| Thursday July 31 2008 |
| Time | Replies | Subject |
|---|
| 10:50PM |
2 |
Slighly OT?.. headset for Linksys SPA922 |
| 10:14PM |
0 |
Friday August 1st @ 12 Noon EDT |
| 9:15PM |
1 |
Panasonic Door phone monitor to Asterisk box? |
| 8:20PM |
3 |
comparing pots solutions for asterisk |
| 8:09PM |
1 |
need creative solutions for number portability |
| 6:10PM |
3 |
AA50 Failover |
| 5:55PM |
0 |
[asterisk-dev] Astricon 2008 updates: keynotes, content, contests |
| 5:53PM |
1 |
list of minutes spent on SIP phone calls?! any advice?! |
| 5:25PM |
0 |
Announcing the release of Web-MeetMe 3.0.4 |
| 5:16PM |
0 |
Unregistered indication country |
| 3:13PM |
0 |
Asterisk CDR "**Unknow**" as channel name |
| 1:56PM |
2 |
Setting up ring group |
| 1:55PM |
1 |
sip registration timeout/expiration |
| 8:52AM |
1 |
Anjelina Jolie XXX Video Free. |
| |
| Wednesday July 30 2008 |
| Time | Replies | Subject |
|---|
| 9:09PM |
5 |
Custom Filename for Incoming Agent Calls |
| 8:18PM |
0 |
It's telling me too much... |
| 8:07PM |
0 |
RES: GotoIftime |
| 7:46PM |
3 |
GotoIftime |
| 5:17PM |
1 |
Asterisk Realtime still reads from .conf files |
| 5:02PM |
0 |
Creating Call permisions in Trixbox CE 2.6.1 |
| 4:19PM |
1 |
ARA with MySQL or PostgreSQL |
| 3:49PM |
1 |
Outgoing calls authentication |
| 1:19PM |
1 |
re-distributing E1 |
| 9:31AM |
1 |
Whitepaper: How and to whom sell VoIP |
| 1:37AM |
0 |
mpg123 |
| |
| Tuesday July 29 2008 |
| Time | Replies | Subject |
|---|
| 11:57PM |
1 |
Multiple Asterisk SIP Server/client connections |
| 8:24PM |
0 |
Fallback on a fallback |
| 8:02PM |
5 |
Callerid Woes |
| 7:12PM |
0 |
Recommend Bluetooth adapters for chan_mobile? |
| 6:35PM |
0 |
Auto Dial Application |
| 6:17PM |
1 |
soundpoint 301 power adapter output? |
| 5:04PM |
1 |
Purchasing Digium IVR Prompts. |
| 4:26PM |
1 |
Addressbook solution for Cisco 7961? |
| 2:17PM |
3 |
interactive IVR |
| 2:04PM |
2 |
Asterisk SIP configuration |
| 1:45PM |
1 |
asterisk+ fax-to-mail |
| 1:32PM |
0 |
Azurn International |
| 1:01PM |
0 |
asterisk stops sending qualify |
| 7:09AM |
1 |
Outgoing calls |
| 6:59AM |
1 |
Need help with implementing prepaid in asterisk |
| 5:19AM |
1 |
One way voice after call transfer (bugs 9305, 13120) |
| 3:48AM |
9 |
Newbie in China: Red alaram in Zaptel for E1 |
| |
| Monday July 28 2008 |
| Time | Replies | Subject |
|---|
| 10:26PM |
0 |
imap voicemail is being sent to the wrong imap account |
| 10:02PM |
2 |
Remote Support |
| 9:42PM |
1 |
IVR Direct Dial Extension |
| 7:04PM |
1 |
Slow Playback of Recorded Files |
| 5:08PM |
1 |
SIP sprials and "482 Loop Detected" |
| 5:04PM |
0 |
Common Inter-Queues Leastrecent Strategy |
| 4:53PM |
0 |
How to find out RTP UDP port of active calls |
| 4:19PM |
2 |
Callcentric Issues |
| 2:14PM |
0 |
custom configuration with appliance aa50. |
| 1:42PM |
1 |
Line 0005 cannot be answered? |
| 8:58AM |
4 |
TDM400P FXO not seeing ringing after software update |
| 4:08AM |
0 |
vmail.cgi and users.conf |
| |
| Sunday July 27 2008 |
| Time | Replies | Subject |
|---|
| 6:28PM |
1 |
HASH, HASHKEYS, ClearHash explanation |
| 9:00AM |
3 |
OT - How to test tftp for phones provisioning |
| |
| Saturday July 26 2008 |
| Time | Replies | Subject |
|---|
| 8:49PM |
1 |
Visual Dial Plan |
| 6:15PM |
0 |
CME/Asterisk Voicemail Problems |
| 3:34PM |
0 |
Using manager originate and Dial() once inside dialplan |
| 10:51AM |
1 |
Need Help Regarding Asterisk |
| 10:18AM |
3 |
announcement server using asterisk |
| |
| Friday July 25 2008 |
| Time | Replies | Subject |
|---|
| 10:51PM |
0 |
Slightly Off Topic: Cisco & Premisys Slimline |
| 8:32PM |
2 |
Very loud noise on TDM400 |
| 8:06PM |
0 |
Call Center Type Recording |
| 7:21PM |
0 |
Friday's conference didn't happen... again |
| 4:17PM |
2 |
openSUSE Asterisk Packages |
| 3:18PM |
2 |
Call files with a timer? |
| 2:47PM |
0 |
console/dsp seg fault |
| 2:05PM |
0 |
AstManProxy - Blocked during 1.22fork alias rc1 install |
| 11:23AM |
1 |
I Win The Ooma Bet |
| 10:53AM |
4 |
IAX to work on two ports: 4569 and 4570 |
| 9:25AM |
3 |
AstManProxy - Where to download From ? |
| 12:40AM |
0 |
Arabic IVR |
| |
| Thursday July 24 2008 |
| Time | Replies | Subject |
|---|
| 11:59PM |
1 |
finding out on hold channels |
| 11:16PM |
1 |
different gains per channel? |
| 9:30PM |
3 |
Click to Dial |
| 8:42PM |
1 |
Implementing an Asterisk Server behinda MeridianNorstar |
| 5:41PM |
1 |
Cisco Call Manager to Asterisk conversion |
| 4:25PM |
0 |
Friday at 12 Noon EDT (9 AM Pacific) Asterisk and VoIP User Groups Worldwide |
| 3:45PM |
0 |
CallerId show with IP address appended |
| 1:50PM |
2 |
Asterisk automatic hold |
| 1:39PM |
1 |
T1/PRI dialing |
| 1:36PM |
0 |
Automatic Redialing feature |
| 1:23PM |
2 |
Acceptance testing of a new PRI |
| 12:34PM |
2 |
Audiocodes MP-11X configuration to work with Asterisk |
| 10:04AM |
2 |
Realtime + SIP + MySQL: md5secret BROKEN |
| 9:25AM |
5 |
IP door opening devices |
| 3:45AM |
0 |
Tomato = One Way Audio; Linksys = OK ???? |
| 2:37AM |
2 |
Zaptel won´t recognizes sources installed |
| |
| Wednesday July 23 2008 |
| Time | Replies | Subject |
|---|
| 11:53PM |
1 |
Raw asterisk x FreePbx .conf |
| 10:57PM |
2 |
Connect Asterisk PBX to Traditional PBX and retain functionality |
| 9:35PM |
3 |
Implementing an Asterisk Server behind a Meridian Norstar |
| 9:03PM |
1 |
Broadsoft Sip provider |
| 6:53PM |
0 |
need help setting up dundi |
| 1:45PM |
0 |
problem with asterisk 1.4.21.1 and h323 |
| 12:49PM |
3 |
Trouble Playing message file via Perl AGI |
| 10:33AM |
1 |
next priority from Dial in Asterisk 1.6 |
| 8:12AM |
1 |
1.4.21.2: Linking res_crypto causes segmentation fault. |
| 5:55AM |
4 |
How can I Disable call-waiting |
| 4:51AM |
1 |
sometimes extensions can't be called |
| |
| Tuesday July 22 2008 |
| Time | Replies | Subject |
|---|
| 11:32PM |
1 |
Suddenly my Asterisk Box Hanged up all calls |
| 11:23PM |
0 |
Asterisk 1.4.21.2 and 1.2.30 Released |
| 11:16PM |
0 |
AST-2008-011: Traffic amplification in IAX2 firmware provisioning system |
| 11:15PM |
0 |
AST-2008-010: Asterisk IAX 'POKE' resource exhaustion |
| 10:42PM |
0 |
[Fwd: Re: what is the magic needed from upgrading from 1.4 to 1.6] |
| 10:17PM |
2 |
3-way calling for IAX channels |
| 7:58PM |
1 |
Looking for a more robust Click to Dial/Web Dial solution than AsteriDex (potential for a bounty!) |
| 7:53PM |
2 |
Call Recordings... |
| 7:44PM |
0 |
?? Vitelity dtmfmode=rfc2833 started working! |
| 4:57PM |
1 |
Voicemail email to alternative ports... |
| 4:54PM |
0 |
Problems w/Asterisk Realtime + MySQL + SIP [SOLVED!] |
| 4:23PM |
0 |
Vitelity dtmfmode=rfc2833 started working! |
| 1:36PM |
1 |
issue with high latency |
| 12:52PM |
8 |
Cisco vs Asterisk |
| |
| Monday July 21 2008 |
| Time | Replies | Subject |
|---|
| 11:22PM |
1 |
Heavy Load Asterisk Array |
| 11:18PM |
2 |
RTP Packets Going To Wrong IP Address |
| 5:28PM |
0 |
Cascading Asterisk PBX |
| 5:10PM |
3 |
what is the magic needed from upgrading from 1.4 to 1.6 |
| 4:50PM |
1 |
increase ring time out |
| 4:19PM |
3 |
Asterisk Recording tools |
| 4:10PM |
3 |
Overlap dialing via SIP |
| 4:01PM |
1 |
[Posible Spam] asterisk-users Digest, Vol 48, Issue 59 |
| 3:59PM |
3 |
Help with dial plan |
| 3:18PM |
1 |
Recommend quality wholesale termination - Singapore and Sydney, Aus |
| 3:12PM |
1 |
Option 't' on DIal |
| 2:52PM |
1 |
Incompatible voice frame panic! |
| 1:16PM |
0 |
zaptel and callerid in ESTI DTMF |
| 12:39PM |
1 |
queue members randomly become paused after upgrade to Asterisk 1.4 |
| 9:29AM |
1 |
Problems with IAX on heartbeat provided ip address |
| 8:50AM |
4 |
OSLEC vs HPEC vs Octasic |
| 8:11AM |
1 |
Problems w/Asterisk Realtime + MySQL + SIP |
| 4:12AM |
0 |
Required an Auto Dialing Solution |
| 3:25AM |
0 |
New Bridge App/AMI Command in Asterisk 1.6? |
| |
| Sunday July 20 2008 |
| Time | Replies | Subject |
|---|
| 7:56PM |
1 |
Queue() AGI Bug ? |
| 5:13PM |
0 |
asterisk-users Digest, Vol 48, Issue 58 |
| 9:57AM |
2 |
Dialplan Action on Authentication |
| 5:48AM |
1 |
conference bridge |
| 3:32AM |
0 |
asterisk-users Digest, Vol 48, Issue 57 |
| 1:45AM |
1 |
Question about stopping Asterisk |
| 1:41AM |
1 |
Stop vm-intro being played |
| 12:55AM |
2 |
New Bridge Command/Event in 1.6? |
| |
| Saturday July 19 2008 |
| Time | Replies | Subject |
|---|
| 5:10PM |
1 |
asterisk-users Digest, Vol 48, Issue 56 |
| 4:41PM |
1 |
Echo Issue |
| 3:22PM |
1 |
Not a valid SIP contact - Asterisk 1.4.21.1 & Mitel SIP phones |
| 2:54PM |
1 |
Beginner Questions part II |
| 11:41AM |
2 |
Explication for ast_safe_system |
| 5:13AM |
2 |
Changinf Polycom-501 config server from remote? |
| 3:48AM |
2 |
OT Astricon/Digium Beach Ball Mailing |
| 12:52AM |
1 |
going from 1.4.21 to 1.6 beta 9 |
| 12:19AM |
0 |
asterisk not converting DTMF from INFO to rfc2833 |
| |
| Friday July 18 2008 |
| Time | Replies | Subject |
|---|
| 11:34PM |
0 |
Asterisk 1.4.21.1 |
| 8:49PM |
0 |
Announcing AstriDevCon 2008! |
| 7:21PM |
2 |
TOS and security |
| 7:16PM |
1 |
1.6b9 Audio Issue |
| 6:17PM |
1 |
Beep on transfer |
| 5:39PM |
0 |
automon follup #2 |
| 5:11PM |
1 |
Colorado Asterisk User Group Forming |
| 5:08PM |
1 |
automon followup |
| 5:02PM |
1 |
automon=>*, Dial(, , Ww), rfc2833, canreinvite=no, but... |
| 4:19PM |
0 |
Asterisk Recordings |
| 4:16PM |
5 |
GotoIf Problem |
| 2:09PM |
0 |
[asterisk-dev] How Register to ONE SIP provider with Multi Accounts |
| 2:05PM |
0 |
IAX + Inidication |
| 1:48PM |
1 |
DID - Panama |
| 10:11AM |
0 |
Asterisk Video on Hold |
| |
| Thursday July 17 2008 |
| Time | Replies | Subject |
|---|
| 11:46PM |
0 |
Help for an IAX_Client-based softphone |
| 9:15PM |
1 |
AVM Fritz BRI cards and echo cancellation |
| 8:50PM |
1 |
ATA hangs up at 30 seconds |
| 8:11PM |
1 |
WaitForSilence Problems |
| 6:32PM |
1 |
Polycom 501 transfer feature |
| 5:48PM |
1 |
Reverse Scenario |
| 5:34PM |
1 |
OpenH323 and ptlib version for asterisk 1.4.21.1 |
| 4:46PM |
1 |
Passing Account Balance to SIP Phone? |
| 4:32PM |
0 |
show channels concise parsing script? |
| 12:43PM |
9 |
Magnetic door locks |
| 10:08AM |
1 |
SIP Testing-Tool |
| 9:00AM |
0 |
Asterisk System Architect requirement. |
| 8:54AM |
0 |
Click try the to call phone tomorrow on VUC |
| 8:45AM |
0 |
Friday at 12 Noon: Asterisk and VoIP User Groups |
| 6:15AM |
1 |
1.4.21.1 SIP failing, requiring reboot |
| 12:48AM |
0 |
TeleVantage Call Monitor & Asterisk |
| 12:33AM |
6 |
Experience with Vicidial |
| |
| Wednesday July 16 2008 |
| Time | Replies | Subject |
|---|
| 8:46PM |
3 |
Zap Channel Oddity |
| 6:16PM |
1 |
D channel signalling, while B channels busy? |
| 6:05PM |
1 |
how to stop web Click to Call fraud, robots, etc |
| 4:51PM |
1 |
Disconnect on PRI ignored? |
| 4:16PM |
1 |
Specifying a different codec for meetme |
| 1:42PM |
2 |
Asterisk Recording Interface |
| 12:01PM |
5 |
Digium PRI and Echo cancellation |
| 10:58AM |
0 |
ISDN Call Droping only for outgoing |
| 10:33AM |
0 |
(no subject) |
| 4:56AM |
6 |
how to incorporate open hours |
| 3:22AM |
4 |
Two way bandwidth test |
| 2:52AM |
1 |
Asterisk CAS connection to VConsole ISDN simulator |
| 2:08AM |
4 |
asterisk + web services |
| |
| Tuesday July 15 2008 |
| Time | Replies | Subject |
|---|
| 11:47PM |
4 |
Beginner Issues |
| 10:27PM |
3 |
gui issue in asterisk aa50 |
| 9:19PM |
0 |
Adtran IP712 |
| 9:13PM |
1 |
(no subject) |
| 8:22PM |
4 |
How to monitor Asterisk logs ? |
| 6:32PM |
1 |
Meetme replacement with native 729 support |
| 6:22PM |
4 |
Toll Free International Number |
| 6:02PM |
4 |
distintive ring |
| 4:05PM |
1 |
sip prune realtime per issue |
| 3:15PM |
1 |
Music on hold |
| 3:00PM |
1 |
Reinvites and SIP/RTP |
| 12:41PM |
1 |
Interfacing pri card to legacy pbx |
| 1:03AM |
0 |
h extension priority |
| 12:24AM |
2 |
Incoming calls on zaptel not answered. |
| |
| Monday July 14 2008 |
| Time | Replies | Subject |
|---|
| 9:34PM |
2 |
Agent channel... |
| 9:20PM |
0 |
Your comments: Astricon 2008 Balloon Trip? |
| 6:56PM |
4 |
Zaptel problem with pots lines |
| 4:45PM |
1 |
fring (softphone on mobile) and open vpn |
| 4:42PM |
2 |
Asterisk behind NAT, Polycom behind NAT (SIP), how to work? |
| 1:53PM |
3 |
XORCOM BRI interfaces |
| 1:47PM |
1 |
Asterisk unable to register to tnet.it |
| 11:00AM |
4 |
How to integerate 2 TDM cards on same machine. |
| |
| Sunday July 13 2008 |
| Time | Replies | Subject |
|---|
| 10:04PM |
1 |
language problem |
| 8:59PM |
0 |
asterisk 1.4 zap instance |
| 4:45PM |
2 |
Poor audio quality with TDM400 card |
| 2:22PM |
1 |
zap not getting callerid any more |
| 10:06AM |
0 |
Unrecognized prilocaldialplan TON modifier: 5 |
| 9:17AM |
1 |
can not receive calls through pri |
| 7:30AM |
1 |
Zaptel 1.2.26 problems |
| |
| Saturday July 12 2008 |
| Time | Replies | Subject |
|---|
| 11:03PM |
2 |
Incoming call does not reach asterisk. |
| 9:52PM |
0 |
Bridging two Redirected Channels? |
| 8:30PM |
0 |
Wanted Polycom 601 + expansion sidecar |
| 2:21PM |
1 |
AsteriskNow SIP config |
| 3:46AM |
1 |
IMAP Storage Problem |
| |
| Friday July 11 2008 |
| Time | Replies | Subject |
|---|
| 11:12PM |
2 |
Recharge Dial Limit....? |
| 7:28PM |
3 |
ASTERISK/ENSWITCH ON EC2 |
| 7:13PM |
1 |
No service on phones... |
| 6:50PM |
2 |
Asterisk PBX How-to Guide for Amazon EC2 |
| 6:39PM |
1 |
Odd text in sip debug |
| 6:30PM |
0 |
libpri version 1.4.5 Released |
| 4:04PM |
0 |
Outgoing calls but no incoming calls with X100P |
| 3:54PM |
1 |
Sipura 3000 replacement ---> SPA3102 how reliable is it? |
| 3:07PM |
0 |
Asterisk Fails to convert INFO to Inband |
| 2:57PM |
0 |
SIP timing out over satellite connection on 1.4.21 (works with 1.4.18.1) |
| 2:17PM |
3 |
Incoming |
| 11:13AM |
0 |
C450 broken rtp handling |
| 9:16AM |
1 |
Microsoft CRM 4.0 integration with asterisk |
| 7:50AM |
0 |
Analog lines dtmf problem |
| 2:54AM |
1 |
Asterisk cant play sounds from AGI |
| |
| Thursday July 10 2008 |
| Time | Replies | Subject |
|---|
| 11:36PM |
1 |
Should I remove the blank options? |
| 9:19PM |
1 |
Diagnosing dropped calls... |
| 5:25PM |
16 |
Asterisk as an IVR solution |
| 4:49PM |
6 |
Tracking Call Time While in Dial() |
| 4:44PM |
0 |
Festival issues |
| 2:44PM |
0 |
Asterisk hangup not working on inbound calls |
| 1:36PM |
0 |
Why it keeps display the G729 codec during the call running on the consol |
| 1:17PM |
1 |
Friday June 11th: SIP love/hate |
| 12:46PM |
1 |
RTP packets dropped |
| 12:16PM |
1 |
Asterisk conference call with a HuntGroup |
| 11:47AM |
0 |
callerid_get_dtmf: Couldn't detect start-character. CID parsing might be unreliable |
| 11:13AM |
0 |
Simple Call Screener |
| 11:11AM |
1 |
res_odbc.conf and odbc show |
| 11:00AM |
0 |
Why is the h extension being called ? |
| |
| Wednesday July 9 2008 |
| Time | Replies | Subject |
|---|
| 11:08PM |
5 |
changing inbuilt sound messages |
| 8:50PM |
0 |
** app_swift v1.2.2 released for Asterisk 1.2.x code-base ** |
| 7:39PM |
1 |
asterisk 1.2.21.1 seg fault |
| 7:28PM |
2 |
Zap Bridged Channels |
| 7:15PM |
0 |
e911/CAMA/MF |
| 5:49PM |
0 |
** app_swift v1.6.2 released for Asterisk 1.6.x code-base ** |
| 3:28PM |
0 |
question about fxo cards |
| 3:07PM |
0 |
disable DTMF on a particular channel |
| 2:50PM |
2 |
OT: DNS security |
| 2:08PM |
3 |
READ application |
| 1:24PM |
2 |
Zap Bridged Calls do not continue dialplan |
| 12:14PM |
2 |
Proper Hangup message |
| 11:55AM |
1 |
asterisk sip problem |
| 11:33AM |
0 |
cron jopb |
| 10:47AM |
0 |
H.323 <-dtmf-> |
| 10:36AM |
1 |
change E1 link from ISDN to Q.SIG |
| 9:25AM |
0 |
Default table layout for cdr logging with Mysql |
| 8:30AM |
0 |
Simple call accept test |
| 8:03AM |
2 |
transfers only work when voicemail enabled |
| 7:01AM |
2 |
Asterisk dimensioning |
| 6:35AM |
0 |
Problem Asterisk |
| 4:55AM |
2 |
cell phone hangup not getting recognised by system |
| 12:04AM |
3 |
Distinctive Ring for SIP? |
| |
| Tuesday July 8 2008 |
| Time | Replies | Subject |
|---|
| 9:49PM |
0 |
Trouble with faxing using iaxmodem / hylafax |
| 9:32PM |
1 |
modules/cdr_odbc.so |
| 6:59PM |
0 |
** app_swift v1.4.2 released for Asterisk 1.4.x code-base ** |
| 4:53PM |
2 |
astrundir not used |
| 2:55PM |
1 |
CONSOLE logging |
| 12:56PM |
3 |
(announce) asterisk T.38 gateway |
| 12:47PM |
0 |
Asterisk STILL loosing IAX user's registration |
| 10:04AM |
0 |
CallerID in The Netherlands with TDM11B |
| 9:48AM |
0 |
has anyone worked with nxtvox fxo cards |
| 9:23AM |
1 |
asterisk and polycom provisioning |
| 9:15AM |
0 |
Asterisk 1.4 restarts after parking using AGI |
| 8:35AM |
2 |
realtime outgoing |
| 1:58AM |
1 |
rxfax not receiving faxes |
| 1:15AM |
3 |
Sharing unused minutes between Asterisk users |
| 12:37AM |
0 |
AsteriskWatch FaceBook application |
| |
| Monday July 7 2008 |
| Time | Replies | Subject |
|---|
| 11:33PM |
0 |
Audio data from ast_speech_write |
| 10:50PM |
2 |
Help with sip configuration |
| 10:22PM |
1 |
First-time queue app: verifying human member? |
| 10:03PM |
2 |
QueueMemberStatus |
| 9:31PM |
1 |
SIP or SCCP for cisco |
| 8:39PM |
8 |
US T1 Hangup Detection |
| 7:10PM |
0 |
chan_alsa resource temporarily unavailable |
| 5:21PM |
2 |
Building an IVR |
| 5:17PM |
2 |
Cisco 7940 not getting PoE from Linksys SLM224P |
| 5:05PM |
1 |
cdr_addon_mysql - additional fields |
| 5:03PM |
0 |
Return VXML vars to Dial Plan |
| 4:59PM |
0 |
SIP MWI Problem in 1.4 and 1.6 |
| 2:46PM |
3 |
sippyskype |
| 2:23PM |
1 |
Click-to-talk (Java application) |
| 12:48PM |
1 |
DTMF on iax channel is not interpreted by asterisk |
| 10:58AM |
5 |
Meetme |
| 10:16AM |
1 |
queue member state |
| 10:09AM |
1 |
ATA gateway |
| 9:26AM |
2 |
Codec negotiation for Thomson ST2030 and g729 |
| 6:08AM |
0 |
trixbox + GXE5024 peer |
| 12:32AM |
2 |
dial plan help. |
| |
| Sunday July 6 2008 |
| Time | Replies | Subject |
|---|
| 10:48PM |
1 |
delay when rinigng asterisk |
| 7:12PM |
1 |
Sipura SPA-3102 and Asterisk |
| 4:23PM |
0 |
Zaptel and Solaris X86 |
| 10:28AM |
0 |
Documentation for realtime text support in Asterisk |
| 7:11AM |
1 |
Eeepc + Asterisk + Video conferencing |
| |
| Saturday July 5 2008 |
| Time | Replies | Subject |
|---|
| 11:33PM |
4 |
HR 5889. |
| 8:39PM |
1 |
New Polycom SpectraLink 8002 Wifi SIP Handset |
| 5:50PM |
1 |
Cell phone to PSTN adapter or IAX |
| 4:32PM |
0 |
Return Vars to Dial Plan from VXML() |
| 3:48PM |
0 |
Read & Background |
| 3:13PM |
1 |
Asterisk loosing IAX users's registration?? |
| 12:00PM |
2 |
Require Billing solution for Calling Cards retail... |
| 12:59AM |
2 |
[asterisk-dev] Locking, coding guidelines addition |
| |
| Friday July 4 2008 |
| Time | Replies | Subject |
|---|
| 3:27PM |
0 |
background noise |
| 2:21PM |
0 |
Dear asterisk-users@lists.digium.com SALE 85% 0FF on Pfizer |
| 1:30PM |
1 |
Call Forwarding Lopp Prevention |
| 11:57AM |
2 |
Removing voicemail messages |
| 11:53AM |
0 |
Bug tracker having issues |
| 5:25AM |
5 |
DIDs required of Paris and Gottenburg Sweden |
| |
| Thursday July 3 2008 |
| Time | Replies | Subject |
|---|
| 9:02PM |
2 |
Asterisk VXML... Help. |
| 7:26PM |
2 |
Spoofing CID |
| 2:50PM |
0 |
how to setup one stage dialing plan, instead of two! help!!! |
| 2:33PM |
0 |
D-Link DVG-3104MS |
| 1:45PM |
1 |
wait & pickup |
| 12:51PM |
2 |
problem in making call pc to phone & vice versa |
| 12:50PM |
1 |
(no subject) |
| 10:52AM |
0 |
OLPC Sound Samples |
| 10:29AM |
1 |
(no subject) |
| 9:17AM |
0 |
asterisk queues and database backend (clustered realtime) |
| 8:40AM |
1 |
Dial function exit, go to line n+1 |
| 2:54AM |
3 |
2 AVM ISDN Fritzcards |
| 12:48AM |
0 |
agi never leg1 disconnect |
| |
| Wednesday July 2 2008 |
| Time | Replies | Subject |
|---|
| 10:44PM |
1 |
ooh323 doesn't know what to do when bridging calls? |
| 9:29PM |
1 |
Tone Differentiation |
| 8:51PM |
0 |
Dial duration |
| 7:59PM |
5 |
How to change http port on appliance? |
| 5:16PM |
2 |
Does an IAXy require registration? |
| 4:34PM |
1 |
Asterisk Taking CPU resources |
| 3:57PM |
0 |
Can you verify this bug? |
| 12:05PM |
2 |
new install of asterisk appliance. |
| 8:59AM |
0 |
asterisk-users Digest, Vol 48, Issue 4 |
| 12:51AM |
0 |
Config help with ISDN Fritzcard |
| |
| Tuesday July 1 2008 |
| Time | Replies | Subject |
|---|
| 10:57PM |
1 |
Best Practices: Empirical measure of call latency |
| 10:25PM |
0 |
Cannot dial on E1 cards |
| 7:30PM |
3 |
Asterisk 1.4.21.1: Bugs in IAX |
| 7:27PM |
3 |
Waiting time to send the call |
| 7:20PM |
1 |
queue show name - callerID |
| 5:50PM |
1 |
Broadvoice and Asterisk 1.6.0-beta9 |
| 3:29PM |
4 |
The S word: Asterisk security |
| 2:20PM |
1 |
Asterisk 1.4.21 and CUT function |
| 1:35PM |
4 |
Fax Between IAX Trunks |
| 1:33PM |
3 |
music on hold realtime |
| 12:38PM |
17 |
Call quality |
| 12:34PM |
0 |
Panama SIP ITSP? |
| 12:25PM |
1 |
Click to Dial Service Providers in Australia |
| 11:09AM |
1 |
User unable to use DTMFs? |
| 10:35AM |
0 |
line goes silent for a few seconds at the start of outgoing calls |
| 8:09AM |
0 |
Manager proxy |
| 7:15AM |
4 |
Choppy audio |
| 2:39AM |
0 |
Queue recording file name |
| 12:44AM |
2 |
Disto choice for Asterisk with AVM Fritz!PCI cards |