Asterisk

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Re: [FreeBSD 6.3] Right-way to recover Zaptel?
- From: Anthony Francis
realtime queue reload
- From: Thomas Winter
Re: dahdi & tdm400p: no luck
- From: sean darcy
[FreeBSD 6.3] Right-way to recover Zaptel?
- From: Vincent
Re: dahdi & tdm400p: no luck
- From: Matt Gibson
Re: dahdi & tdm400p: no luck
- From: Tzafrir Cohen
Re: dahdi & tdm400p: no luck
- From: sean darcy
Re: dahdi & tdm400p: no luck
- From: Matt Gibson
call-limit problem
- From: Nhadie
Re: Transfers on AgentLogin()
- From: Matt Riddell
Re: Transfers on AgentLogin()
- From: Mark Hamilton
Re: Polycom BLF - multiple buddies
- From: Matt Riddell
Re: Transfers on AgentLogin()
- From: James Sneeringer
Re: Polycom BLF - multiple buddies
- From: James Sneeringer
Re: dahdi & tdm400p: no luck
- From: sean darcy
Re: dahdi & tdm400p: no luck
- From: sean darcy
Re: dahdi & tdm400p: no luck
- From: Tzafrir Cohen
Re: dahdi & tdm400p: no luck
- From: sean darcy
Re: (no subject)
- From: Shariq Khan
Re: FAX over T1 Question
- From: Eric \"ManxPower\" Wieling
soft hangup (was: Re: (no subject))
- From: Philipp Kempgen
Re: svn branches for dhadi and its tools
- From: John covici
Re: New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
- From: Tony Mountifield
Re: FAX over T1 Question
- From: Amaru Netapshaak
(no subject)
- From: Bill Andersen
Re: New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
- From: Tzafrir Cohen
Re: svn branches for dhadi and its tools
- From: Tzafrir Cohen
Re: FAX over T1 Question
- From: Vinícius Fontes
Re: FAX over T1 Question
- From: Alex Balashov
Re: FAX over T1 Question
- From: Steve Totaro
Re: FAX over T1 Question
- From: Eric \"ManxPower\" Wieling
Re: Call monitor/barge/train
- From: Steve Totaro
Re: New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
- From: Kevin P. Fleming
Re: Call monitor/barge/train
- From: David Backeberg
Re: FAX over T1 Question
- From: Tim Nelson
Re: FAX over T1 Question
- From: Steve Totaro
Re: FAX over T1 Question
- From: Bob Pierce
FAX over T1 Question
- From: Amaru Netapshaak
Re: PRI Splitter
- From: Craig Guy
Re: Call-leg stays on MusicOnHold forever
- From: Jorge Mendoza
Re: Transfers on AgentLogin()
- From: Mark Hamilton
Re: Polycom BLF - multiple buddies
- From: Robert McNaught
Re: Bridge 2 incoming calls
- From: Tim Panton
Re: Bridge 2 incoming calls
- From: Tim Panton
Re: Dear asterisk-users@xxxxxxxxxxxxxxxx 79% OFF on Pfizer
- From: Joe Pukepail
Re: New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
- From: Steve Repo
Re: Bridge 2 incoming calls
- From: Steve Murphy
Re: New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
- From: Tony Mountifield
Re: Asterisk Crash
- From: Tony Mountifield
Re: Dear asterisk-users@xxxxxxxxxxxxxxxx 79% OFF on Pfizer
- From: VIAGRA Ž
svn branches for dhadi and its tools
- From: John covici
Re: dahdi & tdm400p: no luck
- From: Tzafrir Cohen
Re: Bridge 2 incoming calls
- From: Andreas Brodmann
Re: G722 and Asterisk 1.6
- From: Olivier
Re: dahdi & tdm400p: no luck
- From: sean darcy
FW: Vivox SLim
- From: Dean Collins
Call-leg stays on MusicOnHold forever
- From: Andreas Brodmann
Grandstream Video Phones & Asterisk..
- From: Gordon Henderson
Bridge 2 incoming calls
- From: Tim Panton
Re: Gateway errors
- From: voip crazy
Re: The question about the M(X)option of Dial
- From: Alex Balashov
Re: ringback when the channel is answered
- From: eng. Anatoli Marinov
Re: dahdi & tdm400p: no luck
- From: Tzafrir Cohen
Re: ASTERISK supported Video phone
- From: eng. Anatoli Marinov
Re: Polycom BLF - multiple buddies
- From: Lee, John (Sydney)
Re: Polycom BLF - multiple buddies
- From: James Sneeringer
Re: Polycom BLF - multiple buddies
- From: James Sneeringer
dahdi & tdm400p: no luck
- From: sean darcy
Re: Polycom BLF - multiple buddies
- From: Lee, John (Sydney)
Re: Polycom BLF - multiple buddies
- From: James Sneeringer
Re: G722 and Asterisk 1.6
- From: Michael Graves
Re: G722 and Asterisk 1.6
- From: Darrick Hartman
Linksys 3102 - Call Waiting
- From: Joseph
Re: Asterisk supported Video phone
- From: bilal ghayyad
libpri 1.4.5 priindication
- From: Jorge Mendoza
Re: iLBC codec
- From: ronald
The question about the M(X)option of Dial
- From: larry
Re: Polycom BLF - multiple buddies
- From: Lee, John (Sydney)
Re: extensions.conf programming?
- From: Lee, John (Sydney)
Re: DID number
- From: Philipp Kempgen
Re: G722 and Asterisk 1.6
- From: Steve Underwood
Re: DID number
- From: Marcelo Freitas
Polycom BLF - multiple buddies
- From: Robert McNaught
Re: DAHDI FAQ not up. Anyplace else?
- From: Kevin P. Fleming
Re: Asterisk supported Video phone
- From: Philipp Kempgen
Re: ASTERISK supported Video phone
- From: bilal ghayyad
Re: How to setup SIP so that RTP traffic flows from Source to destination
- From: Terry Wilson
Re: How to setup SIP so that RTP traffic flows from Source to destination
- From: Anthony Francis
Re: ringback when the channel is answered
- From: eng. Anatoli Marinov
How to setup SIP so that RTP traffic flows from Source to destination
- From: Shaun Wingrin
Re: ringback when the channel is answered
- From: Steve Totaro
Re: ringback when the channel is answered
- From: eng. Anatoli Marinov
DAHDI FAQ not up. Anyplace else?
- From: sean darcy
Re: 1.6rc4 chan_iax2 messages
- From: MFH
Re: 1.6rc4 chan_iax2 messages
- From: Tilghman Lesher
Re: G722 and Asterisk 1.6
- From: Gordon Henderson
Re: G722 and Asterisk 1.6
- From: Michael Graves
Re: extensions.conf programming?
- From: Darren Sessions
Re: ASTERISK supported Video phone
- From: Gordon Henderson
Re: extensions.conf programming?
- From: Mark Michelson
Re: ringback when the channel is answered
- From: Eric \"ManxPower\" Wieling
1.6rc4 chan_iax2 messages
- From: MFH
Re: G722 and Asterisk 1.6
- From: Thomas Kenyon
extensions.conf programming?
- From: Ken D'Ambrosio
Re: PRI Splitter
- From: FailSafe Inc.
Re: ASTERISK supported Video phone
- From: eng. Anatoli Marinov
Re: ringback when the channel is answered
- From: eng. Anatoli Marinov
Re: G722 and Asterisk 1.6
- From: Michael Graves
Re: ASTERISK supported Video phone
- From: Steve Repo
Re: G722 and Asterisk 1.6
- From: Steve Repo
Re: ringback when the channel is answered
- From: Eric \"ManxPower\" Wieling
ringback when the channel is answered
- From: eng. Anatoli Marinov
Re: How to check mailbox exists (Received SIP subscribe for peer without mailbox)
- From: Rob Hillis
strange transfer problem
- From: Rizwan Hisham
Re: G722 and Asterisk 1.6
- From: Peder @ NetworkOblivion
Re: iLBC codec
- From: Tilghman Lesher
Re: #include changes in 1.4
- From: Tilghman Lesher
Re: Dial L( x [: y ][: z ]) option truncates colon (:) using AGI /_
- From: Tilghman Lesher
conf files for dahdi
- From: sean darcy
Re: G722 and Asterisk 1.6
- From: Russell Bryant
Re: Stability problems in Asterisk 1.4.18 (and other 1.4.xx versions)
- From: Russell Bryant
Logs: messages, events, queue
- From: bilal ghayyad
Re: How to check mailbox exists (Received SIP subscribe for peer without mailbox)
- From: Olivier
Re: Z-Wave or Zigbee for Office or Home automation using XML Browser enabled Screen Phone
- From: Michael Graves
New Install using DAHDI
- From: Joseph L. Casale
MixMonitor + Originate
- From: Geraint Lee
Re: dial out via fxo gateway
- From: Steve Repo
Re: DID number
- From: michel freiha
Re: G722 and Asterisk 1.6
- From: Steve Repo
dial out via fxo gateway
- From: ACL
Re: PRI Splitter
- From: Igor Hernandez
iLBC codec
- From: Nhadie
Re: How to check mailbox exists (Received SIP subscribe for peer without mailbox)
- From: Rob Hillis
How to check mailbox exists (Received SIP subscribe for peer without mailbox)
- From: Olivier
Installing ValetParking?
- From: Tobias Ahlander
Re: Faxing through Zap cards
- From: z_gringo
#include changes in 1.4
- From: Chris Bagnall
Stability problems in Asterisk 1.4.18 (and other 1.4.xx versions)
- From: z_gringo
Re: Reliable wireless SIP phones
- From: Thomas Kenyon
Re: Asterisk Queue's
- From: Tobias Ahlander
Re: ASTERISK supported Video phone
- From: Gordon Henderson
Dial L( x [: y ][: z ]) option truncates colon (:) using AGI /_
- From: selmak se
Re: Congestion in Outgoing call through PRI
- From: Shariq Khan
Re: MixMonitor-Saving Recorded file with AgentId.
- From: Syed Nasruddin
Re: DID number
- From: Jaswinder Singh
Re: PRI Splitter
- From: Craig Guy
Z-Wave or Zigbee for Office or Home automation using XML Browser enabled Screen Phone
- From: Karl Fife
ASTERISK supported Video phone
- From: Tharanga
Voicemail "from an unknown caller"
- From: Andrew Joakimsen
Re: New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
- From: Russell Bryant
Re: G722 and Asterisk 1.6
- From: Russell Bryant
Re: New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
- From: sean darcy
Re: All calls want to go out only on interface ZAP/g0
- From: Paul Hales
Re: All calls want to go out only on interface ZAP/g0
- From: Paul Hales
All calls want to go out only on interface ZAP/g0
- From: Mark Best
Re: Asterisk with E1 interface vs IP PBX
- From: Luis Morales
Re: G722 and Asterisk 1.6
- From: Thomas Kenyon
Combine sip audio and video from different sources
- From: Artem Makhutov
Re: Faxing through Zap cards
- From: C F
New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
- From: asteriskteam
Re: DID number
- From: michel freiha
Re: DID number
- From: Igor Hernandez
Re: DID number
- From: michel freiha
Re: Ringing on Console after a page
- From: Josiah Bryan
Re: Asterisk Crash
- From: Josiah Bryan
Re: Ringing on Console after a page
- From: Barry L. Kline
Re: DID number
- From: Igor Hernandez
Asterisk with E1 interface vs IP PBX
- From: Alejandro Facultad
Re: G722 and Asterisk 1.6
- From: Michael Graves
Re: Asterisk Crash
- From: Andrew Latham
Re: G722 and Asterisk 1.6
- From: Thomas Kenyon
DID number
- From: michel freiha
Re: Faxing through Zap cards
- From: Luis Morales
Ringing on Console after a page
- From: Josiah Bryan
Re: Congestion in Outgoing call through PRI
- From: Richard Lyman
Re: Asterisk Crash
- From: Josiah Bryan
Re: PRI Splitter
- From: Igor Hernandez
Asterisk voicemail message order
- From: Robert Lister
Re: PRI Splitter
- From: Karl Fife
Re: Faxing through Zap cards
- From: Karl Fife
Re: G722 and Asterisk 1.6
- From: Michael Graves
G722 and Asterisk 1.6
- From: Steve Repo
Re: Inefficient Codec Translation
- From: Kevin P. Fleming
sip to sip unplanned conference! help!!
- From: RoLaNd RoLaNd
Re: Congestion in Outgoing call through PRI
- From: Octavio Ruiz
Re: Problem with Call Forward
- From: Philipp Kempgen
Re: Problem with Call Forward
- From: Nhadie
Re: Asterisk Crash
- From: Josiah Bryan
SIP TLS / Nokia E51
- From: Stefan Gofferje
Re: Live operator as a service?
- From: Michael Graves
Re: Inefficient Codec Translation
- From: Steve Davies
Re: Asterisk Crash
- From: Andrew Latham
Re: Problem with Call Forward
- From: Dpto. Datos Television Costa Blanca
Re: Asterisk Crash
- From: Josiah Bryan
Re: Asterisk Crash
- From: Andrew Latham
Re: Congestion in Outgoing call through PRI
- From: Richard Lyman
res_cepstral.so
- From: Norman Franke
Asterisk Crash
- From: Josiah Bryan
Re: Selectively disable echo cancellation?
- From: Jerry Jones
Re: Live operator as a service?
- From: Dean Collins
Re: Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte & Randal Schwartz
- From: Eric Moniz
Live operator as a service?
- From: Michael Graves
Re: Asterisk Queue's
- From: Atis Lezdins
Re: AgentCallbackLogin AddQueueMember
- From: Atis Lezdins
Re: Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte & Randal Schwartz
- From: randulo
Re: Asterisk Queue's
- From: José Carlos Messias
Re: Inefficient Codec Translation
- From: Jim Boykin
Re: multiple passwords for one meetme!
- From: Philipp Kempgen
Re: Asterisk Queue's
- From: Alejandro Kauffmann
Re: AgentCallbackLogin AddQueueMember
- From: Lee, John (Sydney)
MixMonitor-Saving Recorded file with AgentId.
- From: Syed Nasruddin
Re: AgentCallbackLogin AddQueueMember
- From: Krzysztof Zimnicki
Re: Newbie Polycom: ACD AgentLogin display on phone
- From: Paul Hales
multiple passwords for one meetme!
- From: fateme fatah
Re: Dial timeout to cell phones
- From: Steve Repo
Newbie Polycom: ACD AgentLogin display on phone
- From: Lee, John (Sydney)
Re: Asterisk Queue's
- From: Tobias Ahlander
Re: Asterisk 1.6 beta
- From: Olivier
Offering FIFO service to receptionist with LIFO hardphone ...
- From: Olivier
Re: PRI Splitter
- From: Olivier
Re: Congestion in Outgoing call through PRI
- From: Octavio Ruiz
Re: Selectively disable echo cancellation?
- From: Octavio Ruiz
Re: DUNDI Help
- From: ronald ramos
Re: beta9: how to set callerid on incoming iax?
- From: sean darcy
Re: AgentCallbackLogin AddQueueMember
- From: Lee, John (Sydney)
Re: Asterisk CDR Problem
- From: Steve Murphy
Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte & Randal Schwartz
- From: Michael Graves
Re: Problem with Call Forward
- From: Nhadie
Selectively disable echo cancellation?
- From: Ken D'Ambrosio
Re: Dial timeout to cell phones
- From: Mark G. Thomas
Re: Dial timeout to cell phones
- From: John Novack
Re: Problematic Trunk SIP: Got SIP response 405 "Method not allowed"
- From: Kristian Kielhofner
Dial timeout to cell phones
- From: Steve Repo
Re: SALE 71% OFF on Pfizer
- From: asterisk-users
Re: Faxing through Zap cards
- From: James Sneeringer
Re: play remote file
- From: Atis Lezdins
Re: AgentCallbackLogin AddQueueMember
- From: Atis Lezdins
Re: Asterisk Trunk and normal
- From: Erik Anderson
Re: Asterisk Trunk and normal
- From: bilal ghayyad
Re: Asterisk Trunk and normal
- From: Erik Anderson
Re: PRI Splitter
- From: FailSafe Inc.
Re: Asterisk Trunk and normal
- From: bilal ghayyad
AgentCallbackLogin AddQueueMember
- From: Krzysztof Zimnicki
Re: Redundant PSTN PRI Gateways using Asterisk
- From: Steve Totaro
Re: Asterisk Trunk and normal
- From: Philipp Kempgen
Re: lists.digium.com monthly reminders
- From: Jay R. Ashworth
Re: Pri to sip interfaces
- From: Christian Victor
Re: Asterisk Trunk and normal
- From: Dan Julius
Re: Asterisk Trunk and normal
- From: Doug Lytle
Asterisk Trunk and normal
- From: bilal ghayyad
Re: Redundant PSTN PRI Gateways using Asterisk
- From: Karl Fife
zaptel 1.2.27 ?
- From: Leonardo Gomes Figueira
Re: Problem with Call Forward
- From: Philipp Kempgen
Re: SetCallerPres
- From: Steven Howes
SetCallerPres
- From: Steven Howes
play remote file
- From: Pezhman Lali
Re: Problem with Call Forward
- From: Dpto. Datos Television Costa Blanca
Re: Redundant PSTN PRI Gateways using Asterisk
- From: Alex Balashov
Re: Gateway errors
- From: hatem moiz
Re: Asterisk Queue's
- From: Paul Crane
Dialplan terminates when the caller hangs up
- From: Cong Van Nguyen
Redundant PSTN PRI Gateways using Asterisk
- From: Michael Melia Jr.
Re: PRI Splitter
- From: Karl Fife
Still badly in need for mISDN help!
- From: Julien Claassen
Re: PRI Splitter
- From: Christian Victor
Re: PRI Splitter
- From: Karl Fife
Re: lists.digium.com monthly reminders
- From: Benny Amorsen
Re: Asterisk 1.6 beta
- From: Benny Amorsen
Re: lists.digium.com monthly reminders
- From: Tony Mountifield
Re: PRI Splitter
- From: Karl Fife
Re: lists.digium.com monthly reminders
- From: Ira
Re: Documentation of users.conf
- From: Robert Lister
Re: PRI Splitter
- From: Olivier
Documentation of users.conf
- From: Nestor A. Diaz
Re: Asterisk 1.6 beta
- From: Tim Panton
Re: Asterisk 1.6 beta
- From: Rob Hillis
Re: PRI Splitter
- From: Karl Fife
Re: PRI Splitter
- From: Karl Fife
Re: Get call status and hangup
- From: Loic Didelot
Re: PRI Splitter
- From: Michael Graves
Re: PRI Splitter
- From: Karl Fife
Asterisk 1.6 beta
- From: VoIP Cyprus
Re: lists.digium.com monthly reminders
- From: Tony Mountifield
Re: lists.digium.com monthly reminders
- From: Kevin P. Fleming
Re: lists.digium.com monthly reminders
- From: Tony Mountifield
Re: lists.digium.com monthly reminders
- From: Tzafrir Cohen
Re: lists.digium.com monthly reminders
- From: randulo
Re: lists.digium.com monthly reminders
- From: Michiel van Baak
lists.digium.com monthly reminders
- From: Tony Mountifield
Re: Problematic Trunk SIP: Got SIP response 405 "Method not allowed"
- From: daniele visaggio
Re: Problematic Trunk SIP: Got SIP response 405 "Method not allowed"
- From: Steven Howes
Re: Gateway errors
- From: voip crazy
Re: PRI Splitter
- From: Christian Victor
Problematic Trunk SIP: Got SIP response 405 "Method not allowed"
- From: daniele visaggio
not able to make call to landline no...to mobile works fine
- From: bikrish
Penalties for agents
- From: Tobias Ahlander
Re: Gateway errors
- From: voip crazy
Re: Gateway errors
- From: hatem moiz
Re: Gateway errors
- From: Igor Hernandez
Re: Asterisk Queue's
- From: Philipp Kempgen
Re: Gateway errors
- From: Andrea Spadaccini
Re: Gateway errors
- From: voip crazy
Re: Gateway errors
- From: Igor Hernandez
Gateway errors
- From: voip crazy
Re: Asterisk Queue's
- From: Tobias Ahlander
Re: Asterisk IVR Scalability
- From: Al Baker
PhoneControl integrations
- From: Craig Guy
Re: Transfers on AgentLogin()
- From: Mark Hamilton
Re: PRI Splitter
- From: Olivier
Re: Asterisk IVR Scalability
- From: Luis Morales
Re: Asterisk IVR Scalability
- From: Mike Trest
Asterisk IVR scalability
- From: Sriram
Asterisk IVR Scalability
- From: Sriram
Re: Wi-SIP & 802.11f - Inter Access Point Protocol HANDOFF
- From: Duncan Turnbull
Re: Wi-SIP vs. SIP-DECT
- From: Gordon Henderson
Re: security on localhost connections
- From: Tim Panton
Re: Faxing through Zap cards
- From: C F
Intermittent "rejected because extension not found" On Incoming DID
- From: Robert DeVries
Re: security on localhost connections
- From: Tilghman Lesher
Re: Wi-SIP vs. SIP-DECT
- From: Michael Graves
Re: Heist of MagicJack SIP credentials?
- From: Cory Andrews
security on localhost connections
- From: David Burgess
Re: Problems with DTMF on IVRs
- From: Al lists
Re: Transfers on AgentLogin()
- From: Matt Riddell
Re: Wi-SIP vs. SIP-DECT
- From: Benny Amorsen
Re: Reliable wireless SIP phones (Tzafrir Cohen)
- From: Freddi Hansen
Re: Issue when dialing multiple extensions using & ------Please Help
- From: Mark Michelson
Re: Wi-SIP & 802.11f - Inter Access Point Protocol HANDOFF
- From: Anthony Messina
Re: beta9: how to set callerid on incoming iax?
- From: sean darcy
Re: Wi-SIP & 802.11f - Inter Access Point Protocol HANDOFF
- From: Karl Fife
Re: Congestion in Outgoing call through PRI
- From: Grygoriy Dobrovolskyy
Re: beta9: how to set callerid on incoming iax?
- From: sean darcy
beta9: how to set callerid on incoming iax?
- From: sean darcy
Re: Heist of MagicJack SIP credentials?
- From: Michael Graves
Re: Wi-SIP vs. SIP-DECT
- From: Michael Graves
Congestion in Outgoing call through PRI
- From: Shariq Khan
Re: Wi-SIP & 802.11f - Inter Access Point Protocol HANDOFF
- From: Anthony Messina
Re: Wi-SIP & 802.11f - Inter Access Point Protocol HANDOFF
- From: Michael Graves
Wi-SIP & 802.11f - Inter Access Point Protocol HANDOFF
- From: Karl Fife
Re: PRI Splitter
- From: Karl Fife
Heist of MagicJack SIP credentials?
- From: Karl Fife
Re: Wi-SIP vs. SIP-DECT
- From: Karl Fife
Re: Wi-SIP vs. SIP-DECT
- From: Benny Amorsen
Re: Faxing through Zap cards
- From: Doug Lytle
Re: PRI Splitter
- From: Olivier
Re: music on hold is not working
- From: Lee, John (Sydney)
Incoming Calls via SIP Trunks
- From: Andreas M.
Re: Asterisk CDR Problem
- From: Hiren Mistry
Re: Asterisk CDR Problem
- From: Max Alex
Re: Reliable wireless SIP phones
- From: Tzafrir Cohen
Zap channel DTMF regeneration
- From: drtester@xxxxxxxxxxx
Re: Wi-SIP vs. SIP-DECT
- From: Anthony Francis
Re: Reliable wireless SIP phones
- From: Michael Graves
Re: Reliable wireless SIP phones
- From: Michael Graves
Re: Wi-SIP vs. SIP-DECT
- From: Michael Graves
Re: Audio data between concurrent SIP and PSTN
- From: Alex Balashov
Re: Faxing through Zap cards
- From: James Sneeringer
Audio data between concurrent SIP and PSTN
- From: Allann Jones
Re: Faxing through Zap cards
- From: C F
CDR userfield recording name
- From: Gustavo A Gonzalez
Re: Call monitor/barge/train
- From: Mark Hamilton
Re: Wi-SIP vs. SIP-DECT
- From: Phil Thompson
Re: GSM recordings
- From: Jay R. Ashworth
Re: Connecting two asterisks via IAX
- From: Nuno Marques
Re: Call monitor/barge/train
- From: Robin Rodriguez
Re: Faxing through Zap cards
- From: James Sneeringer
Call monitor/barge/train
- From: Mark Hamilton
Re: Transfers on AgentLogin()
- From: Mark Hamilton
Re: Reliable wireless SIP phones
- From: Anselm Martin Hoffmeister
Re: Connecting two asterisks via IAX
- From: Tom Moore
Re: Reliable wireless SIP phones
- From: Tzafrir Cohen
Re: Faxing through Zap cards
- From: Doug Lytle
Re: Faxing through Zap cards
- From: Doug Lytle
music on hold is not working
- From: Krunal Patel
Issue when dialing multiple extensions using & ------Please Help
- From: Krunal Patel
Re: Reliable wireless SIP phones
- From: Ira
Re: Faxing through Zap cards
- From: James Sneeringer
Re: Faxing through Zap cards
- From: C F
Re: Faxing through Zap cards
- From: James Sneeringer
Re: Wi-SIP vs. SIP-DECT
- From: Gordon Henderson
chan_mobile
- From: michael
Re: Wi-SIP vs. SIP-DECT
- From: Tilghman Lesher
Re: Wi-SIP vs. SIP-DECT
- From: Roderick A. Anderson
Re: Faxing through Zap cards
- From: Gordon Henderson
Connecting two asterisks via IAX
- From: Nuno Marques
Re: Faxing through Zap cards
- From: Doug Lytle
Faxing through Zap cards
- From: James Sneeringer
track 1.6 progress
- From: Sigma Networks
Re: Reliable wireless SIP phones
- From: Karl Fife
Wi-SIP vs. SIP-DECT
- From: Karl Fife
Re: sip conversations overlapping!!!!
- From: Drew Gibson
Re: Reliable wireless SIP phones
- From: Karl Fife
Re: Asterisk Queue's
- From: Mark Michelson
Re: Reliable wireless SIP phones
- From: Fred Posner
Re: Asterisk Tips and Tricks: Dynamic Subroutines inAGI
- From: Darren Sessions
Re: Console softphone
- From: Tilghman Lesher
Re: Reliable wireless SIP phones
- From: Gordon Henderson
Re: Reliable wireless SIP phones
- From: Gordon Henderson
Re: Reliable wireless SIP phones
- From: Karl Fife
Re: Asterisk Tips and Tricks: Dynamic Subroutines inAGI
- From: Watkins, Bradley
Re: Reliable wireless SIP phones
- From: Michael Graves
Re: Problems with DTMF on IVRs
- From: Chris Mason
Re: sip conversations overlapping!!!!
- From: Grygoriy Dobrovolskyy
Asterisk cdr_mysql inexact values
- From: Grygoriy Dobrovolskyy
Re: sip conversations overlapping!!!!
- From: RoLaNd RoLaNd
Re: sip conversations overlapping!!!!
- From: Grygoriy Dobrovolskyy
Re: remove queue call
- From: Steven Howes
Re: Console softphone
- From: Julien Claassen
Re: Reliable wireless SIP phones
- From: Gordon Henderson
Re: Asterisk Queue's
- From: Tobias Ahlander
Re: remove queue call
- From: Rilawich Ango
Re: Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura
- From: Joseph
Asterisk CDR Problem
- From: Hiren Mistry
Asterisk Tips and Tricks: Dynamic Subroutines in AGI
- From: Darren Sessions
Re: Console softphone
- From: Paul Hales
Re: Reliable wireless SIP phones
- From: Michael Graves
Re: Console softphone
- From: Lee, John (Sydney)
Re: Console softphone
- From: Tilghman Lesher
Re: Is including a linefeed in the JabberSend message possible?
- From: Matt Gibson
Re: Console softphone
- From: Lee, John (Sydney)
Is including a linefeed in the JabberSend message possible?
- From: Eric Chamberlain
Re: PRI Splitter
- From: Dan Julius
Re: Transfers on AgentLogin()
- From: Mark Hamilton
Transfers on AgentLogin()
- From: Mark Hamilton
Re: sip conversations overlapping!!!!
- From: RoLaNd RoLaNd
Re: Console softphone
- From: T G
Re: X100P Card in OFFHOOK state
- From: Jay Ray
Re: {Fraud?} {Disarmed} Re: Problems with DTMF on IVRs
- From: Chris Mason (Lists)
Re: Need application, CID number match list to call cell phone
- From: David Backeberg
Re: troubleshooting mISDN...
- From: Julien Claassen
Re: GSM recordings
- From: Javier Prieto Gomez
Re: GSM recordings
- From: Shariq Khan
Re: sip conversations overlapping!!!!
- From: Drew Gibson
Re: GSM recordings
- From: Julien Claassen
troubleshooting mISDN...
- From: Julien Claassen
GSM recordings
- From: Gustavo A Gonzalez
Caller ID in IAX trunk, SIP trunk, between extensions and from FXO
- From: bilal ghayyad
sip conversations overlapping!!!!
- From: RoLaNd RoLaNd
meetme + jitter buffer
- From: Stanisław Pitucha
Re: Reliable wireless SIP phones
- From: Tom Moore
Re: Reliable wireless SIP phones
- From: Fred Posner
VoicePulse Time out?
- From: Fred Posner
Re: Reliable wireless SIP phones
- From: Fred Posner
Re: Reliable wireless SIP phones
- From: Gordon Henderson
Re: Reliable wireless SIP phones
- From: Alan Lord
Re: Newbie: Queue and CDR Reporter and Analyser
- From: Drew Gibson
Re: H323 protocol
- From: Guillermo Salas M.
Re: can not load chan_dahdi.so from asterisk!
- From: Tzafrir Cohen
Re: asterisk linkedin group
- From: Guillermo Salas M.
asterisk linkedin group
- From: BerkHolz, Steven
Re: TDM2400P Voice Quality Problem
- From: Shariq Khan
OT: SEP<mac addr>.cnf.xml file for 7911 with SIP 8.3.5 firmware
- From: Patrick
Re: Reliable wireless SIP phones
- From: Tilghman Lesher
Re: Reliable wireless SIP phones
- From: David Nedved
Re: Reliable wireless SIP phones
- From: Geraint Lee
Re: Reliable wireless SIP phones
- From: Cory Andrews
Re: sip peering between 2 asterisk
- From: Phil Thompson
Re: Reliable wireless SIP phones
- From: Darren Nickerson
Re: Reliable wireless SIP phones
- From: Grygoriy Dobrovolskyy
Re: Reliable wireless SIP phones
- From: Grygoriy Dobrovolskyy
Re: Reliable wireless SIP phones
- From: Tilghman Lesher
Re: can not load chan_dahdi.so from asterisk!
- From: David A. Bandel
Re: Reliable wireless SIP phones
- From: --[ UxBoD ]--
Reliable wireless SIP phones
- From: Jaap Winius
Re: app_jack and calling with pc only
- From: Julien Claassen
Re: VUC Friday: asterisk 1.6 fax, Drawing for Free Astricon Pass
- From: Russell Bryant
Re: TDM2400P Voice Quality Problem
- From: Russell Bryant
Re: app_jack and calling with pc only
- From: Russell Bryant
Re: Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura
- From: Jay R. Ashworth
Re: Asterisk Queue's
- From: Philipp Kempgen
Re: H323 protocol
- From: Paul Catchpole
Re: Problems with DTMF on IVRs
- From: Ruchir
Re: H323 protocol
- From: mahboob zaman
Asterisk Queue's
- From: Tobias Ahlander
Weird asterisk error: ztscan command not found
- From: Steve Repo
Problems with DTMF on IVRs
- From: Chris Mason (Lists)
Re: remove queue call
- From: Steven Howes
Re: remove queue call
- From: Rilawich Ango
Re: ultramonkey and asterisk
- From: ronald
Re: Pri to sip interfaces
- From: Tom Moore
Console softphone
- From: Julien Claassen
Re: H323 protocol
- From: map
How to measure call lenght and act upon it?
- From: Rennes Neps
H323 protocol
- From: mahboob zaman
Re: Asterisk 1.4 -> 1.6
- From: Thomas Kenyon
Re: remove queue call
- From: Steven Howes
Re: remove queue call
- From: Rilawich Ango
Re: execute command after sip register
- From: Steven Howes
Re: Asterisk 1.4 -> 1.6
- From: Chris Maciejewski
execute command after sip register
- From: Andreas M.
Re: Callback voice Quality
- From: michel freiha
Asterisk 1.4 -> 1.6
- From: --[ UxBoD ]--
Re: can not load chan_dahdi.so from asterisk!
- From: Tzafrir Cohen
Re: PRI Splitter
- From: map
Re: Asterisk CLI Show Error :- ("**Unknown**") instead of ("Zap/22-1", )
- From: Max Alex
Asterisk CLI Show Error :- ("**Unknown**") instead of ("Zap/22-1", )
- From: Hiren Mistry
execute command after sip register
- From: Andreas M.
can not load chan_dahdi.so from asterisk!
- From: lizhong zhu
Re: remove queue call
- From: Andy Kuo
Re: X100P Card in OFFHOOK state
- From: Jay Ray
Re: X100P Card in OFFHOOK state
- From: Jay Ray
Re: remove queue call
- From: Lee, John (Sydney)
Re: OT Polycom URI and IP address dialing. Not.
- From: Michael Graves
remove queue call
- From: Rilawich Ango
Re: Pri to sip interfaces
- From: Paul Hales
Re: Pri to sip interfaces
- From: Francisco del rosario
Re: Pri to sip interfaces
- From: Darren Sessions
Re: Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura
- From: Trevor Peirce
Re: Pri to sip interfaces
- From: Tom Moore
Re: Pri to sip interfaces
- From: Paul Hales
Re: Pri to sip interfaces
- From: Darren Sessions
Re: PRI Splitter
- From: George Pajari
Re: PRI Splitter
- From: Anciso, Roy
Pri to sip interfaces
- From: Tom Moore
Re: Newbie: Queue and CDR Reporter and Analyser
- From: Paul Hales
Re: PRI Splitter
- From: Paul Hales
Asterisk and Linksys One (PHB1100)
- From: George Pajari
Re: Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura
- From: Joseph
OT Polycom URI and IP address dialing. Not.
- From: randulo
Re: VUC Friday: asterisk 1.6 fax, Drawing for Free Astricon Pass
- From: randulo
Re: Problem with Call Forward
- From: Philipp Kempgen
Re: Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura
- From: RoLaNd RoLaNd
Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura
- From: Joseph
Problem with Call Forward
- From: Dpto. Datos Television Costa Blanca
Re: Atlanta Asterisk User's Group Conference Tonight Tuesday, August 26th at 7PM EDT
- From: Jay R. Ashworth
Re: VUC Friday: asterisk 1.6 fax, Drawing for Free Astricon Pass
- From: Tzafrir Cohen
Re: Fax issue over cisco gateway
- From: Atis Lezdins
Re: problem making outgoing calls
- From: Tzafrir Cohen
Fax issue over cisco gateway
- From: Enrico Pasqualotto
Re: Asterisk Realtime pounds MySQL
- From: Tilghman Lesher
Callback voice Quality
- From: michel freiha
Re: PRI Splitter
- From: Jeremy Mann
VUC Friday: asterisk 1.6 fax, Drawing for Free Astricon Pass
- From: randulo
Re: Asterisk Realtime pounds MySQL
- From: Tilghman Lesher
Re: FreeTDS Versions?
- From: Tony Mountifield
problem making outgoing calls
- From: bikrish
Re: sip show peers from shell or from CLI [SOLVED]
- From: Olivier
Re: Asterisk Realtime pounds MySQL
- From: J.M.
Call Files
- From: Andy Dixon
Re: PRI Splitter
- From: Kevin P. Fleming
Re: sip show peers from shell or from CLI
- From: Steven Howes
PRI Splitter
- From: Jeremy Mann
Re: sip show peers from shell or from CLI
- From: Olivier
Re: Call transfer over IAX trunk
- From: Andrea Spadaccini
Re: is shared_lastcall available in 1.4
- From: Bob Pierce
Re: DUNDI Help
- From: Bruce Reeves
Re: Codec and CPU load
- From: Miguel Otamendi
Re: Need application, CID number match list to call cell phone
- From: JR Richardson
Re: sip show peers from shell or from CLI
- From: Steven Howes
Re: sip show peers from shell or from CLI
- From: Olivier
Re: sip show peers from shell or from CLI
- From: Olivier
Re: sip show peers from shell or from CLI
- From: Steven Howes
sip show peers from shell or from CLI
- From: Olivier
Re: DUNDI Help
- From: ronald ramos
asterisk-1.6, Remote-Party-ID Header not sent
- From: Alexander Zielke
Re: Newbie: Queue and CDR Reporter and Analyser
- From: Lee, John (Sydney)
Re: is shared_lastcall available in 1.4
- From: Atis Lezdins
Re: compile Dahdi !
- From: Tzafrir Cohen
Re: Digium Coffee anyone? PCI Expresso? WTF?
- From: Anthony Messina
Re: asterisk realtime
- From: Szasz Szabolcs
Re: Need application, CID number match list to call cell phone
- From: Gordon Henderson
Re: Codec and CPU load
- From: Gordon Henderson
compile Dahdi !
- From: lizhong zhu
Asterisk for calling no of users
- From: Samir Ghodasara
Digium Coffee anyone? PCI Expresso? WTF?
- From: Karl Fife
Re: X100P Card in OFFHOOK state
- From: Eric \"ManxPower\" Wieling
Re: X100P Card in OFFHOOK state
- From: Guillermo Salas M.
Re: X100P Card in OFFHOOK state
- From: Jay Ray
Re: Asterisk connected to the PSTN vs. a commercial solution
- From: Noah Miller
Re: Limit to the length of string ?
- From: Paul Hales
Re: Need application, CID number match list to call cell phone
- From: Matt Gibson
Re: Need application, CID number match list to call cell phone
- From: Tilghman Lesher
Re: Atlanta Asterisk User's Group Conference Tonight Tuesday, August 26th at 7PM EDT
- From: SIP
Re: Need application, CID number match list to call cell phone
- From: Bruce Reeves
Need application, CID number match list to call cell phone
- From: JR Richardson
Codec and CPU load
- From: aymen warfalli
[OT] Re: sip peering between 2 asterisk
- From: Philipp Kempgen
FreeTDS Versions?
- From: Norman Franke
Re: sip peering between 2 asterisk
- From: Tariq ..
app_jack and calling with pc only
- From: Julien Claassen
Re: Asterisk connected to the PSTN vs. a commercial solution
- From: JD
Re: implementing an intercom with asterisk
- From: Matt Riddell
Re: Atlanta Asterisk User's Group Conference Tonight Tuesday, August 26th at 7PM EDT
- From: Jay R. Ashworth
Atlanta Asterisk User's Group Conference Tonight Tuesday, August 26th at 7PM EDT
- From: Asterisk
Re: Limit to the length of string ?
- From: Steven Howes
Re: Limit to the length of string ?
- From: Gordon Henderson
Re: implementing an intercom with asterisk
- From: Gordon Henderson
Re: implementing an intercom with asterisk
- From: Bruce Reeves
Re: DUNDI Help
- From: Bruce Reeves
Re: implementing an intercom with asterisk
- From: Jonathan Disher
Re: implementing an intercom with asterisk
- From: Jonathan Disher
Re: DUNDI Help
- From: ronald ramos
Re: DUNDI Help
- From: Bruce Reeves
Re: Limit to the length of string ?
- From: Drew Gibson
Re: Call transfer over IAX trunk
- From: Noah Miller
Re: Asterisk connected to the PSTN vs. a commercial solution
- From: Noah Miller
Re: is shared_lastcall available in 1.4
- From: Bob Pierce
X100P Card in OFFHOOK state
- From: Jay Ray
te410p remains in red-alarm
- From: John Harragin
Re: Limit to the length of string ?
- From: Tilghman Lesher
Re: TDM2400P Voice Quality Problem
- From: Noah Miller
Re: is shared_lastcall available in 1.4
- From: Atis Lezdins
Asterisk connected to the PSTN vs. a commercial solution
- From: Alejandro Cabrera Obed
Re: sip peering between 2 asterisk
- From: Nhadie
Re: is shared_lastcall available in 1.4
- From: Bob Pierce
Re: implementing an intercom with asterisk
- From: Steve Totaro
Re: is shared_lastcall available in 1.4
- From: Atis Lezdins
Re: is shared_lastcall available in 1.4
- From: Bob Pierce
Re: Limit to the length of string ?
- From: Gordon Henderson
Re: Limit to the length of string ?
- From: Tilghman Lesher
Re: sip peering between 2 asterisk
- From: Phil Thompson
Re: implementing an intercom with asterisk
- From: Drew Gibson
Re: Limit to the length of string ?
- From: Steve Totaro
Re: Limit to the length of string ?
- From: Tilghman Lesher
Re: implementing an intercom with asterisk
- From: Chris Mason (Lists)
Re: Get call status and hangup
- From: Andrea Spadaccini
Limit to the length of string ?
- From: Gordon Henderson
Re: Dial Plan Help
- From: Jon Weisman
Re: implementing an intercom with asterisk
- From: Gordon Henderson
Asterisk/Other PBX interconnection
- From: ims.asuser ims.asuser
DUNDI Help
- From: ronald ramos
about iax2 prune realtime
- From: yh_gwd
Re: sip peering between 2 asterisk
- From: Nhadie
Re: implementing an intercom with asterisk
- From: Darren Wiebe
Re: implementing an intercom with asterisk
- From: Robin Rodriguez
implementing an intercom with asterisk
- From: Jonathan Disher
Re: Asterisk Realtime pounds MySQL
- From: Rob Hillis
Re: sip peering between 2 asterisk
- From: Nhadie
Re: ztd-ethmf
- From: Carlos Rojas
Re: sip peering between 2 asterisk
- From: Tariq ..
Re: Asterisk Realtime pounds MySQL
- From: Tilghman Lesher
Re: is shared_lastcall available in 1.4
- From: Bob Pierce
Re: is shared_lastcall available in 1.4
- From: Mark Michelson
is shared_lastcall available in 1.4
- From: Bob Pierce
Re: Asterisk Realtime pounds MySQL
- From: J.M.
Re: Asterisk Realtime pounds MySQL
- From: J.M.
Re: How to query a remote MySQL DB from dialplan
- From: Al Baker
Re: Asterisk Realtime pounds MySQL
- From: Al Baker
Re: Asterisk Realtime pounds MySQL
- From: J.M.
Re: OT - Which rackable case for mini-ITX boards ?
- From: --[ UxBoD ]--
Re: How to query a remote MySQL DB from dialplan
- From: Michiel van Baak
Re: OT - Which rackable case for mini-ITX boards ?
- From: Alan Lord
sip.conf templates and realtime
- From: Charles R. Wadsworth
How to query a remote MySQL DB from dialplan
- From: Rich
Re: subscribemwi
- From: Philipp Kempgen
Get call status and hangup
- From: Loic Didelot
Re: subscribemwi
- From: Russell Bryant
Call transfer over IAX trunk
- From: Andrea Spadaccini
Re: asterisk realtime
- From: Tilghman Lesher
Call transfer over IAX trunk
- From: Andrea Spadaccini
subscribemwi (was: Re: MWI working perfectly. Shouldn't it be broken??)
- From: Philipp Kempgen
Re: MWI working perfectly. Shouldn't it be broken??
- From: Russell Bryant
Re: Really WEIRD: can register but can not call!
- From: David Boyd
Re: Problems with D-channel (PRI)
- From: Chris Brentano
Re: Semi-OT Satellite?
- From: Andrew Kohlsmith (lists)
Re: Problems with D-channel (PRI)
- From: Jakub \"Arkon\" Syrek
Re: Really WEIRD: can register but can not call!
- From: ims.asuser ims.asuser
Problem with dtmf in voicemailmain
- From: Ruchir
Re: SECURITY QUESTION & SANITY CHECK
- From: Philipp Kempgen
sip peering between 2 asterisk
- From: Nhadie
Re: TDM2400P Voice Quality Problem
- From: --[ UxBoD ]--
Re: Global VoIP Calls?
- From: Gavin Henry
Re: Static IP for SIP?
- From: Ruchir
asterisk realtime
- From: Szasz Szabolcs
Re: OT - Which rackable case for mini-ITX boards ?
- From: Joseph L. Casale
Static IP for SIP?
- From: Shariq Khan