Saturday January 31 2004 |
Time | Replies | Subject |
7:39PM |
2 |
Dial via sip gateway? |
5:01PM |
3 |
SUBSCRIBE in chan_sip - anyone? |
3:20PM |
4 |
rtp sound quality? |
2:29PM |
0 |
echo cancellation disabled |
1:58PM |
0 |
The future of VoIP regulation (in the US) |
1:48PM |
0 |
Dial app does not indicate ringing to calling party |
10:49AM |
1 |
Are there any list moderators? |
10:20AM |
2 |
TE410P E1 PRI problem |
8:23AM |
3 |
Caller ID Presentment on PRI... |
7:46AM |
0 |
Using an additional modem to get CallerID information |
6:50AM |
1 |
SIP gateway question |
6:17AM |
1 |
asterisk php status viewer |
3:02AM |
1 |
newbie thinclient env |
12:20AM |
1 |
smtp question |
|
Friday January 30 2004 |
Time | Replies | Subject |
11:40PM |
1 |
Cameron Palmer / voiceglo |
9:21PM |
3 |
Internal Lines Dialing Out |
9:14PM |
1 |
Max messages in VoiceMailMain |
8:34PM |
2 |
Question on setting up asterisk with hunting lines |
7:48PM |
0 |
Voicemail not receiving password or audio |
7:01PM |
0 |
Re: DISA and authcodes (was: t410p) |
5:15PM |
1 |
Voicemail/Playback Questions |
4:51PM |
1 |
Words for Allison(?) |
3:52PM |
2 |
determining legal VoIP service |
2:28PM |
1 |
Help!!!: test Asterisk error: error on IAX1.conf and warning on chan_iax2.c |
2:22PM |
3 |
Call quality questions |
2:17PM |
1 |
SNOM 200 question |
2:10PM |
6 |
Compiling zaptel |
1:47PM |
0 |
recorder |
1:34PM |
3 |
How do you turn on the 7960 msg waiting light? |
1:00PM |
2 |
Extension Questions |
12:57PM |
0 |
X-Lite, X100P, and Speex |
12:54PM |
0 |
error on IAX1.conf and warning on chan_iax2.c |
11:53AM |
0 |
Newbridge Mainstreet 3624 |
11:07AM |
8 |
MeetMe Video option |
10:28AM |
2 |
has Allison said this ? |
10:21AM |
6 |
Compiling while * is running |
9:55AM |
1 |
SIP Transfer problem |
9:34AM |
9 |
Adtran 750 DID question. |
8:39AM |
0 |
newb info needed |
8:23AM |
2 |
PHP developer Wanted ! :-) |
7:59AM |
2 |
IAX1 vs IAX2 for IAXtel |
6:40AM |
2 |
IAX call problems |
6:31AM |
3 |
P2P RTP without SIP re-invites |
6:20AM |
1 |
Auto dial in Off Hook situation. |
6:07AM |
1 |
mediatrix, dtmf |
6:04AM |
1 |
ZAPRTC load error |
5:58AM |
0 |
billing software |
4:57AM |
7 |
Calls dropping off |
4:06AM |
2 |
Asterisk with a laptop with built-in Intel 537 modem |
3:33AM |
2 |
Music on Hold Warnings |
12:58AM |
2 |
Can Asterisk act like a normal sip phone? |
|
Thursday January 29 2004 |
Time | Replies | Subject |
10:25PM |
0 |
Suggestions for debugging an IAX1 one way audio problem |
9:01PM |
0 |
canreinvite and codec negotations... |
7:05PM |
1 |
Grandstream Firmware ? |
5:54PM |
3 |
TDM400 FXO??? |
5:10PM |
3 |
How to delay dialing |
4:30PM |
3 |
Expire old voice mail messages, et al |
3:43PM |
1 |
WTB E100P or better |
3:08PM |
5 |
Echo worsens in 0.7.1 |
2:43PM |
1 |
Duplicates ? |
2:17PM |
0 |
ZAP answering |
12:50PM |
1 |
T1 PRI question |
11:46AM |
3 |
Incoming Voice/Fax Discrimination? |
11:17AM |
0 |
IAX provider in Colorado |
11:05AM |
1 |
Re: Asterisk and gnugk (bam) |
10:37AM |
0 |
Automatically Logging Out Queue Members |
10:02AM |
1 |
Migrating home POTS VM to Asterisk VM |
9:57AM |
1 |
Stable branch - any update available? |
9:19AM |
0 |
Re: Asterisk-Users digest, Vol 1 #2663 - 14 msgs |
9:18AM |
1 |
IP Blue Skinny Client with Asterisk |
9:16AM |
4 |
Multiple Line Appearances |
9:12AM |
2 |
IAX Implementation Problem |
8:28AM |
0 |
Compatibility with Dialogic D4/PCI |
8:01AM |
3 |
good job on the list server! |
7:57AM |
1 |
SIP URI matching |
7:33AM |
4 |
dialing wrong numbers |
6:27AM |
1 |
Send DTMF tone Like "C" on connected call |
4:51AM |
0 |
Cdr call transfer |
4:45AM |
0 |
Register to h323 gk |
4:39AM |
4 |
Asterisk Manager Interface notes |
4:39AM |
1 |
WG: Reference projects which using Asterisk !? |
4:33AM |
3 |
small correction |
4:29AM |
1 |
re: help with voicepulse connect IAX2 |
3:47AM |
0 |
DTMF wrongly recognised |
2:42AM |
1 |
running asterisk under root |
2:34AM |
1 |
asterisk with big number of extentions. |
2:33AM |
1 |
Running Asterisk on FreeBSD. |
1:56AM |
0 |
Asterisk Configuration + MySQL |
1:41AM |
0 |
asterisk as B2BUA with SER? |
1:20AM |
0 |
Latest Article on Asterisk in the press |
|
Wednesday January 28 2004 |
Time | Replies | Subject |
11:33PM |
2 |
specific to X100P with UK telephone lines |
9:58PM |
0 |
Channels |
8:21PM |
0 |
lists.digium.com |
8:18PM |
2 |
Does anyone manage the wiki? |
7:10PM |
1 |
SIP error |
5:47PM |
0 |
DTMF tone stops working with IAX (voicepulse)??? |
5:14PM |
2 |
festival patch missing in latest CVS or stable build |
4:05PM |
1 |
List traffic |
3:47PM |
1 |
cdr_addon_mysql compile error |
8:53AM |
0 |
test list |
3:54AM |
1 |
IAX2 / SIP testing |
|
Tuesday January 27 2004 |
Time | Replies | Subject |
10:56PM |
1 |
Cisco 7960 Problems |
10:08PM |
1 |
Asterisk Appliance |
9:49PM |
1 |
Distinctive ring Issues |
8:19PM |
0 |
ztdummy won't compile |
7:16PM |
0 |
X100P doesn´t hangup |
6:11PM |
4 |
Introducing Firefly |
4:29PM |
1 |
One2One application? |
2:32PM |
0 |
Rolm PBX + Asterisk |
11:38AM |
0 |
Anyone with experience with polish PTSN? |
10:50AM |
0 |
H 323 + Netmeeting test drive |
6:18AM |
0 |
Ethereal IAX2 support |
5:16AM |
0 |
differentiate incoming calls on SIP clients |
2:41AM |
0 |
CDR records on call transfer |
2:40AM |
0 |
Asterisk and BRI ISDN |
|
Monday January 26 2004 |
Time | Replies | Subject |
8:13PM |
0 |
Is this list down? |
7:43PM |
0 |
"NOTICE: Fax detected" on 3'rd DTMF digit on outgoing call to outside line. |
7:22PM |
0 |
I need some clarification on DTMF |
6:45PM |
0 |
Anyone run * on OS X ? |
6:06PM |
0 |
ip num of incoming sip calls? |
5:39PM |
0 |
canreinvite and codec negotations... and NAT |
5:36PM |
1 |
Is there a way to transfer a call from CLI |
5:34PM |
0 |
# transfer with IAX2 |
4:00PM |
0 |
ADPCM support with RECORD FILE |
2:56PM |
1 |
RE: Bluetooth discussions (quick glance to some BT products) |
2:22PM |
0 |
Call Queue wait times |
1:38PM |
1 |
AVM C4 with Asterisk in front of an ISDN PBX |
1:29PM |
0 |
Not sure what I'm looking for to ask correctly. |
1:18PM |
1 |
7960 Problems |
12:18PM |
0 |
IAX2 hold |
11:46AM |
0 |
Asterisk RPMS Updated (FC1,RH9,RH73) |
11:30AM |
1 |
SIP behind NAT - use of "externip" option |
9:40AM |
0 |
Scalability/Benchmarks/Performance |
8:43AM |
2 |
TE410P on Redhat 9 |
8:29AM |
0 |
Detect Answering Machine in Outgoing calls |
7:01AM |
3 |
X-Lite & Asterisk: Speex & iLBC not working? |
6:53AM |
3 |
Questions regarding new echo cancellation features... |
6:02AM |
0 |
Know if a call is answered |
5:59AM |
3 |
app_queue and dialplan |
5:33AM |
0 |
Digium FXO Card |
5:20AM |
0 |
SIP - fax / voicemail |
4:11AM |
1 |
ZAP Problems |
3:52AM |
2 |
Need Europian vendor for Digium hardware. |
3:36AM |
0 |
He really doesn't care |
2:53AM |
1 |
Wildcard X100P usable in Germany? |
2:19AM |
7 |
GSM modems |
1:50AM |
0 |
GSM phone to *? |
|
Sunday January 25 2004 |
Time | Replies | Subject |
11:56PM |
3 |
looking for iax termination |
6:38PM |
3 |
OH323 doesnt hear ringing |
5:17PM |
1 |
Using TDM400P for autodial |
4:12PM |
2 |
Example of TDM20B |
3:51PM |
1 |
Ne machine, build is breaking |
10:19AM |
1 |
iax1.conf ?? |
6:01AM |
2 |
Incoming SIP matching |
12:48AM |
1 |
Announcement: Another test release of chan_sccp |
|
Saturday January 24 2004 |
Time | Replies | Subject |
11:22PM |
2 |
Sipura 2000 Transmit Issues? No Sound being passed to caller |
10:44PM |
0 |
FW: one way choppy sound problem ! |
9:37PM |
1 |
billing systems |
9:14PM |
0 |
Ring/Off-hook Message |
1:17PM |
1 |
Incoming DID call Voice Problems |
12:49PM |
1 |
Warning /:Asterisk.c:255 Listener : Select Returned Error |
12:15PM |
1 |
Asterisk RPMS for RH9 + RH7.3 |
10:25AM |
13 |
Has Nufone gone belly-up |
9:40AM |
0 |
SAFE_ASTERISK DIED - EXIT CODE 127 |
9:27AM |
1 |
Bluetooth discussions |
6:04AM |
0 |
IAX hard phone |
5:48AM |
2 |
Subject: Re: Grandstream 100 sidetone |
5:21AM |
1 |
Searching the archives - new engine demo |
4:17AM |
0 |
strange problem with grandstream software 1.0.4.39 |
2:53AM |
1 |
Is there any plans for Digium ISDN BRI card? |
2:03AM |
2 |
MSN 6.1 |
1:26AM |
4 |
retrans_pkt: Maximum retries exceeded on call |
12:30AM |
3 |
Grandstream 100 sidetone |
|
Friday January 23 2004 |
Time | Replies | Subject |
11:35PM |
1 |
exten=>h and ResetCDR |
11:27PM |
0 |
SIP + ADPCM |
11:04PM |
3 |
Problem installing Asterisk with Mandrake 9.1 |
9:35PM |
1 |
Capabilites of Asterisk |
6:35PM |
3 |
RFC3389 support issue with DG104S |
5:37PM |
2 |
echo problems |
4:30PM |
0 |
Multiple voices on 64K channel (was) simple question... |
4:15PM |
1 |
DG104S firmware has error? |
3:50PM |
0 |
Excternip and FWD |
2:25PM |
0 |
Troubles with the System Attendent Patch. |
1:27PM |
1 |
Back to front logging for calls placed through /var/spool/asterisk/outgoing? |
1:06PM |
2 |
MI2 |
12:40PM |
6 |
Mediatrix 1204 sip experience? |
12:03PM |
2 |
Latest cvs * compile error anyone? |
11:39AM |
1 |
PSTN incoming - both SIP & H323 always arrive in default context :-? |
11:02AM |
0 |
SIP wierdness after upgrade from 0.7.1 to CVS |
9:51AM |
3 |
UK BT Interface with asterisk? |
9:49AM |
6 |
rc.local dont works |
9:08AM |
2 |
Debian Packages and Mirrors |
8:59AM |
1 |
compiling * pipe error |
8:30AM |
12 |
8 lines - best approach |
8:23AM |
0 |
New Asterisk article on O'Reilly's onlamp.com |
7:33AM |
3 |
SIP Absolute Timeout |
7:08AM |
1 |
TE410P/Zaptel |
5:22AM |
1 |
Asterisk + Dialup Modem |
5:16AM |
1 |
Voicemail2 & Mysql Connection |
4:50AM |
1 |
Buying asterisk? |
3:00AM |
3 |
SIP register/auth with Grandstream BudgeTone-100 |
2:48AM |
2 |
chan h323 Compile problem |
1:33AM |
1 |
AW: I got it (was: Cisco 7940 with asterisk) |
12:36AM |
1 |
USB headset recommendations |
12:00AM |
2 |
Maillinglist as newsgroup ? |
|
Thursday January 22 2004 |
Time | Replies | Subject |
10:38PM |
2 |
Polycom Reboot Script - Please wiki-size me |
10:11PM |
3 |
MGCP w/8x8 DTA-310 and as5300 pstn gateway |
8:55PM |
1 |
simple question... |
8:32PM |
3 |
Asterisk vs. Websphere Voice Response? |
8:25PM |
0 |
voiceglo.com and dtmf |
4:30PM |
0 |
Source code for Iax Phone (the new IAX client) now available |
3:43PM |
1 |
Variable to play all gsm files in a directory? |
3:37PM |
0 |
Rtp WARNING Messages on the Cli in safe_asterisk |
3:35PM |
2 |
MGCP Problem. |
3:14PM |
1 |
Grandstream transfer solution + DTMF translation possible? |
2:17PM |
0 |
is the mike back on? |
2:04PM |
5 |
Snom 200 phones not working. |
12:40PM |
0 |
Draytek SIP phones are broken |
12:38PM |
2 |
ETSI PRI ISDN Signalling |
11:44AM |
0 |
Problem with flashing FXO callwaiting from FXS |
11:06AM |
2 |
Using varables in MeetMe? |
9:36AM |
1 |
sidetone issue |
9:30AM |
0 |
Integrating * with a legacy Nec NEAX 1400 |
9:07AM |
0 |
hardware compatibility |
8:54AM |
0 |
Grandstream 10 |
8:45AM |
1 |
OT: Canada's Primus introduces SIP localserv ice |
8:43AM |
0 |
RE: Asterisk-Users digest, Vol 1 #2588 - 11 msgs |
8:39AM |
0 |
Cause of transfer problem (GRANDSTREAM!) |
7:39AM |
0 |
* and rh9 boot problem |
6:25AM |
4 |
Gsm + snom phones |
6:13AM |
3 |
Grandstream 101 |
5:48AM |
2 |
asterisk 0.7.1 - mysql |
5:15AM |
0 |
Switchboard interface |
5:13AM |
0 |
Description of Manager events |
4:04AM |
1 |
Call Queue with no agents -> Congestion or voicebox instead of MOH? |
4:01AM |
3 |
R2 or E&M for E1 CAS pbx to pbx link |
3:31AM |
0 |
DIAX CallMe feature |
3:07AM |
3 |
Data calls (ISDN/64k) through * PRI |
2:45AM |
2 |
Standalone FXO device |
1:35AM |
0 |
Codecs and more analog lines? |
1:07AM |
1 |
chan_capi: suppress calling number on outbound dialing? |
12:40AM |
1 |
Asterisk 0.7.1 RH 7.3 RPMS Released |
|
Wednesday January 21 2004 |
Time | Replies | Subject |
11:34PM |
0 |
Asterisk Management Interface... Do you want one? |
8:56PM |
0 |
X100P remote hangup detection problem |
8:08PM |
1 |
Transfer problem |
8:07PM |
2 |
Diax IAX2 |
8:03PM |
0 |
Asterisk on news.com w00t |
7:15PM |
0 |
CRM Solutions - Incoming Calls & Outgoing Calls |
6:56PM |
0 |
Net2Phone error 407: Unauthorized |
6:48PM |
2 |
Polycom Soundpoint IP400 |
6:26PM |
1 |
need help configuring IAX to make outbound calls through a remote server |
6:17PM |
1 |
mp3player not working |
5:44PM |
3 |
Mailing List Lag |
5:01PM |
9 |
New Windows IAX Client |
5:00PM |
0 |
G729 Codec Error |
4:35PM |
1 |
OT: Canada's Primus introduces SIP localservice |
4:33PM |
2 |
disable transfer on outgoing calls? |
4:10PM |
1 |
Fax problem |
4:09PM |
0 |
Conf files |
2:53PM |
1 |
Strange Zaptel Modprobe driver failure |
2:51PM |
0 |
Asterisk at Open Source-Telephony-Summit 2004 |
2:34PM |
11 |
Digium X100P for $43 |
11:08AM |
1 |
Lucent Definity + Asterisk Success! |
11:07AM |
1 |
h323 with innovaphone ip 400 gatekeeper/innovaphone Ip200 phones |
11:05AM |
0 |
Fw: Word-of-the-Day: wiki |
10:38AM |
1 |
OH323 config file format |
10:16AM |
2 |
Starting with MGCP and Asterisk |
10:07AM |
1 |
Sip phones transfer not working. |
9:58AM |
4 |
What technology could my phone company be using? |
9:36AM |
1 |
Is there a way to # of agents logged into a queue ? |
9:23AM |
1 |
PrePaid App. |
8:16AM |
1 |
help request about wiax2.dll |
8:13AM |
1 |
zaptel - problem solved |
7:51AM |
2 |
zaptel part II |
7:41AM |
3 |
Making a call with sample.call |
7:12AM |
0 |
Modulated tones not working |
6:41AM |
1 |
Parked Calls Settings |
6:39AM |
1 |
ISDN p2p AVM Fritz Card |
5:34AM |
1 |
Reorder tone ...when it should be Busy... |
5:23AM |
0 |
Mediatrix 1104 register problem ? |
2:46AM |
8 |
Calling Card Application |
2:00AM |
0 |
Different classes of MusicOnHold |
1:31AM |
1 |
Zap show channel |
12:05AM |
1 |
Hi |
12:05AM |
2 |
CAS SF Inband tone signalling problem |
|
Tuesday January 20 2004 |
Time | Replies | Subject |
11:59PM |
2 |
Brandwidth for making internet calls |
11:40PM |
2 |
how scalable is digium cards? |
11:16PM |
0 |
Asterisk-Driven Linux Distro |
9:57PM |
1 |
Toll-Free Gateway Beta Test: freenum.org |
9:29PM |
0 |
Out of trunk data space |
8:20PM |
1 |
Can asteric be used with just a voice card |
8:09PM |
1 |
VOIP on linux |
7:23PM |
1 |
G729 - how many needed? |
6:35PM |
0 |
Play volume/speed adjustment on the per call basis |
6:12PM |
2 |
Restricting/Negotiating H323 Port Ranges |
6:06PM |
0 |
installing Asterisk on FreeBSD |
5:33PM |
0 |
Voice Pulse Connect problems Authing |
4:37PM |
1 |
T400P / T100P with Hong Kong IDA-P Lines |
4:30PM |
0 |
Dialplan Sanity Check please |
3:47PM |
2 |
I got it (was: Cisco 7940 with asterisk) |
3:47PM |
1 |
PRI NI2 |
3:40PM |
0 |
New Swissvoice ip10 firmware: 1.0.0 build 3 |
3:20PM |
1 |
help - recording both sides of a conversati on |
3:03PM |
0 |
chan_capi capiECT |
2:12PM |
0 |
Is libpri symmetrical? |
2:08PM |
1 |
Cisco 7940 with asterisk |
1:43PM |
0 |
FLASH TONE |
1:25PM |
1 |
OT: Canada's Primus introduces SIP local service |
1:24PM |
0 |
Agent timeout then Dial() ? |
1:11PM |
0 |
Grandstream cfg.txt hacking? |
1:11PM |
1 |
Music on Hold - can it be done without mpg123? |
12:06PM |
0 |
[A-bit-OT] Power Over Ethernet Discovery process |
11:45AM |
2 |
How to diagnose "pops" and "clicks"? |
11:42AM |
3 |
G.729 Licenses from Digium |
11:37AM |
2 |
DTMF A-D |
11:28AM |
5 |
MeetMe questions |
10:57AM |
0 |
Power Over Ethernet for *any* ethernet switc h (or hub); product idea |
10:44AM |
0 |
Power Over Ethernet for *any* ethernet switch(or hub); product idea |
10:30AM |
4 |
CAPI: Early-B3 working with AVM-B1? |
10:12AM |
1 |
PSTN Gateway |
9:51AM |
2 |
multipledaily digest!!! |
9:51AM |
3 |
Enter Pin followed by Pound key |
9:20AM |
0 |
ADSI phone vs. IP phone (and proper implementation thereof) |
7:59AM |
0 |
wink time |
7:52AM |
1 |
Compiling problems with SuSE |
6:58AM |
0 |
SIP: outbound calls |
6:53AM |
0 |
AG4000C and T100P |
6:41AM |
0 |
Broken macros during transferring call |
6:05AM |
2 |
Re-Invite between SIP phones |
5:27AM |
3 |
Still problems at compiling |
3:18AM |
1 |
open h323 |
1:59AM |
9 |
Power Over Ethernet for *any* ethernet switch (or hub); product idea |
12:43AM |
0 |
Outbound call with Go2Call |
|
Monday January 19 2004 |
Time | Replies | Subject |
10:31PM |
0 |
Call token is ip$localhost |
9:20PM |
1 |
FW: Memory problem |
8:16PM |
4 |
CVS Changes (NAT-SIP) |
3:12PM |
2 |
PLAYBACK multiple files |
1:42PM |
3 |
Getting correct CDR info |
11:55AM |
0 |
Routecall application |
11:16AM |
2 |
Lucent and ISDN-PRI |
10:57AM |
2 |
RE: current version |
9:52AM |
3 |
Residential services |
8:59AM |
2 |
Different Caller ID for each Zap Interface |
7:28AM |
1 |
Concurrents calls on asterisk with H323 |
6:36AM |
0 |
Re: [Asterisk-Dev] benevolent dictatorship, or inclusive developper community? |
6:20AM |
1 |
pri gateways and asterisk |
5:06AM |
1 |
Transferring H.323 Call |
4:34AM |
2 |
Hangup detection failed |
4:33AM |
1 |
SIP: Register that isn't a register? |
4:30AM |
0 |
Best Codec ? |
4:28AM |
0 |
MGCP: condition 14? |
4:19AM |
3 |
Search engine for this list |
4:11AM |
0 |
words for Alison |
3:56AM |
0 |
Dialogic cards with Asterisk |
2:44AM |
1 |
Connecting BRI to PRI card? |
1:42AM |
3 |
configuration to Grandstream via tftp |
12:50AM |
6 |
IAX2 bug in DIAX solved - Great Thanks to Steven! |
|
Sunday January 18 2004 |
Time | Replies | Subject |
9:55PM |
3 |
ATA-186 pass-through Flash |
9:28PM |
1 |
Calls with incoming distinctive ring |
8:44PM |
1 |
California DID Access |
8:16PM |
1 |
RE: current version |
6:28PM |
4 |
[ot] Grandstream hardware |
5:54PM |
3 |
Now: Small Biz Robust Asterisk Solution - SBRAS |
4:40PM |
0 |
Re: newbie ISDN question |
2:22PM |
0 |
Office-wide paging with Asterisk and Cisco 7960 7940 phones |
2:15PM |
5 |
Latest version of asterisk |
1:11PM |
1 |
Asterisk switching ISDN data calls? |
1:08PM |
2 |
Asterisk as SIP Redirect Server -- Implemented - Not Working - Plz Help |
11:45AM |
1 |
Public switches (AXE10) not capable of handling sustained call setup bursts on E100P |
11:01AM |
2 |
Re: ultra-cheap asterisk box -> Small Biz Robust Asterisk Solution - SBRAS |
9:00AM |
0 |
RE: Newbee question (hot phone numbers) |
8:33AM |
2 |
Nufone not taking GSM CALLS |
7:51AM |
1 |
minimum system hardware for Asterisk install |
7:31AM |
1 |
No startup after mpg123 install |
12:04AM |
6 |
ADSI phone vs. IP phone |
|
Saturday January 17 2004 |
Time | Replies | Subject |
10:16PM |
1 |
Channel Bank Woes... |
9:04PM |
4 |
Asterisk Indications |
8:48PM |
1 |
X100P Configs for Australia |
8:03PM |
0 |
Remote reloading Cisco phones... |
7:56PM |
1 |
Registering multiple FWD accounts |
7:42PM |
0 |
Unavailable versus private in extensions.conf? |
6:49PM |
9 |
New sounds also now in CVS |
4:54PM |
3 |
cdr_odbc not logging integers eg duration |
12:54PM |
0 |
New sounds posted |
11:57AM |
0 |
Asterisk/X100 - Sipura Configuration |
9:15AM |
0 |
Combining 2 AUDIO Frames |
9:10AM |
0 |
Long time to detect that an IAX2 user is not logged on |
7:37AM |
3 |
Playing background message |
7:32AM |
0 |
Kernel for 1586 vs i686 for Asterisk? |
7:06AM |
6 |
Zone Paging |
6:56AM |
0 |
Early B3 on PRI channel (Listen free mex without charge) |
5:03AM |
3 |
SS7 over Asterisk ? |
3:18AM |
1 |
Voicetronix OpenLine4: disable answering on a particular channel & delay before dial |
|
Friday January 16 2004 |
Time | Replies | Subject |
7:58PM |
3 |
Class features in dialplan ? |
7:34PM |
0 |
stuck on retransmit |
7:29PM |
1 |
7960 Phone disconnects when dialing using speaker |
6:52PM |
0 |
Re: Asterisk-Users digest, Vol 1 #2525 - 11 msgs |
6:50PM |
0 |
Updated zaprtc, anyone interested? |
6:24PM |
1 |
G.729 quiestion |
5:06PM |
1 |
Caller id and callback |
4:10PM |
1 |
doublehash patch doesn't work in asterisk 0.7.1 |
3:59PM |
1 |
Analog phone help |
3:55PM |
2 |
Hardware for Asterisk |
1:48PM |
1 |
ERROR[8192] |
1:46PM |
1 |
Asterisk Integration with Lucent Definity g3 si |
1:43PM |
0 |
re: hardware requirement -asterisk |
1:10PM |
0 |
RoutCall Info |
1:10PM |
2 |
VoiceMail - no user pre-registration |
12:30PM |
4 |
G.723.1 codec |
12:13PM |
0 |
Analog phone transfer |
12:10PM |
0 |
sound of static removed by hitting flash button |
11:25AM |
1 |
Configuration for modem |
11:22AM |
0 |
ultra-cheap asterisk box -> sorta OT, more a bout Dell |
11:13AM |
11 |
Remote reload Cisco 7960 |
10:59AM |
1 |
Advice Request: 2-4 line, 10 station * system |
10:52AM |
1 |
Asterisk Integration with Lucent Definity g3si |
9:59AM |
2 |
NO DTMF detection in the Outgoing call with GW Cisco5300 |
9:05AM |
2 |
'Intercom' before call transfer |
8:45AM |
0 |
SMP kernel with X100P card |
7:07AM |
7 |
CAPI not installed, after changed from i4l to CAPI |
6:09AM |
1 |
CDR problem with macros |
5:46AM |
0 |
Meetme sound dropped ? Out of trunk data space ?? |
3:53AM |
2 |
Asterisk over WAN |
3:30AM |
2 |
ISDN30 - HW ? |
3:28AM |
0 |
GS Handytone Echo-problem |
1:48AM |
3 |
Odbc not logging |
|
Thursday January 15 2004 |
Time | Replies | Subject |
11:03PM |
3 |
Sending voicemail with qmail |
10:19PM |
3 |
vmail cgi script |
9:41PM |
1 |
meetme - ztdummy |
9:35PM |
0 |
T100P & CA Adit 600 Question |
9:16PM |
0 |
FW: Sending voicemail with qmail and call waiting |
8:50PM |
1 |
SER & Asterisk |
8:36PM |
0 |
announcement using Dial |
7:19PM |
1 |
WANTED: Toll-Free gateways in Europe/Asia/Africa/South America |
7:06PM |
1 |
Help! Asterisk 0.7.1 No Sound in recorded gsm files |
6:18PM |
4 |
meetme without zaptel hardware |
4:12PM |
1 |
Voicetronix Openline 4 + asterisk |
4:03PM |
0 |
Ringback Problem |
3:34PM |
2 |
re: hardware requirement -asterisk |
2:57PM |
3 |
Voicemail Sequence Bug? |
2:19PM |
2 |
re: hardware requirement asterisk |
2:18PM |
3 |
Re Grandstream 1.0.4.38 |
1:56PM |
0 |
t1xxp Unable to request IRQ |
1:22PM |
1 |
SIP Phones - Power over ethernet? |
1:14PM |
2 |
[OT] Commercial conferencing solution? |
1:09PM |
1 |
QoS anyone? |
12:43PM |
1 |
want suggestion to get hardware to learn |
12:39PM |
12 |
capacity testing |
12:38PM |
0 |
Possible Bug: Crash when Parking Calls |
12:32PM |
0 |
establishing before answer "Early B3" libpri |
12:26PM |
0 |
cdr processing |
11:16AM |
0 |
Free Message Signaling |
11:13AM |
1 |
Credit Card Terminal |
11:04AM |
3 |
Cisco FXO as PSTN gateway (updated request for assistance) |
10:37AM |
0 |
Users in sweden |
9:49AM |
0 |
IOS Bug crashing SIP nat sessions |
9:41AM |
4 |
People detected as fax machines |
9:14AM |
2 |
Disturbing trend of * production boxes that shouldn't be |
9:08AM |
4 |
ultra-cheap asterisk box |
8:49AM |
0 |
Choosing a VoIP Protocol. |
8:39AM |
2 |
wav49 voicemail problem with Windows Media Player |
8:12AM |
0 |
ATA186 SIP Outbound Fax Calls |
8:00AM |
0 |
ISDN newbie |
7:37AM |
1 |
Problem at compiling zaptel (again) |
7:19AM |
1 |
Problem at compiling zaptel |
7:13AM |
1 |
Strange sound when fax answers (app_rxfax) |
6:12AM |
2 |
asterisk.org webpage |
5:54AM |
3 |
B-channels restart problem |
5:23AM |
0 |
best SIP-softphone? |
3:48AM |
0 |
Parking extension:700 |
3:39AM |
3 |
ISDN CAPI and anonymous callers |
1:55AM |
0 |
SIP clipping sound |
1:48AM |
1 |
GSM connection for asterisk |
1:07AM |
0 |
H.323 protocol security vulnerability |
12:44AM |
0 |
AW: Re: Again: 7920 Cisco IP Phone Skinny & SIP |
12:22AM |
2 |
hardware requirements - asterisk |
|
Wednesday January 14 2004 |
Time | Replies | Subject |
10:38PM |
1 |
Codec matching weirdness |
10:37PM |
4 |
re hardware requirement - asterisk |
9:08PM |
5 |
* For Call Center |
8:55PM |
1 |
Re Hardware requirement -Asterisk |
7:46PM |
0 |
unload chan_zap.so now possible :) |
7:12PM |
2 |
Single/Dual DS3 - anyone seen this? |
7:06PM |
1 |
zaptel compile erro!(asterisk last version0.7.1) |
6:51PM |
1 |
Cooperate with SIP ITSP |
6:21PM |
1 |
System Attendent |
6:14PM |
0 |
MeetMe, conferencing questions |
5:51PM |
0 |
asterisk & faxing |
5:18PM |
0 |
Windows Call Manager : Formerly [Asterisk-Dev] New Bounty |
5:04PM |
2 |
Static Noise coming from Wildcard FXS: Wildcard TDM400P |
2:37PM |
1 |
Skinny behind NAT? |
1:33PM |
0 |
hungup problems with SIP and x100P card |
1:01PM |
3 |
100% of cpu in an out of the box * |
12:26PM |
1 |
Asterisk as a protocol converter? |
12:09PM |
1 |
Daytime/nighttime broken in asterisk-0.7.0? |
11:46AM |
1 |
How do we updated to the new .7.1 version. |
11:30AM |
0 |
BOUNTY POSTED - Zaptel drivers for *BSD |
11:15AM |
5 |
SNOM IAX image |
10:22AM |
3 |
Basic Asterisk capabilities question |
10:06AM |
0 |
Re: Proposed solution for exit code priority jumps |
9:59AM |
0 |
Re: failover (was Re: voicepulse) |
9:19AM |
2 |
Re: failover (was Re: voicepulse) |
8:52AM |
1 |
DTMF Debug |
8:50AM |
3 |
Asterisk 0.7.1 |
8:41AM |
3 |
NAT friendly TFTP Server |
8:09AM |
4 |
Multiple phonenumbers on one E1 PRI with Digium TE410P ? |
7:46AM |
1 |
... H323 - segmentation fault - core dumped |
6:52AM |
0 |
TDM switching between Digium TE410P ports |
5:40AM |
3 |
grandstream asterisk configuration |
4:48AM |
1 |
Asterisk drops calls - E100P |
4:45AM |
6 |
How to park and pickup a call |
1:41AM |
1 |
hardware requirements of asterisk |
1:36AM |
1 |
always 4 rings before * answers!? |
12:56AM |
7 |
Why I can not use the conference |
12:22AM |
0 |
Kernel 2.6 and ztdummy? |
|
Tuesday January 13 2004 |
Time | Replies | Subject |
9:07PM |
0 |
Asterisk in Linux Journal |
8:20PM |
5 |
linux journal article on asterisk |
8:04PM |
0 |
H.323 security flaws article |
7:48PM |
3 |
Re: Proposed solution for exit code priority jumps |
7:23PM |
2 |
How can I get support about Dialogic hardware |
5:07PM |
7 |
Parking extension not working |
4:48PM |
0 |
Fun (or lack of) with asterisk & T100P |
3:26PM |
0 |
* and signaling (clarification) |
2:53PM |
2 |
Mediatrix 1102 issue after upgrading to CVS |
2:09PM |
0 |
Turning up the volume on outgoing sip to sip gateway calls? |
1:57PM |
0 |
SIP test suite: anyone with spare time? |
1:02PM |
1 |
max queue time; newbie question (fwd) |
11:28AM |
0 |
OH232: cancel call does not stop ringing |
11:25AM |
0 |
* and telco ringback |
10:58AM |
0 |
inbound call routing problem - RESOLVED |
10:41AM |
3 |
threewaycalling ? (Bridge 2 SIP calls?) |
10:19AM |
1 |
E100P works with PCI 3.3V and 5V? |
9:59AM |
0 |
agents and call queueing |
9:55AM |
6 |
SIP and AGI crash... |
9:50AM |
0 |
Memory allocation issues |
9:43AM |
0 |
New software SIP phone released today |
9:38AM |
0 |
Cisco Multiple Products H.323 Protocol Denial of Service Vulnerabilities |
8:50AM |
0 |
CVS problem |
8:13AM |
1 |
24x7x365 asterisk support available? |
8:12AM |
1 |
Specifying a codec to be used in /etc/sip.conf |
7:52AM |
4 |
inbound call routing problem |
7:36AM |
2 |
Voicepulse |
6:53AM |
3 |
How to Order Disconnect Supervision from SBC using Adit 600? |
6:49AM |
4 |
Again: 7920 Cisco IP Phone Skinny & SIP |
6:41AM |
1 |
Symbol NetVision Phone |
6:27AM |
0 |
sme |
6:11AM |
2 |
Asterisk and Festival (* dies with no info) |
4:48AM |
11 |
Best Linux Distribution |
4:46AM |
1 |
GUI client for windows for live monitoring/b arge |
4:42AM |
1 |
cisco 7910 phone |
4:28AM |
1 |
E100P without q931? |
3:40AM |
2 |
KPhone working |
3:31AM |
2 |
FS/OS Telephony Summit 2004 |
12:26AM |
2 |
Nufone.net net wackiness? |
|
Monday January 12 2004 |
Time | Replies | Subject |
11:35PM |
0 |
0.7.0 Release Mirrors |
11:30PM |
1 |
newbie to asterisk |
11:10PM |
4 |
Asterisk 0.7.0 |
10:23PM |
0 |
sip and x-lite |
7:22PM |
0 |
Does asterisk surport old voice card? |
7:02PM |
2 |
GUI client for windows for live monitoring/barge |
6:47PM |
3 |
MeetMe issues? |
6:28PM |
3 |
Fw: problem with safe_asterisk |
5:58PM |
1 |
Asterisk Voicemail that reacts to my AIM status |
5:16PM |
2 |
Re: Nauti miles |
4:01PM |
7 |
3.3v PCI board - TE410P photo |
3:11PM |
1 |
Release 0.7.0 - not yet |
3:01PM |
0 |
Grandstream and VLAN |
2:41PM |
0 |
FW: How to bind RTP when IP alias are configured |
2:37PM |
0 |
Disconnect Supervision, SBC, and Adit 600 |
1:42PM |
1 |
New Installation problem |
12:57PM |
0 |
Wiki - stable -dev |
12:41PM |
0 |
Turning a profit (WAS: More words for Allis on) |
12:27PM |
1 |
ADSI. used beyond own phone network? |
11:37AM |
0 |
Routing packets in and out |
11:34AM |
0 |
Fw: Forward call with response required to accept |
11:16AM |
0 |
Voice Caller Name and NUmber? |
10:33AM |
2 |
How to bind RTP when IP alias are configured |
10:17AM |
2 |
LCR / Trollphone Rate Engine |
9:46AM |
0 |
Fax handled on E&M T1 DEBUG messages |
9:45AM |
4 |
Issue - vmail.cgi on Redhat 9 (Apache) ? |
9:40AM |
4 |
RFC3389 messages with ATA 186 |
9:38AM |
1 |
Advance Options in VoicemailMain() ? |
9:31AM |
3 |
Thank You All |
9:31AM |
2 |
Securing Cisco SIP gateway |
8:27AM |
2 |
'*' call conference? |
8:23AM |
3 |
Linux Sip UAs |
8:16AM |
1 |
CAS Idle definition bits ? |
8:13AM |
1 |
E100P - connected to Cisco |
7:27AM |
0 |
Fax on Cisco ATA |
7:20AM |
2 |
host=dynamic and defaultip=xxx |
7:18AM |
0 |
Using the FLASH key on MGCP sends DTMF to the third party |
7:12AM |
1 |
Message signaling on MGCP (light on the phone) |
6:42AM |
1 |
Cisco FXO as PSTN gateway |
6:09AM |
2 |
bad pstn audio coming from old * processes |
4:53AM |
2 |
A question on codec translation. |
4:28AM |
0 |
OH323: Dropping incompatible voice frame |
3:42AM |
1 |
Install problem (compile error) |
3:00AM |
2 |
SIP-Client for Handheld PC |
1:06AM |
0 |
Queue timeout problem |
12:49AM |
4 |
Bandwidth ? + Doc + cdr |
|
Sunday January 11 2004 |
Time | Replies | Subject |
7:17PM |
1 |
Re: Asterisk-Users digest, Vol 1 #2448 - 10 msgs |
7:07PM |
1 |
possible solution to PRI T100P dropped call issue |
6:50PM |
0 |
WTS (200) AC Power Adapters for Cisco 7910 / 7940 / 7960 IP Phones |
6:34PM |
2 |
Forward call with response required to accept |
6:01PM |
1 |
zttool and errors |
5:36PM |
24 |
More words for Allison |
4:58PM |
0 |
NuFone Network H323 configuration? |
4:41PM |
6 |
T1 Sync clarification |
4:10PM |
0 |
"friendly" dial tone frequency combinations |
4:07PM |
4 |
analog or sip ? was far end disconnect supervision |
3:18PM |
2 |
Asterisk on FreeBSD 4.9? |
3:12PM |
2 |
CONTEST: Top Posters win 80G Hard Drive |
3:09PM |
0 |
Asterisk on FreeBSD 4.9 |
12:33PM |
2 |
SpeakFree |
10:24AM |
1 |
Re: [Asterisk-Dev] More Success on the Cisco 7920 and SCCP !!!!! |
9:55AM |
0 |
Strange problem with call hangup on Budgetone 102 Phones |
9:47AM |
2 |
macro error "exited non-zero" |
9:27AM |
2 |
Cisco 79xx Ringtones |
8:54AM |
1 |
New Version of SJPhone |
4:51AM |
1 |
More Success on the Cisco 7920 and SCCP !!!!! |
3:36AM |
2 |
High Level of CVS activity |
1:50AM |
0 |
a constructive proposal: tie the marshals to a cvs server |
|
Saturday January 10 2004 |
Time | Replies | Subject |
11:24PM |
2 |
WTB / WTS Voip hardware |
10:13PM |
0 |
how do i make this happen [macro-record-cleanup] |
9:26PM |
0 |
Using ACD functionality for main number answer and "music on hold" |
8:45PM |
5 |
Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon) |
7:29PM |
1 |
default music source for SIP channel |
6:55PM |
0 |
REQ: init/startup scripts for asterisk for possible inclusion in version 0.7.0 |
5:25PM |
0 |
Record calls where to put line? |
4:07PM |
5 |
Asterisk + BudgeTone (behind NAT) |
3:48PM |
0 |
My first E1 card is running :) |
2:19PM |
2 |
drop calls with T100P / PRI |
2:03PM |
2 |
far end disconnect supervision |
1:26PM |
1 |
ADSI Configs |
1:04PM |
0 |
IAX v1 Changes |
12:34PM |
2 |
Free Software or not -- that's the question /* New subject */ |
11:11AM |
0 |
E1 - E100P connected to Cisco - problem - HELP |
11:09AM |
0 |
Bridging ethernet over hdlc |
11:01AM |
2 |
Record all phone calls |
10:55AM |
3 |
R2 Digital - Brazil |
10:22AM |
1 |
Oops! |
8:39AM |
2 |
E100P - Error 500 |
8:28AM |
2 |
Forums Need Help |
5:39AM |
0 |
FYI: New SIP Flash Image 7940/7960 IP Phone |
3:22AM |
0 |
Call transfer message |
2:04AM |
0 |
Music_on_hold adjust volume |
12:57AM |
1 |
picking a channel bank |
|
Friday January 9 2004 |
Time | Replies | Subject |
11:48PM |
2 |
Re: Asterisk-Users digest, Vol 1 #2413 - 13 msgs |
11:29PM |
0 |
AW: Problems with Cisco 7920/Skinny/Asterisk |
10:57PM |
7 |
Chagres Technologies, Inc |
9:43PM |
0 |
more VoIP news - "wire"taps |
9:40PM |
1 |
Called Party Identification |
9:37PM |
1 |
crontab |
9:37PM |
1 |
(no subject) |
7:55PM |
3 |
newbie question; can * screen calls? |
7:25PM |
0 |
(no subject) |
6:49PM |
3 |
ChanIsAvail and SIP |
5:45PM |
1 |
chan_iax2.c Ignoring Port For Now |
5:14PM |
3 |
file_inlcude .. why not? |
5:06PM |
0 |
Slightly OT: CNN: FCC cautious on Voice-over-Internet regulation |
5:02PM |
1 |
Fwd: new cvs build failure |
4:51PM |
0 |
new cvs build failure |
4:18PM |
2 |
Broken DNS makes Asterisk whacky! |
4:13PM |
1 |
Asteriks as SIP<>H323 Proxy? |
3:55PM |
1 |
At last!!! :) |
2:38PM |
1 |
Screen Pop & Remote Agents = Telemarketing |
1:20PM |
1 |
* as sip b2bua? |
1:18PM |
3 |
Why * try to codec translate when it can do without during codec negotiation. |
12:44PM |
0 |
soft fax machine |
12:31PM |
1 |
zapbarge w/o the mute |
11:50AM |
12 |
USA dial plan |
11:46AM |
0 |
IConnect audio quality |
11:32AM |
3 |
Screen Pop & Remote Agents |
11:09AM |
0 |
SIP/2.0 487 Request Cancelled |
11:00AM |
5 |
Cisco Gear |
10:49AM |
0 |
Problems with Cisco 7920/Skinny/Asterisk |
10:42AM |
2 |
* dialing before line is open? |
10:34AM |
1 |
DTMF in MeetMe |
10:23AM |
1 |
Help with compiling |
8:22AM |
1 |
log incoming and outgoing call |
7:42AM |
3 |
Very high delay |
7:34AM |
0 |
Receiving faxes from a SIP gateway |
6:36AM |
1 |
Development Process comment and Email list suggestion |
5:51AM |
0 |
SV: Mailing list growth |
4:11AM |
2 |
asterisk sip with voicemail |
4:08AM |
1 |
A question about Linux kernels and Asterisk |
4:07AM |
2 |
max queue time; newbie question |
2:18AM |
0 |
DTMF through H.245 UserIndication |
|
Thursday January 8 2004 |
Time | Replies | Subject |
11:42PM |
2 |
SIP reload configuration problem /* New subject */ |
9:32PM |
1 |
GrandStream giving an RTP Read Error Again |
8:35PM |
3 |
Progress on the Polycom front... |
7:09PM |
5 |
Dialing the Phone from OS X Address Book with AppleScript, XML-RPC, PHP and Asterisk |
7:01PM |
0 |
Hangup detection issue |
5:05PM |
0 |
iaxtel iax.conf entry? |
4:52PM |
0 |
RE: [Asterisk-Dev] Asterisk Development Updates |
3:03PM |
0 |
Error messages during Registeration on CVS Version CVS-01/08/04-14:20:34 |
2:44PM |
2 |
Asterisk Development Updates |
2:24PM |
0 |
Re: Sound Card help -- Solution -- and a question |
2:09PM |
1 |
latest cvs == broken tdmoe |
1:58PM |
3 |
Kedpad less extension |
1:54PM |
4 |
Dial from command line? |
1:20PM |
0 |
Getting VoicePulse # listed in 411 |
1:19PM |
1 |
SIP URI's: possible now? |
12:03PM |
0 |
Clicking until DSP called |
12:02PM |
2 |
Some bugs www.voip-info.org html |
11:30AM |
1 |
Re: 911 and lawsuits and redundancy |
11:12AM |
0 |
Asterisk success stories in small-mediumoffi ce environments? |
10:41AM |
0 |
E100P - R2 support |
10:29AM |
3 |
T100P positional on PCI bus? |
10:27AM |
4 |
2nd call leg status? |
9:09AM |
3 |
Asterisk success stories in small-mediumoffice environments? |
8:45AM |
1 |
Cisco tftp |
8:35AM |
3 |
Asterisk hanging? |
8:26AM |
1 |
Nortel Option 61C PBX? |
8:21AM |
1 |
Billing system experiences with Advanced Communications? |
8:03AM |
0 |
BudgetTone 102D |
6:26AM |
5 |
AbsoluteTimeout Users Messages |
6:22AM |
1 |
Phonejack |
6:02AM |
3 |
Asterisk & Sipura 2000 |
5:36AM |
3 |
Administrative suggestions |
4:40AM |
1 |
Strange Call waiting problems - SNOM 200 & Grandstream Budgetone |
4:25AM |
1 |
AW: IPv6 support |
2:33AM |
0 |
Asterisk Capability Module ? |
2:12AM |
1 |
E100P : Pb with outgoing calls |
1:48AM |
9 |
Mailing list growth |
1:46AM |
2 |
Red Alarms - FXS(Signalling Q) |
|
Wednesday January 7 2004 |
Time | Replies | Subject |
11:42PM |
3 |
SIP and error talking to voicemail |
10:56PM |
2 |
IPv6 support |
10:23PM |
0 |
" Error in buffer handling Message" |
7:40PM |
0 |
2.4 Kernel and Hyperthreading (was Re: P4 processor |
7:39PM |
0 |
RE: Inexpensive Analog Ports |
6:03PM |
2 |
* and Cisco Gateways |
5:59PM |
0 |
Frazzled newbie questions |
4:33PM |
3 |
PRI D Channel and Caller-ID issue...... |
4:13PM |
1 |
Re: 911 and lawsuits and redundancy |
3:52PM |
0 |
DTMF via SIP not working for certain phone systems |
2:04PM |
1 |
E1 - E100P connected to Cisco - problem |
2:01PM |
1 |
yet another question on DID trunks |
1:16PM |
1 |
Test Post-Do Not Read |
1:04PM |
2 |
Asterisk success stories in small-medium office environments? |
12:56PM |
4 |
Newbie Question-Looking for Feedback |
12:56PM |
3 |
Voicemail account size limit ? |
12:32PM |
1 |
(newbie) Hardware sizing question |
11:46AM |
2 |
zaprtc install problem |
10:24AM |
2 |
A Note to GS users.. |
10:01AM |
1 |
Call Rollover |
9:38AM |
1 |
(no subject) |
9:36AM |
4 |
* crashed |
9:24AM |
0 |
IAX2 missing? |
8:59AM |
0 |
Asterisk log messages |
8:44AM |
0 |
Re: 911 and lawsuits and redundancy |
7:25AM |
1 |
DTMF recognized improperly? |
6:59AM |
8 |
Asterisk + fax |
6:42AM |
5 |
Client for P800/P900 |
6:24AM |
2 |
P4 processor with Hyperthreading and Asterisk |
6:04AM |
1 |
Unexpected ISDN hangup on outbound call |
5:06AM |
3 |
manipulating with numbers - StripMSD, Prefix |
2:27AM |
0 |
Asterisk stops responding after about 80 calls |
12:10AM |
0 |
Small scaled VoIP calling card system |
|
Tuesday January 6 2004 |
Time | Replies | Subject |
10:11PM |
0 |
Asterisk interop with Syndeo |
9:12PM |
3 |
no results. |
6:46PM |
0 |
Voicemail scalability |
6:40PM |
0 |
Asterisk support for NEON or Centrex/CLASS/VMWI MW formats ? |
4:47PM |
0 |
HTML tags? |
4:41PM |
1 |
IAX2 Trunk two Asterisk boxes. |
4:15PM |
1 |
Re: 911 and lawsuits and redundancy |
3:41PM |
1 |
ATA call |
3:19PM |
3 |
MWI message not seen on SNOM200 |
2:49PM |
2 |
benevolent dictatorship, or inclusive developper community? |
2:45PM |
3 |
Doorbells & Door Intercoms |
2:17PM |
0 |
Request for Information on Asterisk Functionality |
1:55PM |
1 |
Need Cisco 7940 or 7960s at good price for Asterisk deployment |
1:25PM |
3 |
Voicemail to email file sizes |
1:24PM |
1 |
Fw: Pls confirm |
12:30PM |
2 |
911 |
12:30PM |
0 |
Call Transfer Function in * |
12:25PM |
2 |
Heads up v2.03h on snom 200 |
12:15PM |
5 |
Scaleable Solution for small office |
12:06PM |
1 |
Hpw to enable Voicemail Indicator on IP/Analog Phone ? |
11:00AM |
4 |
Pls confirm |
10:23AM |
1 |
Got SIP response 482 "Loop Detected" |
9:56AM |
7 |
911 and lawsuits |
9:16AM |
0 |
Re: Multi-line help & AOL Messenger Style PBX Navigation |
8:58AM |
1 |
ring tone |
8:37AM |
0 |
[Fwd: reject connect from iaxtel.com] |
8:35AM |
0 |
small question from a new user |
8:17AM |
2 |
URGENT - micronet & asterisk on h323 |
7:35AM |
2 |
How to flash hook when there is no hook ? |
7:17AM |
4 |
Asterisk feature list: spreadsheet |
7:15AM |
3 |
cant load drivers for TE410P cards |
7:08AM |
0 |
Asterisk Nat Issue |
6:42AM |
3 |
Policies - deny some nubers |
5:56AM |
4 |
AGI Scripting |
5:16AM |
2 |
Problems compiling cdr_pgsql |
4:45AM |
1 |
FW: Matrix Orbital (usbl LCD or VFD) (oops, wrong list I think) |
4:24AM |
1 |
Call Queue and Agent Statistics |
4:17AM |
1 |
Asterisk not working with session border controller |
12:53AM |
1 |
IVR Question |
12:09AM |
1 |
"Everyone is busy at this time" message ? |
|
Monday January 5 2004 |
Time | Replies | Subject |
11:48PM |
0 |
Interfacing Asterisk with PSTN network (Nortel SL100 PBX) |
10:24PM |
2 |
I stumbled on this list... |
9:26PM |
0 |
Special variable for AGENT |
8:08PM |
0 |
problems dialing area code |
6:07PM |
2 |
Message waiting indicator |
5:45PM |
1 |
Identifying the Originating Cisco SIP Gateway |
5:42PM |
0 |
asterisk sccp support |
5:41PM |
2 |
How to monitor calls initiated by .call file using manager interface? |
5:19PM |
4 |
Echo on polycom sip phone |
5:17PM |
3 |
This is a test |
3:50PM |
0 |
Need Help... |
3:35PM |
0 |
Lindows ? |
3:23PM |
7 |
Are messages censored on this board? |
3:13PM |
1 |
FW: This newbie gives up for now - sadly (2) |
2:55PM |
0 |
HTML Stripping in mailing lists? |
2:28PM |
2 |
Echo with polycom phones |
2:28PM |
3 |
question re voicemail |
1:50PM |
1 |
reject connect from iaxtel.com |
12:46PM |
0 |
queue questions: max time in queue; customer option to drop out of queue |
12:44PM |
8 |
This newbie gives up for now - sadly |
11:50AM |
0 |
Codec Negotiation Does not seem to work as e xpected ?? Help Please !! |
11:44AM |
0 |
mailbox= wrong context. was: Newbie - MWI |
9:36AM |
1 |
Question about MP3's |
9:24AM |
8 |
Sip Trunking |
9:18AM |
0 |
Hardware to build an Enterprise AsteriskUniversal Gateway |
8:28AM |
3 |
DID Trunk Lines and Caller ID |
4:18AM |
1 |
CLIR and isdn4linux |
3:06AM |
0 |
FW: SIP to SIP redirect while ringing |
1:39AM |
1 |
"Internal" ISDN bus |
12:38AM |
7 |
RE: Inexpensive Analog Ports |
12:29AM |
2 |
Codec Negotiation Does not seem to work as expected ?? Help Please !! |
|
Sunday January 4 2004 |
Time | Replies | Subject |
9:16PM |
8 |
Grandstream Handytone 286 RTP Problems |
7:37PM |
1 |
Hold and transfer problem |
7:36PM |
1 |
Voicepulse DID fast busy |
7:31PM |
1 |
4 X100P Cards |
7:25PM |
4 |
Sun Servers with UltraSparc Processors |
7:11PM |
1 |
pager reminder script |
5:52PM |
0 |
Dutch/DTMF Caller ID |
5:45PM |
4 |
Cisco to Cisco - poor quality |
5:31PM |
2 |
Earpiece Connections |
5:18PM |
5 |
Multi-line help |
4:04PM |
1 |
Cisco 12sp+ program update |
3:07PM |
3 |
Newbie - MWI |
11:09AM |
3 |
OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany? |
10:11AM |
2 |
Two E100P boards - could not load zaptel module - Channel 63 - no such device |
10:07AM |
2 |
Voicemail Out call |
4:46AM |
2 |
POTS interfacing recommendation |
4:33AM |
1 |
Modem Communications thru * |
3:35AM |
3 |
Hardware to build an Enterprise Asterisk Universal Gateway |
3:25AM |
0 |
TDM400P & X101P cards, echo issues? |
3:18AM |
4 |
CAPI, transfering thru a 2nd PBX - keep original CallerID |
|
Saturday January 3 2004 |
Time | Replies | Subject |
10:09PM |
3 |
AW: AW: Snom 200 with two extns defined anyone? |
8:59PM |
0 |
TDM400P driver modprobe failed |
7:31PM |
2 |
STOP THIS THREAD New to asterisk? RUN... don't walk. |
4:29PM |
0 |
Free PSTN calls |
12:05PM |
0 |
expression parsing |
10:14AM |
1 |
Newbie - getting two local phones tocommunicate would be a good start :) |
|
Friday January 2 2004 |
Time | Replies | Subject |
9:42PM |
1 |
Asterisk Gotoif / last called |
8:05PM |
0 |
Newbridge Mainstreet 3624 Manual |
6:48PM |
2 |
Cisco SIP license? |
3:57PM |
4 |
Newbie - getting two local phones to communicate would be a good start :) |
2:59PM |
0 |
Grandstream Flash Button |
1:42PM |
2 |
AgentCallbackLogin. |
1:00PM |
2 |
Newbridge Mainstreet 3624 T1 channel bank no w Alcatel |
11:34AM |
1 |
mini-ITX suggestions |
11:25AM |
6 |
hangup detection |
10:28AM |
3 |
T400P & E400P second source |
9:17AM |
2 |
Malloc debug kills asterisk? |
8:27AM |
1 |
asterisk dies while making calls |
7:24AM |
3 |
* Stresstool Help required |
6:59AM |
3 |
Slow wiki? |
5:35AM |
0 |
SQL Updater Down!!! |
4:23AM |
1 |
License questioni supose ?? |
2:23AM |
4 |
one way choppy sound problem ! |
1:18AM |
0 |
IAXy Release ? |
12:31AM |
1 |
FW: [Asterisk-Dev] Strange bells in voicemail and voicemailmain ? |
|
Thursday January 1 2004 |
Time | Replies | Subject |
9:48PM |
2 |
sound driver advise needed |
8:44PM |
2 |
How to load the driver of TDM400P card! |
7:17PM |
2 |
Prediction for 2004 |
12:31PM |
1 |
Newbridge Mainstreet 3624 T1 channel bank now Alcatel |
11:25AM |
4 |
* crash when forward voicemail --Nicolas Gudino |
10:27AM |
10 |
help |
10:04AM |
1 |
asterisk gateway to other gateways |
1:41AM |
1 |
asterisk reload for FWD to register |