Friday May 31 2013 |
Time | Replies | Subject |
9:57PM |
2 |
Help me understand these log messages |
7:21PM |
1 |
TE410P PCI card in 1U rackount server |
11:53AM |
1 |
WebRTC softphone for Asterisk - any suggestion? |
|
Thursday May 30 2013 |
Time | Replies | Subject |
7:29PM |
0 |
asterisk-users Digest, Vol 106, Issue 41 |
3:06PM |
1 |
how to launch a URl when dialing a number |
12:54PM |
1 |
Queue Periodic Announce not working... |
9:46AM |
2 |
Executing a dynamic sequence of applications |
|
Wednesday May 29 2013 |
Time | Replies | Subject |
8:18PM |
0 |
DAHDI-Linux and DAHDI-Tools 2.7.0-rc1 Now Available |
4:53PM |
0 |
IM through Asterisk |
|
Tuesday May 28 2013 |
Time | Replies | Subject |
6:11PM |
0 |
Initial cut off audio |
4:45PM |
2 |
Asterisk - Soundcard - Recording? |
6:21AM |
1 |
DTMF recognized after call establishment |
|
Monday May 27 2013 |
Time | Replies | Subject |
5:17PM |
0 |
ChanIsAvail function is breaking the round robin strategy |
5:14PM |
2 |
RED on DAHDI channel |
6:56AM |
3 |
Not able to build the chan_sip.c module |
6:19AM |
1 |
G.729 codec in pass-thru mode |
|
Saturday May 25 2013 |
Time | Replies | Subject |
3:14PM |
0 |
Asterisk 1.8 wrong Def. Username |
|
Friday May 24 2013 |
Time | Replies | Subject |
8:29PM |
1 |
Asterisk 11 dtmf not recognised |
5:46PM |
1 |
asterisk-gui-2.1.0-rc1 |
1:34PM |
0 |
Pri-Debug-Log / Is Early Media supported by provider? |
10:32AM |
1 |
Registration timed out - for created users |
|
Thursday May 23 2013 |
Time | Replies | Subject |
3:08PM |
1 |
Asterisk on Solaris |
1:49PM |
1 |
GotoIf function |
1:41PM |
0 |
Seeking for TTS engine supporting Hebrew |
9:57AM |
2 |
Integration with skype |
9:49AM |
1 |
Jabber |
12:04AM |
0 |
Diversion vs. P-Asserted-Id vs. Remote-Party-Id vs. P-Charge-Info vs. From Fields |
|
Wednesday May 22 2013 |
Time | Replies | Subject |
2:39PM |
1 |
Error 488 Not Acceptable Here |
1:56PM |
0 |
Changes to the community service maintenance notifications |
12:11AM |
0 |
Automatic Speech Recognition and Text To Speech using iSpeech |
|
Tuesday May 21 2013 |
Time | Replies | Subject |
5:11PM |
1 |
Failed to authenticate device "Ext 110" |
3:19PM |
4 |
Asterisk Log rotate not working |
1:21PM |
0 |
Planned maintenance for community services on May 23, 2013 |
|
Monday May 20 2013 |
Time | Replies | Subject |
9:05PM |
1 |
Stress testing Asterisk |
1:02PM |
2 |
Passcode |
1:01PM |
1 |
Secure Calling |
1:00PM |
1 |
Loopback question |
12:58PM |
1 |
Question |
|
Saturday May 18 2013 |
Time | Replies | Subject |
6:10PM |
0 |
Asterisk 1.8 vici and the fax, SMS, gtalk, Jaber channels |
6:01PM |
5 |
Performance Asterisk large installation on Vmware/Xen |
4:34PM |
1 |
Asterisk 1.8-cert and AGC |
|
Friday May 17 2013 |
Time | Replies | Subject |
9:03PM |
0 |
Asterisk 11.4.0 Now Available |
9:03PM |
0 |
Asterisk 1.8.22.0 Now Available |
8:35PM |
2 |
Auto dialer scripts and software |
9:47AM |
0 |
Temporarily features (transfer) off during Read |
|
Thursday May 16 2013 |
Time | Replies | Subject |
11:45PM |
1 |
wanpipe and digium, oslec and hardware echo canceller |
6:37PM |
0 |
Planned maintenance for community services on May 16, 2013 |
6:37PM |
0 |
Planned maintenance for community services on May 16, 2013 |
6:09PM |
0 |
asterisk-users Digest, Vol 106, Issue 23 |
5:14PM |
0 |
Asterisk High-availability/failover solutions |
1:41PM |
2 |
11.4: motif can only handle one channel at a time? |
1:26PM |
0 |
AstriCon 2013 (our 10th AstriCon) needs YOU! |
10:13AM |
1 |
Call Transfer question |
|
Wednesday May 15 2013 |
Time | Replies | Subject |
10:12PM |
1 |
SetCallerPres questions |
7:10PM |
2 |
Polycom and forwarding. |
6:18PM |
3 |
Cut offs on outgoing SIP calls |
3:10PM |
1 |
How to allow AMI access to Originate yet deny Application: System |
2:17PM |
0 |
3. mfcr2 channel state IDLE 0x00 and call trace log file not ended ?? (Leonardo Rivanera) |
|
Tuesday May 14 2013 |
Time | Replies | Subject |
7:42PM |
2 |
Using PHPMyAdmin to remotely access Asterisk MySQL Database |
7:21PM |
0 |
mfcr2 channel state IDLE 0x00 and call trace log file not ended ?? |
4:16PM |
4 |
dial and bridge |
3:30PM |
2 |
Monitoring SIP trunk status on call by call basis |
2:55PM |
0 |
Call Diversion Override |
12:16PM |
1 |
Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial |
|
Monday May 13 2013 |
Time | Replies | Subject |
9:27PM |
1 |
Upgrade from 1.0.x to AsteriskNOW 3.0 |
7:44PM |
1 |
amiDebugger - might make your life easier if you program through the AMI |
9:29AM |
1 |
Sangoma Wanpipe Driver |
|
Sunday May 12 2013 |
Time | Replies | Subject |
2:04AM |
2 |
Integrate Astreisk with SIP interface |
1:40AM |
3 |
time zone setting in asterisk |
|
Saturday May 11 2013 |
Time | Replies | Subject |
9:26PM |
0 |
HD Voice -- connecting Asterisk into HD Voice compatible mobile phone |
2:52PM |
1 |
Which channels are required for FAX, GTALK and Jaber |
11:23AM |
1 |
AMI Originate issue |
8:15AM |
2 |
dahdi driver not getting install |
2:53AM |
0 |
11.4: no incoming gv/xmpp |
2:42AM |
2 |
Tier 1 Service Providers (AT&T, Level 3) |
|
Friday May 10 2013 |
Time | Replies | Subject |
10:27PM |
1 |
ISP trunk session ID? |
8:30PM |
0 |
Job Posting |
6:45PM |
1 |
Asterisk 12 and OPUS Codec |
12:21AM |
1 |
qualify=yes: OPTIONS: How to Change?: `From: "asterisk"` |
|
Thursday May 9 2013 |
Time | Replies | Subject |
8:30PM |
0 |
DID providers |
7:36PM |
0 |
Planned maintenance for community services on May 11, 2013 |
7:23PM |
4 |
monitoring Asterisk 1.8 |
7:13PM |
1 |
chanstats console errors |
3:01PM |
1 |
Elastix vs vicidial |
8:46AM |
2 |
question about CDR |
2:01AM |
0 |
No early media on 302 redirect via two servers |
|
Wednesday May 8 2013 |
Time | Replies | Subject |
8:48PM |
0 |
Transfer cmd via AsyncAGI |
12:55PM |
0 |
Confbridge Dynamic video_mode |
|
Tuesday May 7 2013 |
Time | Replies | Subject |
9:54PM |
1 |
Obtaining high voice quality in dahdi channel |
4:23PM |
2 |
Asterisk and hylafax: how to debug ... |
10:52AM |
1 |
passing '302 moved temporarily' back to the SIP provider |
5:25AM |
1 |
Get Channel Variables in AMI Event NewExten |
1:58AM |
0 |
(no subject) |
1:53AM |
1 |
НА: asterisk-users Digest, Vol 105, Issue 40 |
1:31AM |
1 |
chan_alsa and confbridge |
|
Monday May 6 2013 |
Time | Replies | Subject |
10:54PM |
1 |
What is bootstrap.sh for ? Possible bug in 11.3.0 ? |
6:48PM |
1 |
Installing on an OpenVZ instance |
10:37AM |
0 |
MRCPSynth() change voice |
9:56AM |
0 |
OT - Question on Aastra 6735i - Was: Sip phone displaying caller name while on call |
9:24AM |
1 |
OT - Differences between Aastra 6730i and 6750i series |
1:34AM |
3 |
Joining an astablished call |
|
Sunday May 5 2013 |
Time | Replies | Subject |
3:07PM |
1 |
Testing 911 call |
8:33AM |
1 |
Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'" |
5:33AM |
1 |
GotoIf DIALSTATUS - not working |
4:26AM |
0 |
Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
4:02AM |
0 |
BLF and asterisk Queue |
2:01AM |
0 |
11.4.-rc1: new segfault in iksemel ?? |
12:43AM |
2 |
My new Polycom 450's can't xfer to 4-digit extension |
|
Saturday May 4 2013 |
Time | Replies | Subject |
3:07PM |
1 |
AMI help needed |
8:32AM |
2 |
Cisco 9971 help |
|
Friday May 3 2013 |
Time | Replies | Subject |
6:53PM |
0 |
Digium D70 visual voicemail - won't play |
5:14PM |
1 |
changing ringtones to a group of phones |
7:15AM |
0 |
VoIP Incoming Issue |
|
Thursday May 2 2013 |
Time | Replies | Subject |
9:53PM |
1 |
Playing a sound file during a call |
9:34PM |
1 |
Building Asterisk 11.4.0-rc1 with PJSIP 2.1 |
11:19AM |
2 |
debug strategy for one-way audio calls |
3:04AM |
0 |
Queues with different technologies for members, like Khomp Driver |
|
Wednesday May 1 2013 |
Time | Replies | Subject |
6:11PM |
1 |
Call "stuck" in queue |
9:47AM |
1 |
SMS Scenario |
6:54AM |
0 |
asterisk-users Digest, Vol 105, Issue 39 |
4:33AM |
1 |
multiple provider for incoming |