asterisk users - Aug 2005

Wednesday August 31 2005
TimeRepliesSubject
11:15PM 0 Pleiades p32mxi
11:05PM 2 Sangoma card problem with EWSD Exchange
10:48PM 0 first beta of ruby-agi is out !
10:30PM 4 Oracle Realtime Driver and CDR Logger
9:25PM 1 Softphone vmail indicator and TDM400P woes
8:50PM 1 Outgoing Context being mistake for dialplan?
8:38PM 0 DIDs MO DE NY + 8xx#
8:14PM 7 Asterisk Queues and Strategies
7:39PM 0 SIP Registration resets
7:36PM 1 Asterisk@Home: How to changed AMP User Login and Password
6:46PM 4 One way echo canceling?
6:45PM 2 TDM04b and echo
5:25PM 4 How to shorten ringing stop detection on X101P clone?
5:14PM 1 How to speed-up dialnig with X101P clone modem?
4:12PM 9 VoipBuster with astersisk?
3:49PM 1 Call Pickup with Dialog on snom display
2:57PM 3 Open source firmware on an ATA
2:39PM 10 /etc/init.d/asterisk barfing
2:36PM 0 Unprovoked hangups
1:13PM 1 problems with dialing-out with Zap
1:03PM 8 Asterisk for Voicemail Server
12:28PM 1 RE: Is the 2.6 Linux kernel ready for produc tion * environment
12:23PM 0 Asterisk -> Sipura SPA3000 peer behind NAT
12:00PM 2 Need Local HELP!!!
11:59AM 0 Unicall X reload
11:06AM 0 1.2beta and PRI and CDR Corruption
11:03AM 0 webcast
10:42AM 17 RE: Is the 2.6 Linux kernel ready for production * environment
10:35AM 1 Asterisk and eicon diva server 2M as FXO
10:28AM 3 odbc realtime update problem
9:05AM 2 PRI Identity Crisis
8:54AM 1 CallerID Num and Name setting to Asterisk.. Problem
8:44AM 2 Howto disable adsi in app_voicemail.c so I can noload *adsi*.so
8:29AM 0 Uniden UIP200 and Call Queue
7:31AM 1 astcc number not answering
7:22AM 6 detecting extensions in use
7:18AM 0 telextreme and *
7:00AM 4 SpanDSP rxfax TSID variable name?
6:27AM 0 Simpletelecom.com
6:26AM 0 (no subject)
6:23AM 1 Cisco 7920 and Asterisk - How well do they play together?
6:17AM 20 VoIP service recommendation
6:05AM 2 Why it says "all circuits are busy now"
6:04AM 7 why won;t my voice files play?
6:03AM 0 SIP phone status
5:00AM 0 RE: Noise on ZAP channel
2:32AM 0 voipreach.net - are they functioning
2:19AM 0 canreinvite=no being ignored?
1:51AM 3 Sipura SPA-3000 strange behaviour
 
Tuesday August 30 2005
TimeRepliesSubject
11:48PM 2 Manipulate CALLERIDNUM
11:30PM 0 canreinvite = yes with PAP2
10:57PM 0 ANNOUNCEMENT: Asterisk-Java 0.2-rc1 released
6:25PM 1 call attend to spanish
5:48PM 2 Registrar only setup
5:10PM 4 aastra 9133i DTMF tones
4:54PM 2 free open source softphone for windows
4:18PM 5 nested dial, or jump to another line to continue dialing.
3:43PM 0 OT: Monitoring Tools
1:59PM 4 Graphical Management Interface - Comments requested
1:41PM 16 Queues.conf OPTIONALURL within the Queues cmd
12:19PM 0 astcc hangup problem
12:16PM 5 ICD Features
10:48AM 8 (no subject)
9:56AM 0 How to mute DTMF in meetme?
9:30AM 0 RE: [Asterisk-Dev] voicemessages table
8:57AM 0 RE: [Asterisk-Dev] voicemessages table
8:19AM 0 mrth+manager.conf
7:47AM 6 unresolved symbol when loading ztdummy
6:19AM 2 How to use * and # as part of numberindialcommand
6:07AM 2 FAX and AGI
5:31AM 3 Wierd Problem
5:30AM 5 Realtime Queues and Agents
5:26AM 2 TE110p and E1
5:07AM 1 RE: Noise on ZAP channel
5:01AM 1 Extensions started with #
4:22AM 3 queue - ringing members in order
4:12AM 0 re: how to set the voice message as
3:45AM 0 Re: [Asterisk-Dev] voicemessages table
3:38AM 0 Re: [Asterisk-Dev] voicemessages table
2:46AM 2 Asterisk won't listen on different port
2:22AM 0 sending dtmf tones to the caller (not the called)
1:33AM 7 X100P and UK CallerID
12:13AM 0 fedora core 3 kernel source - couldsomeonethrowthe dog a bone!
 
Monday August 29 2005
TimeRepliesSubject
8:23PM 0 SS7 to IPDC
8:19PM 1 Sangoma on Telstra E1 Tx/Rx and Echo Settings
6:32PM 1 Activate/Deactivate Hardware echo cancellation on TE406/TE411 when briging
4:14PM 1 Call waiting setup/Confenencing problems in AAH
3:39PM 6 echo system command and set the results to a new variable
3:23PM 0 delay before dial on TDM04B - continued - poosibly solved
3:17PM 0 Is Asterisk integrated to NEC PBX?
1:53PM 1 TXFAX() status
1:28PM 1 Asterisk + Dualtalk
12:46PM 0 ztdummy and zttest results
12:42PM 2 delay before dial on TDM04B - continued
11:50AM 0 Internal Extensions Busy
11:35AM 0 confifiguration of Asterisk with Cisco hardware?
11:20AM 1 MSG Waiting Off
10:59AM 0 New astGUIclient version released 1.1.6
10:58AM 9 delay before dial on TDM04B
10:49AM 9 grandstream handytone 488 fxo
10:42AM 2 Moving to New Zealand
10:11AM 1 RE: Noise on ZAP channel
9:31AM 3 Asterisk Compile error - x86_64
9:28AM 0 [Announce] Web-MeetMe v1.3.3
9:15AM 1 SER NAT any additional requirement
8:43AM 1 text till answer
8:40AM 5 TDM400 and Phone does not 'ring'
8:01AM 19 Return code of txfax
7:34AM 0 plainvoip provider problem
6:17AM 3 How to use * and # as part of number indialcommand
6:04AM 5 FW: cvs update error?
6:04AM 3 Compile problem with 1.2 beta 1
6:02AM 0 static noise - follow up
2:36AM 0 Call file always redials (grrrrr)
2:31AM 0 SV: Using * in number to chose outgoing peer.
2:20AM 0 Using * in number to chose outgoing peer.
2:15AM 4 GXP-2000 presence
2:05AM 2 Register Asterisk with Gatekeeper - oh323
1:58AM 0 Asterisk truncate my FAX !!!
1:10AM 0 Digi QuadMicro ISDN adapter with asterisk?
1:09AM 0 Conference and HFC card conflict: no solution??
 
Sunday August 28 2005
TimeRepliesSubject
10:57PM 1 monitoring with mrtg
10:49PM 0 Unable to transfer external calls to MeetMeconference (re-post)
10:21PM 7 Polycom Reboot Script
8:43PM 0 GXP-2000 registration issues
7:43PM 2 Japanese ISDN BRI card for asterisk (INS64) where to start?
7:41PM 3 error messages
7:20PM 12 Asterisk 1.2.0-beta1 tarball re-released
5:35PM 0 hfc-pci/zaphfc: Asterisk hangs with signalling bri_net_ptmp but not with bri_net
4:52PM 3 Good Deal on A Good Asterisk Box?
3:06PM 2 SER + ASTERISK voicemail
2:49PM 1 How to use * and # as part of number in dialcommand
2:39PM 0 All extensions now cannot loggin!!!!
1:57PM 0 bid on this small project if you are interested.
1:20PM 10 ztdummy and Linux 2.6.13-rc7
12:46PM 2 Multiple IP's (aliases) on asterisk box?
12:16PM 5 1.2.0 Beta1
12:08PM 4 error compiling on solaris 10
11:13AM 0 way to prevent voicemail dialout/callback from 'outside'
10:56AM 8 Mplayer as replacement to mgp123 in MP3Player cmd
10:46AM 5 Multiple PCI cards
8:24AM 1 Sip pickup
6:55AM 0 Re Invite not working
6:26AM 4 T1 DSU's/Split for voice
5:25AM 2 How to configure Cisco AS5800 - Asterisk ??
5:03AM 0 SER and Asterisk authentication
4:45AM 7 Need quote for Asterisk and billing remote install
3:53AM 2 Spped Dial setup from wiki
3:45AM 1 DIALSTATUS for Originate Command
3:35AM 2 DIALSTATUS for Originate
2:17AM 1 (no subject)
1:39AM 8 Detect Dialtone
12:29AM 0 Asterisk + AstLinux testing images now available
 
Saturday August 27 2005
TimeRepliesSubject
9:55PM 5 Low handset microphone volume with Sipura SPA-841
9:23PM 3 Asterisk ISDN: Problem Setting CallerID as DID Extension Numbers.
6:54PM 0 how can I reduce delays in meetme with zap channels
6:41PM 2 gotoiftime
4:53PM 1 Nortell Release 11 and Asterisk E1
4:15PM 1 Calling PSTN lines from VOIP softphone
3:53PM 0 Passing variables across an IAX2 call
3:31PM 5 Variuos hangup codes in Manager API for failover
3:19PM 11 Problems with registration
2:41PM 4 How to use * and # as part of number in dial command
1:17PM 0 ATComm AG-468 or AG-268
11:56AM 0 chan_sip.c: stale nonce received
11:13AM 3 required packages for asterisk on FC3/FC4
9:06AM 0 TE410P Questions
8:56AM 0 Newbie :SIP ETXTN to SIP EXTN calls
8:31AM 1 SIP Registration failure
8:29AM 0 web app
8:29AM 2 bug tracker down?
7:20AM 7 Asterisk and a Meridian Nortell Release 11
7:12AM 5 storing voice messages in DB SQL
6:41AM 2 Asterisk 1.2.0 fails to hang up using SIP
3:41AM 0 ast_register_file_version in 1.2.0-beta1
3:11AM 4 dtmf not being detected from viatalk
2:50AM 1 Asterisk conection to Nextone, codec error
 
Friday August 26 2005
TimeRepliesSubject
11:48PM 1 Is LDAPget module stable enough for enterprise usage?
9:04PM 14 Satellite Broadband and VOIP
8:56PM 1 isa2004
8:25PM 4 bug tracker bug?
8:23PM 1 Fw: IAX2 Softphone Quality & Network Cards
8:16PM 2 [Announce] Pending update to Web-MeetMe
6:40PM 2 Asterisk 1.2.0-beta1 Released
6:27PM 6 Help Solving Asterisk Lockups
4:50PM 0 motorola vt1000 games
3:29PM 0 Audio Problem when zaptel modules are loaded
2:37PM 1 Re: [Asterisk-Dev] Patch 3644 - subscription states *** IMPORTANT ***
2:35PM 2 HDLC on T1
2:33PM 1 Realtime and database structure
2:24PM 0 Timing issue with call to poll() when running asterisk -rx?
2:09PM 1 AT-320 IAX & MWI?
2:03PM 2 Dial command nor progressing on Zap channels
12:57PM 0 Asterisk on VMWare 4.5, Error Ouch ... error while writing audio data
12:46PM 0 cdr_odbc in CVS-HEAD gives connect error on reload
12:19PM 0 voice modification
12:16PM 0 CVS HEAD HDLC Abort on a TE405P PRI
12:04PM 0 PCI 2.3
11:10AM 7 911 Notices
10:26AM 4 Ztmonitor values when zap channel is onhook
10:23AM 4 ignorepat not working - what might I have done?
10:16AM 6 Fedora Core 4 x86_64
10:11AM 1 When 486 ATA crashes, asterisk does not disconnect the call
10:09AM 1 Attached Voicemail does not play mac/linux
9:49AM 3 Polycom Phone advise
9:46AM 0 fedora core 3 kernel source - could someonethrowthe dog a bone!
9:34AM 0 RE: Voicetronix openline4 quality
9:26AM 0 ChanIsAvail for IAX not working again/still? AKA Redundant IAX connections not working
9:07AM 0 HooDaHek 0.4 Released
9:04AM 0 PhoneCALL version 1.0 Administrative Manual - Released
8:15AM 1 [Asterisk-Dev] SIP Benchmarking / Stress Testing
7:48AM 2 SIP Benchmarking / Stress Testing
7:45AM 1 Asterisk wiht LDAP
7:43AM 0 cvs update error?
7:25AM 2 WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type
6:54AM 8 voip-info - is it alive
6:48AM 1 realtime sip channel configuration -> insecure option
6:29AM 6 system crash
6:09AM 19 IAX2 Softphone Quality & Network Cards
5:21AM 1 bridging sip to capi, no playtones back to caller
4:56AM 2 Asterisk: Unable to read password.
3:29AM 0 SV: Maximum retries error.
1:55AM 2 About asterisk realtime
1:03AM 1 Maximum retries error.
12:54AM 0 fax codec problem
12:18AM 11 Re:TE110P EuroISDN dial out timing out
 
Thursday August 25 2005
TimeRepliesSubject
11:57PM 0 UK Caller ID with TDM400P
10:50PM 2 Caller ID ?
9:37PM 2 Tools for Remote Monitoring and User Management
7:20PM 2 no sound with red alarm?
6:39PM 0 fedora core 3 asterisk startup
6:19PM 0 Need someone to write a console application for us.
5:15PM 1 VoIP Mythbusters Help!! - NATed phone-phone connection without proxy? Possible? Yes/No
4:05PM 0 Aastra 480 CTI?
2:33PM 2 Working NFAS config w 411p anyone?
2:26PM 0 Strange Echo
2:18PM 5 Dell 2850 anyone ...
1:46PM 1 Cisco 3620 NM-HDV-T1 PRI
1:32PM 2 Optipoint 600 Cant boot - development shell active
1:04PM 2 Dial DTMF after bridging call
12:29PM 1 callerid...
10:42AM 5 where can I get low cost g723.1 liscence
10:41AM 0 Detect On-Hook on FXO port
10:39AM 0 CVS-HEAD: KB1 echo canceller -- USE IT
10:34AM 0 Can't call to cellular phones from extensions
10:29AM 0 Internal FXS to SIP problem
9:02AM 3 OT: Are you using a Lucent?
9:01AM 0 Re: [Asterisk-Dev] Patch 3644 - subscription states *** IMPORTANT ***
7:57AM 5 Sipura spa-2000 / 3000: surge protection
7:43AM 1 Loop back cable pinout 15 Pin Serial
7:24AM 0 Automated AgentCallback logon and logoff is possible
7:21AM 0 RealWorld Stats; Not achieving expected results
6:54AM 6 updating display of a hardphone based on agents logging in
3:36AM 2 Custom Application For Asterisk
3:04AM 2 TE110P EuroISDN dial out timing out.
1:39AM 2 Which Card to choose
12:32AM 1 PRI signaling experts please help
12:21AM 0 fedora core 3 kernel source - could someone throwthe dog a bone!
12:16AM 6 VoIP providers -- California, U.S.
 
Wednesday August 24 2005
TimeRepliesSubject
11:10PM 0 Distorted Sound from E1
9:39PM 0 Asterisk hint thing.... what do you do with it?
8:33PM 0 SIP trunk rollover problem
8:17PM 1 Will Echo problems EVER be solved, I'm scare d
7:04PM 0 Digium TDM400 in UK with BT Lines
6:21PM 6 SIP Registration --Giving up forever after very short network outage.
5:34PM 2 Busy number signalling
5:01PM 0 Channel ooh323c and DTMF with Call Manager
4:39PM 5 Motherboards and IRQs
3:38PM 1 dingotel - connect Asterisk to 2-way radio?
3:33PM 1 FW: [Asterisk-Users, Andrew] Will Echo problems EVER be solved, I'm scared
2:58PM 0 ANI2 AKA Info Digits not supported?
2:29PM 0 [Asterisk-Dev] Job Opening - Release Engineer
2:10PM 6 GXP 2000 Firmware 1.0.1.2
2:05PM 0 OT - Packet 8 firmware
2:04PM 0 google talk sniff
1:49PM 2 ASTCC and cdrs
1:44PM 2 Error when answering CAPI
1:43PM 0 Re: Asterisk and MWI
1:37PM 61 Will Echo problems EVER be solved, I'm scared
1:37PM 0 AEL Question
1:24PM 3 SIP Jitter Buffer on Asterisk
12:57PM 0 Music on hold configuration
12:32PM 0 ISDN MSN problems
11:51AM 0 Re: [Serusers] SER IP PBX for multiple clients
11:50AM 0 zapata.conf for a BT phone line with a TDM422P
11:47AM 4 RealTime ignoringswitch=>Realtime/context@re altime_ext
11:26AM 0 saynumber and variables
11:03AM 4 fedora core 3 kernel source - could someone throw the dog a bone!
10:54AM 0 Zaptel Not Sending Tones
10:18AM 0 Replace Aspect by using Asterisk
10:03AM 1 Yellow Alarm issues with second TE410P installed
9:44AM 1 Sql Realtime
9:44AM 8 Cisco 7960 / SIP & tftp configs
9:32AM 7 Lots of console; attach and grep?
9:22AM 2 Can exsiting router handle VoIP traffic?
9:22AM 7 chan_capi on slackware10? cannot compile :-( why?
9:17AM 0 Problems setting up X100 FXO card
8:55AM 13 AGI + Ruby
8:28AM 0 Re: [Serusers] SER IP PBX for multiple clients
8:10AM 0 AstriCon Update: Early Bird Ends Tomorrow
7:54AM 8 asterisk in Taiwan
7:48AM 0 Listening to agent's conversation while waiting in the queue
7:34AM 0 Asterisk, Maximiliano J. Goldsmid has invited you to try Google Talk.
6:44AM 1 Polycom Default .cfg
6:33AM 1 RealTime ignoringswitch=>Realtime/context@realtime_ext
6:20AM 0 (no subject)
5:05AM 0 How do I pick up a specific call from a queue?
5:05AM 1 Exec a Cmd during a dial
4:43AM 1 installing pystre
4:01AM 7 Connection TDM400P to UK PSTN
3:58AM 0 Experienced Sysadmin/Programmer having major troublewith British Telecom Caller ID & Distinctive Ring
3:45AM 1 Storing a number to Dial
3:43AM 1 TDM400P : no dial tone...
3:10AM 3 tdm04b hangup problem
2:33AM 1 [Asterisk-Dev] Cisco 7970 SCCP Configs.
1:55AM 2 Snom 360 - Message waiting and conference keys
1:46AM 0 FOP queue status
1:44AM 22 NAT and SIP.conf update.
1:07AM 0 Answer confirmation via IAX?
12:44AM 1 HOW TO SEND A MESSAGE TO A CHANNEL THAT IS RECEIVING A CALL????
12:38AM 2 SV: Fax to email using mime-contruct
12:31AM 3 Issue in calling mobiles....
12:21AM 1 Wifi UT Starcom F1000: Raising Audio volume Level via asterisk?
 
Tuesday August 23 2005
TimeRepliesSubject
11:42PM 0 Fax to email using mime-contruct
10:08PM 0 Out of Office AutoReply: yet another Asterisk an d VMware question
9:41PM 9 RealTime ignoringswitch=>Realtime/context@realti me_ext
7:55PM 3 Zyxel Prestige 2000W Firmware - EVIL
5:09PM 1 Wait before dialing ( was Pause during dialing to enter another number)
4:32PM 1 Voiceblue and slow dialling
4:27PM 3 call parking timeout
3:32PM 5 Help...... I need the TE-110P NEW ZEALAND E1 SETTINGS
3:18PM 1 Cisco 7940 + no audio after MOH
2:03PM 4 WARRNING REGARDING Support from 2n.cz !!!!
1:46PM 2 OH323 with Asterisk@home - seems incomplete
1:11PM 2 SIP powercycle not hanging up
11:57AM 1 latest CVS on Mandrake 9.2 Mini ITX
11:37AM 0 Meetme using ztdummy on Linux 2.6 sounds scratchy
11:21AM 3 Can't get G729 working after buying a license.
11:03AM 1 Asterisk 1.0.9: TE411P replacement for TE410P 1stgen causes crashes
11:03AM 0 Retreive and Play Voicemail name
11:02AM 0 Sip trunk groups, possible?
10:53AM 0 AreskiCC + Mutliple SIP Gateways for one route
10:46AM 0 Nokia PoC PTT Asterisk
10:14AM 3 AGI nor System working after a dial - Should it work?
9:47AM 9 HDLC/Zaptel/Kernel 2.6.11(.9)
9:35AM 0 Sip channel remains active indefinitely
9:29AM 0 asterisk problem with ODBC
9:18AM 3 app_sms: using * as an smsc
9:07AM 2 YAACID isn't working
8:41AM 2 FW: Register Today for Fall 2005 VON: "The Destination for IP Communications"
8:33AM 12 looking for failover ideas
8:25AM 0 Toll Call Voicemail Ring Timeout (new module????)
8:16AM 0 Embedded HW: asterisk with USB ISDN TA on NSLU2/Debian (fwd)
8:13AM 0 iax and zap interface problem
7:32AM 1 Asterisk set-up for LCR
6:58AM 0 [Asterisk-Dev] Severe ISDN signal distortion in CVS-HEAD with octoBRI
6:56AM 3 compiling CVS-HEAD + Patch from http://bugs.digium.com/view.php?id=3644
6:53AM 0 US based CLEC provider request
6:11AM 4 Echo after running for several days?
6:11AM 1 follow me configuration web page??
6:11AM 5 chan_unical-MFC/R2 CPU usage problem
5:34AM 0 FW: SIP DEADLOCK
5:00AM 0 mailing list
4:29AM 0 header intact
4:02AM 0 X100P Clone not picking up incoming calls. [POTS]
2:33AM 0 call number + tariling suffix
2:23AM 0 Faxing help
2:18AM 2 Asterisk & Alcatel PBX
1:50AM 1 asterisk+realtime
1:44AM 2 [Asterisk-Dev] q931 dial errors
12:43AM 1 Odd problem with sip.conf register command:
12:12AM 15 Music On Hold + canreinvite=yes
12:04AM 1 where is addmailbox now?
 
Monday August 22 2005
TimeRepliesSubject
9:46PM 0 Telstra/Sangoma bouncing E1 D Channel
9:27PM 0 MP3PLayer - rewind, forward, pause functions - feature request
8:04PM 1 Asterisk ISDN CallerID identification failure
7:10PM 2 SIP message re-writing and routing with Asterisk
6:50PM 1 SIP Subscriptions with SNOM
6:14PM 0 BLF for POTS lines via SIP
6:13PM 0 Dial, RING with a digit interrupt
5:24PM 1 dtmf tones
5:10PM 1 IAX2 with g729 ATA Device
4:39PM 0 TE110p Speech?
3:17PM 3 Re: MWI problems on 9133i
3:16PM 16 Small office setup/using analog lines w/ Ast erisk
3:00PM 1 txtcidname usage
2:13PM 0 Recorded sound quiet
2:10PM 5 grandstream bt100 help
1:26PM 0 HELP PANASONIC TDA200 AND ASTERISK
1:15PM 1 Hangup Faster
1:08PM 0 cisco 7960 disconnect problem
11:30AM 9 asterisk -rx (or remote connections in general)
11:20AM 0 SPA3000 dial plan?
11:12AM 1 Public Key
11:08AM 2 Shared Call and Bridged Line appearances on Polycom IP501
11:02AM 3 Make asterisk 1.0.7 fail under FC4
10:58AM 18 Pause during dialing to enter another number
10:39AM 3 Polycom 1.5.2 call waiting focus behaviour change?
10:32AM 1 Asterisk 1.0.7 won't run after upgrade to FC4
10:29AM 0 Stange behavior with g729 and DTMF
10:24AM 1 Polycom 1.5.2 firmware NTP problems
9:09AM 4 Problem with Hangups
8:38AM 1 Question on Zap interfaces
7:13AM 0 Asterisk-UK Website
7:03AM 0 new version of asteriskguru queue statistics released
6:47AM 1 Qualify time +2000ms?
6:22AM 3 TE110P problem
6:10AM 0 Does Asterisk support T1 E&M Wink/Wink voicechannels on any Digium/Sangoma hardware?
5:57AM 1 FW: Nat + Asterisk + Ser (Far end Nat Traversal)
5:52AM 1 Delete function in realtime voicemail?
5:43AM 2 Cut leading digit?
5:35AM 0 Aastra 9133i and MWI
5:29AM 0 Aastra 9133i Phone and MWI
4:35AM 0 SDP media attribute
4:22AM 0 REGEX Function
1:32AM 1 Help with Weird Setup
12:03AM 2 How to start ztmonitor in 'quantitative' mode ?
12:00AM 0 Asterisk as a SIP provider
 
Sunday August 21 2005
TimeRepliesSubject
11:46PM 0 Using locked PAP2 and PAP2-NA with Asterisk
9:41PM 2 hybrid clients
8:46PM 0 Suggestions
8:27PM 0 Re: call waiting beep on PSTN and TDM400P FXO linehook flash
7:07PM 2 perl-cpan
6:17PM 9 Dial Zero to get outside line?
2:54PM 3 Call duration limits not working
2:42PM 0 Nortel Meridian-1 Line Side E1
1:41PM 0 call waiting beep on PSTN and TDM400P FXO line hook flash
12:33PM 0 PrivacyManager not working Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n
11:59AM 2 Broadvoice Issue
11:51AM 0 On Network Usage from the CDR
10:49AM 0 Problem with auto-attendant config, I think..
8:31AM 1 "Not-Registered" Problem
6:57AM 5 Asterisk 1.0.9 on SuSE 9.2 with ISDN BRI & zaphfc?
6:14AM 3 TDM11B modprobe wcfxs fails
4:17AM 4 Warning Unable to allocate socket
 
Saturday August 20 2005
TimeRepliesSubject
11:25PM 2 Asterisk(*) on a Cobalt RaQ2?
11:24PM 1 Re: Asterisk-Users Digest, Vol 13, Issue 131
11:18PM 0 Re: Asterisk-Users Digest, Vol 13, Issue 131
10:05PM 39 Small office setup/using analog lines w/ Asterisk
9:23PM 4 [Asterisk-Dev] IM patch
7:07PM 0 1.0.9 - can't get link up, 1.0.7 works fine.
7:01PM 0 Asterisk aborts => undefined symbol: pri_channel_bridge
6:47PM 0 Re: Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1
6:34PM 5 Problems with Asterisk(*): Not-Registered
5:11PM 0 1 server vs. 2 server config
3:29PM 0 Help needed receiving incoming calls.
2:33PM 0 Asterisk transcoding /Routing
12:43PM 0 X100P compatible
11:21AM 1 Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension
10:44AM 5 Ring more than two isdn phones simultaneously
10:08AM 0 ZAP divert problem
9:11AM 0 ATA186 reguest problem
8:11AM 3 ISDN BRI voice one way only
8:04AM 0 OT? ... Trying to get cid_rewrite script to work
7:33AM 0 need provider with these did's avail: (anybody?)
5:28AM 6 ViaTalk Down?
5:15AM 0 static noise with TDM revision G but not with revision F
4:53AM 0 What is the reason for warning "Unable to allocate socket"
4:20AM 4 Asterisk Zaptel Leading Zero Problem With TE110P
4:05AM 3 Realtime sip_buddies "register=>" how?
3:57AM 1 Call quality problem when using lan
3:47AM 0 Quality problem on LAN when using the network!
 
Friday August 19 2005
TimeRepliesSubject
10:28PM 0 meetme mixer configuration
6:59PM 2 Asterisk and CompactPCI boards??
6:45PM 0 Configuring Asterisk as SIP client behind NAT
5:36PM 6 Where did my DID's go??
4:50PM 2 FXO not picking up; baffled
4:03PM 11 ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
3:38PM 11 [OT] Looking for Web based SIP endpoint
2:44PM 0 Sudenly unable to get incoming from
2:35PM 10 Installing to a prefix.
1:51PM 6 Overriding Caller ID
1:35PM 1 Sound warnings bringing asterisk down.
1:05PM 2 Ascend Pipeline POTS to TDM400P FXO Question..
12:27PM 4 Asterisk and Vonage - Can't call out but can receive calls
11:49AM 2 Asterisk not conforming to the RFC?/Aastra phone delay issue
11:08AM 4 Sending digits from SIP to Asterisk's VoiceMailMain
11:07AM 2 Unexpected hangups when calling Dialogic D/41JTC-LS
10:50AM 0 OT: autoresponders
10:07AM 3 Cisco 7960 Line rollover for secretary's phone.
9:41AM 0 DTMF on Zap / PBX Transfer
8:38AM 1 Persistent variables disappear when dialingLocalextension
8:08AM 0 Overlap digits...
7:42AM 3 Persistent variables disappear when dialingLocal extension
7:42AM 11 Sudenly unable to get incoming from Broadvoice
7:20AM 3 CVS-HEAD Compile Problem
6:45AM 1 Is there a way in dialplan to determine if call is incoming or outgoing if callerid presentation not enabled on line?
6:30AM 0 tdm400 and hfc card problem after ztcfg
6:01AM 5 any ISDN/PRI signaling experts out there?
4:54AM 1 Nat + Asterisk + Ser (Far end Nat Traversal)
4:26AM 0 dialing by CODEC type
3:40AM 1 IPManager now supports SIP, IAX and Zap
2:31AM 0 meetme-icecast2-ice2
1:57AM 0 Has Anybody a working Asterisk auto-startup(Init.d) for SUSE 9.2?
1:14AM 0 Tr: [Asterisk-Dev] Asterisk IM + Presence
12:04AM 4 sccp help
 
Thursday August 18 2005
TimeRepliesSubject
11:55PM 2 Monitoring RTP protocol
11:30PM 0 why asterisk starts listening on all ports
11:20PM 0 SV: Zaphfc.ko module error
10:43PM 1 Unable to transfer external calls to MeetMe conference
8:21PM 14 Vonage locked Motorola VT-1000s
8:09PM 1 Agi Script - sending a message to called party
7:34PM 1 Newbie Trying to make 'catch all extension' but is catching voicemail exit!
7:33PM 0 asterisk command realtime
6:25PM 0 SER, Asterisk, SIP proxy, routing, redirection - confused
6:08PM 3 Optimum online-upload throttling confirmed.
4:01PM 2 asterick and festival...Help!
3:08PM 2 Re:How many TDM22P Card can be used on thesame PC ?
2:03PM 0 MP3Player cmd issue
2:02PM 1 Persistent variables disappear when dialing Local extension
1:45PM 0 Which AGI Development Software is fastest onAsterisk?
1:28PM 4 Updated Patch to chan_agent.c for PREACKANNOUNCE
1:18PM 5 RES: asterisk seems to load but cannot connect using -r?
1:04PM 5 VoipJet Problems - anyone?
1:01PM 5 Which AGI Development Software is fastest on Asterisk?
1:00PM 0 asterisk, Kirk IP600 and Kirk Z-4020
12:57PM 0 asterisk seems to load but cannot connect using -r ?
12:35PM 4 Preventing an extension from dialing certain outbound codes
12:32PM 0 How to get long distance carrier to provide separate billing for several companies that share a PRI to LEC?
12:32PM 0 Set voicemail maximum length by context
12:32PM 0 [Fwd: Re: Set voicemail maximum length by context]
12:27PM 0 Directed pickup troubles
12:20PM 4 static noise with this hardware any advice
12:11PM 2 Hardware echo cancellation
12:01PM 0 Festival sounds too wired !!
11:55AM 3 Searching For a Voip Provider
11:47AM 7 SPA-2100 Analog Telephone Adapter
11:46AM 6 Polycom SoundPoint 501 power adapter
11:26AM 6 Craig R. Saxton/PACE/US is out of the office.
11:14AM 0 re: slightly OT
11:14AM 2 Zaphfc.ko module error
10:56AM 0 help with waning on OSS/dsp, condition 16 and 17
10:47AM 0 HDLC Bad FCS / HDLC Abort solution
10:43AM 0 Question about SIP connection and disconnection events on Asterisk
10:08AM 0 Asterisk -rx causing crashes?
10:00AM 1 ASTCC UPDATEproblem
9:58AM 0 Awesome Job for the Right VoIP Engineer
9:57AM 1 Cisco ATA-186 working peer to peer
9:28AM 3 libpri mwi functionality?
9:13AM 4 Disconnect supervision question
9:07AM 0 Re: Asterisk-Users Digest, Vol 13, Issue 123
8:41AM 4 options for mysql query from dialplan
8:28AM 5 CRM software
8:17AM 1 Epygi QuadroFXO?
6:58AM 0 [ACD]AgentCallBackLogin
6:16AM 1 pins for users
6:08AM 1 RE: Pannel
6:02AM 1 do not appear to have the sources for the 2.6.11.4-20a-default kernel installed
5:26AM 2 Password for Conf Room
5:05AM 0 Limit fax tx speed of 'dumb' faxes??
4:54AM 21 SNMP for Asterisk
4:46AM 2 codec gsm and cisco
4:41AM 0 Tr: [Serusers] SER as "Outbound SIP Proxy"
4:14AM 1 asterisk with odbc
3:34AM 0 bristuff-0.2.0-rc8f-cvs does not work with TDM400P
3:26AM 0 asterisk oh323 not detecting dtmf
3:19AM 5 segfault with chan_capi-cm 0.5.4
3:07AM 2 Lock Extension
2:32AM 0 granstream, vlan, tftp
2:26AM 3 initiating Monitor during call
2:06AM 3 Help on AGI running
1:59AM 0 Asterisk configuration from database
1:58AM 2 Asterisk (OH323) - gnugk connection
1:06AM 0 rotary/pulse
12:44AM 0 Agent Wrap-up status
12:37AM 1 Which external (remote) gateway I can use with * ??
12:09AM 2 Asterisk configuration from database with res_config
12:04AM 3 V.17
 
Wednesday August 17 2005
TimeRepliesSubject
11:28PM 0 Which cards or box for Germany? - Welche Karten / box fuer Deutschland?
9:46PM 0 version 1.0.9 slow in acknowledging agent channel calls
9:10PM 0 sip.conf user entry for ViaTalk
8:51PM 0 Automatic outgoing calls calling twice
8:21PM 1 trouble with IP500
7:30PM 0 TE110P w/ Dell SC1420 ... any problems out there?
5:02PM 2 Choppy Ringing
4:36PM 10 How many TDM22P Card can be used on the same PC ?
3:30PM 2 How "real time" is realtime?
3:27PM 6 IP Cop as a firewall and QOS
3:17PM 0 AstriCon Update: Early Bird Ends Soon - Free Asterisk Book
2:37PM 8 TDM04B, trunk group
2:34PM 2 Patchy audio to and from VOIPBUSTER
2:30PM 2 CAPI problem - need help
2:13PM 0 [Fwd: [SA16438] Grandstream BudgeTone Denial of Service Vulnerability]
1:45PM 1 Comfort Noise incomplete - No translator path exists for channel type MGCP (native 4) to 256
1:18PM 1 Voicemail/Directory, one person, one box, two last names
12:54PM 0 chan_sip2.c compiling
12:38PM 1 DUNDi Install
12:15PM 1 AGI SCRIPTS USING PERL NEED SOME KIND OF COMPILATION TO WORK WITH *
11:51AM 0 TDM400 on BT network in UK
11:36AM 20 DECT gateways
10:53AM 5 XML Revisited
10:47AM 0 FXS on TDM12B suddenly stopped working Properly
10:35AM 0 Asterisk and Port
10:14AM 3 Does intel 865 board works fine with Asterisk
10:06AM 0 Xten & Digum TDP FXO card: No sound
10:03AM 0 (no subject)
9:33AM 4 Iaxy Distinctive Ring
9:27AM 0 canreinvite in sip.conf
9:27AM 0 Any success with Polycom DHCP VLAN discovery?
9:01AM 0 [Asterisk-Dev] New Astmanproxy Mailing List, and New Version 1.11
9:00AM 0 Avaya 4602 SIP Internal Dial Plan
8:38AM 2 DID on TDM400P Question?
8:23AM 2 Voicemail crashes asterisk
7:07AM 2 SIP message 183 and in band info
6:56AM 3 Any one using the new Digium echo cancellation cards
6:48AM 2 OT: PC network down if plugged in Polycom IP600
6:41AM 2 snom hint
5:36AM 2 XORCOM RAPID Asterisk - Suggestions?
5:34AM 2 iaxcomm huge latency
5:20AM 3 X100P dial out problem
5:17AM 6 Echo cancellation again ...
3:28AM 1 zaphfc ptp did problems
3:27AM 6 Automatic start with SuSe linux
2:55AM 0 Asterisk (multiple) + Ser
2:23AM 4 FW: Asterisk-panel
1:34AM 16 1-800 number
1:29AM 8 Voicemail Retrival
1:25AM 0 How to change RINGING style for internal calls
12:29AM 0 Nikotel issues
 
Tuesday August 16 2005
TimeRepliesSubject
11:53PM 3 Can not dial more then 23 calls
11:30PM 0 Re: [Asterisk-Dev] X101P register map data please?
11:27PM 0 IAX compatible phones
11:22PM 0 Re: [Asterisk-Dev] X101P register map data please?
9:57PM 3 florz patch for bristuff breaks compile on x86_64?
9:49PM 3 ASTCC astcc-config.conf card length question
8:18PM 6 TE410P + SPANDSP fax problem
8:10PM 0 Solved: Unable to load module for TE406P
7:15PM 0 3 way calling
7:00PM 5 All Page ??
6:40PM 0 Polycom 501 Firmware
6:33PM 1 Execute script on Answer
6:30PM 1 Does Asterisk support T1 E&M Wink/Wink voice channels on any Digium/Sangoma hardware?
5:59PM 0 Result from TxFax
4:01PM 4 SIP "agent" phone w/ headset
2:42PM 0 hint on parkedcalls
2:10PM 9 realtime caching
1:44PM 0 RE:Asterisk-Users] PhoneCALL v2.6.1 - Released
1:25PM 0 What should my next steps in troubleshooting this TDM04B error be?
12:37PM 7 TxFax -> RxFax on same machine hangs
12:20PM 4 Called Party Identification on Polycom IP501
12:07PM 1 Asterisk and H323 interoperation issue
12:05PM 0 Re: Asterisk-Users Digest, Vol 13, Issue 109
11:55AM 2 Polycom 501 dialing problem
11:40AM 8 5 way calling?
11:39AM 0 MFC/R2 DTMF and digits "*" and "#"
11:36AM 0 [Asterisk-Dev] SIP channels not cleared
11:00AM 0 X-lite and Dell Optiplex
10:29AM 0 asterisk supported compact pci boards
10:20AM 1 how do we block registration based on ip/subnet?
9:38AM 1 Advice on old polycom ip 500
9:37AM 1 calling number type
9:32AM 17 quad t1 / 1U rack server combos
9:01AM 14 Asterisk and LCR
8:56AM 3 TAFM
8:31AM 0 Tr: RE: Maximum remote directory size in Polycom IP501
8:27AM 7 problems with eyebeam - video phone
8:02AM 1 x100p question for incomming calls
7:22AM 1 USB ISDN
7:07AM 0 Echo calibration with ztmonitor and a testlinefrom a telco
7:04AM 4 adding another fxo card
6:57AM 6 PhoneCALL v2.6.1 - Released
6:55AM 1 Asterisk QUEUES ACD Call Back
6:25AM 0 features.conf and CVS
6:19AM 1 DISA over Zap (TE110P) issues on * STABLE 1.0.9
6:17AM 0 Help Asterisk -> Hipath 1500 V3.0
6:12AM 2 Send 12khz or 16khz billing pulse through fxs
5:54AM 3 DTMF, Asterisk, External PSTN gateway, and PAP2 (was: RE: Issue with DTMF Tones - CodecIssues)
5:13AM 5 Issue with DTMF Tones - Codec Issues
5:09AM 1 E1 R2
3:48AM 1 intel 875P chipset ok?
3:22AM 2 Registration with Asterisk server
12:36AM 1 Astcc Problem
12:21AM 5 HP Compatability
 
Monday August 15 2005
TimeRepliesSubject
10:49PM 1 astcc brands table, inc field
10:31PM 3 Transferring from cell phone
10:19PM 7 Plantronics USB Headsets Audio 45
10:10PM 0 DTMF being cancelled
8:26PM 1 Maximum remote directory size in Polycom IP501
8:02PM 21 Voipbuster blocking Asterisk/IAX connections?
7:03PM 0 Mitel 5220 Dial Problem
6:46PM 0 Ast.1.0.9 (only) strange problem with IAX and DDNS
6:25PM 0 Load-balancing / offload question
4:58PM 0 Call waiting beeps
4:46PM 5 Simple Fax question
4:15PM 1 Configuration to get CallerID working in New Zealand
4:09PM 1 NAT'd Snom360 problems
2:54PM 4 dnsmgr
2:35PM 5 bristuff-0.2.0-RC8n problems and kernel panic
2:32PM 7 8 FXS in Asterisk Server
1:31PM 0 Firewall will definatelyincreasejitters inyourvoice conversation
1:30PM 3 Only single channel recorded with Monitor
1:23PM 1 How to remove standard ISDN drivers from RedHat
1:21PM 0 Firewall will definatelyincrease jittersinyourvoice conversation
12:40PM 2 Fax Issues
12:08PM 2 problem with sound device
11:22AM 4 BRI Hunting, using both channels on one msn
11:16AM 13 Switch between FXS ports
9:38AM 3 No translator path exists for channel type MGCP & Comfort noise support incomplete
8:52AM 0 h323 registration problem
8:37AM 3 Re: [Asterisk-Dev] MS Live Communications Server
8:25AM 2 User in two queues receive two calls at once
8:21AM 1 Chan_sccp and dynamic DNS
7:30AM 0 Conference moderator password
7:29AM 0 RocketVoip?
7:19AM 1 (no subject)
6:29AM 7 codecs order
5:17AM 4 asterisk + chan_mISDN = undefined symbol: ast_pickup_call
2:43AM 0 Asterisk Java-Call Problem
2:22AM 2 Security and SIP
2:11AM 1 permission denied when monitoring channel OSS/dsp
1:17AM 0 Unable to load module for TE406P
1:16AM 1 Connecting 2 * servers
 
Sunday August 14 2005
TimeRepliesSubject
11:37PM 0 Sirrix bri card:killing the machine
10:55PM 1 PABX and Asterisk Dial Plan
10:33PM 0 (no subject)
6:50PM 0 anyone use TE4xxP work well with huawei C&C08 switch?
5:15PM 0 h Priority
2:39PM 0 setting up rate-engine?
2:37PM 0 OrderlyQ
1:12PM 3 Problem with FWD connection rejected
1:12PM 9 Bigger problems than ogg
11:23AM 11 Cisco and "protocol application invalid"
10:25AM 0 subscibe FOODFIX digest
8:55AM 2 TELASIP DOWN?
8:34AM 12 Multiple Asterisk Installations + SER
8:16AM 2 ogg causing me heart burn
7:15AM 0 [OT] SPA-3000 loudness
6:11AM 0 IPManager now templated based
6:11AM 1 *confused* - help needed
5:37AM 2 ParkAndAnnounce - No Disconnect
4:33AM 0 ParkAndAnnounce - Any way to not disconnect?
3:37AM 2 Module wcfxs - is it not part of astlinux?
 
Saturday August 13 2005
TimeRepliesSubject
10:41PM 0 HooDaHek 0.3 Released
10:41PM 1 Initiating a transfer from an analog handset?
7:17PM 0 Call Queues and Agent Call Logs/Wrapup logs
5:27PM 0 Asterisk Flash Transfer (callthrough)
4:53PM 2 Asterisk forwarding confirmation?
2:48PM 0 cvs STABLE of 08/10 & gcc4 issue
1:52PM 28 Why NAT problem
1:08PM 2 forward incoming analog call to SIP?
12:20PM 0 Re:(2) Henning G. Schulzrinne quote on IAX2 from von magazine
11:57AM 0 Re: Henning G. Schulzrinne quote on IAX2 from von magazine
9:55AM 1 T.38 decoding
9:20AM 0 Re: Asterisk-Users Digest, Vol 13, Issue 86
9:10AM 2 (no subject)
9:06AM 0 Attended Trasnfer
8:56AM 0 extensions exchange
8:41AM 2 premature call release - SIP 480
8:36AM 2 Cisco IP Phone- 7905G
8:07AM 2 New Beta IAX Statistics Program
7:46AM 5 TDM400P Card (Rev G) with bad FXS module?
6:30AM 0 [Asterisk-Dev] Re: FXO PCI Master abort
6:14AM 2 receiving calls from FWD
5:50AM 6 One more newbie question
5:26AM 0 Flash over SIP Trunk
5:07AM 3 Push to talk and asterisk
4:52AM 0 txfax on strike while rxfax works flawlessly
2:59AM 2 Identify call flow from manager events
2:44AM 5 MISDN callerid
2:05AM 0 Receive fax then send onwards
1:25AM 1 Disable Call Waiting On SIP User Agents
12:59AM 0 Incompatible destination (88) Error Message. Please Help !!!
 
Friday August 12 2005
TimeRepliesSubject
11:29PM 2 Remotely rebooting Sipura SPA-3000 from command line
8:20PM 1 fc3 build after kernel update?
7:06PM 4 Suggestions for mainstream hardware compatible with TE411P.
4:29PM 6 chan_skinny issue
3:48PM 6 Dell Poweredge 1400
3:26PM 0 ubr924
3:20PM 3 Announcement to called party
2:47PM 5 voicemail - 99 message limit
2:03PM 0 Saved Message playback
12:42PM 1 Small Form Factory Machine
12:11PM 0 Asterisk Cell Socket Recommendation
11:37AM 4 PauseQueueMember and UnpauseQueueMember
11:31AM 4 7960 TFTP
11:29AM 0 Zap with fax outbound signature
11:27AM 10 yet another Asterisk and VMware question
11:07AM 0 7960 Stuck booting
11:05AM 2 ChanSpy and Sipura 2100 jitter.
10:55AM 14 FXO port trhoug optimum voice VOIP service
10:42AM 1 I need a Asterisk tech
10:29AM 9 TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
10:23AM 0 New astGUIclient version released 1.1.5
10:09AM 8 Incompatible destination (88) Error Message
9:25AM 0 three questions
9:21AM 1 Polycom IP500 / Registration Question?
9:19AM 0 Forwarding behavior question
8:37AM 1 Comedian annoucment files
8:07AM 3 PC for 8 line system
8:03AM 1 Weird issues with TDM400P
8:00AM 0 Ang: Voipjet experiment
7:57AM 4 Voipjet experiment
7:34AM 0 7960 + 7914 Problems
7:01AM 0 txfax spandsp
5:46AM 2 Possibly bad FXS module in TDM400P?
5:24AM 0 yahoo voice
3:59AM 9 TE405P / TE410P with 2nd generation firmware field upgradable?
3:51AM 1 Call recording, monitor & soxmix in Asterisk 1.0.9
3:42AM 4 v92 modems
2:09AM 9 OT: Sendmail question
1:31AM 0 Status of app_sms in 1.0.9
12:43AM 6 Billion BRI PCI card
12:25AM 0 ZapHFC E1 PRI (cwain)
12:08AM 0 txgain for SIP?
12:08AM 0 My users are using PSTN instead of VoIP
12:05AM 4 TE405P V2 changes?
 
Thursday August 11 2005
TimeRepliesSubject
9:20PM 3 wildcard/FXO config
9:04PM 2 list in asterisk cli is getting too long
6:33PM 0 Call queues bug?
5:52PM 0 Patches 0002838 and 0002924
5:21PM 1 Firewall will definately increasejittersinyourvoice conversation
4:05PM 0 Unexpected On Hook event
3:57PM 0 Join Martin O'Shield on >Yahoo!
3:35PM 4 is this possible with asterisk?
3:12PM 6 Install just to play with experiment
3:10PM 0 NVLineDetect and head after aug 2
3:06PM 1 External channels getting connected
2:22PM 4 Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone
2:14PM 3 Newbie Question: Building an Asterisk systemtoreplace an old PBX but using existing phone
2:09PM 0 Nothing techhy, just a greeting
1:47PM 0 Re: 24. Privacy Manager (Andi Strain)
1:44PM 22 How to fix a Blue Alarm?? Line Noise?
1:25PM 1 PRI dropped calls w/ asterisk dropped betweenpstn & norstar
1:04PM 3 Newbie Question: Building an Asterisk systemto replace an old PBX but using existing phone
12:57PM 0 CVS HEAD 08-11-2005 & Definity G3R
12:25PM 9 Cisco 79XX and VLANS
11:52AM 14 Polycom IP301 and 501 with asterisk...
11:12AM 1 Cisco 7920 boot causes 7940 to release DHCP lease
9:31AM 0 disable initial music for call queue
9:30AM 1 Where to buy Sangoma cards?
9:02AM 0 meetme.conf and realtime
7:58AM 0 is there cdrs for sip
7:35AM 2 Is it mandatory to give power supplytoTDM400Pcard
6:40AM 2 Suggestion for VoIP router with QoS
6:20AM 13 Realtime + MYSQL
5:34AM 14 Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
5:04AM 3 Comedian mail ignores mailbox greetings
4:42AM 1 More then one Tormenta 2 E1/T1 card on system.
3:34AM 3 MS Live Communication Server
3:28AM 2 re: how to set the voice message as email attachment ?
3:07AM 0 Sipura-3000 IP->PSTN scenrio
3:01AM 4 Ignoring the called number in the INVITE message
2:03AM 2 help on receive text
1:02AM 0 How to determine elapsed time of a call in progress?
12:24AM 2 Sip ports
12:05AM 11 Supervised transfer problem with BudgetTone
12:03AM 0 * behind NAT, client behind NAT(handytone 286), very strange behavior
12:02AM 1 IAX setup
 
Wednesday August 10 2005
TimeRepliesSubject
11:58PM 2 Zultys ZIP 4x5
11:33PM 0 No speech path
11:28PM 1 Error while calling
11:20PM 0 tdm400p / outbound zap prob
10:21PM 1 Join Martin O'Shield on Yahoo! Messenger!
10:18PM 1 Help, using SendText cmd sip message...
7:56PM 5 Help with calling Perl AGI interface
6:29PM 1 Addendum to my post re: BrookTrout TR1034 T.38
6:10PM 0 T.38 Faxing w/ BrookTrout TR1034 FOIP Board
5:50PM 2 PRI dropped calls w/ asterisk dropped between pstn & norstar
4:37PM 6 real-time priority
4:34PM 5 ZAP bchan and dchan HELP!!
4:20PM 3 Hitachi wip5000
3:08PM 8 Blank CIDName or CIDNum = "asterisk"
2:29PM 1 Problems with zaptel.conf
2:06PM 0 Blank faxes in rxfax.
1:46PM 5 Hard deskphone via wifi?
1:27PM 0 Problem with setting the right dialplan for german PRI E1 on TE405P from digium
1:24PM 3 does SIP works behind the NAT
1:15PM 2 Is it mandatory to give power supply toTDM400Pcard
1:13PM 0 Error reloading extension!
12:43PM 5 Firewall will definately increase jittersinyourvoice conversation
12:41PM 0 Building on Itanium
12:30PM 2 Is it mandatory to give power supply to TDM400Pcard
12:24PM 5 Firewall will definately increase jitters inyourvoice conversation
12:04PM 5 app_voicemail.c still looking for config file even I try to configure the voicemail from database.
11:32AM 1 Limiting the number of calls
11:27AM 3 Is it mandatory to give power supply to TDM400P card
11:26AM 1 Firewall will definately increase jitters in yourvoice conversation
11:24AM 0 Waring problem with different brand phone
11:18AM 1 Asterisk support of MF trunks?
11:16AM 0 Problem with channel allocation between BRI and PRI cards
11:14AM 0 Asterisk scaling with agent channels
11:06AM 2 Help me how to listen voicemail with SIP 7960
11:04AM 1 T100P Problems
10:59AM 0 Radius + NAS with Asterisk
10:57AM 2 Firewall will definately increase jitters in your voice conversation
10:56AM 0 Vonage Click-2-Call
10:46AM 0 Asterisk mailing lists
10:37AM 2 realtime odbc/mysql eating connections
10:35AM 3 E&M to E&M Dialing - TE410P
10:30AM 0 Polycom 501 Do Not Disturb issue
10:28AM 4 GrandStream GSX-2000 strangeness
10:05AM 3 TDM40B and weird analog problem
9:32AM 0 audio fading in and out?
9:25AM 2 asterisk query mysql problem or bug?
9:14AM 1 chan_oh323.c:2706 oh323_request: Blocking outbound H.323 call due to call-limit violation.
9:09AM 0 Asterisk and Asterisk management portal issue
9:02AM 0 RE: Info / recommendation on either Audiocodes or Vegastream gateways
8:42AM 0 Numeric Pagers & Voicemail
8:14AM 2 App_Queue strategy=ringallfree (feature request, possible bounty)
8:09AM 3 h323 error when trying to start Asterisk
8:05AM 0 Problem with voicemail, invalid extension, no error handler
8:03AM 3 Help with TNT and Asterisk
8:01AM 0 Asterisk Stops Sending Data (CVS 20050809)
7:46AM 4 TE205P installation problem - ZT_SPANCONFIG failed on span 1
7:20AM 1 where the debug log stored?
6:49AM 3 SRV implementation supporting priority
4:39AM 15 USB handset wanted
4:10AM 0 Asterisk and SER and Asterisks Queues
2:55AM 0 Single extension/user registers across multiple asterisk servers
2:28AM 2 Asterisk Call Queue Application
2:20AM 4 No audio when calling between internal phones
2:13AM 0 Asterisk & RTC Client API
2:11AM 7 will a firewall slow down asterisk?
1:54AM 5 Calling Extension directly
1:54AM 0 Yoda VG-400 and Asterisk Settings
 
Tuesday August 9 2005
TimeRepliesSubject
11:46PM 10 Load Testing
11:24PM 0 Incoming call #2 sent to VM immediately whenalready on phone with incoming.
11:06PM 7 error compiling asterisk on solaris
8:51PM 26 call "load balancing"
7:50PM 4 Need some statistics & facts
7:29PM 0 Can I change the call waiting signal tone.
6:05PM 2 How to dial several extensions with different timeouts
5:07PM 0 Console Auto-Completion Lockup
3:56PM 3 Incoming call #2 sent to VM immediately when already on phone with incoming.
3:17PM 2 inbound caller id name pri - tnt - asterisk
3:05PM 2 Playback before Answer
3:02PM 15 ISDN DID
2:45PM 3 SIP-Trunk problem, Please help!!!
2:40PM 2 Asterisk and XML Applications
2:14PM 2 Stable or not?
1:59PM 0 channel_pvt.h not found
12:56PM 1 Com-On-Air (PCI/PCMCIA) chan drivers?
12:15PM 0 Sipura wrong password on invite
12:06PM 4 detaching console from foreground asterisk
11:50AM 0 Registration intervals
11:03AM 2 dvc 1000 support
11:02AM 4 X100P Wildcard - Hassle free clone?
10:12AM 2 Asterisk and Wave files problem
10:02AM 0 Connection Asterisk- Panasonic TDA200
9:26AM 18 Build on Itanium fails
9:07AM 14 QoS General Question
8:25AM 0 Random Zap Channel Resets
8:25AM 2 Both lines in an ATA using the same UID/PASS
7:57AM 3 Playing GSM files in Windows?
7:38AM 0 H.323 vs SIP for small FXO gateways
7:28AM 1 TE110P flashing red/green when PRI connected ... continued
7:22AM 0 Echo during begining of incoming calls
7:08AM 3 CLI and Dial
7:02AM 0 looping through SER
6:53AM 5 First PRI
5:22AM 0 How to configure Outbound Proxy for REGISTER?
3:09AM 1 Incoming call action based on trunk
2:53AM 0 Cannot hear Music On Hold with SIP Phones
2:23AM 1 voip solution with SER, ASTERSIK and CCM
 
Monday August 8 2005
TimeRepliesSubject
11:54PM 0 Calls to Turkey, any good providers?
11:53PM 0 Broadvoice europe plus calling plan quality
11:51PM 1 T1 versus PRI
11:41PM 0 Active channel, no users
11:38PM 1 FXO definition
10:36PM 0 queue-hold time + weight in astersk+acd
9:37PM 0 delay problem
8:04PM 0 FXO gateways / Audiocodes MP-108
8:01PM 0 OT: Anyone having issues with sipphone?
6:55PM 6 SNOM Hint for MeetMe
6:34PM 0 Problems with cmd monitor
6:24PM 2 X100P with Caller-ID in Australia,
6:19PM 3 FXS - Don't want a Dailtone
5:37PM 2 Asterisk and .NET
5:27PM 1 Press # to continue / Findme
5:09PM 0 Re: asterisk rpms (was: Does anyone run Asterisk on FC4? with Digium's TDM40B cards)
5:02PM 1 Snom 360 4.0 firmware issue
4:54PM 0 Help interpreting channel stats?
4:22PM 0 OT: DTMF issues with Vonage forwarded lines
4:06PM 0 Question about agent queuing in Asterisk
3:42PM 10 IAX TO IAX call between two registered servers
3:16PM 2 Detecting hangup - TDM400P / X100P
3:09PM 0 ISDN D-Channel Problem / bristuff / qozap
3:05PM 2 zaphfc syslog flooding
2:45PM 1 FCC to require wiretaps from VoIP providers
2:24PM 0 IAX and Realtime...
2:13PM 0 howto let the media stream not passing saterisk?
2:03PM 3 Call Recording with *
1:49PM 0 Where is the asterisk DB file stored?
1:48PM 1 howto let the stream not passing asterisk
1:40PM 4 AGI perl problem
1:35PM 0 Voicemail Web Access Security
1:30PM 9 [OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)
12:47PM 2 URGENT: Problems with PHP AGI...
12:31PM 0 Failed IAX Connection
12:08PM 0 Polycom IP600 Presence question
11:49AM 0 Call Quality Issues
11:24AM 7 info regarding hardware
10:58AM 0 Failed to authenticate user
10:49AM 0 Screening Sip Calls - Record()
10:33AM 0 Wired Interactions between Asterisk (Public) and Budgetone (behind NAT)
10:22AM 1 Call forward & SER as SIP router
9:27AM 10 DTMF issues with SIPPhone?
8:52AM 0 Config files for zaphfc in nt mode
8:47AM 1 IAX2 Encryption
8:21AM 0 Asterisk-to-IVR Problem
7:26AM 0 Packet loss concealment and G729
7:05AM 0 trouble using variables with included contexts
7:02AM 0 Voicemail web access
6:56AM 0 g729 recording on asterisk using g729 enabledphone
6:33AM 0 g729 recording on asterisk using g729 enabled phone
6:28AM 6 Multiple MWI on a single phone?
6:26AM 0 Configuring TDM40B and X100P
6:01AM 0 Mediatrix 1204 setup
6:01AM 0 problem with callerid ( SetCIDName )
5:49AM 1 Newbie with Cisco 7910 phones
5:22AM 1 Transfer a call from cell phone (pseudo-disa)
5:21AM 3 Stun support
5:12AM 5 Speex QoS
4:37AM 0 Using * and other gateways together
3:59AM 0 Need unique switchboard/op-panel written
3:13AM 1 Same action to multiple numbers
2:58AM 1 Found solution to my PHP AGI Script problem...
2:57AM 7 TE110P flashing red/green when PRI connected
2:53AM 2 problem in inbound calls
2:46AM 3 AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFOproblems
2:42AM 1 CVS not responding
2:34AM 0 Without E1 lines,how to test E400P card?
1:57AM 1 Setup faxing with latest CVS/STABLE
1:52AM 0 NT1 devices with analog ports on HFC based
1:48AM 0 zaptel compile to incorrect directory
1:03AM 10 Digium TE405P, caller id and migration to *
12:26AM 1 CDR TDS
12:23AM 0 RTC Client API & Asterisk
12:04AM 4 X100P with Caller-ID in Australia, anyone?
 
Sunday August 7 2005
TimeRepliesSubject
11:58PM 1 How to config voicemail with mysql?
11:35PM 1 mysql sock location
8:45PM 0 list of T.38 providers on wiki: please contribute
5:23PM 4 http://www.voip-info.org/ front page taken out by spammer
4:51PM 0 VoicePulse Connect down Sunday evening?
4:37PM 0 NT1 devices with analog ports on HFC based ISDN BRI cards in NT mode and asterisk (chan_mISDN)
4:36PM 3 Unable to connect to FWD
2:34PM 3 request for clarification on Asterisk T.38 bounty
12:34PM 0 Using * and 3rd party GW together
11:45AM 3 z-machine + asterisk = fun!
10:39AM 1 voice prompt repository
10:15AM 11 Can call from iax extn but cannot call it - unable to cteate channel iax
9:55AM 5 Configuring Asterisk@home for Sipgate.
9:45AM 0 Calls from Asterisk to CallManager 3.0 how?
9:13AM 0 How to configure * for Net2phone using innomedia settings
9:09AM 0 ASTCC web can't connect to DB
8:40AM 0 How to configure/install ISDN Card
6:58AM 0 zaphfc HFC-S in nt mode but no dial tone after pickup
3:52AM 0 Planet sip phone and asterisk
12:08AM 0 Can't compile asterisk-oh323 on Mandrake 10
 
Saturday August 6 2005
TimeRepliesSubject
11:07PM 9 SPA 841 form SIPURA
5:38PM 2 Dialplan mapping for multiple outbound providers to determine best rates
2:41PM 1 Setup faxing with latest CVS
2:13PM 4 Cisco 7206 and Sample configs (Newbie)
1:36PM 0 Need Help RE Zultys Zip 2+ Basics
12:43PM 3 sip/rtp performance monitoring
11:23AM 0 g729 pass-thru for sip provider and g711 ulaw for conference and voicemail
9:13AM 0 SIP rejecting calls?
9:00AM 2 How to test H.323
8:02AM 14 TDM400P - All extensions have same CallerID
7:57AM 8 BudgeTone 100 Woes
6:52AM 1 Extensions beginning with *
6:04AM 0 Latest Asterisk and Fedora Core 4 question
4:50AM 2 Voicemail -- newbie question
4:08AM 1 Queue_log all calls marked ABANDONED?
3:00AM 13 Does anyone run Asterisk on FC4? with Digium's TDM40B cards
1:28AM 2 low sound
 
Friday August 5 2005
TimeRepliesSubject
9:45PM 3 Phone interface hardware
7:35PM 0 call outside from FXS through FXO
7:01PM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday 8/6/2005
6:10PM 15 Very complicated dialplans?
5:53PM 1 TE405P Dropping Calls
5:30PM 1 starting asterisk with nice -5
4:46PM 3 TE411P problem
3:25PM 0 Uniden UIP 1868 / Asterisk experiences
2:47PM 0 Partner II Expert Needed
2:11PM 0 Seeking Beta testers for enterprise mystery service
12:41PM 2 Switchboards
11:30AM 0 CallerID Problems.
11:29AM 0 number 'register => ' in sip.conf
11:16AM 7 Snom 360 and firmware 4.0 problem
11:16AM 3 how may channels
10:46AM 6 Is this echo problem down to IP Phone hardware?
10:30AM 3 Uniden UIP200 Opinions
10:29AM 1 Abwesenheitsnotiz: Nortel Option 11 and TE110P o f Digium
10:25AM 5 Asterisk MWI and Realtime
9:54AM 0 ATA186 can not generate dtmf
9:47AM 4 Nortel Option 11 and TE110P of Digium
9:35AM 0 Looking for IBM or HP Server Recommendation
9:23AM 3 Zaptel warning
9:20AM 4 Need Help Troubleshooting Broadvoice Connection
9:08AM 0 IAX Phone Pro Beta - New Version Available
9:08AM 1 No dial tone on BT100
8:53AM 0 Masters changes / Line looses
8:33AM 8 Realtime IAX
7:57AM 0 Audio files problem - as usual
7:33AM 0 Another problem on queues
6:36AM 0 Phone hangups after a TEI check request
6:34AM 3 SIP signaling vs Media (Voice) Traffic
6:31AM 1 Asterisk (Comedian Mail) and AUDIX
5:25AM 11 asterisk registered in ser proxy
5:19AM 0 USB ISDN devices
5:07AM 0 IPManager has been released - the ultimate configuration tool for Asterisk
5:04AM 3 Is there a right place for a include_once statement in a PHP AGI script?
4:44AM 0 Roundrobin queue strategy broken ?
2:56AM 1 Problem running Astrerisk after defined card in /etc/asterisk/zapatal.conf
2:02AM 2 Cisco IP Phones on Asterisk: chan_sip or chan_sccp
 
Thursday August 4 2005
TimeRepliesSubject
11:30PM 0 defining range of user in sip.conf
11:18PM 0 Rebooting GS phone thru sip_notify
10:37PM 4 Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone
10:00PM 1 PolyCom SoundPoint 300 and distinctive ring
9:57PM 19 ip phones
9:57PM 9 Cisco IP Phone 30 VIP
9:25PM 0 Asterisk payphone with voipjet for retail/resale purposes
8:04PM 3 HELP! X100P IRQ conflict w/ USB
7:34PM 1 app_txfax.c problem
7:12PM 1 application doesn't dial out...
5:49PM 0 ADIT 600 Expert needed
5:26PM 3 include behavior (word puzzle of the day)
5:24PM 1 Cvs Head
5:22PM 0 Asterisk in ACD configuration
5:21PM 4 no ring to callers?
4:55PM 0 AbsoluteTimeout Problems?
3:53PM 0 h.323 Call problem asterisk to\from lucent(avaya) definity
3:37PM 0 h323 CALL PROBLEM TO / FROM AVAYA(UCENT)inity
3:30PM 2 Asterisk and the IAD2431 via MGCP
3:06PM 9 newbiew extensions.conf question
2:55PM 4 Outbound Extension problem
2:52PM 0 How to log the different extensions dialed within a single call?
2:10PM 2 Some echo?
1:55PM 1 bristuff-0.2.0RC8m
1:13PM 4 Asterisk Voice Mail Server and older Executone PBX..can it be done?
1:04PM 1 Receiving Calls from FWD Network using IAX2
12:15PM 1 Merlin Legend
11:38AM 0 BT102 phones giving strange errors
11:15AM 0 Re: Asterisk-Users Digest, Vol 13, Issue 25
11:05AM 3 How scalable is asterisk
10:34AM 3 Polycom and Presence
10:28AM 4 TFTP - Good or Bad?
10:23AM 0 TDD over Asterisk
10:07AM 0 No rering on misoperation on SIP ATA
9:55AM 3 Callback question
9:46AM 0 Re: [Asterisk-Dev] The killer app for Asterisk in corporate deployment
9:28AM 2 CVS Down
9:24AM 1 Getting asterisk to work with callthroughs?
8:54AM 0 Agent channels
8:47AM 0 weird DTMF problem
8:31AM 3 [Asterisk-Dev] OPAL now supports IAX2
7:58AM 0 RPMS & SRPMS of Asterisk STABLE & HEAD on i686 & PPC
7:51AM 1 send an sms through a gateway GSM (stargate)
7:34AM 19 Features you'd like to see in a GUI?
7:16AM 0 Voicemail advanced options, 5 to send a message not available
6:52AM 1 Asterisk, Tenovis, Fritz, capi problem
6:31AM 2 The killer app for Asterisk in corporate deployment
6:10AM 0 Best common practice for emailing conferences?
5:55AM 7 SIPPeersAction class file not found in the Asterisk-java.jar file
5:53AM 0 Call specified, but not found?
5:25AM 0 vmail.cgi question
4:30AM 0 Calls not cleared down if extra destinations or dial commands added to extension
4:12AM 5 Directory problem
1:22AM 1 REINVITE and Codecs
12:20AM 5 asterisk & cisco 7960 softkeys [Virus checked]
 
Wednesday August 3 2005
TimeRepliesSubject
11:43PM 3 PLEASE REPLY, are you using an X101P
11:39PM 6 Send voicemail notification to SMS
11:22PM 0 Polycom ring volume
10:44PM 0 Asterisk TDM card connected to phone linesAND fax line
10:31PM 1 Attaching data to outgoing INVITE message .
10:19PM 2 ISDN BRI Funkyness
10:17PM 0 Dead spa841
9:43PM 0 Windows client for sending fax using txfax - spandsp
8:50PM 0 Multiple CLI connections
8:48PM 4 MFC/R2 Mexico Unicall Blocked
8:15PM 0 Asterisk on FreeBSD-5.4 RELEASE : H323 audio problem
7:09PM 0 Vmail.cgi and realtime
6:39PM 0 register sip
6:27PM 1 64K ISDN call not passing thru
6:16PM 1 IAXy2 question?
5:40PM 4 inter-asterisk meetme
5:06PM 0 Attaching data to outgoing INVITE message
5:06PM 1 Voicemail Password crashing
3:26PM 1 chan_capi upgrade
3:08PM 0 chanspy not working with Agents
2:49PM 0 iax to iax severs
2:34PM 0 Asterisk Network Troubleshooting Help Needed - Will Pay $$$
2:26PM 0 SIP call termination on PSTN lines
2:09PM 1 Incoming SIP from Cisco 7206
2:07PM 0 OT - but very interesting speech application
2:01PM 0 Compile ZAPTEL warning and Strange Congestion
1:13PM 6 polycom 301 phone advice
1:07PM 0 Line Buttons (Key system behavior)
12:53PM 0 Voicemail Issues
12:28PM 11 Cisco ATA and a PayPhone
12:10PM 2 Asterisk support Shared Call Appearance Signaling?
10:44AM 0 AstriCon 2005 - Early Bird Registration Open (Free IAXy To First 50!)
10:38AM 0 Inter-Tel AXXESS failure: HDLC Bad FCS (8) onPrimary D-channel of span 1
8:58AM 1 7970 SCCP configs?
8:52AM 0 Chan_bluetooth and AudioGateway phone [long]
8:09AM 0 IDSN 30 PRI UK
7:43AM 0 fax <--> grandstream 286 <--> asterisk <--> pstn
7:32AM 0 Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1
7:12AM 14 Transfer to outside line.
6:56AM 2 Asterisk TDM card connected to phone lines AND fax line
6:21AM 1 AstLinux - Anyone running on a Soekris Engineering net4826
6:15AM 2 Generic Question: Why should I use Asterisk over SIPxchange?
6:04AM 4 Is it possible to use CHAN_CAPI with ZAPHFC enabled card ?
5:37AM 0 How to test E400p card without E1 lines?thanks a lot
5:30AM 0 How to config incall ?I have a E400p card
4:38AM 3 Anyone know of an open source sip video phone like eyebeam available?
4:19AM 0 Is there an upper extension limit to Asterisk?
3:46AM 1 app_dbodbc for asterisk stable 1.09
3:03AM 0 LG Goldstar GDK-186/162 question on voicemail
2:51AM 1 Database querie
2:42AM 4 call does not hangup after client quits
1:59AM 0 app_intercept
12:41AM 0 Installing a TE100P (Digium) card over Suse 9.2..
 
Tuesday August 2 2005
TimeRepliesSubject
11:18PM 0 sip ata's
10:46PM 7 TFTP Secondary Ports
10:19PM 5 same extension on multiple sip phones?
9:25PM 3 what phones support this when running with asterisk
8:39PM 3 "invalid extension" dilemma
6:41PM 0 spandsp&nbsp;fax&nbsp;problem
6:03PM 1 Paging systems from the phone...
5:28PM 0 Few questions about Asterisk
5:09PM 0 asterisk e&m echo problem
4:56PM 4 port forwarding ip to ip sip calls
4:06PM 0 Channel Lock problems
3:38PM 2 How to let ZAPHFC work with and act on different incoming MSNs?
3:34PM 2 asterisk.org beta site up!
2:37PM 1 Polycom Soundpoint 600
1:56PM 1 Polycom Soundpoint 500
1:39PM 0 list test - ignore me
1:21PM 0 AstLinux 0.2.8 released
12:55PM 11 Dell Servers
12:33PM 1 stale nonce
12:26PM 0 RE: What does it take?
12:13PM 1 DND Indication
12:02PM 0 Problem with attended transfers...
12:01PM 5 7970 SIP
11:55AM 0 codec question
11:33AM 0 ISDN phone no dialtone
11:26AM 1 Two questions about Asterisk Call Center
11:22AM 0 Oh323 Module - Not Loading Error - Unregistered channel type 'Modem'
11:13AM 0 app_rxfax errors
11:11AM 3 Channel Bank Help Please....
11:02AM 1 Best way to connect asterisk to an traditional PBX
10:59AM 1 AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFO problems
10:52AM 0 TAPI driver: AstTAPI
10:49AM 10 Has Sixtel gone under?
10:00AM 2 Making a call on Asterisk... new thread or not?
9:23AM 0 Re: Minimum CPU required for >60 calls
9:09AM 0 Asterisk to Televantage
9:05AM 10 Polycom phones w/ two lines on different servers
8:51AM 0 Sip over VPN not working
8:28AM 15 WHat does it take
8:27AM 1 Ztdummy or Zaptel card on production server
8:07AM 0 can one specify "talking only" for a participant in app_conference
8:03AM 0 Asterisk as PSTN gateway, voice mail server with SIP
7:29AM 1 How to create a secret code to use myasterisk@home server's long distance plan from a public phone
7:29AM 1 Voicemail/Password Issue
7:27AM 0 New release: Queue Statistics 0.1
6:58AM 0 Re: Asterisk-Users Digest, Vol 13, Issue 7
6:55AM 4 How to create a secret code to use my asterisk@home server's long distance plan from a public phone
6:54AM 0 Polycom SoundPoint 600 : 10 seconds of delay when answering a call.
6:33AM 5 priority "a" in macro to access voicemail
6:23AM 0 Config extentions for ISDNphone (Phone autmatically calls internal extention)
5:55AM 0 Control IAXy Provisioning from a central
5:51AM 3 asterisk@home newbie extensions always busy
5:37AM 1 Strange DTMF issue with callback
5:25AM 0 Suggested System Specs - 20 ext, 8 Incoming Lines - Thanks
4:49AM 0 Dell SC420 and Interrupts
4:45AM 2 call center 20 seats
4:30AM 0 Hang up as soon as other party picks up call
4:25AM 0 Strange beeps in Calls
4:24AM 0 Asterisk & ISDN
4:07AM 0 Dialogic D/300/SC-2E1
4:03AM 3 This should work right??? Any caveats that you guys know about?
3:35AM 1 How does TDM work?
3:16AM 7 Minimum CPU required for 60 calls
3:14AM 0 FW: WEB SIP Dialer
3:10AM 0 Festival not working with Asterisk 1.0.7_7
2:45AM 0 strange asterisk issue
2:14AM 1 [Asterisk-Dev] Getting ISDN line restart problem with TE110P
2:07AM 1 Asterisk PSTN connectivity
1:52AM 0 Asking telephone no from caller
1:48AM 2 Config HFC-card in asterisk.(Config the phone and asterisk)
 
Monday August 1 2005
TimeRepliesSubject
10:40PM 0 register Every user without auth
9:48PM 2 TDM400P REV I issues - ProSLIC vs TDM400P
6:15PM 0 Dialplan to dial SIP, but stop dial on analog pick up?
6:14PM 0 Configuring A@H with Analog Phones UPDATED
6:13PM 3 Configuring A@H with Analog Phones
6:00PM 2 ast_config not updating voicemail password
4:58PM 3 X100P/Caller ID: clidtest works, * complains [repost]
4:13PM 0 Issue with zapata.conf "immediate" setting
3:52PM 1 Voicemail envelope time is 4 hours ahead
3:39PM 1 How to install PHPAGI?
2:37PM 0 Sipura SPA-1001: Bad Outgoing Call Quality
1:38PM 11 Queue/Agents
12:46PM 0 Marc Spindt is out of the office
12:07PM 4 IAX Devices Recommendation
12:03PM 4 *@Home/Grandstream Call Transfer
12:01PM 4 test message - ignore me
12:00PM 0 Polycom IP500 Ringtone howto
11:53AM 4 g729 liscence question
11:46AM 7 List
10:47AM 2 IAX2, can't receive calls
9:53AM 13 two UA with the same usr/pwd
9:10AM 1 Warning: We're Zap/XX-1,
8:17AM 1 sip+nat+asterisk
6:39AM 0 How to force Requested transfer capability on BRI/PRI dial?
5:49AM 3 Is this maillist down?
2:39AM 0 iax2 trunking issues
1:32AM 0 announce-holdtime+ACD+asterisk
12:51AM 0 iax cdr problem
12:37AM 0 Music on hold problem.