| Wednesday November 30 2011 |
| Time | Replies | Subject |
|---|
| 10:39PM |
1 |
Installing asterisk on a server vs appliance(e.g digium mypbx) |
| 9:20PM |
2 |
Walkie talkie to sip phone interface |
| 8:47PM |
3 |
how to find out one way latency |
| 7:55PM |
1 |
Sound files with MixMonitor not playable with Media Player |
| 4:08PM |
0 |
vall directly extensions from E1-PRI line |
| 3:26PM |
2 |
Issue with Polycom SPIP650 and its sidecar |
| 1:38PM |
1 |
Question on PAP2 linksys showing off-hook |
| 12:25PM |
1 |
s/n ratio detection etc... |
| 6:51AM |
1 |
Best VoIP conferencing phone ? |
| 1:05AM |
0 |
SLA and polycom |
| |
| Tuesday November 29 2011 |
| Time | Replies | Subject |
|---|
| 11:03AM |
3 |
When dialing the number, I need to see it in the Cisco LCD Phone |
| 10:47AM |
2 |
SIM to E1 gateway, and SMS gateway |
| 10:38AM |
2 |
OT: Does IEEE 801.2q include VLAN trunking? |
| 9:50AM |
3 |
app_mysql and asterisk 1.4 |
| |
| Monday November 28 2011 |
| Time | Replies | Subject |
|---|
| 8:30PM |
1 |
Recommendations |
| 3:29PM |
2 |
Call Parking Realtime |
| 1:32PM |
0 |
AsteriskGUI - Which version for 1.8 ? |
| 8:47AM |
1 |
Queue-Tip/Adhearsion installation tip |
| 1:22AM |
1 |
centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing? |
| |
| Sunday November 27 2011 |
| Time | Replies | Subject |
|---|
| 10:38PM |
0 |
Displaying entered digits in the LCD of the IP Phone when is requested to enter it |
| 10:04PM |
0 |
SMS problems. |
| 5:47PM |
6 |
Does Asterisk alter the Headers of INVITE Message |
| 6:33AM |
0 |
Automatic IVR generator (alpha) |
| |
| Saturday November 26 2011 |
| Time | Replies | Subject |
|---|
| 11:55AM |
3 |
A new hack? |
| |
| Friday November 25 2011 |
| Time | Replies | Subject |
|---|
| 10:11PM |
1 |
android won't play wav49: how to change format |
| 6:00PM |
0 |
Install Adhearsion on Debian [SOLVED] |
| 5:19PM |
1 |
Install Adhearsion on Debian |
| 3:32PM |
1 |
hwo to stok variable wiith menu |
| 1:18PM |
1 |
Sporadic yellow alarms in dahdi_tool output |
| 7:57AM |
3 |
Call Parking |
| 5:18AM |
1 |
Rgarding asterisk 10 stable release |
| |
| Thursday November 24 2011 |
| Time | Replies | Subject |
|---|
| 10:36AM |
2 |
PSTN Frequency parameters |
| |
| Wednesday November 23 2011 |
| Time | Replies | Subject |
|---|
| 12:03PM |
1 |
MWI for non-subscribed Realtime peers? |
| 11:37AM |
2 |
Is it possible call land into extensions.ael configuration file not in extensions.conf |
| 10:39AM |
1 |
DONT_OPTIMISE, BETTER_BACKTRACES and performance |
| 7:16AM |
3 |
safe_asterisk ? |
| 6:40AM |
0 |
how to call a ring group via the dial plan language in asterisk? |
| 2:46AM |
2 |
Sending more than one call the agent while he is already in a call !! ringinuse=no/yes |
| |
| Tuesday November 22 2011 |
| Time | Replies | Subject |
|---|
| 11:22PM |
1 |
Receive sms via bluetooth with chan_mobile and an Android phone? |
| 11:00PM |
1 |
Follow me unreachable message default |
| 10:34PM |
4 |
How to program a 100ms delay between the ringing of queued calls w/ ringall |
| 3:00PM |
2 |
sip show peers |
| 2:40PM |
1 |
Asterisk refuses INVITE (401) and I don't know why |
| 7:14AM |
2 |
Resell VoIP Servcies |
| 6:57AM |
0 |
no sound with ICES ? |
| |
| Monday November 21 2011 |
| Time | Replies | Subject |
|---|
| 10:17PM |
1 |
AEX410P drops DTMF digits |
| 5:50PM |
3 |
difference between playback and background? |
| 5:43PM |
1 |
CDR uniqueid - across multiple servers? |
| 5:30PM |
0 |
no more trunk with asterisk |
| 4:56PM |
1 |
queue ring delay |
| 12:13PM |
2 |
CDR mysql with asterisk 1.4 |
| 10:35AM |
1 |
video calls not working |
| 10:20AM |
1 |
vigor 2920 problems |
| 10:12AM |
0 |
Dependencies for BETTER_BACKTRACES on Centos 5.6 |
| 6:42AM |
7 |
How to use password file with Authenticate Application |
| 6:38AM |
2 |
Continue AGI after Dial() following caller hang up? |
| |
| Sunday November 20 2011 |
| Time | Replies | Subject |
|---|
| 11:32PM |
0 |
hello |
| 8:49PM |
1 |
check if devices reachable in queue |
| 7:25PM |
4 |
Deleting AstDB family at start |
| 2:43AM |
1 |
SIP registration issues |
| |
| Saturday November 19 2011 |
| Time | Replies | Subject |
|---|
| 8:39PM |
0 |
Queue: The call keep going to agent until the agent drop the call |
| |
| Friday November 18 2011 |
| Time | Replies | Subject |
|---|
| 8:02PM |
2 |
Setting outbound PRI Callerid with Asterisk 10.0.0-beta2 |
| 7:23PM |
4 |
Using asterisk with DSP chips |
| 7:10PM |
1 |
I want to participate in development |
| 2:33PM |
1 |
Polycom Phantom Ringing |
| 10:29AM |
1 |
Call files and spool directiory shared amongst several asterisk servers |
| 10:16AM |
1 |
Asterisk Developer |
| |
| Thursday November 17 2011 |
| Time | Replies | Subject |
|---|
| 6:13PM |
0 |
2 same sip extension number on 2 asterisk - call not passing on certain condition |
| 4:10PM |
1 |
Question about Read() application |
| 3:44PM |
0 |
DTMF dropping in Read Command |
| 2:20PM |
0 |
10-rc2: how to debug dropped calls? |
| 1:17PM |
1 |
How to unregister a sip trunk |
| 1:00PM |
0 |
ast_debug messages not showing up |
| 11:30AM |
2 |
Fax not detected by Asterisk |
| 11:00AM |
3 |
AMI: anything to glue originate to events? |
| 6:34AM |
0 |
Use Polycom FX with Asterisk |
| 1:46AM |
0 |
Grandstream HT503 colgado |
| |
| Wednesday November 16 2011 |
| Time | Replies | Subject |
|---|
| 6:49PM |
5 |
Polycom Attended Transfer |
| 6:06PM |
2 |
polycom soundpint ip650 question |
| 1:46PM |
4 |
Limit monthly calls by context |
| 12:01PM |
9 |
Skype For Asterisk (SFA) |
| 10:18AM |
1 |
Server-to-server BLF |
| 9:23AM |
3 |
Does Asterisk Support SIP Video Call ? |
| 5:06AM |
1 |
Asterisk Send out SIP Invites to external network- howto |
| |
| Tuesday November 15 2011 |
| Time | Replies | Subject |
|---|
| 3:58PM |
3 |
Standard UIDs, especially for asterisk? |
| 3:12PM |
2 |
More than one route to a destination |
| 2:47PM |
2 |
Forcing a CODEC |
| 1:28PM |
4 |
Multiple SIP endpoint registrations |
| 10:56AM |
2 |
Goto Queue, does not work, it should play message or any thing |
| 6:01AM |
2 |
Calling an independent gateway from asterisk |
| 1:45AM |
1 |
asterisk bin file may change when running |
| |
| Monday November 14 2011 |
| Time | Replies | Subject |
|---|
| 11:51PM |
2 |
trouble with sip connection and registration |
| 9:51PM |
2 |
How do extensions "stay registered" |
| 6:47PM |
1 |
Becoming a CLEC |
| 3:40PM |
0 |
Dial Application / hangup with option h or H / featuremap / more than 1 valid key |
| 2:08PM |
2 |
TE122 |
| 1:34PM |
1 |
Monitor() - splitting long calls into several sound files |
| 9:47AM |
0 |
Asterisk 1.6 AEL Macro vs GoSub |
| 5:54AM |
2 |
unavailable state not reported to Cisco SPA50X phone |
| 3:25AM |
1 |
Call to Asterisk registered sofphone from an independent unregistered Endpoint |
| |
| Sunday November 13 2011 |
| Time | Replies | Subject |
|---|
| 10:19PM |
2 |
Logging Specific Verbose Level To Seperate File |
| 8:55PM |
0 |
shared_lastcall for 1.4.42 |
| 10:05AM |
0 |
Music on Hold does not " Kick-in" until second try , on outgoing calls. |
| |
| Saturday November 12 2011 |
| Time | Replies | Subject |
|---|
| 1:39PM |
1 |
Warning from Exchange Calendar |
| |
| Friday November 11 2011 |
| Time | Replies | Subject |
|---|
| 10:50PM |
2 |
10.0.0-rc1: dahdi doesn't see card |
| 10:19PM |
1 |
10.0.0-rc1: won't start: "empty buf size" |
| 8:36PM |
0 |
Text to speech modules (espeak, flite) |
| 5:26PM |
3 |
1.8.7.0 crashing : Can't send 10 type frames with SIP write |
| 12:08AM |
1 |
What the variable that return the IP Phone username to use it for AddQueueMember |
| |
| Thursday November 10 2011 |
| Time | Replies | Subject |
|---|
| 8:44PM |
1 |
FXS - Power Alarms |
| 6:10PM |
1 |
Frequent Asterisk Restarts |
| 5:07PM |
0 |
DTMF issue with 1.8.6.0 and SIP Trunks [WORKING] |
| 4:38PM |
2 |
Asterisk 10.0.0-rc1 Now Available |
| 3:38PM |
5 |
DAHDI has broken by pbx :-( |
| 3:28AM |
1 |
IAX2 availability testing |
| 12:03AM |
1 |
DTMF issue with 1.8.6.0 and SIP Trunks |
| |
| Wednesday November 9 2011 |
| Time | Replies | Subject |
|---|
| 11:01PM |
0 |
ayrv2by jg4yjbf3r |
| 5:09PM |
1 |
ConfBridge 1.6.20 user count |
| 10:22AM |
1 |
Permanent sip and agi debug on? |
| |
| Tuesday November 8 2011 |
| Time | Replies | Subject |
|---|
| 10:47PM |
2 |
Licensing question. |
| 5:09PM |
0 |
Asterisk 1.6.2.20 lost registration bug with NAT keep-alive |
| 2:51PM |
1 |
No call progress sounds |
| 9:45AM |
0 |
Many SIP-480 responses |
| 6:44AM |
1 |
Realtime Queue - changing strategy to linear needs Asterisk restart |
| |
| Monday November 7 2011 |
| Time | Replies | Subject |
|---|
| 2:28PM |
4 |
4 sec delay in voice menu (asterisk) |
| 1:38PM |
1 |
1.8.8.0-rc2 Missing core functions. |
| 8:46AM |
2 |
Dahdi complete 2.5.0.1 - dahdi_dummy not compiled |
| |
| Saturday November 5 2011 |
| Time | Replies | Subject |
|---|
| 9:23AM |
1 |
Where do I find error message descriptions? |
| |
| Friday November 4 2011 |
| Time | Replies | Subject |
|---|
| 12:21PM |
0 |
augeas lens asterisk |
| 12:15PM |
0 |
What does this error mean? |
| 9:34AM |
2 |
9. any live queue monitor recommendation? (Jean Chassoul) chassoul@gmail.com |
| 7:57AM |
1 |
problem when exiting from "record file" function without pressing the escape digit |
| 12:10AM |
0 |
Queue status question. |
| |
| Thursday November 3 2011 |
| Time | Replies | Subject |
|---|
| 11:59PM |
0 |
any live queue monitor recommendation? |
| 11:20PM |
15 |
DID from Direct from Telco |
| 6:36PM |
1 |
2 pbxes |
| 5:04PM |
1 |
Asterisk as SoftSwitch - Hardware |
| 10:10AM |
2 |
duration limits in Dial() not being enforced at correct time |
| |
| Wednesday November 2 2011 |
| Time | Replies | Subject |
|---|
| 5:41PM |
1 |
Change indications in Dialplan |
| 2:56PM |
1 |
Unable to build sip pvt data - Switching equipment congestion |
| 2:06PM |
1 |
libpri : Q.931 Called Party Number interpreted as empty |
| 1:08PM |
1 |
Option 'd' of application Dial not working in 1.8.8-rc2 |
| 4:47AM |
2 |
FFA - Asterisk 1.6.2.6 |
| |
| Tuesday November 1 2011 |
| Time | Replies | Subject |
|---|
| 5:08PM |
10 |
State of Asterisk+Virtualization+Timing |