| 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 |