| Saturday January 31 2009 |
| Time | Replies | Subject |
|---|
| 10:40PM |
2 |
Avaya and Asterisk sound one-way |
| 7:10PM |
1 |
asterisk-users Digest, Vol 54, Issue 107 |
| 5:37PM |
3 |
Zapata.conf |
| 4:47PM |
6 |
Quiet 24 port POE gig switch |
| 4:04PM |
1 |
iax clients were unregistered after 30sec |
| 1:37PM |
3 |
Is http://downloads.digium.com/pub/ down??? |
| 10:24AM |
1 |
where to find STUN Server howto |
| 7:50AM |
1 |
Call without Answer |
| 3:24AM |
1 |
Ideas on how to convert spoken name to text (or wav to text)..speech recognition software? |
| |
| Friday January 30 2009 |
| Time | Replies | Subject |
|---|
| 8:04PM |
0 |
Can't hear audio when Playback(something, noanswer) on Zap |
| 6:32PM |
0 |
Duplicate Radius accounting in Asterisk. |
| 5:17PM |
2 |
Asterisk with Avaya |
| 5:11PM |
1 |
Asterisk VoiceMail: Is there a web interface for checking voicemail? |
| 3:59PM |
2 |
SIP.Conf - bindaddr per peer? |
| 3:26PM |
4 |
TAPI and Asterisk |
| 2:08PM |
3 |
How to elegantly set DEVSTATE values after restarting |
| 1:43PM |
0 |
Use of Re-INVITE and t,T (transfer) options |
| 12:26PM |
0 |
Friday Jan 30th at 12 Noon EST: Design the Phone of the Future |
| 7:23AM |
1 |
Where to find db1_dump185 in debian packages ? |
| 1:56AM |
3 |
looking for a link or pdf ot something about opensip/openser and load balancing |
| |
| Thursday January 29 2009 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
Asterisk 1.6.1 Release Candidate 1 Now Available |
| 11:55PM |
2 |
manager API with no login? |
| 7:29PM |
2 |
Can I use an interact and visa terminal through VoIP? |
| 6:15PM |
1 |
early dial: asterisk and ATA |
| 6:15PM |
3 |
32 bit server is ok? |
| 5:23PM |
2 |
Eyebeam or Xlite |
| 4:19PM |
1 |
Managing codecs |
| 4:00PM |
1 |
howto configure an asterisk to send credentials in a REGISTER message to another asterisk |
| 3:30PM |
2 |
RTP/NAT Traffic to private IP |
| 11:54AM |
1 |
blind transfer on hook-flash from SIP phone |
| 11:01AM |
2 |
Don't get asterisk to run behind NAT router |
| 10:47AM |
5 |
Wanted information |
| 10:32AM |
0 |
Benqtelecom in cdr log |
| 9:36AM |
1 |
CDR when replaces |
| 3:30AM |
0 |
[asterisk-dev] DTMF queuing |
| 2:47AM |
2 |
GTalk Channel |
| 1:15AM |
9 |
Callback / Camp / Extention Free notify? |
| |
| Wednesday January 28 2009 |
| Time | Replies | Subject |
|---|
| 11:40PM |
2 |
How to retrieve a phone number from call forwarding? |
| 10:40PM |
1 |
E1 conection to a Cisco2600 |
| 7:59PM |
1 |
asterisk-users Digest, Vol 54, Issue 94 |
| 7:48PM |
1 |
Scope of variable |
| 7:01PM |
2 |
SIP Registrations broken on 1.4.22.1? |
| 6:43PM |
4 |
Call Recording Alias |
| 4:45PM |
0 |
problem joining a conference room |
| 4:27PM |
5 |
Inbound Call Disconnect in 3 seconds |
| 3:35PM |
1 |
Looking for SIP loud ringer |
| 3:30PM |
1 |
FAX |
| 10:18AM |
2 |
Zapatel early media issue |
| 9:45AM |
1 |
Record and then Read does not found file |
| 9:19AM |
4 |
route based from source |
| 8:10AM |
1 |
dahdi echocancel configuration for dahdi_dummy? |
| 6:56AM |
2 |
Improving asterisk documentation - sources and what the community can do |
| 6:52AM |
0 |
How many bounces does it take before you get unsubscribed? |
| |
| Tuesday January 27 2009 |
| Time | Replies | Subject |
|---|
| 11:46PM |
1 |
Asterisk & Twitter - Release/Announce only 'channel' ? |
| 10:53PM |
2 |
Module res_odbc is not loading |
| 10:21PM |
1 |
dialstatus through a call file |
| 6:38PM |
2 |
Muted sound on a Linksys 962 |
| 5:49PM |
12 |
USA BRI -- any hope at all? |
| 4:55PM |
0 |
Can't start Asterisk after installing Digium G729 licence [SOLVED] |
| 4:46PM |
1 |
RFC -- Improving the quality of the mailinglists |
| 4:04PM |
0 |
Queue time to answer/abandon + OrderlyStats Server Edition. |
| 3:57PM |
2 |
RFC -- Improving the quality of the mailing lists |
| 3:47PM |
2 |
T.38 |
| 3:45PM |
4 |
Asterisk 1.6 dahdi only? |
| 3:31PM |
0 |
SPA-3102 in India - Problem dialing out PTSN |
| 3:17PM |
1 |
Asterisk - Nortel integration via SIP protocol |
| 2:59PM |
1 |
Webcall app needed |
| 12:53PM |
1 |
G726 Codec |
| 11:25AM |
1 |
Queue time to answer/abandon |
| 10:30AM |
2 |
server sizing for ~ 200 simultaneous call |
| 9:24AM |
0 |
asterisk-users Digest, Vol 54, Issue 83 |
| 7:48AM |
1 |
Can't start Asterisk after installing Digium G729 licence |
| 6:20AM |
0 |
hangup problem(for spa400) |
| 3:14AM |
0 |
Help with cdr_odbc |
| |
| Monday January 26 2009 |
| Time | Replies | Subject |
|---|
| 10:28PM |
1 |
Document with differences between 1.2, 1.4 and 1.6? |
| 10:20PM |
3 |
I need help |
| 9:12PM |
1 |
Dial weirdness |
| 7:12PM |
2 |
General Asterisk SIP/IAX provider question |
| 4:22PM |
0 |
goto iax problem |
| 2:51PM |
1 |
Strange Cisco/Asterisk anomaly |
| 2:43PM |
1 |
Suggestion for a new server for E1 line |
| 1:57PM |
1 |
* Queues with legacy pbx extensions ? |
| 12:53PM |
2 |
Network Card |
| 12:08PM |
3 |
Digium TE220 card partially detected |
| 11:54AM |
7 |
Auto Detect |
| 11:28AM |
2 |
German date format in voicemail emails |
| 9:40AM |
2 |
custom cdr userfiled |
| 9:26AM |
1 |
Voicemail |
| 8:32AM |
5 |
Start asterisk on boot |
| 12:20AM |
2 |
dialplan/config basics - analog hangup on keypress |
| |
| Sunday January 25 2009 |
| Time | Replies | Subject |
|---|
| 11:02PM |
2 |
Zaptel transfer using any button or code, but not flash hook |
| 7:02PM |
1 |
simple dial plan - brain dead operator |
| 6:55PM |
5 |
soft phone |
| 4:34PM |
10 |
CentOS and BAT File |
| 2:08PM |
2 |
how to build a small asterisk pbx |
| 9:23AM |
5 |
Ntework Card |
| 1:48AM |
2 |
Choppy Sound On Bridging From SIP->IAX |
| 1:23AM |
2 |
asterisk help |
| 12:12AM |
2 |
monitoring SIP connection |
| |
| Saturday January 24 2009 |
| Time | Replies | Subject |
|---|
| 11:38PM |
3 |
no dial tone tdm400p |
| 10:26PM |
1 |
interesting comment. New Physics? |
| 9:08PM |
2 |
Dahdi Init script for Suse? |
| 7:30PM |
2 |
Zaptel? Dahdi? |
| 6:45PM |
0 |
Having tone in my fxs, and loading the zaptel |
| 6:18PM |
0 |
idle-url for Cisco 7940 using Sip |
| 5:32PM |
2 |
NAT router for Linux |
| 4:00PM |
2 |
Reading/Writing the Astdb |
| 2:00PM |
1 |
Which policy for ISDN BRI support in NT/PtMP ? |
| 11:50AM |
1 |
Asterisk freezes with Fixup failed on channel SIP/...<MASQ> |
| 6:46AM |
1 |
local dialing |
| 5:49AM |
0 |
unistim - no dial tone frequecy, no number display when dialing |
| 4:52AM |
0 |
unistim only recognize "default" context |
| 2:57AM |
3 |
Nortel IP phone i2002 - DHCP server unreachable |
| 1:13AM |
3 |
Passing DTMF |
| 12:22AM |
1 |
Logging outgoing calls |
| |
| Friday January 23 2009 |
| Time | Replies | Subject |
|---|
| 9:55PM |
1 |
Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released |
| 9:46PM |
2 |
Long Delay after sip reload command |
| 7:23PM |
2 |
Cant Find |
| 6:35PM |
2 |
sip based fax |
| 5:21PM |
1 |
OSLEC, no echo, but a big noise in line |
| 9:00AM |
3 |
OT - Is Netgear ProSafe FS108P with PoE silent ? |
| 6:58AM |
1 |
Trying to do a transfer in agi |
| 5:26AM |
3 |
Packet8 hacked |
| |
| Thursday January 22 2009 |
| Time | Replies | Subject |
|---|
| 11:40PM |
0 |
Psssst - hey buddy, wanna' get a job? (follow-up to asterisk-biz please) |
| 10:16PM |
2 |
Dumb question: retrieve values from OS-level commands? |
| 9:22PM |
2 |
Looking for Asterisk admin or related job |
| 8:53PM |
1 |
Newbie in Cisco Phone |
| 8:11PM |
2 |
random Linksys question |
| 7:47PM |
0 |
Asterisk 1.6.0.4 Release Candidate 1 Now Available |
| 6:57PM |
6 |
Vicidialnow |
| 6:39PM |
0 |
Friday Jan 23 at 12 Noon EST: Open Source vs Commercial |
| 5:22PM |
1 |
(Fwd) New problem: "They disconnect your service for no reason |
| 3:52PM |
7 |
Root Password not taking |
| 3:21PM |
2 |
registration problem using asterisk 1.6 |
| 2:28PM |
1 |
Zap connection problem |
| 1:55PM |
2 |
Incoming fax detection on mISDN hfcmulti B410P card |
| 11:10AM |
0 |
Fw: Re: mISDN BRI Asterisk 1.4 |
| 11:01AM |
0 |
Query About Asterisk 1.6.0.1 Dialog Event Package. |
| 9:05AM |
1 |
Help with Avaya integration |
| 8:16AM |
0 |
Few of my phones do not ring when in a queue? |
| 2:31AM |
0 |
DTMF queuing problems |
| 12:35AM |
1 |
oslec + dahdi |
| |
| Wednesday January 21 2009 |
| Time | Replies | Subject |
|---|
| 10:42PM |
1 |
SIP realtime status... |
| 10:30PM |
1 |
recording failed |
| 9:32PM |
0 |
Polycom SoundPoint IP 500 + X100P + Sirrix PCI4S0 + Conrad HFC-S cards |
| 9:19PM |
6 |
soft ATA on linux with zaptel? |
| 8:52PM |
0 |
g729 with Cisco gateways? |
| 8:14PM |
0 |
Asterisk 1.4.23 Now Available! |
| 6:29PM |
0 |
Playfile to both legs of call |
| 6:14PM |
3 |
snap a number now digium? |
| 4:01PM |
3 |
Need Help |
| 3:17PM |
1 |
Fw: Re: mISDN BRI Asterisk 1.4 |
| 11:46AM |
1 |
No Ring on Analog Phone using Rhino Channel Bank in China |
| 10:48AM |
4 |
integration with Microsoft CRM? |
| 10:15AM |
2 |
CDR 0.00 duration |
| 9:33AM |
1 |
Asterisk On Solaris Real Time |
| 9:10AM |
0 |
About Asterisk 1.6.0.1 |
| 9:00AM |
0 |
Prob on DISA |
| 6:37AM |
0 |
Job description |
| 4:58AM |
1 |
Asterisk queues sending calls to members on the phone |
| |
| Tuesday January 20 2009 |
| Time | Replies | Subject |
|---|
| 11:04PM |
2 |
PAP2T provisioning |
| 10:02PM |
2 |
extensions.conf -- what to do when command throws errors? |
| 9:30PM |
0 |
Timestamp on voice mail messages is based on wrong timezone |
| 8:58PM |
3 |
Using centos and kickstart to build a minimum installation |
| 8:30PM |
1 |
Problem with TDM808 |
| 7:07PM |
0 |
Call Dropped in Voicemail / No Reply to Our Critical Packet w/ SIP Debug |
| 6:49PM |
3 |
dead sip channel |
| 5:30PM |
1 |
Setting up an outgoing trunk group |
| 4:56PM |
0 |
Stutter/chopoff first audio played |
| 3:40PM |
0 |
Outgoing CallerID w. DAHDI on ISDN BRI |
| 3:08PM |
2 |
Why does Asterisk not hangup? |
| 2:46PM |
0 |
channel var for Call on hold? |
| 2:32PM |
1 |
CallerID ANI issues |
| 1:29PM |
0 |
Hang up detection problems |
| 1:04PM |
0 |
X-Lite and Asterisk RTP cutting out |
| 11:40AM |
1 |
Called's channel |
| 11:30AM |
1 |
Siemens S685IP registration problems |
| 11:16AM |
3 |
Forwarding calls and trasfer calls |
| 10:57AM |
2 |
SIP DTMF problem with SNOM |
| 10:18AM |
5 |
the FXS ports of Digium and damaging if connected to Tel Line |
| 9:42AM |
1 |
asterisk-users Digest, Vol 54, Issue 53 |
| 9:09AM |
1 |
Skype beta news ? |
| |
| Monday January 19 2009 |
| Time | Replies | Subject |
|---|
| 11:30PM |
4 |
Problems With Playback of Audio On SIP Only System |
| 9:03PM |
1 |
looking for Asterisk experts |
| 8:35PM |
3 |
[somewhat OT] seeking ideas/input for my thesis |
| 8:09PM |
1 |
Suggestions on how to create a hunt or hunt like (rollover, multi-line) group or where to get one? |
| 6:21PM |
3 |
Interesting observation |
| 4:56PM |
1 |
Need help registering Cisco 7960 Phones on Asterisk |
| 4:55PM |
1 |
Server freeze & kernel panic |
| 4:17PM |
1 |
Fring and Asterisk |
| 2:38PM |
0 |
Asterisk On Solaris |
| 1:49PM |
1 |
Cisco 7941G-GE with Asterisk and CTPSEP odyssee |
| 1:45PM |
1 |
How to overwrite CDR(dst) value in h priority? |
| 1:40PM |
0 |
How to add SipAddHeader in outgoing call file. |
| 1:11PM |
3 |
adding numbers in dialplan |
| 12:41PM |
3 |
IAX IP Phone |
| 12:03PM |
6 |
G729 codec |
| 11:51AM |
1 |
indications.conf entry for Iceland |
| 11:26AM |
1 |
how to cancel new recorded message from voicemail menu? |
| 11:08AM |
3 |
followme order field |
| 10:10AM |
4 |
Description of Zaptel/DAHDI E1 alarms |
| 6:15AM |
0 |
Asterisk and PhoneControl |
| |
| Sunday January 18 2009 |
| Time | Replies | Subject |
|---|
| 7:43PM |
3 |
Using a sidecar? Ideas? |
| 6:27PM |
0 |
Asterisk T.38 Passthrough + T38Modem/Hylafax - has anyone had luck with this? |
| 4:28PM |
2 |
Recordin call in asterisk |
| 3:57PM |
0 |
BRI on Solaris/SPARC |
| 12:37PM |
2 |
DAHDI trouble (again) Unable to open master device '/dev/zap/ctl' |
| 12:19PM |
0 |
is multiple contexts in alsa.conf possible |
| 4:48AM |
1 |
caller ID - handle_request_invite: Failed to authenticate user |
| 2:07AM |
0 |
ast_yyerror() |
| |
| Saturday January 17 2009 |
| Time | Replies | Subject |
|---|
| 10:06PM |
1 |
canreinvite per route |
| 9:39PM |
1 |
Sip Trunk registration |
| 7:40PM |
0 |
asterisk support for multiply (two) console dsp devices |
| 6:52PM |
2 |
Call file in the future |
| 4:51PM |
3 |
Asterisk 1.6 T38 to G711 transcoding is this possible? |
| 11:37AM |
1 |
compare Linksys SPA8000 and Grandstream GXW4008 |
| |
| Friday January 16 2009 |
| Time | Replies | Subject |
|---|
| 10:47PM |
2 |
UpdateConfig : Appending line fails |
| 8:26PM |
0 |
Asterisk 1.4.23-rc4 Now Available |
| 8:14PM |
0 |
Crickets. Yes, crickets. |
| 8:05PM |
1 |
ATA gateway with 2 ethernet interfaces |
| 7:58PM |
0 |
about hardware |
| 6:53PM |
2 |
mini-PCI FXS card? |
| 6:12PM |
2 |
CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working. |
| 6:07PM |
1 |
Voicemail message is dialtone |
| 5:42PM |
3 |
How to hangup a call manually... |
| 5:21PM |
1 |
Dialing from E1/T1 |
| 4:38PM |
4 |
Remote RTP |
| 3:52PM |
4 |
Snom 300 vs Grandstream gxp |
| 3:49PM |
1 |
pstn hangs up: MWI no message waiting ?? |
| 12:27PM |
0 |
Can not fetch SIP_HEADER incase of Transfer |
| 12:17PM |
0 |
dialing trunk-to-trunk |
| 12:13PM |
0 |
dialing trunk to trunk |
| 11:13AM |
2 |
want to add SipAddHeader in call out file |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 4:29AM |
0 |
No subject |
| 3:56AM |
1 |
CRTC and FCC Feeds |
| 1:25AM |
1 |
Asterisk Upgrade |
| 1:00AM |
1 |
Portech MV-378 with Asterisk |
| 12:33AM |
0 |
ISDN and routers... |
| 12:29AM |
0 |
gtalk and jingle again... |
| |
| Thursday January 15 2009 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
multiple registration to sip trunking provider. |
| 11:15PM |
2 |
How to transfer a call from one Asterisk Server to another |
| 10:29PM |
1 |
Broadcast Phone system (for radio) |
| 8:45PM |
1 |
how to debug mime-construct with fax2mail? |
| 7:52PM |
0 |
Voicetronix Openswitch 12 + echo problem |
| 7:42PM |
0 |
Warning in CLI: Inringing for peer [PEER] < 0 |
| 7:40PM |
0 |
Up To 20% OFF At Our Signature Style Event + Holiday Weekend Clearance |
| 7:00PM |
2 |
Asterisk - Trixbox |
| 6:02PM |
1 |
Patton SmartNode 4638 and ISDN2e |
| 5:11PM |
2 |
Digium TE220 supported protocol |
| 3:34PM |
1 |
problem with PlayDTMF: no error but no tone |
| 2:44PM |
1 |
R2 |
| 11:11AM |
6 |
Call Stealing |
| 4:09AM |
1 |
call transfer in CDR |
| 2:31AM |
2 |
Has anyone used FaxGateway() |
| 1:13AM |
2 |
OT - Differences between modprobe and insmod |
| 12:18AM |
2 |
Dropping this SIP message, it's incomplete |
| |
| Wednesday January 14 2009 |
| Time | Replies | Subject |
|---|
| 10:55PM |
2 |
Zap problems |
| 10:30PM |
1 |
1.6.1-b4: Can't get fax2mail work from System() |
| 10:25PM |
0 |
AMI API , Editing extensions.conf |
| 10:20PM |
0 |
Nortel files for bankruptcy protection |
| 6:41PM |
0 |
sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? |
| 6:32PM |
0 |
Strange IAX2 registration issue |
| 6:04PM |
0 |
agi and set variable ( accountcode in aserisk 1.4) |
| 5:11PM |
3 |
G.729.1 - any interest? |
| 1:02PM |
8 |
evaluate SIP response codes in dialplan |
| 9:33AM |
1 |
gxp2000 and no sound asterisk 1.6 |
| 9:18AM |
2 |
Any free video (or audio) softphone VOIP client under Linux with touchscreen friendly interface ? |
| 8:24AM |
2 |
Set caller ID to anonymous |
| |
| Tuesday January 13 2009 |
| Time | Replies | Subject |
|---|
| 11:58PM |
1 |
FWD and IPCall |
| 10:36PM |
1 |
Asterisk Appliance |
| 9:10PM |
9 |
FWD and Asterisk |
| 9:08PM |
0 |
test |
| 6:44PM |
0 |
Problem with overhead paging with Alsa and OSS |
| 6:03PM |
3 |
mISDN BRI Asterisk 1.4 |
| 5:40PM |
2 |
0800 UK number |
| 4:12PM |
0 |
[Re: CDR Rewrite -- Questions to the users] |
| 2:04PM |
2 |
404 not found from one ip-adress |
| 1:31PM |
2 |
Zaptel & multiple kernels |
| 1:28PM |
0 |
Realtime MOH |
| 9:20AM |
1 |
Dahdi caused Kernel to segfault |
| 8:29AM |
4 |
What are the various models of DID providers |
| 4:55AM |
1 |
cli reload error |
| 12:05AM |
1 |
Beware of DIDX & Super Technologies |
| |
| Monday January 12 2009 |
| Time | Replies | Subject |
|---|
| 10:28PM |
2 |
FXS Help Needed... |
| 9:15PM |
1 |
u-law file header ? |
| 7:26PM |
1 |
Upgrade to v.1.2.31 ... weird change |
| 6:51PM |
1 |
CDR Rewrite -- Questions to the users (Steve Murphy) |
| 5:15PM |
0 |
WCTDM/Zaptel memory leak |
| 5:03PM |
2 |
a zaptel problem |
| 4:53PM |
4 |
bug 14153 and svn checkout. |
| 3:51PM |
6 |
CDR Rewrite -- Questions to the users |
| 1:54PM |
1 |
bug(?) bandwidth problem |
| 1:05PM |
0 |
Transfer in Asterisk 1.6 |
| 12:39PM |
1 |
problem with dahdi and meetme |
| 12:34PM |
2 |
error messgae |
| 7:45AM |
0 |
no busy here |
| 5:32AM |
1 |
RTCP SR transmission error, rtcp halted |
| |
| Sunday January 11 2009 |
| Time | Replies | Subject |
|---|
| 7:17PM |
2 |
sip peer permit/deny - Need some explanation |
| 1:09PM |
2 |
asterisk 1.4 with h323 for debian |
| 12:06PM |
4 |
chan_sip on non-standard port 5062 - contact has no port |
| 5:35AM |
1 |
Use ZAP/Dahdi channel for outbound only... no inbound? |
| 12:33AM |
1 |
Configuring Linksys spa8000 devices through xml |
| 12:01AM |
2 |
hdmi an console dsp |
| |
| Saturday January 10 2009 |
| Time | Replies | Subject |
|---|
| 7:44PM |
2 |
How to monitor asterisk with SNMP? |
| 6:13PM |
1 |
Pay Phone Controller Project |
| 12:43PM |
0 |
line disconnected after 20 seconds no reply to our critical packet |
| 9:53AM |
1 |
Cisco VoIP QOS |
| 9:36AM |
1 |
Local channel Help required |
| 2:36AM |
3 |
Asterisk/GXW410x IP Analog Gateway |
| 12:52AM |
2 |
Lenny. Where to find zaptel patches |
| |
| Friday January 9 2009 |
| Time | Replies | Subject |
|---|
| 9:33PM |
8 |
Spurious hangups on Sangoma A102d, Trixbox 2.6.1 |
| 9:05PM |
2 |
Security communication dilemma: your help needed |
| 8:57PM |
1 |
fake ringback tone |
| 8:08PM |
1 |
Web Softphone |
| 7:20PM |
0 |
Fw: iax2 bindaddress: how to reload so iax2 can bind to an alias IP |
| 4:36PM |
5 |
lock SIP Account after too many failed logins |
| 3:33PM |
1 |
Queues, SIP channel and "In Use" |
| 12:40PM |
0 |
AmooCon - Call for Papers |
| 12:27PM |
1 |
slow ODBC reconnect |
| 11:36AM |
0 |
Asterisk does not reREGISTER in case of failure |
| 8:22AM |
1 |
iax2 bindaddress: how to reload so iax2 can bind to an alias IP |
| 8:15AM |
1 |
Friday Jan 9th at Noon ET: VoicePHP from TringMe |
| |
| Thursday January 8 2009 |
| Time | Replies | Subject |
|---|
| 11:07PM |
6 |
Not Dialing 9 |
| 10:14PM |
1 |
how many quad T1 cards |
| 9:15PM |
3 |
Playing MP3s... |
| 7:28PM |
0 |
AST-2009-001: Information leak in IAX2 authentication |
| 7:06PM |
0 |
Asterisk 1.2.31, 1.4.22.1, and 1.6.0.3 released |
| 6:24PM |
4 |
AEL question: testing channel variables |
| 5:44PM |
2 |
Could you compile mISDN 1.1.8 on Lenny ? |
| 5:41PM |
1 |
Executive Assistant Guidance |
| 5:35PM |
0 |
Attended transfer problems |
| 4:03PM |
0 |
console/dsp with digital sound |
| 3:56PM |
1 |
Goto Question |
| 3:17PM |
0 |
SIP message routed back to mysql |
| 2:21PM |
2 |
SIP "peer" with different username/password for incoming and outgoing |
| 2:06PM |
2 |
Problem incomming from openser |
| 1:28PM |
3 |
AEL and }; |
| 12:12PM |
1 |
is it possible to store vmsecrets outside of users.conf? |
| 11:02AM |
0 |
mISDN & Numeris Signaling (2 channels for 1 call) |
| 10:00AM |
1 |
Macro arguments seperator |
| 2:50AM |
0 |
dahdi_dummy only compile |
| |
| Wednesday January 7 2009 |
| Time | Replies | Subject |
|---|
| 9:53PM |
1 |
rejected because extension not found |
| 9:19PM |
1 |
SLA and Polycom |
| 9:09PM |
2 |
How to listen in on a SIP channel? |
| 5:19PM |
3 |
mISDN compile problem |
| 5:15PM |
1 |
Are mISDN mailinglists active ? |
| 4:38PM |
0 |
Chan_alsa stops working on 1.4.22 |
| 4:30PM |
5 |
recommendation for German sound files |
| 2:59PM |
2 |
1.6 |
| 2:59PM |
2 |
How to use AMD "Answering Machine Detect" ? |
| 1:52PM |
1 |
CISCO 7940 United_States/7960-tones.xml |
| 10:47AM |
1 |
app_rxfax and app_txfax with Ubuntu? |
| 8:25AM |
1 |
[Asus Eee PC 900] as replacement for legacy BRI phone |
| 6:36AM |
2 |
\iaxclient-2.0.2 compile problem |
| |
| Tuesday January 6 2009 |
| Time | Replies | Subject |
|---|
| 10:16PM |
0 |
If you use Realtime Extensions... READ THIS... |
| 9:41PM |
1 |
Asterisk 1.6 and LUA |
| 7:59PM |
5 |
Queue |
| 6:33PM |
2 |
any SIP client for BlackBerry? |
| 3:53PM |
5 |
Simple CDRs |
| 3:35PM |
1 |
.call file not updating MySQL CDR's |
| 3:22PM |
2 |
[Asus Eeebox] USB FXO adapter? |
| 3:19PM |
0 |
Fwd: A2billing Multiple Servers |
| 3:06PM |
1 |
Call transfer using agi |
| 1:18PM |
3 |
enabling silence suppression in asterisk |
| 12:32PM |
0 |
Asterisk Generating NetworkOOO (ISDN Cause Code 38) |
| 12:21PM |
1 |
"username mismatch, have <x>, digest has <y>" |
| 10:26AM |
1 |
Asterisk CLI got freezed!! |
| 10:21AM |
0 |
G.729 VAD issue |
| 8:28AM |
1 |
Problems getting 1.6 to run with user asterisk and group asterisk |
| 8:02AM |
1 |
R2D2 VOIP Kubuntu 8.4 Ekiga, Ekiga.net voice conference |
| 6:11AM |
4 |
bridge 2 calls |
| 4:11AM |
0 |
chan_sccp and CISCO CP-7914 Module |
| 12:24AM |
3 |
Incoming side of SIP trunk does not work unless I add "insecure=very" |
| |
| Monday January 5 2009 |
| Time | Replies | Subject |
|---|
| 10:12PM |
1 |
queue log parser |
| 8:05PM |
3 |
Agents, Queues and logon/logoff |
| 7:27PM |
1 |
CDR - What Changed? |
| 5:40PM |
1 |
cdr_addon_mysql 'Failed to insert into database' stops * call processing |
| 11:36AM |
0 |
G729 VAD issue |
| 11:04AM |
1 |
B410p, Ast1.4, France Télecom Numeris Double T0 problem |
| |
| Sunday January 4 2009 |
| Time | Replies | Subject |
|---|
| 8:42PM |
2 |
queue log in mysql |
| 8:12AM |
1 |
Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5 |
| 4:12AM |
2 |
Bring India together |
| |
| Saturday January 3 2009 |
| Time | Replies | Subject |
|---|
| 4:05PM |
1 |
OSLEC |
| 3:50PM |
1 |
snom 320's creating inappropriate conference calls |
| 11:18AM |
0 |
BerkeleyTIP TODAY Jan 3 Sat- Party Time :) Video Talks: Asterisk, GPU |
| 11:16AM |
0 |
BerkeleyTIP - Hello, Introduction, Monthly Global GNU(Linux) BSD Free SW HW meeting |
| 11:12AM |
0 |
Hello Asterisk list. BerkeleyTIP & you |
| |
| Friday January 2 2009 |
| Time | Replies | Subject |
|---|
| 4:19PM |
1 |
SIP URI: Allison Smith, Music-on-Hold Parody--outstanding. |
| 3:37PM |
2 |
Deprecated Realtime application, what's to be gained ??? |
| 1:21PM |
4 |
Setting Periodic-Announce filename in the dialplan |
| 9:26AM |
4 |
2008 Post Count |
| 4:41AM |
0 |
Audiocodes MP-11X configuration to work with Asterisk |
| |
| Thursday January 1 2009 |
| Time | Replies | Subject |
|---|
| 11:07AM |
0 |
DISA and the # key |
| 7:28AM |
2 |
New box, reload command takes 1 min. |
| 12:07AM |
5 |
Allison Smith, Music-on-Hold Parody--outstanding. |