| Tuesday October 31 2006 |
| Time | Replies | Subject |
|---|
| 11:39PM |
1 |
wrong password on authentication for INVITE |
| 8:35PM |
0 |
AW: NAT issue ? [More Info] |
| 6:29PM |
2 |
Opinions on the best wholesale origination/term providers |
| 5:38PM |
0 |
NAT issue ? [More Info] |
| 5:13PM |
2 |
Newbie Questions |
| 4:52PM |
0 |
[SPAM HEADER] - Re: Snom or Cisco Phones? - Email found in subject |
| 4:47PM |
1 |
NAT issue ? |
| 4:32PM |
0 |
SIP with Qualify and NAT |
| 2:49PM |
2 |
compilation problem with asterisk-addons |
| 1:24PM |
0 |
FXO Card's vs. T1 |
| 12:42PM |
2 |
AW: Snom or Cisco Phones? |
| 12:15PM |
4 |
DTMF Tones |
| 12:03PM |
1 |
Strange Characters in CLI on TTY9 |
| 11:22AM |
7 |
FXO Cards vs. Channel bank with T1 |
| 11:13AM |
3 |
overlap of zap trunk groups |
| 10:28AM |
2 |
simultaneous ring - call groups or queues or something else? |
| 10:24AM |
5 |
Example Polycom function key config |
| 10:21AM |
3 |
Snom or Cisco Phones? |
| 9:44AM |
1 |
Astricon followup |
| 9:43AM |
2 |
channel.c: Unable to request channel ZAP |
| 9:00AM |
7 |
Asterisk Call Statistics |
| 8:44AM |
6 |
Asterisk web interface is not parsing the PHP pages |
| 8:29AM |
1 |
auto recording extensions |
| 7:55AM |
3 |
Asterisk and ARI (Aterisk Recording Interface) integration problem |
| 7:12AM |
0 |
Asterisk dial out (in SIP) to another asterisk context ! |
| 7:03AM |
2 |
SIP RTP flow |
| 6:34AM |
1 |
+Ura +md3200 nao encaminha ligacao |
| 6:06AM |
0 |
SMS and 1.2.12 |
| 5:27AM |
1 |
dial D option with w for wait? |
| 5:18AM |
1 |
sip realtime broken? |
| 4:19AM |
6 |
best gui |
| 3:44AM |
0 |
Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 2:42AM |
0 |
Setting up UTStarcom F300 |
| 2:41AM |
0 |
Read cmd |
| 2:31AM |
2 |
Bridging Video Calls using Zap |
| 1:59AM |
1 |
Asterisk does not bridge zap channels on outgoing calls |
| 1:29AM |
1 |
Fedora Core 6 (FC6) and Asterisk-1.2.13 and Zaptel-1.2.10 compile problems |
| 1:26AM |
2 |
Asterisk both behind a NAT and outside at the same time |
| 12:37AM |
1 |
S(x) - Hang up the call after 'x' seconds - Not working from queue |
| |
| Monday October 30 2006 |
| Time | Replies | Subject |
|---|
| 9:36PM |
1 |
Audiocodes MP-114 noise |
| 6:52PM |
3 |
Cisco 7960 Skinny calling SIP phone |
| 6:13PM |
1 |
dealing with blind transfers to invalid extensions |
| 5:44PM |
0 |
sip trunk - SIP/2.0 488 Not Acceptable Media |
| 5:14PM |
1 |
Registration problem |
| 5:08PM |
4 |
Architecture for Asterisk |
| 4:52PM |
4 |
IVR |
| 4:24PM |
1 |
Asterisk architecure |
| 4:22PM |
3 |
Server Recommendations |
| 3:38PM |
0 |
Good phones for outside of the office? |
| 3:16PM |
1 |
MFC/R2 patch problems |
| 1:08PM |
3 |
Live creation of trunk groups |
| 12:52PM |
1 |
Forwarding recorded calls to Voicemail |
| 12:51PM |
2 |
light web user interface |
| 12:51PM |
2 |
Fxo box for asterisk ? |
| 12:45PM |
1 |
TE110P Card |
| 12:05PM |
0 |
Asterisk Billing Plataforms |
| 11:42AM |
6 |
Asterisk and Panasonic KX Model |
| 11:30AM |
1 |
Wildcard X100P Suport |
| 10:38AM |
1 |
Anyone got a dialplan for SPA ATAs for ISN? |
| 10:31AM |
3 |
Billing Solution ? |
| 9:27AM |
2 |
operator console |
| 9:17AM |
3 |
Grandstream ATA 286 tdm400 and Asterisk 1.2-13 |
| 8:57AM |
0 |
Asterisk and Siemens C450IP |
| 8:46AM |
1 |
Realtime in the Real World |
| 7:21AM |
6 |
How to do Automatic Daylight Saving on Grandstream GXP-2000 |
| 6:03AM |
1 |
show logged clients |
| 5:46AM |
0 |
Extension Matching with "Match As You Go" Dialing |
| 5:43AM |
0 |
Realtime trouble with contex |
| 5:42AM |
1 |
SIP Server |
| 5:36AM |
1 |
Mac OS X Desktop / Asterisk integration? |
| 4:55AM |
1 |
Need Help in Meetme (Conferencing) |
| 4:43AM |
0 |
Vgsm driver 0.18.0 released today |
| 4:32AM |
0 |
RT Problem: Asterisk & Session Border Controller |
| 3:31AM |
0 |
Problem with Digium 400P and asterisk 1.4 |
| 3:29AM |
2 |
anti ex-girlfriend |
| 3:26AM |
0 |
Problem with incomming calls |
| 3:06AM |
0 |
Intel S3000AHLX - Digium TE110P |
| 2:22AM |
0 |
Information on Asterisk 1.4-beta 3 and ARA |
| 12:17AM |
0 |
Re: Linksys PAP2: calling tone stops after 5 |
| 12:10AM |
0 |
Call from internal num. to VoIP gate |
| |
| Sunday October 29 2006 |
| Time | Replies | Subject |
|---|
| 10:19PM |
1 |
CID and CDR conflict? |
| 10:13PM |
2 |
Incorrect Ring tone. Getting a US tone when it should be AU tone |
| 8:34PM |
3 |
No ring tone when using IAX |
| 6:38PM |
1 |
Multiple dial macros at the same time |
| 5:25PM |
2 |
asterisk-1.2.13 fails to 'Make' in Fedore Core 6' |
| 4:58PM |
0 |
Polycom IP500 Problems |
| 3:47PM |
4 |
blind transfers with IP Polycom 501 |
| 2:41PM |
1 |
AEL2 and the variables |
| 2:27PM |
1 |
Asterisk Voicemail with ODBC Realtime Access |
| 2:00PM |
1 |
Linksys PAP2: calling tone stops after 5 tones |
| 1:43PM |
1 |
Out bound calls 'you must first dial a 1' |
| 1:39PM |
1 |
Something is trashing /var/run |
| 1:22PM |
2 |
No zap* commands? |
| 10:29AM |
0 |
hardware requirements.. |
| 10:13AM |
0 |
H.263 Video Messages |
| 3:44AM |
3 |
Pager Voicemail Message |
| 1:47AM |
2 |
app_meetme not loading |
| |
| Saturday October 28 2006 |
| Time | Replies | Subject |
|---|
| 8:17PM |
1 |
How to make different ext using different trunks? |
| 6:16PM |
4 |
VoIP GSM Gateways |
| 4:40PM |
1 |
Compiling Zaptel 1.2.10 on Ubuntu 6.10 |
| 4:26PM |
3 |
Asterisk behind NAT and without portforwarding for rtp |
| 2:31PM |
0 |
Queues: roundrobin w/ reset ("circular call distribution") |
| 11:47AM |
1 |
Diva server 4bri and Portuguese BRI |
| 10:20AM |
1 |
tx_fax not getting entire fax |
| 9:03AM |
0 |
Asteroid SIP Denial of Service Tool |
| 7:58AM |
0 |
Zap disconnect |
| 7:55AM |
0 |
Is it possible to connect two servers using SIP? |
| 6:45AM |
1 |
translate.c:88 powerof: Powerof 0: No power?? |
| 5:44AM |
0 |
IAX2/SIP Wifi Phones |
| 1:51AM |
0 |
Polycom 501 + Voicemail notification |
| 1:09AM |
1 |
Configuring 2 Asterisk servers with a SIP trunk |
| |
| Friday October 27 2006 |
| Time | Replies | Subject |
|---|
| 10:40PM |
1 |
dialing external number within meetme |
| 9:30PM |
2 |
0 channels configured with tdm400 (tdm04b rev. G) |
| 7:05PM |
1 |
Waiting before executing System command |
| 3:59PM |
0 |
autocreate peer + sippeers table entry => auth required? |
| 3:43PM |
0 |
Voicemail 'exitcontext' |
| 1:27PM |
0 |
Enterprise Asterisk User Group |
| 1:21PM |
0 |
Vancouver Asterisk User Group |
| 12:55PM |
0 |
Zultys Phones w/ Encryption |
| 12:31PM |
0 |
Confused about SIP Realtime Updates |
| 12:14PM |
1 |
New Asterisk-GUI? |
| 11:53AM |
0 |
fully featured and robust * client gui? |
| 11:39AM |
1 |
Digium TE110P |
| 11:08AM |
0 |
detecting ring |
| 9:59AM |
1 |
Voicemail and OSX 10.4 Intel |
| 9:27AM |
1 |
meet me |
| 9:17AM |
0 |
Asterisk stopps matching extensions after first digit |
| 8:49AM |
3 |
[OT] wi-fi ip phone scenario |
| 8:11AM |
2 |
polycom's don't register with 2.6.18 |
| 7:28AM |
1 |
Taking a Polycom IP601 home |
| 7:26AM |
0 |
set outgoing msn on chan_misdn |
| 7:08AM |
0 |
Enhancements for the Queue application |
| 6:12AM |
1 |
Snom, mute and rtptimeout |
| 6:03AM |
2 |
Advice on GUI |
| 5:37AM |
0 |
How to hung up , While in Conference going on. |
| 5:23AM |
4 |
IAX2 show peers - description |
| 3:56AM |
2 |
DTMF detection problem in PABX trunk |
| 3:55AM |
1 |
Direct call vs Block call |
| 3:40AM |
2 |
ISDN-BRI issue |
| 3:21AM |
2 |
asterisk misdn incoming line not working. |
| 2:37AM |
3 |
Would you support a Bristuff mailing list ? |
| 1:57AM |
0 |
lines usage statistics |
| 1:54AM |
0 |
Auto Dial problem! |
| 1:31AM |
1 |
Iax bug ? |
| 1:25AM |
0 |
chan_skype license? |
| |
| Thursday October 26 2006 |
| Time | Replies | Subject |
|---|
| 10:57PM |
0 |
Re: asterisk-users Digest, Vol 27, Issue 140 |
| 10:11PM |
3 |
dialplan issue - 1& 0 should be evaluated false |
| 9:39PM |
0 |
dialplan issue - 1& 0 should be false |
| 9:31PM |
0 |
[Fwd: Asterisk n QoS] |
| 7:44PM |
1 |
simple dialplan trick I can't figure out (smdi, mwi substitute) |
| 6:09PM |
0 |
Make/Break ratio for Pulse Dialing |
| 4:57PM |
0 |
7960 (8.2) - Call Center - REBOOT |
| 3:22PM |
1 |
SipAddHeader |
| 2:22PM |
4 |
Asterisk and ISDN and Hylafax |
| 2:12PM |
0 |
Open SER or DUNDI |
| 12:29PM |
0 |
Call Routing Time Issue |
| 11:50AM |
1 |
Lumenvox speech recognition |
| 11:40AM |
0 |
Cepstral/Swift TTS app |
| 10:39AM |
0 |
Can't Register Client - Multiple Subnets |
| 10:05AM |
0 |
external username conflict in dialplan |
| 9:32AM |
2 |
Is SQLite a replacement for Mysql while using ARA in 1.2.x |
| 7:32AM |
1 |
IPv6 |
| 6:32AM |
0 |
How to disconnect in Conferenceing in between the Confermce ..... |
| 5:25AM |
1 |
channel.c: Avoided initial deadlock |
| 5:10AM |
0 |
Asterisk n QoS |
| 5:01AM |
0 |
question about IF |
| 4:21AM |
2 |
"Cheapest" way to determine channels in a group from outside asterisk? |
| 4:09AM |
0 |
Problem: Dial command with L option |
| 4:00AM |
4 |
porting numbers in UK telewest/bt/adept |
| 3:51AM |
6 |
SIP v IAX2 |
| 3:32AM |
0 |
OOH323 GK Context Help |
| 3:15AM |
10 |
ECHO Cancellation in SIP Calls |
| 3:13AM |
1 |
Query regarding Pulse Dialing at 20 pps |
| 1:54AM |
1 |
PRI (TE205P) allways RED/NOP |
| 12:39AM |
1 |
chan_capi and bristuff |
| |
| Wednesday October 25 2006 |
| Time | Replies | Subject |
|---|
| 11:42PM |
1 |
Phone Rings, Immediate Hangup and then Rings Again. |
| 11:06PM |
0 |
spandsp bug |
| 8:13PM |
1 |
WiFi Phones (was Looking for Wireless Heaset for Polycom 501) |
| 6:42PM |
1 |
Default login information for a ArtDio IPF-2600 |
| 2:58PM |
1 |
Need recommendation for SIP hard phones |
| 2:22PM |
2 |
CSU Support on Digium T1 Cards |
| 1:59PM |
2 |
Cisco 7971G-GE & SEP{MAC}.cnf.xml |
| 1:08PM |
0 |
[SPAM] - Looking for Wireless Heaset for Polycom 501 - Email found in subject |
| 12:30PM |
2 |
"No Authority Found" |
| 11:37AM |
3 |
Add second account to Xlite 3.0 |
| 11:31AM |
2 |
Looking for Wireless Heaset for Polycom 501 |
| 11:29AM |
2 |
Multiple queue_log files based on queue - is it possible?? |
| 11:25AM |
1 |
Trixbox installation - ZAP channels becoming upresponsive |
| 11:12AM |
1 |
Re: Asterisk Manager |
| 9:51AM |
0 |
Re: Meetme... No channel type registered for'zap' |
| 9:14AM |
2 |
Simple example for call transfer. |
| 8:05AM |
0 |
chan_misdn |
| 7:44AM |
2 |
Without ZapTel inferface or Card install , is Conference working or Not |
| 6:53AM |
0 |
Conference is Not Working.... with OpenSER And Asterisk |
| 6:20AM |
2 |
Nerdvittle's Reminders and Zaptel |
| 6:06AM |
3 |
Maximum talktime in a queue? |
| 5:55AM |
2 |
SIP problem - ACT p160s error |
| 5:27AM |
3 |
Quintum DX as gateway to PSTN for Asterisk |
| 4:19AM |
0 |
*****SPAM***** asterisk 1.4 problem with call queues |
| 2:44AM |
2 |
PBAX-Group with QuadBRI card, outgoing call problem |
| 2:24AM |
2 |
Choice of soundfile format |
| 1:54AM |
2 |
Call is not coming through sipgate.co.uk+Asterisk |
| 12:51AM |
5 |
VoiceOne 0.4.0 released: a new web-based and open source GUI |
| |
| Tuesday October 24 2006 |
| Time | Replies | Subject |
|---|
| 11:51PM |
0 |
sip.conf - srvlookup |
| 8:37PM |
6 |
Callmanager 3.3(5) and Asterisk with ooh323 |
| 8:33PM |
1 |
Adit 600 resetting |
| 8:29PM |
1 |
All calls Hangup after receive these logs. |
| 5:03PM |
1 |
AstFax Sending a Fax |
| 4:47PM |
1 |
Basic Conf |
| 3:22PM |
10 |
Meetme... No channel type registered for 'zap' |
| 2:23PM |
3 |
ASterisk Start problem |
| 1:52PM |
0 |
attempting native bridge on TDM2400 |
| 1:44PM |
1 |
update_header: Unable to find our position |
| 1:41PM |
1 |
problem with setting outbound caller id when calling another asterisk |
| 1:04PM |
2 |
IAX2 goes "one way audio" when lag gets bad |
| 12:48PM |
0 |
Problem with CallerID (UK) TDM400P ( CID timed out waiting for ring ) |
| 12:38PM |
0 |
1.4 Beta 3 H323 Video? |
| 11:58AM |
3 |
"Fixing the Caller-ID Problem", by John Todd for O'ReillyNet |
| 11:05AM |
2 |
Voicemail help |
| 10:32AM |
1 |
(no subject) |
| 10:28AM |
5 |
need help using tftp for polycom 501 |
| 10:24AM |
0 |
txfax only getting 1 page of 3. |
| 8:21AM |
0 |
misdn.conf: how to set music on hold |
| 8:16AM |
1 |
Resampling Audio for use with Asterisk |
| 8:06AM |
1 |
Distributing calls among channels in dial group |
| 6:51AM |
1 |
voicemail idea and a question |
| 6:44AM |
3 |
Dynamic Codec Selection |
| 5:22AM |
0 |
something about Agent Transfer |
| 5:13AM |
6 |
Becoming a User on IRC |
| 4:29AM |
0 |
CDR_DISPOSITION_FAILED - Call has been answered correctly |
| 1:41AM |
0 |
Core dumps when Releasing clone lock |
| 12:54AM |
0 |
newbie astdb error, please help |
| 12:24AM |
0 |
mgcp registration with asterisk |
| 12:22AM |
2 |
UA - number assignment |
| |
| Monday October 23 2006 |
| Time | Replies | Subject |
|---|
| 11:25PM |
2 |
T.38 faxing with spandsp and Grandstream HT.486 |
| 7:54PM |
0 |
Callmanager 3.3(5) and Asterisk with ooh323 problem |
| 6:09PM |
1 |
Asterisk conferencing features |
| 6:01PM |
2 |
Digium vs. Sangoma |
| 5:06PM |
1 |
make menuselect question- Module Embedding |
| 3:51PM |
2 |
Polycom SP4000 ftp problem |
| 2:08PM |
0 |
Multiple line phones with different contexts |
| 1:48PM |
4 |
Where to best start looking for voicemail/moh sound quality problem? |
| 1:43PM |
0 |
CBeyond SIP |
| 1:30PM |
1 |
Polycom provision errors still! Arg! |
| 1:21PM |
1 |
Question on one-way-audio with IAX |
| 1:08PM |
8 |
Asterisk and dialer Running on Thin Clients |
| 1:03PM |
0 |
REQ: Astricon Pictures |
| 12:59PM |
3 |
One way audio half way through call |
| 11:24AM |
0 |
call file mechanism |
| 10:08AM |
1 |
INVAL Messages |
| 8:54AM |
1 |
asterisk and HMP |
| 8:49AM |
2 |
asterisk not detecting hangup |
| 8:01AM |
1 |
Macro 'exited non-zero' |
| 7:55AM |
0 |
How to busy out PRI channels? |
| 7:07AM |
1 |
chan_h323.so Asterisk Beta compilation |
| 6:58AM |
0 |
7960/SIP MWI Question |
| 6:52AM |
1 |
Primary D-Channel & channal numbers.... |
| 5:55AM |
0 |
(no subject) |
| 5:45AM |
4 |
Problems with chan-capi and Eicon Diva 4BRI |
| 5:33AM |
0 |
Real Time and Asterisk |
| 5:32AM |
0 |
(no subject) |
| 4:52AM |
2 |
Zap Channel and VM problem |
| 4:33AM |
2 |
spandsp and freebsd |
| 4:26AM |
0 |
SIP_HEADER function; what names are available? |
| 3:34AM |
1 |
astdb error, please help |
| 1:45AM |
3 |
Unicall Installation |
| 1:23AM |
0 |
Can anyone help? Why does One-Touch record mute/disconnect callif not dialed quick enough? |
| 1:12AM |
0 |
Primary D-Channel on span 2 down |
| 12:58AM |
1 |
Why does it take at least 4 flipping days before asterisk tries to resolve a provider? |
| 12:24AM |
0 |
Compiling H323 channel Asterisk 1.4.Beta3? |
| |
| Sunday October 22 2006 |
| Time | Replies | Subject |
|---|
| 9:01PM |
2 |
asterisk guru needed for job in Chicago area |
| 7:35PM |
2 |
How to deploy a PBX in such a condition ? |
| 5:16PM |
2 |
checking 'voicemail" externally - doesn't work |
| 3:47PM |
3 |
Audiocodes MP-20x |
| 11:07AM |
1 |
[SOLVED] 1.2.12.1 crashing |
| 7:14AM |
3 |
G.729 operating on outgoing only |
| 5:29AM |
2 |
Using variable as a context extension ? |
| 3:03AM |
1 |
new g.729a codecs for asterisk 1.2/1.4 and glibc |
| |
| Saturday October 21 2006 |
| Time | Replies | Subject |
|---|
| 6:57PM |
0 |
AGI Help |
| 11:16AM |
2 |
Unique call ID's across several systems |
| 10:18AM |
0 |
forward several times |
| 9:59AM |
0 |
Using the ZOOM 5801 ATA with Asterisk |
| 9:08AM |
1 |
new route by caller id |
| 7:50AM |
0 |
Asterisk 1.4beta3 and Asterisk Manager API Action: ExtensionState |
| 7:48AM |
1 |
zaptel 1.2.10 make problem |
| 6:44AM |
0 |
route by caller id |
| 1:57AM |
2 |
1.4 branch on OSX? |
| |
| Friday October 20 2006 |
| Time | Replies | Subject |
|---|
| 10:38PM |
0 |
(no subject) |
| 1:51PM |
2 |
modprobe Ztdummy is not working |
| 12:41PM |
1 |
Snom 320, Queues and Transfer not working as expected with * 1.2.12.1 |
| 11:47AM |
0 |
centos or rhel and txfax with libtiff |
| 11:12AM |
1 |
some transfers dropped. |
| 11:00AM |
1 |
Escape from Voicemail |
| 10:01AM |
1 |
PRI boards with g729 capable DSPs |
| 9:16AM |
2 |
getting DID info.. |
| 8:13AM |
2 |
Clicking Noise on Pure Voip Calls |
| 7:54AM |
1 |
#Transfer - Timeout is configurable? |
| 7:47AM |
3 |
voicemail usernames can't begin with "j" letter? |
| 6:59AM |
2 |
noise gate for asterisk? |
| 6:45AM |
1 |
Astricon - post show Saturday? |
| 5:30AM |
0 |
using asterisk to do remote control |
| 5:11AM |
1 |
call center status viewer |
| 5:09AM |
3 |
Linksys PAP2 dial plan help please |
| 4:46AM |
0 |
Xorcom Astribank |
| 2:59AM |
1 |
Asterisk Realtime... Help Me!!! |
| 2:28AM |
3 |
using asterisk to do remote control functions |
| 12:44AM |
0 |
Asterisk 1.2.13 make problem |
| 12:34AM |
1 |
Help: Problems about console color (FC5, XTerm) |
| |
| Thursday October 19 2006 |
| Time | Replies | Subject |
|---|
| 9:07PM |
2 |
/dev/zap/channel ownership |
| 8:31PM |
1 |
Getting started with sample dial plans |
| 8:26PM |
2 |
Polycom boot error |
| 8:23PM |
0 |
question about asterisk txFAX |
| 6:05PM |
0 |
FS: Sangoma A200 10 port FXO card |
| 2:20PM |
0 |
DTMF logging |
| 2:04PM |
1 |
bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now |
| 1:21PM |
0 |
DTMF / Silence issues |
| 1:13PM |
7 |
Embedded Asterisk |
| 12:45PM |
3 |
T1 pricing in Oz |
| 12:13PM |
3 |
plainvoip - down ??? |
| 10:04AM |
0 |
Errors in console in every call made when using 1.4b3 |
| 9:45AM |
1 |
How do I configure Asterisk if I need to run Mysql server on second Linux |
| 9:06AM |
2 |
Occasional one-way audio - Sangoma A101 |
| 8:57AM |
1 |
rxfax problem |
| 8:22AM |
1 |
siemens hipath interoperability - PRI/Q.SIG - card recommendation |
| 7:04AM |
1 |
Modifying SIP Stack |
| 5:55AM |
0 |
Please help with these SIP errors |
| 5:53AM |
0 |
Multiple bridge attempts |
| 5:41AM |
3 |
Bristuff qozap drivers problem |
| 4:44AM |
1 |
Access Denied on a Windows share |
| 3:11AM |
0 |
RE gotoiftime and Macro question |
| 3:00AM |
3 |
say Asterisk to answer |
| 2:38AM |
1 |
Zaptel not detecting Tormenta2 PRI Interface card |
| 2:12AM |
2 |
wrong outgoing caller id with PRI lines: maybe usecallingpres involved? |
| 1:55AM |
0 |
Which is the best ? |
| 1:27AM |
0 |
Got reject for frame XX, retransmitting frame XX now, updating n_r! |
| 1:02AM |
0 |
Cisco 7970 - versionStamp |
| 12:37AM |
3 |
accountcode and amaflags? |
| 12:36AM |
1 |
SIP users with Database |
| |
| Wednesday October 18 2006 |
| Time | Replies | Subject |
|---|
| 11:56PM |
0 |
Asterisk Realtime with ODBC/MySql |
| 10:30PM |
3 |
Asterisk hangs up on incoming analog calls after a while |
| 10:20PM |
1 |
How to get the agent id in the recording filename |
| 8:43PM |
1 |
question about CDR command |
| 8:37PM |
0 |
What doe these error messages mean? |
| 8:25PM |
0 |
Asterisk 1.4.0-beta3 released! |
| 8:24PM |
0 |
Asterisk 1.2.13 released - Security Vulnerability Fix |
| 8:24PM |
0 |
Asterisk 1.0.12 released - Security Vulnerability Fix |
| 7:36PM |
4 |
Asterisk + Huawei |
| 6:10PM |
1 |
Speed Dials |
| 4:36PM |
0 |
OT: (Job) Full-Time Asterisk Opportunity |
| 2:40PM |
2 |
echotraining=yes in misdn.conf is invalid or out of range. |
| 2:37PM |
1 |
adding outbound prefix |
| 2:31PM |
1 |
1.4 downgrade |
| 12:34PM |
2 |
Sip Trunks |
| 10:27AM |
2 |
random one way audio and noise betweenSIP phoneson same LAN |
| 10:03AM |
1 |
IAX softphones |
| 9:53AM |
0 |
Help with fxotune |
| 9:10AM |
0 |
ooh323 dtmf problem |
| 8:20AM |
0 |
DTMF problems with legacy PBX |
| 7:52AM |
0 |
list down? |
| 7:41AM |
0 |
QueueMetrics 1.3 released today |
| 7:28AM |
1 |
CAPI channel not available but nobody is usingthe system |
| 7:20AM |
4 |
Findme problem |
| 6:43AM |
1 |
Polycom IP650 |
| 6:20AM |
2 |
random one way audio and noise between SIP phoneson same LAN |
| 5:35AM |
1 |
Windows and file shares |
| 5:31AM |
1 |
Asterisk+SER help |
| 4:52AM |
3 |
identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M |
| 4:44AM |
2 |
gotoiftime and Macro question |
| 4:44AM |
2 |
random one way audio and noise between SIP phones on same LAN |
| 4:15AM |
0 |
cut ip adress from caller id number display (ci$co 7941) |
| 3:45AM |
1 |
Orange Flash Light Mitel 5215 - Asterisk - working ! |
| 2:34AM |
1 |
Blank page when sending faxes (repost) |
| 2:06AM |
1 |
Server power indication |
| 1:54AM |
2 |
Digium on Dell PowerEdge 1850 |
| 1:40AM |
0 |
IAX2 thru NAT problem |
| 12:58AM |
1 |
Netgear WGT Flash-fest at Astricon |
| 12:46AM |
0 |
AGI for authenticating calls with DTMF |
| 12:40AM |
1 |
Re: Is 1.2.12.1 production ready (Mauro Zanin) |
| 12:37AM |
0 |
Please explain these SIP errors |
| 12:25AM |
0 |
[OT] Nokia E60/61/70 and SIP |
| |
| Tuesday October 17 2006 |
| Time | Replies | Subject |
|---|
| 8:00PM |
0 |
Out dialing Integration |
| 6:58PM |
1 |
1.4 gsm files changed?? |
| 3:28PM |
0 |
Blank page when sending faxes |
| 2:03PM |
1 |
CAPI channel not available but nobody is using the system |
| 1:52PM |
3 |
Cisco 2621 NM-HDV VWIC-1MFT1 |
| 1:47PM |
0 |
FW: Why does One-Touch record mute/disconnect call if not dialed quick enough? |
| 11:49AM |
1 |
Unique ID |
| 11:11AM |
4 |
IVR problem |
| 10:49AM |
2 |
considering purchasing a t1 card, any recommendations? |
| 10:18AM |
0 |
lots of registrations, sip problem |
| 10:14AM |
1 |
Gtalk on Asterisk 1.4 |
| 9:59AM |
2 |
Electric usage of a tdm400p |
| 9:31AM |
0 |
Sipura 901? Any experiences |
| 9:14AM |
2 |
install MAGI |
| 8:39AM |
0 |
sipXezphone |
| 8:35AM |
0 |
Authenticate application |
| 8:26AM |
3 |
Locking phones at night... |
| 7:36AM |
0 |
acami |
| 7:02AM |
0 |
Setting the H323 Callerid sent by asterisk (using chan_h323) |
| 7:02AM |
1 |
Help with Dialplan Rules Please! |
| 6:59AM |
2 |
duplicate "ghost" calls with long duration |
| 6:42AM |
1 |
Please help me!! |
| 6:42AM |
0 |
Dial - i parametar |
| 6:38AM |
3 |
what hardware and is it possible |
| 6:19AM |
1 |
One way audio on chan_gtalk |
| 3:25AM |
0 |
TIMEOUT() function missing |
| 3:10AM |
1 |
how to activate recording (automon) |
| 2:31AM |
2 |
Inaccurate CDRs |
| 2:20AM |
1 |
Why the MusicOnHold sound so soft? |
| 2:18AM |
1 |
chan_bluetooth, mobile handset as VoIP terminal? |
| 2:09AM |
3 |
sending sip style messages in response |
| 1:43AM |
1 |
how to config chanspy |
| 1:21AM |
0 |
How to get Linksys-Sipura error codes ? |
| 12:59AM |
1 |
Call Forwarding Using Asterisk |
| |
| Monday October 16 2006 |
| Time | Replies | Subject |
|---|
| 11:52PM |
1 |
1.4 Beta and oracle |
| 11:14PM |
1 |
nat auto detect ? |
| 11:00PM |
1 |
Recording from a script |
| 9:36PM |
0 |
Critical - No audio issue with re-invite (wrong media address) |
| 9:09PM |
1 |
macros reference? |
| 9:07PM |
1 |
1.4 beta voicemail warning |
| 4:13PM |
0 |
Asterisk/VOIP to PSTN (?) |
| 4:10PM |
5 |
Stopping putgoing calls after working hours |
| 2:48PM |
4 |
Is 1.2.12.1 production ready |
| 1:30PM |
2 |
Accessing MySQL DB to set variables in Asterisk |
| 12:43PM |
3 |
Why is this happening? |
| 9:25AM |
1 |
ZapHFC & quadBRI D-Channel going down randomly |
| 9:04AM |
0 |
Do you encounter this REC alarm before? |
| 9:00AM |
0 |
Asterisk-ooh323c Video ? |
| 7:47AM |
4 |
Remote UNIX connection, Remote UNIX disconnected displayed every second |
| 7:10AM |
0 |
Asterisk <-> Live Communications Server Integration |
| 7:08AM |
3 |
Reception Console |
| 7:00AM |
0 |
Some Warning in Asterisk for Voicemail intgreting, |
| 6:45AM |
0 |
Tellabs and PRI |
| 6:01AM |
1 |
Monitor stops recording midstream? |
| 5:08AM |
1 |
Multiple 'routes' to extension in different contextes. How to influence search oder? |
| 4:58AM |
0 |
Weird problem with beep.wav! |
| 3:57AM |
1 |
quality control |
| 3:04AM |
2 |
asterisk upgrade |
| 2:59AM |
1 |
Page hangs up after 5 seconds |
| 2:58AM |
0 |
member queue refresh |
| 2:08AM |
0 |
SV: How do you like TrixBox? |
| 1:08AM |
1 |
FOP run control for CentOS/RHEL |
| 1:05AM |
1 |
Quescom 400 |
| 12:49AM |
2 |
Unable to open Asterisk database |
| 12:07AM |
7 |
tdm2400p question |
| 12:06AM |
0 |
Sipura SPA-481 |
| |
| Sunday October 15 2006 |
| Time | Replies | Subject |
|---|
| 10:54PM |
0 |
chan_bluetooth - one way audio |
| 10:13PM |
3 |
VoicePulse Connect 4 Channel Limit? |
| 9:32PM |
0 |
sip agent stuck in queue even after restarts |
| 8:57PM |
1 |
eagi-sphinx-test how and why |
| 3:39PM |
2 |
detecting the receivers voicemail |
| 12:30PM |
2 |
SPA942 quality for a Bank |
| 10:54AM |
0 |
Ringtones won't work |
| |
| Saturday October 14 2006 |
| Time | Replies | Subject |
|---|
| 10:53PM |
12 |
two SIP phones as one line |
| 8:00PM |
1 |
Codec swap (reinvite) |
| 7:14PM |
0 |
New and Improved |
| 7:09PM |
1 |
Student Research - Asterisk H323 Video |
| 6:04PM |
1 |
Re: Generate Random Numbers in dialplan |
| 5:54PM |
1 |
Re: Centos kernel 34 vs. 42? [was: asterisk-users Digest, Vol 27, Issue 72] |
| 4:57PM |
0 |
Re: Generate Random Numbers in dialplan |
| 11:29AM |
0 |
Test to list |
| 6:25AM |
1 |
12 port FXx PCI card |
| 5:27AM |
0 |
SIP trunk from an Audiocodes mediant 1000 |
| 4:11AM |
0 |
rxfax problem ("Trainability test failed") |
| |
| Friday October 13 2006 |
| Time | Replies | Subject |
|---|
| 11:35PM |
0 |
NAT/firewall/Asterisk/Polycom Phones |
| 11:14PM |
1 |
Looking for a Voicemail Lamp device |
| 10:29PM |
0 |
Problem in Voice Message Storing............... |
| 9:30PM |
2 |
Re: Generate Random Numbers in dialplan |
| 7:57PM |
2 |
DID failover |
| 5:07PM |
1 |
3way calling / codec problem |
| 3:36PM |
1 |
Avaya 8300 - Asterisk integration using H.323 |
| 2:57PM |
1 |
Calls being disconnected across VPN |
| 1:08PM |
1 |
Go to DIGIUMBOARDS.COM |
| 12:15PM |
1 |
OT: Voipsupply.com phones are down. Was: how big is *your* dialplan |
| 11:24AM |
3 |
VoipSupply? [Semi-Urgent] |
| 11:07AM |
1 |
Asterisk (meetme) and SMP/HT OK? |
| 10:58AM |
2 |
Centos kernel 34 vs. 42? |
| 10:51AM |
1 |
Inhouse SIP to ZAP has echo sometimes. |
| 10:49AM |
1 |
Unable to create/find SIP channel for this INVITE & Broadvoice |
| 10:19AM |
1 |
hold drops audio |
| 10:11AM |
2 |
How do I figure out where this connection is coming from? |
| 9:53AM |
5 |
Cisco 7970 SIP won't update? |
| 9:13AM |
3 |
Polycom HDVoice |
| 7:55AM |
3 |
How big is *your* ego? |
| 7:40AM |
5 |
Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls) |
| 6:30AM |
2 |
AEL Question |
| 6:10AM |
3 |
Switchtype,Signalling,rxwink warnings |
| 5:26AM |
3 |
OriginateEvent reason codes. |
| 4:34AM |
3 |
VoIP+RJ11 Phone existed? |
| 4:34AM |
1 |
Digium TE410P LED problem |
| 2:48AM |
0 |
Segmentation fault issue |
| 1:17AM |
0 |
Asterisk 1.4 / install app_bundle problems |
| 12:21AM |
0 |
how i can do auto dialing using mysql |
| |
| Thursday October 12 2006 |
| Time | Replies | Subject |
|---|
| 11:40PM |
2 |
Call Asterisk : It calls me backup with a dial tone |
| 11:07PM |
0 |
AGI scripts |
| 10:03PM |
2 |
Some file aren't loaded its No file in that Directory. |
| 9:41PM |
1 |
Voicemail prompts clipped when retrieving from some SIP phones |
| 9:19PM |
4 |
How do you like TrixBox? |
| 3:45PM |
1 |
AccountCode set in sip.conf but not showing up in CDR |
| 3:26PM |
1 |
OT: jobs for asterisk lovers |
| 2:59PM |
1 |
Attended transfer hanging PRI channel |
| 2:58PM |
2 |
Polycom IP 501 message light |
| 2:53PM |
0 |
Problem when both Proxy-Authenticate and WWW-Authenticate is required |
| 2:11PM |
1 |
unix sysctl config for asterisk |
| 1:47PM |
0 |
Codes negotiation problemsbetweenAsterisk1.4beta2 and Aastra 480i |
| 1:03PM |
5 |
unauthenticated calls |
| 12:58PM |
1 |
How to send correct Caller ID on PRI |
| 12:17PM |
2 |
Issues with Asterisk 1.4 Beta |
| 11:17AM |
1 |
Bridging of PRI calls |
| 10:24AM |
2 |
1.2.12.1 crashing |
| 10:13AM |
1 |
AstriCon Hotel Full - Here are some near-by alternates |
| 9:58AM |
1 |
SPA 3102 |
| 9:19AM |
0 |
OT: BioFuel to power phone networks |
| 7:39AM |
1 |
Call drop and strange CDR records |
| 5:58AM |
0 |
Reg. chanspy |
| 5:26AM |
2 |
Call bridged, but no sound |
| 5:23AM |
0 |
prohibit CallerID presentation |
| 4:22AM |
1 |
Fax receive (rx fax) problem |
| 3:23AM |
2 |
vGSM drivers updated (0.17.2) |
| 3:20AM |
1 |
Urgent Billing |
| 2:27AM |
0 |
Beronet BN4S0 instalation |
| 2:18AM |
0 |
Stripping digits on internal calls |
| 1:51AM |
1 |
Anybody using "inphonex" service? |
| 12:33AM |
1 |
How to enable talking in chanspy while spying? |
| 12:14AM |
0 |
Asterisk -> regitration in DB |
| |
| Wednesday October 11 2006 |
| Time | Replies | Subject |
|---|
| 11:02PM |
4 |
Multiple TE110P cards in one chassis |
| 10:41PM |
0 |
how to setup call center with media gateway? |
| 9:02PM |
3 |
TDM400P incoming route for DID |
| 7:07PM |
1 |
cdr_addon_mysql.c - Asterisk 1.4 - Asterisk Addons |
| 5:41PM |
1 |
XO SIP Origination Services |
| 5:31PM |
0 |
Meeting |
| 4:42PM |
0 |
CDR Help... |
| 2:56PM |
2 |
Test Call Script |
| 2:03PM |
1 |
What alternatives to Asterisk based virtual PBX? |
| 1:32PM |
1 |
Echo problems on ISDN. (mainly incoming call s) |
| 12:38PM |
0 |
Load balance Asterisk server, when it is a SIP client. |
| 12:26PM |
1 |
Urgent Please help |
| 11:54AM |
0 |
SIP Locking Up? |
| 11:53AM |
1 |
Asterisk as SIP Client |
| 11:24AM |
1 |
Echo problems on ISDN. (mainly incoming calls) |
| 10:25AM |
1 |
Problem with ZAPTEL-1.4.0-beta1 and WCT100P card |
| 10:17AM |
1 |
max users |
| 10:02AM |
1 |
1.4 beta2 on intel mac |
| 9:53AM |
1 |
compiling libunicall |
| 9:40AM |
1 |
Asterisk users help |
| 9:27AM |
1 |
average waiting time in a queue |
| 9:23AM |
0 |
Segmentation fault asterisk realtime problem |
| 9:17AM |
0 |
Re: asterisk-users Digest, Vol 27, Issue 49 |
| 9:11AM |
10 |
GPL Softphones |
| 8:30AM |
3 |
zt_chanconfig failed |
| 8:19AM |
3 |
asterisk 1.2.12 lost phone registrations today... why? |
| 8:10AM |
0 |
RE: Welcome to the "asterisk-users" mailing list |
| 8:09AM |
0 |
[Fwd: Re: NFAS Not Passing Audio on B-chan 48, 72, 96] |
| 7:17AM |
0 |
IAX2 outgoing calls delayed before they connect |
| 7:08AM |
6 |
Asterisk + E1 with MFC/R2 in Argentina? |
| 6:56AM |
1 |
cisco 7960 not registering after * restart |
| 6:44AM |
4 |
NFAS Not Passing Audio on B-chan 48,72,96 |
| 6:07AM |
0 |
Digium TE405 card and Matra PBX |
| 5:49AM |
1 |
sending fax with chan-capi |
| 5:37AM |
0 |
"Guest" SIP-Invites not accepted |
| 5:25AM |
1 |
MGCP stuff |
| 5:16AM |
1 |
user address format |
| 3:22AM |
1 |
SIP fails when internet connection lost. |
| 3:11AM |
0 |
Voicemail app. not working... |
| 3:06AM |
0 |
Hicom 150 -- BRI -- Asterisk |
| 1:08AM |
0 |
Asterisk 1.4.0 compile error on AMD64 Opteron server; recompile with -fPIC? |
| 12:14AM |
3 |
Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again |
| 12:00AM |
4 |
Psst... Top secret information: Codename Pineapple |
| |
| Tuesday October 10 2006 |
| Time | Replies | Subject |
|---|
| 11:54PM |
5 |
Billing |
| 11:06PM |
0 |
Cubix / Firefly softphone and Asterisk |
| 10:17PM |
1 |
RE: Welcome to the "asterisk-users" mailing list |
| 10:02PM |
1 |
WRT54GP2 provisioning |
| 8:25PM |
2 |
E164 caller ID |
| 6:05PM |
1 |
how can I detect a DTMF tone while on a bridged call ? anyone knows? |
| 4:35PM |
0 |
asterisk crash in res_features.c |
| 3:42PM |
1 |
Strange FXS disconnection problem. |
| 3:13PM |
1 |
Free copy of "TrixBox Made Easy" |
| 2:58PM |
2 |
Change the background of a conversation |
| 2:07PM |
1 |
voicemail issue |
| 2:03PM |
3 |
Understanding NAT Traversal |
| 1:46PM |
0 |
Destar 0.2.0 released |
| 1:42PM |
1 |
Looking for AudioCodes 2 port FXS gateway - Asterisk compatability info |
| 1:17PM |
28 |
How big is *your* dialplan?? |
| 1:07PM |
2 |
RE: Welcome to the "asterisk-users" mailing list |
| 12:34PM |
1 |
Hangup or busy when the peer answer outgoing calls |
| 12:17PM |
2 |
Increase VoiceMail Messages Recording Gain - Audio Calls are Ok |
| 11:45AM |
1 |
transfer from VM to Cell Phone |
| 11:32AM |
1 |
1.4 and slow sound playback |
| 10:33AM |
5 |
Cisco CCM - Asterisk |
| 10:28AM |
3 |
sequential Dial() commands |
| 10:20AM |
1 |
Mitel 5224/SIP no MWI |
| 10:16AM |
2 |
Connection question... |
| 9:38AM |
0 |
bristuff problem? |
| 8:30AM |
10 |
Voicemail Press '0' |
| 8:26AM |
1 |
Asterisk 1.2.12.1 and snom 360 6.2.3 no audio |
| 8:19AM |
0 |
Xorcom TS-1 and Digium TE110P or TE210P |
| 8:01AM |
2 |
whisper paging |
| 7:43AM |
4 |
Inbound Callcenter with multiple DIDs |
| 7:28AM |
0 |
FYI - Polycom SoundPoint IP 301 Denial of Service] |
| 5:03AM |
2 |
alive check for HA constellation |
| 4:08AM |
0 |
Tutorial: Simple queue and agent debug monitoring |
| 4:07AM |
1 |
help this.... |
| 3:37AM |
8 |
single conference, multiple numbers |
| 1:33AM |
1 |
OT: Hand free solution recommandation |
| |
| Monday October 9 2006 |
| Time | Replies | Subject |
|---|
| 7:18PM |
3 |
T1 Passthrough |
| 5:41PM |
1 |
Echo Cancel Cards |
| 5:28PM |
2 |
Monitor Current outgoing calls |
| 4:57PM |
1 |
Error loading Unicall |
| 4:17PM |
0 |
ooh323 error |
| 3:53PM |
3 |
Home Hardware SIP Proxy with use of POTS in Emergency |
| 3:30PM |
1 |
Problem compiling libmfcr2.0.0.2 on Fedora Core 5 |
| 3:27PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday October 14th 2006 - 10:30am |
| 2:13PM |
1 |
Function ENUMLOOKUP |
| 2:00PM |
1 |
Asterisk RT on Disk On Module PerformanceandDurability |
| 1:41PM |
1 |
how to play pre-recorded file in meetme conference |
| 1:19PM |
3 |
Asterisk RT on Disk On Module Performance andDurability |
| 12:51PM |
0 |
problem in ooh323 |
| 11:23AM |
1 |
Cisco 7970 Unbootable After FW Upgrade |
| 10:47AM |
1 |
GotoIfTime - much slowdown with 90 conditions? |
| 10:47AM |
0 |
External domain |
| 10:30AM |
0 |
Is there a way to collect dtmf digits during a call? (inband) |
| 9:59AM |
2 |
Number Range |
| 9:43AM |
2 |
Range Operator |
| 9:37AM |
2 |
connecting multiple servers with iax - authentication fails |
| 8:52AM |
0 |
PRI TON/pridialplan digit prefixing |
| 8:30AM |
3 |
VOIP with PSTN backup |
| 8:24AM |
1 |
isdn cross-over ... |
| 7:54AM |
0 |
Get user context from dialplan. |
| 7:35AM |
0 |
how to play background music |
| 7:34AM |
3 |
Lots and lots of log files |
| 6:49AM |
1 |
AstriCon Dallas in Two Weeks |
| 6:36AM |
0 |
H323 <-> SIP |
| 3:49AM |
1 |
SIP vz IAX... |
| 2:35AM |
0 |
Beronet card strange log messages |
| 1:20AM |
1 |
Redefinition of transfer |
| 12:57AM |
1 |
Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate |
| |
| Sunday October 8 2006 |
| Time | Replies | Subject |
|---|
| 10:23PM |
3 |
password for vm users |
| 10:04PM |
2 |
Polycom 601 & Expansion Module: Light the LEDs??? |
| 9:07PM |
2 |
CDR - mysql with asterisk 1.2.12 not working |
| 8:58PM |
1 |
question about astdb |
| 8:40PM |
1 |
Asterisk Load balancing |
| 8:30PM |
2 |
How to make this easier |
| 8:20PM |
0 |
TDM22B |
| 7:42PM |
1 |
polycom reboot script |
| 7:01PM |
0 |
SIP vs. SIP-B |
| 6:45PM |
1 |
Blacklist to check http://whocalled.us |
| 9:48AM |
3 |
Tellabs and a PRI |
| 9:37AM |
5 |
PRI issues |
| 9:13AM |
0 |
USA Origination recommended service? |
| 7:44AM |
1 |
DID is not working (call is not routing) |
| 6:27AM |
0 |
OH323 Fake ring |
| 2:54AM |
0 |
Sun Cluster and Asterisk |
| 2:02AM |
0 |
Transfer app and DTMF via SIP info |
| 1:05AM |
2 |
How can i store PAP2 or any device config in Asterisk |
| |
| Saturday October 7 2006 |
| Time | Replies | Subject |
|---|
| 10:30PM |
0 |
polycom auto cfg file |
| 10:30PM |
6 |
ftp server |
| 8:46PM |
1 |
Real-time and priority "n" |
| 6:37PM |
0 |
Asterisk: Can anybody forward anybody's extension? |
| 3:30PM |
2 |
disabling hardware echo can on tdm2400p |
| 2:15PM |
2 |
Xorcom Astribank and 64 bit linux |
| 11:33AM |
1 |
Requirements for Asterisk & SER integration |
| 9:20AM |
1 |
G729 Licence Consumption Problem |
| 7:51AM |
1 |
Outbound FXO call, getting "You must first dial..." |
| 5:48AM |
1 |
AEL2 Catching on? |
| 1:51AM |
2 |
SIP stuck channel soft hangup? |
| |
| Friday October 6 2006 |
| Time | Replies | Subject |
|---|
| 10:39PM |
1 |
astcc help-pleasssssseeee |
| 9:40PM |
1 |
Options for moving to * friendly Business VSP |
| 6:13PM |
1 |
HTTP Connection Closed on 7960 SIP |
| 5:40PM |
1 |
A Call centre module on Asterisk |
| 5:21PM |
1 |
Asterisk access Postgres for Realtime Configuration |
| 5:16PM |
3 |
regexten & regcontext broken for SIP? |
| 5:11PM |
0 |
commercial asterisk |
| 4:04PM |
0 |
Codes negotiation problems betweenAsterisk1.4beta2 and Aastra 480i |
| 3:27PM |
2 |
Odd echo issue with speaker phone |
| 2:03PM |
3 |
ChanIsAvail() in 1.2.12.1 |
| 1:43PM |
1 |
swap CID with DID |
| 1:27PM |
2 |
AGI() in 1.2 and 1.4 |
| 12:38PM |
0 |
Asterisk Postgres Native support |
| 10:49AM |
2 |
Voicemail MWI |
| 10:46AM |
0 |
Match & Chat Author? |
| 9:54AM |
2 |
Asterisk RT on Disk On Module Performance and Durability |
| 8:26AM |
2 |
Voicemail and Forwarding |
| 8:07AM |
0 |
defining trunks in sip.conf |
| 8:02AM |
1 |
Tutorial - avoiding queue_log file rotation |
| 4:19AM |
1 |
Asterisk act as a proxy ? |
| 4:07AM |
2 |
How to forward DID to another Server |
| 3:46AM |
2 |
2x* and realtime |
| 2:39AM |
1 |
New Asterisk StumbleUpon Group |
| 1:55AM |
1 |
asterisk gui sans live cd |
| |
| Thursday October 5 2006 |
| Time | Replies | Subject |
|---|
| 11:46PM |
1 |
Asterisk CDR |
| 11:29PM |
3 |
Asterisk Server : IDE HDD frequent crash |
| 8:00PM |
0 |
different dialtones for DISA |
| 7:22PM |
3 |
Newbie h/w Q, and confirming basic concepts |
| 7:01PM |
3 |
Optus PRI via DSL |
| 6:31PM |
4 |
No Dialtone |
| 6:30PM |
1 |
odd muting issue |
| 5:04PM |
1 |
failed registration |
| 4:30PM |
1 |
IVR menu system external database information collection |
| 4:24PM |
7 |
Asterisk@Home problems |
| 3:41PM |
1 |
DOA IAXy? |
| 3:07PM |
1 |
AW: PoE IP Phone |
| 2:26PM |
1 |
How to create echo for learning purpose |
| 2:05PM |
0 |
AGI PHP |
| 1:54PM |
0 |
ACK when using SIP Proxy |
| 1:15PM |
1 |
Codes negotiation problems between Asterisk 1.4beta2 and Aastra 480i |
| 1:06PM |
0 |
Dial without phone |
| 11:58AM |
2 |
No voice for when using Playback and background |
| 11:56AM |
1 |
Problem with 2 machines connected with IAX |
| 11:53AM |
0 |
Help with gdb bt full results |
| 11:38AM |
8 |
PoE IP Phone |
| 11:26AM |
3 |
OT: Polycom time sync - sorta |
| 10:42AM |
2 |
pop a web page with DID in url |
| 10:24AM |
0 |
we are having trouble detecting the # for making a transfer from an E1, usually under some load, please help |
| 10:16AM |
2 |
Getting Asterisk to work with GoogleTalk |
| 9:22AM |
0 |
sdsl |
| 8:17AM |
0 |
Dial out trhough a FXS channel on a TDM card |
| 7:54AM |
0 |
Re:UPDATE: Zaptel problems |
| 7:19AM |
0 |
Detecting Busy AGI Extensions? |
| 7:11AM |
3 |
AW: asterisk-users Digest, Vol 27, Issue 23 |
| 7:10AM |
0 |
for god's sake somebody help me! ANSWEREDTIME=0 in astcc!! |
| 7:08AM |
2 |
Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute |
| 6:51AM |
2 |
two asterisk and one NBX system |
| 5:58AM |
0 |
GXP - 2000 BLF |
| 5:55AM |
1 |
Call Center requirements |
| 5:35AM |
0 |
India:Reliance - E1configuration using TE110P |
| 5:27AM |
1 |
Extremely choppy sound on some of our POTSnetwork calls; goes away with mute |
| 4:43AM |
0 |
New Version of "Tycho" Voicemail Manager released |
| 2:47AM |
1 |
Message count requested for mailbox 9002@from-sip but voicemail not loaded. |
| 2:14AM |
1 |
Problems with Dial In - Dial Out via SIP - no voice |
| 1:51AM |
0 |
Silience on random calls |
| 1:47AM |
0 |
spandsp logs? |
| 12:54AM |
4 |
"set verbose 4" in SVN trunk? |
| |
| Wednesday October 4 2006 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
fonality acquires trixbox (asterisk@home) ? |
| 11:25PM |
2 |
asterisk-addons-1.2.4 Installation Problem |
| 10:35PM |
1 |
CDR problem with call transfer |
| 10:21PM |
0 |
echo cancellation on hard phones |
| 10:03PM |
1 |
AEL2 #include madness in Asterisk 1.4 - Murf? |
| 5:53PM |
1 |
Clipcomm CG 410 / FXO Gateway |
| 5:38PM |
1 |
ZAP chanel doesn't reset if external caller hangs up in menu |
| 5:03PM |
0 |
Possible to set max voicemail mesasge limit per user |
| 4:51PM |
6 |
Bandwidth requirements |
| 4:11PM |
0 |
snom 360 - how to make record button working ? |
| 3:39PM |
1 |
Re: [asteisk-users]USA DID + trunk |
| 2:58PM |
4 |
no callerid from PSTN using TDM2400P |
| 2:49PM |
3 |
Calling Functions from AEL2 |
| 2:43PM |
2 |
MODEM (data) througt asterisk ? |
| 2:42PM |
2 |
How to make RTP does not go thru asterisk server |
| 1:18PM |
1 |
TNT Max Password reset |
| 1:17PM |
3 |
Voicemail maintenance |
| 12:27PM |
0 |
Intrado V9-1-1 |
| 11:27AM |
3 |
Newbie question about meetme |
| 11:24AM |
4 |
Dialplan Syslog |
| 11:01AM |
2 |
Need USA DID + trunk provider |
| 10:57AM |
0 |
Intel Chipset 945p compatible? |
| 10:37AM |
2 |
Wouldn't Tri-tone detection in Dial() be cool? |
| 10:35AM |
1 |
SIP client that runs on Linux or Solaris through X Windows? |
| 10:33AM |
1 |
snom 360: how to make record button working ? |
| 10:10AM |
3 |
New tutorial - peering two * servers using IAX |
| 10:01AM |
4 |
Transfer feature - howto? |
| 9:53AM |
1 |
digium compatibility notes |
| 9:36AM |
0 |
Asterisk 1.4 moh - mohsuggest |
| 8:50AM |
0 |
FOP v.27 IAX trunks not "ringing" |
| 8:32AM |
3 |
[Asterisk-Java] SipShowPeerAction |
| 7:05AM |
5 |
Where is the PlayDTMF command? |
| 6:53AM |
1 |
Spandsp and tif |
| 4:45AM |
1 |
voicemail maintenance questions |
| 4:11AM |
1 |
Help in MySQL + Asterisk. |
| 3:44AM |
1 |
verbose logging to file in 1.4 |
| 3:16AM |
0 |
Rejecting call |
| 2:32AM |
0 |
Asterisk and Attachment |
| 12:43AM |
2 |
DISA and legacy PBX |
| |
| Tuesday October 3 2006 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
Call Forwarding not working for extension in queue, why? |
| 10:51PM |
1 |
Strange problem(Munin-node-1.2.4-7) |
| 8:22PM |
1 |
(no subject) |
| 3:30PM |
2 |
Digium Interfaces in Tampa? |
| 3:24PM |
1 |
Asterisk Directory listing |
| 3:18PM |
1 |
authenticating forwarded calls |
| 3:11PM |
0 |
AstmanProxy Not Collating Manager Info |
| 2:09PM |
1 |
CDR stats to one mysql database, multiple webstats packages |
| 1:23PM |
1 |
Digium TDM or SPA-3000? |
| 11:32AM |
2 |
Extremely choppy sound on some of our POTS network calls; goes away with mute |
| 11:01AM |
1 |
uniden uip200 phone hangs any ideas? |
| 10:13AM |
3 |
CALEA support within asterisk? |
| 9:19AM |
0 |
Asterisk+Panasonic KX-TDA100+zaphfc NT link problem |
| 8:58AM |
0 |
o extension for voicemail app |
| 8:55AM |
6 |
asterisk to asterisk DID extentions |
| 8:21AM |
0 |
Cisco 7961 - Presence Example? |
| 8:10AM |
1 |
Caller ID on Zap not always working |
| 7:40AM |
1 |
R: Zaphfc woth florz patch |
| 6:30AM |
2 |
SV: Screen pop based on incoming DID |
| 6:27AM |
0 |
Query on Call Parking |
| 6:20AM |
2 |
Problems with automon |
| 6:05AM |
0 |
Zaphfc woth florz patch |
| 5:43AM |
2 |
Screen pop based on incoming DID |
| 5:39AM |
0 |
How to add new codec support? |
| 4:30AM |
1 |
Defining sip users through mysql |
| 4:02AM |
1 |
sip provider not working |
| 3:53AM |
1 |
Which IP Phone is good to use at reception desk? |
| 3:26AM |
0 |
realtimeupdate error |
| 2:07AM |
0 |
pbx call setup to asterisk, behavior context dependend |
| 1:40AM |
0 |
[ast-users] bridging active channels together |
| 1:39AM |
0 |
ZyXEL desktop ethernet switch for QoS |
| |
| Monday October 2 2006 |
| Time | Replies | Subject |
|---|
| 9:30PM |
1 |
tools/techniques/metrics for measurement of end-point quality |
| 8:59PM |
1 |
TDM2400P wiring. |
| 5:01PM |
6 |
Polycom Buddy Watch Broken with 2.0.1 Firmware? |
| 3:12PM |
0 |
Trunks and Outbound Routes |
| 2:33PM |
0 |
Problems with Tormenta 2 quad card |
| 2:01PM |
1 |
Configuration / dialplan problem |
| 12:58PM |
0 |
Minexpiry time - how to set this |
| 11:37AM |
2 |
Passing Arguments to FastAGI |
| 11:03AM |
1 |
g729 Codec for AMD Sempron |
| 10:24AM |
0 |
Conversations Mix |
| 10:13AM |
0 |
!! No channel map, no channel, and no ds1? What am I supposed to identify? |
| 9:54AM |
3 |
t1 voip to failover pri |
| 8:47AM |
0 |
Asterisk 1.2.10 and SCCP |
| 8:32AM |
0 |
480i phone: Is there a trick to registering with *?? <--Solved, first impressions |
| 8:12AM |
0 |
480i phone: Is there a trick to registering with *?? <--Solved |
| 7:21AM |
0 |
asterisk queues with SER, aka "sip show peers" |
| 6:50AM |
1 |
Issues with calling certain phone numbers... |
| 6:46AM |
0 |
Dial and connect to sip provider works, but no audio. |
| 5:06AM |
2 |
attended transfer unreliable (2nd try) |
| 4:23AM |
1 |
Spying a channel in a meetme |
| 4:04AM |
2 |
can't transcode ilbc |
| 3:45AM |
1 |
Call Quality / Echo / Problems |
| 3:19AM |
0 |
Can't get second line of Sangoma A200 to work. |
| 1:57AM |
0 |
suggest a configuration |
| 1:03AM |
3 |
Siemens Hipath <-> asterisk, pri problem |
| 12:46AM |
0 |
I: Sip answer one side , ring other side |
| |
| Sunday October 1 2006 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
Bristuff vs. vISDN vs. mISDN for hfc card ? |
| 10:31PM |
0 |
G729 Codec Loading Error |
| 8:33PM |
1 |
G726 prompts |
| 5:14PM |
0 |
polycomphone (SOLVED) |
| 3:15PM |
0 |
Another Issue with 1.4 |
| 8:32AM |
1 |
detecting busy on queue transfer |
| 5:28AM |
3 |
WiFi SIP handset with Bluetooth required |
| 4:27AM |
1 |
polycomphone |
| 1:39AM |
0 |
Real-time LDAP config |