asterisk users - Jun 2003

Monday June 30 2003
TimeRepliesSubject
9:40PM 7 Conference calls
9:37PM 24 A solution for SIP and NAT
7:07PM 1 Asterisk against 3 Com NBX 100 and Siemens HiPath 3700/3750
5:27PM 1 chan_h323 woes
4:49PM 0 Anyone Try Zultys Zip2 Phones?
3:31PM 2 Internet Telephony, net2phone
3:29PM 3 MGCP with Cisco doesn't work
3:27PM 0 Re: Connections, but no voice paths except by
2:39PM 2 E400P E1 Pin Layout
2:26PM 0 outgoing calls with packet8 and dta310 problems
2:17PM 0 app_queue ringing all available channels
2:04PM 0 thx to Martin :)
12:12PM 0 CVS Broke my sound output
11:12AM 0 Call Queues and Agents, using both devices and agent members
11:07AM 1 T1 slips/BPVs clarifications (was: Help! Problems talking to upstream switch)
9:28AM 3 Connections, but no voice paths except by console
9:06AM 0 stuck channel
8:31AM 1 Beginner Questions
7:12AM 5 * Video changes
6:00AM 0 mec3 - temporary call distortion
5:01AM 3 E100P installation sheet
3:48AM 1 ISDN PRI E1-CLI and DNIS
3:25AM 0 digium drivers for ISDN card
1:33AM 10 asterisk with modem
 
Sunday June 29 2003
TimeRepliesSubject
6:59PM 7 Help! Problems talking to upstream switch
5:56PM 2 Newbie, Loaded Asterisk can't figure out manual
5:33PM 0 Seeking Debugging Help
3:49PM 3 SIP only with no soundcard?
3:35PM 9 Cisco ATA-186 config guide for Asterisk
12:32PM 16 Minimum budget question ...
9:55AM 10 fixed point mec3
7:59AM 4 newbie .conf question
7:58AM 0 (no subject)
6:18AM 3 PGSQL app and pbx parsing :-(
3:24AM 0 sip dial command 484
 
Saturday June 28 2003
TimeRepliesSubject
3:55PM 4 CPU power required - Asterisk
3:48PM 7 Major format changes
2:27PM 0 SV: Newbie questions.....
2:14PM 2 IAX2 trunking: codec bandwidth comparison notes and results
12:14PM 1 is "zap destroy channel" safe?
 
Friday June 27 2003
TimeRepliesSubject
10:41PM 0 Problems with zombies left after calls to Festival
6:59PM 2 Working: TFTPd for NAT'd Cisco 7960 and ATA-186
6:00PM 1 Channel for virtual modem
4:57PM 0 What user-id should Asterisk run under
4:39PM 3 Terrible audio quality using Asterisk and X-Lite?
4:07PM 2 modprobe ? for TDM40B
4:06PM 2 Zultys SIP Phones - NEW?
3:12PM 0 Re: SIP auth usernames
2:48PM 3 defaultip= in sip.conf doesnt work?
1:03PM 1 (no subject)
12:25PM 0 Re: [Asterisk-Dev] PHP Web interface testing and RFC
12:17PM 1 PHP Web interface testing and RFC
10:10AM 7 Can I disable musiconhold for agents
7:23AM 3 IP phone with asterisk
6:46AM 0 IAXTEL numbers and example
6:39AM 0 I Need Consulting Help!
6:18AM 1 BudgeTone 100 Calling Problems
4:43AM 1 Asterisk CPU usage
4:26AM 3 No dial Tone but its registering from remote site! Anyone with idea?
3:30AM 21 Voicemail issue
3:30AM 2 Making calls from snom 100
2:56AM 1 Advanced SIP management
2:14AM 3 x-lite and audio
1:03AM 3 Basic Asterisk questions - personal coments
 
Thursday June 26 2003
TimeRepliesSubject
11:59PM 3 Detecting off-hook state on extension
11:52PM 1 Retry dial when busy
11:40PM 3 IAX -> IETF draft ??
8:50PM 3 PHP Web interface for Asterisk
8:44PM 0 HEY CISB--- re: REMOVE REMOVE REMOVE REMOVE etc...
8:17PM 0 Almost there.... strange error message... anyone???
3:30PM 5 cisco 186 helpp!ยช!!!!
2:28PM 14 Web interface for Asterisk
1:46PM 7 hunt group
1:34PM 0 Kphone not working with Asterisk?
12:55PM 2 Important: PSTN access-number for Dutch gateway changed
12:27PM 1 unsubscribe from list
10:54AM 0 What is Newt?
10:39AM 14 use of Asterisk and T100P as Nortel DSX-1?
10:19AM 1 Removing ones self
9:59AM 1 bug in cdr ?
9:41AM 0 Questions about Voicemail2..
9:38AM 0 MTRG and Asterisk.
8:59AM 0 mec3 experiment
8:27AM 1 T1 or T1PRI? which sould i use?
6:56AM 1 X100P and echo cancelation
6:45AM 2 how to identify user using chan_H323
6:42AM 3 CallerID with chan_capi
3:42AM 2 No busy detection
3:38AM 17 Asterisk, IAX and NAT issue
2:02AM 2 Congestion or Busy app using I4L indicate ringing
 
Wednesday June 25 2003
TimeRepliesSubject
11:34PM 1 Code some * examples for me? I'll pay you! :)
11:16PM 2 Asterisk - first impressions
6:05PM 1 Pattern matching: least-to-most specific PITA
4:36PM 2 Meridian Option 11 with T100P
4:02PM 2 indication tones and callwaiting chirp too loud
3:11PM 0 oh323 and GSM
12:52PM 3 Possible solution to Zaptel panics
12:44PM 5 Asterisk hardphone
12:01PM 8 Fax and SIP
11:26AM 0 Webmin module
11:21AM 0 gastman -- how do you build it?
8:17AM 34 snom 100 and GSM codec
8:04AM 1 Linux-IAX-Client
6:57AM 0 IAX termination in the US
6:29AM 3 no sound pri --> h323
5:56AM 0 RTP stream missing the target - cisco 5300 + mgcp
4:53AM 0 No field 'Via' present to copy
3:56AM 13 Asterisk and FWD
2:34AM 1 Problems with music during tones of dial.
12:27AM 0 Hardware Survey: your configurations
 
Tuesday June 24 2003
TimeRepliesSubject
8:32PM 1 chan_oh323.c Segmentation fault during Openphone/Gnomemeeting connect during module loading...
8:19PM 3 Tor2 modprobe problem
7:23PM 0 Which type of lines to get from the Analog PBX??
5:29PM 1 Problems with # and extensions.
4:28PM 1 Asterisk SIP-to-SIP proxy
3:59PM 3 Compiling Asterisk under Yellow Dog
2:38PM 1 Distinctive Ring Macro Example
1:51PM 1 Analog 2x8
1:02PM 0 Telephony Portal w/ Asterisk
11:47AM 3 Patching Festival
11:40AM 0 SIP REGISTER script
11:17AM 1 Working Clients for Linux?
11:17AM 2 Asterisk ALSA module not working
8:37AM 0 Conference calls on Pingtel Phones
8:36AM 2 parsing bug? (using PGSQL)
8:30AM 1 Asterisk and Polycom
8:18AM 2 PHP MySQL cdr interface?
8:16AM 1 "NoOp" gives an ringing indication ?
8:14AM 0 Chan Local Examples
3:31AM 4 App queue only + waiting call pickup
12:03AM 13 asterisk and passwords
 
Monday June 23 2003
TimeRepliesSubject
11:19PM 3 (no subject)
4:36PM 6 dynamic queue channels
2:12PM 0 Problem with native bridge function.
12:57PM 2 Ringing tones oh323
11:28AM 0 Gastman and New Extension
7:32AM 3 Sip too many open files?
6:58AM 3 Setting up the E100P
6:23AM 1 codecs question ..
6:13AM 2 Asterisk CPU power requirements
6:12AM 0 TDM400P and Caller ID on Call Waiting
4:19AM 0 Dialogic Proline 2v Supported?
3:39AM 0 Budgetone + remote call pickup
2:49AM 0 Process multiple commands on dial out..
2:34AM 2 help with pri configuration..
1:02AM 0 I need IAX sample config..
12:07AM 0 Stopping the ADSI <BBBOOIIIINNNNGGG!!!!> on call wait
 
Sunday June 22 2003
TimeRepliesSubject
9:58PM 1 FWD and registrations
9:07PM 4 Is this possible:
1:51PM 0 new user here
9:27AM 4 Please Help: Trying to build Asterisk - bazillions of errors
8:14AM 2 PCI CARD
4:04AM 0 what hardware to choose from?
3:09AM 2 Sip and dual homed box
2:59AM 2 How can I log SIP debug messages to a file?
12:13AM 9 asteisk, sip & NAT
 
Saturday June 21 2003
TimeRepliesSubject
7:26PM 5 Grandstream BudgeTone?
2:36PM 0 Today's CVS version hangs on alsa module
1:29PM 0 Whats required for unblockable ANI / CLID?
11:21AM 1 Need help with inbound/outbound PRI calls
9:02AM 59 Newbie questions
7:28AM 0 H3500CW
5:39AM 3 PRI & BRI question
4:59AM 0 Kernel Panic: Aiee, Killing Interrupt Handler
12:26AM 2 Grandstream BudgeTone - opinions??
 
Friday June 20 2003
TimeRepliesSubject
10:04PM 1 More than one param to AGI
7:41PM 5 best ISDN BRI solution for DID
6:02PM 1 Moving up... T1 with 50 extensions
5:36PM 0 Specifying Allowed Codecs in iax.conf
2:31PM 0 4 channel TDM40B only has 1 channel now
10:47AM 1 Question :: groundstart and loopstart :: Update
9:59AM 0 Poor quality with FWD - codec selection issue?
9:18AM 11 Newbie questions.....
9:17AM 9 Manager interface, again
8:33AM 4 where to get adsi phones in europe ?
8:16AM 2 SIP registration without password (secret)
8:08AM 1 Error compiling, is it only mee?
7:28AM 0 Free World Dialup change which may be relevant to *
6:32AM 29 Asterisk hogging CPU resources
6:22AM 1 Firewalling, Ports and rtp.conf..
4:28AM 4 Active ISDN PCMCIA card
3:12AM 2 Asterisk VS. Bayonne
2:55AM 1 [HS] results testing asterisk with ISDN BRI & look for help to understand configuring SIP with asterisk
12:19AM 11 databases for billing
 
Thursday June 19 2003
TimeRepliesSubject
9:36PM 0 packet8 and asterisk
9:01PM 0 Problem with CID matching
6:17PM 2 Is it possible to do this with Asterisk?
1:49PM 10 Billsec on CDR
10:57AM 12 festival error
10:49AM 1 Unable to find a path
7:16AM 0 Newbie: Looking to setup calling between 2 analog phones with a TDM20B
7:14AM 1 uclibc enviroment #2
7:09AM 1 Chan_oh323 problem
6:25AM 0 Modems supported by Asterisk
6:22AM 1 soundcore???
5:12AM 0 Asterisk support for Voicetronix
5:08AM 2 chan_capi syntax
4:18AM 0 number of digits from incoming msn on i4l modem
3:58AM 0 "unsubccribe"
2:22AM 3 compile in uclibc enviroment
1:03AM 1 Grandstream and GSM..
1:02AM 0 Dialogic pricing & Natural Micro-systems support
1:00AM 0 Leave one call to pick up another?
 
Wednesday June 18 2003
TimeRepliesSubject
10:07PM 1 New Zealand Users
4:40PM 0 IAXClient news
3:18PM 1 Integration with external ASR engines
3:01PM 11 asterisk -rx under cron?
1:22PM 3 Wrap-up
1:05PM 10 CAC Access Bank
10:32AM 1 chan_agent MOH was (Re: CVS Error 2003-06-19)
9:54AM 0 MP3Player and Ringing (long)
9:53AM 0 Directory Application Context Issue
9:38AM 8 == Everyone is busy at this time problem
9:11AM 1 Newbie making progress... need help with .conf files
9:05AM 2 ISDN BRI
8:46AM 1 Extra parameters in SIP URIs
8:27AM 2 * build problem on older systems
7:54AM 0 list of pbx control sequences?
5:38AM 0 Nevermind about the CVS Error
5:25AM 3 CVS Error 2003-06-19
5:04AM 1 Errors when compiling from CVS this morning
2:22AM 4 Temporized AGI Scripts.
1:55AM 2 Problem with oh323 package for asterisk
 
Tuesday June 17 2003
TimeRepliesSubject
9:18PM 3 newbie needs SIP config examples -- especially soft phones
9:15PM 9 soft phones -- voice quality tuning
9:15PM 0 can I use Nexpath cards with * ?
4:05PM 3 Question :: groundstart and loopstart
3:03PM 13 New busydetect routines for analog channels (FXO mostly)
2:07PM 2 callerid time set
1:51PM 6 Parking causes crash
1:48PM 5 i4l - summary of patches?
1:32PM 4 Directory Application question
9:59AM 4 chan_capi error!!
9:17AM 3 Firewall Silly - anyone can help with a CVS tar ball ?
9:06AM 0 help with SIP softphones
8:23AM 2 Test System?
8:22AM 25 New Module app_perl
8:21AM 2 DTMF with grandstream phones
5:50AM 2 SIP Firmware for Cisco Phones
5:27AM 24 Speex
4:20AM 3 Voice linmodems and Asterisk PBX
4:02AM 4 (no subject)
1:12AM 3 newbie SIP question
12:37AM 6 sip.conf
 
Monday June 16 2003
TimeRepliesSubject
9:14PM 0 ProSLIC error message
7:59PM 0 Fw: X100P questions
2:22PM 0 zttool shows OK for the T400P board which is not configured/start ed
1:24PM 1 1X1 PBX
11:26AM 5 chan_h323 - pwlib 1.4.11, openh 1.11.7 comiple problems
9:35AM 0 newbie: isdn4linux and BRI (FRANCE)
9:19AM 0 Local PBX
9:03AM 5 h323 compile error
8:22AM 0 Setting caller ID on no Caller ID
8:16AM 17 SIP REGISTER
7:58AM 1 Error chan_oh323.so
7:08AM 0 chan_capi and hanging channels
6:42AM 2 queue application
5:44AM 8 G.729 Licencing..
4:54AM 2 Queue App
4:05AM 4 Installing the wcfxs driver
2:24AM 2 The same SIP problems...SORRY!
 
Sunday June 15 2003
TimeRepliesSubject
10:39PM 7 X100P questions
8:41PM 3 GASTMAN AUTH QUESTION
6:20PM 15 VoicemailMain
6:15PM 3 Voicemail with H.323?
4:53PM 0 gnophone experts!
4:49PM 1 SIP REGISTER behavior change: specific domains possible in REGISTER
3:32PM 3 Reminder paging for voicemail (?)
2:03PM 0 Bug with SIP and indications?
12:24PM 2 Whoooaaa!!! Feaky - but in a good way
8:19AM 7 a few questions about sip implementation
7:34AM 10 .gsm files
6:15AM 0 isdn pbx
5:41AM 1 Re: Application, Dialplan not loading
4:34AM 3 Voicemail and DISA fixes
 
Saturday June 14 2003
TimeRepliesSubject
7:22PM 2 Cisco 7960 config?
3:44PM 10 Dialogic D/41E
11:36AM 4 InternetPhoneWizard
10:13AM 0 Asterisk confused when interface has multiple addresses?
8:25AM 1 Intercom/autoanswer, SIP, Cisco
8:21AM 1 show application DISA
 
Friday June 13 2003
TimeRepliesSubject
7:24PM 0 Can asterisk do hoteling?
4:09PM 1 strace shows that files are not accessed
2:26PM 0 Asterisk H323 endpoint (ATA 186)
1:37PM 0 send DTMF digits
11:56AM 7 CallerID forward???
10:03AM 1 Dialogic PCI hardware
9:57AM 5 Call queues for phone operator
9:12AM 8 Asterisk asterisk => statement
7:30AM 8 Disabled echo canceller because of tone (rx)
7:09AM 12 Applications, dialplan not loading
6:39AM 1 Segmentation Fault ... Big problems
6:23AM 4 Problem with outgoing spool...
4:27AM 2 Budgetone Supervised Transfer
2:31AM 0 Hungry channel
12:53AM 1 PSTN and POTS
12:48AM 3 E1 in South Africa
 
Thursday June 12 2003
TimeRepliesSubject
7:37PM 0 Playtones unexpected hangups
6:52PM 3 Convert your FXS port to FXO
6:12PM 4 Telemarketer GSM?
6:00PM 10 E1 cards
11:58AM 1 No way to review Voicemail busy message?
11:52AM 2 Segmentation fault on "reload"
8:21AM 0 ATA losing registration problems solved by setting tftp
8:03AM 2 fxs card not loading in new computer
7:31AM 2 out of curiosity ..
7:14AM 5 E1, E100P
6:35AM 16 Voicemail message as e-mail attachment
6:14AM 1 shutdown cancel?
5:57AM 1 srv.c + srv.h
5:21AM 5 Clock skew detected
3:18AM 1 Info sip/h.323 interoperability
3:00AM 5 Monitor application
2:40AM 0 how can I do unregister?
2:39AM 0 2 different chan_capi core dumps
2:21AM 1 Callerid Modem I4l and outgoing spool
2:07AM 2 Voicemail2 bug (?) saving new messages as new
1:05AM 0 help! I still can't use more than 1 of the 2 BRIs on my AVM C2 (chan_capi)
 
Wednesday June 11 2003
TimeRepliesSubject
11:51PM 2 Asterisk logging questions
10:24PM 0 Thank you very much
5:08PM 5 Telephone Tree
4:35PM 3 AGI and SET VARIABLE
2:00PM 0 Problems configuring Asterisk with SIP
1:16PM 29 Voicemail notification
1:03PM 2 filling suppressed silence with chan_oh323
12:27PM 0 New Asterisk System
11:38AM 6 Busy message with call waiting?
10:46AM 1 segmentation asterisk oh323
10:26AM 2 lost variables
8:43AM 15 How do i make best use of Macro?
6:07AM 4 Configuring zhone zplex to 24 fxs ports
5:03AM 13 Testing two E400P with E1 cross-cable
5:01AM 1 SIP phone behind NAT
3:37AM 5 Dialing out through a Hardware PBX
2:36AM 0 All extensions busy
2:26AM 0 how to receive call on iaxclient
2:11AM 2 Newbie : i try and test to use asterisk
12:47AM 3 E100P Setup
12:37AM 5 some sip questions AGAIN
12:25AM 0 (no subject)
12:23AM 2 some sip questions
 
Tuesday June 10 2003
TimeRepliesSubject
6:08PM 0 a web admin
5:54PM 1 Screenshots of admin GUI
4:16PM 0 chan_h323 + openh323 CVS = no go? (fwd)
2:22PM 1 (no subject)
2:05PM 3 WILDCARD TDM400P or four Wildcard X100P
1:46PM 1 SIP sdp o= and c= fields
12:55PM 0 Directory by names in VMAIL2
12:28PM 1 Re: Adding an app (Steven Critchfield)
11:59AM 1 Using Asterisks with old Rhetorix 4108s?
11:49AM 11 PDA's over SIP channels on Asterisk
10:56AM 2 Slow Faxing
9:40AM 5 chan_h323 + openh323 CVS = no go?
9:03AM 2 paging system (long)
8:57AM 6 s extension don't work on TDM40B
8:19AM 9 Using Linux traffic shaping to prioritise SIP/IAX traffic?
8:07AM 1 IVR
7:21AM 7 Only noise in zap channel
5:18AM 26 chan_oh323
1:38AM 5 NewbieQ: SOHO setup with x100p
12:18AM 9 Opportunistic VoIP
 
Monday June 9 2003
TimeRepliesSubject
11:35PM 1 Att: Bill Zhang (Was: Re: Correction regarding price of Grandstream Budgetone 100 series phone)
6:09PM 4 Setting local IP address for the RTP port
5:08PM 25 Dual T400P, SMP, performance issues
3:24PM 3 Call Back
2:06PM 0 Modem driver question
1:14PM 7 Correction regarding price of Grandstream Budgetone 100 series phone
12:29PM 8 How much to use Dialogic?
12:19PM 0 working with SIP soft phones
11:54AM 0 iLBC, Speex and X-Lite
11:02AM 2 OH323 crashing
11:02AM 1 Question for someone running hylafax off *.
11:01AM 1 Adding an app
10:31AM 0 Premisys channel bank
10:03AM 0 Do not try the Queue App Patch
9:52AM 2 Underwater in 10 - 20 seconds
7:55AM 8 Packet8 VOIP service now 1/2 the price
5:12AM 2 ADSI
 
Sunday June 8 2003
TimeRepliesSubject
11:59PM 1 oh323 and extentions.conf
9:36PM 1 anyone seen this error when running asterisk!
7:33PM 1 FWD<-> *
5:19PM 4 zapata.conf and zaptel.conf
1:07PM 3 busydetect and X100P hangups
9:21AM 1 Asterisk, ATA186 and callerid
9:20AM 19 VoIP Provider
 
Saturday June 7 2003
TimeRepliesSubject
5:19PM 7 Another PRI based question
5:14PM 9 PRI questions
3:33PM 3 Wish-to-have in Asterisk
1:00PM 4 Bandwidth measurement tool: bmtools
12:51PM 0 TDMxx weirdness with 2-line SBC portable phone
11:28AM 0 hardware supported
7:49AM 10 SIP, NAT & Asterisk
6:30AM 0 New cdr_mysql.c
5:46AM 2 VON in London..
5:27AM 1 Queue App Patch, addendum
5:20AM 1 Queue App Patch
4:20AM 0 Newbie question on soft phones with SIP
 
Friday June 6 2003
TimeRepliesSubject
9:04PM 8 install asterisk without FXO PCI or modem? Is it possible! TXT FILE NOW!
7:36PM 0 install asterisk without FXO PCI? is it possible!
4:10PM 1 Where to order the Budgetone 100 phone online?
2:56PM 0 P0S30200.bin file for Cisco 7960 phones
2:09PM 5 small office
2:07PM 0 Colorado Asterisk Users
12:27PM 9 Receptionist phone
10:01AM 0 CallerId CheckSum Errors on x100??
8:17AM 1 more about SIP ...
5:17AM 5 Newbie question on soft phones with SIP and *
5:05AM 5 SIP codecs
2:08AM 0 sendmail invocation in voicemail and voicemail2 applications
 
Thursday June 5 2003
TimeRepliesSubject
8:57PM 0 problems after adding FXS module to TDM400P
1:21PM 7 Asterisk Documentation
1:05PM 7 email notification not working anymore
10:30AM 5 answering calls with SIP phones
7:12AM 2 T100P + Capi + Caller ID issue
6:15AM 1 dl102s again
1:33AM 0 dl102S
12:28AM 3 Valiant Comms VCL 30 Channel bank + Digium E100P
 
Wednesday June 4 2003
TimeRepliesSubject
7:30PM 2 AgentLogin
6:44PM 1 detecting pickup
4:16PM 1 new application Dialtone()
1:14PM 7 RTP codec error???
1:04PM 1 Newbie help getting started
12:31PM 0 Call Waiting not detected
11:49AM 1 Inside .vs. Outside Rings
9:54AM 0 Anyone know about callerid format used by NTL in Cambridge?
9:53AM 1 Maybe a Rehash Call Queues
8:47AM 0 Use of Vina Technologies Integrator 300
8:37AM 2 Eicon and capi
8:19AM 9 h323 and g729
4:48AM 14 Budgettone 100 phone Configuration
4:45AM 12 Getting netmeeting to work with Asterisk
2:46AM 4 chan_capi with avm c2 only uses one BRI
1:25AM 1 7940 SIP upgrade
 
Tuesday June 3 2003
TimeRepliesSubject
11:25PM 2 Clock Sync
9:17PM 16 Modem => Serial ?
7:22PM 1 ata186 and 9 for outgoing line type dialplans
7:02PM 6 Fixed Cellular adapters/terminals
3:50PM 0 Asterisk localization
1:36PM 0 Initial Connection Hangup (T100P) and Ringing Failure
12:12PM 0 Sound: Recording overrun
12:02PM 1 Example of the Transfer application?
11:11AM 3 Asterisk Works on Linux on Sparc
10:17AM 2 Detect hangup on unanswered POTS call
9:27AM 0 ADSI - BT Easicom 1000
8:35AM 0 Few questions from new user...
7:14AM 0 Is there a way to play audio to the callee?
6:10AM 2 Call Parking on 7960
5:08AM 10 Cisco 7905G phone
3:52AM 8 E400P
3:07AM 0 wav49 problem
2:12AM 0 Asterisk terminates unexpectedly with SIP call and G.723 codec
2:03AM 0 SIP default gateway
 
Monday June 2 2003
TimeRepliesSubject
11:08PM 1 Two * Questions
10:57PM 1 (no subject)
10:36PM 2 Conferencing : authentication
8:06PM 0 Unable to start D-Channel
4:02PM 2 Announcing IAXCLIENT v0.02 A cross-platform IAX client.
3:06PM 1 script in perl or PGSQL to predictive dialing - progesive dialing.
1:46PM 6 Dinosaur *
1:11PM 4 Net2Phone SIP
11:11AM 5 MP3Player
10:38AM 0 ArrayVox problems
9:20AM 2 Performance on VoIP
8:28AM 1 G.711 Codec
8:24AM 0 SIP, DTMF, and AudioCodes Mediant 2k
8:01AM 8 Limit Concurrent IAX Channels
6:29AM 3 Configuring spans
6:11AM 1 Does anyone know how to get rid of this warning message?
3:45AM 0 E400P cable
2:50AM 2 enum.conf file
1:39AM 0 What they are used for?
1:01AM 0 asterisk with vocal
 
Sunday June 1 2003
TimeRepliesSubject
9:58PM 12 Voice Modem + Soundcard Driver
9:09PM 10 Call Transfer Problem
4:32PM 1 Fax support over MGCP, H323 and SIP in asterisk
4:16PM 5 Zapata 3.3v PCI version...
3:33PM 8 ISDN4Linux + Asterisk and Europe
1:49PM 3 Any plans for a .....