asterisk users - Jul 2016

Sunday July 31 2016
TimeRepliesSubject
3:42PM 0 Removing mailbox and password prompt for voicemail
5:19AM 3 Removing mailbox and password prompt for voicemail
 
Saturday July 30 2016
TimeRepliesSubject
6:32PM 0 Removing mailbox and password prompt for voicemail
6:27PM 0 Calls are dropped after 15 minutes
6:07PM 5 Calls are dropped after 15 minutes
2:52PM 3 Removing mailbox and password prompt for voicemail
2:18PM 0 Removing mailbox and password prompt for voicemail
6:01AM 1 Remove 'Comedian Mail' Message
5:43AM 4 Removing mailbox and password prompt for voicemail
 
Friday July 29 2016
TimeRepliesSubject
12:55PM 0 WhatsApp feature on Asterisk
9:39AM 0 AsteriskNow and Hyper-V
9:15AM 2 WhatsApp feature on Asterisk
 
Wednesday July 27 2016
TimeRepliesSubject
9:02PM 3 Asterisk 14.0.0-beta1 Now Available
11:59AM 0 Identify endpoint based on Diversion header
11:56AM 2 Identify endpoint based on Diversion header
 
Tuesday July 26 2016
TimeRepliesSubject
5:54PM 0 Yealink T21P E2 bug solved
2:39PM 3 SIP trunk
1:15PM 1 PJSIP - Video Support for WebRTC
6:34AM 0 __sip_xmit Returned -1 Invalid Argument
 
Sunday July 24 2016
TimeRepliesSubject
5:39PM 1 Asterisk 13 High CPU usage
12:39PM 0 Asterisk 13 High CPU usage
 
Friday July 22 2016
TimeRepliesSubject
6:55AM 0 1.8.32.3 - billsec field does not not increment after call answer - what triggers it? (Joshua Colp)
 
Thursday July 21 2016
TimeRepliesSubject
11:02PM 3 Asterisk 13 High CPU usage
9:53PM 0 VoiceMail - Allow * for only some users
9:23PM 2 VoiceMail - Allow * for only some users
9:18PM 0 Asterisk 13.10.0 Now Available
5:38PM 2 Asterisk 13.10.0 Now Available
5:36PM 0 Asterisk 11.23.0 Now Available
4:26PM 0 Is it possible to change the default format for ConfBridge recordings?
 
Wednesday July 20 2016
TimeRepliesSubject
5:14PM 1 No Sangoma ISDN BRI cards detected by goautodial
3:58PM 0 PJSIP_DIAL_CONTACTS issue
3:32PM 0 No Sangoma ISDN BRI cards detected by goautodial
3:05PM 3 No Sangoma ISDN BRI cards detected by goautodial
2:58PM 3 PJSIP_DIAL_CONTACTS issue
11:47AM 1 PJSIP and the pound (#) as %23
10:03AM 1 1.8.32.3 - billsec field does not increment after call answer - what triggers it?
 
Sunday July 17 2016
TimeRepliesSubject
9:19PM 0 PJSIP - State of the art
12:30PM 3 PJSIP - State of the art
 
Friday July 15 2016
TimeRepliesSubject
4:01PM 1 PlaySMS with Chan Dongle?
3:34PM 0 PlaySMS with Chan Dongle?
3:29PM 2 VoiceMail and SMS
3:00PM 2 PlaySMS with Chan Dongle?
12:52PM 0 VoiceMail Audio playing
12:14PM 0 VoiceMail Audio playing
12:06PM 4 VoiceMail Audio playing
12:00PM 0 VoiceMail Audio playing
9:37AM 2 VoiceMail Audio playing
9:29AM 0 play sound without stop it when dtmf is pressed
 
Thursday July 14 2016
TimeRepliesSubject
11:40PM 0 ODBC freezing Asterisk 13
11:26PM 2 ODBC freezing Asterisk 13
9:42PM 0 1 way audio but audio+video is fine
6:42PM 0 Asterisk and Yealink T21P E2
6:14PM 2 Asterisk and Yealink T21P E2
5:06PM 0 Voicemail Mailboxes + Cassandra
1:52PM 0 CDR replacement with CEL
12:45PM 1 PJSIP defaults for endpoints when using realtime
12:21PM 0 PJSIP defaults for endpoints when using realtime
12:09AM 1 Compile of smsq.c failed on Ubuntu Xenial (16.04LTS)
 
Wednesday July 13 2016
TimeRepliesSubject
9:49PM 3 PJSIP defaults for endpoints when using realtime
9:44PM 1 Asterisk 13 MWI
5:37PM 0 Certified Asterisk 13.8-cert1 Now Available
10:22AM 1 Force out-bond call to specific CIC
2:27AM 0 Asterisk 13 MWI
 
Tuesday July 12 2016
TimeRepliesSubject
5:55PM 2 Asterisk 13 MWI
 
Saturday July 9 2016
TimeRepliesSubject
3:37PM 1 dtls_cipher
 
Friday July 8 2016
TimeRepliesSubject
11:22AM 0 need help
 
Thursday July 7 2016
TimeRepliesSubject
8:28PM 0 Getting better Caller ID
8:25PM 2 Getting better Caller ID
8:23PM 0 First call - one way audio
 
Wednesday July 6 2016
TimeRepliesSubject
9:09PM 0 Impossible to use any recent asterisk version with chan_sip
8:48PM 3 Impossible to use any recent asterisk version with chan_sip
7:46PM 0 Impossible to use any recent asterisk version with chan_sip
7:33PM 2 how to read sip debug
7:04PM 0 how to read sip debug
6:43PM 2 how to read sip debug
6:18PM 0 Compiler errors when 'make asterisk' for D100 transcoding board
6:10PM 0 Identify more demanding routine inside Asterisk
3:18PM 1 Impossible to use any recent asterisk version with chan_sip
3:14PM 0 Impossible to use any recent asterisk version with chan_sip
3:10PM 4 Impossible to use any recent asterisk version with chan_sip
12:36PM 1 rasberry pi
12:13PM 0 rasberry pi
11:10AM 0 OpenSIPS or Kamailio based fronting for Asterisk?
10:38AM 0 rasberry pi
9:05AM 0 Function SHELL not registered
8:51AM 3 rasberry pi
8:36AM 0 rasberry pi
8:12AM 1 what is a SIP invite, and who can issue them?
8:10AM 5 rasberry pi
2:28AM 1 how to read sip debug
 
Tuesday July 5 2016
TimeRepliesSubject
4:18PM 2 OpenSIPS or Kamailio based fronting for Asterisk?
3:40PM 4 Function SHELL not registered
2:27PM 0 Function SHELL not registered
2:09PM 0 Some SIP OPTIONS packages seem to be ignored by the peer
2:06PM 2 Function SHELL not registered
 
Monday July 4 2016
TimeRepliesSubject
11:18AM 0 CALLERID on pjsip doesn't work?
7:34AM 0 Function SHELL not registered
6:36AM 2 Function SHELL not registered
2:20AM 2 CALLERID on pjsip doesn't work?
1:37AM 0 Please help me understand lines and extensions little better
 
Saturday July 2 2016
TimeRepliesSubject
8:16PM 0 Registration server with PJSIP
6:59PM 3 Registration server with PJSIP
4:06PM 0 AstriCon 2016 - crowdfunding for participation
 
Friday July 1 2016
TimeRepliesSubject
9:41AM 0 CALLERID on pjsip doesn't work?
6:33AM 2 CALLERID on pjsip doesn't work?
2:25AM 1 how to join 2 channels using AGI/AMI