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asterisk users
79364 threads
Jan 2017
68 threads
Monday January 30 2017
Time
Replies
Subject
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
Time
Replies
Subject
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
Time
Replies
Subject
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
Time
Replies
Subject
2:34PM
0
Spandsp updated
11:03AM
0
asterisk-users Digest, Vol 150, Issue 17
Wednesday January 25 2017
Time
Replies
Subject
12:30PM
0
Asterisk 13.13.1
Tuesday January 24 2017
Time
Replies
Subject
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
Time
Replies
Subject
9:26PM
0
Attended Transfer using AMI on PJSIP
Thursday January 19 2017
Time
Replies
Subject
10:09AM
1
Understanding how LLDP works with DHCP
Wednesday January 18 2017
Time
Replies
Subject
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
Time
Replies
Subject
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
Time
Replies
Subject
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
Time
Replies
Subject
7:04PM
1
Asterisk - Vtiger integration
Thursday January 12 2017
Time
Replies
Subject
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
Time
Replies
Subject
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
Time
Replies
Subject
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
Time
Replies
Subject
6:47PM
0
Connection dropped after 15 minutes with Deutsche Telekom
Friday January 6 2017
Time
Replies
Subject
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
Time
Replies
Subject
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
Time
Replies
Subject
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
Time
Replies
Subject
12:26PM
1
anveo, a different kind of trunk provider?