Mail Index

• Thread Index

• Re: [asterisk-dev] CDR and daylight savings time
  • From: Hans Petter Selasky
• [asterisk-users] Thomson ST2020 and voicemail
  • From: Dante Dante
• Re: [asterisk-dev] CDR and daylight savings time
  • From: Tilghman Lesher
• Re: [asterisk-dev] Detecting DMTF tone prolonged key press
  • From: Tilghman Lesher
• Re: [asterisk-users] X100P "rings" randomly when "phone" line makes call
  • From: Zoilo Gomez
• Re: [asterisk-dev] Cross-compiling for embedded platforms using openWRT
  • From: Kevin P. Fleming
• Re: [asterisk-dev] CDR and daylight savings time
  • From: Hans Petter Selasky
• [asterisk-users] Help needed with Polycom dialplan pattern matching
  • From: John French
• Re: [asterisk-users] Help needed with Polycom dialplan pattern matching
  • From: Doug Lytle
• Re: [asterisk-users] Happy 2007!!!
  • From: Lee Jenkins
• Re: RE : [asterisk-users] Happy 2007!!!
  • From: Bill Hackensack
• Re: RE : [asterisk-users] Happy 2007!!!
  • From: Dovid B
• Re: RE : [asterisk-users] Happy 2007!!!
  • From: Tom Lynn
• Re: [asterisk-dev] Cross-compiling for embedded platforms using openWRT
  • From: Brian Capouch
• Re: [asterisk-users] How to connect two asterisk server
  • From: sunil@xxxxxxxxxxxxxx
• Re: [asterisk-users] How to connect two asterisk server
  • From: Noah Miller
• Re: [asterisk-users] How to connect two asterisk server
  • From: Noah Miller
• Re: [asterisk-users] (OT) Where to post free source for AGI?
  • From: Lee Jenkins
• Re: [asterisk-dev] Cross-compiling for embedded platforms using openWRT
  • From: Kevin P. Fleming
• Re: [asterisk-dev] Cross-compiling for embedded platforms using openWRT
  • From: Kevin P. Fleming
• [asterisk-users] Re: Hi reg. 2 asterisk server
  • From: Thirumal Saminathan
• [asterisk-dev] Communication between two asterisk server
  • From: Dhiraj Kumar
• RE: [asterisk-users] Dialed Number missing from the CDR when usingcallfiles.
  • From: Michael Collins
• [asterisk-users] Problem with centos 4.4 and jabber/gtalk (really iksemel)
  • From: Kenneth Padgett
• Re: [asterisk-users] Dual Ringing Tones
  • From: Florian Overkamp
• [asterisk-dev] How can we continue to the next priority after calling Pickup() ?
  • From: Dung Nguyen Anh
• [asterisk-users] chan_oh323 early media
  • From: Jason Kim
• Re: [asterisk-dev] SIP asterisk proprietary extensions
  • From: Anton
• [asterisk-users] asterisk and mysql
  • From: RdBSD
• Re: [asterisk-users] Realtime multiple registration for a Hard Phone Snom 360
  • From: Olivier
• Re: [asterisk-users] asterisk and mysql
  • From: Ngo Duc Loi
• Re: [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330
  • From: Olivier
• [asterisk-users] Buying
  • From: Khaled
• [asterisk-dev] 1.4.0 SIP failing sporadically on MIPSEL, too
  • From: Brian Capouch
• Re: [asterisk-dev] SIP asterisk proprietary extensions
  • From: Johansson Olle E
• Re: [asterisk-dev] SIP asterisk proprietary extensions
  • From: Anton
• Re: [Asterisk-Users] asterisk + door opener
  • From: Terry Wade
• Re: [asterisk-dev] SIP asterisk proprietary extensions
  • From: Johansson Olle E
• Re: [asterisk-users] Problem with centos 4.4 and jabber/gtalk (really iksemel)
  • From: Bob Chiodini
• [asterisk-users] Avoiding deadlock-line drop problem
  • From: Giannis Margaritis
• [asterisk-users] Best Hardware for Asterisk Server?
  • From: Mark Greene
• [asterisk-users] Save SIP DEBUG output to a file
  • From: Frederico Madeira
• [asterisk-dev] Re: [asterisk-commits] oej: trunk r49152 - in /trunk: ./ configs/features.conf.sample
  • From: Kevin P. Fleming
• Re: [asterisk-dev] Re: [asterisk-commits] oej: trunk r49152 - in /trunk: ./ configs/features.conf.sample
  • From: Johansson Olle E
• Re: [asterisk-users] PRI ANI/CallerID
  • From: Jerry Jones
• Re: [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330
  • From: Jorge Mendoza
• RE: [asterisk-users] asterisk and mysql
  • From: Savoy, Kevin - Williston, ND
• RE: [asterisk-users] (OT) Where to post free source for AGI?
  • From: Ejay Hire
• [asterisk-users] [asterisk-biz] Slightly updated UK English voice prompts
  • From: Steve Kennedy
• [asterisk-users] 802.1x support in wired sip hardphones ?
  • From: Olivier
• Re: [asterisk-users] (OT) Where to post free source for AGI?
  • From: Bruce Reeves
• Re: [asterisk-users] (OT) Where to post free source for AGI?
  • From: Lenz
• Re: [asterisk-users] 802.1x support in wired sip hardphones ?
  • From: richard Coco
• Re: [asterisk-users] (OT) Where to post free source for AGI?
  • From: Tzafrir Cohen
• Re: [asterisk-users] 802.1x support in wired sip hardphones ?
  • From: Olivier
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: chester c young
• Re: [asterisk-users] 802.1x support in wired sip hardphones ?
  • From: richard Coco
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Gordon Henderson
• Re: [asterisk-dev] Re: [asterisk-commits] oej: trunk r49152 - in /trunk: ./ configs/features.conf.sample
  • From: Kevin P. Fleming
• [asterisk-users] How to show a debugging remark in a sip or extensions context?
  • From: Larry Alkoff
• [asterisk-users] (OT) Where to post free source for AGI?
  • From: Michael Collins
• Re: [asterisk-users] How to connect two asterisk server
  • From: Dave Schardin
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Mark Greene
• Re: [asterisk-users] chan_oh323 early media
  • From: Mark Greene
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: joe a.
• Re: [asterisk-users] (OT) Where to post free source for AGI?
  • From: Lee Jenkins
• [asterisk-dev] Asterisk 1.4.0 Segfault
  • From: Anton
• [asterisk-users] SpanDSP and Asterisk 1.4
  • From: Mark Johnson
• RE: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Michael Collins
• Re: [asterisk-users] How to connect two asterisk server
  • From: sunil@xxxxxxxxxxxxxx
• [asterisk-users] yet another faxing issue (outbound only, via ATA)
  • From: Bill Gibbs
• [asterisk-users] Re: Hi reg. 2 asterisk server
  • From: Noah Miller
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Doug
• [asterisk-users] Re: Grandstream GXW-4108 8 port FXO
  • From: Martin Joseph
• Re: [asterisk-dev] Asterisk 1.4.0 Segfault
  • From: Kevin P. Fleming
• [asterisk-users] Call connected, cannot hear or speak - $20 for fix
  • From: zero massive
• Re: [asterisk-users] Problem with centos 4.4 and jabber/gtalk (really iksemel)
  • From: Kenneth Padgett
• RE: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Colin Anderson
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Mark Greene
• [asterisk-dev] Re: [svn-commits] kpfleming: branch 1.4 r49165 - /branches/1.4/channels/chan_zap.c
  • From: Russell Bryant
• Re: [asterisk-users] (OT) Where to post free source for AGI?
  • From: Lee Jenkins
• Re: [asterisk-users] (OT) Where to post free source for AGI?
  • From: Jason Parker
• [asterisk-users] RE: yet another faxing issue (outbound only, via ATA)
  • From: Bill Gibbs
• Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
  • From: Joao Pereira
• [asterisk-users] queues - limiting ringing calls to queue members
  • From: Nikola Ciprich
• RE: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Gordon Henderson
• Re: [asterisk-users] vzaphfc?
  • From: Remco Barendse
• [asterisk-users] OT: Admin manual for Linksys Sipura SPA-2102
  • From: Erick Perez
• Re: [asterisk-dev] Re: [svn-commits] kpfleming: branch 1.4 r49165 - /branches/1.4/channels/chan_zap.c
  • From: Kevin P. Fleming
• [asterisk-users] OnHook Call Announcement...
  • From: Carlos Chavez
• [asterisk-users] extension problems
  • From: Vulpes Velox
• Re: [asterisk-users] queues - limiting ringing calls to queue members
  • From: Ex Vitorino
• [asterisk-users] Double quotes in CDRUserField?
  • From: Michael Collins
• Re: [asterisk-users] vzaphfc?
  • From: Julian J. M.
• Re: [asterisk-users] extension problems
  • From: Mike
• Re: [asterisk-users] OnHook Call Announcement...
  • From: C F
• Re: [asterisk-users] RE: yet another faxing issue (outbound only, via ATA)
  • From: Lacy Moore - Aspendora
• RE: [asterisk-users] RE: yet another faxing issue (outbound only, via ATA)
  • From: Bill Gibbs
• Re: [asterisk-users] Double quotes in CDRUserField?
  • From: Trevor Peirce
• [asterisk-dev] Academic Asterisk Adventure
  • From: Shawn Van Every
• [asterisk-users] Re: [A*UG] How to show a debugging remark in a sip or extensions context?
  • From: Larry Alkoff
• [asterisk-users] connecting asterisk (trixbox) to traditional phone lines?
  • From: blackwater dev
• Re: [asterisk-users] OnHook Call Announcement...
  • From: Yuan LIU
• Re: [asterisk-users] Error compiling chan_vpb
  • From: Kevin P. Fleming
• Re: [asterisk-users] connecting asterisk (trixbox) to traditional phone lines?
  • From: cb
• RE: [asterisk-users] connecting asterisk (trixbox) to traditional phonelines?
  • From: Yuan LIU
• Re: [asterisk-users] connecting asterisk (trixbox) to traditional phone lines?
  • From: Todd H
• [asterisk-dev] Planning for AstriDevCon USA 2007
  • From: Kevin P. Fleming
• Re: [asterisk-users] Buying
  • From: Paul Hales
• Re: [asterisk-users] connecting asterisk (trixbox) to traditional phone lines?
  • From: Alex Robar
• Re: [asterisk-dev] Academic Asterisk Adventure
  • From: Tzafrir Cohen
• Re: [asterisk-users] vzaphfc?
  • From: Tzafrir Cohen
• Re: [asterisk-users] connecting asterisk (trixbox) to traditional phone lines?
  • From: Tzafrir Cohen
• RE: [asterisk-users] Double quotes in CDRUserField?
  • From: Michael Collins
• Re: [asterisk-users] vzaphfc?
  • From: Remco Barendse
• RE: [asterisk-users] Dialed Number missing from the CDR when using callfiles.
  • From: Michael Collins
• Re: [asterisk-users] connecting asterisk (trixbox) to traditional phone lines?
  • From: Doug Crompton
• [asterisk-dev] 1.4.0 SIP weirdness: possible clue?
  • From: Brian Capouch
• Re: [asterisk-users] vzaphfc?
  • From: Tzafrir Cohen
• [asterisk-users] Re: [A*UG] How to show a debugging remark in a sip or extensions context?
  • From: Larry Alkoff
• Re: [asterisk-dev] 1.4.0 SIP weirdness: possible clue?
  • From: Brian Capouch
• [asterisk-users] SNOM loses server registration
  • From: Joao Pereira
• [asterisk-users] Dubai Caller ID
  • From: Mihaly Antal
• Re: [asterisk-users] yet another faxing issue (outbound only, via ATA)
  • From: Thomas Kenyon
• Re: [asterisk-users] yet another faxing issue (outbound only, via ATA)
  • From: Marco Mouta
• Re: [asterisk-users] SNOM loses server registration
  • From: Marco Mouta
• [asterisk-users] Block some number outgoing from joust one extention
  • From: Mattias Andersson
• [asterisk-users] ISA server Issue (Maybe off topic)
  • From: Mattias Andersson
• Fwd: [asterisk-users] Disconnect supervision in India?
  • From: Rajkumar S
• [asterisk-users] voice fax modem and asterisk
  • From: Gregory Machin
• [asterisk-users] native music on hold distortion between files
  • From: Damon Estep
• [asterisk-users] Sangoma Remora A202
  • From: Todd H
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Derek Whitten
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Derek Whitten
• Re: [asterisk-users] Sangoma Remora A202
  • From: Tom
• Re: [asterisk-users] Sangoma Remora A202
  • From: Time Bandit
• Re: [asterisk-users] Sangoma Remora A202
  • From: Rob Schall
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Steve Edwards
• [asterisk-users] MeetMe() not recording calls
  • From: John French
• [asterisk-dev] Asterisk and Cisco AS5350 DTMF Problem
  • From: Juan Carlos Jaramillo B.
• [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo /audio artifacts
  • From: Colin Anderson
• [asterisk-users] Fonebridge2
  • From: Jon Schøpzinsky
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: Derek Whitten
• [asterisk-users] Voicemail to email
  • From: Mark Greene
• Re: [asterisk-users] Best Hardware for Asterisk Server?
  • From: joe a.
• RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent
  • From: Colin Anderson
• RE: [BULK] [asterisk-users] Fonebridge2
  • From: Savoy, Kevin - Williston, ND
• RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts
  • From: Michael L. Young
• Re: [asterisk-users] Sangoma Remora A202
  • From: Bruce Reeves
• Re: [asterisk-users] Block some number outgoing from joust one extention
  • From: Marco Mouta
• Re: [asterisk-users] Voicemail to email
  • From: Doug Crompton
• RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent
  • From: Colin Anderson
• RE: [asterisk-dev] PROPOSAL: new manager scope called "reporting"
  • From: Jonathan k. Creasy
• [asterisk-users] answer machine detection
  • From: Julian Lyndon-Smith
• Re: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo /audio artifacts
  • From: Eric \"ManxPower\" Wieling
• [asterisk-users] SIP Dial out timeout
  • From: Arik Raffael Funke
• Re: [asterisk-users] SIP Dial out timeout
  • From: Eric \"ManxPower\" Wieling
• [asterisk-users] Polycom Power Specs
  • From: Peder @ NetworkOblivion
• [asterisk-users] API: how to bridge originated call?
  • From: chester c young
• Re: [asterisk-users] Polycom Power Specs
  • From: Dave Schardin
• Re: [asterisk-users] Polycom Power Specs
  • From: Michael Welter
• [asterisk-users] Is chan_zap.so loaded?
  • From: John French
• Re: [asterisk-users] Polycom Power Specs
  • From: Alvin Austin
• Re: [asterisk-users] Is chan_zap.so loaded?
  • From: Eric \"ManxPower\" Wieling
• [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
  • From: Dan Austin
• [asterisk-users] Park and Page
  • From: Mike Clark
• Re: [asterisk-users] Is chan_zap.so loaded?
  • From: Yuan LIU
• Re: [asterisk-users] Sangoma Remora A202
  • From: Pierre Marceau
• Re: [asterisk-users] API: how to bridge originated call?
  • From: Moises Silva
• [asterisk-users] Cisco 79x1 Auto-Answer
  • From: Jeremiah Millay
• [asterisk-users] Error on answer a SIP 401 message
  • From: Frederico Madeira
• Re: [asterisk-users] API: how to bridge originated call?
  • From: Moises Silva
• Re: [asterisk-users] Fonebridge2
  • From: Bill Burdick
• [asterisk-users] have a phone number from stanaphone and a working trixbox, how do I connect them?
  • From: blackwater dev
• [asterisk-dev] Support for Agent channels in Bridge manager and dial plan patch
  • From: Moises Silva
• [asterisk-users] over 200 queues, anyone?
  • From: lenz
• Re: [asterisk-users] have a phone number from stanaphone and a working trixbox, how do I connect them?
  • From: Alex Robar
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Joe Dennick
• RE: [asterisk-users] have a phone number from stanaphone and a workingtrixbox, h
  • From: Yuan LIU
• Re: [asterisk-users] Polycom Power Specs
  • From: LST
• [asterisk-dev] pri show span
  • From: Jonathan k. Creasy
• Re: [asterisk-users] Polycom Power Specs
  • From: Bas van der Veen
• [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
  • From: Anton Krall
• Re: [asterisk-users] Polycom Power Specs
  • From: Bas van der Veen
• RE: [asterisk-users] yet another faxing issue (outbound only, via ATA)
  • From: Bill Gibbs
• Re: [asterisk-dev] pri show span
  • From: Andrew Latham
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Alex Robar
• Re: [asterisk-users] Voicemail to email
  • From: Forrest Beck
• Re: [asterisk-dev] pri show span
  • From: Steven Critchfield
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Joe Dennick
• Re: [asterisk-dev] pri show span
  • From: Eric \"ManxPower\" Wieling
• Re: [asterisk-users] Voicemail to email
  • From: Mark Greene
• [asterisk-users] ARI help
  • From: Mark Greene
• [asterisk-users] Gentoo ebuild for 1.4?
  • From: Chris Bagnall
• Re: [asterisk-users] Problem with centos 4.4 and jabber/gtalk (really iksemel)
  • From: Bob Chiodini
• RE: [asterisk-dev] pri show span
  • From: Jonathan k. Creasy
• [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
  • From: Steven
• Re: [asterisk-dev] pri show span
  • From: Tzafrir Cohen
• Re: [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
  • From: Guillermo Salas M.
• Re: [asterisk-users] Sangoma Remora A202
  • From: Todd H
• [asterisk-users] Detect IP path before calling
  • From: Yuan LIU
• [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
  • From: Douglas Garstang
• RE: [asterisk-users] voice fax modem and asterisk
  • From: Yuan LIU
• Re: [asterisk-users] Sangoma Remora A202
  • From: Time Bandit
• RE: [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
  • From: Dan Austin
• [asterisk-users] TDM400 & UK Caller ID problems ...
  • From: Gordon Henderson
• [asterisk-users] Any quiet 24 port POE switches out there?
  • From: John French
• Re: [asterisk-users] Detect IP path before calling
  • From: Paul Hales
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Richard Lyman
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: Brian Roy
• Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
  • From: Moises Silva
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: Jerry Jones
• Re: [asterisk-users] Polycom Power Specs
  • From: Chris Mason (Lists)
• Re: [asterisk-users] Block some number outgoing from joust one extention
  • From: C F
• [asterisk-users] v140 ./configure not finding installed ssl
  • From: snowcrash+asterisk
• [asterisk-users] [Announce] Web-MeetMe 3.0.0 RE-released
  • From: Dan Austin
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: Andres
• Re: [asterisk-users] Error compiling chan_vpb
  • From: DiegoF
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: Anselm Martin Hoffmeister
• [asterisk-users] 1.4 segfaulting when manager client is connected
  • From: Brad Templeton
• Re: [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
  • From: Rob Fugina
• Re: [asterisk-users] API: how to bridge originated call?
  • From: chester c young
• [asterisk-users] ztdummy on 1.6
  • From: chester c young
• RE: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
  • From: Dan Austin
• [asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 4
  • From: Edwin Groothuis
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: Angel Heart
• Re: [asterisk-users] API: how to bridge originated call?
  • From: Moises Silva
• Re: [asterisk-users] API: how to bridge originated call?
  • From: chester c young
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: David Gomillion
• Re: [asterisk-users] Detect IP path before calling
  • From: Yuan LIU
• RE: [asterisk-users] Detect IP path before calling
  • From: Bill Gibbs
• Re: [asterisk-users] Detect IP path before calling
  • From: Eric \"ManxPower\" Wieling
• RE: [asterisk-users] Detect IP path before calling
  • From: Yuan LIU
• [asterisk-users] caller id ring tones for Asterisk Phone
  • From: Jeronimo Romero
• Re: [asterisk-users] Detect IP path before calling
  • From: Paul Hales
• Re: [asterisk-users] caller id ring tones for Asterisk Phone
  • From: Brad Templeton
• Re: [asterisk-users] caller id ring tones for Asterisk Phone
  • From: James Andrewartha
• [asterisk-dev] Asterisk zap channels death
  • From: Anton
• [asterisk-users] asterisk sip peer/user matching methods for authentication backwards?
  • From: Damon Estep
• Re: [asterisk-users] ztdummy on 1.6
  • From: Jason Parker
• RE: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: shadowym
• RE: [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
  • From: Douglas Garstang
• [asterisk-dev] asterisk cross compilation
  • From: Dhiraj Kumar
• Re: [asterisk-dev] Asterisk zap channels death
  • From: Edwin Groothuis
• Re: [asterisk-users] 802.1x support in wired sip hardphones ?
  • From: Olivier
• Re: [asterisk-users] Detect IP path before calling
  • From: Julian J. M.
• [asterisk-users] Required freelancer for installing hylafax on Asterisk Box
  • From: gkanuganti
• [asterisk-users] Cisco AS5300
  • From: yusuf
• Re: [asterisk-users] 802.1x support in wired sip hardphones ?
  • From: richard Coco
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Lenz
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Lenz
• Re: [asterisk-users] Block some number outgoing from joust one extention
  • From: Mattias Andersson
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Gavin Hamill
• [asterisk-users] bypass menu for certain numbers?
  • From: Matt Gibson
• [asterisk-users] Hi reg. asterisk Compilation
  • From: Thirumal Saminathan
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Leo Ann Boon
• [asterisk-dev] Thirumal Saminathan wants to chat
  • From: Thirumal Saminathan
• [asterisk-dev] Re: Hi reg. asterisk Compilation
  • From: Thirumal Saminathan
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Terry Wade
• Re: [asterisk-users] voice fax modem and asterisk
  • From: Thomas Kenyon
• Re: [asterisk-dev] Re: Hi reg. asterisk Compilation
  • From: serva
• [asterisk-users] Maybe a NAT problem
  • From: Facundo Barrera - GMail
• Re: [asterisk-users] Maybe a NAT problem
  • From: Bob Chiodini
• Re: [asterisk-users] caller id ring tones for Asterisk Phone
  • From: Bryan M. Johns
• Re: [asterisk-users] bypass menu for certain numbers?
  • From: Tzafrir Cohen
• Re: [asterisk-users] 802.1x support in wired sip hardphones ?
  • From: Olivier
• [asterisk-users] Re: SIP Dial out timeout
  • From: Arik Raffael Funke
• Re: [asterisk-users] queues - limiting ringing calls to queue members
  • From: Nikola Ciprich
• [asterisk-users] Digium Wildcard B410P
  • From: Henrik Woffinden
• Re: [asterisk-users] queues - limiting ringing calls to queue members
  • From: Todd H
• [asterisk-users] postgres and asterisk
  • From: O . Kamal
• Re: [asterisk-users] Digium Wildcard B410P
  • From: Timothy Parez
• Re: [asterisk-users] Maybe a NAT problem
  • From: Facundo Barrera - GMail
• Re: [asterisk-users] postgres and asterisk
  • From: Thomas Kenyon
• Re: [asterisk-users] over 200 queues, anyone?
  • From: lenz
• Re: [asterisk-users] ztdummy on 1.6
  • From: Thomas Kenyon
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: Walt Reed
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Olivier
• [asterisk-users] Re: Re: [Announce] Web-MeetMe 3.0.0 released
  • From: Steven
• [asterisk-users] Re: ztdummy on 1.6
  • From: Steven
• Re: [asterisk-users] voice fax modem and asterisk
  • From: Gregory Machin
• Re: [asterisk-dev] Re: Hi reg. asterisk Compilation
  • From: Moises Silva
• Re: [asterisk-users] API: how to bridge originated call?
  • From: Moises Silva
• Re: [asterisk-users] Digium Wildcard B410P
  • From: Richard Soderblom
• Re: [asterisk-users] Digium Wildcard B410P
  • From: Tzafrir Cohen
• [asterisk-users] Create a group of SIP acoount for outgoing calls ?
  • From: Noc Phibee
• [asterisk-users] Realtime voicemail passwords
  • From: Bruce Ferrell
• [asterisk-users] mISDN crypto?
  • From: Andreas Anderson
• Re: [asterisk-users] Maybe a NAT problem
  • From: Carlos Rojas
• Re: [asterisk-users] Re: ztdummy on 1.6
  • From: Jason Parker
• [asterisk-users] PRI Problems
  • From: Rob Schall
• [asterisk-users] System() and Trysystem() in extensions.conf => get the result ?
  • From: Noc Phibee
• [asterisk-users] [Fwd: PRI Problems]
  • From: Rob Schall
• Re: [asterisk-users] Digium Wildcard B410P
  • From: Matthew Fredrickson
• [asterisk-users] Re: Any quiet 24 port POE switches out there?
  • From: Allen Casteran
• Re: [asterisk-users] over 200 queues, anyone?
  • From: lists
• [asterisk-users] Re: Disconnect supervision in India?
  • From: Chris Earle
• Re: [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
  • From: Richard Lyman
• Re: [asterisk-users] [Fwd: PRI Problems]
  • From: David Gomillion
• Re: [asterisk-users] Re: Any quiet 24 port POE switches out there?
  • From: David Gomillion
• Re: [asterisk-users] Hi reg. asterisk Compilation
  • From: Noah Miller
• Re: [asterisk-dev] asterisk cross compilation
  • From: Steven
• RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent
  • From: Colin Anderson
• Re: [asterisk-users] [Fwd: PRI Problems]
  • From: Noah Miller
• [asterisk-users] Convert a file from WAV to WAV49 or GSM for Asterisk
  • From: Chris Carey
• [asterisk-users] Asterisk 1.4.0 segfault
  • From: Ondrej Valousek
• RE: [asterisk-users] asterisk sip peer/user matching methods forauthentication backwards?
  • From: Doug Meredith
• Re: [asterisk-users] [Fwd: PRI Problems]
  • From: Rob Schall
• RE: [asterisk-users] caller id ring tones for Asterisk Phone
  • From: Steve Langstaff
• RE: [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
  • From: Douglas Garstang
• [asterisk-users] #include not working in 1.4
  • From: Bruce Ferrell
• Re: [asterisk-users] bypass menu for certain numbers?
  • From: Matt Gibson
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: Noah Miller
• [asterisk-users] SIP peer lookup problems
  • From: Jon Schøpzinsky
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Rob Lith
• Re: [asterisk-users] Re: Codec swap (reinvite)
  • From: Julian J. M.
• [asterisk-users] Best inexpensive home office router for VoIP (QoS with maybe PoE)
  • From: Mike
• Re: [asterisk-users] Maybe a NAT problem
  • From: Facundo Barrera - GMail
• Re: [asterisk-users] Convert a file from WAV to WAV49 or GSM for Asterisk
  • From: Noah Miller
• RE: [asterisk-users] asterisk sip peer/user matching methodsforauthentication backwards?
  • From: Damon Estep
• RE: [asterisk-users] Re: Re: [Announce] Web-MeetMe 3.0.0 released
  • From: Dan Austin
• Re: [asterisk-users] Hi reg. asterisk Compilation
  • From: Tzafrir Cohen
• Re: [asterisk-users] #include not working in 1.4
  • From: Tzafrir Cohen
• Re: [asterisk-users] Best inexpensive home office router for VoIP (QoS with maybe PoE)
  • From: Mike
• [asterisk-users] Trouble compiling asterisk 1.2.14
  • From: Guillermo Salas M.
• [asterisk-dev] asterisk sip authentication flawed?
  • From: Damon Estep
• [asterisk-users] How big a pipe can IAX2 go?
  • From: Adrian Marsh
• [asterisk-users] Re: Alert: Steering Committee Reminder and Agenda
  • From: Flash Love
• Re: [asterisk-users] How big a pipe can IAX2 go?
  • From: Noah Miller
• Re: [asterisk-users] Trouble compiling asterisk 1.2.14
  • From: Tzafrir Cohen
• Re: [asterisk-dev] Academic Asterisk Adventure
  • From: Shawn Van Every
• Re: [asterisk-users] Cisco AS5300
  • From: Noah Miller
• [asterisk-users] proxy howto
  • From: Mark Price
• RE: [asterisk-users] Best inexpensive home office router for VoIP(QoS with maybe PoE)
  • From: Mike
• Re: [asterisk-users] postgres and asterisk
  • From: O . Kamal
• Re: [asterisk-dev] asterisk sip authentication flawed?
  • From: Kevin P. Fleming
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: Pavel Jezek
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: Drew Gibson
• Re: [asterisk-users] Sangoma Remora A202
  • From: Todd H
• RE: [asterisk-users] Best inexpensive home office router for VoIP(QoSwith maybe PoE)
  • From: Robert Augustyn
• Re: [asterisk-users] Best inexpensive home office router for VoIP (QoS with maybe PoE)
  • From: Olivier
• Re: [asterisk-users] bypass menu for certain numbers?
  • From: Anselm Martin Hoffmeister
• RE: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
  • From: Anton Krall
• Re: RE : [asterisk-users] TE110P with Qsig
  • From: Josué Conti
• Re: [asterisk-users] postgres and asterisk
  • From: Paul
• Re: [asterisk-users] Trouble compiling asterisk 1.2.14
  • From: Guillermo Salas M.
• Re: [asterisk-dev] asterisk sip authentication flawed?
  • From: Jared Smith
• Re: [asterisk-dev] Academic Asterisk Adventure
  • From: Steve Kann
• Re: [asterisk-users] [resolved] asterisk 1,4 and google talk
  • From: Ronald Lewis
• Re: [asterisk-users] How big a pipe can IAX2 go?
  • From: Zoa
• Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
  • From: Erick Perez
• Re: [asterisk-users] Any quiet 24 port POE switches out there?
  • From: Noah Miller
• Re: [asterisk-dev] Academic Asterisk Adventure
  • From: Tzafrir Cohen
• [asterisk-users] How to routing call to Quintum.
  • From: Reaz
• Re: [asterisk-users] Cisco AS5300
  • From: Andrew Pogrebennyk
• Re: [asterisk-users] bypass menu for certain numbers?
  • From: Noah Miller
• Re: [asterisk-users] voice fax modem and asterisk
  • From: Yuan LIU
• Re: [asterisk-users] no unicall on 1.4
  • From: Carlos Chavez
• Re: [asterisk-users] bypass menu for certain numbers?
  • From: Anselm Martin Hoffmeister
• [asterisk-users] Dimensioning a 50 sip phone installation
  • From: Erick Perez
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Leo Ann Boon
• Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
  • From: Moises Silva
• Re: [asterisk-users] no unicall on 1.4
  • From: Barzilai Spinak
• RE: [asterisk-users] Realtime multiple registration for a Hard PhoneSnom 360
  • From: Asterisk [Submusic]
• Re: [asterisk-users] no unicall on 1.4
  • From: Moises Silva
• Re: [asterisk-users] no unicall on 1.4
  • From: Lee Howard
• [asterisk-users] HowTO configure voice T1
  • From: Mark Greene
• Re: [asterisk-users] HowTO configure voice T1
  • From: David Gomillion
• Re: [asterisk-users] Realtime voicemail passwords
  • From: Trevor Peirce
• [asterisk-users] DISA Ring Back
  • From: Supa
• [asterisk-users] MusicOnHold Files
  • From: Forrest Beck
• Re: [asterisk-users] no unicall on 1.4
  • From: Moises Silva
• Re: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
  • From: Rob Fugina
• Re: [asterisk-users] HowTO configure voice T1
  • From: Forrest Beck
• Re: [asterisk-users] HowTO configure voice T1
  • From: Forrest Beck
• RE: [asterisk-dev] asterisk sip authentication flawed?
  • From: Damon Estep
• RE: [asterisk-users] HowTO configure voice T1
  • From: Michael Collins
• [asterisk-users] IAX vs SIP trunks between Asterisk boxes
  • From: Damon Estep
• RE: [asterisk-users] Cisco AS5300
  • From: Mark Rounds
• RE: [asterisk-users] Block some number outgoing from joust oneextention
  • From: Mark Rounds
• Re: [asterisk-users] MusicOnHold Files
  • From: Carla Schroder
• Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
  • From: Noah Miller
• [asterisk-users] 3-way calling MGCP capture
  • From: Olga Mill
• RE: [asterisk-users] no unicall on 1.4
  • From: Anton Krall
• Re: [asterisk-users] Dimensioning a 50 sip phone installation
  • From: Noah Miller
• RE: [asterisk-dev] asterisk sip authentication flawed?
  • From: Damon Estep
• Re: [asterisk-users] no unicall on 1.4
  • From: Lee Howard
• Re: [asterisk-dev] Asterisk zap channels death
  • From: Anton
• Re: [asterisk-users] HowTO configure voice T1
  • From: Eric \"ManxPower\" Wieling
• Re: [asterisk-users] no unicall on 1.4
  • From: Eric \"ManxPower\" Wieling
• [asterisk-users] POE draw on Aastra 480i
  • From: Allen Casteran
• Re: [asterisk-users] Dimensioning a 50 sip phone installation
  • From: Erick Perez
• Re: [asterisk-users] MusicOnHold Files
  • From: Trevor Peirce
• RE: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
  • From: Dan Austin
• Re: [asterisk-users] MusicOnHold Files
  • From: Yuan LIU
• [asterisk-users] Which is GUI to edit Asterisk IVR logic
  • From: Olivier
• Re: [asterisk-users] faxing times!
  • From: Thirumal Saminathan
• [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument
  • From: Lee Archer
• Re: [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument
  • From: Eric \"ManxPower\" Wieling
• RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument
  • From: Lee Archer
• [asterisk-users] Which g729 module for HP DL 360 G3 (Xeon CPU's)?
  • From: Eric Bishop
• Re: [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument
  • From: Tzafrir Cohen
• RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument
  • From: Lee Archer
• Re: [asterisk-dev] Academic Asterisk Adventure
  • From: Tim Panton
• Re: [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument
  • From: Tzafrir Cohen
• Re: [asterisk-users] Gentoo ebuild for 1.4?
  • From: Benko
• Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
  • From: Josué Conti
• [asterisk-users] asterisk 1.4.0 didn't compile chan_zap.so
  • From: Ma Zhiyong
• Re: [asterisk-users] How big a pipe can IAX2 go?
  • From: Dinesh Nair
• [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)
  • From: Robbie Hughes
• RE: [asterisk-users] POE draw on Aastra 480i
  • From: Watkins, Bradley
• Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
  • From: Gordon Henderson
• [asterisk-users] How to build 1.4 with res_crypto.so
  • From: Yann Massard
• RE: [asterisk-users] POE draw on Aastra 480i
  • From: shadowym
• Re: [asterisk-dev] asterisk sip authentication flawed?
  • From: Johansson Olle E
• [asterisk-users] Invalid DivertingLegInformation2 component received 0x38
  • From: Andreas Gaufer
• [asterisk-users] fax transmission
  • From: Vieri Di Paola
• RE: [asterisk-users] Block some number outgoing from joust oneextention
  • From: Mattias Andersson
• Re: [asterisk-users] anyone using metermaid / parked call BLF?
  • From: Dr. Michael J. Chudobiak
• [asterisk-users] Asterisk and IM
  • From: Hall, Eric M.
• [asterisk-users] integrating with Asterisk and OpenSER for Voicemail
  • From: raviprakash sunkara
• RE: [asterisk-users] no unicall on 1.4
  • From: Anton Krall
• RE: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
  • From: Anton Krall
• Re: [asterisk-users] Which is GUI to edit Asterisk IVR logic
  • From: Lee Jenkins
• [asterisk-users] radius
  • From: Khaled
• Re: [asterisk-users] POE draw on Aastra 480i
  • From: Drew Gibson
• Re: [asterisk-dev] asterisk sip authentication flawed?
  • From: Olle E Johansson
• Re: [asterisk-users] Which g729 module for HP DL 360 G3 (Xeon CPU's)?
  • From: Kevin P. Fleming
• [asterisk-dev] Documentation about asterisk API
  • From: Simone Ricci
• Re: [asterisk-users] over 200 queues, anyone?
  • From: Lenz
• Re: [asterisk-users] no unicall on 1.4
  • From: Moises Silva
• [asterisk-dev] Re: [svn-commits] mattf: branch 1.4 r47462 - /branches/1.4/channels/chan_zap.c
  • From: Tzafrir Cohen
• [asterisk-users] Re: [Users] integrating with Asterisk and OpenSER for Voicemail
  • From: Steve Blair
• [asterisk-dev] 0008660: Possible memory leak doing only inbound SIP handling
  • From: Andre Courchesne
• [asterisk-users] idle SIP channels problem
  • From: O . Kamal
• Re: [asterisk-dev] 0008660: Possible memory leak doing only inbound SIP handling
  • From: Tzafrir Cohen
• Re: [asterisk-users] Which is GUI to edit Asterisk IVR logic
  • From: Olivier
• Re: [asterisk-dev] 0008660: Possible memory leak doing only inbound SIP handling
  • From: Andre Courchesne
• Re: [asterisk-users] over 200 queues, anyone?
  • From: BJ Weschke
• Re: [asterisk-users] Gentoo ebuild for 1.4?
  • From: Sune Kloppenborg Jeppesen
• [asterisk-dev] Build Asterisk 1.4 addons on a blank new server.
  • From: Roland Ndaka Fru
• [asterisk-users] addons 1.4 and cdr_addon_mysql not installed !
  • From: Luca Lafranchi Lists
• Re: [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
  • From: BJ Weschke
• [asterisk-users] how to transfer calls when analog phone has no transfer button
  • From: Erick Perez
• Re: [asterisk-dev] Support for Agent channels in Bridge manager and dial plan patch
  • From: BJ Weschke
• RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument
  • From: Lee Archer
• [asterisk-users] Re: POE draw on Aastra 480i
  • From: Allen Casteran
• RE: [asterisk-users] no unicall on 1.4
  • From: Anton Krall
• Re: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
  • From: Rob Fugina
• Re: [asterisk-dev] 0008660: Possible memory leak doing only inbound SIP handling
  • From: Andre Courchesne
• Re: [asterisk-users] how to transfer calls when analog phone has no transfer button
  • From: cb
• [asterisk-users] how to register nokia with Asterisk
  • From: Biju
• Re: [asterisk-dev] Build Asterisk 1.4 addons on a blank new server.
  • From: Tzafrir Cohen
• [asterisk-users] ASterisk OOH323c
  • From: Michel
• RE: [asterisk-users] HowTO configure voice T1
  • From: Don Pobanz
• RE: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
  • From: Dan Austin
• Re: [asterisk-users] no unicall on 1.4
  • From: Lee Howard
• RE: [asterisk-users] how to transfer calls when analog phone has notransfer button
  • From: Don Pobanz
• Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
  • From: Josué Conti
• Re: [asterisk-users] addons 1.4 and cdr_addon_mysql not installed !
  • From: Tzafrir Cohen
• [asterisk-users] asterisk (FreePBX) and queues
  • From: Felipe Neuwald
• RE: [asterisk-dev] Build Asterisk 1.4 addons on a blank new server.
  • From: Jonathan k. Creasy
• Re: [asterisk-users] how to transfer calls when analog phone has no transfer button
  • From: Eric \"ManxPower\" Wieling
• Re: [asterisk-users] POE draw on Aastra 480i
  • From: Ira
• [asterisk-users] SIP/TCP?
  • From: Yuan LIU
• Re: [asterisk-users] how to transfer calls when analog phone has notransfer button
  • From: Erick Perez
• [asterisk-users] asterisk 1.4 debian packages
  • From: Juraj Bednar
• Re: [asterisk-users] SIP/TCP?
  • From: Howard Lowndes
• Re: [asterisk-users] Dimensioning a 50 sip phone installation
  • From: Luki
• Re: [asterisk-users] asterisk 1.4 debian packages
  • From: Tzafrir Cohen
• Re: [asterisk-users] asterisk (FreePBX) and queues
  • From: Alex Robar
• [asterisk-users] Has anybody voipstunt working?
  • From: Arik Raffael Funke
• [asterisk-users] Multiple users and a single extension
  • From: Phil Finkler
• Re: [asterisk-users] Multiple users and a single extension
  • From: Gavin Hamill
• RE: [asterisk-users] how to transfer calls when analog phone hasnotransfer button
  • From: Don Pobanz
• [asterisk-users] Re: SIP/TCP?
  • From: Steven
• AW: [asterisk-dev] Build Asterisk 1.4 addons on a blank new server.
  • From: Roland Ndaka Fru
• RE: [asterisk-users] Best inexpensive home office router for VoIP(QoS with maybe PoE)
  • From: Mike
• Re: [asterisk-users] Best inexpensive home office router for VoIP(QoS with maybe PoE)
  • From: Alex Robar
• RE: [asterisk-users] SIP/TCP?
  • From: James R. Stevens
• [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?
  • From: Ray Jackson
• [asterisk-users] Random "unknown" codec format IAX calls
  • From: Max Ochoa
• [asterisk-users] DiD for less then $4
  • From: CM Rahman
• RE: [asterisk-users] how to transfer calls when analog phone hasnotransfer button
  • From: Doug Crompton
• [asterisk-users] Re: Best inexpensive home office router for VoIP(QoS with maybe PoE)
  • From: Allen Casteran
• RE: [asterisk-users] Voicemail personalised greetings using DB/IMAPbackend?
  • From: Douglas Garstang
• Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?
  • From: Bryan M. Johns
• Re: [asterisk-users] SIP/TCP?
  • From: Tzafrir Cohen
• Re: [asterisk-users] Asterisk to a Huawei softX3000 problem has already been solved ！
  • From: Guillermo Salas M.
• Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?
  • From: Kevin P. Fleming
• [asterisk-users] Call waiting notification
  • From: Kevin Smith
• [asterisk-users] .call files no longer generating CDR files
  • From: Michael Collins
• Re: [asterisk-users] fax transmission
  • From: Doug Lytle
• RE: [asterisk-users] SIP/TCP?
  • From: Yuan LIU
• Re: [asterisk-users] Voicemail personalised greetings using DB/IMAPbackend?
  • From: Ray Jackson
• Re: [asterisk-dev] 0008660: Possible memory leak doing only inbound SIP handling
  • From: Marc Blanchet
• Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?
  • From: Ray Jackson
• Re: [asterisk-users] Call waiting notification
  • From: Kevin P. Fleming
• Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?
  • From: Ray Jackson
• Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?
  • From: Kevin P. Fleming