asterisk users - Aug 2007

Friday August 31 2007
TimeRepliesSubject
11:08PM 2 Problems with Polycom 300/500/600
10:59PM 3 Cisco 7960 sccp
6:51PM 2 Latency, Jitter and Lost packets...
6:46PM 1 Cisco Directory Format
6:01PM 1 AEL missing in recent 1.2 releases?
5:34PM 0 chan_sip.c:5495 sip_reg_timeout: ERROR
4:38PM 1 Strange behaviour on Asterisk 1.4.9 with Queues...
3:55PM 0 Sipp scenario for asterisk sip
2:45PM 0 Question on Asterisk and ISDN
2:37PM 3 VoIP+IM with Asterisk+Jabber
1:18PM 1 sip:EXTEN;phone-context in asterisk dial plan
12:55PM 8 E1 to Ethernet Bridge
12:33PM 4 Cisco 7960 Won'
11:24AM 0 about ChanSpy
11:08AM 0 WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register
9:34AM 6 Shortening Context code
7:36AM 0 question about realtime
1:45AM 4 app_conference
 
Thursday August 30 2007
TimeRepliesSubject
10:38PM 2 Round robin behavior for dialing SIP trunks...
9:10PM 0 Digium Asterisk Appliance reviews?
8:13PM 2 Hierarchical Config file (re)writing (bug 8684)
8:03PM 5 Testing Framework
6:57PM 1 FYI
6:41PM 0 FATAL: Module wcdtm not found
6:19PM 4 Channel banks for E1
5:45PM 0 Canada PRI order -- anybody willing to help?
4:59PM 1 FATAL: Module wcdtm not found.
3:54PM 0 Opinions on AsteriskNOW
3:23PM 0 WARNING[22292]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x82f2fe0', 9 retries!
2:52PM 3 How long to detect an "h" exten?
2:25PM 0 DTMF Question
1:31PM 0 Cannot create Incoming Outgoing call through for r2mfc protocol
12:44PM 7 Rechazo de llamada en triangulacion de asterisk.
11:34AM 28 How to handle "+" prefix
10:19AM 5 dialed peer number
9:15AM 5 asterisk at 100% CPU, 1000's of log files
12:03AM 2 Unknown connection error: (2006) MySQL server has gone away
 
Wednesday August 29 2007
TimeRepliesSubject
11:39PM 0 app_conference and asterisk 1.2.24
11:13PM 0 Hangup detection and trombining
9:02PM 2 Members in 'Unknown' status in output of 'queue show'
8:46PM 0 Cisco 7970G App Development?
7:06PM 50 where is 1.4.12?
6:09PM 0 Trying to use "Set Group" correctly
5:48PM 3 AsteriskNOW and config files
5:25PM 4 Queue Agents on Remote Asterisk server?
4:58PM 0 Asterisk with IM (instant messaging)
4:32PM 2 understanding queues
3:38PM 0 Cisco FXS Issue...
2:46PM 5 Monitor System using AGI Scripts
2:44PM 0 (Asterisk_1.4.0 + rxfax + spandsp_0.0.4) - symbol lookup error
2:18PM 0 can anybody tell me about unicall.conf for incoming and outgoing
1:46PM 2 OT - Callto:// tags options
1:38PM 4 Channel Bank Recommendations
1:31PM 2 Voicemail and fax detect
1:03PM 6 Ringing sound doesn't work
12:12PM 2 Best text-to-speech
11:39AM 0 Email to Voice
11:09AM 1 WARNING[11439]
7:53AM 0 Asterisk + MSN Messenger
7:36AM 0 call pickup problem
2:41AM 2 sip authorization problem
 
Tuesday August 28 2007
TimeRepliesSubject
10:42PM 2 Zaptel causes kernel crash - zt_init_tone_state
9:11PM 6 Voicemail Password Issue
9:11PM 7 Load testing/burn-in for Sangoma quad PRI card
8:11PM 25 Fax Problems with SpanDSP
7:51PM 3 ATrpms/Fritz FCPCI CAPI/Fedora 7
3:51PM 2 E911 mf camma Trunks
3:14PM 11 Distributed System
3:00PM 4 Zaptel 1.4.4 compiling problems
2:53PM 2 server recommentation (unique requirements)
2:13PM 15 G729 Confusion
2:04PM 2 HDL F10 brazilian doorbell device + TDM2400
1:53PM 2 Astricon Meetup
1:24PM 1 calls being forwarded to neighbor?? please help, thx :)
1:02PM 0 Saftware RAID1 or Hardware RAID1 with Asterisk (Andrew Joakimsen)
12:57PM 0 (no subject)
10:37AM 0 Asterisk Manager Interface, response types
10:31AM 0 Asterisk call waiting with SIP
8:17AM 3 deadagi and billsec or answeredtime
8:04AM 21 Dell SC1430 + Digium TE110P = Digital Noise in PRI
5:09AM 4 app-conference
 
Monday August 27 2007
TimeRepliesSubject
9:05PM 18 OT: DELL Platforms
7:42PM 1 Detecting tones
2:36PM 1 console/dsp 1.4.11
2:14PM 5 Can't create audio conversation between softphonesthrough Asterisk
1:51PM 0 error in linking libmfcr2
12:55PM 3 voip provider settings problem, please help
12:18PM 7 Stereo Conferences?
11:09AM 1 AstriCon Tutorials
10:11AM 5 Prepaid Billing: A2Billing, AstBill, ASTCC
9:06AM 0 libmfcr2 is giving compilation errors
7:01AM 3 Is it possible to register without sending the password
6:56AM 0 call forwading problem DTMF
5:59AM 0 Bad hangup event cause
2:42AM 4 No LongDistance for 1 Extension?
 
Sunday August 26 2007
TimeRepliesSubject
8:43PM 5 Calling Clients or Tele Marketing
4:31PM 0 Nokia cell connectel to asterisk
1:36PM 2 IAXmodem on Fonality?
10:56AM 2 foneBRIDGE2 setup
9:57AM 1 Is it possible to register without sending the password?
9:44AM 1 TDM400 and TDM800 fxo stop answering
9:15AM 0 Test - pls ignore
2:04AM 0 Davide Marcellan is out of the office.
 
Saturday August 25 2007
TimeRepliesSubject
11:49PM 4 Chan-capi Fedora 7
11:27PM 0 Asterisk & Speechphone/Mandi
9:32PM 0 SIP endpoint registeration problem
6:49PM 1 Avaya IPOffice and a SIP trunk to Asterisk
3:05PM 10 Asterisk 1.2 + Zaptel 1.4 + HPEC = Crash?
2:41PM 1 asterisk and vad/cng
7:32AM 7 Polycom firmware download
12:55AM 0 Help define the Asterisk regression test suite
 
Friday August 24 2007
TimeRepliesSubject
11:19PM 9 IAX2 trunking scalability
11:17PM 4 Restart status
10:24PM 0 AST-2007-021: Crash from invalid/corrupted MIME bodies when using voicemail with IMAP storage
10:11PM 18 which OS would be fine for asterisk
10:08PM 1 asterisk stable 1.2.x or 1.4.x
7:35PM 1 Tuning a ZyWALL for Asterisk
3:29PM 20 AsteriskNOW Web GUI
2:41PM 1 Can't create audio conversation between softphones through Asterisk
2:39PM 10 Keeping queue counters after restarting
12:28PM 2 TE120P digium card PRI_CPE error
11:22AM 0 [Fwd: Re: issues with caller ID , remote-party-id
10:32AM 0 MYSQL problem and configuration
9:45AM 0 DTFM not recognise
9:26AM 1 Error in loading libunicall.so module while running asterisk command
8:29AM 2 Problem compiling Zaptel 1.4.5.1
8:27AM 5 TE210P digim card PRI problem
7:55AM 1 recomend web interface for virual call center
7:49AM 1 Simulating errors (Busy / Out of Order)
 
Thursday August 23 2007
TimeRepliesSubject
11:44PM 2 xPL and Asterisk?
10:58PM 1 Linksys (PAP2) delay time between hung up and line release
10:55PM 9 asterisk as a softswitch
9:23PM 7 Is it posible for an incoming to ring to Polycom and cell at the same time?
8:33PM 9 Speech Rec on Voicemail
8:19PM 32 Stable-Stable Asterisk
7:01PM 4 Asterisk Prompt
6:57PM 0 What is this?
6:26PM 2 channel not hungup (zombie?) so call limit not reset to zero
4:37PM 3 1.4 Branch -- which revision
4:16PM 4 meetme conference problem
3:14PM 11 Asterisk Message Logs
2:51PM 3 [PHP-AGI] Problem executing script
2:11PM 2 libmfcr2 is giving definition error while compiling
2:10PM 1 unable to load chan_unicall.so
11:38AM 1 [Serusers] why combine ser with asterisk
8:37AM 0 asterisk configurator with E120P E1 card
7:18AM 0 B410P and echo
7:15AM 1 contact header is missing in 200OK for SUBSCRIBE
7:12AM 0 ASTCC and IVR
1:57AM 0 How to get callee extension in applicationmap(features.conf)
1:57AM 0 asterisk-users Digest, Vol 37, Issue 88
 
Wednesday August 22 2007
TimeRepliesSubject
11:49PM 0 Users Conference - Friday@12:30 PM EDT: Founders of Voicepulse
8:31PM 0 Queue Agents from Dialplan
6:39PM 0 VoIP encryption with SIP and IAX
6:19PM 2 Zaptel 1.2.20.1 and 1.4.5.1 released
5:05PM 7 Multiple servers using realtime
3:52PM 5 Chan_mobile and Asterisk SVN-branch-1.4-r80183 compile error
3:04PM 1 TDM400P Not hanging up fast enough
2:42PM 1 Agent status on Polycom phone?
2:27PM 3 [OT] IAX2 WiFi phone?
2:17PM 0 Asterisk Home Automation (was: Re: 99 bottles of beer)
12:32PM 0 asterisk with FAX problem
11:26AM 2 Cisco firmwares 3.6.3 vs 3.8.6
9:40AM 0 Which interface?
9:38AM 1 How do I configure asterisk?
7:22AM 2 How to re-read values from database in Trixbox
6:53AM 3 rfc3680, reginfo+xml
1:51AM 1 DUNDi, So Easy A Caveman Could Do It!
12:51AM 3 Polycom and NAT
12:36AM 7 Polycom behind NAT won't register to * server behind ALG
 
Tuesday August 21 2007
TimeRepliesSubject
8:40PM 0 Enable Media Atribute on 180 Ringing
8:30PM 0 Call back or some voicemail notifing.
8:24PM 0 AST-2007-020: Resource Exhaustion vulnerability in SIP channel driver
8:15PM 1 Asterisk 1.4.11 released
7:46PM 0 Mitel 5020 IP phones
7:17PM 1 Contact: header and NAT.
7:00PM 4 Dialogic support
6:15PM 1 SET EXTENSION
5:38PM 4 asterisks addon make problem
4:08PM 3 Problems with overlap dial and Xorcom Astribank BRI
2:28PM 1 Call queue problem
1:36PM 0 Asterisk in Soekris 5501: Is Astlinux the only able solution?
12:22PM 5 TC400B and show transcoder
10:34AM 0 Saftware RAID1 or Hardware RAID1 with Asterisk (Vidura Senadeera)
7:11AM 2 Which GUI for ACD edition ?
4:36AM 3 compatibility of PRI Two B channel transfers TBTC/2BTC
2:03AM 47 Saftware RAID1 or Hardware RAID1 with Asterisk
1:25AM 6 Passing Variables to Voicemail's Email Notification
1:23AM 3 TE405/TE410P help updating from 1.0 to 1.4
 
Monday August 20 2007
TimeRepliesSubject
11:25PM 1 Zaptel 1.2.20 echo cancelling problem
7:42PM 6 Realtime Queue Members
5:57PM 2 Setting caller ID on outgoing calls.
5:14PM 0 SpanDSP/TxFAX FAX Status
4:45PM 4 Cdr reports
3:17PM 2 Asterisk as ISDN PRI Proxy
3:11PM 0 OT - IMAP voicemail statistics
1:33PM 0 Got SUBSCRIBE for extension...., but there is no hint for that extension.
1:24PM 1 Disabling Asterisk Authentication
1:15PM 0 How to configure and use GCE4019VOIP phone using asterisk
11:18AM 1 1.4.4. caller ID not working ?
9:40AM 6 Queues with Dynanic Users (BUG?)
7:47AM 5 Firefly IAX2 configuration
7:47AM 5 Redundancy / Failover
5:27AM 0 asterisk1.2.24 or asterisk1.4.10.1
5:03AM 1 Application for Home Delivery Restaurants
1:00AM 5 Quick DUNDi Poll Questions, For All Asterisk, Users, Please Give Feedback
 
Sunday August 19 2007
TimeRepliesSubject
10:26PM 15 Nokia cell connected to Asterisk
9:29PM 2 How many calls can use the same username
7:49PM 1 Rewriting the From and Subject from voicemail for a MMS Message to a Cell Phone - like visual voicemail
7:39PM 0 Asterisk 2 Speechphone/Mandi
4:46PM 1 Snom 300 Hints and LIne Buttons
3:39PM 0 Increase Volume on AGI
3:36PM 0 flash zap FXO port from SIP device (SPA-2002) using RFC2833 or SIP INFO
3:12PM 1 Someone please explain FXO/FXS module/channel relationships in Zaptel configuration to me
11:23AM 4 Change Packetization Time
9:15AM 5 Asterisk and Client NAT
5:26AM 0 The Future of Telephony, by Leif Madsen, Jared Smith, and Jim Van Meggelen
5:11AM 1 CDR Disposition Value with ODBC
2:37AM 8 GotoIf not working with ${EXTEN} for me in 1.4.8
 
Saturday August 18 2007
TimeRepliesSubject
10:46PM 3 2 asterisk servers, how to connect them together?
7:10PM 2 Best way to detect unknown and/or private incoming caller-id?
7:06PM 6 Blacklisting Toll-Free etc.
12:25PM 1 incoming calls in SIP
11:44AM 2 Forwarding calls, passing Caller ID (or not)
1:41AM 1 Asterisk Manager Proxy - Still required?
1:06AM 2 Asterisk Channel as MusicOnHold
 
Friday August 17 2007
TimeRepliesSubject
11:15PM 2 Zaptel 1.2.20 and 1.4.5 released
9:34PM 11 Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback
7:15PM 2 gsm errors
5:18PM 4 Subscribe/Notify MWI not working for non-numeric accounts w/X-Lite
4:51PM 2 [asterisk-biz] Skype Outage Leaves Millions Speechless
4:16PM 0 MOH being activated in the middle of a call
4:09PM 0 analog lines running agi on hangup question
3:20PM 0 Suggestions on how to debug strange DTMF problems
2:53PM 0 DISA and Ericsson Dialog 3212
2:40PM 0 Jain-Sip-Applet-Phone with Asterisk
1:37PM 10 Call Limits
1:06PM 3 Connecting a GSM gateway to a FXO port
1:03PM 1 Detecting DTMF Tones from Muted app_meetme Participants
12:28PM 2 No audio on ISDN PRI calls
10:54AM 11 Lock extension from asterisk
8:42AM 1 1.4.10.[0,1] crashes when call parked
7:53AM 2 Problem in installing libmfcr2 for configuring MFC/R2
6:49AM 0 Hook flash time problem on TDM400/FXS
 
Thursday August 16 2007
TimeRepliesSubject
6:56PM 0 Friday@12:30 PM EDT: All about DUNDI
6:38PM 23 RAW asterisk!
6:33PM 1 Outbound SIP authentication with dynamic credentials
4:23PM 7 Experimenting- Sip dialing with Zap
4:08PM 1 Asterisk, PAP2T and 2Wire DSL router
4:07PM 8 Heavy duty environment - Is TDM2400P suits?
4:00PM 0 IAX Trunk
3:10PM 0 Asterisk & SNOM Page/Auto Ans - SNOM only beeps intermittently
2:58PM 4 A102 card, BT ISDN30e, silence
2:48PM 1 Set CALLERID(num) to a specific number only if ${CALLERID(num)} is not an NANP number
2:40PM 2 Outbund Route via Extension
2:10PM 1 Error in intalling library for r2mfc support to asterisk
1:50PM 2 Where I will get astersik.spec and zaptel.spec
1:06PM 0 chan-capi in 1.4.10.1
12:14PM 15 Authenticating SIP user in LDAP database instead of SIP.conf file
11:06AM 0 Call back voicemail.
11:01AM 0 About cards for ISDN-PRI in Ireland
10:42AM 1 Introducing myself
9:40AM 3 Incoming and Outgoing zaptel configuration : ISDN30e
8:35AM 49 99 bottles of beer
7:04AM 2 tone in linksys pap2t
5:33AM 0 Mitel IP 5020 phones
4:36AM 6 asterisk multiport
1:52AM 2 Seeking opinions: Polycom IP330 phones?
1:22AM 2 zaptel update locks up computer from 1.2.9.1 to 1.2.19
 
Wednesday August 15 2007
TimeRepliesSubject
10:12PM 11 GUI for Asterisk realtime
8:36PM 0 Asterisk & SNOM Page - SNOM beeps intermittently
8:33PM 2 Load balancing SIP trunks?
8:28PM 0 3-com Model 3102 IP-Phone / Sip firmware download ?
8:25PM 0 Client-negotiated Codec Instead of Transcoding?
7:47PM 0 iaxtel
7:14PM 16 TDM400P FXO click sounds
5:51PM 14 Dialplan / AGI autoanswer question
4:24PM 3 SIP Events
4:08PM 1 Callback DTMF Problem
3:42PM 6 iaxmodem, chan_capi, hylafax problem and faxing in general
3:29PM 0 slightly OT: Polycom SIP phones
2:14PM 2 CallerID Error causes problems for Polycom phones
1:49PM 8 CDR billsec greater than duration
12:11PM 2 Disable MoH for certain phones
12:08PM 3 Sangoma Wanpipe installation problems
11:33AM 1 Dundi x ENUM
10:36AM 5 why is nonce="584760da" used in sip packets?
8:10AM 0 DUNDi limitation?
6:51AM 3 "Remote" extension search?
 
Tuesday August 14 2007
TimeRepliesSubject
10:17PM 5 Some advice
5:16PM 9 Dial plan suggestions
4:48PM 3 asterisk 1.2.24 installation
4:19PM 6 Recognize 800 number
2:52PM 1 DTMF on Bridged ZAP call
1:26PM 0 Maximum retries for seqno 102 when re-inviting.
12:17PM 5 BLF with Aastra
10:53AM 3 Faulty voicemail
10:17AM 9 IVR and MySQL
2:41AM 0 Alert_info for AudioCodes MP-124
2:07AM 28 Patent issues, what features we can't use?
1:42AM 0 REALTIME application vs RealTime function
 
Monday August 13 2007
TimeRepliesSubject
10:33PM 0 Anyone using zaptel under Solaris?
9:17PM 2 FXO Modules and Sip Outbound
8:25PM 2 AGI answering the channel even though I never asked it to
5:59PM 0 Using hints over DUNDi
3:53PM 1 Problem with Page command
3:38PM 0 test list
3:08PM 0 FW: The trixbox Revolution Continues! Sign upforthe Webinar.
1:49PM 11 CDR-CSV Processing
1:24PM 1 FreePBX
1:07PM 1 Asterisk RTP bridging
12:50PM 1 bristuff - qozap dirver bug (and fix?)
11:30AM 0 Originate and tracking
10:51AM 0 about REALTIME application
10:45AM 2 Does Digium TE120P card support MFCR2
9:55AM 1 Does digium TE120P card support for MFC/R2 protocol
9:40AM 0 how to configure for the R2MF
9:32AM 0 From where to download all libraries required for configuring R2MF
8:52AM 0 Weird noise problem on SIP transfers...
8:21AM 0 Problems on SIP gateway (especially Planet VIP-450)
6:34AM 0 Codec issues (ilbc->g729)
3:53AM 1 Can't HANGUP call or channel on 1.4.9
12:52AM 10 How strip +1 from caller id on inbound call
 
Sunday August 12 2007
TimeRepliesSubject
11:01PM 1 Call file & IAX Trunk: Call Failed, Reason 0
10:20PM 0 TDM 2400 ?
9:25PM 1 Asterisk 1.2 TDM24xx and B410P
8:02PM 18 20min waiting time
7:38PM 5 Converting an audio file to a ".gsm" format
3:58PM 1 Shared Line Appearance - Aastra 55i - Does it work?
12:54PM 1 Playback a video file?
12:25PM 12 New Pico-ITX
 
Saturday August 11 2007
TimeRepliesSubject
11:07PM 4 asterisk and telewell isdn hfc problem
4:51PM 1 BLF for Queue
4:09PM 11 indications.c: Can't generate that much data!
2:52PM 18 LumenVox Speech Recognition
 
Friday August 10 2007
TimeRepliesSubject
10:57PM 2 Hardware Platform Recommendations for Digium Card Compatability
10:23PM 2 Faxing through a PAP2
10:22PM 0 asterisk-users Digest, Vol 37, Issue 46
10:02PM 0 Mitel SIP phones
9:05PM 2 misdn and incoming fax detection
6:37PM 3 Pickup command
6:26PM 7 Ordering BRI From AT&T
5:48PM 0 Sending live audio in Asterisk
5:40PM 0 analog fax extension dialing out
5:07PM 3 Locating Asterisk documentation after installation
3:49PM 9 Asterisk Manager to Record Greetings
2:42PM 23 Sort of OT: PBX vs CO
12:49PM 2 Polycom question - removing a soft key functionality
11:51AM 1 How to verify IAX trunking
11:12AM 0 Asterisk action when transfer occurs
10:13AM 4 OT Provisioning http-server-enabled devices (Was: Siemens Gigaset DECT base provisioning)
9:56AM 5 A TrixBox 2.0 seems to be asleep...
9:35AM 6 sip ... codec conversion matrix
2:04AM 0 MINNESOTA - TwinCities Asterisk Users Group 8/11/2007 Echo Cancellation Threat or Menace?
1:12AM 4 Dialplan loop
12:37AM 15 FW: Can you reload only one conf file?
 
Thursday August 9 2007
TimeRepliesSubject
9:48PM 0 VOIP Provider- Callcentric
9:48PM 5 forking from a dial plan?
8:59PM 3 PRI Question
8:35PM 0 False hangups with TDM400P and Kewlstart
8:12PM 2 Forced Ping or re-registration process for SIP devices or accounts/lines
7:25PM 2 Failover Configuration
5:23PM 17 How to use OpenVPN with Asterisk
5:16PM 0 Polycom Phones Call Hold Reminder function problem
5:04PM 4 How to disable DND feature key in Polycom Phone
4:59PM 2 LIBPRI - video calls over ISDN
4:51PM 4 Need Help in changing Voice message
3:39PM 9 Terrible clicking on T1
3:27PM 5 The quest for making "hint" more flexible continues - using Realtime now
3:10PM 4 Asterisk Help
2:23PM 0 transfer/conference
1:55PM 0 Friday Aug 10 @ 12:30 PM EDT - Asterisk Users Conference
1:31PM 5 Call forward at telco
12:51PM 0 Level3 WIreless
12:38PM 0 Polycom 330 Speakerphone
12:22PM 11 Allison Smith?
12:15PM 5 705 DIDs for Collingwood Ontario?
11:21AM 1 strange warning
11:01AM 2 generating a GUID
10:52AM 1 Overlapping Playback() with Dial()?
10:27AM 1 usage of each field
7:41AM 2 a couple of new tutorials
5:37AM 1 how to push callerid for each user from sip phone on one side through asterisk (Digium) to E1 card running application on other side
3:44AM 19 Major Digium Card Problems
 
Wednesday August 8 2007
TimeRepliesSubject
11:58PM 0 Sangoma BRI card -- National ISDN/North America support (Having problems with analog disconnect supervision?)
10:24PM 2 Question on the Monitor command on AMI
10:21PM 1 Using CURL
9:32PM 4 MoH mysteriously stopped working
6:33PM 15 les.net losing DID's
6:20PM 8 FW: The trixbox Revolution Continues! Sign up for the Webinar.
5:56PM 6 Paging Application - Polycom 601
5:45PM 1 RoundRobin Holding Memory?
5:38PM 1 Howto generate a Manager Event from the Dialplan?
4:49PM 0 FW: OT - Callto:// tags inside web pages
4:30PM 11 VoicePulse Connect
3:26PM 1 Order of matching SIP packet to sections in sip.conf
3:24PM 19 How to write a function with a return value in Asterisk
2:29PM 3 PRI Reset
1:35PM 0 Zap Bridge Question
1:23PM 1 Help : problem in SLA (Shared Line Apperence
12:56PM 1 Buddy watch and the hint priority - brain teaser
12:37PM 1 pick sip channel whn two party talking
12:31PM 0 Asterisk AND Cisco Phones in H323 cloud...problems with some models.
12:02PM 4 Siemens Openstage & Asterisk ?
9:59AM 4 asterisk wait for traling digits
9:56AM 4 Monitor doohicky got event Event 160 on channel..
8:51AM 1 OT - P-asserted-identity and remote id
7:23AM 2 Siemens Gigaset DECT base provisioning
4:37AM 4 Method for scripting options specified in make menuconfig
4:35AM 1 E1 or analog line
2:47AM 0 Looking for unified messaging expert
 
Tuesday August 7 2007
TimeRepliesSubject
10:24PM 2 turn off music on hold for a single sip user
10:14PM 0 Asterisk 1.2.24 and 1.4.10 released
9:45PM 0 ASA-2007-019: Remote crash vulnerability in Skinny channel driver
8:22PM 1 Switchtype
7:38PM 5 Outbound dialing
6:13PM 5 Macro Overlap
3:18PM 2 Use of context=... in [default] section of sip.conf
2:51PM 1 Intermittent busy tone detection on loopback setup
2:51PM 10 OT - Callto:// tags inside web pages
2:35PM 0 how to specify a channel inside txfax command
2:16PM 4 OT, I'm looking for SIP/register-enabled softphone
2:06PM 1 caller ID strangeness
2:01PM 4 TE207P Question
2:01PM 8 Which spandsp & unicall version to use with 1.2?
1:24PM 6 Prblem with Page Hight While Faxing over uLaw
11:56AM 1 .call file and logging
9:44AM 7 test the email-list
8:10AM 3 ISDN30 card for UK : sanity check
7:17AM 0 users.conf in 1.4
5:53AM 2 Login info from Active directory
5:28AM 2 How to stack Sangoma Remora cards
2:10AM 0 ????????????????????????
 
Monday August 6 2007
TimeRepliesSubject
11:08PM 2 sip issue with one way audio
8:04PM 7 CDR/MySQL basic config
6:28PM 0 Friday Aug 10th Asterisk Users Conference at 12:30 PM EDT
6:09PM 19 low-level dump for PRI dchan debugging
4:55PM 0 Digium|Asterisk World
4:09PM 1 TAE to RJ11 connector (hope not OT)
3:34PM 18 Free sitting
3:00PM 6 iax2 registration being rejected
2:58PM 0 SIP RegEvent - RFC3680
2:42PM 5 ATA phones ring when they register
2:28PM 0 Setting gain levels with mISDN
2:04PM 1 Cant Play gsm file
12:09PM 0 How to debug OH323 Channel (version 0.7.3)
9:05AM 2 Before Bridging Two Calls
8:42AM 4 help: H323 and SIP
8:06AM 1 Re : Connecting two Asterisk servers with a framerelay
8:05AM 3 A102d samgoma's card
7:45AM 2 Telco is not detecting HangUp w/ TDM400P
12:08AM 3 I am looking for VOIP (SIP/IAX) providers that support sending me RDNIS info on forwarded calls. Are there any providers out there that support this?
 
Sunday August 5 2007
TimeRepliesSubject
11:32PM 0 Linksys 224P switch and Polycom PoE phones
3:46PM 2 How does one use sip_autoreg
3:41PM 0 are there g729 sound files available?
2:01PM 0 chan_alsa - no sound / strange sound - 1.4.9
10:15AM 0 Connecting two Asterisk servers with a frame relay
4:44AM 22 Sangoma PRI
4:42AM 0 Agents being bounced from queues after a call and sometimes randomly...
1:16AM 5 ! Command from -rx?
 
Saturday August 4 2007
TimeRepliesSubject
11:52PM 3 text2wave Voices Improvements?
10:09PM 0 zttool says tdm800 is OK, but it won't recieve calls.
8:06PM 0 * and SIP ocupped
8:02PM 1 Connecting two Asterisk servers with a framerelay connection
7:07PM 0 Update zaptel on business edition.
7:06PM 0 quintum AFT200 connection to Asterisk
7:03PM 2 Pre-recorded first and last names audio database
5:17PM 0 Outcall 1.40 released
2:11PM 1 Time Limit on Call or Conference Room? "NEW ASTERISK PROVERB"
1:51PM 0 Turn off musiconhold
11:07AM 3 Hardware advice for 100 extensions, routing via ISDN
10:10AM 3 Connecting two Asterisk servers with a frame relay connection
9:41AM 2 IAX2 - DualServer Problem
8:42AM 3 asterisk 1.2.14 with GUI
8:21AM 6 asterisk always rining phone
3:54AM 0 VoiceMail Call Limit Messages
3:31AM 2 IAX bat phone.
3:19AM 0 Handling message for SAPI/TEI=0/0 Repeated Quickly
1:55AM 0 queue beep
 
Friday August 3 2007
TimeRepliesSubject
7:31PM 16 Measuring Jitter in Asterisk
6:58PM 3 DIALSTATUS not set
6:50PM 0 Several doubts on Asterisk as an UAC
6:48PM 2 Time Limit on Call or Conference Room?
6:12PM 2 Macro and Arguments
6:04PM 0 CONSOLE=Console/dsp
5:38PM 0 "Asterisk can be attacked using buffer overflow."
3:27PM 0 Asterisk, ISDN AVM C4 and Terrible noise
12:45PM 4 partial ChanSpy
12:27PM 5 Difference between WaitExten and TIMEOUT (response)
10:16AM 1 Knowing zap channel status
8:34AM 0 SIP-6.0 software for Siemens
8:33AM 0 B410P echo cancellation
6:55AM 2 Where to find t38modem
4:07AM 0 Fwd: Re: PRI - DS3 Calls Dropped
3:38AM 5 PRI - DS3 Calls Dropped
3:10AM 0 Asterisk configuration directly with Mandi (Speechphone)
 
Thursday August 2 2007
TimeRepliesSubject
11:17PM 0 Hints and Noop
9:32PM 0 PhonicEQ T100P
9:02PM 1 A simple IVR extension problem
8:31PM 9 Unicall and Private CID
7:59PM 0 callback and bridge problem
7:35PM 10 PRI/T1 data rate...
7:24PM 1 MySQL + Realtime + SIP Registration
6:21PM 2 dtmf get data
4:23PM 49 Teliax Quality of Service
4:08PM 1 AGI SAY TIME
3:29PM 10 asterisk1.2 to 1.4 g711a fax
3:12PM 3 H.323
3:11PM 6 Receiving SIP calls without registeration and dynamic IP address
2:13PM 0 uptime script?
1:42PM 1 Recording calls after queues?
1:32PM 2 TE220B
1:06PM 2 radius support
12:47PM 7 Blip every 30 seconds?
12:34PM 1 problem with rfc2833
9:38AM 0 OT - How to switch headphones between softphones on Linux ?
6:40AM 1 Problem in making SIP call after compiling Asterisk server
2:31AM 0 chan_sip.c error
 
Wednesday August 1 2007
TimeRepliesSubject
9:42PM 0 dtmf issues over sip and pri
9:40PM 6 Polycom 320 - Can it actually be configured?
9:40PM 0 Announcing free (GPL) VXML for Asterisk - Voiceglue
9:31PM 3 Couple installation questions
8:50PM 4 2 Digit Issue
7:42PM 3 Retail DID provider ?
7:11PM 6 Asterisk DTMF Tones
6:36PM 0 perl script to generate new sip.conf users
6:15PM 6 Slightly OT: SNOM & PoE
5:06PM 9 pri "call by call" trunking?
3:47PM 1 Agent Question
3:43PM 5 Problem with the dial command
1:48PM 20 Hardware that can ring my phone?
1:47PM 0 Can you specify a sip UA's codec based on IP?
12:44PM 0 Help on AsteriskNOW
12:15PM 11 How to use stun server?
11:02AM 7 multiple pbxes, multiple domains, same user ids?
4:34AM 1 Asterisk ref book
3:50AM 4 Live Answering Service with Direct SIP Connections and Light Accounts starting at $14.95/mo
2:04AM 11 Problems building zaptel 1.4.4
1:44AM 3 TE120P in Canada