Tuesday May 31 2016 |
Time | Replies | Subject |
5:20PM |
1 |
__sip_xmit Returned -1 Invalid Argument |
4:16PM |
0 |
Performance Note: Creating Local channels with ARI |
3:43PM |
0 |
How to set outgoing sip callid ? |
2:53PM |
2 |
How to set outgoing sip callid ? |
5:53AM |
0 |
Need stronger SRTP ciphers (256 bit) |
|
Monday May 30 2016 |
Time | Replies | Subject |
11:58PM |
0 |
Trying to record incoming calls that go to queues in Asterisk v11 |
9:41PM |
0 |
Asterisk 13 IAX and MoH realtime |
6:49PM |
2 |
Need stronger SRTP ciphers (256 bit) |
|
Sunday May 29 2016 |
Time | Replies | Subject |
8:04PM |
0 |
asterisk odbc segfaults (SOLVED) |
7:31PM |
2 |
asterisk odbc segfaults |
3:48PM |
0 |
asterisk odbc segfaults |
|
Friday May 27 2016 |
Time | Replies | Subject |
10:56PM |
0 |
What this attacks means? |
10:28PM |
2 |
What this attacks means? |
9:15PM |
0 |
registration timeout asterisk polycom sp450 transport=tls port 5061 provision server ftps |
8:05PM |
0 |
Solved !! Siemens Hicom --> Asterisk-Server <-- Telekom |
4:28PM |
2 |
asterisk odbc segfaults |
4:09PM |
0 |
asterisk odbc segfaults |
3:58PM |
2 |
asterisk odbc segfaults |
2:43PM |
0 |
Avaya Phones and Asterisk |
6:08AM |
1 |
Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP |
|
Thursday May 26 2016 |
Time | Replies | Subject |
9:14PM |
0 |
Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP |
8:12PM |
2 |
Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP |
2:11PM |
0 |
pjsip segfault problem |
11:34AM |
1 |
open source pbx free |
9:11AM |
3 |
pjsip segfault problem |
|
Wednesday May 25 2016 |
Time | Replies | Subject |
9:46PM |
0 |
Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP |
9:13PM |
3 |
Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP |
|
Monday May 23 2016 |
Time | Replies | Subject |
8:02PM |
6 |
Wildcard X100P Disconnect Problems |
12:55PM |
1 |
WSS ISSUE |
|
Sunday May 22 2016 |
Time | Replies | Subject |
3:32AM |
0 |
PCI FXO disconnect problems |
|
Saturday May 21 2016 |
Time | Replies | Subject |
1:42AM |
2 |
musiconhold.conf problems |
|
Friday May 20 2016 |
Time | Replies | Subject |
1:54PM |
1 |
Avaya Phones and Asterisk |
1:52PM |
0 |
Test |
5:03AM |
0 |
[SOLVED] AMI issue with Filter |
|
Thursday May 19 2016 |
Time | Replies | Subject |
10:54AM |
0 |
AMI issue with Filter |
|
Wednesday May 18 2016 |
Time | Replies | Subject |
9:32PM |
1 |
Hints realtime table structure Ast 11 |
5:29PM |
1 |
TDM800 just receive calls, but not make |
2:48PM |
0 |
variable to get waittime of caller exiting queue |
2:44PM |
1 |
Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ? |
12:05PM |
2 |
variable to get waittime of caller exiting queue |
8:44AM |
0 |
Way to replay messages recorded by voicemail() |
|
Tuesday May 17 2016 |
Time | Replies | Subject |
1:27PM |
0 |
PJSIP outgoing INVITE and "contact" value |
1:09PM |
0 |
Asterisk-Java library |
12:59PM |
1 |
Strange SIP debug |
9:57AM |
1 |
Asterisk 11 on Centos: Voicemail crashes when recording message |
|
Monday May 16 2016 |
Time | Replies | Subject |
11:54PM |
0 |
asterisk admin interface |
10:29PM |
0 |
asterisk admin interface |
9:55PM |
4 |
asterisk admin interface |
9:15PM |
0 |
asterisk admin interface |
9:03PM |
0 |
JABBER_RECEIVE timeout don't work |
8:23PM |
2 |
asterisk admin interface |
8:08PM |
0 |
Asterisk 11 on Centos: Voicemail crashes when recording message |
8:03PM |
2 |
Asterisk 11 on Centos: Voicemail crashes when recording message |
7:54PM |
0 |
asterisk admin interface |
6:33PM |
6 |
asterisk admin interface |
3:25PM |
0 |
Russian and French sounds |
3:06PM |
0 |
DAHDI press button get fast busy |
12:40PM |
0 |
Asterisk PJSIP Multi-tenant |
4:17AM |
2 |
Asterisk PJSIP Multi-tenant |
12:52AM |
0 |
Asterisk PJSIP Multi-tenant |
|
Sunday May 15 2016 |
Time | Replies | Subject |
6:00PM |
2 |
Asterisk PJSIP Multi-tenant |
|
Saturday May 14 2016 |
Time | Replies | Subject |
7:32PM |
1 |
Questions... connecting Asterisk to the World |
5:39PM |
0 |
Questions... connecting Asterisk to the World |
4:51PM |
3 |
Questions... connecting Asterisk to the World |
|
Friday May 13 2016 |
Time | Replies | Subject |
7:53PM |
0 |
Asterisk 13.9.1 Now Available |
|
Thursday May 12 2016 |
Time | Replies | Subject |
11:24PM |
0 |
"__sip_xmit....Success" every 15 seconds ! |
9:04PM |
2 |
DAHDI press button get fast busy |
12:01PM |
1 |
pjsip module reload problem |
11:38AM |
0 |
pjsip module reload problem |
11:35AM |
2 |
pjsip module reload problem |
9:54AM |
0 |
[asterisk 13.9] pjsip: Extensions always lost after short period of time |
8:12AM |
2 |
[asterisk 13.9] pjsip: Extensions always lost after short period of time |
1:43AM |
1 |
maximum call time |
1:12AM |
0 |
maximum call time |
1:08AM |
2 |
maximum call time |
|
Wednesday May 11 2016 |
Time | Replies | Subject |
8:52PM |
1 |
Call File - CPU spikes |
2:30PM |
0 |
Early Media Dialplan Issue |
12:39PM |
0 |
maximum call time |
11:26AM |
3 |
maximum call time |
10:09AM |
0 |
Switching between Music on Hold streams. [13.8.2] |
10:05AM |
2 |
Russian and French sounds |
10:02AM |
1 |
How is Queue avg holdtime and avg talktime calculated |
9:59AM |
0 |
How is Queue avg holdtime and avg talktime calculated |
9:24AM |
2 |
How is Queue avg holdtime and avg talktime calculated |
|
Tuesday May 10 2016 |
Time | Replies | Subject |
7:42AM |
0 |
VoipRaider is true for FREE calls? |
|
Monday May 9 2016 |
Time | Replies | Subject |
10:44PM |
0 |
VoipRaider is true for FREE calls? |
10:43PM |
4 |
VoipRaider is true for FREE calls? |
10:10PM |
2 |
Switching between Music on Hold streams. [13.8.2] |
5:36PM |
1 |
Early Media Dialplan Issue |
5:22PM |
0 |
Switching between Music on Hold streams. [13.8.2] |
5:18PM |
4 |
Switching between Music on Hold streams. [13.8.2] |
5:00PM |
0 |
Switching between Music on Hold streams. [13.8.2] |
2:49PM |
1 |
Switching between Music on Hold streams. [13.8.2] |
2:45PM |
0 |
Switching between Music on Hold streams. [13.8.2] |
2:42PM |
0 |
Asterisk 13.9.0 Now Available |
2:24PM |
2 |
Switching between Music on Hold streams. [13.8.2] |
2:00PM |
0 |
voicemail: duration while leaving a message |
1:58PM |
0 |
Switching between Music on Hold streams. [13.8.2] |
1:52PM |
2 |
voicemail: duration while leaving a message |
1:50PM |
3 |
Switching between Music on Hold streams. [13.8.2] |
6:46AM |
1 |
Proper way to start Asterisk on CentOS 7? (Carlos Chavez) |
|
Sunday May 8 2016 |
Time | Replies | Subject |
1:56PM |
0 |
Switching between Music on Hold streams. [13.8.2] |
10:36AM |
4 |
Switching between Music on Hold streams. [13.8.2] |
|
Saturday May 7 2016 |
Time | Replies | Subject |
5:19PM |
1 |
Detecting sounds while recording |
|
Friday May 6 2016 |
Time | Replies | Subject |
5:10PM |
1 |
Asterisk Secure SIP session TLS port 5061 |
12:40PM |
0 |
click2call for conferencing two mobile numbers |
10:26AM |
2 |
click2call for conferencing two mobile numbers |
|
Thursday May 5 2016 |
Time | Replies | Subject |
5:46PM |
0 |
cannot find -lasteriskssl |
5:21PM |
2 |
cannot find -lasteriskssl |
5:12PM |
0 |
cannot find -lasteriskssl |
5:09PM |
2 |
cannot find -lasteriskssl |
5:07PM |
0 |
Proper way to start Asterisk on CentOS 7? |
4:33PM |
0 |
cannot find -lasteriskssl |
4:31PM |
2 |
cannot find -lasteriskssl |
2:58AM |
2 |
Double queue calls being delivered to agents |
|
Wednesday May 4 2016 |
Time | Replies | Subject |
11:48PM |
0 |
Double queue calls being delivered to agents |
11:45PM |
0 |
Double queue calls being delivered to agents |
8:22PM |
0 |
UAC and UAS for timer refresher header |
5:11PM |
0 |
Asterisk 1.8 secure SIP session only |
4:43PM |
2 |
Asterisk 1.8 secure SIP session only |
1:23PM |
0 |
Anyone have problems with HPE 5130 EI Switch Series |
12:49PM |
2 |
Compatibilty between agi for asterisk 13.8.0 and php5.6 |
11:25AM |
0 |
Asterisk registers with TLS, but sends out calls via UDP |
9:12AM |
0 |
T.38 with Audiocodes gateway [SOLVED] |
1:59AM |
0 |
Double queue calls being delivered to agents |
|
Tuesday May 3 2016 |
Time | Replies | Subject |
11:48PM |
0 |
Execute an app on the master channel from inside a Macro on the called channel |
11:15PM |
2 |
Double queue calls being delivered to agents |
9:24PM |
1 |
Call a subroutine via Originate? |
7:16PM |
1 |
Migrating asterisk 11 to 13: some callers get no ringback tone any more |
7:07PM |
0 |
Migrating asterisk 11 to 13: some callers get no ringback tone any more |
6:52PM |
2 |
Migrating asterisk 11 to 13: some callers get no ringback tone any more |
6:45PM |
0 |
Migrating asterisk 11 to 13: some callers get no ringback tone any more |
4:56PM |
0 |
TDM804 card |
3:39PM |
1 |
Asterisk (PJSIP) registers with sips Contact URI, but why? |
2:43PM |
0 |
T.38 with Audiocodes gateway |
2:20PM |
0 |
Is MixMonitor command is blocking ? |
11:25AM |
2 |
Is MixMonitor command is blocking ? |
10:10AM |
2 |
Asterisk 13 Realtime Voicemail frustrating issue |
7:57AM |
0 |
my dahdi dont'n start |
7:43AM |
0 |
my dahdi dont'n start |
7:34AM |
0 |
Ubuntu 14 Warning |
3:50AM |
3 |
Migrating asterisk 11 to 13: some callers get no ringback tone any more |
|
Monday May 2 2016 |
Time | Replies | Subject |
4:04PM |
2 |
Ubuntu 14 Warning |
|
Sunday May 1 2016 |
Time | Replies | Subject |
8:45PM |
1 |
Taskprocessors |
3:01PM |
0 |
Homer Captagent 6 - duplicate records. |