May 2007 Archives by date

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Starting: Tue May 1 00:33:47 MST 2007
Ending: Thu May 31 23:46:14 MST 2007
Messages: 2590

• [asterisk-users] ADSL routers with integrated SIP QoS for otherdevices   Francisco Pérez Botella
• [asterisk-users] Improving Asterisk's DNS support   Per Jessen
• [asterisk-users] did we all get spammed by TechnoCo ?   Per Jessen
• [asterisk-users] headsets for linksys/sipura phones?   Wilson Pickett
• [asterisk-users] CDR and Billing Issue   Dovid B
• [asterisk-users] Two Context Residing On The Same Server   broadbandvoice at comcast.net
• [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?   broadbandvoice at comcast.net
• [asterisk-users] CDR and Billing Issue   Grey Man
• [asterisk-users] did we all get spammed by TechnoCo ?   Doug Lytle
• [asterisk-users] Autoattendant press 1 collides with extension numbers...   J. Oquendo
• [asterisk-users] chan_local   Rizwan Hisham
• [asterisk-users] Two Connected Servers Sound Quailty   Thomas Deillon
• [asterisk-users] restrictions on meetme with agi background   Jerry Geis
• [asterisk-users] Re: restrictions on meetme with agi background   Tony Mountifield
• [asterisk-users] My Sip Provider lacks Sip 2.0 183 (Ringing) information   Knud Müller
• [asterisk-users] Autoattendant press 1 collides with extension numbers...   C F
• [asterisk-users] Change Codec   Arun Kumar
• [asterisk-users] did we all get spammed by TechnoCo ?   Lee Jenkins
• [asterisk-users] RE: Voicemail on Different Server (MySQL Replication split thread)   JR Richardson
• [asterisk-users] did we all get spammed by TechnoCo ?   Tzafrir Cohen
• [asterisk-users] did we all get spammed by TechnoCo ?   Stephen Bosch
• [asterisk-users] TDM400P and TE405P   Nitesh Divecha
• [asterisk-users] Test   Wilson Pickett
• [asterisk-users] did we all get spammed by TechnoCo ?   Salvatore Giudice
• [asterisk-users] Autoattendant press 1 collides with extension numbers...   Stephen Bosch
• [asterisk-users] TDM400P and TE405P   Dave Miller
• [asterisk-users] Autoattendant press 1 collides with extension numbers...   J. Oquendo
• [asterisk-users] Autoattendant press 1 collides with extension numbers...   J. Oquendo
• [asterisk-users] TDM400P and TE405P   Nitesh Divecha
• [asterisk-users] Cisco 7940 no outgoing audio   Simon Alman
• [asterisk-users] Change Codec   Salvatore Giudice
• [asterisk-users] How many users can be supported simultaneously?   Antonopoulos Angelos
• [asterisk-users] Email to HP Product Suggestions - Seamless Transparent Fax Gateway   Rob Townley
• [asterisk-users] Delay in Dial()   Steve Finkelstein
• [asterisk-users] Calls in ulaw, not gsm as desired   Rob Schall
• [asterisk-users] Cisco 7940 no outgoing audio   Salvatore Giudice
• [asterisk-users] How many users can be supported simultaneously?   Dean Collins
• [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help   Bruce McAlister
• [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help   Bruce McAlister
• [asterisk-users] How many users can be supported simultaneously?   Salvatore Giudice
• [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help   Bruce McAlister
• [asterisk-users] How do I do this in Asterisk?   Christian
• [asterisk-users] Delay in Dial()   Luki
• [asterisk-users] How many users can be supported simultaneously?   Philipp Kempgen
• [asterisk-users] Calls in ulaw, not gsm as desired   Salvatore Giudice
• [asterisk-users] Delay in Dial()   Edoardo Serra
• [asterisk-users] Channel stuck with call pri flag   Andre Courchesne - Consultant
• [asterisk-users] is dundi worth pursuing in this situation?   Erik Anderson
• [asterisk-users] Calls in ulaw, not gsm as desired   Andreas Sikkema
• [asterisk-users] Delay in Dial()   Eric "ManxPower" Wieling
• [asterisk-users] OT: Capture Asterisk traffic   CSB
• [asterisk-users] TDM400P and TE405P   Nitesh Divecha
• [asterisk-users] Test   Dovid B
• [asterisk-users] Voicemail Creation   Dovid B
• [asterisk-users] Calls in ulaw, not gsm as desired   Rob Schall
• [asterisk-users] Channel stuck with call pri flag   John Treble
• [asterisk-users] How do I do this in Asterisk?   f6hqz-m at hamwlan.net
• [asterisk-users] Channel stuck with call pri flag   Andre Courchesne - Consultant
• [asterisk-users] is dundi worth pursuing in this situation?   Justin Hamade
• [asterisk-users] is dundi worth pursuing in this situation?   Bruce Reeves
• [asterisk-users] MYSQL application in dial plan   Yehavi Bourvine +972-8-9489444
• [asterisk-users] did we all get spammed by TechnoCo ?   Dovid B
• [asterisk-users] Calls in ulaw, not gsm as desired   Salvatore Giudice
• [asterisk-users] OT: Capture Asterisk traffic   Salvatore Giudice
• [asterisk-users] OT: Capture Asterisk traffic   Bruce Reeves
• [asterisk-users] Calls in ulaw, not gsm as desired   Ed Nuñez
• [asterisk-users] TDM400P and TE405P   Tzafrir Cohen
• [asterisk-users] My Sip Provider lacks Sip 2.0 183 (Ringing)information   Yuan LIU
• [asterisk-users] is dundi worth pursuing in this situation?   Salvatore Giudice
• [asterisk-users] is dundi worth pursuing in this situation?   Jonathan Creasy
• [asterisk-users] OT: Capture Asterisk traffic   Andres Paglayan
• [asterisk-users] OT: Capture Asterisk traffic   Stephen Bosch
• [asterisk-users] Delay in Dial()   Stephen Bosch
• [asterisk-users] T1 interface   Bill Michaelson
• [asterisk-users] MYSQL application in dial plan   Doug Garstang
• [asterisk-users] OT: Capture Asterisk traffic   Doug Garstang
• [asterisk-users] is dundi worth pursuing in this situation?   Erik Anderson
• [asterisk-users] Improving Asterisk's DNS support   Kai-Uwe Jensen
• [asterisk-users] Calls in ulaw, not gsm as desired   Rob Schall
• [asterisk-users] chan_sip seems to be hanging   Ken Williams
• [asterisk-users] Delay in Dial()   Eric "ManxPower" Wieling
• [asterisk-users] Stanaphone business ok?   Todd H
• [asterisk-users] Display Caller ID of called party   Savoy, Kevin - Williston, ND
• [asterisk-users] Re: [asterisk-dev] SRTP implementation   marek cervenka
• [asterisk-users] Delay in Dial()   Steve Finkelstein
• [asterisk-users] MYSQL application in dial plan   Remco Post
• [asterisk-users] T1 interface   Salvatore Giudice
• [asterisk-users] OT: Capture Asterisk traffic   Salvatore Giudice
• [asterisk-users] is dundi worth pursuing in this situation?   Remco Post
• [asterisk-users] Calls in ulaw, not gsm as desired   Salvatore Giudice
• [asterisk-users] Delay in Dial()   Remco Post
• [asterisk-users] Stanaphone business ok?   Salvatore Giudice
• [asterisk-users] Stanaphone business ok?   Dean Collins
• [asterisk-users] Display Caller ID of called party   Alex Balashov
• [asterisk-users] Display Caller ID of called party   James FitzGibbon
• [asterisk-users] Zaptel kernel module load order   C. Chad Wallace
• [asterisk-users] Stanaphone business ok?   bkruse
• [asterisk-users] Re: Wildcard TDM11B & Wildcard TDM04B   bilal ghayyad
• [asterisk-users] Re: is dundi worth pursuing in this situation?   JR Richardson
• [asterisk-users] Digital Phones   bilal ghayyad
• [asterisk-users] Applet?   Pablo L. Arturi
• [asterisk-users] TC400B   bilal ghayyad
• [asterisk-users] How many users can be supported simultaneously?   Antonopoulos Angelos
• [asterisk-users] Re: Wildcard TDM11B & Wildcard TDM04B   Yuan LIU
• [asterisk-users] Display Caller ID of called party   Dan Austin
• [asterisk-users] TC400B   Andres
• [asterisk-users] Digital Phones   Dean Collins
• [asterisk-users] Applet?   Dean Collins
• [asterisk-users] Improving Asterisk's DNS support   Kristian Kielhofner
• [asterisk-users] Applet?   Salvatore Giudice
• [asterisk-users] How many users can be supported simultaneously?   Salvatore Giudice
• [asterisk-users] Applet?   Dean Collins
• [asterisk-users] Digital Phones   Salvatore Giudice
• [asterisk-users] Applet?   Salvatore Giudice
• [asterisk-users] Applet?   Dean Collins
• [asterisk-users] Digital Phones   shadowym
• [asterisk-users] Digital Phones   C F
• [asterisk-users] Autoattendant press 1 collides with extension numbers...   C F
• [asterisk-users] Display Caller ID of called party   Shane Young
• [asterisk-users] Trixbox 2 and MFC/R2   Facundo Ameal
• [asterisk-users] using Playback() to play a random sound file   Jay Austad
• [asterisk-users] using Playback() to play a random sound file   Steve Edwards
• [asterisk-users] using Playback() to play a random sound file   Yuan LIU
• [asterisk-users] using Playback() to play a random sound file   Steve Edwards
• [asterisk-users] Applet?   Stephen Bosch
• [asterisk-users] Improving Asterisk's DNS support   Brian Capouch
• [asterisk-users] Digital Phones   Stephen Bosch
• [asterisk-users] Applet?   Remco Post
• [asterisk-users] Applet?   Dean Collins
• [asterisk-users] Improving Asterisk's DNS support   Joseph
• [asterisk-users] Zaptel kernel module load order   Tzafrir Cohen
• [asterisk-users] using Playback() to play a random sound file   Yuan LIU
• [asterisk-users] Runaway MOH/mp3123 process?   Alex Balashov
• [asterisk-users] Change Codec   Arun Kumar
• [asterisk-users] Cisco 7940 no outgoing audio   Simon Alman
• [asterisk-users] ADSL routers with integrated SIP QoS for otherdevices   Tim Koehler
• [asterisk-users] OT: Capture Asterisk traffic   CSB
• [asterisk-users] OT: Capture Asterisk traffic   CSB
• Re Re: [asterisk-users] TC400B   bilal ghayyad
• [asterisk-users] Re: RE: Digital Phones (Dean Collins)   bilal ghayyad
• [asterisk-users] Newbie Question about E1   Luar Roji
• [asterisk-users] How many users can be supported simultaneously?   Antonopoulos Angelos
• [asterisk-users] VPN between Asterisk server and phone client   Biju
• [asterisk-users] Asterisk Integration of XMPP/Jingle   demuel at thephinix.org
• [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help   Bruce McAlister
• [asterisk-users] OT: Capture Asterisk traffic   Gordon Henderson
• [asterisk-users] How many users can be supported simultaneously?   Knud Müller
• [asterisk-users] Applet?   Stephen Wingfield
• [asterisk-users] OT: Capture Asterisk traffic   Joe acquisto
• [asterisk-users] OT: Capture Asterisk traffic   Steve Totaro
• [asterisk-users] Improving Asterisk's DNS support   Per Jessen
• [asterisk-users] Applet?   Steve Totaro
• [asterisk-users] Volume (gain?) on VoIP-only system.   Ken D'Ambrosio
• [asterisk-users] headsets for linksys/sipura phones?   Per Jessen
• [asterisk-users] VPN between Asterisk server and phone client   Steve Totaro
• [asterisk-users] Funky BIND/named errors   Per Jessen
• [asterisk-users] MySQL ** DBI connect failed : Too many connections **   Doug Lytle
• [asterisk-users] Digital Phones   Steve Totaro
• [asterisk-users] make iso image   Andrew Latham
• [asterisk-users] voice mail format   Adam KOSA
• [asterisk-users] make iso image   Giedrius Augys
• [asterisk-users] make iso image   Tzafrir Cohen
• [asterisk-users] OT: Capture Asterisk traffic   Tzafrir Cohen
• [asterisk-users] How do I do this in Asterisk?   Christian
• [asterisk-users] MySQL ** DBI connect failed : Too many connections**   Steve Totaro
• [asterisk-users] ZAP Error: Unable to create channel of type 'Zap'   Markus A Wipfler
• [asterisk-users] Improving Asterisk's DNS support   Darrick Hartman
• [asterisk-users] ADSL routers with integrated SIP QoS for otherdevices   Mike Dent
• [asterisk-users] Digital Phones   Stephen Bosch
• [asterisk-users] Volume (gain?) on VoIP-only system.   Stephen Bosch
• [asterisk-users] Testing Asterisk and Zaptel   Martin Smith
• [asterisk-users] Reinvite after DTMF?   Wilson Pickett
• [asterisk-users] SIPGetHeader in Asterisk 1.4   Jean-Marc Salsa
• [asterisk-users] Applet?   Dean Collins
• [asterisk-users] SIPGetHeader in Asterisk 1.4   Manu Mehta
• Re Re: [asterisk-users] TC400B   Ed Nuñez
• [asterisk-users] ADSL routers with integrated SIP QoS for otherdevices   Tim Koehler
• [asterisk-users] SIPGetHeader in Asterisk 1.4   Jean-Marc Salsa
• [asterisk-users] MySQL ** DBI connect failed : Too many connections**   Doug Lytle
• [asterisk-users] Runaway MOH/mp3123 process?   Matthew J. Roth
• [asterisk-users] cdr on channel lacks end, not posted   Maysara A. Abdulhaq
• [asterisk-users] Applet?   Salvatore Giudice
• [asterisk-users] Returning different SIP Hangup Cause   Jean-Marc Salsa
• [asterisk-users] using Playback() to play a random sound file   Steve Finkelstein
• [asterisk-users] Calls in ulaw, not gsm as desired   Rob Schall
• [asterisk-users] Asterisk-1.4 with agent snmp   Matt Gibson
• [asterisk-users] OT: Capture Asterisk traffic   Salvatore Giudice
• [asterisk-users] Digital Phones   Salvatore Giudice
• [asterisk-users] Re: headsets for linksys/sipura phones?   JR Richardson
• [asterisk-users] CDR changes in 1.4.3?   Andreas Sikkema
• [asterisk-users] delay in switching between contexts   Danish Samad
• [asterisk-users] Returning different SIP Hangup Cause   Kristian Kielhofner
• [asterisk-users] Asterisk-1.4 with agent snmp   Everton Goularth
• [asterisk-users] headsets for linksys/sipura phones?   Wilson Pickett
• [asterisk-users] Re: headsets for linksys/sipura phones?   John Novack
• [asterisk-users] Re: headsets for linksys/sipura phones?   Per Jessen
• [asterisk-users] delay in switching between contexts   Eric "ManxPower" Wieling
• [asterisk-users] Re: OT: Capture Asterisk traffic   Benny Amorsen
• [asterisk-users] delay in switching between contexts   Andrew Kohlsmith
• [asterisk-users] Re: How many users can be supported simultaneously?   Benny Amorsen
• [asterisk-users] Re: How many users can be supported simultaneously?   Salvatore Giudice
• [asterisk-users] Calls in ulaw, not gsm as desired   Jaswinder Singh
• [asterisk-users] Call In queue stucks   Sanjay Rajdev
• [asterisk-users] 1.4 memory leak?   Adam Moffett
• [asterisk-users] 1.4 memory leak?   Steve Finkelstein
• [asterisk-users] Calls in ulaw, not gsm as desired   Jerry Jones
• [asterisk-users] Returning different SIP Hangup Cause   Yuan LIU
• [asterisk-users] Daemontools and holidays macro   Vicente Aguilar
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   San Singhania
• [asterisk-users] Asterisk locked up   shadowym
• [asterisk-users] Reinvite after DTMF?   Yuan LIU
• [asterisk-users] Large dial plans and variables   Doug Garstang
• [asterisk-users] make iso image   Khaled Chehab
• [asterisk-users] PRI T1 Problems   Frank at explido.us
• [asterisk-users] allowing call to my pabx every 15 minutes   Goke Aruna
• [asterisk-users] allowing call every 15mins   Goke Aruna
• [asterisk-users] Answer A Ringing Queue By Dialing An Extension   Brandon Comouche
• [asterisk-users] Re: How many users can be supported simultaneously?   Knud Müller
• [asterisk-users] Large dial plans and variables   Philipp Kempgen
• [asterisk-users] VPN between Asterisk server and phone client   Kai-Uwe Jensen
• [asterisk-users] Large dial plans and variables   Philipp Kempgen
• [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help   Remco Post
• [asterisk-users] allowing call to my pabx every 15 minutes   Remco Post
• [asterisk-users] PRI T1 Problems   Michael Welter
• [asterisk-users] Returning different SIP Hangup Cause   Kristian Kielhofner
• [asterisk-users] allowing call every 15mins   Andrew Kohlsmith
• [asterisk-users] PRI T1 Problems   Steve Totaro
• [asterisk-users] allowing call every 15mins   Steve Totaro
• [asterisk-users] VPN between Asterisk server and phone client   Steve Totaro
• [asterisk-users] Daemontools and holidays macro   Steve Totaro
• [asterisk-users] Large dial plans and variables   Doug Garstang
• [asterisk-users] allowing call every 15mins   Salvatore Giudice
• [asterisk-users] VPN between Asterisk server and phone client   Salvatore Giudice
• [asterisk-users] Calls in ulaw, not gsm as desired   Rob Schall
• [asterisk-users] SIP Proxy   Ronaldo
• [asterisk-users] allowing call every 15mins   SIP
• [asterisk-users] allowing call every 15mins   Luki
• [asterisk-users] allowing call every 15mins   Derek Whitten
• [asterisk-users] Kansas City Asterisk User Group Meeting Announcement   Kyle Sexton
• [asterisk-users] allowing call every 15mins   Salvatore Giudice
• [asterisk-users] OT: USB T1/E1 Interface?   Michael Collins
• [asterisk-users] VPN between Asterisk server and phone client   Salvatore Giudice
• [asterisk-users] VPN between Asterisk server and phone client   Kai-Uwe Jensen
• [asterisk-users] Calls in ulaw, not gsm as desired   Salvatore Giudice
• [asterisk-users] Poor man's High Availability solution   Laurent Caron
• [asterisk-users] using Playback() to play a random sound file   dave cantera
• [asterisk-users] 1.4 memory leak?   Adam Moffett
• [asterisk-users] allowing call every 15mins   Salvatore Giudice
• [asterisk-users] 1.4 memory leak?   Adam Moffett
• [asterisk-users] 1.4 memory leak?   Salvatore Giudice
• [asterisk-users] VPN between Asterisk server and phone client   Kai-Uwe Jensen
• [asterisk-users] VPN between Asterisk server and phone client   Darrick Hartman
• [asterisk-users] IAX and SETLANGUAGE delays   Joel Hill
• [asterisk-users] VPN between Asterisk server and phone client   Salvatore Giudice
• [asterisk-users] Calls in ulaw, not gsm as desired   Dave Miller
• [asterisk-users] IAX and SETLANGUAGE delays   Jonathan Barratt
• [asterisk-users] IAX and SETLANGUAGE delays   Joel Hill
• [asterisk-users] OT: USB T1/E1 Interface?   Remco Post
• [asterisk-users] Poor man's High Availability solution   Laurent Caron
• [asterisk-users] rtpmap encoding parameters & the 'unknown codec' problem?   Ray Jackson
• [asterisk-users] Call Limit with multiple SIP extensions   Michael Hamann
• [asterisk-users] SIP Proxy   Remco Post
• [asterisk-users] Reinvite after DTMF?   Wilson Pickett
• [asterisk-users] allowing call every 15mins   Anselm Martin Hoffmeister
• [asterisk-users] [Announce] Web-MeetMe V3.0.1 released   Ondrej Valousek
• [asterisk-users] 0 duration but non-zero billsec in mysql cdr   Jaswinder Singh
• [asterisk-users] 0 duration but non-zero billsec in mysql cdr   Salvatore Giudice
• [asterisk-users] 0 duration but non-zero billsec in mysql cdr   Jaswinder Singh
• [asterisk-users] UK - 2 port ISDN2e cards ...   Gordon Henderson
• [asterisk-users] 0 duration but non-zero billsec in mysql cdr   Salvatore Giudice
• [asterisk-users] sipura spa9x1 - missed calls not wanted   Per Jessen
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Per Jessen
• [asterisk-users] Large dial plans and variables   Andreas Sikkema
• [asterisk-users] UK - 2 port ISDN2e cards ...   Per Jessen
• [asterisk-users] Get Asterisk to redirect a SIP INVITE   CSB
• [asterisk-users] Double DTMF digits sent on IAX native bridge   Steve Davies
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Mats Karlsson
• [asterisk-users] sipura spa9x1 - missed calls not wanted   Dave Cotton
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Dave Cotton
• [asterisk-users] ADSL routers with integrated SIP QoS for otherdevices   Francisco Pérez Botella In
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xmlgeneration   Gavin Spurgeon
• [asterisk-users] sipura spa9x1 - missed calls not wanted   Per Jessen
• [asterisk-users] My Polycom IP 501 is formatted its file system itself   Crazy Boy
• [asterisk-users] My Polycom IP 501 is formatted its file systemitself   Crazy Boy
• [asterisk-users] My Polycom IP 501 is formatted its file system itself   Crazy Boy
• [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)   Rizwan Hisham
• [asterisk-users] VPN between Asterisk server and phone client   Steve Totaro
• [asterisk-users] Daemontools and holidays macro   William Moore
• [asterisk-users] OT: USB T1/E1 Interface?   Steve Totaro
• [asterisk-users] zttranscode crashes server   Forrest Beck
• [asterisk-users] UK - 2 port ISDN2e cards ...   Gordon Henderson
• [asterisk-users] zttranscode crashes server   Tzafrir Cohen
• [asterisk-users] Daemontools and holidays macro   Tzafrir Cohen
• [asterisk-users] Digital Phones   Jason Fuermann
• [asterisk-users] Answer A Ringing Queue By Dialing An Extension   Chris Bagnall
• [asterisk-users] Called party identification - where to take called name?   Yehavi Bourvine +972-8-9489444
• [asterisk-users] Virtual IP Adresses and SIP requests failing...   Christopher Aloi
• [asterisk-users] Semi-OT: useful things to do with XML browsers in phones   Chris Bagnall
• [asterisk-users] Wildcard TE410P problem   Alexandr Olekhnovich
• [asterisk-users] Semi-OT: useful things to do with XML browsers in phones   Andrew Kohlsmith
• [asterisk-users] zttranscode crashes server   Forrest Beck
• [asterisk-users] Secondary redirect failed   pandi ponnangan
• [asterisk-users] zttranscode crashes server   Forrest Beck
• [asterisk-users] Daemontools and holidays macro   Vicente Aguilar
• [asterisk-users] Semi-OT: useful things to do with XML browsers inphones   Dean Collins
• [asterisk-users] "you have been kicked my this conference"   Jerry Geis
• [asterisk-users] Double DTMF digits sent on IAX native bridge   Steve Davies
• [asterisk-users] using Playback() to play a random sound file   Brandon Comouche
• [asterisk-users] Linksys SPA3012 inbound FXO problems   lenz
• [asterisk-users] IAX Trunk   Ronaldo
• [asterisk-users] IAX Trunk   Dean Collins
• [asterisk-users] IAX Trunk   Bruce Reeves
• [asterisk-users] IAX Trunk   Bruce Reeves
• [asterisk-users] IAX Trunk   William Moore
• [asterisk-users] UK - 2 port ISDN2e cards ...   Stephen Bosch
• [asterisk-users] Linksys SPA3012 inbound FXO problems   Cosmin Prund
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Stephen Bosch
• [asterisk-users] OT: USB T1/E1 Interface?   Michael Collins
• [asterisk-users] IAX Trunk   Ronaldo
• [asterisk-users] IAX Trunk   Ronaldo
• [asterisk-users] Daemontools and holidays macro   Stephen Bosch
• [asterisk-users] Daemontools and holidays macro   Stephen Bosch
• [asterisk-users] Digital Phones   Stephen Bosch
• [asterisk-users] Large dial plans and variables   Doug Garstang
• [asterisk-users] OT: USB T1/E1 Interface?   Frank at explido.us
• [asterisk-users] Linksys SPA3012 inbound FXO problems   Dave Cotton
• [asterisk-users] "you have been kicked my this conference"   Salvatore Giudice
• [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)   Steve Murphy
• [asterisk-users] IAX Trunk   Steve Kennedy
• [asterisk-users] IAX Trunk   Gordon Henderson
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Jim Suber
• [asterisk-users] [Announce] Web-MeetMe V3.0.1 released   Dan Austin
• [asterisk-users] using Playback() to play a random sound file   Steve Edwards
• [asterisk-users] Called party identification - where to take calledname?   Dan Austin
• [asterisk-users] Autologoff   Ed Nuñez
• [asterisk-users] ast_parse_allow_disallow: Cannot disallow unknown format ''   Dima
• [asterisk-users] Semi-OT: useful things to do with XML browsers inphones   Dan Austin
• [asterisk-users] IAX Trunk   Bruce Reeves
• [asterisk-users] OT: USB T1/E1 Interface?   Jorge Mendoza
• [asterisk-users] zttranscode crashes server   Tzafrir Cohen
• [asterisk-users] chan_sip seems to be hanging   Jaswinder Singh
• [asterisk-users] IAX Trunk   Salvatore Giudice
• [asterisk-users] OT: USB T1/E1 Interface?   Steve Edwards
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Erik Anderson
• [asterisk-users] Linksys SPA3012 inbound FXO problems   lenz
• [asterisk-users] IAX Trunk   Ronaldo
• [asterisk-users] OT: USB T1/E1 Interface?   Michael Collins
• [asterisk-users] Linseed   Dean Collins
• [asterisk-users] Strange noise - Polycom   Rob Schall
• [asterisk-users] Wildcard TE410P problem   f6hqz-m at hamwlan.net
• [asterisk-users] Wildcard TE410P problem   f6hqz-m at hamwlan.net
• [asterisk-users] FXO recommendation   Kyle Gordon
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Bruce Reeves
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Gordon Henderson
• [asterisk-users] Semi-OT: useful things to do with XML browsersinphones   Dean Collins
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Stephen Bosch
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Doug Lytle
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Jason Adams
• [asterisk-users] Poor man's High Availability solution   Stephen Davies
• [asterisk-users] Answer A Ringing Queue By Dialing An Extension   Brandon Comouche
• [asterisk-users] Double DTMF digits   Ken Leland III
• [asterisk-users] Linksys SPA3012 inbound FXO problems   Drew Gibson
• [asterisk-users] OT: USB T1/E1 Interface?   Steve Edwards
• [asterisk-users] Re: Asterisk-1.4 with agent snmp   Everton Goularth
• [asterisk-users] OT: USB T1/E1 Interface?   John Treble
• [asterisk-users] IAX and SETLANGUAGE delays   Jonathan Barratt
• [asterisk-users] Balancing interrupts.   Steve Edwards
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Brandon Kruse
• [asterisk-users] OT: USB T1/E1 Interface?   Michael Collins
• [asterisk-users] VoiceXML + Nuance   Eric Rousse
• [asterisk-users] Error messages in console : sipsock_read: SIP MESSAGE JUST IGNORED:   Don Fletcher
• [asterisk-users] Linseed   Salvatore Giudice
• [asterisk-users] IAX Trunk   Salvatore Giudice
• [asterisk-users] Reinvite after DTMF?   Yuan LIU
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Tom Rymes
• [asterisk-users] Get Asterisk to redirect a SIP INVITE   Yuan LIU
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Doug Garstang
• [asterisk-users] Called party identification - where to takecalledname?   Yuan LIU
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Andreas van dem Helge
• [asterisk-users] IAX Trunk   Steve Kennedy
• [asterisk-users] OT: USB T1/E1 Interface?   Michael Collins
• [asterisk-users] Restricting membership to a queue?   Fernando Gleiser
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Eric "ManxPower" Wieling
• [asterisk-users] VoiceXML + Nuance   wendell hamilton
• [asterisk-users] OT: USB T1/E1 Interface?   Michael Collins
• [asterisk-users] Asterisk 1.4 and Cisco Phones 7940   Erick Perez
• [asterisk-users] SIP RealTime Friends   Forrest Beck
• [asterisk-users] Unable to Execute System Command From DialPlan   vcomp
• [asterisk-users] SIP peer / Maximum retries exceeded on transmission   chris at cgb1911.mine.nu
• [asterisk-users] [Announce] Web-MeetMe 3.0.2 and 2.2.2 Released   Dan Austin
• [asterisk-users] Get Asterisk to redirect a SIP INVITE   Salvatore Giudice
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Salvatore Giudice
• [asterisk-users] Asterisk 1.4 and Cisco Phones 7940   Salvatore Giudice
• [asterisk-users] Unable to Execute System Command From DialPlan   Salvatore Giudice
• [asterisk-users] Connections rejected in DUNDi requests   Chris Bagnall
• [asterisk-users] T1/E1 Configuration   Nitesh Divecha
• [asterisk-users] T1/E1 Configuration   Forrest Beck
• [asterisk-users] Called party identification - where to takecalledname?   Yehavi Bourvine +972-8-9489444
• [asterisk-users] OT - robo dialer   Doug Crompton
• [asterisk-users] Large dial plans and variables   Ira
• [asterisk-users] OT - robo dialer   Chris Bennett
• [asterisk-users] OT - robo dialer   Doug Crompton
• [asterisk-users] Call interruption   Andre Wangler
• [asterisk-users] SIP RealTime Friends   0xception
• [asterisk-users] Connections rejected in DUNDi requests   Remco Post
• [asterisk-users] SIP RealTime Friends   Remco Post
• [asterisk-users] Poor man's High Availability solution   FailSafeVOIP
• [asterisk-users] Poor man's High Availability solution   FailSafeVOIP
• [asterisk-users] Balancing interrupts.   Matthew J. Roth
• [asterisk-users] VoiceXML + Nuance   Rob Townley
• [asterisk-users] Asterisk Codec Translation Table   Al
• [asterisk-users] Re: Balancing interrupts.   Daniel Pittman
• [asterisk-users] UK - 2 port ISDN2e cards ...   Per Jessen
• [asterisk-users] Re: headsets for linksys/sipura phones?   Per Jessen
• [asterisk-users] OT - robo dialer   Anselm Martin Hoffmeister
• [asterisk-users] UK - 2 port ISDN2e cards ...   Gordon Henderson
• [asterisk-users] 1.2.x -> 1.4.x upgrade: dialplan block no longer works   Louis-David Mitterrand
• [asterisk-users] FXO recommendation   Gordon Henderson
• [asterisk-users] Double DTMF digits   Steve Davies
• [asterisk-users] FXO recommendation   Phil Reynolds
• [asterisk-users] T1/E1 Configuration   Tzafrir Cohen
• [asterisk-users] Asterisk Users Conference Friday, May 4th at 12:30 PM EDT   Wilson Pickett
• [asterisk-users] Reinvite after DTMF?   Wilson Pickett
• [asterisk-users] Call interruption   Yuan LIU
• [asterisk-users] need more knowledge about asterisk   Iban Lopetegi Zinkunegi
• [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)   Rizwan Hisham
• [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)   Rizwan Hisham
• [asterisk-users] FXO recommendation   Gergo Csibra
• [asterisk-users] Connections rejected in DUNDi requests   Chris Bagnall
• [asterisk-users] need more knowledge about asterisk   Rizwan Hisham
• [asterisk-users] need more knowledge about asterisk   Iban Lopetegi Zinkunegi
• [asterisk-users] SIP RealTime Friends   Sergio (Red)
• [asterisk-users] need more knowledge about asterisk   Rizwan Hisham
• [asterisk-users] IAX Trunk   Ronaldo
• [asterisk-users] Error compiling patched pppd for zapras   Alex
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Christian
• [asterisk-users] UK zaptel and zapata.conf for TDM400P   Wireless
• [asterisk-users] Asterisk Codec Translation Table   Salvatore Giudice
• [asterisk-users] FXO recommendation   Joe acquisto
• [asterisk-users] app_txfax, app_rxfax   mail-lists
• [asterisk-users] app_txfax, app_rxfax   Doug Lytle
• [asterisk-users] app_txfax, app_rxfax   Stefan Wintermeyer
• [asterisk-users] FXO recommendation   Dave Cotton
• [asterisk-users] Console flooded by WARNING app_meetme messages   Heison Chak
• [asterisk-users] SIP RealTime Friends   Forrest Beck
• [asterisk-users] SIP RealTime Friends   Vadim Berezniker
• [asterisk-users] SIP RealTime Friends   Forrest Beck
• [asterisk-users] app_txfax, app_rxfax   Thomas Göttgens
• [asterisk-users] Asterisk Codec Translation Table   Tzafrir Cohen
• [asterisk-users] does Not detected HANGUP and DTMF   pandi ponnangan
• [asterisk-users] Runaway MOH/mp3123 process?   gc
• [asterisk-users] app_txfax, app_rxfax   mail-lists
• [asterisk-users] app_txfax, app_rxfax   Doug Lytle
• [asterisk-users] need more knowledge about asterisk   Tzafrir Cohen
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Tzafrir Cohen
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Stephen Bosch
• [asterisk-users] VoiceXML + Nuance   Eric Rousse
• [asterisk-users] T1/E1 Configuration   Nitesh Divecha
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Darrick Hartman
• [asterisk-users] Headset for Polycom   Mike
• [asterisk-users] Headset for Polycom   Michael Cargile
• [asterisk-users] T1/E1 Configuration   John Treble
• [asterisk-users] Headset for Polycom   Alex Robar
• [asterisk-users] T1/E1 Configuration   Nitesh Divecha
• [asterisk-users] Headset for Polycom   Jim Rice
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Time Bandit
• [asterisk-users] Reinvite after DTMF?   Yuan LIU
• [asterisk-users] ASA-2007-013: IAX2 users can cause unauthorized data disclosure   Kevin P. Fleming
• [asterisk-users] app_txfax, app_rxfax   Gordon Henderson
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Christian
• [asterisk-users] Headset for Polycom   Michael Cargile
• [asterisk-users] FXO recommendation   Gergo Csibra
• [asterisk-users] app_txfax, app_rxfax   Lee Howard
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   James Texter
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Nitesh Divecha
• [asterisk-users] app_txfax, app_rxfax   Kevin Collins
• [asterisk-users] zaptel compile error   mail-lists
• [asterisk-users] app_txfax, app_rxfax   Per Jessen
• [asterisk-users] app_txfax, app_rxfax   Gordon Henderson
• [asterisk-users] AsteriskNow!   Ed Nuñez
• [asterisk-users] cpu usuage   Khaled Chehab
• [asterisk-users] T1/E1 Configuration   Andreas van dem Helge
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Tzafrir Cohen
• [asterisk-users] allowing call to my pabx every 15 minutes   bkruse
• [asterisk-users] question about more than one drop file   shawn bright
• [asterisk-users] Re: Unable to Execute System Command From DialPlan   Victor
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Tom Rymes
• [asterisk-users] RE: Autologoff   Ed Nuñez
• [asterisk-users] T1/E1 Configuration   Tzafrir Cohen
• [asterisk-users] E1 config for chile   laurent schweizer
• [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)   Steve Murphy
• [asterisk-users] app_txfax, app_rxfax   Luki
• [asterisk-users] question about more than one drop file   bkruse
• [asterisk-users] Asterisk registration SIP confusion. Can someone explain this?   Michelle Dupuis
• [asterisk-users] T1/E1 Configuration   Nitesh Divecha
• [asterisk-users] app_txfax, app_rxfax   Doug Lytle
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Christian
• [asterisk-users] Asterisk 1.2 on CentOS 5?   shadowym
• [asterisk-users] app_txfax, app_rxfax   Lee Howard
• [asterisk-users] Semi-OT: useful things to do with XML browsers inphones   Dave Bour
• [asterisk-users] app_txfax, app_rxfax   Luki
• [asterisk-users] OT: USB T1/E1 Interface?   Leo Ann Boon
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Yuan LIU
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Mark Coccimiglio
• [asterisk-users] RXFAX/TXFAX   Cesar Benjamin Garcia Martinez
• [asterisk-users] app_txfax, app_rxfax   Lee Howard
• [asterisk-users] SLA broken in 1.4.3?   Jay Austad
• [asterisk-users] SLA broken in 1.4.3?   Stephen Bosch
• [asterisk-users] Asterisk 1.2 on CentOS 5?   Noah Miller
• [asterisk-users] Re: Balancing interrupts.   Carlos Rojas
• [asterisk-users] AsteriskNow!   Forrest Beck
• [asterisk-users] zaptel compile error   Forrest Beck
• [asterisk-users] Queue and voicemail   Josué Conti
• [asterisk-users] Asterisk x legacy pabx   Josué Conti
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Tzafrir Cohen
• [asterisk-users] Queue Answer   Arun Kumar
• [asterisk-users] Queue and voicemail   Per Jessen
• [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation   Per Jessen
• [asterisk-users] Queue and voicemail   0xception
• [asterisk-users] RXFAX/TXFAX   Per Jessen
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Mark Coccimiglio
• [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)   Rizwan Hisham
• [asterisk-users] Queue Status   Arun Kumar
• [asterisk-users] TLS support   Alexandr Olekhnovich
• [asterisk-users] Asterisk x legacy pabx   Dean Collins
• [asterisk-users] Manager API Output   Arun Kumar
• [asterisk-users] SIP registration problem   Michelle Dupuis
• [asterisk-users] Queue and voicemail   Josué Conti
• [asterisk-users] Manager API Output   Philipp Kempgen
• [asterisk-users] Asterisk x legacy pabx   Josué Conti
• [asterisk-users] Manager API Output   Gordon Henderson
• [asterisk-users] Asterisk x legacy pabx   Salvatore Giudice
• [asterisk-users] TDM400P usada?   Rodrigo Mercado
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Remco Post
• [asterisk-users] Re: Virtual IP Adresses and SIP requests failing...   Christopher Aloi
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Tzafrir Cohen
• [asterisk-users] TDM400P usada?   Tom Rymes
• [asterisk-users] asterisk telemarketer torture sound files   Adam Jacob Muller
• [asterisk-users] asterisk telemarketer torture sound files   Dave Miller
• [asterisk-users] Queue and voicemail   0xception
• [asterisk-users] TDM400P usada?   Rodrigo Mercado
• [asterisk-users] asterisk telemarketer torture sound files   Adam Jacob Muller
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Remco Post
• [asterisk-users] SLA broken in 1.4.3?   David W. Rice
• [asterisk-users] GXV-3000 IP Video Phone   Nitesh Divecha
• [asterisk-users] Asterisk x legacy pabx   Josué Conti
• [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!   Jim Suber
• [asterisk-users] RXFAX/TXFAX   Cesar Benjamin Garcia Martinez
• [asterisk-users] TDM400P usada?   Cesar Benjamin Garcia Martinez
• [asterisk-users] AsteriskNow!   Bill Merriam
• [asterisk-users] ODBC   Forrest Beck
• [asterisk-users] RXFAX/TXFAX   Tzafrir Cohen
• [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?   Tom Lynn
• [asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checking for PQexec in -lpq... no)   Gavin Henry
• [asterisk-users] RXFAX/TXFAX   Andreas van dem Helge
• [asterisk-users] AsteriskNow!   Steve Edwards
• [asterisk-users] I'm looking for solution   Ardit Saliu
• [asterisk-users] asterisk telemarketer torture sound files   Dave Miller
• [asterisk-users] I'm looking for solution   Remco Post
• [asterisk-users] I'm looking for solution   Ardit Saliu
• [asterisk-users] res_config_pgsql.c in * 1.4.4   Gavin Henry
• [asterisk-users] TDM400P usada?   Carlos Rojas
• [asterisk-users] Dial Plan for Multi-Location & Support Queue   Deepak Naidu
• [asterisk-users] OT: USB T1/E1 Interface?   Michael Collins
• [asterisk-users] ${ANSWEREDTIME} Broken on 1.2.13?   Barton Fisher
• [asterisk-users] Channel / Exten Status   Pablo L. Arturi
• [asterisk-users] using Playback() to play a random sound file   dave cantera
• [asterisk-users] RXFAX/TXFAX   Michelle Dupuis
• [asterisk-users] auto call out via drop file ERROR: 'OutgoingSpoolFailed'   dave cantera
• [asterisk-users] GXV-3000 IP Video Phone   dave cantera
• [asterisk-users] question about more than one drop file   dave cantera
• [asterisk-users] ASA-2007-013: IAX2 users can cause unauthorizeddata disclosure   Dovid B
• [asterisk-users] Asterisk x legacy pabx   Salvatore Giudice
• [asterisk-users] asterisk telemarketer torture sound files   Salvatore Giudice
• [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?   Salvatore Giudice
• [asterisk-users] SLA broken in 1.4.3?   Jay Austad
• [asterisk-users] Fwd: Queue Status   Arun Kumar
• Fwd: [asterisk-users] Change Codec   Arun Kumar
• [asterisk-users] asterisk telemarketer torture sound files   Lacy Moore - Aspendora
• [asterisk-users] I'm looking for solution   Ardit Saliu
• [asterisk-users] Polycom 601 - To not make noise when there is VM   Dovid B
• [asterisk-users] Call waiting tone when calling a busy station?   Yehavi Bourvine +972-8-9489444
• [asterisk-users] Were i make mistake   Ardit Saliu
• [asterisk-users] Were i make mistake   Nico Busch
• [asterisk-users] Headset for Polycom   Bryan M. Johns
• [asterisk-users] Polycom 601 - To not make noise when there is VM   Alvin Austin
• [asterisk-users] Buddywatch on Polycom 601 crashes phone   J French
• [asterisk-users] Call waiting tone when calling a busy station?   Edoardo Serra
• [asterisk-users] asterisk telemarketer torture sound files   Steve Finkelstein
• [asterisk-users] I'm looking for solution   Pablo L. Arturi
• [asterisk-users] I'm looking for solution   Pablo L. Arturi
• [asterisk-users] Asterisk x legacy pabx   Josué Conti
• [asterisk-users] SLA broken in 1.4.3?   David W. Rice
• [asterisk-users] Call waiting tone when calling a busy station?   Yehavi Bourvine +972-8-9489444
• [asterisk-users] question about more than one drop file   Lee Jenkins
• [asterisk-users] asterisk telemarketer torture sound files   Adam Jacob Muller
• [asterisk-users] Asterisk x legacy pabx   C F
• [asterisk-users] Asterisk 1.2.14-BRIstuffed-0.3.0-PRE-1y   Lists
• [asterisk-users] telemarketer database ....next stages... Was asterisk telemarketer torture sound files   Dave Bour
• [asterisk-users] telemarketer database ....next stages... Was asterisk telemarketer torture sound files   Steve Finkelstein
• [asterisk-users] SLA broken in 1.4.3?   shadowym
• [asterisk-users] My Polycom IP 501 is formatted its file systemitself   Steven Ringwald
• [asterisk-users] telemarketer database ....next stages... Was asterisk telemarketer torture sound files   Adam Jacob Muller
• [asterisk-users] Problem with conferences, Vlada, Pancevo   Vladimir Kovacevic
• [asterisk-users] telemarketer database ....next stages... Wasasterisk telemarketer torture sound file   Dave Bour
• [asterisk-users] Channel Bank   Forum
• [asterisk-users] asterisk telemarketer torture sound files   Anselm Martin Hoffmeister
• [asterisk-users] Channel Bank   Doug Lytle
• [asterisk-users] Channel Bank   Brian Roy
• [asterisk-users] asterisk telemarketer torture sound files   Adam Jacob Muller
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Tzafrir Cohen
• [asterisk-users] My Polycom IP 501 is formatted its file systemitself   Crazy Boy
• [asterisk-users] My Polycom IP 501 is formatted its file systemitself   Steven Ringwald
• [asterisk-users] Re: [Dundi] Dial Plan for Multi-Location & Support Queue   Deepak Naidu
• [asterisk-users] Re: [Dundi] Dial Plan for Multi-Location & Support Queu   Vamsi Pottangi
• [asterisk-users] app_txfax, app_rxfax   Olivier
• [asterisk-users] HPEC audio clipping   Olivier
• [asterisk-users] Send SIP Re-invite.   Rohan Hathiwala
• [asterisk-users] GXV-3000 IP Video Phone   Andreas van dem Helge
• [asterisk-users] T1/E1 Configuration   Andreas van dem Helge
• [asterisk-users] CDR changes in 1.4.3?   Andreas van dem Helge
• [asterisk-users] zaptel compile error   mail-lists
• [asterisk-users] zaptel compile error   mail-lists
• [asterisk-users] zaptel compile error   mail-lists
• [asterisk-users] Were i make mistake   Ronaldo
• [asterisk-users] Problem with conferences, Vlada, Pancevo   Ronaldo
• [asterisk-users] Starting Asterisk on Ubuntu 7.04   Mark Coccimiglio
• [asterisk-users] Channel Bank   Andrew Kohlsmith
• [asterisk-users] Call waiting tone when calling a busy station?   Edoardo Serra
• [asterisk-users] Channel Bank   Eric "ManxPower" Wieling
• [asterisk-users] Rhino Analog Cards - Questions   Lee Jenkins
• [asterisk-users] RTP Mixer   Akashdeep Dutta
• [asterisk-users] Problem with the loading of the cards in Debian   MCelo
• [asterisk-users] Queues: Play a list of sound file n round-robin at a specific interval   Andre Courchesne - Consultant
• [asterisk-users] RTP Mixer   Akashdeep Dutta
• [asterisk-users] RTP Mixer   Akashdeep Dutta
• [asterisk-users] Queues: Play a list of sound file n round-robin at a specific interval   Lenz
• [asterisk-users] ISDN with Billion   Josu Lazkano Lete
• [asterisk-users] Asterisk to record CDR in DB Oracle   Everton Goularth
• [asterisk-users] GXV-3000 IP Video Phone   Nitesh Divecha
• [asterisk-users] zaptel compile error   Tzafrir Cohen
• [asterisk-users] GXV-3000 IP Video Phone   Jorge Mendoza
• [asterisk-users] Problem with the loading of the cards in Debian   Tzafrir Cohen
• [asterisk-users] Problem with the loading of the cards in Debian   Tzafrir Cohen
• [asterisk-users] H323 to H323 bridging ... failed ... also with chan_local   Cesc
• [asterisk-users] Asterisk to record CDR in DB Oracle   Florian Overkamp
• [asterisk-users] RTP Mixer   Joshua Colp
• [asterisk-users] Channel Bank   Russ Price
• [asterisk-users] Send SIP Re-invite.   Joshua Colp
• [asterisk-users] ${ANSWEREDTIME} Broken on 1.2.13?   Joshua Colp
• [asterisk-users] app_txfax, app_rxfax   Luki
• [asterisk-users] Queues: Play a list of sound file n round-robin at a specific interval   Andre Courchesne - Consultant
• [asterisk-users] GXV-3000 IP Video Phone   Nitesh Divecha
• [asterisk-users] Could two Asterisk servers connect through VPN   Tielin Xu
• [asterisk-users] app_txfax, app_rxfax   Olivier
• [asterisk-users] Could two Asterisk servers connect through VPN   Chris Mason (Lists)
• [asterisk-users] Could two Asterisk servers connect through VPN   Sahil Gupta
• [asterisk-users] h323 problem with asterisk 1.2.18   nik600
• [asterisk-users] Could two Asterisk servers connect through VPN   Stefan Wintermeyer
• [asterisk-users] Problem with the loading of the cards in Debian   MCelo
• [asterisk-users] Problem with the loading of the cards in Debian   Tzafrir Cohen
• [asterisk-users] Could two Asterisk servers connect through VPN   Gordon Henderson
• [asterisk-users] Could two Asterisk servers connect through VPN   Tielin Xu
• [asterisk-users] Could two Asterisk servers connect through VPN   Noah Miller
• [asterisk-users] ${ANSWEREDTIME} Broken on 1.2.13?   Barton Fisher
• [asterisk-users] Problem with the loading of the cards in Debian   MCelo
• [asterisk-users] HPEC audio clipping   Noah Miller
• [asterisk-users] Buddywatch on Polycom 601 crashes phone   Noah Miller
• [asterisk-users] help MWI setting   Ray Chen
• [asterisk-users] help MWI setting   Erik Anderson
• [asterisk-users] iax to iax Reject Connection   chawki hammoud
• [asterisk-users] Problem with the loading of the cards in Debian   Tzafrir Cohen
• [asterisk-users] RTP Mixer   Akashdeep Dutta
• [asterisk-users] RTP Mixer   Alex Balashov
• [asterisk-users] Channel Bank   Eric "ManxPower" Wieling
• [asterisk-users] iax to iax Reject Connection   Matt
• [asterisk-users] RTP Mixer   Eric "ManxPower" Wieling
• [asterisk-users] Send SIP Re-invite.   Rohan Hathiwala
• [asterisk-users] RTP Mixer   Kapil Dhawan
• [asterisk-users] asterisk with festival facing problem!!!!!   Cheikhou DIAW
• [asterisk-users] isup-oli or ani2   JK
• [asterisk-users] Queues: Play a list of sound file n round-robin at a specific interval   Lenz
• [asterisk-users] Problems with SPA3102   Jonson Player
• [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?   Sven Jacobs
• [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?   Per Jessen
• [asterisk-users] Asterisk to record CDR in DB Oracle   Tim Panton
• [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?   Sven Jacobs
• [asterisk-users] Could two Asterisk servers connect through VPN   Tim Panton
• [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?   Per Jessen
• [asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 39   stewart at owd.com
• [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?   Sven Jacobs
• [asterisk-users] zaptel compile error   Arun Kumar
• [asterisk-users] zaptel compile error   Tzafrir Cohen
• [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?   Per Jessen
• [asterisk-users] Asterisk to record CDR in DB Oracle   Bruce McAlister
• [asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checkingfor PQexec in -lpq... no)   Bruce McAlister
• [asterisk-users] iax to iax Reject Connection   Ronaldo
• [asterisk-users] Queue Status   Edoardo Serra
• [asterisk-users] outgoing calls   Josu Lazkano Lete
• [asterisk-users] outgoing calls   Josu Lazkano Lete
• [asterisk-users] outgoing calls   Josu Lazkano Lete
• [asterisk-users] outgoing calls   Dijkstra, Roelof
• [asterisk-users] Beronet card - issue?   Enrico Pasqualotto
• [asterisk-users] Responding to SIP OPTIONS   Alex Lake
• [asterisk-users] asterisk 1.2 from svn ... lock on shutdown   Cesc
• [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?   Sven Jacobs
• [asterisk-users] Queue Status   Arun Kumar
• [asterisk-users] outgoing calls   Josu Lazkano Lete
• [asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checkingfor PQexec in -lpq... no)   Gavin Henry
• [asterisk-users] Sangoma A101 on Freebsd 6.2   Zvonimir Mileta
• [asterisk-users] load modules   Josu Lazkano Lete
• [asterisk-users] Send SIP Re-invite.   Joshua Colp
• [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?   Joshua Colp
• [asterisk-users] asterisk 1.2 from svn ... lock on shutdown   Joshua Colp
• [asterisk-users] isup-oli or ani2   Joshua Colp
• [asterisk-users] iax to iax Reject Connection   Joshua Colp
• [asterisk-users] Sangoma A101 on Freebsd 6.2   Noah Miller
• [asterisk-users] Re: h323 problem with asterisk 1.2.18   nik600
• [asterisk-users] Responding to SIP OPTIONS   Joshua Colp
• [asterisk-users] Sangoma A101 on Freebsd 6.2   Joshua Colp
• [asterisk-users] GXV-3000 IP Video Phone   Noah Miller
• [asterisk-users] Remote Phone and Server Behind NAT   Chris Shipman
• [asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 39   C F
• [asterisk-users] HPEC audio clipping   Jason Parker
• [asterisk-users] Responding to SIP OPTIONS   SIP
• [asterisk-users] Remote Phone and Server Behind NAT   Joshua Colp
• [asterisk-users] G729 - Part cut   Thomas Deillon
• [asterisk-users] Outbound call through a Single Asterisk Server   Davis Sylvester III
• [asterisk-users] GXV-3000 IP Video Phone   Nitesh Divecha
• [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?   Per Jessen
• [asterisk-users] Responding to SIP OPTIONS   Stephen Bosch
• [asterisk-users] G729 - Part cut   EWV2
• [asterisk-users] Asterisk 1.4.2 tanking CPU   Steve Finkelstein
• [asterisk-users] Re: h323 problem with asterisk 1.2.18   Cesc
• [asterisk-users] app_txfax, app_rxfax   mail-lists
• [asterisk-users] app_txfax, app_rxfax   mail-lists
• [asterisk-users] asterisk 1.2 from svn ... lock on shutdown   Cesc
• [asterisk-users] Sangoma cards for sale   Porier, Jeremy M.
• [asterisk-users] app_txfax, app_rxfax   Kevin Collins
• [asterisk-users] Re: Could two Asterisk servers connect through VPN   Benny Amorsen
• [asterisk-users] Re: h323 problem with asterisk 1.2.18   nik600
• [asterisk-users] Sound files   Pedro Silva
• [asterisk-users] random sections lost from call recording   Don Fletcher
• [asterisk-users] MYSQL Query --> PAGE   Forrest Beck
• [asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 42   stewart at owd.com
• [asterisk-users] Nagios/Cacti Plugin   Diego Quintana Cruz
• [asterisk-users] Sound files   James FitzGibbon
• [asterisk-users] Vista compatibilty in SIP softphones   Chris Bagnall
• [asterisk-users] MYSQL Query --> PAGE   Remco Post
• [asterisk-users] Vista compatibilty in SIP softphones   Jason Parker
• [asterisk-users] Sangoma A101 on Freebsd 6.2   Alex Feldman
• [asterisk-users] Problem when PABX call to Asterisk by Unicall   Everton Goularth
• [asterisk-users] MYSQL Query --> PAGE   Forrest Beck
• [asterisk-users] voip-info.org mirrors?   Stephen Bosch
• [asterisk-users] Ericsson dialog 4187   Jose Limeres
• [asterisk-users] Problems witch SPA3102.   Jonson Player
• [asterisk-users] Re: Could two Asterisk servers connect through VPN   Jonson Player
• [asterisk-users] Problem with the loading of the cards in Debian   MCelo
• [asterisk-users] Ringing Volume   Eric "ManxPower" Wieling
• [asterisk-users] Ringing Volume   Jadrien Wauthier
• [asterisk-users] LDAPget or something else?   Klaverstyn, David C
• [asterisk-users] asterisk 1.2 and UDP packet numbering on bridged channels (for jitter buffering)?   Damon Estep
• [asterisk-users] MYSQL Query --> PAGE   Lee Jenkins
• [asterisk-users] MYSQL Query --> PAGE   Lee Jenkins
• [asterisk-users] SIP peer / Maximum retries exceeded on transmission   chris at cgb1911.mine.nu
• [asterisk-users] asterisk 1.2 and UDP packet numbering on bridged channels (for jitter buffering)?   Andres
• [asterisk-users] Aastra phones?   mgraves at mstvp.com
• [asterisk-users] Ringing Volume   Jadrien Wauthier
• [asterisk-users] Vista compatibilty in SIP softphones   Deepak Naidu
• [asterisk-users] Re: Call waiting tone   Yehavi Bourvine +972-8-9489444
• [asterisk-users] SIP peer / Maximum retries exceeded on transmission   0xception
• [asterisk-users] Responding to SIP OPTIONS   Chris Bennett
• [asterisk-users] asterisk with festival facing problem   Cheikhou DIAW
• [asterisk-users] LDAPget or something else?   Matthias Fechner
• [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.   Gavin Henry
• [asterisk-users] Bug no. 8680 (billsec is 0 even when the call is answered) in Asterisk 1.4.2   Roi Stork
• [asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?   Zeeshan Zakaria
• [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.   f6hqz-m at hamwlan.net
• [asterisk-users] Audio going blank for a few seconds and then comesback. What could be the reason?   f6hqz-m at hamwlan.net
• [asterisk-users] Ringing Volume   Bob Chiodini
• [asterisk-users] The purpose of DUNDi   Ronaldo
• [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.   Gavin Henry
• [asterisk-users] app_txfax, app_rxfax   Craig Guy
• [Asterisk-Users] Microsoft CRM & Asterisk   Frank Bobbio
• [asterisk-users] HPEC audio clipping   Olivier
• [asterisk-users] The purpose of DUNDi   Bruce Reeves
• [asterisk-users] The purpose of DUNDi   Bruce Reeves
• [asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 45   stewart at owd.com
• [asterisk-users] The purpose of DUNDi   Olivier
• [asterisk-users] The purpose of DUNDi   Alex Robar
• [asterisk-users] Send SIP Re-invite.   Olle E Johansson
• [asterisk-users] Replaces header   Steve Blair
• [asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?   randulo
• [asterisk-users] The purpose of DUNDi   Bruce Reeves
• [asterisk-users] SIP peer / Maximum retries exceeded on transmission   Olle E Johansson
• [asterisk-users] The purpose of DUNDi   Bruce Reeves
• [asterisk-users] Problem when PABX call to Asterisk by Unicall   Everton Goularth
• [asterisk-users] asterisk 1.2 and UDP packet numbering on bridgedchannels (for jitter buffering)?   Damon Estep
• [asterisk-users] Re: RE: Digital Phones   bilal ghayyad
• [asterisk-users] The 'h' extension problem   Rizwan Hisham
• [asterisk-users] fax receiving   Josu Lazkano Lete
• [asterisk-users] Send SIP Re-invite.   Rohan Hathiwala
• [asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?   Eric "ManxPower" Wieling
• [asterisk-users] The 'h' extension problem   Eric "ManxPower" Wieling
• [asterisk-users] Problems witch SPA3102.   Drew Gibson
• [asterisk-users] select menu   Josu Lazkano Lete
• [asterisk-users] Re: RE: Digital Phones   Steven
• [asterisk-users] using voip software client as public address system. Low volume   Antonio Almodóvar
• [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.   Stelios Koroneos
• [asterisk-users] Double DTMF digits   Remi Quezada
• [asterisk-users] Double DTMF digits   Remi Quezada
• [asterisk-users] The 'h' extension problem   Rafael Rodrigo - NSI
• [asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 46   stewart at owd.com
• [asterisk-users] select menu   franco escalona
• RES: [asterisk-users] select menu   Moacir O. de Souza Junior - Personalsoft Sistemas Ltda.
• [asterisk-users] asterisk 1.2 and UDP packet numbering on bridgedchannels (for jitter buffering)?   Andres
• [asterisk-users] fax receiving   Steve Davies
• [asterisk-users] SIP Problems continue...   Ken Williams
• [asterisk-users] SIP Problems continue...   mail-lists
• [asterisk-users] select menu   franco escalona
• [asterisk-users] SIP Problems continue...   franco escalona
• [asterisk-users] SIP Problems continue...   Ken Williams
• [asterisk-users] asterisk with festival facing problem   Lee Jenkins
• [asterisk-users] additional volume added to sound on CONSOLE/dsp   Jerry Geis
• [asterisk-users] select menu   Adam Moffett
• [asterisk-users] Question about Asterisk 1.4 depoyment.   Vietnhi Phuvan
• [asterisk-users] HPEC audio clipping   Matthew Fredrickson
• [asterisk-users] Re: RE: Digital Phones   Eric "ManxPower" Wieling
• [asterisk-users] SIP Problems continue...   Ken Williams
• [asterisk-users] List of telemarketers??   Ritesh Agrawal
• [asterisk-users] Mobile Number to Mobile carrier mapping   Ritesh Agrawal
• [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.   Gavin Henry
• [asterisk-users] SIP Problems continue...   Adam Moffett
• [asterisk-users] List of telemarketers??   Yossi Ben Hagai
• [asterisk-users] Re: RE: Digital Phones   Noah Miller
• [asterisk-users] Re: RE: Digital Phones   Robert Augustyn
• [asterisk-users] Boost Polycom IP601 headset volume   Alvin Austin
• [asterisk-users] Mobile Number to Mobile carrier mapping   Adam Moffett
• [asterisk-users] List of telemarketers??   Noah Miller
• [asterisk-users] app_txfax, app_rxfax   Paradise Dove
• [asterisk-users] List of telemarketers??   J. Oquendo
• [asterisk-users] SIP Problems continue...   Eric "ManxPower" Wieling
• [asterisk-users] SIP Problems continue...   Ken Williams
• [asterisk-users] Could two Asterisk servers connect through VPN   Tielin Xu
• [asterisk-users] Question about Asterisk 1.4 depoyment.   Remco Post
• [asterisk-users] Mobile Number to Mobile carrier mapping   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk to record CDR in DB Oracle   Everton Goularth
• [asterisk-users] Mobile Number to Mobile carrier mapping   Erik Anderson
• [asterisk-users] Mobile Number to Mobile carrier mapping   George Pajari
• [asterisk-users] Re: Mobile Number to Mobile carrier mapping   Steven
• [asterisk-users] 10 FXS - Channel Bank or PCI Card?   Gavin Henry
• [asterisk-users] List of telemarketers??   J. Oquendo
• [asterisk-users] Ericcson analog phone   Jose Limeres
• [asterisk-users] 10 FXS - Channel Bank or PCI Card?   William Moore
• [asterisk-users] List of telemarketers??   Ritesh Agrawal
• [asterisk-users] 10 FXS - Channel Bank or PCI Card?   Cory Andrews
• [asterisk-users] 10 FXS - Channel Bank or PCI Card?   Robert Hajime Lanning
• [asterisk-users] SPA841 3.1.1(a) firmware file   Nabeel Jafferali
• [asterisk-users] 10 FXS - Channel Bank or PCI Card?   cb
• [asterisk-users] Is there a possiblity to check in the dialplan whether a SIP user is registred?   Doug Garstang
• [asterisk-users] MINNESOTA: Twin Cities Asterisk Users Group - Saturday May 12th 2007 - 11:30am   Don Kelly
• [asterisk-users] Re: RE: Digital Phones   Stephen Bosch
• [asterisk-users] The 'h' extension problem   Stephen Bosch
• [asterisk-users] Boost Polycom IP601 headset volume   Stephen Bosch
• [asterisk-users] Re: RE: Digital Phones   Robert Augustyn
• [asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?   Zeeshan Zakaria
• [asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?   Zeeshan Zakaria
• [asterisk-users] Is there a possiblity to check in the dialplan whether a SIP user is registred?   kalle at esprITschmiede.de
• [asterisk-users] Re: RE: Digital Phones   Andrew Kohlsmith
• [asterisk-users] Re: RE: Digital Phones   Stephen Bosch
• [asterisk-users] The purpose of DUNDi   Rilawich Ango
• [asterisk-users] Re: RE: Digital Phones   Stephen Bosch
• [asterisk-users] Trixbox drops call after running AGI script   Allan Dalton
• [asterisk-users] SIP Problems continue...   Deepak Naidu
• [asterisk-users] Responding to SIP OPTIONS   Chris Bennett
• [asterisk-users] SIP peer / Maximum retries exceeded on transmission   chris at cgb1911.mine.nu
• [asterisk-users] The purpose of DUNDi   Remco Post
• [asterisk-users] 10 FXS - Channel Bank or PCI Card?   Gavin Henry
• [asterisk-users] 10 FXS - Channel Bank or PCI Card?   Gavin Henry
• [asterisk-users] 10 FXS - Channel Bank or PCI Card?   Gavin Henry
• [asterisk-users] ivr and internal voip phones   Thomas Stein
• [asterisk-users] AT530 Telephone   Josu Lazkano Lete
• [asterisk-users] Asterisk 1.4 and AgentMonitorOutgoing   Roi Stork
• [asterisk-users] AT530 Telephone   Alexandre VERNIOL
• [asterisk-users] AT530 Telephone   Josu Lazkano Lete
• [asterisk-users] AT530 Telephone   Alexandre VERNIOL
• [asterisk-users] AT530 Telephone   Josu Lazkano Lete
• [asterisk-users] module zttranscode: what is it?   Giorgio Incantalupo
• [asterisk-users] iax to iax Reject Connection   chawki hammoud
• [asterisk-users] TDM410P   Alexandre VERNIOL
• [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN - Asterisk 1.4   Administrator TOOTAI
• [asterisk-users] Aastra phones?   Christopher Aloi
• [asterisk-users] Anyone Want To Trade a TE405P for a TE410P?   Steve Totaro
• [asterisk-users] Asterisk To GoogleTalk Audio Issues   demuel at thephinix.org
• [asterisk-users] Ericcson analog phone   Steve Totaro
• [asterisk-users] Re: "you have been kicked my this conference"   Tony Mountifield
• [asterisk-users] List of telemarketers??   Steve Totaro
• [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN - Asterisk1.4   Steve Totaro
• [asterisk-users] asterisk with festival facing problem   Steve Totaro
• [asterisk-users] List of telemarketers??   lenz
• [asterisk-users] iax to iax Reject Connection   Steve Totaro
• [asterisk-users] Unbridge the two call.   pandi ponnangan
• [asterisk-users] List of telemarketers??   lenz
• [asterisk-users] List of telemarketers??   Dave Bour
• [asterisk-users] Anyone tried using a SCCP service provider>   Mark Phillips
• [asterisk-users] List of telemarketers??   Tim Koehler
• [asterisk-users] SIP Problems continue...   Ken Williams
• [asterisk-users] Re: RE: Digital Phones   bilal ghayyad
• [asterisk-users] Re: RE: Digital Phones   Tom Lynn
• [asterisk-users] ODBC   Lee Jenkins
• [asterisk-users] Re: zttranscode crashes server   Tony Mountifield
• [asterisk-users] The purpose of DUNDi   Rilawich Ango
• [asterisk-users] call transfer to asterik.. asterisk as an end point   Zahid Mehmood
• [asterisk-users] Re: module zttranscode: what is it?   Tony Mountifield
• [asterisk-users] Re: Outbound call through a Single Asterisk Server   Matthew Harrell
• [asterisk-users] AgentCallbackLogin not working with 1.4.4   Sanjay Rajdev
• [asterisk-users] Softkey config example for Cisco 7941/7961   Joe Pukepail
• [asterisk-users] Redirect Message Waiting Indication   Ben Brown
• [asterisk-users] ivr and internal voip phones   C F
• [asterisk-users] Re: module zttranscode: what is it?   Drew Gibson
• [asterisk-users] CITEL gateway does it work well?   Robert Augustyn
• [asterisk-users] Re: module zttranscode: what is it?   Tony Mountifield
• [asterisk-users] CITEL gateway does it work well?   Alex Balashov
• [asterisk-users] Re: RE: Digital Phones   Stephen Bosch
• [asterisk-users] CITEL gateway does it work well?   FUERMANN, JASON BRYCE
• [asterisk-users] force outgoinc callerid   nik600
• [asterisk-users] Polycom headset button blinking   James Fromm
• [asterisk-users] Polycom headset button blinking   Bruce Reeves
• [asterisk-users] Log CODECS in CDR's   Morgan Gilroy
• [asterisk-users] force outgoinc callerid   Steve Kennedy
• [asterisk-users] Polycom power over ethernet (PoE) cables for 500/501, 600/601 and 650 sets   Stephen Bosch
• [asterisk-users] Responding to SIP OPTIONS   Olle E Johansson
• [asterisk-users] The purpose of DUNDi   Remco Post
• [asterisk-users] Replaces header   Olle E Johansson
• [asterisk-users] Vista compatibilty in SIP softphones   Rob Townley
• [asterisk-users] SIP Problems continue...   Olle E Johansson
• [asterisk-users] CITEL gateway does it work well?   Olle E Johansson
• [asterisk-users] asterisk SIP domain (in LAN or DMZ)?   Joao Pereira
• [asterisk-users] Asttapi Collect   Faisal Ashraf
• [asterisk-users] Redirecting an existing channel?   Lee Jenkins
• [asterisk-users] ices low volume   Jon-o Addleman
• [asterisk-users] Re: RE: Digital Phones   shadowym
• [asterisk-users] The downside of Asterisk and least cost routing...   Francesco Peeters (Asterisk)
• [asterisk-users] socket_process: Received mini frame before first full voice frame   Sanjay Rajdev
• [asterisk-users] Iaxy clicking   Matthew Yingling
• [asterisk-users] SNOM 360 Rejecting Calls   Dovid B
• [asterisk-users] Correct setup for directing already ringing calls to newly available phones   asterisk-users at rogg.is
• [asterisk-users] Iaxy clicking   Sanjay Rajdev
• [asterisk-users] The downside of Asterisk and least cost routing...   Gordon Henderson
• [asterisk-users] HPEC audio clipping   Olivier
• [asterisk-users] The downside of Asterisk and least cost routing...   Francesco Peeters (Asterisk)
• [asterisk-users] HPEC audio clipping   Noah Miller
• [asterisk-users] Re: RE: Digital Phones   Hans Witvliet
• [asterisk-users] SIP Problems continue...   Ken Williams
• [asterisk-users] SNOM 360 Rejecting Calls   Philipp Kempgen
• [asterisk-users] Re: RE: Digital Phones   Philipp Kempgen
• [asterisk-users] Redirecting an existing channel?   Lee Jenkins
• [asterisk-users] SNOM 360 Rejecting Calls   Dovid B
• [asterisk-users] Unbridge the two call.   Moises Silva
• [asterisk-users] TDM400P usada?   Julio Tejera
• [asterisk-users] Log CODECS in CDR's   dave cantera
• [asterisk-users] Voice mail volume   Stephen Bosch
• [asterisk-users] Any other softPBX like Asterisk?   Crazy Boy
• [asterisk-users] Any other softPBX like Asterisk?   Armin Schindler
• [asterisk-users] Any other softPBX like Asterisk?   Francesco Peeters (Asterisk)
• [asterisk-users] The downside of Asterisk and least cost routing...   Gordon Henderson
• [asterisk-users] Any other softPBX like Asterisk?   Stephen Bosch
• [asterisk-users] record voice   Josu Lazkano Lete
• [asterisk-users] The downside of Asterisk and least cost routing...   Chris Bagnall
• [asterisk-users] record voice   Gordon Henderson
• [asterisk-users] Polycom power over ethernet (PoE) cables for 500/501, 600/601 and 650 sets   John Marvin
• [asterisk-users] record voice   Josu Lazkano Lete
• [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN - Asterisk1.4   Administrator TOOTAI
• [asterisk-users] Reminder: Asterisk Users Conference Friday 12:30PM EDT   randulo
• [asterisk-users] Log CODECS in CDR's   Morgan Gilroy
• [asterisk-users] Any other softPBX like Asterisk?   Roberto Pereyra
• [asterisk-users] Dundi and unknown remote peers   Asterisk
• [asterisk-users] Rapid DTMF missing digits   Bryan Laird
• [asterisk-users] A couple of questions for the Mitel gurus (phone-related - not systems)   Nigel Kendrick
• [asterisk-users] Any other softPBX like Asterisk?   Luca Corti
• [asterisk-users] 'Invalid characters in name' with asterisk-gui   Tom Lobato
• [asterisk-users] TDM410P   Nitesh Divecha
• [asterisk-users] Need a RTP/SIP Proxy to be used as SBC (Session Border Controller)   Jean-Marc Salsa
• [asterisk-users] Confference function   gc
• [asterisk-users] TDM410P   Alexandre VERNIOL
• [asterisk-users] SIP Problems continue...   Ken Williams
• [asterisk-users] Module wctdm24xxp not found - TDM808P on debian   Juliano Fernandes Schroeder
• [asterisk-users] Need some help with a very simple Queestion..   Gavin Spurgeon
• [asterisk-users] Log CODECS in CDR's   James FitzGibbon
• [asterisk-users] Module wctdm24xxp not found - TDM808P on debian   William Moore
• [asterisk-users] Strange problem with asterisk   Vitaly Oborsky
• [asterisk-users] Need some help with a very simple Queestion..   Gordon Henderson
• [asterisk-users] Module wctdm24xxp not found - TDM808P on debian   Juliano Fernandes Schroeder
• [asterisk-users] Need some help with a very simple Queestion..   Alex Robar
• [asterisk-users] Dealing with 2 SIP providers   Mike
• [asterisk-users] Need