asterisk users - Sep 2012

Sunday September 30 2012
TimeRepliesSubject
4:09PM 0 asterisk 1.8 realtime queue_log
7:45AM 0 callerid channel variable vs. CDR(src)?
7:33AM 0 ASTERISK VOICEMAIL NOTIFICATION TOM MOBILE
 
Saturday September 29 2012
TimeRepliesSubject
6:17PM 1 Who said asterisk is not to the task
5:08PM 4 AsteriskNOW x86_64 install GPT partitions
2:27PM 0 Extension hints, which info available?
1:56PM 3 Remote SIP Extension Best Practices
9:32AM 9 Reuse h extension?
12:24AM 0 Strategy for custom data in the CDR
 
Friday September 28 2012
TimeRepliesSubject
9:27PM 1 'Training mode'
4:42PM 9 "Call me now" outbound calls in a queue
2:17PM 0 Call Hold problem
12:23PM 3 User expected behavior of musiconhold and AGI's "stream file"
10:50AM 9 ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
8:33AM 3 RealTime table fields ordering
8:22AM 6 Disconnect calls : known reasons
1:01AM 9 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
 
Thursday September 27 2012
TimeRepliesSubject
9:00AM 7 realtime_multi_mysql: MySQL RealTime: Invalid database specified
1:25AM 2 Paetec SIP Trunk
 
Wednesday September 26 2012
TimeRepliesSubject
9:09PM 10 QUEUEHOLDTIME always zero
6:54PM 1 SIP DTMF Flash Event
3:53PM 5 FAX via Asterisk
3:32PM 6 Asterisk and History-Info
3:16PM 0 OT; What happen with voipuser.org ?
2:35PM 5 Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
1:49PM 2 Asterisk call forward to mobile numbers if ringroup is not picking up the call
12:44PM 9 SIP Retransmitting REGISTER message
9:35AM 7 PLAYIN MUSIC WHILE SEARCHING MYSQL
5:01AM 0 asterisk-users Digest, Vol 98, Issue 38
 
Tuesday September 25 2012
TimeRepliesSubject
9:23PM 2 no audio while call forwarding, yes audio with followme
9:15PM 11 MySQL InnoDB or MyISAM for CDR
8:00PM 2 asterisk ip authentication
3:03PM 3 is silk included in asterisk 11?
3:03PM 0 Question about async channel or macro for monitoring a call
2:18PM 1 confbridge video support
11:57AM 1 white noise on conference
6:56AM 2 undefined symbols
6:35AM 0 context local: unexpected KW _LOCAL
5:01AM 0 OpenVox A400P+asterisk=problem with dial recall tone
1:18AM 15 T.38 gateway ATA
 
Monday September 24 2012
TimeRepliesSubject
9:37PM 5 Realtime Hints
7:12PM 1 confbridge command not found
6:55PM 1 Asterisk as a E1/T1 FXO ?
4:35PM 8 Dial plan order of operations
11:47AM 2 Help Required IVR
11:00AM 4 Peculiar problem with failover provision.
6:10AM 6 CDR Unanswered calls
 
Sunday September 23 2012
TimeRepliesSubject
10:17PM 1 How to get SIP Response Code and use it to change destination.
3:31PM 18 Issue with PRI connection
 
Saturday September 22 2012
TimeRepliesSubject
12:47PM 1 Grandstream VoIP phones
 
Friday September 21 2012
TimeRepliesSubject
2:34PM 0 attended transfer caller hears ringing after transfer done
 
Thursday September 20 2012
TimeRepliesSubject
7:59PM 2 XMPP sendtodialplan
7:11PM 5 Hand Raise|Meetme Conf
6:19PM 0 Problem with macros in AEL
5:31PM 8 accept email and make phone call?
4:40PM 2 Asterisk 11.0.0-beta2 Now Available!
12:37PM 8 1.4.43 lost part of dialplan
8:49AM 2 Need to Help for setup Video IVRS on Asterisk
7:39AM 1 chan_motif, xmpp, jabber, jingle
7:28AM 3 Voicemail not working with vm boxes named with a star
4:22AM 1 TLS/SRTP - Cisco SPA-301
 
Wednesday September 19 2012
TimeRepliesSubject
7:05PM 1 Two IP addresses and a SIP Trunk
3:51PM 3 SIP CANCEL, Reason
11:21AM 0 Bridge
11:20AM 0 Manager Bridge
9:39AM 2 SRTP & asterisk 1.8.x & SNOM
 
Tuesday September 18 2012
TimeRepliesSubject
7:41PM 8 Trigger Asterisk after data inserted in mysql
3:36PM 0 Trunk SCCP
2:47PM 1 Any workaround for res_speech_lumenvox.so issue?
7:04AM 2 chan_mobile
6:45AM 4 Hangup not detected
 
Monday September 17 2012
TimeRepliesSubject
9:39PM 2 inboun routing based on area aode
5:22PM 8 AGI HANGUP PROBLEM
2:36PM 0 AGI problem
4:15AM 4 $agi->hangup() Does not hang up the channel
1:47AM 2 iax2 trunks between asterisk servers
 
Friday September 14 2012
TimeRepliesSubject
8:44PM 17 DTMF digits falsely detected
6:56PM 2 opus codec
4:16PM 9 MySQL Query : Calls Answered for < 5 sec
6:46AM 0 Trevlig lampa :)
4:29AM 4 Need to record user voice while play background music
3:26AM 8 Digium AEX410, MTNL Mumbai Caller-ID problems
 
Thursday September 13 2012
TimeRepliesSubject
11:19PM 1 Voice Mail message should transfer to email address
9:18PM 0 Asterisk 10.8.0 Now Available
9:18PM 0 Asterisk 1.8.16.0 Now Available
9:06PM 0 alsa channel
8:52PM 3 Trunk Config
2:20PM 1 Unplanned community service outage Sept 13th 2012
12:13PM 0 Volume issue.
11:06AM 2 Asterisk Streaming MeetMe Conference
9:40AM 2 Trouble phoning via HUAWEI E169
7:47AM 0 Questions about fax detection
3:52AM 8 Asterisk on VM with NO DAHDI hardware
 
Wednesday September 12 2012
TimeRepliesSubject
10:01PM 2 Fax and sending to mail
3:19PM 20 kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
12:25PM 0 Asterisk in the London Olympics
12:12PM 0 deadagi
6:33AM 0 [asterisk-user] INTERNAL_OBJ error in asterisk 1.8.13
 
Tuesday September 11 2012
TimeRepliesSubject
9:53PM 4 multiple users for jabber.conf
7:08PM 2 asterisk boxes looses registration
6:53PM 0 Linebreaks in cdr_custom.conf / cdr_sqlite3_custom.conf
5:06PM 6 Asterisk HangUp not breaking incoming call for caller
1:58PM 0 codec priorities
 
Monday September 10 2012
TimeRepliesSubject
4:58PM 2 Maximum messages in voicemail
4:13PM 1 MixMonitor inserting extra 20ms packets of silence (1.4.43)
1:19PM 1 Asterisk crashing when recording ConfBridge calls (10.7.1)
12:47PM 0 Queue and reinvite
8:33AM 0 Asterisk as a translating proxy only?
5:01AM 0 ConfBridge announce_join_leave custom recording?
 
Sunday September 9 2012
TimeRepliesSubject
2:27PM 0 MusicOnHold (stream) interrupted after DDoS / network issues
 
Saturday September 8 2012
TimeRepliesSubject
6:12PM 0 Calls from talkonaut to pstn Phone
8:49AM 2 how to load our own .wav sound files in the dial plans for playback
 
Friday September 7 2012
TimeRepliesSubject
6:49PM 0 Rolm T1 not passing caller ID to asterisk
5:17PM 0 Automatic reconnect to ODBC sources?
6:49AM 3 AMI Permissions, "all" means different things?
 
Thursday September 6 2012
TimeRepliesSubject
6:19PM 0 OSS booths at AstriCon
2:46PM 1 Polycom Phone Configuration Overrides Not Saved
7:22AM 3 Asterisk Test Suite error
 
Wednesday September 5 2012
TimeRepliesSubject
4:25PM 9 Async AGI
9:30AM 6 Help with GotoIf Command
1:23AM 0 Responsibility for res_speech_lumenvox.so
 
Tuesday September 4 2012
TimeRepliesSubject
10:56PM 3 Repeated Asterisk 10.7.0 crashes
12:34PM 0 One-way audio with media_address
8:54AM 1 CDR Issue
6:56AM 0 Asterisk outbound register messages causes firewall issues
6:30AM 1 Interrupt error
12:25AM 1 Indicate multiple incoming calls from a multi-channel DID on a single phone
 
Monday September 3 2012
TimeRepliesSubject
9:09PM 1 Cascading macros in 1.8. Bug or feature ?
1:49PM 0 new HOWTO - Lumicall/Android with an Asterisk PBX
11:46AM 1 Asterisk 10 deb packages for Ubuntu 12.04?
11:42AM 0 dtmf problem
11:25AM 2 Asterisk 11 WebSockets.
 
Sunday September 2 2012
TimeRepliesSubject
2:28PM 4 My digium card die?
 
Saturday September 1 2012
TimeRepliesSubject
11:29AM 1 any good way to reload "realtime" configuration