asterisk users - Sep 2014

Monday September 29 2014
TimeRepliesSubject
1:52PM 0 Dahdi problem with dahdi_genconf
 
Sunday September 28 2014
TimeRepliesSubject
3:51PM 0 how to make voip client cannot use same username?
8:58AM 0 Intercom Telephone Feature
5:54AM 0 How to append the recording file.
 
Saturday September 27 2014
TimeRepliesSubject
3:28PM 0 can PJSIP_MEDIA_OFFER work like SIP_CODEC?
 
Friday September 26 2014
TimeRepliesSubject
4:15PM 0 Ports leak
 
Thursday September 25 2014
TimeRepliesSubject
2:24PM 0 Re: Realtime ERROR
11:32AM 0 weird behaviour of sip history and sip debug
 
Wednesday September 24 2014
TimeRepliesSubject
10:14PM 0 Asterisk 1.8 - Security Fix Only Notice
9:34PM 0 Asterisk 12.6.0 Now Available
9:34PM 0 Asterisk 11.13.0 Now Available
9:34PM 0 Asterisk 1.8.31.0 Now Available
11:06AM 0 Identifying frequency tone in Asterisk
3:38AM 0 AsteriskCDR
 
Tuesday September 23 2014
TimeRepliesSubject
7:17PM 0 Playback/background audio from MySQL BLOB
5:49PM 0 how can queue agents choose which call to answer?
5:30PM 0 Change codec when dial from SIP to DAHDI
5:25PM 0 read digits from the user through php agi script
1:05PM 0 Multicast AMI?
9:30AM 0 Ubuntu 14.04 LTS Asterisk and ISDN Cologne Chip
 
Monday September 22 2014
TimeRepliesSubject
8:08PM 0 DAHDI v2.10.0.1 Fixes loadzone=us ringback tones.
7:24PM 0 [asterisk-announce] DAHDI-Linux and DAHDI-Tools 2.10.0.1 Now Available
3:02PM 0 PRI answer too fast
2:43PM 0 SIPAddHeader from a realtime databse
10:03AM 0 Install Asterisk 1.4 and Asterisk 12.4 on the same machine
 
Sunday September 21 2014
TimeRepliesSubject
11:27PM 0 MixMonitor with b option recording all calls
12:42PM 0 error receiving a fax ... but with a fax that was received without problems
 
Friday September 19 2014
TimeRepliesSubject
9:29PM 0 Asterisk 13.0.0-beta2 Now Available!
11:22AM 0 Show Log(NOTICE) messages on the console
 
Thursday September 18 2014
TimeRepliesSubject
9:43PM 0 Record call ends in 10min
8:16PM 0 conversation record prematurely
7:17PM 0 AST-2014-010: Remote crash when handling out of call message in certain dialplan configurations
7:17PM 0 AST-2014-009: Remote crash based on malformed SIP subscription requests
7:16PM 0 Asterisk 11.6-cert6, 11.12.1, 12.5.1 Now Available (Security Release)
5:35PM 0 Asterisk prefix code to dial a high fraud country - security mechanism
2:36PM 0 Voice-Recognition / ASR / with barge in
1:33PM 0 Asterisk 11.9.0 PRI no ring indications
9:06AM 0 mixmonitor - convert wav to mp3/aac
 
Wednesday September 17 2014
TimeRepliesSubject
11:34AM 0 ${ANSWEREDTIME} returning null
10:12AM 0 GSM to GSM call with callerid passthrough
2:07AM 0 Polycom DND + Intercom/Paging Override?
 
Tuesday September 16 2014
TimeRepliesSubject
4:42PM 0 Disabling CDR for all dialed parties in Asterisk 12
4:03PM 0 Asterisk- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf
 
Monday September 15 2014
TimeRepliesSubject
9:45PM 0 Record ANSWERED call
7:07AM 0 fail2ban and pjsip in asterisk 12 and 13
 
Sunday September 14 2014
TimeRepliesSubject
2:36PM 0 sip.conf and extension.conf configuration
 
Saturday September 13 2014
TimeRepliesSubject
6:04PM 0 NOT able to call on local extensions while successfully call on external mobile no.(using SONETEL account)
6:03PM 0 on reboot /var/run/asterisk owner changed to root.
4:09PM 0 On kernel 3.16.2 : dahdi_rec: Invalid argument
1:44PM 0 how to store sip.conf and extension.conf into phpmyadmin
 
Friday September 12 2014
TimeRepliesSubject
6:00PM 0 compiling Asterisk
2:07PM 0 Call Flow Documentation Tools
9:27AM 0 Tutorial: compiling and installing Asterisk 13
6:43AM 0 Cisco SX20 disconnecting before call
 
Thursday September 11 2014
TimeRepliesSubject
9:43PM 0 if statement recording - after hours
3:46PM 0 ASTERISK AND CHAT MESSAGES
3:14PM 0 chan_sip.c:23647 handle_request_invite: Failed to authenticate device
7:42AM 0 How to use phpmyadmin to remotely access asterisk mysql database?
 
Wednesday September 10 2014
TimeRepliesSubject
8:11PM 0 SIP 380 Alternative Service with PJSIP
7:14PM 0 Ast to Ast TLS trunk
6:29PM 0 WebRTC meeting Norfolk, 15 October 2014
 
Tuesday September 9 2014
TimeRepliesSubject
7:35PM 0 Suspicious routers
2:52PM 0 Segfault Asterisk 1.4.44 in wmvare ESXi 5.5
 
Monday September 8 2014
TimeRepliesSubject
10:39PM 0 Asterisk failed to authenticate device - attack attempt.
8:48PM 0 is pattern matching inside macro valid?
6:31PM 0 adding IAX headers
2:48PM 0 Asterisk removes ice lines in sdp when calling between webrtc clients
3:55AM 0 Call Transfer Fails - Not a Valid Extension
 
Sunday September 7 2014
TimeRepliesSubject
8:41PM 0 Pattern Extension not working in Dialplan
5:00PM 0 Channel h323 and oh323 fails to match inbound IP
12:14PM 0 PJSIP and Multiple transports per endpoint
 
Saturday September 6 2014
TimeRepliesSubject
9:28AM 0 Question about SIP warning
 
Friday September 5 2014
TimeRepliesSubject
1:26PM 0 Re: unidata incom ICW-1000G - On asterisk
9:55AM 0 Asterisk with PJSIP
7:18AM 0 New to Asterisks, Couple of Questions
 
Thursday September 4 2014
TimeRepliesSubject
7:54PM 0 AstriCon Hackathon
4:57PM 0 Special functionality for Secretary/Boss
2:44PM 0 Asterisk secure fine tune - stop attack
1:12PM 0 opus 11.12.0
 
Wednesday September 3 2014
TimeRepliesSubject
6:57PM 0 (no subject)
6:39PM 0 Failover / modifying response time
11:11AM 0 Fwd: Identifying the multiple cards Digium TE820
 
Tuesday September 2 2014
TimeRepliesSubject
8:03AM 0 Custom SIP-header not present in call Asterisk to Asterisk
6:47AM 0 PJSIP issues with handling incoming calls
5:09AM 0 Re: AGI scripts - delay issue.
 
Monday September 1 2014
TimeRepliesSubject
4:46PM 0 Redundancy: PRI Load share +SIP Acctive/Standby
3:27PM 0 Asterisk 11.Why two NOTIFY while ringing ?
1:23PM 0 SIP Calls Not Working
1:11PM 0 Does Asterisk 1.8. Supports Video Calls
12:34PM 0 More XMPP + Asterisk integration: Send a XMPP message to all extensions logged in an Asterisk queue
11:20AM 0 Media update error flooding the console output
10:31AM 0 Setup Own IP PBX Server
6:24AM 0 Asterisk 11.5.0 T38 Faxing