asterisk users - Feb 2004

Sunday February 29 2004
TimeRepliesSubject
10:22PM 1 Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk
9:00PM 1 AGI/php help needed with variables
8:21PM 0 RE: Problem connecting to ASkterisk Server
8:11PM 0 can * support ANI
7:40PM 0 Background and multiple digits
7:28PM 8 Asterisk rpm packages
7:13PM 3 freeBSD zaptel driver
6:54PM 0 Testing ENUM
6:46PM 1 Hangup to CDR recording timing
2:15PM 6 vmail.cgi -> .php?
1:25PM 2 Unable to specify channel 1: No such device or address
12:29PM 36 Brazilian Protocol
10:22AM 0 TDM400 problems
9:49AM 2 Re: Outgoing parallelism
7:22AM 6 Dialing out after caller leaves message
6:18AM 1 outgoing spool parallelism
 
Saturday February 28 2004
TimeRepliesSubject
9:10PM 1 Load average ...
8:26PM 21 Asterisk on Feebsd , pls. HELP !
7:54PM 2 If one extension is busy...
7:40PM 1 A working number at enum.fierymoon.com?
6:39PM 40 Hotel wake-up
5:15PM 2 PCphoneline FXO to FXS box??
5:04PM 0 zaphfc bri: crackling sound
4:22PM 2 iaxComm updates at sourceforge
3:31PM 0 Help needed setting up H323 gateway.
2:53PM 1 New to T-1/Channel Bank hardware -- help?
2:34PM 1 zaphfc bri with overlap sending/receiving
7:22AM 0 How to compile bri-stuff.0.0.2.rc12
7:17AM 1 sip:user@domain.tld
6:57AM 0 Galaxy Voice - Good or Bad?
5:46AM 5 OTish: Firefly Crashing with *
5:45AM 3 Asterisk PABX switch
4:40AM 0 Cisco 7960 sip v6.2 is out
1:41AM 6 Iconnect behind NAT
 
Friday February 27 2004
TimeRepliesSubject
10:24PM 11 Video Conference
9:47PM 1 CISCO ATA 188
8:02PM 4 DTMF Issues with SJPHONE
7:10PM 0 H323 SETUP ON ASTERISK??
5:35PM 1 cvs update and new x100p cards broke menu playback
5:26PM 1 outdial broadcast
3:51PM 0 Mediatrix 1204 FXO GW ring cadence question
2:39PM 5 retrieve_sip_conf_from_mysql.pl data format
2:28PM 0 Fujitsu 9600
2:02PM 0 Snom 200 Map key lights problem.
1:37PM 0 Asterisk as proxy?
12:26PM 0 Voicemail cutting off messages on SIP
12:11PM 0 Setting up with an Eicon DIVA PCI card?
11:54AM 0 budgetones + G726
9:51AM 2 FXO Gateway of choice is?
9:46AM 11 Anybody managed to call a phone through sipgate.de
9:42AM 0 WTS (20) ATA-186, Various Cisco IP Phones VOIP Gear and P/S
9:42AM 2 wisip firmware, updates, features??
9:32AM 2 Core dump crash
9:24AM 0 Routing NOTIFY SIP messages
8:54AM 0 Agent Queuing on multiple machines
8:52AM 2 Lucent Definity CallerID {Scanned}
8:52AM 6 USB Phones
8:52AM 1 Remote retrieval of voicemail, a question
8:00AM 0 IAX Phone Update - Slight Change
7:26AM 1 Best VOIP Analog adapter ???
7:15AM 0 IAX Phone Bug Fix
4:30AM 0 Request for enhancement - IP dependent ports
3:27AM 2 Failed to start asterisk
3:26AM 0 Re: Asterisk-Dev digest, Vol 1 #507 - 8 msgs
3:25AM 3 chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy
3:24AM 0 chan_sip support for SIP:Remote-Party-ID, sp ecificallyCLID priva cy
3:03AM 1 RE: Message waiting light not coming on
2:53AM 3 Problem connecting to Asterisk Server
 
Thursday February 26 2004
TimeRepliesSubject
11:58PM 4 registration
11:45PM 0 New Insatll of *
9:25PM 1 Does Digium TDM400P + X100P make a switchboard?
8:41PM 9 exit
7:24PM 3 Big Install examples please
7:08PM 0 (OT) HOWTO: 802.3af POE w/ 79xx
6:44PM 3 MWI false light activity - msg0000.txt
6:29PM 1 Delta Three/iConnectHere Outgoing Caller ID?
5:58PM 2 GS Budgetone 101 canot receive calls
5:37PM 3 Lucent Definity CallerID
3:46PM 0 chan_h323 chan_oh323
3:13PM 0 A newbie list question
2:52PM 1 record application in extensions.conf -- how to stop recording?
2:49PM 1 Off topic question
2:12PM 3 chan_sip support for SIP:Remote-Party-ID, specificallyCLID priva cy
1:11PM 1 Re: Grandstream -> firefly call translator problem
1:08PM 2 DTA-310 Outbound Dialing
11:18AM 1 callerid will not be set
10:28AM 0 relaxdtmf - duplicatedigits
9:53AM 0 chan_capi 0.3.1 segfault backtrace
9:28AM 3 Can You Specify Codec Per Extension?
9:08AM 1 chan_sip support for SIP:Remote-Party-ID, specifically CLID priva cy
9:07AM 0 Any schedule for Digium's TDM400 FXO modules, or for IAXy?
8:52AM 0 C7-Hardware
8:09AM 5 Asterisk Venture
8:04AM 23 Simple Front Gate Intercom
6:52AM 0 Distinctive ring on 2 channels?
6:16AM 0 Video Recording
6:04AM 2 Connecting an ISDN DECT phone base
3:49AM 0 RE:Poor Voicemail / Ivr announcement quality
3:07AM 1 SIP Extrange Problem
 
Wednesday February 25 2004
TimeRepliesSubject
11:42PM 2 sip router gateway
11:12PM 0 cannot configure voicemail with mysql
8:36PM 1 how to route based on DNIS
7:18PM 0 Shopping for ISDN card
7:01PM 0 Macro Forward Calls
6:55PM 1 Newbie Qu.
6:18PM 5 Patching Asterisk for OpenH323 ASN.1 Vulnera bilities
6:07PM 3 7960 x SCCP/Skinny (off-topic)
3:14PM 29 Grandstream transfer into outer space
3:01PM 5 Patching Asterisk for OpenH323 ASN.1 Vulnerabilities
12:48PM 5 Calls always parked on 701
11:58AM 0 Some ADSI script programming guide?
11:13AM 5 Web based UA
10:44AM 0 Unreliable Fax
9:15AM 4 Detection of extension
6:26AM 9 Comments on Voice Quality IP Hard Phones
6:03AM 5 Problem with SIP 407
5:48AM 0 segfault with alsa
4:35AM 1 Need some information
2:43AM 4 cisco 7912 problem with chan_sccp
12:01AM 4 dial plan question
 
Tuesday February 24 2004
TimeRepliesSubject
11:40PM 1 Re: 12SP registration
10:55PM 0 parking ?
9:58PM 4 cdr->dst incorrect?
8:45PM 0 Need Origination Number form Bahamas
8:03PM 1 T1 inbound dialplan
7:56PM 0 Avaya question
7:32PM 1 Transferring Incoming Calls Twice
4:22PM 1 "CLASS" codes or "VERTICAL SERVICE CODES"
3:47PM 2 Can't compile Zaptel drivers
3:26PM 1 painful first steps with * and SIP phones
3:23PM 6 Incoming context based on ISDN MSN
3:08PM 2 VoIP Reseller Programs
1:57PM 1 Vegastream 50 BRI
1:53PM 0 InfoElement in chan_capi
1:31PM 19 Simulating the "lighted line in use" type of phone
12:40PM 0 Echo on Diva Server 2M with melware CAPI
11:00AM 5 DSL (DMT) goes down when X100 plugged in
10:34AM 0 FCC forces 911 services
10:13AM 0 RE: qview.pl
9:56AM 1 SIP Re-Invites & Timeout
9:27AM 0 CISCO 7912 SIP Problem
9:25AM 1 Cisco 7940/60 Intercom
8:46AM 8 Anyone working with NUFONE?
8:18AM 0 Re: IAX Voicepulse - no DTMF response
8:17AM 15 calls dropped with grandstream
7:00AM 0 system speed dial
6:42AM 1 RFC 2833 / Timestamp
5:51AM 2 Changes in capi.conf
5:08AM 9 Understanding AgentCallbackLogin
4:51AM 0 more codec negotiation problems
4:10AM 0 Flash application on H323
12:58AM 1 codec translation
 
Monday February 23 2004
TimeRepliesSubject
10:26PM 0 Job Opening
9:42PM 2 Asterisk and Faxing
9:26PM 2 line status
6:20PM 0 Asterisk Hardware Solution
5:56PM 0 Cisco Kit for Grabs
4:07PM 2 cdr_addon_mysql problem linking
4:03PM 9 Nested include statements in extensions.conf?
3:59PM 0 *8# and zaphfc in NT-mode
3:47PM 0 Asterisk and Multicast
3:03PM 3 A missing argument
2:40PM 1 SIP Codec selection order
2:39PM 1 12SP
2:05PM 0 calling between two zap points with zaphfc
1:51PM 1 ztmonitor and the x101p
1:19PM 3 VM: Multilanguage and digits
1:10PM 14 Unable to create channem of type 'Zap'
12:51PM 2 SIP over NAT
12:29PM 0 DevLite problem with ztcfg
12:25PM 1 Call Groups and outgoing line selection
12:09PM 1 Queue Modified ACD for Asterisk 0.7.2
12:03PM 6 Codec Order / Preference
11:57AM 3 Pickup
11:53AM 1 Confusion with IAX PBX-PBX
11:37AM 5 Dual Xeon
11:28AM 3 SPA 2000 ringing
10:56AM 1 Minimum voice mail message limit?
10:05AM 0 Attended Transfer Question
9:15AM 5 Thread-safe applications
8:25AM 4 An example config for using a Wildcard X100P and a SIP phone?
8:19AM 4 Processor load spikes
7:45AM 2 Re: Asterisk-Users digest, Vol 1 #2879 - 10 msgs
3:25AM 0 Re: How to best debug SIP registration failure (Solved)
1:41AM 3 About Grandstream ATA-286 and ring voltage
1:31AM 1 SIP overlap (early dial) 484 response
1:05AM 3 Hyuandi pstn handsets
 
Sunday February 22 2004
TimeRepliesSubject
8:23PM 2 IAX2 Call menu handling problem with Norstar
5:44PM 1 Sip Register Fail - NAT
3:38PM 2 Cisco AS5350 Gateway Intergration
1:47PM 9 OT: SNOM and TAPI
12:31PM 7 How to best debug SIP registration failure
10:21AM 6 2 questions about ISDN BRI
9:46AM 0 What is the best way to debug the DTMF tones on a Zap interface
9:43AM 0 app_directory.c
8:04AM 2 X100P and DTMF sending
6:46AM 0 SEGFAULT (capi amd hfc-s NT)
6:16AM 6 "multicasting" conference calls
3:46AM 3 oh323 codec negotiation
1:04AM 4 Pingtel Opensource PBX Announcement
 
Saturday February 21 2004
TimeRepliesSubject
11:19PM 2 7960 - multiple lines - sip.conf
9:35PM 0 iaxtel to asterisk config help.
8:16PM 3 Voicemail Follow-up
6:44PM 8 SIP extension "busy" when not available ??
5:09PM 9 Using Festival inside an agi
11:33AM 4 New Wiki page: Dimensioning an Asterisk system
10:37AM 2 "Call did not go through"
9:43AM 0 Chan_skinny
9:32AM 3 Using * for large VoIP implementation
8:01AM 1 Re: System call forked - more stuff
7:08AM 3 System called seems forked up
6:07AM 0 Subject: Grandstream / SIP <-> IAX2 / Voicepulse
5:44AM 1 Connection Problem - GrandStream
5:35AM 3 Simple Call Center Setup?
5:25AM 1 switch ivr menu on/off remotely?
5:04AM 1 Voicepulse Connection
1:28AM 0 HFC-Cologne TE mode, non-I4L mode.
12:27AM 0 T100P & T1
 
Friday February 20 2004
TimeRepliesSubject
11:32PM 16 Agents / ackcall
11:09PM 4 Call Redirection
9:13PM 0 Termination in Phoenix
9:06PM 0 Voice mail sound distortion has everyone laughing
8:16PM 1 TE410P PRI T1 problem
4:43PM 2 Not Woodpeckers
2:33PM 0 Problem playing the first voice mail prompt
1:50PM 0 Re: System call succeed, asterisk sees failure
1:41PM 0 Setup ?
12:48PM 0 Connecting Nortel DMS100 to Asterisk ?
12:44PM 0 T1 Ground Start signalling support ?
12:38PM 1 multiple lines on 7960's
12:27PM 0 Disabling echo cancellation on fax ?
11:54AM 0 POTS Distinctive Ring
11:42AM 0 Firefly and IXATEL
10:48AM 1 System cmd usage
9:51AM 0 Snom 100 + H.323
9:22AM 2 Skinny and SIP
7:40AM 0 Zaptel BRI and HFC-S cards in NT-Mode - Dialout problem
6:49AM 0 SwissVoice IP10S can't take back call
6:42AM 3 RE: codec negotiation prob solved
4:13AM 15 voicemail not working with mysql!!!!
3:39AM 0 g729, g723.1 codec translation costs
1:58AM 0 tapi for asterisk*
1:20AM 8 Are IAX2 providers ready for prime time?
12:24AM 10 INFO/DTMF retransmissions in * not absorbed?
 
Thursday February 19 2004
TimeRepliesSubject
8:23PM 0 Minor update of Firefly
6:30PM 0 codec negotiation prob solved?
5:45PM 0 app_sql_postgres doesn't clean up
5:19PM 19 IAX Connection - Voicepulse
4:00PM 23 Woodpeckers
3:31PM 0 WARNING[1142106560]
3:20PM 1 Configuring Pingtel Xpressa
3:15PM 0 moh badness: mpg123 0.59s
3:12PM 2 * dropps outbound calls over PSTN
3:07PM 0 Language setting
3:01PM 0 Re: Zombies got me! - Fixed!
2:53PM 1 International PSTN dialing
1:07PM 1 Re: Zombies got me! - Fixed!
1:06PM 2 Make a phone dial remotely?
12:23PM 1 SIP Behind NAT (sipgate.de)
12:12PM 0 Caller ID Oddity
12:12PM 1 mgcp endpoint question
12:11PM 2 faxing with asterisk
11:51AM 4 Bizarre ring
11:17AM 1 compiling gastman
11:12AM 1 Registering Polycom IP 500 with Asterisk [re vised]
11:02AM 0 Registering Polycom IP 500 with Asterisk [revised]
10:52AM 0 EAGI errors?
10:15AM 0 Enumlookup and Asterisk
10:04AM 0 FW: H-324M/SIP Gateway
9:54AM 1 dtmf recording record and playback
9:39AM 0 sip - t.38 through asterisk
9:38AM 0 Zombies got me!
9:03AM 0 Spanish newbie need help
8:38AM 0 SIP Stuff
8:21AM 4 Enhancements Coming To VoicePulse Connect!
8:12AM 0 H-324M/SIP Gateway
8:01AM 15 Cisco 7960 SIP image (off-topic)
6:27AM 1 Problem with call to IAX
6:25AM 0 check if que has members
6:24AM 8 Zaptel BRI and HFC-S cards in NT-Mode
6:11AM 6 Mac X-Lite and Asterisk
4:38AM 2 IVR (does not exist in any format)(No such file or directory)
3:56AM 0 pri error
3:44AM 0 Flash to PBX with X100P - how to?
3:33AM 0 audiocodes and asterisk interoperability
3:26AM 0 Help a Newbie to conf softphone
3:11AM 2 X100 => T100 Upgrade
1:27AM 3 4 port FXO
12:54AM 3 help a poor newbie out with SIP choppy one-way problem
 
Wednesday February 18 2004
TimeRepliesSubject
10:57PM 3 vm to email ?
7:43PM 1 Gastman doesn't draw lines properly between resources ...
7:28PM 1 Pingtel SIPxchange IP PBX goes Open Source...
7:00PM 0 running script based on if a peer is up or down
6:33PM 0 VoicemailMain2
6:24PM 0 Anyone use the Cisco 12SP+ phone w. asterisk?
6:12PM 0 Microsoft Portrait 2.2
4:36PM 2 OT: Cisco 79XX operation
4:10PM 0 sip-to-sip hangup problems?
3:28PM 0 Re: Asterisk-Users digest, Vol 1 #2849 - 12 msgs
3:24PM 0 Re: Asterisk-Users digest, Vol 1 #2849 - 12 msgs
3:22PM 3 ADSI ports
2:45PM 2 IAX2 dosent work for DIAX
2:31PM 2 Call Pick on Cisco 7960's
2:01PM 3 Round-robin chan_zap groups...
12:25PM 0 Red Alarms and mysterious disconnects
11:31AM 3 softphone configs?
11:07AM 6 Budgetone phones from FWD
10:55AM 2 Where to download app_pgsql
10:44AM 2 Memory usage
10:36AM 0 * nat internet nat sip phone howto
10:11AM 5 agi scripting in perl - dealiing with unexpected disconnects gracefully / spurious DTMF
10:09AM 1 IAX2 queue member no auth found
9:40AM 0 Typo fixes
8:29AM 0 mysql_freinds
6:15AM 5 Callerid & AGI Thougts
4:41AM 4 Executing external script
1:10AM 2 Help setting up asterisk
 
Tuesday February 17 2004
TimeRepliesSubject
10:51PM 3 pstn 800#'s
10:00PM 2 FXO gateways on Asterisk
9:52PM 0 How to Dial-out from USR Voice Modem
9:10PM 4 DIAX does not receive calls...
8:16PM 1 Specials on VOIP Gear
7:06PM 0 Background command, error
7:03PM 3 Howto apply a patch, diff file
5:43PM 1 Wanpipe cards?
5:28PM 19 SIP config documentation
4:54PM 0 Weird sdp output
4:46PM 1 Inbound IAX to SIP
1:42PM 0 Problems to register with SIP provider
1:31PM 3 Telemarketer handling
1:21PM 1 Marketing collateral, etc.
12:22PM 2 Re: Asterisk-Users digest, Vol 1 #2840 - 11 msgs
12:12PM 0 G729 and VoiceMail
12:09PM 1 Asterisk quietly segfaults
12:02PM 7 Dell 1750 server and Asteriks...
11:52AM 6 T1 Help
11:03AM 1 phpagi DIAL command not working
8:57AM 5 Record communication
8:44AM 3 x100p dropping incoming calls
7:54AM 29 max asterisk load
7:45AM 3 Double digits seen using Grandstream phones
7:03AM 1 Anyway to automate an AgentCallBackLogin
6:54AM 0 nasty segfault with previous strange Zaptel warnings
6:17AM 0 Incomming Distinctive ringing
5:55AM 0 Fw: voicemail extension - hangup
5:48AM 0 Queue timeout reason
5:10AM 3 extravagant behavior, nat problem ?
4:41AM 4 Mailbox full ?
2:06AM 5 IAXTEL and the registration traffic
2:04AM 3 PRI error or what?
1:42AM 1 How can you savage a failed call transfer
1:33AM 12 chan_capi problem
12:59AM 6 Analog Cordless Phone Recommendations
12:55AM 0 RFC: Some proposals for the list digests
12:48AM 0 (no subject)
 
Monday February 16 2004
TimeRepliesSubject
8:59PM 2 Asterisk monitor with Daemontools
7:07PM 4 cannot find -lXext when building * ?
6:51PM 15 Room Monitor
6:09PM 0 2 Asterisk Boxes 2 Dev Kits
5:30PM 1 Asterisk for a call center?
4:16PM 0 What can cause a Red alarm?
3:16PM 6 Got my DID, getting an error.
3:16PM 0 Upgrading asterisk yields broken pipe
2:57PM 0 X100P analogue cards and impedance matching
2:04PM 0 FS: Adtran TotalAccess 850 Channel Bank,Router,4x4FXS
12:51PM 1 HFC-S cards?
11:55AM 0 Asterisk - Carrier Access Bank Ring through
11:43AM 0 Eicon Diva Server card, where to purchase ?
10:07AM 0 New to the list -> some (unsolved) questions
10:05AM 14 Speech between Grandstream phones sounds like talking under water
9:12AM 0 Agent / Queue help
8:11AM 0 Good source for moh files
8:05AM 0 IaxTel: Using IaxTel Numbers As Asterisk DIDs
7:57AM 0 SIP Messages (SIMPLE)
7:13AM 3 VOIP Carrier recommendations?
7:10AM 1 Cisco 30VIP Phones
6:41AM 0 Mailing list lag again
6:39AM 5 Analogical FXO vs. BRI dialing speed
6:22AM 0 voicemail extension - hangup
5:15AM 4 ZapRAS + RADIUS authentication
4:51AM 0 re: SIP 481 subscription does not exist with SJPhone
4:50AM 4 Zhone + call transfer
3:57AM 0 LDAP authentication
3:01AM 0 re: SIP 481 subscription does not exist with SJPhone
2:20AM 0 Re: Asterisk-Users digest, Vol 1 #2827 - 16 msgs
2:10AM 21 Need to interface to BRIs
 
Sunday February 15 2004
TimeRepliesSubject
8:59PM 1 merlin legend / * as ld gw
7:25PM 1 Call File Troubles
7:00PM 2 Asterisk and Vonage -- no make that VoicePulse Connect :)
2:39PM 0 Official word from GalaxyVoice customer service
2:00PM 2 Asterisk and Vonage -- possible? Other DID providers with Atlanta presence?
11:58AM 0 Festival patch ?
11:12AM 0 Correct cvs checkout?
9:17AM 3 Pingtel Phones?
8:34AM 0 Looking for Incoming # for Area Code 713 (Houston, TX)
5:38AM 35 Wifi Phones
1:56AM 0 WTB: Grandstream Budgetone
 
Saturday February 14 2004
TimeRepliesSubject
11:37PM 2 Music on Hold - Context
10:07PM 4 Get new PRI working
10:03PM 0 Xlite, GSM, Agent Logins and DTMF
6:13PM 5 TE405P and dual Athlon systems
5:40PM 0 Siemens Gigaset not ringing the cordless phjone for very long
3:46PM 3 Asterisk - oh323 - Cisco CallManager
1:59PM 8 running asterisk as non-root
1:59PM 0 Incoming SIP-calls and Festival
1:41PM 1 Festival: read text from external fil
1:39PM 0 Asterisk and "dial by email?"
11:31AM 0 with soekris?
10:38AM 0 Is there a MaxQueueTime for Queues ?
10:10AM 3 Kansas SIP or IAX Provider?
5:40AM 0 CallerID or Noise ?
3:47AM 0 FWD/Iaxtel/Asterisk codec use
12:47AM 19 Voip in the EU
 
Friday February 13 2004
TimeRepliesSubject
11:41PM 0 Translator 'g729tolinb'
8:09PM 0 RE: Rhino channel bank and aastra PT390 phones
6:33PM 2 GSM codec with Cisco equipment
5:56PM 0 Problem with * - Nat - Internet - Nat - X-lite
3:37PM 11 HELP!!!! Having problems Starting Asterisk
1:33PM 1 GS BT-100 echo
1:32PM 1 Switch brands, speeds, etc.
12:12PM 3 RE: Rhino channel bank and aastra PT390 phones
9:14AM 2 chan_local and variables
8:07AM 3 Codecs compile error on yellowdog
7:46AM 1 Re-Invites and Studder.
7:45AM 0 Wierd Zap Channel Behavior
6:54AM 16 Digium connectivity issue?
6:34AM 2 Adtran 750 - what do I need
4:17AM 1 Spanish indications configurationÂș
4:13AM 10 Hide outgoing CallerId on Zap interface
3:59AM 0 festival in agi?
1:47AM 2 multiple context in sip.conf
1:44AM 2 channel bank - Adit 600
 
Thursday February 12 2004
TimeRepliesSubject
10:52PM 0 [OT] Looking for Manual: Clarent CPG 101
6:48PM 9 X100P / Echo / ZTMONITOR CAN2,3, etc.
6:15PM 4 Direct mailbox transfer
5:11PM 4 x101p beeps/sceeching
4:54PM 9 Anybody going to the Spring VON converence [ OT]
4:28PM 1 Sip problem with IpDialog phone.
4:23PM 1 More external call control
3:30PM 2 Voicemail Password Digit Timeout
3:29PM 2 Anybody going to the Spring VON converence [OT]
3:19PM 0 Why does the DG104S keep sending?
3:12PM 1 AudioCodes MP-104, register
12:56PM 2 Jitter Buffer Configuration (typo in iax.conf)
11:50AM 0 Database items
9:31AM 5 festival voices
8:53AM 1 Playing GSM files(s)
8:52AM 2 billing question
5:44AM 0 Specify address with base=0xNNNNN
4:59AM 0 Mailing list search engine
2:03AM 1 setting up callback
 
Wednesday February 11 2004
TimeRepliesSubject
9:30PM 2 New Zealand
8:46PM 0 Asterisk hangs up when a call comes in
8:08PM 0 "Integrated" T1 PRI (voice and data)
7:27PM 1 Force SIP Phones to Register
4:04PM 3 Asterisk and Wildcard T100P
3:51PM 1 Mediatrix 1204 sip g/w now working
1:59PM 0 Please Explain newchan->pvt->pvt
1:52PM 0 Asterisk Critical Mass: Thursday, Miami, 9:00 PM
1:02PM 2 Constant crashes with Asterisk 0.7.2
12:25PM 1 speex with VoicePulse
11:15AM 2 asterisk-oh323 new update, v0.5.9
11:13AM 4 T1 PRI CallerID
11:03AM 3 "Stuck" TE410P cards
10:24AM 8 TDM card loses Dial tone
9:42AM 5 Can't connect KPhone to asterisk
9:34AM 2 OT: Cisco 7940 Smartnet in the UK
9:26AM 0 Re: Asterisk<->GS and codec selection
9:15AM 1 unable to open ../voicemail/context/exten/msg0000
8:54AM 6 Cisco ATA 186
8:31AM 1 I need patch for musiconhold-multiful format
6:02AM 1 Calling from Iaxtel to FWD users always busy
5:08AM 6 Noise and scratches when there are two concurrent CAPI calls
4:49AM 0 Multiple switch staments
3:21AM 0 [DENICenum-l] Open Workshop on IP voice and associated convergent services]
2:12AM 2 Pls help for Musiconhold
12:41AM 1 Cisco 7960G ordering Question
 
Tuesday February 10 2004
TimeRepliesSubject
11:34PM 1 IAX DTMF question
11:23PM 0 Call center integration - passing caller idinto an external app.
10:22PM 0 Iaxphone problem
9:55PM 2 Residential Plans for Asterisk Users
9:45PM 0 linux 2.6
9:26PM 5 How much processing power is needed?
9:07PM 4 Loading module chan_capi.so failed!
5:01PM 10 Cisco 7960 - how to enable "messages" key
3:24PM 3 I finally did IT!!!! Internal dial tone
3:07PM 7 alert-info and Cisco 7960 phones (6.1)
2:02PM 1 Call center integration - passing caller id into an external app.
1:36PM 4 Wait command in auto attendant causes sched.c error
1:14PM 5 Termination - Cuba
12:38PM 2 TDM400 showing up as Tiger Jet
12:31PM 0 Error Logging (stops Randomly)
12:06PM 5 [Fwd: Having problems with RTP packets and H old]
11:33AM 0 [Fwd: Having problems with RTP packets and Hold]
10:36AM 0 RV: Strange Behaviour with DMZ
9:19AM 4 Callerid detection
8:39AM 0 two phones one host
8:24AM 0 Log entry - solved
8:11AM 2 Log entry
7:53AM 1 Sending DTMF out-of-band over IAX2
7:47AM 0 Make outbound calls only from certain hosts
6:25AM 3 Spurious DTMF tones heard by the person being called
6:03AM 0 Having problems with RTP packets and Hold
6:01AM 5 NIC card failure [was: System freeze]
5:52AM 0 Basic Sip proxy setup question
 
Monday February 9 2004
TimeRepliesSubject
9:44PM 0 Firefly 1.4 released
7:31PM 0 X100P + HP DL380
7:11PM 3 Recording
6:37PM 0 NEC IP phone compatibility?
5:26PM 0 Infite RTP to wrong address from DG104S
4:32PM 0 Re: Asterisk-Users digest, Vol 1 #2785 - 6 msgs
4:13PM 3 Dual line Skinny
4:05PM 0 OS X -- Zaptel
3:01PM 1 Revisit the Cisco 7910
2:47PM 25 Dialing 800 numbers with VOIP
1:59PM 3 X100P Cards have gone belly up?
1:42PM 3 Dial-out and Dial-in modem problems.
1:25PM 0 alternative to mpg123 musiconhold was [Sys tem freeze]
1:15PM 9 asterisk-grandstream call
1:06PM 2 SIP and DTA-310 was [ MGCP w/8x8 DTA-310 and as5300 pstn gateway]
12:57PM 5 Intercom system (not paging system)
12:16PM 10 Calling SIP
11:24AM 3 Unable to create vpb channel
11:20AM 8 Best OS for Asterisk
11:02AM 1 New Firmware for Grandstream Phones -Supports CFG by MAC address
10:21AM 4 Can asterisk make a call to a phone?
9:52AM 3 port number keeps changing
9:45AM 19 System freeze
9:40AM 4 how to password protect a meetme conference?
9:23AM 6 New Firmware for Grandstream Phones - Supports CFG by MAC address
9:14AM 2 incoming call to internal user
8:36AM 12 SIP phones with dual ethernet.
8:35AM 1 OS X -- More Specific
7:46AM 0 /var/spool/asterisk/outgoing issues
6:28AM 7 asterisk and fax over ip - concept
5:29AM 0 long delay before asterisk returns 486 busy with sip
5:07AM 2 asterisk-oh323, new version 0.5.8
4:03AM 0 incoming DTMF on a SIP call
4:00AM 2 Help with Sip call problems - Whats not working?
2:48AM 0 Incomplete dialed number in CDR
2:05AM 0 RTP with ATA186 ?
1:21AM 0 DTMF over SIP to a Cisco gateway
 
Sunday February 8 2004
TimeRepliesSubject
11:12PM 0 Newbie - help
9:42PM 1 Registering SJPhone with Asterisk
8:55PM 0 Call transfer from a queue
8:25PM 1 Motherboard and fxo suggestion.
5:27PM 1 OS X
3:44PM 2 dialout redunancy.
3:07PM 3 Speex == Screech using version 1.1.4
12:16PM 4 Asterisk & Panasonic KXTD - Vonage
11:52AM 0 NanoBGA VIA Eden-N Processor
9:23AM 2 Asterisk pins CPU
5:24AM 0 FW: SNOM 200 silence suppression
12:45AM 1 PCMCIA
 
Saturday February 7 2004
TimeRepliesSubject
9:57PM 3 ringing
8:56PM 13 Problems with ATA's locking up..
6:21PM 4 Snom 200 MWI Button
2:31PM 3 central voicemail with remote offices
2:31PM 2 All incoming Zap calls getting picked up as FAX calls!
1:42PM 2 dial timeout not working
9:05AM 2 play_and_record: No audio available
8:06AM 10 s/asterisk mailinglists/asterisk forum/g ?
5:26AM 0 OpenBSD 3.4 Patching
5:15AM 4 Snom 100 Code Recommendation
1:50AM 1 IAX Softphone Errors
 
Friday February 6 2004
TimeRepliesSubject
10:23PM 16 Caller-ID is being sent wrong. How to fix it?
8:02PM 0 RE:voiceglo sip config
6:02PM 0 Message Not Delivered
4:47PM 2 Asterisk under UML?
4:19PM 1 G.729, show command or log to confirm it's using the G.729 codec.
3:54PM 6 Interrupted musiconhold sound when silence supression is enabled
3:26PM 8 modprobe wcfxs
2:29PM 6 busy status
2:11PM 1 Asterisk on ebay.
1:58PM 4 is it possible to turn auto answer off and on in the dialplan?
1:02PM 1 iax2 jitter stats confusion
12:46PM 4 Annoying Beeps
12:06PM 2 Silencing Background App during touch tone detection
9:37AM 0 passing variables to a macro
9:20AM 14 Conference server
8:53AM 3 SIP - Native Bridge Error
4:06AM 3 DIAX 0.9.6b call reception
2:25AM 3 Trouble emailing Digium
2:18AM 0 Configuring buttons on a CISCO 12SP+ Ip Phone (skinny.conf)
1:44AM 0 ATA in MGCP sometimes dropping calls
1:10AM 0 Re: Asterisk-Users digest, Vol 1 #2749 - 7 msgs
 
Thursday February 5 2004
TimeRepliesSubject
11:48PM 2 chan_sccp: incoming calls on multiple lines
9:30PM 2 Adding another X100P after X100P and TDM400P is already configured
8:32PM 13 zaptel on Debian
8:31PM 1 Sip transfers
8:17PM 3 Re: DISA
7:47PM 0 Current version of gastman precompiled binary
7:14PM 4 ISDN update
7:00PM 5 Re: DISA
6:17PM 0 Re: [Asterisk-Dev] DISA
5:55PM 0 AutoAttendent ON/OFF control by Attendent
5:00PM 0 RE: Apple OS-X
4:16PM 1 Asterisk Randomly Stopping
3:57PM 0 OT Asterisk Sales Questions (Not for Asterisk itself)
3:49PM 1 has Allison recorded "Do Not Disturb"
3:42PM 2 http://www.oneunified.net
3:23PM 7 Voiceglo questions
2:59PM 3 simple test setup
2:45PM 2 Fax with wildcards
2:41PM 8 question for oh323 users
2:33PM 2 Asterisk GUI Client - New verison 0.9
2:24PM 0 sethdlc-new compile, does it?
2:12PM 1 X100P - Asterisk - Asterisk - X100P setup help
12:19PM 1 fwd settings
12:11PM 3 Vegastream 50 FXO with Asterisk
11:01AM 3 Release phone call
10:18AM 0 CallWaiting CallerID: Available on all channel types?
9:14AM 0 The Evil of type=friend explained, again ( wa s Re: Minor Registration Problem With Polycom Soun dpoin t IP 500)
8:26AM 3 (no subject)
8:20AM 2 Record conversation
8:00AM 0 compact fxo device
7:03AM 4 Asterisk as non root
5:36AM 5 Execute command in shell
5:33AM 3 Dialogic D300SC-E1
5:23AM 2 Asterisk + oh323 docs ?
4:10AM 0 H323 calls via provider
2:04AM 2 Data call transfer
 
Wednesday February 4 2004
TimeRepliesSubject
9:51PM 0 Music on hold inside of an agi script whileprocessing programs in background
9:29PM 2 help *** newbie
8:59PM 7 Music on hold inside of an agi script while processing programs in background
4:48PM 1 Audiocodes FXO - loop channels
4:38PM 3 progress on DTMF
2:39PM 0 Integrating with an existing PBX
2:24PM 0 Need knowledgeable comments, bug 981 and dual redirect
1:41PM 14 Asterisk 0.7.2 RPMS Updated
12:45PM 5 Possible Sip logic bug?
12:36PM 4 Adtran 750 Configuration
12:10PM 0 Audio code registration
11:55AM 0 Asuscom HiSax based ISDN BRI card - one way latency
10:45AM 7 Sip flow diagram?
10:32AM 1 New Search engine for the list - Final resting place
10:27AM 3 Interrupted musiconhold sound when silence suppression is enabled
10:11AM 0 Voicemail volume level?
9:51AM 0 7960 MGCP dialtone problems, part 2 [long]
9:50AM 1 7960 MGCP dialtone problems, part 1 [long]
9:37AM 1 Newbie Question. Is asterisk right for my scenario?
8:49AM 0 voicemail auth failure
8:39AM 4 ParkAndAnnounce - Get Parking Extension
8:21AM 4 Asterisk 0.7.2
8:11AM 4 Whats wrong with dialplan?
8:03AM 7 Cepstral TTS Code
7:19AM 13 talking clock
7:17AM 15 Boards falling out...
4:59AM 11 Minor Registration Problem With Polycom Soun dpoint IP 500
3:48AM 0 Newbie: Chan_capi, early b3 in Italy
2:42AM 1 Port bind
2:18AM 21 Code Hosting...
1:57AM 3 Do you Linux softphone..
1:34AM 0 X100P and PSTN line Callwaiting
12:58AM 0 billing information from telecom
12:26AM 6 CALEA?
 
Tuesday February 3 2004
TimeRepliesSubject
11:40PM 5 iax, trunking, etc.
11:40PM 1 Anyone used a Grandstream ATA286 with Asterisk
11:15PM 15 Cisco 7960 bug in 6.1 evident in Asterisk
11:05PM 0 Minor Registration Problem With Polycom Soundpoint IP 500
10:38PM 2 VOIP Deployment Concerns
8:21PM 0 Cisco AC Power Cubes for Sale
7:57PM 9 diax softphone
6:47PM 2 IPKall->FWD->Asterisk
6:17PM 2 Pictures of new multiport FXO/FXS from digum
5:19PM 3 Detecting answer supervison from an AGI app
4:26PM 2 Re: Asterisk-Users digest, Vol 1 #2711 - 15 msgs
4:16PM 0 (no subject)
3:18PM 1 sipphone dialing out problem
3:09PM 10 voip phones
2:56PM 6 [OT] Oldest Telephone
2:17PM 1 RE: Asterisk-Users digest, Vol 1 #2719 - 10 msgs
1:17PM 2 GS and NAT
12:50PM 1 Mediatrix sip fxo gateway workaround?
12:48PM 0 RedHat 9 & VSFTPD & Digium Hardware Oddoties
12:42PM 4 Qualify statement
11:48AM 0 Asterisk compatibility list
11:14AM 4 x100p card conflicts with DSL modem
10:46AM 4 Smallest server continued...
10:46AM 4 sementation fault with mpg123
9:32AM 23 The Smallest Asterisk Server Ever?
9:29AM 0 Transfer of call from a call queue
9:23AM 1 Nortel and Asterisk interconnection
9:20AM 10 Cisco 7960 quick dial
9:03AM 0 Asterisk 0.7.1 RPMS Updated to Rel 4
8:56AM 0 upgrade problems
8:47AM 0 kernel 2.4.x .... which one?
8:24AM 6 SIP debug logs
8:17AM 4 Using a Dial Statement with option m and t
7:48AM 2 Problems with chan_sip: random calls have no sound withouth any errors
7:15AM 5 Still looking for small fxo sip gateway
6:51AM 2 Mediatrix 1102 Auth
5:59AM 10 Dialling Hook Flash on Zaptel
4:54AM 4 Playing announcement to called user prior toConfirmation
4:44AM 5 busy tones
3:36AM 4 cisco 7912 voicemail/dnd issue
 
Monday February 2 2004
TimeRepliesSubject
11:20PM 0 Mark's Asterisk Presentation at Linux-Kongress2003
10:16PM 0 Newbie -- TE410P installation
10:11PM 1 Voicetronix Audio Problems when making two or more simultanoues calls
9:24PM 1 Details on TE410P Digium cards
8:41PM 1 Playing announcement to called user prior to Confirmation
6:50PM 3 extension mobility
6:07PM 0 Carrier Access Access Bank 1, incomming calls only echo problems, and Adit 600
4:34PM 3 Large scale e.g. university
3:26PM 4 agent autologoff
2:29PM 1 New Zealand users/contractors
12:44PM 0 VoicePulse IAX2 lag
12:31PM 0 help with h.323 outgoing calls
12:06PM 5 Automated Dialing / Recording ?
11:49AM 14 cdr mysql problem
11:22AM 1 Problem sip registration
10:30AM 1 Fax Extension
10:09AM 3 compile error (still having problems)
8:40AM 32 Transfer
8:25AM 0 Re: how to dial and accept a call with only
7:59AM 15 Can audio streams go client to cleint with IAX?
7:15AM 0 Re: how to dial and accept a call with only
6:57AM 0 VOIP/IAX Termination
6:21AM 0 ISDN, CISCO and SCCP call forwarding
5:59AM 1 Norstar Integration with Asterisk via FXO or BRI ISDN
5:19AM 13 compile error
3:41AM 3 Channel Bank
3:34AM 0 Hide outgoing CallerID
 
Sunday February 1 2004
TimeRepliesSubject
10:23PM 0 NewB: Cisco 7910
5:46PM 4 Superbowl = Linux Shake up to the world..
4:19PM 0 DNIS on X100P
3:24PM 2 setting up ---- newbie
2:51PM 1 can a variable be redefined within extensions.conf
2:09PM 3 Luxoncomm 3800 series FXO/FXS adapters?
2:05PM 5 How do I provide redundancy and reliability w/ Asterisk?
1:10PM 2 Configuring Firefly Network in *
12:02PM 0 PCI expansion slots.
11:57AM 3 how to dial and accept a call with only x100p card on Redhat linux 9.0?
10:46AM 1 Mediatrix 1204 SIP FXO 4-port gateway review
8:31AM 0 SMDI on *
6:13AM 1 short ringing