asterisk users - Apr 2007

Monday April 30 2007
TimeRepliesSubject
7:59PM 0 Segfaults in 1.2.18
6:08PM 1 Re: Voicemail on Different Server (MySQL Replication split thread)
5:14PM 5 Zaptel kernel module load order
4:05PM 6 Asterisk 1.4.4 VoiceMail ODBC Storage Help
1:51PM 11 Improving Asterisk's DNS support
11:38AM 3 CDR and Billing Issue
11:33AM 1 IVR dictionary dial-plan
10:37AM 1 automatically close a meetme
10:30AM 2 Confference function
10:25AM 6 ZAPTEL PROBLEM
10:13AM 2 Send Variable in Dial
9:38AM 2 Simple dial plan inquiry
8:22AM 0 voicemail + Dynamic mailbox
6:52AM 0 Remodified Asterisk brute force blockers..
5:29AM 13 SugarCRM, NO!, Foxpro, SI?
4:25AM 2 TDM400P and Junghanns QuadBRI issue
3:47AM 2 don't want call to get answered
2:56AM 0 Priority in ACD
 
Sunday April 29 2007
TimeRepliesSubject
7:17PM 2 Polycom 650
12:57PM 2 100 users - voip lan security and qos ?
10:08AM 0 asterisk 1.4 and zap channel flash
9:26AM 2 Early audio(progress) and MOH
6:00AM 7 Polycom 430 , 501 and 550
5:34AM 0 Unable to find a codec translation path from ilbcto ulaw
5:29AM 4 Wildcard TDM11B & Wildcard TDM04B
3:52AM 1 Voicemail Creation
 
Saturday April 28 2007
TimeRepliesSubject
10:08PM 0 app_dictate problems
2:22PM 17 Poor man's High Availability solution
1:48PM 1 Viable using purchasing sip lines
10:33AM 4 Trixbox/FreePBX
8:22AM 13 ADSL routers with integrated SIP QoS for other devices
1:05AM 7 Two Connected Servers Sound Quailty
 
Friday April 27 2007
TimeRepliesSubject
9:21PM 2 Music on Hold issue with asterisk 1.4.2
5:22PM 2 Call Pick Up
3:56PM 0 Asterisk 1.4.4 Released
1:18PM 0 chan_bluetooth as FXS?
12:36PM 1 execute commands after hangup
11:55AM 3 Free seating Agents and logged in / logged out indication
11:00AM 5 Fixed quantity calls per extension
10:59AM 2 New VICIDIAL astGUIclient Release: 2.0.3
10:34AM 0 Problems with Digium TE110P
10:28AM 0 Re: Voicemail on Different Server, Voicemail with NFS
9:59AM 0 Live conference call on now
9:58AM 1 SIP<->H323 calls without proxying RTP
9:37AM 2 Problem of configuring musiconhold.conf file
9:07AM 7 Unable to find a codec translation path from ilbc to ulaw
8:35AM 0 zaptel/pri, early audio, dial()
8:32AM 7 CDR changes in 1.4.3?
8:01AM 0 Utilisation of multiple database tables in Asterisk
7:24AM 9 Best Wireless bridge for Polycoms
6:36AM 2 How to configure a stun server for a sip peer
5:55AM 1 Asterisk hosted Callwaiting???
1:17AM 1 2 cards in a server
1:09AM 1 canĀ“t anserd the call
12:29AM 0 Attended Transfer of a queue call fails
 
Thursday April 26 2007
TimeRepliesSubject
7:02PM 0 E1 Card recommendation
6:52PM 5 Asterisk 1.2.14 will not run without internet connection
5:31PM 1 Re: Voicemail on Different Server, Voicemail with NFS
2:40PM 0 FYI - PRS fraud
2:10PM 3 7970 sip success
1:38PM 2 How does Realtime read config files?
1:20PM 2 Call prority (QUEUE_PRO) in the queues
1:10PM 0 Static in Audio PRI, Got reject for frame 39, retransmitting frame 39 now, updating n_r!
12:52PM 0 problem with A400P01 OpenVox
12:01PM 0 Potential Consulting Opportunity
11:48AM 1 AsteriskNOW generation of zapata.conf file
10:45AM 1 IVR Testing
10:44AM 1 App Read() kills call when file doesn't exist
10:07AM 2 Asterisk cookbook
9:18AM 3 Two devices registrating same extension
8:46AM 5 MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped
8:30AM 3 [OT] How to find out line that you are on from Bezeq
8:20AM 2 Changing Voice from Male to Female
7:59AM 1 Asterisk in support enviroment, some CTI maybe & RT integration?
7:51AM 1 Asterisk IVR and Call Center Agents
7:13AM 0 take-over && transfer question
7:01AM 3 Can asterisk record the duration of users putting on hold?
6:47AM 0 FW: ChanSpy and MeetMe
6:38AM 1 Asterisk Voice sound level
6:37AM 2 asterisk slows down when unplugging internet cable with VoIP lines
5:55AM 16 headsets for linksys/sipura phones?
5:44AM 1 Asterisk brute force watcher (was FYI)
3:59AM 0 Asterisk IAX Configuration Problem
2:21AM 1 Cisco 7920 sccp
2:21AM 2 analog line cards / adapter
1:00AM 0 Open Source Softphone
12:21AM 0 IAX channel unreliable with multiple hops
 
Wednesday April 25 2007
TimeRepliesSubject
8:09PM 2 Asterisk 1.4 Conference with G.722
6:58PM 13 Too many open files, asterisk crash
5:59PM 2 asterisk answering machine
3:13PM 3 No Audio with SIP to only one provider when switching servers
3:12PM 4 call dispatching - legacy application
1:36PM 3 Polycom Provisioning Problems
12:07PM 1 prob with install on ubuntu linux
11:51AM 0 Error compiling Zaptel on CentOS 5
10:56AM 0 Problems to transfer calls when it is ringing
10:53AM 7 How to check my voice mail from outside landline?
10:34AM 0 ZOOM 5806 ATA
10:20AM 0 Zaptel 1.4.2.1 Released
10:19AM 0 Zaptel 1.2.17.1 Released
10:18AM 2 Asterisk-addons 1.4.1 Released
10:17AM 0 Asterisk-addons 1.2.6 Released
10:16AM 0 Asterisk 1.4.3 Released
10:15AM 0 Asterisk 1.2.18 Released
9:58AM 4 Asterisk 1.4.3 segfaults on receiving calls.
8:55AM 12 FYI
8:47AM 9 SLA Appearance between 2 Cisco 7960's (SIP)
7:23AM 0 Call Parking is slow with park orbit on Snom 3xx / 360
7:02AM 5 My Polycom IP 501 is formatted its file systemitself
5:03AM 2 agi and transfer
4:21AM 8 dialplan / problem with extension-length > 1
3:49AM 0 Asterisk Users Conference Friday 12:30 PM EDT
2:44AM 7 Asterisk Business Edition Question
2:40AM 0 OriginateResponse 'reason' property.
12:57AM 2 Problem with SuSe 10.0 and zaptel 1.2.17
12:27AM 1 Calllog
12:20AM 6 German voiceprompts for 1.4 available
 
Tuesday April 24 2007
TimeRepliesSubject
10:52PM 6 My Polycom IP 501 is formatted its file system itself
9:44PM 0 7970 Config Question
8:45PM 0 Re: agi timeout......clarification
6:17PM 3 Funky BIND/named errors
5:28PM 22 Voicemail on Different Server
5:25PM 15 Asterisk & Pix firewalls
5:20PM 1 E&M Wink start problem
4:24PM 0 ASA-2007-012: Remote Crash Vulnerability in Manager Interface
4:24PM 6 Random Asterisk deaths
4:22PM 0 Queue: SIP status not set to busy
4:22PM 0 ASA-2007-011: Multiple problems in SIP channel parser handling response codes
4:21PM 0 ASA-2007-010: Two stack buffer overflows in SIP channel's T.38 SDP parsing code
4:09PM 0 Asterisk Project Security Adivsory Process
3:49PM 0 app_dictate playback problems
2:47PM 0 agi timeout
2:43PM 1 SER/OpenSER, I Finally Get It.............General Observation
2:14PM 5 dundi problem * 1.4.2
1:47PM 11 Marketing 101
12:13PM 0 Free agent while are waiting calls
10:29AM 0 agentcallback login kicking agents out after call completion.
9:35AM 0 7960G + Asterisk auto attendant
9:33AM 6 Make an iso image or a kickstart-Really its too urgent
9:24AM 3 Call Connection Problem
9:23AM 1 TE412P (T1/E1+DSP) digium card cause server crash
8:51AM 8 Polycom SP 601 Reboot Issue- Help!
8:33AM 0 AstLinux 0.4.5 released
7:50AM 0 can't cancel call conference when invited by asterisk
7:44AM 0 Snom 360 Caller ID in missed / recieved calls
7:41AM 0 3 way calls and meetme problem
6:50AM 10 Digium card sale
6:46AM 6 tone generation
4:50AM 0 Hylafax EE and T.38
3:32AM 6 auto dial out multiple destinations
2:43AM 2 Asterisk Problem
2:00AM 2 ast 1.2.x - cisco 7970 behind nat to external asterisk with no nat
1:21AM 6 help please
 
Monday April 23 2007
TimeRepliesSubject
11:14PM 2 Request for Configration details
6:37PM 1 problem when using Dial(Local/extension@context)
6:32PM 2 auto load error in asterisk cli
5:43PM 2 Linking asterisk servers
4:57PM 1 Trixbox 2 and MFC/R2
1:53PM 0 Crackly Prompts but Voice OK
1:47PM 7 Missing "dialplan" commands in Asterisk 1.4.2 CLI
1:37PM 8 Purchasing a Sangoma A102 - should I get the hw echo cancellation or not?
12:38PM 0 End User guide
10:40AM 5 ztdummy
9:01AM 3 echo cancellation and ztdummy
7:22AM 0 Hardware Compatibility list
7:11AM 1 Make an iso image or a kickstart
6:29AM 1 Asterisk+mISDN drops calls after 3-4 secs
6:25AM 1 A400P01 from OpenVox
6:20AM 2 Billion ISDN problem
6:09AM 4 Asterisk codecs retranslation
6:06AM 12 SIP devices with packet loss tolerance
6:02AM 1 polycom boot server...
5:28AM 5 chan_zap not compiling.
5:17AM 0 Pass-thru
4:57AM 1 problem with 3-way conferenicing
3:30AM 11 Asterisk on Debian Etch
3:00AM 1 app_rxfax produces "RTP: Received packet with bad UDP checksum"
2:16AM 7 Asterisk dialing next extension only if first is busy?
2:12AM 1 Internet gateway problem
2:05AM 1 Microsoft Dynamics CRM 3.0 Integration with Asterisk
 
Sunday April 22 2007
TimeRepliesSubject
8:16PM 13 Asterisk & M$ SQL Server
7:44PM 0 Kvin's g.723-gcc4 and asterisk 1.4.1
4:49PM 3 SLES?
1:52PM 1 Extension and language for users/registered ends
1:49PM 0 Re: asterisk-users Digest, Vol 33, Issue 102
11:57AM 3 Digium h/w serial numbers
3:24AM 1 Exten Length
12:41AM 0 Incoming SIP callerid
 
Saturday April 21 2007
TimeRepliesSubject
10:54AM 2 Apple IPhone mobile is released in India?
10:50AM 2 Asterisk config for Apple IPhone
8:00AM 5 UK zaptel and zapata.conf for TDM400P
7:27AM 7 Transer calls hitting #
2:36AM 5 FAX on PRI and TE205P
12:52AM 0 hints howto reset if wrong subscription Asterisk 1.2
 
Friday April 20 2007
TimeRepliesSubject
5:21PM 1 Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemailto text translation)
3:22PM 3 why do I get this message
2:43PM 6 Developing Marketing materials ...
2:33PM 7 Queue problems
2:07PM 2 Big trouble with zap lines
1:59PM 0 FW: Asterisk & PiX devices
1:58PM 2 Asterisk & PiX devices
1:14PM 0 7921G running linux
12:24PM 1 VPM450: Not Present
11:01AM 17 How can I improve call quality?
10:27AM 1 Polycom Phones
8:58AM 0 Call queue problem
8:48AM 57 Softphone that supports central provisioning?
8:26AM 1 adding second TDM400P card causes echo cancellation to fail for all Zap channels
8:25AM 1 iaxComm problems
8:10AM 0 Pika boards - anyone are using it?
7:50AM 0 Polycom not picking up phone transferred phone call.
7:26AM 2 Why duoble digits must be so fast to activate features?
7:24AM 0 Agents.conf feature replication using addqueuemember
6:57AM 1 CallerID Auth
6:28AM 2 Asterisk stops responding to SIP/ZAP
6:07AM 6 Passive E1 Pri Tap for Voice Recording
5:59AM 3 G.729 & Voicemail
5:37AM 3 pci 2.2 - pci-e x16
3:50AM 0 RAD IPmux8
1:41AM 0 Friday April 20th Asterisk Users Conference at 12:30PM EDT
12:54AM 0 app_voicemail.c
 
Thursday April 19 2007
TimeRepliesSubject
10:31PM 2 Cell phone that can be connected to standad phone switch network
9:52PM 2 Newbie Question about E1
8:52PM 0 SLA with SIP only configuration
7:38PM 3 Failed to authenticate on INVITE
4:31PM 2 users.conf SIP registration fails
4:26PM 2 MySQL Update from exten
2:44PM 5 3rd T1 of quad card won't change signaling
2:23PM 5 Polycom SIP Phones On LAN can't register without WAN (Internet) Access
2:15PM 3 Problem with TDM2400 and Polycom 501... Voice Quality Lost...
1:38PM 0 CLI Dialplan options...
12:41PM 3 CallerID masking
12:36PM 1 Asterisk 1.4.2 connection to Nortel CS1000M -followup with log
12:07PM 1 Asterisk 1.4.2 connection to Nortel CS1000M
11:47AM 5 aastra phones with asterisk 1.2.17 - hangup after 20 seconds
10:40AM 2 Setup Asterisk configuration
10:02AM 1 Asterisk 1.2 and mixmonitor stopping short
9:29AM 2 AudioCodes MP-104 MGCP?
9:28AM 14 ztdummy does not load properly at server startup
9:12AM 1 CDR(dst) != CALLERID(dnid)
8:49AM 2 Asterisk - Cisco Call Manager Express Trunk
7:51AM 2 Dial plans
7:48AM 3 Outgoing CallerID
6:47AM 1 Ser as IVR
5:33AM 1 Asterisk Queue Call Transfer
5:03AM 5 Hardware suggestions for 8-10 lines in the UK
4:17AM 13 Polycom IP 501 is displaying wrong time
3:56AM 2 extensions.conf #include behaviour
3:29AM 0 any format
2:34AM 1 Help Astertest - Asterisk stressing tool
2:25AM 0 DTMF issues
2:08AM 19 ZT_CHANCONFIG failed on channel 1: No such device or address
1:47AM 4 SIP kpml DTMF support in *
12:28AM 9 Improve voice quality on Asterisk + chan_capi + DIVA BRI
 
Wednesday April 18 2007
TimeRepliesSubject
11:24PM 1 Re: asterisk-users Digest, Vol 33, Issue 80
8:33PM 8 Trigger for unavailable SIP peer
3:38PM 1 Timestamp in recorded calls filename
3:08PM 14 [OT] OMG Verizon is terrible
3:02PM 1 Monitor application inestability and high load
2:57PM 0 IM
2:47PM 10 MeetMe Error
2:36PM 13 Asterisk 1.4.2 + Cisco 7960G not registering
2:21PM 0 Audio playback problems with FC6 and Zaptel 1.2.16
2:11PM 16 RE: OT (a little): IPV6 Ramifications Article
2:01PM 3 Segmentation Fault
12:26PM 5 gxp2000 expansion module blf leds not working
10:18AM 0 Dial out from AGI and then connect it to another dialled out call
9:55AM 0 Asterisk COLP (COnnected Line Presentation)
9:14AM 5 incoming SIP call
7:57AM 4 asterisk svn and zaptel
5:55AM 3 asterisk unable to create files, too many files open
5:51AM 0 QueueMetrics 1.3.4 released today
5:47AM 0 Phones working with 1.2.17, not with 1.4.2
5:30AM 0 Reminder: HITBSecConf2007 - Malaysia: Call for Papers closing in 2 weeks
3:28AM 14 Feedback on Linksys SPA-921 and GrandStream GXP-2000
12:04AM 5 SIP failover between Sip Providers
 
Tuesday April 17 2007
TimeRepliesSubject
11:08PM 1 Asterisk Billing
8:52PM 4 Can I add distinctive ring with asterisk and TDM400?
8:37PM 1 No Incoming Ring Tone (Even with "r" option)
5:57PM 0 Re: [Asterisk-java-users] asterisk-java.org up again with bonus article on Local/ channels
5:55PM 1 TM Malaysia E1 PRI signaling
5:29PM 0 Change to Sip in a phonecall, when the user registers meanwhile
5:07PM 0 Question regrading IAX
4:53PM 2 queues
3:17PM 4 Transfercapability DIGITAL
2:46PM 5 CDR datasets
2:32PM 5 Querying channel variables via the Manager API
2:29PM 11 internal sounds of asterisk / freePBX
2:03PM 10 Asterisk 1.2.16 - No Caller ID
1:49PM 1 Default lenguage
1:25PM 5 Connection between Asterisk - Cisco 2851
11:44AM 1 What are service activation codes ?
10:56AM 0 TDM24 Cards
10:17AM 3 Transfer via CTI
10:06AM 2 Trigger a wake-up call from the shell?
9:47AM 2 Voicemail files permission
8:39AM 6 peers are using wrong contexts
8:09AM 2 PRAM
7:10AM 2 Whats this about!
6:38AM 0 Queue report statistics
6:09AM 4 Using meetme like call
2:03AM 0 GETVARIABLE and IAX
12:54AM 12 No of Calls
12:40AM 14 sending an SMS via Asterisk?
12:17AM 1 Make an iso image
 
Monday April 16 2007
TimeRepliesSubject
11:50PM 1 Asterisk VS Cisco,Avaya,Siemens,....
8:16PM 2 Having trouble figuring this out...
8:14PM 4 [OT] Nokia E60 firmware update break SIP
6:47PM 3 Zaptel problems in Fedora 6
5:16PM 1 Recommended hardware
5:07PM 0 Re: Dial outbound trunk numbers in a round-robin sequence?
3:53PM 6 Problem with queue
3:18PM 3 Problems with queue announcements under high call volumes
2:36PM 3 Audio Problems - Operating System??
1:43PM 3 ${CALLERIDNUM} DEPRECATED in 1.4.2
1:20PM 1 Stuck on MySQL UPDATE
12:10PM 0 IAX implementation question
11:27AM 3 Redundant * servers
10:53AM 7 BSNL caller ID (India)
10:50AM 0 "jittershrinkrate" equivalent in current (new) iax jb implementation
7:50AM 4 Voicemail: How to send a notification even if Caller does not let any messages?
7:40AM 1 moving from asterisk1.2 to asterisk1.4
7:12AM 1 Need some dialplan help for obscure user request
7:00AM 0 G.729 Pass-Thru & Voicemail
6:43AM 10 duration sec and billing sec in cdr
5:51AM 0 Dial n voicemaile
5:24AM 1 Difference between SCCP and Cisco Call Manager traffic?
5:08AM 6 Instability on Asterisk
4:58AM 3 Queue trouble
3:03AM 5 sip tcp support
2:48AM 4 injecting audio announcements into sip channel
12:48AM 2 Debian asterisk-bristuff
12:12AM 13 New T1 Asterisk installation
 
Sunday April 15 2007
TimeRepliesSubject
6:53PM 11 Loudspeaker
6:39PM 3 Digium TE205P and channelbank
6:17PM 1 G723 problems with TC400B
2:33PM 1 Optipoint 420std SIP Firmware
2:17PM 1 Hardware
2:09PM 2 Is STP wire decent for analog phones?
12:09PM 0 601 Rebooting/Crashing seems to be due to full directory
12:00PM 1 LEDs on Polycom Expansion Modules misbehave when paging
11:11AM 16 Fax with Asterisk + Hylafax
10:59AM 0 transfering calls
10:43AM 0 Call tranfer drops 1st. digit
6:53AM 0 features.conf and blind xfer
5:50AM 2 saydigits in another "language"
2:02AM 4 agents and music on hold with autoanswer..
 
Saturday April 14 2007
TimeRepliesSubject
9:06PM 5 queue report problem
7:32PM 0 Trixbox abd broadvoice
5:11PM 2 Installing Applications
3:11PM 0 Presence on Polycom 301 partially broke?
10:10AM 2 a2billing
9:37AM 5 Fast busy on TDM400P
9:12AM 1 "HTTP Connection Timeout" Trouble with Cisco 7960 Phone
3:27AM 10 what version is running?
 
Friday April 13 2007
TimeRepliesSubject
9:52PM 1 no real ring back
4:37PM 5 Job listing on cisco.com for Asterisk...?
4:28PM 0 Asterisk, nat, gizmo and fwd
3:09PM 6 Hardware requirements question
3:04PM 6 openvz resources
2:37PM 1 SpanDSP (RxFax)
2:25PM 1 Dial outbount trunk numbers in a round-robin sequence?
1:58PM 6 LED does not glow on new Voicemail
1:44PM 4 E1 capacity
1:13PM 0 Parking calls and snom phones
12:48PM 1 Outgoing Calls on PRI ISDN
12:47PM 2 compile error on RHEL5 or CENTOS5
11:43AM 7 MySQL query from extensions?
10:10AM 2 Bristuff and HPEC
9:33AM 10 SIP REGISTRATION TIME OUT
9:01AM 4 Call Recording Servers
8:37AM 3 FreePBX - Vicidial Integration
6:18AM 6 Polycom 501 sluggish keys: found the problem!
4:52AM 6 A question about an install i have been asked about...
4:42AM 2 PAP2T-NA Jitter Buffer
2:13AM 0 meet me monitor
1:38AM 1 How can i add multiple callerids to an inbound route?
1:37AM 2 How can i send voicemail to multiple email IDs?
12:36AM 5 voicemail - "digits/1F does not exist in any format"
12:14AM 0 Asterisks CDMA Cards
 
Thursday April 12 2007
TimeRepliesSubject
11:58PM 0 compilation error in CYGWIN
10:46PM 5 Zap failure: cause 66 - Channel not implemented
10:02PM 2 Recording and Conferencing
8:08PM 6 Huh? IP address ending with 611
6:22PM 0 Outside Network PAP and also Outside Network eyeBeam Soft Phone
6:06PM 1 Re: Which SIP phones...
5:26PM 0 RAGI channel_status() never returnes
4:26PM 1 Spandsp-0.0.3 and asterisk 1.2
4:02PM 17 zaptel/ssh interaction
2:13PM 7 Sharing trunks between asterisk machines
12:43PM 1 Asterisk and hard phone configuration
11:24AM 0 speex codec: Out of Buffer space
10:54AM 4 Asterisk-Java website
10:54AM 1 Destar web interface problem
9:36AM 0 Installing codec g729 on Asterisk 1.2.1 on FreeBSD 6.0
9:31AM 1 Asterisk (1.4) and hints/presence/BLF
9:22AM 12 Fax Blast over IP?
8:58AM 3 Best External PRI Gateway?
8:35AM 1 Delay to start sip registration after asterisk restart
7:54AM 4 SIP: number to names
7:49AM 0 video phones and call files
7:42AM 2 DTMF problem with inbound calls on Toll-Free number
7:17AM 3 Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
6:27AM 1 (no subject)
6:00AM 1 CDR(disposition)
4:25AM 5 Measuring audio file legth
4:07AM 0 Asterisk 1.2.14 and zaptel 1.2.12 ivr hangs every 2 days
4:06AM 1 Automatic Hang
4:06AM 0 (no subject)
3:23AM 10 test
2:25AM 2 compile problem with wavelenght
1:01AM 2 hanguponpolarityswitch - where did it go??
 
Wednesday April 11 2007
TimeRepliesSubject
11:49PM 0 How to set fromuser in sip.conf so each user gets it's own callerid?
8:52PM 0 Asterisk or Trixbox
6:15PM 5 Polycom 301 questions
4:53PM 9 What is your Backup Strategy?
4:02PM 0 ZAP does not disconnect
2:53PM 27 Which SIP phones to buy?
1:09PM 3 FW: Polycom 501 issue with latest firmware : sluggish keys
1:03PM 8 missing chan_zap.so
12:52PM 2 Polycom - Static IP
10:43AM 11 Mediatrix 1204
10:37AM 16 Polycom 501 issue with latest firmware : sluggish keys
10:15AM 4 SIP Jitter Buffer Patch for 1.2.x branch?
9:37AM 3 IMAP Voicemail with MS Exchange
9:28AM 1 Call Pickup with more than one argument
9:27AM 1 outCALL- the open source Asterisk integration applicaiton for Microsoft Outlook
9:00AM 1 Cisco IP Phone services.xml sample?
8:26AM 2 SIP INFO message
8:23AM 0 GTalk and No Audio Problem
7:53AM 7 Execute EAGI script with params from extensions.conf
6:44AM 6 wideband codec support
6:16AM 0 Re: nadi: branch 1.4 r61342 - /branches/1.4/channels/chan_misdn.c
4:16AM 1 Purposely setting red alarm on PRI for testing purposes
4:07AM 1 calls bridging
3:58AM 0 Asterisk Pickup with more than one argument
2:41AM 0 IM on x-lite
1:38AM 13 Nagios asterisk monitoring
 
Tuesday April 10 2007
TimeRepliesSubject
7:18PM 18 Learn some terminalogy before mounting this task.
6:15PM 8 how to install asterisk on redhat ?
12:31PM 8 Help w/ Asterisk Cisco IP phone and SCCP
11:28AM 0 ISDN BRITE support?
10:59AM 1 XO Flex T-1 & Asterisk
9:46AM 6 Maximum retries exceeded on transmission
9:32AM 0 Re: asterisk-users Digest, Vol 33, Issue 38
7:45AM 0 Voicemail: How to send a notification if Caller hags up during announcement
7:27AM 0 Re: asterisk-users Digest, Vol 33, Issue 37
7:01AM 6 Asterisk without PSTN interface cards
6:58AM 9 Reverse-ATA : Using PSTN lines to connect to Asterisk
6:52AM 6 help with Sipura SPA 3000
5:14AM 0 clarification about bridge the call
4:50AM 0 Dialplan help - MeetMe (or ChannelRedirect) and call monitoring
4:26AM 1 Dialplan help - MeetMe and call monitoring
3:48AM 3 Re: asterisk-users Digest, Vol 33, Issue 36
3:42AM 0 checking credit by phone
3:41AM 0 Autoreply: [Posible Spam] asterisk-users Digest, Vol 33, Issue 36
 
Monday April 9 2007
TimeRepliesSubject
11:08PM 2 Strange error, "logger.c: No more room in scheduler..."
10:43PM 7 zapata.conf
10:23PM 0 Call forwarding (from PHONE configuration) with PRI
9:01PM 2 Cisco GW, PRI & CallerID Name
6:54PM 1 Re: asterisk-users Digest, Vol 33, Issue 35
6:18PM 0 Re: asterisk-users Digest, Vol 33, Issue 35
6:13PM 0 no reply to our critical packet
3:50PM 9 Too much silence, perceived delay
3:08PM 1 Date Wise Recordings
2:20PM 9 Play audio and continue to next priority before audio ends...
1:37PM 1 TellMe Voice Recognition in Asterisk working..
1:16PM 1 ${QUEUESTATUS}
1:03PM 0 Asterisk mini conference within IT360 in Toronto Apr30-May2nd
12:50PM 2 T100P -->>> TE120P
12:06PM 0 Re: asterisk-users Digest, Vol 33, Issue 34
11:51AM 0 OT: But telephony related and funny
9:39AM 2 trouble recording calls
9:26AM 3 Re: asterisk-users Digest, Vol 33, Issue 33
9:11AM 5 Polycom 330/320
9:10AM 1 Received mini frame before first full voice frame
7:47AM 4 Privacy Manager w/ No Recording
7:07AM 2 DTMF auto detection bug?
6:56AM 0 Re: asterisk-users Digest, Vol 33, Issue 32
6:46AM 6 incoming zaptel calls fail
6:14AM 8 Upgrade 4 to 8 Analog Lines Question
5:26AM 7 Asterisk installation issue - CLI showing 0 active channels
2:43AM 12 sip_header=value?
 
Sunday April 8 2007
TimeRepliesSubject
7:43PM 2 intermittent choppy sound over wifi link
12:01PM 0 Re: asterisk-users Digest, Vol 33, Issue 31
7:11AM 3 Is there a variable for SIP response codes?
5:40AM 1 Manager Originate and Var to long
5:26AM 2 Adding Noise or background noise
2:37AM 0 Re: asterisk-users Digest, Vol 33, Issue 30
 
Saturday April 7 2007
TimeRepliesSubject
5:49PM 0 Linux IAX client to zaptel voice quality issue
2:13PM 3 Prompt for a PIN number to make long distance call?
12:15PM 0 Re: asterisk-users Digest, Vol 33, Issue 29
10:35AM 2 Follow Me and Transferring Calls
10:09AM 4 Cannot compile 1.4.2 on Slackware 7
9:34AM 0 Re: asterisk-users Digest, Vol 33, Issue 28
8:45AM 4 Verizon Vonage 101
4:11AM 0 Re: asterisk-users Digest, Vol 33, Issue 27
3:59AM 3 Different devices for asterisk!!!
3:36AM 2 CRM integration with Asterisk
1:00AM 0 Wireless Bridge for SNOM360
 
Friday April 6 2007
TimeRepliesSubject
6:23PM 1 Voicemail from GTalk says "from an unknown caller"
5:22PM 0 Re: asterisk-users Digest, Vol 33, Issue 26
5:15PM 0 Yellow alarm TE110P with latest release
5:07PM 10 How well does a celldock work with Asterisk?
3:06PM 53 Verizon-Vonage Lawsuit
1:11PM 5 Balancing the Hybrid
12:14PM 0 Re: asterisk-users Digest, Vol 33, Issue 25
11:37AM 3 Snom 320 voicemail key & MWI
11:01AM 2 hox to connecte two asterisk server
7:43AM 1 Is it possible to Voicemail menus (not just audio files) ?
7:33AM 0 Re: asterisk-users Digest, Vol 33, Issue 24
7:14AM 2 SIP Header fields?
7:10AM 0 sending the dialed no to the peer
7:07AM 1 Poor analog line quality, wireless "base station", FAX-ing
7:03AM 0 Meetme and on demand monitoring
6:26AM 2 pap2 - dtmf works when 'sip debug' is enabled
5:42AM 19 FAX thru TDM400p
5:05AM 0 SVN update
 
Thursday April 5 2007
TimeRepliesSubject
11:06PM 0 Re: asterisk-users Digest, Vol 33, Issue 23
10:55PM 22 HPEC audio clipping
4:59PM 8 Open Source VoIP client (on a webpage)
4:18PM 5 BeroNet HFC-4S card is now detected as only 2 ports
1:32PM 2 Analog phones, dial out
1:06PM 2 Queue call distribution
1:03PM 0 Re: asterisk-users Digest, Vol 33, Issue 22
12:57PM 0 IAX2 threads
12:48PM 3 Asterisknow or Trixbox?
11:37AM 1 Dialplan not reading MySQL table
11:20AM 0 detecting a "beep"
10:56AM 3 IAX Trunk Failover
10:29AM 0 Re: asterisk-users Digest, Vol 33, Issue 21
8:34AM 0 No response on extensions: TDM842
8:17AM 2 What is this error message? (check_auth: stale nonce received from ...)
8:02AM 3 Polycom 601 message waiting indicator
7:23AM 4 PRI DCHAN Errors
3:42AM 0 Asterisk 1.2.17 and BRIstuff - SOLVED
3:26AM 1 Asterisk 1.2.17 and BRIstuff
1:52AM 0 SNOM and Got SUBSCRIBE for extensions without hint. Please add hint to *8 in context inbound_sip
12:56AM 0 Re: asterisk-users Digest, Vol 33, Issue 20
12:06AM 1 How to return dialstatus of second (sub) call
 
Wednesday April 4 2007
TimeRepliesSubject
9:36PM 2 polycom repair
7:09PM 1 Asterisk server hangs on after only few hours again.
4:55PM 0 Voicemail Playback Issue
3:38PM 0 "no gtalk capable clients to talk to."
3:30PM 0 Re: asterisk-users Digest, Vol 33, Issue 19
2:51PM 4 ZAP device reference in Zaptel 1.4 - SIMILAR
2:28PM 0 Bad Line Noise over T1
2:22PM 1 Pound # key not being handled
1:50PM 1 Polycom
1:12PM 3 Tunnel Q.SIG through an IP network
12:41PM 1 Queue application strategy
12:36PM 0 Re: asterisk-users Digest, Vol 33, Issue 18
12:24PM 0 Speex codec in 1.4.2
12:21PM 0 Re: asterisk-users Digest, Vol 33, Issue 17
10:34AM 2 make Zaptel 1.2.16 'struct inode' has no member named 'u'.
10:06AM 0 Console messages
9:53AM 0 Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemail to text translation)
8:51AM 1 Configuring sip.conf to allow guest access
8:48AM 0 Which GUI for call screening ?
8:25AM 1 call files
8:14AM 0 Re: asterisk-users Digest, Vol 33, Issue 16
7:48AM 0 Parked calls and Music on hold
7:07AM 1 disabling authentication
6:50AM 1 Using DUNDi in a failover environment
6:32AM 1 RE: asterisk-users Digest, Vol 33, Issue 15
6:18AM 2 Ring file
5:22AM 0 make a call with IP address
5:19AM 0 Re: asterisk-users Digest, Vol 33, Issue 15
4:41AM 0 SIP - choppy sound on local LAN to T1
4:36AM 15 "remote" SIP, no audio, or one way audio.
3:56AM 0 Asterisk Job in Saudi Arabian Companies?
3:29AM 0 what the cable to connect with digium TE110 and avaya s8300
3:14AM 2 Digium B410P Need Help
2:56AM 0 Localise VM_DATE timestamp like the voicemessage envelope
1:55AM 1 extra field
12:25AM 2 Remastering asterisk
12:17AM 0 Re: asterisk-users Digest, Vol 33, Issue 14
 
Tuesday April 3 2007
TimeRepliesSubject
9:18PM 0 I can't use the 'Group', 'CallGroup' , 'PickupGroup' in SIP channel (asterisk1.4.2)
6:21PM 0 Faxing issues
6:15PM 0 Called Number Issue
5:48PM 11 ZAP device reference in Zaptel 1.4
5:14PM 0 Re: asterisk-users Digest, Vol 33, Issue 13
5:08PM 0 DTMF via IAX ignored after a few seconds
4:47PM 7 stun
2:18PM 1 x100p not showing in core show channels
1:37PM 0 T400P 4 Port T1 Cards for Sale
1:12PM 1 Re: asterisk-users Digest, Vol 33, Issue 12
1:05PM 3 asterisk and mplayer
1:05PM 2 Play "blank" sound while VM recording?
12:56PM 1 Re: asterisk-users Digest, Vol 33, Issue12
12:44PM 9 Re: asterisk-users Digest, Vol 33, Issue 12
12:10PM 3 Re: asterisk-users Digest, Vol 33, Issue 12
10:53AM 1 Interconnecting Cisco 1760 routers with Asterisk
9:26AM 0 Re: asterisk-users Digest, Vol 33, Issue 11
9:09AM 2 RE: Asterisk-Addon-1.4.0 MySQL module
9:02AM 0 Problem with a Zap channel
8:24AM 2 Hints not working using SVN-branch-1.4-r59289
8:19AM 1 master.csv interpretation
7:51AM 3 Require only GSM Codec
7:45AM 13 Adding DND to dialplan
7:43AM 2 realtime problem
7:09AM 0 CDR and RADIUS (cdr_radius) - working
6:53AM 2 kirk Wireless + Asterisk 1.2.7 Mysql Realtime problem
6:35AM 0 Re: asterisk-users Digest, Vol 33, Issue 10
6:28AM 1 Asterisk USER PORTAL
5:30AM 8 ipv6 patch
3:58AM 12 Zaptel 1.4.1 Install Modules CentOS
2:53AM 0 Dial Macros
2:25AM 0 Quad BRI cards
1:04AM 1 SDP bug
12:20AM 0 tdm400 problem
 
Monday April 2 2007
TimeRepliesSubject
11:48PM 1 understanding what h extension does
11:41PM 0 Re: asterisk-users Digest, Vol 33, Issue 9
10:47PM 7 adding chan_celliax
8:45PM 3 DTMF problem with 1.4.1
7:35PM 2 TE120P and Unknown Signalling Method
7:12PM 2 Re: On Topic: Cheapest Asterisk USB Key?
4:36PM 0 Re: asterisk-users Digest, Vol 33, Issue 8
2:50PM 0 automonitor and CDR(userfiled)
1:25PM 1 Asterisk realtime
1:21PM 2 SIP 484 (Early Dial) and International Dialing
12:45PM 3 Replicating SIP Registrations Across Asterisk Servers
12:32PM 1 Quick question about time
12:10PM 0 Re: asterisk-users Digest, Vol 33, Issue 7
12:07PM 4 Fax detection (Sangoma)
12:07PM 7 Aastra 480 i
10:31AM 5 misdn and debian
9:03AM 0 Re: asterisk-users Digest, Vol 33, Issue 6
8:24AM 5 simplify
8:16AM 1 Mysql issue
7:22AM 0 Freeworddialup, no inbound calls
7:14AM 1 Yeastar Cards
7:04AM 15 chan-capi-HEAD and Asterisk 1.4.2
6:12AM 0 Re: asterisk-users Digest, Vol 33, Issue 5
6:12AM 0 Queue
4:48AM 1 LDAP authentication in Asterisk
4:11AM 0 LDAPget in Asterisk
2:04AM 1 Number of calls
2:03AM 0 Interconnection of LDK to an Asterisk server
1:46AM 4 SIP - Automatic Redial on No Answer
1:25AM 0 hardware for 100-200 sip users
1:16AM 2 603 Error
 
Sunday April 1 2007
TimeRepliesSubject
11:16PM 11 Best Hardphone (Subjective?)
9:31PM 0 Re: asterisk-users Digest, Vol 33, Issue 4
8:35PM 2 Trigger and Email in Dial Plan
7:35PM 19 Problems with TE110P
1:56PM 7 [MACRO-SCREEN] and MACRO_RESULT
12:02PM 0 Re: asterisk-users Digest, Vol 33, Issue 3
11:25AM 0 Add/remove international prefix
10:09AM 1 Asterisk 1.2 and res_perl - lock that leads to weird behaviour
9:10AM 0 Re: asterisk-users Digest, Vol 33, Issue 2
8:44AM 3 Announcement: Asterisk Service Provider Edition v1.0 Beta
7:57AM 0 jiaxclient run error
6:28AM 2 Linking incoming calls
6:08AM 8 On Topic: Cheapest Asterisk USB Key? (was: Re: Off Topic: Open Source USB Softphone)
6:00AM 2 Anyone here have opinion on the Linksys SPA-400?
3:16AM 2 Weird extension behavior
2:57AM 0 Re: asterisk-users Digest, Vol 33, Issue 1
12:54AM 2 No Audio with Gtalk
12:02AM 0 ISDN PRI DTMF problem