asterisk users - Jun 2005

Thursday June 30 2005
11:37PM 0 Got this error after my installation when i do ztcfg -vv
11:35PM 1 Caller ID problem..
11:23PM 1 Outbound answer on TDM400P
11:23PM 1 IAX DTMF Problem...
10:58PM 0 Flash Zap Channel
9:55PM 0 Re: Asterisk-Users Digest, Vol 11, Issue 181
8:58PM 1 No BUSY on PRI
6:31PM 0 chan_oh323.c not compiling (cid not found)
4:56PM 3 Linux Firewall Question
3:30PM 3 Computer to use
3:21PM 0 Cell Phones reporting internation calls
2:53PM 2 New Setup with Analog Phone lines
1:54PM 1 Flash and zap and # key
1:34PM 5 wi-fi phone advice
1:20PM 2 Dial Option A(file.gsm)
11:55AM 0 callprogress and queues
11:42AM 0 Asterisk & mp3 playback while dialing
11:37AM 1 spandsp fax out fails
11:25AM 7 passing through MWI info from SBC
11:09AM 2 ser --> sip.conf --->extensions.conf, variable context
9:55AM 5 Failover question
9:48AM 3 Trying to do very simple Zaptel Config. NO LUCK!
9:43AM 1 Pickup pin
9:22AM 0 Strange dropped calls
9:22AM 3 GUI that supports virtual PBX's/users
8:27AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 202
7:50AM 0 Audiocodes MP-1xx cheat sheet
7:42AM 2 Asterisk failover solution
7:22AM 5 Logrotate
7:22AM 1 Cisco Voip Question
6:54AM 4 [Fwd: Asterisk Balancing solution]
6:34AM 0 Daily Asterisk News
6:05AM 2 [Asterisk-Dev] Developing an Application in Asterisk
5:43AM 1 Developing an application in Asterisk.
4:59AM 1 [Asterisk-Dev] C Code of Asterisk
3:43AM 1 Master.csv and MYSQL
3:32AM 1 problems in dialing in routes patterns
2:55AM 7 Voicemail => SMS
2:41AM 1 Fw: Multiple Quad Bri card
2:38AM 4 Getting FOP working with ICD?
12:57AM 3 Resolving groupcalls
12:51AM 1 Do any ITSPs support Speex?
12:29AM 3 AMP - recording call
12:05AM 3 R: Music oh hold
12:01AM 0 Sipura 3k answers then immediate busy signal
Wednesday June 29 2005
11:35PM 1 GnuGK and Asterisk
10:28PM 1 Can't build cdr_addon_mysql.
10:28PM 0 Calls Dropping
10:01PM 1 Asterisk/SER/Call Manager
10:00PM 0 failed IAX, SIP registration - email notification
8:55PM 2 X100P connected as extension to Panasonic 616 EASA-PHONE
8:50PM 2 Problems with zaptel and voice prompts/voicemail
8:36PM 4 Quality of provider: VocTel
8:32PM 0 RE: Asterisk-Users Digest, Vol 11, Issue 198
6:42PM 0 ASTCC Issues - Resolved
5:41PM 11 Asterisk@Home Ver 1.2 Whats new?
5:33PM 1 Kind of Computer to use
5:19PM 0 Access asterisk features from analog phone during call
5:08PM 0 Server hanging
4:33PM 3 UK SIP Provider
3:40PM 2 New Asterisk documentation
3:25PM 1 Problems connecting to and from my Asterisk server :(
2:25PM 0 Parking Position
1:56PM 3 hidecallerid on analog line
1:16PM 5 Extension Matching.
12:53PM 2 Multiple Timezones with Asterisk
12:47PM 1 Welcome
12:39PM 1 AMP or Asterisk
12:27PM 1 Dial ZAP Problem
12:23PM 0 PRI: XXX Missing handling for mandatory IE 12 (cs0, Connected Number) XXX
11:48AM 2 Polycom SoundPoint 501 Problem
10:35AM 4 Music oh hold
10:26AM 0 (no subject)
10:21AM 1 Sangoma and quad card hang up problems
9:47AM 0 dtmfmode=inband still broken in *-1.0.7
8:33AM 10 Setting Caller ID after Dial
8:13AM 1 OrderlyQ installations?
7:49AM 1 Teliax Problems
7:40AM 5 Problems with OR Logic in the GotoIf Statement
7:28AM 2 Play an announcement to the CALLING party
7:12AM 2 timeout on incoming PRI call
7:11AM 0 How to fetch a call not in the same callgroup
6:17AM 1 Machine Sizing
6:13AM 4 PRI got event: HDLC Abort (6) on Primary D-channel of span 1
6:02AM 2 Asterisk LAMP Developer
4:54AM 1 Equipment for small office setup
4:40AM 0 ast_rtp_read: Unknown RTP codec 100 received21 when receiving fax
4:40AM 2 Recommend against Teliax as primary ITSP
4:18AM 1 App_conference in dial plan?
4:14AM 1 CAPI and Caller ID name not showing.
2:31AM 1 GSM Hunting
2:15AM 0 chan_capi-cm-0.5.3 fixup release
1:48AM 0 CallerID Bug?
Tuesday June 28 2005
11:20PM 1 Hop-On WIFI Phone MSRP $40
10:55PM 1 Fw: Shoutcast Music On Hold problems?
10:39PM 1 audiocodes
9:46PM 2 AMP/A@H (asterisk at home) custom incoming routing
8:51PM 4 Anyone using SipP to produce RTP load?
7:43PM 1 Red Hat Enterprise 3.0 issue
7:08PM 0 manager api call number, pause, dial ext
6:48PM 0 ASDI Programming through an ATA/SIP device?
4:32PM 2 SIP Phone Config Generator
4:14PM 1 Net2Phone equipment and different VOIP providers
3:45PM 1 TDM04B Echo on Only One Channel
3:42PM 2 Asterisk RSS list feeder ready
3:04PM 0 BRIstuff/OctoBRI problem: Ring requested on unconfigured channel 255/255 span 5
2:49PM 3 Asterisk with Lucent TNT echo
2:15PM 0 Inter-Tel 8662 configuration problem.
2:04PM 2 Trying to get *8 call pickup to work
1:49PM 0 Dial application timeout
12:02PM 1 VoipJet TOS (was Teliax and also LiveVoip)
12:00PM 1 Linksys WRT54GP2-NA settings for performance and low bandwidth?
11:53AM 4 Revision I Board TDM04b
11:52AM 0 Mitel SX2000 Integration
11:36AM 2 Asterisk Realtime and ODBC
11:34AM 1 enabling stun on asterisk?
11:24AM 1 Voicemail max time length
10:53AM 1 Voice Mail hangup on not messages
10:37AM 1 initial setup: problem
10:07AM 0 CPU load about at max when it should be idle.
10:06AM 1 problems with chan_capi 0.3.5 , divactrl, eicon diva server, and kernel 2.6.10/2.6.12
10:02AM 1 Correction to Janghanns BRI problem
9:55AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 181
9:44AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 183
9:22AM 2 Junghanns 4 port BRI problem
9:21AM 1 This weeks Developer meeting
9:19AM 0 Speech driven crm apps
9:16AM 0 Re: teliax [Was: LiveVoip is Bankrupt]
9:14AM 1 Unable to connect to remote asterisk
9:03AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 183
9:00AM 0 Asterisk dies with Meetme
8:35AM 1 Hangup detection on Panasonic KXTD816
8:29AM 1 TDM400
8:25AM 1 list Searchability
8:16AM 2 MeetMe application in Asterisk V1.07
7:59AM 4 How do you handle NAT?
7:43AM 1 ClueCon, Vote?
7:35AM 1 Re: teliax [Was: LiveVoip is Bankrupt]
7:13AM 0 help, switch off NOTICE in console
6:53AM 0 Asterisk & SpanDSP -> problems by sending a fax
6:16AM 4 Using Conferencing and Meetme
5:28AM 2 Using asterisk as Quality Monitoring Platform?
2:35AM 0 Spinlock with ZAPTEL
2:35AM 0 GSM/PSTN Gateway function of DIAX - feedback request
2:04AM 1 simultaneus calls?
1:58AM 0 cheap HFC card on Bristuff vs cheap HFC card oni4l vs Fritz ISDN BRI card on CAPI
1:56AM 1 pbx_extension_helper: No application 'agi'
1:46AM 0 RE: [Serusers] *** SER - Asterisk
12:01AM 1 cheap HFC card on Bristuff vs cheap HFC card on i4l vs Fritz ISDN BRI card on CAPI
Monday June 27 2005
10:51PM 3 Shoutcast Music On Hold problems?
10:14PM 0 Can anyone guide me regarding h323.cong ???
9:20PM 0 Fw: shoutcast mp3 music onhold with amp portal?
9:07PM 0 How to get the outbound data of agent in queue
8:19PM 2 Eicon equipment, BRI Server or PRI?
8:13PM 1 Newbie Confusion on Call Forward and DBput/DBdel
7:33PM 0 Asterisk ended with exit status 1
6:49PM 0 Disable record busy greeting option in voicemail
6:31PM 2 is teliax down?
5:36PM 1 SixTel?
4:24PM 1 Level 3 SIP <--> asterisk
3:17PM 0 Dialogic D/300pci-E1 card
3:17PM 0 Possible bug in meetme when hangup
2:58PM 0 Just let whiners whine... Please?
2:17PM 0 Bridging and unbridging channels
2:07PM 0 RE: [Serusers] *** SER - Asterisk
1:55PM 4 LiveVoip is Bankrupt - Why this thread
1:49PM 0 how to set agent to busy when agent makes a outgoing call?
1:34PM 1 VoIP provider in Switzerland
1:19PM 1 Polycom & VPN trouble
1:09PM 0 Snom 220 Active Call Lights
1:04PM 3 AGI "say number" but in french
12:42PM 8 OT: Good soft-phone on Linux
12:17PM 0 H.323 (Asterisk@Home)
11:50AM 1 RE: [Serusers] *** SER - Asterisk
11:42AM 2 PSTN IAX Connections / Line Banks
11:00AM 0 Re: teliax [Was: LiveVoip is Bankrupt]
10:27AM 1 Re: teliax [Was: LiveVoip is Bankrupt]
9:51AM 3 Bad Bad Performance; Max 20 Calls on Quad Proc?
9:35AM 0 Re: teliax [Was: LiveVoip is Bankrupt]
9:34AM 1 Passing called number in SIP
9:31AM 1 announced transfer
9:11AM 2 DID in Western Canada
8:50AM 0 dropcount
8:43AM 1 Asterisk and conference bridging...
8:06AM 0 Re: FXO as modem (was: * fax reliability between ISDN PRI andFXS ports)
7:31AM 1 LogWatch for Asterisk
7:31AM 0 ???? WARNING[20313]: channel.c:531 ast_channel_walk_locked ????
6:52AM 2 Accessing SIP username from AGI script
6:16AM 2 Comedian Mail User Setup Prompts
6:00AM 6 TDM card and voicemail volume
5:22AM 1 TE100P
5:22AM 0 Howto answer, hold and transfer a incoming call?
4:46AM 1 Native MoH patch for 1.0.8?
4:46AM 0 Failover Design
4:34AM 0 facing troubles with routes patterns dialplan
3:35AM 3 Fw: linksys rt31p2 test case
3:15AM 2 R: zaphfc: empty HDLC frame or bad CRC received
2:34AM 0 chan_capi-cm-0.5.2 fixup release
2:22AM 1 Strange behaviour with lost internet connection
2:17AM 1 MWI
12:50AM 0 bristuff-0.2.0-RC8h does not compile
Sunday June 26 2005
11:30PM 0 No Sound at all
11:18PM 0 newbie here ...... regarding h323.conf
9:34PM 2 FXO as modem (was: * fax reliability between ISDN PRI and FXS ports)
7:34PM 0 TDM400P-04B fails after reboot
5:53PM 0 APP - ValetParking on CVS-HEAD -- instructions on its use, anyone?
5:51PM 0 Missing first second of voice on outgoing SIP/IAX calls
2:25PM 1 Changing Caller ID
12:57PM 0 Prepaid for mysql and simple auth
12:15PM 3 TDM400P (TDM02B) ceased to work...
9:44AM 0 need help for configuring voicemail with db
9:12AM 1 DID in 513 Cincinnati
8:59AM 1 CDR: source completed with sip domain
6:28AM 0 failure in writing in pattern (routes)
6:01AM 1 help regarding h323.conf
5:53AM 0 Building DUNDi?
5:33AM 0 Bug in Mailman version 2.1.5
4:59AM 2 Asterisk RealTime Voicemail
4:31AM 3 cdr and billing
12:38AM 30 LiveVoip is Bankrupt
Saturday June 25 2005
10:33PM 0 IPSwitchBoard version 0.120 released
9:27PM 2 * fax reliability between ISDN PRI and FXS ports
9:11PM 0 Everyone is busy/congested at this time
8:06PM 0 iaxy device
6:52PM 0 How to bridge 2 calls together
4:24PM 2 iaxy over the public cloud
12:47PM 6 ASTCC not billing
10:33AM 1 Looking for link.exe to compile G729 codec
10:01AM 0 SIP registration fails with realtime
9:38AM 0 help for odbc storage
8:58AM 1 callerid in forwarded call
8:11AM 3 * 1.0.8: no more reacting to callerid?
7:54AM 4 Asterisk and Cisco CallManager Integration
5:47AM 1 isdn channels busy
5:18AM 0 OH323, RxFax and codecs
2:58AM 0 FW: ZAP to SIP Dial Plan
Friday June 24 2005
7:35PM 1 Asterisk with dual WAN router
7:14PM 0 H323 with Asterisk
7:11PM 0 Running 6 copies of Asterisk on my machine
5:26PM 4 UTStarcom F1000 WiFi IP Phone Review
2:33PM 0 Help installing PyAsterisk
2:26PM 1 Qualify Frequency
2:25PM 0 Exposing Zap Channels on Server A to be UsedByServer B
1:32PM 0 UTStarCom F1000 SIP configuration
1:31PM 1 Exposing Zap Channels on Server A to be Used ByServer B
1:17PM 4 Tellabs Echo Canceller
1:14PM 0 Playtones volume control?
12:47PM 0 Help instaling PyAsterisk
11:48AM 0 New astGUIclient version released 1.1.4
11:05AM 2 Set global variables without extension..
10:39AM 1 Unable to open pseudo channel for timing... Sound may be choppy.
9:33AM 0 Distinctive Ring for Agents (Was: Re: Asterisk 1.0.8)
9:07AM 2 Exposing Zap Channels on Server A to be Used By Server B
9:06AM 0 format_base64.c released on
7:32AM 0 Extensions Puzzle: Contexts Confligting with each other.
6:44AM 0 wcte11xp hardlock problem
6:36AM 2 RTP session between two end users
6:21AM 3 SendText
6:00AM 1 Asterisk server with remote monitoringcapabilities
5:37AM 1 Errors on SuSE 9.3 default install.
5:24AM 1 Dial peer preference
4:55AM 0 How to setup two Asterisk boxes - keeping theregistration
3:39AM 1 Whole configuration for SMS
3:07AM 1 BRIstuff/QuadBRI problem: Ring requested on unconfigured channel 255/255 span 5
Thursday June 23 2005
11:34PM 0 asterisk security issue
11:28PM 1 realtime sip confusion
11:08PM 0 Requirement for internal calls
10:36PM 1 More IP address in bindaddr directive
10:05PM 0 Voicemail recording cutoff when silent for 1 second
9:08PM 0 SPA841 Utterly Horrible, are there any good stun hardphones?
6:59PM 1 HDLC abort 6 error
6:55PM 1 Asterisk & Zoom x5v 5565
6:14PM 0 Using 2 x DSL
5:02PM 0 Asterisk server with remote monitoring capab ilities
4:46PM 5 Asterisk server with remote monitoring capabilities
3:13PM 0 Automatically setting mailbox on registration of SIP device by SIP device's line #
1:46PM 4 French Audio Files
1:46PM 1 Driving direction sent to callers mobile phone viatext/sms ?
1:41PM 1 Asterisk 1.0.8 Released
12:51PM 1 *77 does not work ..
12:45PM 0 Driving direction sent to callers mobile phone via text/sms ?
12:44PM 5 SpanDSP - Squished Faxes
12:41PM 3 Chan_Woomera beta released at
12:40PM 2 Asterisk 'losing' upstream provider registration state during small network outages.
12:38PM 0 Looking for Honduras DIDs, Origination, Termination
12:14PM 0 Pls Hlp - Sendmail handoff / relay to cable
11:18AM 4 12 FXO ports into Asterisk
11:05AM 1 PRI auto reset?
10:41AM 0 Asterisk Manager Interface Remote BufferOverflow Vulnerability
10:21AM 0 Dial a specific priority
9:39AM 7 mini itx
8:30AM 4 Monitoring Sirrix quad BRI channels
8:11AM 3 privacy manager
8:09AM 2 ChanSpy on Asterisk v1.0.7
8:07AM 1 Always forward an extension?
8:00AM 0 RES: MFC R2 - Can this problem be solved??????????
7:50AM 0 dialtone conf.of Turkey for ata186 sip
7:39AM 0 This cpu usage doesn't seem right.
7:26AM 1 Polycom display variable
7:23AM 0 BRI signalling Morocco
7:01AM 0 AGI to monitor conenction quality
6:40AM 1 Asterisk @ Home setup & Doc
6:39AM 2 Legal Requirement for Digital PBX
6:25AM 1 Help with Dial multiple channels simultanously
5:58AM 1 Zap lines-inbound,outbound calls intersect
5:57AM 2 Management: Reload performace & Realtime performance ?
5:38AM 1 MGCP Groups
2:48AM 1 Server Load/Capacity
2:23AM 1 Music on Hold Choppy
1:46AM 0 avm c2 correct configuration for two p2p lines
1:03AM 0 Welltech 4 Port FXO - Asterisk
12:57AM 7 Cisco 7960 firmware upgrade promblems
12:19AM 0 Loosing hair on connecting Panasonic PBX- *- Euroisdn Italy
12:11AM 1 SIP DID routing
12:07AM 0 Routing calls by trunk?
Wednesday June 22 2005
11:54PM 3 flash panel only works with IP address
11:50PM 3 Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy
11:38PM 2 ASTCC not making calls
11:20PM 1 OT: MAX TNT and PRI calling name (CNAM) facility message
10:44PM 1 missing cdr records
10:30PM 1 Error on installing oh323 on asterisk
10:23PM 0 Malformed/Missing.URL Error from CallManager
9:26PM 0 Asterisk + Asterisk-Stat as a Employee Time Clock
9:03PM 2 Zaptel card AND Ztdummy together?
8:23PM 1 Sip Sidecar Options
7:34PM 2 asterisk authentication issue
7:17PM 0 DMS-500 CID NAME Problem
6:21PM 0 Zaptel + IBM OpenPower Servers
5:25PM 0 add-on mysql cmd
5:14PM 1 RE: res_cepstral
4:49PM 3 indexing tables for dialing
4:30PM 2 Asterisk Manager Interface Remote Buffer Overflow Vulnerability
3:47PM 1 Question on bridged calls
3:46PM 1 Connecting extern telephones,
3:30PM 0 Adit600-->Asterisk Via MGCP
2:45PM 1 Is anyone using VOIPREACH
2:41PM 0 Setup suggestions/ideas
1:37PM 0 Wireless & Wireline Integration
1:32PM 3 combining calls from 2 queues
1:18PM 1 Zap POTS Line Problem calling outbound
1:15PM 2 Weird ring back
11:46AM 1 Using HEAD version of Zaptel with Asterisk Stable Release
11:18AM 1 A Simple * Answering Machine w/ Caller Screening like the olden days
11:04AM 0 Presentation Number
10:01AM 1 Can I dial a number from handset to pickup voicemail?
9:57AM 0 DIAX 0.9.15a with GSM gateway functionality
9:20AM 2 problem compile
9:06AM 0 Performance Monitoring.
9:03AM 0 ISDN (PRI) in the US and Redirect?
9:01AM 1 Garbled one-way audio only with ulaw
8:58AM 1 volume "fading in and out"
8:58AM 1 TE110P Card
8:49AM 4 automated response
8:46AM 3 TDM400P & Channel Group
7:54AM 0 (no subject)
7:51AM 4 TDM400P DevKit Problem
7:50AM 10 New Asterisk Implementation
7:47AM 4 Asterisk Manager Api
7:34AM 1 Fwd:protocol TCP/UDP question
6:58AM 1 FOP related questions
6:41AM 4 FXS interfaces
6:28AM 0 Asterisk ended with exit status 139
6:23AM 1 gsm gateway
6:13AM 0 TDM400P and Dell Poweredge 1750
5:44AM 1 Re: [Serusers] ASTERISK+SER+MWI
5:43AM 2 meetme mute status
4:27AM 2 Is this server sufficient?
4:19AM 0 is sip:%2321 valid invite?
3:32AM 2 OT: Asterisk and Mambo - help wanted
3:29AM 0 Detecting the active queue agent...
3:28AM 1 Dialplan Q: Dialing with Capi
3:11AM 5 ZapRAS
2:28AM 0 Variables to emailbody of voicemail
2:28AM 2 Spanish doc
2:03AM 2 Asterisk to NEC NEAX
1:58AM 1 call divert to TRUNK , if one number is unregistered?
1:57AM 1 PPPD problem please help
1:43AM 0 using DBGet inside extensions.ael
1:16AM 0 3month Internship between February end July 2006
12:54AM 0 Core Dump
12:45AM 1 zeroconf help
12:25AM 0 3-way conference using zap channels -- how is it done?
Tuesday June 21 2005
11:50PM 1 DID not working? + sendmail problems
7:37PM 3 FXS
7:23PM 1 gxp-2000 tftp cfg
7:09PM 0 new res_js example order status checking script.
6:58PM 1 Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show
6:56PM 0 Questions about FXO Outgoing Dialing
6:37PM 2 Echo Issues
3:47PM 5 logged in agent make an outbound call?
3:43PM 0 Zombie?
3:41PM 0 IAX protocol will not go through firewall after certain time.
3:39PM 0 chan_unicall and /dev/zap/channel
3:19PM 4 unreliable lately?
3:16PM 1 Re: New JAVA application server for Asterisk - OrderlyCalls
2:15PM 5 NVFaxdetect
1:29PM 0 Re: Asterisk-Users Digest, Vol 11, Issue 136
1:20PM 0 Problem with Connecting pbx and asterisk: using TE405P Asterisk -> T1 -> PBX
1:13PM 2 Digium Card: Echo, Echo and more Echo
12:52PM 2 403 forbidden on SIP register
12:47PM 0 Astricon Europe Media Post!
12:47PM 5 Problem with Connecting PBX to Asterisk
12:26PM 0 Intermittent audio issues with Asterisk behind symmetrical firewa ll
12:21PM 1 Asterisk in India?
12:14PM 4 Grandstream 100 pricing question
11:58AM 1 Asterisk answers with high pitch sound
11:54AM 2 Polycom and CallerID
11:48AM 0 Best Echo Canceller.
11:01AM 2 Re: New JAVA application server for Asterisk - OrderlyCalls
10:06AM 0 [ot] wifi 3G gsm phone
9:41AM 1 MeetMe Problems
9:17AM 0 communication between IAX softphones
9:09AM 0 asterisk-api
8:35AM 1 Asterisk died - exactly every 60 minutes
7:24AM 1 GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?
7:22AM 1 chan_unicall, bug in 1.0.X - 99% CPU
7:10AM 5 app_changrab.c released on
6:56AM 1 Cisco 7750
6:47AM 1 modprobe wctdm waiting for ever
6:22AM 2 MFC R2 - Can this problem be solved??????????
6:16AM 0 MFC R2 - Can this be solved???
6:00AM 1 ast_data help
5:28AM 0 Call file calling twice
5:26AM 0 Looking for PRI Outbound Caller ID Configura tion
5:26AM 1 Ground Start on Asterisk
4:22AM 0 Problems with wew FXO modules for TDM400P
2:35AM 0 ipswitchboard
12:37AM 1 Asterisk and Sirrix PCI4S0 echo cancellation
12:18AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 131
12:04AM 0 [ abuse] New JAVA application server for Asterisk - OrderlyCalls
Monday June 20 2005
11:25PM 2 Asterisk does not function without a DNS ser ver
11:03PM 0 asterisk and radius?
10:42PM 1 storing CDR records in a MySQL database
10:09PM 0 Re: app_valetparking.c for * STABLE
9:23PM 3 accountcode not present in cdr
7:27PM 6 Extension Configuration Best Practice
6:05PM 1 the table tariffrate is empty in Areskicc !
6:04PM 1 Unable to reconfigure channel
5:48PM 0 IAXy G729 Translation
5:08PM 0 Prepaid systems considerations
4:37PM 3 service scan
4:24PM 1 Zaptel Disconnect Tone
3:51PM 1 voicemail system
3:14PM 4 ee1000 Ethernet in Dell 1850
3:07PM 0 Contexts Calling Each Other
2:56PM 1 Asterisk does not function without a DNS server
2:49PM 1 Compilation Problem with asterisk-addons
2:31PM 1 How can you check that eg TDM04B hardware installed and drivers OK
1:39PM 1 Looking for PRI Outbound Caller ID Configuration
1:37PM 0 Re: app_valetparking.c for * STABLE
1:21PM 2 FXO/FXS cpu spikes, data loss and ztclock.
1:19PM 2 Automatic Agent Login
12:33PM 0 Suggestions for using AbsoluteTimeout
12:22PM 1 SIP Ad-Hoc Conferencing with Asterisk
11:53AM 1 RCAPI ISDN Support
11:23AM 1 Open for business! / JavaScript module for Asterisk Unveiled!
11:20AM 3 QuadBRI: How to set the outgoing callerid (KPN - NL)
10:46AM 0 What should I get for SOHO TDM card or sipura3000?
10:15AM 1 Re: app_valetparking.c for * STABLE (1.0.X)
8:15AM 0 Can't get TDM04B to work!
7:47AM 1 TxFax: can't get a fax to destination (log inside)
7:44AM 0 LookupCIDName on outgoing calls
7:21AM 1 chan_h323 vs chan_oh323 & chan_ooh323
7:17AM 2 app_valetparking.c
6:33AM 0 MGCP and SIP clients
5:23AM 0 second isdn line doesn't work with avm c2 card
4:11AM 1 call file ignored?
4:11AM 3 AGI/PHP errors
3:27AM 0 CPU load 100% when SIP register
2:53AM 1 oneTouchVoicemail issue with Polycom 1.5.2
2:46AM 1 $0-per-month (pay as you go) provider with T.38?
1:45AM 1 sipredirect question
12:46AM 3 Debian Vs Fedora
12:25AM 2 webvmail debian package
12:12AM 0 No CDR Records
Sunday June 19 2005
11:57PM 1 help for making several calls at the same time..
11:43PM 2 Zaptel HEAD with * Stable?
11:17PM 0 Premptible Linux Kernel
11:02PM 0 Scratchy audio on Bridged PRI Calls
9:40PM 4 Polycom 500 Sound Problem
3:22PM 3 tos problem
3:17PM 2 NAT/Proxy advise
1:42PM 0 chanisavail...not workin with SIP and IAX
1:19PM 0 Zaptel and Zapata Conf's
1:17PM 0 Problem with astperl primitives say... in astcc
12:54PM 1 MySQL to static .conf
12:46PM 1 *67 with Sipura 3000
11:00AM 0 chan_capi-cm-0.5.1 fixup release
10:26AM 2 outgoing call routing
10:11AM 0 asterisk and
7:08AM 0 I don't get a queue_log with my version of asterisk (0.7.1).
6:23AM 0 missing mysql cdr records
6:01AM 0 Can't locate module sound-service-0-3
5:54AM 4 bluetooth audio and asterisk
5:51AM 3 Libtiff 3.5.7 - recommended version for spandsp
5:24AM 0 stale nonce received
4:29AM 0 Now it is working
3:13AM 0 Is actually somebody doing it: load balancing 20 asterisk servers
12:45AM 0 Asterisk on vpbx exited on signal 11. Might want to take a peek.
Saturday June 18 2005
8:35PM 1 channel.c:1884 set_format: Unable to find a path from g729 to gsm
7:45PM 0 How to setup two Asterisk boxes - keeping the registration
5:25PM 0 New mirror/translation
4:45PM 0 Three way calling with Cisco 12SP+
2:29PM 0 H323 implementations
1:24PM 0 [Fwd: IAX with shaw cable not going through]
1:16PM 4 IAX with shaw cable not going through
12:00PM 1 Want to test my * behind firewall
11:55AM 2 Unable to make outbound calls
11:39AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 68
8:00AM 3 FATAL: Error running install command for wctdm
3:45AM 0 UK SMS Config problems
3:18AM 0 E1 to E1 connection
1:54AM 0 TTS
12:59AM 1 How to view all call detail in asterisk server
12:11AM 2 allowing outside dialing only for SIP users
Friday June 17 2005
11:15PM 0 AreskiCC and asterisk
8:20PM 1 Unable to find a path from g729 to gsm
8:17PM 1 Local numbers
6:40PM 0 Queue/How to get the number of incoming calls
5:19PM 2 ASTCC Rate Calculation
4:59PM 1 Asterisk ael files
3:48PM 1 callqueues confused :(
2:04PM 0 MGCP files for Polycom
1:34PM 6 Console ALSA Sound
12:26PM 0 Multiple phones on a Zap FXS extension
11:56AM 0 Phone lookup
11:50AM 0 Spandsp - fax problem
11:48AM 2 PLEASE HELP X100P no responding
11:29AM 0 Phantom problem authenticating IAX2 with RSA
10:29AM 1 PIX Firewall Ports and Access-Lists
10:25AM 0 txfax 18Kb file problem
9:36AM 5 tdm400p not working after cvs-head update
9:15AM 2 Can't switch span to E1-mode
8:51AM 5 Presence and IM?
7:59AM 0 Programinng Aplication with Music on Hold
7:54AM 2 Calculating the lenght of time in a call queue?
6:58AM 0 No ringing tone on outgoing SIP trunk
6:55AM 0 auto-dial dial status
6:41AM 0 Asterisk box as a billing machine in a PSTN network
6:12AM 0 Need Last App-Fax Source
5:48AM 0 Agents/Queues Contexts
5:19AM 4 Analog modems behind an Asterisk server?
5:05AM 2 Dell PowerEdge + TDM
4:52AM 0 Call group channel limits
4:02AM 0 Miax: Digital voice channel when connecting to asterisk
3:31AM 0 Sip INFO DTMF over satellite
2:30AM 1 bristuff-0.2.0-RC8g: zaptel error in suse 9.2
2:08AM 1 Dial timeout when server down
1:58AM 0 ata186 IVR problem
1:06AM 2 Junk at the beginning of frame
Thursday June 16 2005
11:58PM 0 inbound agent recording filename
11:10PM 1 Re: Dell PowerEdge SC420 interrupt issue
10:42PM 0 How to detect telco Message Waiting Indicator (WMI)
8:49PM 0 Problem with monitor.
8:24PM 3 Dial Commands "D" Option Question
7:39PM 1 Routing SIP to Cisco routers running IOS 12.3+
5:37PM 1 Newbie question about pressing a key to, be connected to the caller
5:27PM 1 Viva Madrid!
4:40PM 5 meetme - conf-invalid
3:53PM 3 rxfax problem - libspandsp issue?
3:41PM 2 Multiple Sipura 3000
1:16PM 2 @Home AMP call recording documentation
1:04PM 0 SIP connection
12:40PM 4 Sipura 3000 help
11:35AM 1 faxdetect config issues
11:12AM 1 iax2 registry - auto reconnect ?
10:45AM 0 How to get started, what do I need?
10:23AM 0 Intelligent maximum channels solution?
10:15AM 0 have asterisk box #2 pick up calls.
9:45AM 1 Coding a telemarketing call blocker
9:01AM 0 misdn and call hangup problem
8:28AM 2 Error when compiling in freeTDS support
8:23AM 1 Nobody picked up in 30000 ms
7:43AM 0 Problem with 2 digium cards
7:27AM 6 Case studies for Asterisk Voicemail
7:05AM 0 Subject: asterisk gsm gateway hardware
6:38AM 1 Cisco 7960 (SIP) with Asterisk: how to get # to work during a call
6:13AM 1 reload from dialplan
5:43AM 0 Fall back dialing
5:36AM 3 SER and Asterisk question
5:34AM 0 How to dimension Asterisk - that is used solely as callback server - only sending untranscoded voice between two ISDN channels on PRI ?
4:15AM 1 unamble to dialout to mobiles and others "special" numbers
4:06AM 1 Do includes include the includes
3:39AM 0 Zaphfc unable to dial out
3:38AM 0 Grandstream phones losing registration withserver.
3:27AM 0 chan_h323 context
3:12AM 0 Features.conf Set Language
3:03AM 0 Error on incoming calls
2:53AM 0 Busy, differences between SIP and Zaptel(bristuff)
1:46AM 1 MeetMe ERROR "Unable to dup channel"
1:25AM 1 Grandstream phones losing registration with server.
1:08AM 0 Asterisk Problems with FXO Ground Start Trunks and DID Wink Start Trunks
12:32AM 0 Asterisk on Fedora Cora 3
12:22AM 0 SER with Asterisk Problem
12:10AM 1 How to stop Asterisk from changing the SDP?
12:02AM 9 chan_capi-cm-0.5 release announcement
Wednesday June 15 2005
11:59PM 0 Asterisk Integration with an SBC-410 phone system
9:53PM 2 iax2 can't listen on virtual interface
9:42PM 0 asterisk with
9:03PM 2 terminating DID to FWD
8:40PM 1 Bridged-appearances
8:36PM 2 VoiceXML? question
8:11PM 2 Bill seconds
7:22PM 2 SIP to PRI
7:10PM 0 Problem with slin
7:06PM 1 Strange Inbound Ring Handling
7:00PM 1 Gnet Phones
6:26PM 1 This mailing list is being spam filtered on my site.
5:48PM 3 Includes include the includes?
5:19PM 2 1-800 DID in Alberta
4:04PM 3 Grandstream ATA Toasted
3:52PM 0 zaptel witch smp
2:46PM 0 SIP to ZAP dialout without pre"0" (Asterisk <-> HiCom)
2:44PM 1 echo cancellation on an iax2 channel
2:05PM 1 Changing caller ID on a Zap channel
2:03PM 0 Dundi - Multiple Results
1:48PM 1 Sipura 841 Ringtones
1:18PM 1 Caller ID on TelaSIP SIP Channel
12:41PM 0 Re: Asterisk-Users Digest, Vol 11, Issue 100
12:30PM 1 Problem with overlap dialing
12:27PM 1 CellPhone BlueTooth adapater with Wireless Profile ??
12:13PM 0 Config files under CVS versioning system
12:07PM 1 phantom answer
12:00PM 12 WiFi IP Phones
11:47AM 6 Help with Cron and Reload
11:09AM 1 SIP transfer/REFER to voicemail problem
10:51AM 1 Question on cdr_odbc
9:34AM 0 asterisk gsm gateway hardware recommendation?
8:16AM 0 Error installing Asterisk with zaptel and libpri
8:15AM 0 Handling -1 in dialplans
8:04AM 0 user web interface
7:55AM 2 Asterisk and Max TNT
6:50AM 0 RE: Call being answered, but no audio on either end
6:37AM 1 Broadvoice and Inbound DTMF
6:17AM 1 [Help] ZT_CHANCONFIG failed on channel 25
6:01AM 1 Port Inquiry
5:45AM 0 Load problems
5:43AM 1 asterisk security
5:40AM 2 Nasty little incident ...
5:28AM 0 empty HDLC frame or bad CRC received
5:10AM 0 CDR's -> ODBC and logging IP's
4:13AM 2 SIP call doesn't execute the 's'-extension
3:47AM 0 Asterisk slow transferring calls
3:24AM 1 Sipura SPA 3000 FXO Setting India
3:20AM 4 Dial more then 9 digits
3:13AM 0 Personalised Unavail / busy messages no longer play
3:08AM 1 app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
2:56AM 0 Stable Versions
2:40AM 1 How to restrict access to * for a specific soft/hard phone model?
1:47AM 1 Old but Gold
1:45AM 0 SIP REFER method.
1:35AM 1 newbie question..
12:59AM 0 TAPI for Dos (xbase / Clipper)
12:54AM 0 SV: Asterisk and Panasonic KX-TD1232
12:09AM 0 localize ${VM_DATE} ?
Tuesday June 14 2005
8:12PM 0 upgrade problems...
7:35PM 0 Asterisk & outbound proxy?
7:18PM 1 Newbie question about pressing a key to be connected to the caller
7:17PM 3 Calling on all Polycom Experts
7:07PM 2 [PRI] TE110P
6:52PM 1 canreinvite=yes not working with sipura device.
6:43PM 0 Asterisk@Home and intermittent ring sound on callers end
6:13PM 0 RE: Call being answered, but no audio on either end
6:09PM 1 how to make a dialplan on bases of Caller
6:05PM 0 Snom hardware quality
6:02PM 0 dialplan appdata separators
5:39PM 1 Asterisk and grandstream weird call probs
4:10PM 0 ATA186 & X100P - detect hangup
3:56PM 0 Digit Map for IP500 - prepend digits from phone
3:10PM 1 OH323 Packetization
2:30PM 0 Call being answered, but no audio on either end (Intermittent)
1:04PM 0 AW: Should I choose DSL @ 1.5 or a full T1?
12:59PM 0 how to remove + in CIDNumber
12:56PM 0 chan_sip.c: Maximum retries exceeded on call ........ for seqno 1 01 (Non-critical Response)
12:49PM 0 Has anyone made a purchase from
11:54AM 0 Cannot handle frames in 2 format
11:16AM 2 Call parking in multi user environment
11:14AM 0 RJ45 instead RJ11 in Digium's TDM20B card he lp me please
11:11AM 0 RJ45 instead RJ11 in Digium's TDM20B card help meplease
10:46AM 5 RJ45 instead RJ11 in Digium's TDM20B card help me please
10:36AM 1 outgoing prefix dial plan
10:34AM 0 Zaptel problem on BSD
9:33AM 2 -HEAD/--STABLE using 100% cpu
9:08AM 2 Questions about contexts
9:04AM 8 Making Asterisk NOT Pickup a Line when Ringing?
8:21AM 0 Info on ACD in Asterisk
8:19AM 0 Transfers on PRI connected channel banks and legacy PBX's
7:39AM 4 488 Not Acceptable Here
7:27AM 6 VOIP-INFO down?
6:43AM 2 SIP_HEADER - anybody using it?
6:28AM 5 HT-488 vs. SPA-3000?
6:00AM 3 How to setup a test number to know my extension number
5:56AM 2 Asterisk and Panasonic KX-TD1232
5:48AM 1 Long time to detect hang-up
5:26AM 2 # no longer working
4:57AM 2 Features.conf for secretary function
4:54AM 0 ERROR[6504]: chan_zap.c:6710 mkintf: Channel 24 is reserved for D-channel.
4:44AM 0 Is there a problem when we want to transfer anincoming call to an external phone number
4:10AM 1 Re: Asterisk-Users Digest, Vol 11, Issue 93
4:01AM 3 RTP Forwarding
3:31AM 1 SIP to ZAP Dialplan
2:57AM 0 How to connect to LVDX /
2:35AM 2 ERROR[5539]: chan_zap.c:9750 setup_zap: Unable to load config zapata.conf
2:32AM 2 AVAYA & Asteris & H323 chanel
2:11AM 0 Static CLID
1:59AM 0 Max Retries Exceeded - IAX2. Network problem?
1:16AM 1 Is there a problem when we want to transfer an incoming call to an external phone number
12:06AM 0 No mans problem?
Monday June 13 2005
10:42PM 2 Adtran TA 750 FXO Groundstart Mode
10:22PM 7 Keeping users, extensions, voicemail and so on in DB
9:45PM 9 SIP Listen to multiple ports
8:54PM 0 Re: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition
7:31PM 2 ztcfg server crash
7:00PM 1 Need help connecting two * pcs with *@home
6:58PM 0 Reading about G729
6:19PM 0 IAX Issues...
5:00PM 3 problem with pf and asterisk
3:59PM 7 MCI vs. XO/Allegiance
2:09PM 1 Re: Re: Digium Website Update: Asterisk Busi ness Edition
2:03PM 0 Re: Asterisk forking, Was: Digium Website Up date:Asterisk Business Edition
1:55PM 0 Re: Voicemail and MS Exchange Synchronizatio n
1:27PM 0 Unable to support trunking .... without zaptel timing
1:14PM 0 Asterisk connecting remote villages in westernUganda
12:39PM 0 DID in AMP with 2+ incoming lines
11:52AM 0 Hiss patch
10:55AM 2 Asterisk connecting remote villages in western Uganda
10:42AM 1 Zaptel modules
9:57AM 1 More on the IAD connection
9:01AM 1 DNIS and DID seeking confirmation
8:56AM 1 Interfacing to an IAD
8:43AM 0 T1 multiplexer (or ?) for failover in largeinstallation
8:42AM 2 snom 190: dial tone without registration?
8:35AM 2 T1 multiplexer (or ?) for failover in large installation
8:15AM 1 Components and suggestions for an asterisk server with 9 to 17 POTS.
8:04AM 1 wiki server session limit?
7:20AM 1 presence and video conference
6:56AM 0 nativ bridging problem with ilbc!!
6:04AM 3 Oh323 and Caller ID missing
5:47AM 1 Cepstral partnership with Digium
5:25AM 0 Guidance , for which card to buy
5:17AM 2 SNOM, Asterisk and Attended transfer (bug?)
4:59AM 0 MySQL: max realistic size of extensions table.
3:37AM 1 about timeouts
2:08AM 1 Problem with DTMF Relay and Oh323
1:53AM 0 Asterisk installation error after CVS update
12:32AM 2 Need Help with pickup *8
12:27AM 0 Phantom incoming calls on x100p
Sunday June 12 2005
11:35PM 0 Macro support in realtime
10:41PM 2 POLYCOM IP 500 Setup
7:54PM 0 phone rings but caller doest hear it
5:11PM 0 Asterisk Community Meeting in Sydney Australia
3:48PM 0 *66 auto redial emulation?
2:51PM 0 ZAP channel (X100P) won't detect call waiting
1:40PM 1 how to tell
12:31PM 0 Unable To Register a SIP phone ... Help Needed
10:34AM 3 GSM -> ULAW sound conversion
10:10AM 1 Not answering inbound a line used for outboun
9:50AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 81
8:03AM 0 IAXy Pulse/Flash timing
7:52AM 1 DID Issue
7:32AM 3 Best bet ... IAX vs SIP
6:07AM 1 Changing Email Templates
1:56AM 1 Load balancing for each protocol
Saturday June 11 2005
11:55PM 3 Not answering inbound a line used for outbound
8:07PM 4 PRI Trouble
6:29PM 0 SPA-2001 features on analog side
5:28PM 0 LCDC Integration/bounty
4:32PM 1 ISDN Sub-Address
4:27PM 0 Help with denighing access to certain numbersbyCallerID
3:46PM 2 Help with Oh323
3:45PM 0 Help with denighing access to certain numbers byCallerID
3:39PM 0 Help with denighing access to certain numbers by CallerID
3:14PM 0 Re: ztdummy/rtc - staticy audio
3:05PM 1 SIP-H.323 dial tone and busy tone problem.
2:46PM 0 Re: ztdummy/rtc - staticy audio
2:33PM 1 Problems with IAX Trunks
2:25PM 1 SIP Connection Timing Out BroadVoice
2:04PM 0 Transcoding GSM to G723.1
2:02PM 0 Flash hook not going through SPA-2002
1:33PM 1 AreskiCC Calling Problem
11:13AM 3 how to limit simultaneous calls
11:01AM 0 Voice quality of Softphones vs. IP Phones an d Gateways.
10:56AM 0 Shorewall Configuration for Asterisk Box
10:44AM 0 SIP_HEADER example
10:35AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 77
10:04AM 3 ztdummy/rtc
8:55AM 0 Caller ID transforms
8:49AM 0 Deleting Unavail Message
8:09AM 0 In Dial Application, reading the L(x[:y][:z]) parameter from database.
7:55AM 3 No path to translate from SIP/615-25c8(256) to SIP/601-27b6(4)
7:23AM 0 Asterisk Users & Developers on their way to Madrid - Meet us there!
7:03AM 0 Voice quality of Softphones vs. IP Phones and Gateways.
4:22AM 4 Best platform
3:02AM 1 Why does my name not show in the from address
2:48AM 1 Manager API timestamps of events
12:21AM 0 Newbie Here..... Unable To Register A SIP phone
Friday June 10 2005
10:50PM 2 what is asteriskathome-1.0.iso?
10:35PM 1 VoicePulse DTMF Problems Anyone?
9:36PM 1 Is it necessary that i need to have TDM01B for PC-to-PC intercom calls?
7:45PM 2 Asterisk@Home connecting through firewall/router
6:49PM 1 [newbie] configuration for IAX server to server
6:26PM 19 Should I choose DSL @ 1.5 or a full T1?
6:23PM 0 First Asterisk community meeting in Sydney
4:48PM 1 Wildly inaccurate CDR records
4:12PM 0 Open
3:12PM 0 voiceblue gsm/sip
3:00PM 1 Polycoms Go Silent after a a handful of calls.
2:47PM 1 Convert extensions.conf INTO MySQL script
2:27PM 1 Newie Questions
2:11PM 1 Unable to register Zyxel WIFI Phone as SIP Client to Asterisk
2:05PM 3 DMS-500 CID name not in CDR
1:36PM 0 Unable To Register A SIP phone
11:31AM 2 Toll Free DIDs
10:19AM 0 Call disconnect
10:12AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 71
9:23AM 0 Dropping Frame of G729
9:15AM 0 blindtransfers with IAX
8:58AM 2 G711 ( alaw or ulaw ) pass-thru
8:48AM 0 AAH 1.1 cannot call between extensions (xten lite softphones)
8:44AM 4 Best BootRom & SIP Code for Poly600?
8:08AM 1 config problem
7:45AM 0 asterisk and mpg123 lock up
7:41AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 69
7:41AM 0 Is it possible to have a remote Phone work behindNat without a VPN?
7:20AM 0 D-Link DVG-1402S
7:07AM 1 Re: Voicemail and MS Exchange Synchronizatio n
6:55AM 1 ATTN: Keith - Seriously OT
6:53AM 0 SoftPhone - Solaris
6:48AM 1 404 not found
5:56AM 1 Asterisk Evening in Melbourne (again!) next Thursday
5:51AM 0 g729 support
5:41AM 0 SpanDSP wownt compile
5:24AM 1 TE410P and Siemens HIPATH 3750
5:01AM 2 Cell redirect
4:47AM 0 Handytone-488 FXO - PSTN in calls to Asterisk, sip.conf? (fwd)
4:35AM 3 chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU
3:17AM 1 Call inband progress indication and zaphfc
2:09AM 5 lost g729 lic
1:58AM 1 Request OPTION and 404 Sjphone Xlite
1:53AM 2 G.729AB codec support
12:13AM 0 sirrix NT mode
Thursday June 9 2005
11:45PM 1 PHPAGI Swift Escape Digits
11:30PM 1 Zap Clocking - Frame Slips - tdm400p wcfxo zttest cpu spikes spandsp
11:18PM 2 Is it possible to have a remote Phone work behind Nat without a VPN?
10:21PM 1 IAX2 Max Retries dropped calls Firefly
8:09PM 0 "auto-dial out" not waiting for answer
7:35PM 0 Conversations cuts: "didn't get a frame from Channel: SIP/..."
7:22PM 1 compile error cannot find -lidn
6:58PM 0 Multiple Digium cards?
6:19PM 0 GXP-2000 Wiki update..
6:04PM 0 Handytone-488 FXO - PSTN in calls to Asterisk, sip.conf?
5:40PM 2 Sixtel is still alive?
3:43PM 0 Flash Hook won't work with Asterisk@Home and SPS-2002
2:12PM 0 Parked Call queue function key notify
1:26PM 1 Phantom (ghost) Calls with Wildcard TDM400P
1:25PM 1 voicemail check for busy message
1:07PM 5 Voicemail and MS Exchange
12:53PM 1 Inbound provider in Canada
11:46AM 0 Getting a Quintum AS200 to connect with Asterisk using SIP?
11:04AM 0 OT: SpamFiltering (used to be: ATTN: Keith)
10:49AM 0 Digium vs. Sangoma: Performance
10:39AM 4 ATTN: Keith
10:38AM 1 Asterisk to Cisco Voip System Unity
10:18AM 23 Voicemail and MS Exchange Synchronization
10:18AM 1 astGUIclient installation problem
9:51AM 0 Agent refuses to log out
8:59AM 0 Polycom IP-500 & 600 Nat settings.
8:51AM 8 howto write CDRs on two mysql servers
8:46AM 1 REPOSTED: Polycom 500 "Group Call Pickup Feature" and *
8:16AM 1 Cisco 7960 and Skinny
8:08AM 2 E1 and SS7
8:04AM 2 having to reload asterix after internet connection failure
7:02AM 4 Lingo(.com) and Asterisk
5:49AM 1 3COM NBX SuperStack 3
5:12AM 3 Pickup problem
4:10AM 3 Asterisk to Cisco Unity
1:50AM 3 Softphone for Linux desktops
1:22AM 0 New version 1.013 of Asterisk VConfig
Wednesday June 8 2005
11:44PM 1 TDM400P strangeness
11:26PM 0 Asterisk Engineer/Programmer required
11:05PM 1 Thank you for the timely suggestion
10:48PM 3 Play MP3 during Record
10:23PM 5 GXP2000 and hint LED's
10:01PM 1 tdm04b slow response
7:37PM 2 [ADMIN]: subscription failure
6:15PM 3 More than one account from the same provider?
6:01PM 2 Ringing a few phones
5:39PM 1 Cisco 7960 mic generating noise on other end
5:31PM 2 Incoming call stops at random with Teliax
3:57PM 2 format g729 and
3:04PM 3 AgentCallBacklogin (logout continued...)
2:35PM 0 Load per server?
2:28PM 2 IP PHONE iareaphone x100, tested??
2:08PM 13 Anyone noticed Voipjet voice quality problems?
12:23PM 1 Do I need a ring capacitor to use TDM400P cards in UK
11:25AM 1 rxfax not working
11:11AM 0 Number of AGI's running at the same time
10:54AM 1 EuroISDN Italy - quadbri - zaptel.conf - what settings work ?
10:23AM 8 TDM04B
9:33AM 1 Remote CDR logging on mysql:
9:23AM 0 CVS Head, Flex 2.5.31 or higher? READ THIS!
8:19AM 7 Clicks in audio with TE100P PRI
8:14AM 0 Asterisk and Alcatel 4200 PBX
8:05AM 1 Latest CVS and app_rxfax
7:57AM 3 TDM400P... ignoring hanguponpolarityswitch
7:28AM 10 * @ Home: All Circuits busy
7:00AM 0 sip to sip echo with meetme, timing
6:59AM 0 Polycom 500 "Group Call Pickup Feature" and *
6:40AM 0 Fax + Fritz + Capi + detection
5:38AM 2 Station Lines
4:04AM 0 Faxing error rtp.c:504 ast_rtp_read: Unknown RTP codec 100 received
3:55AM 2 How to handle one incoming call on multiple lines?
2:39AM 1 performance of * in several scenarios
2:21AM 1 no DTMF pass-thru
2:09AM 0 file.c:1073 ast_waitstream_full: Wait failed (Interrupted system call)
1:28AM 5 Xlite not communicating with Asterisk
1:03AM 2 bypass incoming it possible?
12:54AM 1 error message: INIT: Id "s0" respawning toofast:disable for 5 minutes
12:44AM 1 Newbie on asterisk ask for configuratio help
12:33AM 1 Asterisk to Avaya PBX using TDM cards
Tuesday June 7 2005
10:39PM 1 MGCP Useragent
10:07PM 1 Message Playback
10:02PM 2 so what are the additional hardware components needed?
9:13PM 3 Help Connecting Cisco AS5300 to Asterisk
9:05PM 2 Books
8:46PM 1 error message: INIT: Id "s0" respawning too fast: disable for 5 minutes
7:54PM 0 Incoming voice "disappears"
6:55PM 2 codec preference
5:17PM 0 meetme recording of one user in the conference
4:54PM 3 FXO Gateway recommendation
4:49PM 2 Gnudialer
4:22PM 2 ASTCC what has been changed
3:40PM 1 connecting Asterisk to NEC NEAX system
3:20PM 1 Fax problem with Asterisk @home ver 1.0.7
3:11PM 1 DID on SIP channel
2:25PM 1 DISA Help
1:53PM 2 Help! Zap echo on bridged calls
1:19PM 0 Incorrect FAX detection.
12:54PM 0 X100P long delay before dial
11:42AM 1 SPA-2002 and NAT
11:38AM 1 Problem in Reloading the asterisk server !
11:25AM 0 Re: Asterisk-Users Digest, Vol 11, Issue 48
10:59AM 1 TE410P
10:32AM 0 AgentCallbackLogin (logout)
9:24AM 3 Polycom Phones & shorter than /24 netmasks
8:49AM 1 New Asterisk Manager Proxy -- astmanproxy 1.0
8:45AM 3 rxfax not answering
8:13AM 0 Monitor and failing Fax
8:08AM 1 realtime & nat
7:53AM 3 AAH 1.1 - CRM Setup
7:43AM 4 Queue Log
7:13AM 2 Multiple E1s on one box
6:36AM 2 PRI Lines not being answered (No User Responding)
6:18AM 0 Sounds
6:07AM 1 Polycom 500 'SERVICE'S' key
5:50AM 0 (no subject)
5:34AM 0 3com 3105 Attendant DSS Console (SIP??)
5:27AM 1 CallerID/chan_sccp
5:23AM 0 NEWBIE: sip subscriptions
5:01AM 2 Call Routing based on number dialed (using S IP)
4:29AM 1 RE: Asterisk-Users] te405p and dell poweredge
3:26AM 4 I want to move the MySQL server out to another machine
3:20AM 1 How to configure 2 asterisks
3:05AM 1 D-link DPH-80 (SIP) call to asterisk problem
2:22AM 3 te405p and dell poweredge
2:10AM 1 connecting Asterisk with Siemens HiPath4000
1:25AM 0 Duplicate Calls
1:21AM 2 Problems with Junghanns QuadBri
1:05AM 2 How to allow multiple codecs in A@H
1:02AM 3 run a script on completion of call
12:44AM 1 USB phones...
12:14AM 0 Re: chan_sccp / 7960: "External call" and more
Monday June 6 2005
9:46PM 1 RE: LOA for CFA . . work up "pencil copy"
7:04PM 1 Service Unavailble, Sipura 3000, CheckGroup, what the heck??
5:34PM 1 Debugging SIP Connection
5:19PM 1 NAT & RealTime
3:35PM 0 About BillSec when having conferences
3:32PM 1 Transfer differences between BudgeTone101 and Snom190
3:19PM 1 ADSI over SIP
3:03PM 2 ENUM NL dead ?
2:57PM 1 Servers Compatible with Digium HW
2:39PM 1 Double NAT issues with SIP and workaround (?)
2:37PM 4 *@home .conf files request
1:49PM 3 Asterisk eating up 99.8% cpu
12:39PM 1 CLUELESS NEWBIE needs help making an outboundsip call to PSTN
12:36PM 0 dial-a-string (e.g. 2=a,b,c, 3=d,e,f)
12:30PM 2 Erro message - Received mini frame before first full voice frame
11:44AM 0 Newbee, help with cdr/odbc/mysql logging problem
11:36AM 0 Unable to Configure NetPhone IP phone
11:32AM 5 Asterisk Live! CF
11:17AM 0 Degraded voice without packet loss
11:09AM 0 Dial(SIP/xyz&zap/r1/123) with different Caller Ids ?!
11:08AM 0 How to make Polycom phones work with Asterisk asaSIP Client?
10:17AM 5 OT: Please comment on Dvorak's troll
10:16AM 1 Hangupcode == 44
9:43AM 0 How to make Polycom phones work with Asterisk as aSIP Client?
9:08AM 0 D channel initialization
9:08AM 0 How to make Polycom phones work with Asterisk as a SIP Client?
8:24AM 0 OT: WAS: * found in Iraq!! NOW: Asterisk bus iness sightings
8:20AM 1 Asterisk at Home ...
8:01AM 1 IAX Phone Pro - Open Beta Test
7:35AM 5 Polycom 500...
7:09AM 1 Zaptel comple on FC2
6:35AM 0 Any thoughts
6:30AM 5 IRQ Problems
6:12AM 0 Echo Issues via SIP
5:53AM 2 Features.conf - atxfer
5:34AM 0 snom 360 conference button
5:24AM 0 [SPAM] - what hardware components do i need? - Email found in subject
4:45AM 1 AMP and custom application
4:37AM 2 How to Playback a file continuously during conversation?
3:53AM 1 Compiling asterisk-addons-1.0.7 on Debian Sarge with asterisk-packages installed
3:00AM 1 what hardware components do i need?
2:47AM 1 Quotation request: 12 KHz signal generation for billing purposes.
2:29AM 1 Issue with SIP inter-op
2:10AM 2 Variables and status problems in AGI application
1:41AM 2 No DTMF interpretation on outgoing calls
1:28AM 2 mISDN + + winbond issue
12:17AM 0 SIP changes in CVS head
Sunday June 5 2005
11:40PM 1 Little help with MySQL please
11:26PM 0 Re: Bison, Flex, Conditional Expression
9:15PM 0 RXFax and Hangup context Question.
9:11PM 2 TDM400P Polarity reversal detection
8:46PM 1 Problems getting VoicePulse Connect working
7:36PM 1 Accountcode being ignored?
5:48PM 1 TDM20B FXS card configuration?
5:34PM 1 Voice Dtect
4:26PM 0 sipura3000 problems in callcenter
4:04PM 2 Disa - how it returns on user not dialing any numbers ?
3:57PM 0 Adtran 600 channel bank
3:49PM 0 VoiceMail Termporary greeting option
3:41PM 0 Examples of Asterisk deployments with 100-500 users?
2:09PM 1 IAXtel update!
2:06PM 3 ISDN 4 BRI card for UK
1:14PM 2 te410p not working after cvs-head update
12:47PM 0 Outgoing TDM400P FXO calls always answered
11:45AM 0 UK call disconnects during record
11:18AM 0 ACD Login
9:33AM 1 DTMF Tone Lengths
8:31AM 4 Digium G729 licensing - is it worth the trouble?
7:07AM 2 180 Ringing?
2:50AM 2 Compilation on Debian with support for HFC-chip based ISDN-cards
2:38AM 0 CDR records.. How do you deal with them?
2:10AM 1 New version of Asterisk VConfig
1:17AM 1 Unable to create channel of type SIP-please help
Saturday June 4 2005
11:40PM 0 New version of IPSwitchBoard
10:41PM 1 Extension 'hint' info please?
10:39PM 1 SetCallerID based on extension
10:37PM 3 zap to zap bridging not hanging up
3:10PM 0 Satelite Internet connection
2:48PM 0 facing problems with TDM400P
1:39PM 1 Asterisk@Home Forum Suggestions
12:20PM 0 Garbled speech - strange problem.
11:57AM 4 X100P installed OK, after added TDM400P Asterisk would no longer start
11:44AM 1 How to quickly replace ',' with '|' in dialplans?
5:31AM 2 chan_sip + MD5 encryption: WARNING Format for authentication entry is user[:secret]@realm
5:26AM 0 Problem with X100P (ZT_SPANCONFIG failed)
4:25AM 1 call queues problem
4:24AM 0 Apologies
4:02AM 0 chan_sccp / 7960: Messages key, line / speeddial keys
3:31AM 0 Re: [Asterisk-Dev] Is there any SCCP patent issue?
3:28AM 1 Incoming SIP calls with no extension
3:27AM 0 [Fwd: [Fwd: Help Cisco 12+]]
3:27AM 0 [Fwd: [Fwd: Use with Octtel 8 port fxs device]]
3:25AM 0 [Fwd: Use with Octtel 8 port fxs device]
3:25AM 0 [Fwd: Help Cisco 12+]
2:51AM 3 Automatic callback feature *66
2:21AM 0 SMS + SMSQ
2:13AM 0 Is there any SCCP patent issue?
1:52AM 3 SNOM extension lights programmable, eg. based on asterisk variable setting?
12:19AM 2 Zap channel not hangingup
Friday June 3 2005
9:25PM 1 ztdummy errors on WBEL4
9:23PM 1 ARESKICC DOESN'T make a CALL!!!
8:16PM 1 Caller ID Routing using VoicePulseConnect
7:05PM 0 spam filter
4:55PM 1 Asterisk and Audiocodes 108 FXS
4:53PM 1 Problem starting RX_FAX and TX_FAX Module
4:16PM 0 (no subject)
3:07PM 6 Livevoip 800 Choppy Audio
2:56PM 1 login/logout of call queue
1:53PM 0 Call Routing based on number dialed (using SIP)
1:49PM 0 Use with Octtel 8 port fxs device
1:44PM 0 Help Cisco 12+
12:58PM 1 Can an open source project get acquired?
12:51PM 2 Setting up calls through the manager interface
12:48PM 0 add/remove PRI card without rebooting
11:39AM 0 Asterisk - >SIP -> DNIS
11:37AM 1 AgentLogin already on?
11:20AM 1 Call parking on Polycom 500 doesn't transfer, stays on hold
10:39AM 1 Last astcc/* versions working?
10:01AM 0 Dazed and confused on refresh
9:48AM 0 New astGUIclient version released 1.1.1
9:33AM 1 .call files in outgoing dont get run
9:27AM 0 ring requested on channel 0/23 already in use on span
9:11AM 1 chan_sip notices
9:09AM 0 voicemail errors
9:01AM 1 oh-323 / Cisco AS5300 problem
7:58AM 0 * found in Iraq!!
7:22AM 0 Anybody knows how to setup chan_misdn incoming calls
7:09AM 2 Everyone-- the scoop on Bison/Flex --
7:03AM 0 Asterisk @ Home 1.1 Released
6:59AM 1 G.729 with RVA
6:54AM 0 Digium TDM400 Trouble Shooting Tip
6:31AM 2 Simple sip.conf question
6:21AM 0 SIP_CODEC, reinvites, and changing codecs
5:27AM 0 Installation of Asterisk addons 1.0.7 fails (longish)
5:25AM 1 Any ideas on an Interactive IVR?
4:55AM 1 How to use same h323-conf-id in incoming and outgoing legs?
4:47AM 0 PAP2-NA with Panasonic KX-TD1232 CE
4:44AM 3 4 port BRI options ?
3:15AM 3 secretary function
2:50AM 3 911 context, is this right?
2:33AM 1 Asterisk Realtime - How to enable the debug message for SIP users query
2:25AM 2 Inactivity restart segmentation fault
2:01AM 4 Portable USB headset for VoIP
1:17AM 3 Sip UA behind NAT
12:32AM 0 [OT] The Voice of Asterisk
12:11AM 0 Followup: MP3Player cmd issue (for Asterisk OS X users)
12:03AM 0 ISDN Data Calls stop working ?
Thursday June 2 2005
11:51PM 0 Connecting Asterisk with Microsoft LCS (Live Communication Server)
11:08PM 0 MP3Player could not play remote stream
9:59PM 0 Newbie MP3Player() cmd questions
9:27PM 2 Asterisk 1.0.7 on Gentoo
9:08PM 1 Teliax is DOWN
8:43PM 2 Ring but now audio on answer
7:36PM 0 IAX2 and Queues Problem?
7:29PM 3 Pricing for DS3000P
7:26PM 2 voip provider request
7:18PM 1 Asterisk RealTime Voicemail Not Working
7:08PM 0 Re: Asterisk-Users Digest, Vol 11, Issue 17
6:44PM 1 Zaptel not found error during modprobe
6:00PM 1 iax went away
4:18PM 2 Announce: Asterisk virtual configuration
4:13PM 0 what about dCap certification?
3:54PM 1 How to disable Digium card ?
2:54PM 3 CLUELESS NEWBIE needs help making an outbound sip call to PSTN
1:06PM 2 asterisk sipura and g726 codec
1:03PM 0 Host Authentication Problems
12:42PM 1 DID Routing over SIP
11:09AM 0 application sdp message and not answering call
10:59AM 0 Call Manager & Asterisk for VM - MWI not working
10:46AM 1 compile asterisk
10:27AM 0 Asterisk connecting to nortel CS 1000 as sip trunk Need help with final piece (incoming call) outgoing works.
9:51AM 3 asterisk on internet sip phone behind nat - doessomeone even have this working
9:39AM 1 asterisk on internet sip phone behind nat - does someone even have this working
9:21AM 2 bison/flex version warning
9:19AM 5 2 incoming lines and Asterisk@home...
9:14AM 1 Replacing SIP Trunking With IAX Trunking
8:51AM 3 ProSLIC 3210 version 2 is too old.
8:29AM 0 FW: Help with Kpn e1 settings please
8:28AM 0 connect to SIP trunk getting unable to create/find channel
7:34AM 1 Does Debian Bristuffed Asterisk work ignore Beronet cards ?
7:32AM 0 connecting to nortel CS1000 (half way there)
7:23AM 1 asterisk like modems access server
5:04AM 0 gsm call-hunting [OT]
4:59AM 2 Call Meeting VS Call Confrence
4:05AM 0 Astricon Europe :: Tutorial Agenda now published
3:51AM 7 a simple call to my girlfriend
3:21AM 1 H323 trunk with cisco gatekeeper
3:20AM 0 Script to test channel bank
3:07AM 1 trunk timing on 2.6.x
2:06AM 0 How to connect to Asterisk to IPTEL.ORG
1:29AM 1 Will my CPU/RAM be sufficient?
12:52AM 1 Newbie :Call Forwarding problem
12:09AM 0 chan_capi + mISDN + Fritz PTP
Wednesday June 1 2005
11:48PM 2 SIP or IAX
10:51PM 4 4+ Port FXS Analog Device
8:59PM 1 Voice recognition application - VoIP/Open Source
8:57PM 0 RTP Read too short
8:27PM 8 Asterisk Box as a Router, Firewall and DHCP Server
8:26PM 1 Incoming and Outgoing
7:59PM 0 Legacy PBX -> * -> Voip Calls problems
7:54PM 0 Issue with Not Capturing All Key Presses
7:11PM 1 Re: Obtaining Cisco Firmware painlessly and LEGITIMATELY?
6:48PM 1 Supervised/Attended transfers
6:40PM 2 IAX2 analog telephone adapter
6:38PM 1 does asterisk work with other processors
6:35PM 2 Realtime+IAX2 and RSA
6:20PM 5 Reccomendations for connecting to 3-4 PSTN lines?
6:14PM 2 Does Asterisk Realtime require the use of CVS HEAD ???
6:11PM 1 CVS HEAD won't compile for me
4:45PM 0 Re: Obtaining Cisco Firmware painlessly andLEGITIMATELY?
4:36PM 1 Newbie Question: HOWTO make outgoing call on SIP account from internal extensions?
4:34PM 0 Areskicc v2 login issues
3:43PM 0 Inject Audio into Existing Call
3:18PM 4 1.0.8 Release Candidate
2:54PM 2 wrong numbers message
2:34PM 0 chan_zap.c error
2:25PM 5 Broadvoice - Customer feedback
2:00PM 7 SNOM 360 extension lights
1:46PM 0 Cannot find module (NET-SNMP-EXTEND-MIB)
1:24PM 2 SetGroup CheckGroup
1:03PM 1 RFC2833 & firewall problems? (16-byte UDP packets)
12:41PM 2 voice-coloring with asterisk
12:34PM 2 A Way to Write DTMF Digits as text to CDR?
12:33PM 0 Cannot receive incoming calls via ISDN
12:12PM 0 Pri restarting randomly (TE110P or TE405P)
11:59AM 0 99% cpu on asterisk with chan_unicall and low traffic
11:54AM 0 Alternate DID
11:22AM 0 tellme hiring VXML
10:15AM 0 TDM400P Channels stop answering after some time .
10:09AM 4 list down?
9:45AM 1 Astcc does not work - no repeat metering
9:30AM 3 DTMF not working
9:27AM 0 astapi memory errors?
9:25AM 1 rxfax problems - cont.
9:20AM 0 Last of the servers forsale cheap
9:10AM 2 ARESKICC - Another issue
9:04AM 0 Segmentation Fautl / Core Dump with G.729
8:15AM 0 [q] About chan_misdn, latest mISDNuser and asterisk cvs
8:06AM 7 Pass-through
7:51AM 2 MOH Jittery Voice
7:46AM 0 Large installation with Asterisk
7:37AM 0 Setting up a TDM
7:10AM 1 Asterisk Google API applications - $4500 bounties available
6:51AM 1 FW: TellMe pay-as-you-go? - UPDATE
4:43AM 1 R: R: R: R: R: AT-320 + supervised transfer
4:33AM 0 Launching an application from within Asterisk
3:38AM 0 Problem with codec negotiation
3:05AM 0 hang up a SIP channel
2:57AM 0 debugging zap channel
2:54AM 1 send and receive MMS
2:23AM 2 Problems hanging up PSTN line
1:55AM 0 BT101 new firmware problem (
1:50AM 1 Unreliable DTMF detection with DISA on incoming Zap channel on bristuffed * and GSM gateway
1:08AM 0 Hardware questions
1:04AM 2 IVR Load
1:02AM 0 When to use 'Answer' and when NOT to...
12:49AM 1 Dynamic IAX Server
12:21AM 0 newbie with kphone and asterisk