asterisk users - Jan 2005

Monday January 31 2005
11:58PM 0 Playing a file upon pickup (dial command?)
10:49PM 0 Timer for MeetMe on Mac OS X
10:42PM 14 TDM400 stopped working
10:09PM 2 H.323
9:07PM 0 Asterisk MTBF studies
6:52PM 0 Intel chip IA98
6:48PM 0 Telephone Line options in Asterisk
6:31PM 0 PRI got event: HDLC Bad FCS (8)
4:58PM 0 Budgetone ringing volume
4:44PM 6 RE: Answering Machine Function?
3:44PM 4 Cisco 7960 and AutoAnswer.
3:36PM 7 Developing an IP Phone
3:30PM 2 video conferencing bounty
3:28PM 0 Single or Dual Processor? High volume MeetM e
3:16PM 1 Asterisk at CeBit 2005
2:45PM 0 Multiple calls placed in outgoing spool interfer with each other
2:28PM 0 Callerid on blind transfer w/ Cisco 7960
2:19PM 0 Strange sip address?
2:12PM 1 A neat "hot seating" mplementation
2:12PM 0 Delayed echo
2:12PM 0 Caller ID Bug in v1.0.5
2:12PM 2 PRI Dropped Calls - Audit, Restore, Idle state
1:46PM 7 Cisco phones config over internet
1:40PM 0 SRTP support
1:21PM 2 VoIP with Asterix
1:17PM 1 Call recorder based on *
12:40PM 0 PRI not hanging up the channel after Executing Hangup when dialing busy number.
12:32PM 3 Where does a newbie get started?
12:24PM 0 re: cdr_mysql and system time
12:13PM 1 Grandstream stops working after "Register Expiration" period has passed (dynamic registration)
12:09PM 4 line assignment on TDM400P
12:07PM 4 Multiport Fax over softphone
12:04PM 6 Q: PRI leading 0 (area access code) or 00 (country access code) missing on incoming callerid
11:52AM 1 chan_sccp bug / problem
11:24AM 0 Fast busy signal
11:02AM 0 Return call after transfer with no answer
10:47AM 0 Eyebeam Vs. Windows Messanger,
10:37AM 6 cisco 7960 image
10:07AM 2 Dialing out on TDM400p 4 port FXO
9:58AM 0 Sending forwarded calls out to a different provider
9:37AM 0 Linksys RT31P2-NA
9:19AM 0 AGI Processing Order
8:52AM 3 SPA-841 Call Waiting
8:27AM 0 music on hold that starts at beginning of file
8:27AM 0 Tuning MoH Volume
8:01AM 3 Audio Quality over LAN very bad
7:37AM 5 Announcement to caller when called party haspicked up - without initial Answer()?
7:34AM 6 Announcement to caller when called party has picked up - without initial Answer()?
7:24AM 0 Error while trying to execute asterisk
5:28AM 0 ISDN supplemetary services (Hold, Retrieve, 3PTY) on HFC-8S
4:37AM 9 SIP x NAT
4:04AM 0 AW: HDLC for Dummies?
3:40AM 0 Eicon Diva audio problem [Newbie]
3:39AM 1 HDLC for Dummies?
3:30AM 6 TDM400P specs clarification
3:07AM 0 Indication of transfer on display
3:01AM 3 congestion problem with only one number
2:41AM 3 Group Extension
2:29AM 8 NAT and SIP
2:18AM 1 Instant Messaging
12:24AM 4 Trunked IAX or not
Sunday January 30 2005
10:55PM 3 how to stop ringing after congestion.
10:34PM 10 Zap channels in AU hanging up on STD pips
9:46PM 1 DIAX softphone - Asterisk server rejecting
9:16PM 1 Slackware + Asterisk + asterisk-addons
8:41PM 4 detailed asterisk howto
8:33PM 1 x100P wildcard discontinued ?
8:29PM 2 x100p issues + TDM400P
8:27PM 0 Hitting IOCTL??
8:16PM 9 Japan
7:00PM 0 Meetme2 web - nothing happens on click ?
6:59PM 4 Processing incoming calls with multiple contextst over PRI
6:05PM 3 Asterisk friendly VoIP providers
4:35PM 0 OH323 compile error : CVS-HEAD
4:30PM 4 Monitor calls timeout
4:29PM 2 Trying to make but it fails
3:56PM 0 conference room capacity question
3:47PM 0 302 Moved temporarily problem / Sipura 3000
2:32PM 0 Caller ID on H323
2:21PM 0 One way call when the * server and phone in a local network
1:44PM 3 Callgroup with bristuff ISDN?
1:40PM 1 IAX2 firmware for PA168x (Giptel G100, Siptronic ST-100 etc)
12:20PM 4 Single or Dual Processor? High volume MeetMe
12:10PM 5 Caller ID spoofing
10:22AM 0 D/41D
9:44AM 0 Setting call forward for Agent's in a Queue
9:37AM 0 Vservices.inv of Julian Pawlowski anoyne has the macro-dailer for this?
9:24AM 8 agent logoff
8:53AM 0 newlines in application data strings (e.g. userevent)
8:13AM 0 Can I start recording during call - is priority "a" active only in voicemail ?
7:02AM 14 Asterisk on MS Virtual Server
4:17AM 11 where to buy x100p
4:15AM 2 widcard x100P doubt
3:40AM 8 Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe
3:15AM 1 Vocera Badges
2:14AM 0 xten x-lite eyebeam
2:01AM 7 Strange Crash
Saturday January 29 2005
8:13PM 0 Cisco BRI & SIP
7:17PM 2 Silly question: Why multiple lines on SIP phones?
7:01PM 0 RE: Asterisk-Users Digest, Vol 6, Issue 463
6:43PM 4 SIP native bridge problem
6:01PM 7 Asterisk@home and Zap Channels
4:51PM 1 Please help, Zap channel hangup TE405P
3:48PM 0 Unable to remove Monitor IN / OUT wav files - Timing error
2:50PM 2 Subject: RE: Q: Can I over-ride the value of caller ID
2:49PM 1 Asterisk@home problem installing CentOS ..
2:19PM 1 Integration PBX
1:20PM 0 Support for Dialogic 4 or Dialogic Proline2V
1:13PM 2 Call rejected by FWD: Unable to negotiate codec
12:33PM 0 What was the conclussion of the R2 test in Mexico??
12:20PM 0 Cisco/Lucent/Asterisk Guru needed
12:12PM 7 Sipura SPA-841 auto-answer support [patch]
11:17AM 0 Adding more links to the Navigation box in
10:31AM 5 Channel Bank Echo
10:24AM 0 Adding digits to incoming callids depending on context?
10:18AM 3 TE405P w/ Intel SE7210TP1_E Motherboard
9:56AM 1 FS- ideal starter pack. 1 X100P and 1 Grandstream Budgetone-101
9:41AM 1 PyAsterisk Download?
9:38AM 7 ISDN in US?
9:36AM 10 asterisk tries to dial out on lines already in use.
9:13AM 0 SIP Caller ID Number vs. Caller ID Name
9:11AM 2 Server auto Fallback
8:15AM 4 How to use ASTCC with SIP ??
7:48AM 3 IAX2 Asymmetric Latency
6:11AM 8 PRI for Data and Voice
4:51AM 0 MyPBX model-1
4:00AM 2 asterisk+h323+rh9
1:11AM 3 problem in compiling asterisk addon
12:55AM 1 Asterisk @ Home 0.4 w/ Broadvoice + 5 SIP Phones How To
12:38AM 1 *1.0.5 CAN NOT find my sip.conf
12:28AM 1 Disable Reinvite on a per call basis.
12:19AM 0 upgrading to *-1.0.5 on Gentoo; error cdr_mysql.conf': Not found
Friday January 28 2005
11:28PM 3 extensions.conf - redundancy removal
11:06PM 2 Direct MP3 channel Black Hole?
10:11PM 1 incoming calls produce multiple quarter rings andasterisk never answers.
10:09PM 0 incoming calls produce multiple quarter rings and asterisk never answers.
10:01PM 1 Meetme2?
9:23PM 1 Asterisk Prepaid Application Help
9:01PM 6 IP Phone for IP PBX
8:11PM 9 iaxComm version 1.0 released
7:38PM 1 FC3 + udev + Asterisk v1.0.3 - Temporary Fix
7:38PM 31 Speech Recognition
7:16PM 4 Nortel --> Asterisk-------->Asterisk
6:45PM 1 Problmes compilling *
6:24PM 0 Outgoing Call Block
5:44PM 1 Putting IP behind firewall
5:36PM 0 No Video With Eyebeam
5:00PM 5 Call Waiting Audio Prompt
3:52PM 1 Fedora Core 3 / Asterisk / TP100 Wildcard
3:36PM 0 PPP over T100P: Using a subset of channels does not always work correctly.
3:09PM 0 New Polycom SIP offerings
2:44PM 2 Who is in control Voicetronix OR Asterisk
2:31PM 0 fax/data/phone switch interfering with voip
2:30PM 10 Record inbound and outbound calls to and from one number.
2:25PM 4 ISDN Hardware
2:22PM 0 ANNOUNCEMENT : NEW CallingCard ApplicationforAst erisk
2:22PM 1 Integrating with existing 1BRI, 6 POTS Panasonic PBX ?
1:51PM 0 asterisk@home voicemail issue
1:50PM 0 re: Polycom
1:44PM 2 Polycom changing policy - allowing firmware downloads?
1:35PM 0 Asterisk Prepaid Applications Comparison
1:03PM 4 MusicOnHold with no sound card?
12:18PM 4 FWD and IAX2
11:53AM 2 MoH does not de-attach
11:50AM 1 Festival Jittery (bad udp checksum)
11:47AM 12 FW: FAQ missing info? Asterisk@home V 0.4
11:43AM 3 1.0.3-BRIstuffed
11:42AM 0 two OpenH323 vulnerabilities
11:42AM 0 asterisk call flow diagrams for ser voicemail combo
11:30AM 3 reason 24 (Call ended with Q.931 cause)
11:25AM 3 chan_iax2.c problem?
11:23AM 8 Eyebeam - asterisk - Messenger
11:14AM 0 Problems with H323/G729--No NATting and no Dynamic IP involved...
10:46AM 1 * acting as IP-Phone?
10:35AM 3 adit 600 fxo ports immediately "answers" outgoing calls (even if not connected to line)
10:22AM 1 Minimum Setup
9:25AM 1 error while trying to install astcc
8:48AM 2 Fwd and Tollfree
8:20AM 0 Trying to use Dial with D option..
8:16AM 1 Authentication against voicemail password database
8:08AM 2 zap FXO channel - wait for N seconds before answer
7:48AM 2 Sipura SPA-841 with Asterisk
7:31AM 2 Problem with chan_sccp and cisco 7960
7:28AM 3 Where can I find good doc on AGI?
7:22AM 0 [Asterisk-biz] update
7:19AM 0 STUN
7:11AM 3 redirect different phone number to different IP phone
7:01AM 0 Sipua SPA-2000 and liong delay afterdialling number
6:23AM 1 Bristuff and Realtime
5:35AM 2 1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping
4:14AM 7 Sipua SPA-2000 and liong delay after dialling number
3:41AM 4 Command to light MWI on 7940 /7960
3:09AM 0 Continuously ringing Zap/4-1 TDM11B All of a sudden ?[Urgent Pls]
2:15AM 1 does asterisk support instant messaging?
1:40AM 8 Ouch ... error while writing audio data: : Broken pipe
1:18AM 0 asterisk CVS rpms for FC1 updated
12:48AM 0 Register replicaton and HA *
12:15AM 4 I want to display my numbers for incoming calls when some one dials my number from any where
Thursday January 27 2005
11:30PM 10 Caller ID in AU
11:12PM 4 Q: Can I over-ride the value of ${CALLERIDNAME} ?
10:02PM 1 Dial and Macro Do not seem to be working
9:02PM 0 SIP CANCEL problem
7:54PM 0 LiveVoip Expanded Codec Support & Feb Sale 1.2 Cents a Min USA & Canada
7:02PM 1 ChanIsAvail not working
5:35PM 0 Need some advises configuring asterisk to callover INTERNET
4:56PM 4 OT: local DID question
4:42PM 3 Voicemail attachment not being emailed out
4:37PM 0 Problems making SIP URL outgoing dial
4:13PM 2 Trouble with Quicknet Linejack
3:44PM 0 Asterisk CVS on FreeBSD-stable gmaking result
3:02PM 0 Channel Groups?
3:01PM 7 Stumped by BroadVoice SIP
2:46PM 1 Asterisk auto-dial out deliver message
2:21PM 2 Making digital/data calls through asterisk
1:47PM 1 Outgoing call on Zap channel always gives DIALSTATUS ANSWER even when BUSY?
1:43PM 4 Tortoise CVS download for Asterisk Docs
1:23PM 0 Re: Asterisk-Users Digest, Vol 6, Issue 432
1:19PM 3 Digium and Intel Chipset compatability
1:18PM 14 SIP + NAT = horrible mess
1:18PM 1 Bad ECHO problem after upgrade to HEAD version
12:43PM 0 X100P/Zaptel on Gentoo Sparc64
12:35PM 10 Linux Bridge + QoS Shaper HOWTO available
12:35PM 5 Avoiding queue retries without hangs?
12:29PM 0 Re: Asterisk-Users Digest, Vol 6, Issue 431
12:20PM 0 RE: 2 questions regarding call ques.
12:16PM 0 Asterisk @ Home & BroadVoice (Outbound) help
12:11PM 1 Hold music while ControlPlayback is paused?
11:49AM 7 CallerID for incoming SIP calls to Asterisk connected phone
11:46AM 0 Re: Asterisk-Users Digest, Vol 6, Issue 430
11:14AM 1 Random hang ups during long calls
11:03AM 0 Re: Asterisk-Users Digest, Vol 6, Issue 429
10:58AM 0 AW: HEELP!! with Eyebeam
10:28AM 5 CISCO 7905 Phone Weirdness
10:14AM 4 Changing mailbox greeting
10:11AM 0 differentiate the incoming from the outgoing calls in PSTN line
9:53AM 2 Soft phone sound quality help
9:50AM 0 How can I check the selected codec for a call?
9:17AM 0 Need some advises configuring asterisk to call over INTERNET
9:07AM 2 TDM-400P + CallerID
8:25AM 2 Adit 600
7:54AM 4 Am I missing something really basic here?????helpwith Asterisk@home {Scanned}
7:34AM 1 Directory service of voicemail extensions
7:21AM 5 / sixtel are they legitimate?
7:12AM 6 /usr/bin/ld: cannot find -lidn
6:58AM 0 Re: Polycom and call waiting again...
6:21AM 0 HEELP!! with Eyebeam
5:37AM 1 analog lines via channel bank --
5:28AM 3 SoftClient for Pocket PC
5:10AM 1 analog fax on ericsson BP250 - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
5:10AM 2 Channel Restart - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
5:09AM 2 Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
4:52AM 0 Grandstream setup woe and solution
4:51AM 0 Asterisk@home and TDM400P cards...
4:38AM 0 Com-on-Air - DECT card
4:23AM 0 ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk
4:21AM 0 res_python
4:00AM 0 enter/leave sound with meetme adminmenu
3:18AM 0 How to check sip channel with smoething similar to ping ?
3:17AM 2 Moh in meetme doesn't work if I transfer to meetme
2:57AM 3 Asterisk chooses invalid outgoing interface (IAX2, virtual interfaces)
2:35AM 0 DUNDi on Asterisk
1:35AM 0 Re: Howto Setup TFTP server on Linux for Cisco
1:00AM 0 Problem with OpenPhone->Asterisk
12:20AM 3 Festival as background
Wednesday January 26 2005
10:07PM 2 Call Announce, Dial 1 to Accept, Dial 2 to send to VoiceMail
9:36PM 0 dialplan logic for conditional DISA on incom ming 800 number
8:31PM 0 dialplan logic for conditional DISA on incomming 800 number
8:12PM 14 phone rings when I'm using it over VOIP - WHY?
8:10PM 0 Firefly reject problem - it just keeps ringing
8:06PM 0 Void callerid info on iax clients, but OK from local extensions or on SIP clients
7:41PM 0 New version of AMP - 1.10.006
7:15PM 0 IAXy problems -- and no documentation
7:07PM 2 I need Help everyone I just bough my Xten Eyebeam
6:55PM 0 Is it possible to use native transfer in a call file?
5:44PM 0 resolved Asterisk + Broadvoice error
5:37PM 0 7900 Problem with Asterisk 1.0.1 and OH323
5:27PM 0 Asterisk @ Broadvoice (I know it's been covered, but odd error)
5:24PM 1 Cmd READ and #
4:50PM 1 Cisco 7905/7912, SIP, g729 and DTMF setup
4:33PM 0 Cannot get * to work on VIA 800 MOBO
4:30PM 1 Re: - GREATadvance
4:18PM 0 [Fwd: Re: [Asterisk-biz] GREATadvance]
4:09PM 1 How to make channel busy signal?
4:00PM 3 ANNOUNCEMENT : NEW CallingCard Application forAsterisk
3:13PM 1 Inbound analog Telco line not answered
2:59PM 1 mySQL-sipfriend dials to another SIP-endpoint - How to set the from-user
2:28PM 0 priority -1
2:26PM 2 ANNOUNCEMENT:NEWCallingCardApplicationforAsterisk
2:06PM 0 supported ip phones (3com)
1:59PM 5 IAXy Hung, Power-cycle Required
1:59PM 0 VICI dialer help...
1:55PM 0 ANNOUNCEMENT : NEW CallingCardApplication fo rAsterisk
1:48PM 3 Asterisk as root in realtime vs. non-root asterisk ?
1:46PM 1 Firefly as Asterisk SIP client - qualify works ?
1:02PM 0 AMP-IAX2 trunk issue
12:47PM 1 channel numbering
12:42PM 2 BroadVoice Outgoing CallerID
12:30PM 1 IAX/SIP Softphone with G729
12:27PM 0 Simple problem - call another phone on Busy
12:09PM 6 A working BroadVoice config example
12:00PM 2 native MOH with Asterisk 1.0.5 - any news?
12:00PM 2 Asterisk drops calls - why ??
11:48AM 1 SIP called number on incoming call
11:13AM 0 Re: Asterisk-Users Digest, Vol 6, Issue 404
11:11AM 4 Dialogic Boards
10:53AM 0 E100P echo on UK PRI
10:05AM 34 ANNOUNCEMENT : NEW CallingCard Application for Asterisk
9:52AM 1 TDM400P/TDM22B dialing issue
9:48AM 0 ulaw blank spots but gsm fine
9:43AM 33 No ringback on IAX channel after selecting menu option
9:34AM 7 TFTP Server Facing the Internet
9:33AM 0 Need help for a quick fun t-shirt/polo project - graphics artist wanted
9:16AM 0 Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from
8:33AM 0 Restart in the DISA to the beginning
8:18AM 0 ZT_CHANCONFIG failed on channel 11: Function not implemented (38)
7:57AM 1 Rining Issues
7:54AM 0 RE: Howto Setup TFTP server on Linux for Cis co 7960
7:52AM 9 Polycom IP 600 - 1.3.1
7:42AM 0 ParkAndAnnounce +${ALERT_INFO}
7:19AM 0 HFC-S card problems
7:00AM 13 Howto Setup TFTP server on Linux for Cisco 7 960
6:56AM 0 chan_capi audio issue
6:50AM 6 Howto Setup TFTP server on Linux for Cisco 7960
6:36AM 1 Am I missing something really basic here????? help with Asterisk@home
6:23AM 2 Telrad + E&M T1 Trunk
6:16AM 28 Cisco 7960 Message Light on multiple phones
6:11AM 2 off topic - DECT phones with FSK VMWI in the UK
5:59AM 0 Polycom IP600 stuck at "Running App = sip.ld"(was: Re: Polycom 1.4.1 firmware for IP500/IP600)
5:31AM 1 Asterisk with PSTN Help........needed!!!!!!!
5:30AM 0 Cannot get call transfers working
5:11AM 1 VoIP QoS with PIX
5:07AM 2 Issue with in latest CVS
4:52AM 8 [Fwd: Re: [Asterisk-biz] - GREATadvance]
4:24AM 3 setup questions- many users, little use
4:03AM 2 optimumvoice
3:58AM 1 interested in your opinion about FWD and iaxtel
3:19AM 0 Polycom boot server problem
2:39AM 2 ASTCC Trunks
2:06AM 1 Callmanager and Asterisk problem
1:42AM 0 Getting a Wildcard TE110P working on E1's in Australia
12:31AM 3 cant do it in CLI anymore?
Tuesday January 25 2005
10:26PM 0 New RPMS for FC1
9:47PM 0 Caller ID w/Name Providers???
9:28PM 1 Asterisk@Home initial setup
9:23PM 1 softphone headsets
9:11PM 2 Another BroadVoice Problem
8:53PM 8 TDM400 - channel out to lunch?
8:36PM 7 Tall free number via FWD over IXA2
8:10PM 0 Perfect billing solution for *?
7:39PM 1 Asterisk@home with Wildcard TDM400P card.
7:18PM 0 Dial command announcement
6:55PM 5 DTMF digit dropping
6:27PM 3 fwd IAX2 error
5:51PM 0 calleridname from chan_sip (mysql_sipfriends)
5:09PM 8 Interesting bellster issue
5:06PM 12 BroadVoice Help
4:55PM 2 Re: [Asterisk-biz] - GREAT advance
4:49PM 0 E100P vs TE110P & Echo
4:33PM 22 Anyone having problems with LiveVoIP?
4:28PM 14 grandstream budgetone-100 updates
4:18PM 15 Polycom and call waiting again..
3:36PM 1 Server side three-way calling with SIP channel
3:30PM 4 Polycom 1.4.1 firmware for IP500/IP600
2:39PM 4 Problems splicing Asterisk with a TE405P between Arcor E1 PRI and Ericsson Business Phone 250
2:25PM 1 SIP clients and double NAT
1:29PM 0 dial-back, call-back, what, is it called?
1:16PM 2 New ip billing solution?? any updates?
12:57PM 3 R2 in Bolivia
12:54PM 0 RE: Question regarding phones with multiple line appearances.
12:37PM 4 New native assisted transfer (atxfer) usage info required
11:57AM 4 Unable to Specify Channel 1 - no such device or address
11:22AM 1 Bellster and DTMF
11:09AM 4 BroadVoice Or VoicePulse ?
11:05AM 1 Re: I think your problem has to do with how you set the variable.
11:00AM 11 TDM400P Dell 1850 Server
10:56AM 0 probably error in chan_capi
10:53AM 1 HEAD vs STABLE
10:10AM 8 One Ring Mystery
9:54AM 0 Marked users with meetme2 ....
9:53AM 3 Configuring VLAN takes ages
9:48AM 2 SIP UDP ports on firewal to open
9:44AM 3 x-lite with wireless connection
9:34AM 0 coredumping on MusicOnHold
9:23AM 4 Am i in control after i dial?
8:52AM 1 iax java client
8:19AM 0 BackupPc_nightly crashing with Perl chdir errors
8:19AM 7 AMP with SUSE 9.2
8:16AM 0 Goto invalid extension doesn't go to 'I' when in a macro.
8:02AM 0 Re: [Fwd: Re: [Asterisk-biz]
7:42AM 0 Directory() ringing problem
7:19AM 0 Asterisk auto-dial out with .call files: Can I provide caller ID to second extension ?
7:14AM 10 Asterisk HEAD ->> Stable schedule?
7:09AM 1 TE110P yellow errors
7:08AM 10 Cisco 7940/7960
7:06AM 5 OT: pinout for"standard"telephoneheadsetrequired.?
6:22AM 6 Codec mismatch between SIP (BT) and IAX Phone
6:09AM 1 Terminiation in the UK.
5:47AM 0 OH323 Cisco Transfer Key
4:21AM 0 Mediatrix voip gateway 1124 and 1204 in UKsetting
4:05AM 2 Problems with H323 channels
3:07AM 8 BUSY-tone on incoming calls?
2:42AM 1 SER Prob
2:40AM 7 Bristuff ZapHFC and Loosing D-Channel
1:08AM 1 Turn off DTMF recognition pending on CallerID
12:52AM 0 FXO and groups
12:46AM 1 Dialplane slip
Monday January 24 2005
11:54PM 1 who used ser and asterisk?
9:37PM 1 How to reset IP600 with no password?
9:31PM 2 Correct way to update Asterisk
9:08PM 1 (no subject)
8:55PM 2 IP FXS channel bank
8:37PM 5 Nufone and Dialing Out
8:12PM 0 Asterisk@Home 0.3 and the Wildcard TDM400P
7:37PM 1 Asterisk Dial Out Issues - POTS Line
7:29PM 4 PrivacyManager not Working
7:29PM 0 chan_iax2.c:5441 socket_read: Rejected connect attempt from
7:28PM 1 What softphones for commercial use ?
7:22PM 20 [Fwd: Re: [Asterisk-biz] - GREAT advance]
7:19PM 1 Realtime voicemail question
7:17PM 1 (no subject)
6:43PM 0 HFC-S cards in UK
6:41PM 0 Cisco Maintence Contract for my 7960
5:51PM 0 size and quality of audio clips effect the playback??
5:13PM 3 TDM400 in aging Dell Optiplex
4:54PM 2 SetGroup and CheckGroup problems
4:40PM 2 .call file creation
4:04PM 3 asterisk@home and capi
3:38PM 0 Volume on Zap channels (T1)
3:29PM 0 G.729 and mutualphone service
3:09PM 0 Voicetronix OpenSwitch6 with 10-digit Dialing
2:58PM 8 SIP-T Support (I got my head in an SS7 cloud)
2:47PM 4 T1 E&M vs PRI question
2:45PM 0 Asterisk v1.0.1 Cisco 7960 Sip v7.3
2:15PM 2 FX CallerID
2:08PM 2 "Inband DTMF is not supported on codec G.711 u-law. Use RFC2833"
1:57PM 1 anyone got a 405 to work on a DL380?
1:29PM 3 cepstral integration with * using AGI?
12:38PM 10 Damn DTMF Beeps on my calls
12:33PM 4 Network Test Tool?
12:23PM 0 auto-dial out (.call file) failure detection
11:54AM 7 Asterisk -> static nat -> laptop w/siproxd -> cisco 7960
11:16AM 11 Athlon 64 for Asterisk?
11:11AM 0 Missing Variable in Local Channels
11:07AM 2 Wildcard TE405P and TDM400 - TDM not working
10:18AM 6 Is Voice Pulse Connect good ?
10:15AM 2 Inbound Errors
10:03AM 0 Need some help with G729 passthru
9:56AM 2 Multiple X100P
9:47AM 13 OT: pinout for "standard" telephone headset required.?
9:36AM 12 LiveVoip DTMF Issues
9:31AM 7 AVM Fritz crash
9:03AM 0 DTMF tones during a call to OSS/dsp
8:55AM 2 Menu tree for voicemailmain application
8:54AM 1 PRI dchannel in use? (take 2)
8:53AM 1 Hitachi Cable WIP-5000 Wifi phone?
8:45AM 0 Best VPN server for * and woad warriors usin g windows?
8:42AM 10 Dialing Delay
8:42AM 0 budgetone - pattern matching for ringtones - firmware
8:41AM 2 zaptel vanilla kernel
8:19AM 3 XEON or not
8:14AM 5 ISP connection to the PSTN using Asterisk
8:13AM 2 IVR Timing out
7:59AM 1 how to use mysql with asterisk
7:57AM 5 Cisco7905 keeps forwarding to voicemail
7:47AM 0 TDM400P Sync source
7:46AM 13 Asterisk with Grandstream ringback
7:39AM 0 forwarding sip
7:26AM 0 How to display number being dialed
7:16AM 3 Sipura Behind NAT howto
7:13AM 1 Voicemail folders
7:05AM 1 DTMF issues (handytone)
6:27AM 1 Threeway callin
5:40AM 5 Not answering PSTN until SIP answers
5:14AM 1 zaphfc no callerid incoming to SIP phone butvisible in logfile
4:31AM 2 asterisk starting problem
4:20AM 1 Mediatrix voip gateway 1124 and 1204 in UK setting
4:04AM 3 OT: Libnewt sourcecode?
3:37AM 12 Zapata in Australia
3:31AM 2 Short DTMF Tones and Asterisk
3:18AM 42 UPS for Asterisk
3:12AM 2 PSTN and Asterisk
2:33AM 4 Asterisk on sattelite link
2:18AM 0 how to display queue status and/or line status in asterisk
1:57AM 16 Auto callout - reminder - is it possible?
1:49AM 0 about call out : a strange question.
Sunday January 23 2005
11:58PM 1 Looking for a prepaid calling card platform
9:39PM 0 Peculiar one way convesation fault with Asterisk.
9:31PM 2 sip - h323 translation stability & capacity limit
9:28PM 0 zaprtc from bristuff? not there?
8:34PM 4 Asterisk 1.0.5
7:16PM 3 SIP USB Phone?
6:46PM 0 - New Security List -
6:27PM 0 Upgrade to the newest cvs now asterisk will notstart
6:21PM 5 VoIP software for MAC OS older than "X"?
6:02PM 0 Upgrade to the newest cvs now asterisk will not start
5:57PM 0 No music with "Blind" transfer on GS ATA + Sipura-841
5:50PM 1 Data calls with Asterisk
4:34PM 2 Sip Notify and PHP AGI
4:10PM 5 Music On-Hold problem
1:58PM 4 VoIP Providers and Backbone Servers
1:42PM 8 Autio cut off at beginning of call
1:19PM 0 Anybody a patch for oss/alsa to not constantly hog the sound card?
12:17PM 13 Any experience with Sangoma cards?
12:02PM 12 Florz patch for zaphfc
10:49AM 0 grandstream sip phone calling Zap/1 on TDM20Brings and answers but not hear voice
9:06AM 0 simulating multiple lines using ADSI
8:46AM 0 How to debug core-file
8:05AM 0 Delay before dialing extension on Zap channel
2:33AM 6 Best VPN server for * and woad warriors using windows?
1:19AM 1 gsm/wav format not recognized in Background() application
12:56AM 7 can iaxcomm run on the same server as Asterisk?
Saturday January 22 2005
11:37PM 1 zaprtc load issue (different that other postings)
11:32PM 6 chan_skinny and firmware upgrade
10:31PM 2 Some issues with X-Lite and codecs.
8:43PM 0 chan_capi patch: app_capiFax modifications
6:53PM 5 flashing zap using macro
5:41PM 1 grandstream sip phone calling Zap/1 on TDM20B rings and answers but not hear voice
4:22PM 6 Bellster - cool :-)
4:07PM 0 PortaOne's RADIUS client and Appradius
3:00PM 1 ASTCC: potential billing issue and "fix"
2:49PM 0 Anyone know where a good source of mailing l ist stats might be found?
2:29PM 0 how to configure Asterisk is outside and the SIP phone (Xlite) is inside behind NAT/PAT
12:38PM 0 asterisk not starting--sound module
11:42AM 1 te405P and german PMX
10:48AM 10 Asterisk Install Method
10:31AM 0 VoIP service setup help
10:19AM 3 Anyone know where a good source of mailing list stats might be found?
10:14AM 6 Dialogic D/4PCI
8:31AM 0 Asterisk/Sip crash "Failed to grab lock"
8:27AM 1 Re: Bellster - IAX-based interchange -- lets youcallanywhere for free
6:09AM 1 Need help configuring TDM10B / X100P Cards
5:43AM 0 Fwd: Re: chan_misdn 0.0.3-rc5 - new release ! Please test it.
3:42AM 0 Asterisk + TDM04b trouble
2:46AM 4 Cisco ATA186 and Asterisk dialplan
Friday January 21 2005
10:57PM 0 Caller ID Problems after upgrading from 1.0.1 to 1.0.4
7:34PM 4 7960 SIP image
5:55PM 0 incoming calls timing out.
5:08PM 10 IAX Inbound Sound Quality
4:38PM 11 SPA-2000
4:28PM 0 Problem compiling zaptel-1.0.3
3:49PM 30 IAXy's apparantly failing in the field
2:38PM 0 Rotate Logs
2:31PM 0 Incoming zap channels busy
2:26PM 1 Where is the * servers IP defined for sip phones?
2:01PM 0 WellTech 3804 Config anyone??
1:46PM 1 Iaxphone - unreachable if qualify yes ?
1:28PM 3 Powell resigns
1:04PM 0 Rate Engine Examples
12:53PM 0 three way call using sip - SOLVED -
12:31PM 0 Multiple Host IP connections per peer
12:24PM 5 Outbound analog dialing with Internet Line Jack (fwd)
11:51AM 0 Manager API on gives the DIALSTATUS of the first picked up channel?
11:36AM 0 AstTapi - Crashes w/ Windows 2000 - UrgentHelpneeded - May need to hire a developer
11:26AM 17 IAXTEL is dead/dying?
11:20AM 6 three way call using sip
10:54AM 9 zaphfc no callerid incoming to SIP phone but visible in logfile
10:42AM 1 Asterisk+Oracle
10:41AM 0 Codec conversion sip peer <> Asterisk
10:39AM 1 chan_misdn 0.0.3-rc5 - new release ! Please testit.
10:37AM 0 About DeStar, a web frontend for Asterisk
10:26AM 2 Bandwidth, again, can someone check my math?
9:48AM 0 chan_misdn 0.0.3-rc5 - new release ! Please test it.
9:43AM 2 Webmin Module for Asterisk (and thirdlane)
9:39AM 0 IAX2 trunking, Voicepulse Connect, and Outbound Faxing
9:35AM 1 Ignoring callwaiting?
9:25AM 0 Voicemail.conf pin protection
9:23AM 2 SpanDSPpre10 and AsterisK1.0.4 issues
9:22AM 0 Cisco 7960 can't make/receive calls
9:03AM 0 Help DIALSTATUS gives ANSWER when line is BUSY?
8:38AM 4 Can anyone recoment T1/PRI provider in SouthOntario?
8:13AM 2 problem with TE-405P
7:54AM 6 Snom hint for ZAP channels?
6:49AM 1 Asterisk 1.0.4 and broadvoice patch
6:10AM 0 german dialtones for IAXy?
4:29AM 0 Mediatrix III FXO 4 Port
4:16AM 0 Grandstreams+Nat
3:53AM 1 Recording a meetme conference
3:31AM 1 Voicemail Synchronization
3:30AM 1 sip.conf configuration for internal calls
3:21AM 0 3Com SIP Phone - Forbidden
3:19AM 0 Dropping duplicate answer
3:16AM 1 Intermittent breakage with the ISDN4Linux modem driver
2:45AM 0 Caller id with isdn4linux
1:54AM 0 h323 client
1:06AM 9 Adit 600 as VoIP router (MGCP) and Asterisk
12:51AM 0 Stanaphone incoming calls problem.
Thursday January 20 2005
11:46PM 1 Polycom IP 300/500 Conferencing Behavior
11:02PM 3 Zap randomly hanging up
8:57PM 9 Segmentation Fault after Digitnetwork X100P install
8:40PM 7 softswitch dilemma
8:32PM 5 Ring an incoming call in multiple extensions
8:19PM 10 OT: Headset for the Cisco 7960
7:58PM 1 Hopping through iax servers
6:35PM 1 SNOM 190 and dtmf
6:25PM 2 controlling recording
5:36PM 1 Re: zaptel on 2.6.10 kernel - debian.
4:42PM 12 Asterisk 1.0.4 and more ...
4:22PM 2 iax encryption
3:51PM 3 Headset with X-Lite
3:47PM 0 Asterisk@Home and / sixTel
3:42PM 1 H323 and ASTCC
3:22PM 7 Stumped on LD questions......
2:51PM 11 PIX!!!!!
2:17PM 0 ASTCC config Problem
1:48PM 2 AstTapi - Crashes w/ Windows 2000 - Urgent Help needed - May need to hire a developer
1:28PM 0 VICIDIAL and meetme conference help
1:07PM 0 Sound quality poor everywhichway
12:48PM 0 Dialplan - intercoms
12:24PM 1 PRI info digits question
12:15PM 6 ringback
11:57AM 0 SIP debugs
11:45AM 1 Realtime Engine
11:12AM 0 is it possible to use Zaphfc (BRI) exactly like i4l?
10:53AM 28 VoIP-to-TDM processing on-card?
10:43AM 0 What's up with IAXTEL?
10:15AM 1 Newbie question - can't get Asterisk to pick up incoming call
9:49AM 0 BRI Fax out through PRI?
9:45AM 0 Meetme Limitations?
9:32AM 1 Weird Zaphfc - not dialling non-local numbers
9:29AM 2 Tips do update Asterisk and AMP
8:41AM 2 Using Zyxel Analog Telephone adapter with a GSM gateway
7:36AM 0 Dial plan problems with realtime extensions ...
6:59AM 3 RE: how to manage Digium TDM04B outgoing calls
6:55AM 5 SIP Stress Test
6:42AM 5 Chan_Capi initial deadlock
6:30AM 0 latest cvs will not compile
6:25AM 7 Some more hardware and E1 questions
6:13AM 2 Asterisk from flash with dynamic voicemail enable/disable?
5:58AM 0 Park/retrieval of calls
4:47AM 2 monitoring packet loss?
4:23AM 3 ilbc high bandwidth
3:39AM 0 change domain caller
3:37AM 0 ztdummy and meetme conference problem
3:10AM 0 Poor sound quality on ISDN BRI calls
3:09AM 0 (no subject)
3:05AM 0 regexten for realtime sip ?
3:04AM 2 hardware details
2:36AM 0 How to read ISDN messages - URGENT!!!!
2:24AM 1 FW: Asterisk 1.0.3 startup
2:23AM 11 API Call Bridge?
2:22AM 0 Asterisk 1.0.3 startup
2:00AM 0 Authentication Problem
Wednesday January 19 2005
8:46PM 1 could someone please tell me how abstraction is provided in asterisk.
7:12PM 2 AGI Environment Dump Question w/ASTCC
6:59PM 0 AGI crash on 1.0.2 on Wait ...
6:32PM 6 Troubles with Broadvoice (register)
6:00PM 0 Why does bristuff generate PRI errors for a BRI only server
5:17PM 6 Call Screen Macro Not Exiting when call rejected
5:15PM 2 IAXTEL errors !
4:26PM 0 Problems transferring calls - Part 2!
4:18PM 0 can callgroup be used to ring a group of phones?
3:55PM 5 RE: how to manage Digium TDM04B outgoing calls
3:43PM 4 ztdummy issues on new asterisk install
3:19PM 4 how to manage Digium TDM04B outgoing calls correctly
3:04PM 1 Calling Voicemail in an AGI script
2:49PM 1 G.729? Worth it? -- YES --
2:45PM 14 E911 Testing !
2:30PM 1 My dialplan just stopped working one day
12:45PM 0 Re: Asterisk-Users Digest, Vol 6, Issue 284
12:26PM 0 Asterisk B2BUA
12:02PM 0 Asterisk vs Proxy SIP
11:59AM 0 IAX line gets 'Hungup' after period of silence
11:45AM 1 echocancellation in modem.conf
11:36AM 0 very big Echo, isdn -> isdn
11:30AM 1 Re: Asterisk bandwidth tuning?
11:21AM 14 Becoming a VOIP provider
11:19AM 0 no sound transmision
11:00AM 0 Cisco 7940 problems
10:56AM 4 Advanced Agents - Need a nice web interface
10:08AM 0 Re: Asterisk monitoring with Nagios and IAX (RoySigurd Karlsbakk)
10:05AM 0 Asterisk fax-modem
9:33AM 4 Accessing Voice mail
9:02AM 13 G.729? Worth it?
8:57AM 1 who changed the codec?
8:46AM 0 Extension Length
8:41AM 0 Play audio to channel
8:32AM 7 queue log analyser?
8:24AM 0 FAX detection in extentions.conf
8:16AM 0 MeetMe MusicOnHold Volume
6:41AM 4 :: Success Case => Motorola 62802-51 as FXO device ::
5:57AM 6 g729 problem
5:33AM 3 Resellers in Europe
5:31AM 0 PSTN Pabx and asterisk
5:00AM 0 h323 compilation problem
4:23AM 0 what does the "c" option in the zap phone number do
3:25AM 1 Re: Busy message on ISDN cards? (SOLVED)
3:11AM 0 iax.conf bindaddr parameter not working
3:09AM 2 How to change the packet size
2:47AM 6 Re: Asterisk monitoring with Nagios and IAX (Roy Sigurd Karlsbakk)
2:39AM 8 Fax and PRI
1:49AM 7 Asterisk not recognizing key beeps
12:24AM 1 Can IAXy be setup for PPPoE ???
Tuesday January 18 2005
11:49PM 0 sip-sip
10:32PM 22 # Transfers.
10:15PM 5 Open Source QoS .
9:32PM 0 X100P not working: no sound
9:15PM 0 Hardware Requirement & Setup
8:20PM 0 LogWatch emails in /var/spool/mail/root
8:01PM 0 C/C++ SIP Phone Development Lib or Stack
7:36PM 7 Wellgate 3804 Firmware
7:03PM 9 Polycom Call-Waiting
7:01PM 5 Router Recommendations Please
6:45PM 8 Reverse phone lookup interface with asterisk
6:34PM 0 spandsp & tdm400p - recommended hardware
5:43PM 1 Cisco 7940 Configuration
5:05PM 15 Cisco 7940G
4:32PM 2 Asterisk and h323
4:25PM 2 Is an unregistered phone busy?
3:56PM 2 Grandstream BT102
3:48PM 14 Newbie question: Can't start up asterisk
3:28PM 2 problems compiling asterisk-addons
3:21PM 0 asterisk and predictive dialers
3:11PM 2 something between an ATA and a channel bank for a small office?
2:31PM 1 Asterisk and IAX softphone (firefly) problem/question
1:36PM 0 TDM400P card & PCI problems
1:27PM 1 Re: Asterisk bandwidth tuning?
1:19PM 1 R2 - Stable Asterisk
1:18PM 0 CallManager 3.1 (2c) and Asterisk Integration
1:09PM 0 Issue using IAX2 as end-point (IAXComm)
12:45PM 0 RE: mgcp <-> h323 problem
12:39PM 2 Asterisk - libunicall - MFCr2 *** settings problems ??? ***
12:10PM 0 Database of event activity
12:10PM 2 What's the easiest way to call two people at same time and bridge them?
12:01PM 0 Urgent handler messages on * 1.0.3
11:56AM 1 QoS tagging - can Asterisk do this, if not, what do you recommend?
11:39AM 0 Error after switching from 1.0.2 (FreeBSD) to 1.0.3 (Gentoo)
11:36AM 0 DTMF is being MUTED by asterisk to/from SIP channels from SIP or ZAP
11:14AM 1 Quick Question on Wildcard T100P
11:06AM 0 SIP URL ?
10:45AM 1 Delay after Dial Application is Called
10:23AM 6 Versatel PRA in Belgium/Netherlands
10:01AM 1 Re: * compatible with Pulse dialing phones ?
9:54AM 12 Broadvoice Patch Error {Scanned}
9:39AM 0 is it possible to use a sp2000 for intercom/paging?
9:36AM 3 External fax modem takeover of fxo?
9:06AM 0 Out of 5 Grandstream BudgeTone 101 THREE are
8:39AM 3 Outbound Dial via SIP
8:31AM 1 Problem with registering Windows Messanger with asterisk
8:26AM 1 Flat Rate Long Distance Providers
8:23AM 1 No compatible codecs
7:21AM 1 Problem with demo on asterisk
7:06AM 2 8 x 8 Analog System for Auth and Minutes Tracking
6:53AM 2 ISDN + chan_capi
6:40AM 0 No Busy signalled to caller
6:30AM 5 sipura 3000 mwi stutter problem
6:28AM 7 TE110P as E1
6:20AM 0 AMP and Asterisk PSTN extension config
6:11AM 32 Attended call transfer
6:07AM 10 Asterisk monitoring with Nagios and IAX
5:58AM 10 Prefered server hardware
5:13AM 2 Realtime Voicemail ...
4:51AM 0 TDM400 - incomming call is answered but if i hang up asterisk never detects it
4:28AM 3 Outgoing SIP call from Asterisk problem
4:21AM 0 Will queueing only work after answering a ca ll?
4:13AM 0 Vmail
4:02AM 1 Will queueing only work after answering a call?
3:57AM 1 Multiple Alsa Devices
3:20AM 4 Outbound calls unpredictable
2:41AM 2 MFCR2 - LIBUNICALL - Asterisk Problems
1:34AM 23 Best Grandstream firmware to use?
1:24AM 8 fax over tdm400p
1:00AM 1 Number of Calls per Proxy on Cisco 7960G?
12:47AM 0 Dial Plan Agents (2 of 2)
12:46AM 1 Dial Plan Agents (1 of 2) agent-dialplan.conf
12:46AM 3 Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore)
12:15AM 0 Canadian Content: Telus and Shaw...
12:06AM 1 Auto Protocol (depending upon registration....
Monday January 17 2005
11:58PM 6 Is anybody using an IAXy?
11:20PM 3 Planning "hotel" phone system - Need input
11:02PM 6 Sound quality - commercial vs. Asterisk
10:33PM 0 VoIP Routes and Terminations
9:03PM 3 On Hold music
7:32PM 11 internal dial tone on password from outside
6:48PM 0 TDM13B - FXO ports not seeing incoming calls
6:42PM 9 SIP URL for incoming
6:22PM 1 here's my IAX callthrough app and some questions about problems I have.
6:12PM 0 Transferring calls on Asterisk with X-Lite
5:51PM 6 spandsp and app_txfax
5:32PM 4 callers who don't press any keys
5:12PM 4 transfers with zap channel
4:44PM 7 Wait(n) -v- Background(silence/n) ?
4:43PM 2 iaxtel - -- Format for call is ADPCM
4:23PM 1 Looking for Asterisk termination in Russia
3:40PM 1 Re: Any interest in a Canadian Asterisk
3:27PM 2 China direct route
3:14PM 0 Multiple Line Caller Id With Polycom IP500
3:09PM 0 Queue and Normal Transfer.
2:41PM 1 Echo on SIP -- not on analog.
2:38PM 1 X-Ten lite troubles.
2:25PM 12 Media Path Optimization & NAT
2:22PM 0 How to call an extension number from ohphone to astersisk
2:05PM 5 FW: Radius on *
1:49PM 4 Offtopic: improving softphone latency on Linux?
1:45PM 1 ZAP/PRI Error: channel reported in use
1:44PM 2 Directory() Command
1:38PM 4 IAX2 doesn't respect bindaddr?
1:34PM 1 Attempting native bridge
12:46PM 1 spandsp recieve problem
12:23PM 0 Jamaica - My apologies for the second post.
12:13PM 2 RE: Issue compiling zaptel on FC 3 kernel 2. 6.10-1.737
11:57AM 0 Jamaica
11:35AM 3 Asterisk C source code documentation
11:32AM 2 Jamaica DID
11:29AM 1 RE: Issue compiling zaptel on FC 3 kernel 2.6.10-1.737
10:46AM 6 iaxtel - best codec
10:28AM 1 is asterisk a good solution?
10:00AM 0 SIP/H323 modules for netfilter
9:57AM 0 How to implement an audio delay?
9:45AM 4 DIDs anywhere but here?
9:13AM 5 simple over view of the process
9:11AM 3 Is it possible to ID payphone calls?
9:01AM 0 DIAX 0.9.9g more features and higher stabili ty
8:15AM 0 RE: [Asterisk-biz] Guatemala DID's?
8:11AM 11 SIP IOS for cisco 7902G IP Phone
7:52AM 1 ASTCC single stage + no access number + auth usingsip username and password
7:41AM 4 ntp Server and Zultys 4X4
7:20AM 1 Communication Between Phones... I can't test :(
7:12AM 10 TDM400 answers the line all the time!
6:33AM 4 CAS voice signalling?
6:13AM 0 chan_capi outgoing msn
5:55AM 0 voicemail sound distorted - chan_capi, diva, cvs-head
5:53AM 0 Can I start recording channel in the middle ofconversation ?
5:45AM 1 Can I start recording channel in the middle of conversation ?
5:33AM 0 Manager Event Logging
4:56AM 2 error compiling
4:30AM 2 Using a variable for EXTEN
3:26AM 2 Does Asterisk do that?
3:22AM 0 Can I get info about email addresses from voicemail.conf in dialplan or variables ?
3:10AM 0 ASTCC single stage + no access number + auth using sip username and password
2:57AM 0 AGI / Sockets
2:53AM 1 Euro ISDN and Caller ID (Sweden)
1:14AM 2 Adding SIP clients using AGI ?
12:44AM 1 quadBRI asterisk error message message: "not able to open Zap channel"
12:29AM 0 voicemail attach not in 1.0.2 ?
Sunday January 16 2005
11:47PM 30 Any interest in a Canadian Asterisk mailing list?
11:29PM 21 pattern matching problem
11:25PM 1 Asterisk over External Motorola BitSurfR Pro ISDN Modem
11:10PM 0 Registering with IAX provider
9:18PM 0 Looking for a VoIP provider for my Asterisk box. {Scanned}
8:43PM 1 New Sipura-841 phone.Mike volume problem.
8:37PM 2 FWD<->NAT<->*
6:45PM 1 VOIP - INBOUND Call - best setup
4:59PM 1 Meetme conf and Shoutcast
4:37PM 0 X100P with no sound!
4:25PM 3 IAX.conf error
4:10PM 5 IAX1 vs. IAX2
2:23PM 1 Guatemala DID's?
2:12PM 1 chan_sccp and bristuff 1.0.3 weirdness
1:51PM 1 Type of Number
1:22PM 1 Inbound Callerid for SIP Phones
12:29PM 0 MVP110 and *
12:25PM 3 H323 Softphone for iPAQ
11:46AM 0 The BEST? analog phones for *
11:09AM 1 VoIP Newbie
10:25AM 0 Re: asterisk-users list and html posts
9:19AM 2 Looking for help with a Polycom Soundpoint IP 600
9:06AM 0 Re: Asterisk-Users Digest, Vol 6, Issue 227
7:59AM 0 * reports the incoming caller id but not the BT100
7:37AM 3 TDM400 lost after reboot
5:40AM 0 TDD support in Asterisk?
4:32AM 0 sound-recorder crash when I start Asterisk
4:26AM 0 Extension.conf, sip.conf and contexts.
1:59AM 6 announcing caller id?
Saturday January 15 2005
11:16PM 5 TDM400P NO BATTERY & Poopy???
9:03PM 3 failed to compile zaptel on redhat
8:51PM 1 TDM400p FXS not sending caller id info?
6:48PM 6 How to demo wired phone set on a wireless network
6:24PM 1 SayDigits -- ToneDigits??
5:28PM 2 IAX2 one side loses audio
4:57PM 0 X100P no sound problem
4:18PM 13 NuFone help
4:14PM 0 Polycom IP600 - Bridge stops because we're zombie or need a soft hangup
3:37PM 0 oh323 driver - [user] type=user
3:18PM 1 ATA with IAX protocol
2:40PM 0 Is it the 15th or the 16th :)
2:39PM 2 No more loading asterisk...
1:39PM 3 CAC Channel Bank I - FXS
1:02PM 0 Sip registration period
12:58PM 5 IAX2 Channels & Bandwidth
11:20AM 0 Add h323 support to Asterisk
11:05AM 0 oh323 compile error
10:54AM 1 can't install 1.0.3
10:53AM 1 failed to compile zaptel on redhat (kernel 2.4.20-31.9)
10:44AM 0 Asterisk to CCM3.3.4 via H32
10:18AM 0 ADSI unlock codes
9:26AM 0 Newbie - Asterisk Tramsfer Problem?
8:39AM 0 configuring ser for *
7:52AM 5 Voicemail after one ring?
6:20AM 0 call deflect with QuadBRI how to
6:12AM 5 Return of experience : Asterisk more stable with 2.6 or 2.4
5:51AM 0 Problems using chan_capi over Fritz!Bluetooth
5:31AM 1 switches
5:23AM 0 sip.conf
4:51AM 1 Packet8 DTA310 SIP Image
4:01AM 1 Re: Budgetone and MWI
3:06AM 1 spa 2000 phones do not ring
1:49AM 3 voice output
1:36AM 2 No sound with X100P (clone)
1:34AM 0 Anyone use SunRocket with Asterisk?
12:07AM 4 DIAX
Friday January 14 2005
11:47PM 3 Echo Training - how long
11:27PM 1 Asterisk@Home Install Problems
11:25PM 1 voice quality with asterisk
11:18PM 1 voice quality in asterisk
9:03PM 8 Remote Voicemail Retrieval...
8:19PM 5 Asterisk and Voice Pulse Open Access
7:30PM 1 Proxy-auth
6:27PM 2 iaxComm 0.99pre11 binaries posted to Sourceforge
5:08PM 4 Packet8 DTA310 and Asterisk
3:59PM 2 Routing incoming calls to various extensions.
3:50PM 2 ULaw not negotiating
3:38PM 3 Having trouble with T405P and PPP: ZT_SPANCONFIG failed
3:16PM 1 DIAX PC to Phone
2:44PM 1 Asterisk for voicemail -> C2611XM, 7940 & 7960 phones
1:22PM 0 Strange CRCX
1:12PM 1 SIP Registration problem, 403 forbidden
12:49PM 0 app_conference compile?
12:28PM 0 IAX on multiple ports
12:13PM 1 context wide variable scope
12:06PM 0 OT: zaptel kernel mod
11:53AM 1 PrePaid Applications
11:53AM 2 PC to Phone
11:37AM 5 Softphone for Linux recommendation
11:30AM 1 T100P with NEC C2400 IPX switch
11:03AM 0 Re: Grandstream Bugetone 101 & mw
10:57AM 1 Re: Budgetone and MWI
10:11AM 2 gotoiftime - different hours
9:56AM 1 Polycom SoundPoint IP by Shoreline
9:55AM 0 Zultys Phone feature
9:48AM 0 Newer CVS-Stable Asterisk not recognizing G711 ULaw from certain providers
9:38AM 0 IAX2 bridging = one way audio
8:47AM 9 Realtime / sip.conf
8:14AM 5 Spandsp....And garble incoming fax
8:03AM 0 Re: SOS
7:53AM 2 Passing PIN Numbers
7:24AM 1 handle_request registration failed?, Polycom IP500
7:22AM 0 SOS !!!
5:46AM 1 iconecthere and *
5:23AM 0 problem in calling
4:45AM 4 [Slightly OT] SIP/T.38 capable system, anyone?
4:30AM 3 Hardware issues
3:25AM 1 incomplete address
3:17AM 0 Can Asterisk generate a 404 message back to a UA?
3:15AM 1 Suse 9.2 / Latest CVS
3:10AM 0 Environment variables
2:16AM 1 Grouping lines pending on Called ID
2:06AM 0 troubles with getting odbc to load data
2:03AM 0 caller's identity
1:06AM 0 Could I "SET AUTOHANGUP()" count down after the channel state is UP
12:00AM 1 Limit outgoing trunk calls
Thursday January 13 2005
11:37PM 0 Polycom Shared Call Appearance
10:15PM 1 sporadic beeps spa3k-*
9:16PM 4 I Don't Want Asterisk in the Media Path
9:12PM 0 Asterisk@Home systems
9:09PM 2 Updated kphone 4.0.5, asterisk v1.0.3
8:14PM 8 Voice Mail Notification
7:24PM 0 Iaxtel directory
7:10PM 2 Firefly repeats registering to * server
6:05PM 4 REGISTER Problems With Realtime
4:39PM 0 Grandstream Bugetone 101 / documentation
4:37PM 0 Oh323 compilation errors
4:35PM 0 Hook-Flash on Voicetronix
4:22PM 0 voicemail function
3:18PM 43 DIAX 0.9.9g more features and higher stability
2:24PM 13 long delays in list posts?
2:22PM 1 MWI on Zap analog phone not lighting
1:37PM 0 Customer Service Coaching
1:31PM 3 ATA186: SIP/2.0 503 Service Unavailable
1:24PM 10 PRI concentrator
1:18PM 1 problems with astcc
1:17PM 3 High delay with diax099f + Asterisk
1:03PM 0 Looking for a wireless phone... wifiortradit ionalwireless ?
1:00PM 5 Security audit scripts
12:24PM 0 Looking for a wireless phone... wifiortraditionalwireless ?
12:20PM 2 Problem patching asterisk CVS with SpanDSP
12:11PM 0 Re: Budgetone 10x & mwi
12:10PM 5 TDM04B vs Dell revisited
11:49AM 2 1xT1 PCI card for *
11:46AM 2 SIP registration error, lost packets with asterisk
11:38AM 1 Re: Looking for a wireless phone...
11:35AM 1 Build PWLIB
11:25AM 1 Enabling/disabling zaptel echo-can from dialplan.
11:12AM 0 Want to install Oh323 and LOST
11:11AM 2 asterisk won't release line
11:01AM 2 SMS Gateway
10:56AM 0 PRI dchannel in use?
10:54AM 4 Howto DTMF pass-through on a channel
10:48AM 1 Not In Local Context
10:33AM 1 Re: R2/MFC Mexico FREE calls to test chan_unicall (Miguel Cavazos)
10:27AM 0 Asterisk doesn't detect when the caller hangs up
10:19AM 1 Status of latest round of Allison recordings
10:09AM 3 error 488
9:56AM 4 Manager API !!!!!!!!!
9:54AM 5 Cisco 79XX phones not talking to asterisk
9:50AM 4 SER vs Asterisk for SIP
9:24AM 0 Xfering a call
9:15AM 4 How to present a dialtone to a dial-in user?
9:12AM 1 Teleconferencing?
9:09AM 5 Asterisk on a notebook?
9:03AM 2 Queue Log Parser
8:31AM 30 How to set asterisk NOT to answer incoming lines?
7:45AM 2 about AGI command parsing
7:41AM 3 Looking for a wireless phone... wifi ortraditional wireless ?
7:30AM 7 Agentcallbackogin without any user input after extension is dialed.
6:45AM 0 current CVS version
6:30AM 1 SIPGetHeader
6:29AM 1 About HDLC in ISDN
5:31AM 1 Hunt group with Accept/Reject Option
5:19AM 0 oh323 compile problem still
4:42AM 2 ASTCC dimensioning
4:32AM 1 MeetMe does not compile with Asterisk
4:11AM 2 Grandstream bt-100 loosing it!
4:09AM 4 SCCP questions
4:03AM 1 asterisk realtime msql
2:15AM 2 pseudo-realtime??
1:22AM 1 Registration of SIP
1:09AM 0 Replacing Cisco3620 with Asterisk
Wednesday January 12 2005
11:40PM 6 snom220
11:13PM 21 Grandstream Bugetone 101 & mwi
9:23PM 0 moh mp3 streaming problem
8:42PM 0 pass through mode
8:23PM 0 IAX peering between two Asterisk servers, how?
7:09PM 12 Is this a $50 wifi or wireless USB VOIP phone ?
6:48PM 1 no playback audio
6:34PM 1 SNOM 190 Configuration with Asterisk
6:08PM 0 Volume in line for music-on-hold
6:06PM 0 BT keeps open sip channels
5:48PM 5 Bristuff 0.20RC3 loses connectivity after short line interruption?
5:32PM 3 Trouble building appradius
4:40PM 2 New SIP Phone Config
4:29PM 0 IAX2 dropped calls: need debug suggestions
4:20PM 9 Setting channel display in SIP
4:17PM 0 Re: EuroISDN BRI 2 or 4 wires? (Remco Barende)
3:22PM 20 R2/MFC Mexico FREE calls to test chan_unicall
3:19PM 0 Queue and penalties
3:09PM 0 getting * to start on suse 9.1
3:06PM 1 EuroISDN BRI 2 or 4 wires?
2:44PM 4 Cant receive calls after network goes down and up
2:39PM 0 Come join the Asterisk Bookclub
2:14PM 0 SIP Authenication (Simple, Digest, ACL)
2:06PM 3 linphone -> NAT -> * -> NAT -> firefly woes.
2:05PM 5 Using asterisk to convert H.323 to SIP?
1:54PM 0 calling an extension after a voicemail is left
1:50PM 1 Asterisk server stopped - "0-order allocation failed " errors in the log
1:49PM 0 Asterisk variables - size limitation?
1:31PM 0 wctdm and alaw audio quality problem
1:10PM 0 Setting "User Info" in extensions.conf? (ZyXEL P2000W)
12:29PM 0 Asterisk + SER Questions
12:05PM 2 Where to buy a quadBRI?
11:40AM 0 OT: Asterisk hits Slashdot again
11:11AM 0 generating CPC for non-CPC analog FXO lines - suggestion/discussion requested for use with TDM400P X100P
11:02AM 2 Ports to open behind a NAT
10:53AM 10 Polycom IP 500 Dial Issues
10:43AM 1 Asterisk version naming convention!!
10:36AM 2 Call Manager or Asterisk
10:34AM 0 FW: asterisk - oh323 driver
10:02AM 0 So many Asterisk Patches - Which do I choose anduse?
9:56AM 1 HW ? on Getting a new Asterisk Box
9:39AM 1 spandsp on FC3
9:29AM 3 T1 Timing Slips
8:55AM 0 Doc Asterisk
8:21AM 9 So many Asterisk Patches - Which do I choose and use?
7:44AM 2 Unofficial Broadvoice-users query/offer and DID routing question
7:38AM 0 chan_misdn - new release ! Please test it.
7:28AM 0 Problem solved on Xonox Asterisk distribution
7:17AM 1 PRI RLT support
7:10AM 0 astweb cdr's mysql.sock problem
7:07AM 10 Operator Panels?
6:58AM 3 What is the best and easiest flavor to be usedwith Asterisk.
6:28AM 6 chan_capi-0.3.5 error 127
6:10AM 4 Re: [Asterisk-biz] SS7 and Asterisk solution
5:51AM 2 H323 on Asterisk@Home
5:43AM 0 connect asterisk to lingo without ata
5:40AM 6 Re: [Asterisk-biz] SS7 and Asterisk solution
5:25AM 2 Stale mp123 processes??
5:10AM 8 What's the easiest way to get * to call PSTN?
4:22AM 1 Can I use spandsp with Asterisk on Fritz with Capi ?
4:17AM 0 Terminating VOIP calls on EuroISDN PRI interface ?
4:16AM 0 ChanIsAvail + Zap and SIP channels
4:01AM 1 Problem
3:13AM 1 How to configure three ISDN line
2:16AM 0 Attended transfer problem with Atxfer
Tuesday January 11 2005
10:37PM 0 Asterisk User Group in Winnipeg, CA
9:55PM 1 Fast Start , Slow Start , or just Codec
8:58PM 10 What is the best and easiest flavor to be used with Asterisk.
7:56PM 5 AMP Anyone?
6:44PM 46 SS7 and Asterisk solution
5:26PM 0 PA-168(S) - Netweb -301 Phone
5:01PM 1 rxfax troubles..
4:34PM 2 PA-168(S) - Netweb IPweb-301 Phone
4:30PM 12 test-ignore
4:23PM 0 Blank Voice Mail messages
4:19PM 3 No sound for music on hold
4:08PM 0 How to enable debug
4:03PM 0 "Telco power supply" with digium card TE410P?
3:57PM 8 Changes to manager outputs - A discussion
3:51PM 3 Dial Out Errors
3:45PM 7 "o" extension broken?
2:52PM 2 Channel IAX2 Socket Read Error
2:13PM 18 not sharing IRQ's
2:06PM 0 Agent autologoff=15
1:38PM 0 Planet VIP-101T or VIP-150T
1:31PM 1 Direct SIP calls to *
1:22PM 3 BroadVoice outgoing works - now tackle caller ID
1:15PM 3 (UN)structured E1
1:05PM 1 Dlink DPH-80 DONT work with asterisk
12:55PM 1 ACD Bug with AddQueueMember Stable
12:51PM 0 Not hanging up. {Scanned}
12:47PM 3 iax.conf qualify=yes not working?
12:23PM 1 PRI Errors (HDLC Abort (6) on Primary D-channel)
12:06PM 2 How to prevent a call from going to voicemail when one phone is offline?
12:01PM 1 Tool Recommendations for measuring UDP throughput / loss / jitter
11:38AM 2 TDM box Hardware
11:01AM 0 operator says that dial 1 or 0
10:33AM 5 SIP, * and clients behind NAT
10:31AM 0 Sounds cut out problem - HFC-S card, zaphfc, Xlite
10:24AM 4 internal caller id on analog phones connected tozap
10:04AM 1 ACD Queues & Agent Status
9:36AM 0 Re: Asterisk-Users Digest, Vol 6, Issue 144
9:25AM 0 howto dump binary data on zap channel?
9:12AM 0 RE: Asterisk-Users Digest, Vol 6, Issue 142
8:34AM 0 Newbie question: call routing
8:25AM 9 Installing * on fedora 3
7:44AM 2 ASTCC - error on call end
7:30AM 1 internal caller id on analog phones connected to zap
7:26AM 6 sip to h.323
6:01AM 0 Cisco ATA 186 for PSTN dialing
4:43AM 1 How to mark a user for a conference
4:07AM 6 Realtime and include
3:59AM 3 Analogue RAS Server
3:46AM 1 asterisk one number service
3:40AM 0 test source for current xorcom rapid
3:35AM 3 requiring logon for SIP users
3:10AM 5 asterisk-oh323 and outgoing call
2:17AM 0 AGI Application Hangup when using AGI->getdata
1:29AM 0 What is acceptablenetworklatencyforvoipconnection?
1:03AM 0 Asterisk Segmentation Fault - layer3.c/mpg123
Monday January 10 2005
11:29PM 1 dialing into * then forwarded out gets choppy audio
11:07PM 1 TE110P with Telstra E1 PRI in Australia and New Zealand
11:05PM 0 test {Scanned}
10:36PM 1 echo cancelation on Digium T1 cards
10:33PM 0 TE-405P freezing, anyone else?
10:31PM 0 asterisk router problem
9:10PM 0 Russian characters showing up on safe_asterisk console in RedHat 9 and Fedora Core 2
8:47PM 5 Route incoming call on 4 X100P to different Ext. {Scanned}
8:01PM 0 Asterisk not answering calls since oh323 upgrade
7:41PM 0 CallerID presentation
6:23PM 4 Sip to IAX ok, ZAP to IAX FAILS
6:20PM 0 Any movement on IAX being submitted to a standards body?
6:08PM 0 zulty's ZIP 2 IP phone
6:08PM 2 Weir long distance behaviour...
5:50PM 8 Generic modem question
5:40PM 0 SOYO G668
5:03PM 0 Mental Blank: HELP: I cant get any callerid on capi incoming?? WHY
4:16PM 3 Asterisk calls back after phone call
3:30PM 0 64 Bit Support?
3:15PM 2 /usr/bin/ld error on make asterisk with Fedora Core 3
3:10PM 2 Call Waiting + Call Transfer Problem
2:12PM 3 Some questions (maybe Nikotel related)
2:11PM 4 "make clean" DO IT!
1:59PM 2 SIP Reorder tones
1:58PM 0 dead line (no LED) on a TDM400B?
1:48PM 2 Power Failure, Line Switch, Relay device
1:46PM 4 ACD Queue question.
1:27PM 4 Is this a firefly problem? (*78/*79 doesn't work)
1:21PM 2 IAX2 keep alive?
1:09PM 9 Multiple gateways for same dial pattern
12:52PM 0 Connecting Asterisk to a Toshiba Strata syst em
12:17PM 0 sip channel between 2 asterisk box
12:16PM 5 X100P in a soekris 4801
11:54AM 4 Festival Woes
11:38AM 9 Help! - Unintelligible prompts and music
11:29AM 9 Ring Voltage Supplied by Wildcard TDM400P REV E/F & AUTO FXS/DPO
11:17AM 1 Broadvoice call quality?
11:17AM 1 Static/Breaking up after I upgradedAsteriskaswell as a crash - Can't trace bug
11:08AM 0 ChanSpy Usage
11:03AM 1 IAX2 provider in Montreal, Canada
10:37AM 0 IAX Local numbers in Wyoming
10:34AM 0 Extensions config help please..
10:17AM 3 Agent Status on FOP
10:16AM 2 R2 for Mexico?
10:12AM 1 Static/Breaking up after I upgraded Asterisk aswell as a crash - Can't trace bug
9:57AM 0 Static/Breaking up after I upgraded Asterisk as well as a crash - Can't trace bug
9:11AM 5 Any Notices from voiceconduit?
9:00AM 7 Vmail.cgi - "Hrm, can't seem to open /var/spool/asterisk/voicemail ....
8:56AM 2 Connecting a Home based worker with An Iaxy
8:50AM 0 Problems calling between two local SIP extensions
8:43AM 2 Ramifications of Multiple Sip Reloads Within Minutes?
8:37AM 0 [Fwd: Re: Asterisk-Users] very loud scratchy noise!]
8:36AM 4 Execute dialplan command at startup
8:20AM 1 Digi Datafire Micro V ISDN Card
8:19AM 4 Asterisk Setup Documentation
8:12AM 0 conference question...
8:12AM 0 open g723+limiting the out bound calls
8:10AM 2 Asterisk UK Community
8:03AM 3 Connecting Asterisk to a Toshiba Strata system
7:45AM 14 Request to schedule in the past?!?!
7:41AM 12 Asterisk to PSTN
7:26AM 1 Re: Asterisk-Users] very loud scratchy noise!
6:57AM 6 UK * group
6:49AM 1 Re: Toronto
6:38AM 5 FXO PCI Master abort
6:35AM 0 AGI EXEC trouble
6:21AM 11 Zaptel problems
6:17AM 14 Unicall errors
5:57AM 3 very loud scratchy noise!
4:58AM 0 Agents question
4:36AM 0 error?
3:14AM 3 OT: SIP Aware Firewall with Asterisk
2:03AM 11 audio delay ISDN
12:57AM 1 SetGroup
12:46AM 1 I need your feedback related to the DIAX 0.9.9f stability
Sunday January 9 2005
11:23PM 4 TE110P error
10:42PM 0 Asterisk as H323 client?
9:16PM 7 Help in E1-T1 encoding
8:43PM 8 telemarketing application
5:54PM 3 What is acceptable network latencyforvoipconnection?
5:20PM 0 RE: Inbound calls getting disconnected when I answer the phone, using 'SIP'. "FIXED"
5:03PM 1 Can zaphfc (bristuff) do caller id?
4:54PM 0 Quicknet Internet Phonecard
4:30PM 11 Little confused about Caller ID
4:15PM 2 Making a call using MGCP
4:07PM 3 E&M trunk card?
3:57PM 0 call from PSTN, not hearing SIP: 180/RINGING( was call from DID,not hearing RINGTONEs )
3:47PM 2 What is acceptable network latency forvoipconnection?
3:43PM 0 Caller ID in Australia
2:36PM 0 TDM4000 FXS and UK Caller ID
1:55PM 0 Asterisk SIP channel (PSTN Calls)
1:51PM 5 Asterisk Demo
1:47PM 0 Incoming no.s being dropped.
1:42PM 2 ASTCC Trunk and Routes Configuration
1:39PM 1 History of the Zapata Telephony Project as it relates to the Asterisk PBX
12:52PM 5 Bristuffed Asterisk 1.0.3 hfc-s card doesn't work
11:33AM 1 [OFF TOPIC] Voip phone sellers in India
10:38AM 0 RE: Inbound calls getting disconnected when I answer the phone, using 'SIP'
10:09AM 0 Using Goto with Asterisk Realtime configuration
9:47AM 2 Asterisk and InterTel Axxess system?
8:47AM 2 GSM adapter for analog telephone - connect with fxo or fxs to Asterisk
8:42AM 1 Wait indefinitely?
8:38AM 2 passing opermode to the wcfxs module
3:29AM 1 PRI AOC (Advice Of Charge)
3:18AM 2 X100P random hangups - Please help with suggestions
1:43AM 0 Extension No.s not being received correctly.
12:00AM 2 Inbound calls getting disconnected when I answer the phone, using 'SIP'.
Saturday January 8 2005
8:30PM 2 No such extension {Scanned}
8:14PM 5 ASTCC questions
7:20PM 2 Connecting Phone To Asterisk
6:25PM 0 Re: Asterisk-Users Digest, Vol 6, Issue 105
4:51PM 4 OT help with rmdir pls
4:14PM 0 Asterisk and echo
3:02PM 11 Echo on Zaptel FXO :(
2:12PM 0 FastAGi change
12:46PM 1 Monitor command volume
12:05PM 0 MGCP phone
11:16AM 0 FYI: NIST issues recommendations for secure VOIP
10:25AM 0 484 Address Incomplete
10:08AM 5 Best gateway to use for *?
9:45AM 0 How to use a codec depending on call type ?
9:29AM 2 SIP and NAT problems "imagine that :) "
9:17AM 3 virtual pbx
8:43AM 1 What is acceptable network latency for voipconnection?
8:06AM 9 zaptel fxotune.c tool
7:54AM 0 Any experience with Linksys WRT54GP2 as localextensions to Asterisk ?
7:29AM 0 Any experience with Linksys WRT54GP2 as local extensions to Asterisk ?
6:31AM 1 Asterisk calls without soft phones
5:04AM 0 Problem SJPhone+Qtek S100 PDA+Sandisk Wi-Fi 256 MB SD+Asterisk
4:30AM 12 France has their (first?) SIP carrier with "unlimited" calls for 6eu/mo
4:23AM 1 Wildcard x100p and Redhat 9.0: Unable to get parameters
3:40AM 9 Toronto?
1:08AM 10 How do i "talk" to the IAXy...? (Newbie Alert)
Friday January 7 2005
10:43PM 0 asking for readers input into the following config...
10:06PM 0 Inbound Pickup Issue - Sipmedia
9:34PM 5 Connecting Sip phone to asterisk.
9:00PM 4 MINNESOTA: TwinCities Asterisk Users Group - Meeting tomorrow 01/08/05 11:30am
8:19PM 0 New 'n' priority
5:27PM 7 Channel Variable
2:52PM 2 Newbe Can't dial local numbers.
12:20PM 2 Loading module failed!
12:13PM 2 xmitting CallerID
10:13AM 1 New York?
9:48AM 0 mantis password reset link
9:45AM 3 Ringing an extension on multiple phones
9:42AM 4 can the dialtone be changed after pressing 9?
9:11AM 0 Re: [Serusers] softphones
9:08AM 7 Moderator on vacation?
9:01AM 5 Setting up Polycom IP 500 with *
8:40AM 1 oh323 driver installation - It works now
8:35AM 3 Asterisk 1.0.2 - Unable to allocate channel structure
8:34AM 1 Test2
8:31AM 0 Re: kind of Urgent (Fedora Core 3 & Asterisk)
7:40AM 5 Monitoring
7:39AM 5 International area codes (incl. mobile)
7:29AM 0 PolyCom IP3000, gnugk and * audio problems
7:11AM 0 How do I get version 1.x from theDigium CVS orelsewhere?
7:07AM 15 fax e-mail spandsp
7:03AM 0 How do I get version 1.x from theDigium CVS or elsewhere?
6:50AM 0 x100p to X-lite works but x-lite to x-lite not (can not transmit audio)
6:02AM 3 TDM400P - Segmentation fault - Help!
3:39AM 2 Sip protocol question ...
3:36AM 1 off topic - SSH configuration for Digium Support
2:48AM 1 specific call transfer
2:43AM 7 Problem with call pickup
2:04AM 1 NIC irq load balancing
12:42AM 0 how to config call waiting and three way calling
12:19AM 0 Sip Phone Won't Login...
12:18AM 4 Broadvoice Status Check 11:18pm PST
Thursday January 6 2005
10:57PM 1 {Scanned}
10:19PM 9 IAX outgoing redundancy
10:19PM 4 Enhancing performance and utility of an Asterisk machine
9:42PM 0 Reception System
8:11PM 0 TA register to Asterisk and getting down after notify msg, why?
8:06PM 10 Message light on 7960 or in this case no message light
7:58PM 9 Multiple lines on Cisco 7960
7:25PM 0 Asterisk and Samsung DCS integration
7:02PM 9 Queue app following dialplan
6:47PM 0 T.38 Passthrough
6:39PM 7 Sipura SPA-1001 and Tivo Series 1
6:32PM 1 ASTCC Bounties
6:15PM 2 3 site asterisk installation question
5:59PM 0 Internal/External IP
5:53PM 0 Line drops after 5-10 seconds
5:47PM 0 chan_capi compile problem
5:02PM 0 call waiiting and 3 way calling
4:48PM 1 Strange problem with incoming call.
4:43PM 1 Numbering plan for incoming call CLID on chan_zap (PRI)
4:12PM 4 What's wrong with compile
4:11PM 6 TE410P problem (Looping UP Span 1...)
3:56PM 4 Sip providor reference in extentions.conf
3:37PM 13 TDM4000P with 4 FXO's not picking up ringing lines
3:17PM 3 OT: TE405P pins and slots
2:42PM 0 Asterisk and SER security doubts
2:14PM 24 spandsp and app_rxfax (alternative topic: t38modem)
2:12PM 0 Incoming calls from I-net only for IP-address?
1:51PM 0 Re: Asterisk-Users Digest, Vol 6, Issue 76
1:26PM 0 Re: kind of Urgent (Fedora Core 3 & Asterisk)
1:21PM 2 Zaptel Compile
12:59PM 3 Re: Asterisk-Users Digest, Vol 6, Issue 73
12:15PM 1 Number of Zap channels in use
12:11PM 0 re: asterisk and libretel
12:11PM 2 POTS Lines
12:01PM 2 destroy SIP channel??
11:39AM 0 Asterisk latest from CVS: SIP registrations fail
11:10AM 4 ZapRAS with BRI
11:05AM 2 TDM400P - Segmentation fault
10:32AM 25 kind of urgent
10:08AM 1 T100P + Adtran TSU600 + FXO and caller id problems
9:55AM 0 using native moh
9:46AM 0 H.323 to SIP extension
9:46AM 0 Asterisk and multiple default routes (sort of) - does not work
9:42AM 0 Call Pickup Problem
9:04AM 0 Problems with FXO interface on TDM400P
8:46AM 13 Asterisk startup
8:33AM 0 Out of Office AutoReply: asterisk addson
7:58AM 2 Inbound calls (similar problem; ISDN - chan_capi)
7:30AM 4 answer supervision for POTS FXO interfaces
7:24AM 0 Does spandsp work with capi channels ?
7:19AM 1 zaptel service stopped working
7:15AM 2 TDM400P - Problems
7:01AM 5 Problems with MeetMe accepting conference PIN
6:58AM 0 FW: Re: Polycom IP500 - problems with multiplesimultaneous calls
6:30AM 2 Gotoif question
6:19AM 1 Twin Cities Asterisk meeting still on for Saturday?
6:18AM 1 Sip Subscribe
6:13AM 0 Four HFC-S Cards in one System - does it work?
6:10AM 0 ISDN, bristuff and hylafax
5:35AM 4 .call MeetMe
5:22AM 3 Changing caller ID based on the extension dialled?
4:55AM 25 fax to email
3:41AM 0 mp3player - sounds terrible
2:52AM 3 DTMF problems on phonecell
2:36AM 2 calling with out registration
1:29AM 1 Sipura 2000 vs 2100
Wednesday January 5 2005
11:56PM 2 Inbound Calls
11:34PM 6 asterisk addson
10:44PM 0 if ${variable} include xxx ???
10:14PM 2 Glophone/Voiceglo and Asterisk
8:01PM 2 Ouch... Error while writing audio data
6:28PM 2 CVS Compile problem on Solaris
6:09PM 10 modprobe: Can't locate module wctdm
6:06PM 0 Polycom IP500 - problems with multiplesimultaneous calls
5:51PM 7 Polycom IP500 - problems with multiple simultaneous calls
5:14PM 6 "Out the box" solutions?
5:01PM 9 TDM04B vs Dell
4:52PM 1 Read() timeout hangs up the line
4:50PM 4 Streaming Audio - Music On Hold Feature
4:46PM 3 queues - announcements and not busy members
4:42PM 6 Aaargh Gentoo updated some packages now * won't start
4:39PM 14 Realtime
4:08PM 0 sip.conf asterisk to vonage
3:32PM 1 debug channel <n>
2:50PM 0 Twin Cities Asterisk meeting this Saturday?
2:25PM 0 PCI hardware
2:20PM 0 funny little question regarding asterisk as a pbx vs a key system [slightly OT]
2:11PM 2 IP Phone suggestion.
2:03PM 3 X-lIte behind NAT and Asterisk behind NAT
1:50PM 2 chan_oh323 Module for Asterisk
1:12PM 2 Allowing "pooling" or "rollover" for inbound calls on VoicePulse
1:02PM 13 Asterisk with MySQL
12:49PM 1 Forwarding Voicemail Crashes Asterisk
12:12PM 1 ASTCC Compiling Problem
12:05PM 5 Music from Freeplay music included in * ??
11:53AM 3 lcdproc and asterisk
11:09AM 3 Sending DTMF to PSTN problem with SIP
11:05AM 0 (no subject)
10:48AM 4 Broadvoice / * re-register issues
9:54AM 0 Getting Agent Channel information
9:34AM 0 Asterisk consultant wanted - S. California
9:23AM 3 Bootable Asterisk CD ?
9:20AM 0 VoIP Provider Peering
9:01AM 0 does TE405P support 3Bit CAS?
9:00AM 4 Last callers script?
8:40AM 0 One way audio [Asterisk + Innovaphone IP3000 + asterisk-oh323/h323]
8:37AM 10 asterisk - oh323 driver
8:34AM 2 Asterisk Pbx Manager Equivalent (in plain text - apologies to those that dont like HTML mail!!)
8:23AM 0 Re: Speex codec problem (unresolved ?) = Fixed
8:23AM 12 Happy Wednesday Morning SMS question, slightly OT
8:22AM 0 Asterisk Pbx Manager Equivalent
7:56AM 4 ISDN/SS7 book?
7:47AM 0 Problems with msn's, did not find device for msn
7:43AM 0 Asterisk as Nortel option 11 Autoattendant, question
6:52AM 1 TDM400P + Asterisk + zaptel timer ?
6:51AM 2 Versions of * what do they do/where is the change history/docs?
5:41AM 0 Usage Of Additional LEDs For Snom (was; Status of SNOM Intercom)
4:41AM 1 New asterisk installation but no audible voicemail prompts?
4:33AM 1 Cannot Hear at all
1:54AM 1 Speex codec problem (unresolved ?)
1:23AM 0 Asterisk with Euro ISDN, etc
1:01AM 24 Digium T100P T1 Card
12:57AM 0 Some bugs in DIAX 0.9.9f are now solved
12:32AM 4 Can't initiate a call with X-Lite.
12:10AM 1 IAX phones
Tuesday January 4 2005
11:55PM 1 Call(out) routing
11:51PM 0 Group= equivalent for sip channels?
11:00PM 0 Question on behalf of a wannabe new list member
9:40PM 2 =?ISO-8859-1?Q?Re: Polycom Buddy Feature?=
9:14PM 1 Login Incorrect Message
8:04PM 3 Where to start. {Scanned}
7:09PM 1 How can I silently use ASTCC?
6:57PM 2 Which numbers should be blocked?
4:02PM 1 IAXy Static... and other issues
3:43PM 0 Manager API - ExtensionState help please.
3:38PM 2 integrating with panasonic td-1232
3:26PM 17 CallerID in Australia & Analogue PSTN Phone System
2:58PM 6 AVM C2 capi.conf ?
2:42PM 1 Sprint Vision Phones ReadyLink=SIP?
2:33PM 3 Do Not Disturb
2:14PM 0 Ericsson 4422/4425 phones
1:53PM 30 Polycom Buddy Feature
1:13PM 1 Asterisk in a mixed phone environment
1:00PM 5 Asterisk stops - why ?
12:33PM 0 the correct way to stop a CDR?
11:30AM 6 OT: List of VoIP providers?
11:27AM 5 modprobe ztdummy hangs
11:27AM 2 Vonage WiFI Phone...
10:58AM 0 Does congestion exit on a different priority?
10:19AM 7 queue_log
10:11AM 1 DID and Callback - Questions!!!
10:04AM 4 Newb howto request: *, Voice Pulse Connect, & SJPhone
9:57AM 5 IAX service - user experiences
9:48AM 1 ChanSpy - Should I repatch it ?
9:11AM 1 HDLC Bad FCS (8) HDLC Abort on TE410P
9:08AM 2 dialplan question - how to dial an * extension to get an outbound dialtone?
8:44AM 0 sip.conf [externip]
8:38AM 6 Kirk SIP-DECT gateway
8:16AM 0 Asterisk CLI : Scrollback with Putty and Screen
7:55AM 7 Avaya IPO412
7:08AM 0 DIAX 0.9.9f website updated
6:57AM 0 cid_num with Asterisk CVS 1.0.12
6:53AM 4 Don't receive the prefix
6:20AM 4 Displaying incoming e.164 callers number - how?
6:15AM 0 Cisco 7200 One-Way Audio
5:56AM 1 configuring sample time period for codecs?
4:58AM 0 "Hey look ma, it's not an RPM..."
4:47AM 0 Dell Poweredge 6300 & 4 analogue lines
4:22AM 0 Asterisk and rtp:// streams
4:18AM 8 Status of SNOM Intercom
3:09AM 0 OT: Asterisk at CeBIT 2005?
2:47AM 0 [OT] Anyone used Metrowerks PCS to build / distribute Asterisk
2:28AM 0 Making an ISDN call via Asterisk?
1:31AM 0 Re: 8 pstn lines+ on Asterisk supported
1:05AM 0 Re: Re: 8 pstn lines+ on Asterisk supported
12:41AM 0 RE: Asterisk-Users Digest, Vol 6, Issue 29
Monday January 3 2005
11:56PM 0 SPA-3000 as FXO Gateway for * (Was: Qs aboutFXO/FXS cards)
11:32PM 0 Echo problem - (sorry if this is an nmf question)
10:58PM 0 Verisign SIP7 sip<-->ss7 service
10:22PM 0 queue_log wrong?
9:51PM 3 IAX2 (IAXy) and DTMF Question
9:12PM 0 Re: Asterisk won't register with
9:12PM 3 Ignoring a ringing connection
8:49PM 6 Xorcom Rapid CD for Production?
7:40PM 4 Anyone ever get the Polycom Microbrowser XML document?
7:25PM 0 X100P - check channel busy?
7:25PM 1 call transfer to conference call
6:38PM 7 Manager API
5:51PM 3 zaptel error while initiating
4:04PM 0 reliable capacity for a single * box
2:45PM 5 sendURL
2:41PM 2 SIP Jitter buffer(control?)
2:32PM 3 UPS - a little OT
2:18PM 0 queuing questions
1:58PM 2 echo test application delay using the asterisk cli
1:23PM 7 Does Digium work on Mondays?
1:09PM 3 Line-in as MOH source
1:04PM 4 realtime audio for asterisk using jack
12:53PM 9 QOS / Cisco / Asterisk
12:17PM 2 PSTN to VoIP
12:08PM 4 oh323 context for peers
11:45AM 0 followup on FXO Call progress question
11:40AM 0 Cisco As5xxxx audio issues
11:34AM 64 TE410P card in an HP-Compaq DL380 G4 server
10:43AM 7 8 pstn lines+ on Asterisk supported hardware.
10:33AM 0 disable ringback of held call on zap channel
10:33AM 0 new country tone/zone info setup in ZAPTEL and ZAPATA FXO config ?
10:16AM 1 Subject: Re: Dial with no phone line connected
9:13AM 5 DHCP Attribute for TFTP server for Aastra 480i?
8:55AM 0 2 E100P card
8:29AM 0 Dialplan, LCR
8:11AM 0 Checking to see if a dialplan variable is NULL, mysql app addon
7:53AM 8 agent with queues remain unavailable during transferred call
7:50AM 4 Speex codec for 8Kbps setting ?
7:50AM 2 PSTN to VoIP FXO gateways?
7:42AM 0 SPA-3000 as FXO Gateway for * (Was: Qs about FXO/FXS cards)
7:41AM 0 PRI Errors: Here is where I am at...
7:37AM 0 ENUM: which number?
7:24AM 10 Asterisk CPU priorities (nice?)
6:41AM 0 ###SORRY###
6:33AM 1 Registration server changed or down?
5:34AM 6 finding current codec?
5:18AM 13 Just saw your [Asterisk] xJack Segfault in Asterisk
4:49AM 6 SipSak: error: this FQDN or IP is not valid: voicegw
2:37AM 0 How to compile zaprtc on CoLinux Debian Package
2:15AM 8 TE410P - Normal activity ?
1:42AM 0 Limit max calls & call duration
Sunday January 2 2005
11:51PM 1 Configuration details for Asterisk interaction with Vocal
11:21PM 1 extensions.conf sorting
8:39PM 0 ????
7:30PM 5 Call Queue Question
6:56PM 0 Terminal Adaptor down after register
5:49PM 0 Using Asterisk as a TA?
3:59PM 6 Indications UK - cant get away from american sounding dial tone
3:45PM 1 pridialplan=unknown ?
3:14PM 1 Can I receive faxes with Fritz card & Asterisk ?
2:35PM 24 phones with two ethernet ports
2:24PM 1 Subject: Re: Dial with no phone line connected
1:55PM 3 Codec Selection in Asterisk
1:31PM 0 Box unstable after loading zaptel drivers for X100P
1:16PM 2 Clipping on outbound calls via SIP/IAX
12:17PM 1 ArtDio IPF-2000 or Sipura SPA-841
10:21AM 11 Booting * from CF
7:52AM 4 Dialling 9 for an outside line
Saturday January 1 2005
11:50PM 3 Announcements via IAX phones
11:48PM 0 outgoing call (Sip phones to PSTN)
7:45PM 1 ASTCC gsm files
5:58PM 0 Audio breakup problems
5:49PM 0 Asterisk@home ISO install of ISDN card with HFC ?
4:52PM 2 Problems to use asterisk with mysql /odbc
1:52PM 79 Qs about FXO/FXS cards
1:50PM 6 sip reload - Hang
11:05AM 6 spandsp app_rxfax - the sending software loops
10:05AM 0 Asterisk dies every hour
1:28AM 0 Help with AGI script calleridnamelookup.agi