asterisk users - Feb 2009

Saturday February 28 2009
TimeRepliesSubject
7:24PM 4 clone X100p+dahdi dial out works only after receiving call
6:02PM 0 Temporaneamente assente
5:21PM 3 No rtp activity
2:53PM 3 Remote connection to an Asterisk server
11:42AM 0 using an eicon diva server card with asterisk.
2:17AM 0 rfc2833 vs. sipinfo and network weirdness
 
Friday February 27 2009
TimeRepliesSubject
9:32PM 2 dialing timing problem?
9:07PM 3 what is the effect of high LBO settings?
8:47PM 1 TE121B server recommendation
7:45PM 6 Continue in dialplan on hangup
6:05PM 2 Switch Options for a service provider
5:40PM 1 API command monitor to record only the input channel
3:37PM 34 building a phone
1:20PM 0 [HOWTO] Priorize one destination over another on a link
12:53PM 3 change language and playback issue
10:58AM 1 Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available
1:58AM 17 call file concurrency
 
Thursday February 26 2009
TimeRepliesSubject
11:58PM 0 using Cisco IP Communicator with SIP to Asterisk
11:33PM 3 Question about Do Not Disturb
8:44PM 5 changing /etc/dahdi/system.conf
7:33PM 3 Current state of Asterisk and Virtualization?
7:27PM 0 Friday Feb 27th at 12 Noon EST: Polycom Applications
7:16PM 0 asterisk 1.4.23.1 and mISDN 1.1.8 segfaults
6:40PM 4 incoming call problem
6:11PM 1 Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4
5:39PM 0 Residential portals and real world scalability
3:35PM 0 [cdr_odbc] error: Cannot insert the value NULL into column 'calldate'
2:30PM 10 Odd Read App Issues
1:41PM 14 ATA recommendation (wih FTP provisioning)
1:36PM 4 Getting SIP field P-Asserted-Identity from EAGI
1:30PM 0 Patton 5.3. How to get incoming calls ? [SOLVED]
10:59AM 5 call-limit on a per destination basis
10:25AM 5 HP DL380 G5 with TE420
10:19AM 2 Dictate
9:00AM 0 Need US Dialing Account with Asterisk
5:44AM 2 codec_dahdi and Asterisk 1.6.0.6
5:28AM 2 asterisk 1.6.0.5 and IM
3:50AM 9 Problems with Outbound Calls
2:48AM 1 DTMF Forwarking Problems.
 
Wednesday February 25 2009
TimeRepliesSubject
11:07PM 8 AGI problem using mono (.Net)
9:17PM 1 Congestion Tone
8:59PM 6 SheevaPlug Development Kit
8:40PM 1 Realtime database function help
8:16PM 0 Call from '6000' to extension rejected becauseextension not found
8:11PM 2 Call from '6000' to extension rejected because extension not found
7:11PM 0 Patton 5.3. How to get incoming calls ?
6:56PM 5 CDR - Asterisk-Stat and PHP5
6:46PM 20 DID's in a specific rate center
6:29PM 1 cannot allocate memory
3:24PM 8 dahdi wcb4xxp and fax
3:00PM 10 TE121 on Asterisk
2:39PM 1 Stuck Parked Calls?
12:54PM 4 SIP_CODEC variable
12:45PM 0 usegmtime=yes for cdr_custom
10:38AM 3 Asterisk with Internet connectivity
9:45AM 4 bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA
9:02AM 5 switchtype QSIG and Asterisk implementation
8:02AM 0 Call files with extensions.ael : "One app must be specified"
7:01AM 0 HDLC Errors
6:11AM 2 Dropping RTP packets
3:29AM 0 Asterisk security between two servers
3:14AM 0 Problem redirecting user running a Dynamic feature
 
Tuesday February 24 2009
TimeRepliesSubject
10:17PM 1 Incoming call
10:02PM 3 Multiple SIPGate accounts.
8:52PM 5 receiving 1st digit from a variable
8:41PM 3 Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card
7:44PM 18 Gosub behavior change <=1.6.0.5 to 1.6.0.6
7:08PM 3 what is the correct character to separate application parameters: , or |
5:33PM 9 Aastra phones
2:16PM 1 COSTA RICA - E1
1:49PM 1 API hangup command
12:59PM 0 db_dump185.c missing if Asterisk 1.4 source file
12:37PM 5 astdb and Debian : can't use db4.5_dump
9:56AM 4 chan_sip and database integration
9:55AM 5 asterisk -f and restart now
9:15AM 5 Swichted digits on received number from fax on an fxs port
7:59AM 14 multiple asterisks in a server
7:27AM 0 Free US DID
4:12AM 0 Does audio have to flow through vitelity servers ?
3:44AM 0 Legacy Cisco ATA186I1
2:10AM 7 Managing the spiralling costs
2:07AM 10 Polycom Spectralink 8002 Configuration
1:35AM 14 HDD FULLL
1:28AM 2 weird problem
12:17AM 8 building asterisk-1.6.0.6 failed!
 
Monday February 23 2009
TimeRepliesSubject
11:59PM 19 GSM codec is a good choice ???
11:30PM 4 strange text message:)
10:44PM 0 Windows Mobile MWI and asterisk
9:16PM 1 Inbound call to IVR drops after 21 seconds?
8:56PM 8 don't get 2.0 gui to run on asterisk 1.6.0.5
8:26PM 2 Compiling asterisk-addons-1.6.0 under Debian 2.6.18?
8:23PM 0 Asterisk 1.6.0.6 Now Available!
7:29PM 2 Voicemail and ADSI
7:13PM 4 5ess
6:22PM 0 FreePBX in a Cloud With a Click - FreePBX secured and optimized for Amazon EC2
5:37PM 8 Delete all
4:52PM 0 Astlinux 0.6.3 Released
3:55PM 0 302 Redirect Handling
3:19PM 0 problem with nortel 2002 disconecting
3:13PM 3 Asterisk/Skype update
12:19PM 1 Flash Operator Panel with Asterisk 1.6
8:30AM 1 receive fax problem
7:32AM 0 Can't compile Dahdi on SuSE 10.1/Kernel 2.6.16.54-0.2.5-smp/GCC 4.1.2
5:06AM 6 Asterisk 1.6.0.5 as non root and moh perm issue
 
Sunday February 22 2009
TimeRepliesSubject
8:11PM 1 I can`t send DTMFs through FXO lines - dahdi
6:33PM 10 Intel Vs AMD
 
Saturday February 21 2009
TimeRepliesSubject
3:30PM 1 VoIP Information in CDRs
12:55PM 0 Where to find db1_dump185 in debian packages ? [SOLVED]
12:27PM 6 DIAL() application 'g' option
11:39AM 0 Cisco Phone losse regsitrations with Asterisk
4:37AM 0 help needed -- chanspy
2:02AM 3 IAX2 - now known as RFC 5456
 
Friday February 20 2009
TimeRepliesSubject
10:55PM 1 Vacation reply
10:34PM 0 hi
9:18PM 1 CDR fields in 1.6
8:01PM 0 asterisk-users Digest, Vol 55, Issue 64
7:38PM 0 SIP debug messages
5:33PM 13 zaptel telephone cards and asterisk in another pc
5:29PM 0 Table definitions for Realtime 1.6...
4:56PM 0 slip r errors nortel switch
4:41PM 12 Polycom Phones start to break up after being up a LONG time
3:26PM 2 real time connection failed
1:29PM 0 SMS to generate a call back
9:33AM 0 Qualify sip users behind remote registrar
7:02AM 1 SIP Proxy behind NAT talkinf to ASterisk with public IP
 
Thursday February 19 2009
TimeRepliesSubject
11:27PM 53 AGI script
10:00PM 1 TDMOE Timing
8:48PM 4 DTMF
7:43PM 1 Annoying silence suppression effect on my digium E1 card with the VPMADT032 module
2:54PM 1 queue_variables() function
2:33PM 12 check if not human
1:47PM 7 Busy status of a snom connected to two asterisk servers?
1:25PM 3 Not answering call when queue is full or has no members
11:53AM 0 Asterisk BLF to Cisco CME
11:00AM 0 About Hint Configuration in Asterisk.
10:06AM 0 sip phone cant hear the caller
8:32AM 0 Friday Feb 20th 12 Noon EST: Jason Fischl from Counterpath on VUC
6:47AM 4 Managing SIP hardphones call history
5:22AM 1 DeadAgi Application in asterisk 1.6
 
Wednesday February 18 2009
TimeRepliesSubject
11:08PM 2 Understand SIP REFER
9:18PM 1 Asterisk on the Cloud With a Click - pre-built Asterisk Amazon EC2 instance
8:48PM 1 call file FXO channel problem
8:42PM 0 Open Source in an Economic Downturn: Asterisk stories
7:00PM 0 No Audio PlayBack Asterisk 1.6 Dahdi 2.1.0.3
6:28PM 3 trunk to trunk
5:54PM 3 Please help test the gender detection moduleat 575-613-4392
3:27PM 0 connection to siemens hipath
2:58PM 1 directrtpsetup=yes does not work in 1.4.23.1
2:51PM 1 Need help on Forwarding
2:38PM 1 asterisk-users Digest, Vol 55, Issue 52
2:19PM 4 US DID
1:36PM 2 Accumulated call time
1:18PM 0 Ditech API
12:37PM 0 Detecting which party initiates a hangup
11:19AM 35 AGI pdf book
9:04AM 8 Setting SIP header on agent calls made by a queue
8:28AM 1 Distributed presence in 1.6
4:19AM 3 xorcom hardware good idea?
1:47AM 0 life safety system and VOIP
1:05AM 0 AgentCallbackLogin via dialaplan and device state
12:37AM 13 Open Source in an Economic Downturn: Asterisk stories needed
 
Tuesday February 17 2009
TimeRepliesSubject
10:08PM 2 SLA and Flashing BLF
10:00PM 5 ztdummy compile under 2.6.28 ?
9:02PM 0 Caller Hangup detection
8:04PM 3 call file bug?
6:06PM 4 Asterisk supports SIP-T?
5:57PM 0 Lost with Patton 5.3 web server. Registration ?
5:27PM 7 Question regarding custom announcements in queues.conf
5:08PM 0 Swift - detection of multiple digits unreliable on my system
4:54PM 0 Asterisk 1.4.21.1 intermittent presence working with Polycom
4:46PM 0 Questions about OpenSky - Asterisk to Skype Gateway
3:19PM 14 Network architecture
2:57PM 2 Packet Truncated - Choppy Audio
1:06PM 3 Pingable and Unreachable at the same time !
11:05AM 2 What is the purpose of membermacro in queues.conf
9:30AM 0 freemin managment for sim cards
8:33AM 11 zaptel compile kernel problem
7:02AM 0 unistim channel problem
6:51AM 3 Stress Testing IVR
2:37AM 0 Optimizing this script for calling Skype users from Asterisk
2:10AM 2 Obtaining callerid on a PRI for billing purposes (with non toll-free numbers)
12:30AM 0 mp3player() to shuffle playlist
 
Monday February 16 2009
TimeRepliesSubject
8:51PM 0 How to beep before transfer ...
6:57PM 0 vmsecret question
6:38PM 5 DTMF not completely muted
6:29PM 33 Please help test the gender detection module at 575-613-4392
4:55PM 6 command show channels concise
3:14PM 5 AstDB wildard searches
2:43PM 1 TeleKaam - Voice Portal for Students and Parents
10:22AM 1 SpanDSP question for Steve
9:14AM 11 Faxing with asterisk
3:12AM 0 Microsoft Recite
 
Sunday February 15 2009
TimeRepliesSubject
8:43PM 0 call-limit
3:01PM 3 Gizmo SIP / Skype gateway
7:36AM 3 No such command 'core stop now'
2:23AM 2 licensed g729
 
Saturday February 14 2009
TimeRepliesSubject
3:28PM 2 Asterisk CLI problem if run from /etc/inittab
12:28PM 1 Progress() and Proceeding()
12:11PM 1 Call Fowarding and Polycom Phone
12:54AM 1 Asterisk 1.6.x timing API
 
Friday February 13 2009
TimeRepliesSubject
11:17PM 0 Slow hangup - Australia - analog - incoming calls
10:13PM 0 Asterisk 1.6 CDR fields...
9:18PM 0 Asterisk 1.6.0.6 Release Candidate 1 Now Available
8:24PM 3 GUI interface to manage business edition
8:08PM 3 linksys PAP2t and asterisk
7:19PM 5 OpenSky: Digium Skype gateway?
5:59PM 9 Asterisk on EC2 cloud computing - price assumptions - your brain needed
5:39PM 2 Asterisk 1.6.0.5 and Aastra phones...
5:25PM 0 Asterisk and Amazon EC2 cloud service tutorial
3:55PM 2 Continue processing AGI script after hangup
9:42AM 20 Cisco IP Phone 7940G.
9:18AM 24 PRI Test Lab
5:35AM 1 ExitIf() convention?
5:33AM 0 Re : Asterisk Queue and URL Calling
5:04AM 6 MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am
3:51AM 6 Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status
2:51AM 0 zaptel for asterisk
 
Thursday February 12 2009
TimeRepliesSubject
9:34PM 1 Queue problem
9:04PM 18 CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?
6:38PM 1 1.6.1-rc1 errors
5:03PM 4 Asterisk Queue and URL Calling
4:27PM 5 Caller ID replacement
3:59PM 3 OSLEC not being loaded on Ubuntu Intrepid
3:05PM 1 Problem with parking
2:52PM 4 Multiple caller id ...
1:55PM 1 After Monitor() files disappear
11:53AM 1 g723 llicense
8:56AM 5 Siemens Hipath PRI to Asterisk Call Routing?
8:27AM 3 Keep your passwords secure .. (VoIP hacker news)
8:18AM 0 Friday the 13th Muhahaha Allison Smith and more on the Polycom Applications
3:57AM 0 IDAP T1
1:18AM 7 Strange dialplan matching issue
 
Wednesday February 11 2009
TimeRepliesSubject
11:52PM 11 WiFi SIP phone w/VPN?
11:25PM 0 Problem with AMI action userevent
9:35PM 4 call picking and transfers
7:08PM 1 The download link, why server down?
6:29PM 0 Intercom/Doorbell Integration
6:21PM 4 asterisk across a firewall
6:01PM 0 ChanSpy problem
2:28PM 5 DTMF tones mid conversation
2:24PM 0 Looking for 'remote Asterisk hands' support in Mexico
2:22PM 3 Billing and Soft Switch.
2:20PM 0 Asterisk AGX addons compile issues
10:14AM 6 call forward all except the extension it is forwarded to
9:43AM 3 OPTIONS packets
 
Tuesday February 10 2009
TimeRepliesSubject
11:32PM 2 Max person in meetme conference
10:11PM 3 Aastra phone crashes with Asterisk 1.6
5:23PM 7 connecting 66 analog phones to asterisk - hardware suggestions
3:00PM 1 unistim and transfer calls
10:54AM 0 Skip password option in voicemail.conf
10:24AM 10 Disabling Echo Cancellation on a per Call basis
10:20AM 1 Asterisk how many calls handle using H.323 to SIP conversion?
7:24AM 26 What do you use? .conf or AEL?
4:23AM 0 hosted voip?
 
Monday February 9 2009
TimeRepliesSubject
11:39PM 7 SMS /w Asterisk
11:10PM 0 Audiocodes - Disconnect Supervision
9:44PM 0 [asterisk-dev] 1.4 and CDRs -- The Breaking Point
9:43PM 2 Is "a=fmtp:101 0-15" a legal option in SDP ?
9:25PM 3 asterisk registered as UA
9:23PM 5 What t38pt_udptl is ? Explain T.38 in 1.4
9:03PM 3 SIP password encryption
8:28PM 15 Hangup extensions via CLI?
7:44PM 0 Problem with upper case extension names
7:22PM 2 Call drops after a minute on 1.6.0.5
6:52PM 4 How to make the Asterisk-GUI work with DAHDI..please??
5:50PM 1 Transfer Asterisk 1.6 Telephone IP
5:44PM 0 problem getting asterisk behind NAT to run with sipproxd
5:12PM 0 Problem with AMI originate
3:54PM 2 Noisy Ring Back Tone with TE205P card
3:11PM 2 reinvite
2:20PM 10 Michael Graves post
11:33AM 2 InUse&Ringing
6:20AM 4 Asterisk and CIsco 1760 SIP ?
4:39AM 3 Asterisk + voxbone ==> Failed to authenticate user
4:29AM 1 chan_oss.c:585 setformat: Unable to re-open DSP device
2:43AM 4 meetme application
 
Sunday February 8 2009
TimeRepliesSubject
5:33PM 0 Streaming meetings vs conference hardware
 
Saturday February 7 2009
TimeRepliesSubject
10:45PM 6 can anybody tell me how Magic jack can be so cheap ????
9:08PM 4 Minimum version for asterisk and iaxmodem
6:41PM 0 One way audio after IVR tree
5:39PM 0 GROUP() decrement
2:42PM 0 Asterisk 3rd party developed commercial software sales licensing platform
1:38PM 0 A Simple Asterisk Based Toll Fraud Prevention Script
7:31AM 4 put the hostname of asterisk in the callerid uri to avoid NAT problems
3:41AM 4 VPN and Asterisk
2:16AM 1 Running asterisk on ARM (TS-7800) 1.4.23.1
 
Friday February 6 2009
TimeRepliesSubject
10:11PM 47 Credit Card processing machines
9:01PM 13 Security issue
8:53PM 18 AgentCallBackLogin no longer works after installing asterisk 1.6
6:29PM 0 [asterisk-user] $100USD for anyone who can install Chan_SCCP for me
4:29PM 2 upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call
3:36PM 0 H323 stress test
3:25PM 0 Asterisk as a dial in server for internet
3:06PM 3 Add-on for SRTP and SSIP
2:08PM 1 set caller id on outgoing calls through BRIISDNlines
1:28PM 1 set caller id on outgoing calls through BRI ISDNlines
12:49PM 0 Incoming fax detection on mISDN hfcmulti B410Pcard
12:36PM 0 set caller id on outgoing calls through BRI ISDN lines
11:02AM 2 Rewriting numbers while processing dial plan?
9:15AM 2 Monitor and SIP transfers (SIP REFER)
8:52AM 7 asterisk and DNS
7:00AM 0 Java IAX Implementation
3:51AM 0 Getting DIALSTATUS from SIP provider
 
Thursday February 5 2009
TimeRepliesSubject
8:20PM 23 Newbie query: how to write priority n+101
8:12PM 0 Patton M-ATA and T.38
5:02PM 0 Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ? [SOLVED]
4:17PM 3 Amazon Flexible Payment System - micropayments finally cracked?
4:09PM 0 Friday Feb 6th at 12 Noon EST: Polycom and Application Development
3:56PM 1 manager API
2:56PM 17 Crash Hard, Crash Often
2:40PM 0 Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ?
2:17PM 5 musiconhold realtime queue
1:06PM 1 SIP Authentication only on auth-user?
12:00PM 4 no need to dial areacode
10:46AM 6 extensions ending with "#"...
9:08AM 0 sendto syscall: EPERM (Operation not permitted)
8:05AM 3 Configure Asterisk to preserve SIP header?
7:22AM 2 TDM400P Circuit/channel congestion problem
5:46AM 6 Autodialler query
12:17AM 3 hardware that can accomondate 2 TDM24
 
Wednesday February 4 2009
TimeRepliesSubject
9:09PM 0 [asterisk-dev] RFC 2833 DTMF w/ Level 3 Sonus
8:29PM 0 Problems with 9133i config
7:47PM 0 Problem with MOH and streaming music on 1.6.0.5
6:39PM 2 Stopping chanspy followup
4:09PM 0 T1, FoneBRIDGE, and dropped D-Channel
3:53PM 6 Call parking
3:37PM 3 siemens hipath 4000
1:34PM 3 question on originate call
12:17PM 4 escaping regular expression
12:00PM 0 BerkeleyTIP Feb 7 Sat Global Meeting - Ekiga3, Asterisk, KDE, GPGPU, Debian Edu, GStreamer
8:01AM 0 Audio lag on SIP connections...
7:39AM 0 Stopping chanspy
6:16AM 6 AOC-E pass through
2:44AM 3 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Tru nk with Pol o com Video Con ferencin g Un it
2:42AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol o com Video Con ferencin g Un it
2:41AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol o com Video Con ferencin g Un it
2:39AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol o com Video Con ferencin g Unit
2:37AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol ocom Video Con ferencin g Unit
2:35AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol ocom Video Con ferencing Unit
2:34AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Re : Trunk with Pol ocom Video Con ferencing Unit
2:33AM 0 Out of Office: Out of Office: Out of Office: Re : Trunk with Pol ocom Video Conferencing Unit
2:31AM 0 Out of Office: Out of Office: Re: Trunk with Pol ocom Video Conferencing Unit
2:30AM 0 Out of Office: Re: Trunk with Polocom Video Conferencing Unit
1:02AM 1 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : som e kind of t i meou t problem in pbx _sp ool.c
1:00AM 1 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t i meou t problem in pbx _sp ool.c
12:59AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t i meout problem in pbx _sp ool.c
12:58AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t i meout problem in pbx _spool.c
12:57AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t imeout problem in pbx _spool.c
12:55AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Re : some kind of t imeout problem in pbx_spool.c
12:54AM 0 Out of Office: Out of Office: Out of Office: Ou t of Office: Re : some kind of t imeout problem in pbx_spool.c
12:53AM 0 Out of Office: Out of Office: Out of Office: Re : some kind of t imeout problem in pbx_spool.c
12:52AM 0 Out of Office: Out of Office: Re: some kind of t imeout problem in pbx_spool.c
12:50AM 0 Out of Office: Re: some kind of timeout problem in pbx_spool.c
 
Tuesday February 3 2009
TimeRepliesSubject
11:57PM 2 Warnings during a compile
10:26PM 4 some kind of timeout problem in pbx_spool.c
9:59PM 3 How to set udptl.conf ?
8:58PM 7 Broken Pipe error while using UpdateConfig command
7:23PM 1 app_rxfax.c: Channel T30 DONE < 0 -- incommplete fax reception.
5:43PM 2 Warning in CLI
5:34PM 7 dahdi-linux 2.1.0.4 released
4:49PM 0 fake ring again when using SIP
4:15PM 4 n-way conferencing
4:04PM 8 Contact lookup
3:50PM 4 Problem with building dahdi-linux RPM
1:14PM 5 Can't compile on NSLU2 LE
12:39PM 0 may convert SIP call in H.323 to words terminator??
10:42AM 0 analysing tools
9:59AM 1 What's the difference between the Jabber Client Mode And Component Mode?
3:27AM 0 Patch to dahdi Chans.pm
2:24AM 3 RBS T1 DID issue
12:25AM 3 Videoconference one-to-many
 
Monday February 2 2009
TimeRepliesSubject
8:18PM 2 Invalid Extension
7:21PM 2 ChanSpy or other variant
6:51PM 0 SIP presence sample script
5:39PM 11 "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail
3:24PM 2 Configuring Patton SmartNode with ISDN2e and Asterisk
2:25PM 0 dundi negative caching
10:12AM 1 Preferred Clock
7:52AM 0 EVRC support
4:02AM 2 Trunk with Polocom Video Conferencing Unit
 
Sunday February 1 2009
TimeRepliesSubject
8:24PM 0 iChat voice (and maybe video?)
5:32PM 25 Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
5:23PM 1 asterisk-users Digest, Vol 54, Issue 109
10:44AM 0 Strange Packet Behavior
10:26AM 3 Need some information on SS7 parameters