asterisk users - Oct 2006

Tuesday October 31 2006
TimeRepliesSubject
11:39PM 1 wrong password on authentication for INVITE
8:35PM 0 AW: NAT issue ? [More Info]
6:29PM 5 Opinions on the best wholesale origination/term providers
5:38PM 0 NAT issue ? [More Info]
5:13PM 2 Newbie Questions
4:52PM 0 [SPAM HEADER] - Re: Snom or Cisco Phones? - Email found in subject
4:47PM 1 NAT issue ?
4:32PM 0 SIP with Qualify and NAT
2:49PM 2 compilation problem with asterisk-addons
1:24PM 0 FXO Card's vs. T1
12:42PM 5 AW: Snom or Cisco Phones?
12:15PM 8 DTMF Tones
12:03PM 1 Strange Characters in CLI on TTY9
11:22AM 13 FXO Cards vs. Channel bank with T1
11:13AM 3 overlap of zap trunk groups
10:28AM 8 simultaneous ring - call groups or queues or something else?
10:24AM 8 Example Polycom function key config
10:21AM 11 Snom or Cisco Phones?
9:44AM 1 Astricon followup
9:43AM 3 channel.c: Unable to request channel ZAP
9:00AM 7 Asterisk Call Statistics
8:44AM 6 Asterisk web interface is not parsing the PHP pages
8:29AM 1 auto recording extensions
7:55AM 4 Asterisk and ARI (Aterisk Recording Interface) integration problem
7:12AM 0 Asterisk dial out (in SIP) to another asterisk context !
7:03AM 2 SIP RTP flow
6:34AM 1 +Ura +md3200 nao encaminha ligacao
6:06AM 0 SMS and 1.2.12
5:27AM 1 dial D option with w for wait?
5:18AM 2 sip realtime broken?
4:19AM 13 best gui
3:44AM 0 Dropping extra frame of G.729 since we already have a VAD frame at the end
2:42AM 0 Setting up UTStarcom F300
2:41AM 0 Read cmd
2:31AM 2 Bridging Video Calls using Zap
1:59AM 1 Asterisk does not bridge zap channels on outgoing calls
1:29AM 1 Fedora Core 6 (FC6) and Asterisk-1.2.13 and Zaptel-1.2.10 compile problems
1:26AM 10 Asterisk both behind a NAT and outside at the same time
12:37AM 4 S(x) - Hang up the call after 'x' seconds - Not working from queue
 
Monday October 30 2006
TimeRepliesSubject
9:36PM 8 Audiocodes MP-114 noise
6:52PM 3 Cisco 7960 Skinny calling SIP phone
6:13PM 1 dealing with blind transfers to invalid extensions
5:44PM 0 sip trunk - SIP/2.0 488 Not Acceptable Media
5:14PM 3 Registration problem
5:08PM 8 Architecture for Asterisk
4:52PM 4 IVR
4:24PM 7 Asterisk architecure
4:22PM 6 Server Recommendations
3:38PM 0 Good phones for outside of the office?
3:16PM 1 MFC/R2 patch problems
1:08PM 3 Live creation of trunk groups
12:52PM 1 Forwarding recorded calls to Voicemail
12:51PM 9 light web user interface
12:51PM 2 Fxo box for asterisk ?
12:45PM 2 TE110P Card
12:05PM 0 Asterisk Billing Plataforms
11:42AM 7 Asterisk and Panasonic KX Model
11:30AM 1 Wildcard X100P Suport
10:38AM 1 Anyone got a dialplan for SPA ATAs for ISN?
10:31AM 3 Billing Solution ?
9:27AM 7 operator console
9:17AM 3 Grandstream ATA 286 tdm400 and Asterisk 1.2-13
8:57AM 0 Asterisk and Siemens C450IP
8:46AM 1 Realtime in the Real World
7:21AM 9 How to do Automatic Daylight Saving on Grandstream GXP-2000
6:03AM 1 show logged clients
5:46AM 0 Extension Matching with "Match As You Go" Dialing
5:43AM 0 Realtime trouble with contex
5:42AM 1 SIP Server
5:36AM 2 Mac OS X Desktop / Asterisk integration?
4:55AM 1 Need Help in Meetme (Conferencing)
4:43AM 0 Vgsm driver 0.18.0 released today
4:32AM 0 RT Problem: Asterisk & Session Border Controller
3:31AM 0 Problem with Digium 400P and asterisk 1.4
3:29AM 3 anti ex-girlfriend
3:26AM 0 Problem with incomming calls
3:06AM 0 Intel S3000AHLX - Digium TE110P
2:22AM 0 Information on Asterisk 1.4-beta 3 and ARA
12:17AM 0 Re: Linksys PAP2: calling tone stops after 5
12:10AM 0 Call from internal num. to VoIP gate
 
Sunday October 29 2006
TimeRepliesSubject
10:19PM 2 CID and CDR conflict?
10:13PM 6 Incorrect Ring tone. Getting a US tone when it should be AU tone
8:34PM 4 No ring tone when using IAX
6:38PM 1 Multiple dial macros at the same time
5:25PM 3 asterisk-1.2.13 fails to 'Make' in Fedore Core 6'
4:58PM 0 Polycom IP500 Problems
3:47PM 5 blind transfers with IP Polycom 501
2:41PM 1 AEL2 and the variables
2:27PM 1 Asterisk Voicemail with ODBC Realtime Access
2:00PM 1 Linksys PAP2: calling tone stops after 5 tones
1:43PM 1 Out bound calls 'you must first dial a 1'
1:39PM 2 Something is trashing /var/run
1:22PM 4 No zap* commands?
10:29AM 0 hardware requirements..
10:13AM 0 H.263 Video Messages
3:44AM 4 Pager Voicemail Message
1:47AM 2 app_meetme not loading
 
Saturday October 28 2006
TimeRepliesSubject
8:17PM 1 How to make different ext using different trunks?
6:16PM 11 VoIP GSM Gateways
4:40PM 1 Compiling Zaptel 1.2.10 on Ubuntu 6.10
4:26PM 3 Asterisk behind NAT and without portforwarding for rtp
2:31PM 0 Queues: roundrobin w/ reset ("circular call distribution")
11:47AM 13 Diva server 4bri and Portuguese BRI
10:20AM 1 tx_fax not getting entire fax
9:03AM 0 Asteroid SIP Denial of Service Tool
7:58AM 0 Zap disconnect
7:55AM 0 Is it possible to connect two servers using SIP?
6:45AM 2 translate.c:88 powerof: Powerof 0: No power??
5:44AM 0 IAX2/SIP Wifi Phones
1:51AM 0 Polycom 501 + Voicemail notification
1:09AM 1 Configuring 2 Asterisk servers with a SIP trunk
 
Friday October 27 2006
TimeRepliesSubject
10:40PM 1 dialing external number within meetme
9:30PM 3 0 channels configured with tdm400 (tdm04b rev. G)
7:05PM 2 Waiting before executing System command
3:59PM 0 autocreate peer + sippeers table entry => auth required?
3:43PM 0 Voicemail 'exitcontext'
1:27PM 0 Enterprise Asterisk User Group
1:21PM 0 Vancouver Asterisk User Group
12:55PM 0 Zultys Phones w/ Encryption
12:31PM 0 Confused about SIP Realtime Updates
12:14PM 1 New Asterisk-GUI?
11:53AM 0 fully featured and robust * client gui?
11:39AM 1 Digium TE110P
11:08AM 0 detecting ring
9:59AM 2 Voicemail and OSX 10.4 Intel
9:27AM 1 meet me
9:17AM 0 Asterisk stopps matching extensions after first digit
8:49AM 16 [OT] wi-fi ip phone scenario
8:11AM 4 polycom's don't register with 2.6.18
7:28AM 1 Taking a Polycom IP601 home
7:26AM 0 set outgoing msn on chan_misdn
7:08AM 0 Enhancements for the Queue application
6:12AM 1 Snom, mute and rtptimeout
6:03AM 3 Advice on GUI
5:37AM 0 How to hung up , While in Conference going on.
5:23AM 4 IAX2 show peers - description
3:56AM 3 DTMF detection problem in PABX trunk
3:55AM 1 Direct call vs Block call
3:40AM 5 ISDN-BRI issue
3:21AM 3 asterisk misdn incoming line not working.
2:37AM 3 Would you support a Bristuff mailing list ?
1:57AM 0 lines usage statistics
1:54AM 0 Auto Dial problem!
1:31AM 7 Iax bug ?
1:25AM 0 chan_skype license?
 
Thursday October 26 2006
TimeRepliesSubject
10:57PM 0 Re: asterisk-users Digest, Vol 27, Issue 140
10:11PM 3 dialplan issue - 1& 0 should be evaluated false
9:39PM 0 dialplan issue - 1& 0 should be false
9:31PM 0 [Fwd: Asterisk n QoS]
7:44PM 1 simple dialplan trick I can't figure out (smdi, mwi substitute)
6:09PM 0 Make/Break ratio for Pulse Dialing
4:57PM 0 7960 (8.2) - Call Center - REBOOT
3:22PM 1 SipAddHeader
2:22PM 10 Asterisk and ISDN and Hylafax
2:12PM 0 Open SER or DUNDI
12:29PM 0 Call Routing Time Issue
11:50AM 1 Lumenvox speech recognition
11:40AM 0 Cepstral/Swift TTS app
10:39AM 0 Can't Register Client - Multiple Subnets
10:05AM 0 external username conflict in dialplan
9:32AM 5 Is SQLite a replacement for Mysql while using ARA in 1.2.x
7:32AM 1 IPv6
6:32AM 0 How to disconnect in Conferenceing in between the Confermce .....
5:25AM 1 channel.c: Avoided initial deadlock
5:10AM 0 Asterisk n QoS
5:01AM 0 question about IF
4:21AM 3 "Cheapest" way to determine channels in a group from outside asterisk?
4:09AM 0 Problem: Dial command with L option
4:00AM 6 porting numbers in UK telewest/bt/adept
3:51AM 29 SIP v IAX2
3:32AM 0 OOH323 GK Context Help
3:15AM 14 ECHO Cancellation in SIP Calls
3:13AM 2 Query regarding Pulse Dialing at 20 pps
1:54AM 1 PRI (TE205P) allways RED/NOP
12:39AM 6 chan_capi and bristuff
 
Wednesday October 25 2006
TimeRepliesSubject
11:42PM 1 Phone Rings, Immediate Hangup and then Rings Again.
11:06PM 0 spandsp bug
8:13PM 1 WiFi Phones (was Looking for Wireless Heaset for Polycom 501)
6:42PM 1 Default login information for a ArtDio IPF-2600
2:58PM 1 Need recommendation for SIP hard phones
2:22PM 2 CSU Support on Digium T1 Cards
1:59PM 2 Cisco 7971G-GE & SEP{MAC}.cnf.xml
1:08PM 0 [SPAM] - Looking for Wireless Heaset for Polycom 501 - Email found in subject
12:30PM 3 "No Authority Found"
11:37AM 3 Add second account to Xlite 3.0
11:31AM 9 Looking for Wireless Heaset for Polycom 501
11:29AM 2 Multiple queue_log files based on queue - is it possible??
11:25AM 1 Trixbox installation - ZAP channels becoming upresponsive
11:12AM 2 Re: Asterisk Manager
9:51AM 0 Re: Meetme... No channel type registered for'zap'
9:14AM 2 Simple example for call transfer.
8:05AM 0 chan_misdn
7:44AM 2 Without ZapTel inferface or Card install , is Conference working or Not
6:53AM 0 Conference is Not Working.... with OpenSER And Asterisk
6:20AM 3 Nerdvittle's Reminders and Zaptel
6:06AM 5 Maximum talktime in a queue?
5:55AM 3 SIP problem - ACT p160s error
5:27AM 3 Quintum DX as gateway to PSTN for Asterisk
4:19AM 0 *****SPAM***** asterisk 1.4 problem with call queues
2:44AM 2 PBAX-Group with QuadBRI card, outgoing call problem
2:24AM 8 Choice of soundfile format
1:54AM 2 Call is not coming through sipgate.co.uk+Asterisk
12:51AM 7 VoiceOne 0.4.0 released: a new web-based and open source GUI
 
Tuesday October 24 2006
TimeRepliesSubject
11:51PM 0 sip.conf - srvlookup
8:37PM 16 Callmanager 3.3(5) and Asterisk with ooh323
8:33PM 1 Adit 600 resetting
8:29PM 1 All calls Hangup after receive these logs.
5:03PM 3 AstFax Sending a Fax
4:47PM 1 Basic Conf
3:22PM 18 Meetme... No channel type registered for 'zap'
2:23PM 4 ASterisk Start problem
1:52PM 0 attempting native bridge on TDM2400
1:44PM 1 update_header: Unable to find our position
1:41PM 1 problem with setting outbound caller id when calling another asterisk
1:04PM 9 IAX2 goes "one way audio" when lag gets bad
12:48PM 0 Problem with CallerID (UK) TDM400P ( CID timed out waiting for ring )
12:38PM 0 1.4 Beta 3 H323 Video?
11:58AM 9 "Fixing the Caller-ID Problem", by John Todd for O'ReillyNet
11:05AM 4 Voicemail help
10:32AM 2 (no subject)
10:28AM 5 need help using tftp for polycom 501
10:24AM 0 txfax only getting 1 page of 3.
8:21AM 0 misdn.conf: how to set music on hold
8:16AM 1 Resampling Audio for use with Asterisk
8:06AM 1 Distributing calls among channels in dial group
6:51AM 1 voicemail idea and a question
6:44AM 5 Dynamic Codec Selection
5:22AM 0 something about Agent Transfer
5:13AM 6 Becoming a User on IRC
4:29AM 0 CDR_DISPOSITION_FAILED - Call has been answered correctly
1:41AM 0 Core dumps when Releasing clone lock
12:54AM 0 newbie astdb error, please help
12:24AM 0 mgcp registration with asterisk
12:22AM 4 UA - number assignment
 
Monday October 23 2006
TimeRepliesSubject
11:25PM 4 T.38 faxing with spandsp and Grandstream HT.486
7:54PM 0 Callmanager 3.3(5) and Asterisk with ooh323 problem
6:09PM 3 Asterisk conferencing features
6:01PM 6 Digium vs. Sangoma
5:06PM 1 make menuselect question- Module Embedding
3:51PM 10 Polycom SP4000 ftp problem
2:08PM 0 Multiple line phones with different contexts
1:48PM 5 Where to best start looking for voicemail/moh sound quality problem?
1:43PM 0 CBeyond SIP
1:30PM 10 Polycom provision errors still! Arg!
1:21PM 1 Question on one-way-audio with IAX
1:08PM 10 Asterisk and dialer Running on Thin Clients
1:03PM 0 REQ: Astricon Pictures
12:59PM 5 One way audio half way through call
11:24AM 0 call file mechanism
10:08AM 4 INVAL Messages
8:54AM 4 asterisk and HMP
8:49AM 3 asterisk not detecting hangup
8:01AM 2 Macro 'exited non-zero'
7:55AM 0 How to busy out PRI channels?
7:07AM 2 chan_h323.so Asterisk Beta compilation
6:58AM 0 7960/SIP MWI Question
6:52AM 1 Primary D-Channel & channal numbers....
5:55AM 0 (no subject)
5:45AM 21 Problems with chan-capi and Eicon Diva 4BRI
5:33AM 0 Real Time and Asterisk
5:32AM 0 (no subject)
4:52AM 2 Zap Channel and VM problem
4:33AM 3 spandsp and freebsd
4:26AM 0 SIP_HEADER function; what names are available?
3:34AM 1 astdb error, please help
1:45AM 4 Unicall Installation
1:23AM 0 Can anyone help? Why does One-Touch record mute/disconnect callif not dialed quick enough?
1:12AM 0 Primary D-Channel on span 2 down
12:58AM 5 Why does it take at least 4 flipping days before asterisk tries to resolve a provider?
12:24AM 0 Compiling H323 channel Asterisk 1.4.Beta3?
 
Sunday October 22 2006
TimeRepliesSubject
9:01PM 2 asterisk guru needed for job in Chicago area
7:35PM 2 How to deploy a PBX in such a condition ?
5:16PM 22 checking 'voicemail" externally - doesn't work
3:47PM 9 Audiocodes MP-20x
11:07AM 2 [SOLVED] 1.2.12.1 crashing
7:14AM 3 G.729 operating on outgoing only
5:29AM 2 Using variable as a context extension ?
3:03AM 1 new g.729a codecs for asterisk 1.2/1.4 and glibc
 
Saturday October 21 2006
TimeRepliesSubject
6:57PM 0 AGI Help
11:16AM 10 Unique call ID's across several systems
10:18AM 0 forward several times
9:59AM 0 Using the ZOOM 5801 ATA with Asterisk
9:08AM 1 new route by caller id
7:50AM 0 Asterisk 1.4beta3 and Asterisk Manager API Action: ExtensionState
7:48AM 1 zaptel 1.2.10 make problem
6:44AM 0 route by caller id
1:57AM 13 1.4 branch on OSX?
 
Friday October 20 2006
TimeRepliesSubject
10:38PM 0 (no subject)
1:51PM 3 modprobe Ztdummy is not working
12:41PM 1 Snom 320, Queues and Transfer not working as expected with * 1.2.12.1
11:47AM 0 centos or rhel and txfax with libtiff
11:12AM 4 some transfers dropped.
11:00AM 1 Escape from Voicemail
10:01AM 2 PRI boards with g729 capable DSPs
9:16AM 15 getting DID info..
8:13AM 2 Clicking Noise on Pure Voip Calls
7:54AM 1 #Transfer - Timeout is configurable?
7:47AM 10 voicemail usernames can't begin with "j" letter?
6:59AM 2 noise gate for asterisk?
6:45AM 1 Astricon - post show Saturday?
5:30AM 0 using asterisk to do remote control
5:11AM 1 call center status viewer
5:09AM 8 Linksys PAP2 dial plan help please
4:46AM 0 Xorcom Astribank
2:59AM 8 Asterisk Realtime... Help Me!!!
2:28AM 5 using asterisk to do remote control functions
12:44AM 0 Asterisk 1.2.13 make problem
12:34AM 1 Help: Problems about console color (FC5, XTerm)
 
Thursday October 19 2006
TimeRepliesSubject
9:07PM 3 /dev/zap/channel ownership
8:31PM 1 Getting started with sample dial plans
8:26PM 5 Polycom boot error
8:23PM 0 question about asterisk txFAX
6:05PM 0 FS: Sangoma A200 10 port FXO card
2:20PM 0 DTMF logging
2:04PM 18 bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
1:21PM 0 DTMF / Silence issues
1:13PM 23 Embedded Asterisk
12:45PM 6 T1 pricing in Oz
12:13PM 6 plainvoip - down ???
10:04AM 0 Errors in console in every call made when using 1.4b3
9:45AM 1 How do I configure Asterisk if I need to run Mysql server on second Linux
9:06AM 4 Occasional one-way audio - Sangoma A101
8:57AM 17 rxfax problem
8:22AM 2 siemens hipath interoperability - PRI/Q.SIG - card recommendation
7:04AM 1 Modifying SIP Stack
5:55AM 0 Please help with these SIP errors
5:53AM 0 Multiple bridge attempts
5:41AM 3 Bristuff qozap drivers problem
4:44AM 1 Access Denied on a Windows share
3:11AM 0 RE gotoiftime and Macro question
3:00AM 3 say Asterisk to answer
2:38AM 1 Zaptel not detecting Tormenta2 PRI Interface card
2:12AM 7 wrong outgoing caller id with PRI lines: maybe usecallingpres involved?
1:55AM 0 Which is the best ?
1:27AM 0 Got reject for frame XX, retransmitting frame XX now, updating n_r!
1:02AM 0 Cisco 7970 - versionStamp
12:37AM 3 accountcode and amaflags?
12:36AM 2 SIP users with Database
 
Wednesday October 18 2006
TimeRepliesSubject
11:56PM 0 Asterisk Realtime with ODBC/MySql
10:30PM 3 Asterisk hangs up on incoming analog calls after a while
10:20PM 1 How to get the agent id in the recording filename
8:43PM 3 question about CDR command
8:37PM 0 What doe these error messages mean?
8:23PM 0 Asterisk 1.4.0-beta3 released!
8:23PM 0 Asterisk 1.2.13 released - Security Vulnerability Fix
8:23PM 0 Asterisk 1.0.12 released - Security Vulnerability Fix
7:36PM 4 Asterisk + Huawei
6:10PM 1 Speed Dials
4:36PM 0 OT: (Job) Full-Time Asterisk Opportunity
2:40PM 3 echotraining=yes in misdn.conf is invalid or out of range.
2:37PM 1 adding outbound prefix
2:31PM 1 1.4 downgrade
12:34PM 2 Sip Trunks
10:27AM 2 random one way audio and noise betweenSIP phoneson same LAN
10:03AM 4 IAX softphones
9:53AM 0 Help with fxotune
9:10AM 0 ooh323 dtmf problem
8:20AM 0 DTMF problems with legacy PBX
7:52AM 0 list down?
7:41AM 0 QueueMetrics 1.3 released today
7:28AM 1 CAPI channel not available but nobody is usingthe system
7:20AM 4 Findme problem
6:43AM 1 Polycom IP650
6:20AM 2 random one way audio and noise between SIP phoneson same LAN
5:35AM 1 Windows and file shares
5:31AM 1 Asterisk+SER help
4:52AM 4 identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M
4:44AM 2 gotoiftime and Macro question
4:44AM 3 random one way audio and noise between SIP phones on same LAN
4:15AM 0 cut ip adress from caller id number display (ci$co 7941)
3:45AM 1 Orange Flash Light Mitel 5215 - Asterisk - working !
2:34AM 1 Blank page when sending faxes (repost)
2:06AM 1 Server power indication
1:54AM 4 Digium on Dell PowerEdge 1850
1:40AM 0 IAX2 thru NAT problem
12:58AM 2 Netgear WGT Flash-fest at Astricon
12:46AM 0 AGI for authenticating calls with DTMF
12:40AM 4 Re: Is 1.2.12.1 production ready (Mauro Zanin)
12:37AM 0 Please explain these SIP errors
12:25AM 0 [OT] Nokia E60/61/70 and SIP
 
Tuesday October 17 2006
TimeRepliesSubject
8:00PM 0 Out dialing Integration
6:58PM 1 1.4 gsm files changed??
3:28PM 0 Blank page when sending faxes
2:03PM 1 CAPI channel not available but nobody is using the system
1:52PM 6 Cisco 2621 NM-HDV VWIC-1MFT1
1:47PM 0 FW: Why does One-Touch record mute/disconnect call if not dialed quick enough?
11:49AM 1 Unique ID
11:11AM 6 IVR problem
10:49AM 13 considering purchasing a t1 card, any recommendations?
10:18AM 0 lots of registrations, sip problem
10:14AM 2 Gtalk on Asterisk 1.4
9:59AM 7 Electric usage of a tdm400p
9:31AM 0 Sipura 901? Any experiences
9:14AM 2 install MAGI
8:39AM 0 sipXezphone
8:35AM 0 Authenticate application
8:26AM 18 Locking phones at night...
7:36AM 0 acami
7:02AM 0 Setting the H323 Callerid sent by asterisk (using chan_h323)
7:02AM 2 Help with Dialplan Rules Please!
6:59AM 5 duplicate "ghost" calls with long duration
6:42AM 1 Please help me!!
6:42AM 0 Dial - i parametar
6:38AM 5 what hardware and is it possible
6:19AM 3 One way audio on chan_gtalk
3:25AM 0 TIMEOUT() function missing
3:10AM 2 how to activate recording (automon)
2:31AM 8 Inaccurate CDRs
2:20AM 1 Why the MusicOnHold sound so soft?
2:18AM 1 chan_bluetooth, mobile handset as VoIP terminal?
2:09AM 3 sending sip style messages in response
1:43AM 3 how to config chanspy
1:21AM 0 How to get Linksys-Sipura error codes ?
12:59AM 1 Call Forwarding Using Asterisk
 
Monday October 16 2006
TimeRepliesSubject
11:52PM 1 1.4 Beta and oracle
11:14PM 2 nat auto detect ?
11:00PM 1 Recording from a script
9:36PM 0 Critical - No audio issue with re-invite (wrong media address)
9:09PM 2 macros reference?
9:07PM 1 1.4 beta voicemail warning
4:13PM 0 Asterisk/VOIP to PSTN (?)
4:10PM 13 Stopping putgoing calls after working hours
2:48PM 13 Is 1.2.12.1 production ready
1:30PM 3 Accessing MySQL DB to set variables in Asterisk
12:43PM 27 Why is this happening?
9:25AM 4 ZapHFC & quadBRI D-Channel going down randomly
9:04AM 0 Do you encounter this REC alarm before?
9:00AM 0 Asterisk-ooh323c Video ?
7:47AM 7 Remote UNIX connection, Remote UNIX disconnected displayed every second
7:10AM 0 Asterisk <-> Live Communications Server Integration
7:08AM 3 Reception Console
7:00AM 0 Some Warning in Asterisk for Voicemail intgreting,
6:45AM 0 Tellabs and PRI
6:01AM 1 Monitor stops recording midstream?
5:08AM 1 Multiple 'routes' to extension in different contextes. How to influence search oder?
4:58AM 0 Weird problem with beep.wav!
3:57AM 2 quality control
3:04AM 5 asterisk upgrade
2:59AM 1 Page hangs up after 5 seconds
2:58AM 0 member queue refresh
2:08AM 0 SV: How do you like TrixBox?
1:08AM 1 FOP run control for CentOS/RHEL
1:05AM 2 Quescom 400
12:49AM 4 Unable to open Asterisk database
12:07AM 8 tdm2400p question
12:06AM 0 Sipura SPA-481
 
Sunday October 15 2006
TimeRepliesSubject
10:54PM 0 chan_bluetooth - one way audio
10:13PM 6 VoicePulse Connect 4 Channel Limit?
9:32PM 0 sip agent stuck in queue even after restarts
8:57PM 1 eagi-sphinx-test how and why
3:39PM 2 detecting the receivers voicemail
12:30PM 3 SPA942 quality for a Bank
10:54AM 0 Ringtones won't work
 
Saturday October 14 2006
TimeRepliesSubject
10:53PM 22 two SIP phones as one line
8:00PM 2 Codec swap (reinvite)
7:14PM 0 New and Improved
7:09PM 1 Student Research - Asterisk H323 Video
6:04PM 1 Re: Generate Random Numbers in dialplan
5:54PM 1 Re: Centos kernel 34 vs. 42? [was: asterisk-users Digest, Vol 27, Issue 72]
4:57PM 0 Re: Generate Random Numbers in dialplan
11:29AM 0 Test to list
6:25AM 1 12 port FXx PCI card
5:27AM 0 SIP trunk from an Audiocodes mediant 1000
4:11AM 0 rxfax problem ("Trainability test failed")
 
Friday October 13 2006
TimeRepliesSubject
11:35PM 0 NAT/firewall/Asterisk/Polycom Phones
11:14PM 4 Looking for a Voicemail Lamp device
10:29PM 0 Problem in Voice Message Storing...............
9:30PM 2 Re: Generate Random Numbers in dialplan
7:57PM 3 DID failover
5:07PM 9 3way calling / codec problem
3:36PM 1 Avaya 8300 - Asterisk integration using H.323
2:57PM 1 Calls being disconnected across VPN
1:08PM 1 Go to DIGIUMBOARDS.COM
12:15PM 1 OT: Voipsupply.com phones are down. Was: how big is *your* dialplan
11:24AM 16 VoipSupply? [Semi-Urgent]
11:07AM 2 Asterisk (meetme) and SMP/HT OK?
10:58AM 4 Centos kernel 34 vs. 42?
10:51AM 1 Inhouse SIP to ZAP has echo sometimes.
10:49AM 1 Unable to create/find SIP channel for this INVITE & Broadvoice
10:19AM 1 hold drops audio
10:11AM 3 How do I figure out where this connection is coming from?
9:53AM 5 Cisco 7970 SIP won't update?
9:13AM 7 Polycom HDVoice
7:55AM 4 How big is *your* ego?
7:40AM 8 Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)
6:30AM 3 AEL Question
6:10AM 5 Switchtype,Signalling,rxwink warnings
5:26AM 3 OriginateEvent reason codes.
4:34AM 6 VoIP+RJ11 Phone existed?
4:34AM 6 Digium TE410P LED problem
2:48AM 0 Segmentation fault issue
1:17AM 0 Asterisk 1.4 / install app_bundle problems
12:21AM 0 how i can do auto dialing using mysql
 
Thursday October 12 2006
TimeRepliesSubject
11:40PM 2 Call Asterisk : It calls me backup with a dial tone
11:07PM 0 AGI scripts
10:03PM 2 Some file aren't loaded its No file in that Directory.
9:41PM 2 Voicemail prompts clipped when retrieving from some SIP phones
9:19PM 14 How do you like TrixBox?
3:45PM 1 AccountCode set in sip.conf but not showing up in CDR
3:26PM 9 OT: jobs for asterisk lovers
2:59PM 5 Attended transfer hanging PRI channel
2:58PM 3 Polycom IP 501 message light
2:53PM 0 Problem when both Proxy-Authenticate and WWW-Authenticate is required
2:11PM 1 unix sysctl config for asterisk
1:47PM 0 Codes negotiation problemsbetweenAsterisk1.4beta2 and Aastra 480i
1:03PM 8 unauthenticated calls
12:58PM 1 How to send correct Caller ID on PRI
12:17PM 2 Issues with Asterisk 1.4 Beta
11:17AM 3 Bridging of PRI calls
10:24AM 13 1.2.12.1 crashing
10:13AM 1 AstriCon Hotel Full - Here are some near-by alternates
9:58AM 6 SPA 3102
9:19AM 0 OT: BioFuel to power phone networks
7:39AM 1 Call drop and strange CDR records
5:58AM 0 Reg. chanspy
5:26AM 6 Call bridged, but no sound
5:23AM 0 prohibit CallerID presentation
4:22AM 1 Fax receive (rx fax) problem
3:23AM 2 vGSM drivers updated (0.17.2)
3:20AM 1 Urgent Billing
2:27AM 0 Beronet BN4S0 instalation
2:18AM 0 Stripping digits on internal calls
1:51AM 1 Anybody using "inphonex" service?
12:33AM 1 How to enable talking in chanspy while spying?
12:14AM 0 Asterisk -> regitration in DB
 
Wednesday October 11 2006
TimeRepliesSubject
11:02PM 9 Multiple TE110P cards in one chassis
10:41PM 0 how to setup call center with media gateway?
9:02PM 5 TDM400P incoming route for DID
7:07PM 1 cdr_addon_mysql.c - Asterisk 1.4 - Asterisk Addons
5:41PM 1 XO SIP Origination Services
5:31PM 0 Meeting
4:42PM 0 CDR Help...
2:56PM 4 Test Call Script
2:03PM 1 What alternatives to Asterisk based virtual PBX?
1:32PM 1 Echo problems on ISDN. (mainly incoming call s)
12:38PM 0 Load balance Asterisk server, when it is a SIP client.
12:26PM 1 Urgent Please help
11:54AM 0 SIP Locking Up?
11:53AM 1 Asterisk as SIP Client
11:24AM 1 Echo problems on ISDN. (mainly incoming calls)
10:25AM 1 Problem with ZAPTEL-1.4.0-beta1 and WCT100P card
10:17AM 1 max users
10:02AM 18 1.4 beta2 on intel mac
9:53AM 6 compiling libunicall
9:40AM 1 Asterisk users help
9:27AM 1 average waiting time in a queue
9:23AM 0 Segmentation fault asterisk realtime problem
9:17AM 0 Re: asterisk-users Digest, Vol 27, Issue 49
9:11AM 17 GPL Softphones
8:30AM 5 zt_chanconfig failed
8:19AM 12 asterisk 1.2.12 lost phone registrations today... why?
8:10AM 0 RE: Welcome to the "asterisk-users" mailing list
8:09AM 0 [Fwd: Re: NFAS Not Passing Audio on B-chan 48, 72, 96]
7:17AM 0 IAX2 outgoing calls delayed before they connect
7:08AM 6 Asterisk + E1 with MFC/R2 in Argentina?
6:56AM 4 cisco 7960 not registering after * restart
6:44AM 9 NFAS Not Passing Audio on B-chan 48,72,96
6:07AM 0 Digium TE405 card and Matra PBX
5:49AM 5 sending fax with chan-capi
5:37AM 0 "Guest" SIP-Invites not accepted
5:25AM 2 MGCP stuff
5:16AM 1 user address format
3:22AM 1 SIP fails when internet connection lost.
3:11AM 0 Voicemail app. not working...
3:06AM 0 Hicom 150 -- BRI -- Asterisk
1:08AM 0 Asterisk 1.4.0 compile error on AMD64 Opteron server; recompile with -fPIC?
12:14AM 8 Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again
12:00AM 11 Psst... Top secret information: Codename Pineapple
 
Tuesday October 10 2006
TimeRepliesSubject
11:54PM 6 Billing
11:06PM 0 Cubix / Firefly softphone and Asterisk
10:17PM 1 RE: Welcome to the "asterisk-users" mailing list
10:02PM 1 WRT54GP2 provisioning
8:25PM 8 E164 caller ID
6:05PM 2 how can I detect a DTMF tone while on a bridged call ? anyone knows?
4:35PM 0 asterisk crash in res_features.c
3:42PM 10 Strange FXS disconnection problem.
3:13PM 4 Free copy of "TrixBox Made Easy"
2:58PM 2 Change the background of a conversation
2:07PM 1 voicemail issue
2:03PM 10 Understanding NAT Traversal
1:46PM 0 Destar 0.2.0 released
1:42PM 1 Looking for AudioCodes 2 port FXS gateway - Asterisk compatability info
1:17PM 68 How big is *your* dialplan??
1:07PM 16 RE: Welcome to the "asterisk-users" mailing list
12:34PM 2 Hangup or busy when the peer answer outgoing calls
12:17PM 10 Increase VoiceMail Messages Recording Gain - Audio Calls are Ok
11:45AM 3 transfer from VM to Cell Phone
11:32AM 1 1.4 and slow sound playback
10:33AM 8 Cisco CCM - Asterisk
10:28AM 4 sequential Dial() commands
10:20AM 1 Mitel 5224/SIP no MWI
10:16AM 3 Connection question...
9:38AM 0 bristuff problem?
8:30AM 17 Voicemail Press '0'
8:26AM 1 Asterisk 1.2.12.1 and snom 360 6.2.3 no audio
8:19AM 0 Xorcom TS-1 and Digium TE110P or TE210P
8:01AM 5 whisper paging
7:43AM 4 Inbound Callcenter with multiple DIDs
7:28AM 0 FYI - Polycom SoundPoint IP 301 Denial of Service]
5:03AM 4 alive check for HA constellation
4:08AM 0 Tutorial: Simple queue and agent debug monitoring
4:07AM 1 help this....
3:37AM 8 single conference, multiple numbers
1:33AM 2 OT: Hand free solution recommandation
 
Monday October 9 2006
TimeRepliesSubject
7:18PM 3 T1 Passthrough
5:41PM 11 Echo Cancel Cards
5:28PM 4 Monitor Current outgoing calls
4:57PM 3 Error loading Unicall
4:17PM 0 ooh323 error
3:53PM 4 Home Hardware SIP Proxy with use of POTS in Emergency
3:30PM 1 Problem compiling libmfcr2.0.0.2 on Fedora Core 5
3:27PM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday October 14th 2006 - 10:30am
2:13PM 1 Function ENUMLOOKUP
2:00PM 1 Asterisk RT on Disk On Module PerformanceandDurability
1:41PM 4 how to play pre-recorded file in meetme conference
1:19PM 3 Asterisk RT on Disk On Module Performance andDurability
12:51PM 0 problem in ooh323
11:23AM 1 Cisco 7970 Unbootable After FW Upgrade
10:47AM 1 GotoIfTime - much slowdown with 90 conditions?
10:47AM 0 External domain
10:30AM 0 Is there a way to collect dtmf digits during a call? (inband)
9:59AM 2 Number Range
9:43AM 4 Range Operator
9:37AM 3 connecting multiple servers with iax - authentication fails
8:52AM 0 PRI TON/pridialplan digit prefixing
8:30AM 3 VOIP with PSTN backup
8:24AM 5 isdn cross-over ...
7:54AM 0 Get user context from dialplan.
7:35AM 0 how to play background music
7:34AM 6 Lots and lots of log files
6:49AM 1 AstriCon Dallas in Two Weeks
6:36AM 0 H323 <-> SIP
3:49AM 1 SIP vz IAX...
2:35AM 0 Beronet card strange log messages
1:20AM 1 Redefinition of transfer
12:57AM 10 Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate
 
Sunday October 8 2006
TimeRepliesSubject
10:23PM 4 password for vm users
10:04PM 2 Polycom 601 & Expansion Module: Light the LEDs???
9:07PM 8 CDR - mysql with asterisk 1.2.12 not working
8:58PM 5 question about astdb
8:40PM 1 Asterisk Load balancing
8:30PM 2 How to make this easier
8:20PM 0 TDM22B
7:42PM 2 polycom reboot script
7:01PM 0 SIP vs. SIP-B
6:45PM 1 Blacklist to check http://whocalled.us
9:48AM 3 Tellabs and a PRI
9:37AM 14 PRI issues
9:13AM 0 USA Origination recommended service?
7:44AM 6 DID is not working (call is not routing)
6:27AM 0 OH323 Fake ring
2:54AM 0 Sun Cluster and Asterisk
2:02AM 0 Transfer app and DTMF via SIP info
1:05AM 2 How can i store PAP2 or any device config in Asterisk
 
Saturday October 7 2006
TimeRepliesSubject
10:30PM 0 polycom auto cfg file
10:30PM 18 ftp server
8:46PM 3 Real-time and priority "n"
6:37PM 0 Asterisk: Can anybody forward anybody's extension?
3:30PM 2 disabling hardware echo can on tdm2400p
2:15PM 2 Xorcom Astribank and 64 bit linux
11:33AM 1 Requirements for Asterisk & SER integration
9:20AM 2 G729 Licence Consumption Problem
7:51AM 3 Outbound FXO call, getting "You must first dial..."
5:48AM 1 AEL2 Catching on?
1:51AM 9 SIP stuck channel soft hangup?
 
Friday October 6 2006
TimeRepliesSubject
10:39PM 3 astcc help-pleasssssseeee
9:40PM 2 Options for moving to * friendly Business VSP
6:13PM 2 HTTP Connection Closed on 7960 SIP
5:40PM 2 A Call centre module on Asterisk
5:21PM 1 Asterisk access Postgres for Realtime Configuration
5:16PM 6 regexten & regcontext broken for SIP?
5:11PM 0 commercial asterisk
4:04PM 0 Codes negotiation problems betweenAsterisk1.4beta2 and Aastra 480i
3:27PM 4 Odd echo issue with speaker phone
2:03PM 3 ChanIsAvail() in 1.2.12.1
1:43PM 1 swap CID with DID
1:27PM 2 AGI() in 1.2 and 1.4
12:38PM 0 Asterisk Postgres Native support
10:49AM 2 Voicemail MWI
10:46AM 0 Match & Chat Author?
9:54AM 10 Asterisk RT on Disk On Module Performance and Durability
8:26AM 2 Voicemail and Forwarding
8:07AM 0 defining trunks in sip.conf
8:02AM 1 Tutorial - avoiding queue_log file rotation
4:19AM 1 Asterisk act as a proxy ?
4:07AM 2 How to forward DID to another Server
3:46AM 2 2x* and realtime
2:39AM 3 New Asterisk StumbleUpon Group
1:55AM 1 asterisk gui sans live cd
 
Thursday October 5 2006
TimeRepliesSubject
11:46PM 1 Asterisk CDR
11:29PM 6 Asterisk Server : IDE HDD frequent crash
8:00PM 0 different dialtones for DISA
7:22PM 3 Newbie h/w Q, and confirming basic concepts
7:01PM 3 Optus PRI via DSL
6:31PM 13 No Dialtone
6:30PM 1 odd muting issue
5:04PM 3 failed registration
4:30PM 1 IVR menu system external database information collection
4:24PM 11 Asterisk@Home problems
3:41PM 1 DOA IAXy?
3:07PM 1 AW: PoE IP Phone
2:26PM 1 How to create echo for learning purpose
2:05PM 0 AGI PHP
1:54PM 0 ACK when using SIP Proxy
1:15PM 3 Codes negotiation problems between Asterisk 1.4beta2 and Aastra 480i
1:06PM 0 Dial without phone
11:58AM 2 No voice for when using Playback and background
11:56AM 4 Problem with 2 machines connected with IAX
11:53AM 0 Help with gdb bt full results
11:38AM 12 PoE IP Phone
11:26AM 5 OT: Polycom time sync - sorta
10:42AM 5 pop a web page with DID in url
10:24AM 0 we are having trouble detecting the # for making a transfer from an E1, usually under some load, please help
10:16AM 3 Getting Asterisk to work with GoogleTalk
9:22AM 0 sdsl
8:17AM 0 Dial out trhough a FXS channel on a TDM card
7:54AM 0 Re:UPDATE: Zaptel problems
7:19AM 0 Detecting Busy AGI Extensions?
7:11AM 3 AW: asterisk-users Digest, Vol 27, Issue 23
7:10AM 0 for god's sake somebody help me! ANSWEREDTIME=0 in astcc!!
7:08AM 5 Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
6:51AM 3 two asterisk and one NBX system
5:58AM 0 GXP - 2000 BLF
5:55AM 4 Call Center requirements
5:35AM 0 India:Reliance - E1configuration using TE110P
5:27AM 1 Extremely choppy sound on some of our POTSnetwork calls; goes away with mute
4:43AM 0 New Version of "Tycho" Voicemail Manager released
2:47AM 1 Message count requested for mailbox 9002@from-sip but voicemail not loaded.
2:14AM 1 Problems with Dial In - Dial Out via SIP - no voice
1:51AM 0 Silience on random calls
1:47AM 0 spandsp logs?
12:54AM 4 "set verbose 4" in SVN trunk?
 
Wednesday October 4 2006
TimeRepliesSubject
11:56PM 1 fonality acquires trixbox (asterisk@home) ?
11:25PM 2 asterisk-addons-1.2.4 Installation Problem
10:35PM 1 CDR problem with call transfer
10:21PM 0 echo cancellation on hard phones
10:03PM 1 AEL2 #include madness in Asterisk 1.4 - Murf?
5:53PM 1 Clipcomm CG 410 / FXO Gateway
5:38PM 1 ZAP chanel doesn't reset if external caller hangs up in menu
5:03PM 0 Possible to set max voicemail mesasge limit per user
4:51PM 15 Bandwidth requirements
4:11PM 0 snom 360 - how to make record button working ?
3:39PM 1 Re: [asteisk-users]USA DID + trunk
2:58PM 5 no callerid from PSTN using TDM2400P
2:49PM 3 Calling Functions from AEL2
2:43PM 2 MODEM (data) througt asterisk ?
2:42PM 3 How to make RTP does not go thru asterisk server
1:18PM 11 TNT Max Password reset
1:17PM 3 Voicemail maintenance
12:27PM 0 Intrado V9-1-1
11:27AM 4 Newbie question about meetme
11:24AM 5 Dialplan Syslog
11:01AM 2 Need USA DID + trunk provider
10:57AM 0 Intel Chipset 945p compatible?
10:37AM 3 Wouldn't Tri-tone detection in Dial() be cool?
10:35AM 4 SIP client that runs on Linux or Solaris through X Windows?
10:33AM 16 snom 360: how to make record button working ?
10:10AM 6 New tutorial - peering two * servers using IAX
10:01AM 10 Transfer feature - howto?
9:53AM 1 digium compatibility notes
9:36AM 0 Asterisk 1.4 moh - mohsuggest
8:50AM 0 FOP v.27 IAX trunks not "ringing"
8:32AM 4 [Asterisk-Java] SipShowPeerAction
7:05AM 21 Where is the PlayDTMF command?
6:53AM 4 Spandsp and tif
4:45AM 2 voicemail maintenance questions
4:11AM 1 Help in MySQL + Asterisk.
3:44AM 2 verbose logging to file in 1.4
3:16AM 0 Rejecting call
2:32AM 0 Asterisk and Attachment
12:43AM 4 DISA and legacy PBX
 
Tuesday October 3 2006
TimeRepliesSubject
11:56PM 2 Call Forwarding not working for extension in queue, why?
10:51PM 1 Strange problem(Munin-node-1.2.4-7)
8:22PM 1 (no subject)
3:30PM 2 Digium Interfaces in Tampa?
3:24PM 1 Asterisk Directory listing
3:18PM 1 authenticating forwarded calls
3:11PM 0 AstmanProxy Not Collating Manager Info
2:09PM 1 CDR stats to one mysql database, multiple webstats packages
1:23PM 5 Digium TDM or SPA-3000?
11:32AM 2 Extremely choppy sound on some of our POTS network calls; goes away with mute
11:01AM 2 uniden uip200 phone hangs any ideas?
10:13AM 17 CALEA support within asterisk?
9:19AM 0 Asterisk+Panasonic KX-TDA100+zaphfc NT link problem
8:58AM 0 o extension for voicemail app
8:55AM 6 asterisk to asterisk DID extentions
8:21AM 0 Cisco 7961 - Presence Example?
8:10AM 2 Caller ID on Zap not always working
7:40AM 1 R: Zaphfc woth florz patch
6:30AM 4 SV: Screen pop based on incoming DID
6:27AM 0 Query on Call Parking
6:20AM 3 Problems with automon
6:05AM 0 Zaphfc woth florz patch
5:43AM 2 Screen pop based on incoming DID
5:39AM 0 How to add new codec support?
4:30AM 1 Defining sip users through mysql
4:02AM 1 sip provider not working
3:53AM 3 Which IP Phone is good to use at reception desk?
3:26AM 0 realtimeupdate error
2:07AM 0 pbx call setup to asterisk, behavior context dependend
1:40AM 0 [ast-users] bridging active channels together
1:39AM 0 ZyXEL desktop ethernet switch for QoS
 
Monday October 2 2006
TimeRepliesSubject
9:30PM 1 tools/techniques/metrics for measurement of end-point quality
8:59PM 4 TDM2400P wiring.
5:01PM 15 Polycom Buddy Watch Broken with 2.0.1 Firmware?
3:12PM 0 Trunks and Outbound Routes
2:33PM 0 Problems with Tormenta 2 quad card
2:01PM 3 Configuration / dialplan problem
12:58PM 0 Minexpiry time - how to set this
11:37AM 2 Passing Arguments to FastAGI
11:03AM 1 g729 Codec for AMD Sempron
10:24AM 0 Conversations Mix
10:13AM 0 !! No channel map, no channel, and no ds1? What am I supposed to identify?
9:54AM 6 t1 voip to failover pri
8:47AM 0 Asterisk 1.2.10 and SCCP
8:32AM 0 480i phone: Is there a trick to registering with *?? <--Solved, first impressions
8:12AM 0 480i phone: Is there a trick to registering with *?? <--Solved
7:21AM 0 asterisk queues with SER, aka "sip show peers"
6:50AM 2 Issues with calling certain phone numbers...
6:46AM 0 Dial and connect to sip provider works, but no audio.
5:06AM 2 attended transfer unreliable (2nd try)
4:23AM 3 Spying a channel in a meetme
4:04AM 2 can't transcode ilbc
3:45AM 1 Call Quality / Echo / Problems
3:19AM 0 Can't get second line of Sangoma A200 to work.
1:57AM 0 suggest a configuration
1:03AM 7 Siemens Hipath <-> asterisk, pri problem
12:46AM 0 I: Sip answer one side , ring other side
 
Sunday October 1 2006
TimeRepliesSubject
11:56PM 1 Bristuff vs. vISDN vs. mISDN for hfc card ?
10:31PM 0 G729 Codec Loading Error
8:33PM 2 G726 prompts
5:14PM 0 polycomphone (SOLVED)
3:15PM 0 Another Issue with 1.4
8:32AM 5 detecting busy on queue transfer
5:28AM 5 WiFi SIP handset with Bluetooth required
4:27AM 1 polycomphone
1:39AM 0 Real-time LDAP config