asterisk users - Apr 2009

Thursday April 30 2009
10:46PM 0 Voicemail Caller ID
7:22PM 6 Registration of 'cstore' rejected: 'Registration Refused' from: ''
7:12PM 1 test
4:37PM 4 Proper install order for Asterisk and it's related packages..
4:17PM 5 Asterisk and 4G
4:10PM 1 Wanpipe
3:45PM 1 ${CALLERID(name)} question
3:37PM 2 Question with Asterisk and call waiting ${CALLERID(name/num)}
3:31PM 1 rtsp help
3:06PM 1 FW: Update: HD Communications Summit in NYC
1:35PM 0 Saturday May 2 - Asterisk @ Global FSW Conference via VOIP - BerkeleyTIP - 21 Videos - For forwarding
12:44PM 5 need help on asterisk call forwarding
11:50AM 1 Asterisk or Zaptel Issues
11:31AM 0 Asterisk and Shoretel integration
3:34AM 0 valetparking.c
2:01AM 0 automon *1 not working; asterisk-
1:20AM 1 ExtenSpy d option 1.6
Wednesday April 29 2009
9:39PM 9 2nd Parking Lot
9:29PM 3 US Caller ID
7:14PM 15 What do I need to connect landline calls without telephony hardware?
6:32PM 5 Bounty for parking on <slot>@<context>
6:25PM 0 Verifone-Asterisk-AGI
5:46PM 3 Asterisk sudden crash
5:41PM 2 Something wrong with DAHDI signalling according to the CLI
12:49PM 5 problem in upgrading to
12:27PM 4 Replacement of Macro() with Gosub()
6:36AM 2 Cisco SPA525G
1:39AM 3 I am looking for a good source of Monterrey DIDs
Tuesday April 28 2009
11:39PM 0 Asterisk Now Available
11:37PM 0 Asterisk-Addons Now Available
11:36PM 0 Asterisk-Addons 1.4.8 Now Available
8:35PM 2 Asterisk 1.6 and CDR/MySQL
6:19PM 3 asterisk -C option not honored 100%
5:00PM 1 Asterisk-Verifone-Agi
3:49PM 5 How to get PBX's clock with AMI?
1:04PM 0 Explain when DIALSTATUS is set to CANCEL
10:15AM 3 no source on calllogs
9:35AM 1 Call recording - posible to remove recorded file at the end of the call
7:27AM 3 finding the right amd.conf settings
12:18AM 4 POS modems
Monday April 27 2009
9:33PM 7 Who has the clever Polycom upgrade system?
9:26PM 1 Where I get free VoiP-in numbers?
9:08PM 2 IPv6 support?
8:41PM 1 Packet2packet bridging while in sip.conf canreinvite=no
6:38PM 0 SIP infrastructure
5:29PM 3 Change Termination of Read Command
4:17PM 2 No format for saving voicemail?
3:09PM 4 Diference between volume of mp3 and wav files
2:20PM 0 Going to AMOOCON?
9:36AM 3 music on hold using mms
6:37AM 39 [UK SPECIFIC] DAHDI and a OpenVox Card
2:57AM 3 Video Conference Software (Open Source)
Sunday April 26 2009
11:51PM 1 Error, Clue to what?
7:03PM 1 1.6.1: "DNS error" but ping works
6:52PM 1 sipgate doesn't work with sipgate anymore
6:28PM 1 file.c:655 ast_openstream_full: File /tmp/winkel-gesloten.alaw does not exist in any format
2:26PM 4 1.6.1: menuselect has problems with x86_64 ??
12:33PM 4 Digium fax force T38?
9:48AM 0 FW: issue with sip 180 responses
2:21AM 10 Digium fax failing
Saturday April 25 2009
2:37PM 4 Outgoing Queues
10:38AM 0 Information required about ADM
10:03AM 7 64bit: any problems with asterisk?
7:35AM 0 sip.conf RTP settings
4:43AM 1 Callweaver/Asterisk 'outgoing' spool
4:01AM 2 Can't dial out until I dial in once
Friday April 24 2009
10:30PM 9 Asterisk & EC2
9:46PM 1 Digium Fax for Asterisk
8:19PM 2 voicemail number of rings
5:55PM 0 dahdi_tool reports that dahdi_dummy is UNCONFIGURED
5:03PM 0 Dialtones as Progressinband
4:06PM 1 Asterisk 1.6.2 Beta
2:33PM 0 Duplicating existing PBX function
1:59PM 3 listen to prompt before bridging call.
1:55PM 1 FOP and UserEvent()
10:09AM 3 timing source problem
7:39AM 0 Friday Apr 24 @ 12 Noon: Wideband, or HD Voice as Polycom calls it
7:35AM 2 function originate
7:07AM 3 Feature request: manager show events
4:03AM 7 cheap CHEAP ata
2:22AM 1 Hangup Detection After Originate (Asterisk Manager API)
Thursday April 23 2009
11:54PM 2 Dahi-tools Compilation on Ubuntu/Xen
10:01PM 0 about Asterisk and AudioCodes FXO/H323
8:56PM 4 BLINDTRANSFER and SIP hardphones
8:43PM 5 want to set up text based "adventure" for asterisk
8:30PM 0 dial and transfer while ringing
8:27PM 9 Record in mp3
8:08PM 1 Convert file in GSM codec to G729 codec
7:14PM 8 Asterisk on Mac OS X
6:53PM 0 Libpri-1.4.10 Released
5:49PM 2 CDR issue
4:26PM 6 AGI PHP script
3:00PM 21 AMD Not Working
2:38PM 0 Fritz USB 2.1 on Asterisk 1.4.22 / trixbox
2:24PM 4 Dial-out via AMI
2:20PM 0 Do I need G729 codec for wholesale ?
1:18PM 2 Zaptel Not Releasing Channel (PRI)
1:01PM 2 Asterisk Double Invite
12:49PM 0 UserEvent doc : is Uniqueid mandatory in 1.6
11:49AM 1 Cause 34 still there
11:12AM 4 Asterisk Capacity
9:50AM 24 Compact, fanless appliance?
9:46AM 0 Howto see the source ip address of SIP call in cli monitor
9:29AM 2 Asterisk and HUD server
8:53AM 3 Parked calls for multiple customers
3:10AM 2 voice quality
Wednesday April 22 2009
11:42PM 40 Step-by-Step Asterisk and Cisco 1760 Help
9:18PM 0 random hangups: how to debug?
8:19PM 0 how to know the channel from the iax phone side?
6:34PM 2 Upgrading from to breaks sql queries with backslashes?
5:07PM 0 HI
3:56PM 2 Conference problem
2:00PM 0 Zaptel tone debug
1:31PM 1 How to get to 10.000 open calls
12:55PM 0 [asterisk-dev] How to get to 10.000 open calls
10:26AM 1 Should you use UserEvents for monitoring calls ?
10:09AM 0 E1 not synchronized
8:57AM 0 mISDN DTMF endless tone
8:12AM 1 CDR feature not working properly for "failed call attempt"
5:55AM 2 Queue() Ignore Hangup Request
5:14AM 14 Asterisk routine maintenance activities
1:58AM 2 Faxing and TIFF files
Tuesday April 21 2009
8:42PM 0 Asterisk Now Available
5:48PM 1 run dialplan when open line
1:53PM 4 Polycom wideband codecs?
12:36PM 1 Should I go for Asterisk 1.6 ?
11:13AM 8 Asterisk Database
10:57AM 0 Cleared Event Log
10:02AM 13 Asterisk process ended
Monday April 20 2009
10:02PM 7 Peer 'iaxfax' is now UNREACHABLE! Time: 3
7:13PM 3 Asterisk PA system with cepstral
5:33PM 3 AstDB & MixMonitor queries
2:45PM 4 Voice mail does not contain a time?
2:31PM 5 Execute after hangup
1:48PM 0 freezing randomly - what files to use to downgrade
12:24PM 1 Zaptel to Dahdi
9:34AM 1 T38 fax failing
7:47AM 1 Asterisk 'outgoing' directory
6:10AM 2 About Asterisk 1.6 web GUI
3:50AM 4 Asterisk 1.4 to 1.6 extensions.conf
3:06AM 0 Note for all regarding Asterisk Addons
12:29AM 1 Asterisk Addons - part 2
12:01AM 1 Asterisk addons - disable H323
Sunday April 19 2009
7:55AM 2 issue with sip 180 responses
12:04AM 11 asterisk- voicemail: Segmentation fault (core dumped)
Saturday April 18 2009
11:01PM 0 Astlinux 0.6.5 upgrade released
7:32PM 1 Insecure=
5:21PM 5 dialling multiple extensions in an internal context
2:48PM 3 Digium Fax for Asterisk questions
11:26AM 0 do i need to install libpri
4:40AM 0 Callweaver TXfax queuing
1:23AM 1 Origination and Termination
Friday April 17 2009
11:57PM 2 Sangoma A104d and Adtran 850 problems
7:51PM 0 Quick and Dirty HOW-TO for Enum on Asterisk 1.2?
6:47PM 34 Here is Step by Step Example of Asterisk PBX System Install and configuration
4:51PM 8 opening 2 and more channels on 1 SIP account
4:50PM 3 Getting Zaptel and Asterisk Link, also Dahdi
4:20PM 5 Alcatel OmniPCX Enterprise + Asterisk with E1
2:28PM 2 Jabber and Presence
1:37PM 1 2BCT last mile... Hopefully
12:22PM 2 - weird freeze
12:14PM 3 Digium G.729 licenses
11:52AM 5 MOH always plays from the start
6:30AM 1 how to call forward on 1.6
3:12AM 0 Canreinvite after media connection
3:08AM 0 Stun clients and canreinvite
Thursday April 16 2009
7:57PM 1 sending AT commands through the SIP channel to the end device?!
7:12PM 6 Sequential Ring Groups?
6:50PM 1 AMI IAXPeers
6:44PM 1 Set CDR(src) from dialplan
3:06PM 17 Simultaneous Calls at a time
2:52PM 1 Connection to non-human numbers
2:08PM 1 Problem transferring calls between Cisco 7940 with SIP firmware
11:17AM 1 Can Asterisk bridge between a SIP client and a Cisco Call
10:59AM 9 How to send "404 Not found" SIP reply?
9:57AM 0 mISDN ports and dstchannel CDR logging
9:36AM 1 AGI Programming
9:22AM 1 ISDN from Macau CTM
8:37AM 5 Remote BLF / hint on IAX2 trunk
8:09AM 0 reinvite problem
6:50AM 1 Friday Apr 17th @12 Noon ET: Digium's Open Source Asterisk Support
2:41AM 2 TDM2400P dial tone is not present on phones, but the phone ring with incoming calls
Wednesday April 15 2009
3:59PM 10 inbound filed
2:01PM 1 pickupexten *8
9:30AM 0 chan_mISDN with asterisk version 1.4.22 + codec negotiation patch
9:30AM 2 What is "WARNING: Got 200 OK on REGISTER that isn't a register"?
9:25AM 2 astcanary not exiting in asterisk V1.6.1
Tuesday April 14 2009
11:22PM 0 RTCP ports
9:52PM 18 2B Channel Transfer on XO-based T1
6:47PM 0 Gxp 2000 softkey question
6:30PM 0 Asterisk Dial Pagers And Enter Callback Numbers
6:19PM 0 FW: Asterisk-beginner : cannot make phone calls using Asterisk
5:14PM 5 Ring All Queue
4:55PM 1 Vacation reply user nuked
4:42PM 2 OT - snom phone question
4:05PM 3 What means? Correct auth, but based on stale nonce received
3:41PM 0 How are you doing recently?
2:39PM 11 MOH
12:30PM 0 SRTP testers needed
11:39AM 9 .GSM -> .WAV (or ,MP3) Conversion
11:00AM 31 T38modem in loopback mode does not work on asterisk
10:04AM 21 Exit Dial Application
9:48AM 0 OT - Define what substitution is ...
8:37AM 3 SIP and FW settings
6:16AM 7 Ignoring time spent waiting in queue in CDR
1:29AM 4 dynamic menus in dialplan
12:28AM 3 Changing menuselect values from CLI and not TUI
Monday April 13 2009
10:19PM 2 wanpipe Compiling error
9:32PM 14 duration of rfc2833 generated dtmf
8:49PM 0 Re: Asterisk-beginner : cannot make phonecalls using Asterisk (update)
5:37PM 0 MySQL queries
4:18PM 45 Asterisk-beginner : cannot make phonecalls using Asterisk
2:30PM 2 Agents on asterisk
12:33PM 0 Clock problem with TE122
11:05AM 3 Send Re-invite from Dialplan application?
11:02AM 0 Sending Re-Invite with Dialplan application?
10:11AM 0 opensips and asterisk canreinvite
4:19AM 3 FAX reliability
12:30AM 8 retransmision error con asterisk
Sunday April 12 2009
2:38AM 0 problem with asterisk
Saturday April 11 2009
8:44PM 1 asterisknow 1.5 with X100P and TDM400P
4:04PM 2 [OFF TOPIC] wich virtualization solution to use?
10:29AM 5 Voicemail Greetings Will Not Save
7:04AM 2 [astersik-users] ss7 consultancy $1000 USD
3:04AM 6 Is there documentation explaining res_config_curl?
2:55AM 2 OT XEN asterisk and a digium board
Friday April 10 2009
6:22PM 0 one-button call parking/pickup on Asterisk with Polycom phones?
4:04PM 8 IVR Survey
12:25PM 0 Asterisk 1.4.24 and Gtalk audio failure
11:15AM 0 Friday April 10th: Google Voice, Asterisk Open-Source Support, etc.
6:06AM 5 Can Asterisk bridge between a SIP client and a Cisco Call Manager server?
4:54AM 0 IVR and DTMF
Thursday April 9 2009
10:41PM 15 DTMF
9:00PM 15 Looking for good IAX ATA
8:59PM 15 MeetMe not working - was before
4:45PM 1 Check sip availability
4:36PM 7 T.38 ATAs
1:44PM 8 asterisk command line problem
1:42PM 2 notifyringing=no does not work
9:12AM 0 AstManProxy and broadcast
2:50AM 3 Softphone question
Wednesday April 8 2009
11:06PM 6 Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA
9:27PM 2 Perl AGI
5:19PM 6 Zopier Client
3:58PM 1 __ast_read: ast_read() called with no recorded file descriptor
1:13PM 2 Asterisk Trunk billing
7:05AM 5 Siemens Gigaset Phones get mute function.
6:44AM 0 Asterisk and Voice Recognition Sphinx
4:56AM 1 Call Pickup Works w/Linksys ATA, not with Cisco 7940G
2:30AM 0 chan_mobile sms compatible phone
Tuesday April 7 2009
10:05PM 2 dahdi_dummy: Unable to register DAHDI rtc driver
7:53PM 5 i have a probleme and my asterisk and ovh
7:49PM 3 Fwd: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel commands?
3:54PM 8 Best Practice Advice?
2:38PM 9 AEL2, BASE64_DECODE and hexadecimal
2:28PM 2 immediate=yes is populated to all channels
2:26PM 6 Grandstream blind transfer issue
1:51PM 0 Zaptel connectivity issues
1:00PM 4 Logging Asterisk console
9:41AM 1 OT - SIP MESSAGE, newline chars and formatting
8:16AM 5 app_backticks and 1.6
5:23AM 2 asterisk and patton
12:41AM 1 Australian NBN network announced
Monday April 6 2009
10:11PM 4 One way AUDIO
8:55PM 2 Zaptel Config
7:55PM 9 Hacked
7:17PM 0 Asterisk Now Available
7:14PM 0 Asterisk Now Available
7:04PM 5 Sangoma and BT single lines
6:48PM 2 IMAP Voicemail - can't get messages. Arrgh!
4:13PM 3 IOS Interface
3:00PM 1 Douds it
2:40PM 1 Relay ringing sip message 180
1:32PM 0 OT - Call forwarding services for corporate users
1:01PM 7 IPkall
12:45PM 5 SIP Registration and INVITE question
12:39PM 12 25-50-100fxs
10:00AM 3 fail to retrieve the calling party information
9:45AM 0 app_queue.c: No one is answering queue
8:42AM 0 Mysql cache delay
7:24AM 1 Off-topic: SIP DTMF most supported method
Sunday April 5 2009
7:43PM 2 what can we do with lost voice packet on a congestioned VPN?
5:07PM 14 Inexpensive device for bandwidth management
Saturday April 4 2009
10:07PM 2 off topic - voip providers raided by FBI for unpaid telecom bills:
5:51PM 0 OT - Car Warranty Calls - OT
12:31PM 4 Advice
8:02AM 0 TODAY April 4 -Global FSW Voice Meeting BerkeleyTIP -Linus, Guido, Shuttleworth...
Friday April 3 2009
11:27PM 7 Live Support function?
9:41PM 3 Grandstream surveillance devices
7:42PM 1 conference calling
7:39PM 7 Eicon Diva 2.01 PCI Passive BRI ISDN card
6:57PM 13 Using multiple 'peer' identities on one phone with 1.4
5:45PM 2 SIP Warnning Message
5:43PM 8 Radio interfaces for Asterisk - ISO image distro
5:14PM 6 ISDN Timer T309
4:02PM 1 Seg Fault after upgrade to Asterisk
2:40PM 3 Bridging Avaya IP systems and Cisco IP system
2:30PM 5 New ViciDial Call Center Suite Release: 2.0.5
1:24PM 1 Unichan wtih Te201p alarms
1:22PM 2 agi no longer working with 1.4 svn 186229
8:14AM 0 VoIP Farm
6:23AM 0 Asterisk and Call Manager
Thursday April 2 2009
11:36PM 7 Dahdi, TE220 Device, and Asterisk Problem
11:22PM 5 Ring group howto
10:38PM 3 Simple Queue question
9:40PM 1 Anyone actually built h323plus on Fedora?
9:38PM 3 FXS Line Voltage When Dahdi/Zaptel is off?
8:57PM 4 2-3 Calls at a time
8:53PM 3 problema con una x100p
8:51PM 7 cant get a x100p works
6:59PM 0 AST-2009-003: SIP responses expose valid usernames
6:51PM 0 Asterisk 1.2.32,, and Now Available
6:25PM 11 Asterisk G729 codec...
6:20PM 0 VB6 to HUD Pro Integration
5:22PM 10 meetme dahdi and zaptel
4:44PM 1 T1/PRI ignore answer signal
4:39PM 0 Magic List: Thanks Shain Rufeel & Danny Nicholas.
4:22PM 4 opermode=?
4:07PM 3 Asterisk + Cisco Call Manager
3:54PM 4 Nothing at /proc/zaptel with new Digium TE201
3:45PM 2 SIP vs RTP destination IP
3:45PM 3 fxotune and the bug
3:37PM 5 FXO Ignore ring
3:17PM 0 Asterisk SIP trunk to Cisco IAD2400
2:01PM 2 activate telco redirection service from Asterisk
11:39AM 7 Mountain ahead of me!
11:06AM 35 Xorcom and Doorbell
10:38AM 1 Friday April 3rd Gizmo, OpenSky, Skype for Asterisk, SIP for Skype - where are they?
7:30AM 1 Trying to test my voicemail
3:46AM 18 400 calls at g711 how much cpu power
3:27AM 0 SIP topology hiding
Wednesday April 1 2009
9:19PM 0 Asterisk-Addons Now Available
8:55PM 0 Asterisk Now Available
7:45PM 6 SIP Context Confusion
3:45PM 2 Trunk SIP and configuration
3:26PM 1 login-logout asterisk
3:14PM 7 Extract a MOS value from Asterisk CDR
1:06PM 2 Remote host can't match request CANCEL to call
12:18PM 0 stress asterisk voicemail
4:59AM 0 IAX2 transfer=force