| Tuesday November 30 2004 |
| Time | Replies | Subject |
|---|
| 11:00PM |
4 |
After setting up my FXO card, what should I now order from my telco? |
| 9:29PM |
2 |
Can't get x100p to answer the phone |
| 8:58PM |
1 |
HFC-S card for Australia? |
| 8:43PM |
3 |
Cisco Asterisk Integration |
| 8:06PM |
2 |
* Compatible VSP Service in Ukraine? |
| 7:52PM |
2 |
broadvoice and gsm codec |
| 6:29PM |
1 |
kernel: Out of storage space while 900 MB free? |
| 6:26PM |
0 |
conf from database |
| 5:43PM |
1 |
cisco 7960 sccp firmware version? |
| 5:41PM |
0 |
park app vs. extension 700 |
| 4:18PM |
0 |
Pick up call without ringing an extension |
| 3:35PM |
0 |
[BOOK] VoIP Telephony with Asterisk |
| 3:02PM |
0 |
CAll Parking Help needed |
| 2:51PM |
1 |
Issues with zaptel on FC3 - don't know how to fix zaptel after yum update |
| 2:12PM |
1 |
National (US) callerid name resolution for yourasterisk box |
| 1:56PM |
3 |
Asterisk for home office |
| 1:53PM |
1 |
National (US) callerid name resolution for your asterisk box |
| 1:39PM |
1 |
(no subject) |
| 1:13PM |
2 |
Dual NAT for SIP |
| 12:44PM |
0 |
Is it feasible to use 1 SIP account for PSTN connection on an Asterisk gateway accepting IAX2 connections? |
| 12:34PM |
0 |
AW: zaphfc problem |
| 12:33PM |
5 |
Asterisk PBX Manager |
| 12:10PM |
0 |
Trouble-shooting SIP/2.0 482 Loop Detected |
| 11:52AM |
0 |
Simple *69 |
| 11:45AM |
1 |
Agents/Queues - Drops call after 60 seconds |
| 11:32AM |
3 |
Fedora Core 2 firewall rules - NO NAT! |
| 11:27AM |
0 |
Polycom Call Park (with sip debug attached) |
| 11:25AM |
0 |
VoIP Business Weekly Article |
| 11:11AM |
5 |
Re: Asterisk-Users Digest, Vol 4, Issue 405 |
| 10:52AM |
1 |
Pls help me i can't send a voicemail by sendmail |
| 10:32AM |
1 |
Problem with a new italian service provider... |
| 10:24AM |
4 |
Asterisk Process Stop After few hours |
| 9:55AM |
2 |
Really Get 96 Simul Calls? |
| 9:28AM |
3 |
cisco 7902g |
| 8:20AM |
3 |
ASTCC and Pattern question |
| 8:20AM |
2 |
grandstream bt100 |
| 8:01AM |
0 |
H323 -- No Audio |
| 7:44AM |
2 |
Spandsp kind of working |
| 7:29AM |
0 |
Any tool to ease provisioning IAXy? |
| 7:26AM |
0 |
chan_capi compilation problems |
| 7:23AM |
0 |
ParkAndAnnounce Problem, Great Idea, not working consistently |
| 7:06AM |
0 |
E1s ISDN PRI & CPC |
| 6:58AM |
0 |
RE: Parking from call group problems traced to context |
| 6:35AM |
5 |
cisco dial-peer voip |
| 6:32AM |
3 |
Passing Var to PHP AGI script |
| 6:26AM |
1 |
Passing Var to PHP-AGI |
| 6:19AM |
4 |
chan_capi on 2.6 - impossible? |
| 5:41AM |
0 |
Chanspy ? |
| 5:14AM |
3 |
fxo connection in the UK |
| 4:15AM |
0 |
empty username in authorization section ?! |
| 4:14AM |
1 |
realTime configuration help needed |
| 3:08AM |
0 |
SIP client registration ignored by Asterisk |
| 2:52AM |
1 |
Zaprtc seems unsupported, Asterisk in production environment without Digium cards |
| 2:46AM |
0 |
clients behind nat |
| 2:34AM |
3 |
7960 utilize all lines |
| 2:13AM |
0 |
No voice when I dial out |
| 1:58AM |
1 |
Performance problems |
| 1:37AM |
2 |
Is the wcfxo driver sharing an interrupt with Intel 82801DB-ICH4(sound card) |
| 1:08AM |
1 |
wanic 520 with asterisk card |
| 12:19AM |
0 |
Multiple IPs and SIP |
| |
| Monday November 29 2004 |
| Time | Replies | Subject |
|---|
| 11:51PM |
1 |
SIP.Conf help? (srvlookup) |
| 11:04PM |
2 |
Problems with conference on FreeBSD 5.2.1 w/* 1.0.1 |
| 10:55PM |
1 |
Terminal Services + VoIP |
| 9:48PM |
2 |
SPA-2000 Dropped calls |
| 8:45PM |
1 |
IAX port |
| 8:31PM |
3 |
TE410P lights don't blink read after the module is loaded |
| 8:28PM |
1 |
Outbound E&M? |
| 8:22PM |
0 |
IPv6-enabled Asterisk + testing |
| 8:20PM |
3 |
chan_oh323.o |
| 8:06PM |
4 |
Gentoo and Asterisk - any experiences? |
| 7:50PM |
2 |
Cannot Start Asterisk |
| 7:40PM |
2 |
Problems starting Asterisk with TDM22B |
| 7:36PM |
0 |
res_odbc and configuration files |
| 6:09PM |
1 |
Calling from PSTN let exension 601 ring twice, hang up and starts over again to ring twice, ... |
| 6:08PM |
2 |
Compiling zaptel 1.0.2 on Fedora Core |
| 5:54PM |
0 |
Cisco FXO Caller-ID |
| 4:49PM |
3 |
no plain text passwords in iax.conf |
| 4:47PM |
1 |
Cisco gateway help needed |
| 4:26PM |
1 |
T.38 support |
| 3:07PM |
4 |
asterisk newsgrup proposal or phpBB forum |
| 3:05PM |
0 |
I apologize |
| 2:20PM |
1 |
Packet8 integration into Asterisk? |
| 2:15PM |
1 |
CONSOLE/dsp and command line play of wave file |
| 2:03PM |
5 |
Comparision of IAX2, FWD, iaxtel etc etc. |
| 1:48PM |
2 |
Prepaid |
| 12:48PM |
1 |
IAXy power source from Radio Shack |
| 12:45PM |
2 |
Fedora Core 3 & TDM400P cards? |
| 12:38PM |
2 |
Variable substitution - How can I do Dial(${DIALSTRING}) where ${DIALSTRING} is 'SIP/201, 15, tT'? |
| 12:25PM |
1 |
TOS Settings to DSCP |
| 12:12PM |
1 |
[Fwd: Re: Adit 600 channel bank in UK setting] |
| 12:08PM |
1 |
How to rid yourself of Broadvoice |
| 11:41AM |
0 |
FWD * and IAX2... |
| 11:26AM |
2 |
Vonage integration... Hardware or Softphone type acct. |
| 10:57AM |
0 |
Parking from call group problems |
| 10:48AM |
4 |
Small PBX setup |
| 10:35AM |
4 |
Zap gives no ring to the caller... |
| 10:16AM |
1 |
NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' |
| 10:16AM |
0 |
Regular Phones - ISDN NT - FXS Adapters |
| 10:06AM |
1 |
Sending triggers through SIP |
| 9:46AM |
2 |
Asterisk on a notebook: Modem = FXO? |
| 9:43AM |
1 |
IAXy and ADPCM codec problem |
| 9:36AM |
0 |
Subject: IAXy and ADPCM codec problem. |
| 9:18AM |
1 |
Spawn extension |
| 9:02AM |
1 |
Fax pass-throught. |
| 8:44AM |
0 |
IAX2 Warnings - chan_iax2.c:1464 attempt_transmit |
| 7:22AM |
3 |
how to call s extension from SIP phone? |
| 7:08AM |
1 |
Record() and problems converting with sox. |
| 6:17AM |
2 |
Asterisk on a notebook |
| 6:15AM |
1 |
New T100P Pri install suggestions? |
| 5:40AM |
1 |
unable to compile testcpuid.c in spandsp in x86_64 |
| 5:10AM |
3 |
low quality sound samples |
| 4:05AM |
0 |
Problem when I call someone who is busy |
| 3:46AM |
3 |
Audio Drops out at Random - one way |
| 3:34AM |
1 |
Dial plan for TDM22B |
| 1:54AM |
1 |
Polycom Reboot Script PRI errors!! |
| 12:51AM |
0 |
Asterisk A LA MEXICANA!!!! |
| |
| Sunday November 28 2004 |
| Time | Replies | Subject |
|---|
| 11:51PM |
0 |
how to modify dsp.c so that It can detect busytone outside US? |
| 11:27PM |
3 |
D-LINK PoE switch, does it work with cisco or do I need to do the cable trick? |
| 9:54PM |
0 |
Ateus VoiceBlue |
| 9:39PM |
0 |
optipoint400 + MOH |
| 9:21PM |
1 |
optipoint 400 standard + MOH |
| 6:34PM |
17 |
Wiki down? |
| 4:31PM |
0 |
Entire mailing list archive download? |
| 4:01PM |
1 |
OT: mixing monitor files to stereo wav |
| 3:36PM |
3 |
soxmix |
| 1:49PM |
1 |
multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible? |
| 12:10PM |
1 |
IAX2 and FWD problems? |
| 12:07PM |
5 |
IP to IP call without server? |
| 12:01PM |
4 |
PRI Dialing failure? |
| 10:38AM |
4 |
Registering on Gatekeeper |
| 10:36AM |
4 |
Phone Selection |
| 10:36AM |
3 |
OS Choice ? |
| 10:26AM |
0 |
Registering on GK |
| 10:10AM |
4 |
Asterisk not startin anymore. |
| 9:29AM |
2 |
Asterisk/linux 2.6.9 kernel build failure |
| 8:56AM |
2 |
[Fwd: Call Transfer between phones] |
| 8:37AM |
2 |
am i baned or something? |
| 7:30AM |
1 |
asterisk based bbs |
| 5:49AM |
0 |
Real time queue monitoring |
| 4:23AM |
1 |
asterisk compile errors - pbx_dundi.c -help |
| 3:52AM |
0 |
Fwd: Re: very newbie question |
| 3:36AM |
1 |
SetVar ALERT_INFO |
| 1:32AM |
0 |
Hardware performance issues - Zaptel / wct4xxp for TE405P |
| 12:38AM |
2 |
GNUGK + Asterisk consultant requiered |
| 12:29AM |
0 |
Flash Timings |
| 12:07AM |
4 |
Experiences with Termination Providers? |
| |
| Saturday November 27 2004 |
| Time | Replies | Subject |
|---|
| 11:39PM |
0 |
problems setting up cdr_addon_mysql |
| 10:34PM |
0 |
Asterisk and GNUGK Consultant Requiered |
| 10:32PM |
0 |
Asterisk + GNUGK Consultant requiered |
| 7:54PM |
1 |
VoiceMail Outdial? |
| 6:12PM |
0 |
Problems compiling zaprtc on 2.4.27 kernel |
| 5:31PM |
3 |
How to test if PCI 2.2? |
| 4:40PM |
0 |
Contact me Asap! |
| 2:56PM |
0 |
Failed to WWW-authenticate on INVITE |
| 2:34PM |
3 |
Problem with voicemailsystem |
| 2:00PM |
1 |
Low Volume WAV Files in Email Attachments |
| 12:37PM |
4 |
very newbie question |
| 11:30AM |
0 |
Can't Register! |
| 10:56AM |
3 |
newbie problem |
| 8:24AM |
0 |
Zapata: No such device or address |
| 8:16AM |
0 |
Built-in Extension Numbers |
| 8:00AM |
1 |
isdn4linux delay |
| 6:45AM |
1 |
getting TDM400P to work in a system that previously had Voicetronix card |
| 6:13AM |
0 |
RealTime Mysql - error res_config_mysql.so |
| 6:10AM |
2 |
rtp compile error |
| 5:17AM |
2 |
capi question |
| 4:46AM |
1 |
Interfacing T100P with Definity PBX |
| 3:27AM |
1 |
asterix as proxy |
| 2:04AM |
0 |
allow=all in sip.conf [genernal] no longer evil (I think) |
| 1:34AM |
1 |
Meetme Help !!!! |
| 1:15AM |
1 |
Reconfiguring a Zap Channel on the fly |
| |
| Friday November 26 2004 |
| Time | Replies | Subject |
|---|
| 10:18PM |
1 |
Which is the best signalling for FXS |
| 9:54PM |
2 |
Is "Busydetect" obsolete in the latest CVS? |
| 9:25PM |
2 |
Help with broadvoice outbound plz... ;) |
| 8:09PM |
4 |
Grandstream BT102 Busy signal on hangup |
| 7:29PM |
4 |
*67 or *57 |
| 7:28PM |
1 |
FWD with iax2 |
| 7:25PM |
2 |
Uniden UIP200 -- configured, but not working? |
| 7:07PM |
0 |
^5 problem with chan_unicall.c for Asterisk |
| 6:40PM |
0 |
TDM22B - how to setup the extensions ?? |
| 6:32PM |
1 |
Asterisk+ MGCP |
| 5:15PM |
1 |
direct asterisk to asterisk SIP calls without external SIP provider |
| 2:37PM |
0 |
ast_data causes pbx_realtime.so to fail |
| 1:28PM |
0 |
advise for cheap ISDN card which works with chan_capi and supports p2p mode |
| 1:15PM |
2 |
low voice only |
| 1:05PM |
1 |
Voicemail / sendmail |
| 12:45PM |
0 |
^^4 problem with chan_unicall.c for Asterisk |
| 11:20AM |
1 |
OT - how to get BT to present a number |
| 11:16AM |
0 |
"reason 23 (Temporary failure)" when using Dial(OH323) |
| 11:10AM |
4 |
SIP phones cutting out with Asterisk?? |
| 10:18AM |
0 |
sip call test |
| 9:37AM |
2 |
E&M Digium card quotation |
| 9:09AM |
0 |
Exiting out of Voicemail with a '0' |
| 8:56AM |
1 |
How to transfer value to extensions.conf? |
| 8:56AM |
1 |
Asterisk - ACD. |
| 8:42AM |
2 |
T1 and FX CPE |
| 7:24AM |
1 |
AGENTDUMP lines |
| 7:20AM |
0 |
P2P (DDI) mode with chan_capi 0.3.5 |
| 7:03AM |
2 |
Execute a script upon registration |
| 6:16AM |
0 |
Can I trigger an application in * with DTMF tones, during a call? |
| 4:28AM |
0 |
Forwarding SIP calls to another SIP Proxy (Peer) |
| 4:14AM |
2 |
IAXy and DHCP |
| 4:00AM |
4 |
overriding DTMF and codec from dialplan? |
| 3:39AM |
0 |
load wcfxs module |
| 3:36AM |
2 |
problem with chan_unicall.c for MFC/R2 with asterisk |
| 3:31AM |
1 |
Monitoring app. - see whats really going on in asterisk |
| 3:16AM |
0 |
PrepaidAuthCID - nothing happens |
| 2:55AM |
1 |
can anyone will help me regarding autodialing in asterisk |
| 2:48AM |
1 |
Quality of the voicemail sound |
| 2:35AM |
3 |
Billing of outoging calls via CAPI |
| 2:21AM |
0 |
snom - blinking leds on fuction keys when call is not yet established - how? |
| 2:05AM |
1 |
Monitor performance |
| 2:05AM |
1 |
General feature questions |
| 1:51AM |
0 |
Re[4]: [Asterisk-Dev] Asterisk Hardware Platform - Intel x86 versus Intel RISC Xscale (ARM) |
| 12:56AM |
4 |
Where did USE_MYSQL_FRINDS go ? What to use ? |
| |
| Thursday November 25 2004 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
Consultancy service needed urgently ! |
| 11:45PM |
1 |
No Music: Queue Hold and MusicOnHold |
| 10:10PM |
0 |
Problem with IAX2 Unregistered in the chan_iax2.c and data_pgsql.c file |
| 9:55PM |
1 |
Problem with onboard sound card on kphone |
| 8:12PM |
3 |
Playing reveived message WAV file |
| 6:26PM |
3 |
OH323 Rocks :) --- H323 guys, use it to solve no answer at this time problem!!! |
| 1:09PM |
1 |
SNOM telephones and LEDs |
| 12:51PM |
1 |
Fax server (TxFax) fails during transmission |
| 12:04PM |
1 |
Interview with Mark Spencer |
| 12:03PM |
3 |
redhat9 100% CPU |
| 12:02PM |
1 |
allow=SLINR |
| 11:55AM |
0 |
Solution - ISDN-PRI hangup cause |
| 11:38AM |
1 |
Stanaphone down? |
| 11:02AM |
4 |
Opinions on renice or turning off swap or ramdis k as swap? |
| 10:26AM |
0 |
Area Code 514 DIDs |
| 9:39AM |
3 |
configuring voicemail |
| 8:56AM |
2 |
Cannot get two TE410Ps to operate correctly in the same machine |
| 8:38AM |
1 |
astcc newbie question |
| 8:33AM |
0 |
probleme with running lib_unicall with asterisk |
| 7:59AM |
1 |
Module Failure |
| 5:57AM |
1 |
Call to x-lite clients failing? |
| 5:11AM |
0 |
Asterisk Hardware Platform - Intel x86 versus Intel RISC Xscale (ARM) |
| 5:10AM |
0 |
record call on demand |
| 4:09AM |
0 |
How to make/recieve call using asterisk whenthereis a power failure? |
| 3:57AM |
0 |
Forwarding Call |
| 3:53AM |
4 |
Billing (itemized) in the UK |
| 3:46AM |
1 |
No hangup(vpb) |
| 3:26AM |
0 |
ZAP FXS problem - no caller id |
| 3:10AM |
1 |
Connecting a PBX with Asterisk via E1 / PRI |
| 3:06AM |
2 |
How to make/recieve call using asterisk when thereis a power failure? |
| 3:02AM |
1 |
astGUIClient Question |
| 2:56AM |
3 |
How to make/recieve call using asterisk when there is a power failure? |
| 2:03AM |
2 |
oh323 compile issue |
| 1:50AM |
1 |
Can't hear playtones? |
| 1:24AM |
3 |
Zaptel on Suse 9.0 |
| |
| Wednesday November 24 2004 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
supported RFCs |
| 10:25PM |
1 |
asterisk 1.0.1 |
| 10:22PM |
2 |
Changing Asterisk Voicemail Storage Location |
| 9:50PM |
0 |
Unable to open master device |
| 9:44PM |
1 |
Cannot open /dev/dsp |
| 9:21PM |
1 |
I just got my TDM22B - but no data sheet |
| 8:42PM |
2 |
Asterisk Digium FXS |
| 8:04PM |
0 |
H323-Asterisk-SIP-TNT consultant needed |
| 6:57PM |
0 |
Call External Program When SIP Message Arrives |
| 5:18PM |
2 |
asterisk and verizon DSL |
| 2:57PM |
2 |
how to use stop calls |
| 2:48PM |
2 |
Bothering with H323 |
| 2:43PM |
0 |
How to Modify Diversion Header for 3rd Party SIP Vmail? |
| 12:50PM |
1 |
Just upgraded from multiple X100P's to a T100P |
| 12:40PM |
3 |
Haven't got a clue ... |
| 12:23PM |
1 |
Question on IXAy |
| 11:03AM |
5 |
GUI |
| 10:31AM |
1 |
How to decrease the speech volume for record? |
| 9:33AM |
2 |
Graststream ATA 286 Caller ID Europe |
| 9:17AM |
4 |
asterisk and pstn |
| 9:14AM |
4 |
zap fxo hangs after upgrade to stable v1-0 |
| 9:09AM |
1 |
Asterisk/Panasonic PRI Integration |
| 8:45AM |
1 |
Problems with udev on FC3 |
| 8:21AM |
1 |
Re: Asterisk timer for Freebsd |
| 7:51AM |
1 |
Busy Lamp Field |
| 7:04AM |
1 |
Sip test |
| 6:24AM |
2 |
Asterisk and Dialogic LSI161SCREV2 --- Don't kill me ; -) |
| 6:12AM |
2 |
Codec control |
| 5:59AM |
2 |
call forwarding to gsm phones |
| 5:56AM |
1 |
bristuff'ed version doesn't run |
| 5:36AM |
3 |
Asterisk with ISDN |
| 5:31AM |
1 |
gateways failover with asterisk |
| 4:40AM |
1 |
vm notification no longer contains calling party |
| 3:36AM |
3 |
Grandstream Firmware 1.0.5.16 Attended Transfer |
| 3:34AM |
1 |
Find extension from Dial(,M()) macro |
| 3:11AM |
0 |
Have anyone successfully install Daniel G729 test suite ? mine core dumped !! |
| 2:14AM |
1 |
Horrible BUZZZZ noise when sounds/music play on SIP phone? |
| 1:46AM |
1 |
Which modem is known to work with asterisk? |
| 1:27AM |
0 |
No debugging informations on the CLI after patching with ast_data 1.0.2 |
| |
| Tuesday November 23 2004 |
| Time | Replies | Subject |
|---|
| 11:41PM |
7 |
Unable to open master device '/dev/zap/ctl' |
| 11:17PM |
1 |
CLI > h.323 show codecs shows nothing |
| 6:12PM |
2 |
need some advice |
| 5:28PM |
1 |
AstriCon offers a most sincere and humble apology for the barage of mail... |
| 5:19PM |
0 |
SBC ADTSe - Sending DP digits |
| 5:14PM |
1 |
Is there a way to check if an extensions exists in a context before you send the call there. |
| 4:52PM |
2 |
Asterisk on a Linksys WRT54G(S) |
| 4:50PM |
1 |
linking 2 isdn30 and 2 meridian cards |
| 4:36PM |
0 |
rtp.c dtmf issues solved. |
| 4:17PM |
5 |
Fw: Gift for Mark Spencer |
| 3:46PM |
2 |
Re: list proposition |
| 3:13PM |
0 |
using asterisk to bridge H323v1 to SIPv2 |
| 2:58PM |
3 |
Re: List proposition |
| 2:33PM |
2 |
Can isdn data calls routed through 2 t100p's |
| 2:18PM |
1 |
IAX2->SIP->meetme = ZOMBIE |
| 1:41PM |
0 |
meetme2 can't set status |
| 1:31PM |
2 |
Yet another faxing issue.. |
| 1:04PM |
1 |
Queue Patch - estimated hold time announcements |
| 11:10AM |
4 |
ATA186 V2.15.ms upgrade |
| 10:19AM |
0 |
Zombie channels dropping lines |
| 9:53AM |
5 |
ATA186 V2.15.ms |
| 9:17AM |
4 |
Quick Questions - IVR=Auto Attendant? |
| 9:03AM |
1 |
Fax over SIP Problems (sorry for this topic ...) |
| 9:01AM |
4 |
Forwarding calls |
| 8:53AM |
1 |
CP-7960 |
| 8:52AM |
0 |
Fax over TDM400 and E100P disconnects |
| 8:32AM |
4 |
ASTCC Routes |
| 8:19AM |
3 |
Firefly on Linux |
| 8:10AM |
1 |
Error when install E100P |
| 8:04AM |
0 |
please help !! - context for an incoming call |
| 7:47AM |
0 |
SIP Registration failed notices |
| 7:00AM |
4 |
Spandsp and Asterisk |
| 6:34AM |
2 |
Re: Asterisk-Users Digest, Vol 4, Issue 300 |
| 6:31AM |
2 |
PRI Logging |
| 6:17AM |
5 |
NEED HELP!! |
| 6:14AM |
1 |
Polycom 500 bootrom.ld problem |
| 5:34AM |
2 |
Commercial g723.1 license for asterisk |
| 5:14AM |
0 |
RE : -lssl |
| 5:14AM |
0 |
Asterisk not relaying SIP messgaes |
| 5:04AM |
2 |
-lssl |
| 4:48AM |
4 |
oh323/g729 and DTMF |
| 4:45AM |
0 |
astcc db creation |
| 4:41AM |
1 |
Newbie questions from South Africa: Initial setup |
| 4:37AM |
0 |
Huge ten second audio delay on SIP channel |
| 4:23AM |
0 |
Problems with MACRO_EXTEN variable |
| 4:11AM |
0 |
Random Audio Drop out one side |
| 4:11AM |
1 |
a=rtpmap:101 telephone-event/8000 |
| 3:34AM |
1 |
Paul Mahlers Book |
| 3:08AM |
1 |
Firefly:Canreinvite problem |
| 2:33AM |
1 |
Error on install under Fedora Core 3 |
| 2:15AM |
1 |
dail cli |
| 2:03AM |
0 |
DTMF mode autodetect? |
| 12:54AM |
0 |
Asterisk & Windows Messenger |
| |
| Monday November 22 2004 |
| Time | Replies | Subject |
|---|
| 10:17PM |
1 |
Uniden UIP200 configuration -- manual MIA? |
| 9:33PM |
3 |
ChanSpy |
| 9:08PM |
0 |
H323 linking with asterisk |
| 9:07PM |
0 |
How to configure the Asterisk server such that a FXS phone can talk to SIP client? |
| 7:57PM |
2 |
dtmf tones during conversation |
| 7:33PM |
1 |
Anyone use SixNet for IAX termination? |
| 4:42PM |
0 |
SIP phones disconnect frequently |
| 4:19PM |
1 |
T100P -- data? |
| 4:18PM |
0 |
Asterisk with MeritCall |
| 3:59PM |
0 |
new application swait... |
| 3:59PM |
2 |
chan_h323 on AMD64 |
| 3:43PM |
2 |
Granstream BT100 - only partial success |
| 3:25PM |
0 |
Asterisk and Bastille |
| 3:02PM |
0 |
Configuring Asterisk From Postgres |
| 2:46PM |
2 |
sip.conf not paying attention to allow/disallow |
| 2:31PM |
2 |
Polycom Problems |
| 2:11PM |
0 |
Re:SIP Problem |
| 1:57PM |
9 |
asterisk gui? |
| 1:33PM |
1 |
SIP Problem! |
| 1:07PM |
1 |
Cisco 7940 Volume low |
| 12:38PM |
8 |
Patching asterisk for spandsp |
| 11:51AM |
3 |
Cisco 7960 version 7.3 SIP not always able to hear calling person |
| 11:16AM |
1 |
Using IPKall and SIP with insecure=very |
| 10:57AM |
2 |
edirecting calls with Asterisk |
| 10:53AM |
0 |
asterisk manager api to stop a stream file command in an agi |
| 10:33AM |
3 |
IPv6 and Asterisk? |
| 10:24AM |
3 |
Zap - 256 format frames |
| 10:16AM |
1 |
Siemens optiPoint 300 |
| 10:08AM |
2 |
Problem with fax tone (CNG) from TxFax and busy detect |
| 9:47AM |
2 |
RE: Asterisk-Users Digest, Vol 4, Issue 298 |
| 8:56AM |
0 |
Cisco Call Manager and Asterisk |
| 8:47AM |
2 |
Creating CDR's with online connected time |
| 8:47AM |
6 |
Linksys RT31P2 |
| 8:41AM |
2 |
Unknown number CID on SIP phone |
| 8:03AM |
3 |
which ISDN Card? |
| 8:03AM |
1 |
Call Deflection (CD) with ZapHFC |
| 7:56AM |
1 |
callprogress option |
| 7:52AM |
1 |
Test Number in the UK? |
| 6:49AM |
1 |
wiki down ? |
| 5:26AM |
1 |
Strange Fromuser behavior? |
| 4:27AM |
0 |
Problems with not correctly unregistered users... |
| 2:59AM |
1 |
IAX error tolerence?? |
| 1:47AM |
3 |
hangup()??? |
| |
| Sunday November 21 2004 |
| Time | Replies | Subject |
|---|
| 10:26PM |
2 |
SPA-841 / SPA-2100 Canadian Distributor |
| 9:48PM |
0 |
Is there Asterisk module for Logwatch? |
| 8:14PM |
1 |
Mailing List Admin - Remove annoying user [Fwd: RE: Re: Get the Caller-ID without Answering] |
| 7:17PM |
2 |
Examples of hardware implementations |
| 6:04PM |
3 |
Get the Caller-ID without Answering |
| 5:24PM |
1 |
SER is a better NAT solution? |
| 4:01PM |
0 |
Headsets for Polycom Soundpoint 500/600 |
| 4:01PM |
3 |
Headsets for Cisco 7940/7960 |
| 1:49PM |
3 |
Error "WARNING[-150101888]" when starting Asterisk. |
| 1:01PM |
0 |
iax busy / unavailable - not registered |
| 12:50PM |
2 |
Fw: TDMoE over bonded NIC's |
| 12:18PM |
3 |
TDM400 FXO stops handling outgoing calls, but still accepts incoming? |
| 10:49AM |
0 |
HFS in NT mode getting PRI got event: 6 on Primary D-Channel of span 1 |
| 10:22AM |
3 |
I Am Missing Something Somewhere Somehow! |
| 10:02AM |
1 |
Gatway with IAX ? |
| 9:56AM |
0 |
sip debug command? |
| 9:37AM |
1 |
Grandstream Ringtone |
| 9:34AM |
0 |
No incoming calls on skinny phone |
| 9:31AM |
0 |
Asterisk Newsletter :: Back online! |
| 6:54AM |
1 |
incompatible with our capability 0x400. |
| 6:39AM |
1 |
make asterisk accept Register messages |
| 6:31AM |
0 |
Flashing Active ZAP Channels |
| 5:24AM |
4 |
Snom 190 - dhcp - settings_server |
| 2:38AM |
4 |
UK available SIP phone? |
| 1:08AM |
1 |
Using CallingPres to set up CallerID blocking |
| |
| Saturday November 20 2004 |
| Time | Replies | Subject |
|---|
| 9:43PM |
0 |
* and scansoft TTS |
| 9:13PM |
1 |
Asterisk dead but pid file exists - gdb asterisk core.13089 |
| 8:02PM |
1 |
extensions.conf help needed |
| 5:16PM |
1 |
IAX IAX connection |
| 5:10PM |
0 |
zaptel driver problem |
| 5:06PM |
3 |
A new alternative to see who is online |
| 4:44PM |
0 |
Changing simple switch dialtone |
| 4:36PM |
0 |
Odd situation with Cisco 7960 IP phone |
| 4:29PM |
2 |
Problems with call files (/var/spool/asterisk/outgoing) |
| 4:04PM |
1 |
TE410P PRI problems |
| 3:57PM |
1 |
ANY DEVELOPERS HERE? "warning: implicit declaration of function `__use_ast_pthread_create_instead__" |
| 3:46PM |
1 |
Queue Sounds - not working? |
| 2:45PM |
6 |
SIP Phones-Receptionist Setup |
| 2:35PM |
1 |
IAX Dialstatus |
| 2:23PM |
0 |
SIP Call not Approved |
| 2:08PM |
1 |
IAX issue at nufone |
| 12:51PM |
1 |
How to encript SIP comunications? |
| 11:15AM |
0 |
Playing announcement when call is answered |
| 10:39AM |
0 |
Can anyone shed some light on wht these calls were dropped? |
| 7:44AM |
1 |
three way mixing / conferencing |
| 7:32AM |
0 |
Setting the EXTEN variable - is it possible? |
| 6:33AM |
0 |
Fax testing using loop-back |
| 5:53AM |
2 |
zaphfc sound problems |
| 4:37AM |
0 |
Bug with Dial in AGI script? |
| 2:15AM |
3 |
block caller id |
| 1:08AM |
1 |
* and NAT |
| 12:57AM |
0 |
SIP to IAX using G.729 |
| 12:46AM |
1 |
Monitor Command |
| |
| Friday November 19 2004 |
| Time | Replies | Subject |
|---|
| 10:08PM |
4 |
Multiple asterisk process |
| 9:06PM |
2 |
Polycom Soundstation IP 3000 firmware |
| 8:48PM |
2 |
Just getting started... |
| 6:49PM |
1 |
PRI NI2 and callerID name |
| 5:32PM |
5 |
Asterisk and H.323 Gatekeeper |
| 4:40PM |
1 |
Starting AGI when handset is picked up? |
| 4:31PM |
4 |
IAXy Configuration |
| 4:27PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Meeting tomorrow 11/20/04 |
| 3:51PM |
0 |
Question involving Windows Messenger 5.0 and Asterisk (SDP related) |
| 3:13PM |
0 |
SIP Clients other than 200-299 |
| 3:09PM |
0 |
remote iaxy device Ping:OK iax2 Poke: no answer |
| 2:48PM |
0 |
asterisk and level3 |
| 2:38PM |
3 |
Alcatel PBX |
| 2:22PM |
1 |
Newbie Basic Questions |
| 12:27PM |
4 |
Error during installation |
| 12:01PM |
2 |
How to enter billing codes when dialling |
| 11:17AM |
0 |
Mitel 5220 phones |
| 11:04AM |
1 |
Voice + DTMF |
| 10:31AM |
1 |
Broadvoice update |
| 10:18AM |
2 |
Shared line appearances |
| 10:17AM |
1 |
ASTCC MySQL CDR's |
| 10:16AM |
5 |
txfax |
| 10:11AM |
0 |
Cisco 7970 Non-SIP Phone setup with Asterisk |
| 10:08AM |
0 |
differents contexts for a channel |
| 9:40AM |
0 |
Asterisk and Tecom IP2005 phone, problems :( |
| 9:12AM |
1 |
SBC VoIP Tariff to ISP's |
| 8:40AM |
0 |
AgentMonitorOutgoing => is there an opposite ? |
| 8:22AM |
4 |
hello |
| 8:20AM |
0 |
helo |
| 7:52AM |
2 |
Zaptel init script |
| 7:50AM |
0 |
H.323 Status |
| 7:28AM |
0 |
Asterisk crashes with Unicall |
| 7:28AM |
5 |
Fedora Core 3 supported? |
| 7:22AM |
2 |
"Best" line protocol for T1 |
| 6:54AM |
1 |
rtp codec error |
| 6:40AM |
2 |
app_sms: problems sending a sms |
| 6:16AM |
0 |
Fwd: MARIO SPOLJAR is not longer working for PLIVA |
| 5:56AM |
2 |
Routing between different interfaces |
| 5:56AM |
2 |
Need help selecting phones |
| 4:48AM |
0 |
X100P and Siemens Gigaset 4175 |
| 4:19AM |
5 |
Unpredictables Hangups |
| 3:47AM |
1 |
R: problem with zyxel prestige 2002 |
| 2:40AM |
2 |
OT - 3com 3C17205 & cisco 79xx |
| 2:08AM |
0 |
Ericsson or ACC - AXC or Tigris ?? |
| 2:08AM |
7 |
i swtiched to digest |
| 1:40AM |
2 |
compiling error |
| 1:14AM |
2 |
E100 or TE410 card an PRA line |
| 1:10AM |
1 |
Digium E100P or TE410P card |
| |
| Thursday November 18 2004 |
| Time | Replies | Subject |
|---|
| 11:27PM |
0 |
Linking H323 with Asterisk |
| 11:26PM |
2 |
changing configuration file |
| 9:37PM |
1 |
X-Lite and Voicemail |
| 9:02PM |
3 |
Little off topic |
| 8:55PM |
0 |
[perhpas OT] asterisk holding rtp ports open with natted spa-3000 |
| 8:09PM |
2 |
[Asterisk-User] recommendation for IP phones |
| 7:55PM |
3 |
SipTone II |
| 6:48PM |
3 |
Is H323 dying? |
| 6:23PM |
3 |
iaxComm to iaxComm |
| 6:17PM |
0 |
Asterisk with verizon DSL and Westell 2200 DSL router |
| 6:14PM |
2 |
Interrupting MusicOnHold while call in queue ? |
| 5:52PM |
2 |
(Analog Intercom) PagePal by ATT -- was hooked to a Merlin |
| 5:45PM |
0 |
DTMF stopped functioning after upgrade to 1.0.2 |
| 5:21PM |
3 |
Best SIP phone for high quality telemarketing |
| 4:48PM |
3 |
"Lobotomized" Sipura SPA-3000 configuration needed |
| 4:10PM |
0 |
Video Phone recommendations for SIP trunking on * |
| 3:47PM |
1 |
Sparc hardware, Linux and X100P |
| 3:21PM |
1 |
Re: Netgear powered switch |
| 3:16PM |
1 |
[Fwd: Re: Adit 600 channel bank in UK setting] |
| 2:57PM |
2 |
Speaking of DS3s.... |
| 2:46PM |
1 |
[OT] PoE switch question (Netgear FSM7326P works |
| 2:45PM |
1 |
Est. count of deployed Asterisk environments? |
| 2:26PM |
3 |
Spam: I really need help with this!!!!!! |
| 2:23PM |
4 |
Controlling Asterisk from PHP? |
| 2:08PM |
0 |
Polycom 300 registration |
| 1:51PM |
0 |
FW: More than 20 FXS |
| 1:50PM |
0 |
No Voice Path With PSTN Call Forward |
| 1:47PM |
2 |
please unsubscribe all pliva.hr members |
| 12:56PM |
1 |
Incorrect parsing of 'unavailable' caller-ID from Cisco gateway |
| 12:01PM |
0 |
OT |
| 12:00PM |
0 |
DTMF noise |
| 11:54AM |
2 |
More than 20 FXS |
| 11:35AM |
0 |
VoIP engineer and technical/networking support |
| 10:41AM |
1 |
Help wanted getting Busy / Congested working properly |
| 10:02AM |
2 |
VOIP security on an IAX connection. |
| 9:56AM |
0 |
Adit 600 channel bank in UK setting |
| 9:46AM |
5 |
TE410P - How many can I have? |
| 9:43AM |
0 |
app_icd compile problem |
| 8:42AM |
0 |
[OT] but of interest to Grandstream users : firmware .5.18 |
| 8:28AM |
0 |
Playtones problems |
| 8:21AM |
1 |
Find out the reason for dropped calls? |
| 8:20AM |
4 |
please Can some bady help me ??? |
| 8:14AM |
8 |
X100p and 6 second delay |
| 8:08AM |
0 |
safe_asterisk isn't auto-restarting |
| 7:51AM |
1 |
Polycom IP 300 PoE? Sipura instead? |
| 7:35AM |
1 |
AW: Voice in Asterisk with BRI ISDN Any properworking configurations yet? |
| 7:07AM |
0 |
asterisk connecting to cisco call manager using quad T1 card |
| 7:03AM |
1 |
Analog ports via USB |
| 6:45AM |
0 |
AW: Voice in Asterisk with BRI ISDN Any proper workingconfigurations yet? |
| 6:40AM |
2 |
Voice in Asterisk with BRI ISDN Any proper working configurations yet? |
| 6:38AM |
0 |
Asterisk server to asterisk server question |
| 4:53AM |
1 |
Zyxel Prestige 2002/2002L sound quality |
| 4:51AM |
1 |
Problems using AGI->get_data |
| 4:48AM |
0 |
FreeBSD asterisk-addons |
| 4:24AM |
1 |
setup question |
| 4:22AM |
0 |
OH323_OUTCODEC=g729 has influence on chan_iax? |
| 3:51AM |
1 |
mISDN & kernel 2.6.9 |
| 3:32AM |
1 |
Setup/SIP routing |
| 3:11AM |
0 |
H323 and AMD64 |
| 3:01AM |
2 |
configure channels |
| 2:32AM |
5 |
Music on Hold on Debian 2.6 help wanted |
| 2:22AM |
0 |
ISDN BRI one way voice quality problem |
| 12:55AM |
5 |
internet bandwidth |
| 12:52AM |
0 |
Queue calls- multiple to same extension, max extensions? |
| 12:35AM |
0 |
inernet bandwidth |
| 12:04AM |
0 |
Queue using iaxy agent fails? |
| |
| Wednesday November 17 2004 |
| Time | Replies | Subject |
|---|
| 11:58PM |
1 |
[OT] PoE switch question |
| 11:16PM |
3 |
Auto Dialing |
| 10:29PM |
0 |
call delay problem after call recording |
| 10:06PM |
0 |
return codes from extension.conf |
| 9:56PM |
0 |
Anybody got asterisk workin with Diva 4bri and fdora core 2? |
| 9:51PM |
0 |
Call ID WinPopup working one-line example withoutscratch file |
| 9:23PM |
2 |
Call ID WinPopup working one-line example without scratch file |
| 8:56PM |
1 |
E100P Media Gateway With Asterisk |
| 8:37PM |
1 |
Motherboard with TE405p |
| 8:30PM |
1 |
Asterisk Call ID Popup |
| 8:13PM |
1 |
Digits entered ARE NOT RECOGNIZED by bank's IVR's |
| 7:51PM |
1 |
Mini Call-ID Winpopup |
| 6:37PM |
5 |
The Apperiant Death of IAXtel |
| 5:52PM |
2 |
OT: Why "encrypted" config files |
| 5:20PM |
1 |
Removed default indication country 'us' |
| 4:42PM |
2 |
Cisco SIP Firmware HERE!!!! |
| 3:27PM |
3 |
Polycom IP 300 PoE? |
| 3:19PM |
1 |
Problem with an hardware phone: Maximum retries exceeded |
| 3:17PM |
2 |
Call Status |
| 3:07PM |
4 |
Cisco 7970G VOIP phones |
| 2:47PM |
0 |
start_pri: Unable to open D-channel 24 (No suchdevice or address) |
| 2:46PM |
0 |
Strange g729 error. Just now started. |
| 2:34PM |
1 |
Coverting Cisco 7960 to SIP |
| 2:23PM |
6 |
How to generate "ringing tone" to a calling party. |
| 1:44PM |
1 |
Zap card, PRI, Fax detection, and 1.0 stable |
| 1:05PM |
5 |
Call ID Mini-Popup? |
| 12:44PM |
2 |
PowerEdge 17500 with TDM400P - 4 FXO -- NMI, loud noise when dialing out |
| 12:42PM |
0 |
CallerID and Outlook / CSV |
| 12:21PM |
3 |
chan-sccp problem, phone is not registering |
| 12:18PM |
2 |
AstLinux 0.1.3 released |
| 12:15PM |
1 |
Why <ZOMBIE> ? |
| 12:01PM |
0 |
BroadVoice patch on latest CVS snapshot |
| 11:32AM |
3 |
IVR and voice mail using G729 |
| 11:07AM |
4 |
patch for chan_capi to compile with latest CVS |
| 11:06AM |
2 |
Asterisk on Solaris |
| 11:03AM |
1 |
Does ASTCC Require CDR_MySql? |
| 10:45AM |
4 |
Possible to display which extensions are in use on the phone's display? |
| 10:36AM |
0 |
chan_capi dialout problem |
| 9:57AM |
1 |
Polycom phone question |
| 9:52AM |
0 |
H.350 integration |
| 9:47AM |
4 |
Software SIP Phones |
| 9:38AM |
1 |
TDM400P callwaiting, threewaycalling and cancallforward problem |
| 9:36AM |
2 |
Firefly 1.9.5 and 20041117 CVS HEAD -- IAX2 one way audio |
| 8:30AM |
0 |
AP200B Phones |
| 8:09AM |
2 |
Max retries exceeded to host ... |
| 8:08AM |
0 |
AP200B or C |
| 7:14AM |
2 |
Port for Asterisk |
| 6:54AM |
1 |
IAX authenticated transfer |
| 6:12AM |
0 |
Russian Asterisk community |
| 5:50AM |
1 |
Compile error on spandsp-0.0.2-pre6 |
| 5:06AM |
1 |
Re: Asterisk-Users Digest, Vol 4, Issue 222 |
| 4:46AM |
0 |
Ringing tone on calls going out on chan_modem |
| 3:58AM |
1 |
TDM FXS Module & caller ID |
| 2:20AM |
0 |
Cannot create mysql database with TRABAS |
| 2:09AM |
1 |
Hardware selection |
| 1:24AM |
0 |
why dsp.c can not detect busytone? |
| |
| Tuesday November 16 2004 |
| Time | Replies | Subject |
|---|
| 10:00PM |
1 |
Connection of Asterisk to Cisco Callmanager via H.323 |
| 8:12PM |
2 |
Errors Compiling chan_capi 0.3.5 |
| 7:45PM |
0 |
Asterisk-Users Digest, Vol 4, Issue 222 (fwd) |
| 6:43PM |
1 |
IAX2 peers via MySQL DB with Asterisk 1.0.2 |
| 5:47PM |
1 |
sending faxes with asterisk in between |
| 5:14PM |
1 |
Using a Aastra/Nortel 390 Phone with Asterisk |
| 4:18PM |
1 |
Grandstream Dial Tone from PBX |
| 3:30PM |
0 |
no media for VM |
| 3:17PM |
2 |
RJ11 and Digium TDM 400P |
| 3:01PM |
0 |
TDM31B Interrupt Issue SOLVED! :-) |
| 2:43PM |
2 |
Recording from AGI playback is LOW |
| 2:36PM |
1 |
T405P Mulitiple Signalling modes on 1 card. |
| 2:12PM |
9 |
Variables |
| 1:59PM |
0 |
LookupCIDName - 1 vs "" |
| 1:31PM |
10 |
SS7 for * |
| 1:03PM |
1 |
RE: Sending DTMF Digits for DID |
| 1:03PM |
2 |
Interrupts failure on T100P |
| 12:51PM |
3 |
Dial by name |
| 11:48AM |
0 |
Suggestion for SIP video phone for windows CE |
| 11:48AM |
0 |
Newbie - NO Problems!!! - System Info |
| 11:41AM |
0 |
broadvoice connection error message |
| 11:26AM |
0 |
Timing Question:) (Loop/Internal etc). |
| 11:17AM |
0 |
SIP Video Conferencing System to PRI |
| 11:11AM |
0 |
Sending DTMF DID w/ Asterisk |
| 10:33AM |
3 |
SIP register problem |
| 10:31AM |
0 |
IAX2 unable to transfer? |
| 10:26AM |
2 |
Gaps in sound |
| 10:14AM |
0 |
Source for generic linksys phone adapter? |
| 10:03AM |
1 |
Zaptel Compile Problems with 1.0 Stable |
| 9:41AM |
2 |
Asterisk API Docs |
| 9:28AM |
1 |
Asterisk CLI access permissions? |
| 9:26AM |
1 |
Log extension in CDR when forwarding calls to another number |
| 9:00AM |
2 |
Newbe Question |
| 8:07AM |
2 |
Newbie - NO Problems!!! |
| 7:52AM |
2 |
TDM31B has no interrupts? |
| 7:35AM |
1 |
Asterisk with "chan_misdn" (in USA) |
| 7:23AM |
1 |
Using Asterisk as an external MOH for Televantage5? |
| 6:58AM |
0 |
Multi Lines in Asterisk |
| 6:08AM |
0 |
Snom and Stun |
| 6:04AM |
0 |
if NOT SipUser then Dial(Zap/1/${EXTEN}) |
| 5:44AM |
1 |
Capi Deflection (CD) not working |
| 5:29AM |
0 |
404 error found when making SIP point to point calls |
| 5:23AM |
2 |
Problem with sox |
| 4:36AM |
0 |
Agent channel problem |
| 4:35AM |
1 |
freebsd & voicemail everything seems to work?? |
| 4:05AM |
0 |
new version problem |
| 3:49AM |
2 |
Voicemail Digits |
| 3:40AM |
0 |
Unable to get Incoming Calls |
| 3:01AM |
0 |
backtracing ABANDON entries to CID in queue_log? |
| 1:09AM |
0 |
FXO ? |
| |
| Monday November 15 2004 |
| Time | Replies | Subject |
|---|
| 11:40PM |
0 |
MTA 3308 (Innomedia)ipphone does it work with asterisk |
| 10:25PM |
0 |
Asterisk queue |
| 9:31PM |
1 |
How to emulate a multiline phone in Asterisk |
| 8:26PM |
2 |
Is IAXTEL working? |
| 8:09PM |
1 |
OH323 and gatekeeper |
| 5:12PM |
3 |
Memory Consumption |
| 4:08PM |
4 |
Skype API release |
| 3:57PM |
5 |
Question about remote POTS lines |
| 3:46PM |
1 |
Measuring Bandwidth on T1 into * |
| 3:45PM |
2 |
VM Greeting |
| 3:36PM |
0 |
Using Asterisk as an external MOH for Televantage 5? |
| 3:35PM |
6 |
Standard messages instead of MOH during dial |
| 3:24PM |
0 |
Meetme and audio recording/playback |
| 3:23PM |
3 |
Auto dialout |
| 3:09PM |
1 |
ISDN, fax and bristuff |
| 2:33PM |
2 |
Problem with NAT on Asterisk 1.0.1 |
| 1:40PM |
3 |
ADSI questions for a 390 ADSI Phone |
| 1:40PM |
1 |
Traffic shaping script for kernel 2.6 and SIP? |
| 1:37PM |
0 |
Asterisk scalability IVR/Voicemail only |
| 1:35PM |
1 |
Asterisk and ISDN |
| 1:23PM |
4 |
$10 for G.729 ? |
| 1:01PM |
4 |
Broadvoice number always busy |
| 12:19PM |
3 |
Manager API Call Origination & Variables |
| 12:03PM |
1 |
MC3810 IOS |
| 11:54AM |
0 |
Avoiding 2 ring callerid delay for calls that don't go to voicemail |
| 11:54AM |
1 |
Multiple TDM400 vs T1 |
| 11:49AM |
2 |
Odd error at startup |
| 10:40AM |
1 |
Help with this debug output? |
| 10:21AM |
1 |
VICIDIAL in windows xp |
| 9:18AM |
1 |
TMD400 FXO <-> Nokia 32 GSM (Hangup Problems) |
| 9:06AM |
1 |
Transferring calls from a Zyxel P2000w |
| 9:04AM |
4 |
MYSQL Dialplan Question |
| 9:03AM |
1 |
FXO setup |
| 8:41AM |
1 |
IAX2 trunking - timing - ztdummy?? |
| 8:34AM |
2 |
Where can I find searchable version of this list? |
| 8:26AM |
0 |
NETDEV WATCHGOG eth0 timeout |
| 8:14AM |
0 |
(no subject) |
| 8:14AM |
0 |
Transfer # - Intermittent with Cisco 7905 SIP Phone |
| 7:57AM |
0 |
Multiple options to Dial command - what is the correct format? |
| 7:32AM |
0 |
irq CPU state |
| 7:31AM |
2 |
asterisk nagios plugin |
| 5:22AM |
0 |
Re: zap channel won't send/receive calls |
| 4:44AM |
0 |
iax preferred codec question? |
| 3:18AM |
2 |
PSTN -> Asterisk -> PSTN Call quality |
| 3:03AM |
0 |
Sip relay with asterisk |
| 1:45AM |
1 |
Meetme2 - web interface not working |
| 12:46AM |
1 |
AU FreeBSD PRI Hardware |
| 12:34AM |
0 |
SIP (or IAX) modem driver |
| |
| Sunday November 14 2004 |
| Time | Replies | Subject |
|---|
| 11:05PM |
1 |
AU PSTN Tone / Progress Detection |
| 8:32PM |
0 |
WAV file volume in voicemail - anyone actually solve this? |
| 8:30PM |
2 |
Linux Kernel 2.6 Questions - safe_asterisk and udev |
| 8:25PM |
2 |
ResponseTimeout problem |
| 7:38PM |
1 |
Service Providers With Caller ID Name?? |
| 6:47PM |
0 |
Hangup Phone |
| 4:16PM |
0 |
Snom 220 Problem |
| 4:06PM |
0 |
(no subject) |
| 3:57PM |
0 |
SIP Packets stuck in queue |
| 3:11PM |
0 |
asterisk & ser setup consulting needed |
| 2:52PM |
0 |
ERROR: retrans_pkt: Maximum retries exceeded on call |
| 1:51PM |
1 |
problem with zyxel prestige 2002 |
| 12:49PM |
1 |
3 - TDM31B Card Installation Difficulty |
| 12:40PM |
1 |
Asterisk using the wrong peer in sip.conf |
| 12:30PM |
2 |
Asterisk update |
| 12:25PM |
0 |
MacOS/x softphone and g729a |
| 11:02AM |
2 |
Voicemail shorter then (ex) 2sec - don't accept |
| 10:50AM |
7 |
Dial Plan Pattern Matching |
| 10:36AM |
3 |
SysMaster and GPL Violation (lets think before we jump) |
| 9:51AM |
0 |
Asterisk and Digium |
| 9:41AM |
0 |
Does Music On Hold not work on Debian??? |
| 7:41AM |
0 |
ODBC Message Waiting Indicator |
| 7:06AM |
0 |
How to route all incoming call to the defines context in extensions.conf |
| 6:39AM |
0 |
Elesign - ESC2420. |
| 6:24AM |
0 |
Garbled sound - CPU or traffic problem? |
| 6:21AM |
2 |
H323/*/IAX <-> Firewall <-> IAX/*/H323 |
| 4:30AM |
0 |
Asterisk-prepaid |
| 4:07AM |
0 |
AgentCallBackLogin and queue_log |
| 2:19AM |
11 |
(newbie) no dialtone on a TDM400P card |
| 1:42AM |
2 |
skinny error |
| 12:25AM |
1 |
re: DVG-1120 |
| |
| Saturday November 13 2004 |
| Time | Replies | Subject |
|---|
| 11:36PM |
0 |
Queue/AgentCallbackLogin Problems |
| 8:36PM |
3 |
Cisco ATA and G729 |
| 8:33PM |
0 |
my asterisk drops connection when remote side puts me on hold? |
| 6:14PM |
2 |
manager api: how to handle failed calls |
| 5:23PM |
1 |
Best setup for BudgeTone |
| 4:50PM |
2 |
isdn to sip gw |
| 4:13PM |
3 |
Remote answer not detected |
| 2:28PM |
1 |
spandsp problem |
| 11:57AM |
5 |
NAT |
| 10:11AM |
1 |
Cable for T1 connection: Crossover or straight through? |
| 9:03AM |
0 |
New TA from Uniden |
| 8:47AM |
2 |
wctdm to replaces wcfxs module ? |
| 8:43AM |
2 |
Broadvoice Patch issues |
| 4:58AM |
1 |
Cisco IP phones, SIP, Call-Manager & Contracts |
| 2:48AM |
1 |
SPA-3000 Wizard for Asterisk |
| 1:11AM |
5 |
Over 10,000 lines. Will asterisk manage? |
| 1:09AM |
2 |
Extension "follow me" |
| |
| Friday November 12 2004 |
| Time | Replies | Subject |
|---|
| 11:35PM |
1 |
random echo on TA750 |
| 9:34PM |
0 |
Cisco 7940 multiple line capability questions... |
| 7:20PM |
1 |
Advice on starting out |
| 6:48PM |
1 |
Calling an outside number along side other internal extensions? |
| 6:32PM |
1 |
Authenticate or DISA? |
| 6:26PM |
1 |
pressing a key to get out of voicemail? |
| 6:04PM |
1 |
voip to pstn |
| 5:01PM |
0 |
DECT channel |
| 4:51PM |
2 |
CNG Comfort Noise Generation |
| 4:04PM |
1 |
Need low-cost flat-rate incoming DID's throughout the U.S - Anybody competing against VoicePulse? |
| 3:06PM |
1 |
Kirk IP 600 DECT station |
| 2:43PM |
0 |
gold rush? |
| 1:50PM |
1 |
Can someone tell me what is going on from this debug? |
| 1:47PM |
0 |
Answer Confirmation "c" |
| 1:46PM |
0 |
ACD queue timeout problem |
| 1:10PM |
2 |
BRI in the US |
| 1:01PM |
1 |
Quick call group question... |
| 12:58PM |
0 |
Asterisk crashes after call when running as non-root, bug??? |
| 12:21PM |
1 |
Combination Cellular and WiFi/SIP |
| 11:12AM |
0 |
FW: Strange error |
| 11:02AM |
1 |
Asterisk Administration and Management requi rements (splinter from $200 AMP bounty thread) |
| 10:47AM |
1 |
Audio troubles on the Zyxel 2000w |
| 10:43AM |
1 |
SIP REGISTER -- Via 0.0.0.0:5060 -- Oooops?! |
| 10:42AM |
3 |
Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread) |
| 10:33AM |
1 |
SIP & ALERT_INFO for distinctive ring |
| 10:16AM |
5 |
Strange error |
| 10:06AM |
2 |
$200 AMP documentation bounty < - Comments o n the Linux user experience |
| 9:58AM |
0 |
Faster g726 and ADPCM |
| 8:50AM |
0 |
Strange Behavior, static and clicking on outbound calls only. |
| 8:43AM |
0 |
Motherboard whitelist (was Echo - UK Impedan ce problem with X100P?) |
| 8:32AM |
0 |
Ring after hangup with Rhino Channel Bank |
| 8:28AM |
2 |
The BV patch: Some notes |
| 8:21AM |
1 |
CDR & MySQL Problem |
| 8:19AM |
4 |
OT: Grandstream problems |
| 8:18AM |
0 |
DID/PRI sending to the s, extension <-solved it |
| 8:11AM |
2 |
Motherboard whitelist (was Echo - UK Impedance problem with X100P?) |
| 7:59AM |
1 |
Lock the phone when no using it |
| 7:32AM |
3 |
Cisco 7912g SIP firmware |
| 7:27AM |
8 |
$200 AMP documentation bounty |
| 7:26AM |
0 |
Asterisk behind external PBX +enable IVR |
| 7:21AM |
0 |
astGUIclient - 1.0.4 (Running in Windows) an d SQL Updater Down |
| 6:53AM |
2 |
Caller ID for Japan? |
| 6:45AM |
1 |
astGUIclient - 1.0.4 (Running in Windows) and SQL Updater Down |
| 6:44AM |
1 |
Conferencing needs Zaptel ?? |
| 5:51AM |
5 |
Echo - UK Impedance problem with X100P? |
| 5:40AM |
2 |
timeout |
| 5:39AM |
3 |
Calling h@ and Loop Detected |
| 5:08AM |
0 |
Continuing a call to callee after caller has hung up. |
| 4:30AM |
1 |
Siemens voip adapter |
| 3:51AM |
0 |
attempting native bridge error |
| 3:12AM |
0 |
SIP Register with Huawei equipment HELP |
| 2:54AM |
3 |
Dial without bridge |
| 2:37AM |
0 |
SIP clients <--> SE R <--> Asterisk <--> carrier/gateway |
| 2:33AM |
1 |
No ringing with Phonejack Lite - hardware or software problem? |
| 2:06AM |
1 |
Install X-lite automatic with (windows) .ins file |
| 1:40AM |
1 |
Recent * SRPMS |
| 1:10AM |
1 |
wcfxs module gone from CVS head? |
| 12:47AM |
0 |
How to see if I have PCI 2.2 |
| |
| Thursday November 11 2004 |
| Time | Replies | Subject |
|---|
| 11:38PM |
0 |
New Zealand Centrex Service |
| 10:45PM |
7 |
SysMaster and GPL Violation |
| 8:14PM |
0 |
SIP distinctive ring (BroadVoice) |
| 7:49PM |
0 |
SIP no working in 1/4 installations |
| 7:34PM |
1 |
TDM400p module error? |
| 4:53PM |
0 |
DID/PRI sending to the s, extension <-more i nformation |
| 3:59PM |
1 |
DID/PRI sending to the s, extension |
| 3:41PM |
1 |
FXO dialing - all lines dial but one |
| 3:23PM |
0 |
Problems compiling chan_capi with latest CVS |
| 3:09PM |
3 |
ive noticed that our 1.02 stable box's asterisk is taking 100% cpu load.. |
| 3:09PM |
1 |
DHCP from server A and connect to server B messes with SIP call out. |
| 2:26PM |
1 |
ZT_CHANCONFIG failed on channel 1: No |
| 2:19PM |
2 |
DID/PRI sending to the s, extension instead of t he DID extension |
| 2:12PM |
1 |
sometimes problem with dialing ZAP channel |
| 2:02PM |
15 |
Can some bady help me ??? |
| 1:39PM |
0 |
One way audio on calls across a TDM400P |
| 1:07PM |
0 |
Cisco 79XX phone using dhcp can call out but not in |
| 12:55PM |
3 |
Deploying multiple Sipura 3000s with Asterisk |
| 12:39PM |
2 |
ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 12:06PM |
0 |
broadvoice patch and 16 second re-registers |
| 12:02PM |
1 |
Zaptel module load errors under stock FedoraCore 2 (2.6.8-1.521 kernel ) |
| 11:32AM |
0 |
Problem using Digi DataFire Micro V |
| 11:14AM |
3 |
Palm Tungsten and Asterisk |
| 11:00AM |
1 |
Grandstream BT100 - No Sound with Playback() |
| 10:58AM |
4 |
Snom 190/220 dialplan strings? |
| 10:41AM |
0 |
astGUIclient Problem -- http://10.10.10.15/a stguiclient/admin.php |
| 10:28AM |
1 |
astGUIclient Problem -- http://10.10.10.15/astguiclient/admin.php |
| 10:10AM |
0 |
Preventing Call Forwarding by SIP UA |
| 9:53AM |
1 |
setup of cisco 7960 phone tftp asking for unkownfile |
| 9:46AM |
0 |
working Marconi sys X config |
| 9:42AM |
3 |
setup of cisco 7960 phone tftp asking for unkown file |
| 8:40AM |
0 |
Special Characters In Passwords |
| 8:07AM |
6 |
cisco poe |
| 7:40AM |
2 |
Monitor/Record MeetMe Conversations |
| 7:19AM |
1 |
failed to go to next dial command |
| 6:51AM |
0 |
tdm04b outbound call question |
| 6:28AM |
3 |
Multiple NIC's on * box? |
| 6:21AM |
1 |
"Distributed" registration SIP/IAX2 |
| 5:47AM |
1 |
asterisk & xlite codecs |
| 5:46AM |
0 |
Several Problems with PhoneJack |
| 5:45AM |
1 |
Asterisk DNS issue |
| 5:45AM |
0 |
Problems in autnenticating with SER / PortaSIP |
| 3:38AM |
1 |
asterisk support for ISDN 1TR6 ? |
| 1:56AM |
6 |
Top posting |
| 1:52AM |
2 |
No SIP registration but user has dialled out?!? |
| 1:51AM |
0 |
TDM400P / FXO / Polarity Reversal |
| 1:49AM |
1 |
Grandstream BugeTone 101 - Multi-Server setup ??? |
| 1:06AM |
0 |
Frequency Shift |
| |
| Wednesday November 10 2004 |
| Time | Replies | Subject |
|---|
| 11:48PM |
3 |
No Inbound CallerID Name Has me Stumped. |
| 11:35PM |
0 |
Broadvoice Problems.- |
| 9:25PM |
2 |
Aastra/Sayson 480i eval |
| 8:51PM |
1 |
Connecting to Exicom GSX 418/816 |
| 6:31PM |
3 |
Zaptel module load errors under stock Fedora Core 2 (2.6.8-1.521 kernel ) |
| 4:58PM |
0 |
Sip Phone UIP200 Accepts calls but dialing out fails |
| 4:49PM |
1 |
DTMF and Access Codes |
| 4:28PM |
1 |
Callerid is recieved by fxo, but sometimes not passed to extensions |
| 4:24PM |
0 |
AgentCallBackLogin and accepting call using # |
| 3:32PM |
4 |
NoOp |
| 3:15PM |
1 |
Problem flashing zap channel. |
| 3:02PM |
1 |
Sending SMS from ISDN to cellular |
| 2:17PM |
0 |
Analog calls not working |
| 1:57PM |
1 |
No sound with kphone 4.05 on SuSE 8.2 and asterisk |
| 1:29PM |
1 |
Voicemail and MySQL 4.1.x |
| 1:26PM |
3 |
Hooking up a an Adit 600 |
| 12:58PM |
1 |
Broadvoice Patch |
| 12:54PM |
5 |
Broadvoice asterisk patch |
| 12:51PM |
0 |
Problem adding zaprtc to Asterisk CVS on debian sarge |
| 12:19PM |
1 |
iconnect incoming problems |
| 12:05PM |
1 |
GTW V.92 modem work with asterisk? |
| 10:51AM |
0 |
IAXy Call Transfer and X100p audio quality in UK |
| 10:25AM |
7 |
xlite and asterisk |
| 10:16AM |
0 |
the asterisk work with modem generic? |
| 10:04AM |
0 |
Voicemail Outcall Notification App Ready to test |
| 10:01AM |
1 |
Unknown RTP Codec when sending fax |
| 9:50AM |
1 |
asterisk PC hardware reccomendations? |
| 9:27AM |
4 |
Pause during dial |
| 8:12AM |
0 |
SELinux and Asterisk |
| 7:32AM |
4 |
Asterisk, X-Lite, and * and # keys |
| 7:10AM |
0 |
Amount of time asterisk take to pickup incom ing call on ZAP interface |
| 7:00AM |
1 |
Web tool for Connection History |
| 5:04AM |
1 |
4 port ISDN BRI pci card |
| 3:28AM |
0 |
HELP: Asterisk becomes zombie process ... |
| 3:18AM |
1 |
Call failover and redundancy |
| 2:31AM |
2 |
maximum retries error |
| 12:20AM |
0 |
register problem of iaxcomm |
| 12:16AM |
1 |
remove channels |
| |
| Tuesday November 9 2004 |
| Time | Replies | Subject |
|---|
| 9:34PM |
1 |
Asterisk-OH323 OUTCODEC |
| 8:21PM |
3 |
processing power / codecs |
| 7:48PM |
2 |
Auto dial Out |
| 7:15PM |
0 |
Queue Optional URL Problem |
| 5:51PM |
2 |
DISA() context restrictions |
| 5:30PM |
0 |
Problem with agentcallbacklogin and hitting # to accept call |
| 4:49PM |
0 |
TDM04B and T100P driver loading issue |
| 4:03PM |
5 |
Digium Generic Boards - Low Prices / High Quality. |
| 3:36PM |
0 |
Monitor on Asterisk´s Manager API |
| 3:21PM |
1 |
External call initiation |
| 3:02PM |
1 |
linphone |
| 2:20PM |
5 |
E100P - Generic (Clone) - :) |
| 2:13PM |
1 |
Old Dialogic Hardware Questions |
| 2:10PM |
4 |
quasi-skype channel for Asterisk? |
| 1:22PM |
2 |
X100P CLONES again |
| 1:16PM |
3 |
Voicemail questions |
| 1:04PM |
0 |
Segmentation fault on SIP inbound |
| 10:36AM |
1 |
looking for BKW |
| 10:07AM |
0 |
Broken H323 channel |
| 8:50AM |
1 |
Enquiry about Wildcard E100P card |
| 8:24AM |
0 |
Queue Behavior. |
| 8:01AM |
2 |
New Release Asterisk-Stat V 1.3 |
| 7:56AM |
1 |
Zaptel makefile error/bug? |
| 7:05AM |
2 |
Marconi Sys X/TE410P configuration |
| 7:02AM |
2 |
UK CID patch and version 1.0 CVS build |
| 6:24AM |
0 |
(no subject) |
| 6:05AM |
0 |
Intel IPP installation |
| 5:41AM |
3 |
UK BT Caller ID, X100P and Asterisk v1 |
| 5:16AM |
1 |
Alcatel IP Phone |
| 5:11AM |
2 |
Costum ring tones with BT10x |
| 4:47AM |
0 |
DIALEDPEERNUMBER and Queues bugged? |
| 4:23AM |
0 |
how to detect busy tone? |
| 2:21AM |
0 |
Linksys / Cisco does not support the PAP2-NA |
| 1:29AM |
1 |
WRT54GP2 (WiFi + ATA) |
| 12:50AM |
0 |
How to connect Siemens Combiset to Asterisk - fxo or fxs ? |
| 12:49AM |
0 |
X100P, Caller Id and Ireland |
| 12:44AM |
0 |
Running Asterisk in chroot environment ? |
| |
| Monday November 8 2004 |
| Time | Replies | Subject |
|---|
| 11:38PM |
3 |
Faxing issues (no VoIP involved) |
| 10:56PM |
3 |
NAT setup |
| 10:53PM |
2 |
Cisco Unity and Asterisk |
| 10:24PM |
1 |
Change log available? |
| 7:38PM |
0 |
FC3 and udev troubles |
| 5:43PM |
0 |
x100p drive use Tone-based Supervisory Disconnect? |
| 5:21PM |
1 |
bad quality for toll free calls with gafachi |
| 4:48PM |
0 |
IVR functionality Any Idea's how to implement this? |
| 4:03PM |
1 |
IAX2 One way audio PSTN via Gafachi |
| 4:00PM |
1 |
SpanDSP + Lexmark 6170 = Cut off faxes? |
| 4:00PM |
1 |
txfax problem? |
| 3:47PM |
1 |
IAX and ADSI Help |
| 3:40PM |
1 |
CallerID+Distinctive ring in Australia |
| 3:31PM |
0 |
RPMS for Fedora Core 2 now available |
| 3:21PM |
1 |
sip trunking works? |
| 3:02PM |
1 |
Polycom 600 as a Receptionist Phone |
| 2:40PM |
1 |
new RH9 install - no playback audio? |
| 2:40PM |
2 |
calls go silent |
| 2:01PM |
0 |
Xten Video Softphone Gets IM, Presence |
| 1:59PM |
3 |
how to get Stable 1.X via CVS |
| 1:43PM |
5 |
Same Extensions in Multiple contexts |
| 1:08PM |
1 |
iPeya iPHONE-1001M? |
| 12:20PM |
3 |
MWI Doesn't Turn Off |
| 12:19PM |
2 |
Configuring Asterisk As A Sip Server |
| 12:19PM |
2 |
Cordless vs Wireless phones |
| 11:53AM |
0 |
TDM400P card on Mac dialtone problem |
| 11:41AM |
2 |
Voicemail Macro issue. |
| 11:11AM |
0 |
timing and dropped calls |
| 10:44AM |
1 |
Sort of OT: Grandstream Phone and MS Wireless mouse |
| 10:23AM |
0 |
Snom 220 (or other phones) - line |
| 10:23AM |
0 |
Setting DND feature via access code |
| 10:22AM |
1 |
FW: Need a creative solution - Caller ID and a stupidupstream |
| 10:22AM |
0 |
FW: Need a creative solution - stop forwarding from changing caller ID |
| 9:58AM |
0 |
Error forwarding calls to Voicemail from SER |
| 9:46AM |
0 |
Zap FXO channel locked up with steadystatic( white noise) |
| 8:49AM |
0 |
ZyXEL 2000w unregistering and no audio |
| 8:12AM |
0 |
Free World Dialup via IAX2 gives duplicate calls? |
| 6:54AM |
0 |
Quintum vs Asterisk |
| 5:46AM |
2 |
Setting jitterbuffer in with iax |
| 4:12AM |
0 |
Help on "Supervised Call Transfer" |
| 4:00AM |
5 |
AGI Errors |
| 2:06AM |
1 |
re: CallerID for the UK |
| 1:57AM |
0 |
Cisco 1751-V SIP Gateway for Asterisk |
| 1:51AM |
0 |
Have anyone try to use asterisk as a business mode |
| 1:34AM |
1 |
Astricon Brazil. Why not ?! |
| 1:14AM |
0 |
Re: [Asterisk-Dev] Illegal Instruction (Solved) |
| |
| Sunday November 7 2004 |
| Time | Replies | Subject |
|---|
| 11:59PM |
1 |
Aterisk and ISDN |
| 11:24PM |
0 |
Problem with call originating from Cisco |
| 10:17PM |
0 |
how to get CallerId info for call originated from manager API |
| 10:00PM |
1 |
New bounty for voicemail outcall notification- add $$ if interested |
| 9:16PM |
0 |
New bounty for voicemail outcall notification -add $$ if interested |
| 8:59PM |
3 |
Point to Point VOIP |
| 8:29PM |
1 |
Zap FXO channel locked up with steadystatic(white noise) |
| 8:06PM |
2 |
New bounty for voicemail outcall notification - add $$ if interested |
| 7:36PM |
1 |
openhours - include contexts based on time and date |
| 6:17PM |
2 |
Snom 220 (or other phones) - line apperances? |
| 4:28PM |
5 |
getting callerid from spa3k to asterisk |
| 4:23PM |
0 |
Cisco Unity + Asterisk |
| 3:09PM |
1 |
Zap FXO channel locked up with steady static (white noise) |
| 2:08PM |
0 |
Need help from the USB phone owners |
| 1:31PM |
3 |
CallerID Name from SIP to IAX2 |
| 1:30PM |
4 |
"night" mode ideas |
| 1:14PM |
1 |
SMS through Cisco PSTN GW |
| 12:17PM |
3 |
Queue announce behavior for callback agents? |
| 12:03PM |
1 |
Forward incoming SIP calls to H323 ipphone? |
| 11:53AM |
2 |
Clipping at start of call |
| 11:37AM |
1 |
Unable to create channel of type Zap! |
| 10:39AM |
4 |
MAX TNT |
| 9:20AM |
3 |
No busy-tone |
| 6:45AM |
2 |
Siemens GSM terminal with Wildcard FXO |
| 6:06AM |
1 |
FreeBSD asterisk and zaptel versions |
| 6:01AM |
1 |
zaptel (ztdummt) compilation problems |
| 5:39AM |
1 |
CVS RPMs for Mandrake 10 (Zaptel and, Asterisk) |
| 5:08AM |
3 |
press # to execute |
| 4:06AM |
3 |
Problem with call originating from Cisco 7940 SIP phone to a SIP peer |
| 1:15AM |
0 |
ADTRAN 850 and T100P - need some help! |
| 12:22AM |
2 |
I don't know the name of this feature... |
| |
| Saturday November 6 2004 |
| Time | Replies | Subject |
|---|
| 11:45PM |
0 |
how to establish a queue for external agents (was:Need a dial plan as follows) |
| 8:23PM |
5 |
SIP Groups |
| 8:21PM |
0 |
Adit 600 and T100P echo from VOIP clients |
| 7:42PM |
1 |
SIPURA does not register with Asterisk |
| 6:56PM |
1 |
Asterisk X100p can not hangup |
| 6:04PM |
4 |
Need a dial plan as follows |
| 5:46PM |
4 |
Enhanced Audio Support for EAGIs |
| 5:38PM |
1 |
Caller-id |
| 5:33PM |
2 |
Passwords in extra include file |
| 4:30PM |
1 |
fax and echo cancel |
| 1:03PM |
5 |
* does not listen to DTMF during wait ? |
| 11:47AM |
4 |
Polycom 500 software? |
| 11:27AM |
1 |
astGUIClient |
| 7:59AM |
0 |
Giving users the ability to break out of thequeueand go to voicemail |
| 6:35AM |
2 |
Setting up a Fritz AVM PCI card |
| 6:28AM |
0 |
Call Park Bug |
| 3:16AM |
0 |
group limit |
| 1:42AM |
1 |
missing wakeup gsm files |
| 1:41AM |
1 |
Giving users the ability to break out of thequeue and go to voicemail |
| 12:11AM |
1 |
Analog to Digital |
| |
| Friday November 5 2004 |
| Time | Replies | Subject |
|---|
| 10:59PM |
1 |
SIP REGISTER -- Asterisk non-compliant or is it the provider? |
| 10:56PM |
2 |
Giving users the ability to break out of the queue and go to voicemail |
| 8:34PM |
4 |
[OT] Old Building Needs a New Telephone System |
| 7:39PM |
1 |
Grandstream BT100 Message Button |
| 7:05PM |
0 |
X100P Clone - Can't load moddule |
| 5:59PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group. |
| 5:32PM |
1 |
R: sip.conf extensions.conf |
| 4:21PM |
4 |
Cisco 7970 & Firmware for the 7960G |
| 3:06PM |
1 |
chan_zap.c unable to register channel |
| 2:58PM |
0 |
Telephone Call Voicemail Notification |
| 2:46PM |
1 |
Record() help |
| 2:45PM |
0 |
Vovivda.org & VOCAL & Asterisk |
| 2:14PM |
0 |
asterisk + hotel ? |
| 1:58PM |
2 |
Newbie X100P Clone question |
| 1:54PM |
0 |
Questions from an Asterisk newbie - follow-up question. |
| 1:45PM |
0 |
Audiocodes FXO MP104 |
| 1:45PM |
0 |
weird problem with outgoing calls using chan_CAPI |
| 1:41PM |
0 |
Need a creative solution - Caller ID and a stupid upstream |
| 1:27PM |
0 |
Queue only allowing 1 call |
| 1:16PM |
2 |
res_config problems |
| 12:59PM |
1 |
Max retries exceeded with voiceconnect |
| 12:55PM |
2 |
Asterisk Brazillian Community |
| 12:48PM |
0 |
warning: implicit declaration of function `__use_ast_pthread_create_instead__' |
| 12:15PM |
3 |
BudgetTone 100 + NuFone |
| 11:45AM |
3 |
Questions from an Asterisk newbie |
| 11:41AM |
4 |
Adjusting txgain/rxgain |
| 11:31AM |
1 |
Are softphones usable? |
| 11:01AM |
0 |
Wrong return ext from call park? |
| 10:53AM |
1 |
Polycom IP 300 VoiceMail Retrieval |
| 10:26AM |
1 |
unable to create channel of type Zap |
| 9:10AM |
0 |
Asterisk As a Callback Server and Message Dialout Server - Can be Linked to ASTCC |
| 9:05AM |
2 |
VoiceMailMain(s<exten>@<context>) doesn't |
| 9:03AM |
3 |
wcfxs module doesn't load |
| 8:50AM |
1 |
Messanger 6.2 with Asterisk |
| 8:47AM |
3 |
sip.conf extensions.conf |
| 7:59AM |
2 |
Problems with voicemail |
| 7:31AM |
1 |
VoiceMailMain(s<exten>@<context>) doesn't work in CVS 11/03 |
| 7:05AM |
0 |
Cisco 1751-V as SIP Gateway for Asterisk |
| 6:31AM |
1 |
german patches for say.c |
| 6:07AM |
0 |
Transcoding - when and when not? |
| 4:49AM |
2 |
New-B-ish Question |
| 4:20AM |
2 |
Service numbers |
| 2:35AM |
0 |
Sip Error Message, pbx.c: 1938 |
| 2:06AM |
1 |
Asterisk incoming calls |
| 12:42AM |
1 |
voicemail&ilbc |
| 12:04AM |
0 |
Fw: Snom 190/220 |
| |
| Thursday November 4 2004 |
| Time | Replies | Subject |
|---|
| 11:04PM |
0 |
Problem In RTC Client With Asterisk |
| 9:59PM |
2 |
RIM Blackberry WLAN SIP phone |
| 9:45PM |
0 |
Using a Vonage Softphone |
| 9:44PM |
1 |
example Monit control file |
| 6:08PM |
0 |
oh323 0.7.0 don't start |
| 5:36PM |
1 |
ICD status |
| 5:16PM |
1 |
TDM400P and some problems |
| 3:13PM |
0 |
4-port T1 and TDM400 w/FXS in the same chassis problem |
| 2:45PM |
1 |
remote hold. |
| 2:28PM |
4 |
Looking for a SQL or ODBC Application |
| 2:13PM |
1 |
AstLinux posted for testing |
| 1:51PM |
0 |
RE: ZapTel problems ***** Problem solved ***** |
| 1:31PM |
0 |
Asterisk Manager PHP Class |
| 1:18PM |
1 |
FW: ZapTel problems |
| 1:12PM |
1 |
7940/7960's 'talking' through speaker when in headset mode? |
| 1:10PM |
0 |
Light reading SIP webinar |
| 1:08PM |
2 |
T100P <-> Merlin Legend 100D not working |
| 1:02PM |
0 |
PHP AGI and system call weird behaviour |
| 12:59PM |
0 |
Alcatel Enterprise |
| 12:40PM |
1 |
Is it possible to use IAXY device to make 56Kmodem calls |
| 12:35PM |
1 |
Call Leg/Transaction Does Not Exist |
| 12:30PM |
2 |
NAT with Linksys |
| 12:19PM |
0 |
MEETME and PRIORITIES |
| 12:02PM |
0 |
Remote MWI (I know it's possible) |
| 11:55AM |
1 |
Call Leg/Transaction Does Not Exist" back |
| 11:53AM |
3 |
Grandstream BT100 - Does not recognize DTMF |
| 11:50AM |
2 |
Is it possible to use IAXY device to make 56K modem calls |
| 11:28AM |
2 |
Passing callerID info to a forwarded line |
| 11:07AM |
1 |
Asterisk and ISDN HFC-S card (Biilion) instead of Fritz Capi ? |
| 11:04AM |
0 |
Grandstream BT100 - Failed to write frame |
| 10:56AM |
8 |
ATCC - Astcc-Admin.cgi File |
| 10:35AM |
0 |
OT: anyone using pointone? |
| 10:12AM |
1 |
Newbie question: forwarding call from PSTN to VoIP |
| 10:11AM |
3 |
system errors |
| 9:39AM |
3 |
Best Linux base for small Asterisk server? |
| 9:36AM |
0 |
avm fritz box fon |
| 9:31AM |
2 |
chan_capi patch : fax support |
| 8:56AM |
3 |
Limit DTMF tones |
| 8:51AM |
1 |
IAX --> SIP DTMF |
| 8:49AM |
0 |
Here's a tough question |
| 8:22AM |
1 |
Cisco 7910 - Success? |
| 8:18AM |
1 |
Multi-line analog phones with Asterisk? |
| 8:10AM |
8 |
Hardware Support |
| 7:48AM |
0 |
Asterisk 1.0.2-CVS RPM update |
| 7:47AM |
0 |
Perl AGIs & TCP Sockets |
| 7:44AM |
1 |
real-time-clock & asterisk/meetme/ztdummy in 2.6.9 UML |
| 7:34AM |
0 |
CISCO IP Conference Station |
| 7:33AM |
2 |
Multiline (4 or 8) sip phone |
| 7:31AM |
1 |
X100P & Analog PBX - not RING and not answer |
| 7:15AM |
0 |
Video conferencing Meet Me Bounty bumped |
| 7:14AM |
1 |
CVS-HEAD-11/03/04-14:09:34 ALERT_INFO Doesn't Get Passed |
| 7:00AM |
0 |
h323 & dundi problems with 11/04/04 CVS |
| 7:00AM |
1 |
supposable timing problem with TE100P |
| 6:50AM |
1 |
BROADVOICE fails to register |
| 6:47AM |
2 |
what do I ask my provider for when using e&m_w and a T100P? |
| 5:56AM |
1 |
res_config / realtime? |
| 5:56AM |
2 |
G.729 and Voicemail |
| 5:17AM |
1 |
control of calls |
| 5:17AM |
1 |
sipura 2000 flash ? |
| 4:57AM |
1 |
Howto correctly identify the telephone area code? |
| 4:05AM |
0 |
chan_capi on top of mISDN with HFC-8s |
| 3:31AM |
0 |
Voicemail, Cisco and H.323 problems |
| 2:57AM |
0 |
asterisk sip disabled error |
| 2:01AM |
0 |
Capi echo problems solved |
| 1:38AM |
3 |
res_features.so Segmentation fault |
| 1:14AM |
0 |
SIP phones, Asterisk and bandwidth |
| 12:58AM |
3 |
Dynamic DNS causes problems |
| 12:10AM |
2 |
asterisk as sip proxy registrar |
| |
| Wednesday November 3 2004 |
| Time | Replies | Subject |
|---|
| 11:22PM |
1 |
MusicOnhold on Bridged calls "plain text" |
| 9:44PM |
0 |
Little help here... |
| 9:12PM |
0 |
asterisk can not hangup .usrWildcard X100P |
| 9:00PM |
3 |
Voicemail Mailbox Configuration |
| 8:54PM |
0 |
SER-->Asterisk-->GNUGK Accounting Problem |
| 7:45PM |
0 |
Re: Re: [Serusers] asterisk can not hangup .user Wildcard X100P |
| 6:50PM |
1 |
Asterisk X100P doesnot Hangup |
| 5:16PM |
1 |
SIP registration/dialing problem. |
| 4:01PM |
1 |
Cisco 79XX - Using built-in 3way conference |
| 3:40PM |
2 |
How change default law for T100P |
| 3:32PM |
3 |
What do I need to ask my T1 supplier? |
| 3:27PM |
0 |
MusicOnhold on Bridged calls |
| 2:59PM |
0 |
Hookflash with cisco 827-4v |
| 2:22PM |
4 |
Sip clients not longer registering |
| 2:12PM |
0 |
G.729 for Asterisk: new version released |
| 1:41PM |
2 |
Automatically restart asterisk if not running |
| 1:09PM |
2 |
Dropped calls with analog lines using TDM400P |
| 12:37PM |
3 |
problem facing on Firewall, NAT and asterisk |
| 12:11PM |
1 |
SIPGate for outgoing calls |
| 11:34AM |
5 |
FireFly Problems |
| 11:27AM |
0 |
manager api originate doesn't give detailed information |
| 11:19AM |
0 |
RE: IAXys or IAX Softphones cannot call SIP phones |
| 11:02AM |
1 |
Installing X100P Asterisk - Unable to create channel of type 'Zap' |
| 10:52AM |
1 |
addon_mysql_cdr allows fraud by sip or iax users |
| 10:44AM |
1 |
Speed Dial / New Context |
| 10:40AM |
2 |
Asterisk's Fails to start! |
| 10:34AM |
0 |
ASTCC - cdrs database and number-entry timeout questions |
| 10:33AM |
1 |
asterisk port problem? |
| 10:17AM |
0 |
SendDTMFthrough the manager |
| 10:06AM |
0 |
can i call my local phone to IP phone or vice versa |
| 9:53AM |
0 |
Remote MWI |
| 9:29AM |
1 |
oh323 compilation error |
| 9:11AM |
3 |
Good ringing plans for small office |
| 9:01AM |
1 |
Call pickup and snom phones |
| 8:43AM |
0 |
Configuring MTA-V102 through TFTP, HTTP, HTTPS for Asterisk |
| 8:30AM |
3 |
zt hook failed: Device or resource busy |
| 8:26AM |
0 |
launching urls from queues |
| 8:24AM |
1 |
Maddog weighs in on the state of the Linux [Asterisk plug] |
| 8:13AM |
9 |
An anniversary and a lament for FXOs |
| 8:08AM |
0 |
No ISA tormenta card found at d0000 |
| 7:43AM |
0 |
asterisk-oh323: New versions now available! |
| 7:38AM |
1 |
Latest CVS voicemail<->mysql problem |
| 7:36AM |
2 |
chan sip error |
| 7:32AM |
0 |
Voicemail: howto disable vm-intro.gsm at the endof message? |
| 7:24AM |
2 |
Voicemail: howto disable vm-intro.gsm at the end of message? |
| 6:36AM |
2 |
asterisk as a sip registrar and user accounts |
| 6:21AM |
1 |
H323 ISDN |
| 5:57AM |
2 |
Asterix-to-PBX |
| 4:07AM |
1 |
Console Error message |
| 2:42AM |
2 |
H323 Compilation |
| 2:32AM |
1 |
No outgoping calls with ISDN |
| 2:15AM |
0 |
Cisco 2600 Gatekeeper registrations |
| 12:45AM |
3 |
Reject a call if no callerID |
| 12:43AM |
2 |
Using T100P on E1 line |
| |
| Tuesday November 2 2004 |
| Time | Replies | Subject |
|---|
| 8:46PM |
1 |
marginal voicemail prompt sound quality |
| 6:03PM |
0 |
reboot polycom via sip message |
| 5:52PM |
0 |
Dropping last digit when dialling from analogue phone. |
| 5:37PM |
1 |
dialing from mexico mapping numbers. |
| 4:26PM |
3 |
FXO devel Kit Card |
| 4:21PM |
2 |
gastman - documentation? |
| 4:13PM |
0 |
E&M timing |
| 3:58PM |
4 |
FXO module in TDM400P (UK, BT) - Hangup detection failing |
| 3:05PM |
0 |
agents can't hear callers. |
| 2:43PM |
1 |
anyone got a 7910 to work with asterisk? |
| 2:14PM |
1 |
IAX between two * |
| 2:09PM |
1 |
Notification of missed calls |
| 2:00PM |
1 |
The best SIP HW Phone and WLAN Phones for Asterisk |
| 1:57PM |
2 |
Outgoing call fails on pulse dial line |
| 1:50PM |
3 |
Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash |
| 1:40PM |
4 |
ISDN Dialplan |
| 1:16PM |
0 |
Asterisk Hanging! |
| 12:48PM |
2 |
Tone while ringing another IAX Phone |
| 12:45PM |
5 |
MAX TNT SIP / Asterisk |
| 12:16PM |
0 |
Remote Office question, Draytek , recommende d analog phone |
| 12:11PM |
0 |
isdn to isdn data call (bristuff'ed with hfc based card) |
| 12:00PM |
1 |
Fw: Re: How far is IAX to be a Standard |
| 11:59AM |
1 |
Polycom IP-500 Network Problems |
| 11:27AM |
0 |
Multi Freq signalling..key pulse...stop pulse....e&m..... |
| 11:04AM |
1 |
Anyone have bristuff's zaphfc module coexisting with wcfxs for the tdm400p? |
| 10:29AM |
3 |
Best codec for faxes? |
| 10:08AM |
1 |
Remote Office question, Draytek , recommended analog phone |
| 9:53AM |
3 |
FXO Module Error |
| 9:52AM |
2 |
Asterisk refuses to use anything but g729 |
| 9:51AM |
7 |
Zaptel Issue in Fedora Core 2 test 3 |
| 9:23AM |
0 |
Calling any Linksys PAP2-NA users... |
| 9:15AM |
1 |
(no subject) |
| 9:12AM |
1 |
Problems with CISCO, SIP and Asterisk |
| 8:51AM |
2 |
Broadvoice with multiple numbers |
| 8:51AM |
0 |
TDM11B auto configuration issue |
| 8:49AM |
4 |
Wireless VOIP Phone suggestions |
| 8:21AM |
0 |
RE: Question--Eezee phone? |
| 8:06AM |
3 |
Allied Telesyn Residential Gateway 613 |
| 7:44AM |
0 |
g729 passthrough |
| 7:32AM |
1 |
Quintum Tenor DX |
| 7:26AM |
2 |
IAX2 audio problems but SIP OK? |
| 7:20AM |
1 |
FW: ASTCC with password |
| 7:18AM |
2 |
OpenSource Proxies ?. |
| 7:12AM |
0 |
iax bracking up |
| 6:31AM |
3 |
Speech to Text Conversion |
| 6:22AM |
0 |
Unable to include a context in another context for some unknown reason |
| 6:16AM |
3 |
Reading extensions from MySQL database |
| 6:13AM |
0 |
* Sunday News |
| 6:06AM |
0 |
ANSWEREDTIME and DIALEDTIME |
| 5:45AM |
1 |
Enhancing list quality? |
| 4:22AM |
1 |
Urgent handler |
| 3:42AM |
2 |
prioritising codecs per user? |
| 3:02AM |
2 |
ISDN Capi Drivers HELP |
| 2:49AM |
2 |
Unable to get our IP address, Skinny disabled |
| 2:30AM |
2 |
Motherboard compatibility with 6 PCI slots for TDM04B |
| 2:17AM |
0 |
DTMF from TE410P to SIP devices doesn't work |
| 2:17AM |
1 |
Codecs and echo |
| 1:41AM |
0 |
Asterisk terminating VoIP over 20 E1. |
| |
| Monday November 1 2004 |
| Time | Replies | Subject |
|---|
| 11:59PM |
3 |
Hold music while ringing |
| 11:12PM |
1 |
soxmix? |
| 8:29PM |
1 |
MOH whilst waiting for Conference attendees |
| 8:26PM |
1 |
tdm410 driver prevents files being played |
| 6:56PM |
2 |
Is there a way to disable call wating? |
| 5:54PM |
1 |
astcc configure |
| 4:57PM |
0 |
anyuser i-3100 |
| 3:37PM |
1 |
Queue Prioritization |
| 2:55PM |
0 |
'Unregistered Channel Type' when parsing zapata. conf on * startup |
| 2:34PM |
0 |
ASTCC - Anyone has a Dial Plan that is working? |
| 2:30PM |
6 |
calling an iaxy |
| 2:21PM |
2 |
Directory app and extension |
| 2:16PM |
0 |
Asterisk 1.0.2 changes |
| 1:41PM |
2 |
Asterisk & NetCentrex CCS integration |
| 1:38PM |
0 |
One way audio, h.323 cisco call manager |
| 1:34PM |
4 |
Centrex |
| 1:29PM |
1 |
Unable to write frame to channel: Success - MeetMe problem |
| 12:43PM |
1 |
Can anybody explain the meaning of these messages? |
| 11:32AM |
0 |
Call waiting does not work with g729 codec |
| 10:09AM |
0 |
Re: Voicemail with separate greetings based on extension |
| 10:01AM |
1 |
Problem with Cisco 7905 "Not Acceptable Here" |
| 9:57AM |
1 |
User problem |
| 9:35AM |
0 |
Passing a PIN in SIP Parameters |
| 9:34AM |
1 |
Re: loss concealment (Steve Kann) |
| 9:05AM |
0 |
Frequent dropped call on Wildcard E100P |
| 9:02AM |
0 |
MVP130 |
| 8:37AM |
3 |
SIP via Wireless Ethernet Bridge and Double NAT |
| 8:25AM |
0 |
Weird problem with a Cisco call manager using the h.323 channel |
| 8:14AM |
2 |
H323 or SCCP? |
| 8:08AM |
4 |
Voicemail with separate greetings based on extension |
| 7:33AM |
4 |
adding an artificial delay to * |
| 7:23AM |
1 |
ChanSpy(scan) working, but not... any idea? |
| 7:11AM |
3 |
T100P Caller ID UK |
| 7:01AM |
2 |
field description /zaptel/zonedata.c |
| 3:20AM |
2 |
snom200 -> asterisk & dtmf (rfc2833) |