asterisk users - Sep 2010

Thursday September 30 2010
4:31PM 0 Friday 12 Noon EDT: VoIP Abuse Project
4:08PM 0 Same extension on multiple servers confusion
2:51PM 2 Unable to load fax modules
2:03PM 2 Intercom with Dial() works, but not with Page()
1:52PM 0 Unscheduled service outage for various Asterisk community services
12:57PM 1 channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI
12:00PM 1 Asterisk Audio Prompts Stopping
10:16AM 2 Asterisk Internal timing
10:09AM 3 Kernel Panic When restarting the server
7:36AM 0 SIP Registrations
7:30AM 3 Go from *100* to just 100
Wednesday September 29 2010
8:57PM 1 can't get libpri/PRI to work, missing PRI commands
6:57PM 0 Successive Dial apps give hang up within 30s!!
3:52PM 0 asterisk > cisco gateway > westell > isdx
2:13PM 2 Alert-Info advice
12:50PM 0 (no subject)
12:49PM 0 Jin.
12:39PM 1 Weird Behavior with DAHDI
9:56AM 4 DAHDI FXO port only recognizes the "S" extension‏
2:06AM 4 Use modprobe to find E1/T1 jumper setting on PRI card
Tuesday September 28 2010
1:22PM 2 E1 check with nagios, how to?
12:05PM 2 SIP X.25
11:51AM 1 What's the meaning of this?
11:15AM 1 Inbound calls from TRUNK
10:52AM 3 ISDN - Busy signal on 3rd call
8:02AM 1 1.6 and 1.8 version & A2Billing
5:05AM 0 AstLinux 0.7.3 released
1:28AM 2 NAT issue (i think?)
Monday September 27 2010
6:02PM 1 How to pick a codec on the fly
6:00PM 2 SCCP (skinny) phone behind NAT: RTP dest addr wrong
4:09PM 8 Problems compiling Asterisk on Debian
4:02PM 1 propagate sip reinvites with directrtpsetup=yes
3:26PM 0 groupcount - show usage
1:57PM 0 Asterisk and dahdi on Arch linux
10:37AM 1 RFC3329 support in Asterisk
8:08AM 0 PSTN to SMS and SMS to PSTN
3:04AM 1 misc newbie VoIP questions
Sunday September 26 2010
6:08PM 2 Asterisk ODBC Insert issue
5:48PM 5 Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...
5:22PM 1 Downloading the Asterisk as tar.gz file
5:00PM 2 1.4, 1.6, 1.8 versions and the H323 gatekeeper functionality
3:04PM 3 Asterisk Redundancy
Saturday September 25 2010
10:22AM 0 Asterisk Cluster Scenario
9:53AM 0 September 67% OFF
2:09AM 0 can call internal branch , but can not call external numbers with avaya , always get return message : Q931IncompatibleDestination
Friday September 24 2010
7:40PM 5 Asterisk 1.62.13 - CPU spikes every 10 minutes
4:52PM 2 best format for playback/generation
2:55PM 3 should trixbox system hang when ISP drops connection?
2:47PM 2 Debug compile fails
1:52PM 0 Fwd: Can't cross compile asterisk on arm using ltib
1:13PM 4 differential billing
9:28AM 1 RDNIS not passed from one box to another with BRI access
9:11AM 1 tcpdump auto stats script
9:05AM 5 How to test BIG traffic through DAHDI/WANPIPE interfaces
Thursday September 23 2010
11:01PM 1 Asterisk - have asterisk reply from same IP address
9:40PM 2 rtp problem with 1.8.0-rdc1
8:14PM 2 Asterisk 1.8.0 Release Candidate 2 Now Available
7:39PM 0 Unable to make outgoing call on E1
5:56PM 1 OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? (bruce bruce)
4:25PM 0 Asterisk Transfer/call patching support
4:09PM 1 Forking a call
2:59PM 0 Sip from ip address
2:06PM 1 Can't turn debug on in a 1.2 box
2:03PM 1 Net2Phone SIP trunk problem
11:48AM 4 Asterisk and Digium TC400B
11:15AM 3 realm: security issue
10:10AM 2 CDR display in minute
6:21AM 8 Record() Cmd and My SQL
5:19AM 0 Calls stuck in the queue even when ext's are available
4:25AM 0 Installing Asterisk + FreePBX from Repsitory spits out some warnings and errors for ever
Wednesday September 22 2010
9:27PM 2 Recording maximum time and stop on silence
8:59PM 1 Asterisk- speech to text(Voicemail to text message)
6:05PM 0 Sangoma A500 NT BRI PTMP without woomera on asterisk 1.6
4:49PM 4 Asterisk as a distributed paging system
3:45PM 5 naming schema
2:26PM 0 TLS re-negotiation attack on SIP/TLS of Asterisk?
2:05PM 1 Costa Rica Hangup Detection
2:00PM 2 Asterisk T38
1:21PM 2 Can't cross compile asterisk on arm using ltib
11:58AM 1 T38 and codecs negotiation
10:32AM 2 Unable to open vm-INBOXs
9:42AM 1 Cross compile Asterisk for mipsel-linux
5:27AM 5 OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Tuesday September 21 2010
9:56PM 0 Mixing ISDN and R2 in the same card...
9:17PM 1 digits in chan_dahdi
3:48PM 1 random hangups on RBS T1
1:42PM 3 Unexplained message in 1.6.2
9:33AM 0 Dialplan extension pattern matching for '/' character
9:25AM 6 func SHARED, how to use?
1:56AM 0 Asterisk News Accepting Submissions
Monday September 20 2010
5:02PM 0 Commands needed via AMI to find callerid of inbound call to extension
4:15PM 1 Authentication best practice
4:08PM 0 Asterisk stops processing SIP UDP messages
4:01PM 1 Setting 'fname_base' variable doesn't affect 'automon' result file.
12:51PM 2 Playing Audio To One Channel
11:31AM 3 Extension continues ringing after caller hanged up
8:15AM 1 Confused about notifyringing in sip.conf
Saturday September 18 2010
10:46AM 2 Audiocode Median 2000 Gateway with Asterisk ?
4:12AM 1 Asterisk sip attack
Friday September 17 2010
11:54PM 1 externip/localnet
6:43PM 0 quick 1.8 question on console/dsp
6:00PM 0 5-7 second delay in connecting outgoing FXO calls
5:51PM 2 3rd party app store
5:29PM 1 Rotary phone on Asterisk
5:10PM 3 do carriers detect unusual / unauthorized VoIP calling patterns?
5:08PM 0 CallerId: behavior changed between and 1.4.36 with .call files
4:59PM 0 Asterisk 1.8 and CEL logging
3:37PM 0 need help with IVR dialplan
3:25PM 0 ISDN BRI call disconnection issue
3:00PM 5 Initial Audio Cut off
1:24PM 4 Not able to join conference
11:13AM 2 Call restriction for particular extension
11:02AM 3 Sangoma A108 PCIe V2.0
11:00AM 0 Sangoma A108 PCIe 2.0
9:50AM 1 Issue with transfer (sip)
9:31AM 1 Attended Transfer does not release channels
8:57AM 0 Deadlock rendering sip useless
Thursday September 16 2010
6:44PM 5 AGI Delimiter in 1.6
5:58PM 4 one way audio for xlite clients behind NAT
5:03PM 1 Help!! Call waiting issue
3:42PM 4 [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?
2:26PM 0 Indications and tonelist on a SIP channel..
1:33PM 0 DTMF tones too long, for once
11:59AM 1 How to Understand a pri intense debug span X
11:56AM 2 Realtime semi-colon
10:15AM 3 Configure Asterisk with openssl
7:09AM 0 asterisk 1.6 and BLF
12:24AM 5 a2billing
12:20AM 3 Purpose of qualify=yes
Wednesday September 15 2010
10:10PM 3 Echo on Sangoma A400 and background noise
7:41PM 6 Bug with Realtime?
7:31PM 1 Queue member status not changing
6:02PM 0 Asterisk Now Available (Re-Releast of
3:54PM 1 Asterisk Download
3:24PM 0 Asterisk Now Available
3:23PM 0 Asterisk 1.4.36 Now Available
3:09PM 1 Error loading skype_for_asterisk
3:06PM 3 changing from zap to DAHDI
3:04PM 2 Problems with audio
2:47PM 2 Dual WAN with load balancing
12:20PM 2 incoming call FXO
11:14AM 2 Help me Out!!!!
10:12AM 1 One way audio when overlapdial is set to yes
10:05AM 3 Skip Busy Agents/Channels from Queue
3:19AM 2 Digest Username/auth name mismatch‏
2:22AM 0 about yahoo messager work with asterisk
Tuesday September 14 2010
7:56PM 6 How different is implementing Cisco based system than Asterisk based system?
7:28PM 1 conf checkout
5:33PM 2 DTMF
4:32PM 5 sip show channels
3:11PM 0 question on asterisk 1.8 meetme
2:19PM 2 SIP 800 Origination/Termination - International
1:44PM 9 Random File Name
12:30PM 6 Spontaneous reboots on asterisk
9:59AM 1 can asterisk accept "anonymous" register ?
7:06AM 0 3xx redirect response list Noop and capture
6:27AM 2 OT - Gigaset C470IP - How to access SMS settings
5:01AM 9 Speech To Text on linux with asterisk
3:29AM 1 Which 1.6 subversion is Stable one?
Monday September 13 2010
3:47PM 2 PostgreSQL is asterisk friendly with it?
3:22PM 5 Force ip disconnect after register?
10:34AM 1 Changing voicemail.conf file format list
10:29AM 0 Upgrade from 1.4 to 1.6 : problems with realtimemysql
9:57AM 1 Which voicemail file format is the most widely understood ?
9:56AM 7 High volume BLF - Suggestions?
9:04AM 2 How to send SMS to Gigaset phones ?
8:26AM 2 Correct queue agi syntax in
6:38AM 3 doing dnsmgr_lookup
Sunday September 12 2010
7:47PM 1 Synway cards
7:41PM 1 First boot asterisk -vvvvvgcn segfaults
5:43PM 3 Moving from DSL to T1
5:05PM 1 username mismatch with
5:20AM 0 asterisk as POC(push to talk) server?
4:41AM 0 voicemail not working for all extensions in same way
Saturday September 11 2010
3:07AM 8 No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
12:44AM 5 SIP softphones answer but do not connect...
Friday September 10 2010
11:35PM 5 Polycom dhcp boot
8:51PM 7 A way to check against a list of numbers?
7:33PM 0 Cisco or Linksys WRP400 reliability?
7:01PM 0 Anyone can share their experience about Thomson TG784 wireless router/ATA?
2:07PM 1 problem with iax call (chan unavailable)
10:24AM 0 realtime sip registrations disappear from DB
6:17AM 0 Asterisk SIP woes
Thursday September 9 2010
8:40PM 2 DAHDI fxstest?
8:38PM 1 Curious what 'early media' is in terms of Answer()
7:43PM 0 Cisco 7975g running 8.3.4
5:25PM 3 Archive of security advisories?
4:51PM 1 VoIP friendly Internet providers in Dallas and Philadelphia
3:19PM 0 Issues with in-call DTMF using Broadvox and Level 3
3:12PM 0 vegastream 50 BRI-s latest firmware ?
2:04PM 5 info about application not available asterisk
1:55PM 1 Set channel variable from within other channel
1:29PM 2 Mirroring or other arangement to secure *
11:39AM 2 How to avoid interruptions with DIGIUM
10:31AM 1 syntax error, unexpected '<token>'
4:14AM 1 getting error chan_sip.c: Failed to grab lock, trying again..
Wednesday September 8 2010
9:57PM 2 Max TDM calls per asterisk box
8:18PM 1 Asterisk 1.6 and fax
7:43PM 1 Thailand DID
4:52PM 1 Sangnoma + Digium Bridging
4:51PM 0 Asterisk 1.8.0-beta5 Now Available
4:48PM 0 Sip real problem
4:10PM 3 IPSec on asterisk
3:38PM 2 Problem with new AEX800 card dying because of interrupt problems
3:16PM 2 Queue/Dial Recording - Capture answering channel name.
2:48PM 1 Upgrade from 1.4 to 1.6 : problems with realtime mysql
1:42PM 1 asterisk 1.8 Calendar
1:38PM 0 Requirement or just Best Practice
6:23AM 0 How to Set Callerid Of Originate a call?
4:47AM 0 rtcp to cdr for calls from dahdi to sip
Tuesday September 7 2010
7:49PM 1 Solving the CDR mess of attended transfers
7:08PM 3 Losing first DTMF digit (with ASR)
6:56PM 3 Call Center: scripting for call routing, reporting, login and logout, CTI
6:47PM 2 5-7 second connection delay in outgoing FXO calls
5:00PM 4 SPA3102 FAX not working
1:16PM 3 voice mail system
Monday September 6 2010
9:08PM 1 MeetMe errorhandling
7:32PM 3 What can make G.729a codec hostid change?
5:20PM 0 How are shared variables destroyed ?
4:20PM 0 Asterisk stops processing calls...
3:35PM 0 Going to go out on a limb here - regarding Vonage
3:28PM 1 Dial timeout and SIP 302 Moved Temporarily
2:23PM 2 Macro when calling cellphone (GSM) + silence when connecting
2:02PM 2 Is it possible to keep both call legs live after Dial()
1:10PM 4 SMS and fixed land lines
12:45PM 1 Asterisk Fax
Sunday September 5 2010
10:20PM 3 Registering and initiating a SIP call without a SIP client
Saturday September 4 2010
8:08PM 1 Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ?
3:52PM 3 Vitelity offline?
2:30PM 0 Global Outage?
1:40PM 4 fast busy out?
12:42PM 1 Manuplating Queue
9:21AM 1 Snom phones recommended firmware
8:34AM 0 Wanted: UK-specific hardware recommendations (FXOand FXS)
Friday September 3 2010
9:06PM 3 How to tell if there is a transfer from CDR?
2:49PM 1 Faxes
1:45PM 0 [draft] DAHDI-linux & DAHDI-tools 2.4.0 Release Announcement
1:23PM 0 How to use MYSQL(Set timeout x)
1:11PM 4 openvz
9:07AM 2 Wanted: UK-specific hardware recommendations (FXO and FXS)
8:14AM 1 not succeeding to hide callerid with outbound calls
4:34AM 0 Polycom 670 with Extension Module | Busy Lamp Field | Directed Pickup | Speed Dial | etc
3:40AM 0 Asterisk failing when recording calls
2:03AM 0 Asterisk processing URI's
Thursday September 2 2010
8:32PM 1 Channel Signalling
6:53PM 0 asterisk freezes the server
6:26PM 5 How to create a coredump for Asterisk
5:52PM 1 How to finish an AGI
4:32PM 2 Fw: [asterisk-biz] To compete with Avaya - What are their current cost?
2:57PM 5 Google Voice-like feature.
12:31PM 2 Voicemail - disable * 0 and #
10:56AM 2 Call Recording Questions
9:37AM 4 agi playback to execute say.conf settings
6:21AM 1 IAX2 calls getting rejected without a CAUSE CODE. How to debug this?
3:28AM 0 NCS - Cablemodem
Wednesday September 1 2010
7:35PM 0 ITSP with DDIs (or DIDs) from India
7:28PM 0 libpri Now Available
6:28PM 3 MOH in the middle of the call
4:15PM 2 * and mj
1:24PM 1 ChanSpy getting piled up
1:09PM 3 3Com 3102 Phones
11:58AM 2 Freepbx + Asterisk problem - NEED HELP