asterisk users - Dec 2006

Sunday December 31 2006
TimeRepliesSubject
10:08PM 2 Dual Ringing Tones
3:38PM 1 PRI ANI/CallerID
3:07PM 0 Digest of lists on forum asterisk.voicemeup.com
3:00PM 0 PHP C Extension to connect to manager interface
12:12PM 0 IAX timeout if no ringing
10:46AM 9 (OT) Where to post free source for AGI?
8:07AM 0 exec after recording agents
7:59AM 12 Sangoma A102d and Asterisk on Debian 3.1.
6:21AM 0 IAX & WaitExten
12:05AM 1 X100P "rings" randomly when "phone" line makes call
 
Saturday December 30 2006
TimeRepliesSubject
7:34PM 0 Server to Server connectivity on SIP
6:21PM 0 Modem Dial-up internet connection thru Asterisk with T1-PRI
5:22PM 0 Theory behind RDNIS and does it work or not?
4:29PM 0 Best WIFI Phone- Zulty WIP 2 Link
1:20PM 12 Happy 2007!!!
12:58PM 13 WIFI SIP- The Best phone
5:43AM 1 Odd hangup problem TDM400P
 
Friday December 29 2006
TimeRepliesSubject
10:19PM 1 E1 controller
3:21PM 2 Re: Hi reg. 2 asterisk server
3:07PM 1 Need advice on dual core processing with *
2:53PM 4 Avaya to Asterisk via H323
2:26PM 0 Dial - g option
12:22PM 1 trixbox web-administration
11:10AM 0 Toll free numbers
10:04AM 0 How does Sipura route incoming calls?
9:19AM 2 chan_sip loading delay in Asterisk 1.2.10
9:02AM 5 Binary AGI Scripts
8:45AM 0 PHP to call script
8:34AM 0 problem with VoiceMailMain
7:45AM 18 Asterisk and MiniITX setups
7:38AM 4 Disconnect supervision in India?
7:29AM 5 Realtime multiple registration for a Hard Phone Snom 360
6:58AM 2 Presence issues with "Got SUBSCRIBE for extensions without hint. Please add hint to s"
4:03AM 6 Dialed Number missing from the CDR when using call files.
1:24AM 2 asterisk doesn't know version of asterisk-addons?
 
Thursday December 28 2006
TimeRepliesSubject
11:49PM 1 voicemail and ip phones
8:14PM 0 Re: asterisk-users Digest, Vol 29, Issue 114
7:29PM 5 TE110P with Qsig
6:07PM 3 Error compiling chan_vpb
5:24PM 0 Compiling Zaptel 1.4.0 on SuSE 10.0
4:27PM 10 1.4 Random disconnects
3:43PM 1 mIDN question
2:30PM 3 one way rtp stream (Sent alwax to 127.0.0.1)
2:19PM 17 [OT] Wifi SIP phones - LinkSys WIP330
1:58PM 1 1.4 - G729 - Have License - No path to translate from Zap to IAX2
1:04PM 13 vzaphfc?
12:57PM 1 Music On Hold Between Servers
11:34AM 2 FW: cdr_addon_mysql.so did not register itselfduringload
11:02AM 0 cdr_addon_mysql.so did not register itselfduringload
8:18AM 2 Background switch to different context
8:16AM 2 FW: cdr_addon_mysql.so did not register itself duringload
6:48AM 4 Checking voicemail from outside
6:32AM 1 Re: Asterisk Queues
1:50AM 0 res_perl with asterisk 1.4 compile problem
12:34AM 0 Say who is using the PSTN?
 
Wednesday December 27 2006
TimeRepliesSubject
11:09PM 9 How to connect two asterisk server
6:22PM 2 Toll-Free number in India
1:08PM 0 cdr_addon_mysql.so did not register itself duringload
11:44AM 10 1.4.0, IMAP and Dovecot
11:30AM 3 Is ZTDUMMY still required with Asterisk 1.4?
11:19AM 1 Asterisk 1.4 Warnings
11:10AM 0 comand stun (for what) asterisk 1.4
10:05AM 7 Searching the list
8:45AM 2 Verbose and sip invitestate logging question (1.4 release)
8:05AM 3 Polycom 601 Contacts List
7:42AM 0 problem with extentions
6:38AM 9 php agi trixbox help
2:30AM 0 AGI Dial channel status
 
Tuesday December 26 2006
TimeRepliesSubject
11:47PM 4 Agent presence
6:06PM 0 1.4 with a nortel call server 1000 running SIP(sdp headers)
4:44PM 3 I cant install zaptel drivers in suse 10.1
1:53PM 1 Questions about 1.4
1:50PM 0 1.4 with a nortel call server 1000 running SIP (sdp headers)
12:51PM 2 Number forwarding and porting?
12:49PM 0 (no subject)
12:13PM 2 agi+cepstral driving me nuts
11:40AM 2 Asterisk 1.4 missing sound in Spanish
11:32AM 6 Asterisk 1.4.0 (release) and G.729
11:07AM 1 How to limit the duration of the MeetMe conversation?
10:53AM 0 controlled playback for MP3
10:27AM 6 SIP Subscription Bug?
10:14AM 1 cdr_addon_mysql.so did not register itself during load
9:36AM 1 flight and the agi
5:15AM 4 1.4 and unicall
 
Monday December 25 2006
TimeRepliesSubject
7:09PM 4 Question about MWI in Asterisk 1.4.0
4:41PM 5 Asterisk 1.4 - no PRI and no Zap?
10:40AM 1 Asterisk 1.4.0 Released!
4:33AM 0 Siedle DCA612-0 and Asterisk : problem with DTMF
 
Sunday December 24 2006
TimeRepliesSubject
9:45AM 0 asterisk with ldap
7:03AM 5 Escalate Call To Mobile
1:35AM 1 Voicemail hangup by gateway?
 
Saturday December 23 2006
TimeRepliesSubject
8:37PM 2 asterisk + door opener
8:36PM 0 centos4.4 x86_64 and zaptel-1.2.12 compile problems?
5:50PM 2 mySQL and to many connections with SQL statement UPDATE
11:38AM 2 CLI Errors and warnings
7:16AM 2 conditional dialplan
3:02AM 1 SNOM 200 behind NAT and other xmas woes
 
Friday December 22 2006
TimeRepliesSubject
11:46PM 4 UPDATE - Analog Phones with FSK/Stutter MWI
10:33PM 0 VXML in Asterisk HELP!
9:23PM 5 Happy X-mas
6:51PM 20 How accurate is show translation?
2:05PM 1 New astGUIclient VICIDIAL Release: 2.0.2
12:53PM 5 Determining invalid extensions.
12:51PM 1 sangomo
12:35PM 0 RESOLVED: Sangoma Wanpipe 2.3.4-3 compilatio n fails un der FC2 with Zaptel 1.0.9.2
12:24PM 2 spa300 password recovery
12:21PM 1 Answering Machine Detect (AMD) time values
10:09AM 0 Sangoma Wanpipe 2.3.4-3 compilation fails un der FC2 with Zaptel 1.0.9.2
10:03AM 4 meetmejoin example
9:58AM 4 Sangoma Wanpipe 2.3.4-3 compilation fails under FC2 with Zaptel 1.0.9.2
7:39AM 0 RE: RTPTIMEOUT Configuration
6:19AM 5 problems using the 1.4 version of meetme
5:49AM 2 System Application with java
5:17AM 1 Problem for trunk to trunk communication
 
Thursday December 21 2006
TimeRepliesSubject
11:56PM 0 question about astdb
9:09PM 2 Connect many fax lines?
8:40PM 3 Help with SUSE 10.2 and Sangoma A104D
3:23PM 10 International dialplans for Asterisk?
3:03PM 1 IAX calls not ringing
2:30PM 3 Grandstream GXW-4108 8 port FXO
12:57PM 0 GXP-2000 and Asterisk Configuration
12:23PM 1 Re: Match a Numer - then continue with, dialplan
12:09PM 2 asterisk crashed
11:56AM 4 more than 32 callgroups & pickupgroups
7:57AM 2 AELPARSE - Wish/Suggestion
7:41AM 4 Insert 1+areacode for VOIP calls
1:31AM 0 When line in use busy signal?!
12:27AM 0 The parameter of ast_request_and_dial()
 
Wednesday December 20 2006
TimeRepliesSubject
10:28PM 3 Calls disconnected after 1 hour
10:12PM 0 Re: Match a Numer - then continue with, dialplan
7:59PM 3 clear ast database
7:35PM 5 Sangoma A101 with Unicall
5:14PM 0 Re: Match a Numer - then continue with, dialplan
3:48PM 0 RE: spandsp 0.0.3 RxFax fax =?ISO-8859-1?Q?_reception crashes bristuffed_asterisk_1=2E2=2E13_[?= Virusgeprüft]
2:54PM 3 Re: Match a Numer - then continue with, dialplan
2:54PM 6 Agentcallbacklogin deprecation
1:47PM 0 Call Routing
12:34PM 2 Dial own extension to get to voicemail.
11:50AM 5 Re: Match a Numer - then continue with, dialplan
11:23AM 2 Asterisk Now
10:27AM 0 Can't make outgoing calls (T100P)
10:26AM 0 Re: Match a Numer - then continue with, dialplan
10:09AM 1 Incoming Lines Confusion
10:07AM 5 RE: spandsp 0.0.3 RxFax fax =?ISO-8859-1?Q?_reception crashes bristuffed_asterisk_1=2E2=2E13_[?= Virusgeprüft]
9:00AM 4 question about sip account format
7:55AM 2 Dial 9 to get out?
6:39AM 4 AgentCallbackLogin() deprecated in 1.4
6:18AM 62 Need quality toll free 800 number over IAX?
2:37AM 0 asterisk run on vxworks for hardware pbx
1:22AM 6 IAX connection to FWD not working
1:10AM 0 better handling of calls forwarded by SIP phones
12:29AM 0 MATH function
 
Tuesday December 19 2006
TimeRepliesSubject
10:17PM 0 Need a 630 DID
10:02PM 1 Need Wholesale Termination
9:00PM 0 SIP and ZAP
5:02PM 6 Echo problem
4:55PM 0 Using Asterisk/Digium card with Tadiran switch
4:38PM 5 AstManProxy - Manager
4:11PM 0 Cisco devices (without STUN) and dynamic NAT
3:54PM 1 Is logic right?
3:36PM 1 SPAM-LOW: Re: .Call files do not seem to wo rk
3:02PM 1 T1 Pri Question
2:49PM 16 No music on hold?
2:45PM 48 Match a Numer - then continue with dialplan
12:56PM 1 SPAM-LOW: Re: .Call files do not seem to wo rk
10:27AM 0 Is MOH Still Broken in Asterisk 1.4 (beta3)?
9:10AM 0 db.c: Unable to open Asterisk database
8:38AM 6 DTMF Tones "A-B-C-D"
8:35AM 3 Parsing Area Code from CallerID
8:34AM 7 Polycom ring backs and CID
8:17AM 5 .Call files do not seem to work
8:03AM 0 Automatic sip conference
7:35AM 1 G.279 license question
7:04AM 0 dtmf and ivr
4:22AM 2 Distinctive Ring detection and caller ID
4:22AM 1 Asterisk-LDAP Integration?
3:22AM 2 Re: asterisk-users Digest, Vol 29, Issue 71
2:54AM 12 2 devices using same sip account
2:20AM 2 AEL2 on Asterisk 1.2.4
2:15AM 0 features.conf problems
12:54AM 1 [Fwd: Re: spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13]
 
Monday December 18 2006
TimeRepliesSubject
11:58PM 3 Remote Reboot of a Polycom
11:15PM 8 AGI Help Please
11:11PM 3 Changing CALLERIDNUM on the fly
9:09PM 6 Follow-me challenge
8:54PM 0 Cisco 7914 with sccp
8:24PM 0 openwrt wrt54gs running asterisk/pap2
8:10PM 0 HITBSecConf2007 - Dubai - Call for Papers now open!
8:01PM 6 Inform callers on recorded/monitored number.
6:34PM 8 Billing solution
6:27PM 1 RE: Best way to access MySQL data from dial plan
5:21PM 1 MWI, Realtime SIP, Voicemail and Extensions, UAs registered with SER
4:41PM 3 Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
3:38PM 0 Re: Best way to access MySQL data from dial plan
2:25PM 0 pap2/wrt54gs/asterisk
2:22PM 5 ZAP problem
12:41PM 0 Re: Best way to access MySQL data from dial plan
11:55AM 1 stop logging certain error messages
11:54AM 3 GXP2000, Linksys RV082 Firewall / NAT, Registrations
11:47AM 0 Colomachine & TE405P
11:33AM 1 Re: Best way to access MySQL data from dial plan
10:35AM 1 Voicemail delivery
9:06AM 1 Cisco 7940 - NAT Option
8:15AM 5 Asterisk and outlook
7:52AM 0 Wait command
6:23AM 2 asterisk to asterisk - to zap
6:13AM 3 Shared Line Appearances (SLA) in 1.4
6:07AM 0 calls interrupted by music on hold
4:52AM 4 Digium TE405P with French E1 => Red Alert
4:13AM 11 Thomson ST2030S and BLF
3:16AM 4 Asterisk + Orion E1 GSM Gateway
1:18AM 13 spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
12:19AM 0 zap sending fax congested
 
Sunday December 17 2006
TimeRepliesSubject
9:16PM 0 Davox
8:05PM 3 CLI output to file
5:59PM 3 sip peer name channel variable?
5:35PM 1 bridging isdn calls to free up channels
4:49PM 7 BLF on GXP2000
3:31PM 3 Dial 9 For Outside Line?
1:24PM 1 1.4 sounds long space before and after prompt
12:51PM 1 GXP2000 and BLF
10:07AM 8 What web interfaces are available today for debian based Asterisk installation?
6:33AM 9 Day/night service and indications on the phone
3:52AM 1 AGI and php simple example
2:53AM 3 call forward
2:44AM 0 Learning Asterisk Internals
 
Saturday December 16 2006
TimeRepliesSubject
9:44PM 0 PRI debugging outgoing not working, help needed
1:04PM 4 Asterisk 1.4.0 B4 Sounds Directory
11:57AM 2 rxfax detection problems with multiple contexts
9:54AM 0 SIP call not fail over immediately when Server Off Line
9:38AM 1 Asterisk 1.4.0-beta4 compile errors on Fedora 6
6:14AM 10 Linux distro + Asterisk or Trixbox?
4:11AM 0 Asterisk 1.4.0b4 installation
 
Friday December 15 2006
TimeRepliesSubject
8:52PM 1 Asterisk 1.4.0-beta4 Released
8:52PM 0 Asterisk 1.2.14 Released
8:52PM 0 Zaptel 1.4.0-beta3 Released
8:52PM 0 Zaptel 1.2.12 Released
7:44PM 4 Bandwidth requirements for 1, 000, 000 minutes a month
7:41PM 1 ztmonitor displays full bar when idle
7:32PM 3 Boot load wcfxo does not configure self under Ubuntu 6
6:15PM 1 Sipura question
6:07PM 1 International Provider
4:58PM 3 iax2 softphone attended transfers
4:28PM 2 Motherboard 3.3V PCI for TE412P
3:28PM 0 dialing via SIP URI
2:55PM 2 MOH Between Asterisk Servers
2:23PM 9 Good Commercial Grade Service Provider?
12:36PM 1 DTMF Tone Issues
12:22PM 12 Iptables rule help
12:14PM 4 Fast Busy Followup
12:08PM 1 What's up with DATETIME and TIMESTAMP in Asterisk 1.4beta3 ?
11:10AM 1 zapata.conf channel variable question
11:00AM 6 Sip port= not working
10:35AM 3 Trying to forward calls by using the Callee's context as the forward dial context
10:33AM 0 SIP DTMF not acted on for features in 1.4.0b3
10:26AM 0 Cisco Call Manager 4.0 to Asterisk, Anyone haveSIP Reinvite working?
9:26AM 0 Hardware TDM Switching (Out Of Office - on vacation)
7:33AM 1 anyone using metermaid / parked call BLF?
7:11AM 0 AEL: CID match and pattern in switch statement
6:54AM 0 100rel & Prack enable
6:51AM 1 Cisco Call Manager 4.0 to Asterisk, Anyone have SIP Reinvite working?
6:35AM 1 Attended Transfer on queue_log
6:32AM 2 Hardware TDM Switching
6:13AM 2 is it possible to use Asterisk voicemail as anouncement system only?
5:51AM 8 call from h323 to SIP
4:36AM 2 How to know who hangup ?
2:18AM 1 Page + ParkAndAnnounce
2:09AM 2 enum
12:32AM 2 fxotune unable to set impedence
 
Thursday December 14 2006
TimeRepliesSubject
11:52PM 4 FYI Panasonic Wireless Phone MWI
10:23PM 2 Bandwidth.com on asterisk
7:21PM 0 bridging calls on a samsung pbx from asterisk
3:38PM 1 VoipTalk unable to accept calls at present?
2:55PM 3 StripXXX apps missing from asterisk-1.2.13?
2:50PM 3 Fast Busy
2:47PM 4 Console latency
2:32PM 4 Voicemail Live
1:30PM 2 On-Hold
1:24PM 2 Broadvoice registration problems
11:04AM 2 Show agent queue status on the phone?
11:01AM 0 Web-MeetMe ready for prime time?
9:13AM 18 IBM Server / USB Ports
6:32AM 3 Ssh access over a zap channel...
6:29AM 1 agi scripts running slowly
6:23AM 4 (no subject)
6:05AM 6 Zaptel under FC6
3:35AM 3 AOC-D or similar
3:34AM 0 WRAP+astlinux g729
 
Wednesday December 13 2006
TimeRepliesSubject
8:30PM 7 TDM400P won't ring GM phone of mere 0.1B
7:44PM 4 PRI to SIP
7:30PM 1 Phone routing - curious what others are doing?
7:16PM 0 Asterisk, Bluetooth, and wireless phone
5:34PM 5 Asterisk to a Huawei softX3000 problem has already been solved ?
5:32PM 2 Searchable Archives of this list
3:39PM 0 Asterisk Community lost a valuable contributor today
2:27PM 0 Re: Core Dump: create_transaction (p=0x0) atpbx_dundi.c:2787
2:06PM 0 ZAP multiline handset questions
2:00PM 2 ssh access using zaptel channel to dial in.
1:20PM 0 webvoicemail
1:01PM 1 Polycom IP4000 and vsftpd 2.0.1
12:18PM 3 Playing a sound file on handset pickup
11:18AM 8 how to define a secure trunk
11:04AM 0 Remember last IP address of IAX client
11:03AM 1 SRV Entries
10:29AM 0 Help with voicemail
10:28AM 1 record time with phones option buttons
10:20AM 1 Core Dump: create_transaction (p=0x0) at pbx_dundi.c:2787
10:09AM 3 Diva Server V-BRI-2 and internal numbers
10:08AM 0 Remote-Party-ID and CallerID
9:56AM 0 FW: New Software available on Cisco.com P0S3-08-5-00
9:55AM 1 Pickup application
9:40AM 3 Realtime +Mysql +Failover
9:34AM 2 MFC/R2 on chan_zap
9:29AM 0 Annoying echo echo problem problem ...
8:33AM 12 Polycom MyStat
8:15AM 7 MixMonitor and Queues
8:04AM 1 Audiocodes MediaPack MP-118
8:02AM 4 CallerID Issue (asterisk newbie)
7:55AM 4 anyone used vitelity?
7:22AM 0 FW: MeetMe Conferencing and Marked Mode
7:17AM 8 Question about hardware
6:55AM 3 Stress test
6:30AM 1 IAX trunk problem
5:53AM 4 send fax by Iaxmodem ?
5:23AM 3 Multi Operator
5:14AM 3 How to temporarily unload modules.
3:39AM 0 TDM04B and shared IRQ ..but asterisk can work..
2:45AM 1 Problem with asterisk 1.4 Installation (undefined reference to `ast_copy_string')
1:05AM 0 Asterisk and spandsp 0.3
12:07AM 1 Hardware Suggestion for 2 PRI with call recording
 
Tuesday December 12 2006
TimeRepliesSubject
8:13PM 22 Need help getting started with asterisk
7:23PM 5 caller ID authentication
7:00PM 5 Measuring VoIP latency and packet loss
4:03PM 1 Settings CallerId for outgoing calls based on the sip account making them
3:08PM 0 [BULK] Asterisk manager
3:03PM 6 Conference between skinny user and many sip user
2:32PM 0 Disregistering Constantly - message: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181000' is now UNREACHABLE! Last qualify: 0
1:59PM 0 ASTCC and DTMF
1:27PM 7 Cisco 7970 + New Firmware (8.2)
12:08PM 8 MeetMe Conferencing and Marked Mode
12:04PM 5 sip help for newbie
10:48AM 2 zapata.conf: cannot set txgain lower than -6.3 ?
10:18AM 4 long busy()
10:04AM 0 Auto answer when already on a call
9:26AM 7 Asterisk Manager
9:04AM 3 AGI problema
9:00AM 11 Input on Dundi
8:21AM 1 Anyone using Ranch Networks products for Load Balancing in a SIP environment?
8:20AM 1 SIP and IAX configuration from LDAP
8:14AM 5 outgoing call on ISDN PRI
7:57AM 7 repost gain problem with asterisk and zaptel 1.4
6:28AM 2 Hangup Party
5:53AM 2 SPA2100 sends an unexpected BYE message when transmitting a FAX
2:08AM 0 Voicemail App
1:48AM 1 func_curl fails to compile, asterisk1.4
 
Monday December 11 2006
TimeRepliesSubject
11:45PM 0 Asterisk 1.2.12 + fedora core 5 and TDM04B IRQ issues
10:11PM 0 Sip communicator issue
9:31PM 7 Problem in making outbound calls in PRI
8:55PM 0 Asterisk Sends 180-RINGINGto UAeven withprogressinband=yes
6:38PM 1 Asterisk Sends 180-RINGING to UAeven withprogressinband=yes
4:58PM 2 Asterisk Sends 180-RINGING to UA even withprogressinband=yes
4:57PM 3 How to add include statement into Realtime static
4:23PM 1 Asterisk Sends 180-RINGING to UA even with progressinband=yes
3:27PM 8 VPN As SIP Tunneling?
3:22PM 3 Using SIP with NAT (technical code question)
2:18PM 2 Unable to open pseudo channel for timing... Sound may be choppy.
2:02PM 0 Aculab
1:52PM 1 Extending Avaya IP Office ISDN30e with Asterisk
1:52PM 0 zaptel and zapata configuration
1:19PM 1 re: L option in dial command
12:35PM 3 IAX2 to SIP protocol translation overhead?
10:39AM 0 FW: [asterisk-dev] Kernel crash during modprobe wfxco
10:32AM 19 CLI History
10:30AM 2 asterisk PLAR
10:26AM 0 How to manipulate FROM header on Asterisk-DIALPLAN
8:43AM 0 Recall: Re: Recommendations for QoS, PoE Switches
7:32AM 0 OSP peering VOIP servers
7:05AM 0 Asterisk + Zap + CAS Signalling
6:14AM 2 Power requirements on the TDM-400 card
6:01AM 3 Asterisk and Fax How To
5:55AM 0 promotional info in music on hold
5:41AM 0 Cannot find ptlib-config, installing 1.4-beta3
3:12AM 3 Waiting for dial tone in Dial cmd
3:10AM 0 OPS Protocol on Asterisk
2:30AM 3 Asterisk with IM
12:19AM 1 New installation CentOS 4 x86 or X86_64
 
Sunday December 10 2006
TimeRepliesSubject
10:17PM 6 X100P clone dial problems.
10:08PM 0 : Some warnings occur
9:26PM 2 Mediatrix 1124 setup
8:09PM 0 Wifi Phone with Multiple Line Appearances
7:05PM 2 popups, queue & agents
7:00PM 0 gain problems with zaptel 1.4 beta 2
6:51PM 5 Xen, Asterisk & ISDN: Timing Problems
4:16PM 0 tx_fax
3:02PM 1 Problem faxing with SPA2100 in passthru mode.
2:29PM 11 TDM2400
2:10PM 2 Display variables
9:52AM 10 Asterisk from Debian Packages
8:02AM 1 chan_sip.c:5267 sip_reg_timeout Error
5:13AM 4 Asterisk 1.4b3 & Realtime Voicemail
2:11AM 4 NAT and Dial to two channels at once
12:10AM 18 Recommendations for QoS, PoE Switches
 
Saturday December 9 2006
TimeRepliesSubject
9:36PM 1 Anonymous clid ?
8:45PM 1 Jabber Client
1:53PM 7 PCI, PCI-X and PCI-e -- Server / Interface Card Selection
12:54PM 0 Quicknet PhoneJack questions.
10:07AM 5 Zaptel module compile woes
2:15AM 2 RDNIS question
 
Friday December 8 2006
TimeRepliesSubject
9:55PM 1 Asterisk voice recording through TE110p
8:37PM 1 using a mobile phone as a handset via bluetooth
7:29PM 0 trixbox
5:10PM 1 SIP Quality Metrics
4:27PM 1 Polycom soft buttons not working
2:33PM 0 SIP/IAX Fax Detect on Asterisk 1.4
2:25PM 0 Best book to learn SIP details ?
12:33PM 3 Asterisk forgetting about client registration or Polycom phone forgetting to register?
12:28PM 3 Repeated Digits
12:17PM 1 Douglas Garstang <dgarstang@oneeighty.com>
11:55AM 0 Dial groups, groups of phones, multiple line keys
11:49AM 0 Asterisk eating the Asterisk key!
11:15AM 3 downloading asterisk GUI
11:02AM 8 Question on retrieve_file() function in app_voicemail.c
10:19AM 0 Verizon VoiceWing support
10:00AM 0 codec_speex.c: Out of buffer space
9:14AM 7 5.8gig phone MWI
8:52AM 3 CTI: put on hold a call
8:46AM 1 cal recording with email
7:25AM 6 Vonage SIP access via asterisk?
7:02AM 2 AGI interaction with php
6:42AM 6 How to communicated Both SIP and IAX2 each other ?
6:01AM 0 RE: Answer a call that is not ringing on yourextension
5:06AM 0 problem with asterisk 1.4
4:00AM 2 Management GUI
3:35AM 6 Server for 100 concurrent calls
 
Thursday December 7 2006
TimeRepliesSubject
9:56PM 0 nordx designator labels ?
9:56PM 2 ASTERISK y AGC
8:50PM 1 Basic question regarding re-INVITE
8:00PM 1 AMI - Originate Action and Busy, NoAnswer calls - CDR
7:56PM 3 wierd callerid problem
5:03PM 8 Asterisk 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking)
3:54PM 0 Audio Convert Module
2:38PM 0 Need help on AgentCallbackLogin()
2:33PM 1 Asterisk 1.4 + Cisco 7970
1:28PM 0 Session Progress Transmission to Phone
1:15PM 4 queue agent Monitor
12:21PM 2 Polycom buddies question
12:15PM 0 Asterisk stopped Matching Defined Peer
11:53AM 3 Codec Selection in asterisk
11:34AM 4 "illegal" VoIP in India
9:51AM 8 CISCO 2600 - VWIC 1MFT-E1
9:50AM 0 UDP ports
9:16AM 9 FXO USB that works with Asterisk?
8:33AM 0 Job Posting, Asterisk Engineer/Sales Engineer, Dallas TX Area
8:19AM 1 queue member refresh
8:19AM 0 Fax machine detect (akin to AMD)
7:59AM 2 Asterisk accepting calls to fast
7:52AM 1 Standardized IVR UI Pattern (was: Re: Is there any Asterisk controllable thermostat?)
7:25AM 13 Running Asterisk on a Home rotuer
7:25AM 8 Plantronics and Snom RF feedback
7:19AM 1 -- Called 12127773456@OOH323 Segmentation fault (core dumped)
6:09AM 4 eicon diva BRI problems
3:34AM 0 sip qualify unreachable/reachable - ci$co 7940
2:56AM 2 oh323.conf question
2:46AM 1 how to configure Asterisk to support SIP "INFO" method?
1:57AM 0 Requested transfer capability: 0x00 - SPEECH - How to change to 31KAUDIO?
1:39AM 0 calls not terminating (2nd posting)
12:32AM 0 ChanSpy * and 1234# not working
 
Wednesday December 6 2006
TimeRepliesSubject
7:09PM 5 MWI across multiple servers
6:59PM 0 Echo problem with TDM440P and ADSL Line
6:16PM 1 0002475: [patch] Allow app_directory to work with REALTIME
4:56PM 1 Error compiling Eicon Diva from source
2:54PM 0 Avoided initial deadlock asterisk v 1.2.12.1 SIP clients IAX2 termination.
2:40PM 1 Govarion.
12:09PM 9 Setting outgoing caller id on a zap channel for one sip extension only
11:48AM 0 (REPOST DUE TO NO ANSWER) translate.c:88 powerof: Powerof 0: No power?? / translate.c:133 ast_translator_build_path: No translator
10:41AM 2 FW: G.726 on Asterisk 1.4.0
10:03AM 1 Detecting no answers and/or disconnected numbers
9:31AM 0 Dec 6 09:47:52 NOTICE[3263]: chan_iax2.c:1619 iax2_destroy: Avoiding IAX destroy deadlock
8:31AM 0 Error in codec string '=audio 5004 RTP/SAVP 3'
7:53AM 3 problem with asterisk - calls where both sidescannot hear each other
7:36AM 1 Same issue, different way to ask.
7:35AM 1 CAS DID 2way
7:09AM 0 iax/sip registering and real-time
7:07AM 0 MWI/realtime/openSer in 1.4
6:46AM 9 Agent autologoff dynamic queue members - Brain aches please help
6:15AM 1 Ping
6:10AM 1 Can not hear called party
5:23AM 2 G.729E
4:36AM 0 Problems with bridging data calls over Wildcard TE405P
2:47AM 3 ParkAndAnnounce + Paging
2:43AM 5 Asterisk freezes when DNS not working: a BUG??
2:35AM 2 problem with asterisk-1.4+sip communicator
12:00AM 0 asterisk -1.4 with sip communicator
 
Tuesday December 5 2006
TimeRepliesSubject
11:31PM 1 for all Asterisk Users
11:05PM 0 Jumpers and DIP switches on Atcom AX-4S and AX-1E
9:26PM 5 Rejecting a Call
9:08PM 1 Problem loading unicall
8:25PM 0 Melbn Asterisk/Voip get together
8:15PM 0 RE: SOLVED - T1 PRI not announce "this is long distance call, please add 1 for this call..."
7:41PM 2 TE110P Out fine / In Fail
6:08PM 2 Auto dialing: .call file vs. manager interface
5:00PM 0 [Fwd: RE: any possibility of Vonage Integration]
4:49PM 1 Need some examples for configuring Asterisk under Realtime static
4:36PM 25 Switching from FreeBSD to Linux - which distro?
4:23PM 1 Question about Realtime static table
3:40PM 1 Meetme monitoring (once)
2:50PM 1 Install via SVN or tarball?
2:41PM 0 Re: regcontext, NoOp extension vanishes when extension reload, WORKING
2:27PM 6 problem with asterisk - calls where both sides cannot hear each other
2:23PM 0 RE: regcontext, NoOp extension vanishes when extension reload
1:49PM 0 RE: regcontext, NoOp extension vanishes when extension reload
1:30PM 2 SIP firmware for Siemens Optipoint 410 Economy?
1:13PM 7 Attended Transfer
12:29PM 0 G.726 on Asterisk 1.4.0
12:14PM 1 Help with dial plan - two attempts at calling agent before logging agent off?
11:50AM 4 regcontext, NoOp extension vanishes when extension reload and doesn't come back
11:47AM 0 Realtime Error 1045
11:31AM 0 SOLVED: DB9 e1 to RJ45 pinout
10:38AM 1 SetCallingPres propagation
10:32AM 0 nvlinedetect
10:14AM 2 Realtime question
9:39AM 2 installed, stumped on sip registration
9:16AM 1 No ID from the calling party in SIP Header
8:56AM 0 Issues
8:56AM 1 SER/OpenSER + Asterisk + Queue
8:42AM 2 Shared Line Appearances
7:40AM 4 question on tx_fax install for asterisk 1.4
6:59AM 1 calls not terminating
6:23AM 0 Diginetwork X100P card
6:17AM 9 centos 4.4 + asterisk
5:37AM 0 Signalling but no media
4:50AM 1 sip_write warning when executing Pickup of CAPI
2:20AM 4 zaptel-1.4.0-beta2 Getting it to compile on Fedora Core 6 _64bit
1:39AM 2 Max T1s in a server?
 
Monday December 4 2006
TimeRepliesSubject
11:44PM 2 How to stop Asterisk to pick up incoming PSTN signal
10:55PM 34 any possibility of Vonage Integration
4:41PM 2 T1 PRI not announce "this is long distance call, please add 1 for this call..."
1:14PM 5 Digium TE407P vs. Sangoma A104d
12:34PM 0 Registering VoIP providers with realtime
12:03PM 2 ASterisk and SER
11:35AM 3 Answer a call that is not ringing on your extension
10:04AM 1 Sangoma a301 or other DS3 card
9:28AM 0 Addqueuemember and roaming users problem.
8:25AM 0 No answer when press 0 for operator in VM in 1.0 .9?
8:08AM 2 Odd queue issue
8:07AM 0 google talk
7:40AM 1 Moderate setup
7:32AM 0 Codec transcoding and call recording
6:53AM 0 mwi for voicemail not showing up for realtimeconfig.
6:05AM 12 forward skinny call to SIP
5:36AM 6 MySQL cmd % pattern matching
5:12AM 2 Problem with h323 support
5:10AM 2 Nokia E60 problems
1:27AM 0 Can zaptel freak out if you configure 2 trunks but use only one?
1:20AM 0 Extend time in call pickup
12:17AM 3 mwi for voicemail not showing up for realtime config.
12:11AM 1 HOW TO - Asterisk apps/modify and compile
 
Sunday December 3 2006
TimeRepliesSubject
11:04PM 5 How can i processed with Call Snooping,
9:04PM 21 Is there any Asterisk controllable thermostat?
7:12PM 3 TDM01B installation
7:03PM 0 Neat Skype Device
5:05PM 2 G729 Passthru?
4:55PM 5 asterisk manager originate command
3:08PM 0 Asterisk : Numbers Guessing Game
3:06PM 0 Asterisk 1.4: SPANDSP3 (WIP) HOWTO
1:49PM 0 VoIP GSM Gateways
1:04PM 1 Realtime fullcontact field contains nat device private ip
12:01PM 0 * key on Linksys SPA-841
8:39AM 0 G729 Liscence
8:39AM 2 RTP Media Path
8:24AM 0 translate.c:88 powerof: Powerof 0: No power?? / translate.c:133 ast_translator_build_path: No translator
7:08AM 0 RNK
 
Saturday December 2 2006
TimeRepliesSubject
10:49PM 0 Answering Machine detection in Australia
3:09PM 0 rxfax or spandsp problems??
12:29PM 2 "Low" beep on voicemail
12:02PM 5 Linksys PAP2t-NA and Asterisk
9:59AM 1 Detailed description of problem in Poland
9:53AM 3 Problem in Poland
9:42AM 0 RINGNOANSWER on 1.2
7:07AM 7 Help with IAX Trunk
 
Friday December 1 2006
TimeRepliesSubject
5:21PM 0 no tx audio
3:40PM 0 video for call attendant systems
3:03PM 3 sip address in voicemail emails
2:33PM 1 Interesting CALLERID behavior
1:29PM 0 Audiocodes MP104-FXO - Transfer the call only after 3 rings
11:50AM 0 setcallerpres not working
11:44AM 3 direct IP calling with extension
10:45AM 5 Recommendation for FXO
10:18AM 0 feel free to add to the bounty for issue 8064
9:40AM 0 App_Swift
9:31AM 0 Problem with agent AgentCallbackLogin()
9:26AM 1 app_sql_postgres gone in 1.4
9:09AM 3 CALL TRANSFER
8:42AM 4 No caller ID, no incoming call
8:41AM 0 server specs / hardware
8:24AM 0 spa3k dtmf problem asterisk 1.2.x
7:26AM 9 Caller ID Rewrite
6:27AM 0 Asterisk as bridge, strange ${EXTEN} values
6:22AM 0 Music on hold
6:06AM 0 ISDN BRI lines engaged when dialing out
5:46AM 2 Cisco IAXmodem HylaFAX
2:26AM 0 seed vs registration?
1:07AM 1 H323 NAT Problem
1:02AM 4 Asterisk: SIP Gateway or Proxy