asterisk users - Dec 2004

Friday December 31 2004
TimeRepliesSubject
7:49PM 3 FC2 & ztcfg - cannot find channel 2
7:41PM 1 how is a upgrade performed?
7:30PM 1 Help With Configuration From Odbc
7:14PM 2 FC3 compile with new 2.6.10 fails
4:44PM 4 is wiki drunk
3:49PM 11 Softphone in German
2:45PM 2 Mysql-Realtime and ASTCC
2:19PM 0 manager API / weird queue
2:08PM 5 IAX media
10:29AM 0 Segmentation Fault (core dumped)
9:48AM 1 Broken pipe...
9:41AM 9 BroadVoice WiSIP with Asterisk
8:00AM 0 Thanks for help - Almost done - 50% - Can hear
7:24AM 0 Segmentation Fault Problem
6:03AM 2 MGCP parameters
6:00AM 2 Host IP address, Crash on startup, Console grabs soundcard - Newbie needs help
6:00AM 5 IAX users
3:59AM 2 hardened gentoo (selinux) asterisk problem
12:22AM 4 FXO to IAX on ethernet. or FXO to SIP on Ethernet
 
Thursday December 30 2004
TimeRepliesSubject
10:18PM 3 TE410P not Interrupting
7:38PM 0 A Single user
6:19PM 3 A simple scenario
5:47PM 0 TDM04b failures (xpost on purpose)
3:45PM 2 IAXy issues
2:58PM 0 chan_capi and voicemail to cellnumbers crashing asterisk
2:38PM 3 VoiceConduits - Notice
2:37PM 1 Queues strategy
1:58PM 1 Weird..bridging to Zap channel FXS instead of bridging to PSTN FXO on outgoing group
1:39PM 2 RealTime Drivers Connectivity Error
1:34PM 10 VoiceConduits is a scam
12:47PM 6 Agent login state saving?
12:01PM 3 Fw: Cisco 7690 Voicemail Problem
12:00PM 0 Asterisk dialing a Zap channel FXS instead of bridging to PSTN FXO
11:57AM 0 Problems starting *
11:22AM 2 More * weirdness
10:11AM 0 Zapatel ringing multiple SIP devices
10:09AM 2 IAX2 and DTMF
10:08AM 5 Voicemail and Zapatel
9:51AM 1 CDR IAX calls snafu ?
9:05AM 0 This item has been released from quarantine.
8:54AM 12 Nagios and Asterisk
8:53AM 1 DTMF skipped when calling from ISDN to SIP...
8:52AM 11 Sipura 3000 inbound FXO problem
8:32AM 2 IAX hardware
8:28AM 0 Fw: Open ports on router in front of asterisk
8:18AM 5 Helping communications to Asia area.
7:26AM 0 Asterisk with 2 E100P cards behind an Alcatel 440
7:04AM 0 New Diax version 0.9.9f
6:52AM 39 Is asterisk that unstable ????
6:03AM 0 VoDSL without using IAD
4:34AM 0 Re: Asterisk and Capi
4:33AM 1 Doubts about the Monitoring command
 
Wednesday December 29 2004
TimeRepliesSubject
10:59PM 0 12 CANCEL's followed by 12 INVITE's in 5 secs
10:00PM 0 how does ipphone pick up voicemail alert?
9:03PM 3 Problem with Digium TDM04B
6:50PM 6 automatic startup
6:37PM 2 show version
6:00PM 2 So what if I can't dial out ... or in ... Asterisk just blows my mind!
5:52PM 6 PRI Woes continue
5:43PM 0 ISDN4Linux Incoming calls
5:31PM 14 RE: Hook/Flash, Hold, Call Waiting, Three Way Calling
5:07PM 1 Issue with Mediatrix 1124
5:05PM 0 (no subject)
4:35PM 0 Channel Zap/4-1 in prering state
3:46PM 10 Hardphones Console o Secretarial One
3:32PM 4 Asterisk, she no hang uppa the phone!
3:30PM 1 Can I tell if it hung up due to busydetect or disconnect supervision?
3:16PM 1 Dial with no phone line connected
2:51PM 0 queueing question
2:16PM 13 DSLink modem freeze
1:52PM 5 Recording/Monitoring a call mid-stream?
12:13PM 1 RFI: Creating a database of DID providers
11:59AM 6 zapata.conf not being parsed by *
11:26AM 3 Hmmm - anyone seen this before?
8:57AM 0 IAX -> IAX -> SIP problems
8:50AM 15 IP Phone recommendations?
8:30AM 0 Supporting "End User Line Features"
7:44AM 0 trimming messages on reply
7:24AM 1 Polycomm IP500 dropping incoming calls
6:42AM 0 AstTAPI - Incoming Calls
5:56AM 2 Asterisk OH323 acting as a gatekeeper
5:37AM 1 Impossible to compile last version of Asterisk
3:59AM 5 TE110P doesn't appear in /proc/zaptel
3:16AM 0 Problem with musiconhold - No such file or directory
3:08AM 1 API Manager Events
2:53AM 0 Determine UAS on remote SIP phones
2:51AM 9 Final call for departments
1:41AM 8 spandsp-0.0.2pre6
 
Tuesday December 28 2004
TimeRepliesSubject
10:18PM 10 OT: Linux routing with T100P problems
9:18PM 0 service activation code
8:04PM 0 500 "Internal Server Error"
7:34PM 1 Sending e-mail from dialplan
7:28PM 3 Fedora Core 3 app_curl compile error?
7:10PM 5 Invalid Extension
5:45PM 0 How to connect two Asterisks as secure as po ssiblewithout too much additional bandwidth ?
5:22PM 1 PRI & CPU Usage
5:20PM 3 Dialplan variables
5:18PM 2 Meetme scalable to 300 people?
4:39PM 1 ASTCC Expiration
4:32PM 4 caller-id blocking
4:19PM 1 Intercom System with Asterisk and Cisco 7960
2:33PM 0 Calling Card question
2:24PM 7 WARNING[22314]: No such switch 'Realtime'
2:19PM 16 Music instead of Tunes
1:36PM 7 ZtDummy vs Hardware
1:22PM 0 FW: Compile Error
1:17PM 7 Hardware opinions?
1:12PM 0 external Radius Server integration with asterisk
1:09PM 0 Two problems with the Perl AGI
1:01PM 11 Sending call to analog then to Vmail after timeout?
12:59PM 0 Asterisk users manual
12:51PM 4 Dialtone for Software phone?
12:04PM 1 rejected calls from IAX provider
11:52AM 0 Polycom phone stops working
11:37AM 9 DHCP, the TFTP Server setting and the Cisco 79xx phones
11:20AM 2 Wildcard remote looping
11:02AM 0 VoIP Equipment
10:19AM 0 ztdummy necessary?
8:54AM 1 Asterisk consuming 100% CPU - CDR loop
8:05AM 5 Asterisk with T1
7:28AM 0 PCI PERR's w/Digium cards
6:57AM 0 Optional URL param
6:43AM 1 music on hold without sound card
6:03AM 0 H.323 link to provider VoIP with Username and Pass
5:57AM 1 Asterisk / 183 message
5:46AM 0 [Fwd: Callmanager 4.1 and asterisk]
4:45AM 5 Chan IAX2 errors while calling Toll Free numbers using IAXTEL
4:29AM 0 SV: One way audio
4:22AM 0 Packet flow in relaying from SER to Asterisk
4:02AM 0 pickup group
3:55AM 0 My firefly is changing the IP address !!???
3:50AM 0 Does anybody use a video phone ?
3:32AM 0 Asterisk recognize GSM CLI
3:16AM 0 dialplan "not ${VARIABLE}
2:53AM 3 Zaptel ISDN BRI settings for The Netherlands KPN
2:40AM 5 Mysql and Voicemail
1:15AM 0 socksify
12:21AM 3 Callmanager 4.1 and asterisk
12:12AM 0 Re: Help on Register message with Authentication
 
Monday December 27 2004
TimeRepliesSubject
11:40PM 0 Call Placing timeouts
9:44PM 3 Cant get Asterisk server talk with IAX
8:37PM 5 PassThrough mode
7:52PM 0 call parcking failure
6:16PM 1 Selecting Extensions
5:14PM 2 parking.conf
4:31PM 0 Re: Asterisk dying...
3:43PM 12 realtime voicemail
2:59PM 0 help regarding ASTCC
2:45PM 0 Asteriks Compile error
2:33PM 0 IAX -> SIP Call Help; IAX with G729
1:17PM 2 API manager - Redirect with ExtraChannel
1:06PM 3 codec preferences
12:40PM 0 asterisk dies no calls in or out
11:55AM 0 no voice with all sip phones until hold/unhold
11:42AM 2 does a TDM04B (all FXOs) need a power connector?
11:37AM 3 MYSQL_FRIENDS
11:20AM 1 transfer: hookflash vs #
11:11AM 9 how to debug frame slips?
11:01AM 7 TDM400 problem
10:38AM 1 Command-line dialer/recorder for asterisk?
10:03AM 0 Is there a way to avoid bandwidth consumption on sip calls?
9:11AM 0 [chan_capi] can't get it compiled
9:04AM 0 Jeff Pulver quoted talking about Asterisk...
8:21AM 3 Diax echo problem
8:12AM 3 mail function
8:08AM 1 Generic Network profile for VOIP
7:49AM 0 zaptel error : Relocation overflow of type 10
7:15AM 2 SIP client cannot connect to Asterisk
7:11AM 3 restricting SIP access to asterisk
6:16AM 0 Fw: Hookflash timing with TDM400P
5:34AM 1 Make error installing bristuff-0.2.0-rc2b
4:25AM 0 Problem with AgentCallbackLogin
2:40AM 1 incoming & outgoing call
1:06AM 0 ASTCC - setup help please
 
Sunday December 26 2004
TimeRepliesSubject
10:04PM 0 distinctiv ring (Aert-Info)
5:51PM 1 is deadlocking with the Manager API still a problem?
3:13PM 9 OT - Originating Network identity
2:16PM 25 Incoming Calls
1:14PM 9 Asterisk realtime load error
12:57PM 1 Cannot transfer after queue agent picks up c all
9:13AM 1 Cannot transfer after queue agent picks up call
9:09AM 4 Asterisk behind IX66
6:28AM 0 Voice modem + Asterisk
5:28AM 0 HUP signal?
4:21AM 15 IAX Registration Refused
2:02AM 0 SV: Call Completion Snom
 
Saturday December 25 2004
TimeRepliesSubject
11:15PM 0 Asterisk + Voice Modem
9:24PM 1 Alert-Info
9:12PM 5 VM_CALLERID (how to get name+number)
4:46PM 0 Where to get a Polycom IP500 in the UK?
4:37PM 0 safe_asterisk script contains error?
3:59PM 0 patch to build h323 without recompiling pwlib, ...
2:43PM 0 Bri-stuff + TDM 2-Port FXS & 2 Port FXO Card
8:06AM 10 How to connect two Asterisks as secure as possible without too much additional bandwidth ?
5:39AM 3 About CallBack function
4:03AM 0 Automatic calls
3:56AM 0 Bandwidth, computer power
3:47AM 0 TE410P No Interrupts
2:36AM 2 Can Asterisk handle calls that get picked up by answering machines?
2:36AM 2 Dynamic extensions without using DynExtenDB?
2:29AM 2 How to use firefly with Asterisk?
2:07AM 1 Asterisk and Lucent APX8100 Universal Gateway
1:48AM 2 Transcript of sound files?
 
Friday December 24 2004
TimeRepliesSubject
10:17PM 2 ALERT_INFO issue CVS-HEAD-12/24/04
10:01PM 0 Calling Party ringing indicator
8:17PM 13 What do I need to build up DID services?
6:46PM 2 Firefly Transfer call ?
5:12PM 0 VoiceConduits?
4:49PM 5 FC3, TDM11B (DEVPCI) and asterisk
3:06PM 12 Tie web application to VOIP
2:36PM 2 Uniden UIP200 firmware v4.63
10:30AM 3 Registration failure with debug
8:14AM 0 Cisco, Codecs, Sip Phones et al
7:14AM 1 Switch polarity to disconnect a FXS channel
6:44AM 4 Preventing Asterisk from sending 'h' across to SIP Provider
6:09AM 4 Help on Register message with Proxy-Authorization
2:27AM 0 Asterisk Xmas ;-)
1:57AM 0 SIP Multicast Support desperately needed :: Mission critical bug in Asterisk
1:39AM 0 Help:could asterisk work with other sip proxy?
12:32AM 0 help:could asterisk be used such as sip proxy?
 
Thursday December 23 2004
TimeRepliesSubject
11:37PM 7 Record() problem
10:15PM 0 Asked to transmit frame type 2, while native formats is 4???
10:00PM 0 Australian STD "pips" & Telstra pstn
9:09PM 15 Service contract for * in NYC area
8:38PM 0 Asterisk Certification
8:35PM 3 DISA restart from begining
8:03PM 0 txgain / rxgain no effect
7:17PM 4 Special Problem in Australia ??
7:05PM 1 where I can find some learning book about asterisk?
6:32PM 4 error starting asterisk
6:16PM 0 Turning "*" Hangup off in queues
4:01PM 14 asterisk at large
3:58PM 1 Can't Make Outgoing Call
3:45PM 3 rtp channels not through asterisk
3:40PM 0 Asterisk queue_log
3:32PM 3 Voicemail email notification
2:57PM 0 "*" behaviour in agentcallbacklogin
2:43PM 6 Recommended IAX softphone.
2:25PM 0 Cisco 7960 Support Products
2:21PM 3 turn on/off auto/attendant by dialing an extension
2:17PM 3 Asterisk 1.0.3 no RedHat zaptel script?
1:27PM 7 Asterisk in parallel with PSTN
1:06PM 3 IAX cause codes
12:29PM 1 RE: IAX2 calls failing one way
12:26PM 0 changethread: can't change device with no technology!
12:10PM 2 Polycom Buddies
12:07PM 1 Premature DRQ
11:53AM 11 T100P frame slips
11:36AM 0 SV: RedAlarm (t100p - Adtran Total Access 750)
11:28AM 2 Re: Asterisk and Capi
11:08AM 0 Asterisk cannot read DTMF based CallerID from PSTN
10:55AM 0 IAX2 calls failing one way.
10:38AM 1 PRI unable to request channel
10:27AM 0 Call Completion Snom
10:24AM 16 RedAlarm (t100p - Adtran Total Access 750)
10:01AM 0 New astGUIclient version released 1.0.6
9:58AM 5 TDM400 success?
9:28AM 0 switch statement.
9:12AM 4 Queue - roundrobin member order
8:57AM 1 Multiple Registration
8:49AM 3 Incoming calls from Sipgate go through the wrong peer
8:37AM 23 Fw: [digium.com #12961] T100P as bandwidth
8:29AM 0 Need help with cisco 7960 call fwd and dial plan
8:13AM 3 Linksys PAP2-NA Config
8:09AM 4 Polycom 600 problem
7:41AM 0 Registration Failure Directly related to realtime
7:25AM 0 lockup problem with inbound iax calls
7:15AM 1 Softphone x G729 x IAX
7:01AM 12 Realtime sipbuddies table structure why?????
6:27AM 0 Integrating Asterisk and Siemens Hicom 300E with TDM04B
4:37AM 3 One-way audio in incoming calls with Asterisk + OpenGK + Innovaphone IP3000
3:43AM 1 Problems with incoming IAX calls...
3:21AM 0 Passing SIP headers to AGI applications
3:02AM 0 Reservation call on busy
3:00AM 1 Re: Asterisk-Users Digest, Vol 5, Issue 329
2:58AM 0 Re: Asterisk-Users Digest, Vol 5, Issue 333
2:11AM 1 messenger on the mobile phone
2:09AM 1 How to apply patches
2:01AM 1 Qestion about TDM over enthernet
1:00AM 0 Connect attempt rejected error message
12:49AM 9 Asterisk with Dialogic VFX/40ESC plus
12:43AM 1 ignoring signalling
12:39AM 0 Disconnection Problem
 
Wednesday December 22 2004
TimeRepliesSubject
11:12PM 3 Out of G.729 Decoder Licenses!
7:10PM 5 polycom and cdp
6:43PM 13 Still unable to use g729 codec... please HELP
6:20PM 4 New verision of AMP - 1.10.004
5:38PM 0 Iax2 Registration failed
5:38PM 2 Zaptel/Zapata config from T410p to Brooktrout T1
4:49PM 0 FreeBSD, Generic Modem and DIGIUM boards
4:39PM 3 Problem ringing simultaneous channels
4:17PM 0 IAX Peering for PSTN termination Sydney <=> Moscow
4:13PM 1 SIP URI Dialplan?
3:49PM 2 Asterisk Interface to propriotary system and GPL
3:34PM 1 register_verify defined in 2 files?
3:22PM 0 Zap Fxs port always answers?
3:22PM 5 TDM400P install on Debian 2.6.10
3:10PM 1 MGCP Transaction identifiers
3:02PM 7 Can somebody email me the Sipura SPA-2000 and SPA-3000 documentation?
2:59PM 0 Phone Registration Failure Test
2:57PM 4 Asterisk billing solution
2:22PM 0 Softphone with subscribe/notify support
2:15PM 2 711 and 729 with IAX? (IAX Newbie)
1:08PM 4 Can't Receive/Send Calls
12:50PM 0 Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party don't hear)
12:45PM 0 chan_sip errors in CVS stable
12:43PM 1 Asterisk->AS5350 misplaced RTP to 127.0.0.1 (AS5350 party don't hear)
11:50AM 2 txfax failure
11:49AM 0 TE410P to a Rhino CB-24 channel bank
11:31AM 0 RE Zaphfc/BRI Configuration help
10:56AM 0 Macro(dundi-dundi-test, ${ENTEN}) to return +101 on lookup failure ?
10:41AM 0 rtc3389
10:37AM 0 Early media problems...
10:21AM 1 Status of asterisk.xvoip.com?
10:20AM 1 PRI error (HDLC Bad FCS)
10:14AM 13 Another Asterisk Certification
10:14AM 0 FWD + xtraphone and DTMF
9:47AM 1 Grandstream BT100 -> Asterisk -> Voipjet ..... No ring ring when making a call
9:35AM 6 gumstix
9:25AM 7 IAXy playing dead again
9:22AM 0 What is the procedure to test for Caller-ID
8:51AM 0 PassThru mode
8:05AM 0 What is the best commercial soft phone for Asterisk?
8:00AM 10 IAX hardphone
7:40AM 1 Phonecell + wildcard FXO (DTMF problems)
7:37AM 2 Call dies in 180 seconds exactly
7:35AM 3 call from DID, not hearing RINGTONEs
7:32AM 1 Cisco 7960 Hold
7:00AM 1 Link an Asterisk Box with a PBX (E1 connection)
6:57AM 9 ZapBarge
6:54AM 0 Wither ChanSpy ?
6:11AM 13 Why use 'Answer'?
4:53AM 2 Matching Caller ID against a database of knowncallers
4:27AM 3 E1 card for Asterisk
4:27AM 2 Aterisk@Home
4:12AM 0 Ticket: 12775 Multiple IAX client behind a NAT
3:07AM 0 RE: hint extension and Snom phones - CVS or
2:55AM 1 Daily NANPA updates
2:45AM 0 Dialogic MSI cards to FXO port on TDM400P
12:49AM 3 MWI not working on Polycom Phones
 
Tuesday December 21 2004
TimeRepliesSubject
11:12PM 1 Matching Caller ID against a database of known callers
10:31PM 0 opaque= field
10:13PM 0 IAX2 insists on not using port 4569??
9:10PM 1 Dialplan help - Can dial any user but not thePSTN
9:06PM 0 No Ringback tone on Stable 1.0.2
8:36PM 9 Cannot transfer with Cisco or Snom
7:34PM 6 X100P dead?
7:19PM 0 SIP dtmf=rfc2833 not working
6:49PM 3 Lets try this again then! Q: SIP error from dialplan I suspect!
6:44PM 1 Hmm something strange.
5:54PM 0 Voice prompts text & Chinese
5:20PM 0 Status of Queue?
5:10PM 0 Re: problem with calls on hold
4:52PM 0 fxstest cant ring phone, but asterisk can !
4:42PM 3 gateway.lu
4:30PM 2 CallerID returned with error on channel 'Zap/4-1'
3:37PM 1 zaptel ppp HDLC Receiver Overrun messages
2:52PM 2 IAXTEL Configuration
2:23PM 6 Problems installing Zaptel
1:37PM 3 Budgetone is not registering
1:33PM 0 Hung SIP channels in Asterisk
1:28PM 0 Help bridging 2 outbound IAX2 calls !
1:03PM 9 hint extension and Snom phones - CVS or stable?
12:53PM 17 Codec Selection
12:39PM 2 Poor Grammar or is this a bug
11:54AM 1 Small PBX to VoIP transition questions
11:54AM 0 Spandsp 0.0.2pre6 configure fails sanity check.
11:51AM 6 GUI Tool
11:47AM 7 sip seeding vs registration
11:46AM 6 G729, x-pro, and codec ordering
11:07AM 1 Linking 3 Asterisk box, server in the middle type of thing? (IAX?)
11:00AM 2 Call routing based on remote ip address.
10:12AM 1 bri stuff and unknown signalling type
10:07AM 0 Problems with Budgestream and g729 codec
9:49AM 4 asterisk-oh323: New versions available
9:37AM 4 Bug, Feature, or Limitation?
9:23AM 4 Minimal modules.conf (e.g. with autoload=no)?
9:16AM 12 SOHO PBX using asterisk
9:12AM 3 What is sip-friends.sql??????
8:59AM 0 CP7902g SIP IOS
8:54AM 2 TE405P E1 coax cables with balun
8:30AM 2 Paris Meeting on Dec 20, 2004 - réunion à Paris le 20 décembre 2004
8:24AM 1 Incoming call on IP
8:09AM 4 asterisk server to asterisk server
7:44AM 3 h.323 Type=User
6:52AM 7 AMP - Fax Detections
6:40AM 4 upgraded source now ata's ring but stop silence on inbound calls
5:59AM 0 Intel Cards ???
5:45AM 2 Call back when no longer busy
5:40AM 5 Queues without members
4:20AM 0 Showing the name of the country on a Cisco 7960/7912?
4:20AM 1 HELP: agi-test.agi does not return any DTMF!
3:49AM 0 Suggestions for Asterisk + BRI + Data
3:34AM 0 (no subject)
3:02AM 0 Incomming call to asterisk server error
2:49AM 3 Zhone Channel Bank
2:34AM 1 two avm usb isdn fritz v2.0 cards
2:18AM 0 howto disable call waiting ?
1:53AM 3 Channel limits ?
12:13AM 10 Caller ID - TE405P - Telstra Onramp 10 - Australia
 
Monday December 20 2004
TimeRepliesSubject
11:42PM 8 NMI issues...
11:26PM 3 Grouping SIP channels (Sipura 3000)
11:10PM 0 newbie questions / documentation feedback?
10:55PM 0 SIP ringback problem with Polycom phones and CVS HEAD
10:01PM 3 Can asterisk be run as non root anymore?
9:49PM 4 Mysql-Realtime
8:57PM 0 Asterisk mechandise reselers with good reputation
7:45PM 0 Asterisk with RxFAX/TxFax start problem
7:35PM 1 Example config for SPA-1001
6:35PM 0 Q: How do I join an in-progress Zap channel call?
6:16PM 0 x-ten pro and echo cancellation...
5:35PM 0 Patching the source?
5:28PM 0 On Australian News Sites : Open source software set to influence VoIP
4:10PM 4 codec issues
4:02PM 1 A few simple (I hope) questions from a first-timer
3:42PM 0 Asterisk Startup Scripts (My Bad)
3:37PM 1 ATA callwaiting
3:18PM 0 Incoming voicemail and dialtone
3:12PM 1 [Asterisk-Dev] RE: [Asterisk-biz] Asterisktraining andcertification :: AstriconTraining
3:08PM 4 Toshiba DK-40 and Asterisk...possible?
2:59PM 1 RFC3389 support incomplete.
2:42PM 1 RxFAX compile problem
12:44PM 4 Why does * only work with an ancient mpg123?
11:54AM 5 ATA Adaptor
11:45AM 2 What does "t" mean in a CDR entry?
11:16AM 0 Calling SIP Address From Behind NAT
10:45AM 5 Problem using SPA-2000 behind NAT
10:37AM 0 weird problem with IAXphone
10:09AM 0 What is the difference between monitoring and recording???
10:06AM 1 How to allow users to dial certain numbers
10:05AM 4 Asterisk Fails To Start on Reboot Mysql
9:49AM 2 asterisk: webmin or X admin.
9:33AM 0 Unusuall Asterisk Usage Idea...
9:04AM 1 Asterisk A-Z provider from sratch
8:37AM 11 One SIP peer use 2 diff codecs?
8:35AM 0 Testers needed for voicemail ODBC storage patch
8:30AM 29 Updating Asterisk
8:10AM 3 Is there hardware to remote control
7:40AM 2 AW: SMS - how to send one
7:18AM 10 grandstream MWI?
6:31AM 0 Is there hardware to remote control available?
6:24AM 0 Extensions SIP problems.
6:15AM 1 E1 signalling pridialplan
6:12AM 3 [OT] resetting SPA 2000?
6:12AM 2 Realtime voicemail failure
5:59AM 1 how to prevent res_odbc from loading
5:43AM 4 Making a queue menu not exit the queue
4:06AM 0 autovol 0.9
3:32AM 7 Problems with loading TE110 module
3:13AM 1 Help me ($$$) with install h323
2:20AM 28 'I'nvalid extension handling problems, even with workaround
2:18AM 0 Skinny bug / missing feature, who is the maintainer?
1:25AM 1 Fw: pbx.c:1279 pbx_extension_helper: No application 'SetVal' for extension (c819, 1, 1)
12:46AM 1 AW: Zaphfc/BRI Configuration help
12:38AM 9 PA1688 Chipset IP Phones & ATA's
12:29AM 5 Zaphfc/BRI Configuration help
 
Sunday December 19 2004
TimeRepliesSubject
11:28PM 7 Quick questions ( maybe a little confidence building too )
10:46PM 6 MFC/R2 errors
10:37PM 3 OH323 channel compile error
10:13PM 2 Can DPNSS be developed in S/w like libpri ?
8:58PM 0 iax2 event status using asterisk 1.0.3 & iaxfriends
8:43PM 1 Dialplan help - Can dial any user but not the PSTN
7:50PM 0 one way audio on sip channels
7:01PM 1 sip phones in different private networks have one way audio
6:30PM 9 [Asterisk-Dev] RE: [Asterisk-biz] Asterisk training andcertification :: AstriconTraining
6:13PM 0 RE: [Asterisk-biz] Asterisk training and certification :: AstriconTraining
6:09PM 2 OT- Callwave neat app
3:24PM 0 Call Queuing
2:33PM 0 Asterisk SIP transfer(refer)
1:49PM 0 ztcfg seg faulting
1:40PM 1 TE110P - problem with zone from zaptel.conf
12:41PM 15 SMS - how to send one
12:10PM 4 Looking for new hardware
10:58AM 2 dialplan selection
9:30AM 3 QuickNet Internet PhoneJack problem
9:09AM 4 VoicePulse OpenAccess
8:59AM 5 VoicemailMain can't read from phone keyboard!
8:41AM 11 TDM120 card?
7:59AM 1 BRI Error with zaphfc
7:13AM 3 Connecting Siemens HiCom PBX with Asterisk through E1
6:46AM 3 Make asterisk launch script after completing call.
5:52AM 3 Phone choices....opinion request Polycom vs Cisco
4:02AM 4 Per extension/user CDR?
12:56AM 2 TDMoE or IAX?
 
Saturday December 18 2004
TimeRepliesSubject
11:31PM 1 Getting the "real" extension into CDR
11:31PM 0 10-10 dial around
10:30PM 2 audio levels via sip
9:33PM 5 Free World Dialup and Asterisk
9:18PM 0 Configure Asterisk with Radius
7:15PM 0 Zap Channel Group Question
7:09PM 0 New FC1 packages...
6:33PM 0 One-way audio with SIP client only on certaincalls
6:06PM 2 voicemailmain hotkey
5:26PM 6 Q about IAX (and IAXy)
5:21PM 1 call waiting/ 3 way calling
4:51PM 1 One-way audio with SIP client only on certain calls
3:09PM 0 web-based sip / iax client
2:05PM 2 Problem with 302 "Moved Temporarily" Do not disturb
1:52PM 4 3rd party call control / CSTA , JTAPI or TAPI interfaces
1:41PM 0 Using Digium cards for data+voice & Asterisk
1:21PM 0 TEST - Pls Ignore (Unable to see my own posts)
11:53AM 2 Music/Busy Signal Not Heard
10:49AM 0 PRI got event: HDLC Bad FCS
10:31AM 0 Monitor entry not working... please help
10:00AM 2 External Address Books
9:21AM 0 Meetme with video??? + $US 2,000 bounty
9:01AM 0 SIP and IAX Clients for pre OS-X Macs ?
9:00AM 2 It's possible to do a codecs translation during a call in Asterisk?
8:49AM 0 what the heck? codec_gsm.c:135 gsmtolin_framein: Huh?
7:24AM 8 How to increase the performance?
7:22AM 0 Sound problems with iaxcomm and Linux Fedora
7:00AM 4 Problem with a TDM400
6:52AM 5 Setting up asterisk for one user in private ip NAT.
3:17AM 14 Open Ports
2:43AM 1 SV: call billing
12:49AM 3 Re: asterisk - basic hardware and packages
12:42AM 6 X100P card in Australia
12:36AM 0 Call Completion Asterisk and Snom
 
Friday December 17 2004
TimeRepliesSubject
11:03PM 2 Grandstream Voicemail
9:39PM 1 Mysql-Configuration
8:31PM 1 asterisk - basic hardware
8:24PM 0 asterisk packages
7:14PM 2 h323 channel compile error
7:12PM 0 Demo voice hickups.
6:35PM 12 NPA NXX data
5:53PM 2 Optimizing Sipura/Asterisk for DTMF?
5:23PM 4 OT: "Integrated Access T1" voice problems -is this possible?
4:44PM 5 T-1 vs channelised T-1?
4:10PM 1 Second TDM400 card
4:08PM 2 Call Queue Uniden UIP 200 not working
4:04PM 2 Total newbie here looking to do a VoIPconfer ence call?
3:56PM 0 Total newbie here looking to do a VoIPconference call?
3:35PM 4 hdlc + te410p + kernel 2.6.9 - anyone done this?
3:26PM 3 OT: "Integrated Access T1" voice problems - is this possible?
3:12PM 1 Total newbie here looking to do a VoIP confe rence call?
2:31PM 1 Least Cost Routing - Are you doing it? What are you using?
2:29PM 0 Total newbie here looking to do a VoIP conferencecall?
2:00PM 5 Total newbie here looking to do a VoIP conference call?
1:24PM 18 Call on hold disconnects...
1:21PM 5 voicemail without prompt
1:19PM 1 modified prepaid application
12:37PM 4 Snom 190, led and shared lines with asterisk
12:30PM 2 ASTCC in production
12:15PM 2 Asterisk receives busy..but its not...
12:01PM 25 Asterisk Crackly Bad quality
11:42AM 0 Dropping out of Queue to voicemail
11:34AM 1 application meetme
11:23AM 0 Red Alarm / Alarm Cleared Zaptel Issue (bug? )
10:58AM 8 Realtime and PostgreSQL
10:45AM 1 chan_capi - avm card does not work
10:28AM 1 Red Alarm / Alarm Cleared Zaptel Issue (bug?)
10:12AM 0 MusicOnHold. not getting it.-GOT IT!!
9:53AM 3 Old posts and the ability to search...
9:10AM 0 Latest head giving app_queue.c:340 error
9:01AM 1 SNIMTA_SPAM Using the Directory Feature to play a menu
9:00AM 0 Display on OptiPoint400std SIP
8:56AM 0 asterisk clients (need helpdesk solution)
8:54AM 2 New Asterisk Prompts
8:09AM 0 Newbie setup question (Voicepulse, FWD & IAXTEL)
7:59AM 0 German Howto?
7:12AM 1 Masive Fax Sendig with spandsp
7:05AM 1 Forcing E.164ID with chan_h323 & or chan_oh323
7:03AM 8 Definity PBX with a T100P & TN767E
6:43AM 25 OT: DSL without voice
6:42AM 2 Asterisk and HylaFax
6:25AM 2 erroneous errors - registration fails for grandstream phones
6:09AM 7 Cisco 7905g TFTP Configuration
5:32AM 0 can't intstall the webmin module
5:32AM 17 Troubleshooting Asterisk
5:15AM 6 Disabling " !" command
3:47AM 0 s and i in context not invoked
3:40AM 0 Simulate back impulse
3:23AM 0 [Off Topic] humour, XMAS, ground loop - good business strategy
3:14AM 0 AS5xx0: SS7 and SIP?
3:00AM 2 ADSI programming/TDM400P issues
2:55AM 0 instructions to get .bin firmware for 7920
2:49AM 0 chan_sccp and 7920
2:37AM 7 Meetme with video???
2:36AM 3 Paris Meeting Date/Time/Location
2:35AM 0 SayUnixTime
2:19AM 1 MD110 and analog trunks
 
Thursday December 16 2004
TimeRepliesSubject
11:15PM 2 Shorten the recognition time of rings on Wildcard X100P
11:07PM 0 Dialing asterisk from open phone
11:06PM 2 How to generate a SIP NOTIFY for Cisco 7960 remote reboot?
10:08PM 0 Call confirmation on NON Zap channels
9:53PM 1 Public Thanks
9:23PM 8 Hardware based DSP
8:51PM 6 191st simultaneous call fails
8:30PM 0 Call Waiting FXS and *
6:28PM 0 are there any tips/tricks to get the uip200 to register?
5:55PM 2 problem with freebsd 4.9 port
5:08PM 1 ilbc and asterisk 1.0.3 - strange noises.
4:53PM 1 Rapid DTMF entry failure
4:38PM 0 Can I read more than 7 numbers from capi ?
4:13PM 2 Asterisk Cisco CallManager Integration
4:08PM 1 Low-latency kernel?
3:57PM 0 zap, agents, ackcall
3:08PM 1 Which Primary ISDN card to use in Europe ?
2:57PM 2 MusicOnHold. not getting it.
2:47PM 4 Connecting Asterisk to GSM
2:35PM 10 Polycom SIP Phones
2:26PM 2 Dynamically Choose Codec for Bandwidth Management
2:07PM 1 Steps to configure D/41EPCI card
1:51PM 3 Get asterisk out of the RTP stream?
1:37PM 0 STABE, CVS and in between? Confused
1:37PM 0 Re: Re: Cant set H323 up
1:37PM 0 SIP channel groups - is it possible?
1:23PM 0 Has anyone connected to 7960 with console cablefor setup?
1:20PM 0 Good place to get DID's?
1:06PM 9 native MOH with Asterisk 1.0.3
12:04PM 2 working with big blocks of msn's
11:35AM 1 Polycom FX Video Unit - asterisk-oh323
11:34AM 13 Has anyone connected to 7960 with console cable for setup?
11:19AM 2 sox-12.17.6
10:59AM 1 BRI Card not recognized
10:45AM 21 Calculating required bandwidth
10:05AM 0 FW: Cisco 7960 (SIP) hold problems
9:26AM 1 OT: iax.cc hosts - want to do some traceroutes before buying
8:40AM 3 Multiple IAX client behind a NAT
8:01AM 3 Cisco 7960 (SIP) hold problems
7:33AM 4 How to tell "Who's Online"?
7:24AM 0 Codec Negotiation Problem
7:18AM 3 Queueueueuueue position
6:52AM 0 Compile issues: * 1.02 + FreeBSD 5.3
6:06AM 1 send # with transfer enabled
6:02AM 0 Reporting Errors & Mysql
5:30AM 3 Detect line is busy with Zap?
5:16AM 0 SPA-3000 - Stop Message Waiting Indication
5:10AM 0 codec preference?
5:08AM 0 Making "sip show channels" show sane results with sipfriends from mysql?
5:05AM 0 Voicemail Pager Subject?
4:47AM 0 Asterisk <--> Nuera Orca
4:44AM 0 Automated callback with .call file
4:21AM 0 Logging codec in cdr?
4:18AM 0 Channel Groups with SIP
4:16AM 1 Monitoring an active call
3:55AM 8 g711 ulaw vs alaw
3:48AM 4 asterisk on FC3
2:48AM 21 My Boss wants background music!!!!
2:47AM 0 kewlstart - explanation of this method, please ?
2:08AM 0 Asterisk support mailing list in Italian
1:59AM 1 Calls arent handled by asterisk - destruction of call
1:44AM 0 E1 and analog cards FXS in one box.
 
Wednesday December 15 2004
TimeRepliesSubject
11:59PM 1 Asterisk, Capi, Controller
11:10PM 0 Re: Asterisk-Users Digest, Vol 5, Issue 221
10:40PM 6 VoIP bad voice quality
10:03PM 24 VoIP Termination
9:40PM 8 QOS Device?
8:18PM 1 mpg123 exploit
7:39PM 6 VOIP Phone Suggestions
7:19PM 1 Linksys PAP2-NA Screenshot
7:13PM 0 PRI Errors again... sigh.
7:11PM 1 Re: Asterisk-Users Digest, Vol 5, Issue 219
6:49PM 1 Outlook integration?
6:25PM 7 Bugtracker Karma Hall Of Fame
6:09PM 2 SIP Server question / recommendations
5:22PM 3 Voipjet problems
5:14PM 1 Help with transferring a second call from a snom 190
4:43PM 0 Fulfillment, Gold/Platinum Programs
4:32PM 0 Can Directory app read extension numbers?
4:26PM 0 AstLinux - New Version - w/ 1.0.3 what about capi!!!!
4:14PM 0 agi send text option
3:59PM 6 Newbie setup (Hardware questions)
3:56PM 2 Advanced Ring All Hunt Group
3:49PM 1 Using ChanIsAvail with SIP
3:37PM 2 asterisk + freeradius
3:37PM 18 wcfxs causing constant CPU spikes
3:34PM 3 TDM400p FXO module always offhook
1:44PM 0 Digium TDM11B
1:43PM 0 E&M Wink Question
1:30PM 0 Start of conversation lost
12:44PM 2 chan_sccp compile problem w/ CVS head?
11:41AM 1 Sipura 2000 intermitent failure to register
11:09AM 0 Skinny not working?
10:54AM 2 Cisco 7960 SIP + 7914
10:52AM 12 No Caller ID Name PRI NI2.
10:49AM 1 SNOM 190 Call Completion
10:06AM 0 Very strange behaviour, has anybody noticed?
9:22AM 0 AstLinux - New Version - w/ 1.0.3
9:13AM 3 PRI incoming call????
8:27AM 2 IP Conference Units?
8:26AM 22 How "expensive" are the different codecs? (Regarding CPU time)
8:15AM 5 Codecs and RealTime
7:21AM 11 codec order in SIP doesn't work
6:23AM 0 ASTCC and CDR info
6:14AM 0 first 2-3 secs choopy sound
5:38AM 0 SIP INFO vs RFC2833?
4:32AM 1 IAX2 Notify exchanges on port 1024 and 1040 - Normal ?
4:07AM 0 RE: Asterisk-Users Digest, Vol 5, Issue 187
3:31AM 1 Easy question? Get started with the Demo
3:12AM 1 Re: 12.50$ per port ???
2:15AM 0 Digium hardware vendor in Israel?
1:42AM 1 IAX2 tolerance on packet losses
1:31AM 0 Asterisk to sip client behindFirewall/NAT-cancall but cannot receive calls ?
 
Tuesday December 14 2004
TimeRepliesSubject
11:18PM 1 SIP and Windows Messenger
10:45PM 0 Bug 3020 needs supporters :-)
10:24PM 0 Codec "Uknown" with IAX connection
10:24PM 3 Asterisk Realtime IAX - Adding fields
9:08PM 0 Festival 1.95 on 64 bit linux 2.6 FC3
8:14PM 9 Verizon PRI Setup Problems
7:34PM 0 Brian, Mr. West, are you out there?
7:27PM 4 Verizon PRI Setup Problems - Only Busy and Congestion
6:41PM 0 astersik sip routing question
6:37PM 1 terminate sip calls from a 3rd party sip proxy into asterisk. and then to gnugk
6:13PM 3 Sipura 841 delayed: other PoE options?
5:38PM 0 Asterisk make ext. light up?
4:26PM 3 Out of State
2:50PM 6 Realtime problem
2:36PM 0 SIGSEGV, Segmentation fault while debugging asterisk with gdb
2:14PM 4 Confirm MWI doesnt work with SIP RealTime?
2:11PM 16 list broken again?
1:02PM 3 IAX Provider Recommendation - Unlimited
12:50PM 1 SIP and * with dual ethernet cards
12:09PM 5 Digium Hardware in Canada
11:44AM 6 Soekris net4801 for home use?
11:36AM 4 Asterisk Randomly Hanging up on Zap channels
11:29AM 4 ztcfg problems
11:00AM 5 404 "Not Found" Sip Response
10:55AM 1 X100P and Mitel SX-2000 Light
9:20AM 2 Re: Asterisk-Users Digest, Vol 5, Issue 192
9:10AM 0 Kirk IP600 Wireless DECT station setup??
8:58AM 4 numeric caller id display on budgetone 101
8:56AM 3 Virtual Modems
8:53AM 3 CLI Timeout ?
8:49AM 0 voicemail playback problem
8:48AM 13 least sucky FXO interface?
8:35AM 0 Voicetronix FXO on OpenCall 4 vs OpenSwich 6
8:24AM 0 How to do this ?
8:22AM 6 IAXy provisioning
8:21AM 0 Should echo cancellation be a "science" oran"art"?
8:19AM 1 Asterisk to sip client behind Firewall/NAT-cancall but cannot receive calls ?
8:14AM 0 volume problems on zaptel
8:04AM 0 Mixing PRI's and BRI's
7:07AM 1 How to debug? - SIP calls not coming thru
7:04AM 0 AGI Helpdesk/Trouble Ticketing application
7:00AM 2 silence suppression question
6:40AM 3 sip_buddies mysql table
6:17AM 0 Problems with Chan_capi 0.3.5 & Asterisk 1.0.3
6:14AM 1 ISDN HiSax: unauthorized source code changes
6:13AM 1 Suggested Literature
6:07AM 0 Setting ISDN Service Codes chan_capi/zaphfc
5:52AM 6 Asterisk Realtime IAX - Adding fields for database table
5:48AM 0 tetting
5:15AM 1 Softphone features
4:26AM 0 Radius support
4:08AM 0 Snom 190 and lamp field
3:36AM 5 SIP registrations not staying registered
3:32AM 2 Dial Plan Problems
3:30AM 1 Asterisk to sip client behind Firewall/NAT - cancall but cannot receive calls ?
2:51AM 4 Problems with app_realtime
2:23AM 0 Asterisk to sip client behind Firewall/NAT - can call but cannot receive calls ?
2:12AM 0 Analog modem testing
2:05AM 1 Astersik with ISDN up0
1:59AM 4 Help with Queue Cmd
1:18AM 0 Issues with Asterisk
 
Monday December 13 2004
TimeRepliesSubject
11:33PM 4 Caller ID on Snom 190?
11:29PM 1 Newbie-Firewalls?
11:13PM 19 ASTCC
11:02PM 0 Regarding IRQ problems; try googling for "Digital Audio Workstation" or "DAW"
11:01PM 0 Looking for Full or Part time asterisk techs
10:43PM 7 Busy message on ISDN cards?
9:56PM 3 Cisco Router FXO / Skinny
9:16PM 0 What is the purpose or zttest and ztspeed ?
9:04PM 0 Setting up prepaid
7:57PM 1 DS3 Media Gateway
7:54PM 1 Bad Request Connecting SIP
7:36PM 0 hardware IAX to PSTN gateway?
6:56PM 0 Ethernet Channel Bank (Comming Soon to a NOC NearYou!)
6:12PM 0 SIP and IAX login design
5:36PM 0 Transfer and keep variables
5:16PM 0 AstWinPeers - combination of IAX/SIP/Peers/Graph
5:15PM 2 Repost: Cisco 7960 and Asterisk...not working....
4:45PM 1 incoming call from pstn to fxo not working with Asterisk
4:21PM 1 Asterisk up & running, now what?
4:09PM 0 MultiTech VOIP box
3:48PM 2 The correct way to get most recent stable
3:23PM 0 How to connect * to Adtran 600?
2:59PM 1 recommended IP phones and VoIP providers?
2:57PM 4 CPU spikes with wcfxs loaded
2:56PM 10 How can i test a modem with Asterisk?
2:53PM 0 phpconfig - can't locate any of my sections
2:19PM 0 looking for input on broadband router with QoS andVPN support
2:07PM 0 Disa Cdr
2:01PM 0 SIP CGI
1:25PM 0 weird ring behavior
1:14PM 4 Incoming Toll-Free
12:11PM 0 setting up asterisk as voicemail for softswitch
11:50AM 7 Multiline / Console / Receptionist phone
11:47AM 2 Can a TDM21 and a X100P co-exist
11:40AM 0 Asterisk and Sipura SPA-2000
11:28AM 0 Discontinued Firmware?
10:44AM 13 Asterisk on SuSE 9.1?
10:18AM 0 Portuguese (Brazil) configuration setup
10:14AM 5 Asterisk and Cisco 7905G or Cisco 7912G
9:36AM 7 How to create a confrence using SIP channels
9:22AM 0 Music on Hold with Parking
9:17AM 1 CallerID after Supervised Transfer
8:58AM 0 Detect line in use?
8:22AM 1 only allow long distance calls to countries x, y, and z
8:09AM 8 Pitching Asterisk
6:51AM 0 Broadvoice Patch Applied to CVS
6:09AM 0 Reading mysql sip friends
6:08AM 3 CVS zaptel missing files
6:00AM 0 install e100 card errors
5:42AM 11 Dialing out to 2 clients simultaneously
5:42AM 1 MYSQL cmd - preconnect?
5:23AM 0 [oh323] sporadic call setup
5:19AM 0 Asterisk receiving SER calls
5:05AM 5 transferring variables with IAX2?
5:03AM 3 Echo on one E1 line, but not the other
4:46AM 4 Traditional Telephony Interface Card
4:40AM 3 What route do diverted SIP calls travel?
4:16AM 3 IAX.cc / Sixtel?
3:54AM 1 Doing a # transfer on calls needing a #
3:17AM 6 Strange Segmentation fault
2:45AM 0 outgoing call queue.
2:08AM 0 Call Monitor Fails after Transfer
1:56AM 1 "detected NAT type is full cone" for BT behind nat ?
1:25AM 0 Issues getting Asterisk Realtime configured and operational
12:37AM 8 Follow Me & Music on hold
 
Sunday December 12 2004
TimeRepliesSubject
9:45PM 1 Sipura SPA-2000 won't ring
8:42PM 1 Log's Message Codes
8:33PM 0 DUNDi performance
7:57PM 1 patton smartnode integration
7:25PM 0 Any plans for video in oh323?
5:13PM 1 Using SPANDSP for faxes
4:35PM 0 BRI Problem dialing out
4:00PM 2 zaptel 0.9.1 compile problem
2:17PM 0 IAXPeerGraph - a beta of another windows monitor app
1:33PM 1 I'm stumped
1:01PM 6 [OT] Small SIP phones?
11:12AM 3 Re: Cant set H323 up
9:35AM 9 TDM400P FXS polarity reversal?
8:39AM 1 Totally LOST with dialplan and Extensions.
8:10AM 1 SV: How to Playback Mailbox Owners Name?
7:38AM 0 MeetMe performance
7:35AM 0 3com NBX and Asterisk Integration.
6:57AM 2 Pattern-matching in the dial-plan
5:38AM 1 gap in priorities - what happens
5:34AM 1 can a TDM400P FXS drop voltage on hangup?
3:47AM 4 Caller ID info ZAP --> SIP??
3:45AM 2 How to Playback Mailbox Owners Name?
2:06AM 4 Problems getting Asterisk Realtime to work
1:20AM 1 Will Adtran TSU 600 work with *?
1:09AM 0 DIALSTATUS missing an important condition?
 
Saturday December 11 2004
TimeRepliesSubject
9:19PM 0 Variable-length dialing with a Quicknet Inetnet PhoneJACK card
8:39PM 9 modprobe wcfxo causes fc3 box to crash
7:35PM 0 20 BT-100 setup - what firmware is recomended ?
7:32PM 1 Many similar contexts - can I use Macro or some other template concept ?
7:06PM 2 Can't capture "-1" return on Dial command
5:44PM 2 help with detecting fax.
4:45PM 1 Problem with TDM400P and cidstart=polarity
4:39PM 2 ACK from asterisk not matched to transaction by SER / LCS2005
4:17PM 6 does aanyone have an example of how to dial outwith a sip phone on a pstn line?
3:44PM 0 Soyo G668 IP Phone
3:43PM 0 SPA-2000 NAT Problems
2:46PM 0 Background Music via telephone speaker.
2:32PM 3 looking for input on broadband router with QoS and VPN support
2:06PM 0 Cisco 7960 and Asterisk...not working....
1:06PM 0 Migrating from CVS HEAD to Stable 1.0.3?
1:00PM 0 Tormenta PCI - tor2 module not loading
11:23AM 5 Cisco 7960 says "Protocol Application Invalid?"
10:05AM 0 636 Area Code Asterisk Compatible DIDs
9:35AM 2 What might be blocking RTP
9:27AM 1 How to setup private enum server ?
9:13AM 2 voicemail from mysql / change password
9:07AM 0 does aanyone have an example of how to dial out with a sip phone on a pstn line?
8:32AM 0 Re: Asterisk-Users Digest, Vol 5, Issue 158
8:29AM 0 Linux basics and Asterisk basics
8:13AM 0 Monitor, append audio?
7:54AM 0 Newbie MusicOnHold issues
7:53AM 4 OT: canterburyfortmyers.org returned mail
7:00AM 4 IAXy: no dial tone
5:39AM 1 Handling "raw" audio (8000 signed 16bit big-endian)
4:25AM 0 Asterisk 1.0.3 and chan_capi ?
4:01AM 1 RealTime and Macro question?
1:34AM 2 long list of prefixes
 
Friday December 10 2004
TimeRepliesSubject
11:05PM 0 -p real time priority and -U together
11:04PM 3 Can I re-write an incoming caller-id?
10:58PM 3 How to test enum?
10:23PM 0 Voicemail User Reference Guide
9:26PM 0 TXTCIDName
7:26PM 5 Very Cool.........Asterisk Made Wired Magazine
7:12PM 0 Re: Asterisk-Users Digest, Vol 5, Issue 158
5:55PM 1 E100P / Brazilian Telco Problem. (Urgent)
4:56PM 0 Asterisk RealTime Wiki Pages
4:30PM 0 Help setting-up X-Pro behind a proxy
4:26PM 5 New PRI with DID in US?
3:53PM 4 IAXPeers for Windows Beta released
3:42PM 0 Polycom caller id issues
3:34PM 3 Return code from queue app
3:22PM 0 * as a fax/voice switch
3:00PM 11 Voice Prompt Info
2:33PM 2 Should echo cancellation be a "science" or an "art"?
2:31PM 3 Asterisk Training Needed in SouthEast U.S
2:29PM 4 Linux basics
2:25PM 1 ringing after hangup
2:15PM 7 [Fwd: Re: udev or not?]
1:35PM 3 Need an Asterisk Expert for a Project
12:58PM 1 Install Xc-Ast $$$
12:45PM 4 Apply Patch for Broadvoice.
12:40PM 1 T.38 Pass-Thru?
12:17PM 2 include and hint in extensions.conf with new realtime feature - how?
12:04PM 0 providing battery reversal from Asterisk to legacy pbx
11:52AM 0 Confused about proxying and NAT, and seeking guidance
11:50AM 3 Asterisk from CVS
11:43AM 0 Help with configuring CFAS groups
11:39AM 2 static recording
11:05AM 2 using built-in extension numbers on the ZAP channel
11:00AM 17 Ripping CD audio for MOH
10:15AM 0 MySQL - mistake in previous post
9:33AM 2 Integrating * with Mitel SX2000 Lite
9:14AM 0 AGI Perl
9:12AM 2 dtmfmode: inband question
9:08AM 9 Granstream phones message button
9:00AM 4 Intercept and redirect outgoing calls ?
8:58AM 0 Not receiving DTMF from gateway
8:40AM 0 Aditional local number when calling from ISDN thtough Capi to local extension ?
8:18AM 4 OT: How do I know if I should have IO-APIC?
7:58AM 0 Change logs
7:42AM 2 Asterisk 1.0.3 - Signaling on E100P.
7:22AM 10 MySQL Realtime Driver
7:02AM 0 Moving call control to a second server
6:44AM 0 Dialing Problem with Welltech 3806 FXO gateway
6:23AM 2 BT100 how to pickup a parked call
6:01AM 0 SS7 to E1 & CPC
5:51AM 2 ISDN Data calls through *
5:19AM 2 Doubts regarding g726 - 16 bits setup
4:35AM 0 sip phone...direct access...
4:29AM 2 Mysql configuration interface
4:11AM 0 analog FXO debug suggestions
4:10AM 0 voice + data
4:02AM 0 D/41E ISA Card with redhat 8.1
2:46AM 0 variable limit time on Dial
12:39AM 1 udev or not?
12:31AM 4 PoE VOIP phones in Australia
12:27AM 7 polycom phone IP 500/600 conference feature
 
Thursday December 9 2004
TimeRepliesSubject
11:41PM 4 Asterisk started but doesn't register SIP client
11:23PM 1 Lost admin password on Polycom IP500?
11:20PM 5 Lost Password to Polycom IP500
10:35PM 0 Balanced call distribution to agents logged into multiple queues.
8:30PM 11 Cisco AS5XXX to asterisk debugging.
7:50PM 0 Disconnect Via Budgetone and 3com NBX
7:13PM 17 Asterisk@Home
6:56PM 6 SCRIPT: Fax Remvoal Please Call: 1-800...
6:06PM 2 Audio Hung after 1st call
6:06PM 1 Slackware & zttool
6:03PM 5 urgent outbound dialing problem
4:17PM 5 chan_capi question
3:13PM 9 No ring signal when calling internal extensions ?
2:56PM 2 Forward voicemail to *remote* voice mailbox?
2:44PM 0 --SOLVED--Voicemail messages by email
2:23PM 0 solution - running asterisk on box using alsa (FC3) for CONSOLE/dsp and wishing to play audio from browser
2:21PM 0 BT100 cannot park a call properly???
2:19PM 1 prepaid calling card application
2:07PM 1 Cisco IP Conference 7935
1:47PM 6 possible OT - ADIT 600 question
1:16PM 4 very OT - basic newbie networking
12:57PM 6 Voicemail messages by email
12:45PM 0 anyone know anything about audiocodes analog gw's
12:43PM 0 running asterisk on box using alsa (FC3) for CONSOLE/dsp and wishing to play audio from browser
12:28PM 0 New batch of phrases from Allison
12:26PM 1 Changing NICE value for * will it help?
12:15PM 1 can FXS ports on TDM400P provide Battery Reversal or CPC
11:52AM 1 Providers for PSTN Access
11:47AM 1 Asterisk@Home software?
11:00AM 4 Silent IAX calls getting cut off
10:13AM 0 Incomming calls on h323
9:40AM 0 Ser + Asterisk & DMZ
9:36AM 2 Multiple Instances of Asterisk
9:31AM 0 safe_asterisk not working
9:28AM 0 Problem with Accounting and wrong Caller ID
9:21AM 4 MySQL, CDR with MySQL
9:14AM 0 Can asterisk accept cleartext auth (uri user:pass) via SIP
9:08AM 0 Polycom IP400
9:02AM 5 Handsfree Speakerphone
8:58AM 16 Horrible MeetMe performance
8:30AM 3 Adit Asterisk Cabling Connundrum.
8:25AM 9 Sipura SPA-841
8:25AM 0 Reminder: $500 Bounty for Bluetooth
8:19AM 2 pseudo load balancing?
7:53AM 1 OT- Dell Xeon Servers UK Dealy, was Asterisk with SMP hardware
7:45AM 1 res_perl module loading problem
7:36AM 1 sip+nat+bt-100
7:34AM 0 Asterisk Monitor after Call Transfer failing to record the call
7:28AM 0 For all of those wondering about zaptel hardware and interrupts
7:20AM 7 Swissvoice IP 10S VoIP Telephone
6:26AM 1 [OT] Adit 600 Question
6:24AM 0 RE: Re: News about SS7? (Storer, Darren)
6:09AM 0 Base Number and DIDs
5:53AM 0 chan_sip2 multiple outbound proxies
5:43AM 3 hfc card and isdn error E001B
5:23AM 25 four wildcards in a single pc
5:13AM 0 Got SIP response 403 "Anruf nicht erlaubt" back from 194.97.54.97
5:12AM 1 Xorcom Rapid 0.9.0
5:07AM 1 A waning console error
4:35AM 0 Workimg On PostgrSQL
3:00AM 1 IAX midget packets!?
2:59AM 9 MeetMe Features
2:48AM 0 Asterisk and Cisco 5350 - config ?
1:50AM 1 Call Transfer drop.
1:50AM 1 pppd dial-in over asterisk
1:45AM 1 News about SS7?
1:21AM 21 BT-100 Transfer!!
1:16AM 1 Spandsp loading via asterisk app_rxfax.c broken pipe.
12:57AM 44 very OT - basic newbie networking question
12:32AM 5 Get rid of H323 problems for 100$
 
Wednesday December 8 2004
TimeRepliesSubject
10:24PM 8 How to demo the Power of Asterisk
9:32PM 3 Asterisk with SMP hardware
8:41PM 1 ftmp header
7:43PM 42 Ethernet Channel Bank idea
7:15PM 3 NEC Univerge
6:22PM 0 Two Zap Problems with 1.0.2 that appeared at the same time: choppyness and squealing
5:58PM 2 Broadvoice and incoming DTMF
5:45PM 1 PSTN number with callhunt and voicemail we web interface
5:13PM 3 Leadtek BVA8051 / Sipphone.com CallInOne with Asterisk?
4:56PM 4 CAPI, BRI and grouping B channels
3:54PM 0 Asterisk Intermediate-Advanced Administrator wanted in South Florida
3:11PM 0 UA -> SER -> asterisk
3:01PM 14 more then two wildcards in one machine
2:56PM 4 Playing Audio before the Phone is Ready
2:44PM 14 SIP Client for Symbian
2:26PM 8 Asterisk 1.0.1 Too many open files
2:18PM 0 how to make asterisk drop battery on a FXS?
2:05PM 0 Can Dial Calls from an Estara SIP Client, but Cannot Complete Calls to the
1:44PM 12 Asterisk Maintenance
1:15PM 1 ASTCC MySQL CDR
12:47PM 6 Guide to Cisco 79xx
12:34PM 0 Re: Asterisk-Users Digest, Vol 5, Issue 113
11:39AM 9 Polycom 500 - Dialtone while connected
11:33AM 1 3com phones and Asterisk
10:41AM 2 Dead TDM400P ?
10:26AM 0 OT: Polycom IP 400
10:01AM 0 OT: CP-7960's are in for those of you whop purchased them. We are shipping today.
9:48AM 0 Are there any digital phones that runon asteriskyet?
9:30AM 0 Zaprtc seems unsupported, Asterisk in productionenvironment without Digium cards
9:28AM 2 Dropping Calls, irregular interval no logs
9:14AM 2 Using meetme video mode with SIP ? Now a $2000 bounty
9:07AM 18 pc
8:56AM 17 T100P PRI question
8:56AM 0 Asterisk with 3COM phones
8:42AM 0 Does Asterisk support 3com 1102 phones ?
8:31AM 0 small business installation.
7:35AM 3 Voicetronix vs Digium FXO
7:12AM 4 PrivacyManager 10 digit limit.
6:16AM 0 IAXy & Auto-Dial
6:02AM 6 CPC, Calling Party Control, Disconnect supervision, -- how to tell that to Verizon (east coast)?
5:56AM 0 Looking for a Vonage contact
5:36AM 0 /dev/ttyI and few ISDN cards
5:34AM 13 setting the Call Forward Number in Zap?
4:51AM 0 Re: Spandsp loading via asterisk app_rxfax.c brokenpipe.
4:34AM 7 asterisk consultants
3:52AM 3 Asterisk's Empty Folder
3:03AM 0 Dropped calls on IAX connection
2:12AM 2 Spandsp - Libtiff problem
12:40AM 13 sangoma
12:09AM 0 Source/cause of echo delay (on internal stuff network)
 
Tuesday December 7 2004
TimeRepliesSubject
11:35PM 0 Busy Detect
10:12PM 2 dead BT100
9:44PM 0 Zaptel HDLC (NetHDLC) errors on modprobe, Linux 2.6 kernel
9:44PM 3 SIP endpoints ----> RTP stream
7:57PM 3 Cepstral voices
7:51PM 0 Subject: Re: Analog FXO Woes Continue
7:18PM 0 why busydetect can work sometimes, then sometimes not.
6:44PM 2 Segfaults when playing GSM files
5:26PM 4 conferece/Voice Mail features and LBR codecs (G7231, G729)
5:10PM 0 AGI application doing Hangup command and different AGI application running receiving the Hangup - additional
4:56PM 14 :: Migrating to 1.0.3 => Attention. ::
4:42PM 4 Asterisk / VOIP Employment Opportunity
4:14PM 3 Continuance on Polycom issue, not ringing
3:51PM 0 cf gsm adaptors
3:48PM 5 Broadvoice - DTMF
3:31PM 3 Inoming caller id withheld, move to new context, possible?
3:29PM 3 can't compile chan_capi 3.5 after patch applied :-(
3:17PM 2 Ringing multiline phone
3:13PM 0 sip phone to sip phone errors
2:59PM 0 AGI application doing Hangup command and different AGI application running receiving the Hangup
1:56PM 1 asterisk & 3rd party vm
1:38PM 0 ISDN on com port /dev/ttyS0 possible ??
1:37PM 0 Comdial PBX -- can use Asterisk as VM
1:24PM 0 monitor load on (zap)channels ?
1:03PM 1 Monitoring a call in an Call Center Environment
12:50PM 2 Allow calls to certain area codes
11:57AM 0 Asterisk dropping calls when transferred on another PBX
11:51AM 0 asterisk-oh323-0.6.3b and logical Channel
11:06AM 12 Faxing..not 100%
10:39AM 1 Restrict outbound calls on Broadvoice
10:11AM 1 asterisk and kphone (sip soft phone for linux) on same machine
10:01AM 1 IAX DIDs, Illinois
10:00AM 1 Problem on Outgoing Calls (FXO - SIP)
9:28AM 0 Broadvoice patch and latest CVS version
9:27AM 7 How to play messeage when user picks up the phone
9:09AM 0 Dropping calls, Polycom Renegotiation timeout?
9:05AM 0 Avaya 4606 IP Telephone
9:00AM 4 Fine Tuning
8:51AM 1 astcc needs AGI.pm...where is it?
8:40AM 0 Calls dropping, when server sysncs time?
8:37AM 3 SIP URLs
8:31AM 12 Analog FXO Woes Continue
8:21AM 2 Firewall traversal anomalies - AJA
8:12AM 3 TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
7:56AM 2 modprobe ztdummy - failed
7:48AM 0 save dialplan missing in 1.0.2??
7:44AM 11 Website that reads text recently on the list?
7:31AM 3 H.323 trunking
7:30AM 0 Skinny error : Unable to create channel
5:51AM 5 Linking asterisk to an existing small office PBX
4:50AM 5 Comdial PBX -- can use Asterisk as VM box?
4:35AM 4 Question about e1/digium
4:22AM 0 Mini-ITX Mainboard for Asterisk IP PBX, Intel Mobile Celeron 733MHz
4:02AM 5 Transfer on Snom 190
4:00AM 0 IAX2 Hangup Cause
3:52AM 0 callerid PSTN->IAX problem
3:36AM 1 chan_capi 0.3.5 does not compile
3:10AM 0 GrandStream BT VS. IP500 Latency
2:54AM 14 High(er) availability
2:42AM 4 gsm codec, very poor quality.
2:30AM 1 Strange softphone problem
2:24AM 3 Interface analogue exchange line to VOIP phone?
1:23AM 9 Voice mail problem
1:13AM 0 new version problems
12:31AM 0 OT: Two way trunks in Korea?
 
Monday December 6 2004
TimeRepliesSubject
11:45PM 3 Is anyone using Cisco 7905G phones?
11:18PM 2 Asterisk 1.0.3
10:56PM 0 MGCP Gateway
10:15PM 0 pstn <> asterisk -- pstn handled by asterisk box
10:08PM 0 ACT P104SLD (10 Line) phone - "Line Key Settings" ???
9:36PM 8 two questions
9:12PM 0 extension number when calling to registered gateway
8:02PM 11 DTMF via PSTN to * to IAX to * challanges.
5:11PM 2 Asterisk ---> Cisco AS5XXX sip one way audio
5:01PM 9 Kind of off-topic: VoIP services and multipl e callers
4:41PM 0 Phone Giptel G100 with Asterisk?
3:55PM 0 Kind of off-topic: VoIP services and multiplecallers
3:48PM 2 dialplan
3:40PM 1 Another "Unable to create channel of type 'Zap' (cause 0)" error
3:39PM 2 handset to sound card
3:32PM 0 CALLPROGRESS configuration for a X101P
3:25PM 0 Passing SIP digest auth to dialplan
3:11PM 4 I need very fast quick info how to setup ISDN card
3:01PM 8 Are there any digital phones that run on asterisk yet?
1:57PM 3 Kind of off-topic: VoIP services and multiple callers
1:50PM 0 T1 digit timeout when dialing manually
1:48PM 1 how to start with ISDN
1:30PM 0 ASTERISK -> SPANDSP
1:08PM 0 strange caller id and caller name with SIP and ATA186
1:01PM 1 retrieve_extensions_from_mysql.pl
12:14PM 4 Is this possible
11:49AM 1 Asterisk on Macintosh - no sound card support?
11:35AM 0 TDM OnHook/OffHook
11:29AM 0 Firefly prescence + Asterisk
11:04AM 0 Italian Caller ID support in zapata.conf
10:41AM 0 Useful information - UK exchanges and SystemX/SystemY and how ISDN works!
10:34AM 1 Broadvoice - bad quality, dtfm mode
10:30AM 0 How to verify if chan_sccp is working/built correctly?
10:12AM 2 Budgetone 101 phones ? SIP through NAT ?
9:25AM 2 Setting CallerID with ITSPs
9:24AM 1 G.711 Appendix II
9:23AM 5 PRI/Zap premature dialing problem
9:18AM 1 Console as extension problems
9:12AM 0 SoftPhone on * with X-Lite or iaxComm (1 X100Pcard)
9:04AM 0 UK callerid X100P?
8:27AM 0 Dropping calls on IAX2
8:06AM 1 Queue Timeout
7:58AM 0 CVS HEAD h323 no longer builds?
7:06AM 2 SoftPhone on * with X-Lite or iaxComm (1 X100P card)
6:29AM 0 What would I need to do this?
6:01AM 1 SIP response 302 "Moved Temporarily "
5:35AM 1 SIP status lagged
5:10AM 2 h extension in macro
5:02AM 0 fax/voice switch - faxdetect
4:11AM 3 Recomended ISDN for Asterisk ?
3:19AM 0 Voicemail Codec challanges.
2:55AM 0 auto-dialout not doing LCR
2:08AM 1 iax2 nativ bridge question?
2:07AM 1 Users list.
12:16AM 0 Is the list down, or is it just me
 
Sunday December 5 2004
TimeRepliesSubject
11:33PM 4 PRI configuration problem
10:41PM 2 Mysql-cdr
9:25PM 1 Hardware PSTN Gateways?
8:35PM 1 Re: Is Asterisk-users down?
6:34PM 3 List's quiet or down?
5:19PM 0 Dial D option not working?
2:30PM 0 Cisco IAD2421 with Asterisk
10:41AM 0 Sip Channels Left Open
10:05AM 0 Recomended ISDN on Asterisk@home ?
6:34AM 1 Group sip definitions?
6:19AM 3 ANALOG FXO ZAPTEL & WCFXO & WCTDM module issues seen with intermittent analog lines
6:11AM 3 asterisk + chan_sip2 + sipproxd + sipgate
5:33AM 0 just testing please ignore
4:53AM 12 G.729 algorithm?
3:29AM 6 full duplex sound card
1:35AM 0 Planet BRI TA will work ?
 
Saturday December 4 2004
TimeRepliesSubject
10:21PM 0 System hardware requirements for *
9:13PM 2 Billing - which program are you using?
8:49PM 0 Typical Setup for a small/medium office
7:24PM 4 Email to Fax?
7:21PM 7 BLOCKING incoming FAXES on voice line.
5:26PM 0 x100p offhook/onhook states
5:05PM 6 X100P does not detect ringing
5:00PM 9 Door buzzer.
4:57PM 0 Integration to TAPIT/Call tracking software
4:32PM 10 Is Gigabit Ethernet necessary?
3:49PM 0 budge tone 100 caller id
3:22PM 3 Broadvoice outbound 404 error
3:12PM 8 Is this possible?
3:12PM 6 asterisk dabbling...
3:03PM 1 more DIALSTATUS/HANGUPSTATUS woes with IAX2
3:00PM 5 Budgetone 100 Caller ID
2:54PM 0 Asterisk & Gossiptel - 1 way audio???
1:43PM 1 chan_zap.c:6181 mkintf: Unable to get parameters
1:01PM 3 Voicemail for Current Extension?
12:37PM 2 ISDN kernel 2.6 problems chapi isdn4lin
12:31PM 10 SJPhone SIP Tab
12:21PM 0 Remote-Party-ID + CallerID + VoicemailMain
11:33AM 0 NewBie Question Modem Telephone -PSTN
11:11AM 0 IAX Native Transfer
8:49AM 2 iaxy to iaxy call drops out of "show channels"
8:46AM 5 XML to monitor queues on Cisco display ?
8:42AM 0 (no subject)
8:09AM 1 Codec translator problem (g723.1,ilbc => alaw)
5:55AM 0 PRI debug - weird behaviour
5:35AM 0 PRI debug output - still not working :(
4:54AM 3 Udev setup question for zaptel
3:52AM 6 chan_misdn and Dynalink IS64PH ISDN
3:45AM 16 Asterisk and Cisco IP Phones
3:42AM 2 ZAP and IAX Trunks
3:34AM 0 RES: howto install
3:21AM 2 Snom 220 busy lamps [was: Receptionist phone...]
3:18AM 1 howto install
3:00AM 4 Gossiptel with Asterisk?
2:48AM 9 Incoming SIP Address?
12:47AM 0 Asterisk stumbling block
 
Friday December 3 2004
TimeRepliesSubject
9:20PM 2 compiling asterisk-addons for Mysql-cdr
8:09PM 0 Mixing x100p & te405p ??
7:18PM 1 Help with music over intercom.
5:57PM 9 Two zaptel T1 cards: no clock from one
5:47PM 5 SIP SECURITY WARNING: v1-0 (cvs today) sip context in general section ignored goes to default instead - allowing unauthorized sip devices to place calls in default context
5:16PM 0 transfer question
4:58PM 1 Call parking/transfer not working on IAX2 connections
4:13PM 0 Asterisk sms voicemail notification
3:39PM 2 7905G Firmware
3:07PM 0 IO-APIC
3:01PM 1 iaxy not hear ringing
2:30PM 5 Polycom 500, won't ring??
2:24PM 2 CAPI Newbie
2:17PM 11 Why, why, why???
1:36PM 12 DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)
1:35PM 2 FOP Asterisk Manager Login Failed?
12:43PM 0 ZAPHFC, Asterisk does not load with signalling = bri_net_ptmp
12:39PM 1 Umlaut over I on Definity display
12:39PM 10 Unable to create channel of type 'Zap' (cause 0)
12:02PM 2 Alpha Paging
11:25AM 0 feature suggest.: alt. include criteria
11:01AM 0 Initial Chirp while dialing
10:34AM 3 PolyCom MWI Chirp issue
9:27AM 2 Best VM codec for Linux/OS X/Windows environment
9:10AM 0 Digium+asterisk+festival+outgoingcall: How detect a busy line..?
8:57AM 0 ipkall & one way audio
8:33AM 20 Ouch, part reset, quickly
8:06AM 0 IAX2 Codec Pref order.
6:46AM 0 Incoming TypeOfNumber on zap, not just iax2?
6:20AM 2 HasNewVoicemail does not find voicemailbox, but files exist
6:13AM 5 Status of linux 2.6 support
5:49AM 1 Queue without #
5:45AM 0 RE: Asterisk-Users Digest, Vol 5, Issue 42
5:37AM 0 Testing Voip calls only
4:33AM 16 Bluetooth with *
3:51AM 0 Open G723.1 - problems.
3:06AM 1 SMS in Asterisk
2:22AM 0 How stable ist the Asterisk Fax Manager
1:28AM 0 Asterisk and MaxDB
 
Thursday December 2 2004
TimeRepliesSubject
11:54PM 0 Incoming SIP calls not being sent to "s" extension
11:22PM 17 Blank Machine Again.
10:52PM 0 Playing the message when user pickups the phone
10:45PM 0 Taiwan follows ETSI in permitting DTMF and FSK signals. ????
9:50PM 0 asterisk connection problem
8:10PM 0 ParkAndAnnounce Problem
8:09PM 1 Problems with analog line
7:21PM 0 park announcement not working Help!
7:06PM 0 can both chan_h323 and asterisk oh323 be installed on the same machine?
6:38PM 4 IAXy & ADSI ?
6:37PM 10 Codec Conversion
5:02PM 4 Very odd musiconhold
4:24PM 24 drive space for voice mail
3:39PM 0 Getting the right DST in CDR
3:37PM 9 Dial Command M(x) Option
3:30PM 0 Connection Problem
3:30PM 4 fallthrough extension.
1:52PM 0 E100P not starting?
1:06PM 9 No Files Seen via vmail.cgi
12:56PM 4 Multi-Line sip phone?
12:41PM 0 Newbie - Get IVR Informations
12:38PM 4 more than 3 msns with chan_capi
12:18PM 1 SpanDSP 0.0.2pre6 undef symbol on gentoo-ppc
12:02PM 0 Polycom POE Rumor
11:41AM 0 new asterisk installation report and request for mixed voice data apps
11:35AM 5 Asterisk Problem or Polycom Problem
10:12AM 0 [OT] Dutch Asterisk meeting
9:41AM 2 Sipura Blind Transfer - Help
9:40AM 3 firefly and caller id
9:40AM 0 (no subject)
9:37AM 1 Agent Login "Play a file"
9:20AM 2 threeway calling
9:19AM 2 IAX2 and TEXT
8:54AM 22 Asterisk crashes my router!?
8:35AM 1 GUI for Asterisk Configuration
8:22AM 5 Asterisk with SMS
8:21AM 23 Polycom 500, asterisk user opinions?
8:01AM 0 [OT] detect-string.pl
7:56AM 9 Ring all Configured Extension
7:50AM 6 Restarting *
7:30AM 10 Conference
7:26AM 0 ForkCDR app call disposition ALWAYS says ANSWERED??????
6:18AM 1 the pstn line is noisy, busydetect can detect hangup?
6:10AM 4 TE110P + Asterisk
4:44AM 0 Incoming call errors
4:37AM 0 transfering a incoming sip call automaticlly to another number
3:16AM 0 IAX to freshtel
1:15AM 1 900# DID?
12:54AM 0 Re: [Asterisk-Dev] One D channel for multiple spans
12:35AM 0 Newby with no idea
12:25AM 2 Getting a US Number
 
Wednesday December 1 2004
TimeRepliesSubject
11:34PM 6 SV: www.voip-info.org
11:31PM 1 www.voip-info.org
11:29PM 5 Voicemail - Danish, German an French audio files download?
10:54PM 0 asterisk version 0.7.1
10:28PM 2 Newbie Time
9:23PM 4 What exactly does IAX and SIP termination mean???
5:55PM 0 How to get transfer and blind transfer on 7905
5:25PM 0 transparent call routing
4:44PM 1 No version string
4:39PM 0 Diagnosing codecs
4:35PM 0 SIP->IAX->SIP silences
4:29PM 0 Asterisk / Paris Meeting
4:00PM 2 Hypothetical IAX2 situation
3:52PM 0 Interrupt Conflicts
3:26PM 10 Asterisk + Satellite connection
3:22PM 0 threeway calling while conferencing
3:07PM 2 IAX long distance... Re: Asterisk for home office
3:02PM 1 Micronet problem
2:23PM 37 ASTCC configuration problem
2:08PM 5 app_queue question
1:51PM 2 Sometimes calls are silent
1:03PM 1 [OT] [slightly] app lever vs driver level implementation...
12:44PM 0 Re: ASTCC
12:10PM 0 Caller ID showing My Own number
12:07PM 11 grandstream bt100 upgrade 1.0.5.18
12:02PM 0 setting up conference room option
11:48AM 4 conference room possible bug
11:12AM 6 dont write me again
11:01AM 0 Grandstream BT100 / HandyTone 286 and Level 3
10:48AM 16 Sveasoft Alchemy QOS
10:47AM 16 voicemail cuts off / hangs up
10:11AM 10 zaptel and low ring voltage
10:10AM 28 Interrupt latency problems
9:32AM 0 VoIP Dialout issues
8:52AM 1 some infos
8:41AM 5 Getting started with Asterisk
8:22AM 0 X101P interface (asterisk newbie)
7:56AM 2 Asterisk Call Monitor and soxmix error
7:41AM 1 CallerID on X100P in South Africa
7:36AM 3 SIP expiry time
7:20AM 5 Advantage of IAX2 to SIP?
7:00AM 4 PRI litmus test
6:59AM 9 Asterisk without D-Channel possible?
6:56AM 6 Polycom IP 600 status setting in Asterisk
6:52AM 0 (no subject)
6:43AM 6 Asterisk + AS5300
6:31AM 1 SPA-3000 and distinctive ring
6:19AM 3 Japanese FXO card
6:17AM 2 Sip no voice
4:22AM 0 sipgate x asterisk: problems to receive PSTN calls?
4:19AM 2 pre-installation jitters
3:42AM 5 software phones for Asterisk - is there a list?
3:33AM 0 Unable to open pseudo channel for timing... Sound may be choppy
3:24AM 12 Unable to open IAX timing interface: No such file or directory
2:26AM 2 Time announcement
2:15AM 0 ip2ip 302 response
1:56AM 14 Avoided deadlock
12:55AM 0 extension and PSTN connection
12:51AM 1 CVS-HEAD breaks iconnect