asterisk users - Oct 2005

Monday October 31 2005
11:43PM 4 Asterisk 1.2.0-beta2 Released
10:40PM 1 What's the deal with "secret=" vs. "password="?
8:30PM 0 Echo Canceller question- is there aviablesolution?
8:14PM 0 Strange problem with dtmf pound and star
7:47PM 1 HELP PLEASE: What A Pain RSA
6:47PM 0 Fw: Attended transfer restarting asterisk switch
6:14PM 2 echo codec related?
5:58PM 1 musiconhold -vs- musicclass problems setting the differnt class of music
5:31PM 0 problems with 1.2 Beta1
5:14PM 3 spandsp patch
4:51PM 0 No D-channels available! CVS-HEAD-10/31/05-16:01
4:47PM 0 Release Announcement: HooDaHek 0.7
2:46PM 0 Any experiences with Orion hardware echo cancellers?
1:47PM 1 dial-out gives always "not found" (dial-in works fine)
1:32PM 5 queue scheduling...
12:28PM 1 pls help compile rx_fax (patch / Makefile)
12:19PM 1 Agent channels causing problems
11:48AM 4 Asterisk and Zaptel Versions Command?
11:02AM 13 llamdas por 4 lineas a eeuu tarifa plana
10:55AM 1 Caller ID to SIPURA-1000, 2000, 3000 Handset Prlblem in only showing the destination callerId
10:51AM 0 Calling Name Not Displayed On Incoming
10:02AM 2 Release of Asterisk 1.2
9:33AM 0 (no subject)
8:51AM 1 chan.iax2.c errore
8:27AM 1 lucent TNT h323/sip config?
8:26AM 0 Add Contexts Dynamically
8:25AM 1 Adit 600 and Groundstart
8:13AM 2 Rugged VoIP phones for use with asterisk
7:50AM 1 Info on beta1 seem to be broke
7:46AM 1 How to remove a VM greeting - go back to default Allison message
7:30AM 5 Timestamps in Console?
6:37AM 1 FXS Disconnect Supervision (Kewlstart / Open Loop Disconnect)
6:07AM 0 [Asterisk Voicemail] Quota
5:43AM 0 IAX2 trunks encrypted?
4:27AM 3 Tone generator module
3:43AM 8 A2Billing
3:11AM 4 Segfault on latest head 10/31
3:05AM 1 H323 one way audio using oh323
1:48AM 0 can't add zap channels to a group
1:33AM 4 sip show peers
12:40AM 1 lilte help please
Sunday October 30 2005
11:25PM 1 How to specify when to go to 102 priority
10:38PM 0 Re: Automathic call forwarding (Gianni (priv.))
10:18PM 0 Channel Data Access
8:20PM 3 Asterisk to Avaya IP Office
8:05PM 1 zap group channels
7:15PM 0
7:12PM 1 Distinguishing Busy from No Answer
5:27PM 1 Call Transfer problems-am I missing something?
3:44PM 2 Can anyone explain reason for this echo
3:10PM 1 Attended transfer restarting asterisk switch
1:14PM 2 Cisco 7960 Skinny Firware
1:00PM 0 dialout gives 404 (using sjphone (dialin works fine))
10:09AM 2 no sip peers after restarting asterisk?
9:13AM 1 Re: feature usage/digit detection
8:27AM 0 Automathic call forwarding
7:44AM 1 gotta be a dumb question...
7:22AM 1 Trouble with D-Link 2004S and E1 PRI
7:09AM 2 libpri
7:03AM 1 VoiceMailMain() in 1.2-beta
4:57AM 0 Pattern for matching CALLERID
3:09AM 3 SCCP support is making good progress
12:38AM 1 Announce in MeetMe Rooms
Saturday October 29 2005
9:57PM 0 Access to channel data...
8:56PM 0 Asterisk 1.2
7:22PM 1 ericsson pabx and digium card TE110P
6:47PM 1 chan_zap ignoring stuff in beta1?
5:39PM 1 Play music while on incoming connections ringing the internal user
3:06PM 1 I give up - Help with TE410P
1:01PM 2 zaptel + RH3?
11:42AM 1 Set outgoing MSN with chan_capi-cm
10:20AM 0 Meetme streaming a recording
9:31AM 1 Credit card machines, Asterisk and Digium any issues?
7:57AM 3 dropping extra frame of G.729 since we already have a VAD frame at the end
6:07AM 0 ANNOUNCEMENT: Asterisk-Java 0.2-rc2 released
5:18AM 2 CallBack Suggestion
4:40AM 2 Zyxel USB ISDN works with Asterisk
4:25AM 0 Prblem with 723 and 729
1:47AM 2 UK Pounds and pence prompt wanted
1:46AM 0 Asterisk - PRI - Cisco
1:33AM 0 Repost:Generate white noise to avoid RTP timeout
Friday October 28 2005
11:27PM 0 SPA3000 as trunk - no caller ID - solved
10:04PM 4 Sipura SPA 2000 - error using second line
5:49PM 1 Geneys
5:11PM 0 asterisk outgoing does not work
5:07PM 0 anyone using these?
4:55PM 1 Outbound fax solution
4:39PM 0 Warning - AgentCallBackLogin has changed, Possibly will cause 99% cpu
3:27PM 1 I need GoIAX or VoipBuster A@H examples?
2:46PM 0 Why can't I dial - just using SIP internally
2:16PM 1 SIP Host "Unspecified"
1:52PM 0 problem with asterisk-realtime-odbc
1:20PM 0 Metreos
12:15PM 2 Having Meetme call another conference
11:22AM 1 Sipura 841 echo cancel question
10:49AM 4 Opinions on IAX JitterBuffer in old-school 1 .0.0?
10:42AM 2 Mediatrix Gateways
10:24AM 0 Montreal Meet Asterisk Get-Together
9:56AM 2 Help with Zultys
9:25AM 2 Top and asterisk performance
9:19AM 0 Cell phone extension woes
8:46AM 2 2 problems
8:33AM 0 Prevent transcoding
7:53AM 1 Ouch - Error while writing audio data - broken pipe
7:11AM 2 IAX channel options
7:11AM 0 h323 no audio from the sip phone to the outside world.
6:15AM 1 when is 1.2 being released?
5:59AM 2 OT: Suggestions for E1 Service in the UK
5:51AM 2 SV: call queue
5:45AM 0 PhoneCALL v2.7 goes MultiLingual
5:44AM 1 call queue
5:26AM 8 "GSM cards" / "mobile phone cards" for Asterisk?
4:56AM 0 Anyone running zaptel's watchdog in production?
4:35AM 2 [Re] Re: Echo canceller on TE406 & Asterisk
4:00AM 1 Dial with 44 and +44 prefix
3:35AM 2 Help Installing Asterisk.....
3:25AM 3 ADSL
3:08AM 3 Webui to show registered phones
3:07AM 2 Asterisk GUI/web interfaces that don't change config files
3:03AM 0 URL Dialing from SNOM phone
3:02AM 0 Queues with SIP/phones as members, leave when empty?
2:56AM 0 IAX voice problem, no voice at all
2:16AM 1 chan_bluetooth and audio problem
2:14AM 2 Problem With Sipura
12:59AM 2 Console detach.
12:22AM 1 X100P doesn't show Caller-ID
12:17AM 0 AW: problem with receiving faxes over cisco as5300
Thursday October 27 2005
11:29PM 2 Echo Canceller question- is there a viablesolution?
10:16PM 0 {Resend} Process VoiceMail file to attach
9:19PM 2 TDM04B NEW CARD WITH zaptel 1.0.7
8:22PM 2 Where does Asterisk put it's files
7:01PM 5 Echo canceller on TE406 & Asterisk
6:52PM 1 Taking the plung to CVS HEAD
5:36PM 0 call monitoring in external application (newbie)
5:01PM 1 PRI to SIP Problem
4:12PM 1 Opinions on IAX JitterBuffer in old-school 1.0.0 ?
4:06PM 2 Not saving voicemail message
3:40PM 0 polycom ip500 mwi, quite please
3:30PM 1 Zapbarge feature available?
3:28PM 2 Queue Login Out Question
3:24PM 2 problem with receiving faxes over cisco as5300
3:03PM 1 Outgoing fax detect
1:25PM 1 Delay ReInvite
12:54PM 1 Is anyone using OpenSer - A fork of SER?
12:41PM 1 Words for the Asterisk community to live by.
11:59AM 0 sip not working suddenly
11:55AM 1 Is it possible to generate or play some white noise in Asterisk?
11:16AM 0 QoS Monitor
11:03AM 1 TDM01B vs. X100P
10:33AM 0 cannot get dialtone or ring on FXS ports (TDM400p)
10:06AM 1 Network Architecture Question
9:38AM 0 Have IAXy signal busy without losing ongoing call?
8:15AM 1 Test after Hurricane Wilma
7:29AM 0 R: Bristuff question
7:22AM 1 Bristuff question
6:48AM 0 Asterisk 1.2beta and te411p: incorrectly reporting sometimes "all channels busy"
6:39AM 1 Vmail.cgi and realtime?
6:04AM 1 Message Waiting Indicator and PRI
4:58AM 1 spandsp / txfax exit codes / logging?
4:13AM 1 Problems compiling asterisk zaptel for Asterisk 1.0.9
4:05AM 0 PRI Echo - Solved with KB1 Patch
3:21AM 0 Overlap dial and "match as you go" = how to implement early dial on telco line
1:01AM 0 sKinny in database
12:48AM 1 please recommend phones with adsi.
Wednesday October 26 2005
11:29PM 1 faxdetect on voicemail
11:15PM 0 make sipura stop generating stale nonce. Device comes in and goes out every 1 minute
10:38PM 1 Echo Canceller question- is there a viable solution?
9:50PM 0 Backend network, one-way audio...trunking
9:17PM 2 Aussie Call home!
8:00PM 1 Simple SIP only Asterisk Configuration
7:36PM 2 Trying to genereate dial tone, but stop after first digit dialed.
6:53PM 1 Zaptel stop hangs server
6:03PM 1 Asterisk IVR and Cisco Call Manager
5:54PM 1 How to auto-speak agent's number once agent answer the incoming call
5:30PM 1 tellme/skype voice apps go live
3:27PM 1 Asterisk+Nat+sipura (Help)
2:01PM 1 Polycom 601 XHTML microbrowser
1:48PM 3 smp
1:40PM 4 asterisk using tdm400p has echo
12:48PM 3 ANNOUNCEMENT : A2Billing - AreskiCC V3 new release
12:18PM 0 fax2mail script update (includes hoodaheck compatibility)
12:01PM 0 FC4 + ztdummy + timming + trunking
11:46AM 0 Fw: UK BT IDSN30e 'pass through' with TE205P/AvayaArgentOffice?
10:59AM 2 web management interface
10:31AM 1 Incoming CallerID Name display
9:56AM 2 How to do Call Forwarding
9:35AM 0 New Asterisk Mailing List: asterisk-i18n
9:17AM 1 2-line phoneline
8:23AM 5 SPA3000 as trunk - no caller ID
8:22AM 0 AW: Some problem with CAPI support
6:55AM 2 polycom software
6:17AM 1 UK BT IDSN30e 'pass through' with TE205P/Avaya ArgentOffice?
5:38AM 0 Some problem with CAPI support
3:35AM 0 RE: strange behaviour of asterisk sip.conf
1:53AM 0 MeetMe architecture problem
1:33AM 2 Asterisk iptables rules
1:31AM 1 Zaptel + No Hangup
1:02AM 0 Asteriks configuration
Tuesday October 25 2005
9:41PM 2 Asterisk on PPC Linux
9:15PM 0 Asterisk+Nat+Sipura/Linksys
7:31PM 0 clarification please: accountcode, pri channel groups, and CDR
7:29PM 2 Incoming calls via CAPI and AVM Fritz Card
5:46PM 3 Wanted to Swap! TDM400 FXO module(s) for FXS
5:20PM 1 carrier blocking
3:41PM 0 Send tone to src on supervision
3:33PM 2 Fwd: ByVolution
3:09PM 0 Grandstream GXP2000 tftp config
2:20PM 3 FCC VoIP wiretap ruling challenged
1:59PM 3 HELP!
1:31PM 0 OT: Need Asterisk VOIP Support for Customer in Louisiana
12:57PM 1 strange behaviour of asterisk sip.conf type=user vs type=peer
12:27PM 2 Sipura SPA-3000 and Gigaset DECT phone: no ring
12:04PM 0 App_directory + Festival
11:43AM 2 PDA softphone....
11:15AM 5 How to configure the communication between two Asterisk servers
10:44AM 1 variable `oh323_tech' has initializer but incomplete type
10:07AM 2 Echo cancel and fax
9:56AM 0 ECT - Specifying the transfer destination.
9:30AM 1 re: changing protocols and transcoding
7:41AM 1 H323 REGISTRATION PROBLEM: Gatekeeper 'Nortel_H323_Gatekeeper@.. ' found but failed to register
7:36AM 3 Sangoma A104 errors
7:18AM 0 Voicemail prompts not heard on Cisco Phone
7:11AM 2 AudioCodes - TP260
7:00AM 0 Festival() works fine when I call from PSTN and not when I call from XLite... What's going on?
6:13AM 0 Re: FCT-11M
5:54AM 0 MWI for other purpose than voicemail?
5:51AM 1 Question on callingpres and blocked numbers
5:26AM 2 pppoe-server & Asterisk
4:24AM 0 Swissvoice Vizufon firmware
4:10AM 3 Realtime sip register=>
3:29AM 13 iaxmodem
3:16AM 0 Distinguishing National from International Calls on Zap Channel
3:15AM 1 connect 2 phones like in FOP
3:00AM 1 Agent logout
1:59AM 1 Asterisk user meeting in Oslo, Norway
1:42AM 5 Cisco 7905G Power over Ethernet - does it work?
Monday October 24 2005
10:46PM 5 Asterisk & SER for dummies ?
10:01PM 2 X101P and UK CallerID...does it work?
9:55PM 1 Belgium Meetings from Nov 11 to Nov 26?
9:14PM 0 patch for one-way audio for asterisk-oh323
7:16PM 0 How to tell what EC is in place (Was: RE: Terribleecho with Te110P and Adit 600)
7:08PM 4 Format of extensions.conf
5:53PM 0 Unavailable
4:31PM 0 Add SIP extension
4:00PM 2 Red Alarms, No D-Channels, and Crazy People
2:45PM 0 Unicall Error ... T1 Timeout
1:55PM 2 Siemens HI-path to ASTERISK
1:34PM 1 Ticking sound in wildcard tdm400p, Please Help
1:34PM 0 Recommend an LD provider who can use IAX
1:22PM 0 more and more ""
1:20PM 0 Government/Enterprise User Group
10:43AM 1 SIP to CAPI - Soundcard required?
10:31AM 2 Urgent - Need Help - Audio Issues
8:34AM 1 Largest working config files?
8:18AM 3 polycom, call waiting, queues
7:41AM 0 Hangup ZAP channel
7:17AM 0 How to setup parked/on-hold so sorresponding buttons on VoIP phones light up
6:58AM 0 Thounsand of SIP extension
6:33AM 0 could set up only
5:59AM 2 cvs head + spandsp
5:20AM 1 new toy
4:50AM 1 Asterisk vs Sipura SIP problem?
2:53AM 1 "zap show channel 1" says "PRI signalling" on my zaphfc BRI card
2:35AM 1 0.2.0-RC8o (* 1.0.9) + No Caller ID
2:05AM 3 configuring Cisco 7905G for SIP - how?
1:59AM 0 NAT Problem after first call
1:35AM 1 Asterisk Realtime - MySQL Extension registration problem
1:03AM 6 Where is the text of the voicemail email ??
12:24AM 1 Passing parametrs to php agi scripts.
Sunday October 23 2005
11:28PM 0 Queue application problem
9:04PM 1 asterisk -RT
8:43PM 0 Fwd: Re: Problem with asterisk 1.0.9 and sip and dtmf
7:14PM 0 Problems with Festival...
4:27PM 1 T1 Hardware Recommendations
3:22PM 0 iConnectHere (or DeltaThree) trunk settings
2:24PM 1 Hardware setup question
9:23AM 6 Trying to clarify ideas about spands, libtiff & FC4
9:09AM 1 SIP DTMF problem
8:28AM 1 Adit 3104 configuration
3:40AM 1 How to play a voice file for " decline "
2:45AM 1 Asterisk dropping call file without *any* notice
1:30AM 3 Anyone using Java SIP communicator with Asterisk ?
Saturday October 22 2005
9:50PM 0 Asterisk CDR records when a call is transferred
7:12PM 2 Satellite receiver over IP
6:08PM 1 IAX registration with FWD and Teliax - Lost
5:50PM 0 redirecting incoming calls to external phone (cell)
3:05PM 2 Problem with asterisk 1.0.9 and sip and dtmf
12:59PM 3 Modem Over IP: solutions ?
11:15AM 2 CDMA card or module made for Asterisk?
11:07AM 1 Do the quantity of hardware timing devices go up as call volume increases?
6:38AM 1 Linksys pap2 behind Linksys RT31
6:31AM 0 play a voice file voice for " decline "
6:03AM 4 One SIP dead, all SIP dead -- sipmedia gone?
5:36AM 5 voip provider in a box
5:28AM 0 need help:multisite with asterisk?
3:19AM 2 Filenaming for Incoming Queue Call Recordings (Reposted from "changing the filename of incoming call recordings")
3:18AM 0 Call problems using IAX
3:09AM 1 Cisco 7960G and Asterisk
2:23AM 1 chan-capi_cm - 10sec silence before ringing
12:03AM 2 Testing AreskiCC
Friday October 21 2005
10:58PM 0 FXO no release useing fxsks & disconnect supervision from telco
9:14PM 3 Queue Join Event
7:33PM 1 Asterisk Newbee: Fedora Core 4
4:54PM 0 Process VoiceMail file to attach
4:16PM 1 messagenet
3:26PM 0 FS: Single channel GSM to POTS "gateway"
2:46PM 1 OT: How to reach
2:02PM 1 Fax problem with zap trunk...
1:56PM 0 How to configure two Asterisk servers for one callcenter
1:39PM 1 How to wire Asterisk to stations?
1:17PM 3 Getting ztdummy to load on startup for X100P
1:15PM 1 Meetme admin option
1:00PM 3 asterisk locked up
12:52PM 0 Asterisk@Home and Internationalization & Numbers.
12:32PM 1 Voicemail Changes
12:19PM 1 Problem with asterisk-sounds-1.2.0-beta1
10:46AM 1 Polycom 501 can't find TFTP server?
10:25AM 0 Just a test...
10:08AM 2 force 911 over pri line
9:39AM 5 How to configure two Asterisk servers for one call center
9:07AM 0 Whats wrong with incoming
8:51AM 0 Definitive answer: time-range includes
8:26AM 1 Double DTMF with tdm card
8:15AM 0 OT: DID not working anymore? < --i s it appropriate for this list? Yes.
7:45AM 1 Queue_log multiple entries
7:38AM 5 MTP required for CCM integration ?
7:26AM 0 REPOST: Private/Anonymous/Restricted not being passedbyAsterisk Lost in the shuffle?
7:09AM 0 Incoming call and DID routing
6:51AM 3 cvs core dump
6:46AM 0 Force all local numbers and 911 out PRI
6:37AM 2 Re: T1 questions - could I got VoIP instead?
6:36AM 1 Custom handling of SIP 302 redirect?
6:03AM 0 RES: Configure Festival to speak in portuguese
5:56AM 3 Preventing abuse of Goiax
5:52AM 1 Swissvoice IP10S centralized phonebook
5:48AM 0 Asterisk Festival
5:34AM 0 No music on hold for agent channel
4:55AM 0 Configure Festival to speak in portuguese
4:35AM 0 exclude context?
2:39AM 0 Help on func_odbc.c or similar
2:01AM 0 php with oci8 agi script
1:44AM 3 2.6.13 zaptel incompability?
1:43AM 0 Call transfer caller ID
1:35AM 1 Group dial CDR
1:19AM 1 type UniCall
1:00AM 0 Problem with Asterisk newbie
12:58AM 1 SIP gateway: call hangups afer 3 rings
12:14AM 2 Does fwdout even work anymore?
Thursday October 20 2005
10:46PM 3 how many oh323
9:28PM 1 Asterisk Community Participant; Katrina Refugee UPDATE
7:51PM 1 Problem with Asterisk newbie2
7:11PM 4 merchant account
6:47PM 2 Multiple instances of asterisk showing from 'ps aux'
5:52PM 0 intergration with Nortel Meridian 81c
5:18PM 1 Voicemail/Record sending no RTP packets (CNG) back to caller when recording messages
4:04PM 1 Private/Anonymous/Restricted not being passed by Asterisk
3:56PM 0 Diaplan or iax.conf problem
2:52PM 0 Fyi: I-D ACTION:draft-guy-enumiax-00.txt (fwd)
2:32PM 7 DID not working anymore?
1:26PM 2 Help on Asterisk and Client SIP setup
1:17PM 1 areski Problem
1:14PM 1 Sip and autonegotiating codecs
11:47AM 0 Re: T1 questions follow-up
11:45AM 1 Digium list server and spam assassin
11:36AM 2 Re: T1 questions follow-up
10:57AM 1 Application return values
10:18AM 4 Problem with Cisco phone
10:15AM 3 Some questions regarding T1's
9:13AM 0 D-Link DG104S firmware upgrade for flash funcionality on *
8:31AM 3 cdr_odbc with tds
8:23AM 4 E1/T1 failover hardware
8:18AM 0 CallerID PHP Script
8:15AM 4 TDMoE and Badness in Kernel
7:39AM 0 Getting output from agi scripts (python)
7:38AM 0 Manager API - Supervised Transfer
7:01AM 4 Asterisk Billing
6:45AM 1 user name
6:32AM 0 Fwd: Re: [Asterisk-doc] You ASKED for an Asterisk book, you GOT an Asterisk book!
6:06AM 0 E1 (TE405p) SetCallerId problem
5:23AM 3 Asterisk Compilation with H323 working on it
5:20AM 0 codec voice quality ratings
5:17AM 1 wm_w DTMF solution for T1 tie line losing deigits.
4:50AM 3 Why Asterisk documentation is so poor...
4:39AM 2 Siwssvoice IP10S telnet password
3:59AM 0 Context configuration with AstTapi
3:35AM 2 Problem with Swissvoice IP10S and Asterisk
3:34AM 0 changing the filename of incoming call recordings
2:19AM 0 toll free dialing problems using SIP
1:45AM 2 Isdntrace utility
1:00AM 1 Any good docs for latest CVS-HEAD / Stable 1.2?
12:31AM 2 Can't build Asterisk on SuSE
12:30AM 0 Chan-capi sound choppy
12:15AM 4 Asterisk in Croatia - Zagreb
Wednesday October 19 2005
11:22PM 0 Dial Limit Call Options
9:27PM 0 how to edit or delete calleridname in From URI
9:17PM 2 chan_capi-0.6 configuration Query with Eicon Diva 4BRI
7:10PM 1 Re: Asterisk Evening in Melbourne Australia!
5:21PM 1 goiax configuration help please
5:20PM 1 Digium TDM400P (11B) problems
4:29PM 1 OT: Samsung DCS 70
2:38PM 3 New ISDN architecture available for asterisk
1:40PM 0 some faxes get cut off
1:39PM 3 Please recommend a phone
1:33PM 1 Dial 2 channels at onece: Not working anymore at CVS?
1:31PM 2 Asterisk 1.2.0beta1 for Debian Sarge
1:19PM 1 Free DID's
1:19PM 1 Problems Installing MPG123 on a 64 Bit System
1:01PM 0 Extension dialing out
12:51PM 2 sixtel DID
12:49PM 2 more voip patent madness
12:09PM 1 teliax audio issues - response
11:40AM 1 possible bug, what do you think?
10:59AM 0 unable to make connectivity between asterisk to external phone
10:10AM 0 NEWBIE HELP : chan_zap.c: Exception on 16, channel 1, call not being picked up on incoming X1-100P zap
9:39AM 3 uable to establish link between asterisk to external phone
9:04AM 1 Caller-ID via database lookup
8:55AM 0 Fwd: Re: IAX only speech one way
8:51AM 2 E1 PRI error: "!! Got I-frame while link state 2" and "!! Got a UA, but i'm in state 1" (long)
8:10AM 0 Trunk Dialing rules
7:28AM 1 Connection question
6:16AM 2 SIP CallerID
5:58AM 3 SIP to IAX
5:54AM 0 Fw: asterisk shutting down...
5:46AM 0 How can I signal a flash to PABX ...
4:39AM 4 Help with Dial Plan
4:18AM 0 undefined symbol: ast_smoother_feed
4:09AM 0 IAX termination/DID provider in Panama?
4:02AM 1 Asterisk on Slackware ...
3:39AM 0 my SIPURA ATA does not make calls thru teliax
2:59AM 1 what hw/OS to choose [please help]
2:49AM 1 Call queuing question
2:29AM 0 Persistant connection for MYSQL command
2:19AM 0 How to suppres leading zeros in zapata.conf?
1:29AM 0 Realtime - table voicemail
1:00AM 1 Problems Calling PSTN PSTN FROM ASTERISK
12:24AM 1 Asterisk hangs
Tuesday October 18 2005
9:27PM 1 Audiocodes MP-108
8:27PM 2 New TDM Revision in the wild: J
8:02PM 9 more dids added to
8:00PM 1 zaptel.conf config for CAS signalling
7:42PM 12 Terrible echo with Te110P and Adit 600
7:19PM 0 aah digital receptionist weird
7:17PM 0 Newbie IAX
6:54PM 0 Languages, Realtime, German, Finish
6:18PM 0 Language settings not working in astcc
4:01PM 0 whats the difference between and
2:34PM 0 central voicemail storage
2:16PM 1 Priority jump in AEL
1:33PM 0 Monit test for IAX2
1:12PM 2 IAX only speech one way
1:12PM 0 IP300 -> Asterisk -> Broadvoice -> PSTN Choppy / cuts in and out
12:43PM 0 Fwd: {100-1287} RE: DID"s
12:02PM 4 One phone ringing, one phone flashing ?
11:50AM 1 Forwarding Extensions using dialplan
11:48AM 0 zaptel problem
11:16AM 1 strange behavior after turning jitter buffer on
10:47AM 0 Re: Asterisk-Users Digest, Vol 15, Issue 108
10:21AM 0 Problem loading misdn driver
10:20AM 0 Re: Vontage Problems
9:46AM 1 sip rfc bye violated?
9:41AM 2 Fwd: {100-1287} RE: DID"s
9:32AM 7 Asterisk Redundency
8:59AM 4 Polycom IP501 and record on demand
8:26AM 5 Newbie Question: Help with incoming dial plan
8:11AM 1 select codec based on extension
8:02AM 0 Assistance with loging a particular event.
7:39AM 8 Fax2Mail
7:13AM 0 Hang up problem Costa Rica Indications
7:08AM 0 Display number dialled
6:23AM 1 setting a dialplan on a GXP-2000 Grandstream
5:58AM 2 SV: SV: Queues and call waiting indication
5:35AM 2 SV: Queues and call waiting indication
5:27AM 0 411
5:18AM 2 Agent recording and muxmon
5:13AM 1 Queues and call waiting indication
4:50AM 1 Recomendations for utility to generate Asterisk configuration
2:37AM 6 Can IAX be used without going thre a IAX server
1:58AM 1 error while writing audio data: : Broken pipe
1:53AM 2 fax device behind TDM400P
1:32AM 0 Slow dialling from PBX into * via E1
12:50AM 0 Pb musiconhold with G729 codec
12:11AM 3 CAPI - displaying individual MSN
12:05AM 8 free dids on
12:03AM 1 Talkoff (Spurious DTMF) with and TE406P
Monday October 17 2005
11:22PM 0 How to use Use different ports to authenticate SIP/IAX users
7:22PM 0 Fwd: Re: SIP to SIP sadness
6:25PM 2 DID's
5:29PM 1 Can I use ANY port for SIP device?
5:11PM 0 Asterisk 1.0.9 - PortaOne Radius
5:04PM 2 Dial command in extensions
4:45PM 0 Argentina - Vontel - Asterisk
4:26PM 1 Uniden UIP200 Issues
3:28PM 2 Bizarre Echo Problem
3:14PM 2 Teliax IAX problems -- Asterisk doesn't see answer
1:51PM 1 (no subject)
1:50PM 6 Where can I find Polycom 600 config files?
1:42PM 0 FW: ISDN PRI and E1
12:56PM 0 Ztdummy is shutting down my sound
12:49PM 2 CDMA phone line for Asterisk?
12:48PM 1 Middle Ground between POTS and T1?
12:44PM 1 Problem with incoming calls
11:54AM 1 SIP to SIP sadness
11:38AM 3 Multiple calls per phone
11:35AM 1 can't compile ast_*fax
11:15AM 1 How can I get a dialtone calling from outside...
10:27AM 6 initiate call recording from phone.
10:02AM 0 Interface with ability to originate call
9:59AM 0 Ruby module for the Asterisk Manager Interface
9:58AM 0 cmd SIPRedirect for loadbalancing
9:15AM 1 astcc missing to bill random calls?
9:06AM 0 Connecting TIE trunk to Astericks
8:37AM 0 Legacy PBX Integration and Zaptel.conf Timing Source
8:01AM 0 No Audio from Console but mpg123fromshellworksfine.
7:05AM 1 Call transfer - atxfer
6:37AM 0 Transfering calls. Dial plan
6:28AM 0 fax receive problem on zapata channel
6:28AM 1 Double Ringing for PRI Calls
6:19AM 0 Asterisk Busy Detect
5:55AM 4 compiling Asterisk 1.2 with zaptel and h.323
5:30AM 1 module loading error with Ubuntu: insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format
5:09AM 0 ooh323c and calls to pri
5:06AM 0 RxFax dropping line
4:41AM 1 fax - conversion problem
4:14AM 1 integrating asterisk smoothly
3:37AM 0 AstBill- with many new features Released
3:13AM 1 AVM B1
12:39AM 0 Solved? => Playback audio before answered by a queue member
12:35AM 4 Delayed ringing on some SIP phones
12:33AM 4 Polycom MWI
Sunday October 16 2005
11:46PM 1 Newbi stating question
10:57PM 2 Modifying Voicemail App
9:22PM 4 Cannot telnet to port 5038 on asterisk
9:20PM 0 queue to queue to vmail
9:01PM 0 queue to queue to vmail failover
8:52PM 0 failover
8:35PM 1 Need language variable to user account
8:25PM 2 Can you use Polycom 500 with PoE Switch?
6:49PM 0 Playback audio before answered by a queue member
3:55PM 1 Restricting registration for peer '611' to 60 seconds (requested 1200)
3:44PM 1 Can Asterisk "proxy" a SIP phone to make it look like a Cisco skinny softphone?
1:51PM 1 No Audio from Console but mpg123 from shellworksfine.
1:30PM 0 Call to all Astricon attendee's!!!!
12:21PM 2 huge problem compiling * on gcc4.x (SUSE 10.0)
11:43AM 2 Pass variable to context (NOT macro)
10:43AM 1 iax invtation problem
10:38AM 0 Job Offer for working at Madrid, Spain, on Asterisk/SER related issues
10:30AM 1 Routing landline calls to asterisk.
10:21AM 2 Looking for advanced consultant services
9:22AM 1 Incoming SIP connection
8:17AM 3 Dial plan questions
7:20AM 0 No of simultaneous calls in asterisk
6:44AM 0 chan_capi and AVM FritzCard PCI
6:20AM 3 Asterisk and Fedora
4:50AM 0 CDMA USB phone for Linux?
4:29AM 2 No voice - one way - both ways
2:41AM 0 procees
1:41AM 0 IPManager PBX Features
Saturday October 15 2005
9:42PM 2 What would cause a high memory usage in pbx_spool.c ?
8:11PM 0 Problem inAresk GUI installation
7:04PM 6 ACD calls to busy agents
7:00PM 1 Looking for Info on OH323
6:31PM 1 No Audio from Console but mpg123 from shell worksfine.
5:53PM 4 Voicemail 2
3:19PM 3 res_perl - Compiling error
11:58AM 7 You ASKED for an Asterisk book, you GOT an Asterisk book!
9:33AM 0 Planet Vip-150T
7:32AM 1 Maintenance panel
7:21AM 1 Problem with '#' key recognition
5:29AM 4 Quad BRI with Fedora, anyone?
2:38AM 1 Hints and Call Waiting
2:06AM 0 Disconnecting after 1 min while Communicating Clarent class 5 call manager
Friday October 14 2005
5:40PM 3 Problem with compiling spandsp
5:38PM 3 Callerid on t1 lines
4:56PM 1 match a set of numbers in GoToIf against a variable
2:00PM 0 INFO Duration=250
1:32PM 1 2 POTS to
1:00PM 1 Outbound registration expirey
12:42PM 3 Busy not jumping n + 101 anymore
12:37PM 0 soxmix generating mute files
12:22PM 1 How to rewrite a CALLERID on outgoing calls
10:49AM 1 Does anyone Know if tha avaya 4621 IP phone work wiht asteisk?
10:31AM 1 RE: Asterisk-Users Digest, Vol 15, Issue 85
10:22AM 0 warning message when reloading
7:58AM 2 Asterisk/Cisco Call Manager 3.3
7:55AM 0 Don't know what to do if second ROSE componentis of type 0x6
7:51AM 0 multi languages
7:14AM 0 No Audio from Console but mpg123 from shell works fine.
7:12AM 2 "Please Press Any Key to Accept a Call"
7:11AM 0 IAX or IAX2 ? [SOLVED]
7:06AM 0 Sending ANI over SIP
6:48AM 0 DTMF tones not working with SIP
6:35AM 1 Voicemail -> new feature request
6:12AM 0 IAXy Port number. Repost
5:59AM 1 Problem with two hfc-s cards
5:56AM 2 T1/E1 Cards
4:54AM 0 Asterisk IAX config user
3:57AM 1 Incoming call problem - ringing SIP devices report busy
3:06AM 0 Which H323 module to go for?
2:31AM 1 Access to trunks
2:15AM 5 Reset telephone IP PHONE 106
1:15AM 4 [ISDN] Problem: Device '/dev/ttyI1' lacking dialtone
1:07AM 1 SPA-3000 Disconnect tone detection in Spain, Peru and Colombia ?
12:42AM 5 sip accounts
Thursday October 13 2005
11:30PM 0 Sound too loud (saturated). How to change?
11:08PM 0 Call transfer.
10:11PM 2 Enum parse errors
9:45PM 1 TDM04b to SIP extension not ringing (sip to sipworks fine) - resolved but why?
8:12PM 0 TDM04b to SIP extension not ringing (sip to sip works fine)
6:18PM 2 Incomming call line identification (NOT CallerID)
5:45PM 2 what should i select ??????????
5:02PM 2 ztdummy build problems
4:47PM 0 Re: call waiting not working on PAP2 (Andy Kuo)
3:28PM 1 call waiting not working on PAP2
2:37PM 0 calls not ringing
2:34PM 0 Polycom Button Remapping: Part 2
2:11PM 1 New Bug Marshal
12:25PM 2 Sample cisco config for cisco 7206
12:04PM 2 DID on analog line
9:50AM 0 Pose your Sangoma Questions
9:18AM 0 sip channels marked with SIP_NEEDDESTROY but not being removed
9:13AM 0 PRI stopped accepting calls
9:11AM 2 PRI calls to Automated Attendants Dropped
9:00AM 0 R: PA168S/AT320P
8:54AM 0 Not ringing on incoming callls
8:18AM 0 fax consulting
7:37AM 0 fax consult
7:33AM 0 CallerID detection problem
7:29AM 0 RE: Wanting to Make a PocketPC have asecureConnection to asterisk server
7:29AM 0 Moscow Dids
7:23AM 0 Impport script for upgrading to 1.2 SQL Realtime?
7:20AM 1 Noob help with IAX
7:11AM 0 which voip fone will be better
6:34AM 2 PA168S/AT320P
6:12AM 0 [ SOLVED ] ISDN problem: lacking dialtone
6:12AM 1 link quality monitor
6:00AM 0 PickUpChan and Intercept
5:33AM 0 polycom soundpoint ip600 problem
4:58AM 2 Starting simple switch from an extension?
4:55AM 1 IAXy Port number
4:37AM 1 SetCallerID Problem
4:23AM 3 IAX ATA
4:17AM 0 sangoma a104 cards and ss7 signaling
4:07AM 1 Music on hold disappears for Dial(, m) when calling outside numbers
3:52AM 1 USB phone for Linux?
3:40AM 0 pbx_spool Call failed to go through
2:31AM 1 AGI Variable problem
2:16AM 1 TDM400P off-hook detection problem
12:23AM 0 Reset IP PHONE 106
Wednesday October 12 2005
11:01PM 2 Broadvoice Outages?
10:56PM 1 Integrated T1
9:42PM 1 SIP to SIP no audio help
9:18PM 2 Wanting to Make a PocketPC have a secure Connection to asterisk server
9:17PM 2 How can I use different languages (Chinese, Cantoneese)?
7:08PM 3 New Application: Broadcast
5:35PM 0 Notice message meaning for C7960?
3:28PM 1 ASTCC and Asterisk 1.2?
3:11PM 1 TDM04B card with only 3 lines connected using chanisavail
2:39PM 2 Maximum retries exceeded on call.
1:59PM 1 MWI integration between Asterisk and Meridian
1:07PM 2 which hardware should i use??????????
12:21PM 0 RE: faxing to/from asterisk - new
12:19PM 0 Is it possible to listen and respond on more than one IAX port?
12:17PM 2 Canadian Association of VoIP Providers
11:48AM 0 Feature codes work on SIP phone but not analog?
11:43AM 1 send Q931 information element keypadfacility ?!
11:31AM 8 SIP behind NAT to pub Asterisk, best solution?
11:27AM 1 Bulk Buys/Group Buys
11:20AM 1 displaying a message on the Snom 320 using sipsak
10:30AM 5 ACD/queues question
10:11AM 0 sound very loud (saturated) through IAX2 and SIP
9:57AM 3 AGI and set_callerid for number and name
9:44AM 1 Problem with PRI and Ericsson AXE 10
9:05AM 3 Calibrating both RX and TX gain?
8:53AM 2 Polycom: Button Remapping, HELP!
8:40AM 1 Sangoma FXO/FXS cards?
8:28AM 2 Patton SmartNode
7:09AM 0 Zaptel Debug: "T1: Lost our place, resyncing "
6:52AM 0 X100P callerid ETSI - caller*ID failed checksum
6:26AM 0 unloading TE110P bristuffed module cause kernelpanic
5:58AM 0 Second Request for help: hardware requirements
5:32AM 0 arcaplex / horizon isdn and analog multiplex
5:21AM 8 parameters documentation
5:19AM 2 SNOM 360 Unknown SIP command 'PUBLISH'
2:55AM 3 E400P vs te410p vs te411p
2:54AM 2 asterisk log
2:41AM 1 MWI for endpoints not registered at Asterisk
2:24AM 1 detect SIP phone availability before dialing
2:21AM 5 delays with IAX2 and Meetme
1:59AM 2 Asterisk logo
1:57AM 0 Voicemail recording volume control
1:56AM 0 Outgoing Provider Recommendations
1:50AM 2 Modifying cmd VoicemailMain
1:27AM 2 Monitor DTMF problems
1:04AM 1 unloading TE110P bristuffed module cause kernel panic
Tuesday October 11 2005
10:44PM 2 error message when accessing voicemail
10:00PM 2 Large country based dialplan
9:46PM 1 supermicro with asterisk and tdm cards
8:43PM 3 Dual PRI fail over
8:06PM 0 Are Digium serial numbers recorded into boards and modules?
7:14PM 1 Problem w/ Asterisk hanging when caller hangs up in voicemail
6:37PM 1 Areski Calling Card GUI
5:39PM 5 help with broken voicemail
5:22PM 1 Problems with Wait & SIP 486 "DND"
4:59PM 1 migrated to new ver on voip connection vs1 server voicemail problems
3:42PM 1 Wrong caller id in CDR
2:59PM 2 Question on hardware requirements when not using a land-line
2:45PM 1 Voicemail Passwords and RealTime
2:27PM 0 Realtime + OS X = anyone got it working?
12:21PM 1 call to a particular 800 numbernevershowsanswered on Zap channel
11:50AM 0 call to a particular 800 number nevershowsanswered on Zap channel
10:14AM 4 New Sangoma AA Series?
8:49AM 6 PRI echo issues: solvable?
8:47AM 0 TTL
8:45AM 0 FXO tune
8:06AM 1 callerid validation and expression
8:01AM 1 DIDx error
7:41AM 0 Not a local SIP domain
7:32AM 5 Which asterisk-friendly cards are fax-capable?
7:09AM 0 Echo on SIP Side?
6:54AM 0 DID in Bolivia
6:51AM 2 Re: [Chan-sccp-users] Need help with hint and callgroup
6:43AM 2 Voicemail while in queue.
6:30AM 2 nat and wandering phones
5:42AM 3 Asterisk and Mitel SX 200 Slip and Frame Err ors causing Major Ala rms
5:12AM 1 callerid validation and expression parser problems on Solaris 10
3:22AM 1 asterisk to asterisk using mgcp
1:35AM 2 IAX or IAX2 ?
1:08AM 2 CallerID for BSNL (India) phones
Monday October 10 2005
10:12PM 3 country code list
9:50PM 0 Queue delay
9:38PM 5 Soekris and Asterisk
6:51PM 0 cannot load new wctdm module
6:37PM 2 Astricon Podcasts?
6:23PM 2 DTMF detection
5:50PM 2 Errors with new fetched Asterisk cvs
4:25PM 0 Realtime Extensions - DB concepts
2:55PM 2 enable mysql in asterisk
2:52PM 1 Need help with hint and call group
2:48PM 0 Asterisk behaving wierd!!
2:37PM 2 Throroughly confused about SetCallerID
2:30PM 2 Asterisk and Mitel SX 200 Slip and Frame Errors causing Major Ala rms
1:22PM 1 AAH. only 1 ring
12:44PM 2 Beronet app_saynumber-beta-rc1
12:33PM 0 Problem with Oh323 on 1.2Beta on CENTOS 3.5
12:25PM 2 What is this error? Is there a bug?
11:39AM 1 CallerID Outbound on VOXEE
11:18AM 0 Faking it: queue_log and addQueueMember
10:57AM 3 Help, please help -- IAX2 softphone to server on LAN
10:34AM 1 Realtime regseconds update
10:18AM 0 CDR problem with DST Channel
10:15AM 1 2 line SIP ATAs with Asterisk using RealTime
10:08AM 1 Hang up call
9:13AM 1 Incoming SIP getting in, but not ringing.
9:08AM 1 Outgoing quality
9:07AM 0 Fredericksburg ZPUG Meeting will have an Asterisk Flavor this Month
8:54AM 0 Incoming Calls causing Protocol Error (6)
8:10AM 11 Open Source Content Management System - Joomla
7:32AM 1 Multitenant Call Center Setup
7:31AM 2 DTMF Question (misunderstood '*' button)
7:30AM 3 Billing/SPA-841/CDR Log
6:56AM 4 sip register incoming call contexts?
6:45AM 0 does tellular cell phones support answer switching
6:18AM 1 Bandwidth usage for codecs
5:56AM 1 customize the pager email
5:34AM 1 [Fwd: Libpri/chan_zap problems?]
5:12AM 2 My contribution to the issue of code- reversal
4:34AM 0 Dial plan logic documentation?
4:28AM 6 telephony that "just works"
2:25AM 2 TDM400 not working
1:04AM 2 AVM Fritz! + chan_capi + mISDN + PTP
12:20AM 0 where can be find zaptel cvs change log ?
12:03AM 0 Re: faxing to/from asterisk - new scripts
Sunday October 9 2005
11:45PM 1 Problem setting SIP incoming/outgoing
7:36PM 2 Clicks, pops and noise
6:11PM 0 The VoIP Connection has $$$ opportunities for Asterisk experts
5:20PM 1 where to find an asteriak Voice Mail User Manual
4:37PM 1 MPG123 with Asterisk on debian (one of our interesting experiences)
4:21PM 2 Link
3:42PM 8 Zaptel Line Build Out
3:21PM 0 IVR pausing before dialing ext
1:56PM 0 Realm Auth = No?
1:37PM 0 app_txfax not running
1:28PM 4 Avaya 4620/4640 SIP firmware
12:07PM 0 Problem with disable call transfer
10:51AM 0 Problem logging in using domain
10:34AM 1 Dial/goto extension from CLI or BASH script
9:20AM 0 Asterisk, H.323 & Cisco uBR900
8:53AM 0 Incoming Caller ID
8:15AM 2 compiling asterisk on SuSE Linux 9.3 fails: illegal instruction
7:39AM 1 Asterisk, VoiceTronix & UK Caller ID
7:14AM 0 mail2fax and fax2mail updated
5:51AM 0 who has implemented callback function?
4:58AM 0 Anyone Know That !!!
4:32AM 4 *8 and group pickup not working
Saturday October 8 2005
8:44PM 2 Configuring TDM400 in Australia
8:23PM 0 Regcontext/regexten broken??
7:42PM 0 ATA does not register
5:39PM 1 Cannot dial SIP via asterisk
4:57PM 1 Does anyone know what this means
4:15PM 4 Asterisk Log Color Coding
3:01PM 1 How to check what codec translations are in use in a call?
2:12PM 1 Outgoing call: hangup after answer
9:49AM 0 Asterisk on Solaris SPARC
8:52AM 1 need help-can't not register to asterisk from softphone
7:27AM 2 No incoming calls from chan_capi 0.6
3:03AM 1 Extension bracket matching broken in CVS
1:09AM 0 Re: Asterisk-Users Digest, Vol 15, Issue 28
Friday October 7 2005
10:32PM 0 ParkAndAnnounce Question
10:32PM 3 Digium G.729 codec modules updated
9:18PM 0 Prelude to Comfort Noise Generation support on Asterisk
6:58PM 0 * cell phone problem
5:46PM 0 How to speech a text file with festival
5:05PM 0 Asterisk going ahead on a busy call
4:39PM 0 Pingtel applications
4:06PM 1 ASTCC -- semantic note of 'callstart' in cdrs?
3:14PM 0 IBM work with a TE405P Digium card?
2:45PM 3 hardware echo cancellation. sangoma?
2:39PM 1 How do you verify remote registrations
2:37PM 3 call to a particular 800 number never showsanswered on Zap channel
2:28PM 1 PSGw 2.0 Skype<>SIP gateway
2:16PM 0 Variable for codec used?
1:33PM 0 AudioCodes MP-104 FXS
1:14PM 0 BBEdit Language Module for asterisk?
1:10PM 0 'ztcfg -s' causes system hang
12:58PM 0 Asterisk to CCM Message Waiting Indicator
12:54PM 2 call to a particular 800 number never shows answered on Zap channel
12:41PM 3 TDM02B card difficulties
11:03AM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday 10/8/2005
10:46AM 0 asterisk install [colinux]
9:33AM 1 Outbound Mediatrix 1204.
9:32AM 0 txfax (app_txfax) sending issue
9:15AM 1 'make rpm' problem
9:14AM 3 wifi phones - desk
9:13AM 0 []Ouch ... error while writing audio data: : Broken pipe
8:04AM 0 Issue with terra-call today
7:54AM 3 RE: faxing to/from asterisk - new scripts
7:50AM 2 Teliax users, g729 question
6:12AM 1 Distorted VM with iax2 with ilbc and jitterbuffer - bug?
5:24AM 1 overlap zaphfc - dialtone
5:20AM 1 S0 - T0 interfaces question
3:52AM 2 tx(rx)_fax for *-1.2.0.beta
3:05AM 0 Asterisk and chan-spy problems
2:35AM 0 Incoming sip
2:29AM 2 Asterisk on dynamic extrenal IP behind a nat router.
2:29AM 1 Echo cancel on HFC-S cards and CIDNum setting on outgoing calls
2:17AM 3 Where to get the latest SIP Firmware for Polycom Phones?
12:58AM 1 Noise using TE410P & Rhino Channel Bank
Thursday October 6 2005
11:26PM 0 How to send error codes to connected phone?
10:51PM 4 Asterisk PBX in Debian
10:12PM 3 Asterisk and firewall
10:06PM 0 Issue with trunking
8:52PM 3 WCFXO and T1 PRI Card?
8:36PM 2 Latency on bridged PRI calls
7:20PM 1 Outbound CallerID Teliax
4:37PM 1 Snom 360 Phones - Administrator/User Feedback
4:32PM 1 Billing: amaflags and accountcode
4:00PM 2 SIP Dialler
3:40PM 1 7960g 2nd ethernet port cycles on/off
3:37PM 0 chan_capi configuration with AVM C2 card
3:22PM 2 how do I know what codec is being used
3:13PM 1 TDM400 takes Zap/4 line off hook
2:13PM 0 How do I using Hangup?
1:49PM 0 Codec issue? Dropping incompatible voice frame ...
1:27PM 0 transcode or passthrough
1:07PM 1 Results of an incorrect crossover pinout??
12:57PM 0 Fw: Re: Re: inter Asterisk trunking IAX /IAX2
12:21PM 0 Vodavi PRI issues?
11:44AM 0 How can I log call forwards?
9:37AM 1 How can I override *67?
9:05AM 1 Asterisk::AGI Alternate Download
8:23AM 0 Whats the channel name?
8:17AM 2 Asterisk/Debian/VIA EPIA M Howto
7:36AM 1 ast_fax with sendmail
7:21AM 1 adding new indication tones
7:19AM 2 Mediatrix 1204 and Asterisk
7:17AM 0 SIP Realtime Question
6:29AM 1 IP Multimedia Subsystems (IMS)
5:56AM 1 number of did numbers in one channnel?
5:56AM 0 CDR data parsing
5:47AM 0 more calls
5:44AM 1 Fwd: ASTCC - INUSE Flag
5:36AM 2 How do I add a list of cidnames to the asterisk database in one shot ?
5:10AM 1 Selecting outgoing trunk based on extension number
3:29AM 0 AAH
2:52AM 0 SV: Incoming call
2:43AM 14
2:22AM 0 getting called number from a zap channel
1:43AM 1 [help!] asterisk 1.2 beta
1:05AM 1 Incoming call
12:50AM 1 How to Forcing Call Disconnect?
12:28AM 1 Changing IP on Asterisk
Wednesday October 5 2005
11:06PM 3 SIP Attended Transfer using REFER and Replaces: headers
10:41PM 0 IAX2 calls dropped (Max retries)
9:37PM 3 CVS HEAD and Hints
9:36PM 1 IAX2 + Jitter Buffer
9:35PM 1 newbie asterisk build
8:08PM 1 Delay before dialplan is launched?
7:21PM 0 Asterisk Nat solution for such scenario?
6:38PM 1 Clearing Caller-ID from Zaptel Channels
6:04PM 1 Help! Extensions
5:54PM 4 dropped calls when g729 is used on sip leg
4:51PM 2 inter Asterisk trunking IAX /IAX2
4:45PM 0 Unknown or blocked ID now shows up as "asterisk"
4:38PM 1 Extension Always Goes to VoiceMail
4:25PM 2 Sipura SPA-3000 setup in Brazil
3:54PM 1 "AST_LIST_REMOVE" passed 4 arguments but takes just 3
3:40PM 1 Auto-assign CallerID for all my FXS Interfaces
2:53PM 1 DLINK DVG-3004S
2:12PM 1 Config PolyCom SoundStation 4000 help
1:51PM 0 CVS won't compile: res_odbc error
1:50PM 0 Voicemail email issues
1:38PM 1 Attempted to delete none, xistent schedule entry 1! ??
1:25PM 2 Define variable in sip.conf
1:23PM 1 Caching DTMF tones for get_data AGI?
12:46PM 5 Voicemailmain automatic extension detection?
12:09PM 0 Please, help test asynchronous generation patch for inclusion in version 1.2
11:56AM 1 New astGUIclient/VICIDIAL version released 1.1.7
10:35AM 1 What the heck? Sprint sues Vonage
10:28AM 3 IPComms Setup
9:49AM 1 TDMOE Badness in kernel...
9:24AM 2 Sipura Adapter SPA-2002
8:46AM 2 Zaptel tone description
8:10AM 0 Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o memory leak when using call files ?
7:50AM 1 compiling astrisk
6:02AM 2 TE411P and TE406P stability
5:15AM 2 From Database, PHP-Webinterface -> TO flatfileconfiguration
4:52AM 0 Unwieldy outbound macro
4:30AM 0 agi-test.agi question - wierd results
2:31AM 2 can't run app_txfax
2:21AM 0 call transfer problem - something strange
2:14AM 1 Configuration QuadBRI Junghanns
2:02AM 1 how can i let the user in 1th Asterisk can call the user in 2nd Asterisk?
12:58AM 1 How to enter digits using sjphone
12:55AM 1 Easy SIP.conf questien. Incomming call context?
12:39AM 2 Intel Pentium Celeron
Tuesday October 4 2005
7:45PM 2 Hardware vs. Network Inputs
6:27PM 1 Digium hardware echo canceller, zapata.conf settings?
4:35PM 1 Fw: trunking IAX2
3:51PM 3 Transfer directly to voicemail (blind transfer)?
3:23PM 1 Forcing Codec Usage
2:52PM 1 Recommendations for * monitoring?
2:43PM 12 Sprint Nextel sueing over VoIP patents
2:30PM 4 Emergency calls - forcing through on channel
1:43PM 1 Polycom config and DTMF problems
1:14PM 1 Firefly 2 third-party version?
12:49PM 0 DTMF heard at end of AGI Record File
11:33AM 5 PBX 'Personalities' ?
11:25AM 0 compile loop?
11:22AM 0 Connecting two asterisk servers using IAX
11:18AM 0 check_asterisk commands
11:12AM 1 Hanging up on VoiceMailMain w/out putting in password causes call lockup
10:55AM 0 RECAP: 3?
10:26AM 0 forward iax extension
10:05AM 0 app_rxfax module won't load
9:30AM 2 DPH-140S SIP Phone oddities
9:11AM 0 Speed Up SayDigits?
9:03AM 3 ADSI -- is it dead? Worth bothering with?
8:59AM 1 FXS static and noise problem
8:48AM 0 Asterisk w/ BRIstuff compile error
8:32AM 1 Can't compile ast_rxfax with Asterisk 1.2.1b
8:24AM 3 Polycom 501: takes calls, but fast busy on dial out?
8:07AM 1 Announcing – Voice over IP Directory Services (
7:54AM 1 Number Restriction
7:53AM 1 Seeking Asterisk Solution For mid sized corp.
7:52AM 1 SNOM Subscribe/Notify
7:44AM 2 Call-in/Call-out
7:39AM 1 IODBC instead of UNIXODBC
7:37AM 3 Echo Canceling
7:30AM 0 Auto attendant
7:02AM 0 Dynamic feature support recently added to CVS HEAD
6:45AM 0 Error: -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from
6:41AM 0 can't reject call using macro-screen
6:35AM 1 Asterisk Calling Card Platform
6:14AM 1 TDM versions question
5:56AM 0 Three-way calling over SIP and IAX using softphone
5:32AM 2 Quad PRI Problems
3:55AM 0 CallerID octoBRI connected on voxtream parlay i60
2:43AM 1 Asterisk forwarding SIP with Remote-Party-ID
2:28AM 3 Asterisk as H323 gateway
1:45AM 1 Dial pattern sort order
12:25AM 3 Outgoing busy
Monday October 3 2005
11:28PM 2 Voice Quality bad on one side of Frame Relay
10:30PM 3 DIAX not working properly
10:25PM 0 Weird Problem - SIP/POLYCOM/DTMF
9:09PM 4 Snom phones?
7:42PM 0 Inter Asterisk IAX2
7:32PM 0 RTP timing problems? Here's patch...
6:29PM 2 Hang-up Detect - Yet Again
5:40PM 0 Ticking sound in wildcard tdm400p
5:01PM 2 Debian sarge package for 1.2beta1?
3:48PM 2 asterisk, cisco 3640's and DIDs...
3:14PM 1 Realtime and voicemail: request to find out if I'm crazy
3:03PM 0 TDM400P recognised as "Network controller: Unknowndevice"
2:55PM 1 Direct Dial In - second try
2:50PM 3 FreeTDS 0.63
2:42PM 2 TDM400P recognised as "Network controller: Unknown device"
1:11PM 2 sip phones on x86_64
12:56PM 0 Asterisk Ignoring [User] in SIP.CONF
12:13PM 2 Real Life FAX sending receiving
11:27AM 2 Asterisk 1.0.8 and TDM stop acking inbound calls?
11:16AM 0 Console sound output -- shuts off when call from console answered
11:10AM 1 Compiling SpanDSP
10:54AM 0 Need help with Cisco 7960
10:48AM 0 Hangup not detected on callback
10:47AM 0 SIP qualify question.
10:47AM 0 Which hardware configuration? How would this work?
9:56AM 0 Very cheap IP GSM Gateway: Will this work?
9:56AM 0 TDMoE help with Alarms...
9:43AM 1 suse 9.3 pro asterisk install from source problem
9:12AM 1 SIP-CPE Gateway
8:38AM 1 no audio on fxo line
8:19AM 4 R: Diva
8:16AM 0 asterisk behind Linux iptables with masquerading and forwarding on
7:03AM 1 R: codec g723 on Via C3
7:01AM 1 Problem with configuration of Quintum AX with Asterisk
6:38AM 0 fc4 + iax + trunking
6:27AM 1 [Fwd: Eicon Diva 2.01 S/T PCI quality problems]
6:18AM 0 How to establish ISDN port Up
4:50AM 4 SPA-3000 generating one-ring calls
4:05AM 3 codec g723 on Via C3
3:50AM 0 US tollfree DID request
12:46AM 1 *** Community alert :: Do you have open bugs in the bug tracker?
Sunday October 2 2005
9:40PM 0 kjournald and zttest results
8:59PM 0 zttool improvement: histogram
8:18PM 0 is a dual 1.5Ghz server better than a single3Ghz for a 100 Iax users asterisk server
7:59PM 1 IAX2 Group dialing.... Is there something in the horizon?
7:31PM 1 Audiocodes MP108
7:13PM 1 analog phone connects to zaptel fxoks is beeping
3:29PM 3 What does the error "stale nonce' mean?
2:51PM 0 Console Sound: Cuts out, Comes back after restart
2:35PM 0 Avaya 4620 hardphone
2:22PM 0 where can I find this property: answeronpolarityswitch
2:21PM 1 Asterisk-RealTime: sip_friends and register => user:pass@host
1:42PM 0 iax invitation problem
9:22AM 1 Adit 600 FXO card sound quality
7:12AM 3 [Sorta OT] Eicon DIVA with asterisk@home
6:09AM 3 Fw: Channel Banks, what are they for?
4:41AM 1 Adding Voicemail box
2:58AM 0 Grandstream GXP2000
1:24AM 2 DB Function in version 1.2
12:53AM 4 IBM tts engine integration
12:52AM 0 Outgoing rout dialpattern
Saturday October 1 2005
10:58PM 0 sound file installation problem
10:31PM 0 chan_zap vs. Panasonic DTMF integration
6:24PM 1 SIP 400 Bad Request from Cisco 7960/7940
6:16PM 0 Sourcing Eicon Diva V-4BRI/QuadBRI cards in Australia
6:15PM 1 Problem with VM Distribution Groups
3:39PM 0 Hangup half a call?
3:38PM 1 Can't compile zaptel (CVS Head) on Debian
11:41AM 3 Adding Cepstral to Asterisk
10:53AM 0 Callcenter and Softphone hanging
10:29AM 2 Remote call pick-up
10:09AM 1 Swap between callers
9:23AM 0 Faxdetection in IAX? (Missing audio samples)
8:39AM 1 Compiling Zaptel on EM64T machine
8:32AM 2 Asterisk - HEAD , SpanDSP, app_rx/tx Fax..what versions do you have?
8:20AM 0 chan_zap.c: Ring/Off-hook in strange state 6 on channel 1
7:24AM 0 asterisk-oh323-0.6.7
7:17AM 1 OT: RHEL / CentOS Enable APIC
7:05AM 0 How can I tranfer a call form one SIP phone to other during the call (unattended transfer)
6:42AM 0 Now can I tranfer call form one SIP phone to other during call (unattended transfer)
6:36AM 1 error on loading zaptel module
6:33AM 0 How to create IVR system using *
6:32AM 7 Updated presentation of Asterisk 1.2
4:52AM 0 Developer help needed
4:28AM 2 Calls between SIP and IAX
2:15AM 0 VoiceGateway Design - Request for comments/suggestions
12:31AM 0 how to backup asterisk installation for upgrade