asterisk users - Jan 2017

Monday January 30 2017
TimeRepliesSubject
7:52PM 1 PJSIP Real-time Text (T.140)
5:13PM 0 Asterisk 13.13.1
5:04PM 2 Asterisk 13.13.1
4:55PM 0 Asterisk 13.13.1
 
Saturday January 28 2017
TimeRepliesSubject
10:45PM 1 packet loss stats - how does asterisk know about packets sent % lost ?
1:12PM 4 Asterisk 13.13.1
 
Friday January 27 2017
TimeRepliesSubject
9:40PM 0 libpri 1.6.0 Now Available
6:58PM 1 semi-OFF-TOPIC - SIP iptables and NAT - same source, different destination
2:30PM 1 Callback on busy
 
Thursday January 26 2017
TimeRepliesSubject
2:34PM 0 Spandsp updated
11:03AM 0 asterisk-users Digest, Vol 150, Issue 17
 
Wednesday January 25 2017
TimeRepliesSubject
12:30PM 0 Asterisk 13.13.1
 
Tuesday January 24 2017
TimeRepliesSubject
8:03PM 2 Asterisk 13.13.1
5:41PM 1 Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?
5:27PM 0 Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?
5:25PM 2 Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?
11:46AM 2 Setup DID
8:36AM 0 Understanding how LLDP works with DHCP [SOLVED]
 
Monday January 23 2017
TimeRepliesSubject
9:26PM 0 Attended Transfer using AMI on PJSIP
 
Thursday January 19 2017
TimeRepliesSubject
10:09AM 1 Understanding how LLDP works with DHCP
 
Wednesday January 18 2017
TimeRepliesSubject
7:51PM 1 Developing Asterisk Modules
6:35PM 1 How to send SIP_NOTIFY messages with variable content ?
3:13PM 0 How to send SIP_NOTIFY messages with variable content ?
2:58PM 1 How to send SIP_NOTIFY messages with variable content ?
1:54PM 0 How to send SIP_NOTIFY messages with variable content ?
 
Tuesday January 17 2017
TimeRepliesSubject
2:36PM 2 How to send SIP_NOTIFY messages with variable content ?
2:00PM 0 Comunicado Importante!
1:20PM 0 pcapsipdump or general sip debug question - the solution
11:34AM 2 pcapsipdump or general sip debug question
11:27AM 0 pcapsipdump or general sip debug question
12:53AM 0 Advice of Charge for non-Snom SIP phones
 
Monday January 16 2017
TimeRepliesSubject
7:08PM 0 Kernel/Asterisk/DAHDI/Libpri version matrix?
5:53PM 3 Kernel/Asterisk/DAHDI/Libpri version matrix?
1:25PM 0 How to send SIP_NOTIFY messages with variable content ?
1:10PM 4 How to send SIP_NOTIFY messages with variable content ?
 
Friday January 13 2017
TimeRepliesSubject
7:04PM 1 Asterisk - Vtiger integration
 
Thursday January 12 2017
TimeRepliesSubject
5:19PM 0 Replacing PBX during a call in progress
5:05PM 1 Replacing PBX during a call in progress
5:03PM 0 Replacing PBX during a call in progress
4:09PM 5 Replacing PBX during a call in progress
 
Wednesday January 11 2017
TimeRepliesSubject
9:47PM 0 256 bit SRTP ciphers in Asterisk 14.x , only works for outbound call ?
4:59PM 1 PJSIP status check at DB level
3:26PM 0 hangup locked channels
3:09PM 1 sip:ping@noname.com
12:31PM 0 Dial() from the console?
12:20PM 3 Dial() from the console?
10:32AM 2 sip show [general]?
2:42AM 0 Custom INFO for Advice Of Charge
 
Tuesday January 10 2017
TimeRepliesSubject
2:11PM 1 PJSIP status check at DB level (Realtime)
1:24PM 0 Can't comile bundled PJSIP on CentOS 7
9:58AM 2 Can't comile bundled PJSIP on CentOS 7
9:31AM 0 Can't comile bundled PJSIP on CentOS 7
4:30AM 0 Can't comile bundled PJSIP on CentOS 7
12:15AM 6 Can't comile bundled PJSIP on CentOS 7
 
Sunday January 8 2017
TimeRepliesSubject
6:47PM 0 Connection dropped after 15 minutes with Deutsche Telekom
 
Friday January 6 2017
TimeRepliesSubject
6:47PM 0 Issue with handling of 480 DND
6:35PM 0 Thank you Asterisk community!
11:07AM 3 Issue with handling of 480 DND
 
Wednesday January 4 2017
TimeRepliesSubject
8:44PM 1 T1 -Asterisk server - Analog lines
2:29PM 1 TLS certificate warnings in softphone, but not until after successful registration and call placed ?
 
Tuesday January 3 2017
TimeRepliesSubject
6:13PM 1 Saving endpoint statuses to database with pjsip and realtime
6:12PM 0 Saving endpoint statuses to database with pjsip and realtime
6:06PM 0 TLS certificate warnings in softphone, but not until after successful registration and call placed ?
5:58PM 0 183 Session in Progress. Disconnecting channel for lack of RTP activity
4:32PM 1 how to add area code to outgoing number in Asterisk 13.13
3:17PM 0 Does HEP require PJSIP or does it also works with SIP ?
3:04PM 3 Does HEP require PJSIP or does it also works with SIP ?
 
Monday January 2 2017
TimeRepliesSubject
12:26PM 1 anveo, a different kind of trunk provider?