asterisk users - Dec 2008

Wednesday December 31 2008
TimeRepliesSubject
4:28PM 0 End of 2008 Twitter Asterisk, Telephony and VoIP Directory
2:54PM 7 error on alsa
12:03PM 3 Friday VUC 12 Noon ET with Kristian Kielhofner: Identifying Asterisk Quality Issues
11:41AM 2 Troubles with AEL
 
Tuesday December 30 2008
TimeRepliesSubject
11:19PM 7 Attacking DECT
9:51PM 1 Only 8 messages from Asterisk-users Today?
4:34PM 5 Xorcom BRI state NOTOPEN
7:07AM 1 Newbie Polycom: Cannot conference with >10 digit 3rd party
5:53AM 0 connect a LAN server to a WAN server as a sip client
 
Monday December 29 2008
TimeRepliesSubject
11:48PM 2 Bug in contact header from Asterisk 1.6.0.3-rc1 ?
9:55PM 0 SIP host=dynamic help needed for CCME
8:55PM 1 DTMF does not work
6:13PM 8 AEL: how to check if variable is defined
6:09PM 1 Most Digium services are back on-line
4:50PM 1 Asterisk as MGCP client
3:58PM 0 Digium sites down for maintenance
3:10PM 7 Manager API
2:57PM 5 1.6, CDR and h extension
1:20PM 0 Background stress test
12:26PM 4 Join empty queue property
12:19AM 1 noise in Asterisk 1.4 and 1.6 versions
 
Sunday December 28 2008
TimeRepliesSubject
10:11PM 3 DTMF pass-through question
8:28PM 0 Audiocodes MP-11X configuration to work
8:21PM 2 Documentation DID + Asterisk
11:36AM 0 trunk hunt outbound
11:04AM 2 Problems with sip registrations through HP Procurve 7102dl
9:45AM 0 cypromis has invited you to Spokeo
 
Saturday December 27 2008
TimeRepliesSubject
7:43PM 2 help with DAHDI hangup on calling out.
12:32PM 2 asterisk 1.2 and openser 1.4
5:42AM 1 Wich gateway is much better?
4:37AM 4 Meetme - play the name
 
Friday December 26 2008
TimeRepliesSubject
10:10AM 5 Problem: no such extension 'xx' in context 'default'
3:50AM 1 asterisk dedicated server
 
Thursday December 25 2008
TimeRepliesSubject
6:28PM 1 1.6.1-rc4: extension "i" not working??
1:30PM 2 Zaptel vs DAHDI
 
Wednesday December 24 2008
TimeRepliesSubject
4:58PM 0 DTMF Problems
11:39AM 0 Need to implement Multi point Video Conferencing
11:28AM 0 Friday Dec 26th : VUC about Skype for Asterisk
12:06AM 0 DAHDI error
 
Tuesday December 23 2008
TimeRepliesSubject
8:49PM 5 Directory exists when * is pressed....but where?
6:14PM 17 AEL Variable Warning Messages
6:11PM 12 Dailplan code for holiday detection?
5:47PM 10 Pattern Matching
2:02PM 2 second trunk in extensions.conf
11:53AM 1 regarding query registered or online users fro out of asterisk
9:35AM 6 why does users.conf generate SIP peer and SIP user?
7:05AM 3 outging ---asterisk -bug
 
Monday December 22 2008
TimeRepliesSubject
9:59PM 0 interesting problem update
9:52PM 0 queue almost work fine
9:30PM 1 Asterisk SIP URi dialing
8:51PM 0 txfax/rxfax fun
8:30PM 7 IMAP Voicemail and Directory not working?
8:25PM 3 AMI and ExtensionState command returning bogus 'status' number
8:17PM 1 Web-driven SIP call thru Asterisk IPBX
7:07PM 1 No Audio
4:22PM 3 question on connecting speakers
3:45PM 3 Voicepulse down
3:37PM 7 Using Asterisk to measure call quality: Introducing Recqual
3:04PM 2 Manager API - standardization?
11:46AM 2 Disconnect queues members every night
10:46AM 3 Install app_rxfax and app_txfax in 1.4withLenny
9:58AM 0 Asterisk AGX addons
9:36AM 2 Install app_rxfax and app_txfax in 1.4with Lenny
1:39AM 1 Supermicro and onboard Intel e1000 Ethernet controllers ... no longer an issue?
 
Sunday December 21 2008
TimeRepliesSubject
11:13PM 0 IAX2 module hung and unable to unload chan_iax2.so
6:27PM 4 A method to determine PSTN Call Provider?
10:00AM 23 Good comparisons on cheaper VOIP phones
9:52AM 23 Asterisk and Dabatase
9:21AM 4 Outbound fax issues
 
Saturday December 20 2008
TimeRepliesSubject
5:54PM 1 IAX2 softphones keep ringing....
5:27PM 8 SMS text messaging capabilities
1:44PM 0 [FreeBSD 6.3] Upgrading Zaptel messed up host
10:03AM 5 how to set the busy signal usign softphones
4:35AM 3 Needs more cpu usage
3:41AM 2 how to get /var/run/asteris/asterisk.ctl
2:24AM 5 Setup ReceiveFax(), fax2mail, mime-construct - but now Sendmail :(
 
Friday December 19 2008
TimeRepliesSubject
11:59PM 0 New mailing list: digium-announce
7:03PM 5 Pre-routing manipulation of calls
6:30PM 0 OpenSer and MYSQL Lookup Queries!
4:54PM 5 Cut Through DTMF & caller ID on SIP phone
2:55PM 0 Dynamic Feature Playback acting on *both* channels?
2:28PM 0 Asterisk 1.6.1-beta4 released
2:10PM 0 realtime queue change ring strategy
8:36AM 0 Friday Dec 19th at Noon ET: Jazinga pbx appliance
7:56AM 2 Application Layer Gateway for SIP protocol
7:38AM 2 Conference with an AGI inside Queue for password change
3:37AM 15 Authorize & Microsoft SQL
3:02AM 6 Increase DTMF Tone Duration
 
Thursday December 18 2008
TimeRepliesSubject
11:18PM 0 dahdi-linux 2.1.0.3 and dahdi-tools 2.1.0.2 released
7:21PM 8 [Fwd: Asterisk client for ekiga.net NAT problem]
6:48PM 1 canreinvite question
6:21PM 10 Asterisk 1.4.22 Queues problems (Fifo or not ?)
5:29PM 2 Ghost in the Channel-Banks
3:53PM 0 Idle threads
3:49PM 4 Problems with ztdummy
3:42PM 7 DAHDI install dont need download of echo cancel
3:22PM 6 Dial timeout with SIP - how to set timeout for INVITE ACK
3:13PM 0 stream a file on a channel using AMI
3:01PM 0 Asterisk 1.4.23-rc3 Released
2:13PM 1 Call routing in voicemail
1:02PM 0 Latest AstManProxy [SOLVED]
12:59PM 0 Qualify = UNKNOWN
12:48PM 4 ael vs conf
11:44AM 2 (no subject)
11:07AM 5 Latest AstManProxy
9:08AM 4 zaptel-Error
8:58AM 1 Voice Packets latency
2:13AM 0 AsteriskNOW-1.5.0-beta1 Installation Error
2:06AM 3 Asterisk AGX addons compile issues
 
Wednesday December 17 2008
TimeRepliesSubject
8:17PM 1 Asterisk 1.4 to AS5400 using H.323 (ooh323) inbound working but outbound doesn't
7:16PM 1 Asterisk and NAT one way audio
7:06PM 2 ael queue gosub already has PBX structure??
4:28PM 2 How to tell when a issue actually gets in a released version
4:12PM 7 Install app_rxfax and app_txfax in 1.4 with Lenny
9:41AM 5 Alcatel OXE + Asterisk as external IVR
8:41AM 1 WTB: Digium 1 or 4 ports E1 Cards
8:14AM 5 libpri and NT-Point to multi-point
5:27AM 2 user entry as variables
 
Tuesday December 16 2008
TimeRepliesSubject
11:35PM 1 interesting problem
10:55PM 1 Some Good News for VoIP
10:47PM 7 RDNIS and asterisk
10:04PM 2 problems of DNS
7:14PM 7 Installing Asterisk v1.6 on Ubuntu Intrepid?
3:00PM 0 CDR and Agents Call recording
11:35AM 4 starting call recording using AMI or other stuff
8:50AM 0 ERROR[31152] chan_capi.c: Could not write to pipe for ISDN4#02
8:16AM 0 realtime odbc queue member cache problem
6:22AM 0 Netcomm V90s + Asterisk + conference
5:12AM 1 devicestate / inuse issue with 1.4.21.1
4:03AM 6 1.6 upgrade issues
12:37AM 4 Record CMD
 
Monday December 15 2008
TimeRepliesSubject
10:44PM 0 work in Chicago
8:58PM 9 Queue Question
8:35PM 12 tcpdum
8:26PM 0 ALG SIP
7:55PM 8 Dedicated Fax Line
5:16PM 1 D-channel errors and Channelbanks
12:38PM 0 UDPTL setup
5:55AM 14 Zaptel / TDM400P card stopped working
4:36AM 3 Variables for dial plan
1:24AM 4 1.6.1: iax trunk needs "dahdi timing" ??
 
Sunday December 14 2008
TimeRepliesSubject
12:09PM 2 HFC Single port Cards
11:24AM 0 Asterisk and VoIP Links and Sites
 
Saturday December 13 2008
TimeRepliesSubject
7:35PM 1 Asterisk video support
6:30PM 1 Please explain the meaning of the output of lsmod | grep ztdummy?
4:41PM 0 PickupChan()
2:08PM 0 G729 codec files
8:44AM 9 Asterisk / Hylafax
3:10AM 37 Country numbering plan resources
3:03AM 0 VOIP Origination with RDNIS
2:19AM 5 SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
 
Friday December 12 2008
TimeRepliesSubject
10:55PM 6 ring back tone
9:16PM 7 MSet()
6:10PM 0 multivoip bogen mp130
6:08PM 7 docs for rxfax in 1.4 or app_fax in 1.6?
2:23PM 3 prepaid solution
12:52PM 0 Are Cisco SIP phones still non-localizable with an Asterisk server ?
12:18PM 5 Asterisk Problem chan_sip.c: Call''from''to extension rejected because extension not found.
8:34AM 2 say I wish to run tail command on messages file to pick up if any "channels unavailable" messages appear.
7:11AM 3 Asterisk ignoring context= in sip.conf
5:25AM 3 get first month of trixbox free
4:46AM 13 Follow up on parking
4:14AM 2 call to mobiles and it is turn off
12:12AM 2 How to send a call to a Polycom SIP phone with NO callerid whatsoever
 
Thursday December 11 2008
TimeRepliesSubject
11:47PM 3 problem with Asterisk on Ubuntu
11:29PM 1 Virtual PBX
10:21PM 1 Weird problem with parked call expiration
10:01PM 1 Meetme realtime table structure
8:42PM 3 asterisk latency
8:42PM 5 MeetMe echo problems with more than two participants
8:24PM 0 OT: Looking for Dan Toma, author of Diax
7:59PM 0 SNOM Red LED on DND or unregistered Phone
6:47PM 11 service dahdi stop
5:28PM 0 Dialing plan Question
4:51PM 6 SIP CallerID Question
4:01PM 3 dahdi-monitor in France
3:11PM 1 DAHDI help
2:25PM 1 Asterisk spoken digits
2:06PM 1 Dial string required to drop any call not exactly 10 digits long
1:21PM 0 G729 reject call when no more licenses how to?
11:46AM 6 Asterisk dies when external access is lost
11:28AM 1 Dial command
8:59AM 1 top posting again [was: Re: CDR Design]
8:19AM 5 Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
7:40AM 0 Call Pickup (*8) / Attended forward and CallerID
5:25AM 1 CallingCard Applications
12:45AM 1 having problems with asterisk
12:35AM 4 Softphone recommendation
 
Wednesday December 10 2008
TimeRepliesSubject
11:49PM 2 G729 licenses
11:36PM 0 Asterisk 1.2.30.4 released
11:36PM 0 AST-2008-012: Remote crash vulnerability in IAX2
9:29PM 0 Park buttons on Polycom IP501/601
7:25PM 1 SendImage() to Polycom ip550 or ip670
6:36PM 6 asterisk video
5:21PM 4 Execute AGI after answered Dial() has ended
3:05PM 8 a problem on Ubuntu with Asterisk
10:59AM 2 DIY IP hardphone reference design
8:43AM 0 Replace music-on-hold on MeetMe with ringing sound
7:27AM 1 Improving Asterisk french prompts
 
Tuesday December 9 2008
TimeRepliesSubject
11:00PM 0 dahdi-linux 2.1.0 and dahdi-tools 2.1.0 released
10:23PM 2 SIP Registry Problems
9:00PM 1 Is tz working in voicemail.conf [general] section ?
8:20PM 0 Voicemail.conf : concise hour prompts [SOLVED]
7:17PM 10 Asterisk variable for SIP context
5:45PM 2 Func_ODBC question
4:47PM 0 Voicemail.conf: where to fin strftime manual entry ? [SOLVED]
3:14PM 2 Voicemail.conf : concise hour prompts
3:07PM 3 Voicemail.conf: where to fin strftime manual entry ?
2:13PM 12 about trasncoders
2:02PM 3 B410P, dahdi in TE, PtmP mode ?
1:11AM 0 IC3/FBI security announcement - your help needed
 
Monday December 8 2008
TimeRepliesSubject
9:10PM 0 need local upstate ny asterisk tech
9:03PM 6 'dialer' application to trigger call between hardphone and number
8:36PM 1 Voicemail and FreePBX
5:53PM 13 Stability unmatched!
5:27PM 5 DID provider in Sweden
3:40PM 0 asterisk survey
3:11PM 1 Anyone know which vulnerability specifically they are referring to?
1:22PM 2 meetme problem maybe connected to features.conf
1:17PM 0 MedHelp 34189
8:51AM 2 PRI span debug out put - failing international calls
1:21AM 0 Hobart/Tasmanian humans
 
Sunday December 7 2008
TimeRepliesSubject
8:01PM 0 Unexpected behaviour in ForkCDR
6:32PM 1 Echo Cancelation
5:02PM 2 International Calls still failing - Confused!
7:19AM 1 config from DB
2:55AM 5 Question on queue terms
 
Saturday December 6 2008
TimeRepliesSubject
7:16PM 5 Call Recording - Asterisk
2:34PM 0 Pika FAX
1:55PM 2 Sip Node w/ 4 wire audio & AT command set call supervision
9:47AM 1 Visual Dial Plan application: Recommendations?
3:57AM 2 Add volume sip accounts
 
Friday December 5 2008
TimeRepliesSubject
11:30PM 0 Asterisk, OCS and Caller-ID
10:30PM 4 Convert CallerID name to uppercase
10:08PM 4 AMI interface problem
8:28PM 3 Gosubs broken since r160626 (1.6.0 SVN) ?
5:18PM 3 All lines occupied notification from endpoint
4:33PM 0 ligion
4:04PM 2 Asterisk h323 module
4:00PM 9 CLI and choice of messages
2:54PM 0 top posting again [was: Re: CDR Design]
2:33PM 0 top posting again [was: Re: CDR Design] - Or was it top posting?
2:17PM 6 Using DECT phones as SIP phones?
1:49PM 23 top posting again [was: Re: CDR Design]
12:08PM 1 How to connect Asterisk-stat with Asterisk CDRs database
11:49AM 4 How to escape DTMF?
11:30AM 0 call loop in the network
10:52AM 1 Check variables on a live system - Is it possible?
10:13AM 0 H323 crashes Asterisk on high load
9:49AM 2 Linksys SPA922 - hangup problem
9:23AM 4 IAX trunk mixing
9:01AM 21 async agi question
2:01AM 0 remote phones, no audio to PSTN
1:31AM 11 Rate My Dialplan Contest Announced - Win a Phone or Copies of APSTel Visual Dialplan Std or Pro!
12:40AM 0 Web front end for Meetme?
12:24AM 2 polycom no menu
 
Thursday December 4 2008
TimeRepliesSubject
9:16PM 4 Packet size limit for HDLC?
5:05PM 0 407 Proxy Authentication Required
3:49PM 2 ISDN PRI settings for Telus BC network
2:45PM 5 Possible to get "Courtesy Tone" on attended transfer?
11:49AM 3 BT - ISDN30 - International Calls not working, everything else is fine :(
11:25AM 0 Changing the callerid of a mobile
10:44AM 0 Deadlock ? I hope i am wrong
10:04AM 2 OT - Is sourceforge OpenH323 active ?
9:23AM 4 set monitor_filename
9:18AM 2 Friday, Asterisk is 9 years old!
7:35AM 12 We think we are cpe but they think they are cpe too
1:22AM 2 Asterisk 1.6.0.1, IMAP Voicemail storage and temporary greetings.
12:36AM 1 app directory error: libc-client undefined symbol
 
Wednesday December 3 2008
TimeRepliesSubject
9:30PM 0 Asterisk 1.6.0.3-rc1 released
9:17PM 6 disable database
7:35PM 0 Mitel 5340 IP PHONE
7:32PM 5 asterisk ooh323 avaya (URGENT!!!)
5:25PM 0 canreinvite=yes -->problems
5:23PM 8 canreinvite=yes problem
4:33PM 13 Call parking
2:49PM 3 Dynamic loading changed in asterisk 1.4
2:45PM 0 asterisk-users Digest, Vol 53, Issue 5
11:01AM 0 problem with RTP
10:04AM 1 how to improve sound file quality?
9:25AM 0 chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to
8:46AM 2 Asterisk user client for customer service
8:13AM 0 What IRQ field from "dahdi show status" means ?
 
Tuesday December 2 2008
TimeRepliesSubject
10:29PM 6 callcenter supervisor system
10:05PM 15 Dahdi and ztdummy
8:40PM 1 hi from argentina
5:55PM 1 Problem with Bridge Application
4:26PM 2 Paging, Polycom and whispers
3:19PM 12 1.6, t.38 and zoiper - t38_udptl or t38pt_udptl ?
3:08PM 0 1.4.22 crashing on Solaris in ast_dynamic_str_thread_build_va
12:56PM 0 SIP Packets
12:07PM 3 MixMonitor and ChanSpy strangeness...
11:38AM 0 Log file warnings from chan_sip in build_reply_digest
11:34AM 1 Need help for transfer
11:10AM 0 How to get both channel ids from diaplan ?
11:09AM 0 Persistentmembers (Not working with restart)
10:28AM 0 [SPAM] - MySQL Error Message - Email found in subject
10:22AM 0 Using Dial M option from extensions.ael [SOLVED]
10:02AM 1 Dahdi, b410p and looping from 1 port to another - Email found in subject
9:49AM 2 Using Dial M option from extensions.ael
8:52AM 18 cepstral vs festival
8:01AM 0 New release of billing and routing software MOR
7:56AM 1 Is HPEC compliant with B410P ?
7:30AM 4 Which installation policy is behind Asterisk doc delivered with source code ?
7:21AM 2 func_odbc and hash problem
4:58AM 10 Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
1:57AM 12 Parking calls
1:19AM 2 Can asterisk work with a dynamic IP?
 
Monday December 1 2008
TimeRepliesSubject
11:47PM 0 Asternic Call Center and Asterisk 1.4 Queues
10:23PM 2 Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"
9:20PM 3 MySQL Error Message
7:46PM 10 [SPAM] - Re: [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject -Email found in subject - Email found in subject
6:49PM 1 Mitel 3300
6:34PM 23 OT: What do you guys think of this?
5:42PM 3 Difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
5:29PM 0 [SPAM] - Re: CDR Desgin - Email found in subject
4:45PM 3 [SPAM] - Re: CDR Design - Email found in subject
12:15PM 4 func_odbc questions
10:21AM 1 [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject - Email found in subject
9:44AM 1 [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject
8:29AM 0 how to configure free radius server with asterisk
8:12AM 3 Is using dahdi_genconf_parameters recommended to configure dahdi ?
7:39AM 2 Typo in dahdi_genconf man page