asterisk users - Dec 2003

Wednesday December 31 2003
TimeRepliesSubject
1:37PM 33 New to asterisk? RUN... don't walk.
12:42PM 1 Asterisk Web Dialer
12:29PM 0 Audiocode MP108
12:24PM 2 AGI - IVR - Time Clock
12:19PM 5 Java?
11:54AM 11 after hours - is this logic ok ?
9:25AM 3 Snom 200 with two extns defined anyone?
8:50AM 5 Anyone, ideas for incoming call management for CRM system
8:47AM 0 Current database abstraction effort ?
8:26AM 0 grand stream phone and double nat
5:51AM 0 ast gui client error
1:31AM 14 Happy New Year!!
 
Tuesday December 30 2003
TimeRepliesSubject
10:47PM 5 A Head Check
10:47PM 0 Consultancy on Asterisk !!
9:58PM 1 Accountcodes
5:08PM 0 Re: +AFs-Asterisk-Users+AF0- RE: +AFs-Asterisk-Users+AF0- Multi-line, multi-registration phones
4:41PM 1 TDM400P related question
4:36PM 2 playback in [macro-stdexten] problem
4:21PM 6 SIP phone as intercom
4:07PM 3 * crash when forward voicemail message [problem solved]
4:00PM 0 RE: +AFs-Asterisk-Users+AF0- Multi-line, multi-registration phones
2:58PM 0 Quick question on source cvs files
2:31PM 2 7960 Register with 2 * Servers causes phone to reboot over and over
2:08PM 0 Mac OS X
1:59PM 0 X100p always busy - update
1:48PM 21 Programming an unlocked ADSI phone?
1:29PM 1 Multi-line, multi-registration phones
11:49AM 5 Backup Proxy & Automatic Failover
7:12AM 1 Ser and Arterisk works together ?
4:49AM 3 E100P configuration
3:13AM 1 SIP + DTMF problem
3:04AM 1 Routing calls from a T1 based on DNSI.
1:36AM 1 automatic voice dialout call
 
Monday December 29 2003
TimeRepliesSubject
11:36PM 13 Does Asterisk support legacy Dialogic products?
7:03PM 15 CVS Closed?
6:33PM 3 fedora core 1 install problem
3:25PM 0 H.323, MultiVOIP, and DTMF
2:10PM 14 include a file ?
2:05PM 0 Terminiation in Germany or Netherlands?
12:50PM 3 Agent setup
12:40PM 1 Anyone having problems Logging in to Voice Pulse in Iax.conf
12:16PM 8 after hours logic
12:05PM 5 Virtual PC -- Asterisk ?
11:54AM 3 bandwidth requirement
11:10AM 0 FW: Weirdness with CALLERID / CALLERIDNAME from incoming PRI
10:45AM 0 E100P - rxgain / txgain
8:25AM 2 E100P pinouts anyone?
7:26AM 10 asterisk crash
4:51AM 9 transfer with MGCP
 
Sunday December 28 2003
TimeRepliesSubject
9:11PM 0 Is there something wrong with "show manager commands"?
6:48PM 22 TDM Card loses Dialtone and Battery
4:25PM 0 DTMF Error
3:30PM 0 Digium Wildcat E100 card mechanics issue
2:40PM 19 Echo, ISDN And FXS
2:07PM 4 outcall notification
12:56PM 4 Speex Codec - Error IAX2
10:55AM 2 SHARING DIGITAL CONTENT
4:03AM 0 Dianatel Central Office 24 Channel Bank
 
Saturday December 27 2003
TimeRepliesSubject
10:29PM 2 Multiple mpg123 processes when starting asterisk
8:30PM 3 Re: Asterisk-Users digest, Vol 1 #2282 - 15 msgs
4:58PM 1 Dual Athlon 2.4 MP & *
4:16PM 0 Help - after last night update unable to make outbound calls from GS
2:30PM 14 frame buffering
1:16PM 5 mysql cdrs
12:56PM 1 Outgoing call with bad/choppy sound
11:32AM 6 Help with x101P
7:41AM 6 Vocera Communication Badge
2:20AM 3 Setting up asterisk on Rh 9
 
Friday December 26 2003
TimeRepliesSubject
10:46PM 5 Incoming callers aren't hearing ring
6:50PM 1 clicking and popping using T100P resolved
5:58PM 1 what is ztcfg for
5:40PM 0 Re: time to build an open phone?
12:42PM 5 fax detection: false positive
12:26PM 3 Re: time to build an open phone?
11:56AM 2 Polycom Sip Registration
11:38AM 0 fwd problem with *
8:59AM 2 Festival Time and Temperature application
7:05AM 2 Incoming call on LineJack's LINE/FXO is not answered by *
4:58AM 10 Problems with outgoing calls
12:13AM 2 DevKitLite compiles but won't load modules or run asterisk
 
Thursday December 25 2003
TimeRepliesSubject
8:28PM 7 Red Alarm on X100P
2:21PM 1 Calling from * to fwd
1:34PM 0 can't get oss console working.
1:26PM 1 X101P stopped working. Newbie seeks help
11:58AM 2 IAX NOTICE and WARNING messages
6:44AM 6 call pickup via *8 from ata186 (SIP)
5:11AM 6 return of the transfer to a busy number
 
Wednesday December 24 2003
TimeRepliesSubject
10:00PM 0 Merry IAXmas
6:11PM 39 Encryption
4:04PM 17 G729 troubles
2:05PM 0 registration problem
1:04PM 1 Reversing a Firmware Upgrade
12:53PM 7 Unlocking Vonage ATA 186
12:36PM 0 Merry Christmas and Happy New Year from XVOIP
12:21PM 0 chan_skinny Feature set & Development
12:06PM 8 Sip phones on the same extension?
11:56AM 0 Fax capabilities of various services
11:08AM 0 Grandstream budgetTone registration time out
10:21AM 0 Grandstream 102 flashing display
9:41AM 6 CT1 and callerid / DNIS
9:36AM 0 amaflags question
9:19AM 3 Weirdness with CALLERID / CALLERIDNAME from incoming PRI
6:40AM 1 Fw: FAX detection Problem
6:12AM 7 FWD problems
3:49AM 0 offtopic: possible asterisk meeting saturday amsterdam
3:28AM 0 OT: FWD Holiday Promotion: Free Calling to 8 Countries
1:44AM 1 caninvite...
12:16AM 2 when * start at bootup chan_h323 fails to load
 
Tuesday December 23 2003
TimeRepliesSubject
10:43PM 51 Grandstream Quality Survey.... :P
9:42PM 0 DTMF A,B,C and D
7:18PM 3 Merry Christmas!
7:13PM 0 Outdialing with Voicetronix
6:22PM 4 CT1 and callerid
4:50PM 0 Voiceglo SIP configuration
4:37PM 0 configuration files for cisco 7960
4:07PM 0 SIP / FXS - MOH
4:04PM 6 Merry Christmas, all Asterisk users!
3:42PM 0 Packet8 Minus the DTA
3:34PM 2 Cisco 7960 Sounds patchy.
3:19PM 0 Conf file system generation in * for User/Admin update
1:57PM 0 Fw: Fw: Questions and finding
11:54AM 4 Cisco 7960 phones.
11:44AM 2 Fw: perl database get
10:51AM 4 OT: SIP vs. Skinny protocol
10:28AM 11 WEBMIN module for Asterisk
10:13AM 1 turning off IAX registration attempts
9:48AM 1 Re: Asterisk , Video Switching
8:53AM 8 Capi Dial & outgoing msn?
8:48AM 6 PBX Functionality How-to
8:23AM 1 sendmail problems
8:03AM 2 gnophone transfer
7:06AM 1 codes/grandstream/PRI.. few questions :)
5:30AM 6 Cisco 7914 Expansion Unit (for 7960G IP Phone) & Help With 7960's Speed-dials
5:18AM 8 Auto Starting Asterisk
2:57AM 3 Asterisk + CRM
2:52AM 3 Problem - installing TDM400P module
1:27AM 1 abt asterisk
1:15AM 4 Video
 
Monday December 22 2003
TimeRepliesSubject
11:45PM 0 Wishes
9:31PM 1 Authentication
9:15PM 4 MSN to GS - Call drops in 10 secs
8:37PM 0 IAX trunking recomendations
6:46PM 0 What is the bandwidth requirement for IAX
3:00PM 3 Sweet video phone
2:39PM 5 DID trunks -- equipment requirement
2:23PM 3 tor2 does not load
1:15PM 21 Asterisk SIP Packet Time (20ms)
1:01PM 2 Fw: Questions and finding
12:42PM 4 Asterisk as a PSTN gateway for SER
11:10AM 3 Sipura 2000 configuration.
10:52AM 0 Festival sounds like a steam engine
10:10AM 0 Setting audio gain for SIP extensions?
9:54AM 1 ISDN-PRI - WCT1XXP error
8:50AM 4 Audio format for announcements
7:26AM 1 windows messenger and DTMF
6:36AM 17 call files
3:59AM 0 ActiveX DIAX demo available online
2:50AM 1 asterisk with a third party gateway
12:28AM 8 Compile Problem
 
Sunday December 21 2003
TimeRepliesSubject
7:54PM 1 iconnect / asterisk ? calls hang up
5:12PM 9 MSN messenger and *
4:44PM 9 large implementation
3:43PM 0 FWD / Timed out
2:46PM 7 First version of the ActiveX version of DIAX (0.1.0) available for download
1:26PM 2 486 Busy message - SNOM 200
1:19PM 5 Dialing dead SIP peers give misleading (BUSY) voicemail result ...
12:21PM 0 Nortel IP phones?
11:06AM 0 Strange * & ATA behaviour when directly connected
10:34AM 7 ToIP (TDD over IP)
9:47AM 0 FWD via IAXtel?
2:00AM 3 IVR Configuration in *
1:21AM 2 SJphone, Asterisk and DTMF tones ...
12:10AM 9 SS7 API & Card Solution
 
Saturday December 20 2003
TimeRepliesSubject
8:06PM 14 ivr key press?
7:41PM 5 BYEXTENSION and DBPut
4:16PM 1 Cisco 7912 speed dials
3:57PM 16 s, h, t, etc, extensions?
2:44PM 0 Channelized T3 card for linux
1:54PM 2 asterisk on beowulf cluster
1:23PM 0 X101P + TDM400P
12:53PM 4 Level(3) SIP termination services
11:00AM 2 More beginner questions
9:13AM 3 iconnect 480 unavailable msgs
7:37AM 11 IVR sample config?
4:42AM 0 Chan_h323 & gnugk
3:22AM 2 Asterisk MGCP register
2:02AM 3 ZTMonitor - /dev/dsp problem
1:47AM 0 Chan_h323 docs
 
Friday December 19 2003
TimeRepliesSubject
8:09PM 0 SIP - Ringback
8:09PM 0 Asterisk and Zaptel Load on Startup
7:53PM 1 E100P connected to Cisco
7:12PM 0 Level(3) SIP termination services?
5:57PM 0 Excellent service from vendor
4:33PM 10 DIAX phone busy
3:39PM 4 Sip registration change!
2:06PM 4 911 settings.
1:27PM 0 PRIMARY=wcfxo?
12:08PM 3 GotoIfTime help
9:42AM 0 E100P errors with PRI D-channel problem
8:58AM 0 Have HandyTone instock, ready to ship?
7:48AM 0 chan_capi documentation
7:29AM 0 Huge traffic with iaxtel.com (without making calls)!!!
6:06AM 5 Asterisk to H.323 without gatekeeper
4:27AM 0 MGCP call waiting disable?
4:08AM 1 Interconnecting Panasonic KX-TD1232 digital PBX and *
4:01AM 5 nat router + sip phone adaptor (+adsl modem)
 
Thursday December 18 2003
TimeRepliesSubject
10:00PM 0 * crash when forward voicemail message
9:35PM 0 Where can I get some cisco 7960s?
9:18PM 20 asterisk behind NAT
8:43PM 1 What is/isn't affected by "reload"?
8:42PM 5 SIP / X-ten Softphone
7:57PM 4 x100P incoming
7:56PM 0 Re: Sphinx (Karl Putland)
7:26PM 0 Request for offers
7:22PM 0 (no subject)
6:16PM 6 IAX quesitons please.
5:53PM 0 * and Cisco 7910
5:36PM 2 Cisco 7960 - can't traverse NAT?
5:29PM 9 Land line vs. VoIP provider.
5:19PM 1 FYI: ATA-286 Now supports CallerID.
4:48PM 0 TDM40B Port to Grandstream ATA-286 hypersensitive microphone issue.
4:19PM 10 asterisk and nat
3:40PM 14 Sphinx
3:28PM 1 SIP Inuse Count Wrong
3:02PM 19 Headless Linux system for Asterisk
2:56PM 2 3-way calling bug
1:59PM 1 Excessive VNAK's and jitter over IAX2
1:08PM 1 Interesting problem
12:31PM 2 Different Dial tones for internal and external.
11:59AM 8 G729 question
9:30AM 2 Polycom phones update
9:10AM 0 /var/spool/asterisk/outgoing -- call joining ?/
8:26AM 1 Where is D channel in a PRI link?
7:52AM 3 Zaprtc compile error - virtual device for conferencing
7:48AM 2 AGI and broken pipe
5:10AM 1 Configuring DG-104s
5:08AM 8 Expressions
4:34AM 0 CAPI Calls Don't Bridge
1:26AM 11 after hours
12:49AM 4 Unable to detect process 256 frames
 
Wednesday December 17 2003
TimeRepliesSubject
10:08PM 0 Zhone Zplex Hack & Diddling the Battery
9:40PM 0 Share a line with a modem?
9:13PM 3 Any Ideas
8:44PM 2 voip equipment in australia
7:56PM 4 another
7:55PM 0 g729 error - WARNING[1074433504]:
6:28PM 4 (no subject)
6:09PM 31 Grandstream Early Dial
5:44PM 3 Trunk Groups and Multiple Asterisk Machines
4:50PM 0 issue recording files in wav49 from AGI
4:33PM 0 local voip users hanging up
3:36PM 0 CVS and Releases
3:33PM 1 gateway VoIP h323
2:52PM 0 announcements
2:44PM 1 Paris trip delay
2:37PM 0 X100 (when the phone line is not functional) and TDM400
2:16PM 3 Dial plan
1:54PM 3 VoIP Dropouts
1:45PM 0 DIAX 0.9.6d with IAX2 debug support
1:43PM 10 ALL incoming Zap channel calls are getting picked up as FAX calls!
1:14PM 0 Patch to fix vmail.cgi forwarding problem
11:42AM 2 Troubles with voicemail and cisco 7905 SIP
11:19AM 6 Readline & readline-devel installation on RH9
10:51AM 1 PSTN to h323
10:44AM 5 SIP
9:40AM 0 asterisk as media server
9:36AM 4 modprobe -r ztd-eth locks up machine...
9:22AM 4 Polycom SIP Phone config files
9:01AM 3 Residential router w/ QoS support?
8:41AM 1 PRI Error messages
8:40AM 0 h323.conf new try
8:23AM 1 Seting callerID on outgoing calls
7:48AM 24 128 kbs satelite link
7:03AM 1 ASTERISK X SER
5:47AM 2 asterisk phone card application with agi
4:15AM 0 Asterisk and Eicom BRI-2M or 4BRI-8M
1:24AM 0 Install X100P and TDM400P Asterisk cards!
1:15AM 5 Probably not hard but I'm just a no0b with *
 
Tuesday December 16 2003
TimeRepliesSubject
10:57PM 2 Unable to Receive Fax -- RxFAX Application
9:51PM 0 RE: Help! VoiceTronix Multi FXO/FXS Problem (Jacky)
8:38PM 0 Flash Transfer/Voicemail Bug
6:29PM 0 Transcoding CPU usage: surveys?
6:01PM 1 DISA - Zap/DTMF Problem
5:56PM 7 IAX2 using non standard port
5:46PM 7 AT&T access code entry by Asterisk
5:20PM 2 Flash Problem
4:40PM 11 broken pipe - * does not respond
4:17PM 0 Gateway in proxy mode
3:22PM 0 John Brown from Chagres
2:35PM 1 asterisk - scalable ?
2:28PM 2 undefined symbol: ast_moh_stop
2:24PM 1 command Authenticate
12:31PM 0 iconnect incoming works, so what ...
12:29PM 0 Asterisk Support for QSIG
12:26PM 0 SUBSCRIBE in channel_sip.c
12:15PM 1 Cisco AT-18x SIP 3.0 Firmware
12:08PM 1 asterisk and cisco call manager via h.323
11:58AM 5 DIAX-SJPHONE REGISTRATION PROBLEM
11:47AM 1 sip registration send out by asterisk
11:08AM 58 codec negotiation
10:51AM 0 Requesting advice from experienced * users/developers
10:23AM 0 Cisco 7960 Firmware issues
9:59AM 5 Stupid Newbie Questions
9:45AM 0 Voice & Data Systems Engineer Career Opportunity
8:34AM 3 Free Software/Open Source-Telephony-Summit 2004
6:23AM 1 wiax
4:05AM 2 Help! VoiceTronix Multi FXO/FXS Problem
 
Monday December 15 2003
TimeRepliesSubject
9:37PM 0 Packet8 DTA310 Advanced Configuration
9:05PM 5 iaxclients missing calls
8:28PM 0 pyst the Python for Asterisk project
8:21PM 15 more questions
7:15PM 2 Beginner couple of questions
5:07PM 1 Zaptel BRI experimental drivers released.
4:34PM 3 AVM ISDN Fritz!Card USB works
4:01PM 1 ast-ax-snmpd release 0.3
3:13PM 2 Slightly OT and mildly insane: Modems through VoIP :-))
3:02PM 0 IAX and Voicemail
2:33PM 2 Beginners Question
2:26PM 0 Transfer and release?
2:24PM 5 Nagios/measurement with Asterisk - any plugins?
2:16PM 5 voicemail as an attachement
1:40PM 0 Voicemail messages and codecs
1:10PM 7 IP 500/600 1.1.0 Firmware
1:06PM 0 Help Needed - SNOM 200 shows "Forbidden" message
1:02PM 4 Outgoing calls for a fancy address book app
10:05AM 0 SIP Codec problem?
9:41AM 2 noises an Zap channel (TDM20B) while hdd activity
9:23AM 3 snom 200 version 2.03b with changed music on hold
9:10AM 3 voicemail.conf email notification
9:09AM 3 Norstar MICS
8:58AM 4 E400 or TE410 (digium) vs PRI 30M (Eicon)
8:48AM 2 Using asterisk as voicemail with SER
6:34AM 1 Cisco 7960, Nortel MICS, Digital sets, ...
6:18AM 16 transfer with threeway calling
5:06AM 5 Howto to test asterisk applications - VoIP Testing Solution
2:36AM 1 DIAX, chan_capi and busy tone
1:05AM 2 FWD and (multiple) internal IPs
 
Sunday December 14 2003
TimeRepliesSubject
10:49PM 2 MeetMe: Zap channels don't ever disconnect. . .
10:07PM 0 Re: FAX, IAX and *....Maybe I'm dreaming...:-) (Carl Youngblood)
9:00PM 7 can X100P detect phone pick up like an answering machine
8:59PM 1 Error loading modem driver
3:59PM 5 Cisco 7960 lockups - any experiences?
3:57PM 17 Cisco Gateway Integration
3:29PM 0 nexthop: free service to pstn via fwd/iaxtel
2:47PM 3 unable to configure my Grandstream phone
1:18PM 6 ignorepat
1:17PM 0 outbound dialing / wait for keypress?
1:00PM 18 IAX 1/2 registration
12:36PM 0 modem data calls through FXS / FXO digium cards failing
10:53AM 1 Two Stage Dialing for MF CAMA trunk
10:39AM 0 CAMA MF signaling for a 911 Trunk
2:36AM 0 Unable to call from SNOM 200 to IP 7905G
 
Saturday December 13 2003
TimeRepliesSubject
9:18PM 1 extension response
4:38PM 2 IAX Call not transferred - plz help
1:21PM 0 Voicemail notification problem
1:14PM 6 VoiceMail Password problems
12:01PM 2 Wrong voicemail after transfer?
11:19AM 1 Sipura SPA-2000 is shipping, discount for asterisk-users
7:07AM 6 new CVS Checkout
1:29AM 2 voice mail - sip:notify message
 
Friday December 12 2003
TimeRepliesSubject
9:13PM 13 Garbled VoiceMail
4:45PM 0 Translation time
2:11PM 5 RH9 and h323.conf
1:25PM 4 Simultaneous incoming calls
12:28PM 5 estara softphone problem
12:04PM 0 Hang up
11:20AM 4 SIPURA Breaches Contract
10:33AM 0 IAX Stream problem --
10:14AM 1 Chagres Technologies _WHERE IS MY ORDER?
10:01AM 1 Asterisk and Debian
9:41AM 4 chan_capi 0.3 and capiinit questions
8:59AM 2 Dlink DG-104SH
8:47AM 0 dynamically enabling PRI channels
8:43AM 0 Speex codec and X-Lite!
7:35AM 2 How to take over ringing calls
6:25AM 42 FAX, IAX and *....Maybe I'm dreaming...:-)
5:56AM 1 Streaming Hold Music
4:51AM 4 CLIP in Germany
4:26AM 0 MS Messenger RTP
3:20AM 2 Manager API Problem
2:51AM 0 FAX app again
12:05AM 3 simple question on sip.conf
 
Thursday December 11 2003
TimeRepliesSubject
11:15PM 2 restricting one user per account
9:21PM 2 Dialing area question
8:17PM 7 Dial / Ring multiple sip channels
4:42PM 4 ZT_CHANCONFIG failed on channel 2
4:25PM 0 D-channel
3:18PM 19 Queue only ringing one agent at a time
1:30PM 0 Help: codecs and bandwidth
1:17PM 0 Asterisk freezes, no manager traffic, console functions
12:53PM 0 Problem with R2 signaling
12:34PM 3 SIP response 403 "That is ugly"
11:32AM 1 one unified
11:03AM 2 SIP retries
10:31AM 0 problems with Background application
10:12AM 5 Yuck! Error in buffer handling
9:47AM 1 Issues getting a service contract from cisco.
9:27AM 0 Interesting quote about Asterisk
9:27AM 9 Re: * with RADIUS
9:03AM 4 FAX application does not work for me....
7:31AM 0 FW: Iax, Iax2 and Iaxcomm
6:55AM 3 How to return a transfered call
6:50AM 0 g729 and asterisk upgrade
6:18AM 4 Asterisk in a Centrex environment?
6:01AM 0 CONFIG_ZAPATA_NET patch for 2.4.21
5:10AM 3 Very small office install
5:01AM 5 Iax, Iax2 and Iaxcomm
4:01AM 0 Error setting up USB FXS
1:26AM 1 New Installation problems
1:23AM 11 * CVS checkout does not work on one box
12:07AM 6 Re: * with RADIUS
 
Wednesday December 10 2003
TimeRepliesSubject
10:52PM 0 A solution to "free line" notification
10:15PM 2 RTP Codec Error(s) - Is there really a solution for this or these?
9:00PM 0 G.729
6:42PM 0 External Email Notification -2
3:56PM 5 pridump
3:30PM 1 chan_sip.c update to 1.253
3:04PM 5 Sipura SPA2000 & Asterisk & latest firmware (1.0.18)
3:00PM 1 sip.conf and Codecs
2:59PM 0 Native Bridging and Polycom 600 Solved
1:49PM 1 Article on Asterisk: German Linux magazine
12:31PM 0 Search engine
11:47AM 37 Computing horsepower needed
11:18AM 1 Errors after re-plugging T1
10:47AM 15 next stable release?
10:45AM 1 WAV file volume
10:08AM 1 unknown RTP codec 19
10:05AM 3 app_queue bug with call transfer
8:47AM 5 IAX and PDAs
8:28AM 0 Trouble with AGI and SAY DIGITS and WAIT FOR DIGIT using PHP
7:16AM 1 Transfert with IAX
3:05AM 2 DIAX 0.9.6b - available for download - multilingual issue on WinME solved
 
Tuesday December 9 2003
TimeRepliesSubject
10:16PM 12 Telemarketer Torture
9:47PM 7 Sendmail not on localhost
9:45PM 2 On Hold - Talked about before
9:09PM 0 Need advice with "free line" notification
7:48PM 7 was FXO cards
7:06PM 5 SIP "PUBLISH" and "SUBSCRIBE" extensions?
6:25PM 0 Zhone Configuration with T100P
5:04PM 2 Outbound iax dialing to one #
4:34PM 2 Need help with jitter buffer/quality settings
3:29PM 7 voip-info.org DNS seems broken
2:35PM 1 vmail.cgi give Login incorrect error
2:09PM 0 InterStar- can it be used with Asterisk?
1:31PM 0 voicemail minmessage
1:08PM 0 D-channel down
1:06PM 2 Erratic DTMF on E1/PRI (continuation of Strage bip on ISDN/PRI)
12:53PM 0 Asterisk GUI Client Updated
12:34PM 8 IpDialog phone issues.
12:00PM 1 call-waiting caller-id
11:49AM 0 High residual CPU usage after hangup
10:58AM 1 dialling peer problems
10:20AM 3 Strage bip on ISDN/PRI
8:13AM 0 Cisco 7960 Directory - limit?
8:06AM 0 Voicemail2 and outgoint announcement
7:55AM 3 Web Interface for CDRs
7:49AM 8 BT launches consumer VoIP product ...
7:13AM 9 Multilanguage support
5:59AM 0 Weird SIP registration warnings in log
5:43AM 0 IAX termination in the Netherlands
4:18AM 2 Cheap Learning Environment
12:33AM 4 multiple IAX registrys from the same client
 
Monday December 8 2003
TimeRepliesSubject
10:58PM 1 Re: Asterisk-Users digest, Vol 1 #2120 - 14 msgs
10:27PM 2 Call logging In and Out
10:01PM 8 (no subject)
9:35PM 2 chan_h323 readme file
8:57PM 1 www.terracall.com
7:02PM 0 AW: "Phone Unprovisioned" Message in IP 7940 ?
3:39PM 0 problem with gsm codec
3:19PM 2 'asterisk' as caller id
2:47PM 2 New to Asterisk need help with caller id
2:47PM 18 Multiple Asterisk servers sharing/propagating registry ?
2:32PM 4 Problems with voicepulse.com
2:31PM 4 Strange variable chopping from AGI's
2:17PM 8 SIP (peer to peer?)
1:23PM 2 snom X MOH
10:06AM 0 MGCP caller id problems
10:05AM 4 IAX error messages in log
7:38AM 0 Job Opportunities
6:29AM 16 IAX clients
6:18AM 0 ATTN Developers : Features Required on payment
1:49AM 3 DIAX to DIAX call and disconnecting after 50-60 sec.
 
Sunday December 7 2003
TimeRepliesSubject
10:41PM 0 Adtran 750
9:48PM 8 Prefix the * character
7:21PM 3 "Phone Unprovisioned" Message in IP 7940 ?
5:55PM 2 Roaming Users
5:53PM 2 Incoming IAX2 problems with NuFone
4:58PM 2 Call does not terminate correctly
2:50AM 2 Vonage sending Motorola gear now?
1:43AM 0 Off topic: suggestion for call center ?
1:08AM 10 RxFAX application
12:31AM 3 FARFON lives!
12:00AM 0 Diconnectiong after 15s when calling DIAX to DIAX (Tony?)
 
Saturday December 6 2003
TimeRepliesSubject
8:34PM 1 Asterisk Maint.
4:55PM 4 Project Critique
2:41PM 4 unixODBCget/put/del/deltree
1:42PM 2 S100U and *
12:27PM 0 Not Handling NAT correctly
11:13AM 2 Caller ID from Database
11:12AM 0 Problems with asterisk
11:03AM 0 Fw: snom in Wallstreet report
11:01AM 3 H.323 Phone w/ Asteisk
10:10AM 4 console sound
7:06AM 15 some success with linux 2.6 and wcfxo
5:45AM 15 IaxTel seems down
4:50AM 3 CallWaiting CallerID
 
Friday December 5 2003
TimeRepliesSubject
10:14PM 5 MGCP IADs
10:02PM 0 Budgetone phones
6:49PM 0 Please remove this person
6:40PM 4 asterisk codec sizes, data plus overhead
6:19PM 4 USB - FXS for Windows....
1:52PM 7 X100P echo problems - seem to be fixed now
1:14PM 2 Help with setup IpDialog Sip Phones.
10:53AM 0 Native bridging with Polycom 600
8:42AM 20 GrandStream Budgetone Phone & DHCP & General Observations
8:32AM 0 Init.d script was: Operating environment for *
7:53AM 0 BETA: Asterisk Search engine
6:45AM 2 incomminglimit on Xlite not working ...
6:22AM 12 grandstream budgeTone phones or Asterisk ??
4:51AM 0 CDR analysis and Lotus Notes
2:59AM 13 DIAX 0.9.6 now available- some fixes included
2:27AM 8 FAX connected to a TDM400 card port
 
Thursday December 4 2003
TimeRepliesSubject
10:59PM 4 Debian Testing / 2.4.22 / zaptel problems.
9:33PM 31 Asterisk freezing HELP
9:28PM 2 voip-info.org is a great Resource ..BUT
8:16PM 1 Implementing a ringback test function for Zap channels
6:09PM 7 x100p/hangup detection issues?
5:43PM 14 XBOX as and * Dedicated Server
5:26PM 19 Channelbank Recomendation and GS102 question
4:02PM 3 Asterisk and Avaya IP phones
3:41PM 2 Needed - Asterisk Consulting
2:29PM 3 Operating environment for *
2:11PM 0 Another audio file
2:05PM 0 Male voice over work
1:20PM 3 correct way for cvs update?
1:06PM 47 Port density: DS3 cards?
10:49AM 9 Carrier Access Channel Bank Setup -- No hangup
10:35AM 0 The power or *
10:02AM 13 vmail.cgi with Redhat 9.0
8:30AM 4 is there any way to search the mailing list archive and order results by date?
8:27AM 0 how can i play a sound file over the paging system when the phone rings?
8:20AM 8 Experiences with Fedora 1
7:55AM 0 Paradyne Jet Fusion
12:21AM 0 Asterisk and G.729a
 
Wednesday December 3 2003
TimeRepliesSubject
10:01PM 2 Outbound SIP Call
7:26PM 2 Soundblaster
7:00PM 3 Replicating Legacy Phone Behavior
5:33PM 6 How to set the gatekeeper? help me pls.
5:23PM 4 Echo problem on conferencing....no analog interfaces
5:12PM 5 OpenENUM
3:43PM 1 Echo cancel in MeetMe?
11:59AM 4 Forwarding a call to another FXO port
11:52AM 2 DIAX 0.9.5 and some resolutions for the displaty
10:55AM 2 Re-routing of existing calls
10:06AM 1 Cisco and Asterisk 2621
10:02AM 0 Implement missing features in Meetme application
9:26AM 1 Any updates on the Cisco 7920 and SIP?
8:42AM 2 COnfiguring an * system for a non-profit organization
8:23AM 2 Cisco IAD with MGCP
8:01AM 0 show application monitor command
7:04AM 5 New Multilingual DIAX (0.9.5) available for download
6:15AM 0 Kerio SIPPS problems -please help!!!
5:46AM 3 "oh323 calling party number"
5:34AM 1 More voicemodem
5:06AM 0 Bug in MGCP using host=dynamic
4:27AM 1 Transfer via # on Grandstream not always working
4:18AM 3 app_queue different behaviour
2:07AM 4 unable to make it work with MSN Messenger
1:56AM 1 Unable to check my voice mails
1:49AM 0 BOOM! Crash when trying to use SIPDtmfMode on an outgoing call!
1:12AM 2 Asterisk with Voicetronix OpenLine4 card
12:30AM 1 SMS over PRI/E1?
 
Tuesday December 2 2003
TimeRepliesSubject
7:00PM 11 Nortel i2004
6:38PM 0 Play sound to callee
6:34PM 1 Proper use of echotraining=yes
6:22PM 1 CallerId in Voicemail message announcement??
4:46PM 0 iax name resolver
4:26PM 0 How to restart * thru phone "when convenient "
3:50PM 9 Does Asterisk overwrite any libraries?
2:39PM 0 Configuring new system for a non-profitorganization
2:35PM 3 remove me
2:32PM 2 Strange Behavior!
2:26PM 22 John Brown from Chagres!
1:31PM 0 'Stop Now', 'Restart' problems
12:02PM 2 How do you differentiate Busy and Congestion on Dialing PRI
11:53AM 1 ArtDio equipment, anyone tested?
11:42AM 2 G.723.1
11:14AM 1 SIP behind NAT: NAT'ted end has to talk first?
11:14AM 12 Iax Client Library Issues? (DIAX, iaxComm, etc.)
10:44AM 6 IAXTEL configuration for new iaxtel users.
10:36AM 4 Configuring new system for a non-profit organization
10:10AM 0 2 T100P Problem. Broken Pipe
9:32AM 3 IAX port numbers?
9:16AM 4 incominglimit stuck in app_queue
8:04AM 0 Echo: X100P vs. Cisco FXO cards
7:01AM 0 xten to asterisk from outside
6:44AM 6 How to restart * thru phone "when convenient"
6:32AM 0 app_queue and CDR
6:05AM 5 maximum retries exceeded
5:58AM 5 Dedicated * voicemail server
5:02AM 2 Multilingual version of DIAX
4:58AM 9 Meetme Recording
4:28AM 0 Recieving Digits Send by SendDTMF
4:06AM 0 Re: Asterisk European Tour: was RE: * Party in Paris
2:21AM 0 Core voice prompts in french ?
1:38AM 16 CTI/TAPI
 
Monday December 1 2003
TimeRepliesSubject
6:21PM 3 PRI maintenance commands
6:21PM 2 Configuring CISCO IP 7940 for *
4:22PM 5 Tone Detection Problem
3:34PM 14 VoiceGlo
1:51PM 0 Fw: Kerio SIPPS problems
12:57PM 1 Consultant / integrator needed
11:46AM 0 T400P and 2.4.23 kernels
11:35AM 1 Message Waiting Indicator Bugs?
11:19AM 18 PREPAID APPLECATION
10:22AM 0 How do I get caller's number in oh323 ?
9:50AM 7 Call Announcement - How To ...
9:48AM 1 WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
9:35AM 1 Destination number
9:26AM 1 Asterisk Lists (was Re: Asterisk Business discussion again)
8:53AM 1 Re: Asterisk European Tour: was RE: * Party in Paris
8:48AM 0 [offtopic] Re: Re: Asterisk European Tour: w as RE: * Party in Paris
8:46AM 4 Re: Asterisk European Tour: was RE: * Party in Paris
8:10AM 0 Re: Asterisk European Tour: was RE: * Party in Paris
7:39AM 1 Announcment while ringing
6:56AM 3 Re: Asterisk European Tour: was RE: * Party in Paris
4:00AM 0 Why * dont disconnect call.
3:52AM 3 Re: Asterisk behind NAT << How to do it. (Leif Madsen)
12:20AM 1 Another * crash