| Tuesday March 31 2009 |
| Time | Replies | Subject |
|---|
| 8:22PM |
1 |
Queues in memory after startup |
| 7:20PM |
0 |
Dead Call But Still Active |
| 7:18PM |
1 |
dundi show peers - UNREACHABLE but I can ping it! |
| 5:36PM |
2 |
codec payload size |
| 4:38PM |
0 |
Strange voicemail problem when call forwarding off local PBX |
| 4:35PM |
2 |
What is the one thing that polycom can do... |
| 3:07PM |
2 |
dynamic codec preferences |
| 3:02PM |
2 |
DAHDI with OSLEC |
| 1:09PM |
6 |
[Zaptel] Why no driver for PCI voice modems? |
| 12:55PM |
0 |
Dialtones as Inband |
| 10:15AM |
0 |
conference function problems |
| 9:32AM |
2 |
Queue data from within dialplan? |
| 8:43AM |
1 |
iax2 not registering at startup, works on reload |
| 7:42AM |
0 |
USA Pharmacy Discount ID 443 |
| 6:37AM |
1 |
PRI problem |
| 1:17AM |
1 |
PAP2T-na Bricked? |
| |
| Monday March 30 2009 |
| Time | Replies | Subject |
|---|
| 9:26PM |
3 |
Call-limit=1 breaks attended transfer |
| 9:16PM |
2 |
Newbie trying to make calls outside via digium card and POTS line |
| 8:45PM |
1 |
IMAP voicemail storage. |
| 8:24PM |
1 |
Avoid compression with g.729/gsm/etc. |
| 8:06PM |
0 |
Where to find local FXS settings ? [SOLVED] |
| 7:41PM |
0 |
Where to find local FXS settings ? |
| 6:15PM |
1 |
Asterisk doesn't relay remote MOH during hold |
| 4:18PM |
2 |
Ideas for Asterisk load testing, testing trunks etc. |
| 4:07PM |
0 |
Set origin CallerID when forwarding calls to mobile phone |
| 2:14PM |
2 |
no ringtone - just silence/bridging of external calls |
| 11:55AM |
2 |
iphone, skype and asterisk ... |
| 11:13AM |
1 |
The Redirect hangups the call while playing a file |
| 10:29AM |
0 |
Limit on simultaneous manager commands |
| 9:53AM |
0 |
Pickup feature request |
| 6:34AM |
0 |
incoming number information |
| |
| Sunday March 29 2009 |
| Time | Replies | Subject |
|---|
| 2:39PM |
1 |
DUNDi broken in asterisk 1.4-svn? |
| 9:42AM |
2 |
h exten no getting run ... |
| 3:25AM |
1 |
callpickup not working |
| |
| Saturday March 28 2009 |
| Time | Replies | Subject |
|---|
| 10:03PM |
0 |
oh323 to h323 |
| 6:27AM |
2 |
hum noise |
| |
| Friday March 27 2009 |
| Time | Replies | Subject |
|---|
| 11:47PM |
2 |
TE122 |
| 9:37PM |
0 |
ISDN30 Channels Locking |
| 9:02PM |
0 |
UPDATED: Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.9, and Freeplay MoH Update Released |
| 8:51PM |
0 |
Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.8, and Freeplay MoH Update Released |
| 8:31PM |
1 |
Six steps to better SIP security with Asterisk |
| 5:33PM |
1 |
Strange warning message |
| 4:54PM |
0 |
SIP for Skype Solutions: Hosted v Non-hosted |
| 4:43PM |
1 |
Weird sip problem |
| 4:23PM |
2 |
London DDI test request |
| 3:19PM |
2 |
SIP Diversion header |
| 2:06PM |
3 |
AT&T PRI Install - What is outpulsed? |
| 11:07AM |
4 |
Help: RED alarm on Wildcard TE122 card |
| 10:30AM |
3 |
How to Integrate Neospeech with Asterisk |
| 4:02AM |
2 |
Need help on how to programmatically call an extension & test call state |
| |
| Thursday March 26 2009 |
| Time | Replies | Subject |
|---|
| 11:27PM |
3 |
Know who's logged in |
| 10:28PM |
6 |
Need to find small footprint asterisk platform |
| 9:25PM |
0 |
Asterisk 1.6.0.5 no MusicHold REFER |
| 8:41PM |
1 |
Is there a public blacklist of hackers' IPaddresses? |
| 8:35PM |
4 |
out of the box or do it your self? |
| 7:06PM |
3 |
Asterisk multi-cpu |
| 5:51PM |
2 |
PRI dropping #2 |
| 5:24PM |
0 |
Voicemail Problem |
| 4:05PM |
1 |
Sisky to connect Skype to Asterisk |
| 3:41PM |
6 |
Provisioning GXP 2000 |
| 2:19PM |
3 |
show pri usage |
| 1:29PM |
0 |
nat problem in reinvite.. |
| 12:10PM |
0 |
TDMoE in any way related to I-TDM |
| 8:45AM |
1 |
IAX problem through intermediate asterisk box |
| |
| Wednesday March 25 2009 |
| Time | Replies | Subject |
|---|
| 11:38PM |
1 |
help - How to send hangup command to call in progress. |
| 11:31PM |
0 |
How to send hangup command to call in progress. |
| 8:50PM |
1 |
Skype TO SIP (Was SIP to Skype) |
| 6:23PM |
3 |
OT: Accountless, free, skinnable, browser based SIP client wanted |
| 5:22PM |
1 |
More on SIP for Skype |
| 4:35PM |
2 |
New CentOS 5 repo: dahdi, asterisk, freepbx RPMs |
| 2:40PM |
3 |
SIP Asterisk Hacked (1.6.0.6) |
| 1:19PM |
1 |
SIPPEER equivalent for users.conf ? |
| 1:04PM |
8 |
ITSP's no longer supporting IAX? |
| 12:55PM |
1 |
ASTCC and a2billing |
| 12:39PM |
4 |
Recording the calls |
| 10:58AM |
1 |
Defining a call |
| 9:38AM |
3 |
Create separate Voice Recording System.. |
| 5:50AM |
0 |
openIMSCore + asterisk |
| 1:29AM |
0 |
place T1 calls and ignore/override call progress |
| 12:39AM |
0 |
${UNIQUEID} variable and queue log issues on 1.4.22 |
| 12:25AM |
1 |
DISA |
| 12:10AM |
1 |
predictive dialer |
| 12:06AM |
0 |
A Cisco 7960 question |
| |
| Tuesday March 24 2009 |
| Time | Replies | Subject |
|---|
| 10:08PM |
1 |
Inter-Asterisk Using SIP |
| 9:21PM |
4 |
PRI dropping |
| 7:11PM |
2 |
Ebay's SIP for Skype |
| 6:21PM |
2 |
HW-Recommendation: cell/mobile phone, capable of WLAN and SIP ?? |
| 5:56PM |
0 |
originate and local channel problem |
| 5:51PM |
1 |
sip.conf outboundproxy |
| 5:35PM |
0 |
Unrecognized prilocaldialplan error when dialing a SIP call to a PRI trunk |
| 4:00PM |
0 |
T1 issue to analog trunk for paging (intercom) |
| 3:10PM |
5 |
SIP trunk with > 250 lines |
| 2:21PM |
0 |
MWI Asterisk+Openser |
| 1:21PM |
1 |
Error in ReceiveFax with T.38 -- Asterisk 1.6.0.7-rc2 |
| 1:02PM |
0 |
lagrq |
| 10:52AM |
1 |
Relay Register |
| 8:54AM |
0 |
Billing software for 1.6? |
| 8:25AM |
6 |
gpx 2000 Busy Lamp Field |
| 8:16AM |
0 |
Asterisk Realtime Config and SIP/401 Unauthorize: why? |
| 7:14AM |
0 |
Issue with RDNIS |
| 6:16AM |
1 |
Asterisk Originate Command |
| 1:11AM |
6 |
Is there a public blacklist of hackers' IP addresses? |
| 12:14AM |
1 |
strategy ringall |
| |
| Monday March 23 2009 |
| Time | Replies | Subject |
|---|
| 10:44PM |
3 |
usb-phones |
| 8:17PM |
1 |
BACKGROUNDSTATUS not available? |
| 7:42PM |
2 |
Skype for SIP |
| 7:00PM |
2 |
conference and wifi phones |
| 5:35PM |
2 |
Polycoms and BLF |
| 1:33PM |
3 |
Recommended USB Headsets ? |
| 11:01AM |
0 |
sip/iax dialplan extension.. |
| 10:57AM |
0 |
Issue with no change of SIP call ID |
| 10:01AM |
1 |
Dial in / dial out |
| 9:01AM |
1 |
Simple UK Extensions example |
| 7:00AM |
1 |
distictive Ringing in SIP |
| 2:40AM |
1 |
field lastms in 1.4.24 |
| |
| Sunday March 22 2009 |
| Time | Replies | Subject |
|---|
| 8:58PM |
1 |
make script 1.6.0.6 breaks up, need help! |
| 7:22PM |
2 |
Global videoconferencing solution. |
| 7:11PM |
3 |
I need a country, state, city database |
| 5:35PM |
1 |
CID when using WaitExten? |
| 2:12PM |
1 |
Looking for Prepaid Solution |
| 7:39AM |
0 |
Manipulating REGISTER messages |
| 4:09AM |
1 |
Asterisk on iMac G3 Debian5 (powerpc) |
| |
| Saturday March 21 2009 |
| Time | Replies | Subject |
|---|
| 11:37PM |
2 |
music-on-hold kicks in and disconnects/interrupt the call |
| 7:49PM |
2 |
H323plus homepage down? |
| 2:18PM |
0 |
OT - CID with Asterisk and Betamax |
| 11:33AM |
2 |
1.6.0-rc3 Build failure: asterisk.h: No such file or directory |
| 8:39AM |
6 |
OpenBTS chat with David A. Burgess |
| 2:04AM |
0 |
Asterisk with encryption |
| 1:36AM |
2 |
1.6.2 beta 1 crash |
| |
| Friday March 20 2009 |
| Time | Replies | Subject |
|---|
| 11:07PM |
1 |
Music on Hold doesn't play back for external callers |
| 7:22PM |
0 |
Asterisk 1.6.0.7-rc2, 1.6.1.0-rc3, 1.6.2.0-beta1 & Asterisk-Addons 1.6.0.2-rc1, 1.6.1.0-rc3 Now Available |
| 6:05PM |
2 |
Looking for clues to this error message |
| 5:32PM |
3 |
Queues Announce help request. |
| 5:19PM |
0 |
Asterisk Management Application for windows |
| 3:48PM |
0 |
anyone connection to eoncc |
| 2:49PM |
3 |
ATA recommendation?? |
| 2:19PM |
3 |
OpenSIPS on CentOS |
| 2:09PM |
1 |
Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk |
| 9:36AM |
1 |
T38 FAX |
| 8:52AM |
0 |
Asterisk Realtime Configuration and 404 Extension not found |
| 8:27AM |
1 |
Max concurrent calls |
| 7:05AM |
1 |
chan_ss7 with ringing, but no voice stream. |
| 2:31AM |
1 |
Special Information Tones |
| |
| Thursday March 19 2009 |
| Time | Replies | Subject |
|---|
| 11:34PM |
4 |
"The number you have called has been disconnected or is no longer in service" |
| 9:18PM |
1 |
Overriding Queue Wrapup Time |
| 7:56PM |
1 |
Question regarding call queue and penalty's |
| 7:53PM |
0 |
Can I tell if a call picked up on PSTN extension... for example? |
| 7:28PM |
0 |
Flowroute |
| 7:14PM |
3 |
Hardware suggestions |
| 6:09PM |
3 |
Digium and Sangoma Cards PCI express compatibility |
| 5:23PM |
6 |
Asterisk with SRTP and SIP with TLS |
| 4:23PM |
0 |
Is adding "sip show username" easy ? |
| 3:19PM |
3 |
busy lamp filed |
| 3:08PM |
3 |
(no subject) |
| 2:28PM |
2 |
Script to softly restart Asterisk each midnight to clean locked channels |
| 2:27PM |
0 |
DTMF tones mid conversation |
| 2:05PM |
1 |
Polycom MWI. |
| 11:51AM |
1 |
Asterisk and PBX internal numbers |
| 11:19AM |
1 |
VM_DATE in french? |
| 10:29AM |
1 |
IAX trunktimestamps and AST_CONTROL_SRCUPDATE |
| 10:00AM |
0 |
Extensions not found and 401 Unauthorized in realtime configuration (Long post) |
| 9:46AM |
1 |
PRI QSIG Asterisk - Legacy PBX |
| 9:37AM |
1 |
incoming call problem from pri |
| 8:03AM |
0 |
T1 signaling configuration |
| 5:00AM |
1 |
Asterisk crashed!!! |
| 3:55AM |
0 |
AstLinux 0.6.4 available for upgrade |
| |
| Wednesday March 18 2009 |
| Time | Replies | Subject |
|---|
| 11:14PM |
0 |
queued? |
| 10:03PM |
0 |
Recent changes in chan_mobile need testing! |
| 10:02PM |
2 |
Voicemail config help - require password |
| 10:01PM |
1 |
Unable to receive faxes |
| 9:01PM |
0 |
[Fwd: Re: DAHDI or Zaptel doesn't compile against 1.4.24] |
| 5:23PM |
3 |
Global h exten |
| 3:57PM |
1 |
Video phone crashing meetme on asterisk 1.4. |
| 3:46PM |
1 |
Controlling BLF Leds ... |
| 3:34PM |
0 |
Asterisk talking to mysql |
| 2:46PM |
0 |
Asterisk talking to mysql database through odbc |
| 11:56AM |
0 |
Asterisk is not designed for University large scale |
| 11:45AM |
0 |
Callerid charset problems |
| 11:23AM |
1 |
Performance of realtime for millions of SIP user |
| 8:58AM |
3 |
Manager API Originate CDR Problem, all is NO ANSWER |
| 2:55AM |
1 |
Asterisk and G.726 Codec |
| |
| Tuesday March 17 2009 |
| Time | Replies | Subject |
|---|
| 6:16PM |
2 |
DAHDI or Zaptel doesn't compile against 1.4.24 |
| 2:13PM |
0 |
Kewlstart - Busy signal before battery drop. |
| 1:23PM |
2 |
PBX to gate interface |
| 1:13PM |
3 |
SPA3102 - How to save config in a file |
| 12:43PM |
0 |
DTMF troubles |
| 11:44AM |
1 |
Direct Dial-Out and CDR destination numbers |
| 10:33AM |
0 |
ATA react to phone but unresponsive to fax modem [SOLVED] |
| 9:17AM |
0 |
asterisk now and switchvox |
| 8:49AM |
0 |
Weird issue with outbound calls and MOH |
| 8:21AM |
1 |
mobile centrex solution |
| 7:38AM |
1 |
Looking for a patch cable for my SPA941 Phones |
| 6:00AM |
1 |
Test asterisk from behind my firewall |
| 2:49AM |
4 |
Plastic Water Bottles |
| 12:57AM |
2 |
system sizing |
| |
| Monday March 16 2009 |
| Time | Replies | Subject |
|---|
| 11:28PM |
1 |
Bristuff bug or feature ? (Was: Are .call files working with extensions.ael ? bristuff problem) |
| 11:27PM |
2 |
Problem with Verizon Wireless |
| 11:21PM |
2 |
Multi-tenant with receptionist features for managed service |
| 10:42PM |
0 |
Asterisk 1.4.24 Now Available! |
| 10:42PM |
0 |
Uptime for documentation only |
| 10:42PM |
3 |
T1 problem (call using a .call file) |
| 10:24PM |
8 |
Good phone near $125 |
| 10:11PM |
1 |
T.38 - Which endpoint shall reINVITE ? caller or callee ? |
| 9:34PM |
3 |
Asterisk is not designed for University with large user base? |
| 6:42PM |
1 |
ATA react to phone but unresponsive to fax modem |
| 6:11PM |
0 |
Contact id protocol problem |
| 6:11PM |
1 |
asterisk and ericsson e1 connection how to?? |
| 5:54PM |
3 |
Help Inbound number |
| 5:08PM |
1 |
Could Asterisk be rewriting an incoming invite? |
| 4:56PM |
0 |
SIP audio delay after call transfer? |
| 4:56PM |
1 |
Aastra 9133i programmable buttons (* 4.1.23) |
| 4:28PM |
2 |
Busy on SIP |
| 3:34PM |
3 |
A400P + Intel D201GLY2(A) motherboard? |
| 2:50PM |
0 |
Problems on default Attended Transfer |
| 1:39PM |
0 |
Ignore switch to REVERSED Polarity on channel 1, state 4 |
| 12:49PM |
1 |
Transfers on an inter-PBX PRI link |
| 11:30AM |
1 |
ANI with Pickup application |
| 11:09AM |
0 |
SIMPLE |
| 10:51AM |
3 |
url in dial command: how does it work? |
| 10:05AM |
2 |
t38 iax trunk |
| 9:09AM |
0 |
trying asterfax |
| 1:57AM |
3 |
Asterisk 1.6 ReceiveFAX problem |
| |
| Sunday March 15 2009 |
| Time | Replies | Subject |
|---|
| 10:28PM |
5 |
428 Loop Detected |
| 10:03PM |
1 |
No hardware timing source found in /proc/dahdi |
| 4:38PM |
0 |
Too many notify events causing Asterisk crash? |
| 12:19PM |
0 |
Dahdi Error |
| 11:30AM |
1 |
Using PRI_CAUSE to change SIP INVITE rejection response code |
| 10:30AM |
1 |
X-Asterisk-HangupCause - how to disable this? |
| |
| Saturday March 14 2009 |
| Time | Replies | Subject |
|---|
| 6:46PM |
0 |
E&M signalling |
| 4:46PM |
3 |
getting free Did number for asterisk |
| 4:12PM |
1 |
Polycom BLF with Idle State meetme conference |
| 3:26PM |
0 |
Problem with phantom calls |
| 12:13PM |
2 |
BRI cards; JUNGHANNS AND B410P |
| 10:12AM |
1 |
"automatic call bridging when destination is available" feature |
| 1:04AM |
3 |
TRANSFER EVENT ON QUEUE_LOG |
| |
| Friday March 13 2009 |
| Time | Replies | Subject |
|---|
| 10:41PM |
1 |
Realtime dialplan application versus REALTIME dialplan function |
| 10:37PM |
2 |
SendFAX/T.38 question |
| 6:14PM |
0 |
Recording calls and SLA |
| 3:38PM |
2 |
Ast/Hyla/IAX Scalability? |
| 3:00PM |
0 |
VoIP Users Conference today at 12 Noon EDT |
| 2:56PM |
2 |
No reply to our critical packet |
| 2:49PM |
1 |
Asterisk to Ericsson MD110 on E1 with ISDN-USR (not QSIG)? |
| 9:51AM |
1 |
Silence suppression problem with DECT phones and g729 codec |
| 9:45AM |
1 |
AGX Asterisk Addon - Can't find app_fax.c withspandsp-0.0.4 |
| 7:06AM |
3 |
Initial silence during call |
| 1:52AM |
0 |
MOH Realtime |
| |
| Thursday March 12 2009 |
| Time | Replies | Subject |
|---|
| 9:42PM |
4 |
Serving 120 concurrent calls |
| 9:41PM |
1 |
Queue Realtime agents LOGIN for ami |
| 9:23PM |
2 |
Timeout for Queue |
| 8:34PM |
1 |
SetVar (CDR var) from cli |
| 7:02PM |
4 |
log to cdr each dialpan action, not only one record for each call |
| 6:01PM |
1 |
phone emulator for doing interop testing |
| 4:56PM |
1 |
Trying to get sample applicationmap to work (*1.4) |
| 4:13PM |
0 |
chanspy problems (asterisk 1.6.0.6) - When spying starts, the spied parties can't hear each other |
| 3:55PM |
1 |
an easy way to deal with/without leading "1" ? |
| 3:14PM |
1 |
Outgoing call drops |
| 2:11PM |
1 |
Is it possible to get full callin number fromE1? |
| 1:34PM |
2 |
BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai |
| 12:40PM |
5 |
Is it possible to get full callin number from E1? |
| 10:26AM |
4 |
Printing faxes |
| 10:21AM |
8 |
UK ISDN-30 and ANI |
| 10:05AM |
0 |
Manager API and astmanproxy |
| 9:20AM |
3 |
ATCom Phones - AT 510/AT530 |
| 7:45AM |
0 |
(no subject) |
| 5:21AM |
0 |
compile error: implicit declaration of function drv_dbg |
| 3:50AM |
0 |
recording (mixmonitor) stopped of transfer/call parking after queue |
| |
| Wednesday March 11 2009 |
| Time | Replies | Subject |
|---|
| 11:29PM |
4 |
Are .call files working with extensions.ael ? |
| 9:06PM |
3 |
Grandstream speakerphone? |
| 7:02PM |
3 |
Digium B410P: misdn v1 or misdn v2 or dahdi + asterisk 1.6 ? |
| 6:28PM |
1 |
update on Odd occurrence |
| 4:38PM |
2 |
Problem with incoming and outgoing calls via TDM |
| 2:27PM |
2 |
Multiple Agent Login |
| 10:21AM |
2 |
how to configure for incoming message-summary SUBSCRIBE |
| 10:03AM |
0 |
SIP keep-alive with CRLF? |
| 8:56AM |
2 |
VLC |
| 8:09AM |
0 |
How to read installed spandsp version ? |
| 6:16AM |
0 |
AGX Asterisk Addon - Can't find app_fax.c with spandsp-0.0.4 |
| |
| Tuesday March 10 2009 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
MacroExclusive crashed asterisk |
| 8:31PM |
4 |
chan_zap.so missing |
| 6:42PM |
0 |
Does the Asterisk/Digium support REGISTER requests with "Authorization" header field sent by a UAC before receiving 401 Unauthorized |
| 6:32PM |
1 |
Phone Directories/Asterisk/SIP/directory.html |
| 6:21PM |
2 |
1.6.x differences |
| 6:02PM |
1 |
Odd occurrence |
| 5:51PM |
0 |
Asterisk 1.6.1.0-rc2 Now Available |
| 5:50PM |
1 |
Asterisk 1.6.0.7-rc1 Now Available |
| 5:50PM |
0 |
Asterisk 1.4.24-rc1 Now Available |
| 5:38PM |
0 |
AST-2009-002: Remote Crash Vulnerability in SIP channel driver |
| 3:24PM |
2 |
1.4.23 + Realtime Queues/Agents NOT via SIP |
| 2:27PM |
5 |
Sending faxes with T.38 problem. Asterisk - 1.6.0.6 |
| 2:26PM |
1 |
Compiling OSLEC in Dahdi w/ Intel Optimizations |
| 2:20PM |
3 |
configuring channels for dahdi |
| 2:12PM |
0 |
Voxli |
| 12:51PM |
3 |
Update chan_dahdi.conf doc in voip-info.org |
| 11:05AM |
1 |
Asterisk 1.6, B410P and TE/PtmP mode. Could you get it running ? |
| 11:04AM |
1 |
in which asterisk version is zaptel removed? |
| 10:52AM |
3 |
How to do Load-Balancing for Asterisk with OpenSIPS |
| 10:40AM |
1 |
Asterisk and WebIntegration |
| 10:13AM |
1 |
Calling id problem on outgoing call |
| 7:24AM |
0 |
Hold/Resume issue with polycom |
| |
| Monday March 9 2009 |
| Time | Replies | Subject |
|---|
| 11:57PM |
1 |
how to write svn for dahdi-linux and dahdi-tools when using svn 1.4 |
| 7:46PM |
2 |
Job in Atlanta. |
| 7:17PM |
1 |
1.6.0.5 - g729 'locked' by Asterisk |
| 7:01PM |
6 |
MoH - always starting from the beginning |
| 6:55PM |
0 |
Announcements |
| 6:41PM |
0 |
Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4 [SOLVED] |
| 6:33PM |
0 |
How to install spandsp from source in lenny ? [SOLVED] |
| 6:03PM |
0 |
asterisk-users Digest, Vol 56, Issue 23 |
| 5:36PM |
4 |
DAHDI and B410P (BRI) |
| 4:13PM |
1 |
How to install spandsp from source in lenny ? |
| 1:37PM |
0 |
Crash when reloading AEL |
| 1:30PM |
3 |
problem with an agi in PHP |
| 12:18PM |
2 |
Portech MV3770 & Caller-ID |
| 12:03PM |
2 |
I can't receive fax |
| 11:48AM |
0 |
macro on ring group |
| 10:23AM |
0 |
SIP warnings (401) |
| 9:49AM |
0 |
SIP call hangs up after 20 seconds |
| |
| Sunday March 8 2009 |
| Time | Replies | Subject |
|---|
| 9:49PM |
2 |
IAX peer cannot register in Asterisk 1.2.31 |
| 4:11PM |
2 |
Fwd: add a new queue strategy: SBR |
| 1:11PM |
1 |
Simple Meetme Question |
| 12:31PM |
2 |
Server Setup Advice |
| |
| Saturday March 7 2009 |
| Time | Replies | Subject |
|---|
| 11:26PM |
4 |
Compile problems |
| 4:07PM |
0 |
Busy Here |
| 3:27PM |
0 |
QUEUE_MEMBER_COUNT: Bug or functionality ? |
| 2:29PM |
0 |
Join today Asterisk VOIP meeting online 10A-6P PST = 6P-2A GMT Sat Mar 7 BerkeleyTIP |
| 5:24AM |
1 |
Cdr problem |
| |
| Friday March 6 2009 |
| Time | Replies | Subject |
|---|
| 8:40PM |
1 |
question about ringinuse |
| 7:57PM |
1 |
Wideband (G722) MeetMe |
| 7:45PM |
1 |
Parked Calls in 1.4.23.1 |
| 7:41PM |
0 |
Early Media before 200 ok |
| 7:29PM |
3 |
IAX based war dialer |
| 6:26PM |
1 |
GoSub & Queue |
| 5:10PM |
1 |
Asterisk and sip router integration |
| 3:58PM |
5 |
work around the 64 pickupgroups limit |
| 2:41PM |
2 |
Aastra 480i repair? |
| 2:10PM |
2 |
SIP *8 Pickup Problem |
| 2:02PM |
1 |
call pickup and ring groups |
| 12:36PM |
1 |
Making use of SIP "making progress" messages |
| 11:41AM |
0 |
Queue moh problem with 1.4.23.1 |
| 8:59AM |
2 |
colorized logfiles in asterisk 1.6.0.6 |
| 8:58AM |
1 |
Dial command with "r" parameter - no ring tone |
| 7:45AM |
2 |
question about MeetMe performance. |
| 3:22AM |
5 |
How to verify availability of the DID connection? |
| 2:52AM |
1 |
Asterisk dial plan conditional on not busy |
| 1:04AM |
1 |
Fax detection on SIP channel |
| |
| Thursday March 5 2009 |
| Time | Replies | Subject |
|---|
| 11:49PM |
0 |
It took some time... |
| 7:15PM |
0 |
T.38 Problem |
| 5:18PM |
0 |
Invite somebody to a conf call |
| 4:00PM |
1 |
Snom Aler-info Ringtone |
| 3:10PM |
0 |
oslec using sample.c for long(er) dumps |
| 3:04PM |
0 |
asterisk and simple chat protocol |
| 1:10PM |
1 |
use more then one sip-provider to dial out |
| 12:39PM |
0 |
Asterisk 1.6.x and auto-provisioning - Polycom |
| 12:34PM |
0 |
Recognizing the "making progress" notification |
| 12:03PM |
0 |
ael conf and realtime |
| 9:42AM |
1 |
Asterisk Differences |
| 7:42AM |
1 |
Asterisk 1.6.1-rc1 with OpenAIS and different subnets |
| 3:05AM |
0 |
Stun with hosted asterisk solution??? |
| 2:32AM |
2 |
Cisco IP Communicator with Asterisk/Trixbox |
| |
| Wednesday March 4 2009 |
| Time | Replies | Subject |
|---|
| 11:09PM |
0 |
Asterisk 1.6.0.6 sip doesn't work? |
| 5:42PM |
2 |
Outlook integration? |
| 5:34PM |
2 |
SIP attacks |
| 4:06PM |
2 |
Bounty- CDR Bug Fix |
| 3:53PM |
0 |
Asterisk @ Global FreeSW Meeting March 7 Sat BerkeleyTIP -Global - For Forwarding |
| 3:38PM |
1 |
What's the use of sip.conf's notifyringing ? |
| 3:02PM |
2 |
Required:Asterisk Beep tone while call connects |
| 12:22PM |
3 |
Wideband g711-HD vs. g711.1? |
| 12:17PM |
1 |
htable question] |
| 10:44AM |
0 |
Access sip.conf's mailbox from dialplan ? [SOLVED] |
| 9:05AM |
0 |
Master.csv - disposition value (based on?) |
| 8:38AM |
1 |
Question on phone line "pass through" |
| 6:37AM |
2 |
Druid 2.0 released from the Druid Open Source Unified Communications Project |
| 6:25AM |
1 |
faxing via linksys SPA3102 half page goes through |
| 4:51AM |
3 |
Silk for Free |
| 12:20AM |
5 |
AEL2: If-then-else not permitted in Switch-Case |
| 12:12AM |
4 |
$20 Bounty |
| |
| Tuesday March 3 2009 |
| Time | Replies | Subject |
|---|
| 11:30PM |
1 |
Predefined viables |
| 8:05PM |
0 |
Unable to create channel |
| 7:09PM |
0 |
monitoring a channel and redirect to conf |
| 6:39PM |
2 |
Asterisk analog DID with Adit 600 |
| 3:28PM |
2 |
macro-stdexten question |
| 3:03PM |
1 |
Remote Connection to Asterisk |
| 2:11PM |
0 |
patlooptest and TE121P |
| 1:08PM |
0 |
NOTIFY/SUBSCRIBE and MWI in 1.4 |
| 12:24PM |
2 |
CDR |
| 12:19PM |
2 |
Configuring asterisk to revert call back to forwarder if exten is busy |
| 11:59AM |
1 |
Master.csv missing dstchannels values |
| 11:39AM |
0 |
cdr database |
| 10:40AM |
1 |
after install the zaptel but the rtp failed |
| 9:25AM |
1 |
tons of open SIP channel between two snom 360 |
| 9:24AM |
2 |
Access sip.conf's mailbox from dialplan ? |
| 3:38AM |
0 |
Blind transfer from asterisk dialplan (and problems re-parking a call) |
| |
| Monday March 2 2009 |
| Time | Replies | Subject |
|---|
| 11:34PM |
0 |
Retrieve DTMF during Dial |
| 11:13PM |
3 |
Dialing with cli |
| 11:09PM |
1 |
Asterisk Dial plan issue |
| 10:03PM |
0 |
Queue log on MySQL realtime |
| 9:29PM |
1 |
Will A2B worki with asterisk as b2bua? |
| 8:31PM |
2 |
Asterisk realtime |
| 8:21PM |
3 |
How to set PRI line timeout value |
| 7:46PM |
0 |
H323 Call Variables |
| 6:30PM |
0 |
Disabling 180 Messages |
| 2:58PM |
5 |
How to generate core dump? |
| 2:48PM |
2 |
Compiling to use IMAP: how? |
| 12:40PM |
1 |
SIP dialog matching problem? (1.4.23.1) |
| 10:39AM |
1 |
pci cards VS patton |
| 10:12AM |
1 |
early dial (or overlap dial) and Asterisk 1.2 vs. 1.4 |
| 9:27AM |
0 |
how to install the app_meetme2.so and use the web interface |
| 6:27AM |
1 |
Weird segfault |
| |
| Sunday March 1 2009 |
| Time | Replies | Subject |
|---|
| 6:04PM |
1 |
Help T.38 |
| 4:02PM |
0 |
Pharmacy Message 25749 |
| 2:37PM |
1 |
php agi and get_data errors. |
| 2:56AM |
1 |
Called number as variable - how to? |