Sunday November 30 2003 |
Time | Replies | Subject |
9:15PM |
1 |
Sound Breaks |
8:17PM |
1 |
Dial "T" option not obeyed with Grandstream BT101 |
4:48PM |
5 |
cisco 7960 power suplies? |
4:31PM |
1 |
LCR with ENUM and DDNS: half the story |
1:30PM |
1 |
Rate file formats: a standard? |
11:13AM |
2 |
Cisco 6.0 + Asterisk question |
5:21AM |
1 |
app_queue behavior |
|
Saturday November 29 2003 |
Time | Replies | Subject |
11:47PM |
0 |
IAX phones and CPU usage problem |
10:28PM |
14 |
* Party in Paris |
8:52PM |
1 |
Sip Issue |
5:57PM |
1 |
asterisk server crashing |
3:11PM |
1 |
iaxComm Update available [Ringtones, Intercom, UI improvements] |
11:36AM |
0 |
ENUM and DNS/Bind |
3:26AM |
0 |
Asterisk Run Problem |
|
Friday November 28 2003 |
Time | Replies | Subject |
11:58PM |
2 |
Deltathree icomming problem |
5:42PM |
0 |
Can't seem to connect/call fwd network Help! |
3:33PM |
0 |
Asterisck as a Fujistu 9600 VOIP Gateway |
3:13PM |
0 |
Re: Resend: Help for oh323 |
12:40PM |
2 |
Survey says post your 3.3 volt Mother boards used in PRODUCTION withTE410 |
11:56AM |
0 |
QUESTION Ringing Appl. |
11:24AM |
0 |
Iax termination in India |
10:38AM |
0 |
Asterisk install / update script - need testers |
10:02AM |
6 |
3x AVM Fritz!Card PCI for a EuroISDN PBX. |
8:15AM |
3 |
H323.conf |
7:59AM |
1 |
ASTERISK WITHOUT ANY CARD |
7:26AM |
1 |
Problem with SIP-Phones and * audio-files |
7:25AM |
1 |
Request for debug message in ENUM code |
6:50AM |
1 |
channel offset between Asterisk and PBX |
6:15AM |
0 |
TEDAS VoIP DECT PABX |
5:51AM |
2 |
How does Asterisk use CPU? |
4:42AM |
4 |
Mute button in Grandstream? |
1:18AM |
2 |
MGCP Support for NAT |
1:14AM |
4 |
call waiting disable in sip |
|
Thursday November 27 2003 |
Time | Replies | Subject |
11:38PM |
1 |
files for upgrade cisco 7960 phone |
9:57PM |
4 |
RFC3389 support incomplete |
9:10PM |
13 |
Asterisk behind NAT << How to do it. |
7:19PM |
1 |
AGI (IF/ELSE) |
4:49PM |
4 |
Mailing list archives searchable ? |
4:49PM |
4 |
RE: Grandstream BT-100 and |
4:39PM |
0 |
OT: CISCO-ATA-186 |
3:59PM |
0 |
RE: Grandstream BT-100 and latest CVS |
3:24PM |
5 |
IAX2 Ethereal plugin v0.3 is out |
3:19PM |
1 |
ISDN, BRI, PRI, voice over IP, and more... |
1:59PM |
2 |
ENUM regexp replacements |
12:32PM |
0 |
Has anyone else had problems with Chagres? |
12:14PM |
4 |
Multi-line TTS Outbound Dialer |
11:04AM |
0 |
Larger SIP packets |
10:04AM |
0 |
Timeout feature in queues.conf does not seem to work |
9:49AM |
1 |
Agent Logoff inability when calls are being received from queue |
9:28AM |
6 |
Help for oh323 |
8:21AM |
0 |
distinctive ring doesn't work |
6:04AM |
1 |
Crash - What is happening here??? |
5:36AM |
0 |
Problem after re-compiling |
2:06AM |
1 |
App queue and all Agent busy |
1:05AM |
8 |
MGCP problem |
|
Wednesday November 26 2003 |
Time | Replies | Subject |
11:54PM |
3 |
AGI - Freakin Lost |
11:45PM |
0 |
GrandStream BudgeTone 101 Question |
11:09PM |
1 |
Attempting to get SJPhone configured for Asterisk- Help! |
10:23PM |
2 |
unixodbc-vm-routines.h |
6:02PM |
0 |
Asterisk Training |
5:22PM |
3 |
AGI - CallerID ?? |
5:14PM |
0 |
beware supposed PCI 2.2 compatibility! |
2:51PM |
0 |
SIP PBX features |
2:44PM |
5 |
door phone |
2:16PM |
2 |
Symmetric RTP |
1:55PM |
2 |
Issues with Privacy Manager and Zapateller |
11:15AM |
0 |
prob. w/ data (modem) calls through Asterisk |
10:53AM |
0 |
VoIP bandwidth management with linux & CBQ |
10:52AM |
0 |
bugs.digium.com and acknowledged status |
9:01AM |
0 |
jitter buffer for iax |
8:55AM |
2 |
Modem cards?? |
8:49AM |
0 |
ADSI Programming - Found Guide on Designing Apps |
8:29AM |
1 |
perl --> manager problem |
7:33AM |
2 |
Web interface? |
7:10AM |
0 |
Needed - AGI Scripting |
5:09AM |
1 |
Pbx / channel bank install |
4:37AM |
3 |
Virtual PBX (*) |
4:27AM |
0 |
Echo and Call Setup suggestions |
|
Tuesday November 25 2003 |
Time | Replies | Subject |
6:49PM |
0 |
Warning: File chan_sip.c, Line 462 |
6:01PM |
3 |
Handytone 286 - calling out |
4:51PM |
2 |
modem to modem calls through asterisk |
4:24PM |
1 |
About sound modules in Asterisk. And call gnophone-asterisk-h323 |
2:23PM |
5 |
Distinctive ring confusion |
2:04PM |
1 |
Want to get rid of Tormenta error message |
1:23PM |
4 |
How to demo * on a notebook |
1:17PM |
1 |
Crashed Asterisk |
12:51PM |
4 |
Options for 3rd party call control |
12:14PM |
4 |
AGI Rocks!! (A happy camper) |
10:13AM |
2 |
Outgoing-call and enter user in Conference - repost |
9:58AM |
10 |
PCI 3.3 V |
8:37AM |
1 |
Picking a channel (FXO port or SIP) for outb ound calls |
8:30AM |
1 |
ISDN Cards available in Australia |
8:24AM |
8 |
Prompt recording |
6:42AM |
2 |
zt_rec: Unknown error 500 |
4:22AM |
1 |
Problem with fax detection |
4:11AM |
1 |
Ring requested on channel 1 already in use... |
3:49AM |
1 |
How to use * to simply skim off callerid (UK)? |
2:22AM |
4 |
* Configuration |
1:28AM |
6 |
cdr_unixodbc |
12:28AM |
0 |
what is the problem? |
12:27AM |
0 |
For all IAXTEL users of DIAX |
12:05AM |
1 |
SIMPLE support in Asterisk? |
|
Monday November 24 2003 |
Time | Replies | Subject |
10:42PM |
10 |
g729 license |
8:48PM |
1 |
NTT FSK - Japanese Caller ID |
6:07PM |
1 |
MGCP RFC (2705) vs. PacketCable MGCP spec |
1:33PM |
4 |
One voicemail -> multiple recipients? |
1:04PM |
0 |
Pickupgroup and IAX phones |
12:46PM |
1 |
TIME ZONE support |
12:28PM |
4 |
Sip phones! |
12:01PM |
0 |
Picking an open channel (FXO port) for outbo und calls |
11:06AM |
2 |
Ring power on Analog adapters |
10:45AM |
11 |
Picking an open channel (FXO port) for outbound calls |
9:42AM |
3 |
strange SIP authentication/authorization behaviour |
9:10AM |
5 |
test call request |
9:10AM |
3 |
Cisco to asterisk termination with h323 and g729 finally works. |
8:41AM |
2 |
Pressing 0 in Voicemail causes * to hangup |
8:12AM |
1 |
Re: Asterisk-Users digest, Vol 1 #1994 - 14 msgs |
7:34AM |
0 |
SIP channel modification |
6:38AM |
0 |
SIP to SIP redirect while ringing |
5:47AM |
2 |
Netphone SIP phone |
|
Sunday November 23 2003 |
Time | Replies | Subject |
10:46PM |
1 |
Nufone account not registering |
9:14PM |
1 |
CVS update of Asterisk - What did I do wrong? |
7:43PM |
1 |
agi exec problem (followup) |
7:39PM |
1 |
Phone compatibility list |
7:17PM |
2 |
agi exec problem. |
6:47PM |
2 |
SIP Express Router & Asterisk |
6:03PM |
8 |
Re: [Asterisk] GSM access |
4:06PM |
0 |
Call Pickup ??? |
1:15PM |
3 |
Feedback with X100P and SIP fwd.pulver |
8:45AM |
6 |
RxFax |
6:36AM |
1 |
SIP Asterisk -> Nikotel disconnects after 1 Minute |
|
Saturday November 22 2003 |
Time | Replies | Subject |
11:05PM |
0 |
Local numbers to Victorville/Apple Valley, CA |
8:47PM |
0 |
* on 2.4.20-gentoo-r8 linux with eicon-diva-pro-pci-2.0 isdn-bri card how-to ??? |
4:01PM |
1 |
Help Required for Speex |
3:11PM |
0 |
Stability with the Supura SIP Units |
2:20PM |
1 |
g729 codec questions error running asterisk now |
1:42PM |
2 |
New DIAX - version 0.9.4 - a big step forward - available for download |
12:51PM |
3 |
SIP channel improvements |
11:49AM |
0 |
Asterisk - phone docs |
11:39AM |
0 |
Zap MWI method |
10:05AM |
1 |
X100P configuration Problem |
9:11AM |
2 |
How to dial out using OH323? |
7:10AM |
0 |
Experimental Switzerland -> IAX gateway |
4:11AM |
0 |
Opteron - Kernel optimizations |
12:14AM |
2 |
Newbie ... some questions |
|
Friday November 21 2003 |
Time | Replies | Subject |
10:43PM |
2 |
Stutter dialtone but no messages |
7:58PM |
1 |
Echo Cancellation, TDMoE fails, X100P works |
6:39PM |
0 |
status of distinctive ring |
5:33PM |
5 |
MOH - Hold Button - I think I'm going crazy |
4:13PM |
3 |
PRI problems |
2:55PM |
4 |
Unable to create channel of type 'SIP' |
2:12PM |
1 |
making outside call with sip phone |
1:31PM |
5 |
Asterisk Call Manager for Windows 0.0.1 (Alpha) |
11:03AM |
4 |
Current CVS problem |
11:03AM |
0 |
One way sound |
10:49AM |
2 |
DIAX, IAX2 and latency |
10:36AM |
1 |
Can you monitor a call via the asterisk speaker system and do a call pickup if you wish |
10:34AM |
9 |
Outline For Asterisk Book - Please Review & Comment |
9:23AM |
0 |
MGCP with Cisco GW |
8:40AM |
1 |
Outgoing-call and enter user in Conference |
6:54AM |
1 |
SAY NUMBER in AGI? |
6:10AM |
3 |
Upgrade CISCO 7960 Question |
5:53AM |
0 |
Asterisk SIP implementation |
4:45AM |
0 |
Asterisk and Proxy issues |
3:39AM |
1 |
callerid problem...zaptel ppl |
3:06AM |
2 |
Is Asterisk suitable for this use? |
1:48AM |
2 |
Which ISDM BRI Card for Asterisk? |
|
Thursday November 20 2003 |
Time | Replies | Subject |
11:19PM |
1 |
can't get caller id? |
11:07PM |
2 |
Change the all announcement |
10:37PM |
0 |
Unknown RTP codec 72 received NOTICE while using Xlite with * |
9:20PM |
2 |
ADSI Hold |
8:01PM |
1 |
Linux Voice Mail Application?? |
7:03PM |
0 |
FW: Mailing list email masquerading. |
6:00PM |
4 |
T1 / Channel Bank Configuration |
5:47PM |
0 |
Mailing list email masquerading. |
5:07PM |
0 |
SIP URI... d'oh! |
5:07PM |
2 |
Dialup Internet through Asterisk server? |
5:06PM |
2 |
No ringback |
5:02PM |
2 |
Solved! Snom 200 Busy signal |
4:15PM |
2 |
SIP URIs and ENUM or other types of lookup |
3:53PM |
0 |
Asterisk Lists (was Re: Asterisk Business discussion again) |
3:48PM |
0 |
General Email and Mailing List Etiquette |
3:46PM |
2 |
Snom 200 stuck on "Busy" |
2:30PM |
1 |
new iaxComm build available |
2:05PM |
1 |
is possible isdn card in analog line? |
1:59PM |
0 |
SIP<->*/ISDN<->PSTN calls almost impossible because of echo |
1:37PM |
2 |
Voicemail just hanging up... |
12:03PM |
4 |
Tuning the Linux kernel? |
10:32AM |
2 |
Zaptel DAX? |
10:05AM |
0 |
Missing Manager Events/Actions: Hold, Reconnect, Conference |
9:58AM |
2 |
asterisk-oh323 v0.5.7 bugfix release |
9:32AM |
0 |
codec pass-through feature |
8:45AM |
2 |
TE410P ERRORS under load |
8:44AM |
8 |
tunnel iax via gnophone with ssh? |
8:38AM |
1 |
Can I soft-link a voicemailbox? |
8:26AM |
2 |
Scope of the "h" extension.. |
8:02AM |
2 |
Cisco to use * as a gateway? |
7:46AM |
1 |
IAX2 Ethereal Plugin initial release |
6:56AM |
0 |
Stutter Tone all the time? |
5:49AM |
2 |
iaxComm new version installation problem |
5:03AM |
2 |
Cannot do international dial with E1 in Spain |
4:13AM |
1 |
Cisco DTMF Issue |
4:05AM |
2 |
VOIP --> PSTN via. voicemodem/soundcard. |
3:12AM |
5 |
The internet needs a dialing code.. |
2:49AM |
0 |
Tdm400p FXS faults |
2:08AM |
4 |
Still TDM400P problem |
|
Wednesday November 19 2003 |
Time | Replies | Subject |
11:32PM |
0 |
Can anyone give me an example of sip.conf and extensions.conf about asterisk SIP Proxy server? |
10:26PM |
1 |
Controlling asterisk in a dynamic way |
9:26PM |
5 |
Help configuring CISCO 7960 |
8:38PM |
2 |
ATA-186 Double Digit problems |
8:13PM |
0 |
Getting in to h323 |
8:07PM |
2 |
PSTN intercepted announcement |
6:35PM |
8 |
FAQ, Documentation, How-to, etc |
5:03PM |
8 |
Asterisk Business discussion again |
4:23PM |
1 |
Mediatrix 1102 / 1104 authentication problems.... |
3:14PM |
1 |
fritz pci / chan_capi / australia setup |
2:21PM |
1 |
Application CallingPres |
12:51PM |
1 |
Play a "sound" after dialing a user... |
12:10PM |
3 |
RTP timing in a SIP only world (choppy MOH) |
10:50AM |
0 |
GoTo or Dial in AGI?? |
10:20AM |
1 |
FXO card still won't pick up... |
9:52AM |
8 |
Help please |
9:26AM |
0 |
Question on hearing ADSI CAS tone |
9:07AM |
0 |
AMA flags by context |
8:41AM |
7 |
FYI: Simple Small Asterisk install.. |
6:25AM |
1 |
Service codes for MGCP channels |
5:41AM |
3 |
inter diax connection |
5:36AM |
1 |
2 TE410P |
5:31AM |
0 |
T100P and Meetme |
5:25AM |
1 |
VOIP onver the net |
5:10AM |
1 |
Message from * console. |
4:51AM |
10 |
4 Port FXO cards |
4:18AM |
0 |
SIP/IAX2 DTMF |
4:07AM |
2 |
creative VoIP blaster & * |
3:55AM |
1 |
E100P driver overwrites memory used bye linux-kernel |
2:30AM |
0 |
TDM400P channel problem |
2:18AM |
2 |
g723 to g723 SIP call - warning message |
|
Tuesday November 18 2003 |
Time | Replies | Subject |
7:57PM |
1 |
Can't connect to digium cvs |
6:22PM |
0 |
Unable to specify channel 2: Device or resource busy |
6:04PM |
4 |
This is how you Search the Archives |
4:29PM |
1 |
Question about incoming/outgoing |
4:18PM |
1 |
iax vs iax2 question |
4:10PM |
3 |
FXO Card/Interface for Australia |
2:57PM |
3 |
Wifi600 or other Wifi sip phones |
2:40PM |
3 |
Ethereal plugin for IAX2 |
2:16PM |
3 |
hold music =] |
1:15PM |
3 |
"Unable to find path from G729A to ULAW" on Sipphone.com |
1:01PM |
0 |
Bad DTMF detection |
1:00PM |
2 |
SIP Context from domain? |
12:03PM |
6 |
Asterisk GUI Client Released!!! |
11:37AM |
0 |
Musisc on hold insted of Ringing tone |
11:18AM |
4 |
Help with Warnings |
10:56AM |
1 |
Asterisk with External Voicemail |
10:04AM |
1 |
telco access ?s -- PRI, T1, POTS? |
9:22AM |
1 |
DIAX - Can place a call, but can't be called?! |
9:07AM |
1 |
Will Asterisk be supporting RTCP XR in the future? |
9:01AM |
2 |
ISDN Card Types for Europe |
8:56AM |
1 |
capi config |
8:34AM |
0 |
Swissvoice ip10s MGCP questions and experiences |
8:25AM |
0 |
codec problems between * and cisco hardware h323 |
8:01AM |
0 |
Hard & soft phones |
7:45AM |
2 |
ask problem about softphone--asterisk--softphone, Urgent!!! |
7:02AM |
0 |
App Queue |
6:54AM |
0 |
DIGI Datafire QuadMicro |
6:52AM |
2 |
Notice with asterisk System application |
6:43AM |
2 |
Bayonne and Asterisk |
6:13AM |
0 |
Asterisk Festival Perl Net::POP3 |
6:03AM |
0 |
Re: DMTF tones when VOIP call comes in |
3:41AM |
1 |
double-dial in SIP Grandstream |
2:26AM |
0 |
SIP silence detection |
1:35AM |
4 |
PBX (Asterisk) <-> Cellular Phone Network |
1:14AM |
2 |
mysql addon |
|
Monday November 17 2003 |
Time | Replies | Subject |
9:44PM |
1 |
IAX2 Problem |
7:56PM |
5 |
Struggling with grandstream sip to asterisk |
4:45PM |
7 |
Updated iaxComm binaries available for WinXP, Red Hat 9.0 |
4:18PM |
0 |
Asterisks and scripting to get an ACD |
4:16PM |
0 |
RE: Asterisk-Users digest, Vol 1 #1918 - 9 msgs |
3:06PM |
3 |
asterisk and Codec G-723 |
3:02PM |
3 |
3Com NBX phones |
2:16PM |
5 |
Asterisk Cdr |
1:53PM |
1 |
SIP calls no longer work |
1:52PM |
0 |
New FAQ on Echo Cancellation |
1:24PM |
2 |
Hunt groups and SIP? |
12:09PM |
3 |
help voicepulse connect |
11:49AM |
1 |
Wifi600 problem |
11:37AM |
2 |
Static Config? |
11:10AM |
0 |
qtelnet product, FXO gateway |
10:59AM |
1 |
problems with alsa (card ac97) in asterisk |
10:58AM |
1 |
Sample proposal |
10:04AM |
9 |
Radius on * |
9:50AM |
0 |
Transfer directly to voicemail - Solved |
9:32AM |
1 |
mpg123 core when stopping asterisk |
9:22AM |
1 |
IAX2 and MWI |
9:04AM |
0 |
GSM or WAV files for musiconhold? |
8:29AM |
4 |
Transfer directly to voicemail? |
8:12AM |
1 |
iconnecthere incoming |
7:34AM |
2 |
VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk |
7:14AM |
0 |
SIP soft phone registration |
5:38AM |
2 |
IAX2 connectivity problem (qualify=yes) |
5:38AM |
1 |
Voip providers U.S (eastern) ?? |
5:09AM |
2 |
voicepulse working? |
3:58AM |
8 |
DTMF |
3:10AM |
0 |
Posting the data of user agent from extensions.conf to MySql |
3:08AM |
0 |
asterisk iEnsemble H323 communication |
2:51AM |
0 |
mgcp audiocodes mp200 |
2:40AM |
0 |
receive fax with isdn |
1:02AM |
1 |
ISDN debugging and SIP dial-in issue] |
12:57AM |
1 |
Updated Asterisk-NL |
|
Sunday November 16 2003 |
Time | Replies | Subject |
10:01PM |
1 |
wireless |
9:57PM |
5 |
Distinctive Ring |
9:33PM |
3 |
asterisk installation error |
9:16PM |
1 |
strange Music on Hold between SNOM, Grandstream and Asterisk |
6:13PM |
3 |
FXO Cards in Australia |
4:13PM |
0 |
Echo/fault isolation test gear |
3:37PM |
0 |
Enhanced VoiceMail Patch... (vm2) |
3:25PM |
1 |
Message lamp integration with legacy pbx during conversion |
2:07PM |
1 |
Attempting to contact John Brown |
12:32PM |
2 |
two X100P cards, different context |
10:24AM |
0 |
echo probs |
9:56AM |
0 |
Consultant? List yourself on the Asterisk Wiki |
9:36AM |
0 |
Incoming calls randomly hangup and blank calls |
7:53AM |
0 |
[schaefer: Re: ISDN debugging and SIP dial-in issue] |
7:11AM |
0 |
* is crashing, when the call is accepted (H.323 -> SIP) |
5:43AM |
2 |
wcfxo installation error |
5:42AM |
0 |
wcfxo installatio n error |
|
Saturday November 15 2003 |
Time | Replies | Subject |
7:54PM |
0 |
Problem with call pickup -or- what stupid mistake have I made? |
3:16PM |
0 |
asterisk instant hosting --- |
1:11PM |
2 |
Internal server error - cannot align media streams - help needed |
10:33AM |
3 |
Manual Rindown |
8:39AM |
0 |
OSS/dsp sound problem |
7:59AM |
10 |
MeetMe problem |
4:19AM |
2 |
ISDN debugging and SIP dial-in issue |
2:14AM |
3 |
Problem with the Internet LineJACK ISA card... |
|
Friday November 14 2003 |
Time | Replies | Subject |
9:50PM |
1 |
Snom 200, asterisk, MOH and Transfers |
9:49PM |
4 |
MWI and SNOM 200 |
7:34PM |
3 |
Fax over SIP alaw/ulaw |
6:00PM |
0 |
Hidden bug in *8 call pickup with Sip |
3:56PM |
7 |
Cisco 7920 |
3:23PM |
2 |
mpg123 causing Asterisk Freeze? |
2:58PM |
0 |
CallerID when transfering |
1:47PM |
3 |
chan_zap won't load after CVS update |
1:00PM |
1 |
Potential call logging problem for commercial systems.. |
12:55PM |
0 |
Re: 9. Zhone zplex (Angel Gomez Garcia) |
12:42PM |
1 |
Calls drop after 10 seconds |
12:22PM |
0 |
PRI E100P |
11:48AM |
2 |
Confused about Asterisk server with regards to Linux NAT Firewall |
11:19AM |
1 |
How To Transfer From IAX Clients |
11:18AM |
2 |
Streaming channels from Asterisk to the Internet |
10:39AM |
1 |
Re: 9. Zhone zplex (Angel Gomez Garcia) |
10:38AM |
0 |
SIP Intercom & Paging (was Overhead Paging) |
9:48AM |
1 |
dtmfmode & SIPDtmfMode |
9:47AM |
1 |
For Sale - (10) Dialogic D/240SC-T1 REV2 |
8:30AM |
1 |
Looking for recommendations for home office setups |
7:07AM |
0 |
Asterisk and Ser - NEWBIE |
4:13AM |
7 |
Your thoughts.. |
4:05AM |
0 |
RE: Aculab SS7/ISUP |
3:33AM |
0 |
SIP channel mixup |
|
Thursday November 13 2003 |
Time | Replies | Subject |
10:36PM |
3 |
iax configuration |
8:03PM |
1 |
(no subject) |
3:50PM |
1 |
RE: Aculab SS7/ISUP (new subject) |
3:33PM |
6 |
Overhead Paging |
3:09PM |
2 |
Feature request |
2:44PM |
1 |
asterisk solution. |
2:35PM |
0 |
Needed: Digium/Asterisk Resller in Riverside, CA Area |
2:31PM |
0 |
Background only responds to 1 digit |
2:07PM |
0 |
Indications - ring signals etc |
1:23PM |
1 |
Errors in build. |
1:21PM |
0 |
Cisco 2610 as an MGCP gateway |
1:02PM |
0 |
TDM40B > X100P Call Waiting |
12:11PM |
2 |
EU SIP Phone providers |
12:10PM |
0 |
AGI verbose command |
12:05PM |
1 |
box for asterisk |
10:09AM |
2 |
IAX trunk monitoring |
9:49AM |
2 |
Cisco 7910 |
9:10AM |
0 |
Problem in transfer the records of user id to the MySql database |
8:58AM |
1 |
CVS repository changes |
8:33AM |
3 |
multi call iconenct? |
6:59AM |
1 |
how to interconnect gnugk and asterisk? |
6:53AM |
0 |
2 AGI questions.. |
6:41AM |
10 |
Graphical Interface |
6:31AM |
1 |
Exit the Directory Application? |
5:52AM |
3 |
Network Voip Carrier Termination (Off Topic) |
5:38AM |
0 |
New User - Auto Attend / IVR |
5:35AM |
0 |
recommendation? |
5:20AM |
3 |
Limit timeout of outgoing calls?? |
5:07AM |
6 |
I hate to do this but.. |
4:31AM |
2 |
Couple of Questions for Australian Users! |
4:19AM |
2 |
Assignement of extension to Netmeeting with dynamic IP address |
4:10AM |
1 |
IAX2 based software client ..pls help |
2:42AM |
2 |
Open Source Linux PBX! |
|
Wednesday November 12 2003 |
Time | Replies | Subject |
10:06PM |
1 |
(HI,new to asterisk)connecting asterisk to telephonyhardware |
8:33PM |
1 |
SPA 2000 and 404 not found |
8:12PM |
1 |
ADSI Functions |
6:44PM |
1 |
IAX channel and transfering calls |
6:36PM |
1 |
pause after dialed option |
6:28PM |
1 |
vm email notifications |
6:05PM |
5 |
Distintive Ring on x100p |
5:42PM |
0 |
asterisk cvs ebuilds for gentoo portage system |
4:22PM |
0 |
H.323 strange error |
2:46PM |
1 |
Zap timeout not occurring |
2:32PM |
1 |
"488 not acceptable here" message |
1:55PM |
0 |
Sipura / Handytone / Cisco |
12:59PM |
7 |
SoftFax question |
12:41PM |
2 |
Canadian VoIP termination? |
12:40PM |
1 |
D Channel Bonding |
12:38PM |
1 |
Zultys. |
11:48AM |
1 |
No outgoing audio |
10:43AM |
0 |
Echo sometimes with TDM40B / X100P only |
10:37AM |
1 |
X100P random hangups. |
10:06AM |
3 |
Dial Plan Sequencing |
8:41AM |
2 |
Media Negotiation Failed |
8:36AM |
1 |
TAPI development |
8:07AM |
3 |
DIAX 0.93 with some sound improvements and not only... |
7:31AM |
1 |
IAX needs a zaptel device? |
1:58AM |
1 |
MySQL Licence may be changing.. |
1:29AM |
8 |
FreeBSD |
12:07AM |
0 |
sending MWI to a none local client |
|
Tuesday November 11 2003 |
Time | Replies | Subject |
8:22PM |
0 |
DISA questions ? |
8:09PM |
2 |
sip: 401 unauthorized with xlite |
5:00PM |
0 |
(no subject) |
2:37PM |
0 |
xlite- wierd behavior |
12:57PM |
0 |
Codecs and call failure with Grandstream |
12:44PM |
0 |
Help with include files & current CVS |
12:08PM |
3 |
pick up ringing exten |
12:07PM |
2 |
FWD codecs? |
11:49AM |
1 |
Zombie lines. |
11:28AM |
4 |
OT: Document Control System? |
9:41AM |
1 |
zap show channels -> No such command.... |
9:23AM |
3 |
dialing 8 in VM2 causes channel lockup? |
9:04AM |
0 |
VM, IVR, UM |
8:31AM |
4 |
"NO ANSWER" X100P |
6:33AM |
5 |
iaxtel down? |
4:55AM |
0 |
AGI: "set autohangup" |
3:07AM |
1 |
Unable to use voicemail |
2:56AM |
4 |
Registering an application |
2:11AM |
1 |
Call indicators for Brazil and other countries |
|
Monday November 10 2003 |
Time | Replies | Subject |
10:48PM |
3 |
AGI and PHP |
4:42PM |
0 |
Asterisk in the NW |
4:39PM |
2 |
multiple locations |
3:45PM |
1 |
ISDN (isdn4linux) DDI |
2:33PM |
0 |
OT - (Cisco 79xx) SIP ver 6.0?? |
2:28PM |
1 |
Menu's & Sub-Menu's |
1:34PM |
0 |
INTRACOM SIP PHONE |
1:06PM |
1 |
IAX/IAX2 encryption? |
12:55PM |
3 |
Inter-digit minimum |
12:43PM |
0 |
Seeking proposals for large county library voice system |
12:36PM |
1 |
Periodic crash - avoid this syntax... |
12:24PM |
3 |
Asterisk and Polycom Soundpoint IP600 |
11:26AM |
0 |
SIP and Goto failures? |
9:41AM |
0 |
H.323 - SIP Gateway. |
9:05AM |
0 |
Asterisk in Dutch |
8:14AM |
1 |
Jitter Buffer on chan_sip |
7:33AM |
0 |
Cisco router/SIP gateway registration |
6:59AM |
2 |
ISDN TBCT.... |
6:34AM |
4 |
Fedora Core 1 |
6:18AM |
4 |
Asterisk timing |
5:56AM |
2 |
Edit GSM audio files |
5:49AM |
2 |
OFF: Newsgroup gtw |
4:15AM |
0 |
cisco 7960 intercom |
2:56AM |
5 |
OT : For the SQL gurus.. |
12:45AM |
1 |
Problem in MySql-3.23.49 |
|
Sunday November 9 2003 |
Time | Replies | Subject |
8:15PM |
0 |
RX volume |
5:53PM |
1 |
Dialing 800 numbers through FWD or SIPphone? |
5:38PM |
0 |
G.729 confusion |
5:30PM |
0 |
channel bank and sip phone |
2:38PM |
4 |
Multi phone presentation |
1:01PM |
1 |
Iax2 channel usage |
12:10PM |
1 |
chan_capi & Eicon Diva problem |
10:53AM |
10 |
DIAX version 0.9.2 available for download |
8:00AM |
1 |
vertical service codes (US standard) |
7:00AM |
3 |
Text entry by DTMF |
3:21AM |
3 |
unable to find path |
|
Saturday November 8 2003 |
Time | Replies | Subject |
7:46PM |
0 |
hum on z-plex 10 |
6:54PM |
5 |
Eicon Diva Server 4BRI |
6:32PM |
4 |
SIP, Sipura SPA-2000, and Voicemail2 |
3:40PM |
0 |
contact |
2:08PM |
5 |
Call Rate in CDR |
10:27AM |
2 |
Snom200 MWI.. |
|
Friday November 7 2003 |
Time | Replies | Subject |
10:48PM |
4 |
IBM to Run VoIP On Linux |
10:28PM |
0 |
[Asterisk-Dev] Help with Conference |
9:28PM |
0 |
Re: Asterisk-Users digest, Vol 1 #1835 - 12 msgs |
8:28PM |
0 |
Sipura SPA-2000 and Asterisk |
6:05PM |
2 |
Softswitch |
5:02PM |
1 |
Snom 200 Do Not Disturb ?/ |
4:53PM |
6 |
Streaming MOH |
4:12PM |
8 |
Putting call on hold |
3:30PM |
1 |
diax request |
3:29PM |
0 |
++Newbie Question |
2:25PM |
0 |
sipdtmfmode problem |
1:38PM |
1 |
Asterisk can't connect voice |
1:30PM |
0 |
Cisco 6.0 gripes |
1:25PM |
1 |
No communication channel |
12:55PM |
3 |
AGI dialing?? |
10:57AM |
2 |
Differents config files |
10:43AM |
2 |
Modem as a FXO |
10:38AM |
0 |
DTA310 Problems |
10:00AM |
7 |
CDR fields |
9:07AM |
1 |
need Dutch VoIP provider |
8:37AM |
3 |
music on hold (SIP Clients) |
8:17AM |
6 |
SIP protocol bug ??? |
6:48AM |
0 |
RE: msgs archives gsm of asterisk ??? Asterisk-Users digest, Vol 1 #1809 - 16 msgs |
6:41AM |
0 |
RE: Asterisk-Users digest, Vol 1 #1808 - 13 msgs archives gsm of asterisk ??? |
6:40AM |
5 |
grandstream ntp |
6:33AM |
2 |
No ringing tone |
6:24AM |
2 |
MGCP - Repost |
6:22AM |
21 |
Snom 200 |
5:43AM |
3 |
Unable dial out with the new Oh323 0.5.6 |
5:18AM |
0 |
ASTERISK FESTIVAL E-MAIL |
2:55AM |
0 |
Pulsating, Choppy sound using GS.. |
1:30AM |
2 |
Callgroups and Pickupgroups in Console/dsp |
1:09AM |
0 |
fix isn't |
12:24AM |
0 |
Possible fix for grandstream outgoing |
|
Thursday November 6 2003 |
Time | Replies | Subject |
11:20PM |
2 |
this is the code that breaks outgoing calls on grandstream |
9:27PM |
1 |
Strange Problem with Asterisk.... |
8:41PM |
6 |
IAX2 trunking on one side only. |
8:33PM |
0 |
Manager API - howto |
8:02PM |
6 |
5 Channel / Trunk ?? |
4:45PM |
2 |
Dialing an outside number -- QUESTION -- |
3:59PM |
0 |
H323 Gateways |
3:40PM |
40 |
voicemail |
3:22PM |
6 |
Error in Incoming SIP call |
3:05PM |
2 |
Voicemail2 attach question? |
2:35PM |
1 |
Need testers for new STUN build system |
2:32PM |
1 |
Huge SIP traffic!! |
2:08PM |
1 |
asterisk + dual phone lines + cisco + backup |
1:46PM |
0 |
Outgoing calls to SIP provider |
1:13PM |
1 |
processor limits of max concurrent users on a single system |
12:34PM |
1 |
Gnophone URL |
12:29PM |
3 |
Grandstream problem |
12:23PM |
1 |
Can NOT access CVS??? |
11:38AM |
1 |
chan_sip and budgetone |
11:06AM |
0 |
which codec will be used ? |
10:33AM |
2 |
configuring DID trunks |
10:32AM |
5 |
FW: recording calls |
10:29AM |
0 |
ISDN PBX + IVR + Voicemail Configuration - S anity Check ... |
10:13AM |
2 |
To SIP or Not? |
9:11AM |
1 |
CVS compile problem on asterisk |
7:58AM |
2 |
ISDN PBX + IVR + Voicemail Configuration - Sanity Check ... |
7:08AM |
0 |
RE: USB handsets/headsets?? (WipeOut) |
5:22AM |
11 |
USB handsets/headsets?? |
5:10AM |
0 |
Voicemail RFC |
4:49AM |
2 |
Asterisk and SIP Proxy on same machine? |
4:46AM |
0 |
Festiavl Access rejected |
4:16AM |
5 |
MtSQL CDR logging |
2:54AM |
3 |
which channel format number is right? |
2:37AM |
2 |
6.0 image for Cisco 7960's? |
12:44AM |
2 |
Voicemail2 vs Voicemail |
12:09AM |
0 |
SIP nat not working with budgetone (long) |
|
Wednesday November 5 2003 |
Time | Replies | Subject |
11:22PM |
1 |
Manager Server |
11:19PM |
1 |
IAX/SIP Client |
10:43PM |
1 |
Beginners help |
9:49PM |
2 |
Ping AGI Demo |
7:55PM |
1 |
X100P modify DTMF tone length |
6:42PM |
1 |
Asterix - Digium replacing our current key system |
6:28PM |
2 |
asterisk nightmare from hell! |
6:21PM |
2 |
Multi-port FXO anyone? |
6:20PM |
1 |
7960 Directory, WAS: Anyone using * in a live production environment? |
5:57PM |
1 |
X100P + ADSI |
4:14PM |
6 |
recording calls |
4:11PM |
2 |
my first asterisk install |
4:04PM |
1 |
iconnect |
3:12PM |
2 |
Hold, park, transfer, etc-- How is it done? |
3:00PM |
0 |
Asterisk as a Media Gateway Controller |
2:51PM |
2 |
spawn extension (inbound , h, 1) exited non-zero |
2:31PM |
1 |
Authenticating zap callers. (callerid or PIN code) |
2:03PM |
5 |
The Minimum Cost of Setting up an Asterisk Phone System? |
1:40PM |
2 |
archives gsm of asterisk ??? |
1:38PM |
1 |
To anyone with a grandstream budgetone... |
12:25PM |
12 |
Mediatrix 1204 |
12:22PM |
0 |
What the Installation Instructions SHOULD HAVE SAID.. |
12:15PM |
1 |
A real-life production scenario |
11:34AM |
4 |
i4l-modem dtmf detection |
11:24AM |
6 |
Skinny (SCCP) help |
11:11AM |
4 |
error compiling asterisk |
11:06AM |
1 |
Web Interface for adding new users |
11:03AM |
1 |
Error in app_voicemail2.so after CVS update |
9:57AM |
3 |
New Phone Review: Clipcomm 101 |
9:29AM |
1 |
Remote Call Pickup |
9:17AM |
3 |
Apple implementation |
8:42AM |
0 |
Can't connect voice in Zplex 10B |
8:08AM |
10 |
Reasons why I shouldn't use Asterisk? |
8:01AM |
1 |
Using Asterisk as a VOIP gateway |
7:47AM |
2 |
Need info on Gastman/Astman |
7:28AM |
1 |
asterisk-oh323: New version 0.5.6 |
6:14AM |
0 |
SIP with CIC |
5:49AM |
0 |
Agent Logoff question |
5:36AM |
2 |
Best place to order Cisco ATA 186 |
5:19AM |
0 |
Missed calls/activity log in Asterisk |
4:38AM |
1 |
First AGI help.. |
4:14AM |
1 |
g.729 codec registration |
4:12AM |
0 |
SIP broken for budgtone. |
2:20AM |
1 |
SIP and NAT: try, try again. |
1:05AM |
0 |
Client Dev - Newbie questions |
12:50AM |
1 |
Outband DTMF on i4l modem |
12:18AM |
0 |
DIAX users |
|
Tuesday November 4 2003 |
Time | Replies | Subject |
11:38PM |
1 |
asterisk bandwidth management |
10:18PM |
1 |
http://www.skype.com/ |
8:06PM |
1 |
Demo Weather Report AGI v2.0 |
7:59PM |
2 |
IAX clients and the flash button |
2:51PM |
3 |
Asterisk system lock |
2:28PM |
2 |
snatching calls |
2:15PM |
1 |
A little bit of success |
1:11PM |
0 |
Asterisk > Asterisk > PSTN |
12:33PM |
1 |
Best or any VoIP provider that works with *? |
11:48AM |
1 |
asterisk and zplex10b (fwd) |
11:34AM |
1 |
More ringing time on incoming lines |
11:06AM |
1 |
Fw: problem zplex 10 B |
10:43AM |
0 |
Limit of conference rooms. |
10:42AM |
2 |
Does externalip= do anything to help with SIP behind a Linux based NAT router? |
10:28AM |
1 |
Transferring to Meetme |
10:21AM |
0 |
why does context order make a difference in IAX.CONF |
10:02AM |
0 |
Need Help with SIP/H323. |
8:41AM |
1 |
multitech. |
8:39AM |
1 |
Call Transfert with SwissVoice IP10S in MGCP mode |
8:33AM |
1 |
call processing after a PIN |
6:12AM |
3 |
*, Fritz!PCI and strange behavior |
5:31AM |
0 |
Knowledge Sharing on Asterisks issue. |
4:33AM |
2 |
high system load running asterisk |
4:02AM |
0 |
Compil error with Mandrake pwlib |
4:00AM |
1 |
Alert extensions without answering incoming call? |
4:00AM |
0 |
ipphone voicemail problems |
3:39AM |
0 |
Problem in running .gsm file |
3:18AM |
1 |
Does anyone provide inbound UK numbers using IAX ? |
2:48AM |
8 |
Anyone using * in a live production environment? |
2:44AM |
1 |
Flash hook -> SIP device |
2:37AM |
2 |
asterisk does not hang up |
|
Monday November 3 2003 |
Time | Replies | Subject |
8:39PM |
1 |
4 X100P's, 4 7960's Same Box? |
7:30PM |
0 |
(no subject) |
6:01PM |
2 |
ADSI - PowerTouch 350 |
5:15PM |
2 |
Transfer from Grandstream BT100? |
5:08PM |
0 |
Fwd: RE: Asterisk behind LinkSys NAT Routing |
4:43PM |
1 |
isdn, modem, etc. |
4:18PM |
1 |
new voicemail notification by calling #? |
3:15PM |
2 |
Actiontec's Internet Phone Wizard and Digium's S100U |
3:07PM |
0 |
csv phone log |
2:46PM |
5 |
Red Alarm |
2:26PM |
0 |
Re:Looking for CTI/IVR/CallCenter/VoIP project/task as freelance developer |
2:15PM |
4 |
Call waiting on X100P |
2:02PM |
0 |
LGPL IAX2 software phone (for WIndows/Linux platforms) |
1:17PM |
2 |
MWI - I know this has been discussed in depth already |
12:43PM |
0 |
turn off dial tone on a TDM400p channel |
12:37PM |
0 |
NOTICE[16401]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 72 received |
12:17PM |
3 |
DIAX Soft phone v0.9.1 is available for downlaod... |
11:57AM |
5 |
Rollout tips |
11:41AM |
0 |
skipping/jitter in Music on Hold |
11:30AM |
0 |
High Availability and Mass Deployment for Asterisk |
11:13AM |
1 |
Proper syntax for the "Cut" application? |
11:06AM |
1 |
Intel Performance Primitives |
10:32AM |
1 |
<--PRI--> * <--PRI--> modem bank - problems |
10:31AM |
0 |
Subject: Re: Where can i get the g.723 codec? |
10:29AM |
1 |
one way sound with x-lite (sip) -3rd attempt ! |
9:13AM |
0 |
OHT in fxs hates my answering machine + self fix |
9:07AM |
1 |
Aastra 480 ADSI keypad problem |
8:07AM |
1 |
/var/spool/asterisk/outgoing |
7:28AM |
10 |
Where can i get the g.723 codec? |
7:02AM |
1 |
Asterisk compliance with RFC 2617 (qop, nc and cnonce) - in relation to sipcall.co.uk |
6:45AM |
0 |
Gnophone problem |
6:07AM |
9 |
IAX hardphones? anyone? |
3:48AM |
2 |
IAX2 Java library (was Re: New IAX software phone (for WIndows platform)) |
|
Sunday November 2 2003 |
Time | Replies | Subject |
11:20PM |
2 |
Questions from a total beginner |
7:22PM |
1 |
* troubles |
6:11PM |
2 |
Read error on sound device |
5:56PM |
2 |
Recommended places for beginner to start? |
5:14PM |
6 |
Asterisk behind LinkSys NAT Routing |
5:08PM |
0 |
surpress dial tone on TDM400p |
3:35PM |
1 |
Live real extensions.conf samples? |
3:19PM |
2 |
Clearing Queue Stats? |
2:42PM |
3 |
PHP Manager examples |
1:59PM |
3 |
Fw: a bit frightened, guys |
1:21PM |
17 |
New IAX software phone (for WIndows platform) |
12:53PM |
2 |
Threeway calling leaves outside trunks bridged |
12:36PM |
1 |
a bit frightened, guys |
7:44AM |
2 |
one way sound with x-lite (sip) -second attempt |
7:15AM |
0 |
Trustix support.. |
6:43AM |
2 |
Good system board to use with TE410P? |
6:04AM |
1 |
FW: NAT router and off-premise SIP audio problem |
6:00AM |
3 |
recording files for menues |
|
Saturday November 1 2003 |
Time | Replies | Subject |
9:55PM |
1 |
NetJet Cards |
8:59PM |
1 |
which TDM to use? DID line from telco with no dial tone and no voltage |
6:15PM |
13 |
Quick Question |
4:17PM |
2 |
broadcast voicemail msg ?? |
12:15PM |
2 |
Making a Skinny phone talk to Asterisk |
12:08PM |
0 |
sizing - conference room |
11:08AM |
3 |
(no subject) |
8:32AM |
0 |
iax vs iax2 connections |
6:39AM |
4 |
NAT router and off-premise SIP audio problem |
1:11AM |
3 |
FXO modules for TDM400P? |
12:40AM |
0 |
Directory App Weirdness |