Friday October 31 2003 |
Time | Replies | Subject |
11:17PM |
2 |
Asterisk: Reloaded |
9:53PM |
1 |
Polycom Soundpoint IP600 |
8:17PM |
5 |
Huge silence breaks between Cisco 7960 phone & X-Lite |
3:43PM |
1 |
msn messenger |
3:43PM |
3 |
Is iaxtel.com down for 700 #'s? |
3:34PM |
1 |
pcphoneline |
2:15PM |
0 |
Flaky SIP registration |
12:51PM |
0 |
one way sound with x-lite (sip) |
12:51PM |
2 |
problem DG-104S not call |
11:59AM |
2 |
HELP HELP HELP G729 |
11:21AM |
1 |
Echo on remote end when using NuFone |
10:48AM |
2 |
Which ADSI phones to buy? |
10:41AM |
0 |
Dial strings with ; characters |
10:19AM |
1 |
Some problems after an Asterisk update |
9:44AM |
0 |
FAQ on the Wiki |
9:03AM |
1 |
Problems with SIP |
8:48AM |
2 |
Cisco Support Contacts |
7:36AM |
0 |
Free LD/International calling testing, H323/SIP/MGCP and Asterisk |
7:34AM |
0 |
I can hear nothing if call from H323 to SIP. |
7:25AM |
2 |
asterisk and pingtel |
6:53AM |
2 |
asterisk FAQ |
6:36AM |
1 |
Problem in setup of asterisks |
6:28AM |
1 |
Making list of IAX providers |
5:29AM |
0 |
Voicemail storage question |
5:13AM |
2 |
MOH problem |
4:58AM |
0 |
One more QoS question for RH9 |
1:54AM |
1 |
Password in VoiceMail |
12:53AM |
5 |
VoiceMail Configuration |
12:17AM |
0 |
DACS Functionality in Zaptel |
|
Thursday October 30 2003 |
Time | Replies | Subject |
9:46PM |
0 |
Three way calling problems: 2 ea. X100P 1 ea TDM10p |
7:44PM |
0 |
X100P Driver loaded, but no IO on /dev/zap/1 |
6:55PM |
0 |
ADSI Pains |
6:31PM |
9 |
Absolute Minimum Installation Packages |
5:27PM |
0 |
extension exited non-zero... |
4:59PM |
1 |
Question about IAX/DID's... |
4:30PM |
2 |
Asterisk + Video |
1:29PM |
1 |
DTMF x-lite |
12:58PM |
4 |
H.323 and G729: Another sad tale |
12:52PM |
0 |
Newbie Question about MSI 240 Global Station |
12:28PM |
1 |
Compile problem with older ver. of CVS |
12:09PM |
4 |
SwissVoice MGCP IP10S |
11:37AM |
2 |
Fwd: Re: SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients |
10:00AM |
5 |
Newbie hardware question |
9:46AM |
3 |
two things |
9:42AM |
1 |
Newbie with 12sp+ |
9:18AM |
1 |
SIP NAT |
9:00AM |
2 |
critical problem |
8:14AM |
0 |
SIP/REGISTER problems! |
7:38AM |
6 |
Info on UK ISDN30e? |
7:01AM |
1 |
NAT type router database? |
5:15AM |
1 |
Out Of Band DTMF and SIP |
5:07AM |
1 |
G.729 pass thru Asterisk |
5:06AM |
0 |
Ringing .... |
4:52AM |
1 |
ZapRAS docs needed... |
4:33AM |
1 |
install problem |
4:33AM |
0 |
SIP error: Asked to transmit frame type 64 |
1:51AM |
0 |
Communication between 2 UA |
|
Wednesday October 29 2003 |
Time | Replies | Subject |
7:09PM |
1 |
AGI question or something |
6:53PM |
3 |
VMAIL.cgi |
5:52PM |
2 |
Polycom SoundPoint IP 500 |
5:39PM |
6 |
Nortel PowerTouch 350 |
4:28PM |
2 |
Campon feature |
4:17PM |
3 |
Sip bandwidth usage |
3:31PM |
1 |
Asterisk as a load generator |
3:26PM |
1 |
XTEN-Lite Bad sound! |
2:33PM |
1 |
Gnophone and Asterisk |
2:03PM |
1 |
SIP behind NAT problem |
1:51PM |
2 |
Listen to a Call |
1:39PM |
3 |
Channelbanks for use in europe (Sweden) |
12:38PM |
0 |
Re: Large installation [was: SS7 signalling/Softswitch] |
11:51AM |
1 |
Host unspecified ?? |
11:50AM |
0 |
Call pickup and SIP phones |
11:10AM |
2 |
Call transfering, conferencing |
10:48AM |
0 |
Forwarding all calls using SIP |
10:47AM |
0 |
iconnecthere Troubles |
10:23AM |
3 |
call waiting beep |
10:07AM |
1 |
Voicepulse and IAX |
9:07AM |
0 |
unsbscribe |
8:26AM |
0 |
extension dialing in the dial function for PRI ! |
8:07AM |
2 |
16xE1 solution based on * |
7:01AM |
3 |
FW: Voice/Data mixed routing over Digium E1/T1 Card |
4:49AM |
1 |
Some Basic Reading |
4:31AM |
1 |
probs with loading tor2 and wcusb |
2:52AM |
3 |
ATA186 configuration for fax application |
2:24AM |
1 |
Distinguish between voice and data call |
2:14AM |
3 |
Am I missing somthing? |
1:58AM |
6 |
SIP client |
|
Tuesday October 28 2003 |
Time | Replies | Subject |
10:13PM |
1 |
Software Fax Modem. Problem to apply patch to Makefile in apps directory |
9:18PM |
0 |
Dialing long-distance locally |
9:17PM |
4 |
Software FAX |
7:28PM |
2 |
looking for a place to buy SNOM or Cisco Phones (Cheap) |
7:02PM |
0 |
X100P/ATA186 not playing nicely... |
6:58PM |
2 |
Grandstreams can't call out with latest CVS |
6:39PM |
1 |
MOH Mixing tool |
6:25PM |
0 |
Where to find info on #, *67 *82 etc? |
6:03PM |
0 |
VoiceMail and Password |
5:29PM |
5 |
rxfax problem |
5:07PM |
0 |
NAT and v6 |
4:42PM |
2 |
Manager/Originate |
4:35PM |
5 |
RX gain TX gain |
4:31PM |
1 |
Already on the phone? |
3:12PM |
0 |
Asterisk <--> Cisco 2620 |
3:00PM |
1 |
SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients |
2:57PM |
1 |
# TDM 400P signal problem |
2:34PM |
1 |
TDM 400P signal problem |
2:07PM |
1 |
Making PrivacyManager smarter? |
1:49PM |
0 |
Scitel Brix QE |
12:57PM |
0 |
Consultants/Companies in Indianapolis? |
12:40PM |
3 |
Cisco or Snom ??? |
11:29AM |
0 |
cdr - call transfers |
9:38AM |
2 |
SIP Calls Don't Properly Connect (Continue Ringing) After CVS Update |
8:48AM |
0 |
Unable to find a path from G729A to ALAW, Unable to find a path from GSM to G729A |
7:53AM |
1 |
Feature request {with begging} sip debug <ip_address> |
7:50AM |
1 |
Software FAX Modem--One Last Request For Help |
7:46AM |
0 |
Ringing applecation is not working |
7:36AM |
0 |
RE: [Asterisk-Dev] Upcoming Major CVS Changes |
6:32AM |
2 |
Upcoming Major CVS Changes |
5:01AM |
2 |
Today's CVS |
3:43AM |
2 |
Another Segmentation Fault (Recording sound) |
3:12AM |
0 |
Oh323 segmentation fault in asterisk... |
1:56AM |
0 |
voicetronix openline4 echo problem |
|
Monday October 27 2003 |
Time | Replies | Subject |
9:01PM |
3 |
OT Vonage soft phone |
6:02PM |
2 |
PRI & Asterisk Redundancy/Fail-Over |
3:52PM |
0 |
Asterisk behind nat with hole, hardcoding solution |
3:42PM |
1 |
Asterisk + Sip phones on Nat |
3:24PM |
1 |
Fwd: Re: Asterisk on FreeBSD |
3:02PM |
2 |
BOTH UAs behind same FW/NAT |
2:57PM |
14 |
Answering Machine Detection |
2:34PM |
5 |
Voicetronix OpenLine4 |
1:29PM |
0 |
Asterisk on SPARC |
1:21PM |
0 |
Luxon Communications |
1:11PM |
0 |
Stuttered Dialtone for multiple extensions |
12:52PM |
4 |
Groups in * |
12:50PM |
3 |
passing digits for voicemail from sip gateway |
12:49PM |
1 |
SIP Softphone |
12:25PM |
0 |
Using Gastman |
11:55AM |
5 |
QoS What to do? |
10:20AM |
1 |
Is transcoding a bad thing? |
10:13AM |
0 |
Fwd: Re: Asterisk on FreeBSD |
10:12AM |
0 |
ZapBarge for SIP Channels |
10:10AM |
1 |
SIP -> H323 Seg fault. |
8:47AM |
0 |
G723 format compilation errors |
8:07AM |
2 |
Providing PRI to PBX |
7:21AM |
0 |
"Starting simple switch" |
6:25AM |
1 |
get IP Address from caller using oh323 |
6:13AM |
0 |
core dump in app_dial |
5:32AM |
2 |
Anyone got VM2 working with MySQL? |
4:39AM |
1 |
CVS File README.mysql concern.. |
4:31AM |
0 |
dtmf detection on modem-ISDN |
2:00AM |
2 |
SIP & IAX behind NAT |
|
Sunday October 26 2003 |
Time | Replies | Subject |
11:22PM |
0 |
SIPURA SPA-2000 now available |
10:58PM |
0 |
Sipura SPA-2000 anyone? |
9:03PM |
0 |
voicemail broken voice |
8:53PM |
1 |
After start Asterisk, error foung in the messages log file |
6:18PM |
0 |
Anyone have the Cisco 7920 working with Asterisk? |
4:35PM |
5 |
Extensions Problem |
1:48PM |
2 |
Lucent Partner extension to X100P |
1:24PM |
1 |
SIP auth |
12:30PM |
9 |
Asterisk on FreeBSD |
12:20PM |
0 |
PERL scripts: How to trigger? |
11:16AM |
2 |
help: mixing monitored in out files |
11:12AM |
0 |
Re: [Asterisk-Dev] important feature missing?! |
8:53AM |
0 |
alpha/numeric paging |
8:05AM |
1 |
NuFone International Calls |
1:56AM |
4 |
ReplayTV connecting through Asterisk box |
|
Saturday October 25 2003 |
Time | Replies | Subject |
6:48PM |
2 |
Confuson on iax calls (register or not?) |
2:53PM |
0 |
Asterisk External Resources Page |
1:25PM |
2 |
X100P stopped working |
12:31PM |
1 |
Voicemail.conf in MySQL is not functioning |
12:17PM |
1 |
Iconnecthere connect problem |
5:51AM |
6 |
cdr_mysql.so |
2:20AM |
2 |
Voicemail help |
12:59AM |
0 |
Music On Hold with Voicetronix |
|
Friday October 24 2003 |
Time | Replies | Subject |
7:17PM |
1 |
Looking for a 12 to 24 FXO Channel Bank in Colombia |
6:04PM |
0 |
Problem with CDR dst when executing Dial from 's' extension |
4:43PM |
4 |
Help with Dev Kit Lite |
3:44PM |
3 |
Broken Pipe |
1:47PM |
0 |
US source for compatible ISDN cards? |
1:42PM |
3 |
Compiling gastman under Win32 |
1:21PM |
0 |
Looking for supply/installation contractor |
11:56AM |
0 |
OT - Getting Cisco FXS ports to dial thru * |
11:23AM |
3 |
How to use the Cut() command to chop off an ending character |
11:09AM |
8 |
SS7 signaling/Softswitch |
10:33AM |
0 |
Fwd: Re: A software FAX modem |
10:30AM |
1 |
where to buy a SIP phone or adapter in SFBay area? |
10:18AM |
4 |
Nextone softswitch testing and Asterisk long distance |
9:30AM |
1 |
CTI interface(s) for Asterisk? [REALLY LONG MESSAGE - SORRY] |
9:12AM |
4 |
Context restrictions |
9:01AM |
0 |
Asterisk crash on module h323.so loading |
8:50AM |
0 |
anyone with a used analog card for sale or trade? |
8:46AM |
2 |
problems setting up E100P E1 germany |
7:11AM |
1 |
Asterisk behind NAT to SIP provider |
6:53AM |
1 |
mysql-vm-routines.h setup for MySQL |
6:46AM |
1 |
Questions about Zapateller and Privacy Manager |
6:35AM |
0 |
no dial out |
6:32AM |
2 |
asterisk config files |
5:27AM |
1 |
IAX CALLS ONCE MORE |
4:54AM |
6 |
AGI questions.. |
4:14AM |
1 |
Asterisk ??? |
2:40AM |
1 |
Anyone using sipcall.co.uk ? |
2:08AM |
1 |
2 IAX2 calls, bad audio |
|
Thursday October 23 2003 |
Time | Replies | Subject |
10:28PM |
2 |
New here... |
10:27PM |
0 |
music on hold` |
10:12PM |
1 |
New To Asterisk |
9:20PM |
0 |
FW: Voicetronix |
8:37PM |
2 |
CVS update |
7:54PM |
0 |
*69 Last call return function |
5:51PM |
0 |
SIP native bridge |
2:45PM |
0 |
WAS: Call pickup (*8) on SIP devices. Bug #116 |
1:55PM |
0 |
GotoIf Problems |
1:16PM |
6 |
Festival on RH9? |
12:47PM |
2 |
Go back - CVS |
12:10PM |
0 |
Asterisk passwords |
11:58AM |
0 |
DTMF relay with chan_skinny |
11:54AM |
1 |
Extended logic syntax |
11:34AM |
1 |
Number of TDMoE Channels? |
10:53AM |
0 |
ATM/AAL2 |
10:51AM |
3 |
IAXtel problems |
10:50AM |
2 |
IAX peers and NAT |
9:08AM |
0 |
Monitor not in Manager.PM? |
7:16AM |
0 |
VoiceMail delete |
6:52AM |
1 |
asterisk and zplex10b |
6:44AM |
1 |
agi script forcing asterisk reload |
6:27AM |
0 |
G729 help |
5:37AM |
1 |
How to write sound file with G723.1 codec or G729 codec |
5:21AM |
4 |
Gastman crashes on Win32 |
5:05AM |
2 |
wcfxs error |
4:55AM |
0 |
SIP Call Seq Error (SIP/2.0 481 Invalid CSeq Number) |
4:28AM |
2 |
native bridge |
4:26AM |
1 |
Problems with OH323/codecs |
4:21AM |
0 |
* + BRI + Debian = High Utilization |
3:22AM |
6 |
Problems with * and IAXTel/FWD |
12:39AM |
4 |
Call pickup (*8) on SIP devices. |
|
Wednesday October 22 2003 |
Time | Replies | Subject |
11:44PM |
1 |
Placing SIP calls to other SIP domains? |
11:39PM |
0 |
SV: Running Asterisk and NAT on the same box? |
10:40PM |
1 |
Tonight's CVS breaks Grandstream phone |
9:29PM |
4 |
what is the best codec for low bandwidth? for quality? |
8:38PM |
2 |
X100P Manually Answer |
7:53PM |
2 |
new codec for grandstreams |
7:29PM |
0 |
Last call return *69 function problem |
5:25PM |
0 |
Calls out of the PBX |
5:08PM |
0 |
MGCP error for Cisco 7750 FXO card |
2:13PM |
1 |
minimum hardware |
1:31PM |
29 |
Meetme |
1:06PM |
0 |
Asterisk Applications Consulting |
12:14PM |
2 |
Artificially Limiting IAX Calls |
11:57AM |
1 |
Inbound IAXTel failing? |
11:55AM |
2 |
Trouble with loading ztdummy on RH 7.3 |
11:41AM |
2 |
modprobe ztdummy failed |
10:16AM |
2 |
MOH problems |
9:25AM |
0 |
Fwd: Download Asterisk |
8:54AM |
2 |
RE: Grandstream Improvements |
8:22AM |
6 |
Running Asterisk and NAT on the same box? |
7:49AM |
0 |
SIP Carrier |
7:46AM |
0 |
iax over wireless |
7:29AM |
1 |
Download Asterisk |
5:24AM |
3 |
auto voice msg from text? |
4:31AM |
2 |
Slackware 9.1 Install Help |
3:22AM |
1 |
SIP and permit specified ip addresses |
3:16AM |
7 |
Is the X100P a WinModem? |
2:14AM |
1 |
IAX with multiple NIC |
1:13AM |
0 |
Different MGCP issues |
12:39AM |
0 |
capi incoming call |
12:31AM |
2 |
Useful patch in the bugtracker: streaming MOH |
12:16AM |
1 |
Encrypting SIP Phones |
|
Tuesday October 21 2003 |
Time | Replies | Subject |
6:04PM |
3 |
Asterisk with Gentoo |
5:12PM |
3 |
Quick summary of Grandstream survey results |
4:44PM |
0 |
Cisco 7960 MGCP and Skinny phone |
2:29PM |
2 |
Asterisk to SipPhone |
12:45PM |
9 |
Free g.729.1 implementation |
10:14AM |
1 |
SNOM 200 beta build + MOH |
9:53AM |
1 |
"Send to VoiceMail" button |
8:51AM |
0 |
zhone z-plex 10 |
8:44AM |
1 |
"Defragmenting" mailboxes |
8:01AM |
0 |
Iitter Buffer Settings |
8:01AM |
1 |
Grandsteam to support iLBC |
7:36AM |
0 |
CallerID Screening Prohibit |
7:17AM |
0 |
unclear about IAX |
5:51AM |
1 |
Hangup |
3:12AM |
0 |
(no subject) |
1:24AM |
1 |
VM2 and MySQL |
1:02AM |
0 |
active talker in a conference ? |
12:41AM |
0 |
Call pickup -> Change shortcode |
12:03AM |
0 |
Speed for meetme/asterisk? |
|
Monday October 20 2003 |
Time | Replies | Subject |
11:00PM |
1 |
Auto-dial from webpage |
9:03PM |
2 |
Message Indicator Light |
8:55PM |
1 |
Unsubscrip |
7:38PM |
26 |
Survey: Grandstream improvements......... |
7:35PM |
2 |
Setting a variable in extenstions.conf from the phone keypad. |
4:52PM |
1 |
Setvar SIP_CODEC |
4:22PM |
3 |
Authenticate Application Problems |
4:21PM |
3 |
Call Waiting on SIP phones |
3:17PM |
1 |
Conference with MOH or input from computer Mic. |
3:09PM |
1 |
Festival hangs up? |
1:56PM |
3 |
Music Onhold Configuration |
1:15PM |
6 |
Setting up an IAX2 trunk |
12:35PM |
1 |
Polycom IP-600 phone review |
9:39AM |
1 |
Need to partner with someone in Hampstead London on a deal |
9:19AM |
2 |
global vars |
9:13AM |
4 |
MOH different question |
8:30AM |
1 |
No detection of Line Busy |
8:18AM |
0 |
(3) Cisco NM-HDV-2T1-48 for Sale |
8:09AM |
1 |
Queues and Agents |
7:49AM |
4 |
SIP Nat Issue |
7:36AM |
16 |
A software FAX modem |
6:42AM |
4 |
how to escape # |
6:18AM |
1 |
Playing around MSNs |
2:42AM |
3 |
retrieve_?_from_mysql.pl files?? |
1:11AM |
0 |
SIP how to |
12:59AM |
1 |
mgcp transfer takeback with ata186 (logs with comments - long post) |
12:21AM |
1 |
Tested 7905G |
12:00AM |
2 |
Problems on making calls from one Gnophone to another through the local Asterisk Server |
|
Sunday October 19 2003 |
Time | Replies | Subject |
10:39PM |
0 |
Your comment on the following test bed setup ? |
8:39PM |
2 |
Modem |
6:39PM |
0 |
SIP soft phone |
6:36PM |
2 |
newb - want to create a Dialpad like system |
5:39PM |
1 |
Music on hold... |
11:47AM |
0 |
Patch testers needed |
9:57AM |
0 |
Flastman 0.0.1-pre-alpha |
9:55AM |
2 |
Cisco ATA Call Waiting |
9:45AM |
1 |
Asterisk/Freebsd network connections |
9:23AM |
2 |
The Start extension |
7:01AM |
0 |
X100P and Call Waiting Caller ID on the PSTN line |
5:16AM |
2 |
Project Completed [Files Attached] |
3:47AM |
0 |
Feedback request: AGI GET DATA change termination digits |
|
Saturday October 18 2003 |
Time | Replies | Subject |
10:48PM |
1 |
Some questions of heavy * deployment and stability. |
3:09PM |
2 |
Even Newer Patch to app_queue with skillbased strategy |
2:06PM |
0 |
DTA310 Config |
1:32PM |
1 |
Creating new voicemail accounts |
1:14PM |
0 |
Oh323 cisco callamanager |
1:02PM |
0 |
DID line with Adtran TA750 and T100p |
11:33AM |
1 |
latest cvs update |
10:40AM |
6 |
Outgoing call to IVR not being "answered" |
8:30AM |
0 |
We have added an Asterisk Forums to our existing web site. |
8:24AM |
2 |
my asterisk experience (long) |
3:23AM |
1 |
AGI script question |
1:44AM |
6 |
x-lite |
|
Friday October 17 2003 |
Time | Replies | Subject |
7:42PM |
4 |
chan_skinny & XML Files for 7920 |
4:54PM |
0 |
zaptel: [rx|tx]gain on E1/PRI/isdn audio quality problems |
3:36PM |
1 |
Asterisk in a rural project |
3:19PM |
0 |
X100P Echo - not resolved |
3:10PM |
1 |
(no subject) |
2:42PM |
2 |
AGI problem (crash) in RH9 |
2:37PM |
2 |
Polycom IP 600 phone |
2:00PM |
0 |
IAX with dynamic echo cancellation - what do you think? |
1:08PM |
0 |
Dial Multiple extension but require input not just off hook to bridge calls |
12:22PM |
1 |
Festival, Patch, Asterisk, etc. |
12:11PM |
1 |
System layout |
11:59AM |
4 |
Using channel banks |
11:10AM |
1 |
QoS On * |
10:51AM |
2 |
Beta testers for visual configuration tool f or asterisk |
10:23AM |
5 |
Beta testers for visual configuration tool for asterisk |
8:56AM |
4 |
Extension syntax specification - please help! |
7:35AM |
3 |
QoS |
6:50AM |
3 |
Switch statement taking over my local dialplan |
5:11AM |
1 |
Project Completed |
4:33AM |
0 |
Can't get IAX registration to work !#!$ |
3:24AM |
0 |
console sound problem with usb sound card |
|
Thursday October 16 2003 |
Time | Replies | Subject |
9:45PM |
1 |
Prob with Ringing multiple Channels |
7:24PM |
0 |
Some issues with call transfer. Please help. |
5:13PM |
0 |
AddQueueMember penalty patch |
3:28PM |
1 |
Grandstream phone :( |
3:02PM |
1 |
CallerID not passed to Sprint & Verizon Cellphones via XO PRI |
2:44PM |
6 |
MOH and VAD |
2:09PM |
3 |
Starting * with G729 licences |
12:08PM |
0 |
Directory App - excluding users... |
12:08PM |
0 |
Flash vs. 2-B channel transfer |
11:42AM |
1 |
OT - SIP Auto-Answer for Cisco 7940/7960!! |
11:31AM |
2 |
consultative transfer cisco |
10:03AM |
3 |
Adtran TA750 & T100P |
9:56AM |
0 |
Re-2: Some questions for chan_capi |
9:37AM |
2 |
Supervised transfers |
8:47AM |
2 |
Problems with TE410P and E1 line --> Unable to open D-channel 24 (No such device or address) |
8:39AM |
1 |
Calling a registered computer |
8:15AM |
0 |
Trouble with linking 2 Asterisks with IAX |
8:05AM |
0 |
sip registration failed |
7:43AM |
1 |
Weird IAX2 problem |
7:41AM |
1 |
VoIP Monitor |
7:41AM |
5 |
Voicemail File permissions |
7:36AM |
0 |
Zultys Zip 2 Registration / Disabling SIP Authorization |
7:23AM |
2 |
Cisco 7905G phones |
7:18AM |
1 |
[OT] E1 Cable pinout |
6:21AM |
7 |
I give up!! |
5:35AM |
3 |
ISDN BRI card |
4:40AM |
0 |
Caller-ID Spill |
2:48AM |
2 |
AGI problem (crash) |
1:51AM |
0 |
Use of the "hint" modifiers - examples, anyone? |
1:07AM |
0 |
Digium TDM card bad DTMF again |
12:53AM |
0 |
IAX Rejected Connect Attempt |
12:31AM |
0 |
french newbie with asterisk |
|
Wednesday October 15 2003 |
Time | Replies | Subject |
11:50PM |
0 |
AVM ISDN card now working but unable to place calls |
2:30PM |
2 |
Odd ringing conditions |
1:22PM |
1 |
Problem with SIP and DOS attacks... |
12:39PM |
4 |
indications.conf |
12:22PM |
0 |
RE: Manager Interface Needs a protocol amendment |
12:16PM |
5 |
newbie question: Meetme |
11:34AM |
2 |
IAX Clients not connecting |
11:01AM |
4 |
SIP Telephone Quality/Price |
10:50AM |
1 |
e100p in Australia |
10:23AM |
1 |
SER vs STUND with Asterisk.. |
9:51AM |
0 |
Basic questions |
8:56AM |
0 |
app_dial Flag |
8:49AM |
1 |
chan_skinny core dump |
8:49AM |
0 |
Manager Interface Needs a protocol |
8:24AM |
0 |
Real sip fax server |
7:50AM |
1 |
Sip call hang up |
7:46AM |
2 |
x110p -> tdm400p -> answering machine ISSUE |
6:39AM |
0 |
Some questions for chan_capi |
6:30AM |
1 |
No 'ringing' sound to outside callers |
6:17AM |
2 |
My Grandstream works, but my X-Lite doesn't:no sound after 5sec |
6:11AM |
0 |
SIP phone comparisons chart and article |
6:05AM |
2 |
skinny problem |
4:24AM |
0 |
cdr on call transfer |
4:15AM |
0 |
CF loop |
4:13AM |
0 |
Problems with MeetMe SIP / Mobile |
3:26AM |
1 |
Announced Call Transfer |
1:17AM |
2 |
Problem with T100P card in a new chassis |
12:52AM |
0 |
Asterisk running on OpenBSD 3.3 |
12:47AM |
0 |
frensh newbie with asterisk |
|
Tuesday October 14 2003 |
Time | Replies | Subject |
11:51PM |
4 |
Asterisk capacity |
8:37PM |
3 |
My Grandstream works, but my X-Lite doesn't: no sound after 5sec |
8:34PM |
2 |
VAD in Asterisk ? |
8:15PM |
1 |
DISA and ringing tone |
8:06PM |
2 |
Digium should develop and sell just Dummy card. For timing... |
6:57PM |
1 |
outbound caller ID problem on PRI |
6:31PM |
4 |
Wildcard TDM400P - FXO? |
6:10PM |
0 |
No Ringback on Iconnect or Nikotel |
5:58PM |
1 |
Newbie with questions |
4:22PM |
1 |
re: Restoring Cisco 7960 to defaults |
3:55PM |
0 |
Dial-out using Asterisk |
3:20PM |
0 |
pattern matching problem when dialing |
3:14PM |
1 |
Cisco hard IP phones and Skinny vs. SIP |
2:43PM |
1 |
Asterisk Certified Hardware label? |
2:28PM |
1 |
SIP Phone Tone |
2:16PM |
1 |
200-400ms latency |
2:12PM |
2 |
T100P to Adtran TA750 - No dialtone or ring |
2:06PM |
0 |
Turning a regular call into a conference? |
2:00PM |
1 |
IAXTEL - Problem Configuration. |
1:45PM |
5 |
Digium cards just for timing |
1:31PM |
1 |
Iaxtel and Voicepulse |
1:15PM |
6 |
WCFXO echo rexolved for me |
1:00PM |
0 |
list server Delays |
12:39PM |
0 |
AGI function calls |
12:05PM |
1 |
On an RH9 box, where does wcusb get loaded? |
11:37AM |
3 |
use of SIP SHOW CHANNELS question |
11:28AM |
3 |
Mitel 5055 phone |
11:24AM |
2 |
managers.conf Clarification Question |
10:18AM |
1 |
preplanning for a new home installation |
9:07AM |
0 |
Documentation Forbidden!! |
8:02AM |
3 |
H.323 - SIP gateway |
7:14AM |
0 |
QOS Question |
7:04AM |
2 |
RAS protocol |
4:43AM |
1 |
VoiceMail2 warning |
4:02AM |
2 |
Success story |
3:43AM |
3 |
*/SER/FW |
3:39AM |
0 |
Has something changed with AGI recently? |
3:14AM |
1 |
Outgoing CallerID |
2:58AM |
0 |
Meetme error in Mobile/SIP phones. |
1:43AM |
5 |
dialling out |
12:03AM |
1 |
no ring in ear |
|
Monday October 13 2003 |
Time | Replies | Subject |
11:59PM |
6 |
Asterisk Manager |
9:49PM |
1 |
Problem with SIP authentication |
8:40PM |
0 |
MGCP Gateway (Dlink DG104s) |
8:16PM |
0 |
H323 ID's |
8:11PM |
1 |
AGI solution to Grandstream BT102 call waiting problem |
4:57PM |
0 |
Call Parking and Paid Digium software modifi cations |
4:07PM |
1 |
out going calls |
2:44PM |
1 |
Call Parking and Paid Digium software modifications |
1:11PM |
1 |
ACD/IVR dialogs/SIP/client environment |
12:48PM |
7 |
PrePaid Application!!!!! |
12:04PM |
4 |
IAXTEL/ Dial problem |
12:00PM |
1 |
PRI/E1: machine freeze/dies after a few calls |
10:48AM |
1 |
chan_h323 - Segmentation fault (core dumped) |
8:12AM |
4 |
"Gates steps up telecom campaign" |
7:56AM |
2 |
Strange message !! Unknown IE 76 (Unknown Information Element) |
7:35AM |
0 |
Help me please! |
7:22AM |
1 |
test calls between iaxtel & fwd |
6:42AM |
0 |
Ports open |
6:08AM |
0 |
asterisk on the firewall/router |
5:56AM |
0 |
Gatekeeper with Asterisk |
4:34AM |
3 |
bare-bone config |
4:31AM |
2 |
Problems with MeetMe. |
4:30AM |
2 |
replacing sound files |
3:27AM |
2 |
Extension Dialing problem with SIP |
3:08AM |
1 |
[Asterisk-Dev] Cisco ATA 186 chan_mgcp Transfer problem |
2:48AM |
3 |
Error |
2:47AM |
1 |
newbie: need help configuring asterisk and snom |
2:22AM |
1 |
oh323 inband dtmf - Possible bug? |
2:11AM |
2 |
Generating a call with the Manager interface.. |
2:03AM |
2 |
e100p in norway? |
1:54AM |
0 |
IAX authentication code |
|
Sunday October 12 2003 |
Time | Replies | Subject |
10:57PM |
6 |
SIP phone |
10:40PM |
4 |
X100P Echo Problems..What's going to happen? |
8:16PM |
1 |
AGI Test Fails |
7:42PM |
4 |
No sound with SIP Phones on the Internet |
7:03PM |
0 |
Help: Segmentation fault. Something about smoother |
2:16PM |
1 |
Feedback request: AGI GET DATA change - termination digits |
2:14PM |
1 |
Queues and max time in queue timeout? |
11:46AM |
2 |
Beginner |
3:59AM |
2 |
New Processor support.. |
2:16AM |
3 |
Is this Hardaware Enough for Asterisk ? |
12:38AM |
2 |
INFO method and DTMF translation |
|
Saturday October 11 2003 |
Time | Replies | Subject |
10:14PM |
2 |
Fwd: RE: SIP / IAX over satellite |
10:10PM |
2 |
SIP Configuration |
10:08PM |
2 |
"context confusion" internal context 2 context only? |
7:41PM |
1 |
Distortion of voice after cvs upgrade |
5:24PM |
1 |
LINEJACK -- OUTGOING CALLS |
2:41PM |
1 |
SIP / IAX over satellite |
10:15AM |
4 |
Problems with AGI scripts in Perl and Java |
6:24AM |
0 |
We need an consultant |
|
Friday October 10 2003 |
Time | Replies | Subject |
9:01PM |
1 |
Marketing Digium/Asterisk |
8:06PM |
8 |
T100P & Phones Configuration |
6:05PM |
2 |
Grandstream Setup |
4:22PM |
4 |
IAX Not working between machines |
4:21PM |
1 |
X100P check for Dialtone |
4:10PM |
0 |
Asterisk ** Adtran 750, 5xQuad FXS, 1xQuad FXO, Pentium 500MHz, 10GB hard Drive, 384MB RAM..Will that do? |
2:19PM |
1 |
multiple SIP users on one phone? |
1:50PM |
1 |
No ISA tormenta card message] |
11:48AM |
0 |
Echo observations and some questions |
11:39AM |
1 |
NEWBIE looking for advice. |
11:18AM |
3 |
Grandstream wallmount?? |
10:45AM |
1 |
SIP - H323 GAteway |
10:19AM |
3 |
BudgeTone-102 MWI&CID with Asterisk |
9:53AM |
2 |
Actual audio bitrates |
9:20AM |
2 |
ALERT_INFO=1/ Cisco 79x0 |
6:24AM |
4 |
Caller Id AGI Script |
5:53AM |
0 |
Problems with Devkit Lite setup |
5:51AM |
0 |
Error when making a call |
5:10AM |
0 |
modem connection over handy? |
4:39AM |
6 |
one-way audio |
4:23AM |
0 |
[Asterisk-User] Howto get the Caller Phonenumber ? |
4:09AM |
1 |
Asterisk crash on AGI |
2:30AM |
0 |
Supported dialogic hardware ? |
12:18AM |
1 |
Howto configure asterisk with AVM ISDN card |
|
Thursday October 9 2003 |
Time | Replies | Subject |
10:27PM |
1 |
Problem with DTMF 'looping' / mis-dials (X100P card) |
8:31PM |
0 |
University phone system |
6:16PM |
0 |
What is the "pingtime" option in iax chan(iax.conf)? |
6:03PM |
6 |
X100P Config |
5:03PM |
2 |
E100P setup in Switzerland |
3:53PM |
0 |
Asterisk and DMS100 Channelized T-1 |
1:45PM |
2 |
* consultant needed - will pay |
1:32PM |
0 |
Results SUSE 8.2 + server size |
1:29PM |
2 |
No Ringing from PSTN |
1:25PM |
4 |
Record App Paths |
12:55PM |
0 |
Comdial Unisyn Execumail |
12:53PM |
3 |
How to disable native bridge of SIP-to-SIP calls? |
12:16PM |
4 |
Cisco 7914 |
10:55AM |
1 |
Redhat system init and wcusb |
10:39AM |
4 |
IAX2 Trunking confirmation? |
9:15AM |
1 |
7940/60 TFTP Problem |
8:03AM |
9 |
concurrent calls |
6:13AM |
1 |
real billing time for a call |
5:18AM |
0 |
newbe Echo problem |
3:21AM |
0 |
IAX |
12:58AM |
0 |
Cisco 7940/7960 phone and conference calling ? |
12:51AM |
3 |
Sasquatch, the Loch Ness Monster, UFOs and... |
12:49AM |
1 |
5 second latency sip to oh323 |
|
Wednesday October 8 2003 |
Time | Replies | Subject |
11:07PM |
0 |
7940 |
10:48PM |
0 |
Radius + Asterisk |
8:25PM |
0 |
SER versus Asterisk for WAN SIP Phones |
7:34PM |
0 |
Exten delay matching |
7:10PM |
2 |
pbx_spool and contexts |
3:44PM |
2 |
Registering Softphones to Asterisk |
3:03PM |
0 |
Microsoft RTC Client |
1:26PM |
1 |
Mini-PC box to run server |
12:42PM |
2 |
SIP softphone volume control? |
12:27PM |
1 |
Call Error |
10:09AM |
4 |
asterisk & festival problem. |
9:15AM |
7 |
iax2 trunk |
8:52AM |
2 |
Hypothetical : Working across multiple servers?? |
7:55AM |
0 |
SIP Problems with Cisco 5300 - Invalid CSeq Number |
7:40AM |
2 |
Loop counter variable in dialplan? |
7:19AM |
1 |
Asterisk role |
7:08AM |
2 |
Call to "06302" aborted, insufficient bandwidth |
7:01AM |
7 |
chan_capi and latest Debian package |
6:48AM |
1 |
Cisco 7940/7960 phone and conference calling? |
5:30AM |
2 |
Ztdummy Bug |
5:01AM |
1 |
Asterisk CDR reliablility.. |
4:16AM |
4 |
Music On Hold distorted |
12:28AM |
1 |
BudgeTone 102 flakey sound |
|
Tuesday October 7 2003 |
Time | Replies | Subject |
9:20PM |
4 |
Fax Detection |
3:57PM |
4 |
Newbie |
3:49PM |
0 |
AstMan Issues |
3:15PM |
0 |
Is there always data at /dev/zap/1? |
3:11PM |
1 |
[PATCH] allow announcements in app_dial |
3:05PM |
2 |
Dynamic registration to flakey for production system |
2:23PM |
3 |
Second Send: Using PCI backplane |
2:06PM |
0 |
Problem with SIP Client! |
1:30PM |
5 |
auto 'modprobe wct1xxp' on startup? |
12:51PM |
3 |
Line going to Zombie |
12:12PM |
0 |
Large-scale Asterisk deployments: VON panel |
12:10PM |
2 |
Compile problem SuSE 8.2 |
12:04PM |
1 |
Call Park on SIP phones |
11:58AM |
1 |
FXO on AT&T broadband POTS line? |
11:28AM |
2 |
agi exit problem |
10:48AM |
2 |
call parking on specific park number |
10:30AM |
1 |
clocking source for T100P? |
9:14AM |
5 |
IAX and Jitter problem |
8:39AM |
1 |
Dialling problems |
8:25AM |
0 |
Communication between 2 telephones |
8:16AM |
0 |
RE: Asterisk-Users] IVR Questions? |
7:36AM |
0 |
Connect with another PBX |
6:37AM |
2 |
Can AGI be used in this way? |
6:07AM |
1 |
Digium FXO |
4:01AM |
0 |
Vioce Modems |
3:37AM |
5 |
Voicetronics |
|
Monday October 6 2003 |
Time | Replies | Subject |
9:59PM |
5 |
Help with questions for initial Asterisk wizard (GUI) |
7:26PM |
2 |
Message Waiting on Cisco 7960 |
5:39PM |
2 |
ISDN Dialout |
4:13PM |
1 |
SIP X100P Echo Problems |
4:12PM |
1 |
MP3s in /var/lib/asterisk/mohmp3 causes Asterisk crash |
3:43PM |
0 |
Digium TDM400P and Analog DID Trunks |
3:13PM |
1 |
Web Voicemail Permissions |
2:51PM |
1 |
Snom100 H.323 sample config |
2:41PM |
0 |
getting inbound caller-id from sip remote-party-id field |
1:45PM |
7 |
direct-inward-dialing (DID) |
1:45PM |
2 |
callerid name modification (or adding) |
1:13PM |
1 |
X100P too quiet |
11:17AM |
1 |
chan_zap.c - echo cancelation getting in the way of dialing???? |
11:16AM |
1 |
Start... |
10:13AM |
5 |
Remote control IVR |
8:38AM |
0 |
newbie question: 1 or 2 servers |
8:36AM |
0 |
Priority Voicemail |
8:12AM |
2 |
Modem and Fax over VoIP |
7:51AM |
2 |
Asterisk, X-Lite and iLBC..still.. |
7:40AM |
1 |
Data base structure |
6:48AM |
0 |
problems with the extensions of sip in ATA 186 |
6:34AM |
6 |
Alternatives to FXS cards? |
2:38AM |
1 |
Noise with Grandstream/PSTN |
12:21AM |
2 |
Anyone else use Audacity for prompts? |
|
Sunday October 5 2003 |
Time | Replies | Subject |
11:02PM |
2 |
Good W2K softphone |
5:53PM |
1 |
IVR Questions? |
5:21PM |
0 |
Zap Analog Line Hangup Problem |
4:19PM |
0 |
HI.. .Any comments about VoicePulse |
2:41PM |
1 |
ChanIsAvail app setting ${AVAILCHAN} to an unusable value. |
2:29PM |
2 |
HOW TO GET REGISTER WITH NUFONE?? |
7:49AM |
3 |
DB virtualization for multiple database support - Was Re: How to use vmdb.sql in voicemail.conf/extension.conf |
6:43AM |
2 |
Grandstream 102 |
|
Saturday October 4 2003 |
Time | Replies | Subject |
6:02AM |
2 |
Let's TALK ABOUT IT!!! |
5:40AM |
0 |
another newbie question: forwarding delay? |
|
Friday October 3 2003 |
Time | Replies | Subject |
11:39PM |
1 |
TE410P: Double/missed interrupt detected |
11:23PM |
1 |
starting asterisk? |
9:44PM |
1 |
Editting variable contents |
9:30PM |
0 |
codec bridging |
7:44PM |
2 |
Answer on second ring - need it on first. |
4:43PM |
3 |
monitoring the asterisk and safe restart |
4:09PM |
2 |
802.11 phone review: WiSIP |
3:50PM |
3 |
Job Opening at Digium |
11:28AM |
1 |
Problems with Caller ID on FXO |
11:17AM |
1 |
Help Loading a TDM card!! |
11:01AM |
3 |
Message Waiting on Cisco 7940 does not work |
10:38AM |
3 |
Cisco CallManager Image for 7940/7960 |
10:17AM |
4 |
Iconnect Incomming calls |
9:58AM |
2 |
Ascom Ascotel 2050 & Fritz PCI Card (Capi) |
9:47AM |
0 |
Transfer fails periodically |
9:00AM |
1 |
(still) Channel problem - solved partially |
8:27AM |
9 |
No Ringback on Iconnect |
6:52AM |
2 |
suggested hardware especially sound cards |
6:46AM |
2 |
H.323-SIP Gateway |
6:42AM |
2 |
Transfer from IAX call |
6:16AM |
4 |
monitor |
5:54AM |
2 |
where to specify mysql DB USER PASSWD for voicemail2 |
5:13AM |
2 |
Sound file.. |
3:48AM |
1 |
Budgettone + G729 |
12:00AM |
1 |
primuxisdn capi |
|
Thursday October 2 2003 |
Time | Replies | Subject |
9:45PM |
0 |
SIP Date: header |
9:43PM |
0 |
Help-to start Asterik PBX |
8:33PM |
1 |
THE "NAT-MARE" IS OVER test volunteers needed |
7:41PM |
1 |
"New" TDM cards--driver won't load |
7:36PM |
2 |
Voice detection |
5:05PM |
0 |
chan_h323 Ringing Congestion causes * segfault |
3:56PM |
1 |
iaxtel fixes |
2:50PM |
3 |
SIP and DSL Bandwidth queries. |
1:11PM |
0 |
Gastman working in W2Kp. |
12:54PM |
1 |
problem w/ musiconhold & mpg123 |
12:53PM |
7 |
Any way to get out of a remote console without stopping * |
10:01AM |
1 |
Front end |
9:43AM |
2 |
Problem with Dutch PSTN-line on X100P |
8:25AM |
1 |
Version 1 vs Version 2 |
8:23AM |
0 |
WINXP Messenger SIP Client (Good News, Bad News) WINXP authorization with secret |
7:28AM |
2 |
WINXP Messenger SIP Client (Good News, Bad News) |
6:59AM |
0 |
Fw: Call it Asterisk-Addons and let us go have some fun? |
6:57AM |
2 |
Call it Asterisk-Addons and let us go have some fun? |
6:23AM |
2 |
Asterisk friendly IAX/SIP wholesalers in Australia |
6:18AM |
2 |
Has anyone got * working with Xten soft phones |
5:54AM |
0 |
Help with ISA PhoneJack. |
5:52AM |
1 |
What is mySQL used for in Asterisk? |
4:18AM |
3 |
Xten Lite Build 1079 |
1:07AM |
2 |
Zapateller |
|
Wednesday October 1 2003 |
Time | Replies | Subject |
7:14PM |
7 |
eBay Sip Phone Scam. |
7:08PM |
2 |
Voice Mail App |
2:53PM |
3 |
SIP problems fixed? |
2:40PM |
1 |
DTMF weirdness |
1:32PM |
0 |
oss Errors |
1:21PM |
2 |
newbie question: MOH problem |
12:23PM |
1 |
Audiocodes gateway and asterisk |
12:17PM |
0 |
Question: handling fully-qualified SIP dial requests |
12:03PM |
2 |
grandstream phones and Transfer |
11:30AM |
2 |
SIP Provider Question |
11:04AM |
0 |
Codec problems??? (Was: SIP i.e. Is something broken?) |
10:23AM |
1 |
x100p card - detect dialtone? |
8:36AM |
1 |
R2 signalling |
8:12AM |
1 |
MGCP Phone and Asterisk PBX |
7:32AM |
6 |
recording voice calls |
7:14AM |
0 |
Dialogic D/240SC-T1 REV2 Boards for Sale |
6:59AM |
5 |
single dialplan for multiple Asterisk machines |
6:55AM |
2 |
Directory for Cisco 7960 |
6:40AM |
2 |
VOIP long distance providers |
6:37AM |
0 |
Var vs Global Var vs DB |
5:32AM |
2 |
(still) channel problems |
5:27AM |
3 |
IAX and IAXTEL |
5:19AM |
0 |
Feature ver 1/2 Questions |