asterisk users - Mar 2005

Thursday March 31 2005
TimeRepliesSubject
11:38PM 2 Ztdummy is Loaded but Asterisk is not using it
10:51PM 4 Caller ID on voicemail messages
10:36PM 2 VOIP to the PBX
10:09PM 1 NuFone, VoIPJet, circuit (fast) busy question
9:02PM 0 Setting Up @Home 0.8 Guide
8:11PM 0 Calls to outside lines
8:02PM 2 Problem with livevoip dial out
7:09PM 0 Czech ISP Problem Connecting
6:48PM 1 Asterisk - MAX TNT T3 VoIP and PPP T1
5:53PM 1 auto navigate external IVR
5:46PM 3 sangoma S508/FT1 ISA
5:06PM 5 Integrating door intercom?
5:06PM 0 IAXtalk
4:30PM 2 Some phones need about 11 second before they ring
4:28PM 1 additional fields in Realtime
4:25PM 5 Webmin
3:07PM 0 dialparties.agi
2:43PM 3 Preserve g729 registration over reinstall??
2:04PM 19 Livevoip still no DTMF?
1:50PM 4 Timecard application
1:42PM 13 Polycom sound quality problems
1:39PM 1 MFC/R2 in the Philippines
1:23PM 3 Enhanced Queue App Revisited
1:22PM 3 Can this be done?
1:16PM 0 Guidelines for sizing hardware...
12:47PM 0 DVG-1120S
12:41PM 0 Manager API calling DUNDiLookup
12:38PM 0 Calling DUNDiLookup from manager api
12:29PM 0 Are there online forums instead of thisemailforum??
11:50AM 2 System requirements as users grow...
11:46AM 1 patlooptest: Usage, setup?
11:12AM 4 Installing CAPI
11:07AM 4 Online forums vs email list...
9:32AM 2 ATCOM Gateways AG-168, AG-248, AG- 468
9:07AM 2 Asterisk compatible IP Phones
9:03AM 5 Phones "Callwaiting" enable by default?
9:01AM 0 Customized ring tones
8:52AM 22 Are there online forums instead of this emailforum??
8:44AM 0 agent and queue autologoff
8:36AM 3 Echo on internal SIP
8:26AM 88 Are there online forums instead of this email forum??
8:04AM 18 Asterisk-1.0.7 Build - Serious issues
7:46AM 0 one way audio with X-lite for Linux/Suse 9.2
7:32AM 1 chan_capi looking for missing channel_pvt.h
7:19AM 4 Asterisk Realtime - configuration help
7:18AM 2 Business Opportunity for Australia
7:16AM 1 ser, asterisk and conferencing
6:56AM 4 Concurrent Call in Asterisk
6:53AM 3 Many analog lines
6:50AM 5 Time sync on PRI
6:48AM 2 Automatic Configuration Tools?
6:46AM 2 AMP not working in GUI
6:20AM 3 sharing asterisk among several companies
6:14AM 1 ser -> asterisk ->cisco gateway
6:10AM 4 sms and DDI UK
6:08AM 2 cvs-head from 3/31/05 fails to load
5:29AM 1 Installing asterisk and components
5:18AM 2 Problems editing oh323 configuration parameters
5:05AM 4 setting SIP to dial PSTN with TDM400P
5:03AM 0 We require Asterisk configuration and support consultants
5:00AM 1 Music Answer while waiting
4:52AM 0 early B3 connect with TE110P
4:31AM 1 Reject second IAX call
3:49AM 0 External line hangup
2:39AM 0 DTMF detection in dial macro
1:45AM 0 'RFC3261 transaction matching failed' and 'one-way' communication
1:39AM 4 sip.conf match
1:12AM 2 how to call land line number using wireless land line service through asterisk
1:11AM 4 Simple authentication
1:01AM 7 CAPI call fails
12:39AM 0 call timeout
12:14AM 0 RE: TE410P Loadtest problem
 
Wednesday March 30 2005
TimeRepliesSubject
11:51PM 0 Asterisk::AGI script calling macro with arguments - has to be a simpler way...
11:50PM 0 Problem faxing to a fax machine.
11:23PM 5 dial cmd - called party prompted before connect
11:23PM 0 Asterisk to Asterisks
10:53PM 3 Zaptel Periodic Reset
10:32PM 4 SuperMicro X5DE8-GG Motherboard Goes Kaput after Installing TE410P Card - Yikes!
10:26PM 22 Xten-lite for linux
10:08PM 0 Modprobe: FATAL: error running install command for wctdm
9:02PM 2 G729 on Soekris 4801
8:54PM 4 Cisco 7960 and Asterisk, I think I have a curly one here
8:46PM 3 cmd Authenticiation
8:33PM 11 No prompt after installing
8:27PM 0 Problems running asterisk on Solaris 8
8:02PM 0 LNP in Area Code 636 (Missouri)
7:53PM 10 Problem with Music on Hold. Please help
7:38PM 1 using amp with asterisk?
5:14PM 93 Sangoma VS. Digium
5:12PM 1 zoom x5v and *
4:11PM 0 Transferring a Call out...Please help!
4:03PM 4 Asterisk <--> PABX
3:45PM 1 Parked Call Issue with realtime Asterisk version
3:44PM 0 Confused: Qozap is on interrupt 209 alone - is this good or not ?
3:36PM 2 Limitations of aah
3:18PM 1 ISDN question
2:57PM 2 Help with Application Development in Asterisk
2:23PM 6 CheckGroup and transfers
1:42PM 0 How can I solve this?
1:12PM 0 Web-based conference management update
12:57PM 2 Toll Free dialing problems
12:54PM 2 Unable to allocate channel structure
12:39PM 7 Voicepulse connect has doubled their rates
12:29PM 0 Settup of incomming calls
12:18PM 2 Does the Grandstream ship with a power brick
12:05PM 0 Optimal iax.conf settings for VoicePulse COnnect
11:56AM 3 File permissions and ownership
11:51AM 5 Australia and SetCallerID
11:48AM 0 GXP-2000 Grandstream - First Impressions
11:46AM 5 Physically Small Box Asterisk Systems
11:22AM 2 Bristuff
11:06AM 0 Altigen & Asterisk with H.323
10:57AM 0 Transfer a call in the IVR
10:55AM 1 getting boot server working for PolyCom IP500
10:13AM 1 [newbie question]Can I can from a phone through Asterisk to another Asterisk server to call out from the 2nd Asterisk server
9:56AM 0 newline in an sms
9:24AM 1 What the best Asterisk architecture for 900+ users?
9:19AM 4 username/password for PolyCom IP500 webinterface?
9:19AM 0 Monitor command full static
9:10AM 2 Using HFC-S card
9:00AM 0 CISCO 7970 COLOR FROZEN
8:50AM 1 Asterisk@Home 0.8 released
8:28AM 0 (no subject)
8:24AM 6 username/password for PolyCom IP500 web interface?
8:15AM 0 APP CBMYSQL
7:59AM 0 HELP: How to configure h323 channel driver ?
7:49AM 1 job offer - in german only
7:30AM 6 Giving sip users multiple contexts
7:08AM 1 Recommended GSM gateway
6:44AM 0 Polycom IP600 Cannot answer - SOLVED
6:32AM 1 What is ZAP ? newbie question sorry
6:29AM 2 Asterisk @ home
6:18AM 4 Asterisk::AGI script won't work?
6:17AM 1 Bristuff and startup scripts
5:49AM 0 Problem on outgoing calls (quadbri card and bristuffed Asterisk latest) ?
5:44AM 0 Solaris install from HEAD
4:39AM 0 Asterisk GLIB_2.0 Error
3:24AM 1 Troubles with VoIP providers
2:14AM 0 IAX realtime dynamic
1:24AM 7 Comprehensive Asterisk Load Testing
1:13AM 0 (no subject)
1:03AM 0 IPSwitchBoard Version 0.71 Released
12:21AM 1 Combatting echo in VOIP
12:00AM 0 TE410P Outgoing Call Volume
 
Tuesday March 29 2005
TimeRepliesSubject
11:29PM 2 Polycom IP600 Cannot answer
10:40PM 7 Implant GIPS's codec to Asterisk
10:02PM 0 Call Waiting and FXO
9:49PM 1 How do i transfer/forward a call out?
9:07PM 1 Forget Asterisk@Home 0.7 :-) :-) 0.8 is out
8:50PM 7 ACD queue question
8:49PM 3 Test Line
8:15PM 0 re: Problem: Compiling error for SpanDSP
7:43PM 0 Problem: Compiling error for SpanDSP app_rxfax
7:36PM 0 voicemail patch for forcegreetings and forcename?
7:35PM 0 .call Files and Unkown Keywords
6:42PM 5 Soekris products available in the US?
6:33PM 1 Houston DID
5:52PM 3 iax2 & nat
5:41PM 14 Sipura 3000 FXO with Asterisk
5:00PM 0 Astfax questions...
4:42PM 0 Using @Home 0.7 and wanting to debug dial plan problem
4:26PM 5 Sipura SPA 2000 - Miltiple Ring Tones
4:16PM 5 Upgrade *@home to CVS-HEAD
3:58PM 5 help w/ basics
3:26PM 6 Can Asterisk do this ?
2:58PM 1 External voice channels pack up
2:31PM 5 Avaya Partner ACS system, pre 7.0
1:59PM 0 ASTERISK AT HOME USERS -- READ
1:57PM 0 Using * @ Home, all seems to work, but no sound to Softphone
1:43PM 3 With a phone system.
1:30PM 2 Outgoing Volume
1:04PM 1 Newbie question: How do I get Analog Phone to actuall ring
12:53PM 2 Asterisk@Home 0.7 released Question/Problem
12:49PM 2 IAX vs SIP (music on hold)
11:49AM 1 Voicemail sounds
9:47AM 8 Digium - Asterisk Download Ftp Site link Invalid
9:41AM 2 MeetMe flags in * 1.0.7
8:56AM 0 DTMF detection/generation
8:27AM 9 Dell 1750 & TDM400P - Power
8:15AM 0 adding extension ChanSpy
8:08AM 5 -lssl problem on debian
8:02AM 0 Partially receiving a fax
7:45AM 0 Manager API how see if call is on hold
7:18AM 0 Cisco 7970 Color
6:57AM 2 Call-ID and Unique-ID
6:40AM 3 constant ringing on Zap channels
6:08AM 0 rfc2833 cisco 7960 DTMF issue
6:08AM 4 No D-channels available!
6:01AM 11 Fail over
5:59AM 0 Zultys 4x5 phone
5:44AM 0 HFC PCI
5:38AM 8 Asterisk SMS configuration
5:04AM 1 HFC-S
4:33AM 1 app_darthvader.c?
4:28AM 0 Fw: sip provider
4:18AM 0 asterisk-oh323 pre-releases
4:10AM 0 changes to nat =yes?
4:09AM 0 ADTRAN TA 750 + TE405P + PRI with problem to receive or send fax.
3:36AM 16 VoIP Provider problems
3:24AM 5 Erratic CPU load
2:50AM 0 Outgoing call immediately disconnected
2:33AM 0 Asterisk as gateway with oh323 channel to VOIP provider that can provide gateway or gatekeeper feature ?
2:20AM 1 Sending many faxes simultaneously with spandsp
1:58AM 3 Zaptel based timing for VoIP-only Asterisk
1:56AM 2 Spandsp compilation error
1:48AM 2 Voicetronix OpenSwitch12 chan_vpb problem
12:22AM 1 sox
 
Monday March 28 2005
TimeRepliesSubject
11:37PM 6 Asterisk, SER, NAT, STUN and the whole debate
10:24PM 14 SPA-841 Call waiting?
9:49PM 1 Turnkey alternatives to fonality or switchvox?
8:18PM 11 Problem with 401 Unauthorized
7:58PM 1 Problem installing SpanDSP Makefile.patch
7:50PM 2 call center: agents, queues, sip
7:48PM 2 How to do something random?
7:18PM 3 spandsp rxfax under Linux 2.6 w/TDM400?
6:54PM 3 Kernel panic loading second fritz card
6:37PM 0 Asterisk@Home Handbook
6:35PM 25 Asterisk@Home 0.7 released
6:22PM 6 CIC Code
5:15PM 0 CDR to ODBC
4:53PM 13 Open Source Billing Software
4:22PM 10 First second choppy
4:17PM 2 Sounds gets choppy after 30 seconds
4:04PM 4 Debugging Asterisk in GDB (DDD)
4:02PM 0 MWI's for Third Party Softswitch
3:58PM 0 Teliann SIP Firmware
2:58PM 2 Start on system restart
2:55PM 13 call files run at certain times
2:32PM 2 RE: 8 channel fxo setup outgoing call problem (cont)
2:03PM 1 MWI and SIP PHones in Asterisk
1:29PM 0 8 channel fxo setup outgoing call
1:04PM 1 8 channel fxo setup outgoing call problem
12:43PM 12 Verizon ISDN
11:55AM 0 Re: Asterisk-Users Digest, Vol 8, Issue 229
11:20AM 1 voicemail sending blank .WAV file via email
11:19AM 3 problem with 1 dialing (recording says must dial 1 when I thought I did)
11:06AM 1 need to install the openline4 card
10:43AM 2 How to config speex?
10:29AM 3 Which analog phones to use and why?
10:06AM 2 Third party Firefly issue very weird??
10:02AM 1 Remove a channel from receiving inbound calls
9:46AM 0 bristuff-0.2.0-RC7k: error on loading qozap : "qozap: Unknown symbol zt_xxxxx"
9:39AM 10 AMP-1.10.007 Released!
7:53AM 3 CAPI/Dialing out
7:43AM 3 can a sip.conf stanza be shared by several phones?
7:34AM 1 Connecting quadbri to EuroISDN with 2 TE and 2 NT ports - what cables and settings ?
7:30AM 2 AGI STREAM FILE command
7:30AM 1 gsm player for Linux?
6:28AM 0 BroadVoice - "Failed to authenticate on INVITE" error
4:26AM 1 H323: g711-g729 transcoding
4:21AM 3 TDM04B doesn't hang up after Voicemail
2:51AM 0 Bug fixes IPSwitchBoard
12:57AM 1 spandsp-0.0.2pre11
12:32AM 0 Local/Remote * Servers, IAX/SIP mix and voice-mail notifications
 
Sunday March 27 2005
TimeRepliesSubject
11:48PM 2 apps api?
10:16PM 3 Broadvoice getting unregistered
7:57PM 9 pass caller ID to another application or machine.
7:55PM 3 How to park/transfer a call received from a Queue?
7:21PM 1 Strange problems IAX / Monitor / ChanSpy CVS HEAD
7:03PM 11 Asterisk on a dialup connection?
6:48PM 1 Quicknet phonejack connect to telephone line?
4:59PM 0 TDM11B and hook flash
3:11PM 2 Comedian Voicemail Issues
3:02PM 0 MOH Fixed
1:42PM 0 Re: Using call.sample on Zap hardware - Answering problem
12:40PM 0 Voicemail / Dial command issue
10:58AM 6 Music on Hold Broken??
10:28AM 0 analog phone
10:25AM 13 How to use multiple VOIP provider trunks
10:20AM 2 Asterisk and call delivery to connected PABX
9:38AM 0 -Using Reply To on Asterisk List-
9:34AM 0 3 Party Conference & ZapHFC
9:13AM 5 Asterisk and XLite on same machine (OSX)?
9:04AM 0 High Availability on Asterisk
8:55AM 2 sip provider
8:28AM 5 ata vs digium card
8:21AM 0 trying to add the free voipjet test to my
8:13AM 4 missing ring-tone
7:39AM 0 "Unable to get parameters" while configuring FXO cards, any ideas?
7:18AM 1 [Asterix-users] CISCO 7910
5:41AM 5 Can't get format_mp3 to work for music on hold
4:00AM 1 newbie install options
3:37AM 4 Can't Dial Out with TDM04B
1:21AM 11 Sipura 2000 x dual g729 channels x other choices?
1:14AM 6 NPA NXX
 
Saturday March 26 2005
TimeRepliesSubject
10:49PM 2 IPSwitchBoard new Release
10:16PM 0 Setup Zoom V3 Router + VoIP register with Asterisk
10:02PM 0 E1 ISDN Problem
9:58PM 5 Push VLAN to Polycom via DHCP
9:19PM 6 trying to add the free voipjet test to my asterisk at home???
8:38PM 8 context
8:16PM 0 Broadvoice audio problems
7:58PM 0 Echo on Zaptel hardware (Wildcard 100XP)
7:35PM 2 DTMF tones not working
7:01PM 1 AGI "STREAM FILE" issue
4:46PM 1 Soyo G668 + Asterisk
4:35PM 1 Transferred calls CDRs
4:24PM 5 Asterisk with Winmodem
3:53PM 2 about sip and registering
3:31PM 1 Dialout handler with/without leading 1
2:18PM 8 Click-to-Talk with Asterisk?
11:16AM 0 asterisk+voicetronix
8:03AM 0 Newbie Instalation
7:59AM 0 Zap keeps online if caller hangs up
7:16AM 11 Cisco's description of echo
6:59AM 0 ringing CAPI & SIP channels together
6:36AM 0 make a call based on SMS request
5:55AM 1 Major problems with TDM400 and specific
4:49AM 1 Cisco Phones with Asterisk
3:37AM 0 The Sound of Silence on TDM400P
 
Friday March 25 2005
TimeRepliesSubject
10:56PM 2 Asterisk as a dial in server for internet service?
10:53PM 7 Look at that Digium Broadband Modem!
10:47PM 7 Poor pstn line quality
10:10PM 0 OutBound call on Zap with Dial command
10:07PM 2 uniden voip gear
9:59PM 4 faxes
8:52PM 0 Re: Asterisk-Users Digest, Vol 8, Issue 216
8:36PM 2 MeetMe/Conference
8:31PM 6 800 numbers and FWD
7:45PM 0 New Warning in CVS: Format for authentication entry is user[:secret]@realm
6:14PM 5 Does asterisk@home 0.6 really work???
5:51PM 3 gnomemeeting / sip
5:38PM 5 911 & SoftHangup on SPA-3000
5:33PM 4 Openloop disconnect?
5:06PM 3 2 companies - one asterisk
4:38PM 0 Re: Asterisk-Users Digest, Vol 8, Issue 210
4:14PM 0 Remote MWI for Central Voicemail?
4:12PM 0 Re: Square key KTS app on *
4:04PM 2 Eicon DIVA PCI ISDN cards (not server) work withasterisk!
2:40PM 3 WaitExten question
2:14PM 0 OT: AstLinux 0.2.4 Released
1:41PM 1 X100P FXO card-No Dial Tone
1:29PM 0 Re: Two companies - One Asterisk
1:28PM 0 Asterisk vs silence detection
1:28PM 1 Re: Two companies - One Asterisk
12:54PM 2 Re: Two companies - One Asterisk
12:24PM 1 JUST NEED A REPLY
12:05PM 0 Calls from analog/FXS phone?
11:38AM 29 Asterisk compare with Skype
11:33AM 0 CAUTION: Re: grandstream firmware update 1.0.5.23
11:32AM 0 Re: Polycom phones-buggy SIP firmware or am Imissingsomething in the XML configs?
11:00AM 7 Two companies - One Asterisk???
10:24AM 1 We just released our new Asterisk Installation CD set. with 24/7 monitoring
10:23AM 2 Zap Detect called party pickup
10:21AM 1 Audio codec MP108
10:21AM 1 Asterisk@Home Usage
10:13AM 21 small qos switch
10:12AM 0 Outbound audio fades out with IAX Provider
9:57AM 2 Can I get a sip doorbell?
9:34AM 0 ways to get more accuracy from ztdummy
9:25AM 0 Problem with *72
9:12AM 13 Re-write callerid?
9:00AM 6 Square Key system
8:50AM 0 debugging trunks between two asterisk boxes at two different locations
8:47AM 0 Dial command problem(VOIP+*+TDM400P+Legacy PBX)
8:08AM 7 What is web login password for Asteirsk@Home
8:00AM 1 asterisk-addons and 64bit make
7:24AM 0 Re: Dial Out??
6:13AM 1 grandstream firmware update 1.0.5.23
5:38AM 0 re-write statement
5:25AM 2 MGCP issue
4:33AM 1 Hello Everyone
4:14AM 1 Does IAX supports silence suppression?
3:54AM 54 atxfer
1:57AM 2 peering
1:06AM 2 Forwarding to regular numbers?
1:00AM 2 Multiple outgoing calls through VOIP providers
12:33AM 1 Converting 7905G to SIP
 
Thursday March 24 2005
TimeRepliesSubject
11:35PM 2 voicemail problems with CVS-HEAD
11:14PM 5 Xten and NAt Problems
10:26PM 6 Advanced Cisco 7960 Config
9:53PM 1 Can I use my callscreen macro w/ sip?
9:43PM 0 Re: [2] X100p problem
9:27PM 1 Polycom phones-buggy SIP firmware or am I missingsomething in the XML configs?
8:38PM 3 Dynamically limiting the number of outbound calls
8:04PM 9 Backup for linux/asterisk
8:02PM 3 Emailed voicemail
7:27PM 2 SIP/iax routing question
7:26PM 1 realtime - unable to find key
7:08PM 1 Asterisk Hardware Requirements for a 50-100 Seat Call Center
6:57PM 2 Digium T1 Card Questions
6:51PM 2 Best Headsets for a Call Center Environment
6:38PM 0 SPA-3000 disconnect tone
6:21PM 1 Question on framerate
4:50PM 0 Native Bridging drops call on release
4:42PM 11 Forklift a 2000 phone PBX
4:21PM 0 Outlook contacts ->Asteriskdatabase(LookupCI DName)
4:17PM 0 chan_unistim compile failed
3:40PM 0 Any word on when CHanisAvail for IAX2 will be on CVS?
2:05PM 10 * -> SMS w/out PSTN
2:04PM 11 Parking
1:16PM 2 echo using Xlite
12:38PM 0 ChanSpy in CVS !
12:36PM 0 Asterisk@Home version 0.6 forwarding to pstn numbers?
12:28PM 10 Outlook contacts -> Asterisk database (LookupCIDName)
12:06PM 0 AGI commands STDOUT problem
12:00PM 0 IAXy dial tone problem
11:30AM 1 Error cannot record voicemail
10:54AM 0 NetHDLC + PRI
9:51AM 0 Monitor System for T1 failure.
9:24AM 5 Toll-free DID switchover: Get status?
9:24AM 0 No compatible codecs!
9:12AM 3 rxfax trouble on bristuffed capi
9:02AM 0 Properly setup SRV?
8:58AM 1 Third time, is it a charm?
8:57AM 0 Re: IP-500 config
8:50AM 0 Asterisk 1.0.3 Sipura codec error
8:36AM 4 Polycom DTMF
8:34AM 4 Newbie Voicemail Question
8:16AM 3 Question on routes
7:47AM 0 how to bridge two channels ?
7:19AM 2 Fun with CAPI
7:18AM 0 Echo on my TDM fxo
7:18AM 22 Realtime mysql problem?
7:18AM 5 When should I use SER ?
6:58AM 0 Tricky setup
6:34AM 0 "restart gracefully" fails
6:04AM 0 snom220 problem
5:33AM 0 Smal ofice pbx
5:30AM 3 Asterisk as Cisco Call-Manager - dial out to PSTN
4:45AM 21 Newbie pointers
4:16AM 2 RSA interasterisk IAX problems ?
4:10AM 3 Cisco 7905G Firmware
3:11AM 1 direct ip-to-ip call
2:53AM 8 Fax and Voice
2:34AM 0 R: music on hold error
2:23AM 0 Missing CDR data
1:49AM 0 Record(Sip)
1:37AM 3 codec for asterisk
1:21AM 2 Missing CallingPres Application
12:39AM 0 Is there a way to get inserted into an LEC's CLIDB? (fwd)
 
Wednesday March 23 2005
TimeRepliesSubject
11:45PM 1 help understanding sip header
11:33PM 3 WiFi SIP
9:55PM 0 calling an Application
9:31PM 1 VoiceMail Outgoing Calls and Disconnects
9:26PM 3 2 *@home issues away from bliss
8:21PM 0 How set language in Auto-dial out
8:20PM 0 How connect 2 extension by AGI
8:16PM 0 inquery auto monitor in 1.0.3
7:55PM 9 Nortel Option 11
7:13PM 0 Re: IP-500 config
6:53PM 1 Perform Action after X invalid tries
6:50PM 2 Spandsp question ( re: compiling )
6:35PM 0 [Fwd: [Soekris] net5801 & net7501]
6:22PM 0 Re: Asterisk-Users Digest, Vol 8, Issue 198
6:11PM 0 Asterisk Realtime.
6:11PM 1 PRI E1 Questions
6:05PM 0 Help, incoming lines problem!
5:51PM 5 *-1.0.7 DTFM => Not working
5:04PM 1 Polycom phones-buggy SIP firmware or am I missing something in the XML configs?
4:27PM 5 FW: polycom 500 help!!
4:26PM 2 audio outband bad quality
4:18PM 0 polycom 500 help!!
3:51PM 0 agi script for german date / time
3:39PM 0 ASTCC date format
3:20PM 1 cannot dial any extension except xlite
2:58PM 11 Vonage Linksys Router - Life after Vonage
2:54PM 2 Asterisk ChangeLog
2:53PM 0 Direct Dial Into ISDN Line
2:22PM 2 Rejecting ISDN-call without Answering
1:58PM 1 zaptel.o undefined references
1:37PM 0 Local sip client stuttered audio
1:03PM 0 Queue "has X calls (max Y)" problems with INSANE high numbers
12:47PM 4 Problem compiling asterisk-addons
12:30PM 2 Does X100P clone provide Timer?
12:10PM 1 prevent non-free calls
12:03PM 0 [Fwd: newbie DNS problem with BT100
12:00PM 1 call pick up and joining an active call
12:00PM 1 Access comedian mail from imap client
11:52AM 6 Problems with incoming calls
11:39AM 1 Zaphfc + PRI card problem
11:38AM 1 slim server for moh
11:34AM 0 I get "is on the phone" when the client is logged out
11:15AM 1 Multicall
11:15AM 8 Problem parsing unusual SIP/SDP
10:13AM 0 Random use of Sip peers
10:12AM 0 gnophone 0.2.4 and asterisk 1.0.6
10:08AM 1 SIP messagse
9:49AM 4 Need some help
9:22AM 2 Why even have set CallerID option?
9:18AM 5 Eicon DIVA PCI ISDN cards (not server) work with asterisk!
9:17AM 0 Settings to improve voice quality?
9:15AM 3 Where to put the modules to start on boot?
8:45AM 1 speex 1.1.7 crashes asterisk 1.0.6
8:43AM 2 Group channel rotation for outgoing call?
8:26AM 0 Re: [0] Wilcard X100P doesn't hang up when in Voicemail() and calling party hangs up.
7:59AM 7 Chanisavail and IAX2
7:54AM 0 MeetMe Upgrade !
7:06AM 0 Diva Server configuration
7:02AM 1 Wilcard X100P doesn't hang up when in Voicemail() and calling party hangs up.
6:53AM 1 SIP callid
6:53AM 1 * and Cisco Callmanager Interconnection
6:46AM 0 Using Asterisk as VMail server on CCM 3.3.3 System
6:31AM 14 Broadvoice alternatives
6:25AM 3 ADIT 600 "Dynamic Impedance matching"
6:17AM 2 GR-303 from Central Office supported?
6:05AM 0 read dtmf during dial
5:37AM 5 Reg Asterisk
5:32AM 1 snom 220 version
5:18AM 8 Any Software Echo Cancellation in Asterisk?
5:02AM 8 Playback of sound files but no sound
4:30AM 3 FXS FXO
3:30AM 0 Blog post on Asterisk setup
2:59AM 0 features enableing via database per extension number
2:42AM 0 Some audio problems
1:59AM 1 Asterisk Features/Dial Codes (Newbie question?)
1:49AM 1 BV Outbound Drop fixed .
1:38AM 0 Can I change the volume on a sip phone (Snom) from *?
1:23AM 0 Agents priority in queue
12:35AM 0 SIP behavior between different providers
12:11AM 0 Release 0.68 of IPSwitchBoard
 
Tuesday March 22 2005
TimeRepliesSubject
11:29PM 2 asterisk@home print incoming fax
10:56PM 11 Cisco 7940 and multiple simultaneous calls
10:05PM 0 troublshooting DTMF
9:13PM 1 newbe: help with registration
9:02PM 0 silence suppression
8:27PM 2 Asterisk locking up - 99.9% CPU
7:37PM 1 Help Debugging my code?
6:34PM 7 Incoming response and external access
6:17PM 2 asterisk-addons / OS X woes (continued)
5:59PM 6 Chanspy is back !
5:42PM 0 [Fwd: newbie DNS problem with BT100]
5:05PM 0 D() option on Dial
4:22PM 0 sip show peers weirdness
3:58PM 3 Digium support quality: Excellent
3:40PM 1 Mimicking Linksys PAP2?
3:39PM 1 Words of a user, ... what can I make better?
3:39PM 3 Is there a way to get inserted into an LEC's CLIDB?
3:37PM 0 RE: Asterisk-Users Digest, Vol 8, Issue 186
3:12PM 0 sip disconnects
3:06PM 1 Problems loading zapata module under suse 9.2 (cvs stable from 5 days ago) ?
2:38PM 5 TE405P and echo
2:24PM 1 Nat and firewall port forwarding - is it really required?
2:21PM 9 Is there a way to get inserted into an LEC's CLI DB?
1:52PM 1 Zap channels not hanging up...
1:38PM 3 IP PHONE with chip PA1688 and IAX2 Authentication
1:32PM 1 No recorded messages
1:27PM 4 Quick Newbie Question - Auto Call Routing
1:08PM 14 Major problems with TDM400 and specific telephones: suggestions?
12:55PM 0 "Success" report with TDM400 and Via EPIA-MII motherboard
12:55PM 0 "Succes" report for TDM400 and IBM Netfinity 5600
12:54PM 8 Help please for newb on Asterisk to Vonage
12:36PM 13 Rhino Channel Bank or ADIT 600
12:35PM 1 RE: Asterisk-Users Digest, Vol 8, Issue 152
12:34PM 4 multiline, cordless, expandable phone system and asterisk message waiting
12:12PM 3 audio delay in meetme conference using ztdummy
12:12PM 11 X100P interrupt load
11:48AM 0 RE: Asterisk-Users Digest, Vol 8, Issue 150
11:45AM 0 Still no Broadvoice Outbound. (Bump)
11:44AM 0 Callgroups Question
11:38AM 0 RE: Asterisk-Users Digest, Vol 8, Issue 150
10:52AM 1 How to mute a call
10:52AM 0 avm fritz 2.6
10:49AM 1 Reproducible echo on IAX calls to -some- destinations.
10:46AM 4 X100P voicemail volume too low (quiet)
10:41AM 1 Kernel 2.6.11
10:34AM 0 Problems using zaphfc and wct1xxp together...
10:08AM 4 OT: does Sipura SPA 3000 support UK caller id?
9:32AM 3 Setup to dial out only on voip (Broadvoice) not PSTN?
9:20AM 4 VOIP - Billing Solutions with Asterisk?
9:13AM 1 NEWBIE: MWI on 7960
9:01AM 1 *@Home .6 adding a outside number to a group{Scanned}
8:56AM 0 help with registration
8:55AM 0 RE: [Asterisk-uk] Meet
8:46AM 6 Setting MWI on legacy PBX
8:31AM 5 Experience with this radius?
7:42AM 1 Call Transfer Features
7:41AM 6 Enhanced 911
7:10AM 6 Asterisk - SS7 or ISDN
6:28AM 0 te110p sometimes green, sometimes stays red on stable cvs ?
6:28AM 7 IRQ headaches
6:21AM 0 In Call functions
6:00AM 2 Regex howto proof and change a dialed number
5:52AM 0 Phone book
4:59AM 3 SIP response *
4:16AM 0 [info] :: BIOS Motherboard Settings ::
4:07AM 2 bottlenecks
3:59AM 4 Asterisk-addons/OS X woes
3:53AM 0 asterisk + outlook + omniis TAPI driver
3:04AM 1 Call file misbehaviour
2:40AM 6 Feedback on CBMySql, MeetMe2 and web interface
2:35AM 1 RE: [Asterisk-Dev] Problem Making a SIP call over a long latencynetwork- Call rejected: 407 Proxy
2:06AM 5 Review: Asterisk at CeBIT 2005 / Asterisk at Linux-Tag 2005
1:51AM 1 H323 for Asterisk
1:43AM 0 [Asterisk-Dev] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
1:40AM 0 who has purchased a V400 card from Varion ? please help me .I need some document .
1:39AM 0 ANNOUNCEMENT : MeetMe - Web-MeetMe (throughmanager)
 
Monday March 21 2005
TimeRepliesSubject
11:56PM 2 DTMF is not working
11:27PM 1 ISDN EICON Cards
11:16PM 1 Flash pannel: time display
10:36PM 3 how to keep Asterisk up to date on many servers
10:26PM 1 SMS Alert Script - Voice e-mail
8:51PM 1 DISA Hangs up after DTMF is sent
8:14PM 0 asterisk-h323 and h323_id
7:10PM 1 Asterisk as test equipment
7:09PM 3 *@Home .6 adding a outside number to a group
7:07PM 2 Permission issue with outgoing calling
7:05PM 0 Ideas on how to make a script for using random zap channels
5:46PM 2 Asterisk, SER & Jabber
5:17PM 2 I need use sip
5:03PM 0 OSS and ALSA
4:04PM 6 Fax receive issues and NVFaxDetect
2:58PM 2 Hold Pickup
2:37PM 11 Compiling with gcc -shared on OS X
2:30PM 0 Micronet SP5001 ATA
2:06PM 2 Hitachi Cable WIP-5000?
1:53PM 5 Asterisk/Zaptel on Mac G5 or Xserve
1:52PM 0 SIP, NAT, and bindaddr
1:24PM 3 Flash hook & hangup problem
1:07PM 1 Net2Phone / Vonage
12:56PM 0 Compile error for minimal install of Redhat 9.0 [SOLVED]
12:54PM 4 Script to Authenticate User and Dial Out
12:40PM 7 chan-sccp-easter2005 make error with stable 1.0.6?
12:21PM 0 ANNOUNCEMENT : MeetMe - Web-MeetMe (through manager)
12:15PM 0 SIP Dial between two IAX-connected boxes?
12:03PM 4 Ext matching problems
11:53AM 4 Can't hear the caller
11:48AM 5 VoicePulse Issues
11:27AM 0 CallingCaed Application
11:25AM 0 audio frequency with wcfxs and K8t
11:01AM 69 why even use SIP
10:51AM 0 Jabber module for asterisk
10:37AM 0 Doubts Configuration SIP
10:20AM 1 Modify CallerID (on SIP phone) during call
9:52AM 2 iLBC codec and mute issues
9:52AM 0 Unable to get message on hold class to work
9:27AM 2 Why isasterisk's voice mail calledcomedian.
9:01AM 3 G726-16 passthrough...
8:50AM 1 Replacement 7960 Handset
8:09AM 3 US pstn => voip
8:09AM 0 astcc & sip
7:51AM 2 H323 gateway thru NAT
7:29AM 4 CallerID Name with IAX Providers
7:05AM 2 Version 0.67 of IPSwitchBoard Released
6:53AM 0 OT: "No authority found" connecting to Freshtel
6:39AM 0 asterisk outbound to SIP provider problems (still)
6:27AM 7 Why is asterisk's voice mail called comedian.
5:53AM 10 mpg123 home music from stream
4:12AM 1 IAX call rejected.....who was trying to reach 's@'
3:28AM 0 Cdr_odbc asterisk 1.0.6
2:12AM 1 DTMF doesn't seem to get through incoming ZAP channels
1:03AM 1 ASTCC: perl / mysql or me???
 
Sunday March 20 2005
TimeRepliesSubject
11:40PM 0 problems with SLES!
11:18PM 1 asterisk-1.0.7 make install on fedora corre 3 give errors
11:12PM 18 zaptel PRI drivers
10:17PM 1 HELP: Failed start after install asterisk_oh323-0.7.1
9:55PM 18 Soekris net4801 and analog interface?
9:50PM 0 want just few words from the list about SLES!
9:26PM 0 AVM Fritz! Noise / Crackle
8:23PM 0 X100P and Toshiba PBX
7:38PM 8 Choosing an ISP for Asterisk
7:17PM 2 NVBackgroundDetect
5:52PM 4 who has purchased a V400 card from Varion ?
5:46PM 12 Polycom dhcpd.conf? [Or, "Some day, I'll figure this all out."]
5:36PM 6 Follow-Me Script
4:02PM 0 H323 Gatekeeper Registering Question
3:52PM 0 rejected calls
2:59PM 0 i8253 count to high! resetting
2:40PM 1 app_nv_backgrounddetect - how to make module
2:38PM 3 FWD to Vonage not working?
2:33PM 5 TAPI
2:29PM 2 Problem transfering incoming calls
2:21PM 4 Dial from a URL - Possible?
2:19PM 4 OT: VIA Mini-ITX, Asterisk, and hardware
2:16PM 1 Limit incoming calls
1:30PM 0 Question on silcen aware
12:58PM 2 asterisk and outlook
11:10AM 0 Asterisk-addons 1.0.7
10:37AM 5 Cisco 7960 SIP boot takes 2 minutes?
9:59AM 1 Any experience with Dell 1850 Server with PERC 4e/Si
9:38AM 1 IAXY Polarity
8:59AM 5 wctdm fxs ring frequency
7:52AM 0 FW: Can't get more than one SIP device to be able to make outgoing calls
6:17AM 7 virus
5:20AM 5 ISDN-30 in UK
3:48AM 0 Outgoing Call problem with PSTN line
3:46AM 10 IPSwitchBoard-BETA Update
3:35AM 5 I cannot use G711 (ulaw|alaw)
3:00AM 1 softphone with web url support
1:43AM 7 Echo after upgrade * 1.05 -> 1.06
 
Saturday March 19 2005
TimeRepliesSubject
11:46PM 24 VoIP service through Asterisk?
8:30PM 3 Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
8:21PM 1 Ignore incoming calls on X100P
7:55PM 2 RE:Newbie question
6:44PM 1 vmware and asterisk
6:41PM 2 Problem with asterisk-addons/OS X
5:53PM 1 Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP
4:56PM 2 create distinctive ring on FXS
4:06PM 0 X-lite not hanging up / DTMF not present through voipuser.org
3:48PM 6 Asterisk and Cisco AS53xx/54xx Access Server Platform
3:24PM 3 More HEAD wierdness (chan_sip, jitterbuffer/PLC problems)
3:20PM 4 CallingCard Application
2:56PM 1 req: cisco 12sp+ firmware
2:32PM 5 Any Zaurus users??
2:14PM 0 mysql addon and cdr
1:55PM 3 What happened to www.iptel.org?
1:51PM 3 Question on routing table...
12:50PM 0 Polycom Callerid callback
12:36PM 0 DVG-1120S no call display name and time
12:11PM 2 DISA -> macro = congestion
11:37AM 2 MeetMe2 admin functions
10:48AM 1 ANI & DNIS sent to analog FXs Port Possible
10:42AM 2 Polycom Soundpoint boot ROM upgrade: how?
10:00AM 0 How to install /use festival on Asterisk
9:55AM 0 A couple of "dated" questions.
9:53AM 18 Any 24 (or 30) way FXS PCI cards?
9:45AM 2 * and DirecWay
9:06AM 4 ZapBarge restrictions?
9:02AM 3 Routing 911 calls
8:42AM 2 outbound delay
8:19AM 2 Tool for mysql
6:26AM 1 Areskicc installation problems
5:26AM 0 lost newbie requesting help for Asterisk Implementation
5:15AM 2 Goto and E1 line
3:38AM 1 Asterisk's on Suse Linux Enterprise Server(SLESv9)
2:40AM 2 :: What does it take to upgrade? :: Newbie Q ::
2:29AM 1 noice sip to sip only???
2:27AM 0 Excessive indications tone levels (longish)
2:10AM 0 Asterisk 1.0.7 chan_skinny fix port to chan_sccp?
1:46AM 0 my hfc card does not like Siemens
 
Friday March 18 2005
TimeRepliesSubject
11:47PM 5 Asterisk 1.0.7 Released
11:38PM 0 Yet another cisco 9760 7.x firmware failure
11:30PM 0 SIP-T support?
11:14PM 1 best protocol/codec for dialup
10:27PM 4 SIP <-> PSTN DTMF
10:26PM 3 GR303 with *
10:25PM 0 I look for some copartner
10:01PM 1 (no subject)
9:19PM 6 Article on Slashdot
8:41PM 0 Short burst of static then disconnect
7:27PM 3 current asterisk cvs problem with distinctive ring?
6:35PM 0 T100P: Can't Make/Receive Zap Calls (Long Newbie Blah)
6:27PM 1 Broadvoice hangs-up / disconnects after about 30 deconds
6:03PM 4 No sound when calling in from pstn
5:29PM 2 OT: Mexico area codes
4:36PM 2 Te110P initial installation problems ?
4:11PM 0 RXfax / Spandsp bad fax
4:02PM 0 Combine agent inbound and outbound
2:54PM 6 Registration issues with Sipura SPA-841
2:41PM 3 :: BIOS Motherboard Settings ::
1:35PM 1 X100P problem - no responce
1:22PM 6 small Local telco (wifi voip) some experiences with * ??
12:48PM 0 New Version of IPSwitchBoard-Beta
12:34PM 0 Rule of thumb rule for x/x => 1/1 billing
12:33PM 2 Configuring GnomeMeeting for Asterisk
12:09PM 0 Is this a BUG?? Please I need help in this
11:55AM 2 XML config files for Polycom SoundPoint IP 300?
11:41AM 3 Asterisk handling of SIP info
11:34AM 0 HELP: Dose G.729 with IPP only worked on IntelCPU?
11:05AM 0 IAX Peer/auth issues WAS: Netlogic inbound DID issue
10:58AM 3 HELP: Dose G.729 with IPP only worked on Intel CPU?
10:57AM 2 PSTN > Voicemail
10:45AM 1 Problem with Manager Interface
10:40AM 0 seg fault when accessing voicemail via any IAX softphone (diax, iax phone)
10:38AM 3 Some IAX questions
10:30AM 1 Looking for quality inbound DID - IAX providers, UK, USA, Australia
9:59AM 9 Optional URL in App. Queue
9:41AM 0 SNOM 190 Loud Ring While on Speaker
9:28AM 0 I4l + HiSax
9:26AM 1 call a url and get a result in the dialplan
9:22AM 8 echo / delay problem
9:15AM 4 reply a post
9:14AM 2 asterisk reload
8:23AM 1 newbe question sip.conf
7:45AM 0 Meetme2 compilation Err
7:40AM 3 BV This morning
7:38AM 22 Meetme2 compilation problem
7:17AM 6 TDM400P install problems
6:57AM 4 Which linux distribution
6:47AM 1 voicemail.conf extractor?
6:33AM 7 Parking a call in manager interface
6:31AM 0 AGI-like calls in the [globals] section
6:21AM 1 Manager API - Redirect command
6:08AM 2 Where to place calling rule contexts?
5:57AM 9 Group Ring after Timeout
5:40AM 3 Cisco 7940 convert to sip
4:38AM 2 Pattern matching in extensions.conf
2:03AM 5 ANNOUNCEMENT:Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules)
1:58AM 2 TDM400P Not loading Drivers
1:57AM 0 voicemail, busy does not work?
1:16AM 0 ISDN phone Hold-Problem connected to QuadBRI/Zap
 
Thursday March 17 2005
TimeRepliesSubject
11:56PM 1 Cisco 79XX Phones
11:16PM 1 limit about asterisk pstn out
11:11PM 2 gsm cannot be found in any file form... but it's there
11:02PM 5 ANNOUNCEMENT: Updatesforapp_cbmysqlandMeetMe2gui (out of tree modules)
10:56PM 1 leaky reload
10:08PM 14 About the weather..
10:01PM 2 Getting caller-name - cid_rewrite 1.0.0
9:36PM 2 Realtime - how to reload ?
9:02PM 0 seeking GSM 850/1900 gateway
9:00PM 0 Configuring Asterisk with BroadVoice
8:29PM 1 Extension ringing but no ringing sound asterisk
7:00PM 4 Newbie can't dial out to pstn
5:50PM 2 How to make Span Port Selection in "Round Robin" fashion?
5:44PM 4 X-Lite and Asterisk
5:31PM 4 Got 200 OK on REGISTER that isn't a register
5:30PM 1 Broadvoice solution finally
4:39PM 1 Database families and keys
4:16PM 1 My appologies
3:55PM 1 What cable to connect TE110P to telco PRI ?
3:26PM 0 Atxfer not working for called party
3:18PM 2 Backing up configurations and *@home list?
3:17PM 0 Message waiting/station busy conflict?
3:15PM 0 softphones and extensions status
3:12PM 1 IAX2 VOIP HARDPHONE
2:55PM 0 ASTCC dialstatus confusing billing issue
2:48PM 0 adding to asterisk db from a script
2:27PM 2 Agent won't log out!
1:48PM 1 What causes this changethread error message?
1:44PM 6 Undocumented "exten" syntax?
1:23PM 0 MOH and conference calls
1:05PM 1 Test post
12:30PM 0 Re: Last guy to get BV working outbound
12:23PM 2 Different codecs for different numbers via same IAX provider; how? Configs included.
12:17PM 5 Caller ID on E&M Wink
12:16PM 4 Phone ringing and not going to voicemail?
12:14PM 3 Searching the list archives
11:49AM 12 PRI Cause Code Help
11:34AM 1 PRI Test Equipment
11:31AM 1 Include/Macro not working right...
11:29AM 0 h323 problem loading
11:29AM 0 asterisk, callerid name and cisco 2600
11:22AM 6 Channel name (and substring)
11:08AM 13 Realtime Problem = Segmentation faults
11:00AM 5 OT: PC sound hardware for voice recording
10:53AM 5 Codec negociation
10:51AM 3 Redhat 9 Music on hold
10:35AM 1 Strange console call problem
10:08AM 0 MOH patch for bristuffed *
10:03AM 17 Polycom vs. Cisco IP Phones
10:02AM 3 Netlogic inbound DID issue
9:49AM 1 Asterisk dialplan (and/or VM) via LDAP
9:27AM 8 Last guy to get BV working outbound?
9:02AM 1 session border control
8:40AM 3 echo paid support
8:16AM 0 Re: chan_oh323.c ast_oh323_new Internal channel initialization failed [solved]
8:05AM 0 IAX2 Trunking, No connections any more...
7:37AM 1 Using Codec G-726
7:20AM 0 Chan_Spy and MOH - Any Status?
7:06AM 0 ztdummy - no sound in Asterisk@Home
7:03AM 1 Comparing Callmanager to Asterisk
7:03AM 0 Asterisk start problem (automatically)
6:53AM 1 Welltech Welgate 3804 FXO Configs
6:34AM 0 astguiclient error!
6:33AM 1 ZAp channel numbering question
6:32AM 1 Call Quality Detail Record
6:24AM 1 Asterisk with Cisco Call Manager
4:26AM 2 TE110p card with Euro ISDN (Ericsson switch)
4:21AM 4 Hi there..
4:02AM 3 Compilation problem chan_capi and Eicon Diva 4Bri
3:42AM 3 extension.conf dialplan
3:40AM 6 CAC Access Bank Manual
3:09AM 0 asterisk t.38 codec negotiation problems
1:35AM 3 ser+asterisk - security
1:24AM 1 Call Recording and Archiving
12:55AM 4 Snom190 intercom
12:18AM 3 HOW-To write an AGI
 
Wednesday March 16 2005
TimeRepliesSubject
11:53PM 1 who have been fabricated their own cards from Tormenta 2 PCI Card?
11:08PM 1 Pattern Matching?
11:04PM 1 How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to *
9:55PM 2 Connecting Multiple Asterisk Servers!
9:53PM 1 Hong Kong DID
9:52PM 2 music on hold error
9:25PM 0 FXS->FXO Converter??
8:20PM 2 Low cost hardware time for production environment
7:50PM 0 Email MWI Crashes Asterisk - Solved
6:37PM 0 Email MWI crashes Asterisk
6:24PM 1 Re: [Serusers] ser+asterisk - security
5:28PM 0 Problems with TDM400P and asterisk on Linux 2.6
4:55PM 5 Do you need to recompile the Linux 2.6 kernel for zaptel modules?
4:31PM 7 NuFone and CallerID
4:23PM 6 Dial multiple extensions, but different variables/timeouts
3:56PM 2 [Possible SPAM] : about sip, asterisk and cisco ccme
3:49PM 7 (Yet another) Music on hold problemand another...
3:44PM 2 Global Intercom on SIP phones
3:25PM 0 about sip, asterisk and cisco ccme
3:23PM 0 Obscure * command and audio questions
3:19PM 0 Realtime ODBC with cdr_odbc using the same database
3:10PM 8 t.38 support news?
3:10PM 0 OT: AstLinux mailing lists now available!
2:31PM 7 79xx 7-4
2:00PM 1 MGCP Channel Lockup and other probelms
1:54PM 0 Voice cutoffs
1:16PM 1 Pickup extensions for Zap channels does not work
1:00PM 2 ISDN Cards in the USA
12:52PM 0 Help with Audiocodes MP-108-FXO SIP Firmware
12:25PM 0 Mini Manual for IPSwitchBoard published
12:05PM 19 IAX Registration being lost
11:14AM 1 About shadydial
10:59AM 1 No ringing indication to radio phone
10:41AM 5 Asterisk Capabilities
10:36AM 0 Iax register
10:24AM 1 cisco 12sp+/30vip IP phone
9:47AM 0 Meetme doesn't react to DTMF keys
9:46AM 0 Asterisk retains DTMF Control Even whenanExternal IVR System is dialed
9:35AM 10 Asterisk E911?
8:35AM 3 Cisco gateways and hairpinning
8:20AM 7 TxFAX problem
8:04AM 3 Problem starting Asterisk - libssl.so.4 cannot be found
7:27AM 4 problem with musiconhold
7:14AM 0 Asterisk makes the news
6:41AM 0 Two (or more) Asterisk servers, routing calls
5:36AM 4 meetme2 compilation
5:33AM 3 CLI SIP Client
5:13AM 2 Basical question to asterisk
5:12AM 1 Problem with TE405P and Slackware 10.0 (reply this)
4:53AM 0 where is STUN implemented?
4:41AM 1 Re: chan_oh323.c ast_oh323_new Internal channel initialization failed
4:38AM 0 Call Forward
4:33AM 4 Error in placing call file in directory
4:28AM 31 IPSwitchBoard BETA
4:23AM 1 Kernel 2.4 or 2.6 for the latest asterisk ? ?
4:06AM 2 live monitoring of SIP calls chan_spy
3:20AM 6 AGI kill
3:15AM 0 Calling Card Application - which one ?
3:15AM 3 Calls from web interface
3:06AM 0 Help with simple H323 settings
2:07AM 0 chan_oh323.c:2501 ast_oh323_new: Internal channel initialization failed. Bad binary?
1:48AM 0 Agent groups broken in queues? (do not follow strategy)
12:51AM 0 IAX softphone on WinCE/PocketPC
12:34AM 0 Stable CVS or Head CVS for using TE110P ?
 
Tuesday March 15 2005
TimeRepliesSubject
11:54PM 0 ya newbie problem
11:37PM 0 how buy digium card such as TDM400.
11:30PM 0 Is my dialplan wrong?
10:16PM 0 Grandstream BETA Firmware
9:20PM 2 Grandstream and Transfers
9:07PM 1 Problem with presence
8:56PM 1 Background apps that plays music on hold
8:51PM 0 PCI 2.2 question
8:31PM 0 cdr issue
7:39PM 1 Automon Question
7:26PM 7 Voice getting cutoff
7:09PM 6 Flashpannel: How to get more than 28 buttons?
6:57PM 3 Unknown signalling 896?
5:40PM 2 Not ringing phone that are in use
5:02PM 0 SetDigitTimeout question
5:02PM 0 Cisco DTMF problem...
4:48PM 7 Asterisk@Home Install Problem
3:28PM 0 RE: can't hear anything on my side during a SIP call
3:16PM 2 Wiki down: Is there another source for documentation?
2:23PM 1 Which is the "newest" libpri/zaptel?
1:20PM 7 (Yet another) Music on hold problem and another...
1:17PM 0 zaphfc vs. i4l DTMF recognition
12:59PM 1 oh323 and open 729
12:13PM 2 Asterisk retains DTMF Control Even whenan External IVR System is dialed
11:49AM 1 How to see ExtensionStatus in manager
11:38AM 8 Realtime config
11:16AM 1 Asterisk retains DTMF Control Even when an External IVR System is dialed
11:09AM 3 How to connect with a headphone
11:04AM 5 FW: AntiSpam Alert from Rusten McKenzie
10:36AM 1 Learning the Ropes of *
10:34AM 0 Voicemail Question...help
10:19AM 1 Accecpt SIP calls from an IP
10:15AM 0 Incoming calls from Cisco 1760 given wrong context...
10:05AM 1 Transferring calls into MeetMe
9:56AM 11 Asterisk Newbie
8:30AM 6 Call Center software opensource or commercia l
8:29AM 1 Asterisk RealTime
8:29AM 1 Open ports?
8:21AM 3 Setting up Security Groups
8:20AM 2 fcpci - capi driver for Fritz
8:13AM 20 Call Center software opensource or commercial
8:00AM 4 Three way calling with X-Lite / MeetMe
7:38AM 1 PRI: Call Reference Length not supported
7:21AM 1 PRI Card TE110p Question
7:18AM 1 SIP & H323 gateway
7:18AM 0 Re: Asterisk-Users Digest, Vol 8, Issue 119
7:16AM 3 Web triggered calls
6:31AM 1 Site to Site Gateway
6:03AM 3 How to determine the voicemail file name for an AGI script
5:47AM 0 Zombie or soft hangup
5:19AM 4 Asterisk Queue strange behaviour
3:47AM 4 Kernel 2.4 or 2.6 for the latest asterisk ??
3:05AM 0 Forwarding SIP calls to proxy
2:53AM 6 Call Queues and Transfers
2:32AM 0 What different between asterisk-oh323 andastersk's chan_h323 ?
1:51AM 0 dial to h.323
1:43AM 1 Eclipse Plugin for managing Asterisk
1:02AM 2 Queue drop out into context not working?
1:00AM 1 blind xfer works atxfer doesn't...help!
12:39AM 0 trying to get trunk to register with * behind NAT
 
Monday March 14 2005
TimeRepliesSubject
10:48PM 0 Service similiar to VoicePulse-Connect?
10:31PM 1 Newbie - Config Problem ?
10:02PM 1 Creating IAXClient windows component
7:50PM 0 Multitech MVP130 as FXO with asterisk
7:31PM 7 Extentions Variable Dialing QUESTION.
6:57PM 2 I changed some minor things, but how can I contribute it?
6:22PM 6 asterisk-addons OS X
6:00PM 1 Anybody compiled ICD on Asterisk 1.0.6 release..
5:44PM 5 Problem Compiling Spandsp
5:11PM 1 LCR Question - Keep one trunk free
4:47PM 4 Sipura SIP vs. IAX
4:04PM 6 How NuFone.Net's customer service works.
4:01PM 1 meetme2 and meetme
3:46PM 8 Broadvoice's changes last week broke call forwarding
3:04PM 0 Dealing with bandwidth limitations
2:52PM 1 Rhino channel banks
2:49PM 1 VoIP Provider SIP Call Flow
2:37PM 0 dial out using sip via ZAP channel
1:58PM 6 FWD IAX Problem
1:39PM 3 insecure=very
1:31PM 0 Agents without agent channel
1:26PM 2 Broadvoice Busy Issue
12:52PM 3 TDM400P crackel
11:56AM 1 Has anybody tried NVFaxDetect Fax detection SIP/IAX
11:46AM 4 School design question
11:19AM 1 Setting NAT=yes for not NATed clients
10:51AM 0 Asterisk support for SIP REFER message
10:49AM 3 TDM400 audio problems
10:30AM 23 Skype - Bandwidth
10:21AM 3 qualify and NAT....
10:02AM 0 not ringing when place outgoing call
9:23AM 0 1.0.5 / 1.0.6 and oh323 compiling problem
9:17AM 64 Grandstream GXP-2000
8:47AM 1 OT: Recommendation for Dynamic DNS on Meshbox?
8:25AM 3 Has anybody experience with SetGroup / CheckGroup commands?
8:10AM 0 dial script, send variable problem??
8:04AM 1 DS3 with Asterisk
7:50AM 0 busy signal not in cdr
7:47AM 0 ASTCC - Are there some add ons available?
7:36AM 5 How to Flash() a modem line
6:34AM 2 colinux fresh install, zaptel does not compile, size_t error
6:20AM 6 Cisco 7960 SIP 7.4
5:38AM 0 asterisk codec negotiation problem
5:20AM 0 N/A
4:56AM 2 asterisk outbound to SIP provider problems
4:43AM 5 E1/T1 back to back ??
4:39AM 4 Problem with TE405P and Slackware 10.0
3:52AM 0 1.0.5 and h323 compiling problem
2:40AM 1 weird outbound problem through broadvoice (new)
1:50AM 1 snom 220 busy all the time
1:09AM 9 Voicemail SMS Alert - Possible?
 
Sunday March 13 2005
TimeRepliesSubject
11:57PM 0 Doubt about asterisk NOTIFY
10:33PM 0 fxo card not workin in susev9.2!
10:29PM 2 Asterisk, Voicetronix, and Australia
7:51PM 0 Commercial Asterisk Support? (Digium, etc.)
7:10PM 1 g729 Lic ordered from Digium Question.
4:28PM 5 cordless/wireless system with a ip base station?
3:09PM 4 IAX2 and asterisk servers linking to each other
1:21PM 0 Re: possible bug in chan_capi concerning context handling - SOLVED
1:14PM 4 SUSE 9.2 and Zaptel channels
1:07PM 1 sip.conf entry precedence
12:31PM 1 Running asterisk as non-root: Zaptel Permission Probs
12:23PM 2 PRI Call Reference Length not Supported
11:25AM 0 ASTCC Functions
10:47AM 30 Text Messaging or AIM
10:33AM 1 ASTCC sounds
9:55AM 0 looking for DID in spain
9:17AM 7 ASTCC - how to use different brands?
9:17AM 0 Re: Asterisk-Users Digest, Vol 8, Issue 105
9:01AM 0 IAX2 and server links
7:57AM 3 sending a DTMF tone before hangup
5:31AM 4 Sipura 841 issues
4:21AM 7 newbie uk questions...
4:15AM 0 safe_asterisk doesn't restart when called by initlog in fedora
3:21AM 6 possible bug in chan_capi concerning context handling
3:18AM 4 Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues
3:08AM 0 Zaptel problems, Asterisk 1.0.6
3:01AM 2 How can I eveluate trailing numbers in extensions.conf?
1:04AM 0 What different between asterisk-oh323 and astersk's chan_h323 ?
 
Saturday March 12 2005
TimeRepliesSubject
10:24PM 0 ASTCC or should I use something elsefor different rates, depending on the calling card?
10:12PM 0 vars for transfered calls
9:19PM 2 RE: [Asterisk-Dev] SetVarCDR
8:35PM 7 chat line
8:06PM 0 Hang on "making progrogress passing" when dialing out
6:31PM 0 Voice Based Bulletin Board.
6:08PM 1 playing "invalid" to an internal user
5:57PM 0 Does zapateller work in Australia?
5:55PM 0 Question on phones with asterisk
5:47PM 0 Problem with ability to dial out when a channel is used from an external equipment in a point to multi point configuration
4:28PM 1 Asterisk with Skype
4:17PM 1 RE: Asterisk-Users Digest, Vol 8, Issue 88
4:16PM 0 after *-1.0.6 upgrade error: vm_execmain: Unable to read password
3:46PM 3 DISREGARD!! Broadvoice outgoing problems
3:42PM 1 Broadvoice outgoing problems
2:18PM 0 Tracking/Billing Incoming & Outgoing Minutes?
1:50PM 1 Zapping around
1:08PM 0 Where to download the asterisk-oh323?
11:37AM 0 How do I pick up a trailing number in extensions.conf?
11:30AM 15 Advanced conference features, meetme2?
9:33AM 5 checking active SIP members of a queue?
9:31AM 2 ASTCC - Regex: How to "Country" but "special City" different?
9:03AM 0 Looking for an Asterisk Expert/Partner for project
8:04AM 4 Unable to create channel of type 'IAX2'
7:10AM 0 Sipura 2100 and Asterisk one-way audio
5:28AM 2 Signaling on PRI channels
3:59AM 1 ATA 186 Codec Question.
2:57AM 1 X-Lite and * SIP Problem
2:38AM 0 IAX2 Sphone for PocketPC
2:19AM 2 SIP monitor thread is hanged up on a uClinux embeded linux system
1:19AM 1 Simultaneous call to both phones in PAP2-NA
12:58AM 1 ipvolution TDM cards - vaporware?
 
Friday March 11 2005
TimeRepliesSubject
11:30PM 0 SineApps Daily Asterisk News Back Up
10:52PM 4 ASTCC or should I use something else for different rates, depending on the calling card?
9:10PM 0 Sipura 2100 and Asterisk - Faxing
8:49PM 1 digium card
7:49PM 0 Error cant change devie with no technology
6:51PM 0 Re: [Asterisk-biz] Opportunities for good billing solutions
5:49PM 3 SIP-B?
4:30PM 2 Asterisk, IAX2 and iptables
4:08PM 3 DVG-1120 questions
3:42PM 1 CNAM for Asterisk
3:32PM 0 Re: Incoming echo cancel
3:17PM 0 Re: Incoming echo cancel
3:08PM 0 Receiving faxes via SIP
2:56PM 1 Call Transfers
2:14PM 0 SIP -> NAT -> *
2:13PM 9 Parked Call
2:12PM 6 Droping calls
2:00PM 0 Errors using Asterisk as Sip Client behind SER !!!
1:53PM 7 Re: Incoming echo cancel
1:52PM 0 Sipura 2100 and Asterisk and Fax
1:36PM 1 TDM04B lock up
1:15PM 3 Trouble with Realtime
1:08PM 0 Polycom ip600 - how to eliminate echo?
12:27PM 7 VoipJet Terms of Service
12:21PM 9 Sip show registry returning nothing
12:18PM 3 Asterisk Billing System
11:47AM 21 No ringback over IAX - LiveVoip
11:42AM 5 Wireless VoIP
11:38AM 1 agents - queue config
11:15AM 0 Is it an AGI bug in 1.06? IAX Calls going towrong extension with AGI.
11:06AM 12 Asterisk security problem: authorized SIP users can fake any callerid!
10:50AM 2 diffrent area codes for diffrent phones in dialplan
10:33AM 22 Vonage a provider?
10:26AM 1 Is it an AGI bug in 1.06? IAX Calls going to wrong extension with AGI.
10:16AM 6 FC3 Dual Xeon Zaptel PANIC
10:07AM 34 Realtime does not work yet, ...
9:40AM 3 PAP2-NA point to poitn calls ??...(Direct IP Dialing)
9:32AM 0 Festival & Asterisk CVS Head
9:13AM 5 ASTCC and NuFone billing is different!!
8:58AM 4 Phone suggestions
8:45AM 2 1.0.6 music on hold bug ?!
8:27AM 0 CAPI- 2 Cards
8:20AM 7 EADS6550 and asterisk - echo on PSTN call
8:16AM 8 Multiple IAX Phones Behind NAT
7:32AM 1 Am i right by Asterisk?
6:57AM 0 Quescom AS/400 GSM Gateway + Asterisk
6:49AM 2 What is that area code?
6:47AM 1 Some Hardware Advice
6:36AM 1 Manager (5038)
6:16AM 0 Asterisk@home 0.6 + bristuff
6:12AM 1 IAX, double NAT
6:12AM 0 One single record file for a meetme monitor?
5:09AM 0 SIP Phone Unreachable
5:05AM 1 SIP signalling and RTP to different servers
4:58AM 2 CDR database
4:47AM 0 Asterisk@home 0.6 + Modem.conf
4:32AM 1 Unable to create Zap channel when dialing using a bri cellular gateway
4:20AM 1 Incomplete incoming fax using spandsp 0.0.2pre10
4:11AM 1 QuadBRI ,TDM400 and SuSE9.2 (Sencond try)
4:02AM 1 Asterisk + Call hangup
3:50AM 0 Intermittent volume deterioration in conferences
3:47AM 0 FW: IAX Settings
3:27AM 12 Load Balancing b/w 2 asterisk servers using SIP load balancer
3:15AM 1 NuFone Configuration [problem]
3:12AM 1 TE110P experiance
2:08AM 1 from sip to asterisk to h323..how
12:07AM 3 How to register two SIP phones ( e.g. Windows Messenger) from different subnet to *
 
Thursday March 10 2005
TimeRepliesSubject
11:59PM 4 AAH 0.06 - IAX Connection Over NAT Firewall
11:33PM 0 SIP to H.323 no audio
11:22PM 3 E1 LED not lighting up....
10:17PM 1 what is best free softphone.
9:59PM 8 Panasonic TDA200 E1 -> E100P negotiation issues
9:05PM 10 Bandwidth
8:22PM 4 SetCallerID({$NEWCALLERID})
8:20PM 2 multiple enum results
7:32PM 0 One way speech from H.323 incoming calls, but outgoing calls are OK.
6:47PM 0 iconnect here, inbound yes, outbound no
5:26PM 0 Broadvoice Config proper??
5:11PM 2 Transfering calls or using any feature
5:01PM 4 Application SetVarCDR
4:50PM 0 Asterisk@Home - Email to Fax
4:40PM 0 zaptel configuration issues (zaptel.conf vs.zapata.conf)
4:39PM 3 Cisco and Asterisk
4:32PM 2 Asterisk@Home, AMP, and Broadvoice
4:25PM 0 WOW: solved (was: compiling and ssl)
4:12PM 0 MINNESOTA: TwinCities Asterisk Users Group
4:09PM 4 Suse Compiling: next err
3:47PM 1 Odd problem with asterisk
3:37PM 1 Re: Polycom phones do not talk to each other
3:14PM 0 RE: Asterisk-Users Digest, Vol 8, Issue 83
3:07PM 0 Re: Polycom phones do not talk to each other
3:02PM 0 7905 example configs
2:52PM 0 Re: Polycom phones do not talk to each other
2:46PM 1 Polycom phones do not talk to each other andcannot answer when we pickup
2:14PM 1 what replaced app_qcall?
2:08PM 3 Polycom phones do not talk to each other and cannot answer when we pickup
2:07PM 14 IAX2 800 Termination
1:34PM 3 SIP, DNIS, Asterisk
1:10PM 2 QuadBRI ,TDM400 and SuSE9.2
1:04PM 0 Manager Redirect fails on Zap Channels
12:57PM 0 Vonage down in Dallas?
12:48PM 0 Astrisk to legacy Mitel SX2000
12:27PM 2 Listeners in SIP conferences
12:22PM 3 Re: Paging using multiple sound cards
12:03PM 10 asterisk and Broadvoice Outgoing Again :(
11:52AM 2 ***SOLVED*** Broadvoice latest changes andstillnot working- An Additional Server****Solved*****!
11:31AM 3 Re: Do I Need Astrisk
11:11AM 4 OT: AstLinux 0.2.2 released
11:08AM 2 Windows messenger 4.7.3001 does not have Account Tab ?
10:26AM 3 Pictures from the Asterisk Pavilion at Spring VON 2005
10:13AM 3 Asterisk and USB ISDN controllers ...
9:56AM 0 Re: Message Waiting over a IAX trunk
9:40AM 0 OH323 - compilation error (another user, another error)
9:23AM 1 Xlite dont ring on Asterisk
9:17AM 2 NVFaxDetect errors on make
9:10AM 4 tdm400p and dell 2600 poweredge
8:53AM 0 bypassing auth info
8:32AM 0 Avoiding connect signal in two stage dialing
8:22AM 2 hide callerid via presention bits of ISDN
8:06AM 9 NuFone
7:54AM 0 ASTCC - regexpression for country and certain cities?
7:37AM 0 Broadvoice busy message every couple of days.
7:20AM 3 Delay on outgoing calls
7:07AM 6 Location of Voice e-mail Code???
7:02AM 0 BRI: "Unable to create channel of type 'ZAP'"
6:53AM 1 a liitle bit of info required
6:39AM 1 Problem with incoming calls.
6:27AM 0 New Integrics tip: VoIP for ISPs
5:47AM 4 Compiling Asterisk On SUSE 9.2
4:52AM 0 Problem with NOTIFY
4:48AM 1 FWDout credits sharing
4:08AM 2 OT: Active channels bridging with SNOM190
3:42AM 0 Calls hang in a conversation
2:51AM 2 Cisco 7940/60 and 802.3af PoE
1:32AM 0 ISDN to SIP
1:09AM 5 OT: Zap channels intermittently bridging with SNOM190
12:47AM 1 iax,trunking,zap
12:40AM 2 Single port S0 ISDN card to use in Greece
12:32AM 0 Upgraded to Asterisk 1.0.6 now crashes on boot, sql issue?
 
Wednesday March 9 2005
TimeRepliesSubject
11:46PM 1 Asterisk & NOTIFY problem
10:45PM 1 Paging and Intercom using Sipura SPA-841
10:44PM 6 Where can I find all areacodes for USA (accounting purpose)
10:31PM 2 Apple links Asterisk
10:09PM 1 Asterisk@Home Installation Problems
9:36PM 1 Sangoma and other ISA T1 cards
9:26PM 0 Call Parking issue
9:25PM 6 Comparison Charts
8:39PM 2 Voicemail Rap
8:35PM 3 Slightly OT - Snom 190 function keys via subscribed config
7:55PM 7 Asterisk-oh323-0.7.1 compile error
5:53PM 6 Broadvoice Multiple "lines"
5:18PM 2 Server specifications
4:55PM 1 ODBC error ?
4:12PM 3 Asterisk@home silly problem, please help!
3:59PM 1 Support for SIP REFER message
3:52PM 4 Problems with new install voicemail broadcast
3:37PM 0 Problem in Configuring Asterisk Server
3:25PM 3 zaptel configuration issues (zaptel.conf vs. zapata.conf)
3:25PM 1 Paging using multiple sound cards/channels
3:11PM 0 OT: Any interest in Line Powered Amplifiers?
2:46PM 1 Tired of trying to fix this echo problem
2:15PM 1 can I use an external modem such as USR robotics V92
2:13PM 13 VoIPJet
2:06PM 4 voicepulse "silence" during conversations
2:05PM 2 Providing a dialtone
1:54PM 2 Call Progress Analysis
1:43PM 0 New astGUIclient version released 1.1.0
1:21PM 0 IPH-90 and Asterisk , MGCP
12:50PM 86 OT: Best DB
12:43PM 0 joinempty=no
12:10PM 0 [Asterisk-Dev] 1.0.7 Release Candidate
12:08PM 7 Upgrading Asterisk
11:48AM 0 Specifing a linker when building Asterisk
11:41AM 3 Broadvoice latest changes and still not working-An
11:23AM 4 Assistance with Overhead Paging
11:02AM 1 max number of conference rooms, and max number of conference callers in one room
10:16AM 0 Fwd: Re: Broadvoice latest changes and still not working- An Additional Server ****SOLVED****
10:12AM 1 Edit MGCP response
10:01AM 4 Polycom IP 500 bitmaps and Idle Display Animation
9:48AM 0 sip hangup detection problem
9:37AM 1 Asterisk 1.0.6 and chan_sccp problems?
9:31AM 10 Print-to-Fax client
9:07AM 1 Cisco 7960 Protocol Invalid when Upgrading to 7.3
8:47AM 3 Telecom echo cancel disable
8:15AM 0 RE: : RE: Re: MGCP to Inter Tel system
7:36AM 0 iaxy stopped working
7:33AM 0 Unable to dial out using HFC ISDN card
6:59AM 3 Which hardware for this solution?
6:47AM 0 Cicso 7912 phones 3 out of 8 not grabbingthegk<MAC> file
6:42AM 1 Broadvoice latest changes and still not working-An Additional Server
6:41AM 7 NuFone + VoIPJet = busy busy busy
6:39AM 0 Zyxel P2000W - CallerId
6:34AM 2 TDM400P slow getting line tone
6:19AM 3 Echo for first 15 to 20 seconds
5:31AM 9 Which box?
5:26AM 5 Regarding Incoming Calls on PRI
4:51AM 2 Should ICMP port unreachable generate a BYE request?
4:30AM 0 Call through. with 2xT1 .configuration
4:26AM 1 IAX Music on hold
4:14AM 17 how to sip->h323 using asterisk-oh323-0.7.1
4:03AM 0 Asteriks@home
3:24AM 2 Voicemail - No Audio Output!
3:19AM 0 Subject: Re: What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile
12:38AM 4 Another Newbie Question
12:13AM 2 i am missing something!
 
Tuesday March 8 2005
TimeRepliesSubject
11:46PM 2 Connect asterisk on classic pbx with T100P card
10:44PM 14 Broadvoice latest changes and still not working- An Additional Server
8:47PM 4 New Help Site - cut down on Mailing List questions
7:29PM 0 Could dialing long extensions be a problem?
7:20PM 1 All Circuits are Busy Now
7:12PM 1 Dial() out and offer a menu system
7:05PM 6 Polycom IP600 Phantom Ringing
6:38PM 3 Broadvoice-like company in Canada?
6:31PM 0 Sip 400 bad request - broadvoice error
5:05PM 2 STOP NOW not responding
4:56PM 1 Voicetronix Tones
3:34PM 1 Cicso 7912 phones 3 out of 8 not grabbing thegk<MAC> file
3:08PM 1 Cicso 7912 phones 3 out of 8 not grabbing the gk<MAC> file
2:38PM 0 Re: Asterisk-Users Digest, Vol 8, Issue 63
2:30PM 4 Nortel ATA not passing dtmf tones to fxo
2:25PM 3 Please help with install * SOLVED
2:18PM 3 Broadvoice users...
2:15PM 2 Asterisk Interop w/ Level 3
2:04PM 7 DID in the U.S.
1:49PM 0 Broadvoice latest changes and still not working - solved HEYYY
1:36PM 2 Forwarded call flag
1:27PM 0 Play music on hold while waiting for DTMF?
1:24PM 1 SIP - Call Park/Pickup and Canreinvite=yes at the same time??
1:22PM 0 determining an available channel question
1:07PM 2 Adit 600 for asterisk
12:55PM 3 Cisco 7940 Upgrade Failing
12:46PM 2 GotoIf with Authenticate
12:16PM 6 DTMF out to Cell Phone
12:09PM 0 Does anybody have Broadvoice outbound working?
11:50AM 2 7960 Dies when network cable connected
11:29AM 20 GotoIf problem
10:56AM 11 Wildcard X100P or TDM400P?
10:45AM 4 force SIP authentication
10:45AM 2 Incoming Fax Service question
10:19AM 6 using the i extension
10:09AM 3 zaphfc error
9:49AM 2 Asterisk Management API
9:13AM 1 How does asterisk do the routing?
9:02AM 0 2 Asterisk servers (IAX) behind one firewall
8:45AM 13 Please help with install *
8:30AM 1 Asterisk provides ring tone?
7:57AM 0 problem in compiling chan_mISDN
7:45AM 2 problem in compiling openh323
7:10AM 17 Broadvoice latest changes and still not working
6:49AM 11 NAT Far End Traversal
6:15AM 1 CallerID - Broadvoice vs. VoicePulse
6:04AM 1 TDM22B in the UK on BT
5:35AM 0 xc-ast 0.8.0 is out
5:06AM 2 Queue and SetGroup
3:31AM 2 Retreiving the called number
2:43AM 2 looking for cheap 4 port FXS card
12:08AM 3 Cisco 7960 Problem - Phone Unprovisioned
 
Monday March 7 2005
TimeRepliesSubject
11:38PM 0 How work by Asterisk and SER ?
10:15PM 1 What combination of pwlib and openh323 are
9:38PM 1 video confrencing
9:25PM 3 UNISTIM channel driver available
8:41PM 1 What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile
8:19PM 0 SpanDSP Status Messages
7:58PM 2 [Asterisk-Dev] TE405P/410P (Quad-T1/E1) driver
5:50PM 4 [Fwd: RE: Re: TE410P card in an HP-Compaq DL380 G4 server]
5:42PM 11 Question with email notification
4:52PM 4 Polycom SP300 questions
4:49PM 0 CAPI trunks
4:31PM 0 Dock-n-talk connection to asterisk
3:36PM 7 [Asterisk-Dev] Flash Operator Panel
3:22PM 0 Asterisk@Home and VoiceMail
3:21PM 3 grandstream budgetone 101
3:21PM 10 Question about AGI vs. FastAGI vs. straight C/DB development
3:14PM 19 Call Forward or DND
2:19PM 1 working system for months suddenly stopped today with Failed to authenticate on INVITE to - additional
1:57PM 0 Dial, record, save to voicemail
1:56PM 3 [Asterisk-Dev] Polycom IP 600 XML
1:20PM 1 FXO module in TDM400P (UK, BT) - Hangup
1:05PM 10 Asterisk & MySQL Blobs
1:00PM 5 multiple outside phones
11:50AM 3 Setting up asterisk with current PBX?
11:39AM 1 working system for months suddenly stopped today with Failed to authenticate on INVITE to
11:23AM 21 Tweaking AGGRESSIVE_SUPPRESSOR
11:02AM 2 DTMF to Email
10:42AM 1 3COM 3101 SIP
10:41AM 0 iax2 setvars help needed
10:00AM 2 CAPI questions
9:59AM 0 anybody tried Fujitsu-Siemens PRIMERGY RX200 S2 server width te4xx?
8:40AM 2 MP3 stream for MOH
7:56AM 2 2-Ring Delay for CLID
7:31AM 1 chan_sip not 100% RFC3665 compliant - re-REGISTERsfail.
7:27AM 2 SIP and ISDN
7:08AM 2 MGCP howto
7:03AM 2 Where to get (cheap) VoIP
6:54AM 0 chan_sip not 100% RFC3665 compliant - re-REGISTERs fail.
6:17AM 0 Open files / socket leak
6:10AM 0 CVS compile error utils.c
4:25AM 0 Asterisk & Fritz & Capi & isdn PBX integration : Can I dial out on any MSN I declare ?
3:44AM 2 Call transfer questions
3:08AM 2 Exec AGI after hangup.
2:47AM 0 Sip phone service for linux
2:40AM 1 Custom Development
2:10AM 0 DID Functionality with POTS and Digium TDM04B
1:31AM 0 SIP URI
 
Sunday March 6 2005
TimeRepliesSubject
10:16PM 0 How to configure directory within voicemail transfer to search by first name?
10:00PM 0 Zaptel in New Zealand: Caller id vs loadzone
9:18PM 0 Signate is now offering the dCAP test.
6:48PM 0 Loopback
5:39PM 0 Re: Broadvoice configuration changes for outbound calls
5:15PM 0 [Fwd: Re: BroadVoice configuration changes for Outbound]
4:31PM 1 Re: [Asterisk-biz] Livevoip U.S. 800 LNP Starts March 9th 2005
4:22PM 1 IP Providers pass CallerID?
4:09PM 1 SER -> Asterisk voicemail on busy/unavailable. Anyone did it? (googling says NO)
4:02PM 3 Music Volume ?
3:37PM 5 Trying to get 2 SIP phones to work
1:42PM 1 SpanDSP: Training failed (sequence failed)
1:41PM 5 Need help on * anf HFC.
1:27PM 3 SNMP and Astersik
1:00PM 1 Dial option g
9:15AM 6 Zaptel.conf and multiple T1 woes
7:12AM 7 SJphone on PDA registering with Asterisk???
6:31AM 0 Dial Macro
1:20AM 1 IAX - Registration Problems
 
Saturday March 5 2005
TimeRepliesSubject
11:28PM 0 Budgetone 101 Hold/Xfer/Conf/Flash
8:13PM 0 Is anybody having problems with sixtel?
6:00PM 6 Survey: what's the best HTTPd/TFTPd/FTPd to serve up configuration files to sets
5:14PM 2 SIP VoIP Provider problems
4:57PM 0 Asterisk patches - location and use
4:34PM 3 Sayson 480i Fails to Re-register?
3:26PM 1 IAX2 (Variables)
2:40PM 0 DVG-1120M -> S
2:30PM 4 Digium hardware in the UK ?
2:17PM 1 Zultys Zip 2
2:10PM 1 Digium Reseller in the UK ?
1:08PM 1 Dead SCCP client since upgrade to Asterisk 1.0.6-BRIstuffed-0.2.0-RC7j?
1:00PM 0 Capi installation with Fedora Core 3 (AVM Fritz!)
12:57PM 2 IAX Softphones
11:40AM 0 How do I reload extensions included in a switch statement in extensions.conf?
10:50AM 3 Sorry to be a bother ISO root password
10:16AM 3 Asterisk for Live-Stream?
10:13AM 37 BroadVoice configuration changes for Outbound
9:05AM 0 how to optimize sip??
7:34AM 2 Block anonymous calls
7:31AM 4 Newbie guidance requested --- Grandstream Budgetone
7:06AM 0 Unable to transfer timed out calls from call parking
7:04AM 1 Problem with loging on guest account
6:50AM 0 signaling problems
6:19AM 5 Getting asterisk-addons installed on Debian?
6:14AM 25 X100P Clone, Which one?
5:57AM 0 Automatically send monitored call files by e-mail
4:58AM 0 change proxy after timeout
4:18AM 0 Asterisk 1.0.3 Periodically Fails Registrations
4:16AM 0 Re: Is anyone using asterisk in a small call
3:25AM 1 SAY DIGITS problem
2:19AM 3 cant compile app_meetme2
2:15AM 0 ASTCC questions: Userconfig, sip friends, iax friends and multiple trunks in routes
1:17AM 0 Are codec "capabilities bitmasks" different in IAX and SIP?
12:54AM 3 Unable to create channel of type IAX2
 
Friday March 4 2005
TimeRepliesSubject
11:58PM 4 Difference between Snom 190 & Elmeg 290?
9:57PM 2 Asterisk Brochure
9:30PM 2 ANNOUNCEMENT: Updates for app_cbmysqlandMeetMe2gui (out of tree modules)
8:35PM 0 ANNOUNCEMENT: Updates for app_cbmysql andMeetMe2gui (out of tree modules)
8:18PM 31 LiveVoIP Problems?
8:10PM 8 Stutter Tone
8:03PM 5 ANNOUNCEMENT: Updates for app_cbmysql and MeetMe2gui (out of tree modules)
8:00PM 9 Log Error
7:17PM 3 ANNOUNCEMENT: Updates for app_cbmysql and MeetMe2 gui (out of tree modules)
6:32PM 0 Asterisk with mediant 2000 - facing problems
4:42PM 3 Multiple telephone participants
4:07PM 5 music on hold issue
3:51PM 0 Chan_Capi + HFC Card
3:41PM 0 Size of installations
3:37PM 3 Is anyone using asterisk in a small call center
3:27PM 5 PRI HDLC Abort (6) Errors
3:18PM 0 TE405P and quality problem
2:34PM 6 Im a noob
2:21PM 2 Placing a call from command line and passingit to an extension if connected - Is it possible?
2:08PM 1 chan_h323 & codecs
2:00PM 0 Monitor Application with Queued calls
1:58PM 3 IAX Codec
1:55PM 1 Has anyone got early dial working on asterisk ?
1:45PM 3 Options for Attendant Console.
1:44PM 3 Voice over Frame Relay & Asterisk
1:41PM 3 Placing a call from command line and passing it to an extension if connected - Is it possible?
1:39PM 2 Asterisk box and verizon calling it
1:26PM 9 Hardphone deployment recommendation
1:08PM 0 chan_capi patch for the new cvs HEAD
1:01PM 4 SIP MWI and MySQL Realtime
12:43PM 1 Web based tool asterisk real time
12:24PM 3 Bluetooth phone as SIP handset?
11:24AM 3 ANNOUNCEMENT : Asterisk-Stat V2.0 - CDR Analyser
10:53AM 7 budgetphone
9:53AM 0 SMS in 1.0.6
9:33AM 1 defold usernames in asterisk@home version 6
9:04AM 7 [OT] - Why should I answer a Newbie questio n,therethick!
9:01AM 1 [OT] - Why should I answer a Newbie questio n, therethick!
9:00AM 2 IAX on netweb EEZEE phone
8:48AM 3 X100P in the UK - seems to short the dialtone
8:38AM 2 Broadvoice + incoming call works only for ~2 minutes
7:54AM 0 Asterisk ---Toshiba
7:31AM 2 Problem getting Voice Contract script to work
7:02AM 0 Connection time of Transferred Calls
6:23AM 0 Problem with inbound call quality.
6:01AM 12 TE110P module woes
5:51AM 1 chan_capi with patch compilation error
5:48AM 1 mISDN not initialising properly my Fritz cards
5:29AM 6 Problems with g729 codec
4:06AM 1 Zap channels intermittently bridging with SNOM190
3:52AM 2 Answering Machine Detection with app_machinedetect.c
3:27AM 0 * intergation with Panasonic D500 and strange echo
3:27AM 1 Asterisk@home 0.6 + mISDN
3:02AM 2 Bristuff e RealTime: STABLE vs. CVS-HEAD
2:55AM 0 SIP hard phones choice
2:21AM 3 dialing from a website. How to start...?
1:51AM 0 notes: www.voicematch.cc & speex 1.1.7, unrelated
12:06AM 0 why I don't do this test ?
 
Thursday March 3 2005
TimeRepliesSubject
10:41PM 0 Asterisk SIP client problem
9:48PM 0 I have met a message : "No one is available to answer at this time".
6:23PM 10 Options in Brazil
6:03PM 5 Audio pausing over IAX trunk
5:31PM 8 [OT] - Why should I answer a Newbie question, therethick!
5:04PM 5 Problems dialing out - possible settings changes
4:33PM 0 FW: (still problems) Dialing phone number and extension together to avoid listening to voice menu (incoming call)
4:27PM 2 Beginning with Asterisk
4:20PM 0 Vovida Load Balancer.
4:01PM 0 Netphone KE1020A with asterisk
3:57PM 4 defold passwords in asterisk@home version 6
3:21PM 1 Is there a way to find free zap channels on remote servers ??
3:20PM 1 Development help?
3:17PM 3 FWD and SIPPHONE problems after upgrading to CVS HEAD
2:46PM 2 [Asterisk-Dev] CVS-HEAD change: queue/agent persistence
2:33PM 3 Asterisk not relaying back the SIP response messages
2:27PM 0 New user - problem getting dtmf tones through VOIP providers?
2:20PM 2 Help for studying Asterisk source code
2:11PM 8 Why ${EXTEN} variable changes after Goto ?
2:08PM 0 Calling Card Platform
1:46PM 0 fax and codecs
1:38PM 3 Update Asterisk
1:23PM 4 MGCP to Inter Tel system
12:57PM 4 DyDNS + externip
12:24PM 2 Asterisk + SIP + NAT - seriously, what's the secret?
12:10PM 0 Lines to PSTN available in FXO
12:00PM 1 Asterisk@Home .6 Problems with outbound calls using Broadvoice
11:33AM 4 ZAP Line answer questio
11:13AM 0 SIP secret: argument only for outgoing
11:04AM 0 Upgrading the 7960 Image
10:15AM 1 Blacklists.
10:13AM 2 Attended Transfer (ATXFER) with CVS asterisk r 1_
10:04AM 7 kernel error with Zaptel cards
9:47AM 0 Teles GW authentification
9:05AM 2 Voice recognition with Asterisk
8:59AM 0 Warning Message with voicemail CVS 3-3-05
8:48AM 0 Recomended server hardware
8:38AM 0 IAX users in Japan or Taiwan?
8:08AM 4 Detect sound and continue, like BackgroundDetect() for voice
7:37AM 0 Re: More NAT questions -- SOLVED
7:25AM 15 country/city codes
7:22AM 0 problem registering a bt100 with 1.0.5.11 firmware
7:04AM 1 IAXy and Private IP
6:58AM 0 Some errors on sip debug
6:40AM 3 Calling hangup in background
5:50AM 7 Getting phpconfig to work?
5:43AM 2 Installing modules for TDM400p
5:27AM 1 capi debugging
5:02AM 16 how do i get rid of this blasted echo !!!
4:46AM 8 Re : Calling card platform
4:22AM 7 Wrong CVS version ?
3:51AM 0 RE: Getting phpconfig to work?
3:25AM 0 Realtime IAX/SIP with 2 asterisk servers but 1 central iax/sipfriends Database
3:24AM 2 RE: Getting phpconfig to work?
3:17AM 0 RE: Getting phpconfig to work?
3:04AM 0 RE: Getting phpconfig to work?
3:03AM 1 RE: Getting phpconfig to work?
2:47AM 2 Multitenant feature
1:53AM 0 Forward Call from Asterisk to SER
1:12AM 0 Update on the blending of app_cbmysql and app_meetme2 (out of tree modules)
 
Wednesday March 2 2005
TimeRepliesSubject
11:00PM 0 best calling card platform for asterisk
9:02PM 1 Asterisk 1.0.6 music-on-hold
7:50PM 1 Building Asterisk with CentOS
7:00PM 0 super macro
5:14PM 1 Searchable Asterisk-users archive available
4:14PM 0 Windows Messenger and 481 error
3:29PM 1 Dial Application/redirection on demand
3:22PM 3 How use Spanish / English prompts on same box
3:08PM 0 IAX trap question
2:50PM 0 Way to disable "#" as transfer and just take thekey.
2:49PM 10 timing/clock problem
2:39PM 9 Asterisk URL and Callcenter Apps
2:19PM 3 Way to disable "#" as transfer and just take the key.
2:19PM 6 Has anyone seen this before
2:11PM 1 Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)
2:01PM 6 Music on hold on timing sources
1:44PM 0 Call Forwarding to Cell Phone, Pager, etc
1:24PM 5 [OT] stupid firmware question...
1:02PM 6 Polycom Soundpoint 500/600 MiniBrowser
12:32PM 11 Sending Voicemail's to two email addresses
12:05PM 0 IAX & LAGRQ & POKE explanation
11:50AM 0 Asterisk SKINNY with Cisco IP Conference 7935
11:04AM 0 IP300 soft key configuration
10:39AM 1 Asterisk HEAD and Mysql problems
10:34AM 0 OT: Looking for asterisk integrators in Dallas,TX
10:22AM 0 Fax with spandsp + zaphfc
10:04AM 2 Dial application invoked again and again
8:35AM 0 TE405P/zttool
8:22AM 2 Dual Asterisk Servers
8:16AM 3 More NAT questions
8:02AM 4 /dev/zap not created
7:58AM 3 Multiple lines
7:36AM 0 chan_capi - fax patch - crash
6:33AM 0 asterisk-oh323 bugtracker
5:50AM 5 Asterisk Manager API - multi "Originate" cal ls
5:46AM 4 cvs stable and 1.0.5
5:40AM 2 Dual X100P cards
5:28AM 1 Asterisk Manager API - multi "Originate" calls
4:54AM 0 Help needed with installing ZAPHFC
4:53AM 2 e164.org and FWD now have peering arrangement
4:14AM 0 [Asterisk-Dev] Digium's G.729A codec problem
2:40AM 12 wctdm and two tdm cards
2:38AM 21 Why should I answer a Newbie question, there thick!
2:02AM 1 Incorrect CDRs
1:51AM 1 IVR setup problems
1:07AM 0 Send parameters from asterisk to ADSI phone
12:45AM 1 Addons Make Problems! HELP!
 
Tuesday March 1 2005
TimeRepliesSubject
11:08PM 1 Call waiting in Australia
11:06PM 0 MWI work with ast_data?
11:03PM 0 IAX+G729a
9:00PM 0 How could Asterisk help me on a Internet webcast speech!?
8:37PM 0 Echoing Beep on ZAP channel (one sided)
8:04PM 1 "n" priority not in 1.0.6
7:18PM 1 iax notransfer=no and Tt in Dial()
5:33PM 0 AMP with Sipura 3000 PSTN line
2:51PM 1 OH323_OUTCODEC Unsupported
2:17PM 0 IAX or SIP answering services
1:53PM 7 "No compatible codecs!" -- worked with 1.0.0, not 1.0.6 or CVS.
1:29PM 0 Dialing phone number and extension together to avoid listening to voice menu (incoming call)
1:05PM 1 dropping extra frame..already have it????
12:40PM 9 Broadvoice + Videosupport=yes - Fails!
12:33PM 2 www.voicematch.cc
11:58AM 0 Looking for tech in San Francisco
11:42AM 3 agi RECORD FILE with offset
11:31AM 44 MozPhone
10:52AM 1 Re: FRS over *
10:32AM 2 [Asterisk-biz] IAX2 web client that works withg723 / g729. We got One
10:05AM 0 RE: Big increase in SPAM lately
10:03AM 0 [Asterisk-biz] IAX2 web client that works with g723 / g729. We got One
10:00AM 1 Big Increase in SPAM over the last few weeks
10:00AM 0 New Integrics Tip: Recording Voice Prompts
9:47AM 0 FW: SIP Phone Choices
9:35AM 0 SIP Client at outside and connect to an Asterisk Server sit behind NAT with SER
9:26AM 9 Connecting Asterisks via SIP
9:08AM 4 Ordering a Voice PRI for Asterisk
8:24AM 3 Important :: Please support the development of a new Jitterbuffer for SIP
8:11AM 9 Polycom Auto-Answer
7:43AM 2 multiple Fritz ISDN/BRI PCI
7:41AM 5 mini atx and asterisk (EPIA and the like)
7:41AM 0 Addons compile errors
7:16AM 3 Problems Starting Asterisk - FOP AM Portal
7:14AM 2 Newbie - What Do I Need?
7:00AM 2 Some asterisk ser problems
6:55AM 2 openh323
6:21AM 0 chan_sccp and 7912
6:15AM 12 What my IAXy could have been...
5:49AM 0 Is the Siemens SX353 (DECT) Base Station compatible with *?
5:34AM 4 Sipura 3000 Inbound Dialing Problem
5:03AM 0 Voicemail advanced options
5:02AM 4 Cisco 7940, Voicemail & DTMF
4:51AM 1 Music on hold..Mar error "res_musiconhold.c:309 monmp3thread: Request to schedule in the past" ?
4:33AM 2 Cisco 7960 x g729 x Unable to create/find channel
4:07AM 3 Park Craches asterisk
3:57AM 0 Incoming problem of Asterisk and Broadvoice
3:02AM 1 NoCDR Warning
3:00AM 0 UK CLID Asterisk CVS
2:53AM 1 in calling
2:25AM 1 DIAX 0.9.10f available for download
2:02AM 0 SV: chan_capi compile error on FC3
1:59AM 0 RE: Pb DTMF with Asterisk vs Cirpack Transit, , Node
12:47AM 0 Advanced Conferencing optionswithout-of-treemodules?