asterisk users - Nov 2006

Thursday November 30 2006
11:29PM 1 CAPI module issue
7:10PM 3 1.4beta3 help
6:56PM 1 upgrading grandstream GXP-2000 from to
5:30PM 2 Force re-read of sip.conf
2:38PM 2 PAP2 and Asterisk
2:30PM 0 incominglimit and outgoinglimit
2:25PM 2 voicemailmain
2:22PM 3 Pickup *8 with CallerID
1:56PM 1 2nd attempt - Return code - How to?
12:56PM 0 Voicemail callback bug?
11:04AM 1 Asterisk 1.4 : App_Swift (Cepstral) Howto
10:58AM 1 Live call monitoring
10:57AM 0 Problem with ZapRAS and asterisk
10:41AM 1 T1's in St. Lucia
10:21AM 6 zaptel compilation problems with linux 2.6.19
9:29AM 2 Billing Software
9:14AM 0 meetme monitoring
8:52AM 0 zombie SIP channels after CURL cnam lookup
8:29AM 1 IP call to extensions off my server
6:52AM 0 SIP transfer from agent fails
5:33AM 1 Server Compatibility questions... IBM and Dell
3:24AM 0 codec error message
2:50AM 1 AGI PHP Issues (AGI script runs but phone hangs up too quickly)
2:15AM 6 200+ analog phones connected to FXS modules
2:13AM 4 Trouble with regexten
2:11AM 0 Digium TE405P dtmf issue
2:01AM 0 Distinctive ring
1:41AM 1 Cut function on semicolon separator
Wednesday November 29 2006
10:31PM 1 MeetMe announcements and SIP channels
10:24PM 0 Return code - How to?
10:24PM 1 extension launch into AGI
9:09PM 0 register history
8:18PM 2 Trouble using 2 IAX2 DiDs provided by different ITSPs
8:15PM 0 Re: asterisk-users Digest, Vol 28, Issue 152
8:08PM 0 Conferencing Issue please help
4:55PM 2 Setting RTP ports for Asterisk?
4:07PM 0 Call dropping
3:43PM 1 Call recording with Asterisk BE
3:23PM 1 Cisco 7940 Firmware 8.2
3:19PM 0 g726 voice prompts
3:10PM 0 beeping noise in background
2:42PM 1 voicemail.conf locking problem
2:36PM 3 Polycom 601 Second Incoming Call
2:25PM 0 Playing streaming MOH in Asterisk
12:47PM 1 Asterisk connection to a PBX
12:07PM 0 I am unable to find any included rpms with hudlite...
11:28AM 0 Call Recording and Call Transfers
10:55AM 12 What's up with the Manager Interface?!?!
9:21AM 2 Loosing IAX connection between offices
9:02AM 1 Getting app_cepstral to work with Asterisk 1.4.0-beta3
8:48AM 0 b410p hangup detection - Portugal
8:38AM 2 Asterisk + Avaya S8700
8:34AM 3 Blind transfer # not working for forwarded or picked calls
7:38AM 1 AGI PHP Issues (Not new to Asterisk but new to AGI)
7:10AM 1 Answer Supervision problem
6:55AM 0 keep line on hook
6:43AM 0 Desktop application for zap/agent call control
5:13AM 0 Re: SIP Port 5060 (Tom Lynn)
4:56AM 3 Siemens Gigaset C450 IP vs S450 IP
4:02AM 1 Monitoring an asterisk server during off hours
3:45AM 1 chan_misdn on a junghanns card
2:48AM 1 sendmail or postfix?
2:36AM 0 Something similar or better than HUD Pro?
2:33AM 0 Play an announcement while receiving DTMF?
1:35AM 1 Which SIP transport from France and termination services in the Nederlands
12:54AM 1 iptables example
12:08AM 1 Custom Voicemail Notification Email
Tuesday November 28 2006
9:16PM 1 Best text to speech program
8:19PM 4 SIP Port 5060
7:34PM 2 accountcode= placement in zapata.conf
7:29PM 0 Re: newbie question-asterisk username/password
6:11PM 1 Billing software with reseller accounts
5:19PM 2 No sound: X-Lite -> Asterisk -> VoIP Provider -> Cellphone
4:31PM 1 Bad Voice Quality - IAX2 redirect
4:19PM 0 channel ending with /n
12:17PM 1 Call recording filename
12:03PM 1 Why is * continually "destroying call"
11:38AM 0 Best email client for Asterfax
11:22AM 0 WANTED : Zaptel Patch - Dtmfthreshold
11:13AM 1 hang up detection
10:54AM 1 Return codes
9:30AM 1 SIP ATA Device Problems
9:07AM 0 Asterisk "Generating" SIP 486
8:23AM 1 cmd Record doesn't resume Dialplan if phone Hangs-Up.
7:45AM 0 aastra 480i xml interface for "comedian" mail
6:31AM 4 Zaptel drivers for Solaris?
5:39AM 2 Symbian Softphone
5:22AM 1 Different click2call?
5:18AM 1 Use of VPNs
4:19AM 1 Attn: DISA Experts(Strange problem with DISA)
1:22AM 1 Modprobe zaptel reports FATAL: Module zaptel not found
12:26AM 1 vm_change_password shell?
Monday November 27 2006
8:54PM 3 Do extra CPU's help?
8:20PM 0 SayDecimal Number
8:05PM 2 AsteriskNow console access
2:05PM 0 CTI
1:44PM 5 Manage Users in LDAP
1:34PM 1 Asterisk server reports
1:30PM 1 Memory leak
1:18PM 1 Caller ID issues
12:55PM 1 Sangoma & Dell 750
12:18PM 3 Voicemail, SQL & ODBC
11:58AM 2 Busy signal from IAXy when not connecting to my Asterisk box
9:53AM 0 Queues and Flash/SendDTMF in hybrid PBX
9:46AM 2 registration ip address
9:44AM 0 Feature and multiple application
9:23AM 0 [VoIP Trunk] No such host
9:15AM 0 BRIcard not sending DTMF
8:23AM 0 Script on hold
8:19AM 5 Trunk Alcatel - Ring problem and call disconnection
8:10AM 1 Incoming calls don't arrive for correct number
7:44AM 0 Announce Queue Position variable
7:43AM 1 Asterisk Feature Codes won't work
7:42AM 1 wip5000 crash AP
7:22AM 2 SIP group management
7:08AM 0 calls hang up even after Background() messageeventhough response timeout is set to 10 sec
6:33AM 0 flash transfer problem in asterisk with old PBX
6:19AM 1 AgentCallbackLogin deprecated?
5:05AM 0 Asterisk is taking the first digit of my entered number twice. Why?
2:22AM 3 bristuff error: "received SETUP message for call that is not a new call"
1:42AM 1 Click to dial apps always show from "asterisk"
12:49AM 1 Junghanns Bristuff PRI indication
12:11AM 1 calls hang up even after Background() message eventhough response timeout is set to 10 sec
Sunday November 26 2006
11:58PM 2 3xx redirect from asterisk?
11:40PM 0 Dialout to Meetme Fails?
6:37PM 1 Odd blip when playinv IVR over IAX
6:16PM 0 Asterisk 1.4 : AstDB Callerid Rolodex Tool
6:15PM 0 Asterisk 1.4 : Gentoo, MYSQL CDR, and GUI howto
6:15PM 0 MWI - Message Waiting Indicator free software routines
3:13PM 0 spandsp
11:39AM 0 port rtp problem
11:35AM 3 Looking for toll-free US did
9:11AM 0 way to get extension's hint?
8:57AM 1 Setting up FastAGI in Asterisk?
2:30AM 1 Streaming MoH working example required
Saturday November 25 2006
9:11PM 0 SOLVED - 1.4 svn voicemail bug / crash
6:46PM 2 Passing PRI traffic to remote * over IAX
6:31PM 0 Digium Iaxy S100 Factory Default?
6:23PM 1 Asterisknow
4:53PM 0 Re:VOIP Consultants wanted to build a Scalable ITSP Architecture
2:30PM 0 Sip reinvite
2:03PM 0 MeetMe, background agi and playing sounds
1:56PM 0 Linking Asterisk Servers using SIP instead of IAX
12:00PM 0 Re: asterisk-users Digest, Vol 28, Issue 132
8:57AM 2 1.4 svn voicemail bug / crash
8:38AM 1 dialing with different speed
7:35AM 5 DID Provider
7:06AM 0 Problems with sound quality
5:51AM 0 Re: asterisk-users Digest, Vol 28, Issue 131
5:49AM 0 How to do Call barging with SIP channel
5:47AM 0 VOIP Consultants wanted to build a Scalable ITSP Architecture Using OpenSource Softwares
5:09AM 0 Modem and TDM400P
Friday November 24 2006
10:36PM 0 Help needed - Can anyone please explain to me what is causing this - TDM2400P
7:59PM 2 Correct syntax to access a shell variable?
6:00PM 1 Re:Call Transfers in SER + Asterisk
5:38PM 2 Card don't hangup but Asterisk hangup
11:45AM 1 FS: Sangoma 10 port FXO card
11:29AM 2 cisco 7961 , asterisk and busy lamp
11:22AM 1 mfcr/R2
9:52AM 1 Monitoring awareness
8:41AM 0 Doubling up; redunancy with DUNDi
8:29AM 3 Junk faxes
7:21AM 0 Snom 360 / firmware 6.5.1 / registration problems with Asterisk
7:14AM 1 Encrypted password for voicemail
5:31AM 0 Caller Id not propagated to the analog line
4:39AM 2 DB9 e1 to RJ45 pinout
3:34AM 1 upgraded polycom to and...
2:00AM 1 Server Configuration for E1's
1:31AM 1 Installing the b410p card, unable to install mISDN
12:18AM 0 Dial() cmd seams unable to detect caller hangup
Thursday November 23 2006
10:54PM 1 (no subject)
10:44PM 0 asterisk and MISDN on a core2 Duo x64 system
10:32PM 2 Asterisk and TDM400P ?
9:47PM 0 Direct UA to UA RTP connection
8:12PM 0 MWI from ITSP
8:03PM 0 asterisk 1.4 variable list
7:12PM 0 asterisk-users, Matt has invited you to open a Google mail account
6:35PM 0 Asterisk voicemail and hotel software integration
5:54PM 0 Passing arguments to AGI script
4:27PM 0 Re: asterisk-users Digest, Vol 28, Issue 122
3:30PM 0 Store voicemal data in mysql DB
3:20PM 1 Call Transfers in SER + Asterisk architecture
2:02PM 0 Asterisk 1.4 Error
11:58AM 2 FREE DOWNLOAD - PRI / T1 Circuit monitoring
11:55AM 1 When does voicemail authentication take place?
11:54AM 2 Cisco 7970 SIP upgrade issues
10:02AM 0 festival problem using IAX (chan_iax2.c:2995 iax2_read)
9:44AM 2 Digium through Octasic
8:50AM 1 FOP is not displaying all my SIP extensions neither all E1 channels , why?
8:38AM 1 Error uninstalling freepbx-panel
8:18AM 3 Cisco 7970
7:47AM 1 Asterisk with SER
6:03AM 1 (OT) HylaFAX, IAXModem, Asterisk
4:53AM 0 AGI info
4:36AM 1 Re: How to change IAX default port 4569 to some other port :Debug Message Attached
4:18AM 2 How to change IAX default port 4569 to some other port
1:27AM 0 snom subscriptions issue on WRT (2)
1:18AM 1 How to kill a meet me room at midnight
1:04AM 1 asterisk 1.4 chan_h323, help please...
1:00AM 1 Calls "from asterisk"
12:18AM 0 OriginateEvent reason codes.
12:14AM 0 Exact definition of ASR
Wednesday November 22 2006
10:07PM 0 in Asterisk Manger its Unauthentication User and Host ..........
9:17PM 1 gotoiftime and blocking calls
8:26PM 1 Hold calling channel and ask called channel before connect???
8:19PM 2 G722?
7:09PM 1 Sipura phone does not ring
6:54PM 1 queuemetrics
4:14PM 2 How to park calls on a specific extension
4:03PM 1 Zaptel - make b410p fails on Ubuntu 6.10
3:21PM 2 Terrible, horrible firewall issues in * to * setup
2:04PM 1 aastra 480i configuration help
1:18PM 0 Call park on Linksys 922 and similar phones?
1:02PM 4 More than one asterisk process
11:51AM 5 TE110P and TDM400P
11:45AM 4 Asterisk On FreeBSD
10:36AM 0 channel_find_locked: Avoided deadlock ... messages - What to do?
9:57AM 1 G729 issues on 1.4 beta 3
8:48AM 2 How ecord all calls?
7:43AM 1 Recordings for VR analysis
7:33AM 2 Send event from dialplan
7:23AM 0 iax2 - wildiax phone & myself puzzled
7:15AM 1 Asterisk incoming call behaviour
6:54AM 1 DTMF detection during Call
6:08AM 8 Recordings.
6:02AM 11 Rewriting caller ID from database?
5:57AM 0 help in Call parking......
3:32AM 0 Ast 1.4 and B410p
3:21AM 1 Request for working config for DISA
3:18AM 1 about voicemail setting
3:15AM 1 Zaptel error
3:15AM 0 SOLVED: Digium TE405 card and Matra PBX
2:45AM 1 qualify=yes
2:29AM 0 Is it easy to route SIP/SDP and SIP/RTP through different routes ?
2:04AM 0 asterisk-cluster with one database
2:00AM 2 snom subscriptions issue on WRT
1:56AM 1 Agent Channel SIP transfer
1:26AM 1 Welcome to Join Asterisk MSN Groups!
Tuesday November 21 2006
9:59PM 1 Attn:Peter, Gsalas, Tim-Help me to configure my NOKIA E70 Mobile with my Asterisk server
8:51PM 0 codec information
8:47PM 0 Is this a PRI problem, *, or the phone???
8:27PM 2 Can anyone enlighten me as to what this means?
6:57PM 5 Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V
5:54PM 1 Hints no longer working in 1.4beta3 with Polycoms
5:20PM 4 IP601 Expansion Module HELP!!!
4:49PM 1 Is this possible?
4:28PM 0 OT: Reflash Mitel 5220 from MiNet to SIP
3:04PM 2 cmd Record
2:54PM 1 VM mail notification and locale
2:09PM 0 QoS on Linksys SWR208P?
1:40PM 0 voice breaking problems
1:40PM 2 Answer Machine Detection
1:39PM 1 Call to disconnected number on PRI just rings
11:51AM 3 IAX access to FWD broken?
11:30AM 1 Prefix/Suffix/StripLSD/StripMSD gone?
11:18AM 0 Re: Choppy sound in voicemail usingAsterisk1.2.11 on CENTOS4 guest on vmware server
10:50AM 3 Parking on an extension
10:45AM 0 RESOLVED - Snom 360 Multiple calls on hold help
10:23AM 0 Setting the default AMAflag
10:17AM 0 QueueMetrics 1.3.1 released today
8:14AM 3 Diva Server, chan_capi and tone detection
8:02AM 3 Cisco media gateways in general
7:50AM 2 FW: CISCO 7960G & Asterisk
7:12AM 0 Nortel CS1000 Asterisk with SIP
7:04AM 0 Which reliable source for ToIP security alerts ?
5:36AM 0 Signalogic SigC5561 PCI card
5:21AM 2 Handle Options Method
5:14AM 0 Callback agents without chan_agent issues (queue recording)
4:49AM 0 Forums
3:08AM 1 Hairping calls and Originating CLI
2:07AM 0 Can i have two asterisk versions running on samePC??
Monday November 20 2006
11:28PM 1 Reliable European SIP/IAX Providers?
7:27PM 2 QB intergreation
6:09PM 2 TDM400 native bridge echo
6:05PM 1 CAPI (Eicon Diva V-4BRI), Hylafax & IAXModem
5:10PM 0 is there a queue size call limit on the ACD?
4:47PM 2 How to secure access to PSTN line through Linksys gateway?
3:25PM 3 Spandsp rxfax txtax fails no errors
3:20PM 1 alert_info + Linksys 9xx + custom ringtone
2:54PM 0 install asterisk ad zaptel from source on red hat enterprise 3
2:32PM 1 Reset Extension Preferens
2:18PM 1 AW: Snom 360 Multiple calls on hold help
1:41PM 7 Snom 360 Multiple calls on hold help
12:52PM 4 Auto recording calls?
12:40PM 0 iSoftSwitch:: Asterisk & GnuGK Integretion
10:57AM 2 email etiquette (was: Re: Unicall MFC problems in 0.0.3+asterisk 1.2)
10:50AM 0 Set a feature within AgentLogin
10:30AM 1 How to accept All incomings calls from One Special Host (like a proxy)
10:09AM 2 Call limits and VoIP providers
9:45AM 1 SIP Multi-Domain
9:20AM 1 T.38 - By reinvitation only?
7:39AM 2 Help me to configure my NOKIA E70 Mobile with my Asterisk server
7:39AM 0 Compilation problem
7:16AM 0 .call file unable to hear or speak
6:27AM 1 Call-limit
6:04AM 1 g729 registered
5:22AM 2 Recording g729
Sunday November 19 2006
10:50PM 4 reduce dialtone volume on zap channel.
9:24PM 1 Vonage uses Cisco
8:23PM 2 switching trunks based on quality
10:20AM 1 PHP to .call file
10:05AM 2 WaitExten only reading 1 digit.
6:53AM 4 What card for E1R2?
4:28AM 0 MS-GSM codec issues - Anybody seen anything similar?
3:59AM 1 G723 pass-through and codec negotiation
3:27AM 2 Question on CDR Database
Saturday November 18 2006
7:37PM 0 Cant record phone calls
6:05PM 3 odd issue with IP tables
4:56PM 1 Re: Asterisk to listen for sip traffic on 80 and 5060
1:24PM 0 Statistics on Number of Minutes
12:05PM 0 Need help with a function
11:45AM 2 Dialout Conferences?
11:00AM 0 Cisco 2801 and asterisk
10:35AM 0 TDD/TTY device for the deaf
10:21AM 5 Asterisk Manager: equivalent of 'show channels'?
9:07AM 1 Hardware Echo cancelation
8:34AM 0 Using ChanSpy for spying voicemail
3:52AM 2 AdvancedVoIP Billing ?
3:50AM 0 If of external small box supply fxs Isdn and E1 ?
1:15AM 0 H323 no audio
Friday November 17 2006
11:26PM 5 spc.exe
10:56PM 2 strip + sign from incoming ${EXTEN} var?
5:43PM 0 Jitter Buffers in Zapata
4:52PM 3 Ringing a group of phones but not if they are busy
4:28PM 0 Destar release!
3:30PM 1 Asterisk - Do Not Call List
3:06PM 1 Extension Response Slow
2:07PM 0 automated response
2:06PM 1 TDM2400p and HW echo canceller
1:32PM 11 wget from within asterisk?
12:33PM 0 metermaid and 1.2.13?
12:12PM 5 Freepbx changes dont reflect in asterisk
12:08PM 3 voice quality of Aastra 480i CT and cordless
11:09AM 1 specify codec by domain?
10:50AM 0 Understanding the CDR with forwards...
9:51AM 2 1 FXO termination device
7:27AM 0 redhat enterprise 3
6:41AM 2 Need help on Music on Hold
5:35AM 0 Problem with Asterisk 1.4.0-beta3 and Digium TE405P
2:19AM 15 Siemens Gigaset SL75
Thursday November 16 2006
10:37PM 0 Call forwarding....
10:26PM 1 Asterisk on Solars?
6:26PM 0 asterisk OSX -astmasters site is gone
4:39PM 1 Multi-site Redundancy. Possible?
3:36PM 2 dialplan "*" and "0" key detection, not working
3:03PM 3 Nokia E70
1:48PM 1 AEL2 Confusion
1:41PM 1 asterisk billing software
1:28PM 1 Asterisk 1.2.13 can't load module
12:27PM 1 FXO PCI Master abort
11:43AM 0 Celliax LiveCD 0.0.32 released (Asterisk managing cellular phones, and Skype calls to/from cellphones, via chan_celliax)
11:31AM 1 hosted asterisk
10:20AM 0 jitterbuffer in pure voip (sip/iax) - what is best practice
9:34AM 1 zaptel, bristuff zaphfc, and florz question
9:14AM 0 call from cisco router to asterisk gets auto attendant
9:02AM 1 make: execvp: build_tools/make_svn_branch_name: Permission denied
8:52AM 2 POS Terminals
7:37AM 0 Backup and mail on trixox
7:16AM 0 Asterisk call recording
7:14AM 2 installing asterisk for Ubuntu Synaptic
7:12AM 0 upgading to install-misdn 0.3.1-rc23 broke dtmf detection on some calls
6:14AM 1 Sangoma A101 gives 'no PRI configured on span 1' error
5:46AM 0 dialing channel late
4:19AM 0 turning off DTMF detection on Zap channels
4:16AM 2 T.38 - make conclusion
3:27AM 5 spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
3:23AM 1 chanspy crash the asterisk 1.4
1:46AM 1 Trunk outcall line ?
1:10AM 1 queue management
Wednesday November 15 2006
10:40PM 0 Zaptel 1.4.0-beta2 compile error
10:24PM 2 Found GSM version, but any better WAV or ULAW recordings of "Steve" or "Ian" out there?
10:18PM 7 Do Not Call List
9:56PM 2 Installing Ztdummy on Fedora Core 5
9:12PM 0 Anyone using the directory.agi app in AGI perl
7:30PM 3 Regular audio fade-out fade-in on IAX2 calls Asterisk 1.2.4 Hi all, One of my users has a problem with many of his calls via my Asteriskā„¢ server. He describes the problem as having the sound slowly fade out and then fade back at a regular frequency. Has
6:42PM 0 Intercom function on eyebeam xten softphones.
6:23PM 2 Grandstream GXP2000 -- What's the Catch?
5:49PM 0 chan_unicall.c isntallation problem
5:30PM 1 simple mainmenu ivr tones not recognized
4:12PM 1 Attempting native bridge of
3:35PM 1 Queue - how to provide a caller ringing tone when some agent become available
3:34PM 0 Grandstream Programmable Buttons & Retrieving On Hold Lines
3:31PM 0 Auth Issue using Asterisk as Voicemail AND as Normal SIP Extension.
2:28PM 0 web interface to control zap interface
2:15PM 3 Set port to which Asterisk should send its answer
1:56PM 1 Monitor Zap Status - Full E-mail...
1:52PM 0 Monitor Zap Status
1:40PM 3 PortSip and Astericks new install
1:31PM 0 Huawei Videophone
1:11PM 2 Question about TFTPD server
12:28PM 2 PHPAGI example usage of input.php
12:18PM 2 safe_asterisks pawning multiple asterisk process???
11:48AM 0 Handy tip for intercom with FreePBX & Grandstream phones
11:40AM 1 quadbri + kernel
11:38AM 2 Got 200 OK on REGISTER that isn't a register
11:00AM 0 dtmf tones not always recognized
10:34AM 2 Page() Function Timeout
10:01AM 2 Problems with language support
10:00AM 0 Disabling Features Temporarily
9:51AM 1 Setting the CallerID
8:42AM 2 ODBC Voicemail Storage
7:54AM 1 State of a public number
7:28AM 0 SIP NOTIFY routing problem
6:56AM 1 How to disable the 482 Loop Detected messages sent by Asterisk
6:41AM 1 Asterisk - big installation
6:13AM 0 The best available CAPI BRI card for Asterisk ?
5:54AM 2 some questions about atxfer usage
5:40AM 2 T38 problem
4:58AM 0 Asterisk as a SIP client, Need to auto-answer
4:42AM 5 Time Based Voicemail Messages
3:14AM 1 How to do the Call Snooping
3:01AM 0 How to use Voipjet or any Voip provider Trunk from my mobile through fxo and fxs ports?
1:19AM 0 Condensing queue CDRs into single entry
Tuesday November 14 2006
11:13PM 0 Retain call control: Avoid letting call get
10:31PM 0 TDD - stops receiving characters
10:16PM 0 Caller Initiated Conference
10:08PM 2 Add Apps to Asterisk?
8:51PM 0 [SPAM HEADER] - trixbox + agi - Email found in subject
8:15PM 1 trixbox + agi
6:47PM 1 Call log reveals redundant calls!
6:04PM 2 ATA with reliable FAX?
5:58PM 1 How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
5:06PM 3 Caller ID in Sweden not working and looking for and voices
3:42PM 0 Voice mail transfer between 2 asterisk servers
3:34PM 0 How do I change the rtp packet size in a Cisco 7490 from 10ms to 20ms
2:50PM 6 unable to get channel lock BAD BAD BAD
1:36PM 4 In the beginning-The first question.
1:09PM 2 DUNDi Asterisk Cluster
10:41AM 0 asterisk sip doesn't see other asterisk-sip
10:34AM 1 Retain call control: Avoid letting call get into cellular voicemail
10:28AM 1 Dialplan options
9:15AM 0 Problems with voicemail
8:55AM 2 Problem with FXS ports of TDM400P
8:21AM 0 Fax killed on all zaptel devices
8:14AM 1 Broken Call Screening
7:31AM 7 900 rules
6:56AM 0 (no subject)
5:53AM 0 Zaptel and limiting number off channels channels
4:33AM 0 [Voicemail] Change the format of the VM_DATE
2:46AM 3 Is asterisk able to integrate with MS SQL
1:17AM 0 Redirecting Calls
12:29AM 2 Installation of Unicall for MFC/R2
Monday November 13 2006
11:35PM 0 chanspy -coredump( asterisk 1.4)
11:03PM 1 Newbie Questions . . .
7:45PM 3 Load balance Asterisk servers?
7:41PM 0 Application Directory question
7:33PM 2 Linksys doesn't resync properly and doesn't get provisioning from TFTP and HTTP
7:30PM 0 CDR shows NO ANSWER when call is really ANSWERED
6:50PM 0 Data over zaptel
6:50PM 6 Dual Wan Router with Failover
6:04PM 0 establish meetme limit for a single room
5:31PM 1 Bellsouth issue ?
4:51PM 2 STUN with one public and one private IP?
4:03PM 0 Recommend software version for Cisco IOS Gateways and SIP Phones?
3:42PM 3 "Username/auth name mismatch" + SIP phone can't connect?
3:41PM 0 Zaptel
3:16PM 0 Survey: In what ways do you use Asterisk at your house?
2:49PM 0 Question about MySQL Fetch foundRow from the dial plan
1:47PM 1 SIP Ports (1000 to 2000 works)
1:33PM 1 Dial/Continue/Announce
1:29PM 0 MWI not working in 1.4
1:14PM 0 Native TDM Bridge
12:27PM 1 Voicemail argument size limit
12:24PM 2 FAX using T38
12:18PM 0 Asterisk with ss7 and sip-t
11:48AM 1 Can AGI do this?
11:41AM 1 asterisk as a Media Gateway
11:37AM 3 Mysql 6 second rounding
11:29AM 1 DSl and more then 1 call
11:18AM 1 Music on hold question
11:11AM 0 Fast Busy with autodial using a call file
10:53AM 0 newbie question
10:02AM 0 2 * servers Host=ip - doesn't work Host=dynamic with register is OK, why?
9:52AM 0 Slow playback of sound prompts
9:49AM 2 Recording outbound analog calls with X100P
9:14AM 1 problem with redirects
8:37AM 2 Custom voicemail extension greeting
8:31AM 1 Defunct / zombie AGI after some execution time
6:44AM 2 Problem with internet down
6:15AM 3 FW: Desktop integration
5:27AM 0 Question about the GUI for 1.4
4:46AM 4 Asterisk IVR functionality
4:28AM 8 Desktop integration
3:48AM 1 Dial : Executing context/priority after bridge?
3:07AM 0 Voicemail and realtime : the emailbody option ...
2:32AM 0 bindport
1:43AM 1 Sending '#' with Dial
12:23AM 0 Can i have two asterisk versions running on same PC??
12:15AM 2 Can i have two asterisk vcersions running on same PC??
12:08AM 1 Moh stops immediately
Sunday November 12 2006
9:58PM 0 Asterisk VM with Cisco routing
8:58PM 0 Trixbox dialout problems
7:40PM 3 Slow to get dialtone when going off hook - big problem for me :(
6:26PM 1 Zaptel compile problems
4:45PM 2 Headaches with Video over SIP
3:48PM 2 IAX2 one way audio
3:44PM 0 cadences zapata.conf
2:29PM 2 same extension on softphones and hardphones
1:59PM 0 Asterisk billing
1:43PM 3 Determine if Call is from a cellular phone
1:08PM 1 outgoing works, incoming fails on asterisk passthrough to inter-tel
10:42AM 0 VM problems...
9:27AM 1 Some pictures from Astricon 2006 in Dallas
8:48AM 0 Asterisk Media Gateway
8:27AM 3 Looking for a simple TFTP server for Linux
7:39AM 0 Speeding up SayDigits?
7:33AM 0 asterisk-addons 1.4 SVN fails to compile
6:47AM 2 dynamically modifying the dialplan?
1:50AM 1 Knowing when an answerphone answers
Saturday November 11 2006
10:10PM 2 CLI message: remote unix connection disconnected
7:31PM 1 No sounds in svn version?
12:46PM 1 sip forward behind a nat
10:11AM 0 can't hear MusicOnHold when zap answers
8:35AM 1 Call file: CallerID problem
3:43AM 0 chansp core dump
12:49AM 1 Soundfiles adding during phone calls
Friday November 10 2006
10:05PM 0 app_swift: Failed to set voice
6:20PM 2 Dialing from "Placed Calls" on Polycom IP501 doesn't always work
5:37PM 0 Push to Talk settings.
5:25PM 3 SPA-941 (and others ) Transmit Sound Quality
3:20PM 1 (no subject)
2:55PM 0 monitor-join does not seem to work.
1:53PM 2 config template for Grandstreams
1:45PM 2 WIFI phones on asterisk
12:43PM 0 app_pppd - Could not read send data
12:30PM 1 Harris picking up before extension
12:22PM 0 Returncode from command
11:58AM 0 Realtime & sippeers using NAT
11:11AM 1 Choppy sound in voicemail using Asterisk 1.2.11 on CENTOS4 guest on vmware server
10:56AM 1 Need to automatically park an incoming call and then connect to an extension.
10:52AM 3 How to get CDR to show answered calls only
10:01AM 0 Re: Asterisk and Max TNT SIP Authentication Issue, WORKING
9:03AM 1 Question about Mitel phones
7:36AM 2 Outgoing problem on PRI
7:22AM 0 Pointers/suggestions?
6:35AM 2 Presence-awareness in Asterisk
6:33AM 0 VM notification to pager and phone
6:01AM 1 Queues and Timeouts.
5:59AM 9 Stable clock with 2.6 and without Digium hardware.
5:14AM 1 Looking for IP phone / ATA that has builtin VPN support
5:07AM 0 SV: Dropping Connections
4:56AM 1 Dropping Connections
3:21AM 1 EuroISDN+ and Callers name
12:52AM 0 Asterisk BlindTransfer behaves differently in version 1.0 and 1.2
Thursday November 9 2006
10:07PM 3 announcing inbound PSTN calls
7:12PM 1 DTMF problems with IVR - What DMTF Tx method
7:11PM 2 Powering SNOM 200 phones?
4:38PM 2 Latest Debian and latest zaptel
4:24PM 0 Mitel 5224 & Asterisk Distinctive Ring -- Anyone have it working?
4:20PM 7 Modprobe Zaptel
3:35PM 1 Zaptel 1.2.11 released
1:49PM 1 porting numbers away from packet 8?
1:07PM 0 Harris 20-20
12:28PM 2 register suddenly fails
11:36AM 1 wip5000 roaming
11:28AM 1 New Asterisk 1.4 GUI
10:22AM 1 Station Voip Brazil
10:15AM 0 special characters in alphanumeric extension s
10:07AM 2 A couple of new tutorials: installing * 1.4 and the Asterisk GUI
9:58AM 1 Quick Q...
9:50AM 0 Bug ???
9:21AM 1 special characters in alphanumeric extensions
9:14AM 2 asterisk and norstar
9:00AM 1 unsubscribe
8:36AM 5 DUNDi precache
8:32AM 5 Voxee lag problems ?
8:19AM 1 Problem with register command in SIP.conf
5:23AM 2 Alcatel trunk with asterisk problem on dialing digit-by-digit
4:53AM 3 SRTP
3:22AM 0 TDM, loopstart and modules GSM Nokia32
1:56AM 1 Problem with CDR interpretation
Wednesday November 8 2006
11:22PM 0 OT - Polycom https provisioning
9:54PM 1 DID billing with a2billing
9:51PM 0 Unknown caller id problem
7:42PM 1 Auto record a call?
7:15PM 1 Ask users.conf
6:36PM 0 Queues: member order vs. defines in queues.conf
4:45PM 1 Reg errors? Other anomalies? Check those capacitors!
4:25PM 1 Still problems with Asterisk on latest Debian
3:42PM 1 I LOVE IT
2:42PM 0 [FC5] How to update kernel/kernel-develop for Athlon?
2:32PM 0 sms script on receive
2:07PM 2 Off-Site Extensions That Would Show As In-Use?
1:35PM 0 Warning: "Channel does not have a CDR" when doing ForkCDR
1:29PM 1 Microsoft will enter VoIP market in earnest
1:18PM 1 Re: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer
1:15PM 5 DTMF Corruption Problem
12:09PM 1 talking caller ID
11:38AM 1 Re: asterisk iax2 monitoring
11:35AM 2 One-Way-Audio After placing call on hold
10:06AM 1 Re: Asterisk and Max TNT PRI to SIP Authentication Issue
9:52AM 1 Delay between DTMF Down & Detected Digit
9:15AM 1 FIC-GTA001
8:35AM 0 jpeglib
8:14AM 1 VLANs and Quality
7:58AM 1 Performance issues in Realtime
7:56AM 1 HANGUPCAUSE for unalocated number?
7:51AM 0 Asterisk 1.2.x and video
7:30AM 1 Ringing phones
7:27AM 1 I need (some) help in configuring PAP2.
6:22AM 0 Asterisk 1.4 and Queues RealTime
5:55AM 0 Odd results from fxotune?
4:30AM 0 asterisk and peep tone (network tone)
4:17AM 1 faxing times!
3:55AM 0 no sound when bridging 2 asterisk SIP connections
2:49AM 0 Asterisk CTI - SAP R/3 Intergration Certification
2:19AM 1 Agents that handle calls from multiple queues
2:19AM 0 Queue forks asterisk and then leaves theextraprocesses lying around
2:00AM 1 Queue forks asterisk and then leaves the extraprocesses lying around
1:54AM 0 Queue forks asterisk and then leaves the extra processes lying around
1:34AM 2 flash transfer problem in asterisk integration with old PBX
1:33AM 1 Operating queues with clients on a legacy PABX
Tuesday November 7 2006
11:31PM 0 Follow Me problems
10:58PM 0 RxFAX - How to catch errors in the dialplan
6:42PM 0 test please ignore
6:24PM 1 Help with latest Asterisk on latest Debian
6:11PM 1 Glitches in sound every time that Asterisk receives reINVITEs
6:11PM 1 Fax & Digium
6:11PM 1 Why dont my messages get through
6:11PM 1 [resolved] asterisk 1,4 and google talk
6:10PM 0 test message please ignore
6:10PM 2 Microsoft will enter VoIP market in earnest next year, says Ballmer
6:10PM 0 astertest
6:10PM 0 incoming call destination: IVR not working
6:09PM 2 Pressing "*" makes Asterisk destroy my call
6:09PM 3 Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
6:08PM 0 How is ANI "Usually" sent on an ISDN PRI?
6:08PM 1 loosening voicemail file permissions for msg????.txt and msg????.wav
11:39AM 4 Queues and multiple lines
11:10AM 1 Grandstream TFTP system wide settings
11:08AM 0 my future count on this please help
11:06AM 2 g729
9:45AM 0 Generating Recall/Flash using Zaptel
9:14AM 3 connect Sipura with Asterisk - both behind NAT
9:02AM 4 "Sticky" Polycom 501 keys and handset
8:59AM 2 hicecaller ID
8:21AM 0 failed to authenticate on invite
8:14AM 1 How do I make this stop? (Bridging of IAX channels?)
8:03AM 3 capiAnswerFax
6:23AM 0 Asterisk SMS: Experience with EMS?
5:42AM 1 Problem: 2 second silence at the beginning of most calls
3:28AM 2 Mapping CLI'S in Dialplan
3:15AM 1 Upgrading sox
2:23AM 0 Asterisk Showing 404 not found when calling from third party SIP server (newbie question)
1:54AM 2 Snom 360 flickering screen
1:27AM 0 Asterisk and FreeTDS 0.64 or >0.63
1:12AM 0 Desired apps
12:29AM 1 Dial plan Question
Monday November 6 2006
9:05PM 0 silencedetecthangup=
6:39PM 1 asterisk 1,4 and google talk
6:19PM 2 Polycom autoprovision behind a NAT
5:34PM 3 Question on Aastra phones and Astrisk
5:23PM 0 help for recording
3:14PM 1 Polycom dealers in Toronto/London ON
2:21PM 2 how to indicate an non-existent number?
2:16PM 1 Do my messages come through?
1:29PM 0 IAX FWD - down, running own proxy or stun server
1:27PM 0 TrixBox and MP104 FXO (AudioCodes GW)
12:40PM 0 Disappearing voicemail?
12:13PM 0 OT: BarCamp USA
11:31AM 1 Polycom unable to answer more than 3 calls at a time
11:30AM 0 Re: Definity ISDN PRI
10:40AM 1 Asterisk servers being greedy and not letting go of the media path. (using IAX2 channels)
10:12AM 2 receptionist - large number of concurrent calls - example needed
10:03AM 2 Queue time out
8:28AM 7 several behind NAT
8:26AM 1 Register vs. Host=IPADDR
8:22AM 1 Page system using the sound card
8:15AM 1 Amending CLI in Dialplan
6:47AM 4 Port Range
6:31AM 1 Is it possible have multiple ip numbers for an extension?
4:04AM 2 Ring locally when home or roadwarrior via IAX when away
3:53AM 1 Audio goes one way during the call for a fewseconds. Is it RTP, NAT, dyndns, or what it is?
3:14AM 7 DTMF Tones occuring randomly
2:36AM 2 Fast detection of unreachable SIP clients?
Sunday November 5 2006
8:30PM 1 asterisk DTMF detection
7:41PM 0 xfsound=beep is not beeping
6:02PM 0 Use astbill to bill Trixbox
4:09PM 1 Definity Asterisk Caller ID Issue
2:54PM 9 names of SIP aware firewalls
2:45PM 3 Very high translation costs for g729
1:57PM 1 Call Quality Issues with IAX?
11:18AM 0 Voicemail.conf multi languages
9:31AM 0 Free PBX, was - Re: best gui
7:52AM 2 Definity Asterisk CallerID Issue
5:05AM 0 call transfer problem
4:34AM 1 Reading Voicemail Config from MySQL
2:51AM 1 Asterisk and FXO Digium Card for Analog line
2:00AM 1 skype and SIP hardware for linux
1:40AM 3 Anybody used Asterfax?
12:02AM 1 Hang up on SIP calls if connected to long
Saturday November 4 2006
9:54PM 1 Newbie questions about Voice mail
9:37PM 4 SPA3k wired to PAP2 for echo testing
7:16PM 1 FXO lines taking several rings to answer, always two
6:40PM 1 Only one out of 10 remote extensions expiring registry
6:34PM 4 g729 codec help
5:30PM 1 Pass through
4:02PM 1 Redirect problems using IAX2 and SIP
10:53AM 0 Upgrading from to 1.2.13
8:25AM 0 Asterlink Down?
8:24AM 0 I suggest using TFTP.
8:10AM 1 Hairpinning problems using IAX2 and SIP
5:19AM 2 app_prepaid won't load - undefined symbol mysql_num_fields
4:39AM 1 My first Asterisk - Not recognizing X100P clone
3:29AM 0 iax2 qualify - false "peer unreachable"
2:14AM 2 Asterisk upgrade from 1.0.9 to 1.2.6 not working
Friday November 3 2006
9:59PM 1 Polycom SIP 2.0.2 firmware
9:58PM 1 Patton 1400
8:56PM 0 DID with extensions
7:40PM 1 Why only one out of many IP Phones re-registering every one minute
5:46PM 1 SendDTMF() behaves strangely
5:30PM 1 Unicall's MFCR2 with Asterisk 1.4
5:17PM 1 Polycom provisioning and Pure-FTP : problems
1:44PM 1 Experiment: Dialplan size vs. Speed
12:56PM 1 Random 'no audio' problem
12:07PM 1 International dialing with GPX-2000 and "early dial"
11:36AM 3 Extension Spy
11:27AM 1 SIP - IAX Attended transfer
11:20AM 0 Problem with Realtime/ODBC
9:17AM 0 Binding a peer context to a specific IP address
9:12AM 0 Configure Max TNT PRI to SIP with Asterisk
8:45AM 1 Monitor, MixMonitor and volume levels
8:30AM 2 Is fax bridging with TDM2400 working (or about to work) ?
8:27AM 2 AEL2 in 1.2
8:24AM 3 Problems Overwriting CallerID with True ANI
8:02AM 1 Clearing Outgoing Call Queue
7:57AM 1 In bound SIP context issue
7:22AM 3 Nortel Option 11C and SIP gateway integration
6:43AM 0 Caller ID 1.2.10
6:39AM 1 TDM400 hungup problem
6:26AM 0 Pass-through any codecs
5:57AM 1 Error updating bootrom on Polycom phones..doesn't even download the bootrom!
5:48AM 4 some simple newbie help with dialplan needed...
5:43AM 1 Help for registration with "sipdiscount"
4:26AM 4 Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?
4:22AM 0 *****SPAM***** Meetme Conference Rooms
4:20AM 1 How do i redirect a call without answering it? SIP channel
4:04AM 0 The most cost effective and Asterisk friendly T.38 gateway ?
3:45AM 0 How to reduce latency and first real ring timing?
2:16AM 0 Asterisk IAX Trunk and Queues
1:26AM 2 PAP2 to use on my asterisk.
1:21AM 1 SV: ip address in CDR
1:16AM 1 Cisco 7960 - Fast dial
Thursday November 2 2006
11:05PM 1 is IAX required for firewall and router?
10:17PM 0 ip address in CDR
9:57PM 2 fax eater
6:42PM 0 a extension intentionally dropped in favor of * ?
4:08PM 0 Recall: regexten & regcontext broken for SIP?
1:42PM 0 Polycom 501 supports now FTPS?
1:17PM 0 Asterisk 1.2.16 AIX2 - SIP Attended transfer
12:38PM 3 Polycom latest version
12:31PM 2 Tampa Bay Asterisk Users Meetup on Monday
11:31AM 4 Out Dial Interface for Asterisk
11:19AM 0 testing
11:14AM 1 Voicemail issues
10:52AM 0 Problem with MYSQL commands in dialplan
9:46AM 0 Static Realtime Select from Database
9:13AM 1 AGI Problems
9:12AM 4 Running asterisk with 'sudo'
8:51AM 2 Grandstream HandyTone-488 with Asterisk ?
8:13AM 1 AstLinux 0.4.4 Released!
6:39AM 3 How to determine which version is running
6:18AM 0 Subscriptions and call back on busy problems with Snom phones
6:09AM 2 Error installing asterisk, module zaptel not found
4:48AM 1 VM Language
4:46AM 2 Can some moderator kick this person out of the list
4:41AM 1 How to clear trixbox configuration
4:27AM 1 Lucent TNT Help
4:18AM 0 blindtransfer and initiator hangup
4:11AM 0 Macro variables and redirects
4:05AM 0 Wait for an extension and dial. Why does this not work?
3:59AM 0 sound-files not playing?
2:59AM 0 Extending a call limited by L in Dial app
2:20AM 1 Auto dial out and auto answer
2:09AM 1 ZAPtel channel dance
12:47AM 3 mpg123 new version
Wednesday November 1 2006
11:55PM 3 Polycom 601 Phone can not find TFTP server
11:25PM 0 Using asterisk as a call router between pbxs
10:54PM 1 Asterisk Manager and Ruby
10:45PM 1 Videoconferencing solutions with Asterisk-
10:08PM 2 echo with spa-3000
9:26PM 4 My Phone Review- Large Scale Corp Deployment.
8:48PM 0 Problem with libpri?
8:27PM 0 New Dell range
7:39PM 2 Two Sipura 3000s
7:38PM 1 IAX problem
5:26PM 0 Fwd: Benachrichtung zum +ANw-bermittlungsstatus (Fehlgeschlagen)
5:11PM 1 PURE OUTBOUND setup (how do I proceed from here?)
5:11PM 2 Realtime, DUNDi and regexten
4:18PM 2 Echo Issues
3:47PM 3 Sound breaking. Because of Tormenta2 PRI Interface Card or something else
3:32PM 1 connecting internal line with external line
2:45PM 0 TE110P Card Little help
2:34PM 0 Can I use Realtime entries to do multiple registers to same trunk/peer
2:05PM 6 Java Web Phone
12:47PM 1 imap on debian
12:14PM 3 Remote-Party-Id and Attended Transfers
11:12AM 2 Still no CLI in 1.4 branch (OSX)
11:09AM 0 Cisco 7960 password/shared secret problem --- Related to OS X ?
11:04AM 2 Polycom Managment tools
10:06AM 0 AW: Which IP phones have best voice quality, preferably under $150
9:32AM 0 [SPAM HEADER] - Which IP phones have best voice quality, preferably under $150 - Email found in subject
9:28AM 5 DTMF over IAX
9:28AM 1 Upgrading from 1.0.9 to 1.2.6
9:23AM 0 [SPAM HEADER] - RE: Re: Newbie Questions - Grandstorm phones? - Email found in subject
9:20AM 2 Asterisk manager
9:17AM 4 Which IP phones have best voice quality, preferably under $150
9:01AM 3 Re: Newbie Questions - Grandstorm phones?
8:26AM 1 [SPAM HEADER] - Re: Snom or Cisco Phones? - Email found in subject
7:33AM 0 Neat Application for Text to Speech
7:19AM 2 a2billing
7:10AM 0 wav format isn't compatible with Windows Media Player
6:22AM 0 AEL2 - CUT function usage
4:55AM 2 Help me on Call parking
4:23AM 3 Manager API - Originate Call - Need Help
4:00AM 0 SIP realtime issues
1:25AM 0 Need help connecting Alcatel 4400 PBX to Asterisk