asterisk users - Jun 2014

Monday June 30 2014
TimeRepliesSubject
8:11PM 0 Re: recording in mp3
7:23PM 0 Fwd: Regarding packet2packet bridging
6:46PM 0 Sippeers realtime with minimum table
9:55AM 0 Asterisk 11.10.2 update breaks snom TLS transport
 
Sunday June 29 2014
TimeRepliesSubject
6:47PM 0 Passing parameters to voiceglue.conf
 
Saturday June 28 2014
TimeRepliesSubject
3:17PM 0 200 OK however still rinnging
3:28AM 0 PJSIP endpoint max-calls limit missing
1:04AM 0 Re: Popup URL for outgoing calls.
1:01AM 0 Re: Popup URL for outgoing calls.
 
Friday June 27 2014
TimeRepliesSubject
8:33PM 0 [asterisk-announce] DAHDI-Linux and DAHDI-Tools 2.9.2-rc1 Now Available
7:32PM 0 How to execute an AGI script for each call.
3:46PM 0 Re: AGI script VERBOSE cmd
3:24PM 0 Re: AGI script VERBOSE cmd
2:37PM 0 Attack on Sip server.
9:00AM 0 Early media recognition
 
Thursday June 26 2014
TimeRepliesSubject
8:29PM 0 Originate with Caller ID Name
5:30PM 0 PJSIP Include not working
3:19PM 0 Executing an AGI python script in Asterisk after call is bridged.
2:42PM 0 Changing recorded file storage directory.
2:10PM 0 CLID Presentation & Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info
4:57AM 0 PJSIP Dial via IP fails
 
Wednesday June 25 2014
TimeRepliesSubject
9:38PM 0 Asterisk 12 and chan_local
5:04PM 0 Echo Cancellation when calling from softphone to mobile.
2:30PM 0 OPTIONS Request without username <-> Forbidden
1:11PM 0 Play announcement only once in a Call Queue after 10 seconds
8:40AM 0 Multiple Servers: Multiple Peers: call-limit
 
Tuesday June 24 2014
TimeRepliesSubject
9:37PM 0 share mailbox Asterisk 1.8.22
6:05PM 0 Redfone FoneBridge2 Quad T1/E1 Alternative
9:25AM 0 T1 Card RED ALARM
 
Thursday June 19 2014
TimeRepliesSubject
9:20AM 0 BLF (hints) in Asterisk 12 PJSIP how to?
2:11AM 0 SugarAsterisk vs. ________
 
Wednesday June 18 2014
TimeRepliesSubject
10:39PM 0 Start/Stop recording in confbridge
9:06PM 0 Asterisk and LDAP
6:44PM 0 Making sense of SDP for debugging of missing audio in SIP trunk
5:16PM 0 France DID num2sip setup
11:05AM 0 PJSIP question
 
Tuesday June 17 2014
TimeRepliesSubject
3:29PM 0 DTMF transmitting letter A
1:26PM 0 Request for spandsp paid support
7:04AM 0 quickstart
 
Monday June 16 2014
TimeRepliesSubject
9:03PM 0 libss7 2.0.0 Now Available
9:02PM 0 libpri 1.4.15 Now Available
1:51PM 0 OT - How to configure Apache2 server to receive Polycom log files ?
8:48AM 0 Explicit Call Transfer(ECT)
 
Friday June 13 2014
TimeRepliesSubject
6:15PM 0 pull a call from a queue
2:32PM 0 Need to spoof the callerid using the AMI Originate
6:40AM 0 Asterisk 1.8.15-cert7, 1.8.28.2, 11.6-cert4, 11.10.2, 12.3.2 Now Available (Security/Regression Release)
2:54AM 0 dahdi-linux 2.6.2 failing to compile with linux 3.13
 
Thursday June 12 2014
TimeRepliesSubject
8:46PM 0 AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions
8:45PM 0 AST-2014-007: Exhaustion of Allowed Concurrent HTTP Connections
8:45PM 0 AST-2014-006: Asterisk Manager User Unauthorized Shell Access
8:43PM 0 AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe Framework
8:41PM 0 Asterisk 1.8.15-cert6, 1.8.28.1, 11.6-cert3, 11.10.1, 12.3.1 Now Available (Security Release)
 
Wednesday June 11 2014
TimeRepliesSubject
6:35PM 0 Asterisk 12 AMI Hold Event
5:42PM 0 Hold
7:58AM 0 WSS over Asterisk
 
Tuesday June 10 2014
TimeRepliesSubject
11:14PM 0 Fwd: Asterisk 12.1.1 on Ubuntu 12.04 crashing
9:44PM 0 SSL/TLS weakness impact on Asterisk authentication
2:39PM 0 CDR custom variable on second call leg - via originate or .call file
10:05AM 0 Mixing res_mysql and res_odbc
4:27AM 0 Asterisk realtime peer registration
 
Sunday June 8 2014
TimeRepliesSubject
11:02PM 0 Inter-Digit delay when dialing out
9:01PM 0 SIP Softphone
1:49PM 0 iPhone TLS reg problem: FILE * open failed
 
Saturday June 7 2014
TimeRepliesSubject
5:45PM 0 Re: asterisk-users Digest, Vol 119, Issue 7
 
Friday June 6 2014
TimeRepliesSubject
5:04PM 0 Shorten time between DTMF
1:03PM 0 Problem reload queue dynamical members
7:48AM 0 Using macros in extensions.lua?
 
Thursday June 5 2014
TimeRepliesSubject
10:24PM 0 Using asterisk as voicemail for cisco call manager
6:28PM 0 Outgoing call queues
10:38AM 0 Voicemail Realtime
1:50AM 0 Change time between DTMF
 
Wednesday June 4 2014
TimeRepliesSubject
4:10PM 0 Sip Channel Ring Detect
3:14PM 0 Channel is answered by FXO card before callee answered the phone(pick up phone)
2:47PM 0 Renegotiate SIP audio codec after call is up
 
Tuesday June 3 2014
TimeRepliesSubject
9:31AM 0 Get last dialed number in a context?
 
Monday June 2 2014
TimeRepliesSubject
2:00PM 0 Voicemail Realtime issue "Failed to obtain database object for"
10:30AM 0 H323 Transfer
 
Sunday June 1 2014
TimeRepliesSubject
12:41PM 0 wct4xxp Excessive Interrupts Resulting in Unusable System or Card
10:28AM 0 s4 built in sip client and 481 call/transation does not exist error