Sunday February 29 2004 |
Time | Replies | Subject |
10:22PM |
1 |
Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk |
9:00PM |
1 |
AGI/php help needed with variables |
8:21PM |
0 |
RE: Problem connecting to ASkterisk Server |
8:11PM |
0 |
can * support ANI |
7:40PM |
0 |
Background and multiple digits |
7:28PM |
1 |
Asterisk rpm packages |
7:13PM |
2 |
freeBSD zaptel driver |
6:54PM |
0 |
Testing ENUM |
6:46PM |
1 |
Hangup to CDR recording timing |
2:15PM |
3 |
vmail.cgi -> .php? |
1:25PM |
1 |
Unable to specify channel 1: No such device or address |
12:29PM |
8 |
Brazilian Protocol |
10:22AM |
0 |
TDM400 problems |
9:49AM |
1 |
Re: Outgoing parallelism |
7:22AM |
1 |
Dialing out after caller leaves message |
6:18AM |
1 |
outgoing spool parallelism |
|
Saturday February 28 2004 |
Time | Replies | Subject |
9:10PM |
1 |
Load average ... |
8:26PM |
3 |
Asterisk on Feebsd , pls. HELP ! |
7:54PM |
2 |
If one extension is busy... |
7:40PM |
1 |
A working number at enum.fierymoon.com? |
6:39PM |
8 |
Hotel wake-up |
5:15PM |
2 |
PCphoneline FXO to FXS box?? |
5:04PM |
0 |
zaphfc bri: crackling sound |
4:22PM |
2 |
iaxComm updates at sourceforge |
3:31PM |
0 |
Help needed setting up H323 gateway. |
2:53PM |
1 |
New to T-1/Channel Bank hardware -- help? |
2:34PM |
1 |
zaphfc bri with overlap sending/receiving |
7:22AM |
0 |
How to compile bri-stuff.0.0.2.rc12 |
7:17AM |
1 |
sip:user@domain.tld |
6:57AM |
0 |
Galaxy Voice - Good or Bad? |
5:46AM |
1 |
OTish: Firefly Crashing with * |
5:45AM |
2 |
Asterisk PABX switch |
4:40AM |
0 |
Cisco 7960 sip v6.2 is out |
1:41AM |
1 |
Iconnect behind NAT |
|
Friday February 27 2004 |
Time | Replies | Subject |
10:24PM |
6 |
Video Conference |
9:47PM |
1 |
CISCO ATA 188 |
8:02PM |
2 |
DTMF Issues with SJPHONE |
7:10PM |
0 |
H323 SETUP ON ASTERISK?? |
5:35PM |
1 |
cvs update and new x100p cards broke menu playback |
5:26PM |
1 |
outdial broadcast |
3:51PM |
0 |
Mediatrix 1204 FXO GW ring cadence question |
2:39PM |
2 |
retrieve_sip_conf_from_mysql.pl data format |
2:28PM |
0 |
Fujitsu 9600 |
2:02PM |
0 |
Snom 200 Map key lights problem. |
1:37PM |
0 |
Asterisk as proxy? |
12:26PM |
0 |
Voicemail cutting off messages on SIP |
12:11PM |
0 |
Setting up with an Eicon DIVA PCI card? |
11:54AM |
0 |
budgetones + G726 |
9:51AM |
2 |
FXO Gateway of choice is? |
9:46AM |
4 |
Anybody managed to call a phone through sipgate.de |
9:42AM |
0 |
WTS (20) ATA-186, Various Cisco IP Phones VOIP Gear and P/S |
9:42AM |
1 |
wisip firmware, updates, features?? |
9:32AM |
2 |
Core dump crash |
9:24AM |
0 |
Routing NOTIFY SIP messages |
8:54AM |
0 |
Agent Queuing on multiple machines |
8:52AM |
2 |
Lucent Definity CallerID {Scanned} |
8:52AM |
3 |
USB Phones |
8:52AM |
1 |
Remote retrieval of voicemail, a question |
8:00AM |
0 |
IAX Phone Update - Slight Change |
7:26AM |
1 |
Best VOIP Analog adapter ??? |
7:15AM |
0 |
IAX Phone Bug Fix |
4:30AM |
0 |
Request for enhancement - IP dependent ports |
3:27AM |
2 |
Failed to start asterisk |
3:26AM |
0 |
Re: Asterisk-Dev digest, Vol 1 #507 - 8 msgs |
3:25AM |
2 |
chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy |
3:24AM |
0 |
chan_sip support for SIP:Remote-Party-ID, sp ecificallyCLID priva cy |
3:03AM |
1 |
RE: Message waiting light not coming on |
2:53AM |
2 |
Problem connecting to Asterisk Server |
|
Thursday February 26 2004 |
Time | Replies | Subject |
11:58PM |
3 |
registration |
11:45PM |
0 |
New Insatll of * |
9:25PM |
1 |
Does Digium TDM400P + X100P make a switchboard? |
8:41PM |
3 |
exit |
7:24PM |
2 |
Big Install examples please |
7:08PM |
0 |
(OT) HOWTO: 802.3af POE w/ 79xx |
6:44PM |
3 |
MWI false light activity - msg0000.txt |
6:29PM |
1 |
Delta Three/iConnectHere Outgoing Caller ID? |
5:58PM |
2 |
GS Budgetone 101 canot receive calls |
5:37PM |
1 |
Lucent Definity CallerID |
3:46PM |
0 |
chan_h323 chan_oh323 |
3:13PM |
0 |
A newbie list question |
2:52PM |
1 |
record application in extensions.conf -- how to stop recording? |
2:49PM |
1 |
Off topic question |
2:12PM |
2 |
chan_sip support for SIP:Remote-Party-ID, specificallyCLID priva cy |
1:11PM |
1 |
Grandstream -> firefly call translator problem |
1:08PM |
2 |
DTA-310 Outbound Dialing |
11:18AM |
1 |
callerid will not be set |
10:28AM |
0 |
relaxdtmf - duplicatedigits |
9:53AM |
0 |
chan_capi 0.3.1 segfault backtrace |
9:28AM |
1 |
Can You Specify Codec Per Extension? |
9:08AM |
1 |
chan_sip support for SIP:Remote-Party-ID, specifically CLID priva cy |
9:07AM |
0 |
Any schedule for Digium's TDM400 FXO modules, or for IAXy? |
8:52AM |
0 |
C7-Hardware |
8:09AM |
5 |
Asterisk Venture |
8:04AM |
3 |
Simple Front Gate Intercom |
6:52AM |
0 |
Distinctive ring on 2 channels? |
6:16AM |
0 |
Video Recording |
6:04AM |
1 |
Connecting an ISDN DECT phone base |
3:49AM |
0 |
RE:Poor Voicemail / Ivr announcement quality |
3:07AM |
1 |
SIP Extrange Problem |
|
Wednesday February 25 2004 |
Time | Replies | Subject |
11:42PM |
2 |
sip router gateway |
11:12PM |
0 |
cannot configure voicemail with mysql |
8:36PM |
1 |
how to route based on DNIS |
7:18PM |
0 |
Shopping for ISDN card |
7:01PM |
0 |
Macro Forward Calls |
6:55PM |
1 |
Newbie Qu. |
6:18PM |
2 |
Patching Asterisk for OpenH323 ASN.1 Vulnera bilities |
6:07PM |
1 |
7960 x SCCP/Skinny (off-topic) |
3:14PM |
14 |
Grandstream transfer into outer space |
3:01PM |
1 |
Patching Asterisk for OpenH323 ASN.1 Vulnerabilities |
12:48PM |
2 |
Calls always parked on 701 |
11:58AM |
0 |
Some ADSI script programming guide? |
11:13AM |
1 |
Web based UA |
10:44AM |
0 |
Unreliable Fax |
9:15AM |
2 |
Detection of extension |
6:26AM |
6 |
Comments on Voice Quality IP Hard Phones |
6:03AM |
1 |
Problem with SIP 407 |
5:48AM |
0 |
segfault with alsa |
4:35AM |
1 |
Need some information |
2:43AM |
1 |
cisco 7912 problem with chan_sccp |
12:01AM |
4 |
dial plan question |
|
Tuesday February 24 2004 |
Time | Replies | Subject |
11:40PM |
1 |
Re: 12SP registration |
10:55PM |
0 |
parking ? |
9:58PM |
2 |
cdr->dst incorrect? |
8:45PM |
0 |
Need Origination Number form Bahamas |
8:03PM |
1 |
T1 inbound dialplan |
7:56PM |
0 |
Avaya question |
7:32PM |
1 |
Transferring Incoming Calls Twice |
4:22PM |
1 |
"CLASS" codes or "VERTICAL SERVICE CODES" |
3:47PM |
2 |
Can't compile Zaptel drivers |
3:26PM |
1 |
painful first steps with * and SIP phones |
3:23PM |
4 |
Incoming context based on ISDN MSN |
3:08PM |
2 |
VoIP Reseller Programs |
1:57PM |
1 |
Vegastream 50 BRI |
1:53PM |
0 |
InfoElement in chan_capi |
1:31PM |
4 |
Simulating the "lighted line in use" type of phone |
12:40PM |
0 |
Echo on Diva Server 2M with melware CAPI |
11:00AM |
4 |
DSL (DMT) goes down when X100 plugged in |
10:34AM |
0 |
FCC forces 911 services |
10:13AM |
0 |
RE: qview.pl |
9:56AM |
1 |
SIP Re-Invites & Timeout |
9:27AM |
0 |
CISCO 7912 SIP Problem |
9:25AM |
1 |
Cisco 7940/60 Intercom |
8:46AM |
4 |
Anyone working with NUFONE? |
8:18AM |
0 |
Re: IAX Voicepulse - no DTMF response |
8:17AM |
5 |
calls dropped with grandstream |
7:00AM |
0 |
system speed dial |
6:42AM |
1 |
RFC 2833 / Timestamp |
5:51AM |
2 |
Changes in capi.conf |
5:08AM |
4 |
Understanding AgentCallbackLogin |
4:51AM |
0 |
more codec negotiation problems |
4:10AM |
0 |
Flash application on H323 |
12:58AM |
1 |
codec translation |
|
Monday February 23 2004 |
Time | Replies | Subject |
10:26PM |
0 |
Job Opening |
9:42PM |
1 |
Asterisk and Faxing |
9:26PM |
2 |
line status |
6:20PM |
0 |
Asterisk Hardware Solution |
5:56PM |
0 |
Cisco Kit for Grabs |
4:07PM |
2 |
cdr_addon_mysql problem linking |
4:03PM |
3 |
Nested include statements in extensions.conf? |
3:59PM |
0 |
*8# and zaphfc in NT-mode |
3:47PM |
0 |
Asterisk and Multicast |
3:03PM |
2 |
A missing argument |
2:40PM |
1 |
SIP Codec selection order |
2:39PM |
1 |
12SP |
2:05PM |
0 |
calling between two zap points with zaphfc |
1:51PM |
1 |
ztmonitor and the x101p |
1:19PM |
1 |
VM: Multilanguage and digits |
1:10PM |
5 |
Unable to create channem of type 'Zap' |
12:51PM |
2 |
SIP over NAT |
12:29PM |
0 |
DevLite problem with ztcfg |
12:25PM |
1 |
Call Groups and outgoing line selection |
12:09PM |
1 |
Queue Modified ACD for Asterisk 0.7.2 |
12:03PM |
4 |
Codec Order / Preference |
11:57AM |
3 |
Pickup |
11:53AM |
1 |
Confusion with IAX PBX-PBX |
11:37AM |
3 |
Dual Xeon |
11:28AM |
1 |
SPA 2000 ringing |
10:56AM |
1 |
Minimum voice mail message limit? |
10:05AM |
0 |
Attended Transfer Question |
9:15AM |
1 |
Thread-safe applications |
8:25AM |
3 |
An example config for using a Wildcard X100P and a SIP phone? |
8:19AM |
3 |
Processor load spikes |
7:45AM |
2 |
Re: Asterisk-Users digest, Vol 1 #2879 - 10 msgs |
3:25AM |
0 |
Re: How to best debug SIP registration failure (Solved) |
1:41AM |
2 |
About Grandstream ATA-286 and ring voltage |
1:31AM |
1 |
SIP overlap (early dial) 484 response |
1:05AM |
3 |
Hyuandi pstn handsets |
|
Sunday February 22 2004 |
Time | Replies | Subject |
8:23PM |
1 |
IAX2 Call menu handling problem with Norstar |
5:44PM |
1 |
Sip Register Fail - NAT |
3:38PM |
2 |
Cisco AS5350 Gateway Intergration |
1:47PM |
2 |
OT: SNOM and TAPI |
12:31PM |
6 |
How to best debug SIP registration failure |
10:21AM |
5 |
2 questions about ISDN BRI |
9:46AM |
0 |
What is the best way to debug the DTMF tones on a Zap interface |
9:43AM |
0 |
app_directory.c |
8:04AM |
1 |
X100P and DTMF sending |
6:46AM |
0 |
SEGFAULT (capi amd hfc-s NT) |
6:16AM |
3 |
"multicasting" conference calls |
3:46AM |
2 |
oh323 codec negotiation |
1:04AM |
1 |
Pingtel Opensource PBX Announcement |
|
Saturday February 21 2004 |
Time | Replies | Subject |
11:19PM |
1 |
7960 - multiple lines - sip.conf |
9:35PM |
0 |
iaxtel to asterisk config help. |
8:16PM |
3 |
Voicemail Follow-up |
6:44PM |
2 |
SIP extension "busy" when not available ?? |
5:09PM |
4 |
Using Festival inside an agi |
11:33AM |
4 |
New Wiki page: Dimensioning an Asterisk system |
10:37AM |
2 |
"Call did not go through" |
9:43AM |
0 |
Chan_skinny |
9:32AM |
3 |
Using * for large VoIP implementation |
8:01AM |
1 |
Re: System call forked - more stuff |
7:08AM |
2 |
System called seems forked up |
6:07AM |
0 |
Subject: Grandstream / SIP <-> IAX2 / Voicepulse |
5:44AM |
1 |
Connection Problem - GrandStream |
5:35AM |
2 |
Simple Call Center Setup? |
5:25AM |
1 |
switch ivr menu on/off remotely? |
5:04AM |
1 |
Voicepulse Connection |
1:28AM |
0 |
HFC-Cologne TE mode, non-I4L mode. |
12:27AM |
0 |
T100P & T1 |
|
Friday February 20 2004 |
Time | Replies | Subject |
11:32PM |
8 |
Agents / ackcall |
11:09PM |
4 |
Call Redirection |
9:13PM |
0 |
Termination in Phoenix |
9:06PM |
0 |
Voice mail sound distortion has everyone laughing |
8:16PM |
1 |
TE410P PRI T1 problem |
4:43PM |
2 |
Not Woodpeckers |
2:33PM |
0 |
Problem playing the first voice mail prompt |
1:50PM |
0 |
Re: System call succeed, asterisk sees failure |
1:41PM |
0 |
Setup ? |
12:48PM |
0 |
Connecting Nortel DMS100 to Asterisk ? |
12:44PM |
0 |
T1 Ground Start signalling support ? |
12:38PM |
1 |
multiple lines on 7960's |
12:27PM |
0 |
Disabling echo cancellation on fax ? |
11:54AM |
0 |
POTS Distinctive Ring |
11:42AM |
0 |
Firefly and IXATEL |
10:48AM |
1 |
System cmd usage |
9:51AM |
0 |
Snom 100 + H.323 |
9:22AM |
1 |
Skinny and SIP |
7:40AM |
0 |
Zaptel BRI and HFC-S cards in NT-Mode - Dialout problem |
6:49AM |
0 |
SwissVoice IP10S can't take back call |
6:42AM |
1 |
RE: codec negotiation prob solved |
4:13AM |
1 |
voicemail not working with mysql!!!! |
3:39AM |
0 |
g729, g723.1 codec translation costs |
1:58AM |
0 |
tapi for asterisk* |
1:20AM |
5 |
Are IAX2 providers ready for prime time? |
12:24AM |
2 |
INFO/DTMF retransmissions in * not absorbed? |
|
Thursday February 19 2004 |
Time | Replies | Subject |
8:23PM |
0 |
Minor update of Firefly |
6:30PM |
0 |
codec negotiation prob solved? |
5:45PM |
0 |
app_sql_postgres doesn't clean up |
5:19PM |
10 |
IAX Connection - Voicepulse |
4:00PM |
6 |
Woodpeckers |
3:31PM |
0 |
WARNING[1142106560] |
3:20PM |
1 |
Configuring Pingtel Xpressa |
3:15PM |
0 |
moh badness: mpg123 0.59s |
3:12PM |
1 |
* dropps outbound calls over PSTN |
3:07PM |
0 |
Language setting |
3:01PM |
0 |
Re: Zombies got me! - Fixed! |
2:53PM |
1 |
International PSTN dialing |
1:07PM |
1 |
Re: Zombies got me! - Fixed! |
1:06PM |
2 |
Make a phone dial remotely? |
12:23PM |
1 |
SIP Behind NAT (sipgate.de) |
12:12PM |
0 |
Caller ID Oddity |
12:12PM |
1 |
mgcp endpoint question |
12:11PM |
1 |
faxing with asterisk |
11:51AM |
3 |
Bizarre ring |
11:17AM |
1 |
compiling gastman |
11:12AM |
1 |
Registering Polycom IP 500 with Asterisk [re vised] |
11:02AM |
0 |
Registering Polycom IP 500 with Asterisk [revised] |
10:52AM |
0 |
EAGI errors? |
10:15AM |
0 |
Enumlookup and Asterisk |
10:04AM |
0 |
FW: H-324M/SIP Gateway |
9:54AM |
1 |
dtmf recording record and playback |
9:39AM |
0 |
sip - t.38 through asterisk |
9:38AM |
0 |
Zombies got me! |
9:03AM |
0 |
Spanish newbie need help |
8:38AM |
0 |
SIP Stuff |
8:21AM |
3 |
Enhancements Coming To VoicePulse Connect! |
8:12AM |
0 |
H-324M/SIP Gateway |
8:01AM |
8 |
Cisco 7960 SIP image (off-topic) |
6:27AM |
1 |
Problem with call to IAX |
6:25AM |
0 |
check if que has members |
6:24AM |
5 |
Zaptel BRI and HFC-S cards in NT-Mode |
6:11AM |
1 |
Mac X-Lite and Asterisk |
4:38AM |
2 |
IVR (does not exist in any format)(No such file or directory) |
3:56AM |
0 |
pri error |
3:44AM |
0 |
Flash to PBX with X100P - how to? |
3:33AM |
0 |
audiocodes and asterisk interoperability |
3:26AM |
0 |
Help a Newbie to conf softphone |
3:11AM |
2 |
X100 => T100 Upgrade |
1:27AM |
1 |
4 port FXO |
12:54AM |
3 |
help a poor newbie out with SIP choppy one-way problem |
|
Wednesday February 18 2004 |
Time | Replies | Subject |
10:57PM |
2 |
vm to email ? |
7:43PM |
1 |
Gastman doesn't draw lines properly between resources ... |
7:28PM |
1 |
Pingtel SIPxchange IP PBX goes Open Source... |
7:00PM |
0 |
running script based on if a peer is up or down |
6:33PM |
0 |
VoicemailMain2 |
6:24PM |
0 |
Anyone use the Cisco 12SP+ phone w. asterisk? |
6:12PM |
0 |
Microsoft Portrait 2.2 |
4:36PM |
1 |
OT: Cisco 79XX operation |
4:10PM |
0 |
sip-to-sip hangup problems? |
3:28PM |
0 |
Re: Asterisk-Users digest, Vol 1 #2849 - 12 msgs |
3:24PM |
0 |
Re: Asterisk-Users digest, Vol 1 #2849 - 12 msgs |
3:22PM |
3 |
ADSI ports |
2:45PM |
2 |
IAX2 dosent work for DIAX |
2:31PM |
1 |
Call Pick on Cisco 7960's |
2:01PM |
2 |
Round-robin chan_zap groups... |
12:25PM |
0 |
Red Alarms and mysterious disconnects |
11:31AM |
2 |
softphone configs? |
11:07AM |
3 |
Budgetone phones from FWD |
10:55AM |
2 |
Where to download app_pgsql |
10:44AM |
2 |
Memory usage |
10:36AM |
0 |
* nat internet nat sip phone howto |
10:11AM |
1 |
agi scripting in perl - dealiing with unexpected disconnects gracefully / spurious DTMF |
10:09AM |
1 |
IAX2 queue member no auth found |
9:40AM |
0 |
Typo fixes |
8:29AM |
0 |
mysql_freinds |
6:15AM |
5 |
Callerid & AGI Thougts |
4:41AM |
2 |
Executing external script |
1:10AM |
2 |
Help setting up asterisk |
|
Tuesday February 17 2004 |
Time | Replies | Subject |
10:51PM |
3 |
pstn 800#'s |
10:00PM |
2 |
FXO gateways on Asterisk |
9:52PM |
0 |
How to Dial-out from USR Voice Modem |
9:10PM |
1 |
DIAX does not receive calls... |
8:16PM |
1 |
Specials on VOIP Gear |
7:06PM |
0 |
Background command, error |
7:03PM |
3 |
Howto apply a patch, diff file |
5:43PM |
1 |
Wanpipe cards? |
5:28PM |
1 |
SIP config documentation |
4:54PM |
0 |
Weird sdp output |
4:46PM |
1 |
Inbound IAX to SIP |
1:42PM |
0 |
Problems to register with SIP provider |
1:31PM |
2 |
Telemarketer handling |
1:21PM |
1 |
Marketing collateral, etc. |
12:22PM |
2 |
Re: Asterisk-Users digest, Vol 1 #2840 - 11 msgs |
12:12PM |
0 |
G729 and VoiceMail |
12:09PM |
1 |
Asterisk quietly segfaults |
12:02PM |
5 |
Dell 1750 server and Asteriks... |
11:52AM |
3 |
T1 Help |
11:03AM |
1 |
phpagi DIAL command not working |
8:57AM |
2 |
Record communication |
8:44AM |
2 |
x100p dropping incoming calls |
7:54AM |
7 |
max asterisk load |
7:45AM |
3 |
Double digits seen using Grandstream phones |
7:03AM |
1 |
Anyway to automate an AgentCallBackLogin |
6:54AM |
0 |
nasty segfault with previous strange Zaptel warnings |
6:17AM |
0 |
Incomming Distinctive ringing |
5:55AM |
0 |
Fw: voicemail extension - hangup |
5:48AM |
0 |
Queue timeout reason |
5:10AM |
1 |
extravagant behavior, nat problem ? |
4:41AM |
2 |
Mailbox full ? |
2:06AM |
2 |
IAXTEL and the registration traffic |
2:04AM |
2 |
PRI error or what? |
1:42AM |
1 |
How can you savage a failed call transfer |
1:33AM |
5 |
chan_capi problem |
12:59AM |
2 |
Analog Cordless Phone Recommendations |
12:55AM |
0 |
RFC: Some proposals for the list digests |
12:48AM |
0 |
(no subject) |
|
Monday February 16 2004 |
Time | Replies | Subject |
8:59PM |
1 |
Asterisk monitor with Daemontools |
7:07PM |
2 |
cannot find -lXext when building * ? |
6:51PM |
3 |
Room Monitor |
6:09PM |
0 |
2 Asterisk Boxes 2 Dev Kits |
5:30PM |
1 |
Asterisk for a call center? |
4:16PM |
0 |
What can cause a Red alarm? |
3:16PM |
5 |
Got my DID, getting an error. |
3:16PM |
0 |
Upgrading asterisk yields broken pipe |
2:57PM |
0 |
X100P analogue cards and impedance matching |
2:04PM |
0 |
FS: Adtran TotalAccess 850 Channel Bank,Router,4x4FXS |
12:51PM |
1 |
HFC-S cards? |
11:55AM |
0 |
Asterisk - Carrier Access Bank Ring through |
11:43AM |
0 |
Eicon Diva Server card, where to purchase ? |
10:07AM |
0 |
New to the list -> some (unsolved) questions |
10:05AM |
4 |
Speech between Grandstream phones sounds like talking under water |
9:12AM |
0 |
Agent / Queue help |
8:11AM |
0 |
Good source for moh files |
8:05AM |
0 |
IaxTel: Using IaxTel Numbers As Asterisk DIDs |
7:57AM |
0 |
SIP Messages (SIMPLE) |
7:13AM |
2 |
VOIP Carrier recommendations? |
7:10AM |
1 |
Cisco 30VIP Phones |
6:41AM |
0 |
Mailing list lag again |
6:39AM |
2 |
Analogical FXO vs. BRI dialing speed |
6:22AM |
0 |
voicemail extension - hangup |
5:15AM |
2 |
ZapRAS + RADIUS authentication |
4:51AM |
0 |
re: SIP 481 subscription does not exist with SJPhone |
4:50AM |
3 |
Zhone + call transfer |
3:57AM |
0 |
LDAP authentication |
3:01AM |
0 |
re: SIP 481 subscription does not exist with SJPhone |
2:20AM |
0 |
Re: Asterisk-Users digest, Vol 1 #2827 - 16 msgs |
2:10AM |
6 |
Need to interface to BRIs |
|
Sunday February 15 2004 |
Time | Replies | Subject |
8:59PM |
1 |
merlin legend / * as ld gw |
7:25PM |
1 |
Call File Troubles |
7:00PM |
2 |
Asterisk and Vonage -- no make that VoicePulse Connect :) |
2:39PM |
0 |
Official word from GalaxyVoice customer service |
2:00PM |
2 |
Asterisk and Vonage -- possible? Other DID providers with Atlanta presence? |
11:58AM |
0 |
Festival patch ? |
11:12AM |
0 |
Correct cvs checkout? |
9:17AM |
1 |
Pingtel Phones? |
8:34AM |
0 |
Looking for Incoming # for Area Code 713 (Houston, TX) |
5:38AM |
8 |
Wifi Phones |
1:56AM |
0 |
WTB: Grandstream Budgetone |
|
Saturday February 14 2004 |
Time | Replies | Subject |
11:37PM |
2 |
Music on Hold - Context |
10:07PM |
2 |
Get new PRI working |
10:03PM |
0 |
Xlite, GSM, Agent Logins and DTMF |
6:13PM |
2 |
TE405P and dual Athlon systems |
5:40PM |
0 |
Siemens Gigaset not ringing the cordless phjone for very long |
3:46PM |
3 |
Asterisk - oh323 - Cisco CallManager |
1:59PM |
3 |
running asterisk as non-root |
1:59PM |
0 |
Incoming SIP-calls and Festival |
1:41PM |
1 |
Festival: read text from external fil |
1:39PM |
0 |
Asterisk and "dial by email?" |
11:31AM |
0 |
with soekris? |
10:38AM |
0 |
Is there a MaxQueueTime for Queues ? |
10:10AM |
2 |
Kansas SIP or IAX Provider? |
5:40AM |
0 |
CallerID or Noise ? |
3:47AM |
0 |
FWD/Iaxtel/Asterisk codec use |
12:47AM |
5 |
Voip in the EU |
|
Friday February 13 2004 |
Time | Replies | Subject |
11:41PM |
0 |
Translator 'g729tolinb' |
8:09PM |
0 |
RE: Rhino channel bank and aastra PT390 phones |
6:33PM |
2 |
GSM codec with Cisco equipment |
5:56PM |
0 |
Problem with * - Nat - Internet - Nat - X-lite |
3:37PM |
2 |
HELP!!!! Having problems Starting Asterisk |
1:33PM |
1 |
GS BT-100 echo |
1:32PM |
1 |
Switch brands, speeds, etc. |
12:12PM |
3 |
RE: Rhino channel bank and aastra PT390 phones |
9:14AM |
2 |
chan_local and variables |
8:07AM |
2 |
Codecs compile error on yellowdog |
7:46AM |
1 |
Re-Invites and Studder. |
7:45AM |
0 |
Wierd Zap Channel Behavior |
6:54AM |
6 |
Digium connectivity issue? |
6:34AM |
1 |
Adtran 750 - what do I need |
4:17AM |
1 |
Spanish indications configurationÂș |
4:13AM |
5 |
Hide outgoing CallerId on Zap interface |
3:59AM |
0 |
festival in agi? |
1:47AM |
2 |
multiple context in sip.conf |
1:44AM |
2 |
channel bank - Adit 600 |
|
Thursday February 12 2004 |
Time | Replies | Subject |
10:52PM |
0 |
[OT] Looking for Manual: Clarent CPG 101 |
6:48PM |
5 |
X100P / Echo / ZTMONITOR CAN2,3, etc. |
6:15PM |
4 |
Direct mailbox transfer |
5:11PM |
4 |
x101p beeps/sceeching |
4:54PM |
3 |
Anybody going to the Spring VON converence [ OT] |
4:28PM |
1 |
Sip problem with IpDialog phone. |
4:23PM |
1 |
More external call control |
3:30PM |
2 |
Voicemail Password Digit Timeout |
3:29PM |
2 |
Anybody going to the Spring VON converence [OT] |
3:19PM |
0 |
Why does the DG104S keep sending? |
3:12PM |
1 |
AudioCodes MP-104, register |
12:56PM |
2 |
Jitter Buffer Configuration (typo in iax.conf) |
11:50AM |
0 |
Database items |
9:31AM |
1 |
festival voices |
8:53AM |
1 |
Playing GSM files(s) |
8:52AM |
2 |
billing question |
5:44AM |
0 |
Specify address with base=0xNNNNN |
4:59AM |
0 |
Mailing list search engine |
2:03AM |
1 |
setting up callback |
|
Wednesday February 11 2004 |
Time | Replies | Subject |
9:30PM |
2 |
New Zealand |
8:46PM |
0 |
Asterisk hangs up when a call comes in |
8:08PM |
0 |
"Integrated" T1 PRI (voice and data) |
7:27PM |
1 |
Force SIP Phones to Register |
4:04PM |
1 |
Asterisk and Wildcard T100P |
3:51PM |
1 |
Mediatrix 1204 sip g/w now working |
1:59PM |
0 |
Please Explain newchan->pvt->pvt |
1:52PM |
0 |
Asterisk Critical Mass: Thursday, Miami, 9:00 PM |
1:02PM |
1 |
Constant crashes with Asterisk 0.7.2 |
12:25PM |
1 |
speex with VoicePulse |
11:15AM |
1 |
asterisk-oh323 new update, v0.5.9 |
11:13AM |
1 |
T1 PRI CallerID |
11:03AM |
3 |
"Stuck" TE410P cards |
10:24AM |
6 |
TDM card loses Dial tone |
9:42AM |
3 |
Can't connect KPhone to asterisk |
9:34AM |
1 |
OT: Cisco 7940 Smartnet in the UK |
9:26AM |
0 |
Re: Asterisk<->GS and codec selection |
9:15AM |
1 |
unable to open ../voicemail/context/exten/msg0000 |
8:54AM |
5 |
Cisco ATA 186 |
8:31AM |
1 |
I need patch for musiconhold-multiful format |
6:02AM |
1 |
Calling from Iaxtel to FWD users always busy |
5:08AM |
3 |
Noise and scratches when there are two concurrent CAPI calls |
4:49AM |
0 |
Multiple switch staments |
3:21AM |
0 |
[DENICenum-l] Open Workshop on IP voice and associated convergent services] |
2:12AM |
1 |
Pls help for Musiconhold |
12:41AM |
1 |
Cisco 7960G ordering Question |
|
Tuesday February 10 2004 |
Time | Replies | Subject |
11:34PM |
1 |
IAX DTMF question |
11:23PM |
0 |
Call center integration - passing caller idinto an external app. |
10:22PM |
0 |
Iaxphone problem |
9:55PM |
1 |
Residential Plans for Asterisk Users |
9:45PM |
0 |
linux 2.6 |
9:26PM |
3 |
How much processing power is needed? |
9:07PM |
4 |
Loading module chan_capi.so failed! |
5:01PM |
3 |
Cisco 7960 - how to enable "messages" key |
3:24PM |
3 |
I finally did IT!!!! Internal dial tone |
3:07PM |
4 |
alert-info and Cisco 7960 phones (6.1) |
2:02PM |
1 |
Call center integration - passing caller id into an external app. |
1:36PM |
3 |
Wait command in auto attendant causes sched.c error |
1:14PM |
4 |
Termination - Cuba |
12:38PM |
2 |
TDM400 showing up as Tiger Jet |
12:31PM |
0 |
Error Logging (stops Randomly) |
12:06PM |
2 |
[Fwd: Having problems with RTP packets and H old] |
11:33AM |
0 |
[Fwd: Having problems with RTP packets and Hold] |
10:36AM |
0 |
RV: Strange Behaviour with DMZ |
9:19AM |
2 |
Callerid detection |
8:39AM |
0 |
two phones one host |
8:24AM |
0 |
Log entry - solved |
8:11AM |
2 |
Log entry |
7:53AM |
1 |
Sending DTMF out-of-band over IAX2 |
7:47AM |
0 |
Make outbound calls only from certain hosts |
6:25AM |
3 |
Spurious DTMF tones heard by the person being called |
6:03AM |
0 |
Having problems with RTP packets and Hold |
6:01AM |
2 |
NIC card failure [was: System freeze] |
5:52AM |
0 |
Basic Sip proxy setup question |
|
Monday February 9 2004 |
Time | Replies | Subject |
9:44PM |
0 |
Firefly 1.4 released |
7:31PM |
0 |
X100P + HP DL380 |
7:11PM |
3 |
Recording |
6:37PM |
0 |
NEC IP phone compatibility? |
5:26PM |
0 |
Infite RTP to wrong address from DG104S |
4:32PM |
0 |
Re: Asterisk-Users digest, Vol 1 #2785 - 6 msgs |
4:13PM |
2 |
Dual line Skinny |
4:05PM |
0 |
OS X -- Zaptel |
3:01PM |
1 |
Revisit the Cisco 7910 |
2:47PM |
5 |
Dialing 800 numbers with VOIP |
1:59PM |
1 |
X100P Cards have gone belly up? |
1:42PM |
3 |
Dial-out and Dial-in modem problems. |
1:25PM |
0 |
alternative to mpg123 musiconhold was [Sys tem freeze] |
1:15PM |
6 |
asterisk-grandstream call |
1:06PM |
2 |
SIP and DTA-310 was [ MGCP w/8x8 DTA-310 and as5300 pstn gateway] |
12:57PM |
3 |
Intercom system (not paging system) |
12:16PM |
4 |
Calling SIP |
11:24AM |
2 |
Unable to create vpb channel |
11:20AM |
7 |
Best OS for Asterisk |
11:02AM |
1 |
New Firmware for Grandstream Phones -Supports CFG by MAC address |
10:21AM |
4 |
Can asterisk make a call to a phone? |
9:52AM |
3 |
port number keeps changing |
9:45AM |
6 |
System freeze |
9:40AM |
4 |
how to password protect a meetme conference? |
9:23AM |
4 |
New Firmware for Grandstream Phones - Supports CFG by MAC address |
9:14AM |
1 |
incoming call to internal user |
8:36AM |
6 |
SIP phones with dual ethernet. |
8:35AM |
1 |
OS X -- More Specific |
7:46AM |
0 |
/var/spool/asterisk/outgoing issues |
6:28AM |
3 |
asterisk and fax over ip - concept |
5:29AM |
0 |
long delay before asterisk returns 486 busy with sip |
5:07AM |
1 |
asterisk-oh323, new version 0.5.8 |
4:03AM |
0 |
incoming DTMF on a SIP call |
4:00AM |
2 |
Help with Sip call problems - Whats not working? |
2:48AM |
0 |
Incomplete dialed number in CDR |
2:05AM |
0 |
RTP with ATA186 ? |
1:21AM |
0 |
DTMF over SIP to a Cisco gateway |
|
Sunday February 8 2004 |
Time | Replies | Subject |
11:12PM |
0 |
Newbie - help |
9:42PM |
1 |
Registering SJPhone with Asterisk |
8:55PM |
0 |
Call transfer from a queue |
8:25PM |
1 |
Motherboard and fxo suggestion. |
5:27PM |
1 |
OS X |
3:44PM |
2 |
dialout redunancy. |
3:07PM |
1 |
Speex == Screech using version 1.1.4 |
12:16PM |
3 |
Asterisk & Panasonic KXTD - Vonage |
11:52AM |
0 |
NanoBGA VIA Eden-N Processor |
9:23AM |
1 |
Asterisk pins CPU |
5:24AM |
0 |
FW: SNOM 200 silence suppression |
12:45AM |
1 |
PCMCIA |
|
Saturday February 7 2004 |
Time | Replies | Subject |
9:57PM |
1 |
ringing |
8:56PM |
3 |
Problems with ATA's locking up.. |
6:21PM |
3 |
Snom 200 MWI Button |
2:31PM |
2 |
central voicemail with remote offices |
2:31PM |
1 |
All incoming Zap calls getting picked up as FAX calls! |
1:42PM |
1 |
dial timeout not working |
9:05AM |
1 |
play_and_record: No audio available |
8:06AM |
6 |
s/asterisk mailinglists/asterisk forum/g ? |
5:26AM |
0 |
OpenBSD 3.4 Patching |
5:15AM |
3 |
Snom 100 Code Recommendation |
1:50AM |
1 |
IAX Softphone Errors |
|
Friday February 6 2004 |
Time | Replies | Subject |
10:23PM |
2 |
Caller-ID is being sent wrong. How to fix it? |
8:02PM |
0 |
RE:voiceglo sip config |
6:02PM |
0 |
Message Not Delivered |
4:47PM |
2 |
Asterisk under UML? |
4:19PM |
1 |
G.729, show command or log to confirm it's using the G.729 codec. |
3:54PM |
3 |
Interrupted musiconhold sound when silence supression is enabled |
3:26PM |
3 |
modprobe wcfxs |
2:29PM |
1 |
busy status |
2:11PM |
1 |
Asterisk on ebay. |
1:58PM |
3 |
is it possible to turn auto answer off and on in the dialplan? |
1:02PM |
1 |
iax2 jitter stats confusion |
12:46PM |
1 |
Annoying Beeps |
12:06PM |
1 |
Silencing Background App during touch tone detection |
9:37AM |
0 |
passing variables to a macro |
9:20AM |
4 |
Conference server |
8:53AM |
1 |
SIP - Native Bridge Error |
4:06AM |
1 |
DIAX 0.9.6b call reception |
2:25AM |
1 |
Trouble emailing Digium |
2:18AM |
0 |
Configuring buttons on a CISCO 12SP+ Ip Phone (skinny.conf) |
1:44AM |
0 |
ATA in MGCP sometimes dropping calls |
1:10AM |
0 |
Re: Asterisk-Users digest, Vol 1 #2749 - 7 msgs |
|
Thursday February 5 2004 |
Time | Replies | Subject |
11:48PM |
1 |
chan_sccp: incoming calls on multiple lines |
9:30PM |
2 |
Adding another X100P after X100P and TDM400P is already configured |
8:32PM |
9 |
zaptel on Debian |
8:31PM |
1 |
Sip transfers |
8:17PM |
3 |
Re: DISA |
7:47PM |
0 |
Current version of gastman precompiled binary |
7:14PM |
2 |
ISDN update |
7:00PM |
1 |
Re: DISA |
6:17PM |
0 |
Re: [Asterisk-Dev] DISA |
5:55PM |
0 |
AutoAttendent ON/OFF control by Attendent |
5:00PM |
0 |
RE: Apple OS-X |
4:16PM |
1 |
Asterisk Randomly Stopping |
3:57PM |
0 |
OT Asterisk Sales Questions (Not for Asterisk itself) |
3:49PM |
1 |
has Allison recorded "Do Not Disturb" |
3:42PM |
2 |
http://www.oneunified.net |
3:23PM |
6 |
Voiceglo questions |
2:59PM |
2 |
simple test setup |
2:45PM |
2 |
Fax with wildcards |
2:41PM |
4 |
question for oh323 users |
2:33PM |
2 |
Asterisk GUI Client - New verison 0.9 |
2:24PM |
0 |
sethdlc-new compile, does it? |
2:12PM |
1 |
X100P - Asterisk - Asterisk - X100P setup help |
12:19PM |
1 |
fwd settings |
12:11PM |
2 |
Vegastream 50 FXO with Asterisk |
11:01AM |
1 |
Release phone call |
10:18AM |
0 |
CallWaiting CallerID: Available on all channel types? |
9:14AM |
0 |
The Evil of type=friend explained, again ( wa s Re: Minor Registration Problem With Polycom Soun dpoin t IP 500) |
8:26AM |
2 |
(no subject) |
8:20AM |
2 |
Record conversation |
8:00AM |
0 |
compact fxo device |
7:03AM |
3 |
Asterisk as non root |
5:36AM |
1 |
Execute command in shell |
5:33AM |
1 |
Dialogic D300SC-E1 |
5:23AM |
2 |
Asterisk + oh323 docs ? |
4:10AM |
0 |
H323 calls via provider |
2:04AM |
2 |
Data call transfer |
|
Wednesday February 4 2004 |
Time | Replies | Subject |
9:51PM |
0 |
Music on hold inside of an agi script whileprocessing programs in background |
9:29PM |
2 |
help *** newbie |
8:59PM |
4 |
Music on hold inside of an agi script while processing programs in background |
4:48PM |
1 |
Audiocodes FXO - loop channels |
4:38PM |
1 |
progress on DTMF |
2:39PM |
0 |
Integrating with an existing PBX |
2:24PM |
0 |
Need knowledgeable comments, bug 981 and dual redirect |
1:41PM |
1 |
Asterisk 0.7.2 RPMS Updated |
12:45PM |
1 |
Possible Sip logic bug? |
12:36PM |
3 |
Adtran 750 Configuration |
12:10PM |
0 |
Audio code registration |
11:55AM |
0 |
Asuscom HiSax based ISDN BRI card - one way latency |
10:45AM |
5 |
Sip flow diagram? |
10:32AM |
1 |
New Search engine for the list - Final resting place |
10:27AM |
2 |
Interrupted musiconhold sound when silence suppression is enabled |
10:11AM |
0 |
Voicemail volume level? |
9:51AM |
0 |
7960 MGCP dialtone problems, part 2 [long] |
9:50AM |
1 |
7960 MGCP dialtone problems, part 1 [long] |
9:37AM |
1 |
Newbie Question. Is asterisk right for my scenario? |
8:49AM |
0 |
voicemail auth failure |
8:39AM |
1 |
ParkAndAnnounce - Get Parking Extension |
8:21AM |
3 |
Asterisk 0.7.2 |
8:11AM |
4 |
Whats wrong with dialplan? |
8:03AM |
2 |
Cepstral TTS Code |
7:19AM |
3 |
talking clock |
7:17AM |
9 |
Boards falling out... |
4:59AM |
3 |
Minor Registration Problem With Polycom Soun dpoint IP 500 |
3:48AM |
0 |
Newbie: Chan_capi, early b3 in Italy |
2:42AM |
1 |
Port bind |
2:18AM |
9 |
Code Hosting... |
1:57AM |
2 |
Do you Linux softphone.. |
1:34AM |
0 |
X100P and PSTN line Callwaiting |
12:58AM |
0 |
billing information from telecom |
12:26AM |
3 |
CALEA? |
|
Tuesday February 3 2004 |
Time | Replies | Subject |
11:40PM |
4 |
iax, trunking, etc. |
11:40PM |
1 |
Anyone used a Grandstream ATA286 with Asterisk |
11:15PM |
1 |
Cisco 7960 bug in 6.1 evident in Asterisk |
11:05PM |
0 |
Minor Registration Problem With Polycom Soundpoint IP 500 |
10:38PM |
1 |
VOIP Deployment Concerns |
8:21PM |
0 |
Cisco AC Power Cubes for Sale |
7:57PM |
4 |
diax softphone |
6:47PM |
2 |
IPKall->FWD->Asterisk |
6:17PM |
2 |
Pictures of new multiport FXO/FXS from digum |
5:19PM |
2 |
Detecting answer supervison from an AGI app |
4:26PM |
1 |
Re: Asterisk-Users digest, Vol 1 #2711 - 15 msgs |
4:16PM |
0 |
(no subject) |
3:18PM |
1 |
sipphone dialing out problem |
3:09PM |
4 |
voip phones |
2:56PM |
3 |
[OT] Oldest Telephone |
2:17PM |
1 |
RE: Asterisk-Users digest, Vol 1 #2719 - 10 msgs |
1:17PM |
1 |
GS and NAT |
12:50PM |
1 |
Mediatrix sip fxo gateway workaround? |
12:48PM |
0 |
RedHat 9 & VSFTPD & Digium Hardware Oddoties |
12:42PM |
2 |
Qualify statement |
11:48AM |
0 |
Asterisk compatibility list |
11:14AM |
3 |
x100p card conflicts with DSL modem |
10:46AM |
4 |
Smallest server continued... |
10:46AM |
3 |
sementation fault with mpg123 |
9:32AM |
7 |
The Smallest Asterisk Server Ever? |
9:29AM |
0 |
Transfer of call from a call queue |
9:23AM |
1 |
Nortel and Asterisk interconnection |
9:20AM |
3 |
Cisco 7960 quick dial |
9:03AM |
0 |
Asterisk 0.7.1 RPMS Updated to Rel 4 |
8:56AM |
0 |
upgrade problems |
8:47AM |
0 |
kernel 2.4.x .... which one? |
8:24AM |
4 |
SIP debug logs |
8:17AM |
3 |
Using a Dial Statement with option m and t |
7:48AM |
1 |
Problems with chan_sip: random calls have no sound withouth any errors |
7:15AM |
3 |
Still looking for small fxo sip gateway |
6:51AM |
1 |
Mediatrix 1102 Auth |
5:59AM |
2 |
Dialling Hook Flash on Zaptel |
4:54AM |
2 |
Playing announcement to called user prior toConfirmation |
4:44AM |
2 |
busy tones |
3:36AM |
2 |
cisco 7912 voicemail/dnd issue |
|
Monday February 2 2004 |
Time | Replies | Subject |
11:20PM |
0 |
Mark's Asterisk Presentation at Linux-Kongress2003 |
10:16PM |
0 |
Newbie -- TE410P installation |
10:11PM |
1 |
Voicetronix Audio Problems when making two or more simultanoues calls |
9:24PM |
1 |
Details on TE410P Digium cards |
8:41PM |
1 |
Playing announcement to called user prior to Confirmation |
6:50PM |
1 |
extension mobility |
6:07PM |
0 |
Carrier Access Access Bank 1, incomming calls only echo problems, and Adit 600 |
4:34PM |
2 |
Large scale e.g. university |
3:26PM |
4 |
agent autologoff |
2:29PM |
1 |
New Zealand users/contractors |
12:44PM |
0 |
VoicePulse IAX2 lag |
12:31PM |
0 |
help with h.323 outgoing calls |
12:06PM |
4 |
Automated Dialing / Recording ? |
11:49AM |
7 |
cdr mysql problem |
11:22AM |
1 |
Problem sip registration |
10:30AM |
1 |
Fax Extension |
10:09AM |
2 |
compile error (still having problems) |
8:40AM |
6 |
Transfer |
8:25AM |
0 |
Re: how to dial and accept a call with only |
7:59AM |
3 |
Can audio streams go client to cleint with IAX? |
7:15AM |
0 |
Re: how to dial and accept a call with only |
6:57AM |
0 |
VOIP/IAX Termination |
6:21AM |
0 |
ISDN, CISCO and SCCP call forwarding |
5:59AM |
1 |
Norstar Integration with Asterisk via FXO or BRI ISDN |
5:19AM |
11 |
compile error |
3:41AM |
1 |
Channel Bank |
3:34AM |
0 |
Hide outgoing CallerID |
|
Sunday February 1 2004 |
Time | Replies | Subject |
10:23PM |
0 |
NewB: Cisco 7910 |
5:46PM |
1 |
Superbowl = Linux Shake up to the world.. |
4:19PM |
0 |
DNIS on X100P |
3:24PM |
2 |
setting up ---- newbie |
2:51PM |
1 |
can a variable be redefined within extensions.conf |
2:09PM |
2 |
Luxoncomm 3800 series FXO/FXS adapters? |
2:05PM |
2 |
How do I provide redundancy and reliability w/ Asterisk? |
1:10PM |
1 |
Configuring Firefly Network in * |
12:02PM |
0 |
PCI expansion slots. |
11:57AM |
1 |
how to dial and accept a call with only x100p card on Redhat linux 9.0? |
10:46AM |
1 |
Mediatrix 1204 SIP FXO 4-port gateway review |
8:31AM |
0 |
SMDI on * |
6:13AM |
1 |
short ringing |