asterisk users - Nov 2013

Saturday November 30 2013
TimeRepliesSubject
6:26AM 2 AGI Script not working
 
Friday November 29 2013
TimeRepliesSubject
5:35PM 2 Answering agent
 
Thursday November 28 2013
TimeRepliesSubject
10:45AM 1 RTP packets send, but no audio
10:10AM 0 Direct Media and message "SIP/SipAgent-00000bf9 requested media update control 26, passing it to SIP/ead14-00000bfb"
 
Wednesday November 27 2013
TimeRepliesSubject
9:57PM 3 issue with speech in IVR
4:31PM 0 SaySentence/SoundPack Proposal
2:12PM 1 Asterisk RTP Questions
11:26AM 2 Asterisk uses 105% CPU
11:08AM 0 Asterisk is delaying DTMF INFO in meetme
 
Tuesday November 26 2013
TimeRepliesSubject
7:35PM 1 Outgoing phone calls muffled
2:22PM 1 Outgoing phone calls "muffled"
 
Monday November 25 2013
TimeRepliesSubject
11:00PM 0 Asterisk 12.0.0-beta2 Now Available!
5:23PM 4 Voicemail greeting playback issues?
4:22PM 1 Asterisk 11.6.0 not starting up
6:36AM 1 terminating the call, when transferer hangs up the call during attended transfer
 
Sunday November 24 2013
TimeRepliesSubject
9:44PM 2 combine external video source and audio call to make SIP video call?
 
Saturday November 23 2013
TimeRepliesSubject
10:11PM 0 how to answer a Panasonic PBX extension with Asterisk?
7:47PM 1 DAHDI Missing '/sys/bus/astribanks/drivers/xppdrv/sync'
1:58PM 0 11.6 voicemail message cropped off?
1:31PM 0 11.6 voicemail message cropped off?
 
Friday November 22 2013
TimeRepliesSubject
7:41PM 1 DAHDI-Linux and DAHDI-Tools 2.8.0-rc2 Now Available
2:35PM 1 Res corosync.
1:50PM 0 Channel not releasing immediately for Attended Transfer
7:54AM 1 Sangoma transcoding card bug - drops audio samples
1:25AM 0 SIP FXS ATA with Gigabit ethernet bridge port,
1:04AM 0 Caller's phone keeps ringing after 200 OK
 
Thursday November 21 2013
TimeRepliesSubject
9:11PM 0 Dialing directly with username and password
7:47PM 0 Monitor extension status
4:04PM 3 Call files without permission for asterisk to read
3:45PM 1 Question about Management Interface
 
Wednesday November 20 2013
TimeRepliesSubject
10:03PM 2 userfield not logged to CDR
7:32PM 5 Movistar sip Mexico
1:06PM 2 Asterisk 1.8.24 : illegal instruction
9:49AM 0 Welcome to the "asterisk-users" mailing list
 
Tuesday November 19 2013
TimeRepliesSubject
6:15PM 0 How to check ISDN / MFCR2 Trunk Status
4:44PM 1 Ast11: How to see call progress like in Ast <= 1.8
7:32AM 2 Communicate with barge agent
6:31AM 0 Redirecting a channel to Meetme fails with Hangup.
12:33AM 1 Amazon, Asterisk and reliability beyond a hobby system?
 
Monday November 18 2013
TimeRepliesSubject
11:23PM 0 Asterisk 10 EOL Approaching
10:20PM 0 app_swift on centos 6 X64
7:21AM 1 CONNECTEDLINE and panasonic 500
5:03AM 1 CEL for attented transfer
 
Sunday November 17 2013
TimeRepliesSubject
7:16PM 0 DTMF relay in meetme is not reliable
7:04AM 2 Bulk forwarding to another Asterisk
 
Saturday November 16 2013
TimeRepliesSubject
11:23AM 3 Make phone ring through webserver using Asterisk
10:54AM 0 Help - DTMF relay in meetme is not reliable
 
Friday November 15 2013
TimeRepliesSubject
12:37PM 0 overlapdialing and no digits in setup problem
 
Thursday November 14 2013
TimeRepliesSubject
11:27PM 1 recieve fax from PRI using spandsp 65% failure rate
10:52PM 1 DAHDI with (CDR(userfield)
6:41PM 1 Queue linear "unordered" feature when using realtime
5:20PM 0 Adding SIP method MESSAGE to Allow header
4:35PM 2 Add SIP Header for 1 SIP peer when calling a group of SIP peers
3:46PM 1 Integration with NEC DSX - help with dial line
5:15AM 1 e1 , hdlc data link?
1:53AM 1 AMI version vs. AST version
 
Wednesday November 13 2013
TimeRepliesSubject
8:35PM 1 SIP Presence across two servers
6:29PM 1 SIP Mass exodus
3:20PM 1 calendar.conf include
12:37AM 2 Recurring SIP problem with asterisk 11.6 & 11.7
 
Tuesday November 12 2013
TimeRepliesSubject
3:18PM 3 VoIP sound quality : highroad sound
1:56PM 1 Asterisk 1.8.20 crashing
 
Monday November 11 2013
TimeRepliesSubject
6:47PM 0 Asterisk Real-time Static Voicemail
7:10AM 2 how determine mandatory modules to slimming asterisk
6:28AM 1 Asterisk Realtime Static Voicemail
5:35AM 0 MCID
 
Friday November 8 2013
TimeRepliesSubject
7:35PM 1 Automated Call Testing - end-to-end - SIP Provider
4:28PM 1 Asterisk 1.8.22
4:27PM 0 T.38 termination
3:39PM 1 11.5.0 - SIP registration not retrying after failures
12:51AM 3 Capture dead phone?
 
Thursday November 7 2013
TimeRepliesSubject
11:20AM 1 Unix connections not always disconnecting
 
Tuesday November 5 2013
TimeRepliesSubject
10:37PM 0 sip show channelstats shows all 0
5:02PM 1 Asterisk 1.4 and DAHDI 2.7
10:09AM 1 How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ?
4:22AM 1 two steps when calling from web!
 
Monday November 4 2013
TimeRepliesSubject
8:24PM 1 CallerID settings
3:25PM 1 No matching peers message has gone (1.8.23.1)
11:01AM 0 set different codec for different sip calls
 
Saturday November 2 2013
TimeRepliesSubject
12:52PM 1 Register Sip extension with out Sip phone
1:51AM 0 Redirect a GSM call through Wifi to a SIPphone
 
Friday November 1 2013
TimeRepliesSubject
4:27PM 1 Redirect a GSM call through Wifi to a SIP phone
10:02AM 1 TE420, is it possible do disable span (red blinking)?