| Thursday July 31 2003 |
| Time | Replies | Subject |
|---|
| 10:08PM |
1 |
24port or higher fxs |
| 9:48PM |
1 |
PHP API for Manager - Plaintext auth needed? |
| 6:40PM |
0 |
one way audio h323 callmanager |
| 5:56PM |
1 |
Best Analog sets for use w/* |
| 5:12PM |
3 |
Mutex problem in sip? |
| 3:45PM |
3 |
Queue and Agents in CVS |
| 1:21PM |
1 |
Zaptel cards, working FXS and SIP, no audio? |
| 12:29PM |
1 |
retrieving dialed number when overlap dialing? |
| 10:53AM |
1 |
AddQueueMember and RemoveQueueMember |
| 9:56AM |
0 |
Re: Asterisk-Users digest, Vol 1 #944 - 3 msgs |
| 9:23AM |
4 |
'System' application exit with error even if it performs the job as expected |
| 8:32AM |
4 |
SIP calls cause Segmentation Fault |
| 7:14AM |
0 |
Newbie - Looking for pointers |
| 6:32AM |
1 |
(no subject) |
| 5:31AM |
1 |
Sound Quality. |
| 5:28AM |
1 |
Parking calls - why doesn't work? |
| 5:25AM |
1 |
Manager |
| 4:31AM |
2 |
RFC2833 problems with X-Lite |
| 1:46AM |
1 |
RTP codec 13 received - Cisco incompatibilit y? |
| 1:19AM |
1 |
Help with ON-Hold, and call-transfer. |
| 1:08AM |
1 |
RTP codec 13 received - Cisco incompatibility? |
| 1:05AM |
1 |
Congestion |
| |
| Wednesday July 30 2003 |
| Time | Replies | Subject |
|---|
| 7:59PM |
3 |
Manager.pm port |
| 4:17PM |
4 |
Grandstream Budgettone 100 & 102 |
| 4:07PM |
4 |
SCO/Linux concerns |
| 1:55PM |
0 |
X100P and incoming Context + CDR? |
| 1:08PM |
7 |
sip -> h323 -> ptsn |
| 12:09PM |
2 |
MGCP behind NAT |
| 11:41AM |
1 |
voicemail file access problems |
| 9:23AM |
2 |
X-Lite and Call transfer using Asterisk |
| 9:23AM |
6 |
CVS Problem? |
| 9:06AM |
0 |
asterisk,ata186 and Panasonic TD1232 |
| 8:55AM |
0 |
rxgain and txgain in zapata.conf |
| 8:00AM |
16 |
Need help |
| 7:40AM |
2 |
ADSI and SoftKeys |
| 7:40AM |
2 |
Some stats |
| 7:15AM |
1 |
isdn4linux/Teles16.3 |
| 6:55AM |
5 |
Dummy account/extension |
| 6:50AM |
0 |
X100P call detection |
| 6:46AM |
0 |
Voicetronix Hardware |
| 3:31AM |
0 |
ISDN Random Hangup Problems |
| 2:55AM |
1 |
SetCIDName |
| 2:37AM |
5 |
chan_sip.c problems problems from cvs 1.134 |
| 2:16AM |
2 |
Call Transfer, Budgettone 100 |
| 1:29AM |
1 |
Voicemail message forwarded to another extension and file format changing |
| |
| Tuesday July 29 2003 |
| Time | Replies | Subject |
|---|
| 5:51PM |
1 |
RE Pingtel Phones |
| 5:27PM |
1 |
Variable Substitution |
| 5:01PM |
1 |
Asterisk Developer's Kit (TDM) help |
| 2:43PM |
0 |
memory leak in voicemail.c |
| 2:28PM |
0 |
dialogic drivers |
| 1:43PM |
2 |
CAPI & CLID |
| 12:44PM |
0 |
IRQ Misses? |
| 10:55AM |
10 |
Asterisk installation |
| 8:23AM |
0 |
stutter tone for voicemail on SIP |
| 6:58AM |
0 |
7960 SIP problem when calling from outside o f LAN |
| 6:01AM |
1 |
Linux flavor? |
| 5:20AM |
2 |
[Solved] CAPI with hanging channels |
| 5:03AM |
1 |
Call Dropping |
| 2:24AM |
3 |
stupid questions .. |
| 1:59AM |
1 |
7960 SIP problem when calling from outside of LAN |
| 1:05AM |
0 |
Contact header empty in SIP-message |
| |
| Monday July 28 2003 |
| Time | Replies | Subject |
|---|
| 10:19PM |
1 |
Call Forwarding and DND conf |
| 6:50PM |
1 |
iax2 and reinvites |
| 6:34PM |
0 |
Welltech FXS SIP registering with Asterisk |
| 5:05PM |
0 |
Hunt group examples? |
| 2:16PM |
5 |
VoiceMail2 Wish List |
| 2:15PM |
0 |
Re: Asterisk-Users digest, Vol 1 #882 - 11 msgs |
| 1:18PM |
1 |
Following completion when Dialing. |
| 12:37PM |
1 |
D-link 102s and g723 parameters |
| 12:17PM |
4 |
Offering an Asterisk Documentation and FAQ Portal |
| 9:03AM |
2 |
Asterisk user guide .. |
| 8:58AM |
2 |
Call transfer on ATA186 |
| 7:43AM |
0 |
Hardware support for TDM |
| 7:31AM |
0 |
Call transfer between two phones on the same ATA |
| 7:29AM |
8 |
RTP session traversing Asterisk server ... |
| 6:00AM |
1 |
Problems with two B channels |
| 4:27AM |
2 |
"immediate=yes or Compleate recieved" with intcoming calls with new CVS |
| 4:17AM |
0 |
Loop Drop on vpb/1-7 |
| 2:49AM |
4 |
Zaptel |
| 12:08AM |
1 |
go on in current context after destination channels hung up ? |
| |
| Sunday July 27 2003 |
| Time | Replies | Subject |
|---|
| 2:48PM |
1 |
Channel Language |
| 12:25PM |
4 |
ISDN Fritz & RedHat 8.0 |
| 11:31AM |
0 |
Festival talks fast... |
| 11:29AM |
1 |
Ordering digital trunks? |
| 6:31AM |
3 |
Australian Options |
| 5:52AM |
1 |
* behind ISDN pbx - Forwarding to extensions with in primary pbx |
| 5:48AM |
1 |
FWD-gateway prefix |
| 1:32AM |
3 |
Nortel 350 |
| 12:28AM |
20 |
g729 Codec |
| |
| Saturday July 26 2003 |
| Time | Replies | Subject |
|---|
| 10:16PM |
1 |
TE410P startup |
| 8:39PM |
2 |
moh/playback for non-zap interfaces |
| 4:37PM |
1 |
can't compile asterisk |
| 11:48AM |
0 |
Bug Tracker Official Launch |
| 8:47AM |
0 |
Problem with AGI "Record File" |
| 8:46AM |
1 |
Asterisk SIP + Grandstream 100 phone |
| 8:27AM |
0 |
ISDN Callout problem |
| 7:47AM |
1 |
PCM Voice Quality Issue on CVS Version |
| 6:53AM |
0 |
meetme room |
| 6:50AM |
0 |
Database usage? |
| 6:30AM |
0 |
app_voicemail2 became a bit silent, lately... |
| 5:30AM |
0 |
Chan_oh323 Dial format / voice latency 4 to 5 secs |
| 5:18AM |
0 |
Problems with chan_sip on multi-homed hosts |
| 12:13AM |
2 |
Direct Indial with ISDN and Netjet-S |
| |
| Friday July 25 2003 |
| Time | Replies | Subject |
|---|
| 6:14PM |
1 |
Busy detect on pri channel? |
| 4:23PM |
7 |
can't get musiconhold to work |
| 1:06PM |
1 |
reconnecting |
| 11:23AM |
0 |
Web conf files |
| 5:09AM |
3 |
chan_capi error |
| 4:34AM |
0 |
7940 & AS5300 codec issues/questions G.729 & G.711 |
| 3:15AM |
1 |
SetLanguage application doesn;t seem to work in latest Asterisk |
| 2:55AM |
2 |
Configuration sample for isdn4linux? |
| 2:50AM |
0 |
go on in context after the destination channel hung up? |
| 2:30AM |
3 |
Dialogic hardware |
| 2:21AM |
1 |
Asterisk /SIP .. nat |
| 12:47AM |
0 |
IAX and Call format |
| |
| Thursday July 24 2003 |
| Time | Replies | Subject |
|---|
| 10:02PM |
2 |
audiocodes fxs |
| 9:46PM |
0 |
INFO: How the T410P sets the number of channels per span |
| 9:09PM |
1 |
Instant hangup on busy Zap channel. |
| 8:38PM |
2 |
Voicemail() problems - Long pause after incoming message recording ended. |
| 6:24PM |
0 |
X100P - FO card |
| 4:17PM |
5 |
Configuration |
| 3:04PM |
0 |
MeetMe hangup |
| 2:50PM |
1 |
time and date stamp in voicemail |
| 1:59PM |
0 |
AVM C4 and external and internal ISDN bus and * |
| 11:44AM |
2 |
Adtran TSU 600E |
| 11:16AM |
2 |
isdn4linux |
| 11:15AM |
4 |
the 'pound' and '#' are the same? |
| 11:01AM |
2 |
Changes to reset method for ATA186? |
| 10:58AM |
0 |
SIP/H.323 Phone with intercom |
| 10:50AM |
2 |
Debian Package asterisk-oh323? |
| 9:37AM |
1 |
Cisco's CallManager and * (was: Cisco 7960g) (fwd) |
| 9:32AM |
1 |
FWD no longer works.. but nothing has changed? Wierd DEBUG errors. |
| 8:40AM |
0 |
IAXTel Connect Problem - Mini Frame |
| 8:16AM |
1 |
voicemail enhancements |
| 8:02AM |
2 |
Cisco ATA Advanced CallerID |
| 6:51AM |
0 |
the 'pound' and '#' are the same? (OT Rambli ng) |
| 5:03AM |
1 |
Asterisk <--> TTS server |
| 2:46AM |
1 |
compilation error |
| |
| Wednesday July 23 2003 |
| Time | Replies | Subject |
|---|
| 10:07PM |
2 |
T410P and zaptel.conf |
| 8:45PM |
1 |
AGI.pm? |
| 8:35PM |
0 |
shared line-appearance |
| 8:29PM |
0 |
Re: [Asterisk] help with extension switching |
| 7:27PM |
3 |
fxs without fxo |
| 5:41PM |
3 |
iaxclient (Activex) |
| 4:39PM |
0 |
Asterisk & X-Lite |
| 3:14PM |
1 |
Cisco 7960 upgrade from SKINNY load |
| 3:04PM |
2 |
executing an agi script after a successful Dial |
| 2:55PM |
2 |
h323 gateway call lost after 74sec always |
| 2:41PM |
0 |
how to start |
| 2:14PM |
3 |
how do I do s extensions with PRI |
| 11:07AM |
4 |
Problems with g729 |
| 10:35AM |
0 |
asterisk-oh323 v0.5.4 |
| 9:51AM |
4 |
h323 and oh323 modules |
| 9:34AM |
5 |
Asterisk as a stand alone voice mail server |
| 8:11AM |
1 |
newbie - simple dialout server |
| 6:41AM |
0 |
Integrate Asterisk with Meridian phone syste m |
| 6:22AM |
0 |
Self-testing E1 cards? |
| 5:53AM |
2 |
Integrate Asterisk with Meridian phone system |
| 4:55AM |
3 |
2 B channels for ISDN cards |
| 4:55AM |
3 |
Asterisk IVR and Hicom 300 |
| 3:16AM |
1 |
Newbie Help |
| 1:26AM |
2 |
SIP info |
| |
| Tuesday July 22 2003 |
| Time | Replies | Subject |
|---|
| 11:56PM |
4 |
Codecs for use with Cisco 7960 and ATA-186 |
| 11:49PM |
2 |
Cisco 802.11b VoIP phone? |
| 8:09PM |
2 |
QoS for Asterisk |
| 4:37PM |
2 |
Supplementing Current phone system |
| 3:58PM |
0 |
Delays with g729 and SIP. How come? |
| 3:46PM |
2 |
No callerid on outgoing call over chan_h323 |
| 2:58PM |
1 |
chan_capi and poor voice quality |
| 1:19PM |
0 |
new voicemail messages |
| 1:11PM |
3 |
busydetect and random hangups |
| 1:00PM |
0 |
IAX / MeetMe problem |
| 12:24PM |
3 |
Ideal Prompt Recording Setup? |
| 12:02PM |
4 |
extensions |
| 11:44AM |
3 |
capi_chan error - CAPI not loaded. |
| 11:17AM |
0 |
TDM400P card only |
| 9:16AM |
0 |
* as a softswitch for pri interfaces |
| 7:51AM |
2 |
enabling dtmf detection on zap channel? |
| 6:55AM |
0 |
Verizon, SBC local company? |
| 6:41AM |
1 |
*--IAX--* problems. (chan_capi problem) |
| 5:55AM |
0 |
*--IAX--* problems. |
| 5:14AM |
2 |
~3 seconds of silence when picking up a call |
| 3:17AM |
2 |
interfacing asterisk with a legacy PBX |
| 2:03AM |
3 |
SIP Call Forwarding/Transfer support ? |
| 1:45AM |
0 |
ADPCM Codec |
| 1:29AM |
0 |
calls per second with asterisk |
| 1:13AM |
0 |
iConnect Here PCPhone application and Asterisk |
| |
| Monday July 21 2003 |
| Time | Replies | Subject |
|---|
| 9:21PM |
1 |
MYSQL Table Structure |
| 8:28PM |
4 |
Using asterisk for a 911 call center.... |
| 7:17PM |
1 |
PAnasonic And Asterisk |
| 5:46PM |
0 |
Robbed bit signalling debugging |
| 2:58PM |
0 |
RE: Asterisk-Users digest, Vol 1 #873 - 16 msgs |
| 2:14PM |
1 |
Outgoing calls through a calling card |
| 12:14PM |
1 |
URGENT! brandly new Wildcard E400P for sale at $1000 |
| 10:25AM |
4 |
anyone with X100P & Callerid working outside US ? |
| 9:50AM |
3 |
CDR question |
| 9:32AM |
2 |
E911 and asterisk |
| 7:32AM |
1 |
X-Lite Build 1016 |
| 7:19AM |
0 |
7960 / MGCP |
| 6:52AM |
4 |
Dynamically setting up/tearing down extensions |
| 5:43AM |
1 |
UK call termination.. |
| 5:05AM |
0 |
Asterisk -> SIP -> AS5300 signalling missing on connect/clear cal l |
| 3:30AM |
8 |
Best software SIP client |
| 2:22AM |
1 |
Best E1 channel bank? |
| 2:21AM |
4 |
Phones |
| 1:10AM |
3 |
SIP Authentication bug? |
| |
| Sunday July 20 2003 |
| Time | Replies | Subject |
|---|
| 5:50PM |
0 |
Summary of VoIP options for Asterisk and request for more? |
| 5:16PM |
3 |
Music on hold & Read error on sound device |
| 4:54PM |
1 |
DTMF crashes chan_capi |
| 12:25PM |
0 |
No audio in Messenger |
| 4:43AM |
3 |
Self parked but avaliable |
| |
| Saturday July 19 2003 |
| Time | Replies | Subject |
|---|
| 10:18PM |
5 |
Asterisk crashes when trying to load G.729 module. |
| 5:58PM |
0 |
file repository |
| 11:50AM |
2 |
how to set specific codec ? |
| 8:38AM |
0 |
Dlink dg102s and G.729 |
| 8:30AM |
2 |
Analog phone not ringing |
| 4:20AM |
0 |
XS4ALL Gateway now also does FWD |
| 3:29AM |
0 |
Actiontec's InternetPhoneWizard (USB) and Asterisk |
| 3:15AM |
0 |
Call Transfer Anouncement |
| 2:41AM |
0 |
IAX can be used on a different UDP port? |
| |
| Friday July 18 2003 |
| Time | Replies | Subject |
|---|
| 11:49PM |
5 |
Again Asterisk and VMWare - it works now! |
| 10:01PM |
16 |
Call Transfer |
| 6:20PM |
1 |
VoIP in hotels |
| 3:05PM |
8 |
questions |
| 2:29PM |
5 |
cdr_mysql |
| 10:35AM |
2 |
Budgetone and NTP (redux) |
| 9:59AM |
1 |
Grandstream BudgeTone 102 initial experiences |
| 9:24AM |
1 |
Techfone VOIP phone |
| 8:54AM |
8 |
"Best" VoIP provider for Asterisk? |
| 7:40AM |
0 |
FW: Sip codec preferences |
| 7:02AM |
7 |
OT: list format vs newsgroup format |
| 6:36AM |
2 |
H3500CW recommendation |
| 3:29AM |
2 |
Correct syntax to call using IAX and a different UDP port |
| 1:28AM |
0 |
IAX and Speex? |
| |
| Thursday July 17 2003 |
| Time | Replies | Subject |
|---|
| 8:51PM |
2 |
serious dtmf recognition problem. |
| 8:12PM |
0 |
Music while waiting for agent to free. |
| 8:04PM |
7 |
Speex support |
| 6:06PM |
1 |
ATA-186 software upgrade 2.16.1 - notes? |
| 4:23PM |
1 |
queue bug? |
| 3:32PM |
1 |
TE410P startup (2 boards) |
| 3:14PM |
2 |
AGI & Silence detection |
| 2:29PM |
0 |
Cisco interoperability? |
| 2:23PM |
0 |
Example: Writing a click-to-call application using pbx_spool |
| 12:16PM |
2 |
Silly questions due to ingrained knowledge of analog phone use. |
| 11:28AM |
0 |
Video Phones? |
| 10:14AM |
3 |
random hangups |
| 10:06AM |
0 |
Sip call question |
| 9:52AM |
0 |
error "WARNING[28697]: File app_dial.c, Line 304 (wait_for_answer): Unable to forward voice" |
| 8:58AM |
1 |
Can I interoperate with public PSTN gateways ? |
| 8:38AM |
3 |
Any dialing tricks... |
| 8:23AM |
4 |
AVM Fritz! to connect LAN with ISDN line? |
| 7:23AM |
0 |
FW: Echo on incoming calls (PRI->SIP) but not on outgoing (SIP->PRI) (no html crap, sorry) |
| 7:21AM |
0 |
Echo on incoming calls (PRI->SIP) but not on outgoing (SIP->PRI) |
| 7:05AM |
0 |
grandstream sip phone (NTP) |
| 6:52AM |
0 |
Manager/gastman |
| 6:27AM |
7 |
Help Needed |
| 6:17AM |
0 |
UK Gateway |
| 5:57AM |
1 |
outgoing callerid string |
| 5:50AM |
0 |
slightly OT /how to obtain 900 number |
| 5:00AM |
3 |
E1 R2 on Asterisk |
| 2:33AM |
3 |
Asterisk -> AS5300 SIP Interoperability |
| 12:00AM |
2 |
conference problem without zapata interface |
| |
| Wednesday July 16 2003 |
| Time | Replies | Subject |
|---|
| 10:21PM |
8 |
Call Pickup |
| 5:42PM |
0 |
Question on peer to peer config |
| 5:31PM |
1 |
Back-to-back connected boards load test |
| 3:44PM |
3 |
Segmentation fault with chan_oh323 |
| 2:49PM |
1 |
FXS and PBX Integration |
| 12:45PM |
0 |
Cisco 7910 compatibility |
| 11:32AM |
1 |
Problems getting 7960's to play nice with Asterisk |
| 10:20AM |
2 |
Multiple Phones for 1 Extension |
| 8:51AM |
0 |
Analog features over the ATA-186 |
| 8:35AM |
0 |
X100P in Australia (was Asterisk-Users digest, Vol 1 #840 - 13 msgs) |
| 8:31AM |
0 |
Sip codec preferences |
| 7:10AM |
1 |
Vendors for phones |
| 6:44AM |
4 |
grandstream sip phone |
| 4:56AM |
4 |
voicemail instructions |
| 4:52AM |
0 |
Timeout in Call Transfering |
| 4:28AM |
1 |
Cisco 7905G vs ATA186 |
| 3:47AM |
0 |
addmailbox2 (Attached) |
| 12:33AM |
2 |
Cisco 7960g |
| |
| Tuesday July 15 2003 |
| Time | Replies | Subject |
|---|
| 11:23PM |
3 |
Asterisk on Cygwin? |
| 9:31PM |
0 |
user agent with auto-pickup support - is there any? |
| 7:52PM |
1 |
Stream Phone Call: Sound on Consule OSS to Helix server? |
| 6:44PM |
1 |
X100P in Australia |
| 6:41PM |
5 |
Text to Speech - Someone needs to do this |
| 5:02PM |
2 |
G729 quality |
| 10:22AM |
1 |
Phoneserve SIP provider |
| 10:08AM |
1 |
g723.1 voicemail/conference files segfault * |
| 9:10AM |
3 |
Conditional Contexts |
| 9:07AM |
9 |
Poll - Would you pay $30-$50 for high quality speech synthesis? |
| 6:08AM |
1 |
Chan_H323, G729 (minor problem) |
| 3:38AM |
0 |
Budgetone Transfer (The answer) |
| 2:37AM |
1 |
Alphanumerical digits |
| 1:41AM |
2 |
Analog commands |
| |
| Monday July 14 2003 |
| Time | Replies | Subject |
|---|
| 10:20PM |
0 |
ebs mbs p-phone question |
| 10:14PM |
2 |
insmod wcfxo failed ( b8zs, esf, wink start is what I'm trying to do.) |
| 6:35PM |
1 |
VXML? |
| 3:52PM |
1 |
Making Analog Phones Work |
| 2:21PM |
0 |
payload framesize |
| 1:32PM |
1 |
Fwd:[Vocal] Question about Cisco IP hard phones |
| 12:45PM |
2 |
Using 2 PhoneJacks with Asterisk for Data calls. |
| 12:42PM |
3 |
New budgetone firmware |
| 11:48AM |
2 |
Remote Agents |
| 11:38AM |
0 |
Cisco 7960 Transfer Call drop problem |
| 10:49AM |
0 |
Cisco 7960 Transfer & Conference |
| 10:30AM |
2 |
G729 licensing |
| 10:24AM |
2 |
MSN Messenger 4.7 vs 5.0 |
| 10:02AM |
1 |
Odd output from X100P |
| 9:37AM |
3 |
Hardware Vendors |
| 9:19AM |
11 |
Getting started |
| 8:53AM |
0 |
MGCP-H323 interoperability |
| 8:39AM |
0 |
h323 Ringing sound |
| 8:34AM |
3 |
EZ-Install |
| 7:58AM |
0 |
(no subject) |
| 6:58AM |
1 |
asterisk and modem |
| 3:33AM |
1 |
.gsm voice format |
| 3:23AM |
0 |
* with external sip proxy |
| |
| Sunday July 13 2003 |
| Time | Replies | Subject |
|---|
| 11:35PM |
1 |
DTMF control for TDM device? |
| 9:38PM |
1 |
Setting up A TDM400P |
| 9:09PM |
0 |
H323 & Transfer |
| 2:57PM |
5 |
unresolved symbols in /lib/modules/2.4.18/misc/zaptel.o |
| 11:16AM |
2 |
Line Override Device |
| 10:07AM |
1 |
Question #3 |
| 8:05AM |
1 |
something is wrong with gsm prompts format |
| 6:24AM |
1 |
AUSTEL Certified |
| |
| Saturday July 12 2003 |
| Time | Replies | Subject |
|---|
| 8:57PM |
2 |
Question About VOIP |
| 4:53PM |
2 |
Phone System Questions |
| 3:21PM |
2 |
Asterisk takes over |
| 1:42PM |
1 |
AGI script sample using bash shell script |
| 12:42PM |
0 |
Any Carrier Accessbank I users |
| 8:01AM |
2 |
VIP 30 phone |
| 8:00AM |
2 |
New Member |
| 12:05AM |
0 |
what is wrong with gsm files |
| |
| Friday July 11 2003 |
| Time | Replies | Subject |
|---|
| 9:12PM |
2 |
Hook Flash INFO messages |
| 8:42PM |
1 |
SIP immediate hangups with latest CVS |
| 8:09PM |
2 |
Weird experience with MOH |
| 7:52PM |
3 |
What does "callerid=" in sip.conf do? |
| 7:13PM |
1 |
audio pause/delay problems |
| 5:07PM |
1 |
No Sound via Sip Phone |
| 12:56PM |
4 |
module : cdr_sybase.so |
| 12:44PM |
1 |
Configuring BudgeTone and ringer over TFTP |
| 12:28PM |
1 |
SIP call from one extention to another |
| 11:57AM |
0 |
Sip: problem authenticating (with Cisco VoIP IOS 12.x) [long] |
| 8:52AM |
0 |
Client Call Management Application? |
| 7:50AM |
0 |
[Q]: Dialin problems over E1 on a Digium E100P |
| 6:34AM |
1 |
Unable to find IP address??? |
| 5:28AM |
3 |
mgcp problems |
| 4:47AM |
8 |
G729 codec problems |
| 4:22AM |
1 |
mod tor2 takes 20-30% from CPU (20-30% System) |
| 4:09AM |
7 |
ISDN PRI E1 configuration with E100P |
| 3:52AM |
2 |
wait and user input.. |
| 3:35AM |
2 |
Compile Problems with gcc 3.3 |
| 2:02AM |
0 |
More voice prompts available now |
| 1:21AM |
2 |
Cisco 7960s |
| 12:52AM |
2 |
Wildcard E100P resellers in Europe ? |
| |
| Thursday July 10 2003 |
| Time | Replies | Subject |
|---|
| 11:20PM |
1 |
BudgeTone-100 Date and Time |
| 10:53PM |
1 |
msn authentication |
| 7:49PM |
0 |
system alias |
| 6:15PM |
0 |
channel bank setup "What do I do now?" |
| 4:33PM |
6 |
Channel Bank configuration |
| 1:56PM |
1 |
Why mp3 (licensing issues) as opposed to Open Source OGG |
| 1:23PM |
2 |
OH323 + G729 + Go2Call |
| 12:52PM |
1 |
Cisco 7960 SIP Craziness... |
| 12:43PM |
0 |
Cisco 7960 And Firmware Upgrades |
| 11:27AM |
1 |
Sip CANCEL or BYE when picking up a call ? |
| 11:01AM |
0 |
connect 2 asterisk boxes |
| 11:00AM |
0 |
Getting 488's between two c7940's |
| 10:05AM |
1 |
Cellphone as an exchange line |
| 9:05AM |
0 |
-- Got SIP response 481 "Invalid CSeq Number" back from 216.52.153.207 |
| 8:51AM |
1 |
TDM10B - Dies after a few hours |
| 8:26AM |
2 |
Transfers on the Cisco 7960 |
| 8:17AM |
1 |
Voicemail answers, but drops SIP call after about 3 seconds. |
| 8:09AM |
1 |
SIP call transfers - any other way than using '#' ? |
| 6:58AM |
1 |
Using Efax for virtual fax? |
| 6:49AM |
0 |
2003-06-10 CVS: softphone connection failures |
| 6:39AM |
3 |
T1 config for robbed-bit E&M AMI |
| 1:41AM |
0 |
Problem with meetme. |
| |
| Wednesday July 9 2003 |
| Time | Replies | Subject |
|---|
| 4:06PM |
2 |
IAX G729 Codec |
| 2:45PM |
1 |
IAX2 Warning |
| 1:54PM |
2 |
H450 problems |
| 1:31PM |
0 |
Asterisk Call Manager doc |
| 1:25PM |
1 |
OpenBSD version??? |
| 12:22PM |
1 |
E1-RJ45 pin configuration |
| 11:36AM |
2 |
experience with multi-port SIP/FXS gateways? |
| 11:19AM |
7 |
Asterisk basic how-to on O'Reilly's site |
| 11:16AM |
1 |
Asterisk as SIP <-> PSTN gateway |
| 10:41AM |
1 |
PRI with variable length numbers |
| 9:17AM |
0 |
SUMMARY: Problems with Hangup Detection in VoiceMail2. |
| 8:24AM |
2 |
chan_h323, Asterisk and DTMF issue |
| 8:21AM |
2 |
incoming callerid on FXO |
| 8:16AM |
2 |
sip jitter buffer |
| 8:15AM |
1 |
MGCP-H323v2 transcoder? |
| 8:10AM |
1 |
callerid= being ignored |
| 7:23AM |
17 |
caller id |
| 7:05AM |
1 |
more abou msn |
| 6:57AM |
2 |
Music on hold quality.. |
| 6:01AM |
1 |
PBX / Asterisk integration |
| 5:08AM |
2 |
It's true - Nikotel charge for not-completed calls |
| 4:54AM |
1 |
Use dialing plan from h.323 gatekeeper? |
| 4:08AM |
3 |
Asterix Manual |
| 2:56AM |
1 |
Matching winth asterisk-oh323 |
| 2:29AM |
2 |
error on web page for msn |
| 2:15AM |
2 |
Chan_capi hanging channels |
| 1:40AM |
0 |
Newbe Questions. |
| 1:16AM |
4 |
ignorepat doesn't work |
| 12:53AM |
1 |
Error on PRI channel : Call specified but not found! |
| 12:31AM |
2 |
modules.conf again |
| |
| Tuesday July 8 2003 |
| Time | Replies | Subject |
|---|
| 11:40PM |
0 |
dbget & dbput |
| 7:57PM |
2 |
voip |
| 5:26PM |
0 |
RE: IAXTEL toll-free From: Asterisk-Users digest, Vol 1 #791 - 10 msgs |
| 5:00PM |
0 |
DID number assignment to SIP phones |
| 4:15PM |
0 |
codec problems with asterisk |
| 2:28PM |
0 |
SIP Problem (previous post) .. information might be relevant |
| 1:21PM |
0 |
SIP disconnecting : response 481 |
| 1:09PM |
1 |
oh323 prob :) |
| 11:21AM |
4 |
Budgetone and Voicemail |
| 10:07AM |
4 |
Call Accounting |
| 10:06AM |
1 |
Debug PRI! |
| 8:29AM |
3 |
line battery check |
| 8:10AM |
5 |
Using multiple iconnecthere accounts |
| 7:26AM |
0 |
ENUM lookups |
| 6:46AM |
2 |
oh323 problem (small one) |
| 6:42AM |
1 |
RTP.C codec error 19 |
| 6:22AM |
1 |
Agent in new CVS |
| 6:20AM |
3 |
Answering on an zap device |
| 4:07AM |
2 |
asterisk-oh323 v0.5.3 |
| 3:38AM |
0 |
Patch to fix some segfaults in Asterisk |
| 3:12AM |
0 |
ECHO on sip- call |
| 3:06AM |
0 |
re. rtp.c RTP codec 19 |
| 2:56AM |
1 |
chanh323 dialling |
| 1:35AM |
2 |
Transfert call |
| 1:06AM |
0 |
FW: ATA 186 in Australia |
| 12:54AM |
0 |
Conferences with CAPI and H323 |
| 12:07AM |
1 |
Switch issues with non-dedicated comms.. (My experience) |
| |
| Monday July 7 2003 |
| Time | Replies | Subject |
|---|
| 11:23PM |
2 |
msn |
| 6:21PM |
1 |
Loaded latest CVS and get Broken PIPE!!! |
| 6:14PM |
0 |
Problems with Hangup Detection in VoiceMail2. |
| 4:39PM |
1 |
ATA 186 in Australia |
| 4:28PM |
1 |
Lot's of errors and warnings. |
| 4:28PM |
0 |
SIP canreinvite=yes Broke? |
| 3:23PM |
1 |
Dial plan doesn't seem to save properly |
| 2:57PM |
0 |
Fw: IAX Bandwidth Question |
| 2:38PM |
0 |
conection with other PBX's |
| 2:30PM |
4 |
BudgeTone-100 Early Dial |
| 1:31PM |
0 |
Asterisk crashing after Voicemail box creation |
| 1:15PM |
3 |
PCI Master Abort |
| 1:15PM |
0 |
Follow-up -- Using Asterisk with Nikotel |
| 12:36PM |
1 |
overlap dialing on a pri span |
| 12:03PM |
0 |
One-way talk paths (without INVITE?) and other issues |
| 11:44AM |
1 |
Getting Started with Digium T100/E100 cards |
| 11:31AM |
1 |
Can't access outside voicemail services through asterisk |
| 10:43AM |
3 |
System command.. |
| 10:33AM |
3 |
Network design question |
| 9:39AM |
0 |
register line on sip.conf |
| 9:27AM |
1 |
callgroup and pickupgroup |
| 9:16AM |
0 |
modules.conf |
| 9:11AM |
0 |
Initiations in IP voice/Hybrid Voice/etc... |
| 9:07AM |
12 |
Asterisk and VMWare |
| 8:47AM |
5 |
Direct entry to your own voice mailbox |
| 8:32AM |
1 |
three way calling and cisco ata 186 |
| 7:21AM |
3 |
Newbie Doubts |
| 7:05AM |
1 |
Problems with TDM40P |
| 5:32AM |
0 |
problems with new FXS module |
| 4:08AM |
1 |
Remote * Using IAX |
| 2:58AM |
2 |
IAX Bandwidth Question |
| 2:56AM |
1 |
Voicemail2 Contexts |
| 12:07AM |
1 |
Problem with SIP Phone with outgoing phone call |
| |
| Sunday July 6 2003 |
| Time | Replies | Subject |
|---|
| 10:54PM |
0 |
Ringing in sequence |
| 8:07PM |
9 |
Accurate Billing |
| 12:04PM |
3 |
Digital phones |
| 11:11AM |
0 |
Hardware / resources |
| |
| Saturday July 5 2003 |
| Time | Replies | Subject |
|---|
| 11:20PM |
3 |
Activate MySQL logging |
| 7:18PM |
2 |
TDM400P noise? |
| 5:30PM |
1 |
Fw: SIP Client X-Lite |
| 4:49PM |
0 |
SIP show channels display |
| 12:53PM |
1 |
FWD trouble - 407 error |
| 11:57AM |
1 |
E&M DID config question |
| 10:42AM |
2 |
Please help -- Syntax for dialing VoIP provider |
| 9:23AM |
3 |
Runtime error: Undefined symbol, have fetched new CVS and recompiled everything |
| 6:42AM |
4 |
Integratting * With Database(Newbie) |
| 5:59AM |
2 |
macro-record-cleanup in extensions.conf |
| 3:41AM |
0 |
Zaptel Alarms/Manager interface |
| 3:00AM |
1 |
Cllecting digits. |
| |
| Friday July 4 2003 |
| Time | Replies | Subject |
|---|
| 11:20PM |
3 |
Virtual fax on the Asterisk box |
| 3:54PM |
1 |
CDR Information and Pipes |
| 10:38AM |
1 |
How to make * send RTCP reports |
| 9:24AM |
3 |
zt_pri_errors: PRI got event: 8 / 6 |
| 9:03AM |
0 |
gastman and queues |
| 6:09AM |
4 |
Crossover T1 cable |
| 5:38AM |
1 |
IVR problem from PSTN phone |
| 5:23AM |
5 |
Asterisk Sacrifice? |
| 5:05AM |
1 |
LD accontability |
| 2:38AM |
1 |
Accounting info for SIP Calls |
| 1:21AM |
1 |
[Newbie] SIP via fwd |
| 1:01AM |
3 |
switch => priority in the dialplan.. (probably an issue for Mark) |
| 12:41AM |
2 |
dst number |
| |
| Thursday July 3 2003 |
| Time | Replies | Subject |
|---|
| 10:03PM |
1 |
Coding Time/Date announce on voicemail |
| 5:23PM |
0 |
The Budgetone 100 |
| 2:36PM |
1 |
res parking patch |
| 2:11PM |
1 |
No ringing when I dial an extension |
| 1:11PM |
4 |
Migration to Asterisk - Running off of Merlin Legend system |
| 1:01PM |
2 |
Drops due to codecs? |
| 11:17AM |
1 |
That is not a valid conference number meesage |
| 11:09AM |
0 |
ATA-186 re INVITE message turn off |
| 11:01AM |
0 |
How do I make Asterisk login at/use VoIP provider? |
| 10:29AM |
2 |
Asterisk - Protocol Converter from SIP/H.323 |
| 10:16AM |
5 |
Need a recommendation on a good motherboard/processor combination |
| 8:34AM |
2 |
ATA-186 de-register |
| 8:23AM |
1 |
How does Asterisk handle connecting two IP end points? |
| 8:20AM |
4 |
How do you force Asterisk to use only specific codecs? |
| 8:06AM |
0 |
Is there any real asterisk documentation ? |
| 7:33AM |
1 |
Bugetone SIP Transfer |
| 4:38AM |
0 |
Bugetone NTP problem.. |
| 3:57AM |
0 |
Modem channel in Asterisk |
| 3:14AM |
3 |
Using switch => |
| 1:15AM |
0 |
CDR->dst and immediate=yes in zapata.conf |
| 12:51AM |
0 |
(no subject) |
| 12:28AM |
0 |
app_festival not cleaning up properly? |
| 12:28AM |
2 |
Problem with digit 0 X-lite |
| |
| Wednesday July 2 2003 |
| Time | Replies | Subject |
|---|
| 11:35PM |
0 |
Asteriks, GnuGk and outgoing calls |
| 7:41PM |
0 |
Sorry 'bout that |
| 3:35PM |
2 |
client reinvitation problem |
| 11:49AM |
0 |
Re: [Asterisk-Dev] ANNOUNCE: CLASS-like features for Asterisk |
| 11:34AM |
9 |
BIG problem with multiple rings before pickup |
| 10:48AM |
1 |
Dialout Lines ??? |
| 8:36AM |
1 |
More switch => stuff |
| 8:05AM |
2 |
Sip call dropping |
| 7:57AM |
4 |
Linejack strikes again. |
| 5:22AM |
2 |
Seg Fault!! |
| 3:56AM |
2 |
Problems with musiconhold |
| 3:21AM |
4 |
Asterisk and Hot Desks?? |
| 2:48AM |
3 |
Asterisk PBX Billing |
| 12:22AM |
2 |
record a conversation |
| |
| Tuesday July 1 2003 |
| Time | Replies | Subject |
|---|
| 4:24PM |
2 |
Today's Message from linphone; update on Khpone and SJPhone and X-Lite |
| 4:06PM |
0 |
CVS fixed |
| 1:27PM |
1 |
FGB not waiting for digits |
| 12:31PM |
0 |
Actiontec's InternetPhoneWizard and Asterisk |
| 12:02PM |
0 |
H.323 CallerID |
| 11:44AM |
0 |
Zap outgoing hangups callprogress=yes? |
| 10:37AM |
0 |
Acom 200 B/D/E |
| 10:35AM |
1 |
*8 pickup then transfer drops call |
| 10:16AM |
0 |
Large-scale voicemail deployment: any experiences? |
| 9:30AM |
3 |
H.323 Gateway Connection |
| 7:53AM |
2 |
Problem with echo |
| 7:11AM |
6 |
Enhanced queue app |
| 7:08AM |
1 |
gotoiftime error |
| 6:04AM |
0 |
Friendly Slow Faxing Reminder |
| 5:37AM |
0 |
chan_h323.c compile error |
| 3:58AM |
2 |
Unable to get SetMusicOnHold working... |
| 3:47AM |
0 |
"Forbidden" problem!! |
| 3:43AM |
0 |
Logfile rotation |
| 2:52AM |
3 |
picking up a ringing extension |
| 2:09AM |
0 |
More mec3 feedback |