| Sunday May 31 2009 |
| Time | Replies | Subject |
|---|
| 8:45AM |
1 |
Ekiga, Twinkle and from where to start with open source |
| 8:32AM |
0 |
Network settings and quality of voice |
| 7:11AM |
1 |
An outside Caller ID not shown, |
| 7:11AM |
1 |
h323 guide for asterisk |
| 6:57AM |
0 |
Multile IP addresses for SIP device |
| 2:15AM |
1 |
Problem releasing call from a SIP extension |
| |
| Saturday May 30 2009 |
| Time | Replies | Subject |
|---|
| 11:16PM |
0 |
question about reinvite |
| 10:09PM |
0 |
POS Modems |
| 9:38PM |
1 |
Problem T.38 |
| 6:35PM |
2 |
Simplex voice on TDM410P |
| 4:17AM |
5 |
Understanding Call Handling In Asterisk |
| 1:10AM |
0 |
compile error for chan_h323 |
| 12:50AM |
2 |
Queue - Multiple Transfer |
| |
| Friday May 29 2009 |
| Time | Replies | Subject |
|---|
| 7:55PM |
2 |
SIP CALL: RTP ENCRYPTION |
| 4:18PM |
0 |
dial out context from incoming iax trunk |
| 3:39PM |
1 |
Asterisk Clustering |
| 3:28PM |
1 |
Attended transfer and dialplan |
| 2:56PM |
0 |
How to read values from another channel ? |
| 2:50PM |
0 |
Logging into queue homed off remote system |
| 9:21AM |
2 |
regarding to field of accountcode |
| 8:33AM |
1 |
IAX2 trunking with Older Asterisk version ? |
| 8:22AM |
3 |
AMI and Originate on 1.6.0.5 |
| 7:57AM |
1 |
connection fail between Service provider's proxy server and my asterisk server |
| 6:27AM |
1 |
how to detect dtmf in meetme |
| 5:22AM |
4 |
asterisk 1.6.1.0 and dial plan changes |
| 5:10AM |
1 |
CAll-limit or incominglimit ????? |
| 4:26AM |
2 |
To: Field |
| 12:38AM |
1 |
Call telco transfer q931 |
| |
| Thursday May 28 2009 |
| Time | Replies | Subject |
|---|
| 10:12PM |
0 |
Best Current Release for Long Term Use |
| 7:50PM |
2 |
zaptel installation |
| 7:08PM |
0 |
Friday at 12 Noon EDT: Jim Van Meggelen on the VoIP Users Conference |
| 6:59PM |
1 |
SIP CALL ENCRYPTION |
| 6:16PM |
2 |
probably an rtfm but... need to dial out to 2 PSTN lines from AMI |
| 1:52PM |
1 |
asterisk 1.4.X, T.38 and NAT |
| 12:49PM |
0 |
Reg AsteriskNow 1.5 Beta Release |
| |
| Wednesday May 27 2009 |
| Time | Replies | Subject |
|---|
| 7:15PM |
0 |
RLT Transfers and NI2 |
| 5:52PM |
1 |
Auto-congesting call due to slow response |
| 5:51PM |
1 |
Playtones Volume |
| 5:24PM |
1 |
setting CDR values on failed calls |
| 2:46PM |
3 |
1.6.0.9: Now "Unable to create ... 'DAHDI'" |
| 1:37PM |
2 |
problem with T.38 media headers |
| 12:54PM |
1 |
TDM400P in PCI-X Slot |
| 12:15PM |
0 |
No full duplex communication ? |
| 11:49AM |
2 |
Pressing number 2 in dialplan |
| 11:32AM |
1 |
stucked calls in asterisk 1.4 |
| 11:09AM |
2 |
AstDB wildcards |
| 8:19AM |
1 |
DAHDI and hangup issue when playing the IVR |
| 6:58AM |
0 |
Delay and Zombie Channels Problem |
| 6:33AM |
1 |
Asterisk memory problems |
| 6:10AM |
1 |
PHP AGI Problems |
| 12:46AM |
3 |
Call in progress tones |
| |
| Tuesday May 26 2009 |
| Time | Replies | Subject |
|---|
| 8:03PM |
0 |
multiple bind ports with TCP and UDP |
| 7:57PM |
1 |
Fax Machines across carrier SIP trunk? General recommendation? |
| 7:30PM |
1 |
Bug or feature in 1.6.1 (Was: How to register with TCP transport) ? |
| 6:30PM |
3 |
Silly (??) question about chan_dahdi |
| 6:30PM |
0 |
Strange message in CLI |
| 5:34PM |
1 |
STUN setting in Asterisk 1.6.X |
| 5:20PM |
0 |
No Voice - only "noisy audio" |
| 4:37PM |
0 |
Suggest good calling service for London |
| 4:08PM |
0 |
How to register with TCP transport ? |
| 3:20PM |
1 |
SIP over VPN |
| 3:13PM |
0 |
CDR after SIP blind transfer. |
| 2:41PM |
3 |
FXS |
| 2:09PM |
5 |
Maximum cable length for analog phone from FXS port |
| 2:07PM |
1 |
A problem in playing sound files |
| 12:46PM |
1 |
Hanging up a call by DTMF |
| 11:39AM |
2 |
Converting Cisco 7961 to SIP |
| 11:32AM |
8 |
Bandwidth management and ADSL router |
| 11:14AM |
2 |
Domains |
| 11:11AM |
0 |
Logging calls made/lost |
| 10:48AM |
1 |
RDNIS question |
| 9:57AM |
1 |
h extension and channel variables |
| 7:52AM |
2 |
asterisk-addon 1.6.1 problem |
| |
| Monday May 25 2009 |
| Time | Replies | Subject |
|---|
| 9:48PM |
1 |
Placing a MWI on-off call... |
| 9:29PM |
4 |
howto store local exchange prefixes ? |
| 5:46PM |
0 |
DNS issues again |
| 4:42PM |
1 |
New tutorial: storing audio recordings per day |
| 12:40PM |
1 |
SIP Trunk groups |
| 12:14PM |
3 |
Problem running Dahdi |
| 9:54AM |
0 |
Connected Number on incoming calls with mISDN |
| 5:34AM |
1 |
Basic Config |
| |
| Sunday May 24 2009 |
| Time | Replies | Subject |
|---|
| 8:00PM |
7 |
Asterisk, SQL Database Update |
| 7:01PM |
2 |
Can I run two instances of asterisk |
| 12:50PM |
0 |
Duplicate DTMF digits |
| 7:16AM |
0 |
RPID on SNOM phones? |
| 4:16AM |
0 |
HDD FULLL |
| |
| Saturday May 23 2009 |
| Time | Replies | Subject |
|---|
| 8:41PM |
0 |
[OT]I like this community |
| 5:57PM |
0 |
hey |
| 4:23PM |
1 |
1.6.0.9: Unknown signalling method 'pri_cpe' ?? |
| 4:03PM |
2 |
1.6.0.9 sip.c: "Serious Network Trouble" ?? |
| 7:39AM |
0 |
Asterisk automatically closing the file descriptor |
| 6:41AM |
1 |
integrating CTI |
| 12:01AM |
1 |
Faxing issues |
| |
| Friday May 22 2009 |
| Time | Replies | Subject |
|---|
| 10:39PM |
1 |
visp multiaccount + firewall configuration problem |
| 9:28PM |
4 |
How to stop a background music |
| 7:31PM |
9 |
DTMF |
| 5:51PM |
2 |
BT ISDN-30 Pri getting 'stuck' on outgoing calls. |
| 5:36PM |
3 |
No response to our critical packet problem |
| 2:49PM |
0 |
Alison Keenan (free British English voice) |
| 1:40PM |
2 |
Indications.conf and tone generation volume |
| 12:33PM |
1 |
VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working... |
| 11:33AM |
1 |
/etc/asterisk/startup.d |
| 10:23AM |
1 |
rasterisk r processes take the rest of my cpu |
| 9:43AM |
1 |
Can't get G.726 to work. |
| 8:27AM |
1 |
Error ON SIP Incoming TOS |
| 8:04AM |
2 |
Memory leak on asterisk 1.6.0.6 |
| 7:23AM |
3 |
Parsing Asterisk's .conf files from Perl, Java or PHP file |
| 7:06AM |
0 |
"...is circuit-busy" message |
| |
| Thursday May 21 2009 |
| Time | Replies | Subject |
|---|
| 9:55PM |
0 |
Free Fax for Asterisk Receiving problem |
| 9:13PM |
0 |
Cheapest price to cuba route !!! |
| 9:06PM |
0 |
calls stuck in AMD even after analysis time |
| 9:02PM |
3 |
Monitor problem, Asterisk 1.2.13 |
| 8:02PM |
0 |
Asterisk-Addons 1.6.0.2 Now Available |
| 8:01PM |
0 |
Asterisk 1.4.25 Now Available |
| 3:36PM |
0 |
Page/Intercom problem |
| 3:27PM |
1 |
Calling party category |
| 2:27PM |
2 |
Zaptel Error |
| 1:05PM |
1 |
reg static build |
| 11:41AM |
1 |
playing media(moh,prompts) from flash player |
| 9:20AM |
1 |
FW: Writing Hangup causes to CDR record |
| 9:18AM |
0 |
Writing Hangup causes to CDR record |
| 9:01AM |
2 |
Jitter buffer question |
| 8:56AM |
2 |
MeetMe not working with GSM codec? |
| 8:04AM |
2 |
Polycom Productivity Suite |
| 6:10AM |
1 |
interruption in queue |
| 3:06AM |
3 |
PSTN Connection |
| 1:52AM |
4 |
Bridging INBOUND PRI to OUTBOUND PRI fails with Monitor() |
| 12:28AM |
1 |
Voicemail playback NEWEST first vs. OLDEST first |
| 12:22AM |
0 |
1.4.24.1 -> 1.6.0.9: segfault |
| |
| Wednesday May 20 2009 |
| Time | Replies | Subject |
|---|
| 8:31PM |
3 |
...is circuit busy message |
| 8:31PM |
1 |
DAHDI fun and games |
| 6:39PM |
2 |
Do I need a SIP Proxy for this? |
| 4:49PM |
1 |
Macro with DIALSTATUS |
| 3:32PM |
2 |
play with varibles |
| 2:56PM |
2 |
MeetMe - Different pin for different user |
| 2:52PM |
2 |
asterisk memory (issue) |
| 2:38PM |
2 |
How to detect switch to voicemail when calling to mobile phone |
| 2:35PM |
0 |
Step-by-Step Asterisk and MeetMe Help |
| 2:04PM |
2 |
asterisk crash on DAHDI error: No more room in scheduler |
| 1:58PM |
0 |
dtmf=info and canreinvite=yes |
| 1:16PM |
1 |
Queue and Dial operation - Common Variables? |
| 12:49PM |
1 |
Pickup with *8 is not working... |
| 12:04PM |
4 |
FritzBox 7270 |
| 10:58AM |
2 |
Problems receiving some faxes in T.38 |
| 10:56AM |
0 |
inbound SIP funnies |
| 10:03AM |
1 |
Channels configuration with DAHDI |
| 9:49AM |
0 |
Feature request: "database show" from manager API [SOLVED] |
| 9:30AM |
1 |
TC400 |
| 6:07AM |
0 |
FaxIn problems |
| 4:44AM |
3 |
Asterisk CCM, CME Integration |
| 3:06AM |
2 |
Manager ExtensionState function |
| 3:04AM |
1 |
What codec/sample rate/resolution...? |
| |
| Tuesday May 19 2009 |
| Time | Replies | Subject |
|---|
| 10:26PM |
9 |
Hang at 5:34 pm EST |
| 10:08PM |
0 |
announcement: chan_nms - channel driver for NMS Communications hardware |
| 8:50PM |
1 |
Dialplan matching problem |
| 8:47PM |
2 |
Feature request: "database show" from manager API |
| 6:57PM |
0 |
cdr record disposition always FAILED |
| 6:22PM |
1 |
Alternative to Adobe Audition 3 for G723 > G711 uLaw ? (old Cool Edit Pro) |
| 6:16PM |
8 |
Ghost ?? |
| 5:51PM |
2 |
Question |
| 3:37PM |
3 |
Dialplan Priorities and Sort Order... |
| 2:47PM |
2 |
Unable to make outbound calls |
| 10:36AM |
0 |
How to access voicemail from deskphone |
| 9:02AM |
1 |
SPA941 |
| 6:04AM |
5 |
OT: SIP hardphone with multi-color BLF |
| 5:47AM |
0 |
Playing audio messages to the callee. |
| 4:35AM |
0 |
Realtime Static on 1.6.1.0 |
| |
| Monday May 18 2009 |
| Time | Replies | Subject |
|---|
| 9:41PM |
1 |
1.6.1.0 with Realtime Mysql |
| 8:30PM |
2 |
From 1.4 to 1.6.0 |
| 8:19PM |
7 |
callcenter / dialer / predictive dialer / vicidial program is now open |
| 7:31PM |
3 |
Number of max SIP calls. |
| 6:36PM |
1 |
meetme |
| 5:48PM |
2 |
Some direction for creating sound files for Asterisk |
| 5:44PM |
0 |
${HANGUPCAUSE} is not printed when call ends or is interrupted |
| 2:57PM |
0 |
no video by Originate |
| 10:57AM |
1 |
Panasonic SIP Phone |
| 10:57AM |
1 |
Problem with Free Fax For Asterisk |
| 9:56AM |
4 |
Open source SIP client |
| 8:36AM |
2 |
Manager API in PHP |
| |
| Sunday May 17 2009 |
| Time | Replies | Subject |
|---|
| 8:35PM |
2 |
Calls Declined |
| 7:07PM |
4 |
Can YOU find a trailing parenthesis? |
| 2:10PM |
1 |
SHARED() variables and <ZOMBIE> channel |
| 10:43AM |
1 |
Capture "Server" header in SIP reply. |
| 8:26AM |
0 |
Correction IRC Channel name - was TODAY May 17 Sunday Asterisk VOIP Conference server & Ekiga for BerkeleyTIP |
| 8:20AM |
0 |
TODAY May 17 Sunday Asterisk VOIP Conference server & Ekiga for BerkeleyTIP |
| |
| Saturday May 16 2009 |
| Time | Replies | Subject |
|---|
| 1:49PM |
2 |
Agent-Login/out in 1.6 |
| 1:21PM |
1 |
howto set up persistent dynamic meetme |
| 11:46AM |
4 |
Fwd: Asterisk With Cisco Voice Router |
| 8:37AM |
1 |
Queue Load, Asterisk Disconnected |
| |
| Friday May 15 2009 |
| Time | Replies | Subject |
|---|
| 9:40PM |
2 |
Logging In / Out Agents on Asterisk 6 ??? |
| 8:15PM |
2 |
change AGI script return result |
| 7:27PM |
0 |
What happened here when transfering a call ? Circuit-busy ??? |
| 6:13PM |
0 |
Mediant 1000 audiocodes and Trixbox |
| 5:39PM |
1 |
meetme dies looking for conf-getconfno |
| 4:27PM |
0 |
Strange SIP Activity |
| 2:38PM |
1 |
Spiral SIP Request problem |
| 1:35PM |
1 |
help a bald guy |
| 1:24PM |
0 |
Asterisk open source project servers have new names! |
| 10:54AM |
0 |
Zap Transfer |
| 10:22AM |
0 |
DNS host name resolution in iax.conf |
| 9:57AM |
1 |
DTMF Recognition |
| 9:06AM |
1 |
Fax t38 capability |
| 6:38AM |
0 |
Friday May 15 @12 Noon EDT with Askozia pbx |
| 4:47AM |
1 |
how to ignore a ring on a line |
| 12:59AM |
0 |
howto build oslec with dahdi-linux-2.1.0.4 or svn? |
| |
| Thursday May 14 2009 |
| Time | Replies | Subject |
|---|
| 11:27PM |
0 |
Cross-compiling asterisk |
| 10:57PM |
1 |
comedian mail |
| 10:44PM |
0 |
polycom soundpoint question |
| 10:37PM |
3 |
how to avoid call waiting? Or check DIALSTATUS before Dial()? |
| 10:11PM |
0 |
Asterisk 1.4.25-rc1 Now Available |
| 9:12PM |
1 |
Goto not matching |
| 8:04PM |
1 |
Parked Calls Problem |
| 3:06PM |
1 |
Digium TDM 400 or Openvox A400P |
| 2:26PM |
0 |
SIP error message |
| 1:25PM |
0 |
FW: [Dean Collins] Joint BarcampNYC4 sessions? |
| 12:58PM |
0 |
Problem with Asterisk 1.4 and Linksys Spa941/962 |
| 11:47AM |
2 |
DAHDI [USERUSERINFO] |
| 12:46AM |
2 |
Problem with Asterisk + TDM410 FXO |
| |
| Wednesday May 13 2009 |
| Time | Replies | Subject |
|---|
| 10:59PM |
1 |
Double dial. |
| 10:55PM |
2 |
Help need to do Lookup from odbc database |
| 9:05PM |
1 |
High Volume US Traffic? Claim DIP Compensation! |
| 8:04PM |
1 |
#-all.gsm |
| 5:39PM |
1 |
Asterisk+a2billing for over 10,000 ext |
| 5:35PM |
0 |
Why asterisk changes RTP destination port when it receives first RTP packet in opposite direction despite canreinvite=no |
| 5:08PM |
2 |
Voicemail and remote directory with SSHFS |
| 4:21PM |
0 |
AstriCon 2009 speaker submissions open! |
| 3:53PM |
2 |
Add Monitor application to call suppresses audio |
| 2:34PM |
1 |
Sangoma FXS dialmap |
| 2:21PM |
0 |
Request for feedback/testing on Multicast RTP Paging |
| 1:00PM |
4 |
Switchvox |
| 12:54PM |
2 |
Proxying from one server to another |
| 9:22AM |
0 |
AGI scripts in Groovy, JavaScript, JRuby or PHP running on the Java Virtual Machine |
| 7:43AM |
4 |
Free Fax for asterisk |
| 7:30AM |
1 |
Asterisk 1.6 T.38 generation towards a Cisco voice router |
| 5:46AM |
1 |
no source on cdr logs in some cases!! |
| |
| Tuesday May 12 2009 |
| Time | Replies | Subject |
|---|
| 11:07PM |
1 |
enum agi interesting problem |
| 8:08PM |
2 |
Hangup()-command does not hang up the line |
| 6:31PM |
2 |
Is anyone keeping up with the versions? |
| 4:26PM |
1 |
Wanting to manipulate SIP response headers |
| 6:18AM |
0 |
Help with radius |
| 3:44AM |
2 |
Asterisk Manager API Action Originate |
| 12:09AM |
1 |
[asterisk-user] Which policy for ISDN BRI support in NT/PtMP ? |
| |
| Monday May 11 2009 |
| Time | Replies | Subject |
|---|
| 8:44PM |
1 |
Anyone with a working pfSense firewall configuration? |
| 5:54PM |
1 |
Ready to put the box on the net |
| 4:02PM |
1 |
PauseMonitor() Hanging Up Call |
| 3:54PM |
1 |
Problems with res_odbc |
| 3:24PM |
3 |
Asterisk w/ Nokia "e" Series Handsets |
| 12:43PM |
0 |
Asterisk 1.6.2.0-beta2 Now Available |
| 12:12PM |
2 |
DTMF received twice |
| 12:09PM |
0 |
No CDR generated for calls to queues with no agents |
| 11:53AM |
0 |
Polycom-330 not displaying line & buddy label? |
| 7:39AM |
1 |
Support of /* */ comments in ael.vim |
| 7:30AM |
3 |
How to write custom functions in AEL2 , |
| 3:21AM |
6 |
Building a System. |
| |
| Saturday May 9 2009 |
| Time | Replies | Subject |
|---|
| 8:22PM |
1 |
Special Dialplan |
| 2:51PM |
2 |
asterisk blade server |
| 9:39AM |
5 |
Rusting Snoms? |
| 8:42AM |
1 |
Incompatible changes to asterisk 1.6 MYSQL addon query syntax |
| 7:49AM |
1 |
A side of Digium you may have never seen |
| 5:56AM |
5 |
Professional Setup.. |
| 4:47AM |
2 |
Unable to run asterisk CLI commands from php |
| 3:30AM |
4 |
VoIP over satellite internet |
| 2:17AM |
1 |
determination of where a call is placed from (physical location) |
| |
| Friday May 8 2009 |
| Time | Replies | Subject |
|---|
| 10:46PM |
2 |
Possible to add Voice delay? |
| 9:39PM |
2 |
Storage capacity for call recording |
| 9:31PM |
1 |
Record all calls |
| 6:55PM |
0 |
Leg-based CDR proposal updated; Major mods |
| 5:54PM |
0 |
G279 install in 1.6.0.9 ? [SOLVED] |
| 5:24PM |
0 |
The efficient way to add MeetMe to pure SIP install ? |
| 4:35PM |
0 |
G279 install in 1.6.0.9 ? |
| 4:11PM |
1 |
Asterisk 1.6.1.0 can't dial out on Sangoma b600 |
| 3:39PM |
2 |
AMOOCON debriefing |
| 3:26PM |
0 |
Proxying comparison |
| 2:23PM |
2 |
Override sip.conf settings in extensions.conf? Possible? |
| 1:20PM |
0 |
DNID Truncated |
| 1:03PM |
0 |
Numeric Hangup Code |
| 1:01PM |
2 |
Configuring SIP Trunk |
| 12:38PM |
0 |
Difference between Transfer and Dial applications |
| 12:26PM |
0 |
Can't GOSUB_RESULT with Dial U() option ... |
| 11:55AM |
2 |
"pri show spans" shows nothing |
| 10:11AM |
0 |
G.722, 1.4 and IAX trunking ... |
| 8:32AM |
2 |
Not receiving voicemail message in mailbox |
| 6:03AM |
0 |
CALL SETUP TIME |
| |
| Thursday May 7 2009 |
| Time | Replies | Subject |
|---|
| 11:21PM |
1 |
func_odbc.c: Unable to execute query |
| 10:45PM |
0 |
Default dahdi fxs behavior |
| 9:03PM |
0 |
[ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/1-1] |
| 7:54PM |
3 |
QoS & VPN |
| 7:41PM |
0 |
Voicemail format - no transcode? |
| 6:20PM |
1 |
Macro arguments on app_queue |
| 5:49PM |
0 |
asterisk-users Digest, Vol 58, Issue 17 |
| 3:44PM |
1 |
How to get meetme participants in dialplan? |
| 2:04PM |
4 |
Voicemail Alert |
| 1:29PM |
3 |
Master.csv |
| 9:50AM |
0 |
How ro store Reject cause |
| 8:48AM |
0 |
pri errors.. |
| 6:14AM |
7 |
Sangoma a104d and channel banks |
| 3:19AM |
3 |
Messaging System |
| 1:00AM |
1 |
Voice Mail Delete Notification |
| |
| Wednesday May 6 2009 |
| Time | Replies | Subject |
|---|
| 10:21PM |
2 |
Cisco 7960G with static config |
| 10:18PM |
1 |
Asterisk with Sphinx |
| 7:50PM |
3 |
Polycom Dialplan Digitmaps |
| 7:05PM |
2 |
Where are 2 letter language values defined? |
| 6:36PM |
0 |
How to get SIP resposnse codes |
| 5:04PM |
4 |
shut down a single PRI on a running Asterisk system? |
| 5:00PM |
0 |
astcc - outgoing call does not hangup properly |
| 4:53PM |
2 |
Understanding Codecs |
| 4:48PM |
2 |
Cisco 7940 phones become unreachable over VPN after a time |
| 4:15PM |
0 |
Bridge() and Goto() and dialplan contexts, oh my! |
| 4:04PM |
1 |
ConfBridge versus MeetMe |
| 12:43PM |
3 |
Questions on X100P/X101P cards |
| 12:17PM |
1 |
precision of wait dialplan application |
| 9:15AM |
0 |
After transfer context |
| 7:16AM |
1 |
Voicemails do not email through asterisk |
| 4:31AM |
0 |
Caller information in Web |
| 2:37AM |
0 |
problems in h323 channels |
| 1:10AM |
0 |
John Todd, Moises Silva Speaking At ClueCon 2009 |
| |
| Tuesday May 5 2009 |
| Time | Replies | Subject |
|---|
| 6:11PM |
1 |
SIP _call_ to Asterisk box |
| 3:38PM |
2 |
chan_mobile and DTMF |
| 2:02PM |
1 |
stop the MOH since asterisk knows that channel is ringing |
| 2:00PM |
0 |
OT: Polycom handset cord detangler |
| 1:52PM |
1 |
"Asterisk cmd MYSQL" app_addon_sql_mysql / performance ? |
| 12:08PM |
0 |
Dial with MOH |
| 10:05AM |
1 |
Beginning to use Asterisk and tests with extensions |
| 6:52AM |
6 |
Preferred language for Asterisk AGIs development ? |
| 2:56AM |
0 |
asterisk-users Digest, Vol 58, Issue 9 |
| 2:52AM |
2 |
1.6.1 app_fax: WARNING T.30 ECM carrier not found ?? |
| 1:45AM |
0 |
need BT102 firmware (current version) |
| 1:12AM |
1 |
Asterisk cdr_odbc problems |
| 1:10AM |
4 |
AMI + AGI for outbound click to dial |
| |
| Monday May 4 2009 |
| Time | Replies | Subject |
|---|
| 8:17PM |
3 |
AGI PHP |
| 6:59PM |
4 |
Cisco phone - can Call manager reflash automatically if we test in Asterisk with SIP? |
| 6:50PM |
3 |
wireless ATA |
| 5:27PM |
1 |
AOC (advice of charge) current status. |
| 4:57PM |
0 |
Channel/Mute |
| 12:08PM |
1 |
hint limitation |
| 9:57AM |
1 |
advice on OrderlyStats (or other cc software) |
| 8:03AM |
1 |
Can someone help me with my IAX-registration |
| 7:08AM |
0 |
how to add applications to 1.6??? |
| 3:53AM |
0 |
question of flite installation |
| |
| Sunday May 3 2009 |
| Time | Replies | Subject |
|---|
| 5:11PM |
0 |
asterisk-users Digest, Vol 58, Issue 5 |
| 12:58PM |
4 |
System noise reduction |
| 12:30PM |
2 |
Asterisk not starting up due to database problems |
| 4:31AM |
0 |
SIP Extension Registration and Security |
| 1:37AM |
2 |
Sangoma Wanpipe Driver Compile for DAHDI Failure |
| |
| Saturday May 2 2009 |
| Time | Replies | Subject |
|---|
| 7:56PM |
1 |
Module not loading |
| 6:30PM |
2 |
Asterisk and ODBC |
| 4:46PM |
0 |
ISDN Error Code 42 |
| |
| Friday May 1 2009 |
| Time | Replies | Subject |
|---|
| 10:49PM |
0 |
2 phone extensions on a single conference room |
| 3:02PM |
1 |
AGI - Ways to create a call |
| 9:57AM |
9 |
LoadAvg , Codec and Bandwidth Utilisation |
| 8:30AM |
5 |
New system for recording - SCSI, SAS or SATA? |
| 8:26AM |
0 |
May 1st @12 Noon: VoIP and home automation and control |
| 8:12AM |
0 |
Failed log in |
| 6:26AM |
1 |
cdr_mysql custom fields ? |