Thursday December 31 2009 |
Time | Replies | Subject |
8:12PM |
3 |
Daily Thousands of files in recording calls in Device mode |
6:53PM |
1 |
AudioCodes Caller ID |
3:46PM |
3 |
Dialplans & Holiday Dates |
2:11PM |
0 |
Friday Jan 1 Voip Users Conference |
9:47AM |
1 |
Asterisk recieves "11" when pressing "1" from SIPphone |
5:40AM |
0 |
Dahdi install issues |
5:10AM |
1 |
Inquiry:Asterisk different codec schemes? |
1:19AM |
2 |
Twilio |
|
Wednesday December 30 2009 |
Time | Replies | Subject |
9:10PM |
1 |
BLF status of sip trunk |
7:43PM |
2 |
Skype for Asterisk |
6:16PM |
0 |
Asterik with out registration. |
6:11PM |
1 |
Force Jitter Buffer for SIP to SIP calls |
4:52PM |
1 |
CDR_MYSQL 1.4 Database Structure |
3:12PM |
1 |
Monitoring SIP & Skype connections |
2:56PM |
4 |
Per user voicemail greeting |
2:56PM |
2 |
CID not working. |
1:04PM |
1 |
problem with ring being sent to caller |
12:36PM |
2 |
Inquiry:Asterisk Dictate? |
11:59AM |
2 |
multiple instances of asterisk on same machine |
11:29AM |
0 |
Inquiry:Asterisk festival? |
11:28AM |
1 |
Parked Call Ringback |
10:38AM |
0 |
wcte12xp0: Missed interrupt. when disable echocanceller |
6:06AM |
2 |
Auto-provisioining Polycom 430 wth dd-wrt router |
12:14AM |
0 |
Context Switches and Load Average spike - Asterisk Version 1.4.22 |
|
Tuesday December 29 2009 |
Time | Replies | Subject |
4:55PM |
0 |
asterisk billing transferred calls |
2:50PM |
0 |
asterisk 1.6.2.0 sip channel to sip channel call dtmf inband not work. |
12:55PM |
1 |
ReceiveFAX G.711 + Realtime |
12:06PM |
0 |
set box IP from which send sip traffic |
11:44AM |
1 |
T.38 and Linksys SPA8000 |
8:56AM |
0 |
CDR is "NO ANSWER" when it should be "ANSWERED" |
8:21AM |
0 |
codecs and volume |
8:18AM |
1 |
SkyHost is set to expire |
7:23AM |
1 |
CDR |
6:39AM |
1 |
error when open a2billing web page! |
6:07AM |
1 |
Does A2Billing has mial list? |
6:01AM |
1 |
identifying channel for softhangup |
5:37AM |
1 |
de-latinisation of the web - http://blog.collins.net.pr/2009/12/de-latinisation-of-web.html |
2:13AM |
1 |
Any good dialplan code out there to implement vertical service codes? |
1:58AM |
0 |
asterisk-users Digest, Vol 65, Issue 68 |
12:09AM |
2 |
Realtime mysql extensions mutiple queries for each priority? |
|
Monday December 28 2009 |
Time | Replies | Subject |
9:22PM |
3 |
cheap ip phone with auto-answer |
7:59PM |
1 |
Off Topic: Aastra BLF limit... |
6:11PM |
2 |
SIP Issue |
5:45PM |
2 |
Multiple Digium cards with one NFAS trunkgroup |
1:39PM |
0 |
Avaya 96xx handset with SIP 2.5, no name in display |
8:47AM |
2 |
Registering with a static peer? |
3:59AM |
1 |
AudioCodes MP-114 making calls via FXO |
|
Sunday December 27 2009 |
Time | Replies | Subject |
5:27PM |
0 |
Macro only accepts 1 argument |
5:04PM |
0 |
Parking function problem ? |
2:10PM |
2 |
rxgain / txgain for iaxmodem or hylafax |
11:51AM |
2 |
UA send 404 Not found |
10:42AM |
2 |
Call ends when picked up |
10:22AM |
0 |
Q; Recording when a bypass phone is used |
7:39AM |
3 |
Audiocodes MP-114 2FXO/2FXS help registering with Asterisk |
|
Saturday December 26 2009 |
Time | Replies | Subject |
1:36PM |
2 |
pattern matching |
|
Friday December 25 2009 |
Time | Replies | Subject |
10:31PM |
1 |
Newbie Looking For Login/Password |
8:31PM |
0 |
state_interface how to |
9:15AM |
0 |
How freeradius mapping the attributes when save records into radacct table? |
6:50AM |
2 |
compile issues. |
6:23AM |
2 |
SIP Incoming / Inbound not working for Broadvoice (Asterisk PBX 1.6.1.6) |
|
Thursday December 24 2009 |
Time | Replies | Subject |
4:24PM |
2 |
Recording the Calls to a USB Drive |
1:44PM |
0 |
Asterisk Manager API - DTMF issues |
9:49AM |
0 |
X100P clone card problem |
9:30AM |
1 |
Tel uri Support |
7:46AM |
0 |
redial script... |
7:18AM |
2 |
1.6 Troubleshooting help |
6:43AM |
3 |
Asterisk with gdb |
4:52AM |
1 |
How to create MeetME room with dialplan? |
2:50AM |
0 |
Extension.conf |
2:42AM |
0 |
NICE analog recorders |
2:19AM |
0 |
Failed to record Radius CDR record! |
12:24AM |
0 |
MixMonitor stops audio in SIP to SIP call |
|
Wednesday December 23 2009 |
Time | Replies | Subject |
11:28PM |
1 |
How to send variables through AMI originate and read those variables in context? |
11:19PM |
1 |
AMI originate and PHP |
9:53PM |
0 |
Happy Holidays from OpSys Consulting Group |
9:21PM |
4 |
Dell Server suggestion |
7:08PM |
2 |
how to check Asterisk SIP registration |
6:52PM |
2 |
Core show function? |
5:26PM |
0 |
Can't place 2nd call to provider |
4:49PM |
4 |
fax problem |
3:00PM |
1 |
How to exchange/get $variables from/to each channel on cmd Dial |
2:32PM |
4 |
Asterisk and Faxing |
2:04PM |
0 |
Issue calling a TFN |
10:20AM |
1 |
Problems with chan_sip |
10:09AM |
1 |
Can't load cdr_radius.so module? |
9:16AM |
1 |
Can't do make menuselect? |
7:53AM |
1 |
SIP realm |
5:17AM |
1 |
Session Refresh or Codec change |
|
Tuesday December 22 2009 |
Time | Replies | Subject |
5:37PM |
0 |
Account Code Inbound |
3:01PM |
1 |
call queue with external numbers?? |
2:22PM |
0 |
Asterisk Release Time Frames |
2:08PM |
2 |
E1 R2 Congestion Status |
2:05PM |
4 |
Showing "name of extension" when calling |
1:47PM |
1 |
AsteriskNow and language |
11:11AM |
0 |
Available Agent on Queue |
10:57AM |
1 |
Making a data connection with Asterisk |
9:26AM |
4 |
asterisk & x-lite |
8:20AM |
0 |
Asterisk news :: Next release of Asterisk will be 1.8 Long Term Support |
6:41AM |
1 |
Every one Busy Problem |
|
Monday December 21 2009 |
Time | Replies | Subject |
11:06PM |
3 |
TDM 400 hardware(?) issue |
10:39PM |
1 |
Asterisk 1.2.14 - Play an audio or signal |
8:21PM |
0 |
Asterisk chan_skype GUI released |
6:13PM |
0 |
res_cepstral for 1.6.2 |
6:04PM |
3 |
Looking for some example dialplans |
5:35PM |
0 |
possible to list a conference in the database? |
1:41PM |
2 |
Asterisk depend on postgresql files? |
1:04PM |
3 |
script |
12:48PM |
1 |
Yealink vs Aastra |
12:47PM |
1 |
Inquiry:Installing Asterisk 1.4 on CentOS 5.2? |
11:20AM |
1 |
sip show peers returns several notices |
11:19AM |
1 |
anonymous calls code |
11:00AM |
1 |
Incoming calls coming into default context |
8:22AM |
1 |
FreeRadius or A2Billing , which is the better choice of Asterisk for billing? |
6:39AM |
1 |
New 1.4 system: registered, but not responding to invite? |
5:24AM |
1 |
Asterisk Heartbeat Monitor for Fail safe. |
|
Sunday December 20 2009 |
Time | Replies | Subject |
11:04PM |
1 |
What changed in Directed PickUp between 1.6.1 and 1.6.2 ? |
2:50PM |
2 |
Live CD - do you think they are worth doing? |
11:14AM |
0 |
Dec 20 Global All Free SW HW Culture meeting - BerkeleyTIP |
10:15AM |
2 |
Rewrite of calling number for all extensions |
8:10AM |
1 |
Manager command that equal to database show CFIM |
|
Saturday December 19 2009 |
Time | Replies | Subject |
4:20PM |
5 |
sendmail |
2:29PM |
1 |
PAP2 Dialing Delay |
12:05PM |
0 |
E1 ingress to SIP egress problem with 183 response |
11:51AM |
4 |
Inquiry:Connect my Asterisk to external sip? |
10:57AM |
1 |
Nortel BCM - Call Accounting Interface? |
|
Friday December 18 2009 |
Time | Replies | Subject |
11:14PM |
2 |
Asterisk 1.6.2.0 Now Available! |
10:59PM |
1 |
Asterisk 1.6.1.12 Now Available |
10:54PM |
0 |
Asterisk 1.6.0.20 Now Available |
10:49PM |
0 |
Asterisk 1.4.28 Now Available |
6:28PM |
3 |
Call Waiting With Draytek ATA |
5:05PM |
1 |
HOW to record saynumber output |
2:21PM |
0 |
SAP-BCM Sip trunking |
1:53PM |
1 |
Could Asterisk be crashing under high context switches? |
1:24PM |
0 |
IAX/NEW delays |
11:25AM |
0 |
test request for new event "Pickup" when a call is picked up from an other phone |
10:49AM |
0 |
DTMF doubler when using READ() |
10:25AM |
1 |
wrapuptime? |
7:18AM |
0 |
Friday @12 Noon ET: Kamailio, Open SER and Asterisk |
3:36AM |
2 |
FAX for Asterisk |
1:56AM |
0 |
calls ending up in default context |
1:29AM |
2 |
To Asterisk AMI Gurus - Tacking issue with originate |
|
Thursday December 17 2009 |
Time | Replies | Subject |
11:23PM |
6 |
Feature Request: GotoIfTimeWithOffset |
5:47PM |
2 |
Integrate a CPE with Asterisk in MGCP |
9:30AM |
0 |
Setting the Request URI In registration |
8:53AM |
1 |
iax no way sound |
8:39AM |
1 |
Asterisk IPv6 update - we need an update |
4:16AM |
1 |
SIP to Analog Devices |
|
Wednesday December 16 2009 |
Time | Replies | Subject |
8:02PM |
0 |
Help forcing crosstalk |
7:24PM |
1 |
sip show channels display |
5:21PM |
0 |
Federal, State, and Local government installations of Asterisk |
4:08PM |
1 |
Mixing commercial/SVN Asterisk |
1:27PM |
1 |
FW: question on how to connect 2 boxes |
1:17PM |
0 |
Replacing AgentCallBackLogin for Asterisk 1.6 |
1:05PM |
0 |
Shorr/Call quality issues |
8:47AM |
1 |
announce prompt to user |
7:26AM |
3 |
About Asterisk Manager (C# Sharp) |
3:43AM |
0 |
asterisk-users Digest, Vol 65, Issue 38 |
|
Tuesday December 15 2009 |
Time | Replies | Subject |
6:41PM |
2 |
monitor-type=MixMonitor |
6:15PM |
2 |
Can't get G.729 to work... |
3:09PM |
2 |
member (In use) |
2:07PM |
0 |
OT - SPA3102 - Provisioning with config file [SOLVED] |
12:37PM |
1 |
dahdi-channels.conf -v- chan_dahdi.conf |
12:07PM |
1 |
cdr question |
10:42AM |
1 |
OT - SPA3102 - Provisioning with config file |
9:44AM |
0 |
simple sip question (I think) |
8:22AM |
0 |
how apps/enter.h |
6:27AM |
0 |
digest authentication method and the realm domain |
1:16AM |
3 |
Best way ro run 2 or more asterisk servers? |
|
Monday December 14 2009 |
Time | Replies | Subject |
11:59PM |
1 |
Queue still tries to ring agent when busy |
10:43PM |
1 |
3 ed party sip client for Nokia sy |
9:02PM |
0 |
What version of libpri and zaptel work best with 1.4.24 |
8:44PM |
0 |
question on how to connect 2 boxes |
8:06PM |
0 |
USB ISDN30 |
8:02PM |
0 |
pickupexten on chan_dahdi |
7:34PM |
0 |
iaxmodem Disconnect time and ISDN Service |
7:02PM |
3 |
Asterisk throws error using the alsa, module |
6:21PM |
1 |
Rewrite calling number of incoming call |
6:19PM |
3 |
hints through a Local channel |
5:59PM |
3 |
Question regarding digital card TE412p |
4:52PM |
2 |
ISDN: Inband DTMF doesn't trigger transfer feature |
4:28PM |
3 |
Is this bad hardware? Dahdi-v-X100 clone |
3:53PM |
2 |
Asterisk & Zaptel setup on vserver |
1:41PM |
1 |
meetme with review of the entered conference number |
1:32PM |
1 |
Asterisk ZAP/DAHDI reads phantom digit on overlap PRI |
11:52AM |
0 |
ISDN Remote HOLD |
10:11AM |
1 |
Call on hold through DTMF |
8:35AM |
1 |
RTCP SR transmission |
2:10AM |
1 |
AGI with PHP |
12:20AM |
2 |
question on queues |
|
Sunday December 13 2009 |
Time | Replies | Subject |
7:52PM |
1 |
Random DTMF tones generated from speech |
1:25PM |
1 |
Dial with timeout don't end call |
10:15AM |
3 |
iphone client app |
9:15AM |
0 |
Avaya 9650 SIP phone and dial timeout |
8:01AM |
1 |
Asterisk Queue Dialplan |
2:50AM |
1 |
Unable to open file... |
|
Saturday December 12 2009 |
Time | Replies | Subject |
10:17PM |
2 |
Auto Attendant / Receptionist system |
5:49PM |
3 |
Random DTMF tones generated from speech in conversations |
4:56PM |
1 |
how to randomly use provider? |
11:25AM |
3 |
DEVICE_STATE |
9:44AM |
0 |
Inquiry:Asterisk sip server? |
12:12AM |
0 |
T38 Passthrough 1.6.1.12-rc1 Good Results |
12:03AM |
1 |
Playing a message if my call lands in their voicemail |
|
Friday December 11 2009 |
Time | Replies | Subject |
10:03PM |
1 |
Terminate T.38 to PSTN |
6:06PM |
4 |
G729 Pass through |
5:37PM |
3 |
ATA FXO |
5:25PM |
1 |
chan_dahdi.conf for TDM404E |
4:18PM |
1 |
question on register |
4:08PM |
1 |
Free Fax for Asterisk |
3:53PM |
0 |
zttool don't show NT mode with OctoBRI |
3:17PM |
1 |
Asterisk Unregisteres IAX Friend Randomly |
12:26PM |
0 |
VUC Dec 11 @ 12 Noon EST: g729 transcoding, software & hardware |
11:19AM |
3 |
Calls Dropping |
9:25AM |
1 |
ANNOUNCE: New version of Activa TAPI driver |
8:33AM |
0 |
How to get LEG B channel info? |
8:02AM |
2 |
sip realtime question |
7:00AM |
1 |
max. no. of conferences supported |
5:57AM |
0 |
How can Write simple AGI in JAVA |
|
Thursday December 10 2009 |
Time | Replies | Subject |
8:21PM |
0 |
Splash ring on PAP2t |
7:56PM |
0 |
need help to setup a sip trunk between a Nortel CS1000 and asterisk |
6:18PM |
1 |
interdigit timeout chan_dahdi |
4:04PM |
0 |
Encrypted GSM |
3:15PM |
1 |
Sangoma card reports HDLC errors |
1:47PM |
0 |
Avaya 950 one-X Deskphone |
1:19PM |
1 |
Asterisk 1.6.1.11 Fax |
12:57PM |
2 |
Hangs up after 16 minutes on a call. |
7:53AM |
3 |
switchvox 305 Appliance |
6:00AM |
0 |
Asterisk Meetme on XEN virtual machine recording is 2x times faster than normal |
3:15AM |
1 |
Realtime Database Tables |
2:39AM |
2 |
asterisk-users@lists.digium.com Hello! |
12:49AM |
1 |
Asterisk as a PSTN simulator |
|
Wednesday December 9 2009 |
Time | Replies | Subject |
9:08PM |
5 |
Can't restart asterisk from script |
6:29PM |
0 |
FreePBX IP Phone Recommendations |
5:53PM |
4 |
Need help/suggestions for DialPlan |
4:48PM |
0 |
AEX800P on HP Prolaint ML115 - Error on Module Load - |
2:35PM |
1 |
Problem with Asterisk and SPA-3000 |
11:34AM |
1 |
Configure DAHDI with TDM410 for analog |
10:24AM |
1 |
SkypeForAsterisk |
8:11AM |
1 |
How to backup Trixbox 2.8.0.3 |
8:11AM |
0 |
SIP_CODEC related question |
7:54AM |
1 |
app_voicemail. Help me to find typo source ... |
7:04AM |
1 |
Recording from billsec |
2:54AM |
2 |
Interesting problem with IP's |
12:06AM |
4 |
dahdi restart kills server |
|
Tuesday December 8 2009 |
Time | Replies | Subject |
9:45PM |
5 |
Easy way to see what dahdi channels are being used |
9:12PM |
1 |
meetme.conf adminpin - what does it do? |
7:23PM |
1 |
network config |
6:43PM |
2 |
Sangoma A101DE with Dell PE 2850 |
5:47PM |
2 |
E1 Channel Numbering - Your Comments. |
5:25PM |
2 |
Asterisk throws error using the alsa module |
5:14PM |
0 |
Directory application: First DTMF digit is missed if pressed during "using your touch tone keypad..." announcement |
3:53PM |
0 |
might have found and issue |
3:44PM |
1 |
Voicemail issues |
2:51PM |
1 |
Asterisk Voicemail |
1:15PM |
2 |
Starting and installing Dahdi (2.2.0)? |
7:03AM |
1 |
G729 with IAX |
12:58AM |
1 |
Configure DAHDI with TDM410 for analog Modem calls |
|
Monday December 7 2009 |
Time | Replies | Subject |
10:14PM |
0 |
SPA921 Help |
8:34PM |
2 |
realm authentication |
8:30PM |
1 |
Automon -> Voicemail |
7:30PM |
1 |
REFER to trunk |
6:05PM |
1 |
automon => *1 "one touch recording" |
2:58PM |
0 |
Why no re-register when sip register status is UNREACHABLE |
2:06PM |
1 |
g722 question |
8:00AM |
3 |
show queue's name and other info in incoming call to queue member |
5:04AM |
1 |
Error : SIP/2.0 401 Unauthorized |
|
Sunday December 6 2009 |
Time | Replies | Subject |
9:52PM |
0 |
realm |
7:11PM |
1 |
Linksys SPA9x2 echo problem |
4:22PM |
1 |
ABCTI: first usable beta |
3:38PM |
1 |
sequential dialing preferences |
1:49PM |
3 |
Call Limits |
8:56AM |
1 |
Example to handle incoming calls without callerid at home? |
7:40AM |
0 |
Sangoma U100 |
7:24AM |
1 |
Can Asterik act as a SIP Proxy |
1:25AM |
1 |
Asterisk to Email |
|
Saturday December 5 2009 |
Time | Replies | Subject |
10:38PM |
2 |
Setting up skype |
9:02AM |
2 |
G729: TC400B vs Software Encoder |
4:57AM |
2 |
How to use SIP hints and BLF for realtime extensions on Aastra phones? |
|
Friday December 4 2009 |
Time | Replies | Subject |
11:39PM |
0 |
The SIP in the Mobile Phones are not able to register on asterisk |
7:36PM |
0 |
DAHDI - Split data voice use |
7:01PM |
2 |
DAHDI outgoing |
6:49PM |
0 |
No audio - using g729 codec altogether |
4:43PM |
1 |
DAHDI issues on 1.4.26.1 |
4:41PM |
3 |
MWI count wrong when using IMAP and VM |
4:37PM |
1 |
IAX2 Port issue |
4:27PM |
0 |
Today in 30 minutes: VoIP on Social Networks |
2:15PM |
0 |
Get back in dialplan with number-parsing |
1:42PM |
1 |
Dahdi_genconf does not generate NT/TE configuration |
10:54AM |
1 |
spandsp version |
8:41AM |
1 |
Get Queue values from dialplan (Was: queue_variables() function) |
6:32AM |
2 |
hey please help me my 3rd email of how to change From fileld username in sip packet |
5:56AM |
0 |
chan_sip Error |
12:22AM |
2 |
Multiple Channel Variables with AMI Originate |
|
Thursday December 3 2009 |
Time | Replies | Subject |
11:20PM |
0 |
queue_variables() function |
10:51PM |
1 |
only the first ResetCDR works after upgrade to 1.6 |
7:48PM |
2 |
Source-IP on Asterisk DRBD/-HA-Cluster wrong |
7:45PM |
0 |
Repost: Working in useful examples... and freenum/e.164 dialing in extensions.conf.example |
7:15PM |
1 |
multiple sip trunks |
1:49PM |
2 |
Wi-Fi sip phones with auto provisioning |
1:01PM |
0 |
ChanSpy gets stuck |
11:15AM |
0 |
AEL, 1.6, CUT and commas [SOLVED] |
11:09AM |
0 |
AEL, 1.6, CUT and commas |
9:03AM |
3 |
Fax throughput - Asterisk 1.6.1.9 |
8:21AM |
0 |
softphone @handheld |
7:57AM |
1 |
Dial application with M option |
7:27AM |
0 |
sip show channels shows non-existent channels on 1.6.0.19 and 1.4.27.1 ? |
6:39AM |
1 |
Feature Request: "SayNumberFiles" |
5:28AM |
1 |
Flashing Cisco 7941 to SIP |
12:13AM |
2 |
dahdi_tool shows no alarms, but no line connected |
|
Wednesday December 2 2009 |
Time | Replies | Subject |
10:10PM |
0 |
AsteriskNOW sip module not installed |
10:02PM |
6 |
Rsrvd state and off hook dahdi issue |
9:10PM |
0 |
FW: Variable Name needed |
8:05PM |
2 |
Variable Name needed |
6:25PM |
0 |
Asterisk-Addons 1.4.10, 1.6.0.4, and 1.6.1.2 Now Available |
5:59PM |
0 |
dialout problem with analog phone |
5:32PM |
2 |
Help configuring Audiocodes MP-104 FXO |
4:43PM |
0 |
Asterisk Queues Tutorial updated - Hot-Desking without Agent Channels |
2:58PM |
0 |
fsk callerid with DTAS start(like dtmf in issue 9096)? |
2:32PM |
3 |
b option in Directory |
11:47AM |
1 |
Problem with Timeout |
1:20AM |
2 |
Featuremap help |
|
Tuesday December 1 2009 |
Time | Replies | Subject |
11:23PM |
6 |
Question about g729 |
11:06PM |
2 |
Slightly OT - Oreka Call Recording |
9:55PM |
0 |
Codecs negotiation |
8:56PM |
2 |
OpenSBC |
8:49PM |
0 |
SafiServer and SafiWorkshop 1.2 With Web Services Released |
3:32PM |
2 |
Asterisk registers with private IP |
1:09PM |
1 |
"Dropping incompatible voice frame" error |
12:58PM |
0 |
Asterisk - Segmentation fault |
11:00AM |
2 |
Issue with T38 fax Calls |
8:40AM |
2 |
Patch for app_dial.c: exit when just one ext is busy. |
8:21AM |
0 |
Asterisk radius configuration |
7:16AM |
1 |
Asterisk Configuration with Sphinx speech engine |
12:46AM |
1 |
AGI |