October 2009 Archives by date

• Messages sorted by: [ thread ] [ subject ] [ author ]
• More info on this list...

Starting: Thu Oct 1 00:35:04 CDT 2009
Ending: Sat Oct 31 20:09:22 CDT 2009
Messages: 1536

• [asterisk-users] Choose IAX or SIP trunking?   Kirill 'Big K' Katsnelson
• [asterisk-users] Choose IAX or SIP trunking?   Alan Lord (News)
• [asterisk-users] Choose IAX or SIP trunking?   Kirill 'Big K' Katsnelson
• [asterisk-users] Softphone in Web   ABBAS SHAKEEL
• [asterisk-users] Sending Dialled number down a sip channel to a PBX   Ishfaq Malik
• [asterisk-users] help on ${RTPAUDIOQOS}   Asterisk User
• [asterisk-users] Softphone in Web   Administrator TOOTAI
• [asterisk-users] Softphone in Web   ABBAS SHAKEEL
• [asterisk-users] "got stuck at 150 calls, above that not working in stress test"   Matt Riddell
• [asterisk-users] Sending Dialled number down a sip channel to a PBX   Matt Riddell
• [asterisk-users] INVITE Sending Local IP   Mike Bessette
• [asterisk-users] INVITE Sending Local IP   Steve Howes
• [asterisk-users] INVITE Sending Local IP   Mike Bessette
• [asterisk-users] Friday Oct 2: Digium's new Speech Recognition for Asterisk   randulo
• [asterisk-users] RTP Delayed during RTCP   Cyprus VoIP
• [asterisk-users] Choose IAX or SIP trunking?   Mindaugas Kezys
• [asterisk-users] portech MV-378 SIP GSM Gateway   robert boardman
• [asterisk-users] Busy app timeout   Julian Lyndon-Smith
• [asterisk-users] "got stuck at 150 calls, above that not working in stress test"   das sandesh
• [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed   DHAVAL INDRODIYA
• [asterisk-users] INVITE Sending Local IP   Scott L. Lykens
• [asterisk-users] INVITE Sending Local IP   Mike Bessette
• [asterisk-users] Sending Dialled number down a sip channel to a PBX   Lyle Giese
• [asterisk-users] RTP Delayed during RTCP   Kevin P. Fleming
• [asterisk-users] Busy app timeout   Kevin P. Fleming
• [asterisk-users] Is there a way to get info who disconnected the call into CDR?   Rennes Neps
• [asterisk-users] INVITE Sending Local IP   Scott L. Lykens
• [asterisk-users] Is there a way to get info who disconnected thecall into CDR?   Rennes Neps
• [asterisk-users] What are the reasons for VoIP echo?   Myles Wakeham
• [asterisk-users] INVITE Sending Local IP   Mike Bessette
• [asterisk-users] INVITE Sending Local IP   Scott L. Lykens
• [asterisk-users] INVITE Sending Local IP   Mike Bessette
• [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed   Moises Silva
• [asterisk-users] INVITE Sending Local IP   Scott L. Lykens
• [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed   Kevin P. Fleming
• [asterisk-users] What are the reasons for VoIP echo?   Ira
• [asterisk-users] DTMF problems during a message play   Barton Fisher
• [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed   Martin
• [asterisk-users] portech MV-378 SIP GSM Gateway   Martin
• [asterisk-users] QOS/DSCP for IAX?   Michelle Dupuis
• [asterisk-users] QOS/DSCP for IAX?   Michelle Dupuis
• [asterisk-users] Voicemail - remove option to save in different folders   Kyle Kienapfel
• [asterisk-users] QOS/DSCP for IAX?   Danny Nicholas
• [asterisk-users] QOS/DSCP for IAX?   Dave Fullerton
• [asterisk-users] QOS/DSCP for IAX?   Michelle Dupuis
• [asterisk-users] QOS/DSCP for IAX?   Dave Fullerton
• [asterisk-users] "got stuck at 150 calls, above that not working in stress test"   Matt Riddell
• [asterisk-users] TDM410P - False Answer Supervision   Nitesh Divecha
• [asterisk-users] TDM410P - False Answer Supervision   Danny Nicholas
• [asterisk-users] What are the reasons for VoIP echo?   Myles Wakeham
• [asterisk-users] Bringing people into a conference   Harley Holcombe
• [asterisk-users] What are the reasons for VoIP echo?   Ira
• [asterisk-users] TDM410P - False Answer Supervision   Nitesh Divecha
• [asterisk-users] IAX2 Call rejected, CallToken Support required   Klaverstyn, David C
• [asterisk-users] What are the reasons for VoIP echo?   Martin
• [asterisk-users] TDM410P - False Answer Supervision   Martin
• [asterisk-users] What are the reasons for VoIP echo?   John A. Sullivan III
• [asterisk-users] What are the reasons for VoIP echo?   Martin
• [asterisk-users] What are the reasons for VoIP echo?   John A. Sullivan III
• [asterisk-users] IAX2 Call rejected, CallToken Support required   covici at ccs.covici.com
• [asterisk-users] TDM410P - False Answer Supervision   Nitesh Divecha
• [asterisk-users] IAX2 Call rejected, CallToken Support required   Klaverstyn, David C
• [asterisk-users] Busy() returns immediately on IAX trunk   James Lamanna
• [asterisk-users] Music On Hold   Cyprus VoIP
• [asterisk-users] Free version of softswitch with billing and routing released   Mindaugas Kezys
• [asterisk-users] srtp issue   Szasz Szabolcs
• [asterisk-users] Bringing people into a conference   Ioan Indreias
• [asterisk-users] Music On Hold   Ioan Indreias
• [asterisk-users] Music On Hold   Cyprus VoIP
• [asterisk-users] Problem with inbound calls - asterisk 1.6.1.6   Carlo Dimaggio
• [asterisk-users] Creating a clear channel on zaptel   Zeeshan Zakaria
• [asterisk-users] MeetMe Hints   Paul Dugas
• [asterisk-users] How to call extensions and add them to a conference room   Zeeshan Zakaria
• [asterisk-users] How to call extensions and add them to aconference room   Danny Nicholas
• [asterisk-users] Music On Hold   Danny Nicholas
• [asterisk-users] Followme   Anahi Ludueña
• [asterisk-users] Followme   Danny Nicholas
• [asterisk-users] How to call extensions and add them to aconference room   Zeeshan Zakaria
• [asterisk-users] "got stuck at 150 calls, above that not working in stress test"   das sandesh
• [asterisk-users] "got stuck at 150 calls, above that not working in stress test"   Matt Riddell
• [asterisk-users] Creating a clear channel on zaptel   Tzafrir Cohen
• [asterisk-users] One side SIP goes dead on length conversation   David Cook
• [asterisk-users] Sending a DTMF remotely with PlayDTMF problem.   Pablo Bernasconi
• [asterisk-users] Followme   Anahi Ludueña
• [asterisk-users] "got stuck at 150 calls, above that not working in stress test"   das sandesh
• [asterisk-users] Followme   Danny Nicholas
• [asterisk-users] Followme   Anahi Ludueña
• [asterisk-users] Followme   Danny Nicholas
• [asterisk-users] Followme   Anahi Ludueña
• [asterisk-users] Followme   Jim Dickenson
• [asterisk-users] Followme   Danny Nicholas
• [asterisk-users] One side SIP goes dead on length conversation   Steven Stromer
• [asterisk-users] Asterisk + Skype deployment   Alan Lord (News)
• [asterisk-users] What are the reasons for VoIP echo?   Steve Underwood
• [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed   Steve Underwood
• [asterisk-users] What are the reasons for VoIP echo?   Steve Underwood
• [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)   James Hankins
• [asterisk-users] What are the reasons for VoIP echo?   Martin
• [asterisk-users] What are the reasons for VoIP echo?   Peder
• [asterisk-users] Extra Sounds Missing on 1.6.1.6 install   Mark Hulber
• [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)   Jonathan Thurman
• [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)   Fred Posner
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   Michelle Dupuis
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   Doug Lytle
• [asterisk-users] Extra Sounds Missing on 1.6.1.6 install   Mark Hulber
• [asterisk-users] Extra Sounds Missing on 1.6.1.6 install   Kyle Kienapfel
• [asterisk-users] Extra Sounds Missing on 1.6.1.6 install   Jason Parker
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   C F
• [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)   Jonathan Thurman
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   Michelle Dupuis
• [asterisk-users] TDM410P - False Answer Supervision   Nitesh Divecha
• [asterisk-users] What are the reasons for VoIP echo?   Steve Underwood
• [asterisk-users] TDM410P - False Answer Supervision   Danny Nicholas
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   Danny Nicholas
• [asterisk-users] TDM410P - False Answer Supervision   Nitesh Divecha
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   Michelle Dupuis
• [asterisk-users] TDM410P - False Answer Supervision   Danny Nicholas
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   Danny Nicholas
• [asterisk-users] Restarting of B-channel on span 1   Chris Miller
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   John A. Sullivan III
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   Danny Nicholas
• [asterisk-users] help on ${RTPAUDIOQOS}   Klaus Darilion
• [asterisk-users] Asterisk,click2talk, webphone   Doddle WebPhone
• [asterisk-users] Problem sending a DTMF remotely. Please need help...   Pablo Bernasconi
• [asterisk-users] help on ${RTPAUDIOQOS}   Asterisk User
• [asterisk-users] Music On Hold   Cyprus VoIP
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   Michiel van Baak
• [asterisk-users] Restarting of B-channel on span 1   Tzafrir Cohen
• [asterisk-users] Asterisk and Jack   Fabien COMTE
• [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image   James Stocks
• [asterisk-users] Asterisk on NSLU2 : Grandstream not registering   jonas kellens
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   Tilghman Lesher
• [asterisk-users] Restarting of B-channel on span 1   Tilghman Lesher
• [asterisk-users] Asterisk on NSLU2 : Grandstream not registering   Tzafrir Cohen
• [asterisk-users] Asterisk and Jack   Tzafrir Cohen
• [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image   Jonathan Thurman
• [asterisk-users] [SOLVED] Asterisk on NSLU2 : Grandstream not registering   jonas kellens
• [asterisk-users] ERROR[1499]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR transmission error   jonas kellens
• [asterisk-users] Asterisk and Jack   Fabien COMTE
• [asterisk-users] Ideasip   David at ULC
• [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image   James Stocks
• [asterisk-users] After call into console/dsp hangup hear ringing   Jerry Geis
• [asterisk-users] x100p card   Alan Zheng
• [asterisk-users] Zaptel problems on SUSE 9.3   Angus Asterisk
• [asterisk-users] x100p card   Olivier
• [asterisk-users] Hacking the network   Vijay Gandhi
• [asterisk-users] Zaptel problems on SUSE 9.3   Josué Conti
• [asterisk-users] Zaptel problems on SUSE 9.3   Angus Asterisk
• [asterisk-users] After call into console/dsp hangup hear ringing   Jerry Geis
• [asterisk-users] Zaptel problems on SUSE 9.3   Josué Conti
• [asterisk-users] (OT) Zaptel, SuSE 9.3, Debian   Philipp Kempgen
• [asterisk-users] Zaptel problems on SUSE 9.3   Tzafrir Cohen
• [asterisk-users] (OT) Zaptel, SuSE 9.3, Debian   Hans Witvliet
• [asterisk-users] tdm outgoing   B.Masoud at SH
• [asterisk-users] tdm outgoing   Paul Hales
• [asterisk-users] Peculiar error message when using Q-SIG   Richard Kenner
• [asterisk-users] After call into console/dsp hangup hear ringing   Jerry Geis
• [asterisk-users] "got stuck at 150 calls, above that not working in stress test"   Matt Riddell
• [asterisk-users] tdm outgoing   Ira
• [asterisk-users] Peculiar error message when using Q-SIG   Tilghman Lesher
• [asterisk-users] Drop calls when using Flash Operator Panel   hin lee
• [asterisk-users] Peculiar error message when using Q-SIG   Richard Kenner
• [asterisk-users] After call into console/dsp hangup hear ringing   Matt Riddell
• [asterisk-users] tdm outgoing   Tzafrir Cohen
• [asterisk-users] AEL problem: bug or feature?   Klaus Darilion
• [asterisk-users] Problems using chan_sebi and Huawei E169G   Thomas Kenyon
• [asterisk-users] Grandstream GXW4024 experience   Olivier
• [asterisk-users] tdm outgoing   B.Masoud at SH
• [asterisk-users] Drop calls when using Flash Operator Panel   Doug Lytle
• [asterisk-users] tdm outgoing   B.Masoud at SH
• [asterisk-users] tdm outgoing   Ioan Indreias
• [asterisk-users] tdm outgoing   B.Masoud at SH
• [asterisk-users] Problem sending a DTMF remotely. Please need help!!   Pablo Bernasconi
• [asterisk-users] AEL problem: bug or feature?   Klaus Darilion
• [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling   Klaus Darilion
• [asterisk-users] digium fax: can't indicate condition 19?   Scott L. Lykens
• [asterisk-users] AEL problem: bug or feature?   Steve Edwards
• [asterisk-users] AEL problem: bug or feature?   Philipp Kempgen
• [asterisk-users] Questions about app_jack.c   Fabien COMTE
• [asterisk-users] digium fax: can't indicate condition 19?   Kevin P. Fleming
• [asterisk-users] tdm outgoing   Ioan Indreias
• [asterisk-users] What dahdi_dynamic and dahdi_transcode modules are for?   Gonzalo Marcote Peña
• [asterisk-users] OriginateResponse Event   Anahi Ludueña
• [asterisk-users] AEL problem: bug or feature?   Klaus Darilion
• [asterisk-users] DTMF problem during read()   Bart Fisher
• [asterisk-users] OriginateResponse Event   Danny Nicholas
• [asterisk-users] AEL problem: bug or feature?   Klaus Darilion
• [asterisk-users] Asterisk and QSIG   Vadim Lebedev
• [asterisk-users] *****SPAM***** DTMF problem during read()   Danny Nicholas
• [asterisk-users] dahdi dies with "No more room in scheduler"   James Lamanna
• [asterisk-users] OriginateResponse Event   Anahi Ludueña
• [asterisk-users] Grandstream GXW4024 experience   Jonathan Thurman
• [asterisk-users] Drop calls when using Flash Operator Panel   hin lee
• [asterisk-users] dahdi dies with "No more room in scheduler"   Barry L. Kline
• [asterisk-users] OriginateResponse Event   Danny Nicholas
• [asterisk-users] tdm outgoing   Ira
• [asterisk-users] dahdi dies with "No more room in scheduler"   Jeff LaCoursiere
• [asterisk-users] web module for video calls   Jeff LaCoursiere
• [asterisk-users] Method to downgrade asterisk   Bart Fisher
• [asterisk-users] OriginateResponse Event   Jared Smith
• [asterisk-users] *****SPAM***** Method to downgrade asterisk   Danny Nicholas
• [asterisk-users] OriginateResponse Event   Danny Nicholas
• [asterisk-users] OriginateResponse Event   Philipp Kempgen
• [asterisk-users] OriginateResponse Event   Jared Smith
• [asterisk-users] Receptionist GUI?   Ken D'Ambrosio
• [asterisk-users] Receptionist GUI?   Steve Totaro
• [asterisk-users] tdm outgoing   B.Masoud at SH
• [asterisk-users] Method to downgrade asterisk   Ryan Wagoner
• [asterisk-users] tdm outgoing   Philipp Kempgen
• [asterisk-users] Receptionist GUI?   Jason Baker
• [asterisk-users] Receptionist GUI?   Danny Nicholas
• [asterisk-users] OriginateResponse Event   Anthony Messina
• [asterisk-users] Zaptel problems on SUSE 9.3   Angus Asterisk
• [asterisk-users] (OT) Zaptel, SuSE 9.3, Debian   Angus Asterisk
• [asterisk-users] Receptionist GUI?   CunningPike
• [asterisk-users] Receptionist GUI?   troxlinux
• [asterisk-users] Receptionist GUI?   Danny Nicholas
• [asterisk-users] Zaptel problems on SuSE 9.3   Philipp Kempgen
• [asterisk-users] Networking Concept   B.Masoud at SH
• [asterisk-users] Networking Concept   Ivan Stepaniuk
• [asterisk-users] Receptionist GUI?   Nicolás Gudiño
• [asterisk-users] "got stuck at 150 calls, above that not working in stress test"   das sandesh
• [asterisk-users] Grandstream GXW4024 experience   Olivier
• [asterisk-users] T38 REINVITe issue   Ujjval Karihaloo
• [asterisk-users] Networking Concept   B.Masoud at SH
• [asterisk-users] ooh323 and h323   Dovid Bender
• [asterisk-users] Questions about app_jack.c [solved]   Fabien COMTE
• [asterisk-users] How to answer to an incoming call with alsa.   Fabien COMTE
• [asterisk-users] Networking Concept   Leif Neland
• [asterisk-users] fsk callerid with DTAS start?   d tbsky
• [asterisk-users] Networking Concept   B.Masoud at SH
• [asterisk-users] Networking Concept   Faraz Khan
• [asterisk-users] Lancom 1722 and Asterisk (i need HELP)   Thomas Janzen
• [asterisk-users] What happened to MACRO_EXTEN in AEL macros since 1.6?   Klaus Darilion
• [asterisk-users] parse transparent ISUP parameters   Giedrius Augys
• [asterisk-users] video support over iax   Ron
• [asterisk-users] Asterisk + Monitor() Poor quality   Kim Delevati
• [asterisk-users] Asterisk + Monitor() Poor quality   Danny Nicholas
• [asterisk-users] Asterisk + Monitor() Poor quality   Kim Delevati
• [asterisk-users] Cent OS 5.3 All Updated Asterisk Installation Giving Error   David Home
• [asterisk-users] Cent OS 5.3 All Updated Asterisk Installation Giving Error   Tim Nelson
• [asterisk-users] Cent OS 5.3 All Updated Asterisk Installation Giving Error   Jim Dickenson
• [asterisk-users] Cent OS 5.3 All Updated Asterisk InstallationGiving Error   Danny Nicholas
• [asterisk-users] Is anyone doing real time updates to where asterisk registers?   Eric Chamberlain
• [asterisk-users] How to answer to an incoming call with alsa.   Philipp Kempgen
• [asterisk-users] Cent OS 5.3 All Updated Asterisk Installation Giving Error   Tzafrir Cohen
• [asterisk-users] Networking Concept   Gordon Henderson
• [asterisk-users] T38 REINVITe issue   Ujjval Karihaloo
• [asterisk-users] Asterisk + Monitor() Poor quality   Danny Nicholas
• [asterisk-users] adding modules   mickael ropars
• [asterisk-users] adding modules   Tzafrir Cohen
• [asterisk-users] Receptionist GUI?   Xavier
• [asterisk-users] (OT) Asterisk + Monitor() Poor quality   Philipp Kempgen
• [asterisk-users] adding modules   mickael ropars
• [asterisk-users] Asterisk + Monitor() Poor quality   Kim Delevati
• [asterisk-users] Asterisk Integration with RDP Property Management Software?   David Wathen
• [asterisk-users] Asterisk + Monitor() Poor quality   Kim Delevati
• [asterisk-users] Asterisk + Monitor() Poor quality   Danny Nicholas
• [asterisk-users] Asterisk + Monitor() Poor quality   Kim Delevati
• [asterisk-users] Networking Concept   Hans Witvliet
• [asterisk-users] Problems using chan_sebi and Huawei E169G   Martin Stubbs
• [asterisk-users] Receptionist GUI?   Danny Nicholas
• [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)   astgroups at comcast.net
• [asterisk-users] Transfers from Queue Calls   Darrin Henshaw
• [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)   Danny Nicholas
• [asterisk-users] MPG123 Dying   --[ UxBoD ]--
• [asterisk-users] MPG123 Dying   Danny Nicholas
• [asterisk-users] Asterisk Dying [was: Re: MPG123 Dying]   Tzafrir Cohen
• [asterisk-users] Asterisk 1.6.1.6 suddently restarts ...   Olivier
• [asterisk-users] adding modules   Tzafrir Cohen
• [asterisk-users] MPG123 Dying   Mindaugas Kezys
• [asterisk-users] Asterisk 1.4.27-rc2, 1.6.0.16-rc2, 1.6.1.7-rc2, and 1.6.2.0-rc3 Now Available   Asterisk Development Team
• [asterisk-users] MPG123 Dying   Trevor Peirce
• [asterisk-users] Receptionist GUI?   Danny Nicholas
• [asterisk-users] Problem sending a DTMF remotely. Please need help!!!   Pablo Bernasconi
• [asterisk-users] Asterisk 1.6.1.6 suddently restarts ...   Leif Madsen
• [asterisk-users] Choose IAX or SIP trunking?   Kirill 'Big K' Katsnelson
• [asterisk-users] Transfers from Queue Calls   Kirill 'Big K' Katsnelson
• [asterisk-users] T38 REINVITe issue   Trevor Peirce
• [asterisk-users] T38 REINVITe issue   Ujjval Karihaloo
• [asterisk-users] Choose IAX or SIP trunking?   Hans Witvliet
• [asterisk-users] How to answer to an incoming call with alsa (or OSS).   Fabien COMTE
• [asterisk-users] MPG123 Dying   --[ UxBoD ]--
• [asterisk-users] Is anyone doing real time updates to where asterisk registers?   Ishfaq Malik
• [asterisk-users] Networking Concept   Ivan Stepaniuk
• [asterisk-users] Drop calls when using Flash Operator Panel   Doug Lytle
• [asterisk-users] Asterisk Integration with RDP Property Management Software?   Ivan Stepaniuk
• [asterisk-users] system cmd + fax line   Alex Samad
• [asterisk-users] tdm outgoing   B.Masoud at SH
• [asterisk-users] tdm outgoing   Tzafrir Cohen
• [asterisk-users] Asterisk 1.6.1.6 suddently restarts ...   Olivier
• [asterisk-users] system cmd + fax line   Danny Nicholas
• [asterisk-users] Transfers from Queue Calls   Lenz Emilitri
• [asterisk-users] put some IVR into a queue after the call queuing   nik600
• [asterisk-users] Need provider recommendations for the UK   Ed W
• [asterisk-users] DTMF Issues   Barton Fisher
• [asterisk-users] Asterisk 1.6.1.6 suddently restarts ...   Olivier
• [asterisk-users] system cmd + fax line   Dovid Bender
• [asterisk-users] MPG123 Dying   Dovid Bender
• [asterisk-users] MPG123 Dying   Danny Nicholas
• [asterisk-users] DTMF Issues   Dovid Bender
• [asterisk-users] Need provider recommendations for the UK   Dovid Bender
• [asterisk-users] Asterisk,click2talk, webphone   Doddle WebPhone
• [asterisk-users] VPS Server   David at ULC
• [asterisk-users] Can dial long distance but not local?   Ben Schorr
• [asterisk-users] Can dial long distance but not local?   Cary Fitch
• [asterisk-users] Can dial long distance but not local?   Doug Lytle
• [asterisk-users] Can dial long distance but not local?   Ben Schorr
• [asterisk-users] Can dial long distance but not local?   Ben Schorr
• [asterisk-users] Asterisk 1.6.1.6 suddently restarts ...   Leif Madsen
• [asterisk-users] Asterisk debug message --- stopped sounds ???   B.Masoud at SH
• [asterisk-users] Drop Call on ICMP Port Unreachable?   Dan Mahoney, System Admin
• [asterisk-users] Choose IAX or SIP trunking?   Kirill 'Big K' Katsnelson
• [asterisk-users] MeetMe + SLA   Chandrakant Solanki
• [asterisk-users] limiting number of channels to be accessed   B.Masoud at SH
• [asterisk-users] limiting number of channels to be accessed   Ivan Stepaniuk
• [asterisk-users] VPS Server   Ivan Stepaniuk
• [asterisk-users] Problem sending a DTMF remotely. Please need help!!   Ivan Stepaniuk
• [asterisk-users] DTMF problems during a message play   Ivan Stepaniuk
• [asterisk-users] Problem with inbound calls - asterisk 1.6.1.6   Ivan Stepaniuk
• [asterisk-users] MeetMe option question   Richard Kenner
• [asterisk-users] Asterisk and Sheeva "wall wart".   Ken D'Ambrosio
• [asterisk-users] How to keep difference between 2 SIP-accounts/trunks from same server ??   jonas kellens
• [asterisk-users] Asterisk and Sheeva "wall wart".   Vincent
• [asterisk-users] Problem sending a DTMF remotely. Please need help!!   Pablo Bernasconi
• [asterisk-users] Asterisk and Sheeva "wall wart".   Tzafrir Cohen
• [asterisk-users] Dialplan problem   Anahi Ludueña
• [asterisk-users] Having trouble with "IF" and blanks   Richard Kenner
• [asterisk-users] Server-side scripting when SIP phones register   Olivier
• [asterisk-users] Server-side scripting when SIP phones register   Danny Nicholas
• [asterisk-users] How to keep difference between 2 SIP-accounts/trunks from same server ??   Ioan Indreias
• [asterisk-users] Dialplan problem   Ioan Indreias
• [asterisk-users] Having trouble with "IF" and blanks   Ivan Stepaniuk
• [asterisk-users] Best afordable router with QOS for *   hbk
• [asterisk-users] Friday Noon VUC with guest Alex Robar   randulo
• [asterisk-users] Best afordable router with QOS for *   John Knight
• [asterisk-users] g729 free codec any idea   DHAVAL INDRODIYA
• [asterisk-users] Drop Call on ICMP Port Unreachable?   John Todd
• [asterisk-users] Suggestions for low level RTP stream generator?   Stephen Davies
• [asterisk-users] Best afordable router with QOS for *   Michelle Dupuis
• [asterisk-users] Best afordable router with QOS for *   John Knight
• [asterisk-users] Best afordable router with QOS for *   Fred Posner
• [asterisk-users] Best afordable router with QOS for *   Cory Andrews
• [asterisk-users] Server-side scripting when SIP phones register   Olivier
• [asterisk-users] Best afordable router with QOS for *   Ira
• [asterisk-users] Server-side scripting when SIP phones register   Elliot Otchet
• [asterisk-users] Server-side scripting when SIP phones register   Danny Nicholas
• [asterisk-users] Fuori ufficio   Pierluigi Frullani
• [asterisk-users] g729 free codec any idea   Tim Nelson
• [asterisk-users] g729 free codec any idea   Danny Nicholas
• [asterisk-users] g729 free codec any idea   DHAVAL INDRODIYA
• [asterisk-users] asterisk 2bct/rlt calling   Steve Mathers
• [asterisk-users] Asterisk debug message --- stopped sounds ???   B.Masoud at SH
• [asterisk-users] limiting number of channels to be accessed   B.Masoud at SH
• [asterisk-users] g729 free codec any idea   Moises Silva
• [asterisk-users] Best afordable router with QOS for *   John Novack
• [asterisk-users] limiting number of channels to be accessed   Steve Mathers
• [asterisk-users] limiting number of channels to be accessed   Michelle Dupuis
• [asterisk-users] No sound on voicemail from analog line   Landy Landy
• [asterisk-users] Best QoS for Linux   Michelle Dupuis
• [asterisk-users] Best QoS for Linux   Michelle Dupuis
• [asterisk-users] No sound on voicemail from analog line   Tzafrir Cohen
• [asterisk-users] Problem sending a DTMF remotely. Please need help!!   Matt Riddell
• [asterisk-users] limiting number of channels to be accessed   B.Masoud at SH
• [asterisk-users] Help setting up IMAP_STORAGE on CentOS 5   Noah I. Engelberth
• [asterisk-users] Help setting up IMAP_STORAGE on CentOS 5   Jason Parker
• [asterisk-users] Best QoS for Linux   John A. Sullivan III
• [asterisk-users] Help setting up IMAP_STORAGE on CentOS 5   John A. Sullivan III
• [asterisk-users] Having trouble with "IF" and blanks   Richard Kenner
• [asterisk-users] How to keep difference between 2 SIP-accounts/trunks from same server ??   Dovid Bender
• [asterisk-users] Realtime static does not work in 1.6.1 or 1.6.2   Carlos Chavez
• [asterisk-users] Dialplan problem   C F
• [asterisk-users] Realtime static does not work in 1.6.1 or 1.6.2   Tilghman Lesher
• [asterisk-users] Dialplan problem   Ivan Stepaniuk
• [asterisk-users] Dialplan problem   Anahi Ludueña
• [asterisk-users] adding modules   Dovid Bender
• [asterisk-users] g729 free codec any idea   Dovid Bender
• [asterisk-users] Best afordable router with QOS for *   Darrick Hartman
• [asterisk-users] g729 free codec any idea   Michelle Dupuis
• [asterisk-users] g729 free codec any idea   Matt Riddell
• [asterisk-users] SIP Hard Phone with SMS   "Juan E. Rodríguez"
• [asterisk-users] g729 free codec any idea   Ujjval Karihaloo
• [asterisk-users] g729 free codec any idea   Tzafrir Cohen
• [asterisk-users] g729 free codec any idea   Tzafrir Cohen
• [asterisk-users] SIP Hard Phone with SMS   randulo
• [asterisk-users] SIP Hard Phone with SMS   Johann Steinwendtner
• [asterisk-users] Server-side scripting when SIP phones register   Olivier
• [asterisk-users] Asterisk 1.6.1.6 suddently restarts ...   Olivier
• [asterisk-users] How to keep difference between 2 SIP-accounts/trunks from same server ??   jonas kellens
• [asterisk-users] Best QoS for Linux   RSCL Mumbai
• [asterisk-users] Digium G729 licence unattended install   Olivier
• [asterisk-users] Today's problem: Inbound call routing   Ben Schorr
• [asterisk-users] MPG123 Dying   --[ UxBoD ]--
• [asterisk-users] Today's problem: Inbound call routing   Tzafrir Cohen
• [asterisk-users] Asterisk Queue & Agent   Marco Sambo
• [asterisk-users] Digium G729 licence unattended install   Gordon Henderson
• [asterisk-users] G.729 and Voicemail   Gordon Henderson
• [asterisk-users] Trunk and Pstn line   ABBAS SHAKEEL
• [asterisk-users] How to keep difference between 2 SIP-accounts/trunks from same server ??   Ioan Indreias
• [asterisk-users] MPG123 Dying   --[ UxBoD ]--
• [asterisk-users] How to keep difference between 2 SIP-accounts/trunks from same server ??   Dovid Bender
• [asterisk-users] MPG123 Dying   Dovid Bender
• [asterisk-users] MPG123 Dying   --[ UxBoD ]--
• [asterisk-users] MPG123 Dying   --[ UxBoD ]--
• [asterisk-users] Best QoS for Linux   Jason Baker
• [asterisk-users] MeetMe option question   Robert McGilvray
• [asterisk-users] No sound on voicemail from analog line   Landy Landy
• [asterisk-users] wrond DTMF detection on Zap channel   nik600
• [asterisk-users] g729 free codec any idea   Michelle Dupuis
• [asterisk-users] G.729 and Voicemail   Kevin P. Fleming
• [asterisk-users] MeetMe option question   Richard Kenner
• [asterisk-users] MeetMe option question   covici at ccs.covici.com
• [asterisk-users] VoiceMail and IMAP   --[ UxBoD ]--
• [asterisk-users] g729 free codec any idea   Jeff LaCoursiere
• [asterisk-users] G.729 and Voicemail   Gordon Henderson
• [asterisk-users] VoiceMail and IMAP   John A. Sullivan III
• [asterisk-users] Billing applications   voip crazy
• [asterisk-users] Billing applications   voip crazy
• [asterisk-users] Billing applications   "Juan E. Rodríguez"
• [asterisk-users] G.729 and Voicemail   Kevin P. Fleming
• [asterisk-users] calls ansowered for 1 second or less   B.Masoud at SH
• [asterisk-users] Today's problem: Inbound call routing   Ben Schorr
• [asterisk-users] G.729 and Voicemail   Gordon Henderson
• [asterisk-users] G.729 and Voicemail   Jeff LaCoursiere
• [asterisk-users] wrond DTMF detection on Zap channel   C F
• [asterisk-users] G.729 and Voicemail   Moises Silva
• [asterisk-users] G.729 and Voicemail   Jeff LaCoursiere
• [asterisk-users] How to keep difference between 2 SIP-accounts/trunks from same server ??   jonas kellens
• [asterisk-users] Chanspy   Torintino T
• [asterisk-users] Chanspy   Chris Brentano
• [asterisk-users] Chanspy   Torintino T
• [asterisk-users] No sound on voicemail from analog line   Ivan Stepaniuk
• [asterisk-users] Today's problem: Inbound call routing   Tzafrir Cohen
• [asterisk-users] Chanspy   Danny Nicholas
• [asterisk-users] ${REASON} not getting set.   Mike Diehl
• [asterisk-users] ${REASON} not getting set.   David Klaverstyn
• [asterisk-users] Today's problem: Inbound call routing   Ben Schorr
• [asterisk-users] choppy sound   B.Masoud at SH
• [asterisk-users] choppy sound   Danny Nicholas
• [asterisk-users] ${REASON} not getting set.   Mike Diehl
• [asterisk-users] Incoming extension not working.   Ken D'Ambrosio
• [asterisk-users] choppy sound   B.Masoud at SH
• [asterisk-users] choppy sound   B.Masoud at SH
• [asterisk-users] lawnmower man "attack" sip tag=Zerogij34 some one else notice this in 20th september or recently?   Marco Mouta
• [asterisk-users] lawnmower man "attack" ??   Marco Mouta
• [asterisk-users] Wifi GSM handover   Patrick
• [asterisk-users] Incoming extension not working.   Warren Selby
• [asterisk-users] Wifi GSM handover   Steve Kennedy
• [asterisk-users] Today's problem: Inbound call routing   Duncan Turnbull
• [asterisk-users] Wifi GSM handover   Frank Bulk
• [asterisk-users] Wifi GSM handover   Patrick
• [asterisk-users] Wifi GSM handover   Steve Kennedy
• [asterisk-users] Billing applications   Mindaugas Kezys
• [asterisk-users] wrond DTMF detection on Zap channel   nik600
• [asterisk-users] lawnmower man "attack" sip tag=Zerogij34 some one else notice this in 20th september or recently?   Ivan Stepaniuk
• [asterisk-users] delay to dial   B.Masoud at SH
• [asterisk-users] Wifi GSM handover   Frank Bulk
• [asterisk-users] Slightly OT: Astricon and Google Wave   Randy R
• [asterisk-users] delay to dial   Doug Lytle
• [asterisk-users] Mp3 for IVR prompts   Steve Edwards
• [asterisk-users] paging/intercom   lists at contacttel.com
• [asterisk-users] delay to dial   B.Masoud at SH
• [asterisk-users] delay to dial   John Novack
• [asterisk-users] No sound on voicemail from analog line   Landy Landy
• [asterisk-users] delay to dial   Ivan Stepaniuk
• [asterisk-users] delay to dial   Doug Lytle
• [asterisk-users] Grandstream GXP 2010 : multiple accounts not working   jonas kellens
• [asterisk-users] Grandstream GXP 2010 : multiple accounts not working   Gordon Henderson
• [asterisk-users] delay to dial   B.Masoud at SH
• [asterisk-users] delay to dial   Doug Lytle
• [asterisk-users] Mp3 for IVR prompts   RSCL Mumbai
• [asterisk-users] Incoming extension not working.   Steve Edwards
• [asterisk-users] No sound on voicemail from analog line   Ivan Stepaniuk
• [asterisk-users] delay to dial   Ivan Stepaniuk
• [asterisk-users] Mp3 for IVR prompts   Steve Edwards
• [asterisk-users] delay to dial   B.Masoud at SH
• [asterisk-users] delay to dial   Ivan Stepaniuk
• [asterisk-users] Mp3 for IVR prompts   gergis.rasmy
• [asterisk-users] outgoing sip calls work; incoming calls fail   listmail at websage.ca
• [asterisk-users] Asterisk to Asterisk access voicemail - not working   Joseph
• [asterisk-users] Method to use SOX inside a Dialplan   Bart Fisher
• [asterisk-users] Method to use SOX inside a Dialplan   Martin
• [asterisk-users] delay to dial   covici at ccs.covici.com
• [asterisk-users] choppy sound   Dovid Bender
• [asterisk-users] Method to use SOX inside a Dialplan   covici at ccs.covici.com
• [asterisk-users] Method to use SOX inside a Dialplan   Steve Edwards
• [asterisk-users] Asterisk to Asterisk access voicemail - not working   Ivan Stepaniuk
• [asterisk-users] outgoing sip calls work; incoming calls fail   Ivan Stepaniuk
• [asterisk-users] Method to use SOX inside a Dialplan   Barton Fisher
• [asterisk-users] outgoing sip calls work; incoming calls fail   listmail at websage.ca
• [asterisk-users] outgoing sip calls work; incoming calls fail   Ivan Stepaniuk
• [asterisk-users] outgoing sip calls work; incoming calls fail   Ivan Stepaniuk
• [asterisk-users] Asterisk to Asterisk access voicemail - not working   Joseph
• [asterisk-users] outgoing sip calls work; incoming calls fail   listmail at websage.ca
• [asterisk-users] Method to use SOX inside a Dialplan   Steve Edwards
• [asterisk-users] Mp3 for IVR prompts   RSCL Mumbai
• [asterisk-users] Mp3 for IVR prompts   Tilghman Lesher
• [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image   James Stocks
• [asterisk-users] Mp3 for IVR prompts   Michael Graves
• [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image   Jonathan Thurman
• [asterisk-users] Zaptel problems on SuSE 9.3   Angus Asterisk
• [asterisk-users] Asterisk to Asterisk access voicemail - not working   Joseph
• [asterisk-users] Call Recording and Posting   Dan Journo
• [asterisk-users] Grandstream 2010   Cary Fitch
• [asterisk-users] Call Recording and Posting   Elliot Otchet
• [asterisk-users] Call Recording and Posting   Steve Edwards
• [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4   Jeff Johnson
• [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4   Lee, John (Sydney)
• [asterisk-users] Call Recording and Posting   Dovid Bender
• [asterisk-users] SPRINTF option : format %1$s not supported   Olivier
• [asterisk-users] Asterisk to Asterisk access voicemail - not working   Ivan Stepaniuk
• [asterisk-users] Call Recording and Posting   Ivan Stepaniuk
• [asterisk-users] Call Recording and Posting   Dan Journo
• [asterisk-users] meetme and confbridge   Robor Oghene
• [asterisk-users] tealtime static   Robor Oghene
• [asterisk-users] SPRINTF option : format %1$s not supported   Tilghman Lesher
• [asterisk-users] libss7 problem with dialing a non numeric string   Rennes Neps
• [asterisk-users] SPRINTF option : format %1$s not supported   Steve Murphy
• [asterisk-users] live audio streaming using monitor, mixmonitor or chanspy   nik600
• [asterisk-users] Asterisk 1.6 with TDM2400 and 2 FXS modules   Eckhard Jokisch
• [asterisk-users] Asterisk 1.6 with TDM2400 and 2 FXS modules   Tim Nelson
• [asterisk-users] G729 in asterisk upgrade issue   Luis Silva
• [asterisk-users] G729 in asterisk upgrade issue   Danny Nicholas
• [asterisk-users] G729 in asterisk upgrade issue   Kevin P. Fleming
• [asterisk-users] Asterisk to Asterisk access voicemail - not working   Joseph
• [asterisk-users] How to send a digit to a channel??   Pablo Bernasconi
• [asterisk-users] Asterisk 1.6 with TDM2400 and 2 FXS modules   Silvère Maugain
• [asterisk-users] video support with voicemail question   Gallmeier, Jonathan
• [asterisk-users] Call Recording and Posting   Ivan Stepaniuk
• [asterisk-users] Call Recording and Posting   Dan Journo
• [asterisk-users] asterisk dialplan to share fax line   Alex Samad
• [asterisk-users] Call Recording and Posting   Steve Edwards
• [asterisk-users] Call Recording and Posting   Dan Journo
• [asterisk-users] [SOLVED] Asterisk to Asterisk access voicemail - not working   Joseph
• [asterisk-users] Call Recording and Posting   Steve Edwards
• [asterisk-users] Call Recording and Posting   Ivan Stepaniuk
• [asterisk-users] Call Recording and Posting   Ivan Stepaniuk
• [asterisk-users] [SOLVED] Asterisk to Asterisk access voicemail - not working   Ivan Stepaniuk
• [asterisk-users] strange transcoding values   Gianluca Baù
• [asterisk-users] missing CDR records in cdr while kick from meetme   DHAVAL INDRODIYA
• [asterisk-users] strange transcoding values   Jeff LaCoursiere
• [asterisk-users] strange transcoding values   Gianluca Baù
• [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4   Miguel Molina
• [asterisk-users] strange transcoding values   Tzafrir Cohen
• [asterisk-users] G729 in asterisk upgrade issue   Luis Silva
• [asterisk-users] Dial Delay   Ron
• [asterisk-users] Dial Delay   Danny Nicholas
• [asterisk-users] Call Recording and Posting   Elliot Otchet
• [asterisk-users] Best Firewall Suggestions?   David Wathen
• [asterisk-users] Best Firewall Suggestions?   Doug Lytle
• [asterisk-users] Dial Delay   Ron
• [asterisk-users] Best Firewall Suggestions?   Pascal Bruno
• [asterisk-users] Dial Delay   Danny Nicholas
• [asterisk-users] Call Recording and Posting   Steve Edwards
• [asterisk-users] Best Firewall Suggestions?   Peder
• [asterisk-users] Bridge command in 1.6   Matthew Harrell
• [asterisk-users] Best Firewall Suggestions?   Gordon Henderson
• [asterisk-users] Best Firewall Suggestions?   Fred Posner
• [asterisk-users] Best Firewall Suggestions?   SIP
• [asterisk-users] Best Firewall Suggestions?   Steve Totaro
• [asterisk-users] Best Firewall Suggestions?   Karl Fife
• [asterisk-users] Best Firewall Suggestions?   "Juan E. Rodríguez"
• [asterisk-users] Best Firewall Suggestions?   Grygoriy Dobrovolskyy
• [asterisk-users] Best Firewall Suggestions?   Steve Totaro
• [asterisk-users] Dial Delay   Matt Riddell
• [asterisk-users] Best Firewall Suggestions?   Darren Wiebe
• [asterisk-users] tealtime static   Carlos Chavez
• [asterisk-users] Best Firewall Suggestions?   Michiel van Baak
• [asterisk-users] wrond DTMF detection on Zap channel   nik600
• [asterisk-users] Best Firewall Suggestions?   Hans Witvliet
• [asterisk-users] Best Firewall Suggestions?   Michiel van Baak
• [asterisk-users] Best Firewall Suggestions?   Alex Samad
• [asterisk-users] Best Firewall Suggestions?   Karl Fife
• [asterisk-users] AMI input streams limit?   Jose P. Espinal
• [asterisk-users] Asterisk in the Cloud   Dan Journo
• [asterisk-users] outgoing sip calls work; incoming calls fail   listmail at websage.ca
• [asterisk-users] T38 negotiations in RTP   Ujjval Karihaloo
• [asterisk-users] Asterisk in the Cloud   C F
• [asterisk-users] AMI input streams limit?   David Backeberg
• [asterisk-users] AMI input streams limit?   Jose P. Espinal
• [asterisk-users] AMI input streams limit?   David Backeberg
• [asterisk-users] AMI input streams limit?   Jose P. Espinal
• [asterisk-users] AMI input streams limit?   Matt Riddell
• [asterisk-users] T38 negotiations in RTP   Kevin P. Fleming
• [asterisk-users] Best Firewall Suggestions?   Myles Wakeham
• [asterisk-users] AMI input streams limit?   Jose P. Espinal
• [asterisk-users] ChanSpy on asterisk 1.6   Jorge Gutiérrez
• [asterisk-users] T38 negotiations in RTP   Ujjval Karihaloo
• [asterisk-users] T38 negotiations in RTP   Kevin P. Fleming
• [asterisk-users] Asterisk in the Cloud   Dovid Bender
• [asterisk-users] Asterisk in the Cloud   Dovid Bender
• [asterisk-users] Best Firewall Suggestions?   Patrick Plattes
• [asterisk-users] DAHDI Dummy for Linux VServers   kn0x
• [asterisk-users] Asterisk in the Cloud   Eric Chamberlain
• [asterisk-users] FXS to SIP gateway   jonas kellens
• [asterisk-users] Asterisk in the Cloud   Dan Journo
• [asterisk-users] Asterisk in the Cloud   Dan Journo
• [asterisk-users] SIP - busy tone detection   Asterisk
• [asterisk-users] DTMF failing in some calls   abdelkader
• [asterisk-users] FXS to SIP gateway   Ivan Stepaniuk
• [asterisk-users] outgoing sip calls work; incoming calls fail   Ivan Stepaniuk
• [asterisk-users] SIP - busy tone detection   Ivan Stepaniuk
• [asterisk-users] Best Firewall Suggestions?   Benny Amorsen
• [asterisk-users] SIP RealTime defaultuser Field Cleared   Stuart Elvish
• [asterisk-users] ChanSpy on asterisk 1.6   DHAVAL INDRODIYA
• [asterisk-users] Queues with unavailable members   Benny Amorsen
• [asterisk-users] Asterisk 1.4 vs 1.6   Dan Journo
• [asterisk-users] Asterisk 1.4 vs 1.6   David Backeberg
• [asterisk-users] Queues with unavailable members   Lenz Emilitri
• [asterisk-users] no outbound calls   Ott Rose
• [asterisk-users] Queues with unavailable members   RSCL Mumbai
• [asterisk-users] multiple call   Miguel Molina
• [asterisk-users] Queues with unavailable members   Elliot Otchet
• [asterisk-users] DAHDI Dummy for Linux VServers   Vinícius Fontes
• [asterisk-users] Config Files   Matt
• [asterisk-users] ChanSpy   Torintino T
• [asterisk-users] DAHDI Dummy for Linux VServers   Tzafrir Cohen
• [asterisk-users] FXS to SIP gateway   Jonathan Thurman
• [asterisk-users] ChanSpy   Rennes Neps
• [asterisk-users] multiple call   Faheem
• [asterisk-users] multiple call   Danny Nicholas
• [asterisk-users] Config Files   Faheem
• [asterisk-users] ACD & ASR   B.Masoud at SH
• [asterisk-users] ACD & ASR   Grygoriy Dobrovolskyy
• [asterisk-users] PostgreSQL problems   David A. Bandel
• [asterisk-users] PostgreSQL problems   Roderick A. Anderson
• [asterisk-users] Best Firewall Suggestions?   David Wathen
• [asterisk-users] Cisco router   Julian Lyndon-Smith
• [asterisk-users] Best Firewall Suggestions?   Dr. Michael J. Chudobiak
• [asterisk-users] Cisco router   Jonathan Thurman
• [asterisk-users] multiple call   kaustuvak_b at bbsr.syscomes.com
• [asterisk-users] Config Files   C. Chad Wallace
• [asterisk-users] ChanSpy   Torintino T
• [asterisk-users] WaitForSilence doesn't work unless Background is called?   Devin Jacobs
• [asterisk-users] Cisco router   Julian Lyndon-Smith
• [asterisk-users] Cisco router   Peder
• [asterisk-users] ACD & ASR   Ivan Stepaniuk
• [asterisk-users] Door Phones   Dan Journo
• [asterisk-users] Cisco router   Jonathan Thurman
• [asterisk-users] Door Phones   Jonathan Thurman
• [asterisk-users] Door Phones   Cory Andrews
• [asterisk-users] Door Phones   John A. Sullivan III
• [asterisk-users] Door Phones   Dan Journo
• [asterisk-users] Extension Paging   Dan Journo
• [asterisk-users] Extension Paging   Jimmy Godbout
• [asterisk-users] Extension Paging   Dan Journo
• [asterisk-users] Extension Paging   C F
• [asterisk-users] outgoing sip calls work; incoming calls fail   listmail at websage.ca
• [asterisk-users] MWI for multiple voice mail boxes   John A. Sullivan III
• [asterisk-users] Callpickup works for outside calls but not inside calls   John A. Sullivan III
• [asterisk-users] multiple call   Lenz Emilitri
• [asterisk-users] multiple call   Matt Riddell
• [asterisk-users] Calls hang up after 20 seconds   Gianni Fioretta
• [asterisk-users] Calls hang up after 20 seconds   Ishfaq Malik
• [asterisk-users] Queues with unavailable members   Benny Amorsen
• [asterisk-users] Queues with unavailable members   Benny Amorsen
• [asterisk-users] Calls hang up after 20 seconds   Vieri
• [asterisk-users] Asterisk with a Cisco AS5300 gateway   Phibee Network Operation Center
• [asterisk-users] Calls hang up after 20 seconds   SIP
• [asterisk-users] hi   as asd
• [asterisk-users] hi   Mike Bessette
• [asterisk-users] Queues with unavailable members   Elliot Otchet
• [asterisk-users] Where to find IMAP storage doc ?   Olivier
• [asterisk-users] Queues with unavailable members   Benny Amorsen
• [asterisk-users] Asterisk with a Cisco AS5300 gateway   Jonathan Thurman
• [asterisk-users] Where to find IMAP storage doc ?   Matthew Harrell
• [asterisk-users] Does ADA 1.1 or ADA Pro exists ?   Olivier
• [asterisk-users] ChanSpy on asterisk 1.6   Jorge Gutiérrez
• [asterisk-users] Asterisk with a Cisco AS5300 gateway   David Backeberg
• [asterisk-users] PSTN to SIP line ratio   Shahnawaz Mir
• [asterisk-users] sporadic one-way audio   Brent Davidson
• [asterisk-users] best way to make 5-10 simultaneous calls to the same did at a set time of day   Eric Fort
• [asterisk-users] PSTN to SIP line ratio   Ivan Stepaniuk
• [asterisk-users] PSTN to SIP line ratio   Steve Edwards
• [asterisk-users] best way to make 5-10 simultaneous calls to the same did at a set time of day   Steve Edwards
• [asterisk-users] PSTN to SIP line ratio   Tim Nelson
• [asterisk-users] PSTN to SIP line ratio   Jorge Mendoza
• [asterisk-users] MWI for multiple voice mail boxes   Olivier
• [asterisk-users] Where to find IMAP storage doc ?   Olivier
• [asterisk-users] MWI for multiple voice mail boxes   John A. Sullivan III
• [asterisk-users] MWI for multiple voice mail boxes   Danny Nicholas
• [asterisk-users] PSTN to SIP line ratio   Shahnawaz Mir
• [asterisk-users] ChanSpy   Rennes Neps
• [asterisk-users] Where to find IMAP storage doc ?   John A. Sullivan III
• [asterisk-users] MWI for multiple voice mail boxes   John A. Sullivan III
• [asterisk-users] MWI for multiple voice mail boxes   Danny Nicholas
• [asterisk-users] MWI for multiple voice mail boxes   Danny Nicholas
• [asterisk-users] Queues with unavailable members   C. Chad Wallace
• [asterisk-users] no outbound calls   das sandesh
• [asterisk-users] Queues with unavailable members   C. Chad Wallace
• [asterisk-users] Testing the Timing Device   Danny Nicholas
• [asterisk-users] Testing the Timing Device   Dan Journo
• [asterisk-users] Best way to detect fax in Asterisk 1.6??   Pablo Bernasconi
• [asterisk-users] DS3 capacity calls using asterisk   das sandesh
• [asterisk-users] OT - Can't upgrade Cisco 7942 to SIP   Olivier
• [asterisk-users] DS3 capacity calls using asterisk   Tim Nelson
• [asterisk-users] Testing the Timing Device   Danny Nicholas
• [asterisk-users] Testing the Timing Device   Dan Journo
• [asterisk-users] A little OT but need an opinion on Aastra 57i CT   John Millican
• [asterisk-users] PSTN to SIP line ratio   Steve Edwards
• [asterisk-users] 2 IPs for an Asterisk server.   Guillaume Yziquel
• [asterisk-users] Testing the Timing Device   Dan Journo
• [asterisk-users] Testing the Timing Device   Danny Nicholas
• [asterisk-users] A little OT but need an opinion on Aastra 57i CT   Ira
• [asterisk-users] 2 IPs for an Asterisk server.   Jorge Gutiérrez
• [asterisk-users] A little OT but need an opinion on Aastra 57i CT   Michelle Dupuis
• [asterisk-users] PSTN to SIP line ratio   Tim Nelson
• [asterisk-users] DS3 capacity calls using asterisk   Steve Edwards
• [asterisk-users] 2 IPs for an Asterisk server.   Guillaume Yziquel
• [asterisk-users] Asterisk with a Cisco AS5300 gateway   F6HQZ
• [asterisk-users] DS3 capacity calls using asterisk   Alex Balashov
• [asterisk-users] DS3 capacity calls using asterisk   David Backeberg
• [asterisk-users] DS3 capacity calls using asterisk   David Backeberg
• [asterisk-users] Where to find IMAP storage doc ?   Matthew Harrell
• [asterisk-users] DS3 capacity calls using asterisk   Alex Balashov
• [asterisk-users] question on SIP and call manager   Jerry Geis
• [asterisk-users] question on SIP and call manager   Danny Nicholas
• [asterisk-users] Asterisk in the Cloud   Eric Chamberlain
• [asterisk-users] Asterisk and FreePBX Amazon EC2 instances are now available in Europe   Eric Chamberlain
• [asterisk-users] OT wanted old Sipura firmware 2.0.13   Joseph
• [asterisk-users] MWI for multiple voice mail boxes   Jared Smith
• [asterisk-users] Testing the Timing Device   Tzafrir Cohen
• [asterisk-users] question on SIP and call manager   Jerry Geis
• [asterisk-users] MWI for multiple voice mail boxes   John A. Sullivan III
• [asterisk-users] Callpickup works for outside calls but not inside calls   John A. Sullivan III
• [asterisk-users] Mixing SIP/TDM in MeetMe   Richard Kenner
• [asterisk-users] The City of Amsterdam has been deploying asterisk throughout the city!   Ron Arts
• [asterisk-users] Linux/Asterisk on game consoles?   Vincent
• [asterisk-users] Linux/Asterisk on game consoles?   Stelios Koroneos
• [asterisk-users] The City of Amsterdam has been deploying asterisk throughout the city!   Ivan Stepaniuk
• [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4   Dovid Bender
• [asterisk-users] sporadic one-way audio   Ishfaq Malik
• [asterisk-users] multiple call   Dovid Bender
• [asterisk-users] DTMF failing in some calls   Dovid Bender
• [asterisk-users] Queues with unavailable members   Benny Amorsen
• [asterisk-users] Check if a variable is set   Ishfaq Malik
• [asterisk-users] Invite after bye?   Josip Djuricic
• [asterisk-users] Check if a variable is set   ABBAS SHAKEEL
• [asterisk-users] Check if a variable is set   ABBAS SHAKEEL
• [asterisk-users] Check if a variable is set   Ishfaq Malik
• [asterisk-users] Soft phone not registering   Rakesh Sabharwal
• [asterisk-users] Invite after bye?   Kevin P. Fleming
• [asterisk-users] Check if a variable is set   ABBAS SHAKEEL
• [asterisk-users] Check if a variable is set   Darrin Henshaw
• [asterisk-users] Check if a variable is set   Ishfaq Malik
• [asterisk-users] Check if a variable is set   Darrin Henshaw
• [asterisk-users] Check if a variable is set   Danny Nicholas
• [asterisk-users] Check if a variable is set   Darrin Henshaw
• [asterisk-users] Check if a variable is set   Darrin Henshaw
• [asterisk-users] Soft phone not registering   Darrin Henshaw
• [asterisk-users] question on SIP and call manager   Danny Nicholas
• [asterisk-users] app_swift issue   Jeremy Kister
• [asterisk-users] Check if a variable is set   Ishfaq Malik
• [asterisk-users] Check if a variable is set   Ishfaq Malik
• [asterisk-users] Check if a variable is set   Danny Nicholas
• [asterisk-users] OT wanted old Sipura firmware 2.0.13   Dave Fullerton
• [asterisk-users] Queues with unavailable members   Benny Amorsen
• [asterisk-users] inquire if SIP connections are active or not   Jerry Geis
• [asterisk-users] Check if a variable is set   Ishfaq Malik
• [asterisk-users] inquire if SIP connections are active or not   Darrin Henshaw
• [asterisk-users] Asterisk with a Cisco AS5300 gateway   Jonathan Thurman
• [asterisk-users] MWI for multiple voice mail boxes   Danny Nicholas
• [asterisk-users] OT wanted old Sipura firmware 2.0.13   Joseph
• [asterisk-users] Origin of "Exceptionally long voice queue length queuing to IAX2/blahblah" messages   daren ferreira
• [asterisk-users] OT Old Sipura OK - Linksys (junk)   Joseph
• [asterisk-users] MWI for multiple voice mail boxes   John A. Sullivan III
• [asterisk-users] MWI for multiple voice mail boxes   Danny Nicholas
• [asterisk-users] Queues with unavailable members   C. Chad Wallace
• [asterisk-users] MWI for multiple voice mail boxes   John A. Sullivan III
• [asterisk-users] Linux/Asterisk on game consoles?   Noah Miller
• [asterisk-users] Queues with unavailable members   Benny Amorsen
• [asterisk-users] Where to find IMAP storage doc ?   Noah Miller
• [asterisk-users] MWI for multiple voice mail boxes   Danny Nicholas
• [asterisk-users] can i use Asterisk to send sms to my database users?   kazabe
• [asterisk-users] can i use Asterisk to send sms to my databaseusers?   Danny Nicholas
• [asterisk-users] Queues with unavailable members   C. Chad Wallace
• [asterisk-users] SIP to IAX to SIP   George Farris
• [asterisk-users] SIP to IAX to SIP   Ivan Stepaniuk
• [asterisk-users] Linux/Asterisk on game consoles?   Adam Moffett
• [asterisk-users] Linux/Asterisk on game consoles?   C. Chad Wallace
• [asterisk-users] Whither asterisk-addons?   Chris Brentano
• [asterisk-users] IVR   Nazir Ahmed Vaid
• [asterisk-users] Nehalem & Digium Wildcard issues?   Chris Brentano
• [asterisk-users] Whither asterisk-addons?   Chris Brentano
• [asterisk-users] IVR   Lenz Emilitri
• [asterisk-users] IVR   Alex Balashov
• [asterisk-users] OT - In which countries are ISDN subaddressesused ?   Olivier
• [asterisk-users] OT - KVM and PCI telephony cards   Olivier
• [asterisk-users] how to limit the calls leaving a queue?   Niccolò Belli
• [asterisk-users] OT - DECT SIP Phones   --[ UxBoD ]--
• [asterisk-users] OT - DECT SIP Phones   Gordon Henderson
• [asterisk-users] OT - DECT SIP Phones   --[ UxBoD ]--
• [asterisk-users] sched_settime: Request to schedule in the past?!?!   Bart Fisher
• [asterisk-users] OT - DECT SIP Phones   --[ UxBoD ]--
• [asterisk-users] OT - DECT SIP Phones   Randy R
• [asterisk-users] Possible bug in app_meetme.c   Richard Kenner
• [asterisk-users] OT - DECT SIP Phones   Darrick Hartman
• [asterisk-users] Possible bug in app_meetme.c   Moises Silva
• [asterisk-users] how to limit the calls leaving a queue?   Paul Hales
• [asterisk-users] OT - DECT SIP Phones   Nick Morrott
• [asterisk-users] Best practices for reliable carrier grade telephony   Alistair Cunningham
• [asterisk-users] Asterisk Monitoring   Alex Balashov
• [asterisk-users] Asterisk Monitoring   Dan Journo
• [asterisk-users] Asterisk Monitoring   Jarrod Lash
• [asterisk-users] Asterisk Monitoring   Jai Rangi
• [asterisk-users] Possible bug in app_meetme.c   David Backeberg
• [asterisk-users] OT - Can't upgrade Cisco 7942 to SIP   Jonathan Thurman
• [asterisk-users] Possible bug in app_meetme.c   David Backeberg
• [asterisk-users] Soft phone not registering   Rakesh Sabharwal
• [asterisk-users] Possible bug in app_meetme.c   Richard Kenner
• [asterisk-users] Astricon   Jeff LaCoursiere
• [asterisk-users] Astricon   Alex Balashov
• [asterisk-users] Astricon   Michael Graves
• [asterisk-users] Possible bug in app_meetme.c   Richard Kenner
• [asterisk-users] Asterisk Monitoring   John A. Sullivan III
• [asterisk-users] Asterisk Monitoring   Steve Totaro
• [asterisk-users] SIP debugging enabled : written to log ??   jonas kellens
• [asterisk-users] Sunday 18th 12N-3P PDST Global Asterisk Mtg - BerkeleyTIP - for forwarding   john_re
• [asterisk-users] Where to find IMAP storage doc ?   Tzafrir Cohen
• [asterisk-users] OT - DECT SIP Phones   --[ UxBoD ]--
• [asterisk-users] Linux/Asterisk on game consoles?   Tzafrir Cohen
• [asterisk-users] Snom870 sidecar   Olivier
• [asterisk-users] Where to find IMAP storage doc ?   Olivier
• [asterisk-users] Snom870 sidecar   Christian Stredicke
• [asterisk-users] Where to find IMAP storage doc ?   Tzafrir Cohen
• [asterisk-users] how to limit the calls leaving a queue?   Niccolò Belli
• [asterisk-users] OT - DECT SIP Phones   Michael Graves
• [asterisk-users] OT - DECT SIP Phones   Alan Lord (News)
• [asterisk-users] Kernel Timer   Dan Journo
• [asterisk-users] Asterisk+Sphinx4 for simple mobile phone <-> server speech recognition   johny jj2
• [asterisk-users] OT - DECT SIP Phones   --[ UxBoD ]--
• [asterisk-users] Asterisk Monitoring   realmj at gmail.com
• [asterisk-users] OT - DECT SIP Phones   Michael Graves
• [asterisk-users] OT - DECT SIP Phones   Gordon Henderson
• [asterisk-users] Call from 'sip-id' to extension 'sip-id' rejected because extension not found ?   sean darcy
• [asterisk-users] Call from 'sip-id' to extension 'sip-id' rejected because extension not found ?   sean darcy
• [asterisk-users] Asterisk Expert Freelancer in Karachi   Nazir Ahmed Vaid
• [asterisk-users] OT - DECT SIP Phones   Mr. James W. Laferriere
• [asterisk-users] Snom870 sidecar   Doug Lytle
• [asterisk-users] Asterisk Expert Freelancer in Karachi   Alex Balashov
• [asterisk-users] DTMF   didier.cuffaut
• [asterisk-users] SIP debugging enabled : written to log   Neeraj Chand
• [asterisk-users] OT - DECT SIP Phones   Barry Miller
• [asterisk-users] SIP Headers   Dan Journo
• [asterisk-users] Customising Firmware   Dan Journo
• [asterisk-users] BTS   Dean Collins
• [asterisk-users] BTS   Steve Totaro
• [asterisk-users] BTS   Eric Fort
• [asterisk-users] SIP Headers   Michelle Dupuis
• [asterisk-users] BTS   Dean Collins
• [asterisk-users] MWI for multiple voice mail boxes   Tilghman Lesher
• [asterisk-users] MWI for multiple voice mail boxes   John A. Sullivan III
• [asterisk-users] Where to find IMAP storage doc ?   Olivier
• [asterisk-users] Snom870 sidecar   Olivier
• [asterisk-users] Customising Firmware   Steve Edwards
• [asterisk-users] OT - DECT SIP Phones   --[ UxBoD ]--
• [asterisk-users] Asterisk Monitoring   Lee Archer
• [asterisk-users] SIP Headers   Lee Archer
• [asterisk-users] OT - DECT SIP Phones   Gordon Henderson
• [asterisk-users] announcement tone to callees of app_page   Jeremy Kister
• [asterisk-users] Invite after bye?   Alex Balashov
• [asterisk-users] SIP debugging enabled : written to log   jonas kellens
• [asterisk-users] question about getting instance ringing member in queue   Rilawich Ango
• [asterisk-users] sporadic one-way audio   Steve Davies
• [asterisk-users] Mixing SIP/TDM in MeetMe   Steve Davies
• [asterisk-users] Calls hang up after 20 seconds   Kevin P. Fleming
• [asterisk-users] Calls hang up after 20 seconds   SIP
• [asterisk-users] Calls hang up after 20 seconds   Alex Balashov
• [asterisk-users] digium fax: can't indicate condition 19?   Scott L. Lykens
• [asterisk-users] digium fax: can't indicate condition 19?   Kevin P. Fleming
• [asterisk-users] Customising Firmware   Ivan Stepaniuk
• [asterisk-users] Asterisk+Sphinx4 for simple mobile phone <->server speech recognition   Danny Nicholas
• [asterisk-users] Snom870 sidecar   Usman Tahir
• [asterisk-users] *****SPAM***** sched_settime: Request to schedule in the past?!?!   Danny Nicholas
• [asterisk-users] Calls hang up after 20 seconds   SIP
• [asterisk-users] Missing digits from CallerID on TDM400P?   Remco Barendse
• [asterisk-users] Customising Firmware   Dan Journo
• [asterisk-users] VoIP interconnection with Acme packet SBC   Kasun Daminda
• [asterisk-users] Customising Firmware   Steve Totaro
• [asterisk-users] Problems replaying G.726 - only noise   Michael Hirschbichler
• [asterisk-users] VoIP interconnection with Acme packet SBC   Vijay Gandhi
• [asterisk-users] SIP to IAX to SIP   George Farris
• [asterisk-users] MWI for multiple voice mail boxes   Tilghman Lesher
• [asterisk-users] update CDRs in mysql during a call   mickael ropars
• [asterisk-users] update CDRs in mysql during a call   Danny Nicholas
• [asterisk-users] Astricon talk on wideband codecs   Zoaaaaa
• [asterisk-users] update CDRs in mysql during a call   Scott L. Lykens
• [asterisk-users] Astricon talk on wideband codecs   Fred Posner
• [asterisk-users] asterisk services not starting up   Ott Rose
• [asterisk-users] update CDRs in mysql during a call   mickael ropars
• [asterisk-users] Snom870 sidecar   Olivier
• [asterisk-users] update CDRs in mysql during a call   Steve Edwards
• [asterisk-users] asterisk services not starting up   Steve Edwards
• [asterisk-users] delay in processing dtmf   Waruna de Silva
• [asterisk-users] delay in processing dtmf   Danny Nicholas
• [asterisk-users] Astricon talk on wideband codecs   John Todd
• [asterisk-users] delay in processing dtmf   Waruna de Silva
• [asterisk-users] asterisk services not starting up   Ott Rose
• [asterisk-users] Looking for the asterisk 'off' sound file   JR Richardson
• [asterisk-users] asterisk services not starting up   Steve Edwards
• [asterisk-users] Looking for the asterisk 'off' sound file   Steve Edwards
• [asterisk-users] delay in processing dtmf   Danny Nicholas
• [asterisk-users] Looking for the asterisk 'off' sound file   Doug Lytle
• [asterisk-users] delay in processing dtmf   Ioan Indreias
• [asterisk-users] delay in processing dtmf   Danny Nicholas
• [asterisk-users] IMAP voicemail using subfolders fails.   Barry L. Kline
• [asterisk-users] Dial a external number with extension   Anahi Ludueña
• [asterisk-users] delay in processing dtmf   Kristian Kielhofner
• [asterisk-users] ANN: Asterisk-Java 1.0.0.M3 Released   Stefan Reuter
• [asterisk-users] delay in processing dtmf   Danny Nicholas
• [asterisk-users] IMAP voicemail using subfolders fails.   John A. Sullivan III
• [asterisk-users] Cisco 1751 setup with asterisk   Joseph
• [asterisk-users] IMAP voicemail using subfolders fails.   Barry L. Kline
• [asterisk-users] IMAP voicemail using subfolders fails.   Barry L. Kline
• [asterisk-users] IMAP voicemail using subfolders fails.   John A. Sullivan III
• [asterisk-users] IMAP voicemail using subfolders fails.   Barry L. Kline
• [asterisk-users] all our circuits are busy now   B.Masoud at SH
• [asterisk-users] all our circuits are busy now   Paul Hales
• [asterisk-users] all our circuits are busy now   Matt Riddell
• [asterisk-users] Astricon   John Todd
• [asterisk-users] delay in processing dtmf   Waruna de Silva
• [asterisk-users] SIP debugging enabled : written to log ??   Dovid Bender
• [asterisk-users] Voip interconnection with ACME SBC-asterisk-users Digest, Vol 63, Issue 52   Kasun Daminda
• [asterisk-users] Astricon   Darrick Hartman
• [asterisk-users] (no subject)   mickael ropars
• [asterisk-users] Dial a external number with extension   mickael ropars
• [asterisk-users] Problems replaying G.726 - only noise   Michael Hirschbichler
• [asterisk-users] Cisco 7921   Torintino T
• [asterisk-users] Cisco 7921   Michiel van Baak
• [asterisk-users] need help with card for IVR   Roshan Singh
• [asterisk-users] SOLVED: Problems replaying G.726 - only noise   Michael Hirschbichler
• [asterisk-users] AMI 1.0 -> 1.1 with originate.   Guillaume Yziquel
• [asterisk-users] how to limit the calls leaving a queue?   Niccolò Belli
• [asterisk-users] Dial a external number with extension   Anahi Ludueña
• [asterisk-users] cascaded pickup   Christophorus Laube
• [asterisk-users] Dial a external number with extension   Marc Charbonneau
• [asterisk-users] how to limit the calls leaving a queue?   Ron Arts
• [asterisk-users] Music On Hold   Cyprus VoIP
• [asterisk-users] Syncronizing files on different Asterisk servers   ABBAS SHAKEEL
• [asterisk-users] Astricon   Jeff LaCoursiere
• [asterisk-users] Syncronizing files on different Asterisk servers   Jeff LaCoursiere
• [asterisk-users] Astricon   Barry L. Kline
• [asterisk-users] Syncronizing files on different Asterisk servers   ABBAS SHAKEEL
• [asterisk-users] Syncronizing files on different Asterisk servers   Jeff LaCoursiere
• [asterisk-users] Syncronizing files on different Asteriskservers   Arjan Kroon | Mobillion
• [asterisk-users] IMAP voicemail using subfolders fails.   Kevin P. Fleming
• [asterisk-users] Syncronizing files on different Asteriskservers   Robin
• [asterisk-users] AMI 1.0 -> 1.1 with originate.   Miguel Molina
• [asterisk-users] srtp crypto lifetime   Szasz Szabolcs
• [asterisk-users] how to limit the calls leaving a queue?   Miguel Molina
• [asterisk-users] IMAP voicemail using subfolders fails.   --[ UxBoD ]--
• [asterisk-users] (no subject)   Danny Nicholas
• [asterisk-users] how to limit the calls leaving a queue?   Niccolò Belli
• [asterisk-users] Cisco 1751 setup with asterisk   Jonathan Thurman
• [asterisk-users] IMAP voicemail using subfolders fails.   Barry L. Kline
• [asterisk-users] troubleshooting NAT   Ott Rose
• [asterisk-users] Dial a external number with extension   Alec Davis
• [asterisk-users] (no subject)   Steve Edwards
• [asterisk-users] High Volume Call Center SIP versus IAX2   Robert Grignon
• [asterisk-users] High Volume Call Center SIP versus IAX2   Danny Nicholas
• [asterisk-users] High Volume Call Center SIP versus IAX2   David Gibbons
• [asterisk-users] High Volume Call Center SIP versus IAX2   Matt Riddell
• [asterisk-users] High Volume Call Center SIP versus IAX2   Danny Nicholas
• [asterisk-users] High Volume Call Center SIP versus IAX2   Steve Edwards
• [asterisk-users] Cisco 1751 setup with asterisk   Joseph
• [asterisk-users] Cisco 1751 setup with asterisk   B.Masoud at SH
• [asterisk-users] Libpri-1.4.10.2 Released   Asterisk Development Team
• [asterisk-users] Cisco 1751 setup with asterisk   Carlos Chavez
• [asterisk-users] High Volume Call Center SIP versus IAX2   Zoaaaaa
• [asterisk-users] Kernel panic w/ DAHDI 2.x/Digium TE220B   Chris Brentano
• [asterisk-users] Is there a way to force a codec on an incoming sip uri call?   Eric Chamberlain
• [asterisk-users] Cisco 1751 setup with asterisk   Joseph
• [asterisk-users] OutCALL   Dan Journo
• [asterisk-users] OutCALL   Dan Journo
• [asterisk-users] IMAP voicemail using subfolders fails.   Kevin P. Fleming
• [asterisk-users] Linksys 962   Jeff LaCoursiere
• [asterisk-users] all our circuits are busy now   B.Masoud at SH
• [asterisk-users] Linksys 962   Jimmy Godbout
• [asterisk-users] all our circuits are busy now   C. Chad Wallace
• [asterisk-users] Is there a way to force a codec on an incoming sip uri call?   Martin
• [asterisk-users] troubleshooting NAT   Martin
• [asterisk-users] ChannelStateDesc: Ring ?   Guillaume Yziquel
• [asterisk-users] ChannelStateDesc: Ring ?   Martin
• [asterisk-users] Syncronizing files on different Asterisk servers   Joseph
• [asterisk-users] Syncronizing files on different Asterisk servers   ABBAS SHAKEEL
• [asterisk-users] Syncronizing files on different Asterisk servers   Robin
• [asterisk-users] Syncronizing files on different Asteriskservers   Arjan Kroon | Mobillion
• [asterisk-users] error - sources for the 2.6.18-92.1.22.el5xen kernel   PATRICK KANGETHE
• [asterisk-users] wrond DTMF detection on Zap channel   nik600
• [asterisk-users] Linksys 962   Stefan Schmidt
• [asterisk-users] Searching on how to keep local calls... local   jonas kellens
• [asterisk-users] Searching on how to keep local calls... local   Kyle Kienapfel
• [asterisk-users] error - sources for the 2.6.18-92.1.22.el5xen kernel   Chandrakant Solanki
• [asterisk-users] RAMDisk vs Extarnal server for recording   Robin
• [asterisk-users] Calls hang up after 20 seconds   Gianni Fioretta
• [asterisk-users] Searching on how to keep local calls... local   jonas kellens
• [asterisk-users] Need Help   kiran.reddy at mpowerglobal.in
• [asterisk-users] Need Help   kiran.reddy at mpowerglobal.in
• [asterisk-users] Need Help   kiran.reddy at mpowerglobal.in
• [asterisk-users] RAMDisk vs Extarnal server for recording   Zoaaaaa
• [asterisk-users] error - sources for the 2.6.18-92.1.22.el5xen kernel   PATRICK KANGETHE
• [asterisk-users] RAMDisk vs Extarnal server for recording   Robin
• [asterisk-users] TxFax works only with one of 2 PRI   martin cabrera
• [asterisk-users] IMAP voicemail using subfolders fails.   Barry L. Kline
• [asterisk-users] ChannelStateDesc: Ring ?   Guillaume Yziquel
• [asterisk-users] ringing... or lack thereof   Jeff LaCoursiere
• [asterisk-users] Linksys 962   Jeff LaCoursiere
• [asterisk-users] Linksys 962   Jeff LaCoursiere
• [asterisk-users] AMI 1.0 -> 1.1 with originate.   Guillaume Yziquel
• [asterisk-users] Need Help   kiran.reddy at mpowerglobal.in
• [asterisk-users] Need Help   Steve Edwards
• [asterisk-users] troubleshooting NAT   Ott Rose
• [asterisk-users] Astricon   Bob Pierce
• [asterisk-users] Linksys 962   Andres
• [asterisk-users] Astricon   Randy R
• [asterisk-users] polarity on some channels   B.Masoud at SH
• [asterisk-users] troubleshooting NAT   Ott Rose
• [asterisk-users] Intermittent Low volume   Robert Grignon
• [asterisk-users] RAMDisk vs Extarnal server for recording   Matt Florell
• [asterisk-users] RAMDisk vs Extarnal server for recording   Robin
• [asterisk-users] troubleshooting NAT   Warren Selby
• [asterisk-users] RAMDisk vs Extarnal server for recording   Matt Florell
• [asterisk-users] DAHDI: TCM PCI Master abort   Greg Woods
• [asterisk-users] RAMDisk vs Extarnal server for recording   Robin
• [asterisk-users] troubleshooting NAT   Ott Rose
• [asterisk-users] OT - Gigaset Chagall - How to download firmware without Internet access ?   Olivier
• [asterisk-users] OT - Gigaset Chagall - How to download firmware without Internet access ?   Leif Madsen
• [asterisk-users] Syncronizing files on different Asterisk servers   lists at torrenga.com
• [asterisk-users] Astricon   Barry L. Kline
• [asterisk-users] Searching on how to keep local calls... local   Kyle Kienapfel
• [asterisk-users] DS3 capacity calls using asterisk   das sandesh
• [asterisk-users] Astricon   Randy R
• [asterisk-users] Astricon   Danny Nicholas
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   das sandesh
• [asterisk-users] RAMDisk vs Extarnal server for recording   David Backeberg
• [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine   Vela Sivasankaran
• [asterisk-users] polarity on some channels   Lyle Giese
• [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine   Danny Nicholas
• [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine   Christophorus Laube
• [asterisk-users] RAMDisk vs Extarnal server for recording   Matt Florell
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Steve Edwards
• [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine   Steve Edwards
• [asterisk-users] Unstable PRI interface: Link restart after few min::   research at businesstz.com
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   David Backeberg
• [asterisk-users] Astricon   SIP
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   das sandesh
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Jai Rangi
• [asterisk-users] Concurrent calls including mysql taking lot oftime for execution   Danny Nicholas
• [asterisk-users] Concurrent calls including mysql taking lot oftime for execution   Jeff LaCoursiere
• [asterisk-users] Incorrect voice mail format on transfer   John A. Sullivan III
• [asterisk-users] Concurrent calls including mysql taking lot oftime for execution   Danny Nicholas
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Matt Riddell
• [asterisk-users] Concurrent calls including mysql taking lot oftime for execution   Matt Riddell
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Matt Riddell
• [asterisk-users] Incorrect voice mail format on transfer   Kyle Kienapfel
• [asterisk-users] Incorrect voice mail format on transfer   John A. Sullivan III
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Steve Edwards
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   das sandesh
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Steve Edwards
• [asterisk-users] polarity on some channels   B.Masoud at SH
• [asterisk-users] Incorrect voice mail format on transfer   Kyle Kienapfel
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Matt Riddell
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Jai Rangi
• [asterisk-users] IMAP voicemail using subfolders fails.   Darrick Hartman
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   covici at ccs.covici.com
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Matt Riddell
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Steve Edwards
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   covici at ccs.covici.com
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Jeff LaCoursiere
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Andrew Hakman
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Matt Riddell
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Paul Hales
• [asterisk-users] Asterisk 1.6.1.6 crashing -- multiple phone entries   David A. Bandel
• [asterisk-users] Concurrent calls including mysql taking lot oftime for execution   Tilghman Lesher
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Steve Edwards
• [asterisk-users] ivr menu not hanging up call   Landy Landy
• [asterisk-users] ivr menu not hanging up call   Steve Edwards
• [asterisk-users] OT - Gigaset Chagall - How to download firmware without Internet access ?   Olivier
• [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine   Olivier
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   das sandesh
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Matt Riddell
• [asterisk-users] AGI STREAM FILE and not blocking execution   Patrick
• [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine   Christophorus Laube
• [asterisk-users] VoIP interconnection with Acme packet SBC   Kasun Daminda
• [asterisk-users] polarity on some channels   Tzafrir Cohen
• [asterisk-users] Astricon   Randy R
• [asterisk-users] Astricon   Randy R
• [asterisk-users] Invite after bye?   Josip Djuricic
• [asterisk-users] OT - Gigaset Chagall - How to download firmware without Internet access ?   Kyle Kienapfel
• [asterisk-users] Audio issue in skype for asterisk   Samir Doshi
• [asterisk-users] RAMDisk vs Extarnal server for recording   Robin
• [asterisk-users] queues autopause   Rilawich Ango
• [asterisk-users] IMAP voicemail using subfolders fails.   Barry L. Kline
• [asterisk-users] ivr menu not hanging up call   Landy Landy
• [asterisk-users] AGI STREAM FILE and not blocking execution   Steve Edwards
• [asterisk-users] queues autopause   Miguel Molina
• [asterisk-users] carefulwrite: write() returned error: Broken pipe   Josip Djuricic
• [asterisk-users] carefulwrite: write() returned error: Broken pipe   Danny Nicholas
• [asterisk-users] OT - How to organize TFTP root directory ?   Olivier
• [asterisk-users] carefulwrite: write() returned error: Broken pipe   Tilghman Lesher
• [asterisk-users] ChanSpy in Asterisk 1.2.24   Joao Gomes Pereira
• [asterisk-users] ChanSpy in Asterisk 1.2.24   Danny Nicholas
• [asterisk-users] OT - How to organize TFTP root directory ?   Dave Fullerton
• [asterisk-users] Incorrect voice mail format on transfer   Robert Grignon
• [asterisk-users] Incorrect voice mail format on transfer   John A. Sullivan III
• [asterisk-users] OT - How to organize TFTP root directory ?   Jared Smith
• [asterisk-users] AGI STREAM FILE and not blocking execution   Jared Smith
• [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine   Jean-Denis Girard
• [asterisk-users] Can't configure Cisco 7942 avec factory reset   Olivier
• [asterisk-users] ChanSpy in Asterisk 1.2.24   Joao Gomes Pereira
• [asterisk-users] OT - How to organize TFTP root directory ?   Olivier
• [asterisk-users] carefulwrite: write() returned error: Brokenpipe   Danny Nicholas
• [asterisk-users] queues autopause   Rilawich Ango
• [asterisk-users] Incorrect voice mail format on transfer   Robert Grignon
• [asterisk-users] IAX Hardphones.   Albert Culleton
• [asterisk-users] IAX Hardphones.   Jorge Gutiérrez
• [asterisk-users] GSM 6.10 codec for Asterisk   Alejandro Cabrera Obed
• [asterisk-users] carefulwrite: write() returned error: Brokenpipe   Tilghman Lesher
• [asterisk-users] (SOLVED) Kernel panic w/ DAHDI 2.x/Digium TE220B   Chris Brentano
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   jonas kellens
• [asterisk-users] hangup from which side   B.Masoud at SH
• [asterisk-users] GSM 6.10 codec for Asterisk   Gordon Henderson
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Steve Edwards
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Danny Nicholas
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Steve Edwards
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Danny Nicholas
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Steve Edwards
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Danny Nicholas
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Steve Edwards
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Danny Nicholas
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Steve Edwards
• [asterisk-users] (SOLVED) Kernel panic w/ DAHDI 2.x/Digium TE220B   John Knight
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Tzafrir Cohen
• [asterisk-users] OT - How to organize TFTP root directory ?   Benny Amorsen
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Steve Edwards
• [asterisk-users] OT - How to organize TFTP root directory ?   Steve Edwards
• [asterisk-users] AstriCon videos: a question of method   John Todd
• [asterisk-users] AstriCon videos: a question of method   Robin
• [asterisk-users] Poor VoIP voice quality in one direction from three providers   Robert L Mathews
• [asterisk-users] AstriCon videos: a question of method   Ron Arts
• [asterisk-users] AstriCon videos: a question of method   Kyle Kienapfel
• [asterisk-users] OSLEC with DAHDI and Linksys/Sipura   Joseph
• [asterisk-users] OSLEC with DAHDI and Linksys/Sipura   Tzafrir Cohen
• [asterisk-users] AstriCon videos: a question of method   Ron Arts
• [asterisk-users] Poor VoIP voice quality in one direction from three providers   John A. Sullivan III
• [asterisk-users] Asterisk MOH playing old audio for first 30 to 60 seconds   OrangeCell Center Inc.
• [asterisk-users] ChanSpy in Asterisk 1.2.24   Joao Gomes Pereira
• [asterisk-users] OSLEC with DAHDI and Linksys/Sipura   Joseph
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Benaiad
• [asterisk-users] Asterisk MOH playing old audio for first 30 to 60 seconds   Tilghman Lesher
• [asterisk-users] AGI STREAM FILE and not blocking execution   Patrick
• [asterisk-users] Linksys 962   Michel Verbraak
• [asterisk-users] IAX Hardphones.   Andrew Higgs
• [asterisk-users] AstriCon videos: a question of method (Robin)   Michael Collins
• [asterisk-users] interfacing asterisk with a legacy PBX   PATRICK KANGETHE
• [asterisk-users] OT - DECT SIP Phones   --[ UxBoD ]--
• [asterisk-users] OT - DECT SIP Phones   Alan Lord (News)
• [asterisk-users] OT - DECT SIP Phones   --[ UxBoD ]--
• [asterisk-users] AstriCon videos: a question of method   Randy R
• [asterisk-users] AstriCon videos: a question of method (Robin)   Robin
• [asterisk-users] carefulwrite: write() returned error:Brokenpipe   Josip Djuricic
• [asterisk-users] OT - How to organize TFTP root directory ?   Benny Amorsen
• [asterisk-users] OSLEC with DAHDI and Linksys/Sipura   Tzafrir Cohen
• [asterisk-users] asterisk crashes when calling gtalk user   Giorgio Incantalupo
• [asterisk-users] How to generate 183 Session Progress   Marc Leurent
• [asterisk-users] Recording management for IVR   Mail list
• [asterisk-users] Recording management for IVR   ABBAS SHAKEEL
• [asterisk-users] OT - How to organize TFTP root directory ?   Steve Edwards
• [asterisk-users] interfacing asterisk with a legacy PBX   Lyle Giese
• [asterisk-users] How to generate 183 Session Progress   Klaus Darilion
• [asterisk-users] hangup from which side   Klaus Darilion
• [asterisk-users] hangup from which side   Robert Grignon
• [asterisk-users] GUI for asterix management   giancarlo lombardo
• [asterisk-users] Cisco 7921   Torintino T
• [asterisk-users] Cisco 7921   Michiel van Baak
• [asterisk-users] Strange IAX2 / Iaxmodem problem   Alexandru Oniciuc
• [asterisk-users] Strange IAX2 / Iaxmodem problem   Doug Lytle
• [asterisk-users] AstriCon videos: a question of method   Eric Chamberlain
• [asterisk-users] AstriCon videos: a question of method (Robin)   covici at ccs.covici.com
• [asterisk-users] Asterisk SIP to Cisco IAD2430 Series?   JR Richardson
• [asterisk-users] hangup from which side   Martin
• [asterisk-users] How to generate 183 Session Progress   Martin
• [asterisk-users] OSLEC with DAHDI and Linksys/Sipura   Martin
• [asterisk-users] Crash with app_mixmonitor   Darrin Henshaw
• [asterisk-users] SIREN14 call setup and record/playback   Tom Browning
• [asterisk-users] how to announce the agent answering in a queue to the caller   nik600
• [asterisk-users] SIREN14 call setup and record/playback   Peder
• [asterisk-users] SIREN14 call setup and record/playback   Kevin P. Fleming
• [asterisk-users] IVR reports?   Magnus Kelly
• [asterisk-users] IVR reports?   Carlos Chavez
• [asterisk-users] IVR reports?   Magnus Kelly
• [asterisk-users] OT - How to organize TFTP root directory ?   Warren Selby
• [asterisk-users] GUI for asterix management   Warren Selby
• [asterisk-users] interfacing asterisk with a legacy PBX   Paul Hales
• [asterisk-users] interfacing asterisk with a legacy PBX   Kyle Kienapfel
• [asterisk-users] interfacing asterisk with a legacy PBX   Steve Edwards
• [asterisk-users] interfacing asterisk with a legacy PBX   Torintino T
• [asterisk-users] GUI for asterix management   giancarlo lombardo
• [asterisk-users] AMI script..   Ritesh A
• [asterisk-users] GUI for asterix management   Steve Howes
• [asterisk-users] Manage users & administrator for the asterisk-gui   jonas kellens
• [asterisk-users] GUI for asterix management   Alan Lord (News)
• [asterisk-users] IAX Hardphones.   M Shokuie
• [asterisk-users] OT - How to organize TFTP root directory ?   Olivier
• [asterisk-users] Tim Panton Astricon Presentation "recreated"   Randy R
• [asterisk-users] SVN newbie - No trunk/ in http://svn.asterisk.org/svn/libpri/   Olivier
• [asterisk-users] OT - mISDN and B410P questions   Olivier
• [asterisk-users] OT - How to organize TFTP root directory ?   Dave Platt
• [asterisk-users] OT - How to organize TFTP root directory ?   Olivier
• [asterisk-users] SVN newbie - No trunk/ in http://svn.asterisk.org/svn/libpri/   Kyle Kienapfel
• [asterisk-users] OT - How to organize TFTP root directory ?   Steve Edwards
• [asterisk-users] OT - mISDN and B410P questions   Darrick Hartman
• [asterisk-users] GUI for asterix management   giancarlo lombardo
• [asterisk-users] OT - mISDN and B410P questions   Jean-Denis Girard
• [asterisk-users] OT - mISDN and B410P questions   Paul Hales
• [asterisk-users] test   Matt
• [asterisk-users] OSLEC with DAHDI and Linksys/Sipura   Tzafrir Cohen
• [asterisk-users] test   Alex Balashov
• [asterisk-users] SVN newbie - No trunk/ in http://svn.asterisk.org/svn/libpri/   Tzafrir Cohen
• [asterisk-users] OT - mISDN and B410P questions   Tzafrir Cohen
• [asterisk-users] OT - How to organize TFTP root directory ?   Tzafrir Cohen
• [asterisk-users] OT - mISDN and B410P questions   Jean-Denis Girard
• [asterisk-users] test   Andrew Furey
• [asterisk-users] help sip show on CLI : no such command   jonas kellens
• [asterisk-users] help sip show on CLI : no such command   Jonathan Moore
• [asterisk-users] help sip show on CLI : no such command   Tzafrir Cohen
• [asterisk-users] SIP interconnection problem   Robert Bielik
• [asterisk-users] Asterisk-stat! - help needed (once again due to mailserver problem)   Lukasz Pakula
• [asterisk-users] Queue Transfers   Sriram
• [asterisk-users] FW: Queue Transfers   Sriram
• [asterisk-users] SIP interconnection problem   Tarek Sawah
• [asterisk-users] help sip show on CLI : no such command   jonas kellens
• [asterisk-users] OSLEC with DAHDI and Linksys/Sipura   Joseph
• [asterisk-users] Asterisk as the recording server for Avaya Definity   Research
• [asterisk-users] chan_echolink   Matt
• [asterisk-users] Asterisk as the recording server for Avaya Definity   Doug Lytle
• [asterisk-users] some issue with libpri cant go past 1.4.1   Jerry Geis
• [asterisk-users] Pharmacy Online Sale 89% OFF!   VIAGRA Inc.
• [asterisk-users] OT - mISDN and B410P questions   Olivier
• [asterisk-users] OT - mISDN and B410P questions   Olivier
• [asterisk-users] SVN newbie - No trunk/ in http://svn.asterisk.org/svn/libpri/   Olivier
• [asterisk-users] How to generate 183 Session Progress   Marc Leurent
• [asterisk-users] SIP interconnection problem   Robert Bielik
• [asterisk-users] Common Community for exchange the routes via Asterisk boxes   bilal ghayyad
• [asterisk-users] Common Community for exchange the routes via Asterisk boxes   Alex Balashov
• [asterisk-users] SIP interconnection problem   Robert Bielik
• [asterisk-users] No tone, one way communcation.   PATRICK KANGETHE
• [asterisk-users] Call Record.   rajeev at sgcricket.com
• [asterisk-users] No tone, one way communcation.   Tzafrir Cohen
• [asterisk-users] IAX jitterbufer oddity   Steve Davies
• [asterisk-users] hangup from which side   Danny Nicholas
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   das sandesh
• [asterisk-users] OT - mISDN and B410P questions   Jean-Denis Girard
• [asterisk-users] Cancel attended transfer   Miguel Molina
• [asterisk-users] state_interface backport issue   Robert Broyles
• [asterisk-users] Answer call from another device   Elliot Murdock
• [asterisk-users] Answer call from another device   Danny Nicholas
• [asterisk-users] DAHDI not detecting RINGING Status on the Channel   Mariano Lecuona
• [asterisk-users] Cancel attended transfer   Danny Nicholas
• [asterisk-users] What is the best way to configure this?   Robert Augustyn
• [asterisk-users] DAHDI not detecting RINGING Status on the Channel   Danny Nicholas
• [asterisk-users] What is the best way to configure this?   Danny Nicholas
• [asterisk-users] Answer call from another device   John A. Sullivan III
• [asterisk-users] AST-2009-007: ACL not respected on SIP INVITE   Asterisk Security Team
• [asterisk-users] What is the best way to configure this?   Robert Augustyn
• [asterisk-users] What is the best way to configure this?   Danny Nicholas
• [asterisk-users] What is the best way to configure this?   Danny Nicholas
• [asterisk-users] Asterisk 1.6.1.8 Now Available   Asterisk Development Team
• [asterisk-users] DAHDI not detecting RINGING Status on the Channel   Mariano Lecuona
• [asterisk-users] No tone, one way communcation.   PATRICK KANGETHE
• [asterisk-users] No tone, one way communcation.   Tzafrir Cohen
• [asterisk-users] Installing Asterisk   asterisk at opensourcesolution.in
• [asterisk-users] DAHDI not detecting RINGING Status on the Channel   Tzafrir Cohen
• [asterisk-users] installing   asterisk at opensourcesolution.in
• [asterisk-users] Installing Asterisk   Dan Journo
• [asterisk-users] installing   Steve Howes
• [asterisk-users] installing   Alex Balashov
• [asterisk-users] [OT] Snom M9   --[ UxBoD ]--
• [asterisk-users] RTP timestamps   Liivo Vöörmann
• [asterisk-users] Installing Asterisk   PATRICK KANGETHE
• [asterisk-users] hangup from which side   Martin
• [asterisk-users] Fax with a AEx410P and Beronet BN4S0 => Sending Problem   Vincent Renaville
• [asterisk-users] hangup from which side   Danny Nicholas
• [asterisk-users] RTP timestamps   Alex Balashov
• [asterisk-users] DAHDI not detecting RINGING Status on the Channel   Mariano Lecuona
• [asterisk-users] installing   Pascal Bruno
• [asterisk-users] Installing Asterisk   John Novack
• [asterisk-users] pri intense debug span 1   Jerry Geis
• [asterisk-users] pri intense debug span 1   Danny Nicholas
• [asterisk-users] How to dial multiple extensions at once like in a ring group and put them in conference?   Zeeshan Zakaria
• [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?   Danny Nicholas
• [asterisk-users] SIP interconnection problem   Robert Bielik
• [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?   Zeeshan Zakaria
• [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?   Grygoriy Dobrovolskyy
• [asterisk-users] No tone, one way communcation.   Jorge Gutiérrez
• [asterisk-users] How to dial multiple extensions at once likein aring group and put them in conference?   Danny Nicholas
• [asterisk-users] RTP timestamps   Liivo Vöörmann
• [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?   Miguel Molina
• [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?   Zeeshan Zakaria
• [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?   Miguel Molina
• [asterisk-users] SIP interconnection problem   Robert Bielik
• [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?   Danny Nicholas
• [asterisk-users] SIP interconnection problem   Danny Nicholas
• [asterisk-users] Asterisk as the recording server for Avaya Definity   Muro, Sam
• [asterisk-users] Codecs with MixMonitor (,a) option   Miguel Molina
• [asterisk-users] The Mobile devices are not able to register on my asterisk   bilal ghayyad
• [asterisk-users] The Mobile devices are not able to register on myasterisk   Danny Nicholas
• [asterisk-users] Software for PC-PC voice comunication   giancarlo lombardo
• [asterisk-users] Software for PC-PC voice comunication   Danny Nicholas
• [asterisk-users] Installing Asterisk   Hans Witvliet
• [asterisk-users] Software for PC-PC voice comunication   Steve Edwards
• [asterisk-users] need to find firmware for cisco ata-188   Erick Perez
• [asterisk-users] need to find firmware for cisco ata-188   Steve Howes
• [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?   Zeeshan Zakaria
• [asterisk-users] Confusion on caller-ID with SIP provider   Richard Kenner
• [asterisk-users] Confusion on caller-ID with SIP provider   Alex Balashov
• [asterisk-users] Confusion on caller-ID with SIP provider   Richard Kenner
• [asterisk-users] How to dial multiple extensions at once likein aring group and put them in conference?   Matt Riddell
• [asterisk-users] Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)   Phibee Network Operation Center
• [asterisk-users] Dialing out a T1   trebaum
• [asterisk-users] Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)   Phibee Network Operation Center
• [asterisk-users] Dialing out a T1   trebaum
• [asterisk-users] SIP client MAC address.   DHAVAL INDRODIYA
• [asterisk-users] SIP client MAC address.   Klaverstyn, David C
• [asterisk-users] SIP client MAC address.   DHAVAL INDRODIYA
• [asterisk-users] SIP client MAC address.   Klaverstyn, David C
• [asterisk-users] SIP client MAC address.   DHAVAL INDRODIYA
• [asterisk-users] RTP timestamps   Liivo Vöörmann
• [asterisk-users] chan_echolink   Michael Maxwell
• [asterisk-users] Asterisk Server with Panasonic PBX   ABBAS SHAKEEL
• [asterisk-users] Syncronizing files on different Asterisk servers   ABBAS SHAKEEL
• [asterisk-users] Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)   Alex Balashov
• [asterisk-users] SIP client MAC address.   Alex Balashov
• [asterisk-users] Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)   Alex Balashov
• [asterisk-users] The SIP in the Mobile Phones are not able to register on asterisk   bilal ghayyad
• [asterisk-users] How to dial multiple extensions at once likein aring group and put them in conference?   Zeeshan Zakaria
• [asterisk-users] The SIP in the Mobile Phones are not able to register on asterisk   Xavier Mesquida
• [asterisk-users] SIP Peers still ping with SIP OPTIONS on a reload   Marc Leurent
• [asterisk-users] CDR(billsec)   Anahi Ludueña
• [asterisk-users] sip fullcontact and port values   Ishfaq Malik
• [asterisk-users] Software for PC-PC voice comunication   Ramu
• [asterisk-users] Ravindra K (ravibth at gmail.com) has sent you a private message   Ravindra K
• [asterisk-users] Ravindra K (ravibth at gmail.com) has sent you a private message   Alex Balashov
• [asterisk-users] CDR(billsec)   Danny Nicholas
• [asterisk-users] Software for PC-PC voice comunication   Zoaaaaa
• [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?   Danny Nicholas
• [asterisk-users] need to find firmware for cisco ata-188   Erick Perez
• [asterisk-users] Software for PC-PC voice comunication   Doddle WebPhone
• [asterisk-users] R: CDR(billsec)   Alexandru Oniciuc
• [asterisk-users] deploying asterisk   asterisk at opensourcesolution.in
• [asterisk-users] deploying asterisk   Danny Nicholas
• [asterisk-users] Clear pending SIP channels   Aggio Alberto
• [asterisk-users] Clear pending SIP channels   Jose P. Espinal
• [asterisk-users] need a local tech   Ott Rose
• [asterisk-users] need a local tech   Danny Nicholas
• [asterisk-users] R: CDR(billsec)   Danny Nicholas
• [asterisk-users] deploying asterisk   Darrick Hartman
• [asterisk-users] deploying asterisk   Randy R
• [asterisk-users] deploying asterisk   Steve Edwards
• [asterisk-users] R: CDR(billsec)   Ishfaq Malik
• [asterisk-users] R: CDR(billsec)   Danny Nicholas
• [asterisk-users] R: R: CDR(billsec)   Alexandru Oniciuc
• [asterisk-users] Software for PC-PC voice comunication   giancarlo lombardo
• [asterisk-users] deploying asterisk   Alex Balashov
• [asterisk-users] Having a heck of a time   Robert Grignon
• [asterisk-users] SIP 18x Messages   Tim King
• [asterisk-users] Having a heck of a time   BJ Weschke
• [asterisk-users] Having a heck of a time   Tim King
• [asterisk-users] Having a heck of a time   Robert Grignon
• [asterisk-users] Having a heck of a time   Robert Grignon
• [asterisk-users] Having a heck of a time   Robert Grignon
• [asterisk-users] SIP 18x Messages   Kevin P. Fleming
• [asterisk-users] SIP 18x Messages   Tim King
• [asterisk-users] MOH   Peder
• [asterisk-users] MOH   Kevin P. Fleming
• [asterisk-users] need a local tech   Hans Witvliet
• [asterisk-users] Asterisk/Cisco AS5300 => Two problems in   Neeraj Chand
• [asterisk-users] need a local tech   Tzafrir Cohen
• [asterisk-users] how to announce the agent answering in a queue to the caller   nik600
• [asterisk-users] Asterisk 302 Moved Temporarily   "Juan E. Rodríguez"
• [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID   Carlos Chavez
• [asterisk-users] Asterisk Server with Panasonic PBX   C F
• [asterisk-users] GUI for hunt groups?   Ken D'Ambrosio
• [asterisk-users] GUI for hunt groups?   Duncan Turnbull
• [asterisk-users] Dynamic DNS trunk   B.Masoud at SH
• [asterisk-users] Asterisk Server with Panasonic PBX   ABBAS SHAKEEL
• [asterisk-users] Dynamic DNS trunk   "Juan E. Rodríguez"
• [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID   Tilghman Lesher
• [asterisk-users] Async Agi problem   Robert Bielik
• [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?   DHAVAL INDRODIYA
• [asterisk-users] R: CDR(billsec)   Ishfaq Malik
• [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?   Zeeshan Zakaria
• [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?   DHAVAL INDRODIYA
• [asterisk-users] Zap inbound hangup problem   Robin
• [asterisk-users] Zap inbound hangup problem   Tzafrir Cohen
• [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?   Matt Riddell
• [asterisk-users] R: R: CDR(billsec)   Matt Riddell
• [asterisk-users] Dynamic DNS trunk   Matt Riddell
• [asterisk-users] Zap inbound hangup problem   Robin
• [asterisk-users] R: R: R: CDR(billsec)   Alexandru Oniciuc
• [asterisk-users] R: R: R: CDR(billsec)   Matt Riddell
• [asterisk-users] R: R: R: CDR(billsec)   Anahi Ludueña
• [asterisk-users] R: R: R: CDR(billsec)   Danny Nicholas
• [asterisk-users] R: R: R: CDR(billsec)   Ishfaq Malik
• [asterisk-users] R: R: R: CDR(billsec)   Danny Nicholas
• [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?   John Novack
• [asterisk-users] R: R: R: R: CDR(billsec)   Alexandru Oniciuc
• [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?   Danny Nicholas
• [asterisk-users] R: R: R: R: R: CDR(billsec)   Alexandru Oniciuc
• [asterisk-users] GUI for hunt groups?   Robert Grignon
• [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID   Carlos Chavez
• [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID   Tilghman Lesher
• [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID   Carlos Chavez
• [asterisk-users] Dynamic DNS trunk   B.Masoud at SH
• [asterisk-users] need a local tech   Ott Rose
• [asterisk-users] need a local tech   Ott Rose
• [asterisk-users] Async Agi problem   Moises Silva
• [asterisk-users] Delayed answer when calling out   Robert Grignon
• [asterisk-users] Delayed answer when calling out   Danny Nicholas
• [asterisk-users] Delayed answer when calling out   Robert Grignon
• [asterisk-users] Delayed answer when calling out   Danny Nicholas
• [asterisk-users] !! Unknown IE 50 (cs5, Unknown Information Element) on console.   Alec Davis
• [asterisk-users] Booting Error for /dev/kmem   Torintino T
• [asterisk-users] Booting Error for /dev/kmem   Danny Nicholas
• [asterisk-users] AsteriskForge Now Open   Steve Sokol
• [asterisk-users] Booting Error for /dev/kmem   Steve Edwards
• [asterisk-users] Astreicon presentations   Neeraj Chand
• [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?   Paul Hales
• [asterisk-users] need a local tech   Don Kelly
• [asterisk-users] Asterisk Server with Panasonic PBX   C F
• [asterisk-users] Dynamic DNS trunk   Matt Riddell
• [asterisk-users] Unable to set TOS to 184?   Bart Fisher
• [asterisk-users] Unable to set TOS to 184?   "Juan E. Rodríguez"
• [asterisk-users] Dynamic DNS trunk   B.Masoud at SH
• [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?   Zeeshan Zakaria
• [asterisk-users] Unable to set TOS to 184?   John A. Sullivan III
• [asterisk-users] asterisk 1.6 enable cdr_mysql   Joseph
• [asterisk-users] asterisk 1.6 enable cdr_mysql   Prince Singh
• [asterisk-users] Async Agi problem   Robert Bielik
• [asterisk-users] R: How to dial multiple extensions at once likeinaring group and put them in conference?   Alexandru Oniciuc
• [asterisk-users] Unable to set TOS to 184?   Karsten Wemheuer
• [asterisk-users] Asterisk Server with Panasonic PBX   ABBAS SHAKEEL
• [asterisk-users] Unable to set TOS to 184?   John A. Sullivan III
• [asterisk-users] inbound routes   PATRICK KANGETHE
• [asterisk-users] inbound routes   Steve Howes
• [asterisk-users] DAHDI/ZAP overlap dialing   Vieri
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Vincent
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Alex Balashov
• [asterisk-users] Linux/Asterisk on game consoles?   Vincent
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Michelle Dupuis
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Alex Balashov
• [asterisk-users] DAHDI/ZAP overlap dialing   Martin
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Michelle Dupuis
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Alex Balashov
• [asterisk-users] Queue device state problem   Alexandre Rodrigues
• [asterisk-users] DAHDI/ZAP overlap dialing   Vieri
• [asterisk-users] [IAX] Recommended soft- and hardphones?   John Todd
• [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID   Tilghman Lesher
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Michael Graves
• [asterisk-users] Queue device state problem   Danny Nicholas
• [asterisk-users] Astreicon presentations   Tilghman Lesher
• [asterisk-users] asterisk 1.6 enable cdr_mysql   Tilghman Lesher
• [asterisk-users] DAHDI/ZAP overlap dialing   Martin
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Gordon Henderson
• [asterisk-users] Cancel attended transfer   Miguel Molina
• [asterisk-users] Async Agi problem   Moises Silva
• [asterisk-users] Voicemail file   Anahi Ludueña
• [asterisk-users] DAHDI/ZAP overlap dialing   Vieri
• [asterisk-users] Voicemail file   Danny Nicholas
• [asterisk-users] Voicemail file   Anahi Ludueña
• [asterisk-users] Voicemail file   Steve Howes
• [asterisk-users] Voicemail file   Danny Nicholas
• [asterisk-users] Voicemail file   Anahi Ludueña
• [asterisk-users] Voicemail file   Danny Nicholas
• [asterisk-users] Voicemail file   Danny Nicholas
• [asterisk-users] Voicemail file   Ishfaq Malik
• [asterisk-users] asterisk 1.6 enable cdr_mysql   Joseph
• [asterisk-users] Voicemail file   Danny Nicholas
• [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6   Mariano Lecuona
• [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID   Carlos Chavez
• [asterisk-users] Voicemail file   Jim Dickenson
• [asterisk-users] SNOM 870   --[ UxBoD ]--
• [asterisk-users] Real replacement for AgentCallBackLogin() onAsterisk 1.6   Danny Nicholas
• [asterisk-users] Voicemail file   Danny Nicholas
• [asterisk-users] Cannot make calls   Cliconnect
• [asterisk-users] asterisk 1.6 - doing dnsmgr lookup for... / call fails   Joseph
• [asterisk-users] asterisk 1.6 - doing dnsmgr lookup for... / call fails   Joseph
• [asterisk-users] Real replacement for AgentCallBackLogin() onAsterisk 1.6   Warren Selby
• [asterisk-users] Cannot make calls   Warren Selby
• [asterisk-users] Cannot make calls   Cliconnect
• [asterisk-users] voicmail: no entry in voicemail config   Joseph
• [asterisk-users] voicmail: no entry in voicemail config   Danny Nicholas
• [asterisk-users] [SOLVED] voicmail: no entry in voicemail config   Joseph
• [asterisk-users] voicmail: no entry in voicemail config   Jared Smith
• [asterisk-users] voicmail: no entry in voicemail config   Joseph
• [asterisk-users] voicmail: no entry in voicemail config   Joseph
• [asterisk-users] voicmail: no entry in voicemail config   Danny Nicholas
• [asterisk-users] Cannot make calls   Warren Selby
• [asterisk-users] Cannot make calls   Cliconnect
• [asterisk-users] Cannot make calls   Ott Rose
• [asterisk-users] SNOM 870   hbk
• [asterisk-users] AEX800P on HP Prolaint ML115 kernel panic   David Shauger
• [asterisk-users] AEX800P on HP Prolaint ML115 kernel panic   Mariano Lecuona
• [asterisk-users] DAHDI/ZAP overlap dialing   Martin
• [asterisk-users] Asterisk-1.6.1.8 DTMF with SIP is not working   Joseph
• [asterisk-users] strange dialing HELP !   B.Masoud at SH
• [asterisk-users] Asterisk-1.6.1.8 DTMF with SIP is not working   Joseph
• [asterisk-users] some issue with libpri cant go past 1.4.1   Jerry Geis
• [asterisk-users] Asterisk-1.6.1.8 DTMF with SIP is not working   Moises Silva
• [asterisk-users] Asterisk-1.6.1.8 DTMF with SIP is not working   Joseph
• [asterisk-users] Local channel that runs a custom app... why immediate hangup?   eric weaver
• [asterisk-users] Asterisk-1.6.1.8 DTMF with SIP is not working   Ivan Stepaniuk
• [asterisk-users] DAHDI/ZAP overlap dialing   Tzafrir Cohen
• [asterisk-users] OT - Number Portability   Jeff LaCoursiere
• [asterisk-users] DAHDI/ZAP overlap dialing   Martin
• [asterisk-users] OT - Number Portability   Cary Fitch
• [asterisk-users] OT - Number Portability   Cary Fitch
• [asterisk-users] AEX800P on HP Prolaint ML115 kernel panic   David Shauger
• [asterisk-users] OT - Number Portability   Jeff LaCoursiere
• [asterisk-users] Disconnecting during the call, analog lines   bilal ghayyad
• [asterisk-users] Calls disconnects after short time   B.Masoud at SH
• [asterisk-users] !<command> from Manager   cbulist
• [asterisk-users] Asterisk-1.6.1.8 DTMF with SIP is not working   Joseph
• [asterisk-users] PRI line resetting on incoming call   Samuel Nair
• [asterisk-users] Asterisk, Realtime and specify MySQL Table Name ?   Phibee Network Operation Center
• [asterisk-users] Long pause during dialing to IVR   Joseph
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Joseph
• [asterisk-users] AEX800P on HP Prolaint ML115 kernel panic   David Shauger
• [asterisk-users] Disconnecting during the call, analog lines   Joseph
• [asterisk-users] Calls disconnects after short time   B.Masoud at SH
• [asterisk-users] Long pause during dialing to IVR   Joseph
• [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID   Joseph
• [asterisk-users] Determining extension's sip.conf default mailbox   Steve Johnson
• [asterisk-users] !<command> from Manager   Tzafrir Cohen
• [asterisk-users] Calls disconnects after short time   B.Masoud at SH
• [asterisk-users] Determining SIP peer's default mailbox   Philipp Kempgen

Last message date: Sat Oct 31 20:09:22 CDT 2009
Archived on: Sat Oct 31 20:09:29 CDT 2009

• Messages sorted by: [ thread ] [ subject ] [ author ]
• More info on this list...