| Sunday August 31 2003 |
| Time | Replies | Subject |
|---|
| 1:39PM |
5 |
Newbie IVR question |
| 10:43AM |
2 |
DBSaveTree & DBLoadTree |
| 7:04AM |
4 |
Newbie - setup help |
| 12:54AM |
0 |
(no subject) |
| 12:47AM |
1 |
Message-waiting-indicator thru ZAP interfaces - how to? |
| 12:24AM |
2 |
ENUM, iax,iax2 and h323? |
| |
| Saturday August 30 2003 |
| Time | Replies | Subject |
|---|
| 7:36PM |
2 |
ATA 186 & DynExtenDB (query extensions vía sql) |
| 3:35PM |
0 |
Caller Id Issues |
| 1:40PM |
1 |
Filling PHP Variable from EXTENSION in AGI |
| 10:38AM |
1 |
Incomming call issue |
| 6:28AM |
3 |
Conference without zaptel?? |
| 3:23AM |
7 |
Installation Problem |
| 2:31AM |
0 |
OT: My congestion music. |
| |
| Friday August 29 2003 |
| Time | Replies | Subject |
|---|
| 11:04PM |
2 |
Packet8 DTA310 |
| 7:54PM |
0 |
Queue timeouts |
| 7:35PM |
4 |
sample configs |
| 5:24PM |
6 |
Asterisk and Cisco 7960 |
| 1:58PM |
0 |
voicemail.conf emailbody fromaddress |
| 1:33PM |
3 |
G729 error messages |
| 1:12PM |
3 |
Restricting concurrent SIP calls |
| 11:16AM |
2 |
sip and pix |
| 9:10AM |
1 |
Version number not updating after CVS |
| 7:47AM |
6 |
Festival and Asterisk |
| 6:41AM |
1 |
X100P not hanging up. |
| 5:22AM |
1 |
Buffering DTMF input |
| 3:50AM |
1 |
additional digit in front of the dialed extenesion by outgoing pri/E1 call |
| |
| Thursday August 28 2003 |
| Time | Replies | Subject |
|---|
| 7:44PM |
1 |
(no subject) |
| 3:43PM |
2 |
Video conference apps/appliances? |
| 3:12PM |
1 |
Problems with TDM400P & X100P |
| 12:36PM |
0 |
Re: Three way calling on outgoing FXO line (Martin Pycko) |
| 12:27PM |
0 |
AgentLogin and Huntgroups |
| 10:26AM |
1 |
X100P in Spain & Busy Detect |
| 9:18AM |
1 |
Three way calling on outgoing FXO line |
| 9:08AM |
12 |
Asterisk stops responding |
| 8:16AM |
6 |
SIP and ECHO |
| 8:00AM |
0 |
Echo cancellation problem from SIP to PSTN |
| 2:13AM |
0 |
Asterisk -> Intertex |
| |
| Wednesday August 27 2003 |
| Time | Replies | Subject |
|---|
| 5:15PM |
0 |
* newbie: overhead paging and nbsd |
| 1:22PM |
0 |
Chan_h323/g729 - X100P connecting to non-Digium Partner |
| 12:38PM |
2 |
Configuration Adtran TA 750 |
| 11:46AM |
1 |
Default Flash Time |
| 11:28AM |
1 |
PCI X100P card interrupt problems |
| 9:42AM |
3 |
Question About BRI Cards |
| 9:31AM |
3 |
ADSI Programs |
| 8:59AM |
1 |
Polycom SoundPoint 500 with Asterisk |
| 7:27AM |
1 |
sample configs / load module failure |
| 6:46AM |
2 |
include context |
| 2:32AM |
0 |
Registering via IAX2 succeeds, but bridging to the registered peer fails |
| 1:58AM |
0 |
H323 caller ID |
| 12:39AM |
2 |
SNOM 200 bugs |
| 12:10AM |
3 |
conference authorization |
| |
| Tuesday August 26 2003 |
| Time | Replies | Subject |
|---|
| 10:00PM |
0 |
Chan_h323 does not seem to send the destimation number to gateway |
| 8:55PM |
0 |
Chan_h323 support for phone numbers via gateway? |
| 6:33PM |
1 |
H.323 channel problems |
| 6:13PM |
0 |
have everyone use asterisk to setup a network as iconnecthere or quicknet? |
| 5:19PM |
0 |
Ordering ISDN PRI |
| 2:55PM |
1 |
Dialed Number Identification in analog hunt group |
| 1:30PM |
2 |
x100P: Ring/off-hook in strange state 6 on channel1 |
| 11:47AM |
0 |
Pickup groups with SIP |
| 10:02AM |
0 |
Hardware Requirement for Asterisk PBX |
| 9:36AM |
1 |
More questions. Call Waiting and Threeway |
| 8:13AM |
2 |
Asterisk internal database access |
| 8:11AM |
1 |
Problem starting Asterisk after abnormal shutdown |
| 7:36AM |
0 |
Accountcode and cdr-csv |
| 6:35AM |
0 |
Forward but wait for acknowledgement |
| 6:25AM |
0 |
phonecore in gnophone? |
| 5:04AM |
0 |
bug report: whitespaces in uris |
| 4:22AM |
4 |
* server based Phonebook |
| 4:12AM |
0 |
Decent DECT cordless compatible with Asterisk/ATA? |
| 3:44AM |
0 |
TDM10M && Siemens Euroset 2015 |
| 3:02AM |
1 |
Alias limitation in asterisk-oh323.0.5.5 |
| |
| Monday August 25 2003 |
| Time | Replies | Subject |
|---|
| 11:13PM |
2 |
Chan_h323 and a Cisco Gateway |
| 10:21PM |
1 |
gnophone connection |
| 8:21PM |
0 |
Voicetronix V4PCI |
| 3:45PM |
6 |
SIP vs SCCP vs XML |
| 3:02PM |
0 |
FXO gateway experience? |
| 2:59PM |
1 |
Secondary gatekeeper support by asterisk h323 drivers |
| 2:15PM |
1 |
Intercom with Cisco SIP 796x phones? |
| 2:09PM |
2 |
0 out of voicemail to different secretaries |
| 1:58PM |
0 |
Fwd: Data calls through * |
| 1:35PM |
1 |
Problems reloading |
| 12:13PM |
2 |
Warning from chan_zap ring requested |
| 12:10PM |
2 |
Data calls through * |
| 11:28AM |
4 |
T100P/ TSU 600 installation problem |
| 11:20AM |
1 |
Unified messaging. |
| 11:16AM |
1 |
Is the DTMF bug in bugs.digium.com what number. |
| 10:37AM |
0 |
te410p with serial console fails with error: TE410P: Double/missed interrupt detected |
| 10:33AM |
4 |
Cisco 7940 SIP |
| 10:28AM |
0 |
RE: Asterisk-Users digest, Vol 1 #1133 - 18 msgs |
| 10:05AM |
0 |
Caller ID and Call Waiting. |
| 9:20AM |
6 |
Syncronize Monitored Calls |
| 8:39AM |
0 |
No Audio on SIP Phone Connection |
| 8:34AM |
0 |
Manager interface & Event:Leave |
| 8:22AM |
0 |
ISDN Sub Address |
| 6:54AM |
2 |
SetVar on sample.call |
| 6:11AM |
2 |
call center - operators not using phone keys |
| 4:33AM |
11 |
Why doesnt anyone reply me ? |
| 4:19AM |
1 |
I4L CallerID not working |
| 2:54AM |
13 |
SIP phones |
| 2:51AM |
3 |
Grandstream firmware update DMTF Payload Type |
| 2:26AM |
0 |
Grandstream firmware update. {HTTP error} |
| 2:21AM |
0 |
ENUM on Asterisk |
| 1:33AM |
1 |
chan_zap.c zt_rec: Unknown error 500 |
| |
| Sunday August 24 2003 |
| Time | Replies | Subject |
|---|
| 10:42PM |
2 |
GS on ebay... |
| 9:49PM |
1 |
GS geek info |
| 9:39PM |
5 |
T1 to T1 on asterisk? |
| 8:18PM |
0 |
Music on hold - multiple formats |
| 1:12PM |
1 |
Any way to distinguish between... |
| 12:46PM |
2 |
line numbering and gosub |
| 7:30AM |
2 |
Sound files for internal functions in another place?? |
| 6:47AM |
1 |
Grandstream firmware update. |
| 1:52AM |
4 |
fatal embrace control in menus ? |
| 1:22AM |
0 |
Non Asterisk - Apology to the list |
| 12:05AM |
0 |
X100P disconnecting without any reason |
| |
| Saturday August 23 2003 |
| Time | Replies | Subject |
|---|
| 10:43PM |
2 |
DIAL via CLI missing |
| 8:09PM |
3 |
Grandstream and CallerID not working |
| 5:41PM |
1 |
Private ENUM examples? |
| 5:09PM |
0 |
[Asterisk-Dev] Re: SIP change... |
| 12:32PM |
1 |
There is any cache for sound files? |
| 11:43AM |
1 |
Webmin and incoming call recording |
| 11:09AM |
0 |
One-way audio using console |
| 5:28AM |
0 |
callerid, callwaiting callerid, Asterisk and ATA |
| 12:09AM |
2 |
SIP change... |
| |
| Friday August 22 2003 |
| Time | Replies | Subject |
|---|
| 11:05PM |
0 |
Intresting Vonage story... |
| 9:44PM |
10 |
Intresting.. hrm |
| 9:01PM |
6 |
Caller ID problem |
| 8:41PM |
1 |
pardon the newbie question |
| 8:39PM |
0 |
Warning message in /var/log/asterisk/messages |
| 8:01PM |
2 |
Game time is over gang |
| 7:01PM |
4 |
CVS Question |
| 4:39PM |
0 |
dtmf/audio before going offhook |
| 11:40AM |
0 |
"Frame rejections" on E1 trucks |
| 9:05AM |
5 |
DTMF tones not long enough on out going call s |
| 8:34AM |
1 |
cdr_csv actual duaration |
| 8:33AM |
0 |
DTMF tones not long enough on out going calls |
| 7:08AM |
3 |
Need a "trick" to generate calls |
| 4:26AM |
1 |
Slowly get it ... Hardware |
| 4:16AM |
6 |
sox and wav to gsm conversion quality issue |
| 2:43AM |
1 |
Best SIP phone? |
| 2:03AM |
1 |
IAXtel + NAT |
| 1:54AM |
0 |
Crash using alsa |
| 12:47AM |
2 |
Re: ATAs |
| 12:43AM |
0 |
Pager support |
| |
| Thursday August 21 2003 |
| Time | Replies | Subject |
|---|
| 10:32PM |
1 |
Structured release, Maillists |
| 9:22PM |
0 |
Dial in modem speeds over VoIP? |
| 4:36PM |
0 |
problem with manager: Response error, Missing action in request |
| 1:39PM |
1 |
Voicemail2 and RFC2833 DTMF |
| 1:26PM |
4 |
Asterisk + SNOM + Pound and star keys |
| 1:25PM |
1 |
Working example of "switch"? |
| 1:20PM |
0 |
Which linux soft phone is best with asterisk. |
| 1:17PM |
5 |
Background Noise |
| 12:34PM |
1 |
Status of ISDN && DTMF (AFAIK): Please add corrections and comments |
| 12:27PM |
1 |
Minnesota PUC: Phone rules apply to VoIP |
| 11:46AM |
2 |
Re: Some questions about Asterisk and reliability |
| 10:49AM |
0 |
RTP channel |
| 10:24AM |
0 |
ATA 186 - X-Lite and Asterisk |
| 9:51AM |
1 |
Subject: Provisioning CO lines |
| 8:36AM |
2 |
Grandstream Budgetone Defective Units |
| 8:25AM |
0 |
Zaptel.conf & digium E100P |
| 8:21AM |
2 |
Xphone Lite Cannot make work on Asterisk |
| 8:20AM |
7 |
Provisioning CO lines |
| 8:08AM |
7 |
AGI Channel Status |
| 8:06AM |
1 |
Question on setting up MeetMe conference bridge |
| 7:50AM |
3 |
Sending dtmf over an ougoing call from asterisk |
| 7:14AM |
1 |
Multi-extension buttoned phones |
| 6:54AM |
2 |
Newbie Question / ISDN |
| 6:33AM |
1 |
Cisco 79xx XML carriage returns/line feeds |
| 4:58AM |
1 |
asterisk-oh323 v0.5.5 |
| 3:56AM |
0 |
No audio in either direction, sip channels hanging, asterisk will not shut down. |
| 2:12AM |
3 |
Conference + time limit |
| 1:52AM |
0 |
Asterisk BoF: Boston, Sept _2[2-4] - interested? |
| 1:32AM |
2 |
911, networks of * servers, etc. (was: VOIP Dialtone?) |
| 12:43AM |
1 |
Asterisk and RTP flow |
| 12:31AM |
1 |
Configurable auto forward in Asterisk |
| |
| Wednesday August 20 2003 |
| Time | Replies | Subject |
|---|
| 11:32PM |
2 |
BudgeTone Firmware 1.0.3.78? |
| 8:19PM |
0 |
DTMF Tone length |
| 1:52PM |
0 |
reload problem fixed |
| 1:20PM |
13 |
VoIP dialtone? |
| 12:39PM |
2 |
PRI CallerID problem |
| 12:01PM |
1 |
Asterisk introductory talk: Portland, OR USA |
| 11:48AM |
2 |
RTP header compression? |
| 11:27AM |
1 |
ATA-186 locking: implausible unlock method |
| 11:16AM |
1 |
IAX to zaptel echo |
| 10:49AM |
2 |
Strange happenings |
| 10:15AM |
0 |
Adtran TA 750 |
| 9:45AM |
1 |
IAX <> IAX trunking... DP cache? |
| 9:33AM |
1 |
AudioCodes MP108 8-Port FXO Analog Gateway (SIP) |
| 8:42AM |
1 |
X-Lite Build 1059 problems |
| 8:42AM |
14 |
Is Asterisk ready for "real" use? |
| 8:29AM |
1 |
VAD (silence suppression) on Asterisk |
| 8:14AM |
0 |
Re: Asterisk diskless server, a web page with more info? |
| 8:11AM |
9 |
Hardware question |
| 6:53AM |
0 |
App Directory issues-again? |
| 6:49AM |
2 |
ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank |
| 6:49AM |
4 |
Conference call |
| 6:04AM |
2 |
reload not working |
| 4:38AM |
1 |
Queue |
| 2:47AM |
0 |
snom100(with latest firmware) screeching noise when doing transfers, |
| 2:09AM |
0 |
Dialogic cards... |
| 12:20AM |
1 |
SIP using which codec? |
| 12:07AM |
2 |
weird error message with zaptel |
| |
| Tuesday August 19 2003 |
| Time | Replies | Subject |
|---|
| 11:42PM |
8 |
Where to find correct ver of OpenH323 & PWLIB for Chan_h323 |
| 11:23PM |
1 |
echo on the sip side |
| 11:17PM |
6 |
Limit Number of user in Conference |
| 9:03PM |
3 |
Compile problems |
| 6:46PM |
0 |
Re: Open source IP phone, maybe? |
| 5:40PM |
1 |
Speex & openh323 |
| 5:16PM |
1 |
Problem with * server and FWD |
| 2:21PM |
0 |
# Transfer context problem |
| 2:12PM |
2 |
trying to make a X100P work |
| 2:06PM |
2 |
Re: Open source IP phone, maybe? |
| 1:23PM |
0 |
current status of i4l and dtmf stuff |
| 12:30PM |
0 |
SPAMWARNING:216.207.245.21:RE: Re: LAN switches with PoE? PoE phones? |
| 12:09PM |
2 |
Vonage locked ATA-186 question |
| 11:35AM |
1 |
Oen source IP phone, maybe? |
| 10:54AM |
0 |
RE: IAX, Asterisk, GSM,SPEEX and ILBC (fwd) |
| 10:48AM |
1 |
[OT] Virus propagation by asterisk user member. |
| 9:17AM |
5 |
SIP QUESTION |
| 9:06AM |
4 |
Analog lines |
| 7:41AM |
1 |
MWI question |
| 6:59AM |
1 |
How Do I disable faststart? |
| 6:08AM |
6 |
PrePaid and IVR |
| 3:11AM |
3 |
MusicOnHold |
| 2:08AM |
1 |
Brooktrout PRI-ISA48 card... info.. |
| 2:03AM |
1 |
Hiding and Changing Caller ID |
| |
| Monday August 18 2003 |
| Time | Replies | Subject |
|---|
| 4:45PM |
1 |
zaptel does not compile anymore |
| 4:36PM |
1 |
Asterisk Outbound Calling Warning: Unable To Forward Voice |
| 4:34PM |
0 |
screeching and MOH bleed on PRI |
| 2:51PM |
3 |
Call transfer ATA186 |
| 2:31PM |
3 |
Cisco 7940 7960 |
| 2:06PM |
3 |
Voicemail2 vs. Voicemail |
| 1:57PM |
2 |
Grandstream, SIP encryption |
| 12:44PM |
1 |
* and IAX as a gateway to video conferencing |
| 12:19PM |
8 |
PRI Question |
| 11:03AM |
1 |
chan_h323.c |
| 10:59AM |
1 |
Asterisk's configuration : Which signalling in France with an E1 ? |
| 9:48AM |
1 |
dumb x100p question |
| 7:35AM |
6 |
sound problem |
| 7:35AM |
0 |
Setting a minimum 'on-hook' interval? |
| 7:31AM |
3 |
403 FORBIDDEN Help! |
| 5:34AM |
3 |
Pops |
| 5:13AM |
1 |
Can I runAsterisk remotely from telnetsession? |
| 4:19AM |
0 |
(ATTENTION ) Quicknet Lan jack and phone jack |
| 3:31AM |
2 |
Cisco 7920 phone |
| 3:09AM |
3 |
MOH with SIP |
| 2:29AM |
0 |
Re: FW: Fax from 925 603 5512 (18 pages) |
| 2:27AM |
0 |
Re: FW: Fax from 925 603 5512 (18 pages) |
| |
| Sunday August 17 2003 |
| Time | Replies | Subject |
|---|
| 11:52PM |
0 |
Receptionist Console |
| 11:46PM |
1 |
Java SIP Client |
| 10:48PM |
3 |
Monitor application temporary hack |
| 5:25PM |
1 |
Asterix Newbie |
| 1:12PM |
2 |
no incoming packets & Sound: Recording overrun |
| 1:05PM |
2 |
Recomendations for an ISDN-PBX to use with asterisk |
| 12:28PM |
0 |
HP300 phone |
| 11:50AM |
1 |
BudgeTone NAT issues |
| 10:44AM |
5 |
LAN switches with PoE? PoE phones? |
| 10:18AM |
1 |
pre-newbie - some basic questions... |
| 9:29AM |
1 |
Configuring iptables to allow sip and dynamically allocate rtp ports |
| 8:58AM |
1 |
Has anyone got sip/IAX working behind a firewall? |
| 6:32AM |
2 |
chan_capi compile errors with latest CVS |
| 5:56AM |
1 |
Chan_h323 one way audio |
| 5:22AM |
4 |
Grandstream Budgetone |
| 12:18AM |
1 |
call routing based on dnis |
| |
| Saturday August 16 2003 |
| Time | Replies | Subject |
|---|
| 6:49PM |
2 |
Voicemail cliping digits via sip |
| 6:45PM |
1 |
music on hold help |
| 6:44PM |
0 |
Great concept but a few issues unresolved |
| 2:13PM |
1 |
Voicemail2 patches |
| 7:17AM |
1 |
Questions regarding CDR's |
| 5:12AM |
1 |
Chan_h323.so native? |
| |
| Friday August 15 2003 |
| Time | Replies | Subject |
|---|
| 5:21PM |
0 |
Registring soft phones in Asterisk |
| 2:29PM |
3 |
FXO/FXS hotline |
| 10:21AM |
3 |
Can I runAsterisk remotely from telnet session? |
| 9:02AM |
0 |
Autodialer / bulk dialer application |
| 8:55AM |
1 |
DTMF SIP |
| 3:49AM |
1 |
Asterisk H323 Trunk |
| 1:48AM |
0 |
Unable to detect process 2 frames |
| |
| Thursday August 14 2003 |
| Time | Replies | Subject |
|---|
| 5:03PM |
1 |
ast_channel_alloc() losing pvt struct |
| 4:08PM |
1 |
*-openh323 faststart |
| 4:05PM |
0 |
*-openh323 & faststart |
| 2:42PM |
1 |
make: warning: Clock skew detected. Your build may be incomplete. |
| 12:21PM |
1 |
Asterisk SIP calls failing - not a proxy? What of RTP codec transcoding? |
| 11:21AM |
6 |
.:. .: .. .Stottering audio ?? |
| 11:17AM |
1 |
chan_capi in the US |
| 10:45AM |
0 |
Which version of MySQL are you running? |
| 10:29AM |
0 |
Twisted Idea |
| 9:38AM |
1 |
Re: The Almighty X-Lite DTMF Problem (patch tested) |
| 8:36AM |
1 |
Problem with latest cdr Makefile??? |
| 8:35AM |
0 |
PHP Web Interface helpers |
| 7:33AM |
2 |
Don't know how to calculate timelen |
| 6:52AM |
0 |
New Asterisk user. |
| 6:47AM |
0 |
Alogirthm Used for Extension Matching ? |
| 6:36AM |
1 |
Virtual extension as local modem |
| 3:24AM |
7 |
What is the highest quality codec I can use for recording voice messages? |
| 1:35AM |
2 |
'#' doesn't work for me |
| 1:28AM |
1 |
How can I know if a user is busy or not connected? |
| 12:33AM |
0 |
Extension and phone management bestpractices?? |
| |
| Wednesday August 13 2003 |
| Time | Replies | Subject |
|---|
| 11:03PM |
0 |
Conference Number + CDR |
| 10:58PM |
0 |
Fwd: FW: SIP NAT question |
| 4:07PM |
0 |
Voicemail group, adding to the vm when forwarding |
| 3:50PM |
1 |
Receiving iaxtel calls |
| 3:20PM |
1 |
"Double" transfer? |
| 1:06PM |
5 |
Can't compile cdr_mysql |
| 12:25PM |
1 |
SIP NAT question |
| 12:16PM |
2 |
CLASS feature syntax |
| 11:58AM |
0 |
Which GS IP product to buy?? |
| 11:58AM |
1 |
Park and out-going trunk calls. |
| 11:46AM |
1 |
I can't get a two way conversation going? |
| 10:10AM |
0 |
Fwd: Stable versions of Asterisk (Was: Re: Fair comparison (John Todd)) |
| 9:43AM |
3 |
h extension seems to wipe variables? |
| 8:55AM |
0 |
Asterisk and AT&T 964 phones... |
| 8:46AM |
4 |
FXO mode |
| 7:56AM |
1 |
FWD SIP phone format=2, FWD call format=4, why? |
| 7:56AM |
1 |
Mixing audio from Music on Hold and IVR |
| 4:57AM |
1 |
Extension and phone management best practices?? |
| 3:27AM |
1 |
How do i configure so an incoming call triggers an http request? |
| 2:39AM |
2 |
reload |
| |
| Tuesday August 12 2003 |
| Time | Replies | Subject |
|---|
| 8:39PM |
0 |
Stable versions of Asterisk (Was: Re: Fair comparison (John Todd)) |
| 8:06PM |
1 |
Conference + E100P + H323 |
| 3:15PM |
1 |
usrobotics modem and pstn |
| 2:22PM |
4 |
X100P Ringing/Answering |
| 2:09PM |
0 |
RE: Asterisk-Users digest, Vol 1 #1033 - 7 msgs |
| 1:49PM |
1 |
New-ish list of hardware phone vendors |
| 1:32PM |
3 |
Weird DTMF issue |
| 12:47PM |
1 |
URI for dialing |
| 11:24AM |
1 |
Using Asterisk with FWD through NAT |
| 10:06AM |
0 |
Upgrading Queue App |
| 10:05AM |
1 |
Working with FWD, IPTel, SIPPhone? |
| 9:44AM |
1 |
Codec? |
| 9:38AM |
12 |
IP phone recommendation |
| 9:34AM |
0 |
Future Grandstream codecs?? |
| 7:43AM |
0 |
Xten-Lite and Asterisk. |
| 7:23AM |
0 |
Fw: Fax Handled |
| 7:22AM |
1 |
new on E100P |
| 6:57AM |
1 |
New gastman clone + what else? |
| 6:05AM |
4 |
Sip and One Way Audio |
| 5:09AM |
6 |
OT: Grandstream power supplies.. |
| 3:57AM |
2 |
problem with Wildcard 100XP and hangup signal |
| 3:37AM |
3 |
Fair comparison |
| 3:35AM |
1 |
Open G.729A codec |
| 3:29AM |
1 |
Malicious Call Trace |
| 12:25AM |
2 |
How to Asterisk |
| 12:12AM |
0 |
CVS version build error |
| |
| Monday August 11 2003 |
| Time | Replies | Subject |
|---|
| 5:35PM |
0 |
Ring when leaveing queue? |
| 10:59AM |
1 |
zaptel sync |
| 9:19AM |
3 |
Ring while on phone |
| 9:03AM |
1 |
avm fritz pci |
| 8:40AM |
1 |
ANI/DNIS call routing |
| 7:01AM |
0 |
InternetPhoneWiazard |
| 5:34AM |
0 |
chan_capi ptp mode |
| 12:03AM |
0 |
Faking Ring tone |
| |
| Sunday August 10 2003 |
| Time | Replies | Subject |
|---|
| 11:11PM |
0 |
Outdial digits - non TDM trunk |
| 6:41PM |
0 |
"Out of area" displayed as caller-id |
| 4:03PM |
4 |
Windows Messenger |
| 3:12PM |
3 |
Registering SIP with FWD and ICONNECTHERE |
| 2:20PM |
0 |
Asterisk (g729) termination on CISCO |
| 9:59AM |
2 |
SNOM200 firmware roll back!! |
| 6:13AM |
3 |
Asterisk Newbie ... |
| |
| Saturday August 9 2003 |
| Time | Replies | Subject |
|---|
| 5:03PM |
3 |
Need help with installation of H323 chanel driver |
| 4:29PM |
2 |
Gatekeeper |
| 3:13PM |
0 |
dialogic D/41 ESC with asterisk |
| 2:23PM |
3 |
help please with single t1 configuration |
| 1:47PM |
2 |
H323 and SIP |
| 12:46PM |
0 |
Something very strange regarding callerid |
| 12:06PM |
5 |
app_queue, fewestcalls and leastrecent logic |
| 11:12AM |
1 |
callerid (Bell type) in Europe |
| 9:43AM |
0 |
Call Center RFP |
| 8:19AM |
0 |
This is how to set ATA186 for different standards of CallerID format |
| 7:06AM |
1 |
To Switch or not to Switch... that is the question.... |
| 6:43AM |
0 |
callerid, british and french type DECTs |
| 6:07AM |
1 |
Does Wildcard x100p support Caller ID outside the US? (fwd) |
| 5:54AM |
1 |
Chan_Capi questions?? |
| 2:55AM |
2 |
Asterisk as a stand alone voice mail server (fwd) |
| 2:42AM |
0 |
Digium & PCI-X |
| 1:19AM |
2 |
Multiple E1 configuration question |
| 12:46AM |
0 |
ATT: marrandy - Re: Grandstream Budgettone 102 |
| 12:19AM |
0 |
IAX protocol description |
| |
| Friday August 8 2003 |
| Time | Replies | Subject |
|---|
| 6:06PM |
1 |
UNIX command-line interaction with astdb |
| 2:50PM |
3 |
Killing runaway PBX |
| 2:04PM |
2 |
G.729 licensing -- an opinion |
| 1:34PM |
0 |
VoicemailMain2, inband digits detection, rcf2833 digits detection (rtp issue, I think) |
| 12:34PM |
3 |
queue / agent documentation |
| 12:26PM |
0 |
Workaround for BudgeTone "ringing in your ear?" |
| 12:09PM |
1 |
X-Lite - No sound + chan_sip issue |
| 11:37AM |
4 |
Voicemail2 - auto fill the dialing extension? |
| 11:25AM |
5 |
list proposal |
| 10:15AM |
0 |
re: Web GUI |
| 8:55AM |
3 |
segfaults with queue |
| 8:36AM |
1 |
g729 problems |
| 8:29AM |
2 |
Fax Handled |
| 7:28AM |
2 |
Re2: Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO] |
| 6:39AM |
0 |
dtmf detection from AS5350 over SIP |
| 6:35AM |
5 |
ip phones and intercom/paging |
| 6:22AM |
2 |
Call Waiting and Call Parking Together?? |
| 5:49AM |
1 |
Play Music On Hold just for a fixed period of time |
| 3:45AM |
1 |
CallerID, DECT phones and ATA |
| 1:46AM |
1 |
Snome-200 with Asterisk |
| 1:22AM |
0 |
ISDN BRI outgoing call instant hangup |
| |
| Thursday August 7 2003 |
| Time | Replies | Subject |
|---|
| 11:27PM |
2 |
Problem -ATA-711-723-Oh323-Asterisk |
| 11:14PM |
3 |
SIP Lines |
| 10:59PM |
1 |
h323 and cvs one way audio |
| 5:36PM |
3 |
Call routing question |
| 5:35PM |
1 |
TE401P driver warning |
| 3:24PM |
2 |
Newbie Issue |
| 3:10PM |
1 |
How to determine line signalling? |
| 2:51PM |
1 |
3xx SIP messages |
| 2:31PM |
1 |
Busy detect options |
| 1:31PM |
1 |
Error loading latest CVS |
| 12:30PM |
1 |
Sip Trunk config |
| 8:33AM |
0 |
list of sip phones? |
| 8:26AM |
1 |
MWI bug ? |
| 8:18AM |
1 |
Hardware for a Big PBX |
| 4:57AM |
0 |
Minimum system requirement for .... |
| 3:54AM |
1 |
Warning Messages |
| 2:06AM |
0 |
Asterisks integration with pre-existing PBXs |
| 1:53AM |
1 |
cdr_mysql uncompress |
| 12:56AM |
2 |
Leftover Budgettone issues |
| |
| Wednesday August 6 2003 |
| Time | Replies | Subject |
|---|
| 11:45PM |
2 |
Semi-newbie question "Softswitch" and Asterisk - Is there a difference? |
| 9:13PM |
1 |
Festival 1.4.3 |
| 8:06PM |
0 |
H323 + DTMF detection |
| 7:46PM |
1 |
Feedback for Asterisk Handbook? |
| 5:32PM |
2 |
FYI: G723.1 Licensing Prices |
| 5:08PM |
1 |
BRI newbie queries. |
| 3:27PM |
1 |
Unregister SIP connection? |
| 2:28PM |
8 |
CDR MySQL |
| 1:38PM |
4 |
New SIP Phone |
| 1:36PM |
0 |
Standard Analoge modem - can it be used? |
| 1:35PM |
1 |
Budgettone Newbie |
| 1:12PM |
10 |
AgentCallbackLogin |
| 11:13AM |
0 |
indications.conf settings for Belgium |
| 11:04AM |
1 |
Behind Firewalls, SonicWalls, etc.. |
| 10:42AM |
0 |
FAQs and Helpful URLs? |
| 10:26AM |
1 |
X100P CallerID issue solved for my PSTN connection |
| 9:55AM |
0 |
Bad sound quality with G729A on SNOMs |
| 8:55AM |
0 |
ISDN Examples |
| 8:28AM |
0 |
Push to Talk |
| 7:45AM |
0 |
Intermittant IAX Call Failures |
| 6:26AM |
0 |
Config files - examples |
| 6:08AM |
9 |
R2 support |
| 4:45AM |
3 |
X-Lite <-> Snom200 |
| 2:08AM |
1 |
X100P and Caller ID (again and again...) |
| 1:11AM |
2 |
iax.conf / Registration rejected |
| 1:02AM |
1 |
chan_oh323 + dtmf |
| |
| Tuesday August 5 2003 |
| Time | Replies | Subject |
|---|
| 11:23PM |
4 |
Windows IAX soft phone |
| 9:19PM |
1 |
So now I'm playing around with Queues.... |
| 8:40PM |
0 |
CVS troubles |
| 7:15PM |
0 |
WipeOut - gateway access with pin solution |
| 4:33PM |
1 |
T-shirt ideas |
| 2:05PM |
5 |
(no subject) |
| 1:55PM |
1 |
Someone used ADIT 600 Channel Bank. |
| 1:33PM |
2 |
Wierd Message |
| 1:27PM |
0 |
Asterisk and authentication |
| 9:36AM |
2 |
Call Monitor |
| 9:29AM |
3 |
Newbie just starting out with * |
| 6:42AM |
0 |
GSM file format |
| 6:41AM |
2 |
chan_capi: Hanging channels - again |
| 6:10AM |
1 |
Zhone Zplex |
| 5:53AM |
3 |
Does Wildcard x100p support BT Caller ID in UK? |
| 4:59AM |
2 |
Why are FXO so expensive? |
| 2:40AM |
0 |
usable/affordable usb phone? |
| 1:53AM |
4 |
SendDtmf |
| 12:45AM |
1 |
Zhone Zplex 10 units |
| |
| Monday August 4 2003 |
| Time | Replies | Subject |
|---|
| 8:40PM |
2 |
Channel banks, etc. |
| 4:56PM |
0 |
Transitioning from existing PBX |
| 4:54PM |
4 |
SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers |
| 3:39PM |
2 |
Fix for Redhat 9 zombie AGI processes |
| 3:12PM |
3 |
FW: Cisco 7960, SIP, NAT, Voicemal |
| 2:00PM |
0 |
Repost MS Messenger 4.7 docs? |
| 12:16PM |
6 |
bugs.digium.com |
| 11:42AM |
0 |
Bridged trunks stuck off hook. |
| 9:56AM |
3 |
Syntax for hiding caller ID but still passing ANI? |
| 9:05AM |
3 |
newbie question - devices |
| 9:01AM |
0 |
Some questions about a potential usage scena rio for asterisk |
| 8:26AM |
3 |
limiting out going calls to a maximum duration |
| 8:21AM |
1 |
Some questions about a potential usage scenario for asterisk |
| 5:51AM |
2 |
H323 CallerID |
| 5:31AM |
3 |
CDR |
| 5:03AM |
0 |
mem leak in logger.c? |
| 4:33AM |
0 |
Any pointers for setting up PRI for incoming and outgoing calls? |
| 4:11AM |
0 |
small fix in chan_mgcp.c |
| 2:37AM |
1 |
SIP clients not sending audio |
| 12:23AM |
14 |
Mysql CDR |
| |
| Sunday August 3 2003 |
| Time | Replies | Subject |
|---|
| 3:17PM |
0 |
g.729 licenses do not release when used in Voicemail |
| 1:11PM |
0 |
RTP / SIP routing issues |
| 1:10PM |
2 |
AGI accountcode. |
| 11:57AM |
1 |
Prepaid calling card |
| 9:56AM |
1 |
Fax Detection? |
| |
| Saturday August 2 2003 |
| Time | Replies | Subject |
|---|
| 7:43PM |
17 |
call waiting |
| 7:27PM |
1 |
SIP app_queue |
| 4:29PM |
0 |
Webalizer for CDR logs.... |
| 11:00AM |
1 |
Asterisk agi interface leaves zombie processes? |
| 8:33AM |
1 |
Patch - transfer with two rather than one # |
| 5:37AM |
1 |
GSM codec |
| 12:45AM |
1 |
Asterisk + SER |
| |
| Friday August 1 2003 |
| Time | Replies | Subject |
|---|
| 7:25PM |
2 |
Hangup after a Timeout |
| 4:13PM |
1 |
HELP!!!! Ringback oh323 |
| 3:56PM |
1 |
HELP!!!! |
| 3:46PM |
7 |
Using OH323 and Gatekeeper |
| 2:52PM |
1 |
ztdummy & usb-ohci? |
| 2:16PM |
0 |
Cisco AS5300 -- Not hearing anything |
| 1:58PM |
1 |
Musiconhold interrupted sound |
| 1:24PM |
5 |
Seting up TDM40B |
| 12:50PM |
0 |
Overlap on PRI to PSTN |
| 11:50AM |
0 |
pcphoneline producs |
| 11:32AM |
1 |
Monitor app |
| 9:09AM |
1 |
phone rings while already on a call |
| 8:19AM |
0 |
Background messages while waiting for pick-up |
| 8:04AM |
0 |
Problem with SIP Native Bridging and UPnP |
| 7:21AM |
0 |
DTMF onto the reall world |
| 7:17AM |
2 |
Asterisk community input: FreeTDS (cdr_tds.c) or unixODBC (cdr_unixodbc.c) ? |
| 6:08AM |
1 |
segmentation fault with asterisk and OH323 |
| 5:37AM |
1 |
Asterisk SIP bug with Net2Phone |
| 5:25AM |
1 |
Extension handling. |
| 4:28AM |
2 |
DTMF modes and external IVR systems over ISDN |
| 2:36AM |
1 |
memory leak? |
| 1:03AM |
1 |
SIP with an iptables fiewall |