| Saturday February 28 2009 |
| Time | Replies | Subject |
|---|
| 7:24PM |
2 |
clone X100p+dahdi dial out works only after receiving call |
| 6:02PM |
0 |
Temporaneamente assente |
| 5:21PM |
2 |
No rtp activity |
| 2:53PM |
2 |
Remote connection to an Asterisk server |
| 11:42AM |
0 |
using an eicon diva server card with asterisk. |
| 2:17AM |
0 |
rfc2833 vs. sipinfo and network weirdness |
| |
| Friday February 27 2009 |
| Time | Replies | Subject |
|---|
| 9:32PM |
1 |
dialing timing problem? |
| 9:07PM |
1 |
what is the effect of high LBO settings? |
| 8:47PM |
1 |
TE121B server recommendation |
| 7:45PM |
3 |
Continue in dialplan on hangup |
| 6:05PM |
1 |
Switch Options for a service provider |
| 5:40PM |
1 |
API command monitor to record only the input channel |
| 3:37PM |
11 |
building a phone |
| 1:20PM |
0 |
[HOWTO] Priorize one destination over another on a link |
| 12:53PM |
1 |
change language and playback issue |
| 10:58AM |
1 |
Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available |
| 1:58AM |
9 |
call file concurrency |
| |
| Thursday February 26 2009 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
using Cisco IP Communicator with SIP to Asterisk |
| 11:33PM |
3 |
Question about Do Not Disturb |
| 8:44PM |
3 |
changing /etc/dahdi/system.conf |
| 7:33PM |
1 |
Current state of Asterisk and Virtualization? |
| 7:27PM |
0 |
Friday Feb 27th at 12 Noon EST: Polycom Applications |
| 7:16PM |
0 |
asterisk 1.4.23.1 and mISDN 1.1.8 segfaults |
| 6:40PM |
1 |
incoming call problem |
| 6:11PM |
1 |
Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4 |
| 5:39PM |
0 |
Residential portals and real world scalability |
| 3:35PM |
0 |
[cdr_odbc] error: Cannot insert the value NULL into column 'calldate' |
| 2:30PM |
2 |
Odd Read App Issues |
| 1:41PM |
5 |
ATA recommendation (wih FTP provisioning) |
| 1:36PM |
3 |
Getting SIP field P-Asserted-Identity from EAGI |
| 1:30PM |
0 |
Patton 5.3. How to get incoming calls ? [SOLVED] |
| 10:59AM |
3 |
call-limit on a per destination basis |
| 10:25AM |
3 |
HP DL380 G5 with TE420 |
| 10:19AM |
1 |
Dictate |
| 9:00AM |
0 |
Need US Dialing Account with Asterisk |
| 5:44AM |
1 |
codec_dahdi and Asterisk 1.6.0.6 |
| 5:28AM |
2 |
asterisk 1.6.0.5 and IM |
| 3:50AM |
2 |
Problems with Outbound Calls |
| 2:48AM |
1 |
DTMF Forwarking Problems. |
| |
| Wednesday February 25 2009 |
| Time | Replies | Subject |
|---|
| 11:07PM |
5 |
AGI problem using mono (.Net) |
| 9:17PM |
1 |
Congestion Tone |
| 8:59PM |
2 |
SheevaPlug Development Kit |
| 8:40PM |
1 |
Realtime database function help |
| 8:16PM |
0 |
Call from '6000' to extension rejected becauseextension not found |
| 8:11PM |
2 |
Call from '6000' to extension rejected because extension not found |
| 7:11PM |
0 |
Patton 5.3. How to get incoming calls ? |
| 6:56PM |
1 |
CDR - Asterisk-Stat and PHP5 |
| 6:46PM |
4 |
DID's in a specific rate center |
| 6:29PM |
1 |
cannot allocate memory |
| 3:24PM |
2 |
dahdi wcb4xxp and fax |
| 3:00PM |
4 |
TE121 on Asterisk |
| 2:39PM |
1 |
Stuck Parked Calls? |
| 12:54PM |
1 |
SIP_CODEC variable |
| 12:45PM |
0 |
usegmtime=yes for cdr_custom |
| 10:38AM |
3 |
Asterisk with Internet connectivity |
| 9:45AM |
3 |
bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA |
| 9:02AM |
4 |
switchtype QSIG and Asterisk implementation |
| 8:02AM |
0 |
Call files with extensions.ael : "One app must be specified" |
| 7:01AM |
0 |
HDLC Errors |
| 6:11AM |
1 |
Dropping RTP packets |
| 3:29AM |
0 |
Asterisk security between two servers |
| 3:14AM |
0 |
Problem redirecting user running a Dynamic feature |
| |
| Tuesday February 24 2009 |
| Time | Replies | Subject |
|---|
| 10:17PM |
1 |
Incoming call |
| 10:02PM |
2 |
Multiple SIPGate accounts. |
| 8:52PM |
2 |
receiving 1st digit from a variable |
| 8:41PM |
2 |
Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card |
| 7:44PM |
3 |
Gosub behavior change <=1.6.0.5 to 1.6.0.6 |
| 7:08PM |
2 |
what is the correct character to separate application parameters: , or | |
| 5:33PM |
5 |
Aastra phones |
| 2:16PM |
1 |
COSTA RICA - E1 |
| 1:49PM |
1 |
API hangup command |
| 12:59PM |
0 |
db_dump185.c missing if Asterisk 1.4 source file |
| 12:37PM |
2 |
astdb and Debian : can't use db4.5_dump |
| 9:56AM |
2 |
chan_sip and database integration |
| 9:55AM |
1 |
asterisk -f and restart now |
| 9:15AM |
5 |
Swichted digits on received number from fax on an fxs port |
| 7:59AM |
7 |
multiple asterisks in a server |
| 7:27AM |
0 |
Free US DID |
| 4:12AM |
0 |
Does audio have to flow through vitelity servers ? |
| 3:44AM |
0 |
Legacy Cisco ATA186I1 |
| 2:10AM |
7 |
Managing the spiralling costs |
| 2:07AM |
3 |
Polycom Spectralink 8002 Configuration |
| 1:35AM |
8 |
HDD FULLL |
| 1:28AM |
1 |
weird problem |
| 12:17AM |
1 |
building asterisk-1.6.0.6 failed! |
| |
| Monday February 23 2009 |
| Time | Replies | Subject |
|---|
| 11:59PM |
3 |
GSM codec is a good choice ??? |
| 11:30PM |
1 |
strange text message:) |
| 10:44PM |
0 |
Windows Mobile MWI and asterisk |
| 9:16PM |
1 |
Inbound call to IVR drops after 21 seconds? |
| 8:56PM |
3 |
don't get 2.0 gui to run on asterisk 1.6.0.5 |
| 8:26PM |
1 |
Compiling asterisk-addons-1.6.0 under Debian 2.6.18? |
| 8:23PM |
0 |
Asterisk 1.6.0.6 Now Available! |
| 7:29PM |
2 |
Voicemail and ADSI |
| 7:13PM |
3 |
5ess |
| 6:22PM |
0 |
FreePBX in a Cloud With a Click - FreePBX secured and optimized for Amazon EC2 |
| 5:37PM |
6 |
Delete all |
| 4:52PM |
0 |
Astlinux 0.6.3 Released |
| 3:55PM |
0 |
302 Redirect Handling |
| 3:19PM |
0 |
problem with nortel 2002 disconecting |
| 3:13PM |
1 |
Asterisk/Skype update |
| 12:19PM |
1 |
Flash Operator Panel with Asterisk 1.6 |
| 8:30AM |
1 |
receive fax problem |
| 7:32AM |
0 |
Can't compile Dahdi on SuSE 10.1/Kernel 2.6.16.54-0.2.5-smp/GCC 4.1.2 |
| 5:06AM |
2 |
Asterisk 1.6.0.5 as non root and moh perm issue |
| |
| Sunday February 22 2009 |
| Time | Replies | Subject |
|---|
| 8:11PM |
1 |
I can`t send DTMFs through FXO lines - dahdi |
| 6:33PM |
3 |
Intel Vs AMD |
| |
| Saturday February 21 2009 |
| Time | Replies | Subject |
|---|
| 3:30PM |
1 |
VoIP Information in CDRs |
| 12:55PM |
0 |
Where to find db1_dump185 in debian packages ? [SOLVED] |
| 12:27PM |
2 |
DIAL() application 'g' option |
| 11:39AM |
0 |
Cisco Phone losse regsitrations with Asterisk |
| 4:37AM |
0 |
help needed -- chanspy |
| 2:02AM |
3 |
IAX2 - now known as RFC 5456 |
| |
| Friday February 20 2009 |
| Time | Replies | Subject |
|---|
| 10:55PM |
1 |
Vacation reply |
| 10:34PM |
0 |
hi |
| 9:18PM |
1 |
CDR fields in 1.6 |
| 8:01PM |
0 |
asterisk-users Digest, Vol 55, Issue 64 |
| 7:38PM |
0 |
SIP debug messages |
| 5:33PM |
2 |
zaptel telephone cards and asterisk in another pc |
| 5:29PM |
0 |
Table definitions for Realtime 1.6... |
| 4:56PM |
0 |
slip r errors nortel switch |
| 4:41PM |
1 |
Polycom Phones start to break up after being up a LONG time |
| 3:26PM |
1 |
real time connection failed |
| 1:29PM |
0 |
SMS to generate a call back |
| 9:33AM |
0 |
Qualify sip users behind remote registrar |
| 7:02AM |
1 |
SIP Proxy behind NAT talkinf to ASterisk with public IP |
| |
| Thursday February 19 2009 |
| Time | Replies | Subject |
|---|
| 11:27PM |
3 |
AGI script |
| 10:00PM |
1 |
TDMOE Timing |
| 8:48PM |
3 |
DTMF |
| 7:43PM |
1 |
Annoying silence suppression effect on my digium E1 card with the VPMADT032 module |
| 2:54PM |
1 |
queue_variables() function |
| 2:33PM |
4 |
check if not human |
| 1:47PM |
3 |
Busy status of a snom connected to two asterisk servers? |
| 1:25PM |
2 |
Not answering call when queue is full or has no members |
| 11:53AM |
0 |
Asterisk BLF to Cisco CME |
| 11:00AM |
0 |
About Hint Configuration in Asterisk. |
| 10:06AM |
0 |
sip phone cant hear the caller |
| 8:32AM |
0 |
Friday Feb 20th 12 Noon EST: Jason Fischl from Counterpath on VUC |
| 6:47AM |
2 |
Managing SIP hardphones call history |
| 5:22AM |
1 |
DeadAgi Application in asterisk 1.6 |
| |
| Wednesday February 18 2009 |
| Time | Replies | Subject |
|---|
| 11:08PM |
1 |
Understand SIP REFER |
| 9:18PM |
1 |
Asterisk on the Cloud With a Click - pre-built Asterisk Amazon EC2 instance |
| 8:48PM |
1 |
call file FXO channel problem |
| 8:42PM |
0 |
Open Source in an Economic Downturn: Asterisk stories |
| 7:00PM |
0 |
No Audio PlayBack Asterisk 1.6 Dahdi 2.1.0.3 |
| 6:28PM |
1 |
trunk to trunk |
| 5:54PM |
2 |
Please help test the gender detection moduleat 575-613-4392 |
| 3:27PM |
0 |
connection to siemens hipath |
| 2:58PM |
1 |
directrtpsetup=yes does not work in 1.4.23.1 |
| 2:51PM |
1 |
Need help on Forwarding |
| 2:38PM |
1 |
asterisk-users Digest, Vol 55, Issue 52 |
| 2:19PM |
3 |
US DID |
| 1:36PM |
1 |
Accumulated call time |
| 1:18PM |
0 |
Ditech API |
| 12:37PM |
0 |
Detecting which party initiates a hangup |
| 11:19AM |
6 |
AGI pdf book |
| 9:04AM |
2 |
Setting SIP header on agent calls made by a queue |
| 8:28AM |
1 |
Distributed presence in 1.6 |
| 4:19AM |
1 |
xorcom hardware good idea? |
| 1:47AM |
0 |
life safety system and VOIP |
| 1:05AM |
0 |
AgentCallbackLogin via dialaplan and device state |
| 12:37AM |
2 |
Open Source in an Economic Downturn: Asterisk stories needed |
| |
| Tuesday February 17 2009 |
| Time | Replies | Subject |
|---|
| 10:08PM |
2 |
SLA and Flashing BLF |
| 10:00PM |
3 |
ztdummy compile under 2.6.28 ? |
| 9:02PM |
0 |
Caller Hangup detection |
| 8:04PM |
3 |
call file bug? |
| 6:06PM |
2 |
Asterisk supports SIP-T? |
| 5:57PM |
0 |
Lost with Patton 5.3 web server. Registration ? |
| 5:27PM |
1 |
Question regarding custom announcements in queues.conf |
| 5:08PM |
0 |
Swift - detection of multiple digits unreliable on my system |
| 4:54PM |
0 |
Asterisk 1.4.21.1 intermittent presence working with Polycom |
| 4:46PM |
0 |
Questions about OpenSky - Asterisk to Skype Gateway |
| 3:19PM |
4 |
Network architecture |
| 2:57PM |
2 |
Packet Truncated - Choppy Audio |
| 1:06PM |
1 |
Pingable and Unreachable at the same time ! |
| 11:05AM |
1 |
What is the purpose of membermacro in queues.conf |
| 9:30AM |
0 |
freemin managment for sim cards |
| 8:33AM |
1 |
zaptel compile kernel problem |
| 7:02AM |
0 |
unistim channel problem |
| 6:51AM |
2 |
Stress Testing IVR |
| 2:37AM |
0 |
Optimizing this script for calling Skype users from Asterisk |
| 2:10AM |
2 |
Obtaining callerid on a PRI for billing purposes (with non toll-free numbers) |
| 12:30AM |
0 |
mp3player() to shuffle playlist |
| |
| Monday February 16 2009 |
| Time | Replies | Subject |
|---|
| 8:51PM |
0 |
How to beep before transfer ... |
| 6:57PM |
0 |
vmsecret question |
| 6:38PM |
1 |
DTMF not completely muted |
| 6:29PM |
7 |
Please help test the gender detection module at 575-613-4392 |
| 4:55PM |
3 |
command show channels concise |
| 3:14PM |
2 |
AstDB wildard searches |
| 2:43PM |
1 |
TeleKaam - Voice Portal for Students and Parents |
| 10:22AM |
1 |
SpanDSP question for Steve |
| 9:14AM |
4 |
Faxing with asterisk |
| 3:12AM |
0 |
Microsoft Recite |
| |
| Sunday February 15 2009 |
| Time | Replies | Subject |
|---|
| 8:43PM |
0 |
call-limit |
| 3:01PM |
1 |
Gizmo SIP / Skype gateway |
| 7:36AM |
1 |
No such command 'core stop now' |
| 2:23AM |
1 |
licensed g729 |
| |
| Saturday February 14 2009 |
| Time | Replies | Subject |
|---|
| 3:28PM |
1 |
Asterisk CLI problem if run from /etc/inittab |
| 12:28PM |
1 |
Progress() and Proceeding() |
| 12:11PM |
1 |
Call Fowarding and Polycom Phone |
| 12:54AM |
1 |
Asterisk 1.6.x timing API |
| |
| Friday February 13 2009 |
| Time | Replies | Subject |
|---|
| 11:17PM |
0 |
Slow hangup - Australia - analog - incoming calls |
| 10:13PM |
0 |
Asterisk 1.6 CDR fields... |
| 9:18PM |
0 |
Asterisk 1.6.0.6 Release Candidate 1 Now Available |
| 8:24PM |
3 |
GUI interface to manage business edition |
| 8:08PM |
1 |
linksys PAP2t and asterisk |
| 7:19PM |
2 |
OpenSky: Digium Skype gateway? |
| 5:59PM |
2 |
Asterisk on EC2 cloud computing - price assumptions - your brain needed |
| 5:39PM |
2 |
Asterisk 1.6.0.5 and Aastra phones... |
| 5:25PM |
0 |
Asterisk and Amazon EC2 cloud service tutorial |
| 3:55PM |
2 |
Continue processing AGI script after hangup |
| 9:42AM |
2 |
Cisco IP Phone 7940G. |
| 9:18AM |
5 |
PRI Test Lab |
| 5:35AM |
1 |
ExitIf() convention? |
| 5:33AM |
0 |
Re : Asterisk Queue and URL Calling |
| 5:04AM |
3 |
MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am |
| 3:51AM |
2 |
Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status |
| 2:51AM |
0 |
zaptel for asterisk |
| |
| Thursday February 12 2009 |
| Time | Replies | Subject |
|---|
| 9:34PM |
1 |
Queue problem |
| 9:04PM |
5 |
CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ? |
| 6:38PM |
1 |
1.6.1-rc1 errors |
| 5:03PM |
4 |
Asterisk Queue and URL Calling |
| 4:27PM |
2 |
Caller ID replacement |
| 3:59PM |
2 |
OSLEC not being loaded on Ubuntu Intrepid |
| 3:05PM |
1 |
Problem with parking |
| 2:52PM |
4 |
Multiple caller id ... |
| 1:55PM |
1 |
After Monitor() files disappear |
| 11:53AM |
1 |
g723 llicense |
| 8:56AM |
5 |
Siemens Hipath PRI to Asterisk Call Routing? |
| 8:27AM |
1 |
Keep your passwords secure .. (VoIP hacker news) |
| 8:18AM |
0 |
Friday the 13th Muhahaha Allison Smith and more on the Polycom Applications |
| 3:57AM |
0 |
IDAP T1 |
| 1:18AM |
3 |
Strange dialplan matching issue |
| |
| Wednesday February 11 2009 |
| Time | Replies | Subject |
|---|
| 11:52PM |
4 |
WiFi SIP phone w/VPN? |
| 11:25PM |
0 |
Problem with AMI action userevent |
| 9:35PM |
1 |
call picking and transfers |
| 7:08PM |
1 |
The download link, why server down? |
| 6:29PM |
0 |
Intercom/Doorbell Integration |
| 6:21PM |
3 |
asterisk across a firewall |
| 6:01PM |
0 |
ChanSpy problem |
| 2:28PM |
2 |
DTMF tones mid conversation |
| 2:24PM |
0 |
Looking for 'remote Asterisk hands' support in Mexico |
| 2:22PM |
3 |
Billing and Soft Switch. |
| 2:20PM |
0 |
Asterisk AGX addons compile issues |
| 10:14AM |
3 |
call forward all except the extension it is forwarded to |
| 9:43AM |
2 |
OPTIONS packets |
| |
| Tuesday February 10 2009 |
| Time | Replies | Subject |
|---|
| 11:32PM |
1 |
Max person in meetme conference |
| 10:11PM |
1 |
Aastra phone crashes with Asterisk 1.6 |
| 5:23PM |
4 |
connecting 66 analog phones to asterisk - hardware suggestions |
| 3:00PM |
1 |
unistim and transfer calls |
| 10:54AM |
0 |
Skip password option in voicemail.conf |
| 10:24AM |
1 |
Disabling Echo Cancellation on a per Call basis |
| 10:20AM |
1 |
Asterisk how many calls handle using H.323 to SIP conversion? |
| 7:24AM |
5 |
What do you use? .conf or AEL? |
| 4:23AM |
0 |
hosted voip? |
| |
| Monday February 9 2009 |
| Time | Replies | Subject |
|---|
| 11:39PM |
2 |
SMS /w Asterisk |
| 11:10PM |
0 |
Audiocodes - Disconnect Supervision |
| 9:44PM |
0 |
[asterisk-dev] 1.4 and CDRs -- The Breaking Point |
| 9:43PM |
1 |
Is "a=fmtp:101 0-15" a legal option in SDP ? |
| 9:25PM |
2 |
asterisk registered as UA |
| 9:23PM |
1 |
What t38pt_udptl is ? Explain T.38 in 1.4 |
| 9:03PM |
2 |
SIP password encryption |
| 8:28PM |
3 |
Hangup extensions via CLI? |
| 7:44PM |
0 |
Problem with upper case extension names |
| 7:22PM |
2 |
Call drops after a minute on 1.6.0.5 |
| 6:52PM |
1 |
How to make the Asterisk-GUI work with DAHDI..please?? |
| 5:50PM |
1 |
Transfer Asterisk 1.6 Telephone IP |
| 5:44PM |
0 |
problem getting asterisk behind NAT to run with sipproxd |
| 5:12PM |
0 |
Problem with AMI originate |
| 3:54PM |
1 |
Noisy Ring Back Tone with TE205P card |
| 3:11PM |
1 |
reinvite |
| 2:20PM |
3 |
Michael Graves post |
| 11:33AM |
2 |
InUse&Ringing |
| 6:20AM |
1 |
Asterisk and CIsco 1760 SIP ? |
| 4:39AM |
2 |
Asterisk + voxbone ==> Failed to authenticate user |
| 4:29AM |
1 |
chan_oss.c:585 setformat: Unable to re-open DSP device |
| 2:43AM |
2 |
meetme application |
| |
| Sunday February 8 2009 |
| Time | Replies | Subject |
|---|
| 5:33PM |
0 |
Streaming meetings vs conference hardware |
| |
| Saturday February 7 2009 |
| Time | Replies | Subject |
|---|
| 10:45PM |
2 |
can anybody tell me how Magic jack can be so cheap ???? |
| 9:08PM |
2 |
Minimum version for asterisk and iaxmodem |
| 6:41PM |
0 |
One way audio after IVR tree |
| 5:39PM |
0 |
GROUP() decrement |
| 2:42PM |
0 |
Asterisk 3rd party developed commercial software sales licensing platform |
| 1:38PM |
0 |
A Simple Asterisk Based Toll Fraud Prevention Script |
| 7:31AM |
1 |
put the hostname of asterisk in the callerid uri to avoid NAT problems |
| 3:41AM |
3 |
VPN and Asterisk |
| 2:16AM |
1 |
Running asterisk on ARM (TS-7800) 1.4.23.1 |
| |
| Friday February 6 2009 |
| Time | Replies | Subject |
|---|
| 10:11PM |
14 |
Credit Card processing machines |
| 9:01PM |
4 |
Security issue |
| 8:53PM |
1 |
AgentCallBackLogin no longer works after installing asterisk 1.6 |
| 6:29PM |
0 |
[asterisk-user] $100USD for anyone who can install Chan_SCCP for me |
| 4:29PM |
2 |
upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call |
| 3:36PM |
0 |
H323 stress test |
| 3:25PM |
0 |
Asterisk as a dial in server for internet |
| 3:06PM |
3 |
Add-on for SRTP and SSIP |
| 2:08PM |
1 |
set caller id on outgoing calls through BRIISDNlines |
| 1:28PM |
1 |
set caller id on outgoing calls through BRI ISDNlines |
| 12:49PM |
0 |
Incoming fax detection on mISDN hfcmulti B410Pcard |
| 12:36PM |
0 |
set caller id on outgoing calls through BRI ISDN lines |
| 11:02AM |
2 |
Rewriting numbers while processing dial plan? |
| 9:15AM |
1 |
Monitor and SIP transfers (SIP REFER) |
| 8:52AM |
2 |
asterisk and DNS |
| 7:00AM |
0 |
Java IAX Implementation |
| 3:51AM |
0 |
Getting DIALSTATUS from SIP provider |
| |
| Thursday February 5 2009 |
| Time | Replies | Subject |
|---|
| 8:20PM |
6 |
Newbie query: how to write priority n+101 |
| 8:12PM |
0 |
Patton M-ATA and T.38 |
| 5:02PM |
0 |
Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ? [SOLVED] |
| 4:17PM |
2 |
Amazon Flexible Payment System - micropayments finally cracked? |
| 4:09PM |
0 |
Friday Feb 6th at 12 Noon EST: Polycom and Application Development |
| 3:56PM |
1 |
manager API |
| 2:56PM |
11 |
Crash Hard, Crash Often |
| 2:40PM |
0 |
Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ? |
| 2:17PM |
1 |
musiconhold realtime queue |
| 1:06PM |
1 |
SIP Authentication only on auth-user? |
| 12:00PM |
2 |
no need to dial areacode |
| 10:46AM |
1 |
extensions ending with "#"... |
| 9:08AM |
0 |
sendto syscall: EPERM (Operation not permitted) |
| 8:05AM |
2 |
Configure Asterisk to preserve SIP header? |
| 7:22AM |
2 |
TDM400P Circuit/channel congestion problem |
| 5:46AM |
2 |
Autodialler query |
| 12:17AM |
2 |
hardware that can accomondate 2 TDM24 |
| |
| Wednesday February 4 2009 |
| Time | Replies | Subject |
|---|
| 9:09PM |
0 |
[asterisk-dev] RFC 2833 DTMF w/ Level 3 Sonus |
| 8:29PM |
0 |
Problems with 9133i config |
| 7:47PM |
0 |
Problem with MOH and streaming music on 1.6.0.5 |
| 6:39PM |
1 |
Stopping chanspy followup |
| 4:09PM |
0 |
T1, FoneBRIDGE, and dropped D-Channel |
| 3:53PM |
2 |
Call parking |
| 3:37PM |
3 |
siemens hipath 4000 |
| 1:34PM |
2 |
question on originate call |
| 12:17PM |
2 |
escaping regular expression |
| 12:00PM |
0 |
BerkeleyTIP Feb 7 Sat Global Meeting - Ekiga3, Asterisk, KDE, GPGPU, Debian Edu, GStreamer |
| 8:01AM |
0 |
Audio lag on SIP connections... |
| 7:39AM |
0 |
Stopping chanspy |
| 6:16AM |
1 |
AOC-E pass through |
| 2:44AM |
3 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Tru nk with Pol o com Video Con ferencin g Un it |
| 2:42AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol o com Video Con ferencin g Un it |
| 2:41AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol o com Video Con ferencin g Un it |
| 2:39AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol o com Video Con ferencin g Unit |
| 2:37AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol ocom Video Con ferencin g Unit |
| 2:35AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Re : Trunk with Pol ocom Video Con ferencing Unit |
| 2:34AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Re : Trunk with Pol ocom Video Con ferencing Unit |
| 2:33AM |
0 |
Out of Office: Out of Office: Out of Office: Re : Trunk with Pol ocom Video Conferencing Unit |
| 2:31AM |
0 |
Out of Office: Out of Office: Re: Trunk with Pol ocom Video Conferencing Unit |
| 2:30AM |
0 |
Out of Office: Re: Trunk with Polocom Video Conferencing Unit |
| 1:02AM |
1 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : som e kind of t i meou t problem in pbx _sp ool.c |
| 1:00AM |
1 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t i meou t problem in pbx _sp ool.c |
| 12:59AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t i meout problem in pbx _sp ool.c |
| 12:58AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t i meout problem in pbx _spool.c |
| 12:57AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : some kind of t imeout problem in pbx _spool.c |
| 12:55AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Re : some kind of t imeout problem in pbx_spool.c |
| 12:54AM |
0 |
Out of Office: Out of Office: Out of Office: Ou t of Office: Re : some kind of t imeout problem in pbx_spool.c |
| 12:53AM |
0 |
Out of Office: Out of Office: Out of Office: Re : some kind of t imeout problem in pbx_spool.c |
| 12:52AM |
0 |
Out of Office: Out of Office: Re: some kind of t imeout problem in pbx_spool.c |
| 12:50AM |
0 |
Out of Office: Re: some kind of timeout problem in pbx_spool.c |
| |
| Tuesday February 3 2009 |
| Time | Replies | Subject |
|---|
| 11:57PM |
1 |
Warnings during a compile |
| 10:26PM |
2 |
some kind of timeout problem in pbx_spool.c |
| 9:59PM |
1 |
How to set udptl.conf ? |
| 8:58PM |
2 |
Broken Pipe error while using UpdateConfig command |
| 7:23PM |
1 |
app_rxfax.c: Channel T30 DONE < 0 -- incommplete fax reception. |
| 5:43PM |
1 |
Warning in CLI |
| 5:34PM |
1 |
dahdi-linux 2.1.0.4 released |
| 4:49PM |
0 |
fake ring again when using SIP |
| 4:15PM |
3 |
n-way conferencing |
| 4:04PM |
2 |
Contact lookup |
| 3:50PM |
1 |
Problem with building dahdi-linux RPM |
| 1:14PM |
1 |
Can't compile on NSLU2 LE |
| 12:39PM |
0 |
may convert SIP call in H.323 to words terminator?? |
| 10:42AM |
0 |
analysing tools |
| 9:59AM |
1 |
What's the difference between the Jabber Client Mode And Component Mode? |
| 3:27AM |
0 |
Patch to dahdi Chans.pm |
| 2:24AM |
2 |
RBS T1 DID issue |
| 12:25AM |
3 |
Videoconference one-to-many |
| |
| Monday February 2 2009 |
| Time | Replies | Subject |
|---|
| 8:18PM |
2 |
Invalid Extension |
| 7:21PM |
1 |
ChanSpy or other variant |
| 6:51PM |
0 |
SIP presence sample script |
| 5:39PM |
5 |
"No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail |
| 3:24PM |
2 |
Configuring Patton SmartNode with ISDN2e and Asterisk |
| 2:25PM |
0 |
dundi negative caching |
| 10:12AM |
1 |
Preferred Clock |
| 7:52AM |
0 |
EVRC support |
| 4:02AM |
2 |
Trunk with Polocom Video Conferencing Unit |
| |
| Sunday February 1 2009 |
| Time | Replies | Subject |
|---|
| 8:24PM |
0 |
iChat voice (and maybe video?) |
| 5:32PM |
5 |
Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine |
| 5:23PM |
1 |
asterisk-users Digest, Vol 54, Issue 109 |
| 10:44AM |
0 |
Strange Packet Behavior |
| 10:26AM |
3 |
Need some information on SS7 parameters |