asterisk users - Oct 2003

Friday October 31 2003
11:17PM 2 Asterisk: Reloaded
9:53PM 1 Polycom Soundpoint IP600
8:17PM 7 Huge silence breaks between Cisco 7960 phone & X-Lite
3:43PM 2 msn messenger
3:43PM 7 Is down for 700 #'s?
3:34PM 1 pcphoneline
2:15PM 0 Flaky SIP registration
12:51PM 0 one way sound with x-lite (sip)
12:51PM 2 problem DG-104S not call
11:21AM 2 Echo on remote end when using NuFone
10:48AM 6 Which ADSI phones to buy?
10:41AM 0 Dial strings with ; characters
10:19AM 2 Some problems after an Asterisk update
9:44AM 0 FAQ on the Wiki
9:03AM 1 Problems with SIP
8:48AM 3 Cisco Support Contacts
7:36AM 0 Free LD/International calling testing, H323/SIP/MGCP and Asterisk
7:34AM 0 I can hear nothing if call from H323 to SIP.
7:25AM 2 asterisk and pingtel
6:53AM 8 asterisk FAQ
6:36AM 1 Problem in setup of asterisks
6:28AM 1 Making list of IAX providers
5:29AM 0 Voicemail storage question
5:13AM 3 MOH problem
4:58AM 0 One more QoS question for RH9
1:54AM 3 Password in VoiceMail
12:53AM 5 VoiceMail Configuration
12:17AM 0 DACS Functionality in Zaptel
Thursday October 30 2003
9:46PM 0 Three way calling problems: 2 ea. X100P 1 ea TDM10p
7:44PM 0 X100P Driver loaded, but no IO on /dev/zap/1
6:55PM 0 ADSI Pains
6:31PM 19 Absolute Minimum Installation Packages
5:27PM 0 extension exited non-zero...
4:59PM 1 Question about IAX/DID's...
4:30PM 6 Asterisk + Video
1:29PM 6 DTMF x-lite
12:58PM 12 H.323 and G729: Another sad tale
12:52PM 0 Newbie Question about MSI 240 Global Station
12:28PM 3 Compile problem with older ver. of CVS
12:09PM 41 SwissVoice MGCP IP10S
11:37AM 3 Fwd: Re: SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
10:00AM 9 Newbie hardware question
9:46AM 5 two things
9:42AM 4 Newbie with 12sp+
9:18AM 1 SIP NAT
9:00AM 5 critical problem
8:14AM 0 SIP/REGISTER problems!
7:38AM 23 Info on UK ISDN30e?
7:01AM 1 NAT type router database?
5:15AM 1 Out Of Band DTMF and SIP
5:07AM 1 G.729 pass thru Asterisk
5:06AM 0 Ringing ....
4:52AM 3 ZapRAS docs needed...
4:33AM 1 install problem
4:33AM 0 SIP error: Asked to transmit frame type 64
1:51AM 0 Communication between 2 UA
Wednesday October 29 2003
7:09PM 3 AGI question or something
6:53PM 5 VMAIL.cgi
5:52PM 9 Polycom SoundPoint IP 500
5:39PM 10 Nortel PowerTouch 350
4:28PM 3 Campon feature
4:17PM 10 Sip bandwidth usage
3:31PM 1 Asterisk as a load generator
3:26PM 5 XTEN-Lite Bad sound!
2:33PM 6 Gnophone and Asterisk
2:03PM 1 SIP behind NAT problem
1:51PM 3 Listen to a Call
1:39PM 4 Channelbanks for use in europe (Sweden)
12:38PM 0 Re: Large installation [was: SS7 signalling/Softswitch]
11:51AM 4 Host unspecified ??
11:50AM 0 Call pickup and SIP phones
11:10AM 6 Call transfering, conferencing
10:48AM 0 Forwarding all calls using SIP
10:47AM 0 iconnecthere Troubles
10:23AM 5 call waiting beep
10:07AM 2 Voicepulse and IAX
9:07AM 0 unsbscribe
8:26AM 0 extension dialing in the dial function for PRI !
8:07AM 3 16xE1 solution based on *
7:01AM 8 FW: Voice/Data mixed routing over Digium E1/T1 Card
4:49AM 1 Some Basic Reading
4:31AM 4 probs with loading tor2 and wcusb
2:52AM 8 ATA186 configuration for fax application
2:24AM 1 Distinguish between voice and data call
2:14AM 3 Am I missing somthing?
1:58AM 8 SIP client
Tuesday October 28 2003
10:13PM 1 Software Fax Modem. Problem to apply patch to Makefile in apps directory
9:18PM 0 Dialing long-distance locally
9:17PM 7 Software FAX
7:28PM 2 looking for a place to buy SNOM or Cisco Phones (Cheap)
7:02PM 0 X100P/ATA186 not playing nicely...
6:58PM 5 Grandstreams can't call out with latest CVS
6:39PM 1 MOH Mixing tool
6:25PM 0 Where to find info on #, *67 *82 etc?
6:03PM 0 VoiceMail and Password
5:29PM 7 rxfax problem
5:07PM 0 NAT and v6
4:42PM 2 Manager/Originate
4:35PM 7 RX gain TX gain
4:31PM 5 Already on the phone?
3:12PM 0 Asterisk <--> Cisco 2620
3:00PM 7 SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
2:57PM 1 # TDM 400P signal problem
2:34PM 1 TDM 400P signal problem
2:07PM 4 Making PrivacyManager smarter?
1:49PM 0 Scitel Brix QE
12:57PM 0 Consultants/Companies in Indianapolis?
12:40PM 10 Cisco or Snom ???
11:29AM 0 cdr - call transfers
9:38AM 2 SIP Calls Don't Properly Connect (Continue Ringing) After CVS Update
8:48AM 0 Unable to find a path from G729A to ALAW, Unable to find a path from GSM to G729A
7:53AM 1 Feature request {with begging} sip debug <ip_address>
7:50AM 3 Software FAX Modem--One Last Request For Help
7:46AM 0 Ringing applecation is not working
7:36AM 0 RE: [Asterisk-Dev] Upcoming Major CVS Changes
6:32AM 2 Upcoming Major CVS Changes
5:01AM 4 Today's CVS
3:43AM 2 Another Segmentation Fault (Recording sound)
3:12AM 0 Oh323 segmentation fault in asterisk...
1:56AM 0 voicetronix openline4 echo problem
Monday October 27 2003
9:01PM 5 OT Vonage soft phone
6:02PM 9 PRI & Asterisk Redundancy/Fail-Over
3:52PM 0 Asterisk behind nat with hole, hardcoding solution
3:42PM 4 Asterisk + Sip phones on Nat
3:24PM 4 Fwd: Re: Asterisk on FreeBSD
3:02PM 5 BOTH UAs behind same FW/NAT
2:57PM 35 Answering Machine Detection
2:34PM 5 Voicetronix OpenLine4
1:29PM 0 Asterisk on SPARC
1:21PM 0 Luxon Communications
1:11PM 0 Stuttered Dialtone for multiple extensions
12:52PM 5 Groups in *
12:50PM 4 passing digits for voicemail from sip gateway
12:49PM 5 SIP Softphone
12:25PM 0 Using Gastman
11:55AM 11 QoS What to do?
10:20AM 1 Is transcoding a bad thing?
10:13AM 0 Fwd: Re: Asterisk on FreeBSD
10:12AM 0 ZapBarge for SIP Channels
10:10AM 4 SIP -> H323 Seg fault.
8:47AM 0 G723 format compilation errors
8:07AM 3 Providing PRI to PBX
7:21AM 0 "Starting simple switch"
6:25AM 1 get IP Address from caller using oh323
6:13AM 0 core dump in app_dial
5:32AM 9 Anyone got VM2 working with MySQL?
4:39AM 1 CVS File README.mysql concern..
4:31AM 0 dtmf detection on modem-ISDN
2:00AM 5 SIP & IAX behind NAT
Sunday October 26 2003
11:22PM 0 SIPURA SPA-2000 now available
10:58PM 0 Sipura SPA-2000 anyone?
9:03PM 0 voicemail broken voice
8:53PM 1 After start Asterisk, error foung in the messages log file
6:18PM 0 Anyone have the Cisco 7920 working with Asterisk?
4:35PM 15 Extensions Problem
1:48PM 10 Lucent Partner extension to X100P
1:24PM 1 SIP auth
12:30PM 33 Asterisk on FreeBSD
12:20PM 0 PERL scripts: How to trigger?
11:16AM 2 help: mixing monitored in out files
11:12AM 0 Re: [Asterisk-Dev] important feature missing?!
8:53AM 0 alpha/numeric paging
8:05AM 15 NuFone International Calls
1:56AM 5 ReplayTV connecting through Asterisk box
Saturday October 25 2003
6:48PM 3 Confuson on iax calls (register or not?)
2:53PM 0 Asterisk External Resources Page
1:25PM 4 X100P stopped working
12:31PM 1 Voicemail.conf in MySQL is not functioning
12:17PM 3 Iconnecthere connect problem
5:51AM 16
2:20AM 3 Voicemail help
12:59AM 0 Music On Hold with Voicetronix
Friday October 24 2003
7:17PM 1 Looking for a 12 to 24 FXO Channel Bank in Colombia
6:04PM 0 Problem with CDR dst when executing Dial from 's' extension
4:43PM 4 Help with Dev Kit Lite
3:44PM 3 Broken Pipe
1:47PM 0 US source for compatible ISDN cards?
1:42PM 3 Compiling gastman under Win32
1:21PM 0 Looking for supply/installation contractor
11:56AM 0 OT - Getting Cisco FXS ports to dial thru *
11:23AM 3 How to use the Cut() command to chop off an ending character
11:09AM 29 SS7 signaling/Softswitch
10:33AM 0 Fwd: Re: A software FAX modem
10:30AM 1 where to buy a SIP phone or adapter in SFBay area?
10:18AM 4 Nextone softswitch testing and Asterisk long distance
9:30AM 1 CTI interface(s) for Asterisk? [REALLY LONG MESSAGE - SORRY]
9:12AM 9 Context restrictions
9:01AM 0 Asterisk crash on module loading
8:50AM 0 anyone with a used analog card for sale or trade?
8:46AM 3 problems setting up E100P E1 germany
7:11AM 18 Asterisk behind NAT to SIP provider
6:53AM 1 mysql-vm-routines.h setup for MySQL
6:46AM 1 Questions about Zapateller and Privacy Manager
6:35AM 0 no dial out
6:32AM 3 asterisk config files
4:54AM 18 AGI questions..
4:14AM 3 Asterisk ???
2:40AM 5 Anyone using ?
2:08AM 4 2 IAX2 calls, bad audio
Thursday October 23 2003
10:28PM 2 New here...
10:27PM 0 music on hold`
10:12PM 1 New To Asterisk
9:20PM 0 FW: Voicetronix
8:37PM 11 CVS update
7:54PM 0 *69 Last call return function
5:51PM 0 SIP native bridge
2:45PM 0 WAS: Call pickup (*8) on SIP devices. Bug #116
1:55PM 0 GotoIf Problems
1:16PM 22 Festival on RH9?
12:47PM 2 Go back - CVS
12:10PM 0 Asterisk passwords
11:58AM 0 DTMF relay with chan_skinny
11:54AM 1 Extended logic syntax
11:34AM 1 Number of TDMoE Channels?
10:53AM 0 ATM/AAL2
10:51AM 6 IAXtel problems
10:50AM 6 IAX peers and NAT
9:08AM 0 Monitor not in Manager.PM?
7:16AM 0 VoiceMail delete
6:52AM 2 asterisk and zplex10b
6:44AM 4 agi script forcing asterisk reload
6:27AM 0 G729 help
5:37AM 2 How to write sound file with G723.1 codec or G729 codec
5:21AM 5 Gastman crashes on Win32
5:05AM 3 wcfxs error
4:55AM 0 SIP Call Seq Error (SIP/2.0 481 Invalid CSeq Number)
4:28AM 3 native bridge
4:26AM 1 Problems with OH323/codecs
4:21AM 0 * + BRI + Debian = High Utilization
3:22AM 10 Problems with * and IAXTel/FWD
12:39AM 12 Call pickup (*8) on SIP devices.
Wednesday October 22 2003
11:44PM 1 Placing SIP calls to other SIP domains?
11:39PM 0 SV: Running Asterisk and NAT on the same box?
10:40PM 3 Tonight's CVS breaks Grandstream phone
9:29PM 5 what is the best codec for low bandwidth? for quality?
8:38PM 2 X100P Manually Answer
7:53PM 2 new codec for grandstreams
7:29PM 0 Last call return *69 function problem
5:25PM 0 Calls out of the PBX
5:08PM 0 MGCP error for Cisco 7750 FXO card
2:13PM 3 minimum hardware
1:31PM 93 Meetme
1:06PM 0 Asterisk Applications Consulting
12:14PM 2 Artificially Limiting IAX Calls
11:57AM 1 Inbound IAXTel failing?
11:55AM 2 Trouble with loading ztdummy on RH 7.3
11:41AM 2 modprobe ztdummy failed
10:16AM 4 MOH problems
9:25AM 0 Fwd: Download Asterisk
8:54AM 2 RE: Grandstream Improvements
8:22AM 7 Running Asterisk and NAT on the same box?
7:49AM 0 SIP Carrier
7:46AM 0 iax over wireless
7:29AM 1 Download Asterisk
5:24AM 8 auto voice msg from text?
4:31AM 3 Slackware 9.1 Install Help
3:22AM 1 SIP and permit specified ip addresses
3:16AM 56 Is the X100P a WinModem?
2:14AM 1 IAX with multiple NIC
1:13AM 0 Different MGCP issues
12:39AM 0 capi incoming call
12:31AM 2 Useful patch in the bugtracker: streaming MOH
12:16AM 2 Encrypting SIP Phones
Tuesday October 21 2003
6:04PM 4 Asterisk with Gentoo
5:12PM 3 Quick summary of Grandstream survey results
4:44PM 0 Cisco 7960 MGCP and Skinny phone
2:29PM 2 Asterisk to SipPhone
12:45PM 15 Free g.729.1 implementation
10:14AM 1 SNOM 200 beta build + MOH
9:53AM 3 "Send to VoiceMail" button
8:51AM 0 zhone z-plex 10
8:44AM 17 "Defragmenting" mailboxes
8:01AM 0 Iitter Buffer Settings
8:01AM 1 Grandsteam to support iLBC
7:36AM 0 CallerID Screening Prohibit
7:17AM 0 unclear about IAX
5:51AM 3 Hangup
3:12AM 0 (no subject)
1:24AM 1 VM2 and MySQL
1:02AM 0 active talker in a conference ?
12:41AM 0 Call pickup -> Change shortcode
12:03AM 0 Speed for meetme/asterisk?
Monday October 20 2003
11:00PM 6 Auto-dial from webpage
9:03PM 2 Message Indicator Light
8:55PM 1 Unsubscrip
7:38PM 89 Survey: Grandstream improvements.........
7:35PM 2 Setting a variable in extenstions.conf from the phone keypad.
4:52PM 2 Setvar SIP_CODEC
4:22PM 3 Authenticate Application Problems
4:21PM 13 Call Waiting on SIP phones
3:17PM 1 Conference with MOH or input from computer Mic.
3:09PM 1 Festival hangs up?
1:56PM 15 Music Onhold Configuration
1:15PM 8 Setting up an IAX2 trunk
12:35PM 3 Polycom IP-600 phone review
9:39AM 1 Need to partner with someone in Hampstead London on a deal
9:19AM 2 global vars
9:13AM 5 MOH different question
8:30AM 5 No detection of Line Busy
8:18AM 0 (3) Cisco NM-HDV-2T1-48 for Sale
8:09AM 6 Queues and Agents
7:49AM 12 SIP Nat Issue
7:36AM 56 A software FAX modem
6:42AM 10 how to escape #
6:18AM 2 Playing around MSNs
2:42AM 10 retrieve_? files??
1:11AM 0 SIP how to
12:59AM 1 mgcp transfer takeback with ata186 (logs with comments - long post)
12:21AM 3 Tested 7905G
12:00AM 2 Problems on making calls from one Gnophone to another through the local Asterisk Server
Sunday October 19 2003
10:39PM 0 Your comment on the following test bed setup ?
8:39PM 3 Modem
6:39PM 0 SIP soft phone
6:36PM 2 newb - want to create a Dialpad like system
5:39PM 2 Music on hold...
11:47AM 0 Patch testers needed
9:57AM 0 Flastman 0.0.1-pre-alpha
9:55AM 4 Cisco ATA Call Waiting
9:45AM 2 Asterisk/Freebsd network connections
9:23AM 3 The Start extension
7:01AM 0 X100P and Call Waiting Caller ID on the PSTN line
5:16AM 2 Project Completed [Files Attached]
3:47AM 0 Feedback request: AGI GET DATA change termination digits
Saturday October 18 2003
10:48PM 1 Some questions of heavy * deployment and stability.
3:09PM 2 Even Newer Patch to app_queue with skillbased strategy
2:06PM 0 DTA310 Config
1:32PM 2 Creating new voicemail accounts
1:14PM 0 Oh323 cisco callamanager
1:02PM 0 DID line with Adtran TA750 and T100p
11:33AM 1 latest cvs update
10:40AM 10 Outgoing call to IVR not being "answered"
8:30AM 0 We have added an Asterisk Forums to our existing web site.
8:24AM 2 my asterisk experience (long)
3:23AM 1 AGI script question
1:44AM 23 x-lite
Friday October 17 2003
7:42PM 5 chan_skinny & XML Files for 7920
4:54PM 0 zaptel: [rx|tx]gain on E1/PRI/isdn audio quality problems
3:36PM 1 Asterisk in a rural project
3:19PM 0 X100P Echo - not resolved
3:10PM 1 (no subject)
2:42PM 2 AGI problem (crash) in RH9
2:37PM 2 Polycom IP 600 phone
2:00PM 0 IAX with dynamic echo cancellation - what do you think?
1:08PM 0 Dial Multiple extension but require input not just off hook to bridge calls
12:22PM 1 Festival, Patch, Asterisk, etc.
12:11PM 2 System layout
11:59AM 5 Using channel banks
11:10AM 1 QoS On *
10:51AM 3 Beta testers for visual configuration tool f or asterisk
10:23AM 8 Beta testers for visual configuration tool for asterisk
8:56AM 7 Extension syntax specification - please help!
7:35AM 6 QoS
6:50AM 6 Switch statement taking over my local dialplan
5:11AM 1 Project Completed
4:33AM 0 Can't get IAX registration to work !#!$
3:24AM 0 console sound problem with usb sound card
Thursday October 16 2003
9:45PM 1 Prob with Ringing multiple Channels
7:24PM 0 Some issues with call transfer. Please help.
5:13PM 0 AddQueueMember penalty patch
3:28PM 2 Grandstream phone :(
3:02PM 2 CallerID not passed to Sprint & Verizon Cellphones via XO PRI
2:44PM 10 MOH and VAD
2:09PM 9 Starting * with G729 licences
12:08PM 0 Directory App - excluding users...
12:08PM 0 Flash vs. 2-B channel transfer
11:42AM 9 OT - SIP Auto-Answer for Cisco 7940/7960!!
11:31AM 2 consultative transfer cisco
10:03AM 4 Adtran TA750 & T100P
9:56AM 0 Re-2: Some questions for chan_capi
9:37AM 2 Supervised transfers
8:47AM 2 Problems with TE410P and E1 line --> Unable to open D-channel 24 (No such device or address)
8:39AM 1 Calling a registered computer
8:15AM 0 Trouble with linking 2 Asterisks with IAX
8:05AM 0 sip registration failed
7:43AM 2 Weird IAX2 problem
7:41AM 1 VoIP Monitor
7:41AM 21 Voicemail File permissions
7:36AM 0 Zultys Zip 2 Registration / Disabling SIP Authorization
7:23AM 2 Cisco 7905G phones
7:18AM 1 [OT] E1 Cable pinout
6:21AM 14 I give up!!
5:35AM 3 ISDN BRI card
4:40AM 0 Caller-ID Spill
2:48AM 3 AGI problem (crash)
1:51AM 0 Use of the "hint" modifiers - examples, anyone?
1:07AM 0 Digium TDM card bad DTMF again
12:53AM 0 IAX Rejected Connect Attempt
12:31AM 0 french newbie with asterisk
Wednesday October 15 2003
11:50PM 0 AVM ISDN card now working but unable to place calls
2:30PM 2 Odd ringing conditions
1:22PM 1 Problem with SIP and DOS attacks...
12:39PM 4 indications.conf
12:22PM 0 RE: Manager Interface Needs a protocol amendment
12:16PM 13 newbie question: Meetme
11:34AM 2 IAX Clients not connecting
11:01AM 4 SIP Telephone Quality/Price
10:50AM 5 e100p in Australia
10:23AM 20 SER vs STUND with Asterisk..
9:51AM 0 Basic questions
8:56AM 0 app_dial Flag
8:49AM 4 chan_skinny core dump
8:49AM 0 Manager Interface Needs a protocol
8:24AM 0 Real sip fax server
7:50AM 5 Sip call hang up
7:46AM 2 x110p -> tdm400p -> answering machine ISSUE
6:39AM 0 Some questions for chan_capi
6:30AM 1 No 'ringing' sound to outside callers
6:17AM 2 My Grandstream works, but my X-Lite doesn't:no sound after 5sec
6:11AM 0 SIP phone comparisons chart and article
6:05AM 8 skinny problem
4:24AM 0 cdr on call transfer
4:15AM 0 CF loop
4:13AM 0 Problems with MeetMe SIP / Mobile
3:26AM 12 Announced Call Transfer
1:17AM 3 Problem with T100P card in a new chassis
12:52AM 0 Asterisk running on OpenBSD 3.3
12:47AM 0 frensh newbie with asterisk
Tuesday October 14 2003
11:51PM 15 Asterisk capacity
8:37PM 16 My Grandstream works, but my X-Lite doesn't: no sound after 5sec
8:34PM 5 VAD in Asterisk ?
8:15PM 1 DISA and ringing tone
8:06PM 31 Digium should develop and sell just Dummy card. For timing...
6:57PM 1 outbound caller ID problem on PRI
6:31PM 5 Wildcard TDM400P - FXO?
6:10PM 0 No Ringback on Iconnect or Nikotel
5:58PM 1 Newbie with questions
4:22PM 1 re: Restoring Cisco 7960 to defaults
3:55PM 0 Dial-out using Asterisk
3:20PM 0 pattern matching problem when dialing
3:14PM 2 Cisco hard IP phones and Skinny vs. SIP
2:43PM 1 Asterisk Certified Hardware label?
2:28PM 2 SIP Phone Tone
2:16PM 1 200-400ms latency
2:12PM 2 T100P to Adtran TA750 - No dialtone or ring
2:06PM 0 Turning a regular call into a conference?
2:00PM 2 IAXTEL - Problem Configuration.
1:45PM 12 Digium cards just for timing
1:31PM 3 Iaxtel and Voicepulse
1:15PM 8 WCFXO echo rexolved for me
1:00PM 0 list server Delays
12:39PM 0 AGI function calls
12:05PM 1 On an RH9 box, where does wcusb get loaded?
11:37AM 17 use of SIP SHOW CHANNELS question
11:28AM 5 Mitel 5055 phone
11:24AM 2 managers.conf Clarification Question
10:18AM 1 preplanning for a new home installation
9:07AM 0 Documentation Forbidden!!
8:02AM 16 H.323 - SIP gateway
7:14AM 0 QOS Question
7:04AM 2 RAS protocol
4:43AM 5 VoiceMail2 warning
4:02AM 4 Success story
3:43AM 11 */SER/FW
3:39AM 0 Has something changed with AGI recently?
3:14AM 6 Outgoing CallerID
2:58AM 0 Meetme error in Mobile/SIP phones.
1:43AM 7 dialling out
12:03AM 1 no ring in ear
Monday October 13 2003
11:59PM 13 Asterisk Manager
9:49PM 2 Problem with SIP authentication
8:40PM 0 MGCP Gateway (Dlink DG104s)
8:16PM 0 H323 ID's
8:11PM 1 AGI solution to Grandstream BT102 call waiting problem
4:57PM 0 Call Parking and Paid Digium software modifi cations
4:07PM 2 out going calls
2:44PM 3 Call Parking and Paid Digium software modifications
1:11PM 1 ACD/IVR dialogs/SIP/client environment
12:48PM 8 PrePaid Application!!!!!
12:04PM 4 IAXTEL/ Dial problem
12:00PM 4 PRI/E1: machine freeze/dies after a few calls
10:48AM 2 chan_h323 - Segmentation fault (core dumped)
8:12AM 8 "Gates steps up telecom campaign"
7:56AM 6 Strange message !! Unknown IE 76 (Unknown Information Element)
7:35AM 0 Help me please!
7:22AM 2 test calls between iaxtel & fwd
6:42AM 0 Ports open
6:08AM 0 asterisk on the firewall/router
5:56AM 0 Gatekeeper with Asterisk
4:34AM 11 bare-bone config
4:31AM 3 Problems with MeetMe.
4:30AM 2 replacing sound files
3:27AM 3 Extension Dialing problem with SIP
3:08AM 2 [Asterisk-Dev] Cisco ATA 186 chan_mgcp Transfer problem
2:48AM 6 Error
2:47AM 1 newbie: need help configuring asterisk and snom
2:22AM 1 oh323 inband dtmf - Possible bug?
2:11AM 2 Generating a call with the Manager interface..
2:03AM 2 e100p in norway?
1:54AM 0 IAX authentication code
Sunday October 12 2003
10:57PM 13 SIP phone
10:40PM 9 X100P Echo Problems..What's going to happen?
8:16PM 1 AGI Test Fails
7:42PM 32 No sound with SIP Phones on the Internet
7:03PM 0 Help: Segmentation fault. Something about smoother
2:16PM 7 Feedback request: AGI GET DATA change - termination digits
2:14PM 1 Queues and max time in queue timeout?
11:46AM 2 Beginner
3:59AM 4 New Processor support..
2:16AM 3 Is this Hardaware Enough for Asterisk ?
12:38AM 3 INFO method and DTMF translation
Saturday October 11 2003
10:14PM 2 Fwd: RE: SIP / IAX over satellite
10:10PM 2 SIP Configuration
10:08PM 2 "context confusion" internal context 2 context only?
7:41PM 2 Distortion of voice after cvs upgrade
2:41PM 8 SIP / IAX over satellite
10:15AM 4 Problems with AGI scripts in Perl and Java
6:24AM 0 We need an consultant
Friday October 10 2003
9:01PM 4 Marketing Digium/Asterisk
8:06PM 11 T100P & Phones Configuration
6:05PM 4 Grandstream Setup
4:22PM 4 IAX Not working between machines
4:21PM 1 X100P check for Dialtone
4:10PM 0 Asterisk ** Adtran 750, 5xQuad FXS, 1xQuad FXO, Pentium 500MHz, 10GB hard Drive, 384MB RAM..Will that do?
2:19PM 1 multiple SIP users on one phone?
1:50PM 1 No ISA tormenta card message]
11:48AM 0 Echo observations and some questions
11:39AM 1 NEWBIE looking for advice.
11:18AM 4 Grandstream wallmount??
10:45AM 2 SIP - H323 GAteway
10:19AM 5 BudgeTone-102 MWI&CID with Asterisk
9:53AM 6 Actual audio bitrates
9:20AM 2 ALERT_INFO=1/ Cisco 79x0
6:24AM 5 Caller Id AGI Script
5:53AM 0 Problems with Devkit Lite setup
5:51AM 0 Error when making a call
5:10AM 0 modem connection over handy?
4:39AM 8 one-way audio
4:23AM 0 [Asterisk-User] Howto get the Caller Phonenumber ?
4:09AM 1 Asterisk crash on AGI
2:30AM 0 Supported dialogic hardware ?
12:18AM 1 Howto configure asterisk with AVM ISDN card
Thursday October 9 2003
10:27PM 1 Problem with DTMF 'looping' / mis-dials (X100P card)
8:31PM 0 University phone system
6:16PM 0 What is the "pingtime" option in iax chan(iax.conf)?
6:03PM 21 X100P Config
5:03PM 6 E100P setup in Switzerland
3:53PM 0 Asterisk and DMS100 Channelized T-1
1:45PM 2 * consultant needed - will pay
1:32PM 0 Results SUSE 8.2 + server size
1:29PM 4 No Ringing from PSTN
1:25PM 5 Record App Paths
12:55PM 0 Comdial Unisyn Execumail
12:53PM 3 How to disable native bridge of SIP-to-SIP calls?
12:16PM 10 Cisco 7914
10:55AM 1 Redhat system init and wcusb
10:39AM 19 IAX2 Trunking confirmation?
9:15AM 1 7940/60 TFTP Problem
8:03AM 10 concurrent calls
6:13AM 1 real billing time for a call
5:18AM 0 newbe Echo problem
3:21AM 0 IAX
12:58AM 0 Cisco 7940/7960 phone and conference calling ?
12:51AM 7 Sasquatch, the Loch Ness Monster, UFOs and...
12:49AM 5 5 second latency sip to oh323
Wednesday October 8 2003
11:07PM 0 7940
10:48PM 0 Radius + Asterisk
8:25PM 0 SER versus Asterisk for WAN SIP Phones
7:34PM 0 Exten delay matching
7:10PM 5 pbx_spool and contexts
3:44PM 4 Registering Softphones to Asterisk
3:03PM 0 Microsoft RTC Client
1:26PM 5 Mini-PC box to run server
12:42PM 3 SIP softphone volume control?
12:27PM 3 Call Error
10:09AM 7 asterisk & festival problem.
9:15AM 8 iax2 trunk
8:52AM 4 Hypothetical : Working across multiple servers??
7:55AM 0 SIP Problems with Cisco 5300 - Invalid CSeq Number
7:40AM 2 Loop counter variable in dialplan?
7:19AM 1 Asterisk role
7:08AM 2 Call to "06302" aborted, insufficient bandwidth
7:01AM 15 chan_capi and latest Debian package
6:48AM 1 Cisco 7940/7960 phone and conference calling?
5:30AM 2 Ztdummy Bug
5:01AM 1 Asterisk CDR reliablility..
4:16AM 7 Music On Hold distorted
12:28AM 2 BudgeTone 102 flakey sound
Tuesday October 7 2003
9:20PM 15 Fax Detection
3:57PM 9 Newbie
3:49PM 0 AstMan Issues
3:15PM 0 Is there always data at /dev/zap/1?
3:11PM 1 [PATCH] allow announcements in app_dial
3:05PM 3 Dynamic registration to flakey for production system
2:23PM 3 Second Send: Using PCI backplane
2:06PM 0 Problem with SIP Client!
1:30PM 5 auto 'modprobe wct1xxp' on startup?
12:51PM 6 Line going to Zombie
12:12PM 0 Large-scale Asterisk deployments: VON panel
12:10PM 3 Compile problem SuSE 8.2
12:04PM 1 Call Park on SIP phones
11:58AM 1 FXO on AT&T broadband POTS line?
11:28AM 2 agi exit problem
10:48AM 2 call parking on specific park number
10:30AM 1 clocking source for T100P?
9:14AM 27 IAX and Jitter problem
8:39AM 1 Dialling problems
8:25AM 0 Communication between 2 telephones
8:16AM 0 RE: Asterisk-Users] IVR Questions?
7:36AM 0 Connect with another PBX
6:37AM 2 Can AGI be used in this way?
6:07AM 1 Digium FXO
4:01AM 0 Vioce Modems
3:37AM 5 Voicetronics
Monday October 6 2003
9:59PM 13 Help with questions for initial Asterisk wizard (GUI)
7:26PM 8 Message Waiting on Cisco 7960
5:39PM 7 ISDN Dialout
4:13PM 1 SIP X100P Echo Problems
4:12PM 1 MP3s in /var/lib/asterisk/mohmp3 causes Asterisk crash
3:43PM 0 Digium TDM400P and Analog DID Trunks
3:13PM 3 Web Voicemail Permissions
2:51PM 1 Snom100 H.323 sample config
2:41PM 0 getting inbound caller-id from sip remote-party-id field
1:45PM 28 direct-inward-dialing (DID)
1:45PM 2 callerid name modification (or adding)
1:13PM 1 X100P too quiet
11:17AM 1 chan_zap.c - echo cancelation getting in the way of dialing????
11:16AM 1 Start...
10:13AM 5 Remote control IVR
8:38AM 0 newbie question: 1 or 2 servers
8:36AM 0 Priority Voicemail
8:12AM 6 Modem and Fax over VoIP
7:51AM 7 Asterisk, X-Lite and iLBC..still..
7:40AM 1 Data base structure
6:48AM 0 problems with the extensions of sip in ATA 186
6:34AM 8 Alternatives to FXS cards?
2:38AM 1 Noise with Grandstream/PSTN
12:21AM 12 Anyone else use Audacity for prompts?
Sunday October 5 2003
11:02PM 5 Good W2K softphone
5:53PM 9 IVR Questions?
5:21PM 0 Zap Analog Line Hangup Problem
4:19PM 0 HI.. .Any comments about VoicePulse
2:41PM 4 ChanIsAvail app setting ${AVAILCHAN} to an unusable value.
7:49AM 9 DB virtualization for multiple database support - Was Re: How to use vmdb.sql in voicemail.conf/extension.conf
6:43AM 4 Grandstream 102
Saturday October 4 2003
6:02AM 18 Let's TALK ABOUT IT!!!
5:40AM 0 another newbie question: forwarding delay?
Friday October 3 2003
11:39PM 8 TE410P: Double/missed interrupt detected
11:23PM 1 starting asterisk?
9:44PM 2 Editting variable contents
9:30PM 0 codec bridging
7:44PM 8 Answer on second ring - need it on first.
4:43PM 3 monitoring the asterisk and safe restart
4:09PM 5 802.11 phone review: WiSIP
3:50PM 5 Job Opening at Digium
11:28AM 5 Problems with Caller ID on FXO
11:17AM 2 Help Loading a TDM card!!
11:01AM 13 Message Waiting on Cisco 7940 does not work
10:38AM 3 Cisco CallManager Image for 7940/7960
10:17AM 5 Iconnect Incomming calls
9:58AM 2 Ascom Ascotel 2050 & Fritz PCI Card (Capi)
9:47AM 0 Transfer fails periodically
9:00AM 5 (still) Channel problem - solved partially
8:27AM 10 No Ringback on Iconnect
6:52AM 4 suggested hardware especially sound cards
6:46AM 2 H.323-SIP Gateway
6:42AM 3 Transfer from IAX call
6:16AM 5 monitor
5:54AM 2 where to specify mysql DB USER PASSWD for voicemail2
5:13AM 4 Sound file..
3:48AM 3 Budgettone + G729
12:00AM 1 primuxisdn capi
Thursday October 2 2003
9:45PM 0 SIP Date: header
9:43PM 0 Help-to start Asterik PBX
8:33PM 1 THE "NAT-MARE" IS OVER test volunteers needed
7:41PM 5 "New" TDM cards--driver won't load
7:36PM 7 Voice detection
5:05PM 0 chan_h323 Ringing Congestion causes * segfault
3:56PM 3 iaxtel fixes
2:50PM 4 SIP and DSL Bandwidth queries.
1:11PM 0 Gastman working in W2Kp.
12:54PM 1 problem w/ musiconhold & mpg123
12:53PM 7 Any way to get out of a remote console without stopping *
10:01AM 1 Front end
9:43AM 3 Problem with Dutch PSTN-line on X100P
8:25AM 3 Version 1 vs Version 2
8:23AM 0 WINXP Messenger SIP Client (Good News, Bad News) WINXP authorization with secret
7:28AM 9 WINXP Messenger SIP Client (Good News, Bad News)
6:59AM 0 Fw: Call it Asterisk-Addons and let us go have some fun?
6:57AM 2 Call it Asterisk-Addons and let us go have some fun?
6:23AM 3 Asterisk friendly IAX/SIP wholesalers in Australia
6:18AM 6 Has anyone got * working with Xten soft phones
5:54AM 0 Help with ISA PhoneJack.
5:52AM 1 What is mySQL used for in Asterisk?
4:18AM 7 Xten Lite Build 1079
1:07AM 7 Zapateller
Wednesday October 1 2003
7:14PM 23 eBay Sip Phone Scam.
7:08PM 2 Voice Mail App
2:53PM 4 SIP problems fixed?
2:40PM 2 DTMF weirdness
1:32PM 0 oss Errors
1:21PM 2 newbie question: MOH problem
12:23PM 1 Audiocodes gateway and asterisk
12:17PM 0 Question: handling fully-qualified SIP dial requests
12:03PM 2 grandstream phones and Transfer
11:30AM 3 SIP Provider Question
11:04AM 0 Codec problems??? (Was: SIP i.e. Is something broken?)
10:23AM 1 x100p card - detect dialtone?
8:36AM 1 R2 signalling
8:12AM 2 MGCP Phone and Asterisk PBX
7:32AM 6 recording voice calls
7:14AM 0 Dialogic D/240SC-T1 REV2 Boards for Sale
6:59AM 7 single dialplan for multiple Asterisk machines
6:55AM 2 Directory for Cisco 7960
6:40AM 3 VOIP long distance providers
6:37AM 0 Var vs Global Var vs DB
5:32AM 13 (still) channel problems
5:27AM 7 IAX and IAXTEL
5:19AM 0 Feature ver 1/2 Questions