asterisk users - Nov 2014

Friday November 21 2014
TimeRepliesSubject
4:36PM 0 Not able to register an Extension
3:53PM 0 High resident memory with 11.14.0 ?
10:19AM 0 One way audio internal
12:16AM 0 AST-2014-018: AMI permission escalation through DB dialplan function
12:15AM 0 AST-2014-017: <font size="3" style="font-size: 12pt">Permission escalation through ConfBridge actions/dialplan functions</font>
12:15AM 0 AST-2014-016: Remote Crash Vulnerability in PJSIP channel driver
12:15AM 0 AST-2014-015: Remote Crash Vulnerability in PJSIP channel driver
12:15AM 0 AST-2014-014: High call load may result in hung channels in ConfBridge.
12:15AM 0 AST-2014-013: PJSIP ACLs are not loaded on startup
12:14AM 0 AST-2014-012: Mixed IP address families in access control lists may permit unwanted traffic.
12:12AM 0 Asterisk 1.8.28-cert3, 1.8.32.1, 11.6-cert8, 11.14.1, 12.7.1, 13.0.1 Now Available (Security Release)
 
Thursday November 20 2014
TimeRepliesSubject
6:10PM 0 Error saving cdr at h exten in Asterisk13
5:27PM 0 Asterisk13 don't execute h exten inside macros
8:18AM 0 Asterisk problems
 
Wednesday November 19 2014
TimeRepliesSubject
12:36AM 0 Upgraded to 13 and now "Mailbox" is empty in sip show peers
 
Tuesday November 18 2014
TimeRepliesSubject
8:33PM 0 AGI and AMI in PHP -- What's current?
12:37AM 0 google voice
 
Monday November 17 2014
TimeRepliesSubject
10:56PM 0 motif and other xmpp
8:52PM 0 Get the status of a PJSIP endpoint?
2:36PM 0 Code send by asterisk when a peer hangs up
10:42AM 0 OT - Is T.38 possible on SPA8800 FXO port ?
12:36AM 0 from: rschroe@gmail.com
 
Friday November 14 2014
TimeRepliesSubject
3:32PM 0 Asterisk 13 confbridge recordings not working
1:26PM 0 SLA (Shared Line Appearance) and realtime
 
Thursday November 13 2014
TimeRepliesSubject
7:11PM 0 pjsip phoneprov realtime?
2:23PM 0 Erratic calls through NAT-ed server
3:20AM 0 [SOLVED] Re: Incoming calls to a GSM gateway & "SIP/2.0 401 Unauthorized" response when dial 7777 to Asterisk
 
Wednesday November 12 2014
TimeRepliesSubject
10:24PM 0 Como unir webrtc con asterisk???
4:02PM 0 E1 - Cisco - Asterisk and vice verso
3:03PM 0 Asterisk 12 crashes on CANCEL received during ANSWER handlingl
4:45AM 0 Incoming calls to a GSM gateway & "SIP/2.0 401 Unauthorized" response when dial 7777 to Asterisk
3:48AM 0 ITSP Gateway Solution?
 
Tuesday November 11 2014
TimeRepliesSubject
9:51AM 0 odbc connection timeout varable
 
Monday November 10 2014
TimeRepliesSubject
5:12PM 0 Asterisk 12.7.0 Now Available
5:11PM 0 Asterisk 11.14.0 Now Available
5:11PM 0 Asterisk 1.8.32.0 Now Available
3:53PM 0 Webinar GratuĂ­to, Como evitar fraudes em telefonia
9:02AM 0 Subscribe event "ua-profile"
 
Sunday November 9 2014
TimeRepliesSubject
5:21AM 0 One thread per peer
5:12AM 0 Asterisk 12 is broken
3:59AM 0 taskprocessor fails to allocate memory
 
Saturday November 8 2014
TimeRepliesSubject
1:57PM 0 How to find RTP address of ongoing call?
 
Friday November 7 2014
TimeRepliesSubject
4:17PM 0 Asterisk 12 - MGCP realtime gw load
2:17PM 0 detect volume level change on SIP channel
 
Thursday November 6 2014
TimeRepliesSubject
11:16PM 0 Function to get mailbox for a PJSIP Endpoint?
2:05PM 0 Configure Asterisk as SIP UA using NAT
 
Tuesday November 4 2014
TimeRepliesSubject
3:11PM 0 Hangup Chanel when a peer unregisters
2:36PM 0 Asterisk SIP UUI Protocol
10:40AM 0 queue log realtime mysql
 
Monday November 3 2014
TimeRepliesSubject
12:28PM 0 issue with NAT
 
Sunday November 2 2014
TimeRepliesSubject
4:35PM 0 sslv3 alert handshake failure error