asterisk users - Feb 2010

Sunday February 28 2010
TimeRepliesSubject
6:29PM 0 Compatible IP Phones for Asterisk
4:58PM 6 AUTHENTICATE Command customized prompts - Work around
4:56PM 0 ISDN Options
1:17PM 0 Lower kernel version for mISDN
10:48AM 6 Premicell solutions?
5:43AM 8 Server response time
 
Saturday February 27 2010
TimeRepliesSubject
8:42PM 4 No RTP from asterisk?
8:08PM 2 Conference Calling
7:51PM 1 Asterisk AUTHENTICATE Command
12:58PM 0 Increasing the dahdi chunk size with Sangoma cards
12:43PM 0 New patch for app_queue to show all call attempts, even failing ones
1:29AM 0 "Unexpected message received" when receiving Fax
 
Friday February 26 2010
TimeRepliesSubject
11:29PM 3 Fun with virtual asterisks ...
8:48PM 0 Qeuee/Agent Question
6:11PM 6 hi
4:30PM 0 qsigchannelmapping parameter
4:24PM 8 Asterisk RPM's
4:01PM 1 SPA941 WMI not lighting up when natted
1:35PM 10 : PSTN calls
1:34PM 2 Web operator/softphone with integration features
10:21AM 0 Problem with BLF's
10:06AM 0 record a user call while playing a background music
8:17AM 0 realtime modules not load ?
5:26AM 5 Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
1:27AM 2 How to tell if asterisk is handling media or not?
12:41AM 0 How can we pickup a call that is not going to a real extension?
 
Thursday February 25 2010
TimeRepliesSubject
10:40PM 1 Asterisk 1.6.0.17 PBX with two interfaces does not routes RTP packets - SIP Conf Problem likely
10:39PM 0 Asterisk 1.6.0.25, 1.6.1.17, and 1.6.2.5 Now Available
10:38PM 0 DTMF timing - first # keypress not registering
10:28PM 0 AST-2010-003: Invalid parsing of ACL rules can compromise security
8:51PM 2 Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid
7:54PM 0 Asterisk Crashs due to some Sip messages
6:45PM 4 Morse Code
6:08PM 5 Followme broken
5:19PM 7 MeetMe() and dahdi_dummy on an embedded system
4:50PM 0 Intermittent DAHDI issue with a PRI line causing asterisk to crash!
4:19PM 3 Deadlock while using MGCP on Asterisk
3:59PM 1 AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences?
3:30PM 2 Problems installing dahdi : kernel sources
3:21PM 0 IAX peers one way voice
2:35PM 3 X-Lite won't register
10:17AM 7 Redirect call based on CLI???
9:26AM 2 Getting: Can't fix up channel from 5 to 7 because 7 is already in use, and pri_dchannel: Answer requested on channel 0/7 not in use on span 1
8:45AM 0 CDR duration/billsec
7:29AM 1 curl and ssl certificate
6:38AM 1 Asterisk n-way DTMF detection
2:16AM 3 Do i need install Dahdi or libpri ?
 
Wednesday February 24 2010
TimeRepliesSubject
11:51PM 4 Problems with Linksys IP Phone SPA 942
11:30PM 9 audio glitches in conference
9:49PM 0 Question
9:05PM 1 Encrypted calls between mobile gsm and isdn (asterisk)
6:23PM 2 AMD: HANGUP
4:35PM 4 identify the costumer
3:44PM 3 Re-INVITE on BYE
3:42PM 2 Problems in Asterisk Real Time (Urgent help )
2:20PM 6 subject: 1.4 vs 1.6
1:55PM 0 Manager Logged off
11:41AM 0 Wrong MOH
11:37AM 0 Looping over AstDB
6:04AM 0 IAX devices not registering after upgrade to asterisk
3:35AM 1 Macros, GoSub & StackPop
 
Tuesday February 23 2010
TimeRepliesSubject
7:28PM 2 IAX devices not registering after upgrade to
7:23PM 4 Which H.323 to use in Ast 1.6
5:41PM 4 SIP provider registration attempts
1:22PM 5 directrtp with SIP + H.323
11:59AM 11 Calls per second limit in manager
11:30AM 1 Codec translation in Asterisk
10:39AM 1 IAX devices not registering after upgrade to asterisk 1.4.29
8:18AM 5 Running safe_asterisk
 
Monday February 22 2010
TimeRepliesSubject
10:59PM 4 Load balance outgoing calls
9:13PM 13 SIP Disconnects from Network - Asterisk Does not hangup
8:31PM 0 Avaya with Asterisk
7:20PM 2 Problem w/ MoH
7:13PM 5 Problems with SIP realtime
7:02PM 2 Open source or low-budget recommendation for call-center software
6:57PM 0 init.d error when installing trunk
6:02PM 7 Multiple instances of Asterisk on the same host...
4:18PM 14 [OT] Asterisk 1.6 and DECT Phones
3:31PM 4 Caller ID question
1:20PM 4 4 PCIe cards in one asterisk server
11:52AM 5 Denying call transfer to certain extensions
11:50AM 2 AMI Originate differences between 1.4 and 1.6.1
11:16AM 1 Sending back the BYE code gotten on second leg
8:28AM 2 TE410P Spans offline/red after power down/restart
7:36AM 1 Does Playback will answer the call?
6:23AM 2 Audio to remote AGI server
2:40AM 3 Free iPhone Asterisk Function and Application Reference
 
Sunday February 21 2010
TimeRepliesSubject
7:55PM 13 HFC-S card
6:54PM 0 Trouble with externalIVR socket connection
5:22PM 1 Dahdi & Congestion status
3:14PM 2 add Reason header on hangup
 
Saturday February 20 2010
TimeRepliesSubject
10:22PM 0 IPKall NOT coming on Asterisk
9:28PM 0 Moh help needed
9:16PM 2 Sending a hook flash to a DAHDI channel
6:02PM 2 Slightly OT: Has SILK codec gotten anywhere?
4:49PM 0 outgoing callerid problem
1:14PM 3 Fax, T38 and NAT
2:43AM 0 Hung channel problem with 1.4.26.2
12:35AM 1 Error redirecting an incoming call of a SIP provider to a local extension
 
Friday February 19 2010
TimeRepliesSubject
9:54PM 0 asterisk sudden restart - 1.4.18.1 - Resolved after upgrade to 1.4.22.1
8:05PM 4 string length in dialplan
4:42PM 15 Virtual machine timing (KVM)
4:21PM 5 splitting sip.conf to two files
4:01PM 1 transcoding with TC400P
3:16PM 1 mISDN (HFC-S) and TDM400P - isac xdu no tx_busy
2:21PM 0 AMI + device status (patch 0016732) + remote control
1:57PM 1 Volume of Playback() application
6:51AM 4 Dial Plan configuration in asterisk
1:19AM 1 directmedia/canreinvite/native bridging question
 
Thursday February 18 2010
TimeRepliesSubject
11:51PM 0 Asterisk 1.2.40, 1.4.29.1, 1.6.0.24, 1.6.1.16, and 1.6.2.4 Now Available
11:46PM 0 AST-2010-002: Dialplan injection vulnerability
8:57PM 0 How to transfer call using function T
8:47PM 2 BRI vs. PRI?
8:05PM 8 OpenVPN/SNOM 820: a review.
6:46PM 7 Realtime extensions
5:52PM 2 Product offerings from DIDforSale
1:51PM 0 ISDN phone not ringing. ISDN PBX not answering?!
7:45AM 0 Feb 19th @12 noon EST: Voxeo's Tropo
7:40AM 3 Asterisk t38modem Fax gateway evaluation
1:49AM 0 Asterisk cluster in Active/Active mode
1:40AM 8 how asterisk knows which context forward the call to?
12:53AM 6 Registering of Asterisk against a SIP provider
 
Wednesday February 17 2010
TimeRepliesSubject
9:23PM 0 Asterisk answers inbound call during ringing
9:15PM 11 Static IP
9:06PM 3 Setting up only one caller at a time
9:03PM 6 Access to header field: event
8:47PM 2 queue.conf - Set(MONITOR_FILENAME=${})
8:21PM 1 One-Way Audio after Hold
7:50PM 1 some newbie questions about gcc
7:17PM 0 sending call to correct context
6:39PM 0 Asterisk in Active/Active mode
6:12PM 6 sip.conf - sort order, does it matter
4:50PM 1 1.6.1 Voicemail users.conf
4:11PM 8 asterisk dahdi fax problem
3:00PM 6 chan_local and Originate
2:57PM 5 Help with Dictate app
11:37AM 10 Unrecognized prilocaldialplan NPI modifier
10:13AM 2 how to remove asterisk from this string X-Asterisk-HangupCauseCode
8:13AM 1 call parking
3:39AM 1 Ideasip
 
Tuesday February 16 2010
TimeRepliesSubject
10:50PM 6 Asterisk listens on all NICs
7:51PM 1 chan_sip.c: Disconnecting call 'SIP/302-b720dd78' for lack of RTP activity in 301 seconds
6:47PM 0 Voicemail IMAP storage enhancement
6:05PM 2 Cannot built kmod-dahdi-linux for PAE kvariant from SRPM
5:48PM 1 call is not going to wrong "context"
5:36PM 2 rawplayer in asterisk 1.0.0
5:29PM 1 CODECS: Best practice question: Avoid transcode when calling out?
4:39PM 1 How does holdtime get calculated for queues
4:17PM 0 Users of the SMS application?
3:50PM 3 Stupid question: Why Cmd Dial and Queue haven't same options?
3:04PM 2 Issue with trying to dial two different servers at the same time.
11:55AM 2 call transfer
9:40AM 1 Empty SIP Packet
7:54AM 2 OT- Using TR-069
 
Monday February 15 2010
TimeRepliesSubject
7:31PM 4 video voicemail
4:05PM 4 Maximum call handling capacity on single server
10:55AM 1 strange asterisk behaviour on XEN
8:26AM 0 Zaptel/DAHDI error's on PRI
6:54AM 1 signal problem
1:25AM 2 Capture
12:35AM 4 insecure=invite - not working for different dial plan
 
Sunday February 14 2010
TimeRepliesSubject
9:59PM 2 how to have disconnect signals enabled in line
4:42PM 7 Asterisk Redundancy
4:34PM 0 transmit_silence_during_record
1:37PM 0 voicemail problem
12:43PM 1 issues.asterisk.org
12:24PM 5 Cisco 7940: showing FWD in display.
9:33AM 3 Line DC
6:56AM 0 Domain Authentication - Caller ID Failed to authenticate
3:50AM 3 agi debug in Asterisk 1.6?
 
Saturday February 13 2010
TimeRepliesSubject
11:04PM 44 Important security alert: update your dialplans now!
8:59PM 1 how to create voicemail
3:57PM 2 1.6.x SIP allow incoming calls based on from ip address?
7:24AM 6 extension not found
3:09AM 2 Call Pickup with 1.6.2.1 and Snom
 
Friday February 12 2010
TimeRepliesSubject
9:29PM 11 Robotic sound sometimes
9:23PM 2 PAP2
8:57PM 2 how to allow some extenstions to call outside and some extensions cant call outside
8:54PM 1 PRI Problems with 1.6.0.10
8:49PM 1 dropping line (s) for 911
8:32PM 4 how to allow some extensions to make call outside and some extensions cant call outside
5:24PM 10 Asterisk Cepstral TTS
3:52PM 2 T.38 with reinvite
2:25PM 3 parked calls
7:55AM 0 [Fwd: SIP tunnel]
5:41AM 0 g722 IP Phone
1:39AM 1 Wierdness in AGI file
 
Thursday February 11 2010
TimeRepliesSubject
9:23PM 0 nortle BCM450 & SIP-Trunking
8:57PM 0 Asterisk 1.2.39 Now Available
5:38PM 0 IP Kall One-Way Audio
2:33PM 0 dnd sorta working
1:37PM 13 SIP tunnel
12:19PM 0 Asterisk ignores BYE messages
10:21AM 13 SIP RTP ports not released when channel is hung up
7:20AM 9 app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
12:36AM 0 Sending "Progress" during dialing
12:16AM 0 wellgate 3804A with frying
 
Wednesday February 10 2010
TimeRepliesSubject
10:37PM 3 How to avoid AGI script is canceled if caller HangUp
7:32PM 2 problems with 1.6
5:24PM 9 1.6.2 : global vars not read/set after #include w/ globals
4:55PM 1 asterisk sudden restart - 1.4.18.1
4:08PM 0 EAGI delay
3:39PM 33 IP Phone recommendation
2:47PM 2 Optimization of call from server 1 to 2 and then back to 1
2:11PM 1 Muted calls occasionally dropping after 30 seconds
1:36PM 2 problems with creating a call
11:57AM 18 Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
11:53AM 1 Nat Issue - is this Draytek || Asterisk?
9:47AM 0 PMS (SDMR, ...) support in Asterisk
8:58AM 2 asterisk and mysql connection
7:21AM 0 VUC Friday Feb 12th: HD Communications Summit
6:52AM 2 forward incomming line to modem
4:02AM 1 billing based on local access number
 
Tuesday February 9 2010
TimeRepliesSubject
11:47PM 0 Callerid problem in 1.6.2.2
9:33PM 6 Security Logging
6:48PM 7 ways of initiating a call
6:19PM 0 ? chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received
5:45PM 1 test
5:32PM 0 ISDN users: 1.6.x users, I need some testing done please, regarding Overlap Receiving
12:43PM 0 asterisk-users Digest, Vol 67, Issue 20 Re: Asterisk going down
12:42PM 2 Not able to receive fax
10:26AM 1 Lua status in asterisk.
10:01AM 3 Get Talk Time
4:42AM 17 VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
12:40AM 2 E71
 
Monday February 8 2010
TimeRepliesSubject
9:23PM 1 Strange Problem
8:45PM 0 moving a bridged call to a conference room
5:04PM 7 IVR Demo / Create file / Move file / Demo all
5:00PM 1 billsec is set to duration if call is not answered
4:09PM 1 Asterisk how install speex support
12:54PM 3 High codec translation times on x64
12:52PM 6 GSM Gateway
11:29AM 10 conferencing without DAHDI
9:27AM 1 queue with strategy=linear
9:24AM 0 Help with iax.conf {tesco|freshtel} 1.6
9:16AM 0 Asterisk going down (Josiah Bryan)
6:19AM 0 Call doesn't disconnect in SIP
5:10AM 5 Not able to compile asterisk, zaptel, libpri in /usr/src
3:14AM 0 Asterisk going dow
3:06AM 0 originate, local channel and failure extension
2:54AM 7 How to run a remote PHP script and still have access to audio stream?
12:18AM 3 syntax
 
Sunday February 7 2010
TimeRepliesSubject
12:13PM 0 Pickup the call ringing at SIP Phone but was transferred from Zap channel
 
Saturday February 6 2010
TimeRepliesSubject
11:19PM 3 A2Billing and other prepaid Billing like ASTCC, who is better?
12:30PM 1 CONNECTEDLINE
11:18AM 14 Asterisk 1.4.26.2 died after 80 days uptime
7:31AM 2 TOS bits, DSCP, Asterisk & Polycom
6:48AM 2 Website Down ?
2:54AM 27 Dial script
1:37AM 0 Recording Calls
 
Friday February 5 2010
TimeRepliesSubject
11:07PM 0 strange issue with iptables + Asterisk
10:59PM 4 Asterisk going down
9:50PM 10 large scale paging
9:23PM 0 Sipgate.co.uk on Asterisk 1.6.2.2
4:55PM 5 2 Asterisk Boxes, Single Voicemail
11:40AM 19 Still on spandsp/app_fax and T.38
8:38AM 2 Ongoing calls interface
4:46AM 0 Do the Linksys Sipura series have a known problem with Asterisk?
2:05AM 41 Losing local SIP phones when internet goes down?
12:39AM 5 Know what would be killer?
 
Thursday February 4 2010
TimeRepliesSubject
7:11PM 0 pickup the call: No target channel found
6:25PM 0 Audiocodes MP-114 MWI Stutter Tone
4:08PM 19 OpenVPN on phones?
3:42PM 4 Gotoif Question
1:48PM 2 SS7 and Asterisk
12:41PM 0 OT: VUC Feb 5th @ 12 Noon Open VPN
11:09AM 0 OT - MWI, Polycom/kirk and Gigaset handsets
10:59AM 0 pickup with gxp2000 does not work..
10:00AM 22 Running a script after Dial() ?
9:55AM 1 Aastra 50-limit blf
12:34AM 4 1.6.2.1: DTMF trouble with PSTN
 
Wednesday February 3 2010
TimeRepliesSubject
10:24PM 1 aastra 9480i dtmf ?
9:00PM 0 Routing inbound call to correct sip trunk
8:43PM 3 500 Internal Server Error on Cisco 7940 after INVITE
7:17PM 3 calling into server with cell and originate a call
3:24PM 1 ast_cdr_setvar: Attempt to set the 'src' read-only variable!
2:52PM 1 Asterisk core sounds in English by June Wallack
10:57AM 0 CDR and Queue Reporting windows application looking for Beta testers!
10:26AM 0 Pri HDLC aborts and choppy audio when dialling into pri, caused by BIOS option ["CPU enhanced halt" c1e]
7:19AM 7 Intel Atom based Asterisk server?
12:43AM 2 CDR / billsec / originate / local chan
12:09AM 0 asterisk video support and IPTV
 
Tuesday February 2 2010
TimeRepliesSubject
11:56PM 0 Queue problem, ringing agents.
10:40PM 0 AST-2010-001: T.38 Remote Crash Vulnerability
10:28PM 0 Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 Released
10:05PM 1 # as dial key - chan_dahdi
8:02PM 5 Semi-Transfer
7:58PM 2 codec conversion
7:20PM 3 sip realtime md5secret
5:25PM 1 Codec coversion
5:10PM 2 Two Extensions showing as Busy
1:53PM 0 Issue when reloading
1:00PM 16 Asterisk 1.6.1.13 and T.38 faxing
12:40PM 3 Asterisk 1.6.2 ?
10:20AM 4 uri tel: instead of sip:accepted ?
7:53AM 0 Realtime queue strategy issue
5:41AM 6 Smallest possible Asterisk VM
1:22AM 4 Use a BLF for monitoring
1:22AM 1 (no subject)
 
Monday February 1 2010
TimeRepliesSubject
8:33PM 0 One way audio with Grandstream HT503
1:55PM 2 Problems with recordings of call using Monitor
12:54PM 5 NVFaxDetect
12:52PM 0 Stuck logger rotation
12:22PM 0 mysterious rippled sound with IAX
9:25AM 0 Asterisk for productive Calling Card System
8:51AM 0 call count per peer
7:38AM 2 connect problem unless when verbose
5:44AM 0 FCT for 3G Video calls
2:09AM 1 Odd error mssage on DAHDI lines
12:12AM 2 app_directory broken in 1.6