asterisk users - Jan 2016

Sunday January 31 2016
TimeRepliesSubject
7:27PM 0 11.21.0 : echo woes : can't install canceller (sean)
6:51PM 0 11.21.0 : echo woes : can't installcanceller (sean darcy)
1:52PM 0 Android native SIP client and 183 (Session Progress) call Declined
 
Friday January 29 2016
TimeRepliesSubject
9:23PM 1 PJSIP RTP Timeout - Calls not ending
8:59PM 1 11.21.0 : echo woes : can't install canceller (sean darcy)
2:47PM 1 Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API
12:15PM 0 Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API
12:11PM 1 PJSIP Stun/ICE
10:46AM 0 asterisk 13 mixmonitor - random missing syllables
10:39AM 2 asterisk 13 mixmonitor - random missing syllables
2:11AM 0 PJSIP Stun/ICE
1:58AM 2 PJSIP Stun/ICE
12:34AM 3 Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API
 
Thursday January 28 2016
TimeRepliesSubject
8:39PM 1 11.21.0 : echo woes : can't install canceller
5:19PM 0 Resource List Subscriptions/BLF List and Aastra phones
2:50PM 0 Caller ID Sent in PAI header.
2:46PM 2 Caller ID Sent in PAI header.
12:37PM 0 asterisk 13 mixmonitor - random missing syllables
9:57AM 2 asterisk 13 mixmonitor - random missing syllables
5:22AM 0 tlsverifyclient=yes option not working
2:10AM 0 Cisco BLF c7975 notifications not working with asterisk realtime
 
Wednesday January 27 2016
TimeRepliesSubject
6:36PM 0 PJSIP Stun/ICE
6:10PM 4 PJSIP Stun/ICE
4:50PM 0 asterisk 13 mixmonitor - random missing syllables
4:42PM 0 Asterisk 13.7.0 AutoMixMonitor
2:45PM 0 Asterisk 13.7.0 losing database connection
1:21PM 2 asterisk 13 mixmonitor - random missing syllables
12:14PM 0 asterisk 13 mixmonitor - random missing syllables
11:59AM 2 asterisk 13 mixmonitor - random missing syllables
2:31AM 1 PJSIP Stun/ICE
1:39AM 0 PJSIP Stun/ICE
1:28AM 4 PJSIP Stun/ICE
 
Tuesday January 26 2016
TimeRepliesSubject
3:39PM 0 PJSIP Stun/ICE
3:14PM 2 PJSIP Stun/ICE
3:02PM 0 PJSIP - Realtime - Transports module?
1:21PM 0 PJSIP Stun/ICE
1:18PM 3 PJSIP Stun/ICE
1:09PM 0 PJSIP Stun/ICE
1:07PM 2 PJSIP Stun/ICE
12:38PM 0 PJSIP Stun/ICE
12:36PM 2 PJSIP Stun/ICE
9:07AM 1 GSM Gateway behind SIP ATA?
 
Monday January 25 2016
TimeRepliesSubject
8:10PM 0 PJSIP NAT traversal.
6:37PM 2 asterisk-users Digest, Vol 138, Issue 19
5:34PM 0 Asterisk 13.7.0 failed to start - PJSIP 2.4.5
9:44AM 1 t.38 fax over IAX2?
 
Saturday January 23 2016
TimeRepliesSubject
1:22AM 1 set framing on dynamic interface DAHDI
 
Friday January 22 2016
TimeRepliesSubject
12:36AM 1 NAME/USERNAME conflict
 
Thursday January 21 2016
TimeRepliesSubject
11:26PM 0 Mixing PJSIP realtime and flat files
11:18PM 2 Mixing PJSIP realtime and flat files
6:14PM 0 is there some blocking in 11.21.0
3:58PM 0 NAME/USERNAME conflict
3:52PM 0 Queue logfile txt format in mySQL needed
3:49PM 2 NAME/USERNAME conflict
3:11PM 4 is there some blocking in 11.21.0
2:02PM 0 is there some blocking in 11.21.0
1:42PM 4 is there some blocking in 11.21.0
10:17AM 2 Queue logfile txt format in mySQL needed
 
Wednesday January 20 2016
TimeRepliesSubject
11:33PM 0 Incoming webrtc call succeeds in Firefox but fails in Google Chrome
9:25PM 2 Incoming webrtc call succeeds in Firefox but fails in Google Chrome
8:46PM 0 PJSIP TLS sometimes RTP, sometimes no RTP
12:50PM 0 488 Not acceptable here
12:19PM 2 488 Not acceptable here
 
Tuesday January 19 2016
TimeRepliesSubject
8:20PM 1 PJSIP TLS sometimes RTP, sometimes no RTP
5:53PM 1 how to flush user input before READ()
5:02PM 0 how to flush user input before READ()
4:22PM 0 Statsd Dialplan Application
2:46PM 2 Statsd Dialplan Application
12:38PM 0 Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)
11:53AM 2 how to flush user input before READ()
 
Monday January 18 2016
TimeRepliesSubject
7:17PM 2 Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)
6:09PM 0 how to flush user input before READ()
5:38PM 2 how to flush user input before READ()
1:36PM 0 How to get PJSIP SIP messages in a log file and not in console ?
1:32PM 0 how to flush user input before READ()
1:22PM 0 best practices - ari reconnect
12:53PM 2 How to get PJSIP SIP messages in a log file and not in console ?
11:57AM 0 Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".
11:40AM 2 Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".
 
Sunday January 17 2016
TimeRepliesSubject
4:19PM 0 asterisk-users Digest, Vol 138, Issue 10
 
Friday January 15 2016
TimeRepliesSubject
9:27PM 0 Asterisk 13.7.0 Now Available
9:22PM 0 Asterisk 11.21.0 Now Available
7:03PM 1 Help me please i am facing much trouble
5:37PM 2 how to flush user input before READ()
 
Thursday January 14 2016
TimeRepliesSubject
9:25PM 0 Digium board considerations
5:53PM 1 X-RTP-Stat SIP header
 
Wednesday January 13 2016
TimeRepliesSubject
7:08PM 0 warble or clicking sound with 11.20.0 with Console/dsp
6:58PM 1 PJSIP Returning 421 Extension Required
6:05PM 1 asterisk-users Digest, Vol 138, Issue 8
2:34PM 0 "pjsip show endpoints" returns "No Objects Found" in 13.7.0-rc2
2:26PM 2 "pjsip show endpoints" returns "No Objects Found" in 13.7.0-rc2
1:48PM 1 cdr_odbc: Error in ExecDirect: -1
 
Monday January 11 2016
TimeRepliesSubject
1:52PM 1 Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11
1:35PM 1 res_pjsip/pjsip_configuration.c: Unable to create ast_sip_contact_status for contact
1:21PM 0 Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11
 
Sunday January 10 2016
TimeRepliesSubject
9:39PM 0 Call Recording
7:55PM 2 Call Recording
 
Friday January 8 2016
TimeRepliesSubject
10:16AM 0 SNMP order of channel types
8:18AM 0 ST2030 replacement
7:46AM 1 ST2030 replacement
 
Thursday January 7 2016
TimeRepliesSubject
11:00PM 0 ST2030 replacement
8:23PM 1 The To header was truncated in call... Whats this means?
4:35PM 5 ST2030 replacement
2:59PM 2 Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11
9:55AM 0 Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11
5:04AM 1 Virtual domain redirects
4:21AM 1 No joy with my first AGI Python script
 
Wednesday January 6 2016
TimeRepliesSubject
8:19PM 0 No joy with my first AGI Python script
5:19PM 2 No joy with my first AGI Python script
4:03PM 2 Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11
10:29AM 1 Getting Asterisk to use the SIP Path header
10:27AM 1 placing calls with linphone.org SIP account
 
Tuesday January 5 2016
TimeRepliesSubject
3:23PM 1 Detected alarm on channel 3: Red Alarm
2:46PM 0 Detected alarm on channel 3: Red Alarm
2:41PM 2 Detected alarm on channel 3: Red Alarm
2:36PM 0 Detected alarm on channel 3: Red Alarm
2:20PM 3 Detected alarm on channel 3: Red Alarm
 
Monday January 4 2016
TimeRepliesSubject
6:31PM 0 Asterisk Behind Firewall
6:15PM 3 Asterisk Behind Firewall
6:10PM 1 Forwarding call if extension busy
3:22PM 0 Forwarding call if extension busy
2:55PM 4 Forwarding call if extension busy