asterisk users - Sep 2005

Friday September 30 2005
7:23PM 0 How to get names into the *411 directory
6:53PM 1 is a dual 1.5Ghz server better than a single 3Ghz for a 100 Iax users asterisk server
5:52PM 2 Asterisk and RTP streams
4:33PM 1 Music on hold not initiating RTP stream?
1:17PM 3 SPA-841 "Decode Latency"?
12:57PM 4 Revieving some fax problems
12:28PM 0 Polycom IP301 Hangs on boot.
11:56AM 0 voip alarm circuit
11:53AM 1 Linksys register hangs Asterisk!
11:28AM 2 quick question on ztdummy
11:16AM 0 Co-author of O'Reilly's Asterisk book presenting in Utah Valley
11:02AM 1 X100p Problem, randomly hungup pstn line
10:46AM 0 oh323 implementation 0.67 has call-id problem
10:21AM 1 Best way to create IVR/voicemail system
10:17AM 1 (no subject)
9:52AM 2 SIP make outside call
9:51AM 4 C Manager Interface Client
9:41AM 1 Maximum number of Digium Trunk Cards
9:06AM 0 mISDN, HFC, W6692, one-way-voice problem
8:51AM 0 It is possible to have 2 AVM Fritz! USB for multiple BRI access?
8:47AM 0 Calls Dropping w/ Cisco 7960 Phones
8:26AM 1 Question about 3Com(r) 3101 Basic Phone
7:58AM 1 strange wave like noise on sip handset
7:55AM 1 Asterisk and telephone volume
7:53AM 1 No ringback tone generated by Asterisk with OH323connections
7:31AM 1 No ringback tone generated by Asterisk with OH323 connections
7:26AM 1 ?
7:20AM 7 911 Q
7:13AM 2 OT: SIPSAK usage
6:57AM 2 Echo Cancellation not working in Zapata.conf
6:47AM 1 Not Authenticate
6:37AM 1 TE410P not working
5:59AM 0 R: chan_capi-0.3.5
5:40AM 2 analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly
5:32AM 2 Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?
4:45AM 1 VideoConference with UMTS
4:43AM 1 Register times out on internet outage
4:36AM 2 chan_capi-0.3.5
4:12AM 2 Why does the s extension not work in my extensions.conf file
3:58AM 2 Diva
3:27AM 0 Compile broken on FreeBSD ?
2:35AM 0 [Fwd: TDM40B - "Unable to play dialtone on channel X" ?]
2:20AM 3 Zaptel TDM questions
2:02AM 1 Empty ACK
2:01AM 0 IAXPhone
1:19AM 4 G.729 patent in France
12:31AM 1 Siemens TC35 GSM gateway
12:14AM 0 * T.38 fax
Thursday September 29 2005
11:49PM 0 [Asterisk-User] linux/Asterisk change ip address
10:18PM 0 please help on FreeTDS (writing CDR to MS-SQL or MySQL)
8:49PM 2 Is this normal?
7:52PM 0 Can't make outside call with SIP softphone
5:56PM 1 Voice Prompts, what do you think? Good voice.
5:27PM 1 SIP Gateway wants T38, Asterisk rejects but media path not established.
4:08PM 1 Meet me conferencing without blind transfers (Asterisk@home)
3:29PM 1 Using Realtime queues and queue members
3:20PM 1 Mathematicians wanted (was RE: Best echo canceller?)
2:58PM 1 files conflict after CVS update
2:39PM 0 dtmfmode type
2:04PM 2 Best echo canceller?
1:52PM 3 Auto Answer Fax
1:39PM 0 FWD via Trunk from DMZ to LAN
1:30PM 1 Cisco AS5300 --> [SIP] --> Asterisk - NO AUDIO
1:28PM 3 Broadvoice inbound issues
12:59PM 0 TDM40B - "Unable to play dialtone on channel X" ?
12:34PM 3 Problems using SIPURA and MFC/R2
12:27PM 3 FWD: '486 Busy here' and 'All Circuits are busy now'
12:13PM 0 Yada table in oracle
12:12PM 2 Asterisk for "Man-In-The-Middle" Trunk Side Call Recording?
11:53AM 0 Asterisk as a Voice Logger alternative to NICE or Witness Systems
11:35AM 2 Hardware Specifications
11:31AM 2 Unable to send fax using BroadVoice
11:04AM 4 Any way to not overwrite sound files on compile?
10:42AM 0 DTMF tones from PSTN not reaching SIP device
9:15AM 1 minor(? ) Grandstream phone issue
7:44AM 2 R: PRI value
7:32AM 1 Cannot figure out why calls from my Asterisk appear to be from country code +34?
7:28AM 4 OOH323C
7:01AM 0 Prueba
7:01AM 1 Re: [Asterisk-biz] Problem with sending fax froma SIP extension
6:53AM 2 Remotely dialing calls from a polycom phone
6:49AM 2 Getting asterisk to send e-mail to mailbox-users
6:35AM 0 Caller ID, Attended Transfers, Polycom
6:13AM 4 chan_cap-cm-0.6 deflect support
5:53AM 1 Asterisk Echo problems, Urgent, please help,
5:49AM 1 Audio Files, Filtering, and Formats for Asterisk
5:17AM 0 Major bug solved in IPSwitchBoard
4:50AM 1 sip calleridnum
4:45AM 0 Asterisk registering with vonage
4:16AM 1 chan_cap-cm-0.6 is not working for incomming calls
4:02AM 1 Variable in call parking
3:00AM 1 digits won't play
2:38AM 2 Don't call
1:43AM 4 Calling voicemail from external phone.
1:36AM 0 Re: Asterisk-Users Digest, Vol 14, Issue 178
1:10AM 2 PRI value
12:36AM 1 zttest - 100% ?
12:19AM 0 Voice Prompts, what do you think? Good voice. Should we record a new prompt-set?
12:15AM 1 Dealt with IAreaNet before?
Wednesday September 28 2005
11:05PM 1 Recording channels
9:16PM 2 * mod core dump help
9:03PM 0 Recommended wireless router to run Asterisk on OpenWRT
5:38PM 1 Does the 1.0.9 release contain the Broadvoice patches?
5:35PM 2 chan_capi-cm, Euro ISDN bus: 2 extensions on same BRI port not working
5:08PM 0 Problem redirecting to voicemail through a SIP proxy (Looks like a bug)
4:46PM 2 TE205P in loopback?
4:17PM 3 cisco phones problems
4:07PM 0 ISO SIP Based Conference Bridge Solution
4:03PM 0 No audio non channels and choopy sound to PSTN network
3:12PM 1 Motherboard for Digium card
3:01PM 2 asterisk 1.0.9 + spandsp 0.0.2pre20 = crash on boot
1:40PM 0 Upgrading *
1:17PM 1 Can I install latest oH323 on *@home
12:49PM 0 To get phone to ring in two or more places
12:42PM 4 T.38 Faxing
12:18PM 1 Tiny Echo on PSTN via Zaptel
11:38AM 1 Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net'
11:18AM 6 Music on Hold Quality
11:18AM 2 Zap FXO/FXS issues, 1.2.0-beta1
9:47AM 3 ASTCC - INUSE Flag
9:17AM 1 Monitor in AGI
9:13AM 0 TDM-400 cards, technical limitations
9:11AM 0 DID's in CA, WA, BC, FL and NY
9:10AM 1 Correction: Asterisk sound files, audio bandwidth, and sound quality
9:06AM 1 Asterisk sound files, audio bandwidth, and sound quality
8:57AM 2 Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unkn own signalling method 'pri_net'
8:56AM 0 BAD echo problems with Sangoma and, Telstra
8:49AM 4 Delay in dial
8:24AM 5 Roll back from CVS Head to v1.09
7:53AM 1 adit 600 mgcp.conf
7:33AM 0 digital receptionist pick up time
7:26AM 1 Where MeetMe application
7:14AM 0 Trying to cut out the paper work...
7:10AM 0 Does Asterisk just pass thru RTP if the codec is the same between two extensions?
7:07AM 0 Does Asterisk just pass thru RTP if the codec is the same between two extension?
7:04AM 0 [Asterisk-User] Does Asterisk just pass thru RTP if the codec is the same between two extension?
6:34AM 2 setting up asterisk as an sms central?
6:28AM 0 problems accessing directory
6:09AM 1 Asterisk in Production
6:03AM 1 Asterisk does not send "Setup acknowledge" on euroISDN E1
5:14AM 15 Asterisk on windows
3:10AM 0 SV: Turn off echo-cancellation when fax is detected?
3:10AM 1 MeetMe error
2:50AM 0 call wating and call transfer
Tuesday September 27 2005
10:58PM 4 Voice Encryption
10:49PM 2 Auto CallBack on busy
10:33PM 1 oH323 Voice in one direction only
8:50PM 1 IAX2 encryption of data packets?
8:27PM 0 linux dist. and kernel version
7:02PM 0 7960 show queue status
5:23PM 2 Sipura 2000 Dial Plan
5:09PM 4 BAD echo problems with Sangoma and Telstra
4:19PM 1 Re: [Asterisk-biz] Problem with sending fax from a SIP extension
4:17PM 1 Extensions go straight to voicemail
2:43PM 5 Canada VOIP provider quality
2:17PM 4 Hook Flapping on Cisco 7960
2:00PM 1 Creating an OPX from a traditional PBX using Asterisk and a SIP device
1:38PM 0 AstriCon 2005 - Now With Free Beer!
12:40PM 2 Review: Digium TE405P v2
12:17PM 0 cgi-bin/vmail.cgi - - Invalid Context
11:54AM 1 SIP Tandem Inbound only.
11:46AM 0 asterisk@home inbound call problem to SIP trunk. (voipfone UK)
11:22AM 2 One-way audio with VPN
10:51AM 1 blindxfer & atxfer not working?
10:21AM 2 Polycom IP 500 - problem dialing extra numbers
10:18AM 1 [MSG]TDM Error on ASUS Pundit-R
10:10AM 3 analogue phone with asterisk
9:49AM 2 How to change ${VM_DATE} in voicemail.conf
9:20AM 1 VoIP Buster stopped working?
8:18AM 0 Asterisk & European Digital CAS Help
8:12AM 10 Software only Asterisk PBX (commercial)
8:04AM 1 Moaning dog...
7:51AM 0 405 "Method Not Allowed" error
7:45AM 0 function LEN missing
6:06AM 2 IAX2 hard phone
4:42AM 1 wait before accepting the call
3:33AM 1 R: Best drivers for HFC-S ISDN cards
3:23AM 0 Turn off echo-cancellation when fax is detected?
3:18AM 0 * Accounting with Oracle
3:06AM 0 radius and *
2:42AM 0 Listening for DTMF when dialling (sorry, accidentally sent the previous message too early!)
2:35AM 0 Listening for DTMF when dialling
2:23AM 1 R: Problem setting up TDM22B card
2:19AM 1 failed make install on Solaris 10
1:20AM 2 Integration with NMS AG-E1/T1
12:36AM 1 undefined symbol:
Monday September 26 2005
11:23PM 0 "Non-blocking" Dial (and other commands): is there a way?
11:22PM 1 Bad FCS nightmare to Nortel SL100 with TE410P
9:22PM 1 IAX provider w/Toronto & Detroit termination
9:08PM 5 SPA-3000 and incoming faxes
8:41PM 0 ICD with asterisk
8:39PM 0 asterisk fifo
8:27PM 1 StripMSD or extension parser bug?
8:10PM 0 Flash Panal
7:30PM 0 system() app changed drastically! How do I useit now?
7:27PM 3 re: DTMF woes, continued
7:14PM 1 AsteriskJava - Queue
7:10PM 0 system() app changed drastically! How do I use itnow?
6:15PM 1 system() app changed drastically! How do I use it now?
6:01PM 0 Faxing via a sip extension with a digium e1 card
4:47PM 0 TE110P Hanging up & sometimes not picking up on E&M T1
4:09PM 1 Socket 478 Motherboard for use with TDM400P
3:33PM 0 CPU spiking with TDM400 cards fixed
3:26PM 1 voipbuster advise
3:18PM 0 netappel
3:00PM 0 Asterisk Realtime.. : Unixodbc drivers
2:39PM 2 What ISDN hardware would you recommend?
2:32PM 4 Polycom Setup Questions
2:31PM 0 ZapHFC Channel unavailable
2:24PM 1 Grandstream 496 not working on cordless phone
2:17PM 1 how to connect two SIP channels
2:10PM 0 Areskicc LCR problem
2:03PM 0 Performance tuning on dual Xeon EM64T and x86_64 Linux
1:27PM 1 Dialogic Cards Will they be available to NON AsteriskBE
1:06PM 3 IBM x306 - some progress
12:35PM 1 FSX/UK analogue Phone rings all the time
11:38AM 0 CAS Question
11:28AM 1 Re: Ring requested on channel already in use
11:25AM 1 Carrier Access - Access Bank I config
11:17AM 3 asterisk SMS and sprintpcs
10:56AM 6 Extension availabilty
10:48AM 2 Early Media in 180 Ringing
10:46AM 0 IptablesAsterisk
10:43AM 0 ? In CLI not working
10:37AM 1 goiax caller ID
10:07AM 3 Sangoma and Digium same machine?
10:00AM 1 Early Media in 100 Ringing
9:33AM 1 I want to send oH323 calls to our Quintum D3000 which is connected to a PSTN
8:51AM 0 BRI ISDN on USB
8:25AM 0 Recent Sphinx integration work?
7:37AM 0 CheckGroup accross multiple servers
6:45AM 1 Call Back On Busy?
6:41AM 2 Subject: Vonage-type service
5:30AM 0 Asterisk::AGI - What license ???
4:37AM 0 Will Digium Wildard work with PCI-Xor PCI Express
4:05AM 0 dialing selected text with asterisk under windows ...
2:35AM 1 Date based context inclusion
2:08AM 1 sip, call ransfer and call waiting
2:03AM 1 IAX Registry problems
12:55AM 1 VOIP in Japan using Freebit
Sunday September 25 2005
11:22PM 2 change codec based on callerid (sip/iax)
9:55PM 0 compute traffic intensity from CDR?
9:48PM 1 Can an outside caller dial an extension before someone answer?
9:30PM 3 TE405P V2 - Fantastic!
6:54PM 0 Emergency Asterisk Guru help needed -- Yucky sound with MOH
6:06PM 3 Vonage-type service
5:38PM 0 Unable to Transfer an outbound call
5:31PM 1 Digium T-1 and FXO cards for sale
5:31PM 0 Cisco phone ports
12:48PM 1 WRT54GP2 SIP server on LAN port
11:04AM 2 Pager Notification Script
10:25AM 0 CALLERID to Sipura Devices (or others for that matter).. CVS-Latest Version
10:18AM 0 VPB Driver Question
8:19AM 0 pound/hash key not recognized
6:27AM 0 Problem Asterisk: can't make call but can receive calls
3:32AM 2 iax problem
12:48AM 1 Codec routing?
Saturday September 24 2005
10:46PM 2 Extension Mobility (roaming) Cisco 7960
10:08PM 1 dialplan game
10:06PM 4 didgium card in india
10:00PM 0 IPSpeedDial has just been released
8:22PM 2 CDR problem
7:23PM 1 Cheap Time sources which is best?
7:00PM 0 Software to generate an SRTP key pair?
6:45PM 1 Need good explanation on contexts and extensions
5:51PM 3 IBM x306
5:49PM 0 PA1688 Phones using IAX MWI
4:55PM 0 Pictures from VON Fall 2005 Digium/Asterisk booth
3:29PM 0 Falsh Panel in Xorcom Rapid
3:28PM 2 Directed pickup syntax?
12:21PM 2 Send DTMF after call bridge
11:25AM 1 ASTCC on Fedora 4 and MySQL 4.1.12
10:42AM 1 unable to use misdn group dial
9:29AM 2 Asterisk returns 484 ADDRESS INCOMPLETE for incoming SIP calls
9:19AM 1 Help!! trying to use an MTA
8:55AM 0 BT100 can't register
5:11AM 0 HP DL360 G4 EM64T and hyperthreading options
3:14AM 0 Seperate siptrunks
1:19AM 1 wrong password on authentication for INVITE to '"asterisk"
12:22AM 0 Do Sifira use Asterisk?
Friday September 23 2005
10:22PM 1 Message Waiting Indicator (MWI) for remote voice mail?
9:53PM 0 Is background() fax detect broken?
9:18PM 0 delay SIP answer
8:18PM 1 Skye gateway?
6:14PM 1 context question
3:56PM 1 Wildcard TE110P in Mexico
3:31PM 0 DTMF detection problems.
2:27PM 1 RE: [Asterisk-Dev] Open source time card application for Asterisk
1:46PM 0 X-Lit not picking up callgroup call with *8
1:44PM 1 FW: channel offhook state
1:38PM 1 Play sound on connect
1:14PM 1 Asterisk CMD MySQL
12:58PM 1 Asterisk - Dying Signal 11
12:22PM 3 Removing "-" (Dash) from Dialed Numbers
12:15PM 2 Can't receive Faxes with Asterisk (help)
12:03PM 0 Call Queue ANI
11:38AM 4 CallerID issue
11:37AM 2 asterisk invitation problem
11:28AM 2 Continue dialtone after pressing 9
11:12AM 4 goiax expanded with free us domestic calling
10:18AM 0 voicetronix openline4 comments
9:53AM 1 ChanSpy performance sub-optimal
9:17AM 2 ZAP ISDN losing digits
9:13AM 0 Trunks greyed-out on Flash Operator Panel?
9:10AM 1 retry times
8:58AM 0 voicemail operation modification
8:22AM 0 Problem with outbound calls
8:15AM 2 Problems with queue and remote agents
7:35AM 2 Execute php agi after channel hangup
7:26AM 0 RE: SNOM 190 '486/Busy here' after upgrade to re 3.60s
7:23AM 0 DTMF translation
6:04AM 0 SIP Hangup via Call Files
5:55AM 1 ztdummy compile again
5:44AM 1 dial (iax/X&sip/y) get y fraction earlier
5:12AM 1 Double cpu
5:03AM 2 Dialtone problems with phpagi and asterisk
4:27AM 1 chan_capi-cm-0.6: hangup is detected really late
2:54AM 1 Dial multiple phones
2:48AM 10 Problem setting up TDM22B card
2:19AM 6 Which codec?
2:14AM 1 Dial() and BackGround()
1:41AM 1 zaphfc problem: overlapdial don't work after update bristuff
1:39AM 1 Fax detection question
12:28AM 0 Hangup when dial via Mobile Interface
Thursday September 22 2005
11:58PM 0 Keytouch without effect
11:05PM 0 SNOM 190 '486/Busy here' after upgrade to firmware 3.60s
10:59PM 2 Asterisk + GNUGK + Asterisk-Addons ooh323
9:16PM 0 CVS-HEAD and Caller ID -- Pulling my hair out!
8:47PM 1 SayUnixTime in CVS?
7:12PM 2 Recently reported ASTCC audio issues
6:17PM 1 anyone know about this company?
4:21PM 0 Extended SIP registration failures
4:12PM 0 SNOM 190 '486/Busy here' after upgrade to firmwa re 3.60s
2:49PM 0 priindication passthru TE410P EuroISDN?
2:32PM 0 problems with sending fax from SIP channels
1:45PM 0 rtp problems
1:33PM 2 Set Log Level for Messages log file
12:15PM 1 Will Digium Wildard work with PCI-X or PCI Express
12:05PM 0 logging in problem
11:15AM 4 Polycom IP500 Quickstart page or files?
10:49AM 1 WaitExten
10:39AM 0 OT: Sangoma A102u available
10:25AM 1 externpass
10:15AM 0 cdr_custom?
10:10AM 12 custom ring tone
10:05AM 0 AGI Script to interact with ACCESS Databse a nd Set CID info on the fly.
9:52AM 3 AGI Script to interact with ACCESS Databse and Set CID info on the fly.
9:10AM 0 Hardware Recommendations for Junghanns card QuadBRI PCI.
8:00AM 0 Multiple SIP Phone Calls Overlapping on the Same Phone
7:31AM 0 ASTCC error when using silent=5
6:40AM 1 Initial release of AMPortal Debian/Xorcom-Rapid packages
6:34AM 0 Call Pickup issue
6:18AM 1 Asterisk with
5:52AM 1 AgentRecord In and Out streams
5:18AM 2 SOHO Survey / Creative Asterisk Solutions
4:44AM 1 IAX client for Linux text console
3:26AM 1 Early Media with Asterisk
12:29AM 1 Compile problems on Solaris SPARC
12:26AM 1 Any problems with Asterisk and "nice"
Wednesday September 21 2005
11:48PM 2 Submitting ISDN-MSN from a SIP-Phone
9:44PM 2 Web based application for call History
6:53PM 0 new spandsp-0.0.3pre1 missing tx and rx fax apps?
6:36PM 1 I got "403", "Forbidden"... please help
6:28PM 3 Cisco AS5XXX + CallerID Name
6:13PM 2 down?
5:22PM 5 Tux/Asterisk logo for Cisco phones
4:28PM 0 Soyo Phones Crashing
2:56PM 4 WMI problem
2:24PM 0 Asterisk Platform - Success Strories - iAreanet in the news.
1:34PM 4 POP3 and TTS (Festival?)
1:10PM 0 Callprogress and TDM400 in Brasil
1:04PM 1 Problem with meetme monitor (recording)
12:58PM 1 Asterisk and a SPA3000 behind NAT peer registration
12:51PM 0 Problem with monitor application meetme
12:33PM 0 IAX2 vs SIP Phones and adapters
12:31PM 1 Problems with sipura 1001's and 2002's
12:25PM 0 problem with monitor meetme
12:21PM 0 re: Problems with Queues
12:10PM 2 Get SIP to work over very limited network access
11:46AM 1 oh323 driver and RFC2833
10:52AM 3 How can i call to a cellphone here in Mexico?
10:11AM 1 Weird Over Lapping Asterisk Calls via SIP Phones
9:21AM 0 is possible connect?
9:13AM 0 ODBC Voicemail WEB Retrieval V1.1
8:39AM 0 HOWTO: A simple AGI application to modify incomi ng CallerID on the fly using SQL Server and *not* UnixODBC
8:38AM 1 Addendum to Problem with Queues question
8:35AM 2 Problem with Queues
8:27AM 2 Macro exists if an application returned -1
8:24AM 0 qualify=yes
8:07AM 3 Caller ID and Call Parking on an analog PSTN line?
8:03AM 1 Does Asterisk know if the trunks are busy?
7:56AM 2 ISDN-forwarding to intern without cost?
7:51AM 0 Cellphones and Asterisk Bluetooth
7:50AM 0 HELP: E1 ChannelBank and UniCall
7:28AM 2 maximum concurrent ZAP channels .... max conf ports ...
7:26AM 1 Ask for config files of Nortell Meridian Op 11 & Asterisk for PRI
7:24AM 0 Using *0 to flash an external trunk on bridged channel
7:17AM 0 permit syntax question
7:17AM 0 Packetization period for CODECs
7:10AM 7 add 0 (zero) to incoming callerID - how?
5:12AM 0 First release of the Asteriskguru Operator Panel
4:35AM 0 IAX2 registration
2:54AM 4 How to retrieve voicemail from an IP phone?
2:23AM 0 Intermitant delays on call setup.
1:22AM 1 Call getting disconnected in queue
1:14AM 0 DID problem with calls from analog to ISDN
12:56AM 0 Brand New IPSwitchBoard
Tuesday September 20 2005
11:45PM 1 Asterisk PBX
11:16PM 6 iax2 trunking wackyness
11:03PM 0 Phone lines
10:24PM 1 automon wav format problems
10:01PM 0 Anyone using Asterisk to take credit card payments?
9:03PM 0 Can I connect an IAXy to my Panasonic PBX?
8:41PM 0 DIDx
7:45PM 1 HooDaHek w/AST 1.2
7:45PM 0 ODBC VM Playback from Web Page
4:57PM 0 Handling SIP 404 event
4:56PM 1 cvs-head and unicall with r2mfc
4:46PM 3 sipuras 841 bad sound
3:38PM 1 MOH failures (bad quality with interruptions)
3:37PM 4 SUCCESS - 512 Simultaneous Calls with Digital Recording
3:16PM 0 TE110P hybrid configuration for data and voice
2:33PM 0 fixlocalprefix error
1:16PM 1 [Fwd: ASTCC speaks and cut RTP channel, => Kind of solution...
1:06PM 5 MySQL and Asterisk
1:03PM 1 Asterisk vertical service activation codes
12:52PM 2 Snom-320 badly garbled audio
12:09PM 3 [ANNOUNCE] chan_capi-cm-0.6 released
10:54AM 0 agent channel busy - how to stop it?
10:37AM 1 ODBC Voicemail WEB Retrieval
10:30AM 4 how to distinguish the "ringing" and "connected" for zap channel
10:03AM 0 BackgroundDetect problem
8:50AM 9 HooDaHek 0.6 Released
8:34AM 1 one way voice
8:07AM 0 Aterisk App ICES Question
7:02AM 0 using a voip cable modem
6:58AM 0 Asterisk@Home Music on Hold
6:35AM 0 Red or Yellow alarm monitoring
5:48AM 0 What hardware would you recommend?
5:34AM 0 General Config information
5:30AM 0 asterisk-oh323: New versions 0.6.7 and 0.7.3
4:35AM 0 Connect not signalled (SIP -> Zap)
4:21AM 0 HELP: Valiant E1 CB and UniCall
4:05AM 1 Is there a clever way to page a group of extensions?
4:03AM 0 sipp examples
3:56AM 0 PTN calls into asterisk slow release
3:23AM 0 Hangup after voicemail not detected
2:15AM 1 Cisco 7960 Locking Up
Monday September 19 2005
11:48PM 1 Resolving QOS problems
10:27PM 1 "Stopping retransmission on" messages
10:24PM 0 TE410 stop responding
8:24PM 1 Buy a digium hardware
7:43PM 2 MWI indicator HINT on Snom thru IAX?
6:37PM 1 need example about sjphone with asterisk
5:34PM 0 Call dropped 100% of time when incoming IAX routed to outgoing CAPI
5:10PM 1 Re: Welcome to the "Asterisk-Users" mailing list
4:37PM 1 Zap calls dropping just after answer
4:34PM 0 Voicemail() application returning -1 on a hangup
3:52PM 0 MSNs don't work for me... :(
3:48PM 1 hfc card unplug & plug not working?
3:07PM 0 pridialplan per call or per channel group?
2:38PM 1 [Fwd: ASTCC speaks and cut RTP channel => Kind of solution...
2:37PM 0 H.263 Format video
1:57PM 1 Point to Point with Fritz Card ...
1:53PM 0 Dial time limit doesn't work when calling party transfers
1:24PM 1 Silence suppression /RTP VAD and Asterisk? Dropped calls on IP-500
1:21PM 3 T.38 & Canreinvite (yes, again)
12:53PM 1 Asterisk Keep Crashing need Help please
12:41PM 1 Most desireable Linux distribution for Aster isk?
11:58AM 1 Asterisk monitoring availability
11:44AM 4 IAX dialplan problem?
11:43AM 2 kill a .call file
11:42AM 2 Looking for firmware for Cisco 12sp+ and 30VIP
11:05AM 1 Most desireable Linux distribution for Asterisk?
11:05AM 0 HooDaHek Version 0.5 Release
10:53AM 2 ztdummy configuration help
10:09AM 4 Pinging ...
9:57AM 0 Sip and ISDN problem
9:49AM 1 Complete NPA-NXX list for USA/Canada npanxx,
9:48AM 2 hints and the sNOM 360
9:38AM 4 VM low volume - testers needed
9:30AM 1 i4l ring indication problem, again...
9:22AM 6 SIP audio port usage
9:01AM 0 Anyone have the firmware for WRT54GP2?
8:59AM 0 hints not working on CVS HEAD
8:45AM 3 OT: Hardware Interrupts; Who is it?
8:45AM 0 Asterisk ISDN: Problem Setting CallerID as DIDExtension Numbers.
8:41AM 1 OT: Xoops Skype module
8:27AM 0 Round-robin with Queue
8:11AM 0 sip invite question
8:10AM 0 Unable to open space (format ulaw)?
8:07AM 0 chan_alsa.c blocking sound port - solution
7:54AM 0 FW: ADTRAN Virtual Classes: Ensuring QoS for VoIP & Total Access 900 Series
7:05AM 1 problems with PRI
6:13AM 0 clear SIP channel
5:53AM 1 Prompt translation: can't find "please wait try ext" prompt filename
4:49AM 0 need a simply configuration for calling in/out to PSTN
3:43AM 1 Voipbuster in Australia -- delay problem
3:01AM 0 ISDN BRI 2 pci cards and mISDN
12:53AM 0 problems with remote access to PSTN
Sunday September 18 2005
9:28PM 7 Cisco Callmanager & Asterisk for Voicemail revisited
9:27PM 2 HW Question (TDM400)
9:12PM 5 Monitor and sox mix quality
4:43PM 1 sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?
4:00PM 0 Julien COURTEMANCHE/TELINTRANS/FR est absent(e).
3:39PM 6 Differ between "private" and "out of area"?
3:27PM 0 ChanSpy not loading
12:12PM 1 Re: Asterisk-Users Digest, Vol 14, Issue 108
12:09PM 2 limiting calls per day based on amount of time
10:24AM 1 Two POTS in, but only want one out?
8:52AM 2 Asterisk Won't Process Call
8:03AM 0 voicemail context. macro, and directory
7:26AM 0 (no subject)
7:15AM 1 TFTP and DHCP...
4:24AM 1 DID from an analog phone
Saturday September 17 2005
9:52PM 2 Complete NPA-NXX list for USA/Canada npanxx, ratecenters, etc (attached)
7:17PM 1 Who is going to AstriCon (TheAsteriskConference)?
6:38PM 1 How does one set-up incoming/outgoing SIP with no registration and only IP authentication?
5:33PM 0 bounty partners and/or possible coder? queues.conf ackcall and pre-ack announce
5:17PM 1 unlocking cisco 7940 phone
2:31PM 2 checking voice mail from different phone
2:26PM 0 Anybody using SIP Interaction Proxy 2.X and Asterisk CVS head?
1:09PM 2 moh - turn off randomization?
12:17PM 1 capiFax causes segfault on asterisk
11:30AM 0 (no subject)
11:05AM 22 AstriCon 2006 Location
6:07AM 1 Flash Operator Panel Help
5:41AM 2 MGCP service from Free Télécom
2:31AM 2 AgentCallbackLogin and calling outside
Friday September 16 2005
9:52PM 1 How to make Basic authenticatuion in Asterisk server.
6:13PM 11 wav instead of gsm for vm-sounds?
6:03PM 8 Who is going to AstriCon (The Asterisk Conference)?
5:51PM 0 free IAX calling platform
5:25PM 15 Double Ring
4:45PM 1 Grandstream
4:23PM 1 TDM400P Dialing Out - "Cannot be completed as dialed"
3:06PM 2 Orinoco Injectors
3:01PM 0 linux sip or iax phone that will autoanswer and route to console
1:57PM 0 Anyone using iPlan Networks in Argentina?
12:51PM 5 How to access * thru router when ip address is not known
12:38PM 0 asterisk mixing sound card with anybody?
12:10PM 0 Weird behaviour
11:27AM 0 lastest spandsp-0.03pre1 don't compile
10:23AM 0 Zap failed
10:12AM 1 Sipura 2k voice quality
9:48AM 1 Easier way for end user to change main greeting?
9:14AM 2 R: direct sip call pickup
9:08AM 1 New version of idefisk softphone released.
8:56AM 7 mpg123 on x86_64 (Opteron MP)
8:19AM 1 direct sip call pickup
8:07AM 0 alsa issue with asound.conf
8:06AM 4 queue_log on mysql
7:43AM 0 SIP port assignment for user agents registering to Asterisk.
7:23AM 4 Caller Name: Asterisk reading too fast
7:09AM 0 Extension Restrictions
7:04AM 0 broadvoice incoming caller ID is wierd when calling from voipjet
5:45AM 1 7 digit dialing to e.164 format
2:15AM 0 How to suppress Local/Zombie channels?
2:00AM 2 Call Forward - 7940 Asterisk - Help
12:56AM 0 Unable to create ZAP channel - All circuits are busy
12:52AM 0 Wildcard TE110P
12:29AM 0 auto restart
Thursday September 15 2005
11:27PM 0 Transfering from a device to a queue crashesAsterisk
10:26PM 2 Help on RealTime Extensions on Oracle DB
9:25PM 0 Changing the sip port in sip.conf does not work
9:25PM 0 Send SIP NOTIFY frequency
8:44PM 0 Sip recording
7:58PM 2 Is digium supporting new te405p and te406p install?
7:15PM 0 triggering automatic dial-outs with Zap interface
7:01PM 1 Asterisk and Zyxel Prestige 2000W_v2
6:42PM 2 SIP reinvite asterisk and NAT
6:04PM 1 ZyXEL P662HW / SIP / Crashing
5:41PM 3 USB Phones for use with Asterisk
3:26PM 2 Asterisk CDRs
2:13PM 0 Console/dsp and mplayer
2:02PM 2 Caller ID for auto outgoing calls
1:29PM 1 Faxibility in NZ
1:17PM 2 Still having hangup problems in NZ
12:38PM 1 Unable to call some numbers with I4L
12:21PM 0 dialing sip before answering pstn line
12:07PM 0 If call fails, then try again with something else
12:03PM 0 Call Pickup between ZAP and SIP technologies
11:21AM 3 internet connection between Africa and Europe
10:42AM 1 Can not get realtime static voicemail.conf to work
10:28AM 0 Polycom oddities: Mixed up digits -> *8 Call Pickup
9:04AM 3 Seperate Incoming calls on TDM02?
8:57AM 0 Comfort Noise Generation with Zap-IAX
8:50AM 0 Siemens Hi-Path help
8:28AM 0 Transfering from a device to a queue crashes Asterisk
8:19AM 1 Getting email of voicemail to work
8:09AM 2 Fax->Email for Hosted PBX
7:53AM 0 Configuring GR303 trunks from Asterisk to a Taqua/TEKELEC T7000
7:48AM 1 Don't install asterisk-chan-capi
7:24AM 1 USB ISDN (OT question)
7:24AM 2 Asterisk CDR information into Oracle DB
7:10AM 5 Asterisk don't start
7:07AM 0 dialplan to try VOIP providers if they can't terminate call
6:24AM 2 cdr server
5:58AM 0 linux kernel tweaking for Asterisk
5:29AM 3 MusicOnHold not working
5:27AM 0 TE110P - Asterisk@Home Install Problems - Televantage 3 T1
5:03AM 0 Looking for China DID
4:18AM 0 Why isn't 3-way calling a standard feature?
4:09AM 0 No sounds on Playback()
4:07AM 0 TxFAX don't work
3:24AM 0 SIP rogue channel
3:08AM 1 iax phone and asterisk server on different LANs
3:06AM 4 PSTN calls are quiet
2:59AM 0 Incoming / Outgoing call problems on TDM card.
2:33AM 0 AW: ***SPAM*** actionID on manager events
2:28AM 3 ${DIALSTATUS} problems
1:46AM 1 Originate not understanding 2 vars in setvars
1:12AM 0 SV: RxFax problems
Wednesday September 14 2005
10:27PM 0 compile problems with yada
8:54PM 2 Starting From Scratch
7:52PM 0 Cannot hear teleco side error message
7:04PM 1 Liquidation: Cisco; Polycom; D-Link; MediaTrix, Colubris - Highly Reduced Prices
6:16PM 1 Distinctive Ring Tones
4:45PM 0 How to uninstall
4:25PM 0 Weird SIP behavior or I need a shrink?
3:22PM 0 Interop with Cisco T1/PRI on the 2811 and PSTN
2:48PM 1 Routes IPSEc And Asterisk.
2:03PM 1 RE: Asterisk-Users Digest, Vol 14, Issue 86
1:13PM 0 Compile error on cdr_yada for asterisk on centos with Oracle
12:58PM 0 # dialplan not working...
12:15PM 0 ${VM_CIDNUM} shows up but ${VM_CALLERID} & ${VM_CIDNAME} don't?
11:42AM 4 Echo on SPA-3000 FXO
11:38AM 1 ASTCC issues
11:25AM 0 sox conversion has introduces background hiss for both 8k and 41K recordings to gsm
11:10AM 11 RxFax/TxFax - Compile Problem
11:01AM 1 Asterisk Consulting Project ISO Hired Gun
10:57AM 1 Indications for Ireland
10:35AM 1 Re: Polycom randomly fails outbound calls,
9:56AM 0 RES: How to create IVR menu and transfer to anothersip extensions.
9:50AM 0 Anyone knows how to receive a SIP call withoutregistering gateway?
9:46AM 0 RxFax problems.
9:34AM 7 Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
8:45AM 1 TE110P - Asterisk@Home Install Problems
8:36AM 1 Asterisk as a gateway. 'flash for transfers transparency?'
8:10AM 1 IAX Registration with servers
8:09AM 3 Asterisk 1.0.9 long term stability
7:53AM 1 SMS using a PRI channel
7:46AM 3 (no subject)
7:24AM 1 timeout with queue
7:22AM 0 MAX PRI for single server (was:Not enoughlinesavailable for Asterisk implemetation)
5:54AM 0 Dial Application Return Codes - Help needed
5:51AM 2 STUN vs NAT Helper
5:40AM 2 PRI to PRI passthrough with DID intact
3:58AM 6 T.38 ATA
1:47AM 1 call restrictions
1:47AM 0 oh323 and Asterisk: Calls always hang up
1:05AM 2 pri release cause code mismatch
Tuesday September 13 2005
11:20PM 1 Anyone knows how to receive a SIP call without registering gateway?
10:33PM 0 spandsp frame slip tolerance.
10:15PM 0 Zap Clocking - Frame Slips - tdm400p wcfxozttest cpu spikes spandsp
10:01PM 1 Limiting call minutes on a GSM SIM
9:50PM 1 slight echo via sip provider
8:27PM 0 PRI zap channels not cleared when nomatchincontext for dialed number on inbound call
8:25PM 1 PRI zap channels not cleared whennomatchincontext for dialed number on inbound call
8:11PM 0 PRI zap channels not cleared when no matchincontext for dialed number on inbound call
8:07PM 0 PRI zap channels not cleared when no match incontext for dialed number on inbound call
7:56PM 0 PRI zap channels not cleared when no match in context for dialed number on inbound call
7:49PM 3 Call Wrapup time for agents.
7:28PM 2 Digium Cards in Australia
7:26PM 1 wctdm, issue w/outbound calls
7:09PM 1 Asterisk@home with Eyebeam
6:21PM 1 populating asterisk realtime tables from configfiles
5:39PM 0 populating asterisk realtime tables from config files
4:37PM 0 CVS vs CVS-HEAD
3:45PM 0 TDM400P stops answering
3:37PM 1 callfile: How to invoke SetCallerPres ?
3:35PM 1 make * listen on a specific ethernet interface
3:17PM 0 callfiles: how to query current dial attempt nr in extensions.conf?
2:52PM 1 How to IGNORE distinctive ring
2:43PM 1 Cisco AS5400 Configuration as a SIP Peer - URGENT
2:37PM 0 First PRI Installed - WOOT
2:22PM 0 MTA V102
2:09PM 1 sometimes dtmf passed, sometimes not (cisco 7960 SIP)
1:49PM 1 Oh323 and Asterisk with MERA
1:44PM 0 Integration Nortel x Asterisk
1:35PM 1 Dialplan Design Q
1:15PM 1 disable chan_skinny and chan_oss
12:37PM 1 Not able to access asterisk from internet via ip-forwarding
12:24PM 0 ZoomTel x5v Model 5565: is it any good?
12:21PM 4 Fedora Core 4 not recognizing X100P cards
12:20PM 0 AMP created extensions busy when dialed.
12:03PM 1 Polycom IP500 Mass Configurations
11:33AM 0 Asterisk + NEC IPK 192 integration
11:32AM 1 translate letters into digits
11:22AM 1 asterisk hangup detection on a pbx analog port]
11:12AM 0 Can anyone explain why this is happening? Odd CUT Problem
10:47AM 5 How to create IVR menu and transfer to another sip extensions.
10:08AM 1 FW: Nat & Sip & Pain
9:26AM 1 TDMoE Configuration problems
9:17AM 2 actionID on manager events
9:13AM 1 problem with FXS module
9:01AM 0 asterisk callerid problems
8:31AM 0 Bristuff version for use with 1.2.0beta1
8:09AM 0 Real-time Linux claims single-digit microsecond responsiveness
8:05AM 2 passing variables to h extension
7:46AM 1 SetCIDName question
7:24AM 0 Micro-cuts in MusicOnHold
7:01AM 0 [Re: civil emergency comms: Asterisk + HAM]]
6:14AM 1 Integration between Asterisk and Siemens HiCom 150e over ISDN
5:47AM 0 show queue callcenter output?
4:50AM 1 Monitoring status of ISDN lines
4:44AM 2 Nat & Sip & Pain
1:55AM 0 Coexistence of zaphfc and hisax?
1:00AM 1 2 box single Asterisk
Monday September 12 2005
10:18PM 0 get dialstatus variable when returning No such context/extension
10:18PM 1 Phonecall or something as robust
8:31PM 1 Is "ChanIsAvail" thread safe?
7:22PM 0 High system load and system freezes
6:23PM 0 Subject: '#' dialplan pattern matching
5:10PM 2 Firmware upgrade Aastra 480i CT
5:08PM 0 Whisper Mode
5:03PM 3 monitor peak channel use
4:47PM 0 early dial (grandstream bt100)
3:10PM 1 Meetme Dial Out
2:21PM 2 Stupid tricks: preventable?
1:59PM 5 What have I misconfigured?
1:53PM 5 OT: Online TTS engines?
1:49PM 1 compile error with postgres and voicemail
1:39PM 1 LiveVOIP - I win :)
1:33PM 4 CallerID Name in dialplan
1:17PM 0 WaitExten?
11:55AM 0 New York Asterisk User Group - Established
10:08AM 13 Skype purchased by Ebay 2.6 Billion
9:44AM 1 optimizing for via C3
9:26AM 1 AW: Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card
8:58AM 1 callfiles: set variables ?
8:42AM 0 Re: Asterisk-Users Digest, Vol 14, Issue 70
8:34AM 0 [Fwd: SwissVoice IP10S not able to dial calls with protocol SIP]
8:32AM 2 Callerid fails in any release after beta1 fails
8:17AM 0 Canspy listening to SIP channels
8:11AM 1 Other Voicemail systems
7:57AM 1 Callerid UK patches (from Lusyn)
6:38AM 1 Can't pickup inbound calls with TDM400P Fxo
6:19AM 0 Voicemail Not Recognizing user and password?
6:12AM 1 wctdm module won't load after kernel upgrade
5:41AM 1 Montreal asterisk usergroup meeting today 6pm
4:53AM 4 Hotel Setup?
4:46AM 3 Asterisk Registration as Client to OpenSER
4:07AM 0 Configure asterisk to dial user and notify if new voicemail
3:25AM 2 Zap Channel
2:22AM 1 chan_zap.c:8050 pri_dchannel: Ring requested on unconfigured channel 255/255 span 2
2:05AM 0 asteriskathome and cisco 2600
1:54AM 0 ChanSpy with asterisk 1.0.9
1:32AM 0 tdm400p wattage
12:50AM 0 Sip phone will not connect
12:34AM 1 How to remove the voice mail greeting...
12:12AM 2 Hang up not hanging up (New Zealand Indications??)
Sunday September 11 2005
9:50PM 2 Asterisk and AMP installed now what?
9:16PM 1 Anyone using Telasip, Caller ID presentation outbound??
8:39PM 0 extensions.conf for VOXEE using SIP!!
8:15PM 1 Syslog file size
7:58PM 4 Asterisk on AMD64
4:56PM 1 first character in line 11 missing
3:57PM 1 Presence Fully Supported?
3:04PM 0 Call Waiting Tracking?
1:40PM 0 H323 with asterisk-ooh323c
1:06PM 3 David Choo/eServices/eSpore is overseas
12:39PM 0 Australian Dial tone TDM400P
11:11AM 2 pb
11:08AM 1 ruby-agi 0.0.2 released
9:26AM 5 rotate * log file?
6:16AM 2 Using RedirectAction with queues
4:05AM 0 Ignore incomingcall?
4:02AM 2 Make asterisk call out
3:11AM 0 OpenH323-Channel Q.931-Problems with Gatekeeper
2:48AM 5 TE406p no interrupts
1:56AM 1 Integrating with existing analog PBX
12:46AM 6 SIP Connection Problems
Saturday September 10 2005
5:41PM 2 Echo Issue
3:13PM 1 TE110P reset
2:14PM 1 Configuring SIPURA 2002 to work wih Asterisk
11:43AM 0 Broadcasting via Asterisk
9:58AM 1 False Zap answer problem (Again)
9:56AM 1 AGI problem with library path
9:11AM 0 Distinctive Ring Problems
8:59AM 0 Problems with TE205P
7:05AM 0 call tests
5:33AM 2 AGI programming work required
5:32AM 1 Required hardware
4:57AM 2 VoipBuster again
4:18AM 4 Fritz, mISDN, Help
3:35AM 0 Need some HFC-S help
12:51AM 1 PRI echo
12:46AM 2 GotoIf Syntax to match first digits
Friday September 9 2005
8:08PM 2 call volume
6:27PM 0 Queue "abandon" count increments incorrectly?
5:19PM 0 Transferred calls dropping out of MeetMe
4:12PM 1 vm notif
3:33PM 0 Announcement: FOP 0.23 released
3:21PM 1 Wait for dialtone
3:02PM 0 did edmonton
2:51PM 0 Asterisk Extension Language
2:35PM 1 Special handling of IAX circuit-busy vs busy
2:09PM 1 ASTCC speaks and cut RTP channel
1:58PM 0 RTP ports in use grows and grows...
1:42PM 2 AMP 1.10.009 released!
1:39PM 1 Polycom 501 Multiple Line Instances
1:14PM 0 realtime and presence
11:51AM 1 Setting Account Code?
11:48AM 1 musiconhold errors in 1.2.0-beta1
11:26AM 9 adding DNIS digits
10:56AM 2 FW: Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist"
10:31AM 1 RE:NewCUT()
10:01AM 0 Asterisk connected to Concept XI520
9:53AM 0 woomera doesn't work (same OpenH323 problem as with chan_h323)
8:44AM 1 Changing User-Agent: Asterisk PBX
8:43AM 2 Storing extension prefs. in MySQL
8:21AM 1 OH323 for HEAD? 0.7.1 doesn't compile.
8:08AM 0 Detecting retries in call files
7:55AM 2 "Registered SIP '202' ... expires 1800". Why does it expire
7:34AM 1 Motherboard and processor recommendations
7:15AM 0 VIP-050
7:09AM 1 siemens pbx what i ask techinician?
6:16AM 1 New CUT()
5:08AM 0 Doesn't finishes callerid spill
5:07AM 1 spandsp txfax multi page problem
4:24AM 0 remote SIP phones
4:17AM 0 BRI debug, national ISDN speech call problem
4:14AM 4 Huge Echo
4:06AM 0 Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card
2:53AM 0 OT Humo[u]r IVR Menu sample
1:13AM 0 the number of incoming calls in queue
Thursday September 8 2005
11:10PM 0 Using E1 without power off simence pbx
10:29PM 1 can not make call with Unicall (MFC/R2)
9:46PM 0 T1 DSP Card to T1 - TXFAX RXFAX Posible Solved
9:30PM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday 9/10/2005
9:25PM 2 T400P vs TE405P
8:49PM 1 Montreal usergroup
8:27PM 0 Question about setup Grandstream HandyTone 488 SIP with Astersik to Travel throught NAT.
7:27PM 4 Solution for 12 to 16 FXO to asterisk connection
7:18PM 0 PRI and Caller ID when immediate=yes
7:11PM 2 TDM PCI Master abort
5:05PM 2 sip log messages every few seconds
4:29PM 1 IAX Trunking Weirdness
4:17PM 2 How do you change the festival voice
4:16PM 1 FW: Adtran TA 616
3:45PM 0 Announcement: ASTPP-1.2-Beta
2:47PM 2 TE411P zapata.conf, monitoring echo cancellation and echo tail size
2:40PM 1 MAX PRI for single server (was: Not enoughlinesavailable for Asterisk implemetation)
2:25PM 1 SIP/2.0 487 Request Terminated problem on Cisco 7960
2:18PM 1 Siupra-2002 with astersik
1:49PM 10 voice over atlantic
1:17PM 1 TDM400P not detecting hangup and not hanging up
12:11PM 1 Problem with IAXy
11:40AM 0 CVSHEAD callerid not working
11:08AM 2 play each person's voicemail
9:34AM 0 How to cascade dial status back through IAX
8:49AM 1 Multiple Line Appearances / Why use this?
8:46AM 2 Server Brand
8:37AM 1 Call goes through, but no audio
8:14AM 2 Pass through of T.38
8:00AM 0 Slight OT: Multi WAN Router and SIP Calls
7:56AM 0 Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist"
7:39AM 1 Multiple Instances of Asterisk (no contexts)
7:28AM 0 2 X100P and SIP inbound routing
7:07AM 2 All Circuits are busy
7:06AM 0 IVR Documentation and Samples.
6:56AM 0 Asterisk & Euro-ISDN
6:53AM 3 power over ethernet hub/switch
6:47AM 6 Not enough lines available for Asterisk implemetation
6:19AM 0 cvs head and seqno 102 (Critical Response) messages for Cisco 7960
5:56AM 1 Hangup problem
5:34AM 2 Distinctive ringing on Cisco 79xx
5:26AM 0 Yuxin hardphones feedback
5:22AM 0 Sip clients through proxy
5:02AM 1 Additional: Several SIP clients behind router registerwiththe same IP, messing up call routing, any ideas?
4:29AM 0 sending fax....i'm in trouble !
3:56AM 0 Extension a
2:42AM 1 pri gateway
2:31AM 0 Contexts are not being created - Asterisk BT100 Password Issue
2:27AM 0 who use astlinux with booting from DOM?
1:50AM 1 How to increase delay before incoming call answer with tdm400p
1:33AM 0 Setting up multiple trunk groups with different internal ring groups
1:26AM 2 Transfer calls from cellphone
1:19AM 1 (OT) Dialplan Standards for Business/Offices
12:31AM 1 (no subject)
Wednesday September 7 2005
11:15PM 2 410P upgrade to 411P?
10:25PM 0 I should never be called!
9:43PM 1 OT: Differences between test equipment
8:53PM 0 Sipura-2002 Can not make outgoing calls, incoming calls works OK
8:38PM 1 Not can call to PSTN
7:56PM 1 asterisk frequently dead
6:22PM 0 Hack for Canadian weather
5:53PM 0 Problem with PRI channels, restarted after every call.
5:46PM 0 Need Help - Losing first few seconds of call when using Broadvoice
4:18PM 2 Want to use a remotely location POTS phone
3:32PM 2 g729 test
3:30PM 0 IVR Documentation an Sample.
3:19PM 0 Remote Provisioning for the PA1688 phones.
3:11PM 1 IAXy - no dailtone
2:48PM 0 asterisk-statv2 showing blank screens
2:12PM 1 externpass in voicemail
1:46PM 1 Several SIP clients behind router register with the same IP, messing up call routing, any ideas?
1:30PM 0 sip - aastra 9133i
1:16PM 0 Asterisk with Vonage problems
12:50PM 1 TDM400P not detecting hangup and not hanging up.
12:40PM 1 blocked - rejecting connections
9:50AM 1 ztcfg Kills My Dial Tone
9:28AM 0 Second Line does not Connect - HELP - misdn,sip
9:08AM 0 ArtDio IPF-2000 unable to send audio to Cisco 7940 until placed on hold and resumed
8:45AM 1 Asterisk crashed?
6:56AM 3 Extensions - Realtime
6:34AM 1 Polycom 300 with latest 1.5.3 firmware not registering
6:30AM 1 Speex codec - Out of buffer space
6:29AM 0 IAX PBX responds to IAX registration with expires time=0
6:05AM 1 2 X100P and SIP outbound routing
5:53AM 1 Eeven Stranger - Asterisk BT100 Password Issue
5:21AM 2 Desincripcion de la lista de Asterisk
4:47AM 1 "-- PROGRESS with cause code 34 received"?
4:35AM 1 Packet Cable
4:10AM 3 channels VHF/ HF radio in asterisk
4:05AM 1 ISDN PBX integration
3:26AM 1 presence settings and Eyebeam
2:20AM 4 How to connect many analog lines to Asterisk?
1:27AM 0 Max concurrent faxes with txfax/spandsp?
12:55AM 3 Hosted PBX (vPBX) and Call/PickUP Groups
12:18AM 0 Some info about Cisco's 79xx, and Sipura's phones
12:04AM 2 asterisk, SIP, Re-INVITEs and different contexts
Tuesday September 6 2005
9:38PM 0 IAX2 Problems causing server to hang
9:36PM 4 Working example of ALERT_INFO with Cisco ATAs?
9:06PM 4 Which Linux distribution?
8:05PM 5 PRI in and out
7:08PM 1 Some problems (SendDTMF, Wait, Parked Calls)
6:13PM 1 CTI and Asterisk
4:55PM 0 /dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8
4:17PM 1 Occasional quiet voicemails
3:47PM 1 Asterisk as SIP/H.323 Signalling Gateway
3:41PM 2 Speaking of Polycom phones...updated ROM: ouch!
2:43PM 5 Good Polycom Dealer?
2:15PM 1 Routing depending on sip response code?
2:02PM 1 one extension goes straight to voicemail, others don't
1:48PM 1 Queue AgentCallBackLogin
1:44PM 1 Asterisk overheating on VIA Epia MSeriesmotherboard
1:42PM 0 AstriCon Update: Please Register ASAP - Free Phones
1:41PM 1 Utility to find length of wav49 file
1:40PM 0 asterisk handling of old voicemail messages
1:09PM 4 Sipura Devices and Asterisk?
1:03PM 3 Asterisk scenario
12:59PM 0 Wireless router with built-in VOIP(FXS) ports forAnsterisk
12:48PM 2 Polycom ip301 hangs at Running "sip.ld"
12:26PM 0 Transfering to voicemail problem with 1.2beta
12:16PM 2 Wireless router with built-in VOIP(FXS) ports for Ansterisk
12:04PM 0 Loging agents in
11:42AM 1 Threeway calling uses up two FXO lines
11:40AM 0 IP PBX Market Share and Growth
11:39AM 0 Weird SIP behaviour
10:08AM 1 Asterisk BT100 Password Issue
10:05AM 4 PHP and ASterisk Manager
9:33AM 1 /dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8)
9:11AM 2 Asterisk overheating on VIA Epia MSeriesmoth erboard
8:34AM 1 "all lines are busy"
8:32AM 1 Application rxfax missing ?
8:27AM 9 civil emergency comms: Asterisk + HAM
8:24AM 1 Can get IAX connection but no SIP connection?
8:17AM 3 TE406P audio drops
8:10AM 2 Business telephones
7:24AM 0 Help evacuees from LA, MS, AL locate lived ones
2:35AM 1 TDM 400p
2:04AM 2 Going crazy with FAX :-(
1:47AM 1 SIP Callgroups
Monday September 5 2005
11:38PM 0 atxfer featuremap
5:55PM 0 Heartbeat with Broadvoice
4:57PM 1 unicall and cvs head
2:13PM 3 TDM11B pinout
2:05PM 0 Agentlogin transfer calls
1:52PM 2 Asterisk overheating on VIA Epia M Series motherboard
12:46PM 0 Asterisk as a GSM-Gateway? Possible or not??
12:38PM 2 Zaptel issue
12:35PM 3 Assessing network quality
12:19PM 3 Cisco 7960 upgrades
12:14PM 3 Asterisk architecture
11:48AM 1 (Call Features Resource) not loading
11:13AM 1 Unexpected results with "While" and "EndWhile" applications
11:13AM 2 Asterisk won't listen on another port
9:39AM 0 putty and winscp
9:30AM 2 "Provisioned, Down, Active", but D-channel seems to be fine
9:20AM 1 BT100 and BETA
9:17AM 9 Asterisk Follow ME
8:42AM 1 User authentication and privileges
8:41AM 0 more accounts
8:27AM 0 ooh323c h323_convertAsteriskCapToH323Cap Don't know how to deal with mode 0x40 (slin)
7:45AM 0 Re: Asterisk-Users Digest, Vol 14, Issue 22
6:34AM 1 A good HW
6:16AM 1 TDM Card FXO Question
6:13AM 6 asterisk CAPI dial-in issues
5:40AM 0 Asterisk clustering with SIP proxy?
5:17AM 0 asterisk@home and zaphfc dial out not working
4:27AM 1 SV: sending fax
3:53AM 0 queue transfers always get EXITWITHKEY
3:19AM 3 GotoIf sample...
3:09AM 0 Tr: MWI - message waiting indication
2:35AM 2 No DID on ZAP
2:18AM 2 DTMF issue on IVR
2:08AM 0 ReInvite not working
1:40AM 0 WG: Timeout when Dialing - HELP
1:31AM 0 (no subject)
1:27AM 2 Billing - Disable accounts when balance gets 0 value
1:24AM 0 Asterisk and SCCP unofficial site
12:08AM 4 sending fax
Sunday September 4 2005
11:59PM 1 Problem with Asterisk app command Read...
11:59PM 1 hints and polycom IP 300 phones
11:23PM 0 help on 2 X-Lite: call failed: 404 not found
10:31PM 3 A few questions before final proposal...
6:31PM 1 Unable to hear.
6:10PM 1 kernel panic
4:51PM 3 Asterisk Real-Time Voicemail Configuration
4:50PM 0 Updated Chan Unistim?
3:51PM 0 FW: OH323 with Asterisk@home - seems incomplete
2:55PM 0
2:38PM 0 sipura spc.exe ?
2:09PM 1 Option 1 in IVR menu
1:37PM 0 SIP, NAT and MySQL support (sipfriends)
1:25PM 0 OT: Sipura SPA 200 Caller ID Problem
12:14PM 3 Nokia 32 Terminal
11:34AM 0 chan_sip.c:946 __sip_xmit
8:34AM 0 Asterisk SMS via IAX2?
6:37AM 0 donating VOIP gear to the relief efforts.
6:18AM 0 Any hardphones with SIP API?
5:39AM 2 HELP - How Do I Separate incoming channels from the others on a PRI
5:14AM 0 dial rule / prefix with #
3:35AM 0 Open G.729 / G.723.1 update, fixed memory leak
3:03AM 1 FW: Asterisk@home - requesting help on oh323, ISDN BRI and iConnectHere DID
2:38AM 0 IPSwichBoard designers wanted
Saturday September 3 2005
10:04PM 5 Asterisk Community Participant; Katrina Refugee
5:33PM 0 Sipura spa841 problems
3:42PM 0 MWI - message waiting indication
3:12PM 0 How To Separate incoming channels from the others on a PRI
3:06PM 0 How Separate a few channels from the others on a PRI
2:54PM 2 Argentina - zapata.conf switchtype for Argentina
2:20PM 1 I connected my quicknet phonejack to the wall phone outlet and .......
1:59PM 0 chan_iax2.c:7672 iax2_poke_noanswer
1:04PM 3 unicall deploy
12:24PM 1 *81, block CID, using ATA
8:58AM 0 DNS SRV and new Asterisk install
8:53AM 0 stale nonce?
8:35AM 1 equipment and network advice
8:28AM 1 newbie install problem. And I already searched everywhere!
8:05AM 1 Current status on _outgoing_ Swedish/Dutch DTMF CLIP for TDM400 FXS interfaces?
7:39AM 0 Debug info from txfax - howto?
7:10AM 0 How to tell reason for hangup or busy in SIP or IAX
3:31AM 1 Multiple ASTCC Cards Configuration
1:46AM 1 chan_capi [0.4.0|-cm-0.5.4] and Asterisk 1.2.0-beta1 - early B3 not early enough sometimes
Friday September 2 2005
10:31PM 2 IVR Prompts
8:51PM 2 Sipura 3000 setup
8:43PM 0 STUN on PAP2-NA 2.0.12(LS)
7:20PM 0 CVS-HEAD Inband Ringing?
6:31PM 0 X101P ringing too long !
6:21PM 0 Web-voicemail doesn't play files nor display default pictures
5:02PM 0 Need * Setup Help
2:20PM 1 Asterisk and Eyebeam
2:03PM 0 SER+ASTERISK voicemail
1:19PM 1 Dlink dph-140s/ACT P104SLD
11:53AM 0 Notification of new voicemail by various met hods
11:52AM 1 Call Return
11:47AM 2 Notification of new voicemail by various methods
11:35AM 0 chan_oh323.conf (inAccess version)
11:09AM 0 How to locate Toll Free Ownership
11:04AM 4 Receptionist
10:08AM 0 CallerID and CDR
9:46AM 1 Linux-HA Heartbeat2 and Asterisk
9:16AM 0 Recommendations for a low cost GSM phone
9:04AM 1 AG-468 4xFXS - my personal review
8:45AM 1 No application 'AgentsLogin'
7:56AM 0 TDM400 w/ FXS S110M pinout on RJ11 connector?
7:52AM 0 Unable to create RTP session
7:26AM 1 how to execute something after Dial() ?
7:13AM 1 G711u sound quality decrease with upgrade from 1.0.7 to CVS-HEAD?
7:03AM 1 Semi-OT: An idea for New Orleanstemporarycommunications infrastructure
7:01AM 0 Semi-OT: An idea for New Orleans temporary communications infrastructure
7:00AM 0 Zapata help needed howto configure nationalprefix for a single card
7:00AM 2 FW: defunct email kill list
6:52AM 0 Semi-OT: An idea for New Orleans temporarycommunications infrastructure
6:30AM 0 monitoring VM via speaker and grabbing connection
6:03AM 0 Why is that: Sep 2 08:25:03 NOTICE[1403]: -- Registration for '1096377@' timed out, trying again
5:59AM 3 DTMF and "breaking through" voice prompts
4:29AM 0 sip SUBSCRIPTION bug in 1.0.9
3:47AM 2 chan_capi hfcpci mISDN linux 2.6.12 not working
1:52AM 0 Call drops
1:44AM 6 Looking for better "Follow Me"
1:40AM 0 ASTCC-adding more than one trunk to one route
12:47AM 1 Italy FastWeb problem: ISDN line crashes every time cisco router turns off
12:46AM 1 Fax problem, missing/compressed lines
12:42AM 1 Setting wcte11xp card to use IRQ
12:11AM 1 Snom 360 problem
Thursday September 1 2005
8:28PM 0 Re: Asterisk-Users Digest, Vol 14, Issue 1
8:19PM 2 Any one in Toronto / Canada can help me!
8:07PM 1 TE406P seg fault on Stable
7:15PM 0 extra ring after answer on sip calls
6:15PM 1 RE: Hardware dimensioning issues To: <>
5:58PM 0 Help on second dial
4:40PM 0 How to set CLIR when using call files ?
4:02PM 1 OT: SCALE 4x -- Call For Papers
2:31PM 0 Question about Asterisk connections
1:33PM 3 Automon filenames
1:18PM 1 Skipping problems on outgoing calls (using uLaw with an internal * server through Voxee)
1:16PM 1 Best costs effective solution...
12:51PM 2 Contact Directory on Polycom IP-501 phones
12:34PM 0 IAX2 how to disable VAD ?
11:27AM 2 ipvolution t1 cards
11:13AM 0 Two devices behind nat
11:10AM 1 dialparties.agi is returning no extensions to dial
11:07AM 1 Speed Questiosn
10:56AM 0 Fax trouble with HP 3330mfp (again)
10:42AM 1 sip jitter buffer in 1.2?
10:37AM 1 TOS bit and DSCP
10:21AM 0 Buying DIDs
10:20AM 1 Problem with include
10:14AM 0 How to resolve SMS/WAP/MMS/VoIP gateways on a shoestring?
10:05AM 0 *66 with Sipura devices.
10:01AM 0 Overhead Paging Systems...[More Info]
9:58AM 0 dialing extension, which context is searched
9:47AM 2 ztcfg problem
9:18AM 0 Outbound Authentication
9:16AM 1 Loop error when compiling CVS version of 1.2-Beta
9:16AM 0 Astaro SIP Proxy
9:10AM 0 RE: Asterisk with Meridian1 option11 in the UK
8:22AM 0 zapata nationalprefix-problem [Virus checked]
7:53AM 0 HELP - Queue Transfer
7:45AM 1 Snom 360 hold problem
7:33AM 4 Overhead Paging Systems...
7:22AM 0 Re: Polycom 301 second line registration
6:36AM 1 oh323 or h323
6:00AM 6 Grandstream GXP-2000 Poor sound Quality
5:39AM 0 Mobilephone users get echo of them self when calling in to our asterisk server.
5:24AM 3 Snom 360 and hints
5:20AM 1 What this little red cross mean in AAH
5:16AM 1 Sipura 1001 Adapter with two lines using one RG11 jack
5:05AM 0 Help setting up trunk on AAH
4:41AM 1 How to execute StopPlayTones when a SIP phone is answered
3:26AM 0 Mulig_SPAM: More than one outgoing call
3:13AM 0 Micronet 5050s FXO gateway and hookflash transfers.
1:46AM 1 How to require a keypress on answer?
1:06AM 2 TE cards with ISDN BRI?
12:49AM 0 Asterisk@Home: How to changed AMP User Login andPassword
12:33AM 1 Asterisk run problem, was working, rebooted server, now nothing
12:31AM 2 Recommendation for 8 lines analog card in Australia