asterisk users - Jun 2008

Monday June 30 2008
10:54PM 0 dnsmgr.conf, I do not see Refreshing DNS lookups
10:25PM 3 Centos-5.2 and zaptel-1.4.11 do not get along well
9:46PM 2 Milliwatt-sounding tone recorded over voicemail message
7:51PM 12 Windows Mobile 6 IAX/SIP client?
6:32PM 0 how to have an agi check for dial tone on analog lines before dialing
5:33PM 0 Asterisk Released
5:24PM 4 Voicemail- Recorded Mesage Low Volume
4:15PM 18 sip extension compromised, need help blocking brute force attempts
4:03PM 2 Spam Filter
12:09PM 0 capture call within same callgroup with *8
12:02PM 0 Interesting use of IVR
11:40AM 0 FXS: two rings, then it answer and hangup
7:55AM 2 asterisk and 802.1Q
7:19AM 3 queue welcome message
5:26AM 0 Asterisk to Broadvoice SIP peer fails in 1.6.9-beta9
2:04AM 0 Hangup?
Sunday June 29 2008
10:35PM 1 Timeout between digits for fxs station
5:51PM 0 [VOIP-Users-Conference] Re: A Flood Of Asterisk Appliances
2:32PM 2 indicating call on d channel when no b chan available
12:11PM 0 hint() extension in AEL
11:48AM 1 [FreeBSD 6.3] Why not use safe_asterisk?
9:07AM 0 Druid Open Source Events - Druid Miami Meetup (18 Jul), OSCON (21-25 Jul), Druid London Meetup (22 Jul) & LinuxWorld (4-7 Aug)
7:59AM 0 CTI Intergration with the CRM
7:48AM 1 sendmail file
Saturday June 28 2008
1:15PM 6 Asterisk as an IVR
12:34PM 3 Palyback and CDR records
9:39AM 0 AMI extenstion state
Friday June 27 2008
11:26PM 4 Debug dropped calls
9:20PM 2 FW: Do not update to Firefox 3, yet?
9:08PM 1 Asterisk 1.2 app_vxml
9:03PM 9 measuring network quality in the field
6:31PM 1 Set Language not working!
5:20PM 2 polycom with http/https basic authentication
3:07PM 2 How to pass variable between 2 Asterisk servers over IAX2
2:54PM 4 Asterisk as a component in Jabber network
2:36PM 2 Asterisk's ZRTP patch
2:02PM 3 Do not update to Firefox 3, yet?
1:39PM 1 Maximum number of SIP peers in Asterisk 1.4
1:24PM 1 gxp2000 time.
10:57AM 2 usb - audio asterisk crashes
8:35AM 1 Asterisk cuts off intial voice path on bridging SIP channel
4:08AM 8 DNS Query Overload
1:11AM 6 Asterisk, POTS and plain handsets
12:20AM 2 is it possible? 1 VOIP Provider Multiple registrations <to> multiple inbound contexts
Thursday June 26 2008
10:52PM 0 Cepstral ... Swift... weird result
10:18PM 0 start valgrind and asterisk via init.d script
9:36PM 9 SIP/IAX2 Provider with fallback dialing?
8:51PM 4 queues and MEMBERINTERFACE for AGI script
6:38PM 0 Astricon: Early Bird Special ends next week
5:37PM 0 Console/dsp in 1.4.X
5:24PM 0 Hangup channel
4:17PM 3 Echo Cancelation
3:55PM 1 VoIP Users Conference June 27th @ 12 Noon EDT Scaling and Clustering
3:05PM 0 Error while Compiling zaptel-1.4.11
2:35PM 3 Asterisk With Web meetme
2:10PM 1 chan_zap.c:4747 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1
1:46PM 0 disconnection from caller did not recognized
11:55AM 1 Fw: Outbound video Calls
9:56AM 1 Outbound video Calls
8:17AM 3 where can I found documentation about channel drivers
5:44AM 0 about the Dial application
Wednesday June 25 2008
10:07PM 0 iax2_trunk_queue: Maximum trunk data space exceeded
9:16PM 2 SIP vs. SKINNY
7:59PM 2 Cisco Presence
7:10PM 0 Res: Asterisk with Nextone using H323
6:35PM 0 Cisco 7960 Promiscuous Redirect?
4:40PM 2 Google Apps IMAP
3:54PM 3 included context not being prioritized properly
3:20PM 2 Any SLA alternatives?
2:49PM 10 Number portability in other parts of the world.
12:42PM 3 asterisk seg fault
12:16PM 0 [Fwd: Bridging an existing PBX in with Asterisk]
12:04PM 0 Bridging an existing PBX in with Asterisk
9:21AM 0 misdn issues
7:23AM 2 AS5400 E1 SS7
5:58AM 0 unable to send a fax to a given FAX number
4:56AM 14 Major problem with 1.4.21 asterisk
1:26AM 5 Can asterisk support using different ip for rtp?
Tuesday June 24 2008
5:39PM 5 does asterisk 1.4.20 run on a 486 sx
3:37PM 2 Calls drop + "Didn't get a frame from channel" log message
3:28PM 6 Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!
3:20PM 3 Asterisk with Nextone using H323
1:30PM 16 Chef-secretary scenario
11:44AM 5 Lockups with IAX2 and cdr_odbc in 1.4.21 is it my weird config ?
9:22AM 4 Queue with different music for each caller
8:43AM 0 Can I use X-Lite from local and external ip (when I'm not at home) ?
8:29AM 1 Softphone accepting sip messages
7:32AM 0 GXW4024
6:54AM 5 Loose connection with MySql.
6:46AM 0 Kirk 600v3 Server with sip secret
6:41AM 0 retrieve the status of a sip user using AMI
5:57AM 1 No Codecs and app
12:06AM 6 GotoIfTime Function
Monday June 23 2008
10:02PM 0 Zaptel version on Asterisk website...
5:20PM 14 Centile ipbx, anyone heard of this?
4:54PM 11 Building a Complex IVR
3:57PM 0 new bounty CURL timeout
3:21PM 0 how to restart asterisk after it crashes
12:08PM 7 Controlling cell phone VM / Fax waiting notification icon for asterisk VM
4:19AM 1 Replace music-on-hold on MeetMe with ringing sound
Sunday June 22 2008
4:38PM 3 Telco MWI with Asterisk 1.6-beta9
3:23PM 7 Send cell phone #VM waiting, just like cell carrier
2:21PM 6 SIP over TCP
2:00PM 1 multi-asterisk server implementations
10:47AM 0 Software loop on ZAP trunk - Sangoma
9:25AM 0 (no subject)
9:25AM 1 voicemail didn't send voice message to my email
7:24AM 5 mpg123 problem
Saturday June 21 2008
9:38PM 20 Asterisk GSM Gateway Project
9:32PM 7 1.4.21 + Realtime Queues = Agents Not Ringing?
7:39PM 1 Realtime and OOH323
6:00PM 0 One VOIP Provider Multiple registrations <to> multiple inbound contexts ?
4:11PM 7 DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP
12:30PM 1 iax2 trunk becomes unreachable (asterisk 1.4.21)
11:23AM 1 Fwd: Detection of Answer, hangup, busy etc while using Dial command
11:02AM 3 Continued TAPI Trouble
10:38AM 0 asterisk v1.6 monitor_exec
Friday June 20 2008
9:42PM 5 Recommendations for Motel Instalation.
7:17PM 1 Asterisk and remote phone.
7:13PM 9 Voice only works from one way.
5:47PM 4 Asterisk Openfire Asterisk-IM Plugin Performance Observation
8:15AM 6 FXS port doesn't provide dialtone
7:33AM 3 Asterisk 1.4.21 stalls?
2:13AM 2 Can't make asterisk to test?
Thursday June 19 2008
7:50PM 2 SIP over TCP development in 1.6 branch?
7:23PM 1 Asterisk + zap + sangoma A104D - how to setup call using particular timeslot
7:06PM 4 CLI> show queues NOT WORKING WELL
4:51PM 16 Trouble with PRI config
6:22AM 10 Grandstream Busy Light Fields
12:21AM 2 Mapping multimedia keys: "pressed key not recognized"
Wednesday June 18 2008
10:34PM 3 Adding ;password=foo;method=bar to SIP uri
9:02PM 4 error: conflicting types for ‘bool’
8:34PM 4 Interesting Directory Behaviour (not)
8:32PM 0 RES: GXW 4108 asterisk configuration
7:45PM 12 Website callback
7:43PM 0 Question on T1 OPS
4:53PM 0 T.38 Passthru w/ MediaGateway | Fax <-Analog Line-> ATA <-SIP-> Ast1.4T.38Passthru <-SIP-> MAX TNT <-PRI-> PSTN
4:35PM 0 sending DTMF during PROGRESS
4:21PM 3 [FreeBSD 6.3] Zaptel stops responding
3:41PM 2 zaptel 1.4.11 install
1:25PM 7 Who has the best call recording solution!
1:05PM 1 TRANSFER_CONTEXT ignored?
8:00AM 0 Connect caller and callee after Dial with G
1:08AM 4 Canadian Whitepage Listing Capability
Tuesday June 17 2008
9:56PM 4 Zaptel and OSLEC on Ubuntu 8.04
9:53PM 5 GXW 4108 asterisk configuration
8:26PM 0 suggestions for IAX ATA device or phone in US
6:55PM 0 connectivity with oracle database and astreisk
6:54PM 2 'Together Everywhere'
6:31PM 0 Reg recording of calls
5:55PM 0 asterisk v1.6 queue() continue after answered call
4:45PM 3 Packages for ubuntu
4:04PM 1 Putting incoming sip call leg on MOH while dialing out other party**********NEED HELP************
3:39PM 3 strange SIP-SIP delay
3:15PM 1 looking for help / input with Blind transfer from asterisk to zap
2:54PM 4 Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan
1:23PM 2 Problem with realtime?
11:21AM 0 Audiocodes
9:05AM 1 voicemail problem
5:10AM 3 Reg call recording
4:03AM 29 need ata suggestion
12:24AM 10 Call Center
Monday June 16 2008
7:01PM 1 FW: Request to mailing list asterisk-users rejected
1:44PM 1 Euro_isdn PRI Line, callerid and usecallingpres"
12:42PM 0 Astribank and Celular Interface Module
11:33AM 5 Help! - Double NAT issue
10:39AM 2 Transfers with TE12xp
9:30AM 1 Agents getting "stuck" busy
5:03AM 1 asterisk was discunnected suddenly
4:05AM 0 Asterisk Manager Telnet Times Out?
3:18AM 0 Problem connecting to another server, Failed to authenticate on INVITE
Sunday June 15 2008
4:04PM 25 OT How Digium Saved My Bacon!
3:35PM 11 *OT* DLI Ethernet Power Controller $289 (I paid $200 for a two port "webswitch")
2:06PM 17 Please Advice on Best High traffic fxo gateway/cards
1:34AM 1 [asterisk-dev] Astricon question: four or five tracks?
Saturday June 14 2008
6:32PM 1 World Most Economical Predictive Dialer!
9:28AM 8 How to append "#" to the number before sending to Zap
7:50AM 2 play sound on a specific channel
6:07AM 0 cpu and ram requirements
12:08AM 13 Idiot's question
Friday June 13 2008
9:43PM 0 strange iax authentication behavior
7:23PM 18 cdr-custom/Master.csv rotation
7:08PM 2 AEL Help
6:31PM 2 Need a SIP trunk provider for US - Dallas/TX
5:44PM 0 TCP & UDP path not the same
5:26PM 0 how to make a ip to ip call
4:52PM 2 PRI crashing Asterisk
1:29PM 11 start n run an agi script on hangup
10:16AM 2 Behind NAT: source is fring software (SIP)
5:18AM 53 World Cheapest Predictive Dialer!
12:45AM 0 funny search engine terms
Thursday June 12 2008
9:37PM 4 Astricon question: four or five tracks?
8:36PM 0 DUNDi question
8:35PM 1 Really destroying SIP dialog
8:32PM 4 Odd Polycom Reboot Issue
5:58PM 0 problems getting dialed information on asterisk
5:58PM 0 Phone selective variable setting?
5:41PM 0 Asterisk Unified communication features
5:28PM 0 [asterisk-biz] New faxing protocol. Good/Bad ?
4:50PM 0 Asterisk 1.4.21 Released
4:49PM 0 On Hold "Context"?
4:41PM 0 custom functions is voicemail
3:55PM 0 Fwd: Complimentary Subscription to VoIP Industry Publication
3:16PM 10 Using Asterisk Only as Voice Recording Solution.
2:48PM 0 iax2 qualify problem - PONG ignored
12:36PM 5 multiple CDRs for one call (multiple dial attempts during one call)
11:41AM 3 AGI after Hangup
10:25AM 3 Dial command and its g option
9:09AM 0 Friday the 13th lucky asterisk appliance day
8:43AM 3 Dial Command Option D Early Bridged
8:23AM 5 IAX2 phones, BRI and Analogue cards
8:14AM 3 Monitoring QoS
4:16AM 1 g729 codec for asterisk-1.6.0?
1:47AM 19 aSTERISK / Vicidial systems over 4MB fiber
1:46AM 1 Echo on PRI even with H/W echo cancel
1:31AM 1 ?
1:23AM 4 Asterisk on SLOW solid state disk
Wednesday June 11 2008
10:52PM 14 time on asterisk
10:11PM 4 Asterisk and XMPP (Jabber) : testing new application JabberReceive
9:29PM 7 asterisk calls per second
6:21PM 5 Losing CDR(accountcode)
3:20PM 5 Asterisk Data Calls
1:43PM 0 use of AJAM wth high load
1:38PM 7 decrease the time it takes for asterisk (fxsks) to answer
1:17PM 22 SIP call, updated with CID as it becomes available
12:00PM 2 Zaptel timer failure
11:39AM 2 SIP channel lock issues
10:23AM 2 hang -- extra info + gdb hangs
7:40AM 13 How to turn on the H323 logging on Asterisk
5:30AM 3 TE110P with 40,000 IRQ missess
1:17AM 2 Sound files custom path
Tuesday June 10 2008
9:28PM 1 Delaying SIP disconnect after incoming call hangs up?
9:22PM 4 Problems configuring a PRI...
9:16PM 0 Seeking Collaboration in Development and Validation of an Anomaly Detection System for Asterisk
8:00PM 3 zaptel issue
7:48PM 1 Zaptel config
7:37PM 0 Blind transfers and ringback tone
5:40PM 6 Weird one way Audio situation
5:03PM 7 SayNumber while reading DTMF?
2:34PM 2 Camp / Callback feature in 1.4
2:11PM 0 Debugging SIP call hangup reasons
10:16AM 1 meetme recording with security?
6:13AM 14 g729 open source codec and sample size
1:01AM 5 Asterisk : using setvar with IP Realtime and variable inheritance
Monday June 9 2008
10:51PM 18 Interoffice phone setup
8:46PM 0 [OFFTOPIC][SPANISH] Creando una comunidad de asterisk en español
8:36PM 1 RFC2833 DTMF -- with an RTP debug log -- need some analysis/interpretation
7:54PM 1 Call hold in dialplan
7:30PM 1 Long call setup with non-PRI T1
7:29PM 5 Polycom SIP and DHCP problem
7:06PM 1 3g video call using h324m_loopback not connecting
7:01PM 1 redfone fonebridge2
5:48PM 0 fring and g729
5:36PM 4 Asterisk Installation with Radius Support
4:01PM 1 SIP over M$ ISA
12:45PM 0 Asterisk 1.4.21-rc2 Now Available
12:29PM 2 Remote-Party-ID and selective CLI withold
5:26AM 5 Asterisk 1.2.28 + Realtime Queues - Thinks Queue is empty
Sunday June 8 2008
9:10PM 2 Diverted Call Information on PRI
8:42PM 1 Asterisk can handle only 200 to 300 SIP device registrations
2:13PM 0 How to set name of call wav recording file in outgoing/call file?
4:16AM 9 MeetMe Limits
Saturday June 7 2008
10:37PM 2 Manager Originate CDR problem
3:37PM 16 Fax on FXS
1:03AM 5 Logitech DiNovo Mini keyboard with myth
Friday June 6 2008
9:33PM 10 features.conf not working
9:15PM 0 Anyone using zaptel analogue hardware in Singapore?
7:15PM 4 MiixMonitor filename for queue calls.
7:13PM 2 SIP call recording
5:37PM 6 Asterisk not picking up incoming calls from TDM400P
5:23PM 3 Bad ringback tone on zap channel
4:23PM 1 Zap channels state
2:35PM 0 MixMonitor Not recording whole calls
1:57PM 1 Block on hold
1:52PM 0 Reminder TODAY Friday June 6th at 12 Noon EDT VoIP Users Conference
1:34PM 4 bad call quality
1:24PM 1 Disable sending CNAM over facility for 2bct
1:18PM 0 Asterisk and TDD
12:20PM 1 Sending texts questions.
12:01PM 5 hang -- three times in 1.5 days (TC400B at fault ?)
4:43AM 5 fxotune vs rxgain/txgain
12:02AM 0 SIP Phone, multiple line, but one call at a time ?
Thursday June 5 2008
10:24PM 9 Asterisk video alternatives
10:08PM 7 PoE budget
10:01PM 4 Similar extension numbers for multiple users
9:08PM 0 RECALL: Lithium batteries for Polycom Soundstation 2W
4:25PM 0 Asterisk -> Nortel CS1K via NRS
3:57PM 1 detecting which party hung up
1:58PM 5 Default ringtone
9:31AM 0 About H323 configuration on Asterix
7:56AM 1 remote server with Snom 190
6:55AM 5 handling SIP trunk with limited concurent calls
3:02AM 18 fxotune question
Wednesday June 4 2008
11:36PM 0 Codec troubles
11:01PM 1 = sorry no "s"
10:52PM 2 Browser based VoIP client? None of them are very full featured
10:46PM 0 Browser based VoIP client? -
10:24PM 3 Avaya IP Phones with *
10:18PM 0 AST-2008-009: (Corrected subject) Remote crash vulnerability in ooh323 channel driver
10:07PM 0 Patch for app_asr.c: DTMF instead of goto
10:04PM 0 Asterisk-Addons 1.2.9 and 1.4.7 released; Asterisk-Addons 1.6.0-beta4 now available
10:03PM 0 AST-2008-009: AST-2008-007 Cryptographic keys generated by OpenSSL on Debian-based systems compromised
7:34PM 3 Lumenvox - Gentoo
7:04PM 2 queue delay between calls to agents
6:51PM 0 disable "send reply" in asterisk voicemail
6:42PM 9 Browser based VoIP client?
3:52PM 24 Asterisk 1.6 vs 1.4?
3:14PM 2 G.722?
2:55PM 2 init.d script no longer uses safe_asterisk
1:51PM 0 Asterisk problems
12:42PM 0 911 via MAX TNT
11:30AM 0 This AEL vs. Dialplan thing ...
10:43AM 2 connecting 2 FXS together
10:30AM 0 busydetect=yes, busycount=5: hangup automtically without reason, why?
8:47AM 1 Error Wile starting AsterFax
6:36AM 0 How to improve group call User Interface ?
4:47AM 0 SPA 3102 disconnect tone setting for China
Tuesday June 3 2008
11:53PM 3 What does reason 8 for failure means in Manager
11:43PM 16 911 via MAX TNT ??
10:56PM 8 handling jabber status
9:36PM 0 Videos of online
9:22PM 0 Asterisk 1.4.21-rc1 Now Available
8:05PM 0 Asterisk 1.2.29 Released
7:53PM 0 AST-2008-008: Remote Crash Vulnerability in SIP channel driver when run in pedantic mode
7:37PM 2 #include
6:54PM 13 Queue is sending calls to Agents even when they are in use
6:41PM 0 unload zaptel if asterisk is not running (or crashes)
4:46PM 2 Problem with several includes in ARA
4:43PM 3 Asterisk Seg faulting.... No core dump.
3:37PM 9 Trouble with Polycom phones
3:12PM 9 Asterisk with bad gsm file playback
2:52PM 0 Asterisk and Realtime
1:41PM 1 setting channel variable in DeadAGI script
1:21PM 28 Any reason to *not* use AEL? (Also, MixMonitor q)
12:22PM 0 externip not setting extern ip/ loop detected issues
9:50AM 1 G.722 over ISDN PRI/BRI
7:57AM 0 artificial
7:55AM 2 Media time out for SIP and IAX Trunk
5:37AM 0 Final call for Astricon talk topics
1:05AM 4 Date in Dialplan
Monday June 2 2008
11:44PM 4 What do I Copy to Migrate to New Machine?
10:11PM 0 For all your office needs... and voip-greylisting
9:27PM 0 queues with callback members
9:04PM 1 PBX Functionality for Less than the Price of a Key System (3Com Asterisk IP Telephony Appliance)
8:24PM 7 Source for Digium TDM400P card
8:15PM 5 ast_compile_ael2: Warning: file /etc/asterisk/extensions.ael, line 932-932: Empty Extension!
6:07PM 1 Why doesn't Pickup() work??
4:10PM 4 QSIG transfer of calls away from Asterisk?
1:32PM 0 Voicemail question
1:27PM 0 Good Asterisk HA Load balancing document.
1:20PM 0 looking for Asterisk smarties in Bangalore, India.
12:09PM 4 passing a call to the call queue
11:25AM 13 Help with E1s
9:55AM 1 Query about AsterFax Installation
9:45AM 0 OT - Intel IPP 5.3
8:25AM 6 RTCP debugs and stats
Sunday June 1 2008
10:37PM 6 Trouble with unicall
9:54PM 7 New faxing protocol. Good/Bad ?
9:00PM 9 How to limit extension to allow only internal calls
5:26PM 6 Mysql and extensions.conf
12:13AM 2 Asterisk and V.22bis