asterisk users - Nov 2003

Sunday November 30 2003
TimeRepliesSubject
9:15PM 2 Sound Breaks
8:17PM 3 Dial "T" option not obeyed with Grandstream BT101
4:48PM 5 cisco 7960 power suplies?
4:31PM 2 LCR with ENUM and DDNS: half the story
1:30PM 1 Rate file formats: a standard?
11:13AM 2 Cisco 6.0 + Asterisk question
5:21AM 5 app_queue behavior
 
Saturday November 29 2003
TimeRepliesSubject
11:47PM 0 IAX phones and CPU usage problem
10:28PM 44 * Party in Paris
8:52PM 1 Sip Issue
5:57PM 10 asterisk server crashing
3:11PM 5 iaxComm Update available [Ringtones, Intercom, UI improvements]
11:36AM 0 ENUM and DNS/Bind
3:26AM 0 Asterisk Run Problem
 
Friday November 28 2003
TimeRepliesSubject
11:58PM 4 Deltathree icomming problem
5:42PM 0 Can't seem to connect/call fwd network Help!
3:33PM 0 Asterisck as a Fujistu 9600 VOIP Gateway
3:13PM 0 Re: Resend: Help for oh323
12:40PM 3 Survey says post your 3.3 volt Mother boards used in PRODUCTION withTE410
11:56AM 0 QUESTION Ringing Appl.
11:24AM 0 Iax termination in India
10:38AM 0 Asterisk install / update script - need testers
10:02AM 14 3x AVM Fritz!Card PCI for a EuroISDN PBX.
8:15AM 3 H323.conf
7:59AM 3 ASTERISK WITHOUT ANY CARD
7:26AM 1 Problem with SIP-Phones and * audio-files
7:25AM 1 Request for debug message in ENUM code
6:50AM 1 channel offset between Asterisk and PBX
6:15AM 0 TEDAS VoIP DECT PABX
5:51AM 3 How does Asterisk use CPU?
4:42AM 9 Mute button in Grandstream?
1:18AM 2 MGCP Support for NAT
1:14AM 8 call waiting disable in sip
 
Thursday November 27 2003
TimeRepliesSubject
11:38PM 1 files for upgrade cisco 7960 phone
9:57PM 4 RFC3389 support incomplete
9:10PM 32 Asterisk behind NAT << How to do it.
7:19PM 1 AGI (IF/ELSE)
4:49PM 9 Mailing list archives searchable ?
4:49PM 4 RE: Grandstream BT-100 and
4:39PM 0 OT: CISCO-ATA-186
3:59PM 0 RE: Grandstream BT-100 and latest CVS
3:24PM 6 IAX2 Ethereal plugin v0.3 is out
3:19PM 3 ISDN, BRI, PRI, voice over IP, and more...
1:59PM 4 ENUM regexp replacements
12:32PM 0 Has anyone else had problems with Chagres?
12:14PM 11 Multi-line TTS Outbound Dialer
11:04AM 0 Larger SIP packets
10:04AM 0 Timeout feature in queues.conf does not seem to work
9:49AM 2 Agent Logoff inability when calls are being received from queue
9:28AM 8 Help for oh323
8:21AM 0 distinctive ring doesn't work
6:04AM 2 Crash - What is happening here???
5:36AM 0 Problem after re-compiling
2:06AM 3 App queue and all Agent busy
1:05AM 11 MGCP problem
 
Wednesday November 26 2003
TimeRepliesSubject
11:54PM 3 AGI - Freakin Lost
11:45PM 0 GrandStream BudgeTone 101 Question
11:09PM 3 Attempting to get SJPhone configured for Asterisk- Help!
10:23PM 2 unixodbc-vm-routines.h
6:02PM 0 Asterisk Training
5:22PM 4 AGI - CallerID ??
5:14PM 0 beware supposed PCI 2.2 compatibility!
2:51PM 0 SIP PBX features
2:44PM 21 door phone
2:16PM 6 Symmetric RTP
1:55PM 2 Issues with Privacy Manager and Zapateller
11:15AM 0 prob. w/ data (modem) calls through Asterisk
10:53AM 0 VoIP bandwidth management with linux & CBQ
10:52AM 0 bugs.digium.com and acknowledged status
9:01AM 0 jitter buffer for iax
8:55AM 11 Modem cards??
8:49AM 0 ADSI Programming - Found Guide on Designing Apps
8:29AM 1 perl --> manager problem
7:33AM 2 Web interface?
7:10AM 0 Needed - AGI Scripting
5:09AM 1 Pbx / channel bank install
4:37AM 3 Virtual PBX (*)
4:27AM 0 Echo and Call Setup suggestions
 
Tuesday November 25 2003
TimeRepliesSubject
6:49PM 0 Warning: File chan_sip.c, Line 462
6:01PM 6 Handytone 286 - calling out
4:51PM 4 modem to modem calls through asterisk
4:24PM 1 About sound modules in Asterisk. And call gnophone-asterisk-h323
2:23PM 6 Distinctive ring confusion
2:04PM 1 Want to get rid of Tormenta error message
1:23PM 5 How to demo * on a notebook
1:17PM 13 Crashed Asterisk
12:51PM 5 Options for 3rd party call control
12:14PM 8 AGI Rocks!! (A happy camper)
10:13AM 3 Outgoing-call and enter user in Conference - repost
9:58AM 14 PCI 3.3 V
8:37AM 1 Picking a channel (FXO port or SIP) for outb ound calls
8:30AM 1 ISDN Cards available in Australia
8:24AM 17 Prompt recording
6:42AM 3 zt_rec: Unknown error 500
4:22AM 1 Problem with fax detection
4:11AM 1 Ring requested on channel 1 already in use...
3:49AM 2 How to use * to simply skim off callerid (UK)?
2:22AM 4 * Configuration
1:28AM 18 cdr_unixodbc
12:28AM 0 what is the problem?
12:27AM 0 For all IAXTEL users of DIAX
12:05AM 8 SIMPLE support in Asterisk?
 
Monday November 24 2003
TimeRepliesSubject
10:42PM 14 g729 license
8:48PM 1 NTT FSK - Japanese Caller ID
6:07PM 1 MGCP RFC (2705) vs. PacketCable MGCP spec
1:33PM 4 One voicemail -> multiple recipients?
1:04PM 0 Pickupgroup and IAX phones
12:46PM 1 TIME ZONE support
12:28PM 5 Sip phones!
12:01PM 0 Picking an open channel (FXO port) for outbo und calls
11:06AM 6 Ring power on Analog adapters
10:45AM 22 Picking an open channel (FXO port) for outbound calls
9:42AM 3 strange SIP authentication/authorization behaviour
9:10AM 9 test call request
9:10AM 19 Cisco to asterisk termination with h323 and g729 finally works.
8:41AM 3 Pressing 0 in Voicemail causes * to hangup
8:12AM 1 Re: Asterisk-Users digest, Vol 1 #1994 - 14 msgs
7:34AM 0 SIP channel modification
6:38AM 0 SIP to SIP redirect while ringing
5:47AM 2 Netphone SIP phone
 
Sunday November 23 2003
TimeRepliesSubject
10:46PM 5 Nufone account not registering
9:14PM 1 CVS update of Asterisk - What did I do wrong?
7:43PM 1 agi exec problem (followup)
7:39PM 1 Phone compatibility list
7:17PM 4 agi exec problem.
6:47PM 4 SIP Express Router & Asterisk
6:03PM 10 Re: [Asterisk] GSM access
4:06PM 0 Call Pickup ???
1:15PM 4 Feedback with X100P and SIP fwd.pulver
8:45AM 10 RxFax
6:36AM 4 SIP Asterisk -> Nikotel disconnects after 1 Minute
 
Saturday November 22 2003
TimeRepliesSubject
11:05PM 0 Local numbers to Victorville/Apple Valley, CA
8:47PM 0 * on 2.4.20-gentoo-r8 linux with eicon-diva-pro-pci-2.0 isdn-bri card how-to ???
4:01PM 1 Help Required for Speex
3:11PM 0 Stability with the Supura SIP Units
2:20PM 2 g729 codec questions error running asterisk now
1:42PM 26 New DIAX - version 0.9.4 - a big step forward - available for download
12:51PM 5 SIP channel improvements
11:49AM 0 Asterisk - phone docs
11:39AM 0 Zap MWI method
10:05AM 1 X100P configuration Problem
9:11AM 2 How to dial out using OH323?
7:10AM 0 Experimental Switzerland -> IAX gateway
4:11AM 0 Opteron - Kernel optimizations
12:14AM 3 Newbie ... some questions
 
Friday November 21 2003
TimeRepliesSubject
10:43PM 3 Stutter dialtone but no messages
7:58PM 1 Echo Cancellation, TDMoE fails, X100P works
6:39PM 0 status of distinctive ring
5:33PM 16 MOH - Hold Button - I think I'm going crazy
4:13PM 7 PRI problems
2:55PM 7 Unable to create channel of type 'SIP'
2:12PM 1 making outside call with sip phone
1:31PM 12 Asterisk Call Manager for Windows 0.0.1 (Alpha)
11:03AM 4 Current CVS problem
11:03AM 0 One way sound
10:49AM 2 DIAX, IAX2 and latency
10:36AM 2 Can you monitor a call via the asterisk speaker system and do a call pickup if you wish
10:34AM 16 Outline For Asterisk Book - Please Review & Comment
9:23AM 0 MGCP with Cisco GW
8:40AM 1 Outgoing-call and enter user in Conference
6:54AM 1 SAY NUMBER in AGI?
6:10AM 4 Upgrade CISCO 7960 Question
5:53AM 0 Asterisk SIP implementation
4:45AM 0 Asterisk and Proxy issues
3:39AM 4 callerid problem...zaptel ppl
3:06AM 8 Is Asterisk suitable for this use?
1:48AM 6 Which ISDM BRI Card for Asterisk?
 
Thursday November 20 2003
TimeRepliesSubject
11:19PM 9 can't get caller id?
11:07PM 4 Change the all announcement
10:37PM 0 Unknown RTP codec 72 received NOTICE while using Xlite with *
9:20PM 2 ADSI Hold
8:01PM 1 Linux Voice Mail Application??
7:03PM 0 FW: Mailing list email masquerading.
6:00PM 4 T1 / Channel Bank Configuration
5:47PM 0 Mailing list email masquerading.
5:07PM 0 SIP URI... d'oh!
5:07PM 2 Dialup Internet through Asterisk server?
5:06PM 3 No ringback
5:02PM 2 Solved! Snom 200 Busy signal
4:15PM 3 SIP URIs and ENUM or other types of lookup
3:53PM 0 Asterisk Lists (was Re: Asterisk Business discussion again)
3:48PM 0 General Email and Mailing List Etiquette
3:46PM 2 Snom 200 stuck on "Busy"
2:30PM 2 new iaxComm build available
2:05PM 1 is possible isdn card in analog line?
1:59PM 0 SIP<->*/ISDN<->PSTN calls almost impossible because of echo
1:37PM 2 Voicemail just hanging up...
12:03PM 21 Tuning the Linux kernel?
10:32AM 2 Zaptel DAX?
10:05AM 0 Missing Manager Events/Actions: Hold, Reconnect, Conference
9:58AM 3 asterisk-oh323 v0.5.7 bugfix release
9:32AM 0 codec pass-through feature
8:45AM 11 TE410P ERRORS under load
8:44AM 17 tunnel iax via gnophone with ssh?
8:38AM 1 Can I soft-link a voicemailbox?
8:26AM 2 Scope of the "h" extension..
8:02AM 4 Cisco to use * as a gateway?
7:46AM 1 IAX2 Ethereal Plugin initial release
6:56AM 0 Stutter Tone all the time?
5:49AM 9 iaxComm new version installation problem
5:03AM 3 Cannot do international dial with E1 in Spain
4:13AM 4 Cisco DTMF Issue
4:05AM 5 VOIP --> PSTN via. voicemodem/soundcard.
3:12AM 7 The internet needs a dialing code..
2:49AM 0 Tdm400p FXS faults
2:08AM 4 Still TDM400P problem
 
Wednesday November 19 2003
TimeRepliesSubject
11:32PM 0 Can anyone give me an example of sip.conf and extensions.conf about asterisk SIP Proxy server?
10:26PM 8 Controlling asterisk in a dynamic way
9:26PM 7 Help configuring CISCO 7960
8:38PM 5 ATA-186 Double Digit problems
8:13PM 0 Getting in to h323
8:07PM 4 PSTN intercepted announcement
6:35PM 41 FAQ, Documentation, How-to, etc
5:03PM 39 Asterisk Business discussion again
4:23PM 1 Mediatrix 1102 / 1104 authentication problems....
3:14PM 1 fritz pci / chan_capi / australia setup
2:21PM 2 Application CallingPres
12:51PM 1 Play a "sound" after dialing a user...
12:10PM 3 RTP timing in a SIP only world (choppy MOH)
10:50AM 0 GoTo or Dial in AGI??
10:20AM 1 FXO card still won't pick up...
9:52AM 8 Help please
9:26AM 0 Question on hearing ADSI CAS tone
9:07AM 0 AMA flags by context
8:41AM 13 FYI: Simple Small Asterisk install..
6:25AM 19 Service codes for MGCP channels
5:41AM 8 inter diax connection
5:36AM 1 2 TE410P
5:31AM 0 T100P and Meetme
5:25AM 1 VOIP onver the net
5:10AM 1 Message from * console.
4:51AM 16 4 Port FXO cards
4:18AM 0 SIP/IAX2 DTMF
4:07AM 2 creative VoIP blaster & *
3:55AM 4 E100P driver overwrites memory used bye linux-kernel
2:30AM 0 TDM400P channel problem
2:18AM 6 g723 to g723 SIP call - warning message
 
Tuesday November 18 2003
TimeRepliesSubject
7:57PM 5 Can't connect to digium cvs
6:22PM 0 Unable to specify channel 2: Device or resource busy
6:04PM 12 This is how you Search the Archives
4:29PM 4 Question about incoming/outgoing
4:18PM 4 iax vs iax2 question
4:10PM 5 FXO Card/Interface for Australia
2:57PM 3 Wifi600 or other Wifi sip phones
2:40PM 4 Ethereal plugin for IAX2
2:16PM 5 hold music =]
1:15PM 14 "Unable to find path from G729A to ULAW" on Sipphone.com
1:01PM 0 Bad DTMF detection
1:00PM 3 SIP Context from domain?
12:03PM 9 Asterisk GUI Client Released!!!
11:37AM 0 Musisc on hold insted of Ringing tone
11:18AM 4 Help with Warnings
10:56AM 1 Asterisk with External Voicemail
10:04AM 1 telco access ?s -- PRI, T1, POTS?
9:22AM 1 DIAX - Can place a call, but can't be called?!
9:07AM 1 Will Asterisk be supporting RTCP XR in the future?
9:01AM 2 ISDN Card Types for Europe
8:56AM 1 capi config
8:34AM 0 Swissvoice ip10s MGCP questions and experiences
8:25AM 0 codec problems between * and cisco hardware h323
8:01AM 0 Hard & soft phones
7:45AM 2 ask problem about softphone--asterisk--softphone, Urgent!!!
7:02AM 0 App Queue
6:54AM 0 DIGI Datafire QuadMicro
6:52AM 9 Notice with asterisk System application
6:43AM 15 Bayonne and Asterisk
6:13AM 0 Asterisk Festival Perl Net::POP3
6:03AM 0 Re: DMTF tones when VOIP call comes in
3:41AM 3 double-dial in SIP Grandstream
2:26AM 0 SIP silence detection
1:35AM 13 PBX (Asterisk) <-> Cellular Phone Network
1:14AM 2 mysql addon
 
Monday November 17 2003
TimeRepliesSubject
9:44PM 1 IAX2 Problem
7:56PM 18 Struggling with grandstream sip to asterisk
4:45PM 21 Updated iaxComm binaries available for WinXP, Red Hat 9.0
4:18PM 0 Asterisks and scripting to get an ACD
4:16PM 0 RE: Asterisk-Users digest, Vol 1 #1918 - 9 msgs
3:06PM 3 asterisk and Codec G-723
3:02PM 18 3Com NBX phones
2:16PM 5 Asterisk Cdr
1:53PM 3 SIP calls no longer work
1:52PM 0 New FAQ on Echo Cancellation
1:24PM 4 Hunt groups and SIP?
12:09PM 3 help voicepulse connect
11:49AM 5 Wifi600 problem
11:37AM 4 Static Config?
11:10AM 0 qtelnet product, FXO gateway
10:59AM 8 problems with alsa (card ac97) in asterisk
10:58AM 1 Sample proposal
10:04AM 24 Radius on *
9:50AM 0 Transfer directly to voicemail - Solved
9:32AM 1 mpg123 core when stopping asterisk
9:22AM 1 IAX2 and MWI
9:04AM 0 GSM or WAV files for musiconhold?
8:29AM 4 Transfer directly to voicemail?
8:12AM 2 iconnecthere incoming
7:34AM 8 VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk
7:14AM 0 SIP soft phone registration
5:38AM 4 IAX2 connectivity problem (qualify=yes)
5:38AM 1 Voip providers U.S (eastern) ??
5:09AM 2 voicepulse working?
3:58AM 15 DTMF
3:10AM 0 Posting the data of user agent from extensions.conf to MySql
3:08AM 0 asterisk iEnsemble H323 communication
2:51AM 0 mgcp audiocodes mp200
2:40AM 0 receive fax with isdn
1:02AM 2 ISDN debugging and SIP dial-in issue]
12:57AM 3 Updated Asterisk-NL
 
Sunday November 16 2003
TimeRepliesSubject
10:01PM 3 wireless
9:57PM 6 Distinctive Ring
9:33PM 3 asterisk installation error
9:16PM 1 strange Music on Hold between SNOM, Grandstream and Asterisk
6:13PM 23 FXO Cards in Australia
4:13PM 0 Echo/fault isolation test gear
3:37PM 0 Enhanced VoiceMail Patch... (vm2)
3:25PM 1 Message lamp integration with legacy pbx during conversion
2:07PM 1 Attempting to contact John Brown
12:32PM 2 two X100P cards, different context
10:24AM 0 echo probs
9:56AM 0 Consultant? List yourself on the Asterisk Wiki
9:36AM 0 Incoming calls randomly hangup and blank calls
7:53AM 0 [schaefer: Re: ISDN debugging and SIP dial-in issue]
7:11AM 0 * is crashing, when the call is accepted (H.323 -> SIP)
5:43AM 17 wcfxo installation error
5:42AM 0 wcfxo installatio n error
 
Saturday November 15 2003
TimeRepliesSubject
7:54PM 0 Problem with call pickup -or- what stupid mistake have I made?
3:16PM 0 asterisk instant hosting ---
1:11PM 2 Internal server error - cannot align media streams - help needed
10:33AM 4 Manual Rindown
8:39AM 0 OSS/dsp sound problem
7:59AM 15 MeetMe problem
4:19AM 2 ISDN debugging and SIP dial-in issue
2:14AM 4 Problem with the Internet LineJACK ISA card...
 
Friday November 14 2003
TimeRepliesSubject
9:50PM 1 Snom 200, asterisk, MOH and Transfers
9:49PM 4 MWI and SNOM 200
7:34PM 4 Fax over SIP alaw/ulaw
6:00PM 0 Hidden bug in *8 call pickup with Sip
3:56PM 8 Cisco 7920
3:23PM 2 mpg123 causing Asterisk Freeze?
2:58PM 0 CallerID when transfering
1:47PM 3 chan_zap won't load after CVS update
1:00PM 3 Potential call logging problem for commercial systems..
12:55PM 0 Re: 9. Zhone zplex (Angel Gomez Garcia)
12:42PM 1 Calls drop after 10 seconds
12:22PM 0 PRI E100P
11:48AM 17 Confused about Asterisk server with regards to Linux NAT Firewall
11:19AM 1 How To Transfer From IAX Clients
11:18AM 2 Streaming channels from Asterisk to the Internet
10:39AM 1 Re: 9. Zhone zplex (Angel Gomez Garcia)
10:38AM 0 SIP Intercom & Paging (was Overhead Paging)
9:48AM 5 dtmfmode & SIPDtmfMode
9:47AM 2 For Sale - (10) Dialogic D/240SC-T1 REV2
8:30AM 3 Looking for recommendations for home office setups
7:07AM 0 Asterisk and Ser - NEWBIE
4:13AM 10 Your thoughts..
4:05AM 0 RE: Aculab SS7/ISUP
3:33AM 0 SIP channel mixup
 
Thursday November 13 2003
TimeRepliesSubject
10:36PM 4 iax configuration
8:03PM 4 (no subject)
3:50PM 1 RE: Aculab SS7/ISUP (new subject)
3:33PM 33 Overhead Paging
3:09PM 2 Feature request
2:44PM 1 asterisk solution.
2:35PM 0 Needed: Digium/Asterisk Resller in Riverside, CA Area
2:31PM 0 Background only responds to 1 digit
2:07PM 0 Indications - ring signals etc
1:23PM 2 Errors in build.
1:21PM 0 Cisco 2610 as an MGCP gateway
1:02PM 0 TDM40B > X100P Call Waiting
12:11PM 2 EU SIP Phone providers
12:10PM 0 AGI verbose command
12:05PM 1 box for asterisk
10:09AM 2 IAX trunk monitoring
9:49AM 2 Cisco 7910
9:10AM 0 Problem in transfer the records of user id to the MySql database
8:58AM 1 CVS repository changes
8:33AM 4 multi call iconenct?
6:59AM 1 how to interconnect gnugk and asterisk?
6:53AM 0 2 AGI questions..
6:41AM 16 Graphical Interface
6:31AM 4 Exit the Directory Application?
5:52AM 10 Network Voip Carrier Termination (Off Topic)
5:38AM 0 New User - Auto Attend / IVR
5:35AM 0 recommendation?
5:20AM 6 Limit timeout of outgoing calls??
5:07AM 27 I hate to do this but..
4:31AM 3 Couple of Questions for Australian Users!
4:19AM 2 Assignement of extension to Netmeeting with dynamic IP address
4:10AM 1 IAX2 based software client ..pls help
2:42AM 3 Open Source Linux PBX!
 
Wednesday November 12 2003
TimeRepliesSubject
10:06PM 1 (HI,new to asterisk)connecting asterisk to telephonyhardware
8:33PM 2 SPA 2000 and 404 not found
8:12PM 1 ADSI Functions
6:44PM 1 IAX channel and transfering calls
6:36PM 5 pause after dialed option
6:28PM 2 vm email notifications
6:05PM 9 Distintive Ring on x100p
5:42PM 0 asterisk cvs ebuilds for gentoo portage system
4:22PM 0 H.323 strange error
2:46PM 2 Zap timeout not occurring
2:32PM 1 "488 not acceptable here" message
1:55PM 0 Sipura / Handytone / Cisco
12:59PM 9 SoftFax question
12:41PM 4 Canadian VoIP termination?
12:40PM 5 D Channel Bonding
12:38PM 1 Zultys.
11:48AM 1 No outgoing audio
10:43AM 0 Echo sometimes with TDM40B / X100P only
10:37AM 1 X100P random hangups.
10:06AM 12 Dial Plan Sequencing
8:41AM 2 Media Negotiation Failed
8:36AM 2 TAPI development
8:07AM 5 DIAX 0.93 with some sound improvements and not only...
7:31AM 1 IAX needs a zaptel device?
1:58AM 11 MySQL Licence may be changing..
1:29AM 14 FreeBSD
12:07AM 0 sending MWI to a none local client
 
Tuesday November 11 2003
TimeRepliesSubject
8:22PM 0 DISA questions ?
8:09PM 2 sip: 401 unauthorized with xlite
5:00PM 0 (no subject)
2:37PM 0 xlite- wierd behavior
12:57PM 0 Codecs and call failure with Grandstream
12:44PM 0 Help with include files & current CVS
12:08PM 6 pick up ringing exten
12:07PM 9 FWD codecs?
11:49AM 1 Zombie lines.
11:28AM 11 OT: Document Control System?
9:41AM 1 zap show channels -> No such command....
9:23AM 7 dialing 8 in VM2 causes channel lockup?
9:04AM 0 VM, IVR, UM
8:31AM 4 "NO ANSWER" X100P
6:33AM 10 iaxtel down?
4:55AM 0 AGI: "set autohangup"
3:07AM 13 Unable to use voicemail
2:56AM 4 Registering an application
2:11AM 1 Call indicators for Brazil and other countries
 
Monday November 10 2003
TimeRepliesSubject
10:48PM 7 AGI and PHP
4:42PM 0 Asterisk in the NW
4:39PM 2 multiple locations
3:45PM 4 ISDN (isdn4linux) DDI
2:33PM 0 OT - (Cisco 79xx) SIP ver 6.0??
2:28PM 2 Menu's & Sub-Menu's
1:34PM 0 INTRACOM SIP PHONE
1:06PM 26 IAX/IAX2 encryption?
12:55PM 3 Inter-digit minimum
12:43PM 0 Seeking proposals for large county library voice system
12:36PM 1 Periodic crash - avoid this syntax...
12:24PM 3 Asterisk and Polycom Soundpoint IP600
11:26AM 0 SIP and Goto failures?
9:41AM 0 H.323 - SIP Gateway.
9:05AM 0 Asterisk in Dutch
8:14AM 4 Jitter Buffer on chan_sip
7:33AM 0 Cisco router/SIP gateway registration
6:59AM 4 ISDN TBCT....
6:34AM 7 Fedora Core 1
6:18AM 8 Asterisk timing
5:56AM 3 Edit GSM audio files
5:49AM 2 OFF: Newsgroup gtw
4:15AM 0 cisco 7960 intercom
2:56AM 39 OT : For the SQL gurus..
12:45AM 1 Problem in MySql-3.23.49
 
Sunday November 9 2003
TimeRepliesSubject
8:15PM 0 RX volume
5:53PM 5 Dialing 800 numbers through FWD or SIPphone?
5:38PM 0 G.729 confusion
5:30PM 0 channel bank and sip phone
2:38PM 8 Multi phone presentation
1:01PM 4 Iax2 channel usage
12:10PM 3 chan_capi & Eicon Diva problem
10:53AM 33 DIAX version 0.9.2 available for download
8:00AM 1 vertical service codes (US standard)
7:00AM 9 Text entry by DTMF
3:21AM 3 unable to find path
 
Saturday November 8 2003
TimeRepliesSubject
7:46PM 0 hum on z-plex 10
6:54PM 9 Eicon Diva Server 4BRI
6:32PM 4 SIP, Sipura SPA-2000, and Voicemail2
3:40PM 0 contact
2:08PM 10 Call Rate in CDR
10:27AM 4 Snom200 MWI..
 
Friday November 7 2003
TimeRepliesSubject
10:48PM 28 IBM to Run VoIP On Linux
10:28PM 0 [Asterisk-Dev] Help with Conference
9:28PM 0 Re: Asterisk-Users digest, Vol 1 #1835 - 12 msgs
8:28PM 0 Sipura SPA-2000 and Asterisk
6:05PM 5 Softswitch
5:02PM 1 Snom 200 Do Not Disturb ?/
4:53PM 14 Streaming MOH
4:12PM 10 Putting call on hold
3:30PM 1 diax request
3:29PM 0 ++Newbie Question
2:25PM 0 sipdtmfmode problem
1:38PM 1 Asterisk can't connect voice
1:30PM 0 Cisco 6.0 gripes
1:25PM 1 No communication channel
12:55PM 3 AGI dialing??
10:57AM 2 Differents config files
10:43AM 3 Modem as a FXO
10:38AM 0 DTA310 Problems
10:00AM 10 CDR fields
9:07AM 1 need Dutch VoIP provider
8:37AM 4 music on hold (SIP Clients)
8:17AM 16 SIP protocol bug ???
6:48AM 0 RE: msgs archives gsm of asterisk ??? Asterisk-Users digest, Vol 1 #1809 - 16 msgs
6:41AM 0 RE: Asterisk-Users digest, Vol 1 #1808 - 13 msgs archives gsm of asterisk ???
6:40AM 7 grandstream ntp
6:33AM 4 No ringing tone
6:24AM 3 MGCP - Repost
6:22AM 45 Snom 200
5:43AM 6 Unable dial out with the new Oh323 0.5.6
5:18AM 0 ASTERISK FESTIVAL E-MAIL
2:55AM 0 Pulsating, Choppy sound using GS..
1:30AM 3 Callgroups and Pickupgroups in Console/dsp
1:09AM 0 fix isn't
12:24AM 0 Possible fix for grandstream outgoing
 
Thursday November 6 2003
TimeRepliesSubject
11:20PM 2 this is the code that breaks outgoing calls on grandstream
9:27PM 4 Strange Problem with Asterisk....
8:41PM 10 IAX2 trunking on one side only.
8:33PM 0 Manager API - howto
8:02PM 10 5 Channel / Trunk ??
4:45PM 4 Dialing an outside number -- QUESTION --
3:59PM 0 H323 Gateways
3:40PM 69 voicemail
3:22PM 7 Error in Incoming SIP call
3:05PM 2 Voicemail2 attach question?
2:35PM 1 Need testers for new STUN build system
2:32PM 1 Huge SIP traffic!!
2:08PM 2 asterisk + dual phone lines + cisco + backup
1:46PM 0 Outgoing calls to SIP provider
1:13PM 1 processor limits of max concurrent users on a single system
12:34PM 1 Gnophone URL
12:29PM 10 Grandstream problem
12:23PM 1 Can NOT access CVS???
11:38AM 1 chan_sip and budgetone
11:06AM 0 which codec will be used ?
10:33AM 3 configuring DID trunks
10:32AM 5 FW: recording calls
10:29AM 0 ISDN PBX + IVR + Voicemail Configuration - S anity Check ...
10:13AM 2 To SIP or Not?
9:11AM 1 CVS compile problem on asterisk
7:58AM 2 ISDN PBX + IVR + Voicemail Configuration - Sanity Check ...
7:08AM 0 RE: USB handsets/headsets?? (WipeOut)
5:22AM 19 USB handsets/headsets??
5:10AM 0 Voicemail RFC
4:49AM 2 Asterisk and SIP Proxy on same machine?
4:46AM 0 Festiavl Access rejected
4:16AM 8 MtSQL CDR logging
2:54AM 3 which channel format number is right?
2:37AM 4 6.0 image for Cisco 7960's?
12:44AM 3 Voicemail2 vs Voicemail
12:09AM 0 SIP nat not working with budgetone (long)
 
Wednesday November 5 2003
TimeRepliesSubject
11:22PM 1 Manager Server
11:19PM 5 IAX/SIP Client
10:43PM 1 Beginners help
9:49PM 5 Ping AGI Demo
7:55PM 1 X100P modify DTMF tone length
6:42PM 2 Asterix - Digium replacing our current key system
6:28PM 2 asterisk nightmare from hell!
6:21PM 2 Multi-port FXO anyone?
6:20PM 3 7960 Directory, WAS: Anyone using * in a live production environment?
5:57PM 1 X100P + ADSI
4:14PM 7 recording calls
4:11PM 3 my first asterisk install
4:04PM 2 iconnect
3:12PM 2 Hold, park, transfer, etc-- How is it done?
3:00PM 0 Asterisk as a Media Gateway Controller
2:51PM 2 spawn extension (inbound , h, 1) exited non-zero
2:31PM 2 Authenticating zap callers. (callerid or PIN code)
2:03PM 15 The Minimum Cost of Setting up an Asterisk Phone System?
1:40PM 24 archives gsm of asterisk ???
1:38PM 1 To anyone with a grandstream budgetone...
12:25PM 19 Mediatrix 1204
12:22PM 0 What the Installation Instructions SHOULD HAVE SAID..
12:15PM 1 A real-life production scenario
11:34AM 4 i4l-modem dtmf detection
11:24AM 17 Skinny (SCCP) help
11:11AM 7 error compiling asterisk
11:06AM 3 Web Interface for adding new users
11:03AM 6 Error in app_voicemail2.so after CVS update
9:57AM 5 New Phone Review: Clipcomm 101
9:29AM 1 Remote Call Pickup
9:17AM 4 Apple implementation
8:42AM 0 Can't connect voice in Zplex 10B
8:08AM 35 Reasons why I shouldn't use Asterisk?
8:01AM 3 Using Asterisk as a VOIP gateway
7:47AM 2 Need info on Gastman/Astman
7:28AM 2 asterisk-oh323: New version 0.5.6
6:14AM 0 SIP with CIC
5:49AM 0 Agent Logoff question
5:36AM 2 Best place to order Cisco ATA 186
5:19AM 0 Missed calls/activity log in Asterisk
4:38AM 2 First AGI help..
4:14AM 2 g.729 codec registration
4:12AM 0 SIP broken for budgtone.
2:20AM 1 SIP and NAT: try, try again.
1:05AM 0 Client Dev - Newbie questions
12:50AM 1 Outband DTMF on i4l modem
12:18AM 0 DIAX users
 
Tuesday November 4 2003
TimeRepliesSubject
11:38PM 1 asterisk bandwidth management
10:18PM 6 http://www.skype.com/
8:06PM 3 Demo Weather Report AGI v2.0
7:59PM 10 IAX clients and the flash button
2:51PM 5 Asterisk system lock
2:28PM 7 snatching calls
2:15PM 1 A little bit of success
1:11PM 0 Asterisk > Asterisk > PSTN
12:33PM 1 Best or any VoIP provider that works with *?
11:48AM 1 asterisk and zplex10b (fwd)
11:34AM 1 More ringing time on incoming lines
11:06AM 1 Fw: problem zplex 10 B
10:43AM 0 Limit of conference rooms.
10:42AM 3 Does externalip= do anything to help with SIP behind a Linux based NAT router?
10:28AM 1 Transferring to Meetme
10:21AM 0 why does context order make a difference in IAX.CONF
10:02AM 0 Need Help with SIP/H323.
8:41AM 1 multitech.
8:39AM 9 Call Transfert with SwissVoice IP10S in MGCP mode
8:33AM 1 call processing after a PIN
6:12AM 7 *, Fritz!PCI and strange behavior
5:31AM 0 Knowledge Sharing on Asterisks issue.
4:33AM 4 high system load running asterisk
4:02AM 0 Compil error with Mandrake pwlib
4:00AM 2 Alert extensions without answering incoming call?
4:00AM 0 ipphone voicemail problems
3:39AM 0 Problem in running .gsm file
3:18AM 1 Does anyone provide inbound UK numbers using IAX ?
2:48AM 36 Anyone using * in a live production environment?
2:44AM 1 Flash hook -> SIP device
2:37AM 2 asterisk does not hang up
 
Monday November 3 2003
TimeRepliesSubject
8:39PM 3 4 X100P's, 4 7960's Same Box?
7:30PM 0 (no subject)
6:01PM 2 ADSI - PowerTouch 350
5:15PM 5 Transfer from Grandstream BT100?
5:08PM 0 Fwd: RE: Asterisk behind LinkSys NAT Routing
4:43PM 1 isdn, modem, etc.
4:18PM 1 new voicemail notification by calling #?
3:15PM 5 Actiontec's Internet Phone Wizard and Digium's S100U
3:07PM 0 csv phone log
2:46PM 22 Red Alarm
2:26PM 0 Re:Looking for CTI/IVR/CallCenter/VoIP project/task as freelance developer
2:15PM 5 Call waiting on X100P
2:02PM 0 LGPL IAX2 software phone (for WIndows/Linux platforms)
1:17PM 2 MWI - I know this has been discussed in depth already
12:43PM 0 turn off dial tone on a TDM400p channel
12:37PM 0 NOTICE[16401]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 72 received
12:17PM 9 DIAX Soft phone v0.9.1 is available for downlaod...
11:57AM 6 Rollout tips
11:41AM 0 skipping/jitter in Music on Hold
11:30AM 0 High Availability and Mass Deployment for Asterisk
11:13AM 1 Proper syntax for the "Cut" application?
11:06AM 1 Intel Performance Primitives
10:32AM 1 <--PRI--> * <--PRI--> modem bank - problems
10:31AM 0 Subject: Re: Where can i get the g.723 codec?
10:29AM 1 one way sound with x-lite (sip) -3rd attempt !
9:13AM 0 OHT in fxs hates my answering machine + self fix
9:07AM 3 Aastra 480 ADSI keypad problem
8:07AM 1 /var/spool/asterisk/outgoing
7:28AM 28 Where can i get the g.723 codec?
7:02AM 1 Asterisk compliance with RFC 2617 (qop, nc and cnonce) - in relation to sipcall.co.uk
6:45AM 0 Gnophone problem
6:07AM 12 IAX hardphones? anyone?
3:48AM 15 IAX2 Java library (was Re: New IAX software phone (for WIndows platform))
 
Sunday November 2 2003
TimeRepliesSubject
11:20PM 7 Questions from a total beginner
7:22PM 1 * troubles
6:11PM 3 Read error on sound device
5:56PM 2 Recommended places for beginner to start?
5:14PM 39 Asterisk behind LinkSys NAT Routing
5:08PM 0 surpress dial tone on TDM400p
3:35PM 1 Live real extensions.conf samples?
3:19PM 3 Clearing Queue Stats?
2:42PM 4 PHP Manager examples
1:59PM 3 Fw: a bit frightened, guys
1:21PM 67 New IAX software phone (for WIndows platform)
12:53PM 2 Threeway calling leaves outside trunks bridged
12:36PM 15 a bit frightened, guys
7:44AM 3 one way sound with x-lite (sip) -second attempt
7:15AM 0 Trustix support..
6:43AM 3 Good system board to use with TE410P?
6:04AM 1 FW: NAT router and off-premise SIP audio problem
6:00AM 17 recording files for menues
 
Saturday November 1 2003
TimeRepliesSubject
9:55PM 4 NetJet Cards
8:59PM 5 which TDM to use? DID line from telco with no dial tone and no voltage
6:15PM 21 Quick Question
4:17PM 2 broadcast voicemail msg ??
12:15PM 4 Making a Skinny phone talk to Asterisk
12:08PM 0 sizing - conference room
11:08AM 5 (no subject)
8:32AM 0 iax vs iax2 connections
6:39AM 5 NAT router and off-premise SIP audio problem
1:11AM 3 FXO modules for TDM400P?
12:40AM 0 Directory App Weirdness