asterisk users - Aug 2010

Tuesday August 31 2010
11:27PM 2 Logging the CID from the Privacy Manager
10:50PM 0 STUN
6:22PM 6 Pickup parcked call from Aastra 9480i ct cordless
5:03PM 0 Asterisk with Blockhosts
3:37PM 7 Yes it is a dimensioning question! Atom CPU
2:52PM 2 Running System() after call completion, not in 'h'?
9:57AM 2 asterisk core dump
8:04AM 8 No audio on call forward after upgrade from Asterisk 1.4 to 1.6
2:20AM 1 Cisco 9971
Monday August 30 2010
8:43PM 0 Voicemail prompts fuzzy and quiet
5:26PM 0 Wifi + SIP + Asterisk
3:31PM 3 Maximum Wait Time queue option
2:37PM 3 help with dialplan
11:16AM 29 Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny
7:34AM 0 How to Billing for MeetMe Conference?
7:19AM 1 SIP Debug Messages
3:29AM 5 Asterisk routing to SoftSwitch
2:52AM 2 Could MeetMe invite someone to the conference?
2:48AM 2 Digest Username/auth name mismatch
1:01AM 1 Web-meetme
Sunday August 29 2010
5:31PM 1 asterisk-users Digest, Vol 73, Issue 63
7:25AM 1 evil disconnect of call with cisco 1760
1:27AM 2 Why does Digium not respect their own development guidelines?
12:43AM 0 Problem routing incoming from-pstn calls using different contexts
Saturday August 28 2010
9:20PM 10 Play a number of files to a caller
3:32PM 0 $250 Asterisk app install bounty
9:22AM 5 Asterisk does not translate from wav to alaw
8:55AM 5 only part of dialplan available
Friday August 27 2010
10:45PM 0 Asterisk 1.6 Displaying BackGround() in call trace but no audio is heard from caller
10:18PM 2 Early media and IAX2
10:13PM 0 Compiling into AsteriskNow install
9:58PM 1 Migrating 1.4 to 1.6.2
6:49PM 3 Asterisk Crashed - But why?
3:48PM 21 ASterisk CDR file Master.csv
2:55PM 1 Protect yourself
2:17PM 0 Asterisk DTMF RFC2833 issues
2:04PM 1 asterisk-users Digest, Vol 73, Issue 58
1:13PM 0 Duplicate channel variables after transfer
12:51PM 2 Call Forwarding
12:05PM 0 queue agent and blind transfer
11:39AM 1 music on hold in blind transfer
9:27AM 3 dynamic MeetMe, min. digits
1:58AM 0 Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround() but no audio is heard from the phone
Thursday August 26 2010
6:55PM 3 double DTMF digits
5:25PM 4 CDR on Transfer...
4:38PM 5 Use of AGISIGHUP
3:09PM 0 OrderlyStats or QueueMetrics
2:32PM 1 Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010.
10:19AM 0 sms - your suggestions
10:10AM 1 MusicOnHold class working for internal calls, not for external
8:41AM 1 Timecondition fallthrough on 2nd GSM Modem, First modem and ZAP's are all fine
Wednesday August 25 2010
10:03PM 3 help for a new user
9:42PM 4 CDR Help
8:06PM 2 Quick Question - Jabra Headset and Aastra 53i - Where is the speaker/headset enable setting on Aastra UI?
6:08PM 2 Asterisk ACK/BYE question
6:04PM 1 AMR Codec
3:23PM 8 AEL - what is error: ael.flex:647 ael_yylex: Unhandled char(s):
12:27PM 2 ODBC Voicemail storage
8:54AM 5 asterisk-1.8 problem with one-way audio with no nat
8:12AM 0 Asterisk codec config issue with ALAW
2:03AM 2 Looking for MIB description
12:02AM 3 Asterisk 1.6.1 Won't Play Default ULAW Files
Tuesday August 24 2010
11:17PM 6 1.6 and asterisk gui
11:04PM 0 DAHDI compile warning
9:44PM 0 Announcing Adhearsion 0.8.5
9:13PM 0 Tones of dtmf during call
7:21PM 0 Fwd: Re: Make a transfer for external line.
6:43PM 1 asterisk-1.8.0-beta4 - compile error
5:45PM 0 Asterisk + SMS + PRI
3:35PM 2 Asterisk 1.8.0-beta4 Now Available
1:51PM 21 Should I move to 1.6 or 1.8, or stay with 1.4?
1:05PM 2 asterisk + cisco 3825 with ISDN
12:53PM 2 Attempted SIP connection by foreign host. Help!
12:48PM 9 Include and Realtime
12:40PM 0 Transfer + speed dial button problem?
8:34AM 2 How to do voice barge in using asterisk server
7:31AM 0 OT - How to blacklist a driver in /etc/modprobe.d without reboot [SOLVED]
7:11AM 1 OT - How to blacklist a driver in /etc/modprobe.d without reboot
3:25AM 4 IAX2 - Separate Signaling and Media?
Monday August 23 2010
11:46PM 1 sip probe syntax
8:00PM 5 Asterisk, HylaFax and Cardiff
6:13PM 1 channel stay up when extension unreachable
5:16PM 0 Transfer to non registered extension creates call hangup
5:03PM 2 All phones ringing when temporary loss of Internet
4:42PM 10 How to debug this specific issue?
4:34PM 3 can't build resODBC on SUSE 11.3
4:05PM 3 Dahdi install gone wrong
3:57PM 2 Asterisk voicemail server - gsm notifications
3:56PM 9 DAHDI not detecting caller hangup
2:06PM 2 outbound SIP trunk hunting (or any fxo for that matter)
1:28PM 5 Make a transfer for external line.
1:14PM 15 problem with mssql and Asterisk 1.8.0 beta3
11:14AM 2 How to prevent soft hangup from being necessary ?
10:35AM 1 EMail on Missed Call
6:58AM 1 How to do barging using asterisk server.
3:50AM 0 Asterisk dialup connection?
Sunday August 22 2010
2:11PM 1 .call files with application/data are not generating correct CDR
12:33AM 1 NVidia component out
Saturday August 21 2010
10:38PM 4 Mobile answer machine cut off
8:25PM 24 Opensource Speech recognition for Asterisk
9:49AM 0 iax stresstest client
Friday August 20 2010
2:12PM 0 Push to talk over cellular
2:28AM 3 not work!
Thursday August 19 2010
9:56PM 3 Executing system commands through Manager API
6:54PM 0 Call-limit field
5:20PM 11 asterisk + openBTS
3:58PM 0 Loop Detection / SIP
3:58PM 1 AstriCon approaches: Innovation Awards, your attendance wanted!
3:29PM 9 setting variable for a DID number
1:52PM 4 AMD message
12:14PM 0 3g call support for ISDN line
8:14AM 13 Codec choice
7:21AM 15 Calling Line Identity - any ideas
Wednesday August 18 2010
10:46PM 2 IXJ issues on 1.4.35
8:54PM 3 CDR variables
5:51PM 15 WaitExten() always times out
5:14PM 2 Fwd: AsteriskNow REGISTER'ing s@ extension for all inbound trunks
4:10PM 5 Playing with sipvicious ..
1:12AM 0 Pfsense and IAX2 - What is the proper firewall NAT setup?
12:36AM 0 Polling DND status of a Linksys SPA9xx/5xx phone?
Tuesday August 17 2010
7:43PM 1 Directory routing to wrong extension if dial tones are pressed too quick.
7:39PM 1 Asterisk with Motorola Canopy
6:25PM 3 Add & play moh-files without reload
2:09PM 3 Create File Directory
1:00PM 3 Convert wav-file to alaw-file
1:00PM 9 sending sms from Asterisk server
11:59AM 7 MySQL Connect problem...
9:12AM 0 MP3Player audio format
8:36AM 5 dial_exec_full problems with TDM400
Monday August 16 2010
8:21PM 4 Polycom 331 freezes connecting to FreePBX
4:10PM 6 parkcall: How to remove announcement.
12:42PM 4 colored CLI with reattach
10:46AM 1 Asterisk Hardwares
Sunday August 15 2010
6:35PM 11 Realtime Context
3:09PM 1 Use of Storage Area Network with Asterisk
1:46PM 1 603 error
1:32PM 0 Timing on Asterisk
6:46AM 1 Removing `chan_dahdi.conf`
Saturday August 14 2010
11:01PM 3 BLF/Call Pickup using SPA942, SPA962, SPA932
Friday August 13 2010
8:20PM 12 Asterisk on AMD
8:13PM 3 IXJ Quicknet PhoneJack issues
6:08PM 6 installing with yum
4:36PM 0 Enhancing snmp mib
3:43PM 3 4 Port FXO interface
1:08PM 9 install asterisk
7:57AM 12 realtime sip peers : musiconhold class
3:15AM 0 How to Record with Konference when it has no record option?
12:13AM 3 OT: UK PPP certification -- what is it?
Thursday August 12 2010
7:08PM 3 Callback script anyone
6:52PM 2 Problems with meetme in 1.4.26
2:50PM 0 VUC Friday 13th: Skype, then Video
11:52AM 0 Skype
9:29AM 0 BRI line issue on third call
9:01AM 2 Recording the conversation with MixMonitor() ends when the call is transfered
Wednesday August 11 2010
9:54PM 1 Youmail RDNIS
9:40PM 3 PRI errors no D channel
6:04PM 1 DAHDI config file system.conf
5:53PM 3 VM Extension : asterisk
4:30PM 0 Aastra 6739i Support
4:08PM 4 channel variables in AGI
2:12PM 3 How to set up Asterisk to deliver a trunk sip connection?
10:14AM 12 Asterisk 1.8 beta3 - Unable to stop/start/restart deamon
8:59AM 6 billsec exceeds duration on some calls
8:36AM 8 asterisk on Vmware
6:36AM 0 No CDR with originate from manager and then an redirect to a dial from manager
Tuesday August 10 2010
11:28PM 2 IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?
9:49PM 0 Asterisk 1.8.0-beta3 Now Available
9:49PM 0 Asterisk Now Available
9:48PM 0 Asterisk 1.4.35 Now Available
5:17PM 0 asterisk-users Digest, Vol 73, Issue 24
4:51PM 6 How to determine which party hangup the call? cause of Hang-up needed.?
3:44PM 1 Dial option 'r' not working anymore?
1:33PM 2 PRI D-channel bouncing
11:42AM 6 Asterisk on Ben NanoNote?
7:04AM 2 Call agent when queue is empty and there is a voicemail left
6:13AM 0 MeetMe will record automaticlly even without 'r' option??
5:44AM 5 speciality of SIPp and SER(Sip Express Router)
2:13AM 1 DEBUG: Cannot find variable 'XXX' ??
12:19AM 2 Playback during call
Monday August 9 2010
7:03PM 10 Correct Caller-ID
6:56PM 6 check channels
6:08PM 8 'System' application in asterisk
4:25PM 0 Allison Smith Hilarity
3:41PM 6 Connecting two calls with Originate
3:03PM 3 SIP response 500 "Server Internal Error"
1:04PM 1 redirect based on incoming number
12:27PM 1 op_div: non-numeric argument
9:01AM 2 Prepay Limited Calls.
8:36AM 1 MeetMe VS. Conference
7:31AM 0 [SIP/H.264] Codec negotiation problem ?
Sunday August 8 2010
8:29PM 0 PBX Status-like module for AsteriskNow?
1:10PM 5 How to track a call result originated from originate AMI command
11:49AM 0 Asterisk 1.6.2 FastAGI Hangup Problem
Saturday August 7 2010
7:25PM 11 Monitor asterisk
2:06PM 3 AMD setup in Astersik
7:16AM 1 Scilence problem on running call
7:09AM 0 Should external hosts be able to register on Asterisk behind a firewall?
5:45AM 3 Dahdi issue on sangoma A200
5:27AM 0 shrinkcallerid
1:17AM 0 Set outgoing number in filename of the recordings
Friday August 6 2010
6:15PM 1 Reinstalling Asterisk due to hardware changes
3:33PM 3 Asterisk 1.4 and TE420P
3:19PM 5 OT: Grandstream GXV3140
2:45PM 7 How does deny/permit work in sip.conf?
2:36PM 3 Using a 1.4 config with 1.6
2:13PM 1 How do I install speex for
11:59AM 4 How do I install speex for asterisk?
6:08AM 4 Security - What inbound variables can attackers populate or use when calling?
5:54AM 2 How to reuse mysql connection between AGI's
4:58AM 0 Missing Mailboxes on SIP
Thursday August 5 2010
8:52PM 3 Asterisk 1.6 without DAHDI
8:26PM 2 rolling over Master.csv CDR File
8:11PM 1 CDR report
9:12AM 17 Codec Conversion
8:28AM 6 AMI Command
7:56AM 1 Can ChanIsAvail return status from sip uri using router ip
6:08AM 0 Which Python binding to use ?
5:52AM 0 No Mailbox Subscription in SIP Users Suddenly
1:52AM 5 Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
Wednesday August 4 2010
7:12PM 4 Identify remote prompts: Partial audio matching?
3:57PM 5 Asterisk and RAID
2:44PM 1 Asterisk (1.8-beta2) and SIP IPv4/IPv6 dual-stack possibilities
2:08PM 6 Tweaking AMD in Asterisk
1:45PM 0 Queue to queue transfer error
12:59PM 0 can't write to queues_additional.conf
12:28PM 5 Asterisk not working with Festival
11:28AM 2 callerid between 2 asterisk servers
9:12AM 2 How to record a file and play some other file at the same time
4:40AM 1 CDR: MySQL query
Tuesday August 3 2010
8:57PM 4 Using SIP to dial extension that will give an outside line
6:19PM 9 Dial() M parameter in
5:39PM 4 outboundproxy timeout or qualify
5:20PM 0 ConfBridge
5:06PM 1 real-time queue problems
4:25PM 1 Garbled messages - format_wav_gsm.c:414 wav_read: Short read (60) (No such file or directory)!
3:22PM 0 asterisk-users Digest, Vol 73, Issue 5
2:48PM 3 Fax/Modem, Asterisk, Channel Banks
1:28PM 7 FYI: Seen the 2600Hz announcement?
8:13AM 6 sip.conf register in realtime DB
6:05AM 2 RTP stream not passing through router with port forwarding
12:58AM 2 chinaroby fxo card - never heard of them
12:40AM 2 Asterisk 1.6 and PrivacyManager with SIP
Monday August 2 2010
8:24PM 0 alaw.h in app_meetme.c
7:56PM 6 Caller ID issue
7:36PM 6 Femtocell to VoIP?
7:15PM 0 Whither app_nv_faxdetect
6:59PM 6 FAX Options
6:26PM 18 What do you use for Invoicing?
4:56PM 1 asterisknow
4:35PM 3 IAX softphone
1:35PM 10 asterisk compatible cards?
12:53PM 5 mapping of disconnect reasons
12:37PM 7 Asterisk and TV media server
12:12PM 6 Codec negotiation : expecting G726, getting G711a (alaw)
7:34AM 1 Any Free software that can connect to an Asterisk Server and Do video Conferencing?
4:46AM 3 how to place a call on hold and play music on hold using agi
2:12AM 1 SIP Status: 401 Unauthorized (0 bindings)
Sunday August 1 2010
11:47PM 2 Exporting Blacklist database
9:27PM 5 fail2ban does not work for my asterisk installation
1:58PM 4 # -key not to be 'transfer'
6:25AM 0 How to read span debug ?