asterisk users - May 2009

Sunday May 31 2009
8:45AM 1 Ekiga, Twinkle and from where to start with open source
8:32AM 0 Network settings and quality of voice
7:11AM 1 An outside Caller ID not shown,
7:11AM 1 h323 guide for asterisk
6:57AM 0 Multile IP addresses for SIP device
2:15AM 1 Problem releasing call from a SIP extension
Saturday May 30 2009
11:16PM 0 question about reinvite
10:09PM 0 POS Modems
9:38PM 1 Problem T.38
6:35PM 2 Simplex voice on TDM410P
4:17AM 5 Understanding Call Handling In Asterisk
1:10AM 0 compile error for chan_h323
12:50AM 2 Queue - Multiple Transfer
Friday May 29 2009
4:18PM 0 dial out context from incoming iax trunk
3:39PM 1 Asterisk Clustering
3:28PM 1 Attended transfer and dialplan
2:56PM 0 How to read values from another channel ?
2:50PM 0 Logging into queue homed off remote system
9:21AM 2 regarding to field of accountcode
8:33AM 1 IAX2 trunking with Older Asterisk version ?
8:22AM 3 AMI and Originate on
7:57AM 1 connection fail between Service provider's proxy server and my asterisk server
6:27AM 1 how to detect dtmf in meetme
5:22AM 4 asterisk and dial plan changes
5:10AM 1 CAll-limit or incominglimit ?????
4:26AM 2 To: Field
12:38AM 1 Call telco transfer q931
Thursday May 28 2009
10:12PM 0 Best Current Release for Long Term Use
7:50PM 2 zaptel installation
7:08PM 0 Friday at 12 Noon EDT: Jim Van Meggelen on the VoIP Users Conference
6:16PM 2 probably an rtfm but... need to dial out to 2 PSTN lines from AMI
1:52PM 1 asterisk 1.4.X, T.38 and NAT
12:49PM 0 Reg AsteriskNow 1.5 Beta Release
Wednesday May 27 2009
7:15PM 0 RLT Transfers and NI2
5:52PM 1 Auto-congesting call due to slow response
5:51PM 1 Playtones Volume
5:24PM 1 setting CDR values on failed calls
2:46PM 3 Now "Unable to create ... 'DAHDI'"
1:37PM 2 problem with T.38 media headers
12:54PM 1 TDM400P in PCI-X Slot
12:15PM 0 No full duplex communication ?
11:49AM 2 Pressing number 2 in dialplan
11:32AM 1 stucked calls in asterisk 1.4
11:09AM 2 AstDB wildcards
8:19AM 1 DAHDI and hangup issue when playing the IVR
6:58AM 0 Delay and Zombie Channels Problem
6:33AM 1 Asterisk memory problems
6:10AM 1 PHP AGI Problems
12:46AM 3 Call in progress tones
Tuesday May 26 2009
8:03PM 0 multiple bind ports with TCP and UDP
7:57PM 1 Fax Machines across carrier SIP trunk? General recommendation?
7:30PM 1 Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?
6:30PM 3 Silly (??) question about chan_dahdi
6:30PM 0 Strange message in CLI
5:34PM 1 STUN setting in Asterisk 1.6.X
5:20PM 0 No Voice - only "noisy audio"
4:37PM 0 Suggest good calling service for London
4:08PM 0 How to register with TCP transport ?
3:20PM 1 SIP over VPN
3:13PM 0 CDR after SIP blind transfer.
2:41PM 3 FXS
2:09PM 5 Maximum cable length for analog phone from FXS port
2:07PM 1 A problem in playing sound files
12:46PM 1 Hanging up a call by DTMF
11:39AM 2 Converting Cisco 7961 to SIP
11:32AM 8 Bandwidth management and ADSL router
11:14AM 2 Domains
11:11AM 0 Logging calls made/lost
10:48AM 1 RDNIS question
9:57AM 1 h extension and channel variables
7:52AM 2 asterisk-addon 1.6.1 problem
Monday May 25 2009
9:48PM 1 Placing a MWI on-off call...
9:29PM 4 howto store local exchange prefixes ?
5:46PM 0 DNS issues again
4:42PM 1 New tutorial: storing audio recordings per day
12:40PM 1 SIP Trunk groups
12:14PM 3 Problem running Dahdi
9:54AM 0 Connected Number on incoming calls with mISDN
5:34AM 1 Basic Config
Sunday May 24 2009
8:00PM 7 Asterisk, SQL Database Update
7:01PM 2 Can I run two instances of asterisk
12:50PM 0 Duplicate DTMF digits
7:16AM 0 RPID on SNOM phones?
Saturday May 23 2009
8:41PM 0 [OT]I like this community
5:57PM 0 hey
4:23PM 1 Unknown signalling method 'pri_cpe' ??
4:03PM 2 sip.c: "Serious Network Trouble" ??
7:39AM 0 Asterisk automatically closing the file descriptor
6:41AM 1 integrating CTI
12:01AM 1 Faxing issues
Friday May 22 2009
10:39PM 1 visp multiaccount + firewall configuration problem
9:28PM 4 How to stop a background music
7:31PM 9 DTMF
5:51PM 2 BT ISDN-30 Pri getting 'stuck' on outgoing calls.
5:36PM 3 No response to our critical packet problem
2:49PM 0 Alison Keenan (free British English voice)
1:40PM 2 Indications.conf and tone generation volume
12:33PM 1 VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...
11:33AM 1 /etc/asterisk/startup.d
10:23AM 1 rasterisk r processes take the rest of my cpu
9:43AM 1 Can't get G.726 to work.
8:27AM 1 Error ON SIP Incoming TOS
8:04AM 2 Memory leak on asterisk
7:23AM 3 Parsing Asterisk's .conf files from Perl, Java or PHP file
7:06AM 0 " circuit-busy" message
Thursday May 21 2009
9:55PM 0 Free Fax for Asterisk Receiving problem
9:13PM 0 Cheapest price to cuba route !!!
9:06PM 0 calls stuck in AMD even after analysis time
9:02PM 3 Monitor problem, Asterisk 1.2.13
8:02PM 0 Asterisk-Addons Now Available
8:01PM 0 Asterisk 1.4.25 Now Available
3:36PM 0 Page/Intercom problem
3:27PM 1 Calling party category
2:27PM 2 Zaptel Error
1:05PM 1 reg static build
11:41AM 1 playing media(moh,prompts) from flash player
9:20AM 1 FW: Writing Hangup causes to CDR record
9:18AM 0 Writing Hangup causes to CDR record
9:01AM 2 Jitter buffer question
8:56AM 2 MeetMe not working with GSM codec?
8:04AM 2 Polycom Productivity Suite
6:10AM 1 interruption in queue
3:06AM 3 PSTN Connection
1:52AM 4 Bridging INBOUND PRI to OUTBOUND PRI fails with Monitor()
12:28AM 1 Voicemail playback NEWEST first vs. OLDEST first
12:22AM 0 -> segfault
Wednesday May 20 2009
8:31PM 3 circuit busy message
8:31PM 1 DAHDI fun and games
6:39PM 2 Do I need a SIP Proxy for this?
4:49PM 1 Macro with DIALSTATUS
3:32PM 2 play with varibles
2:56PM 2 MeetMe - Different pin for different user
2:52PM 2 asterisk memory (issue)
2:38PM 2 How to detect switch to voicemail when calling to mobile phone
2:35PM 0 Step-by-Step Asterisk and MeetMe Help
2:04PM 2 asterisk crash on DAHDI error: No more room in scheduler
1:58PM 0 dtmf=info and canreinvite=yes
1:16PM 1 Queue and Dial operation - Common Variables?
12:49PM 1 Pickup with *8 is not working...
12:04PM 4 FritzBox 7270
10:58AM 2 Problems receiving some faxes in T.38
10:56AM 0 inbound SIP funnies
10:03AM 1 Channels configuration with DAHDI
9:49AM 0 Feature request: "database show" from manager API [SOLVED]
9:30AM 1 TC400
6:07AM 0 FaxIn problems
4:44AM 3 Asterisk CCM, CME Integration
3:06AM 2 Manager ExtensionState function
3:04AM 1 What codec/sample rate/resolution...?
Tuesday May 19 2009
10:26PM 9 Hang at 5:34 pm EST
10:08PM 0 announcement: chan_nms - channel driver for NMS Communications hardware
8:50PM 1 Dialplan matching problem
8:47PM 2 Feature request: "database show" from manager API
6:57PM 0 cdr record disposition always FAILED
6:22PM 1 Alternative to Adobe Audition 3 for G723 > G711 uLaw ? (old Cool Edit Pro)
6:16PM 8 Ghost ??
5:51PM 2 Question
3:37PM 3 Dialplan Priorities and Sort Order...
2:47PM 2 Unable to make outbound calls
10:36AM 0 How to access voicemail from deskphone
9:02AM 1 SPA941
6:04AM 5 OT: SIP hardphone with multi-color BLF
5:47AM 0 Playing audio messages to the callee.
4:35AM 0 Realtime Static on
Monday May 18 2009
9:41PM 1 with Realtime Mysql
8:30PM 2 From 1.4 to 1.6.0
8:19PM 7 callcenter / dialer / predictive dialer / vicidial program is now open
7:31PM 3 Number of max SIP calls.
6:36PM 1 meetme
5:48PM 2 Some direction for creating sound files for Asterisk
5:44PM 0 ${HANGUPCAUSE} is not printed when call ends or is interrupted
2:57PM 0 no video by Originate
10:57AM 1 Panasonic SIP Phone
10:57AM 1 Problem with Free Fax For Asterisk
9:56AM 4 Open source SIP client
8:36AM 2 Manager API in PHP
Sunday May 17 2009
8:35PM 2 Calls Declined
7:07PM 4 Can YOU find a trailing parenthesis?
2:10PM 1 SHARED() variables and <ZOMBIE> channel
10:43AM 1 Capture "Server" header in SIP reply.
8:26AM 0 Correction IRC Channel name - was TODAY May 17 Sunday Asterisk VOIP Conference server & Ekiga for BerkeleyTIP
8:20AM 0 TODAY May 17 Sunday Asterisk VOIP Conference server & Ekiga for BerkeleyTIP
Saturday May 16 2009
1:49PM 2 Agent-Login/out in 1.6
1:21PM 1 howto set up persistent dynamic meetme
11:46AM 4 Fwd: Asterisk With Cisco Voice Router
8:37AM 1 Queue Load, Asterisk Disconnected
Friday May 15 2009
9:40PM 2 Logging In / Out Agents on Asterisk 6 ???
8:15PM 2 change AGI script return result
7:27PM 0 What happened here when transfering a call ? Circuit-busy ???
6:13PM 0 Mediant 1000 audiocodes and Trixbox
5:39PM 1 meetme dies looking for conf-getconfno
4:27PM 0 Strange SIP Activity
2:38PM 1 Spiral SIP Request problem
1:35PM 1 help a bald guy
1:24PM 0 Asterisk open source project servers have new names!
10:54AM 0 Zap Transfer
10:22AM 0 DNS host name resolution in iax.conf
9:57AM 1 DTMF Recognition
9:06AM 1 Fax t38 capability
6:38AM 0 Friday May 15 @12 Noon EDT with Askozia pbx
4:47AM 1 how to ignore a ring on a line
12:59AM 0 howto build oslec with dahdi-linux- or svn?
Thursday May 14 2009
11:27PM 0 Cross-compiling asterisk
10:57PM 1 comedian mail
10:44PM 0 polycom soundpoint question
10:37PM 3 how to avoid call waiting? Or check DIALSTATUS before Dial()?
10:11PM 0 Asterisk 1.4.25-rc1 Now Available
9:12PM 1 Goto not matching
8:04PM 1 Parked Calls Problem
3:06PM 1 Digium TDM 400 or Openvox A400P
2:26PM 0 SIP error message
1:25PM 0 FW: [Dean Collins] Joint BarcampNYC4 sessions?
12:58PM 0 Problem with Asterisk 1.4 and Linksys Spa941/962
12:46AM 2 Problem with Asterisk + TDM410 FXO
Wednesday May 13 2009
10:59PM 1 Double dial.
10:55PM 2 Help need to do Lookup from odbc database
9:05PM 1 High Volume US Traffic? Claim DIP Compensation!
8:04PM 1 #-all.gsm
5:39PM 1 Asterisk+a2billing for over 10,000 ext
5:35PM 0 Why asterisk changes RTP destination port when it receives first RTP packet in opposite direction despite canreinvite=no
5:08PM 2 Voicemail and remote directory with SSHFS
4:21PM 0 AstriCon 2009 speaker submissions open!
3:53PM 2 Add Monitor application to call suppresses audio
2:34PM 1 Sangoma FXS dialmap
2:21PM 0 Request for feedback/testing on Multicast RTP Paging
1:00PM 4 Switchvox
12:54PM 2 Proxying from one server to another
9:22AM 0 AGI scripts in Groovy, JavaScript, JRuby or PHP running on the Java Virtual Machine
7:43AM 4 Free Fax for asterisk
7:30AM 1 Asterisk 1.6 T.38 generation towards a Cisco voice router
5:46AM 1 no source on cdr logs in some cases!!
Tuesday May 12 2009
11:07PM 1 enum agi interesting problem
8:08PM 2 Hangup()-command does not hang up the line
6:31PM 2 Is anyone keeping up with the versions?
4:26PM 1 Wanting to manipulate SIP response headers
6:18AM 0 Help with radius
3:44AM 2 Asterisk Manager API Action Originate
12:09AM 1 [asterisk-user] Which policy for ISDN BRI support in NT/PtMP ?
Monday May 11 2009
8:44PM 1 Anyone with a working pfSense firewall configuration?
5:54PM 1 Ready to put the box on the net
4:02PM 1 PauseMonitor() Hanging Up Call
3:54PM 1 Problems with res_odbc
3:24PM 3 Asterisk w/ Nokia "e" Series Handsets
12:43PM 0 Asterisk Now Available
12:12PM 2 DTMF received twice
12:09PM 0 No CDR generated for calls to queues with no agents
11:53AM 0 Polycom-330 not displaying line & buddy label?
7:39AM 1 Support of /* */ comments in ael.vim
7:30AM 3 How to write custom functions in AEL2 ,
3:21AM 6 Building a System.
Saturday May 9 2009
8:22PM 1 Special Dialplan
2:51PM 2 asterisk blade server
9:39AM 5 Rusting Snoms?
8:42AM 1 Incompatible changes to asterisk 1.6 MYSQL addon query syntax
7:49AM 1 A side of Digium you may have never seen
5:56AM 5 Professional Setup..
4:47AM 2 Unable to run asterisk CLI commands from php
3:30AM 4 VoIP over satellite internet
2:17AM 1 determination of where a call is placed from (physical location)
Friday May 8 2009
10:46PM 2 Possible to add Voice delay?
9:39PM 2 Storage capacity for call recording
9:31PM 1 Record all calls
6:55PM 0 Leg-based CDR proposal updated; Major mods
5:54PM 0 G279 install in ? [SOLVED]
5:24PM 0 The efficient way to add MeetMe to pure SIP install ?
4:35PM 0 G279 install in ?
4:11PM 1 Asterisk can't dial out on Sangoma b600
3:39PM 2 AMOOCON debriefing
3:26PM 0 Proxying comparison
2:23PM 2 Override sip.conf settings in extensions.conf? Possible?
1:20PM 0 DNID Truncated
1:03PM 0 Numeric Hangup Code
1:01PM 2 Configuring SIP Trunk
12:38PM 0 Difference between Transfer and Dial applications
12:26PM 0 Can't GOSUB_RESULT with Dial U() option ...
11:55AM 2 "pri show spans" shows nothing
10:11AM 0 G.722, 1.4 and IAX trunking ...
8:32AM 2 Not receiving voicemail message in mailbox
Thursday May 7 2009
11:21PM 1 func_odbc.c: Unable to execute query
10:45PM 0 Default dahdi fxs behavior
9:03PM 0 [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/1-1]
7:54PM 3 QoS & VPN
7:41PM 0 Voicemail format - no transcode?
6:20PM 1 Macro arguments on app_queue
5:49PM 0 asterisk-users Digest, Vol 58, Issue 17
3:44PM 1 How to get meetme participants in dialplan?
2:04PM 4 Voicemail Alert
1:29PM 3 Master.csv
9:50AM 0 How ro store Reject cause
8:48AM 0 pri errors..
6:14AM 7 Sangoma a104d and channel banks
3:19AM 3 Messaging System
1:00AM 1 Voice Mail Delete Notification
Wednesday May 6 2009
10:21PM 2 Cisco 7960G with static config
10:18PM 1 Asterisk with Sphinx
7:50PM 3 Polycom Dialplan Digitmaps
7:05PM 2 Where are 2 letter language values defined?
6:36PM 0 How to get SIP resposnse codes
5:04PM 4 shut down a single PRI on a running Asterisk system?
5:00PM 0 astcc - outgoing call does not hangup properly
4:53PM 2 Understanding Codecs
4:48PM 2 Cisco 7940 phones become unreachable over VPN after a time
4:15PM 0 Bridge() and Goto() and dialplan contexts, oh my!
4:04PM 1 ConfBridge versus MeetMe
12:43PM 3 Questions on X100P/X101P cards
12:17PM 1 precision of wait dialplan application
9:15AM 0 After transfer context
7:16AM 1 Voicemails do not email through asterisk
4:31AM 0 Caller information in Web
2:37AM 0 problems in h323 channels
1:10AM 0 John Todd, Moises Silva Speaking At ClueCon 2009
Tuesday May 5 2009
6:11PM 1 SIP _call_ to Asterisk box
3:38PM 2 chan_mobile and DTMF
2:02PM 1 stop the MOH since asterisk knows that channel is ringing
2:00PM 0 OT: Polycom handset cord detangler
1:52PM 1 "Asterisk cmd MYSQL" app_addon_sql_mysql / performance ?
12:08PM 0 Dial with MOH
10:05AM 1 Beginning to use Asterisk and tests with extensions
6:52AM 6 Preferred language for Asterisk AGIs development ?
2:56AM 0 asterisk-users Digest, Vol 58, Issue 9
2:52AM 2 1.6.1 app_fax: WARNING T.30 ECM carrier not found ??
1:45AM 0 need BT102 firmware (current version)
1:12AM 1 Asterisk cdr_odbc problems
1:10AM 4 AMI + AGI for outbound click to dial
Monday May 4 2009
8:17PM 3 AGI PHP
6:59PM 4 Cisco phone - can Call manager reflash automatically if we test in Asterisk with SIP?
6:50PM 3 wireless ATA
5:27PM 1 AOC (advice of charge) current status.
4:57PM 0 Channel/Mute
12:08PM 1 hint limitation
9:57AM 1 advice on OrderlyStats (or other cc software)
8:03AM 1 Can someone help me with my IAX-registration
7:08AM 0 how to add applications to 1.6???
3:53AM 0 question of flite installation
Sunday May 3 2009
5:11PM 0 asterisk-users Digest, Vol 58, Issue 5
12:58PM 4 System noise reduction
12:30PM 2 Asterisk not starting up due to database problems
4:31AM 0 SIP Extension Registration and Security
1:37AM 2 Sangoma Wanpipe Driver Compile for DAHDI Failure
Saturday May 2 2009
7:56PM 1 Module not loading
6:30PM 2 Asterisk and ODBC
4:46PM 0 ISDN Error Code 42
Friday May 1 2009
10:49PM 0 2 phone extensions on a single conference room
3:02PM 1 AGI - Ways to create a call
9:57AM 9 LoadAvg , Codec and Bandwidth Utilisation
8:30AM 5 New system for recording - SCSI, SAS or SATA?
8:26AM 0 May 1st @12 Noon: VoIP and home automation and control
8:12AM 0 Failed log in
6:26AM 1 cdr_mysql custom fields ?