asterisk users - Aug 2009

Monday August 31 2009
11:21PM 11 queue issue
8:05PM 5 Upgrading Asterisk in Trixbox installation
7:01PM 2 List Access
5:11PM 27 Asterisk MWI issue
4:37PM 4 Versions of Asterisk 1.6
3:33PM 3 Strange problem
1:26PM 5 How to stop IVR once system receives DTMF?
12:52PM 0 Clarifying RX and TX gains
12:41PM 0 asterisk-users Digest, Vol 61, Issue 85
11:01AM 0 Asterisk 1.4 and GUI Configuration Help
9:53AM 2 Asterisk Regular expression to validate any phonenumber
6:36AM 2 SIPP how can we give delays between 2 calls
5:24AM 14 Inquiry:How to hide Caller Id
5:17AM 5 CDR to Postgres Centos
2:45AM 7 How to deal with PayPal frauds?
2:33AM 16 Asterisk Web Meetme module not loading
12:54AM 2 Question of resiliance
Sunday August 30 2009
6:54PM 1 I find this incomprehensible ?!
1:30PM 10 Need help - CDR MySQL
12:11PM 2 MySQL syntax error : I really don't see where...
9:57AM 3 Asterisk/app_rpt and bandwidth
5:52AM 3 Help me testing this webphone at
12:50AM 1 unable to execute command from
Saturday August 29 2009
7:19PM 2 GoToIfTime : how to define sep 25th till oct 10th ?
5:18PM 0 asterisk-users Digest, Vol 61, Issue 84
4:01PM 2 Asterisk and Now Available
12:29PM 5 cannot run agi scripts
6:59AM 2 Flite module for asterisk 1.6.x
6:16AM 0 Training failed with hylafax
Friday August 28 2009
11:46PM 2 Error loading module ''
9:24PM 0 realtime voicemail and imap user
8:41PM 5 Report
6:28PM 0 chan_mobile on macosx ppc
5:35PM 1 Help needed with getting a maxed-out Asterisk to gracefully deny calls.
4:14PM 0 2 Asterisk boxes via IAX : always calling as IAX guest
3:21PM 6 QuickPhones QA-342 and SIP/RTP Flash Event for Transferring
2:52PM 5 Crystal Recording Interface
1:50PM 2 PRI worked fine for months, nowit stopps working after 2-3 hours
1:02PM 2 Help with call scenario
11:09AM 1 Zap / dahdi errors
9:25AM 6 application order when you make a call
Thursday August 27 2009
10:57PM 0 Fotos 18/08 .
8:47PM 0 Universal Services Fund taxes now apply to VoIP end-users.
7:39PM 0 transfer two gsm mobile calls
6:44PM 6 how does "wrapuptime" work in queue.conf
6:30PM 1 Bad Gateway
6:23PM 6 Selective canreinvite in multi-tenant environment
5:52PM 4 Sticky Park
5:32PM 0 password length of sip peer
3:14PM 7 POTS supervision with DAHDI in 1.4 releases
1:08PM 0 How to set call record file name
12:39PM 2 Problems using chan_sebi and Huawei E169G
10:57AM 0 create applicationmap and use it in dialplan
9:40AM 1 Documentation on RSA key authentication ?? (No way to send secret to peer)
9:24AM 6 Measuring voice quality with Asterisk
7:01AM 3 asterisk with realtime DB , issue with MWI
5:34AM 3 Digium Echo cancellation.
Wednesday August 26 2009
9:29PM 1 2 Asterisk boxes : 1 can see the other, not vica versa
9:21PM 0 Company in Los Angeles looking for Asterisk & Network Administration/Maintenance Engineer
7:21PM 0 408 error
6:03PM 4 Fw: app_swift issue
3:48PM 2 TE4XXP: Version Synchronization Error!
3:07PM 0 Swift application and DTMF
2:07PM 16 Multiple user registration ...
2:03PM 2 application missed in asterisk 1.6.1 - SetCallerID()
12:36PM 0 Header from REFFER
12:05PM 0 looking for user:Zoa for T38...
11:42AM 1 Bria / eyebeam: no RTCP while on hold
6:53AM 12 PRI worked fine for months, now it stopps working after 2-3 hours
6:48AM 1 Open Source Visual Call-Flow and IVR Dev Tool v1.0 Released!
6:33AM 1 app_swift issue
5:20AM 3 ACD, call barge, recording
3:07AM 0 Timeout func ignored if inside a macro and when Dial cmd has limit (L). Bug ?
2:39AM 3 User-invoked call restrictions
1:07AM 1 netfilter conntrack mangling canreinvite?
Tuesday August 25 2009
8:43PM 5 Echo
8:14PM 1 How to detect if the call is being answered by Voice Mail?
7:47PM 3 Realtime with "rtcachefriends=no" problems...
3:34PM 2 Linuz from asterisk
2:28PM 3 followme app
1:47PM 0 DTMF duplicated when Waitexten
12:59PM 21 Breaking news, but what happened? 11.000 channels on one server
10:54AM 2 Authenticating SIP peer on IP address only
10:39AM 0 Patton ISDN 2e boxes as media converters...
9:28AM 0 No translator path exists for channel type dahdi
8:54AM 7 Set up SER as a SIP extension on asterisk server
6:13AM 0 Stop Originate
Monday August 24 2009
7:52PM 2 E1 w/ TE420B EC
7:51PM 1 Bottlenecks with my asterisk setup.
4:12PM 9 install the digium card TC400P howto?
4:09PM 0 Autodial not waiting for voicemail
2:11PM 2 asterisk + Asterisk GUI v2.0
1:55PM 20 Problems sending voicemail emails
1:21PM 1 Show queue-name near the callerId
12:44PM 0 SIP doesn't recognize hangup
11:49AM 3 Follow me IVR sounds
11:48AM 5 Need to now my "Asterisk User ID"
10:17AM 1 problem on compiling asterisk-addons-
10:13AM 0 exchanging CDR data between Asterisk servers
8:51AM 1 Request Pending retransmitions
6:27AM 12 Dailing any number PSTN or MObile number
5:04AM 1 disconnection silent channels
Sunday August 23 2009
11:34PM 2,,
Saturday August 22 2009
2:29PM 7 Patton smartnode 463x (BRI) 25ms tail echo cancellation
1:23AM 0 Interview with John Todd
Friday August 21 2009
9:40PM 0 Valet Park with Hint - Button Support
8:22PM 6 how to install asterisk
8:00PM 1 Queue Question
2:16PM 2 Incoming caller presentation doesn't work - out of ideas
12:39PM 3 stutter playback
12:27PM 7 Asterisk 1.6 t 38 passthrough
9:27AM 4 codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory
2:17AM 5 Core dump gets created while accessing voicemail
12:18AM 1 problem with asterisk hylafax and sangoma A200D
Thursday August 20 2009
6:51PM 3 2 single span TDM cards in asterisk
6:45PM 1 Asterisk - Monitor / MixMonitor Recording
6:06PM 0 thanks!
6:05PM 0 Friday Aug 21st @ 12 noon EDT: VoIP Users Conference, More Wideband Madness
5:57PM 19 IPKall and FWD
5:14PM 0 Nokia E90 video via Asterisk
4:22PM 1 multiple call dialing and playback an message
3:47PM 14 Cannot play soundfile, doesnt find it or wrong format? Weird, worked yesterday! :-)
3:12PM 2 Sip Tunneling
3:06PM 1 Pause/Unpause agent based on devstate
2:54PM 4 HUD display?
1:35PM 1 Post recording command to be executed after the end of recording
12:41PM 12 Save directly to database , and play directly from database
11:54AM 21 mysql sip realtime
11:38AM 1 Call routing between two Asterisk boxes using SIP not working ...
7:49AM 0 asterisk followme feature code
6:00AM 4 LDAP Get for Asterisk 1.6.x
3:45AM 1 Create VoiceMenu SNAFU
Wednesday August 19 2009
11:20PM 7 PRI Connected to definity errors
8:25PM 0 AsteriskGUI Create VoiceMenu SNAFU
7:33PM 1 Dial plan sample for detecting Voice Mail
3:56PM 17 * 1.4 -> 1.6, zaptel -> dahdi
1:59PM 6 mysql error (err 2002)
1:54PM 8 outbound calls not ringing
1:08PM 0 Newbie: Mac OS X - Asterisk Gui 2.0 (svn) loops at " Verifying Dialplan Contexts needed for GUI"
11:33AM 1 Individual PIN Code per Extension
10:54AM 12 CAP_FOWNER=ep for asterisk
9:22AM 0 CDR record for call originated from manager
9:22AM 0 ISDN Calling Sub Address and Called Sub Address for the branches
7:35AM 5 Newbie: How to copy track from CD for MOH without getting "Junk at beginning of frame ..."
7:16AM 1 MEETME how to lock the conference if no admin are connected
3:45AM 4 Multi operator platform Asterisk {manage}
2:02AM 0 AsteriskGui 2.0.4 and TDM400P
Tuesday August 18 2009
11:43PM 0 Moderator access to meetme allowed despite pin
9:55PM 1 Cisco IAD's
9:49PM 0 How to send Caller ID Extension to Trunk?
9:44PM 4 Asterisk project changes Music-On-Hold provider
9:30PM 2 Monitor-join not joining files in the queues.conf file
8:54PM 0 Date/time in queue_log
8:10PM 0 Asterisk Realtime : use different family names family => mysql, database, table
5:52PM 7 Platform decision ...
4:09PM 7 DAHDI - better to have card?
3:38PM 1 Play Fake ring in phpagi
3:25PM 3 IAX2 ActiveX Control
2:55PM 1 Get SS7 Hangup Code as Asterisk variable.
2:43PM 0 res_ldap.conf
1:25PM 7 Channels don't go away with soft hangup
1:25PM 1 avoid indicate condition 9 and starting music on hold
12:50PM 19 Zaptel -> DAHDI: now echo
12:29PM 0 Paging with Pickup
12:29PM 3 Speech Recg and TTS
12:07PM 3 You do not appear to have the sources for the 2.6.20-prep kernel installed
8:33AM 0 Asterisk + realtime applications
7:28AM 2 Execute some kind of script when something happens with Asterisk
7:08AM 9 OT - DECT handset with Line key
7:03AM 4 Call variables(dialstatus?)
3:45AM 57 Skype for Asterisk???
Monday August 17 2009
10:06PM 0 SipDroid
6:42PM 0 Echo on TE121B with hardware echo module
6:23PM 0 Call back DIALSTATUS is empty
6:12PM 6 Accessing to through Asterisk
5:24PM 3 Accessing Asterisk gosub arguments in extensions.lua
4:58PM 4 Goto mask
4:18PM 1 - Is Asterisk Zaptel Compatible? -
2:44PM 3 Same number for each caller, but should reach different zap-channels, how?
2:44PM 4 queue_log in mysql and file
2:10PM 2 Problems with pstn cards
12:21PM 1 /usr/bin/ld: cannot find -lpq
12:59AM 4 Newbie: How to find the serial number of Digium card?
Sunday August 16 2009
1:48AM 1 Detecting Called party Ring indication (and act on it)
Saturday August 15 2009
12:56PM 11 BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]
12:30PM 1 OT - Using new Asterisk calendars features
2:20AM 3 bare minimum /etc/asterisk for sip based *
Friday August 14 2009
8:27PM 1 Number of Phone Numbers per Outgoing CALL File
6:02PM 1 Meaning of " requested special control 20, passing it to SIP"
5:35PM 2 Complete neutral Spanish sounds
4:46PM 6 Vicidial now extension setup
3:52PM 3 chan_dahdi refuses to build
2:58PM 5 i have a error in ivr
2:18PM 6 no ring tone
2:11PM 0 Call no reject when receive 'PROGRESS with cause code 27 received' in zap channel
2:04PM 16 CURL function with SSL
2:01PM 3 Cdr error? Help!
10:20AM 3 onnecting two asterisk using B410p BRI cards
9:39AM 7 play prompt after hanup
8:19AM 0 Astricon 2009 - dCAP
7:01AM 0 CPU Spikes in asterisk connected via IAX trunk
3:22AM 1 Stale auth messages
Thursday August 13 2009
8:43PM 13 Time of Day Routing
7:53PM 0 Looking for recommendations - US SIP provider for T.38 Faxing
7:21PM 2 RealTime in dialplan - proper way?
5:42PM 3 outbound mail slow?
4:44PM 1 Autofallthrough delays before hanging up calling channel?
2:35PM 1 Help for Alcatel asterisk
2:21PM 0 Cdr error??
2:10PM 4 Snom Phones Registration/Failover Feature
1:48PM 4 Areski CDR + Mysql + asterisk 1.6
11:19AM 0 39 Free Softphones
10:25AM 4 PRI Gateway - Worth it?
9:58AM 0 asterisk conference error/bug?
8:55AM 0 Conferencing and web front-end
8:33AM 2 Database Access from dialplan.
6:51AM 2 "Channels stuck up even without use"
Wednesday August 12 2009
7:33PM 4 Creating an IAX/SIP-to-ISDN PRI gateway
5:50PM 4 Cdr src field fail??
4:21PM 1 Why do CDR dstchannel have a strange number after them? IAX2/XXXX-????
3:49PM 0 meetme conference hangs in silence after dialing
3:03PM 10 call drops after a few seconds
1:30PM 10 Asterisk + CDRTool
1:25PM 0 maximum dahdi tdm concurent calls and max iax trunk calls
5:28AM 21 Twitter is Suing me!!!
4:33AM 5 Cisco 79XX, SIP and Asterisk
3:22AM 1 undefinied symbol: global_app_buf
12:05AM 0 Different From and contact header
Tuesday August 11 2009
8:57PM 3 SIP app for iPhone that works well with Asterisk?
4:33PM 6 func_odbc insert with mssql
4:15PM 10 Cisco 1760 Multiline phone
4:11PM 1 Unable to compile
3:20PM 2 MixMonitor and Transcoding..
2:16PM 0 FSK UK Problems
2:00PM 0 AST-2009-005: Remote Crash Vulnerability in SIP channel driver
1:59PM 0 Asterisk 1.2.34,,, and release announcement
1:54PM 0 asterisk 1.4 segfaults when trying to use mixmonitor
1:22PM 0 chan_iax2.c:1219 __send_lagrq mesages
11:03AM 1 testing music
10:00AM 0 Xfer extension to extension call, flash hookpass through by Asterisk needed via quintum and X-lite/Eyebeam
4:02AM 0 waitfordialtone patch
3:52AM 0 How to adjust the timeout to send CANCEL?
2:37AM 1 sflphone questions
Monday August 10 2009
4:51PM 4 SNOM 870
4:30PM 1 7940g
1:34PM 0 Transfer after pickup
10:55AM 12 "context" does not work
8:36AM 0 Issues with sound quality and HDLC
6:16AM 0 X100P FXO PCI card not receiving calls
Sunday August 9 2009
6:11PM 0 Killing the Server
2:11AM 1 queue need very long can start music
Saturday August 8 2009
7:08PM 0 Second voice caller notification
7:06PM 3 bad(?) TDM400 card?
4:16PM 0 DeadAgi application not exiting
9:14AM 4 Question: How to contribute to Asterisk-addons
6:32AM 6 A problem with recoding agents calls via monitor
3:35AM 2 30 Great free Asterisk applications
Friday August 7 2009
8:07PM 5 Placing a SIP Call on Hold
7:08PM 2 realtime config and extensions.conf
5:48PM 4 caller id problem
4:04PM 4 Going to VM after 180 seconds in queue
3:47PM 8 Asterisk in VMWare, how does it perform and what is the limit?
3:25PM 9 Anyone had any luck with SIP clients on theiPhoneplatform?
3:18PM 4 regcontext regexten
2:46PM 1 Anyone had any luck with SIP clients on the iPhoneplatform?
2:01PM 2 Linksys SPA922
11:05AM 0 asterisk crashes!!!
9:43AM 0 A problem with monitoring calls
8:44AM 5 Host-ID.
6:18AM 0 iax2_read: I should never be called - issue 8286
4:43AM 7 Anyone had any luck with SIP clients on the iPhone platform?
2:46AM 0 Friday Aug 7th @12 Noon EDT Mobile VoIP
Thursday August 6 2009
8:56PM 6 OT -
7:31PM 0 Extensions patterns algorithm
6:40PM 5 No audio on remote SIP calls
5:57PM 2 Inbound Call coding
3:54PM 2 ForkCDR and setting the account info?
3:32PM 7 Setting up Outgoing Trunk
2:28PM 3 Monitoring Asterisk uptime
2:21PM 4 Set PHP binary location for AGI
12:24PM 4 Asterisk dont detects hangup by phone
10:33AM 1 Can't delete voicemail messages
10:33AM 0 chan_mobile handle 92 log flood
10:27AM 1 Asterisk 1.2 -> 1.4 CDR change?
10:07AM 0 ntop and Asterisk
10:02AM 8 E1 line simulation for Asterisk
7:15AM 0 Regarding XRMS support
Wednesday August 5 2009
7:36PM 0 Asterisk with gizmo5 and google voice only takes one call at a time.
7:17PM 13 Best ISDN BRI solutions?
6:12PM 7 Several mailboxes on SIP peer
2:25PM 2 [asterisk]q: asterisk 1.6.1 install
12:32PM 3 sip.conf parameter and sip msg between server <-> client
12:23PM 0 Asterisk & sending an sms
7:16AM 2 original & reformat extension
2:23AM 4 Gizmo Dial Out No CALLED PARTY AUDIO??
Tuesday August 4 2009
9:35PM 3 ChangeLog revision question
9:29PM 0 chan_mobile future
9:13PM 5 Calling issue for non-extension numbers
8:50PM 1 MWI
6:07PM 0 Dahadi - Grouping Issues
5:51PM 2 Transfer Issue with IAX Trunk
5:32PM 6 Message Waiting Indicator on DAHDI line
3:44PM 0 Need case study for Open Source presentation
2:36PM 10 Anyone actively using RLT for mobile phone forwarding?
1:25PM 1 question on nortel CS 1000 PBX and PRI connection to built in PA system
11:32AM 0 dahdi_pri_error No more room in scheduler
10:34AM 6 CDR Problem - No CDRs when call is not bridged
10:11AM 0 SIP server behind NAT
9:54AM 4 how to implement CLONED LINE Feature in asterisk?
9:10AM 3 setting verbosity for asterisk cli..
8:40AM 1 dahdi_scan doesn't recognize an OpenVox A400E
12:05AM 3 undefined symbol: ast_speech_register
Monday August 3 2009
9:45PM 0 dahdi_dummy soft lockup in dahdi-linux-
8:29PM 9 Difference between 1.4.x and 1.6.x?
5:04PM 7 Upgrading from to
3:55PM 2 ami
3:29PM 1 PMR446 interface
2:08PM 4 single port voip gateways
12:45PM 1 New Digium TE207P for sale
12:20PM 2 Outbound calls drop after 15 to 30 seconds.
4:52AM 3 User Authentication in sip.conf
4:30AM 0 AST-2009-004: Remote Crash Vulnerability in RTP stack
3:50AM 4 Asterisk,,, and Release Announcement
Sunday August 2 2009
10:12PM 1 AstLinux 0.6.7 released
7:30PM 4 T.38 and reinvite
6:54PM 11 Modem
2:30PM 3 Converting sound files
12:01AM 1 Broadvoice versus Asterisk and 1.4.26
Saturday August 1 2009
7:32PM 4 SNOM Phones Displays NR Frequently
7:02PM 10 Different codecs for reading and writing
1:06PM 1 H248 support
10:46AM 2 Maximum number of concurrent calls
8:54AM 0 Aug 1 & 16- Global VOIP Free SW HW Culture meeting, BerkeleyTIP, For Forwarding
7:32AM 4 Dialplan strategy suggestions needed
7:22AM 1 how to setup incoming calls not to use authentication
5:42AM 2 Inquiry : Asterisk hash key
5:19AM 0 Inquiry:Asterisk supporting hash (#) key