Sunday October 31 2010 |
Time | Replies | Subject |
11:49PM |
0 |
setting standard with asterisk |
11:45PM |
0 |
Junghanns douBRI miniPCI (b559) DAHDI drivers |
10:19PM |
1 |
billsec=0 when using Local channel |
|
Saturday October 30 2010 |
Time | Replies | Subject |
10:22PM |
2 |
Exceptionally long queue length queuing . . . . |
10:10PM |
1 |
Tormenta 3 (Tor3e) - Driver. |
6:28PM |
8 |
Under heavy attack |
2:32PM |
0 |
Gtalk and asterisk 1.6 |
|
Friday October 29 2010 |
Time | Replies | Subject |
9:55PM |
1 |
Updating asteriskcdrdb with uniqueid field from Master.csv, Master.csv.1....Master.csv.5 - Must I bring all files together first? |
9:14PM |
2 |
Video based Asterisk Training |
4:46PM |
1 |
Asterisk 1.8 and character sets and AMI |
2:57PM |
0 |
New tutorial: Compiling Asterisk 1.8 on CentOS 64 |
11:01AM |
0 |
Asterisk 1.6 Overlap dialling timeout? |
10:37AM |
1 |
trixbox - sip trunk with voipwise |
9:08AM |
1 |
BLF in Asterisk 1.4.* |
8:21AM |
2 |
MGCP |
7:25AM |
0 |
asterisk 1.6 and Firefox 4 Beta |
|
Thursday October 28 2010 |
Time | Replies | Subject |
5:44PM |
1 |
generic_odbc and ltdl are not available to enable ODBC support |
3:28PM |
0 |
Adhearsion 1.0 - Now Showing |
3:08PM |
8 |
SIP Load Balancing |
2:43PM |
0 |
SIP Communicator Friday at 12 Noon EDT |
7:41AM |
5 |
being bombarded with SIP packets |
7:38AM |
3 |
SIP client floods port 5060 and gets blocked |
4:37AM |
0 |
ss7_channel or ss7lib |
1:21AM |
1 |
what interface for ISDN-10/20/30? |
12:48AM |
0 |
[asterisk-biz] D-Link Wifi Phones |
12:41AM |
0 |
Intermittent failure when placing calls - unable to create channel of type SIP |
|
Wednesday October 27 2010 |
Time | Replies | Subject |
10:37PM |
1 |
Astribank Configuration Issues |
7:43PM |
1 |
Extension notation in default ViciDial installation |
2:29PM |
0 |
Send INVITES and REFERs from OpenSIPS to Asterisk with multiple Contexts |
10:59AM |
1 |
Asterisk died without any message, segfault |
10:39AM |
1 |
phoneprov |
9:11AM |
1 |
Fax Degium channel License |
6:58AM |
0 |
Asterisk Strange Problem while call received from customer On PRI. |
5:59AM |
0 |
Test numbers Worldwide |
|
Tuesday October 26 2010 |
Time | Replies | Subject |
7:41PM |
2 |
OT: SMS inbound |
6:25PM |
2 |
No media being sent in SIP call |
6:00PM |
1 |
need to be able to pass a call to the pstn from another pbx trunk |
5:57PM |
2 |
Trim the RDNIS |
3:31PM |
11 |
Auto provisioning from public server |
12:57PM |
3 |
Channel Bank ? Simple Switch Hangup? |
11:38AM |
1 |
2 HB8 cards in one server - first one is not recognized, the second is |
10:59AM |
0 |
IAX2 call dropped when a second call comes in |
2:41AM |
5 |
Mobile Phones and Asterisk |
|
Monday October 25 2010 |
Time | Replies | Subject |
8:21PM |
3 |
Extension Exists |
6:14PM |
2 |
Pop-up for MS Outlook 2007 recommended |
3:52PM |
2 |
Re : thousands Hangup per second /saturation of bandwidth |
3:51PM |
1 |
Re : saturation of bandwidth because of HANGUP |
3:43PM |
1 |
particular sip registry and outbound proxy |
12:59PM |
0 |
CDR updating |
6:03AM |
4 |
google voice + asterisk: calls made to GV# processed but weird |
5:02AM |
0 |
xpp_fxloader fails to load Astribank firmware on Ubuntu Lucid |
12:28AM |
1 |
E1 configuration |
|
Sunday October 24 2010 |
Time | Replies | Subject |
10:23PM |
5 |
Integrating Asterisk 1.8 with Google Talk and Google Voice |
9:55PM |
2 |
Chan variables for peer |
9:47PM |
0 |
baffled by defaultuser on aastra 9133i |
5:02PM |
1 |
How to have failover sip interface? |
4:46PM |
0 |
Can't hear MOH from PSTN [SOLVED] |
4:44PM |
0 |
Default MOH not working on 1.6.1 [SOLVED] |
3:52PM |
1 |
ISDN & SS7 |
3:35PM |
1 |
Can't hear MOH from PSTN |
1:42PM |
1 |
Cepstral voice quality |
11:18AM |
0 |
Does any one uses PortSIP VoIP SDK? |
10:07AM |
0 |
[OT] Friday funny |
|
Saturday October 23 2010 |
Time | Replies | Subject |
11:07PM |
2 |
Just Take dCAP at Astricon? |
9:26PM |
3 |
Cepstral voice quality not good |
7:33PM |
0 |
NAT issues |
7:14PM |
2 |
1.8 Console Welcome Message |
5:56PM |
0 |
Parinya Sirisang invited you to Dropbox |
4:31PM |
3 |
Why such high latency on internal lan? |
4:27PM |
1 |
SipSak: Send SIP OPTION with password |
2:36PM |
4 |
Asterisk 1.8 IAX Registration |
12:43PM |
1 |
How to properly re-configure dahdi |
12:35PM |
1 |
Problem |
12:31PM |
5 |
a2billing muting "enter the phone number" |
12:20PM |
0 |
hangup delayed very much on fastagi appliaction of asterisk 1.6 |
12:07PM |
2 |
B410P - BRI NT 100 Ohm terminator |
9:43AM |
1 |
RealTime Voicemail |
9:35AM |
7 |
Dial plan help |
12:53AM |
2 |
killing asterisk 1.8 |
|
Friday October 22 2010 |
Time | Replies | Subject |
9:05PM |
1 |
E1 and T1 on the same card, or on the same server |
9:00PM |
0 |
CEL ODBC problem in 1.8.0 |
6:43PM |
0 |
E1 and Pt on the same card, on in the same asterisk box |
5:28PM |
0 |
488 Not acceptable here |
2:02PM |
2 |
OpenVPN over TCP 1194 rather than UDP 1194 - Is there an adverse effect when running Asterisk? |
1:48PM |
0 |
Best practices to edit multi-versions config files ? |
1:02PM |
1 |
SIP Channel naming conventions |
11:10AM |
5 |
dials a trunk when off hook |
9:44AM |
3 |
Licensing of Default MOH |
3:31AM |
0 |
Counterpath Presence Patent and Android VoIP app |
12:16AM |
1 |
MS-SQL / Freetds -- func_odbc |
|
Thursday October 21 2010 |
Time | Replies | Subject |
7:59PM |
2 |
Incoming calls |
6:03PM |
1 |
Why high latency on internal lan? |
5:41PM |
0 |
Asterisk 1.8.0 Now Available! |
5:30PM |
1 |
Hardware Compatibility HP Proliant - Sangoma PCI Express |
3:55PM |
0 |
saturation of bandwidth because of HANGUP |
3:41PM |
5 |
SIP Blacklisting |
3:35PM |
2 |
1 way audio asterisk 1.6 |
2:56PM |
1 |
Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician |
2:40PM |
10 |
Asterisk 1.80-rc5 |
2:27PM |
1 |
Asterisk 1.8.0-rc5: Blind transfer failed, SIP REFER Method |
1:31PM |
1 |
asterisk 1.8 SIP register uri: peer field ? |
12:41PM |
2 |
dialing from asterisk console? |
12:31PM |
1 |
Busy detection in dialplan - Asterisk 1.6 |
9:16AM |
8 |
Dial Plan Conf |
7:59AM |
2 |
DIALSTATUS always returns NOANSWER |
5:24AM |
3 |
Asterisk Realtime Billing Question??? |
12:49AM |
1 |
How to kill AMI ORIGINATE on-the-fly |
|
Wednesday October 20 2010 |
Time | Replies | Subject |
10:41PM |
4 |
Email from Dialplan |
9:56PM |
2 |
Adaptive CDR and default fields |
8:02PM |
5 |
Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk? |
4:58PM |
1 |
SIP 401 |
4:04PM |
1 |
2 step dialing |
2:38PM |
0 |
Audio Playback randomly stops |
1:52PM |
4 |
Recommendation for a new server |
1:51PM |
3 |
Using Calls Rejection Reasons |
1:33PM |
2 |
DAHDI weather quirk |
11:05AM |
2 |
Playback in the middle of a call though AMI |
8:20AM |
1 |
echo on TE122 |
7:30AM |
1 |
Best way to recording the hold time for a Queue agent or an extension |
7:22AM |
2 |
Is Asterix right tool for me? |
6:02AM |
5 |
Queue member status - BUSY |
3:03AM |
3 |
dahdi_genconf |
1:14AM |
1 |
Parked calls drop asterisk-1.4.22.1 |
|
Tuesday October 19 2010 |
Time | Replies | Subject |
10:50PM |
0 |
Distortion and block on analog lines |
6:40PM |
1 |
E1 channels real time monitoring |
2:46PM |
1 |
dahdi vmware query |
2:36PM |
1 |
FFA SendFax rejects T.38 reINVITE (488 Not acceptable here) |
|
Monday October 18 2010 |
Time | Replies | Subject |
11:54PM |
2 |
Asterisk 1.8.0 Release Candidate 5 Now Available |
7:40PM |
5 |
Same extension registering over eth0 and eth1 |
7:36PM |
1 |
Recording |
6:54PM |
2 |
CEL Documentation |
6:35PM |
1 |
a2billing |
6:03PM |
0 |
Asterisk 1.8.0 Release Candidate 4 Now Available |
4:43PM |
0 |
How to check if Agent is logged into a specific Queue using dial-plan? |
3:42PM |
0 |
Problems detecting hangup |
11:09AM |
8 |
Asterisk to switch on electric heaters remotely? |
10:59AM |
15 |
SIP DNS SRV |
8:57AM |
0 |
How to execute Asterisk Functions in PHPAGI |
1:23AM |
0 |
app_swift for Asterisk 1.8 |
1:13AM |
5 |
IAX2 works one direction, but not the other... |
|
Sunday October 17 2010 |
Time | Replies | Subject |
9:18PM |
4 |
Meetme |
5:18PM |
2 |
Error with Connecting Two Asterisk BOX with IAX |
1:03PM |
0 |
Good Day! 2010.10.17.21.4.4 |
|
Saturday October 16 2010 |
Time | Replies | Subject |
9:36PM |
6 |
Remote Unix Connection |
9:35PM |
1 |
(no subject) |
8:59PM |
3 |
Detect incoming fax on PSTN and route to fax machine on DADHI extension? |
3:28PM |
1 |
fraud advice (Also advice on using ipbanning) |
3:06PM |
1 |
DAHDI, PRI and callerid |
10:42AM |
4 |
How to find ".gsm" audio file length or duration |
|
Friday October 15 2010 |
Time | Replies | Subject |
4:22PM |
3 |
SIP - no audio behind nat problem |
3:22PM |
0 |
how to insert dynamic hostname into shared CDR database |
2:53PM |
4 |
Audio problems on cable modem link |
1:59PM |
8 |
drop dead fix |
1:55PM |
2 |
Echo on PRI after upgrading to Asterisk 1.6.2 from 1.2 |
1:38PM |
1 |
app_meetme build option is XXX'ed out |
10:17AM |
1 |
warning diego viola the trouble maker for the world |
9:00AM |
2 |
Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4) |
3:13AM |
1 |
Queue Agent Getting Additional Calls When on the Phone |
1:10AM |
8 |
fraud advice |
|
Thursday October 14 2010 |
Time | Replies | Subject |
7:25PM |
6 |
Audiocodes firmware |
4:18PM |
5 |
Routers that do not show external IPs... |
4:12PM |
0 |
AstriCon update - less than two weeks! |
4:00PM |
1 |
Default MOH not working on 1.6.1 |
3:43PM |
1 |
Explain "core show translation" |
3:01PM |
2 |
clustering |
2:17PM |
1 |
Passing variables into macros? |
1:35PM |
5 |
How to connect asterisk PBX to PSTN |
9:29AM |
1 |
Using hint priority with LDAP extensions and users |
2:46AM |
1 |
advice re: Page() application |
1:31AM |
1 |
MySQL and Channel Event Logging |
|
Wednesday October 13 2010 |
Time | Replies | Subject |
11:54PM |
1 |
advice re: Page() application |
10:48PM |
1 |
Some give 603 Declined |
7:40PM |
3 |
call forwarding callerID |
5:52PM |
0 |
Unable to specify channel 5: No such device or address |
5:50PM |
1 |
SIP disconnects after 20 seconds behind NAT |
5:11PM |
4 |
checking CDR |
3:26PM |
2 |
Configuring & Setting up Asterisk |
2:56PM |
0 |
Getting last 2 Sip registrations of same user |
2:46PM |
0 |
innomedia ATA's |
2:43PM |
3 |
GXP-21XX |
1:06PM |
11 |
DMTF Mode |
9:21AM |
1 |
realtime users call problem |
8:10AM |
0 |
Asterisk Hangup Issue in Ringing State with Incoming call |
|
Tuesday October 12 2010 |
Time | Replies | Subject |
8:02PM |
1 |
sound file debug |
2:14PM |
0 |
REFER method |
2:10PM |
1 |
how to fake asterisk register? |
1:51PM |
1 |
src_mysql problem |
12:12PM |
0 |
Application Map Not Working |
12:08PM |
1 |
chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049 |
11:53AM |
0 |
rtpip patch |
|
Monday October 11 2010 |
Time | Replies | Subject |
11:37PM |
4 |
SIP and ANI |
10:21PM |
1 |
MWI Assistance |
10:14PM |
2 |
user number in conference |
8:20PM |
2 |
Second time Parking issue |
8:02PM |
0 |
don't leave meetme conf if key pressed |
2:48PM |
8 |
Create channel bank with TDMoE |
1:01PM |
0 |
Synchron Playback to caller AND callee ? |
12:36PM |
1 |
iax2 users calls limit for outgoing / incoming |
12:33PM |
1 |
Call Failed Audio |
9:48AM |
1 |
About Action Originate |
8:22AM |
1 |
Quintum Tenor AX and Echo |
4:08AM |
1 |
OpenR2 |
12:57AM |
1 |
Unable to find a codec translation path from ulaw|h261 to slin |
|
Sunday October 10 2010 |
Time | Replies | Subject |
8:55PM |
1 |
TDM 400p and Noise on the line |
1:46PM |
1 |
Modifying cid.cid_name in app_parkandannounce.c |
4:54AM |
1 |
Dahdi missing |
|
Friday October 8 2010 |
Time | Replies | Subject |
11:25PM |
0 |
SIP NOTIFY to make linksys/cisco SPA BLF go yellow |
10:28PM |
3 |
looking for a better ATA |
3:06PM |
0 |
Weird stalling of playback on IAX2 channels on 1.8svn |
2:33PM |
2 |
Weird stalling of playback on IAX2 channels on 1.8 svn |
2:10PM |
3 |
How to use Atxfer in AMI |
9:12AM |
1 |
Voice quality assessment in Asterisk |
6:37AM |
2 |
Polycom getting DCHP address from wrong VLAN |
6:25AM |
1 |
asterisk-users Digest, Vol 75, Issue 7 |
5:15AM |
0 |
Radius client support |
1:35AM |
1 |
REINVITE with Auth Credentials has different SDP Codec |
|
Thursday October 7 2010 |
Time | Replies | Subject |
10:57PM |
1 |
asterisk router |
8:41PM |
0 |
Asterisk 1.8.0 Release Candidate 3 Now Available |
7:07PM |
2 |
Dahdi error |
4:55PM |
0 |
Voice drop out |
3:36PM |
1 |
RTP Read too short |
3:31PM |
0 |
How to change features.conf's atxfer dialing tone ? |
3:18PM |
0 |
convert g729A-g729B and vice-versa |
2:34PM |
1 |
Polycom: full caller ID? |
11:54AM |
2 |
401 Unauthorized with Snom but not with Zoiper softphone |
8:52AM |
2 |
SIP authentication - Thoughts please |
8:44AM |
0 |
Fw: asterisk > cisco gateway > westell > isdx |
|
Wednesday October 6 2010 |
Time | Replies | Subject |
9:00PM |
3 |
integrate Intertel Axxess with Asterisk |
8:51PM |
0 |
How to learn encrypted VoIP development for embedded systems |
8:35PM |
1 |
CALLERPRES() with Queue |
7:50PM |
2 |
SPA-2102 sending local IP instead of WAN IP in SIP packets |
7:25PM |
2 |
ADA: DOA? |
5:39PM |
1 |
AMI connection limit |
3:07PM |
1 |
2 way intercom recommendation for restaurant kitchens |
2:03PM |
1 |
using better quality wav or mp3 in Asterisk 1.2.x |
1:11PM |
2 |
Asterisk 1.8: Warning messages in CLI while putting a SIP-Call on hold |
12:26PM |
0 |
Page minimum number of extensions |
12:15PM |
2 |
Difference |
11:56AM |
2 |
AMI getting related channels in Ringing state |
10:35AM |
3 |
MYSQL ADDON INSTALLATION ERROR |
6:28AM |
2 |
Which virtualization tech to get PCI assignment ? |
6:14AM |
3 |
How to test BRI lines energy saving mode ? |
|
Tuesday October 5 2010 |
Time | Replies | Subject |
11:41PM |
0 |
Web-meetme |
9:13PM |
2 |
Setting up realtime config. |
9:08PM |
2 |
Cisco SIP 8.5 and 9.0 Issues |
8:40PM |
1 |
Asterisk sharing a line with POTS handsets: how to interoperate cleanly? |
8:16PM |
3 |
Asterisk CDR Radius error |
8:04PM |
0 |
meetme don't play conf-invalid if room does not exist |
10:12AM |
5 |
Implementing more than one asterisk instance in the same hardware machine? |
7:08AM |
2 |
CDR record for call originated from CLI originate |
3:48AM |
0 |
Chage Asterisk 1.6.1 to 1.6.2 |
3:13AM |
2 |
Checking SIP Headers existence and content |
|
Monday October 4 2010 |
Time | Replies | Subject |
9:44PM |
0 |
DAHDI 2.2.1, Asterisk 1.6.2.6 - Channel unacceptable (6) |
7:27PM |
3 |
take input and store in variable |
7:17PM |
1 |
DISA does not accept "pause" from cellphones when upgrading from 1.4 to 1.6 |
6:48PM |
0 |
session border controller |
6:32PM |
1 |
Registering Multiple Trunks to Service Provider |
1:44PM |
3 |
Module reload |
12:26PM |
1 |
asterisk-users Digest, Vol 75, Issue 2 |
10:10AM |
1 |
Inter pbx communication via BRI |
9:24AM |
3 |
Phantom phone ringing |
|
Sunday October 3 2010 |
Time | Replies | Subject |
8:19PM |
3 |
SIP flood attacK |
6:20PM |
1 |
more condition check for gotoif |
2:34PM |
1 |
other end hangup |
2:29PM |
1 |
Flash WAV Player |
|
Saturday October 2 2010 |
Time | Replies | Subject |
6:59PM |
2 |
Attempts to hack Asterisk - What do these lines means |
6:56PM |
2 |
Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T? |
4:24PM |
4 |
minimum card for dahdi timing source ? |
10:09AM |
1 |
RE : Re: differential billing |
|
Friday October 1 2010 |
Time | Replies | Subject |
8:50PM |
2 |
AMI Originate |
6:00PM |
0 |
Looking for a PSTN DTMF echo test |
2:08PM |
2 |
No translator path exists for channel type DAHDI (native 76) to 256 |
12:58PM |
0 |
Need some info on cmd Bridge (Confbridge) |
12:49PM |
1 |
debian/dahdi/zaphfc - Unable to receive TEI fromnetwork! |
12:11PM |
0 |
Asterisk 1.6.1 Realtime Extensions => Limited ? |
10:37AM |
0 |
debian/dahdi/zaphfc - Unable to receive TEI from network! |
8:52AM |
2 |
Asterisk/Realtime and MySQL |