asterisk users - Feb 2005

Monday February 28 2005
TimeRepliesSubject
11:05PM 0 snom220 *8 hangup
10:29PM 0 how to increase max number of simulatneousoutgoing calls
10:10PM 3 SNOM Call Diversion
9:17PM 0 chan_capi compile error on FC3
8:23PM 0 strange CDR problem
8:06PM 0 what is phone: Linux Telephony channel
6:50PM 2 AMP with FC3
6:37PM 0 about dial parameter L
6:08PM 0 New SMS gateway command
5:34PM 2 Advanced Conferencing options with out-of-treemodules?
4:05PM 0 What about Asterisk and handling switchtype qsig (zaptel) ?
3:58PM 9 Zap channel calling back after hangup (due to polarity CID detection)
2:24PM 2 Asterisk-OH323 no ringing
2:23PM 8 Advanced FollowMe or Forwarding Application Suggestions
2:19PM 0 Newbie---Inquiring.
2:18PM 2 No such host when trying to register
1:58PM 12 Strange text on Asterisk console
1:47PM 17 Grandstream and VLANs
1:35PM 8 I can't load modules (ztdummy, wcfxo.o)
1:22PM 7 Cannot compile (app.c)
12:40PM 7 Recommendation for dialplan in case of DDoS atta cks?
12:17PM 0 Advanced Conferencing options with out-of-tree modules?
11:06AM 0 Ring state patch
10:59AM 0 How to charge incoming calls with ASTCC ?
10:57AM 1 Asterisk network architecture
10:54AM 0 how to increase max number of simulatneous outgoing calls
10:47AM 1 Manager "Message: Originate failed" beinggenerated when callee does not pick up
10:35AM 0 RE: Asterisk-Users Digest, Vol 7, Issue 323
10:33AM 0 Manager "Message: Originate failed" being generated when callee does not pick up
10:18AM 0 How to limit a peer to one connection only?
10:12AM 1 x101p + Nortel ATA2
10:06AM 1 Suse 9.2 + CAPI Driver
9:58AM 0 Passing additional information to an AGI via a call file
9:46AM 2 Weird behaviour on incoming DIDs
9:44AM 7 FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format
9:41AM 0 queue_log and exitwithkey
8:53AM 1 Anybody using X-Lite Softphone ? tryedtoforwarda call to X-Lite....
8:37AM 0 Anybody using X-Lite Softphone ? tryed toforwarda call to X-Lite....
8:22AM 10 Sipura SPA-841 autodial?
8:16AM 1 Unable to handle ROSE operation 34
7:56AM 4 phpconfig
7:34AM 2 Fax Failing
7:10AM 0 New Instalation
7:07AM 0 Secure IAX Interasterisk authentication ?
6:40AM 2 dialing application - newbie question
6:37AM 1 SIP video problems
6:22AM 1 Problem with call hold
5:49AM 0 SIP broadband phone addon for asterisk
5:04AM 0 ASTERISKBRASIL.ORG
3:55AM 0 Pb DTMF with Asterisk vs Cirpack Transit, Node
3:31AM 0 X100P with Analogue DDI Trunks
3:13AM 0 calling sdp
2:38AM 3 Two offices connection
2:35AM 1 call from two sip phones registered in different asterisk server
1:37AM 4 Digium E1/T1 card with mgetty+sendfax
1:20AM 0 Bad soundquality on inbound calls.
12:58AM 3 Digium Card Problems
12:43AM 0 Pb DTMF with Asterisk vs Cirpack Transit Node
12:11AM 1 setting up fromuser
 
Sunday February 27 2005
TimeRepliesSubject
10:18PM 1 context of transfer
9:51PM 3 [Asterisk-Dev] Asterisk 1.0.6
9:18PM 2 No Agents Catch
7:46PM 4 music on hold trouble
6:53PM 19 where is voice conduits
4:53PM 2 CDR's are not stored in mysql
4:51PM 0 Barter studio time for asterisk lessons Brooklyn NY
4:49PM 2 IAX2 (Stupid question)
4:43PM 2 dialout with PPP on ISDN to an ISP
4:19PM 0 test - no msg
4:00PM 1 Beronet BN4S0 (quad BRI) card, echo cancel, zaptel timing, bristuff ...
3:54PM 3 Possibility of getting someone to delete a user from the list???
3:51PM 0 FW: DISA and a long delay; ideas?
3:48PM 0 IAX2 web client that works with g723 / g729
3:23PM 0 email enviado sextafeira. sobre a lista IMPORTANTE
3:12PM 5 Problem selecting E1 on TE405P
1:29PM 0 Interface * with ATA from ATA FXS port? (Here I go again)
12:39PM 1 limit SIP extention outgoing calls
12:34PM 1 Which codecs are used?
11:58AM 2 Jumb between macro's and variables
11:57AM 4 Grandest Free Softphone
11:24AM 1 Suggestions for what to do with a Dialogic D/41EPCI?
11:00AM 6 Outbound call on TDM400P
10:30AM 0 not connecting with X-Lite
10:10AM 6 DISA and a long delay; ideas?
8:50AM 5 Introducing the Asterisk Realtime Architecture - ARA
6:47AM 2 Weird Delay (> 30 sec)
6:38AM 1 DIALSTATUS with X100P
6:08AM 0 ATA 286 downgrade failure
4:10AM 0 g723 issue+asterisk impropoer shutdown
1:27AM 2 opencall.org is changing to soft-switch.org
1:23AM 1 astguiclient gives me Object not found
1:17AM 0 Astcc installation
12:43AM 0 Transfer not working
 
Saturday February 26 2005
TimeRepliesSubject
10:19PM 7 Wierd asterisk-perl compilation problem
9:38PM 16 Limit the call & recording when pressing *1
8:56PM 0 snom 190 funtion buttons
6:22PM 3 call pickup with Sipura-3000
3:41PM 0 'asterisk' displays on 2nd line (CID Number Line) on Cisco 79x0 phones
1:23PM 10 Dial out through Broadvoice
11:55AM 0 SIP phone speaker phone mic cutting out
11:40AM 2 BRIstuff - synchronization with PSTN?
11:28AM 0 How to grab CallerId information
10:24AM 0 Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite....
9:47AM 0 Re: FRS over *
8:27AM 0 NAT= setting for a public proxy
8:23AM 3 Error Message
7:58AM 3 ERROR: compile asterisk(from CVS HEAD) and got an error
7:27AM 0 ERROR: compile asterisk (from CVS HEAD)
6:37AM 2 Interface * with ATA from ATA FXS port?
5:30AM 1 Which is best : Chan_capi or chan_misdn ???
4:54AM 0 Wildcard failing to load on asterisk@home
4:34AM 1 ERROR: when compile app_addon_sql_mysql.c of asterisk_addon
3:32AM 1 Determine IP addres of a AIP/IAX user
3:01AM 3 listening to gsm files
1:59AM 1 Queue Auto fallthrough
1:54AM 0 Polycom SP300 problem solved
12:47AM 3 Language Problems
12:45AM 0 Polycpm SP300 problems
12:06AM 6 FRS & *: an actual business use
 
Friday February 25 2005
TimeRepliesSubject
11:50PM 1 playing "i" invalid context to an internal user
11:08PM 7 Seting up for afirst time -- can not call
8:45PM 1 open 723
8:14PM 0 Asterisk with regular analog phones
4:29PM 9 weather asterisk@home
4:06PM 1 Re: FRS radios on *
3:59PM 0 SER vs. Asterisk - call in progress to PSTN
3:37PM 2 VM+Realtime config
3:33PM 0 CallerID Name and Digium TE405P
3:13PM 1 Asterisk in front of Toshiba CTX
1:47PM 3 SetCIDNum using SIP?
1:19PM 1 Re: Asterisk-Users Digest, Vol 7, Issue 304
1:13PM 3 Festival - Asterisk@home
1:09PM 1 Transposed ringing
12:41PM 0 Video Support Not Working
12:33PM 0 Wheres the Math application
12:07PM 7 HELP NEEDED ASTERISK AND MEDIATRIX 1102
10:57AM 2 Fax on Asterisk
9:54AM 0 Speex transcoding for Cisco / Polycom
9:44AM 2 Avaya Partner ACS3 and Asterisk
9:19AM 2 WebVMail Woirks but No Audio
9:00AM 2 Directory config...
8:59AM 15 How does the g.729 registration program work?
8:23AM 0 Vonage <---> Asterisk Complete Config
8:05AM 1 Working SIP phone for linux and windows
8:00AM 5 407 Proxy Authentication Required
7:54AM 0 call waiting notification and cisco 7960 phone
7:49AM 0 Anyone had a Cisco 7970 working with
7:25AM 1 SIP Errors
7:04AM 0 Asterisk with PortaOne Radius client- problem in accounting script with OH323
6:53AM 27 T.38 fax summary
6:41AM 1 r2 signalling in east europe
6:32AM 29 FW: Getting PHP Config to work?
6:02AM 6 Fedora Core 3?
5:18AM 1 msic while ringing
5:00AM 6 CDR writing incorrect data to pgsql tables
4:10AM 3 "click to dial extension number" functionality ?
3:42AM 1 Asterisk and 723,729
3:06AM 0 international calls and NOANSWER
2:37AM 0 Which version of ast_data for Asterisk v1.0.5?
2:36AM 0 about caller sdp
2:32AM 4 cascaded ringing
2:02AM 0 WG: AW: Transfer a call ? Am I looking for theflashcommand ?
1:34AM 0 help me : about dial to PSTN
 
Thursday February 24 2005
TimeRepliesSubject
11:04PM 0 Re: Radio over *
10:10PM 6 VoIP/Asterisk presentation
10:04PM 1 IPCB
9:21PM 1 RESELER ON INDONESIA
9:12PM 16 Asterisk With Broadvoice
9:00PM 0 Question of SER to Asterisk to PSTN
8:26PM 4 softphone has problem to call out via X100P card
8:12PM 5 Re: FRS and GMRS via *
7:59PM 0 Hope cooperate
7:16PM 17 asterisk supports VXML?
6:44PM 0 Connect to siemens pbx with misdn NT mode
5:39PM 1 Which Codec(s) to use..?
4:50PM 12 Transfer a call ? Am I looking for the flash command ?
4:49PM 6 Delay after entering digits with IVR
4:42PM 0 hint and contexts
3:56PM 5 What is an E400P-SS7??
3:10PM 1 Call Xfer and other features..
3:04PM 3 IAXY DNS possibilities??
2:57PM 2 No audio when h323 calls are incoming
2:49PM 3 [Asterisk-Dev] How to monitor Agen Voice channal?
2:03PM 0 Weird Issu: Figuired it out
1:36PM 2 Weird Issue: Call will not go into VM
1:28PM 2 Making two * servers share same dial plan?
1:25PM 0 Get SPA-2000 to dial out on one * and get calls in from a different *?
12:56PM 2 choppy and cracking sound from zyxel prestige 2002
11:55AM 9 CallerID problem
11:15AM 0 Caller in meetme room quiet (low level?)
11:10AM 1 How does Asterisk choose the CDr backend to use?
11:08AM 0 is this stuff for me? need some help
11:01AM 2 Can you set up a phone via MAC address?
10:59AM 0 transfer ringback
10:54AM 2 Park Call timeout
10:51AM 3 Asterisk and #
9:59AM 4 SIP Phone with headset
9:51AM 1 Zap Channels Disappear???
9:41AM 5 OT - C structure question
9:38AM 2 do i have to reload asterisk every thing i add a new extension
9:26AM 1 Servidor SIP
8:54AM 10 Polycom Call Parking
8:40AM 0 Queue Questions
8:31AM 6 Inheriting variables
8:24AM 1 Queue Announcement
7:51AM 1 Bug in SUBSCRIBE handling : running out of RTP ports
7:40AM 0 Is using Sipura 2100 as SOHO main router good solution?
7:23AM 0 asterisk & proxies...
7:19AM 1 VideoMail & Asterisk
6:53AM 0 SV: SV: SV: QSIG, Asterisk and Eicon DIVA
6:32AM 2 Aastra 480i and Telnet - anyone know how to log in?
6:29AM 2 Asterisk and Welltech USB SIP phone K1000A
6:12AM 4 High capacity voicemail
5:55AM 1 Re: Asterisk-Users Digest, Vol 7, Issue 296
5:47AM 3 Ericsson MD-110 and Dig-410
5:31AM 0 MGCP transfer and CDR
5:30AM 0 Any $CALLER
5:16AM 0 FW: SIP echo on LAN
5:03AM 0 a silly question regarding call monitoring!
4:55AM 3 Brainstorm: Running Asterisk as cool as poss ible - AKA solid state.
4:48AM 1 Problems with SIP codec selection
3:36AM 0 Introduce bridged calls with a beep ...
2:27AM 0 Analogue Extension Hold Sequence
2:13AM 0 Strange problem with h323
2:10AM 6 SV: SV: QSIG, Asterisk and Eicon DIVA
1:57AM 1 Call recording stopped when call transferred
1:25AM 1 SV: QSIG, Asterisk and Eicon DIVA
1:20AM 10 CallTransfer
12:50AM 1 Azatel Azacall 200 issue with asterisk
12:41AM 1 Meetme with video & audio phone mixed
 
Wednesday February 23 2005
TimeRepliesSubject
9:18PM 1 Brainstorm: Running Asterisk as cool as possible - AKA solid state.
9:04PM 1 Cannot compile latest version from CVS
8:03PM 2 Trouble installing TE405P with asterisk@home
7:00PM 3 multiple sip phones behind firewall
6:46PM 1 Asterisk as a voicemail for a central office switch
5:56PM 0 Serious Audio Problem.
4:01PM 0 confirm use of INSTALL_PREFIX
3:54PM 1 AST -> Channel Bank Hangup Problem
3:33PM 0 ZAPHFC is back in bristuff 0.2.0-RC7d+
3:14PM 0 Asterisk wont accept tone from Norstar 3X8 ATA port ?
3:11PM 2 Sound files quality and volume
2:50PM 5 Vonage <---> Asterisk Working Config!
2:43PM 4 SIP NOTIFY in stable branch?
2:41PM 24 FRS / FRS/GMRS 2-way radios as SIP clients
2:29PM 0 sharing parking lots
1:12PM 0 Cant connect to sjphone
1:02PM 2 Dialogic cards
12:44PM 3 Request for PRI Dump
12:44PM 1 Sipura 2000 w/fax machine oddities
12:32PM 1 AreskiCC - pass card number?
12:30PM 1 Re: Some simple voicemail questions...
12:22PM 0 Uniden, Polycom or SwissVoice???
11:58AM 1 Best practices direction
11:48AM 0 Success stories - Asterisk + Video support
11:33AM 0 do i have to reload asterisk every thing i add a neww extension
10:50AM 0 Problem with dialing out and chan_capi
10:27AM 10 Able to tell if phone is registered?
10:07AM 0 Streaming Phone Calls
9:39AM 0 avaya 4602
9:38AM 15 Problem connecting a TE410P to an E1/IP equipment
9:28AM 3 Send outgoing calls to a SIP gateway
9:20AM 1 List tips for new subscribers <--sorry for 2nd post, missed this.
9:18AM 0 Newbie Help - Auto Fallthrough
8:56AM 0 Question about DTMF
8:54AM 15 List tips for new subscribers <--sorry for 2 nd post, missed this.
8:43AM 3 7960 Not Picking up new firmware.
8:36AM 0 Digium TE405P and Cirpack Switch
8:28AM 0 logger reload/restart hanging
8:27AM 2 Creating extension groups
8:20AM 10 List tips for new subscribers
7:31AM 4 Using as FAX 100% IP
7:03AM 6 storing cdr in two databases
6:22AM 8 Zaptel (Junghanns 4BRI card) to cell phone problem
5:56AM 0 Re: Asterisk-Users Digest, Vol 7, Issue 284
5:27AM 0 Subject: Welltech with Asterisk Registration
5:21AM 1 Error connecting to remote mysql database.
5:15AM 9 Difference between E1 and PRI
5:03AM 6 Digium BRI or quad BRI
4:57AM 25 Help With Adit 600 Configuration
4:31AM 1 Chanspy and current version of cvs
4:01AM 1 mixing sound files?
3:54AM 0 cdr_odbc logging insane integer values
3:37AM 2 Anyone had a Cisco 7970 working with Asterisk?
2:37AM 0 Teleconferencing using Zapta cards.
2:35AM 8 Zaptel Red Alarm
2:15AM 11 IVR stats
1:36AM 0 IAX Trunking capacity enforcement
1:00AM 0 hylafax
12:35AM 2 FW: What do I still need?
 
Tuesday February 22 2005
TimeRepliesSubject
9:24PM 2 Voicemail as email attachment not working individually i.e. extensions specific
9:18PM 8 * or X100P dropping analog calls
8:58PM 0 Extension Design in Visio
8:55PM 12 SpanDSP - Still can't send
7:50PM 0 Do ser + asterisk_b2bua work ?
7:44PM 0 asterisk@home 0.6
6:54PM 0 Connecting Broadvox Direct TA to *
5:56PM 0 How do I do this ?
5:06PM 2 Settings for SIP to dial PSTN with TDM400P w/FXO module
4:44PM 3 Call Manager Express Peer
4:21PM 0 register failed with 2nd Sipura-2000
3:23PM 2 install BRIstuff on *@home?
3:22PM 0 bridging <ZOMBIE> ?
2:49PM 1 Anybody using X-Lite Softphone ? tryed to forward a call to X-Lite....
2:38PM 0 Manager API problems
2:33PM 0 Grandstream 486 Sending Faxes issue out TDM400P
2:19PM 0 queue estimated hold time.
2:13PM 0 H.323 problem, calls don't get answered by asterisk
1:32PM 2 Repost: How do I install Skinny support for non sip cisco phones
12:19PM 1 Astersik CVS HEAD + T1 e&m wink + IAX client doesnt detect call answered on Zap channel
11:56AM 0 Asterisk-HEAD more stable than Asterisk-1.0. 5
11:51AM 6 Multiple Parking Lots.
11:49AM 0 Asterisk-HEAD more stable than Asterisk-1.0.5
11:24AM 15 TFTP Server
11:17AM 2 Voicemail call notification of voicemail
10:34AM 2 newbie needs advice
10:29AM 1 how do I dial extensions with oh323?
9:55AM 4 mp3 to gsm?
9:46AM 1 Finding the true src in CDR
9:17AM 0 PSTN tones with ISDN4Linux
8:43AM 1 Noob question on connection
8:31AM 5 QSIG, Asterisk and Eicon DIVA
8:29AM 1 Polycom IP 500 : Displaying digits dialed after connection
8:28AM 3 [PBX]: New message 1 in mailbox 1000
7:49AM 3 Sip billing
7:41AM 0 Monitor and Record : audio quality
7:33AM 4 Zap timing device
6:47AM 0 setting caller id number and usingsip type=peerfor incomming calles.
6:17AM 5 Amphenol cables?
5:54AM 0 send fax with pri
5:41AM 0 [Fwd: Asterisk to Asterisk via IAX2 Help]
5:35AM 4 does asterisk support menus?
5:34AM 0 CDR - is this possible
5:01AM 0 DID, Sending dialled number to PBX
4:56AM 5 Sound of breathing
4:38AM 0 VMS - AGI
4:29AM 0 manager interface, get callerid number??
4:23AM 3 ISDN/SIP videophone gatewaying?
4:08AM 1 what is problem in odbc
3:41AM 8 asterisk -vvvvvvvgrc?
3:25AM 0 SPEEX installation problems
2:15AM 2 asterisk to pbx dialing
2:12AM 2 OH323 and CDR
1:55AM 0 Segfault when using res_config_odbc on x86_64
12:48AM 6 Custom Menu Not Working
12:07AM 1 route outgoing call
 
Monday February 21 2005
TimeRepliesSubject
11:29PM 0 Textron voip gateway
10:26PM 2 Noise during calls
9:34PM 11 sip wifi phone?
8:57PM 0 Asterisk Video Phones <-> Cisco Call manager 4.0
8:34PM 0 app_groupcount
8:06PM 0 SIP registration timeout
7:14PM 4 Canadian DIDs...
6:55PM 0 Multiple multiline sip phones ringing.
6:55PM 0 LiveVoip digit loss
6:49PM 4 Unable to call FWD user via IAX servers
6:18PM 1 NAT-helping outbound proxy
6:01PM 0 Re: list SNR
5:54PM 1 zaprtc on Debian Sarge 2.4.27
5:35PM 11 Minimal hardware requirements
4:02PM 3 Suggestion for noise reduction on Asterisk-U sers
3:54PM 2 Problem with Avaya 4602 / SIP response 481
3:53PM 0 Brian Elton / Avaya 4602
3:50PM 1 some questions about busy detection
3:47PM 0 FWD problem
3:35PM 8 IAX ATA's
3:32PM 0 Call terminaison Tools
3:13PM 7 Suggestion for noise reduction on Asterisk-Users
2:50PM 0 FWD using IAX2
2:12PM 1 ZAP FXS vs ethernet FXS
2:04PM 4 Asterisk@home Linux has no KDE
1:45PM 0 Hitachi Wireless SIP handset
1:27PM 0 Problems with the FXS module in a TDMxxx card (no sound when receiving a call
1:05PM 0 VoIP Test Phone
12:56PM 2 Dns problems with digium and asterisk.org?
11:46AM 2 voice recognition xml
11:27AM 0 South Korea DID wanted
11:16AM 1 why can't I make toll free calls via IAXTEL
11:04AM 3 Why can't I make inter IAX calls between 2 Asterisk servers
11:00AM 1 IAX channel unable to create
10:54AM 0 Call Announce
10:32AM 0 Terminating problem
10:32AM 1 setting caller id number and using sip type=peerfor incomming calles.
10:05AM 3 Zap call bridge drops randomly
9:34AM 0 Asterisk to Asterisk via IAX2 Help
9:22AM 1 X-IMail-SPAM-Connection DNS Sudo ANI vs True ANI
8:33AM 2 Adit 600 MGCP configuration
8:25AM 3 Anyone using SuperMicro SuperServer 6014P-8R?
8:25AM 5 Polycom Phone Calling Party ID
8:08AM 1 setting caller id number and using sip type=peer for incomming calles.
7:26AM 2 Illegal instruction on startup
7:06AM 0 [SOLVED] Problem with ISDN Dialin via CAPI
6:00AM 6 * Call Monitoring
5:55AM 0 Any luck with attended transfer and ATA186?
5:52AM 2 Monitoring calls through a transfer
5:52AM 0 bug? Unterminated comment detected beginning on line 0
5:36AM 5 compiling cvs-head today?
4:43AM 0 LineJACK dial problem
4:10AM 0 How to ECT (explicit call transfer) ?
3:47AM 10 Conecting to asterisk server through NAT usingIAX
3:36AM 1 Problem with ISDN Dialin via CAPI
3:07AM 0 CallingCard application AreskiCC RELEASE v1.1
2:53AM 0 Disable musiconhold
2:30AM 0 ZAP libpri issue crashes PRI?
2:12AM 1 Problems with the FXS module in a TDMxxx card (no sound when receiving a call)
1:42AM 6 MOH clicks
12:47AM 3 SIP echo on LAN
12:40AM 0 Thank You Note
12:37AM 1 Conference between 2 lines
 
Sunday February 20 2005
TimeRepliesSubject
11:48PM 0 Fwd: res_config_mysql & chan_iax2 socket_read error
11:13PM 1 Re: Ring/Off-hook in strange state 6 on channel...
11:09PM 1 Sangoma A101
10:26PM 0 SIP to SIP calls have no audio until put on hold and taken back off - SOLVED
10:19PM 21 HELP NEEDED! - Asterisk GUI
10:14PM 4 Asterisk H323 support
9:53PM 2 How many line appearance can Snom 200 handle?
9:50PM 1 How to announce the DNID to the called party
8:52PM 5 * > Mobile Phone > Mobile Network
8:41PM 1 Where to contrib the sound files ?
8:17PM 4 Modem as PSTN interface?
8:00PM 2 Adding zap channels under *@Home
7:53PM 2 PLease help: Asterisk to Quintum interconnection
7:25PM 1 Phones for vitural office business
5:42PM 12 help with @home
5:13PM 0 Traditional Ringback Tone
5:12PM 3 NAT and FWD
4:07PM 1 HFC-S ISDN card on *@home
3:31PM 0 Sparc hardware, Linux and X100P REVISITED
2:58PM 3 Conecting to asterisk server through NAT using IAX
1:23PM 1 What happens if quadbri or octobri loses power - do they have power failure feature ?
1:02PM 0 Recording of calls stopped - normal behaviour?
1:01PM 0 Re: Asterisk-Users Digest, Vol 7, Issue 260
11:51AM 3 Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?
11:06AM 3 Voice Prompts with no sound
9:41AM 1 Adtran Total Access MGCP Config?
9:11AM 0 SIP to SIP calls have no audio until put on hold and taken back off
8:43AM 0 SIP peer registration interval - SOLUTION
8:14AM 0 CDR for callback
5:00AM 20 Mandrake & CAPI
4:00AM 2 External relay triggered by Asterisk extension-question
3:02AM 1 making ASTCC web page secure ???
2:43AM 12 bridging iaxtel calls to PSTN
1:20AM 14 Simulated dialtone like in other PBX
12:10AM 4 Soundcard problems?
 
Saturday February 19 2005
TimeRepliesSubject
10:33PM 1 External relay triggered by Asterisk extension - question
9:29PM 2 asterisk setup
9:22PM 0 ROUTING INCOMING CALL BASED ON CADENCE?
4:35PM 0 Can't Dial-out
3:38PM 2 Anyone used the ACT P104SLD SIP Phone
3:02PM 0 X-IMail-SPAM-Phrase X-IMail-SPAM-Connection DNS Problem with T1 and international calls
2:58PM 1 video conferencing
2:16PM 2 sending traffic to LiveVoip
2:02PM 3 Terminating IAX calls in E1 PRI interface - what do I need to be able to send arbitrary caller id to called party ?
1:49PM 0 HFC/zaphfc/zaptel: issues with multiple inbound calls
1:42PM 2 I have a odd question...
12:53PM 17 No Sounds
10:43AM 1 Asterisk with Multitech H323 Gateway MVP400
9:59AM 4 I need to dial multiple numbers concurently but with delays.
9:06AM 1 Can I exchange datas between two Asterisk servers ?
9:00AM 1 Uniden UIP200, please help
8:56AM 6 simpletelecom.com??? are they a SCAM?
8:49AM 6 Still asterisk startup crash plz help
8:12AM 5 Hi Newbie question
5:58AM 2 This is NUTS!!SOLVED
1:19AM 2 Sip question - allow only 1 incoming call to sip phone
12:30AM 0 Understanding and Troubleshooting Analog E&M
12:24AM 0 TOUCH_MONITOR
12:14AM 30 Snom phone hint exten question
 
Friday February 18 2005
TimeRepliesSubject
10:54PM 0 Asterisk to Quintum gateway interconnection
7:16PM 0 ACD softphones?
6:22PM 3 VoIP Test Samples to test Asterisk
5:04PM 1 Asterisk@home festival weather report
4:30PM 0 Time to beg on my knees for help!!!
4:06PM 0 Using * to connect to database and to modify said database
4:01PM 0 FXS signalling for Ireland
3:47PM 1 wikki problem
3:18PM 0 Installing Asterisk on Mandrake 10.1 Official
3:11PM 7 Which PRI card for EuroISDN ?
3:10PM 0 Looking for Asterisk setup and maintainance (terminating calls to EuroISDN PRI interface) in Frankfurt Germany
2:45PM 1 GotoIfTime Discrete weekdays (Mon,Wed,Fri)
2:39PM 0 Feb 18 15:22:33 WARNING[1862]: chan_zap.c:3428 zt_handle_event: Ring/Off-hook in strange state 6 on channel 2
2:33PM 0 *** Important *** About the bug tracker
2:21PM 0 Monitoring stops when call is transferred
2:08PM 1 Power failure + which card must i choose
2:06PM 6 Sending DTMF after a call is set up
11:55AM 2 VoIP Service Provider
11:49AM 0 Grandstreams ATA286
11:48AM 2 Difference between a TE410P and TE405P?
11:33AM 0 More asymmetrical call quality discussion
11:11AM 2 defining the zap channel used on inbound analogue calls
11:06AM 2 Send CallerID to PBX via PRI NI2
10:35AM 5 MSG WAITING OFF on cordless handset not going away
10:27AM 1 Safecom SIP-300 Information?
10:04AM 3 need info
9:56AM 10 HDLC Bad FCS / HDLC Abort
9:54AM 4 Help asterisk startup errors
9:26AM 1 Calls directed via queue to unavailable device result in call acceptance
9:04AM 17 W&M Wink timings for Nortel
8:53AM 8 Asterisk GUI
8:44AM 0 VAD (Silence suppresion problem)
8:41AM 0 Process incoming faxes in Asterisk
8:25AM 6 VONAGE <----> ASTERISK SIP TERMINATION?????
8:21AM 31 A bit of a survey: What do do if you need more than 4 C.O. lines
8:21AM 2 This is NUTS!!
8:20AM 0 Asterisk 1.0.5 an MySQL CDR
8:16AM 0 More on W6692pci NT mode under chan_misdn
8:13AM 0 Asterisk on Solaris 10
8:09AM 1 Help with config.
8:04AM 9 Budgetone 101
8:02AM 0 Monitoring a telco line for MWI through a TDM400 FXO
6:52AM 0 TDM 2 FXO + Traditional PABX
6:43AM 1 wrapuptime + agents.conf
6:29AM 2 Asterisk + RedHat9 - Libpri problem
6:26AM 3 Wiring question for Digium card
6:18AM 5 Q.SIG support in CVS
6:14AM 1 Disable Loop Detection
6:13AM 1 ISDN channel bank
6:12AM 0 Asterisk with SER
6:08AM 0 mISDN+w6692pci errors while loading
4:46AM 5 quadbri and spandsp
4:46AM 0 Asterisk Can't Run
4:27AM 0 TDM400P and SOHO traditional (analog) telephones
3:54AM 0 Voice Message Matching?
3:47AM 8 MultiLine Sip Phones
3:42AM 1 Is this a bug or by design? Workaround?
3:15AM 1 Asterisk Performance in comparission of SER
2:54AM 4 Vonage, broadvoice et al
2:14AM 2 Timing device OpenBSD
2:00AM 1 Problem with starting music on hold when cal l connects to phone via queue
12:53AM 3 Problems compiling on mandrake
12:48AM 3 any good redhat 9.0 rpm reposiroty?
12:42AM 0 can't see calling number
12:20AM 6 Astricon 2004 tutorials available?
12:14AM 3 SER/Asterisk consultants in Denver
 
Thursday February 17 2005
TimeRepliesSubject
10:04PM 0 SIP "catchall"
9:33PM 1 Zultys Paging Solution / App for Multicast
9:29PM 0 Problem with the IAXy and Netgear Hubs!
8:57PM 0 DTMF Problems with Asterisk
7:01PM 4 Zaptel Needed
5:44PM 0 SIP Seeding peers from Astdb - jam the console
5:15PM 2 Problem with starting music on hold when call connects to phone via queue
5:03PM 1 Voicepulse Open Access & Asterisk Problems
5:02PM 0 MGCP - Unicall
4:34PM 0 Warning messages error
4:22PM 0 List of VoIP provider codes
4:20PM 0 TDM400 FXO not responding to inbound rings a fter 30ish days?
4:10PM 7 Mac Mini and chan_bluetooth, has anyone told The o if it works?
3:54PM 11 functional difference: canreinvite=yes, no, or update
3:54PM 2 Problems compiling pridump utility
3:10PM 1 TDM400 FXO not responding to inbound rings after 30ish days?
2:59PM 0 E1/PRi Hardware echo canceller
2:41PM 4 (Kphone) Registration Failed: Forbidden
2:34PM 1 RE: Asterisk-Users Digest, Vol 7, Issue 239
2:24PM 0 asterisk@home greek letters and suggestions
2:07PM 13 Trying to install X100P
1:16PM 4 Accountcode and SIP Peers Part 2
1:14PM 1 X100P DID
1:04PM 0 Accountcode and SIP Peers
12:47PM 0 Festival and french language
11:23AM 2 arrgghhh dialparties.agi
10:37AM 5 IAXy Provisioning Using Windows
10:29AM 6 Digium TDM 400P and Dell 1750
10:24AM 5 PRI and echocancel
10:22AM 1 Re: Cisco 7970 Won't boot after factory rese t
9:49AM 3 Packet 8
9:29AM 1 UIP-200, registers, 4 seconds pass, then #1 disconnected
8:46AM 2 The 'sipfriends' table is obsolete - ????
8:25AM 4 SIP peer registration interval
8:07AM 4 Sangoma A104 - D-Channel problem
7:21AM 2 Problem with asterisk-addons: libmysqlclient.so.14: cannot open shared object file
7:20AM 1 Cyclades-PC300/TE 1 Compatibility?
7:02AM 6 Strange MSN issue with HFC-s
6:59AM 0 Brand New Digium T100P for sale
6:39AM 5 Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
6:26AM 1 Sirrix ISDN Card
5:05AM 2 VoipJet issues?
4:55AM 0 (no subject)
4:41AM 0 started asterisk with chan_misdn
4:35AM 0 Error loading wcfxs module
4:35AM 4 Voicemail and busy tone
4:08AM 4 can't enable trunking :(
3:29AM 10 Call termination database
2:48AM 1 change the caller id number
1:54AM 3 asterisk functions without voIP
 
Wednesday February 16 2005
TimeRepliesSubject
11:37PM 0 asterisk and gatekeeper
11:27PM 6 problem : undefined symbol.
11:09PM 1 RTP Stream on Multicast
10:45PM 20 Anyone having trouble with VoicePulse Connect?
9:55PM 0 zap a sip channel
9:51PM 3 Zap/g0/ to a Telstra Mobile
9:24PM 7 festival text for weather report
6:25PM 4 Monitoring Conferences
5:11PM 0 Outbound calling timeout
3:22PM 0 Melbourne Asterisk Users meet TONIGHT
3:20PM 4 DIAX 0.9.10d with Eutectics USB phone suport
3:10PM 3 Cisco 7970 Won't boot after factory reset
2:57PM 0 More jitter buffer questions
2:07PM 5 Zaptel DACS and FDL
1:51PM 0 Using zaphfc and wcte11xp at the same time problem
1:40PM 10 IAX2: Connection rejected
1:28PM 0 Verizon BroadBandAccess and *
12:58PM 4 capiECT problem
12:46PM 0 zaphfc buffer underflow/overflow messages
12:41PM 0 Agent Logoff not generating event messages
12:39PM 2 Sip Notify PAP2-NA?
12:33PM 2 Help Please!!!!
12:25PM 0 Polycom MGCP firmware
12:03PM 1 WLAN-Voip phones anyone?
11:45AM 1 IAX Hardphone AT-320EE
11:34AM 1 Inter-asterisk conferencing delays - IAX2 configuration problem?
10:54AM 0 TDM card and Call recognition
10:31AM 5 [patch] fix libpri problem in Q931_INFORMATION handling
10:03AM 0 G729, NAT and Transcoding (all-in-one)
10:02AM 1 Can't connect Snom 190 to Asterix PBX. Sugge stions?
9:17AM 0 How do I match a "D"? (Was: RE: In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
9:06AM 0 When callerid changes its value ?
9:04AM 0 Can't connect Snom 190 to Asterix PBX. Suggestions?
8:57AM 7 Monitor does not like variable subsitutions
8:56AM 5 Dutch VOIP-PSTN provider
8:49AM 12 Why Asterisk can't cope with silence suppression?
8:08AM 0 Attended xfer
8:04AM 9 HELP!!!!!!!!
7:42AM 0 ZAP channel on TE410P doesn't hang up (Plain Text this time)
7:25AM 0 ZAP channel on TE410P doesn't hang up
6:40AM 10 DTMF inband detection improvement
5:40AM 2 chan_sip errors on CVS HEAD
4:40AM 4 Asterisk exist with error
4:27AM 1 Strict Routing vs Loose Routing
2:27AM 1 Passthrough and reInvite
12:49AM 0 Login OK but NO SOUND
12:38AM 0 bristuff-0.2.0-RC7a error messages
 
Tuesday February 15 2005
TimeRepliesSubject
11:27PM 0 Re: card dialer phone (thanks for the info!)
9:11PM 0 How does Asterisk use ALSA?
9:11PM 2 Re: X100P problems
8:43PM 0 X100p + cell socket no callerid
7:10PM 0 VM and MeetMe Stopped working! HELP
6:30PM 6 Dialplan + Registrar DB
6:29PM 6 Help With Broadvoice
6:27PM 4 Queue strategy
6:04PM 1 newbie: help two cisco phones (sip)
5:10PM 3 Dial (Local/.....)
5:05PM 3 Asterisk "no one is available to take your call"
4:55PM 0 Supermicro P4SGA board?
4:46PM 3 iax.cc and/or Sixtel.net seems like IT IS A SCAM.
4:33PM 0 Making ZAP Trunk groups
4:01PM 0 Problem with IAX and codecs
3:37PM 5 iax.cc and/or Sixtel.net ,, IS IT A SCAM???
3:34PM 0 Re: Asterisk-Users Digest, Vol 7, Issue 216
3:29PM 7 A hypothetical question...
3:15PM 2 Solaris 10
2:13PM 0 HFC-S and TE110P at the same time
1:41PM 3 Ser 0.9.0 adding a user?
1:39PM 0 Playtones segfaults?
1:30PM 0 asterisk@home and grandstream display
1:09PM 0 E&M and other Radio-based signalling
1:07PM 2 Sixtel.net / IAX.CC - Vanity Toll-Free Numbe r
12:54PM 3 Teles PCI and chan_capi, possible ???
12:44PM 1 Put call on hold
12:28PM 8 Stop now, well it doesn't :)
12:25PM 3 "i" extension with invalid context
12:22PM 2 Mandrake 9.2 and CAPI
12:05PM 3 Asterisk Integration with ALCATEL 4400
11:59AM 0 Spectralink SVP server - Asterisk
11:44AM 1 More *@Home puzzle
11:40AM 0 CVS Head ADSI Voicemail Busted ?
11:40AM 3 Strange error in debug file
11:32AM 0 perl poe::component::client::asterisk::manager usage
11:07AM 3 Virtual PBX setup.
10:35AM 0 Queue Abandoned and DND
10:18AM 1 IAX2 bugs...
10:10AM 0 Call Recognition. Which TDM card?
10:01AM 0 OT: Comments on Vonage SIP port blocking com plai nts??
9:56AM 25 X-Lite Softphone
9:36AM 1 7912G via SIP, looking for comments
9:29AM 0 Fail to detect DTMF over direct ISDN pri lin k
9:17AM 2 OT: Comments on Vonage SIP port blocking complai nts??
9:01AM 6 Asterisk@Home .5 Setup help with 4 X100P
8:49AM 4 Autostart Asterisk on Slackware?
8:39AM 0 Asterisk Users in Madrid?
8:09AM 3 Asterisk, inband DTMF send by a GSM mobile
8:01AM 0 Mobile operator message
7:54AM 0 oh323 question
7:48AM 11 Extra sounds (Weather)
7:46AM 5 E1 and/or Euro-ISDN specifications?
7:40AM 3 app_rxfax creating bad faxes? (StripOffsets)
7:38AM 0 extension matching in gastman
7:27AM 9 Sixtel.net / IAX.CC - Vanity Toll-Free Number
7:09AM 6 Integration Panasonic PBX
6:01AM 1 "System" command causes core dump Warning: Newbie help :)
5:54AM 3 4xHFC-s cards vs 1 quadbri HFC-4S card ?
5:54AM 0 Asterisk hangs the establised calls
5:41AM 0 [OT] Anyone that knows this ATA?
5:34AM 4 make of asterisk doesn't do anything...
5:27AM 1 asterisk qualified
5:04AM 1 (no subject)
4:44AM 7 solid-state asterisk pbx?
4:17AM 1 Question regarding SER/Asterisk functionality
3:52AM 4 CAPI not installed
3:28AM 0 Problems with SIP Registration at PSTN Provider
3:21AM 1 Asterisk and Call recognition (call id)
3:06AM 0 asterisk@home in production env
2:54AM 0 prblem in compileing asterisk-prepaid
2:45AM 2 Capi channel - can I route call to another channel or back to PBX and free current channel ?
1:52AM 9 Sip phones how to dial a # sign?
1:38AM 5 why does the Polycom IP600 check FTP every 60 seconds...
1:32AM 0 Asterisk restart alone
12:03AM 0 Asterisk@Home 0.5
 
Monday February 14 2005
TimeRepliesSubject
11:42PM 0 No Sound???
11:13PM 41 Which IP phone to use in Australia
10:59PM 1 Native vs Intl calls
7:42PM 5 Clarification on Fax capability?
7:32PM 1 Flash Operator Panel - lots of problems
6:21PM 4 TFTP Serer ????
6:07PM 11 Outbound Caller ID on PRI
5:07PM 2 Can't run AGI for outbound call
4:43PM 0 ASTCC Auth and Dialing problem
4:39PM 0 CVS with attended transfers
3:11PM 2 usb phones in linux, any??
2:16PM 4 ztdummy on Gentoo 2.6.10 Box
1:42PM 0 Asterisk as a Protocol Converter from E1 to T1
12:48PM 0 VXML support
12:30PM 2 Bristuff-0.2.0-RC5 florz patched weird error and no outgoing calls?
12:23PM 2 Asterisk@Home ... the next step
11:50AM 0 H323 no sound
11:31AM 0 cdr_mysql losing logs
11:25AM 15 Sipura g729 call quality to PSTN
10:49AM 1 (no subject)
10:44AM 0 H323 registration
10:26AM 0 Italian speaking. Asterisk configuration and needs
9:45AM 8 Uptime/reliability with SER, Asterisk
9:18AM 2 ztmonitor
9:06AM 0 Asterisk as SIP UAC !!!
8:46AM 0 APP_QUEUE MYSQL LOGGING
8:24AM 22 ATA that actually work with T.38
8:13AM 2 FW: SER Asterisk Voicemail
8:13AM 1 E1-PRI: Warning Message: Unable to handle ROSE operation 36
8:10AM 7 Asterisk-H323
7:55AM 1 Asterisk@home .5 and meetme
6:44AM 0 SIP configurations
6:38AM 2 Asterisk in Singapore.
6:24AM 6 Digium Cards connecting to BT
6:21AM 3 asterisk in New-Zealand
5:31AM 0 Re: Asterisk-Users Digest, Vol 7, Issue 202
5:27AM 0 Error: Unknown RTP codec 72 received???
5:05AM 0 Error: Unknown RTP codec 72 received
4:41AM 1 Sipura 841 and paging function
4:22AM 21 Linphone / Kphone
3:33AM 3 ISDN zaphfc - What kernel are you using successfully?
2:48AM 0 spandsp asterisk 3/5
1:27AM 0 Re: card dialer phone
 
Sunday February 13 2005
TimeRepliesSubject
11:18PM 2 OT: Aastra 390 - weird problem
11:13PM 1 Mysql and SIP real time configuration...
7:52PM 9 Who makes these phones?
7:40PM 4 Q: Does anyone have a WE multi-line card dialer phone working with *?
2:59PM 1 Snom 190's vs Softphone
2:32PM 0 zaphfc NOTICE[6799]: chan_zap.c:7685 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
1:55PM 2 TDM-400P alternatives?
1:54PM 2 TDM-400P Sound Quality issues
12:14PM 2 Broadvoice international dialling question
12:00PM 0 No CallerID on TDM11B?
11:57AM 2 connect asterisk to ISDN in China
11:39AM 1 Dlink VPNs??
10:00AM 1 MusicOnHold Native Mode, Please Clarify
5:01AM 13 Sangoma A102 cards testing
4:14AM 0 Caller IP-Addr from agi ?
3:56AM 4 OT: Open source CRM systems with * integration
3:25AM 0 problems detecting hangup events
12:55AM 1 bad sound ISDN bristuff
12:16AM 2 TDMOE + kernel badness
 
Saturday February 12 2005
TimeRepliesSubject
8:51PM 5 Cannot reset an IAXy box!!!
8:46PM 4 Asterisk+GNOMEMeeting=No Sound.
8:15PM 0 anyone patched CVS Asterisk with ast_data?
7:52PM 0 IAX2-FWD
7:10PM 3 Intermediary jitter buffering
4:30PM 1 Installation of Zatel
4:02PM 0 Asterisk as B2BUA. New application!!!
3:56PM 6 ast_data does not patch
3:55PM 0 Re: Asterisk as b2bua
3:54PM 0 *@home .5 Double Dial Tone
2:50PM 0 uninstall Asterisk?
2:38PM 12 soho fax suggestions?
1:36PM 0 How stable are cheap HFC-s cards in NT mode ?
1:29PM 1 What quad/octo BRI cards are best/stable for EuroISDN and Asterisk ?
12:46PM 2 Mobile Wireless IP Phone
11:39AM 0 Finding exact build version
11:04AM 0 French CallerID
11:01AM 1 ASTCC vs AreskiCC
10:46AM 8 7912G: Takes the same firmware as 7940/60?
10:05AM 9 PLEASE HELP Adit 600 went kaput?
10:01AM 1 return code of app in dialplan
8:36AM 2 Flash Pane - Monitor Parked Calls?
8:31AM 2 Re: Codec Issue on IAX trunk? (Solved)
7:49AM 0 bristuff-0.2.0 RC7 and RC7a
7:05AM 1 MGCP, Asterisk & Cisco VG200
6:52AM 0 Delay on zap channel
6:29AM 0 Outbound calls on a busy Zap/1: BUSY vs. CHANUNAVAIL
5:20AM 5 Initializing two ISDN cards in isdn4linux
4:44AM 0 Asterisk as B2BUA - New Application!!!
3:20AM 0 Possible to use CAPI PBX as interface to analog phone?
2:33AM 9 fax with asterisk
2:02AM 0 Missed Call List on SIP Phones
1:20AM 4 iax.conf config and iax based clients
12:46AM 3 Is there a Caller ID issue in the latest CVSStable
 
Friday February 11 2005
TimeRepliesSubject
10:52PM 2 Asterisk won't answer incoming analog line
10:50PM 0 Q: Can Zap channels be arbitrarily numbered?
7:39PM 1 Is no one using MySQL on stable asterisk?
6:13PM 0 Playing Dialtones
6:06PM 0 Polycom 300 -- "No compatible codecs!"
6:05PM 0 SendText application
5:55PM 16 Asterisk@home .05 release questions on setup.
5:39PM 2 Problem with # Transfer from queue
5:35PM 0 ASTCC one stage dialing problem
5:04PM 1 Re: Codec Issue on IAX trunk? (Solved)
4:42PM 0 Polycom headset tweaking
4:37PM 0 MINNESOTA: TwinCities Asterisk Users Group - Meeting tomorrow Feb 12, 2005
4:22PM 0 Audio issues with greetings & messages
4:22PM 0 /var/run/asterisk.ctl configuration
3:08PM 2 Still stuck trying to make Asterisk read MySQL
2:39PM 0 Quick How-To Guide for getting a Cisco 7960 going.
2:22PM 0 Delay answering inbound calls
2:16PM 3 Can agents login be permanent across Asterisk restarts ?
2:06PM 3 Codec Issue on IAX trunk?
1:55PM 2 Re: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins
1:54PM 3 differentiating busy & not connected
1:35PM 0 Cisco PRI to Asterisk & CallerID
12:47PM 8 Polycom IP 3000 configuration
11:16AM 1 SIP in the Philippines
11:11AM 0 polycom ip phones + asterisk
10:48AM 7 Setting a "Forward" to an external number on your phone
10:35AM 2 *.conf files not parsing
9:36AM 3 Asterisk-MySQL: Not loading voicemail config from MySQL
9:05AM 2 Question about DID
8:28AM 2 RE:mandrake linux install of zaptel
8:07AM 0 Asterisk as a UAC forwarded by SER
7:20AM 7 Weird Echo Problem
7:16AM 5 Menu Selections Only Work Internally
6:55AM 9 chan_capi and asterisk
6:33AM 2 transferring a IAX call into a conference
6:10AM 7 chan_capi or chan_mISDN vs bristuff
6:04AM 0 Not register SIP and IAX
5:56AM 3 Newbie: ISDN E1 the same in all countries?
5:45AM 0 How can agent logout manually ?
5:31AM 0 zaphfc - problems with hangup detection?
5:25AM 0 Transfers to engaged extensions
5:24AM 5 Dial and congestion
4:26AM 0 Help with dial command and h, H and g parameters
3:29AM 0 Multiple incomming contexts
2:54AM 0 Proper handling of incoming IAX/SIP callerids to be able to call back - why is calleridnum stripping dots out of number ?
2:40AM 1 How to monitor externip automatically?
2:09AM 6 i want to load chan_h323.so
 
Thursday February 10 2005
TimeRepliesSubject
10:35PM 3 Bri problem
10:15PM 1 [Asterisk-Dev] Asterisk not accepting multiple SIP phone logins
10:07PM 2 Asterisk not accepting multiple SIP phone logins
9:35PM 4 asterisk as sip client behind nat
8:45PM 0 CISCO CP-7902G and chan_skinny.
8:25PM 0 FW: really easy FOP asterisk@home question
7:26PM 3 Voice Recognition
6:57PM 28 Why echo occurs
5:34PM 2 Fail to detect DTMF over direct ISDN pri link
5:29PM 2 TelIAX troubles
5:00PM 16 Searchable Mailing Lists & NooB Question
4:31PM 3 No dialtone in a E1
4:30PM 1 Proper Contexts in extensions.conf
4:29PM 2 Codec passthrough patch for IAX
3:31PM 0 Context fails so falling back to extension " s" ?
3:28PM 0 asterisk GUI's that supports zap fxs extensi ons
3:05PM 3 Asterisk - SER Configuration
2:52PM 0 Logging agents in and out via manager API or other utility. Is it possible?
2:33PM 1 Dial SIP peers
2:31PM 5 Round Robin Strategy doesn't seem to work
2:12PM 0 Context fails so falling back to extension "s" ?
2:00PM 1 WAS: Strategy for a stable IAXy NOW: IAXy vs old P-3
12:10PM 1 really easy FOP asterisk@home question
11:57AM 1 Problem with SPA-2000 and Asterisk 1.0.5
11:43AM 0 ADM 0.7 Released
11:39AM 14 Wireless LANs and Asterisk
11:30AM 0 asterisk features
11:29AM 0 (no subject)
10:55AM 5 Debian way of compiling zaptel kernel modules
10:46AM 17 dtmfmode and IAX protocol
10:41AM 2 Asterisk on RedHat/AMD
10:29AM 9 Strategy for a stable IAXy
10:25AM 4 General Inbound Calls
10:14AM 1 /dev/dsp blocked
9:45AM 1 SER Asterisk Voicemail
8:47AM 0 asterisk GUI's that supports zap fxs extensions
8:32AM 0 Asterisk 1.0.5 won't pick up incoming calls
7:56AM 0 Using asterisk on a single phone line
7:47AM 4 Need help with a Cisco 7960
7:24AM 0 Tormenta 2 Card number rotary switch
7:12AM 2 Configuring Asterisk
7:10AM 72 asterisk@home scary log
6:44AM 0 A working config for For FX100P Cards in United Kingdom ?
6:09AM 2 Cisco7960/SCCP Transfer Help?
4:49AM 2 Detect hangup
4:07AM 2 Asterisk and Fedora Core 3
3:50AM 0 7940 VM DTMF not detecting
3:11AM 2 Softphone..easy to use ?
2:55AM 2 SIP proxies & Asterisk ?
1:55AM 0 Please share the experience on VoIP phones heavyusing.
1:44AM 4 why asterisk is replying 404 Not Found
1:30AM 0 Manager API - Call Transfer/Blind Transfer
 
Wednesday February 9 2005
TimeRepliesSubject
10:43PM 0 Voicemail timeouts after 30sec's everytime no matter what I set in the configs. CVS Dec 04
10:33PM 3 CallPickup from SIP phone
9:31PM 6 reboot polycom 1.4.1
8:19PM 1 Asterisk consultants directory
8:14PM 5 Please share the experience on VoIP phones heavy using.
6:54PM 2 Melbourne Asterisk Users meet next Thursday
5:28PM 3 Asterisk and Sipura SPA-841 SIP phones
5:05PM 2 sample REGEX's for astcc
3:27PM 1 looking for responsible iax provider, aftermath
3:22PM 1 TDM400P FXO lines problem
3:15PM 0 logging events with time stamps
3:02PM 11 Zombie SIP channels
2:05PM 0 How to map zap channels to ISDN extensions on queues?
2:04PM 2 voice delay after call setup, outgoing calls
1:35PM 1 wcte11xp Trouble
1:20PM 13 Cisco 7960 Beating a Dead Horse
12:59PM 7 Multiple SIP registrations for one account?
12:46PM 3 Startup Question
12:43PM 1 Re: Asterisk Compile Problem on Red Hat 9 solved
12:19PM 0 FireFly + G729 license
12:04PM 0 Why does Asterisk Hangup cause server to freeze?
12:02PM 25 Getting SPEEX to work
12:02PM 1 Asterisk Compile Problem on Red Hat 9 resolved
11:54AM 2 Asterisk Versioning
11:49AM 0 Wait for Digits.. solved
11:44AM 1 Wait for Digits
11:42AM 1 Re: Newbie help/pointers required -configure xlite with asterisk
11:34AM 2 Analogue Line to Asterisk (Which Digium Model???)
11:26AM 2 SIP / IAX ActiveX
10:52AM 0 polycom ip300
10:48AM 0 TDM400P FXO - Any one got it working well in UK without Hangup problems
10:30AM 7 IAX Voice Quality Issues
10:07AM 4 ISDN in Spain
9:54AM 2 How do I match a "D"? (Was: RE: In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
9:49AM 0 Background() ignoring digits A-D (Was: RE: How do I match a "D"?)
9:45AM 0 Using Asterisk as sip user agent with more than one device
9:44AM 1 Asterisk and SER Integration together
9:23AM 2 problem with running ztcfg
9:21AM 0 How do I match a "D"? (Was: RE: In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
8:54AM 16 How do I match a "D"? (Was: RE: In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
8:46AM 2 SIP ActiveX
8:45AM 12 Is there a Caller ID issue in the latest CVS Stable
8:44AM 1 loader.c:301 __load_resource: libpt_linux_x86_r.so.1.8.1: cannot open shared object file... [solution found, but quick question]
8:21AM 0 Asterisk CVS stable (current) crashes on remote user (over CAPI) pressing # or * when in conference
8:11AM 0 VoIP guide for business people
8:10AM 12 polycom soundpoint ip 300
8:03AM 19 Problem with meetMe
8:00AM 9 G.729 codec for X-lite soft phone
7:17AM 9 IAX <=> FWD down again?
6:58AM 1 DTMF Payload Type Compatability
6:01AM 18 Web based Asterisk management tool
5:44AM 1 limit iax calls
5:36AM 1 calling problem in cvs verison on fedora core2
4:57AM 0 Asterisk and SIPphone won't cooperate
2:50AM 0 incoming h323 calls, routed to SIP/H323 drop after connection
2:40AM 3 Problem using TDM400P FXS card
2:21AM 1 Error compiling app_icd
1:35AM 2 Asterisk Compile Problem on Red Hat 9
1:20AM 1 add_pppd dialout problems
1:04AM 1 Asterisk as VoIP gateway
12:35AM 0 incoming call high failure rate on pickup of call.
 
Tuesday February 8 2005
TimeRepliesSubject
11:52PM 6 sip_notify.conf
11:09PM 1 Fastagi question
10:24PM 2 Asterisk connected to pbx
10:24PM 9 bri dropping calls
9:44PM 1 Unable to load module iax.conf
8:56PM 2 SPA-841 MWI
8:33PM 2 giving up on x100p in Australia
8:10PM 0 InterFone IF-102/104?
7:48PM 1 Voip as a secure service?
7:26PM 2 Caller ID Question
5:34PM 1 Callerid to set time on phone?
4:40PM 3 Asterisk causing server to hang ... any hints?
3:57PM 1 breaking friends into users & peers
3:52PM 1 SIP Qualify/Status – What kind of numbers are you getting?
3:30PM 2 astcc with multiple access
3:23PM 0 Codec negotiation problems
3:20PM 0 DIAX version 0.9.10a available for download
1:48PM 1 No dial tone...
1:14PM 1 TDMO4B, GSM Gateways and CallerID
1:04PM 0 SPEEX CODEC and Voicepulse
12:56PM 0 Polycom/sip.conf/voicemail configurator
12:29PM 1 Digium TDM400P 4xFXO
12:16PM 0 Can someone tell me why I'm getting these? ( mailing list probe message)
12:15PM 1 Bug? Background() doesn't recognize D tone.
12:06PM 0 CODEC declarations in IAX.conf
12:06PM 8 Can someone tell me why I'm getting these? (mailing list probe message)
12:05PM 0 Confusing Contexts using AMP
11:53AM 2 Spaces in config files??
11:44AM 3 Looking for FXS device - CISCO ATA 186
11:38AM 0 attended call transfer in 1.0.5
11:32AM 1 Can only call VoIP SIP Providers (Weird)
11:23AM 6 announcement: astfax 1.0
11:10AM 1 Asterisk performance monitoring
11:09AM 2 Polycom screwed up Messages button in 1.4.1?
11:05AM 26 More complicated huntgroups / delayed ringing
10:34AM 4 stable combination of versions for asterisk and chan_oh323?
10:19AM 2 Digium TDM400p troubles
10:11AM 1 how to make g.729 preferred, but failover to gsm
9:10AM 0 Re: Asterisk-Users Digest, Vol 7, Issue 113
9:09AM 4 Music on hold is a durge
8:59AM 1 faxing digium?
8:40AM 21 SRV lookups
8:10AM 1 How do I match a "D"? (Was: RE: In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
8:09AM 1 ASTCC simultenous calls per card
7:55AM 0 codec order, does it matter
7:43AM 17 SER Interaction: Agents and Extensions
7:35AM 1 DASS II cards supported
7:23AM 0 Bristuff - analogue communication over ISDN
7:21AM 1 Linux OS platforms
7:17AM 1 AreskiCC Installation -- Please Help
7:16AM 3 Using a Dual WAN Load Balancing Device
7:04AM 0 Asterisk FXS & SMDI for Octel access
6:44AM 6 In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
6:21AM 0 Fw: Help on Load Testing
6:14AM 7 how to pop up called number details using php scripts in agi scripts
6:04AM 3 Q: ISDN / E1-PRI - fax problems - Receiving and setting of Service Indicator (SIN) / Bearer Capability (BC) / High Level Compatibility (HLC) / Low Level Compatibility (LLC)
5:47AM 1 Snom programmable leds / keys usage for pickup groups?
5:27AM 1 VoIP extn number planning
4:50AM 3 How to xfer calls or is my setup wrong?
4:26AM 8 high-quality, high-bandwidth codecs?
4:23AM 6 live monitoring (SIP only)
4:21AM 2 Voicemail not working properly
4:00AM 18 SIP jitter?
3:56AM 1 DTMF CLIP in Sweden and others
3:17AM 0 Help on Load Testing
2:56AM 1 Question about TDM11B Configuration
2:47AM 3 CVS or release?
2:37AM 2 MD5 in SIP's "register => ..."
2:28AM 4 Asterisk and Sipgate problem...
2:01AM 2 bristuff and audio drop outs (5 sec and longer)
12:46AM 5 jitterbuffers - suggested settings
12:14AM 0 Drop and Insert ?
 
Monday February 7 2005
TimeRepliesSubject
11:35PM 0 Howto( CLI or called number is attached to a database which automatically updates records let suppose if some dials xxxxxxx number so Company X's database record pops up on the computer screen of agent)
10:34PM 2 Best OS for Asterisk--newbie!!!
7:50PM 1 Conferencing without Meetme
6:57PM 0 kphone and *
5:34PM 10 Voicemail timeouts after 30sec's everytime.
4:28PM 0 Call Monitoring on IAX Channels - ChanSpy
4:28PM 3 SIPP load testing - unexpected message - anyone using sipp sucessfully ?
3:48PM 6 SIP port blocked in Dubai ?
3:46PM 11 How to number extensions - Which way is best?
3:32PM 1 In-band disconnect problem (legacy PBX) - asterisk doesn't hear t he touchtone?
3:28PM 1 Asterisk and SER differences
2:58PM 0 ot: WiSIP with zyxel firmware
2:53PM 2 asterisk to asterisk communication
2:38PM 0 IAXy Heat? Aluminum case anyone?
2:32PM 0 seeking references
1:38PM 0 newbie setup
1:31PM 0 Informations
1:20PM 6 2 x Fritz!pci card
12:58PM 0 RE: Asterisk-Users Digest, Vol 7, Issue 93
12:45PM 6 Record() cut off after 40 sec
12:45PM 0 Asterisk 1.0.1 - CCM 3.0.3 - GNUGK 2.0.8 - OpenH323
12:40PM 6 Broadvoice issues
12:16PM 5 Zaptel down after upgrade.
11:51AM 2 Asterisk on a single phone line
11:36AM 4 *HOWTO* : using mime-construct with outlook - send fax to email recipient
11:27AM 0 can not dial problem
11:23AM 12 no sound playing vm greetings and options
11:17AM 1 Provider with GA DIDs and LNP
11:00AM 0 Calls from SIP ATA to CO
10:57AM 0 most popular addons?
10:53AM 5 TDM-400P and Grandstream Question
10:28AM 0 Asterisk success histories in business?
10:04AM 2 CDR, RingTo Number, and DST
9:48AM 0 Incomming Fax deteccion
9:14AM 0 Recording already started on that call
8:53AM 1 Group=????
8:45AM 1 Asterisk => SKYPE
8:10AM 5 incoming calls in h323 do not come to right dialplan
6:48AM 1 Festival patch
6:36AM 1 Incoming Call Problem
6:19AM 3 callback agents cannot transfer calls
6:04AM 1 PocketPC Softphone?
5:29AM 5 ADM 0.5 - Asterisk Desktop Manager (alpha)
4:23AM 2 Pro biz Asterisk
4:20AM 1 multiple nics and internet
4:06AM 3 How to Create customized audio file to use withASTCC??
3:22AM 1 Remote MWI via IAX?
2:48AM 6 Newbie help/pointers required - configure xlite with asterisk
2:45AM 0 Small PHP script for displaying * CID database in Cisco 7940/60 XML
2:08AM 1 Failed to query database. Check debug for more info
1:28AM 0 RealTime Configuration for extensions.conf
12:55AM 0 TDM400P FXS works only if two lines are off hook?
12:45AM 12 IAX2 Trunk Problems with NAT
 
Sunday February 6 2005
TimeRepliesSubject
11:49PM 0 wanted: sample config' using GOTOIF's for all features for a roll-out
11:46PM 0 re: difference between STUN servers and far-end solutions
11:44PM 0 Which version of asterisk-oh323 should i use with asterisk v1-0-5.
10:38PM 0 Intel 537EP is NOT the MD3200 aka X100P [Re: Intel 537EP chipset, revisited]
9:23PM 0 "whispering" mode in Meetme?
5:42PM 1 IAX2 Bandwidth Study
4:56PM 1 Soft keys and transfer problem on Sayson 480i
4:37PM 0 passing "*" into a dial plan
3:45PM 1 Understanding the "Hint" priority.
2:47PM 1 no caller ID presented from 12SP+
2:19PM 0 Fax-modem
2:08PM 0 Using Asterisk to monitor in/out calls (single line)
2:01PM 0 SIP URI modified unexpectedly! Is that a router problem?
1:52PM 3 Voicepulse DNID is blank - Any other options?
1:51PM 8 snom soft phone
1:31PM 1 Call forwarding of IAX inbound call
1:03PM 1 Call status after Answer
12:55PM 18 Autodetecting faxes
12:17PM 0 blindxfer not in stable 1.0.5?
11:24AM 2 Cisco 12SP+ firware anyone?
10:45AM 14 iax2-jitter-trunking?
7:07AM 0 FYI - New firmware from Sipura
6:06AM 4 Proxied SIP
5:56AM 3 Question about X100P card
5:13AM 5 Help with extensions
3:20AM 0 Xorcom Rapid 1.0 released
2:56AM 0 IAXy ring frequency
1:46AM 3 inter asterisk
1:45AM 2 Need help with perl script/agi for ringback
 
Saturday February 5 2005
TimeRepliesSubject
4:17PM 0 ISDN Phones With Asterisk
3:13PM 0 Problems with SIP invite due to long ping round trips
1:43PM 3 Question about VoIP providers
1:16PM 0 Beep every 5 seconds
12:35PM 4 RTC Client (maybe VAD related)
9:52AM 1 cannot dial non-local numbers (junghanns QuadBRI cards)
9:10AM 1 OT: FWD and IAX: down?
8:48AM 2 Siemens C200 phone - callerid not visible on FXS
6:14AM 3 asterisk@home basic
5:55AM 0 [Fwd: D-Link DVG-1402S VoIP Router]
5:19AM 3 ISDN X-Over
5:09AM 0 Question about VoIP Solution
5:01AM 0 Inbound SIP to demo context
4:49AM 1 TAPI integration with * using Identapop software
3:11AM 1 CallerID and anonymous SIP calls
 
Friday February 4 2005
TimeRepliesSubject
10:03PM 2 ASTCC error on free calls
8:40PM 0 Need some Advise
8:23PM 2 MYSQL Failed
6:26PM 16 Encrypted VOIP?
6:24PM 0 Re: Can't get Polycom auto-answer to work (Solved)
5:06PM 0 codec0 = 516 is not codec1 = 216
4:42PM 13 FIX YOUR AUTO-RESPONDERS!!!
4:19PM 1 toll-free anonymous
4:10PM 1 Polycom Auto-Answer and Call Transfers
3:40PM 5 No ring tone on Outgoing calls
3:33PM 1 Multi Office Configuration
3:19PM 8 Limit MOH processes
2:44PM 1 vicidial and mysql ........help
2:30PM 1 External Callforward (Vanity CLI)
2:28PM 2 AU caller ID with Sipura SPA-3000
2:18PM 1 Call pickup across technologies (SIP, IAX, MGCP)?
2:18PM 1 autoAnswer and autoAnswerLogin?
2:16PM 0 Conference Bridge?
1:47PM 0 DTMF Problem with analog phones
1:00PM 0 Updateing to Stable from CVS
12:21PM 3 Server Criteria
12:03PM 0 patch for chan_capi error condition report when receiving CAPI_CONF:CAPI_LISTEN message
11:49AM 0 FC2 RPMS are updated
11:16AM 11 callback on busy
10:32AM 0 2 x100p + Static + echo
10:23AM 1 Snom Phones Volume
9:43AM 7 BRI in the US?
9:29AM 1 *, BeroNet BN4S0 and misdn - problems
9:23AM 10 Callerid problems with 1.0.5
9:15AM 2 zapata.conf ERROR?????? please help
9:05AM 3 PCMCIA card
8:52AM 5 IAX2 register Refresh
8:44AM 0 Specify a codec in dial plan?
8:40AM 0 TMD card to buy.
8:25AM 0 manager api - Async:True?
8:19AM 12 HP ProLiant server for Asterisk
8:12AM 1 echo's + cheap phones
8:10AM 4 No Playback() when Digicom TE110P enabled
8:02AM 1 X-lite to Cisco ATA - no RTP
7:43AM 8 How to Create customized audio file to use with ASTCC??
6:45AM 11 T.38 bounty
6:44AM 3 gsm audio files
5:59AM 1 Microsoft RTC Client SDK with Asterisk
4:45AM 2 Swap Memory get used totally
4:16AM 8 Bristuff and incoming call problems
3:22AM 1 Intertex IX66 incoming IAX
2:49AM 5 ASTCC Apllication
2:08AM 2 New Asterisk user with a goal
2:07AM 1 Q: how to receice the number of the called party back?
2:00AM 1 Q: charge info on E1-PRI
1:35AM 5 why asterisk and ser
12:36AM 0 (no subject)
 
Thursday February 3 2005
TimeRepliesSubject
11:54PM 1 Help with chan_h323
7:32PM 3 MWI with IAX
7:13PM 0 Dial timer problem? Short rings.
6:21PM 2 queue-timeout- press button to remain on hold
4:44PM 0 zttool user's manual & HDLC Abort errors
4:18PM 10 Q: How to get the preset callerid from a CLID-no-screen E1-PRI
3:55PM 0 Australian Caller ID with Sipura SPA-3000
2:16PM 9 Can't get Polycom auto-answer to work
1:57PM 2 How to charge for Asterisk installations and ongoing support?
12:45PM 3 Good 800 Number provider
12:44PM 12 OT: How to "own" a telephone number?
12:37PM 2 FastAgi Help
12:34PM 1 DTMF Payload type
12:22PM 1 Incoming call not ringing
12:09PM 1 AMP with SUSE9.2 (Apache2)
12:06PM 4 Mi extensions keeps ringing
12:01PM 0 AsteriskBrasil.org - We have an email list!!!
11:59AM 4 E&M Wink problems
11:56AM 1 Multiple mailbox on the same SIP extension
11:56AM 4 E1's and span - what questions to ask my service provider
11:44AM 1 Forwarding voicemail messages
11:41AM 0 Everyone is busy/congested
11:25AM 3 Odd behaviour between Grandstream and Xlite
11:16AM 4 Question about wildcard T1 card
11:03AM 0 DTMF Payload Type:
10:02AM 4 astcc digit timeout
9:51AM 3 Where are chan_capi bug reports and bugfixes sent?
9:28AM 0 Difference between Asterisk and VOCAL
9:08AM 0 Call Forward Loop
9:04AM 0 Different rings
8:57AM 4 Asterisk Dialplan command "PPPD" released
8:47AM 1 free pocketPC softphone (toshiba e750)
8:17AM 0 Automated CallbackLogin
8:15AM 2 Individual contexts pending on Caller-ID?
7:51AM 0 Busy Extension Ring to alternate.
7:37AM 4 good god! stop the damn auto-replys!
6:20AM 5 Asterisk crashes from time to time
6:18AM 11 Cisco 7960G phone crashes during SIP upgrade
5:32AM 0 Error on compiling oh323
5:30AM 0 Grandstream ATA 486 works only with ulaw and alaw codecs.
4:37AM 4 Nortel i2004 support asterisk?
3:50AM 0 Incoming SIP calls with different signaling and RTP IP addresses
3:43AM 1 How to forward a call to the same ISDN box ?
3:40AM 1 403 Forbidden when registering sip user database on backend
2:22AM 0 Special "error" numbers
12:37AM 3 IAX dns lookups
12:01AM 0 key in number after 'h' extension
 
Wednesday February 2 2005
TimeRepliesSubject
11:10PM 1 BRI only 2 calls
10:48PM 1 TDM series + kernel 2.6
10:38PM 0 tuning for ulaw g.711 - Polycom IP500
10:02PM 2 using the MYSQL command to insert a record
9:58PM 1 Asterisk problems behind firewall
8:08PM 5 Calling Asterisk Autoattendant With SIP Phone
7:49PM 3 outbound 911 calling
6:56PM 6 HEEEELP!!!!!!!! with file CODEC_G729.SO
6:33PM 2 How to download CVS with attended transfers
5:00PM 2 different IAX ports for different contexts
3:21PM 2 Using Asterisk to Find a Live Person
3:09PM 13 MeetMe & ztdummy
2:49PM 9 Broadvoice problems with outbound calls {Scanned}
2:44PM 0 Problemas with Basic Services.
2:30PM 0 Speex pass through on SIP
2:15PM 17 911 and Cops knocking on my door
1:55PM 1 Reproducible crash with CVS stable (from about 5 days ago...) - but only from iax clients
1:36PM 2 Asterisk@home - problem getting console output ...
1:10PM 0 RES: AgentLogin / AgentCallbackLogin transfer pro blem
12:46PM 2 how to add more TDM04B
12:34PM 3 Installation on Fedora 3
12:07PM 4 ZapTel Errors on boot
10:49AM 2 Asterisk with SourdCard
10:45AM 0 32 FXO 32 FXS and record call
10:21AM 0 rxgain won't always ring extension
10:19AM 3 [OT - somewhat] chan_sccp status
10:17AM 0 AgentLogin / AgentCallbackLogin transfer pro blem
10:07AM 0 DTMF outbound problem with ata 186
10:05AM 0 Does any Cisco VoIP kit support IAX?
9:57AM 0 Paging Zultys Phones
9:39AM 1 (OT:) Tool for trying/troubleshooting WAN/LAN
9:21AM 0 AgentLogin / AgentCallbackLogin transfer problem
8:01AM 4 * not hanging up when call from POTS to IAX phone
7:55AM 13 howto answer a call in a queue
7:55AM 0 clicktocall via manager with cisco 7905
7:50AM 0 403 forbidden error
7:38AM 0 Ignoring too old packet packet
7:13AM 9 Cisco 7940 [SIP], DTMF and Voicemail
7:07AM 0 IAXy Configuration for Alternate Server
7:04AM 1 Transfer call digit length
7:02AM 5 Disabling native bridging for IAX calls
6:45AM 0 ExtensionState problems using Manager.conf API
6:23AM 0 SIP Call through Asterisk
6:05AM 8 problem in compiling asterisk-addons
5:54AM 0 Integration Asterisk and Siemens Hicom 150
5:29AM 1 Asterisk waits 4 rings before FXO answers incomingcall
5:03AM 2 Hangup detection with TDM400 in UK
4:43AM 0 Asterisk waits 4 rings before FXO answers incoming call
4:33AM 1 Asterisk cmd SayNumber : how to pronounce in another language - we say "one-and-twenty" instead of "twenty-one"
4:21AM 2 X100P Setup
3:49AM 13 Reccomendation for reliable handsets
3:40AM 3 Installing ASTERIS@HOME, How to install on text mode same help?
3:01AM 1 SIP with Delay
3:00AM 0 ZAPHFC Drop calls
2:59AM 1 Astrerisk + Conversation OneWay
2:23AM 3 Forbidding ZAP interface bridging
12:23AM 0 How to continue execution after called party hangs up?
12:07AM 4 new install
 
Tuesday February 1 2005
TimeRepliesSubject
11:05PM 0 chan_capi and G711u
10:19PM 0 Outbound proxy
10:10PM 0 call forwarding with code
9:56PM 2 Is Bell HDSL in Ontario good solution for VOIP?
9:24PM 46 load balancing 20 asterisk servers
9:19PM 5 astGUIclient users should not upgrade to Asterisk 1.0.5
8:33PM 3 Problems compiling zaptel on SuSE V9.2
8:23PM 2 X100P not hanging up...
7:57PM 2 Why is host= being ignored in sip.conf ?
6:51PM 1 chan_sip.c:7296 handle_request: Unable to create/find channel
5:53PM 1 3G Video Mobile Phone
5:32PM 3 list administrator.....???
5:25PM 0 Realtime and callforwarding
4:52PM 1 PCI CARD X100P CLONE FXO WORK ALSO AS FXS ?
4:36PM 0 how to make a call with asterisk from shell ? (orwith a .sh file )
4:29PM 1 Play tone till first digit read
4:21PM 17 Terrible inbound call quality vs. outbound
4:02PM 0 Help with DIAL command
3:41PM 1 how to make a call with asterisk from shell ? (or with a .sh file )
3:36PM 6 *ASTERISK* Install and configure Step by Step.
3:15PM 1 Custom MusicOnHold
2:37PM 1 choppy sound after 15 minutes in a call
2:10PM 4 Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 caller id?
1:46PM 1 AGI global style variables
1:44PM 0 Odd error.. sip_xmit Bad file descriptor
1:12PM 7 Soft phones that _actually_ work under Linux?
1:02PM 0 Crash: Call from IAX-client to a distribution where the IAX-Client is in
12:54PM 2 Re: Asterisk-Users Digest, Vol 6, Issue 325
12:50PM 19 Outlook Integration
12:42PM 5 IAX2 Softphone
12:30PM 2 Asterisk Not hanging up DS0 when number called is busy.
12:07PM 0 AGI two calls - one hangs up - othere gets "interrupted system call"
11:32AM 0 newbie: questions
11:27AM 0 TBM400 no callerid on incoming calls?
11:21AM 7 Linksys PAP2 / RT31P2 + multiple G.729 calls
11:13AM 5 HFC-5/S + Asterisk
10:59AM 0 OT: IAX provider for business
10:52AM 0 manager api events (pri vs pstn)
10:43AM 1 Zap channel occasionally misses dialing thefirst digit
10:40AM 1 Scope of definitions
10:35AM 3 Outbound calling with TDM400P
10:13AM 0 how to add extension to mysql database
10:12AM 0 ChanSpy?
9:55AM 1 FW: Messaging with * and eyeBeam
9:34AM 0 Messaging with * and eyeBeam
9:31AM 0 One extension, multiple endpoints
9:13AM 0 Call Forward - Need Help
9:03AM 5 Zap channel occasionally misses dialing the first digit
8:55AM 0 Limiting no. of calls on one channel
8:49AM 1 Germany specific settings for Grandstream ATA286 - Polarity reversal, impedence and onhook voltage
8:46AM 0 VoiceMail ANI question
8:43AM 3 Different ring when called by door entry
8:35AM 13 IAX native transfers
8:26AM 2 mysql based adressbook with agi and web interface ?
8:16AM 0 No Sound Playback
8:12AM 0 RE: Re: RE: Answering Machine Function?
8:03AM 2 Feature automon
6:55AM 1 SIP Challenge response bug?
6:40AM 4 X100P Clone
6:34AM 1 i4l: Quality of Voice
6:05AM 0 Troubles with Macro-stdexten and dial
6:00AM 0 Asterisk Services working with SER !!!
5:09AM 6 IAX registration keep alives
4:51AM 6 Error on compiling oh323 0.6.5 on cvs stable asterisk
4:45AM 1 MeetMe missing?
4:26AM 1 i4l + SIP: Audio One-way
4:17AM 2 asterisk remote monitor
4:13AM 0 How to mark calls for inclusion in CDR ?
3:58AM 0 24 CTU ringtone for grandstream 101?
3:36AM 0 Call queue ackcall doesnt work
3:17AM 0 Bangkok DID?
2:43AM 0 Actions taken drugin calls - are there any other keys active beside # for transfer ?
2:12AM 2 How to compile "iaxclient" with MinGW/Cygwin
1:26AM 1 Where to download the soxmix please?
1:23AM 2 broken message waiting indicator on Polycom IP600?
12:19AM 2 AGI Script for CID Rewrite and CID Name lookup