Monday February 28 2005 |
Time | Replies | Subject |
11:05PM |
0 |
snom220 *8 hangup |
10:29PM |
0 |
how to increase max number of simulatneousoutgoing calls |
10:10PM |
1 |
SNOM Call Diversion |
9:17PM |
0 |
chan_capi compile error on FC3 |
8:23PM |
0 |
strange CDR problem |
8:06PM |
0 |
what is phone: Linux Telephony channel |
6:50PM |
1 |
AMP with FC3 |
6:37PM |
0 |
about dial parameter L |
6:08PM |
0 |
New SMS gateway command |
5:34PM |
2 |
Advanced Conferencing options with out-of-treemodules? |
4:05PM |
0 |
What about Asterisk and handling switchtype qsig (zaptel) ? |
3:58PM |
1 |
Zap channel calling back after hangup (due to polarity CID detection) |
2:24PM |
2 |
Asterisk-OH323 no ringing |
2:23PM |
2 |
Advanced FollowMe or Forwarding Application Suggestions |
2:19PM |
0 |
Newbie---Inquiring. |
2:18PM |
1 |
No such host when trying to register |
1:58PM |
5 |
Strange text on Asterisk console |
1:47PM |
5 |
Grandstream and VLANs |
1:35PM |
1 |
I can't load modules (ztdummy, wcfxo.o) |
1:22PM |
3 |
Cannot compile (app.c) |
12:40PM |
4 |
Recommendation for dialplan in case of DDoS atta cks? |
12:17PM |
0 |
Advanced Conferencing options with out-of-tree modules? |
11:06AM |
0 |
Ring state patch |
10:59AM |
0 |
How to charge incoming calls with ASTCC ? |
10:57AM |
1 |
Asterisk network architecture |
10:54AM |
0 |
how to increase max number of simulatneous outgoing calls |
10:47AM |
1 |
Manager "Message: Originate failed" beinggenerated when callee does not pick up |
10:35AM |
0 |
RE: Asterisk-Users Digest, Vol 7, Issue 323 |
10:33AM |
0 |
Manager "Message: Originate failed" being generated when callee does not pick up |
10:18AM |
0 |
How to limit a peer to one connection only? |
10:12AM |
1 |
x101p + Nortel ATA2 |
10:06AM |
1 |
Suse 9.2 + CAPI Driver |
9:58AM |
0 |
Passing additional information to an AGI via a call file |
9:46AM |
1 |
Weird behaviour on incoming DIDs |
9:44AM |
1 |
FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format |
9:41AM |
0 |
queue_log and exitwithkey |
8:53AM |
1 |
Anybody using X-Lite Softphone ? tryedtoforwarda call to X-Lite.... |
8:37AM |
0 |
Anybody using X-Lite Softphone ? tryed toforwarda call to X-Lite.... |
8:22AM |
1 |
Sipura SPA-841 autodial? |
8:16AM |
1 |
Unable to handle ROSE operation 34 |
7:56AM |
2 |
phpconfig |
7:34AM |
2 |
Fax Failing |
7:10AM |
0 |
New Instalation |
7:07AM |
0 |
Secure IAX Interasterisk authentication ? |
6:40AM |
2 |
dialing application - newbie question |
6:37AM |
1 |
SIP video problems |
6:22AM |
1 |
Problem with call hold |
5:49AM |
0 |
SIP broadband phone addon for asterisk |
5:04AM |
0 |
ASTERISKBRASIL.ORG |
3:55AM |
0 |
Pb DTMF with Asterisk vs Cirpack Transit, Node |
3:31AM |
0 |
X100P with Analogue DDI Trunks |
3:13AM |
0 |
calling sdp |
2:38AM |
2 |
Two offices connection |
2:35AM |
1 |
call from two sip phones registered in different asterisk server |
1:37AM |
3 |
Digium E1/T1 card with mgetty+sendfax |
1:20AM |
0 |
Bad soundquality on inbound calls. |
12:58AM |
3 |
Digium Card Problems |
12:43AM |
0 |
Pb DTMF with Asterisk vs Cirpack Transit Node |
12:11AM |
1 |
setting up fromuser |
|
Sunday February 27 2005 |
Time | Replies | Subject |
10:18PM |
1 |
context of transfer |
9:51PM |
2 |
[Asterisk-Dev] Asterisk 1.0.6 |
9:18PM |
1 |
No Agents Catch |
7:46PM |
3 |
music on hold trouble |
6:53PM |
4 |
where is voice conduits |
4:53PM |
2 |
CDR's are not stored in mysql |
4:51PM |
0 |
Barter studio time for asterisk lessons Brooklyn NY |
4:49PM |
1 |
IAX2 (Stupid question) |
4:43PM |
1 |
dialout with PPP on ISDN to an ISP |
4:19PM |
0 |
test - no msg |
4:00PM |
1 |
Beronet BN4S0 (quad BRI) card, echo cancel, zaptel timing, bristuff ... |
3:54PM |
1 |
Possibility of getting someone to delete a user from the list??? |
3:51PM |
0 |
FW: DISA and a long delay; ideas? |
3:48PM |
0 |
IAX2 web client that works with g723 / g729 |
3:23PM |
0 |
email enviado sextafeira. sobre a lista IMPORTANTE |
3:12PM |
5 |
Problem selecting E1 on TE405P |
1:29PM |
0 |
Interface * with ATA from ATA FXS port? (Here I go again) |
12:39PM |
1 |
limit SIP extention outgoing calls |
12:34PM |
1 |
Which codecs are used? |
11:58AM |
2 |
Jumb between macro's and variables |
11:57AM |
4 |
Grandest Free Softphone |
11:24AM |
1 |
Suggestions for what to do with a Dialogic D/41EPCI? |
11:00AM |
5 |
Outbound call on TDM400P |
10:30AM |
0 |
not connecting with X-Lite |
10:10AM |
1 |
DISA and a long delay; ideas? |
8:50AM |
2 |
Introducing the Asterisk Realtime Architecture - ARA |
6:47AM |
2 |
Weird Delay (> 30 sec) |
6:38AM |
1 |
DIALSTATUS with X100P |
6:08AM |
0 |
ATA 286 downgrade failure |
4:10AM |
0 |
g723 issue+asterisk impropoer shutdown |
1:27AM |
2 |
opencall.org is changing to soft-switch.org |
1:23AM |
1 |
astguiclient gives me Object not found |
1:17AM |
0 |
Astcc installation |
12:43AM |
0 |
Transfer not working |
|
Saturday February 26 2005 |
Time | Replies | Subject |
10:19PM |
2 |
Wierd asterisk-perl compilation problem |
9:38PM |
2 |
Limit the call & recording when pressing *1 |
8:56PM |
0 |
snom 190 funtion buttons |
6:22PM |
1 |
call pickup with Sipura-3000 |
3:41PM |
0 |
'asterisk' displays on 2nd line (CID Number Line) on Cisco 79x0 phones |
1:23PM |
1 |
Dial out through Broadvoice |
11:55AM |
0 |
SIP phone speaker phone mic cutting out |
11:40AM |
1 |
BRIstuff - synchronization with PSTN? |
11:28AM |
0 |
How to grab CallerId information |
10:24AM |
0 |
Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite.... |
9:47AM |
0 |
Re: FRS over * |
8:27AM |
0 |
NAT= setting for a public proxy |
8:23AM |
2 |
Error Message |
7:58AM |
2 |
ERROR: compile asterisk(from CVS HEAD) and got an error |
7:27AM |
0 |
ERROR: compile asterisk (from CVS HEAD) |
6:37AM |
2 |
Interface * with ATA from ATA FXS port? |
5:30AM |
1 |
Which is best : Chan_capi or chan_misdn ??? |
4:54AM |
0 |
Wildcard failing to load on asterisk@home |
4:34AM |
1 |
ERROR: when compile app_addon_sql_mysql.c of asterisk_addon |
3:32AM |
1 |
Determine IP addres of a AIP/IAX user |
3:01AM |
3 |
listening to gsm files |
1:59AM |
1 |
Queue Auto fallthrough |
1:54AM |
0 |
Polycom SP300 problem solved |
12:47AM |
3 |
Language Problems |
12:45AM |
0 |
Polycpm SP300 problems |
12:06AM |
2 |
FRS & *: an actual business use |
|
Friday February 25 2005 |
Time | Replies | Subject |
11:50PM |
1 |
playing "i" invalid context to an internal user |
11:08PM |
1 |
Seting up for afirst time -- can not call |
8:45PM |
1 |
open 723 |
8:14PM |
0 |
Asterisk with regular analog phones |
4:29PM |
1 |
weather asterisk@home |
4:06PM |
1 |
Re: FRS radios on * |
3:59PM |
0 |
SER vs. Asterisk - call in progress to PSTN |
3:37PM |
1 |
VM+Realtime config |
3:33PM |
0 |
CallerID Name and Digium TE405P |
3:13PM |
1 |
Asterisk in front of Toshiba CTX |
1:47PM |
1 |
SetCIDNum using SIP? |
1:19PM |
1 |
Re: Asterisk-Users Digest, Vol 7, Issue 304 |
1:13PM |
3 |
Festival - Asterisk@home |
1:09PM |
1 |
Transposed ringing |
12:41PM |
0 |
Video Support Not Working |
12:33PM |
0 |
Wheres the Math application |
12:07PM |
5 |
HELP NEEDED ASTERISK AND MEDIATRIX 1102 |
10:57AM |
2 |
Fax on Asterisk |
9:54AM |
0 |
Speex transcoding for Cisco / Polycom |
9:44AM |
2 |
Avaya Partner ACS3 and Asterisk |
9:19AM |
1 |
WebVMail Woirks but No Audio |
9:00AM |
1 |
Directory config... |
8:59AM |
3 |
How does the g.729 registration program work? |
8:23AM |
0 |
Vonage <---> Asterisk Complete Config |
8:05AM |
1 |
Working SIP phone for linux and windows |
8:00AM |
2 |
407 Proxy Authentication Required |
7:54AM |
0 |
call waiting notification and cisco 7960 phone |
7:49AM |
0 |
Anyone had a Cisco 7970 working with |
7:25AM |
1 |
SIP Errors |
7:04AM |
0 |
Asterisk with PortaOne Radius client- problem in accounting script with OH323 |
6:53AM |
4 |
T.38 fax summary |
6:41AM |
1 |
r2 signalling in east europe |
6:32AM |
15 |
FW: Getting PHP Config to work? |
6:02AM |
2 |
Fedora Core 3? |
5:18AM |
1 |
msic while ringing |
5:00AM |
4 |
CDR writing incorrect data to pgsql tables |
4:10AM |
2 |
"click to dial extension number" functionality ? |
3:42AM |
1 |
Asterisk and 723,729 |
3:06AM |
0 |
international calls and NOANSWER |
2:37AM |
0 |
Which version of ast_data for Asterisk v1.0.5? |
2:36AM |
0 |
about caller sdp |
2:32AM |
1 |
cascaded ringing |
2:02AM |
0 |
WG: AW: Transfer a call ? Am I looking for theflashcommand ? |
1:34AM |
0 |
help me : about dial to PSTN |
|
Thursday February 24 2005 |
Time | Replies | Subject |
11:04PM |
0 |
Re: Radio over * |
10:10PM |
3 |
VoIP/Asterisk presentation |
10:04PM |
1 |
IPCB |
9:21PM |
1 |
RESELER ON INDONESIA |
9:12PM |
5 |
Asterisk With Broadvoice |
9:00PM |
0 |
Question of SER to Asterisk to PSTN |
8:26PM |
2 |
softphone has problem to call out via X100P card |
8:12PM |
1 |
Re: FRS and GMRS via * |
7:59PM |
0 |
Hope cooperate |
7:16PM |
2 |
asterisk supports VXML? |
6:44PM |
0 |
Connect to siemens pbx with misdn NT mode |
5:39PM |
1 |
Which Codec(s) to use..? |
4:50PM |
1 |
Transfer a call ? Am I looking for the flash command ? |
4:49PM |
2 |
Delay after entering digits with IVR |
4:42PM |
0 |
hint and contexts |
3:56PM |
4 |
What is an E400P-SS7?? |
3:10PM |
1 |
Call Xfer and other features.. |
3:04PM |
3 |
IAXY DNS possibilities?? |
2:57PM |
2 |
No audio when h323 calls are incoming |
2:49PM |
2 |
[Asterisk-Dev] How to monitor Agen Voice channal? |
2:03PM |
0 |
Weird Issu: Figuired it out |
1:36PM |
2 |
Weird Issue: Call will not go into VM |
1:28PM |
2 |
Making two * servers share same dial plan? |
1:25PM |
0 |
Get SPA-2000 to dial out on one * and get calls in from a different *? |
12:56PM |
1 |
choppy and cracking sound from zyxel prestige 2002 |
11:55AM |
7 |
CallerID problem |
11:15AM |
0 |
Caller in meetme room quiet (low level?) |
11:10AM |
1 |
How does Asterisk choose the CDr backend to use? |
11:08AM |
0 |
is this stuff for me? need some help |
11:01AM |
2 |
Can you set up a phone via MAC address? |
10:59AM |
0 |
transfer ringback |
10:54AM |
1 |
Park Call timeout |
10:51AM |
2 |
Asterisk and # |
9:59AM |
4 |
SIP Phone with headset |
9:51AM |
1 |
Zap Channels Disappear??? |
9:41AM |
2 |
OT - C structure question |
9:38AM |
2 |
do i have to reload asterisk every thing i add a new extension |
9:26AM |
1 |
Servidor SIP |
8:54AM |
2 |
Polycom Call Parking |
8:40AM |
0 |
Queue Questions |
8:31AM |
3 |
Inheriting variables |
8:24AM |
1 |
Queue Announcement |
7:51AM |
1 |
Bug in SUBSCRIBE handling : running out of RTP ports |
7:40AM |
0 |
Is using Sipura 2100 as SOHO main router good solution? |
7:23AM |
0 |
asterisk & proxies... |
7:19AM |
1 |
VideoMail & Asterisk |
6:53AM |
0 |
SV: SV: SV: QSIG, Asterisk and Eicon DIVA |
6:32AM |
1 |
Aastra 480i and Telnet - anyone know how to log in? |
6:29AM |
2 |
Asterisk and Welltech USB SIP phone K1000A |
6:12AM |
3 |
High capacity voicemail |
5:55AM |
1 |
Re: Asterisk-Users Digest, Vol 7, Issue 296 |
5:47AM |
2 |
Ericsson MD-110 and Dig-410 |
5:31AM |
0 |
MGCP transfer and CDR |
5:30AM |
0 |
Any $CALLER |
5:16AM |
0 |
FW: SIP echo on LAN |
5:03AM |
0 |
a silly question regarding call monitoring! |
4:55AM |
2 |
Brainstorm: Running Asterisk as cool as poss ible - AKA solid state. |
4:48AM |
1 |
Problems with SIP codec selection |
3:36AM |
0 |
Introduce bridged calls with a beep ... |
2:27AM |
0 |
Analogue Extension Hold Sequence |
2:13AM |
0 |
Strange problem with h323 |
2:10AM |
4 |
SV: SV: QSIG, Asterisk and Eicon DIVA |
1:57AM |
1 |
Call recording stopped when call transferred |
1:25AM |
1 |
SV: QSIG, Asterisk and Eicon DIVA |
1:20AM |
7 |
CallTransfer |
12:50AM |
1 |
Azatel Azacall 200 issue with asterisk |
12:41AM |
1 |
Meetme with video & audio phone mixed |
|
Wednesday February 23 2005 |
Time | Replies | Subject |
9:18PM |
1 |
Brainstorm: Running Asterisk as cool as possible - AKA solid state. |
9:04PM |
1 |
Cannot compile latest version from CVS |
8:03PM |
2 |
Trouble installing TE405P with asterisk@home |
7:00PM |
2 |
multiple sip phones behind firewall |
6:46PM |
1 |
Asterisk as a voicemail for a central office switch |
5:56PM |
0 |
Serious Audio Problem. |
4:01PM |
0 |
confirm use of INSTALL_PREFIX |
3:54PM |
1 |
AST -> Channel Bank Hangup Problem |
3:33PM |
0 |
ZAPHFC is back in bristuff 0.2.0-RC7d+ |
3:14PM |
0 |
Asterisk wont accept tone from Norstar 3X8 ATA port ? |
3:11PM |
1 |
Sound files quality and volume |
2:50PM |
4 |
Vonage <---> Asterisk Working Config! |
2:43PM |
2 |
SIP NOTIFY in stable branch? |
2:41PM |
8 |
FRS / FRS/GMRS 2-way radios as SIP clients |
2:29PM |
0 |
sharing parking lots |
1:12PM |
0 |
Cant connect to sjphone |
1:02PM |
2 |
Dialogic cards |
12:44PM |
1 |
Request for PRI Dump |
12:44PM |
1 |
Sipura 2000 w/fax machine oddities |
12:32PM |
1 |
AreskiCC - pass card number? |
12:30PM |
1 |
Re: Some simple voicemail questions... |
12:22PM |
0 |
Uniden, Polycom or SwissVoice??? |
11:58AM |
1 |
Best practices direction |
11:48AM |
0 |
Success stories - Asterisk + Video support |
11:33AM |
0 |
do i have to reload asterisk every thing i add a neww extension |
10:50AM |
0 |
Problem with dialing out and chan_capi |
10:27AM |
3 |
Able to tell if phone is registered? |
10:07AM |
0 |
Streaming Phone Calls |
9:39AM |
0 |
avaya 4602 |
9:38AM |
3 |
Problem connecting a TE410P to an E1/IP equipment |
9:28AM |
3 |
Send outgoing calls to a SIP gateway |
9:20AM |
1 |
List tips for new subscribers <--sorry for 2nd post, missed this. |
9:18AM |
0 |
Newbie Help - Auto Fallthrough |
8:56AM |
0 |
Question about DTMF |
8:54AM |
4 |
List tips for new subscribers <--sorry for 2 nd post, missed this. |
8:43AM |
2 |
7960 Not Picking up new firmware. |
8:36AM |
0 |
Digium TE405P and Cirpack Switch |
8:28AM |
0 |
logger reload/restart hanging |
8:27AM |
2 |
Creating extension groups |
8:20AM |
6 |
List tips for new subscribers |
7:31AM |
2 |
Using as FAX 100% IP |
7:03AM |
2 |
storing cdr in two databases |
6:22AM |
1 |
Zaptel (Junghanns 4BRI card) to cell phone problem |
5:56AM |
0 |
Re: Asterisk-Users Digest, Vol 7, Issue 284 |
5:27AM |
0 |
Subject: Welltech with Asterisk Registration |
5:21AM |
1 |
Error connecting to remote mysql database. |
5:15AM |
5 |
Difference between E1 and PRI |
5:03AM |
2 |
Digium BRI or quad BRI |
4:57AM |
3 |
Help With Adit 600 Configuration |
4:31AM |
1 |
Chanspy and current version of cvs |
4:01AM |
1 |
mixing sound files? |
3:54AM |
0 |
cdr_odbc logging insane integer values |
3:37AM |
1 |
Anyone had a Cisco 7970 working with Asterisk? |
2:37AM |
0 |
Teleconferencing using Zapta cards. |
2:35AM |
5 |
Zaptel Red Alarm |
2:15AM |
7 |
IVR stats |
1:36AM |
0 |
IAX Trunking capacity enforcement |
1:00AM |
0 |
hylafax |
12:35AM |
2 |
FW: What do I still need? |
|
Tuesday February 22 2005 |
Time | Replies | Subject |
9:24PM |
1 |
Voicemail as email attachment not working individually i.e. extensions specific |
9:18PM |
3 |
* or X100P dropping analog calls |
8:58PM |
0 |
Extension Design in Visio |
8:55PM |
2 |
SpanDSP - Still can't send |
7:50PM |
0 |
Do ser + asterisk_b2bua work ? |
7:44PM |
0 |
asterisk@home 0.6 |
6:54PM |
0 |
Connecting Broadvox Direct TA to * |
5:56PM |
0 |
How do I do this ? |
5:06PM |
1 |
Settings for SIP to dial PSTN with TDM400P w/FXO module |
4:44PM |
3 |
Call Manager Express Peer |
4:21PM |
0 |
register failed with 2nd Sipura-2000 |
3:23PM |
1 |
install BRIstuff on *@home? |
3:22PM |
0 |
bridging <ZOMBIE> ? |
2:49PM |
1 |
Anybody using X-Lite Softphone ? tryed to forward a call to X-Lite.... |
2:38PM |
0 |
Manager API problems |
2:33PM |
0 |
Grandstream 486 Sending Faxes issue out TDM400P |
2:19PM |
0 |
queue estimated hold time. |
2:13PM |
0 |
H.323 problem, calls don't get answered by asterisk |
1:32PM |
2 |
Repost: How do I install Skinny support for non sip cisco phones |
12:19PM |
1 |
Astersik CVS HEAD + T1 e&m wink + IAX client doesnt detect call answered on Zap channel |
11:56AM |
0 |
Asterisk-HEAD more stable than Asterisk-1.0. 5 |
11:51AM |
1 |
Multiple Parking Lots. |
11:49AM |
0 |
Asterisk-HEAD more stable than Asterisk-1.0.5 |
11:24AM |
13 |
TFTP Server |
11:17AM |
1 |
Voicemail call notification of voicemail |
10:34AM |
2 |
newbie needs advice |
10:29AM |
1 |
how do I dial extensions with oh323? |
9:55AM |
4 |
mp3 to gsm? |
9:46AM |
1 |
Finding the true src in CDR |
9:17AM |
0 |
PSTN tones with ISDN4Linux |
8:43AM |
1 |
Noob question on connection |
8:31AM |
2 |
QSIG, Asterisk and Eicon DIVA |
8:29AM |
1 |
Polycom IP 500 : Displaying digits dialed after connection |
8:28AM |
2 |
[PBX]: New message 1 in mailbox 1000 |
7:49AM |
1 |
Sip billing |
7:41AM |
0 |
Monitor and Record : audio quality |
7:33AM |
2 |
Zap timing device |
6:47AM |
0 |
setting caller id number and usingsip type=peerfor incomming calles. |
6:17AM |
2 |
Amphenol cables? |
5:54AM |
0 |
send fax with pri |
5:41AM |
0 |
[Fwd: Asterisk to Asterisk via IAX2 Help] |
5:35AM |
4 |
does asterisk support menus? |
5:34AM |
0 |
CDR - is this possible |
5:01AM |
0 |
DID, Sending dialled number to PBX |
4:56AM |
4 |
Sound of breathing |
4:38AM |
0 |
VMS - AGI |
4:29AM |
0 |
manager interface, get callerid number?? |
4:23AM |
2 |
ISDN/SIP videophone gatewaying? |
4:08AM |
1 |
what is problem in odbc |
3:41AM |
3 |
asterisk -vvvvvvvgrc? |
3:25AM |
0 |
SPEEX installation problems |
2:15AM |
1 |
asterisk to pbx dialing |
2:12AM |
1 |
OH323 and CDR |
1:55AM |
0 |
Segfault when using res_config_odbc on x86_64 |
12:48AM |
2 |
Custom Menu Not Working |
12:07AM |
1 |
route outgoing call |
|
Monday February 21 2005 |
Time | Replies | Subject |
11:29PM |
0 |
Textron voip gateway |
10:26PM |
2 |
Noise during calls |
9:34PM |
4 |
sip wifi phone? |
8:57PM |
0 |
Asterisk Video Phones <-> Cisco Call manager 4.0 |
8:34PM |
0 |
app_groupcount |
8:06PM |
0 |
SIP registration timeout |
7:14PM |
1 |
Canadian DIDs... |
6:55PM |
0 |
Multiple multiline sip phones ringing. |
6:55PM |
0 |
LiveVoip digit loss |
6:49PM |
2 |
Unable to call FWD user via IAX servers |
6:18PM |
1 |
NAT-helping outbound proxy |
6:01PM |
0 |
Re: list SNR |
5:54PM |
1 |
zaprtc on Debian Sarge 2.4.27 |
5:35PM |
8 |
Minimal hardware requirements |
4:02PM |
2 |
Suggestion for noise reduction on Asterisk-U sers |
3:54PM |
2 |
Problem with Avaya 4602 / SIP response 481 |
3:53PM |
0 |
Brian Elton / Avaya 4602 |
3:50PM |
1 |
some questions about busy detection |
3:47PM |
0 |
FWD problem |
3:35PM |
3 |
IAX ATA's |
3:32PM |
0 |
Call terminaison Tools |
3:13PM |
2 |
Suggestion for noise reduction on Asterisk-Users |
2:50PM |
0 |
FWD using IAX2 |
2:12PM |
1 |
ZAP FXS vs ethernet FXS |
2:04PM |
2 |
Asterisk@home Linux has no KDE |
1:45PM |
0 |
Hitachi Wireless SIP handset |
1:27PM |
0 |
Problems with the FXS module in a TDMxxx card (no sound when receiving a call |
1:05PM |
0 |
VoIP Test Phone |
12:56PM |
1 |
Dns problems with digium and asterisk.org? |
11:46AM |
1 |
voice recognition xml |
11:27AM |
0 |
South Korea DID wanted |
11:16AM |
1 |
why can't I make toll free calls via IAXTEL |
11:04AM |
2 |
Why can't I make inter IAX calls between 2 Asterisk servers |
11:00AM |
1 |
IAX channel unable to create |
10:54AM |
0 |
Call Announce |
10:32AM |
0 |
Terminating problem |
10:32AM |
1 |
setting caller id number and using sip type=peerfor incomming calles. |
10:05AM |
2 |
Zap call bridge drops randomly |
9:34AM |
0 |
Asterisk to Asterisk via IAX2 Help |
9:22AM |
1 |
X-IMail-SPAM-Connection DNS Sudo ANI vs True ANI |
8:33AM |
2 |
Adit 600 MGCP configuration |
8:25AM |
2 |
Anyone using SuperMicro SuperServer 6014P-8R? |
8:25AM |
4 |
Polycom Phone Calling Party ID |
8:08AM |
1 |
setting caller id number and using sip type=peer for incomming calles. |
7:26AM |
2 |
Illegal instruction on startup |
7:06AM |
0 |
[SOLVED] Problem with ISDN Dialin via CAPI |
6:00AM |
3 |
* Call Monitoring |
5:55AM |
0 |
Any luck with attended transfer and ATA186? |
5:52AM |
1 |
Monitoring calls through a transfer |
5:52AM |
0 |
bug? Unterminated comment detected beginning on line 0 |
5:36AM |
2 |
compiling cvs-head today? |
4:43AM |
0 |
LineJACK dial problem |
4:10AM |
0 |
How to ECT (explicit call transfer) ? |
3:47AM |
2 |
Conecting to asterisk server through NAT usingIAX |
3:36AM |
1 |
Problem with ISDN Dialin via CAPI |
3:07AM |
0 |
CallingCard application AreskiCC RELEASE v1.1 |
2:53AM |
0 |
Disable musiconhold |
2:30AM |
0 |
ZAP libpri issue crashes PRI? |
2:12AM |
1 |
Problems with the FXS module in a TDMxxx card (no sound when receiving a call) |
1:42AM |
1 |
MOH clicks |
12:47AM |
1 |
SIP echo on LAN |
12:40AM |
0 |
Thank You Note |
12:37AM |
1 |
Conference between 2 lines |
|
Sunday February 20 2005 |
Time | Replies | Subject |
11:48PM |
0 |
Fwd: res_config_mysql & chan_iax2 socket_read error |
11:13PM |
1 |
Re: Ring/Off-hook in strange state 6 on channel... |
11:09PM |
1 |
Sangoma A101 |
10:26PM |
0 |
SIP to SIP calls have no audio until put on hold and taken back off - SOLVED |
10:19PM |
10 |
HELP NEEDED! - Asterisk GUI |
10:14PM |
2 |
Asterisk H323 support |
9:53PM |
2 |
How many line appearance can Snom 200 handle? |
9:50PM |
1 |
How to announce the DNID to the called party |
8:52PM |
3 |
* > Mobile Phone > Mobile Network |
8:41PM |
1 |
Where to contrib the sound files ? |
8:17PM |
2 |
Modem as PSTN interface? |
8:00PM |
1 |
Adding zap channels under *@Home |
7:53PM |
1 |
PLease help: Asterisk to Quintum interconnection |
7:25PM |
1 |
Phones for vitural office business |
5:42PM |
3 |
help with @home |
5:13PM |
0 |
Traditional Ringback Tone |
5:12PM |
1 |
NAT and FWD |
4:07PM |
1 |
HFC-S ISDN card on *@home |
3:31PM |
0 |
Sparc hardware, Linux and X100P REVISITED |
2:58PM |
1 |
Conecting to asterisk server through NAT using IAX |
1:23PM |
1 |
What happens if quadbri or octobri loses power - do they have power failure feature ? |
1:02PM |
0 |
Recording of calls stopped - normal behaviour? |
1:01PM |
0 |
Re: Asterisk-Users Digest, Vol 7, Issue 260 |
11:51AM |
3 |
Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11? |
11:06AM |
2 |
Voice Prompts with no sound |
9:41AM |
1 |
Adtran Total Access MGCP Config? |
9:11AM |
0 |
SIP to SIP calls have no audio until put on hold and taken back off |
8:43AM |
0 |
SIP peer registration interval - SOLUTION |
8:14AM |
0 |
CDR for callback |
5:00AM |
1 |
Mandrake & CAPI |
4:00AM |
2 |
External relay triggered by Asterisk extension-question |
3:02AM |
1 |
making ASTCC web page secure ??? |
2:43AM |
7 |
bridging iaxtel calls to PSTN |
1:20AM |
8 |
Simulated dialtone like in other PBX |
12:10AM |
2 |
Soundcard problems? |
|
Saturday February 19 2005 |
Time | Replies | Subject |
10:33PM |
1 |
External relay triggered by Asterisk extension - question |
9:29PM |
2 |
asterisk setup |
9:22PM |
0 |
ROUTING INCOMING CALL BASED ON CADENCE? |
4:35PM |
0 |
Can't Dial-out |
3:38PM |
2 |
Anyone used the ACT P104SLD SIP Phone |
3:02PM |
0 |
X-IMail-SPAM-Phrase X-IMail-SPAM-Connection DNS Problem with T1 and international calls |
2:58PM |
1 |
video conferencing |
2:16PM |
1 |
sending traffic to LiveVoip |
2:02PM |
3 |
Terminating IAX calls in E1 PRI interface - what do I need to be able to send arbitrary caller id to called party ? |
1:49PM |
0 |
HFC/zaphfc/zaptel: issues with multiple inbound calls |
1:42PM |
2 |
I have a odd question... |
12:53PM |
2 |
No Sounds |
10:43AM |
1 |
Asterisk with Multitech H323 Gateway MVP400 |
9:59AM |
4 |
I need to dial multiple numbers concurently but with delays. |
9:06AM |
1 |
Can I exchange datas between two Asterisk servers ? |
9:00AM |
1 |
Uniden UIP200, please help |
8:56AM |
3 |
simpletelecom.com??? are they a SCAM? |
8:49AM |
3 |
Still asterisk startup crash plz help |
8:12AM |
3 |
Hi Newbie question |
5:58AM |
2 |
This is NUTS!!SOLVED |
1:19AM |
2 |
Sip question - allow only 1 incoming call to sip phone |
12:30AM |
0 |
Understanding and Troubleshooting Analog E&M |
12:24AM |
0 |
TOUCH_MONITOR |
12:14AM |
16 |
Snom phone hint exten question |
|
Friday February 18 2005 |
Time | Replies | Subject |
10:54PM |
0 |
Asterisk to Quintum gateway interconnection |
7:16PM |
0 |
ACD softphones? |
6:22PM |
2 |
VoIP Test Samples to test Asterisk |
5:04PM |
1 |
Asterisk@home festival weather report |
4:30PM |
0 |
Time to beg on my knees for help!!! |
4:06PM |
0 |
Using * to connect to database and to modify said database |
4:01PM |
0 |
FXS signalling for Ireland |
3:47PM |
1 |
wikki problem |
3:18PM |
0 |
Installing Asterisk on Mandrake 10.1 Official |
3:11PM |
5 |
Which PRI card for EuroISDN ? |
3:10PM |
0 |
Looking for Asterisk setup and maintainance (terminating calls to EuroISDN PRI interface) in Frankfurt Germany |
2:45PM |
1 |
GotoIfTime Discrete weekdays (Mon,Wed,Fri) |
2:39PM |
0 |
Feb 18 15:22:33 WARNING[1862]: chan_zap.c:3428 zt_handle_event: Ring/Off-hook in strange state 6 on channel 2 |
2:33PM |
0 |
*** Important *** About the bug tracker |
2:21PM |
0 |
Monitoring stops when call is transferred |
2:08PM |
1 |
Power failure + which card must i choose |
2:06PM |
2 |
Sending DTMF after a call is set up |
11:55AM |
1 |
VoIP Service Provider |
11:49AM |
0 |
Grandstreams ATA286 |
11:48AM |
2 |
Difference between a TE410P and TE405P? |
11:33AM |
0 |
More asymmetrical call quality discussion |
11:11AM |
2 |
defining the zap channel used on inbound analogue calls |
11:06AM |
1 |
Send CallerID to PBX via PRI NI2 |
10:35AM |
2 |
MSG WAITING OFF on cordless handset not going away |
10:27AM |
1 |
Safecom SIP-300 Information? |
10:04AM |
3 |
need info |
9:56AM |
4 |
HDLC Bad FCS / HDLC Abort |
9:54AM |
3 |
Help asterisk startup errors |
9:26AM |
1 |
Calls directed via queue to unavailable device result in call acceptance |
9:04AM |
6 |
W&M Wink timings for Nortel |
8:53AM |
5 |
Asterisk GUI |
8:44AM |
0 |
VAD (Silence suppresion problem) |
8:41AM |
0 |
Process incoming faxes in Asterisk |
8:25AM |
2 |
VONAGE <----> ASTERISK SIP TERMINATION????? |
8:21AM |
4 |
A bit of a survey: What do do if you need more than 4 C.O. lines |
8:21AM |
2 |
This is NUTS!! |
8:20AM |
0 |
Asterisk 1.0.5 an MySQL CDR |
8:16AM |
0 |
More on W6692pci NT mode under chan_misdn |
8:13AM |
0 |
Asterisk on Solaris 10 |
8:09AM |
1 |
Help with config. |
8:04AM |
5 |
Budgetone 101 |
8:02AM |
0 |
Monitoring a telco line for MWI through a TDM400 FXO |
6:52AM |
0 |
TDM 2 FXO + Traditional PABX |
6:43AM |
1 |
wrapuptime + agents.conf |
6:29AM |
2 |
Asterisk + RedHat9 - Libpri problem |
6:26AM |
2 |
Wiring question for Digium card |
6:18AM |
2 |
Q.SIG support in CVS |
6:14AM |
1 |
Disable Loop Detection |
6:13AM |
1 |
ISDN channel bank |
6:12AM |
0 |
Asterisk with SER |
6:08AM |
0 |
mISDN+w6692pci errors while loading |
4:46AM |
3 |
quadbri and spandsp |
4:46AM |
0 |
Asterisk Can't Run |
4:27AM |
0 |
TDM400P and SOHO traditional (analog) telephones |
3:54AM |
0 |
Voice Message Matching? |
3:47AM |
3 |
MultiLine Sip Phones |
3:42AM |
1 |
Is this a bug or by design? Workaround? |
3:15AM |
1 |
Asterisk Performance in comparission of SER |
2:54AM |
1 |
Vonage, broadvoice et al |
2:14AM |
1 |
Timing device OpenBSD |
2:00AM |
1 |
Problem with starting music on hold when cal l connects to phone via queue |
12:53AM |
1 |
Problems compiling on mandrake |
12:48AM |
2 |
any good redhat 9.0 rpm reposiroty? |
12:42AM |
0 |
can't see calling number |
12:20AM |
3 |
Astricon 2004 tutorials available? |
12:14AM |
3 |
SER/Asterisk consultants in Denver |
|
Thursday February 17 2005 |
Time | Replies | Subject |
10:04PM |
0 |
SIP "catchall" |
9:33PM |
1 |
Zultys Paging Solution / App for Multicast |
9:29PM |
0 |
Problem with the IAXy and Netgear Hubs! |
8:57PM |
0 |
DTMF Problems with Asterisk |
7:01PM |
2 |
Zaptel Needed |
5:44PM |
0 |
SIP Seeding peers from Astdb - jam the console |
5:15PM |
1 |
Problem with starting music on hold when call connects to phone via queue |
5:03PM |
1 |
Voicepulse Open Access & Asterisk Problems |
5:02PM |
0 |
MGCP - Unicall |
4:34PM |
0 |
Warning messages error |
4:22PM |
0 |
List of VoIP provider codes |
4:20PM |
0 |
TDM400 FXO not responding to inbound rings a fter 30ish days? |
4:10PM |
4 |
Mac Mini and chan_bluetooth, has anyone told The o if it works? |
3:54PM |
4 |
functional difference: canreinvite=yes, no, or update |
3:54PM |
1 |
Problems compiling pridump utility |
3:10PM |
1 |
TDM400 FXO not responding to inbound rings after 30ish days? |
2:59PM |
0 |
E1/PRi Hardware echo canceller |
2:41PM |
1 |
(Kphone) Registration Failed: Forbidden |
2:34PM |
1 |
RE: Asterisk-Users Digest, Vol 7, Issue 239 |
2:24PM |
0 |
asterisk@home greek letters and suggestions |
2:07PM |
8 |
Trying to install X100P |
1:16PM |
2 |
Accountcode and SIP Peers Part 2 |
1:14PM |
1 |
X100P DID |
1:04PM |
0 |
Accountcode and SIP Peers |
12:47PM |
0 |
Festival and french language |
11:23AM |
2 |
arrgghhh dialparties.agi |
10:37AM |
4 |
IAXy Provisioning Using Windows |
10:29AM |
5 |
Digium TDM 400P and Dell 1750 |
10:24AM |
5 |
PRI and echocancel |
10:22AM |
1 |
Re: Cisco 7970 Won't boot after factory rese t |
9:49AM |
2 |
Packet 8 |
9:29AM |
1 |
UIP-200, registers, 4 seconds pass, then #1 disconnected |
8:46AM |
2 |
The 'sipfriends' table is obsolete - ???? |
8:25AM |
4 |
SIP peer registration interval |
8:07AM |
2 |
Sangoma A104 - D-Channel problem |
7:21AM |
1 |
Problem with asterisk-addons: libmysqlclient.so.14: cannot open shared object file |
7:20AM |
1 |
Cyclades-PC300/TE 1 Compatibility? |
7:02AM |
4 |
Strange MSN issue with HFC-s |
6:59AM |
0 |
Brand New Digium T100P for sale |
6:39AM |
1 |
Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl |
6:26AM |
1 |
Sirrix ISDN Card |
5:05AM |
1 |
VoipJet issues? |
4:55AM |
0 |
(no subject) |
4:41AM |
0 |
started asterisk with chan_misdn |
4:35AM |
0 |
Error loading wcfxs module |
4:35AM |
2 |
Voicemail and busy tone |
4:08AM |
4 |
can't enable trunking :( |
3:29AM |
4 |
Call termination database |
2:48AM |
1 |
change the caller id number |
1:54AM |
1 |
asterisk functions without voIP |
|
Wednesday February 16 2005 |
Time | Replies | Subject |
11:37PM |
0 |
asterisk and gatekeeper |
11:27PM |
5 |
problem : undefined symbol. |
11:09PM |
1 |
RTP Stream on Multicast |
10:45PM |
2 |
Anyone having trouble with VoicePulse Connect? |
9:55PM |
0 |
zap a sip channel |
9:51PM |
2 |
Zap/g0/ to a Telstra Mobile |
9:24PM |
4 |
festival text for weather report |
6:25PM |
3 |
Monitoring Conferences |
5:11PM |
0 |
Outbound calling timeout |
3:22PM |
0 |
Melbourne Asterisk Users meet TONIGHT |
3:20PM |
1 |
DIAX 0.9.10d with Eutectics USB phone suport |
3:10PM |
2 |
Cisco 7970 Won't boot after factory reset |
2:57PM |
0 |
More jitter buffer questions |
2:07PM |
1 |
Zaptel DACS and FDL |
1:51PM |
0 |
Using zaphfc and wcte11xp at the same time problem |
1:40PM |
3 |
IAX2: Connection rejected |
1:28PM |
0 |
Verizon BroadBandAccess and * |
12:58PM |
3 |
capiECT problem |
12:46PM |
0 |
zaphfc buffer underflow/overflow messages |
12:41PM |
0 |
Agent Logoff not generating event messages |
12:39PM |
2 |
Sip Notify PAP2-NA? |
12:33PM |
1 |
Help Please!!!! |
12:25PM |
0 |
Polycom MGCP firmware |
12:03PM |
1 |
WLAN-Voip phones anyone? |
11:45AM |
1 |
IAX Hardphone AT-320EE |
11:34AM |
1 |
Inter-asterisk conferencing delays - IAX2 configuration problem? |
10:54AM |
0 |
TDM card and Call recognition |
10:31AM |
1 |
[patch] fix libpri problem in Q931_INFORMATION handling |
10:03AM |
0 |
G729, NAT and Transcoding (all-in-one) |
10:02AM |
1 |
Can't connect Snom 190 to Asterix PBX. Sugge stions? |
9:17AM |
0 |
How do I match a "D"? (Was: RE: In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?) |
9:06AM |
0 |
When callerid changes its value ? |
9:04AM |
0 |
Can't connect Snom 190 to Asterix PBX. Suggestions? |
8:57AM |
2 |
Monitor does not like variable subsitutions |
8:56AM |
4 |
Dutch VOIP-PSTN provider |
8:49AM |
4 |
Why Asterisk can't cope with silence suppression? |
8:08AM |
0 |
Attended xfer |
8:04AM |
3 |
HELP!!!!!!!! |
7:42AM |
0 |
ZAP channel on TE410P doesn't hang up (Plain Text this time) |
7:25AM |
0 |
ZAP channel on TE410P doesn't hang up |
6:40AM |
4 |
DTMF inband detection improvement |
5:40AM |
1 |
chan_sip errors on CVS HEAD |
4:40AM |
4 |
Asterisk exist with error |
4:27AM |
1 |
Strict Routing vs Loose Routing |
2:27AM |
1 |
Passthrough and reInvite |
12:49AM |
0 |
Login OK but NO SOUND |
12:38AM |
0 |
bristuff-0.2.0-RC7a error messages |
|
Tuesday February 15 2005 |
Time | Replies | Subject |
11:27PM |
0 |
Re: card dialer phone (thanks for the info!) |
9:11PM |
0 |
How does Asterisk use ALSA? |
9:11PM |
2 |
Re: X100P problems |
8:43PM |
0 |
X100p + cell socket no callerid |
7:10PM |
0 |
VM and MeetMe Stopped working! HELP |
6:30PM |
2 |
Dialplan + Registrar DB |
6:29PM |
1 |
Help With Broadvoice |
6:27PM |
1 |
Queue strategy |
6:04PM |
1 |
newbie: help two cisco phones (sip) |
5:10PM |
3 |
Dial (Local/.....) |
5:05PM |
1 |
Asterisk "no one is available to take your call" |
4:55PM |
0 |
Supermicro P4SGA board? |
4:46PM |
1 |
iax.cc and/or Sixtel.net seems like IT IS A SCAM. |
4:33PM |
0 |
Making ZAP Trunk groups |
4:01PM |
0 |
Problem with IAX and codecs |
3:37PM |
3 |
iax.cc and/or Sixtel.net ,, IS IT A SCAM??? |
3:34PM |
0 |
Re: Asterisk-Users Digest, Vol 7, Issue 216 |
3:29PM |
6 |
A hypothetical question... |
3:15PM |
1 |
Solaris 10 |
2:13PM |
0 |
HFC-S and TE110P at the same time |
1:41PM |
2 |
Ser 0.9.0 adding a user? |
1:39PM |
0 |
Playtones segfaults? |
1:30PM |
0 |
asterisk@home and grandstream display |
1:09PM |
0 |
E&M and other Radio-based signalling |
1:07PM |
2 |
Sixtel.net / IAX.CC - Vanity Toll-Free Numbe r |
12:54PM |
1 |
Teles PCI and chan_capi, possible ??? |
12:44PM |
1 |
Put call on hold |
12:28PM |
2 |
Stop now, well it doesn't :) |
12:25PM |
1 |
"i" extension with invalid context |
12:22PM |
2 |
Mandrake 9.2 and CAPI |
12:05PM |
2 |
Asterisk Integration with ALCATEL 4400 |
11:59AM |
0 |
Spectralink SVP server - Asterisk |
11:44AM |
1 |
More *@Home puzzle |
11:40AM |
0 |
CVS Head ADSI Voicemail Busted ? |
11:40AM |
1 |
Strange error in debug file |
11:32AM |
0 |
perl poe::component::client::asterisk::manager usage |
11:07AM |
3 |
Virtual PBX setup. |
10:35AM |
0 |
Queue Abandoned and DND |
10:18AM |
1 |
IAX2 bugs... |
10:10AM |
0 |
Call Recognition. Which TDM card? |
10:01AM |
0 |
OT: Comments on Vonage SIP port blocking com plai nts?? |
9:56AM |
14 |
X-Lite Softphone |
9:36AM |
1 |
7912G via SIP, looking for comments |
9:29AM |
0 |
Fail to detect DTMF over direct ISDN pri lin k |
9:17AM |
2 |
OT: Comments on Vonage SIP port blocking complai nts?? |
9:01AM |
1 |
Asterisk@Home .5 Setup help with 4 X100P |
8:49AM |
3 |
Autostart Asterisk on Slackware? |
8:39AM |
0 |
Asterisk Users in Madrid? |
8:09AM |
2 |
Asterisk, inband DTMF send by a GSM mobile |
8:01AM |
0 |
Mobile operator message |
7:54AM |
0 |
oh323 question |
7:48AM |
7 |
Extra sounds (Weather) |
7:46AM |
2 |
E1 and/or Euro-ISDN specifications? |
7:40AM |
1 |
app_rxfax creating bad faxes? (StripOffsets) |
7:38AM |
0 |
extension matching in gastman |
7:27AM |
2 |
Sixtel.net / IAX.CC - Vanity Toll-Free Number |
7:09AM |
1 |
Integration Panasonic PBX |
6:01AM |
1 |
"System" command causes core dump Warning: Newbie help :) |
5:54AM |
3 |
4xHFC-s cards vs 1 quadbri HFC-4S card ? |
5:54AM |
0 |
Asterisk hangs the establised calls |
5:41AM |
0 |
[OT] Anyone that knows this ATA? |
5:34AM |
2 |
make of asterisk doesn't do anything... |
5:27AM |
1 |
asterisk qualified |
5:04AM |
1 |
(no subject) |
4:44AM |
4 |
solid-state asterisk pbx? |
4:17AM |
1 |
Question regarding SER/Asterisk functionality |
3:52AM |
2 |
CAPI not installed |
3:28AM |
0 |
Problems with SIP Registration at PSTN Provider |
3:21AM |
1 |
Asterisk and Call recognition (call id) |
3:06AM |
0 |
asterisk@home in production env |
2:54AM |
0 |
prblem in compileing asterisk-prepaid |
2:45AM |
2 |
Capi channel - can I route call to another channel or back to PBX and free current channel ? |
1:52AM |
3 |
Sip phones how to dial a # sign? |
1:38AM |
2 |
why does the Polycom IP600 check FTP every 60 seconds... |
1:32AM |
0 |
Asterisk restart alone |
12:03AM |
0 |
Asterisk@Home 0.5 |
|
Monday February 14 2005 |
Time | Replies | Subject |
11:42PM |
0 |
No Sound??? |
11:13PM |
18 |
Which IP phone to use in Australia |
10:59PM |
1 |
Native vs Intl calls |
7:42PM |
4 |
Clarification on Fax capability? |
7:32PM |
1 |
Flash Operator Panel - lots of problems |
6:21PM |
3 |
TFTP Serer ???? |
6:07PM |
7 |
Outbound Caller ID on PRI |
5:07PM |
2 |
Can't run AGI for outbound call |
4:43PM |
0 |
ASTCC Auth and Dialing problem |
4:39PM |
0 |
CVS with attended transfers |
3:11PM |
1 |
usb phones in linux, any?? |
2:16PM |
2 |
ztdummy on Gentoo 2.6.10 Box |
1:42PM |
0 |
Asterisk as a Protocol Converter from E1 to T1 |
12:48PM |
0 |
VXML support |
12:30PM |
1 |
Bristuff-0.2.0-RC5 florz patched weird error and no outgoing calls? |
12:23PM |
1 |
Asterisk@Home ... the next step |
11:50AM |
0 |
H323 no sound |
11:31AM |
0 |
cdr_mysql losing logs |
11:25AM |
5 |
Sipura g729 call quality to PSTN |
10:49AM |
1 |
(no subject) |
10:44AM |
0 |
H323 registration |
10:26AM |
0 |
Italian speaking. Asterisk configuration and needs |
9:45AM |
1 |
Uptime/reliability with SER, Asterisk |
9:18AM |
2 |
ztmonitor |
9:06AM |
0 |
Asterisk as SIP UAC !!! |
8:46AM |
0 |
APP_QUEUE MYSQL LOGGING |
8:24AM |
5 |
ATA that actually work with T.38 |
8:13AM |
2 |
FW: SER Asterisk Voicemail |
8:13AM |
1 |
E1-PRI: Warning Message: Unable to handle ROSE operation 36 |
8:10AM |
4 |
Asterisk-H323 |
7:55AM |
1 |
Asterisk@home .5 and meetme |
6:44AM |
0 |
SIP configurations |
6:38AM |
2 |
Asterisk in Singapore. |
6:24AM |
3 |
Digium Cards connecting to BT |
6:21AM |
3 |
asterisk in New-Zealand |
5:31AM |
0 |
Re: Asterisk-Users Digest, Vol 7, Issue 202 |
5:27AM |
0 |
Error: Unknown RTP codec 72 received??? |
5:05AM |
0 |
Error: Unknown RTP codec 72 received |
4:41AM |
1 |
Sipura 841 and paging function |
4:22AM |
6 |
Linphone / Kphone |
3:33AM |
3 |
ISDN zaphfc - What kernel are you using successfully? |
2:48AM |
0 |
spandsp asterisk 3/5 |
1:27AM |
0 |
Re: card dialer phone |
|
Sunday February 13 2005 |
Time | Replies | Subject |
11:18PM |
1 |
OT: Aastra 390 - weird problem |
11:13PM |
1 |
Mysql and SIP real time configuration... |
7:52PM |
6 |
Who makes these phones? |
7:40PM |
3 |
Q: Does anyone have a WE multi-line card dialer phone working with *? |
2:59PM |
1 |
Snom 190's vs Softphone |
2:32PM |
0 |
zaphfc NOTICE[6799]: chan_zap.c:7685 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
1:55PM |
1 |
TDM-400P alternatives? |
1:54PM |
1 |
TDM-400P Sound Quality issues |
12:14PM |
1 |
Broadvoice international dialling question |
12:00PM |
0 |
No CallerID on TDM11B? |
11:57AM |
2 |
connect asterisk to ISDN in China |
11:39AM |
1 |
Dlink VPNs?? |
10:00AM |
1 |
MusicOnHold Native Mode, Please Clarify |
5:01AM |
3 |
Sangoma A102 cards testing |
4:14AM |
0 |
Caller IP-Addr from agi ? |
3:56AM |
2 |
OT: Open source CRM systems with * integration |
3:25AM |
0 |
problems detecting hangup events |
12:55AM |
1 |
bad sound ISDN bristuff |
12:16AM |
2 |
TDMOE + kernel badness |
|
Saturday February 12 2005 |
Time | Replies | Subject |
8:51PM |
3 |
Cannot reset an IAXy box!!! |
8:46PM |
2 |
Asterisk+GNOMEMeeting=No Sound. |
8:15PM |
0 |
anyone patched CVS Asterisk with ast_data? |
7:52PM |
0 |
IAX2-FWD |
7:10PM |
2 |
Intermediary jitter buffering |
4:30PM |
1 |
Installation of Zatel |
4:02PM |
0 |
Asterisk as B2BUA. New application!!! |
3:56PM |
1 |
ast_data does not patch |
3:55PM |
0 |
Re: Asterisk as b2bua |
3:54PM |
0 |
*@home .5 Double Dial Tone |
2:50PM |
0 |
uninstall Asterisk? |
2:38PM |
2 |
soho fax suggestions? |
1:36PM |
0 |
How stable are cheap HFC-s cards in NT mode ? |
1:29PM |
1 |
What quad/octo BRI cards are best/stable for EuroISDN and Asterisk ? |
12:46PM |
2 |
Mobile Wireless IP Phone |
11:39AM |
0 |
Finding exact build version |
11:04AM |
0 |
French CallerID |
11:01AM |
1 |
ASTCC vs AreskiCC |
10:46AM |
3 |
7912G: Takes the same firmware as 7940/60? |
10:05AM |
1 |
PLEASE HELP Adit 600 went kaput? |
10:01AM |
1 |
return code of app in dialplan |
8:36AM |
1 |
Flash Pane - Monitor Parked Calls? |
8:31AM |
1 |
Re: Codec Issue on IAX trunk? (Solved) |
7:49AM |
0 |
bristuff-0.2.0 RC7 and RC7a |
7:05AM |
1 |
MGCP, Asterisk & Cisco VG200 |
6:52AM |
0 |
Delay on zap channel |
6:29AM |
0 |
Outbound calls on a busy Zap/1: BUSY vs. CHANUNAVAIL |
5:20AM |
3 |
Initializing two ISDN cards in isdn4linux |
4:44AM |
0 |
Asterisk as B2BUA - New Application!!! |
3:20AM |
0 |
Possible to use CAPI PBX as interface to analog phone? |
2:33AM |
5 |
fax with asterisk |
2:02AM |
0 |
Missed Call List on SIP Phones |
1:20AM |
1 |
iax.conf config and iax based clients |
12:46AM |
3 |
Is there a Caller ID issue in the latest CVSStable |
|
Friday February 11 2005 |
Time | Replies | Subject |
10:52PM |
1 |
Asterisk won't answer incoming analog line |
10:50PM |
0 |
Q: Can Zap channels be arbitrarily numbered? |
7:39PM |
1 |
Is no one using MySQL on stable asterisk? |
6:13PM |
0 |
Playing Dialtones |
6:06PM |
0 |
Polycom 300 -- "No compatible codecs!" |
6:05PM |
0 |
SendText application |
5:55PM |
5 |
Asterisk@home .05 release questions on setup. |
5:39PM |
1 |
Problem with # Transfer from queue |
5:35PM |
0 |
ASTCC one stage dialing problem |
5:04PM |
1 |
Re: Codec Issue on IAX trunk? (Solved) |
4:42PM |
0 |
Polycom headset tweaking |
4:37PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Meeting tomorrow Feb 12, 2005 |
4:22PM |
0 |
Audio issues with greetings & messages |
4:22PM |
0 |
/var/run/asterisk.ctl configuration |
3:08PM |
1 |
Still stuck trying to make Asterisk read MySQL |
2:39PM |
0 |
Quick How-To Guide for getting a Cisco 7960 going. |
2:22PM |
0 |
Delay answering inbound calls |
2:16PM |
2 |
Can agents login be permanent across Asterisk restarts ? |
2:06PM |
2 |
Codec Issue on IAX trunk? |
1:55PM |
1 |
Re: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins |
1:54PM |
1 |
differentiating busy & not connected |
1:35PM |
0 |
Cisco PRI to Asterisk & CallerID |
12:47PM |
3 |
Polycom IP 3000 configuration |
11:16AM |
1 |
SIP in the Philippines |
11:11AM |
0 |
polycom ip phones + asterisk |
10:48AM |
4 |
Setting a "Forward" to an external number on your phone |
10:35AM |
1 |
*.conf files not parsing |
9:36AM |
1 |
Asterisk-MySQL: Not loading voicemail config from MySQL |
9:05AM |
2 |
Question about DID |
8:28AM |
1 |
RE:mandrake linux install of zaptel |
8:07AM |
0 |
Asterisk as a UAC forwarded by SER |
7:20AM |
4 |
Weird Echo Problem |
7:16AM |
2 |
Menu Selections Only Work Internally |
6:55AM |
8 |
chan_capi and asterisk |
6:33AM |
2 |
transferring a IAX call into a conference |
6:10AM |
2 |
chan_capi or chan_mISDN vs bristuff |
6:04AM |
0 |
Not register SIP and IAX |
5:56AM |
3 |
Newbie: ISDN E1 the same in all countries? |
5:45AM |
0 |
How can agent logout manually ? |
5:31AM |
0 |
zaphfc - problems with hangup detection? |
5:25AM |
0 |
Transfers to engaged extensions |
5:24AM |
3 |
Dial and congestion |
4:26AM |
0 |
Help with dial command and h, H and g parameters |
3:29AM |
0 |
Multiple incomming contexts |
2:54AM |
0 |
Proper handling of incoming IAX/SIP callerids to be able to call back - why is calleridnum stripping dots out of number ? |
2:40AM |
1 |
How to monitor externip automatically? |
2:09AM |
6 |
i want to load chan_h323.so |
|
Thursday February 10 2005 |
Time | Replies | Subject |
10:35PM |
1 |
Bri problem |
10:15PM |
1 |
[Asterisk-Dev] Asterisk not accepting multiple SIP phone logins |
10:07PM |
2 |
Asterisk not accepting multiple SIP phone logins |
9:35PM |
4 |
asterisk as sip client behind nat |
8:45PM |
0 |
CISCO CP-7902G and chan_skinny. |
8:25PM |
0 |
FW: really easy FOP asterisk@home question |
7:26PM |
3 |
Voice Recognition |
6:57PM |
4 |
Why echo occurs |
5:34PM |
1 |
Fail to detect DTMF over direct ISDN pri link |
5:29PM |
2 |
TelIAX troubles |
5:00PM |
2 |
Searchable Mailing Lists & NooB Question |
4:31PM |
1 |
No dialtone in a E1 |
4:30PM |
1 |
Proper Contexts in extensions.conf |
4:29PM |
1 |
Codec passthrough patch for IAX |
3:31PM |
0 |
Context fails so falling back to extension " s" ? |
3:28PM |
0 |
asterisk GUI's that supports zap fxs extensi ons |
3:05PM |
1 |
Asterisk - SER Configuration |
2:52PM |
0 |
Logging agents in and out via manager API or other utility. Is it possible? |
2:33PM |
1 |
Dial SIP peers |
2:31PM |
4 |
Round Robin Strategy doesn't seem to work |
2:12PM |
0 |
Context fails so falling back to extension "s" ? |
2:00PM |
1 |
WAS: Strategy for a stable IAXy NOW: IAXy vs old P-3 |
12:10PM |
1 |
really easy FOP asterisk@home question |
11:57AM |
1 |
Problem with SPA-2000 and Asterisk 1.0.5 |
11:43AM |
0 |
ADM 0.7 Released |
11:39AM |
6 |
Wireless LANs and Asterisk |
11:30AM |
0 |
asterisk features |
11:29AM |
0 |
(no subject) |
10:55AM |
4 |
Debian way of compiling zaptel kernel modules |
10:46AM |
2 |
dtmfmode and IAX protocol |
10:41AM |
2 |
Asterisk on RedHat/AMD |
10:29AM |
2 |
Strategy for a stable IAXy |
10:25AM |
3 |
General Inbound Calls |
10:14AM |
1 |
/dev/dsp blocked |
9:45AM |
1 |
SER Asterisk Voicemail |
8:47AM |
0 |
asterisk GUI's that supports zap fxs extensions |
8:32AM |
0 |
Asterisk 1.0.5 won't pick up incoming calls |
7:56AM |
0 |
Using asterisk on a single phone line |
7:47AM |
1 |
Need help with a Cisco 7960 |
7:24AM |
0 |
Tormenta 2 Card number rotary switch |
7:12AM |
2 |
Configuring Asterisk |
7:10AM |
12 |
asterisk@home scary log |
6:44AM |
0 |
A working config for For FX100P Cards in United Kingdom ? |
6:09AM |
1 |
Cisco7960/SCCP Transfer Help? |
4:49AM |
2 |
Detect hangup |
4:07AM |
1 |
Asterisk and Fedora Core 3 |
3:50AM |
0 |
7940 VM DTMF not detecting |
3:11AM |
2 |
Softphone..easy to use ? |
2:55AM |
1 |
SIP proxies & Asterisk ? |
1:55AM |
0 |
Please share the experience on VoIP phones heavyusing. |
1:44AM |
4 |
why asterisk is replying 404 Not Found |
1:30AM |
0 |
Manager API - Call Transfer/Blind Transfer |
|
Wednesday February 9 2005 |
Time | Replies | Subject |
10:43PM |
0 |
Voicemail timeouts after 30sec's everytime no matter what I set in the configs. CVS Dec 04 |
10:33PM |
1 |
CallPickup from SIP phone |
9:31PM |
2 |
reboot polycom 1.4.1 |
8:19PM |
1 |
Asterisk consultants directory |
8:14PM |
1 |
Please share the experience on VoIP phones heavy using. |
6:54PM |
2 |
Melbourne Asterisk Users meet next Thursday |
5:28PM |
2 |
Asterisk and Sipura SPA-841 SIP phones |
5:05PM |
2 |
sample REGEX's for astcc |
3:27PM |
1 |
looking for responsible iax provider, aftermath |
3:22PM |
1 |
TDM400P FXO lines problem |
3:15PM |
0 |
logging events with time stamps |
3:02PM |
2 |
Zombie SIP channels |
2:05PM |
0 |
How to map zap channels to ISDN extensions on queues? |
2:04PM |
1 |
voice delay after call setup, outgoing calls |
1:35PM |
1 |
wcte11xp Trouble |
1:20PM |
6 |
Cisco 7960 Beating a Dead Horse |
12:59PM |
3 |
Multiple SIP registrations for one account? |
12:46PM |
2 |
Startup Question |
12:43PM |
1 |
Re: Asterisk Compile Problem on Red Hat 9 solved |
12:19PM |
0 |
FireFly + G729 license |
12:04PM |
0 |
Why does Asterisk Hangup cause server to freeze? |
12:02PM |
5 |
Getting SPEEX to work |
12:02PM |
1 |
Asterisk Compile Problem on Red Hat 9 resolved |
11:54AM |
1 |
Asterisk Versioning |
11:49AM |
0 |
Wait for Digits.. solved |
11:44AM |
1 |
Wait for Digits |
11:42AM |
1 |
Re: Newbie help/pointers required -configure xlite with asterisk |
11:34AM |
1 |
Analogue Line to Asterisk (Which Digium Model???) |
11:26AM |
1 |
SIP / IAX ActiveX |
10:52AM |
0 |
polycom ip300 |
10:48AM |
0 |
TDM400P FXO - Any one got it working well in UK without Hangup problems |
10:30AM |
4 |
IAX Voice Quality Issues |
10:07AM |
3 |
ISDN in Spain |
9:54AM |
1 |
How do I match a "D"? (Was: RE: In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?) |
9:49AM |
0 |
Background() ignoring digits A-D (Was: RE: How do I match a "D"?) |
9:45AM |
0 |
Using Asterisk as sip user agent with more than one device |
9:44AM |
1 |
Asterisk and SER Integration together |
9:23AM |
1 |
problem with running ztcfg |
9:21AM |
0 |
How do I match a "D"? (Was: RE: In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?) |
8:54AM |
2 |
How do I match a "D"? (Was: RE: In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?) |
8:46AM |
1 |
SIP ActiveX |
8:45AM |
1 |
Is there a Caller ID issue in the latest CVS Stable |
8:44AM |
1 |
loader.c:301 __load_resource: libpt_linux_x86_r.so.1.8.1: cannot open shared object file... [solution found, but quick question] |
8:21AM |
0 |
Asterisk CVS stable (current) crashes on remote user (over CAPI) pressing # or * when in conference |
8:11AM |
0 |
VoIP guide for business people |
8:10AM |
5 |
polycom soundpoint ip 300 |
8:03AM |
2 |
Problem with meetMe |
8:00AM |
4 |
G.729 codec for X-lite soft phone |
7:17AM |
6 |
IAX <=> FWD down again? |
6:58AM |
1 |
DTMF Payload Type Compatability |
6:01AM |
9 |
Web based Asterisk management tool |
5:44AM |
1 |
limit iax calls |
5:36AM |
1 |
calling problem in cvs verison on fedora core2 |
4:57AM |
0 |
Asterisk and SIPphone won't cooperate |
2:50AM |
0 |
incoming h323 calls, routed to SIP/H323 drop after connection |
2:40AM |
2 |
Problem using TDM400P FXS card |
2:21AM |
1 |
Error compiling app_icd |
1:35AM |
2 |
Asterisk Compile Problem on Red Hat 9 |
1:20AM |
1 |
add_pppd dialout problems |
1:04AM |
1 |
Asterisk as VoIP gateway |
12:35AM |
0 |
incoming call high failure rate on pickup of call. |
|
Tuesday February 8 2005 |
Time | Replies | Subject |
11:52PM |
1 |
sip_notify.conf |
11:09PM |
1 |
Fastagi question |
10:24PM |
2 |
Asterisk connected to pbx |
10:24PM |
2 |
bri dropping calls |
9:44PM |
1 |
Unable to load module iax.conf |
8:56PM |
1 |
SPA-841 MWI |
8:33PM |
2 |
giving up on x100p in Australia |
8:10PM |
0 |
InterFone IF-102/104? |
7:48PM |
1 |
Voip as a secure service? |
7:26PM |
2 |
Caller ID Question |
5:34PM |
1 |
Callerid to set time on phone? |
4:40PM |
1 |
Asterisk causing server to hang ... any hints? |
3:57PM |
1 |
breaking friends into users & peers |
3:52PM |
1 |
SIP Qualify/Status – What kind of numbers are you getting? |
3:30PM |
1 |
astcc with multiple access |
3:23PM |
0 |
Codec negotiation problems |
3:20PM |
0 |
DIAX version 0.9.10a available for download |
1:48PM |
1 |
No dial tone... |
1:14PM |
1 |
TDMO4B, GSM Gateways and CallerID |
1:04PM |
0 |
SPEEX CODEC and Voicepulse |
12:56PM |
0 |
Polycom/sip.conf/voicemail configurator |
12:29PM |
1 |
Digium TDM400P 4xFXO |
12:16PM |
0 |
Can someone tell me why I'm getting these? ( mailing list probe message) |
12:15PM |
1 |
Bug? Background() doesn't recognize D tone. |
12:06PM |
0 |
CODEC declarations in IAX.conf |
12:06PM |
2 |
Can someone tell me why I'm getting these? (mailing list probe message) |
12:05PM |
0 |
Confusing Contexts using AMP |
11:53AM |
2 |
Spaces in config files?? |
11:44AM |
3 |
Looking for FXS device - CISCO ATA 186 |
11:38AM |
0 |
attended call transfer in 1.0.5 |
11:32AM |
1 |
Can only call VoIP SIP Providers (Weird) |
11:23AM |
3 |
announcement: astfax 1.0 |
11:10AM |
1 |
Asterisk performance monitoring |
11:09AM |
2 |
Polycom screwed up Messages button in 1.4.1? |
11:05AM |
11 |
More complicated huntgroups / delayed ringing |
10:34AM |
3 |
stable combination of versions for asterisk and chan_oh323? |
10:19AM |
1 |
Digium TDM400p troubles |
10:11AM |
1 |
how to make g.729 preferred, but failover to gsm |
9:10AM |
0 |
Re: Asterisk-Users Digest, Vol 7, Issue 113 |
9:09AM |
1 |
Music on hold is a durge |
8:59AM |
1 |
faxing digium? |
8:40AM |
12 |
SRV lookups |
8:10AM |
1 |
How do I match a "D"? (Was: RE: In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?) |
8:09AM |
1 |
ASTCC simultenous calls per card |
7:55AM |
0 |
codec order, does it matter |
7:43AM |
1 |
SER Interaction: Agents and Extensions |
7:35AM |
1 |
DASS II cards supported |
7:23AM |
0 |
Bristuff - analogue communication over ISDN |
7:21AM |
1 |
Linux OS platforms |
7:17AM |
1 |
AreskiCC Installation -- Please Help |
7:16AM |
2 |
Using a Dual WAN Load Balancing Device |
7:04AM |
0 |
Asterisk FXS & SMDI for Octel access |
6:44AM |
4 |
In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone? |
6:21AM |
0 |
Fw: Help on Load Testing |
6:14AM |
4 |
how to pop up called number details using php scripts in agi scripts |
6:04AM |
3 |
Q: ISDN / E1-PRI - fax problems - Receiving and setting of Service Indicator (SIN) / Bearer Capability (BC) / High Level Compatibility (HLC) / Low Level Compatibility (LLC) |
5:47AM |
1 |
Snom programmable leds / keys usage for pickup groups? |
5:27AM |
1 |
VoIP extn number planning |
4:50AM |
2 |
How to xfer calls or is my setup wrong? |
4:26AM |
4 |
high-quality, high-bandwidth codecs? |
4:23AM |
3 |
live monitoring (SIP only) |
4:21AM |
2 |
Voicemail not working properly |
4:00AM |
3 |
SIP jitter? |
3:56AM |
1 |
DTMF CLIP in Sweden and others |
3:17AM |
0 |
Help on Load Testing |
2:56AM |
1 |
Question about TDM11B Configuration |
2:47AM |
1 |
CVS or release? |
2:37AM |
2 |
MD5 in SIP's "register => ..." |
2:28AM |
2 |
Asterisk and Sipgate problem... |
2:01AM |
1 |
bristuff and audio drop outs (5 sec and longer) |
12:46AM |
5 |
jitterbuffers - suggested settings |
12:14AM |
0 |
Drop and Insert ? |
|
Monday February 7 2005 |
Time | Replies | Subject |
11:35PM |
0 |
Howto( CLI or called number is attached to a database which automatically updates records let suppose if some dials xxxxxxx number so Company X's database record pops up on the computer screen of agent) |
10:34PM |
2 |
Best OS for Asterisk--newbie!!! |
7:50PM |
1 |
Conferencing without Meetme |
6:57PM |
0 |
kphone and * |
5:34PM |
1 |
Voicemail timeouts after 30sec's everytime. |
4:28PM |
0 |
Call Monitoring on IAX Channels - ChanSpy |
4:28PM |
3 |
SIPP load testing - unexpected message - anyone using sipp sucessfully ? |
3:48PM |
6 |
SIP port blocked in Dubai ? |
3:46PM |
2 |
How to number extensions - Which way is best? |
3:32PM |
1 |
In-band disconnect problem (legacy PBX) - asterisk doesn't hear t he touchtone? |
3:28PM |
1 |
Asterisk and SER differences |
2:58PM |
0 |
ot: WiSIP with zyxel firmware |
2:53PM |
2 |
asterisk to asterisk communication |
2:38PM |
0 |
IAXy Heat? Aluminum case anyone? |
2:32PM |
0 |
seeking references |
1:38PM |
0 |
newbie setup |
1:31PM |
0 |
Informations |
1:20PM |
2 |
2 x Fritz!pci card |
12:58PM |
0 |
RE: Asterisk-Users Digest, Vol 7, Issue 93 |
12:45PM |
2 |
Record() cut off after 40 sec |
12:45PM |
0 |
Asterisk 1.0.1 - CCM 3.0.3 - GNUGK 2.0.8 - OpenH323 |
12:40PM |
2 |
Broadvoice issues |
12:16PM |
2 |
Zaptel down after upgrade. |
11:51AM |
2 |
Asterisk on a single phone line |
11:36AM |
2 |
*HOWTO* : using mime-construct with outlook - send fax to email recipient |
11:27AM |
0 |
can not dial problem |
11:23AM |
2 |
no sound playing vm greetings and options |
11:17AM |
1 |
Provider with GA DIDs and LNP |
11:00AM |
0 |
Calls from SIP ATA to CO |
10:57AM |
0 |
most popular addons? |
10:53AM |
5 |
TDM-400P and Grandstream Question |
10:28AM |
0 |
Asterisk success histories in business? |
10:04AM |
1 |
CDR, RingTo Number, and DST |
9:48AM |
0 |
Incomming Fax deteccion |
9:14AM |
0 |
Recording already started on that call |
8:53AM |
1 |
Group=???? |
8:45AM |
1 |
Asterisk => SKYPE |
8:10AM |
3 |
incoming calls in h323 do not come to right dialplan |
6:48AM |
1 |
Festival patch |
6:36AM |
1 |
Incoming Call Problem |
6:19AM |
2 |
callback agents cannot transfer calls |
6:04AM |
1 |
PocketPC Softphone? |
5:29AM |
3 |
ADM 0.5 - Asterisk Desktop Manager (alpha) |
4:23AM |
2 |
Pro biz Asterisk |
4:20AM |
1 |
multiple nics and internet |
4:06AM |
1 |
How to Create customized audio file to use withASTCC?? |
3:22AM |
1 |
Remote MWI via IAX? |
2:48AM |
4 |
Newbie help/pointers required - configure xlite with asterisk |
2:45AM |
0 |
Small PHP script for displaying * CID database in Cisco 7940/60 XML |
2:08AM |
1 |
Failed to query database. Check debug for more info |
1:28AM |
0 |
RealTime Configuration for extensions.conf |
12:55AM |
0 |
TDM400P FXS works only if two lines are off hook? |
12:45AM |
7 |
IAX2 Trunk Problems with NAT |
|
Sunday February 6 2005 |
Time | Replies | Subject |
11:49PM |
0 |
wanted: sample config' using GOTOIF's for all features for a roll-out |
11:46PM |
0 |
re: difference between STUN servers and far-end solutions |
11:44PM |
0 |
Which version of asterisk-oh323 should i use with asterisk v1-0-5. |
10:38PM |
0 |
Intel 537EP is NOT the MD3200 aka X100P [Re: Intel 537EP chipset, revisited] |
9:23PM |
0 |
"whispering" mode in Meetme? |
5:42PM |
1 |
IAX2 Bandwidth Study |
4:56PM |
1 |
Soft keys and transfer problem on Sayson 480i |
4:37PM |
0 |
passing "*" into a dial plan |
3:45PM |
1 |
Understanding the "Hint" priority. |
2:47PM |
1 |
no caller ID presented from 12SP+ |
2:19PM |
0 |
Fax-modem |
2:08PM |
0 |
Using Asterisk to monitor in/out calls (single line) |
2:01PM |
0 |
SIP URI modified unexpectedly! Is that a router problem? |
1:52PM |
1 |
Voicepulse DNID is blank - Any other options? |
1:51PM |
8 |
snom soft phone |
1:31PM |
1 |
Call forwarding of IAX inbound call |
1:03PM |
1 |
Call status after Answer |
12:55PM |
4 |
Autodetecting faxes |
12:17PM |
0 |
blindxfer not in stable 1.0.5? |
11:24AM |
1 |
Cisco 12SP+ firware anyone? |
10:45AM |
3 |
iax2-jitter-trunking? |
7:07AM |
0 |
FYI - New firmware from Sipura |
6:06AM |
1 |
Proxied SIP |
5:56AM |
3 |
Question about X100P card |
5:13AM |
1 |
Help with extensions |
3:20AM |
0 |
Xorcom Rapid 1.0 released |
2:56AM |
0 |
IAXy ring frequency |
1:46AM |
3 |
inter asterisk |
1:45AM |
2 |
Need help with perl script/agi for ringback |
|
Saturday February 5 2005 |
Time | Replies | Subject |
4:17PM |
0 |
ISDN Phones With Asterisk |
3:13PM |
0 |
Problems with SIP invite due to long ping round trips |
1:43PM |
2 |
Question about VoIP providers |
1:16PM |
0 |
Beep every 5 seconds |
12:35PM |
1 |
RTC Client (maybe VAD related) |
9:52AM |
1 |
cannot dial non-local numbers (junghanns QuadBRI cards) |
9:10AM |
1 |
OT: FWD and IAX: down? |
8:48AM |
2 |
Siemens C200 phone - callerid not visible on FXS |
6:14AM |
1 |
asterisk@home basic |
5:55AM |
0 |
[Fwd: D-Link DVG-1402S VoIP Router] |
5:19AM |
3 |
ISDN X-Over |
5:09AM |
0 |
Question about VoIP Solution |
5:01AM |
0 |
Inbound SIP to demo context |
4:49AM |
1 |
TAPI integration with * using Identapop software |
3:11AM |
1 |
CallerID and anonymous SIP calls |
|
Friday February 4 2005 |
Time | Replies | Subject |
10:03PM |
1 |
ASTCC error on free calls |
8:40PM |
0 |
Need some Advise |
8:23PM |
2 |
MYSQL Failed |
6:26PM |
2 |
Encrypted VOIP? |
6:24PM |
0 |
Re: Can't get Polycom auto-answer to work (Solved) |
5:06PM |
0 |
codec0 = 516 is not codec1 = 216 |
4:42PM |
3 |
FIX YOUR AUTO-RESPONDERS!!! |
4:19PM |
1 |
toll-free anonymous |
4:10PM |
1 |
Polycom Auto-Answer and Call Transfers |
3:40PM |
3 |
No ring tone on Outgoing calls |
3:33PM |
1 |
Multi Office Configuration |
3:19PM |
7 |
Limit MOH processes |
2:44PM |
1 |
vicidial and mysql ........help |
2:30PM |
1 |
External Callforward (Vanity CLI) |
2:28PM |
2 |
AU caller ID with Sipura SPA-3000 |
2:18PM |
1 |
autoAnswer and autoAnswerLogin? |
2:18PM |
1 |
Call pickup across technologies (SIP, IAX, MGCP)? |
2:16PM |
0 |
Conference Bridge? |
1:47PM |
0 |
DTMF Problem with analog phones |
1:00PM |
0 |
Updateing to Stable from CVS |
12:21PM |
3 |
Server Criteria |
12:03PM |
0 |
patch for chan_capi error condition report when receiving CAPI_CONF:CAPI_LISTEN message |
11:49AM |
0 |
FC2 RPMS are updated |
11:16AM |
9 |
callback on busy |
10:32AM |
0 |
2 x100p + Static + echo |
10:23AM |
1 |
Snom Phones Volume |
9:43AM |
4 |
BRI in the US? |
9:29AM |
1 |
*, BeroNet BN4S0 and misdn - problems |
9:23AM |
3 |
Callerid problems with 1.0.5 |
9:15AM |
2 |
zapata.conf ERROR?????? please help |
9:05AM |
3 |
PCMCIA card |
8:52AM |
5 |
IAX2 register Refresh |
8:44AM |
0 |
Specify a codec in dial plan? |
8:40AM |
0 |
TMD card to buy. |
8:25AM |
0 |
manager api - Async:True? |
8:19AM |
4 |
HP ProLiant server for Asterisk |
8:12AM |
1 |
echo's + cheap phones |
8:10AM |
2 |
No Playback() when Digicom TE110P enabled |
8:02AM |
1 |
X-lite to Cisco ATA - no RTP |
7:43AM |
2 |
How to Create customized audio file to use with ASTCC?? |
6:45AM |
4 |
T.38 bounty |
6:44AM |
2 |
gsm audio files |
5:59AM |
1 |
Microsoft RTC Client SDK with Asterisk |
4:45AM |
2 |
Swap Memory get used totally |
4:16AM |
3 |
Bristuff and incoming call problems |
3:22AM |
1 |
Intertex IX66 incoming IAX |
2:49AM |
4 |
ASTCC Apllication |
2:08AM |
2 |
New Asterisk user with a goal |
2:07AM |
1 |
Q: how to receice the number of the called party back? |
2:00AM |
1 |
Q: charge info on E1-PRI |
1:35AM |
3 |
why asterisk and ser |
12:36AM |
0 |
(no subject) |
|
Thursday February 3 2005 |
Time | Replies | Subject |
11:54PM |
1 |
Help with chan_h323 |
7:32PM |
1 |
MWI with IAX |
7:13PM |
0 |
Dial timer problem? Short rings. |
6:21PM |
2 |
queue-timeout- press button to remain on hold |
4:44PM |
0 |
zttool user's manual & HDLC Abort errors |
4:18PM |
1 |
Q: How to get the preset callerid from a CLID-no-screen E1-PRI |
3:55PM |
0 |
Australian Caller ID with Sipura SPA-3000 |
2:16PM |
3 |
Can't get Polycom auto-answer to work |
1:57PM |
2 |
How to charge for Asterisk installations and ongoing support? |
12:45PM |
2 |
Good 800 Number provider |
12:44PM |
5 |
OT: How to "own" a telephone number? |
12:37PM |
1 |
FastAgi Help |
12:34PM |
1 |
DTMF Payload type |
12:22PM |
1 |
Incoming call not ringing |
12:09PM |
1 |
AMP with SUSE9.2 (Apache2) |
12:06PM |
1 |
Mi extensions keeps ringing |
12:01PM |
0 |
AsteriskBrasil.org - We have an email list!!! |
11:59AM |
2 |
E&M Wink problems |
11:56AM |
1 |
Multiple mailbox on the same SIP extension |
11:56AM |
1 |
E1's and span - what questions to ask my service provider |
11:44AM |
1 |
Forwarding voicemail messages |
11:41AM |
0 |
Everyone is busy/congested |
11:25AM |
2 |
Odd behaviour between Grandstream and Xlite |
11:16AM |
3 |
Question about wildcard T1 card |
11:03AM |
0 |
DTMF Payload Type: |
10:02AM |
4 |
astcc digit timeout |
9:51AM |
3 |
Where are chan_capi bug reports and bugfixes sent? |
9:28AM |
0 |
Difference between Asterisk and VOCAL |
9:08AM |
0 |
Call Forward Loop |
9:04AM |
0 |
Different rings |
8:57AM |
4 |
Asterisk Dialplan command "PPPD" released |
8:47AM |
1 |
free pocketPC softphone (toshiba e750) |
8:17AM |
0 |
Automated CallbackLogin |
8:15AM |
2 |
Individual contexts pending on Caller-ID? |
7:51AM |
0 |
Busy Extension Ring to alternate. |
7:37AM |
3 |
good god! stop the damn auto-replys! |
6:20AM |
3 |
Asterisk crashes from time to time |
6:18AM |
5 |
Cisco 7960G phone crashes during SIP upgrade |
5:32AM |
0 |
Error on compiling oh323 |
5:30AM |
0 |
Grandstream ATA 486 works only with ulaw and alaw codecs. |
4:37AM |
1 |
Nortel i2004 support asterisk? |
3:50AM |
0 |
Incoming SIP calls with different signaling and RTP IP addresses |
3:43AM |
1 |
How to forward a call to the same ISDN box ? |
3:40AM |
1 |
403 Forbidden when registering sip user database on backend |
2:22AM |
0 |
Special "error" numbers |
12:37AM |
3 |
IAX dns lookups |
12:01AM |
0 |
key in number after 'h' extension |
|
Wednesday February 2 2005 |
Time | Replies | Subject |
11:10PM |
1 |
BRI only 2 calls |
10:48PM |
1 |
TDM series + kernel 2.6 |
10:38PM |
0 |
tuning for ulaw g.711 - Polycom IP500 |
10:02PM |
2 |
using the MYSQL command to insert a record |
9:58PM |
1 |
Asterisk problems behind firewall |
8:08PM |
1 |
Calling Asterisk Autoattendant With SIP Phone |
7:49PM |
2 |
outbound 911 calling |
6:56PM |
2 |
HEEEELP!!!!!!!! with file CODEC_G729.SO |
6:33PM |
2 |
How to download CVS with attended transfers |
5:00PM |
2 |
different IAX ports for different contexts |
3:21PM |
1 |
Using Asterisk to Find a Live Person |
3:09PM |
2 |
MeetMe & ztdummy |
2:49PM |
2 |
Broadvoice problems with outbound calls {Scanned} |
2:44PM |
0 |
Problemas with Basic Services. |
2:30PM |
0 |
Speex pass through on SIP |
2:15PM |
9 |
911 and Cops knocking on my door |
1:55PM |
1 |
Reproducible crash with CVS stable (from about 5 days ago...) - but only from iax clients |
1:36PM |
2 |
Asterisk@home - problem getting console output ... |
1:10PM |
0 |
RES: AgentLogin / AgentCallbackLogin transfer pro blem |
12:46PM |
2 |
how to add more TDM04B |
12:34PM |
2 |
Installation on Fedora 3 |
12:07PM |
1 |
ZapTel Errors on boot |
10:49AM |
2 |
Asterisk with SourdCard |
10:45AM |
0 |
32 FXO 32 FXS and record call |
10:21AM |
0 |
rxgain won't always ring extension |
10:19AM |
1 |
[OT - somewhat] chan_sccp status |
10:17AM |
0 |
AgentLogin / AgentCallbackLogin transfer pro blem |
10:07AM |
0 |
DTMF outbound problem with ata 186 |
10:05AM |
0 |
Does any Cisco VoIP kit support IAX? |
9:57AM |
0 |
Paging Zultys Phones |
9:39AM |
1 |
(OT:) Tool for trying/troubleshooting WAN/LAN |
9:21AM |
0 |
AgentLogin / AgentCallbackLogin transfer problem |
8:01AM |
4 |
* not hanging up when call from POTS to IAX phone |
7:55AM |
8 |
howto answer a call in a queue |
7:55AM |
0 |
clicktocall via manager with cisco 7905 |
7:50AM |
0 |
403 forbidden error |
7:38AM |
0 |
Ignoring too old packet packet |
7:13AM |
1 |
Cisco 7940 [SIP], DTMF and Voicemail |
7:07AM |
0 |
IAXy Configuration for Alternate Server |
7:04AM |
1 |
Transfer call digit length |
7:02AM |
2 |
Disabling native bridging for IAX calls |
6:45AM |
0 |
ExtensionState problems using Manager.conf API |
6:23AM |
0 |
SIP Call through Asterisk |
6:05AM |
6 |
problem in compiling asterisk-addons |
5:54AM |
0 |
Integration Asterisk and Siemens Hicom 150 |
5:29AM |
1 |
Asterisk waits 4 rings before FXO answers incomingcall |
5:03AM |
1 |
Hangup detection with TDM400 in UK |
4:43AM |
0 |
Asterisk waits 4 rings before FXO answers incoming call |
4:33AM |
1 |
Asterisk cmd SayNumber : how to pronounce in another language - we say "one-and-twenty" instead of "twenty-one" |
4:21AM |
1 |
X100P Setup |
3:49AM |
3 |
Reccomendation for reliable handsets |
3:40AM |
2 |
Installing ASTERIS@HOME, How to install on text mode same help? |
3:01AM |
1 |
SIP with Delay |
3:00AM |
0 |
ZAPHFC Drop calls |
2:59AM |
1 |
Astrerisk + Conversation OneWay |
2:23AM |
2 |
Forbidding ZAP interface bridging |
12:23AM |
0 |
How to continue execution after called party hangs up? |
12:07AM |
4 |
new install |
|
Tuesday February 1 2005 |
Time | Replies | Subject |
11:05PM |
0 |
chan_capi and G711u |
10:19PM |
0 |
Outbound proxy |
10:10PM |
0 |
call forwarding with code |
9:56PM |
1 |
Is Bell HDSL in Ontario good solution for VOIP? |
9:24PM |
11 |
load balancing 20 asterisk servers |
9:19PM |
4 |
astGUIclient users should not upgrade to Asterisk 1.0.5 |
8:33PM |
2 |
Problems compiling zaptel on SuSE V9.2 |
8:23PM |
2 |
X100P not hanging up... |
7:57PM |
1 |
Why is host= being ignored in sip.conf ? |
6:51PM |
1 |
chan_sip.c:7296 handle_request: Unable to create/find channel |
5:53PM |
1 |
3G Video Mobile Phone |
5:32PM |
1 |
list administrator.....??? |
5:25PM |
0 |
Realtime and callforwarding |
4:52PM |
1 |
PCI CARD X100P CLONE FXO WORK ALSO AS FXS ? |
4:36PM |
0 |
how to make a call with asterisk from shell ? (orwith a .sh file ) |
4:29PM |
1 |
Play tone till first digit read |
4:21PM |
5 |
Terrible inbound call quality vs. outbound |
4:02PM |
0 |
Help with DIAL command |
3:41PM |
1 |
how to make a call with asterisk from shell ? (or with a .sh file ) |
3:36PM |
6 |
*ASTERISK* Install and configure Step by Step. |
3:15PM |
1 |
Custom MusicOnHold |
2:37PM |
1 |
choppy sound after 15 minutes in a call |
2:10PM |
2 |
Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 caller id? |
1:46PM |
1 |
AGI global style variables |
1:44PM |
0 |
Odd error.. sip_xmit Bad file descriptor |
1:12PM |
2 |
Soft phones that _actually_ work under Linux? |
1:02PM |
0 |
Crash: Call from IAX-client to a distribution where the IAX-Client is in |
12:54PM |
1 |
Re: Asterisk-Users Digest, Vol 6, Issue 325 |
12:50PM |
8 |
Outlook Integration |
12:42PM |
2 |
IAX2 Softphone |
12:30PM |
2 |
Asterisk Not hanging up DS0 when number called is busy. |
12:07PM |
0 |
AGI two calls - one hangs up - othere gets "interrupted system call" |
11:32AM |
0 |
newbie: questions |
11:27AM |
0 |
TBM400 no callerid on incoming calls? |
11:21AM |
3 |
Linksys PAP2 / RT31P2 + multiple G.729 calls |
11:13AM |
1 |
HFC-5/S + Asterisk |
10:59AM |
0 |
OT: IAX provider for business |
10:52AM |
0 |
manager api events (pri vs pstn) |
10:43AM |
1 |
Zap channel occasionally misses dialing thefirst digit |
10:40AM |
1 |
Scope of definitions |
10:35AM |
2 |
Outbound calling with TDM400P |
10:13AM |
0 |
how to add extension to mysql database |
10:12AM |
0 |
ChanSpy? |
9:55AM |
1 |
FW: Messaging with * and eyeBeam |
9:34AM |
0 |
Messaging with * and eyeBeam |
9:31AM |
0 |
One extension, multiple endpoints |
9:13AM |
0 |
Call Forward - Need Help |
9:03AM |
3 |
Zap channel occasionally misses dialing the first digit |
8:55AM |
0 |
Limiting no. of calls on one channel |
8:49AM |
1 |
Germany specific settings for Grandstream ATA286 - Polarity reversal, impedence and onhook voltage |
8:46AM |
0 |
VoiceMail ANI question |
8:43AM |
1 |
Different ring when called by door entry |
8:35AM |
2 |
IAX native transfers |
8:26AM |
2 |
mysql based adressbook with agi and web interface ? |
8:16AM |
0 |
No Sound Playback |
8:12AM |
0 |
RE: Re: RE: Answering Machine Function? |
8:03AM |
2 |
Feature automon |
6:55AM |
1 |
SIP Challenge response bug? |
6:40AM |
3 |
X100P Clone |
6:34AM |
1 |
i4l: Quality of Voice |
6:05AM |
0 |
Troubles with Macro-stdexten and dial |
6:00AM |
0 |
Asterisk Services working with SER !!! |
5:09AM |
5 |
IAX registration keep alives |
4:51AM |
2 |
Error on compiling oh323 0.6.5 on cvs stable asterisk |
4:45AM |
1 |
MeetMe missing? |
4:26AM |
1 |
i4l + SIP: Audio One-way |
4:17AM |
2 |
asterisk remote monitor |
4:13AM |
0 |
How to mark calls for inclusion in CDR ? |
3:58AM |
0 |
24 CTU ringtone for grandstream 101? |
3:36AM |
0 |
Call queue ackcall doesnt work |
3:17AM |
0 |
Bangkok DID? |
2:43AM |
0 |
Actions taken drugin calls - are there any other keys active beside # for transfer ? |
2:12AM |
2 |
How to compile "iaxclient" with MinGW/Cygwin |
1:26AM |
1 |
Where to download the soxmix please? |
1:23AM |
1 |
broken message waiting indicator on Polycom IP600? |
12:19AM |
2 |
AGI Script for CID Rewrite and CID Name lookup |