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- asterisk app store,
Dean Collins
- Question about Asterisk and Java,
Santiago Panchi
- Cisco 7911g,
Sean Lowry
- OT: real 2 line phone vs. 1 line and call waiting,
Brian J. Murrell
- problem with my softphone,
Abel Monzon
- Maybe OT - routing calls in PSTN,
Bill Michaelson
- Channel variables "materializing" ...,
Julian Lyndon-Smith
- Cheap FXO Card?,
Andrew Joakimsen
- How can Block a pri channel,
Alireza Shokoienia
- Source of SIP "Remote host can't match request NOTIFY",
Olivier
- Zaptel Lines - How many are in use..,
Singer Wang
- Re: CLI and verbosity level [SOLVED],
Olivier
- Re: Problem with pickup extension *8 from features.conf using IAX,
Serghei Gutanu
- CLI and verbosity level,
Olivier
- SIP/IAX Interworking ip CANCEL behavior,
Andreas M.
- OT - Avantages of ISDN PtP and PtmP,
Olivier
- Disable CDR?,
Vincent
- Re: Knowing incoming call technology and channel [SOLVED],
Olivier
- Knowing incoming call technology and channel,
Olivier
- identify/find a channel to pick it up,
Stefan Schmidt
- ATA for large networks,
Vieri
- Creating Asterisk Binary Package,
Jim Boykin
- AGI defunct processes + GSM Playback - HELP!,
Hakan C
- uk tole-free dids?,
Babcock, Michael Alex
- Conferencing Hardware,
Jim Boykin
- Vividial issue,
Brad
- Eye P Media Soft Phone?,
Michael Graves
- G.722 between Eyebeam and a Polycom IP650,
Michael Graves
- credit card processing,
Ruddy Gbaguidi
- Keeps Ringing After Answer,
broadband Voice
- Troubleshooting one-way voice... how to peek into SIP RTP?,
Philip Prindeville
- Re: PRI TE110P Configuration (Solved),
Shyju K
- rtpkeepalive problem ?,
tic tac
- FW: Google Alert - "dean collins",
Dean Collins
- Asterisk and VoIP educational resources,
randulo
- Philippines,
Dean Collins
- Set A-Number in Sip Header,
Michael Litzel
- running out of disk space,
Mr surfit
- Problem with pickup extension *8 from features.conf using IAX,
coco
- test call generator,
Sam Tam
- Split incoming call volume across queues on several asterisk servers,
Haider Raza
- iPhone Sip App,
Forrest Beck
- Audio Files,
Abel Monzon
- Extremely OT: I need someone who can parse a MS Word or PDF or RTF document,
randulo
- Voicemail retention,
Asterisk User List
- Bizarre international call problem.,
Ken D'Ambrosio
- Dial issue,
equis software
- server and 2 uniden phones no ringing,
Jerry Geis
- Friday 2008-09-26 12:00:00 Asterisk + Skype on your box,
randulo
- Incoming URL handling Problem (Asterisk problem ?),
Fabio Mosti
- Get Call Length of Calls,
Ali Jawad
- setting DNID,
Giorgio Incantalupo
- Push presence from one asterisk to another,
Loic Didelot
- T38 fax gateway announcement,
Daniel Ferenci
- Skype channel beta,
randulo
- ZAP not answering call,
Daniel Johnson
- Monitoring simul calls,
carl Lougher
- Music on hold for sub tenants,
carl Lougher
- Sip reload casuing issues,
carl Lougher
- Mysql Command and number rows returned,
David Murphy
- users.conf behavior,
Dave Poirier
- Skype + Asterisk Interview at Astricon,
Ming Yong
- Create virtual extension,
Manolet Gmail
- sip forking needed for ekiga 3.0,
Brian J. Murrell
- PRI TE110P Configuration,
Shyju K
- Skype-asterisk connection announced (was Astricon people please post the announcement),
randulo
- Astricon people please post the announcement,
randulo
- Dial Plan Issues,
Tariq ..
- Monitoring trunk,
Rodrigo Pinto
- OT: Do You Know What the Problem With CDMA is?,
Dean Collins
- Ringing after console dsp hangup,
Jerry Geis
- SIP TLS,
Rafael Puga
- IMAP voicemail import,
Andreas Anderson
- appconference low quality g729,
Pezhman Lali
- Current available allarms in the Asterisk,
bilal ghayyad
- Terrible Experience Net2phone A-Z termination,
broadband Voice
- Problem making international calls,
f.beijer
- What happened to the register= setting in sip.conf?,
David Kerr
- g729 capacity,
Robert McNaught
- Asterisk 1.4 is asking me for Mailbox #,
Joseph
- Asterisk on VMware Workstation 6,
Michael J. Liberatore
- Zaptel/DAHDI ztdummy only,
Roderick A. Anderson
- Astricon 08 Videos & interviews & Voiceroute twitters on astricon,
Ming Yong
- Timeout question,
Vadim Lebedev
- DID mode,
michel freiha
- Asterisk is covering the peers IP address in SIP and SDP messages,
Arno Scholz
- Voicemail cutting out after about 30 seconds,
Chris Bagnall
- IAX Hangup floods link with repeated VNAK and HANGUP,
Nathan Dennis
- asterisk console: "quit" is twice in history,
Klaus Darilion
- Asterisk mysql CDR,
Nhadie
- How to send indicating call privacy using P-Asserted-Identity?,
Zeeshan Zakaria
- A2Billing Callback Hangup after/about 20 sec!,
Muhammad Zulqarnain
- "No route to destination" error,
Martin Seebach
- Asterisk 1.4 or 1.6,
Joseph
- extension definition,
michel freiha
- Short question: CPU hardware requirements for Asterisk,
Alejandro Cabrera Obed
- Connecting TE212p to NEC XenMaster,
Mikel Lindsaar
- Linksys 3102 with rfc2833 - NOT WORKING,
Joseph
- Fwd: more on Free World Dialup groups and FWDLive,
Steve Totaro
- PRI incoming call forward / call redirect,
Andrew Kohlsmith (lists)
- Extension registration,
michel freiha
- chan_misdn troubles,
Thanos Koukoulis
- AGI and prepaid billing,
bilal ghayyad
- Registration by IP address,
michel freiha
- t38modem on OpenSuse,
Marco Signorini
- Transcoding G.729 files,
Thomas Kenyon
- [1.4.21.2] Checking that already off-hook?,
Vincent
- PSTN Simulator,
mark morreny
- Send us your suggestions on exhibits & tutorials to cover (video) at Voiceroute,
Ming Yong
- How to hangup a channel immediately so that it doesn't get charged on cell phone,
Zeeshan Zakaria
- ast_func_write: Function not registered,
Anis Maatoug
- E&M wink/no audio,
Bill Michaelson
- I can't call my remote users?,
Steve Anness
- GotoIfTime and timezone specification,
Klaus Darilion
- Problem using AJAM on asterisk 1.4.17,
Jason Martin
- setvar for outgoing SIP channels?,
Klaus Darilion
- Astricon news online?,
randulo
- Seemingly easy question: NPA/NXX,
Karl Fife
- How to notify an event to every user,
Olivier
- Asterisk weird behavior after upgrading,
Andrew Nowrot
- Astricon: Throw the dice, give a talk,
John Todd
- how to add extensions and sip registrations dynamically,
George Williams
- broadcast ability,
DesRae Mason
- callwaiting callerid,
robb
- Cisco acquires Jabber,
Dean Collins
- 1.6.0-rc6 - SIP hold logic broken?,
Stefan Gofferje
- [CID] Unknown IE 18/21?,
Vincent
- Specific SIP answers on incoming calls?,
Stefan Gofferje
- SVN 1.6.0 / current does not compile,
Stefan Gofferje
- Loud noise on Zap port...,
Carlos Chavez
- Last 2 days for early bird tickets to DruidCON 2008, 1-2 Oct in Atlanta GA,
Ming Yong
- getting results messages from CLI commands via -rx,
George Williams
- Dropping Phone Calls,
Todd Reese
- Weird "permissions" issue when permissions check out...,
Jason Lixfeld
- TE110P or TE120P,
Gordon Henderson
- VoIP Users Friday Conference @12 Noon EDT: Astricon run up meeting and more,
randulo
- dundi and zap devices,
Giorgio Incantalupo
- Preventing a call forward,
Julian Lyndon-Smith
- SIP request send me 482 error,
remi . druilhe
- Dialing a 60anything number issue!,
Brad
- T38 FAX over a Broadsoft,
eng. Anatoli Marinov
- PRI E1 Inbound calls hangup with busy after a few seconds,
Daniel Johnson
- T100P detection.,
Alex Balashov
- what codec is sip using?,
sean darcy
- Follow Me app question,
Mark Phillips
- Polycom phones and DNS SRV,
CunningPike
- Old voicemail bounces users,
David A. Bandel
- device probe order question,
Jason T. Nelson
- OT: Cisco 1841 - Can it be made SIP aware?,
Barton Fisher
- OT - How to stream a A-Law/wav file to a browser ?,
Olivier
- Custom Voicemail emails,
Steve Anness
- Pre-paid Billing,
Jim Boykin
- rxfax and txfax,
Rizwan Hisham
- How to make a Outgoing Call from Asterisk ?,
Hiren Mistry
- Verbosity best practice,
Olivier
- Get rid of "Really destroying SIP dialog",
Olivier
- BRI or PRI callerid,
Loic Didelot
- 482 Loop Detected,
remi . druilhe
- how to detect pickup...,
Gergo Csibra
- Speech recognition on simultaneous SIP / PSTN calls,
Allann Jones
- strategy for measuring conference audio delay,
George Williams
- Understanding of SIP Info Messages,
robb
- app_confrence with loud voices,
Pezhman Lali
- Digium training course,
Pascal Bruno
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