asterisk users - Jul 2003

Thursday July 31 2003
TimeRepliesSubject
10:08PM 1 24port or higher fxs
9:48PM 1 PHP API for Manager - Plaintext auth needed?
6:40PM 0 one way audio h323 callmanager
5:56PM 2 Best Analog sets for use w/*
5:12PM 3 Mutex problem in sip?
3:45PM 3 Queue and Agents in CVS
1:21PM 1 Zaptel cards, working FXS and SIP, no audio?
12:29PM 2 retrieving dialed number when overlap dialing?
10:53AM 1 AddQueueMember and RemoveQueueMember
9:56AM 0 Re: Asterisk-Users digest, Vol 1 #944 - 3 msgs
9:23AM 17 'System' application exit with error even if it performs the job as expected
8:32AM 8 SIP calls cause Segmentation Fault
7:14AM 0 Newbie - Looking for pointers
6:32AM 1 (no subject)
5:31AM 2 Sound Quality.
5:28AM 6 Parking calls - why doesn't work?
5:25AM 10 Manager
4:31AM 4 RFC2833 problems with X-Lite
1:46AM 1 RTP codec 13 received - Cisco incompatibilit y?
1:19AM 1 Help with ON-Hold, and call-transfer.
1:08AM 1 RTP codec 13 received - Cisco incompatibility?
1:05AM 3 Congestion
 
Wednesday July 30 2003
TimeRepliesSubject
7:59PM 4 Manager.pm port
4:17PM 10 Grandstream Budgettone 100 & 102
4:07PM 6 SCO/Linux concerns
1:55PM 0 X100P and incoming Context + CDR?
1:08PM 7 sip -> h323 -> ptsn
12:09PM 3 MGCP behind NAT
11:41AM 11 voicemail file access problems
9:23AM 6 X-Lite and Call transfer using Asterisk
9:23AM 9 CVS Problem?
9:06AM 0 asterisk,ata186 and Panasonic TD1232
8:55AM 0 rxgain and txgain in zapata.conf
8:00AM 26 Need help
7:40AM 13 ADSI and SoftKeys
7:40AM 2 Some stats
7:15AM 2 isdn4linux/Teles16.3
6:55AM 5 Dummy account/extension
6:50AM 0 X100P call detection
6:46AM 0 Voicetronix Hardware
3:31AM 0 ISDN Random Hangup Problems
2:55AM 4 SetCIDName
2:37AM 8 chan_sip.c problems problems from cvs 1.134
2:16AM 3 Call Transfer, Budgettone 100
1:29AM 2 Voicemail message forwarded to another extension and file format changing
 
Tuesday July 29 2003
TimeRepliesSubject
5:51PM 1 RE Pingtel Phones
5:27PM 1 Variable Substitution
5:01PM 1 Asterisk Developer's Kit (TDM) help
2:43PM 0 memory leak in voicemail.c
2:28PM 0 dialogic drivers
1:43PM 2 CAPI & CLID
12:44PM 0 IRQ Misses?
10:55AM 16 Asterisk installation
8:23AM 0 stutter tone for voicemail on SIP
6:58AM 0 7960 SIP problem when calling from outside o f LAN
6:01AM 7 Linux flavor?
5:20AM 3 [Solved] CAPI with hanging channels
5:03AM 1 Call Dropping
2:24AM 4 stupid questions ..
1:59AM 1 7960 SIP problem when calling from outside of LAN
1:05AM 0 Contact header empty in SIP-message
 
Monday July 28 2003
TimeRepliesSubject
10:19PM 1 Call Forwarding and DND conf
6:50PM 1 iax2 and reinvites
6:34PM 0 Welltech FXS SIP registering with Asterisk
5:05PM 0 Hunt group examples?
2:16PM 8 VoiceMail2 Wish List
2:15PM 0 Re: Asterisk-Users digest, Vol 1 #882 - 11 msgs
1:18PM 1 Following completion when Dialing.
12:37PM 1 D-link 102s and g723 parameters
12:17PM 12 Offering an Asterisk Documentation and FAQ Portal
9:03AM 4 Asterisk user guide ..
8:58AM 8 Call transfer on ATA186
7:43AM 0 Hardware support for TDM
7:31AM 0 Call transfer between two phones on the same ATA
7:29AM 13 RTP session traversing Asterisk server ...
6:00AM 4 Problems with two B channels
4:27AM 2 "immediate=yes or Compleate recieved" with intcoming calls with new CVS
4:17AM 0 Loop Drop on vpb/1-7
2:49AM 8 Zaptel
12:08AM 3 go on in current context after destination channels hung up ?
 
Sunday July 27 2003
TimeRepliesSubject
2:48PM 8 Channel Language
12:25PM 7 ISDN Fritz & RedHat 8.0
11:31AM 0 Festival talks fast...
11:29AM 2 Ordering digital trunks?
6:31AM 13 Australian Options
5:52AM 10 * behind ISDN pbx - Forwarding to extensions with in primary pbx
5:48AM 4 FWD-gateway prefix
1:32AM 5 Nortel 350
12:28AM 63 g729 Codec
 
Saturday July 26 2003
TimeRepliesSubject
10:16PM 4 TE410P startup
8:39PM 2 moh/playback for non-zap interfaces
4:37PM 1 can't compile asterisk
11:48AM 0 Bug Tracker Official Launch
8:47AM 0 Problem with AGI "Record File"
8:46AM 1 Asterisk SIP + Grandstream 100 phone
8:27AM 0 ISDN Callout problem
7:47AM 2 PCM Voice Quality Issue on CVS Version
6:53AM 0 meetme room
6:50AM 0 Database usage?
6:30AM 0 app_voicemail2 became a bit silent, lately...
5:30AM 0 Chan_oh323 Dial format / voice latency 4 to 5 secs
5:18AM 0 Problems with chan_sip on multi-homed hosts
12:13AM 2 Direct Indial with ISDN and Netjet-S
 
Friday July 25 2003
TimeRepliesSubject
6:14PM 3 Busy detect on pri channel?
4:23PM 12 can't get musiconhold to work
1:06PM 1 reconnecting
11:23AM 0 Web conf files
5:09AM 3 chan_capi error
4:34AM 0 7940 & AS5300 codec issues/questions G.729 & G.711
3:15AM 1 SetLanguage application doesn;t seem to work in latest Asterisk
2:55AM 2 Configuration sample for isdn4linux?
2:50AM 0 go on in context after the destination channel hung up?
2:30AM 4 Dialogic hardware
2:21AM 1 Asterisk /SIP .. nat
12:47AM 0 IAX and Call format
 
Thursday July 24 2003
TimeRepliesSubject
10:02PM 3 audiocodes fxs
9:46PM 0 INFO: How the T410P sets the number of channels per span
9:09PM 2 Instant hangup on busy Zap channel.
8:38PM 8 Voicemail() problems - Long pause after incoming message recording ended.
6:24PM 0 X100P - FO card
4:17PM 6 Configuration
3:04PM 0 MeetMe hangup
2:50PM 20 time and date stamp in voicemail
1:59PM 0 AVM C4 and external and internal ISDN bus and *
11:44AM 2 Adtran TSU 600E
11:16AM 2 isdn4linux
11:15AM 12 the 'pound' and '#' are the same?
11:01AM 2 Changes to reset method for ATA186?
10:58AM 0 SIP/H.323 Phone with intercom
10:50AM 2 Debian Package asterisk-oh323?
9:37AM 2 Cisco's CallManager and * (was: Cisco 7960g) (fwd)
9:32AM 1 FWD no longer works.. but nothing has changed? Wierd DEBUG errors.
8:40AM 0 IAXTel Connect Problem - Mini Frame
8:16AM 1 voicemail enhancements
8:02AM 4 Cisco ATA Advanced CallerID
6:51AM 0 the 'pound' and '#' are the same? (OT Rambli ng)
5:03AM 1 Asterisk <--> TTS server
2:46AM 1 compilation error
 
Wednesday July 23 2003
TimeRepliesSubject
10:07PM 2 T410P and zaptel.conf
8:45PM 1 AGI.pm?
8:35PM 0 shared line-appearance
8:29PM 0 Re: [Asterisk] help with extension switching
7:27PM 4 fxs without fxo
5:41PM 5 iaxclient (Activex)
4:39PM 0 Asterisk & X-Lite
3:14PM 1 Cisco 7960 upgrade from SKINNY load
3:04PM 8 executing an agi script after a successful Dial
2:55PM 6 h323 gateway call lost after 74sec always
2:41PM 0 how to start
2:14PM 4 how do I do s extensions with PRI
11:07AM 6 Problems with g729
10:35AM 0 asterisk-oh323 v0.5.4
9:51AM 8 h323 and oh323 modules
9:34AM 20 Asterisk as a stand alone voice mail server
8:11AM 1 newbie - simple dialout server
6:41AM 0 Integrate Asterisk with Meridian phone syste m
6:22AM 0 Self-testing E1 cards?
5:53AM 2 Integrate Asterisk with Meridian phone system
4:55AM 3 2 B channels for ISDN cards
4:55AM 3 Asterisk IVR and Hicom 300
3:16AM 1 Newbie Help
1:26AM 6 SIP info
 
Tuesday July 22 2003
TimeRepliesSubject
11:56PM 4 Codecs for use with Cisco 7960 and ATA-186
11:49PM 2 Cisco 802.11b VoIP phone?
8:09PM 3 QoS for Asterisk
4:37PM 2 Supplementing Current phone system
3:58PM 0 Delays with g729 and SIP. How come?
3:46PM 5 No callerid on outgoing call over chan_h323
2:58PM 3 chan_capi and poor voice quality
1:19PM 0 new voicemail messages
1:11PM 5 busydetect and random hangups
1:00PM 0 IAX / MeetMe problem
12:24PM 3 Ideal Prompt Recording Setup?
12:02PM 7 extensions
11:44AM 3 capi_chan error - CAPI not loaded.
11:17AM 0 TDM400P card only
9:16AM 0 * as a softswitch for pri interfaces
7:51AM 2 enabling dtmf detection on zap channel?
6:55AM 0 Verizon, SBC local company?
6:41AM 2 *--IAX--* problems. (chan_capi problem)
5:55AM 0 *--IAX--* problems.
5:14AM 5 ~3 seconds of silence when picking up a call
3:17AM 2 interfacing asterisk with a legacy PBX
2:03AM 3 SIP Call Forwarding/Transfer support ?
1:45AM 0 ADPCM Codec
1:29AM 0 calls per second with asterisk
1:13AM 0 iConnect Here PCPhone application and Asterisk
 
Monday July 21 2003
TimeRepliesSubject
9:21PM 2 MYSQL Table Structure
8:28PM 14 Using asterisk for a 911 call center....
7:17PM 1 PAnasonic And Asterisk
5:46PM 0 Robbed bit signalling debugging
2:58PM 0 RE: Asterisk-Users digest, Vol 1 #873 - 16 msgs
2:14PM 1 Outgoing calls through a calling card
12:14PM 1 URGENT! brandly new Wildcard E400P for sale at $1000
10:25AM 9 anyone with X100P & Callerid working outside US ?
9:50AM 8 CDR question
9:32AM 2 E911 and asterisk
7:32AM 5 X-Lite Build 1016
7:19AM 0 7960 / MGCP
6:52AM 10 Dynamically setting up/tearing down extensions
5:43AM 1 UK call termination..
5:05AM 0 Asterisk -> SIP -> AS5300 signalling missing on connect/clear cal l
3:30AM 20 Best software SIP client
2:22AM 6 Best E1 channel bank?
2:21AM 13 Phones
1:10AM 3 SIP Authentication bug?
 
Sunday July 20 2003
TimeRepliesSubject
5:50PM 0 Summary of VoIP options for Asterisk and request for more?
5:16PM 8 Music on hold & Read error on sound device
4:54PM 1 DTMF crashes chan_capi
12:25PM 0 No audio in Messenger
4:43AM 3 Self parked but avaliable
 
Saturday July 19 2003
TimeRepliesSubject
10:18PM 5 Asterisk crashes when trying to load G.729 module.
5:58PM 0 file repository
11:50AM 2 how to set specific codec ?
8:38AM 0 Dlink dg102s and G.729
8:30AM 2 Analog phone not ringing
4:20AM 0 XS4ALL Gateway now also does FWD
3:29AM 0 Actiontec's InternetPhoneWizard (USB) and Asterisk
3:15AM 0 Call Transfer Anouncement
2:41AM 0 IAX can be used on a different UDP port?
 
Friday July 18 2003
TimeRepliesSubject
11:49PM 22 Again Asterisk and VMWare - it works now!
10:01PM 59 Call Transfer
6:20PM 2 VoIP in hotels
3:05PM 13 questions
2:29PM 20 cdr_mysql
10:35AM 2 Budgetone and NTP (redux)
9:59AM 2 Grandstream BudgeTone 102 initial experiences
9:24AM 3 Techfone VOIP phone
8:54AM 17 "Best" VoIP provider for Asterisk?
7:40AM 0 FW: Sip codec preferences
7:02AM 17 OT: list format vs newsgroup format
6:36AM 7 H3500CW recommendation
3:29AM 2 Correct syntax to call using IAX and a different UDP port
1:28AM 0 IAX and Speex?
 
Thursday July 17 2003
TimeRepliesSubject
8:51PM 6 serious dtmf recognition problem.
8:12PM 0 Music while waiting for agent to free.
8:04PM 15 Speex support
6:06PM 1 ATA-186 software upgrade 2.16.1 - notes?
4:23PM 1 queue bug?
3:32PM 1 TE410P startup (2 boards)
3:14PM 3 AGI & Silence detection
2:29PM 0 Cisco interoperability?
2:23PM 0 Example: Writing a click-to-call application using pbx_spool
12:16PM 2 Silly questions due to ingrained knowledge of analog phone use.
11:28AM 0 Video Phones?
10:14AM 6 random hangups
10:06AM 0 Sip call question
9:52AM 0 error "WARNING[28697]: File app_dial.c, Line 304 (wait_for_answer): Unable to forward voice"
8:58AM 2 Can I interoperate with public PSTN gateways ?
8:38AM 7 Any dialing tricks...
8:23AM 6 AVM Fritz! to connect LAN with ISDN line?
7:23AM 0 FW: Echo on incoming calls (PRI->SIP) but not on outgoing (SIP->PRI) (no html crap, sorry)
7:21AM 0 Echo on incoming calls (PRI->SIP) but not on outgoing (SIP->PRI)
7:05AM 0 grandstream sip phone (NTP)
6:52AM 0 Manager/gastman
6:27AM 17 Help Needed
6:17AM 0 UK Gateway
5:57AM 2 outgoing callerid string
5:50AM 0 slightly OT /how to obtain 900 number
5:00AM 7 E1 R2 on Asterisk
2:33AM 3 Asterisk -> AS5300 SIP Interoperability
12:00AM 2 conference problem without zapata interface
 
Wednesday July 16 2003
TimeRepliesSubject
10:21PM 23 Call Pickup
5:42PM 0 Question on peer to peer config
5:31PM 1 Back-to-back connected boards load test
3:44PM 4 Segmentation fault with chan_oh323
2:49PM 1 FXS and PBX Integration
12:45PM 0 Cisco 7910 compatibility
11:32AM 4 Problems getting 7960's to play nice with Asterisk
10:20AM 2 Multiple Phones for 1 Extension
8:51AM 0 Analog features over the ATA-186
8:35AM 0 X100P in Australia (was Asterisk-Users digest, Vol 1 #840 - 13 msgs)
8:31AM 0 Sip codec preferences
7:10AM 1 Vendors for phones
6:44AM 13 grandstream sip phone
4:56AM 6 voicemail instructions
4:52AM 0 Timeout in Call Transfering
4:28AM 1 Cisco 7905G vs ATA186
3:47AM 0 addmailbox2 (Attached)
12:33AM 10 Cisco 7960g
 
Tuesday July 15 2003
TimeRepliesSubject
11:23PM 5 Asterisk on Cygwin?
9:31PM 0 user agent with auto-pickup support - is there any?
7:52PM 1 Stream Phone Call: Sound on Consule OSS to Helix server?
6:44PM 2 X100P in Australia
6:41PM 11 Text to Speech - Someone needs to do this
5:02PM 2 G729 quality
10:22AM 3 Phoneserve SIP provider
10:08AM 2 g723.1 voicemail/conference files segfault *
9:10AM 7 Conditional Contexts
9:07AM 17 Poll - Would you pay $30-$50 for high quality speech synthesis?
6:08AM 5 Chan_H323, G729 (minor problem)
3:38AM 0 Budgetone Transfer (The answer)
2:37AM 2 Alphanumerical digits
1:41AM 2 Analog commands
 
Monday July 14 2003
TimeRepliesSubject
10:20PM 0 ebs mbs p-phone question
10:14PM 3 insmod wcfxo failed ( b8zs, esf, wink start is what I'm trying to do.)
6:35PM 1 VXML?
3:52PM 1 Making Analog Phones Work
2:21PM 0 payload framesize
1:32PM 1 Fwd:[Vocal] Question about Cisco IP hard phones
12:45PM 8 Using 2 PhoneJacks with Asterisk for Data calls.
12:42PM 6 New budgetone firmware
11:48AM 5 Remote Agents
11:38AM 0 Cisco 7960 Transfer Call drop problem
10:49AM 0 Cisco 7960 Transfer & Conference
10:30AM 5 G729 licensing
10:24AM 5 MSN Messenger 4.7 vs 5.0
10:02AM 2 Odd output from X100P
9:37AM 3 Hardware Vendors
9:19AM 20 Getting started
8:53AM 0 MGCP-H323 interoperability
8:39AM 0 h323 Ringing sound
8:34AM 7 EZ-Install
7:58AM 0 (no subject)
6:58AM 2 asterisk and modem
3:33AM 13 .gsm voice format
3:23AM 0 * with external sip proxy
 
Sunday July 13 2003
TimeRepliesSubject
11:35PM 1 DTMF control for TDM device?
9:38PM 3 Setting up A TDM400P
9:09PM 0 H323 & Transfer
2:57PM 7 unresolved symbols in /lib/modules/2.4.18/misc/zaptel.o
11:16AM 13 Line Override Device
10:07AM 4 Question #3
8:05AM 1 something is wrong with gsm prompts format
6:24AM 12 AUSTEL Certified
 
Saturday July 12 2003
TimeRepliesSubject
8:57PM 3 Question About VOIP
4:53PM 2 Phone System Questions
3:21PM 4 Asterisk takes over
1:42PM 2 AGI script sample using bash shell script
12:42PM 0 Any Carrier Accessbank I users
8:01AM 6 VIP 30 phone
8:00AM 3 New Member
12:05AM 0 what is wrong with gsm files
 
Friday July 11 2003
TimeRepliesSubject
9:12PM 2 Hook Flash INFO messages
8:42PM 3 SIP immediate hangups with latest CVS
8:09PM 3 Weird experience with MOH
7:52PM 5 What does "callerid=" in sip.conf do?
7:13PM 8 audio pause/delay problems
5:07PM 1 No Sound via Sip Phone
12:56PM 5 module : cdr_sybase.so
12:44PM 1 Configuring BudgeTone and ringer over TFTP
12:28PM 1 SIP call from one extention to another
11:57AM 0 Sip: problem authenticating (with Cisco VoIP IOS 12.x) [long]
8:52AM 0 Client Call Management Application?
7:50AM 0 [Q]: Dialin problems over E1 on a Digium E100P
6:34AM 1 Unable to find IP address???
5:28AM 7 mgcp problems
4:47AM 14 G729 codec problems
4:22AM 4 mod tor2 takes 20-30% from CPU (20-30% System)
4:09AM 9 ISDN PRI E1 configuration with E100P
3:52AM 2 wait and user input..
3:35AM 2 Compile Problems with gcc 3.3
2:02AM 0 More voice prompts available now
1:21AM 3 Cisco 7960s
12:52AM 2 Wildcard E100P resellers in Europe ?
 
Thursday July 10 2003
TimeRepliesSubject
11:20PM 1 BudgeTone-100 Date and Time
10:53PM 2 msn authentication
7:49PM 0 system alias
6:15PM 0 channel bank setup "What do I do now?"
4:33PM 12 Channel Bank configuration
1:56PM 2 Why mp3 (licensing issues) as opposed to Open Source OGG
1:23PM 7 OH323 + G729 + Go2Call
12:52PM 1 Cisco 7960 SIP Craziness...
12:43PM 0 Cisco 7960 And Firmware Upgrades
11:27AM 1 Sip CANCEL or BYE when picking up a call ?
11:01AM 0 connect 2 asterisk boxes
11:00AM 0 Getting 488's between two c7940's
10:05AM 1 Cellphone as an exchange line
9:05AM 0 -- Got SIP response 481 "Invalid CSeq Number" back from 216.52.153.207
8:51AM 4 TDM10B - Dies after a few hours
8:26AM 3 Transfers on the Cisco 7960
8:17AM 1 Voicemail answers, but drops SIP call after about 3 seconds.
8:09AM 1 SIP call transfers - any other way than using '#' ?
6:58AM 1 Using Efax for virtual fax?
6:49AM 0 2003-06-10 CVS: softphone connection failures
6:39AM 3 T1 config for robbed-bit E&M AMI
1:41AM 0 Problem with meetme.
 
Wednesday July 9 2003
TimeRepliesSubject
4:06PM 10 IAX G729 Codec
2:45PM 1 IAX2 Warning
1:54PM 4 H450 problems
1:31PM 0 Asterisk Call Manager doc
1:25PM 1 OpenBSD version???
12:22PM 1 E1-RJ45 pin configuration
11:36AM 2 experience with multi-port SIP/FXS gateways?
11:19AM 7 Asterisk basic how-to on O'Reilly's site
11:16AM 1 Asterisk as SIP <-> PSTN gateway
10:41AM 2 PRI with variable length numbers
9:17AM 0 SUMMARY: Problems with Hangup Detection in VoiceMail2.
8:24AM 3 chan_h323, Asterisk and DTMF issue
8:21AM 4 incoming callerid on FXO
8:16AM 4 sip jitter buffer
8:15AM 1 MGCP-H323v2 transcoder?
8:10AM 2 callerid= being ignored
7:23AM 33 caller id
7:05AM 2 more abou msn
6:57AM 6 Music on hold quality..
6:01AM 1 PBX / Asterisk integration
5:08AM 3 It's true - Nikotel charge for not-completed calls
4:54AM 1 Use dialing plan from h.323 gatekeeper?
4:08AM 3 Asterix Manual
2:56AM 1 Matching winth asterisk-oh323
2:29AM 2 error on web page for msn
2:15AM 3 Chan_capi hanging channels
1:40AM 0 Newbe Questions.
1:16AM 9 ignorepat doesn't work
12:53AM 2 Error on PRI channel : Call specified but not found!
12:31AM 3 modules.conf again
 
Tuesday July 8 2003
TimeRepliesSubject
11:40PM 0 dbget & dbput
7:57PM 3 voip
5:26PM 0 RE: IAXTEL toll-free From: Asterisk-Users digest, Vol 1 #791 - 10 msgs
5:00PM 0 DID number assignment to SIP phones
4:15PM 0 codec problems with asterisk
2:28PM 0 SIP Problem (previous post) .. information might be relevant
1:21PM 0 SIP disconnecting : response 481
1:09PM 1 oh323 prob :)
11:21AM 8 Budgetone and Voicemail
10:07AM 8 Call Accounting
10:06AM 1 Debug PRI!
8:29AM 4 line battery check
8:10AM 11 Using multiple iconnecthere accounts
7:26AM 0 ENUM lookups
6:46AM 2 oh323 problem (small one)
6:42AM 3 RTP.C codec error 19
6:22AM 1 Agent in new CVS
6:20AM 5 Answering on an zap device
4:07AM 2 asterisk-oh323 v0.5.3
3:38AM 0 Patch to fix some segfaults in Asterisk
3:12AM 0 ECHO on sip- call
3:06AM 0 re. rtp.c RTP codec 19
2:56AM 1 chanh323 dialling
1:35AM 5 Transfert call
1:06AM 0 FW: ATA 186 in Australia
12:54AM 0 Conferences with CAPI and H323
12:07AM 1 Switch issues with non-dedicated comms.. (My experience)
 
Monday July 7 2003
TimeRepliesSubject
11:23PM 2 msn
6:21PM 1 Loaded latest CVS and get Broken PIPE!!!
6:14PM 0 Problems with Hangup Detection in VoiceMail2.
4:39PM 1 ATA 186 in Australia
4:28PM 9 Lot's of errors and warnings.
4:28PM 0 SIP canreinvite=yes Broke?
3:23PM 1 Dial plan doesn't seem to save properly
2:57PM 0 Fw: IAX Bandwidth Question
2:38PM 0 conection with other PBX's
2:30PM 8 BudgeTone-100 Early Dial
1:31PM 0 Asterisk crashing after Voicemail box creation
1:15PM 9 PCI Master Abort
1:15PM 0 Follow-up -- Using Asterisk with Nikotel
12:36PM 3 overlap dialing on a pri span
12:03PM 0 One-way talk paths (without INVITE?) and other issues
11:44AM 1 Getting Started with Digium T100/E100 cards
11:31AM 2 Can't access outside voicemail services through asterisk
10:43AM 5 System command..
10:33AM 3 Network design question
9:39AM 0 register line on sip.conf
9:27AM 1 callgroup and pickupgroup
9:16AM 0 modules.conf
9:11AM 0 Initiations in IP voice/Hybrid Voice/etc...
9:07AM 26 Asterisk and VMWare
8:47AM 7 Direct entry to your own voice mailbox
8:32AM 4 three way calling and cisco ata 186
7:21AM 5 Newbie Doubts
7:05AM 2 Problems with TDM40P
5:32AM 0 problems with new FXS module
4:08AM 1 Remote * Using IAX
2:58AM 2 IAX Bandwidth Question
2:56AM 1 Voicemail2 Contexts
12:07AM 1 Problem with SIP Phone with outgoing phone call
 
Sunday July 6 2003
TimeRepliesSubject
10:54PM 0 Ringing in sequence
8:07PM 14 Accurate Billing
12:04PM 9 Digital phones
11:11AM 0 Hardware / resources
 
Saturday July 5 2003
TimeRepliesSubject
11:20PM 5 Activate MySQL logging
7:18PM 4 TDM400P noise?
5:30PM 1 Fw: SIP Client X-Lite
4:49PM 0 SIP show channels display
12:53PM 2 FWD trouble - 407 error
11:57AM 2 E&M DID config question
10:42AM 7 Please help -- Syntax for dialing VoIP provider
9:23AM 4 Runtime error: Undefined symbol, have fetched new CVS and recompiled everything
6:42AM 4 Integratting * With Database(Newbie)
5:59AM 2 macro-record-cleanup in extensions.conf
3:41AM 0 Zaptel Alarms/Manager interface
3:00AM 5 Cllecting digits.
 
Friday July 4 2003
TimeRepliesSubject
11:20PM 15 Virtual fax on the Asterisk box
3:54PM 1 CDR Information and Pipes
10:38AM 2 How to make * send RTCP reports
9:24AM 4 zt_pri_errors: PRI got event: 8 / 6
9:03AM 0 gastman and queues
6:09AM 4 Crossover T1 cable
5:38AM 1 IVR problem from PSTN phone
5:23AM 12 Asterisk Sacrifice?
5:05AM 1 LD accontability
2:38AM 1 Accounting info for SIP Calls
1:21AM 1 [Newbie] SIP via fwd
1:01AM 5 switch => priority in the dialplan.. (probably an issue for Mark)
12:41AM 3 dst number
 
Thursday July 3 2003
TimeRepliesSubject
10:03PM 3 Coding Time/Date announce on voicemail
5:23PM 0 The Budgetone 100
2:36PM 5 res parking patch
2:11PM 1 No ringing when I dial an extension
1:11PM 5 Migration to Asterisk - Running off of Merlin Legend system
1:01PM 2 Drops due to codecs?
11:17AM 1 That is not a valid conference number meesage
11:09AM 0 ATA-186 re INVITE message turn off
11:01AM 0 How do I make Asterisk login at/use VoIP provider?
10:29AM 2 Asterisk - Protocol Converter from SIP/H.323
10:16AM 7 Need a recommendation on a good motherboard/processor combination
8:34AM 2 ATA-186 de-register
8:23AM 2 How does Asterisk handle connecting two IP end points?
8:20AM 5 How do you force Asterisk to use only specific codecs?
8:06AM 0 Is there any real asterisk documentation ?
7:33AM 1 Bugetone SIP Transfer
4:38AM 0 Bugetone NTP problem..
3:57AM 0 Modem channel in Asterisk
3:14AM 11 Using switch =>
1:15AM 0 CDR->dst and immediate=yes in zapata.conf
12:51AM 0 (no subject)
12:28AM 0 app_festival not cleaning up properly?
12:28AM 3 Problem with digit 0 X-lite
 
Wednesday July 2 2003
TimeRepliesSubject
11:35PM 0 Asteriks, GnuGk and outgoing calls
7:41PM 0 Sorry 'bout that
3:35PM 2 client reinvitation problem
11:49AM 0 Re: [Asterisk-Dev] ANNOUNCE: CLASS-like features for Asterisk
11:34AM 21 BIG problem with multiple rings before pickup
10:48AM 1 Dialout Lines ???
8:36AM 3 More switch => stuff
8:05AM 2 Sip call dropping
7:57AM 5 Linejack strikes again.
5:22AM 4 Seg Fault!!
3:56AM 2 Problems with musiconhold
3:21AM 4 Asterisk and Hot Desks??
2:48AM 9 Asterisk PBX Billing
12:22AM 2 record a conversation
 
Tuesday July 1 2003
TimeRepliesSubject
4:24PM 6 Today's Message from linphone; update on Khpone and SJPhone and X-Lite
4:06PM 0 CVS fixed
1:27PM 1 FGB not waiting for digits
12:31PM 0 Actiontec's InternetPhoneWizard and Asterisk
12:02PM 0 H.323 CallerID
11:44AM 0 Zap outgoing hangups callprogress=yes?
10:37AM 0 Acom 200 B/D/E
10:35AM 1 *8 pickup then transfer drops call
10:16AM 0 Large-scale voicemail deployment: any experiences?
9:30AM 3 H.323 Gateway Connection
7:53AM 8 Problem with echo
7:11AM 25 Enhanced queue app
7:08AM 3 gotoiftime error
6:04AM 0 Friendly Slow Faxing Reminder
5:37AM 0 chan_h323.c compile error
3:58AM 3 Unable to get SetMusicOnHold working...
3:47AM 0 "Forbidden" problem!!
3:43AM 0 Logfile rotation
2:52AM 3 picking up a ringing extension
2:09AM 0 More mec3 feedback