September 2007 Archives by author

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Starting: Sat Sep 1 00:59:45 CDT 2007
Ending: Sun Sep 30 23:45:02 CDT 2007
Messages: 1803

• [asterisk-users] Wireless VOIP Keysets? Recommendations?   William Stillwell (Ki4swy)
• [asterisk-users] Grandstream GXW-4104 ???   William Stillwell (Ki4swy)
• [asterisk-users] Mutipoint Conferencing?   William Stillwell (Ki4swy)
• [asterisk-users] Recommend Digium Hardware?   William Stillwell (Ki4swy)
• [asterisk-users] Linux-HA and Asterisk   Chris Mason (Lists)
• [asterisk-users] Call Center SoftPhone with Auto Answer   Chris Mason (Lists)
• [asterisk-users] Call Center SoftPhone with Auto Answer   Chris Mason (Lists)
• [asterisk-users] VoIP Provider for business   Chris Mason (Lists)
• [asterisk-users] Interesting Conference Request - Can this be done ?   Chris Mason (Lists)
• [asterisk-users] MOH Files Volume   Darrick Hartman (lists)
• [asterisk-users] g729 on 1.4.10.1   Darrick Hartman (lists)
• [asterisk-users] What's the deal with ATAcomm?   Darrick Hartman (lists)
• [asterisk-users] Show Callee name on Display   Yehavi Bourvine +972-8-9489444
• [asterisk-users] online active call watching   Yehavi Bourvine +972-8-9489444
• [asterisk-users] Polycom 330 + Asterisk, phone locks up. * key will do it   telmnstr at 757.org
• [asterisk-users] Polycom 330 + Asterisk, phone locks up. * key will do it   telmnstr at 757.org
• [asterisk-users] Asterisk MESSAGE Method for SMS   Abdul
• [asterisk-users] Installing Asterisk on to CentOS 4   Abdul
• [asterisk-users] Cisco 7960 sccp   Joe Acquisto
• [asterisk-users] Cisco 7960 or 7960G   Joe Acquisto
• [asterisk-users] Cisco 7960 or 7960G   Joe Acquisto
• [asterisk-users] OT: DELL Platforms   Joe Acquisto
• [asterisk-users] TDM400P (TDM22P) and aux power.   Joe Acquisto
• [asterisk-users] TDM400P (TDM22P) and aux power.   Joe Acquisto
• [asterisk-users] TDM400P (TDM22P) and aux power.   Joe Acquisto
• [asterisk-users] FAX detection not working   Joe Acquisto
• [asterisk-users] FAX detection not working   Joe Acquisto
• [asterisk-users] FAX detection not working   Joe Acquisto
• [asterisk-users] OT: DELL Platforms   Nick Adams
• [asterisk-users] Problems sending more than 2 SMS with asterisk / smsq   Christoph Adomeit
• [asterisk-users] Manager Originate Action and Cancel   Santiago Aguiar
• [asterisk-users] ztdummy kills audio   John Albano
• [asterisk-users] ztdummy kills audio   John Albano
• [asterisk-users] ztdummy kills audio   John Albano
• [asterisk-users] ztdummy kills audio   John Albano
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Brian Alexander
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Brian Alexander
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Brian Alexander
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Brian Alexander
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Brian Alexander
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Brian Alexander
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Brian Alexander
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Brian Alexander
• [asterisk-users] How to handle "+" prefix   Benny Amorsen
• [asterisk-users] Multitenant or Multiple virtual machines   Benny Amorsen
• [asterisk-users] Multiple Home system with SIP   Benny Amorsen
• [asterisk-users] Multiple Home system with SIP   Benny Amorsen
• [asterisk-users] Ping   Bill Andersen
• [asterisk-users] Softkeys wrong with chan_skinny   Andreas Anderson
• [asterisk-users] Different Networks   Erik Anderson
• [asterisk-users] Different Networks   Erik Anderson
• [asterisk-users] ISDN PRI debug in Asterisk   Erik Anderson
• [asterisk-users] sip.conf best practices?   Erik Anderson
• [asterisk-users] sip.conf best practices?   Erik Anderson
• [asterisk-users] Anyone use the Linksys phones?   Erik Anderson
• [asterisk-users] What's the deal with ATAcomm?   Erik Anderson
• [asterisk-users] Recommend Digium Hardware?   Erik Anderson
• [asterisk-users] Cisco 7940G licensing with asterisk   Cory Andrews
• [asterisk-users] Asterisk Appliance with VoIPStreet   Cory Andrews
• [asterisk-users] call transfer detection in dial plan   Rilawich Ango
• [asterisk-users] prepaid application recommendation   Rilawich Ango
• [asterisk-users] asterisk crash   Rilawich Ango
• [asterisk-users] 3-way calling   Rilawich Ango
• [asterisk-users] 3-way calling   Rilawich Ango
• [asterisk-users] 3-way calling   Rilawich Ango
• [asterisk-users] call relation in call transfer   Rilawich Ango
• [asterisk-users] 3-way calling   Rilawich Ango
• [asterisk-users] call relation in call transfer   Rilawich Ango
• [asterisk-users] Networking Question   Brian M. Arlinghaus
• [asterisk-users] AGI script fails on IAX channels (from call file).   Jonas Arndt
• [asterisk-users] AGI script fails on IAX channels (from call file).   Jonas Arndt
• [asterisk-users] AGI script fails on IAX channels (from call file).   Jonas Arndt
• [asterisk-users] AGI script fails on IAX channels (from call file).   Jonas Arndt
• [asterisk-users] NAT-troubles with RTP   Florian Arthofer
• [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk   Jay R. Ashworth
• [asterisk-users] off-hook warning tone   Jay R. Ashworth
• [asterisk-users] Another State Of The Punctuation Mark question - Vonage   Jay R. Ashworth
• [asterisk-users] Another State Of The Punctuation Mark question - Vonage   Jay R. Ashworth
• [asterisk-users] New Project: AskoziaPBX   Jay R. Ashworth
• [asterisk-users] Another State Of The Punctuation Mark question - Vonage   Jay R. Ashworth
• [asterisk-users] Linux-HA and Asterisk   Jay R. Ashworth
• [asterisk-users] Generating an old-fashioned dialtone   Jay R. Ashworth
• [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?   Jay R. Ashworth
• [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?   Jay R. Ashworth
• [asterisk-users] Linux limits   Jay R. Ashworth
• [asterisk-users] (Getting OT) Re: Call Center SoftPhone with Auto Answer   Jay R. Ashworth
• [asterisk-users] Why does everyone seem to dislike *now?   Jay R. Ashworth
• [asterisk-users] OT: DELL Platforms   Lacy Moore - Aspendora
• [asterisk-users] OT: DELL Platforms   Lacy Moore - Aspendora
• [asterisk-users] Cisco 79xx XML Apps (was: Re: Cisco Directory Format)   Lacy Moore - Aspendora
• [asterisk-users] Show Callee name on Display   Lacy Moore - Aspendora
• [asterisk-users] Paging to external speaker like in airports etc...   Lacy Moore - Aspendora
• [asterisk-users] Why does everyone seem to dislike *now?   Lacy Moore - Aspendora
• [asterisk-users] Chan_SCCP vs. Chan_Skinny   Lacy Moore - Aspendora
• [asterisk-users] Dificult macro, please advise   Atis
• [asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension   Atis
• [asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension   Atis
• [asterisk-users] Dialplan regexp   Atis
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Atis
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Atis
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Atis
• [asterisk-users] Asterisk + Realtime + Manager reload = crash   Atis
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Atis
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Atis
• [asterisk-users] Asterisk Manager API - Originate command   Atis
• [asterisk-users] Prevent multiple sip registrations   Atis
• [asterisk-users] Prevent multiple sip registrations   Atis
• [asterisk-users] Prevent multiple sip registrations   Atis
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   Atis
• [asterisk-users] Callback for unanswered transfers...   Atis
• [asterisk-users] asterisk call back dail plan   Atis
• [asterisk-users] call transfer detection in dial plan   Atis
• [asterisk-users] No Sound on Zap Channels   Atis
• [asterisk-users] Asterisk cli   Atis
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Atis
• [asterisk-users] Prompt for extension with standard dial-tone.   Atis
• [asterisk-users] Prompt for extension with standard dial-tone.   Atis
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Atis
• [asterisk-users] Installing Asterisk on to CentOS 4   Ove Aursand
• [asterisk-users] Installing Asterisk on to CentOS 4   Ove Aursand
• [asterisk-users] Linux-HA and Asterisk   Ove Aursand
• [asterisk-users] Show Callee name on Display   Dan Austin
• [asterisk-users] Show Callee name on Display   Dan Austin
• [asterisk-users] Chan_SCCP vs. Chan_Skinny   Dan Austin
• [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO   Troy Ayers
• [asterisk-users] How to handle "+" prefix   Dovid B
• [asterisk-users] OT: DELL Platforms   Dovid B
• [asterisk-users] OT: DELL Platforms   Dovid B
• [asterisk-users] What is the difference between increasing theverbose level and the debug level?   Dovid B
• [asterisk-users] USA Termination   Dovid B
• [asterisk-users] Linux-HA and Asterisk   Dovid B
• [asterisk-users] SIP Debugging to separate log file   Dovid B
• [asterisk-users] Interesting Conference Request - Can this be done ?   Dovid B
• [asterisk-users] Linux-HA and Asterisk   Dovid B
• [asterisk-users] Problems with Asterisk behind a firewall   Dovid B
• [asterisk-users] OT: Samsung Sprint CDMAoIP   Dovid B
• [asterisk-users] New Installed X100p   G B
• [asterisk-users] New Installed X100p   G B
• [asterisk-users] New Installed X100p   G B
• [asterisk-users] New Installed X100p   G B
• [asterisk-users] New Installed X100p   G B
• [asterisk-users] New Installed X100p   G B
• [asterisk-users] Which Asterisk version to use?   Eric B.
• [asterisk-users] Which Asterisk version to use?   Eric B.
• [asterisk-users] Which Asterisk version to use?   Eric B.
• [asterisk-users] Cisco 7960 sccp   Michiel van Baak
• [asterisk-users] Cisco 7960 sccp   Michiel van Baak
• [asterisk-users] Softkeys wrong with chan_skinny   Michiel van Baak
• [asterisk-users] DTMF Relay Problems   Michiel van Baak
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Michiel van Baak
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Michiel van Baak
• [asterisk-users] DECT SIP phones   Michiel van Baak
• [asterisk-users] bug in 1.2.24   Michiel van Baak
• [asterisk-users] Anyone else having problems with the list   Michiel van Baak
• [asterisk-users] Dutch Number for Inbound   Michiel van Baak
• [asterisk-users] Asterisk Redundancy   Michiel van Baak
• [asterisk-users] Asterisk Redundancy   Michiel van Baak
• [asterisk-users] CID debugging help needed   Jeff Bachtel
• [asterisk-users] Another State Of The Punctuation Mark question - Vonage   Jeff Bachtel
• [asterisk-users] Linux-HA and Asterisk   Jeff Bachtel
• [asterisk-users] CID spill after second ring   Jeff Bachtel
• [asterisk-users] Cisco 7960 or 7960G   Chris Bagnall
• [asterisk-users] Multitenant or Multiple virtual machines   Chris Bagnall
• [asterisk-users] Connecting Asterisk to Alcatel OmniPCX   Chris Bagnall
• [asterisk-users] Anyone use the Linksys phones?   Chris Bagnall
• [asterisk-users] Anyone use the Linksys phones?   Chris Bagnall
• [asterisk-users] Point-to-Point SIP link without registration   Chris Bagnall
• [asterisk-users] Grandstream GXP2020 / 2000   Chris Bagnall
• [asterisk-users] Point-to-Point SIP link without registration   Chris Bagnall
• [asterisk-users] T1 to SIP conversion, standalone device   Joseph Bajin
• [asterisk-users] T1 to SIP conversion, standalone device   Joseph Bajin
• [asterisk-users] how to route outgoing calls on IP-level   Joseph Bajin
• [asterisk-users] Rechazo de llamada en triangulacion de asterisk.   Alex Balashov
• [asterisk-users] Rechazo de llamada en triangulacion de asterisk.   Alex Balashov
• [asterisk-users] Meridian S1 to Asterisk via T1   Alex Balashov
• [asterisk-users] Meridian S1 to Asterisk via T1   Alex Balashov
• [asterisk-users] T1 to SIP conversion, standalone device   Alex Balashov
• [asterisk-users] T1 to SIP conversion, standalone device   Alex Balashov
• [asterisk-users] Meridian S1 to Asterisk via T1   Alex Balashov
• [asterisk-users] Another State Of The Punctuation Mark question - Vonage   Alex Balashov
• [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?   Alex Balashov
• [asterisk-users] Conference bridge.   Alex Balashov
• [asterisk-users] Conference bridge.   Alex Balashov
• [asterisk-users] Linux limits   Alex Balashov
• [asterisk-users] Limiting Simultaneous calls   Alex Balashov
• [asterisk-users] (Getting OT) Re: Call Center SoftPhone with Auto Answer   Alex Balashov
• [asterisk-users] what is softswitch   Alex Balashov
• [asterisk-users] what is softswitch   Alex Balashov
• [asterisk-users] Multi-sip rings   Alex Balashov
• [asterisk-users] how to route outgoing calls on IP-level   Alex Balashov
• [asterisk-users] what is softswitch   Alex Balashov
• [asterisk-users] what is softswitch   Alex Balashov
• [asterisk-users] what is softswitch   Alex Balashov
• [asterisk-users] what is softswitch   Alex Balashov
• [asterisk-users] (Getting OT) Re: Call Center SoftPhone with Auto Answer   Alex Balashov
• [asterisk-users] what is softswitch   Alex Balashov
• [asterisk-users] what is softswitch   Alex Balashov
• [asterisk-users] Dial() Command Parameter L Overflow?   Alex Balashov
• [asterisk-users] crash after callbackagent ackcall   Alex Balashov
• [asterisk-users] what is softswitch   Alex Balashov
• [asterisk-users] Outgoing SIP packets out of order?   Alex Balashov
• [asterisk-users] Point-to-Point SIP link without registration   Alex Balashov
• [asterisk-users] Do I need to run #modprobe zaptel for Digium   Alex Balashov
• [asterisk-users] call relation in call transfer   Alex Balashov
• [asterisk-users] Cisco Directory Format   Time Bandit
• [asterisk-users] Callback for unanswered transfers...   Luis Antonio Prata Barbosa
• [asterisk-users] Callback for unanswered transfers...   Luis Antonio Prata Barbosa
• [asterisk-users] DISA and DTMF detection problem w/ FXO port on a TDM400   Luis Antonio Prata Barbosa
• [asterisk-users] nat=yes   Marco Bartholomew
• [asterisk-users] [EVENT] Asterisk and VoIP in enterprise   Raphaël Bauduin
• [asterisk-users] Limiting Simultaneous calls   Forrest Beck
• [asterisk-users] How to cancel the password check in VoicemailMain()   Forrest Beck
• [asterisk-users] Paging MEETME_RECORDINGFILE Variable   Forrest Beck
• [asterisk-users] Paging MEETME_RECORDINGFILE Variable   Forrest Beck
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Forrest Beck
• [asterisk-users] ExternNotify Voicemail   Forrest Beck
• [asterisk-users] ExternNotify Voicemail   Forrest Beck
• [asterisk-users] Zaptel-1.4.5.1 Compile Error   Forrest Beck
• [asterisk-users] Music On Hold   Forrest Beck
• [asterisk-users] DTMF Relay Problems   Joseph Begumisa
• [asterisk-users] DTMF Relay Problems   Joseph Begumisa
• [asterisk-users] DTMF Relay Problems   Joseph Begumisa
• [asterisk-users] DTMF Relay Problems   Joseph Begumisa
• [asterisk-users] DTMF Relay Problems   Joseph Begumisa
• [asterisk-users] dtmf issues on PRI and 1.4.11   Joseph Begumisa
• [asterisk-users] Asterisk 1.2 + Zaptel 1.4 + HPEC = Crash?   Nic Bellamy
• [asterisk-users] Asterisk and MS Exchange 2007   Jared Bellows
• [asterisk-users] Direct dialing to correct extension from analog lines   Lars Bensmann
• [asterisk-users] Direct dialing to correct extension from analog lines   Lars Bensmann
• [asterisk-users] GROUP() issues for me   Nicholas Blasgen
• [asterisk-users] GROUP() issues for me   Nicholas Blasgen
• [asterisk-users] Asterisk 1.2.24 simultaneous call limits.   Nicholas Blasgen
• [asterisk-users] Dialing an external number and then passing it to an extension...   Nicholas Blasgen
• [asterisk-users] call limit   Nicholas Blasgen
• [asterisk-users] Paging to external speaker like in airports etc...   Nicholas Blasgen
• [asterisk-users] error messages related to mysql in asterisk CLI   Nicholas Blasgen
• [asterisk-users] Polycom 501 Phones Rebooting   Gregory Boehnlein
• [asterisk-users] Polycom 501 Phones Rebooting   Gregory Boehnlein
• [asterisk-users] Polycom 501 Phones Rebooting   Gregory Boehnlein
• [asterisk-users] Polycom 501 Phones Rebooting   Gregory Boehnlein
• [asterisk-users] off-hook warning tone   Stephen Bosch
• [asterisk-users] Build your own "appliance" concept   Stephen Bosch
• [asterisk-users] New Installed X100p   Stephen Bosch
• [asterisk-users] New Installed X100p   Stephen Bosch
• [asterisk-users] Meridian S1 to Asterisk via T1   Stephen Bosch
• [asterisk-users] Flash IDE   Stephen Bosch
• [asterisk-users] Generating an old-fashioned dialtone   Stephen Bosch
• [asterisk-users] Generating an old-fashioned dialtone   Stephen Bosch
• [asterisk-users] DECT SIP phones   Stephen Bosch
• [asterisk-users] DECT SIP phones   Stephen Bosch
• [asterisk-users] DECT SIP phones   Stephen Bosch
• [asterisk-users] DECT SIP phones   Stephen Bosch
• [asterisk-users] Wondering why I can't post   Stephen Bosch
• [asterisk-users] Wondering why I can't post   Stephen Bosch
• [asterisk-users] Wondering why I can't post   Stephen Bosch
• [asterisk-users] Astricon 2007 -- does anybody need a ride?   Stephen Bosch
• [asterisk-users] Polycom 501 Phones Rebooting   Stephen Bosch
• [asterisk-users] Polycom 501 Phones Rebooting   Stephen Bosch
• [asterisk-users] Digium Appliance   David Boyd
• [asterisk-users] swift.conf - cepstral voice quality adjustment options   David Boyd
• [asterisk-users] Comfort noice sample (gsm/mp3)   Jim Boykin
• [asterisk-users] Limiting Simultaneous calls   Jim Boykin
• [asterisk-users] Comfort noice sample (gsm/mp3)   Jim Boykin
• [asterisk-users] VoIP Provider for business   Jim Boykin
• [asterisk-users] Limiting Simultaneous calls   Jim Boykin
• [asterisk-users] Manager Originate without phone off hook?   Russell Brown
• [asterisk-users] Manager Originate without phone off hook?   Russell Brown
• [asterisk-users] Manager Originate without phone off hook?   Russell Brown
• [asterisk-users] Ringing Groups, SIP Forward and looping problem   Russell Brown
• [asterisk-users] Include zaptel in kernel...   Russell Bryant
• [asterisk-users] Fwd: Bad FCS error   Russell Bryant
• [asterisk-users] Force a new user to configure Comedian mail?   Russell Bryant
• [asterisk-users] Force a new user to configure Comedian mail?   Russell Bryant
• [asterisk-users] Dell Power Edge 1900   Russell Bryant
• [asterisk-users] Interesting Conference Request - Can this be done ?   Russell Bryant
• [asterisk-users] Asterisk 1.4.12 Release?   Russell Bryant
• [asterisk-users] Proximity Detection: Motorola Q + Bluetooth + Asterisk   Chuck Bunn
• [asterisk-users] Proximity detection versus GSM receiver   Chuck Bunn
• [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk   Zane C.B.
• [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk   Zane C.B.
• [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk   Zane C.B.
• [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk   Zane C.B.
• [asterisk-users] how to route outgoing calls on IP-level   Rafael Canchola
• [asterisk-users] Why does everyone seem to dislike *now?   Jim Canfield
• [asterisk-users] Why does everyone seem to dislike *now?   Jim Canfield
• [asterisk-users] Why does everyone seem to dislike *now?   Jim Canfield
• [asterisk-users] Which Asterisk version to use?   Jim Canfield
• [asterisk-users] Nano syntax highlighting.   Jim Canfield
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   Brian Capouch
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   Brian Capouch
• [asterisk-users] What's the deal with ATAcomm?   Brian Capouch
• [asterisk-users] Authenticate() application and CDR   Ricardo Carvalho
• [asterisk-users] Authenticate() application and CDR   Ricardo Carvalho
• [asterisk-users] Grandstream GXP2020 / 2000   Ricardo Carvalho
• [asterisk-users] faster timeout in ENUMLOOKUP() function   Ricardo Carvalho
• [asterisk-users] Proper trunk to connect two systems.   Dan Casey
• [asterisk-users] Wireless VOIP Keysets? Recommendations?   Eric Chamberlain
• [asterisk-users] T1 to SIP conversion, standalone device   Eric Chamberlain
• [asterisk-users] Another State Of The Punctuation Mark question - Vonage   Eric Chamberlain
• [asterisk-users] OT: Samsung Sprint CDMAoIP   Eric Chamberlain
• [asterisk-users] OT: Samsung Sprint CDMAoIP   Eric Chamberlain
• [asterisk-users] OT: Samsung Sprint CDMAoIP   Eric Chamberlain
• [asterisk-users] Point-to-Point SIP link without registration   Eric Chamberlain
• [asterisk-users] Point-to-Point SIP link without registration   Eric Chamberlain
• [asterisk-users] Overhead paging over IP...   Carlos Chavez
• [asterisk-users] ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ...   Carlos Chavez
• [asterisk-users] Asterisk + Realtime + Manager reload = crash   Carlos Chavez
• [asterisk-users] DECT SIP phones   Carlos Chavez
• [asterisk-users] Dell Power Edge 1900   Carlos Chavez
• [asterisk-users] Dialing an external number and then passing it to an extension...   Carlos Chavez
• [asterisk-users] UNICALL MFC/R2 + Asterisk 1.4   Carlos Chavez
• [asterisk-users] Grandstream GXP2020 / 2000   Carlos Chavez
• [asterisk-users] Asterisk w/MS SQL Server 2005   Mik Cheez
• [asterisk-users] Asterisk realtime error   Mik Cheez
• [asterisk-users] Dead SIP channels   Gary Chen
• [asterisk-users] prepaid application recommendation   Sarfaraz Chougule
• [asterisk-users] Asterisk on Ubuntu Feisty   Christian
• [asterisk-users] Asterisk on Ubuntu Feisty   Christian
• [asterisk-users] Asterisk on Ubuntu Feisty   Christian
• [asterisk-users] Asterisk on Ubuntu Feisty   Christian
• [asterisk-users] Problems with Asterisk behind a firewall   Christian
• [asterisk-users] Problems with Asterisk behind a firewall   Christian
• [asterisk-users] Extensions Configuration   Max Clark
• [asterisk-users] Linux-HA and Asterisk   Mike Clark
• [asterisk-users] Linux-HA and Asterisk   Mike Clark
• [asterisk-users] Linux-HA and Asterisk   Mike Clark
• [asterisk-users] Linux-HA and Asterisk   Mike Clark
• [asterisk-users] IAX Java Softphone?   Mike Clark
• [asterisk-users] Zaptel modules are being installed in different directory   Tzafrir Cohen
• [asterisk-users] E1 to Ethernet Bridge   Tzafrir Cohen
• [asterisk-users] Zaptel modules are being installed in different directory   Tzafrir Cohen
• [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk   Tzafrir Cohen
• [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk   Tzafrir Cohen
• [asterisk-users] stop log/debug messages into /var/log/messages   Tzafrir Cohen
• [asterisk-users] stop log/debug messages into /var/log/messages   Tzafrir Cohen
• [asterisk-users] Udev issue on zaptel install   Tzafrir Cohen
• [asterisk-users] E1 Line Tapping   Tzafrir Cohen
• [asterisk-users] Asterisk Died message   Tzafrir Cohen
• [asterisk-users] ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ...   Tzafrir Cohen
• [asterisk-users] No Dial tone came from fxs modules   Tzafrir Cohen
• [asterisk-users] 14. Re: ztcfg error : TE110p error with " CAS signalling on span1 conflicts with HDLC with ... (Carlos Chavez)   Tzafrir Cohen
• [asterisk-users] ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ...   Tzafrir Cohen
• [asterisk-users] asterisk voicemail to email and relaying   Tzafrir Cohen
• [asterisk-users] asterisk voicemail to email and relaying   Tzafrir Cohen
• [asterisk-users] Multitenant or Multiple virtual machines   Tzafrir Cohen
• [asterisk-users] Udev issue on zaptel install   Tzafrir Cohen
• [asterisk-users] Build your own "appliance" concept   Tzafrir Cohen
• [asterisk-users] Multitenant or Multiple virtual machines   Tzafrir Cohen
• [asterisk-users] Build your own "appliance" concept   Tzafrir Cohen
• [asterisk-users] New Installed X100p   Tzafrir Cohen
• [asterisk-users] New Installed X100p   Tzafrir Cohen
• [asterisk-users] Multitenant or Multiple virtual machines   Tzafrir Cohen
• [asterisk-users] New Installed X100p   Tzafrir Cohen
• [asterisk-users] Configure extension by software   Tzafrir Cohen
• [asterisk-users] Udev issue on zaptel install   Tzafrir Cohen
• [asterisk-users] Asterisk on Ubuntu Feisty   Tzafrir Cohen
• [asterisk-users] online active call watching   Tzafrir Cohen
• [asterisk-users] TDM400P periodic sound clicks on FXS   Tzafrir Cohen
• [asterisk-users] Astribank 32 and Far End Disconnection   Tzafrir Cohen
• [asterisk-users] Direct dialing to correct extension from analog lines   Tzafrir Cohen
• [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?   Tzafrir Cohen
• [asterisk-users] Zap channels: no sound with certain call paths   Tzafrir Cohen
• [asterisk-users] Asterisk cli   Tzafrir Cohen
• [asterisk-users] how to disable wctdm auto modprobe during boot   Tzafrir Cohen
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Tzafrir Cohen
• [asterisk-users] ztdummy kills audio   Tzafrir Cohen
• [asterisk-users] ztdummy kills audio   Tzafrir Cohen
• [asterisk-users] ztdummy kills audio   Tzafrir Cohen
• [asterisk-users] Force a new user to configure Comedian mail?   Tzafrir Cohen
• [asterisk-users] ztdummy kills audio   Tzafrir Cohen
• [asterisk-users] Astribank and caller ID from PSTN   Tzafrir Cohen
• [asterisk-users] ztdummy kills audio   Tzafrir Cohen
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Tzafrir Cohen
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Tzafrir Cohen
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Tzafrir Cohen
• [asterisk-users] stop log/debug messages into /var/log/messages   Tzafrir Cohen
• [asterisk-users] how to disable wctdm auto modprobe during boot   Tzafrir Cohen
• [asterisk-users] how to disable wctdm auto modprobe during boot   Tzafrir Cohen
• [asterisk-users] Useful GUI? [Was: Why does everyone seem to dislike *now?]   Tzafrir Cohen
• [asterisk-users] Randomly half-voice at sip/zap   Tzafrir Cohen
• [asterisk-users] Randomly half-voice at sip/zap   Tzafrir Cohen
• [asterisk-users] Why does everyone seem to dislike *now?   Tzafrir Cohen
• [asterisk-users] Problem with asterisk-perl-0.08 and Asterisk >= 1.2.20   Tzafrir Cohen
• [asterisk-users] what is softswitch   Tzafrir Cohen
• [asterisk-users] Building an RPM from Asterisk 1.4   Tzafrir Cohen
• [asterisk-users] Horrible problem - calls losing sound   Tzafrir Cohen
• [asterisk-users] Building an RPM from Asterisk 1.4   Tzafrir Cohen
• [asterisk-users] Horrible problem - calls losing sound   Tzafrir Cohen
• [asterisk-users] Horrible problem - calls losing sound   Tzafrir Cohen
• [asterisk-users] Asterisk 1.2.13 and presence   Tzafrir Cohen
• [asterisk-users] Asterisk 1.2.13 and presence   Tzafrir Cohen
• [asterisk-users] errors messages in asterisk CLI   Tzafrir Cohen
• [asterisk-users] asterisk cli - vi keybindings ?   Tzafrir Cohen
• [asterisk-users] Asterisk 1.4.12 Release?   Tzafrir Cohen
• [asterisk-users] Do I need to run #modprobe zaptel for Digium   Tzafrir Cohen
• [asterisk-users] Zaptel-1.4.5.1 Compile Error   Tzafrir Cohen
• [asterisk-users] Do I need to run #modprobe zaptel for Digium   Tzafrir Cohen
• [asterisk-users] Zaptel-1.4.5.1 Compile Error   Tzafrir Cohen
• [asterisk-users] Music On Hold - How to increase volume ?   Tzafrir Cohen
• [asterisk-users] Zap channel stuck in conference   Tzafrir Cohen
• [asterisk-users] Changing contexts "on the fly"   Tzafrir Cohen
• [asterisk-users] Asterisk Redundancy   Tzafrir Cohen
• [asterisk-users] . (period): Wildcard match; matches one or more characters   Tzafrir Cohen
• [asterisk-users] FAX detection not working   Tzafrir Cohen
• [asterisk-users] Selecting a specific line from Zap/g   Tzafrir Cohen
• [asterisk-users] OT: DELL Platforms   Dean Collins
• [asterisk-users] OT: DELL Platforms   Dean Collins
• [asterisk-users] voice to text   Dean Collins
• [asterisk-users] Bug labs   Dean Collins
• [asterisk-users] Call Center SoftPhone with Auto Answer   Dean Collins
• [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO   Dean Collins
• [asterisk-users] IAX Java Softphone?   Dean Collins
• [asterisk-users] IAX Java Softphone?   Dean Collins
• [asterisk-users] Announcing: Click-to-Call with VIDEO   Dean Collins
• [asterisk-users] Asterisk and MS Exchange 2007   Dean Collins
• [asterisk-users] Dutch Number for Inbound   Dean Collins
• [asterisk-users] voip hacking article   Dean Collins
• [asterisk-users] Music On Hold - How to increase volume ?   Dean Collins
• [asterisk-users] Which Asterisk version to use?   Michael Collins
• [asterisk-users] Digium acquires Switchvox   Michael Collins
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Joshua Colp
• [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault   Joshua Colp
• [asterisk-users] phone as control interface (was 99 bottles of beer)   David Cook
• [asterisk-users] [on-asterisk] Configure one call per line on Cisco 7941/7961   David Cook
• [asterisk-users] [on-asterisk] Configure one call per line on Cisco 7941/7961   David Cook
• [asterisk-users] ZAP to invalid SIP device call looping   Gustavo Cordeiro
• [asterisk-users] Newcomer Question   Jan De Coster
• [asterisk-users] Asterisk w/MS SQL Server 2005   Larry Costigan
• [asterisk-users] swift.conf - cepstral voice quality adjustment options   Larry Costigan
• [asterisk-users] Very fast playback   Andre Courchesne
• [asterisk-users] Ping   Jonathan Creasy
• [asterisk-users] stop log/debug messages into /var/log/messages   Jonathan K. Creasy
• [asterisk-users] CallWithUs Service?   Doug Crompton
• [asterisk-users] USA Termination   Claude Cunningham
• [asterisk-users] Voicemail in 1.4?   Ken D'Ambrosio
• [asterisk-users] Useful GUI? [Was: Why does everyone seem to dislike *now?]   Ken D'Ambrosio
• [asterisk-users] openser/ser/Asterisk user meeting (beer drinking in Vienna)   Klaus Darilion
• [asterisk-users] Anyone use the Linksys phones?   Steve Davies
• [asterisk-users] Help with log entries.   Mike Diehl
• [asterisk-users] Help with log entries.   Mike Diehl
• [asterisk-users] Doesn't seem to want to transcode.   Mike Diehl
• [asterisk-users] Asterisk Died message   Nitesh Divecha
• [asterisk-users] DTMF Problem with International Calls   Nitesh Divecha
• [asterisk-users] Trunk & Outbound Route for a Cisco VOIP router?   Doug
• [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?   Doug
• [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?   Doug
• [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?   Doug
• [asterisk-users] What's the deal with ATAcomm?   Doug
• [asterisk-users] What's the deal with ATAcomm?   Doug
• [asterisk-users] Polycom 501 won't take new bootrom.ld or sip.ld...   Doug
• [asterisk-users] Asterisk Redundancy   Doug
• [asterisk-users] Asterisk Redundancy   Doug
• [asterisk-users] Asterisk Redundancy   Doug
• [asterisk-users] Meridian S1 to Asterisk via T1   Michelle Dupuis
• [asterisk-users] T1 to SIP conversion, standalone device   Michelle Dupuis
• [asterisk-users] Meridian S1 to Asterisk via T1   Michelle Dupuis
• [asterisk-users] [mythtv-users] Real Time Clock Alarm Broken with 2.6.22+ kernel   Michelle Dupuis
• [asterisk-users] phone as control interface (was 99 bottles of beer)   Steve Edwards
• [asterisk-users] Hints / State change on outgoing calls   Alex Epshteyn
• [asterisk-users] crash after callbackagent ackcall   Damon Estep
• [asterisk-users] crash after callbackagent ackcall   Damon Estep
• [asterisk-users] Escape characer for Digit Timeout   C F
• [asterisk-users] Overhead paging over IP...   C F
• [asterisk-users] SIP endpoint that does not register   C F
• [asterisk-users] canreinvite   C F
• [asterisk-users] nat=yes   C F
• [asterisk-users] What is the difference between increasing the verbose level and the debug level?   C F
• [asterisk-users] What is the difference between increasing the verbose level and the debug level?   C F
• [asterisk-users] nat=yes   C F
• [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.   C F
• [asterisk-users] nat=yes   C F
• [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.   C F
• [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.   C F
• [asterisk-users] canreinvite   C F
• [asterisk-users] Paging to external speaker like in airports etc...   C F
• [asterisk-users] sip.conf best practices?   C F
• [asterisk-users] OT: Samsung Sprint CDMAoIP   C F
• [asterisk-users] OT: Samsung Sprint CDMAoIP   C F
• [asterisk-users] OT: Samsung Sprint CDMAoIP   C F
• [asterisk-users] OT: Samsung Sprint CDMAoIP   C F
• [asterisk-users] OT: Samsung Sprint CDMAoIP   C F
• [asterisk-users] Paging to external speaker like in airports etc...   C F
• [asterisk-users] Asterisk with app_RPT question   F6HQZ
• [asterisk-users] sip.conf best practices?   John Faubion
• [asterisk-users] Grandstream GXP2020 / 2000   John Faubion
• [asterisk-users] Include zaptel in kernel...   Robert La Ferla
• [asterisk-users] CallerID problem Asterisk 1.4.2   Bruce Ferrell
• [asterisk-users] ADIT 600 & CMG <=> Asterisk question   Barton Fisher
• [asterisk-users] Append Extension number sounds to Voice Mail Message?   Barton Fisher
• [asterisk-users] ADIT 600 & CMG <=> Asterisk question   Barton Fisher
• [asterisk-users] Channels in use?   Barton Fisher
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Barton Fisher
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Barton Fisher
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Barton Fisher
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Barton Fisher
• [asterisk-users] DTMF dropping digits   Barton Fisher
• [asterisk-users] DTMF dropping digits   Barton Fisher
• [asterisk-users] ADIT TDM T1 <> Asterisk MGCP   Barton Fisher
• [asterisk-users] E1 to Ethernet Bridge   James FitzGibbon
• [asterisk-users] Asterisk Died message   James FitzGibbon
• [asterisk-users] Dialplan regexp   James FitzGibbon
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   James FitzGibbon
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   James FitzGibbon
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   James FitzGibbon
• [asterisk-users] Which cause less CPU usage: GSM or wav??   James FitzGibbon
• [asterisk-users] exit ChanSpy with DTMF   James FitzGibbon
• [asterisk-users] Voicemail in 1.4?   James FitzGibbon
• [asterisk-users] Agent Callback Login in 1.4   James FitzGibbon
• [asterisk-users] Force a new user to configure Comedian mail?   James FitzGibbon
• [asterisk-users] Why does everyone seem to dislike *now?   James FitzGibbon
• [asterisk-users] Enabling MySQL UNIQUE from cdr.conf   James FitzGibbon
• [asterisk-users] Call Center SoftPhone with Auto Answer   James FitzGibbon
• [asterisk-users] Softphone RTP Session Start-up Delay   James FitzGibbon
• [asterisk-users] Building an RPM from Asterisk 1.4   James FitzGibbon
• [asterisk-users] Hints / State change on outgoing calls   James FitzGibbon
• [asterisk-users] show queue (queue name)   James FitzGibbon
• [asterisk-users] Agent Callback Login in 1.4   Kevin P. Fleming
• [asterisk-users] Agent Callback Login in 1.4   Kevin P. Fleming
• [asterisk-users] Agent Callback Login in 1.4   Kevin P. Fleming
• [asterisk-users] Agent Callback Login in 1.4   Kevin P. Fleming
• [asterisk-users] Digium Appliance   Kevin P. Fleming
• [asterisk-users] Agent Callback Login in 1.4   Kevin P. Fleming
• [asterisk-users] Force a new user to configure Comedian mail?   Kevin P. Fleming
• [asterisk-users] g729 on 1.4.10.1   Kevin P. Fleming
• [asterisk-users] How to cancel the password check in VoicemailMain()   Kevin P. Fleming
• [asterisk-users] G.722: ast_channel_make_compatible failure   Kevin P. Fleming
• [asterisk-users] How to handle "+" prefix   Anthony Francis
• [asterisk-users] How to handle "+" prefix   Anthony Francis
• [asterisk-users] Installing Asterisk on to CentOS 4   Anthony Francis
• [asterisk-users] Installing Asterisk on to CentOS 4   Anthony Francis
• [asterisk-users] Agent Callback Login in 1.4   Anthony Francis
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   Anthony Francis
• [asterisk-users] Agent Callback Login in 1.4   Anthony Francis
• [asterisk-users] Agent Callback Login in 1.4   Anthony Francis
• [asterisk-users] Agent Callback Login in 1.4   Anthony Francis
• [asterisk-users] TDM400P   Anthony Francis
• [asterisk-users] Agent Callback Login in 1.4   Anthony Francis
• [asterisk-users] DECT SIP phones   Anthony Francis
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Anthony Francis
• [asterisk-users] outgoing call restriction in extention.conf   Anthony Francis
• [asterisk-users] Agent Callback Login in 1.4   Anthony Francis
• [asterisk-users] Help Drop Calls   Anthony Francis
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Anthony Francis
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Anthony Francis
• [asterisk-users] Why does everyone seem to dislike *now?   Anthony Francis
• [asterisk-users] Call Center SoftPhone with Auto Answer   Anthony Francis
• [asterisk-users] Comfort noice sample (gsm/mp3)   Anthony Francis
• [asterisk-users] what is softswitch   Anthony Francis
• [asterisk-users] what is softswitch   Anthony Francis
• [asterisk-users] AMI extension states   Anthony Francis
• [asterisk-users] AMI extension states   Anthony Francis
• [asterisk-users] AMI extension states   Anthony Francis
• [asterisk-users] AMI extension states   Anthony Francis
• [asterisk-users] Announcing: Click-to-Call with VIDEO ***SPAM***   Anthony Francis
• [asterisk-users] How to "busy out" zap channels   Anthony Francis
• [asterisk-users] 3-way calling   Anthony Francis
• [asterisk-users] Building an RPM from Asterisk 1.4   Marcus Franke
• [asterisk-users] Anyone use the Linksys phones? (Zeeshan Zakaria)   Norman Franke
• [asterisk-users] TDM400P (TDM22P) and aux power.   Matthew Fredrickson
• [asterisk-users] ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ...   Matthew Fredrickson
• [asterisk-users] No Dial tone came from fxs modules   Matthew Fredrickson
• [asterisk-users] New Installed X100p   Matthew Fredrickson
• [asterisk-users] 56k modem configuration   Matthew Fredrickson
• [asterisk-users] New Installed X100p   Matthew Fredrickson
• [asterisk-users] 56k modem configuration   Matthew Fredrickson
• [asterisk-users] TDM400P periodic sound clicks on FXS   Matthew Fredrickson
• [asterisk-users] TDM400P periodic sound clicks on FXS   Matthew Fredrickson
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   Matthew Fredrickson
• [asterisk-users] TE405P intermittent yellow alarm   Matthew Fredrickson
• [asterisk-users] g729 on 1.4.10.1   Matthew Fredrickson
• [asterisk-users] ISDN data packets   Matthew Fredrickson
• [asterisk-users] ISDN data packets   Matthew Fredrickson
• [asterisk-users] ISDN PRI debug in Asterisk   Matthew Fredrickson
• [asterisk-users] ISDN PRI debug in Asterisk   Matthew Fredrickson
• [asterisk-users] Hfcmulti and B410P Digium Card   Matthew Fredrickson
• [asterisk-users] what is softswitch   Matthew Fredrickson
• [asterisk-users] HOWTO/FAQ question (Location: Sweden)   Turbo Fredriksson
• [asterisk-users] HOWTO/FAQ question (Location: Sweden)   Turbo Fredriksson
• [asterisk-users] DTMF signalling, SIP, and Background()   Bastian Friedrich
• [asterisk-users] SIP interface status   James Fromm
• [asterisk-users] Overhead paging over IP...   Dave Fullerton
• [asterisk-users] Ping   Dave Fullerton
• [asterisk-users] Force a new user to configure Comedian mail?   Dave Fullerton
• [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?   Dave Fullerton
• [asterisk-users] How can i send my sip channel 3 to mailbox 2? Please Help!   Jonathan GF
• [asterisk-users] Rechazo de llamada en triangulacion de asterisk.   Jonathan GF
• [asterisk-users] Dificult macro, please advise   Jonathan GF
• [asterisk-users] Dificult macro, please advise   Jonathan GF
• [asterisk-users] How can i send my sip channel 3 to mailbox 2? Please Help!   Jonathan GF
• [asterisk-users] [Serusers] user meeting (beer drinking in Vienna)   Christian Gansberger
• [asterisk-users] Dial() Command Parameter L Overflow?   Douglas Garstang
• [asterisk-users] Building an RPM from Asterisk 1.4   Douglas Garstang
• [asterisk-users] Building an RPM from Asterisk 1.4   Douglas Garstang
• [asterisk-users] Confused about Asterisk 1.4 RTPQOS...   Douglas Garstang
• [asterisk-users] Polycom 501 Phones Rebooting   Douglas Garstang
• [asterisk-users] Asterisk Redundancy   Douglas Garstang
• [asterisk-users] Asterisk Redundancy   Douglas Garstang
• [asterisk-users] Asterisk Redundancy   Douglas Garstang
• [asterisk-users] Asterisk Redundancy   Douglas Garstang
• [asterisk-users] Asterisk Redundancy   Douglas Garstang
• [asterisk-users] DTMF bug in dsp.c and 1.4.11   Jerry Geis
• [asterisk-users] dtmfmode rfc2833 and info   Jerry Geis
• [asterisk-users] dtmf issues on PRI and 1.4.11   Jerry Geis
• [asterisk-users] E1 Line Tapping   Ricardo Gemignani
• [asterisk-users] E1 Line Tapping   Ricardo Gemignani
• [asterisk-users] E1 Line Tapping   Ricardo Gemignani
• [asterisk-users] Cisco UC 500   Drew Gibson
• [asterisk-users] how to route outgoing calls on IP-level   Drew Gibson
• [asterisk-users] sip.conf best practices?   Drew Gibson
• [asterisk-users] Grandstream GXP2020 / 2000   Drew Gibson
• [asterisk-users] Backports to 1.2.14 of 1.4.0 app_queue features.   Gary T. Giesen
• [asterisk-users] Configure one call per line on Cisco 7941/7961   Gary T. Giesen
• [asterisk-users] [on-asterisk] Configure one call per line on Cisco 7941/7961   Gary T. Giesen
• [asterisk-users] IAX Java Softphone?   Jean-Denis Girard
• [asterisk-users] IAX Java Softphone?   Jean-Denis Girard
• [asterisk-users] Escape characer for Digit Timeout   Giuffredi
• [asterisk-users] Escape characer for Digit Timeout   Giuffredi
• [asterisk-users] SIP Panel?   Terry Giufre-Sweetser
• [asterisk-users] Asterisk with app_RPT question   Mohamed A. Gombolaty
• [asterisk-users] Meridian S1 to Asterisk via T1   David Gomillion
• [asterisk-users] Partitioning DSL input   David Gomillion
• [asterisk-users] T1/PRI pricing   David Gomillion
• [asterisk-users] Call Center SoftPhone with Auto Answer   David Gomillion
• [asterisk-users] Music On Hold   David Gomillion
• [asterisk-users] Changing contexts "on the fly"   David Gomillion
• [asterisk-users] Routing issue   David Gonzalez
• [asterisk-users] Fwd: Routing issue   David Gonzalez
• [asterisk-users] TDM400P   Gustavo Gonzalez
• [asterisk-users] show queue (queue name)   Everton Goularth
• [asterisk-users] Presentation and mISDN   Giordano Grandis
• [asterisk-users] Build your own "appliance" concept   Michael Graves
• [asterisk-users] Build your own "appliance" concept   Michael Graves
• [asterisk-users] Flash IDE   Michael Graves
• [asterisk-users] Skype + Asterisk   Michael Graves
• [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.   Sanspareils Greenlans
• [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.   Sanspareils Greenlans
• [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.   Sanspareils Greenlans
• [asterisk-users] External FXO port.   Sanspareils Greenlans
• [asterisk-users] g729 on 1.4.10.1   Luke Groeneveld
• [asterisk-users] Multitenant or Multiple virtual machines   Edgar Guadamuz
• [asterisk-users] PCI: UNABLE TO HANDLE 64-bit ADDRESS SPACE FOR   Edgar Guadamuz
• [asterisk-users] errors messages in asterisk CLI   Jody Gugelhupf
• [asterisk-users] error messages related to mysql in asterisk CLI   Jody Gugelhupf
• [asterisk-users] Anyone use the Linksys phones?   Craig Guy
• [asterisk-users] Asterisk Redundancy   Guilherme Loch Waltrick Góes
• [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI SOLVED!!   Marc Patino Gómez
• [asterisk-users] Backuping VoIP provider with PRI   Marc Patino Gómez
• [asterisk-users] Backuping VoIP provider with PRI   Marc Patino Gómez
• [asterisk-users] Backuping VoIP provider with PRI   Marc Patino Gómez
• [asterisk-users] VSP authentication to incorrect context   Paul Hales
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Paul Hales
• [asterisk-users] TDM400P (TDM22P) and aux power.   Paul Hales
• [asterisk-users] Issue with calling queues   Paul Hales
• [asterisk-users] Problems with two trunks   Paul Hales
• [asterisk-users] Conference bridge.   Paul Hales
• [asterisk-users] Agent Callback Login in 1.4   Paul Hales
• [asterisk-users] Agent Callback Login in 1.4   Paul Hales
• [asterisk-users] FW: Problems with two trunks   Paul Hales
• [asterisk-users] Agent Callback Login in 1.4   Paul Hales
• [asterisk-users] Agent Callback Login in 1.4   Paul Hales
• [asterisk-users] Voicemail.conf   Paul Hales
• [asterisk-users] Voicemail.conf   Paul Hales
• [asterisk-users] Voicemail.conf   Paul Hales
• [asterisk-users] sip.conf best practices?   Paul Hales
• [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?   Paul Hales
• [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?   Paul Hales
• [asterisk-users] 3-way calling   Paul Hales
• [asterisk-users] 3-way calling   Paul Hales
• [asterisk-users] Ping   Mike Hammett
• [asterisk-users] Ping   Mike Hammett
• [asterisk-users] Ping   Mike Hammett
• [asterisk-users] Ping   Mike Hammett
• [asterisk-users] Ping   Mike Hammett
• [asterisk-users] Different Networks   Mike Hammett
• [asterisk-users] Different Networks   Mike Hammett
• [asterisk-users] Different Networks   Mike Hammett
• [asterisk-users] Different Networks   Mike Hammett
• [asterisk-users] Different Networks   Mike Hammett
• [asterisk-users] Asterisk 1.4 and Cepstral   Russell Handorf
• [asterisk-users] Siemans SIP/PSTN phone S450   Paul Hayes
• [asterisk-users] Random Double Digits   Daniel Hazelbaker
• [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk   Gordon Henderson
• [asterisk-users] Build your own "appliance" concept   Gordon Henderson
• [asterisk-users] Siemans SIP/PSTN phone S450   Gordon Henderson
• [asterisk-users] Flash IDE   Gordon Henderson
• [asterisk-users] Generating an old-fashioned dialtone   Gordon Henderson
• [asterisk-users] Flash IDE   Gordon Henderson
• [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?   Gordon Henderson
• [asterisk-users] how to route outgoing calls on IP-level   Gordon Henderson
• [asterisk-users] Asterisk 1.2.24 simultaneous call limits.   Gordon Henderson
• [asterisk-users] "HiarPinning" via TDM400 in the UK ...   Gordon Henderson
• [asterisk-users] Grandstream GXP2020 / 2000   Gordon Henderson
• [asterisk-users] IAX configuration   Gordon Henderson
• [asterisk-users] Astribank 32 and Far End Disconnection   Gleidson Antonio Henriques
• [asterisk-users] Anyone else having problems with the list   Carlos Hernandez
• [asterisk-users] help with channelbank audiocodes MP-124   Carlos Hernandez
• [asterisk-users] Spur error with Siemens Hi Path   Ruairi Hickey
• [asterisk-users] Music On Hold   Joel Hill
• [asterisk-users] Music On Hold   Joel Hill
• [asterisk-users] Voicemail.conf   Rob Hillis
• [asterisk-users] Agent Callback Login in 1.4   Rob Hillis
• [asterisk-users] Paging to external speaker like in airports etc...   Lorentz Hinrichsen
• [asterisk-users] Manager connection timeout   Rizwan Hisham
• [asterisk-users] Prevent multiple sip registrations   Rizwan Hisham
• [asterisk-users] Prevent multiple sip registrations   Rizwan Hisham
• [asterisk-users] Prevent multiple sip registrations   Rizwan Hisham
• [asterisk-users] Prevent multiple sip registrations   Rizwan Hisham
• [asterisk-users] Prevent multiple sip registrations   Rizwan Hisham
• [asterisk-users] Asterisk cli   Rizwan Hisham
• [asterisk-users] Asterisk cli   Rizwan Hisham
• [asterisk-users] Asterisk cli   Rizwan Hisham
• [asterisk-users] TE405P intermittent yellow alarm   Richard van der Hoff
• [asterisk-users] TE405P intermittent yellow alarm   Richard van der Hoff
• [asterisk-users] TE405P intermittent yellow alarm   Richard van der Hoff
• [asterisk-users] unnumbered priorities   Anselm Martin Hoffmeister
• [asterisk-users] FAX machine connect with audiocode SIP device   Anselm Martin Hoffmeister
• [asterisk-users] alphabetical extension patterns   Anselm Martin Hoffmeister
• [asterisk-users] asterisk voicemail to email and relaying   Anselm Martin Hoffmeister
• [asterisk-users] Strange Behaviour   Anselm Martin Hoffmeister
• [asterisk-users] Strange Behaviour   Anselm Martin Hoffmeister
• [asterisk-users] alphabetical extension patterns   Anselm Martin Hoffmeister
• [asterisk-users] Prevent multiple sip registrations   Anselm Martin Hoffmeister
• [asterisk-users] Filesharing + video + voice supported Soft phone   Anselm Martin Hoffmeister
• [asterisk-users] Softphone RTP Session Start-up Delay   Anselm Martin Hoffmeister
• [asterisk-users] (Getting OT) Re: Call Center SoftPhone with Auto Answer   Anselm Martin Hoffmeister
• [asterisk-users] VoIP Provider for business   Anselm Martin Hoffmeister
• [asterisk-users] Newcomer Question   Anselm Martin Hoffmeister
• [asterisk-users] Problems sending more than 2 SMS with asterisk / smsq   Anselm Martin Hoffmeister
• [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO   Anselm Martin Hoffmeister
• [asterisk-users] Completing my Configuration   Anselm Martin Hoffmeister
• [asterisk-users] Hola Jonathan, a ver si tre suena...   Anselm Martin Hoffmeister
• [asterisk-users] DTMF signalling, SIP, and Background()   Anselm Martin Hoffmeister
• [asterisk-users] FAX detection not working   Lee Howard
• [asterisk-users] Video doesn't work for outgoing call?   Chih-Wei Huang
• [asterisk-users] Sort of OT: PBX vs CO   John Hughes
• [asterisk-users] SMS in France - allways get "NAK"   John Hughes
• [asterisk-users] SMS in France - allways get "NAK"   John Hughes
• [asterisk-users] Horrible problem - calls losing sound   John Hughes
• [asterisk-users] Horrible problem - calls losing sound   John Hughes
• [asterisk-users] Horrible problem - calls losing sound   John Hughes
• [asterisk-users] Horrible problem - calls losing sound   John Hughes
• [asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt   John Hughes
• [asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt   John Hughes
• [asterisk-users] ISDN2#02: too much voice to send for NCCI=0x40502   John Hughes
• [asterisk-users] Call Center SoftPhone with Auto Answer   Kenneth T. Van Wie II
• [asterisk-users] Call Center SoftPhone with Auto Answer   Kenneth T. Van Wie II
• [asterisk-users] Problems bringing up ZAP trunks via PRI   Norman D Eisley III
• [asterisk-users] Flash IDE   Diego Iastrubni
• [asterisk-users] Useful GUI? [Was: Why does everyone seem to dislike *now?]   Diego Iastrubni
• [asterisk-users] New Project: AskoziaPBX   Michael Iedema
• [asterisk-users] AGI/PHP: missing arguments   Nasir Iqbal
• [asterisk-users] Astribank and caller ID from PSTN   Nasir Iqbal
• [asterisk-users] AGI/PHP: missing arguments   Nasir Iqbal
• [asterisk-users] Asterisk realtime error   Nasir Iqbal
• [asterisk-users] Partitioning DSL input   Ira
• [asterisk-users] DTMF   Ira
• [asterisk-users] CallWithUs Service?   Ira
• [asterisk-users] Paging to external speaker like in airports etc...   Ira
• [asterisk-users] RTP Read too short with T.38   Irmantas
• [asterisk-users] E1 to Ethernet Bridge   Arinze Izukanne
• [asterisk-users] E1 to Ethernet Bridge   Arinze Izukanne
• [asterisk-users] Howto pickup call from queue?   Jack
• [asterisk-users] Skype + Asterisk   Eric Jacksch
• [asterisk-users] Anyone use the Linksys phones?   Eric Jacksch
• [asterisk-users] stop log/debug messages into /var/log/messages   Benjamin Jacob
• [asterisk-users] stop log/debug messages into /var/log/messages   Benjamin Jacob
• [asterisk-users] stop log/debug messages into /var/log/messages   Benjamin Jacob
• [asterisk-users] stop log/debug messages into /var/log/messages   Benjamin Jacob
• [asterisk-users] stop log/debug messages into /var/log/messages   Benjamin Jacob
• [asterisk-users] alphabetical extension patterns   Benjamin Jacob
• [asterisk-users] nat=yes   Benjamin Jacob
• [asterisk-users] alphabetical extension patterns   Benjamin Jacob
• [asterisk-users] stop log/debug messages into /var/log/messages   Benjamin Jacob
• [asterisk-users] Linux limits   Benjamin Jacob
• [asterisk-users] prepaid application recommendation   Benjamin Jacob
• [asterisk-users] running twice   Benjamin Jacob
• [asterisk-users] Prevent multiple sip registrations   Raj Jain
• [asterisk-users] Multi-sip rings   Raj Jain
• [asterisk-users] Cepstral's Allison is having troublespeaking clearly   Kai-Uwe Jensen
• [asterisk-users] Cepstral's Allison is having troublespeaking clearly   Kai-Uwe Jensen
• [asterisk-users] swift.conf - cepstral voice quality adjustment options   Kai-Uwe Jensen
• [asterisk-users] Asterisk Redundancy   Per Jessen
• [asterisk-users] Asterisk Redundancy   Per Jessen
• [asterisk-users] Asterisk Redundancy   Per Jessen
• [asterisk-users] Asterisk Redundancy   Per Jessen
• [asterisk-users] Asterisk Redundancy   Per Jessen
• [asterisk-users] Non-USASCII chars in sip.conf?   Per Jessen
• [asterisk-users] E1 Line Tapping   Andrew Joakimsen
• [asterisk-users] special kind of billing   Andrew Joakimsen
• [asterisk-users] rxfax() problem - fax signal seems to be ignored   Andrew Joakimsen
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Andrew Joakimsen
• [asterisk-users] How to "busy out" zap channels   Andrew Joakimsen
• [asterisk-users] IAX gsm bandwith calls   Andrew Joakimsen
• [asterisk-users] Proximity Detection: Motorola Q + Bluetooth + Asterisk   Andrew Joakimsen
• [asterisk-users] What's the deal with ATAcomm?   Andrew Joakimsen
• [asterisk-users] redendent asterisk server for backup   Ron Joffe
• [asterisk-users] Overhead paging over IP...   Bryan M. Johns
• [asterisk-users] Wondering why I can't post   Bryan M. Johns
• [asterisk-users] Prevent multiple sip registrations   Jerry Jones
• [asterisk-users] Linux-HA and Asterisk   Jerry Jones
• [asterisk-users] What's the deal with ATAcomm?   Jerry Jones
• [asterisk-users] Multitenant or Multiple virtual machines   Senad Jordanovic
• [asterisk-users] Multitenant or Multiple virtual machines   Senad Jordanovic
• [asterisk-users] Multitenant or Multiple virtual machines   Senad Jordanovic
• [asterisk-users] Which Asterisk version to use?   Joseph
• [asterisk-users] Help with Sip Registration   Treesa Fairy Joseph
• [asterisk-users] SIP Panel?   Walt Joyce
• [asterisk-users] outgoing call restriction in extention.conf   Keshav K.
• [asterisk-users] Enabling MySQL UNIQUE from cdr.conf   Keshav K.
• [asterisk-users] Voicemail.conf   Adam KOSA
• [asterisk-users] Backuping VoIP provider with PRI   Adam KOSA
• [asterisk-users] AGI/PHP: missing arguments   Michael Kamleitner
• [asterisk-users] AGI/PHP: missing arguments   Michael Kamleitner
• [asterisk-users] Grandstream GXW-4008   Don Kelly
• [asterisk-users] remove unnecessary text (was: Re: Can asterisk give half-ring periodically for MWI?)   Philipp Kempgen
• [asterisk-users] remove unnecessary text   Philipp Kempgen
• [asterisk-users] AGI/PHP: missing arguments   Philipp Kempgen
• [asterisk-users] AMI extension states   Philipp Kempgen
• [asterisk-users] Asterisk on Fedora Core 4   Philipp Kempgen
• [asterisk-users] AMI extension states   Philipp Kempgen
• [asterisk-users] AMI extension states   Philipp Kempgen
• [asterisk-users] Asterisk Redundancy   Philipp Kempgen
• [asterisk-users] show queue (queue name)   Philipp Kempgen
• [asterisk-users] Asterisk Redundancy   Philipp Kempgen
• [asterisk-users] Asterisk Redundancy   Philipp Kempgen
• [asterisk-users] . (period): Wildcard match; matches one or more characters   Philipp Kempgen
• [asterisk-users] echo problems   Philipp Kempgen
• [asterisk-users] TDM400P (TDM22P) and aux power.   Thomas Kenyon
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Thomas Kenyon
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Thomas Kenyon
• [asterisk-users] Replacing an SPA 3000   Thomas Kenyon
• [asterisk-users] Paging to external speaker like in airports etc...   Anthony Kepler
• [asterisk-users] special kind of billing   Mindaugas Kezys
• [asterisk-users] Build your own "appliance" concept   Kristian Kielhofner
• [asterisk-users] AstLinux 0.4.8 Released   Kristian Kielhofner
• [asterisk-users] Followme app_followme   Kevin Kiely
• [asterisk-users] Followme app_followme   Kevin Kiely
• [asterisk-users] ATS X10001P   Kevin Kiely
• [asterisk-users] Tone detection while Dialing   Jason Kim
• [asterisk-users] Assistance needed.   Tim King
• [asterisk-users] VSP authentication to incorrect context   Klaverstyn, David C
• [asterisk-users] VSP authentication to incorrect context   Klaverstyn, David C
• [asterisk-users] Voxalot User and Peer details.   Klaverstyn, David C
• [asterisk-users] Anyone use the Linksys phones?   Klaverstyn, David C
• [asterisk-users] Asterisk on Fedora Core 4   Barry L. Kline
• [asterisk-users] h.323 out of media path   Lars Knopf
• [asterisk-users] What's the deal with ATAcomm?   Andrew Kohlsmith
• [asterisk-users] Flash IDE   Stelios Koroneos
• [asterisk-users] app_conference   Anton Krall
• [asterisk-users] app_conference   Anton Krall
• [asterisk-users] bug in 1.2.24   Anton Krall
• [asterisk-users] bug in 1.2.24   Anton Krall
• [asterisk-users] bug in 1.2.24   Anton Krall
• [asterisk-users] CallerID problem Asterisk 1.4.2   Peter Kranz
• [asterisk-users] special kind of billing   Kate Kretz
• [asterisk-users] Asterisk + LDAP or RADIUS   Kate Kretz
• [asterisk-users] how to route outgoing calls on IP-level   Kate Kretz
• [asterisk-users] how to route outgoing calls on IP-level   Kate Kretz
• [asterisk-users] how to route outgoing calls on IP-level   Kate Kretz
• [asterisk-users] how to route outgoing calls on IP-level   Kate Kretz
• [asterisk-users] How to handle "+" prefix   Brandon Kruse
• [asterisk-users] Useful GUI? [Was: Why does everyone seem to dislike *now?]   Brandon Kruse
• [asterisk-users] Useful GUI? [Was: Why does everyone seem to dislike *now?]   Brandon Kruse
• [asterisk-users] RTP Call Disconnect   Arun Kumar
• [asterisk-users] DECT SIP phones   Håkan Källberg
• [asterisk-users] loop detected   Pezhman Lali
• [asterisk-users] running twice   Pezhman Lali
• [asterisk-users] running twice   Pezhman Lali
• [asterisk-users] What is the difference between increasingtheverbose level and the debug level?   Steve Langstaff
• [asterisk-users] how to determine if a SIP extension has DND on oroff   Steve Langstaff
• [asterisk-users] how to determine if a SIP extension has DND onoroff   Steve Langstaff
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Steve Langstaff
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Steve Langstaff
• [asterisk-users] Overhead paging over IP...   Andrew Latham
• [asterisk-users] E1 Line Tapping   Andrew Latham
• [asterisk-users] E1 Line Tapping   Andrew Latham
• [asterisk-users] DECT SIP phones   Andrew Latham
• [asterisk-users] AAI2UUI - how?   Christophorus Laube
• [asterisk-users] Digium 4FXO PCI card or Grandstream 4FXO Gateway?   Alejandro Lengua
• [asterisk-users] Skype + Asterisk   Alejandro Lengua
• [asterisk-users] Looking for recommendations on Nufone and Gamachi   Alejandro Lengua
• [asterisk-users] Looking for a reliable source for DIDs from Mexico, Colombia and Venezuela   Alejandro Lengua
• [asterisk-users] Voicemail in 1.4?   Tilghman Lesher
• [asterisk-users] DECT SIP phones   Tilghman Lesher
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Tilghman Lesher
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Tilghman Lesher
• [asterisk-users] Voicemail.conf   Tilghman Lesher
• [asterisk-users] Enabling MySQL UNIQUE from cdr.conf   Tilghman Lesher
• [asterisk-users] ATS X10001P   Tilghman Lesher
• [asterisk-users] Interesting Conference Request - Can this be done ?   Tilghman Lesher
• [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO   Tilghman Lesher
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Tilghman Lesher
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Tilghman Lesher
• [asterisk-users] Realtime table columns   Tilghman Lesher
• [asterisk-users] Zaptel-1.4.5.1 Compile Error   Tilghman Lesher
• [asterisk-users] Zaptel-1.4.5.1 Compile Error   Tilghman Lesher
• [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828   Tilghman Lesher
• [asterisk-users] asterisk audits   Tilghman Lesher
• [asterisk-users] Music On Hold   Tilghman Lesher
• [asterisk-users] asterisk crash and core dump   Atis Lezdins
• [asterisk-users] Call Center SoftPhone with Auto Answer   Atis Lezdins
• [asterisk-users] AgentCalbackLogin not loging in race condition ?   Atis Lezdins
• [asterisk-users] AgentCalbackLogin not loging in race condition ?   Atis Lezdins
• [asterisk-users] Queue serializes call delivery ?   Atis Lezdins
• [asterisk-users] How to cancel the password check in VoicemailMain()   Atis Lezdins
• [asterisk-users] asterisk crash and core dump: format_mp3.so   Atis Lezdins
• [asterisk-users] Asterisk and MS Exchange 2007   Atis Lezdins
• [asterisk-users] Authenticate() application and CDR   Atis Lezdins
• [asterisk-users] # to transfer calls   Atis Lezdins
• [asterisk-users] Asterisk crash and debug   Atis Lezdins
• [asterisk-users] Asterisk Redundancy   Atis Lezdins
• [asterisk-users] Asterisk Redundancy   Atis Lezdins
• [asterisk-users] call relation in call transfer   Atis Lezdins
• [asterisk-users] 3-way calling   Atis Lezdins
• [asterisk-users] Cisco 7960 or 7960G   Robert Lister
• [asterisk-users] How can i send my sip channel 3 to mailbox 2? Please Help!   Robert Lister
• [asterisk-users] off-hook warning tone   Robert Lister
• [asterisk-users] off-hook warning tone   Robert Lister
• [asterisk-users] Limiting Simultaneous calls   Robert Lister
• [asterisk-users] Queue agents w/ DUNDi   Robert Lister
• [asterisk-users] Ringing Groups, SIP Forward and looping problem   Robert Lister
• [asterisk-users] Ringing Groups, SIP Forward and looping problem   Robert Lister
• [asterisk-users] Asterisk Died message   Perssy Llamosas
• [asterisk-users] IAX2 NAT issues   Perssy Llamosas
• [asterisk-users] . (period): Wildcard match; matches one or more characters   Perssy Llamosas
• [asterisk-users] Mutipoint Conferencing?   Alexander Lopez
• [asterisk-users] Interesting Conference Request - Can this be done ?   Alexander Lopez
• [asterisk-users] OT: Samsung Sprint CDMAoIP   Alexander Lopez
• [asterisk-users] OT: Samsung Sprint CDMAoIP   Alexander Lopez
• [asterisk-users] Networking Question   Alexander Lopez
• [asterisk-users] Music On Hold   Alexander Lopez
• [asterisk-users] TE405P intermittent yellow alarm   Richard Lyman
• [asterisk-users] AGI script fails on IAX channels (from call file).   Richard Lyman
• [asterisk-users] CallerID problem Asterisk 1.4.2   Richard Lyman
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Richard Lyman
• [asterisk-users] Help needed - ISDN is "redialling"   Julian Lyndon-Smith
• [asterisk-users] GROUP() issues for me   Julian Lyndon-Smith
• [asterisk-users] Anyone else having problems with the list   Julian Lyndon-Smith
• [asterisk-users] Maximum retries exceeded on transmission   Tom Lynn
• [asterisk-users] Overhead paging over IP...   Doug Lytle
• [asterisk-users] Ping   Doug Lytle
• [asterisk-users] Build your own "appliance" concept   Doug Lytle
• [asterisk-users] online active call watching   Doug Lytle
• [asterisk-users] Voicemail in 1.4?   Doug Lytle
• [asterisk-users] Force a new user to configure Comedian mail?   Doug Lytle
• [asterisk-users] Force a new user to configure Comedian mail?   Doug Lytle
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Doug Lytle
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Doug Lytle
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Doug Lytle
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Doug Lytle
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Doug Lytle
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Doug Lytle
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Doug Lytle
• [asterisk-users] Busy problem   Doug Lytle
• [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME   Doug Lytle
• [asterisk-users] # to transfer calls   Doug Lytle
• [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME   Doug Lytle
• [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME   Doug Lytle
• [asterisk-users] Ast_log   Doug Lytle
• [asterisk-users] DISA and DTMF detection problem w/ FXO port on a TDM400   Benjamin M.
• [asterisk-users] special kind of billing   Guillermo Salas M.
• [asterisk-users] special kind of billing   Guillermo Salas M.
• [asterisk-users] External FXO port.   Guillermo Salas M.
• [asterisk-users] Astribank and caller ID from PSTN   Guillermo Salas M.
• [asterisk-users] Astribank and caller ID from PSTN   Guillermo Salas M.
• [asterisk-users] Astribank and caller ID from PSTN   Guillermo Salas M.
• [asterisk-users] Astribank and caller ID from PSTN   Guillermo Salas M.
• [asterisk-users] IAX Java Softphone?   Guillermo Salas M.
• [asterisk-users] Configuring Loose routing method   Frederico Madeira
• [asterisk-users] Asterisk as Media Server   Frederico Madeira
• [asterisk-users] Asterisk 1.4 Ignoring SIP ACK's on 487 Responses   Grey Man
• [asterisk-users] Inbound SIP issues   Jeremy Mann
• [asterisk-users] Failover SIP logic   Jeremy Mann
• [asterisk-users] Failover SIP logic   Jeremy Mann
• [asterisk-users] Cisco UC 500   Jeremy Mann
• [asterisk-users] g729 on 1.4.10.1   Jeremy Mann
• [asterisk-users] Queue Question   Jeremy Mann
• [asterisk-users] Multiple Home system with SIP   Jeremy Mann
• [asterisk-users] Multiple Home system with SIP   Jeremy Mann
• [asterisk-users] Multiple Home system with SIP   Jeremy Mann
• [asterisk-users] Voicemail.conf   Ajay Mansingka
• [asterisk-users] Udev issue on zaptel install   Markham, Craig (FRTC Contractor)
• [asterisk-users] Udev issue on zaptel install   Markham, Craig (FRTC Contractor)
• [asterisk-users] Udev issue on zaptel install   Markham, Craig (FRTC Contractor)
• [asterisk-users] How to handle "+" prefix   Adrian Marsh
• [asterisk-users] stop log/debug messages into /var/log/messages   Adrian Marsh
• [asterisk-users] SIPBroker vs SIPgate   Adrian Marsh
• [asterisk-users] stop log/debug messages into /var/log/messages   Adrian Marsh
• [asterisk-users] SIPBroker vs SIPgate   Adrian Marsh
• [asterisk-users] Dialplan regexp   Adrian Marsh
• [asterisk-users] SIPBroker vs SIPgate   Adrian Marsh
• [asterisk-users] Dialplan regexp   Adrian Marsh
• [asterisk-users] Broken UDP streams   Adrian Marsh
• [asterisk-users] Forgotten SIP session   Adrian Marsh
• [asterisk-users] Siemans SIP/PSTN phone S450   Adrian Marsh
• [asterisk-users] Prevent multiple sip registrations   Adrian Marsh
• [asterisk-users] Prevent multiple sip registrations   Adrian Marsh
• [asterisk-users] Prevent multiple sip registrations   Adrian Marsh
• [asterisk-users] Asterisk voice quality tuning   Adrian Marsh
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Adrian Marsh
• [asterisk-users] Multi-sip rings   Adrian Marsh
• [asterisk-users] Asterisk Redundancy   Adrian Marsh
• [asterisk-users] Asterisk Redundancy   Adrian Marsh
• [asterisk-users] SIP Debugging to separate log file   Jason Martin
• [asterisk-users] Outgoing SIP packets out of order?   Jason Martin
• [asterisk-users] Zap channel stuck in conference   Jason Martin
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Matt
• [asterisk-users] unsuscribe   Matt
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Matt
• [asterisk-users] Digium Appliance   Matt
• [asterisk-users] Digium Appliance   Matt
• [asterisk-users] G729 copy protection   Bruce McAlister
• [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault   Bruce McAlister
• [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault   Bruce McAlister
• [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault   Bruce McAlister
• [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault   Bruce McAlister
• [asterisk-users] g729 on 1.4.10.1   Bruce McAlister
• [asterisk-users] Asterisk 1.4.12 Release?   Bruce McAlister
• [asterisk-users] Grandstream HT 502 ATA stops receiving calls   Antoine Megalla
• [asterisk-users] ISDN data packets   Arpit Mehta
• [asterisk-users] ISDN PRI debug in Asterisk   Arpit Mehta
• [asterisk-users] ISDN data packets   Arpit Mehta
• [asterisk-users] How to make call from asterisk?   Bhrugu Mehta
• [asterisk-users] Asterisk cli   Bhrugu Mehta
• [asterisk-users] Looking for Asterisk Consultant in San Franicsco   Bhrugu Mehta
• [asterisk-users] Skype + Asterisk   John Meksavan
• [asterisk-users] CallWithUs Service?   John Meksavan
• [asterisk-users] CallWithUs Service?   John Meksavan
• [asterisk-users] Skype + Asterisk   John Meksavan
• [asterisk-users] UNICALL MFC/R2 + Asterisk 1.4   Ricardo Melendez
• [asterisk-users] PCI: UNABLE TO HANDLE 64-bit ADDRESS SPACE FOR   Ricardo Melendez
• [asterisk-users] IAX gsm bandwith calls   Dario Mendez
• [asterisk-users] AgentCalbackLogin not loging in race condition ?   Carlos G Mendioroz
• [asterisk-users] AgentCalbackLogin not loging in race condition ?   Carlos G Mendioroz
• [asterisk-users] AgentCalbackLogin not loging in race condition ?   Carlos G Mendioroz
• [asterisk-users] Queue serializes call delivery ?   Carlos G Mendioroz
• [asterisk-users] off-hook warning tone   Anthony Messina
• [asterisk-users] off-hook warning tone   Anthony Messina
• [asterisk-users] off-hook warning tone   Anthony Messina
• [asterisk-users] off-hook warning tone   Anthony Messina
• [asterisk-users] No Dial tone came from fxs modules   Anthony Messina
• [asterisk-users] Musiconhold instead ringing   Anthony Messina
• [asterisk-users] Generating an old-fashioned dialtone   Anthony Messina
• [asterisk-users] CallWithUs Service?   Anthony Messina
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Anthony Messina
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Anthony Messina
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Anthony Messina
• [asterisk-users] Realtime table columns   Anthony Messina
• [asterisk-users] Realtime table columns   Anthony Messina
• [asterisk-users] IAX configuration   Anthony Messina
• [asterisk-users] VoIP, What Do you really need ?   Giovanni Miano
• [asterisk-users] VoIP, Asterisk, What Do you really need ?   Giovanni Miano
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Mark Michelson
• [asterisk-users] queue static agents   Mark Michelson
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Mark Michelson
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Mark Michelson
• [asterisk-users] Asterisk cli   Mark Michelson
• [asterisk-users] Force a new user to configure Comedian mail?   Mark Michelson
• [asterisk-users] How to cancel the password check in VoicemailMain()   Mark Michelson
• [asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt   Mark Michelson
• [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?   Mike
• [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?   Mike
• [asterisk-users] Digium Appliance   Noah Miller
• [asterisk-users] Generating an old-fashioned dialtone   Clayton Milos
• [asterisk-users] Maximum retries exceeded on transmission   Apa Minerala
• [asterisk-users] Maximum retries exceeded on transmission   Apa Minerala
• [asterisk-users] prepaid application recommendation   Apa Minerala
• [asterisk-users] echo problems   Apa Minerala
• [asterisk-users] off-hook warning tone   Mojo with Horan & Company, LLC
• [asterisk-users] Escape characer for Digit Timeout   Mojo with Horan & Company, LLC
• [asterisk-users] No Dial tone came from fxs modules   Mojo with Horan & Company, LLC
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Mojo with Horan & Company, LLC
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Mojo with Horan & Company, LLC
• [asterisk-users] online active call watching   Mojo with Horan & Company, LLC
• [asterisk-users] Flash IDE   Mojo with Horan & Company, LLC
• [asterisk-users] Flash IDE   Mojo with Horan & Company, LLC
• [asterisk-users] online active call watching   Mojo with Horan & Company, LLC
• [asterisk-users] Filesharing + video + voice supported Soft phone   Mojo with Horan & Company, LLC
• [asterisk-users] Slightly OT: Help choosing a free software license?   Mojo with Horan & Company, LLC
• [asterisk-users] Music On Hold   Mojo with Horan & Company, LLC
• [asterisk-users] IAX configuration   Mojo with Horan & Company, LLC
• [asterisk-users] Point-to-Point SIP link without registration   Mojo with Horan & Company, LLC
• [asterisk-users] Changing contexts "on the fly"   Mojo with Horan & Company, LLC
• [asterisk-users] . (period): Wildcard match; matches one or more characters   Mojo with Horan & Company, LLC
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Lacy Moore
• [asterisk-users] IAX gsm bandwith calls   Tom Moore
• [asterisk-users] IAX gsm bandwith calls   Tom Moore
• [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME   Luis Morales
• [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME   Luis Morales
• [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME   Luis Morales
• [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME   Luis Morales
• [asterisk-users] g729 on 1.4.10.1   Scott Moseman
• [asterisk-users] g729 on 1.4.10.1   Scott Moseman
• [asterisk-users] g729 on 1.4.10.1   Scott Moseman
• [asterisk-users] g729 on 1.4.10.1   Scott Moseman
• [asterisk-users] g729 on 1.4.10.1   Scott Moseman
• [asterisk-users] g729 on 1.4.10.1   Scott Moseman
• [asterisk-users] g729 on 1.4.10.1   Scott Moseman
• [asterisk-users] g729 on 1.4.10.1   Scott Moseman
• [asterisk-users] g729 on 1.4.10.1   Scott Moseman
• [asterisk-users] g729 on 1.4.10.1   Scott Moseman
• [asterisk-users] Asterisk Redundancy   Scott Moseman
• [asterisk-users] My G729 problem re-visited   Scott Moseman
• [asterisk-users] unnumbered priorities   Tony Mountifield
• [asterisk-users] DTMF Relay Problems   Tony Mountifield
• [asterisk-users] Random Double Digits   Tony Mountifield
• [asterisk-users] DTMF Relay Problems   Tony Mountifield
• [asterisk-users] DTMF bug in dsp.c and 1.4.11   Tony Mountifield
• [asterisk-users] AGI script fails on IAX channels (from call file).   Tony Mountifield
• [asterisk-users] AGI script fails on IAX channels (from call file).   Tony Mountifield
• [asterisk-users] AGI script fails on IAX channels (from call file).   Tony Mountifield
• [asterisk-users] extensions for conference call   Tony Mountifield
• [asterisk-users] Limiting Simultaneous calls   Tony Mountifield
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Tony Mountifield
• [asterisk-users] DTMF dropping digits   Tony Mountifield
• [asterisk-users] Multiple Meet me conferences   Tony Mountifield
• [asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension   Devraj Mukherjee
• [asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension   Devraj Mukherjee
• [asterisk-users] How to make call from asterisk?   Devraj Mukherjee
• [asterisk-users] How to make call from asterisk?   Devraj Mukherjee
• [asterisk-users] Installing Asterisk on to CentOS 4   Devraj Mukherjee
• [asterisk-users] How to handle "+" prefix   Steve Murphy
• [asterisk-users] AEL missing in recent 1.2 releases?   Steve Murphy
• [asterisk-users] Dialplan regexp   Steve Murphy
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Steve Murphy
• [asterisk-users] Problem with asterisk-perl-0.08 and Asterisk >= 1.2.20   Benoît Mérouze
• [asterisk-users] Problem with asterisk-perl-0.08 and Asterisk >= 1.2.20   Benoît Mérouze
• [asterisk-users] Problem with asterisk-perl-0.08 and Asterisk >= 1.2.20   Benoît Mérouze
• [asterisk-users] Paging to external speaker like in airports etc...   Deepak Naidu
• [asterisk-users] online active call watching   Dinesh Nair
• [asterisk-users] Sysmaster and Asterisk   Mani Nair
• [asterisk-users] Sysmaster and Asterisk   Mani Nair
• [asterisk-users] Solution: Sysmaster and Asterisk   Mani Nair
• [asterisk-users] SIP INFO request in asterisk   Irakli Natsvlishvili
• [asterisk-users] Strange Behaviour   Il Neofita
• [asterisk-users] Strange Behaviour   Il Neofita
• [asterisk-users] Strange Behaviour   Il Neofita
• [asterisk-users] Call hangup after 60seconds   Il Neofita
• [asterisk-users] Zaptel ztdummy module causes playback to fail   Chris Nestrud
• [asterisk-users] ztdummy kills audio   Chris Nestrud
• [asterisk-users] ztdummy kills audio   Chris Nestrud
• [asterisk-users] ztdummy kills audio   Chris Nestrud
• [asterisk-users] Problem with asterisk 1.4.11 and playing files to meetme conference   Chris Nestrud
• [asterisk-users] Show Callee name on Display   Peder at NetworkOblivion
• [asterisk-users] T1 to SIP conversion, standalone device   Peder at NetworkOblivion
• [asterisk-users] CallWithUs Service?   Peder at NetworkOblivion
• [asterisk-users] MOH Files Volume   Peder at NetworkOblivion
• [asterisk-users] Asterisk realtime error   Peder at NetworkOblivion
• [asterisk-users] Cisco 7940G licensing with asterisk   Peder at NetworkOblivion
• [asterisk-users] Cisco 7940G licensing with asterisk   Peder at NetworkOblivion
• [asterisk-users] Asterisk and HPC Cluster   Felipe Neuwald
• [asterisk-users] # to transfer calls   VoIP Newbie
• [asterisk-users] # to transfer calls   VoIP Newbie
• [asterisk-users] Musiconhold instead ringing   Hoai-Anh Ngo-Vi
• [asterisk-users] No Sound on Zap Channels   Hoai-Anh Ngo-Vi
• [asterisk-users] Polycom 330 + Asterisk, phone locks up. * key will do it   Darren Nickerson
• [asterisk-users] Polycom 501 Phones Rebooting   Darren Nickerson
• [asterisk-users] Paging to external speaker like in airports etc...   John Novack
• [asterisk-users] Ast_log   Ed Nuñez
• [asterisk-users] ChanSpy issue   Ed Nuñez
• [asterisk-users] ChanSpy issue   Ed Nuñez
• [asterisk-users] Errors: Too many SIP headers and Unknown SDP media type in offer: video 10702 RTP/AVP 34 31   Alejandro Cabrera Obed
• [asterisk-users] Asterisk 1.2.13 and presence   Alejandro Cabrera Obed
• [asterisk-users] Call Center SoftPhone with Auto Answer   Daniel Varella de Oliveira
• [asterisk-users] OT - How to script softphone installation   Olivier
• [asterisk-users] Build your own "appliance" concept   Olivier
• [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.   Olivier
• [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.   Olivier
• [asterisk-users] Licensing and provisionning   Olivier
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Olivier
• [asterisk-users] Asterisk and MS Exchange 2007   Olivier
• [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828   Olivier
• [asterisk-users] Changing contexts "on the fly"   J. Oquendo
• [asterisk-users] AsteriskNOW + legacy PBX integration   Shina Owolabi
• [asterisk-users] Build your own "appliance" concept   Jeremy P
• [asterisk-users] Build your own "appliance" concept   Jeremy P
• [asterisk-users] Enabling MySQL UNIQUE from cdr.conf   Luís Palma
• [asterisk-users] Enabling MySQL UNIQUE from cdr.conf   Luís Palma
• [asterisk-users] Enabling MySQL UNIQUE from cdr.conf   Luís Palma
• [asterisk-users] Enabling MySQL UNIQUE from cdr.conf   Luís Palma
• [asterisk-users] Choppy sound while converting alaw to ulaw   Benoit Panizzon
• [asterisk-users] Register Extension   Tim Panton
• [asterisk-users] Flash IDE   Tim Panton
• [asterisk-users] Mutipoint Conferencing?   Tim Panton
• [asterisk-users] Paging to external speaker like in airports etc...   Tim Panton
• [asterisk-users] Why does everyone seem to dislike *now?   Tim Panton
• [asterisk-users] Call Center SoftPhone with Auto Answer   Tim Panton
• [asterisk-users] IAX Java Softphone?   Tim Panton
• [asterisk-users] Register Extension   Tim H. Panton
• [asterisk-users] IAX2 NAT issues   Tim H. Panton
• [asterisk-users] res_snmp   Tim H. Panton
• [asterisk-users] online active call watching   Tim H. Panton
• [asterisk-users] res_snmp   Tim H. Panton
• [asterisk-users] TDM400P (TDM22P) and aux power.   Jason Parker
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Jason Parker
• [asterisk-users] Followme app_followme   Jason Parker
• [asterisk-users] g729 on 1.4.10.1   Jason Parker
• [asterisk-users] OT: Samsung Sprint CDMAoIP   Jason Parker
• [asterisk-users] Call Center SoftPhone with Auto Answer   Ed Pastore
• [asterisk-users] ztdummy problem in fedora7, kernel 2.6.22.5-76.fc7   Patrick
• [asterisk-users] Horrible problem - calls losing sound   Patrick
• [asterisk-users] Cisco 7940G licensing with asterisk   Patrick
• [asterisk-users] Rechazo de llamada en triangulacion de asterisk.   Paul
• [asterisk-users] OT: Samsung Sprint CDMAoIP   Paul
• [asterisk-users] OT: Samsung Sprint CDMAoIP   Paul
• [asterisk-users] Help with log entries.   Paul
• [asterisk-users] Music On Hold - How to increase volume ?   Rogerio Pazini
• [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?   Euler Pereira
• [asterisk-users] Call Center SoftPhone with Auto Answer   Joao Pereira
• [asterisk-users] Call Center SoftPhone with Auto Answer   Joao Pereira
• [asterisk-users] Asterisk crash and debug   Joao Pereira
• [asterisk-users] Cisco 7940G licensing with asterisk   Erick Perez
• [asterisk-users] Cisco 7940G licensing with asterisk   Erick Perez
• [asterisk-users] odd audio problem   Peter
• [asterisk-users] Virtual server Solution   Bincy K. Philip
• [asterisk-users] Analog Telephone Adapter   Bincy K. Philip
• [asterisk-users] Which Asterisk version to use?   Bob Pierce
• [asterisk-users] rxfax() problem - fax signal seems to be ignored   Pirlouwi
• [asterisk-users] rxfax() problem - fax signal seems to be ignored   Pirlouwi
• [asterisk-users] (txfax+spandsp) fax is successfully sent, but Asterisk keeps sending DelayedRetries.   Pirlouwi
• [asterisk-users] Installing Asterisk on to CentOS 4   Stanisław Pitucha
• [asterisk-users] TDM400P not answering or making calls   Tom Playford
• [asterisk-users] TDM400P not answering or making calls   Tom Playford
• [asterisk-users] TE405P intermittent yellow alarm   Don Pobanz
• [asterisk-users] phone as control interface (was 99 bottles of beer)   Jon Pounder
• [asterisk-users] Overhead paging over IP   Jon Pounder
• [asterisk-users] Flash IDE   Jon Pounder
• [asterisk-users] Linux-HA and Asterisk   Jon Pounder
• [asterisk-users] Flash IDE   Jon Pounder
• [asterisk-users] Interesting Conference Request - Can this be done ?   Jon Pounder
• [asterisk-users] HITBSecConf2007 - Malaysia Materials & Photos are up !   Praburaajan
• [asterisk-users] sip:EXTEN;phone-context in asterisk dial plan   Prathapkumar S - TLS , Chennai
• [asterisk-users] Cepstral's Allison is having troublespeaking clearly   Steve Prior
• [asterisk-users] Fwd: Bad FCS error   Jan Prunk
• [asterisk-users] Asterisk voice quality tuning   Joe Pukepail
• [asterisk-users] HOWTO/FAQ question (Location: Sweden)   Mark Quitoriano
• [asterisk-users] HOWTO/FAQ question (Location: Sweden)   Mark Quitoriano
• [asterisk-users] asterisk audits   Mark Quitoriano
• [asterisk-users] asterisk audits   Mark Quitoriano
• [asterisk-users] meetme conference using g729?   Mark Quitoriano
• [asterisk-users] ATrpms/Fritz FCPCI CAPI/Fedora 7   Razza
• [asterisk-users] Which Asterisk version to use?   Razza
• [asterisk-users] Cepstral's Allison is having trouble speaking clearly   Todd Reese
• [asterisk-users] Cepstral's Allison is having troublespeaking clearly   Todd Reese
• [asterisk-users] Cepstral's Allison is having troublespeakingclearly   Todd Reese
• [asterisk-users] Asterisk 1.4 and Cepstral   Todd Reese
• [asterisk-users] Overhead paging over IP...   Bruce Reeves
• [asterisk-users] how to DUNDi branch office with area code?   Bruce Reeves
• [asterisk-users] ANNOUNCEMENT: Asterisk-Java 0.3.1 released   Stefan Reuter
• [asterisk-users] Generating an old-fashioned dialtone   Phil Reynolds
• [asterisk-users] Generating an old-fashioned dialtone   Phil Reynolds
• [asterisk-users] Generating an old-fashioned dialtone   Phil Reynolds
• [asterisk-users] Generating an old-fashioned dialtone   Phil Reynolds
• [asterisk-users] stanaphone issues. can someone verify my config?   Richard
• [asterisk-users] Music On Hold - How to increase volume ?   Richard
• [asterisk-users] Overhead paging over IP   JR Richardson
• [asterisk-users] Build your own "appliance" concept   JR Richardson
• [asterisk-users] SIPAddHeader cmd from Realtime MySQL, not getting all the 'appdata' field   JR Richardson
• [asterisk-users] Astricon Ride From Airport to Conf Hotel   JR Richardson
• [asterisk-users] Anyone use the Linksys phones?   JR Richardson
• [asterisk-users] Zaptel modules are being installed in different directory   Matt Riddell
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Matt Riddell
• [asterisk-users] Manager connection timeout   Matt Riddell
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Matt Riddell
• [asterisk-users] Asterisk Manager API - Originate command   Matt Riddell
• [asterisk-users] HA - How to detect software failure?   Matt Riddell
• [asterisk-users] Conference bridge.   Matt Riddell
• [asterisk-users] Agent Callback Login in 1.4   Matt Riddell
• [asterisk-users] Agent Callback Login in 1.4   Matt Riddell
• [asterisk-users] Agent Callback Login in 1.4   Matt Riddell
• [asterisk-users] TCP connection to AMI broken after 15 minutes   Matt Riddell
• [asterisk-users] TDM400P   Matt Riddell
• [asterisk-users] Wondering why I can't post   Matt Riddell
• [asterisk-users] Astricon 2007 -- does anybody need a ride?   Matt Riddell
• [asterisk-users] Why does everyone seem to dislike *now?   Matt Riddell
• [asterisk-users] Useful GUI? [Was: Why does everyone seem to dislike *now?]   Matt Riddell
• [asterisk-users] Useful GUI? [Was: Why does everyone seem to dislike *now?]   Matt Riddell
• [asterisk-users] Comfort noice sample (gsm/mp3)   Matt Riddell
• [asterisk-users] mISDN NPI setting with b410p   Matt Riddell
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Justin Ridge
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Justin Ridge
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Justin Ridge
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Justin Ridge
• [asterisk-users] Partitioning DSL input   Alex Robar
• [asterisk-users] Linux limits   Alex Robar
• [asterisk-users] Polycom 501 won't take new bootrom.ld or sip.ld...   Anthony Rodgers
• [asterisk-users] Rechazo de llamada en triangulacion de asterisk.   Guillermo Rodriguez
• [asterisk-users] Rechazo de llamada en triangulacion de asterisk.   Guillermo Rodriguez
• [asterisk-users] 31 seconds because it is directly bridged to another RTP stream   Guillermo Rodriguez
• [asterisk-users] TDM400P not answering or making calls   Carlos Rojas
• [asterisk-users] Asterisk with multi-line appearence? How?   Lucian Romi
• [asterisk-users] Sangoma or digium ?   Leon de Rooij
• [asterisk-users] Sangoma or digium ?   Leon de Rooij
• [asterisk-users] Queue serializes call delivery ?   Matthew J. Roth
• [asterisk-users] How to "busy out" zap channels   Brian Roy
• [asterisk-users] Asterisk on Fedora Core 4   Martin Roy
• [asterisk-users] Asterisk on Fedora Core 4   Martin Roy
• [asterisk-users] Cisco 79xx XML Apps (was: Re: Cisco Directory Format)   Matthew Rubenstein
• [asterisk-users] DECT SIP phones   Matthew Rubenstein
• [asterisk-users] IAX Java Softphone?   Matthew Rubenstein
• [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file   Alessandro Russo
• [asterisk-users] Asterisk + LDAP or RADIUS   Alessandro Russo
• [asterisk-users] Support of simple E1 CAS signaling (MFCR-like)   Lukas Rypl
• [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file   Abhishek M S
• [asterisk-users] How to handle "+" prefix   SIP
• [asterisk-users] SIPBroker vs SIPgate   SIP
• [asterisk-users] SIPBroker vs SIPgate   SIP
• [asterisk-users] [Serusers] user meeting (beer drinking in Vienna)   SIP
• [asterisk-users] Why does everyone seem to dislike *now?   SIP
• [asterisk-users] Why does everyone seem to dislike *now?   SIP
• [asterisk-users] what is softswitch   SIP
• [asterisk-users] Asterisk Redundancy   SIP
• [asterisk-users] asterisk-users Digest, Vol 38, Issue 83   RENZZO SOTOMAYOR
• [asterisk-users] Asterisk realtime error   RENZZO SOTOMAYOR
• [asterisk-users] Asterisk realtime error   RENZZO SOTOMAYOR
• [asterisk-users] Asterisk realtime error   RENZZO SOTOMAYOR
• [asterisk-users] Flash IDE   Juan Sandro
• [asterisk-users] Flash IDE   Juan Sandro
• [asterisk-users] Flash IDE   Juan Sandro
• [asterisk-users] Flash IDE   Juan Sandro
• [asterisk-users] off-topic: Avaya 46xx, release 032207 ... help   Cesc Santa
• [asterisk-users] Cisco 7940G licensing with asterisk   Sasa
• [asterisk-users] Cisco 7940G licensing with asterisk   Sasa
• [asterisk-users] Partitioning DSL input   C. Savinovich
• [asterisk-users] Partitioning DSL input   C. Savinovich
• [asterisk-users] Announcing: Click-to-Call with VIDEO   C. Savinovich
• [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO   C. Savinovich
• [asterisk-users] Announcing: Click-to-Call with VIDEO   C. Savinovich
• [asterisk-users] Announcing: Click-to-Call with VIDEO   C. Savinovich
• [asterisk-users] [PHISH] Re: ***SPAM*** Announcing: Click-to-Call with VIDEO   C. Savinovich
• [asterisk-users] bridge on DIVA card and how to see it   Armin Schindler
• [asterisk-users] bridge on DIVA card and how to see it   Armin Schindler
• [asterisk-users] bridge on DIVA card and how to see it   Armin Schindler
• [asterisk-users] bridge on DIVA card and how to see it   Armin Schindler
• [asterisk-users] Grandstream GXP2020 / 2000   Ben Schorr
• [asterisk-users] Asterisk and OCS integration   Jon Schøpzinsky
• [asterisk-users] Flash IDE   Bill Seddon
• [asterisk-users] (no subject)   Niki Selken
• [asterisk-users] Looking for Asterisk Consultant in San Franicsco   Niki Selken
• [asterisk-users] ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ...   Vidura Senadeera
• [asterisk-users] 14. Re: ztcfg error : TE110p error with " CAS signalling on span1 conflicts with HDLC with ... (Carlos Chavez)   Vidura Senadeera
• [asterisk-users] Asterisk Dropping Calls (Richard Young)   Vidura Senadeera
• [asterisk-users] asterisk audits   Gonzalo Servat
• [asterisk-users] Testing Framework   Kyle Sexton
• [asterisk-users] Queue agents w/ DUNDi   Kyle Sexton
• [asterisk-users] Benchmark   Seysan
• [asterisk-users] Show Callee name on Display   Seysan
• [asterisk-users] Show Callee name on Display   Seysan
• [asterisk-users] Show Callee name on Display   Seysan
• [asterisk-users] AsteriskNOW   Seysan
• [asterisk-users] 3 way Calling   Seysan
• [asterisk-users] how to route outgoing calls on IP-level   Seysan
• [asterisk-users] how to route outgoing calls on IP-level   Seysan
• [asterisk-users] Issue with Asterisk realtime   Mohammad Shokuie
• [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0   Shonga_Kerz
• [asterisk-users] help with channelbank audiocodes MP-124   Leonardo Silva
• [asterisk-users] app_conference   Moises Silva
• [asterisk-users] Manager Originate without phone off hook?   Moises Silva
• [asterisk-users] Asterisk Died message   Moises Silva
• [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault   Moises Silva
• [asterisk-users] A102d sangoma's card and ztdummy   Jaswinder Singh
• [asterisk-users] Configure extension by software   Jaswinder Singh
• [asterisk-users] Difference in show channels   Jaswinder Singh
• [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?   Jaswinder Singh
• [asterisk-users] errors messages in asterisk CLI   Jaswinder Singh
• [asterisk-users] prepaid application recommendation   Jaswinder Singh
• [asterisk-users] SIP and Firewall   Jaswinder Singh
• [asterisk-users] Multiple Home system with SIP   Jaswinder Singh
• [asterisk-users] echo problems   Jaswinder Singh
• [asterisk-users] Issue with calling queues   Joshua Small
• [asterisk-users] Problems with two trunks   Joshua Small
• [asterisk-users] FW: Problems with two trunks   Joshua Small
• [asterisk-users] FW: Problems with two trunks   Joshua Small
• [asterisk-users] FW: Problems with two trunks   Joshua Small
• [asterisk-users] Ping   Sander Smeenk
• [asterisk-users] Ping   Sander Smeenk
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Sander Smeenk
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Sander Smeenk
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Sander Smeenk
• [asterisk-users] Cascading queues & calls not joining unavailable queues.   Sander Smeenk
• [asterisk-users] Ping   Jared Smith
• [asterisk-users] No Dial tone came from fxs modules   Jared Smith
• [asterisk-users] SIP Debugging to separate log file   Jared Smith
• [asterisk-users] Digium Innovation Awards   Jared Smith
• [asterisk-users] Asterisk on NGINX Server?   Jared Smith
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Jared Smith
• [asterisk-users] Prompt for extension with standard dial-tone.   Jared Smith
• [asterisk-users] Spam filter woes [was: Astricon 2007 -- does anybody need a ride?]   Jared Smith
• [asterisk-users] asterisk crash and core dump   Jared Smith
• [asterisk-users] Limiting Simultaneous calls   Jared Smith
• [asterisk-users] Supermicro PDSME+ and TE110P   Jared Smith
• [asterisk-users] Short Audio Drop Out During Calls   Jared Smith
• [asterisk-users] Building an RPM from Asterisk 1.4   Jared Smith
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Jared Smith
• [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards   Jared Smith
• [asterisk-users] Polycom 501 Phones Rebooting   Jared Smith
• [asterisk-users] PCI: UNABLE TO HANDLE 64-bit ADDRESS SPACE FOR   Jared Smith
• [asterisk-users] Anyone else having problems with the list   Jared Smith
• [asterisk-users] Help with Sip Registration   Jared Smith
• [asterisk-users] Do I need to run #modprobe zaptel for Digium   Jared Smith
• [asterisk-users] Asterisk Redundancy   Jared Smith
• [asterisk-users] Digium acquires Switchvox   Jared Smith
• [asterisk-users] Digium acquires Switchvox   Jared Smith
• [asterisk-users] Queue Question   Kevin Smith
• [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO   Martin Smith
• [asterisk-users] Agent Callback Login in 1.4   Alexandre Snarskii
• [asterisk-users] Newcomer Question   Guenther Sohler
• [asterisk-users] Newcomer Question   Guenther Sohler
• [asterisk-users] Newcomer Question   Guenther Sohler
• [asterisk-users] Newcomer Question   Guenther Sohler
• [asterisk-users] SIP and Firewall   Guenther Sohler
• [asterisk-users] SIP and Firewall   Guenther Sohler
• [asterisk-users] Completing my Configuration   Guenther Sohler
• [asterisk-users] Completing my Configuration   Guenther Sohler
• [asterisk-users] Anyone else having problems with the list   Guenther Sohler
• [asterisk-users] Dead SIP channels   Andrey Solovjov
• [asterisk-users] Asterisk on NGINX Server?   Dominic Son
• [asterisk-users] 56k modem configuration   Andrea Spadaccini
• [asterisk-users] Failover SIP logic   Andrea Spadaccini
• [asterisk-users] 56k modem configuration   Andrea Spadaccini
• [asterisk-users] error messages related to mysql in asterisk CLI   Andrea Spadaccini
• [asterisk-users] Agent Callback Login in 1.4   Ryan Stark
• [asterisk-users] IAX Java Softphone?   Steven
• [asterisk-users] IAX Java Softphone?   Steven
• [asterisk-users] faster timeout in ENUMLOOKUP() function   James R. Stevens
• [asterisk-users] unsuscribe   Moshe at Talk'n'Save
• [asterisk-users] Asterisk - Spandsp Fax not working?   Jonn R Taylor
• [asterisk-users] Asterisk - Spandsp Fax not working?   Jonn R Taylor
• [asterisk-users] rtptimeout on Asterisk 1.4.x   Rodrigo P. Telles
• [asterisk-users] Asterisk on UML (User Mode Linux)   Simon Tennant
• [asterisk-users] Asterisk 1.2.24 simultaneous call limits.   James Texter
• [asterisk-users] Filesharing + video + voice supported Soft phone   Goltsios Theodore
• [asterisk-users] Asterisk 1.4, h.323, OpenCom 1010   Stefan Tichy
• [asterisk-users] Non-USASCII chars in sip.conf?   Stefan Tichy
• [asterisk-users] How to "busy out" zap channels   Tomás Laureano Peralta Tormey
• [asterisk-users] How to "busy out" zap channels   Tomás Laureano Peralta Tormey
• [asterisk-users] Short Audio Drop Out During Calls   Brent Torrenga
• [asterisk-users] E1 to Ethernet Bridge   Steve Totaro
• [asterisk-users] Zaptel modules are being installed in different directory   Steve Totaro
• [asterisk-users] OT: DELL Platforms   Steve Totaro
• [asterisk-users] Overhead paging over IP...   Steve Totaro
• [asterisk-users] E1 Line Tapping   Steve Totaro
• [asterisk-users] asterisk voicemail to email and relaying   Steve Totaro
• [asterisk-users] Choppy sound while converting alaw to ulaw   Steve Totaro
• [asterisk-users] Multitenant or Multiple virtual machines   Steve Totaro
• [asterisk-users] New Installed X100p   Steve Totaro
• [asterisk-users] T1 to SIP conversion, standalone device   Steve Totaro
• [asterisk-users] New Installed X100p   Steve Totaro
• [asterisk-users] Channels in use?   Steve Totaro
• [asterisk-users] T1 to SIP conversion, standalone device   Steve Totaro
• [asterisk-users] New Installed X100p   Steve Totaro
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Steve Totaro
• [asterisk-users] Partitioning DSL input   Steve Totaro
• [asterisk-users] E1 Line Tapping   Steve Totaro
• [asterisk-users] E1 Line Tapping   Steve Totaro
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   Steve Totaro
• [asterisk-users] New Installed X100p   Steve Totaro
• [asterisk-users] TE405P intermittent yellow alarm   Steve Totaro
• [asterisk-users] Digium Appliance   Steve Totaro
• [asterisk-users] CallWithUs Service?   Steve Totaro
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   Steve Totaro
• [asterisk-users] Astricon 2007 -- does anybody need a ride?   Steve Totaro
• [asterisk-users] (Getting OT) Re: Call Center SoftPhone with Auto Answer   Steve Totaro
• [asterisk-users] Interesting Conference Request - Can this be done ?   Steve Totaro
• [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO   Steve Totaro
• [asterisk-users] Sangoma or digium ?   Steve Totaro
• [asterisk-users] Backuping VoIP provider with PRI   Steve Totaro
• [asterisk-users] . (period): Wildcard match; matches one or more characters   Steve Totaro
• [asterisk-users] How can I know if I wrote the configuration like correctly   Steve Totaro
• [asterisk-users] How to "busy out" zap channels   Steve Totaro
• [asterisk-users] IAX gsm bandwith calls   Steve Totaro
• [asterisk-users] Proximity Detection: Motorola Q + Bluetooth + Asterisk   Steve Totaro
• [asterisk-users] VoIP, Asterisk, What Do you really need ?   Steve Totaro
• [asterisk-users] TDM400P periodic sound clicks on FXS   Costa Tsaousis
• [asterisk-users] TDM400P periodic sound clicks on FXS   Costa Tsaousis
• [asterisk-users] TDM400P periodic sound clicks on FXS   Costa Tsaousis
• [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?   Duncan Turnbull
• [asterisk-users] Randomly half-voice at sip/zap   Péter Tóth
• [asterisk-users] Randomly half-voice at sip/zap   Péter Tóth
• [asterisk-users] Randomly half-voice at sip/zap   Péter Tóth
• [asterisk-users] Horrible problem - calls losing sound   Péter Tóth
• [asterisk-users] Horrible problem - calls losing sound   Péter Tóth
• [asterisk-users] G.722: ast_channel_make_compatible failure   Ondrej Valousek
• [asterisk-users] Testing Framework   Hariharan Veerappan
• [asterisk-users] Udev issue on zaptel install   Hariharan Veerappan
• [asterisk-users] Anyone use the Linksys phones?   Karl J. Vesterling
• [asterisk-users] Changing contexts "on the fly"   Ade Vickers
• [asterisk-users] queue static agents   Vieri
• [asterisk-users] queue static agents   Vieri
• [asterisk-users] how to determine if a SIP extension has DND on or off   Vieri
• [asterisk-users] how to determine if a SIP extension has DND on oroff   Vieri
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Vieri
• [asterisk-users] how to determine if a SIP extension has DND onoroff   Vieri
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Vieri
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Vieri
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Vieri
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Vieri
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Vieri
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Vieri
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Vieri
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Vieri
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Vieri
• [asterisk-users] asterisk crash and core dump   Vieri
• [asterisk-users] asterisk crash and core dump: format_mp3.so   Vieri
• [asterisk-users] asterisk crash and core dump: format_mp3.so   Vieri
• [asterisk-users] call limit   Vieri
• [asterisk-users] asterisk cli - vi keybindings ?   Ex Vito
• [asterisk-users] asterisk cli - vi keybindings ?   Ex Vito
• [asterisk-users] Flash IDE   Ed W
• [asterisk-users] Force a new user to configure Comedian mail?   Jeremy Wadhams
• [asterisk-users] Force a new user to configure Comedian mail?   Jeremy Wadhams
• [asterisk-users] ztdummy problem in fedora7, kernel 2.6.22.5-76.fc7   Dave Walker
• [asterisk-users] Asterisk cli   Dave Walker
• [asterisk-users] DECT SIP phones   Dave Walker
• [asterisk-users] Anyone use the Linksys phones?   Dave Walker
• [asterisk-users] Asterisk Redundancy   Dave Walker
• [asterisk-users] Asterisk Redundancy   Dave Walker
• [asterisk-users] Ghost calls from phones   Erik Wartusch
• [asterisk-users] Grandstream GXP2020 / 2000   Erik Wartusch
• [asterisk-users] Busy problem   Erik Wartusch
• [asterisk-users] Agent Callback Login in 1.4   Watkins, Bradley
• [asterisk-users] g729 on 1.4.10.1   Matt Watson
• [asterisk-users] Music On Hold   Wayne
• [asterisk-users] Music On Hold   Wayne
• [asterisk-users] Anyone use the Linksys phones?   Robert Webb
• [asterisk-users] Zap channels: no sound with certain call paths   Christian Weeks
• [asterisk-users] 3-way calling   Pamela Weis
• [asterisk-users] No Sound on Zap Channels   Jon Weisman
• [asterisk-users] Trunk & Outbound Route for a Cisco VOIP router?   Jon Weisman
• [asterisk-users] No Sound on Zap Channels   Jon Weisman
• [asterisk-users] Digium acquires Switchvox   Aubrey Wells
• [asterisk-users] chan_misdn   Ron Wellsted
• [asterisk-users] Asterisk on Ubuntu Feisty   Ron Wellsted
• [asterisk-users] Asterisk on Ubuntu Feisty   Ron Wellsted
• [asterisk-users] Asterisk on Ubuntu Feisty   Ron Wellsted
• [asterisk-users] Musiconhold instead ringing   Karsten Wemheuer
• [asterisk-users] Asterisk w/MS SQL Server 2005   BJ Weschke
• [asterisk-users] Polycom 501 Phones Rebooting   BJ Weschke
• [asterisk-users] Cepstral's Allison is having trouble speaking clearly   Brian West
• [asterisk-users] FW: Account Registration Failed   Brian West
• [asterisk-users] Asterisk Died message   Brian West
• [asterisk-users] remove unnecessary text (was: Re: Can asterisk give half-ring periodically for MWI?)   Brian West
• [asterisk-users] VoIP Provider for business   Brian West
• [asterisk-users] what is softswitch   Brian West
• [asterisk-users] Freeswitch Vs Asterisk   Brian West
• [asterisk-users] what is softswitch   Brian West
• [asterisk-users] what is softswitch   Brian West
• [asterisk-users] . (period): Wildcard match; matches one or more characters   Brian West
• [asterisk-users] Zaptel modules are being installed in different directory   Eric "ManxPower" Wieling
• [asterisk-users] Zaptel modules are being installed in different directory   Eric "ManxPower" Wieling
• [asterisk-users] off-hook warning tone   Eric "ManxPower" Wieling
• [asterisk-users] Can asterisk give half-ring periodically for MWI?   Eric "ManxPower" Wieling
• [asterisk-users] Show Callee name on Display   Eric "ManxPower" Wieling
• [asterisk-users] Show Callee name on Display   Eric "ManxPower" Wieling
• [asterisk-users] Prevent multiple sip registrations   Eric "ManxPower" Wieling
• [asterisk-users] Linux-HA and Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Linux-HA and Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Linux-HA and Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Fwd: Bad FCS error   Eric "ManxPower" Wieling
• [asterisk-users] DTMF error on asterisk   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Eric "ManxPower" Wieling
• [asterisk-users] Caller ID on Channelized T1 (E&M Wink)   Eric "ManxPower" Wieling
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Eric "ManxPower" Wieling
• [asterisk-users] how to determine if a SIP extension has DNDonoroff   Eric "ManxPower" Wieling
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Eric "ManxPower" Wieling
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   Eric "ManxPower" Wieling
• [asterisk-users] Force a new user to configure Comedian mail?   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX   Eric "ManxPower" Wieling
• [asterisk-users] RTP Call Disconnect   Eric "ManxPower" Wieling
• [asterisk-users] Why does everyone seem to dislike *now?   Eric "ManxPower" Wieling
• [asterisk-users] sip.conf best practices?   Eric "ManxPower" Wieling
• [asterisk-users] GROUP() issues for me   Eric "ManxPower" Wieling
• [asterisk-users] CallerID problem Asterisk 1.4.2   Eric "ManxPower" Wieling
• [asterisk-users] ChanSpy issue   Eric "ManxPower" Wieling
• [asterisk-users] problem   Walter Willis
• [asterisk-users] Asterisk 1.4.11 over NAT   Walter Willis
• [asterisk-users] Flash IDE   Hans Witvliet
• [asterisk-users] 56k modem configuration   Tobias Wolf
• [asterisk-users] DECT SIP phones   Tobias Wolf
• [asterisk-users] Asterisk 1.2.13 and presence   Scott Wolfe
• [asterisk-users] Asterisk 1.2.13 and presence   Scott Wolfe
• [asterisk-users] Caller ID on Channelized T1 (E&M Wink)   Willy Wouters
• [asterisk-users] ADIT 600 & CMG <=> Asterisk question   Darren Wright
• [asterisk-users] Asterisk Manager API - Originate command   Wai Wu
• [asterisk-users] Asterisk Manager API - Originate command   Wai Wu
• [asterisk-users] Asterisk Manager API - Originate command   Wai Wu
• [asterisk-users] Asterisk Manager API - Originate command   Wai Wu
• [asterisk-users] canreinvite   Wai Wu
• [asterisk-users] TCP connection to AMI broken after 15 minutes   Wai Wu
• [asterisk-users] Linux limits   Wai Wu
• [asterisk-users] Linux limits   Wai Wu
• [asterisk-users] Linux limits   Wai Wu
• [asterisk-users] Asterisk 1.2.24 simultaneous call limits.   Wai Wu
• [asterisk-users] Asterisk 1.2.24 simultaneous call limits.   Wai Wu
• [asterisk-users] Asterisk 1.2.24 simultaneous call limits.   Wai Wu
• [asterisk-users] Asterisk 1.2.24 simultaneous call limits.   Wai Wu
• [asterisk-users] Ast_log   Wai Wu
• [asterisk-users] How to "busy out" zap channels   Wai Wu
• [asterisk-users] Networking Question   Wai Wu
• [asterisk-users] Ast_log   Wai Wu
• [asterisk-users] ChanSpy issue   Wai Wu
• [asterisk-users] ChanSpy issue   Wai Wu
• [asterisk-users] Problems Connecting Two Asterisk Installs ViaISDN PRI Cards   Wai Wu
• [asterisk-users] bug in 1.2.24   Isaac Xiao
• [asterisk-users] bug in 1.2.24   Isaac Xiao
• [asterisk-users] Cepstral's Allison is having trouble speaking clearly   Vahan Yerkanian
• [asterisk-users] What's the deal with ATAcomm?   Vahan Yerkanian
• [asterisk-users] Asterisk Dropping Calls   Richard Young
• [asterisk-users] Show Callee name on Display   Shane Young
• [asterisk-users] Show Callee name on Display   Shane Young
• [asterisk-users] Show Callee name on Display   Shane Young
• [asterisk-users] Zaptel-1.4.5.1 Compile Error   Jeng Yu
• [asterisk-users] Zaptel-1.4.5.1 Compile Error   Jeng Yu
• [asterisk-users] Zaptel-1.4.5.1 Compile Error   Jeng Yu
• [asterisk-users] Zaptel-1.4.5.1 Compile Error   Jeng Yu
• [asterisk-users] Failover SIP logic   Yusuf
• [asterisk-users] Sangoma or digium ?   Zeeshan Zakaria
• [asterisk-users] Anyone use the Linksys phones?   Zeeshan Zakaria
• [asterisk-users] Adding allowed codecs to Asterisk   Matti Zemack
• [asterisk-users] Call Center SoftPhone with Auto Answer   Zoa
• [asterisk-users] IAX Java Softphone?   Zoa
• [asterisk-users] Help Drop Calls   paul aldee
• [asterisk-users] # to transfer calls   bails
• [asterisk-users] off-topic: Avaya 46xx, release 032207 ... help   robert boardman
• [asterisk-users] Asterisk - Spandsp Fax not working?   marco britannio
• [asterisk-users] Asterisk - Spandsp Fax not working?   marco britannio
• [asterisk-users] chan_sip falls over with undefined symbol ast_pickup_ext   stephen.hindmarch at bt.com
• [asterisk-users] chan_sip falls over with undefined symbolast_pickup_ext   stephen.hindmarch at bt.com
• [asterisk-users] ztdummy problem in fedora7, kernel 2.6.22.5-76.fc7   saqib butt
• [asterisk-users] Testing Framework   dave cantera
• [asterisk-users] Grandstream GXP2020 / 2000   cb
• [asterisk-users] Setting Callerid with chan_misdn   jan at codejunky.org
• [asterisk-users] ChanSpy issue   John covici
• [asterisk-users] ChanSpy issue   John covici
• [asterisk-users] Hfcmulti and B410P Digium Card   voip crazy
• [asterisk-users] Hfcmulti and B410P Digium Card   voip crazy
• [asterisk-users] Virtual server Solution   voip crazy
• [asterisk-users] Virtual server Solution   voip crazy
• [asterisk-users] Zaptel modules are being installed in different directory   wassim darwish
• [asterisk-users] Zaptel modules are being installed in different directory   wassim darwish
• [asterisk-users] Zaptel modules are being installed in different directory   wassim darwish
• [asterisk-users] Zaptel modules are being installed in different directory   wassim darwish
• [asterisk-users] Zaptel modules are being installed in different directory   wassim darwish
• [asterisk-users] No Dial tone came from fxs modules   wassim darwish
• [asterisk-users] No Dial tone came from fxs modules   wassim darwish
• [asterisk-users] Musiconhold instead ringing   wassim darwish
• [asterisk-users] Musiconhold instead ringing   wassim darwish
• [asterisk-users] Musiconhold instead ringing   wassim darwish
• [asterisk-users] astcc sometimes doesnt write on mysql   wassim darwish
• [asterisk-users] Astcc does'nt count all calls   wassim darwish
• [asterisk-users] SIP over TCP   dadsadsadf dsadasdsa
• [asterisk-users] Asterisk and OCS integration   dadsadsadf dsadasdsa
• [asterisk-users] Asterisk and OCS integration   dadsadsadf dsadasdsa
• [asterisk-users] Asterisk and OCS integration   dadsadsadf dsadasdsa
• [asterisk-users] Hola Jonathan, a ver si tre suena...   dadsadsadf dsadasdsa
• [asterisk-users] A102d sangoma's card and ztdummy   fateme fatah
• [asterisk-users] asterisk 1.2 or 1.4 for conference call service   fateme fatah
• [asterisk-users] unnumbered priorities   fateme fatah
• [asterisk-users] enter menu   fateme fatah
• [asterisk-users] extensions for conference call   fateme fatah
• [asterisk-users] configuration of wanpipe for asterisk.   fateme fatah
• [asterisk-users] Chan_sip Entry   Kutman.DK at forces.gc.ca
• [asterisk-users] Chan_sip Entry   Kutman.DK at forces.gc.ca
• [asterisk-users] Softphone RTP Session Start-up Delay   Kutman.DK at forces.gc.ca
• [asterisk-users] Softphone RTP Session Start-up Delay   Kutman.DK at forces.gc.ca
• [asterisk-users] Softphone RTP Session Start-up Delay   Kutman.DK at forces.gc.ca
• [asterisk-users] Timeout issues   Kutman.DK at forces.gc.ca
• [asterisk-users] Multiple Meet me conferences   Kutman.DK at forces.gc.ca
• [asterisk-users] SIP Debugging to separate log file   bilal ghayyad
• [asterisk-users] special kind of billing   bilal ghayyad
• [asterisk-users] SIP Debugging to separate log file   bilal ghayyad
• [asterisk-users] What is the difference between increasing the verbose level and the debug level?   bilal ghayyad
• [asterisk-users] nat=yes   bilal ghayyad
• [asterisk-users] canreinvite   bilal ghayyad
• [asterisk-users] SIP endpoint that does not register   bilal ghayyad
• [asterisk-users] nat=yes   bilal ghayyad
• [asterisk-users] canreinvite   bilal ghayyad
• [asterisk-users] stop log/debug messages into /var/log/messages   bilal ghayyad
• [asterisk-users] stop log/debug messages into /var/log/messages   bilal ghayyad
• [asterisk-users] stop log/debug messages into /var/log/messages   bilal ghayyad
• [asterisk-users] stop log/debug messages into /var/log/messages   bilal ghayyad
• [asterisk-users] Do I need to run #modprobe zaptel for Digium   bilal ghayyad
• [asterisk-users] Do I need to run #modprobe zaptel for Digium   bilal ghayyad
• [asterisk-users] How can I know if I wrote the configuration like correctly   bilal ghayyad
• [asterisk-users] . (period): Wildcard match; matches one or more characters   bilal ghayyad
• [asterisk-users] Selecting a specific line from Zap/g   bilal ghayyad
• [asterisk-users] TDM2400P: Power alarm error on boot   gincantalupo
• [asterisk-users] TDM2400P: Power alarm error on boot   gincantalupo
• [asterisk-users] how to disable wctdm auto modprobe during boot   gincantalupo
• [asterisk-users] DTMF error on asterisk   gincantalupo
• [asterisk-users] how to disable wctdm auto modprobe during boot   gincantalupo
• [asterisk-users] how to disable wctdm auto modprobe during boot   gincantalupo
• [asterisk-users] how to disable wctdm auto modprobe during boot   gincantalupo
• [asterisk-users] analog topex and Digium cards (maybe OT??)   gincantalupo
• [asterisk-users] SIP over TCP   jonathangf at gmail.com
• [asterisk-users] exit ChanSpy with DTMF   GDrayer at guesswho.com
• [asterisk-users] Multitenant or Multiple virtual machines   lists at infoway.net
• [asterisk-users] Multitenant or Multiple virtual machines   lists at infoway.net
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting - This Saturday Sep 8th, 2007 (Only hours away)   asterisk_help at iwishi.nu
• [asterisk-users] Supermicro PDSME+ and TE110P   kido
• [asterisk-users] Supermicro PDSME+ and TE110P   kido
• [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828   kido
• [asterisk-users] Account Registration Failed   neoh kumyee
• [asterisk-users] FW: Account Registration Failed   neoh kumyee
• [asterisk-users] FW: Account Registration Failed   neoh kumyee
• [asterisk-users] How to make call from asterisk?   neoh kumyee
• [asterisk-users] How to make call from asterisk?   neoh kumyee
• [asterisk-users] How to make call from asterisk?   neoh kumyee
• [asterisk-users] bridge on DIVA card and how to see it   lemmel
• [asterisk-users] fax machine detection for outgoing call on DIVA card   lemmel
• [asterisk-users] bridge on DIVA card and how to see it   lemmel lemmel
• [asterisk-users] bridge on DIVA card and how to see it   lemmel lemmel
• [asterisk-users] bridge on DIVA card and how to see it   lemmel lemmel
• [asterisk-users] bridge on DIVA card and how to see it   lemmel lemmel
• [asterisk-users] bridge on DIVA card and how to see it   lemmel lemmel
• [asterisk-users] fax and answer machine detection for outgoing call on DIVA card   lemmel lemmel
• [asterisk-users] Problems with Asterisk behind a firewall   lemmel lemmel
• [asterisk-users] [SOLVED] fax machine detection for outgoing call on DIVAcard   lemmel lemmel
• [asterisk-users] [SOLVED] fax machine detection for outgoingcall on DIVAcard   lemmel lemmel
• [asterisk-users] IAX2 trunking scalability   Al lists
• [asterisk-users] asterisk voicemail to email and relaying   Al lists
• [asterisk-users] Broken UDP streams   Al lists
• [asterisk-users] Which cause less CPU usage: GSM or wav??   Al lists
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   Al lists
• [asterisk-users] Cisco UC 500   Al lists
• [asterisk-users] Partitioning DSL input   Al lists
• [asterisk-users] DECT SIP phones   Al lists
• [asterisk-users] DECT SIP phones   Al lists
• [asterisk-users] DISA and DTMF detection problem w/ FXO port on a TDM400   Al lists
• [asterisk-users] CallWithUs Service?   Al lists
• [asterisk-users] CallWithUs Service?   Al lists
• [asterisk-users] stanaphone issues. can someone verify my config?   Al lists
• [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?   Al lists
• [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?   Al lists
• [asterisk-users] canreinvite   mail-lists
• [asterisk-users] asterisk cli - vi keybindings ?   mail-lists
• [asterisk-users] res_snmp   yonoko molomo
• [asterisk-users] res_snmp   yonoko molomo
• [asterisk-users] IAX configuration   yonoko molomo
• [asterisk-users] IAX configuration   yonoko molomo
• [asterisk-users] Cisco 7940G licensing with asterisk   yonoko molomo
• [asterisk-users] IAX configuration   yonoko molomo
• [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0   nik600
• [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0   nik600
• [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0   nik600
• [asterisk-users] [phpAGI] generate a call from a SIP device to Asterisk   nik600
• [asterisk-users] outgoing call restriction   satish patel
• [asterisk-users] FAX with asterisk   satish patel
• [asterisk-users] ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ...   satish patel
• [asterisk-users] FAX machine connect with audiocode SIP device   satish patel
• [asterisk-users] FAX machine connect with audiocode SIP device   satish patel
• [asterisk-users] redendent asterisk server for backup   satish patel
• [asterisk-users] online active call watching   satish patel
• [asterisk-users] DTMF error on asterisk   satish patel
• [asterisk-users] Asterisk voice quality tuning   satish patel
• [asterisk-users] Asterisk voice quality tuning   satish patel
• [asterisk-users] outgoing call restriction in extention.conf   satish patel
• [asterisk-users] Filesharing + video + voice supported Soft phone   satish patel
• [asterisk-users] asterisk directory dialing   satish patel
• [asterisk-users] Vedio confrancing with asterisk   satish patel
• [asterisk-users] what is softswitch   satish patel
• [asterisk-users] How many SIP phone connect with asterisk   satish patel
• [asterisk-users] Freeswitch Vs Asterisk   satish patel
• [asterisk-users] asterisk canreinvite option questions   satish patel
• [asterisk-users] Register Extension   phananhvu
• [asterisk-users] Register Extension   phananhvu
• [asterisk-users] Configure extension by software   phananhvu
• [asterisk-users] Configure extension by software   phananhvu
• [asterisk-users] asterisk 1.2 or 1.4 for conference call service   ram
• [asterisk-users] Difference in show channels   ram
• [asterisk-users] Sysmaster and Asterisk   ram
• [asterisk-users] SIP Debugging to separate log file   ram
• [asterisk-users] Difference in show channels   ram
• [asterisk-users] online active call watching   ram
• [asterisk-users] Asterisk Users Conference Friday @ 12:30PM EDT   randulo
• [asterisk-users] Asterisk Users Conference Friday @ 12:30PM EDT   randulo
• [asterisk-users] VOIP Users Conference Friday September 14 @ 12:30PM EDT   randulo
• [asterisk-users] Why does everyone seem to dislike *now?   randulo
• [asterisk-users] Problems sending more than 2 SMS with asterisk / smsq   randulo
• [asterisk-users] Anyone use the Linksys phones?   randulo
• [asterisk-users] Why does everyone seem to dislike *now?   randulo
• [asterisk-users] Conference call today at 12:30 PM EDT   randulo
• [asterisk-users] How to cancel the password check in VoicemailMain()   rrgv
• [asterisk-users] TE120p and music on hold   salvatore at open-sys.it
• [asterisk-users] OT: Samsung Sprint CDMAoIP   schmerold at eml.cc
• [asterisk-users] OT: Samsung Sprint CDMAoIP   schmerold at eml.cc
• [asterisk-users] Wanted: VoIP Engineer for Warsaw!   laurent schweizer
• [asterisk-users] Wanted: VoIP Engineer for Warsaw !   laurent schweizer
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   shadowym
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   shadowym
• [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?   shadowym
• [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)   shadowym
• [asterisk-users] DECT SIP phones   shadowym
• [asterisk-users] Why does everyone seem to dislike *now?   shadowym
• [asterisk-users] Supermicro PDSME+ and TE110P   shadowym
• [asterisk-users] TDM2400 answer detection   mccoy silva
• [asterisk-users] Big problems with TDM2400 :(   mccoy silva
• [asterisk-users] how to DUNDi branch office with area code?   d tbsky
• [asterisk-users] how to DUNDi branch office with area code?   d tbsky
• [asterisk-users] what is the usable feature in DUNDi?   d tbsky
• [asterisk-users] Can Asterisk match a literal "*" in extensions.conf   d tbsky
• [asterisk-users] about ChanSpy,thank you!!!   weizhao_hou
• [asterisk-users] HOWTO/FAQ question (Location: Sweden)   zoachien

Last message date: Sun Sep 30 23:45:02 CDT 2007
Archived on: Sun Sep 30 23:45:18 CDT 2007

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