Monday October 31 2011 |
Time | Replies | Subject |
11:03PM |
0 |
(no subject) |
9:59PM |
11 |
custom automated meeting |
8:47PM |
1 |
Calls from PSTN on SPA3102 |
8:31PM |
1 |
Nat Phone in Asterisk 10 |
2:53PM |
0 |
asterisk and iax2 errors |
2:05PM |
1 |
Starting asterisk turns bash console text white in rxvt |
12:53PM |
0 |
Endpoint Capabilities? |
12:33PM |
3 |
sip issue |
11:09AM |
1 |
Problem with Atxfer for the calling party |
10:14AM |
1 |
Temporarily disabling voicemail recordings (but not greetings) |
|
Sunday October 30 2011 |
Time | Replies | Subject |
10:23PM |
1 |
Meetme does not return back to the dialplan |
1:33PM |
1 |
raju@linux-delhi.org |
|
Saturday October 29 2011 |
Time | Replies | Subject |
6:14PM |
2 |
Issues with Xorcom astribanks when running Ubuntu 11.10 (oneiric) as a quemu/libvirt guest |
8:43AM |
1 |
googleapps calendar |
5:13AM |
3 |
Queue announcements and MOH blanking out on calls from PSTN over IAX2 |
|
Friday October 28 2011 |
Time | Replies | Subject |
8:13PM |
0 |
File permissions mysteriously changed |
7:10PM |
1 |
Network testing for VoIP |
|
Thursday October 27 2011 |
Time | Replies | Subject |
11:02PM |
5 |
Asterisk Executing outbound dial number twice |
3:53PM |
2 |
Recording a meetme conference |
3:30PM |
7 |
Sangoma Card with 16E1 SS7 signaling |
2:48PM |
0 |
DAHDI spans up without physical connections |
2:36PM |
2 |
Check which client access Asterisk using AMI |
12:15PM |
0 |
OPTIONS support for SDP |
10:04AM |
1 |
Unknown warning |
8:58AM |
0 |
Still having trouble to configure gxw4108 with asterisk 1.8 need enlightenment |
12:53AM |
1 |
Asterisk 1.8 RealTime problem with ipaddr field |
12:16AM |
1 |
Tips & best practices for asterisk troubleshooting & parsing logs |
|
Wednesday October 26 2011 |
Time | Replies | Subject |
10:11AM |
0 |
OPTIONS to determine codec capability before an INVITE |
|
Tuesday October 25 2011 |
Time | Replies | Subject |
6:13PM |
0 |
OPTIONS to query endpoint capability |
4:17PM |
0 |
Asterisk 1.8 and CDR Mysql: Unqiue ID and if any commands required in extensions.conf |
1:28PM |
1 |
Concurrent call monitoring |
12:25PM |
1 |
bug in queuemanager? |
11:30AM |
2 |
Asterisk does not accepts SIP registration |
|
Monday October 24 2011 |
Time | Replies | Subject |
2:32PM |
0 |
[asterisk-dev] SIP, NAT, security concerns, oh my! |
11:46AM |
0 |
device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable |
9:46AM |
4 |
Storing a variable at a context and using it in another context |
9:35AM |
1 |
Voicemail: playing a message to give option if need to transfer for operator |
9:19AM |
2 |
${CALLERID(num)} after doing transfer from extension to extension |
|
Sunday October 23 2011 |
Time | Replies | Subject |
2:36PM |
1 |
Questions on IAX client |
9:46AM |
1 |
Peer and User Clarification |
9:22AM |
1 |
how to know RTP por of a SIP client in |
|
Saturday October 22 2011 |
Time | Replies | Subject |
8:38PM |
2 |
Strange behavior over Zap chennels |
12:29AM |
0 |
DAHDI-Linux 2.5.0.2 and DAHDI-Tools 2.5.0.2 Released |
|
Friday October 21 2011 |
Time | Replies | Subject |
3:45PM |
2 |
Skype Messaging with Asterisk 10? |
1:40PM |
0 |
No Voice path during NCS call with Asterisk 10.0.0 |
12:46PM |
0 |
Video Softphone |
11:54AM |
1 |
Asterisk dialplan macro output |
7:46AM |
2 |
how to know RTP por of a SIP client in the dialplan |
|
Thursday October 20 2011 |
Time | Replies | Subject |
10:16PM |
1 |
Astricon: GPG Key signing event |
9:41PM |
1 |
10.0 CallerID question |
7:58PM |
0 |
problems getting chan_alsa.so to run |
7:45PM |
2 |
Cutting noise and voice |
4:01PM |
2 |
question about queues.conf |
2:28PM |
1 |
Unable to build sip pvt data |
1:28PM |
3 |
RTP ports used by Asterisk in dialplan |
11:09AM |
0 |
Video call Setup in Asterisk 1.4.17 |
10:53AM |
1 |
elegant way to change codec whn failing over to another line |
10:35AM |
0 |
Call popups with Thunderbird (and potentially other PIMs) |
9:47AM |
1 |
Sipp and asterisk 10 |
9:05AM |
0 |
ASR & ACD Analysis & Monitoring from Master.csv |
8:12AM |
2 |
Monitor does not work well (little cuts in the audio file) |
|
Wednesday October 19 2011 |
Time | Replies | Subject |
9:35PM |
3 |
Can we use MySQL native connector for ARA? |
8:22PM |
1 |
Asterisk replying 491 |
7:45PM |
1 |
G729 and Dahdi: Inbound forcing ulaw! |
7:25PM |
1 |
Problem E1 PRI |
3:37PM |
1 |
Problem with video phone call, error in sdp media handling? |
3:33PM |
1 |
Asterisk call transfers not working |
1:47PM |
1 |
DTMF fun |
12:41PM |
1 |
strange delay behaviour in SIP call with same codec |
11:58AM |
1 |
Asterisk sponteanous reboot : core dump file |
10:34AM |
0 |
Detecting Special Information Tone in Asterisk |
9:50AM |
5 |
Running as non-root |
5:13AM |
2 |
Outgoing call failure |
1:22AM |
2 |
How to use menuselect.makeopts? |
|
Tuesday October 18 2011 |
Time | Replies | Subject |
11:44PM |
2 |
GoogleTalk Calls |
11:15PM |
0 |
DID and how caller id will appear |
9:21PM |
1 |
nvfaxdetect in 10.0 |
3:46PM |
0 |
How does asterisk parse configuration files |
2:29PM |
0 |
Asterisk IAX Trunking Stops - Too much delay in IAX2 calltoken timestamp from address |
1:35PM |
2 |
Problems during calls |
12:27PM |
1 |
make progdocs |
12:26PM |
1 |
voicemail |
7:30AM |
1 |
Chanspy() not working with group in asterisk 1.4.42 |
7:08AM |
2 |
How does Asterisk parse extensions.conf file? |
5:58AM |
0 |
Chanspy() on group |
|
Monday October 17 2011 |
Time | Replies | Subject |
11:48PM |
1 |
Conference solution to handle 10, 000 participants - possible at all? |
7:22PM |
1 |
Asterisk Centos RPM packages question |
6:36PM |
0 |
Asterisk 1.8.7.1 Now Available (Security Release) |
5:44PM |
0 |
AST-2011-012: Remote crash vulnerability in SIP channel driver |
5:30PM |
1 |
SIP Device and ZAP device |
4:32PM |
1 |
forwarding early media |
2:33PM |
2 |
Request hangup on local channel |
1:29PM |
0 |
Integration asterisk with pabx |
1:28PM |
0 |
chanspy() with group |
|
Sunday October 16 2011 |
Time | Replies | Subject |
9:49PM |
0 |
Voicemail in asterisk 1.8.7, stop working |
7:05PM |
0 |
PRI E1 call termination issue |
6:55PM |
0 |
**OT** Fwd: oFono 1.0 has been released |
5:51AM |
1 |
Any help with these error messages??? |
|
Saturday October 15 2011 |
Time | Replies | Subject |
9:12AM |
3 |
Emulate and script emulation of users calling in/receiving calls, transferring calls &etc |
|
Friday October 14 2011 |
Time | Replies | Subject |
8:00PM |
0 |
Correction: Scheduled Maintenance for Asterisk Project community services |
3:29PM |
2 |
Problem with outbound dialing from remote phone |
10:21AM |
2 |
one way voice with IVR |
9:17AM |
1 |
Get the total amount of lines/channels for a SIP-trunk? |
|
Thursday October 13 2011 |
Time | Replies | Subject |
9:04PM |
7 |
Free ticket for Astricon |
7:25PM |
1 |
MEETME_AGI_BACKGROUND |
7:03PM |
0 |
Cisco 3925 Integrated Services Router |
4:21PM |
2 |
Phones flapping with * and Sonicwall. |
9:09AM |
0 |
many sip dialog/ opened channels. |
12:24AM |
1 |
Asterisk 10 'database show' CLI command |
|
Wednesday October 12 2011 |
Time | Replies | Subject |
9:27PM |
6 |
Binding asterisk to two static IPs |
8:25PM |
0 |
Scheduled Maintenance for Asterisk Project community services |
5:49PM |
1 |
Asterisk HoneyPot |
5:24PM |
1 |
AGI not Installed? |
8:13AM |
1 |
failed to extend from 512 to 676 |
3:42AM |
3 |
FXS ports on TDM410P card... |
|
Tuesday October 11 2011 |
Time | Replies | Subject |
8:15PM |
5 |
permit -- deny not working |
5:11PM |
2 |
Question on meetme and t option |
4:58PM |
11 |
Reporting for Asterisk Call Center |
4:13PM |
0 |
Call deflection with Libpri/Dahdi on BRI/PRI lines |
4:06PM |
3 |
CallerID inconsistently presented through ISDN/cellular networks |
1:53PM |
2 |
Asterisk to asterisk IAX trunk |
11:50AM |
0 |
Failure to write to tcp/tls socket |
11:41AM |
0 |
Asterisk 1.8.7 and VoiceMailMain |
11:16AM |
2 |
BT line: unavailable vs withheld numbers? |
10:55AM |
0 |
Is it recommended to let Asterisk run with "backtrace options" |
9:24AM |
2 |
Queuing strategy |
6:50AM |
1 |
t.38 interop with metaswitch |
5:58AM |
1 |
Asterisk 1.8.7.0- Incorrect information in Queue events-AMI |
|
Monday October 10 2011 |
Time | Replies | Subject |
11:32PM |
2 |
Queue calls to agent end prematurely with diastatus cancel |
9:35PM |
4 |
Maybe slightly OT but.. |
3:40PM |
1 |
Dialout from MeetMe to another conference (Asterisk 1.4) |
12:08AM |
1 |
Which mISDN required for chan_misdn in 1.8 & 10? |
|
Sunday October 9 2011 |
Time | Replies | Subject |
1:40PM |
4 |
Grandstream GXP2000 - copy configuration from handset |
|
Friday October 7 2011 |
Time | Replies | Subject |
5:17PM |
0 |
DIDs in Singapore |
3:03PM |
2 |
Add SIP diversion header in originate from AMI? |
1:01PM |
1 |
Asterisk 1.8.7 and client outside network |
12:46PM |
4 |
Asterisk 1.8.7 and ReceiveFAX |
11:20AM |
0 |
Problem With Playing Busy Tone |
|
Thursday October 6 2011 |
Time | Replies | Subject |
9:25PM |
2 |
Cisco AS5400XM |
8:37PM |
3 |
Digium FFA + Gafachi T38 outgoing issues |
4:25PM |
1 |
Which SIP phone LCD expansion module and >100 asterisk-compatible BLF ? |
2:57PM |
1 |
dahdi show status command not avilable in CLI |
7:12AM |
2 |
A manager event whenever an hint value changes |
12:28AM |
1 |
Questions on Dahdi |
|
Wednesday October 5 2011 |
Time | Replies | Subject |
8:06PM |
0 |
Asterisk 1.8 Manager Perl Script Problem [SOLVED] |
7:36PM |
4 |
making announcements |
7:04PM |
1 |
Different revisions of Digium cards |
6:42PM |
1 |
call pickup |
4:17PM |
4 |
meetme |
8:41AM |
0 |
Passive wait in dialplan |
8:35AM |
1 |
Passive wait in dialplan? |
6:36AM |
1 |
Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing] |
6:30AM |
1 |
Reduce the wav file size |
|
Tuesday October 4 2011 |
Time | Replies | Subject |
6:40PM |
3 |
Asterisk (Trixbox) - VirtualBox - Linux Host |
6:25PM |
1 |
Database Lookup Advice |
5:48PM |
1 |
music on hold |
3:15PM |
0 |
OT - SIP - Toggle to autoanswer after ringing |
2:21PM |
3 |
Beep file with Record |
11:54AM |
0 |
Lag with Call Transfer (Patching) |
8:33AM |
2 |
rtp.conf and Asterisk as a sip agent/client |
|
Monday October 3 2011 |
Time | Replies | Subject |
11:38PM |
3 |
Delay before ringing from PSTN`s call |
10:01PM |
0 |
Asterisk 1.8 Manager Perl Script Problem |
5:51PM |
2 |
Keeping Voice Call Active During Data Connectivity Loss |
|
Sunday October 2 2011 |
Time | Replies | Subject |
4:05PM |
2 |
Asterisk in the Cloud with Diamonds |
3:20PM |
2 |
Sipgate trunk doesn't bridge with other trunk, but works with local extensions |
11:53AM |
1 |
asterisk_1.8.7.0-1digium1 100% CPU |
|
Saturday October 1 2011 |
Time | Replies | Subject |
10:30PM |
1 |
Make asterisk cluster appear and operate as a single server? |
10:02PM |
1 |
Converting dahdi_monitor unit to dbm0 |