under Solaris?   Ken Godee
• [asterisk-users] Zaptel-1.2.9 compile error   Nigel Godfrey
• [asterisk-users] ztdummy installed but choppy audio warning on load   Nigel Godfrey
• [asterisk-users] mISDN, 2 Billion HFC ISDN cards, cannot dial or receive   Nigel Godfrey
• [asterisk-users] Re: mISDN, 2 Billion HFC ISDN cards, cannot dial or receive   Nigel Godfrey
• [asterisk-users] Asterisk vicidial question   Gustavo Alejandro Gonzalez
• [asterisk-users] Asterisk with cisco 7935   Julien Goodwin
• [asterisk-users] Asterisk with cisco 7935   Julien Goodwin
• [asterisk-users] Switch Experiences   Ben Gore
• [asterisk-users] Polycom Soundpoint Key Remap   Adam Goryachev
• [asterisk-users] HFC isdn card and bristuff 0.2.0 rc8n   Giordano Grandis
• [asterisk-users] mpg123   Giordano Grandis
• [asterisk-users] Whcih phones are better for mass deployment   Michael Graves
• [asterisk-users] Whcih phones are better for mass deployment   Michael Graves
• [asterisk-users] using residential voip for business?   Michael Graves
• [asterisk-users] using residential voip for business?   Michael Graves
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Michael Graves
• [asterisk-users] VMware and Digium TDM400P card   Andy Green
• [asterisk-users] --- Dlink DVC-2000 VideoPhone (H.323) with Asterisk ---   Ed Greenberg
• [asterisk-users] Dell hardware ...   Ed Greenberg
• [asterisk-users] Is asterisk's mgcp support(NAS) Network access server package   Davor Grgicevic
• [asterisk-users] Maximum retries exceeded on transmission   AJ Grinnell
• [asterisk-users] gTalk no audio issue   Alex Guan
• [asterisk-users] Re: gTalk no audio issue   Alex Guan
• [asterisk-users] Asterisk Hangups on PRI Interface   Sahil Gupta
• [asterisk-users] Asterisk Hangups on PRI Interface   Sahil Gupta
• [asterisk-users] Cisco 7940 Problem (Mess)   Gary Guthary
• [asterisk-users] SOS building fastagi C   Sebastian Gutierrez
• [asterisk-users] ChanSpy issue   Sebastian Gutierrez
• [asterisk-users] Grandstream Budgetone phones don't show   Craig Guy
• [asterisk-users] rxfax, spandsp and lack of ecm   Craig Guy
• [asterisk-users] rxfax, spandsp and lack of ecm   Craig Guy
• [asterisk-users] rxfax, spandsp and lack of ecm   Craig Guy
• [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?   Craig Guy
• [asterisk-users] Re: Can you explain why multiple registrationisan important (missing) feature ?   Craig Guy
• [asterisk-users] Extensions busy/congested and "circuit-busy"   Guy M Guyadeen
• [asterisk-users] Extensions busy/congested and "circuit-busy"   Guy M Guyadeen
• [asterisk-users] Grandstream GXV 3000   Chris HARIGA
• [asterisk-users] Problems with KG1000 voip gateway and DTMF   Terence Haddock
• [asterisk-users] Voip providers and sip origination and termination?   Yair Hakak
• [asterisk-users] Voip providers and sip origination and termination?   Yair Hakak
• [asterisk-users] re: asterisk/SER integration - HELP   Yair Hakak
• [asterisk-users] PBX -> VoIP migration   Paul Hales
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Paul Hales
• [asterisk-users] TDM400 and T100 config on same asterisk   Paul Hales
• [asterisk-users] question...   Paul Hales
• [asterisk-users] Grandstream Budgetone phones don't show   Paul Hales
• [asterisk-users] How to download asterisk 1.3 development version   Paul Hales
• [asterisk-users] SHSU asterisk installation?   Paul Hales
• [asterisk-users] Forwarding   Paul Hales
• [asterisk-users] Forwarding   Paul Hales
• [asterisk-users] Forwarding   Paul Hales
• [asterisk-users] E1 crossover system   Paul Hales
• [asterisk-users] E1 crossover system   Paul Hales
• [asterisk-users] E1 crossover system   Paul Hales
• [asterisk-users] Auto Dialer question   Hall, Eric M.
• [asterisk-users] FollowMe question   Hall, Eric M.
• [asterisk-users] RE: FollowMe question   Hall, Eric M.
• [asterisk-users] HINT problems with SVN-trunk-r43322   Hall, Eric M.
• [asterisk-users] HINT problems with SVN-trunk-r43322   Hall, Eric M.
• [asterisk-users] HINT problems with SVN-trunk-r43322   Hall, Eric M.
• [asterisk-users] HINT problems with SVN-trunk-r43322   Hall, Eric M.
• [asterisk-users] Question about SVN-trunk-r43322 and Asterisk Recording Interface   Hall, Eric M.
• [asterisk-users] Question about SVN-trunk-r43322 and Asterisk Recording Interface   Hall, Eric M.
• Spam? [asterisk-users] OT: Opinions on Aastra 480i CT?   Hall, Eric M.
• [asterisk-users] Re: Set hint status from dialplan?   Hall, Eric M.
• [asterisk-users] Cisco 7970 behind NAT   Evan P. Hall
• [asterisk-users] ASTTAPI   Mike Hammett
• [asterisk-users] Trouble with rxfax multi-page printing with cups   Steve Hanselman
• [asterisk-users] RPID   Freddi Hansen
• [asterisk-users] Budgetones - multiple phones losing IP addressduring day   Harden, Bob
• [asterisk-users] Does a "HST Saphir III ML PCI" work with Asterisk?   James Harper
• [asterisk-users] Call is dead after featuredigittimeout   Florian Hars
• [asterisk-users] A1200+fxo, anyone using this?   Darrick Hartman
• [asterisk-users] Fedora   Darrick Hartman
• [asterisk-users] RE: FollowMe question   Darrick Hartman
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Barry D. Hassler
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Barry D. Hassler
• [asterisk-users] ZAP: psuedo camped on channel 1?   Barry D. Hassler
• [asterisk-users] Spurious hangups on zaptel interface   Barry D. Hassler
• [asterisk-users] Spurious hangups on zaptel interface   Barry D. Hassler
• [asterisk-users] Spurious hangups on zaptel interface   Barry D. Hassler
• [asterisk-users] Queue timeout problems   Guido Hecken
• [asterisk-users] Queue timeout problems   Guido Hecken
• [asterisk-users] Experience Patton BRI gateways and Asterisk?   Guido Hecken
• [asterisk-users] Incoming call problem-calling part is busy(I PKall)   Guido Hecken
• [asterisk-users] Asterisk 1.2 and SATA drives   Guido Hecken
• [asterisk-users] Asterisk Server Down   Guido Hecken
• [asterisk-users] Asterisk Server Down   Guido Hecken
• [asterisk-users] Fedora   Guido Hecken
• [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2   Guido Hecken
• [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0   Guido Hecken
• [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0   Guido Hecken
• [asterisk-users] Can QUEUE member be assigned from a GlobalVar set in EXTENSIONS.CONF?   Gary G. Hendershot
• [asterisk-users] Can QUEUE member be assigned from a GlobalVar set in EXTENSIONS.CONF?   Gary G. Hendershot
• [asterisk-users] Branch office interconnect - IAX :vs: SIP?   Gary G. Hendershot
• [asterisk-users] Very Loud (+Echo/Reverb) issue with TDM400P+fax detection (UK)   Gordon Henderson
• [asterisk-users] Correct settings for UK (BT) FXO   Gordon Henderson
• [asterisk-users] A Caller ID question (UK)   Gordon Henderson
• [asterisk-users] Extension Numbering   Gordon Henderson
• [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic   Paul Hewlett
• [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problemwith module versionmagic   Paul Hewlett
• [asterisk-users] rtc: lost some interrupts at 1024 when loading ztdummy   Paul Hewlett
• [asterisk-users] Sound Quality.   Joel Hill
• [asterisk-users] Asterisk as a SER client   David Hindmarsh
• [asterisk-users] trixbox t38 pass through   David Hindmarsh
• [asterisk-users] Polycom IP430 HUM and Echo   Lorentz Hinrichsen
• [asterisk-users] Uninstalling Trixbox   Rizwan Hisham
• [asterisk-users] Uninstalling Trixbox   Rizwan Hisham
• [asterisk-users] Uninstalling Trixbox   Rizwan Hisham
• [asterisk-users] Uninstalling Trixbox   Rizwan Hisham
• [asterisk-users] Uninstalling Trixbox   Rizwan Hisham
• [asterisk-users] Trixbox Documentation   Rizwan Hisham
• [asterisk-users] Codec Thread   Jean-Michel Hiver
• [asterisk-users] Can not hear the telco System Announcement   Jean-Michel Hiver
• [asterisk-users] OT -- echo cancellation of an audio file   Jean-Michel Hiver
• [asterisk-users] callback without agi   Jean-Michel Hiver
• [asterisk-users] Re: [asterisk-dev] open letter   Jean-Michel Hiver
• [asterisk-users] Re: [asterisk-dev] open letter   Jean-Michel Hiver
• [asterisk-users] Setting QOS settings in asterisk and/or CentOS?   Nick Hoffman
• [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's   Nick Hoffman
• [asterisk-users] SER with multiple asterisk deployment   Nick Hoffman
• [asterisk-users] Grandstream Budgetone phones don't show   Jessee Holmes
• [asterisk-users] Budgetones - multiple phones losing IP address during day   Jessee J Holmes
• [asterisk-users] Budgetones - multiple phones losing IP addressduring day   Jessee J Holmes
• [asterisk-users] Polycom new firmware and bootrom   Jessee J Holmes
• [asterisk-users] Polycom new firmware and bootrom   Jessee J Holmes
• [asterisk-users] Polycom new firmware and bootrom   Jessee J Holmes
• [asterisk-users] Grandstream Budgetone phones don't show   Jessee J Holmes
• [asterisk-users] problems with Polycom 500 boot up   Jessee J Holmes
• [asterisk-users] Polycom default handset volume   Jessee J Holmes
• [asterisk-users] problems with Polycom 500 boot up   Jessee J Holmes
• [asterisk-users] problems with Polycom 500 boot up   Jessee J Holmes
• [asterisk-users] problems with Polycom 500 boot up   Jessee J Holmes
• [asterisk-users] problems with Polycom 500 boot up   Jessee J Holmes
• [asterisk-users] problems with Polycom 500 boot up   Jessee J Holmes
• [asterisk-users] Asterisk and "305 Use Proxy"   Wolfgang Hottgenroth
• [asterisk-users] rxfax, spandsp and lack of ecm   Lee Howard
• [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Lee Howard
• [hylafax-users] [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Lee Howard
• [asterisk-users] spandsp (foip)   Lee Howard
• [asterisk-users] spandsp (foip)   Lee Howard
• [asterisk-users] Conditional IF based on IP address?   Steve Hsieh
• [asterisk-users] Native Chinese speaker needed   Steve Hsieh
• [asterisk-users] Conditional IF based on IP address?   Steve Hsieh
• [asterisk-users] Native Chinese speaker needed   Steve Hsieh
• [asterisk-users] Re: Uninstalling Trixbox   Nic Hughes
• [asterisk-users] Re: Uninstalling Trixbox   Nic Hughes
• [asterisk-users] Cisco 7960 Double Natted   Hughes, Sam
• [asterisk-users] Static RealTime - SIP.CONF   Hugo
• [asterisk-users] Static RealTime - SIP.CONF   Hugo
• [asterisk-users] Building Zaptel 1.2.9 with Octasic   Mark Hulber
• [asterisk-users] Building Zaptel 1.2.9 with Octasic   Mark Hulber
• [asterisk-users] MWI on 1.4 Beta   Mark Hulber
• [asterisk-users] Help with blind transfer   George A. Roberts IV
• [asterisk-users] Help with blind transfer   George A. Roberts IV
• [asterisk-users] Help with blind transfer   George A. Roberts IV
• [asterisk-users] Help with blind transfer   George A. Roberts IV
• [asterisk-users] Asterisk "Clusters"   Koen Van Impe
• [asterisk-users] How to enable jingle in 1.4beta2?   Koen Van Impe
• [asterisk-users] quadbri & TDM400P on same pbx ?   Giorgio Incantalupo
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] Digits are played in english in french voicemail   Giorgio Incantalupo
• [asterisk-users] PRI: sometimes Asterisk drop calls   Giorgio Incantalupo
• [asterisk-users] Dell hardware ...   Giorgio Incantalupo
• [asterisk-users] PRI: sometimes Asterisk drop calls   Giorgio Incantalupo
• [Asterisk-Users] Junghanns BRI cards and misdn   Giorgio Incantalupo
• [asterisk-users] WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner.   Giorgio Incantalupo
• [asterisk-users] PRI: sometimes Asterisk drop calls   Giorgio Incantalupo
• [asterisk-users] WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner.   Giorgio Incantalupo
• [asterisk-users] PRI: sometimes Asterisk drop calls   Giorgio Incantalupo
• [asterisk-users] One way audio problem on gateway to PSTN after some time, no NAT involved   Giorgio Incantalupo
• [asterisk-users] Re: PRI: sometimes Asterisk drop calls   Giorgio Incantalupo
• [asterisk-users] two safe_asterisk processes on the same PBX???   Giorgio Incantalupo
• [asterisk-users] two safe_asterisk processes on the same PBX???   Giorgio Incantalupo
• [asterisk-users] two safe_asterisk processes on the same PBX???   Giorgio Incantalupo
• [asterisk-users] Re: PRI: sometimes Asterisk drop calls   Giorgio Incantalupo
• [asterisk-users] is chanisavail command reliable?   Giorgio Incantalupo
• [asterisk-users] Channel kept busy when creating ssh tunnel via AGI   Giorgio Incantalupo
• [asterisk-users] Channel kept busy when creating ssh tunnel via   Giorgio Incantalupo
• [asterisk-users] Asteisk plays music on hold starting from random point   Giorgio Incantalupo
• [asterisk-users] Asteisk plays music on hold starting from randompoint   Giorgio Incantalupo
• [asterisk-users] PRI Problems   Giorgio Incantalupo
• [asterisk-users] alternatives to mpg123: format_mp3, rawplayer or madplay?   Giorgio Incantalupo
• [asterisk-users] two safe_asterisk processes on the same PBX???   Giorgio Incantalupo
• [asterisk-users] Cannot make outbound calls to some numbers with PRI line from legacy PBX thru Asterisk with Sangoma   Giorgio Incantalupo
• [asterisk-users] Cannot make outbound calls to some numbers with PRI line from legacy PBX thru Asterisk with Sangoma   Giorgio Incantalupo
• [asterisk-users] Asterisk Hangups on PRI Interface   Giorgio Incantalupo
• [asterisk-users] Asterisk Hangups on PRI Interface   Giorgio Incantalupo
• [asterisk-users] Outgoing Call group ?   Ira
• [asterisk-users] Caller ID has extra digits to strip   Ira
• [asterisk-users] How to manipulate a plus in a phone number   Ira
• [asterisk-users] Which phones are better for mass deployment   Ira
• [asterisk-users] using residential voip for business?   Ira
• [asterisk-users] WG: Asterisk and Agents   Ira
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Ira
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Ira
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Ira
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Ira
• [asterisk-users] Re: Voip Buster - CID   Ira
• [asterisk-users] Asterisk / Audiocodes annoying issue - Seeking Suggestions   Morten Isaksen
• [asterisk-users] Does Asterisk 1.4 going to support realtime ex-girlfriend logic?   Morten Isaksen
• [asterisk-users] Audiocodes 124, Mediatrix 1124, or Citel field reports?   Morten Isaksen
• [asterisk-users] Audiocodes 124, Mediatrix 1124, or Citel field reports?   Morten Isaksen
• [asterisk-users] Problem with zaptel 1.4b2 and X101P Wildcard   Morten Isaksen
• [asterisk-users] Problem with zaptel 1.4b2 and X101P Wildcard   Morten Isaksen
• [asterisk-users] Problem with zaptel 1.4b2 and X101P Wildcard   Morten Isaksen
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] macros in Realtime   Benjamin Jacob
• [asterisk-users] cmd SET time value   Benjamin Jacob
• [asterisk-users] macros in Realtime   Benjamin Jacob
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] cmd SET time value   Benjamin Jacob
• [asterisk-users] cmd SET time value   Benjamin Jacob
• [asterisk-users] cmd SET time value   Benjamin Jacob
• [asterisk-users] distinguishing users by their domain   Benjamin Jacob
• [asterisk-users] Asterisk Realtime Arch - static or realtime?   Benjamin Jacob
• [asterisk-users] Asterisk Realtime Arch - static or realtime?   Benjamin Jacob
• [asterisk-users] realtime static config include contexts   Benjamin Jacob
• [asterisk-users] Static RealTime - SIP.CONF   Benjamin Jacob
• [asterisk-users] Static RealTime - SIP.CONF   Benjamin Jacob
• [asterisk-users] realtime static config include contexts   Benjamin Jacob
• [asterisk-users] voicemailmain errors on CLI   Benjamin Jacob
• [asterisk-users] voicemail access thru apache on another server   Benjamin Jacob
• [asterisk-users] voicemail access thru apache on another server   Benjamin Jacob
• [asterisk-users] voicemail access thru apache on another server   Benjamin Jacob
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Benjamin Jacob
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Benjamin Jacob
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Benjamin Jacob
• [asterisk-users] enumlookup - deprecated working - but appreciated one duznt :-(   Benjamin Jacob
• [asterisk-users] enumlookup - deprecated working - but appreciated one duznt :-(   Benjamin Jacob
• [asterisk-users] VoicemailMain()   Benjamin Jacob
• [asterisk-users] I doubt it...   Benjamin Jacob
• [asterisk-users] media stream count   Benjamin Jacob
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] Has anyone tried to install both digital card and analog card in one machine   Raphael Jacquot
• [asterisk-users] Asterisk 1.2 and SATA drives   Raphael Jacquot
• [asterisk-users] Asterisk "Clusters"   Raphael Jacquot
• [asterisk-users] Rack for Asterisk with TDM2400 Digium board   Raphael Jacquot
• [asterisk-users] sound file length   Raphael Jacquot
• [asterisk-users] sound file length   Raphael Jacquot
• [asterisk-users] WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner.   Raphael Jacquot
• [asterisk-users] University switches to Asterisk   Raphael Jacquot
• [asterisk-users] Enabling Second Processor Trashes Audio Quality   Raphael Jacquot
• [asterisk-users] Enabling Second Processor Trashes Audio Quality   Raphael Jacquot
• [asterisk-users] how to configure a sip service   Raphael Jacquot
• [asterisk-users] Is this phone any good?   Nabeel Jafferali
• [asterisk-users] Callback + dtmf problem   James
• [asterisk-users] Forcing Transcode   James
• [asterisk-users] PRI Outbound CallerID Question   James
• [asterisk-users] zttest output   Jason
• [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's   Robert Jenkins
• [asterisk-users] Call is dead after featuredigittimeout   Kai-Uwe Jensen
• [asterisk-users] RE: 1.4 Beta 2 Config Problem   Kai-Uwe Jensen
• [asterisk-users] RE: 1.4 Beta 2 Config Problem   Kai-Uwe Jensen
• [Asterisk-Users] Network impairment tools   Kai-Uwe Jensen
• [asterisk-users] Attended transfer and parking calls   Andrew Joakimsen
• [asterisk-users] Voicemail adjustments   Andrew Joakimsen
• [asterisk-users] Internal message being heard on pstn line   Andrew Joakimsen
• [asterisk-users] Reliability of the newer IAXy's   Andrew Joakimsen
• [asterisk-users] Reliability of the newer IAXy's   Andrew Joakimsen
• [asterisk-users] saved.gsm -> Voicemail greeting ??   Andrew Joakimsen
• [asterisk-users] Calling to PSTN newbie question   Andrew Joakimsen
• [asterisk-users] call across 2 asterisks   Andrew Joakimsen
• [asterisk-users] Ericsson MD110   Andrew Joakimsen
• [asterisk-users] ATA with wireless client   Andrew Joakimsen
• [asterisk-users] DSL router with integrated SIP proxy?   Andrew Joakimsen
• [asterisk-users] Want to support a better SIP stack in Asterisk?   Olle E Johansson
• [asterisk-users] Take 3 -- Trying to get SIP firmware on a 7970G   Chris Jones
• [asterisk-users] Copyright issues with libcurl and OpenSSL   James Jones
• [asterisk-users] Copyright issues with libcurl and OpenSSL   James Jones
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   James Jones
• [asterisk-users] Re: Adit 3104 randomly reboot   Jerry Jones
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Jerry Jones
• [asterisk-users] Polycom Expansion Module   Jerry Jones
• [asterisk-users] Polycom Expansion Module   Jerry Jones
• [asterisk-users] polycom 501 digitmap   Jerry Jones
• [asterisk-users] Aastra 9133i and Atcom AT-320 - Comments please   Jerry Jones
• [asterisk-users] Polycom 2.0.1 Software   Jerry Jones
• [asterisk-users] Re: Two phones, same number   Jerry Jones
• [asterisk-users] Polycom (and others) digitmap info   Jerry Jones
• [asterisk-users] DSL router with integrated SIP proxy?   Jerry Jones
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Mr. Jones
• [asterisk-users] Queue timeout problems   Mr. Jones
• [asterisk-users] Queue timeout problems   Mr. Jones
• [asterisk-users] Queue timeout problems   Mr. Jones
• [asterisk-users] SIP 415 messagse   Mr. Jones
• [asterisk-users] codecs/voicemail/DTMF   Mr. Jones
• [asterisk-users] codecs/voicemail/DTMF   Mr. Jones
• [asterisk-users] Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.209   Mr. Jones
• [asterisk-users] Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.209   Mr. Jones
• [asterisk-users] Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.209   Mr. Jones
• [asterisk-users] Forcing Transcode   Mr. Jones
• [asterisk-users] Forcing Transcode   Mr. Jones
• [asterisk-users] T1 timing errors (Frame Slips) on Nortel 61C to TE110P   Ronnie Jones
• [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P   Ronnie Jones
• [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P   Ronnie Jones
• [asterisk-users] Hardware question   Jeremy Jongsma
• [asterisk-users] LinkSys PAP2 ATA & Siemens Cordless 3010   Jopi at fastwebnet.it
• R: Re: [asterisk-users] LinkSys PAP2 ATA & Siemens Cordless 3010   Jopi at fastwebnet.it
• [asterisk-users] Multiple asterisk same GUI   Senad Jordanovic
• [asterisk-users] Re: Softphones IAX vs. SIP, remote connectivity.   Martin Joseph
• [asterisk-users] Re: the sounds quality of IAX2 channels are not good as SIP channels?   Martin Joseph
• [asterisk-users] Re: Streaming MoH Problem, starts and then stops immediately   Martin Joseph
• [asterisk-users] [OT] Nokia E60/61/70 and SIP   Martin Joseph
• [asterisk-users] Re: Reliability of the newer IAXy's   Martin Joseph
• [asterisk-users] Re: codecs/voicemail/DTMF   Martin Joseph
• [asterisk-users] Re: codecs/voicemail/DTMF   Martin Joseph
• [asterisk-users] Dynamic DNS asterisk server?   Martin Joseph
• [asterisk-users] Re: [asterisk-dev] Re: OT But So Ungodly Important   Martin Joseph
• [asterisk-users] Re: e911   Martin Joseph
• [asterisk-users] Re: Digium G.729 codec binaries updated for Asterisk 1.4 beta   Martin Joseph
• [asterisk-users] TDM400P Problem or Not?   Eddie Johnson Jr
• [asterisk-users] Verify Database Installation   Areski K
• [asterisk-users] Asterisk and REFER authentication   KEN KANGAN
• [asterisk-users] Asterisk 1.4 Beta ODBC connection   KC
• [asterisk-users] Callback + dtmf problem   Insider KT
• [asterisk-users] Termination Rates   Insider KT
• [asterisk-users] inbound call from GSM gateway: handle_request_invite: Failed to authenticate user   Allan Kamau
• [asterisk-users] SPA 3102 does not even attempt to register   Allan Kamau
• [asterisk-users] SPA-3102 PSTN->VoIP Gateway (quest for one stage dialing)   Allan Kamau
• [asterisk-users] Invite issues   Allan Kamau
• [asterisk-users] rtc: lost some interrupts at 1024 when loading ztdummy   Geoff Karl
• [asterisk-users] Repost: Register message received from realtime peer crashes Asterisk   Cameron and Karlene
• [asterisk-users] incompatible hardware?   Roy Sigurd Karlsbakk
• [asterisk-users] SHSU asterisk installation?   Roy Sigurd Karlsbakk
• [asterisk-users] SHSU asterisk installation?   Roy Sigurd Karlsbakk
• [asterisk-users] Call forward with CFU?   Roy Sigurd Karlsbakk
• [asterisk-users] When does Scalability requests Asterisk   Roy Sigurd Karlsbakk
• [asterisk-users] new in 1.4?   Roy Sigurd Karlsbakk
• [asterisk-users] Re: [asterisk-announce] Digium G.729 codec binaries updated for Asterisk 1.4 beta   Roy Sigurd Karlsbakk
• [asterisk-users] [asterisk-dev] Re: [asterisk-announce] Digium G.729 codec binariesupdated for Asterisk 1.4 beta   Roy Sigurd Karlsbakk
• [asterisk-users] RE: Asterisk 1.4 Docs   Jason A. Kates
• [asterisk-users] Polycom Soundpoint Key Remap   Shawn Kelley
• [asterisk-users] Polycom Soundpoint Key Remap   Shawn Kelley
• [asterisk-users] PRI Outbound CallerID Question   Shawn Kelley
• [asterisk-users] PRI Outbound CallerID Question   Shawn Kelley
• [asterisk-users] pstn failback   Shawn Kelley
• [asterisk-users] RE: Setting QOS settings in asterisk and/or CentOS?   Greg Kennedy
• [asterisk-users] iaxyprov downloading problems   Sean Kennedy
• [asterisk-users] iaxyprov downloading problems   Sean Kennedy
• [asterisk-users] iaxyprov downloading problems   Sean Kennedy
• [asterisk-users] iaxy configuration problems   Sean Kennedy
• [asterisk-users] iaxy will register, but doesn't detect POTS line   Sean Kennedy
• [asterisk-users] iaxy will register, but doesn't detect POTS line   Sean Kennedy
• [asterisk-users] iaxy: one way audio   Sean Kennedy
• [asterisk-users] iaxy: one way audio   Sean Kennedy
• [asterisk-users] Sipura SPA3000   Steve Kennedy
• [asterisk-users] Cisco MWI   Steve Kennedy
• [asterisk-users] Cisco MWI   Steve Kennedy
• [asterisk-users] updated zaptel tarball   Steve Kennedy
• [asterisk-users] Building Zaptel 1.2.9 with Octasic   Steve Kennedy
• [asterisk-users] Building Zaptel 1.2.9 with Octasic   Steve Kennedy
• [asterisk-users] amr codec   Steve Kennedy
• [asterisk-users] amr codec   Steve Kennedy
• [asterisk-users] CSR introduces UniVox reference platform   Steve Kennedy
• [asterisk-users] Available channels   Steve Kennedy
• [asterisk-users] Grandstream SX2000 attended tranfer   Steve Kennedy
• [asterisk-users] ChanIsAvail   Steve Kennedy
• [asterisk-users] DSL router with integrated SIP proxy?   Steve Kennedy
• [Asterisk-Users] UK Male English Voices   Steve Kennedy
• [asterisk-users] =?unknown-8bit?Q?GSM_?= =?unknown-8bit?Q?Gateway_Promotion_from_=A369GBP?=   Steve Kennedy
• [asterisk-users] Whcih phones are better for mass deployment   Thomas Kenyon
• [asterisk-users] Whcih phones are better for mass deployment   Thomas Kenyon
• [asterisk-users] SIP trunk   Thomas Kenyon
• [asterisk-users] g729 problem   Thomas Kenyon
• [asterisk-users] g729 problem   Thomas Kenyon
• [asterisk-users] Leave Queue when all agents busy   Mindaugas Kezys
• [asterisk-users] asterisk and PowerEdge 1950   Stas Khromoy
• [asterisk-users] Asterisk + Samsung OffServ 500   Eugeniy Khvastunov
• [asterisk-users] Samsung OfficeServ 500 + Asterisk(Tormenta 2) via PRI   Eugeniy Khvastunov
• [asterisk-users] Digium G.729 codec binaries updated   Kristian Kielhofner
• [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)   Kristian Kielhofner
• [asterisk-users] More Zaptel build problems   Kristian Kielhofner
• [asterisk-users] More Zaptel build problems   Kristian Kielhofner
• [asterisk-users] Via Epia platforms and asterisk   Kristian Kielhofner
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Kristian Kielhofner
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Kristian Kielhofner
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Kristian Kielhofner
• [asterisk-users] PRI Outbound CallerID Question   Kristian Kielhofner
• [asterisk-users] Asterisk behind Sonicwall firewall   Kristian Kielhofner
• [asterisk-users] PRI Outbound CallerID Question   Kristian Kielhofner
• [asterisk-users] PRI Outbound CallerID Question   Kristian Kielhofner
• [asterisk-users] PRI Outbound CallerID Question   Kristian Kielhofner
• [asterisk-users] PRI Outbound CallerID Question   Kristian Kielhofner
• [asterisk-users] RPID   Kristian Kielhofner
• [asterisk-users] SER with multiple asterisk deployment   Kristian Kielhofner
• [asterisk-users] Re: RPID   Kristian Kielhofner
• [asterisk-users] Voicemail callback bug?   Kristian Kielhofner
• [asterisk-users] RPID   Kristian Kielhofner
• [asterisk-users] RPID   Kristian Kielhofner
• [asterisk-users] polycom expansion module   Kevin Kiely
• [asterisk-users] Polycom Expansion Module   Kevin Kiely
• [asterisk-users] Dell hardware ...   Kevin Kiely
• [asterisk-users] pay as you go t.t38 fax termination and origination   John Kington
• [asterisk-users] Sound (or lack of it) problems   Jordan Kirby
• [asterisk-users] Disappearing Voicemail   Andrew Kirch
• [asterisk-users] sip show peers   Andrew Kirch
• [asterisk-users] Disappearing Voicemail   Andrew Kirch
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Richard Klingler
• [asterisk-users] PBX -> VoIP migration   Richard Klingler
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Richard Klingler
• [asterisk-users] Matra 6501   Richard Klingler
• [asterisk-users] Wrong CallerID passed to SIP phone   Richard Klingler
• [asterisk-users] Wrong CallerID passed to SIP phone   Richard Klingler
• [asterisk-users] netmask   Richard Klingler
• [asterisk-users] Cisco 7970 directories and services xml   Richard Klingler
• [asterisk-users] Re: Cisco 7970 directories and services xml   Richard Klingler
• [asterisk-users] Caller ID display on 7970G   Richard Klingler
• [asterisk-users] SIP trunk   Richard Klingler
• [asterisk-users] Register 2 times with same host   Richard Klingler
• [asterisk-users] Problems getting 7970G upgraded to SIP   Richard Klingler
• [asterisk-users] Forcing Marker bit, because SSRC has changed   Richard Klingler
• [asterisk-users] No channels available after reloading config   Richard Klingler
• [asterisk-users] Forcing Marker bit, because SSRC has changed   Richard Klingler
• [asterisk-users] 7970G SIP8-0-4 not registering with asterisk   Richard Klingler
• [asterisk-users] Any Hardphone with VPNClient embedded?   Philipp von Klitzing
• [asterisk-users] Dropped call question - "Maximum retries exceeded on transmission"   Kohler, Jeffrey
• [asterisk-users] Response to KP Flemming...   Andrew Kohlsmith
• [asterisk-users] Whcih phones are better for mass deployment   Andrew Kohlsmith
• [asterisk-users] 9 becomes 99 ? And other strangeness   Andrew Kohlsmith
• [asterisk-users] is chanisavail command reliable?   Andrew Kohlsmith
• [asterisk-users] HINT problems with SVN-trunk-r43322   Andrew Kohlsmith
• [asterisk-users] University dumps CISCO VoIP for Asterisk   Andrew Kohlsmith
• [asterisk-users] Re: Set hint status from dialplan?   Andrew Kohlsmith
• [asterisk-users] Re: Set hint status from dialplan?   Andrew Kohlsmith
• [asterisk-users] Re: Set hint status from dialplan?   Andrew Kohlsmith
• [asterisk-users] Set hint status from dialplan?   Andrew Kohlsmith
• [asterisk-users] Blind transfer 3/4 digits   Koopmann, Jan-Peter
• [asterisk-users] Blind transfer 3/4 digits   Koopmann, Jan-Peter
• [asterisk-users] Blind transfer 3/4 digits   Koopmann, Jan-Peter
• [asterisk-users] Experience Patton BRI gateways and Asterisk?   Koopmann, Jan-Peter
• [asterisk-users] CDR question with SIP/IAX trunks   Koopmann, Jan-Peter
• [asterisk-users] CDR question with SIP/IAX trunks   Koopmann, Jan-Peter
• [asterisk-users] CDR question with SIP/IAX trunks   Koopmann, Jan-Peter
• [asterisk-users] app_amd and voicemail   David Koski
• [asterisk-users] voice detection during playback   David Koski
• [asterisk-users] SIP client with video???   Blake Krone
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Blake Krone
• [asterisk-users] connect with two servers multiple time   Arjan Kroon
• [asterisk-users] Dell hardware ...   Arjan Kroon
• [asterisk-users] Asterisk Outgoing Spool Failed   Arun Kumar
• [asterisk-users] Asterisk Outgoing Spool Failed   Arun Kumar
• [asterisk-users] Iax Netstat Output   Arun Kumar
• [asterisk-users] Iax Netstat Output   Arun Kumar
• [asterisk-users] Iax2 show netstat   Arun Kumar
• [asterisk-users] Iax Netstat Output   Arun Kumar
• [asterisk-users] Iax Netstat Output   Arun Kumar
• [asterisk-users] how to setup poxy sip server   Ranjeet Kumar
• [asterisk-users] strange problem with calls between MGCP and SIP clients(ATA's)   Andy Kuo
• [asterisk-users] Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?   Andy Kuo
• [asterisk-users] Google talk and Asterisk 1.4   Robert LaPoint
• [asterisk-users] Google talk and Asterisk 1.4   Robert LaPoint
• [asterisk-users] Why not g726-32?   Steve Langstaff
• [asterisk-users] 488 Not acceptable here sent by Asterisk - SIPdebug follows   Steve Langstaff
• [asterisk-users] Alcatel OXO Sip   Steve Langstaff
• [asterisk-users] Alcatel OXO Sip   Steve Langstaff
• [asterisk-users] Alcatel OXO Sip   Steve Langstaff
• [asterisk-users] SIP "Lines" Example Citel   Steve Langstaff
• [asterisk-users] Forcing Marker bit, because SSRC has changed   Steve Langstaff
• [asterisk-users] ISDN30 and digital phones   Steve Langstaff
• [asterisk-users] I doubt it...   Steve Langstaff
• [asterisk-users] WAIT FOR DIGIT not working   Joel Lansden
• [asterisk-users] WAIT FOR DIGIT not working   Joel Lansden
• [asterisk-users] Asterisk / Audiocodes annoying issue - Seeking Suggestions   Andrew Latham
• [asterisk-users] Comments on new system plan.   Andrew Latham
• [asterisk-users] asterisk logging per day   Christophorus Laube
• [asterisk-users] read variable from shell script   Christophorus Laube
• [asterisk-users] asterisk skills in the philippines   Josel Layno
• [asterisk-users] Getting Incoming called from trxtel.com   Howard Leadmon
• [asterisk-users] Rpath PoundKey 1.2   Bobby Lee
• [asterisk-users] Intel Based G.729 and SVN-trunk-r42453   Jason Lee
• [asterisk-users] Intel Based G.729 and SVN-trunk-r42453   Jason Lee
• [asterisk-users] Intel Based G.729 and SVN-trunk-r42453   Jason Lee
• [asterisk-users] Intel Based G.729 and SVN-trunk-r42453   Jason Lee
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Problems compil 1.2.11   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Faxing ..   Lenny
• [asterisk-users] University switches to Asterisk   Ronald Lewis
• [asterisk-users] Help spread the word about Asterisk!   Ronald Lewis
• [asterisk-users] Issues with Monitor in 1.4?   Ronald Lewis
• [asterisk-users] Issues with Monitor in 1.4?   Ronald Lewis
• [asterisk-users] Has anyone tried to install both digital card and analog card in one machine   Xue Liangliang
• [asterisk-users] Intel 945G and Digium TE110P compatibility issue   Xue Liangliang
• [asterisk-users] Flexible Wrap Up Time for Queue   Xue Liangliang
• [asterisk-users] Has anyone tried to install both digital card and analog card in one machine   Xue Liangliang
• [asterisk-users] Queue Status via Dialplan   Michael J. Liberatore
• [asterisk-users] Re: Mediatrix 1204 trix   Ralph Liebessohn
• [asterisk-users] beginners question....   Sharon Lim
• [asterisk-users] How to integrate freepbx with a2billing?   Sharon Lim
• [asterisk-users] How to integrate freepbx with a2billing?   Sharon Lim
• [asterisk-users] SkypeOut with Asterisk?   Sharon Lim
• [asterisk-users] Uninstalling Trixbox   Sharon Lim
• [asterisk-users] Uninstalling Trixbox   Sharon Lim
• [asterisk-users] Help in Reloading of Asterisk...   Sharon Lim
• [asterisk-users] Multiple asterisk same GUI   Sharon Lim
• [asterisk-users] Multiple asterisk same GUI   Sharon Lim
• [asterisk-users] FAX handling   Jose Limeres
• [asterisk-users] Re: FAX handling   Jose Limeres
• [asterisk-users] Modem calls   Jose Limeres
• [asterisk-users] Modem calls   Jose Limeres
• [asterisk-users] A1200+fxo, anyone using this?   Peter Lindquist
• [asterisk-users] Incoming SIP provider goes unregistered and never recovers   John Lingate
• [asterisk-users] Anyone working on VXML, CCXML support for asterisk?   Asterisk Mail List
• [asterisk-users] Reload question   Mailing List
• [asterisk-users] grandstream gxp 2000 does not display names whencalling out   Mailing List
• [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out   Mailing List
• [asterisk-users] Very high ping times from 7960 phones   Mailing List
• [asterisk-users] Very high ping times from 7960 phones   Mailing List
• [asterisk-users] Reliability of the newer IAXy's   Lists
• [asterisk-users] Reliability of the newer IAXy's   Lists
• [asterisk-users] Reliability of the newer IAXy's   Lists
• [asterisk-users] Reliability of the newer IAXy's   Lists
• [asterisk-users] "Too many files..." error - best way to fix?   Matt Arnilo S. Baluyos (Mailing Lists)
• [asterisk-users] "Too many files..." error - best way to fix?   Matt Arnilo S. Baluyos (Mailing Lists)
• [asterisk-users] Termination   Duracom Lists
• [asterisk-users] SER+Asterisk integration   Rob Lith
• [asterisk-users] Budgetones - multiple phones losing IP address during day   Rob Lith
• [asterisk-users] Asterisk "Clusters"   Rob Lith
• [asterisk-users] TE411P or TE412P?   Rob Lith
• [asterisk-users] Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???   Rob Lith
• [asterisk-users] Meet-me recording formats   Michael Lively
• [asterisk-users] Any way to go from factory reset 7970 to SIP without Call Manager?   Jason Lixfeld
• [asterisk-users] Re: Any way to go from factory reset 7970 to SIP without Call Manager?   Jason Lixfeld
• [asterisk-users] Re: Any way to go from factory reset 7970 to SIP without Call Manager?   Jason Lixfeld
• [asterisk-users] Take 3 -- Trying to get SIP firmware on a 7970G   Jason Lixfeld
• [asterisk-users] Take 3 -- Trying to get SIP firmware on a 7970G   Jason Lixfeld
• [asterisk-users] Problems getting 7970G upgraded to SIP   Jason Lixfeld
• [asterisk-users] Problems getting 7970G upgraded to SIP   Jason Lixfeld
• [Asterisk-Users] No Sound from VoicemailMain - Device Linksys PAP2T-NA   Tracy Lofton
• [asterisk-users] 1.4 Beta 2 Config Problem   Alexander Lopez
• [asterisk-users] Realtime Queues and Postgres.   Fernando Lujan
• [asterisk-users] IAX or SIP termination provider that reaches6421xxxxxxx?   Luki
• [asterisk-users] fax over ip   Luki
• [asterisk-users] How to notify an ACD agent before he/she picks up   Richard Lyman
• [asterisk-users] set global variable   Richard Lyman
• [asterisk-users] Digium GUI?   Richard Lyman
• [asterisk-users] How do I reset a password?   Jim Lynch
• [asterisk-users] How do I reset a password?   Jim Lynch
• [asterisk-users] Where are the kernel sources?   Jim Lynch
• [asterisk-users] LinkSys PAP2 ATA & Siemens Cordless 3010   Mike Lynchfield
• [asterisk-users] Use PauseQueueMember   Julian Lyndon-Smith
• [asterisk-users] Use PauseQueueMember   Julian Lyndon-Smith
• [asterisk-users] Cisco 79xx and vlan   Julian Lyndon-Smith
• [asterisk-users] two safe_asterisk processes on the same PBX???   Julian Lyndon-Smith
• [asterisk-users] Cisco Distinctive ring using alert-info   Julian Lyndon-Smith
• [asterisk-users] "Ping" a phone   Julian Lyndon-Smith
• [asterisk-users] "Ping" a phone   Julian Lyndon-Smith
• [asterisk-users] Format_MP3, Streaming, File Formats, MOH   Julian Lyndon-Smith
• [asterisk-users] SNOM 320 - 404 "Not Found"   Julian Lyndon-Smith
• [asterisk-users] Running Multiple Instances of Asterisk   Julian Lyndon-Smith
• [asterisk-users] SMS Text Send working with BT Text in the UK??   Julian Lyndon-Smith
• [asterisk-users] 7940 vs. 7941   Julian Lyndon-Smith
• [asterisk-users] Queue AddQueueMember()   Julian Lyndon-Smith
• [asterisk-users] e911   Tom Lynn
• [asterisk-users] Re: 911 versus 9.911   Doug Lytle
• [asterisk-users] Re: Re: 911 versus 9.911   Doug Lytle
• [asterisk-users] How to manipulate a plus in a phone number   Doug Lytle
• [Asterisk-Users] T1 echo canceller   Doug Lytle
• [asterisk-users] Different MOH between waiting calls and transfer   Doug Lytle
• [asterisk-users] config include issues   Doug Lytle
• [asterisk-users] config include issues   Doug Lytle
• [asterisk-users] app_rxfax Only Receives One Page   Doug Lytle
• [asterisk-users] app_rxfax Only Receives One Page   Doug Lytle
• [asterisk-users] Cisco MWI   Doug Lytle
• [asterisk-users] Cisco MWI   Doug Lytle
• [asterisk-users] Different MOH between waiting calls and transfer   Doug Lytle
• [asterisk-users] Cisco MWI   Doug Lytle
• [asterisk-users] Cisco MWI   Doug Lytle
• [asterisk-users] Incoming call problem-calling part is busy(IPKall)   Doug Lytle
• [asterisk-users] Trouble with rxfax multi-page printing with cups   Doug Lytle
• [asterisk-users] Scope of contexts   Doug Lytle
• [asterisk-users] How to send correct Caller ID on PRI   Doug Lytle
• [asterisk-users] How to send correct Caller ID on PRI   Doug Lytle
• [asterisk-users] Grandstream Budgetone phones don't show   Doug Lytle
• [asterisk-users] Verizon ISDN service in NY & Hunt Groups   Doug Lytle
• [asterisk-users] rxfax, spandsp and lack of ecm   Doug Lytle
• [asterisk-users] voicemailmain errors on CLI   Doug Lytle
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   Doug Lytle
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   Doug Lytle
• [asterisk-users] IVR not able to Play the Balance.. need some   Doug Lytle
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   Doug Lytle
• [asterisk-users] University switches to Asterisk   Doug Lytle
• [asterisk-users] Controlling the channel   Doug Lytle
• [asterisk-users] Disappearing Voicemail   Doug Lytle
• [asterisk-users] Controlling the channel   Doug Lytle
• [asterisk-users] Page() paging application problem   Doug Lytle
• [asterisk-users] Tracking the source of a disconnect?   Doug Lytle
• [asterisk-users] Tracking the source of a disconnect?   Doug Lytle
• [asterisk-users] Tracking the source of a disconnect?   Doug Lytle
• [asterisk-users] Asterisk with cisco 7935   Doug Lytle
• [asterisk-users] Asterisk with cisco 7935   Doug Lytle
• [asterisk-users] A1200+fxo, anyone using this?   Doug Lytle
• [asterisk-users] Issues with AGI+Dial command   Doug Lytle
• [asterisk-users] iax2 trunk call limits   Doug Lytle
• [asterisk-users] rxfax, spandsp and lack of ecm   Doug Lytle
• [asterisk-users] Pri Event 6 and 8   Doug Lytle
• [asterisk-users] Channel kept busy when creating ssh tunnel via   Doug Lytle
• [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Doug Lytle
• [asterisk-users] Polycom phone references needed   Doug Lytle
• [asterisk-users] Callback   Doug Lytle
• [asterisk-users] Callback   Doug Lytle
• [asterisk-users] Asterisk with cisco 7935   Doug Lytle
• [asterisk-users] Asterisk with cisco 7935   Doug Lytle
• [asterisk-users] How do I reset a password?   Doug Lytle
• [asterisk-users] Fax detection ...   Doug Lytle
• [asterisk-users] Issues with Monitor in 1.4?   Doug Lytle
• [asterisk-users] real time billing system   Guillermo Salas M.
• [asterisk-users] Re: Really bad phone line.. possible causes?   M.Hockings
• [asterisk-users] How to notify an ACD agent before he/she picks up   MF
• [asterisk-users] Has anyone tried to install both digital card and analog card in one machine   MF
• [asterisk-users] How to notify an ACD agent before he/she picks up   MF
• [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?   MF
• [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?   MF
• [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?   MF
• [asterisk-users] Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???   MF
• [asterisk-users] What I always get asked in SME * deployments   Colin MacMillan
• [asterisk-users] Dell hardware ...   Brodie Macleod
• [asterisk-users] How to install HUDLite Server   Brodie Macleod
• [asterisk-users] Sip answer one side , ring other side   Brodie Macleod
• [asterisk-users] Asterisk Trunk with Alcatel 4200 PABX   Frederico Madeira
• [asterisk-users] asterisk - alcatel   Frederico Madeira
• [asterisk-users] asterisk - alcatel   Frederico Madeira
• [asterisk-users] Zaptel-1.2.9 compile error   Bill Maidment
• [asterisk-users] Zaptel-1.2.9 compile error   Bill Maidment
• [asterisk-users] Zaptel-1.2.9 compile error   Bill Maidment
• [asterisk-users] Zaptel-1.2.9 compile error   Bill Maidment
• [asterisk-users] Zaptel-1.2.9 compile error   Bill Maidment
• [asterisk-users] Zaptel-1.2.9 compile error   Bill Maidment
• [asterisk-users] How to detect dial tone on ZAP channel before dialling using TDM2400P   Naija Man
• [asterisk-users] Re: Voip Buster - CID   Naija Man
• [asterisk-users] Dial-9 (was Extension Numbering)   Naija Man
• [asterisk-users] help chan_bluetooth   Mauricio Mantilla
• [asterisk-users] help connecting cell phone, chan_bluetooth   Mauricio Mantilla
• [asterisk-users] How to Dial a number with Sangoma PRI card?   Mario
• [asterisk-users] How to Dial a number with Sangoma PRI card?   Mario
• [asterisk-users] How to Dial a number with Sangoma PRI card?   Mario
• [asterisk-users] Application of Asterisk Packetization Patch   Mark
• [asterisk-users] T38 in 1.4?   Ricardo Martins
• [asterisk-users] Is there T.38 support on asterisk 1.4 beta2 ???   Ricardo Martins
• [asterisk-users] What don't I get about SIP?   John Marvin
• [asterisk-users] What don't I get about SIP?   John Marvin
• [asterisk-users] What don't I get about SIP?   John Marvin
• [asterisk-users] Polycom related question   John Marvin
• [asterisk-users] Polycom related question   John Marvin
• [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)   John Marvin
• [asterisk-users] Polycom related question   John Marvin
• [asterisk-users] A Caller ID question (UK)   John Marvin
• [asterisk-users] sound file length   John Marvin
• [asterisk-users] Looped message playback   John Marvin
• [asterisk-users] IAX or SIP termination provider that reaches6421xxxxxxx?   John Marvin
• [asterisk-users] Extension Numbering   John Marvin
• [asterisk-users] Extension Numbering   John Marvin
• [asterisk-users] Wrong outgoing port   Master_PE
• [asterisk-users] Re: Re: 911 versus 9.911   Matt
• [asterisk-users] Asterisk 1.2.11 and # key   Matt
• [asterisk-users] music onhold choppy music problems   Matt
• [asterisk-users] Zero length queue   Artifex Maximus
• [asterisk-users] Trouble with rxfax multi-page printing with cups   Artifex Maximus
• [asterisk-users] Trouble with rxfax multi-page printing with cups   Artifex Maximus
• [asterisk-users] How to setup announce attibute in queues.conf   Artifex Maximus
• [asterisk-users] rxfax, spandsp and lack of ecm   Artifex Maximus
• [asterisk-users] rxfax, spandsp and lack of ecm   Artifex Maximus
• [asterisk-users] Queue - static members   Artifex Maximus
• [asterisk-users] rxfax, spandsp and lack of ecm   Artifex Maximus
• [asterisk-users] rxfax, spandsp and lack of ecm   Artifex Maximus
• [asterisk-users] where download app_txfax?   Artifex Maximus
• [asterisk-users] rxfax, spandsp and lack of ecm   Artifex Maximus
• [asterisk-users] rxfax, spandsp and lack of ecm   Artifex Maximus
• [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Artifex Maximus
• [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Artifex Maximus
• [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Artifex Maximus
• [hylafax-users] [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Artifex Maximus
• [hylafax-users] [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Artifex Maximus
• [asterisk-users] Included context   Artifex Maximus
• [asterisk-users] Included context   Artifex Maximus
• [asterisk-users] Polycom programmable buttons   Ron McCarthy
• [asterisk-users] Polycom programmable buttons   Ron McCarthy
• [asterisk-users] Termination Rates   Ron McCarthy
• [asterisk-users] Sound Quality.   Callum McGillivray
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Timothy R. McKee
• [asterisk-users] Asterisk not sending RTP   Jeremy McNamara
• [asterisk-users] Asterisk not sending RTP   Jeremy McNamara
• [asterisk-users] Looks like Nufone is changing around...   Jeremy McNamara
• [asterisk-users] Asterisk as a gateway to SER   Jeremy McNamara
• [asterisk-users] SER with multiple asterisk deployment   Jeremy McNamara
• [asterisk-users] Rack for Asterisk with TDM2400 Digium board   Antoine Megalla
• [asterisk-users] Any suggestions about VoIP provider?   Antoine Megalla
• [asterisk-users] Configuring new IAX2 Jitter Buffer for IVR application.   John Melody
• [asterisk-users] Hangup on Panasonic KX-TEM824   Jorge Mendoza
• [asterisk-users] Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???   Jorge Mendoza
• [asterisk-users] Setting system time via Asterisk   Bill Merriam
• [asterisk-users] Page() paging application problem   Michael
• [asterisk-users] Page() paging application problem   Michael
• [asterisk-users] Mediatrix 1204   Bill Michaelson
• [asterisk-users] Re: Mediatrix 1204 trix   Bill Michaelson
• [asterisk-users] Query about Call Detail Record in Asterisk   Chan Kwang Mien
• [asterisk-users] is there anyone working with 5ESS?   Luiz Miguel
• [asterisk-users] is there anyone working with 5ESS?   Luiz Miguel
• [asterisk-users] Adding custom fields (more than one) to CDR DB   Mike
• [asterisk-users] Is this a warning or not...MYSQL Fetch   Mike
• [asterisk-users] Asterisk hangs up after 10-15 minutes when SIP Phone is on mute   Mike
• [asterisk-users] Re: Asterisk hangs up after 10-15 minutes when SIPPhone is on mute   Mike
• [asterisk-users] Re: Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute   Mike
• [asterisk-users] Asterisk hangs up after 10-15 minuteswhenSIPPhone is on mute   Mike
• [asterisk-users] How can I set CDR data in dialplan? Set(CDR(src)=foo)   Mike
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] Another (quick) Polycom 501 question   Mike
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] Polycom 501 - message waiting LED manipulation   Mike
• [asterisk-users] Asterisk variables   Mike
• [asterisk-users] How to make Polycom 501 go off hook when pressing any digits   Mike
• [asterisk-users] Astbill DIALSTRING doesn't work   Sebastian Milioto
• [asterisk-users] One way audio problem on gateway to PSTN after some time, no NAT involved   Kai Militzer
• [asterisk-users] 911 versus 9.911   Jay Milk
• [asterisk-users] Toll-Free numbers   Jay Milk
• [asterisk-users] US Toll-Free DID Providers with Caller ID NAME?   Jay Milk
• {Fraud?} RE: [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC isarrested at JFK!!   Jay Milk
• [asterisk-users] FW: Peter Dicks Chairman ofSportingbet PLC is arrested at JFK!!   Jay Milk
• [asterisk-users] Cisco 7970 behind NAT   Jeremiah Millay
• [asterisk-users] Cisco 7970 behind NAT   Jeremiah Millay
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Avi Miller
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Avi Miller
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Avi Miller
• [asterisk-users] latest CentOS-asterisk-freepbx installation procedure   Avi Miller
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   Avi Miller
• [asterisk-users] Noob question: Packet size   Avi Miller
• [asterisk-users] Uninstalling Trixbox   Avi Miller
• [asterisk-users] Trixbox Documentation   Avi Miller
• [asterisk-users] channel.c: Nobody there, continuing...   Chris Miller
• [asterisk-users] channel.c: Nobody there, continuing...   Chris Miller
• [asterisk-users] Unrecognized frames   Chris Miller
• [asterisk-users] Polycom Soundpoint Key Remap   Noah Miller
• [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry   Noah Miller
• [asterisk-users] Polycom Soundpoint Key Remap   Noah Miller
• [asterisk-users] Polycom programmable buttons   Noah Miller
• [asterisk-users] Polycom Expansion Module   Noah Miller
• [asterisk-users] Polycom programmable buttons   Noah Miller
• [asterisk-users] Digium GUI?   Noah Miller
• [asterisk-users] Polycom 2.0.1 Software   Noah Miller
• [asterisk-users] Polycom phone help needed   Noah Miller
• [asterisk-users] balance anouncement   John Millican
• [asterisk-users] balance anouncement   John Millican
• [asterisk-users] Zapata.conf   Danko Miocevic
• [asterisk-users] Submenus   Mir
• [asterisk-users] Submenus   Mir
• [asterisk-users] Outgoing callerid in AMI   Mir
• [asterisk-users] Calls on hold   Mir
• [asterisk-users] Adding own info in AMI   Mir
• [asterisk-users] Re: Calls on hold   Mir
• [asterisk-users] ISDN config EWSD   Virmones Pereira Tavares de Miranda
• [asterisk-users] Polycom IP430 sound level too low?   Louis-David Mitterrand
• [asterisk-users] corrupt faxes   Louis-David Mitterrand
• [asterisk-users] importance of crc4 in zaptel.conf?   Louis-David Mitterrand
• [asterisk-users] DSL router with integrated SIP proxy?   Roy-Magne Mo
• [asterisk-users] changed behaviour of status indication on incoming pri lines in asterisk 1.2?   Christian Mohrbacher
• [Asterisk-Users] Can you explain why multiple registration is an important (missing) feature ?   Christian Mohrbacher
• [Asterisk-Users] Can you explain why multiple registration is an important (missing) feature ?   Christian Mohrbacher
• [asterisk-users] Re: Re: 911 versus 9.911   Mojo with Horan & Company, LLC
• [asterisk-users] Submenus   Mojo with Horan & Company, LLC
• [asterisk-users] Submenus   Mojo with Horan & Company, LLC
• [asterisk-users] Really bad phone line.. possible causes?   Mojo with Horan & Company, LLC
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   Mojo with Horan & Company, LLC
• [asterisk-users] Re: Really bad phone line.. possible causes?   Mojo with Horan & Company, LLC
• [asterisk-users] Re: Really bad phone line.. possible causes?   Mojo with Horan & Company, LLC
• [asterisk-users] Remote tone access   Mojo with Horan & Company, LLC
• [Asterisk-Users] How to learn or teach VoIP QoE   Mojo with Horan & Company, LLC
• [asterisk-users] IAX or SIP termination provider that reaches 6421xxxxxxx?   Mojo with Horan & Company, LLC
• [asterisk-users] IAX or SIP termination provider that reaches 6421xxxxxxx?   Mojo with Horan & Company, LLC
• [asterisk-users] Polycom 500 power supply   Mojo with Horan & Company, LLC
• [asterisk-users] Looped message playback   Mojo with Horan & Company, LLC
• [asterisk-users] IAX or SIP termination provider that reaches 6421xxxxxxx?   Mojo with Horan & Company, LLC
• [asterisk-users] "does /var/run/asterisk.ctl exist?" -- but Asterisk *is* running.   Mojo with Horan & Company, LLC
• [asterisk-users] File structure question   Jay Moore
• [asterisk-users] File structure question   Jay Moore
• [asterisk-users] File structure question   Jay Moore
• [asterisk-users] Asterisk with cisco 7935   Lacy Moore
• [asterisk-users] Re: Voip Buster - CID   Lacy Moore
• [asterisk-users] pstn failback   Lacy Moore
• [asterisk-users] SHSU asterisk installation?   Melcon Moraes
• [asterisk-users] How to Dial a number with Sangoma PRI card?   Melcon Moraes
• [asterisk-users] señalizacion te110p, signaling te110p   Melcon Moraes
• [asterisk-users] Asterisk IVR .wav issue   David Moring
• [asterisk-users] Asterisk IVR .wav issue   David Moring
• [asterisk-users] Merlin Legend - Working Now!   Sterling Moses
• [asterisk-users] Merlin Legend - Working Now!   Sterling Moses
• [asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...   Tony Mountifield
• [asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...   Tony Mountifield
• [asterisk-users] Re: Asterisk hangs up after 10-15 minutes when SIP Phone is on mute   Tony Mountifield
• [asterisk-users] Re: Tracking the source of a disconnect?   Tony Mountifield
• [asterisk-users] Re: ztdummy installed but choppy audio warning on load   Tony Mountifield
• [asterisk-users] TE411P or TE412P?   Tony Mountifield
• [asterisk-users] Re: chan_zap.so stopped working after upgrading CentOS   Tony Mountifield
• [asterisk-users] Re: chan_zap.so stopped working after upgrading CentOS   Tony Mountifield
• [asterisk-users] 4-wire analogue interfaces?   Tony Mountifield
• [asterisk-users] Re: 4-wire analogue interfaces?   Tony Mountifield
• [asterisk-users] help me!!Problem on incoming calls   Marco Mouta
• [asterisk-users] help me!!Problem on incoming calls   Marco Mouta
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Marco Mouta
• [asterisk-users] help me!!Problem on incoming calls   Marco Mouta
• [asterisk-users] Probelm with incoming calls to my DID-Please help me   Marco Mouta
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Marco Mouta
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Marco Mouta
• [asterisk-users] Architecture:MainOffice(AstServer)-SmallOffices(ATA.-LegacyPBX)   Marco Mouta
• [asterisk-users] Any Hardphone with VPNClient embedded?   Marco Mouta
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Marco Mouta
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Marco Mouta
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Marco Mouta
• [asterisk-users] FAX handling   Marco Mouta
• [asterisk-users] File structure question   Marco Mouta
• [asterisk-users] why executed Hangup doesn't exit DialPlan?look my dialplan...   Marco Mouta
• [asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...   Marco Mouta
• [asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...   Marco Mouta
• [asterisk-users] latest CentOS-asterisk-freepbx installation procedure   Marco Mouta
• [asterisk-users] File structure question   Marco Mouta
• [asterisk-users] ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server   Marco Mouta
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Marco Mouta
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Marco Mouta
• [asterisk-users] core dumps   Marco Mouta
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Marco Mouta
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Marco Mouta
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Marco Mouta
• [asterisk-users] app_rxfax Only Receives One Page   Marco Mouta
• [asterisk-users] app_rxfax Only Receives One Page   Marco Mouta
• [asterisk-users] Handling incoming calls from VoIPbuster   Marco Mouta
• [asterisk-users] Forward recorded voicemail message to more than one extension using sendvoicemail=yes   Marco Mouta
• [asterisk-users] Deploying an IVR - direct extensions.conf or AGI scripts?   Marco Mouta
• [asterisk-users] SHSU asterisk installation?   Marco Mouta
• [asterisk-users] When does Scalability requests Asterisk to Use SER ?   Marco Mouta
• [asterisk-users] grandstream gxp 2000 does not display names when calling out   Marco Mouta
• [asterisk-users] UK (BT) Problem with TDM 400P   Mark Muffett
• [asterisk-users] UK (BT) Problem with TDM 400P   Mark Muffett
• [asterisk-users] UK (BT) Problem with TDM 400P   Mark Muffett
• [asterisk-users] UK (BT) Problem with TDM 400P   Mark Muffett
• [asterisk-users] SkypeOut with Asterisk?   Devraj Mukherjee
• [asterisk-users] SkypeOut with Asterisk?   Devraj Mukherjee
• [asterisk-users] Asterisk Core dump   Anthony Musaluke
• [asterisk-users] core dumps   Anthony Musaluke
• [asterisk-users] Queue and Pickup/DPickup   Ole Myhre
• [asterisk-users] modifying the INVITE headers   Dinesh Nair
• [asterisk-users] Makefile.moddir_rules: No such file or directory   Dinesh Nair
• [asterisk-users] 488 Not acceptable here sent by Asterisk - SIP debug follows   Dinesh Nair
• [asterisk-users] 488 Not acceptable here sent by Asterisk - SIPdebug follows   Dinesh Nair
• [asterisk-users] 488 Not acceptable here sent by Asterisk - SIPdebug follows   Dinesh Nair
• [asterisk-users] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] Re: [asterisk-dev] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] Re: [asterisk-dev] G729 Replacement Codec - FREE or may ne cheaperthan existing one.   Kannaiyan Natesan
• [asterisk-users] Re: [asterisk-biz] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] Re: [asterisk-biz] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] [asterisk-dev] Re: [asterisk-biz] G729 Replacement Codec - FREE ormay ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] Digum g729 and g723   Kannaiyan Natesan
• [asterisk-users] Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-dev] Re: [asterisk-users] Digum g729 and g723   Kannaiyan Natesan
• [asterisk-dev] Re: [asterisk-users] Digum g729 and g723   Kannaiyan Natesan
• [asterisk-users] Forwarding   Leif Neland
• [asterisk-users] Satellite link-IAX Jitter Buffer.   ANANGWE Nelson
• [asterisk-users] Cisco PIX firewall and nat=yes   Peder at NetworkOblivion
• [asterisk-users] Cisco GW & CID Name   Peder at NetworkOblivion
• [Asterisk-Users] Suggestion for directed pickup in bristuffed 1.2 Asterisk   Michael Neuhauser
• [asterisk-users] is chanisavail command reliable?   Michael Neuhauser
• [asterisk-users] grandstream gxp 2000 does not display names when calling out   Michael Neuhauser
• [asterisk-users] grandstream gxp 2000 does not display names when calling out   Michael Neuhauser
• [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out   Michael Neuhauser
• [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out   Michael Neuhauser
• [asterisk-users] extensions.conf strangeness   Michael Neuhauser
• [asterisk-users] Looks like Nufone is changing around...   Justin Newman
• [asterisk-users] Re: Nufone making changes   Justin Newman
• [asterisk-users] Re: FAX handling   Justin Newman
• [asterisk-users] Find-Me/Follow-ME   Marnus van Niekerk
• [asterisk-users] USA Regulatons   Marnus van Niekerk
• [asterisk-users] WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long]   Marnus van Niekerk
• [asterisk-users] Sirrix to Legacy PBX   Marnus van Niekerk
• [asterisk-users] BLF across asterisk trunks   Norris, Sam
• [asterisk-users] BLF across asterisk trunks   Norris, Sam
• [asterisk-users] question...   John Novack
• [asterisk-users] Reliability of the newer IAXy's   John Novack
• [asterisk-users] outgoing call problem   John Novack
• [asterisk-users] TDM400P Problem or Not?   John Novack
• [asterisk-users] polycom 501 digitmap   Jordan Novak
• [asterisk-users] TDM2400P   Jordan Novak
• [asterisk-users] multiple zaptel cards   Jordan Novak
• [asterisk-users] IAX and rsa   Andrew Nowrot
• [asterisk-users] amr codec   Net Nut
• [asterisk-users] amr codec   Net Nut
• [asterisk-users] amr codec   Net Nut
• [asterisk-users] Re: Dual core   Massimo Nuvoli
• [asterisk-users] 1.4 Beta 2 Config Problem   Keith O'Brien
• [asterisk-users] RE: 1.4 Beta 2 Config Problem   Keith O'Brien
• [asterisk-users] RE: 1.4 Beta 2 Config Problem   Keith O'Brien
• [asterisk-users] RE: 1.4 Beta 2 Config Problem   Keith O'Brien
• [asterisk-users] LDAP athentication   Andre O.
• [asterisk-users] DIAL and automatic/manual co line acces   Kai Ober
• [asterisk-users] mISDN versus ZapHFC with BRIstuff   Kai Ober
• [asterisk-users] WAIT FOR DIGIT not working   Kai Ober
• [asterisk-users] callback without agi   Kai Ober
• [asterisk-users] User authentication   Kai Ober
• [asterisk-users] Playtones   Kai Ober
• [asterisk-users] ANI and Meetme...   Natambu Obleton
• [asterisk-users] Fedora   Natambu Obleton
• [asterisk-users] Habitual set of number   Vitaly Oborsky
• [asterisk-users] Asterisk Appliance, will Asterisk Business Edition be mandatory?   Info Oceania
• [asterisk-users] 7940 vs. 7941   Greg Oliver
• [asterisk-users] Cisco CAll Manger and H323   Greg Oliver
• [asterisk-users] Cisco CAll Manger and H323   Greg Oliver
• [Asterisk-Users] Which SIP hardphone with embedded VPNClient ?   Olivier
• [Asterisk-Users] SIP parameter to prevent a call from being added in missed calls logs   Olivier
• [Asterisk-Users] SIP hardphones and BLF monitoring keys   Olivier
• [Asterisk-Users] SIP parameter to prevent a call from being added in missed calls logs   Olivier
• [Asterisk-Users] Browsing distant missed call list   Olivier
• [Asterisk-Users] The best way to design local-only off-hours ringing   Olivier
• [Asterisk-Users] Junghanns BRI cards and misdn   Olivier
• [Asterisk-Users] Which SIP hardphone implements RTCP XR (aka RFC3611)   Olivier
• [Asterisk-Users] Suggestion for directed pickup in bristuffed 1.2 Asterisk   Olivier
• [Asterisk-Users] Junghanns BRI cards and misdn   Olivier
• [Asterisk-Users] Suggestion for directed pickup in bristuffed 1.2 Asterisk   Olivier
• [Asterisk-Users] Can you explain why multiple registration is an important (missing) feature ?   Olivier
• [Asterisk-Users] Shared Line Appearance, Snom and trunk   Olivier
• [asterisk-users] pickup call little complicated   Olivier
• [Asterisk-Users] How to learn or teach VoIP QoE   Olivier
• [Asterisk-Users] CallerID retain on internal transfer   Olivier
• [asterisk-users] BRI: Asterisk disconnecting on 'call diverted' message?   Olivier
• [asterisk-users] University dumps CISCO VoIP for Asterisk   Olivier
• [Asterisk-Users] Can you explain why multiple registration is an important (missing) feature ?   Olivier
• [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?   Olivier
• [Asterisk-Users] How to learn or teach VoIP QoE   Olivier
• [Asterisk-Users] Where to find error codes   Olivier
• [Asterisk-Users] Network impairment tools   Olivier
• [Asterisk-Users] The best way to track no-audio calls   Olivier
• asterisk t.38 (was RE: [asterisk-users] trixbox t38 pass through)   Olivier
• [Asterisk-Users] Off-hooking Snom hanset doesn't answer incoming call   Olivier
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   J. Oquendo
• [asterisk-users] Call parking and ringbacks   J. Oquendo
• [asterisk-users] sip and iax over the internet (asterisk to asterisk) drop outs normal???   J. Oquendo
• [asterisk-users] asterisk to cell phone network   J. Oquendo
• [asterisk-users] Asterisk behind Sonicwall firewall   J. Oquendo
• [asterisk-users] Mediatrix 1204   Florian Overkamp
• [asterisk-users] blf aastra 9133i working but can't pickup calls   Gareth Owen
• [asterisk-users] blf aastra 9133i working but can't pickup calls   Gareth Owen
• [asterisk-users] 480i phone: Is there a trick to registering with *??   Gareth Owen
• [asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54   FRANCISCO PEREZ-LANDAETA
• [asterisk-users] Quintum tenor configuration with asterisk help   FRANCISCO PEREZ-LANDAETA
• [asterisk-users] QUINTUM CONFIGURATION.-   FRANCISCO PEREZ-LANDAETA
• [asterisk-users] QUINTUM TENOR ASM200 Configuration   FRANCISCO PEREZ-LANDAETA
• [asterisk-users] about 'zap show channels'   PSPunch
• [asterisk-users] about 'zap show channels'   PSPunch
• [asterisk-users] Unable to make calls from CallManager to Asterisk   Anantha Padmanabha.M.L
• [asterisk-users] Urgent !!!! Unable to make calls from cisco callmanager to asterisk   Anantha Padmanabha.M.L
• [asterisk-users] How to send correct Caller ID on PRI   George Pajari
• [asterisk-users] Asterisk / Patton SmartNode SN2400 Strangeness FYI   George Pajari
• [asterisk-users] Enabling Second Processor Trashes Audio Quality   George Pajari
• [asterisk-users] Anyone Using a Patton (Inalp) SmartNode 2400 for T.38?   George Pajari
• [asterisk-users] Long Delay in IAX Calls   Jonathan Palley
• [asterisk-users] BRI: Asterisk disconnecting on 'call diverted' message?   Benoit Panizzon
• [asterisk-users] Toll-Free numbers   Tim Panton
• [asterisk-users] How to manipulate a plus in a phone number   Tim Panton
• [asterisk-users] How to manipulate a plus in a phone number   Tim Panton
• [asterisk-users] Re: How to manipulate a plus in a phone number   Tim Panton
• [asterisk-users] MSSQL connection   Tim Panton
• [asterisk-users] Problems with outgoing calls   Tim Panton
• [asterisk-users] Via Epia platforms and asterisk   Tim Panton
• [asterisk-users] amr codec   Tim Panton
• [asterisk-users] Starting out   Tim Panton
• [asterisk-users] amr codec   Tim Panton
• [asterisk-dev] Re: [asterisk-users] OT But So Ungodly Important   Tim Panton
• [asterisk-users] ISDN30 and digital phones   Tim Panton
• [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P   Tim Panton
• [asterisk-users] Yet another processor question   Tim Panton
• [asterisk-users] AGI Errors   Tim Panton
• [asterisk-users] t1-pri or sip trunk?   Tim Panton
• [asterisk-users] recommended application for salesman using asterisk   Tim Panton
• [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN   Tim Panton
• [asterisk-users] Building the Perfect Box   Tim Panton
• [asterisk-users] Starting out   Timothy Parez
• [asterisk-users] why executed Hangup doesn't exit DialPlan?look my dialplan...   Jason Parker
• [asterisk-users] Digium G.729 codec binaries updated for Asterisk 1.4 beta   Jason Parker
• [asterisk-users] Missing sound in spanish from 1.4 beta2   Jason Parker
• [asterisk-users] MOH in 1.4 - Still Broken?   Jason Parker
• [asterisk-users] Digium G.729 codec binaries updated for Asterisk 1.4 on Solaris   Jason Parker
• [asterisk-users] IAX2 trunk problem   Enrico Pasqualotto
• [asterisk-users] Bad number - is not in inbound speed dial   Enrico Pasqualotto
• [asterisk-users] Reload question   Patrick
• [asterisk-users] How to configure Fritz ISDN2 card with Trixbox?   Patrick
• [asterisk-users] HFC isdn card and bristuff 0.2.0 rc8n   Patrick
• [asterisk-users] g729 problem   Patrick
• [asterisk-users] Silence Call {very very urgent plz}   Patrick
• [asterisk-users] [asterisk-dev] 491 request pending   Patrick
• [asterisk-users] Alcatel OXO Sip   Patrick
• [asterisk-users] Cisco 7960 part numbers ...   Patrick
• [asterisk-users] Problem with Asterisk Realtime (MySQL)   Edwin Pauli
• [asterisk-users] Cisco 7960 won't download dialplan.xml   Peter Pauly
• [asterisk-users] SIP client with video???   Joao Pereira
• [asterisk-users] Asterisk video support   Joao Pereira
• [asterisk-users] How much SIP calls can I squeeze from this box   Erick Perez
• [asterisk-users] Mini call center only 15 seats fxs to sip suggestion   Erick Perez
• [asterisk-users] Are you using app_meetme or app_conference   Erick Perez
• [asterisk-users] Outgoing Call group ?   Noc Phibee
• [asterisk-users] Dropping extra frame of G.729 ?   Noc Phibee
• [asterisk-users] Asterisk and NAT ?   Noc Phibee
• [asterisk-users] Asterisk and NAT ?   Noc Phibee
• [asterisk-users] Asterisk and "Maximum retries exceeded"   Noc Phibee
• [asterisk-users] Asterisk and "Maximum retries exceeded"   Noc Phibee
• [asterisk-users] Bri Card for Asterisk ?   Noc Phibee
• [asterisk-users] g729 failover when out of licenses   Mark Phillips
• [asterisk-users] Capacity for transcode G711 to G729   Mark Phillips
• [asterisk-users] iaxy will register, but doesn't detect POTS line   Wilson Pickett
• [asterisk-users] voicemail as email and attachment   Tim St. Pierre
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Tim St. Pierre
• [asterisk-users] Can QUEUE member be assigned from a GlobalVar set in EXTENSIONS.CONF?   Tim St. Pierre
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Tim St. Pierre
• [asterisk-users] What does 'trunk' mean in outgoing and incoming?   Tim St. Pierre
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Tim St. Pierre
• [asterisk-users] Blind transfer 3/4 digits   Tim St. Pierre
• [asterisk-users] How to send correct Caller ID on PRI   Tim St. Pierre
• [asterisk-users] Roundrobin not working on PRI   Tim St. Pierre
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Tim St. Pierre
• [asterisk-users] Caller ID has extra digits to strip   Tim St. Pierre
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Tim St. Pierre
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Tim St. Pierre
• [asterisk-users] using SIP to connect remote other VoIP server   Tim St. Pierre
• [asterisk-users] cmd SET time value   Tim St. Pierre
• [asterisk-users] Call Forwarding in SIP.conf   Tim St. Pierre
• [asterisk-users] cmd SET time value   Tim St. Pierre
• [asterisk-users] sip peer question   Tim St. Pierre
• [asterisk-users] Call Forwarding in SIP.conf   Tim St. Pierre
• [asterisk-users] Call Forwarding in SIP.conf   Tim St. Pierre
• [asterisk-users] What don't I get about SIP?   Tim St. Pierre
• [asterisk-users] distinguishing users by their domain   Tim St. Pierre
• [asterisk-users] No dialtone, just directly busy   Tim St. Pierre
• [asterisk-users] What don't I get about SIP?   Tim St. Pierre
• [asterisk-users] Auto Dialer question   Tim St. Pierre
• [asterisk-users] No dialtone, just directly busy   Tim St. Pierre
• [asterisk-users] Call Forwarding in SIP.conf   Tim St. Pierre
• [asterisk-users] Scope of contexts   Tim St. Pierre
• [asterisk-users] Whcih phones are better for mass deployment   Tim St. Pierre
• [asterisk-users] Whcih phones are better for mass deployment   Tim St. Pierre
• [asterisk-users] Voip providers and sip origination and termination?   Tim St. Pierre
• [asterisk-users] music onhold choppy music problems   Tim St. Pierre
• [asterisk-users] can someone recommend a voip provider that...   Tim St. Pierre
• [asterisk-users] Max Size of Conf Files   Tim St. Pierre
• [asterisk-users] using residential voip for business?   Tim St. Pierre
• [asterisk-users] Setting system time via Asterisk   Tim St. Pierre
• [asterisk-users] Context   Tim St. Pierre
• [Asterisk-Users] SIP parameter to prevent a call from being added in missed calls logs   Tim St. Pierre
• [asterisk-users] SIP trunk   Tim St. Pierre
• [asterisk-users] Remote tone access   Tim St. Pierre
• [asterisk-users] missing pri connect (wwomera to pri)   Rosario Pingaro
• [asterisk-users] rxfax, spandsp and lack of ecm   Rosario Pingaro
• [asterisk-users] rxfax, spandsp and lack of ecm   Rosario Pingaro
• [asterisk-users] Ringtones   Scott Pinhorne
• [asterisk-users] ASTERFAX   Scott Pinhorne
• [asterisk-users] Realtime madness   Scott Pinhorne
• [asterisk-users] Outgoing DialPlan   Scott Pinhorne
• [asterisk-users] Missing Agent Function   William Piper
• [asterisk-users] question of CLI   William Piper
• [asterisk-users] Outgoing Call group ?   William Piper
• [asterisk-users] Help with blind transfer   William Piper
• [asterisk-users] Query on Call Forward Feature codes for SIP users..   William Piper
• [asterisk-users] Asterisk and NAT ?   William Piper
• [asterisk-users] Call Forwarding in SIP.conf   William Piper
• [asterisk-users] Grandstream, how to use the configuration tool   William Piper
• [asterisk-users] Grandstream GX-2000 Remote Login Problem   William Piper
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   William Piper
• [asterisk-users] How to integrate freepbx with a2billing?   William Piper
• [asterisk-users] A simple goal, help me please!   William Piper
• [asterisk-users] sip origination and termination   William Piper
• [asterisk-users] SPA 3102 does not even attempt to register   William Piper
• [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN   William Piper
• [asterisk-users] Open source G.729 and G.723.1 release for 1.2 and 1.4   Daniel Pocock
• [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute   Daniel Pocock
• [asterisk-users] Re: FW: Peter Dicks Chairman ofSportingbet PLC isarrested at JFK!!   Daniel Pocock
• [asterisk-users] Intel Based G.729 and SVN-trunk-r42453   Daniel Pocock
• [asterisk-users] Intel Based G.729 and SVN-trunk-r42453   Daniel Pocock
• [asterisk-users] ztdummy installed but choppy audio warning on load   Daniel Pocock
• [asterisk-users] University switches to Asterisk   Porier, Jeremy M.
• [asterisk-users] ztdummy installed but choppy audio warning   Juanjo Portela
• [asterisk-users] Configuring Asterisk 1.4-beta2 to work with jingle   Raffaele Porzio
• [asterisk-users] How can I unistall Asterisk?   Raffaele Porzio
• [asterisk-users] How to enable jingle in 1.4beta2?   Raffaele Porzio
• [asterisk-users] How to enable jingle in 1.4beta2?   Raffaele Porzio
• [asterisk-users] Silence Call {very very urgent plz}   Vamsi Pottangi
• [asterisk-users] DSL router with integrated SIP proxy?   Vamsi Pottangi
• [asterisk-users] HITBSecConf2006 Final Call !   Praburaajan
• [asterisk-users] new in 1.4?   Joe Pukepail
• [asterisk-users] Re: Dual core   Joe Pukepail
• [asterisk-users] Asterisk Design Question   Remi Quezada
• [asterisk-users] Early rtp bridge and reINVITE in 1.4b2   H Quintana
• [asterisk-users] A simple goal, help me please!   David R.
• [asterisk-users] Asterisk AGI question   David R.
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] includes in realtime ??   RR
• [asterisk-users] includes in realtime ??   RR
• [asterisk-users] includes in realtime ??   RR
• [asterisk-users] includes in realtime ??   RR
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Capacity for transcode G711 to G729   RR
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Capacity for transcode G711 to G729   RR
• [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?   RR
• [asterisk-users] voicemail access thru apache on another server   RR
• [asterisk-users] G729 and Tribox   RR
• [asterisk-users] Why not g726-32?   RR
• [asterisk-users] ASTERISK HIGH AVAILABILITY   RR
• [asterisk-users] Why not g726-32?   RR
• [asterisk-users] Why not g726-32?   RR
• [asterisk-users] Why not g726-32?   RR
• [asterisk-users] Fedora   Aryanto Rachmad
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Elpidio Ramos
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Elpidio Ramos
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Elpidio Ramos
• [asterisk-users] using SIP to connect remote other VoIP server   Elpidio Ramos
• [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)   Elpidio Ramos
• [asterisk-users] Attended transfer and parking calls   Elpidio Ramos
• [asterisk-users] Attended transfer and parking calls   Elpidio Ramos
• [asterisk-users] CURL   Elpidio Ramos
• [asterisk-users] CURL   Elpidio Ramos
• [asterisk-users] CURL   Elpidio Ramos
• [asterisk-users] Correct settings for UK (BT) FXO   Faris Raouf
• [asterisk-users] Grandstream SX2000 attended tranfer   Faris Raouf
• [asterisk-users] Asterisk AGI and Firebird   Steve Rawlings
• [asterisk-users] Dell hardware ...   Steve Rawlings
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Bruce Reeves
• [asterisk-users] Polycom new firmware and bootrom   Bruce Reeves
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Bruce Reeves
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Bruce Reeves
• [asterisk-users] rxfax, spandsp and lack of ecm   Bruce Reeves
• [asterisk-users] new in 1.4?   Bruce Reeves
• [asterisk-users] Master.csv has stopped writing call logs.   Richard Reina
• [asterisk-users] Master.csv has stopped writing call logs.   Richard Reina
• [asterisk-users] Scope of contexts   Rene
• [asterisk-users] SIP/2.0 403 Relaying denied   Rene
• [asterisk-users] Virtualise asterisk on Xen   Rene
• [asterisk-users] Asterisk AGI question   Stefan Reuter
• [asterisk-users] SOS building fastagi C   Stefan Reuter
• [asterisk-users] TDM2400P   Robson Ribeiro
• [asterisk-users] TDM2400P   Robson Ribeiro
• [asterisk-users] TDM2400P   Robson Ribeiro
• [asterisk-users] TDM2400 wired description and skiping frames   Robson Ribeiro
• [asterisk-users] TDM2400P   Robson Ribeiro
• [asterisk-users] TDM2400 wired description and skiping frames   Robson Ribeiro
• [asterisk-users] TDM2400 problem isolated with POLYCOM IP301 phones!!!   Robson Ribeiro
• [asterisk-users] TDM2400P and Polycom phones   Robson Ribeiro
• [asterisk-users] SER with multiple asterisk deployment   Simone Ricci
• [asterisk-users] Re: Re: 911 versus 9.911   Jim Rice
• [asterisk-users] Polycom 501 display bug   Jim Rice
• [asterisk-users] TDM400 and T100 config on same asterisk   Rich
• [asterisk-users] iaxyprov downloading problems   Hadley Rich
• [asterisk-users] Chanspy crashing the server, again   Richard
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Richard
• [asterisk-users] Unable to make calls from CallManager to Asterisk   Gary Richardson
• [asterisk-users] LDAP athentication   Gary Richardson
• [asterisk-users] Cable Systems ICS-G302, Anyone have an Admin Guide Please?   JR Richardson
• [asterisk-users] Fedora   Jaymz Ringler
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Steven Ringwald
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Steven Ringwald
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Steven Ringwald
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Steven Ringwald
• [asterisk-users] Adding custom fields (more than one) to CDR DB   Steven Ringwald
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   Steven Ringwald
• [asterisk-users] Re: uConnect Voip device   Alex Robar
• [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC is arrested at JFK!!   Alex Robar
• [asterisk-users] FW: Peter Dicks Chairman ofSportingbet PLC is arrested at JFK!!   Alex Robar
• Re: [asterisk-users] GSM Gateway Promotion from £69GBP   Alex Robar
• [asterisk-users] Blind transfer 3/4 digits   Anthony Rodgers
• [asterisk-users] Asterisk and "Maximum retries exceeded"   Anthony Rodgers
• [asterisk-users] What don't I get about SIP?   Anthony Rodgers
• [asterisk-users] How to install HUDLite Server   Anthony Rodgers
• [asterisk-users] How to make Polycom 501 go off hook when pressingany digits   Anthony Rodgers
• [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P   Anthony Rodgers
• [asterisk-users] Issues with AGI+Dial command   Brian Rogan
• [asterisk-users] Starting out   Brian Rogan
• [asterisk-users] Issues with AGI+Dial command   Brian Rogan
• [asterisk-users] OT: Opinions on Aastra 480i CT?   Brian Rogan
• [asterisk-users] Running Multiple Instances of Asterisk   Brian Rogan
• [asterisk-users] Running Multiple Instances of Asterisk   Brian Rogan
• [asterisk-users] latest CentOS-asterisk-freepbx installation procedure   Roland
• [asterisk-users] MOH distorted on Pound Key Linux on asterisk 1.2.8   Jeronimo Romero
• [asterisk-users] MOH distorted on Pound Key Linux on asterisk1.2.8   Jeronimo Romero
• [asterisk-users] MOH distorted on Pound Key Linux on asterisk1.2.8   Jeronimo Romero
• [asterisk-users] WAS: 64 analog phones NOW: Selection criteria and recipie for a good Asterisk install [long]   Jeronimo Romero
• [asterisk-users] sip show peers   Eric Rousse
• [asterisk-users] sip show peers   Eric Rousse
• [asterisk-users] include "context"   Eric Rousse
• [asterisk-users] Rpath PoundKey 1.2   Brian Roy
• [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic   Robert Rozman
• [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problemwith module versionmagic   Robert Rozman
• [asterisk-users] How to stream audio to external app for speech recognition and recognize dtmf in parallel ?   Robert Rozman
• [asterisk-users] Help compiling asterisk-addons on Debian?   Rushowr
• [asterisk-users] Keys pressed not registering ...   Rushowr
• [asterisk-users] Zaptel-1.2.8 compile problem   Rushowr
• [asterisk-users] includes in realtime ??   Rushowr
• [asterisk-users] Conditional IF based on IP address?   Rushowr
• [asterisk-users] Conditional IF based on IP address?   Rushowr
• [asterisk-users] Conditional IF based on IP address?   Rushowr
• [asterisk-users] Response to KP Flemming...   Rushowr
• [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?   Rushowr
• [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?   Rushowr
• [asterisk-users] sip peer question   Rushowr
• [asterisk-users] What don't I get about SIP?   Rushowr
• [asterisk-users] What don't I get about SIP?   Rushowr
• [asterisk-users] What don't I get about SIP?   Rushowr
• [asterisk-users] Auto Dialer question   Rushowr
• [asterisk-users] using residential voip for business?   Rushowr
• [asterisk-users] using residential voip for business?   Rushowr
• [asterisk-users] Asterisk Realtime Arch - static or realtime?   Rushowr
• [asterisk-users] Asterisk Realtime Arch - static or realtime?   Rushowr
• [asterisk-users] Static RealTime - SIP.CONF   Rushowr
• [asterisk-users] Fedora   Rushowr
• [asterisk-users] ANI and Meetme...   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to Use SER ?   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Rushowr
• [asterisk-users] RE: FollowMe question   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to Use SER ?   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Rushowr
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   Rushowr
• [asterisk-users] OT But So Ungodly Important   Rushowr
• [asterisk-users] OT But So Ungodly Important   Rushowr
• [asterisk-users] OT But So Ungodly Important   Rushowr
• [asterisk-dev] Re: [asterisk-users] OT But So Ungodly Important   Rushowr
• [asterisk-users] OT But So Ungodly Important   Rushowr
• [asterisk-users] call notification for queues?   Rajkumar S
• [asterisk-users] Mediant 1000   Rajkumar S
• [asterisk-users] Picking up a call from queue?   Rajkumar S
• [asterisk-users] recommended application for salesman using asterisk   Yu Safin
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Alberto Sagredo
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Alberto Sagredo
• [asterisk-users] How to Install H323   Alberto Sagredo
• [asterisk-users] Call Processing Slow 11 seconds   Alberto Sagredo
• [asterisk-users] Whcih phones are better for mass deployment   Alberto Sagredo
• [asterisk-users] QUINTUM TENOR ASM200 Configuration   Alberto Sagredo
• [asterisk-users] asterisk logging per day   Alberto Sagredo
• [asterisk-users] Choppy MOH (Cisco gateway)   Alberto Sagredo
• [asterisk-users] Attended Transfer Asterisk 1.2.11   Alberto Sagredo
• [asterisk-users] Linksys SPA400   Alberto Sagredo
• [asterisk-users] Linksys SPA400   Alberto Sagredo
• [asterisk-users] señalizacion te110p, signaling te110p   Alberto Sagredo
• [asterisk-users] Google talk and Asterisk 1.4   Alberto Sagredo
• [asterisk-users] Google talk and Asterisk 1.4   Alberto Sagredo
• [asterisk-users] Grandstream GX-2000, doesn't send calls to free lines   Daniel Salama
• [asterisk-users] GXP2000 - Blind Transfer Hangs Up Call   Daniel Salama
• [asterisk-users] Grandstream SX2000 attended tranfer   Daniel Salama
• [asterisk-users] Zap channel digit.   Luca Salemmi
• [asterisk-users] AGI Errors   Edmilson Santana
• [asterisk-users] AGI Errors   Edmilson Santana
• [asterisk-users] Re: Dropping extra frame of G.729 ?   Rosa De Santis
• [asterisk-users] Query on Call Forward Feature codes for SIP users..   A C Sathish-a22713
• [asterisk-users] Catch an event   Olivier Saulnier
• [asterisk-users] balance anouncement   Kevin Savoy
• [BULK] Re: [asterisk-users] Prompts recording for Asterisk   Savoy, Kevin - Williston, ND
• [asterisk-users] Detect PBX vs Network message   Savoy, Kevin - Williston, ND
• [asterisk-users] Re: Detect PBX vs Network message   Savoy, Kevin - Williston, ND
• [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P   Savoy, Kevin - Williston, ND
• [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry   Gunnar Schaller
• [asterisk-users] asterisk / chan_capi problems   Armin Schindler
• [asterisk-users] asterisk / chan_capi problems   Armin Schindler
• [asterisk-users] core dump with 1.2.7.1 and chan-capi-cm 0.6.5   Armin Schindler
• [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0   Armin Schindler
• [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0   Armin Schindler
• [asterisk-users] ISDN config EWSD   Roger Schreiter
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Richard Scobie
• [asterisk-users] Asterisk with cisco 7935   Miles Scruggs
• [asterisk-users] Asterisk with cisco 7935   Miles Scruggs
• [asterisk-users] "Asterisk Developers Mailing List" <asterisk-dev@lists.digium.com>,   Vidura Senadeera
• [asterisk-users] Zaptel-1.2.8 compile problem   Vidura Senadeera
• [asterisk-users] Sipura 3000 and Asterisk   Francisco Seratti
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Dan Serban
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Dan Serban
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Dan Serban
• [asterisk-users] AEL if/else/IFTIME fun.   Dan Serban
• [asterisk-users] ISDN HFC card cannot 'detect remote answer'   Edoardo Serra
• [asterisk-users] Volume events causing talk off on Asterisk with Digium 411P   Servetas, Andrew
• [asterisk-users] RE: Volume events causing talk off on Asterisk with Digium 411P   Servetas, Andrew
• [asterisk-users] Re: Volume events causing talk off on Asterisk with Digium 411P   Servetas, Andrew
• [asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 39   Servetas, Andrew
• [asterisk-users] RE: Volume events causing talk off on Asterisk with Digium 411P   Servetas, Andrew
• [asterisk-users] config include issues   Curt Shaffer
• [asterisk-users] config include issues   Curt Shaffer
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Tim Sharp
• [asterisk-users] Zaptel-1.2.9 compile error   Stuart Sheldon
• [asterisk-users] Problems with DISA   Shidan
• [asterisk-users] Problems with DISA   Shidan
• [asterisk-users] Response to KP Flemming...   Joe Shmoe
• [asterisk-users] Response to KP Flemming...   Joe Shmoe
• [asterisk-users] Response to KP Flemming...   Joe Shmoe
• [asterisk-users] How could i get bridged channel partner   Mohammad Shokuie
• [asterisk-users] Quintum tenor configuration with asterisk help   Mohammad Shokuie
• [asterisk-users] SER+Asterisk integration   Siqhamo Sifo
• [asterisk-users] Asterisk+ser+docs   Siqhamo Sifo
• [asterisk-users] IAX2 trunking   Siqhamo Sifo
• [asterisk-users] Asterisk as a gateway to SER   Siqhamo Sifo
• [asterisk-users] IAX2 audio problem   Siqhamo Sifo
• [asterisk-users] User authentication   Siqhamo Sifo
• [asterisk-users] Asterisk+Astbill   Siqhamo Sifo
• [asterisk-users] Stupid question about FXS/FXO   Moises Silva
• [asterisk-users] Scope of contexts   Moises Silva
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   Moises Silva
• [asterisk-users] Re: chan_zap.so stopped working after upgrading CentOS   Moises Silva
• [asterisk-users] How to send DTMF down a channel   Moises Silva
• [asterisk-users] How to send DTMF down a channel   Moises Silva
• [asterisk-users] two safe_asterisk processes on the same PBX???   Moises Silva
• [asterisk-users] Asterisk variables   Moises Silva
• [asterisk-users] How to send DTMF down a channel   Moises Silva
• [asterisk-users] How to send DTMF down a channel   Moises Silva
• [asterisk-users] DTMF Tone Not Passing Help   Moises Silva
• [asterisk-users] Chanspy crashing the server, again   Moises Silva
• [asterisk-users] Asterisk server crashes after two years   Nir Simionovich
• [asterisk-users] Asterisk not sending RTP   Nir Simionovich
• [asterisk-users] Asterisk not sending RTP   Nir Simionovich
• [asterisk-users] Asterisk not sending RTP   Nir Simionovich
• [asterisk-users] Asterisk not sending RTP   Nir Simionovich
• [asterisk-users] Asterisk not sending RTP   Nir Simionovich
• [asterisk-users] SER with multiple asterisk deployment   Adi Simon
• [asterisk-users] SER with multiple asterisk deployment   Adi Simon
• [asterisk-users] SER with multiple asterisk deployment   Adi Simon
• [asterisk-users] AgentCallBackLogin and cdrupdate   Garth van Sittert
• [asterisk-users] Budgetones - multiple phones losing IP address during day   Garth van Sittert
• [asterisk-users] Asterisk + Samsung OffServ 500   Garth van Sittert
• [asterisk-users] How to test TE405P T1   Garth van Sittert
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Kevin Smith
• [asterisk-users] Blind transfer 3/4 digits   Kevin Smith
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Kevin Smith
• [asterisk-users] MSSQL connection   Kevin Smith
• [asterisk-users] MSSQL connection   Kevin Smith
• [asterisk-users] Another (quick) Polycom 501 question   Kevin Smith
• [asterisk-users] Polycom related question   Kevin Smith
• [asterisk-users] Polycom related question   Kevin Smith
• [asterisk-users] Polycom related question   Kevin Smith
• [asterisk-users] how to transfer a caller out of a queue ?   Rick Smith
• [asterisk-users] Running Multiple Instances of Asterisk   Rick Smith
• [asterisk-users] Queue Status via Dialplan   Rick Smith
• [asterisk-users] AstriCon 2006 Reminder / Hotel Selling Out   Steven Sokol
• [asterisk-users] asterisk t.38 fax failed   Kokfoo Soo
• [asterisk-users] asterisk t.38 fax failed   Kokfoo Soo
• [asterisk-users] asterisk t.38 fax failed   Kokfoo Soo
• [asterisk-users] netmask   Kokfoo Soo
• [asterisk-users] Capacity for transcode G711 to G729   Kokfoo Soo
• [asterisk-users] Speex Codex - Eyebean to Asterisk   Kokfoo Soo
• [asterisk-users] Transcode Speex to G711-ulaw   Kokfoo Soo
• [asterisk-users] [Slightly OT] Grandstream configurator tool   Andrea Spadaccini
• [asterisk-users] [Slightly OT] Grandstream configurator tool   Andrea Spadaccini
• [asterisk-users] Asterisk as a SER client   Andrea Spadaccini
• [asterisk-users] [Slightly OT] Grandstream configurator tool   Andrea Spadaccini
• [asterisk-users] Problems compil 1.2.11   Andrea Spadaccini
• [asterisk-users] Asterisk and UMTS phones   Andrea Spadaccini
• [asterisk-users] Re: asterisk to cell phone network   Andrea Spadaccini
• [asterisk-users] X100P Clone card in JAPAN   Miroslav Spasovski
• [asterisk-users] freepbx dial plan, add and remove at the same time   Mat Stace
• [asterisk-users] freepbx dial plan, add and remove at the same time   Mat Stace
• [asterisk-users] Help with Tieing Outbound calls to Zap Channels   Kevin Steil
• [asterisk-users] Voip Buster - CID   Chris Stenton
• [asterisk-users] Re: 911 versus 9.911   Steven
• [asterisk-users] Re: Re: 911 versus 9.911   Steven
• [asterisk-users] Re: Re: 911 versus 9.911   Steven
• [asterisk-users] Re: Re: 911 versus 9.911   Steven
• [asterisk-users] Re: Re: Re: 911 versus 9.911   Steven
• [asterisk-users] Re: Re: Re: 911 versus 9.911   Steven
• [asterisk-users] Re: Volume events causing talk off on Asterisk withDigium 411P   Steven
• [asterisk-users] Re: FW: Peter Dicks Chairman of Sportingbet PLCisarrested at JFK!!   Steven
• [asterisk-users] Re: FW: Peter Dicks Chairman ofSportingbet PLC isarrested at JFK!!   Steven
• [asterisk-users] Re: Re: FW: Peter Dicks Chairman ofSportingbet PLCisarrested at JFK!!   Steven
• [asterisk-users] Re: All circuits are busy now???   Steven
• [asterisk-users] Re: Detect PBX vs Network message   Steven
• [asterisk-users] Re: SIP "Lines" Example Citel   Steven
• [asterisk-users] Re: asterisk and PowerEdge 1950   Steven
• [asterisk-users] Re: Setting QOS settings in asterisk and/or CentOS?   Steven
• [asterisk-users] Re: "does /var/run/asterisk.ctl exist?" -- butAsterisk *is* running.   Steven
• [asterisk-users] Set hint status from dialplan?   Steven
• [asterisk-users] Re: Set hint status from dialplan?   Steven
• [asterisk-users] Re: Set hint status from dialplan?   Steven
• [asterisk-users] Re: Re: Set hint status from dialplan?   Steven
• [asterisk-users] SMS Text Send working with BT Text in the UK??   Scott Stingel
• [asterisk-users] SMS Text Send working with BT Text in the UK??   Scott Stingel
• [asterisk-users] E1 crossover system   Scott Stingel
• [asterisk-users] SNOM 320 - 404 "Not Found"   Christian Stredicke
• [asterisk-users] Asterisk 1.2.11 and # key   Michael Strelnikov
• [asterisk-users] Asterisk 1.2.11 and # key   Michael Strelnikov
• [asterisk-users] Asterisk 1.2.11 and # key   Michael Strelnikov
• [asterisk-users] End of call   Michael Strelnikov
• [asterisk-users] End of call   Michael Strelnikov
• [asterisk-users] Asterisk 1.2.11 and # key   Michael Strelnikov
• [asterisk-users] Asterisk 1.2.11 and # key   Michael Strelnikov
• [asterisk-users] Problem with # locking up call   Michael Strelnikov
• [asterisk-users] Asterisk speaks Italian!   Stuart
• [asterisk-users] PRI Backup   Massimiliano Stucchi
• [asterisk-users] loosing Sipura 841 almost exactly on the hour   Jim Sturtevant
• [asterisk-users] Call center reports   Technical Support
• [asterisk-users] FAX handling   Technical Support
• [asterisk-users] 64 analog phones   Carlo Taguinod
• [asterisk-users] Finally a VoIP GSM Gateway single port..   Sam Tam
• RE: [asterisk-users] GSM Gateway Promotion from £69GBP   Sam Tam
• [asterisk-users] 64 analog phones   Sam Tam
• FW: [asterisk-users] Re: asterisk to cell phone network   Sam Tam
• [asterisk-users] automatic dialer with number pool and auto dialing   Sam Tam
• [asterisk-users] Replacing mpg123 with madplay under Solaris?   Frank Tarczynski
• [asterisk-users] Re: asterisk-users Digest, Vol 26, Issue 172   Frank Tarczynski
• [Asterisk-Users] UK Male English Voices   Will Tatam
• [Asterisk-Users] Re: [asterisk-biz] UK Male English Voices   Will Tatam
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Jonn R Taylor
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Jonn R Taylor
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Jonn R Taylor
• [asterisk-users] Asterisk, Asterisk-Addons, Zaptel and Libpri 1.4 betas released!   Asterisk Development Team
• [asterisk-users] Asterisk 1.2.12 and Zaptel 1.2.9 released!   The Asterisk Development Team
• [asterisk-users] Asterisk 1.2.12.1 and Zaptel 1.2.9.1 Released   The Asterisk Development Team
• [asterisk-users] zaptel 1.2.9 won't compile   Earl Terwilliger
• [asterisk-users] Asterisk 1.4-beta2 Spanish Sounds missing vm-youhaveno?   Earl Terwilliger
• [asterisk-users] Chanspy crashing the server, again   James Texter
• [asterisk-users] Running Multiple Instances of Asterisk   James Texter
• [asterisk-users] Running Multiple Instances of Asterisk   James Texter
• [asterisk-users] Asterisk 1.2 and SATA drives   Tharanga
• [asterisk-users] trying to understand siprealtime & nat/MWI issues   Arne Van Theemsche
• [asterisk-users] IAX2 & SIP Monitoring Solution for Asterisk   David Thomas
• [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry   Matthew Thompson
• [asterisk-users] 64 analog phones   Matthew Thompson
• [asterisk-users] Dial-9 (was Extension Numbering)   Matthew Thompson
• [asterisk-users] Where are the kernel sources?   Matthew Thompson
• [asterisk-users] Asterisk "Clusters"   Mitch Thompson
• [asterisk-users] Re: How to manipulate a plus in a phone number   Stefan Tichy
• [asterisk-users] X100P and zaptel 1.2.8   Tim
• [asterisk-users] X100P compile problems   Tim
• [asterisk-users] Is this phone any good?   Tim
• [asterisk-users] Is this phone any good?   Tim
• [asterisk-users] Is this phone any good   Tim
• [asterisk-users] Re: help connecting cell phone, chan_bluetooth   Todd
• [asterisk-users] Re: [asterisk-dev] UUI in calls   John Todd
• [asterisk-users] Changing the recording dir of MeetMe recordings.   Jan du Toit
• [asterisk-users] 7940 vs. 7941   Tom
• [asterisk-users] Asterisk+ser+docs   Victor Toofic
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Victor Toofic
• [asterisk-users] Faxing ..   Steve Totaro
• [asterisk-users] Caller ID has extra digits to strip   Steve Totaro
• [asterisk-users] Merlin Legend - Working Now!   Steve Totaro
• [asterisk-users] Merlin Legend - Working Now!   Steve Totaro
• [asterisk-users] app_rxfax Only Receives One Page   Steve Totaro
• [asterisk-users] app_rxfax Only Receives One Page   Steve Totaro
• [asterisk-users] app_rxfax Only Receives One Page   Steve Totaro
• [asterisk-users] Quintum tenor configuration with asterisk help   Steve Totaro
• [asterisk-users] QUINTUM CONFIGURATION.-   Steve Totaro
• [asterisk-users] Max Size of Conf Files   Steve Totaro
• [asterisk-users] QUINTUM TENOR ASM200 Configuration   Steve Totaro
• [asterisk-users] How to integrate freepbx with a2billing?   Steve Totaro
• [asterisk-users] Max Size of Conf Files   Steve Totaro
• [asterisk-users] QUINTUM TENOR ASM200 Configuration   Steve Totaro
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya Definity Experts out there?)   Steve Totaro
• [asterisk-users] "Too many files..." error - best way to fix?   Steve Totaro
• [asterisk-users] IVR not able to Play the Balance.. need some helphere   Steve Totaro
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   Steve Totaro
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   Steve Totaro
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   Steve Totaro
• [asterisk-users] IVR not able to Play the Balance.. need some helphere   Steve Totaro
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   Steve Totaro
• [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?   Steve Totaro
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   Steve Totaro
• [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?   Steve Totaro
• [asterisk-users] IVR not able to Play the Balance.. need some helphere   Steve Totaro
• [asterisk-users] Dropped Calls on TDM400p   Steve Totaro
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   Steve Totaro
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   Steve Totaro
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   Steve Totaro
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   Steve Totaro
• [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy   Steve Totaro
• [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy   Steve Totaro
• [asterisk-users] "Ping" a phone   Steve Totaro
• [asterisk-users] SHSU asterisk installation?   Steve Totaro
• [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy   Steve Totaro
• [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy   Steve Totaro
• [asterisk-users] Asterisk Server Down   Steve Totaro
• [asterisk-users] Issues with AGI+Dial command   Steve Totaro
• [asterisk-users] Digium GUI?   Steve Totaro
• [asterisk-users] SIP "Lines" Example Citel   Steve Totaro
• [asterisk-users] University dumps CISCO VoIP for Asterisk   Steve Totaro
• [asterisk-users] Re: SIP "Lines" Example Citel   Steve Totaro
• [asterisk-users] alternatives to mpg123: format_mp3, rawplayer or madplay?   Steve Totaro
• [asterisk-users] 64 analog phones   Steve Totaro
• [asterisk-users] alternatives to mpg123: format_mp3, rawplayer or madplay?   Steve Totaro
• [asterisk-users] OT: Anybody remember this from last Dec?   Steve Totaro
• [asterisk-users] Cisco 7960 Double Natted   Steve Totaro
• [asterisk-users] Trixbox Documentation   Steve Totaro
• [asterisk-users] 2 CPU's, Only 1 taking IRQ's   Steve Totaro
• [asterisk-users] PRI Outbound CallerID Question   Steve Totaro
• [asterisk-users] Building Zaptel 1.2.9 with Octasic   Steven Totaro
• [asterisk-users] Dropped Calls on TDM400p   Steven Totaro
• [asterisk-users] asterisk server to server using sip question   Steven Totaro
• [asterisk-users] Controlling the channel   Steven Totaro
• [asterisk-users] 9 becomes 99 ? And other strangeness   Steven Totaro
• [asterisk-users] University switches to Asterisk   Steven Totaro
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   Steven Totaro
• [asterisk-users] clustering asterisk is possible ?   Información Capa Tres
• [asterisk-users] How to Dial a number with Sangoma PRI card?   Tristan
• [asterisk-users] Analog Modem through ISDN E1 Primay Line   Tron
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Justin Tunney
• [asterisk-users] Starting Asterisk PBX: FATAL: Module ixj not found.   Justin Tunney
• [asterisk-users] SER with multiple asterisk deployment   Justin Tunney
• [asterisk-users] Really bad phone line.. possible causes?   Jeff Turner
• [asterisk-users] Re: Really bad phone line.. possible causes?   Jeff Turner
• [asterisk-users] fxotune failure! "Could not fill input buffer - got -1 bytes, expected 4000 bytes"   Jeff Turner
• [asterisk-users] g729 and polycoms problem   Alyed Tzompa
• [asterisk-users] g729 and polycoms problem   Alyed Tzompa
• [asterisk-users] Cisco 7970 behind NAT   Alyed Tzompa
• [asterisk-users] Configuring Codecs   Alyed Tzompa
• [asterisk-users] g729 and polycoms problem   Alyed Tzompa
• [asterisk-users] fw: Uniden - TVUNIDEN_UIP300   Alyed Tzompa
• [asterisk-users] H323 IP phones   Alyed Tzompa
• [asterisk-users] Are you using app_meetme or app_conference   Alyed Tzompa
• [asterisk-users] Spurious hangups on zaptel interface   Alyed Tzompa
• [asterisk-users] Spurious hangups on zaptel interface   Alyed Tzompa
• [asterisk-users] Native Chinese speaker needed   Steve Underwood
• [asterisk-users] app_rxfax Only Receives One Page   Steve Underwood
• [asterisk-users] Native Chinese speaker needed   Steve Underwood
• [asterisk-users] app_rxfax Only Receives One Page   Steve Underwood
• [asterisk-users] rxfax, spandsp and lack of ecm   Steve Underwood
• [asterisk-users] rxfax, spandsp and lack of ecm   Steve Underwood
• [asterisk-users] rxfax, spandsp and lack of ecm   Steve Underwood
• [asterisk-users] amr codec   Steve Underwood
• [Asterisk-Users] How to learn or teach VoIP QoE   Steve Underwood
• [asterisk-users] spandsp (foip)   Steve Underwood
• [asterisk-users] spandsp (foip)   Steve Underwood
• [asterisk-users] trixbox t38 pass through   Steve Underwood
• asterisk t.38 (was RE: [asterisk-users] trixbox t38 pass through)   Steve Underwood
• [asterisk-users] quadbri + tdm400p + modem-fax   Steve Underwood
• [asterisk-users] quadbri + tdm400p + modem-fax   Steve Underwood
• [asterisk-users] usereqphone=yes seems to don't work   Alexandre VERNIOL
• [asterisk-users] Realtime madness   Michel Vaillancourt
• [asterisk-users] Sip configuration using mysql   Michel Vaillancourt
• [asterisk-users] Sip configuration using mysql   Michel Vaillancourt
• [asterisk-users] ztcfg / X100P question   Michel Vaillancourt
• [asterisk-users] ztcfg / X100P question   Michel Vaillancourt
• [asterisk-users] ztcfg / X100P question   Michel Vaillancourt
• [asterisk-users] Good Book on Asterisk   Michel Vaillancourt
• [asterisk-users] quadbri + tdm400p + modem-fax   Michel Vaillancourt
• [asterisk-users] WAS: 64 analog phones NOW: Selection criteria and recipie for a good Asterisk install [long]   Michel Vaillancourt
• [asterisk-users] Any Suggestions for Election Polling Application?   Michel Vaillancourt
• [asterisk-users] callback without agi   Pato Valarezo
• [asterisk-users] WAIT FOR DIGIT not working   Pato Valarezo
• [asterisk-users] Line Pickup Problem   Pato Valarezo
• [asterisk-users] real time billing system   Pato Valarezo
• [asterisk-users] real time billing system   Pato Valarezo
• [asterisk-users] callback without agi   Patricio Valarezo
• [asterisk-users] Find-Me/Follow-ME   Joel Vandal
• [asterisk-users] sound file length   Race Vanderdecken
• [asterisk-users] sound file length   Race Vanderdecken
• [asterisk-users] Good Book on Asterisk   Race Vanderdecken
• [asterisk-users] Asterisk as a SER client   Arnd Vehling
• [asterisk-users] SER+Asterisk integration   Arnd Vehling
• [asterisk-users] Anyone working on VXML, CCXML support for asterisk?   Arnd Vehling
• [asterisk-users] Anyone working on VXML, CCXML support for asterisk?   Arnd Vehling
• [asterisk-users] Anyone using Voicemail with IMAP Support?   Arnd Vehling
• [asterisk-users] voicemail access thru apache on another server   Arnd Vehling
• [asterisk-users] Setting up imap based voicemail / invalid remote specification   Arnd Vehling
• [asterisk-users] real time billing system   Bernardo Vieira
• [asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right   Tom Vile
• [asterisk-users] trixbox t38 pass through   Tom Vile
• [asterisk-users] How do I reset a password?   Tom Vile
• [asterisk-users] Change Payload   Yelson Vivas
• [asterisk-users] SOS building fastagi C   Yelson Vivas
• [asterisk-users] outgoing call problem   Alexandru Voinescu
• [asterisk-users] stress a server with a tool   Charles Wang
• [asterisk-users] Wrong call handling   Erik Wartusch
• [asterisk-users] How to Install H323   Wasif
• [asterisk-users] How to Install H323   Wasif
• [asterisk-users] How to notify an ACD agent before he/she picks up   Watkins, Bradley
• [asterisk-users] Polycom Soundpoint Key Remap   Watkins, Bradley
• [asterisk-users] HINT problems with SVN-trunk-r43322   Watkins, Bradley
• [asterisk-users] HINT problems with SVN-trunk-r43322   Watkins, Bradley
• [asterisk-users] Asterisk -> Tekelec T6000 (Vocaldata, voiss)   Watkins, Bradley
• [asterisk-users] Unable to open pseudo channel for timing... Sound may be choppy.   Dominik Weber
• [asterisk-users] Receive Fax with rxfax on asterisk with debian   Dominik Weber
• [asterisk-users] Receive Fax with rxfax on asterisk with debian   Dominik Weber
• [asterisk-users] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Hermann Wecke
• [asterisk-users] Dropping extra frame of G.729 ?   Hermann Wecke
• [asterisk-users] Call center reports   Hermann Wecke
• [asterisk-users] voicemailmain menu   Jack Wei
• [asterisk-users] Correct settings for UK (BT) FXO   Ron Wellsted
• [asterisk-users] Deadlock   Michael Welter
• [asterisk-users] PRI channel hangup   Michael Welter
• [asterisk-users] PRI channel hangup   Michael Welter
• [asterisk-users] Re: PRI: sometimes Asterisk drop calls   Michael Welter
• [asterisk-users] Re: PRI: sometimes Asterisk drop calls   Michael Welter
• [asterisk-users] University switches to Asterisk   Michael Welter
• [asterisk-users] new in 1.4?   Michael Welter
• [asterisk-users] Deadlock   BJ Weschke
• [asterisk-users] Use PauseQueueMember   BJ Weschke
• [asterisk-users] Rack for Asterisk with TDM2400 Digium board   BJ Weschke
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   BJ Weschke
• [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya   BJ Weschke
• [asterisk-users] Polycom Firmware   BJ Weschke
• [asterisk-users] Channel kept busy when creating ssh tunnel via AGI   BJ Weschke
• [asterisk-users] stress a server with a tool   BJ Weschke
• [asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe   BJ Weschke
• [asterisk-users] CURL   BJ Weschke
• [asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe   BJ Weschke
• [asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe   BJ Weschke
• [asterisk-users] Queue failover and wrap time   BJ Weschke
• [asterisk-users] unable to call AT&T audio conference bridge   BJ Weschke
• [asterisk-users] Queue AddQueueMember()   BJ Weschke
• [asterisk-users] Re: Re: 911 versus 9.911   Derek Whitten
• [asterisk-users] Asterisk calling through FWD?   Derek Whitten
• [asterisk-users] why executed Hangup doesn't exit DialPlan?look my dialplan...   Eric "ManxPower" Wieling
• [asterisk-users] Response to KP Flemming...   Eric "ManxPower" Wieling
• [asterisk-users] What don't I get about SIP?   Eric "ManxPower" Wieling
• [asterisk-users] No dialtone, just directly busy   Eric "ManxPower" Wieling
• [asterisk-users] No dialtone, just directly busy   Eric "ManxPower" Wieling
• [asterisk-users] What don't I get about SIP?   Eric "ManxPower" Wieling
• [asterisk-users] I'm I wrong - No 3-way calling for Single line sets?   Eric "ManxPower" Wieling
• [asterisk-users] What don't I get about SIP?   Eric "ManxPower" Wieling
• [asterisk-users] What don't I get about SIP?   Eric "ManxPower" Wieling
• [asterisk-users] What don't I get about SIP?   Eric "ManxPower" Wieling
• [asterisk-users] Re: No such device -> TDM13B   Eric "ManxPower" Wieling
• [asterisk-users] What don't I get about SIP?   Eric "ManxPower" Wieling
• [asterisk-users] All circuits are busy now???   Eric "ManxPower" Wieling
• [asterisk-users] callback without agi   Eric "ManxPower" Wieling
• [asterisk-users] University switches to Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] University switches to Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Getting 'i' functionality on internal extensions   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk 1.4 Docs   Eric "ManxPower" Wieling
• [asterisk-users] University switches to Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Getting 'i' functionality on internal extensions   Eric "ManxPower" Wieling
• [asterisk-users] 9 becomes 99 ? And other strangeness   Eric "ManxPower" Wieling
• [asterisk-users] Tracking the source of a disconnect?   Eric "ManxPower" Wieling
• [asterisk-users] Tracking the source of a disconnect?   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk / Patton SmartNode SN2400 Strangeness FYI   Eric "ManxPower" Wieling
• [asterisk-users] problems with Polycom 500 boot up   Eric "ManxPower" Wieling
• [Asterisk-Users] Can you explain why multiple registration is an important (missing) feature ?   Eric "ManxPower" Wieling
• [asterisk-users] Cisco Distinctive ring using alert-info   Eric "ManxPower" Wieling
• [asterisk-users] transcoding error?   Eric "ManxPower" Wieling
• [asterisk-users] Problem with # locking up call   Eric "ManxPower" Wieling
• [asterisk-users] codecs/voicemail/DTMF   Eric "ManxPower" Wieling
• [asterisk-users] Problem with # locking up call   Eric "ManxPower" Wieling
• [asterisk-users] MOH distorted on Pound Key Linux on asterisk 1.2.8   Eric "ManxPower" Wieling
• [asterisk-users] University dumps CISCO VoIP for Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] MOH distorted on Pound Key Linux on asterisk1.2.8   Eric "ManxPower" Wieling
• Asterisk capabilities, was [asterisk-users] University dumps CISCO VoIP for Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Tracking the source of a disconnect? - SOLVED   Eric "ManxPower" Wieling
• [asterisk-users] Tracking the source of a disconnect? - SOLVED   Eric "ManxPower" Wieling
• [asterisk-users] problems with Polycom 500 boot up   Eric "ManxPower" Wieling
• [asterisk-users] codecs/voicemail/DTMF   Eric "ManxPower" Wieling
• [asterisk-users] problems with Polycom 500 boot up   Eric "ManxPower" Wieling
• [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?   Eric "ManxPower" Wieling
• [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?   Eric "ManxPower" Wieling
• [asterisk-users] VoicemailMain()   Eric "ManxPower" Wieling
• [asterisk-users] Looped message playback   Eric "ManxPower" Wieling
• [asterisk-users] TDM2400P   Eric "ManxPower" Wieling
• [asterisk-users] CURL   Eric "ManxPower" Wieling
• [asterisk-users] TDM2400 wired description and skiping frames   Eric "ManxPower" Wieling
• [asterisk-users] Re: Can you explain why multiple registrationisan important (missing) feature ?   Eric "ManxPower" Wieling
• [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?   Eric "ManxPower" Wieling
• [asterisk-users] Looped message playback   Eric "ManxPower" Wieling
• [asterisk-users] Running Multiple Instances of Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Running Multiple Instances of Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Running Multiple Instances of Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Extensions busy/congested and "circuit-busy"   Eric "ManxPower" Wieling
• [asterisk-users] TDM2400P vs Sangoma A200   Eric "ManxPower" Wieling
• [asterisk-users] How can I stop lost DNS from killing Asterisk?   Eric "ManxPower" Wieling
• [asterisk-users] How can I stop lost DNS from killing Asterisk?   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk behind Sonicwall firewall   Eric "ManxPower" Wieling
• [asterisk-users] Right way to prevent analog channel from answering the phone?   Eric "ManxPower" Wieling
• [asterisk-users] 64 analog phones   Eric "ManxPower" Wieling
• [asterisk-users] Right way to prevent analog channel from answering the phone?   Eric "ManxPower" Wieling
• [asterisk-users] Spurious hangups on zaptel interface   Eric "ManxPower" Wieling
• [asterisk-users] How to detect dial tone on ZAP channel before dialling using TDM2400P   Eric "ManxPower" Wieling
• [asterisk-users] Spurious hangups on zaptel interface   Eric "ManxPower" Wieling
• [asterisk-users] extensions.conf strangeness   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk -> Tekelec T6000 (Vocaldata, voiss)   Eric "ManxPower" Wieling
• [asterisk-users] Re: Voip Buster - CID   Eric "ManxPower" Wieling
• [asterisk-users] RPID   Eric "ManxPower" Wieling
• [asterisk-users] extensions.conf strangeness   Eric "ManxPower" Wieling
• [asterisk-users] t1-pri or sip trunk?   Eric "ManxPower" Wieling
• [asterisk-users] Problems with DISA   Eric "ManxPower" Wieling
• [asterisk-users] Re: SIP Gateway   Eric "ManxPower" Wieling
• [asterisk-users] Call Forward Problem   James Williams
• [asterisk-users] Native Chinese speaker needed   John Williams
• [asterisk-users] Native Chinese speaker needed   John Williams
• [asterisk-users] freepbx dial plan, add and remove at the same time   Mike Williams
• [asterisk-users] freepbx dial plan, add and remove at the same time   Mike Williams
• [asterisk-users] ISDN30 and digital phones   Mike Williams
• [asterisk-users] SPA3000 and asterisk   Walter Willis
• [asterisk-users] SPA3000 register in asterisk   Walter Willis
• [asterisk-users] Multiple asterisk same GUI   Stephen Wingfield
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Keys pressed not registering ...   Ronald Wiplinger
• [asterisk-users] Keys pressed not registering ...   Ronald Wiplinger
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Need somebody for video phone testing   Ronald Wiplinger
• [asterisk-users] How to check which rtp ports my firewall let through?   Ronald Wiplinger
• [asterisk-users] ast_parse_allow_disallow: Cannot allow unknown format 'h264'   Ronald Wiplinger
• [asterisk-users] svn trunk or branches ???   Ronald Wiplinger
• [asterisk-users] Makefile.moddir_rules: No such file or directory   Ronald Wiplinger
• [asterisk-users] ASTCC: change from no pin to pin request?   Ronald Wiplinger
• [asterisk-users] pickupgroup 1   Ronald Wiplinger
• [asterisk-users] Accounting and re-invite   Ronald Wiplinger
• [asterisk-users] Context default & incoming ENUM   Ronald Wiplinger
• [asterisk-users] Priority "n"   Ronald Wiplinger
• [asterisk-users] WARNING: chan_sip.c add_realm_authentication: ???   Ronald Wiplinger
• [asterisk-users] Pri Event 6 and 8   Dave Wise
• [asterisk-users] How to configure Fritz ISDN2 card with Trixbox?   Hans Witvliet
• [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?   Hans Witvliet
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Henrik Woffinden
• [asterisk-users] No dialtone, just directly busy   Henrik Woffinden
• [asterisk-users] No dialtone, just directly busy   Henrik Woffinden
• [asterisk-users] No dialtone, just directly busy   Henrik Woffinden
• [asterisk-users] mISDN versus ZapHFC with BRIstuff   Henrik Woffinden
• [asterisk-users] mISDN versus ZapHFC with BRIstuff   Henrik Woffinden
• [asterisk-users] Compile error in Asterisk 1.2.12.1   Henrik Woffinden
• [asterisk-users] mISDN versus ZapHFC with BRIstuff   Henrik Woffinden
• [asterisk-users] Dial and Timeout   Tobias Wolf
• [asterisk-users] Dial and Timeout   Tobias Wolf
• [asterisk-users] sound file length   Tobias Wolf
• [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN   Wolfgang_Borgon
• [asterisk-users] Cisco 7961 "dropouts"   Nat Wong
• [asterisk-users] manager api redirect dropping calls   Wing Wong
• [asterisk-users] Snom 360 Function Keys   Conrad Wood
• [asterisk-users] Grandstream Budgetone phones don't show   Conrad Wood
• [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s   Conrad Wood
• [asterisk-users] DSL router with integrated SIP proxy?   Conrad Wood
• [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s   Conrad Wood
• [asterisk-users] PRI Backup   Conrad Wood
• [asterisk-users] PRI Backup   Conrad Wood
• [asterisk-users] Forcing Transcode   Conrad Wood
• [asterisk-users] Forcing Transcode   Conrad Wood
• [asterisk-users] Building the Perfect Box   Conrad Wood
• [asterisk-users] macros in Realtime   Simon Woodhead
• [asterisk-users] Accounting and re-invite   Simon Woodhead
• [asterisk-users] DUNDi Servers   Simon Woodhead
• [asterisk-users] UK Colocation services   Simon Woodhead
• [asterisk-users] Find-Me/Follow-ME   Roger Workman
• [asterisk-users] Find-Me/Follow-ME   Roger Workman
• [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature   Tielin Xu
• [asterisk-users] ZT_SPANCONFIG failed on span 1: No such device or address (6)   Juan Miguel Yamakawa
• [asterisk-users] Linksys/Sipura 3K, Calls Timing Out   Iain Young
• [asterisk-users] 4-wire analogue interfaces?   Shane Young
• [asterisk-users] Satellite link-IAX Jitter Buffer.   Yusuf
• [asterisk-users] Has anyone tried to install both digital card   Yusuf
• [asterisk-users] new in 1.4?   Yusuf
• [asterisk-users] Cisco CAll Manger and H323   Yusuf
• [asterisk-users] Cisco CAll Manger and H323   Yusuf
• [asterisk-users] asterisk - alcatel   Sylvain ZUCCA
• [asterisk-users] How to send correct Caller ID on PRI   Zeeshan Zakaria
• [asterisk-users] Roundrobin not working on PRI   Zeeshan Zakaria
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Zeeshan Zakaria
• [asterisk-users] Roundrobin not working on PRI   Zeeshan Zakaria
• [asterisk-users] Grandstream GX-2000, doesn't send calls to free lines   Zeeshan Zakaria
• [asterisk-users] Grandstream, how to use the configuration tool   Zeeshan Zakaria
• [asterisk-users] Grandstream GX-2000, doesn't send calls to free lines   Zeeshan Zakaria
• [asterisk-users] How to send correct Caller ID on PRI   Zeeshan Zakaria
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Zeeshan Zakaria
• [asterisk-users] Grandstream, how to use the configuration tool   Zeeshan Zakaria
• [asterisk-users] Roundrobin not working on PRI   Zeeshan Zakaria
• [asterisk-users] Using option 'r' in queue doesn't announce frequeny etc.   Zeeshan Zakaria
• [asterisk-users] Whcih phones are better for mass deployment   Zeeshan Zakaria
• [asterisk-users] Whcih phones are better for mass deployment   Zeeshan Zakaria
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Zeeshan Zakaria
• [asterisk-users] Whcih phones are better for mass deployment   Zeeshan Zakaria
• [asterisk-users] Grandstream GX-2000 Remote Login Problem   Zeeshan Zakaria
• [asterisk-users] Streaming audio for MoH   Zeeshan Zakaria
• [asterisk-users] Grandstream GX-2000 Remote Login Problem   Zeeshan Zakaria
• [asterisk-users] Choppy MOH (Cisco gateway)   Zeeshan Zakaria
• [asterisk-users] Re: All circuits are busy now???   Zeeshan Zakaria
• [asterisk-users] Streaming MoH Problem, starts and then stops immediately   Zeeshan Zakaria
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   Zeeshan Zakaria
• [asterisk-users] chan_zap.so stopped working after upgrading CentOS   Zeeshan Zakaria
• [asterisk-users] How to install HUDLite Server   Zeeshan Zakaria
• [asterisk-users] Re: Streaming MoH Problem, starts and then stops immediately   Zeeshan Zakaria
• [asterisk-users] Re: chan_zap.so stopped working after upgrading CentOS   Zeeshan Zakaria
• [asterisk-users] How to install HUDLite Server   Zeeshan Zakaria
• [asterisk-users] How to install HUDLite Server   Zeeshan Zakaria
• [asterisk-users] FOP Installation help   Zeeshan Zakaria
• [asterisk-users] FOP Installation help   Zeeshan Zakaria
• [asterisk-users] How to install HUDLite Server   Zeeshan Zakaria
• [asterisk-users] How to install HUDLite Server   Zeeshan Zakaria
• [asterisk-users] How to install HUDLite Server   Zeeshan Zakaria
• [asterisk-users] Forwarding the Ring Group and Calls coming in to Queues   Zeeshan Zakaria
• [asterisk-users] Two phones, same number   Zeeshan Zakaria
• [asterisk-users] Trixbox Documentation   Zeeshan Zakaria
• [asterisk-users] Good Book on Asterisk   Norbert Zawodsky
• [asterisk-users] Good Book on Asterisk   Norbert Zawodsky
• [asterisk-users] Extension Numbering   Norbert Zawodsky
• [asterisk-users] Extension Numbering   Norbert Zawodsky
• [asterisk-users] Asterisk and SIP Redirect message   Michel Zenone
• [asterisk-users] VoicemailMain()   Michel Zenone
• [asterisk-users] Re:sip giving problems, please help.   Ma Zhiyong
• [asterisk-users] the sounds quality of IAX2 channels are not good as SIP channels?   Ma Zhiyong
• [asterisk-users] Re: the sounds quality of IAX2 channels are notgood as SIP channels?   Ma Zhiyong
• [asterisk-users] iax2 warning!   Ma Zhiyong
• [asterisk-users] IAX2 trunk voice quality: how many calls cause jitter?   Ma Zhiyong
• [asterisk-users] IAX2 trunking   Ma Zhiyong
• [asterisk-users] How to setup multiple iax2 trunks between two asterisk server?   Ma Zhiyong
• *****SPAM***** [asterisk-users] iax2 trunk call limits   Ma Zhiyong
• [asterisk-users] IAX2 register refuse but Dial cmd works!   Ma Zhiyong
• [asterisk-users] RTCP and RTP packetization in 1.4   Ma Zhiyong
• [asterisk-users] RTCP and RTP packetization in 1.4   Ma Zhiyong
• [asterisk-users] Iax Netstat Output   Ma Zhiyong
• [asterisk-users] asterisk, iaxmodem, hylafax quality problem   Ma Zhiyong
• [asterisk-users] real time billing system   Ma Zhiyong
• [asterisk-users] beginners question....   Panagiotis Zikos
• [asterisk-users] (no subject)   Panagiotis Zikos
• [asterisk-users] sample configuration   Panagiotis Zikos
• [asterisk-users] Clustering architecture and echo cancellation issue   Panagiotis Zikos
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   Zoa
• [asterisk-users] Volume events causing talk off on Asterisk with Digium 411P   Zoa
• [asterisk-users] Re: Volume events causing talk off on Asterisk with Digium 411P   Zoa
• [asterisk-users] new in 1.4?   Zoa
• [asterisk-users] 64 analog phones   Zoa
• [asterisk-users] spandsp (foip)   Zoa
• [asterisk-users] Re: Very high ping times from 7960 phones   Zoa
• [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's   Zoa
• [Asterisk-Users] External calls from Asteris over a legacy Siemens BusinessPhone 250 PBX   Wolfgang Zweimueller
• [asterisk-users] Format_MP3, Streaming, File Formats, MOH   Asterisk [Submusic]
• [asterisk-users] Re: Re: 911 versus 9.911   ahester at galacticltd.com
• [asterisk-users] IAX and rsa   andrutto
• [asterisk-users] ip address incoming call   antonio
• [asterisk-users] incoming call h323 cdr   antonio
• [asterisk-users] Sip answer one side , ring other side   antonio
• [asterisk-users] Sip answer one side , ring other side   antonio
• [asterisk-users] Sip answer one side , ring other side   antonio
• [asterisk-users] Conference bridge problem   Bartosz Wegrzyn - asterisk
• [asterisk-users] G729 and Tribox   Bartosz Wegrzyn - asterisk
• [asterisk-users] G729 and Tribox   Bartosz Wegrzyn - asterisk
• [asterisk-users] DTMF Detection Problems with certain phones incoming zap channels   asterisk asterisk
• [asterisk-users] DTMF Detection Problems with certain phones incoming zap channels   asterisk asterisk
• [asterisk-users] unable to call AT&T audio conference bridge   asterisk-user
• [asterisk-users] [Fwd: asterisk-users Digest, Vol 26, Issue 166]   asterisk-user
• [asterisk-users] [Fwd: [Fwd: [Fwd: asterisk-users Digest, Vol 26, Issue 166]]]   asterisk-user
• [asterisk-users] (no subject)   asterisk at teladdict.com
• [asterisk-users] A-Z termination   asterisk at teladdict.com
• [asterisk-users] voicemail greeting - How to access vociemail   asterisk_help at iwishi.nu
• [asterisk-users] IVR not able to Play the Balance.. need some helphere   bails
• [asterisk-users] Aastra 9133i and Atcom AT-320 - Comments please   bails
• [asterisk-users] Uninstalling Trixbox   bails
• [asterisk-users] how to setup poxy sip server   brian at neotiq.com
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] Call Processing Slow 11 seconds   broadbandvoice at comcast.net
• [asterisk-users] DID Provider in Thailand   broadbandvoice at comcast.net
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] Call Processing Slow 11 seconds   broadbandvoice at comcast.net
• [asterisk-users] Call Processing Slow 11 seconds   broadbandvoice at comcast.net
• [asterisk-users] Whcih phones are better for mass deployment   broadbandvoice at comcast.net
• [asterisk-users] Whcih phones are better for mass deployment   broadbandvoice at comcast.net
• [asterisk-users] su - postgres -bash-3.00$   broadbandvoice at comcast.net
• [asterisk-users] can someone recommend a voip provider that...   broadbandvoice at comcast.net
• [asterisk-users] using residential voip for business?   broadbandvoice at comcast.net
• [asterisk-users] su - postgres -bash-3.00$   broadbandvoice at comcast.net
• [asterisk-users] su - postgres -bash-3.00$   broadbandvoice at comcast.net
• [asterisk-users] Problems Unpacking tarball For Asterisk Application   broadbandvoice at comcast.net
• [asterisk-users] Problems Unpacking tarball For Asterisk Application   broadbandvoice at comcast.net
• [asterisk-users] Problems Unpacking tarball For Asterisk Application   broadbandvoice at comcast.net
• [asterisk-users] Verify Database Installation   broadbandvoice at comcast.net
• [asterisk-users] Verify Database Installation   broadbandvoice at comcast.net
• [asterisk-users] sip origination and termination   broadbandvoice at comcast.net
• [asterisk-users] University switches to Asterisk   broadbandvoice at comcast.net
• [asterisk-users] Asterisk Server Down   broadbandvoice at comcast.net
• [asterisk-users] Asterisk Server Down   broadbandvoice at comcast.net
• [asterisk-users] Termination Rates   broadbandvoice at comcast.net
• [asterisk-users] Starting Asterisk PBX: FATAL: Module ixj not found.   broadbandvoice at comcast.net
• [asterisk-users] create_addr: No such host:   broadbandvoice at comcast.net
• [asterisk-users] Starting Asterisk PBX: FATAL: Module ixj not found.   broadbandvoice at comcast.net
• [asterisk-users] Termination Rates   broadbandvoice at comcast.net
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   burke at tailorhosting.com
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   burke at tailorhosting.com
• [asterisk-users] Quintum tenor configuration with asterisk help   c.savinovich at itntelecom.com
• [asterisk-users] OT But So Ungodly Important   calvis
• [asterisk-users] looking for GXV-3000 users   marek cervenka
• asterisk t.38 (was RE: [asterisk-users] trixbox t38 pass through)   marek cervenka
• [asterisk-users] Support for Intel Boards On Asterisk   cleviton.araujo at caixa.gov.br
• [asterisk-users] Keys pressed not registering ...   John covici
• [asterisk-users] Keys pressed not registering ...   John covici
• [asterisk-users] Remote tone access   John covici
• [asterisk-users] Problem withfeature introducer as first digit on a call   John covici
• [asterisk-users] saved.gsm -> Voicemail greeting ??   John covici
• [asterisk-users] 407 Proxy Authentication Required   John covici
• [asterisk-users] WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long]   John covici
• [asterisk-users] blf aastra 9133i working but can't pickup calls   Jean-Louis curty
• [asterisk-users] blf aastra 9133i working but can't pickup calls   Jean-Louis curty
• [asterisk-users] blf aastra 9133i working but can't pickup calls   Jean-Louis curty
• [asterisk-users] Install issues on Freebsd 6.1 with Asterisk 1.4.0-beta2   boon dash
• [asterisk-users] 911 versus 9.911   end1r
• [asterisk-users] asterisk - alcatel   et pourquoi pas ? epp
• [asterisk-users] asterisk - alcatel   et pourquoi pas ? epp
• [asterisk-users] University switches to Asterisk   f6hqz-m at hamwlan.net
• [asterisk-users] Astmanproxy authentication problems   fho at inmedias.it
• [asterisk-users] Unexpected delay   flavio
• [asterisk-users] Unexpected delay: problem with outgoing calls   flavio
• [asterisk-users] ougoing calls problem   flavio
• [asterisk-users] pay as you go t.t38 fax termination and origination   stan ford
• [asterisk-users] t1-pri or sip trunk?   stan ford
• [asterisk-users] pstn failback   stan ford
• [asterisk-users] pstn failback   stan ford
• [asterisk-users] pstn failback   stan ford
• [asterisk-users] pay as you go t.t38 fax termination and origination   stan ford
• [asterisk-users] Use PauseQueueMember   gc
• [asterisk-users] Use PauseQueueMember   gc
• [asterisk-users] How to setup announce attibute in queues.conf   gc
• [asterisk-users] SNOM 320 - 404 "Not Found"   oi geli
• [asterisk-users] Asterisk vicidial question   ggonzalez at telviso.com.ar
• [asterisk-users] Jitter Buffer on SIP   ggonzalez at telviso.com.ar
• [asterisk-users] Hangup on Panasonic KX-TEM824   ggonzalez at telviso.com.ar
• [asterisk-users] Fedora   bilal ghayyad
• [asterisk-users] Registration doubt   god
• [asterisk-users] Register doubt   god
• [asterisk-users] How to register from asterisk server to an xlite.   god
• [asterisk-users] Blind transfer 3/4 digits   wendell hamilton
• [asterisk-users] asterisk presence (from manager API)   harrygaillac-sip at yahoo.fr
• [asterisk-users] New polycom firmware / presence   harrygaillac-sip at yahoo.fr
• [asterisk-users] Polycom new firmware and bootrom   harrygaillac-sip at yahoo.fr
• [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990   harrygaillac-sip at yahoo.fr
• [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990 [SOLVED]   harrygaillac-sip at yahoo.fr
• [asterisk-users] New polycom firmware / presence   harrygaillac-sip at yahoo.fr
• [asterisk-users] Digits are played in english in french voicemail   harrygaillac-sip at yahoo.fr
• [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990   harrygaillac-sip at yahoo.fr
• [asterisk-users] How to Install H323   harrygaillac-sip at yahoo.fr
• [asterisk-users] [Serusers] MS LCS 2005 / SER / Asterisk Integration   harrygaillac-sip at yahoo.fr
• [asterisk-users] SIP DOMAIN SUPPORT   harrygaillac-sip at yahoo.fr
• [asterisk-users] RE : Re: [asterisk-dev] Forwarding sip requests from none local domains   harrygaillac-sip at yahoo.fr
• [asterisk-users] RE : Re: [asterisk-dev] Forwarding sip requests from none localdomains   harrygaillac-sip at yahoo.fr
• [asterisk-users] test   harrygaillac-sip at yahoo.fr
• [asterisk-users] 491 request pending   harrygaillac-sip at yahoo.fr
• [asterisk-users] [asterisk-dev] 491 request pending   harrygaillac-sip at yahoo.fr
• [asterisk-users] [asterisk-dev] 491 request pending   harrygaillac-sip at yahoo.fr
• [asterisk-users] 491 request pending [2]   harrygaillac-sip at yahoo.fr
• [asterisk-users] open letter   harrygaillac-sip at yahoo.fr
• [asterisk-users] [asterisk-dev] open letter   harrygaillac-sip at yahoo.fr
• [asterisk-users] RE : Re: [asterisk-dev] open letter   harrygaillac-sip at yahoo.fr
• [asterisk-users] Polycom SoundPoint 2.0.1 SIP firmware?   harrygaillac-sip at yahoo.fr
• [asterisk-users] 404 not Found   harrygaillac-sip at yahoo.fr
• [asterisk-users] Problem with # locking up call   marvin horst
• [asterisk-users] Problem with # locking up call   marvin horst
• [asterisk-users] dialplan for confrencing   imthiyaz at peopletech.co.in
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] How to send and receiving fax with asterisk?   Andrea infoteam
• [asterisk-users] spandsp fax using Asterisk 1.2.X   isamar at isamarmaia.org
• [asterisk-users] Re: IAX2 & SIP Monitoring Solution for Asterisk   jaw+asterisk at tcp4me.com
• [asterisk-users] SIP on Asterisk, new install   joe, at j4computers
• [asterisk-users] SIP on Asterisk, new install   joe, at j4computers
• [asterisk-users] SIP phones not talking   joe, at j4computers
• [asterisk-users] Incoming calls, identify   joea, j4computers
• [asterisk-users] University dumps CISCO VoIP for Asterisk   joea, j4computers
• Asterisk capabilities, was [asterisk-users] University dumps CISCO VoIP for Asterisk   joea, j4computers
• [asterisk-users] Incoming calls, identify   joea, j4computers
• [asterisk-users] make error   joea, j4computers
• [asterisk-users] Trixbox Documentation   joea, j4computers
• [asterisk-users] make error   joea, j4computers
• [asterisk-users] libpri current extracts as beta1   joea, j4computers
• [asterisk-users] Trixbox Documentation   joea, j4computers
• [asterisk-users] libpri current extracts as beta1   joea, j4computers
• [Asterisk-Users] Asterisk Cygwin Port.   joshnet at nbnet.nb.ca
• [asterisk-users] How does Asterisk determine an incoming SIP Channel name?   kjcsb
• [asterisk-users] Register message received from realtime peer crashes Asterisk   kjcsb
• [asterisk-users] Segmentation fault on Asterisk startup: res_config_mysql.so problem?   kjcsb
• [asterisk-users] Segmentation fault on Asterisk startup:res_config_mysql.so problem?   kjcsb
• [asterisk-users] Segmentation fault on Asteriskstartup:res_config_mysql.so problem?   kjcsb
• [asterisk-users] agi   Gopal krishnan
• [asterisk-users] Digum g729 and g723   brandon kruz
• [asterisk-users] MOH distorted on Pound Key Linux on asterisk 1.2.8   brandon kruz
• [asterisk-users] Pri Event 6 and 8   brandon kruz
• [asterisk-users] Digium GUI?   brandon kruz
• [asterisk-users] AGI Errors   leitstelle at bnet.li
• [asterisk-users] AGI Errors   leitstelle at bnet.li
• [asterisk-users] AGI Errors   leitstelle at bnet.li
• [asterisk-users] Dell hardware ...   Warren (mailing lists)
• [asterisk-users] Voicemailmail hanging after entering password   Warren (mailing lists)
• [asterisk-users] Garbled (quality probs) IAX2 & SIP calls Asterisk-to-Asterisk   lists.digium.com at tgice.com
• [asterisk-users] my (SIP) INVITE is ignored   lokotes
• [asterisk-users] Grandstream SX2000 attended tranfer   magnus
• [asterisk-users] WG: Asterisk and Agents   mbodbg at gmx.net
• [asterisk-users] Log out an Agent on RNA   mbodbg at gmx.net
• [asterisk-users] Asterisk crashing when monitoring SIP device with Chanspy   mezzmor at aim.com
• [asterisk-users] Chanspy crashing the server, again   mezzmor at aim.com
• [asterisk-users] Chanspy crashing the server, again   mezzmor at aim.com
• [asterisk-users] Chanspy crashing server, again   mezzmor at aim.com
• [asterisk-users] UK Colocation services   mezzmor at aim.com
• [asterisk-users] quadbri & TDM400P on same pbx ?   mike
• [asterisk-users] 64 analog phones   mike
• [asterisk-users] Re: 64 analog phones   mike
• [asterisk-users] Digium GUI?   mitcheloc
• [asterisk-users] Digium GUI?   mitcheloc
• [asterisk-users] res_osp.c not compiled   neil
• [asterisk-users] What minimum required packages for 1.4   news.gmane.org
• [asterisk-users] stress a server with a tool   nik600
• [asterisk-users] g729 problem   o o
• [asterisk-users] g729 problem   o o
• [asterisk-users] Max Size of Conf Files   adebayo omo-dare
• [asterisk-users] When does Scalability requests Asterisk to U se SER ?   adebayo omo-dare
• [asterisk-users] PRI Backup   adebayo omo-dare
• [asterisk-users] PRI Backup   adebayo omo-dare
• [asterisk-users] PRI Backup   adebayo omo-dare
• [asterisk-users] Sangoma a301   adebayo omo-dare
• [asterisk-users] PAP2-NA + Asterisk   joy panlilio
• [asterisk-users] Warning about using PAP2-NA ATA recent firmware 3.1.12 LS   joy panlilio
• [asterisk-users] Application of Asterisk Packetization Patch   pannel at optusnet.com.au
• [asterisk-users] BLF via metermaid on 1.2.7.1 and aastra 9133i   mike pham
• [asterisk-users] pickup call little complicated   mike pham
• [asterisk-users] pickup call little complicated   mike pham
• [asterisk-users] FAX handling   phil.dawson at marnock.com
• [asterisk-users] PAP2-UK and Asterisk   phil.dawson at marnock.com
• [asterisk-users] Asterisk 1.2 & snom 360 MWI   phil.dawson at marnock.com
• ( SOLVED ) Re: [asterisk-users] Asterisk 1.2 & snom 360 MWI   phil.dawson at marnock.com
• [asterisk-users] Fax detection ...   phil.dawson at marnock.com
• [asterisk-users] IAX and rsa   picciuX
• [asterisk-users] Max Size of Conf Files   picciuX
• [asterisk-users] Call Forward Problem   picciuX
• [asterisk-users] Whcih phones are better for mass deployment   picciuX
• [asterisk-users] How to notify an ACD agent before he/she picks up   picciuX
• [asterisk-users] Two phones, same number   picciuX
• [asterisk-users] balance anouncement   ram
• [asterisk-users] balance anouncement   ram
• [asterisk-users] balance anouncement   ram
• [asterisk-users] balance anouncement   ram
• [asterisk-users] Asterisk+ser+docs   ram
• [asterisk-users] IVR not able to Play the Balance.. need some help here   ram
• [asterisk-users] IVR not able to Play the Balance.. need some helphere   ram
• [asterisk-users] IVR not able to Play the Balance.. need some helphere   ram
• [asterisk-users] IVR not able to Play the Balance.. need some   ram
• [asterisk-users] IVR not able to Play the Balance.. need some helphere   ram
• [asterisk-users] IVR not able to Play the Balance.. need some helphere   ram
• [asterisk-users] asterisk to cell phone network   yrving rivas
• [asterisk-users] data pass through   robb at boardman.me.uk
• [asterisk-users] voicemail ,MWI problem   tanzeel sarfaraz
• [asterisk-users] High utilization with SIP registration   sdallan at ozemail.com.au
• [asterisk-users] High CPU usage when Internet goes down   sdallan at ozemail.com.au
• [asterisk-users] Sipura SPA3000   sdcharly at gmail.com
• [asterisk-users] Calling Card and Billing   sdcharly at gmail.com
• [asterisk-users] Re: Calling Card and Billing   sdcharly at gmail.com
• [asterisk-users] urgently requires help regarding MWI   Tanzeel serfaraz
• [asterisk-users] MWI problem on asterisk   Tanzeel serfaraz
• [asterisk-users] Query on MWI   Tanzeel serfaraz
• [asterisk-users] Query ,NEED help regarding MWI   Tanzeel serfaraz
• [asterisk-users] blf aastra 9133i working but can't pickup calls   shadowym
• [asterisk-users] Asterisk 1.2 and SATA drives   shadowym
• [asterisk-users] Whcih phones are better for mass deployment   shadowym
• [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)   shadowym
• [asterisk-users] Building Zaptel 1.2.9 with Octasic   shadowym
• [asterisk-users] Asterisk 1.4 Docs   shadowym
• [Asterisk-Users] Shared Line Appearance, Snom and trunk   shadowym
• [asterisk-users] Digium GUI?   shadowym
• [asterisk-users] Asterisk Appliance, will Asterisk Business Edition be mandatory?   shadowym
• [asterisk-users] Digium GUI?   shadowym
• [asterisk-users] Digium GUI?   shadowym
• [asterisk-users] Digium GUI?   shadowym
• [asterisk-users] OT: Opinions on Aastra 480i CT?   shadowym
• [asterisk-users] Issues with Monitor in 1.4?   shadowym
• [asterisk-users] Digum g729 and g723   Joe shmoe
• [asterisk-users] Codec Thread   Joe shmoe
• [asterisk-users] Fedora   sip
• [asterisk-users] SER with multiple asterisk deployment   sip
• [asterisk-users] SER with multiple asterisk deployment   sip
• [asterisk-users] Re: Voip Buster - CID   sip
• [asterisk-users] Different MOH in parked calls??   equis software
• [asterisk-users] Different MOH in waiting calls and parked calls   equis software
• [asterisk-users] Different MOH between waiting calls and transfer calls   equis software
• [asterisk-users] Different MOH between waiting calls and transfer   equis software
• [asterisk-users] Is anybody using autofill option in queue.conf?   equis software
• [asterisk-users] Round Robin + Ringall   equis software
• [asterisk-users] Extensions problems   equis software
• [asterisk-users] Asterisk ramdonly crash using Realtime Static   equis software
• [asterisk-users] Can not hear the telco System Announcement   stoffell
• [asterisk-users] telco error message on PRI and BRI   stoffell
• [asterisk-users] Can not hear the telco System Announcement   stoffell
• [asterisk-users] Polycom Firmware   stoffell
• [asterisk-users] Polycom Firmware   stoffell
• [asterisk-users] Bri Card for Asterisk ?   stoffell
• [asterisk-users] non-technical, dealing with users giving feedback   stoffell
• [asterisk-users] non-technical, dealing with users giving feedback   stoffell
• [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s   stoffell
• [asterisk-users] Integrating the Openser for VoiceMail and PBX with Asterisk, For Account   raviprakash sunkara
• [asterisk-users] Help in Reloading of Asterisk...   raviprakash sunkara
• [asterisk-users] How can the User Know he has voicemail in the Databases.   raviprakash sunkara
• [asterisk-users] playback some digits to the caller from the callee (involves DTMF) prob   umar tarar
• [asterisk-users] flag 'g' in Dail() is'nt working with agentcallbacklogin()   umar tarar
• [asterisk-users] VoiceXML browser for Asterisk available !   tech at i6net.com
• [asterisk-users] using SIP to connect remote other VoIP server   tengulre
• [asterisk-users] asterisk skills in the philippines   tubongpeyups
• [asterisk-users] call across 2 asterisks   unplug
• [asterisk-users] call across 2 asterisks   unplug
• [asterisk-users] system cmd   unplug
• [asterisk-users] system cmd   unplug
• [asterisk-users] prompt playing problem   unplug
• [asterisk-users] voicemail greeting   unplug
• [asterisk-users] voicemail greeting   unplug
• [asterisk-users] voicemail greeting   unplug
• [asterisk-users] error 407 authenticate on INVITE   unplug
• [asterisk-users] queue information   unplug
• [asterisk-users] get the value from CDR   unplug
• [asterisk-users] 407 Proxy Authentication Required   unplug
• [asterisk-users] 407 Proxy Authentication Required   unplug
• [asterisk-users] Getting Music On Hold working in * 1.2.12.1 with Fedora?   voiplist
• [asterisk-users] Configuring Codecs   wyatt.wmvg at comcast.net
• [asterisk-users] Debugging and Outbound SIP Trunk   wyatt.wmvg at comcast.net
• [asterisk-users] Conference Call Delay & Quality   wyatt.wmvg at comcast.net
• [asterisk-users] voicemail greeting   wyatt.wmvg at comcast.net
• [asterisk-users] AgentCallbacklogin in Asterisk1.4 beta2   Li yuqian
• [asterisk-users] Zaptel-1.2.8 compile problem   yusuf
• [asterisk-users] Asterisk and NAT ?   yusuf
• [asterisk-users] 0005162: RTP Packetization : Few questions   yusuf
• [asterisk-users] 0005162: RTP Packetization : Few questions   yusuf
• [asterisk-users] 0005162: RTP Packetization : Few questions   yusuf
• [asterisk-users] SOLVED: ringback on box with E1 and premicell   yusuf
• [asterisk-users] How to Dial a number with Sangoma PRI card?   yusuf
• [asterisk-users] How to Dial a number with Sangoma PRI card?   yusuf
• [asterisk-users] RTCP and RTP packetization in 1.4   yusuf
• [asterisk-users] RTCP and RTP packetization in 1.4   yusuf
• [asterisk-users] RTCP and RTP packetization in 1.4   yusuf
• [asterisk-users] Iax Netstat Output   yusuf
• [asterisk-users] Cisco CAll Manger and H323   yusuf