| Tuesday July 31 2007 |
| Time | Replies | Subject |
| 10:45PM |
0 |
Number disappears when picking up a call |
| 10:35PM |
5 |
Dropouts and echo |
| 10:34PM |
1 |
DTMF integration pana d500 |
| 9:24PM |
1 |
not hearing dtmf tones |
| 8:13PM |
1 |
Problems using TE412P and TDM400B in a IBM x3650 |
| 7:55PM |
5 |
.call file problem |
| 5:43PM |
0 |
AsteriskNOW and Custom VoIP |
| 5:41PM |
0 |
Repost: Aastra BLF directed pickup: anyone have this pickup-mgernoth file? |
| 5:23PM |
2 |
Connecting GSM Phone to Asterisk Box |
| 5:21PM |
0 |
AstriCon -- Last chance to speak! |
| 5:20PM |
1 |
Wrinkled faxes or missing lines with Hylafax + IAXModem + Asterisk |
| 5:14PM |
0 |
FW: NYC Special Event - Join Us |
| 4:35PM |
0 |
OT: Polycom Directory XML via PHP |
| 3:42PM |
1 |
Zaptel update trouble |
| 3:08PM |
1 |
g729 setup help |
| 2:16PM |
1 |
Queue Time to Speak to Agent Algorithm? |
| 1:48PM |
0 |
ExternalIVR() broken in 1.4.9? |
| 1:26PM |
1 |
Turn off SIP 183 Session Progress in Asterisk 1.4.8 |
| 1:21PM |
2 |
Welcome to the "asterisk-users" mailing list (Digest mode) |
| 12:39PM |
3 |
1and1 dedicated servers have been down for a few hours . |
| 12:37PM |
0 |
SIP Refer ... rejected? |
| 9:15AM |
3 |
asterisk on 64-bit? |
| 5:36AM |
3 |
Royalty for On Hold Music ? |
| 5:06AM |
2 |
asterisk or asterisknow |
| 2:53AM |
0 |
Zaptel compiling broken: error: conflicting types for '__kernel_dev_t' |
| |
| Monday July 30 2007 |
| Time | Replies | Subject |
| 10:53PM |
0 |
Asterisk with Speechphone |
| 9:40PM |
1 |
Manager - QueueAdd |
| 9:29PM |
7 |
software bloat - is this really useful to anyone? |
| 8:46PM |
0 |
RTP Session Streaming |
| 8:05PM |
0 |
String truncation problems on FreeBSD Sparc64 |
| 6:59PM |
0 |
Questions about SPA3102. |
| 5:30PM |
1 |
MeetMe through DeadAGI has changed to return -1 on Hangup |
| 4:23PM |
0 |
AGI Que "Say Time" |
| 4:10PM |
0 |
Zombie (Masqueraded) Channel CDR Problem |
| 4:00PM |
5 |
Silly MeetMe() question. |
| 3:10PM |
0 |
Trouble getting sound from a call |
| 3:09PM |
1 |
Dial plan question: PSTN via Linksys SPA3102 then IAX if busy? |
| 2:46PM |
2 |
Strange ISDN Troubles |
| 2:21PM |
1 |
iax2 trunk registration with auth rsa |
| 1:55PM |
1 |
AGI and exec Playback |
| 1:54PM |
2 |
Creating an SIP softphone |
| 1:45PM |
3 |
Description for each sound files |
| 1:32PM |
0 |
announcement server |
| 1:32PM |
0 |
What is SIP conntrack status ? |
| 1:00PM |
1 |
G729 licenses installed - voicemail has no audio... |
| 12:38PM |
1 |
How to use 1 channel from TE110P for data transmission |
| 12:24PM |
6 |
outbound caller ID |
| 12:01PM |
7 |
Zaptel channel reservation |
| 11:19AM |
3 |
Lightweight IAX balancer |
| 9:40AM |
2 |
TE212 or TE220 |
| 8:14AM |
1 |
how to configure zaptel for incoming call |
| 6:34AM |
1 |
Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box |
| 6:10AM |
4 |
Huntgroup with asterisk feature |
| 6:09AM |
0 |
codian with asterisk voice confrance |
| 5:13AM |
0 |
asterisk 1.4.8 and google talk - no audio |
| 4:19AM |
1 |
Programming with libiax2 |
| 4:03AM |
1 |
Queues with logged in agents that are not reachable |
| |
| Sunday July 29 2007 |
| Time | Replies | Subject |
| 2:05PM |
2 |
Dial from Phonebook of Evolution or Thunderbird |
| 3:27AM |
0 |
Asterisk 1.4.X support for Solaris 10? |
| |
| Saturday July 28 2007 |
| Time | Replies | Subject |
| 10:35PM |
1 |
Calling to users in other asterisk servers |
| 3:07PM |
0 |
New York Asterisk Meetup Aug 9th |
| 9:18AM |
3 |
global variables and updates |
| 6:41AM |
1 |
Query2 |
| 6:33AM |
1 |
Query1 |
| 2:34AM |
0 |
SVP support |
| |
| Friday July 27 2007 |
| Time | Replies | Subject |
| 8:31PM |
0 |
Astricon 2007 |
| 6:02PM |
1 |
Nufone problems |
| 5:49PM |
2 |
SIP "Max Channels" Setup |
| 4:09PM |
2 |
Unicall/Dont know how to handle Accepted |
| 3:57PM |
1 |
Locking a device to a codec |
| 3:15PM |
1 |
Will the Sangoma A40003X fit in a 2950? |
| 3:08PM |
1 |
Avaya SIP phones (4610SW) and MWI |
| 1:58PM |
0 |
Asterisk Users Conference Friday at 12:30 PM, EDT |
| 1:46PM |
4 |
CONSOLE=Phone/phone0 and CONSOLE=Console/dsp and Zap/g2 |
| 1:33PM |
0 |
Telco Testing locks up asterisk |
| 1:27PM |
1 |
chan_mISDN module does not load |
| 1:26PM |
4 |
Asterisk Wiki |
| 12:26PM |
1 |
Asterisk advanced concepts |
| 12:21PM |
1 |
asterisk meetme confrance problem |
| 10:09AM |
1 |
Queues strategy "leastrecent" |
| 8:40AM |
1 |
ISDN: Problems starting off [another attempt] |
| 8:26AM |
0 |
Keep playing Background while dialling invalid dtmf extensions |
| 8:16AM |
1 |
Asterisk Users Conference Friday at 12:30 PM EDT |
| 7:57AM |
0 |
auto dialout call status |
| 7:06AM |
1 |
Can someone Stop this autoreply stuff????? |
| 6:53AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 6:50AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ? |
| 6:50AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails |
| 6:48AM |
0 |
Please unsubscribe or moderate rp@raha.cc?!?!... |
| 6:47AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE |
| 6:44AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 6:43AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ? |
| 6:43AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails |
| 6:42AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE |
| 6:40AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 6:40AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ? |
| 6:39AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails |
| 6:37AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE |
| 6:37AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 6:36AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ? |
| 6:36AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails |
| 6:32AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE |
| 6:31AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 6:30AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ? |
| 6:29AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails |
| 6:26AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 6:26AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE |
| 6:26AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ? |
| 6:25AM |
0 |
Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails |
| 6:23AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 6:23AM |
0 |
Autoreply: Autoreply: Autoreply: Re: Display IE |
| 6:22AM |
0 |
Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails |
| 6:22AM |
0 |
Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ? |
| 6:19AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 6:19AM |
0 |
Autoreply: Autoreply: Re: Display IE |
| 6:19AM |
0 |
Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ? |
| 6:14AM |
0 |
Autoreply: Re: Attaching VoiceMails on E-Mails |
| 6:11AM |
0 |
Autoreply: Is it possible transcode (ilbc <-> g.729) in software ? |
| 6:09AM |
0 |
Autoreply: Re: Display IE |
| 6:06AM |
0 |
Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 6:01AM |
0 |
Is it possible transcode (ilbc <-> g.729) in software ? |
| 5:59AM |
0 |
Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 5:56AM |
0 |
Autoreply: Autoreply: Re: Dialtone when automatically picking up. |
| 5:53AM |
2 |
Sangoma on Fedora 7 x86_64 |
| 5:52AM |
0 |
Autoreply: Re: Dialtone when automatically picking up. |
| 5:39AM |
0 |
Autoreply: Re: Newbie Advice on Asterisk and Linux |
| 5:33AM |
1 |
Autoreply: Queue stats |
| 5:25AM |
0 |
Autoreply: Re: SunRocket / ALLO / etc special offer |
| 4:36AM |
6 |
polycom custom ring tones (slightly OT) |
| 4:18AM |
0 |
Autoreply: Re: Queue stats |
| 2:31AM |
3 |
Need help with inbound IAX |
| 2:24AM |
3 |
Default Asterisk Numbers |
| 2:23AM |
1 |
Problems with new logic being 'n' option to Queue in 1.4.9 |
| 12:55AM |
2 |
Attaching VoiceMails on E-Mails |
| |
| Thursday July 26 2007 |
| Time | Replies | Subject |
| 11:08PM |
2 |
ISDN: Problems starting off |
| 10:17PM |
0 |
(Update) Digium FTP server will be replaced with HTTP server |
| 8:57PM |
2 |
SetCallerPres and Cisco PRI |
| 8:51PM |
1 |
Ring forever |
| 8:20PM |
1 |
vm-duration announcement missing? |
| 8:08PM |
3 |
Newbie Advice on Asterisk and Linux |
| 6:29PM |
0 |
missing digits on PRI |
| 6:23PM |
1 |
Grandstream RTP keepalive packets causing Asterisk warning |
| 5:10PM |
3 |
Digium FTP server will be replaced with HTTP server |
| 2:37PM |
7 |
Queue stats |
| 1:43PM |
8 |
IAX connections broken |
| 1:28PM |
2 |
Asterisk Conference Call |
| 12:37PM |
1 |
Asterisk 1.2.23 and Sangoma a102 no incoming calldetected |
| 12:01PM |
0 |
Queue stats from the dial plan |
| 11:55AM |
10 |
Query |
| 10:13AM |
1 |
tdm400p fxs module busy |
| 6:06AM |
0 |
strange problem in asterisk + mediant2k setup |
| 4:09AM |
0 |
Asterisk 1.4.9 reproducibly dumps core on Solaris 10 |
| 4:03AM |
1 |
Asterisk 1.2.23 and Sangoma a102 no incoming call detected |
| |
| Wednesday July 25 2007 |
| Time | Replies | Subject |
| 9:25PM |
1 |
Call report by pinset |
| 9:07PM |
0 |
Problem with asterisk-addons - checking for mysql_init in -lmysqlclient... no |
| 9:05PM |
1 |
Dialtone when automatically picking up. |
| 7:53PM |
1 |
Asterisk-1.2 and Centos 5 |
| 7:26PM |
0 |
Asterisk with RFC 3313 support |
| 7:11PM |
1 |
WAV49 output in sox |
| 5:48PM |
1 |
Asterisk Supported Harware Architecture |
| 5:35PM |
5 |
IAX2 INBAND DTMF? |
| 5:20PM |
3 |
Asterisk Vm functionality question |
| 4:50PM |
2 |
X100P pass through questions |
| 4:10PM |
3 |
Asterisk 1.4.9.tar.gz download fails |
| 3:27PM |
0 |
Asterisk Freeze Problem |
| 2:46PM |
1 |
international calls with national rates |
| 2:32PM |
1 |
help with mfcr2 and pri |
| 2:22PM |
0 |
Parking calls via cli / manager / dialplan |
| 1:52PM |
1 |
Add prefix digits in dialplan extention |
| 12:55PM |
2 |
Display IE |
| 12:13PM |
1 |
SunRocket / ALLO / etc special offer |
| 9:37AM |
0 |
Help with SIP to PBX |
| 4:36AM |
1 |
Post voicemail processing. |
| |
| Tuesday July 24 2007 |
| Time | Replies | Subject |
| 11:36PM |
0 |
Asterisk 1.2.23 and 1.4.9 released |
| 11:26PM |
0 |
ASA-2007-018: Resource Exhaustion vulnerability in IAX2 channel driver |
| 9:23PM |
0 |
How I can configure asterisk to register as gatekeeper server with another gatekeeper |
| 9:19PM |
1 |
Testers needed for VoIP router solution |
| 9:09PM |
10 |
What is the best softphone work with Asterisk |
| 8:27PM |
2 |
SIP IP Trunk, between Asterisk and Softswitch |
| 8:02PM |
3 |
rxFAX core dumps |
| 6:32PM |
0 |
mISDN & Asterisk 1.4: HFC-S card not responsive |
| 4:51PM |
0 |
Problem Hangup Help |
| 4:48PM |
1 |
Digium adaper S101I - IAXy Losing connection |
| 4:45PM |
0 |
VoiceTronix 12 + Far-End Hangup Detection |
| 4:25PM |
1 |
SIP jitter buffer and asterisk native bridge |
| 3:44PM |
1 |
[beginner] Problem of detecting call |
| 12:17PM |
1 |
MySQL components in asterisk-addons not being built |
| 9:43AM |
0 |
Diva Server BRI hangs up after about 25 seconds |
| 9:27AM |
2 |
Dial out through multiple Zap groups |
| 3:47AM |
1 |
CallerID from POTS to SIP |
| 2:21AM |
2 |
Astribank-8BRI |
| |
| Monday July 23 2007 |
| Time | Replies | Subject |
| 10:54PM |
1 |
G729 with SIP and H.323 |
| 8:19PM |
1 |
TDM04B & FIOS No Hangups Often |
| 8:09PM |
1 |
is there a tool that gives DTMF information on T1 |
| 7:31PM |
0 |
Can Asterisk hear on two IP addresses? |
| 7:30PM |
0 |
Problem w/ MySQL update from perl AGI script |
| 6:56PM |
0 |
chan_mobile - no Audio (Asterisk + bluetooth + chan_mobile) |
| 6:33PM |
3 |
Dialplan |
| 5:40PM |
0 |
Fwd: Asterisk and COS bits |
| 4:41PM |
2 |
Voicemail .lock- files voicemail box not accessible |
| 3:49PM |
0 |
app_changrab, replacement for meetme and conference: returning to dialplan |
| 3:36PM |
0 |
Asterisknow recompile kernel module |
| 2:55PM |
2 |
Problem Hangup |
| 2:53PM |
2 |
IAX Encryption |
| 1:44PM |
6 |
phone directory with asterisk |
| 1:26PM |
7 |
Polycom IP 4000 Soundstation SIP Conference Phone Question |
| 9:54AM |
0 |
users in sip.conf or in users.conf? what is the difference? |
| 8:42AM |
0 |
1.4.8 jabber integration |
| 8:40AM |
2 |
ODBC Connection failed |
| 6:59AM |
1 |
VPN on Asterisk |
| 6:38AM |
1 |
Can Asterisk hear on two IP addresses? And can I do |
| 4:16AM |
3 |
extension.conf doesn't reload? |
| 4:15AM |
2 |
Upgrade and keep the configuration |
| 4:11AM |
0 |
IP Trunk between Asterisk and another IP PBX |
| 2:28AM |
0 |
CAS signalling and FAX solution |
| 12:33AM |
1 |
Viable Alternatives to TDM400P |
| |
| Sunday July 22 2007 |
| Time | Replies | Subject |
| 5:58PM |
1 |
DTMF recognition problem with PSTN |
| 4:45PM |
3 |
Debian etch and web voice mail - how to configure it? |
| 3:54PM |
2 |
Asterisk CTI interface to control legacy PBX |
| 2:29PM |
1 |
Wake-Up Call didn't work |
| 1:07PM |
0 |
te110p stays in red alarm |
| 11:32AM |
3 |
Music on Hold and Announcements |
| 10:37AM |
1 |
IMAP and ODBC voicemail storage |
| 9:53AM |
1 |
Asterisk-1.2.22 DeadAGI Hangup |
| 7:58AM |
1 |
Problem X100P - clone asterisk 1.4.8 no hangup |
| 1:34AM |
0 |
AstLinux 0.4.7 |
| 12:24AM |
1 |
Asterisk and COS bits |
| |
| Saturday July 21 2007 |
| Time | Replies | Subject |
| 5:11PM |
0 |
700Mhz Spectrum |
| 1:38PM |
1 |
Call Initiation with Asterisk |
| 12:51PM |
0 |
New tutorial: compiling Asterisk 1.4 with zaptel and H323 support |
| 10:52AM |
10 |
Best and easiest soft phone for my Dad.. |
| 3:22AM |
0 |
asterisk-users Digest, Vol 36, Issue 61 |
| 1:03AM |
1 |
Configuring Sangoma A101D with Asterisk 1.2.18 & zaptel-1.2.17.1 |
| 12:44AM |
3 |
Has anybody used fanless computers of logic supply with asterisk? |
| |
| Friday July 20 2007 |
| Time | Replies | Subject |
| 11:55PM |
1 |
POE injector |
| 9:34PM |
1 |
Aastra phones loosing service... |
| 9:01PM |
2 |
Announcing Digium/Asterisk World's Conference Program |
| 8:33PM |
2 |
Problem |
| 7:16PM |
1 |
ulaw to g726 conversion |
| 4:13PM |
1 |
Asterisk IVR Performance |
| 2:30PM |
1 |
asterisk novice needs help. |
| 12:28PM |
2 |
asterisk hang up the Caller after recording voicemail |
| 12:15PM |
0 |
sip softphone for PDA window mobile 2003 / 5.0 ? |
| 12:10PM |
1 |
Which IP Phones will work with non-Asterisk PBX systems too? |
| 11:32AM |
0 |
Asterisk Channel and VLC |
| 10:31AM |
0 |
Caller is hanged up after recording voicemail |
| 9:45AM |
2 |
priorityjumping not working, Dial goes to n+1 not n+101 |
| 8:38AM |
3 |
Asterisk Freeze |
| 6:41AM |
3 |
pattern base call routing |
| 3:32AM |
2 |
* core file not recognized |
| 1:15AM |
2 |
Any plans for proper faxing support |
| |
| Thursday July 19 2007 |
| Time | Replies | Subject |
| 10:34PM |
1 |
Polycom 650 freezing on Transfer |
| 9:16PM |
0 |
job opportunity |
| 9:03PM |
0 |
Any open source OSS system for asterisk? |
| 6:37PM |
2 |
Problem after upgrading from 1.2.21.1 to 1.2.22 |
| 6:01PM |
0 |
Blank Voicemails/Vonage Problem |
| 5:24PM |
1 |
Idefisk softphone - official 2.0 release - Zoiper |
| 4:55PM |
4 |
Why using usecallerid=no? |
| 4:05PM |
1 |
iaxtel.com down? |
| 3:28PM |
2 |
PRI Card |
| 2:41PM |
8 |
Blank Voicemails |
| 2:05PM |
1 |
1.4.X howto disable able xpp with ./configure |
| 1:55PM |
2 |
open up firewall ports for Asterisk - safe? |
| 1:29PM |
0 |
Does anyone have the file: pickup-mgernoth-2006-07-28.patch.txt |
| 1:20PM |
5 |
G729 copy protection |
| 1:16PM |
1 |
Parsing IAXPeers from Asterisk Manager (PHP API) |
| 12:52PM |
2 |
Upgrade Procedure |
| 10:58AM |
1 |
asterisk libraries dependecies |
| 9:30AM |
0 |
Asterisk with 2 Public IP-Is it possible? |
| 7:44AM |
1 |
Force asterisk to re-resolve dns names? |
| 5:30AM |
1 |
New book "Asterisk Cookbook" any good? |
| 4:07AM |
2 |
Gtalk/Jabber connect issues in 1.4.8 |
| |
| Wednesday July 18 2007 |
| Time | Replies | Subject |
| 11:41PM |
1 |
cdr_addon_mysql.so is not created |
| 11:37PM |
5 |
In Vancouver is it a local to call from 778 to 604, and vice versa? |
| 9:23PM |
0 |
Ncurses dependencies |
| 9:14PM |
2 |
Flash(), Centrex Lines, and 3 way calling |
| 9:01PM |
3 |
Redundancy / Failover |
| 8:47PM |
0 |
Does Asterisk support STUN or TURN? How to configure asterisk NAT traversal? |
| 7:39PM |
2 |
Force SIP hang up. |
| 5:07PM |
1 |
Error Configuring Asterisk (FREEPBX) |
| 4:40PM |
3 |
Remote vm system message pickup |
| 3:32PM |
3 |
Problem building Asterisk 1.2.22 |
| 2:12PM |
2 |
what codecs for LAN |
| 2:11PM |
0 |
calls dropped with te110p E1 |
| 2:09PM |
1 |
large setup for asterisk |
| 2:09PM |
1 |
Issue in insatlling addons-1.4.2 |
| 2:06PM |
3 |
how to use call transfer |
| 12:18PM |
0 |
PRI Change Channel Identification from Exclusive to Preferred or Negotiation? |
| 12:00PM |
1 |
Asterisk Voicemail Imap Storage with MS Exchange |
| 11:25AM |
0 |
Queue to outgoing Zap channels when congested |
| 10:31AM |
2 |
E1 Virtual Callcenter |
| 9:51AM |
1 |
blind transfer on hook-flash from SIP phone |
| 7:47AM |
0 |
How to change Zap channel negotiation/exclusive etc..? |
| 4:00AM |
1 |
AudioCodec MP114 |
| 1:04AM |
1 |
Any way to determine remote Asterisk version |
| |
| Tuesday July 17 2007 |
| Time | Replies | Subject |
| 10:44PM |
4 |
bristuff for hfc card on Xscale 80219 |
| 10:30PM |
0 |
ASA-2007-017: Remote crash vulnerability in STUN implementation |
| 10:28PM |
0 |
ASA-2007-016: Remote crash vulnerability in Skinny channel driver |
| 10:27PM |
0 |
ASA-2007-015: Remote Crash Vulnerability in IAX2 channel driver |
| 10:25PM |
0 |
ASA-2007-014: Stack buffer overflow in IAX2 channel driver |
| 10:22PM |
0 |
Critical Updates: Asterisk 1.2.22 and 1.4.8 released |
| 8:46PM |
1 |
No sound from Festival, but *something* is happening |
| 7:39PM |
7 |
Asterisk 1.4, Unicall and Nextel... |
| 7:22PM |
0 |
Can Asterisk hear on two IP addresses? And can Ido routing for calls from private to public or public toprivate IP addresses |
| 5:54PM |
1 |
Asterisk 1.4.6 crash using queue app |
| 4:26PM |
2 |
media not accpetable with outgoing call on cisco |
| 2:09PM |
2 |
2 PRI on asterisk |
| 2:09PM |
1 |
Asterisk and ATA-186 question-- calling one port from the other port.. |
| 1:43PM |
5 |
Zap channels unavailable? |
| 12:59PM |
1 |
chan_isdn with HFC-compatible |
| 12:24PM |
1 |
Problem with H option of Dial() |
| 12:06PM |
1 |
Not hearing the caller after 2 x Flash |
| 12:04PM |
1 |
asterisk web interface |
| 12:02PM |
0 |
help with sip configuration for sipgate.de on asterisk 1.4 |
| 11:57AM |
0 |
No music on hold on ISDN line |
| 11:04AM |
0 |
Suggestion for installation |
| 9:26AM |
0 |
Digitized audio at the beginning of a call |
| 9:25AM |
1 |
Music on hold problem |
| 8:56AM |
5 |
Asterisk PRI Busy Problem |
| 5:15AM |
2 |
Asterisk Hosting (Dedicated Servers) |
| 2:23AM |
1 |
MultiParking |
| 1:32AM |
1 |
DTMF regeneration on PRI |
| |
| Monday July 16 2007 |
| Time | Replies | Subject |
| 11:43PM |
2 |
CID on Polycom Phones |
| 10:38PM |
1 |
OT: How to compile Zaptel for different kernel version... |
| 10:27PM |
0 |
No audio after updated Asterisk 1.4.7.1 |
| 6:51PM |
3 |
Zaptel 1.2.19 and 1.4.4 released |
| 6:38PM |
0 |
Pri problems |
| 5:58PM |
2 |
OT - Cisco Callmanager System Prompts |
| 4:57PM |
0 |
Dial and option G |
| 4:14PM |
0 |
Dual dtmfmode? |
| 3:30PM |
0 |
Registration Interval |
| 3:13PM |
1 |
asterisk 1.4 and gnugk with ooh323 |
| 2:44PM |
4 |
USB Cordless |
| 2:31PM |
3 |
Crontab script to check health of Asterisk server? |
| 12:48PM |
1 |
Cisco 7940 log on/off |
| 10:01AM |
1 |
[Asterisk]Asterisk's behavior of a simple call |
| 8:36AM |
4 |
I want to record each phone call |
| |
| Sunday July 15 2007 |
| Time | Replies | Subject |
| 7:38PM |
0 |
choppy sound when transcoding (after os update) |
| 6:49PM |
2 |
1.4.7 chan_alsa : snd_pcm_open failed |
| 2:50PM |
2 |
Asterisk with iax2 over satellite |
| 11:14AM |
0 |
asterisk ncurses dependencies |
| 3:31AM |
1 |
TimeStamp a Recording |
| 2:17AM |
3 |
surge protector? |
| |
| Saturday July 14 2007 |
| Time | Replies | Subject |
| 8:51PM |
0 |
1.4 Crashes |
| 8:20PM |
3 |
tT in callparking |
| 7:56PM |
2 |
's' extension Asterisk 1.2.18 |
| 11:23AM |
4 |
Zaptel/mISDN and call transfer |
| 9:58AM |
2 |
gui for conferencing |
| 3:54AM |
2 |
open source screen pop software for asterisk |
| 2:27AM |
1 |
calling from ACT |
| 1:30AM |
1 |
Info about Providers |
| |
| Friday July 13 2007 |
| Time | Replies | Subject |
| 7:57PM |
1 |
Media Proxy Mode in Asterik: SIP and |
| 7:52PM |
0 |
b410p and DTMF: dtmfthreshold in 1.2.18 zaptel drivers please? |
| 7:15PM |
0 |
Selling a Digium TDM400P w/ 4 FXO cards |
| 4:52PM |
1 |
Can Asterisk hear on two IP addresses? And can I do routing for calls from private to public or public to private IP addresses |
| 4:19PM |
1 |
Media Proxy Mode in Asterik: SIP and H.323 |
| 4:16PM |
3 |
Macro: s-NOANSWER, _s-. |
| 2:06PM |
3 |
asterisk-addons compilation "error: dereferencing pointer to incomplete type" |
| 1:58PM |
2 |
limit simultaneous calls |
| 1:41PM |
0 |
no ringback from SIP server when originating call |
| 1:25PM |
1 |
Transfer Question |
| 1:20PM |
2 |
Distribution lists for voicemail |
| 12:33PM |
0 |
Problems with RNDIS |
| 10:06AM |
0 |
asterisk snmp |
| 8:24AM |
1 |
QUEUE_WAITING_COUNT |
| |
| Thursday July 12 2007 |
| Time | Replies | Subject |
| 10:12PM |
0 |
Queues monitoring software - OrderlyStats now FREE |
| 9:27PM |
0 |
AstLinux 0.4.6.1 Released |
| 9:08PM |
0 |
Outpulse with Asterisk |
| 8:58PM |
0 |
PRI and Local numbers |
| 5:50PM |
2 |
Different SIP From and Auth? |
| 5:01PM |
2 |
Trials with 1.4 |
| 4:40PM |
0 |
analog call progress - simplified I hope |
| 4:22PM |
1 |
Test |
| 2:25PM |
2 |
how to load phone registration information |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
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| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
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| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
No subject |
| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
0 |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
0 |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
0 |
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| 2:23PM |
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| 2:23PM |
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| 2:23PM |
0 |
No subject |
| 1:23PM |
0 |
Additional Wildcard TDM2400P Setup |
| 12:43PM |
1 |
exit ChanSpy with DTMF |
| 9:12AM |
1 |
Queues monitoring software |
| 7:55AM |
4 |
Lines Not being Hung UP Major |
| 3:35AM |
2 |
USB Modem with asterisk |
| |
| Wednesday July 11 2007 |
| Time | Replies | Subject |
| 10:25PM |
1 |
Asterisk as outbound proxy |
| 10:24PM |
1 |
Access specific port of Mediatrix 1204 from Asterisk |
| 10:08PM |
3 |
Codec Negotiation |
| 7:39PM |
1 |
Voicemail messages not deleting |
| 7:32PM |
2 |
Queue property |
| 6:59PM |
2 |
Pass Dialed number to a script |
| 5:57PM |
0 |
analog call progress - simplified ( I hope) |
| 3:04PM |
2 |
Call Waiting |
| 1:55PM |
1 |
MOH stop and resume when i hold |
| 1:30PM |
0 |
VoIP + IM unified client |
| 12:10PM |
2 |
iax2 peer become UNREACHABLE |
| 11:00AM |
0 |
how to force asterisk to read registration information from DB |
| 8:32AM |
1 |
Asterisk and Hardware Requirements |
| 7:51AM |
2 |
Music on hold stops on blind transfer |
| 4:37AM |
1 |
Looking for a USB phone handset or headset |
| 1:20AM |
1 |
extensions reload -- what impact? |
| |
| Tuesday July 10 2007 |
| Time | Replies | Subject |
| 10:11PM |
0 |
mixing video and non-video clients |
| 10:08PM |
2 |
video calls - Windows / Linux interoperability ? |
| 7:52PM |
0 |
Macro Goofiness |
| 7:48PM |
0 |
Odd AGI Issue - STREAM FILE, GET DATA not playing file |
| 6:15PM |
1 |
Asterisk 1.2.21.1 and 1.4.7.1 released |
| 5:05PM |
0 |
G722 and Polycom 550 |
| 4:17PM |
0 |
Asterisk, AudioCodes, Caller ID |
| 4:13PM |
1 |
PlayDTMF and Asterisk Manager |
| 2:24PM |
6 |
Edit ulaw file |
| 12:28PM |
2 |
video phones on 1.4.7 |
| 12:09PM |
0 |
VAD/CNG |
| 11:01AM |
0 |
sharing phone registration information between asterisk servers |
| 9:47AM |
2 |
DUNDI behind NAT? |
| 6:16AM |
1 |
asterisk-users Digest, Vol 36, Issue 25 |
| 5:42AM |
3 |
ZAP TDM and DTMF issue |
| 4:45AM |
1 |
Asterisk 1.4.7 and MOH |
| 4:25AM |
0 |
how to register several clients with different number but using single authentication account ? |
| |
| Monday July 9 2007 |
| Time | Replies | Subject |
| 11:40PM |
0 |
Meetme delay? |
| 10:45PM |
0 |
Asterisk 1.2.21, 1.4.7 and Libpri 1.2.5, 1.4.1 released |
| 8:35PM |
0 |
Digium cards for sale in Pakistan |
| 7:15PM |
3 |
Basic asterisk Autodialer? |
| 6:41PM |
0 |
allow third party registration/invitaion |
| 6:40PM |
1 |
Setting Appearance on Outbound Calls? |
| 3:21PM |
4 |
Problems sending more than 2 SMS with asterisk / smsq |
| 1:06PM |
2 |
Very bad TDMF tone ! |
| 10:39AM |
2 |
Background transfers with callback |
| 10:22AM |
10 |
Monitor events? |
| 8:28AM |
0 |
CTI application controling Asteridk |
| |
| Sunday July 8 2007 |
| Time | Replies | Subject |
| 11:43PM |
1 |
Asterisk and Mitel 3300 ICP |
| 11:41PM |
0 |
Sip trunk between Asterisk and Mitel 3300 ICP |
| 1:22PM |
3 |
Zapata, Junghanns Card and a leading 0 on inbound calls |
| 12:58PM |
1 |
Asterisk Help |
| 8:25AM |
1 |
Early Media Handling |
| 5:56AM |
2 |
asterisk is not sip proxy |
| 5:33AM |
2 |
Auto Fall Through when kicking users in MeetMe |
| 4:15AM |
0 |
SIT tone detection on Zap channel (PRI) |
| |
| Saturday July 7 2007 |
| Time | Replies | Subject |
| 10:50PM |
1 |
Call Waiting curiosity... |
| 8:07PM |
1 |
Polycom multiple registrations |
| 6:29PM |
2 |
Fax and Asterisk |
| 4:20PM |
0 |
Help doing one modification to libmfcr2.c of Unicall |
| 4:11PM |
9 |
Sip Providers |
| 4:02PM |
6 |
installing * from source |
| 3:04PM |
2 |
Corporate Feedback to OSS (was: Re: Mushtaq Ahmed is out of the office.) |
| 2:06PM |
2 |
Which features are lost when canreinvite is turned on ? |
| 8:55AM |
0 |
Asterisk TV is about to go live |
| 6:16AM |
2 |
Unable to install Asterisk Now Beta 6 |
| 12:15AM |
1 |
Channel name in queue log replaced by a manager event? |
| |
| Friday July 6 2007 |
| Time | Replies | Subject |
| 8:30PM |
0 |
LinkSys ATAs with GSM Support? |
| 6:14PM |
0 |
PRI CallerID and Apple iPhone |
| 4:21PM |
6 |
OT: Blackberry and Asterisk voicemail files. |
| 3:27PM |
0 |
Blind transfer from Queue in AGI script failuire |
| 2:32PM |
0 |
SIP peer unrechable when using an aliased interface |
| 11:58AM |
1 |
Asterisk Manager |
| 8:05AM |
3 |
Mushtaq Ahmed is out of the office. |
| 6:10AM |
2 |
Issue using zaptel's dynamic spans. |
| |
| Thursday July 5 2007 |
| Time | Replies | Subject |
| 10:20PM |
1 |
Missing TRANSFER event in queue log when using Local Channels |
| 10:13PM |
1 |
sounds |
| 7:32PM |
3 |
Call Queues |
| 7:13PM |
1 |
sometimes half audio on 7960 |
| 6:17PM |
2 |
Call Screening Not Working |
| 5:10PM |
0 |
IAX-Voicemail |
| 4:50PM |
1 |
IAX additional call-data |
| 4:12PM |
0 |
Visually impaired employees |
| 3:51PM |
1 |
Simple CDRs w/Asterisk/OpenSER. |
| 3:18PM |
2 |
REGEX expression for NXXNXXXXXX? |
| 3:16PM |
0 |
exits in NJ |
| 2:50PM |
0 |
connecting 1.2 and 1.4 using SIP |
| 1:16PM |
1 |
SIP / STUN / Network - Help!! |
| 12:37PM |
1 |
G729 on Solaris SPARC/x86/x64 Codec |
| 11:08AM |
2 |
sometimes calls drop during attended transfer |
| 9:42AM |
2 |
Asterisk E1 card support Q.SIG |
| 9:02AM |
1 |
AgentCallBackLogin vsAddQueueMember |
| 4:09AM |
2 |
Asterisk console filtering and logging |
| 12:04AM |
1 |
Need Help in Asterisk BLF/Presence/Hints |
| |
| Wednesday July 4 2007 |
| Time | Replies | Subject |
| 11:17PM |
0 |
FW: Openmoko ads now on youtube |
| 11:14PM |
2 |
Xorcom Bri and asterisk crashes |
| 10:56PM |
0 |
Problems with SIP Registration on VPN Link |
| 9:14PM |
0 |
ANNOUNCEMENT : A2Billing (Asterisk2Billing) - V1.3.0 STABLE (Yellowjacket) |
| 7:55PM |
2 |
reliaclear.com |
| 4:59PM |
2 |
Upgrade Asterisk |
| 4:38PM |
2 |
Call still in queue after Reject Signal |
| 3:31PM |
0 |
H263-2000 video format |
| 2:46PM |
1 |
Problems with misdn and ChanIsAvail |
| 12:43PM |
7 |
List delays |
| 10:32AM |
1 |
Dialout Macro and transfer call in progress |
| 9:26AM |
1 |
Asterisk TV will go live this Friday |
| 6:35AM |
0 |
asterisk hardware E1 pri card |
| 5:47AM |
0 |
single digit dial extension |
| 5:39AM |
1 |
call transfer not working |
| 5:30AM |
0 |
system recording problem using wav file |
| 4:58AM |
2 |
Need advice to get wcte11xp and wcfxo to load |
| 2:05AM |
1 |
Asterisk Support Question |
| 1:11AM |
1 |
QueueMemberStatus |
| |
| Tuesday July 3 2007 |
| Time | Replies | Subject |
| 10:25PM |
1 |
Distinctive ring detection not detecting ring cadences |
| 10:23PM |
1 |
Digit Convesion and Digit Insertion |
| 10:12PM |
2 |
Putting a password on the international call |
| 10:06PM |
5 |
Determining the used codec for the IP Trunk (SIP Trunk) |
| 10:00PM |
0 |
TDM800P cards with one way voice |
| 6:11PM |
1 |
res_config_mysql.c: MySQL RealTime: Failed to connect database server .. |
| 5:44PM |
1 |
Asterisk and Panasonic TDA200 |
| 5:29PM |
2 |
got 404 when route calls through Asterisk to another proxy |
| 4:14PM |
1 |
Session Border Controller time... |
| 2:44PM |
1 |
lookup a anonymous internal caller |
| 2:36PM |
1 |
help with internal extensions |
| 1:42PM |
3 |
garbled calls |
| 12:17PM |
4 |
Google acquires Grand Central |
| 11:43AM |
1 |
Configuring BLF or Asterisk presence/Hints feature |
| 11:20AM |
3 |
Suing Dell||Dull Computers for CID abuse |
| 7:00AM |
6 |
Need Advice/Suggestion |
| 6:51AM |
0 |
Configuring BLF or Asterisk presence feature |
| |
| Monday July 2 2007 |
| Time | Replies | Subject |
| 9:02PM |
1 |
DID providers in Toronto |
| 8:54PM |
2 |
Sip phones using the wrong context for an outbound call |
| 8:45PM |
0 |
trying to get vpb to compile |
| 7:27PM |
1 |
Asterisk 1.2 TDM24xx and B410P |
| 5:37PM |
1 |
Question about dnsmgr |
| 5:13PM |
0 |
FXO interface modules in Bombay, India |
| 2:45PM |
4 |
Help. Cannot compile version 1.4.6 with the following error |
| 12:34PM |
5 |
softphone with g729 codec |
| 10:45AM |
3 |
Call transfer in asterisk |
| 10:18AM |
3 |
Gigaset 450IP loses registration |
| 9:56AM |
1 |
"Random" all circuits busy now message |
| 7:40AM |
0 |
Authenticaion on incoming calls |
| 1:40AM |
0 |
Single ringer phone for incoming calls, that every one can answer |
| |
| Sunday July 1 2007 |
| Time | Replies | Subject |
| 11:47PM |
1 |
Asterisk strange behaviour |
| 10:39PM |
0 |
Rockwell ACD - "Take back and transfer" |
| 10:22PM |
1 |
Installing AJAM |
| 10:00PM |
0 |
ANNOUNCEMENT: Asterisk-Java 0.3 released |
| 7:00PM |
1 |
How can we block the calls for specific code |
| 6:09PM |
4 |
Not able to find the file zaptel.conf after compiling asterisk and zaptel |
| 5:52PM |
1 |
Creating a Voicemail ... |
| 5:00PM |
3 |
Suppress MusicOnHold in Queue |
| 3:14PM |
1 |
problems with dtmf using asterisk-1.4 rev r 6745 |
| 1:21PM |
0 |
Transfer outgoing call - macro |
| 12:58PM |
0 |
asterisk / MSN & DID |
| 10:04AM |
2 |
the-asterisk-book.com online (unstable version) |
| 9:37AM |
0 |
Solved: fail tp load modules |
| 1:11AM |
2 |
G729 , upgrade asterisk |