Friday July 31 2015 |
Time | Replies | Subject |
6:10PM |
0 |
showing sip number insted of pri number |
4:10PM |
4 |
showing sip number insted of pri number |
1:04PM |
0 |
How to configure through the GUI 35 cisco ip phones, -spa502g (jg) |
|
Thursday July 30 2015 |
Time | Replies | Subject |
11:20PM |
0 |
PJSIP T.38 issues |
8:56AM |
0 |
Insecure Meaning. |
6:51AM |
0 |
CEL eventtime incorrect, but CDR times are correct - 1.8.11.0 |
4:02AM |
0 |
Windows Asterisk Help |
12:56AM |
1 |
How to configure through the GUI 35 cisco ip phones -spa502g |
|
Wednesday July 29 2015 |
Time | Replies | Subject |
10:40PM |
0 |
Asterisk 1.8.22.0 built - encrypt authentication |
8:30PM |
3 |
Asterisk 1.8.22.0 built - encrypt authentication |
4:53PM |
0 |
Queues don't follow dialplan if no members are registered |
4:47PM |
2 |
Windows Asterisk Help |
3:16PM |
0 |
Windows Asterisk Help |
2:26PM |
2 |
Windows Asterisk Help |
2:11PM |
0 |
Windows Asterisk Help |
1:59PM |
3 |
Windows Asterisk Help |
4:13AM |
2 |
PJSIP T.38 issues |
3:51AM |
2 |
Queues don't follow dialplan if no members are registered |
|
Tuesday July 28 2015 |
Time | Replies | Subject |
9:26PM |
0 |
Siren7 and Asterisk 13 |
5:12PM |
0 |
Queues don't follow dialplan if no members are registered |
4:58PM |
2 |
Queues don't follow dialplan if no members are registered |
10:05AM |
1 |
re-invite update dialog |
|
Monday July 27 2015 |
Time | Replies | Subject |
6:38PM |
1 |
Why no CentOS 7 repos? |
6:19PM |
0 |
Why no CentOS 7 repos? |
5:51PM |
2 |
Why no CentOS 7 repos? |
4:54PM |
0 |
No audio on SIP over WebRTC |
4:30PM |
0 |
Asterisk 11.19.0-rc1 Now Available |
11:22AM |
0 |
PJSIP T.38 issues |
9:51AM |
1 |
Filters |
3:15AM |
2 |
PJSIP T.38 issues |
|
Friday July 24 2015 |
Time | Replies | Subject |
12:41AM |
0 |
Update of dialed number on sip phones |
|
Thursday July 23 2015 |
Time | Replies | Subject |
9:54AM |
1 |
Centos 6.5 Asterisk 1.8.11.0 - starts in rc.local, but not contactible? |
9:13AM |
1 |
Centos 6.5 Asterisk 1.8.11.0 - starts in rc.local, but not contactible? |
3:12AM |
0 |
Cisco 7940 and PJSIP registration |
12:50AM |
2 |
Cisco 7940 and PJSIP registration |
|
Wednesday July 22 2015 |
Time | Replies | Subject |
11:29AM |
0 |
Cisco 7940 and PJSIP registration |
5:38AM |
2 |
Cisco 7940 and PJSIP registration |
1:45AM |
0 |
Cisco 7940 and PJSIP registration |
1:37AM |
2 |
Cisco 7940 and PJSIP registration |
|
Tuesday July 21 2015 |
Time | Replies | Subject |
1:58PM |
0 |
Queue handling : does every call have to be accounted twice? |
8:43AM |
1 |
asterisk segfault debian jessie asterisk 11.13 |
8:27AM |
1 |
Always 486 Busy Here for anonymous calls |
7:19AM |
0 |
sip can not transmit fax receive from chan dahdi |
|
Monday July 20 2015 |
Time | Replies | Subject |
5:53AM |
0 |
RES: Can I use PJSIP_HEADER to read the SIP 183 message header? |
|
Friday July 17 2015 |
Time | Replies | Subject |
2:09PM |
1 |
Recording INCOMING calls |
2:06PM |
0 |
Recording INCOMING calls |
|
Thursday July 16 2015 |
Time | Replies | Subject |
2:09PM |
0 |
Asterisk and Vitelity's vMobile service |
1:54PM |
0 |
How to create direct media with PJSIP.conf configurations in Asterisk 13? |
1:49PM |
2 |
How to create direct media with PJSIP.conf configurations in Asterisk 13? |
9:06AM |
1 |
How to enable group call |
8:51AM |
0 |
how to return a transfered call to the transferrer? |
8:37AM |
2 |
Recording INCOMING calls |
|
Wednesday July 15 2015 |
Time | Replies | Subject |
9:35PM |
1 |
How to dial extensions asynchronous-sequentially ? |
9:24PM |
0 |
How to dial extensions asynchronous-sequentially ? |
9:16PM |
2 |
How to dial extensions asynchronous-sequentially ? |
7:51PM |
2 |
how to return a transfered call to the transferrer? |
5:46PM |
1 |
Problem "no voice" |
5:41PM |
0 |
Problem "no voice" |
5:01PM |
2 |
Problem "no voice" |
4:57PM |
0 |
How to call a group of peers all registered with the same login? |
|
Tuesday July 14 2015 |
Time | Replies | Subject |
8:08PM |
1 |
ConfBridge play message to all in conf |
7:38PM |
0 |
pjsip.conf question |
7:34PM |
2 |
pjsip.conf question |
7:03PM |
0 |
pjsip.conf question |
6:56PM |
2 |
pjsip.conf question |
8:29AM |
0 |
Dial L options and attended tranfer |
|
Monday July 13 2015 |
Time | Replies | Subject |
9:57PM |
0 |
RES: RES: How to dial extensions asynchronous-sequentially ? |
9:32PM |
2 |
RES: RES: How to dial extensions asynchronous-sequentially ? |
8:43PM |
0 |
RES: How to dial extensions asynchronous-sequentially ? |
7:51PM |
3 |
RES: How to dial extensions asynchronous-sequentially ? |
7:24PM |
0 |
How to dial extensions asynchronous-sequentially ? |
6:28PM |
3 |
How to dial extensions asynchronous-sequentially ? |
12:38PM |
0 |
CEL eventtime drift |
|
Sunday July 12 2015 |
Time | Replies | Subject |
7:04AM |
0 |
Does the asterisk support instance mes saging ? |
7:03AM |
0 |
Does the asterisk support instance messaging ? |
7:02AM |
0 |
Does the asterisk support instance messaging ? |
|
Saturday July 11 2015 |
Time | Replies | Subject |
12:11AM |
0 |
Tdm4010p 4 port card |
12:11AM |
0 |
Tdm4010p 4 port card |
12:10AM |
0 |
Tdm4010p 4 port card |
|
Friday July 10 2015 |
Time | Replies | Subject |
6:36PM |
2 |
RES: Can I use PJSIP_HEADER to read the SIP 183 message header? |
6:29PM |
0 |
Messages out of calls. Is it really possible? |
6:14PM |
0 |
Can I use PJSIP_HEADER to read the SIP 183 message header? |
4:53PM |
2 |
Can I use PJSIP_HEADER to read the SIP 183 message header? |
4:51PM |
2 |
Messages out of calls. Is it really possible? |
3:37PM |
3 |
Tdm4010p 4 port card |
2:45PM |
2 |
Sending E-Mail from voicemail with AND without attachment |
11:09AM |
0 |
11.18.0 patch against 11.17.0 running version failed to apply |
11:09AM |
0 |
11.18.0 patch against 11.17.0 running version failed to apply |
9:22AM |
0 |
Asterisk SMS |
8:30AM |
2 |
Asterisk SMS |
7:26AM |
2 |
Sending E-Mail from voicemail |
|
Thursday July 9 2015 |
Time | Replies | Subject |
7:51PM |
0 |
What Dial Plan function can access the contents of the SDP ? |
6:18PM |
1 |
Can't install gmime22 |
3:34PM |
0 |
Asterisk 13 / realtime voicemail creation |
3:28PM |
2 |
Asterisk 13 / realtime voicemail creation |
3:05PM |
0 |
11.18.0 patch against 11.17.0 running version failed to apply |
8:02AM |
0 |
Call Return |
5:10AM |
0 |
PJSIP, T.38 fax gateway |
|
Wednesday July 8 2015 |
Time | Replies | Subject |
10:28PM |
2 |
11.18.0 patch against 11.17.0 running version failed to apply |
7:11PM |
0 |
tls on asterisk 13 |
7:09PM |
0 |
tls on asterisk 13 |
7:05PM |
6 |
tls on asterisk 13 |
6:58PM |
0 |
How to handle multiple lines call |
6:13PM |
2 |
Call Return |
4:02PM |
1 |
How to handle multiple lines call |
3:45PM |
1 |
How to enable IM over the asterisk server |
3:36PM |
0 |
11.18.0 patch against 11.17.0 running version failed to apply |
2:53PM |
0 |
How may SIP 183 messages a caller receives when many callee rings? |
2:51PM |
0 |
Bug in ast_frame_adjust_volume in 12.2.0? |
1:14PM |
2 |
11.18.0 patch against 11.17.0 running version failed to apply |
11:55AM |
0 |
How to enable IM over the asterisk server |
9:27AM |
1 |
DTMF issue |
9:21AM |
0 |
Asterisk how to setup alarm too many outgoing calls from same user |
8:24AM |
3 |
How to enable IM over the asterisk server |
5:43AM |
0 |
How to enable IM over the asterisk server |
|
Tuesday July 7 2015 |
Time | Replies | Subject |
9:23PM |
0 |
DTMF issue |
8:53PM |
2 |
DTMF issue |
8:28PM |
1 |
Can I use ARI to update the builtin database, without executing the dial plan? |
7:44PM |
0 |
DTMF issue |
7:22PM |
0 |
Asterisk pin code for out-going international calls (safeguard against fraud) |
7:03PM |
2 |
DTMF issue |
6:14PM |
0 |
DTMF issue |
6:01PM |
2 |
Bug in ast_frame_adjust_volume in 12.2.0? |
4:09PM |
2 |
Asterisk pin code for out-going international calls (safeguard against fraud) |
2:58PM |
0 |
What database should I use, for simple data storing? SQLite or the buitin one? |
2:51PM |
1 |
Fwd: What database should I use, for simple data storing? SQLite or the buitin one? |
2:32PM |
0 |
What database should I use, for simple data storing? SQLite or the buitin one? |
2:26PM |
4 |
What database should I use, for simple data storing? SQLite or the buitin one? |
9:28AM |
2 |
How to enable IM over the asterisk server |
8:17AM |
0 |
How to enable IM over the asterisk server |
7:33AM |
0 |
Voicemail: saycid without prefix |
5:58AM |
1 |
Issue call quality: Asterisk call quality on trunks |
4:07AM |
2 |
How to enable IM over the asterisk server |
12:32AM |
0 |
CDR in an MySQL-Database |
|
Monday July 6 2015 |
Time | Replies | Subject |
11:54PM |
0 |
Asterisk pin code for out-going international calls (safeguard against fraud) |
9:53PM |
4 |
DTMF issue |
9:43PM |
3 |
Asterisk pin code for out-going international calls (safeguard against fraud) |
8:20PM |
1 |
CDR in an MySQL-Database |
8:14PM |
0 |
CDR in an MySQL-Database |
7:22PM |
4 |
CDR in an MySQL-Database |
6:23PM |
0 |
SIP/2.0 401 Unauthorized when calling from one SIP extension to another |
4:25PM |
2 |
Voicemail: saycid without prefix |
4:23PM |
0 |
Voicemail: saycid without prefix |
3:06PM |
0 |
Asterisk how to setup alarm too many outgoing calls from same user |
2:27PM |
2 |
Asterisk how to setup alarm too many outgoing calls from same user |
2:17PM |
0 |
Asterisk 13.4.0 - mixmonitor only records one side's perspective |
1:26PM |
2 |
How may SIP 183 messages a caller receives when many callee rings? |
10:19AM |
0 |
Unisteam not showing callerid |
9:55AM |
0 |
Choosing codecs |
8:57AM |
2 |
Choosing codecs |
8:38AM |
0 |
Choosing codecs |
8:17AM |
3 |
Choosing codecs |
8:05AM |
0 |
Choosing codecs |
7:56AM |
2 |
Choosing codecs |
7:48AM |
0 |
Choosing codecs |
12:53AM |
0 |
Can't install gmime22 |
|
Sunday July 5 2015 |
Time | Replies | Subject |
8:57PM |
0 |
Choosing codecs |
7:34PM |
2 |
Choosing codecs |
|
Saturday July 4 2015 |
Time | Replies | Subject |
6:06PM |
0 |
from: rschroe@gmail.com |
5:53AM |
2 |
Voicemail: saycid without prefix |
|
Friday July 3 2015 |
Time | Replies | Subject |
6:46PM |
2 |
Action Originate in Asterisk 13 creates 2 calls in core show channels |
2:10PM |
2 |
Asterisk 11 and pulseaudio setup as local user |
11:17AM |
0 |
Asterisk 11 and pulseaudio setup as local user |
|
Thursday July 2 2015 |
Time | Replies | Subject |
5:54PM |
5 |
Asterisk 11 and pulseaudio setup as local user |
5:02PM |
0 |
confbridge play tone before speaking |
3:28PM |
0 |
multiple sip trunks with the same ITSP |
12:53PM |
0 |
Custom header when busy |
12:49PM |
3 |
Custom header when busy |
12:31PM |
0 |
Custom header when busy |
12:05PM |
2 |
Custom header when busy |
12:02PM |
0 |
asterisk email to fax |
11:45AM |
2 |
asterisk email to fax |
11:02AM |
0 |
Custom header when busy |
8:29AM |
0 |
For a failed retransmission - what were the IP addresses? |
|
Wednesday July 1 2015 |
Time | Replies | Subject |
6:44PM |
0 |
Dell portability |
6:36PM |
2 |
Dell portability |
12:43PM |
0 |
Sip registrations question |
9:46AM |
2 |
Custom header when busy |
7:10AM |
0 |
Help With Physical Layer |
3:03AM |
1 |
Question on permit/deny |