| Saturday May 31 2008 |
| Time | Replies | Subject |
|---|
| 6:34AM |
1 |
IAX2 hardware video phone |
| 3:02AM |
0 |
SLA & Polycom |
| |
| Friday May 30 2008 |
| Time | Replies | Subject |
|---|
| 6:18PM |
2 |
cdr_odbc not working for some reason |
| 6:12PM |
1 |
How to have a difference dial tone after dialing the ignorepat digit |
| 4:45PM |
2 |
Dial() and the timeout to jump for next priority |
| 3:33PM |
2 |
Calls to Teliax dropping... |
| 3:21PM |
3 |
CDMA Phones |
| 1:49PM |
1 |
SPA 3102 unable to detect hangup |
| 11:46AM |
1 |
3Com Phones |
| 11:24AM |
1 |
Recommend a speaker/conference room SIP phone? |
| 6:57AM |
1 |
Problems CallerID in Call Out |
| 5:27AM |
0 |
Friday @12 PM EDT VOIP Users Conference |
| 12:02AM |
2 |
OT: Good phone to work with Asterisk/NAT |
| |
| Thursday May 29 2008 |
| Time | Replies | Subject |
|---|
| 10:09PM |
0 |
FIXED @1.4.11 - zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ? |
| 8:55PM |
1 |
polycom 501 call waiting. can hear it but can't pick it up |
| 4:29PM |
1 |
New York based Asterisk Users event this monday the 2nd of June at 6.30pm |
| 2:32PM |
2 |
Ideas for call back when all lines are busy |
| 2:26PM |
1 |
Channelized T1 |
| 12:42PM |
0 |
Skype connectivity? Siskyee or chanskype? |
| 10:25AM |
0 |
ILBC sh script throwing error |
| 9:44AM |
0 |
Asterisk Cnfiguration requirements? |
| 5:38AM |
2 |
Polycom 601 BLF multiple servers |
| 5:30AM |
2 |
Dialplan questions... |
| 3:58AM |
0 |
Zaptel 1.2.26 and 1.4.11 Released |
| 2:27AM |
2 |
Wait for input |
| |
| Wednesday May 28 2008 |
| Time | Replies | Subject |
|---|
| 9:09PM |
2 |
Does Digium support the B410P? |
| 6:13PM |
1 |
[Extensions.conf] Interval + discrete extension? |
| 5:56PM |
1 |
Zambia Africa |
| 4:53PM |
2 |
Calling '**1' through Asterisk |
| 4:23PM |
7 |
Cisco Gateway sending call to * without CID Name |
| 4:11PM |
4 |
transferring a not yet answered call |
| 3:13PM |
4 |
iLBC and Asterisk |
| 1:49PM |
3 |
Asterisk VoIP in Dubai/UAE? |
| 1:25PM |
1 |
Callgroup phone Notificacion |
| 10:31AM |
2 |
any pointers to mute/unmute a channel |
| 8:57AM |
3 |
Making remote calls via my Asterisk server |
| 8:56AM |
1 |
About Meetme |
| 7:31AM |
1 |
Linksys 942 and pickup function |
| 7:21AM |
1 |
length voicefile |
| 5:26AM |
1 |
why is CLIP(CallerID presentation) facility charged? |
| |
| Tuesday May 27 2008 |
| Time | Replies | Subject |
|---|
| 11:05PM |
0 |
callerid.c callerid_feed: Unknown IE |
| 10:02PM |
1 |
CDR "forking" for DUNDi calls |
| 9:43PM |
2 |
Trunk/Peering provider in Canada |
| 8:05PM |
2 |
Problem with Asterisk to Asterisk IAX trunk |
| 6:12PM |
2 |
ForkCDR |
| 6:09PM |
1 |
asterisk 1.4 QUEUES Variables |
| 5:11PM |
4 |
Asterisk just stops working... |
| 3:55PM |
1 |
SIP to SIP calls |
| 2:46PM |
4 |
Can Asterisk run fine on 64 bit CentOS 5.1? |
| 1:02PM |
0 |
Max SIP Registrations with and without Realtime |
| 12:37PM |
1 |
Max number of Registrations with and without Realtime |
| 10:05AM |
0 |
Is it possible to specify a timeout value to AGI or FastAGI applications ? |
| 9:28AM |
2 |
Why "should not run FastAGI on the same server" ? |
| 7:04AM |
3 |
Group pickup and CDR Logging / TAPI |
| |
| Monday May 26 2008 |
| Time | Replies | Subject |
|---|
| 11:19PM |
9 |
Voice low on ZAP |
| 9:36PM |
1 |
TE2XXP and modems |
| 9:23PM |
3 |
Card loading order... |
| 8:58PM |
1 |
Loud, obnoxious screech heard during ring |
| 7:44PM |
0 |
SIP Debug information |
| 4:15PM |
0 |
queues MEMBERINTERFACE variable |
| 4:10PM |
1 |
Long Distance Access |
| 4:06PM |
5 |
Skype Howto |
| 3:57PM |
3 |
Registration of multiple SIP-clients for the same extensions |
| 2:15PM |
1 |
Kevin Flemming live from Berlin NOW |
| 2:11PM |
1 |
Asterisk and Linksys SPA8000? |
| 1:01PM |
0 |
callback |
| 12:47PM |
1 |
Require a Virtual Private Server (VPS) to terminate calls |
| 12:00PM |
2 |
Installing a TDM410 card |
| 9:52AM |
1 |
Zaptel does not hangup |
| 9:15AM |
1 |
How to do not use Asterisk internal DB for SIP register? |
| 8:40AM |
0 |
realtime problem with two Asterisk servers - UPDATE |
| 6:30AM |
0 |
realtime problem with two Asterisk servers |
| 6:25AM |
0 |
Question about ISDNguard |
| 3:59AM |
1 |
Ubuntu as remote MySQL server |
| 12:36AM |
0 |
problem with e1 connection |
| 12:35AM |
1 |
AMI Redirect between MeetMe rooms |
| |
| Sunday May 25 2008 |
| Time | Replies | Subject |
|---|
| 9:51PM |
0 |
I invite you to join my Ziki Network ! |
| 5:06PM |
0 |
Incoming SIP call ring timeout |
| 4:58PM |
3 |
End call behaviour |
| 3:29PM |
1 |
No Audio on Meetme |
| 11:17AM |
1 |
Logical AND (resent due to bounces) |
| 10:47AM |
2 |
Logical AND |
| 9:45AM |
3 |
trying directrtpsetup |
| 8:49AM |
0 |
Cmd Dial (a group) and 1) who pick up the call 2) How to use the G option |
| 6:46AM |
1 |
Asterisk Tag Berlin live notes page |
| 1:33AM |
1 |
Error after upgrading from 1.2.18 to 1.4.20 |
| |
| Saturday May 24 2008 |
| Time | Replies | Subject |
|---|
| 6:09PM |
0 |
(Newbie)How to reduce security risks inopening IAX & Sip Ports |
| 6:04PM |
2 |
One way Speech issue - only in PAP2T to SIP device attached to Asterisk but not PAP2T to Voip service provider |
| 2:03PM |
1 |
Asterisk not answering calls |
| 1:52PM |
3 |
excessive bounces??? |
| 11:55AM |
2 |
Asterisk semi-hangs |
| 10:53AM |
2 |
digium cards with sangoma cards |
| 10:09AM |
2 |
Incoming calls not being answered by asterisk |
| |
| Friday May 23 2008 |
| Time | Replies | Subject |
|---|
| 10:43PM |
1 |
Strange ring or moh quality |
| 9:27PM |
2 |
*#&%! Polycom... |
| 9:08PM |
1 |
dialplan syntax error: need new eyes |
| 5:42PM |
1 |
[asterisk-dev] Asterisk 1.6 Realtime Database must use ', ' not '|' |
| 4:51PM |
0 |
OOH323 to Avaya S8500? |
| 4:24PM |
2 |
Strange State 6 on Channel X |
| 3:28PM |
0 |
Proposed changes for queue timeout |
| 3:26PM |
0 |
Asterisk chan Skype |
| 3:17PM |
1 |
Transfer |
| 2:15PM |
1 |
(no subject) |
| 1:56PM |
0 |
Asterisk/OpenSER users in Porto, Portugal? |
| 12:56PM |
2 |
New York Asterisk Users |
| 12:41PM |
2 |
B410P install |
| 9:02AM |
0 |
Todays at 12 Noon EDT: Mike Trest on large volume calling with asterisk |
| 8:05AM |
3 |
H.323 video support |
| 4:45AM |
3 |
forwarding pots lines |
| 4:02AM |
1 |
application sendtext |
| |
| Thursday May 22 2008 |
| Time | Replies | Subject |
|---|
| 8:36PM |
0 |
SIP configuration issues |
| 7:59PM |
1 |
Handling multiple fax machines and the fax extension, and general call routing |
| 7:35PM |
0 |
Continued NTL ISDN issues with Vox D210E card |
| 5:49PM |
7 |
reload stopping EVERYTHING on CLI and causing havoc. |
| 5:48PM |
1 |
Telco intercept prompts |
| 5:48PM |
1 |
Asterisk Wackyness |
| 5:42PM |
1 |
Adit 600 password reset |
| 5:37PM |
1 |
asterisk-addons 1.6.0 "Command 'realtime mysql status' |
| 5:23PM |
0 |
Call Park Logic? |
| 4:53PM |
0 |
asterisk-addons 1.6.0 "Command 'realtime mysql status' failed"? |
| 2:56PM |
2 |
More fun but with Wireshark capture |
| 2:54PM |
0 |
/home/putnopvut/asa/AST-2008-007/AST-2008-007: AST-2008-007 Cryptographic keys generated by OpenSSL on Debian-based systems compromised |
| 1:48PM |
2 |
Grandstream |
| 9:33AM |
3 |
call pick up |
| 8:48AM |
2 |
upgrade of asterisk .... to what? |
| 7:28AM |
1 |
Require Customization of SugarCRM or Vtiger |
| 7:02AM |
0 |
Require software like StarTel for CTI Pop Up ? |
| 7:00AM |
1 |
Fwd: - Asterisk Local channel |
| 6:59AM |
0 |
- |
| 6:57AM |
2 |
Meet you at Asterisk-tag in Berlin! |
| 6:44AM |
0 |
iax test |
| 12:46AM |
1 |
Call Placed through Manager connecting before the call connects. |
| 12:39AM |
2 |
1.4.20 delay |
| |
| Wednesday May 21 2008 |
| Time | Replies | Subject |
|---|
| 11:55PM |
1 |
Asterisk 3 way calling |
| 9:57PM |
1 |
using gtalk received instant messages in the dialplan |
| 9:37PM |
11 |
Asterisk Database Handling |
| 9:06PM |
9 |
ASP web phone |
| 7:31PM |
4 |
addons-1.6 not seeing installed MySQL packages |
| 4:22PM |
0 |
AsteriskNow Wizard not running? |
| 4:13PM |
0 |
Language Change on Polycom S IP 300 |
| 2:54PM |
3 |
3 ways |
| 2:39PM |
1 |
T38 fax solution with Asterisk possible? |
| 2:01PM |
0 |
Gentilini, Paul is out of the office. |
| 1:55PM |
6 |
Fax solution for Asterisk |
| 1:00PM |
5 |
asterisk and sipura 3102 (pstn to sip/sip to pstn calls) |
| 12:23PM |
0 |
iax2 received mini frame before first full voice frame |
| 11:48AM |
0 |
ACTIONTEC VOSKY EXCHANGE |
| 11:28AM |
1 |
speex, ilbc and g729 codecs, GSM with IAX |
| 6:22AM |
1 |
T.38 w/ MAX TNT & ASTERISK |
| 5:00AM |
0 |
Require US Toll free number |
| |
| Tuesday May 20 2008 |
| Time | Replies | Subject |
|---|
| 11:35PM |
2 |
Asterisk 1.4.20 Released |
| 8:33PM |
7 |
Busy out a zap channel? |
| 8:25PM |
2 |
Error Counters on PRI Circuit |
| 8:05PM |
0 |
Troubleshoot in-bound DTMF over PRI |
| 7:35PM |
0 |
asterisk black holing h263 |
| 5:41PM |
5 |
Digium announcement: new community manager - John Todd |
| 4:09PM |
0 |
183 Session Progress |
| 3:51PM |
2 |
At whit's end was 'DHCP Failure screws up system ' |
| 2:55PM |
1 |
Problem with Polycom forwarding |
| 1:03PM |
2 |
AsteriskNow: No Analog Hardware Detected |
| 12:55PM |
5 |
Server recommendation help |
| 12:14PM |
1 |
IVR for callee (called party) |
| 12:12PM |
0 |
[svn-commits] file: branch 1.4 r117081 - /branches/1.4/channels/h323/ast_h323.cxx |
| 12:10PM |
3 |
Newbie Voicemail: Just use one [context] invoicemail.conf?! |
| 9:58AM |
2 |
karaoke functionality |
| 8:41AM |
2 |
(Newbie)How to reduce security risks in opening IAX & Sip Ports |
| 7:58AM |
2 |
Newbie Voicemail: Just use one [context] in voicemail.conf?! |
| 3:44AM |
0 |
mute a call/ re-invite mid-session? |
| 3:27AM |
1 |
Fax Machine Options |
| 12:39AM |
1 |
Zaptel project being renamed to DAHDI |
| |
| Monday May 19 2008 |
| Time | Replies | Subject |
|---|
| 11:19PM |
3 |
Concept Clarifications |
| 8:45PM |
0 |
Not hearing first prompt |
| 8:33PM |
2 |
Recording problems, reinvites |
| 8:01PM |
0 |
Asterisk Jobs Spring Special - Free Postings for Employers! |
| 7:24PM |
0 |
VoIP Termination To Togo and Cameroon |
| 7:21PM |
1 |
Asterisk first time user |
| 6:45PM |
1 |
Dialplan Visualization, Rating System Unveiled! |
| 6:42PM |
1 |
DHCP Failure screws up system |
| 6:12PM |
3 |
Understanding Incoming sip DID handling |
| 6:09PM |
0 |
outbound calls on PRI all congested |
| 5:38PM |
3 |
Fedora 9 + Asterisk |
| 5:29PM |
3 |
Which best practices to build and deploy Asterisk on different hardware ? |
| 5:28PM |
1 |
nokia 770 has a build in mic, asterisk and iphone |
| 4:47PM |
3 |
Wireless headsets for Polycom phones |
| 2:17PM |
2 |
Is this the forum for Business Edition of Asterisk |
| 11:38AM |
0 |
zaptel errors update |
| 10:59AM |
1 |
(Newbie)Security Risks in opening IAX & Sip Ports |
| 10:59AM |
1 |
where did the switch statement come from? |
| 9:39AM |
0 |
- Failed to authenticate user |
| 9:20AM |
0 |
callerid |
| 8:58AM |
0 |
Dutch Asterisk mailing list |
| 6:44AM |
2 |
Extension not found |
| 5:08AM |
6 |
time limit |
| 4:53AM |
0 |
max retry |
| 4:42AM |
1 |
Recall: Newbie Asterisk: Install Asterisk as non-root |
| |
| Sunday May 18 2008 |
| Time | Replies | Subject |
|---|
| 9:17PM |
3 |
Inbound Answer not working |
| 8:46PM |
2 |
Dutch Asterisk mailing list? |
| 7:33PM |
1 |
strange name alt.asterisk.canary.tweet.tweet.tweet |
| 10:57AM |
1 |
Bridging a call on hold with an active call |
| 7:43AM |
0 |
Asterisk users and Twitter |
| 12:40AM |
1 |
Paging |
| |
| Saturday May 17 2008 |
| Time | Replies | Subject |
|---|
| 9:50PM |
1 |
Using a Loopback Plug for an RJ-45 Ethernet Interface for testing a Digium Card |
| 8:15PM |
1 |
Hangup issue |
| 7:21PM |
0 |
Local loopback vs SIP/IAX2 |
| 7:01PM |
1 |
zaptel 1.4.10 doesn't build on debian etch epia itx system |
| 6:18PM |
1 |
asterisk virtualization on VMWARE SX infrastructure |
| 5:40PM |
0 |
sipbroker CLI |
| 5:38PM |
2 |
Googles 411 services |
| 1:11PM |
0 |
trixbox, sangoma a200, dell poweredge |
| 11:47AM |
1 |
One way sound when Using Dial cmd without "t" option (SOLVED) Need explanation |
| 10:08AM |
3 |
Implementation of Video Conferencing using Asterisk |
| 6:40AM |
1 |
More dialplan visualization (neat graphs!) |
| 3:35AM |
0 |
CFP For HITBSecConf2008 - Malaysia Now Open |
| 12:17AM |
0 |
Asterisk peer definition with multiple host ip addresses |
| |
| Friday May 16 2008 |
| Time | Replies | Subject |
|---|
| 11:50PM |
3 |
Queue Stats |
| 9:10PM |
2 |
Fetching Binary data from SQL Server |
| 8:42PM |
1 |
Alternate names in Directory (dial-by-name) |
| 7:40PM |
0 |
Digium TDM4xx CID problem |
| 5:48PM |
1 |
trixbox, sangoma a200, dell poweredge 2550 issue |
| 4:35PM |
1 |
PRI debugging ... |
| 2:28PM |
2 |
Connecting a PSTN gateway to Asterisk using PRI |
| 1:46PM |
1 |
Problem with cisco 7970G [j.munoz@epsilontel.com] |
| 9:37AM |
3 |
PBX deployment big problems: Voip traffic analysis |
| 7:47AM |
7 |
Asterisk concurrent calls count |
| 7:36AM |
3 |
Problems passing variables from a macro |
| 3:39AM |
1 |
queue autopause |
| 3:38AM |
1 |
A couple of newbie questions |
| |
| Thursday May 15 2008 |
| Time | Replies | Subject |
|---|
| 11:26PM |
7 |
playing .gsm sounds through a web browser |
| 9:42PM |
3 |
Not hearing first prompts |
| 9:41PM |
0 |
Asterisk for Larg (Al Baker) |
| 9:39PM |
2 |
QOS and Asterisk |
| 9:04PM |
1 |
Received SIP subscribe for peer without mailbox |
| 7:14PM |
1 |
Where does menuselect save your choices? |
| 6:59PM |
1 |
Citel Gateways |
| 4:23PM |
0 |
how to find the logs for this problem |
| 4:11PM |
7 |
Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling? |
| 4:06PM |
1 |
*72 Telco Call Forwarding |
| 3:47PM |
1 |
ChanSpy not working - "transmit frame type 64" warning |
| 1:37PM |
1 |
caller-id on X100P fails frequently |
| 11:05AM |
4 |
are channel names unique |
| 10:57AM |
1 |
Problem while running Flash Operator Panel |
| 8:17AM |
4 |
Newbie Asterisk: Install Asterisk as non-root |
| 5:50AM |
0 |
Friday May 16th @1 Noon EDT: VoIP Users Conference is about Click 2 Call |
| 4:49AM |
0 |
Listen And Talk mode differentiation of meetme() conference |
| 4:08AM |
0 |
Fw: voicemail not sending emails |
| 3:31AM |
0 |
Polycom IP300 Language |
| 1:54AM |
0 |
monitoring names |
| |
| Wednesday May 14 2008 |
| Time | Replies | Subject |
|---|
| 11:45PM |
9 |
Polycom XML Files / asterisk |
| 7:52PM |
3 |
Question about SS7 |
| 7:39PM |
6 |
anyone from Joplin, MO |
| 5:02PM |
0 |
Setting CallerID UNKNOWN on an outgoing |
| 4:16PM |
1 |
Asterisk 1.4.20-rc3 and 1.6.0-beta9 Now Available |
| 2:31PM |
3 |
Announcing the first North America Druid Meetups happening Chicago 22 May 2008 and Altanta 27 May 2008 |
| 1:57PM |
2 |
Understanding Asterisk |
| 9:32AM |
0 |
[asterisk-dev] UWB Codec / Command-line softphone help |
| 8:48AM |
0 |
No sound with Playback() and Background() |
| 6:08AM |
2 |
Setting CallerID UNKNOWN on an outgoing call |
| 2:22AM |
3 |
No-mobo PC for USB Drives Enclosure? |
| |
| Tuesday May 13 2008 |
| Time | Replies | Subject |
|---|
| 11:51PM |
1 |
Zaptel Install Error |
| 11:48PM |
6 |
voicemail not sending emails |
| 9:55PM |
2 |
[asterisk-biz] ANI |
| 8:56PM |
2 |
Installation Question |
| 8:44PM |
0 |
Call retard from a softphone to a hardphone |
| 7:50PM |
7 |
BLF Compatible Phones |
| 6:31PM |
3 |
Call only for registered sip users... |
| 6:18PM |
0 |
Asterisk 1.4.19.2 Released |
| 5:27PM |
1 |
queue problem |
| 4:38PM |
1 |
More one way audio... |
| 3:47PM |
0 |
Fwd: [asterisk-dev] Paging intercom extensions |
| 3:46PM |
0 |
Asterisk-Tag.org conference, May 26th/27th, Berlin, Germany |
| 3:45PM |
2 |
How to test dialplan w/o a trunk |
| 12:44PM |
3 |
Queuing if no one available to answer |
| 12:35PM |
0 |
Paging for analoge devices |
| 12:35PM |
1 |
Control of individual call legs |
| 11:55AM |
0 |
Extension Auto Fall through help when matching fails. |
| 11:17AM |
1 |
New Asterisk Deployment - Need some tips |
| 10:29AM |
1 |
cannot get calls with Tellfree brazilian provider |
| 10:11AM |
2 |
Asterisk stops MOH on transfer |
| 7:44AM |
1 |
chan_mobile install with Asterisk 1.4.19 |
| 6:18AM |
0 |
Newbie Polycom: Cannot Disable Services button |
| |
| Monday May 12 2008 |
| Time | Replies | Subject |
|---|
| 11:43PM |
2 |
Newbie Dialplan: Best Practice in using Context - Do not use Default?? |
| 10:42PM |
0 |
Require a Touch-Tone to Connect? proof of concept with meetme() |
| 10:21PM |
0 |
business class sip provider with a SIP proxy server in India ? |
| 8:57PM |
2 |
Which sound file formats? |
| 8:44PM |
1 |
Is there a way to have Manager Bridge Channel without being connected |
| 7:05PM |
1 |
Crappy sound on Console (chan_oss) |
| 6:07PM |
1 |
Problem with SIP Subscription Status |
| 5:32PM |
0 |
externip not working... |
| 5:30PM |
1 |
"module reload" CLI Asterisk question |
| 4:19PM |
1 |
Using multiple variables in SIP.CONF setvar |
| 3:29PM |
2 |
3U server chassis & Digium TE405P? |
| 3:02PM |
1 |
exten => pattern match query |
| 2:42PM |
1 |
x100p card or similar in India |
| 2:36PM |
1 |
Sangoma and Voicetronix cards |
| 2:28PM |
1 |
Lone worker system |
| 1:34PM |
0 |
"module reload" question |
| 1:09PM |
0 |
test message please do not reply and clog up the list |
| 6:36AM |
1 |
Escape characters or replace function |
| |
| Sunday May 11 2008 |
| Time | Replies | Subject |
|---|
| 5:51PM |
0 |
Digium AEX410 |
| 4:24PM |
3 |
Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect? |
| 7:23AM |
2 |
Use safe_asterisk manually, you get colors in CLI. Crontab it, you don't. |
| |
| Saturday May 10 2008 |
| Time | Replies | Subject |
|---|
| 6:26AM |
1 |
G.722 for polycom |
| 4:33AM |
1 |
ztdummy problems |
| |
| Friday May 9 2008 |
| Time | Replies | Subject |
|---|
| 10:36PM |
1 |
Unable to build Hpec on Zaptel 1.4.10.1 |
| 9:32PM |
3 |
Calls on E1 TDMoE span are dropped at random |
| 8:24PM |
2 |
Polycom Advanced Features |
| 3:52PM |
1 |
Polycom causes conference to fail |
| 3:36PM |
2 |
DTMF lose with TE-121F |
| 3:19PM |
11 |
Best Linux distribution to use in Asterisk server |
| 2:14PM |
1 |
Asterisk ZRTP? |
| 8:49AM |
2 |
app_queue New Function ToggleQueueMemberUse() |
| 8:09AM |
0 |
Dial Command with the L switch again |
| 6:44AM |
0 |
VUC Friday May 9 @ 12 Noon EDT: Asterisk 3rd party licensing platform |
| 5:07AM |
1 |
-zapg729toulaw did not update samples 160 |
| 4:09AM |
0 |
t38modem |
| 3:07AM |
0 |
Zaptel ring voltage detection |
| 2:38AM |
1 |
Text for built-in recordings |
| |
| Thursday May 8 2008 |
| Time | Replies | Subject |
|---|
| 11:43PM |
0 |
I hear noise in the line |
| 10:17PM |
0 |
Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA) |
| 9:46PM |
1 |
MOH and Licensed G729 codec |
| 9:28PM |
2 |
Out-Going Callerid |
| 9:22PM |
1 |
Zap Channels Collide (Incoming & Outgoing) |
| 8:54PM |
3 |
Which Cepstral Voice to license |
| 6:06PM |
0 |
chan_sip Maximum retries exceeded on transmission |
| 4:39PM |
0 |
(no subject) |
| 4:22PM |
1 |
Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA) Excessive Echo (only to the sip party) |
| 3:46PM |
3 |
help with rotating number plan |
| 2:39PM |
1 |
Manager API - Setvar not working |
| 10:00AM |
2 |
SLN File Format |
| 8:25AM |
3 |
Newbie Queue: tricky problem with MOH |
| 5:31AM |
3 |
Looking for a Snom expert |
| 5:22AM |
0 |
md5secret for IAX? |
| 2:08AM |
2 |
dundi network - redundancy / fault tolerance ? |
| |
| Wednesday May 7 2008 |
| Time | Replies | Subject |
|---|
| 10:38PM |
1 |
Big difference in CPU utilization with MeetMe |
| 10:07PM |
1 |
RE:Asterisk 3rd party developed commercial software sales licensing platform |
| 9:39PM |
1 |
show CODEC in CDR |
| 9:36PM |
1 |
Mediatrix 2102's |
| 9:11PM |
2 |
Realtime status feature - user feedback needed |
| 8:49PM |
1 |
URGENT |
| 4:49PM |
4 |
VOICEMAIL OPTIONS help needed |
| 2:24PM |
0 |
Problem using the sip_header-function |
| 1:39PM |
0 |
SLA in 1.4.18: i'm going crazy. |
| 12:57PM |
1 |
voice mail indicator on phone |
| 12:39PM |
0 |
reINVITE with Dial() options -- bug 0010647 |
| 12:20PM |
1 |
Ubuntu 8.04 + Astribank |
| 11:59AM |
2 |
Setting the TOS using IPtables screws up the DSCP field |
| 10:08AM |
1 |
update DB on ringing/ catch ringing event |
| 9:36AM |
1 |
cdr question |
| 8:48AM |
0 |
IAX IP Trunk + GSM Codec and Noise in the Polycom IP Phone 320 |
| 5:34AM |
0 |
queue |
| 5:10AM |
4 |
Newbie alert: VoIP hardware |
| 3:42AM |
0 |
phone status question |
| 2:34AM |
3 |
better enumlookup handler |
| 1:44AM |
1 |
Melbourne Asterisk night |
| |
| Tuesday May 6 2008 |
| Time | Replies | Subject |
|---|
| 10:54PM |
2 |
Receptionist SNOM-360 |
| 7:33PM |
2 |
Storing voicemail on samba share |
| 7:21PM |
3 |
asterisk queue cluster |
| 5:45PM |
1 |
how do I set callerid for incoming iax? |
| 5:24PM |
2 |
Mixmonitor recording issue |
| 4:01PM |
2 |
PRI D-Channel reconfiguration = crash asterisk? |
| 3:57PM |
0 |
finding Asterisk user group users and enthusiasts in the (Salt Lake City Utah, Chicago IL, Boston MA, Tampa FL) |
| 3:03PM |
0 |
Asterisk in Production |
| 2:20PM |
3 |
Performance issues |
| 1:59PM |
2 |
Basic modules of Asterisk |
| 1:55PM |
1 |
DUNDi call impossible in one direction |
| 11:38AM |
11 |
Asterisk in Production ? |
| 1:29AM |
1 |
using cell phone as an FXO port |
| 12:18AM |
0 |
Passing Dial URL argument through Asterisk |
| |
| Monday May 5 2008 |
| Time | Replies | Subject |
|---|
| 11:59PM |
4 |
Running Asterisk as root |
| 11:09PM |
2 |
AGI - Choppy Sound |
| 8:32PM |
1 |
Call manager using Asterisk as voicemail server (SIP) not working ... |
| 7:59PM |
2 |
DTMF |
| 7:04PM |
0 |
Salesforce.com - ribbit mashup |
| 6:47PM |
0 |
Adtran TA-750 channels go onhook |
| 4:54PM |
2 |
T38 Passthrough Verification |
| 3:53PM |
1 |
Asterisk & Bluetooth |
| 3:29PM |
0 |
FW: [Red5] Open-source SIP phone with Red5 and Flex3 |
| 2:51PM |
5 |
Asterisk Restarting due to segfault |
| 2:19PM |
0 |
TDM410 and TE205P in a dell 2850 |
| 2:14PM |
2 |
Unicall - How to automatically block collect calls |
| 1:59PM |
3 |
TDM410P driver? |
| 1:26PM |
0 |
Problem with transfer (and asterisk -r) |
| 12:29PM |
4 |
[OT] wireless headphone that can answer a call? |
| 11:24AM |
4 |
FW: Asterisk 3rd party developed commercial software sales licensing platform |
| 9:44AM |
3 |
MeetMeAdmin() working problem |
| 8:47AM |
2 |
ResetCDR() - v 1.4.19.1 |
| 8:41AM |
1 |
Playback don't play the beginning if a sound file |
| 8:40AM |
3 |
simple realtime question |
| 8:07AM |
0 |
hangup occurs after 2 rings with B410P |
| |
| Sunday May 4 2008 |
| Time | Replies | Subject |
|---|
| 7:38PM |
2 |
Astricon? |
| 5:55PM |
3 |
Dialplan, Extensions, etc. Worksheet |
| 5:09PM |
1 |
segmentation fault |
| 3:55PM |
1 |
externnotify php script |
| 5:49AM |
0 |
Director Server |
| 12:26AM |
1 |
UK BT ISDN30e PRI Problem |
| |
| Saturday May 3 2008 |
| Time | Replies | Subject |
|---|
| 10:29PM |
1 |
AGI asterisk high balance |
| 6:48PM |
2 |
RTP and Sip Provider |
| 5:05PM |
0 |
Apple iPhone visual voicemail |
| 4:19PM |
3 |
Anonymous statistics collection tool for Asterisk servers? |
| 3:27PM |
2 |
China vaults past USA in Internet users - now 220 million users in China |
| 3:18PM |
2 |
toggling recordings on / off using MeetMe() |
| 10:32AM |
0 |
Zap channels hang |
| 7:54AM |
0 |
Attended transfers with original CID information - Polycom |
| 1:51AM |
0 |
Photos and Presentation Materials from HITBSecConf2008 - Dubai Released |
| 1:13AM |
1 |
Asending or Round robin with trunks sip |
| |
| Friday May 2 2008 |
| Time | Replies | Subject |
|---|
| 10:34PM |
0 |
OT: VoIP 911 Call Misrouted; Child Dies |
| 8:14PM |
1 |
DTMF issues in 1.4.19 with missing digits |
| 6:28PM |
0 |
Problems with a BRI Line |
| 6:22PM |
0 |
One Way Audio After Dial |
| 6:15PM |
0 |
voice mail indicator command |
| 5:25PM |
1 |
rebooting newer cisco phones |
| 4:46PM |
0 |
SRTP between 2 asterisks |
| 2:57PM |
0 |
12 Noon Friday VoIP Users Conference "Weak-ly" Reminder |
| 2:51PM |
2 |
sip show peers |
| 1:58PM |
1 |
Asymmetric codecs in IAX2 trunk |
| 1:46PM |
0 |
Macro |
| 1:45PM |
0 |
Connecting Elite H323 gateways |
| 10:07AM |
7 |
Digium Card: Power Connector, from SATA to NORMAL |
| 5:04AM |
1 |
Asterisk - get Caller String(as per key action) |
| 2:27AM |
2 |
call pickup - Asterisk 1.4.19.1 - |
| 1:35AM |
0 |
sound quality drop after call transfer |
| 12:34AM |
2 |
Stupid Timeout Question |
| 12:25AM |
4 |
e164 Format Numbers |
| |
| Thursday May 1 2008 |
| Time | Replies | Subject |
|---|
| 11:47PM |
0 |
Asterisk 1.4.20-rc1 Now Available |
| 10:00PM |
8 |
New generic sounds |
| 9:35PM |
1 |
Sound Prompt 'per' |
| 8:41PM |
0 |
Providers |
| 8:19PM |
3 |
Minimum upload speed for Asterisk? |
| 8:10PM |
1 |
http://www.asteriskdocs.org/html/apas02.html |
| 5:07PM |
3 |
Zaptel 1.4.10.1 Released |
| 3:39PM |
3 |
Digium PRI card hi-Z for sniffing? |
| 3:38PM |
3 |
How i know the version of my vpmadt032 firmware |
| 2:05PM |
0 |
Background ring |
| 11:59AM |
0 |
PCI ISDN as a PSTN gateway card |
| 9:45AM |
0 |
TEST MAIL |
| 9:45AM |
1 |
ast_indicate_data: Unable to handle indication 3 |
| 8:49AM |
0 |
Penalty based Cascading Queue - possible ? |
| 7:17AM |
3 |
Newbie: How to remote test a call prolem in an Asterisk site? |
| 6:54AM |
1 |
Remote host can't match request NOTIFY??? |