March 2010 Archives by date

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Starting: Mon Mar 1 04:22:45 CST 2010
Ending: Wed Mar 31 21:22:33 CDT 2010
Messages: 1385

• [asterisk-users] Is answer() necessary ?   jonas kellens
• [asterisk-users] Is answer() necessary ?   Håkon Nessjøen
• [asterisk-users] Premicell solutions?   Steve Davies
• [asterisk-users] Is answer() necessary ?   Ishfaq Malik
• [asterisk-users] Premicell solutions?   Gordon Henderson
• [asterisk-users] MeetMe and usernum   Emrah
• [asterisk-users] yahoo   Ciprian ARSENIE
• [asterisk-users] Swift from eagi, problems with prosody rate   equis software
• [asterisk-users] MeetMe and usernum   Steve Edwards
• [asterisk-users] rtcachefriends & qualify   jonas kellens
• [asterisk-users] Asterisk and Cisco DTMF   Szasz Szabolcs
• [asterisk-users] SPA3102 Firmware Upgrade via TFTP fails   Georghy
• [asterisk-users] rtcachefriends & qualify   Nic Colledge
• [asterisk-users] Attended transfer: transferring a call as soon as the destination starts ringing   A. B.
• [asterisk-users] Asterisk and Cisco DTMF   David Backeberg
• [asterisk-users] Asterisk and Cisco DTMF   David Backeberg
• [asterisk-users] MeetMe and usernum   David Backeberg
• [asterisk-users] Asterisk and Cisco DTMF   Szasz Szabolcs
• [asterisk-users] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject   DLeese at LStelcom.com
• [asterisk-users] Swift from eagi, problems with prosody rate   equis software
• [asterisk-users] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject   Armin Schindler
• [asterisk-users] Fwd: Erika DeBenedictis-Recommendation   drew einhorn
• [asterisk-users] MeetMe and usernum   Emrah
• [asterisk-users] Erika DeBenedictis-Recommendation   drew einhorn
• [asterisk-users] MeetMe and usernum   David Backeberg
• [asterisk-users] Unable to register a sip account with x-lite   Tim Culhane
• [asterisk-users] Unable to register a sip account with x-lite   Danny Nicholas
• [asterisk-users] Unable to register a sip account with x-lite   Ahmed Ossama
• [asterisk-users] Asterisk / Trixbox 2.6 Streaming MOH Problems   Gary T. Giesen
• [asterisk-users] Server response time   Juan C. Villa
• [asterisk-users] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject   Armin Schindler
• [asterisk-users] MeetMe and usernum   Steve Edwards
• [asterisk-users] No RTP from asterisk?   --[ UxBoD ]--
• [asterisk-users] Server response time   Administrator TOOTAI
• [asterisk-users] OT:4 Line DECT Cordless phone without answering machine   C F
• [asterisk-users] OT:4 Line DECT Cordless phone without answering machine   Gordon Henderson
• [asterisk-users] Solved:Re: OT:4 Line DECT Cordless phone without answering machine   C F
• [asterisk-users] OT:4 Line DECT Cordless phone without answering machine   C F
• [asterisk-users] User on PC?   Leif Neland
• [asterisk-users] OT:4 Line DECT Cordless phone without answering machine   Michael Graves
• [asterisk-users] User on PC?   Gordon Henderson
• [asterisk-users] help!!! Internal extensions not connect   carem gyssell nieto
• [asterisk-users] User on PC?   Jared Smith
• [asterisk-users] help!!! Internal extensions not connect   lesouvage
• [asterisk-users] User on PC?   Stelios Koroneos
• [asterisk-users] help with install   Givon Zirkind
• [asterisk-users] help with install   Givon Zirkind
• [asterisk-users] Is answer() necessary ?   sean darcy
• [asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid   D Tucny
• [asterisk-users] help!!! Internal extensions not connect   Ioan Indreias
• [asterisk-users] Is answer() necessary ?   Ioan Indreias
• [asterisk-users] Does Asterisk 1.6.2.1 Support SIP TLS encryption   Zhang Shukun
• [asterisk-users] SIP Trunk with "multiple" remote ip-addresses   Magnus Benngård
• [asterisk-users] Server response time   Gordon Henderson
• [asterisk-users] Does Asterisk 1.6.2.1 Support SIP TLS encryption   Klaus Darilion
• [asterisk-users] SIP Trunk with "multiple" remote ip-addresses   Klaus Darilion
• [asterisk-users] rtcachefriends & qualify   jonas kellens
• [asterisk-users] MeetMe and usernum   Tony Mountifield
• [asterisk-users] Echo cancellation on DAHDI   DHAVAL INDRODIYA
• [asterisk-users] rtcachefriends & qualify   Mindaugas Kezys
• [asterisk-users] 1.4 chan_sip use internal IP for dialog-info+xml SUBSCRIBE, why?   Kristijan Vrban
• [asterisk-users] rtcachefriends & qualify & sip reload   jonas kellens
• [asterisk-users] Sip module problem   Luis Silva
• [asterisk-users] rtcachefriends & qualify & sip reload   Ishfaq Malik
• [asterisk-users] cli_originate malfunction after upgrade from 1.6.2.0 to 1.6.2.1-5   Andreas Brodmann
• [asterisk-users] Sip module problem   Philipp von Klitzing
• [asterisk-users] Echo cancellation on DAHDI   Vinícius Fontes
• [asterisk-users] SIP Trunk with "multiple" remote ip-addresses   Magnus Benngård
• [asterisk-users] dialplan reload: not working with large dialplans   Andreas Brodmann
• [asterisk-users] rtcachefriends & qualify & sip reload   jonas kellens
• [asterisk-users] help!!! Internal extensions not connect   carem gyssell nieto
• [asterisk-users] help!!! Internal extensions not connect   Tzafrir Cohen
• [asterisk-users] dialplan reload: not working with large dialplans   Tzafrir Cohen
• [asterisk-users] dialplan reload: not working with large dialplans   Andreas Brodmann
• [asterisk-users] rtcachefriends & qualify & sip reload   Ishfaq Malik
• [asterisk-users] X-Lite won't register   carem gyssell nieto
• [asterisk-users] Echo cancellation on DAHDI   Brian
• [asterisk-users] rtcachefriends & qualify & sip reload   jonas kellens
• [asterisk-users] Hide time consuming processed by prompt   Patrick
• [asterisk-users] rtcachefriends & qualify & sip reload   Mindaugas Kezys
• [asterisk-users] Hide time consuming processed by prompt   Steve Edwards
• [asterisk-users] Echo cancellation on DAHDI   Vinícius Fontes
• [asterisk-users] Server response time   Juan C. Villa
• [asterisk-users] Hide time consuming processed by prompt   Kevin P. Fleming
• [asterisk-users] Echo cancellation on DAHDI   Brian
• [asterisk-users] rtcachefriends & qualify & sip reload   Ishfaq Malik
• [asterisk-users] Echo cancellation on DAHDI   Gordon Henderson
• [asterisk-users] realtime call peers status   lore
• [asterisk-users] Server response time   Juan C. Villa
• [asterisk-users] Server response time   Gordon Henderson
• [asterisk-users] Echo cancellation on DAHDI   Brian
• [asterisk-users] realtime call peers status   Ishfaq Malik
• [asterisk-users] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject - Bayesian Filter detected spam   DLeese at LStelcom.com
• [asterisk-users] Echo cancellation on DAHDI   Brian
• [asterisk-users] ARI problem with monitor   Luis campo
• [asterisk-users] Echo cancellation on DAHDI   Carlos Chavez
• [asterisk-users] realtime call peers status   lore
• [asterisk-users] rtcachefriends & qualify & sip reload   Carlos Chavez
• [asterisk-users] asterisk-users Digest, Vol 68, Issue 4   Luis Silva
• [asterisk-users] Echo cancellation on DAHDI   Brian
• [asterisk-users] MWI and 1.6.1   Dave Poirier
• [asterisk-users] Asterisk and cellphone/GSM voicemailbox   jonas kellens
• [asterisk-users] Asterisk and cellphone/GSM voicemailbox   Fred Posner
• [asterisk-users] FW: ARI problem with monitor   Luis campo
• [asterisk-users] dialplan reload: not working with large dialplans   Warren Selby
• [asterisk-users] Uverse, Asterisk and SIP   sean darcy
• [asterisk-users] Uverse, Asterisk and SIP   Fred Posner
• [asterisk-users] dialplan reload: not working with large dialplans   Andreas Brodmann
• [asterisk-users] Dial timeout problem with OpenVox A1200P Card / FXS module   Fábio da Silva Cunha
• [asterisk-users] Uverse, Asterisk and SIP   sean darcy
• [asterisk-users] Uverse, Asterisk and SIP   Warren Selby
• [asterisk-users] asterisk-users] how to create a dummy call   Pham Quy
• [asterisk-users] how to play background music during record   Pham Quy
• [asterisk-users] Echo cancellation on DAHDI   DHAVAL INDRODIYA
• [asterisk-users] dahdi and oslec   Chandrakant Solanki
• [asterisk-users] dahdi and oslec   wins mallow
• [asterisk-users] dahdi and oslec   wins mallow
• [asterisk-users] rtcachefriends & qualify & sip reload   Mindaugas Kezys
• [asterisk-users] realtime call peers status   Ishfaq Malik
• [asterisk-users] rtcachefriends & qualify & sip reload   Ishfaq Malik
• [asterisk-users] dahdi and oslec   wins mallow
• [asterisk-users] Getting verbose or debug tracing in Asterisk   Tim Culhane
• [asterisk-users] how can I release trunks after transferring 2 calls connected on trunks between the same machines.   Raoul Trevisi
• [asterisk-users] Echo cancellation on DAHDI   Vinícius Fontes
• [asterisk-users] cli_originate malfunction after upgrade from 1.6.2.0 to 1.6.2.1-5   Tzafrir Cohen
• [asterisk-users] dialplan reload: not working with large dialplans   Tzafrir Cohen
• [asterisk-users] Getting verbose or debug tracing in Asterisk   Jim Dickenson
• [asterisk-users] Getting verbose or debug tracing in Asterisk   Danny Nicholas
• [asterisk-users] dahdi and oslec   Danny Nicholas
• [asterisk-users] CALLERID(num) not working   Jim Dickenson
• [asterisk-users] dahdi and oslec   Tzafrir Cohen
• [asterisk-users] Asterisk and cellphone/GSM voicemailbox   jonas kellens
• [asterisk-users] Is this a bug?   Danny Nicholas
• [asterisk-users] Getting verbose or debug tracing in Asterisk   Tim Culhane
• [asterisk-users] Getting verbose or debug tracing in Asterisk   Danny Nicholas
• [asterisk-users] Getting verbose or debug tracing in Asterisk   Tim Culhane
• [asterisk-users] Deadlock while using MGCP on Asterisk   Adrien Lemoine
• [asterisk-users] Getting verbose or debug tracing in Asterisk   Barry L. Kline
• [asterisk-users] forward problem!   BERGANZ Francois
• [asterisk-users] Getting verbose or debug tracing in Asterisk   Tim Culhane
• [asterisk-users] Echo cancellation on DAHDI   Carlos Chavez
• [asterisk-users] 911, channel full   mir shahnawaz
• [asterisk-users] Best practise for ISDN Video Conferencing..   Mark Adams
• [asterisk-users] asterisk SIP, SIPAddHeader() and Cisco GED-125   David Backeberg
• [asterisk-users] 911, channel full   Steve Howes
• [asterisk-users] Best practise for ISDN Video Conferencing..   Vinícius Fontes
• [asterisk-users] Uverse, Asterisk and SIP   sean darcy
• [asterisk-users] Uverse, Asterisk and SIP   Warren Selby
• [asterisk-users] Uverse, Asterisk and SIP   Fred Posner
• [asterisk-users] 911, channel full   mir shahnawaz
• [asterisk-users] Free 'Locked up' Channels   Brian Chamberlain
• [asterisk-users] Looking for a configuration guru to collaborate with   Philip A. Prindeville
• [asterisk-users] Best practise for ISDN Video Conferencing..   Mark Adams
• [asterisk-users] Identify scripts connecting to the asterisk manager   Jason Marble
• [asterisk-users] forward problem!   Dave Poirier
• [asterisk-users] CallerID and distinctive ring detection   Barry Miller
• [asterisk-users] Best practise for ISDN Video Conferencing..   Alec Davis
• [asterisk-users] how to create a dummy call   Pham Quy
• [asterisk-users] how to create a dummy call   Pham Quy
• [asterisk-users] how to create a dummy call   Pascal Bruno
• [asterisk-users] [asterisk-user] SIP / Echo Cancellation   Chandrakant Solanki
• [asterisk-users] No Audio on pstn call   Siti Zalifah Md Yatim
• [asterisk-users] No Audio on pstn call   Siti Zalifah Md Yatim
• [asterisk-users] how to create a dummy call   Tilghman Lesher
• [asterisk-users] Getting: Can't fix up channel from 5 to 7 because 7 is already in use, and pri_dchannel: Answer requested on channel 0/7 not in use on span 1   Håkon Nessjøen
• [asterisk-users] anonymous   Ciprian ARSENIE
• [asterisk-users] Asterisk and cellphone/GSM voicemailbox   jonas kellens
• [asterisk-users] SIP / Echo Cancellation   Chandrakant Solanki
• [asterisk-users] Identify scripts connecting to the asterisk manager   Tzafrir Cohen
• [asterisk-users] Caller ID in Asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] Best practise for ISDN Video Conferencing..   Mark Adams
• [asterisk-users] Caller ID in Asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] Caller ID in Asterisk   Jimmy Godbout
• [asterisk-users] Caller ID in Asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] Best practise for ISDN Video Conferencing..   Vinícius Fontes
• [asterisk-users] Caller ID in Asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] Availstatus returns 20 ?   jonas kellens
• [asterisk-users] Asterisk & Sofaware & Polycom   Darrin Henshaw
• [asterisk-users] UK CallerID -v- Wildcard W100P   Brian
• [asterisk-users] UK CallerID -v- Wildcard W100P   Gordon Henderson
• [asterisk-users] No Audio on pstn call   LATEEF, IRFAN (ATTSI)
• [asterisk-users] time/date over POTS?   Jeff LaCoursiere
• [asterisk-users] UK CallerID -v- Wildcard W100P   Brian
• [asterisk-users] UK CallerID -v- Wildcard W100P   Gordon Henderson
• [asterisk-users] time/date over POTS?   Dave Fullerton
• [asterisk-users] UK CallerID -v- Wildcard W100P   Ade Vickers
• [asterisk-users] UK CallerID -v- Wildcard W100P   Brian
• [asterisk-users] Hardware   Aditya Kumar
• [asterisk-users] Hardware   Steve Howes
• [asterisk-users] No Audio on pstn call   Siti Zalifah Md Yatim
• [asterisk-users] Hardware   Siti Zalifah Md Yatim
• [asterisk-users] time/date over POTS?   Jeff LaCoursiere
• [asterisk-users] time/date over POTS?   Dave Fullerton
• [asterisk-users] InterPBX communication using SIP   khalid touati
• [asterisk-users] 30 mins GSM file   David at ULC
• [asterisk-users] 30 mins GSM file   Tim Nelson
• [asterisk-users] 30 mins GSM file   Danny Nicholas
• [asterisk-users] 30 mins GSM file   Steve Edwards
• [asterisk-users] 30 mins GSM file   David at ULC
• [asterisk-users] time/date over POTS?   Warren Selby
• [asterisk-users] 30 mins GSM file   David at ULC
• [asterisk-users] 30 mins GSM file   Steve Edwards
• [asterisk-users] 30 mins GSM file   Steve Howes
• [asterisk-users] 30 mins GSM file   Carlos Chavez
• [asterisk-users] Remote Agents   Matt
• [asterisk-users] Remote Agents   Steve Edwards
• [asterisk-users] Remote Agents   Matt
• [asterisk-users] 30 mins GSM file   Hans Witvliet
• [asterisk-users] PHPAGI and Asterisk 1.6   Carlos Chavez
• [asterisk-users] dialplan reload: not working with large dialplans   Leif Madsen
• [asterisk-users] Remote Agents   Leif Madsen
• [asterisk-users] cli_originate malfunction after upgrade from 1.6.2.0 to 1.6.2.1-5   Leif Madsen
• [asterisk-users] MWI and 1.6.1   Matt Watson
• [asterisk-users] SIP / Echo Cancellation   Vinícius Fontes
• [asterisk-users] Remote Agents   Matt
• [asterisk-users] 30 mins GSM file   Tilghman Lesher
• [asterisk-users] Codec translation in Asterisk   Asterisk User
• [asterisk-users] SIP / Echo Cancellation   Vineet Bhojnagarwala
• [asterisk-users] Caller ID in Asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] iLBC installation problem   nedo nodo
• [asterisk-users] Asterisk Management API   Peter Childs
• [asterisk-users] Having problems with BLF   John
• [asterisk-users] Deadlock in Asterisk 1.4.29.1   Adrien Lemoine
• [asterisk-users] FAX configuration for DAHDI lines   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] MGCP FXO endpoint   Ignacio
• [asterisk-users] FAX configuration for DAHDI lines   Peter Gelencser
• [asterisk-users] FAX configuration for DAHDI lines   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] Caller ID in Asterisk   Jimmy Godbout
• [asterisk-users] Caller ID in Asterisk   Peter Gelencser
• [asterisk-users] FAX configuration for DAHDI lines   Peter Gelencser
• [asterisk-users] Asterisk Management API   Jim Dickenson
• [asterisk-users] 30 mins GSM file   Danny Nicholas
• [asterisk-users] Playback in h extension   Anahi Ludueña
• [asterisk-users] SIP / Echo Cancellation   Steve Underwood
• [asterisk-users] Playback in h extension   Danny Nicholas
• [asterisk-users] Follow-up to CALLERID(num) not working   Jim Dickenson
• [asterisk-users] Having problems with BLF   Philipp von Klitzing
• [asterisk-users] Deadlock in Asterisk 1.4.29.1   Moises Silva
• [asterisk-users] Having problems with BLF   John
• [asterisk-users] Asterisk 1.4 Followme Question   Cory Andrews
• [asterisk-users] SIP / Echo Cancellation   Vinícius Fontes
• [asterisk-users] 30 mins GSM file   Jeff LaCoursiere
• [asterisk-users] FollowMe / Asterisk 1.4 Question   Cory Andrews
• [asterisk-users] Deadlock in Asterisk 1.4.29.1   Adrien Lemoine
• [asterisk-users] 30 mins GSM file   Vinícius Fontes
• [asterisk-users] Hardware requirements question.   David Little
• [asterisk-users] AMI logs   Anahi Ludueña
• [asterisk-users] iLBC installation problem   Philipp von Klitzing
• [asterisk-users] Denial of Service Attack   Dan Journo
• [asterisk-users] NeoSpeech & Asterisk?   Byron J. Lee
• [asterisk-users] InterPBX communication using SIP   khalid touati
• [asterisk-users] Hardware requirements question.   Tim Nelson
• [asterisk-users] 30 mins GSM file   David at ULC
• [asterisk-users] Denial of Service Attack   Tilghman Lesher
• [asterisk-users] Hardware requirements question.   Gordon Henderson
• [asterisk-users] 30 mins GSM file   Steve Edwards
• [asterisk-users] Hardware requirements question.   Steve Edwards
• [asterisk-users] State of 64 bits applications in Asterisk   Administrator TOOTAI
• [asterisk-users] FollowMe / Asterisk 1.4 Question   Warren Selby
• [asterisk-users] Regarding - P-Asserted identity   das sandesh
• [asterisk-users] Hardware requirements question.   Gordon Henderson
• [asterisk-users] AMI logs   Jim Dickenson
• [asterisk-users] State of 64 bits applications in Asterisk   Jordan Kirby
• [asterisk-users] Observation about DAHDI, FAX and Echo cancellation   Håkon Nessjøen
• [asterisk-users] Uverse, Asterisk and SIP   sean darcy
• [asterisk-users] Observation about DAHDI, FAX and Echo cancellation   Vinícius Fontes
• [asterisk-users] Having problems with BLF   John
• [asterisk-users] Uverse, Asterisk and SIP   Fred Posner
• [asterisk-users] NeoSpeech & Asterisk?   LATEEF, IRFAN (ATTSI)
• [asterisk-users] app_confbridge production ready?   Robert McGilvray
• [asterisk-users] Hardware requirements question.   Christian Victor
• [asterisk-users] Playback in h extension   Christian Victor
• [asterisk-users] Asterisk 1.4 Followme Question   Dovey Forman
• [asterisk-users] Having problems with BLF   Philipp von Klitzing
• [asterisk-users] MOH Oddity   Matt
• [asterisk-users] MOH Oddity   Matt
• [asterisk-users] Having problems with BLF   Philipp von Klitzing
• [asterisk-users] MOH Oddity   Håkon Nessjøen
• [asterisk-users] MOH Oddity   Matt
• [asterisk-users] MOH Oddity   Steve Edwards
• [asterisk-users] MOH Oddity   Tilghman Lesher
• [asterisk-users] SIPit 26 in Sweden - organized by Edvina   Olle E. Johansson
• [asterisk-users] Having problems with BLF   John
• [asterisk-users] Audio problems ins conference zap->sip   Sara Pavón
• [asterisk-users] MOH Oddity   Matt
• [asterisk-users] Having problems with BLF   John
• [asterisk-users] MOH over IAX2 - NOT working   Joseph
• [asterisk-users] SIP, internet calling, per-peer contexts, and multitenancy   Philip A. Prindeville
• [asterisk-users] saving pressed keys   Necati Demir
• [asterisk-users] Denial of Service Attack   Dan Journo
• [asterisk-users] saving pressed keys   Steve Edwards
• [asterisk-users] MOH over IAX2 - NOT working   Alex Balashov
• [asterisk-users] 30 mins GSM file   Tzafrir Cohen
• [asterisk-users] Remote Agents   Tzafrir Cohen
• [asterisk-users] Custom App   Sascha Ferley
• [asterisk-users] Mail-2-Fax and Fax-2-Mail solution for Asterisk with T38   Thorolf Godawa
• [asterisk-users] Custom App   Steve Edwards
• [asterisk-users] dahdi-2.2.1 & kernel-2.6.32: working for anyone?   sean darcy
• [asterisk-users] dahdi-2.2.1 & kernel-2.6.32: working for anyone?   Anthony Messina
• [asterisk-users] dahdi-2.2.1 & kernel-2.6.32: working for anyone?   Tzafrir Cohen
• [asterisk-users] indications.conf   ciprian at carsenie.ro
• [asterisk-users] saving pressed keys   Necati Demir
• [asterisk-users] indications.conf   Ciprian ARSENIE
• [asterisk-users] Mail-2-Fax and Fax-2-Mail solution for Asterisk with T38   Zoa
• [asterisk-users] dahdi-2.2.1 & kernel-2.6.32: working for anyone?   Eckhard Jokisch
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.   Zoa
• [asterisk-users] indications.conf   Ciprian ARSENIE
• [asterisk-users] Callcenter open source program   wassim darwich
• [asterisk-users] Callcenter open source program   Bayardo Sanchez
• [asterisk-users] indications.conf   Lyle Giese
• [asterisk-users] dahdi-2.2.1 & kernel-2.6.32: working for anyone?   sean darcy
• [asterisk-users] saving pressed keys   Steve Edwards
• [asterisk-users] Caller Presentation Confusion   Jim Dickenson
• [asterisk-users] indications.conf   Ciprian ARSENIE
• [asterisk-users] Attended transfer broken in 1.6.0.25   Theo Band
• [asterisk-users] Denial of Service Attack   Dan Journo
• [asterisk-users] Caller Presentation Confusion   Gordon Henderson
• [asterisk-users] Asterisk Redundancy   Dan Journo
• [asterisk-users] Asterisk Redundancy   Steve Totaro
• [asterisk-users] Caller Presentation Confusion   Jim Dickenson
• [asterisk-users] FollowMe / Asterisk 1.4 Question   BJ Weschke
• [asterisk-users] Asterisk Redundancy   Dan Journo
• [asterisk-users] Grandstream HT 503 Outoing 403 Forbidden   Jim Rosenberg
• [asterisk-users] Mail-2-Fax and Fax-2-Mail solution for Asterisk with T38   Grygoriy Dobrovolskyy
• [asterisk-users] dahdi-2.2.1 & kernel-2.6.32: working for anyone?   Anthony Messina
• [asterisk-users] Grandstream HT 503 Outoing 403 Forbidden   Edoardo Sabadelli
• [asterisk-users] dahdi-2.2.1 & kernel-2.6.32: working for anyone?   sean darcy
• [asterisk-users] Asterisk Redundancy   Steve Totaro
• [asterisk-users] Followme broken   Sean Brady
• [asterisk-users] dahdi-2.2.1 & kernel-2.6.32: working for anyone?   Anthony Messina
• [asterisk-users] Callcenter open source program   Edwin Quijada
• [asterisk-users] Attended transfer broken in 1.6.0.25   Leif Madsen
• [asterisk-users] Having problems with BLF   Philipp von Klitzing
• [asterisk-users] dahdi not available in Asterisk   Klaverstyn, David C
• [asterisk-users] Attended transfer broken in 1.6.0.25   Theo Band
• [asterisk-users] Dail of meetme options   Arjan Kroon | Mobillion
• [asterisk-users] Having problems with BLF   John
• [asterisk-users] Play an audio file from a remote host   Pham Quy
• [asterisk-users] Play an audio file from a remote host   Pascal Bruno
• [asterisk-users] Followme broken   --[ UxBoD ]--
• [asterisk-users] Calculating R Factor and MOS metrics for VoIP   mosbah.abdelkader
• [asterisk-users] Asterisk Management API   Peter Childs
• [asterisk-users] dahdi not available in Asterisk   Grygoriy Dobrovolskyy
• [asterisk-users] Followme broken   Sean Brady
• [asterisk-users] dahdi not available in Asterisk   Doug Lytle
• [asterisk-users] Calculating R Factor and MOS metrics for VoIP   Marc LEURENT
• [asterisk-users] dahdi not available in Asterisk   Klaus Darilion
• [asterisk-users] Calculating R Factor and MOS metrics for VoIP   Klaus Darilion
• [asterisk-users] Free 'Locked up' Channels   David Gibbons
• [asterisk-users] Is it possible to configure Asterisk so that it does the Q.SIG “Path Replacement Feature” ?   Raoul Trevisi
• [asterisk-users] Asterisk Management API   Olle E. Johansson
• [asterisk-users] Calculating R Factor and MOS metrics for VoIP   Steve Underwood
• [asterisk-users] Calculating R Factor and MOS metrics for VoIP   Kevin P. Fleming
• [asterisk-users] Turning off DNIS on T1 set to FXO_LS protocol   Dean Hoover
• [asterisk-users] Calculating R Factor and MOS metrics for VoIP   Mindaugas Kezys
• [asterisk-users] Calculating R Factor and MOS metrics for VoIP   Steve Underwood
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.   Matt Watson
• [asterisk-users] dialplan reload: not working with large dialplans   Andreas Brodmann
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.   Matt Watson
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.   Zoa
• [asterisk-users] Voip Users Conference March 26th   Randy R
• [asterisk-users] asterisk SIP, SIPAddHeader() and Cisco GED-125   Franklin Webb
• [asterisk-users] Getting: Can't fix up channel from 5 to 7 because 7 is already in use, and pri_dchannel: Answer requested on channel 0/7 not in use on span 1   Håkon Nessjøen
• [asterisk-users] Turning off DNIS on T1 set to FXO_LS protocol   Kevin P. Fleming
• [asterisk-users] asterisk SIP, SIPAddHeader() and Cisco GED-125   David Backeberg
• [asterisk-users] Dialplan behaviour   equis software
• [asterisk-users] Dialplan behaviour   Danny Nicholas
• [asterisk-users] MWI and 1.6.1   Dave Poirier
• [asterisk-users] Uverse, Asterisk and SIP   sean darcy
• [asterisk-users] fax & spandsp   Edwin Lam
• [asterisk-users] fax & spandsp   Philip A. Prindeville
• [asterisk-users] SIP handset + SLA example   Philip A. Prindeville
• [asterisk-users] SIP to IAX to SIP Jitterbuffer question   Karl Fife
• [asterisk-users] Aastra, Asterisk 1.4 and Voicemail   Mike
• [asterisk-users] Having problems with BLF   Philipp von Klitzing
• [asterisk-users] fax & spandsp   Edwin Lam
• [asterisk-users] Having problems with BLF   John
• [asterisk-users] Aastra, Asterisk 1.4 and Voicemail   Bob Pierce
• [asterisk-users] FAX configuration for DAHDI lines   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] Asterisk Management API   Peter Childs
• [asterisk-users] Asterisk Management API   Peter Childs
• [asterisk-users] MWI and 1.6.1   Will Payne
• [asterisk-users] DUNDI Sip authentication failure   Georghy
• [asterisk-users] Callcenter open source program   Emanuele Carbone
• [asterisk-users] app_queue problem with Ringing state   Håkon Nessjøen
• [asterisk-users] FAX configuration for DAHDI lines   Håkon Nessjøen
• [asterisk-users] FAX configuration for DAHDI lines   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] FAX configuration for DAHDI lines   Håkon Nessjøen
• [asterisk-users] Asterisk 1.6.2.5 crash with chan_capi upon calling to PSTN   DLeese at LStelcom.com
• [asterisk-users] asterisk peer uses 5060 to send and 5061 to receive   Joao Gomes Pereira
• [asterisk-users] asterisk peer uses 5060 to send and 5061 to receive   Joao Gomes Pereira
• [asterisk-users] Disable echo canceller Fonebridge   spv spv
• [asterisk-users] MWI and 1.6.1   SIP
• [asterisk-users] MWI and 1.6.1   Will Payne
• [asterisk-users] CallerID presented in Asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] CallerID presented in Asterisk   Doug Lytle
• [asterisk-users] MWI and 1.6.1   SIP
• [asterisk-users] CallerID presented in Asterisk   Steve Howes
• [asterisk-users] Turning off DNIS on T1 set to FXO_LS protocol   Dean Hoover
• [asterisk-users] Aastra, Asterisk 1.4 and Voicemail   Mike
• [asterisk-users] Snom Provisioning   voip crazy
• [asterisk-users] Snom Provisioning   Philipp von Klitzing
• [asterisk-users] Snom Provisioning   Alexander Samad
• [asterisk-users] Snom Provisioning   Ishfaq Malik
• [asterisk-users] Uverse, Asterisk and SIP   Fred Posner
• [asterisk-users] confbridge manager/cli   Jonathan Addleman
• [asterisk-users] Snom Provisioning   --[ UxBoD ]--
• [asterisk-users] fax & spandsp   Klaus Darilion
• [asterisk-users] Asterisk SMDI for Nortel Option 11   Carlos Chavez
• [asterisk-users] Queue Member stuck in Ring+InUse?   William Stillwell (Lists)
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.   Klaus Darilion
• [asterisk-users] Which spandsp to use with 1.6.2?   sean darcy
• [asterisk-users] MWI and 1.6.1   Matt Watson
• [asterisk-users] CallerID presented in Asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] CLI not working properly - Asterisk Freez   Danny Dias
• [asterisk-users] call features affected by native bridging between sip phones   MURALI V
• [asterisk-users] fax & spandsp   Edwin Lam
• [asterisk-users] func odbc and mult iquery   voipas
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.   Zoa
• [asterisk-users] fax & spandsp   Tommy Botten Jensen
• [asterisk-users] callerid change name   Georghy
• [asterisk-users] Extensions.conf changed but not take effect   Zhang Shukun
• [asterisk-users] Extensions.conf changed but not take effect   Emanuele Carbone
• [asterisk-users] Extensions.conf changed but not take effect   Håkon Nessjøen
• [asterisk-users] I loose incoming call after transfer   jonas kellens
• [asterisk-users] Which spandsp to use with 1.6.2?   Tzafrir Cohen
• [asterisk-users] Passing a parameter to voicemail   Carlo Dimaggio
• [asterisk-users] callerid change name   Georghy
• [asterisk-users] CallerID presented in Asterisk   Steve Howes
• [asterisk-users] Passing a parameter to voicemail   Emanuele Carbone
• [asterisk-users] dtmf payload 100   Katerina Borin
• [asterisk-users] Extensions.conf changed but not take effect   Danny Nicholas
• [asterisk-users] Dialplan behaviour   equis software
• [asterisk-users] Dialplan behaviour   Danny Nicholas
• [asterisk-users] dialplan reload: not working with large dialplans   Leif Madsen
• [asterisk-users] Passing a parameter to voicemail   Carlo Dimaggio
• [asterisk-users] 00h323 cant get gatekeeper to connect   Michelle Dupuis
• [asterisk-users] multiple RTP port ranges for SIP   Michelle Dupuis
• [asterisk-users] multiple RTP port ranges for SIP   Klaus Darilion
• [asterisk-users] multiple RTP port ranges for SIP   Kevin P. Fleming
• [asterisk-users] 00h323 cant get gatekeeper to connect   Magnus Benngård
• [asterisk-users] BLF and realtime SIP buddies   jonas kellens
• [asterisk-users] 00h323 cant get gatekeeper to connect   Michelle Dupuis
• [asterisk-users] Dialplan behaviour   equis software
• [asterisk-users] func odbc and mult iquery   Tilghman Lesher
• [asterisk-users] app_confbridge production ready?   David Backeberg
• [asterisk-users] 00h323 cant get gatekeeper to connect   Magnus Benngård
• [asterisk-users] Extensions.conf changed but not take effect   Warren Selby
• [asterisk-users] 00h323 cant get gatekeeper to connect   Michelle Dupuis
• [asterisk-users] Extensions.conf changed but not take effect   Philipp von Klitzing
• [asterisk-users] PGSQL application   Vinícius Fontes
• [asterisk-users] Meetme Closes Conference After One Hour   Carlos A. Alvarez
• [asterisk-users] Extensions.conf changed but not take effect   Warren Selby
• [asterisk-users] PGSQL application   Tilghman Lesher
• [asterisk-users] Phishing attempt posing as digium   Thomas Kenyon
• [asterisk-users] Phishing attempt posing as digium   Darren Nickerson
• [asterisk-users] Asterisk SMDI for Nortel Option 11   James Noble
• [asterisk-users] Diaplan reload command not working   ayodele abejide
• [asterisk-users] Extensions.conf changed but not take effect   Philipp von Klitzing
• [asterisk-users] Asterisk SMDI for Nortel Option 11   Carlos Chavez
• [asterisk-users] Asterisk SMDI for Nortel Option 11   Cory Andrews
• [asterisk-users] PGSQL application   Vinícius Fontes
• [asterisk-users] confbridge manager/cli   Jonathan Addleman
• [asterisk-users] Diaplan reload command not working   Tilghman Lesher
• [asterisk-users] Asterisk SMDI for Nortel Option 11   Alec Davis
• [asterisk-users] Asterisk SMDI for Nortel Option 11   Carlos Chavez
• [asterisk-users] Diaplan reload command not working   ayodele abejide
• [asterisk-users] Diaplan reload command not working   Warren Selby
• [asterisk-users] Dialplan behaviour   C. Chad Wallace
• [asterisk-users] Diaplan reload command not working   ayodele abejide
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license   JR Richardson
• [asterisk-users] Unable to forward voice or dtmf   Alejandro Recarey
• [asterisk-users] Phones won't stop ringing   Chris Owen
• [asterisk-users] Diaplan reload command not working   Warren Selby
• [asterisk-users] Phones won't stop ringing   Warren Selby
• [asterisk-users] Phones won't stop ringing   Jeff Brower
• [asterisk-users] Phones won't stop ringing   Chris Owen
• [asterisk-users] Phones won't stop ringing   Jason Aarons (US)
• [asterisk-users] Phones won't stop ringing   cb
• [asterisk-users] How to add custom CDR fields to MySQL   Alejandro Recarey
• [asterisk-users] CallerID presented in Asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] CallerID presented in Asterisk   Tzafrir Cohen
• [asterisk-users] Diaplan reload command not working   Tzafrir Cohen
• [asterisk-users] Phones won't stop ringing   Tzafrir Cohen
• [asterisk-users] CallerID presented in Asterisk   Steve Howes
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license   Zoa
• [asterisk-users] Is there a way for a peer to clear its registration from a server?   Frank Church
• [asterisk-users] SIP Phone for conference room use.   Tommy Botten Jensen
• [asterisk-users] CallerID presented in Asterisk   Gopalakrishnaiyer Venugopal-Q16770
• [asterisk-users] app_queue problem with Ringing state   Håkon Nessjøen
• [asterisk-users] SIP Phone for conference room use.   Gordon Henderson
• [asterisk-users] SIP Phone for conference room use.   Will Payne
• [asterisk-users] CallerID presented in Asterisk   Tzafrir Cohen
• [asterisk-users] Is there a way for a peer to clear its registration from a server?   Kevin P. Fleming
• [asterisk-users] Testers Need Issue #0016965: [patch] DBGet response does not end with a 'Complete' event   Ryan Bullock
• [asterisk-users] Is there a way for a peer to clear its registration from a server?   Philipp von Klitzing
• [asterisk-users] Xorcom Astribank Versus Rhino ChannelBank   Robert Grignon
• [asterisk-users] SIP Phone for conference room use.   Michael Graves
• [asterisk-users] BLF and realtime SIP buddies   jonas kellens
• [asterisk-users] BLF and realtime SIP buddies   Danny Nicholas
• [asterisk-users] Is there a way for a peer to clear its registration from a server?   Olle E. Johansson
• [asterisk-users] Is there a way for a peer to clear its registration from a server?   Philipp von Klitzing
• [asterisk-users] Codec preference   jonas kellens
• [asterisk-users] Is there a way for a peer to clear its registration from a server?   Frank Church
• [asterisk-users] Diaplan reload command not working   Carlos Chavez
• [asterisk-users] Digium TE4xx T1 Bonding   Eric Wheeler
• [asterisk-users] Asterisk Management API   Matt Riddell
• [asterisk-users] Digium TE4xx T1 Bonding   Christian Victor
• [asterisk-users] Fwd: Switchvox SOHO 4.5 is Here   Angelito Manansala
• [asterisk-users] Fwd: Switchvox SOHO 4.5 is Here   Jeff LaCoursiere
• [asterisk-users] 00h323 cant get gatekeeper to connect   Michelle Dupuis
• [asterisk-users] Which spandsp to use with 1.6.2?   sean darcy
• [asterisk-users] 00h323 cant get gatekeeper to connect   Michelle Dupuis
• [asterisk-users] Running DEADAGI from h extension   Carlos Chavez
• [asterisk-users] Codec preference   Prince Singh
• [asterisk-users] How to add custom CDR fields to MySQL   Emanuele Carbone
• [asterisk-users] how to create a dummy call   Pham Quy
• [asterisk-users] Codec preference   jonas kellens
• [asterisk-users] Fri March 12th @ 12 noon EST: SIP scanning, security and attacks + Hosted vs on-site voip   Randy R
• [asterisk-users] Time counting down and # detect   Pham Quy
• [asterisk-users] Fwd: Switchvox SOHO 4.5 is Here   Lito Manansala
• [asterisk-users] Can not enable sip debug because CLI flooded   jonas kellens
• [asterisk-users] dtmf payload 100   Katerina Borin
• [asterisk-users] dtmf payload 100   Olle E. Johansson
• [asterisk-users] R: Can not enable sip debug because CLI flooded   Alexandru Oniciuc
• [asterisk-users] Time counting down and # detect   Pham Quy
• [asterisk-users] R: Can not enable sip debug because CLI flooded   Steve Howes
• [asterisk-users] R: Can not enable sip debug because CLI flooded   jonas kellens
• [asterisk-users] Time counting down and # detect   Gordon Henderson
• [asterisk-users] multiple RTP port ranges for SIP   Klaus Darilion
• [asterisk-users] Codec preference   jonas kellens
• [asterisk-users] SIP Trunk with "multiple" remote ip-addresses   Klaus Darilion
• [asterisk-users] Phones won't stop ringing   Phil Reynolds
• [asterisk-users] SIP Trunk with "multiple" remote ip-addresses   Olle E. Johansson
• [asterisk-users] func odbc and mult iquery   voipas
• [asterisk-users] 1.2 to 1.6 and bristuff   Steve Davies
• [asterisk-users] Asterisk 1.2 crash: gdb trace on core dump   Vieri
• [asterisk-users] Codec preference   jonas kellens
• [asterisk-users] Regarding - P-Asserted identity and Privacy - SOLVED   das sandesh
• [asterisk-users] ExtenSpy Problem   Ishfaq Malik
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license   Noman Siddiqui
• [asterisk-users] t38 ATA   Alexandru Oniciuc
• [asterisk-users] multiple RTP port ranges for SIP   Kevin P. Fleming
• [asterisk-users] func odbc and mult iquery   Tilghman Lesher
• [asterisk-users] R: Can not enable sip debug because CLI flooded   Warren Selby
• [asterisk-users] t38 ATA   Steve Underwood
• [asterisk-users] How to add custom CDR fields to MySQL   Michael Silveus
• [asterisk-users] MWI and 1.6.1   Matt Watson
• [asterisk-users] t38 ATA   Jeff Brower
• [asterisk-users] R: t38 ATA   Alexandru Oniciuc
• [asterisk-users] Polycom not updating the directory list   hin lee
• [asterisk-users] modem config & pots & documentation   Givon Zirkind
• [asterisk-users] Installing chan_H323 by yum?   Michelle Dupuis
• [asterisk-users] Asterisk 1.6.2.5 x64 with Skype and DTMF on skype-out.   Joakim Eriksson
• [asterisk-users] Asterisk 1.6.2.5 x64 with Skype and DTMF on skype-out.   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.2.5 x64 with Skype and DTMF on skype-out.   Joakim Eriksson
• [asterisk-users] app_confbridge production ready?   Robert McGilvray
• [asterisk-users] Asterisk 1.4.30 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.0.26 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.1.18 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.6 Now Available   Asterisk Development Team
• [asterisk-users] fax & spandsp   Edwin Lam
• [asterisk-users] R: Can not enable sip debug because CLI flooded   jonas kellens
• [asterisk-users] Setting up RTP to flow between endpoints directly bypassing Asterisk   Vikram Ragukumar
• [asterisk-users] Time counting down and # detect   Pham Quy
• [asterisk-users] Time counting down and # detect   Pham Quy
• [asterisk-users] t38 ATA   Steve Underwood
• [asterisk-users] Skype for Asterisk and regular expressions   Richard Kenner
• [asterisk-users] fax & spandsp   Steve Underwood
• [asterisk-users] DUNDILOOKUP doesn't return record   Asterisk User
• [asterisk-users] PBX_DUNDI question   DHAVAL INDRODIYA
• [asterisk-users] adding agent with 2 phones to a queue   Magnus Benngård
• [asterisk-users] adding agent with 2 phones to a queue   Rob Hillis
• [asterisk-users] Asterisk on MPLS VPN   Sriram
• [asterisk-users] Asterisk on MPLS VPN   Steve Totaro
• [asterisk-users] SIP debug on a per call base   jonas kellens
• [asterisk-users] Asterisk on MPLS VPN   Sriram
• [asterisk-users] IAX2 peer question   Jeff LaCoursiere
• [asterisk-users] IAX2 peer question   Jeff LaCoursiere
• [asterisk-users] Asterisk on MPLS VPN   Tilghman Lesher
• [asterisk-users] func_devstate with latest 1.4...   Carlos Chavez
• [asterisk-users] How to test my Dial(SIP/...) ?   tjoen
• [asterisk-users] DID forwarding ?   Thomas Perron
• [asterisk-users] DID forwarding ?   Kyle Kienapfel
• [asterisk-users] Asterisk on MPLS VPN   mir shahnawaz
• [asterisk-users] Asterisk Management API   Peter Childs
• [asterisk-users] install asterisk from binaries   Joao Gomes Pereira
• [asterisk-users] install asterisk from binaries   Tzafrir Cohen
• [asterisk-users] adding agent with 2 phones to a queue   Magnus Benngård
• [asterisk-users] adding agent with 2 phones to a queue - SOLVED   Magnus Benngård
• [asterisk-users] DECT phone wont stop ringing   Magnus Benngård
• [asterisk-users] adding agent with 2 phones to a queue - SOLVED   Rob Hillis
• [asterisk-users] adding agent with 2 phones to a queue - SOLVED   Magnus Benngård
• [asterisk-users] ooh323_indicate: Don't know how to indicate condition 20   Michelle Dupuis
• [asterisk-users] Debugging log rotation problem   Richard Kenner
• [asterisk-users] Debugging log rotation problem   Tzafrir Cohen
• [asterisk-users] Debugging log rotation problem   Richard Kenner
• [asterisk-users] dahdi-linux-complete-2.2.1+2.2.1 failed to compile   Nitesh Divecha
• [asterisk-users] Strange audio problem with Digium Wildcard B410   Stefan-Michael Guenther
• [asterisk-users] dahdi-linux-complete-2.2.1+2.2.1 failed to compile   tjoen
• [asterisk-users] dahdi-linux-complete-2.2.1+2.2.1 failed to compile   Nitesh Divecha
• [asterisk-users] dahdi-linux-complete-2.2.1+2.2.1 failed to compile   Tzafrir Cohen
• [asterisk-users] dahdi-linux-complete-2.2.1+2.2.1 failed to compile   Nitesh Divecha
• [asterisk-users] Help with playing a recorded message in a conference.   Sean Brady
• [asterisk-users] queue MOH   Thomas Perron
• [asterisk-users] Change SIP Release Code   Nitesh Divecha
• [asterisk-users] Help with playing a recorded message in a conference.   Philipp von Klitzing
• [asterisk-users] Help with playing a recorded message in a conference.   Philipp von Klitzing
• [asterisk-users] Digium TE4xx T1 Bonding   Eric Wheeler
• [asterisk-users] Time counting down and # detect   Pham Quy
• [asterisk-users] USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER   RESEARCH
• [asterisk-users] High Availability Asterisk PBX   RESEARCH
• [asterisk-users] USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER   Alex Balashov
• [asterisk-users] High Availability Asterisk PBX   Jai Rangi
• [asterisk-users] High Availability Asterisk PBX   Alex Balashov
• [asterisk-users] Time counting down and # detect   Steve Edwards
• [asterisk-users] Time counting down and # detect   Pham Quy
• [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL   RSCL Mumbai
• [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL   Alex Balashov
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Mohit Saxena
• [asterisk-users] ExtenSpy Problem   Ishfaq Malik
• [asterisk-users] 1.2 to 1.6 and bristuff   Klaus Darilion
• [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL   Lee Archer
• [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL   RSCL Mumbai
• [asterisk-users] AEL in 1.6 and Gosub   Klaus Darilion
• [asterisk-users] AEL in 1.6 and Gosub   Kevin P. Fleming
• [asterisk-users] Android Phones ;-)   Conrad Wood
• [asterisk-users] Android Phones ;-)   Ishfaq Malik
• [asterisk-users] Android Phones ;-)   Jeff LaCoursiere
• [asterisk-users] asterisk-users Digest, Vol 68, Issue 33   Muro, Sam
• [asterisk-users] USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER   Muro, Sam
• [asterisk-users] Android Phones ;-)   Ishfaq Malik
• [asterisk-users] Using asterisk as avaya definity recording server   Miguel Molina
• [asterisk-users] Using asterisk as avaya definity recording server   Muro, Sam
• [asterisk-users] ExtenSpy Problem   David Backeberg
• [asterisk-users] Asterisk to be used with Ciscs media gateways   David Backeberg
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Jeff LaCoursiere
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Mohit Saxena
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Mohit Saxena
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Tim Nelson
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Jeff LaCoursiere
• [asterisk-users] ExtenSpy Problem   Ishfaq Malik
• [asterisk-users] ExtenSpy Problem   David Gibbons
• [asterisk-users] Asterisk to be used with Ciscs media gateways   David Gibbons
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Mohit Saxena
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Mohit Saxena
• [asterisk-users] dnd   Ott Rose
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Tim Nelson
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Peder
• [asterisk-users] ExtenSpy Problem   David Backeberg
• [asterisk-users] AEL in 1.6 and Gosub   Klaus Darilion
• [asterisk-users] ExtenSpy Problem   Ishfaq Malik
• [asterisk-users] ExtenSpy Problem   Ishfaq Malik
• [asterisk-users] Installing cdr_pgsql on asterisk 1.6.0.26   Miguel Molina
• [asterisk-users] How to find Asterisk compile time options for building app_swift module   LATEEF, IRFAN (ATTSI)
• [asterisk-users] Article - a method on how to evaluate an Asterisk server   Ioan Indreias
• [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL   Sean Brady
• [asterisk-users] Article - a method on how to evaluate an Asteriskserver   Jeff Brower
• [asterisk-users] Setting up RTP to flow between endpoints directlybypassing Asterisk   Jeff Brower
• [asterisk-users] Time counting while playback   Pham Quy
• [asterisk-users] Asterisk 1.4.24 DUNDi CLI commands not found   John Haigh
• [asterisk-users] Time counting while playback   Jeff LaCoursiere
• [asterisk-users] Time counting while playback   Steve Edwards
• [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL   RSCL Mumbai
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Mohit Saxena
• [asterisk-users] Setting up RTP to flow between endpoints directlybypassing Asterisk   Klaus Darilion
• [asterisk-users] R: t38 ATA   Klaus Darilion
• [asterisk-users] t38 ATA   Klaus Darilion
• [asterisk-users] Asterisk 1.4.24 DUNDi CLI commands not found   Klaus Darilion
• [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID)field into MySQL   Lee Archer
• [asterisk-users] DID/CID doesn't match "." (dot) in CID field   Daniel Grotti
• [asterisk-users] DID/CID doesn't match "." (dot) in CID field   Steve Murphy
• [asterisk-users] different authentication requirement   Eckhard Jokisch
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Peder
• [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID)field into MySQL   Steve Edwards
• [asterisk-users] DID/CID doesn't match "." (dot) in CID field   Daniel Grotti
• [asterisk-users] different authentication requirement   Philipp von Klitzing
• [asterisk-users] Asterisk + Sip Phone + BLF   damiano bertuna
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Tim Nelson
• [asterisk-users] DID/CID doesn't match "." (dot) in CID field   Richard Kenner
• [asterisk-users] [solved] Installing cdr_pgsql on asterisk 1.6.0.26   Miguel Molina
• [asterisk-users] dnd   Matt Watson
• [asterisk-users] dnd   Ott Rose
• [asterisk-users] different authentication requirement   Eckhard Jokisch
• [asterisk-users] DID/CID doesn't match "." (dot) in CID field   Tilghman Lesher
• [asterisk-users] [SPAM] - Asterisk + Sip Phone + BLF - Email found in subject   DLeese at LStelcom.com
• [asterisk-users] DID/CID doesn't match "." (dot) in CID field   Richard Kenner
• [asterisk-users] Asterisk to be used with Ciscs media gateways   Mohit Saxena
• [asterisk-users] Asterisk 1.4.24 DUNDi CLI commands not found   Leif Madsen
• [asterisk-users] DID/CID doesn't match "." (dot) in CID field   Steve Murphy
• [asterisk-users] softhangup   mir shahnawaz
• [asterisk-users] softhangup   mir shahnawaz
• [asterisk-users] Asterisk hangup all incoming calls after 10 seconds   Bruno Camargo
• [asterisk-users] fax & spandsp   Edwin Lam
• [asterisk-users] Outbound route prefixes   Alejandro Cabrera Obed
• [asterisk-users] Asterisk 1.4.24 DUNDi CLI commands not found   John Haigh
• [asterisk-users] AEL in 1.6 and Gosub   Steve Murphy
• [asterisk-users] Outbound route prefixes   Alejandro Cabrera Obed
• [asterisk-users] [NEWBIE] Simple hunt group on SIP -- need confirmation   Raj Mathur
• [asterisk-users] sip send image   bhrugu mehta
• [asterisk-users] sip send image   Steve Howes
• [asterisk-users] asterisk fax handeling   Peter den Hartog
• [asterisk-users] asterisk fax handeling   Tommy Botten Jensen
• [asterisk-users] Asterisk hangup all incoming calls after 10 seconds   Giorgio Incantalupo
• [asterisk-users] asterisk fax handeling   Klaus Darilion
• [asterisk-users] R: Asterisk hangup all incoming calls after 10 seconds   Alexandru Oniciuc
• [asterisk-users] AEL in 1.6 and Gosub   Klaus Darilion
• [asterisk-users] fax & spandsp   Klaus Darilion
• [asterisk-users] Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM   DHAVAL INDRODIYA
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   David Backeberg
• [asterisk-users] asterisk fax handeling   David Backeberg
• [asterisk-users] Asterisk as a skinny/sccp "client"?   Brian J. Murrell
• [asterisk-users] sip send image   bhrugu mehta
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Daniel Leite de Abreu
• [asterisk-users] AEL in 1.6 and Gosub   Elliot Murdock
• [asterisk-users] subject: 1.4 vs 1.6   Elliot Murdock
• [asterisk-users] Call Filtering   Dan Journo
• [asterisk-users] SIP codec negotiation / manipulation   Kevin Sandy
• [asterisk-users] Call Filtering   Kevin Sandy
• [asterisk-users] Asterisk as a skinny/sccp "client"?   Jason Parker
• [asterisk-users] Adding an external dial code   Alejandro Cabrera Obed
• [asterisk-users] Adding an external dial code   Zeeshan Zakaria
• [asterisk-users] monitor SIP jitter buffer   lore
• [asterisk-users] Using asterisk as avaya definity recording server   Muro, Sam
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Warren Selby
• [asterisk-users] Asterisk hangup all incoming calls after 10 seconds   Bruno Camargo
• [asterisk-users] Asterisk hangup all incoming calls after 10 seconds   Fred Posner
• [asterisk-users] Asterisk as a skinny/sccp "client"?   Brian J. Murrell
• [asterisk-users] BT ISDN-30 Call Failures   Russell Brown
• [asterisk-users] Call Filtering   Dan Journo
• [asterisk-users] Call Filtering   Kevin Sandy
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.   Matt Watson
• [asterisk-users] Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM   Barry L. Kline
• [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!   Danny Dias
• [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!   Zeeshan Zakaria
• [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!   Zeeshan Zakaria
• [asterisk-users] spandsp with asterisk 1.4.x   Joao Gomes Pereira
• [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!   Zeeshan Zakaria
• [asterisk-users] spandsp with asterisk 1.4.x   Doug Lytle
• [asterisk-users] SIP codec negotiation / manipulation   Vinícius Fontes
• [asterisk-users] SIP codec negotiation / manipulation   Steve Totaro
• [asterisk-users] asterisk fax handeling   Peter den Hartog
• [asterisk-users] spandsp with asterisk 1.4.x   Vinícius Fontes
• [asterisk-users] Door Phone Assistance   Robert Grignon
• [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!   Danny Dias
• [asterisk-users] SIP codec negotiation / manipulation   Jeff Brower
• [asterisk-users] SIP codec negotiation / manipulation   Jeff Brower
• [asterisk-users] Setting up RTP to flow between endpoints directlybypassingAsterisk   Jeff Brower
• [asterisk-users] Need help with auto-forwarding virtual extensions (Asterisk 1.4/GUI 2.0)   Cory Andrews
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Karl Fife
• [asterisk-users] Asterisk as a skinny/sccp "client"?   Michiel van Baak
• [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!   Zeeshan Zakaria
• [asterisk-users] DID number   Mike
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Mike Diehl
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Sean Brady
• [asterisk-users] DID number   Joe Greco
• [asterisk-users] DID number   Steve Edwards
• [asterisk-users] Wanted: free DID number and provider feedback   Mike
• [asterisk-users] Asterisk hangup all incoming calls after 10 seconds   Bruno Camargo
• [asterisk-users] Wanted: free DID number and provider feedback   SIP
• [asterisk-users] Wanted: free DID number and provider feedback   Mike
• [asterisk-users] Wanted: free DID number and provider feedback   SIP
• [asterisk-users] Door Phone Assistance   C F
• [asterisk-users] Wanted: free DID number and provider feedback   Steve Edwards
• [asterisk-users] Asterisk hangup all incoming calls after 10 seconds   Bruno Camargo
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Lee, John (Sydney)
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Karl Fife
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Lee, John (Sydney)
• [asterisk-users] DID number   Ira
• [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!   Danny Dias
• [asterisk-users] asterisk fax handeling   Olivier
• [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!   Zeeshan Zakaria
• [asterisk-users] Asterisk and OOo Smart Tags   Olivier
• [asterisk-users] Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM   DHAVAL INDRODIYA
• [asterisk-users] Polycom not updating the directory list   hin lee
• [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID)field into MySQL   RSCL Mumbai
• [asterisk-users] Polycom not updating the directory list   Lee, John (Sydney)
• [asterisk-users] How to detect a PSTN telephone is busy or not?   Zhang Shukun
• [asterisk-users] asterisk fax handeling   Klaus Darilion
• [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.   Klaus Darilion
• [asterisk-users] How to detect a PSTN telephone is busy or not?   ABBAS SHAKEEL
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   ABBAS SHAKEEL
• [asterisk-users] Call Filtering   Dan Journo
• [asterisk-users] Voicemail Remote Access   Dan Journo
• [asterisk-users] Door Phone Assistance   Robert Grignon
• [asterisk-users] SIP codec negotiation / manipulation   Kevin Sandy
• [asterisk-users] spandsp with asterisk 1.4.x   Joao Gomes Pereira
• [asterisk-users] Door Phone Assistance   Danny Nicholas
• [asterisk-users] Call Filtering   Danny Nicholas
• [asterisk-users] sip send image   Danny Nicholas
• [asterisk-users] spandsp with asterisk 1.4.x   Joao Gomes Pereira
• [asterisk-users] Software for my laptop to send Fax via H.323 ?   Jason Aarons (US)
• [asterisk-users] spandsp with asterisk 1.4.x   Vinícius Fontes
• [asterisk-users] Asterisk DIES with no trace. PLEASE   Danny Dias
• [asterisk-users] Voicemail Remote Access   Doug Lytle
• [asterisk-users] Voicemail Remote Access   Dan Journo
• [asterisk-users] SIP Router Project   Randy R
• [asterisk-users] Asterisk DIES with no trace. PLEASE   Zeeshan Zakaria
• [asterisk-users] Asterisk DIES with no trace. PLEASE   Zeeshan Zakaria
• [asterisk-users] Setting up RTP to flow between endpoints directlybypassingAsterisk   Klaus Darilion
• [asterisk-users] Problem with forwarding: Now forwarding SIP/ XX to Local/   Alex Rendour
• [asterisk-users] Setting up RTP to flow between endpoints directlybypassingAsterisk   Jeff Brower
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Karl Fife
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Mike Diehl
• [asterisk-users] Asterisk DIES with no trace. PLEASE   Danny Dias
• [asterisk-users] Asterisk DIES with no trace. PLEASE   Zeeshan Zakaria
• [asterisk-users] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)   Sebastian Milioto
• [asterisk-users] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)   Joseph
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   Philipp von Klitzing
• [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app   Matt Riddell
• [asterisk-users] queue MOH   Matt Riddell
• [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app   Zeeshan Zakaria
• [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app   Zeeshan Zakaria
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Warren Selby
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Tzafrir Cohen
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Daniel Leite de Abreu
• [asterisk-users] (no subject)   Adrian Marsh
• [asterisk-users] (no subject)   Steve Edwards
• [asterisk-users] (no subject)   Matt Riddell
• [asterisk-users] (no subject)   Zeeshan Zakaria
• [asterisk-users] Define an array of sip number in sip.conf   huu giang
• [asterisk-users] Define an array of sip number in sip.conf   Zeeshan Zakaria
• [asterisk-users] Better SIP security please! Was: (no subject)   Philipp von Klitzing
• [asterisk-users] Better SIP security please! Was: (no subject)   Zeeshan Zakaria
• [asterisk-users] confbridge not working?   Kelvin Chan
• [asterisk-users] How to detect a PSTN telephone is busy or not?   Zhang Shukun
• [asterisk-users] confbridge not working?   Alex Balashov
• [asterisk-users] confbridge not working?   Magnus Benngård
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   ABBAS SHAKEEL
• [asterisk-users] how to configure caller id   cool dude
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Warren Selby
• [asterisk-users] rtp connection remained when call busy using agi for call control   CHEN XUEQIN
• [asterisk-users] (no subject)   Ioan Indreias
• [asterisk-users] Strange initial RING   Alexandru Oniciuc
• [asterisk-users] too much sockets open by asterisk   Ilya Pichugin
• [asterisk-users] Strange initial RING   Gordon Henderson
• [asterisk-users] R: Strange initial RING   Alexandru Oniciuc
• [asterisk-users] ExtenSpy Problem [SOLVED]   Ishfaq Malik
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   tjoen
• [asterisk-users] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)   Sebastian Milioto
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   Jonathan Addleman
• [asterisk-users] Better SIP security please! Was: (no subject)   Ryan Bullock
• [asterisk-users] too much sockets open by asterisk   Leif Madsen
• [asterisk-users] how to configure caller id   Leif Madsen
• [asterisk-users] Define an array of sip number in sip.conf   Leif Madsen
• [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app   Leif Madsen
• [asterisk-users] Define an array of sip number in sip.conf   Zeeshan Zakaria
• [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app   Zeeshan Zakaria
• [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app   Zeeshan Zakaria
• [asterisk-users] Free iPhone Asterisk Function and Application Reference   Zeeshan Zakaria
• [asterisk-users] SPA3102 + asterisk drop call and loop (was SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) )   Sebastian Milioto
• [asterisk-users] how to configure caller id   cool dude
• [asterisk-users] register => 2345:password at sip_proxy/1234   tjoen
• [asterisk-users] too much sockets open by asterisk   CHEN XUEQIN
• [asterisk-users] Setting Caller ID for attended transfer   Daniel - Asterisk
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   ABBAS SHAKEEL
• [asterisk-users] Polycom not updating the directory list   hin lee
• [asterisk-users] register => 2345:password at sip_proxy/1234   Christian Victor
• [asterisk-users] Call Drops while doing assisted transfer from remote location   das sandesh
• [asterisk-users] Free iPhone Asterisk Function and Application Reference   Tzafrir Cohen
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Tzafrir Cohen
• [asterisk-users] register => 2345:password at sip_proxy/1234   tjoen
• [asterisk-users] Elastix 1.6 continuos ring   Bülent YILDIZ,EMPATIQ
• [asterisk-users] basic pc to pc voip in lan   kartik manocha
• [asterisk-users] basic pc to pc voip in lan   Zeeshan Zakaria
• [asterisk-users] 8Port Junghanns BRI card under Dahdi   Loic Didelot
• [asterisk-users] 8Port Junghanns BRI card under Dahdi   Jeff LaCoursiere
• [asterisk-users] 8Port Junghanns BRI card under Dahdi   Loic Didelot
• [asterisk-users] Asterisk general Timeout for digits   bruce bruce
• [asterisk-users] SIP signal through one IP and media through different IPs   bruce bruce
• [asterisk-users] Asterisk general Timeout for digits   Doug Lytle
• [asterisk-users] Asterisk general Timeout for digits   Zeeshan Zakaria
• [asterisk-users] how to configure caller id   Leif Madsen
• [asterisk-users] Call Drops while doing assisted transfer from remote location   Leif Madsen
• [asterisk-users] too much sockets open by asterisk   Leif Madsen
• [asterisk-users] basic pc to pc voip in lan   kartik manocha
• [asterisk-users] Asterisk general Timeout for digits   bruce bruce
• [asterisk-users] Asterisk general Timeout for digits   Zeeshan Zakaria
• [asterisk-users] how to start callerid for india   cool dude
• [asterisk-users] how to start callerid for india   Zeeshan Zakaria
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Daniel Leite de Abreu
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Daniel Leite de Abreu
• [asterisk-users] ZTdummy   Joao Gomes Pereira
• [asterisk-users] ZTdummy   Zeeshan Zakaria
• [asterisk-users] ZTdummy   Zeeshan Zakaria
• [asterisk-users] 1.6.1.18 -> 1.6.2.6 T38 Fax: call drops   sean darcy
• [asterisk-users] how to start callerid for india   Goke M Aruna
• [asterisk-users] Voicemail, Asterisk and Grandstream BT200   Daniel Bareiro
• [asterisk-users] Early audio problem in chan_dahdi   Roger Schreiter
• [asterisk-users] SIP codec negotiation / manipulation   Olle E. Johansson
• [asterisk-users] Better SIP security please! Was: (no subject)   Olle E. Johansson
• [asterisk-users] Voicemail, Asterisk and Grandstream BT200   Gordon Henderson
• [asterisk-users] Do i really need Dahdi and Libpri.   Randy R
• [asterisk-users] Do i really need Dahdi and Libpri.   Alex Balashov
• [asterisk-users] Do i really need Dahdi and Libpri.   Alex Balashov
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   tjoen
• [asterisk-users] Do i really need Dahdi and Libpri.   Tzafrir Cohen
• [asterisk-users] Do i really need Dahdi and Libpri.   Gordon Henderson
• [asterisk-users] Do i really need Dahdi and Libpri.   Tzafrir Cohen
• [asterisk-users] Do i really need Dahdi and Libpri.   Zeeshan Zakaria
• [asterisk-users] Do i really need Dahdi and Libpri.   Daniel Leite de Abreu
• [asterisk-users] Do i really need Dahdi and Libpri.   Daniel Leite de Abreu
• [asterisk-users] How to get Asterisk to make batch calls?   Leo Burd
• [asterisk-users] Do i really need Dahdi and Libpri.   Alex Balashov
• [asterisk-users] Do i really need Dahdi and Libpri.   Daniel Leite de Abreu
• [asterisk-users] test   card support asterisk
• [asterisk-users] test   Zeeshan Zakaria
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Daniel Leite de Abreu
• [asterisk-users] dahdi_monitor doesn't show data on RX & TX: broken card or cable?   Stefan-Michael Guenther
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Daniel Leite de Abreu
• [asterisk-users] Better SIP security please! Was: (no subject)   Philipp von Klitzing
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Theo Band
• [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!   Tzafrir Cohen
• [asterisk-users] Do i really need Dahdi and Libpri.   Tzafrir Cohen
• [asterisk-users] Asterisk Died - Ver-1.6.2.6.   Nitesh Divecha
• [asterisk-users] Asterisk Manager Interface (AMI) proxy recommendation   Leo Burd
• [asterisk-users] Asterisk Manager Interface (AMI) proxy recommendation   Tiago Geada
• [asterisk-users] Invalid Makefiles to install asterisk with ldap   mickael
• [asterisk-users] Do i really need Dahdi and Libpri.   Gordon Henderson
• [asterisk-users] Asterisk Died - Ver-1.6.2.6.   Stefan Schmidt
• [asterisk-users] Better SIP security please! Was: (no subject)   Gavin Henry
• [asterisk-users] Do i really need Dahdi and Libpri.   Zeeshan Zakaria
• [asterisk-users] Early audio problem in chan_dahdi   Roger Schreiter
• [asterisk-users] Voicemail, Asterisk and Grandstream BT200   Daniel Bareiro
• [asterisk-users] Voicemail, Asterisk and Grandstream BT200   Alyed
• [asterisk-users] Do i really need Dahdi and Libpri.   Hans Witvliet
• [asterisk-users] Voicemail, Asterisk and Grandstream BT200   Daniel Bareiro
• [asterisk-users] Invalid Makefiles to install asterisk with ldap   tjoen
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Watkins, Bradley
• [asterisk-users] PRI lines do not have CallerID activated yet it is   Nathanial Allan
• [asterisk-users] voicemail problem   Tamer Higazi
• [asterisk-users] PRI lines do not have CallerID activated yet it is   Rob Hillis
• [asterisk-users] voicemail problem   Danny Nicholas
• [asterisk-users] SIP signal through one IP and media through different IPs   Kevin Sandy
• [asterisk-users] voicemail problem   Ishfaq Malik
• [asterisk-users] How to get Asterisk to make batch calls?   Kevin Sandy
• [asterisk-users] Call files : call multiple SIP-accounts   jonas kellens
• [asterisk-users] SIP codec negotiation / manipulation   Kevin Sandy
• [asterisk-users] Context vs. Custom Context   Alejandro Cabrera Obed
• [asterisk-users] How to get Asterisk to make batch calls?   Danny Nicholas
• [asterisk-users] Call files : call multiple SIP-accounts   Zeeshan Zakaria
• [asterisk-users] DUNDi Confusion   Shina Owolabi
• [asterisk-users] Context vs. Custom Context   Leif Madsen
• [asterisk-users] Context vs. Custom Context   Alejandro Cabrera Obed
• [asterisk-users] too much sockets open by asterisk   CHEN XUEQIN
• [asterisk-users] requirecalltoken & receiving IAX calls   bilal ghayyad
• [asterisk-users] Transcoding question   Jim Dickenson
• [asterisk-users] requirecalltoken & receiving IAX calls   Jim Dickenson
• [asterisk-users] requirecalltoken & receiving IAX calls   Steve Howes
• [asterisk-users] Context vs. Custom Context   Leif Madsen
• [asterisk-users] Voicemail, Asterisk and Grandstream BT200   Alyed
• [asterisk-users] Play music to caller after answer, before dial   Michelle Dupuis
• [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app   Matt Riddell
• [asterisk-users] Free iPhone Asterisk Function and Application Reference   Matt Riddell
• [asterisk-users] Play music to caller after answer, before dial   Danny Nicholas
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Karl Fife
• [asterisk-users] Play music to caller after answer, before dial   Michelle Dupuis
• [asterisk-users] Asterisk running on a Xen Centos Serverchallenge!!!   Rafael Prado Rocchi
• [asterisk-users] Can I call myself on the same machine   ayodele abejide
• [asterisk-users] Transcoding question   Jim Dickenson
• [asterisk-users] Can I call myself on the same machine   Carlos Chavez
• [asterisk-users] Using asterisk as avaya definity recordingserver   Rafael Prado Rocchi
• [asterisk-users] Can I call myself on the same machine   ayodele abejide
• [asterisk-users] Transcoding question   Rafael Prado Rocchi
• [asterisk-users] PGSQL application   Rafael Prado Rocchi
• [asterisk-users] Play music to caller after answer, before dial   Thomas Perron
• [asterisk-users] Which folder for sounds?   sean darcy
• [asterisk-users] Play music to caller after answer, before dial   Michelle Dupuis
• [asterisk-users] Which folder for sounds?   Zeeshan Zakaria
• [asterisk-users] How to make upgrades with Asterisk   Danny Dias
• [asterisk-users] (no subject)   Aaron chen
• [asterisk-users] How to make upgrades with Asterisk   Zeeshan Zakaria
• [asterisk-users] Which folder for sounds?   Philipp von Klitzing
• [asterisk-users] Can I call myself on the same machine   Gordon Henderson
• [asterisk-users] chan_ss7 issue   Kasun Daminda
• [asterisk-users] [asterisk-ss7]Chan_ss7 issue   Kasun Daminda
• [asterisk-users] Which folder for sounds?   Tzafrir Cohen
• [asterisk-users] Better SIP security please! Was: (no subject)   Olle E. Johansson
• [asterisk-users] SIP codec negotiation / manipulation   Olle E. Johansson
• [asterisk-users] Integrate a CPE with Asterisk in MGCP   Nenad Kljajic
• [asterisk-users] Install dahdi on Xen virtual console   Vidura Senadeera
• [asterisk-users] Install dahdi on Xen virtual console   Tzafrir Cohen
• [asterisk-users] Install dahdi on Xen virtual console   Vinícius Fontes
• [asterisk-users] Install dahdi on Xen virtual console   Zeeshan Zakaria
• [asterisk-users] Using asterisk as avaya definity recordingserver   Moises Silva
• [asterisk-users] Install dahdi on Xen virtual console   Zeeshan Zakaria
• [asterisk-users] Asterisk crash - segmentation fault   Vieri
• [asterisk-users] How to make upgrades with Asterisk   Danny Dias
• [asterisk-users] distribuited ACD on many asterisk nodes   nik600
• [asterisk-users] How to make upgrades with Asterisk   Zeeshan Zakaria
• [asterisk-users] Install dahdi on Xen virtual console   Daniel Leite de Abreu
• [asterisk-users] Minimalize jitter in VoIP calls   jonas kellens
• [asterisk-users] Classic NO AUDIO problem - DD-WRT and NAT forwarding - HELP PLEASE!   bruce bruce
• [asterisk-users] Asterisk crash - segmentation fault   Vieri
• [asterisk-users] In Berlin this week? Kamailio/Asterisk community dinner on Thursday   Olle E. Johansson
• [asterisk-users] Strange Meetme disconnects   Tim McKee
• [asterisk-users] Sip module and dns   Luis Silva
• [asterisk-users] G.711a or G.711u ???   Alejandro Cabrera Obed
• [asterisk-users] G.711a or G.711u ???   Matt Riddell
• [asterisk-users] G.711a or G.711u ???   Zeeshan Zakaria
• [asterisk-users] G.711a or G.711u ???   Christian Victor
• [asterisk-users] G.711a or G.711u ???   Karl Fife
• [asterisk-users] Asterisk crash - segmentation fault   Tilghman Lesher
• [asterisk-users] G.711a or G.711u ???   Andrew Hakman
• [asterisk-users] permit/deny in sip.conf iax.conf   Karl Fife
• [asterisk-users] permit/deny in sip.conf iax.conf   Kevin P. Fleming
• [asterisk-users] permit/deny in sip.conf iax.conf   Steve Edwards
• [asterisk-users] permit/deny in sip.conf iax.conf   Kevin P. Fleming
• [asterisk-users] permit/deny in sip.conf iax.conf   Karl Fife
• [asterisk-users] permit/deny in sip.conf iax.conf   Steve Edwards
• [asterisk-users] Safe_asterisk doesn't exists???   Danny Dias
• [asterisk-users] Safe_asterisk doesn't exists???   Steve Edwards
• [asterisk-users] Asterisk 1.6.1.12 with Grandstream HT502 T38 Fax   JR Richardson
• [asterisk-users] AMD reporting NOTSURE most of the time   Steve Moran
• [asterisk-users] Mobile phone shut down, but Queue() Ring as usual   Zhang Shukun
• [asterisk-users] pstn calls not picked up   Balu Raman
• [asterisk-users] Mobile phone shut down, but Queue() Ring as usual   Alyed
• [asterisk-users] pstn calls not picked up   ABBAS SHAKEEL
• [asterisk-users] Asterisk crash - segmentation fault   Vieri
• [asterisk-users] G.729 Codec problem.   Arun Sasidhar
• [asterisk-users] permit/deny in sip.conf iax.conf   Olle E. Johansson
• [asterisk-users] Safe_asterisk doesn't exists???   Prince Singh
• [asterisk-users] Hook playback or ControlPlayBack cmd   huu giang
• [asterisk-users] Firewall & audio : need a wide range to work !   jonas kellens
• [asterisk-users] Firewall & audio : need a wide range to work !   Alex Balashov
• [asterisk-users] Firewall & audio : need a wide range to work !   jonas kellens
• [asterisk-users] Firewall & audio : need a wide range to work !   Danny Nicholas
• [asterisk-users] Using asterisk as avaya definity recordingserver   Muro, Sam
• [asterisk-users] pstn calls not picked up   Balu Raman
• [asterisk-users] software version   Ott Rose
• [asterisk-users] software version   Amine Mrichcha
• [asterisk-users] software version   Tzafrir Cohen
• [asterisk-users] permit/deny in sip.conf iax.conf   Karl Fife
• [asterisk-users] software version   Amine Mrichcha
• [asterisk-users] Restarting Asterisk using a script - Thanks to all -   Amine Mrichcha
• [asterisk-users] software version   Steve Edwards
• [asterisk-users] Restarting Asterisk using a script - Thanks to all -   Steve Edwards
• [asterisk-users] Restarting Asterisk using a script - Thanks to all -   Alyed
• [asterisk-users] Restarting Asterisk using a script - Thanks to all -   Jose P. Espinal
• [asterisk-users] Firewall & audio : need a wide range to work !   jonas kellens
• [asterisk-users] Restarting Asterisk using a script - Thanks to all -   Amine Mrichcha
• [asterisk-users] software version   Ott Rose
• [asterisk-users] chan_h323 and ToS   Daniel Grotti
• [asterisk-users] software version (lets try it again)   Ott Rose
• [asterisk-users] software version   Tzafrir Cohen
• [asterisk-users] software version (lets try it again)   Alyed
• [asterisk-users] Restarting Asterisk using a script - Thanks to all -   Jose P. Espinal
• [asterisk-users] Restarting Asterisk using a script - Thanks to all -   Steve Edwards
• [asterisk-users] software version   Alyed
• [asterisk-users] software version   Ott Rose
• [asterisk-users] Safe_asterisk doesn't exists???   Danny Dias
• [asterisk-users] Safe_asterisk doesn't exists???   Andrew Latham
• [asterisk-users] AstLinux 0.7.1 released   Darrick Hartman
• [asterisk-users] Safe_asterisk doesn't exists???   Andrew Latham
• [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem   mosbah.abdelkader
• [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem   Doug Lytle
• [asterisk-users] installing dahdi card   Ott Rose
• [asterisk-users] This is a test, hijack this   Gergo Csibra
• [asterisk-users] installing dahdi card   Tzafrir Cohen
• [asterisk-users] Dahdi-linux & Dahdi-tools 2.2.1.1 Release Announcement   Russ Meyerriecks
• [asterisk-users] This is a test, hijack this   Miguel Molina
• [asterisk-users] installing dahdi card   Ott Rose
• [asterisk-users] installing dahdi card   Tzafrir Cohen
• [asterisk-users] installing dahdi card   Ott Rose
• [asterisk-users] This is a test, hijack this   Dave Fullerton
• [asterisk-users] This is a test, hijack this   Gergo Csibra
• [asterisk-users] This is a test, hijack this   Karl Fife
• [asterisk-users] AMD reporting NOTSURE most of the time   Matt Riddell
• [asterisk-users] AMD reporting NOTSURE most of the time   Matt Riddell
• [asterisk-users] Which folder for sounds?   sean darcy
• [asterisk-users] This is a test, hijack this   Gergo Csibra
• [asterisk-users] new server install errors starting asterisk   Ott Rose
• [asterisk-users] new server install errors starting asterisk   Matt Riddell
• [asterisk-users] new server install errors starting asterisk   Steve Edwards
• [asterisk-users] Aastra weirds IP 169.x.x.x   Danny Dias
• [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem   Dave Platt
• [asterisk-users] Aastra weirds IP 169.x.x.x   Matt Watson
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   James Lamanna
• [asterisk-users] Polycom DHCP Option 66 FTP provisioning   Andrew Joakimsen
• [asterisk-users] Mobile phone shut down, but Queue() Ring as usual   Zhang Shukun
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Zeeshan Zakaria
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Ryan Wagoner
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Zeeshan Zakaria
• [asterisk-users] Music class default requested but no musiconhold loaded   675842709
• [asterisk-users] How to get Sip response codes in Dialplan?   Zhang Shukun
• [asterisk-users] How to get Sip response codes in Dialplan?   Juan E. Rodríguez
• [asterisk-users] How to get Sip response codes in Dialplan?   Zhang Shukun
• [asterisk-users] Attended transfer and callerID updates forSiemens Openstage phones   Loic Didelot
• [asterisk-users] Voicemail, Asterisk and Grandstream BT200   Daniel Bareiro
• [asterisk-users] configure the sound for inbound calls   salaheddine elharit
• [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE   Asterisk
• [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE   Steve Howes
• [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE   Asterisk
• [asterisk-users] permit/deny in sip.conf iax.conf   Olle E. Johansson
• [asterisk-users] Which folder for sounds?   Tzafrir Cohen
• [asterisk-users] rtp.conf ports for inbound or outbound?   Michelle Dupuis
• [asterisk-users] Attended transfer and callerID updates forSiemens Openstage phones   Kevin P. Fleming
• [asterisk-users] intergration of Diameter   Tushar Jain
• [asterisk-users] new server install errors starting asterisk   Ott Rose
• [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem   mosbah.abdelkader
• [asterisk-users] new server install errors starting asterisk   Steve Howes
• [asterisk-users] new server install errors starting asterisk   Ott Rose
• [asterisk-users] new server install errors starting asterisk   Steve Howes
• [asterisk-users] Static linking   Jiri Uncovsky
• [asterisk-users] Metasphere?   David Gibbons
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   James Lamanna
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Christian Victor
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Carlos Chavez
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Zeeshan Zakaria
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Zeeshan Zakaria
• [asterisk-users] Metasphere?   Daryl Jones
• [asterisk-users] new server install errors starting asterisk   Ott Rose
• [asterisk-users] call not routed   Balu Raman
• [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE   --[ UxBoD ]--
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Tzafrir Cohen
• [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem   Dave Platt
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Steve Edwards
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Zeeshan Zakaria
• [asterisk-users] Background noise   khalid touati
• [asterisk-users] new server install errors starting asterisk   Steve Howes
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Christian Victor
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Christian Victor
• [asterisk-users] new server install errors starting asterisk   Ott Rose
• [asterisk-users] How to get Sip response codes in Dialplan?   Philipp von Klitzing
• [asterisk-users] Attended transfer and callerID updates forSiemens Openstage phones   Philipp von Klitzing
• [asterisk-users] new server install errors starting asterisk   Ott Rose
• [asterisk-users] Background noise   Philipp von Klitzing
• [asterisk-users] new server install errors starting asterisk   Philipp von Klitzing
• [asterisk-users] new server install errors starting asterisk   Tzafrir Cohen
• [asterisk-users] Transcoding question   Jeff Brower
• [asterisk-users] Metasphere?   Jeff Brower
• [asterisk-users] Transcoding question   Jim Dickenson
• [asterisk-users] BT ISDN-30 Call Failures   Gavin Henry
• [asterisk-users] send a call from A to B use sip trunk prablem   Aaron chen
• [asterisk-users] Time counting while playback   Pham Quy
• [asterisk-users] "Failed to play transfer sound! " during attended transfer   kamrun nahar bina
• [asterisk-users] 1.6.1.18 -> 1.6.2.6 T38 Fax: call drops   sean darcy
• [asterisk-users] "Failed to play transfer sound! " during attended transfer   Alyed
• [asterisk-users] send a call from A to B use sip trunk prablem   Alyed
• [asterisk-users] SIP/2.0 403 Forbidden   Aaron chen
• [asterisk-users] Transcoding question   Jeff Brower
• [asterisk-users] [VUC] Voipathon 24-hour online party begins in 30 mintes   Randy R
• [asterisk-users] send a call from A to B use sip trunk prablem   Aaron chen
• [asterisk-users] "Failed to play transfer sound! " during attended transfer   kamrun nahar bina
• [asterisk-users] [VUC] Voipathon 24-hour online party begins in 30 mintes   covici at ccs.covici.com
• [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem   mosbah.abdelkader
• [asterisk-users] Asterisk load balancing and failover   huu giang
• [asterisk-users] Asterisk load balancing and failover   Zeeshan Zakaria
• [asterisk-users] Delay on sip channel   Asterisk User
• [asterisk-users] need help on setup rtp directly between 2 sip clients   haloha
• [asterisk-users] Asterisk load balancing and failover   huu giang
• [asterisk-users] Is there any Diguim distributor in Lahore   Faheem
• [asterisk-users] Is there any Diguim distributor in Lahore   ABBAS SHAKEEL
• [asterisk-users] Asterisk load balancing and failover   Zeeshan Zakaria
• [asterisk-users] Not hearing Telco Operator messages   Robert Grignon
• [asterisk-users] problem with polarity reverse   Justas Gulbinskas
• [asterisk-users] Transcoding question   Jim Dickenson
• [asterisk-users] rtp.conf ports for inbound or outbound?   Olle E. Johansson
• [asterisk-users] Is there any Diguim distributor in Lahore   Leif Madsen
• [asterisk-users] no voicemail on pstn line   Landy Landy
• [asterisk-users] Not hearing Telco Operator messages   Zeeshan Zakaria
• [asterisk-users] Sip module and dns   Luis Silva
• [asterisk-users] Background noise   khalid touati
• [asterisk-users] Asterisk load balancing and failover   Eric Wheeler
• [asterisk-users] "Failed to play transfer sound! " during attended transfer   Alyed
• [asterisk-users] Sip module and dns   Alyed
• [asterisk-users] rtp.conf ports for inbound or outbound?   Mark Michelson
• [asterisk-users] rtp.conf ports for inbound or outbound?   Mark Michelson
• [asterisk-users] need help on setup rtp directly between 2 sip clients   Alyed
• [asterisk-users] Not hearing Telco Operator messages   Robert Grignon
• [asterisk-users] new server install errors starting asterisk   Ott Rose
• [asterisk-users] Re :Re: Sip module and dns (Alyed)   Luis Silva
• [asterisk-users] Re :Re: Sip module and dns (Alyed)   Alyed
• [asterisk-users] no voicemail on pstn line   Sil
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Alejandro Kauffmann
• [asterisk-users] What does this error message mean   Ira
• [asterisk-users] dnd not working correctly   Ott Rose
• [asterisk-users] Background noise   Philipp von Klitzing
• [asterisk-users] dnd not working correctly   Alyed
• [asterisk-users] What does this error message mean   Alyed
• [asterisk-users] What does this error message mean   Ira
• [asterisk-users] need help on setup rtp directly between 2 sip clients   haloha
• [asterisk-users] need help on setup rtp directly between 2 sip clients   Alyed
• [asterisk-users] need help on setup rtp directly between 2 sip clients   haloha
• [asterisk-users] need help on setup rtp directly between 2 sip clients   Alyed
• [asterisk-users] What does this error message mean   Juan E. Rodríguez
• [asterisk-users] need help on setup rtp directly between 2 sipclients   Juan E. Rodríguez
• [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)   Stephen Davies
• [asterisk-users] configure the sound for inbound calls   Aurimas Skirgaila
• [asterisk-users] Background noise   khalid touati
• [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall   James Lamanna
• [asterisk-users] chan_ss7 issue   Kasun Daminda
• [asterisk-users] problem with polarity reverse   Justas Gulbinskas
• [asterisk-users] migration   Thomas Perron
• [asterisk-users] Trying to configure xorcom on Suse 11   Tim Panton
• [asterisk-users] migration   Steve Edwards
• [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall   Alyed
• [asterisk-users] Trying to configure xorcom on Suse 11   JD Austin
• [asterisk-users] SIP/2.0 403 Forbidden   Dovid Bender
• [asterisk-users] Trying to configure xorcom on Suse 11   Tim Panton
• [asterisk-users] Attended transfer and callerID updates forSiemens Openstage phones   Loic Didelot
• [asterisk-users] Libtonezone   Joseph L. Casale
• [asterisk-users] Trying to configure xorcom on Suse 11   Tzafrir Cohen
• [asterisk-users] Updating Asterisk and its use with MySQL   Daniel Bareiro
• [asterisk-users] Updating Asterisk and its use with MySQL   Jim Dickenson
• [asterisk-users] Updating Asterisk and its use with MySQL   Daniel Bareiro
• [asterisk-users] Updating Asterisk and its use with MySQL   Jim Dickenson
• [asterisk-users] problem with polarity reverse   Tzafrir Cohen
• [asterisk-users] Trying to configure xorcom on Suse 11   JD Austin
• [asterisk-users] Updating Asterisk and its use with MySQL   Daniel Bareiro
• [asterisk-users] Updating Asterisk and its use with MySQL   Alyed
• [asterisk-users] Updating Asterisk and its use with MySQL   Daniel Bareiro
• [asterisk-users] Updating Asterisk and its use with MySQL   Alyed
• [asterisk-users] Libtonezone   Anthony Francis - Handy Networks LLC
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Joseph Begumisa
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Zeeshan Zakaria
• [asterisk-users] "Failed to play transfer sound! " during attended transfer   kamrun nahar bina
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Doug
• [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall   James Lamanna
• [asterisk-users] Trying to configure xorcom on Suse 11   Tim Panton
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Steve Edwards
• [asterisk-users] Trying to configure xorcom on Suse 11   Jim Dickenson
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   James Lamanna
• [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall   Troy Davis
• [asterisk-users] dnd not working correctly   Ioan Indreias
• [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall   Warren Selby
• [asterisk-users] is it possible to connect Digium TE420 and Cisco card?   Aurimas Skirgaila
• [asterisk-users] How to add custom CDR fields to MySQL   Robert Price
• [asterisk-users] Restarting Asterisk using a script - Thanks to all -   Aurimas Skirgaila
• [asterisk-users] Asterisk load balancing and failover   huu giang
• [asterisk-users] Asterisk load balancing and failover   huu giang
• [asterisk-users] Continue a dialplan when the client hang up the call   huu giang
• [asterisk-users] MixMonitor and StopMixMonitor   jonas kellens
• [asterisk-users] Continue a dialplan when the client hang up the call   Ishfaq Malik
• [asterisk-users] Continue a dialplan when the client hang up the call   huu giang
• [asterisk-users] Continue a dialplan when the client hang up the call   huu giang
• [asterisk-users] queue autopause status   Christian Gansberger
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   nik600
• [asterisk-users] Continue a dialplan when the client hang up the call   Zeeshan Zakaria
• [asterisk-users] How to add custom CDR fields to MySQL   Zeeshan Zakaria
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   John Novack
• [asterisk-users] Slightly more advanced dialling..   Andy Dixon
• [asterisk-users] Slightly more advanced dialling..   Danny Nicholas
• [asterisk-users] Slightly more advanced dialling..   Andy Dixon
• [asterisk-users] Trying to configure xorcom on Suse 11   Tzafrir Cohen
• [asterisk-users] Realtime Issue   Jason Walker
• [asterisk-users] Slightly more advanced dialling..   Philipp von Klitzing
• [asterisk-users] Background noise   khalid touati
• [asterisk-users] Background noise   Philipp von Klitzing
• [asterisk-users] dnd not working correctly   Ott Rose
• [asterisk-users] Background noise   khalid touati
• [asterisk-users] Realtime Issue   D Tucny
• [asterisk-users] Libtonezone   Joseph L. Casale
• [asterisk-users] Foip solution   Mike Diehl
• [asterisk-users] Foip solution   jon pounder
• [asterisk-users] [asterisk-biz] Foip solution   Lee Howard
• [asterisk-users] Background noise   Jeff Brower
• [asterisk-users] No audio when calling via PSTN, before remote answers (with polarity reversal)   Luar Roji
• [asterisk-users] Foip solution   Mike Diehl
• [asterisk-users] Asterisk, IAX, & Sub interfaces   trebaum
• [asterisk-users] dnd not working correctly   Ott Rose
• [asterisk-users] Foip solution   David Backeberg
• [asterisk-users] Slightly more advanced dialling..   Zeeshan Zakaria
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Joseph Begumisa
• [asterisk-users] Foip solution   Zeeshan Zakaria
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Doug
• [asterisk-users] Libtonezone   Tzafrir Cohen
• [asterisk-users] Asterisk, IAX, & Sub interfaces   trebaum
• [asterisk-users] Asterisk, IAX, & Sub interfaces   Tim Nelson
• [asterisk-users] Asterisk, IAX, & Sub interfaces   trebaum
• [asterisk-users] Asterisk system for church call center   Frank Church
• [asterisk-users] Asterisk, IAX, & Sub interfaces   Steve Edwards
• [asterisk-users] Asterisk system for church call center   Frank Church
• [asterisk-users] Asterisk system for church call center   Gondar Monn
• [asterisk-users] amr   Hans Witvliet
• [asterisk-users] Asterisk system for church call center   Duncan Turnbull
• [asterisk-users] Trying to get reason for ending of AGI call recording   Jeff Johnson
• [asterisk-users] Trying to get reason for ending of AGI call recording   Jeff Johnson
• [asterisk-users] [asterisk-biz] Asterisk system for church call center   Alex Balashov
• [asterisk-users] Trying to get reason for ending of AGI call recording   Steve Edwards
• [asterisk-users] Asterisk and Call files   Anthony Geoffron
• [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)   James Lamanna
• [asterisk-users] [asterisk-biz] Asterisk system for church call center   Mark Phillips
• [asterisk-users] Diameter for Asterisk, Traffix Diameter stack ?   mara greenberg
• [asterisk-users] a2billing wont pass the number   Nathanial Allan
• [asterisk-users] Trying to configure xorcom on Suse 11   Tim Panton
• [asterisk-users] Asynchronous play music   Pham Quy
• [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)   Matt Watson
• [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)   James Lamanna
• [asterisk-users] a2billing wont pass the number   Juan E. Rodríguez
• [asterisk-users] Asynchronous play music   Steve Edwards
• [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)   James Lamanna
• [asterisk-users] dnd not working correctly   Alyed
• [asterisk-users] Inbound configuration   chima s
• [asterisk-users] Asynchronous play music   Quy Pham Sy
• [asterisk-users] How can install and use Async AGI   Quy Pham Sy
• [asterisk-users] [asterisk-biz] Asterisk system for church call center   Peter Childs
• [asterisk-users] Asterisk realtime ldap:active directory   mickael
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Robert Grignon
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Juan Miguel
• [asterisk-users] dnd not working correctly   Ott Rose
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   mir shahnawaz
• [asterisk-users] Minimalize jitter in VoIP calls   jonas kellens
• [asterisk-users] Continue a dialplan when the client hang up the call   Ngo-Vi Hoai-Anh
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   hin lee
• [asterisk-users] FAX 2 mail configuration   Joao Gomes Pereira
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Darrick Hartman
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Andrew Latham
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Gordon Henderson
• [asterisk-users] dnd not working correctly   Alyed
• [asterisk-users] Confusion on call forwarding   Richard Kenner
• [asterisk-users] Confusion on call forwarding   Juan E. Rodríguez
• [asterisk-users] dnd not working correctly (resolved)   Ott Rose
• [asterisk-users] Priority based softhangup   mir shahnawaz
• [asterisk-users] DAHDI 2.2.1, Asterisk 1.6.2.6 - Channel unacceptable (6)   Stefan Tichy
• [asterisk-users] Confusion on call forwarding   Richard Kenner
• [asterisk-users] Priority based softhangup   Steve Edwards
• [asterisk-users] Asynchronous play music   Steve Edwards
• [asterisk-users] Priority based softhangup   Danny Nicholas
• [asterisk-users] Asynchronous play music   Danny Nicholas
• [asterisk-users] Priority based softhangup   mir shahnawaz
• [asterisk-users] Dropped Calls   Brent Davidson
• [asterisk-users] Priority based softhangup   Danny Nicholas
• [asterisk-users] Dropped Calls   Danny Nicholas
• [asterisk-users] convert from wav or mp3 to gsm   salaheddine elharit
• [asterisk-users] convert from wav or mp3 to gsm   Danny Nicholas
• [asterisk-users] E-mails from Asterisk coming from root   Mike A. Leonetti
• [asterisk-users] Priority based softhangup   Steve Edwards
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Joseph Begumisa
• [asterisk-users] Priority based softhangup   mir shahnawaz
• [asterisk-users] E1 card w/o echo cancellation   M. Ehsanul Karim
• [asterisk-users] Dropped Calls   Brent Davidson
• [asterisk-users] Dropped Calls   JR Richardson
• [asterisk-users] app_txfax.c   Jerry Geis
• [asterisk-users] Confusion on call forwarding   Juan E. Rodríguez
• [asterisk-users] [asterisk-biz] Asterisk system for church call center   Frank Church
• [asterisk-users] Asterisk system for church call center   Frank Church
• [asterisk-users] Asterisk hangup all outging calls after 32 seconds   Ing CIP. Alejandro Celi Mariátegui
• [asterisk-users] Confusion on call forwarding   Richard Kenner
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   Jonathan Addleman
• [asterisk-users] Asynchronous play music   Pham Quy
• [asterisk-users] Asterisk system for church call center   Lenz Emilitri
• [asterisk-users] Slightly more advanced dialling..   Andy Dixon
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   covici at ccs.covici.com
• [asterisk-users] Asterisk system for church call center   bruce bruce
• [asterisk-users] a2billing wont pass the number   bruce bruce
• [asterisk-users] Asterisk load balancing and failover   Tobias Wolf
• [asterisk-users] Asterisk load balancing and failover   Zeeshan Zakaria
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   nik600
• [asterisk-users] Jitter Buffer and MeetMe.   russian qwerty
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   covici at ccs.covici.com
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   Randy R
• [asterisk-users] Unable to login to voicemail with Ekiga   Alejandro Imass
• [asterisk-users] Unable to login to voicemail with Ekiga   Zeeshan Zakaria
• [asterisk-users] Unable to login to voicemail with Ekiga   Alejandro Imass
• [asterisk-users] convert from wav or mp3 to gsm   Robert Grignon
• [asterisk-users] convert from wav or mp3 to gsm   David Backeberg
• [asterisk-users] convert from wav or mp3 to gsm   Danny Nicholas
• [asterisk-users] 24 FXS Port Voip Gateway and Asterisk   Karl Fife
• [asterisk-users] convert from wav or mp3 to gsm   salaheddine elharit
• [asterisk-users] Dropped Calls   Michael L. Young
• [asterisk-users] Reset personal voicemail settings   Felix Tiefenthaler
• [asterisk-users] Dropped Calls   Brent Davidson
• [asterisk-users] Unable to login to voicemail with Ekiga   Alejandro Imass
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   Philipp von Klitzing
• [asterisk-users] Live Audio Streaming- From Aux interface-Online resource   Randy R
• [asterisk-users] Dropped Calls   Danny Nicholas
• [asterisk-users] Dropped Calls   Philipp von Klitzing
• [asterisk-users] meetme() and dahdi_dummy on an embedded system   Darko Bodnaruk
• [asterisk-users] Safe_asterisk doesn't exists???   Danny Dias
• [asterisk-users] DAHDI 2.2.1, Asterisk 1.6.2.6 - Channel unacceptable (6)   Stefan Tichy
• [asterisk-users] Multicast Paging   Jonathan C. Bailey
• [asterisk-users] Live Audio Streaming- From Auxinterface-Online resource   Danny Nicholas
• [asterisk-users] Multicast Paging   Leif Madsen
• [asterisk-users] Upcoming Asterisk 1.6.0 and 1.6.1 Maintenance Changes   Asterisk Development Team
• [asterisk-users] How to run Music while looking for the caller in Database   Bharath B. Reddy Bynagari
• [asterisk-users] How to run Music while looking for the caller in Database   Mark Michelson
• [asterisk-users] Multicast Paging   Philipp von Klitzing
• [asterisk-users] How to run Music while looking for the caller in Database   Steve Edwards
• [asterisk-users] Reset personal voicemail settings   Mark Michelson
• [asterisk-users] Multicast Paging   Jonathan C. Bailey
• [asterisk-users] Necessary hardware   Kosa
• [asterisk-users] Dropped Calls   Brent Davidson
• [asterisk-users] Dropped Calls   Brent Davidson
• [asterisk-users] Asterisk load balancing and failover   huu giang

Last message date: Wed Mar 31 21:22:33 CDT 2010
Archived on: Wed Mar 31 21:22:43 CDT 2010

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