| Monday August 31 2009 |
| Time | Replies | Subject |
|---|
| 11:21PM |
5 |
queue issue |
| 8:05PM |
1 |
Upgrading Asterisk in Trixbox installation |
| 7:01PM |
2 |
List Access |
| 5:11PM |
1 |
Asterisk MWI issue |
| 4:37PM |
2 |
Versions of Asterisk 1.6 |
| 3:33PM |
1 |
Strange problem |
| 1:26PM |
4 |
How to stop IVR once system receives DTMF? |
| 12:52PM |
0 |
Clarifying RX and TX gains |
| 12:41PM |
0 |
asterisk-users Digest, Vol 61, Issue 85 |
| 11:01AM |
0 |
Asterisk 1.4 and GUI Configuration Help |
| 9:53AM |
2 |
Asterisk Regular expression to validate any phonenumber |
| 6:36AM |
2 |
SIPP how can we give delays between 2 calls |
| 5:24AM |
4 |
Inquiry:How to hide Caller Id |
| 5:17AM |
5 |
CDR to Postgres Centos |
| 2:45AM |
3 |
How to deal with PayPal frauds? |
| 2:33AM |
2 |
Asterisk Web Meetme module not loading |
| 12:54AM |
1 |
Question of resiliance |
| |
| Sunday August 30 2009 |
| Time | Replies | Subject |
|---|
| 6:54PM |
1 |
I find this incomprehensible ?! |
| 1:30PM |
1 |
Need help - CDR MySQL |
| 12:11PM |
2 |
MySQL syntax error : I really don't see where... |
| 9:57AM |
2 |
Asterisk/app_rpt and bandwidth |
| 5:52AM |
1 |
Help me testing this webphone at www.VisionVoIP.com |
| 12:50AM |
1 |
unable to execute command from |
| |
| Saturday August 29 2009 |
| Time | Replies | Subject |
|---|
| 7:19PM |
1 |
GoToIfTime : how to define sep 25th till oct 10th ? |
| 5:18PM |
0 |
asterisk-users Digest, Vol 61, Issue 84 |
| 4:01PM |
2 |
Asterisk 1.6.0.14 and 1.6.1.5 Now Available |
| 12:29PM |
2 |
cannot run agi scripts |
| 6:59AM |
2 |
Flite module for asterisk 1.6.x |
| 6:16AM |
0 |
Training failed with hylafax |
| |
| Friday August 28 2009 |
| Time | Replies | Subject |
|---|
| 11:46PM |
2 |
Error loading module 'res_config_odbc.so' |
| 9:24PM |
0 |
realtime voicemail and imap user |
| 8:41PM |
4 |
Report |
| 6:28PM |
0 |
chan_mobile on macosx ppc |
| 5:35PM |
1 |
Help needed with getting a maxed-out Asterisk to gracefully deny calls. |
| 4:14PM |
0 |
2 Asterisk boxes via IAX : always calling as IAX guest |
| 3:21PM |
1 |
QuickPhones QA-342 and SIP/RTP Flash Event for Transferring |
| 2:52PM |
1 |
Crystal Recording Interface |
| 1:50PM |
1 |
PRI worked fine for months, nowit stopps working after 2-3 hours |
| 1:02PM |
2 |
Help with call scenario |
| 11:09AM |
1 |
Zap / dahdi errors |
| 9:25AM |
1 |
application order when you make a call |
| |
| Thursday August 27 2009 |
| Time | Replies | Subject |
|---|
| 10:57PM |
0 |
Fotos 18/08 . |
| 8:47PM |
0 |
Universal Services Fund taxes now apply to VoIP end-users. |
| 7:39PM |
0 |
transfer two gsm mobile calls |
| 6:44PM |
1 |
how does "wrapuptime" work in queue.conf |
| 6:30PM |
1 |
Bad Gateway |
| 6:23PM |
2 |
Selective canreinvite in multi-tenant environment |
| 5:52PM |
3 |
Sticky Park |
| 5:32PM |
0 |
password length of sip peer |
| 3:14PM |
2 |
POTS supervision with DAHDI in 1.4 releases |
| 1:08PM |
0 |
How to set call record file name |
| 12:39PM |
1 |
Problems using chan_sebi and Huawei E169G |
| 10:57AM |
0 |
create applicationmap and use it in dialplan |
| 9:40AM |
1 |
Documentation on RSA key authentication ?? (No way to send secret to peer) |
| 9:24AM |
6 |
Measuring voice quality with Asterisk |
| 7:01AM |
2 |
asterisk 1.6.0.13 with realtime DB , issue with MWI |
| 5:34AM |
3 |
Digium Echo cancellation. |
| |
| Wednesday August 26 2009 |
| Time | Replies | Subject |
|---|
| 9:29PM |
1 |
2 Asterisk boxes : 1 can see the other, not vica versa |
| 9:21PM |
0 |
Company in Los Angeles looking for Asterisk & Network Administration/Maintenance Engineer |
| 7:21PM |
0 |
408 error |
| 6:03PM |
4 |
Fw: app_swift issue |
| 3:48PM |
1 |
TE4XXP: Version Synchronization Error! |
| 3:07PM |
0 |
Swift application and DTMF |
| 2:07PM |
4 |
Multiple user registration ... |
| 2:03PM |
2 |
application missed in asterisk 1.6.1 - SetCallerID() |
| 12:36PM |
0 |
Header from REFFER |
| 12:05PM |
0 |
looking for user:Zoa for T38... |
| 11:42AM |
1 |
Bria / eyebeam: no RTCP while on hold |
| 8:54AM |
0 |
DAHDI PRI ERROR |
| 6:53AM |
6 |
PRI worked fine for months, now it stopps working after 2-3 hours |
| 6:48AM |
1 |
Open Source Visual Call-Flow and IVR Dev Tool v1.0 Released! |
| 6:33AM |
1 |
app_swift issue |
| 5:20AM |
1 |
ACD, call barge, recording |
| 3:07AM |
0 |
Timeout func ignored if inside a macro and when Dial cmd has limit (L). Bug ? |
| 2:39AM |
3 |
User-invoked call restrictions |
| 1:07AM |
1 |
netfilter conntrack mangling canreinvite? |
| |
| Tuesday August 25 2009 |
| Time | Replies | Subject |
|---|
| 8:43PM |
2 |
Echo |
| 8:14PM |
1 |
How to detect if the call is being answered by Voice Mail? |
| 7:47PM |
1 |
Realtime with "rtcachefriends=no" problems... |
| 3:34PM |
2 |
Linuz from asterisk |
| 2:28PM |
1 |
followme app |
| 1:47PM |
0 |
DTMF duplicated when Waitexten |
| 12:59PM |
6 |
Breaking news, but what happened? 11.000 channels on one server |
| 10:54AM |
2 |
Authenticating SIP peer on IP address only |
| 10:39AM |
0 |
Patton ISDN 2e boxes as media converters... |
| 9:28AM |
0 |
No translator path exists for channel type dahdi |
| 8:54AM |
1 |
Set up SER as a SIP extension on asterisk server |
| 6:13AM |
0 |
Stop Originate |
| |
| Monday August 24 2009 |
| Time | Replies | Subject |
|---|
| 7:52PM |
1 |
E1 w/ TE420B EC |
| 7:51PM |
1 |
Bottlenecks with my asterisk setup. |
| 4:12PM |
2 |
install the digium card TC400P howto? |
| 4:09PM |
0 |
Autodial not waiting for voicemail |
| 2:11PM |
1 |
asterisk 1.6.1.1 + Asterisk GUI v2.0 |
| 1:55PM |
1 |
Problems sending voicemail emails |
| 1:21PM |
1 |
Show queue-name near the callerId |
| 12:44PM |
0 |
SIP doesn't recognize hangup |
| 11:49AM |
1 |
Follow me IVR sounds |
| 11:48AM |
1 |
Need to now my "Asterisk User ID" |
| 10:17AM |
1 |
problem on compiling asterisk-addons-1.6.2.0-rc1 |
| 10:13AM |
0 |
exchanging CDR data between Asterisk servers |
| 8:51AM |
1 |
Request Pending retransmitions |
| 6:27AM |
12 |
Dailing any number PSTN or MObile number |
| 5:04AM |
1 |
disconnection silent channels |
| |
| Sunday August 23 2009 |
| Time | Replies | Subject |
|---|
| 11:34PM |
2 |
1.4.26.1, 1.6.0.13, 1.6.1.4 |
| |
| Saturday August 22 2009 |
| Time | Replies | Subject |
|---|
| 2:29PM |
1 |
Patton smartnode 463x (BRI) 25ms tail echo cancellation |
| 1:23AM |
0 |
Interview with John Todd |
| |
| Friday August 21 2009 |
| Time | Replies | Subject |
|---|
| 9:40PM |
0 |
Valet Park with Hint - Button Support |
| 8:22PM |
5 |
how to install asterisk |
| 8:00PM |
1 |
Queue Question |
| 2:16PM |
1 |
Incoming caller presentation doesn't work - out of ideas |
| 12:39PM |
2 |
stutter playback |
| 12:27PM |
2 |
Asterisk 1.6 t 38 passthrough |
| 9:27AM |
2 |
codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory |
| 2:17AM |
3 |
Core dump gets created while accessing voicemail |
| 12:18AM |
1 |
problem with asterisk hylafax and sangoma A200D |
| |
| Thursday August 20 2009 |
| Time | Replies | Subject |
|---|
| 6:51PM |
2 |
2 single span TDM cards in asterisk |
| 6:45PM |
1 |
Asterisk 1.6.2.0-beta4 - Monitor / MixMonitor Recording |
| 6:06PM |
0 |
thanks! |
| 6:05PM |
0 |
Friday Aug 21st @ 12 noon EDT: VoIP Users Conference, More Wideband Madness |
| 5:57PM |
12 |
IPKall and FWD |
| 5:14PM |
0 |
Nokia E90 video via Asterisk |
| 4:22PM |
1 |
multiple call dialing and playback an message |
| 3:47PM |
6 |
Cannot play soundfile, doesnt find it or wrong format? Weird, worked yesterday! :-) |
| 3:12PM |
1 |
Sip Tunneling |
| 3:06PM |
1 |
Pause/Unpause agent based on devstate |
| 2:54PM |
1 |
HUD display? |
| 1:35PM |
1 |
Post recording command to be executed after the end of recording |
| 12:41PM |
2 |
Save directly to database , and play directly from database |
| 11:54AM |
8 |
mysql sip realtime |
| 11:38AM |
1 |
Call routing between two Asterisk boxes using SIP not working ... |
| 7:49AM |
0 |
asterisk followme feature code |
| 6:00AM |
3 |
LDAP Get for Asterisk 1.6.x |
| 3:45AM |
1 |
Create VoiceMenu SNAFU |
| |
| Wednesday August 19 2009 |
| Time | Replies | Subject |
|---|
| 11:20PM |
2 |
PRI Connected to definity errors |
| 8:25PM |
0 |
AsteriskGUI Create VoiceMenu SNAFU |
| 7:33PM |
1 |
Dial plan sample for detecting Voice Mail |
| 3:56PM |
7 |
* 1.4 -> 1.6, zaptel -> dahdi |
| 1:59PM |
2 |
mysql error (err 2002) |
| 1:54PM |
2 |
outbound calls not ringing |
| 1:08PM |
0 |
Newbie: Mac OS X - Asterisk Gui 2.0 (svn) loops at " Verifying Dialplan Contexts needed for GUI" |
| 11:33AM |
1 |
Individual PIN Code per Extension |
| 10:54AM |
1 |
CAP_FOWNER=ep for asterisk |
| 9:22AM |
0 |
CDR record for call originated from manager |
| 9:22AM |
0 |
ISDN Calling Sub Address and Called Sub Address for the branches |
| 7:35AM |
2 |
Newbie: How to copy track from CD for MOH without getting "Junk at beginning of frame ..." |
| 7:16AM |
1 |
MEETME how to lock the conference if no admin are connected |
| 3:45AM |
2 |
Multi operator platform Asterisk {manage} |
| 2:02AM |
0 |
AsteriskGui 2.0.4 and TDM400P |
| |
| Tuesday August 18 2009 |
| Time | Replies | Subject |
|---|
| 11:43PM |
0 |
Moderator access to meetme allowed despite pin |
| 9:55PM |
1 |
Cisco IAD's |
| 9:49PM |
0 |
How to send Caller ID Extension to Trunk? |
| 9:44PM |
4 |
Asterisk project changes Music-On-Hold provider |
| 9:30PM |
2 |
Monitor-join not joining files in the queues.conf file |
| 8:54PM |
0 |
Date/time in queue_log |
| 8:10PM |
0 |
Asterisk Realtime : use different family names family => mysql, database, table |
| 5:52PM |
2 |
Platform decision ... |
| 4:09PM |
2 |
DAHDI - better to have card? |
| 3:38PM |
1 |
Play Fake ring in phpagi |
| 3:25PM |
3 |
IAX2 ActiveX Control |
| 2:55PM |
1 |
Get SS7 Hangup Code as Asterisk variable. |
| 2:43PM |
0 |
res_ldap.conf |
| 1:25PM |
2 |
Channels don't go away with soft hangup |
| 1:25PM |
1 |
avoid indicate condition 9 and starting music on hold |
| 12:50PM |
3 |
Zaptel -> DAHDI: now echo |
| 12:29PM |
0 |
Paging with Pickup |
| 12:29PM |
2 |
Speech Recg and TTS |
| 12:07PM |
2 |
You do not appear to have the sources for the 2.6.20-prep kernel installed |
| 8:33AM |
0 |
Asterisk + realtime applications |
| 7:28AM |
2 |
Execute some kind of script when something happens with Asterisk |
| 7:08AM |
5 |
OT - DECT handset with Line key |
| 7:03AM |
2 |
Call variables(dialstatus?) |
| 3:45AM |
7 |
Skype for Asterisk??? |
| |
| Monday August 17 2009 |
| Time | Replies | Subject |
|---|
| 10:06PM |
0 |
SipDroid |
| 6:42PM |
0 |
Echo on TE121B with hardware echo module |
| 6:23PM |
0 |
Call back DIALSTATUS is empty |
| 6:12PM |
2 |
Accessing to ekiga.net through Asterisk |
| 5:24PM |
2 |
Accessing Asterisk gosub arguments in extensions.lua |
| 4:58PM |
1 |
Goto mask |
| 4:18PM |
1 |
- Is Asterisk 1.4.21.2 Zaptel Compatible? - |
| 2:44PM |
2 |
Same number for each caller, but should reach different zap-channels, how? |
| 2:44PM |
3 |
queue_log in mysql and file |
| 2:10PM |
1 |
Problems with pstn cards |
| 12:21PM |
1 |
/usr/bin/ld: cannot find -lpq |
| 12:59AM |
3 |
Newbie: How to find the serial number of Digium card? |
| |
| Sunday August 16 2009 |
| Time | Replies | Subject |
|---|
| 1:48AM |
1 |
Detecting Called party Ring indication (and act on it) |
| |
| Saturday August 15 2009 |
| Time | Replies | Subject |
|---|
| 12:56PM |
1 |
BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0] |
| 12:30PM |
1 |
OT - Using new Asterisk calendars features |
| 2:20AM |
2 |
bare minimum /etc/asterisk for sip based * |
| |
| Friday August 14 2009 |
| Time | Replies | Subject |
|---|
| 8:27PM |
1 |
Number of Phone Numbers per Outgoing CALL File |
| 6:02PM |
1 |
Meaning of " requested special control 20, passing it to SIP" |
| 5:35PM |
2 |
Complete neutral Spanish sounds |
| 4:46PM |
1 |
Vicidial now extension setup |
| 3:52PM |
1 |
chan_dahdi refuses to build |
| 2:58PM |
1 |
i have a error in ivr |
| 2:18PM |
2 |
no ring tone |
| 2:11PM |
0 |
Call no reject when receive 'PROGRESS with cause code 27 received' in zap channel |
| 2:04PM |
2 |
CURL function with SSL |
| 2:01PM |
2 |
Cdr error? Help! |
| 10:20AM |
2 |
onnecting two asterisk using B410p BRI cards |
| 9:39AM |
1 |
play prompt after hanup |
| 8:19AM |
0 |
Astricon 2009 - dCAP |
| 7:01AM |
0 |
CPU Spikes in asterisk connected via IAX trunk |
| 3:22AM |
1 |
Stale auth messages |
| |
| Thursday August 13 2009 |
| Time | Replies | Subject |
|---|
| 8:43PM |
4 |
Time of Day Routing |
| 7:53PM |
0 |
Looking for recommendations - US SIP provider for T.38 Faxing |
| 7:21PM |
1 |
RealTime in dialplan - proper way? |
| 5:42PM |
2 |
lists.digium.com outbound mail slow? |
| 4:44PM |
1 |
Autofallthrough delays before hanging up calling channel? |
| 2:35PM |
1 |
Help for Alcatel asterisk |
| 2:21PM |
0 |
Cdr error?? |
| 2:10PM |
4 |
Snom Phones Registration/Failover Feature |
| 1:48PM |
3 |
Areski CDR + Mysql + asterisk 1.6 |
| 11:19AM |
0 |
39 Free Softphones |
| 10:25AM |
2 |
PRI Gateway - Worth it? |
| 9:58AM |
0 |
asterisk conference error/bug? |
| 8:55AM |
0 |
Conferencing and web front-end |
| 8:33AM |
2 |
Database Access from dialplan. |
| 6:51AM |
2 |
"Channels stuck up even without use" |
| |
| Wednesday August 12 2009 |
| Time | Replies | Subject |
|---|
| 7:33PM |
3 |
Creating an IAX/SIP-to-ISDN PRI gateway |
| 5:50PM |
2 |
Cdr src field fail?? |
| 4:21PM |
1 |
Why do CDR dstchannel have a strange number after them? IAX2/XXXX-???? |
| 3:49PM |
0 |
meetme conference hangs in silence after dialing |
| 3:03PM |
2 |
call drops after a few seconds |
| 1:30PM |
3 |
Asterisk + CDRTool |
| 1:25PM |
0 |
maximum dahdi tdm concurent calls and max iax trunk calls |
| 5:28AM |
4 |
Twitter is Suing me!!! |
| 4:33AM |
4 |
Cisco 79XX, SIP and Asterisk |
| 3:22AM |
1 |
app_voicemail.so: undefinied symbol: global_app_buf |
| 12:05AM |
0 |
Different From and contact header |
| |
| Tuesday August 11 2009 |
| Time | Replies | Subject |
|---|
| 8:57PM |
3 |
SIP app for iPhone that works well with Asterisk? |
| 4:33PM |
4 |
func_odbc insert with mssql |
| 4:15PM |
1 |
Cisco 1760 Multiline phone |
| 4:11PM |
1 |
Unable to compile 1.6.0.12 |
| 3:20PM |
1 |
MixMonitor and Transcoding.. |
| 2:16PM |
0 |
FSK UK Problems |
| 2:00PM |
0 |
AST-2009-005: Remote Crash Vulnerability in SIP channel driver |
| 1:59PM |
0 |
Asterisk 1.2.34, 1.4.26.1, 1.6.0.12, and 1.6.1.4 release announcement |
| 1:54PM |
0 |
asterisk 1.4 segfaults when trying to use mixmonitor |
| 1:22PM |
0 |
chan_iax2.c:1219 __send_lagrq mesages |
| 11:03AM |
1 |
testing music |
| 10:00AM |
0 |
Xfer extension to extension call, flash hookpass through by Asterisk needed via quintum and X-lite/Eyebeam |
| 4:02AM |
0 |
waitfordialtone patch |
| 3:52AM |
0 |
How to adjust the timeout to send CANCEL? |
| 2:37AM |
1 |
sflphone questions |
| |
| Monday August 10 2009 |
| Time | Replies | Subject |
|---|
| 4:51PM |
3 |
SNOM 870 |
| 4:30PM |
1 |
7940g |
| 1:34PM |
0 |
Transfer after pickup |
| 10:55AM |
6 |
"context" does not work |
| 8:36AM |
0 |
Issues with sound quality and HDLC |
| 6:16AM |
0 |
X100P FXO PCI card not receiving calls |
| |
| Sunday August 9 2009 |
| Time | Replies | Subject |
|---|
| 6:11PM |
0 |
Killing the Server |
| 2:11AM |
1 |
queue need very long can start music |
| |
| Saturday August 8 2009 |
| Time | Replies | Subject |
|---|
| 7:08PM |
0 |
Second voice caller notification |
| 7:06PM |
1 |
bad(?) TDM400 card? |
| 4:16PM |
0 |
DeadAgi application not exiting |
| 9:14AM |
4 |
Question: How to contribute to Asterisk-addons |
| 6:32AM |
1 |
A problem with recoding agents calls via monitor |
| 3:35AM |
1 |
30 Great free Asterisk applications |
| |
| Friday August 7 2009 |
| Time | Replies | Subject |
|---|
| 8:07PM |
2 |
Placing a SIP Call on Hold |
| 7:08PM |
2 |
realtime config and extensions.conf |
| 5:48PM |
2 |
caller id problem |
| 4:04PM |
3 |
Going to VM after 180 seconds in queue |
| 3:47PM |
5 |
Asterisk in VMWare, how does it perform and what is the limit? |
| 3:25PM |
2 |
Anyone had any luck with SIP clients on theiPhoneplatform? |
| 3:18PM |
1 |
regcontext regexten |
| 2:46PM |
1 |
Anyone had any luck with SIP clients on the iPhoneplatform? |
| 2:01PM |
1 |
Linksys SPA922 |
| 11:05AM |
0 |
asterisk crashes!!! |
| 9:43AM |
0 |
A problem with monitoring calls |
| 8:44AM |
1 |
Host-ID. |
| 6:18AM |
0 |
iax2_read: I should never be called - issue 8286 |
| 4:43AM |
3 |
Anyone had any luck with SIP clients on the iPhone platform? |
| 2:46AM |
0 |
Friday Aug 7th @12 Noon EDT Mobile VoIP |
| |
| Thursday August 6 2009 |
| Time | Replies | Subject |
|---|
| 8:56PM |
1 |
OT - Opensourcesip.org |
| 7:31PM |
0 |
Extensions patterns algorithm |
| 6:40PM |
1 |
No audio on remote SIP calls |
| 5:57PM |
2 |
Inbound Call coding |
| 3:54PM |
1 |
ForkCDR and setting the account info? |
| 3:32PM |
5 |
Setting up Outgoing Trunk |
| 2:28PM |
3 |
Monitoring Asterisk uptime |
| 2:21PM |
3 |
Set PHP binary location for AGI |
| 12:24PM |
2 |
Asterisk dont detects hangup by phone |
| 10:33AM |
1 |
Can't delete voicemail messages |
| 10:33AM |
0 |
chan_mobile handle 92 log flood |
| 10:27AM |
1 |
Asterisk 1.2 -> 1.4 CDR change? |
| 10:07AM |
0 |
ntop and Asterisk |
| 10:02AM |
6 |
E1 line simulation for Asterisk |
| 7:15AM |
0 |
Regarding XRMS support |
| |
| Wednesday August 5 2009 |
| Time | Replies | Subject |
|---|
| 7:36PM |
0 |
Asterisk with gizmo5 and google voice only takes one call at a time. |
| 7:17PM |
3 |
Best ISDN BRI solutions? |
| 6:12PM |
3 |
Several mailboxes on SIP peer |
| 2:25PM |
1 |
[asterisk]q: asterisk 1.6.1 install |
| 12:32PM |
2 |
sip.conf parameter and sip msg between server <-> client |
| 12:23PM |
0 |
Asterisk & sending an sms |
| 7:16AM |
2 |
original & reformat extension |
| 2:23AM |
1 |
Gizmo Dial Out No CALLED PARTY AUDIO?? |
| |
| Tuesday August 4 2009 |
| Time | Replies | Subject |
|---|
| 9:35PM |
1 |
ChangeLog revision question |
| 9:29PM |
0 |
chan_mobile future |
| 9:13PM |
4 |
Calling issue for non-extension numbers |
| 8:50PM |
1 |
MWI |
| 6:07PM |
0 |
Dahadi - Grouping Issues |
| 5:51PM |
2 |
Transfer Issue with IAX Trunk |
| 5:32PM |
2 |
Message Waiting Indicator on DAHDI line |
| 3:44PM |
0 |
Need case study for Open Source presentation |
| 2:36PM |
5 |
Anyone actively using RLT for mobile phone forwarding? |
| 1:25PM |
1 |
question on nortel CS 1000 PBX and PRI connection to built in PA system |
| 11:32AM |
0 |
dahdi_pri_error No more room in scheduler |
| 10:34AM |
4 |
CDR Problem - No CDRs when call is not bridged |
| 10:11AM |
0 |
SIP server behind NAT |
| 9:54AM |
3 |
how to implement CLONED LINE Feature in asterisk? |
| 9:10AM |
3 |
setting verbosity for asterisk cli.. |
| 8:40AM |
1 |
dahdi_scan doesn't recognize an OpenVox A400E |
| 12:05AM |
3 |
res_speech_lumenvox.so: undefined symbol: ast_speech_register |
| |
| Monday August 3 2009 |
| Time | Replies | Subject |
|---|
| 9:45PM |
0 |
dahdi_dummy soft lockup in dahdi-linux-2.2.0.2 |
| 8:29PM |
5 |
Difference between 1.4.x and 1.6.x? |
| 5:29PM |
3 |
SIP AND NAT |
| 5:04PM |
2 |
Upgrading from 1.6.1.1 to 1.6.1.2 |
| 3:55PM |
1 |
ami |
| 3:29PM |
1 |
PMR446 interface |
| 2:08PM |
4 |
single port voip gateways |
| 12:45PM |
1 |
New Digium TE207P for sale |
| 12:20PM |
1 |
Outbound calls drop after 15 to 30 seconds. |
| 4:52AM |
1 |
User Authentication in sip.conf |
| 4:30AM |
0 |
AST-2009-004: Remote Crash Vulnerability in RTP stack |
| 3:50AM |
2 |
Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1, and 1.6.2.0-beta4 Release Announcement |
| |
| Sunday August 2 2009 |
| Time | Replies | Subject |
|---|
| 10:12PM |
1 |
AstLinux 0.6.7 released |
| 7:30PM |
1 |
T.38 and reinvite |
| 6:54PM |
5 |
Modem |
| 2:30PM |
3 |
Converting sound files |
| 12:01AM |
1 |
Broadvoice versus Asterisk 1.4.25.1 and 1.4.26 |
| |
| Saturday August 1 2009 |
| Time | Replies | Subject |
|---|
| 7:32PM |
1 |
SNOM Phones Displays NR Frequently |
| 7:02PM |
1 |
Different codecs for reading and writing |
| 1:06PM |
1 |
H248 support |
| 10:46AM |
1 |
Maximum number of concurrent calls |
| 8:54AM |
0 |
Aug 1 & 16- Global VOIP Free SW HW Culture meeting, BerkeleyTIP, For Forwarding |
| 7:32AM |
3 |
Dialplan strategy suggestions needed |
| 7:22AM |
1 |
how to setup incoming calls not to use authentication |
| 5:42AM |
1 |
Inquiry : Asterisk hash key |
| 5:19AM |
0 |
Inquiry:Asterisk supporting hash (#) key |