Friday August 31 2012 |
Time | Replies | Subject |
6:48PM |
1 |
Receiving and processing unsolicited XMPP messages with Asterisk 11 |
5:53PM |
3 |
asterisk on arm |
3:22PM |
5 |
Record calls as BLOB into MySQL? |
2:36PM |
4 |
Good way to query data from asterisk realtime with Asterisk Manager API |
2:18PM |
2 |
Question about cli |
10:38AM |
4 |
Automatic ODBC reconnect? |
5:50AM |
0 |
failed to extend from 512 to 676 message on console |
|
Thursday August 30 2012 |
Time | Replies | Subject |
8:45PM |
0 |
AST-2012-013: ACL rules ignored when placing outbound calls by certain IAX2 users |
8:45PM |
0 |
AST-2012-012: Asterisk Manager User Unauthorized Shell Access |
3:26PM |
0 |
Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, 10.7.1-digiumphones Now Available (Security Release) |
11:31AM |
0 |
Spa3102 info about tones an frecuency for Brasil's analog line |
9:34AM |
2 |
change channel variable to a user chosen value during a call |
|
Wednesday August 29 2012 |
Time | Replies | Subject |
10:28PM |
2 |
Install AsteriskNow |
5:17PM |
2 |
Click-to-call software in a hosted environment |
2:14AM |
2 |
Asterisk Package Question |
|
Tuesday August 28 2012 |
Time | Replies | Subject |
8:23PM |
2 |
FAX detection in chan_dahdi 1.8.15 |
5:05PM |
1 |
Call Recording |
4:51PM |
2 |
Best practices for hints management in extensions.conf |
2:39PM |
5 |
How do you convert your prompts to an asterisk-friendly format? |
10:40AM |
0 |
Increase Asterisk AGI commands length |
|
Monday August 27 2012 |
Time | Replies | Subject |
5:21PM |
1 |
Getting hold status via AMI ? |
5:02PM |
1 |
Asterisk 1.8.15 distintive ringtone for internal calls |
2:08PM |
3 |
Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012 |
12:08PM |
6 |
can we install 10 PCI card on asterisk |
|
Sunday August 26 2012 |
Time | Replies | Subject |
11:43PM |
1 |
One leg in a conference and adjusting stream volume of other leg |
9:05PM |
2 |
the lenght of the uri affects on dialplan? |
4:10AM |
0 |
Is it a BUG |
|
Saturday August 25 2012 |
Time | Replies | Subject |
12:31PM |
2 |
Understanding CHANNEL function values |
7:08AM |
1 |
Basic GotoIf question |
|
Friday August 24 2012 |
Time | Replies | Subject |
2:18PM |
2 |
SIP Question - Early audio one-way or 2-way? |
12:42PM |
2 |
Log faulty calls? |
10:37AM |
1 |
CHANNEL arguments documentation? |
8:13AM |
1 |
xmpp / sip |
6:27AM |
1 |
Asterisk and Wave files. |
|
Thursday August 23 2012 |
Time | Replies | Subject |
10:31PM |
0 |
Bug or Not |
7:24PM |
1 |
GotoIf redirection to label not working correctly |
5:50PM |
1 |
Easy to install CDR-Viewer? |
4:26PM |
1 |
Japanese voicefiles |
3:30PM |
1 |
sip trunk failing to register causes sip phones to become unreachable |
2:31PM |
2 |
quick questions on version 10 |
2:05PM |
1 |
RemoveQueueMember and realtime queues |
1:28PM |
0 |
Asterisk 1.6 / voicemail / final voice auth-thankyou |
|
Wednesday August 22 2012 |
Time | Replies | Subject |
6:04PM |
3 |
Asterisk 1.8 and 11 |
10:09AM |
2 |
TE110P Wildcard does not work with Ubuntu 12.04 server |
5:13AM |
1 |
Load test for FXS and FXO cards |
1:09AM |
1 |
recording calls |
12:23AM |
1 |
comma issue with func_odbc |
|
Tuesday August 21 2012 |
Time | Replies | Subject |
7:51PM |
1 |
Asterisk 11 - XMPP and JabberSend() |
4:42PM |
1 |
Check for the voicemail |
1:33PM |
2 |
Which card to get? |
10:04AM |
1 |
version compatible with centos 5.7 (2.6.18-308.8.2.el5) |
|
Monday August 20 2012 |
Time | Replies | Subject |
8:38PM |
1 |
Extensions mask as variable? |
7:20PM |
1 |
Asterisk 11 - BLF on Custom devices |
6:57PM |
1 |
Asterisk 11 queue calls - emulate Dial(b) functionality |
3:14PM |
1 |
Digium Phones |
2:23PM |
1 |
Asterisk as TLS server as well as TLS client |
12:19PM |
1 |
DTMF Issue. |
12:37AM |
6 |
using analog phones |
|
Sunday August 19 2012 |
Time | Replies | Subject |
8:50PM |
1 |
CDR default table specification? |
4:51PM |
1 |
Enable CDR logging? |
2:49PM |
1 |
new How-to guide: using repro SIP proxy for TLS with Asterisk |
1:26PM |
2 |
Verifying if Asterisk is connected using ODBC? |
|
Saturday August 18 2012 |
Time | Replies | Subject |
10:39PM |
3 |
graceful restart |
8:55AM |
1 |
asterisk tries reinvite when incompatible codecs on call legs |
8:45AM |
1 |
Make outgoing calls through BroadWorks/BroadSoft SIP gateway from Asterisk |
|
Friday August 17 2012 |
Time | Replies | Subject |
7:26PM |
1 |
BLF and Call Queues |
6:45PM |
2 |
How to test Websocket support in SIP in Asterisk trunk? |
6:11PM |
1 |
Hosted Softswitch Integration |
9:29AM |
0 |
OpenVox G400P SMS messages character set issues |
6:27AM |
0 |
Trouble with call pickup using RPID with Cisco |
12:07AM |
4 |
Grandstream VoIP phones |
|
Thursday August 16 2012 |
Time | Replies | Subject |
10:08PM |
1 |
TDM Fax |
8:12PM |
1 |
Requiring agent to confirm queue calls only when forwarded to external device |
6:47PM |
1 |
Fax Detect on Demand |
5:01AM |
0 |
Still having CDR problems. |
12:45AM |
1 |
How to input NULL in CDR report |
12:13AM |
1 |
UDP miss a hangup on SIP |
|
Wednesday August 15 2012 |
Time | Replies | Subject |
8:42PM |
0 |
DTMF detection issues |
6:35PM |
1 |
Send Fax from Asterisk |
5:57PM |
1 |
Incompatible voice frame ulaw/alaw |
1:42PM |
1 |
Extensions DTMF |
|
Tuesday August 14 2012 |
Time | Replies | Subject |
9:30PM |
0 |
Revoking a TLS certificat created with ast_tls_cert |
7:44PM |
2 |
VOIP over Metro Ethernet |
7:41PM |
0 |
SIP client that supports T.38 Fax |
7:26PM |
0 |
SayUnixTime quandry |
5:43PM |
2 |
Call in the queue to listen to the Channel |
4:44PM |
0 |
Virgin Meda VMDG280 and SIP Asterisk |
1:50PM |
1 |
Email to Fax solution |
1:45PM |
0 |
Console/Dsp |
|
Monday August 13 2012 |
Time | Replies | Subject |
6:58PM |
1 |
Websockets on Asterisk 11 and SipML5 |
12:19PM |
8 |
Asterisk hangs while starting in OpenSuse 12.2 |
9:32AM |
0 |
Reverse phone lookup service |
|
Sunday August 12 2012 |
Time | Replies | Subject |
4:11AM |
0 |
AstLinux 1.0.4 Released |
|
Saturday August 11 2012 |
Time | Replies | Subject |
12:05PM |
5 |
best free fax solution with asterisk |
10:16AM |
4 |
Segmenting A Configration File |
|
Friday August 10 2012 |
Time | Replies | Subject |
8:47PM |
4 |
Debian 7/Asterisk TLS bug and others |
8:43PM |
3 |
ConfBridge |
5:02PM |
0 |
iCall service any good? |
4:25PM |
0 |
Asterisk 11.0.0-beta1 Now Available! |
3:23PM |
1 |
Question on app_confbridge |
1:00PM |
1 |
asterisk and meetme |
12:42PM |
0 |
chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway! |
|
Thursday August 9 2012 |
Time | Replies | Subject |
9:21PM |
1 |
Asterisk to control just one phone within current CCM? |
8:12PM |
4 |
Asterisk on Rackspace, My SIP phone behind NAT |
5:59PM |
1 |
Multi-tenant IVR |
6:54AM |
2 |
No CDR after upgrade (1.6.x -> 10.2.1) |
5:36AM |
1 |
IAX with two asterisk boxes |
12:11AM |
0 |
tls is up but no audio |
|
Wednesday August 8 2012 |
Time | Replies | Subject |
6:11PM |
1 |
alwaysauthreject=yes not working as expected |
4:51PM |
0 |
qualifysmoothing |
2:17PM |
0 |
OT - Patton - FXO - Reduce incoming call delay |
11:30AM |
1 |
RFC List |
|
Tuesday August 7 2012 |
Time | Replies | Subject |
4:42PM |
0 |
Asterisk and SNMP. No resource graphs in OpenNMS. |
12:00PM |
1 |
Asterisk & Websockets |
7:31AM |
1 |
asterisk debian package and digium repository |
|
Monday August 6 2012 |
Time | Replies | Subject |
10:43PM |
4 |
Showing the name of the called number at the source IP Phone, how? |
10:31PM |
1 |
Background, Playback wave files in asterisk |
6:40PM |
1 |
Block outbound calls based on IP address |
5:59PM |
1 |
Asterisk 1.6 and Outbound SIP Proxy configuration |
1:48PM |
2 |
Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial |
8:57AM |
0 |
SIP register refresh time |
8:03AM |
2 |
asterisk.ctl file |
|
Sunday August 5 2012 |
Time | Replies | Subject |
11:26PM |
0 |
Background, Playback wave files in Asterisk 1.8.11-cert1 |
5:52PM |
3 |
Voice Mail beep / tone detection |
|
Saturday August 4 2012 |
Time | Replies | Subject |
6:09AM |
0 |
Message could not be delivered |
12:11AM |
1 |
Suggestion of Server Specifications for Asterisk |
|
Friday August 3 2012 |
Time | Replies | Subject |
9:42PM |
1 |
Talk detection during call |
9:19PM |
0 |
Scheduled Maintenance for Asterisk Project community services |
8:24PM |
1 |
Unplanned Asterisk community service outage |
12:10PM |
1 |
Voicemail full. |
7:49AM |
1 |
asterisk realtime database structure |
4:27AM |
6 |
Asterisk realtime don't support 'n' as extension's next priority |
|
Thursday August 2 2012 |
Time | Replies | Subject |
5:24PM |
1 |
Originate call from cli does not work for SIP line... |
12:45PM |
1 |
DTMF transmission problem |
12:28PM |
1 |
can't get libpri/PRI to work |
2:27AM |
4 |
html/js/flash/air SIP clients? |
|
Wednesday August 1 2012 |
Time | Replies | Subject |
5:45PM |
2 |
Problem with callfile and CDR |
3:15PM |
1 |
Asterisk Dahdi 1.6.2.23 Iaxmodem |
2:52PM |
0 |
Planned service outage for community services on August 2, 2012 |
1:54PM |
0 |
CallerID |
1:31PM |
0 |
Call Completion Supplementary Services (CCSS) sound files? |
12:11PM |
1 |
app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close |
10:24AM |
0 |
16kHz sampling |
6:05AM |
2 |
Problem provisioning Cisco SPA303 |