Tuesday July 31 2012 |
Time | Replies | Subject |
3:41PM |
0 |
As Kevin Fleming says "So long, and thanks for all the fish!", we say thank you - and look to the future |
1:59PM |
4 |
So long, and thanks for all the fish! |
12:22PM |
3 |
Digium IP Phone D40 quality, very bad |
8:44AM |
1 |
Static noise on bridged calls to PSTN, although the trunk line is clean on its own |
8:43AM |
0 |
AGI not generating sip 180/183 status |
|
Monday July 30 2012 |
Time | Replies | Subject |
7:42PM |
0 |
Asterisk 10.7.0 Now Available |
7:42PM |
1 |
Asterisk 1.8.15.0 Now Available |
7:06PM |
0 |
111 Useful and/or Funny New Prompts For Asterisk, Courtesy of Allison Smith |
11:02AM |
1 |
problem with 8 port card |
10:08AM |
2 |
libpri error |
9:36AM |
4 |
Multi-Tenant PBX with Asterisk |
9:11AM |
1 |
(no subject) |
8:43AM |
0 |
(no subject) |
|
Sunday July 29 2012 |
Time | Replies | Subject |
7:34PM |
0 |
just did sched_add waitid Warnings 1.8.14.1 |
|
Saturday July 28 2012 |
Time | Replies | Subject |
10:43PM |
3 |
best PRI gateway? |
3:38PM |
1 |
How to send a SIP MESSAGE outside a call |
2:18PM |
2 |
Asterisk + Google Voice |
9:43AM |
3 |
No audio playing back voicemail from odbc |
9:19AM |
2 |
Multiple DID for SIP Trunk |
6:58AM |
2 |
MixMonitor creating file on non-bridged calls with option b |
4:58AM |
3 |
Asterisk on Dynamic IP to a SIP extension |
|
Friday July 27 2012 |
Time | Replies | Subject |
4:55PM |
1 |
CAS T1 - No Ringback |
2:07PM |
0 |
MWI not working - Asterisk 1.8.9.2 |
11:31AM |
0 |
(no subject) |
3:33AM |
1 |
still got ReceiveFax() problem, how to properly setup asterisk fax? |
|
Thursday July 26 2012 |
Time | Replies | Subject |
11:21PM |
1 |
asterisk crash |
10:03PM |
2 |
Call ID of the second call leg |
8:28PM |
2 |
Polycom Presence with Asterisk 1.8.12.0 |
5:34PM |
1 |
Asterisk Realtime issue after registering with x-lite |
1:53PM |
0 |
Realtime Queue and Queue_members |
1:02PM |
3 |
callback on busy |
10:37AM |
1 |
callback - disa |
10:08AM |
2 |
What TTS to use? |
5:22AM |
1 |
Confbridge examples for Asterisk 10? |
1:09AM |
1 |
Dahdi+Redfone+Channel Bank+E&M |
|
Wednesday July 25 2012 |
Time | Replies | Subject |
7:35PM |
1 |
SIP/GSM-gateway recommendation? |
6:24PM |
4 |
Video conferencing? |
1:42PM |
0 |
How to play DTMF digits without blocking |
1:10PM |
1 |
res_odbc crashing asterisk after freetds dsn reconnects |
9:17AM |
2 |
Asterisk |
|
Tuesday July 24 2012 |
Time | Replies | Subject |
9:06PM |
1 |
passing arguments to macros from originate command |
8:46PM |
1 |
echo canceler query |
4:45PM |
5 |
DAHDI problems |
2:19PM |
2 |
Video call using Asterisk |
9:22AM |
0 |
Play announcement during Dial just once |
8:37AM |
2 |
Finding the position of a character in a string |
|
Monday July 23 2012 |
Time | Replies | Subject |
4:23PM |
1 |
Asterisk 1.8.12 and Fax? |
3:40PM |
0 |
Digium Phones: Heads Up |
11:30AM |
2 |
T.38 Gateway |
7:55AM |
2 |
file and on SayNumber() app |
4:22AM |
8 |
PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk |
|
Saturday July 21 2012 |
Time | Replies | Subject |
10:35PM |
1 |
- SIP retransmission problem |
5:46PM |
2 |
Less good call quality using Asterisk |
12:24PM |
0 |
Asterisk and IPTV |
|
Friday July 20 2012 |
Time | Replies | Subject |
5:53PM |
4 |
Voicemail Emails |
2:48PM |
1 |
T.38 (PRI) Fax Debugging |
12:22PM |
0 |
SIP Inband DTMF problem |
7:15AM |
2 |
freepbx asterisk |
|
Thursday July 19 2012 |
Time | Replies | Subject |
11:30PM |
1 |
Route incoming calls |
8:49PM |
1 |
Channel is rsrvd and does not turn off |
2:34PM |
1 |
Agent receives call while making calls |
|
Wednesday July 18 2012 |
Time | Replies | Subject |
9:35PM |
0 |
Can not get my Eicon Diva running with Asterisk... |
4:08PM |
4 |
Remote party ID - sort of working... |
3:06PM |
1 |
Asterisk 1.8.13 / res_fax / res_fax_digium |
11:44AM |
5 |
How to work around asterisk ss7 |
11:30AM |
3 |
Using Asterisk 10.6 as a T38 Fax gateway |
10:53AM |
1 |
Telecom HU cannot callforward to external number |
6:27AM |
4 |
asterisk 1.8 on Solaris/sparc |
|
Tuesday July 17 2012 |
Time | Replies | Subject |
6:10PM |
0 |
Unplanned service outage within next hour for Asterisk community services |
1:29PM |
0 |
How I could Insert a record into the table with REALTIME |
10:24AM |
0 |
Asterisk Capacity |
|
Monday July 16 2012 |
Time | Replies | Subject |
6:47PM |
2 |
any working calling card solution "open source" |
4:44PM |
0 |
Asterisk 10.6.1 Now Available |
4:44PM |
0 |
Asterisk 1.8.14.1 Now Available |
11:48AM |
1 |
Changing auto mixmonitor output file name |
11:35AM |
1 |
Getting error while dialing outside (PRI Connection card TE110P ASTERISK1.8 AND FREEPBX 2.10 |
1:23AM |
1 |
QoS : tos and cos settings |
|
Sunday July 15 2012 |
Time | Replies | Subject |
4:40PM |
0 |
Mediatrix 4400plus ISDN setup |
2:17AM |
0 |
asterisk-users list testing - hopefull2@hotmail.com |
1:46AM |
0 |
asterisk-users list testing - msegovia.ins@gmail.com |
|
Friday July 13 2012 |
Time | Replies | Subject |
9:42AM |
8 |
How to set SIP to auto answer in the dial plan . |
9:00AM |
2 |
Recommended VOIP Monitoring Tools |
8:56AM |
1 |
trouble with asterisk behind router |
4:17AM |
7 |
How to Auto Answer a sip phone |
|
Thursday July 12 2012 |
Time | Replies | Subject |
8:36PM |
2 |
Issue with a ticket system subscribed to asterisk-users |
5:56PM |
0 |
chan_sip sending from wrong source, address when multiple interfaces are used |
5:38PM |
1 |
chan_sip sending from wrong source address when multiple interfaces are used |
12:55PM |
1 |
Asterisk with OpenBTS and mobile phone |
12:53PM |
1 |
weird dect beheaviour multiple handsets |
12:07PM |
0 |
chan_ss7 quick patch to enable RBT |
4:18AM |
0 |
Audiocodes 310HD - on Asterisk Server |
2:08AM |
0 |
wrong RTT QoS information always reported |
|
Wednesday July 11 2012 |
Time | Replies | Subject |
2:08PM |
1 |
Inconsistency in CDR between NO ANSWER and BUSY calls |
7:35AM |
3 |
click to call |
|
Tuesday July 10 2012 |
Time | Replies | Subject |
9:09PM |
0 |
Planned service outage for community services on July 12, 2012 |
8:15PM |
0 |
Asterisk 10.6.0 Now Available |
8:15PM |
0 |
Asterisk 1.8.14.0 Now Available |
4:20PM |
2 |
Flowroute: howto set outbound callerid (ast 1.4)? |
3:42PM |
1 |
10.6.0-rc2: tmp full of core.PBX |
3:07PM |
1 |
connections to manager |
2:04PM |
3 |
channel not available and jump to next group channels |
1:45PM |
1 |
NO AUDIO |
|
Monday July 9 2012 |
Time | Replies | Subject |
3:04PM |
0 |
Queue timeoutpriority=app doesn't working as explained in conf.sample |
1:24PM |
2 |
seems like call is picked and returned to me |
9:26AM |
2 |
SRTP Encryption Per Device |
|
Saturday July 7 2012 |
Time | Replies | Subject |
6:48PM |
2 |
Rookie / sip and extensions |
9:46AM |
1 |
Trixbox or FreePBX or Elastix or PBX In a Flash |
|
Friday July 6 2012 |
Time | Replies | Subject |
9:18PM |
1 |
sip.conf and binaddr issue |
9:18PM |
1 |
Trixbox or FreePBX? |
9:16PM |
1 |
Can I install Asterisk normally and then installing the GUI |
9:01PM |
1 |
Maximum concurrent calls using call files |
3:53PM |
2 |
DAHDI DTMF problem? |
3:15PM |
1 |
Asterisk trying to call a queue with no members |
1:34PM |
1 |
Can I install Asterisk normally and then installing the GUI "asterisk now" |
10:56AM |
2 |
call file and NFS server |
|
Thursday July 5 2012 |
Time | Replies | Subject |
10:25PM |
2 |
FreePBX: How to hangup if the caller did not press # after the voicemail message |
10:20PM |
7 |
FreePBX: using context other than the default context and the generation for the configuration |
9:44PM |
2 |
sip and extensions |
9:04PM |
0 |
AST-2012-011: Remote crash vulnerability in voice mail application |
9:04PM |
0 |
AST-2012-010: Possible resource leak on uncompleted re-invite transactions |
5:47PM |
1 |
touch command not behaving for future calls in asterisk 1.4.41 |
4:00PM |
0 |
Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, 10.5.2-digiumphones Now Available (Security Release) |
2:15PM |
1 |
OT - Multi Function Printer with one touch scanning/emailing |
1:21PM |
4 |
OT - Integration with building intercom systems |
9:34AM |
1 |
sip set debug on always showing error |
7:19AM |
1 |
Elastix 2.3.0.1 |
6:52AM |
1 |
Regrading Speech Recognition. |
2:20AM |
2 |
basic sip quesiton |
|
Wednesday July 4 2012 |
Time | Replies | Subject |
5:50PM |
0 |
Asterisk 1.8.13 PlayTones App |
11:32AM |
1 |
Timer1 RFC and SIP.CONF |
9:32AM |
1 |
Queue Member login from IAX trunk |
|
Tuesday July 3 2012 |
Time | Replies | Subject |
10:35PM |
0 |
QUEUEMEMBER_STATUS incorrect? |
9:08PM |
1 |
Outbound Asterisk calls default directmedia specifications |
8:21PM |
0 |
Free PBX: hangup even if did not dial # in the voicemail |
3:56PM |
1 |
IAX trunking stopped working |
12:30PM |
3 |
How to play different different hold music. |
11:43AM |
3 |
AMR - Segmentation Fault |
7:38AM |
3 |
Centos 6 mISDN |
|
Monday July 2 2012 |
Time | Replies | Subject |
9:15PM |
1 |
Call recording and hosted PBX platform using Asterisk |
5:05PM |
0 |
(no subject) |
2:55PM |
0 |
Allworx 9212 |
|
Sunday July 1 2012 |
Time | Replies | Subject |
8:46AM |
1 |
T.30 Fax session error: Received bad response to DCS or training |
7:34AM |
5 |
port 5060 is blocked by ISP |