asterisk users - Sep 2009

Wednesday September 30 2009
11:57PM 3 Choose IAX or SIP trunking?
11:40PM 0 Calls at 2 different locations
8:12PM 2 E1/T1 Tapping call recording in Asterisk - Testing needed
7:08PM 1 SIPAddHeader into the SDP?
6:52PM 1 EXTENSION_STATE Asterisk 1.6
5:34PM 0 PBXNSIP Registration Issue
3:08PM 6 question on pri intense debug
1:07PM 1 How to finish a Meetme
10:48AM 1 put some IVR into a queue after the call queuing
Tuesday September 29 2009
9:22PM 0 Dumb Question - Dialing internal and external
8:11PM 2 kill sip user
8:07PM 3 UpdateConfig
6:19PM 1 Asterisk on DD-WRT : modules.conf not found
6:16PM 2 dialing 0 in directory()
3:40PM 0 Static on the line randomly
3:31PM 1 Retrieve Call setup - QoS
3:04PM 3 chanspy and DISA
3:02PM 0 Wrong hint, ringing when idle. after hangup.
2:01PM 3 LDAP integration
11:57AM 1 Native bridging analog phones trouble DAHDI channels.
11:20AM 1 Fax and dial-up connection issues
9:37AM 1 Who am xxx talking to.agi
8:35AM 2 play audio file within an active call
Monday September 28 2009
8:24PM 2 DAHDI channel congested busy
7:07PM 1 How to get "Call-ID" SIP header outside "chan_sip" scope ...
5:14PM 0 strange cisco nat issue
5:10PM 2 Disable/enable CDR in dialplan
3:54PM 0 Asterisk complaning about no such host -- never asked to contact the host it complains about
3:41PM 1 Voicemail - remove option to save in different folders
2:21PM 3 GoTo IF
1:20PM 1 ding Dialled number down a sip channel to a PBX
11:56AM 0 simulate calls for testing sripts
11:32AM 1 Firefox Plugin for Sip Click2Call
10:39AM 0 AGI script
7:05AM 1 TE121P Blue Alarm/Recovering
Sunday September 27 2009
10:38PM 1 DAHDI Question/Choppy Sound
10:31PM 1 MeetMe Hints
9:05PM 2 DAHDI congestion problem
9:02PM 0 Issue with incoming caller-ID to NEC SV8300 with QSIG
9:01PM 0 channel.c:780 channel_find_locked: Avoided deadlock
7:26PM 0 Is channel local what I need?
4:17PM 3 Problems with Digium TDM400 card
10:38AM 0 FW: New in asterisk
9:31AM 1 callfile to auto-answering extension
6:00AM 1 Peers Listed in "sip show channels"
5:47AM 1 New thread - SIP over VPN
1:05AM 1 digium fax: failed to queue document
Saturday September 26 2009
11:07PM 1 Know for how long an agent is talking?
5:59PM 1 Where are phone registrations kept?
2:20PM 3 VOIP solutions
1:35PM 0 Voiced E-mail
12:06PM 0 VoiceMail.conf reading variables
8:41AM 6 Inquiry:Asterisk server remote access
5:44AM 8 Inquiry:How to convert *.wav files ?
Friday September 25 2009
9:58PM 1 "multiple contexts for multiple locations"
9:05PM 1 OT - In which countries are ISDN subaddresses used ?
3:30PM 3 disable dtmf on SIP peer
3:19PM 4 DAHDI disconnect supervision timing
3:15PM 2 How to remove peers from channels
10:01AM 1 Choppy sound, SIP calls within LAN
10:00AM 1 Asterisk Manager Problem
7:02AM 0 ignore flash hook
6:14AM 0 (solved) CPU Spikes in asterisk connected via IAX trunk
Thursday September 24 2009
5:57PM 0 Asterisk and VoIP Users Friday Meeting
5:41PM 4 Polycom push application for microbrowser
4:16PM 1 rtp.conf dtmftimeout
12:56PM 1 Asterisk 1.6 Transfer issue[Edited]
12:54PM 0 Asterisk 1.6 Transfer issue
12:47PM 6 Connecting home intercom to Asterisk?
8:42AM 2 Digium transcoding card
7:14AM 1 CDRs on call forward
Wednesday September 23 2009
10:27PM 1 SFA - No channel cause 66
9:29PM 0 DYNAMIC FEATURES, AEL2 - how to use Goto, Gosub or Macro ?
7:31PM 1 Parking - How to transfer the other party to a given slot
5:12PM 0 Testers Wanted for IMAP Voicemail patch
4:30PM 0 About bug 13115
1:34PM 3 SIP/WiFi handsets?
1:08PM 0 Asterisk success
10:49AM 4 Error When Using Postgresql Schema With Realtime Sip
10:18AM 0 unknown error ON DAHDI please help me...
8:14AM 0 Inquiry:Which codec to get higher download rate on dialup connection
5:58AM 0 SIPP + Duration
5:07AM 3 Bringing people into a conference
4:19AM 4 International Numbering plan ?
3:32AM 3 Simple dialplan issue
12:32AM 1 I need a really simple analog SendFax dialplan
Tuesday September 22 2009
6:29PM 0 Asterisk with Cisco 5300/E1/DSP
5:12PM 5 New Xorcom FXS USB Bank is not loading firmware
4:23PM 1 not hearing audio on console/dsp
3:25PM 0 How to associate a custom script to "core stop now"
2:53PM 0 AstManProxy - No pid file created when run
2:12PM 1 setting up a IP based voip carrier account
12:56PM 4 Asterisk on a Beagleboard?
10:13AM 2 SIPP question
8:43AM 2 Problem with dialplan -> gotoif ?
8:08AM 1 Call deflection on Asterisk
2:52AM 0 Which oslec.h should will work?
1:26AM 1 digium fax: can't indicate condition 19?
Monday September 21 2009
10:20PM 0 Enterprise users going to VoiceCon?
8:37PM 2 MySQL cmd
7:18PM 1 Voicemail to email transcribed
6:46PM 1 Voicemail Crash - ODBC Realtime
3:54PM 2 Atcom AG188N as FXO?
9:23AM 0 Asterisk 1.6 dynamic agents
8:55AM 0 Newbie: How to detect an "*" in Read()?
8:44AM 1 GSM cellphone as cheap gateway?
Sunday September 20 2009
4:51PM 0 Recipe: Automatically Create Dial-able Extensions For Skype Callers
4:27PM 1 A in ACL of sip show peers.
2:58PM 2 different verbose level for full log than to console?
1:20PM 1 DAHDI installation warning
5:42AM 1 Experience with Sangoma's USBfxo
1:40AM 0 Asterisk + Nonoh
12:07AM 0 Stop / Resume in Dialplan / AMI
Saturday September 19 2009
1:22PM 0 "Switchboard"/"advanced answering machine setup" with Asterisk?
10:40AM 0 Sunday 20th Global Asterisk Mtg via VOIP - BerkeleyTIP - for forwarding
7:33AM 3 Sangoma A200 and battery removal detection ??!!!
7:11AM 0 Echo cancellation on DAHDI
6:07AM 1 DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657
6:07AM 0 IPKall using iax
4:43AM 1 "Channels got stuck in asterisk"
Friday September 18 2009
11:45PM 1 digium fax: is this even close to working?
6:33PM 1 No more room in scheduler
6:26PM 2 IAX2 order
5:09PM 3 DUNDi + SIP Realtime
3:43PM 0 Queue Call Disconnection
2:28PM 4 console color
2:23PM 0 Blind Transfer Won't Hangup
2:16PM 0 DAHDI TDM440E still has echo on bridged connections
2:15PM 0 calls drop during attended transfer with PRI line
12:45PM 1 Audio Files
6:46AM 2 Help sending call to local server
2:29AM 1 404 for SUBSCRIBE
1:03AM 1 DAHDI Caller ID problem
Thursday September 17 2009
11:27PM 1 Custom auto-install asterisk using ks.cfg
10:36PM 0 Anyone having issues with res_snmp?
10:19PM 1 CDR Records for MeetMe
7:58PM 2 DeadAgi
5:42PM 0 VoIP Users Conference Friday: Andy Abramson of VoIP Watch
3:31PM 1 I'm not getting the ability to leave a voicemail-message
2:20PM 1 Freepbx database
1:49PM 2 Voice Playback cutting first word or so of audio file
1:12PM 1 Changing or Adding a Line to the Extensions.conf in Asterisk
6:04AM 1 ZAP and line disconnection detection
5:16AM 2 limit concurrent calls on trunk supporting multiple DID
1:50AM 0 web-meetme cbEnd.php not running - error
Wednesday September 16 2009
11:51PM 1 H323 RTP Transmission error of packet
10:18PM 3 call-limit on dahdi channel
9:58PM 1 Connected Line ID for Asterisk 1.4
9:17PM 3 Music on Hold
7:34PM 1 res-crypto dependencies
7:08PM 1 Meetme feature
5:05PM 0 asterisk-users Digest, Vol 62, Issue 44
4:35PM 5 custom voicemail e-mail
3:31PM 1 ACR Anonymous Call Rejection
2:34PM 3 [asterisk-dev] MeetMe in Macro
1:02PM 2 IVR seleCtion
11:46AM 4 G729
3:41AM 2 Reproducible crash - known bug?
Tuesday September 15 2009
10:44PM 1 How to list ongoing calls from dialplan
9:44PM 1 Detecting Transfer
6:41PM 0 intermittent voicemail problem ?
5:35PM 0 Call forwarding, callerID, and e911
4:20PM 0 FW: Blind transfer on Queue-CDR
4:17PM 1 DAHDI hangup detection
11:54AM 0 Sound quality issue
10:31AM 0 (no subject)
7:54AM 3 Which is best provider for G.729
7:49AM 0 Dynamically Move caller to different dial plan
3:31AM 3 dCAP Exam
3:15AM 1 Simple Time of Day Branching problem
Monday September 14 2009
11:22PM 1 The "o" dial option
9:54PM 1 Aastra - Alert-Info : how to stop auto-answer on call second leg ?
7:49PM 0 DAHDI Dial 9 Receiving Setup Acknowledge
5:02PM 0 AstriCon 2009: 30 days, 5 reasons & discount code
10:22AM 0 Something with Dahdi and reversal event...
9:48AM 3 G.729 for Asterisk
8:05AM 0 user=phone
3:35AM 0 Odd sip error
Sunday September 13 2009
4:24PM 0 Blind transfer on Queue-CDR
3:45PM 3 custom ip phone interface
Saturday September 12 2009
3:31PM 1 Queue_logs
3:11PM 1 OT: Question about Wifi sniffing on network
1:54PM 1 Simulscribe/Ditech
8:08AM 1 E65 fails registration, soft phone works
7:03AM 2 zaptel kernel configuration error on vmware
Friday September 11 2009
10:29PM 2 No 64 bit binary for Fax for Asterisk
7:22PM 0 seems dead ?
7:17PM 1 Asterisk & Faxing
5:47PM 0 Aastra 51i and PAP2T behind NAT
4:58PM 1 Hiding voiemailbox/entry from directory
4:14PM 2 Parser for Asterisk Queue Logs
4:05PM 1 MySql and custom CDR
2:36PM 1 Voicemail by email with HTML
12:48PM 0 Asterisk Crash when accessing Directory
11:55AM 0 183 early media
6:58AM 0 asterisk addons don't compile using non standart prefix
6:31AM 1 DIAL IAX2 vs. SIP
Thursday September 10 2009
8:16PM 2 ASR & ACD
7:25PM 1 Operation of ATAs in a call shop type set-up
7:11PM 2 CDR Reporting
6:45PM 1 googlevoice player
6:13PM 1 Friday 11th: Aswath Rao: "Trapezoidal VoIP is Evil" on VoIP Users Conference at Noon EDT
5:21PM 4 Looking for a way to show caller id information on the desktop
2:00PM 1 Help with dialparties.agi
10:33AM 1 RTPAUDIOQOS On DAHDI is it possible
5:49AM 2 Asterisk With Broadvoice
4:35AM 1 g723 to wav conversion
3:04AM 0 A Strange Exception/Notice
2:35AM 1 SPA2102 with Public IP no NAT getting one way audio between Asterisk Phones.
2:32AM 2 How to catch isdn progress message
2:22AM 2 Duplicate DTMF
Wednesday September 9 2009
9:59PM 3 Call Aanalyzer
9:25PM 2 streaming meetme conference
7:34PM 0 Custom CDR Help
6:12PM 1 Blind transfers security
6:11PM 2 Call getting stucked !!
4:31PM 6 how take over an ongoing conversation?
4:10PM 1 MySQL question
3:35PM 0 Hung Lines on GXP2000s?
3:25PM 0 Voice Broadcast Software
3:18PM 0 Query result is array of elements, how to iterate over it ??
2:33PM 1 CLI file convert from alaw to g729 with TC400B transcoding card results to an empty file
2:19PM 0 Call Parking - BLF - AEL2
2:07PM 1 Dial multiple extensions and know who picks up call
1:33PM 1 SIP reply CALL-ID from ITSP has internal address in host part
12:55PM 1 UNIQUEID not the same in Dialplan as passed to AGI
12:42PM 2 asterisk and CCM cisco call manager
6:47AM 2 All the four lights blinking
Tuesday September 8 2009
10:50PM 1 Should digium build a 2FXO / 2FXS 4-port daughter board?
9:51PM 2 Fax For Asterisk and SendFax question
9:33PM 1 Function to query ASTDB families
6:59PM 1 SIP Error
4:27PM 1 Caller ID from POTS lines
4:24PM 2 Realtime static with Asterisk
3:34PM 2 1.2 AGI Deadlock
2:23PM 1 Asterisk remote calls with low bandwith and high latency
2:11PM 2 Manage a E1 system
1:40PM 1 Strange extension state changes in
1:28PM 0 hang up problem while calling
10:14AM 0 Intermittent metallic voice SIP->ISDN ISDN<-SIP
9:22AM 2 CallerID app for Symbian?
7:39AM 3 Asterisk CLI commands not running !!!!!
7:19AM 0 asterisk and link spa942 provisioning
6:56AM 0 Shared Call Appearance - Polycom Phones
5:50AM 1 confBridge in Asterisk doesn't stable
5:29AM 1 Inquiry:Asterisk sound files
Monday September 7 2009
10:16PM 1 Is not yet available ODBC support for queue_log in asterisk 1.6?
9:27PM 1 Older Aastra phones and Asterisk 1.6
6:40PM 2 features.conf : feature map ==> getting feature to work
5:35PM 2 All hints say Hold
5:12PM 2 Aastra phones and Asterisk
5:08PM 1 invalid extension
2:36PM 2 Echo and Playtones not working on SIP after upgrade
2:15PM 0 automatic calls
1:20PM 1 dahdi/DTMF problem
11:48AM 0 Feature request: Meetme and invisible users
10:55AM 3 Help setting IAX variables.
10:14AM 5 TE420P configuration
9:20AM 0 Freepbx database followme disable/enable value
7:45AM 0 Record conversations and place soundfile in user-directory
7:00AM 0 one touch recording not working lately in asterisk
6:40AM 2 The identifier parameter in Dial() command
2:47AM 3 Using asterisk as the recording server
Sunday September 6 2009
11:51PM 0 Preserve userfield on CDR on attended transfer
7:29PM 1 1.6.2-RC1 question
2:33PM 4 Digium hardware support ?
9:45AM 1 running a asterisk -rx command in bash backgroun
4:10AM 1 Chanskype Support
Saturday September 5 2009
6:02PM 1 Asterisk- and Instant Message sending
5:22PM 0 Remote attended transfer
2:58AM 2 Need some help/Suggestions for multiple invites from Asterisk
Friday September 4 2009
9:23PM 3 T1 TE121 cable connected to a TN2464BP AVAYA card
7:45PM 0 Problem with NAT settings?: SIP reply CALL-ID from ITSP has internal address in host part
7:00PM 5 cisco call manager version 6.1.3
6:50PM 1 SUN and PRI ?
5:39PM 1 1.4.26-2, DAHDI-2.2.0, B410P and BRI
5:08PM 2 requirecalltoken and Realtime
4:37PM 1 starfish - pbx
3:52PM 1 Asterisk PBX causes mysql to take more CPU time
3:47PM 0 [Fwd: AST-2009-006: IAX2 Call Number Resource Exhaustion]
1:41PM 0 RFC 3578 in Asterisk
1:27PM 0 Today @12 Noon EDT: Skype for Asterisk, Floor Show at Astricon
12:02PM 2 [ANNOUNCEMENT] Amatix Office 2.0
11:40AM 1 Send 200 OK with SDP instead of 183 with SDP when ringing starts
9:36AM 1 Incremented UniqueId
8:28AM 1 Strange beep when using VoiceMailMain application
6:34AM 1 OT - log rotation [solved]
1:03AM 0 DTMF with duration = 0
Thursday September 3 2009
11:05PM 0 chan_mobile -- bluetooth
10:47PM 1 AST-2009-006: IAX2 Call Number Resource Exhaustion
10:46PM 0 Asterisk 1.2.35,,, and Now Available
10:24PM 1 setvar=CDR(accountcode)=${EXTEN} in sip.conf ???
8:13PM 1 Noises on Batphones
7:35PM 0 transcoder card
3:24PM 1 probleme with web-meetme.3.1.0
2:41PM 1 Originate calls with AMI.
12:56PM 2 How to Disable CDR for callfile?
10:39AM 2 OT - log rotation
9:37AM 1 MeetMe unactive pin access
9:30AM 1 Recommendations about infrastructure to use with Asterisk
9:24AM 3 GTalk functionality Asterisk
8:17AM 0 sql error on trunk qualify....??
7:47AM 1 G.722 problems with IAX
7:01AM 1 passing commands asterisk cli and getting output using PHP AGI
Wednesday September 2 2009
11:18PM 1 Voipbuster not ringing, other SIP peers are ringing...
9:31PM 1 outbound calls not ringing still
8:40PM 1 Payload size of 30ms
8:37PM 2 DISA() fails to recognize dtmf where WaitExten() succeeds (DAHDI-PRI)
8:33PM 0 problem with agi script not getting variable
8:17PM 2 Does L(x:y:z) "Dial" option work on Asterisk version 1.4 ?
8:12PM 0 [UOL - Manutenões Desktop] Controlling call duration ...
6:37PM 0 New Languages: Call for contributions
6:34PM 4 More Echo
5:19PM 4 Allowing multiple callers to join a public speaking session...?
4:32PM 0 followme Script
3:42PM 1 Very simple callback application needed
8:35AM 3 internet connection lagged -> * lagged ...
8:26AM 0 Problem with Cisco 7911G and ABE 2.1.2C - randomly cannot DIAL
8:14AM 1 web meetme PHP undefined variable
7:45AM 1 Skype for Asterisk callfile question
6:27AM 2 Prevent Agent Login from a second extension
3:22AM 1 AMI Originate Commands executed in sequential Order problem
1:53AM 2 DAHDI selective install
12:44AM 2 Configuring Parallel SIP Trunks
Tuesday September 1 2009
10:41PM 2 fails to load : Inappropriate ioctl for device
9:35PM 0 mISDN NT mode config setting
7:33PM 0 MeetMe and dedicated conference room phone
7:22PM 1 espeak app for asterisk 1.6
2:14PM 1 set language in asterisk-1.6.x
1:16PM 7 Dahdi configuraion / error
9:28AM 0 Congratulations to Kamailio - Infoworld Best of Open Source Awards
7:48AM 1 SIP and other phones other then local network
6:17AM 4 jitterbuffer for chan_sip on asterisk 1.2
6:14AM 1 Inquiry:Problem with VoiceMail
4:39AM 4 Inquiry:Problem with Call Parking
2:59AM 1 Digium PRI cards for data usage?
12:14AM 2 1.6.1 + TDM840 FSK MWI problem