Saturday January 31 2015 |
Time | Replies | Subject |
4:40PM |
0 |
Question on multicast source |
|
Friday January 30 2015 |
Time | Replies | Subject |
6:02PM |
2 |
JITTERBUFFER function |
2:37PM |
2 |
SSL traffic on RTP instance without an SSL session |
8:48AM |
0 |
Remote Attended Transfer |
5:25AM |
1 |
Dialplan for receiving faxes on Asterisk |
1:16AM |
1 |
What conditions allow the use of dahdi native bridge? |
|
Thursday January 29 2015 |
Time | Replies | Subject |
7:56PM |
1 |
JITTERBUFFER function |
5:07PM |
1 |
Semi OT - LDAP multi-valued attributes support in SIP phones |
4:34PM |
0 |
What conditions allow the use of dahdi native bridge? |
3:52PM |
0 |
any valid up-to-date info about Kamailio-Asterisk integration ? |
3:41PM |
0 |
JITTERBUFFER function |
1:37PM |
1 |
Investigating international calls fraud |
11:51AM |
0 |
Investigating international calls fraud |
10:56AM |
2 |
JITTERBUFFER function |
8:43AM |
2 |
any valid up-to-date info about Kamailio-Asterisk integration ? |
6:10AM |
2 |
Investigating international calls fraud |
2:27AM |
2 |
What conditions allow the use of dahdi native bridge? |
|
Wednesday January 28 2015 |
Time | Replies | Subject |
11:29PM |
0 |
AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability |
11:29PM |
0 |
AST-2015-001: File descriptor leak when incompatible codecs are offered |
11:19PM |
0 |
Investigating international calls fraud |
11:14PM |
0 |
Asterisk 1.8.28-cert4, 1.8.32.2, 11.6-cert10, 11.15.1, 12.8.1, 13.1.1 Now Available (Security Release) |
11:07PM |
2 |
Investigating international calls fraud |
10:30PM |
1 |
Investigating international calls fraud |
10:07PM |
0 |
Investigating international calls fraud |
9:38PM |
1 |
Investigating international calls fraud |
9:23PM |
0 |
Investigating international calls fraud |
9:03PM |
5 |
Investigating international calls fraud |
6:37PM |
0 |
queue show <queue-name> vs queue log for calculating average hold time |
5:40PM |
1 |
subscriber absent |
5:23PM |
2 |
queue show <queue-name> vs queue log for calculating average hold time |
4:19PM |
0 |
Asterisk Java API - Up to date |
1:27PM |
1 |
Cannot get my first WebRTC experiment to work. |
|
Tuesday January 27 2015 |
Time | Replies | Subject |
9:14PM |
2 |
Asterisk Java API - Up to date |
11:27AM |
1 |
Inline transfer |
10:09AM |
0 |
Dialing from phonebook, and hiding the dialed number from the user. |
7:55AM |
1 |
asterisk 11.14 - voicemail incorrect duration |
1:47AM |
0 |
Need some help interpreting SDP on a failing WebRTC connection |
|
Monday January 26 2015 |
Time | Replies | Subject |
10:56PM |
0 |
asterisk 11.14 - voicemail incorrect duration |
7:29PM |
3 |
Dialing from phonebook, and hiding the dialed number from the user. |
6:44PM |
1 |
PJSIP vs SIP channeltype |
6:00PM |
0 |
Need help interpreting SDP on failing WebRTC connection |
3:37PM |
2 |
asterisk 11.14 - voicemail incorrect duration |
10:43AM |
0 |
Call Recording doesn't work |
|
Sunday January 25 2015 |
Time | Replies | Subject |
10:53PM |
1 |
Wiki (pjsip+realtime) says don't put the transports into realtime. Still true? |
10:49PM |
0 |
Wiki (pjsip+realtime) says don't put the transports into realtime. Still true? |
10:25PM |
2 |
Wiki (pjsip+realtime) says don't put the transports into realtime. Still true? |
11:11AM |
1 |
customizing Asterisk CLI |
10:15AM |
0 |
1/25/2015 10:15:09 AM |
3:33AM |
1 |
Best way to get dahdi status |
|
Saturday January 24 2015 |
Time | Replies | Subject |
7:05AM |
0 |
h235 for authenticating RAS message |
|
Friday January 23 2015 |
Time | Replies | Subject |
4:38PM |
1 |
Polycom SoundStation 6000 Dropping Registration |
4:29PM |
0 |
Polycom SoundStation 6000 Dropping Registration |
4:24PM |
3 |
Polycom SoundStation 6000 Dropping Registration |
|
Thursday January 22 2015 |
Time | Replies | Subject |
10:47PM |
0 |
DAHDI-Linux and DAHDI-Tools 2.10.1-rc2 Now Available |
9:32PM |
0 |
New Feature CALLERID(ani2) read/write |
7:57PM |
0 |
CDR and confbridge |
7:45PM |
0 |
Maintenance for community services tonight (January 22nd) |
4:22PM |
1 |
CALLERID(ani2) inserting |
10:31AM |
1 |
Problem with Cisco Phones |
|
Wednesday January 21 2015 |
Time | Replies | Subject |
3:55PM |
1 |
PJ SIP realtime with Kamailio / opensips |
2:24PM |
0 |
Return SIP 401 on hangup |
1:38PM |
0 |
asterisk-users Digest, Vol 126, Issue 18 mtr |
|
Tuesday January 20 2015 |
Time | Replies | Subject |
10:35PM |
1 |
Mailbox password change problem on realtime engine |
5:26PM |
0 |
Problem with Cisco Phones |
5:16PM |
2 |
Problem with Cisco Phones |
5:03PM |
0 |
Problem with Cisco Phones |
4:41PM |
2 |
Problem with Cisco Phones |
4:32PM |
0 |
Problem with Cisco Phones |
4:03PM |
2 |
Problem with Cisco Phones |
2:55PM |
0 |
sip show channelstats reliable? |
1:59PM |
0 |
MWI issue |
|
Monday January 19 2015 |
Time | Replies | Subject |
7:13PM |
2 |
sip show channelstats reliable? |
7:00PM |
0 |
sip show channelstats reliable? |
6:55PM |
2 |
sip show channelstats reliable? |
6:44PM |
0 |
sip show channelstats reliable? |
6:17PM |
2 |
sip show channelstats reliable? |
4:24PM |
2 |
SEMI-OFFTOPIC openvox |
2:06PM |
1 |
Meaning of core show hint output |
|
Saturday January 17 2015 |
Time | Replies | Subject |
2:31PM |
1 |
dahdi_genconf fails with "Empty configuration - no spans" |
1:35PM |
1 |
Google Voice |
8:31AM |
0 |
dahdi_genconf fails with "Empty configuration - no spans" |
6:04AM |
1 |
Fwd: Asterisk pjsip auto dtmf mode |
12:00AM |
2 |
dahdi_genconf fails with "Empty configuration - no spans" |
|
Friday January 16 2015 |
Time | Replies | Subject |
10:13PM |
0 |
Disable fax detect on specific incoming DID |
3:33PM |
0 |
agitator - FastAGI reverse proxy |
10:58AM |
2 |
Disable fax detect on specific incoming DID |
|
Thursday January 15 2015 |
Time | Replies | Subject |
11:03PM |
0 |
Showing sip subscriptions in Manager |
11:00PM |
2 |
Showing sip subscriptions in Manager |
6:58PM |
0 |
dahdi_genconf fails with "Empty configuration - no spans" |
6:27PM |
0 |
(no subject) |
8:05AM |
2 |
dahdi_genconf fails with "Empty configuration - no spans" |
|
Wednesday January 14 2015 |
Time | Replies | Subject |
6:21PM |
0 |
Asterisk - WiMax Island Use |
4:32PM |
3 |
Asterisk - WiMax Island Use |
5:24AM |
1 |
WSS Socket Configuration |
|
Tuesday January 13 2015 |
Time | Replies | Subject |
10:02AM |
0 |
WSS Socket Configuration |
|
Monday January 12 2015 |
Time | Replies | Subject |
3:52PM |
0 |
Polycom instant messages |
2:11PM |
0 |
Which ATA supporting T.38 and 802.1x ? |
1:41PM |
1 |
SEMI OFF-TOPIC - Fail2ban |
6:39AM |
0 |
Asterisk executable suddenly about 40KB |
5:19AM |
3 |
Polycom instant messages |
|
Saturday January 10 2015 |
Time | Replies | Subject |
10:57AM |
0 |
Asterisk e Busca Automatica |
|
Friday January 9 2015 |
Time | Replies | Subject |
11:24PM |
0 |
SEMI OFF-TOPIC - Fail2ban |
8:55PM |
0 |
[Elastix-user] hide files |
8:49PM |
0 |
[Elastix-users] file with passwords for default |
8:02PM |
2 |
SEMI OFF-TOPIC - Fail2ban |
3:05PM |
0 |
SEMI OFF-TOPIC - Fail2ban |
9:53AM |
0 |
SEMI OFF-TOPIC - Fail2ban |
7:29AM |
1 |
Asterisk executable suddenly about 40KB larger - modules (Andres) |
1:03AM |
0 |
Asterisk 13.1.0/PJSIP peer IP address issue |
|
Thursday January 8 2015 |
Time | Replies | Subject |
11:53PM |
0 |
Allison Smith AMA |
9:37PM |
4 |
SEMI OFF-TOPIC - Fail2ban |
7:32PM |
4 |
Asterisk 13.1.0/PJSIP peer IP address issue |
4:48PM |
0 |
Asterisk 13.1.0/PJSIP peer IP address issue |
4:15PM |
2 |
Asterisk 13.1.0/PJSIP peer IP address issue |
2:23PM |
0 |
queue reload command |
12:10PM |
2 |
queue reload command |
7:24AM |
1 |
Asterisk executable suddenly about 40KB larger - modules |
|
Wednesday January 7 2015 |
Time | Replies | Subject |
7:18PM |
0 |
Adding an Event on chan_sip.c (asterisk 1.8.22) |
12:31PM |
3 |
Asterisk executable suddenly about 40KB larger - modules not working |
|
Tuesday January 6 2015 |
Time | Replies | Subject |
8:59PM |
0 |
Participant unable to hear other participants in ConfBridge |
12:44PM |
0 |
PJSIP / T.38 - Asterisk not passing on v21 preamble and data |
|
Monday January 5 2015 |
Time | Replies | Subject |
11:39PM |
0 |
Planned maintenance for community services tomorrow night, Tuesday, Jan 6th 2015 |
1:15PM |
0 |
Asterisk removes a charachter from sip peer name |
5:52AM |
0 |
weird stasis_cache.c error in Asterisk 13.0/13.1 version |
3:52AM |
0 |
A little OT: AstDemo - Openfire and Asterisk integration |
2:27AM |
1 |
Confused by concepts behind pjsip: endpoint, aor, contact |
1:48AM |
0 |
Confused by concepts behind pjsip: endpoint, aor, contact |
1:41AM |
2 |
Confused by concepts behind pjsip: endpoint, aor, contact |
|
Sunday January 4 2015 |
Time | Replies | Subject |
11:39PM |
0 |
Confused by concepts behind pjsip: endpoint, aor, contact |
11:31PM |
2 |
Confused by concepts behind pjsip: endpoint, aor, contact |
10:45PM |
0 |
Confused by concepts behind pjsip: endpoint, aor, contact |
10:29PM |
3 |
Confused by concepts behind pjsip: endpoint, aor, contact |
|
Saturday January 3 2015 |
Time | Replies | Subject |
8:03AM |
2 |
Asterisk removes a charachter from sip peer name |
|
Friday January 2 2015 |
Time | Replies | Subject |
10:16AM |
2 |
using feature from applicationmap while ringing in queue |
|
Thursday January 1 2015 |
Time | Replies | Subject |
6:09PM |
2 |
PJSIP / T.38 - Asterisk not passing on v21 preamble and data |