asterisk users - Aug 2003

Sunday August 31 2003
TimeRepliesSubject
1:39PM 14 Newbie IVR question
10:43AM 5 DBSaveTree & DBLoadTree
7:04AM 6 Newbie - setup help
12:54AM 0 (no subject)
12:47AM 3 Message-waiting-indicator thru ZAP interfaces - how to?
12:24AM 2 ENUM, iax,iax2 and h323?
 
Saturday August 30 2003
TimeRepliesSubject
7:36PM 3 ATA 186 & DynExtenDB (query extensions vía sql)
3:35PM 0 Caller Id Issues
1:40PM 3 Filling PHP Variable from EXTENSION in AGI
10:38AM 1 Incomming call issue
6:28AM 3 Conference without zaptel??
3:23AM 12 Installation Problem
2:31AM 0 OT: My congestion music.
 
Friday August 29 2003
TimeRepliesSubject
11:04PM 4 Packet8 DTA310
7:54PM 0 Queue timeouts
7:35PM 5 sample configs
5:24PM 11 Asterisk and Cisco 7960
1:58PM 0 voicemail.conf emailbody fromaddress
1:33PM 4 G729 error messages
1:12PM 3 Restricting concurrent SIP calls
11:16AM 2 sip and pix
9:10AM 1 Version number not updating after CVS
7:47AM 12 Festival and Asterisk
6:41AM 1 X100P not hanging up.
5:22AM 1 Buffering DTMF input
3:50AM 2 additional digit in front of the dialed extenesion by outgoing pri/E1 call
 
Thursday August 28 2003
TimeRepliesSubject
7:44PM 2 (no subject)
3:43PM 4 Video conference apps/appliances?
3:12PM 2 Problems with TDM400P & X100P
12:36PM 0 Re: Three way calling on outgoing FXO line (Martin Pycko)
12:27PM 0 AgentLogin and Huntgroups
10:26AM 3 X100P in Spain & Busy Detect
9:18AM 2 Three way calling on outgoing FXO line
9:08AM 28 Asterisk stops responding
8:16AM 12 SIP and ECHO
8:00AM 0 Echo cancellation problem from SIP to PSTN
2:13AM 0 Asterisk -> Intertex
 
Wednesday August 27 2003
TimeRepliesSubject
5:15PM 0 * newbie: overhead paging and nbsd
1:22PM 0 Chan_h323/g729 - X100P connecting to non-Digium Partner
12:38PM 2 Configuration Adtran TA 750
11:46AM 1 Default Flash Time
11:28AM 1 PCI X100P card interrupt problems
9:42AM 4 Question About BRI Cards
9:31AM 4 ADSI Programs
8:59AM 1 Polycom SoundPoint 500 with Asterisk
7:27AM 1 sample configs / load module failure
6:46AM 2 include context
2:32AM 0 Registering via IAX2 succeeds, but bridging to the registered peer fails
1:58AM 0 H323 caller ID
12:39AM 2 SNOM 200 bugs
12:10AM 7 conference authorization
 
Tuesday August 26 2003
TimeRepliesSubject
10:00PM 0 Chan_h323 does not seem to send the destimation number to gateway
8:55PM 0 Chan_h323 support for phone numbers via gateway?
6:33PM 3 H.323 channel problems
6:13PM 0 have everyone use asterisk to setup a network as iconnecthere or quicknet?
5:19PM 0 Ordering ISDN PRI
2:55PM 5 Dialed Number Identification in analog hunt group
1:30PM 3 x100P: Ring/off-hook in strange state 6 on channel1
11:47AM 0 Pickup groups with SIP
10:02AM 0 Hardware Requirement for Asterisk PBX
9:36AM 1 More questions. Call Waiting and Threeway
8:13AM 2 Asterisk internal database access
8:11AM 1 Problem starting Asterisk after abnormal shutdown
7:36AM 0 Accountcode and cdr-csv
6:35AM 0 Forward but wait for acknowledgement
6:25AM 0 phonecore in gnophone?
5:04AM 0 bug report: whitespaces in uris
4:22AM 6 * server based Phonebook
4:12AM 0 Decent DECT cordless compatible with Asterisk/ATA?
3:44AM 0 TDM10M && Siemens Euroset 2015
3:02AM 1 Alias limitation in asterisk-oh323.0.5.5
 
Monday August 25 2003
TimeRepliesSubject
11:13PM 3 Chan_h323 and a Cisco Gateway
10:21PM 1 gnophone connection
8:21PM 0 Voicetronix V4PCI
3:45PM 11 SIP vs SCCP vs XML
3:02PM 0 FXO gateway experience?
2:59PM 4 Secondary gatekeeper support by asterisk h323 drivers
2:15PM 4 Intercom with Cisco SIP 796x phones?
2:09PM 6 0 out of voicemail to different secretaries
1:58PM 0 Fwd: Data calls through *
1:35PM 2 Problems reloading
12:13PM 2 Warning from chan_zap ring requested
12:10PM 3 Data calls through *
11:28AM 8 T100P/ TSU 600 installation problem
11:20AM 1 Unified messaging.
11:16AM 4 Is the DTMF bug in bugs.digium.com what number.
10:37AM 0 te410p with serial console fails with error: TE410P: Double/missed interrupt detected
10:33AM 9 Cisco 7940 SIP
10:28AM 0 RE: Asterisk-Users digest, Vol 1 #1133 - 18 msgs
10:05AM 0 Caller ID and Call Waiting.
9:20AM 8 Syncronize Monitored Calls
8:39AM 0 No Audio on SIP Phone Connection
8:34AM 0 Manager interface & Event:Leave
8:22AM 0 ISDN Sub Address
6:54AM 4 SetVar on sample.call
6:11AM 9 call center - operators not using phone keys
4:33AM 12 Why doesnt anyone reply me ?
4:19AM 1 I4L CallerID not working
2:54AM 19 SIP phones
2:51AM 5 Grandstream firmware update DMTF Payload Type
2:26AM 0 Grandstream firmware update. {HTTP error}
2:21AM 0 ENUM on Asterisk
1:33AM 4 chan_zap.c zt_rec: Unknown error 500
 
Sunday August 24 2003
TimeRepliesSubject
10:42PM 6 GS on ebay...
9:49PM 1 GS geek info
9:39PM 13 T1 to T1 on asterisk?
8:18PM 0 Music on hold - multiple formats
1:12PM 1 Any way to distinguish between...
12:46PM 2 line numbering and gosub
7:30AM 3 Sound files for internal functions in another place??
6:47AM 1 Grandstream firmware update.
1:52AM 5 fatal embrace control in menus ?
1:22AM 0 Non Asterisk - Apology to the list
12:05AM 0 X100P disconnecting without any reason
 
Saturday August 23 2003
TimeRepliesSubject
10:43PM 18 DIAL via CLI missing
8:09PM 5 Grandstream and CallerID not working
5:41PM 4 Private ENUM examples?
5:09PM 0 [Asterisk-Dev] Re: SIP change...
12:32PM 1 There is any cache for sound files?
11:43AM 1 Webmin and incoming call recording
11:09AM 0 One-way audio using console
5:28AM 0 callerid, callwaiting callerid, Asterisk and ATA
12:09AM 2 SIP change...
 
Friday August 22 2003
TimeRepliesSubject
11:05PM 0 Intresting Vonage story...
9:44PM 12 Intresting.. hrm
9:01PM 12 Caller ID problem
8:41PM 1 pardon the newbie question
8:39PM 0 Warning message in /var/log/asterisk/messages
8:01PM 2 Game time is over gang
7:01PM 5 CVS Question
4:39PM 0 dtmf/audio before going offhook
11:40AM 0 "Frame rejections" on E1 trucks
9:05AM 6 DTMF tones not long enough on out going call s
8:34AM 1 cdr_csv actual duaration
8:33AM 0 DTMF tones not long enough on out going calls
7:08AM 3 Need a "trick" to generate calls
4:26AM 1 Slowly get it ... Hardware
4:16AM 6 sox and wav to gsm conversion quality issue
2:43AM 1 Best SIP phone?
2:03AM 3 IAXtel + NAT
1:54AM 0 Crash using alsa
12:47AM 2 Re: ATAs
12:43AM 0 Pager support
 
Thursday August 21 2003
TimeRepliesSubject
10:32PM 1 Structured release, Maillists
9:22PM 0 Dial in modem speeds over VoIP?
4:36PM 0 problem with manager: Response error, Missing action in request
1:39PM 1 Voicemail2 and RFC2833 DTMF
1:26PM 7 Asterisk + SNOM + Pound and star keys
1:25PM 1 Working example of "switch"?
1:20PM 0 Which linux soft phone is best with asterisk.
1:17PM 6 Background Noise
12:34PM 1 Status of ISDN && DTMF (AFAIK): Please add corrections and comments
12:27PM 1 Minnesota PUC: Phone rules apply to VoIP
11:46AM 5 Re: Some questions about Asterisk and reliability
10:49AM 0 RTP channel
10:24AM 0 ATA 186 - X-Lite and Asterisk
9:51AM 1 Subject: Provisioning CO lines
8:36AM 9 Grandstream Budgetone Defective Units
8:25AM 0 Zaptel.conf & digium E100P
8:21AM 2 Xphone Lite Cannot make work on Asterisk
8:20AM 12 Provisioning CO lines
8:08AM 9 AGI Channel Status
8:06AM 1 Question on setting up MeetMe conference bridge
7:50AM 3 Sending dtmf over an ougoing call from asterisk
7:14AM 1 Multi-extension buttoned phones
6:54AM 5 Newbie Question / ISDN
6:33AM 1 Cisco 79xx XML carriage returns/line feeds
4:58AM 2 asterisk-oh323 v0.5.5
3:56AM 0 No audio in either direction, sip channels hanging, asterisk will not shut down.
2:12AM 7 Conference + time limit
1:52AM 0 Asterisk BoF: Boston, Sept _2[2-4] - interested?
1:32AM 7 911, networks of * servers, etc. (was: VOIP Dialtone?)
12:43AM 1 Asterisk and RTP flow
12:31AM 2 Configurable auto forward in Asterisk
 
Wednesday August 20 2003
TimeRepliesSubject
11:32PM 8 BudgeTone Firmware 1.0.3.78?
8:19PM 0 DTMF Tone length
1:52PM 0 reload problem fixed
1:20PM 34 VoIP dialtone?
12:39PM 3 PRI CallerID problem
12:01PM 2 Asterisk introductory talk: Portland, OR USA
11:48AM 4 RTP header compression?
11:27AM 1 ATA-186 locking: implausible unlock method
11:16AM 1 IAX to zaptel echo
10:49AM 3 Strange happenings
10:15AM 0 Adtran TA 750
9:45AM 6 IAX <> IAX trunking... DP cache?
9:33AM 3 AudioCodes MP108 8-Port FXO Analog Gateway (SIP)
8:42AM 1 X-Lite Build 1059 problems
8:42AM 30 Is Asterisk ready for "real" use?
8:29AM 8 VAD (silence suppression) on Asterisk
8:14AM 0 Re: Asterisk diskless server, a web page with more info?
8:11AM 27 Hardware question
6:53AM 0 App Directory issues-again?
6:49AM 7 ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
6:49AM 4 Conference call
6:04AM 5 reload not working
4:38AM 3 Queue
2:47AM 0 snom100(with latest firmware) screeching noise when doing transfers,
2:09AM 0 Dialogic cards...
12:20AM 7 SIP using which codec?
12:07AM 4 weird error message with zaptel
 
Tuesday August 19 2003
TimeRepliesSubject
11:42PM 8 Where to find correct ver of OpenH323 & PWLIB for Chan_h323
11:23PM 1 echo on the sip side
11:17PM 6 Limit Number of user in Conference
9:03PM 4 Compile problems
6:46PM 0 Re: Open source IP phone, maybe?
5:40PM 1 Speex & openh323
5:16PM 1 Problem with * server and FWD
2:21PM 0 # Transfer context problem
2:12PM 5 trying to make a X100P work
2:06PM 2 Re: Open source IP phone, maybe?
1:23PM 0 current status of i4l and dtmf stuff
12:30PM 0 SPAMWARNING:216.207.245.21:RE: Re: LAN switches with PoE? PoE phones?
12:09PM 5 Vonage locked ATA-186 question
11:35AM 3 Oen source IP phone, maybe?
10:54AM 0 RE: IAX, Asterisk, GSM,SPEEX and ILBC (fwd)
10:48AM 6 [OT] Virus propagation by asterisk user member.
9:17AM 10 SIP QUESTION
9:06AM 7 Analog lines
7:41AM 1 MWI question
6:59AM 1 How Do I disable faststart?
6:08AM 13 PrePaid and IVR
3:11AM 10 MusicOnHold
2:08AM 3 Brooktrout PRI-ISA48 card... info..
2:03AM 1 Hiding and Changing Caller ID
 
Monday August 18 2003
TimeRepliesSubject
4:45PM 1 zaptel does not compile anymore
4:36PM 2 Asterisk Outbound Calling Warning: Unable To Forward Voice
4:34PM 0 screeching and MOH bleed on PRI
2:51PM 11 Call transfer ATA186
2:31PM 5 Cisco 7940 7960
2:06PM 6 Voicemail2 vs. Voicemail
1:57PM 8 Grandstream, SIP encryption
12:44PM 1 * and IAX as a gateway to video conferencing
12:19PM 11 PRI Question
11:03AM 4 chan_h323.c
10:59AM 1 Asterisk's configuration : Which signalling in France with an E1 ?
9:48AM 1 dumb x100p question
7:35AM 9 sound problem
7:35AM 0 Setting a minimum 'on-hook' interval?
7:31AM 7 403 FORBIDDEN Help!
5:34AM 5 Pops
5:13AM 1 Can I runAsterisk remotely from telnetsession?
4:19AM 0 (ATTENTION ) Quicknet Lan jack and phone jack
3:31AM 2 Cisco 7920 phone
3:09AM 6 MOH with SIP
2:29AM 0 Re: FW: Fax from 925 603 5512 (18 pages)
2:27AM 0 Re: FW: Fax from 925 603 5512 (18 pages)
 
Sunday August 17 2003
TimeRepliesSubject
11:52PM 0 Receptionist Console
11:46PM 1 Java SIP Client
10:48PM 14 Monitor application temporary hack
5:25PM 1 Asterix Newbie
1:12PM 3 no incoming packets & Sound: Recording overrun
1:05PM 2 Recomendations for an ISDN-PBX to use with asterisk
12:28PM 0 HP300 phone
11:50AM 1 BudgeTone NAT issues
10:44AM 8 LAN switches with PoE? PoE phones?
10:18AM 1 pre-newbie - some basic questions...
9:29AM 1 Configuring iptables to allow sip and dynamically allocate rtp ports
8:58AM 1 Has anyone got sip/IAX working behind a firewall?
6:32AM 2 chan_capi compile errors with latest CVS
5:56AM 10 Chan_h323 one way audio
5:22AM 5 Grandstream Budgetone
12:18AM 1 call routing based on dnis
 
Saturday August 16 2003
TimeRepliesSubject
6:49PM 8 Voicemail cliping digits via sip
6:45PM 1 music on hold help
6:44PM 0 Great concept but a few issues unresolved
2:13PM 4 Voicemail2 patches
7:17AM 5 Questions regarding CDR's
5:12AM 1 Chan_h323.so native?
 
Friday August 15 2003
TimeRepliesSubject
5:21PM 0 Registring soft phones in Asterisk
2:29PM 4 FXO/FXS hotline
10:21AM 6 Can I runAsterisk remotely from telnet session?
9:02AM 0 Autodialer / bulk dialer application
8:55AM 1 DTMF SIP
3:49AM 1 Asterisk H323 Trunk
1:48AM 0 Unable to detect process 2 frames
 
Thursday August 14 2003
TimeRepliesSubject
5:03PM 1 ast_channel_alloc() losing pvt struct
4:08PM 1 *-openh323 faststart
4:05PM 0 *-openh323 & faststart
2:42PM 1 make: warning: Clock skew detected. Your build may be incomplete.
12:21PM 1 Asterisk SIP calls failing - not a proxy? What of RTP codec transcoding?
11:21AM 11 .:. .: .. .Stottering audio ??
11:17AM 1 chan_capi in the US
10:45AM 0 Which version of MySQL are you running?
10:29AM 0 Twisted Idea
9:38AM 1 Re: The Almighty X-Lite DTMF Problem (patch tested)
8:36AM 1 Problem with latest cdr Makefile???
8:35AM 0 PHP Web Interface helpers
7:33AM 6 Don't know how to calculate timelen
6:52AM 0 New Asterisk user.
6:47AM 0 Alogirthm Used for Extension Matching ?
6:36AM 1 Virtual extension as local modem
3:24AM 9 What is the highest quality codec I can use for recording voice messages?
1:35AM 11 '#' doesn't work for me
1:28AM 4 How can I know if a user is busy or not connected?
12:33AM 0 Extension and phone management bestpractices??
 
Wednesday August 13 2003
TimeRepliesSubject
11:03PM 0 Conference Number + CDR
10:58PM 0 Fwd: FW: SIP NAT question
4:07PM 0 Voicemail group, adding to the vm when forwarding
3:50PM 3 Receiving iaxtel calls
3:20PM 1 "Double" transfer?
1:06PM 10 Can't compile cdr_mysql
12:25PM 1 SIP NAT question
12:16PM 2 CLASS feature syntax
11:58AM 0 Which GS IP product to buy??
11:58AM 1 Park and out-going trunk calls.
11:46AM 2 I can't get a two way conversation going?
10:10AM 0 Fwd: Stable versions of Asterisk (Was: Re: Fair comparison (John Todd))
9:43AM 5 h extension seems to wipe variables?
8:55AM 0 Asterisk and AT&T 964 phones...
8:46AM 24 FXO mode
7:56AM 1 FWD SIP phone format=2, FWD call format=4, why?
7:56AM 1 Mixing audio from Music on Hold and IVR
4:57AM 15 Extension and phone management best practices??
3:27AM 4 How do i configure so an incoming call triggers an http request?
2:39AM 2 reload
 
Tuesday August 12 2003
TimeRepliesSubject
8:39PM 0 Stable versions of Asterisk (Was: Re: Fair comparison (John Todd))
8:06PM 5 Conference + E100P + H323
3:15PM 1 usrobotics modem and pstn
2:22PM 4 X100P Ringing/Answering
2:09PM 0 RE: Asterisk-Users digest, Vol 1 #1033 - 7 msgs
1:49PM 2 New-ish list of hardware phone vendors
1:32PM 11 Weird DTMF issue
12:47PM 1 URI for dialing
11:24AM 1 Using Asterisk with FWD through NAT
10:06AM 0 Upgrading Queue App
10:05AM 3 Working with FWD, IPTel, SIPPhone?
9:44AM 5 Codec?
9:38AM 41 IP phone recommendation
9:34AM 0 Future Grandstream codecs??
7:43AM 0 Xten-Lite and Asterisk.
7:23AM 0 Fw: Fax Handled
7:22AM 1 new on E100P
6:57AM 3 New gastman clone + what else?
6:05AM 9 Sip and One Way Audio
5:09AM 6 OT: Grandstream power supplies..
3:57AM 3 problem with Wildcard 100XP and hangup signal
3:37AM 6 Fair comparison
3:35AM 19 Open G.729A codec
3:29AM 2 Malicious Call Trace
12:25AM 6 How to Asterisk
12:12AM 0 CVS version build error
 
Monday August 11 2003
TimeRepliesSubject
5:35PM 0 Ring when leaveing queue?
10:59AM 1 zaptel sync
9:19AM 3 Ring while on phone
9:03AM 1 avm fritz pci
8:40AM 2 ANI/DNIS call routing
7:01AM 0 InternetPhoneWiazard
5:34AM 0 chan_capi ptp mode
12:03AM 0 Faking Ring tone
 
Sunday August 10 2003
TimeRepliesSubject
11:11PM 0 Outdial digits - non TDM trunk
6:41PM 0 "Out of area" displayed as caller-id
4:03PM 6 Windows Messenger
3:12PM 14 Registering SIP with FWD and ICONNECTHERE
2:20PM 0 Asterisk (g729) termination on CISCO
9:59AM 2 SNOM200 firmware roll back!!
6:13AM 10 Asterisk Newbie ...
 
Saturday August 9 2003
TimeRepliesSubject
5:03PM 3 Need help with installation of H323 chanel driver
4:29PM 4 Gatekeeper
3:13PM 0 dialogic D/41 ESC with asterisk
2:23PM 3 help please with single t1 configuration
1:47PM 2 H323 and SIP
12:46PM 0 Something very strange regarding callerid
12:06PM 30 app_queue, fewestcalls and leastrecent logic
11:12AM 1 callerid (Bell type) in Europe
9:43AM 0 Call Center RFP
8:19AM 0 This is how to set ATA186 for different standards of CallerID format
7:06AM 1 To Switch or not to Switch... that is the question....
6:43AM 0 callerid, british and french type DECTs
6:07AM 2 Does Wildcard x100p support Caller ID outside the US? (fwd)
5:54AM 1 Chan_Capi questions??
2:55AM 6 Asterisk as a stand alone voice mail server (fwd)
2:42AM 0 Digium & PCI-X
1:19AM 2 Multiple E1 configuration question
12:46AM 0 ATT: marrandy - Re: Grandstream Budgettone 102
12:19AM 0 IAX protocol description
 
Friday August 8 2003
TimeRepliesSubject
6:06PM 2 UNIX command-line interaction with astdb
2:50PM 16 Killing runaway PBX
2:04PM 3 G.729 licensing -- an opinion
1:34PM 0 VoicemailMain2, inband digits detection, rcf2833 digits detection (rtp issue, I think)
12:34PM 4 queue / agent documentation
12:26PM 0 Workaround for BudgeTone "ringing in your ear?"
12:09PM 4 X-Lite - No sound + chan_sip issue
11:37AM 4 Voicemail2 - auto fill the dialing extension?
11:25AM 15 list proposal
10:15AM 0 re: Web GUI
8:55AM 4 segfaults with queue
8:36AM 1 g729 problems
8:29AM 28 Fax Handled
7:28AM 3 Re2: Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]
6:39AM 0 dtmf detection from AS5350 over SIP
6:35AM 8 ip phones and intercom/paging
6:22AM 4 Call Waiting and Call Parking Together??
5:49AM 2 Play Music On Hold just for a fixed period of time
3:45AM 4 CallerID, DECT phones and ATA
1:46AM 2 Snome-200 with Asterisk
1:22AM 0 ISDN BRI outgoing call instant hangup
 
Thursday August 7 2003
TimeRepliesSubject
11:27PM 2 Problem -ATA-711-723-Oh323-Asterisk
11:14PM 3 SIP Lines
10:59PM 3 h323 and cvs one way audio
5:36PM 5 Call routing question
5:35PM 1 TE401P driver warning
3:24PM 2 Newbie Issue
3:10PM 1 How to determine line signalling?
2:51PM 3 3xx SIP messages
2:31PM 1 Busy detect options
1:31PM 1 Error loading latest CVS
12:30PM 7 Sip Trunk config
8:33AM 0 list of sip phones?
8:26AM 2 MWI bug ?
8:18AM 1 Hardware for a Big PBX
4:57AM 0 Minimum system requirement for ....
3:54AM 1 Warning Messages
2:06AM 0 Asterisks integration with pre-existing PBXs
1:53AM 1 cdr_mysql uncompress
12:56AM 3 Leftover Budgettone issues
 
Wednesday August 6 2003
TimeRepliesSubject
11:45PM 2 Semi-newbie question "Softswitch" and Asterisk - Is there a difference?
9:13PM 1 Festival 1.4.3
8:06PM 0 H323 + DTMF detection
7:46PM 1 Feedback for Asterisk Handbook?
5:32PM 6 FYI: G723.1 Licensing Prices
5:08PM 1 BRI newbie queries.
3:27PM 1 Unregister SIP connection?
2:28PM 15 CDR MySQL
1:38PM 12 New SIP Phone
1:36PM 0 Standard Analoge modem - can it be used?
1:35PM 1 Budgettone Newbie
1:12PM 20 AgentCallbackLogin
11:13AM 0 indications.conf settings for Belgium
11:04AM 5 Behind Firewalls, SonicWalls, etc..
10:42AM 0 FAQs and Helpful URLs?
10:26AM 2 X100P CallerID issue solved for my PSTN connection
9:55AM 0 Bad sound quality with G729A on SNOMs
8:55AM 0 ISDN Examples
8:28AM 0 Push to Talk
7:45AM 0 Intermittant IAX Call Failures
6:26AM 0 Config files - examples
6:08AM 33 R2 support
4:45AM 10 X-Lite <-> Snom200
2:08AM 2 X100P and Caller ID (again and again...)
1:11AM 4 iax.conf / Registration rejected
1:02AM 11 chan_oh323 + dtmf
 
Tuesday August 5 2003
TimeRepliesSubject
11:23PM 27 Windows IAX soft phone
9:19PM 1 So now I'm playing around with Queues....
8:40PM 0 CVS troubles
7:15PM 0 WipeOut - gateway access with pin solution
4:33PM 1 T-shirt ideas
2:05PM 6 (no subject)
1:55PM 1 Someone used ADIT 600 Channel Bank.
1:33PM 8 Wierd Message
1:27PM 0 Asterisk and authentication
9:36AM 4 Call Monitor
9:29AM 10 Newbie just starting out with *
6:42AM 0 GSM file format
6:41AM 5 chan_capi: Hanging channels - again
6:10AM 1 Zhone Zplex
5:53AM 17 Does Wildcard x100p support BT Caller ID in UK?
4:59AM 5 Why are FXO so expensive?
2:40AM 0 usable/affordable usb phone?
1:53AM 8 SendDtmf
12:45AM 3 Zhone Zplex 10 units
 
Monday August 4 2003
TimeRepliesSubject
8:40PM 3 Channel banks, etc.
4:56PM 0 Transitioning from existing PBX
4:54PM 10 SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers
3:39PM 3 Fix for Redhat 9 zombie AGI processes
3:12PM 5 FW: Cisco 7960, SIP, NAT, Voicemal
2:00PM 0 Repost MS Messenger 4.7 docs?
12:16PM 14 bugs.digium.com
11:42AM 0 Bridged trunks stuck off hook.
9:56AM 8 Syntax for hiding caller ID but still passing ANI?
9:05AM 10 newbie question - devices
9:01AM 0 Some questions about a potential usage scena rio for asterisk
8:26AM 3 limiting out going calls to a maximum duration
8:21AM 6 Some questions about a potential usage scenario for asterisk
5:51AM 9 H323 CallerID
5:31AM 11 CDR
5:03AM 0 mem leak in logger.c?
4:33AM 0 Any pointers for setting up PRI for incoming and outgoing calls?
4:11AM 0 small fix in chan_mgcp.c
2:37AM 1 SIP clients not sending audio
12:23AM 21 Mysql CDR
 
Sunday August 3 2003
TimeRepliesSubject
3:17PM 0 g.729 licenses do not release when used in Voicemail
1:11PM 0 RTP / SIP routing issues
1:10PM 3 AGI accountcode.
11:57AM 1 Prepaid calling card
9:56AM 2 Fax Detection?
 
Saturday August 2 2003
TimeRepliesSubject
7:43PM 30 call waiting
7:27PM 3 SIP app_queue
4:29PM 0 Webalizer for CDR logs....
11:00AM 10 Asterisk agi interface leaves zombie processes?
8:33AM 1 Patch - transfer with two rather than one #
5:37AM 5 GSM codec
12:45AM 2 Asterisk + SER
 
Friday August 1 2003
TimeRepliesSubject
7:25PM 2 Hangup after a Timeout
4:13PM 1 HELP!!!! Ringback oh323
3:56PM 1 HELP!!!!
3:46PM 11 Using OH323 and Gatekeeper
2:52PM 1 ztdummy & usb-ohci?
2:16PM 0 Cisco AS5300 -- Not hearing anything
1:58PM 10 Musiconhold interrupted sound
1:24PM 14 Seting up TDM40B
12:50PM 0 Overlap on PRI to PSTN
11:50AM 0 pcphoneline producs
11:32AM 1 Monitor app
9:09AM 6 phone rings while already on a call
8:19AM 0 Background messages while waiting for pick-up
8:04AM 0 Problem with SIP Native Bridging and UPnP
7:21AM 0 DTMF onto the reall world
7:17AM 5 Asterisk community input: FreeTDS (cdr_tds.c) or unixODBC (cdr_unixodbc.c) ?
6:08AM 3 segmentation fault with asterisk and OH323
5:37AM 1 Asterisk SIP bug with Net2Phone
5:25AM 1 Extension handling.
4:28AM 12 DTMF modes and external IVR systems over ISDN
2:36AM 2 memory leak?
1:03AM 1 SIP with an iptables fiewall