asterisk users - Jul 2009

Friday July 31 2009
TimeRepliesSubject
6:48PM 2 Ignoring MOH directory and using default
3:34PM 1 Faxing over Carrier SIP trunk/g711 ?
3:11PM 0 Friday July 31st at 12 Noon EDT: Dave Nelsen, Skype for Asterisk beta opens, Gizmo Voice + Google Voice = free SIP calls
2:44PM 0 Friday July 31 @ 12 Noon EDT: Talkshoe former CEO Dave Nelsen, Skype for Asterisk open beta, Gizmo Voice+Google Voice
12:21PM 2 asterisk 1.6 call forwarding
12:19PM 1 timeout for trunk in failover
11:00AM 3 DAHDI - analogue, not seeing ringing (UK)
10:38AM 10 Voicemail feature: enable or disable the ability to leave a message
7:22AM 6 BT IP Exchange interconnect
 
Thursday July 30 2009
TimeRepliesSubject
10:42PM 0 odd T1 issue
8:07PM 1 Voicemail Error
5:50PM 24 Skype for Asterisk: Public Beta available
4:50PM 1 Dialplan SIP call back problem
3:19PM 2 Sound through NAT issue
8:34AM 0 Request for information about Asterisk Business Edition
5:56AM 0 Asterisk 1.6 and RFC4235
4:09AM 1 Out of office
2:27AM 5 Looking for wisdom - One Asterisk system - Multi-incoming trunks
 
Wednesday July 29 2009
TimeRepliesSubject
10:01PM 2 Open Source Pavilion at AstriCon: Your project wanted!
9:25PM 3 Recording calls again
7:14PM 2 Recording Calls
1:52PM 0 Instant messaging (yeah, again)
1:51PM 6 HPEC > VPM ?
12:49PM 1 Matching Originate action with its NewChannel event
6:55AM 0 SIP client Resp code
5:41AM 0 question about Asterisk-GUI
12:53AM 4 Misunderstood thing
 
Tuesday July 28 2009
TimeRepliesSubject
11:06PM 6 Possibly I don't understand sip peers
9:14PM 3 CIsco 7960 + asterisk: hepl needed
8:32PM 2 chan_dahdi.conf parser question
7:37PM 2 Updated patch for 8824?
6:59PM 2 AGI with queues status
6:33PM 2 outbound calls not reaching vitelity
2:54PM 2 sip realtime with caching
12:50PM 1 CDR.C
11:06AM 0 Call history problems from B2BUA
11:05AM 0 Asked to transmit frame type 256, while native formats is 0x4
11:01AM 2 sip trunk that fails over time
10:52AM 0 Meetme Enter/Leave Sounds
9:06AM 0 Asterisk Crashing on chan_h323
8:52AM 0 Inquiry:Asterisk "*" character dialing for IN service
7:23AM 0 crd wrong destination....
5:05AM 1 Inquiry:Asterisk pbx announcements
5:01AM 7 Inquiry:Asterisk Inter digit delay
 
Monday July 27 2009
TimeRepliesSubject
11:49PM 4 Fax for Asterisk quick question
8:30PM 0 Milkfish
6:19PM 0 Cell phones and (no) rings
4:28PM 14 Asterisk core dumps files
4:15PM 2 Player to listen to WAV files using an hardphone
1:50PM 2 Asterisk and Kamailio NAT problem
1:29PM 2 disposition "answered" after authenticate??????????
1:28PM 5 INVITE Privacy Information
12:11PM 1 dahdi kernel panic
10:09AM 0 authenticate with password file on asterisk
9:37AM 0 mobile hosted pbx
7:04AM 0 Emulating attended transfer through the dialplan
2:37AM 1 AMI not show originate on CLI
 
Sunday July 26 2009
TimeRepliesSubject
5:19PM 3 Not getting inbound CallerID name on Asterisk
3:51PM 0 after 1.4.26 upgrade: "ast_carefulwrite: write() returned error: Broken pipe"
3:13PM 4 Verbose() messages go unnoticed
5:50AM 0 MeetMe time doesn't show up in CDRs?
12:22AM 0 Audiocodes MP114, 2xFXS, @xFXO - does any one have configuration files they can share for trixbox?
 
Saturday July 25 2009
TimeRepliesSubject
2:03PM 0 DeadAgi application issue
9:04AM 0 how to remove MWI from a Polycom phone
1:43AM 0 Set custom file name for automon recordings
 
Friday July 24 2009
TimeRepliesSubject
11:08PM 4 TLS Manager
11:02PM 1 EVERY toll free number appears to be in e164.org??
9:43PM 3 How determine extension of who initiated call
8:13PM 13 Goto from a feature macro is not working?
7:22PM 2 Russia Calls Skype/VoIP Security Threat
6:37PM 0 Asterisk-Addons 1.4.9, 1.6.0.3, and 1.6.1.1 Now Available
4:01PM 1 Anonymous Michigan Calls, Skype/Other
2:22PM 3 asterisk users
12:44PM 12 dialplan tips
10:12AM 13 Asterisk on OpenWRT
9:16AM 1 FAX Machine Testing ...
9:04AM 5 Web Browser Pop-up
8:17AM 7 how to match "no callerid" in 1.6 ?
2:30AM 0 best option for Conference timing with native Dahdi support
 
Thursday July 23 2009
TimeRepliesSubject
11:05PM 14 using asterisk on a shared line
9:35PM 1 nortel cs 1000 swtich
5:44PM 3 PRI call progress issue
3:51PM 2 x-lite settings to reach asterisk
3:28PM 0 detect keys before agi starts
2:52PM 3 Analog FXO or IAX DIDS for new facility?
2:35PM 7 Music on hold based on user
12:54PM 3 Asterisk 1.4.25 and attended transfer
8:25AM 6 Test Function if SIP Device is Still Alive
7:51AM 0 how to activate DND on 1.6.0.9
7:30AM 0 Friday 2009-07-24 12:00 EDT: Voxeo Labs on VoIP Users Conference
7:05AM 1 Using Of function SHARED
3:14AM 2 odd behaviour with AGI and dial agent
 
Wednesday July 22 2009
TimeRepliesSubject
8:03PM 8 Asterisk CSTA
7:25PM 2 grandstream and jitter buffer
5:44PM 0 Attended transfer and 'pbx-invalid' - 1.4.26
5:44PM 0 Asterisk as a "gateway"
3:44PM 10 OT - Do analog gateways detect a phone is plugged in or out ?
3:31PM 11 A reason TO run Asterisk as root
2:42PM 5 CallerPres SIP headers Analog Phone
12:43PM 8 ExecIf and empty variables (early evaluation)
12:20PM 3 german voiceprompts
11:27AM 3 Callin Numbers.
8:30AM 3 sip configuration masking the peers
7:24AM 3 Waiting for a call to complete with AMI Originate
5:51AM 1 voicemail does not work from local calls!!!
5:48AM 6 Inquiry abount Asterisk "extensions.conf"
 
Tuesday July 21 2009
TimeRepliesSubject
11:29PM 2 Phone system "ping" checker
10:48PM 0 MWI using Asterisk and external mail server
8:53PM 0 logging cdr to mysql does not fill clid field
8:37PM 4 Free Fax for Asterisk -- benchfax utility hangs.
5:54PM 3 Asterisk 1.4.26 Now Available
3:04PM 2 Dialplan step that I do not have
2:49PM 0 Vdex-40 for sale
2:15PM 3 astmanproxy?
1:46PM 2 Connecting multiple office with multiple servers
12:51PM 5 best practices for running asterisk as SIP B2BUA
12:46PM 2 Graphical Call Manager Allowing Transfer of Any Call?
11:46AM 7 Channel Variables in a Call file?
11:06AM 1 Asterisk and G.729 codec: short questions
10:48AM 3 Scalability and stability matters
10:43AM 1 externalIVR() and how to do actions
10:09AM 0 Audio lost on reinvite
9:05AM 0 Gatekeeper Routing Mode not Working
5:51AM 0 Asterisk Call Transsfer
4:01AM 4 how to use patgen and pattest for PRI card?
 
Monday July 20 2009
TimeRepliesSubject
11:18PM 0 Error: Invalid SIP message - rejected , no call id
10:10PM 2 How to restrict registrations by useragent?
9:50PM 0 Vote on whether SipPhone should support ISN routing.
6:16PM 3 Event Log
6:10PM 0 [asterisk-dev] MeetMe feature request: bypass pincode
5:09PM 12 Digium TDM400P in Soekris net5501-70?
3:36PM 5 What am I doing wrong?
2:13PM 1 callforward with asterisk-gui.problem with stdexten
10:18AM 2 asterisk freepbx difference or solutions..
3:29AM 0 [asterisk-user] MeetMe feature request: bypass pincode
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
12:54AM 0 No subject
 
Sunday July 19 2009
TimeRepliesSubject
10:35PM 1 CyberData SIP-enabled VoIP Intercom
7:47PM 0 Asterisk-gui 2.0 Asterisk 1.4.26-RC6 Analog trunks
1:36PM 0 Asterisk not ACKing some 407 Proxy Auth Required requests?
11:31AM 0 Asterisk or 3CX?
9:57AM 4 DAHDI Error and poor audio quality
 
Saturday July 18 2009
TimeRepliesSubject
7:43PM 1 wcte12xp0: Missed interrupt
4:05PM 3 perhaps libpri issue (thought it was a dahdi issue )
3:11PM 15 Count Available Queue members
2:22PM 1 chan_mobile one device per dongle?
1:27PM 9 Latest chan_mobile
4:47AM 9 Asterisk to PBX
12:31AM 0 Possible WaitUntil Bug
 
Friday July 17 2009
TimeRepliesSubject
11:35PM 8 Truecall
9:25PM 2 Voicemail ODBC storage table schema
7:19PM 0 SPAM
6:53PM 1 Realtime difference sipusers sippeers
5:42PM 0 dahdi_tool question for PRI or T1
4:30PM 13 Delete voicemail after couple of days
2:29PM 7 quenstion about asterisk
1:22PM 0 How to Play IVR and Read DTMF During Active Call?
11:13AM 0 Friday reminder
9:58AM 0 Queue member (Agent) does not Dial
9:11AM 7 dialplan number matching
8:22AM 10 How do I create an IVR/Dial Group that works properly?
6:48AM 12 Skill based routing
6:41AM 1 [HELP] - Conference bridge
6:33AM 1 MoH - can the volume be adjusted
6:08AM 10 Asterisk Error
1:19AM 16 2 Problems with 1.6.2
 
Thursday July 16 2009
TimeRepliesSubject
10:59PM 1 Compilation error
9:45PM 2 Stop recording on SIP attended transfer
9:39PM 5 800 number portability
9:38PM 0 Unique id used for call recording missing from CDR data for transferred call
8:00PM 1 possible to configure 2 servers - one is backup system for the other?
5:26PM 2 iax.conf, IP-based access control
4:51PM 0 early-dial SIP 484 "incomplete address", dialplan patterns and international calls
4:00PM 6 T38 negotiation, the last step !
2:57PM 2 Voicemail login incorrect
2:46PM 1 H323 situation
1:51PM 1 Sending faxes with T.38 problem. Fax for Asterisk (no SpanDSP) - 1.6.1.1
12:44PM 0 AGI to announce temperature from weather.com XMLfile
12:08PM 0 Struggling with Macros and "s" Extension
11:49AM 5 AGI to announce temperature from weather.com XML file
10:21AM 1 Sending things to Jabber but not within an extension
7:16AM 3 Mexican ITSP needed
4:44AM 1 advices on how to debridge/rebridge a call?
 
Wednesday July 15 2009
TimeRepliesSubject
11:57PM 0 16.7.2009 ID69656 71% 0FF on PFIZER !
10:25PM 5 Iphone setup
10:19PM 14 PRI hunt group
9:26PM 7 DEVICE_STATE() and Asterisk 1.6.0.10
7:50PM 0 Queue wrapuptime as Global option
6:47PM 0 Read/Write Codec formats
2:42PM 4 Phantom CallerID on transfers
2:18PM 2 ResetCDR after GotoIf doesn't set dst correctly, Is this a bug?
1:40PM 2 Generic question about PBX PRI installs
1:14PM 3 USB phone with Asterisk under Linux
12:42PM 0 Door Phone
12:04PM 0 [asterisk-dev] Question
12:02PM 0 Howto change CDR record on calling channel from called thread?
6:19AM 4 How to ask questions the smart way
4:34AM 3 how to enable dial to alex@asterisk.blurb.com
3:07AM 2 call transfer using DTMF
 
Tuesday July 14 2009
TimeRepliesSubject
11:06PM 0 TE120P loosing link...
10:03PM 6 QoS
6:45PM 0 Help in oh323 Gatekeeper + does not know what to do when bridging the call
5:20PM 5 Polycom Spectralink 8002 WiFi Phones
4:09PM 2 Error
3:39PM 9 How to block inbound call with Asterisk?
2:14PM 6 Is Enum safe from spammers?
10:44AM 2 Asterisk 1.4.26 final release - What is blocking?
10:10AM 6 Asterisk and several clients behind NAT
8:40AM 3 Help in oh323 Gatekeeper
8:31AM 1 How to count Parked calls?
7:55AM 1 unknown RTP codec 126 ??
2:00AM 0 ooh323 doesn't know what to do when bridging calls
12:10AM 9 Why CDR is recording dst value = h?
 
Monday July 13 2009
TimeRepliesSubject
7:27PM 0 chan_ooh323.so and chan_h323.so
5:58PM 12 #exec in #include'd file
4:37PM 0 ooh323 and h323, it accept the call even not added in h323.conf
3:27PM 0 Polarity Reversal Incorrect
12:19PM 3 open source call center application for Asterisk
12:02PM 0 Push-To-Talk?
10:47AM 0 Go t SIP response 420 "Bad Extension" back from
6:56AM 10 How to Change size of CDR(accountcode) variable?
5:06AM 3 transfer option and pressing #
4:23AM 4 is Asterisk reliable for a call center application??
2:23AM 3 Trouble with originating a call through Asterisk Manager Interface
 
Sunday July 12 2009
TimeRepliesSubject
4:32PM 0 1.6.0.10: server locks up on iax max_retries
8:20AM 0 Missing CLI
 
Saturday July 11 2009
TimeRepliesSubject
9:57PM 0 MACRO-INCOMING-CALL-TO-EXTENSION
4:20PM 4 ooh323 and h323
12:55AM 6 Suggestions for web based soft phones
 
Friday July 10 2009
TimeRepliesSubject
7:46PM 3 Originate (Executing a System Command)
7:19PM 2 beeping in headsets from queue callers
6:30PM 0 Meetme problem (talk detection/opt) in 1.6.1.1
4:19PM 0 ID0151 C-A-N-A-D-l-A-N P-H-A-R-M-A-C-Y
3:57PM 8 Dropped Call Problem -- Looking for ideas and a consultant.
3:55PM 0 Dell poweredge T100 & TE420
3:19PM 4 Lagged Extension
2:30PM 5 [Fwd: confirm f1ab6c493110edited]
1:39PM 2 Video Call
12:45PM 1 Kate AEL syntax ?
9:22AM 1 Friday July 10th: Gigaset DECT/SIP phones have come to the USA
3:50AM 1 Educational institutions: Your Asterisk experiences wanted!
 
Thursday July 9 2009
TimeRepliesSubject
10:34PM 7 can 2 quad T1 cards work in 1 quad core amd server
10:10PM 16 Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
8:37PM 8 PRI failover to SIP trunk
7:20PM 1 Connecting two Asterisk together via SIP + DISA
7:13PM 2 1.6 macro deprecation, dial macros
6:51PM 36 setting up phones
5:13PM 1 Weird audio problem with remote IVRs + DMTF
1:38PM 0 q: port forwarding or NAT
12:05PM 3 Setting up a "secure" AMI?
10:11AM 0 Failed to read gains: Invalid argument
9:30AM 0 Rtp keepalive
9:26AM 6 is possible to sen sms with asterisk in Spain?
7:47AM 1 OT - How to indent AEL file
7:01AM 2 CIDlookup
6:35AM 1 DIDForSale July Special (No Activation on new DID Purchases)
2:06AM 2 Dial stops trying after ~30s regardless
1:39AM 1 q: am i mixing somethign up?
1:27AM 0 q: sip registration fails...
 
Wednesday July 8 2009
TimeRepliesSubject
9:05PM 4 Queue autopause
8:29PM 0 Fwd: q: install asterisk + asteris-gui: SOLVED
6:50PM 3 q: which Browser-GUI do u guys use?
6:49PM 15 q: install asterisk + asteris-gui
2:40PM 0 Grandstream GXP-1200 & G.722?
1:35PM 7 Restarting of B-channel on span 1
12:11PM 1 calculate data traffic
11:36AM 0 asterisk + cisco as5400 t.38 fax sending.
5:31AM 3 Asterisk and Skype
2:06AM 5 g.722 + loudness
1:55AM 1 One Way Audio from External Sip Soft & Hard Phone
1:09AM 0 [asterisk-user] AGI control stream file
 
Tuesday July 7 2009
TimeRepliesSubject
10:26PM 2 asterisk addon mysql - is mysql connection persistent
6:31PM 1 Play a recorded message when a fax is detected ?
6:05PM 3 MixMonitor/Queue and Tranfers
5:40PM 20 Caller ID (name) - where does it come from?
3:26PM 4 Automatic Gain Control
1:44PM 10 Resetting Day/Night setting
1:42PM 9 documentation of DAHDI dial options
1:05PM 3 Answering the nTh call ...
6:29AM 2 How to debug "Nothing to pick up" ?
12:30AM 2 Voicemail attachments not working
 
Monday July 6 2009
TimeRepliesSubject
8:59PM 1 Bug or Not?
7:00PM 2 Get channel string
4:11PM 0 Listed agents in queue not ringing
3:37PM 5 Variable using AMI
3:28PM 4 Small site survivability
2:00PM 4 migrate from zaptel to dahdi
1:51PM 2 false answer on zaptel
1:27PM 6 Asterisk & Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error
9:55AM 0 Iax trunk quality
9:53AM 1 Monitor
9:39AM 0 asterisk and mISDN on Solaris
8:37AM 7 SIP registry fails during night
7:04AM 3 How to notify that a message is waiting using RingTones with a Dahdi-connected analog phone ?
5:07AM 1 Asterisk + kamaili MWI(Message waiting Indication)
3:57AM 5 Dial cmd help
2:19AM 2 asterisk-GUI
12:09AM 8 What is the best way to share extension state
 
Sunday July 5 2009
TimeRepliesSubject
9:06PM 1 SIP IP-Trunk to be authenticated based on username and password, not IP address
7:02PM 0 (slightly OT) SIP redirect
5:11PM 3 Queues recording & CDR
5:09PM 8 chan_mobile help.
1:15PM 2 Fax for Asterisk download selector broken?
11:25AM 3 Source for OpenVox cards?
 
Saturday July 4 2009
TimeRepliesSubject
1:17PM 3 Call parking with ISDN
10:54AM 0 Asterisk and ENUM
10:24AM 1 Channel / peer available
2:14AM 0 debian lenny, asterisk1.6 + freepbx2.5.1
12:12AM 2 Music on Hold
 
Friday July 3 2009
TimeRepliesSubject
7:10PM 0 DAHDI CDR problem
6:38PM 2 dahdi_dummy configure
5:39PM 1 Asterisk + Openfire
3:55PM 1 New attempt : Trigger an action when B number answers the call
3:42PM 0 Fw: Trigger an action when B number answers the call
3:30PM 2 Trigger an action when B number answers the call
3:29PM 0 e164.org and tollfree ENUM records
2:14PM 1 *Sort of Commercial* TracFone's $45 unlimited offer to 'stun' rivals
1:46PM 2 DTMF is not working occasionally over IAX Trunk
12:26PM 6 MISDN/asterisk problem (not sure where from)
11:24AM 9 Asterisk capacity
10:58AM 3 Problem configuring TDM400
9:50AM 0 Typ of Number / Modify in Peer-Definition
7:47AM 0 asterisk-users Digest, Vol 60, Issue 9
6:18AM 13 Some IAX calls do not disconnect.
4:53AM 0 Converged mail box sizes
3:15AM 2 DAHDI
12:59AM 1 Zimbra IMAP authentication - SOLVED
 
Thursday July 2 2009
TimeRepliesSubject
11:59PM 10 Dial
7:36PM 1 Rajkiran Reddy sent you a Friend Request on Yaari
7:20PM 12 Using a mobile phone via USB as an extension
6:55PM 6 need help, service unavailable, registered but call does not get through
6:05PM 3 Why Asterisk + Kamailio ?
4:33PM 5 Grandstream 2010 and blinky lights
2:53PM 8 Using the PBX Directory from a Blackberry
12:34PM 1 AGI Transfer?
8:47AM 0 Ext1: Channel X parameter on PRI
8:09AM 2 Nortel pbx & dtmf issues
 
Wednesday July 1 2009
TimeRepliesSubject
11:03PM 2 /var/lib/asterisk/sounds does not exist
8:51PM 26 g729a compatibility
7:18PM 5 Registrations problems to SIP-provider.
6:19PM 4 UK Vodafone femtocells now available
5:25PM 2 Testing the manager.conf: sending and receiving commands
3:16PM 13 * as VM for legacy PBX?
8:54AM 5 Fwd: Unknown udp ports listening experts calling !
6:17AM 10 Multi-tenant parking broken in 1.6.1.1?
12:49AM 11 Welcome Message