asterisk users - Dec 2009

Thursday December 31 2009
TimeRepliesSubject
8:12PM 3 Daily Thousands of files in recording calls in Device mode
6:53PM 3 AudioCodes Caller ID
3:46PM 3 Dialplans & Holiday Dates
2:11PM 0 Friday Jan 1 Voip Users Conference
9:47AM 2 Asterisk recieves "11" when pressing "1" from SIPphone
5:40AM 0 Dahdi install issues
5:10AM 8 Inquiry:Asterisk different codec schemes?
1:19AM 2 Twilio
 
Wednesday December 30 2009
TimeRepliesSubject
9:10PM 1 BLF status of sip trunk
7:43PM 3 Skype for Asterisk
6:16PM 0 Asterik with out registration.
6:11PM 2 Force Jitter Buffer for SIP to SIP calls
4:52PM 4 CDR_MYSQL 1.4 Database Structure
3:12PM 1 Monitoring SIP & Skype connections
2:56PM 7 Per user voicemail greeting
2:56PM 8 CID not working.
1:04PM 1 problem with ring being sent to caller
12:36PM 2 Inquiry:Asterisk Dictate?
11:59AM 4 multiple instances of asterisk on same machine
11:29AM 0 Inquiry:Asterisk festival?
11:28AM 3 Parked Call Ringback
10:38AM 0 wcte12xp0: Missed interrupt. when disable echocanceller
6:06AM 2 Auto-provisioining Polycom 430 wth dd-wrt router
12:14AM 0 Context Switches and Load Average spike - Asterisk Version 1.4.22
 
Tuesday December 29 2009
TimeRepliesSubject
4:55PM 0 asterisk billing transferred calls
2:50PM 0 asterisk 1.6.2.0 sip channel to sip channel call dtmf inband not work.
12:55PM 4 ReceiveFAX G.711 + Realtime
12:06PM 0 set box IP from which send sip traffic
11:44AM 1 T.38 and Linksys SPA8000
8:56AM 0 CDR is "NO ANSWER" when it should be "ANSWERED"
8:21AM 0 codecs and volume
8:18AM 7 SkyHost is set to expire
7:23AM 1 CDR
6:39AM 3 error when open a2billing web page!
6:07AM 2 Does A2Billing has mial list?
6:01AM 12 identifying channel for softhangup
5:37AM 1 de-latinisation of the web - http://blog.collins.net.pr/2009/12/de-latinisation-of-web.html
2:13AM 2 Any good dialplan code out there to implement vertical service codes?
1:58AM 0 asterisk-users Digest, Vol 65, Issue 68
12:09AM 4 Realtime mysql extensions mutiple queries for each priority?
 
Monday December 28 2009
TimeRepliesSubject
9:22PM 9 cheap ip phone with auto-answer
7:59PM 1 Off Topic: Aastra BLF limit...
6:11PM 4 SIP Issue
5:45PM 4 Multiple Digium cards with one NFAS trunkgroup
1:39PM 0 Avaya 96xx handset with SIP 2.5, no name in display
8:47AM 2 Registering with a static peer?
3:59AM 2 AudioCodes MP-114 making calls via FXO
 
Sunday December 27 2009
TimeRepliesSubject
5:27PM 0 Macro only accepts 1 argument
5:04PM 0 Parking function problem ?
2:10PM 2 rxgain / txgain for iaxmodem or hylafax
11:51AM 2 UA send 404 Not found
10:42AM 4 Call ends when picked up
10:22AM 0 Q; Recording when a bypass phone is used
7:39AM 8 Audiocodes MP-114 2FXO/2FXS help registering with Asterisk
 
Saturday December 26 2009
TimeRepliesSubject
1:36PM 2 pattern matching
 
Friday December 25 2009
TimeRepliesSubject
10:31PM 2 Newbie Looking For Login/Password
8:31PM 0 state_interface how to
9:15AM 0 How freeradius mapping the attributes when save records into radacct table?
6:50AM 12 compile issues.
6:23AM 7 SIP Incoming / Inbound not working for Broadvoice (Asterisk PBX 1.6.1.6)
 
Thursday December 24 2009
TimeRepliesSubject
4:24PM 4 Recording the Calls to a USB Drive
1:44PM 0 Asterisk Manager API - DTMF issues
9:49AM 0 X100P clone card problem
9:30AM 2 Tel uri Support
7:46AM 0 redial script...
7:18AM 3 1.6 Troubleshooting help
6:43AM 3 Asterisk with gdb
4:52AM 2 How to create MeetME room with dialplan?
2:50AM 0 Extension.conf
2:42AM 0 NICE analog recorders
2:19AM 0 Failed to record Radius CDR record!
12:24AM 0 MixMonitor stops audio in SIP to SIP call
 
Wednesday December 23 2009
TimeRepliesSubject
11:28PM 2 How to send variables through AMI originate and read those variables in context?
11:19PM 3 AMI originate and PHP
9:53PM 0 Happy Holidays from OpSys Consulting Group
9:21PM 6 Dell Server suggestion
7:08PM 8 how to check Asterisk SIP registration
6:52PM 2 Core show function?
5:26PM 0 Can't place 2nd call to provider
4:49PM 8 fax problem
3:00PM 1 How to exchange/get $variables from/to each channel on cmd Dial
2:32PM 8 Asterisk and Faxing
2:04PM 0 Issue calling a TFN
10:20AM 1 Problems with chan_sip
10:09AM 4 Can't load cdr_radius.so module?
9:16AM 2 Can't do make menuselect?
7:53AM 1 SIP realm
5:17AM 1 Session Refresh or Codec change
 
Tuesday December 22 2009
TimeRepliesSubject
5:37PM 0 Account Code Inbound
3:01PM 2 call queue with external numbers??
2:22PM 0 Asterisk Release Time Frames
2:08PM 2 E1 R2 Congestion Status
2:05PM 20 Showing "name of extension" when calling
1:47PM 1 AsteriskNow and language
11:11AM 0 Available Agent on Queue
10:57AM 1 Making a data connection with Asterisk
9:26AM 4 asterisk & x-lite
8:20AM 0 Asterisk news :: Next release of Asterisk will be 1.8 Long Term Support
6:41AM 2 Every one Busy Problem
 
Monday December 21 2009
TimeRepliesSubject
11:06PM 9 TDM 400 hardware(?) issue
10:39PM 2 Asterisk 1.2.14 - Play an audio or signal
8:21PM 0 Asterisk chan_skype GUI released
6:13PM 0 res_cepstral for 1.6.2
6:04PM 4 Looking for some example dialplans
5:35PM 0 possible to list a conference in the database?
1:41PM 2 Asterisk depend on postgresql files?
1:04PM 5 script
12:48PM 1 Yealink vs Aastra
12:47PM 11 Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
11:20AM 1 sip show peers returns several notices
11:19AM 5 anonymous calls code
11:00AM 1 Incoming calls coming into default context
8:22AM 3 FreeRadius or A2Billing , which is the better choice of Asterisk for billing?
6:39AM 4 New 1.4 system: registered, but not responding to invite?
5:24AM 2 Asterisk Heartbeat Monitor for Fail safe.
 
Sunday December 20 2009
TimeRepliesSubject
11:04PM 4 What changed in Directed PickUp between 1.6.1 and 1.6.2 ?
2:50PM 9 Live CD - do you think they are worth doing?
11:14AM 0 Dec 20 Global All Free SW HW Culture meeting - BerkeleyTIP
10:15AM 2 Rewrite of calling number for all extensions
8:10AM 3 Manager command that equal to database show CFIM
 
Saturday December 19 2009
TimeRepliesSubject
4:20PM 10 sendmail
2:29PM 2 PAP2 Dialing Delay
12:05PM 0 E1 ingress to SIP egress problem with 183 response
11:51AM 10 Inquiry:Connect my Asterisk to external sip?
10:57AM 8 Nortel BCM - Call Accounting Interface?
 
Friday December 18 2009
TimeRepliesSubject
11:14PM 6 Asterisk 1.6.2.0 Now Available!
10:59PM 4 Asterisk 1.6.1.12 Now Available
10:54PM 0 Asterisk 1.6.0.20 Now Available
10:49PM 0 Asterisk 1.4.28 Now Available
6:28PM 8 Call Waiting With Draytek ATA
5:05PM 4 HOW to record saynumber output
2:21PM 0 SAP-BCM Sip trunking
1:53PM 1 Could Asterisk be crashing under high context switches?
1:24PM 0 IAX/NEW delays
11:25AM 0 test request for new event "Pickup" when a call is picked up from an other phone
10:49AM 0 DTMF doubler when using READ()
10:25AM 1 wrapuptime?
7:18AM 0 Friday @12 Noon ET: Kamailio, Open SER and Asterisk
3:36AM 2 FAX for Asterisk
1:56AM 0 calls ending up in default context
1:29AM 5 To Asterisk AMI Gurus - Tacking issue with originate
 
Thursday December 17 2009
TimeRepliesSubject
11:23PM 11 Feature Request: GotoIfTimeWithOffset
5:47PM 3 Integrate a CPE with Asterisk in MGCP
9:30AM 0 Setting the Request URI In registration
8:53AM 1 iax no way sound
8:39AM 3 Asterisk IPv6 update - we need an update
4:16AM 1 SIP to Analog Devices
 
Wednesday December 16 2009
TimeRepliesSubject
8:02PM 0 Help forcing crosstalk
7:24PM 1 sip show channels display
5:21PM 0 Federal, State, and Local government installations of Asterisk
4:08PM 9 Mixing commercial/SVN Asterisk
1:27PM 1 FW: question on how to connect 2 boxes
1:17PM 0 Replacing AgentCallBackLogin for Asterisk 1.6
1:05PM 0 Shorr/Call quality issues
8:47AM 16 announce prompt to user
7:26AM 3 About Asterisk Manager (C# Sharp)
3:43AM 0 asterisk-users Digest, Vol 65, Issue 38
 
Tuesday December 15 2009
TimeRepliesSubject
6:41PM 2 monitor-type=MixMonitor
6:15PM 29 Can't get G.729 to work...
3:09PM 2 member (In use)
2:07PM 0 OT - SPA3102 - Provisioning with config file [SOLVED]
12:37PM 1 dahdi-channels.conf -v- chan_dahdi.conf
12:07PM 1 cdr question
10:42AM 5 OT - SPA3102 - Provisioning with config file
9:44AM 0 simple sip question (I think)
8:22AM 0 how apps/enter.h
6:27AM 0 digest authentication method and the realm domain
1:16AM 7 Best way ro run 2 or more asterisk servers?
 
Monday December 14 2009
TimeRepliesSubject
11:59PM 1 Queue still tries to ring agent when busy
10:43PM 1 3 ed party sip client for Nokia sy
9:02PM 0 What version of libpri and zaptel work best with 1.4.24
8:44PM 0 question on how to connect 2 boxes
8:06PM 0 USB ISDN30
8:02PM 0 pickupexten on chan_dahdi
7:34PM 0 iaxmodem Disconnect time and ISDN Service
7:02PM 3 Asterisk throws error using the alsa, module
6:21PM 2 Rewrite calling number of incoming call
6:19PM 7 hints through a Local channel
5:59PM 3 Question regarding digital card TE412p
4:52PM 6 ISDN: Inband DTMF doesn't trigger transfer feature
4:28PM 9 Is this bad hardware? Dahdi-v-X100 clone
3:53PM 2 Asterisk & Zaptel setup on vserver
1:41PM 1 meetme with review of the entered conference number
1:32PM 3 Asterisk ZAP/DAHDI reads phantom digit on overlap PRI
11:52AM 0 ISDN Remote HOLD
10:11AM 1 Call on hold through DTMF
8:35AM 1 RTCP SR transmission
2:10AM 1 AGI with PHP
12:20AM 2 question on queues
 
Sunday December 13 2009
TimeRepliesSubject
7:52PM 1 Random DTMF tones generated from speech
1:25PM 1 Dial with timeout don't end call
10:15AM 17 iphone client app
9:15AM 0 Avaya 9650 SIP phone and dial timeout
8:01AM 1 Asterisk Queue Dialplan
2:50AM 9 Unable to open file...
 
Saturday December 12 2009
TimeRepliesSubject
10:17PM 2 Auto Attendant / Receptionist system
5:49PM 3 Random DTMF tones generated from speech in conversations
4:56PM 1 how to randomly use provider?
11:25AM 4 DEVICE_STATE
9:44AM 0 Inquiry:Asterisk sip server?
12:12AM 0 T38 Passthrough 1.6.1.12-rc1 Good Results
12:03AM 2 Playing a message if my call lands in their voicemail
 
Friday December 11 2009
TimeRepliesSubject
10:03PM 1 Terminate T.38 to PSTN
6:06PM 4 G729 Pass through
5:37PM 16 ATA FXO
5:25PM 1 chan_dahdi.conf for TDM404E
4:18PM 1 question on register
4:08PM 2 Free Fax for Asterisk
3:53PM 0 zttool don't show NT mode with OctoBRI
3:17PM 4 Asterisk Unregisteres IAX Friend Randomly
12:26PM 0 VUC Dec 11 @ 12 Noon EST: g729 transcoding, software & hardware
11:19AM 3 Calls Dropping
9:25AM 2 ANNOUNCE: New version of Activa TAPI driver
8:33AM 0 How to get LEG B channel info?
8:02AM 2 sip realtime question
7:00AM 1 max. no. of conferences supported
5:57AM 0 How can Write simple AGI in JAVA
 
Thursday December 10 2009
TimeRepliesSubject
8:21PM 0 Splash ring on PAP2t
7:56PM 0 need help to setup a sip trunk between a Nortel CS1000 and asterisk
6:18PM 1 interdigit timeout chan_dahdi
4:04PM 0 Encrypted GSM
3:15PM 1 Sangoma card reports HDLC errors
1:47PM 0 Avaya 950 one-X Deskphone
1:19PM 5 Asterisk 1.6.1.11 Fax
12:57PM 2 Hangs up after 16 minutes on a call.
7:53AM 3 switchvox 305 Appliance
6:00AM 0 Asterisk Meetme on XEN virtual machine recording is 2x times faster than normal
3:15AM 3 Realtime Database Tables
2:39AM 5 asterisk-users@lists.digium.com Hello!
12:49AM 2 Asterisk as a PSTN simulator
 
Wednesday December 9 2009
TimeRepliesSubject
9:08PM 23 Can't restart asterisk from script
6:29PM 0 FreePBX IP Phone Recommendations
5:53PM 7 Need help/suggestions for DialPlan
4:48PM 0 AEX800P on HP Prolaint ML115 - Error on Module Load -
2:35PM 2 Problem with Asterisk and SPA-3000
11:34AM 1 Configure DAHDI with TDM410 for analog
10:24AM 3 SkypeForAsterisk
8:11AM 2 How to backup Trixbox 2.8.0.3
8:11AM 0 SIP_CODEC related question
7:54AM 2 app_voicemail. Help me to find typo source ...
7:04AM 1 Recording from billsec
2:54AM 5 Interesting problem with IP's
12:06AM 4 dahdi restart kills server
 
Tuesday December 8 2009
TimeRepliesSubject
9:45PM 5 Easy way to see what dahdi channels are being used
9:12PM 1 meetme.conf adminpin - what does it do?
7:23PM 2 network config
6:43PM 2 Sangoma A101DE with Dell PE 2850
5:47PM 5 E1 Channel Numbering - Your Comments.
5:25PM 3 Asterisk throws error using the alsa module
5:14PM 0 Directory application: First DTMF digit is missed if pressed during "using your touch tone keypad..." announcement
3:53PM 0 might have found and issue
3:44PM 1 Voicemail issues
2:51PM 1 Asterisk Voicemail
1:15PM 5 Starting and installing Dahdi (2.2.0)?
7:03AM 1 G729 with IAX
12:58AM 1 Configure DAHDI with TDM410 for analog Modem calls
 
Monday December 7 2009
TimeRepliesSubject
10:14PM 0 SPA921 Help
8:34PM 2 realm authentication
8:30PM 3 Automon -> Voicemail
7:30PM 1 REFER to trunk
6:05PM 10 automon => *1 "one touch recording"
2:58PM 0 Why no re-register when sip register status is UNREACHABLE
2:06PM 2 g722 question
8:00AM 4 show queue's name and other info in incoming call to queue member
5:04AM 1 Error : SIP/2.0 401 Unauthorized
 
Sunday December 6 2009
TimeRepliesSubject
9:52PM 0 realm
7:11PM 1 Linksys SPA9x2 echo problem
4:22PM 2 ABCTI: first usable beta
3:38PM 1 sequential dialing preferences
1:49PM 4 Call Limits
8:56AM 2 Example to handle incoming calls without callerid at home?
7:40AM 0 Sangoma U100
7:24AM 1 Can Asterik act as a SIP Proxy
1:25AM 8 Asterisk to Email
 
Saturday December 5 2009
TimeRepliesSubject
10:38PM 10 Setting up skype
9:02AM 5 G729: TC400B vs Software Encoder
4:57AM 15 How to use SIP hints and BLF for realtime extensions on Aastra phones?
 
Friday December 4 2009
TimeRepliesSubject
11:39PM 0 The SIP in the Mobile Phones are not able to register on asterisk
7:36PM 0 DAHDI - Split data voice use
7:01PM 4 DAHDI outgoing
6:49PM 0 No audio - using g729 codec altogether
4:43PM 1 DAHDI issues on 1.4.26.1
4:41PM 3 MWI count wrong when using IMAP and VM
4:37PM 3 IAX2 Port issue
4:27PM 0 Today in 30 minutes: VoIP on Social Networks
2:15PM 0 Get back in dialplan with number-parsing
1:42PM 4 Dahdi_genconf does not generate NT/TE configuration
10:54AM 3 spandsp version
8:41AM 4 Get Queue values from dialplan (Was: queue_variables() function)
6:32AM 3 hey please help me my 3rd email of how to change From fileld username in sip packet
5:56AM 0 chan_sip Error
12:22AM 3 Multiple Channel Variables with AMI Originate
 
Thursday December 3 2009
TimeRepliesSubject
11:20PM 0 queue_variables() function
10:51PM 1 only the first ResetCDR works after upgrade to 1.6
7:48PM 8 Source-IP on Asterisk DRBD/-HA-Cluster wrong
7:45PM 0 Repost: Working in useful examples... and freenum/e.164 dialing in extensions.conf.example
7:15PM 11 multiple sip trunks
1:49PM 6 Wi-Fi sip phones with auto provisioning
1:01PM 0 ChanSpy gets stuck
11:15AM 0 AEL, 1.6, CUT and commas [SOLVED]
11:09AM 0 AEL, 1.6, CUT and commas
9:03AM 16 Fax throughput - Asterisk 1.6.1.9
8:21AM 0 softphone @handheld
7:57AM 2 Dial application with M option
7:27AM 0 sip show channels shows non-existent channels on 1.6.0.19 and 1.4.27.1 ?
6:39AM 4 Feature Request: "SayNumberFiles"
5:28AM 1 Flashing Cisco 7941 to SIP
12:13AM 2 dahdi_tool shows no alarms, but no line connected
 
Wednesday December 2 2009
TimeRepliesSubject
10:10PM 0 AsteriskNOW sip module not installed
10:02PM 9 Rsrvd state and off hook dahdi issue
9:10PM 0 FW: Variable Name needed
8:05PM 4 Variable Name needed
6:25PM 0 Asterisk-Addons 1.4.10, 1.6.0.4, and 1.6.1.2 Now Available
5:59PM 0 dialout problem with analog phone
5:32PM 6 Help configuring Audiocodes MP-104 FXO
4:43PM 0 Asterisk Queues Tutorial updated - Hot-Desking without Agent Channels
2:58PM 0 fsk callerid with DTAS start(like dtmf in issue 9096)?
2:32PM 5 b option in Directory
11:47AM 3 Problem with Timeout
1:20AM 2 Featuremap help
 
Tuesday December 1 2009
TimeRepliesSubject
11:23PM 9 Question about g729
11:06PM 5 Slightly OT - Oreka Call Recording
9:55PM 0 Codecs negotiation
8:56PM 2 OpenSBC
8:49PM 0 SafiServer and SafiWorkshop 1.2 With Web Services Released
3:32PM 5 Asterisk registers with private IP
1:09PM 2 "Dropping incompatible voice frame" error
12:58PM 0 Asterisk - Segmentation fault
11:00AM 2 Issue with T38 fax Calls
8:40AM 9 Patch for app_dial.c: exit when just one ext is busy.
8:21AM 0 Asterisk radius configuration
7:16AM 1 Asterisk Configuration with Sphinx speech engine
12:46AM 1 AGI