| Sunday January 31 2010 |
| Time | Replies | Subject |
|---|
| 10:51PM |
1 |
SIP Registration Failure Logging |
| 9:08PM |
0 |
request for testing: MixMonitor Mute |
| 8:54PM |
2 |
sip to dahdi and billsec |
| 8:47PM |
0 |
asterisk-users Digest, Vol 66, Issue 75 |
| |
| Saturday January 30 2010 |
| Time | Replies | Subject |
|---|
| 8:53PM |
2 |
FAX over ISDN PRI |
| 1:48PM |
8 |
MATH |
| 12:29PM |
0 |
Aastra RFP-32 and CLID |
| 8:57AM |
1 |
forward call back up same trunk to external cell phone problem |
| 3:57AM |
4 |
Astribank problem |
| 2:27AM |
0 |
Asterisk status "488 Not acceptable here" on receiving fax |
| |
| Friday January 29 2010 |
| Time | Replies | Subject |
|---|
| 11:20PM |
0 |
Caller ID not working properly on some phones... |
| 11:07PM |
1 |
callerid not working over sip |
| 10:25PM |
2 |
Help for MOH - sounding scratchy/static on hold |
| 8:09PM |
1 |
Digium fax - sending fax call file vs manager originate |
| 8:07PM |
0 |
Broker lines on a T1 : Signaling convention? |
| 7:03PM |
0 |
New feature: Asterisk Manager Interface commands for DeviceState |
| 6:07PM |
2 |
microphone on Polycom 550/650 |
| 5:54PM |
2 |
Questions about asterisk and spa2102 |
| 5:12PM |
0 |
smsq command |
| 3:29PM |
1 |
Problem with ringing (or absence of...) with Polycom forwarding |
| 3:13PM |
2 |
Cell phone redialer? |
| 2:09PM |
1 |
1 Asterisk server, multiple registrations to ITSP |
| 1:25PM |
1 |
disable comfort noise |
| 1:10PM |
0 |
Address family not supported by protocol |
| 11:39AM |
0 |
VUC Today at 1 PM EST: Counterpath/Bria |
| 11:30AM |
0 |
chan_mobile problem with audio (distorted) |
| 12:15AM |
1 |
Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| |
| Thursday January 28 2010 |
| Time | Replies | Subject |
|---|
| 11:08PM |
2 |
beroNet BN4S0e PCI Express ISDN Card with chan_dahdi |
| 10:57PM |
1 |
Cell Phone dialing |
| 9:31PM |
0 |
How to set sip client idle or busy in Asterisk ? |
| 9:29PM |
0 |
AsyncGoto/DAHDI ? |
| 9:17PM |
1 |
Use of "603 Declined" |
| 8:28PM |
3 |
Dial cellphone from one PBX1 to PBX2? is it possible? |
| 7:25PM |
3 |
TDM2400 card FXS problems |
| 5:09PM |
2 |
911, location |
| 4:37PM |
2 |
rtp.c:883 ast_rtcp_read: RTCP Read too short |
| 4:25PM |
0 |
New feature in app_queue: Give members a penalty time for not answering (help testing) |
| 2:16PM |
0 |
Polycom Soundpoint 300IP |
| 12:21PM |
4 |
Fw: OfficeSIP Softphone |
| 11:23AM |
1 |
Inserting white noise / music / sound file into mixmonitor |
| 10:02AM |
0 |
Asterisk Database |
| 8:14AM |
1 |
iax client for symbian s60 |
| 8:12AM |
1 |
iax softphones - not reconnecting |
| 7:14AM |
0 |
Database Configration |
| 2:32AM |
2 |
Linux-based hard phones? |
| 1:23AM |
0 |
yum install asterisk16 for Fedora Core 8 |
| |
| Wednesday January 27 2010 |
| Time | Replies | Subject |
|---|
| 9:18PM |
3 |
Data transfer |
| 4:39PM |
1 |
Asterisk Database Configuration |
| 3:48PM |
2 |
Mitel integration |
| 3:28PM |
0 |
Need recommendation for ISDN-BRI cards for use with Asterisk |
| 1:59PM |
2 |
CDR problems with Queue |
| 1:23PM |
1 |
Asterisk, NAT, and RTP? |
| 11:07AM |
2 |
astdb |
| 10:47AM |
3 |
Unregistred users can pass calls, peer being static |
| 10:00AM |
2 |
Connecting to an External EPBX without an SIP provider |
| 9:25AM |
2 |
CDR messed up when using queue |
| 2:27AM |
1 |
Realtime Queue not work in 1.6.2.1 |
| |
| Tuesday January 26 2010 |
| Time | Replies | Subject |
|---|
| 3:54PM |
0 |
Problem with Digium card, not transfering outgoing calls [Solved] |
| 3:48PM |
2 |
Attended Transfer with REFER |
| 3:39PM |
0 |
Anyone going to HD Communications Summit - Europe Feb 12th? |
| 3:10PM |
2 |
[inter-pbx commnication] trying to make PBX1 talk to PBX2 |
| 12:22PM |
0 |
settings for soft phones |
| 11:11AM |
2 |
Error and call drops |
| 9:03AM |
1 |
Sip Trunk takes incomming calls for 2 minutes and then stops |
| 4:21AM |
1 |
Asterisk 1.2.37 + BLF + ParkedCalls + SPA962 |
| 2:16AM |
0 |
StopPlayTones() after first digit? |
| |
| Monday January 25 2010 |
| Time | Replies | Subject |
|---|
| 10:58PM |
1 |
Disa not fully bridging outbound call |
| 8:57PM |
1 |
How to make SpeechBackground keep playing if utterance doesn't match our grammar |
| 8:50PM |
0 |
[OT] spa3000 (Regional & Line1) NL settings required |
| 3:20PM |
0 |
Problem with Digium card, not transfering outgoing calls |
| 2:52PM |
1 |
ASTSBINDIR not being picked up by safe_asterisk |
| 2:26PM |
0 |
Call tagging |
| 2:22PM |
5 |
Call recordings and sensitive information |
| 1:27PM |
3 |
sip.conf with versatel and two NICs very strange problem |
| 12:10PM |
3 |
queue |
| 10:24AM |
1 |
Web-Meetme 4.0 and Asterisk 1.6.2 |
| 9:51AM |
2 |
Detected digit 'f' |
| 9:12AM |
1 |
MYSQL grammar diff in 1.6.2.1? |
| 7:46AM |
1 |
MySQL RealTime Error |
| 5:44AM |
0 |
Adminpin for conference room |
| |
| Sunday January 24 2010 |
| Time | Replies | Subject |
|---|
| 10:05PM |
1 |
[OT] Snom M3s |
| 9:19PM |
1 |
two stage dialing in a SIP dial plan |
| 7:51AM |
2 |
ReceiveFAX and SendFAX questions |
| 7:25AM |
0 |
AOC advise of charge |
| 2:08AM |
3 |
odd issue with the with SIP over VPN |
| |
| Saturday January 23 2010 |
| Time | Replies | Subject |
|---|
| 5:47PM |
0 |
Jabber Server |
| 2:11PM |
2 |
fax over IP - http/ftp-provisioning - intercom |
| 1:14PM |
1 |
Xorcom problem after update from zaptel to dahdi-2.2.1 |
| 12:12AM |
1 |
IAX ans SS7 |
| |
| Friday January 22 2010 |
| Time | Replies | Subject |
|---|
| 11:58PM |
2 |
Siemens Gigaset + Asterisk Query? |
| 11:43PM |
0 |
Handling SIP error codes/ISDN codes |
| 3:50PM |
2 |
Polycom phone DND state |
| 2:22PM |
5 |
Set CDR userfield for Queues |
| 12:26PM |
4 |
Snom vs Polycom |
| 12:14PM |
0 |
Asterisk 1.6 mysql 'NO ANSWER' disposition problem |
| 11:53AM |
0 |
Meetme conferencing - large deployment SIP or ZAP? |
| 11:49AM |
0 |
OT - SPA3102 not detecting CID - Which settings to tune ? |
| 9:35AM |
0 |
FW: Call Xfer issue between DataCenter and User Site |
| 9:13AM |
4 |
OfficeSIP Softphone |
| 8:51AM |
1 |
GoToIfTime issue |
| 8:06AM |
2 |
MYSQL problem |
| 2:08AM |
2 |
Trouble getting feature codes to work |
| 12:56AM |
1 |
Popular Gigabit Phones |
| |
| Thursday January 21 2010 |
| Time | Replies | Subject |
|---|
| 8:40PM |
1 |
pri CLI command not available |
| 5:42PM |
0 |
chan_ss7 or libss7, which is more stable? |
| 5:35PM |
1 |
Echo cancellation in a sip channel |
| 3:51PM |
1 |
odbc question |
| 1:48PM |
0 |
Feature codes not detected |
| 1:46PM |
2 |
Caller hang up not detected |
| 10:59AM |
1 |
DTMF reception during WaitForSilence |
| 10:50AM |
1 |
Asterisk & LDAP authentification |
| 4:27AM |
1 |
Pass-through Call Recording Transfer Information |
| 3:17AM |
1 |
Asterisk 403 Forbidden message with port translation |
| |
| Wednesday January 20 2010 |
| Time | Replies | Subject |
|---|
| 11:20PM |
1 |
Setting MixMonitor options from Queue |
| 10:28PM |
6 |
Virtual Asterisk Installation |
| 10:13PM |
1 |
queue groups in asterisk 1.4 |
| 10:06PM |
0 |
DAHDI-Linux 2.2.1 and DAHDI-Tools 2.2.1 Released |
| 8:57PM |
1 |
Using SIPPEER status with CUT function? SOLVED |
| 8:42PM |
1 |
Using SIPPEER status with CUT function? |
| 7:16PM |
1 |
DTMF Issue? |
| 6:24PM |
0 |
More than a line with same extension + Polycom + Provision Tool |
| 5:57PM |
2 |
Call Xfer issue between DataCenter and User Site |
| 4:51PM |
1 |
More than a line with same extension + Polycom 320 + Provision Tool |
| 3:40PM |
1 |
AstLinux 0.7.0 Released |
| 2:39PM |
1 |
Polycom Soundstation Conferencing Unit |
| 2:19PM |
2 |
Odd message: "correct auth, but ..." |
| 2:17PM |
0 |
Selecting IP address for RTP |
| 12:16PM |
0 |
sendtext() SIP MESSAGE to Bria or Eyebeam |
| 8:56AM |
0 |
Which to choose? Realtime extension OR Static extension with MYSQL command |
| |
| Tuesday January 19 2010 |
| Time | Replies | Subject |
|---|
| 9:17PM |
1 |
How to enable a range of IP addresses in realtime sip_buddies |
| 6:56PM |
0 |
Initialize mailbox greeting message |
| 6:48PM |
0 |
Call drop-out on second incoming call. |
| 5:33PM |
1 |
ast_queue_log to mysql asterisk < 1.4 ? |
| 4:46PM |
0 |
Detecting incoming faxes and forwarding to phone fax machine |
| 2:59PM |
1 |
test case with queues and system() |
| 7:08AM |
0 |
B2bua |
| 6:46AM |
1 |
wav to gsm can't play |
| 2:07AM |
0 |
Asterisk answers with wrong sip entry |
| |
| Monday January 18 2010 |
| Time | Replies | Subject |
|---|
| 8:27PM |
0 |
TDM2400P "Unable to set SW Companding on channel .." |
| 5:13PM |
10 |
Dahdi/callerid issue |
| 7:23AM |
1 |
How to play the voicemail recorded? |
| 7:12AM |
0 |
What's customer_id mean? |
| 2:11AM |
0 |
Will SIP connection stop automaticlly when detect no voice between the channel after a period of time? |
| |
| Sunday January 17 2010 |
| Time | Replies | Subject |
|---|
| 8:25PM |
5 |
help with picking out a digium card. |
| 5:39PM |
1 |
Dial String command after audio background |
| 2:34PM |
1 |
receive text |
| 11:09AM |
2 |
How to escape characters in Dialplan |
| 11:05AM |
0 |
How to escape the Pound-Char in Callfiles |
| |
| Saturday January 16 2010 |
| Time | Replies | Subject |
|---|
| 10:55PM |
0 |
Do any providers support speex codec? |
| 5:47PM |
0 |
Anyone have provisioning documentation for LeadTek devices? |
| 3:54PM |
2 |
Cross compiling Asterisk, Dahdi.. |
| 12:04PM |
1 |
Hint for realtime peers |
| 10:17AM |
0 |
Asterisk 1.4 or 1.6 automated install |
| 10:02AM |
4 |
Howto regret blind transfer? |
| 3:49AM |
0 |
Realtime cached values |
| 12:02AM |
1 |
Echo on Polycom phones |
| |
| Friday January 15 2010 |
| Time | Replies | Subject |
|---|
| 6:11PM |
0 |
TE410P generates only 1 interrupt |
| 5:27PM |
5 |
Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4 |
| 5:23PM |
0 |
info for Busy for incoming internal call but not for exterrnal |
| 5:17PM |
1 |
Digium Asterisk World at ITEXPO - Yahoo keynote update |
| 4:52PM |
2 |
Changing ring cadence on FXS lines |
| 4:06PM |
1 |
DAHDI and Analogue lines (UK) |
| 3:29PM |
0 |
Asterisk 1.6.2.1 Now Available |
| 3:29PM |
0 |
Asterisk 1.6.1.13 Now Available |
| 3:28PM |
0 |
Asterisk 1.6.0.21 Now Available |
| 3:27PM |
0 |
Asterisk 1.4.29 Now Available |
| 2:50PM |
0 |
Getting Answered Stations instead of Group in cdr? |
| 2:25PM |
0 |
: Asterik with out registration. |
| 1:43PM |
0 |
Logs problem of queue_log-mysql |
| 12:32PM |
0 |
OT: Inbound South America numbers |
| 11:56AM |
1 |
jitterbuffer and PLC |
| 10:48AM |
1 |
Realtime queue not work |
| 7:23AM |
1 |
Question about Presence and IM feature |
| 6:54AM |
3 |
10/100 voip phones and gigabit connection |
| |
| Thursday January 14 2010 |
| Time | Replies | Subject |
|---|
| 6:41PM |
0 |
Friday Jan 15 @12 Noon EST: Hacking VoIP |
| 6:38PM |
2 |
GXV3140 and Xlite video |
| 5:35PM |
5 |
Fax Detection on SIP |
| 4:20PM |
1 |
Dahdi and FreePBX |
| 4:11PM |
0 |
Ringing for incoming call |
| 4:03PM |
1 |
Can not play WAV-files attached to mail from my own Asterisk |
| 3:33PM |
1 |
Languages |
| 3:22PM |
2 |
Dahdi issues |
| 2:53PM |
3 |
iaxmodem / hylafax receive problem |
| 2:15PM |
2 |
Followme Options |
| 1:48PM |
0 |
different between ring groups and queue? |
| 10:53AM |
1 |
Ringing issue |
| 10:08AM |
1 |
Lagged Extension |
| 9:55AM |
4 |
how to strip + from the caller-ID |
| 9:10AM |
0 |
What about the performance visit MYSQL in DialPlan code? |
| 9:07AM |
0 |
Attend CampKDE Jan 15-18 via Voice over Internet (VOIP), BerkeleyTIP |
| 4:45AM |
0 |
ISDN Cause codes for unanswered calls |
| 3:29AM |
2 |
is there some Chinese version of sounds available? |
| |
| Wednesday January 13 2010 |
| Time | Replies | Subject |
|---|
| 11:48PM |
0 |
asterisk / NEC2400 / PRI |
| 10:07PM |
0 |
Asterisk 1.4.28 intermittent one way audio? |
| 4:49PM |
0 |
FW: [mythtv-users] VMWare on the backend. Viable solution? |
| 5:56AM |
3 |
Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6? |
| 3:56AM |
2 |
Polycom Mute Problem |
| 2:59AM |
1 |
Odd Voicemail Issue |
| |
| Tuesday January 12 2010 |
| Time | Replies | Subject |
|---|
| 11:05PM |
1 |
Xorcom 32 channel FXS gateway |
| 10:48PM |
1 |
AudioCodes MP-114 - SAS (Stand Alone Survivability) configuration |
| 7:07PM |
1 |
Problem logs queue_log-mysql |
| 5:57PM |
0 |
VMs & IMAP Storage |
| 5:56PM |
2 |
Minimal Asterisk Web Interface? |
| 5:43PM |
2 |
SIP Security |
| 5:38PM |
1 |
Inserting a wait in a sip dial |
| 5:16PM |
2 |
Question about SIP registration |
| 4:28PM |
1 |
Send 503 or 603 error after answer() |
| 11:57AM |
0 |
Virtual ISDN device /dev/XYZ |
| 10:55AM |
5 |
Beginners Guide to setting up a Call Centre |
| 5:57AM |
5 |
Multi-Tenant Parking |
| 5:56AM |
0 |
Why agent log out automaticly？ |
| 3:26AM |
2 |
is roundrobin and rrmemory the same meaning? |
| 2:44AM |
0 |
Interfacing to NEC Xen Master PBX |
| |
| Monday January 11 2010 |
| Time | Replies | Subject |
|---|
| 9:49PM |
0 |
Problem with call transfer and Polycom 430 |
| 7:05PM |
4 |
SIP over VPN -- no audio to other remote/VPN connected phones |
| 5:21PM |
0 |
ChanSpy doesn't hangs up |
| 5:05PM |
1 |
MeetMe Conferencing - Announce your own join/leave to yourself and other conference members |
| 4:58PM |
0 |
Custom date formats with new mode say.conf? |
| 4:26PM |
2 |
Sipgate > DTMF not detected |
| 3:23PM |
2 |
Asterisk core dumps when using PrivacyManager |
| 12:27PM |
0 |
TONIGHT Join 5-6P Mon 11th - 1st Evening Meeting test IRC & VOIP online Asterisk at BerkeleyTIP-Global - for forwarding |
| 11:25AM |
2 |
Extension Status |
| 10:57AM |
0 |
Temporary loss of audio on all SIP channels |
| 10:45AM |
2 |
Attempted break in ? |
| 7:43AM |
0 |
asterisk-users archive |
| 7:16AM |
0 |
PHP-Script (AGI) doesn't finish after upgrading to 1.6.0.15 |
| 2:37AM |
0 |
How to test if a telephone is busy now? |
| 2:19AM |
1 |
How to use AGI php script function $agi -> exec_dial |
| 2:15AM |
0 |
Zhang Shukun ????? |
| |
| Sunday January 10 2010 |
| Time | Replies | Subject |
|---|
| 11:00PM |
1 |
Weird Polycom SP 650 |
| 10:33PM |
2 |
app_swift 1.6.2 DTMF issue |
| 9:31PM |
1 |
Problem with my dialplan |
| 9:17PM |
1 |
Grandstream GXW-4024 |
| 6:58PM |
0 |
Directory and Voicemail Problems after upgrading from 1.4 to 1.6 |
| 3:29PM |
0 |
Off-line subscribed phone amber on SPA942? |
| 10:24AM |
1 |
You won't help me anymore? |
| 7:33AM |
2 |
No dial-tone with X101P FXO card |
| 1:00AM |
0 |
Music / Background |
| 12:06AM |
0 |
Queue - Update CDR |
| |
| Saturday January 9 2010 |
| Time | Replies | Subject |
|---|
| 9:22PM |
1 |
Using HASH() and REALTIME_HASH() |
| 6:03PM |
1 |
UK dialing tone |
| 3:22PM |
1 |
Quick Installing Asterisk-1.4 on Ubuntu |
| 1:00PM |
2 |
Choppy MOH |
| 9:57AM |
0 |
Asterisk CallerId problem? |
| |
| Friday January 8 2010 |
| Time | Replies | Subject |
|---|
| 11:11PM |
1 |
How can I get codec info on active calls |
| 10:37PM |
1 |
Multicast RTP Paging |
| 8:57PM |
0 |
Queue_log file and mysql logs together! |
| 4:47PM |
0 |
Semi-OT: Configuring SIP trunks with Cisco UCM 7.0. |
| 4:07PM |
0 |
[VUC] Today at 12 Noon EST (6PM CEST, 9AM PST) iNum with Voxbone |
| 3:33PM |
0 |
Cheap femtocell's ahead |
| 9:14AM |
1 |
How to recieve number returned by $AGI->wait_for_digit($timeout) |
| |
| Thursday January 7 2010 |
| Time | Replies | Subject |
|---|
| 11:27PM |
4 |
AGI perl script set timeout within script? |
| 10:20PM |
1 |
voicemail /odbc problem |
| 8:28PM |
1 |
Crash in Asterisk |
| 6:19PM |
0 |
dns messages on console |
| 4:15PM |
2 |
Sip REFER failes w/603 Decline (Policy), Polycom Phone |
| 3:49PM |
7 |
Please remove me from the mailing list. |
| 2:06PM |
0 |
Dialing OutBound SIP trunk using Dial() command |
| 1:38PM |
1 |
How to dial a number make two phone Ring at the same time? |
| 11:11AM |
0 |
queue and linear strategy |
| 11:00AM |
2 |
Explain what asterisk.conf's "internal timing" option is |
| 8:19AM |
1 |
error compile dahdi with latest kernels. |
| 6:08AM |
1 |
compile one additional module without recompiling all asterisk |
| 4:16AM |
0 |
Question about PLC of Asterisk |
| 1:53AM |
1 |
How to see STDERR message? |
| 1:07AM |
0 |
video with x-lite |
| |
| Wednesday January 6 2010 |
| Time | Replies | Subject |
|---|
| 11:23PM |
1 |
iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P |
| 10:52PM |
0 |
Urgent: Which spandsp version is recommended for 1.6.1 ? |
| 9:18PM |
0 |
DEVICE STATE "In use" |
| 7:45PM |
2 |
question on makefile |
| 7:15PM |
0 |
Asterisk 1.6.1.x SMDI MWI w/Fujitsu F9600 Problem |
| 11:43AM |
1 |
Zaptel compilation problems: Data Mode!! |
| 10:44AM |
1 |
Inquiry:How to define incoming route for sip? |
| 9:36AM |
0 |
MITEL |
| 9:16AM |
1 |
Fastagi-mapping problem |
| 3:09AM |
0 |
Originate from the Dialplan |
| 2:40AM |
1 |
Merlin Legend integration not routing calls back to PSTN. |
| |
| Tuesday January 5 2010 |
| Time | Replies | Subject |
|---|
| 10:24PM |
6 |
Faxing: Anyone have a compiled executable? |
| 9:41PM |
0 |
send faxes as "3,1 kHz Audio" |
| 7:53PM |
6 |
Really Silly Question From Total Newbie |
| 7:25PM |
1 |
Canadian call quality issue |
| 2:38PM |
1 |
DTMF detection on dahdi with b4xxp (again, some more details) |
| 2:16PM |
0 |
Newbie: MITEL and Asterisk |
| 1:24PM |
5 |
CallerID on Indian PSTN is not working. |
| 12:33PM |
1 |
Realtime LDAP Queues crashes |
| 9:43AM |
0 |
(no subject) |
| 9:38AM |
0 |
Get Queue Info |
| 9:30AM |
0 |
{Spam?} MeetMe/Dahdi for FreeBSD |
| 6:00AM |
0 |
automatic dial from database |
| 4:53AM |
0 |
Inquiry:Asterisk sending dialed digits in one-by-one digit format? |
| 1:41AM |
3 |
AGI and embargeability |
| |
| Monday January 4 2010 |
| Time | Replies | Subject |
|---|
| 11:44PM |
1 |
T.38 ITSP? |
| 9:03PM |
0 |
lpc10 |
| 8:09PM |
0 |
Dialout from Meetme conference |
| 6:48PM |
0 |
Register sip FXO per gateway |
| 5:57PM |
0 |
H323 Disconnects after 15+ minutes |
| 4:46PM |
1 |
Script to show asterisk stuff |
| 3:21PM |
2 |
caller getting cut off intermittently |
| 1:42PM |
1 |
ZapRAS priviledge error |
| 1:16PM |
4 |
Dahdi and oslec |
| 1:08PM |
1 |
Realtime Queue Members Not Ringing |
| 8:51AM |
1 |
Some minor configuration issues with queues |
| 8:34AM |
2 |
DNS reload on trunks for outgoing calls |
| 6:17AM |
3 |
MYSQL queries from dial plan |
| 6:14AM |
0 |
Asterisk realtime chat |
| 6:10AM |
0 |
differences between asterisk 1.6.1.x and 1.6.0.x |
| 5:24AM |
1 |
Free FaxForAsterisk ReceiveFAX not working |
| 3:13AM |
0 |
Dahdi causes panic on server restart |
| 2:14AM |
2 |
Outgoing Calls Only -- Firewall Rules |
| |
| Sunday January 3 2010 |
| Time | Replies | Subject |
|---|
| 9:08PM |
0 |
asterisk-users Digest, Vol 66, Issue 4 |
| 12:04PM |
0 |
Inquiry:How to join Asterisk real time chat? |
| |
| Saturday January 2 2010 |
| Time | Replies | Subject |
|---|
| 2:01PM |
4 |
Help getting info from caller |
| 7:06AM |
4 |
verifying correct loading of VPMADT032 |
| 6:22AM |
0 |
Inquiry:Asterisk sip ? |
| |
| Friday January 1 2010 |
| Time | Replies | Subject |
|---|
| 9:57PM |
0 |
AudioCodes MP-114 2xFXS/2xFXO - FXO not working correctly |
| 4:34PM |
7 |
SIP Listen Multiple Ports |
| 1:50PM |
0 |
Happy New year 2010 |
| 12:01PM |
1 |
PBX Extension Help |
| 2:17AM |
0 |
AudioCodes MWI |