asterisk users - Jan 2004

Saturday January 31 2004
TimeRepliesSubject
7:39PM 16 Dial via sip gateway?
5:01PM 8 SUBSCRIBE in chan_sip - anyone?
3:20PM 9 rtp sound quality?
2:29PM 0 echo cancellation disabled
1:58PM 0 The future of VoIP regulation (in the US)
1:48PM 0 Dial app does not indicate ringing to calling party
10:49AM 1 Are there any list moderators?
10:20AM 2 TE410P E1 PRI problem
8:23AM 3 Caller ID Presentment on PRI...
7:46AM 0 Using an additional modem to get CallerID information
6:50AM 3 SIP gateway question
6:17AM 1 asterisk php status viewer
3:02AM 2 newbie thinclient env
12:20AM 2 smtp question
 
Friday January 30 2004
TimeRepliesSubject
11:40PM 1 Cameron Palmer / voiceglo
9:21PM 8 Internal Lines Dialing Out
9:14PM 3 Max messages in VoiceMailMain
8:34PM 2 Question on setting up asterisk with hunting lines
7:48PM 0 Voicemail not receiving password or audio
7:01PM 0 Re: DISA and authcodes (was: t410p)
5:15PM 3 Voicemail/Playback Questions
4:51PM 5 Words for Allison(?)
3:52PM 6 determining legal VoIP service
2:28PM 4 Help!!!: test Asterisk error: error on IAX1.conf and warning on chan_iax2.c
2:22PM 3 Call quality questions
2:17PM 1 SNOM 200 question
2:10PM 7 Compiling zaptel
1:47PM 0 recorder
1:34PM 3 How do you turn on the 7960 msg waiting light?
1:00PM 2 Extension Questions
12:57PM 0 X-Lite, X100P, and Speex
12:54PM 0 error on IAX1.conf and warning on chan_iax2.c
11:53AM 0 Newbridge Mainstreet 3624
11:07AM 11 MeetMe Video option
10:28AM 2 has Allison said this ?
10:21AM 26 Compiling while * is running
9:55AM 1 SIP Transfer problem
9:34AM 10 Adtran 750 DID question.
8:39AM 0 newb info needed
8:23AM 10 PHP developer Wanted ! :-)
7:59AM 3 IAX1 vs IAX2 for IAXtel
6:40AM 7 IAX call problems
6:31AM 8 P2P RTP without SIP re-invites
6:20AM 6 Auto dial in Off Hook situation.
6:07AM 1 mediatrix, dtmf
6:04AM 1 ZAPRTC load error
5:58AM 0 billing software
4:57AM 31 Calls dropping off
4:06AM 2 Asterisk with a laptop with built-in Intel 537 modem
3:33AM 2 Music on Hold Warnings
12:58AM 14 Can Asterisk act like a normal sip phone?
 
Thursday January 29 2004
TimeRepliesSubject
10:25PM 0 Suggestions for debugging an IAX1 one way audio problem
9:01PM 0 canreinvite and codec negotations...
7:05PM 7 Grandstream Firmware ?
5:54PM 3 TDM400 FXO???
5:10PM 16 How to delay dialing
4:30PM 4 Expire old voice mail messages, et al
3:43PM 2 WTB E100P or better
3:08PM 13 Echo worsens in 0.7.1
2:43PM 1 Duplicates ?
2:17PM 0 ZAP answering
12:50PM 1 T1 PRI question
11:46AM 4 Incoming Voice/Fax Discrimination?
11:17AM 0 IAX provider in Colorado
11:05AM 1 Re: Asterisk and gnugk (bam)
10:37AM 0 Automatically Logging Out Queue Members
10:02AM 1 Migrating home POTS VM to Asterisk VM
9:57AM 1 Stable branch - any update available?
9:19AM 0 Re: Asterisk-Users digest, Vol 1 #2663 - 14 msgs
9:18AM 1 IP Blue Skinny Client with Asterisk
9:16AM 10 Multiple Line Appearances
9:12AM 6 IAX Implementation Problem
8:28AM 0 Compatibility with Dialogic D4/PCI
8:01AM 12 good job on the list server!
7:57AM 1 SIP URI matching
7:33AM 4 dialing wrong numbers
6:27AM 2 Send DTMF tone Like "C" on connected call
4:51AM 0 Cdr call transfer
4:45AM 0 Register to h323 gk
4:39AM 6 Asterisk Manager Interface notes
4:39AM 1 WG: Reference projects which using Asterisk !?
4:33AM 3 small correction
4:29AM 1 re: help with voicepulse connect IAX2
3:47AM 0 DTMF wrongly recognised
2:42AM 1 running asterisk under root
2:34AM 14 asterisk with big number of extentions.
2:33AM 2 Running Asterisk on FreeBSD.
1:56AM 0 Asterisk Configuration + MySQL
1:41AM 0 asterisk as B2BUA with SER?
1:20AM 0 Latest Article on Asterisk in the press
 
Wednesday January 28 2004
TimeRepliesSubject
11:33PM 2 specific to X100P with UK telephone lines
9:58PM 0 Channels
8:21PM 0 lists.digium.com
8:18PM 5 Does anyone manage the wiki?
7:10PM 3 SIP error
5:47PM 0 DTMF tone stops working with IAX (voicepulse)???
5:14PM 2 festival patch missing in latest CVS or stable build
4:05PM 1 List traffic
3:47PM 9 cdr_addon_mysql compile error
8:53AM 0 test list
3:54AM 1 IAX2 / SIP testing
 
Tuesday January 27 2004
TimeRepliesSubject
10:56PM 1 Cisco 7960 Problems
10:08PM 1 Asterisk Appliance
9:49PM 1 Distinctive ring Issues
8:19PM 0 ztdummy won't compile
7:16PM 0 X100P doesn´t hangup
6:11PM 17 Introducing Firefly
4:29PM 1 One2One application?
2:32PM 0 Rolm PBX + Asterisk
11:38AM 0 Anyone with experience with polish PTSN?
10:50AM 0 H 323 + Netmeeting test drive
6:18AM 0 Ethereal IAX2 support
5:16AM 0 differentiate incoming calls on SIP clients
2:41AM 0 CDR records on call transfer
2:40AM 0 Asterisk and BRI ISDN
 
Monday January 26 2004
TimeRepliesSubject
8:13PM 0 Is this list down?
7:43PM 0 "NOTICE: Fax detected" on 3'rd DTMF digit on outgoing call to outside line.
7:22PM 0 I need some clarification on DTMF
6:45PM 0 Anyone run * on OS X ?
6:06PM 0 ip num of incoming sip calls?
5:39PM 0 canreinvite and codec negotations... and NAT
5:36PM 1 Is there a way to transfer a call from CLI
5:34PM 0 # transfer with IAX2
4:00PM 0 ADPCM support with RECORD FILE
2:56PM 2 RE: Bluetooth discussions (quick glance to some BT products)
2:22PM 0 Call Queue wait times
1:38PM 1 AVM C4 with Asterisk in front of an ISDN PBX
1:29PM 0 Not sure what I'm looking for to ask correctly.
1:18PM 3 7960 Problems
12:18PM 0 IAX2 hold
11:46AM 0 Asterisk RPMS Updated (FC1,RH9,RH73)
11:30AM 1 SIP behind NAT - use of "externip" option
9:40AM 0 Scalability/Benchmarks/Performance
8:43AM 2 TE410P on Redhat 9
8:29AM 0 Detect Answering Machine in Outgoing calls
7:01AM 6 X-Lite & Asterisk: Speex & iLBC not working?
6:53AM 3 Questions regarding new echo cancellation features...
6:02AM 0 Know if a call is answered
5:59AM 3 app_queue and dialplan
5:33AM 0 Digium FXO Card
5:20AM 0 SIP - fax / voicemail
4:11AM 1 ZAP Problems
3:52AM 2 Need Europian vendor for Digium hardware.
3:36AM 0 He really doesn't care
2:53AM 1 Wildcard X100P usable in Germany?
2:19AM 9 GSM modems
1:50AM 0 GSM phone to *?
 
Sunday January 25 2004
TimeRepliesSubject
11:56PM 3 looking for iax termination
6:38PM 4 OH323 doesnt hear ringing
5:17PM 3 Using TDM400P for autodial
4:12PM 2 Example of TDM20B
3:51PM 3 Ne machine, build is breaking
10:19AM 3 iax1.conf ??
6:01AM 2 Incoming SIP matching
12:48AM 1 Announcement: Another test release of chan_sccp
 
Saturday January 24 2004
TimeRepliesSubject
11:22PM 21 Sipura 2000 Transmit Issues? No Sound being passed to caller
10:44PM 0 FW: one way choppy sound problem !
9:37PM 1 billing systems
9:14PM 0 Ring/Off-hook Message
1:17PM 3 Incoming DID call Voice Problems
12:49PM 1 Warning /:Asterisk.c:255 Listener : Select Returned Error
12:15PM 2 Asterisk RPMS for RH9 + RH7.3
10:25AM 52 Has Nufone gone belly-up
9:40AM 0 SAFE_ASTERISK DIED - EXIT CODE 127
9:27AM 8 Bluetooth discussions
6:04AM 0 IAX hard phone
5:48AM 2 Subject: Re: Grandstream 100 sidetone
5:21AM 3 Searching the archives - new engine demo
4:17AM 0 strange problem with grandstream software 1.0.4.39
2:53AM 1 Is there any plans for Digium ISDN BRI card?
2:03AM 2 MSN 6.1
1:26AM 12 retrans_pkt: Maximum retries exceeded on call
12:30AM 5 Grandstream 100 sidetone
 
Friday January 23 2004
TimeRepliesSubject
11:35PM 1 exten=>h and ResetCDR
11:27PM 0 SIP + ADPCM
11:04PM 3 Problem installing Asterisk with Mandrake 9.1
9:35PM 1 Capabilites of Asterisk
6:35PM 6 RFC3389 support issue with DG104S
5:37PM 5 echo problems
4:30PM 0 Multiple voices on 64K channel (was) simple question...
4:15PM 1 DG104S firmware has error?
3:50PM 0 Excternip and FWD
2:25PM 0 Troubles with the System Attendent Patch.
1:27PM 6 Back to front logging for calls placed through /var/spool/asterisk/outgoing?
1:06PM 2 MI2
12:40PM 9 Mediatrix 1204 sip experience?
12:03PM 6 Latest cvs * compile error anyone?
11:39AM 2 PSTN incoming - both SIP & H323 always arrive in default context :-?
11:02AM 0 SIP wierdness after upgrade from 0.7.1 to CVS
9:51AM 7 UK BT Interface with asterisk?
9:49AM 10 rc.local dont works
9:08AM 5 Debian Packages and Mirrors
8:59AM 2 compiling * pipe error
8:30AM 24 8 lines - best approach
8:23AM 0 New Asterisk article on O'Reilly's onlamp.com
7:33AM 4 SIP Absolute Timeout
7:08AM 1 TE410P/Zaptel
5:22AM 1 Asterisk + Dialup Modem
5:16AM 1 Voicemail2 & Mysql Connection
4:50AM 1 Buying asterisk?
3:00AM 5 SIP register/auth with Grandstream BudgeTone-100
2:48AM 2 chan h323 Compile problem
1:33AM 7 AW: I got it (was: Cisco 7940 with asterisk)
12:36AM 1 USB headset recommendations
12:00AM 2 Maillinglist as newsgroup ?
 
Thursday January 22 2004
TimeRepliesSubject
10:38PM 2 Polycom Reboot Script - Please wiki-size me
10:11PM 4 MGCP w/8x8 DTA-310 and as5300 pstn gateway
8:55PM 1 simple question...
8:32PM 3 Asterisk vs. Websphere Voice Response?
8:25PM 0 voiceglo.com and dtmf
4:30PM 0 Source code for Iax Phone (the new IAX client) now available
3:43PM 3 Variable to play all gsm files in a directory?
3:37PM 0 Rtp WARNING Messages on the Cli in safe_asterisk
3:35PM 2 MGCP Problem.
3:14PM 1 Grandstream transfer solution + DTMF translation possible?
2:17PM 0 is the mike back on?
2:04PM 6 Snom 200 phones not working.
12:40PM 0 Draytek SIP phones are broken
12:38PM 4 ETSI PRI ISDN Signalling
11:44AM 0 Problem with flashing FXO callwaiting from FXS
11:06AM 2 Using varables in MeetMe?
9:36AM 2 sidetone issue
9:30AM 0 Integrating * with a legacy Nec NEAX 1400
9:07AM 0 hardware compatibility
8:54AM 0 Grandstream 10
8:45AM 2 OT: Canada's Primus introduces SIP localserv ice
8:43AM 0 RE: Asterisk-Users digest, Vol 1 #2588 - 11 msgs
8:39AM 0 Cause of transfer problem (GRANDSTREAM!)
7:39AM 0 * and rh9 boot problem
6:25AM 10 Gsm + snom phones
6:13AM 3 Grandstream 101
5:48AM 12 asterisk 0.7.1 - mysql
5:15AM 0 Switchboard interface
5:13AM 0 Description of Manager events
4:04AM 1 Call Queue with no agents -> Congestion or voicebox instead of MOH?
4:01AM 3 R2 or E&M for E1 CAS pbx to pbx link
3:31AM 0 DIAX CallMe feature
3:07AM 17 Data calls (ISDN/64k) through * PRI
2:45AM 11 Standalone FXO device
1:35AM 0 Codecs and more analog lines?
1:07AM 4 chan_capi: suppress calling number on outbound dialing?
12:40AM 21 Asterisk 0.7.1 RH 7.3 RPMS Released
 
Wednesday January 21 2004
TimeRepliesSubject
11:34PM 0 Asterisk Management Interface... Do you want one?
8:56PM 0 X100P remote hangup detection problem
8:08PM 1 Transfer problem
8:07PM 2 Diax IAX2
8:03PM 0 Asterisk on news.com w00t
7:15PM 0 CRM Solutions - Incoming Calls & Outgoing Calls
6:56PM 0 Net2Phone error 407: Unauthorized
6:48PM 2 Polycom Soundpoint IP400
6:26PM 1 need help configuring IAX to make outbound calls through a remote server
6:17PM 1 mp3player not working
5:44PM 14 Mailing List Lag
5:01PM 13 New Windows IAX Client
5:00PM 0 G729 Codec Error
4:35PM 1 OT: Canada's Primus introduces SIP localservice
4:33PM 3 disable transfer on outgoing calls?
4:10PM 3 Fax problem
4:09PM 0 Conf files
2:53PM 1 Strange Zaptel Modprobe driver failure
2:51PM 0 Asterisk at Open Source-Telephony-Summit 2004
2:34PM 20 Digium X100P for $43
11:08AM 1 Lucent Definity + Asterisk Success!
11:07AM 1 h323 with innovaphone ip 400 gatekeeper/innovaphone Ip200 phones
11:05AM 0 Fw: Word-of-the-Day: wiki
10:38AM 2 OH323 config file format
10:16AM 2 Starting with MGCP and Asterisk
10:07AM 1 Sip phones transfer not working.
9:58AM 5 What technology could my phone company be using?
9:36AM 4 Is there a way to # of agents logged into a queue ?
9:23AM 1 PrePaid App.
8:16AM 1 help request about wiax2.dll
8:13AM 1 zaptel - problem solved
7:51AM 3 zaptel part II
7:41AM 3 Making a call with sample.call
7:12AM 0 Modulated tones not working
6:41AM 1 Parked Calls Settings
6:39AM 1 ISDN p2p AVM Fritz Card
5:34AM 1 Reorder tone ...when it should be Busy...
5:23AM 0 Mediatrix 1104 register problem ?
2:46AM 10 Calling Card Application
2:00AM 0 Different classes of MusicOnHold
1:31AM 1 Zap show channel
12:05AM 1 Hi
12:05AM 2 CAS SF Inband tone signalling problem
 
Tuesday January 20 2004
TimeRepliesSubject
11:59PM 2 Brandwidth for making internet calls
11:40PM 8 how scalable is digium cards?
11:16PM 0 Asterisk-Driven Linux Distro
9:57PM 2 Toll-Free Gateway Beta Test: freenum.org
9:29PM 0 Out of trunk data space
8:20PM 2 Can asteric be used with just a voice card
8:09PM 1 VOIP on linux
7:23PM 1 G729 - how many needed?
6:35PM 0 Play volume/speed adjustment on the per call basis
6:12PM 4 Restricting/Negotiating H323 Port Ranges
6:06PM 0 installing Asterisk on FreeBSD
5:33PM 0 Voice Pulse Connect problems Authing
4:37PM 2 T400P / T100P with Hong Kong IDA-P Lines
4:30PM 0 Dialplan Sanity Check please
3:47PM 18 I got it (was: Cisco 7940 with asterisk)
3:47PM 1 PRI NI2
3:40PM 0 New Swissvoice ip10 firmware: 1.0.0 build 3
3:20PM 1 help - recording both sides of a conversati on
3:03PM 0 chan_capi capiECT
2:12PM 0 Is libpri symmetrical?
2:08PM 2 Cisco 7940 with asterisk
1:43PM 0 FLASH TONE
1:25PM 2 OT: Canada's Primus introduces SIP local service
1:24PM 0 Agent timeout then Dial() ?
1:11PM 0 Grandstream cfg.txt hacking?
1:11PM 1 Music on Hold - can it be done without mpg123?
12:06PM 0 [A-bit-OT] Power Over Ethernet Discovery process
11:45AM 2 How to diagnose "pops" and "clicks"?
11:42AM 18 G.729 Licenses from Digium
11:37AM 2 DTMF A-D
11:28AM 16 MeetMe questions
10:57AM 0 Power Over Ethernet for *any* ethernet switc h (or hub); product idea
10:44AM 0 Power Over Ethernet for *any* ethernet switch(or hub); product idea
10:30AM 4 CAPI: Early-B3 working with AVM-B1?
10:12AM 4 PSTN Gateway
9:51AM 2 multipledaily digest!!!
9:51AM 4 Enter Pin followed by Pound key
9:20AM 0 ADSI phone vs. IP phone (and proper implementation thereof)
7:59AM 0 wink time
7:52AM 1 Compiling problems with SuSE
6:58AM 0 SIP: outbound calls
6:53AM 0 AG4000C and T100P
6:41AM 0 Broken macros during transferring call
6:05AM 12 Re-Invite between SIP phones
5:27AM 3 Still problems at compiling
3:18AM 1 open h323
1:59AM 18 Power Over Ethernet for *any* ethernet switch (or hub); product idea
12:43AM 0 Outbound call with Go2Call
 
Monday January 19 2004
TimeRepliesSubject
10:31PM 0 Call token is ip$localhost
9:20PM 1 FW: Memory problem
8:16PM 4 CVS Changes (NAT-SIP)
3:12PM 4 PLAYBACK multiple files
1:42PM 4 Getting correct CDR info
11:55AM 0 Routecall application
11:16AM 3 Lucent and ISDN-PRI
10:57AM 2 RE: current version
9:52AM 4 Residential services
8:59AM 3 Different Caller ID for each Zap Interface
7:28AM 3 Concurrents calls on asterisk with H323
6:36AM 0 Re: [Asterisk-Dev] benevolent dictatorship, or inclusive developper community?
6:20AM 1 pri gateways and asterisk
5:06AM 1 Transferring H.323 Call
4:34AM 3 Hangup detection failed
4:33AM 3 SIP: Register that isn't a register?
4:30AM 0 Best Codec ?
4:28AM 0 MGCP: condition 14?
4:19AM 7 Search engine for this list
4:11AM 0 words for Alison
3:56AM 0 Dialogic cards with Asterisk
2:44AM 1 Connecting BRI to PRI card?
1:42AM 7 configuration to Grandstream via tftp
12:50AM 11 IAX2 bug in DIAX solved - Great Thanks to Steven!
 
Sunday January 18 2004
TimeRepliesSubject
9:55PM 3 ATA-186 pass-through Flash
9:28PM 3 Calls with incoming distinctive ring
8:44PM 1 California DID Access
8:16PM 5 RE: current version
6:28PM 21 [ot] Grandstream hardware
5:54PM 5 Now: Small Biz Robust Asterisk Solution - SBRAS
4:40PM 0 Re: newbie ISDN question
2:22PM 0 Office-wide paging with Asterisk and Cisco 7960 7940 phones
2:15PM 14 Latest version of asterisk
1:11PM 1 Asterisk switching ISDN data calls?
1:08PM 2 Asterisk as SIP Redirect Server -- Implemented - Not Working - Plz Help
11:45AM 1 Public switches (AXE10) not capable of handling sustained call setup bursts on E100P
11:01AM 4 Re: ultra-cheap asterisk box -> Small Biz Robust Asterisk Solution - SBRAS
9:00AM 0 RE: Newbee question (hot phone numbers)
8:33AM 2 Nufone not taking GSM CALLS
7:51AM 1 minimum system hardware for Asterisk install
7:31AM 1 No startup after mpg123 install
12:04AM 35 ADSI phone vs. IP phone
 
Saturday January 17 2004
TimeRepliesSubject
10:16PM 3 Channel Bank Woes...
9:04PM 29 Asterisk Indications
8:48PM 1 X100P Configs for Australia
8:03PM 0 Remote reloading Cisco phones...
7:56PM 1 Registering multiple FWD accounts
7:42PM 0 Unavailable versus private in extensions.conf?
6:49PM 24 New sounds also now in CVS
4:54PM 4 cdr_odbc not logging integers eg duration
12:54PM 0 New sounds posted
11:57AM 0 Asterisk/X100 - Sipura Configuration
9:15AM 0 Combining 2 AUDIO Frames
9:10AM 0 Long time to detect that an IAX2 user is not logged on
7:37AM 5 Playing background message
7:32AM 0 Kernel for 1586 vs i686 for Asterisk?
7:06AM 7 Zone Paging
6:56AM 0 Early B3 on PRI channel (Listen free mex without charge)
5:03AM 18 SS7 over Asterisk ?
3:18AM 2 Voicetronix OpenLine4: disable answering on a particular channel & delay before dial
 
Friday January 16 2004
TimeRepliesSubject
7:58PM 5 Class features in dialplan ?
7:34PM 0 stuck on retransmit
7:29PM 2 7960 Phone disconnects when dialing using speaker
6:52PM 0 Re: Asterisk-Users digest, Vol 1 #2525 - 11 msgs
6:50PM 0 Updated zaprtc, anyone interested?
6:24PM 1 G.729 quiestion
5:06PM 1 Caller id and callback
4:10PM 1 doublehash patch doesn't work in asterisk 0.7.1
3:59PM 1 Analog phone help
3:55PM 4 Hardware for Asterisk
1:48PM 1 ERROR[8192]
1:46PM 1 Asterisk Integration with Lucent Definity g3 si
1:43PM 0 re: hardware requirement -asterisk
1:10PM 0 RoutCall Info
1:10PM 3 VoiceMail - no user pre-registration
12:30PM 5 G.723.1 codec
12:13PM 0 Analog phone transfer
12:10PM 0 sound of static removed by hitting flash button
11:25AM 1 Configuration for modem
11:22AM 0 ultra-cheap asterisk box -> sorta OT, more a bout Dell
11:13AM 13 Remote reload Cisco 7960
10:59AM 1 Advice Request: 2-4 line, 10 station * system
10:52AM 4 Asterisk Integration with Lucent Definity g3si
9:59AM 2 NO DTMF detection in the Outgoing call with GW Cisco5300
9:05AM 2 'Intercom' before call transfer
8:45AM 0 SMP kernel with X100P card
7:07AM 7 CAPI not installed, after changed from i4l to CAPI
6:09AM 1 CDR problem with macros
5:46AM 0 Meetme sound dropped ? Out of trunk data space ??
3:53AM 2 Asterisk over WAN
3:30AM 4 ISDN30 - HW ?
3:28AM 0 GS Handytone Echo-problem
1:48AM 5 Odbc not logging
 
Thursday January 15 2004
TimeRepliesSubject
11:03PM 3 Sending voicemail with qmail
10:19PM 3 vmail cgi script
9:41PM 1 meetme - ztdummy
9:35PM 0 T100P & CA Adit 600 Question
9:16PM 0 FW: Sending voicemail with qmail and call waiting
8:50PM 6 SER & Asterisk
8:36PM 0 announcement using Dial
7:19PM 7 WANTED: Toll-Free gateways in Europe/Asia/Africa/South America
7:06PM 1 Help! Asterisk 0.7.1 No Sound in recorded gsm files
6:18PM 4 meetme without zaptel hardware
4:12PM 1 Voicetronix Openline 4 + asterisk
4:03PM 0 Ringback Problem
3:34PM 2 re: hardware requirement -asterisk
2:57PM 6 Voicemail Sequence Bug?
2:19PM 3 re: hardware requirement asterisk
2:18PM 6 Re Grandstream 1.0.4.38
1:56PM 0 t1xxp Unable to request IRQ
1:22PM 1 SIP Phones - Power over ethernet?
1:14PM 2 [OT] Commercial conferencing solution?
1:09PM 6 QoS anyone?
12:43PM 1 want suggestion to get hardware to learn
12:39PM 24 capacity testing
12:38PM 0 Possible Bug: Crash when Parking Calls
12:32PM 0 establishing before answer "Early B3" libpri
12:26PM 0 cdr processing
11:16AM 0 Free Message Signaling
11:13AM 1 Credit Card Terminal
11:04AM 5 Cisco FXO as PSTN gateway (updated request for assistance)
10:37AM 0 Users in sweden
9:49AM 0 IOS Bug crashing SIP nat sessions
9:41AM 4 People detected as fax machines
9:14AM 12 Disturbing trend of * production boxes that shouldn't be
9:08AM 22 ultra-cheap asterisk box
8:49AM 0 Choosing a VoIP Protocol.
8:39AM 11 wav49 voicemail problem with Windows Media Player
8:12AM 0 ATA186 SIP Outbound Fax Calls
8:00AM 0 ISDN newbie
7:37AM 1 Problem at compiling zaptel (again)
7:19AM 1 Problem at compiling zaptel
7:13AM 3 Strange sound when fax answers (app_rxfax)
6:12AM 5 asterisk.org webpage
5:54AM 5 B-channels restart problem
5:23AM 0 best SIP-softphone?
3:48AM 0 Parking extension:700
3:39AM 3 ISDN CAPI and anonymous callers
1:55AM 0 SIP clipping sound
1:48AM 1 GSM connection for asterisk
1:07AM 0 H.323 protocol security vulnerability
12:44AM 0 AW: Re: Again: 7920 Cisco IP Phone Skinny & SIP
12:22AM 2 hardware requirements - asterisk
 
Wednesday January 14 2004
TimeRepliesSubject
10:38PM 8 Codec matching weirdness
10:37PM 5 re hardware requirement - asterisk
9:08PM 7 * For Call Center
8:55PM 1 Re Hardware requirement -Asterisk
7:46PM 0 unload chan_zap.so now possible :)
7:12PM 2 Single/Dual DS3 - anyone seen this?
7:06PM 1 zaptel compile erro!(asterisk last version0.7.1)
6:51PM 1 Cooperate with SIP ITSP
6:21PM 1 System Attendent
6:14PM 0 MeetMe, conferencing questions
5:51PM 0 asterisk & faxing
5:18PM 0 Windows Call Manager : Formerly [Asterisk-Dev] New Bounty
5:04PM 3 Static Noise coming from Wildcard FXS: Wildcard TDM400P
2:37PM 2 Skinny behind NAT?
1:33PM 0 hungup problems with SIP and x100P card
1:01PM 5 100% of cpu in an out of the box *
12:26PM 2 Asterisk as a protocol converter?
12:09PM 1 Daytime/nighttime broken in asterisk-0.7.0?
11:46AM 1 How do we updated to the new .7.1 version.
11:30AM 0 BOUNTY POSTED - Zaptel drivers for *BSD
11:15AM 6 SNOM IAX image
10:22AM 4 Basic Asterisk capabilities question
10:06AM 0 Re: Proposed solution for exit code priority jumps
9:59AM 0 Re: failover (was Re: voicepulse)
9:19AM 2 Re: failover (was Re: voicepulse)
8:52AM 1 DTMF Debug
8:50AM 6 Asterisk 0.7.1
8:41AM 4 NAT friendly TFTP Server
8:09AM 4 Multiple phonenumbers on one E1 PRI with Digium TE410P ?
7:46AM 1 ... H323 - segmentation fault - core dumped
6:52AM 0 TDM switching between Digium TE410P ports
5:40AM 11 grandstream asterisk configuration
4:48AM 2 Asterisk drops calls - E100P
4:45AM 6 How to park and pickup a call
1:41AM 2 hardware requirements of asterisk
1:36AM 1 always 4 rings before * answers!?
12:56AM 26 Why I can not use the conference
12:22AM 0 Kernel 2.6 and ztdummy?
 
Tuesday January 13 2004
TimeRepliesSubject
9:07PM 0 Asterisk in Linux Journal
8:20PM 9 linux journal article on asterisk
8:04PM 0 H.323 security flaws article
7:48PM 7 Re: Proposed solution for exit code priority jumps
7:23PM 2 How can I get support about Dialogic hardware
5:07PM 17 Parking extension not working
4:48PM 0 Fun (or lack of) with asterisk & T100P
3:26PM 0 * and signaling (clarification)
2:53PM 2 Mediatrix 1102 issue after upgrading to CVS
2:09PM 0 Turning up the volume on outgoing sip to sip gateway calls?
1:57PM 0 SIP test suite: anyone with spare time?
1:02PM 1 max queue time; newbie question (fwd)
11:28AM 0 OH232: cancel call does not stop ringing
11:25AM 0 * and telco ringback
10:58AM 0 inbound call routing problem - RESOLVED
10:41AM 3 threewaycalling ? (Bridge 2 SIP calls?)
10:19AM 1 E100P works with PCI 3.3V and 5V?
9:59AM 0 agents and call queueing
9:55AM 6 SIP and AGI crash...
9:50AM 0 Memory allocation issues
9:43AM 0 New software SIP phone released today
9:38AM 0 Cisco Multiple Products H.323 Protocol Denial of Service Vulnerabilities
8:50AM 0 CVS problem
8:13AM 1 24x7x365 asterisk support available?
8:12AM 1 Specifying a codec to be used in /etc/sip.conf
7:52AM 4 inbound call routing problem
7:36AM 10 Voicepulse
6:53AM 13 How to Order Disconnect Supervision from SBC using Adit 600?
6:49AM 9 Again: 7920 Cisco IP Phone Skinny & SIP
6:41AM 1 Symbol NetVision Phone
6:27AM 0 sme
6:11AM 3 Asterisk and Festival (* dies with no info)
4:48AM 17 Best Linux Distribution
4:46AM 1 GUI client for windows for live monitoring/b arge
4:42AM 2 cisco 7910 phone
4:28AM 2 E100P without q931?
3:40AM 3 KPhone working
3:31AM 13 FS/OS Telephony Summit 2004
12:26AM 5 Nufone.net net wackiness?
 
Monday January 12 2004
TimeRepliesSubject
11:35PM 0 0.7.0 Release Mirrors
11:30PM 1 newbie to asterisk
11:10PM 18 Asterisk 0.7.0
10:23PM 0 sip and x-lite
7:22PM 0 Does asterisk surport old voice card?
7:02PM 2 GUI client for windows for live monitoring/barge
6:47PM 3 MeetMe issues?
6:28PM 4 Fw: problem with safe_asterisk
5:58PM 1 Asterisk Voicemail that reacts to my AIM status
5:16PM 2 Re: Nauti miles
4:01PM 13 3.3v PCI board - TE410P photo
3:11PM 1 Release 0.7.0 - not yet
3:01PM 0 Grandstream and VLAN
2:41PM 0 FW: How to bind RTP when IP alias are configured
2:37PM 0 Disconnect Supervision, SBC, and Adit 600
1:42PM 2 New Installation problem
12:57PM 0 Wiki - stable -dev
12:41PM 0 Turning a profit (WAS: More words for Allis on)
12:27PM 2 ADSI. used beyond own phone network?
11:37AM 0 Routing packets in and out
11:34AM 0 Fw: Forward call with response required to accept
11:16AM 0 Voice Caller Name and NUmber?
10:33AM 2 How to bind RTP when IP alias are configured
10:17AM 2 LCR / Trollphone Rate Engine
9:46AM 0 Fax handled on E&M T1 DEBUG messages
9:45AM 4 Issue - vmail.cgi on Redhat 9 (Apache) ?
9:40AM 5 RFC3389 messages with ATA 186
9:38AM 1 Advance Options in VoicemailMain() ?
9:31AM 4 Thank You All
9:31AM 3 Securing Cisco SIP gateway
8:27AM 2 '*' call conference?
8:23AM 3 Linux Sip UAs
8:16AM 1 CAS Idle definition bits ?
8:13AM 1 E100P - connected to Cisco
7:27AM 0 Fax on Cisco ATA
7:20AM 3 host=dynamic and defaultip=xxx
7:18AM 0 Using the FLASH key on MGCP sends DTMF to the third party
7:12AM 1 Message signaling on MGCP (light on the phone)
6:42AM 3 Cisco FXO as PSTN gateway
6:09AM 3 bad pstn audio coming from old * processes
4:53AM 5 A question on codec translation.
4:28AM 0 OH323: Dropping incompatible voice frame
3:42AM 1 Install problem (compile error)
3:00AM 4 SIP-Client for Handheld PC
1:06AM 0 Queue timeout problem
12:49AM 4 Bandwidth ? + Doc + cdr
 
Sunday January 11 2004
TimeRepliesSubject
7:17PM 1 Re: Asterisk-Users digest, Vol 1 #2448 - 10 msgs
7:07PM 1 possible solution to PRI T100P dropped call issue
6:50PM 0 WTS (200) AC Power Adapters for Cisco 7910 / 7940 / 7960 IP Phones
6:34PM 2 Forward call with response required to accept
6:01PM 1 zttool and errors
5:36PM 56 More words for Allison
4:58PM 0 NuFone Network H323 configuration?
4:41PM 26 T1 Sync clarification
4:10PM 0 "friendly" dial tone frequency combinations
4:07PM 9 analog or sip ? was far end disconnect supervision
3:18PM 22 Asterisk on FreeBSD 4.9?
3:12PM 3 CONTEST: Top Posters win 80G Hard Drive
3:09PM 0 Asterisk on FreeBSD 4.9
12:33PM 2 SpeakFree
10:24AM 1 Re: [Asterisk-Dev] More Success on the Cisco 7920 and SCCP !!!!!
9:55AM 0 Strange problem with call hangup on Budgetone 102 Phones
9:47AM 3 macro error "exited non-zero"
9:27AM 6 Cisco 79xx Ringtones
8:54AM 2 New Version of SJPhone
4:51AM 1 More Success on the Cisco 7920 and SCCP !!!!!
3:36AM 3 High Level of CVS activity
1:50AM 0 a constructive proposal: tie the marshals to a cvs server
 
Saturday January 10 2004
TimeRepliesSubject
11:24PM 6 WTB / WTS Voip hardware
10:13PM 0 how do i make this happen [macro-record-cleanup]
9:26PM 0 Using ACD functionality for main number answer and "music on hold"
8:45PM 10 Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)
7:29PM 7 default music source for SIP channel
6:55PM 0 REQ: init/startup scripts for asterisk for possible inclusion in version 0.7.0
5:25PM 0 Record calls where to put line?
4:07PM 19 Asterisk + BudgeTone (behind NAT)
3:48PM 0 My first E1 card is running :)
2:19PM 3 drop calls with T100P / PRI
2:03PM 13 far end disconnect supervision
1:26PM 1 ADSI Configs
1:04PM 0 IAX v1 Changes
12:34PM 10 Free Software or not -- that's the question /* New subject */
11:11AM 0 Re: E1 - E100P connected to Cisco - problem - HELP
11:09AM 0 Bridging ethernet over hdlc
11:01AM 2 Record all phone calls
10:55AM 6 R2 Digital - Brazil
10:22AM 1 Oops!
8:39AM 11 E100P - Error 500
8:28AM 2 Forums Need Help
5:39AM 0 FYI: New SIP Flash Image 7940/7960 IP Phone
3:22AM 0 Call transfer message
2:04AM 0 Music_on_hold adjust volume
12:57AM 1 picking a channel bank
 
Friday January 9 2004
TimeRepliesSubject
11:48PM 4 Re: Asterisk-Users digest, Vol 1 #2413 - 13 msgs
11:29PM 0 AW: Problems with Cisco 7920/Skinny/Asterisk
10:57PM 20 Chagres Technologies, Inc
9:43PM 0 more VoIP news - "wire"taps
9:40PM 4 Called Party Identification
9:37PM 2 crontab
9:37PM 1 (no subject)
7:55PM 7 newbie question; can * screen calls?
7:25PM 0 (no subject)
6:49PM 3 ChanIsAvail and SIP
5:45PM 1 chan_iax2.c Ignoring Port For Now
5:14PM 7 file_inlcude .. why not?
5:06PM 0 Slightly OT: CNN: FCC cautious on Voice-over-Internet regulation
5:02PM 4 Fwd: new cvs build failure
4:51PM 0 new cvs build failure
4:18PM 3 Broken DNS makes Asterisk whacky!
4:13PM 1 Asteriks as SIP<>H323 Proxy?
3:55PM 1 At last!!! :)
2:38PM 1 Screen Pop & Remote Agents = Telemarketing
1:20PM 2 * as sip b2bua?
1:18PM 3 Why * try to codec translate when it can do without during codec negotiation.
12:44PM 0 soft fax machine
12:31PM 1 zapbarge w/o the mute
11:50AM 14 USA dial plan
11:46AM 0 IConnect audio quality
11:32AM 3 Screen Pop & Remote Agents
11:09AM 0 SIP/2.0 487 Request Cancelled
11:00AM 7 Cisco Gear
10:49AM 0 Problems with Cisco 7920/Skinny/Asterisk
10:42AM 2 * dialing before line is open?
10:34AM 2 DTMF in MeetMe
10:23AM 2 Help with compiling
8:22AM 2 log incoming and outgoing call
7:42AM 3 Very high delay
7:34AM 0 Receiving faxes from a SIP gateway
6:36AM 1 Development Process comment and Email list suggestion
5:51AM 0 SV: Mailing list growth
4:11AM 2 asterisk sip with voicemail
4:08AM 6 A question about Linux kernels and Asterisk
4:07AM 2 max queue time; newbie question
2:18AM 0 DTMF through H.245 UserIndication
 
Thursday January 8 2004
TimeRepliesSubject
11:42PM 3 SIP reload configuration problem /* New subject */
9:32PM 1 GrandStream giving an RTP Read Error Again
8:35PM 3 Progress on the Polycom front...
7:09PM 8 Dialing the Phone from OS X Address Book with AppleScript, XML-RPC, PHP and Asterisk
7:01PM 0 Hangup detection issue
5:05PM 0 iaxtel iax.conf entry?
4:52PM 0 RE: [Asterisk-Dev] Asterisk Development Updates
3:03PM 0 Error messages during Registeration on CVS Version CVS-01/08/04-14:20:34
2:44PM 9 Asterisk Development Updates
2:24PM 0 Re: Sound Card help -- Solution -- and a question
2:09PM 4 latest cvs == broken tdmoe
1:58PM 7 Kedpad less extension
1:54PM 4 Dial from command line?
1:20PM 0 Getting VoicePulse # listed in 411
1:19PM 2 SIP URI's: possible now?
12:03PM 0 Clicking until DSP called
12:02PM 5 Some bugs www.voip-info.org html
11:30AM 1 Re: 911 and lawsuits and redundancy
11:12AM 0 Asterisk success stories in small-mediumoffi ce environments?
10:41AM 0 E100P - R2 support
10:29AM 5 T100P positional on PCI bus?
10:27AM 6 2nd call leg status?
9:09AM 4 Asterisk success stories in small-mediumoffice environments?
8:45AM 1 Cisco tftp
8:35AM 10 Asterisk hanging?
8:26AM 1 Nortel Option 61C PBX?
8:21AM 1 Billing system experiences with Advanced Communications?
8:03AM 0 BudgetTone 102D
6:26AM 16 AbsoluteTimeout Users Messages
6:22AM 2 Phonejack
6:02AM 7 Asterisk & Sipura 2000
5:36AM 4 Administrative suggestions
4:40AM 2 Strange Call waiting problems - SNOM 200 & Grandstream Budgetone
4:25AM 1 AW: IPv6 support
2:33AM 0 Asterisk Capability Module ?
2:12AM 1 E100P : Pb with outgoing calls
1:48AM 33 Mailing list growth
1:46AM 2 Red Alarms - FXS(Signalling Q)
 
Wednesday January 7 2004
TimeRepliesSubject
11:42PM 10 SIP and error talking to voicemail
10:56PM 8 IPv6 support
10:23PM 0 " Error in buffer handling Message"
7:40PM 0 2.4 Kernel and Hyperthreading (was Re: P4 processor
7:39PM 0 RE: Inexpensive Analog Ports
6:03PM 4 * and Cisco Gateways
5:59PM 0 Frazzled newbie questions
4:33PM 4 PRI D Channel and Caller-ID issue......
4:13PM 1 Re: 911 and lawsuits and redundancy
3:52PM 0 DTMF via SIP not working for certain phone systems
2:04PM 2 E1 - E100P connected to Cisco - problem
2:01PM 4 yet another question on DID trunks
1:16PM 2 Test Post-Do Not Read
1:04PM 2 Asterisk success stories in small-medium office environments?
12:56PM 4 Voicemail account size limit ?
12:56PM 6 Newbie Question-Looking for Feedback
12:32PM 1 (newbie) Hardware sizing question
11:46AM 2 zaprtc install problem
10:24AM 3 A Note to GS users..
10:01AM 5 Call Rollover
9:38AM 3 (no subject)
9:36AM 6 * crashed
9:24AM 0 IAX2 missing?
8:59AM 0 Asterisk log messages
8:44AM 0 Re: 911 and lawsuits and redundancy
7:25AM 1 DTMF recognized improperly?
6:59AM 20 Asterisk + fax
6:42AM 6 Client for P800/P900
6:24AM 2 P4 processor with Hyperthreading and Asterisk
6:04AM 1 Unexpected ISDN hangup on outbound call
5:06AM 5 manipulating with numbers - StripMSD, Prefix
2:27AM 0 Asterisk stops responding after about 80 calls
12:10AM 0 Small scaled VoIP calling card system
 
Tuesday January 6 2004
TimeRepliesSubject
10:11PM 0 Asterisk interop with Syndeo
9:12PM 5 no results.
6:46PM 0 Voicemail scalability
6:40PM 0 Asterisk support for NEON or Centrex/CLASS/VMWI MW formats ?
4:47PM 0 HTML tags?
4:41PM 2 IAX2 Trunk two Asterisk boxes.
4:15PM 17 Re: 911 and lawsuits and redundancy
3:41PM 10 ATA call
3:19PM 14 MWI message not seen on SNOM200
2:49PM 10 benevolent dictatorship, or inclusive developper community?
2:45PM 3 Doorbells & Door Intercoms
2:17PM 0 Request for Information on Asterisk Functionality
1:55PM 1 Need Cisco 7940 or 7960s at good price for Asterisk deployment
1:25PM 5 Voicemail to email file sizes
1:24PM 1 Fw: Pls confirm
12:30PM 2 911
12:30PM 0 Call Transfer Function in *
12:25PM 7 Heads up v2.03h on snom 200
12:15PM 6 Scaleable Solution for small office
12:06PM 2 Hpw to enable Voicemail Indicator on IP/Analog Phone ?
11:00AM 5 Pls confirm
10:23AM 1 Got SIP response 482 "Loop Detected"
9:56AM 29 911 and lawsuits
9:16AM 0 Re: Multi-line help & AOL Messenger Style PBX Navigation
8:58AM 1 ring tone
8:37AM 0 [Fwd: reject connect from iaxtel.com]
8:35AM 0 small question from a new user
8:17AM 4 URGENT - micronet & asterisk on h323
7:35AM 2 How to flash hook when there is no hook ?
7:17AM 7 Asterisk feature list: spreadsheet
7:15AM 3 cant load drivers for TE410P cards
7:08AM 0 Asterisk Nat Issue
6:42AM 3 Policies - deny some nubers
5:56AM 4 AGI Scripting
5:16AM 2 Problems compiling cdr_pgsql
4:45AM 2 FW: Matrix Orbital (usbl LCD or VFD) (oops, wrong list I think)
4:24AM 1 Call Queue and Agent Statistics
4:17AM 4 Asterisk not working with session border controller
12:53AM 1 IVR Question
12:09AM 6 "Everyone is busy at this time" message ?
 
Monday January 5 2004
TimeRepliesSubject
11:48PM 0 Interfacing Asterisk with PSTN network (Nortel SL100 PBX)
10:24PM 2 I stumbled on this list...
9:26PM 0 Special variable for AGENT
8:08PM 0 problems dialing area code
6:07PM 3 Message waiting indicator
5:45PM 1 Identifying the Originating Cisco SIP Gateway
5:42PM 0 asterisk sccp support
5:41PM 2 How to monitor calls initiated by .call file using manager interface?
5:19PM 4 Echo on polycom sip phone
5:17PM 4 This is a test
3:50PM 0 Need Help...
3:35PM 0 Lindows ?
3:23PM 7 Are messages censored on this board?
3:13PM 1 FW: This newbie gives up for now - sadly (2)
2:55PM 0 HTML Stripping in mailing lists?
2:28PM 3 Echo with polycom phones
2:28PM 3 question re voicemail
1:50PM 1 reject connect from iaxtel.com
12:46PM 0 queue questions: max time in queue; customer option to drop out of queue
12:44PM 19 This newbie gives up for now - sadly
11:50AM 0 Codec Negotiation Does not seem to work as e xpected ?? Help Please !!
11:44AM 0 mailbox= wrong context. was: Newbie - MWI
9:36AM 1 Question about MP3's
9:24AM 13 Sip Trunking
9:18AM 0 Hardware to build an Enterprise AsteriskUniversal Gateway
8:28AM 5 DID Trunk Lines and Caller ID
4:18AM 1 CLIR and isdn4linux
3:06AM 0 FW: SIP to SIP redirect while ringing
1:39AM 3 "Internal" ISDN bus
12:38AM 8 RE: Inexpensive Analog Ports
12:29AM 3 Codec Negotiation Does not seem to work as expected ?? Help Please !!
 
Sunday January 4 2004
TimeRepliesSubject
9:16PM 9 Grandstream Handytone 286 RTP Problems
7:37PM 2 Hold and transfer problem
7:36PM 1 Voicepulse DID fast busy
7:31PM 1 4 X100P Cards
7:25PM 5 Sun Servers with UltraSparc Processors
7:11PM 1 pager reminder script
5:52PM 0 Dutch/DTMF Caller ID
5:45PM 12 Cisco to Cisco - poor quality
5:31PM 2 Earpiece Connections
5:18PM 11 Multi-line help
4:04PM 2 Cisco 12sp+ program update
3:07PM 18 Newbie - MWI
11:09AM 3 OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany?
10:11AM 2 Two E100P boards - could not load zaptel module - Channel 63 - no such device
10:07AM 2 Voicemail Out call
4:46AM 4 POTS interfacing recommendation
4:33AM 1 Modem Communications thru *
3:35AM 38 Hardware to build an Enterprise Asterisk Universal Gateway
3:25AM 0 TDM400P & X101P cards, echo issues?
3:18AM 14 CAPI, transfering thru a 2nd PBX - keep original CallerID
 
Saturday January 3 2004
TimeRepliesSubject
10:09PM 6 AW: AW: Snom 200 with two extns defined anyone?
8:59PM 0 TDM400P driver modprobe failed
7:31PM 3 STOP THIS THREAD New to asterisk? RUN... don't walk.
4:29PM 0 Free PSTN calls
12:05PM 0 expression parsing
10:14AM 1 Newbie - getting two local phones tocommunicate would be a good start :)
 
Friday January 2 2004
TimeRepliesSubject
9:42PM 1 Asterisk Gotoif / last called
8:05PM 0 Newbridge Mainstreet 3624 Manual
6:48PM 3 Cisco SIP license?
3:57PM 20 Newbie - getting two local phones to communicate would be a good start :)
2:59PM 0 Grandstream Flash Button
1:42PM 2 AgentCallbackLogin.
1:00PM 2 Newbridge Mainstreet 3624 T1 channel bank no w Alcatel
11:34AM 5 mini-ITX suggestions
11:25AM 15 hangup detection
10:28AM 3 T400P & E400P second source
9:17AM 2 Malloc debug kills asterisk?
8:27AM 1 asterisk dies while making calls
7:24AM 3 * Stresstool Help required
6:59AM 3 Slow wiki?
5:35AM 0 SQL Updater Down!!!
4:23AM 4 License questioni supose ??
2:23AM 17 one way choppy sound problem !
1:18AM 0 IAXy Release ?
12:31AM 2 FW: [Asterisk-Dev] Strange bells in voicemail and voicemailmain ?
 
Thursday January 1 2004
TimeRepliesSubject
9:48PM 3 sound driver advise needed
8:44PM 2 How to load the driver of TDM400P card!
7:17PM 2 Prediction for 2004
12:31PM 2 Newbridge Mainstreet 3624 T1 channel bank now Alcatel
11:25AM 5 * crash when forward voicemail --Nicolas Gudino
10:27AM 12 help
10:04AM 1 asterisk gateway to other gateways
1:41AM 1 asterisk reload for FWD to register