asterisk users - Feb 2016

Monday February 29 2016
TimeRepliesSubject
9:04PM 1 PJSIP signaling question
4:52PM 2 Asterisk 13 and WebRTC. Is wiki page still valid ?
3:40PM 0 Zoiper on Windows Phone
12:42AM 0 Crash asterisk res_odbc
 
Sunday February 28 2016
TimeRepliesSubject
8:49PM 0 Crash asterisk res_odbc
6:16PM 0 Handle a call if one phone of a ring, group is busy
3:47AM 1 app_swift crash asterisk 11.20.0-rc1
12:43AM 1 Handle a call if one phone of a ring group is busy
 
Saturday February 27 2016
TimeRepliesSubject
1:01PM 0 Ast 13 always uses slin internally?
 
Thursday February 25 2016
TimeRepliesSubject
10:54PM 0 DAHDI-Linux and DAHDI-Tools 2.11.1-rc1 Now Available
10:23PM 0 11.21,2 : how to transfer to Jolly Roger ?
10:13PM 2 11.21,2 : how to transfer to Jolly Roger ?
4:53PM 0 Queues - periodic announce while ringing members
3:58PM 2 Queues - periodic announce while ringing members
 
Wednesday February 24 2016
TimeRepliesSubject
11:55PM 1 load test docker images?
11:01PM 0 Asterisk 13.6.0: ChannelDtmfReceived message generated twice towards the ARI application
10:49PM 0 load test docker images?
9:55PM 1 Error compiling dahdi on CentOS 7
6:10PM 3 FAX Detection.
4:29PM 4 Crash asterisk res_odbc
 
Tuesday February 23 2016
TimeRepliesSubject
10:05PM 0 Voice recognition IVR Is it possible?
9:56PM 3 Voice recognition IVR Is it possible?
9:01PM 1 pri channels locked
8:33PM 0 pri channels locked
8:04PM 2 pri channels locked
5:20PM 1 Windstream SIP Trunk settings
5:06PM 0 Voice recognition IVR Is it possible?
 
Monday February 22 2016
TimeRepliesSubject
11:58PM 3 Voice recognition IVR Is it possible?
11:43PM 0 Voice recognition IVR Is it possible?
11:39PM 2 Voice recognition IVR Is it possible?
8:17PM 0 Troubles with MessageSend command
8:03PM 1 Voice recognition IVR Is it possible?
7:58PM 0 Voice recognition IVR Is it possible?
7:56PM 0 Voice recognition IVR Is it possible?
7:53PM 2 Voice recognition IVR Is it possible?
6:40PM 0 Voice recognition IVR Is it possible?
6:34PM 5 Voice recognition IVR Is it possible?
6:17PM 0 Voice recognition IVR Is it possible?
6:00PM 2 Voice recognition IVR Is it possible?
2:06PM 0 Windstream SIP Trunk settings
1:20PM 4 Windstream SIP Trunk settings
 
Friday February 19 2016
TimeRepliesSubject
10:24PM 0 Grandstream Early Dial
5:14PM 1 Passing Caller ID through Digium Gateway
4:52PM 2 Grandstream Early Dial
4:32PM 4 load test docker images?
1:31PM 0 Grandstream Early Dial
1:04PM 1 How to execute a macro after dial but before connect
11:01AM 0 Asterisk 13 and WebRTC. Is wiki page still valid ?
6:37AM 0 TDMoE with wmware
3:44AM 0 No matching endpoint found for incoming call from SIP trunk
3:20AM 2 No matching endpoint found for incoming call from SIP trunk
2:56AM 0 No matching endpoint found for incoming call from SIP trunk
2:25AM 2 No matching endpoint found for incoming call from SIP trunk
1:02AM 2 Grandstream Early Dial
 
Thursday February 18 2016
TimeRepliesSubject
10:54PM 0 Planned maintenance for community services Thursday night, February 18th 2016
9:52PM 1 1000 analogue lines with asterisk
9:42PM 0 Asterisk behind RTPproxy | On-Demand SDP engagement
9:05PM 2 Asterisk behind RTPproxy | On-Demand SDP engagement
9:03PM 0 Grandstream Early Dial
8:42PM 2 Grandstream Early Dial
3:09PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
3:01PM 2 Asterisk 13 and WebRTC. Is wiki page still valid ?
2:42PM 0 Asterisk 13 and WebRTC. Is wiki page still valid ?
2:36PM 2 Asterisk 13 and WebRTC. Is wiki page still valid ?
1:57PM 0 Asterisk 13 and WebRTC. Is wiki page still valid ?
1:43PM 2 Asterisk 13 and WebRTC. Is wiki page still valid ?
12:41PM 0 Blocking transfer by SIP REFER on a call by call basis
12:30PM 0 Asterisk 13 and WebRTC. Is wiki page still valid ?
11:29AM 1 Typo in http.conf sample file ?
11:10AM 2 Asterisk 13 and WebRTC. Is wiki page still valid ?
2:38AM 0 1000 analogue lines with asterisk
12:28AM 0 Problem compiling res_fax_spandsp.c on Debian server.
 
Wednesday February 17 2016
TimeRepliesSubject
11:56PM 2 Problem compiling res_fax_spandsp.c on Debian server.
11:39PM 2 1000 analogue lines with asterisk
11:32PM 0 Problem compiling res_fax_spandsp.c on Debian server.
11:15PM 2 Problem compiling res_fax_spandsp.c on Debian server.
10:02PM 1 res_pjsip trunk between Asterisk servers
8:48PM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
7:13PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
5:43PM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
3:56PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
1:38PM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
1:36PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
1:31PM 3 Asterisk 13.6.0/The simplest TCP configuration does not work
1:28PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
1:23PM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
1:20PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
1:15PM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
1:13PM 0 SIP URI set 'telephone-context='
1:01PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
12:57PM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
12:55PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
11:50AM 2 SIP URI set 'telephone-context='
11:37AM 0 SIP URI set 'telephone-context='
11:35AM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
10:18AM 0 siemens openstage provisioning
9:22AM 0 1000 analogue lines with asterisk
8:27AM 0 1000 analogue lines with asterisk
8:09AM 2 1000 analogue lines with asterisk
7:32AM 0 1000 analogue lines with asterisk
7:16AM 2 1000 analogue lines with asterisk
7:14AM 0 1000 analogue lines with asterisk
7:12AM 2 1000 analogue lines with asterisk
7:07AM 0 1000 analogue lines with asterisk
7:02AM 5 1000 analogue lines with asterisk
3:08AM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
 
Tuesday February 16 2016
TimeRepliesSubject
8:03PM 2 SIP URI set 'telephone-context='
6:03PM 0 SIP URI set 'telephone-context='
5:02PM 2 SIP URI set 'telephone-context='
12:12PM 0 Voicemail using object storage?
12:05AM 2 Voicemail using object storage?
 
Monday February 15 2016
TimeRepliesSubject
11:01PM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
10:31PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
10:29PM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
8:06PM 1 Multiple protocols for transport in PJSIP
7:53PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
7:22PM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
7:08PM 0 Multiple protocols for transport in PJSIP
6:58PM 2 Multiple protocols for transport in PJSIP
6:50PM 0 Multiple protocols for transport in PJSIP
6:48PM 2 Multiple protocols for transport in PJSIP
6:40PM 0 Asterisk 13.6.0/The simplest TCP configuration does not work
6:37PM 2 Asterisk 13.6.0/The simplest TCP configuration does not work
3:28PM 0 Error making dahdi linux compete 2.11.0
2:15PM 2 Error making dahdi linux compete 2.11.0
 
Sunday February 14 2016
TimeRepliesSubject
9:06PM 0 Determining and setting TLS cipher ?
 
Friday February 12 2016
TimeRepliesSubject
6:17PM 0 [dongle0] timedout while waiting 'OK' in response to 'AT'
5:39PM 2 [dongle0] timedout while waiting 'OK' in response to 'AT'
5:12PM 0 [dongle0] timedout while waiting 'OK' in response to 'AT'
4:35PM 0 [dongle0] timedout while waiting 'OK' in response to 'AT'
4:33PM 4 [dongle0] timedout while waiting 'OK' in response to 'AT'
4:31PM 0 [dongle0] timedout while waiting 'OK' in response to 'AT'
4:29PM 2 [dongle0] timedout while waiting 'OK' in response to 'AT'
4:27PM 0 [dongle0] timedout while waiting 'OK' in response to 'AT'
4:01PM 1 NAT on IPsec Tunnel
3:51PM 2 [dongle0] timedout while waiting 'OK' in response to 'AT'
 
Thursday February 11 2016
TimeRepliesSubject
9:41PM 0 Asterisk 11.21.2 Now Available
8:19PM 0 dahdi complete 2.11.0 on linux 4.4.0
6:36PM 0 res_odbc crashes asterisk
6:30PM 3 res_odbc crashes asterisk
3:50PM 0 Unexpected termination of the call when pick up (res_pjsip)
3:40PM 0 Best place to issue tickets for Digium phones ?
1:56PM 1 WhatsApp VoIP in Asterisk integration?
1:02PM 0 CDR ODBC error
10:08AM 0 D70 phone dials 800 when pressing Msgs button. How to change that ?
8:56AM 1 Ignoring audio media offer because port number is zero
7:32AM 3 Unexpected termination of the call when pick up (res_pjsip)
 
Wednesday February 10 2016
TimeRepliesSubject
10:47PM 0 Unexpected termination of the call when pick up (res_pjsip)
9:20PM 2 Unexpected termination of the call when pick up (res_pjsip)
9:09PM 0 looking for soft phone can be manged like Snom phones
2:20PM 2 Authenticate() 11.21.0
2:11PM 2 Best place to issue tickets for Digium phones ?
 
Tuesday February 9 2016
TimeRepliesSubject
10:39PM 2 CDR ODBC error
5:57PM 2 Voicemail issue on Grandstream GXP2000 phones
9:25AM 0 pjsip extension state on outgoing calls
3:08AM 0 res_pjsip trunk between Asterisk servers
2:16AM 2 res_pjsip trunk between Asterisk servers
 
Monday February 8 2016
TimeRepliesSubject
11:03PM 0 Asterisk 13 realtime static not working
9:22PM 0 Class 5 and softphone app supporting ZRTP
3:10PM 0 sql schema without alembic
9:54AM 2 sql schema without alembic
9:03AM 0 Delayed start of video with WebRTC - Missed FIR due to DTLS?
 
Sunday February 7 2016
TimeRepliesSubject
3:32PM 1 Nube question: where is chan_sip.so?
2:55PM 0 Nube question: where is chan_sip.so?
10:29AM 5 Nube question: where is chan_sip.so?
 
Friday February 5 2016
TimeRepliesSubject
10:50PM 0 Asterisk 13.7.2 Now Available
1:09PM 0 Phone audio sound routing through workstation audio ports
12:50PM 1 Panic Button SMS Asterisk Integration
9:30AM 0 [SOLVED] Re: How to simulate 100 phones in a lab ?
8:39AM 0 Asterisk & Docker
12:44AM 1 NAT traversal for mobile app softphones - best strategy?
 
Thursday February 4 2016
TimeRepliesSubject
9:57PM 0 include => parkedcalls but nonexistent context 'parkedcalls'
5:55PM 2 include => parkedcalls but nonexistent context 'parkedcalls'
2:10PM 0 What is SIP Early Media useful for ?
1:00PM 0 Peer Reachable / Unreachable on TLS
11:19AM 1 How to simulate 100 phones in a lab ?
11:17AM 0 sql schema without alembic
10:31AM 2 sql schema without alembic
10:05AM 1 missing https://github.com/asterisk/asterisk/blob/13.7/asterisk-13.7.0-summary
8:26AM 0 Call hangup on transfer when originated from a Queue
1:59AM 0 AST-2016-003: Remote crash vulnerability when receiving UDPTL FAX data.
1:59AM 0 AST-2016-002: File descriptor exhaustion in chan_sip
1:59AM 0 AST-2016-001: BEAST vulnerability in HTTP server
1:56AM 0 Asterisk 11.6-cert12, 11.21.1, 13.1-cert3, 13.7.1 Now Available (Security Release)
 
Wednesday February 3 2016
TimeRepliesSubject
9:12PM 1 How to deal with error messages passed as Early Media
7:42PM 0 include => parkedcalls but nonexistent context 'parkedcalls'
7:32PM 2 include => parkedcalls but nonexistent context 'parkedcalls'
7:27PM 0 include => parkedcalls but nonexistent context 'parkedcalls'
7:19PM 2 include => parkedcalls but nonexistent context 'parkedcalls'
7:15PM 0 include => parkedcalls but nonexistent context 'parkedcalls'
7:05PM 2 include => parkedcalls but nonexistent context 'parkedcalls'
5:00PM 0 How to deal with error messages passed as Early Media
3:29PM 1 How to deal with error messages passed as Early Media
2:59PM 0 How to deal with error messages passed as Early Media
2:56PM 2 What is SIP Early Media useful for ?
2:41PM 4 How to deal with error messages passed as Early Media
7:43AM 1 Dial command: channel type detection
 
Tuesday February 2 2016
TimeRepliesSubject
8:56PM 0 Asterisk not matching peer of incoming call
6:11PM 0 Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup
5:32PM 2 Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup
4:57PM 1 Compile error with libpri 1.4.15
3:38PM 0 Compile error with libpri 1.4.15
3:07PM 2 dahdi on systemd (CentOS 7)
2:58PM 4 Compile error with libpri 1.4.15
 
Monday February 1 2016
TimeRepliesSubject
6:49PM 0 11.21.0 : echo woes : can't installcanceller (sean darcy)