asterisk users - Aug 2004

Tuesday August 31 2004
11:18PM 2 DeadAGI Application
10:27PM 0 Dial/Zap doesn't work
10:15PM 5 Line death not recognized on TDM400P?
8:32PM 1 install software version to mediatrix 1204 (how to)
7:13PM 3 All you polycom folks.....
7:09PM 1 T100P Configuration for Mixed Voice & Data
5:12PM 0 MP3Player strange error
5:01PM 3 Cisco 79XX SIP Ring Tones
4:32PM 2 Asterisk codecs and packet size
4:22PM 0 good Dutch TTS ?
4:06PM 0 Can only call asterisk once
2:16PM 1 Hardware suggestion
2:16PM 1 Why is it called 'Comedian Mail?
1:32PM 4 T100P No D-channels
1:22PM 0 Streaming an audio file to a Zap channel before answer
1:04PM 5 OT: Headset for Cisco 7960?
12:57PM 1 Going to voicemail instead of queue if no agent is logged in ?
11:14AM 4 IAX Client
11:12AM 3 Can asterisk detect BUSY signal?
11:10AM 0 detect telco voicemail stutter-tone
11:05AM 0 error: CDR on channel '<unknown>' has not started
10:52AM 1 Analog lines and TDM card
10:51AM 1 Losing voice on Digium demo server - how to spot problem ?
10:35AM 0 Snom Programmable button Mini Howto and ringstate patch
10:24AM 2 multiple lines with SIP like MGCP?
9:09AM 1 PSTN noob question
8:37AM 1 Polycom IP 300 - Displaying Only CallerNAME... What about NUMBER?
8:35AM 0 answer from wrong port
8:31AM 0 Can i send calling costs to a SIP IP phone display
8:26AM 0 Polycom SoundPoint... Gains -Which is for speakerphone
8:10AM 0 Polycom SoundPoint... Gains - Whichis for speakerphone
8:08AM 1 Polycom IP 300 - Displaying Only Caller NAME... What about NUMBER?
7:53AM 0 X100P Questions: Voicemail and Phone Port questions
7:29AM 0 newbie question about PBX Call Pickup
7:14AM 2 Harddisk noise on TE410P
7:07AM 4 which distro for asterisk?
7:05AM 2 limit the length of extensions
6:53AM 1 SIP registration with public dynamic ip address
6:18AM 0 BRI numbers
5:13AM 0 extensions => s,1,Dial(Zap/2/number) noise
4:42AM 3 pattern matching problems
4:14AM 0 Transfer from MOH to MOH doesn't work.
3:45AM 0 Polycom SoundPoint... Gains - Which isfor speakerphone
3:35AM 0 Transfer to queue
2:00AM 1 Do not get calldeflection (capiCD) to work.
1:17AM 0 transferring call to another line
Monday August 30 2004
11:10PM 4 Newbie - Voicemail Password Help
9:23PM 0 My Three-way calls work backwards
9:10PM 2 VoicePulse Connect DTMF with IAX2
8:43PM 1 Asterisk and Citrix
6:14PM 0 Reload crashes Asterisk ?
5:43PM 0 Re: New to Asterisk and a question
5:06PM 0 Re: New to Asterisk and a question
4:41PM 1 Polycom SoundPoint... Gains - Which is for speakerphone
4:29PM 0 Delays while playing a message
3:33PM 0 [Fwd: [Asterisk-Dev] Snom Programmable button Mini Howto and ring state patch]
3:07PM 1 AstriCon Reminder: Please register today
1:56PM 2 number of simultaneous calls with E&M
1:44PM 0 Asterisk with Sayson 480 ADSI
12:51PM 0 MWI Light On SoundPoint IP 300
12:16PM 0 SOLVED - Problems compiling zaptel driver
11:25AM 2 How record conversation to sound file ?
11:23AM 1 Snom Programmable button Mini Howto and ring state patch
11:23AM 0 Redirect SIP calls to the SIP provider
11:12AM 1 SIPJack
10:57AM 1 Faxing with an IAXy
10:17AM 0 Anyone used Tecom IP2004 SIP phone?
9:55AM 7 Polycom SoundPoint IP 300 Configuration
9:40AM 1 Problems with T100P card not releasing channels.
9:27AM 1 Does anyone have a working GR-303 config?
8:25AM 1 How does call routing actually work with SIP?
7:53AM 0 Parsing problem with oh323
7:36AM 1 Voiceronix and asterisk
6:31AM 0 New Error Log Messages
5:07AM 0 zaphfc success stories
5:06AM 2 Suitable for Dynamic IVR Platform?
5:01AM 0 AVM Fritz on Fedora 1
4:26AM 0 SIP and IAX Registration Problem with Dynamic IP
4:13AM 1 IAX.conf problem (NEWBIE ALERT!)
2:33AM 0 Problem with modified-Prepaid-Application
2:33AM 1 Voicetronix OpenLine4 immediately hangs up on every call
1:16AM 1 X100 and call duration
Sunday August 29 2004
11:30PM 2 Mix Data and SIP Phones
10:38PM 0 Help debugging voicemail problem
9:32PM 2 zaptel configuration
8:56PM 2 AgentCallbackLogin by other means
8:06PM 2 Still unacceptable echo on X101P
7:44PM 1 ${CONTEXT}
6:00PM 0 Asterisk and codecs?
5:34PM 1 Bridging audio in cmd_dial() before connect completes?
3:21PM 0 Static Problem (t100p - Channel Bank)
3:18PM 0 Asterisk H.323 channel...
3:06PM 5 Broadvoice BYOD Plans - 3-way and Call Waiting
2:58PM 7 SMS & Asterisk
2:57PM 0 Which Zaptel release goes with Asterisk-1.0-RC2 ???
2:44PM 3 Revert to dial tone?
2:39PM 0 System freezes when using Festival with usecache
2:15PM 0 SMS and asterisk
1:41PM 1 not getting ringing/busy/answer feedback on my PRI
1:08PM 0 Python and AGI
1:07PM 2 Jitter buffer
12:20PM 2 Servers
11:11AM 2 Sip device not login or register calls to that device go to busy voicemail not un-available
9:34AM 1 Empty Queues
2:51AM 0 Asterisk Assistants for Linux or Windoze???
2:34AM 0 IAXy died
2:10AM 1 Termination in Holland.
12:51AM 1 Mobile phone integration via bluetooth
Saturday August 28 2004
11:20PM 1 where can I find spandsp?
7:27PM 5 Distinctive ring detection problem
10:57AM 3 POE
9:23AM 10 Broadvoice problem
8:38AM 3 SIP Provider for Reseller
8:37AM 1 UK Disconnect supervision with TDM400P
8:24AM 1 asterisks and vonage
8:13AM 1 IAX dialing indication tone (PI = 8)
6:18AM 0 switch statement in extensions.conf
4:22AM 4 incomming call rejected using IAX2 with FWD
2:34AM 4 G729 licenses
1:31AM 0 ISDN BRI card exepriences in UK
1:07AM 0 FXO probs in Aus. Should I give up?
Friday August 27 2004
8:35PM 3 Disconnection From IAXTel
4:53PM 3 "7" Dialing gives a busy signal
4:40PM 2 Zap & ANSWER the Call
3:55PM 0 mysql-vm-routines and Directory app
2:38PM 5 IAXy Power in Australia?
2:12PM 1 does agi wait for digit work in a meetme room ?
11:59AM 1 Retrieve Info from Cisco Call Manager
11:58AM 5 iaxtel and jitterbuffer
11:42AM 0 regex and gotoif question
10:26AM 4 Speech Recognition and Asterisk
10:22AM 1 libr2
10:14AM 0 Asterisk Assistants Custom Icon
10:07AM 0 Re: how to fetch a call? (Tony Mountifield)
9:59AM 1 Help with a fax via Grandstream Handytone 286?
9:45AM 0 Asterisk & Max TNTs
9:32AM 2 Someone please try MeetMe MOH with latest CVS and GS phone
9:31AM 2 No audio on PRI channel answered by Playback() orMeetMe()
9:19AM 1 No audio on PRI channel answered by Playback() or MeetMe()
8:51AM 3 Can a Macro call another Macro ?
8:47AM 2 Are there any graphic designers on this list?
8:41AM 6 FXOs
8:34AM 1 IAX2 --> IAX2 confusion, it doesn't work...
8:31AM 0 questions and recommendations
8:28AM 0 Release 1.01 of FWD Assistant available (bugfix release)
8:24AM 1 Problem dialing out to Free World Dialup
8:04AM 0 Cisco 7940 SIP Firmware - Help.
7:58AM 0 auto-gain, or different gain between incoming and outgoing calls (EURO ISDN PRI) ?
7:55AM 1 xlite Problems
7:45AM 4 Queue Announcement not until after # accept call pressed
7:44AM 0 ACD ringall + roundrobin
7:42AM 1 Re: sip change? (Rich Adamson)
7:37AM 0 Cisco 7940 Sip Firmware
7:15AM 1 Problems dialing out with T100P and Adtran
7:14AM 3 sip change?
7:11AM 2 how to fetch a call?
7:06AM 0 h323 with Fedora 2 & GCC 3.3
7:06AM 0 Voicetronix Segmentation Fault
7:03AM 1 Asterisk compatible E1 cards
6:24AM 0 OT re: sip change?
6:16AM 1 Cisco 7940 - SCCP or SIP?
6:13AM 0 Queues - CallbackLoging Automaically?
6:05AM 2 Using regular expression in dialplan
5:29AM 0 Touch tone problem
3:15AM 2 FXO interfaces used in UK?
2:48AM 1 Can't flash 7960: P0S30200 .bin not found
2:32AM 0 Updated app_mysql.c, enabling use of INSERT and UPDATE
2:20AM 0 'set verbose 3' or other way to get '-vvv' level debugging out of running background asterisk?
1:46AM 0 Hangup() doesn't always when talking to Nortel Norstar over CT1 E &M wink-start trunk line?
1:28AM 3 Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits?
12:28AM 0 1stwave
Thursday August 26 2004
9:32PM 1 Hey admin: Do we have to have a 92-char reply-to header?
9:02PM 0 Newbie with IAX2
7:46PM 2 VoIP Telephony with Asterisk book
7:24PM 1 * and VoIP for LD
6:57PM 0 Problems with ISDN (NT-Mode) - Error Messages inside
5:36PM 0 Polycom SoundPoint IP 300 w/ SIP Software - Where Can I Get One?
5:13PM 1 Polycom non-IP phones
5:11PM 5 TDM400P Problems
4:14PM 4 PLC (Packet loss cancel) questions
3:29PM 2 Asterisk mysql database
2:47PM 0 Error in app_callingcard.c with database connection
2:35PM 1 Newbie needs help - Dev_Kit_Lite installation problem
12:49PM 1 Linux keeps deleting the ZAP files??
10:16AM 1 No signal from ISDN-phone connected to hfc card in NT-mode
10:00AM 0 ShoreTel (ShoreLine) branded phones work with Asterisk?
9:30AM 1 RC2 and VoicePulse
9:15AM 0 IPDialog call transfer
8:31AM 2 Sip Channel CLI
8:25AM 2 Sound card
8:22AM 1 Anyone using Asterisk on Slackware 9?
7:51AM 2 Asterisk+IVR functions trouble
7:42AM 0 ilbc asterisk and handytone/Xpro
7:37AM 1 Wil spandsp work with i4l driver?
7:15AM 2 Astricon hotel recommendations.....?
6:26AM 4 ISDN Card Recommendation
6:22AM 0 chan_oh323 build (resubmit w/ new title)
6:16AM 0 Can't make zaptel on red hat 9
6:04AM 1 GRSecurity and ALSA on a Gentoo Server
6:04AM 2 <<< AGI and EXEC function CONFIRMATION >>>
6:04AM 1 chan_oh323: __use_ast_pthread_create_instead __ (was: chan_oh323 loading error)
4:32AM 6 chan_capi module
4:25AM 0 Asterisk media problem behind NAT
2:49AM 0 "for Lack of RTP activity in 0 seconds"
2:38AM 0 TDM400P + 1FXS + 1FXO for sale
2:03AM 0 R: Grandstream Firmware
1:28AM 1 Grandstream Firmware
1:24AM 0 I have on isdn phone and one isdn-card (ast-pbx)attached (to my line)...
1:13AM 4 Codec
1:09AM 0 I have on isdn phone and one isdn-card (ast-pbx) attached (to my line)...
1:00AM 0 Out Dial Problem
Wednesday August 25 2004
8:21PM 0 Dial by name across 2 systems
8:13PM 0 Could use some advice/reality check from someone knowledgeable
7:25PM 1 home lab setup
6:15PM 0 Phone recommendations
6:15PM 3 IP Phone Recommendation ?
6:02PM 3 Fax detect
4:46PM 0 Error when closing asterisk
4:22PM 1 7960 Looses DHCP Lease when 7920 boots!?
4:13PM 0 New wiki pages: Avaya 4602 upgrade and configuration
3:47PM 0 spamdsp capacity
2:41PM 3 Distinctive Ring Cadences
1:54PM 0 unknown SMS Messages
1:51PM 0 chan_sccp with multi-lines and 7960's
1:38PM 2 Voip phones & headsets
1:32PM 1 Voicemail forwarding from SER & extensions.conf
1:24PM 2 asterisk & chan_sccp
12:16PM 1 chan_oh323: __use_ast_pthread_create_instead__ (was: chan_oh323 loading error)
11:38AM 1 Which end hungup?
11:11AM 5 Something broken in voicemail app??
11:09AM 0 Read and dtmf ?
10:42AM 2 Avaya dialing problems
10:12AM 1 2x HFC ISDN Cards - SuSE 9.1 - Problems with making calls
10:02AM 1 Sound deformation during conversation or IVR message
9:29AM 1 WTS: Just arrived Brand New Cisco IP phones
8:40AM 0 oh323_indicate: Ignoring PROGRESS indication
8:37AM 2 chan_oh323 and cdr
8:29AM 2 GrandStream HT-486 ATA as VoIP Gateway
8:28AM 0 *, Sipura, and cisco dtmf
7:57AM 3 chan_sccp2 & 7960 -- documentation and example request.
7:15AM 4 YAAN (Yet Another Asterisk Newbie)
7:08AM 7 TDM400P lockups (FXO)
6:52AM 0 KS-Like controlling
6:50AM 1 Asterisk PBX and backup Circuits
6:48AM 1 External BRI - is there such thing?
6:35AM 1 Problem of set up asterisk-1.0-RC2.tar.gz with asterisk-prepaid-0.3.1
6:25AM 0 How to be taken out of the queue?
6:02AM 0 Incorrect sound
5:44AM 0 ericsson drg22 - will not connect to asterisk
5:17AM 0 Asterisks
4:23AM 2 Advice on BT ISDN Services (UK)
3:58AM 0 freebsd 4.10 and port misc/zaptel
3:56AM 1 Individual call-forwarding on ISDN
3:37AM 0 sip history - which file ?
3:10AM 0 Loading module
3:07AM 4 GSM to BRI ISDN Gateway
2:35AM 3 Blocking a channel on T1
2:34AM 2 spandsp and certain (e.g. Canon) fax machines
1:46AM 2 Parallel T1 cable?
1:06AM 3 Compressing a dialplan
12:01AM 1 Closing bug reports without fixing the repor ted problem
Tuesday August 24 2004
11:12PM 0 asterisk directory service
11:01PM 0 Closing bug reports without fixing the reported problem
10:02PM 3 desparate for help DEV LITE KIT
9:18PM 0 RTP and NAT
8:24PM 2 Voicepulse incoming / dial extension
7:59PM 0 Perl AGI - no output from agi script to Aste risk
7:23PM 1 Zaptel/Zapata and SIP relationship
6:08PM 0 How can i configure extensions.conf.
5:20PM 1 Can I have same numbers in different contexts ?
4:11PM 3 Hardware for PBX with 4 incoming/outgoing lines and 20 phones
3:56PM 2 Grandstream Budgetone BT-101 and VoipJet
3:20PM 1 [Asterisk Users] Help with SIP Hosted Billing Service
3:07PM 2 Parking and Extensions
2:55PM 2 Remotely change call forward
2:49PM 1 RC2 and Netmeeting 3.01 ?
2:31PM 3 Asterisk with Adit 600
2:09PM 0 off list
1:45PM 0 Using Lucent/Avaya 64XX sets with asterisk
1:04PM 1 X100P connected to Meridan-1 system will not disconnect call
1:00PM 7 SMP Performance
12:46PM 0 Monitor() hangs
12:21PM 0 Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs
12:06PM 1 CPU utilization
12:04PM 1 MailMan slowness...
12:01PM 0 R X100P connected to Meridan-1 system will not disconnect call
11:33AM 0 Astricon - call for help
11:12AM 1 CVS RPMs for Mandrake 10 (Zaptel and Asterisk)
11:10AM 3 ex-girlfriend logic not working in latest CVS?
10:52AM 0 Warning when I use iax2 for inbound and outbound calls
10:49AM 2 Voicemail & "Couldn't read username" error
10:42AM 0 R X100P connected to Meridan-1 system will notdisconnect call
10:27AM 2 call queue help
10:16AM 1 [Asterisk-Dev] Asterisks
10:10AM 1 Autoattend detecting same digit twice
10:09AM 1 Swissvoice IP10S and RTP Port Operation
9:19AM 3 Asterisk to Vonage
8:32AM 0 problems with the mailing list??
8:31AM 2 SIP Provider in India/Pakistan/Bengladesh
7:52AM 0 MGCP 1.0 NCS 1.0 on a motorola SBV4200
7:50AM 0 Problem with sound on Wildcard TE410P
7:11AM 0 Sangoma Card Support
7:07AM 3 Bell Canada Caller-ID
6:49AM 0 If you run Archlinux...
5:50AM 0 h323 direct calls
4:56AM 1 hfc-s card, brii-stuff.0.1.0-RC4a, zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.
3:17AM 3 Inband DTMF is not supported on codec G.711 u-law. Use RFC2833
1:00AM 0 g729 Codec on Max OS X
Monday August 23 2004
11:33PM 1 strange problem PBX-Asterisk
10:34PM 4 Telemarketer screening
9:53PM 1 H323 outgoing calls
9:10PM 0 spandsp make error mmx.h
5:25PM 6 Dell PowerEdge 750 rackmount
4:58PM 2 Hold the phone!
4:20PM 3 newb question regarding DTMF
4:20PM 2 VoicePluse DID problem
4:12PM 4 Asterisk WITH Swyx... Any Idea?
3:04PM 0 multiple X100P config?
2:38PM 1 Asterisk <------- Quintum SIP Registration
2:00PM 3 Cisco 7960G, Skinny.conf, and reboots
1:32PM 1 Firmware Update IAXy
12:14PM 6 Asterisk Install on Kernel 2.6.x
10:45AM 2 Queue Monitor
10:30AM 0 Guide to getting started with Asterisk on MacOSX
9:40AM 1 (no subject)
9:31AM 0 MGCP and dialing out
7:55AM 5 SIP "unphones"
7:52AM 0 Swissvoice MGCP Error 502
7:15AM 2 HFC-S in NT mode, wiring?
7:07AM 2 Cisco 7940 Question
6:38AM 1 routing telephone calls via "switchboard/asterisk".
6:00AM 1 zaptel installation
5:32AM 2 Question about dial out via Zap
4:38AM 2 Adit 600 FXO & FXS
3:54AM 0 How to recover the problem with pgsql and asterisk
3:15AM 1 using ChanIsAvail
2:25AM 1 Choosing between TE405P and TE410P
2:17AM 6 2 servers
2:03AM 1 Problem with asterisk and postgresql
1:05AM 0 Bug in recording uavarible
12:10AM 1 Problem with mysql and with asterisk
Sunday August 22 2004
9:36PM 3 zap show channels - no such command
8:02PM 1 Queue Calls without using the
7:07PM 1 Pulse dialed digit recognization
6:44PM 3 SIP Phone recommendation for Receptionist
4:58PM 1 MusicOnHold problem
10:59AM 4 Error compiling meetme2
9:24AM 3 asterisk T100P to Merlin Legend
9:11AM 5 skinny or sccp?
8:46AM 0 We have thousands of U ISDN interface phones
8:40AM 0 Release 1.0 of FWD Assistant for MacOSX now available
8:23AM 0 tsu 120
7:27AM 0 app_mp3 with bri-stuff.0.1.0RC4a does not work
4:35AM 1 Spandsp - offline
Saturday August 21 2004
5:08PM 1 IAX2 DTMF not recognized - Bug report - Help sought
4:59PM 0 Cisco IP Phone- disjoin conference
3:42PM 1 need help with zaptel.
2:10PM 1 ActXPhone Active X control link is dead - has anyone cached files ?
1:53PM 1 Number and name for SIP extension at the same time ?
1:49PM 0 Can I place calls to arbitrary sip uri address from windows messenger 5 ?
1:19PM 0 cmd Monitor creating sound notification on channel
11:47AM 5 Asterisk and software Raid
10:37AM 0 How to minimally configure modules.conf loading?
10:36AM 0 autocreatepeer and sip peer options
10:15AM 1 Zultys 4x4 or 4x5
7:03AM 3 zaptel config
6:01AM 0 welltech fxo and *
4:42AM 2 system reboot often?
4:33AM 0 Asterisk and AVM FritzBox Fon (Germany)
12:49AM 1 just-added second X100P
12:07AM 0 hangup question
Friday August 20 2004
9:57PM 1 PoE injectors
9:09PM 0 spandsp tiff debian question
7:00PM 1 Sipura partners with Linksys for new combo router/SIP ATA
5:17PM 1 Forwarding PSTN to PSTN?
4:35PM 1 Adding macros causes ringing to fail
4:14PM 3 Strange problem with Dial
4:07PM 1 Testing a channel's status
3:31PM 3 BT Communicator (SIP???) and Asterisk
3:23PM 0 Max retries exceeded on call - seqno 102 (critical request)
3:06PM 0 snom 200 and * question
2:22PM 0 Invalid module format
1:34PM 1 TE410P - ZT_CHANCONFIG failed
1:20PM 0 Is posissble TE405P ?
1:00PM 4 Help with upgrading 7960 SCCP to SIP
12:38PM 1 Silence on incoming monitored calls
12:28PM 3 determining what number was dialed?
12:23PM 1 CDR problems with MySQL
12:08PM 0 chan_h323 doesn't pass audio before call is answered
11:34AM 1 Incoming MSN via ZapHFC -> to SIP
11:11AM 0 Oh323 installing problem make[1]: *** [asteriskaudio.o] Error 1
10:19AM 7 how to collect user entered digits
9:47AM 0 Stream File (AGI) question
9:32AM 2 Handling invalid extensions
9:09AM 6 Asterisk PBX Functions via SIP phone
6:52AM 2 tT funktions
6:44AM 0 Re: problem in with mysql modules.conf to load (Lerale Erwan)
6:39AM 1 dual servers
6:03AM 2 Creating 79xx Configs
5:43AM 6 Sipura endpoints
5:19AM 1 Finding operator from ISDN signalling?
5:14AM 0 Zaptel Problem after Upgrade
3:42AM 0 IAXY S100I noise
3:42AM 2 Multi-bitrate codecs
3:35AM 1 x100p won't answer
3:02AM 0 Problem with asterisk and pgsql
2:55AM 2 problem in with mysql modules.conf to load
2:39AM 4 telnet and Root
12:18AM 0 Operator-type phone
12:03AM 1 from Newcomtech Co,. Ltd Help us.
Thursday August 19 2004
10:28PM 2 Atick Certification on FXO Modules (Australia)
9:21PM 7 Where to purchase ISDN (BRI) cards in Australia (preferably)
6:13PM 2 IAX2 Port strangeness
4:42PM 4 Request for help designing an unusual * application
4:15PM 0 ATA with Built-in Switch
3:56PM 1 AGI Script: calleridnamelookup.agi
3:13PM 3 GrandStream BT101 Attended Transfers
3:11PM 6 How to run different codecs between the same endpoints on an IAX trunk?
3:10PM 2 residential sip phone
3:03PM 1 Received packet with bad UDP checksum
2:59PM 1 More on Broadvox
2:32PM 1 Inband announcement of parking slot from app _parkandannounce?
2:29PM 1 Granstream BT100 Rings Once and Waits for Call Pickup?
1:43PM 2 False Hangups on Asterisk
12:53PM 4 Does Granstream BT100 Conference Button Work?
12:12PM 1 Pingtel registration failing
11:51AM 1 Isdn4Linux and DTMF
11:38AM 0 UserAgent support for MGCP
11:30AM 2 and SpandDSP
11:05AM 1 Problems loading chan_h323 on Opteron 64 bit
10:47AM 0 Andre Bierwirth's ring state patches for SNOM 200 programable buttons
10:26AM 2 Dial from AGI [MSG]
10:02AM 1 Debit/Credit Card Terminals
10:00AM 1 No Success with SwissVoice.
9:25AM 1 SpanDSP/RxFax help...
9:07AM 7 Can PSTN CallerID be fowarded to a SIP phone extension?
8:47AM 2 T100P PCI or PCI-x
6:26AM 3 Echo SIP-T100P-PRI
6:14AM 0 fax output from Asterisk into file
5:33AM 0 Using Asterisk as a voicemail only with legacy switch
5:02AM 0 SIP reinvite code negotiation
4:59AM 2 Floating point exception help
4:40AM 1 chan_phone - compilation problem
4:02AM 2 Multiple SIP phones ringing for same extension
3:57AM 0 Error message "phone_mini_packet Read returned -1" and strange Internet PhoneCard behavior
1:42AM 0 Asterisk & SER
12:42AM 1 not yet a new user, some questions
12:02AM 1 Festival Issues
Wednesday August 18 2004
7:37PM 1 Not a User Yet - but have some questions
6:55PM 7 CID on internal extensions
5:17PM 1 Another small suggestion patch
4:56PM 0 SIPp and asterisk question
4:31PM 1 Three tdm400p's (loaded with FXOs)
4:28PM 2 Help! Cisco 7960 - dialing any digit makes the phone go nuts!
4:00PM 0 static noise TDM400P+T100P+Adtran Channel bank(TA750)
3:31PM 0 Automated FWD configuration of Asterisk on OSX
2:35PM 1 Small patch to zaptel Makefile
2:00PM 1 PCI Express and Digium Cards
1:34PM 0 SIP/IAX2 mysql auth + FreeBSD
1:29PM 1 service zaptel start
12:38PM 1 Restart Digium Cards
12:34PM 0 my iaxy makes a strange noise...
12:14PM 1 Newbie physical layout question
11:46AM 1 RE: New $85 VOIP Phone
11:27AM 2 Problem compiling zaphfc
9:56AM 1 Pingtel and some chinese company
9:00AM 1 paging/intercom
8:58AM 1 Config for sipgate?
8:47AM 1 Mpg123 clarification
7:56AM 1 Testing null values: ast_yyerror(): syntax error
7:14AM 1 Choppiness/Ticking sounds over LAN
7:05AM 1 Asterisk as SMS Service Center
7:00AM 3 How to make RTP Packets NOT passing thru Asterisk?
6:49AM 1 Asterisk and Dial-Up ISP
6:44AM 3 [OT] What's changing /etc/hosts?
6:44AM 3 call-back example
6:22AM 0 Network Crashed when we try to start TDMoE
5:43AM 1 asterisk start
5:38AM 0 List-etiquette * AGAIN *
4:14AM 1 Channel bank for asterisk
3:43AM 0 Help Needed on these doubts
3:23AM 1 DID Terminations
2:58AM 27 SpanDSP
2:46AM 0 Adtran power consumption
2:43AM 1 Audio problem
2:17AM 0 [OT] RE: SIP / IAX provider in the Netherlands.
1:58AM 1 SIP / IAX provider in the Netherlands.
1:54AM 1 pickup any call
1:45AM 3 How to accept the call and without billing the caller?
1:16AM 1 How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work
12:05AM 2 Festival Installation - Asterisk 1.0-RC2 && Debian Woody
12:01AM 1 Hangups - SIGFPE in dsp.c
Tuesday August 17 2004
11:58PM 1 SIP providers USA
11:05PM 1 Commercial deployments of Asterix
10:12PM 1 Asterisk and MEGACO
10:06PM 3 Digium Hardware Question from Newbie
8:44PM 1 Internal extensions giving weird static-like dialtone
8:09PM 0 1.0 RC2 - FXS issues - no ring, no MWI
7:24PM 3 [probably OT] wireless voip converter
6:24PM 1 budgetone 101 and buttons
6:00PM 4 asterisk and qsig
4:44PM 0 zaphfc in mode TE can't dialout (dialin is OK)
3:47PM 0 multiple sound cards - howto have IP phone on each ?
3:03PM 6 dialplan woes
2:53PM 0 Fax Detect over Analog
12:50PM 0 SpanDSP RxFax receives junk - gets Fax3Decode2d: Warning, (FakeInput): .... messages
12:44PM 1 Dialplan problem - incoming calls get MOH, not ringing.
12:00PM 4 Hunt Groups
11:37AM 6 couple basic questions
11:08AM 2 Inter-digit timers on t100
9:52AM 1 BroadVOX
9:00AM 1 CPC on Zaptel
8:44AM 3 asterisk-wide variables
7:57AM 0 RE: RE: dialing out
7:19AM 2 Inbound IAX2 calls has no music on hold
6:33AM 1 Faxing over ulaw
6:28AM 0 Compiling Zaptel under Bering 1.2
6:26AM 0 RE: dialing out
5:19AM 0 (no subject)
4:34AM 1 Cisco 7.2 firmware for SIP 7940/7960 release d
3:36AM 0 has no CRC! error messages while compiling zaptel
2:57AM 0 spandsp + files
1:06AM 2 Problems with DTMF
12:52AM 0 TDMoE crash the Asterisk and the Network
12:44AM 0 dropouts
Monday August 16 2004
11:28PM 2 Cisco 7.2 firmware for SIP 7940/7960 released
11:21PM 0 re: asterisk as VM for SER
11:14PM 0 (no subject)
10:49PM 0 RC2 - RH9
9:36PM 0 Re: using mysql and asterisk
7:22PM 0 Answer a specific channel
7:13PM 2 taking asterisk out of nat?
5:38PM 0 HELP with HFC Card, please
5:28PM 0 SpanDSP - Training failed error / timing problem
5:22PM 3 Digium TE410P and RedHat Enterprise Server 3.0
5:16PM 1 RC2: Where is parking.conf and MeetMe app ?
5:12PM 0 mysql version of Directory app
5:07PM 0 How to run qozap with wcfxo and wsfxs together?
4:40PM 0 Music on hold question..
4:11PM 1 Soft DSS for Asterisk
4:04PM 1 * and answering machine
3:07PM 1 Performance testing of asterisk
2:46PM 2 Problem compiling chan_sccp
2:39PM 4 Polycom SoundPoint IP 500/600 XML minibrowser
2:21PM 1 Set-up for 2 Asterisk System
1:25PM 3 DID Questions
1:13PM 3 Problems compiling chan_capi-0.3.5
12:57PM 4 Avaya firmware
10:43AM 2 1.0 RC2 External AGI Issues
9:29AM 1 Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk)
8:37AM 3 Formatting in sip.conf...can you have 2 @ signs for register?
8:04AM 2 dialing out and ringing issue
7:15AM 1 local echo using SPA-3000 as FXO port
6:25AM 3 What is the best Linux for asterisk
6:21AM 2 randomize Dial() target
5:56AM 0 DTMoE crash the Asterisk and the Network
5:23AM 1 no hangup
4:41AM 1 world incoming calls
4:07AM 2 disable console channels
3:29AM 2 Call stealing
3:29AM 1 Using Asterisk in home environment - room with speakers&mic as IP phone ?
2:35AM 1 Is "Meetme" a generic term?
12:54AM 0 down?
12:43AM 0 Help: SMS in UK
12:25AM 0 Differences in CDR files
Sunday August 15 2004
11:29PM 2 consultative transfer with zaptel
11:00PM 0 Modified Prepaid doesn't update the balance
7:52PM 0 how can i config a Cisco IAD 2430 config as a sip client
5:35PM 3 123 Basic configuration files
5:21PM 1 Newbie with missing .conf files
3:57PM 1 Teliax TOS copied from Vonage?
2:14PM 1 no tones detected
1:53PM 2 __use_ast_pthread_create_instead__
1:50PM 0 Internal Distinctive Ringing + Caller ID
12:10PM 0 asterisk with InPhonex?
10:57AM 3 Asterisk MIBS
10:52AM 2 GrandStream ATA286 & RC2 (was RC2 - H323 channel broken)
10:40AM 5 New $89 VOIP phone
6:36AM 0 Sip to Sip Calls via Asterisk
4:17AM 3 Vlan question
4:02AM 7 chan_oh323 loading error
1:56AM 0 Do you speak Czech?
1:29AM 1 Inbound Free World Dialup - extension not ringing?
Saturday August 14 2004
8:34PM 3 7960 help
7:25PM 0 Questions on various and sundry IP phones, and cabling
4:50PM 7 Free MOH MP3
1:08PM 7 Help - is voip good for in-house calls?
12:03PM 2 List traffic/Software
10:22AM 1 Howto remove digits from a called number
Friday August 13 2004
3:53PM 0 HELP: BYE-request not sent to SIP-peer
3:18PM 1 [Q] DIDs
2:09PM 1 Using a TE405P to connect to an existing PBX
2:00PM 0 Cisco Router as Gateway for FXO Ports
1:33PM 2 Static on outgoing calls using either X100P or TDM400P
12:52PM 1 Asterisk and softswitch
11:21AM 3 voice choppy
10:59AM 1 Euro isdn caller id problem
10:15AM 6 Dial command problems
10:14AM 0 Broadvoice User hung up on voicemail
9:48AM 3 External MW Lamp On/Off
9:46AM 2 How to detect answering machine
9:43AM 3 Cisco 79xx series IP phones
9:34AM 0 *** Asterisk Summer News: Forget numbers, dial by domain!
9:14AM 0 SIP<->H323 "Failed to create smoother"
8:58AM 1 Problem with ougoing Zap calls
8:45AM 3 Will this ISDN card work for me?
8:44AM 1 voicemail email messages
8:10AM 0 asterisk & gnugk & voip phone
6:34AM 0 MGCP & RFC3149 & Swissvoice IP10
6:25AM 1 SIP <->h.323
6:07AM 1 SpanDSP - Training failed (convergence failed) error
5:45AM 1 small asterisk system
5:06AM 2 te410p and Telstra Onramp 10
4:58AM 1 Interop RTP "Extension headers" for QOS?
4:46AM 1 OH.323 Dialout Problem
4:35AM 11 asterisk in india
3:58AM 0 Need help for GUI Client
3:58AM 0 Need help for GUI Clieny
3:49AM 0 Problem with grandstream devices and DTMF signalling [RESOLVED]
3:34AM 2 Lost 7960 time display on upgrade
3:32AM 1 queue name too long when sending sms over 32 chars
3:12AM 2 not hangup
2:43AM 1 Re: Send DTMF tone Like 'C' on connected call
2:41AM 1 Problem with grandstream devices and DTMF signalling
2:10AM 0 * -> Lucent PBX
2:07AM 0 ZAPHFC for Euro-ISDN
1:17AM 1 RC2 - H323 channel broken
1:15AM 0 incomingcall braking all
Thursday August 12 2004
10:27PM 0 Grandstream HandyTone-486 FXO port
4:59PM 0 Blind Call Transfer using Sipura 3000 + aste risk
4:11PM 10 H323 problems
3:40PM 0 Calling Card App
3:19PM 2 outgoing ZAP cannot connect using E1 isdn
2:35PM 5 Question about TE405P
2:14PM 9 Asterisk and SER
12:43PM 0 Zip2 configuration via tftp?
12:40PM 0 Default endian for signed linear
11:41AM 0 Enterprise users of asterisk?
11:40AM 2 Interruptable SayUnixTime
11:36AM 0 Message lamp integration with legacy pbx -- revisited
11:33AM 2 X100P a winmodem?
11:06AM 0 Voicemail.conf not being re-read / Updates (e.g. pin numbers) ok.
9:08AM 1 Problem installing Software Fax SpanDSP support into Asterisk
9:08AM 0 New office hardware set up question.
8:53AM 1 CCM <->(H323) <-> *
8:50AM 5 ::::: Pssst. Rc2! :::::::
8:37AM 0 TDM400P and Adtran 616 problems
8:23AM 2 How Many Calls On This Config
8:08AM 1 Re: Asterisk-Users digest, Vol 1 #4901 - 10 msgs
7:51AM 2 caller id over iax
7:49AM 1 AgentLogin issue
6:50AM 1 zaptel wont compile
5:45AM 0 RES: Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted
3:40AM 1 BRI and E1 in same system
3:21AM 4 Problems receiving SIP calls
2:47AM 9 Convert Cisco 7960 to sip
2:18AM 3 Pulse dialing...
1:36AM 0 AW: Starting with digium cards
Wednesday August 11 2004
9:21PM 0 X-Lite behind NATed ADSL router
8:15PM 1 Starting with digium cards
7:19PM 1 BroadVoice Voicemail
6:40PM 0 Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted (syslog)
6:39PM 0 Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted (tethereal)
5:17PM 1 persistant SABME
4:56PM 7 H323 call dropped when answered
4:40PM 2 Asterisk & (aka Talk(n))
4:32PM 1 Asterisk & Recommended Distro
3:51PM 1 Does IConnectHere still work with asterisk?
3:19PM 2 StanaPhone and Asterisks
1:10PM 0 Inband announcement of parking slot from app_parkandannounce?
12:46PM 0 Problem with Audio Play out in Voice-mail
12:05PM 1 Avaya IP Phones and Asterisk
12:00PM 1 Blind Call Transfer using Sipura 3000 + asterisk
11:53AM 2 Avaya and Asterisk
11:39AM 1 Grandstream Budgetone-102 client cannot register
11:32AM 0 Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted
10:43AM 0 Fedora Core 2 (kernel 2.6.5), CAPI and Fritz PCI
10:24AM 2 a few question about asterisk
10:21AM 1 CallerID Debug On Zap/POTS Channel
9:47AM 2 Autoattendant Configuration
8:39AM 3 X100P outbound only (Don't answer)
8:03AM 5 Asterisk and SMP
7:45AM 1 limit incoming calls to sip extens
7:21AM 0 asterisk -r and -rx questions
7:17AM 1 number unavailable
6:16AM 4 Waiting till picks up Zap line
5:25AM 0 zap channels and sip channels problem
4:34AM 4 zaphfc problems...
4:22AM 1 Ringing() doesn't play sound while phone is ringing
4:17AM 1 Analog Phones with Status Light Indicators
4:09AM 7 Static on outgoing calls (Quad E1)
3:30AM 1 stun and only one external ip
3:00AM 2 2.4.x-SMP vs. 2.6.x-SMP
2:48AM 1 external mailbox
1:51AM 1 is gatekeeper required?
1:33AM 0 Zaptel Dial Out Issues
1:11AM 0 No cdr entries for calls coming via chan_oh323
1:04AM 1 RxFax - tiff file problem
12:49AM 1 can sip users login two times?
Tuesday August 10 2004
8:41PM 4 Cisco 12sp+ and 30VIP
7:37PM 0 codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes from the end)
5:46PM 0 Re: [Asterisk-Dev] VoIP SPAM, what's next ?
5:44PM 0 Personal Meetme conferences; is there a better way to do this?
5:17PM 0 CVS version tags
3:57PM 1 SIP Transfers (Possibly reinvite)
3:41PM 2 Kernel 2.6 and zaptel data
3:24PM 0 UK SMS troubles
3:08PM 1 Parked Call Viewer Application - Freeware
2:26PM 2 SNOM 200 and Asterisk Woes
1:33PM 3 Polycom IP 500 - MWI Not Working
12:30PM 3 Semi-OT: Splitting a PRI into two PRI's?
11:29AM 0 iconnect inbound - FIXED (kinda)
11:13AM 2 Re: VoIP SPAM, what's next ?
10:35AM 0 Answer Call Waiting from Call Forward to Cell Phone
10:00AM 0 Intriguing * problem with voicemail signalling
9:22AM 5 Blocking the 'Do Not Call" List
8:59AM 2 Compile error H323
8:30AM 3 agent login
7:59AM 1 HowTo test asterisk in internal network?
7:55AM 4 Asterisk in a DMZ
7:54AM 1 DTMF issues
7:28AM 0 Can Incomming CallerID be fowarded to a SIP phone extension?
7:14AM 2 WiFi phone radiation regulation?
7:05AM 1 [OT]Google and the Asterisk list
6:37AM 0 TxtCIDName returning mangled text
6:36AM 0 distinguish rining tone
6:30AM 0 Config for Asterisk with Sipgate behind Linksys Router
6:29AM 0 T400P modprobe/ztcfg failure
5:35AM 4 asterisk mirror
3:59AM 0 Sjphone Troubles :
3:27AM 0 Sound problems ..linux red hat
3:19AM 0 RTP-problems with
3:18AM 1 Firefly and *... Argh!
2:26AM 5 just a few newbie questions
2:12AM 11 CAPI call transfer
1:51AM 0 help me in voice jittering problem
1:29AM 6 Problem with EuroISDN E1
1:09AM 0 h.323 channel problem: I hear nothing
1:00AM 1 AVM B1, chan_capi, Kernel 2.6
Monday August 9 2004
11:37PM 1 TE410P-RED Alarm
10:37PM 2 831 Santa Cruz/Watsoncille, Calif. DIDs
10:15PM 0 introduced Agents and * stops answering calls
4:23PM 2 CVS download
3:41PM 1 Inbound Call Errors...
2:58PM 2 Snom Intercom
2:42PM 2 Application asterisk uses obsolete OSS audio interface
2:41PM 1 called and callers buttons on bt100
1:52PM 0 H323 under asterisk RC1 ?
1:33PM 1 quadBRI + FAX
1:15PM 3 AbsoluteTimeout Inside A Macro
1:06PM 0 uniden phones
12:12PM 0 inbound/outbound trunk groups
12:07PM 3 Fedora FC2 and Zaptel (Torisa)
11:51AM 0 Strange H323 problem
9:50AM 2 Call File Routing
9:18AM 5 Questionaire :
8:24AM 2 ChangeMonitor syntax
7:51AM 0 New feature request
6:26AM 1 Click to Call
6:25AM 1 asterisk with H.323 phone
6:09AM 0 RC1 - callparking
5:57AM 0 sip endpoint not ringing
5:47AM 1 How do folks handle NAT routing?
4:42AM 0 Which is PCI HDLC Chip used for TE405P / TE410P / E100P ?
4:37AM 0 Some Errors on Asterisk server
4:31AM 0 Some Errors on Asterisk
4:26AM 0 Variables In a Context
4:25AM 1 h323 direkt call instead over GK
4:07AM 2 Sound file quality
3:18AM 1 Some Error on Asterisk....
2:51AM 0 traffic termination around the globe?
1:56AM 0
1:06AM 0 FW: problems with asterisk and the IAX protocol
Sunday August 8 2004
11:36PM 1 Howto configure TE410P card and channels
10:29PM 0 Extension Status on different phone
10:00PM 0 Straight to VoiceMail
8:52PM 3 iconnect inbound - so do we know how to fix it
6:58PM 2 pbx answers after answering from analog phone
3:17PM 0 Douglas Telecom & Asterisk
1:28PM 0 stable-RC1
12:44PM 1 asterisk-update script - and the script - Fixed typo
12:12PM 1 asterisk-update script - and the script
12:04PM 2 asterisk-update script
11:26AM 0 Zaptel.conf and Zapata.conf for TDM12B
11:14AM 1 truncated extensions
8:28AM 6 Voicepulse problems?
6:29AM 1 No Sound and Jungle:
3:03AM 2 System Reqirements HELP
Saturday August 7 2004
11:02PM 0 SOLVED: 100% cpu usage causes big problems
9:53PM 1 WARNING[1264581056]
9:39PM 0 voice mail greeting not updating?
6:07PM 2 Asterisk : No Sound No Dial
5:41PM 4 Generic X100P setup issues
5:04PM 1 Mediatrix 1204 - Error: Operation not permitted
4:59PM 1 Mediatrix 1204, where I can get the last firmware and mib file ?
3:59PM 0 H.323 IVR with avaya multivantage.
3:47PM 0 RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio) (fwd)
3:03PM 0 Where to get latest SIP bootrom/firmware for Polycom IP500 phones?
10:32AM 3 Message waiting
10:00AM 0 Getting a modem connection working via Asterisk
9:56AM 1 Lots of FXS ports / Channel Bank ?
9:51AM 0 call for help from uip200 users
7:55AM 2 astcc help
6:16AM 2 Howto do something after a Dial command is Answered?
6:06AM 0 (no subject)
4:33AM 0 Video Calls between SIP and H.323
2:39AM 1 res_config_odbc not working
2:22AM 0 Asterisk and Douglas Telecom
1:15AM 0 Re: Invalid data ( 4 bytes at the end) message - from g729codec
1:03AM 1 Confused --> Hardware specs
12:42AM 2 Asterisk : No Sound Issues
Friday August 6 2004
10:41PM 0 iaxtel, asterisk, and sipura 1000 am having trouble with codecs
9:29PM 0 Asterisk, channel banks and SDH
8:38PM 1 Asterisk and Cisco Call Manager.
7:03PM 1 oem x100p undefined symbol ast_get_txt
6:27PM 1 Asterisk Dry Run
5:51PM 0 Need help on installing an Asterisk PBX
4:38PM 1 Problems loading chan_h323 on Opteron 64 bit
4:31PM 1 redhat 9 and oh323
4:07PM 1 termcapsupport not found
3:59PM 0 no mail sent on voice message
2:05PM 1 Interesting catalog: Viking Electronics
2:03PM 2 RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)
1:55PM 0 t100 error on FC2
12:56PM 0 ShoreTel Phones
12:37PM 1 Zaptel CVS HEAD 08-05-2004 problems
11:27AM 2 FXO Problems
10:55AM 2 Inbound not working with iconnect
10:06AM 2 Asterisk not starting
8:41AM 0 Asterisk as SIP proxy?
8:19AM 3 tdm400p lockups
8:14AM 3 E1 monochannel :-(
7:02AM 2 Difficulty evaluating the return value of PlayBack (or any other extensions.conf command
5:16AM 2 DTMF after answer
4:33AM 0 RE: Zaptel X100P Kernel Panic
4:12AM 2 Lots of Echo with SIP -> Asterisk -> PSTN
3:12AM 0 iax->asterisk->capi isdn; crackle overlay after about 1 minute
3:09AM 0 Entry point & Minimal requirements for Linux Device Driver and connection with Asterisk IP PBX
2:48AM 0 Microsoft RTC Authentication
2:48AM 0 Re: Invalid data ( 4 bytes at the end) message - g729codec
2:10AM 0 Urgent help with Sip <------> H323 on FREEBSD
Thursday August 5 2004
11:58PM 1 Sip dialback
9:46PM 1 iptables, Cisco 7960 and TFTP
4:38PM 1 users
3:49PM 1 Skinny and CISCO 7905G
3:01PM 1 is asterisk and/or spandsp what I need for integration with Cisco?
2:37PM 0 Strange message, and one-way audio between sip and H.323
2:10PM 0 Re: Asterisk-Users digest, Vol 1 #4831 - 4 msgs
1:01PM 0 ASTRICON 2004
12:41PM 1 iConnectHere and CallerId
12:08PM 0 Asterisk used like SIP to h323 convertor
11:56AM 0 PRI Errors... Ouch
11:03AM 2 new bounty for modifying calling card application to mysql
10:44AM 1 transfering incoming message from app_queue
10:31AM 5 Anyone use AdvancedVOIP ?
9:21AM 0 problems with asterisk and the IAX protocol
8:54AM 2 PRI protocol question...
8:33AM 1 Advice on possible set-up
8:27AM 0 Bridging Calls
8:27AM 3 Avaya/Lucent Definity -> Asterisk interop question
7:24AM 2 X100P Kernel Panic
6:57AM 2 shared voicemail
4:30AM 1 h323 gnugk to h323 asterisk and then to endpoint
4:25AM 2 personal voicemail
3:27AM 2 Call Transfer Problems with Grandstream Budgetone 100 Phone
2:00AM 3 QuadBri in NT mode not working.
1:30AM 1 AW: Integrating an old PBX with Asterisk
Wednesday August 4 2004
9:11PM 4 Using answering machine in my phone
7:11PM 0 New Head Appears to Break SIP to iConnect
7:10PM 1 how to integrate an addon module to asterisk
6:47PM 0 Configure E1 PRI
4:19PM 0 Can't get T or t option to work with two IAX2 channels.
4:02PM 4 FCC Rules VoIP Must Be Tappable
3:56PM 2 Color in console
2:56PM 1 Identifying which call an event belongs to
2:14PM 2 Snom 200 Programmable Keys
2:10PM 0 Integrating an old PBX with Asterisk
12:54PM 5 H323 Call Dropping
12:24PM 0 calling card on Zap outbound
12:22PM 2 PSTN Access Providers for Asterisk
12:07PM 0 Zultys ZIP2
11:47AM 3 Auto-attendant with an IP trunk
11:36AM 2 Asterisk & ISDN-card
11:34AM 4 Asterisk Integration
11:21AM 2 Get MWI from Telco's voicemail
9:50AM 1 BT100 bad handset?
8:24AM 0 Bogus patent applications (was: [RANT] ... Broadvoice)
7:35AM 1 Barge in on to agents conversation
6:46AM 4 Problems with E100P
6:34AM 0 Astetrisk connectet to PBX
6:22AM 1 SIP pickupgroup
6:05AM 3 No incoming audio on incoming SIP calls
6:03AM 2 2 sip servers
6:02AM 0 Congested link
5:48AM 1 Who is calling me ?
5:18AM 3 German sounds
4:51AM 1 PRI/H323 gateway
2:26AM 5 Asterisk QOS working perfect using sveasoft 3.11g
2:18AM 4 rxfax killed asterisk
12:54AM 1 capturing a call
12:35AM 3 Cisco SIP Phone 7960 & DTMF Problem
12:24AM 1 about sip.conf
12:04AM 0 Call pickup (group)
Tuesday August 3 2004
10:06PM 1 CAC AB1 and Asterisk
9:04PM 0 Very decent book - "VoIP Telephony with Asterisk"
8:46PM 3 Re: problems with'#' transfer after hold
7:50PM 1 CID Blocked vs. Unknown
6:37PM 1 Logging into Multiple Call Queues on two * Servers and Voice Mail option.
3:48PM 2 problems with'#' transfer after hold...
3:19PM 1 UK VoIP-PSTN gateway recommendations
3:09PM 0 Can Zap detect line is already off-hook?
2:17PM 2 SPA-3000 as a regular Asterisk FXO device?
2:14PM 4 After RC1 upgrade, temporary loss of voice
1:38PM 1 instable Modem-Module in CVS ?
1:36PM 0 snom 200 - custom melody
1:09PM 1 Play audio into meetme conference?
11:48AM 0 Asterisk and RT
11:36AM 2 Integration with Altigen
11:03AM 0 Asterisk Tapi Driver for Windows
9:07AM 2 VoIP experiences with Cable and DSL
9:01AM 0 ZyXEL 2000w In Call Menu/Hold configs
8:53AM 1 Analog channel stays offhook
8:40AM 3 PRI Call Redirection / Transfers
8:22AM 1 Emailing phone messages?
8:14AM 0 DID Trunk
7:53AM 1 astguiclient: blank php pages
7:27AM 6 features.conf
7:06AM 0 Configure Makefile to run with older iax Protocol
6:48AM 0 VON Magazine article.
6:03AM 0 avm c4: DISCONNECT_IND ID=001 #0x0193 LEN=0014
5:47AM 0 Fw: Digium FXO Interfaces don't support groundstart???
5:21AM 1 Any small colleges/universities using PBX or Voicemail?
3:30AM 1 bearer Capability
3:24AM 3 Called ID in Australia
3:09AM 0 Echo problems with mISDN?
1:43AM 1 Using Clustering/TDMoE
12:52AM 0 OH323 not dial Modem[i4l]/g1
Monday August 2 2004
11:08PM 5 Making asterisk distributed
8:55PM 1 help with digium E1 card
8:23PM 2 Cisco PRI no CallerID
8:21PM 0 Help with Cisco PRI
7:55PM 1 G729 or GSM
7:52PM 0 Fax on demand
5:01PM 1 MPG123, Music On Hold and Variable Bit Rate
4:13PM 0 One way voice with following error
3:40PM 9 asterisk+radius
3:04PM 0 Help with ParkAndAnnounce command
2:23PM 2 CallPres screening DDI
2:01PM 4 First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?
1:23PM 1 asterisk call parking + SNOM lighted buttons?
1:07PM 0 Pre-release of OSX GUI tool to add extensions and phones
12:54PM 1 Asterisk as Front-End for Artisoft Televantage 6
11:06AM 7 System Requirements
10:09AM 2 New CVS and Sipuras
9:32AM 3 App.c
9:28AM 0 bri-stuff.0.1.0-RC2k + hfc card: dropouts on IAX2 & MP3Player quits on streams
9:05AM 1 avm c4, ptmp
8:49AM 3 How STUN work?
8:41AM 1 Performance of queues
8:30AM 0 Stripping characters from SIP dial strings
8:24AM 1 Vonage catastrophic failure...
8:21AM 0 Multiple Line SIP Phones?
8:20AM 1 (no subject)
8:13AM 1 Selling asterisk-based solutions
8:09AM 2 Cisco MC3810
6:41AM 0 Fwd: Help with Quicknet PhoneJack@Asterisk
6:40AM 0 Clustering in Asterisk
6:21AM 1 DID's in the Czech Republic
6:09AM 0 Help with Quicknet PhoneJack@Asterisk
5:56AM 1 Win2000 DUN via Asterisk (Is it possible)
4:39AM 1 RC1 - error message : Request to schedule in the past
3:48AM 4 CDR with MySQL and Asterisk PID File
2:11AM 0 detect FAX terminal
2:04AM 0 h.323 debug
Sunday August 1 2004
9:53PM 0 About CDR billsec when used TE410P
5:25PM 1 SIP- PSTN Gateway
2:57PM 0 Message Lamps across IAX connected switches.
12:13PM 1 Zaphfc CallerID problem...
11:32AM 0 Writing messages to ISDN phone displays?
11:28AM 2 Parking & SIP Phones
10:20AM 1 Grandstream Message Waiting light
7:47AM 1 Snom 220
7:13AM 1 cdr record for recording location
6:32AM 1 distinctive ring on SNOM 200
6:09AM 1 Does anyone know how to use the DND feature oc Cisco 7940/7960
3:17AM 2 Cisco 7960 backlight update and prices.
2:41AM 2 Zaptel - incoming delay
12:55AM 1 X100P wants to use g2