asterisk users - Dec 2005

Saturday December 31 2005
9:47PM 0 Voicemail through outlook
7:12PM 0 SPA-3K FXO: Incoming and outgoing calls in different contexts?
5:55PM 0 Need HT488 FXO example for both inbound and outbound.
3:11PM 0 New Manager Client Program
1:46PM 0 OT Maybe: Anyone have any knowledge of v5.1/v5.2 in connection with Asterisk?
11:41AM 5 voicemail/privacy system
7:52AM 25 How to set features.conf to change the hangup key.
4:57AM 0 [Announcement] chan_capi-cm 0.6.2 released
3:34AM 13 GXP-2000 fw and NTP
12:33AM 1 Multiple Realm Definitions?
12:10AM 0 AGI Variable
Friday December 30 2005
10:57PM 0 answer supervision and POTS
10:06PM 2 Sip man in the middle
10:04PM 1 RE:problem with X100P card
9:13PM 0 Re: Go directly to new messagesfromVoiceMailMain?
6:53PM 18 name that vendor...
6:49PM 0 Dicate not completing the DialPlan?
6:17PM 0 Best way to terminate calls
5:49PM 0 sip through nat problem
4:01PM 20 GXP-2000 any good with * ?
2:50PM 3 Outputting human readable info on a VoIP call's quality?
2:49PM 1 voicemail .wav filename
2:29PM 3 Recording Calls for Specific ACD Agents
2:16PM 1 NOOB: Need Help Learning How to Debug PRI (U.S.)
2:14PM 6 Aterisk 1.2.1 zaptel module not found
2:02PM 0 Motherboard choice for large opteron based asterisk server?
1:58PM 1 Cheap FXS/USB terminal SE-B2K, can it work with asterisk?
1:32PM 0 Passing authentication to an analog adapter
12:38PM 2 Which Asterisk GUI?
12:17PM 6 Fax Support
12:12PM 0 Re: Asterisk-Users Digest, Vol 17, Issue 176
11:26AM 1 ENUM trees
11:16AM 0 Manually Opening and Closing a Queue
11:09AM 0 Vonage Sip Peering
10:52AM 0 call
10:41AM 2 Playback after Page()
10:38AM 1 IAX problem - Bug or Compatibility issue?
10:20AM 2 FOP Maximum extensions?
10:11AM 0 TDM400 FXO outbound issue
9:34AM 3 RPID Issue
9:28AM 13 using a Gigaset SX440isdn on a Diva 4BRI ?
9:26AM 1 Notifications when host fails qualify
9:18AM 0 Has anyone used the applicationmap in features.conf?
9:15AM 0 TBCT For PRI support
8:58AM 0 MYSQL Fetch Warning
8:52AM 0 No RTP Warning
8:23AM 3 wctdm module goes missing after a reboot - Gentoo?
7:34AM 5 Problem on ZAP channel
6:38AM 3 Queue features
5:16AM 5 Can we dial agents from extensions.conf
5:06AM 4 Howto config tdm2400
4:12AM 3 Asterisk connect to voicemaster configuration 1.7
3:43AM 0 Outbound call using ISDN extension disconnected after *exactly* 30 seconds
2:33AM 0 Re: CALLERIDNUM (Rehan AllahWala)
Thursday December 29 2005
11:19PM 0 RE:probelem in working of X100P
8:34PM 2 Re: Go directly to new messagesfromVoiceMailMain?
7:58PM 0 silent dial/ring?
7:00PM 0 Sending Polarity/DTMF Caller ID in chan_zap (Sweden etc...)
6:58PM 3 SetAccount missing?
6:47PM 36 Semi-OT: porting numbers away
3:22PM 4 Allison on Free 411
2:43PM 2 transfers using # in asterisk
2:09PM 2 Getting Yoda unit to register all four ports
1:41PM 2 voicemail storage over odbc and postgres
1:19PM 4 Regular modems?
12:56PM 2 Linksys SPA-942
12:41PM 28 Realtime Multiple Asterisk boxes andrtcachefriends MWI
12:15PM 0
12:14PM 2 Realtime Multiple Asterisk boxes and rtcachefriends MWI
11:49AM 0 Gnet VP168S
11:35AM 3 Problem getting D channel up on Sangoma A102
11:18AM 0 PRI Hangup cause
11:04AM 1 Asterisk SIP PORTS
10:49AM 3 Asterisk 1.2 + DMZ + NAT clients
10:21AM 0 CoreDump
9:24AM 8 What does "Page" application do?
9:16AM 3 zaptel TDM21B 4-5 second pause
9:07AM 1 Asterisk Server Hangs
9:00AM 1 Re: Go directly to new messages fromVoiceMailMain?
7:32AM 5 Easiest way to use HFC-S?
7:31AM 1 SNOM 360 locked up SOLVED
7:28AM 1 smsq
7:23AM 3 SPA-3000 + call waiting
4:57AM 0 Driver not configuring correctly on TE210P forCCS
4:20AM 3 Congestion problem
4:00AM 1 HELP! Asterisk 1.2.1 stops immediately - voicemail problem?
1:50AM 1 I thought they weren't charging - FW: [] Happy holidays wishes from
Wednesday December 28 2005
11:16PM 5 TDM2400 wierdness
9:56PM 5 Grandstream Configuration Utility available
9:45PM 2 Go directly to new messages from VoiceMailMain?
4:06PM 3 Conditional CODEC translation
3:25PM 7 Regular crashes
2:54PM 4 Polycom check-sync
2:29PM 2 Soundstation 4000
2:22PM 4 sip debug > file.txt
2:16PM 0 Static (Distortion?) and noise on FXO (TDM04b)
2:09PM 13 Asterisk as a Gateway
1:43PM 0 spool calls always failing with congestion (AST_CONTROL_CONGESTION)
1:30PM 1 extension not ringing when dialed from DID
12:38PM 3 Problems with multiple outbound calls going to PSTN - Wildcard TE405P
12:10PM 0 how to alter cdr dst info?
12:05PM 1 Iterfacing with a Mitel PBX
12:02PM 4 Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem
11:06AM 0 GSM-gateway setup
10:51AM 0 Re: 26. RE: Stay away from Grandstream! (Bjorn Asmul)
10:40AM 2 SIP to SIP calls
10:10AM 2 Most Stable Version of Asterisk
10:04AM 0 Tr: Re: call test
9:48AM 1 Driver not configuring correctly on TE210P for CCS
9:40AM 0 BUG? AGI stuck in ast_waitfor_nandfds()
9:39AM 6 What setup
9:18AM 0 MWI problem
9:17AM 3 call test
9:08AM 2 CallerID info needed
9:07AM 7 who is online
7:51AM 9 voip-info: Asterisk record calls
7:48AM 4 oh323 configuration
7:16AM 1 ipVolution
6:44AM 3 Sipura 2002 codec preferences
3:48AM 1 Wrong Password?????
3:36AM 1 CLI execute extensions
2:14AM 1 billing system
2:09AM 4 Bad Checksum answering inbound call
1:10AM 2 PHP Manager
Tuesday December 27 2005
11:36PM 0 Trial Edition of Druid Asterisk Web-interface
11:33PM 1 Maximizing audio quality
9:54PM 12 4-port external sip fxo which doesnt suck?
9:36PM 0 Asterisk seg fault (SVN-branch-1.2-r7641)
5:26PM 0 [Announce] Pending Web-MeetMe update
5:07PM 0 Difference between CDR dispositions..
4:22PM 10 Automatic logoff of all agents at set time
2:33PM 3 Play soundfile before snswer
2:06PM 0 Polycom Soundpoint 501 outbound calls always show NO ANSWER
1:57PM 0 How to register a sip user/peer in real time
1:31PM 0 Asterisk Realtime Database Redundancy
1:25PM 1 agent logs
1:12PM 0 MSN Messenger / Windows messenger Passport service With asterisk any one ?
1:11PM 1 polycom sip slower than grandstream
12:03PM 5 UK, Disconnect supervision
11:28AM 1 Asterisk does not handle call from a Cisco IAD correctly
10:50AM 7 Blackberry SIM card
10:26AM 6 Realtime Static/Dynamic Preference
10:23AM 1 Cisco 7912G through NAT, problems with tones detection.
10:18AM 1 Login incorrect on ZIP2 phones when checking voicemail
8:59AM 1 Strange IAX messages on the console
8:49AM 0 Callerid ID lookup program updated (CID_rewrite v1.2)
8:33AM 1 "one touch record" on asterisk 1.2.1 uses monitor and not mixmonitor
8:27AM 0 TDD/TTY - How does one use this?
8:05AM 2 Cisco dtmf
7:40AM 2 CDR_CSV stops writing, help!
7:30AM 0 RE: [Asterisk- Pls. explain what happens...
7:24AM 0 Asterisk 1.2.1 and X100 clone Zap problem
7:23AM 13 spandsp & fax
7:05AM 2 Asterisk on VPS
7:03AM 3 How to check Digium TE410P firmware version?
6:51AM 1 Polycom IP301 time changing
6:18AM 5 IAX media path, forcing server to stay in the middle
4:38AM 2 Changing Automon filenames?
4:12AM 1 SIP ENUM Daemon
3:32AM 0 Asterisk+mgcp setup+vrg 121
1:26AM 2 Pls. explain what happens...
Monday December 26 2005
11:37PM 1 iptables rules for forwarding SIP/RTP to Asterisk server from behind nat firewall/router
11:20PM 0 SIP "403 Forbidden" Errors...
7:40PM 26 Stay away from Grandstream!
3:18PM 1 Operator breakout from voicemail
1:12PM 2 64 bit Zaptel?
11:58AM 2 Asterisk lines go into PBX?
11:56AM 10 Delays in IVR
10:24AM 0 NEW Asterisk Management Interface withJavaManager Live Console.
8:31AM 3 channel monitoring whisper mode?
7:52AM 0 NEW Asterisk Management Interface with JavaManager Live Console.
6:04AM 1 RE: how to make contribution in asterisk
5:54AM 4 Eicon DIVA Server V-BRI questions
2:32AM 0 NEW Asterisk Management Interface with Java Manager Live Console.
1:05AM 10 Asterisk Christmas Help request
12:20AM 0 No of records in calls table
Sunday December 25 2005
9:58PM 6 Problem with date & time on Aastra 480i since release 1.3
4:39PM 10 Channel bank timing
4:14PM 3 weird problem with sipura spa2000 and soundcardpa setup
3:11PM 1 weird problem with sipura spa2000 and sound cardpa setup
2:48PM 0 weird problem with sipura spa2000 and sound card pa setup
12:32PM 8 Cisco PGW-2200 OR Asterisk
11:07AM 1 newbie question about making outbound call
Saturday December 24 2005
7:03PM 0 Re: Asterisk-Users Digest, Vol 17, Issue 148
5:08PM 3 System(...) but how to pass parameters?
12:04PM 3 PRI outgoing caller ID stopped working
9:18AM 2 Dialling out with clone X100P board
5:02AM 7 CAPI and *
4:53AM 0 Laptop PCMCIA ISDN card
Friday December 23 2005
9:52PM 0 Feature: Attendet transfer with original caller ID
9:02PM 1 Problem with Xlite free phone(Xten)
8:07PM 0 Asterisk With Yahoo messenger
4:19PM 4 Aastra firmware 1.3.x (Far-End sound level issue)
3:22PM 1 problem with tdm400 fxo
3:18PM 1 AMP stuff via CLI?
3:16PM 6 tdm400 fxo problem
3:15PM 4 ASterisk and home lines.. DGM-TDM01B or x100 ?
3:04PM 0 Probs with outbound calls
12:24PM 2 TDM2400P driver change
11:00AM 9 Matching SIP users and peers
9:51AM 5 List Of Defined Variables
9:18AM 9 SIP permit/deny
9:18AM 0 provu 2100 videophone and asterisk
9:03AM 0 FYI on zttool output on SMP system
7:13AM 0 no have dial tone
6:55AM 0 how to get the number of an external phone with Asterisk Manager
6:41AM 4 How to make Asterisk to generate and terminate calls
2:23AM 2 Is there a GUI for asterisk realtime
2:14AM 2 chinese asterisk related web site opened......
1:54AM 4 Virtual Memory Usage
1:25AM 0 Feature: Attendet transfer with original caller ID
1:22AM 7 Merry Xmas to everybody!
1:03AM 10 Grandstream Budge Tone 102
12:52AM 0 dialing outbound using te411p
12:43AM 1 No sound problem, chan_sip.c:3451
Thursday December 22 2005
11:44PM 2 Malformed CallerID freaks out SIP channel
9:56PM 5 Emergency Information Needed: sip.conf - bindport allow multiple ports?
9:17PM 4 What hardware fits my needs?
5:23PM 2 automon doesn't work with 1.0.9
4:27PM 4 Asterisk@Home Fax to Email problems
4:08PM 0 FOLLOWUP: .call files on PRI / Zombie AGI proces ses in FC2 / 1.2 Beta 1 under load
3:07PM 11 Creating conf files from db
2:18PM 0 Re: Fw: Legacy PBX -> * -> Voip Calls problems
1:52PM 5 SNOM 360 locked up
12:51PM 0 Fwd: Legacy PBX -> * -> Voip Calls problems
12:45PM 0 forwarding a caller to a conference room
12:23PM 0 chan_sip.c error message
12:01PM 4 ast_sock_cmd: pipe commands to asterisk
11:31AM 12 wav to g729
11:03AM 0 Asterfax beta4, Asterisk 1.2.0 and issue sending FAX
10:57AM 0 [POSSIBLE SPAM] RE: Identifying Frame Slips from PRI debug
10:30AM 1 how to follow a call in the console
10:21AM 1 Manager API connections - crashes?
10:19AM 0 Make Asterisk explicitly UNREGISTER from a SIP service?
10:09AM 0 Zap Error
9:21AM 5 recording queue calls
9:01AM 8 TDM2400
8:12AM 10 asterisk AVM C2 again
7:48AM 3
7:02AM 1 anybody getting "No authority found" with teliaxnow?
6:58AM 1 anybody getting "No authority found" with teliax now?
6:26AM 1 need help in building dynamic conference
5:55AM 0 DTMF - > FSK CallerID problems
5:48AM 6 Anyone doing NAT through m0n0Wall?
5:47AM 6 snom Firmware 5.0.
5:29AM 0 Codec selection in dialplan
5:23AM 1 PRI problems: B-Channel restart
4:47AM 0 *1.2.1 setcidnum from Zap
4:40AM 2 IAX No Authority found
4:28AM 0 realtime & SIP
3:51AM 3 unplugging E1 cables while asterisk running
1:48AM 5 Problem with octobri and x100p clone
Wednesday December 21 2005
8:34PM 17 How to record a call
7:53PM 1 Daily Phreak - Daily Telecom, Asterisk and Phreaking Updates
7:45PM 4 Calls not incoming to any extension from remote proxy server
7:37PM 1 Weird rtpmap issue
6:14PM 10 Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!
4:29PM 0 (no subject)
2:57PM 0 SIP configuration for uip200
2:39PM 5 caller_id and law
12:56PM 1 DaemonTools Supervise
12:26PM 1 FAX Problems - PRI, Adtran and ZetaFax
12:16PM 0 Asterisk/Zaptel on Kernel 2.6 and ACPI
11:32AM 2 New To Asterisk/POTS - Hardware Setup Questi on
11:28AM 1 Extension cannot match ! receiving call mISDN ...
11:26AM 1 New To Asterisk/POTS - Hardware Setup Question
11:13AM 6 MWI not working - using seperate vm and call routers:
11:11AM 2 chan_capi-cm 0.6.1 won't load
11:02AM 0 enabling while/endwhile
10:55AM 0 Asterisk's VoiceMail server accessed through various DIDs
10:10AM 0 no subject
10:09AM 0 Crash
10:02AM 0 Name file automatic
9:40AM 1 realtime sip firends not being updated
9:31AM 0 Some values ignored when using static realtime
8:44AM 0 Need help with script from
8:40AM 1 recieve mutiple inbound calls
8:18AM 10 Identifying Frame Slips from PRI debug
7:46AM 0 Broken MOH
7:31AM 1 Polycom 500 IP and problems with show hints
7:05AM 0 port vs bindport
6:50AM 0 show queue
6:47AM 0 Using mgcp get/generate "message waiting indication"
6:45AM 1 Re: Re: RFC 3262 PRACK (Olle E. Johansson)
6:33AM 4 Asterisk Call Forwarding
6:15AM 2 php agi problem (perhaps problem..)
6:11AM 12 Asterisk server to provide virtuals IPBX
6:04AM 1 Postgres
5:47AM 15 [offtopic] Asterisk <-IP-> Siemens HiPath 4000
5:28AM 2 Instalar Ubuntu
5:09AM 5 Tracing a crash with CAPI calls
4:53AM 1 turn off message "Silence suppression ..." on Asterisk console
4:44AM 0 MP3 problems: MP3Player and Musiconhold
4:36AM 0 Problem with CDR
1:10AM 0 WG: Goto after Dial PRoblem
Tuesday December 20 2005
11:05PM 0 Unicall Problem with fax
9:15PM 0 Digium E1 Card Modprobe problems
7:28PM 2 Asterisk based IVR/VoiceMail Server for a Unified Messaging suite
5:22PM 26 Latest Source
5:01PM 5 Unicall E1 Error in Mexico
4:43PM 8 Got SUBSCRIBE for extensions without hint
4:36PM 2 Meetme and ztdummy
3:57PM 1 SPANDSP & TX RX Fax paid support wanted.
3:35PM 4 Help Debugging Dropped Call Audio
3:04PM 54 SIP Subscriptions
2:30PM 2 RFC 3262 PRACK
2:22PM 9 IVR Capacity
2:01PM 3 3 Phone Call Qualtiy Issues
1:17PM 2 Analog terminals and modems? does it work
12:33PM 4 Linking existing channels through Manager interface. Is it possible?
12:17PM 7 Asterisk & FXO & Panasonic PBX
11:28AM 5 IVR and db
11:16AM 0 meet me room status
11:03AM 0 SunFire X4100
10:58AM 3 G729 and Cisco IOS 12.4
10:12AM 11 1.2.1 Queues
10:00AM 0 MOH engaged while holding for ANOTHER party (1.2.1)
9:59AM 0 SIP/IAX to PSTN
7:04AM 0 Goto after Dial PRoblem
7:03AM 5 Rolling dialplan... best practice?
6:43AM 0 TE205P E1 PRI card and other problems
6:39AM 1 482 Loop Detected when transferring calls back to Asterisk
6:05AM 1 How to get received digits from console channel
6:05AM 6 Soporte
5:26AM 2 inbound routing with amp and TDM400
4:38AM 1 messages of Mobile Operator
Monday December 19 2005
11:58PM 1 Fast AGi Variables
11:38PM 1 Digium TDM2400 Series Server Compatability
11:29PM 1 To write Sphinx Interface in EAGI or app_xxx.c?
9:06PM 2 SIP - SIP bridge dropping calls?
8:34PM 6 Asterisk with Uniden uip200
5:45PM 0 memory not being released
5:33PM 0 Polycom retry interval and DNS SRV failover
5:10PM 7 Handling SIP clients behind NAT on a semi-dynamic IP
4:34PM 2 Handytone 486 Outbound problem
3:59PM 5 ALERT_INFO Not Working Upon Upgrade to 1.2.1
3:13PM 0 Asterisk & NAT behaviour
2:33PM 0 queues and redirection.
2:19PM 13 IBM eServers?
2:07PM 4 Mulitple voicemail on mulitple phones
1:31PM 2 Simulate incoming line
1:23PM 1 VoIP/VPN providers in Switzerland
12:57PM 1 Originate a call to a Queue?
11:25AM 1 Variable Help
11:16AM 0 problem with automatic attender calls
10:44AM 1 Re: Asterisk-Users Digest, Vol 17, Issue 111
9:55AM 2 iax2 on a server behind a linux based stateful firewall
9:15AM 0 MixMonitor error exit
9:08AM 4 Can't pass variables using Originate in PHPAGI 2.14
7:40AM 1 Problem using Queue and Sip Soft
7:29AM 7 Can't call out on ZAP channel - need help
6:47AM 0 Callware VoiceOne released: a new, easy web GUI
5:34AM 0 ACD with polycom ip phones (resent)
4:45AM 3 DTMFMODE with grandstream
4:20AM 0 Looking for a spare LCD display SNOM 220
3:33AM 26 Asterisk <-> Skype anywhere/anyhow?
Sunday December 18 2005
9:21PM 1 Asterisk Voice mail-reg
7:51PM 0 SIP Remote Call Control
7:48PM 1 [Fwd: Odd problem with Encore 201-SA (r2 converter) with asterisk]
7:47PM 0 Odd problem with Encore 201-SA (r2 converter) with asterisk
6:47PM 1 Anybody having trouble terminating calls at Voxee? <eom>
5:20PM 0 Extension processing misunderstanding
3:28PM 0 iaxmodem through zaphfc
2:15PM 0 FOP led Colors
12:05PM 3 Is it me, or is 1.2.1 slower than 1.0.9?
10:39AM 30 ACD with polycom ip phones
10:24AM 3 New voicemail alert options for Cisco 7960 SIP phones
10:19AM 1 SIP Watchdog
8:27AM 0 Asterisk <-> Avaya system
7:35AM 9 Is this possible in Asterisk?
6:25AM 1 Too high volume on Music on Hold
5:23AM 2 asterisk 1.2.1 and mixmonitor problem
5:22AM 0 Can't pickup call when dialing *8 extension (resent)
Saturday December 17 2005
10:35PM 5 ztdummy problem !!!
7:59PM 1 aastra.cfg & mac.cfg examples Firmware version 1.3
7:11PM 8 SIP and echo cancel
6:28PM 8 Toll Free Providers
4:42PM 12 Teliax billing question
3:48PM 2 Can't pickup call when dialing *8 extension
1:43PM 0 placing a call in one or several call groups
1:05PM 2 Grandstream GXP-2000 Auto Answer
11:41AM 1 Linksys PAP2 and Asterisk
11:14AM 0 Terminating calls externally via SER
9:21AM 0 i can't register to my sip service(but x-lite can)
9:20AM 3 Alarm panel through ATA
9:10AM 7 Strange problem with sjphone and 1.2.1
9:05AM 0 Cisco 79xx display as busy-lamp field
8:44AM 0 Re: [Astguiclient-users] [PATCH][RFC] Quiet debugging messages in Net::MySQL Perl module
8:42AM 0 multiple ALSA devices and Asterisk
7:57AM 0 Cid_rewrite update
7:06AM 0 A2billing Trunk
6:41AM 0 asterisk 1.2.1 realtime mysql.4.1.xx report errors
5:19AM 0 Key R (Flash) and Asterisk
4:33AM 0 I need syntax on applicationmap in features.conf
2:00AM 5 Can Asterisk replace Cisco 5350?
12:24AM 0 asterisk and h323 problems
Friday December 16 2005
8:42PM 36 What is the best Dell Machine for Asterisk?
7:13PM 2 Redency of Asterisk
5:08PM 6 Codecs.
5:02PM 8 Asterisk Redundancy
4:58PM 24 TDM01B answering issue
1:24PM 0 Asterisk-1.2.1 incomplete DID number on incoming T1 line
1:06PM 1 .call files on PRI not waiting for answer in de sired context <--ResponseTimeout the best answer?
12:54PM 0 .call files on PRI not waiting for answer in desired context
12:35PM 0 Merlin Legend mode codes
12:35PM 1 .call files on PRI not waiting for answer in de sired context
11:43AM 0 Digium TE205 Card
11:39AM 0 Amtelco Infinity
11:29AM 4 CID lookup from an Exchange Public folder
9:33AM 0 Having trouble calling out from Zap channel
9:19AM 2 1.2.0 queue.conf exit context
8:04AM 3 Mediatrix 1204 help please.
7:20AM 0 Connecting Meridian M8x24-DS to Asterisk - NoDTMFtones
6:41AM 2 Experience sharing on Planet VIP-450 + Asterisk
6:37AM 1 incoming dtmf handling by ATA devices ?
6:04AM 3 Central Registration mechanism
5:29AM 2 ztdummy / timer problem with kernel 2.6.14
4:28AM 2 Romania/Rumania setup
3:52AM 22 HW Echo Cancellers
3:48AM 16 Does hardware like this exist...?
2:59AM 2 Configuration of two Asterisk server
2:35AM 1 Meetme option Ax
2:19AM 0 asterisk 1.2 mysql cdr garbage
Thursday December 15 2005
11:29PM 1 Raltime database schemas
10:57PM 1 CallerID/Extension Matching with RealtimeExtensions
10:50PM 1 CallerID/Extension Matching with Realtime Extensions
8:51PM 2 Alternative source for Asterisk-IM
8:25PM 0 Echo & TDM11B
7:31PM 4 SIP Trunk please help
7:23PM 3 Connecting Meridian M8x24-DS to Asterisk - No DTMFtones
7:06PM 4 Weird IAX trunking/7960/ILBC quality issue
5:04PM 0 Connecting Meridian M8x24-DS to Asterisk - No DTMF tones
3:50PM 5 Will ooh323 ever move from addons?
3:44PM 3 looking for hardphone configuration info
1:57PM 0 Script to detect corrupted faxes from SpanDSP
1:05PM 3 Echo Canceller usage
12:27PM 4 Disposition Failed still happening
11:36AM 0 Asterisk Realtime connection failed
11:28AM 0 Originating calls to a channel groups
11:26AM 9 Google Analytics and
11:02AM 1 E1 Echo (was: Small explanation of txgain rx gain statement please)
10:55AM 3 Voipsupply - my experience
10:44AM 0 Handyton 486 Outbound problem
10:38AM 8 Shutting down Asterisk when not in RTP Stream
10:12AM 1 Can you time limit access to a trunk?
10:10AM 3 Outbound Routing
9:04AM 11 E1 Echo (was: Small explanation of txgain rxgain statement please)
8:19AM 1 astcc issue
7:50AM 2 function cut()
7:14AM 2 ChanIsAvail()
7:13AM 0 Sip configuration for make and receive calls
6:33AM 1 voicemail cutting out
6:26AM 12 hint on Zap channels
6:09AM 2 chan-capi avm b1 and capi.conf problems
5:58AM 0 RE: how to forward call within office
5:45AM 2 screen safe_asterisk does'nt spawn asterisk
5:38AM 1 Firewall Ports forward
5:31AM 0 EXITWITHQUEUE on queue_log
5:22AM 1 Small explanation of txgain rxgain statement please
5:18AM 0 oh323 : which versions recommended for asterisk 1.2?
5:16AM 1 RE: how to forward call within office
4:55AM 0 RE: Asterisk-Users Digest, Vol 17, Issue 89
4:50AM 4 How to change the Dial command H option to ## ?
4:30AM 0 again - show queue info
4:14AM 0 Help with mgcp
4:02AM 0 How to tell if Authenticate failed without using j in 1.2
4:00AM 1 background music...
3:50AM 0 QueueMetrics 1.0 rc 1 out today
2:42AM 8 AoC (Advice of Charge)
1:14AM 0 Anyone with VIP-450
Wednesday December 14 2005
10:21PM 3 2 PBX linked via internet
10:12PM 2 I don't want ilbc, i just want G.711
9:44PM 4 Starting RTP with Dial and MusicOnHold
9:33PM 3 Asterisk & STUN
9:29PM 0 why sql error in asterisk 1.2.1 with realtime with mysql 4.1.x
9:22PM 0 WishList - Devices that are (probably) not available yet [OFF-TOPIC]
5:24PM 0 Problem with bridging SIP to OH323 and SIP to SIP: Bridge stops bridging
3:57PM 5 How to disable sip Native bridge
3:56PM 2 "Context Picker" for interception and redirection
1:55PM 0 Prevent Logging "reload" verbose
12:43PM 1 1.2.1 Compile Error
12:36PM 3 ChanIsAvail() and SIP
11:33AM 3 HOWOT transfer call from mobile back to extension?
10:56AM 2 asterisk + H323 + 723
10:20AM 8 traffic shaping
10:20AM 1 Background() followed by Read - something wrong?
10:11AM 2 Headset Phones?
10:08AM 1 Cisco 7940 Time Source
9:44AM 9 OT: Linux on treo 650
9:11AM 1 appradius
9:02AM 26 hardware echo cancellation for TDM card
8:51AM 0 Gateway crashes when transferring to external lines
8:49AM 4 Best way to automatically stop and start Asterisk nightly
8:44AM 1 Blind transferred user does not hear phone ring while waiting for phone to be picked up.
7:50AM 0 Need help with Sipura 3000
7:50AM 0 Video calls (MS Messenger, Tandberg)
7:42AM 2 Need help with sipura
7:37AM 0 MGCP Unable to find key
7:19AM 0 '#' (fast foward) and '*' (Rewind) not working in VoicemailMain
5:58AM 0 Exceptionally long queue in SIP Channel
5:42AM 1 subscription
5:30AM 3 Unable to find key
4:58AM 0 quadbri, isnd, netherlands: callerid not working
4:48AM 0 Help:asterisk 1.2.1 release compile
4:35AM 0 [help] problem in astersik
3:57AM 1 [Fwd: Re: Re: [helpp] Problem in astersik]
3:31AM 2 Dial multiple destinations
3:26AM 5 voicemail boxes
3:09AM 5 Wildcard TDM2400P: comments
2:19AM 2 Join when empty problem, in queue
1:46AM 3 capi.conf - AVM C4 P2P or P2MP
1:45AM 0 PSTN gateway & Asterisk -> Virtual Switchboard???
12:59AM 0 SIP peer vs. user-- how is the USER ever selected?
12:41AM 1 send SMS via own SMS Service
12:22AM 0 RealTime and automatic extension registration.
Tuesday December 13 2005
11:10PM 2 fxs woes...
10:59PM 4 SIP Subscription Storage Location
10:28PM 0 Cisco 7960/ATA/MultiTech MVP200 FXS/FXO to H323 gateways on ebay
8:07PM 0 Pattern Matching, speed and memory....
6:24PM 2 VizuFon CIP-4500 with Asterisk through SIP
5:53PM 0 Asterisk 1.2 SIP register problems
4:59PM 0 nice -n 19 called from shell script through Syst em() gives "Permission Denied"
3:01PM 0 Entering Digits
2:57PM 2 Asterisk 1.2.1
2:51PM 0 empty line sip_notify.conf
2:33PM 2 December VON Magazine
1:39PM 2 pb !Astrisk 1.2 Card TE411P
1:18PM 2 IAX error message
12:56PM 0 Meetme Conference sound problems
12:41PM 0 Question on having asterisk put calls into a meetme.
12:12PM 5 format_mp3 & uninstalling mpg123
11:43AM 0 queues & music on hold
11:21AM 1 cdr_addon_mysql can't find
11:12AM 0 CID name & number contain unwanted quotes in CDR
10:59AM 1 Very high memory consumption when high number of calls are processed
10:47AM 1 mISDN & chan_misdn on Fedora Core 4 - problems
10:33AM 2 Tellabs manuals
10:28AM 0 Asterisk Feature Request: app_bridgeme
9:55AM 8 Bonded ethernet ports and *
9:44AM 1 mISDN Caller ID problem
8:49AM 2 extension seen as busy when it is not
8:43AM 0 FXOTUNE Error on channel 2
8:39AM 0 408 Request Timeout vs. 403 Forbidden
8:20AM 13 Partial PRI pass thru?
8:05AM 1 RE: 1.2.1 has broken voicemail realtime
7:39AM 1 Testing with
7:20AM 2 calls forwarded to busy agent
6:34AM 1 1.2.1 has broken voicemail realtime switching
5:57AM 0 NAT/Qualify/RTP bug
5:47AM 3 IAX2 show channels show Channel (NONE)
5:34AM 2 g729 translation to zap (ISDN) doesn´t work
4:18AM 0 OOH323 -> IAX2 : no sound
4:08AM 0 queue_log Vs show queue abandon calls discrepancy
3:54AM 0 Call Disconnecting
3:05AM 6 Info request from Sangoma users
2:10AM 11 chan_capi AVM C2
1:35AM 7 SPA-3000: Dual Registrations?
12:49AM 1 AGI GET Variable Problem
12:33AM 1 Setting Language
Monday December 12 2005
11:47PM 1 ENUM For Presence
11:26PM 5 Patch zaptel.init to support debian
10:47PM 0 NAT Issues?
10:31PM 0 subscribing
9:57PM 0 Patch to zaptel Makefile
8:44PM 2 X100p echo guide
7:13PM 8 No outgoing sound...sometimes
6:54PM 0 Invoking blind SIP REFER transfer asterisk
6:35PM 13 Skips and Pops in Call Recordings
6:16PM 0 (Montreal Users) Call for technical presentations
5:02PM 1 new asterisk 1.2 setup doesn't react when I press any numbers
4:42PM 0 Interesting Article on Echo Cancellation
4:39PM 0 analog FXS card for dialup modem and fax
4:30PM 0 E1 PRI cause codes
4:29PM 0 NOTIFY Messages
4:24PM 2 Softphone with Hint support?
3:45PM 0 Is clearglobalvars=no really working in asterisk 1.2.1?
2:31PM 1 busypattern tones?
2:17PM 10 asterisk in real estate developments
2:02PM 5 Turning off hardware echo can on TE411P
12:41PM 0 Call Monitoring / Ext to Ext with Sipura-841
12:22PM 3 trying to get SIP to work remotly.
11:55AM 1 Dlink DI-102 QOS Thingy?
11:41AM 3 Make list of incoming and outgoing calls
11:40AM 0 capi incoming call timeout
11:33AM 1 Need advice on BRI
11:30AM 0 Zultys ZIP2 + asterisk + DTMF on other end? (i.e. ivrs, autoattendants, etc)
10:41AM 1 executing a reload under stress in Asterisk
10:36AM 8 Cisco 7940 Reboot
10:33AM 1 uniqueid with multiple asterisk hosts
10:04AM 0 Outgoing data call
8:52AM 0 Unable to prevent SIP to SIP calls from removing Asterisk from Media path
8:42AM 0 Dial Cmd Outbound CLID Failure (* 1.2.1)
7:47AM 6 How do I remove the temp greeting?!?!
7:16AM 3 ASTCC/ASTCC anything wrong with that?
7:07AM 1 Zap Transfer
7:01AM 0 ChefSec function
6:36AM 0 persistentagents, persistentmembers
5:00AM 4 Digium PCI-X timeline
4:57AM 1 Production Upgrades
3:19AM 5 PRI E1 - HDLC Bad FCS / HDLC Abort errors
2:45AM 2 Problem with Speex
2:40AM 1 click to dial applications
2:35AM 0 Variables naming, may be a BUG??
2:27AM 3 Cisco 7941 difference
2:14AM 1 "Got clone lock for masquerade" crash
2:11AM 7 Problems with current chan-capi-cm
1:56AM 10 [helpp] Problem in astersik
1:17AM 0 asterisk1.2.1+realtimedb+voicemail+contexts
12:59AM 2 CallerID Transfer
12:15AM 4 Long and variable echo
Sunday December 11 2005
11:32PM 2 Mechanisms for Implementing a Common ContactDatabase
11:03PM 2 New Product ID.
10:53PM 1 Mechanisms for Implementing a Common Contact Database
7:56PM 0 Originate action script causes Asterisk to hang
7:20PM 1 Cleaning up CLID on incoming PRI lines
3:23PM 12 Small / embedded system recommendations
3:22PM 8 Realtime Subscribecontext
1:37PM 25 Regexten
12:39PM 15 Attack dialing
11:52AM 0 [Asterisk-Dev] C++ AGI debuggin
11:50AM 10 Asterisk Dynamic DNS
11:47AM 13 SIP Qualify
8:59AM 6 Dialing analog extensions from SIP?
8:34AM 9 SRV Lookups *ARRGH!*
7:18AM 3 outgoing calls that last an unreasonably long time
Saturday December 10 2005
9:42PM 3 bristuff use without BRI/PRI
6:58PM 2 Setting Request URI
4:31PM 1 extensions and regular expressions ( probablyan easy question )
3:06PM 1 Using SIP_HEADER() Function correctly
2:24PM 0 RE: chan_grab.c for version 1.0X
1:42PM 0 Echo on incoming sip provider- asterisk - sip phone / ata
1:40PM 10 extensions and regular expressions ( probably an easy question )
11:09AM 0 Good Dialing Macros
10:56AM 0 Problems with zaptel channels not properly beinganswered...
10:52AM 1 agi variables list
10:46AM 1 Problems with zaptel channels not properly being answered...
10:22AM 3 Re: Asterisk-Users Digest, Vol 17, Issue 61
9:08AM 1 Re: Asterisk-Users Digest, Vol 17, Issue 61
6:10AM 1 Asterisk not Replying on Port Specified in the VIA header
5:48AM 1 Possible bug in record?
5:44AM 8 Email to voice?
5:34AM 4 Channel 0/1, span 1 got hangup request
4:56AM 0 Sending a recorded message to voicemail
4:13AM 0 Q regarding dialling multiple phones (fwd)
3:49AM 0 Predefined Channel Variables
1:19AM 10 Via Epia
Friday December 9 2005
10:23PM 1 Dial Command Doesn't return Correctly! (Bug?)
9:23PM 2 What's the best opensource web interface for customer portal
8:55PM 0 Asterisk without RTP streams vs. SER in statefull mode
8:17PM 4 Synthesized Voice for Asterisk
6:39PM 0 Major Advantages of Asterisk vs Nortel PBX Systems
6:36PM 0 service
5:34PM 9 Asterisk, Small Business, and Teliax
4:31PM 5 Daily Reboot Script for Asterisk Question
4:13PM 0 Anyone else experiencing Voicepulse outage?
4:04PM 3 IAX Jitterbuffer and trunking
3:45PM 0 Unable to receive DTMF inbound
3:41PM 0 Stop Dial but no Hangup
3:30PM 0 NY Clec Listing
2:55PM 1 Wait for X rings before answering?
2:52PM 0 Asterisk Consulant - Call Centre Proposal
2:46PM 1 callerid international-format
2:03PM 2 Call Routing from GnuGK to Asterisk
1:31PM 4 Adit 600 and Centrex
1:05PM 0 number of users in a meetme conference
1:04PM 2 Asterisk 1.2 chan_misdn modules.conf
12:40PM 3 IAX2 Status monitoring
12:34PM 1 T.38/PRI Gateway Recommendations
12:24PM 0 Unknown RTP codec 96 received
11:19AM 0 Hardware based echo can from NMS Communications
10:23AM 4 Re: Asterisk-Users Digest, Vol 17, Issue 56
10:13AM 0 testers needed for channel.c jitter buffer (better known as SIP jitter buffer)
9:53AM 0 t38 support in latest asterisk release
9:36AM 2 Asterisk vs Nortel, Northstar and Mitel
9:10AM 0 Asteriskguru Queue Statistics version 0.7 released
8:48AM 5 Echo PSTN Asterisk@Home 2.0 Digium TDM11B & DSL
8:16AM 7 Teliax experiences
7:59AM 4 Phone Information
7:07AM 1 Low Layer Compatibility (LLC) not forwarded?
7:02AM 27 Change time when * is running
5:22AM 0 connection between asterisk and cisco
3:40AM 2 Queue routing - calls return to agent which previously handled call
3:26AM 0 Hangup after dialing
2:59AM 3 /dev/zap/ctl or /dev/zapctl cause ztdummy in init.d failed
2:33AM 0 Aastra firmware 1.3.x. Solution to Far-End sound level issue
2:05AM 0 Re: Is Polycom 500CS with P# 2201.11500.001 SIP capable?
2:04AM 0 PRI billing signalization
1:09AM 1 DID Providers
Thursday December 8 2005
10:36PM 7 Porting a phone number to a voip provider
9:51PM 2 Can Asterisk accept and relay calls
9:03PM 19 Asterisk Dial Failover
8:01PM 1 Core dumps since 1.2.0
7:55PM 2 does asterisk-oh323-0.6.7support asterisk1.2
7:14PM 1 cant start the conference bride
7:11PM 8 ztdummy on FC4
6:27PM 4 Why Won't Asterisk REINVITE?
5:40PM 2 Asterisk and Adtran TA 750 Channel Bank -- odd behavior (help!)
4:52PM 1 Re: Meetme and Sipura SPA-941 -badjitter/distortion
4:45PM 2 OOH323 towards cisco gateway(2691)callsetupfailsat q931: Mandatory information element ismissing (96)
4:04PM 1 Where to find Digium products in Mexico?
2:54PM 1 OOH323 towards cisco gateway (2691)callsetupfails at q931: Mandatory information element is missing (96)
1:20PM 2 Call screening script
12:39PM 0 question about priorities?
12:18PM 0 compile problem
12:11PM 0 Compile modules app_rxfax.c app_txfax.c for asterisk 1.2.1
11:53AM 0 AstLinux 0.3.0 Released
11:16AM 0 Re: [Asterisk-biz] Help with learning Asterisk for the real world..
11:15AM 1 How do I set up extensions.conf to dial out on analog telephone line?
11:08AM 2 Leave one voicemail for multiple recipients
10:51AM 0 Lucent MAX TNT - how do I route a DID to my sip trunk
10:43AM 8 Asterisk Bounty Pool
10:13AM 1 OOH323 towards cisco gateway (2691) call setupfails at q931: Mandatory information element is missing (96)
9:55AM 3 Realtime Replication of a Single File
9:40AM 1 SIP.conf Technical Documentation - Help
9:27AM 6 Meetme and Sipura SPA-941 - bad jitter/distortion
9:23AM 0 Voicemail context
9:23AM 1 New GSM 1-8 ports Gateway / Terminal for sale(with SMS Feature and Many more)
9:21AM 2 Maximum Calls handled
9:20AM 0 OOH323 towards cisco gateway (2691) call setup fails at q931: Mandatory information element is missing (96)
9:17AM 0 Zombie AGI processes in FC2 / 1.2 Beta 1 under l oad
8:56AM 3 Exit Voicemail
8:41AM 3 New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more)
8:24AM 0 Polycom SIP part numbers
8:21AM 2 Octo Bri card together te405p and bristuff
7:54AM 0 Hardware combination and type of asterisk configuration
7:26AM 1 Dynamic IAX2 hosting in the UK
6:43AM 12 Asterisk Call Recording and SIP canreinvite
6:15AM 3 Call simulators
5:21AM 1 Forwarding only at certain times
4:27AM 2 about g729
4:15AM 0 about * and CM/CME
4:14AM 1 multiple line registrations on attendant console
4:09AM 3 No application 'MeetMe' for extension
3:48AM 0 IAX stress test
3:35AM 1 Change Inbound CALL ID "Asterisk"
3:11AM 0 CDR manipulation in macros
3:01AM 0 Wrong caller id num on Swissvoice IP10S
2:34AM 0 SVN Revision 7230
2:30AM 0 MYSQL cmd with Asterisk Realtime
2:00AM 1 Asterisk as sipclient
1:55AM 0 Integration of external (ZAP) agents into queue
12:38AM 1 Asterisk as a gatekeeper
12:19AM 2 Recording Volume on Zap Channel
Wednesday December 7 2005
11:53PM 2 RE:how to listen voicemail messages
9:31PM 0 ZT_CHANCONFIG failed on channel 1: No such device or address
8:46PM 0 local not ring,,,,
7:56PM 0 London DID 30 Cents a Number available - 60 Channels, ULAW
7:45PM 6 Aastra 9133i Configurations - are the file names to be lower case or upper case or does it matter?
7:32PM 2 HOW TO: CDR Customer IP address where call came in from
5:47PM 7 Asterisk Hardware recomendation
4:33PM 6 A company that sells Toll Free Number in USA
3:37PM 0 Asterisk 1.2.1 and queue_log
2:55PM 0 AstManProxy Segmentation Faults
2:49PM 5 Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing
2:18PM 4 IAX2: Don't know any of 0xf800 formats
1:43PM 24 Sip behind the NAT
1:33PM 3 app_queue on 1.2 ?
1:20PM 4 Unable to compile zaptel / ztdummy
12:38PM 4 Door Phones
12:23PM 3 Can Asterisk act as a media gateway?
12:02PM 14 Recording a call
11:42AM 0 chan_bluetooth Audio Sensitivity
11:21AM 0 Zaphfc as a timing source?
10:48AM 0 VoIP US - Toll Free Origination / Termination Providers
10:38AM 3 Asterisk 1.2.1 released
10:10AM 4 IConnecthere dial out problems
10:07AM 2 Lucent TNT / Asterisk Help
9:34AM 3 Polycom 501 remapping keys
9:04AM 0 Wanted Japan DID
8:41AM 3 E1/T1 configurations
8:28AM 0 how to detect hangup
8:22AM 0 Config Attended Transfer
6:58AM 3 asterisk with EWSD v16
6:45AM 0 Sending credit card thorugh network/sipura 3000
5:57AM 0 VoIP to GSM?
5:41AM 3 Busy recognition
5:32AM 10 UK ISDN2e with DDI?
5:10AM 0 Goldstar GDK 186 voicemail
4:00AM 3 HDLC link unstable, yellow alarm on
3:58AM 2 Asterisk modules description
3:18AM 1 Feature implemention
2:31AM 0 res_perl error when loading asterisk
2:11AM 2 Ringtone when dialing
12:36AM 0 audio lost on incoming calls
Tuesday December 6 2005
11:42PM 3 Connecting asterisk over consumer wifi network
10:51PM 0 dialplan activated Toll restriction
10:31PM 2 Win up to $2000 for AsteriskEnterpriseReferences!
10:01PM 1 Win up to $2000 for Asterisk EnterpriseReferences!
8:41PM 0 Snom 360 and 320 AutoAnswer
8:17PM 5 Hint Priority for Polycom Phones
8:09PM 2 Asterisk as a Softswitch
7:30PM 4 Win up to $2000 for Asterisk Enterprise References!
6:24PM 1 Flash operation on a call on a ZAP interface...
5:35PM 1 DSP-based echo cancellation (T1).
5:14PM 0 Re: VoipBuster / Finarea
3:59PM 11 Nortel Meridian Option81C to TE405P
3:33PM 2 Snom monitoring of extensions not working
3:24PM 0 ATA Registration problems
2:05PM 0 Outgoing fax detection
1:11PM 6 Packeteer ? Edgemark ? How to not re-cable ?
12:43PM 0 How to setup Connected number
12:35PM 1 Re: Asterisk-Users Digest, Vol 17, Issue 7
12:19PM 1 Toll-free number on a PRI
11:46AM 2 Re: VoipBuster / Finarea
11:10AM 5 VoIPJet issue == No one is available to answer at this time
11:03AM 2 snom 320 'retrieve' button
10:35AM 2 can * translate DTMF from rfc2833 to inband?
10:17AM 4 Per Extension Password for Outgoing Routing
10:08AM 0 E1 and hardware Test.
9:40AM 17 Complicated Dialing plan routing
8:54AM 1 Credit Card Terminals
8:47AM 4 Odd DTMF issue over PRI
8:12AM 0 Dial application "g" option
8:00AM 0 What would prevent logs from being recreated if they are deleted?
7:33AM 7 Call Forwarding with Account Code.. can it be done?
6:27AM 0 PRI and Dialed Number
5:33AM 0 Primary D-Channel on span 1 up
3:38AM 0 Problem with a second incoming call on a BRIZapChannel
2:59AM 0 CallParking and chan_capi-cm-0.6
2:48AM 1 RE:Is it possible to install ZAPTEL after installation of Asterisk
2:42AM 1 Problem with a second incoming call on a BRI ZapChannel
2:05AM 0 Asterisk and Video
1:34AM 2 zaptel : unresolved symbol zt_unregister ...
1:30AM 18 SNOM Shared line DevState
12:37AM 4 OH323 user configuration
Monday December 5 2005
11:31PM 1 How to restric user to call only specified country
10:58PM 7 Echo cancellation over satellite link
10:48PM 2 EAGI Audio Capture
10:03PM 3 Realtime SIP Lookups
9:50PM 0 2 leg bridged call not hanging up until both legs hangup
9:26PM 0 Please help in writing AGI script
6:40PM 6 Asterisk on PPC & chan_capi issue
6:37PM 0 sipura "Vertical Service Activation Codes"
6:04PM 3 uip200 phone not work with 1.2
5:44PM 2 logging performance, important impact?
5:01PM 1 ADIT 600 T1 with DNIS digits problem
4:38PM 7 Messages button on a Polycom 501
3:17PM 5 Best Switch for VOIP Applications
3:00PM 5 PRI indications.
2:08PM 2 Preventing incoming calls from ringing SIP lines
2:07PM 1 Linksys SPA-941 DTMF failure with asterisk v.1.2
1:21PM 0 transfers from Polycom 501 involving Sipura 300 and asterisk 1.2
12:28PM 2 Anyone know anything about the new Linksys One product - does it use Asterisk?
12:26PM 0 video phones
11:30AM 0 Panasonic DBS DISA
11:21AM 0 Asterisk Queues Tutorial updated...
10:54AM 3 Include a variable from another file in configfiles
10:25AM 0 asterisk won't answer malformed caller id
10:09AM 7 Looking for advice on cell carrier's default "Un avaliable" message
8:45AM 0 kernel lockup with Fedora Core 4.0 2.6.14-1.1637
8:36AM 2 Restore logging functionality...
7:36AM 10 h323 vs oh323
7:35AM 4 Transfer/take call to/from other phone
6:50AM 0 Problem with a second incoming call on a BRI Zap Channel
5:10AM 1 A@H with a2Billing
3:56AM 2 Include a variable from another file in config files
3:51AM 0 oh323 installation
3:35AM 3 SIP INVITE with no 'Contact' field and RealTime support.
3:31AM 0 Calls to DISA over ISDN PRI don't get CONNECT ACKNOLEDGE
3:19AM 0 (warning) iaxy.bin fails checksum
2:28AM 2 Chat Lines/ Party Line Solutions for Asterisk
2:28AM 0 asterisk.h
1:04AM 0 VegaStream 400
1:02AM 0 Asterisk 1.2.0 - TE210P - "...Control Frame 15..."
12:42AM 6 Error when compiling asterisk
12:37AM 0 voicexml vendors
12:21AM 1 [Amportal-users] AMP queues, AddQueueMember and 'Wrapup Time'
12:18AM 2 Re: sound problem in X-Lite phone with asterisk server
Sunday December 4 2005
11:26PM 5 Sipura 3000 Call waiting on the PSTN line
10:11PM 10 Connecting 2 Asterisk using SIP
3:51PM 1 UK DID 0208 £1 per month
3:44PM 3 DISA function
2:23PM 0 Sending data over ZAPHFC D-channel?
2:19PM 7 Getting started with Asterisk and Aastra 9133i
1:16PM 0 RE: how to remove asterisk 1.2 from Red Hat 9
1:00PM 0 Why does musiconhold.conf changes require a reboot?
11:39AM 6 ISDN 2e Cards
8:48AM 1 re: Help required on asterisk
7:01AM 1 replace hard-phone if soft-phone is online
2:42AM 1 UK Patches for Asterisk 1.2
2:25AM 1 Sipura 3000 Disconnect Singnel
Saturday December 3 2005
11:41PM 3 Re: Asterisk-Users Digest, Vol 17, Issue 18
8:52PM 1 Converted mp3 files to raw for musiconhold and still does not work...
3:48PM 1 Call queues, agents with DND status set.
1:23PM 0 Order of ports on rear of Sangoma card and pictures in a mini-itx chassis.
12:06PM 6 Can I escape queue with a '*'?
8:42AM 6 Asterisk 1.2 and weird ZAP interface behaviour
8:23AM 0 Fwd: Queue Statistics
7:13AM 0 No CID Info an TE405P with zaptel 1.2.0
6:50AM 0 Rates for Asian countries
2:21AM 1 IAX Conf Realtime?
2:18AM 0 Voip providers with trunked iax2
2:14AM 11 Iax2 connection failed
2:09AM 8 Can't compile chan_zap.c on fresh cvs 1.20 checkout
Friday December 2 2005
11:15PM 0 Softphone setup
11:06PM 1 Fw: Re: Re: Zaptel errors on Debian
10:12PM 0 Need Good UAE calling rates
8:54PM 3 Broadband VoIP Startup with Asterisk
8:47PM 1 Anyone have experience with
8:23PM 0 Suggestions on Echo Problems
6:24PM 2 chan_blutooth
5:56PM 7 Linksys SPA-841 Missing Calls
5:51PM 0 Problem, to register an ata on an asterisk
4:48PM 0 Testing Sangoma A2022-SO card with Asterisk 1.2
4:30PM 0 Prodding channel failed
4:28PM 1 echo canceling algorithm
3:06PM 0 USB/Bluetooth Speakerphones
2:43PM 1 Re: Asterisk 1.2 problems (
2:20PM 0 What kind of extension numbers can be used in the exit context of a queue?
2:07PM 3 Context confict question??
1:54PM 7 Fax Service
1:47PM 1 callback script
1:43PM 0 Subject: Re: Linksys SPA-941 Admin Guide
1:17PM 10 ZapHFC cards not maintaining sync?!
12:53PM 1 Can I get to a menu system while in a queue??
12:22PM 5 Asterisk 1.2 problems
12:00PM 4 Failover Registration
11:22AM 4 Kernel upgrade causes ztdummy to refuse to run
11:18AM 1 DIAXY to DIAXY problems
10:50AM 2 "hint" priority in AEL?
10:44AM 0 Help with a Company or Site for a DEMO. AYUDA con una empresa para una DEMO
10:40AM 0 v1.2 and cdr badly written
10:03AM 10 Sangoma & Asterisk at home
9:46AM 0 Re: Asterisk-Users Digest, Vol 17, Issue 11
9:38AM 2 Meetme option 'b'
9:30AM 2 dial-out and variable inheritance problems
9:30AM 0 Originate calls but can't receive them on a SIP trunk
8:46AM 2 sip invite timeouts
7:53AM 2 Queue and agent transfer
7:17AM 3 what is your echo solution
7:12AM 0 ISDN card Sirrix.PCI4S0
7:12AM 0 DTMF is choppy on the receive
7:04AM 0 Polycom DTMF after connection not working
5:03AM 1 change priority by time
4:59AM 1 equal priority trunks for balancing
4:54AM 1 RE:how to solve error : cannot find extension context 'from-sip'
3:55AM 0 Asterisk-users
3:33AM 0 BT - DSS
3:00AM 1 Limiting DID calls
2:26AM 1 voice problems under 8 concurent calles
2:22AM 3 Asterisk Users Newsgroup
1:10AM 0 DTMF on Planet VIP153
12:47AM 1 (no subject)
Thursday December 1 2005
9:43PM 11 IAX trunking frequency parameter works only on initiator side
8:27PM 1 Hint: how to include dialplan files from remotesystems
7:51PM 2 Hint: how to include dialplan files from remote systems
6:32PM 4 Running asterisk within screen
6:04PM 1 call center dial plan
4:05PM 0 config Polycom with both SIP provider and Asterisk
3:58PM 0 Transfer problem...
3:51PM 0 Asterisk with ooh323 registers with a cisco gatekeeper but disconnects after 300
3:51PM 1 Very Weird problem with MeetMe, SIP, Zap and the combo of the three
3:36PM 2 Can Asterisk do This?
3:32PM 2 Asterisk as a gateway to Index PBX
3:28PM 0 ITSP in a box demo updated
3:14PM 6 Two Phones - Same extension?
3:09PM 4 Linksys SPA-941 Admin Guide
3:02PM 0 Shoutcast For MOH with Asterisk 1.2
2:13PM 0 A worrying article
2:01PM 0 showing the hardware status of an * system
1:56PM 2 version 1.2 with chan_bluetooth
1:52PM 4 Write to text file in dialplan
1:10PM 1 default user name and password for a2billing
1:06PM 0 CID text stripped over IAX
12:25PM 10 sixtel
12:24PM 1 Sip trunk between Avaya S8700 and Asterisk
12:21PM 1 chan_bluetooth and Ericsson/SonyEricsson models
10:48AM 6 Altering Incoming CallerID
10:31AM 2 Problem compiling libmfcr2 on FC4
9:36AM 1 Asterisk Perl AGI, bug with stream_file() ?
8:31AM 0 Asterisk Realtime 2 Servers calling each other
8:31AM 0 meet me message
8:20AM 12 voipbuster
7:50AM 3 Better transfer
7:23AM 3 Complete Removal of Asterisk
7:09AM 0 Call transfer error
6:46AM 11 WG: App_rxfax problem
6:31AM 0 show queue in BE
6:27AM 5 Error on using queue.
6:17AM 0 mail2fax and fax2mail
4:55AM 1 MGCP problem when through internet
4:24AM 0 Compiling Asterisk 1.2 from Source on
4:04AM 8 Codec Problem
3:16AM 1 cannot dial on console on asterisk 1.2
2:53AM 8 unable to make calls out using ast@home
2:24AM 1 Compiling Asterisk 1.2 from Source on Debian Sarge- Problems
2:22AM 0 optipoint 410 and MWI
12:51AM 5 prepaid application
12:34AM 0 App_rxfax problem