asterisk users - Dec 2011

Saturday December 31 2011
10:19AM 4 Outbound Dialer, Agent Login and Logout
Friday December 30 2011
11:11PM 14 High verbose set at console effects the logger file "Full" - Why is that?
2:40PM 1 wait_for_answer: Unable to write frametype: 2 on Asterisk *CLI>
2:00PM 0 Asterisk Now Available
11:17AM 0 Best practise for operator
10:07AM 13 Asterisk 1.4.42 NOTIFY replies ignore NAT setting
8:44AM 2 Problem installing B410P BRI card for asterisk
3:13AM 0 Asterisk Video Playback - MP4 ad 3GP files
1:39AM 3 Block Specific Number on Inbound
Thursday December 29 2011
11:51PM 0 Asterisk Registrar / Trunk
9:55PM 0 can't set up tcp sip - sip connection : digest <s> problem
9:16PM 0 How to create SIP INVITE with different To: Header field than Request-Line URI
3:03PM 4 Asterisk fail2ban filters - show us yours
2:21PM 0 Softphones
11:49AM 1 IAX2 woes
9:37AM 0 Help_In Voicemail , vedio play but voice is not here out.
9:12AM 0 Function TESTTIME example [SOLVED]
4:16AM 7 Client - registers but unreachable
4:07AM 10 Interesting attack tonight & fail2ban them
12:01AM 0 func_odbc not returning whole smalldatetime MS Sql field.
Wednesday December 28 2011
9:32PM 1 MFCR2 Long distance calls not connected
8:57PM 1 Question on hung channel
8:33PM 12 1.6 and 1.8
8:16PM 18 Monitor Command Records separate channales
7:59PM 3 Asterisk forcing uLaw bug NOT fixed yet
10:47AM 0 Chan_ss7 clustering config with single point
10:14AM 0 Direct media path on Avaya IPOFFICE and Asterisk with H323 Trunk
8:25AM 9 DTMF Testing software to test IVR system
12:50AM 0 sendvoicemail=yes not quite working SOLVED
12:46AM 1 cdr call time
Tuesday December 27 2011
7:43PM 3 Call going into s-extension
7:34PM 1 maximizing sound quality in 10.0
12:48PM 3 how to used SIPp for sip load testing
9:26AM 0 read dtmf digits on connected calls
6:54AM 13 how to stop hacking of my server
3:05AM 1 odd "secret" problem
Monday December 26 2011
6:19PM 0 Working on web based IVR Designer for asterisk and Freeswitch
4:27PM 0 Debugging call quality issues in Asterisk
2:55PM 9 how to listen on different sip port for a device?
11:35AM 5 Function TESTTIME example
11:03AM 0 Which language and tool to write diaplans ?
10:52AM 0 AEL2: How to get rid of "does not end with a return; I will insert one" warnings
5:18AM 5 Not able to play wav files in asterisk
Sunday December 25 2011
10:41AM 6 Is Asterisk 1.4 compatible with 1.8.7 ?
Saturday December 24 2011
5:36AM 0 Vicidial license
Friday December 23 2011
3:57PM 0 AEL2: How get rid of "expression has operators, but no variables" warnings
3:40PM 8 Codec warnings after upgrade to 1.8
12:54PM 4 execute command just after Dial()
10:12AM 0 Any one using VICIDIAL?
9:33AM 3 GotoIfTime days query
9:03AM 2 Dahdi not installed and application's details is missing in Asterisk
7:48AM 0 2011 The Year in Review - where were you when...
Thursday December 22 2011
7:49PM 0 How to read doc in /var/lib/asterisk/documentation
5:33PM 10 Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf
4:34PM 3 Properly Escaping Quotes and Commas
10:31AM 5 Using shell script output into phoneprov.conf's custom variables
8:51AM 0 Asterisk Developer
7:43AM 1 Number of Calls
12:14AM 6 dahdi_tool missing
Wednesday December 21 2011
8:57PM 1 Diagnosing call hangups
3:03PM 3 Suppress -- Remote UNIX connection message
1:41PM 1 Asterisk call file size calculation
11:48AM 2 Why **CONGESTION** not *****NOANSWER****** ?
10:36AM 0 Asterisk 1.8 - RTPAUDIOQOSLOSS does not show audio quality
8:50AM 4 dahdi: Unknown symbol kasprintf
7:13AM 1 OT - Which switch to play with LLDP-MED
1:03AM 2 sendvoicemail=yes not quite working
Tuesday December 20 2011
7:39PM 4 GOIP GSM to SIP Gateway?
4:21PM 1 OOH323 config file
1:47PM 4 Help_video call not run
12:56PM 0 Asterisk Sip Media Call Type
12:04PM 3 File Convert
12:44AM 4 Limit # of inbound calls on SIP trunk
12:38AM 5 Use different local IP for each SIP trunk
Monday December 19 2011
9:08PM 1 Asterisk Now Available
8:06PM 8 India Telecom regulations
8:02PM 0 A lot of 603 Declined Error form iCall - Are they going down or is it just bad service?
6:24PM 0 Which Dahdi/Libpri version are you using ?
6:13PM 1 Dahdi - Strange Warning
6:13PM 0 ChanSpy in whisper mode - low quality audio
2:47PM 0 Sending Fax Dialplan with Retry Attempt
11:14AM 5 AMI and Dialplan
Sunday December 18 2011
7:19PM 1 asterisk and heartbeat
3:07PM 0 Called peer IP
7:42AM 7 Asterisk now sends rport always
5:26AM 3 How to monitor SIP Trunk on production server
Saturday December 17 2011
11:55PM 9 No rtpmap codec info in 200 OK
5:56PM 1 asterisk-users Digest, Vol 89, Issue 32
9:11AM 1 AstLinux 1.0.0 release
12:38AM 4 Set Caller Number in E1 PRI ISDN Lines
Friday December 16 2011
9:02PM 6 Dialing problem with Polycom phones after SIP update
8:06PM 4 FreePBX not updating configs on 1.8 RPM install
5:39PM 0 Problems with DAHDI Channels
5:32PM 3 fromstring in voicemail.conf
5:00PM 2 Errors on RBS T1
3:57PM 0 ConfBridge 10 How can I playback a soundfile to an existing conference
2:52PM 0 asterisk-users Digest, Vol 89, Issue 29
2:38PM 6 CDR END TIME in correct in 1.8+
12:06PM 4 ODBC problem - static realtime file not loading
10:29AM 2 SIP Trunk
4:50AM 0 Contexts and Extensions
2:31AM 6 followme forking/parallel dialing breaks when 1 sip device unreachable
2:03AM 2 Asterisk console suddenly extremely verbose...
1:03AM 6 Which device auto-registered an extension?
Thursday December 15 2011
9:58PM 0 Asterisk 10.0.0 Is Released!
9:55PM 0 Asterisk Now Available
9:21PM 0 Struggling with Extensions in Realtime
8:10PM 3 Call pickup on Asterisk 1.8 and Polycom IP550s?
6:33PM 7 Best PBX for Call Centers?
5:46PM 1 Asterisk log format
4:28PM 7 Play audio file for both Caller and Callee in a call
3:41PM 4 Partner Keys on Innovaphone
2:47PM 0 app_swift tts module - new home.
2:05PM 5 Digium TE205P leds flash red on startup
11:59AM 0 how to get the Record_ID
11:51AM 1 Wrong call information on B leg
Wednesday December 14 2011
12:56PM 3 Asterisk 1.4.x segfaulting daily
9:18AM 8 A few (simple?) questions
7:28AM 6 get start-time of all active calls
Tuesday December 13 2011
9:53PM 0 [hint state][BLF] Asterisk 1.8.7 does not send RINGING notifications, even with notifyringing=yes
9:44PM 0 Login agents on asterisk startup according to hints state
6:00PM 1 AEL x LUA
Monday December 12 2011
11:40PM 2 SIP MESSAGE outside calls - state of the art?
10:31PM 0 TLS bug in asterisk?
10:11PM 0 Asterisk Configuration GUI Question
7:55PM 1 ATA with TCP/TLS support?
7:42PM 0 Looking for partners to develop Asterisk Call Centre Applications - A call to investors and programmers
6:44PM 2 How to query Microsoft SQL server for caller-id source
4:55PM 0 How to see initiall dialled extension in CDR records ?
4:36PM 6 VoiceMail and IMAP
2:51PM 1 Help needed for chan_ss7 for Digium device
2:23PM 3 MySql Custom CDR issues
11:16AM 0 Asterisks Statistics (Albert)
10:21AM 2 Which port should be open for asterisk communication
9:21AM 0 How to count ongoing calls from the dialplan
5:36AM 0 How is everything doing?
3:59AM 8 What version to upgrade to...?
12:22AM 5 Multiple route failover zaps registration
Sunday December 11 2011
7:32PM 0 Chan_sip How to store Register Call ID?
Saturday December 10 2011
8:53PM 0 Asterisk 1.8 and measuring Post Dial Delay
Friday December 9 2011
5:16PM 0 Asterisk 10.0.0-rc3 Now Available
4:51PM 2 asterisk-users Digest, Vol 89, Issue 13
4:48PM 0 Scheduled Maintenance for Asterisk Project community services
3:52PM 2 Issue with dahdi 2.5.0 and Digium HA8-B400M
1:27PM 1 Mixing asterisk.conf, asterisk.ael and asterisk realtime
6:10AM 1 Preparing to store vm in database
5:55AM 2 Trying to send customer mwi updates
Thursday December 8 2011
11:34PM 0 Asterisk 1.4.43,, and Now Available (Security Release)
10:58PM 0 Fax client for Windows
10:48PM 0 AST-2011-014: Remote crash possibility with SIP and the “automon� feature enabled
10:47PM 3 AST-2011-013: Possible remote enumeration of SIP endpoints with differing NAT settings
6:21PM 0 Sip.conf and extensions.conf configuration for Exchange 2010 U.M.
2:26PM 2 random digits dialing during call
2:06PM 4 libpri / ISDN feature ECT (explicit call transfer)
12:46PM 3 [OT]: Require suggestions - GSM Gateway <-> Asterisk
8:26AM 9 Issues with dahdi show status output (and check IRQs)
7:46AM 1 How to make app_meetme enable
5:42AM 4 Confrence call is not make
Wednesday December 7 2011
9:55PM 24 ISDN PRI configuration
8:31PM 6 Help! Logs filling up with errors!
8:10PM 1 ChanSpy() and Spygroup
5:56PM 5 Realtime Registration
3:16PM 1 redirect a ringing phone
2:24PM 1 SS7 + T1
12:15PM 1 ss7 installation and configuration
4:26AM 0 IRC Client
Tuesday December 6 2011
5:58PM 0 Proper sip.conf and extensions.conf for Exchange 2010 U.M.
4:57PM 3 rasterisk not knowing config path?
4:34PM 0 Asterisk 1.4 - Help/Doc for Park() application [SOLVED]
10:11AM 0 Data provided by pri show spans and dahdi_tool do not match
10:08AM 5 How to use Hints in asterisk
9:24AM 0 help please (D 300 JCT)
9:03AM 2 Talk detection in meetme
Monday December 5 2011
9:13AM 2 Asterisk 1.4 - Help/Doc for Park() application
7:28AM 1 How to count available parking slots from diaplan
Sunday December 4 2011
6:30PM 0 PRI cards D 300 JCT E1
Saturday December 3 2011
8:36PM 8 Simple Generic IVR to get us up an running Quick
1:10PM 4 Hint'ing with XMPP?
5:59AM 6 google voice calling dial plan question.
Friday December 2 2011
11:48PM 0 skype connect & early media
11:28PM 0 Max channel analyser from asteriskcdrdb?
11:27PM 0 DHCP Option 43 and pfSense + Asterisk
11:24PM 2 How can I decipher password in SIP Packet?
11:13PM 1 Where to download sample video file for Asterisk 1.8x playback?
9:44PM 0 IAX - An informative question
5:37PM 1 CSipSimple audio issue with DAHDI/IAX2 calls
10:18AM 11 DIALSTATUS Values
Thursday December 1 2011
9:48PM 1 Locally bridging channels when using SRTP?
8:35PM 2 Can't get off Europe/Bucharest timezone
12:44PM 3 Issue with Polycom SPIP650 and its sidecar [SOLVED]
11:57AM 8 Populate CDR issues
12:42AM 11 AGI script that uses google's text to speech engine