asterisk users - Jul 2008

Thursday July 31 2008
TimeRepliesSubject
10:50PM 9 Slighly OT?.. headset for Linksys SPA922
10:14PM 0 Friday August 1st @ 12 Noon EDT
9:15PM 4 Panasonic Door phone monitor to Asterisk box?
8:20PM 4 comparing pots solutions for asterisk
8:09PM 1 need creative solutions for number portability
6:10PM 8 AA50 Failover
5:55PM 0 [asterisk-dev] Astricon 2008 updates: keynotes, content, contests
5:53PM 1 list of minutes spent on SIP phone calls?! any advice?!
5:25PM 0 Announcing the release of Web-MeetMe 3.0.4
5:16PM 0 Unregistered indication country
3:13PM 0 Asterisk CDR "**Unknow**" as channel name
1:56PM 6 Setting up ring group
1:55PM 1 sip registration timeout/expiration
8:52AM 1 Anjelina Jolie XXX Video Free.
 
Wednesday July 30 2008
TimeRepliesSubject
9:09PM 5 Custom Filename for Incoming Agent Calls
8:18PM 0 It's telling me too much...
8:07PM 0 RES: GotoIftime
7:46PM 7 GotoIftime
5:17PM 1 Asterisk Realtime still reads from .conf files
5:02PM 0 Creating Call permisions in Trixbox CE 2.6.1
4:19PM 1 ARA with MySQL or PostgreSQL
3:49PM 1 Outgoing calls authentication
1:19PM 1 re-distributing E1
9:31AM 2 Whitepaper: How and to whom sell VoIP
1:37AM 0 mpg123
 
Tuesday July 29 2008
TimeRepliesSubject
11:57PM 2 Multiple Asterisk SIP Server/client connections
8:24PM 0 Fallback on a fallback
8:02PM 7 Callerid Woes
7:12PM 0 Recommend Bluetooth adapters for chan_mobile?
6:35PM 0 Auto Dial Application
6:17PM 1 soundpoint 301 power adapter output?
5:04PM 2 Purchasing Digium IVR Prompts.
4:26PM 4 Addressbook solution for Cisco 7961?
2:17PM 3 interactive IVR
2:04PM 2 Asterisk SIP configuration
1:45PM 1 asterisk+ fax-to-mail
1:32PM 0 Azurn International
1:01PM 0 asterisk stops sending qualify
7:09AM 1 Outgoing calls
6:59AM 1 Need help with implementing prepaid in asterisk
5:19AM 4 One way voice after call transfer (bugs 9305, 13120)
3:48AM 49 Newbie in China: Red alaram in Zaptel for E1
 
Monday July 28 2008
TimeRepliesSubject
10:26PM 0 imap voicemail is being sent to the wrong imap account
10:02PM 4 Remote Support
9:42PM 1 IVR Direct Dial Extension
7:04PM 1 Slow Playback of Recorded Files
5:08PM 1 SIP sprials and "482 Loop Detected"
5:04PM 0 Common Inter-Queues Leastrecent Strategy
4:53PM 0 How to find out RTP UDP port of active calls
4:19PM 6 Callcentric Issues
2:14PM 0 custom configuration with appliance aa50.
1:42PM 1 Line 0005 cannot be answered?
8:58AM 10 TDM400P FXO not seeing ringing after software update
4:08AM 0 vmail.cgi and users.conf
 
Sunday July 27 2008
TimeRepliesSubject
6:28PM 3 HASH, HASHKEYS, ClearHash explanation
9:00AM 7 OT - How to test tftp for phones provisioning
 
Saturday July 26 2008
TimeRepliesSubject
8:49PM 5 Visual Dial Plan
6:15PM 0 CME/Asterisk Voicemail Problems
3:34PM 0 Using manager originate and Dial() once inside dialplan
10:51AM 6 Need Help Regarding Asterisk
10:18AM 9 announcement server using asterisk
 
Friday July 25 2008
TimeRepliesSubject
10:51PM 0 Slightly Off Topic: Cisco & Premisys Slimline
8:32PM 3 Very loud noise on TDM400
8:06PM 0 Call Center Type Recording
7:21PM 0 Friday's conference didn't happen... again
4:17PM 3 openSUSE Asterisk Packages
3:18PM 4 Call files with a timer?
2:47PM 0 console/dsp seg fault
2:05PM 0 AstManProxy - Blocked during 1.22fork alias rc1 install
11:23AM 1 I Win The Ooma Bet
10:53AM 9 IAX to work on two ports: 4569 and 4570
9:25AM 8 AstManProxy - Where to download From ?
12:40AM 0 Arabic IVR
 
Thursday July 24 2008
TimeRepliesSubject
11:59PM 2 finding out on hold channels
11:16PM 1 different gains per channel?
9:30PM 3 Click to Dial
8:42PM 1 Implementing an Asterisk Server behinda MeridianNorstar
5:41PM 6 Cisco Call Manager to Asterisk conversion
4:25PM 0 Friday at 12 Noon EDT (9 AM Pacific) Asterisk and VoIP User Groups Worldwide
3:45PM 0 CallerId show with IP address appended
1:50PM 2 Asterisk automatic hold
1:39PM 1 T1/PRI dialing
1:36PM 0 Automatic Redialing feature
1:23PM 22 Acceptance testing of a new PRI
12:34PM 13 Audiocodes MP-11X configuration to work with Asterisk
10:04AM 3 Realtime + SIP + MySQL: md5secret BROKEN
9:25AM 11 IP door opening devices
3:45AM 0 Tomato = One Way Audio; Linksys = OK ????
2:37AM 3 Zaptel won´t recognizes sources installed
 
Wednesday July 23 2008
TimeRepliesSubject
11:53PM 1 Raw asterisk x FreePbx .conf
10:57PM 2 Connect Asterisk PBX to Traditional PBX and retain functionality
9:35PM 13 Implementing an Asterisk Server behind a Meridian Norstar
9:03PM 1 Broadsoft Sip provider
6:53PM 0 need help setting up dundi
1:45PM 0 problem with asterisk 1.4.21.1 and h323
12:49PM 5 Trouble Playing message file via Perl AGI
10:33AM 3 next priority from Dial in Asterisk 1.6
8:12AM 3 1.4.21.2: Linking res_crypto causes segmentation fault.
5:55AM 9 How can I Disable call-waiting
4:51AM 8 sometimes extensions can't be called
 
Tuesday July 22 2008
TimeRepliesSubject
11:32PM 8 Suddenly my Asterisk Box Hanged up all calls
11:23PM 0 Asterisk 1.4.21.2 and 1.2.30 Released
11:16PM 0 AST-2008-011: Traffic amplification in IAX2 firmware provisioning system
11:15PM 0 AST-2008-010: Asterisk IAX 'POKE' resource exhaustion
10:42PM 0 [Fwd: Re: what is the magic needed from upgrading from 1.4 to 1.6]
10:17PM 7 3-way calling for IAX channels
7:58PM 1 Looking for a more robust Click to Dial/Web Dial solution than AsteriDex (potential for a bounty!)
7:53PM 7 Call Recordings...
7:44PM 0 ?? Vitelity dtmfmode=rfc2833 started working!
4:57PM 1 Voicemail email to alternative ports...
4:54PM 0 Problems w/Asterisk Realtime + MySQL + SIP [SOLVED!]
4:23PM 0 Vitelity dtmfmode=rfc2833 started working!
1:36PM 3 issue with high latency
12:52PM 28 Cisco vs Asterisk
 
Monday July 21 2008
TimeRepliesSubject
11:22PM 3 Heavy Load Asterisk Array
11:18PM 2 RTP Packets Going To Wrong IP Address
5:28PM 0 Cascading Asterisk PBX
5:10PM 14 what is the magic needed from upgrading from 1.4 to 1.6
4:50PM 5 increase ring time out
4:19PM 5 Asterisk Recording tools
4:10PM 5 Overlap dialing via SIP
4:01PM 1 [Posible Spam] asterisk-users Digest, Vol 48, Issue 59
3:59PM 5 Help with dial plan
3:18PM 1 Recommend quality wholesale termination - Singapore and Sydney, Aus
3:12PM 3 Option 't' on DIal
2:52PM 1 Incompatible voice frame panic!
1:16PM 0 zaptel and callerid in ESTI DTMF
12:39PM 7 queue members randomly become paused after upgrade to Asterisk 1.4
9:29AM 1 Problems with IAX on heartbeat provided ip address
8:50AM 9 OSLEC vs HPEC vs Octasic
8:11AM 8 Problems w/Asterisk Realtime + MySQL + SIP
4:12AM 0 Required an Auto Dialing Solution
3:25AM 0 New Bridge App/AMI Command in Asterisk 1.6?
 
Sunday July 20 2008
TimeRepliesSubject
7:56PM 1 Queue() AGI Bug ?
5:13PM 0 asterisk-users Digest, Vol 48, Issue 58
9:57AM 5 Dialplan Action on Authentication
5:48AM 1 conference bridge
3:32AM 0 asterisk-users Digest, Vol 48, Issue 57
1:45AM 3 Question about stopping Asterisk
1:41AM 1 Stop vm-intro being played
12:55AM 2 New Bridge Command/Event in 1.6?
 
Saturday July 19 2008
TimeRepliesSubject
5:10PM 1 asterisk-users Digest, Vol 48, Issue 56
4:41PM 5 Echo Issue
3:22PM 4 Not a valid SIP contact - Asterisk 1.4.21.1 & Mitel SIP phones
2:54PM 1 Beginner Questions part II
11:41AM 2 Explication for ast_safe_system
5:13AM 3 Changinf Polycom-501 config server from remote?
3:48AM 8 OT Astricon/Digium Beach Ball Mailing
12:52AM 1 going from 1.4.21 to 1.6 beta 9
12:19AM 0 asterisk not converting DTMF from INFO to rfc2833
 
Friday July 18 2008
TimeRepliesSubject
11:34PM 0 Asterisk 1.4.21.1
8:49PM 0 Announcing AstriDevCon 2008!
7:21PM 2 TOS and security
7:16PM 2 1.6b9 Audio Issue
6:17PM 1 Beep on transfer
5:39PM 0 automon follup #2
5:11PM 4 Colorado Asterisk User Group Forming
5:08PM 1 automon followup
5:02PM 1 automon=>*, Dial(, , Ww), rfc2833, canreinvite=no, but...
4:19PM 0 Asterisk Recordings
4:16PM 5 GotoIf Problem
2:09PM 0 [asterisk-dev] How Register to ONE SIP provider with Multi Accounts
2:05PM 0 IAX + Inidication
1:48PM 4 DID - Panama
10:11AM 0 Asterisk Video on Hold
 
Thursday July 17 2008
TimeRepliesSubject
11:46PM 0 Help for an IAX_Client-based softphone
9:15PM 1 AVM Fritz BRI cards and echo cancellation
8:50PM 2 ATA hangs up at 30 seconds
8:11PM 3 WaitForSilence Problems
6:32PM 3 Polycom 501 transfer feature
5:48PM 2 Reverse Scenario
5:34PM 2 OpenH323 and ptlib version for asterisk 1.4.21.1
4:46PM 2 Passing Account Balance to SIP Phone?
4:32PM 0 show channels concise parsing script?
12:43PM 14 Magnetic door locks
10:08AM 4 SIP Testing-Tool
9:00AM 0 Asterisk System Architect requirement.
8:54AM 0 Click try the to call phone tomorrow on VUC
8:45AM 0 Friday at 12 Noon: Asterisk and VoIP User Groups
6:15AM 8 1.4.21.1 SIP failing, requiring reboot
12:48AM 0 TeleVantage Call Monitor & Asterisk
12:33AM 14 Experience with Vicidial
 
Wednesday July 16 2008
TimeRepliesSubject
8:46PM 4 Zap Channel Oddity
6:16PM 1 D channel signalling, while B channels busy?
6:05PM 1 how to stop web Click to Call fraud, robots, etc
4:51PM 1 Disconnect on PRI ignored?
4:16PM 7 Specifying a different codec for meetme
1:42PM 2 Asterisk Recording Interface
12:01PM 8 Digium PRI and Echo cancellation
10:58AM 0 ISDN Call Droping only for outgoing
10:33AM 0 (no subject)
4:56AM 9 how to incorporate open hours
3:22AM 5 Two way bandwidth test
2:52AM 1 Asterisk CAS connection to VConsole ISDN simulator
2:08AM 6 asterisk + web services
 
Tuesday July 15 2008
TimeRepliesSubject
11:47PM 6 Beginner Issues
10:27PM 3 gui issue in asterisk aa50
9:19PM 0 Adtran IP712
9:13PM 1 (no subject)
8:22PM 12 How to monitor Asterisk logs ?
6:32PM 6 Meetme replacement with native 729 support
6:22PM 4 Toll Free International Number
6:02PM 6 distintive ring
4:05PM 2 sip prune realtime per issue
3:15PM 1 Music on hold
3:00PM 2 Reinvites and SIP/RTP
12:41PM 3 Interfacing pri card to legacy pbx
1:03AM 0 h extension priority
12:24AM 4 Incoming calls on zaptel not answered.
 
Monday July 14 2008
TimeRepliesSubject
9:34PM 4 Agent channel...
9:20PM 0 Your comments: Astricon 2008 Balloon Trip?
6:56PM 9 Zaptel problem with pots lines
4:45PM 1 fring (softphone on mobile) and open vpn
4:42PM 2 Asterisk behind NAT, Polycom behind NAT (SIP), how to work?
1:53PM 3 XORCOM BRI interfaces
1:47PM 9 Asterisk unable to register to tnet.it
11:00AM 4 How to integerate 2 TDM cards on same machine.
 
Sunday July 13 2008
TimeRepliesSubject
10:04PM 2 language problem
8:59PM 0 asterisk 1.4 zap instance
4:45PM 10 Poor audio quality with TDM400 card
2:22PM 8 zap not getting callerid any more
10:06AM 0 Unrecognized prilocaldialplan TON modifier: 5
9:17AM 1 can not receive calls through pri
7:30AM 2 Zaptel 1.2.26 problems
 
Saturday July 12 2008
TimeRepliesSubject
11:03PM 3 Incoming call does not reach asterisk.
9:52PM 0 Bridging two Redirected Channels?
8:30PM 0 Wanted Polycom 601 + expansion sidecar
2:21PM 1 AsteriskNow SIP config
3:46AM 1 IMAP Storage Problem
 
Friday July 11 2008
TimeRepliesSubject
11:12PM 3 Recharge Dial Limit....?
7:28PM 3 ASTERISK/ENSWITCH ON EC2
7:13PM 1 No service on phones...
6:50PM 5 Asterisk PBX How-to Guide for Amazon EC2
6:39PM 2 Odd text in sip debug
6:30PM 0 libpri version 1.4.5 Released
4:04PM 0 Outgoing calls but no incoming calls with X100P
3:54PM 10 Sipura 3000 replacement ---> SPA3102 how reliable is it?
3:07PM 0 Asterisk Fails to convert INFO to Inband
2:57PM 0 SIP timing out over satellite connection on 1.4.21 (works with 1.4.18.1)
2:17PM 6 Incoming
11:13AM 0 C450 broken rtp handling
9:16AM 1 Microsoft CRM 4.0 integration with asterisk
7:50AM 0 Analog lines dtmf problem
2:54AM 7 Asterisk cant play sounds from AGI
 
Thursday July 10 2008
TimeRepliesSubject
11:36PM 1 Should I remove the blank options?
9:19PM 8 Diagnosing dropped calls...
5:25PM 30 Asterisk as an IVR solution
4:49PM 10 Tracking Call Time While in Dial()
4:44PM 0 Festival issues
2:44PM 0 Asterisk hangup not working on inbound calls
1:36PM 0 Why it keeps display the G729 codec during the call running on the consol
1:17PM 1 Friday June 11th: SIP love/hate
12:46PM 1 RTP packets dropped
12:16PM 4 Asterisk conference call with a HuntGroup
11:47AM 0 callerid_get_dtmf: Couldn't detect start-character. CID parsing might be unreliable
11:13AM 0 Simple Call Screener
11:11AM 1 res_odbc.conf and odbc show
11:00AM 0 Why is the h extension being called ?
 
Wednesday July 9 2008
TimeRepliesSubject
11:08PM 6 changing inbuilt sound messages
8:50PM 0 ** app_swift v1.2.2 released for Asterisk 1.2.x code-base **
7:39PM 2 asterisk 1.2.21.1 seg fault
7:28PM 4 Zap Bridged Channels
7:15PM 0 e911/CAMA/MF
5:49PM 0 ** app_swift v1.6.2 released for Asterisk 1.6.x code-base **
3:28PM 0 question about fxo cards
3:07PM 0 disable DTMF on a particular channel
2:50PM 6 OT: DNS security
2:08PM 4 READ application
1:24PM 5 Zap Bridged Calls do not continue dialplan
12:14PM 6 Proper Hangup message
11:55AM 3 asterisk sip problem
11:33AM 0 cron jopb
10:47AM 0 H.323 <-dtmf->
10:36AM 1 change E1 link from ISDN to Q.SIG
9:25AM 0 Default table layout for cdr logging with Mysql
8:30AM 0 Simple call accept test
8:03AM 2 transfers only work when voicemail enabled
7:01AM 6 Asterisk dimensioning
6:35AM 0 Problem Asterisk
4:55AM 6 cell phone hangup not getting recognised by system
12:04AM 3 Distinctive Ring for SIP?
 
Tuesday July 8 2008
TimeRepliesSubject
9:49PM 0 Trouble with faxing using iaxmodem / hylafax
9:32PM 1 modules/cdr_odbc.so
6:59PM 0 ** app_swift v1.4.2 released for Asterisk 1.4.x code-base **
4:53PM 3 astrundir not used
2:55PM 3 CONSOLE logging
12:56PM 14 (announce) asterisk T.38 gateway
12:47PM 0 Asterisk STILL loosing IAX user's registration
10:04AM 0 CallerID in The Netherlands with TDM11B
9:48AM 0 has anyone worked with nxtvox fxo cards
9:23AM 1 asterisk and polycom provisioning
9:15AM 0 Asterisk 1.4 restarts after parking using AGI
8:35AM 2 realtime outgoing
1:58AM 5 rxfax not receiving faxes
1:15AM 4 Sharing unused minutes between Asterisk users
12:37AM 0 AsteriskWatch FaceBook application
 
Monday July 7 2008
TimeRepliesSubject
11:33PM 0 Audio data from ast_speech_write
10:50PM 5 Help with sip configuration
10:22PM 2 First-time queue app: verifying human member?
10:03PM 8 QueueMemberStatus
9:31PM 1 SIP or SCCP for cisco
8:39PM 60 US T1 Hangup Detection
7:10PM 0 chan_alsa resource temporarily unavailable
5:21PM 2 Building an IVR
5:17PM 2 Cisco 7940 not getting PoE from Linksys SLM224P
5:05PM 4 cdr_addon_mysql - additional fields
5:03PM 0 Return VXML vars to Dial Plan
4:59PM 0 SIP MWI Problem in 1.4 and 1.6
2:46PM 6 sippyskype
2:23PM 1 Click-to-talk (Java application)
12:48PM 3 DTMF on iax channel is not interpreted by asterisk
10:58AM 13 Meetme
10:16AM 2 queue member state
10:09AM 2 ATA gateway
9:26AM 2 Codec negotiation for Thomson ST2030 and g729
6:08AM 0 trixbox + GXE5024 peer
12:32AM 2 dial plan help.
 
Sunday July 6 2008
TimeRepliesSubject
10:48PM 2 delay when rinigng asterisk
7:12PM 2 Sipura SPA-3102 and Asterisk
4:23PM 0 Zaptel and Solaris X86
10:28AM 0 Documentation for realtime text support in Asterisk
7:11AM 2 Eeepc + Asterisk + Video conferencing
 
Saturday July 5 2008
TimeRepliesSubject
11:33PM 5 HR 5889.
8:39PM 5 New Polycom SpectraLink 8002 Wifi SIP Handset
5:50PM 4 Cell phone to PSTN adapter or IAX
4:32PM 0 Return Vars to Dial Plan from VXML()
3:48PM 0 Read & Background
3:13PM 1 Asterisk loosing IAX users's registration??
12:00PM 3 Require Billing solution for Calling Cards retail...
12:59AM 22 [asterisk-dev] Locking, coding guidelines addition
 
Friday July 4 2008
TimeRepliesSubject
3:27PM 0 background noise
2:21PM 0 Dear asterisk-users@lists.digium.com SALE 85% 0FF on Pfizer
1:30PM 1 Call Forwarding Lopp Prevention
11:57AM 2 Removing voicemail messages
11:53AM 0 Bug tracker having issues
5:25AM 9 DIDs required of Paris and Gottenburg Sweden
 
Thursday July 3 2008
TimeRepliesSubject
9:02PM 2 Asterisk VXML... Help.
7:26PM 9 Spoofing CID
2:50PM 0 how to setup one stage dialing plan, instead of two! help!!!
2:33PM 0 D-Link DVG-3104MS
1:45PM 2 wait & pickup
12:51PM 2 problem in making call pc to phone & vice versa
12:50PM 9 (no subject)
10:52AM 0 OLPC Sound Samples
10:29AM 2 (no subject)
9:17AM 0 asterisk queues and database backend (clustered realtime)
8:40AM 5 Dial function exit, go to line n+1
2:54AM 5 2 AVM ISDN Fritzcards
12:48AM 0 agi never leg1 disconnect
 
Wednesday July 2 2008
TimeRepliesSubject
10:44PM 1 ooh323 doesn't know what to do when bridging calls?
9:29PM 1 Tone Differentiation
8:51PM 0 Dial duration
7:59PM 7 How to change http port on appliance?
5:16PM 6 Does an IAXy require registration?
4:34PM 1 Asterisk Taking CPU resources
3:57PM 0 Can you verify this bug?
12:05PM 19 new install of asterisk appliance.
8:59AM 0 asterisk-users Digest, Vol 48, Issue 4
12:51AM 0 Config help with ISDN Fritzcard
 
Tuesday July 1 2008
TimeRepliesSubject
10:57PM 3 Best Practices: Empirical measure of call latency
10:25PM 0 Cannot dial on E1 cards
7:30PM 4 Asterisk 1.4.21.1: Bugs in IAX
7:27PM 10 Waiting time to send the call
7:20PM 1 queue show name - callerID
5:50PM 1 Broadvoice and Asterisk 1.6.0-beta9
3:29PM 20 The S word: Asterisk security
2:20PM 3 Asterisk 1.4.21 and CUT function
1:35PM 6 Fax Between IAX Trunks
1:33PM 6 music on hold realtime
12:38PM 25 Call quality
12:34PM 0 Panama SIP ITSP?
12:25PM 2 Click to Dial Service Providers in Australia
11:09AM 3 User unable to use DTMFs?
10:35AM 0 line goes silent for a few seconds at the start of outgoing calls
8:09AM 0 Manager proxy
7:15AM 13 Choppy audio
2:39AM 0 Queue recording file name
12:44AM 3 Disto choice for Asterisk with AVM Fritz!PCI cards