Wednesday August 31 2011 |
Time | Replies | Subject |
9:12PM |
0 |
phone + video |
7:49PM |
2 |
Faxes suddenly failing |
6:36PM |
0 |
Asterisk 1.8.6.0 Now Available |
3:43PM |
5 |
cli command show codecs |
|
Tuesday August 30 2011 |
Time | Replies | Subject |
10:19PM |
0 |
Transfer to VoiceMail Asterisk 1.6 |
9:48PM |
0 |
subscriptions from ekiga to asterisk |
6:53PM |
1 |
MOH making calls appear hung up |
4:32PM |
1 |
dahdi with isdn nt_mode, phone no signal still. |
2:44PM |
2 |
Avaya to Asterisk Voice mail |
2:16PM |
2 |
same sip peer as user and provider |
1:44PM |
2 |
T.38 passthru on 1.8.5 |
9:16AM |
1 |
FREE webinar video about Auto-Dialer Business Model (Telemarketing) |
|
Monday August 29 2011 |
Time | Replies | Subject |
8:15PM |
0 |
http manager getconfig crashes on reading large files/categories (50+ lines) |
6:56PM |
3 |
Dragging the dialup customers along, possible? |
3:49PM |
0 |
Scheduled Maintenance for Asterisk Project community services |
10:04AM |
1 |
Asterisk is delaying DTMF (SIP INFO) relay in MeetMe |
8:30AM |
0 |
Presence for channels other than SIP. |
7:25AM |
0 |
Create ou update values ins ASTB when Asterisk is stopped |
7:15AM |
2 |
Dialing multiple endpoints and CallerID presentation |
|
Sunday August 28 2011 |
Time | Replies | Subject |
4:26AM |
0 |
chan_dahdi.conf waitfordialtone |
|
Saturday August 27 2011 |
Time | Replies | Subject |
3:31PM |
2 |
USB or Ethernet based FXO device ? |
12:33PM |
1 |
how to play wav files to all members in konference |
|
Friday August 26 2011 |
Time | Replies | Subject |
6:02PM |
4 |
Good, inexpensive wireless VOIP handsets ? |
5:52PM |
1 |
Looking for ideas for nice **Asterisk** home phone system |
3:53PM |
2 |
SIP Trace to troubleshoot one way of communications |
2:29PM |
1 |
app_sms testers required |
2:18PM |
0 |
Queue Group not forwaring calls to agents |
10:05AM |
2 |
Wanted a modified SIP message body |
9:41AM |
1 |
DMTF via rfc2833 and SIP-INFO simultaneously |
6:21AM |
1 |
Asterisk spontaneous reboot |
12:51AM |
2 |
red alarm on tdm400 fxo (fxs signalled) |
|
Thursday August 25 2011 |
Time | Replies | Subject |
7:58PM |
1 |
"Core Show" being assumed before commands |
6:36PM |
1 |
Thunderbird extension using AMI to dial |
2:20PM |
0 |
Possibly odd sip.conf security requirements. Possible? |
1:26PM |
4 |
Bind SIP over TCP port in asterisk 1.4.42. |
10:33AM |
1 |
security: SIP header spoofing CHANNEL(recvip)? |
6:27AM |
1 |
SIP client on a mobile? |
4:37AM |
0 |
What sort of information does LIDB provide? |
|
Wednesday August 24 2011 |
Time | Replies | Subject |
7:12PM |
0 |
Softhungup missing from Asterisk 1.6.20-1 - *without any notice* |
5:21PM |
3 |
System Command not executing php |
5:18PM |
3 |
1.8.5 Voicemail duration incorrect |
2:17PM |
1 |
[OT] Yealink T26/28/38 and Open-VPN |
2:01PM |
2 |
Asterisk Integration with Android device |
1:16PM |
2 |
How to know how many user is connected |
12:16PM |
0 |
Fwd: Re: Welcome to the "asterisk-users" mailing list |
9:15AM |
0 |
Time-limited calls -- revisited |
4:40AM |
1 |
Invite with replaces handling issue |
|
Tuesday August 23 2011 |
Time | Replies | Subject |
10:28PM |
3 |
Help with pri call giving error. |
|
Monday August 22 2011 |
Time | Replies | Subject |
8:11PM |
12 |
Looking for ideas for nice home phone system |
5:47PM |
0 |
netsock error? some sip clients crashing! |
5:17PM |
1 |
res_odbc with informix |
5:38AM |
1 |
allow anonymous call |
|
Sunday August 21 2011 |
Time | Replies | Subject |
9:29PM |
2 |
espeak module for asterisk |
9:26PM |
0 |
Flite module for asterisk |
|
Saturday August 20 2011 |
Time | Replies | Subject |
4:46PM |
3 |
Outgoing calls fail in chan_gtalk |
4:23PM |
1 |
How is a ping test delay "ms" different from status in Asterisk "sip show peers"? |
12:00PM |
1 |
Sytem Commands not executing |
5:09AM |
1 |
Gtalk channel problem |
|
Friday August 19 2011 |
Time | Replies | Subject |
7:37PM |
5 |
Outbound Dial |
1:14PM |
2 |
Possible Bug? .call files executing multiple times |
11:08AM |
0 |
Can't use SmartVoip/JustVoip after update |
3:11AM |
0 |
ChanSpy on 1.6.2.20 |
1:12AM |
3 |
Playback while dialing out |
|
Thursday August 18 2011 |
Time | Replies | Subject |
3:29PM |
0 |
cisco 7945 sip lines on 2 different asterisk servers |
12:42PM |
2 |
Asterisk 1.8 SIP_CAUSE performance regression |
11:42AM |
1 |
issue with the detection of the call status after sending it using Orginate (DAHDI/1/...., app, ... |
10:31AM |
1 |
1.8.5 CLI colors are gone? |
10:20AM |
1 |
How to get presence using AMI |
|
Wednesday August 17 2011 |
Time | Replies | Subject |
10:01PM |
3 |
Polycoms rebooting themselves |
6:35PM |
1 |
Spy just a range of extensions |
1:08PM |
1 |
Asterisk 1.8.5.0 |
|
Tuesday August 16 2011 |
Time | Replies | Subject |
3:16PM |
2 |
Asterisk scaling |
2:31PM |
1 |
PRI Problem |
1:37PM |
1 |
Strange Asterisk notices |
9:38AM |
1 |
Celebrating 10 years SER SIP Router in Vienna: 8th September 2011 |
9:20AM |
0 |
AGI dialplan |
8:34AM |
1 |
Speed Dials Management.... |
12:40AM |
1 |
Asterisk -> Office 365 Unified Messaging... anyone done it? |
|
Monday August 15 2011 |
Time | Replies | Subject |
11:29PM |
0 |
1.4.38 passing a Regular expression containing a pipe character to a macro ? |
10:54PM |
0 |
SIP trunk trouble. Please help. |
9:48PM |
3 |
AMI Commands - not working as Expected, Maybe??? |
5:46PM |
5 |
Linksys/Cisco 504G randomly restarts |
5:20PM |
1 |
issue with some numbers |
2:11PM |
0 |
app_swift for Asterisk 10 |
1:09AM |
3 |
Queue Breakout Input being Ignored |
|
Sunday August 14 2011 |
Time | Replies | Subject |
10:17PM |
0 |
dynamically alter list of offered codecs (for faxing) |
5:26PM |
0 |
Adhearsion 1.2.0 Released at Lone Star Ruby Conference V |
4:26PM |
2 |
DAHDI-linux-complete on CENTOS kernel 2.6.32 |
7:36AM |
1 |
1.6.2.20 ${DIALSTATUS} disagrees with CDR(answered) |
7:08AM |
1 |
callgroup more than 63 |
12:12AM |
2 |
asterisk-users Digest, Vol 85, Issue 23 |
|
Saturday August 13 2011 |
Time | Replies | Subject |
11:34PM |
3 |
Echo problem in the analoge lines |
11:28PM |
4 |
DID to display the calling number |
|
Friday August 12 2011 |
Time | Replies | Subject |
8:59PM |
1 |
One way audio when using originate... |
8:23PM |
1 |
Message prints even if verbose level is Zero |
3:59PM |
0 |
Conference calls through web-interface with moderation using Asterisk? |
3:23PM |
1 |
.call files in /var/spool/asterisk/outgoing |
1:57PM |
5 |
Queuing outgoing calls |
12:37PM |
0 |
Interrupting a call in progress? |
11:06AM |
1 |
Queue agent login notification |
|
Thursday August 11 2011 |
Time | Replies | Subject |
10:40PM |
0 |
Asterisk 1.8.5 - Ubuntu Pkg from diguim Repo - OPENH323 error |
9:31PM |
0 |
experiences sharing |
8:55PM |
1 |
TLS Error on 1.6 and 1.8 |
7:04PM |
2 |
Where to proceed next |
9:23AM |
0 |
Background music during a call |
8:07AM |
5 |
Trouble with *8 Pickup |
7:03AM |
3 |
Problem setting for incoming termination |
6:27AM |
1 |
Any Method for capturing ISUP packets in DAHDI/ASTERISK |
5:50AM |
1 |
Asterisk reporting |
1:12AM |
1 |
Asterisk+internal phones+recorded messages |
|
Wednesday August 10 2011 |
Time | Replies | Subject |
12:30PM |
1 |
Asterisk 1.8 Install Problem |
11:57AM |
0 |
Unable to enable echo cancellation on channel 1 (No such device) |
9:59AM |
1 |
BT killed Ribbit |
7:23AM |
3 |
ulimit |
|
Tuesday August 9 2011 |
Time | Replies | Subject |
8:38PM |
0 |
A lA orden los libros!!! |
7:38PM |
2 |
FAX Issues |
12:16PM |
5 |
block all numbers begin by 00 and 1 |
10:36AM |
0 |
Look c8m9kl |
|
Monday August 8 2011 |
Time | Replies | Subject |
6:57PM |
1 |
1.8 issues with Local Bridging |
4:37PM |
4 |
SRV question |
2:52PM |
3 |
CEL and MySQL |
1:41PM |
1 |
Version 1.8 strange expression error |
9:46AM |
1 |
DAHDI Callerid and transfer problem |
8:43AM |
0 |
Timer B in sip.conf cannot be changed |
6:30AM |
0 |
DAHDI-Linux 2.5.0 and DAHDI-Tools 2.5.0 Released |
5:18AM |
0 |
Reconfiguring smoother module in 1.8 |
2:32AM |
2 |
Polycom and auto answer |
|
Sunday August 7 2011 |
Time | Replies | Subject |
6:49PM |
1 |
fail to correctly build 1.8.5 ?? |
2:49AM |
2 |
Dahdi does not build against Kernel 3.0.0 |
|
Saturday August 6 2011 |
Time | Replies | Subject |
11:11AM |
0 |
openh323 or ooh323 |
6:35AM |
10 |
Firewall Issue |
|
Friday August 5 2011 |
Time | Replies | Subject |
8:32PM |
0 |
Asterisk 1.6.2.20 Now Available |
8:31PM |
1 |
Receptionist Extension cannot be Pickup()'ed |
8:10PM |
0 |
asterisk-users Digest, Vol 85, Issue 10 |
6:20PM |
1 |
No more CDR record for simple Hangup? |
3:41PM |
2 |
Assistance sending mass sms to cellphones |
2:43PM |
2 |
error: Autodestruct on dialog |
2:18PM |
0 |
Fax Detection with chan_ss7 |
12:53PM |
1 |
Ring delay problem |
12:26PM |
0 |
Audio when a call is on hold. |
12:11PM |
0 |
Asterisk 1.8.5 eventfilter not working in manager.conf |
11:17AM |
1 |
ASterisk is Going stop whenever restart the server |
11:16AM |
0 |
Call Park announcing to Caller rather than callee. |
11:02AM |
0 |
(no subject) |
10:55AM |
1 |
Send Refer with replaces from asterisk |
10:53AM |
2 |
display name |
8:32AM |
1 |
Custom Dialplan |
|
Thursday August 4 2011 |
Time | Replies | Subject |
7:39PM |
1 |
Answering machine answers after pickup a phone. |
4:59PM |
1 |
REGISTER forwarding problem |
3:20PM |
4 |
pickupgroup |
2:36PM |
1 |
burned module X400M |
8:21AM |
2 |
Customizing sip response codes for PBX Sip trunk |
7:28AM |
1 |
wctdm24xx IRQ missing |
2:02AM |
0 |
UK BT ISDN30 settings? |
|
Wednesday August 3 2011 |
Time | Replies | Subject |
2:49PM |
2 |
Need a volunteer for a Patch |
2:12PM |
16 |
Increasing volume ? |
1:20PM |
2 |
Debugging Sip |
12:45PM |
0 |
trustrpid in sip.conf |
11:36AM |
1 |
Changing sip response codes |
9:55AM |
1 |
Know the number of concurrent dial ? |
9:32AM |
0 |
Barging in PBX |
9:31AM |
0 |
*8 pickup not releasing channel |
8:06AM |
1 |
dundi |
7:28AM |
2 |
Asterisk reload, to execute file |
6:01AM |
1 |
Segmentation Fault |
5:41AM |
0 |
CHANSPY |
2:15AM |
2 |
snom and srtp |
1:38AM |
0 |
Outgoing call issue in D-link DPH-80 ip phones |
1:22AM |
1 |
TE410P hardware problems |
|
Tuesday August 2 2011 |
Time | Replies | Subject |
7:51PM |
1 |
Codec negotiation issue (no audio format found to offer) |
5:06PM |
3 |
Problem with (asterisk1.8-iksemel1.4-GoogleVoice) |
4:42PM |
2 |
use ILBC installed from asterisk yum repositories |
2:23PM |
0 |
SMS within asterisk users |
9:58AM |
3 |
MixMonitor and attended transfers |
8:43AM |
0 |
How to stop Dial from waiting for extra digits if number is incomplete. |
8:29AM |
0 |
asterisk speech to text and text to speech? |
|
Monday August 1 2011 |
Time | Replies | Subject |
8:24PM |
1 |
Problems with AMI connections (Asterisk 1.8.3.2) |
5:02PM |
2 |
T38 Fax |