asterisk users - Mar 2009

Tuesday March 31 2009
TimeRepliesSubject
8:22PM 2 Queues in memory after startup
7:20PM 0 Dead Call But Still Active
7:18PM 3 dundi show peers - UNREACHABLE but I can ping it!
5:36PM 4 codec payload size
4:38PM 0 Strange voicemail problem when call forwarding off local PBX
4:35PM 3 What is the one thing that polycom can do...
3:07PM 3 dynamic codec preferences
3:02PM 9 DAHDI with OSLEC
1:09PM 14 [Zaptel] Why no driver for PCI voice modems?
12:55PM 0 Dialtones as Inband
10:15AM 0 conference function problems
9:32AM 2 Queue data from within dialplan?
8:43AM 3 iax2 not registering at startup, works on reload
7:42AM 0 USA Pharmacy Discount ID 443
6:37AM 5 PRI problem
1:17AM 3 PAP2T-na Bricked?
 
Monday March 30 2009
TimeRepliesSubject
9:26PM 5 Call-limit=1 breaks attended transfer
9:16PM 3 Newbie trying to make calls outside via digium card and POTS line
8:45PM 1 IMAP voicemail storage.
8:24PM 1 Avoid compression with g.729/gsm/etc.
8:06PM 0 Where to find local FXS settings ? [SOLVED]
7:41PM 0 Where to find local FXS settings ?
6:15PM 11 Asterisk doesn't relay remote MOH during hold
4:18PM 2 Ideas for Asterisk load testing, testing trunks etc.
4:07PM 0 Set origin CallerID when forwarding calls to mobile phone
2:14PM 3 no ringtone - just silence/bridging of external calls
11:55AM 5 iphone, skype and asterisk ...
11:13AM 2 The Redirect hangups the call while playing a file
10:29AM 0 Limit on simultaneous manager commands
9:53AM 0 Pickup feature request
6:34AM 0 incoming number information
 
Sunday March 29 2009
TimeRepliesSubject
2:39PM 1 DUNDi broken in asterisk 1.4-svn?
9:42AM 11 h exten no getting run ...
3:25AM 1 callpickup not working
 
Saturday March 28 2009
TimeRepliesSubject
10:03PM 0 oh323 to h323
6:27AM 6 hum noise
 
Friday March 27 2009
TimeRepliesSubject
11:47PM 2 TE122
9:37PM 0 ISDN30 Channels Locking
9:02PM 0 UPDATED: Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.9, and Freeplay MoH Update Released
8:51PM 0 Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.8, and Freeplay MoH Update Released
8:31PM 1 Six steps to better SIP security with Asterisk
5:33PM 1 Strange warning message
4:54PM 0 SIP for Skype Solutions: Hosted v Non-hosted
4:43PM 4 Weird sip problem
4:23PM 3 London DDI test request
3:19PM 4 SIP Diversion header
2:06PM 3 AT&T PRI Install - What is outpulsed?
11:07AM 4 Help: RED alarm on Wildcard TE122 card
10:30AM 6 How to Integrate Neospeech with Asterisk
4:02AM 2 Need help on how to programmatically call an extension & test call state
 
Thursday March 26 2009
TimeRepliesSubject
11:27PM 8 Know who's logged in
10:28PM 16 Need to find small footprint asterisk platform
9:25PM 0 Asterisk 1.6.0.5 no MusicHold REFER
8:41PM 1 Is there a public blacklist of hackers' IPaddresses?
8:35PM 6 out of the box or do it your self?
7:06PM 7 Asterisk multi-cpu
5:51PM 9 PRI dropping #2
5:24PM 0 Voicemail Problem
4:05PM 1 Sisky to connect Skype to Asterisk
3:41PM 9 Provisioning GXP 2000
2:19PM 3 show pri usage
1:29PM 0 nat problem in reinvite..
12:10PM 0 TDMoE in any way related to I-TDM
8:45AM 6 IAX problem through intermediate asterisk box
 
Wednesday March 25 2009
TimeRepliesSubject
11:38PM 3 help - How to send hangup command to call in progress.
11:31PM 0 How to send hangup command to call in progress.
8:50PM 1 Skype TO SIP (Was SIP to Skype)
6:23PM 8 OT: Accountless, free, skinnable, browser based SIP client wanted
5:22PM 1 More on SIP for Skype
4:35PM 10 New CentOS 5 repo: dahdi, asterisk, freepbx RPMs
2:40PM 7 SIP Asterisk Hacked (1.6.0.6)
1:19PM 1 SIPPEER equivalent for users.conf ?
1:04PM 29 ITSP's no longer supporting IAX?
12:55PM 2 ASTCC and a2billing
12:39PM 4 Recording the calls
10:58AM 1 Defining a call
9:38AM 3 Create separate Voice Recording System..
5:50AM 0 openIMSCore + asterisk
1:29AM 0 place T1 calls and ignore/override call progress
12:39AM 0 ${UNIQUEID} variable and queue log issues on 1.4.22
12:25AM 1 DISA
12:10AM 1 predictive dialer
12:06AM 0 A Cisco 7960 question
 
Tuesday March 24 2009
TimeRepliesSubject
10:08PM 1 Inter-Asterisk Using SIP
9:21PM 4 PRI dropping
7:11PM 12 Ebay's SIP for Skype
6:21PM 4 HW-Recommendation: cell/mobile phone, capable of WLAN and SIP ??
5:56PM 0 originate and local channel problem
5:51PM 8 sip.conf outboundproxy
5:35PM 0 Unrecognized prilocaldialplan error when dialing a SIP call to a PRI trunk
4:00PM 0 T1 issue to analog trunk for paging (intercom)
3:10PM 23 SIP trunk with > 250 lines
2:21PM 0 MWI Asterisk+Openser
1:21PM 6 Error in ReceiveFax with T.38 -- Asterisk 1.6.0.7-rc2
1:02PM 0 lagrq
10:52AM 3 Relay Register
8:54AM 0 Billing software for 1.6?
8:25AM 16 gpx 2000 Busy Lamp Field
8:16AM 0 Asterisk Realtime Config and SIP/401 Unauthorize: why?
7:14AM 0 Issue with RDNIS
6:16AM 1 Asterisk Originate Command
1:11AM 27 Is there a public blacklist of hackers' IP addresses?
12:14AM 1 strategy ringall
 
Monday March 23 2009
TimeRepliesSubject
10:44PM 5 usb-phones
8:17PM 2 BACKGROUNDSTATUS not available?
7:42PM 3 Skype for SIP
7:00PM 15 conference and wifi phones
5:35PM 4 Polycoms and BLF
1:33PM 22 Recommended USB Headsets ?
11:01AM 0 sip/iax dialplan extension..
10:57AM 0 Issue with no change of SIP call ID
10:01AM 3 Dial in / dial out
9:01AM 1 Simple UK Extensions example
7:00AM 1 distictive Ringing in SIP
2:40AM 3 field lastms in 1.4.24
 
Sunday March 22 2009
TimeRepliesSubject
8:58PM 3 make script 1.6.0.6 breaks up, need help!
7:22PM 2 Global videoconferencing solution.
7:11PM 4 I need a country, state, city database
5:35PM 1 CID when using WaitExten?
2:12PM 1 Looking for Prepaid Solution
7:39AM 0 Manipulating REGISTER messages
4:09AM 2 Asterisk on iMac G3 Debian5 (powerpc)
 
Saturday March 21 2009
TimeRepliesSubject
11:37PM 6 music-on-hold kicks in and disconnects/interrupt the call
7:49PM 3 H323plus homepage down?
2:18PM 0 OT - CID with Asterisk and Betamax
11:33AM 4 1.6.0-rc3 Build failure: asterisk.h: No such file or directory
8:39AM 7 OpenBTS chat with David A. Burgess
2:04AM 0 Asterisk with encryption
1:36AM 7 1.6.2 beta 1 crash
 
Friday March 20 2009
TimeRepliesSubject
11:07PM 1 Music on Hold doesn't play back for external callers
7:22PM 0 Asterisk 1.6.0.7-rc2, 1.6.1.0-rc3, 1.6.2.0-beta1 & Asterisk-Addons 1.6.0.2-rc1, 1.6.1.0-rc3 Now Available
6:05PM 2 Looking for clues to this error message
5:32PM 3 Queues Announce help request.
5:19PM 0 Asterisk Management Application for windows
3:48PM 0 anyone connection to eoncc
2:49PM 3 ATA recommendation??
2:19PM 3 OpenSIPS on CentOS
2:09PM 7 Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
9:36AM 1 T38 FAX
8:52AM 0 Asterisk Realtime Configuration and 404 Extension not found
8:27AM 1 Max concurrent calls
7:05AM 2 chan_ss7 with ringing, but no voice stream.
2:31AM 7 Special Information Tones
 
Thursday March 19 2009
TimeRepliesSubject
11:34PM 25 "The number you have called has been disconnected or is no longer in service"
9:18PM 2 Overriding Queue Wrapup Time
7:56PM 4 Question regarding call queue and penalty's
7:53PM 0 Can I tell if a call picked up on PSTN extension... for example?
7:28PM 0 Flowroute
7:14PM 9 Hardware suggestions
6:09PM 3 Digium and Sangoma Cards PCI express compatibility
5:23PM 10 Asterisk with SRTP and SIP with TLS
4:23PM 0 Is adding "sip show username" easy ?
3:19PM 6 busy lamp filed
3:08PM 3 (no subject)
2:28PM 5 Script to softly restart Asterisk each midnight to clean locked channels
2:27PM 0 DTMF tones mid conversation
2:05PM 1 Polycom MWI.
11:51AM 4 Asterisk and PBX internal numbers
11:19AM 1 VM_DATE in french?
10:29AM 3 IAX trunktimestamps and AST_CONTROL_SRCUPDATE
10:00AM 0 Extensions not found and 401 Unauthorized in realtime configuration (Long post)
9:46AM 1 PRI QSIG Asterisk - Legacy PBX
9:37AM 3 incoming call problem from pri
8:03AM 0 T1 signaling configuration
5:00AM 1 Asterisk crashed!!!
3:55AM 0 AstLinux 0.6.4 available for upgrade
 
Wednesday March 18 2009
TimeRepliesSubject
11:14PM 0 queued?
10:03PM 0 Recent changes in chan_mobile need testing!
10:02PM 3 Voicemail config help - require password
10:01PM 1 Unable to receive faxes
9:01PM 0 [Fwd: Re: DAHDI or Zaptel doesn't compile against 1.4.24]
5:23PM 4 Global h exten
3:57PM 1 Video phone crashing meetme on asterisk 1.4.
3:46PM 2 Controlling BLF Leds ...
3:34PM 0 Asterisk talking to mysql
2:46PM 0 Asterisk talking to mysql database through odbc
11:56AM 0 Asterisk is not designed for University large scale
11:45AM 0 Callerid charset problems
11:23AM 1 Performance of realtime for millions of SIP user
8:58AM 4 Manager API Originate CDR Problem, all is NO ANSWER
2:55AM 3 Asterisk and G.726 Codec
 
Tuesday March 17 2009
TimeRepliesSubject
6:16PM 5 DAHDI or Zaptel doesn't compile against 1.4.24
2:13PM 0 Kewlstart - Busy signal before battery drop.
1:23PM 4 PBX to gate interface
1:13PM 8 SPA3102 - How to save config in a file
12:43PM 0 DTMF troubles
11:44AM 2 Direct Dial-Out and CDR destination numbers
10:33AM 0 ATA react to phone but unresponsive to fax modem [SOLVED]
9:17AM 0 asterisk now and switchvox
8:49AM 0 Weird issue with outbound calls and MOH
8:21AM 1 mobile centrex solution
7:38AM 1 Looking for a patch cable for my SPA941 Phones
6:00AM 1 Test asterisk from behind my firewall
2:49AM 6 Plastic Water Bottles
12:57AM 2 system sizing
 
Monday March 16 2009
TimeRepliesSubject
11:28PM 1 Bristuff bug or feature ? (Was: Are .call files working with extensions.ael ? bristuff problem)
11:27PM 10 Problem with Verizon Wireless
11:21PM 15 Multi-tenant with receptionist features for managed service
10:42PM 0 Asterisk 1.4.24 Now Available!
10:42PM 0 Uptime for documentation only
10:42PM 16 T1 problem (call using a .call file)
10:24PM 14 Good phone near $125
10:11PM 11 T.38 - Which endpoint shall reINVITE ? caller or callee ?
9:34PM 25 Asterisk is not designed for University with large user base?
6:42PM 1 ATA react to phone but unresponsive to fax modem
6:11PM 0 Contact id protocol problem
6:11PM 2 asterisk and ericsson e1 connection how to??
5:54PM 11 Help Inbound number
5:08PM 1 Could Asterisk be rewriting an incoming invite?
4:56PM 0 SIP audio delay after call transfer?
4:56PM 2 Aastra 9133i programmable buttons (* 4.1.23)
4:28PM 8 Busy on SIP
3:34PM 4 A400P + Intel D201GLY2(A) motherboard?
2:50PM 0 Problems on default Attended Transfer
1:39PM 0 Ignore switch to REVERSED Polarity on channel 1, state 4
12:49PM 1 Transfers on an inter-PBX PRI link
11:30AM 2 ANI with Pickup application
11:09AM 0 SIMPLE
10:51AM 6 url in dial command: how does it work?
10:05AM 3 t38 iax trunk
9:09AM 0 trying asterfax
1:57AM 3 Asterisk 1.6 ReceiveFAX problem
 
Sunday March 15 2009
TimeRepliesSubject
10:28PM 8 428 Loop Detected
10:03PM 6 No hardware timing source found in /proc/dahdi
4:38PM 0 Too many notify events causing Asterisk crash?
12:19PM 0 Dahdi Error
11:30AM 1 Using PRI_CAUSE to change SIP INVITE rejection response code
10:30AM 2 X-Asterisk-HangupCause - how to disable this?
 
Saturday March 14 2009
TimeRepliesSubject
6:46PM 0 E&M signalling
4:46PM 4 getting free Did number for asterisk
4:12PM 1 Polycom BLF with Idle State meetme conference
3:26PM 0 Problem with phantom calls
12:13PM 12 BRI cards; JUNGHANNS AND B410P
10:12AM 6 "automatic call bridging when destination is available" feature
1:04AM 3 TRANSFER EVENT ON QUEUE_LOG
 
Friday March 13 2009
TimeRepliesSubject
10:41PM 1 Realtime dialplan application versus REALTIME dialplan function
10:37PM 2 SendFAX/T.38 question
6:14PM 0 Recording calls and SLA
3:38PM 15 Ast/Hyla/IAX Scalability?
3:00PM 0 VoIP Users Conference today at 12 Noon EDT
2:56PM 12 No reply to our critical packet
2:49PM 1 Asterisk to Ericsson MD110 on E1 with ISDN-USR (not QSIG)?
9:51AM 2 Silence suppression problem with DECT phones and g729 codec
9:45AM 1 AGX Asterisk Addon - Can't find app_fax.c withspandsp-0.0.4
7:06AM 5 Initial silence during call
1:52AM 0 MOH Realtime
 
Thursday March 12 2009
TimeRepliesSubject
9:42PM 4 Serving 120 concurrent calls
9:41PM 1 Queue Realtime agents LOGIN for ami
9:23PM 2 Timeout for Queue
8:34PM 2 SetVar (CDR var) from cli
7:02PM 13 log to cdr each dialpan action, not only one record for each call
6:01PM 1 phone emulator for doing interop testing
4:56PM 13 Trying to get sample applicationmap to work (*1.4)
4:13PM 0 chanspy problems (asterisk 1.6.0.6) - When spying starts, the spied parties can't hear each other
3:55PM 7 an easy way to deal with/without leading "1" ?
3:14PM 1 Outgoing call drops
2:11PM 4 Is it possible to get full callin number fromE1?
1:34PM 4 BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai
12:40PM 5 Is it possible to get full callin number from E1?
10:26AM 4 Printing faxes
10:21AM 13 UK ISDN-30 and ANI
10:05AM 0 Manager API and astmanproxy
9:20AM 3 ATCom Phones - AT 510/AT530
7:45AM 0 (no subject)
5:21AM 0 compile error: implicit declaration of function drv_dbg
3:50AM 0 recording (mixmonitor) stopped of transfer/call parking after queue
 
Wednesday March 11 2009
TimeRepliesSubject
11:29PM 8 Are .call files working with extensions.ael ?
9:06PM 4 Grandstream speakerphone?
7:02PM 4 Digium B410P: misdn v1 or misdn v2 or dahdi + asterisk 1.6 ?
6:28PM 2 update on Odd occurrence
4:38PM 3 Problem with incoming and outgoing calls via TDM
2:27PM 3 Multiple Agent Login
10:21AM 2 how to configure for incoming message-summary SUBSCRIBE
10:03AM 0 SIP keep-alive with CRLF?
8:56AM 4 VLC
8:09AM 0 How to read installed spandsp version ?
6:16AM 0 AGX Asterisk Addon - Can't find app_fax.c with spandsp-0.0.4
 
Tuesday March 10 2009
TimeRepliesSubject
11:37PM 0 MacroExclusive crashed asterisk
8:31PM 8 chan_zap.so missing
6:42PM 0 Does the Asterisk/Digium support REGISTER requests with "Authorization" header field sent by a UAC before receiving 401 Unauthorized
6:32PM 1 Phone Directories/Asterisk/SIP/directory.html
6:21PM 2 1.6.x differences
6:02PM 2 Odd occurrence
5:51PM 0 Asterisk 1.6.1.0-rc2 Now Available
5:50PM 1 Asterisk 1.6.0.7-rc1 Now Available
5:50PM 0 Asterisk 1.4.24-rc1 Now Available
5:38PM 0 AST-2009-002: Remote Crash Vulnerability in SIP channel driver
3:24PM 2 1.4.23 + Realtime Queues/Agents NOT via SIP
2:27PM 14 Sending faxes with T.38 problem. Asterisk - 1.6.0.6
2:26PM 2 Compiling OSLEC in Dahdi w/ Intel Optimizations
2:20PM 10 configuring channels for dahdi
2:12PM 0 Voxli
12:51PM 4 Update chan_dahdi.conf doc in voip-info.org
11:05AM 1 Asterisk 1.6, B410P and TE/PtmP mode. Could you get it running ?
11:04AM 3 in which asterisk version is zaptel removed?
10:52AM 3 How to do Load-Balancing for Asterisk with OpenSIPS
10:40AM 6 Asterisk and WebIntegration
10:13AM 1 Calling id problem on outgoing call
7:24AM 0 Hold/Resume issue with polycom
 
Monday March 9 2009
TimeRepliesSubject
11:57PM 1 how to write svn for dahdi-linux and dahdi-tools when using svn 1.4
7:46PM 2 Job in Atlanta.
7:17PM 1 1.6.0.5 - g729 'locked' by Asterisk
7:01PM 10 MoH - always starting from the beginning
6:55PM 0 Announcements
6:41PM 0 Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4 [SOLVED]
6:33PM 0 How to install spandsp from source in lenny ? [SOLVED]
6:03PM 0 asterisk-users Digest, Vol 56, Issue 23
5:36PM 5 DAHDI and B410P (BRI)
4:13PM 1 How to install spandsp from source in lenny ?
1:37PM 0 Crash when reloading AEL
1:30PM 10 problem with an agi in PHP
12:18PM 9 Portech MV3770 & Caller-ID
12:03PM 3 I can't receive fax
11:48AM 0 macro on ring group
10:23AM 0 SIP warnings (401)
9:49AM 0 SIP call hangs up after 20 seconds
 
Sunday March 8 2009
TimeRepliesSubject
9:49PM 3 IAX peer cannot register in Asterisk 1.2.31
4:11PM 15 Fwd: add a new queue strategy: SBR
1:11PM 3 Simple Meetme Question
12:31PM 10 Server Setup Advice
 
Saturday March 7 2009
TimeRepliesSubject
11:26PM 27 Compile problems
4:07PM 0 Busy Here
3:27PM 0 QUEUE_MEMBER_COUNT: Bug or functionality ?
2:29PM 0 Join today Asterisk VOIP meeting online 10A-6P PST = 6P-2A GMT Sat Mar 7 BerkeleyTIP
5:24AM 8 Cdr problem
 
Friday March 6 2009
TimeRepliesSubject
8:40PM 1 question about ringinuse
7:57PM 2 Wideband (G722) MeetMe
7:45PM 2 Parked Calls in 1.4.23.1
7:41PM 0 Early Media before 200 ok
7:29PM 10 IAX based war dialer
6:26PM 2 GoSub & Queue
5:10PM 1 Asterisk and sip router integration
3:58PM 12 work around the 64 pickupgroups limit
2:41PM 2 Aastra 480i repair?
2:10PM 7 SIP *8 Pickup Problem
2:02PM 1 call pickup and ring groups
12:36PM 1 Making use of SIP "making progress" messages
11:41AM 0 Queue moh problem with 1.4.23.1
8:59AM 9 colorized logfiles in asterisk 1.6.0.6
8:58AM 1 Dial command with "r" parameter - no ring tone
7:45AM 3 question about MeetMe performance.
3:22AM 8 How to verify availability of the DID connection?
2:52AM 1 Asterisk dial plan conditional on not busy
1:04AM 1 Fax detection on SIP channel
 
Thursday March 5 2009
TimeRepliesSubject
11:49PM 0 It took some time...
7:15PM 0 T.38 Problem
5:18PM 0 Invite somebody to a conf call
4:00PM 1 Snom Aler-info Ringtone
3:10PM 0 oslec using sample.c for long(er) dumps
3:04PM 0 asterisk and simple chat protocol
1:10PM 2 use more then one sip-provider to dial out
12:39PM 0 Asterisk 1.6.x and auto-provisioning - Polycom
12:34PM 0 Recognizing the "making progress" notification
12:03PM 0 ael conf and realtime
9:42AM 2 Asterisk Differences
7:42AM 5 Asterisk 1.6.1-rc1 with OpenAIS and different subnets
3:05AM 0 Stun with hosted asterisk solution???
2:32AM 3 Cisco IP Communicator with Asterisk/Trixbox
 
Wednesday March 4 2009
TimeRepliesSubject
11:09PM 0 Asterisk 1.6.0.6 sip doesn't work?
5:42PM 10 Outlook integration?
5:34PM 2 SIP attacks
4:06PM 17 Bounty- CDR Bug Fix
3:53PM 0 Asterisk @ Global FreeSW Meeting March 7 Sat BerkeleyTIP -Global - For Forwarding
3:38PM 1 What's the use of sip.conf's notifyringing ?
3:02PM 2 Required:Asterisk Beep tone while call connects
12:22PM 5 Wideband g711-HD vs. g711.1?
12:17PM 1 htable question]
10:44AM 0 Access sip.conf's mailbox from dialplan ? [SOLVED]
9:05AM 0 Master.csv - disposition value (based on?)
8:38AM 1 Question on phone line "pass through"
6:37AM 2 Druid 2.0 released from the Druid Open Source Unified Communications Project
6:25AM 13 faxing via linksys SPA3102 half page goes through
4:51AM 11 Silk for Free
12:20AM 7 AEL2: If-then-else not permitted in Switch-Case
12:12AM 16 $20 Bounty
 
Tuesday March 3 2009
TimeRepliesSubject
11:30PM 1 Predefined viables
8:05PM 0 Unable to create channel
7:09PM 0 monitoring a channel and redirect to conf
6:39PM 2 Asterisk analog DID with Adit 600
3:28PM 7 macro-stdexten question
3:03PM 1 Remote Connection to Asterisk
2:11PM 0 patlooptest and TE121P
1:08PM 0 NOTIFY/SUBSCRIBE and MWI in 1.4
12:24PM 2 CDR
12:19PM 5 Configuring asterisk to revert call back to forwarder if exten is busy
11:59AM 1 Master.csv missing dstchannels values
11:39AM 0 cdr database
10:40AM 6 after install the zaptel but the rtp failed
9:25AM 1 tons of open SIP channel between two snom 360
9:24AM 2 Access sip.conf's mailbox from dialplan ?
3:38AM 0 Blind transfer from asterisk dialplan (and problems re-parking a call)
 
Monday March 2 2009
TimeRepliesSubject
11:34PM 0 Retrieve DTMF during Dial
11:13PM 4 Dialing with cli
11:09PM 1 Asterisk Dial plan issue
10:03PM 0 Queue log on MySQL realtime
9:29PM 1 Will A2B worki with asterisk as b2bua?
8:31PM 2 Asterisk realtime
8:21PM 3 How to set PRI line timeout value
7:46PM 0 H323 Call Variables
6:30PM 0 Disabling 180 Messages
2:58PM 13 How to generate core dump?
2:48PM 3 Compiling to use IMAP: how?
12:40PM 4 SIP dialog matching problem? (1.4.23.1)
10:39AM 5 pci cards VS patton
10:12AM 3 early dial (or overlap dial) and Asterisk 1.2 vs. 1.4
9:27AM 0 how to install the app_meetme2.so and use the web interface
6:27AM 2 Weird segfault
 
Sunday March 1 2009
TimeRepliesSubject
6:04PM 2 Help T.38
4:02PM 0 Pharmacy Message 25749
2:37PM 1 php agi and get_data errors.
2:56AM 1 Called number as variable - how to?