Thursday January 31 2013 |
Time | Replies | Subject |
5:56PM |
0 |
(SOLVED) Call parking in a multi-tenant system |
4:11PM |
0 |
3Com 3101SP phone on Asterisk? |
7:25AM |
3 |
OT - Chan-mobile -Bluetooth dongle on remote LAN workstation |
1:57AM |
1 |
IVR Menu Sounds |
|
Wednesday January 30 2013 |
Time | Replies | Subject |
6:21PM |
3 |
Asterisk Messaging Refuses To Work! |
5:26PM |
5 |
sip register peer (the quest for near 100% availability) |
6:28AM |
2 |
#!/usr/bin/php -q unknown command |
|
Tuesday January 29 2013 |
Time | Replies | Subject |
7:44PM |
1 |
Auto Provisioning |
4:01PM |
0 |
Modify from header for anonymous call |
11:58AM |
1 |
Fast AGI library/support for C & C++ |
10:34AM |
2 |
round-robin in asterisk 1.4 |
|
Monday January 28 2013 |
Time | Replies | Subject |
6:06PM |
1 |
Configuration Required for Remove Queue Member |
1:55PM |
3 |
RPM updates |
|
Saturday January 26 2013 |
Time | Replies | Subject |
2:06PM |
1 |
Complex Call Distribution |
4:52AM |
1 |
asterisk 11's app_page options |
|
Friday January 25 2013 |
Time | Replies | Subject |
4:39PM |
1 |
How to implement "priority queuing" within a single queue ? [SOLVED] |
4:22PM |
2 |
How to implement "priority queuing" within a single queue ? |
4:09PM |
2 |
Quoting error with gotoiftime |
2:20PM |
1 |
Frames with invalid timing info |
1:09PM |
0 |
CEL / CELGenUserEvent via AGI / no error and no cel entry |
|
Thursday January 24 2013 |
Time | Replies | Subject |
10:43PM |
0 |
(no subject) |
8:44PM |
5 |
"clicking" sound with alaw codec |
6:12PM |
0 |
Planned maintenance for community services on January 24, 2013 |
6:11PM |
1 |
How to assign the button on the IP Phone |
4:46PM |
7 |
question on SIP trunk and AMI to place call |
3:34PM |
2 |
g723 transcoding |
3:03PM |
1 |
How to assign the button on the IP Phone to a feature? |
2:30PM |
2 |
Asterisk 11 / Missing Application SetCallerPres |
10:11AM |
1 |
How configure asterisk server extension.conf. |
9:37AM |
3 |
DECT Solution |
2:13AM |
2 |
Uninitialized variable in main/pbx.c? |
|
Wednesday January 23 2013 |
Time | Replies | Subject |
10:32PM |
2 |
Asterisk 11 with t38modem 2.0: "488 Not acceptable here" |
9:58PM |
1 |
DAHDI: How to supress notification of changing CallerID on transfer? |
5:20PM |
3 |
Is there a need to secure RTP ports? |
5:04PM |
1 |
Problems with 'i' extension |
4:33PM |
0 |
Digium Phones. Can BLF keys be made to function during conversation? |
10:09AM |
1 |
Execute a script outside Asterisk |
9:41AM |
2 |
Realtime vs Static Files |
3:23AM |
1 |
DPMA and Sending fake auth rejection for device |
2:36AM |
0 |
2. Re: Does Asterisk support remove header from sip message? |
|
Tuesday January 22 2013 |
Time | Replies | Subject |
11:22PM |
2 |
Asterisk voicemail minimum length / silence settings |
10:27PM |
5 |
Integration with Social Media, Email and Web call center |
10:25PM |
0 |
Asterisk 11.2.1 Now Available |
10:24PM |
0 |
Asterisk 10.12.1 Now Available |
10:24PM |
0 |
Asterisk 1.8.20.1 Now Available |
8:27PM |
4 |
Asterisk, Digium phones, and voicemail. |
7:26PM |
0 |
Audio not decrypted between Asterisk and encrypted client |
9:40AM |
2 |
Blind transfer behavior - Asterisk 1.8 and 10 |
7:57AM |
1 |
two steps when calling from web! |
7:54AM |
2 |
Details process to configure Asterisk in CENTOS |
3:38AM |
4 |
Google voice with no voice |
|
Monday January 21 2013 |
Time | Replies | Subject |
11:22PM |
1 |
Queues and distributed device state over WAN |
10:10PM |
1 |
Function DB_KEYS() |
10:07PM |
2 |
MoH with message on intervals |
7:34PM |
0 |
Planned service outage for community services on January 21, 2013 |
6:03PM |
8 |
Capture queue agent drop and put caller back in queue |
5:35PM |
0 |
Minimal pass-through T1 configuration? |
1:21PM |
2 |
OT - Desktop SIP phone with OpenVPN client |
7:22AM |
1 |
Does Asterisk support remove header from sip message? |
|
Saturday January 19 2013 |
Time | Replies | Subject |
1:25AM |
2 |
recrding calls |
|
Friday January 18 2013 |
Time | Replies | Subject |
8:32PM |
0 |
'Slower but cleaner' G711 option |
7:26PM |
1 |
Any timeframe for the release of the Asterisk 11<->Lumenvox connector bridge? |
5:44PM |
0 |
Voicemail and recordings storage: best practices |
4:06PM |
3 |
Annoying delay after main server goes down |
3:22PM |
2 |
rtptimeout: how to detect it in dialplan? |
3:22PM |
0 |
Only silence trying to play streaming MOH |
5:26AM |
2 |
Delay in call asterisk |
3:28AM |
1 |
Open source asterisk GUI options |
|
Thursday January 17 2013 |
Time | Replies | Subject |
11:32PM |
2 |
Mail list settings? |
11:26PM |
0 |
fw: Re: Conf Bridge |
8:42PM |
0 |
Email and web chat call center |
8:05PM |
3 |
Need Help |
8:02PM |
1 |
Conf Bridge |
4:29PM |
1 |
How to give users the capability to set CDR userfield for some calls |
12:29PM |
0 |
How to exclude non-queue calls from recording ? |
11:27AM |
2 |
Question about "directmedia" or "canreinvite" in sip.conf |
8:54AM |
1 |
g729 codec over SIP Trunk between CCM and Asterisk |
|
Wednesday January 16 2013 |
Time | Replies | Subject |
9:44PM |
1 |
N Priority in Mysql |
6:20PM |
1 |
Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start |
1:28PM |
1 |
Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop |
11:29AM |
1 |
OT - Which Call Center class wireless headet with bluetooth connectivity ? |
12:06AM |
2 |
special conference room |
|
Tuesday January 15 2013 |
Time | Replies | Subject |
9:02PM |
4 |
Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable |
8:58PM |
1 |
Call parking in a multi-tenant system |
7:11PM |
0 |
Planned service outage for community services on January 16th, 2013 |
6:32PM |
0 |
Asterisk, DNS SRV, 1.8 |
3:08PM |
0 |
Reporting Utility |
3:08PM |
0 |
Telephony card in Thecus N4800 |
11:01AM |
1 |
POSTing recorded audio stream |
9:05AM |
1 |
AGI command |
8:59AM |
0 |
param sayduration of mailbox |
4:36AM |
1 |
Followme Killing Asterisk |
|
Monday January 14 2013 |
Time | Replies | Subject |
10:29PM |
1 |
gtalk only working with ulaw??? |
9:22PM |
0 |
Asterisk 11.2.0 Now Available |
9:21PM |
0 |
Asterisk 10.12.0 Now Available |
9:21PM |
0 |
Asterisk 1.8.20.0 Now Available |
3:42PM |
0 |
Asterisk 10.12.0 - Final Maintenance Release of Asterisk 10 |
3:33PM |
8 |
block one number in incoming calls |
7:33AM |
0 |
asterisk is responding with 200 ok to Subscribe. |
6:23AM |
1 |
php programming for working with asterisk |
|
Sunday January 13 2013 |
Time | Replies | Subject |
2:17AM |
2 |
Recorded reminders |
|
Friday January 11 2013 |
Time | Replies | Subject |
9:06PM |
3 |
How often to restart Asterisk... |
3:50PM |
2 |
Single = sign and double == sign.What is the difference and when to use the two properly? |
3:34PM |
2 |
FW: Correct auth, but based on stale nonce received from |
11:35AM |
4 |
Set Language for VoiceMailMain |
10:22AM |
2 |
Which tool to edit custom reports from CDR and queues logs ? |
8:22AM |
1 |
Undefined problem Asterisk problem |
1:28AM |
1 |
Playing music through VoIP handsets while on hook |
|
Thursday January 10 2013 |
Time | Replies | Subject |
10:32PM |
0 |
Manager event for hint subscribe |
7:22PM |
1 |
Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2 |
5:00PM |
2 |
Asterisk |
2:23PM |
4 |
Call Disconnected by Caller or Agent |
2:03PM |
1 |
Your thoughts and opinions on Asterisk 11 for production use |
1:04PM |
0 |
403 for SUBSCRIBE methos |
|
Wednesday January 9 2013 |
Time | Replies | Subject |
6:16PM |
13 |
DIDForSale spam |
3:05PM |
6 |
IVR platform for a mobile operator |
6:06AM |
0 |
.call file retry issue in Asterisk-10.11.1 |
|
Tuesday January 8 2013 |
Time | Replies | Subject |
3:24PM |
7 |
Streaming/Recording audio |
11:36AM |
1 |
Monitor extensions status. |
7:14AM |
0 |
Asterisk 11; WEBRTC firefox nightly build fingeprint |
|
Monday January 7 2013 |
Time | Replies | Subject |
10:17PM |
1 |
echo from channel bank |
9:57PM |
5 |
IAX2 support of video |
8:10PM |
5 |
Paging unit suggestions |
6:22PM |
7 |
Outoing Calls Motif Google Voice Calls Ring After Pick-up |
1:30PM |
0 |
Member stay busy after hangup a call in queue |
|
Sunday January 6 2013 |
Time | Replies | Subject |
11:36PM |
1 |
Malicious traffic comming from 37.75.210.90 |
9:18AM |
2 |
PRI (Primary-NTT) |
1:55AM |
1 |
Get CONNECTEDLINE info from other Asterisk system via IAX2 |
|
Saturday January 5 2013 |
Time | Replies | Subject |
10:16PM |
3 |
Limit registration concurrency per friend |
1:37AM |
8 |
Detect Low Quality Calls - Realtime |
|
Friday January 4 2013 |
Time | Replies | Subject |
10:45PM |
2 |
Calender and EWS with shared calenders |
8:56PM |
1 |
Unable to build DAHDI |
3:16PM |
0 |
Asterisk + Huawei K3765 |
2:21PM |
0 |
T38MaxBitRate issue on fax passthrough |
2:17PM |
1 |
Polycom IP6000 upgrading and looping |
1:39PM |
2 |
MaxCallBR Peer Setting |
10:26AM |
0 |
WebM / VP8 support |
|
Thursday January 3 2013 |
Time | Replies | Subject |
9:13PM |
3 |
faxdetect on/off on the fly? |
5:47PM |
5 |
Moving User Agent To Remote Location |
3:38PM |
1 |
Build asterisk for VIA C3 |
3:13PM |
2 |
Verizon SIP "trunking" Field Trial |
11:08AM |
0 |
Asterisk 11.1.2 Now Available (Security Release) |
6:23AM |
1 |
User busy issue in A400P 4 FXO card |
|
Wednesday January 2 2013 |
Time | Replies | Subject |
11:39PM |
3 |
DAHDI: How to know since when it is used? How to shutdown after max time? |
11:08PM |
0 |
Telecom Best Practices |
10:49PM |
8 |
Auto ban IP addresses |
10:45PM |
0 |
Speaking opportunities at Digium Asterisk World/IT Expo - Miami Beach - 1/31 and 2/1 |
9:24PM |
0 |
AST-2012-015: Denial of Service Through Exploitation of Device State Caching |
9:23PM |
0 |
AST-2012-014: Crashes due to large stack allocations when using TCP |
8:30PM |
3 |
Asterisk as answering machine |
3:27PM |
0 |
Asterisk 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones, 11.1.1 Now Available (Security Release) |
2:55PM |
6 |
Asterisk for Razberry Pi |
2:01PM |
3 |
Dialing out and recording |
|
Tuesday January 1 2013 |
Time | Replies | Subject |
12:29AM |
0 |
Question on Confbridge menu item dialplan_exec |