asterisk users - Nov 2004

Tuesday November 30 2004
11:00PM 4 After setting up my FXO card, what should I now order from my telco?
9:29PM 2 Can't get x100p to answer the phone
8:58PM 1 HFC-S card for Australia?
8:43PM 3 Cisco Asterisk Integration
8:06PM 2 * Compatible VSP Service in Ukraine?
7:52PM 2 broadvoice and gsm codec
6:29PM 1 kernel: Out of storage space while 900 MB free?
6:26PM 0 conf from database
5:43PM 1 cisco 7960 sccp firmware version?
5:41PM 0 park app vs. extension 700
4:18PM 0 Pick up call without ringing an extension
3:35PM 0 [BOOK] VoIP Telephony with Asterisk
3:02PM 0 CAll Parking Help needed
2:51PM 1 Issues with zaptel on FC3 - don't know how to fix zaptel after yum update
2:12PM 1 National (US) callerid name resolution for yourasterisk box
1:56PM 3 Asterisk for home office
1:53PM 1 National (US) callerid name resolution for your asterisk box
1:39PM 1 (no subject)
1:13PM 2 Dual NAT for SIP
12:44PM 0 Is it feasible to use 1 SIP account for PSTN connection on an Asterisk gateway accepting IAX2 connections?
12:34PM 0 AW: zaphfc problem
12:33PM 5 Asterisk PBX Manager
12:10PM 0 Trouble-shooting SIP/2.0 482 Loop Detected
11:52AM 0 Simple *69
11:45AM 1 Agents/Queues - Drops call after 60 seconds
11:32AM 3 Fedora Core 2 firewall rules - NO NAT!
11:27AM 0 Polycom Call Park (with sip debug attached)
11:25AM 0 VoIP Business Weekly Article
11:11AM 5 Re: Asterisk-Users Digest, Vol 4, Issue 405
10:52AM 1 Pls help me i can't send a voicemail by sendmail
10:32AM 1 Problem with a new italian service provider...
10:24AM 4 Asterisk Process Stop After few hours
9:55AM 2 Really Get 96 Simul Calls?
9:28AM 3 cisco 7902g
8:20AM 3 ASTCC and Pattern question
8:20AM 2 grandstream bt100
8:01AM 0 H323 -- No Audio
7:44AM 2 Spandsp kind of working
7:29AM 0 Any tool to ease provisioning IAXy?
7:26AM 0 chan_capi compilation problems
7:23AM 0 ParkAndAnnounce Problem, Great Idea, not working consistently
7:06AM 0 E1s ISDN PRI & CPC
6:58AM 0 RE: Parking from call group problems traced to context
6:35AM 5 cisco dial-peer voip
6:32AM 3 Passing Var to PHP AGI script
6:26AM 1 Passing Var to PHP-AGI
6:19AM 4 chan_capi on 2.6 - impossible?
5:41AM 0 Chanspy ?
5:14AM 3 fxo connection in the UK
4:15AM 0 empty username in authorization section ?!
4:14AM 1 realTime configuration help needed
3:08AM 0 SIP client registration ignored by Asterisk
2:52AM 1 Zaprtc seems unsupported, Asterisk in production environment without Digium cards
2:46AM 0 clients behind nat
2:34AM 3 7960 utilize all lines
2:13AM 0 No voice when I dial out
1:58AM 1 Performance problems
1:37AM 2 Is the wcfxo driver sharing an interrupt with Intel 82801DB-ICH4(sound card)
1:08AM 1 wanic 520 with asterisk card
12:19AM 0 Multiple IPs and SIP
Monday November 29 2004
11:51PM 1 SIP.Conf help? (srvlookup)
11:04PM 2 Problems with conference on FreeBSD 5.2.1 w/* 1.0.1
10:55PM 1 Terminal Services + VoIP
9:48PM 2 SPA-2000 Dropped calls
8:45PM 1 IAX port
8:31PM 3 TE410P lights don't blink read after the module is loaded
8:28PM 1 Outbound E&M?
8:22PM 0 IPv6-enabled Asterisk + testing
8:20PM 3 chan_oh323.o
8:06PM 4 Gentoo and Asterisk - any experiences?
7:50PM 2 Cannot Start Asterisk
7:40PM 2 Problems starting Asterisk with TDM22B
7:36PM 0 res_odbc and configuration files
6:09PM 1 Calling from PSTN let exension 601 ring twice, hang up and starts over again to ring twice, ...
6:08PM 2 Compiling zaptel 1.0.2 on Fedora Core
5:54PM 0 Cisco FXO Caller-ID
4:49PM 3 no plain text passwords in iax.conf
4:47PM 1 Cisco gateway help needed
4:26PM 1 T.38 support
3:07PM 4 asterisk newsgrup proposal or phpBB forum
3:05PM 0 I apologize
2:20PM 1 Packet8 integration into Asterisk?
2:15PM 1 CONSOLE/dsp and command line play of wave file
2:03PM 5 Comparision of IAX2, FWD, iaxtel etc etc.
1:48PM 2 Prepaid
12:48PM 1 IAXy power source from Radio Shack
12:45PM 2 Fedora Core 3 & TDM400P cards?
12:38PM 2 Variable substitution - How can I do Dial(${DIALSTRING}) where ${DIALSTRING} is 'SIP/201, 15, tT'?
12:25PM 1 TOS Settings to DSCP
12:12PM 1 [Fwd: Re: Adit 600 channel bank in UK setting]
12:08PM 1 How to rid yourself of Broadvoice
11:41AM 0 FWD * and IAX2...
11:26AM 2 Vonage integration... Hardware or Softphone type acct.
10:57AM 0 Parking from call group problems
10:48AM 4 Small PBX setup
10:35AM 4 Zap gives no ring to the caller...
10:16AM 1 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap'
10:16AM 0 Regular Phones - ISDN NT - FXS Adapters
10:06AM 1 Sending triggers through SIP
9:46AM 2 Asterisk on a notebook: Modem = FXO?
9:43AM 1 IAXy and ADPCM codec problem
9:36AM 0 Subject: IAXy and ADPCM codec problem.
9:18AM 1 Spawn extension
9:02AM 1 Fax pass-throught.
8:44AM 0 IAX2 Warnings - chan_iax2.c:1464 attempt_transmit
7:22AM 3 how to call s extension from SIP phone?
7:08AM 1 Record() and problems converting with sox.
6:17AM 2 Asterisk on a notebook
6:15AM 1 New T100P Pri install suggestions?
5:40AM 1 unable to compile testcpuid.c in spandsp in x86_64
5:10AM 3 low quality sound samples
4:05AM 0 Problem when I call someone who is busy
3:46AM 3 Audio Drops out at Random - one way
3:34AM 1 Dial plan for TDM22B
1:54AM 1 Polycom Reboot Script PRI errors!!
12:51AM 0 Asterisk A LA MEXICANA!!!!
Sunday November 28 2004
11:51PM 0 how to modify dsp.c so that It can detect busytone outside US?
11:27PM 3 D-LINK PoE switch, does it work with cisco or do I need to do the cable trick?
9:54PM 0 Ateus VoiceBlue
9:39PM 0 optipoint400 + MOH
9:21PM 1 optipoint 400 standard + MOH
6:34PM 17 Wiki down?
4:31PM 0 Entire mailing list archive download?
4:01PM 1 OT: mixing monitor files to stereo wav
3:36PM 3 soxmix
1:49PM 1 multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
12:10PM 1 IAX2 and FWD problems?
12:07PM 5 IP to IP call without server?
12:01PM 4 PRI Dialing failure?
10:38AM 4 Registering on Gatekeeper
10:36AM 4 Phone Selection
10:36AM 3 OS Choice ?
10:26AM 0 Registering on GK
10:10AM 4 Asterisk not startin anymore.
9:29AM 2 Asterisk/linux 2.6.9 kernel build failure
8:56AM 2 [Fwd: Call Transfer between phones]
8:37AM 2 am i baned or something?
7:30AM 1 asterisk based bbs
5:49AM 0 Real time queue monitoring
4:23AM 1 asterisk compile errors - pbx_dundi.c -help
3:52AM 0 Fwd: Re: very newbie question
3:36AM 1 SetVar ALERT_INFO
1:32AM 0 Hardware performance issues - Zaptel / wct4xxp for TE405P
12:38AM 2 GNUGK + Asterisk consultant requiered
12:29AM 0 Flash Timings
12:07AM 4 Experiences with Termination Providers?
Saturday November 27 2004
11:39PM 0 problems setting up cdr_addon_mysql
10:34PM 0 Asterisk and GNUGK Consultant Requiered
10:32PM 0 Asterisk + GNUGK Consultant requiered
7:54PM 1 VoiceMail Outdial?
6:12PM 0 Problems compiling zaprtc on 2.4.27 kernel
5:31PM 3 How to test if PCI 2.2?
4:40PM 0 Contact me Asap!
2:56PM 0 Failed to WWW-authenticate on INVITE
2:34PM 3 Problem with voicemailsystem
2:00PM 1 Low Volume WAV Files in Email Attachments
12:37PM 4 very newbie question
11:30AM 0 Can't Register!
10:56AM 3 newbie problem
8:24AM 0 Zapata: No such device or address
8:16AM 0 Built-in Extension Numbers
8:00AM 1 isdn4linux delay
6:45AM 1 getting TDM400P to work in a system that previously had Voicetronix card
6:13AM 0 RealTime Mysql - error
6:10AM 2 rtp compile error
5:17AM 2 capi question
4:46AM 1 Interfacing T100P with Definity PBX
3:27AM 1 asterix as proxy
2:04AM 0 allow=all in sip.conf [genernal] no longer evil (I think)
1:34AM 1 Meetme Help !!!!
1:15AM 1 Reconfiguring a Zap Channel on the fly
Friday November 26 2004
10:18PM 1 Which is the best signalling for FXS
9:54PM 2 Is "Busydetect" obsolete in the latest CVS?
9:25PM 2 Help with broadvoice outbound plz... ;)
8:09PM 4 Grandstream BT102 Busy signal on hangup
7:29PM 4 *67 or *57
7:28PM 1 FWD with iax2
7:25PM 2 Uniden UIP200 -- configured, but not working?
7:07PM 0 ^5 problem with chan_unicall.c for Asterisk
6:40PM 0 TDM22B - how to setup the extensions ??
6:32PM 1 Asterisk+ MGCP
5:15PM 1 direct asterisk to asterisk SIP calls without external SIP provider
2:37PM 0 ast_data causes to fail
1:28PM 0 advise for cheap ISDN card which works with chan_capi and supports p2p mode
1:15PM 2 low voice only
1:05PM 1 Voicemail / sendmail
12:45PM 0 ^^4 problem with chan_unicall.c for Asterisk
11:20AM 1 OT - how to get BT to present a number
11:16AM 0 "reason 23 (Temporary failure)" when using Dial(OH323)
11:10AM 4 SIP phones cutting out with Asterisk??
10:18AM 0 sip call test
9:37AM 2 E&M Digium card quotation
9:09AM 0 Exiting out of Voicemail with a '0'
8:56AM 1 How to transfer value to extensions.conf?
8:56AM 1 Asterisk - ACD.
8:42AM 2 T1 and FX CPE
7:24AM 1 AGENTDUMP lines
7:20AM 0 P2P (DDI) mode with chan_capi 0.3.5
7:03AM 2 Execute a script upon registration
6:16AM 0 Can I trigger an application in * with DTMF tones, during a call?
4:28AM 0 Forwarding SIP calls to another SIP Proxy (Peer)
4:14AM 2 IAXy and DHCP
4:00AM 4 overriding DTMF and codec from dialplan?
3:39AM 0 load wcfxs module
3:36AM 2 problem with chan_unicall.c for MFC/R2 with asterisk
3:31AM 1 Monitoring app. - see whats really going on in asterisk
3:16AM 0 PrepaidAuthCID - nothing happens
2:55AM 1 can anyone will help me regarding autodialing in asterisk
2:48AM 1 Quality of the voicemail sound
2:35AM 3 Billing of outoging calls via CAPI
2:21AM 0 snom - blinking leds on fuction keys when call is not yet established - how?
2:05AM 1 Monitor performance
2:05AM 1 General feature questions
1:51AM 0 Re[4]: [Asterisk-Dev] Asterisk Hardware Platform - Intel x86 versus Intel RISC Xscale (ARM)
12:56AM 4 Where did USE_MYSQL_FRINDS go ? What to use ?
Thursday November 25 2004
11:56PM 1 Consultancy service needed urgently !
11:45PM 1 No Music: Queue Hold and MusicOnHold
10:10PM 0 Problem with IAX2 Unregistered in the chan_iax2.c and data_pgsql.c file
9:55PM 1 Problem with onboard sound card on kphone
8:12PM 3 Playing reveived message WAV file
6:26PM 3 OH323 Rocks :) --- H323 guys, use it to solve no answer at this time problem!!!
1:09PM 1 SNOM telephones and LEDs
12:51PM 1 Fax server (TxFax) fails during transmission
12:04PM 1 Interview with Mark Spencer
12:03PM 3 redhat9 100% CPU
12:02PM 1 allow=SLINR
11:55AM 0 Solution - ISDN-PRI hangup cause
11:38AM 1 Stanaphone down?
11:02AM 4 Opinions on renice or turning off swap or ramdis k as swap?
10:26AM 0 Area Code 514 DIDs
9:39AM 3 configuring voicemail
8:56AM 2 Cannot get two TE410Ps to operate correctly in the same machine
8:38AM 1 astcc newbie question
8:33AM 0 probleme with running lib_unicall with asterisk
7:59AM 1 Module Failure
5:57AM 1 Call to x-lite clients failing?
5:11AM 0 Asterisk Hardware Platform - Intel x86 versus Intel RISC Xscale (ARM)
5:10AM 0 record call on demand
4:09AM 0 How to make/recieve call using asterisk whenthereis a power failure?
3:57AM 0 Forwarding Call
3:53AM 4 Billing (itemized) in the UK
3:46AM 1 No hangup(vpb)
3:26AM 0 ZAP FXS problem - no caller id
3:10AM 1 Connecting a PBX with Asterisk via E1 / PRI
3:06AM 2 How to make/recieve call using asterisk when thereis a power failure?
3:02AM 1 astGUIClient Question
2:56AM 3 How to make/recieve call using asterisk when there is a power failure?
2:03AM 2 oh323 compile issue
1:50AM 1 Can't hear playtones?
1:24AM 3 Zaptel on Suse 9.0
Wednesday November 24 2004
11:58PM 0 supported RFCs
10:25PM 1 asterisk 1.0.1
10:22PM 2 Changing Asterisk Voicemail Storage Location
9:50PM 0 Unable to open master device
9:44PM 1 Cannot open /dev/dsp
9:21PM 1 I just got my TDM22B - but no data sheet
8:42PM 2 Asterisk Digium FXS
8:04PM 0 H323-Asterisk-SIP-TNT consultant needed
6:57PM 0 Call External Program When SIP Message Arrives
5:18PM 2 asterisk and verizon DSL
2:57PM 2 how to use stop calls
2:48PM 2 Bothering with H323
2:43PM 0 How to Modify Diversion Header for 3rd Party SIP Vmail?
12:50PM 1 Just upgraded from multiple X100P's to a T100P
12:40PM 3 Haven't got a clue ...
12:23PM 1 Question on IXAy
11:03AM 5 GUI
10:31AM 1 How to decrease the speech volume for record?
9:33AM 2 Graststream ATA 286 Caller ID Europe
9:17AM 4 asterisk and pstn
9:14AM 4 zap fxo hangs after upgrade to stable v1-0
9:09AM 1 Asterisk/Panasonic PRI Integration
8:45AM 1 Problems with udev on FC3
8:21AM 1 Re: Asterisk timer for Freebsd
7:51AM 1 Busy Lamp Field
7:04AM 1 Sip test
6:24AM 2 Asterisk and Dialogic LSI161SCREV2 --- Don't kill me ; -)
6:12AM 2 Codec control
5:59AM 2 call forwarding to gsm phones
5:56AM 1 bristuff'ed version doesn't run
5:36AM 3 Asterisk with ISDN
5:31AM 1 gateways failover with asterisk
4:40AM 1 vm notification no longer contains calling party
3:36AM 3 Grandstream Firmware Attended Transfer
3:34AM 1 Find extension from Dial(,M()) macro
3:11AM 0 Have anyone successfully install Daniel G729 test suite ? mine core dumped !!
2:14AM 1 Horrible BUZZZZ noise when sounds/music play on SIP phone?
1:46AM 1 Which modem is known to work with asterisk?
1:27AM 0 No debugging informations on the CLI after patching with ast_data 1.0.2
Tuesday November 23 2004
11:41PM 7 Unable to open master device '/dev/zap/ctl'
11:17PM 1 CLI > h.323 show codecs shows nothing
6:12PM 2 need some advice
5:28PM 1 AstriCon offers a most sincere and humble apology for the barage of mail...
5:19PM 0 SBC ADTSe - Sending DP digits
5:14PM 1 Is there a way to check if an extensions exists in a context before you send the call there.
4:52PM 2 Asterisk on a Linksys WRT54G(S)
4:50PM 1 linking 2 isdn30 and 2 meridian cards
4:36PM 0 rtp.c dtmf issues solved.
4:17PM 5 Fw: Gift for Mark Spencer
3:46PM 2 Re: list proposition
3:13PM 0 using asterisk to bridge H323v1 to SIPv2
2:58PM 3 Re: List proposition
2:33PM 2 Can isdn data calls routed through 2 t100p's
2:18PM 1 IAX2->SIP->meetme = ZOMBIE
1:41PM 0 meetme2 can't set status
1:31PM 2 Yet another faxing issue..
1:04PM 1 Queue Patch - estimated hold time announcements
11:10AM 4 ATA186 upgrade
10:19AM 0 Zombie channels dropping lines
9:53AM 5 ATA186
9:17AM 4 Quick Questions - IVR=Auto Attendant?
9:03AM 1 Fax over SIP Problems (sorry for this topic ...)
9:01AM 4 Forwarding calls
8:53AM 1 CP-7960
8:52AM 0 Fax over TDM400 and E100P disconnects
8:32AM 4 ASTCC Routes
8:19AM 3 Firefly on Linux
8:10AM 1 Error when install E100P
8:04AM 0 please help !! - context for an incoming call
7:47AM 0 SIP Registration failed notices
7:00AM 4 Spandsp and Asterisk
6:34AM 2 Re: Asterisk-Users Digest, Vol 4, Issue 300
6:31AM 2 PRI Logging
6:17AM 5 NEED HELP!!
6:14AM 1 Polycom 500 bootrom.ld problem
5:34AM 2 Commercial g723.1 license for asterisk
5:14AM 0 RE : -lssl
5:14AM 0 Asterisk not relaying SIP messgaes
5:04AM 2 -lssl
4:48AM 4 oh323/g729 and DTMF
4:45AM 0 astcc db creation
4:41AM 1 Newbie questions from South Africa: Initial setup
4:37AM 0 Huge ten second audio delay on SIP channel
4:23AM 0 Problems with MACRO_EXTEN variable
4:11AM 0 Random Audio Drop out one side
4:11AM 1 a=rtpmap:101 telephone-event/8000
3:34AM 1 Paul Mahlers Book
3:08AM 1 Firefly:Canreinvite problem
2:33AM 1 Error on install under Fedora Core 3
2:15AM 1 dail cli
2:03AM 0 DTMF mode autodetect?
12:54AM 0 Asterisk & Windows Messenger
Monday November 22 2004
10:17PM 1 Uniden UIP200 configuration -- manual MIA?
9:33PM 3 ChanSpy
9:08PM 0 H323 linking with asterisk
9:07PM 0 How to configure the Asterisk server such that a FXS phone can talk to SIP client?
7:57PM 2 dtmf tones during conversation
7:33PM 1 Anyone use SixNet for IAX termination?
4:42PM 0 SIP phones disconnect frequently
4:19PM 1 T100P -- data?
4:18PM 0 Asterisk with MeritCall
3:59PM 0 new application swait...
3:59PM 2 chan_h323 on AMD64
3:43PM 2 Granstream BT100 - only partial success
3:25PM 0 Asterisk and Bastille
3:02PM 0 Configuring Asterisk From Postgres
2:46PM 2 sip.conf not paying attention to allow/disallow
2:31PM 2 Polycom Problems
2:11PM 0 Re:SIP Problem
1:57PM 9 asterisk gui?
1:33PM 1 SIP Problem!
1:07PM 1 Cisco 7940 Volume low
12:38PM 8 Patching asterisk for spandsp
11:51AM 3 Cisco 7960 version 7.3 SIP not always able to hear calling person
11:16AM 1 Using IPKall and SIP with insecure=very
10:57AM 2 edirecting calls with Asterisk
10:53AM 0 asterisk manager api to stop a stream file command in an agi
10:33AM 3 IPv6 and Asterisk?
10:24AM 3 Zap - 256 format frames
10:16AM 1 Siemens optiPoint 300
10:08AM 2 Problem with fax tone (CNG) from TxFax and busy detect
9:47AM 2 RE: Asterisk-Users Digest, Vol 4, Issue 298
8:56AM 0 Cisco Call Manager and Asterisk
8:47AM 2 Creating CDR's with online connected time
8:47AM 6 Linksys RT31P2
8:41AM 2 Unknown number CID on SIP phone
8:03AM 3 which ISDN Card?
8:03AM 1 Call Deflection (CD) with ZapHFC
7:56AM 1 callprogress option
7:52AM 1 Test Number in the UK?
6:49AM 1 wiki down ?
5:26AM 1 Strange Fromuser behavior?
4:27AM 0 Problems with not correctly unregistered users...
2:59AM 1 IAX error tolerence??
1:47AM 3 hangup()???
Sunday November 21 2004
10:26PM 2 SPA-841 / SPA-2100 Canadian Distributor
9:48PM 0 Is there Asterisk module for Logwatch?
8:14PM 1 Mailing List Admin - Remove annoying user [Fwd: RE: Re: Get the Caller-ID without Answering]
7:17PM 2 Examples of hardware implementations
6:04PM 3 Get the Caller-ID without Answering
5:24PM 1 SER is a better NAT solution?
4:01PM 0 Headsets for Polycom Soundpoint 500/600
4:01PM 3 Headsets for Cisco 7940/7960
1:49PM 3 Error "WARNING[-150101888]" when starting Asterisk.
1:01PM 0 iax busy / unavailable - not registered
12:50PM 2 Fw: TDMoE over bonded NIC's
12:18PM 3 TDM400 FXO stops handling outgoing calls, but still accepts incoming?
10:49AM 0 HFS in NT mode getting PRI got event: 6 on Primary D-Channel of span 1
10:22AM 3 I Am Missing Something Somewhere Somehow!
10:02AM 1 Gatway with IAX ?
9:56AM 0 sip debug command?
9:37AM 1 Grandstream Ringtone
9:34AM 0 No incoming calls on skinny phone
9:31AM 0 Asterisk Newsletter :: Back online!
6:54AM 1 incompatible with our capability 0x400.
6:39AM 1 make asterisk accept Register messages
6:31AM 0 Flashing Active ZAP Channels
5:24AM 4 Snom 190 - dhcp - settings_server
2:38AM 4 UK available SIP phone?
1:08AM 1 Using CallingPres to set up CallerID blocking
Saturday November 20 2004
9:43PM 0 * and scansoft TTS
9:13PM 1 Asterisk dead but pid file exists - gdb asterisk core.13089
8:02PM 1 extensions.conf help needed
5:16PM 1 IAX IAX connection
5:10PM 0 zaptel driver problem
5:06PM 3 A new alternative to see who is online
4:44PM 0 Changing simple switch dialtone
4:36PM 0 Odd situation with Cisco 7960 IP phone
4:29PM 2 Problems with call files (/var/spool/asterisk/outgoing)
4:04PM 1 TE410P PRI problems
3:57PM 1 ANY DEVELOPERS HERE? "warning: implicit declaration of function `__use_ast_pthread_create_instead__"
3:46PM 1 Queue Sounds - not working?
2:45PM 6 SIP Phones-Receptionist Setup
2:35PM 1 IAX Dialstatus
2:23PM 0 SIP Call not Approved
2:08PM 1 IAX issue at nufone
12:51PM 1 How to encript SIP comunications?
11:15AM 0 Playing announcement when call is answered
10:39AM 0 Can anyone shed some light on wht these calls were dropped?
7:44AM 1 three way mixing / conferencing
7:32AM 0 Setting the EXTEN variable - is it possible?
6:33AM 0 Fax testing using loop-back
5:53AM 2 zaphfc sound problems
4:37AM 0 Bug with Dial in AGI script?
2:15AM 3 block caller id
1:08AM 1 * and NAT
12:57AM 0 SIP to IAX using G.729
12:46AM 1 Monitor Command
Friday November 19 2004
10:08PM 4 Multiple asterisk process
9:06PM 2 Polycom Soundstation IP 3000 firmware
8:48PM 2 Just getting started...
6:49PM 1 PRI NI2 and callerID name
5:32PM 5 Asterisk and H.323 Gatekeeper
4:40PM 1 Starting AGI when handset is picked up?
4:31PM 4 IAXy Configuration
4:27PM 0 MINNESOTA: TwinCities Asterisk Users Group - Meeting tomorrow 11/20/04
3:51PM 0 Question involving Windows Messenger 5.0 and Asterisk (SDP related)
3:13PM 0 SIP Clients other than 200-299
3:09PM 0 remote iaxy device Ping:OK iax2 Poke: no answer
2:48PM 0 asterisk and level3
2:38PM 3 Alcatel PBX
2:22PM 1 Newbie Basic Questions
12:27PM 4 Error during installation
12:01PM 2 How to enter billing codes when dialling
11:17AM 0 Mitel 5220 phones
11:04AM 1 Voice + DTMF
10:31AM 1 Broadvoice update
10:18AM 2 Shared line appearances
10:16AM 5 txfax
10:11AM 0 Cisco 7970 Non-SIP Phone setup with Asterisk
10:08AM 0 differents contexts for a channel
9:40AM 0 Asterisk and Tecom IP2005 phone, problems :(
9:12AM 1 SBC VoIP Tariff to ISP's
8:40AM 0 AgentMonitorOutgoing => is there an opposite ?
8:22AM 4 hello
8:20AM 0 helo
7:52AM 2 Zaptel init script
7:50AM 0 H.323 Status
7:28AM 0 Asterisk crashes with Unicall
7:28AM 5 Fedora Core 3 supported?
7:22AM 2 "Best" line protocol for T1
6:54AM 1 rtp codec error
6:40AM 2 app_sms: problems sending a sms
6:16AM 0 Fwd: MARIO SPOLJAR is not longer working for PLIVA
5:56AM 2 Routing between different interfaces
5:56AM 2 Need help selecting phones
4:48AM 0 X100P and Siemens Gigaset 4175
4:19AM 5 Unpredictables Hangups
3:47AM 1 R: problem with zyxel prestige 2002
2:40AM 2 OT - 3com 3C17205 & cisco 79xx
2:08AM 0 Ericsson or ACC - AXC or Tigris ??
2:08AM 7 i swtiched to digest
1:40AM 2 compiling error
1:14AM 2 E100 or TE410 card an PRA line
1:10AM 1 Digium E100P or TE410P card
Thursday November 18 2004
11:27PM 0 Linking H323 with Asterisk
11:26PM 2 changing configuration file
9:37PM 1 X-Lite and Voicemail
9:02PM 3 Little off topic
8:55PM 0 [perhpas OT] asterisk holding rtp ports open with natted spa-3000
8:09PM 2 [Asterisk-User] recommendation for IP phones
7:55PM 3 SipTone II
6:48PM 3 Is H323 dying?
6:23PM 3 iaxComm to iaxComm
6:17PM 0 Asterisk with verizon DSL and Westell 2200 DSL router
6:14PM 2 Interrupting MusicOnHold while call in queue ?
5:52PM 2 (Analog Intercom) PagePal by ATT -- was hooked to a Merlin
5:45PM 0 DTMF stopped functioning after upgrade to 1.0.2
5:21PM 3 Best SIP phone for high quality telemarketing
4:48PM 3 "Lobotomized" Sipura SPA-3000 configuration needed
4:10PM 0 Video Phone recommendations for SIP trunking on *
3:47PM 1 Sparc hardware, Linux and X100P
3:21PM 1 Re: Netgear powered switch
3:16PM 1 [Fwd: Re: Adit 600 channel bank in UK setting]
2:57PM 2 Speaking of DS3s....
2:46PM 1 [OT] PoE switch question (Netgear FSM7326P works
2:45PM 1 Est. count of deployed Asterisk environments?
2:26PM 3 Spam: I really need help with this!!!!!!
2:23PM 4 Controlling Asterisk from PHP?
2:08PM 0 Polycom 300 registration
1:51PM 0 FW: More than 20 FXS
1:50PM 0 No Voice Path With PSTN Call Forward
1:47PM 2 please unsubscribe all members
12:56PM 1 Incorrect parsing of 'unavailable' caller-ID from Cisco gateway
12:01PM 0 OT
12:00PM 0 DTMF noise
11:54AM 2 More than 20 FXS
11:35AM 0 VoIP engineer and technical/networking support
10:41AM 1 Help wanted getting Busy / Congested working properly
10:02AM 2 VOIP security on an IAX connection.
9:56AM 0 Adit 600 channel bank in UK setting
9:46AM 5 TE410P - How many can I have?
9:43AM 0 app_icd compile problem
8:42AM 0 [OT] but of interest to Grandstream users : firmware .5.18
8:28AM 0 Playtones problems
8:21AM 1 Find out the reason for dropped calls?
8:20AM 4 please Can some bady help me ???
8:14AM 8 X100p and 6 second delay
8:08AM 0 safe_asterisk isn't auto-restarting
7:51AM 1 Polycom IP 300 PoE? Sipura instead?
7:35AM 1 AW: Voice in Asterisk with BRI ISDN Any properworking configurations yet?
7:07AM 0 asterisk connecting to cisco call manager using quad T1 card
7:03AM 1 Analog ports via USB
6:45AM 0 AW: Voice in Asterisk with BRI ISDN Any proper workingconfigurations yet?
6:40AM 2 Voice in Asterisk with BRI ISDN Any proper working configurations yet?
6:38AM 0 Asterisk server to asterisk server question
4:53AM 1 Zyxel Prestige 2002/2002L sound quality
4:51AM 1 Problems using AGI->get_data
4:48AM 0 FreeBSD asterisk-addons
4:24AM 1 setup question
4:22AM 0 OH323_OUTCODEC=g729 has influence on chan_iax?
3:51AM 1 mISDN & kernel 2.6.9
3:32AM 1 Setup/SIP routing
3:11AM 0 H323 and AMD64
3:01AM 2 configure channels
2:32AM 5 Music on Hold on Debian 2.6 help wanted
2:22AM 0 ISDN BRI one way voice quality problem
12:55AM 5 internet bandwidth
12:52AM 0 Queue calls- multiple to same extension, max extensions?
12:35AM 0 inernet bandwidth
12:04AM 0 Queue using iaxy agent fails?
Wednesday November 17 2004
11:58PM 1 [OT] PoE switch question
11:16PM 3 Auto Dialing
10:29PM 0 call delay problem after call recording
10:06PM 0 return codes from extension.conf
9:56PM 0 Anybody got asterisk workin with Diva 4bri and fdora core 2?
9:51PM 0 Call ID WinPopup working one-line example withoutscratch file
9:23PM 2 Call ID WinPopup working one-line example without scratch file
8:56PM 1 E100P Media Gateway With Asterisk
8:37PM 1 Motherboard with TE405p
8:30PM 1 Asterisk Call ID Popup
8:13PM 1 Digits entered ARE NOT RECOGNIZED by bank's IVR's
7:51PM 1 Mini Call-ID Winpopup
6:37PM 5 The Apperiant Death of IAXtel
5:52PM 2 OT: Why "encrypted" config files
5:20PM 1 Removed default indication country 'us'
4:42PM 2 Cisco SIP Firmware HERE!!!!
3:27PM 3 Polycom IP 300 PoE?
3:19PM 1 Problem with an hardware phone: Maximum retries exceeded
3:17PM 2 Call Status
3:07PM 4 Cisco 7970G VOIP phones
2:47PM 0 start_pri: Unable to open D-channel 24 (No suchdevice or address)
2:46PM 0 Strange g729 error. Just now started.
2:34PM 1 Coverting Cisco 7960 to SIP
2:23PM 6 How to generate "ringing tone" to a calling party.
1:44PM 1 Zap card, PRI, Fax detection, and 1.0 stable
1:05PM 5 Call ID Mini-Popup?
12:44PM 2 PowerEdge 17500 with TDM400P - 4 FXO -- NMI, loud noise when dialing out
12:42PM 0 CallerID and Outlook / CSV
12:21PM 3 chan-sccp problem, phone is not registering
12:18PM 2 AstLinux 0.1.3 released
12:15PM 1 Why <ZOMBIE> ?
12:01PM 0 BroadVoice patch on latest CVS snapshot
11:32AM 3 IVR and voice mail using G729
11:07AM 4 patch for chan_capi to compile with latest CVS
11:06AM 2 Asterisk on Solaris
11:03AM 1 Does ASTCC Require CDR_MySql?
10:45AM 4 Possible to display which extensions are in use on the phone's display?
10:36AM 0 chan_capi dialout problem
9:57AM 1 Polycom phone question
9:52AM 0 H.350 integration
9:47AM 4 Software SIP Phones
9:38AM 1 TDM400P callwaiting, threewaycalling and cancallforward problem
9:36AM 2 Firefly 1.9.5 and 20041117 CVS HEAD -- IAX2 one way audio
8:30AM 0 AP200B Phones
8:09AM 2 Max retries exceeded to host ...
8:08AM 0 AP200B or C
7:14AM 2 Port for Asterisk
6:54AM 1 IAX authenticated transfer
6:12AM 0 Russian Asterisk community
5:50AM 1 Compile error on spandsp-0.0.2-pre6
5:06AM 1 Re: Asterisk-Users Digest, Vol 4, Issue 222
4:46AM 0 Ringing tone on calls going out on chan_modem
3:58AM 1 TDM FXS Module & caller ID
2:20AM 0 Cannot create mysql database with TRABAS
2:09AM 1 Hardware selection
1:24AM 0 why dsp.c can not detect busytone?
Tuesday November 16 2004
10:00PM 1 Connection of Asterisk to Cisco Callmanager via H.323
8:12PM 2 Errors Compiling chan_capi 0.3.5
7:45PM 0 Asterisk-Users Digest, Vol 4, Issue 222 (fwd)
6:43PM 1 IAX2 peers via MySQL DB with Asterisk 1.0.2
5:47PM 1 sending faxes with asterisk in between
5:14PM 1 Using a Aastra/Nortel 390 Phone with Asterisk
4:18PM 1 Grandstream Dial Tone from PBX
3:30PM 0 no media for VM
3:17PM 2 RJ11 and Digium TDM 400P
3:01PM 0 TDM31B Interrupt Issue SOLVED! :-)
2:43PM 2 Recording from AGI playback is LOW
2:36PM 1 T405P Mulitiple Signalling modes on 1 card.
2:12PM 9 Variables
1:59PM 0 LookupCIDName - 1 vs ""
1:31PM 10 SS7 for *
1:03PM 1 RE: Sending DTMF Digits for DID
1:03PM 2 Interrupts failure on T100P
12:51PM 3 Dial by name
11:48AM 0 Suggestion for SIP video phone for windows CE
11:48AM 0 Newbie - NO Problems!!! - System Info
11:41AM 0 broadvoice connection error message
11:26AM 0 Timing Question:) (Loop/Internal etc).
11:17AM 0 SIP Video Conferencing System to PRI
11:11AM 0 Sending DTMF DID w/ Asterisk
10:33AM 3 SIP register problem
10:31AM 0 IAX2 unable to transfer?
10:26AM 2 Gaps in sound
10:14AM 0 Source for generic linksys phone adapter?
10:03AM 1 Zaptel Compile Problems with 1.0 Stable
9:41AM 2 Asterisk API Docs
9:28AM 1 Asterisk CLI access permissions?
9:26AM 1 Log extension in CDR when forwarding calls to another number
9:00AM 2 Newbe Question
8:07AM 2 Newbie - NO Problems!!!
7:52AM 2 TDM31B has no interrupts?
7:35AM 1 Asterisk with "chan_misdn" (in USA)
7:23AM 1 Using Asterisk as an external MOH for Televantage5?
6:58AM 0 Multi Lines in Asterisk
6:08AM 0 Snom and Stun
6:04AM 0 if NOT SipUser then Dial(Zap/1/${EXTEN})
5:44AM 1 Capi Deflection (CD) not working
5:29AM 0 404 error found when making SIP point to point calls
5:23AM 2 Problem with sox
4:36AM 0 Agent channel problem
4:35AM 1 freebsd & voicemail everything seems to work??
4:05AM 0 new version problem
3:49AM 2 Voicemail Digits
3:40AM 0 Unable to get Incoming Calls
3:01AM 0 backtracing ABANDON entries to CID in queue_log?
1:09AM 0 FXO ?
Monday November 15 2004
11:40PM 0 MTA 3308 (Innomedia)ipphone does it work with asterisk
10:25PM 0 Asterisk queue
9:31PM 1 How to emulate a multiline phone in Asterisk
8:26PM 2 Is IAXTEL working?
8:09PM 1 OH323 and gatekeeper
5:12PM 3 Memory Consumption
4:08PM 4 Skype API release
3:57PM 5 Question about remote POTS lines
3:46PM 1 Measuring Bandwidth on T1 into *
3:45PM 2 VM Greeting
3:36PM 0 Using Asterisk as an external MOH for Televantage 5?
3:35PM 6 Standard messages instead of MOH during dial
3:24PM 0 Meetme and audio recording/playback
3:23PM 3 Auto dialout
3:09PM 1 ISDN, fax and bristuff
2:33PM 2 Problem with NAT on Asterisk 1.0.1
1:40PM 3 ADSI questions for a 390 ADSI Phone
1:40PM 1 Traffic shaping script for kernel 2.6 and SIP?
1:37PM 0 Asterisk scalability IVR/Voicemail only
1:35PM 1 Asterisk and ISDN
1:23PM 4 $10 for G.729 ?
1:01PM 4 Broadvoice number always busy
12:19PM 3 Manager API Call Origination & Variables
12:03PM 1 MC3810 IOS
11:54AM 0 Avoiding 2 ring callerid delay for calls that don't go to voicemail
11:54AM 1 Multiple TDM400 vs T1
11:49AM 2 Odd error at startup
10:40AM 1 Help with this debug output?
10:21AM 1 VICIDIAL in windows xp
9:18AM 1 TMD400 FXO <-> Nokia 32 GSM (Hangup Problems)
9:06AM 1 Transferring calls from a Zyxel P2000w
9:04AM 4 MYSQL Dialplan Question
9:03AM 1 FXO setup
8:41AM 1 IAX2 trunking - timing - ztdummy??
8:34AM 2 Where can I find searchable version of this list?
8:26AM 0 NETDEV WATCHGOG eth0 timeout
8:14AM 0 (no subject)
8:14AM 0 Transfer # - Intermittent with Cisco 7905 SIP Phone
7:57AM 0 Multiple options to Dial command - what is the correct format?
7:32AM 0 irq CPU state
7:31AM 2 asterisk nagios plugin
5:22AM 0 Re: zap channel won't send/receive calls
4:44AM 0 iax preferred codec question?
3:18AM 2 PSTN -> Asterisk -> PSTN Call quality
3:03AM 0 Sip relay with asterisk
1:45AM 1 Meetme2 - web interface not working
12:46AM 1 AU FreeBSD PRI Hardware
12:34AM 0 SIP (or IAX) modem driver
Sunday November 14 2004
11:05PM 1 AU PSTN Tone / Progress Detection
8:32PM 0 WAV file volume in voicemail - anyone actually solve this?
8:30PM 2 Linux Kernel 2.6 Questions - safe_asterisk and udev
8:25PM 2 ResponseTimeout problem
7:38PM 1 Service Providers With Caller ID Name??
6:47PM 0 Hangup Phone
4:16PM 0 Snom 220 Problem
4:06PM 0 (no subject)
3:57PM 0 SIP Packets stuck in queue
3:11PM 0 asterisk & ser setup consulting needed
2:52PM 0 ERROR: retrans_pkt: Maximum retries exceeded on call
1:51PM 1 problem with zyxel prestige 2002
12:49PM 1 3 - TDM31B Card Installation Difficulty
12:40PM 1 Asterisk using the wrong peer in sip.conf
12:30PM 2 Asterisk update
12:25PM 0 MacOS/x softphone and g729a
11:02AM 2 Voicemail shorter then (ex) 2sec - don't accept
10:50AM 7 Dial Plan Pattern Matching
10:36AM 3 SysMaster and GPL Violation (lets think before we jump)
9:51AM 0 Asterisk and Digium
9:41AM 0 Does Music On Hold not work on Debian???
7:41AM 0 ODBC Message Waiting Indicator
7:06AM 0 How to route all incoming call to the defines context in extensions.conf
6:39AM 0 Elesign - ESC2420.
6:24AM 0 Garbled sound - CPU or traffic problem?
6:21AM 2 H323/*/IAX <-> Firewall <-> IAX/*/H323
4:30AM 0 Asterisk-prepaid
4:07AM 0 AgentCallBackLogin and queue_log
2:19AM 11 (newbie) no dialtone on a TDM400P card
1:42AM 2 skinny error
12:25AM 1 re: DVG-1120
Saturday November 13 2004
11:36PM 0 Queue/AgentCallbackLogin Problems
8:36PM 3 Cisco ATA and G729
8:33PM 0 my asterisk drops connection when remote side puts me on hold?
6:14PM 2 manager api: how to handle failed calls
5:23PM 1 Best setup for BudgeTone
4:50PM 2 isdn to sip gw
4:13PM 3 Remote answer not detected
2:28PM 1 spandsp problem
11:57AM 5 NAT
10:11AM 1 Cable for T1 connection: Crossover or straight through?
9:03AM 0 New TA from Uniden
8:47AM 2 wctdm to replaces wcfxs module ?
8:43AM 2 Broadvoice Patch issues
4:58AM 1 Cisco IP phones, SIP, Call-Manager & Contracts
2:48AM 1 SPA-3000 Wizard for Asterisk
1:11AM 5 Over 10,000 lines. Will asterisk manage?
1:09AM 2 Extension "follow me"
Friday November 12 2004
11:35PM 1 random echo on TA750
9:34PM 0 Cisco 7940 multiple line capability questions...
7:20PM 1 Advice on starting out
6:48PM 1 Calling an outside number along side other internal extensions?
6:32PM 1 Authenticate or DISA?
6:26PM 1 pressing a key to get out of voicemail?
6:04PM 1 voip to pstn
5:01PM 0 DECT channel
4:51PM 2 CNG Comfort Noise Generation
4:04PM 1 Need low-cost flat-rate incoming DID's throughout the U.S - Anybody competing against VoicePulse?
3:06PM 1 Kirk IP 600 DECT station
2:43PM 0 gold rush?
1:50PM 1 Can someone tell me what is going on from this debug?
1:47PM 0 Answer Confirmation "c"
1:46PM 0 ACD queue timeout problem
1:10PM 2 BRI in the US
1:01PM 1 Quick call group question...
12:58PM 0 Asterisk crashes after call when running as non-root, bug???
12:21PM 1 Combination Cellular and WiFi/SIP
11:12AM 0 FW: Strange error
11:02AM 1 Asterisk Administration and Management requi rements (splinter from $200 AMP bounty thread)
10:47AM 1 Audio troubles on the Zyxel 2000w
10:43AM 1 SIP REGISTER -- Via -- Oooops?!
10:42AM 3 Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)
10:33AM 1 SIP & ALERT_INFO for distinctive ring
10:16AM 5 Strange error
10:06AM 2 $200 AMP documentation bounty < - Comments o n the Linux user experience
9:58AM 0 Faster g726 and ADPCM
8:50AM 0 Strange Behavior, static and clicking on outbound calls only.
8:43AM 0 Motherboard whitelist (was Echo - UK Impedan ce problem with X100P?)
8:32AM 0 Ring after hangup with Rhino Channel Bank
8:28AM 2 The BV patch: Some notes
8:21AM 1 CDR & MySQL Problem
8:19AM 4 OT: Grandstream problems
8:18AM 0 DID/PRI sending to the s, extension <-solved it
8:11AM 2 Motherboard whitelist (was Echo - UK Impedance problem with X100P?)
7:59AM 1 Lock the phone when no using it
7:32AM 3 Cisco 7912g SIP firmware
7:27AM 8 $200 AMP documentation bounty
7:26AM 0 Asterisk behind external PBX +enable IVR
7:21AM 0 astGUIclient - 1.0.4 (Running in Windows) an d SQL Updater Down
6:53AM 2 Caller ID for Japan?
6:45AM 1 astGUIclient - 1.0.4 (Running in Windows) and SQL Updater Down
6:44AM 1 Conferencing needs Zaptel ??
5:51AM 5 Echo - UK Impedance problem with X100P?
5:40AM 2 timeout
5:39AM 3 Calling h@ and Loop Detected
5:08AM 0 Continuing a call to callee after caller has hung up.
4:30AM 1 Siemens voip adapter
3:51AM 0 attempting native bridge error
3:12AM 0 SIP Register with Huawei equipment HELP
2:54AM 3 Dial without bridge
2:37AM 0 SIP clients <--> SE R <--> Asterisk <--> carrier/gateway
2:33AM 1 No ringing with Phonejack Lite - hardware or software problem?
2:06AM 1 Install X-lite automatic with (windows) .ins file
1:40AM 1 Recent * SRPMS
1:10AM 1 wcfxs module gone from CVS head?
12:47AM 0 How to see if I have PCI 2.2
Thursday November 11 2004
11:38PM 0 New Zealand Centrex Service
10:45PM 7 SysMaster and GPL Violation
8:14PM 0 SIP distinctive ring (BroadVoice)
7:49PM 0 SIP no working in 1/4 installations
7:34PM 1 TDM400p module error?
4:53PM 0 DID/PRI sending to the s, extension <-more i nformation
3:59PM 1 DID/PRI sending to the s, extension
3:41PM 1 FXO dialing - all lines dial but one
3:23PM 0 Problems compiling chan_capi with latest CVS
3:09PM 3 ive noticed that our 1.02 stable box's asterisk is taking 100% cpu load..
3:09PM 1 DHCP from server A and connect to server B messes with SIP call out.
2:26PM 1 ZT_CHANCONFIG failed on channel 1: No
2:19PM 2 DID/PRI sending to the s, extension instead of t he DID extension
2:12PM 1 sometimes problem with dialing ZAP channel
2:02PM 15 Can some bady help me ???
1:39PM 0 One way audio on calls across a TDM400P
1:07PM 0 Cisco 79XX phone using dhcp can call out but not in
12:55PM 3 Deploying multiple Sipura 3000s with Asterisk
12:39PM 2 ZT_CHANCONFIG failed on channel 1: No such device or address (6)
12:06PM 0 broadvoice patch and 16 second re-registers
12:02PM 1 Zaptel module load errors under stock FedoraCore 2 (2.6.8-1.521 kernel )
11:32AM 0 Problem using Digi DataFire Micro V
11:14AM 3 Palm Tungsten and Asterisk
11:00AM 1 Grandstream BT100 - No Sound with Playback()
10:58AM 4 Snom 190/220 dialplan strings?
10:41AM 0 astGUIclient Problem -- stguiclient/admin.php
10:28AM 1 astGUIclient Problem --
10:10AM 0 Preventing Call Forwarding by SIP UA
9:53AM 1 setup of cisco 7960 phone tftp asking for unkownfile
9:46AM 0 working Marconi sys X config
9:42AM 3 setup of cisco 7960 phone tftp asking for unkown file
8:40AM 0 Special Characters In Passwords
8:07AM 6 cisco poe
7:40AM 2 Monitor/Record MeetMe Conversations
7:19AM 1 failed to go to next dial command
6:51AM 0 tdm04b outbound call question
6:28AM 3 Multiple NIC's on * box?
6:21AM 1 "Distributed" registration SIP/IAX2
5:47AM 1 asterisk & xlite codecs
5:46AM 0 Several Problems with PhoneJack
5:45AM 1 Asterisk DNS issue
5:45AM 0 Problems in autnenticating with SER / PortaSIP
3:38AM 1 asterisk support for ISDN 1TR6 ?
1:56AM 6 Top posting
1:52AM 2 No SIP registration but user has dialled out?!?
1:51AM 0 TDM400P / FXO / Polarity Reversal
1:49AM 1 Grandstream BugeTone 101 - Multi-Server setup ???
1:06AM 0 Frequency Shift
Wednesday November 10 2004
11:48PM 3 No Inbound CallerID Name Has me Stumped.
11:35PM 0 Broadvoice Problems.-
9:25PM 2 Aastra/Sayson 480i eval
8:51PM 1 Connecting to Exicom GSX 418/816
6:31PM 3 Zaptel module load errors under stock Fedora Core 2 (2.6.8-1.521 kernel )
4:58PM 0 Sip Phone UIP200 Accepts calls but dialing out fails
4:49PM 1 DTMF and Access Codes
4:28PM 1 Callerid is recieved by fxo, but sometimes not passed to extensions
4:24PM 0 AgentCallBackLogin and accepting call using #
3:32PM 4 NoOp
3:15PM 1 Problem flashing zap channel.
3:02PM 1 Sending SMS from ISDN to cellular
2:17PM 0 Analog calls not working
1:57PM 1 No sound with kphone 4.05 on SuSE 8.2 and asterisk
1:29PM 1 Voicemail and MySQL 4.1.x
1:26PM 3 Hooking up a an Adit 600
12:58PM 1 Broadvoice Patch
12:54PM 5 Broadvoice asterisk patch
12:51PM 0 Problem adding zaprtc to Asterisk CVS on debian sarge
12:19PM 1 iconnect incoming problems
12:05PM 1 GTW V.92 modem work with asterisk?
10:51AM 0 IAXy Call Transfer and X100p audio quality in UK
10:25AM 7 xlite and asterisk
10:16AM 0 the asterisk work with modem generic?
10:04AM 0 Voicemail Outcall Notification App Ready to test
10:01AM 1 Unknown RTP Codec when sending fax
9:50AM 1 asterisk PC hardware reccomendations?
9:27AM 4 Pause during dial
8:12AM 0 SELinux and Asterisk
7:32AM 4 Asterisk, X-Lite, and * and # keys
7:10AM 0 Amount of time asterisk take to pickup incom ing call on ZAP interface
7:00AM 1 Web tool for Connection History
5:04AM 1 4 port ISDN BRI pci card
3:28AM 0 HELP: Asterisk becomes zombie process ...
3:18AM 1 Call failover and redundancy
2:31AM 2 maximum retries error
12:20AM 0 register problem of iaxcomm
12:16AM 1 remove channels
Tuesday November 9 2004
9:34PM 1 Asterisk-OH323 OUTCODEC
8:21PM 3 processing power / codecs
7:48PM 2 Auto dial Out
7:15PM 0 Queue Optional URL Problem
5:51PM 2 DISA() context restrictions
5:30PM 0 Problem with agentcallbacklogin and hitting # to accept call
4:49PM 0 TDM04B and T100P driver loading issue
4:03PM 5 Digium Generic Boards - Low Prices / High Quality.
3:36PM 0 Monitor on AsteriskĀ“s Manager API
3:21PM 1 External call initiation
3:02PM 1 linphone
2:20PM 5 E100P - Generic (Clone) - :)
2:13PM 1 Old Dialogic Hardware Questions
2:10PM 4 quasi-skype channel for Asterisk?
1:22PM 2 X100P CLONES again
1:16PM 3 Voicemail questions
1:04PM 0 Segmentation fault on SIP inbound
10:36AM 1 looking for BKW
10:07AM 0 Broken H323 channel
8:50AM 1 Enquiry about Wildcard E100P card
8:24AM 0 Queue Behavior.
8:01AM 2 New Release Asterisk-Stat V 1.3
7:56AM 1 Zaptel makefile error/bug?
7:05AM 2 Marconi Sys X/TE410P configuration
7:02AM 2 UK CID patch and version 1.0 CVS build
6:24AM 0 (no subject)
6:05AM 0 Intel IPP installation
5:41AM 3 UK BT Caller ID, X100P and Asterisk v1
5:16AM 1 Alcatel IP Phone
5:11AM 2 Costum ring tones with BT10x
4:47AM 0 DIALEDPEERNUMBER and Queues bugged?
4:23AM 0 how to detect busy tone?
2:21AM 0 Linksys / Cisco does not support the PAP2-NA
1:29AM 1 WRT54GP2 (WiFi + ATA)
12:50AM 0 How to connect Siemens Combiset to Asterisk - fxo or fxs ?
12:49AM 0 X100P, Caller Id and Ireland
12:44AM 0 Running Asterisk in chroot environment ?
Monday November 8 2004
11:38PM 3 Faxing issues (no VoIP involved)
10:56PM 3 NAT setup
10:53PM 2 Cisco Unity and Asterisk
10:24PM 1 Change log available?
7:38PM 0 FC3 and udev troubles
5:43PM 0 x100p drive use Tone-based Supervisory Disconnect?
5:21PM 1 bad quality for toll free calls with gafachi
4:48PM 0 IVR functionality Any Idea's how to implement this?
4:03PM 1 IAX2 One way audio PSTN via Gafachi
4:00PM 1 SpanDSP + Lexmark 6170 = Cut off faxes?
4:00PM 1 txfax problem?
3:47PM 1 IAX and ADSI Help
3:40PM 1 CallerID+Distinctive ring in Australia
3:31PM 0 RPMS for Fedora Core 2 now available
3:21PM 1 sip trunking works?
3:02PM 1 Polycom 600 as a Receptionist Phone
2:40PM 1 new RH9 install - no playback audio?
2:40PM 2 calls go silent
2:01PM 0 Xten Video Softphone Gets IM, Presence
1:59PM 3 how to get Stable 1.X via CVS
1:43PM 5 Same Extensions in Multiple contexts
1:08PM 1 iPeya iPHONE-1001M?
12:20PM 3 MWI Doesn't Turn Off
12:19PM 2 Configuring Asterisk As A Sip Server
12:19PM 2 Cordless vs Wireless phones
11:53AM 0 TDM400P card on Mac dialtone problem
11:41AM 2 Voicemail Macro issue.
11:11AM 0 timing and dropped calls
10:44AM 1 Sort of OT: Grandstream Phone and MS Wireless mouse
10:23AM 0 Snom 220 (or other phones) - line
10:23AM 0 Setting DND feature via access code
10:22AM 1 FW: Need a creative solution - Caller ID and a stupidupstream
10:22AM 0 FW: Need a creative solution - stop forwarding from changing caller ID
9:58AM 0 Error forwarding calls to Voicemail from SER
9:46AM 0 Zap FXO channel locked up with steadystatic( white noise)
8:49AM 0 ZyXEL 2000w unregistering and no audio
8:12AM 0 Free World Dialup via IAX2 gives duplicate calls?
6:54AM 0 Quintum vs Asterisk
5:46AM 2 Setting jitterbuffer in with iax
4:12AM 0 Help on "Supervised Call Transfer"
4:00AM 5 AGI Errors
2:06AM 1 re: CallerID for the UK
1:57AM 0 Cisco 1751-V SIP Gateway for Asterisk
1:51AM 0 Have anyone try to use asterisk as a business mode
1:34AM 1 Astricon Brazil. Why not ?!
1:14AM 0 Re: [Asterisk-Dev] Illegal Instruction (Solved)
Sunday November 7 2004
11:59PM 1 Aterisk and ISDN
11:24PM 0 Problem with call originating from Cisco
10:17PM 0 how to get CallerId info for call originated from manager API
10:00PM 1 New bounty for voicemail outcall notification- add $$ if interested
9:16PM 0 New bounty for voicemail outcall notification -add $$ if interested
8:59PM 3 Point to Point VOIP
8:29PM 1 Zap FXO channel locked up with steadystatic(white noise)
8:06PM 2 New bounty for voicemail outcall notification - add $$ if interested
7:36PM 1 openhours - include contexts based on time and date
6:17PM 2 Snom 220 (or other phones) - line apperances?
4:28PM 5 getting callerid from spa3k to asterisk
4:23PM 0 Cisco Unity + Asterisk
3:09PM 1 Zap FXO channel locked up with steady static (white noise)
2:08PM 0 Need help from the USB phone owners
1:31PM 3 CallerID Name from SIP to IAX2
1:30PM 4 "night" mode ideas
1:14PM 1 SMS through Cisco PSTN GW
12:17PM 3 Queue announce behavior for callback agents?
12:03PM 1 Forward incoming SIP calls to H323 ipphone?
11:53AM 2 Clipping at start of call
11:37AM 1 Unable to create channel of type Zap!
10:39AM 4 MAX TNT
9:20AM 3 No busy-tone
6:45AM 2 Siemens GSM terminal with Wildcard FXO
6:06AM 1 FreeBSD asterisk and zaptel versions
6:01AM 1 zaptel (ztdummt) compilation problems
5:39AM 1 CVS RPMs for Mandrake 10 (Zaptel and, Asterisk)
5:08AM 3 press # to execute
4:06AM 3 Problem with call originating from Cisco 7940 SIP phone to a SIP peer
1:15AM 0 ADTRAN 850 and T100P - need some help!
12:22AM 2 I don't know the name of this feature...
Saturday November 6 2004
11:45PM 0 how to establish a queue for external agents (was:Need a dial plan as follows)
8:23PM 5 SIP Groups
8:21PM 0 Adit 600 and T100P echo from VOIP clients
7:42PM 1 SIPURA does not register with Asterisk
6:56PM 1 Asterisk X100p can not hangup
6:04PM 4 Need a dial plan as follows
5:46PM 4 Enhanced Audio Support for EAGIs
5:38PM 1 Caller-id
5:33PM 2 Passwords in extra include file
4:30PM 1 fax and echo cancel
1:03PM 5 * does not listen to DTMF during wait ?
11:47AM 4 Polycom 500 software?
11:27AM 1 astGUIClient
7:59AM 0 Giving users the ability to break out of thequeueand go to voicemail
6:35AM 2 Setting up a Fritz AVM PCI card
6:28AM 0 Call Park Bug
3:16AM 0 group limit
1:42AM 1 missing wakeup gsm files
1:41AM 1 Giving users the ability to break out of thequeue and go to voicemail
12:11AM 1 Analog to Digital
Friday November 5 2004
10:59PM 1 SIP REGISTER -- Asterisk non-compliant or is it the provider?
10:56PM 2 Giving users the ability to break out of the queue and go to voicemail
8:34PM 4 [OT] Old Building Needs a New Telephone System
7:39PM 1 Grandstream BT100 Message Button
7:05PM 0 X100P Clone - Can't load moddule
5:59PM 0 MINNESOTA: TwinCities Asterisk Users Group.
5:32PM 1 R: sip.conf extensions.conf
4:21PM 4 Cisco 7970 & Firmware for the 7960G
3:06PM 1 chan_zap.c unable to register channel
2:58PM 0 Telephone Call Voicemail Notification
2:46PM 1 Record() help
2:45PM 0 & VOCAL & Asterisk
2:14PM 0 asterisk + hotel ?
1:58PM 2 Newbie X100P Clone question
1:54PM 0 Questions from an Asterisk newbie - follow-up question.
1:45PM 0 Audiocodes FXO MP104
1:45PM 0 weird problem with outgoing calls using chan_CAPI
1:41PM 0 Need a creative solution - Caller ID and a stupid upstream
1:27PM 0 Queue only allowing 1 call
1:16PM 2 res_config problems
12:59PM 1 Max retries exceeded with voiceconnect
12:55PM 2 Asterisk Brazillian Community
12:48PM 0 warning: implicit declaration of function `__use_ast_pthread_create_instead__'
12:15PM 3 BudgetTone 100 + NuFone
11:45AM 3 Questions from an Asterisk newbie
11:41AM 4 Adjusting txgain/rxgain
11:31AM 1 Are softphones usable?
11:01AM 0 Wrong return ext from call park?
10:53AM 1 Polycom IP 300 VoiceMail Retrieval
10:26AM 1 unable to create channel of type Zap
9:10AM 0 Asterisk As a Callback Server and Message Dialout Server - Can be Linked to ASTCC
9:05AM 2 VoiceMailMain(s<exten>@<context>) doesn't
9:03AM 3 wcfxs module doesn't load
8:50AM 1 Messanger 6.2 with Asterisk
8:47AM 3 sip.conf extensions.conf
7:59AM 2 Problems with voicemail
7:31AM 1 VoiceMailMain(s<exten>@<context>) doesn't work in CVS 11/03
7:05AM 0 Cisco 1751-V as SIP Gateway for Asterisk
6:31AM 1 german patches for say.c
6:07AM 0 Transcoding - when and when not?
4:49AM 2 New-B-ish Question
4:20AM 2 Service numbers
2:35AM 0 Sip Error Message, pbx.c: 1938
2:06AM 1 Asterisk incoming calls
12:42AM 1 voicemail&ilbc
12:04AM 0 Fw: Snom 190/220
Thursday November 4 2004
11:04PM 0 Problem In RTC Client With Asterisk
9:59PM 2 RIM Blackberry WLAN SIP phone
9:45PM 0 Using a Vonage Softphone
9:44PM 1 example Monit control file
6:08PM 0 oh323 0.7.0 don't start
5:36PM 1 ICD status
5:16PM 1 TDM400P and some problems
3:13PM 0 4-port T1 and TDM400 w/FXS in the same chassis problem
2:45PM 1 remote hold.
2:28PM 4 Looking for a SQL or ODBC Application
2:13PM 1 AstLinux posted for testing
1:51PM 0 RE: ZapTel problems ***** Problem solved *****
1:31PM 0 Asterisk Manager PHP Class
1:18PM 1 FW: ZapTel problems
1:12PM 1 7940/7960's 'talking' through speaker when in headset mode?
1:10PM 0 Light reading SIP webinar
1:08PM 2 T100P <-> Merlin Legend 100D not working
1:02PM 0 PHP AGI and system call weird behaviour
12:59PM 0 Alcatel Enterprise
12:40PM 1 Is it possible to use IAXY device to make 56Kmodem calls
12:35PM 1 Call Leg/Transaction Does Not Exist
12:30PM 2 NAT with Linksys
12:02PM 0 Remote MWI (I know it's possible)
11:55AM 1 Call Leg/Transaction Does Not Exist" back
11:53AM 3 Grandstream BT100 - Does not recognize DTMF
11:50AM 2 Is it possible to use IAXY device to make 56K modem calls
11:28AM 2 Passing callerID info to a forwarded line
11:07AM 1 Asterisk and ISDN HFC-S card (Biilion) instead of Fritz Capi ?
11:04AM 0 Grandstream BT100 - Failed to write frame
10:56AM 8 ATCC - Astcc-Admin.cgi File
10:35AM 0 OT: anyone using pointone?
10:12AM 1 Newbie question: forwarding call from PSTN to VoIP
10:11AM 3 system errors
9:39AM 3 Best Linux base for small Asterisk server?
9:36AM 0 avm fritz box fon
9:31AM 2 chan_capi patch : fax support
8:56AM 3 Limit DTMF tones
8:51AM 1 IAX --> SIP DTMF
8:49AM 0 Here's a tough question
8:22AM 1 Cisco 7910 - Success?
8:18AM 1 Multi-line analog phones with Asterisk?
8:10AM 8 Hardware Support
7:48AM 0 Asterisk 1.0.2-CVS RPM update
7:47AM 0 Perl AGIs & TCP Sockets
7:44AM 1 real-time-clock & asterisk/meetme/ztdummy in 2.6.9 UML
7:34AM 0 CISCO IP Conference Station
7:33AM 2 Multiline (4 or 8) sip phone
7:31AM 1 X100P & Analog PBX - not RING and not answer
7:15AM 0 Video conferencing Meet Me Bounty bumped
7:14AM 1 CVS-HEAD-11/03/04-14:09:34 ALERT_INFO Doesn't Get Passed
7:00AM 0 h323 & dundi problems with 11/04/04 CVS
7:00AM 1 supposable timing problem with TE100P
6:50AM 1 BROADVOICE fails to register
6:47AM 2 what do I ask my provider for when using e&m_w and a T100P?
5:56AM 1 res_config / realtime?
5:56AM 2 G.729 and Voicemail
5:17AM 1 control of calls
5:17AM 1 sipura 2000 flash ?
4:57AM 1 Howto correctly identify the telephone area code?
4:05AM 0 chan_capi on top of mISDN with HFC-8s
3:31AM 0 Voicemail, Cisco and H.323 problems
2:57AM 0 asterisk sip disabled error
2:01AM 0 Capi echo problems solved
1:38AM 3 Segmentation fault
1:14AM 0 SIP phones, Asterisk and bandwidth
12:58AM 3 Dynamic DNS causes problems
12:10AM 2 asterisk as sip proxy registrar
Wednesday November 3 2004
11:22PM 1 MusicOnhold on Bridged calls "plain text"
9:44PM 0 Little help here...
9:12PM 0 asterisk can not hangup .usrWildcard X100P
9:00PM 3 Voicemail Mailbox Configuration
8:54PM 0 SER-->Asterisk-->GNUGK Accounting Problem
7:45PM 0 Re: Re: [Serusers] asterisk can not hangup .user Wildcard X100P
6:50PM 1 Asterisk X100P doesnot Hangup
5:16PM 1 SIP registration/dialing problem.
4:01PM 1 Cisco 79XX - Using built-in 3way conference
3:40PM 2 How change default law for T100P
3:32PM 3 What do I need to ask my T1 supplier?
3:27PM 0 MusicOnhold on Bridged calls
2:59PM 0 Hookflash with cisco 827-4v
2:22PM 4 Sip clients not longer registering
2:12PM 0 G.729 for Asterisk: new version released
1:41PM 2 Automatically restart asterisk if not running
1:09PM 2 Dropped calls with analog lines using TDM400P
12:37PM 3 problem facing on Firewall, NAT and asterisk
12:11PM 1 SIPGate for outgoing calls
11:34AM 5 FireFly Problems
11:27AM 0 manager api originate doesn't give detailed information
11:19AM 0 RE: IAXys or IAX Softphones cannot call SIP phones
11:02AM 1 Installing X100P Asterisk - Unable to create channel of type 'Zap'
10:52AM 1 addon_mysql_cdr allows fraud by sip or iax users
10:44AM 1 Speed Dial / New Context
10:40AM 2 Asterisk's Fails to start!
10:34AM 0 ASTCC - cdrs database and number-entry timeout questions
10:33AM 1 asterisk port problem?
10:17AM 0 SendDTMFthrough the manager
10:06AM 0 can i call my local phone to IP phone or vice versa
9:53AM 0 Remote MWI
9:29AM 1 oh323 compilation error
9:11AM 3 Good ringing plans for small office
9:01AM 1 Call pickup and snom phones
8:43AM 0 Configuring MTA-V102 through TFTP, HTTP, HTTPS for Asterisk
8:30AM 3 zt hook failed: Device or resource busy
8:26AM 0 launching urls from queues
8:24AM 1 Maddog weighs in on the state of the Linux [Asterisk plug]
8:13AM 9 An anniversary and a lament for FXOs
8:08AM 0 No ISA tormenta card found at d0000
7:43AM 0 asterisk-oh323: New versions now available!
7:38AM 1 Latest CVS voicemail<->mysql problem
7:36AM 2 chan sip error
7:32AM 0 Voicemail: howto disable vm-intro.gsm at the endof message?
7:24AM 2 Voicemail: howto disable vm-intro.gsm at the end of message?
6:36AM 2 asterisk as a sip registrar and user accounts
6:21AM 1 H323 ISDN
5:57AM 2 Asterix-to-PBX
4:07AM 1 Console Error message
2:42AM 2 H323 Compilation
2:32AM 1 No outgoping calls with ISDN
2:15AM 0 Cisco 2600 Gatekeeper registrations
12:45AM 3 Reject a call if no callerID
12:43AM 2 Using T100P on E1 line
Tuesday November 2 2004
8:46PM 1 marginal voicemail prompt sound quality
6:03PM 0 reboot polycom via sip message
5:52PM 0 Dropping last digit when dialling from analogue phone.
5:37PM 1 dialing from mexico mapping numbers.
4:26PM 3 FXO devel Kit Card
4:21PM 2 gastman - documentation?
4:13PM 0 E&M timing
3:58PM 4 FXO module in TDM400P (UK, BT) - Hangup detection failing
3:05PM 0 agents can't hear callers.
2:43PM 1 anyone got a 7910 to work with asterisk?
2:14PM 1 IAX between two *
2:09PM 1 Notification of missed calls
2:00PM 1 The best SIP HW Phone and WLAN Phones for Asterisk
1:57PM 2 Outgoing call fails on pulse dial line
1:50PM 3 Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash
1:40PM 4 ISDN Dialplan
1:16PM 0 Asterisk Hanging!
12:48PM 2 Tone while ringing another IAX Phone
12:45PM 5 MAX TNT SIP / Asterisk
12:16PM 0 Remote Office question, Draytek , recommende d analog phone
12:11PM 0 isdn to isdn data call (bristuff'ed with hfc based card)
12:00PM 1 Fw: Re: How far is IAX to be a Standard
11:59AM 1 Polycom IP-500 Network Problems
11:27AM 0 Multi Freq signalling..key pulse...stop pulse....e&m.....
11:04AM 1 Anyone have bristuff's zaphfc module coexisting with wcfxs for the tdm400p?
10:29AM 3 Best codec for faxes?
10:08AM 1 Remote Office question, Draytek , recommended analog phone
9:53AM 3 FXO Module Error
9:52AM 2 Asterisk refuses to use anything but g729
9:51AM 7 Zaptel Issue in Fedora Core 2 test 3
9:23AM 0 Calling any Linksys PAP2-NA users...
9:15AM 1 (no subject)
9:12AM 1 Problems with CISCO, SIP and Asterisk
8:51AM 2 Broadvoice with multiple numbers
8:51AM 0 TDM11B auto configuration issue
8:49AM 4 Wireless VOIP Phone suggestions
8:21AM 0 RE: Question--Eezee phone?
8:06AM 3 Allied Telesyn Residential Gateway 613
7:44AM 0 g729 passthrough
7:32AM 1 Quintum Tenor DX
7:26AM 2 IAX2 audio problems but SIP OK?
7:20AM 1 FW: ASTCC with password
7:18AM 2 OpenSource Proxies ?.
7:12AM 0 iax bracking up
6:31AM 3 Speech to Text Conversion
6:22AM 0 Unable to include a context in another context for some unknown reason
6:16AM 3 Reading extensions from MySQL database
6:13AM 0 * Sunday News
5:45AM 1 Enhancing list quality?
4:22AM 1 Urgent handler
3:42AM 2 prioritising codecs per user?
3:02AM 2 ISDN Capi Drivers HELP
2:49AM 2 Unable to get our IP address, Skinny disabled
2:30AM 2 Motherboard compatibility with 6 PCI slots for TDM04B
2:17AM 0 DTMF from TE410P to SIP devices doesn't work
2:17AM 1 Codecs and echo
1:41AM 0 Asterisk terminating VoIP over 20 E1.
Monday November 1 2004
11:59PM 3 Hold music while ringing
11:12PM 1 soxmix?
8:29PM 1 MOH whilst waiting for Conference attendees
8:26PM 1 tdm410 driver prevents files being played
6:56PM 2 Is there a way to disable call wating?
5:54PM 1 astcc configure
4:57PM 0 anyuser i-3100
3:37PM 1 Queue Prioritization
2:55PM 0 'Unregistered Channel Type' when parsing zapata. conf on * startup
2:34PM 0 ASTCC - Anyone has a Dial Plan that is working?
2:30PM 6 calling an iaxy
2:21PM 2 Directory app and extension
2:16PM 0 Asterisk 1.0.2 changes
1:41PM 2 Asterisk & NetCentrex CCS integration
1:38PM 0 One way audio, h.323 cisco call manager
1:34PM 4 Centrex
1:29PM 1 Unable to write frame to channel: Success - MeetMe problem
12:43PM 1 Can anybody explain the meaning of these messages?
11:32AM 0 Call waiting does not work with g729 codec
10:09AM 0 Re: Voicemail with separate greetings based on extension
10:01AM 1 Problem with Cisco 7905 "Not Acceptable Here"
9:57AM 1 User problem
9:35AM 0 Passing a PIN in SIP Parameters
9:34AM 1 Re: loss concealment (Steve Kann)
9:05AM 0 Frequent dropped call on Wildcard E100P
9:02AM 0 MVP130
8:37AM 3 SIP via Wireless Ethernet Bridge and Double NAT
8:25AM 0 Weird problem with a Cisco call manager using the h.323 channel
8:14AM 2 H323 or SCCP?
8:08AM 4 Voicemail with separate greetings based on extension
7:33AM 4 adding an artificial delay to *
7:23AM 1 ChanSpy(scan) working, but not... any idea?
7:11AM 3 T100P Caller ID UK
7:01AM 2 field description /zaptel/zonedata.c
3:20AM 2 snom200 -> asterisk & dtmf (rfc2833)