asterisk users - Jun 2006

Friday June 30 2006
TimeRepliesSubject
8:34PM 0 AudioCodes MP-124
8:33PM 1 Call back features
7:51PM 0 multiple includes
2:49PM 0 How to register a Motorola VT1005
2:19PM 2 Dial Macro timeout fails
2:15PM 0 Asterisk-1.2.9.1 with QSIG Protocol
1:38PM 1 SIP qualify time - best practices?
12:05PM 3 Auto answer an IAXY how
11:47AM 2 Auto NOTIFY
11:45AM 1 Switchtype
10:55AM 0 Asterisk x Qsig - messages
10:44AM 1 Cannot get back chan_zap.so module!??
9:38AM 2 Asterisk -x option in 1.2.9.1
8:53AM 1 recording all calls patch through asterisk
7:20AM 0 Does anyone know what this means?
6:38AM 0 (no subject)
6:35AM 2 New Digium Card b410p
6:03AM 2 Integrate asterisk with Database
5:54AM 0 IAX2 Jitterbuffer and trunking
5:41AM 0 FOSS, Science, and Public activism
5:32AM 2 Surge Protector for T1/PRI ?
5:10AM 1 Best GPL Gui?
4:45AM 2 BLINDTRANSFER
4:15AM 1 Problems with dial status...
3:46AM 1 ISDN: 3° incoming call
3:44AM 2 IAX jitter / clocking problem
2:31AM 1 Limiting a group of phones available channels
2:03AM 1 OH323 issue on AT320 Phones
1:43AM 2 Queue - Log if caller disconnects
1:30AM 2 cheapest Cisco Smartnet contract?
1:08AM 0 voting,suggestiuon,your input needed to all
 
Thursday June 29 2006
TimeRepliesSubject
10:00PM 0 Asterisk behind dynamic IP
7:16PM 0 What is the --> priexclusive <-- setting for in zapata.conf?
7:10PM 1 Recommended FXO device
6:59PM 0 dlink wifi dph-540 and text messaging
6:38PM 11 Digium Hardware Reliability
6:01PM 1 SIP reinvite still does not occour
5:25PM 0 need help troubleshooting clipping and garbledVOIP calls
4:37PM 0 additional calling party number
4:32PM 2 Help with JIAXClient
3:57PM 2 ISDN (E1) Hardware Echo Cancellation
2:53PM 0 IAX2 debug info
2:20PM 0 Queue errors when phones are down, and possible solution
2:07PM 0 Sangoma A104D is dropping DTMF digits, during IVR
1:33PM 1 need help troubleshooting clipping and garbl ed VOIP calls
1:26PM 3 need help troubleshooting clipping and garbled VOIP calls
12:26PM 1 Sangoma A104D is dropping DTMF digits during IVR
12:18PM 0 Really need some help on IAX2 destroy to prevent deadlock
12:06PM 0 DTMF Tones not coming in clear
11:08AM 2 quadBRI in bri_net mode - t3 timer expired
10:50AM 4 DTMF and ivr systems
10:33AM 0 Any one with sending and receiving Sucessfull SMS PTSN Portugal?
10:08AM 0 Cisco 7905G SIP firmware needed
10:08AM 0 (no subject)
8:49AM 1 username in Real-time changes all the time
8:45AM 0 GXP-2000 and transferring call directly to voicemail
7:57AM 1 Call Queue NOT using RoundRobin ?!?
7:07AM 1 beronet BNS40 led blinking: not working or not connected?
6:31AM 1 iax2 group pickup
6:11AM 1 Digium TE410P configuration to connect with CIsco 3800
6:08AM 1 Very bad quality with AVM Fritz!cardPCIandchan_capi
5:18AM 0 MixMonitor Problems
5:17AM 1 Very bad quality with AVM Fritz!card PCI andchan_capi
4:52AM 0 *** Spam *** recommended telephones
4:47AM 0 hipath 3750 + hg1500 + asterisk
4:26AM 0 hipath 3750
3:52AM 0 Slightly OT: SQL query to find max load
3:27AM 1 Issue with using dialing PBX digits after call is connected
2:43AM 1 app_sms not working anymore
2:19AM 2 Sangoma card A101 Card troubles.
2:16AM 4 Very bad quality with AVM Fritz!card PCI and chan_capi
2:02AM 0 Sangoma A200 Caller ID in UK
1:59AM 1 using kannel with asterisk
1:55AM 1 recommended telephones
1:50AM 0 Asterisk with Sipbroker calling / routing problem
1:43AM 3 bristuff hangup issue
12:15AM 1 Sangoma A200 hangup detection
 
Wednesday June 28 2006
TimeRepliesSubject
11:46PM 2 SNOM Softphone on windows 2000
11:40PM 2 2 or more ISDN cards: which comes first ??
11:05PM 0 IAX2 Destroying channel to avoid deadlock
9:09PM 0 ITSP in Atlanta?
9:00PM 1 Realtime patch
7:35PM 2 s / i extension difficulty
7:10PM 1 Wiki Voip Phone reviews
7:07PM 0 question about the register/invite call flow
6:08PM 4 Realtime SIP Registrations
3:20PM 1 Help with incoming SIP routing
1:42PM 2 Asterisk-Addons compile problem (cdr_addon_mysql.c)
12:55PM 1 G729 Code
12:05PM 6 Suggested Phone
12:05PM 0 Problems with hangup on TE110P and "Unexpected Channel selection 3" messages
11:16AM 2 Standard Sound Files Distortion
11:14AM 0 Re: [asterisk-biz] India Routes
10:35AM 1 asterisk -> my cell phone's voicemail sound problems
9:56AM 2 WIFI sip phone
9:52AM 2 Ztdummy and Debian on Intel Macmini
9:39AM 1 h263 Video Support Questions
9:25AM 0 Remote employees using Polycom 501 lose
9:04AM 9 Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
8:56AM 0 asterisk 1.2.8 compilation problem
8:54AM 2 (no subject)
8:47AM 3 asterisk shutdown
8:34AM 1 Mysql Trixbox
8:29AM 0 Dial Tone + E&M
8:08AM 0 Getting at SIP error with SIP_HEADER() ?
7:38AM 1 Realtime: how to use column setvar?
6:15AM 0 h323 phone
5:48AM 2 point to point T hookup?
5:23AM 3 Trixbox maunual configuration
4:53AM 0 Asterisk auto-dial Help
3:39AM 1 Work required - modify Asterisk + SEMS
3:32AM 1 HDLC Bad FCS (8)
3:04AM 1 password on radius authentication
2:31AM 1 getting agentID and DNID help
12:54AM 1 can Asterisk act as a H.323 Gatekeeper?
 
Tuesday June 27 2006
TimeRepliesSubject
11:45PM 1 zaptel.conf settings for Singtel ISDN-2
11:18PM 2 Changing standard Voicemail behavior
6:48PM 2 Addon-ooh323 install problem
6:29PM 6 FXO for PSTN
6:08PM 3 Most stable Asterisk version
5:19PM 1 Meetme + Sangoma issue?
3:29PM 1 Asterisk 1.2.7.1 with Polycom 501 on SIP -> Conf Calling
2:48PM 4 Mail loop?
2:41PM 0 Wierd bug with MD3200
2:33PM 0 a command to dump all callers in queues preferably from asterisk console
1:27PM 2 trunk rollover
1:25PM 0 Realtime Voicemail Broken?
12:50PM 3 Voicemail volume adjustment
11:29AM 4 PRI - Ring requested on channel errors - inbound & outbound stop working.
10:59AM 0 RE: Asterisk-Users Digest, Vol 23, Issue 182
9:51AM 1 Modifying Voicemail menus?
9:30AM 1 Voip / AudioCodes MP-108 Help Needed
8:19AM 1 ExternalIVR vs AGI
7:56AM 2 7960 help: transferring calls
7:47AM 1 F3000 registering to asterisk
7:37AM 0 can Asterisk act as a H.323 Gatekeeper.
7:33AM 1 isdn-data over iax
7:14AM 3 Call length limitation
6:59AM 7 asterisk to mobile phone
6:48AM 2 voicemail number of recorded messages
6:00AM 2 Problem with callerid in sip to isdn gateway
5:54AM 5 WebPhone
5:50AM 2 Callstatus on bridge IAX2 <-> ZAPTEL is always "answer" even if the call fails
4:33AM 0 (no subject)
3:40AM 1 Help Asterisk crashes
3:01AM 2 Background + Dial
2:52AM 0 dss1 progressing message on zap channel
2:16AM 8 Avaya 4610sw SIP setup problem
1:44AM 0 Globe7
1:18AM 4 siemens pbx and asterisk
12:43AM 1 DID in United Arab Emirates, Iran, Kuwaiti, Iraq, Bahrain, Jordan, Saudi Arabia.
12:22AM 2 SV: Error in config sample for GoToIf?
12:10AM 1 Error in config sample for GoToIf?
 
Monday June 26 2006
TimeRepliesSubject
9:13PM 2 using variable
8:15PM 1 Question about ring groups and ext. busy in call
5:24PM 1 SRST type functionality
5:16PM 2 x100p buying advice
5:00PM 1 M() option to Dial
3:53PM 0 Microsoft unified communications
3:26PM 1 ASTCC: customer wants 100 accounts
2:23PM 0 AGI script can not print out error message toconsole
1:58PM 0 "Say" Applications fail
1:27PM 1 AGI script can not print out error message to console
11:11AM 1 Email notification
10:16AM 4 Oh oh. Micro$oft just noticed VoIP
10:07AM 0 EuroISDN and Sangoma Card
9:36AM 0 Soekris net4801-50 + IAXY
9:32AM 1 STUN?
9:28AM 7 '500 Internal Server' Error on SIP NOTIFY
9:23AM 1 registering a Motorola vt1005
9:00AM 1 asterisk-stat display problems
8:55AM 0 MeetMe Volume Issues
8:52AM 0 Pickup zap issue
8:30AM 0 AEL scripting, CUT use and string concatenation
7:32AM 2 1.2.9.1 SIP/Local/Queue behaviours weird
6:33AM 1 struggling with the "g" flag
6:16AM 0 chan_sip.c: Insufficient information for SDP
5:51AM 3 This is getting really annoying - re: POSTFIX
4:42AM 2 Asterisk x Siemens HiPath 4000
4:40AM 0 Asterisk and Qsig Protocol
4:07AM 0 Agent Dump
1:15AM 0 Re: What happens if the soekris hardware is defective upon arrival? The Cortex Systems way.
 
Sunday June 25 2006
TimeRepliesSubject
10:25PM 1 News: Asterisk VOIP Jobs Site - Revision 3.0 up!
9:51PM 3 Asterisk Startups
9:13PM 2 [ISSUE] Unable to divert external calls.
4:11PM 5 Signaling and media
4:11PM 8 AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
2:28PM 0 Announcement : A2Billing V1.2.1 released today
1:15PM 0 RE : Re: [Serusers] CDRTool +Asterisk + Ser
12:28PM 3 Zaptel answering the Line
12:00PM 0 DTMF Detection: Where it happens actually?
11:51AM 1 Testing a FastAGI script
11:37AM 5 FW: Asterisk Quintum A800 SIP Mode
4:01AM 1 Gizmo and Asterisk analysis
3:34AM 0 AstriCon London Starts Tomorrow
 
Saturday June 24 2006
TimeRepliesSubject
9:06PM 0 DTMF Detection Problems on VGSM channel
4:26PM 2 Playing sound before dialing
11:14AM 0 Caller ID info for DID calls?
9:42AM 2 Polycom 601 question
9:23AM 0 CDRTool +Asterisk + Ser
7:31AM 0 Call stays mute
1:02AM 2 Asterisk ACD with Polycom IP501
12:54AM 2 Is anybody using XEN in conjunction with Asterisk and/or Openser?
12:34AM 5 ASTCC: How to reset periodically all "card in use" flag back?
 
Friday June 23 2006
TimeRepliesSubject
8:14PM 0 Best settings for Unicall and Fax
2:35PM 2 Include Text file in Dial Plan
2:28PM 0 Question about the SET(CALLERID(all)) Function
1:35PM 0 Connection issues
1:18PM 3 Asterisk-1.2.9.1 with Siemens HiPath 4000
12:51PM 5 Asking for phone number to dial
12:28PM 6 Caller ID Matching in extensions.conf
12:26PM 1 Can I get caller id passed to a phone connected to a Supura 2100?
11:52AM 1 RES: Meetme max users
11:51AM 7 Voice calls sent to fax extension
11:42AM 0 QueueMetrics 1.2 released today
11:35AM 0 Odd SIP error message
11:29AM 1 Asterisk home on VMWare time sync issues
10:47AM 3 troubleshooting echo on speakerphone
9:05AM 0 New to the list.
8:39AM 1 Asterisk Users Group - Portugal
8:24AM 0 Echocancelwhenbridged
8:23AM 1 call quality statistics?
7:53AM 0 Tribox - Unistim9.4 Makefile
7:39AM 0 How to use G729 decoded voice files?
6:47AM 0 Asterisk 1.4 on schedule?
6:46AM 1 Meetme max users
6:37AM 1 Kernel 2.4 / 2.6 and timer
6:04AM 1 SIP -> PSTN calls not connecting properly
5:42AM 0 UK English Sounds
5:39AM 0 Dial(ZAP with t option for call transfer via *2)
5:33AM 1 calling between contexts
5:15AM 0 Antek EGW-804 e *
5:13AM 0 Trunk failover
4:42AM 2 asterisk sip listening port
4:28AM 0 Call accounting where calls cross charge zones (code fragment request)
3:57AM 9 best hardphone for Asterisk?
3:16AM 0 TE405P Dropping Calls. !! Got I-frame while linkstate 0
1:59AM 4 GXP-2000 and Shared Line Appearances
12:23AM 2 Snom 360 with Firmware 6.1?
 
Thursday June 22 2006
TimeRepliesSubject
10:29PM 1 GXP 2000 - BLF and Hold/Hangup Answering
10:19PM 1 Asterisk-1.2.9.1 e MOH
8:13PM 0 Subject: Passing DID to external number?
7:06PM 2 problem - DSL line and Digium card
6:56PM 0 RTA, jitter, MOS et al over the internet
6:47PM 0 Cisco IP Phones - FYI
5:35PM 0 Motherboard Selection For TE110P & TDM400P
4:28PM 0 Voip* 300 minutes limit, credit expires
3:55PM 0 TE405P Dropping Calls. !! Got I-frame while link state 0
3:11PM 1 Routing inboud from ISDN to second * server.
2:51PM 1 PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM
2:46PM 0 Asterisk Users Group
2:19PM 0 Troncal SIP
2:17PM 1 Thoughts on building a Voicemail only Asterisk server?
1:57PM 0 uniden uip 200 phones lockup but rare - anyo ne seen this
1:54PM 2 Dell PowerEdge 1650
1:27PM 7 SE Michigan asterisk users group
1:18PM 1 How to set overlap dial timeout in bristuff zaptel?
1:17PM 2 iax2 registration problems
1:11PM 2 *** Spam *** Don't use CDRTool From AG-projescts
12:45PM 1 Re: Can I enter an extension to dial whilevoicemail is playing?
12:43PM 0 Realtime monitor of a channel
11:52AM 2 Soekris net4801 and IAXy dhcp issue
11:31AM 4 Don't use CDRTool From AG-projescts
11:21AM 3 Showing Current Calls
11:18AM 0 Playing sounds from the CLI
11:01AM 0 php-snmp
10:40AM 4 Passing DID to external number?
10:22AM 2 PRI Issue - Calls being rejected with unacceptable channel
9:56AM 0 New VICIDIAL astGUIclient Release: 1.1.12
9:53AM 4 Quality monitoring
9:47AM 1 South Africa DIDs
8:53AM 0 CDRTool / asterisk billing based on realtime
8:45AM 5 Out of Office Auto Reply:
8:40AM 0 Sharing experiences
8:33AM 0 disconnect with mute
8:24AM 4 when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on
7:36AM 1 SV: periodic-announce not working
7:08AM 0 periodic-announce not working
5:14AM 0 Toll free number comaptible with Voicepulse
4:25AM 0 Using Asterisk to better detect hangups when using ATA'S or Analog Gateways'
4:11AM 3 SIP Multi Call Generation
3:30AM 1 Action: Originate PROBLEM
12:50AM 1 SIP Channel hangup problem with re-INVITE enabled - ugrent
 
Wednesday June 21 2006
TimeRepliesSubject
8:23PM 0 How to configure ptime for certain codec
7:33PM 3 Time Based Goto Ifs Act Strange?
6:18PM 0 detecting 1-900 and like exchanges
5:43PM 0 direct a call to a busy channel
5:28PM 1 new asterisk server...welcome message cut off
4:25PM 3 Debian Sarge or CentOS4.3
3:57PM 2 Packet8 and Asterisk, do they play nice?
3:46PM 1 How to configure asterisk to emulate FXO signaling ?
3:26PM 0 Re: User Loses Ability to Make Outgoing Call s
3:19PM 1 Monitor / StopMonitor => MixMonitor / ??
3:06PM 0 Multiple Sound Files Folders - Spanish Syntax. AstCC AGI. 3/3
3:05PM 0 Multiple Sound Files Folders - Spanish Syntax. AstCC AGI. 2/3
3:03PM 0 Multiple Sound Files Folders - Spanish Syntax. AstCC AGI. 1/3
1:35PM 1 Calling same queue member all the time
1:27PM 0 uniden uip 200 phones lockup but rare - anyone seen this
12:16PM 0 Agent channel X SIP Transfer on 1.2.9.1
12:07PM 5 Polycom Intercom - almost there
11:11AM 3 me, voip.trxtel.com and early media
11:11AM 0 AEL Status
10:54AM 2 Snom 360 Passsword Issue
10:36AM 2 Can Asterisk Send a TEL URI INVITE?
10:09AM 4 Polycom 601 problems with multiple registrations
9:27AM 0 asterisk compiling
9:23AM 1 AMD Machine Detect
9:17AM 1 SIP or IAX client written in C
8:25AM 2 Asterisk queue log solution?
8:23AM 0 Telsey CPV
7:56AM 1 forward a call to a SIP account on a remote server
7:34AM 0 MySQL Realtime Voicemail Connection Lost
6:44AM 1 FW: zapata.conf: recent changes?
6:42AM 2 FW: syntax error
6:06AM 1 Asterisk 1.2.7/9.1 mp3 volume is good, wav file of same volume are too loud!
6:04AM 2 database copy in asterisk
5:38AM 0 AW: syntax error
5:16AM 1 syntax error
4:58AM 2 database space
4:57AM 4 zapata.conf: recent changes?
4:49AM 1 SPA-2002 call HANGUP. May be a SIP bug.
3:22AM 0 IVR Applications
3:20AM 1 getting zap peer of sip channel
2:55AM 3 H.323 soft phone known to be run with asterisk.
2:49AM 1 Monitor a particular SIP call for training purposes
 
Tuesday June 20 2006
TimeRepliesSubject
11:36PM 3 disabling modules - how?
8:50PM 1 Avaya phone 4610sw message waiting indicator and other settings
8:09PM 1 voip-magazine article "Using DUNDi with a Cluster of Asterisk Servers"
6:29PM 1 show register users
2:44PM 1 AGI: Dial and Recording my own CDR
1:30PM 0 ChanSpy on a specific channel.
1:03PM 1 Voicemail cut short?
12:28PM 2 TrixBox
12:08PM 0 Queues - Configuration Help needed
11:31AM 0 Voicemail beep doesn't end
11:24AM 0 bristuff chan_zap.c zt_pri_error line errors?
11:20AM 0 Anyone using VoIP WiFi phones?
11:12AM 0 5.8GHz phone and DTMF
9:58AM 5 1.2.9.1 crashed today
9:55AM 3 TDM400P bad echo problem, tried lots of things
9:54AM 0 Provisional problem with SIP channel
9:18AM 2 Snom 360 doesn't register after reboot
9:15AM 1 asterisk-backports.org
9:13AM 0 teste E1 card
8:30AM 3 Fun with Echo -- Follow up
8:28AM 0 Is the current G729 compatible with Asterisk trunk?
8:12AM 1 Caller-ID Info with Voice Mail -- Can it display to the phone?
7:55AM 0 Asterisk realtime and metrics
7:49AM 1 Add Country to CDR's
7:22AM 2 Conferencing with multiple servers
7:17AM 1 IAX2 Dial command
6:51AM 6 IAX FXS.. Any experience with...
6:47AM 0 call rejected tone within dialplan
6:39AM 0 AstriCon Paris Starts Wednesday
6:21AM 1 Integrating H.323 gateways with Asterisk?
4:42AM 10 TE420P/TE415P?
4:40AM 0 Working with Asterisk and SIP? Register for the Asterisk SIP Master class!
4:33AM 1 Bug in asterisk "static" realtime?
3:55AM 5 SIP Softphone on Thinclient?
3:43AM 8 fail to make call
3:39AM 1 manager DBDel action
3:06AM 1 Newest Asterisk doesn't compile
3:06AM 1 Which is the best user GUI ?
3:04AM 0 ooh323 issues
3:02AM 1 voiceone?
12:20AM 2 Call limit function on sip channel to external pop
12:07AM 0 How would you tet a FastAGI script
 
Monday June 19 2006
TimeRepliesSubject
11:51PM 1 Video phones probem
9:50PM 0 Call Not Disconnecting
7:34PM 2 massive screetch and echo from Treo 700w
6:10PM 0 Re: Asterisk-Users Digest, Vol 23, Issue 135
5:45PM 1 software to do sip stress tests
5:31PM 1 Asterisk --> BV: Incoming does not work....
4:47PM 3 Looking for SIP provider with minimal call setup time
3:27PM 1 Asterisk 1.2.9 cli "-x" doesn't flush?
3:23PM 5 faxdetect questions - Please HELP!
3:18PM 3 ECHO Tutorial
3:08PM 2 chat with asterisk
1:52PM 6 User Loses Ability to Make Outgoing Calls
12:37PM 2 home routers
11:18AM 1 Can I enter an extension to dial while voicemail is playing?
10:21AM 10 finding mac addresses
10:14AM 0 Act-Tel G11112DS Telephony Gateway
10:09AM 0 Question about context from-internal
10:02AM 3 sip to h323 ... direct RTP?
10:00AM 0 Linksys PAP2NA Configuration / Asterisk / Voip consultant wanted
9:57AM 6 sangoma unicall m2rfc
8:55AM 4 Polycom Buddies in 1.6.6
8:23AM 2 Asterisk 1.07 crash under Debian Sarge
8:08AM 0 Meetme Dumping Call's
7:41AM 8 How to use a data T-1?
6:55AM 1 Setting caller-id when parking call
6:50AM 0 suggestions for Wireless phone that receives text messages
6:41AM 3 Bristuff-0.3.0-PRE-1q and & florz patch compile trouble
5:16AM 7 Read command
4:06AM 2 "sip show inuse" is useless!
2:48AM 0 asttapi 0.10
2:07AM 2 show queue ... Invalid
1:42AM 2 Asterisk voicemail problem with isdn avm fritz!card
12:31AM 7 Transfer call via AMI or dialplan
 
Sunday June 18 2006
TimeRepliesSubject
8:39PM 1 multiple port
7:23PM 0 Fwd: FW: Creating Queues on Asterisk server - Sendingingress calls off-net to either PSTN or another VoIPapplication - thoughts?
6:01PM 11 DTMF Talk off
5:19PM 1 Creating Queues on Asterisk server - Sending ingress calls off-net to either PSTN or another VoIP application - thoughts?
11:46AM 1 agi, STREAM FILE and SIGHUP
6:44AM 1 302 Redirecting support
4:07AM 0 AstriCon Berlin Starts Tomorrow (Montag)
 
Saturday June 17 2006
TimeRepliesSubject
5:35PM 4 Which phones are good, or at least acceptable, for home and office
5:14PM 6 Canreinvite
3:53PM 0 MeetMe with recording - bitrate too low
1:31PM 1 Using HINT with Cisco 7960/SIP
1:16PM 1 Sipura SPA-2000 & Asterisk 1.24 w/incoming calls
12:40PM 1 Custom Extension halting execution upon caller hanging up
12:29PM 0 Voicemail with NFS (working, I think)
11:51AM 1 What ever happened to the LTAPI, the Linux Telephony API?
10:59AM 0 E&M + Dial tone
10:58AM 0 Nuvio SIP config
10:53AM 0 T1 + E&M
10:36AM 4 free sun boxes
10:33AM 3 ISDN BRI NetJet
9:01AM 2 Echo Cancelling VoIP traffic
6:31AM 0 Zap problem when calling out
5:00AM 0 Trouble somewhere with lib compilation
2:00AM 0 hanging up call after launching a script, script should continue independently
12:55AM 0 DTMF Twist
12:44AM 1 ODBC cdr tearing my hair out
 
Friday June 16 2006
TimeRepliesSubject
10:26PM 2 MOS Scores and LCR
10:22PM 3 Echo and crackle
8:41PM 5 asterisk load balance
6:33PM 1 reinvite, DISA, and switching codec's.
2:52PM 0 planet VIP 152 T
1:17PM 17 Voicemail with NFS
1:10PM 0 no IVR audio but phone to phone fine
12:20PM 0 linksys WIP300 and SMS text messaging
12:06PM 1 Incoming PSTN calls not routing to Asterisk? (using Sipura 3000)
11:06AM 0 One problem (MOH) and one question (incoming SIP calls)
10:21AM 2 DTMF in the middle of a call
10:17AM 2 SIPCALLID, but which callid?
8:41AM 0 French prompts for calling-card app ?
8:31AM 9 Two FXO: How to dial a number when a RING comes in?
8:00AM 1 VoIP Cheap & Asterisk
7:49AM 2 Zaptel dialing too fast?
7:35AM 3 Zaptel HZ Warning
7:18AM 0 Multiple Sound Folder Support for Same Language Syntax
7:16AM 0 CALLERID problems asterisk segfaults
6:14AM 2 Music On Hold troubleshooting
6:01AM 2 Bridging two existing calls (MeetMe, Sip Reinvite)
5:40AM 1 T1 Copper or T1 Fiber Line
5:21AM 0 H323 to SIP connection problem
4:27AM 0 isdn and PARK
4:09AM 0 SIP Registrations and DUNDi
4:07AM 2 Receiving faxes and then sending them on
3:38AM 3 Queues and hangup caller on Agent hangup
2:58AM 0 Soundwin S2400 standalone 24FXS/FXO SIP gateways
2:19AM 1 sangoma card test
2:14AM 0 Sip re-invite
1:21AM 0 no ring from zap channel
12:09AM 1 nortel meridian option 11c and asterisk
 
Thursday June 15 2006
TimeRepliesSubject
11:29PM 0 queue always hangs up/skip the next agent after ringing a agent -- help!!!
10:46PM 1 d & e options in meetme()
10:27PM 1 dial if
10:26PM 0 Multiple Sound Folders Support for Same Language (Syntax)
9:12PM 0 what are the elements of a good asterisk set up?
9:09PM 1 Gumstix!
8:58PM 6 FAX + Digium + SpanDSP
8:36PM 0 New version of NVBackgroundDetect:
7:47PM 0 Surprise!!! New sound files auto-downloaded to my system
5:50PM 2 rollover simulation
5:48PM 3 Problem trying to SayDigits when an invalid extension is dialed
5:37PM 1 what are the elements of a good asterisk setup?
2:30PM 0 pix 501
2:21PM 7 Executing a Function from AGI
1:47PM 1 Dropped calls continued
12:35PM 0 Re: Asterisk-Users Digest, Vol 23, Issue 114
12:12PM 0 asterisk+cdrtool
12:01PM 0 DUNDILOOKUP and DundiLookup()
11:34AM 0 Strange one-way audio
11:15AM 5 DUNDi Not Able to HandleComplexFailoverSituations
10:18AM 1 Asterisk & Cisco 3800
10:03AM 2 Bearer capabilities on PRI
9:36AM 4 DUNDi Not Able to Handle ComplexFailoverSituations
9:01AM 6 Comedian Mail not deleting .txt file
8:59AM 1 Odd Asterisk Stress Test Results
8:55AM 0 ACD Distributed Scenario....
8:41AM 1 Distributed ACD Queues
8:23AM 0 help in create user group
8:19AM 2 Cisco 7936 Conference Phone - SIP or SCCP?
8:16AM 5 Anyone see this?
8:09AM 1 Need to Hire: PHP Programmer for PhoneCALL
7:57AM 3 SIP codec preference order ineffective
7:47AM 1 No "ringing" being played to remote caller?
7:46AM 1 Strange Zaptel issue
7:31AM 10 Best $300 VoIP phone for asterisk?
7:30AM 2 MWI not working
7:22AM 4 EC needed in all-digital situation?
7:18AM 1 Broadvoice - Last Straw!
7:17AM 1 username/auth name mismatch
7:14AM 2 AGI to read MySQL
5:27AM 2 Trying to find good VOIP provider.
5:20AM 1 Backup Question?
4:41AM 2 Single T1 card with Echo CancellationtoworkwithDell?
4:29AM 7 Echo Problem with T411P
4:20AM 1 sip to h323 gateway ...
4:01AM 0 Bus Mastering
3:49AM 3 Auto-pickup cisco phones
2:25AM 1 Digital Receptionist
1:33AM 1 Update
12:32AM 1 Queues and local channels
 
Wednesday June 14 2006
TimeRepliesSubject
11:30PM 2 TigerJet PCI PPG FXO Card
9:00PM 7 open source sip softphone (Window OS version )
8:41PM 0 Easiest (best?) linux distribution for dedic atedAsterisk box?
8:07PM 4 DUNDi Not Able to Handle Complex FailoverSituations
6:59PM 3 WRTG54GS Capacity
6:31PM 1 analog call progress - can I use backgrounddetect
6:28PM 1 SPA941 and Echo
6:24PM 3 GXP-2000 addressbook
5:05PM 1 Please Help - Polycom IP 601 Buddy Watch problems
4:30PM 2 New Asteresk VOIP forum Buy Sell Discuss
4:21PM 2 DUNDi Not Able to Handle Complex Failover Situations
4:13PM 0 Sip stuck
3:49PM 1 Need to track dropped calls
3:16PM 1 Asterisk and multiple SIP registrations to the same host (team/oej/register)
3:13PM 0 Echo Cancel with sangoma o digium
2:32PM 0 CDR Billing
1:56PM 0 A dual Asterisk server question
1:38PM 1 Determining if extension exists
1:13PM 2 Calls keep ringing after being picked up
1:08PM 0 Easiest (best?) linux distribution for dedicatedAsterisk box?
12:17PM 0 Directory - First Name/Last Name - How to, use both? a@h?
11:31AM 4 kiax - iax2 softphone
10:45AM 1 MBX Servers?
10:12AM 3 Directory - First Name/Last Name - How to use both? a@h?
10:11AM 0 loading realtime peers
10:00AM 2 DUNDi Users
9:50AM 1 transcoding problem
9:47AM 1 dial plan return values
9:28AM 2 Sangoma driver and zaptel
9:05AM 0 QSIG
8:51AM 2 Web UI - Best practices?
8:32AM 0 Dynamic features on call waiting
8:09AM 6 DUNDi Docs
7:31AM 1 SIP call disconnected after answer
7:27AM 2 asterisk auto conference
6:59AM 0 Asterisk & wengophone
6:48AM 2 Which application to open Zap channel?
6:30AM 4 100 lines PBX + system config - repost
6:24AM 0 SV: DTMF when using g.729
6:09AM 1 SPA-941 Disable call waiting or Disable Call waiting via asterisk
6:05AM 6 GXP-2000 and Configdownload via TFTP
5:45AM 0 NCS patch
4:49AM 0 Sangoma driver update?
4:19AM 1 Realtime queue_members and penalties nost escalating (clue anyone?)
4:16AM 2 AddQueueMember and Local channels
4:01AM 0 How to find out which line in extensions.conf?
3:57AM 2 GXP-2000 1.1.0.13 Issues
3:50AM 1 AW: Eicon Diva Server with v3.0 drivers
3:38AM 0 RES: DISA Password Authenntication with Grandstream 488
3:33AM 0 FW: Issue in configuring TDM400P
3:32AM 0 Strange problem with MusicOnHold - works outgoing - works with extension - but not incoming!
3:00AM 1 Eicon Diva Server with v3.0 drivers
2:26AM 3 nortel meridian option 11c and asterisk te110p
2:22AM 4 Asterisk server
2:00AM 5 How much bandwidth needed?
12:51AM 1 DTMF when using g.729
12:43AM 3 SIP, Microsoft RTC, and Originate problem
12:28AM 0 Asterisk Zap/QSig with ChanIsAvailable
 
Tuesday June 13 2006
TimeRepliesSubject
11:25PM 0 AW: Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06
9:25PM 0 ISDN in Japan
9:08PM 0 Asterisk-1.0.9 Atxfer
8:38PM 1 Will 200KB/s drive access be OK for voicemailstorage?
8:17PM 1 GXP-2000 Audio Quality
8:03PM 4 how to hang the zap channel
8:00PM 1 voip to voip bridge
6:54PM 3 Easiest (best?) linux distribution for dedicated Asterisk box?
6:03PM 0 Will 200KB/s drive access be OK for voicemail storage?
3:56PM 0 AGI and Video
3:24PM 0 DISA Password Authenntication with Grandstream 488
3:01PM 10 OPENSER / SER and Asterisk
2:18PM 1 Cisco 7960 BLA
1:30PM 1 Polycom Queues
1:22PM 1 [REPOST] Asterisk Realtime and "Ex-Girlfriend"
1:06PM 1 Are zttest results relevant on a system with no telephony hardware?
11:42AM 1 calleridname.agi patch to only overwrite name if it is missing
11:14AM 0 Grandstream BT101 Auto-Answer
10:22AM 2 No incoming sip calls
10:13AM 0 Intel 600SM FXS card
9:56AM 0 Do I need to store voicemail locally?
9:37AM 0 Asterisk keeps running after hungup untill I press #
9:28AM 0 Asterisk Bounty Doubling program
9:14AM 1 [Repost] Asterisk realtime
8:54AM 1 sound quality problem on mISDN
8:47AM 1 Festival RPM?
8:08AM 0 WG: Dialplan problem with Digium tdm04p card
8:00AM 0 Problem with VoicemailMain
7:43AM 1 echo sidetone grandstream and tdm400p
7:14AM 8 IAX2 Vs SIP cpu load
7:04AM 1 Which simple billing application
6:51AM 2 Compiling zaptel on FC5
6:24AM 0 Asterisk and TBCT
4:29AM 0 Asterisk Realtime and "Ex-Girlfriend"
4:09AM 3 Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06
3:46AM 7 delay in MeetMe
2:39AM 1 Sipura SPA2100 ringing without phone
2:12AM 3 FW: conference
2:06AM 1 VOCAL + Asterisk
1:46AM 0 voicemail suddenly exits on DTMF: a bug?
1:43AM 3 Queues and macros and agents
1:40AM 3 Asterisk & Eyebeam chat function
1:39AM 1 timeout 't'
 
Monday June 12 2006
TimeRepliesSubject
11:27PM 2 How to retrieve voicemail
11:24PM 2 Bug in Voicemail ??
11:21PM 0 asterisk and nortel meredian option 11c
10:04PM 5 What is Echo?
9:47PM 2 /var/log/asterisk/full ?
8:21PM 1 MOH too loud
7:49PM 2 transferring calls from ekiga to asterisk
7:42PM 2 Unable to connect to Asterisk? (simple[?] question)
7:19PM 3 Help with Audicodes MP-104
5:34PM 10 Hard drive write cache
5:29PM 0 Good explanation somewhere of SIP security?
5:06PM 2 No reinvite - reason?
4:42PM 0 ICLID or CNAM calling name and number through a cisco isdn gateway
2:59PM 7 Can this config sustain 30 users?
1:35PM 5 Asterisk as Wholesale
1:17PM 3 Linksys SPA-941 NAT?
12:39PM 2 How can I use my regular phones with Asterisk running on my Linksys WRT54G router?
11:58AM 1 TTS to read from Database
11:33AM 0 TDM01B Card Install Problems
11:30AM 3 Snom high SIP ping time
11:27AM 0 freevoip.gedameurope.com - dial out
10:49AM 5 use AT320 international call
10:40AM 1 IP/SS7 gateway on Sun Ultra 20 amd64
10:39AM 3 asterisk on AMD 64 BIT
10:20AM 1 TDM-400 and dialplan -- how to ring a SIP ex tension *before* answering the PSTN line?
10:03AM 2 TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?
9:55AM 2 TDM Fax Problems
9:53AM 1 FW: TTS from MySQL
9:46AM 3 get value from DB directly
9:31AM 0 RAGI + Sphinx + Festival
9:30AM 5 IAX DID channels as incoming hunt group?
8:22AM 0 Re: CallerID name inbound from PRI
8:04AM 0 Presentation + Asterisk Realtime doubts
8:01AM 1 problem dialing out thru sip - using isdn on internal
7:32AM 2 AGI Stderr
6:59AM 0 SIP auth failed "wrong pw" but pw is correct
6:47AM 1 AstriCon Europe - Only 1 Week Away
6:22AM 7 spa3102 vs spa3000 differences?
6:02AM 2 Cell gateway for T-Mobile US??
5:19AM 2 Hitting * in a queue call hangs up?
4:50AM 1 Single agent multiple queues....
3:02AM 2 Attended transfer and queue
2:53AM 1 - SOLVED - Trouble getting SMS working
2:03AM 0 fixed ring strategy
1:40AM 0 enable/disable user
 
Sunday June 11 2006
TimeRepliesSubject
11:35PM 2 Rxfax with Sirrix quad BRI
9:10PM 1 TTS engine query
6:02PM 3 JIAX status
5:02PM 0 SOLVED - Cisco router and "488 Not acceptable here" messages
4:54PM 0 ISDN and DVO
4:35PM 0 Changing RO vars like SRC
4:12PM 0 Cisco router and "488 Not acceptable here"messages
8:08AM 0 hook flash call transfer
7:58AM 1 Cisco router and "488 Not acceptable here" messages
4:48AM 1 asterisk-1.2.9.1
4:23AM 0 to china: good voip service providers?
3:24AM 2 OLD PA system.
2:32AM 2 Nokai E60 and E61 , working fine with Asterisk , with new access points
12:14AM 2 Callback Application: Suggestions Please.
 
Saturday June 10 2006
TimeRepliesSubject
11:54PM 0 SIP quality monitoring
7:17PM 0 Question setting up a
7:01PM 0 Any good voip providers lately?
5:41PM 4 Question setting up a "bat phone" extension.
5:04PM 0 Reorganizing menus in Polycom 301? Was: [asterisk-biz] New Polycom SoundPoint Series IP-430
10:29AM 0 Problems with 7960 + callwaiting
7:21AM 1 Detecting gateways which time out
6:47AM 1 Voicemail records nonsense, but record() works (??)
6:43AM 1 ADSL modem, TDM400P, zaptel and not hanging up
12:43AM 1 record until silence, playback, repeat
 
Friday June 9 2006
TimeRepliesSubject
10:49PM 2 Unicall acting really funny
10:36PM 0 Asterisk,mISDN and a Fritz card -- kernel
10:24PM 3 VGSM Trouble: Kind people, help me please...
8:28PM 0 What's the current state of using shared lines in asterisk?
7:17PM 1 RE: Digium pound key software appliance opinions
7:10PM 3 FXO registration and VegaStream
7:08PM 1 Broken firewall or brain damaged admin?
6:02PM 1 SBC/ATT Supertrunk configuration
3:26PM 1 Grandstream BT100 lockup after attended transfer on 1.2.8 and 1.2.9.1
2:38PM 3 Trouble getting SMS working
2:05PM 1 shutting down a mysql server renders cdr_mysqldead and asterisk nolonger makes or receives calls
1:56PM 0 Why are sip-channels too lagged?
1:49PM 3 g729 or another
1:33PM 2 T1 passthrough/middleman
1:30PM 2 shutting down a mysql server renders cdr_mysql dead and asterisk nolonger makes or receives calls
1:25PM 0 Auto dialer
12:28PM 1 logrotate and logger reload
11:29AM 3 SIP 486 "Busy Here"
11:10AM 1 Polycom subscriptions
10:49AM 0 spandsp with t.38
10:18AM 3 Using "#include" on zaptel.conf
10:10AM 2 Stupid question zaptel-1.2.6 vs. svn/trunk
9:18AM 3 Compiling SVN Trunk
8:46AM 0 Bad call quality using a certain channel.
8:40AM 2 100 lines + system config
8:34AM 1 Anyway to customize ring tones on aastra phones?
8:23AM 1 SV: Call status subscriptions on multiple servers
8:13AM 0 Monitoring transcoding and other heavy activities
8:07AM 0 exactly what ports are required for sip phone to sip voip connection ?
7:57AM 2 No CID on ZAP
7:56AM 2 Dial Plan rules
7:48AM 0 Dead FXO Interface?
7:28AM 1 hangup extension
7:23AM 1 Re: Audio problems on Zap & SIP, local netwo rk, not IRQ related?
6:41AM 3 GXP-2000 MultiPurpose Keys
6:32AM 1 incoming call from Zap: "early audio" problem
6:02AM 4 long distance ask for pin
5:36AM 1 click to call features on asterisk
5:36AM 1 Asterisk, mISDN and a Fritz card -- kernel crashes
4:35AM 0 pickup a call from a group
3:59AM 2 H.264 and Motorola Ojo
3:54AM 0 error with tdm11b
3:45AM 0 SV: TSP on linux
3:37AM 0 SRTP/SIPS
3:32AM 1 TSP on linux
2:44AM 3 SV: Database file to copy for active sessions.
2:37AM 1 Database file to copy for active sessions.
2:11AM 1 Registered SIP:
2:00AM 1 Call status subscriptions on multiple servers
1:57AM 0 RxFax & Asterisk possible bug?
1:49AM 2 who is the mantainer ....
1:43AM 1 remote setting - AGI or what?
1:34AM 1 Asterisk, mISDN and a Fritz card
1:18AM 1 Sip transfer, Sip on hold
1:17AM 1 Random Zap Channel Drops to SIP
12:30AM 0 Duplicate asterisk processes
12:18AM 0 registration SIP softphone:who is the file who makes the registration?how can I set more proxy than 1?
 
Thursday June 8 2006
TimeRepliesSubject
10:58PM 1 Running a poll server with asterisk
10:24PM 0 Sending Fax on local host using IAXmodem
10:14PM 1 Asterisk + Zimbra when?
10:06PM 0 APIC error on CPU0: 60(60) and asterisk crashes
9:43PM 2 hangup lag causing the answering of already answered calls
9:15PM 1 Virtual PBX Billing and Management Software
8:08PM 4 PRI & Fax Passthrough
7:26PM 1 Disabling debug output
6:55PM 1 AEL2
6:30PM 1 Anyone know anything about VoiceWing?
6:26PM 0 ringback tone or signal on the phone somehow?
6:15PM 0 Queues with really short timeouts
6:14PM 0 Polycom IP-601 Microbrowser encountered HTTP error 406
6:02PM 1 "Reserving" a conference room
5:29PM 1 Vega 50 10 FXO
3:16PM 0 Astricon alive and well
1:47PM 7 Fun with Echo
1:46PM 2 no dialtone on channel banks
1:43PM 1 bug? asterisk -rx "show dialplan default"
1:42PM 6 revisit to legacy PBX and CID over PRI
1:10PM 1 Anyone have success using LIMIT_PLAYAUDIO_CALLER or LIMIT_PLAYAUDIO_CALLER variables
1:00PM 0 Two FXO Astralis X101P cards in older PC?
12:42PM 1 [CAVPDiscussion] OT: BT to replace legacy tele com infrastructure with open, standards-based VoIP switches
12:33PM 2 Linksys PAP2T-NA - call goes through but phone doesn't ring
12:30PM 2 Phone recommendations?
12:26PM 1 BN8S0 problem - Extension can never match, so disconnecting
12:19PM 0 new DID's
12:04PM 11 Linksys SRW224P POE Switch
11:59AM 2 Bullet-proof FXO?
11:51AM 1 Small form factor system w/PCI slot
11:37AM 1 FreePBX 2.1.0: Manually rewriting
10:47AM 4 h323 with asterisk problem
10:24AM 0 ipPhone and ATA with UPNP
10:17AM 1 set file path
10:16AM 2 Turning off a temporary message in voicemail
10:08AM 0 Problems with IAX
9:24AM 2 FreePBX 2.1.0: Manually rewriting extensions_additional.conf
9:00AM 1 early session audio on zap channel
9:00AM 3 Voicemail to Email on Blackberry
8:50AM 1 [HELP] - Sound cutting and dropping out - 1.2.7.1/Sangoma/PowerEdge 2850
7:48AM 0 Where has the outbound call directory gone
7:28AM 3 dial pattern
7:25AM 6 how to identify agi crash cause
7:14AM 0 Latest SVN with downloaded sounds. Update
7:12AM 1 RSA Signature (key ***) failed
6:52AM 0 chan_sip.c on debian testing - weird
6:49AM 1 Anyone with GSM488 experience?
6:45AM 1 chan-capi and dtmf
6:07AM 0 RE: help required plzzzzzzzzzz
5:59AM 2 gsm file
5:40AM 0 hangup don't realease analog line
5:23AM 1 FW: asterisk and nortel meredian option 11c
5:12AM 2 Native Music On Hold Volume LOUD! How to adjust?
5:12AM 4 increase the volume ?
5:01AM 0 SIP/2.0 484 Address Incomplete
4:43AM 5 Plainvoip problem.
4:00AM 1 MeetMe - Annouce user join/leave without recording the name
3:35AM 0 "I can hear them but they can't hear me" with VoipBuster
3:25AM 1 zap calls drop suddenly + tremendous noise when answering a call
3:13AM 2 What does RELAXDTMF do?
3:13AM 0 SV: Using regcontext
2:57AM 1 Using regcontext
2:55AM 2 Nokia N80 and asterisk?
2:21AM 1 Hardware to connect analog and ISDN fax devices
2:19AM 0 How to check NAT behaviour before installing Asterisk
1:59AM 0 Astricon No More...
1:40AM 0 Simple Speeddial AGI
1:40AM 0 FW: Quality of Asterisk
1:16AM 3 how to delete a key from database in extensions.conf
12:47AM 1 Query
12:26AM 1 Latest SVN with downloaded sounds.
12:01AM 1 SV: SV: I can hear only one way when I use nokiae-60withX-lite
 
Wednesday June 7 2006
TimeRepliesSubject
10:08PM 0 PRI and BRI
6:55PM 1 SIP to SIP connection problem
4:10PM 1 Many asterisk server behind a redirector?
2:51PM 0 Caller ID issue solved (for now)
2:02PM 2 Unlock / install of Cisco 7940 IP Phone ?
1:58PM 1 MWI on the PA168V in IAX mode?
1:50PM 1 TBCT - Two B-Channel Transfer
1:26PM 0 music on hold Madplay and Files not working
1:16PM 0 Opposite iaxy?
12:59PM 1 Good ATAs from companies other than Sipura/Linksys?
11:56AM 1 Unicall local_unblocking_expired error
11:06AM 0 New York Times article on VoIP Hacker
10:22AM 1 Analog Line "Static" and Low Volume
10:22AM 1 Supporter needed
10:06AM 5 Block access to number@domain.com
10:04AM 0 bewan phonebox
9:55AM 1 Controlling Cisco 7960 Ringtone from Asterisk
9:50AM 3 PHP UnixODBC MS SQl 2000
9:41AM 0 How-To monitor a specific channel?
8:56AM 0 polycom ftp
8:24AM 0 Asterisk not waiting for E&M Wink (I think)
7:53AM 1 meetme public
7:35AM 2 SV: I can hear only one way when I use nokia e-60withX-lite
7:02AM 1 Music On Hold not working with new 1.2.7.1 install
6:42AM 1 Notice Question
6:03AM 0 voipbuster & dtmf tones?
5:53AM 19 Quad T1 Card
5:38AM 0 regexp issue
5:36AM 0 SpeedTouch 780WL
5:12AM 0 CLI comand to register softphones without close them:
5:08AM 2 SV: I can hear only one way when I use nokia e-60 withX-lite
4:58AM 0 I can hear only one way when I use nokia e-60 with X-lite
4:44AM 1 a new asterisk version
4:08AM 1 Delay on calls
2:41AM 1 IAX2 channel problems
2:00AM 0 asterisk load balancing setup
1:10AM 1 asterisk-1.2.9 / res_snmp.so
 
Tuesday June 6 2006
TimeRepliesSubject
11:46PM 5 HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.
11:24PM 0 Need help with two-stage ringing macro
10:07PM 2 A@H / Trixbox Question
8:24PM 1 Reception softphone suggestions?
6:14PM 1 Problem with simple incoming calls
5:32PM 0 This is what I want to do...
4:05PM 0 Voicemail normalization
3:15PM 0 pbx_spool - outgoing qcall failure upon call progress
3:15PM 0 [asterisk-dev] UK Male English Voices
2:27PM 2 UK Male English Voices
1:32PM 1 Asterisk 1.2.7.1 bad file descriptor
1:25PM 1 asterisk-1.2.9 is not stable
1:11PM 10 GXP-2000
1:09PM 4 Zork and Asterisk
12:48PM 0 Sip bug...problem seem to be fixed in trunk. How do I find the patch for 1.2
12:15PM 4 Avaya 4624 Ip phone
12:01PM 1 Customer's voice not compatible with service?
11:48AM 0 Asterisk + Linksys PAP2-NA / Call Clearing
11:19AM 1 OT: Cellular boosters
10:59AM 2 Transcoding g.711 -> g.729
10:37AM 1 Weird Can-Reinvite problem
9:46AM 1 SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com
9:43AM 0 Asterisk 1.2.9.1 and 1.0.11.1 Released -- Security Fix
9:39AM 1 wav49 size for a 3 minute voicemail
9:38AM 5 DTMF feedthru again...
9:22AM 1 Vonage and FXO
9:10AM 1 Asterisk exit on startup
8:02AM 5 syslog server
7:10AM 0 FW: voice mail
5:28AM 3 weather
5:16AM 0 Personal Inquiry
4:39AM 1 PABX Setup
4:29AM 0 What to do on a national celebration day? Test, test, test!
3:31AM 1 Asterisk Realtime and SIP Registration
3:25AM 1 Change in dial command behaviour between 1.2.7.1 and 1.2.8?
3:15AM 5 Playback welcome message while phones ring, please help
2:09AM 2 Can I use an onboard modem?
1:09AM 0 Help - DTMF feedthru
12:57AM 0 Query: IAXModem
12:17AM 5 STNU spport
 
Monday June 5 2006
TimeRepliesSubject
9:29PM 1 Compile install error.
8:32PM 2 show channel issue with 1.2.9
7:12PM 6 ISDN BRI (I.430) over ethernet
5:22PM 1 Asterisk 1.2.9 and 1.0.11 Released -- Security Fix
3:42PM 9 IAX Passing Variables
3:36PM 0 Multiple SIP Accounts Between Asterisk Boxes (Unreachable)
3:36PM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday June 10th - 2006
3:33PM 0 Recurring Wakeup Call Schedule & play Weather Forecast
3:28PM 2 Polycom SIP 1.6.6
2:29PM 4 How many TE405 ...
2:13PM 0 In-bound faxing working ~1/3 of time.
12:15PM 0 Asterisk/Metaswitch trunk, no inbound RTP stream on inbound calls
11:54AM 0 Asterisk & iSeries AS/400
11:47AM 4 Local vs. toll Dial Plan
11:37AM 0 Multiple sip proxy per * server.
11:10AM 1 This should be easy: What happens when the Calling Party hangs up
11:02AM 2 Wanted: CISCO 186 ATAs
10:40AM 2 Outgoing call bridging
10:22AM 2 DTMF and DISA
9:50AM 2 Looking for postpaid quality A-Z termination
9:44AM 2 Asterisk chroot
9:28AM 1 Mixing meetme conferences
9:12AM 2 Configuring behaviour of flash hook
8:51AM 1 More Level QueueSystem
8:23AM 0 SpanDSP and analog Digium channels (TDM400P)
7:33AM 6 Can´t send emails
6:59AM 0 collect call
4:36AM 2 Duplicate CDRs
2:44AM 0 Tr: RE : Openser+Asterisk+voice mail
2:21AM 0 change of calls control with VRRP protocol
1:21AM 1 asterisk clustering
12:56AM 1 Allowing multiple exchanges
 
Sunday June 4 2006
TimeRepliesSubject
10:59PM 1 Campusing two Asterisk boxes?
9:41PM 5 chan_capi-cm-0.6 and incoming calls problem
2:25PM 2 TDM-400 doesn't detect far-end hangup
2:21PM 1 Compiling VD_app_conference for x86_64
1:42PM 5 WCTDM-24xxp woes
1:19PM 2 Call-pickup function in Queue application
1:05PM 3 Configuring Polycom 501 IP phones via the console
12:08PM 3 reinvite
11:30AM 6 fine-tuning asterisk questions
11:28AM 3 Asterisk and SATA Raid 1
10:12AM 1 Inconsistency with ANI and channel callerid
9:47AM 0 asterisk+voicemail+openser
9:35AM 1 statistics
7:19AM 1 Xlite and # code after call is connected
7:10AM 0 capi drivers for suse-10.1
7:07AM 2 Asterisk on Mini-Box M300
6:55AM 0 Asterisk Memory leak
4:28AM 0 ASTCC Developer
3:54AM 0 Sound playback problems
3:34AM 3 asterisk behind cisco pix 506
3:01AM 2 Monitor application and e-mailing attachment
2:50AM 3 How to make this into a Macro?
2:46AM 3 transfer & other features
12:13AM 0 ISDN call-progress IE in SETUP frames
12:02AM 1 Help with compilation of app_conference in x86_64
 
Saturday June 3 2006
TimeRepliesSubject
11:07PM 1 PSTN outgoing DTMF vs. transfer Problem
11:02PM 1 New Member, saying Hi. :)
9:20PM 4 Meetme versus app_conference
9:01PM 1 Asterisk 1.2.8 - WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed
7:12PM 1 Sipura SPA-941 not available after Asterisk & Freepbx upgrade
4:35PM 3 Sangoma A101 configuration
3:11PM 2 ADIT 600 <=> Asterisk Help
2:03PM 4 Size limitations of extensions.conf
1:13PM 1 Asterisk 1.2.8
1:07PM 2 Recommended Web Interface
1:02PM 0 Bullet-proof System
1:01PM 1 Fw: Compiling chan_bluetooth
12:53PM 1 Integrating Asterisk
11:06AM 2 Busy Signals after hangup
10:08AM 3 Asterisk + PRI Card -> Nortel BCM
10:06AM 1 is '9' needed for "outside" numbers
9:01AM 0 What's asterisk on FreeBSD like now a days?
8:09AM 0 "X-Asterisk-HangupCause: Normal Clearing"
3:55AM 1 MWI lost after migration
 
Friday June 2 2006
TimeRepliesSubject
11:55PM 2 BN8S0 Installation problem - 0 devices registrered
9:42PM 1 lspci doesn't show digium card te405p
6:13PM 1 Asterisk - Qsig
4:37PM 3 All non US 48 area codes?
4:09PM 4 Problems and questions with setting up a Feature Group D trunk to a Nortel DMS-10 switch
3:16PM 2 NFS and voicemail
2:41PM 0 Limiting the size of a Queue
1:42PM 17 Config Revision Control
12:56PM 2 Restricting amount of incoming calls
10:35AM 0 Limited Queue Overflow Puzzle
10:05AM 0 OT recommend an IAX phone or IAX softphone+USB handset?
10:00AM 1 DID from Latvia?
9:47AM 0 New => Asterisk Queue (and CDR) Log Analyzer
8:42AM 2 frame.c:128 ast_smoother_feed
8:36AM 0 Re: Asterisk-Users Digest, Vol 23, Issue 11
8:31AM 1 stuck call on asterisk
8:00AM 20 Prices of g729 codec
7:32AM 1 PHP-AGI help
6:54AM 1 Any ideas why I can't dial this SIP phone (sometimes)?
6:29AM 0 Asterisk trunk cisco 2851
6:15AM 0 Re: Asterisk-Users Digest, Vol 23, Issue 10
5:29AM 0 misdn and dtmf problem resolved
3:35AM 0 Small Asterisk Weather / Cepstral Howto
3:17AM 0 Anyway to set maximum wait time when there's only 1 user in Meetme?
3:01AM 1 very slow network from GXP-2000 switch port
2:31AM 2 Audio problems on Zap & SIP, local network, not IRQ related?
1:12AM 0 Ordered my first phones :)
1:02AM 0 Re: Asterisk-Users Digest, Vol 23, Issue 9
1:00AM 0 using mediaproxy for both ASTERISK and SER
12:37AM 1 Grandstream BT101/102 lost register with asterisk ?
 
Thursday June 1 2006
TimeRepliesSubject
10:45PM 2 addons trunk make error
8:49PM 1 Example config files for Snom mass updating?
7:30PM 1 DID in Houston 713?
6:41PM 0 Re: Asterisk-Users Digest, Vol 23, Issue 8
5:37PM 1 IAX multiport ATA
4:13PM 0 Asterisk-addons 1.2.3 released
1:50PM 1 sip channel monitoring
1:46PM 1 email a message
1:42PM 0 Fran Oliveira desea chatear
1:28PM 1 Chanspy Jitter?
12:12PM 0 OT but relevant to SMS
12:09PM 1 Page cmd & FOP
12:08PM 2 Large Asterisk System
12:03PM 1 AEL #include (Labels and Goto app)
11:58AM 1 G729 + Native (files) MOH
11:39AM 0 TDM11B FXS port stops working after a reload?
11:37AM 1 Voice Mail or MWI notify as a (windows) tray icon
11:15AM 5 Converting Voicemail wav to mp3
9:24AM 0 Optimal Hardware
9:03AM 0 Re: Asterisk-Users Digest, Vol 23, Issue 4
8:47AM 6 Asterisk: T1 hunt group setup
8:41AM 3 app_queue and Real roundrobin
8:06AM 2 skype out
7:55AM 0 HDI remove a key from the Asterisk database with a <null> key, but a value?
7:20AM 0 IAX2 and dialin
7:20AM 4 G729, voicemail, no codec_g729
7:05AM 0 SIP Delayed Answer
7:04AM 3 How to redirect an incoming call to an external phone numer
6:56AM 0 Problem when i call to asterisk from traditional phones
6:55AM 1 Several asterisk processes starting with safe_asterisk
6:45AM 1 SIP Jitter buffer. What version of Asterisk PLEASE?
6:44AM 0 How can I use features without enabling 'call parking'?
6:41AM 17 Polycom-Asterisk hints/presence
5:46AM 0 Re: Asterisk-Users Digest, Vol 23, Issue 2
3:30AM 1 audio streaming points different with VRRP
3:11AM 0 choppy audio sip <-> capi
2:33AM 2 unknown host cvs.digium.com
2:18AM 2 Looking for very basic example
1:54AM 1 connecting asterisk to pstn help
1:20AM 0 dealing with trafication tone
1:18AM 2 Change g729 payload
12:39AM 4 astdb entry in sip.conf
12:33AM 0 Problems with misdn and BN8S0