asterisk users - Oct 2010

Sunday October 31 2010
11:49PM 0 setting standard with asterisk
11:45PM 0 Junghanns douBRI miniPCI (b559) DAHDI drivers
10:19PM 1 billsec=0 when using Local channel
Saturday October 30 2010
10:22PM 2 Exceptionally long queue length queuing . . . .
10:10PM 1 Tormenta 3 (Tor3e) - Driver.
6:28PM 8 Under heavy attack
2:32PM 0 Gtalk and asterisk 1.6
Friday October 29 2010
9:55PM 1 Updating asteriskcdrdb with uniqueid field from Master.csv, Master.csv.1....Master.csv.5 - Must I bring all files together first?
9:14PM 2 Video based Asterisk Training
4:46PM 1 Asterisk 1.8 and character sets and AMI
2:57PM 0 New tutorial: Compiling Asterisk 1.8 on CentOS 64
11:01AM 0 Asterisk 1.6 Overlap dialling timeout?
10:37AM 1 trixbox - sip trunk with voipwise
9:08AM 1 BLF in Asterisk 1.4.*
8:21AM 2 MGCP
7:25AM 0 asterisk 1.6 and Firefox 4 Beta
Thursday October 28 2010
5:44PM 1 generic_odbc and ltdl are not available to enable ODBC support
3:28PM 0 Adhearsion 1.0 - Now Showing
3:08PM 8 SIP Load Balancing
2:43PM 0 SIP Communicator Friday at 12 Noon EDT
7:41AM 5 being bombarded with SIP packets
7:38AM 3 SIP client floods port 5060 and gets blocked
4:37AM 0 ss7_channel or ss7lib
1:21AM 1 what interface for ISDN-10/20/30?
12:48AM 0 [asterisk-biz] D-Link Wifi Phones
12:41AM 0 Intermittent failure when placing calls - unable to create channel of type SIP
Wednesday October 27 2010
10:37PM 1 Astribank Configuration Issues
7:43PM 1 Extension notation in default ViciDial installation
2:29PM 0 Send INVITES and REFERs from OpenSIPS to Asterisk with multiple Contexts
10:59AM 1 Asterisk died without any message, segfault
10:39AM 1 phoneprov
9:11AM 1 Fax Degium channel License
6:58AM 0 Asterisk Strange Problem while call received from customer On PRI.
5:59AM 0 Test numbers Worldwide
Tuesday October 26 2010
7:41PM 2 OT: SMS inbound
6:25PM 2 No media being sent in SIP call
6:00PM 1 need to be able to pass a call to the pstn from another pbx trunk
5:57PM 2 Trim the RDNIS
3:31PM 11 Auto provisioning from public server
12:57PM 3 Channel Bank ? Simple Switch Hangup?
11:38AM 1 2 HB8 cards in one server - first one is not recognized, the second is
10:59AM 0 IAX2 call dropped when a second call comes in
2:41AM 5 Mobile Phones and Asterisk
Monday October 25 2010
8:21PM 3 Extension Exists
6:14PM 2 Pop-up for MS Outlook 2007 recommended
3:52PM 2 Re : thousands Hangup per second /saturation of bandwidth
3:51PM 1 Re : saturation of bandwidth because of HANGUP
3:43PM 1 particular sip registry and outbound proxy
12:59PM 0 CDR updating
6:03AM 4 google voice + asterisk: calls made to GV# processed but weird
5:02AM 0 xpp_fxloader fails to load Astribank firmware on Ubuntu Lucid
12:28AM 1 E1 configuration
Sunday October 24 2010
10:23PM 5 Integrating Asterisk 1.8 with Google Talk and Google Voice
9:55PM 2 Chan variables for peer
9:47PM 0 baffled by defaultuser on aastra 9133i
5:02PM 1 How to have failover sip interface?
4:46PM 0 Can't hear MOH from PSTN [SOLVED]
4:44PM 0 Default MOH not working on 1.6.1 [SOLVED]
3:52PM 1 ISDN & SS7
3:35PM 1 Can't hear MOH from PSTN
1:42PM 1 Cepstral voice quality
11:18AM 0 Does any one uses PortSIP VoIP SDK?
10:07AM 0 [OT] Friday funny
Saturday October 23 2010
11:07PM 2 Just Take dCAP at Astricon?
9:26PM 3 Cepstral voice quality not good
7:33PM 0 NAT issues
7:14PM 2 1.8 Console Welcome Message
5:56PM 0 Parinya Sirisang invited you to Dropbox
4:31PM 3 Why such high latency on internal lan?
4:27PM 1 SipSak: Send SIP OPTION with password
2:36PM 4 Asterisk 1.8 IAX Registration
12:43PM 1 How to properly re-configure dahdi
12:35PM 1 Problem
12:31PM 5 a2billing muting "enter the phone number"
12:20PM 0 hangup delayed very much on fastagi appliaction of asterisk 1.6
12:07PM 2 B410P - BRI NT 100 Ohm terminator
9:43AM 1 RealTime Voicemail
9:35AM 7 Dial plan help
12:53AM 2 killing asterisk 1.8
Friday October 22 2010
9:05PM 1 E1 and T1 on the same card, or on the same server
9:00PM 0 CEL ODBC problem in 1.8.0
6:43PM 0 E1 and Pt on the same card, on in the same asterisk box
5:28PM 0 488 Not acceptable here
2:02PM 2 OpenVPN over TCP 1194 rather than UDP 1194 - Is there an adverse effect when running Asterisk?
1:48PM 0 Best practices to edit multi-versions config files ?
1:02PM 1 SIP Channel naming conventions
11:10AM 5 dials a trunk when off hook
9:44AM 3 Licensing of Default MOH
3:31AM 0 Counterpath Presence Patent and Android VoIP app
12:16AM 1 MS-SQL / Freetds -- func_odbc
Thursday October 21 2010
7:59PM 2 Incoming calls
6:03PM 1 Why high latency on internal lan?
5:41PM 0 Asterisk 1.8.0 Now Available!
5:30PM 1 Hardware Compatibility HP Proliant - Sangoma PCI Express
3:55PM 0 saturation of bandwidth because of HANGUP
3:41PM 5 SIP Blacklisting
3:35PM 2 1 way audio asterisk 1.6
2:56PM 1 Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician
2:40PM 10 Asterisk 1.80-rc5
2:27PM 1 Asterisk 1.8.0-rc5: Blind transfer failed, SIP REFER Method
1:31PM 1 asterisk 1.8 SIP register uri: peer field ?
12:41PM 2 dialing from asterisk console?
12:31PM 1 Busy detection in dialplan - Asterisk 1.6
9:16AM 8 Dial Plan Conf
7:59AM 2 DIALSTATUS always returns NOANSWER
5:24AM 3 Asterisk Realtime Billing Question???
12:49AM 1 How to kill AMI ORIGINATE on-the-fly
Wednesday October 20 2010
10:41PM 4 Email from Dialplan
9:56PM 2 Adaptive CDR and default fields
8:02PM 5 Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
4:58PM 1 SIP 401
4:04PM 1 2 step dialing
2:38PM 0 Audio Playback randomly stops
1:52PM 4 Recommendation for a new server
1:51PM 3 Using Calls Rejection Reasons
1:33PM 2 DAHDI weather quirk
11:05AM 2 Playback in the middle of a call though AMI
8:20AM 1 echo on TE122
7:30AM 1 Best way to recording the hold time for a Queue agent or an extension
7:22AM 2 Is Asterix right tool for me?
6:02AM 5 Queue member status - BUSY
3:03AM 3 dahdi_genconf
1:14AM 1 Parked calls drop asterisk-
Tuesday October 19 2010
10:50PM 0 Distortion and block on analog lines
6:40PM 1 E1 channels real time monitoring
2:46PM 1 dahdi vmware query
2:36PM 1 FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Monday October 18 2010
11:54PM 1 Asterisk 1.8.0 Release Candidate 5 Now Available
7:40PM 5 Same extension registering over eth0 and eth1
7:36PM 1 Recording
6:54PM 2 CEL Documentation
6:35PM 1 a2billing
6:03PM 0 Asterisk 1.8.0 Release Candidate 4 Now Available
4:43PM 0 How to check if Agent is logged into a specific Queue using dial-plan?
3:42PM 0 Problems detecting hangup
11:09AM 8 Asterisk to switch on electric heaters remotely?
10:59AM 15 SIP DNS SRV
8:57AM 0 How to execute Asterisk Functions in PHPAGI
1:23AM 0 app_swift for Asterisk 1.8
1:13AM 5 IAX2 works one direction, but not the other...
Sunday October 17 2010
9:18PM 4 Meetme
5:18PM 2 Error with Connecting Two Asterisk BOX with IAX
1:03PM 0 Good Day! 2010.
Saturday October 16 2010
9:36PM 6 Remote Unix Connection
9:35PM 1 (no subject)
8:59PM 3 Detect incoming fax on PSTN and route to fax machine on DADHI extension?
3:28PM 1 fraud advice (Also advice on using ipbanning)
3:06PM 1 DAHDI, PRI and callerid
10:42AM 4 How to find ".gsm" audio file length or duration
Friday October 15 2010
4:22PM 3 SIP - no audio behind nat problem
3:22PM 0 how to insert dynamic hostname into shared CDR database
2:53PM 4 Audio problems on cable modem link
1:59PM 8 drop dead fix
1:55PM 2 Echo on PRI after upgrading to Asterisk 1.6.2 from 1.2
1:38PM 1 app_meetme build option is XXX'ed out
10:17AM 1 warning diego viola the trouble maker for the world
9:00AM 2 Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)
3:13AM 1 Queue Agent Getting Additional Calls When on the Phone
1:10AM 8 fraud advice
Thursday October 14 2010
7:25PM 6 Audiocodes firmware
4:18PM 5 Routers that do not show external IPs...
4:12PM 0 AstriCon update - less than two weeks!
4:00PM 1 Default MOH not working on 1.6.1
3:43PM 1 Explain "core show translation"
3:01PM 2 clustering
2:17PM 1 Passing variables into macros?
1:35PM 5 How to connect asterisk PBX to PSTN
9:29AM 1 Using hint priority with LDAP extensions and users
2:46AM 1 advice re: Page() application
1:31AM 1 MySQL and Channel Event Logging
Wednesday October 13 2010
11:54PM 1 advice re: Page() application
10:48PM 1 Some give 603 Declined
7:40PM 3 call forwarding callerID
5:52PM 0 Unable to specify channel 5: No such device or address
5:50PM 1 SIP disconnects after 20 seconds behind NAT
5:11PM 4 checking CDR
3:26PM 2 Configuring & Setting up Asterisk
2:56PM 0 Getting last 2 Sip registrations of same user
2:46PM 0 innomedia ATA's
2:43PM 3 GXP-21XX
1:06PM 11 DMTF Mode
9:21AM 1 realtime users call problem
8:10AM 0 Asterisk Hangup Issue in Ringing State with Incoming call
Tuesday October 12 2010
8:02PM 1 sound file debug
2:14PM 0 REFER method
2:10PM 1 how to fake asterisk register?
1:51PM 1 src_mysql problem
12:12PM 0 Application Map Not Working
12:08PM 1 chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049
11:53AM 0 rtpip patch
Monday October 11 2010
11:37PM 4 SIP and ANI
10:21PM 1 MWI Assistance
10:14PM 2 user number in conference
8:20PM 2 Second time Parking issue
8:02PM 0 don't leave meetme conf if key pressed
2:48PM 8 Create channel bank with TDMoE
1:01PM 0 Synchron Playback to caller AND callee ?
12:36PM 1 iax2 users calls limit for outgoing / incoming
12:33PM 1 Call Failed Audio
9:48AM 1 About Action Originate
8:22AM 1 Quintum Tenor AX and Echo
4:08AM 1 OpenR2
12:57AM 1 Unable to find a codec translation path from ulaw|h261 to slin
Sunday October 10 2010
8:55PM 1 TDM 400p and Noise on the line
1:46PM 1 Modifying cid.cid_name in app_parkandannounce.c
4:54AM 1 Dahdi missing
Friday October 8 2010
11:25PM 0 SIP NOTIFY to make linksys/cisco SPA BLF go yellow
10:28PM 3 looking for a better ATA
3:06PM 0 Weird stalling of playback on IAX2 channels on 1.8svn
2:33PM 2 Weird stalling of playback on IAX2 channels on 1.8 svn
2:10PM 3 How to use Atxfer in AMI
9:12AM 1 Voice quality assessment in Asterisk
6:37AM 2 Polycom getting DCHP address from wrong VLAN
6:25AM 1 asterisk-users Digest, Vol 75, Issue 7
5:15AM 0 Radius client support
1:35AM 1 REINVITE with Auth Credentials has different SDP Codec
Thursday October 7 2010
10:57PM 1 asterisk router
8:41PM 0 Asterisk 1.8.0 Release Candidate 3 Now Available
7:07PM 2 Dahdi error
4:55PM 0 Voice drop out
3:36PM 1 RTP Read too short
3:31PM 0 How to change features.conf's atxfer dialing tone ?
3:18PM 0 convert g729A-g729B and vice-versa
2:34PM 1 Polycom: full caller ID?
11:54AM 2 401 Unauthorized with Snom but not with Zoiper softphone
8:52AM 2 SIP authentication - Thoughts please
8:44AM 0 Fw: asterisk > cisco gateway > westell > isdx
Wednesday October 6 2010
9:00PM 3 integrate Intertel Axxess with Asterisk
8:51PM 0 How to learn encrypted VoIP development for embedded systems
8:35PM 1 CALLERPRES() with Queue
7:50PM 2 SPA-2102 sending local IP instead of WAN IP in SIP packets
7:25PM 2 ADA: DOA?
5:39PM 1 AMI connection limit
3:07PM 1 2 way intercom recommendation for restaurant kitchens
2:03PM 1 using better quality wav or mp3 in Asterisk 1.2.x
1:11PM 2 Asterisk 1.8: Warning messages in CLI while putting a SIP-Call on hold
12:26PM 0 Page minimum number of extensions
12:15PM 2 Difference
11:56AM 2 AMI getting related channels in Ringing state
6:28AM 2 Which virtualization tech to get PCI assignment ?
6:14AM 3 How to test BRI lines energy saving mode ?
Tuesday October 5 2010
11:41PM 0 Web-meetme
9:13PM 2 Setting up realtime config.
9:08PM 2 Cisco SIP 8.5 and 9.0 Issues
8:40PM 1 Asterisk sharing a line with POTS handsets: how to interoperate cleanly?
8:16PM 3 Asterisk CDR Radius error
8:04PM 0 meetme don't play conf-invalid if room does not exist
10:12AM 5 Implementing more than one asterisk instance in the same hardware machine?
7:08AM 2 CDR record for call originated from CLI originate
3:48AM 0 Chage Asterisk 1.6.1 to 1.6.2
3:13AM 2 Checking SIP Headers existence and content
Monday October 4 2010
9:44PM 0 DAHDI 2.2.1, Asterisk - Channel unacceptable (6)
7:27PM 3 take input and store in variable
7:17PM 1 DISA does not accept "pause" from cellphones when upgrading from 1.4 to 1.6
6:48PM 0 session border controller
6:32PM 1 Registering Multiple Trunks to Service Provider
1:44PM 3 Module reload
12:26PM 1 asterisk-users Digest, Vol 75, Issue 2
10:10AM 1 Inter pbx communication via BRI
9:24AM 3 Phantom phone ringing
Sunday October 3 2010
8:19PM 3 SIP flood attacK
6:20PM 1 more condition check for gotoif
2:34PM 1 other end hangup
2:29PM 1 Flash WAV Player
Saturday October 2 2010
6:59PM 2 Attempts to hack Asterisk - What do these lines means
6:56PM 2 Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
4:24PM 4 minimum card for dahdi timing source ?
10:09AM 1 RE : Re: differential billing
Friday October 1 2010
8:50PM 2 AMI Originate
6:00PM 0 Looking for a PSTN DTMF echo test
2:08PM 2 No translator path exists for channel type DAHDI (native 76) to 256
12:58PM 0 Need some info on cmd Bridge (Confbridge)
12:49PM 1 debian/dahdi/zaphfc - Unable to receive TEI fromnetwork!
12:11PM 0 Asterisk 1.6.1 Realtime Extensions => Limited ?
10:37AM 0 debian/dahdi/zaphfc - Unable to receive TEI from network!
8:52AM 2 Asterisk/Realtime and MySQL