asterisk users - May 2007

Thursday May 31 2007
TimeRepliesSubject
11:09PM 0 Urgent-- Error while installing app_dtmftotext.
10:29PM 6 Auto Dial Problem
9:39PM 1 Compilation after Source code changes in Asterisk
8:37PM 1 CARD FOR inband signal
7:33PM 0 FreePbx/asterisk/openser
3:06PM 0 Custom CLI (spoofing) ?
1:44PM 2 CDR timing
1:39PM 0 Chan_sip max channels limit?
1:21PM 10 High Port Count ATA
12:45PM 0 Track Agent login/logoff status
12:11PM 2 asterisk auto dial does not wait for answer
11:42AM 3 applicationmap on features
10:14AM 10 RF to IP bridge
9:32AM 0 HP OfficeJet 6110, Sipura 2102, T.38, and Clarent
9:26AM 9 Net2Phone Multiple SIP Trunk Not Working
9:19AM 0 Asterisk Release Maintenance News
9:18AM 2 Duplicate UNIQUEID on CDR
9:15AM 3 linksys pap2 version2 ata DTMF issue
8:29AM 4 moh backround?
8:25AM 0 Meetme context.
8:03AM 35 click to call
7:17AM 1 Who are XO and L3?
5:58AM 0 Asterisk Users Conference for Friday June 1st 2007 @ 12:30 PM EDT
5:44AM 3 'asterisk' shown on display
4:35AM 6 Context documentation for the newbie!
3:59AM 1 ringback detection
2:53AM 4 Passing call duration to an AGI Script
1:37AM 1 Diva and Asterisk
12:58AM 7 How to read SIP debug?
12:10AM 1 Subscriptions to Agent hints stopped working after a changing the numbering
 
Wednesday May 30 2007
TimeRepliesSubject
7:32PM 2 Application Developer
6:09PM 0 montavista and Asterisk
5:17PM 1 Delays on E1 Delivered via SHDSL
12:00PM 3 (no subject)
11:46AM 1 FW: Help with IAX
9:54AM 7 Dial plan inquiry using GotoIf()
8:37AM 8 Help with IAX
8:32AM 7 any codec passthru mode
5:35AM 1 Asternic Flash panel
4:58AM 0 Call transfer while dialing
4:19AM 0 SIP SendURL
3:18AM 0 *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
3:03AM 18 False ring problem
2:47AM 1 fax2mail ann missing CallerID number
2:15AM 9 multiple host= in sip.conf
2:06AM 0 Configuring Asterisk as Gateway SIP-H.323 via ooh323
2:02AM 0 Allow for context includes in realtime (ARA)
 
Tuesday May 29 2007
TimeRepliesSubject
10:08PM 0 Play sound file on keypress (bridged call)
3:23PM 11 Problem on incoming call from Zap channel to SIP phones...
2:36PM 4 channel_find_locked: Avoided deadlock
2:08PM 5 SBC
1:24PM 15 *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
12:13PM 0 Sending a SIP INVITE without SDP from Asterisk
11:19AM 0 Theoretical and Received SIP addresses causing no audio
10:23AM 3 Asterisk as a call recorder for ISDN30 ?
8:56AM 1 Monitor application inestability and high load
8:53AM 2 Agents.conf from realtime static
6:49AM 0 SIP OPTIONS triggering some action in case of no reply
6:38AM 27 *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
5:55AM 5 OpenVox A400P01on thin client?
5:22AM 1 disable musiconhold
2:38AM 0 Billion on Debian Etch
2:19AM 0 * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
2:11AM 4 Zaptel linux26
12:11AM 5 zaptel module dependences
 
Monday May 28 2007
TimeRepliesSubject
7:52PM 7 Alcatel - Asterisk setup
7:45PM 14 ekiga register problems
7:38PM 3 [1.2.18] Wrong steps in extensions.conf?
6:07PM 0 Multiple TDM400p cards in one machine -- nolonger an issue?
2:09PM 4 Language in Zaptel.conf
1:40PM 0 Send parked call extension to set
12:55PM 3 Polycom Static IP
11:12AM 1 Asterisk and cell phones
10:30AM 1 Octasic echo cancellation
10:10AM 5 Multiple TDM400p cards in one machine -- no longer an issue?
9:57AM 1 Astmanproxy
7:18AM 15 Blindside Web Conferencing
6:09AM 2 help on asterisk sipp
5:27AM 0 Limit outgoing call for sip peer
3:29AM 0 include=>context in REALTIME!!!!!!
3:17AM 1 Queues with announce
2:42AM 1 Meet me
1:56AM 0 Progress passing problem.
 
Sunday May 27 2007
TimeRepliesSubject
8:13PM 0 Start recording automatically when
7:51PM 6 Please advise: Asterisk on Dell PowerEdge 1750 w/ hyperthreading
2:21PM 3 Cisco remote reboot
12:59PM 2 Divitas
9:50AM 1 h323friends & peer realtime
8:52AM 7 Zonbu
8:44AM 2 Asterisk 1.2.18 problem
2:16AM 3 SIP accounts from MYSQL.
 
Saturday May 26 2007
TimeRepliesSubject
6:13PM 2 Connect two Asterisk boxes through IVR Menu
3:33PM 17 reset Polycom phones remotely
12:33PM 3 test tools of Asterisk server
10:59AM 0 Voicemail and Time Conditions
6:06AM 12 Asterisk in Xen domu with tdm400 hardware
2:45AM 1 Wi-Fi+Wireless Router
2:36AM 0 Asterisk+IMATE PDAL Configuration+Softphone
1:31AM 5 chan_capi install problems
 
Friday May 25 2007
TimeRepliesSubject
7:33PM 2 Linksys WRTP54G-NA with SIP
6:16PM 0 (OT) Interesting and Cheap Device BAFO VoIP Internet Telephony Device Messenger CallBox
3:16PM 1 CDR not recording accountcode on SIP Response 302 Call Forward From Phone
2:40PM 0 GS BT200 dialling PC501
2:17PM 8 TDM bus extension.
1:14PM 2 Queue help: Extending RRMEMORY strategy to use penalty
12:21PM 0 mysql connect
11:49AM 31 Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
10:09AM 1 Suggested BRI cards?
10:03AM 0 Automated outbound call retries
9:58AM 1 wait for rings, answer on outdial via SIP
9:45AM 3 Asterisk with Multiple Network Interfaces
9:13AM 0 standard TDM interface cards that work in Asterisk?
8:54AM 1 Start recording automatically when xferring to an extension?
8:32AM 2 H Parameter in Dial Command
8:31AM 5 Polycom or Linksys phones bootp tftp config setup
7:14AM 4 Urgent: DTMF does not work with, rtpmap:101 telephone-event/8000
5:56AM 1 Problem with call parking
4:32AM 3 how to use sable (festival) markup with asterisk
4:19AM 0 Asterisk to Alcatel 4400 via PRI: analog extensions work - digital do not
4:15AM 6 Matching "+" at the beginning of the line
2:58AM 0 rxgain/txgain in chan_sip
2:04AM 0 Asterisk Users Conference Friday May 25th 12:30 PM EDT
 
Thursday May 24 2007
TimeRepliesSubject
10:31PM 0 Re: asterisk-users Digest, Vol 34, Issue 114
9:40PM 3 Urgent: DTMF does not work with rtpmap:101 telephone-event/8000
8:57PM 3 Echo on hard SIP devices...
6:37PM 2 vmoutcall]
4:28PM 2 Call Center Application
3:15PM 2 Login log out support
2:54PM 3 Cisco CP-7970G
2:04PM 2 transfer call sip to zap
1:46PM 2 Conference room as Music on Hold
11:21AM 2 Basic connection between Mitel 3300 ICP and Asterisk (trixbox) - from a clueless newbie....
10:01AM 3 vmoutcall
9:35AM 28 Bottom line on fax reception
7:46AM 2 SCCP
7:44AM 9 Integrated T1
7:32AM 2 Additional commands for MeetMeAdmin
7:28AM 5 meetme sounds
7:22AM 0 SLA with SIP-only environment
6:41AM 1 Nokia release
6:18AM 1 Parking Lot CallerID
6:04AM 0 bridging calls between two numbers with extensions
5:52AM 1 PSP Voip
4:54AM 3 PRI problem, pri_fixup_principle: Call specified, but not found?
2:17AM 8 modprobe
2:03AM 0 asterisk-backports.org giveaway
1:53AM 0 redirect on AT-530 IP Phone
1:44AM 3 There is no tone on an outgoing call
 
Wednesday May 23 2007
TimeRepliesSubject
7:57PM 0 AW: WiFi SIP phones
7:56PM 0 Realtime Queues and Agents
7:51PM 2 CDR on channel 'IAX2/u92613106-3' already started
7:35PM 0 Fax Detection Using Nvdetect
5:47PM 0 IVR Loop on invalid input
3:32PM 0 Deadlock problem with agents, queues and PRI (stop accepting incoming calls in PRI line)
1:37PM 1 Call limit per sip account user.
1:32PM 3 Using gizmo as softphone for Linux
12:53PM 0 ITSP that honors Dial Around Compensation
12:34PM 0 Problems compiling res_config_mysql (asterisk addons)
12:28PM 1 - SOLVED - stream file not working but get data and exec background work
11:48AM 1 stream file not working but get data and exec background work
11:34AM 3 TE205P, E1, Panasonic PBX and hang-up issues
8:23AM 0 asterisk+nortel3904
8:15AM 1 Asterisk Realtime problem
7:23AM 3 Asterisk and CCM 5.x SIP trunk
7:10AM 0 SIP.CONF: incominglimit and outgoinglimit
6:51AM 1 Bristuff with Billion ISDN
6:31AM 2 problem with attended call transfer
6:31AM 1 voicemail notification.
5:59AM 0 KCAUG Meeting Reminder!
5:27AM 9 What replaces SetCallerPres in 1.4
5:01AM 1 None random SIP channel names
4:53AM 2 SCCP + hint
4:52AM 4 showing camera on video phone
3:33AM 0 Need starter information (newbie)
3:01AM 49 WiFi SIP phones
12:41AM 7 SIP Dial Command to a non-Asterisk url
 
Tuesday May 22 2007
TimeRepliesSubject
7:12PM 0 Mix Dial, Chanspy and MixMonitor or Monitor
3:14PM 3 Fax detection
3:14PM 0 (no subject)
2:43PM 11 FXS + Pots Extensions Help
1:49PM 0 FW: autologoff
11:50AM 10 Working softphone for poket PC
11:38AM 0 how to disable global authentication for registration
10:44AM 1 SMS
10:06AM 0 Astsee v0.5 now available, X/Linux Asterisk Usage Auditor and Monitor
10:05AM 2 Phones fail to ring
9:23AM 1 Net 2 Phone - Asterisk - Problem
7:07AM 2 Kernel Panic in wct4xxp during unload on Zaptel-1.4.4
6:22AM 5 how can I catch the event generated when a parked call is hung up?
5:11AM 3 Dial out issues.
5:08AM 15 SIP & Echo
4:53AM 0 Dialplan Problem - Outgoing
4:31AM 0 asterisk TAPI interface
3:21AM 7 Local SMS how-to.
2:32AM 1 Why 2 branches of asterisk development?
2:30AM 1 Blackberry 8800+VoIP Configuration
 
Monday May 21 2007
TimeRepliesSubject
8:02PM 1 Voice mail issue
2:02PM 1 Windows Media streaming for MOH?
1:38PM 1 getting a call back from voicemail?
12:42PM 3 Originate and bridge Can it be done? Best Way?
12:01PM 4 MoH WAY too loud
11:54AM 0 AGI: Festival & Ringing on Screening not working properly
11:39AM 8 Aastra MWI
10:08AM 5 CAS signalling conflicts with Clear channel
8:18AM 1 FW: Re install
7:46AM 0 Grandstream FXS Gateway star codes
7:42AM 2 Help installing on OpenSuSE 10.2
7:07AM 6 Delete voicemails after X days
6:20AM 5 VoiceMail Access
5:18AM 2 DTMFToText Installation process
2:58AM 1 MusicOnHold() stops after exactly 60 seconds
2:51AM 4 Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage
2:49AM 4 asterisk and fax machine
1:56AM 0 Asterisk Users Conference this Friday: Kerry from Trixbox
1:41AM 0 "dtmf transcoding" with asterisk
1:01AM 1 Vicidial
12:07AM 0 compile asterisk in arm-linux!
12:01AM 0 Gustavo Souza Queiroz está ausente do escritório.
 
Sunday May 20 2007
TimeRepliesSubject
11:53PM 3 MySQL/IVR Integration
11:02PM 2 Queuemetrics and Asterisknow
9:17AM 5 Caller ID matching
7:41AM 3 OpenWengo + Asterisk?
 
Saturday May 19 2007
TimeRepliesSubject
8:03PM 0 Astersik+ss7
7:18PM 3 Call someone to instantly join conference using MeetMe
10:26AM 5 Zaptel hangs machine...
10:05AM 2 Asterisk On Solaris 10
9:34AM 0 SIP NOTIFY on voicemail
9:05AM 2 (OT) Anyone Ever Use http://shopfort1.com as a Broker
8:01AM 5 Asterisk and iBasis
7:54AM 0 a/b door phone
7:20AM 3 Asterisk on OpenSuSE 10.2
7:07AM 0 Branch differences
5:56AM 0 asterisk and snmp
5:34AM 2 Ser vs. DUNDi
5:16AM 1 asterisk not sending ACK after reinvite
2:37AM 0 G729 codec problems
 
Friday May 18 2007
TimeRepliesSubject
8:36PM 1 xten will not send tones to * and i from sip phone
4:39PM 2 unsubscribe
2:36PM 5 Who picked up with *8?
12:06PM 0 IAX2 sniffer and player
10:42AM 0 4-port ATA
9:50AM 4 zap fallback
9:47AM 1 Fwd: Asterisk console loop?
8:06AM 3 Query about DTMF generate
7:53AM 4 TE212P octastic initialization failure
7:51AM 1 web app to playback recorded phone calls.
7:40AM 0 cpu usage for G.729 codec
7:38AM 0 Re: asterisk-users Digest, Vol 34, Issue 82
6:54AM 0 mISDN: long delay when making outbound calls
5:51AM 9 Phone losing IP address for a few seconds but doesn't drop call
5:34AM 1 Asterisk vs. Shoretel
5:29AM 0 Asterisk as SIP Provider
5:25AM 0 call-limit=2 , call counter not reset to zero after hangup
5:22AM 2 CallerID not detected by TDM22B
12:44AM 0 EUA voip provider
 
Thursday May 17 2007
TimeRepliesSubject
11:31PM 3 Call to an arbitrary outbound number by asterisk
10:32PM 0 astertisk for dummies
7:11PM 1 Anyone tested the new Sony Ericsson P1 phones..
2:24PM 0 Digium HPEC 9.00.003 released
1:42PM 7 OK to have Asterisk and clients behind firewalls?
1:22PM 0 AGI and fork()
1:08PM 1 Multiple lines on Linksys/Sipura phones
12:50PM 1 Cascading Queues
11:37AM 2 Blacklist
11:12AM 0 Compiling DBQuery
10:37AM 6 GUI: Not Found. Move along
10:08AM 0 Call waiting / hook flash on ZAP trunk from SIP phone?
9:59AM 7 FastAGI hangs up channel if server is not available
9:54AM 0 Busy tone with different length tone
9:54AM 0 Re: asterisk setup for church / conference call
9:21AM 24 DUNDi configuration problem
8:27AM 0 emergency call called party control
7:23AM 7 asterisk setup for church / conference call / speaker system integration
4:02AM 2 Asterisk Queue MOH
2:47AM 2 Quadbri Cellular Issue
1:40AM 6 how to define a key to decline incoming call
12:17AM 0 [SPAM]RE: CDR is not written
 
Wednesday May 16 2007
TimeRepliesSubject
11:54PM 2 Anyone Installed a Digium TE110P or TE120P card in Canada?
11:50PM 5 CDR is not written
10:47PM 0 NO ANSWER, When openser make an oubound SIP call to my asterisk
9:37PM 1 How to remote reboot Grandstream GXP-2000
8:39PM 0 IAX certificate-based authentication
3:48PM 5 Please help me finding good A-Z provider
3:38PM 1 Digium TE120P and Canada FCC or DOC
3:19PM 6 Get sip response code
3:19PM 0 AGI "record_file" issue... bug?
1:42PM 2 getting call status using Manager API
1:21PM 3 MeetMe and ChannelRedirect
1:06PM 0 Save Key tone to MySQL DB
9:40AM 8 Microsoft's Move Into IP PBX Market
9:23AM 8 SIP Hardware Phone
9:08AM 1 Video Door Phone
7:17AM 0 Passing dialstatus back through an IAX chain ..
7:12AM 14 Which KDE editor to edit Asterisk config files ?
6:21AM 1 SIP INVITE failing and AgentCallBackLogin()
6:17AM 9 GSM Cards for Asterisk (UK)
6:08AM 0 FW: Play a file on a channel from the Manager API
6:04AM 1 G729 Transcoding problems
5:09AM 0 Problem with CDR and DeadAGI
5:07AM 0 Busy tone with the different length tone
4:45AM 3 Asterisk Queue Problem - Automatic Call Distribution
4:34AM 1 Sip client registers then unregisters
4:32AM 18 asterisk manager interface stability
4:03AM 1 WaitExten not responding on key presses
3:35AM 1 Asterisk SRTP certificates
3:08AM 8 voice recording on legacy PBX
2:25AM 3 PRI got event
1:20AM 3 HUDlite Server on Debian Etch/Asterisk 1.4
 
Tuesday May 15 2007
TimeRepliesSubject
8:58PM 3 Asterisk is not showing the correct Incomming CallerID
8:23PM 0 PATH_MAX' undeclared here (not in a function) in asterisk!
8:10PM 0 [RTP] PSTN -> Gateway -> Phone
8:01PM 1 Asterisk 1.4.4 reproducibly dumps core on Solaris 10
12:33PM 4 Feasibility Request
12:04PM 1 Astsee v0.1 released - an Asterisk channel monitor for linux/X windows
11:22AM 2 Zaptel 1.4.2.1 and TE212P
9:47AM 14 Outside lines are just not happening...
8:46AM 9 Trixbox problems
8:39AM 0 IAX2 peer unreachable in one direction - NATproblem?
8:05AM 1 polycom 501 configuration setting
8:03AM 1 finding the sipp soft phone list on the wikey
7:02AM 5 Mr. Spencer Written
5:48AM 0 How to set Name/username to something like 229/john instead of 229/229
4:45AM 2 Originate and ForkCDR()
 
Monday May 14 2007
TimeRepliesSubject
11:56PM 0 Req-Installation process for app_dtmftotext.c
10:45PM 6 [*Win32 0.60] Sending call notification by e-mail/web?
10:36PM 4 How to write data to astdb?
10:01PM 1 `PATH_MAX' undeclared here (not in a function) in asterisk!
8:45PM 3 Web based call control
5:59PM 1 Blind Transfer - Who transferred the call?
3:52PM 0 Asterisk Now
2:40PM 4 Proper AGI use with MySQL
12:52PM 1 ast_yyerror - Help
12:48PM 0 Junghanns DuoBRI Card HELP !
12:34PM 1 Some problems with mysql CDR
11:38AM 1 IAX2 peer unreachable in one direction - NAT problem?
11:09AM 0 Areski CDR
10:07AM 1 DTMF not recognizing *
9:54AM 2 OT (semi) E60 problem
9:11AM 1 Difference between making a call and Originate
9:07AM 5 Simultaneous Capacity
8:39AM 0 How is Context Determined when Transferring a Call?
8:02AM 0 ChanSpy
7:41AM 1 How obtain the slot position when a call is parked?
7:28AM 23 zaptel huge irq problem
7:12AM 0 Codename Pineapple - Chan_sip3 - what's the status?
7:08AM 6 OT ? Number portability, land line to Cell
6:43AM 0 Play a file on a channel from the Manager API
6:37AM 1 Asterisk and unicall + mfcr2 signalling
5:06AM 8 How to bring MoH volume down
4:56AM 4 queue_exec: Unable to join queue
4:47AM 6 function_db_read: DB requires an argument, DB(<family>/<key>)
4:33AM 4 dialplan: execute on hangup
2:12AM 0 quadbri and bristuff : no answer to isdn setup message
12:57AM 0 Decrease group counter without hangup?
 
Sunday May 13 2007
TimeRepliesSubject
7:43PM 1 Sudden appearance of SIP/2.0 401 Unauthorized
3:39PM 0 Asterisknow b5 - trouble registering at voip provider
6:40AM 4 TC400B load problem
4:58AM 3 Zapateller and IAX2
 
Saturday May 12 2007
TimeRepliesSubject
10:00PM 1 Confirmation key to answer -- for a queue
6:35PM 5 Call to Skype network
6:23PM 2 zonedata.c
3:15PM 2 Hardware Echo cancellor Digium Wildcard TE212P
1:11PM 12 Asterisk High-Capacity Stability
10:58AM 0 Recent zaptel versions break CLIP?
8:32AM 0 Preparing music on hold
7:38AM 0 DTMF detection problem on wctdm24xxp
6:29AM 0 1.2.17 -> 1.2.18 asterisk crash
3:27AM 3 Remote extensions not working on provider's wireless Internet connection
2:19AM 0 ser problem
 
Friday May 11 2007
TimeRepliesSubject
11:49PM 4 Fwd: SER as a Session Border Controller
2:33PM 0 Asterisk crashes
11:36AM 49 Dry Copper Pair
11:18AM 1 Swissvoice IP10s setup
8:12AM 1 Problems with outbound calls through VSP
8:06AM 11 Dealing with 2 SIP providers
7:32AM 2 Strange problem with asterisk
7:04AM 3 Module wctdm24xxp not found - TDM808P on debian
6:38AM 2 Need a RTP/SIP Proxy to be used as SBC (Session Border Controller)
6:18AM 1 'Invalid characters in name' with asterisk-gui
4:27AM 3 A couple of questions for the Mitel gurus (phone-related - not systems)
4:14AM 2 Rapid DTMF missing digits
3:39AM 2 Dundi and unknown remote peers
1:50AM 0 Reminder: Asterisk Users Conference Friday 12:30PM EDT
12:18AM 2 record voice
 
Thursday May 10 2007
TimeRepliesSubject
10:20PM 6 Any other softPBX like Asterisk?
8:17PM 1 Voice mail volume
2:41PM 0 Correct setup for directing already ringing calls to newly available phones
2:37PM 2 SNOM 360 Rejecting Calls
2:30PM 3 Iaxy clicking
2:01PM 3 socket_process: Received mini frame before first full voice frame
1:54PM 16 The downside of Asterisk and least cost routing...
12:31PM 1 ices low volume
12:23PM 1 Redirecting an existing channel?
11:59AM 0 Asttapi Collect
11:51AM 1 asterisk SIP domain (in LAN or DMZ)?
11:02AM 1 Polycom power over ethernet (PoE) cables for 500/501, 600/601 and 650 sets
10:23AM 1 Polycom headset button blinking
9:36AM 13 force outgoinc callerid
8:49AM 5 CITEL gateway does it work well?
8:21AM 1 Softkey config example for Cisco 7941/7961
8:10AM 0 AgentCallbackLogin not working with 1.4.4
7:22AM 1 call transfer to asterik.. asterisk as an end point
5:03AM 0 Anyone tried using a SCCP service provider>
4:47AM 1 Unbridge the two call.
4:26AM 15 Cutted audio or 2/3s blanks on EuroISDN - Asterisk1.4
4:06AM 0 Asterisk To GoogleTalk Audio Issues
3:58AM 0 Anyone Want To Trade a TE405P for a TE410P?
3:23AM 0 Cutted audio or 2/3s blanks on EuroISDN - Asterisk 1.4
2:54AM 16 TDM410P
1:59AM 3 module zttranscode: what is it?
12:50AM 0 Asterisk 1.4 and AgentMonitorOutgoing
12:50AM 4 AT530 Telephone
12:38AM 1 ivr and internal voip phones
 
Wednesday May 9 2007
TimeRepliesSubject
9:14PM 0 Trixbox drops call after running AGI script
5:01PM 0 MINNESOTA: Twin Cities Asterisk Users Group - Saturday May 12th 2007 - 11:30am
4:33PM 1 Is there a possiblity to check in the dialplan whether a SIP user is registred?
2:42PM 0 SPA841 3.1.1(a) firmware file
1:25PM 1 Ericcson analog phone
12:45PM 7 10 FXS - Channel Bank or PCI Card?
11:52AM 1 Boost Polycom IP601 headset volume
10:39AM 10 Mobile Number to Mobile carrier mapping
10:36AM 15 List of telemarketers??
10:19AM 1 Question about Asterisk 1.4 depoyment.
10:08AM 0 additional volume added to sound on CONSOLE/dsp
9:13AM 12 SIP Problems continue...
8:33AM 0 Re: asterisk-users Digest, Vol 34, Issue 46
8:01AM 0 using voip software client as public address system. Low volume
7:49AM 4 select menu
6:40AM 1 fax receiving
6:37AM 3 The 'h' extension problem
5:46AM 1 Replaces header
5:45AM 0 Re: asterisk-users Digest, Vol 34, Issue 45
4:57AM 0 Microsoft CRM & Asterisk
4:07AM 12 The purpose of DUNDi
3:11AM 15 Audio going blank for a few seconds and then comes back. What could be the reason?
12:51AM 0 Bug no. 8680 (billsec is 0 even when the call is answered) in Asterisk 1.4.2
12:39AM 4 Testing ISDN T1/E1 Bri and Pri etc.
 
Tuesday May 8 2007
TimeRepliesSubject
10:14PM 2 asterisk with festival facing problem
9:33PM 0 Re: Call waiting tone
8:53PM 1 Aastra phones?
7:02PM 3 asterisk 1.2 and UDP packet numbering on bridged channels (for jitter buffering)?
4:40PM 1 LDAPget or something else?
3:55PM 3 Ringing Volume
3:05PM 1 Problems witch SPA3102.
2:18PM 0 Ericsson dialog 4187
2:16PM 0 voip-info.org mirrors?
1:47PM 1 Problem when PABX call to Asterisk by Unicall
11:34AM 3 Vista compatibilty in SIP softphones
10:51AM 0 Re: asterisk-users Digest, Vol 34, Issue 42
10:47AM 4 MYSQL Query --> PAGE
10:25AM 0 random sections lost from call recording
10:19AM 1 Sound files
9:35AM 0 Sangoma cards for sale
8:48AM 0 Asterisk 1.4.2 tanking CPU
8:07AM 1 G729 - Part cut
7:54AM 1 Outbound call through a Single Asterisk Server
7:42AM 1 Remote Phone and Server Behind NAT
6:23AM 1 load modules
5:53AM 3 Sangoma A101 on Freebsd 6.2
5:09AM 2 asterisk 1.2 from svn ... lock on shutdown
5:07AM 6 Responding to SIP OPTIONS
4:28AM 4 outgoing calls
3:09AM 0 Beronet card - issue?
2:10AM 1 Re: asterisk-users Digest, Vol 34, Issue 39
12:04AM 8 Bug in voicemail module of Asterisk 1.4.2?
 
Monday May 7 2007
TimeRepliesSubject
11:54PM 0 Problems with SPA3102
11:11PM 1 isup-oli or ani2
11:08PM 0 asterisk with festival facing problem!!!!!
6:53PM 5 iax to iax Reject Connection
4:10PM 1 help MWI setting
10:40AM 4 h323 problem with asterisk 1.2.18
10:04AM 10 Could two Asterisk servers connect through VPN
8:34AM 0 H323 to H323 bridging ... failed ... also with chan_local
7:49AM 0 ISDN with Billion
7:45AM 4 Asterisk to record CDR in DB Oracle
7:29AM 2 Queues: Play a list of sound file n round-robin at a specific interval
7:16AM 7 Problem with the loading of the cards in Debian
7:07AM 13 RTP Mixer
6:43AM 0 Rhino Analog Cards - Questions
2:52AM 5 Send SIP Re-invite.
2:48AM 21 HPEC audio clipping
 
Sunday May 6 2007
TimeRepliesSubject
3:42PM 6 Channel Bank
2:42PM 2 Problem with conferences, Vlada, Pancevo
10:59AM 0 Asterisk 1.2.14-BRIstuffed-0.3.0-PRE-1y
8:33AM 1 Buddywatch on Polycom 601 crashes phone
6:42AM 2 Were i make mistake
2:29AM 3 Call waiting tone when calling a busy station?
2:21AM 2 Polycom 601 - To not make noise when there is VM
 
Saturday May 5 2007
TimeRepliesSubject
6:07PM 0 Channel / Exten Status
5:16PM 2 Dial Plan for Multi-Location & Support Queue
4:38PM 2 ${ANSWEREDTIME} Broken on 1.2.13?
4:24PM 0 res_config_pgsql.c in * 1.4.4
3:11PM 5 I'm looking for solution
2:30PM 2 Asterisk 1.4.4 and Custom Postgres 8.2.4 (checking for PQexec in -lpq... no)
1:02PM 1 ODBC
10:06AM 13 asterisk telemarketer torture sound files
9:07AM 5 TDM400P usada?
6:09AM 1 SIP registration problem
6:04AM 2 Manager API Output
5:49AM 0 TLS support
4:17AM 5 Queue Status
 
Friday May 4 2007
TimeRepliesSubject
11:39PM 0 Queue Answer
10:38PM 7 Asterisk x legacy pabx
10:31PM 4 Queue and voicemail
5:47PM 5 SLA broken in 1.4.3?
4:08PM 0 Semi-OT: useful things to do with XML browsers inphones
2:28PM 1 Asterisk 1.2 on CentOS 5?
1:28PM 0 Asterisk registration SIP confusion. Can someone explain this?
12:27PM 0 E1 config for chile
11:37AM 4 question about more than one drop file
11:03AM 3 AsteriskNow!
10:55AM 7 zaptel compile error
9:20AM 1 ASA-2007-013: IAX2 users can cause unauthorized data disclosure
8:15AM 5 Headset for Polycom
6:33AM 0 does Not detected HANGUP and DTMF
6:07AM 0 Console flooded by WARNING app_meetme messages
5:45AM 23 app_txfax, app_rxfax
5:04AM 30 Starting Asterisk on Ubuntu 7.04
4:51AM 0 Error compiling patched pppd for zapras
2:46AM 4 need more knowledge about asterisk
2:34AM 0 Asterisk Users Conference Friday, May 4th at 12:30 PM EDT
1:22AM 4 cpu usuage
12:55AM 0 1.2.x -> 1.4.x upgrade: dialplan block no longer works
12:07AM 2 Asterisk Codec Translation Table
 
Thursday May 3 2007
TimeRepliesSubject
10:35PM 1 Call interruption
9:48PM 3 OT - robo dialer
9:12PM 0 Called party identification - where to takecalledname?
7:59PM 2 Connections rejected in DUNDi requests
6:17PM 0 [Announce] Web-MeetMe 3.0.2 and 2.2.2 Released
5:37PM 2 Unable to Execute System Command From DialPlan
5:35PM 7 SIP RealTime Friends
5:34PM 4 SIP peer / Maximum retries exceeded on transmission
5:32PM 1 Asterisk 1.4 and Cisco Phones 7940
3:57PM 0 Restricting membership to a queue?
2:32PM 0 Error messages in console : sipsock_read: SIP MESSAGE JUST IGNORED:
2:19PM 3 VoiceXML + Nuance
2:01PM 3 Balancing interrupts.
1:34PM 5 Double DTMF digits
12:17PM 6 FXO recommendation
11:52AM 0 Strange noise - Polycom
11:37AM 1 Linseed
11:08AM 0 ast_parse_allow_disallow: Cannot disallow unknown format ''
10:02AM 20 Asterisk-Polycom HELLLLPPP!!!!
9:56AM 1 Autologoff
9:05AM 14 IAX Trunk
8:56AM 4 Linksys SPA3012 inbound FXO problems
8:26AM 2 "you have been kicked my this conference"
7:57AM 0 Secondary redirect failed
7:18AM 2 Wildcard TE410P problem
7:18AM 4 Semi-OT: useful things to do with XML browsers in phones
7:14AM 1 Virtual IP Adresses and SIP requests failing...
7:03AM 2 Called party identification - where to take called name?
5:38AM 5 zttranscode crashes server
4:47AM 5 CDR(accountcode) empty in * 1.4.4 (for local chan)
2:51AM 2 Get Asterisk to redirect a SIP INVITE
2:35AM 2 sipura spa9x1 - missed calls not wanted
1:44AM 5 UK - 2 port ISDN2e cards ...
1:05AM 3 0 duration but non-zero billsec in mysql cdr
 
Wednesday May 2 2007
TimeRepliesSubject
11:15PM 0 Call Limit with multiple SIP extensions
11:09PM 0 rtpmap encoding parameters & the 'unknown codec' problem?
6:26PM 3 IAX and SETLANGUAGE delays
3:12PM 14 OT: USB T1/E1 Interface?
2:55PM 0 Kansas City Asterisk User Group Meeting Announcement
2:33PM 1 SIP Proxy
12:04PM 9 allowing call every 15mins
12:02PM 2 allowing call to my pabx every 15 minutes
11:56AM 2 PRI T1 Problems
10:54AM 6 Large dial plans and variables
10:51AM 1 Asterisk locked up
10:51AM 13 IP Phone Provisioning Tool by voip.com.sg - xml generation
10:48AM 6 Daemontools and holidays macro
10:35AM 0 Call In queue stucks
10:24AM 4 1.4 memory leak?
8:49AM 2 delay in switching between contexts
7:31AM 2 Asterisk-1.4 with agent snmp
7:28AM 3 Returning different SIP Hangup Cause
7:15AM 0 cdr on channel lacks end, not posted
6:33AM 2 SIPGetHeader in Asterisk 1.4
6:30AM 5 Reinvite after DTMF?
5:15AM 0 ZAP Error: Unable to create channel of type 'Zap'
5:10AM 2 Volume (gain?) on VoIP-only system.
5:05AM 1 MySQL ** DBI connect failed : Too many connections**
4:57AM 0 voice mail format
4:19AM 0 MySQL ** DBI connect failed : Too many connections **
4:14AM 9 VPN between Asterisk server and phone client
2:35AM 0 Asterisk Integration of XMPP/Jingle
2:13AM 1 Re: RE: Digital Phones (Dean Collins)
2:10AM 1 Re Re: TC400B
1:07AM 3 make iso image
 
Tuesday May 1 2007
TimeRepliesSubject
11:34PM 2 Runaway MOH/mp3123 process?
9:01PM 9 using Playback() to play a random sound file
3:16PM 1 TC400B
3:00PM 12 Applet?
3:00PM 26 Digital Phones
1:56PM 0 Re: [asterisk-dev] SRTP implementation
1:54PM 4 Display Caller ID of called party
1:47PM 3 Stanaphone business ok?
1:37PM 1 chan_sip seems to be hanging
12:37PM 1 T1 interface
11:23AM 2 MYSQL application in dial plan
10:51AM 14 OT: Capture Asterisk traffic
9:46AM 7 is dundi worth pursuing in this situation?
9:44AM 2 Channel stuck with call pri flag
9:17AM 5 How do I do this in Asterisk?
8:47AM 7 Delay in Dial()
8:33AM 0 Email to HP Product Suggestions - Seamless Transparent Fax Gateway
8:31AM 10 How many users can be supported simultaneously?
8:26AM 2 Cisco 7940 no outgoing audio
6:27AM 0 RE: Voicemail on Different Server (MySQL Replication split thread)
6:24AM 3 Change Codec
5:59AM 1 restrictions on meetme with agi background
5:10AM 5 chan_local
5:02AM 19 Autoattendant press 1 collides with extension numbers...
2:21AM 0 Re: Anyone having trouble with claling US Domesticon Sellvoip?
2:16AM 0 Two Context Residing On The Same Server
1:10AM 6 did we all get spammed by TechnoCo ?