asterisk users - Apr 2004

Friday April 30 2004
TimeRepliesSubject
10:59PM 1 IAX2 * -> * handoff
10:50PM 1 Montoring for digits during actvie call
9:57PM 1 Asterisk missing DTMF tones from some cell phones
7:02PM 0 RE: E164 updater Client
6:20PM 1 T100P & Integrated (D&V) T1 -> Public IP Range
5:55PM 1 strange sound when bridging Zap
5:49PM 0 calling in for Voicemailmain - what does "o" do?
5:35PM 2 festival and gcc 3.3.2 (Fedora Core 1)
5:07PM 1 file.c weirdness
5:02PM 2 IAX Channel Capacity
3:47PM 2 Using IAXTel to dial FWD
3:10PM 1 Flexible Call Parking Solution
12:59PM 1 Is g.711 supported for transcoding in *?
12:27PM 1 TDM400 FXO Disconnect problem?
12:05PM 6 app_dbodbc segfault
11:34AM 1 sip notify from iconnect
11:29AM 0 FW: x100p drivers work on FreeBSD
11:25AM 0 Asterisk: problems and good things
11:17AM 0 Redhat 9 MySQL Apache PHP Installation Guide
11:07AM 1 Error compiling asterisk-oh323-0.6.0
10:23AM 0 FW: spandsp...can't compile *
10:00AM 0 ENUM Porject
9:40AM 0 SMC and Grandstream
9:34AM 1 Configuring Digium TE405P for use in Germany
9:25AM 1 Timeout Gives T in cdr.
9:22AM 0 Configuration on a Vonage Motorola VT-1000
9:15AM 0 Firefly: voicemail notification
8:46AM 0 Gnophone and Asterisk Errors
7:19AM 2 Can not compile zaptel at SuSE 9.0
7:12AM 3 Asterisk <--> Cisco router
7:00AM 2 South-Africa
6:52AM 1 FXS card dial digit wrong
6:13AM 1 (no subject)
5:41AM 10 Second X100P Card
5:36AM 0 music on hold with sip...
5:20AM 0 Spurious ring burst on TDM400 FXS port
4:48AM 2 Site for Asterisk-Ethernet Only-Sip Implementation
4:19AM 1 Security - Encryption
3:15AM 0 Find Me/Follow Me AGI Scripts?
3:07AM 2 Playing with time ranges...
1:12AM 0 Réf.: IAX Example Needed
12:57AM 0 RE: E164 updater Client
 
Thursday April 29 2004
TimeRepliesSubject
11:53PM 4 Outgoing DTMF on BRI
10:56PM 1 Stop thinking - just do it! *** Speak at Astricon 2004!
8:25PM 2 RE: E164 updater Client
5:53PM 5 Vonage and * (and what about those ATAs?)
5:49PM 1 clicks at beginning of call
5:45PM 2 IAX voicemail notification
5:38PM 0 Cisco CP-7905 phones
4:09PM 9 Asterisk VS. Skype
3:58PM 2 Flash on X100P does not really flash.
3:52PM 0 Queues and IAX2
3:23PM 3 ZYXEL wifi phones
2:54PM 0 SIPCALL and [*]
2:23PM 7 Cisco Message Waiting Indicator
1:59PM 1 IAX Example Needed
1:08PM 1 ZT_CHANCONFIG - function not implemented (38)
12:58PM 1 i4l --> capi move - how?
12:15PM 0 compiling asterisk on freebsd 5.2-release
11:32AM 1 CAPI ptp does not work
10:52AM 3 Same username on SIP & IAX?
10:32AM 2 spandsp compile error: PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here
10:14AM 2 Dlink DVG-1120s and Asterisk
9:40AM 2 e100p installation
9:25AM 1 User picks up phone, hears another call, not dialtone
9:19AM 0 OT: softswitch or otherwise?
9:09AM 1 pstn routing via cisco 1760 interface
8:44AM 0 4 Port FXO is listed on Digium
8:17AM 1 Asterisk integration with Meridian 1 Option 11 / ISDN30
8:16AM 1 SIP DTMF signaling to VM
8:00AM 3 Dropped calls -> reproducing scenario
6:49AM 8 GrandStream 1.0.4.55 Firmware
6:21AM 1 Contacting a list member
6:19AM 1 Asttapi in Terminalserver/Muliuser Setup ?
6:15AM 1 Need an explanation about different protocols
4:57AM 2 web yet?
4:12AM 5 Start recording during call by pressing button sequence
3:51AM 2 conference & sip
 
Wednesday April 28 2004
TimeRepliesSubject
11:30PM 5 Asterisk goes international :-)
7:02PM 1 cdr_mysql and macro use for outbound call issue
6:58PM 9 chan_sip.c max number of retries?
3:59PM 1 Asterisk wont start
3:36PM 3 Beeps clicks and volume problems
2:29PM 0 Enhanced Voicemail Features + IAX
2:10PM 1 spandsp rxfax crashes *
2:02PM 0 RTP problems resurfacing?
1:47PM 0 bri-stuff.0.0.2rc20a compile issue
12:49PM 2 Extra digit needed for outbound call
12:38PM 2 I love you!
12:13PM 2 Asterisk and Iconnecthere pause
12:04PM 0 Enhanced voicemail added to CVS
11:54AM 2 Polycom SIP files
10:59AM 0 Re: Re: Message
10:46AM 1 dual x100p and x-lite help for newbie
10:33AM 3 Timing
10:18AM 4 Best echo-free and trouble-free system?
9:47AM 0 weird SIP authentication problem
9:33AM 1 AGI
9:23AM 2 chan_sip.c bad file descriptor error??
8:37AM 1 Voicemail for Toshiba dk280
7:27AM 2 About TE405P Digium Card
7:18AM 1 (no subject)
6:39AM 0 (no subject)
5:25AM 0 Zaphfc random problem
4:45AM 1 do not pickup/ignore msn functionality
4:10AM 0 Are Zaptel and Asterisk out of sync in CVS?
4:05AM 0 Asterisk Segmentation Fault
3:20AM 0 FAX 25% of the first page and only via Zap
2:02AM 4 Mysql Confusion..
1:59AM 0 E100P issues
12:54AM 0 Proxy-Authorization sip
12:53AM 1 Call forwarding and Caller ID
12:35AM 0 SetMusicOnHold
12:27AM 1 Softfax/spandsp compilation
 
Tuesday April 27 2004
TimeRepliesSubject
11:48PM 2 T1 DID problem
10:29PM 1 Channel Modem Failure?
7:43PM 0 Voicemail not hanging up?
6:06PM 0 Issues with Asterisk & siproxd
5:43PM 3 Security Issue in Asterisk with sip.conf configuration.
2:51PM 1 parsing to compare
2:43PM 1 T1 passthru
2:42PM 2 Voicemail From Field
2:07PM 0 Zaphfc unable to configure (ZT_CHANCONFIG failed)
1:50PM 3 New ASTGUICLIENT released: 1.0.1
11:39AM 0 Hookflash woes
11:06AM 0 Strange Warnings and dropped sip calls.
10:12AM 1 exten fax and capi
9:46AM 2 help ---IAX2 with zaptel timming.
9:40AM 2 Second Hand Servers - How Powerful?
9:21AM 1 Queue() with H option
8:42AM 0 mainmenu problems
8:36AM 12 VOIP providers
8:04AM 1 multiple instances of asterisk spawning
8:03AM 7 Can´t compile the source!
7:45AM 0 - Re: Re: Support Digium - Email found in subject
7:15AM 1 void your warranty and get a 3.3V/5V TE405P
5:13AM 1 chan_sip2 install instructions.
3:31AM 0 cisco sp12+ how to address?
3:26AM 2 Getting started woes and an archive question
3:09AM 0 chan_h323: Different ports for both media channels (in, out)
2:13AM 0 Implementation of asterisk
 
Monday April 26 2004
TimeRepliesSubject
11:23PM 0 fax receiving problem
11:03PM 2 billable seconds
9:57PM 0 Macros and Redirect with Manager API
7:39PM 4 e164.org proudly announces PSTN support
5:47PM 1 Cisco 12 sp+ setup
5:45PM 1 using outbound sip proxy in asterisk
5:39PM 3 Sipura SPA-3000
5:03PM 1 g729 licence
2:56PM 0 Unable to play dialtone on channel xx
2:53PM 1 exten => s,1,SetVar(ALERT_INFO=<Bellcore-dr4>)
1:38PM 0 Record-route Issues
1:31PM 3 Skinny protocol documentation
11:46AM 0 Forwarding outgoing through an exsiting system
11:30AM 3 Using ',' character in applications data
11:06AM 1 Weird, weird, weird thing with my Asterisk box...
11:06AM 4 Resetting Asterisk
10:32AM 0 Voicetronix Openswitch/6
10:21AM 3 spandsp Makefile.patch
9:53AM 2 Echo Training Question From a Newbie
9:37AM 2 g.729 or g.723
9:30AM 0 Asterisk on OS-X or Darwin 7.3.0 ???
9:00AM 0 Help with connecting 2 servers via iax
8:43AM 3 dtmf tone clamping in calls to external ivr
8:07AM 1 Complete CID with zaphfc
8:04AM 0 multi-user * installation
7:50AM 1 Gnophone installation
7:50AM 0 Some Grandstream news
7:26AM 1 new sipura firmware
7:24AM 1 Problems registering with Sipphone
6:45AM 8 Intel 537ep
6:36AM 0 Re: [Asterisk-cvs] asterisk BUGS,1.7.2.1,1.7.2.2
6:27AM 2 The Asterisk Fax Manager
6:21AM 0 using an ascm office 30 phone with *?
5:59AM 1 troubles working with Voicetronix Openswitch12
5:24AM 0 SpanDSP Noise every 300 ms
5:11AM 2 Registering a Grandstream Budgetone with Asterisk from Home
4:15AM 0 asterisk-oh323, new version 0.6.0
2:39AM 3 Compiling asterisk
 
Sunday April 25 2004
TimeRepliesSubject
11:04PM 6 sip show registry question
9:16PM 2 PRI Error T100P
8:57PM 0 Cisco 7960 using Skinny protocol
7:33PM 2 Sound files in Chinese ?
2:33PM 0 Pulsed clicking heard
1:54PM 2 asterisk dials wrong numbers ?!?
1:09PM 0 Strange IAX behaviors
12:36PM 1 Fw: Stutter tone when voicemail in box
12:11PM 0 Newbie Setting up Voicetronix OpenLine4
11:43AM 3 Grandstream Budgetone G723, G729 or any compression
11:39AM 1 MusicOnHold spawns everlasting mpg123 processes
11:35AM 0 EXEC Transfer from AGI and detect busy run AGI.
11:32AM 1 ZyXEL Prestige 2000W
11:22AM 8 Using Exchange to send voicemail message
4:48AM 0 Building a PBX on spain, europe
3:10AM 0 BugTracker Information - REPOST
3:01AM 1 no audio using isdn4linux channel
 
Saturday April 24 2004
TimeRepliesSubject
10:53PM 1 \ Adtran Channel Bank? - Email found in subject
10:51PM 0 Cisco 7970 and Skinny
9:03PM 2 Galaxy Voice
6:00PM 3 Re: Hardware for handling large call volume
5:50PM 0 Choppy ringing audo
2:36PM 0 asterisk-oh323 and video
1:57PM 0 Asterisk unable to receive iax or sip calls
12:26PM 2 Is SIP BROKEN?
11:35AM 0 [patch] Binding rtp to specific interface
11:29AM 0 test message, do not reply
11:04AM 1 snom reporting busy when it shouldn't - Email found in subject
10:30AM 2 snom reporting busy when it shouldn't
5:57AM 0 Default Language support in IAX2 channels
5:27AM 0 Messengers calls dropped (SIP problem?)
2:46AM 0 compile error in chan_oh323.c
1:13AM 0 Ett, två, three, four, cinq... saying numbers
 
Friday April 23 2004
TimeRepliesSubject
6:17PM 1 newbie install problems
5:35PM 0 TekDigitel iPRO and *?
5:12PM 1 IAXPHONE failures in calls to Cisco Phones
4:38PM 1 Call Queues, Call groups
4:21PM 0 Hangup in AGI
2:44PM 2 zaprtc on 2.6
2:29PM 2 Asterisk configuration inside a DMZ w/SIP
1:43PM 0 PSTN Call drops randomly - Email found in subject
1:16PM 1 WARNING[1074420448]
12:51PM 3 zaptel on Fedora (Core 1) RedHat Linux-2.4
12:45PM 1 Busy error
12:39PM 4 call initiation
12:38PM 4 PSTN Call drops randomly
11:32AM 2 UK ISDN PRI Problems
10:41AM 0 oh323 goes silent after 5 seconds
10:09AM 6 Polycom registration
9:59AM 1 Planning Asterisk
9:08AM 1 H323 error
8:33AM 3 MP3 encoding of Monitor files
8:18AM 0 SIP to H323 with no joy
8:10AM 0 Info abaut zaphfc
8:09AM 0 CLI command
8:08AM 2 3com SIP phone working with asterisk
7:16AM 0 Adtran TA750 Noise - Email found in subject
6:26AM 1 call transfer with consultation
6:25AM 0 Réf.: Re: Asterisk with UUI support ?
6:00AM 0 Indications for New Zealand
5:28AM 1 list batching frequency
4:27AM 0 481 Call Leg/Transaction Does Not Exist
4:08AM 1 Play a file
3:42AM 3 Problem With zaphfc
2:39AM 0 NCS signaling
2:20AM 1 CAPI and Extensions.conf Security problem
1:31AM 0 Problem at night
1:13AM 1 3 companies 1 card
12:55AM 0 Réf.: Re: Asterisk with UUI support ?
 
Thursday April 22 2004
TimeRepliesSubject
11:55PM 1 Question of Asterisk timer to get Conference work
11:08PM 0 RE: Music on hold for first person in a conference room
10:25PM 2 :)
10:12PM 0 Cross/native compile asterisk for arm
8:00PM 0 ATA (private) --> Xp (NAT/DHCP) --> Internet -->Asterisk (Public IP)
7:25PM 0 Re: Hum on a TA750
7:22PM 7 smallest phone
6:09PM 2 Problems with ADIT600 and T100P
5:33PM 4 Extension buttons
5:28PM 1 IAX or SIP termination provider
4:26PM 2 SIP/IAX termination provider in NZ
3:35PM 0 [SPAM] - Re: Adtran TA750 Noise - Email found in subject
3:05PM 2 MWI indicator on SNOM200 doesn't disappear
2:44PM 2 Adtran TA750 Noise
2:26PM 1 Echo Cancellation Feature
1:36PM 2 Avoiding IAX destroy deadlock
1:22PM 2 Trouble Compiling "zaptel"
12:28PM 0 Modems compatible with NTL caller id
12:05PM 2 Interfacing with an existing phone system
11:47AM 0 Looking for IAX/SIP termination in Israel
11:45AM 0 Russia calling
11:29AM 3 D/41 ESC dialogic ISA CARD
11:17AM 1 Music on Music on Hold Distorted
10:55AM 15 Cisco phones
10:50AM 1 Channel Bank - New * install
10:09AM 0 no signal, mgcp
9:11AM 0 Re: VoiceTronix Openline 4
9:06AM 1 inbound calls better quality than outbound calls on X100P
7:58AM 3 How to get call back when transfer fails
6:38AM 0 Asterisk with 3rd party voicemail
5:55AM 0 Detecting Distinctive ring on a POTS line
5:54AM 0 Double echo cancellation disable?
5:44AM 1 Flash panel
4:57AM 2 asterisk no card
4:38AM 1 ALSA help required !
2:54AM 1 Asterisk with UUI support ?
12:12AM 1 PC based Switchboard application files??
12:06AM 0 IAX2 call causes SEGFAULT
12:02AM 3 Asterisk & RedHat Enterprise
 
Wednesday April 21 2004
TimeRepliesSubject
11:41PM 1 Install Timer to run MeetMe service
8:23PM 0 Remote Call Pickup Problem
6:01PM 0 Do you have sound file in Chinese?
5:59PM 0 Is anyone successfully using Queues and ACD?
4:01PM 0 Problem with Operator Unallocated number message
3:31PM 1 weird IAX2 things going on
3:11PM 0 Make an H323 phone act like a SIP ohone
1:56PM 0 MWI forwarding
12:47PM 0 * and CCM Voicemail questions
12:20PM 9 Cisco 7940/7960 SIP functionality questions
12:10PM 0 g729 problem HELP!
11:31AM 1 one-way audio and isdn4linux
11:21AM 0 FWD <> SIP <> Asterisk <> IAX <> Firefly
10:16AM 1 Fw: Interconnecting to an Altigen PBX?
10:15AM 3 Questions about alarm reporting in Asterisk
8:35AM 3 Webvmail
8:19AM 1 TxFax/SpanDSP problems
7:24AM 6 Help choosing a UK IAX provider
6:42AM 2 Ser and Asterisk together
6:37AM 3 T100P + Zap Errors
6:21AM 1 Repeated Notice: (UN/REACHABLE)
5:03AM 1 Asterisk from scratch
4:46AM 7 Asttapi
3:34AM 0 SIP ACK // CSeq 0 => ZAP Channel hangup
3:25AM 12 A few questions
2:22AM 0 APPRADIUS ANNOUNCE
1:44AM 1 About IAX channels
1:34AM 3 Very basic questions
1:17AM 1 uClibc patch?
12:24AM 0 Alsa driver doesn't initialize
12:20AM 1 sip 4 fedora
 
Tuesday April 20 2004
TimeRepliesSubject
9:46PM 1 Repeated Notice:
9:02PM 1 Stable from 4/20 launching many processes
7:59PM 1 does voice mail require a timer like music on hold and conferencing?
4:55PM 2 ANI II/Payphone indication
4:07PM 1 Re: Auto Answering PSTN --> Asterisk using X 100PCard
3:43PM 2 [OT] Using GS to create .tif files
3:25PM 0 Fax can't pass trough alaw
3:21PM 0 VoiceMail Web interface problems
3:21PM 3 Pattern matching rules for least cost routing
3:06PM 3 IAX clients are Unmonitored / UNREACHABLE
2:43PM 20 Cisco 7970
12:38PM 1 TE410P zaptel Driver Situation
12:30PM 1 Re: SIP re-invite
11:21AM 1 iaxtel and d-link router
10:23AM 0 fwd:Re: Asterisk prepaid debug
10:23AM 1 Extention pickup
10:01AM 0 SIP re-INVITES problem
9:46AM 3 Auto Answering PSTN --> Asterisk using X100P Card
9:23AM 2 zaprtc
8:01AM 1 ** WANTED: FreeBSD or OpenBSD programmer
7:08AM 1 asterisk/oh323 segfaults
6:27AM 1 Channels Idle Status Ring // cdr entries
6:19AM 0 TDM400P funny noise and data calls
4:04AM 1 h323 and oh323 g711 to g729 please help
3:19AM 3 Limiting incoming SIP calls & Original CallerID on transfer
2:04AM 0 Tedas hardware
1:53AM 1 Help for Asterisk and kphone
12:50AM 0 Recall: reboots
12:50AM 3 reboots
12:07AM 1 TDM400 seems healthy, but no dialtone??
12:06AM 1 notransfer=yes but still tryin to bridged
 
Monday April 19 2004
TimeRepliesSubject
11:44PM 1 'Answered' at wrong time.
10:31PM 1 [Fwd: Re: IAX config documentation]
8:19PM 0 RV: Not working!
7:59PM 1 Not working!
7:21PM 1 Connecting PBX to Asterisk
6:25PM 2 Need Help with Dial Plan
5:41PM 0 Coredump while txfax - case2
4:35PM 4 -- MARK --
3:26PM 0 Zap Outgoing
3:20PM 0 ixj module
3:05PM 3 spandsp/rxfax terminates asterisk
3:03PM 1 queue out
2:46PM 1 Load module chan_zap.so failed
2:34PM 0 Asterisk on Mac OS X 10.3
2:08PM 1 IAX config documentation
2:01PM 1 capi_request: didn't find capi device with outgoing msn =
1:38PM 1 Question about prepaid db
12:59PM 1 Playback problems with T100P
12:53PM 3 One, två, tre, quatre, cinq ... International numbers in say.c
12:22PM 2 (no subject)
11:49AM 0 SIP call between 2 *
10:32AM 4 zaphfc
9:18AM 0 *****SPAM***** Spanish translation
7:48AM 2 Advanced queueing
7:04AM 1 Speaking digits and time...
6:47AM 1 SIP dropouts
3:53AM 2 Strange CallerId behaviour with SIP
12:29AM 0 strange problem with SIP/voicemail
 
Sunday April 18 2004
TimeRepliesSubject
9:06PM 1 spandsp...can't compile *
8:40PM 0 AGI Module
7:35PM 1 Does RTP traffic go through Asterisk IP PBX ?
4:24PM 6 libspandsp.so.0
3:50PM 0 SMS receiving & sending
2:16PM 0 Using PCI cards with a laptop
1:27PM 2 Incoming on Zap
1:26PM 2 Dynamic recording function?
1:08PM 0 OpenPhone <-> Asterisk w/H.323
12:26PM 0 asterisk demo (was: x100p config)
10:51AM 1 Database for extensions+vm+sip
10:21AM 0 FWD registration problems
8:09AM 4 PRI: This number has been disconnected
7:43AM 4 Intel 536ep as a FXO?
3:19AM 1 h323 oh323 g729 please help !
2:26AM 2 grandstream and stun
1:24AM 1 DECT PC interface for cordless phones?
 
Saturday April 17 2004
TimeRepliesSubject
10:46PM 0 FWD<->NAT<->* config info
4:56PM 0 Capi & MSN routing.
12:07PM 1 Different UK Caller ID question!
11:39AM 0 SIP incoming distinctive ring
10:59AM 1 E100P for Bandwidth Termination
8:51AM 2 FW: Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)
7:17AM 0 : Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)
6:58AM 2 SIP device rings once on busy before giving busy tone with dialplan
6:47AM 2 Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)
6:27AM 1 Snom 200 Admin Password
3:34AM 1 Problem with x-ten lite
3:15AM 4 asterisk database support
3:03AM 2 no sound when connected
 
Friday April 16 2004
TimeRepliesSubject
8:43PM 0 Agent Cleanup Time?
7:31PM 2 VoIP SIP SoftPhone Recommendations
2:58PM 2 (Newbie) help please?
2:51PM 1 tor2 driver panics with 2 sticks of memory
2:11PM 1 Transfer through AGI
2:08PM 1 DLINK DPH-70 with asterisk
1:48PM 0 T1 going down
1:32PM 0 Polycom SoundStation IP 3000 conference phone on *?
12:57PM 0 Flash Operator Panel new version and Mailing List
12:27PM 0 Cisco 7940 no audio - sip debug
10:04AM 8 Cisco 7940 no audio
10:03AM 0 No read routine on channel AsyncGoto/Zap/1-1<ZOMBIE>
9:44AM 2 Warning from Asterisk
9:40AM 2 Newbie alert: Cannot get voicemail to answer (have scoured the web for help)
8:45AM 2 Strange T1 Problem - FIXED plus new question
8:42AM 1 Windows Drivers for Wildcard FXO Card
8:12AM 1 errors on Pri
8:02AM 0 Proble with sample.calla and setvar
7:02AM 0 Voice lenght option
6:26AM 1 IAX firmware for snom 200s?
6:12AM 0 OT: sorry ... list problem
5:29AM 0 SIP IAX2 MySQL Config
4:07AM 2 SoundPointR IP 300
3:19AM 1 Matching variable-length extensions with chan_zap in overlap dialling
1:01AM 0 * <-> FWD behind NAT
12:05AM 0 Réf.: Re: Re: External access to voicemail
 
Thursday April 15 2004
TimeRepliesSubject
10:29PM 2 TE405P + Adit 600 and FXO module - should this work?
8:00PM 2 FXO cards for TDM400P....
6:49PM 1 sip videosupport
5:53PM 1 ATA 186 SIP behind XP Dynamic IP Firewall to Static Public Asterisk
4:05PM 0 Sipura SPA-2000 VOIP Telephone adapters
3:18PM 7 Strange T1 Problem
2:58PM 0 problem with greek leters in CLI
2:17PM 0 SIP response 404 "Not Found" AND circuit-busy ??
1:07PM 0 All mates in Australia: Check this
1:07PM 2 music on hold problems
11:52AM 1 Unable to process inband DTMF
11:43AM 1 ATA 188 and fax
11:17AM 0 What's in a number? say.c internationalization!
9:45AM 1 Missing vm feature - turn off voicemail
9:19AM 2 too many arguments to function `ast_queue_hangup' compiling asterisk-oh323
9:18AM 0 Severe hum in recording
9:12AM 0 Call transfer with sipura
8:32AM 3 VOIP Spam
8:04AM 2 t1 won't dial outbound
7:53AM 0 external voicemail access - solved (mostly)
7:36AM 2 T1 Line install.. (UK Muppet)
7:24AM 0 Registering Asterisk to Lucent's MVAM Gatekeeper
6:58AM 0 onhold bug?
6:53AM 6 Warning message
6:19AM 1 [semi-OT] Channelbanks for european market / Alternatives
6:16AM 3 * Announcement * Astricon 2004 - call for speakers!
5:50AM 0 Re: Asterisk + Fritz!PCI + CAPI
5:13AM 1 Asterisk in pass-thru mode
1:37AM 1 Calls to Cisco PSTN gateway
1:09AM 1 Upgrade firmware on iaxy?
12:56AM 0 Asterisk List Digest down
12:44AM 1 How many lines of IP phone can Asterisk support?
 
Wednesday April 14 2004
TimeRepliesSubject
10:29PM 1 Most Reliable Proxy Server?
8:11PM 1 same extension numbers ??
7:51PM 1 ACD Functionality
6:03PM 0 RE: Protected message
4:56PM 2 freebsd?
4:06PM 1 PBX <-> AST <-> AST <-> PBX
4:02PM 1 Quality Suffers on Outgoing Only
4:01PM 4 Fax Over VoIP
3:39PM 2 voicemail notification - LED solution
3:35PM 0 RTP Read error
3:31PM 0 No CVS STABLE for zaptel and libpri?
2:52PM 2 CallerID over IAX
2:44PM 1 background / goto commands
2:38PM 1 Asterisk and Pleiades P32mxi
2:26PM 3 IAX2 update - timestamp issue within iax pkts
2:11PM 1 FAX?
11:52AM 4 sip software
11:26AM 0 Frame too large?
11:17AM 0 Cisco 12SP+/30VIP phone support on Asterisk
10:42AM 1 Cisco Call Manager 3.2 and Asterisk..
10:32AM 0 ADDPAC 200 w/SIP
10:23AM 1 Asterisk, GalaxyVoice and Humble Pie
9:06AM 2 Spoofing CallerID on Demain
9:00AM 3 VoIP Phone Recommendations
8:49AM 0 Spoofing CallerID on Demand
8:25AM 2 No ringing sound on GS phones
8:13AM 2 3com Ethernet Power Supply
8:00AM 1 Run Asterisk without any .conf file ??
7:53AM 1 MeetMe - new e and E flags?
5:03AM 0 Quality
4:30AM 1 caller id not working (zap)
4:15AM 0 h323 and * question
3:40AM 0 H263 SIP Video Playback
1:49AM 2 dtmf for public telephony access
1:47AM 0 Asterisk and SER - choppy sound with G.729
12:34AM 1 CND (CID) woe on a TDM400P
12:08AM 1 IAX - lan ip phones.
 
Tuesday April 13 2004
TimeRepliesSubject
10:55PM 1 SIP: missing "180 ringing"
8:21PM 1 SIP->h323 problem DTMF
6:22PM 1 DNID Digits - Australia
4:38PM 2 T100P E&M Wink Trunk
3:40PM 2 Vonage goes to .ca
3:31PM 1 Polycom phones noise cancellation
2:01PM 0 Upcoming 1.0 Release Suggestions
1:54PM 0 sphinx voice recognisation
11:55AM 7 Upgraded to latest CVS, now no IAX1?
10:55AM 0 Call parking on central asterisk system
10:28AM 0 Dialout from SIP to PSTN
10:07AM 0 Bug with 'r' in dial
9:14AM 6 VoicePulse Connect Problems
9:05AM 0 VideoMail
8:46AM 1 small question 3 way calling
8:33AM 0 *** List etiquette - digest readers
8:29AM 1 Quality Problem
8:11AM 1 Internationalisation/Internationalization
7:35AM 0 RE: Asterisk-Users digest, Vol 1 #3413 - 14 msgs
7:05AM 3 CallerID in Australia
6:13AM 4 Dial Plan Format Strings
5:49AM 2 controlling call duration
4:58AM 1 T100P Timing Was:T100P/ ZAP / PRI errors
3:27AM 0 AW: IP Phones that support G.723 on H.323
2:21AM 0 E100P Leds
 
Monday April 12 2004
TimeRepliesSubject
11:14PM 2 TDM400P Issues
10:17PM 0 strange error at extension.conf
7:57PM 0 Re: RE: RxFax/spandsp: not disconnecting
5:39PM 1 tcp/ip stack tweaks
5:19PM 4 X100P and NTL (ex Cable + Wireless)
4:55PM 0 webmin ?
4:42PM 3 Hunting S(n)IPs
4:19PM 4 Invalid module format in 2.6.5 after running make linux26
3:08PM 1 time of waiting in queues
2:44PM 0 RE: Asterisk-Users digest, Vol 1 #3408 - 12 msgs
2:13PM 1 call queue list members using sql query
2:00PM 3 TAPI driver
1:38PM 1 Dial Outside SIP address from AGI
1:29PM 2 SwissVoice IP10S not able to dial calls
12:47PM 3 Zapateller issues
12:30PM 1 OT appologies to list
11:34AM 0 oob to inband dtmf over rtp
11:29AM 1 Voicemail config from database
11:00AM 1 G.723
10:49AM 1 Trouble compiling chan_capi on Suse 9.0
10:35AM 2 Voicemail storage in DB
9:19AM 2 Asterisk systems
9:04AM 0 RE: Asterisk-Users digest, Vol 1 #3387 - 9 msgs
8:39AM 1 Re: Booting error - Unable to specify channel 2:
8:35AM 2 Random disconnect of calls
8:30AM 5 T100P / ZAP / PRI errors
8:04AM 0 CVS head on FreeBSD 4.9 will not compile
3:05AM 0 Re: Asterisk-Users digest, Vol 1 #3402 - 17 msgs
12:34AM 0 *** MGCP on the menu? Check today's special!
 
Sunday April 11 2004
TimeRepliesSubject
9:28PM 1 editing errors/typos in rev 2 of The Asterisk Handbook (current version on digium's site)
5:57PM 1 newbie - Asterix and modem cards
4:52PM 2 Booting error - Unable to specify channel 2: No such device
3:32PM 3 Caller ID via IAX
2:45PM 0 SIP Software video client
2:35PM 0 installation failure
2:11PM 2 Help getting started?
11:25AM 1 problem with SIP configuration AND EXTENSION.
11:01AM 1 Cisco 7940G/7960G SIP phones local echo on * box.
10:23AM 0 incomming call x100p
8:16AM 5 Voicemail Question
6:20AM 1 *** Hang on, we're on our way to 1.0
5:44AM 1 Config docu for SIP<->PSTN gw ?
2:42AM 1 X100P card issues - noise, volume, etc
 
Saturday April 10 2004
TimeRepliesSubject
3:49PM 0 Where to get IAXY firmware and documentation
2:45PM 0 Nwebie Config Problem
2:26PM 1 How to set the jitter buffer
1:04PM 5 Sipura SPA-2000
12:31PM 0 SoundCard and Voice Quality
12:01PM 2 X100P FXO PCI Card
11:56AM 1 VoicePulse 1-800 numbers sound problem
11:24AM 2 how to add prefix to calling number
9:20AM 1 Hum/bux on line
9:17AM 4 Time/Date missing on Cisco 7940G and 7960G SIP phone display
9:11AM 2 Obtaining the stable version
8:17AM 1 Archive Post ISDN Q.931 disconnect cause codes
7:39AM 4 Woodpeckers Revisited
7:24AM 5 Newbie Issues => SIP won't stay connected, and IAX Unable to Create Channel
5:44AM 0 Nothing to do? Go bounty-hunting!
4:36AM 4 No ringing tone with IAXY (and other bits and bobs)
 
Friday April 9 2004
TimeRepliesSubject
10:14PM 5 vm e-mail notification stopped
7:35PM 0 RedHat/Fedora RPMS Update
5:29PM 0 syslog error
5:09PM 0 IAX phone for Pocket PC
2:07PM 6 Analogue telephone cards for the UK
10:58AM 1 Voice mail notifications?
9:44AM 0 wcfxo module fail to load (Unable to request IRQ 0)
9:12AM 1 New Zealand indications.conf
9:02AM 9 small linux distro to run * in old boxes
8:56AM 1 default caller id from X100P
8:46AM 1 NuFone and international dialing
8:12AM 2 IAX2 DTMF Problem
7:44AM 0 Clearpath
3:58AM 3 Ignorepat with capi
3:50AM 0 Réf. : RE: Réf. : Re: Fritz ISDN PCI v2 and CAPI
2:56AM 0 app_queue dialback cdr problem
12:56AM 2 g729 and dtmf
 
Thursday April 8 2004
TimeRepliesSubject
10:59PM 1 AGI -> GET DATA not working on current stable cvs (anyone else?)
10:01PM 0 Latency and 'Scratchy' Voice...
9:46PM 3 Asterisk Server Crashing with New Application
9:26PM 0 IAX2 Trunk to PSTN (voicepulse) questions...
9:17PM 3 Re: : External access to voicemail
9:13PM 0 Zapateller problems solved
8:54PM 1 Problems with Zpateller on incoming external calls
8:41PM 1 application Directory (Modified by Ryan Thrash)
8:41PM 0 application Directory
5:02PM 0 AgentCallBackLogin agent-incorrect play right away
4:13PM 1 Two operators, 10 rollover lines, Cisco 7960G chanisavail problem
4:10PM 1 Different PTSN Hangup problem
3:36PM 0 Re: [Iaxclient-devel] codec negotiation ?
3:03PM 0 call progress on x100p
1:49PM 1 Live Music on Hold
12:42PM 0 RE: Asterisk-Users digest, Vol 1 #3373 - 14 msgs
12:37PM 1 Hangup on SIP unreachable?
11:41AM 4 External access to voicemail
11:04AM 4 Local Calling Area database?
10:55AM 1 FreeBSD port of asterisk
10:54AM 0 TDM Stater kit all working - WOOHOO - wondering about Asterisk FAX Support
10:48AM 2 Auto Attendant??
10:31AM 0 Using Skinny with a 7905G phone
9:10AM 1 Adding two FXO cards - not working
8:52AM 2 i'm looking for reference guide for Skinny SCCP
8:30AM 1 Asterisk & 3com nbx 100 support
8:30AM 3 TigerJet ISDN card
8:11AM 0 RE: Asterisk-Users digest, Vol 1 #3368 - 12 msgs
8:01AM 1 can't hear vm audio
7:55AM 0 NTL PRI Config required please
7:32AM 2 Zapata required?
7:18AM 0 transfer sip
7:00AM 3 Fwd: Sasquatch, the Loch Ness Monster, UFOs and...
6:48AM 5 Restart Asterisk
6:43AM 4 PC based Switchboard application
6:31AM 0 Caller ID on TDM400P Quad FXS
3:10AM 2 Réf. : Re: Fritz ISDN PCI v2 and CAPI
2:06AM 0 WAN Side Calling
1:40AM 0 DIAX and CallMe feature
1:09AM 0 Réf. : Re: Fritz ISDN PCI v2 and CAPI
1:07AM 1 error compiling cdr_mysql support
12:58AM 1 GUI?
12:28AM 2 Fritz ISDN PCI v2 and CAPI
 
Wednesday April 7 2004
TimeRepliesSubject
11:06PM 6 dreaded Caller*ID failed checksum
9:32PM 0 Channelized T1, T100P problems
8:46PM 1 SIP <--> PSTN gateways
8:19PM 0 Adtran 850 questions
6:57PM 0 Cell Phone, *, Portability
5:29PM 2 Presence
4:32PM 1 Voice Mail Email problem
3:46PM 2 Problems with ADIT 600 - latency, loss, etc
2:30PM 0 Presence (was FW: pda skype)
2:25PM 1 H.323 Seg faulting
2:01PM 0 inband dtmfmode, SIP to VoicePulse, > 1 digit extentions do not work?
1:18PM 2 error 488 - Not Acceptable Here
1:07PM 0 Toshiba Digital Phones -> Asterisk
12:43PM 3 dropped calls from queue
12:43PM 0 Asterisk dimensioning (IVR, mass calling)
12:27PM 3 Asterisk call manager
11:28AM 1 attendent transfer on ZAP channels
10:54AM 0 callback with 3 way call?
10:42AM 0 Call hangs up after a fiew seconds with a quad BRI
10:38AM 5 Lucent Phones
10:24AM 1 Asterisk / SMP / Scalability
10:02AM 0 Struggling with ISDN4Linux and Asterisk config
9:46AM 4 B-channels resetting every 60 minutes?
9:20AM 3 Getting info about changes in CVS
7:59AM 0 no good day today ! :(
7:56AM 1 ZAPRTC question(s)
7:49AM 1 Out of trunk data space on call number 16386, dropping
7:29AM 0 Dial Capi Question / Problem
7:08AM 3 Dial-In/Out Modem Zap Channel Config. Adtran 750
6:58AM 2 Siemens EWSD 13
6:24AM 0 Bug? Asterisk crashes if SIP UA hangs up first
6:04AM 1 errror compiling asterisk from cvs
5:39AM 1 Strange SIP issue (again)
4:48AM 0 SIP flashhook transfer
4:37AM 4 quadBRI and UK ISDN2e
3:34AM 1 chan_oss.c:461: error: too many arguments to function `ast_queue_frame'
3:02AM 0 indications.conf for Portugal
2:25AM 1 PSTN calls do NOT hang up
12:59AM 4 Callerid + Zaphfc
 
Tuesday April 6 2004
TimeRepliesSubject
10:54PM 5 FW: pda skype
8:31PM 17 res_motv: Request for Comment
6:35PM 0 RE: Siemens usb developer kit
5:54PM 3 Software to test a LAN for possible VoIP Install
5:39PM 1 Zap channel still in use after MeetMe conference ends
5:28PM 1 Looking for a Polycom IP phone --> Asterisk expert
4:54PM 0 queston for res_radius users
4:20PM 1 indications.conf settings for spain
3:56PM 0 Compiling Zaptel 0.9.0 drivers
2:21PM 1 SIP phone registering problem
1:30PM 0 quad BRI. Outgoing calls droped in 10 seconds.
1:29PM 1 voicemail-hangup issue
1:19PM 5 registration failure
1:18PM 1 Non working 800 numbers
12:47PM 0 HELP! - weird 7960 problem - phone goes nuts - display flashes - phone reboots
11:58AM 0 zapHFC in TE mode with multiple hfc cards
10:48AM 1 Quick Caller ID and Voicemail ?s
10:29AM 3 Passing DTMF
10:20AM 1 grandstream distinctive ring
10:03AM 1 SIP Friends and MySql
9:43AM 1 Asterisk CLI Issues - CVS-03/30/04-14:34:01
9:37AM 4 Need a list of asterisk built-in variables
9:31AM 1 How to use ZapHFC ?
8:39AM 1 Softphone (with USB headset) for Mac recommendations
8:09AM 3 SIP Proxy Problem (NAT Environment)
8:02AM 3 Problems with IAX2?
7:19AM 0 no ringback - stable not fixed
6:46AM 2 Largescale Asterisk setup - 1000 external lines
6:29AM 1 softphone (SIP) with multiple profiles
6:02AM 6 swissvoice ip10s
4:49AM 0 Ericsson Webswitch 100 Model G4 with * ?
3:15AM 2 Astersik and Europe
1:31AM 1 SIP soft?
1:23AM 1 Agi and bridging problem when codecs differ
1:01AM 1 gsm playback garbled over sip
 
Monday April 5 2004
TimeRepliesSubject
9:36PM 2 WAMi - Windows Asterisk Manager
9:32PM 4 mpg123 issue and solution
7:11PM 0 Level3 and resellers (was: Spring VON Wrap Up)
7:02PM 0 SRTP (was: Spring VON Wrap Up)
6:26PM 2 iax2 trunk - unable to accept trunk packet
4:28PM 0 SingTel ready to break into web telephony
3:37PM 5 Stable Relase Broken ?
2:17PM 0 Dropped calls, 5-10 seconds of silence
2:02PM 1 Extensions.conf sending calls to Cisco AS5300
1:57PM 5 Auto connect to voicemail
1:55PM 2 Disambiguating incoming IAXTel calls
1:54PM 0 DTMF Passing
1:20PM 0 Receiving H323 calls
1:18PM 0 RTP dataflow directly from a SIP phone to a H323 phone
1:06PM 0 Re: Asterisk IAX gatewway
12:58PM 2 Change IP info.
12:28PM 2 ADPCM 4-bit, 6 kHz
11:44AM 4 Spring VON Wrap Up
11:41AM 3 Buzzing on TDM400P FXS?
11:37AM 3 ZAP channels
10:29AM 1 RPM packages
9:25AM 0 Segmentation fault, exit status 139, ...
8:49AM 4 Cisco QoS Howto
7:47AM 0 iax2 reload - how ?
5:50AM 4 Redhat 9 OVER, Fidora Support, comments please.
5:45AM 4 The maximum capacity of MeetMe
12:51AM 1 sip no sound?
 
Sunday April 4 2004
TimeRepliesSubject
9:49PM 1 IAX2 Problem and Question
9:41PM 0 * behind NAT - FWD or ICH but not both
9:29PM 0 Strangeness
8:21PM 2 Voice Mail Service
8:10PM 1 Wildcard TDM400P
7:47PM 1 What is the most popular pre-paid billing system?
7:09PM 0 cronjob to reboot gs101
4:36PM 1 Silence suppression on SIP calls generated from Asterisk?
4:07PM 1 SetCDRUserField actually works?
12:29PM 3 SIP Registration Errors
9:31AM 0 Using externip in sip.conf with DNS name
7:13AM 2 Problem with Manager Originate
7:03AM 0 Asterisk PBX -> RT Integration
5:50AM 0 Balance my customers
2:28AM 3 Please help
 
Saturday April 3 2004
TimeRepliesSubject
11:44PM 1 Asterisk - Cisco 7960 - NAT
9:38PM 1 Direct connection to Packet8 without DTA
8:01PM 1 Unabled to exit console
7:00PM 0 Question receiving calls via SIP
5:41PM 0 Grandstream and codec G.711
3:32PM 0 Another Newbie Question: Does Asterisk allow for a hot failover solution in case of failure?
9:38AM 1 Archive heading broken
9:36AM 2 STABLE 1.0 Branch CVS repository
7:30AM 2 FireFly Problem
6:28AM 2 Ztdummy - is it requirement?
 
Friday April 2 2004
TimeRepliesSubject
11:50PM 0 The Windows Asterisk Management interface is ready for beta testing.
11:00PM 0 * server acting as SIP/IAX gateway problem
9:36PM 3 cron job to reboot GS101
8:27PM 1 ANNOUNCE: Flash Operator Panel - Extensions fixed
8:14PM 2 All calls go to Voice mail and never ring.
4:22PM 1 IAX/SIP in 604?
4:12PM 1 problems getting inbound to work @ voicepulse
3:46PM 5 Seattle IAX Termination
3:33PM 1 Asterisk and SIP Communicator
3:05PM 3 WiSIP Firmware Version F?
2:19PM 0 auto-attendent
12:47PM 4 avaya and linux
11:32AM 1 T100P specs
11:30AM 0 Newbie Question: ISDN and Capacity Planning
11:13AM 1 Newbie: ISDN and Capacity Planning
11:04AM 2 One voicemail -> multiple boxes?
11:04AM 2 H.323 vs SIP?
10:13AM 0 ms messenger problems
10:08AM 0 modprobe wcfxs ------ fail
10:05AM 1 error with asterisk -vvvvc
9:33AM 0 First approach to Asterisk - need help
9:13AM 7 Welltech FXO: initial tests
9:11AM 0 Asterisk and Zapata... which kernels?
8:54AM 1 Firefly Client can't receive incoming calls
8:01AM 2 Gnophone installation problems
7:52AM 1 X-Lite -> Asterisk: Cannot transmit Audio
7:38AM 2 SIP register and externip
7:27AM 1 Voicemail Indication Software
7:22AM 1 dtmfmode=inband with G.729
7:14AM 0 VON show report - Wi Fi Phones
5:37AM 0 Office Space Quotes : Get Office Space Quotes
5:36AM 0 Best Web Hosting Resource
4:00AM 0 SIP call troubleshooting
2:12AM 4 checkout ztdummy
1:00AM 0 MGCP and IPH-90
 
Thursday April 1 2004
TimeRepliesSubject
8:23PM 4 CISCO 7940 and directory/services problem
6:06PM 0 I'm still a little lost...
5:29PM 0 ISDN BRI-U card suggestion for use in USA
5:16PM 1 quadBRI card installation issues
5:12PM 0 MP3Player problems...
4:49PM 1 sipura fade to static
4:18PM 1 Is asterisks the best for a simple DTMF response system?
3:35PM 1 Still trying program -> phone call
3:31PM 0 DG104S (MGCP) requies me to reboot often
3:07PM 1 Can't block CallerID outbound
2:57PM 1 dialout with chan_capi
2:32PM 2 PRI integration with Marconi switch
2:29PM 0 Dialing PIN from console
2:08PM 4 Asterisk call forwarding / remote dial-in/out?
1:52PM 15 ANNOUNCE: Flash Operator Panel
12:40PM 1 how to config Windows Messanger?
12:20PM 2 Where is the archive?
11:48AM 1 how to add a wiki page?
10:44AM 1 Echo's and dropped calls
10:07AM 4 sip problems
10:01AM 2 SoftFAX/spandsp cvs access
9:51AM 1 Just static on TDM400P (not even a dialtone)
9:23AM 0 Nasty one way distortion between TDM400 and SIP Phone.
9:20AM 1 I didn't want to bother the list with this, but...
9:08AM 0 Drayteck Vigor Router with traffic shaping
9:06AM 0 (no subject)
8:37AM 2 H323 - SIP Interoperability
7:49AM 1 Asterisk + Cisco 7920 + chan_sccp or chan_skinny
6:58AM 0 Linphone connecting to default Asterisk Samples
6:42AM 1 European and/or US DTMF Tones?
6:24AM 2 Meet Me and G.729
6:08AM 0 * to * sip authentication failure
4:06AM 0 RxFax/spandsp Crash
1:51AM 2 Anyone has a working * with E1 in Mexico E1 R2 modified?
1:01AM 5 Zap Channels Hang