| Monday October 31 2005 |
| Time | Replies | Subject |
|---|
| 11:43PM |
4 |
Asterisk 1.2.0-beta2 Released |
| 10:40PM |
1 |
What's the deal with "secret=" vs. "password="? |
| 8:30PM |
0 |
Echo Canceller question- is there aviablesolution? |
| 8:14PM |
0 |
Strange problem with dtmf pound and star |
| 7:47PM |
1 |
HELP PLEASE: What A Pain RSA |
| 6:47PM |
0 |
Fw: Attended transfer restarting asterisk switch |
| 6:14PM |
2 |
echo codec related? |
| 5:58PM |
1 |
musiconhold -vs- musicclass problems setting the differnt class of music |
| 5:31PM |
0 |
problems with 1.2 Beta1 |
| 5:14PM |
3 |
spandsp patch |
| 4:51PM |
0 |
No D-channels available! CVS-HEAD-10/31/05-16:01 |
| 4:47PM |
0 |
Release Announcement: HooDaHek 0.7 |
| 2:46PM |
0 |
Any experiences with Orion hardware echo cancellers? |
| 1:47PM |
1 |
dial-out gives always "not found" (dial-in works fine) |
| 1:32PM |
5 |
queue scheduling... |
| 12:28PM |
1 |
pls help compile rx_fax (patch / Makefile) |
| 12:19PM |
1 |
Agent channels causing problems |
| 11:48AM |
4 |
Asterisk and Zaptel Versions Command? |
| 11:02AM |
13 |
llamdas por 4 lineas a eeuu tarifa plana |
| 10:55AM |
1 |
Caller ID to SIPURA-1000, 2000, 3000 Handset Prlblem in only showing the destination callerId |
| 10:51AM |
0 |
Calling Name Not Displayed On Incoming |
| 10:02AM |
2 |
Release of Asterisk 1.2 |
| 9:33AM |
0 |
(no subject) |
| 8:51AM |
1 |
chan.iax2.c errore |
| 8:27AM |
1 |
lucent TNT h323/sip config? |
| 8:26AM |
0 |
Add Contexts Dynamically |
| 8:25AM |
1 |
Adit 600 and Groundstart |
| 8:13AM |
2 |
Rugged VoIP phones for use with asterisk |
| 7:50AM |
1 |
Info on beta1 seem to be broke |
| 7:46AM |
1 |
How to remove a VM greeting - go back to default Allison message |
| 7:30AM |
5 |
Timestamps in Console? |
| 6:37AM |
1 |
FXS Disconnect Supervision (Kewlstart / Open Loop Disconnect) |
| 6:07AM |
0 |
[Asterisk Voicemail] Quota |
| 5:43AM |
0 |
IAX2 trunks encrypted? |
| 4:27AM |
3 |
Tone generator module |
| 3:43AM |
8 |
A2Billing |
| 3:11AM |
4 |
Segfault on latest head 10/31 |
| 3:05AM |
1 |
H323 one way audio using oh323 |
| 1:48AM |
0 |
can't add zap channels to a group |
| 1:33AM |
4 |
sip show peers |
| 12:40AM |
1 |
lilte help please |
| |
| Sunday October 30 2005 |
| Time | Replies | Subject |
|---|
| 11:25PM |
1 |
How to specify when to go to 102 priority |
| 10:38PM |
0 |
Re: Automathic call forwarding (Gianni (priv.)) |
| 10:18PM |
0 |
Channel Data Access |
| 8:20PM |
3 |
Asterisk to Avaya IP Office |
| 8:05PM |
1 |
zap group channels |
| 7:15PM |
0 |
app_txfax.so app_rxfax.so |
| 7:12PM |
1 |
Distinguishing Busy from No Answer |
| 5:27PM |
1 |
Call Transfer problems-am I missing something? |
| 3:44PM |
2 |
Can anyone explain reason for this echo |
| 3:10PM |
1 |
Attended transfer restarting asterisk switch |
| 1:14PM |
2 |
Cisco 7960 Skinny Firware |
| 1:00PM |
0 |
dialout gives 404 (using sjphone (dialin works fine)) |
| 10:09AM |
2 |
no sip peers after restarting asterisk? |
| 9:13AM |
1 |
Re: feature usage/digit detection |
| 8:27AM |
0 |
Automathic call forwarding |
| 7:44AM |
1 |
gotta be a dumb question... |
| 7:22AM |
1 |
Trouble with D-Link 2004S and E1 PRI |
| 7:09AM |
2 |
libpri |
| 7:03AM |
1 |
VoiceMailMain() in 1.2-beta |
| 4:57AM |
0 |
Pattern for matching CALLERID |
| 3:09AM |
3 |
SCCP support is making good progress |
| 12:38AM |
1 |
Announce in MeetMe Rooms |
| |
| Saturday October 29 2005 |
| Time | Replies | Subject |
|---|
| 9:57PM |
0 |
Access to channel data... |
| 8:56PM |
0 |
Asterisk 1.2 |
| 7:22PM |
1 |
ericsson pabx and digium card TE110P |
| 6:47PM |
1 |
chan_zap ignoring stuff in beta1? |
| 5:39PM |
1 |
Play music while on incoming connections ringing the internal user |
| 3:06PM |
1 |
I give up - Help with TE410P |
| 1:01PM |
2 |
zaptel + RH3? |
| 11:42AM |
1 |
Set outgoing MSN with chan_capi-cm |
| 10:20AM |
0 |
Meetme streaming a recording |
| 9:31AM |
1 |
Credit card machines, Asterisk and Digium any issues? |
| 7:57AM |
3 |
dropping extra frame of G.729 since we already have a VAD frame at the end |
| 6:07AM |
0 |
ANNOUNCEMENT: Asterisk-Java 0.2-rc2 released |
| 5:18AM |
2 |
CallBack Suggestion |
| 4:40AM |
2 |
Zyxel omni.net USB ISDN works with Asterisk |
| 4:25AM |
0 |
Prblem with 723 and 729 |
| 1:47AM |
2 |
UK Pounds and pence prompt wanted |
| 1:46AM |
0 |
Asterisk - PRI - Cisco |
| 1:33AM |
0 |
Repost:Generate white noise to avoid RTP timeout |
| |
| Friday October 28 2005 |
| Time | Replies | Subject |
|---|
| 11:27PM |
0 |
SPA3000 as trunk - no caller ID - solved |
| 10:04PM |
4 |
Sipura SPA 2000 - error using second line |
| 5:49PM |
1 |
Geneys |
| 5:11PM |
0 |
asterisk outgoing does not work |
| 5:07PM |
0 |
anyone using these? |
| 4:55PM |
1 |
Outbound fax solution |
| 4:39PM |
0 |
Warning - AgentCallBackLogin has changed, Possibly will cause 99% cpu |
| 3:27PM |
1 |
I need GoIAX or VoipBuster A@H examples? |
| 2:46PM |
0 |
Why can't I dial - just using SIP internally |
| 2:16PM |
1 |
SIP Host "Unspecified" |
| 1:52PM |
0 |
problem with asterisk-realtime-odbc |
| 1:20PM |
0 |
Metreos |
| 12:15PM |
2 |
Having Meetme call another conference |
| 11:22AM |
1 |
Sipura 841 echo cancel question |
| 10:49AM |
4 |
Opinions on IAX JitterBuffer in old-school 1 .0.0? |
| 10:42AM |
2 |
Mediatrix Gateways |
| 10:24AM |
0 |
Montreal Meet Asterisk Get-Together |
| 9:56AM |
2 |
Help with Zultys |
| 9:25AM |
2 |
Top and asterisk performance |
| 9:19AM |
0 |
Cell phone extension woes |
| 8:46AM |
2 |
2 problems |
| 8:33AM |
0 |
Prevent transcoding |
| 7:53AM |
1 |
Ouch - Error while writing audio data - broken pipe |
| 7:11AM |
2 |
IAX channel options |
| 7:11AM |
0 |
h323 no audio from the sip phone to the outside world. |
| 6:15AM |
1 |
when is 1.2 being released? |
| 5:59AM |
2 |
OT: Suggestions for E1 Service in the UK |
| 5:51AM |
2 |
SV: call queue |
| 5:45AM |
0 |
PhoneCALL v2.7 goes MultiLingual |
| 5:44AM |
1 |
call queue |
| 5:26AM |
8 |
"GSM cards" / "mobile phone cards" for Asterisk? |
| 4:56AM |
0 |
Anyone running zaptel's watchdog in production? |
| 4:35AM |
2 |
[Re] Re: Echo canceller on TE406 & Asterisk |
| 4:00AM |
1 |
Dial with 44 and +44 prefix |
| 3:35AM |
2 |
Help Installing Asterisk..... |
| 3:25AM |
3 |
ADSL |
| 3:08AM |
3 |
Webui to show registered phones |
| 3:07AM |
2 |
Asterisk GUI/web interfaces that don't change config files |
| 3:03AM |
0 |
URL Dialing from SNOM phone |
| 3:02AM |
0 |
Queues with SIP/phones as members, leave when empty? |
| 2:56AM |
0 |
IAX voice problem, no voice at all |
| 2:16AM |
1 |
chan_bluetooth and audio problem |
| 2:14AM |
2 |
Problem With Sipura |
| 12:59AM |
2 |
Console detach. |
| 12:22AM |
1 |
X100P doesn't show Caller-ID |
| 12:17AM |
0 |
AW: problem with receiving faxes over cisco as5300 |
| |
| Thursday October 27 2005 |
| Time | Replies | Subject |
|---|
| 11:29PM |
2 |
Echo Canceller question- is there a viablesolution? |
| 10:16PM |
0 |
{Resend} Process VoiceMail file to attach |
| 9:19PM |
2 |
TDM04B NEW CARD WITH zaptel 1.0.7 |
| 8:22PM |
2 |
Where does Asterisk put it's files |
| 7:01PM |
5 |
Echo canceller on TE406 & Asterisk |
| 6:52PM |
1 |
Taking the plung to CVS HEAD |
| 5:36PM |
0 |
call monitoring in external application (newbie) |
| 5:01PM |
1 |
PRI to SIP Problem |
| 4:12PM |
1 |
Opinions on IAX JitterBuffer in old-school 1.0.0 ? |
| 4:06PM |
2 |
Not saving voicemail message |
| 3:40PM |
0 |
polycom ip500 mwi, quite please |
| 3:30PM |
1 |
Zapbarge feature available? |
| 3:28PM |
2 |
Queue Login Out Question |
| 3:24PM |
2 |
problem with receiving faxes over cisco as5300 |
| 3:03PM |
1 |
Outgoing fax detect |
| 1:25PM |
1 |
Delay ReInvite |
| 12:54PM |
1 |
Is anyone using OpenSer - A fork of SER? |
| 12:41PM |
1 |
Words for the Asterisk community to live by. |
| 11:59AM |
0 |
sip not working suddenly |
| 11:55AM |
1 |
Is it possible to generate or play some white noise in Asterisk? |
| 11:16AM |
0 |
QoS Monitor |
| 11:03AM |
1 |
TDM01B vs. X100P |
| 10:33AM |
0 |
cannot get dialtone or ring on FXS ports (TDM400p) |
| 10:06AM |
1 |
Network Architecture Question |
| 9:38AM |
0 |
Have IAXy signal busy without losing ongoing call? |
| 8:15AM |
1 |
Test after Hurricane Wilma |
| 7:29AM |
0 |
R: Bristuff question |
| 7:22AM |
1 |
Bristuff question |
| 6:48AM |
0 |
Asterisk 1.2beta and te411p: incorrectly reporting sometimes "all channels busy" |
| 6:39AM |
1 |
Vmail.cgi and realtime? |
| 6:04AM |
1 |
Message Waiting Indicator and PRI |
| 4:58AM |
1 |
spandsp / txfax exit codes / logging? |
| 4:13AM |
1 |
Problems compiling asterisk zaptel for Asterisk 1.0.9 |
| 4:05AM |
0 |
PRI Echo - Solved with KB1 Patch |
| 3:21AM |
0 |
Overlap dial and "match as you go" = how to implement early dial on telco line |
| 1:01AM |
0 |
sKinny in database |
| 12:48AM |
1 |
please recommend phones with adsi. |
| |
| Wednesday October 26 2005 |
| Time | Replies | Subject |
|---|
| 11:29PM |
1 |
faxdetect on voicemail |
| 11:15PM |
0 |
make sipura stop generating stale nonce. Device comes in and goes out every 1 minute |
| 10:38PM |
1 |
Echo Canceller question- is there a viable solution? |
| 9:50PM |
0 |
Backend network, one-way audio...trunking |
| 9:17PM |
2 |
Aussie Call home! |
| 8:00PM |
1 |
Simple SIP only Asterisk Configuration |
| 7:36PM |
2 |
Trying to genereate dial tone, but stop after first digit dialed. |
| 6:53PM |
1 |
Zaptel stop hangs server |
| 6:03PM |
1 |
Asterisk IVR and Cisco Call Manager |
| 5:54PM |
1 |
How to auto-speak agent's number once agent answer the incoming call |
| 5:30PM |
1 |
tellme/skype voice apps go live |
| 3:27PM |
1 |
Asterisk+Nat+sipura (Help) |
| 2:01PM |
1 |
Polycom 601 XHTML microbrowser |
| 1:48PM |
3 |
smp |
| 1:40PM |
4 |
asterisk using tdm400p has echo |
| 12:48PM |
3 |
ANNOUNCEMENT : A2Billing - AreskiCC V3 new release |
| 12:18PM |
0 |
fax2mail script update (includes hoodaheck compatibility) |
| 12:01PM |
0 |
FC4 + ztdummy + timming + trunking |
| 11:46AM |
0 |
Fw: UK BT IDSN30e 'pass through' with TE205P/AvayaArgentOffice? |
| 10:59AM |
2 |
web management interface |
| 10:31AM |
1 |
Incoming CallerID Name display |
| 9:56AM |
2 |
How to do Call Forwarding |
| 9:35AM |
0 |
New Asterisk Mailing List: asterisk-i18n |
| 9:17AM |
1 |
2-line phoneline |
| 8:23AM |
5 |
SPA3000 as trunk - no caller ID |
| 8:22AM |
0 |
AW: Some problem with CAPI support |
| 6:55AM |
2 |
polycom software |
| 6:17AM |
1 |
UK BT IDSN30e 'pass through' with TE205P/Avaya ArgentOffice? |
| 5:38AM |
0 |
Some problem with CAPI support |
| 3:35AM |
0 |
RE: strange behaviour of asterisk sip.conf |
| 1:53AM |
0 |
MeetMe architecture problem |
| 1:33AM |
2 |
Asterisk iptables rules |
| 1:31AM |
1 |
Zaptel + No Hangup |
| 1:02AM |
0 |
Asteriks configuration |
| |
| Tuesday October 25 2005 |
| Time | Replies | Subject |
|---|
| 9:41PM |
2 |
Asterisk on PPC Linux |
| 9:15PM |
0 |
Asterisk+Nat+Sipura/Linksys |
| 7:31PM |
0 |
clarification please: accountcode, pri channel groups, and CDR |
| 7:29PM |
2 |
Incoming calls via CAPI and AVM Fritz Card |
| 5:46PM |
3 |
Wanted to Swap! TDM400 FXO module(s) for FXS |
| 5:20PM |
1 |
carrier blocking |
| 3:41PM |
0 |
Send tone to src on supervision |
| 3:33PM |
2 |
Fwd: ByVolution |
| 3:09PM |
0 |
Grandstream GXP2000 tftp config |
| 2:20PM |
3 |
FCC VoIP wiretap ruling challenged |
| 1:59PM |
3 |
HELP! |
| 1:31PM |
0 |
OT: Need Asterisk VOIP Support for Customer in Louisiana |
| 12:57PM |
1 |
strange behaviour of asterisk sip.conf type=user vs type=peer |
| 12:27PM |
2 |
Sipura SPA-3000 and Gigaset DECT phone: no ring |
| 12:04PM |
0 |
App_directory + Festival |
| 11:43AM |
2 |
PDA softphone.... |
| 11:15AM |
5 |
How to configure the communication between two Asterisk servers |
| 10:44AM |
1 |
variable `oh323_tech' has initializer but incomplete type |
| 10:07AM |
2 |
Echo cancel and fax |
| 9:56AM |
0 |
ECT - Specifying the transfer destination. |
| 9:30AM |
1 |
re: changing protocols and transcoding |
| 7:41AM |
1 |
H323 REGISTRATION PROBLEM: Gatekeeper 'Nortel_H323_Gatekeeper@.. ' found but failed to register |
| 7:36AM |
3 |
Sangoma A104 errors |
| 7:18AM |
0 |
Voicemail prompts not heard on Cisco Phone |
| 7:11AM |
2 |
AudioCodes - TP260 |
| 7:00AM |
0 |
Festival() works fine when I call from PSTN and not when I call from XLite... What's going on? |
| 6:13AM |
0 |
Re: FCT-11M |
| 5:54AM |
0 |
MWI for other purpose than voicemail? |
| 5:51AM |
1 |
Question on callingpres and blocked numbers |
| 5:26AM |
2 |
pppoe-server & Asterisk |
| 4:24AM |
0 |
Swissvoice Vizufon firmware |
| 4:10AM |
3 |
Realtime sip register=> |
| 3:29AM |
13 |
iaxmodem |
| 3:16AM |
0 |
Distinguishing National from International Calls on Zap Channel |
| 3:15AM |
1 |
connect 2 phones like in FOP |
| 3:00AM |
1 |
Agent logout |
| 1:59AM |
1 |
Asterisk user meeting in Oslo, Norway |
| 1:42AM |
5 |
Cisco 7905G Power over Ethernet - does it work? |
| |
| Monday October 24 2005 |
| Time | Replies | Subject |
|---|
| 10:46PM |
5 |
Asterisk & SER for dummies ? |
| 10:01PM |
2 |
X101P and UK CallerID...does it work? |
| 9:55PM |
1 |
Belgium Meetings from Nov 11 to Nov 26? |
| 9:14PM |
0 |
patch for one-way audio for asterisk-oh323 |
| 7:16PM |
0 |
How to tell what EC is in place (Was: RE: Terribleecho with Te110P and Adit 600) |
| 7:08PM |
4 |
Format of extensions.conf |
| 5:53PM |
0 |
Unavailable |
| 4:31PM |
0 |
Add SIP extension |
| 4:10PM |
2 |
PROBLEM WITH A PRI INCOMING CALLS |
| 4:00PM |
2 |
Red Alarms, No D-Channels, and Crazy People |
| 2:45PM |
0 |
Unicall Error ... T1 Timeout |
| 1:55PM |
2 |
Siemens HI-path to ASTERISK |
| 1:34PM |
1 |
Ticking sound in wildcard tdm400p, Please Help |
| 1:34PM |
0 |
Recommend an LD provider who can use IAX |
| 1:22PM |
0 |
more and more "017...ec@127.0.0.1" |
| 1:20PM |
0 |
Government/Enterprise User Group |
| 10:43AM |
1 |
SIP to CAPI - Soundcard required? |
| 10:31AM |
2 |
Urgent - Need Help - Audio Issues |
| 8:34AM |
1 |
Largest working config files? |
| 8:18AM |
3 |
polycom, call waiting, queues |
| 7:41AM |
0 |
Hangup ZAP channel |
| 7:17AM |
0 |
How to setup parked/on-hold so sorresponding buttons on VoIP phones light up |
| 6:58AM |
0 |
Thounsand of SIP extension |
| 6:33AM |
0 |
could set up only messagenet.it |
| 5:59AM |
2 |
cvs head + spandsp |
| 5:20AM |
1 |
new toy |
| 4:50AM |
1 |
Asterisk vs Sipura SIP problem? |
| 2:53AM |
1 |
"zap show channel 1" says "PRI signalling" on my zaphfc BRI card |
| 2:35AM |
1 |
0.2.0-RC8o (* 1.0.9) + No Caller ID |
| 2:05AM |
3 |
configuring Cisco 7905G for SIP - how? |
| 1:59AM |
0 |
NAT Problem after first call |
| 1:35AM |
1 |
Asterisk Realtime - MySQL Extension registration problem |
| 1:03AM |
6 |
Where is the text of the voicemail email ?? |
| 12:24AM |
1 |
Passing parametrs to php agi scripts. |
| |
| Sunday October 23 2005 |
| Time | Replies | Subject |
|---|
| 11:28PM |
0 |
Queue application problem |
| 9:04PM |
1 |
asterisk -RT |
| 8:43PM |
0 |
Fwd: Re: Problem with asterisk 1.0.9 and sip and dtmf |
| 7:14PM |
0 |
Problems with Festival... |
| 4:27PM |
1 |
T1 Hardware Recommendations |
| 3:22PM |
0 |
iConnectHere (or DeltaThree) trunk settings |
| 2:24PM |
1 |
Hardware setup question |
| 9:23AM |
6 |
Trying to clarify ideas about spands, libtiff & FC4 |
| 9:09AM |
1 |
SIP DTMF problem |
| 8:28AM |
1 |
Adit 3104 configuration |
| 3:40AM |
1 |
How to play a voice file for " decline " |
| 2:57AM |
1 |
ASTBILL |
| 2:45AM |
1 |
Asterisk dropping call file without *any* notice |
| 1:30AM |
3 |
Anyone using Java SIP communicator with Asterisk ? |
| |
| Saturday October 22 2005 |
| Time | Replies | Subject |
|---|
| 9:50PM |
0 |
Asterisk CDR records when a call is transferred |
| 7:12PM |
2 |
Satellite receiver over IP |
| 6:08PM |
1 |
IAX registration with FWD and Teliax - Lost |
| 5:50PM |
0 |
redirecting incoming calls to external phone (cell) |
| 3:05PM |
2 |
Problem with asterisk 1.0.9 and sip and dtmf |
| 12:59PM |
3 |
Modem Over IP: solutions ? |
| 11:15AM |
2 |
CDMA card or module made for Asterisk? |
| 11:07AM |
1 |
Do the quantity of hardware timing devices go up as call volume increases? |
| 6:38AM |
1 |
Linksys pap2 behind Linksys RT31 |
| 6:31AM |
0 |
play a voice file voice for " decline " |
| 6:03AM |
4 |
One SIP dead, all SIP dead -- sipmedia gone? |
| 5:36AM |
5 |
voip provider in a box |
| 5:28AM |
0 |
need help:multisite with asterisk? |
| 3:19AM |
2 |
Filenaming for Incoming Queue Call Recordings (Reposted from "changing the filename of incoming call recordings") |
| 3:18AM |
0 |
Call problems using IAX |
| 3:09AM |
1 |
Cisco 7960G and Asterisk |
| 2:23AM |
1 |
chan-capi_cm - 10sec silence before ringing |
| 12:03AM |
2 |
Testing AreskiCC |
| |
| Friday October 21 2005 |
| Time | Replies | Subject |
|---|
| 10:58PM |
0 |
FXO no release useing fxsks & disconnect supervision from telco |
| 9:14PM |
3 |
Queue Join Event |
| 7:33PM |
1 |
Asterisk Newbee: Fedora Core 4 |
| 4:54PM |
0 |
Process VoiceMail file to attach |
| 4:16PM |
1 |
messagenet |
| 3:26PM |
0 |
FS: Single channel GSM to POTS "gateway" |
| 2:46PM |
1 |
OT: How to reach Junghanns.net? |
| 2:02PM |
1 |
Fax problem with zap trunk... |
| 1:56PM |
0 |
How to configure two Asterisk servers for one callcenter |
| 1:39PM |
1 |
How to wire Asterisk to stations? |
| 1:17PM |
3 |
Getting ztdummy to load on startup for X100P |
| 1:15PM |
1 |
Meetme admin option |
| 1:00PM |
3 |
asterisk locked up |
| 12:52PM |
0 |
Asterisk@Home and Internationalization & Numbers. |
| 12:32PM |
1 |
Voicemail Changes |
| 12:19PM |
1 |
Problem with asterisk-sounds-1.2.0-beta1 |
| 10:46AM |
1 |
Polycom 501 can't find TFTP server? |
| 10:25AM |
0 |
Just a test... |
| 10:08AM |
2 |
force 911 over pri line |
| 9:39AM |
5 |
How to configure two Asterisk servers for one call center |
| 9:07AM |
0 |
Whats wrong with incoming |
| 8:51AM |
0 |
Definitive answer: time-range includes |
| 8:26AM |
1 |
Double DTMF with tdm card |
| 8:15AM |
0 |
OT: Goiax.com DID not working anymore? < --i s it appropriate for this list? Yes. |
| 7:45AM |
1 |
Queue_log multiple entries |
| 7:38AM |
5 |
MTP required for CCM integration ? |
| 7:26AM |
0 |
REPOST: Private/Anonymous/Restricted not being passedbyAsterisk Lost in the shuffle? |
| 7:09AM |
0 |
Incoming call and DID routing |
| 6:51AM |
3 |
cvs core dump |
| 6:46AM |
0 |
Force all local numbers and 911 out PRI |
| 6:37AM |
2 |
Re: T1 questions - could I got VoIP instead? |
| 6:36AM |
1 |
Custom handling of SIP 302 redirect? |
| 6:03AM |
0 |
RES: Configure Festival to speak in portuguese |
| 5:56AM |
3 |
Preventing abuse of Goiax |
| 5:52AM |
1 |
Swissvoice IP10S centralized phonebook |
| 5:48AM |
0 |
Asterisk Festival |
| 5:34AM |
0 |
No music on hold for agent channel |
| 4:55AM |
0 |
Configure Festival to speak in portuguese |
| 4:35AM |
0 |
exclude context? |
| 2:39AM |
0 |
Help on func_odbc.c or similar |
| 2:01AM |
0 |
php with oci8 agi script |
| 1:44AM |
3 |
2.6.13 zaptel incompability? |
| 1:43AM |
0 |
Call transfer caller ID |
| 1:35AM |
1 |
Group dial CDR |
| 1:19AM |
1 |
R2...channel type UniCall |
| 1:00AM |
0 |
Problem with Asterisk newbie |
| 12:58AM |
1 |
SIP gateway: call hangups afer 3 rings |
| 12:14AM |
2 |
Does fwdout even work anymore? |
| |
| Thursday October 20 2005 |
| Time | Replies | Subject |
|---|
| 10:46PM |
3 |
how many oh323 |
| 9:28PM |
1 |
Asterisk Community Participant; Katrina Refugee UPDATE |
| 7:51PM |
1 |
Problem with Asterisk newbie2 |
| 7:11PM |
4 |
merchant account |
| 6:47PM |
2 |
Multiple instances of asterisk showing from 'ps aux' |
| 5:52PM |
0 |
intergration with Nortel Meridian 81c |
| 5:18PM |
1 |
Voicemail/Record sending no RTP packets (CNG) back to caller when recording messages |
| 4:04PM |
1 |
Private/Anonymous/Restricted not being passed by Asterisk |
| 3:56PM |
0 |
Diaplan or iax.conf problem |
| 2:52PM |
0 |
Fyi: I-D ACTION:draft-guy-enumiax-00.txt (fwd) |
| 2:32PM |
7 |
Goiax.com DID not working anymore? |
| 1:26PM |
2 |
Help on Asterisk and Client SIP setup |
| 1:17PM |
1 |
areski Problem |
| 1:14PM |
1 |
Sip and autonegotiating codecs |
| 11:47AM |
0 |
Re: T1 questions follow-up |
| 11:45AM |
1 |
Digium list server and spam assassin |
| 11:36AM |
2 |
Re: T1 questions follow-up |
| 10:57AM |
1 |
Application return values |
| 10:18AM |
4 |
Problem with Cisco phone |
| 10:15AM |
3 |
Some questions regarding T1's |
| 9:13AM |
0 |
D-Link DG104S firmware upgrade for flash funcionality on * |
| 8:31AM |
3 |
cdr_odbc with tds |
| 8:23AM |
4 |
E1/T1 failover hardware |
| 8:18AM |
0 |
CallerID PHP Script |
| 8:15AM |
4 |
TDMoE and Badness in Kernel |
| 7:39AM |
0 |
Getting output from agi scripts (python) |
| 7:38AM |
0 |
Manager API - Supervised Transfer |
| 7:01AM |
4 |
Asterisk Billing |
| 6:45AM |
1 |
user name |
| 6:32AM |
0 |
Fwd: Re: [Asterisk-doc] You ASKED for an Asterisk book, you GOT an Asterisk book! |
| 6:06AM |
0 |
E1 (TE405p) SetCallerId problem |
| 5:23AM |
3 |
Asterisk Compilation with H323 working on it |
| 5:20AM |
0 |
codec voice quality ratings |
| 5:17AM |
1 |
wm_w DTMF solution for T1 tie line losing deigits. |
| 4:50AM |
3 |
Why Asterisk documentation is so poor... |
| 4:39AM |
2 |
Siwssvoice IP10S telnet password |
| 3:59AM |
0 |
Context configuration with AstTapi |
| 3:35AM |
2 |
Problem with Swissvoice IP10S and Asterisk |
| 3:34AM |
0 |
changing the filename of incoming call recordings |
| 2:19AM |
0 |
toll free dialing problems using SIP |
| 1:45AM |
2 |
Isdntrace utility |
| 1:00AM |
1 |
Any good docs for latest CVS-HEAD / Stable 1.2? |
| 12:31AM |
2 |
Can't build Asterisk on SuSE |
| 12:30AM |
0 |
Chan-capi sound choppy |
| 12:15AM |
4 |
Asterisk in Croatia - Zagreb |
| |
| Wednesday October 19 2005 |
| Time | Replies | Subject |
|---|
| 11:22PM |
0 |
Dial Limit Call Options |
| 9:27PM |
0 |
how to edit or delete calleridname in From URI |
| 9:17PM |
2 |
chan_capi-0.6 configuration Query with Eicon Diva 4BRI |
| 7:10PM |
1 |
Re: Asterisk Evening in Melbourne Australia! |
| 5:21PM |
1 |
goiax configuration help please |
| 5:20PM |
1 |
Digium TDM400P (11B) problems |
| 4:29PM |
1 |
OT: Samsung DCS 70 |
| 2:38PM |
3 |
New ISDN architecture available for asterisk |
| 1:40PM |
0 |
some faxes get cut off |
| 1:39PM |
3 |
Please recommend a phone |
| 1:36PM |
1 |
INBOUND DID SERVICE FOR THE ASTERISK COMMUNITY |
| 1:33PM |
1 |
Dial 2 channels at onece: Not working anymore at CVS? |
| 1:31PM |
2 |
Asterisk 1.2.0beta1 for Debian Sarge |
| 1:19PM |
1 |
Free DID's |
| 1:19PM |
1 |
Problems Installing MPG123 on a 64 Bit System |
| 1:01PM |
0 |
Extension dialing out |
| 12:51PM |
2 |
sixtel DID |
| 12:49PM |
2 |
more voip patent madness |
| 12:09PM |
1 |
teliax audio issues - response |
| 11:40AM |
1 |
possible bug, what do you think? |
| 10:59AM |
0 |
unable to make connectivity between asterisk to external phone |
| 10:10AM |
0 |
NEWBIE HELP : chan_zap.c: Exception on 16, channel 1, call not being picked up on incoming X1-100P zap |
| 9:39AM |
3 |
uable to establish link between asterisk to external phone |
| 9:04AM |
1 |
Caller-ID via database lookup |
| 8:55AM |
0 |
Fwd: Re: IAX only speech one way |
| 8:51AM |
2 |
E1 PRI error: "!! Got I-frame while link state 2" and "!! Got a UA, but i'm in state 1" (long) |
| 8:10AM |
0 |
Trunk Dialing rules |
| 7:28AM |
1 |
Connection question |
| 6:33AM |
1 |
DNIS/DNID |
| 6:16AM |
2 |
SIP CallerID |
| 5:58AM |
3 |
SIP to IAX |
| 5:54AM |
0 |
Fw: asterisk shutting down... |
| 5:46AM |
0 |
How can I signal a flash to PABX ... |
| 4:39AM |
4 |
Help with Dial Plan |
| 4:18AM |
0 |
chan_capi.so: undefined symbol: ast_smoother_feed |
| 4:09AM |
0 |
IAX termination/DID provider in Panama? |
| 4:02AM |
1 |
Asterisk on Slackware ... |
| 3:39AM |
0 |
my SIPURA ATA does not make calls thru teliax |
| 2:59AM |
1 |
what hw/OS to choose [please help] |
| 2:49AM |
1 |
Call queuing question |
| 2:29AM |
0 |
Persistant connection for MYSQL command |
| 2:19AM |
0 |
How to suppres leading zeros in zapata.conf? |
| 1:29AM |
0 |
Realtime - table voicemail |
| 1:00AM |
1 |
Problems Calling PSTN PSTN FROM ASTERISK |
| 12:24AM |
1 |
Asterisk hangs |
| |
| Tuesday October 18 2005 |
| Time | Replies | Subject |
|---|
| 9:27PM |
1 |
Audiocodes MP-108 |
| 8:27PM |
2 |
New TDM Revision in the wild: J |
| 8:02PM |
9 |
more dids added to goiax.com |
| 8:00PM |
1 |
zaptel.conf config for CAS signalling |
| 7:42PM |
12 |
Terrible echo with Te110P and Adit 600 |
| 7:19PM |
0 |
aah digital receptionist weird |
| 7:17PM |
0 |
Newbie IAX |
| 6:54PM |
0 |
Languages, Realtime, German, Finish |
| 6:18PM |
0 |
Language settings not working in astcc |
| 4:01PM |
0 |
whats the difference between http://www.teliax.com and http://www.starvox.com |
| 2:34PM |
0 |
central voicemail storage |
| 2:16PM |
1 |
Priority jump in AEL |
| 1:33PM |
0 |
Monit test for IAX2 |
| 1:12PM |
2 |
IAX only speech one way |
| 1:12PM |
0 |
IP300 -> Asterisk -> Broadvoice -> PSTN Choppy / cuts in and out |
| 12:43PM |
0 |
Fwd: {100-1287} RE: DID"s |
| 12:02PM |
4 |
One phone ringing, one phone flashing ? |
| 11:50AM |
1 |
Forwarding Extensions using dialplan |
| 11:48AM |
0 |
zaptel problem |
| 11:16AM |
1 |
strange behavior after turning jitter buffer on |
| 10:47AM |
0 |
Re: Asterisk-Users Digest, Vol 15, Issue 108 |
| 10:21AM |
0 |
Problem loading misdn driver |
| 10:20AM |
0 |
Re: Vontage Problems |
| 9:46AM |
1 |
sip rfc bye violated? |
| 9:41AM |
2 |
Fwd: {100-1287} RE: DID"s |
| 9:32AM |
7 |
Asterisk Redundency |
| 8:59AM |
4 |
Polycom IP501 and record on demand |
| 8:26AM |
5 |
Newbie Question: Help with incoming dial plan |
| 8:11AM |
1 |
select codec based on extension |
| 8:02AM |
0 |
Assistance with loging a particular event. |
| 7:39AM |
8 |
Fax2Mail |
| 7:13AM |
0 |
Hang up problem Costa Rica Indications |
| 7:08AM |
0 |
Display number dialled |
| 6:23AM |
1 |
setting a dialplan on a GXP-2000 Grandstream |
| 5:58AM |
2 |
SV: SV: Queues and call waiting indication |
| 5:35AM |
2 |
SV: Queues and call waiting indication |
| 5:27AM |
0 |
411 |
| 5:18AM |
2 |
Agent recording and muxmon |
| 5:13AM |
1 |
Queues and call waiting indication |
| 4:50AM |
1 |
Recomendations for utility to generate Asterisk configuration |
| 2:37AM |
6 |
Can IAX be used without going thre a IAX server |
| 1:58AM |
1 |
error while writing audio data: : Broken pipe |
| 1:53AM |
2 |
fax device behind TDM400P |
| 1:32AM |
0 |
Slow dialling from PBX into * via E1 |
| 12:50AM |
0 |
Pb musiconhold with G729 codec |
| 12:11AM |
3 |
CAPI - displaying individual MSN |
| 12:05AM |
8 |
free dids on goiax.com |
| 12:03AM |
1 |
Talkoff (Spurious DTMF) with 1.0.9.2 and TE406P |
| |
| Monday October 17 2005 |
| Time | Replies | Subject |
|---|
| 11:22PM |
0 |
How to use Use different ports to authenticate SIP/IAX users |
| 7:22PM |
0 |
Fwd: Re: SIP to SIP sadness |
| 6:25PM |
2 |
DID's |
| 5:29PM |
1 |
Can I use ANY port for SIP device? |
| 5:11PM |
0 |
Asterisk 1.0.9 - PortaOne Radius |
| 5:04PM |
2 |
Dial command in extensions |
| 4:45PM |
0 |
Argentina - Vontel - Asterisk |
| 4:26PM |
1 |
Uniden UIP200 Issues |
| 3:28PM |
2 |
Bizarre Echo Problem |
| 3:14PM |
2 |
Teliax IAX problems -- Asterisk doesn't see answer |
| 1:51PM |
1 |
(no subject) |
| 1:50PM |
6 |
Where can I find Polycom 600 config files? |
| 1:42PM |
0 |
FW: ISDN PRI and E1 |
| 12:56PM |
0 |
Ztdummy is shutting down my sound |
| 12:49PM |
2 |
CDMA phone line for Asterisk? |
| 12:48PM |
1 |
Middle Ground between POTS and T1? |
| 12:44PM |
1 |
Problem with incoming calls |
| 11:54AM |
1 |
SIP to SIP sadness |
| 11:38AM |
3 |
Multiple calls per phone |
| 11:35AM |
1 |
can't compile ast_*fax |
| 11:15AM |
1 |
How can I get a dialtone calling from outside... |
| 10:27AM |
6 |
initiate call recording from phone. |
| 10:02AM |
0 |
Interface with ability to originate call |
| 9:59AM |
0 |
Ruby module for the Asterisk Manager Interface |
| 9:58AM |
0 |
cmd SIPRedirect for loadbalancing |
| 9:15AM |
1 |
astcc missing to bill random calls? |
| 9:06AM |
0 |
Connecting TIE trunk to Astericks |
| 8:37AM |
0 |
Legacy PBX Integration and Zaptel.conf Timing Source |
| 8:01AM |
0 |
No Audio from Console but mpg123fromshellworksfine. |
| 7:05AM |
1 |
Call transfer - atxfer |
| 6:37AM |
0 |
Transfering calls. Dial plan |
| 6:28AM |
0 |
fax receive problem on zapata channel |
| 6:28AM |
1 |
Double Ringing for PRI Calls |
| 6:19AM |
0 |
Asterisk Busy Detect |
| 5:55AM |
4 |
compiling Asterisk 1.2 with zaptel and h.323 |
| 5:30AM |
1 |
module loading error with Ubuntu: insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format |
| 5:09AM |
0 |
ooh323c and calls to pri |
| 5:06AM |
0 |
RxFax dropping line |
| 4:41AM |
1 |
fax - conversion problem |
| 4:14AM |
1 |
integrating asterisk smoothly |
| 3:37AM |
0 |
AstBill-0.9.0.7 with many new features Released |
| 3:13AM |
1 |
AVM B1 |
| 12:39AM |
0 |
Solved? => Playback audio before answered by a queue member |
| 12:35AM |
4 |
Delayed ringing on some SIP phones |
| 12:33AM |
4 |
Polycom MWI |
| |
| Sunday October 16 2005 |
| Time | Replies | Subject |
|---|
| 11:46PM |
1 |
Newbi stating question |
| 10:57PM |
2 |
Modifying Voicemail App |
| 9:22PM |
4 |
Cannot telnet to port 5038 on asterisk |
| 9:20PM |
0 |
queue to queue to vmail |
| 9:01PM |
0 |
queue to queue to vmail failover |
| 8:52PM |
0 |
failover |
| 8:35PM |
1 |
Need language variable to user account |
| 8:25PM |
2 |
Can you use Polycom 500 with PoE Switch? |
| 6:49PM |
0 |
Playback audio before answered by a queue member |
| 4:15PM |
1 |
GROUP and GROUP_COUNT |
| 3:55PM |
1 |
Restricting registration for peer '611' to 60 seconds (requested 1200) |
| 3:44PM |
1 |
Can Asterisk "proxy" a SIP phone to make it look like a Cisco skinny softphone? |
| 1:51PM |
1 |
No Audio from Console but mpg123 from shellworksfine. |
| 1:30PM |
0 |
Call to all Astricon attendee's!!!! |
| 12:21PM |
2 |
huge problem compiling * on gcc4.x (SUSE 10.0) |
| 11:43AM |
2 |
Pass variable to context (NOT macro) |
| 10:43AM |
1 |
iax invtation problem |
| 10:38AM |
0 |
Job Offer for working at Madrid, Spain, on Asterisk/SER related issues |
| 10:30AM |
1 |
Routing landline calls to asterisk. |
| 10:21AM |
2 |
Looking for advanced consultant services |
| 9:22AM |
1 |
Incoming SIP connection |
| 8:17AM |
3 |
Dial plan questions |
| 7:20AM |
0 |
No of simultaneous calls in asterisk |
| 6:44AM |
0 |
chan_capi and AVM FritzCard PCI |
| 6:20AM |
3 |
Asterisk and Fedora |
| 4:50AM |
0 |
CDMA USB phone for Linux? |
| 4:29AM |
2 |
No voice - one way - both ways |
| 2:41AM |
0 |
procees |
| 1:41AM |
0 |
IPManager PBX Features |
| |
| Saturday October 15 2005 |
| Time | Replies | Subject |
|---|
| 9:42PM |
2 |
What would cause a high memory usage in pbx_spool.c ? |
| 8:11PM |
0 |
Problem inAresk GUI installation |
| 7:04PM |
6 |
ACD calls to busy agents |
| 7:00PM |
1 |
Looking for Info on OH323 |
| 6:31PM |
1 |
No Audio from Console but mpg123 from shell worksfine. |
| 5:53PM |
4 |
Voicemail 2 |
| 3:19PM |
3 |
res_perl - Compiling error |
| 11:58AM |
7 |
You ASKED for an Asterisk book, you GOT an Asterisk book! |
| 9:33AM |
0 |
Planet Vip-150T |
| 7:32AM |
1 |
Maintenance panel |
| 7:21AM |
1 |
Problem with '#' key recognition |
| 5:29AM |
4 |
Quad BRI with Fedora, anyone? |
| 2:38AM |
1 |
Hints and Call Waiting |
| 2:06AM |
0 |
Disconnecting after 1 min while Communicating Clarent class 5 call manager |
| |
| Friday October 14 2005 |
| Time | Replies | Subject |
|---|
| 5:40PM |
3 |
Problem with compiling spandsp |
| 5:38PM |
3 |
Callerid on t1 lines |
| 4:56PM |
1 |
match a set of numbers in GoToIf against a variable |
| 2:00PM |
0 |
INFO Duration=250 |
| 1:32PM |
1 |
2 POTS to |
| 1:00PM |
1 |
Outbound registration expirey |
| 12:42PM |
3 |
Busy not jumping n + 101 anymore |
| 12:37PM |
0 |
soxmix generating mute files |
| 12:22PM |
1 |
How to rewrite a CALLERID on outgoing calls |
| 10:49AM |
1 |
Does anyone Know if tha avaya 4621 IP phone work wiht asteisk? |
| 10:31AM |
1 |
RE: Asterisk-Users Digest, Vol 15, Issue 85 |
| 10:22AM |
0 |
warning message when reloading chan_sap.so |
| 7:58AM |
2 |
Asterisk/Cisco Call Manager 3.3 |
| 7:55AM |
0 |
Don't know what to do if second ROSE componentis of type 0x6 |
| 7:51AM |
0 |
multi languages |
| 7:14AM |
0 |
No Audio from Console but mpg123 from shell works fine. |
| 7:12AM |
2 |
"Please Press Any Key to Accept a Call" |
| 7:11AM |
0 |
IAX or IAX2 ? [SOLVED] |
| 7:06AM |
0 |
Sending ANI over SIP |
| 6:48AM |
0 |
DTMF tones not working with SIP |
| 6:35AM |
1 |
Voicemail -> new feature request |
| 6:12AM |
0 |
IAXy Port number. Repost |
| 5:59AM |
1 |
Problem with two hfc-s cards |
| 5:56AM |
2 |
T1/E1 Cards |
| 4:54AM |
0 |
Asterisk IAX config user |
| 3:57AM |
1 |
Incoming call problem - ringing SIP devices report busy |
| 3:06AM |
0 |
Which H323 module to go for? |
| 2:31AM |
1 |
Access to trunks |
| 2:15AM |
5 |
Reset telephone IP PHONE 106 |
| 1:15AM |
4 |
[ISDN] Problem: Device '/dev/ttyI1' lacking dialtone |
| 1:07AM |
1 |
SPA-3000 Disconnect tone detection in Spain, Peru and Colombia ? |
| 12:42AM |
5 |
sip accounts |
| |
| Thursday October 13 2005 |
| Time | Replies | Subject |
|---|
| 11:30PM |
0 |
Sound too loud (saturated). How to change? |
| 11:08PM |
0 |
Call transfer. |
| 10:11PM |
2 |
Enum parse errors |
| 9:45PM |
1 |
TDM04b to SIP extension not ringing (sip to sipworks fine) - resolved but why? |
| 8:12PM |
0 |
TDM04b to SIP extension not ringing (sip to sip works fine) |
| 6:18PM |
2 |
Incomming call line identification (NOT CallerID) |
| 5:45PM |
2 |
what should i select ?????????? |
| 5:02PM |
2 |
ztdummy build problems |
| 4:47PM |
0 |
Re: call waiting not working on PAP2 (Andy Kuo) |
| 3:28PM |
1 |
call waiting not working on PAP2 |
| 2:37PM |
0 |
calls not ringing |
| 2:34PM |
0 |
Polycom Button Remapping: Part 2 |
| 2:11PM |
1 |
New Bug Marshal |
| 12:25PM |
2 |
Sample cisco config for cisco 7206 |
| 12:04PM |
2 |
DID on analog line |
| 9:50AM |
0 |
Pose your Sangoma Questions |
| 9:18AM |
0 |
sip channels marked with SIP_NEEDDESTROY but not being removed |
| 9:13AM |
0 |
PRI stopped accepting calls |
| 9:11AM |
2 |
PRI calls to Automated Attendants Dropped |
| 9:00AM |
0 |
R: PA168S/AT320P |
| 8:54AM |
0 |
Not ringing on incoming callls |
| 8:18AM |
0 |
fax consulting |
| 7:37AM |
0 |
fax consult |
| 7:33AM |
0 |
CallerID detection problem |
| 7:29AM |
0 |
RE: Wanting to Make a PocketPC have asecureConnection to asterisk server |
| 7:29AM |
0 |
Moscow Dids |
| 7:23AM |
0 |
Impport script for upgrading to 1.2 SQL Realtime? |
| 7:20AM |
1 |
Noob help with IAX |
| 7:11AM |
0 |
which voip fone will be better |
| 6:34AM |
2 |
PA168S/AT320P |
| 6:12AM |
0 |
[ SOLVED ] ISDN problem: lacking dialtone |
| 6:12AM |
1 |
link quality monitor |
| 6:00AM |
0 |
PickUpChan and Intercept |
| 5:33AM |
0 |
polycom soundpoint ip600 problem |
| 4:58AM |
2 |
Starting simple switch from an extension? |
| 4:55AM |
1 |
IAXy Port number |
| 4:37AM |
1 |
SetCallerID Problem |
| 4:23AM |
3 |
IAX ATA |
| 4:17AM |
0 |
sangoma a104 cards and ss7 signaling |
| 4:07AM |
1 |
Music on hold disappears for Dial(, m) when calling outside numbers |
| 3:52AM |
1 |
USB phone for Linux? |
| 3:40AM |
0 |
pbx_spool Call failed to go through |
| 2:31AM |
1 |
AGI Variable problem |
| 2:16AM |
1 |
TDM400P off-hook detection problem |
| 12:23AM |
0 |
Reset IP PHONE 106 |
| |
| Wednesday October 12 2005 |
| Time | Replies | Subject |
|---|
| 11:01PM |
2 |
Broadvoice Outages? |
| 10:56PM |
1 |
Integrated T1 |
| 9:42PM |
1 |
SIP to SIP no audio help |
| 9:18PM |
2 |
Wanting to Make a PocketPC have a secure Connection to asterisk server |
| 9:17PM |
2 |
How can I use different languages (Chinese, Cantoneese)? |
| 7:08PM |
3 |
New Application: Broadcast |
| 5:35PM |
0 |
Notice message meaning for C7960? |
| 3:28PM |
1 |
ASTCC and Asterisk 1.2? |
| 3:11PM |
1 |
TDM04B card with only 3 lines connected using chanisavail |
| 2:39PM |
2 |
Maximum retries exceeded on call. |
| 1:59PM |
1 |
MWI integration between Asterisk and Meridian |
| 1:07PM |
2 |
which hardware should i use?????????? |
| 12:21PM |
0 |
RE: faxing to/from asterisk - new |
| 12:19PM |
0 |
Is it possible to listen and respond on more than one IAX port? |
| 12:17PM |
2 |
Canadian Association of VoIP Providers |
| 11:48AM |
0 |
Feature codes work on SIP phone but not analog? |
| 11:43AM |
1 |
send Q931 information element keypadfacility ?! |
| 11:31AM |
8 |
SIP behind NAT to pub Asterisk, best solution? |
| 11:27AM |
1 |
Bulk Buys/Group Buys |
| 11:20AM |
1 |
displaying a message on the Snom 320 using sipsak |
| 10:30AM |
5 |
ACD/queues question |
| 10:11AM |
0 |
sound very loud (saturated) through IAX2 and SIP |
| 9:57AM |
3 |
AGI and set_callerid for number and name |
| 9:44AM |
1 |
Problem with PRI and Ericsson AXE 10 |
| 9:05AM |
3 |
Calibrating both RX and TX gain? |
| 8:53AM |
2 |
Polycom: Button Remapping, HELP! |
| 8:40AM |
1 |
Sangoma FXO/FXS cards? |
| 8:28AM |
2 |
Patton SmartNode |
| 7:09AM |
0 |
Zaptel Debug: "T1: Lost our place, resyncing " |
| 6:52AM |
0 |
X100P callerid ETSI - caller*ID failed checksum |
| 6:26AM |
0 |
unloading TE110P bristuffed module cause kernelpanic |
| 5:58AM |
0 |
Second Request for help: hardware requirements |
| 5:32AM |
0 |
arcaplex / horizon isdn and analog multiplex |
| 5:21AM |
8 |
parameters documentation |
| 5:19AM |
2 |
SNOM 360 Unknown SIP command 'PUBLISH' |
| 5:10AM |
0 |
CHANNEL HANGUP ASSISTANCE |
| 2:55AM |
3 |
E400P vs te410p vs te411p |
| 2:54AM |
2 |
asterisk log |
| 2:41AM |
1 |
MWI for endpoints not registered at Asterisk |
| 2:24AM |
1 |
detect SIP phone availability before dialing |
| 2:21AM |
5 |
delays with IAX2 and Meetme |
| 1:59AM |
2 |
Asterisk logo |
| 1:57AM |
0 |
Voicemail recording volume control |
| 1:56AM |
0 |
Outgoing Provider Recommendations |
| 1:50AM |
2 |
Modifying cmd VoicemailMain |
| 1:27AM |
2 |
Monitor DTMF problems |
| 1:04AM |
1 |
unloading TE110P bristuffed module cause kernel panic |
| |
| Tuesday October 11 2005 |
| Time | Replies | Subject |
|---|
| 10:44PM |
2 |
error message when accessing voicemail |
| 10:00PM |
2 |
Large country based dialplan |
| 9:46PM |
1 |
supermicro with asterisk and tdm cards |
| 8:43PM |
3 |
Dual PRI fail over |
| 8:06PM |
0 |
Are Digium serial numbers recorded into boards and modules? |
| 7:14PM |
1 |
Problem w/ Asterisk hanging when caller hangs up in voicemail |
| 6:37PM |
1 |
Areski Calling Card GUI |
| 5:39PM |
5 |
help with broken voicemail |
| 5:22PM |
1 |
Problems with Wait & SIP 486 "DND" |
| 4:59PM |
1 |
migrated to new ver on voip connection vs1 server voicemail problems |
| 3:42PM |
1 |
Wrong caller id in CDR |
| 2:59PM |
2 |
Question on hardware requirements when not using a land-line |
| 2:45PM |
1 |
Voicemail Passwords and RealTime |
| 2:27PM |
0 |
Realtime + OS X = anyone got it working? |
| 12:21PM |
1 |
call to a particular 800 numbernevershowsanswered on Zap channel |
| 11:50AM |
0 |
call to a particular 800 number nevershowsanswered on Zap channel |
| 10:14AM |
4 |
New Sangoma AA Series? |
| 8:49AM |
6 |
PRI echo issues: solvable? |
| 8:47AM |
0 |
TTL |
| 8:45AM |
0 |
FXO tune |
| 8:06AM |
1 |
callerid validation and expression |
| 8:01AM |
1 |
DIDx error |
| 7:41AM |
0 |
Not a local SIP domain |
| 7:32AM |
5 |
Which asterisk-friendly cards are fax-capable? |
| 7:09AM |
0 |
Echo on SIP Side? |
| 6:54AM |
0 |
DID in Bolivia |
| 6:51AM |
2 |
Re: [Chan-sccp-users] Need help with hint and callgroup |
| 6:43AM |
2 |
Voicemail while in queue. |
| 6:30AM |
2 |
nat and wandering phones |
| 5:42AM |
3 |
Asterisk and Mitel SX 200 Slip and Frame Err ors causing Major Ala rms |
| 5:12AM |
1 |
callerid validation and expression parser problems on Solaris 10 |
| 3:22AM |
1 |
asterisk to asterisk using mgcp |
| 1:35AM |
2 |
IAX or IAX2 ? |
| 1:08AM |
2 |
CallerID for BSNL (India) phones |
| |
| Monday October 10 2005 |
| Time | Replies | Subject |
|---|
| 10:12PM |
3 |
country code list |
| 9:50PM |
0 |
Queue delay |
| 9:38PM |
5 |
Soekris and Asterisk |
| 6:51PM |
0 |
cannot load new wctdm module |
| 6:37PM |
2 |
Astricon Podcasts? |
| 6:23PM |
2 |
DTMF detection |
| 5:50PM |
2 |
Errors with new fetched Asterisk cvs |
| 4:25PM |
0 |
Realtime Extensions - DB concepts |
| 2:55PM |
2 |
enable mysql in asterisk |
| 2:52PM |
1 |
Need help with hint and call group |
| 2:48PM |
0 |
Asterisk behaving wierd!! |
| 2:37PM |
2 |
Throroughly confused about SetCallerID |
| 2:30PM |
2 |
Asterisk and Mitel SX 200 Slip and Frame Errors causing Major Ala rms |
| 1:22PM |
1 |
AAH. only 1 ring |
| 12:44PM |
2 |
Beronet app_saynumber-beta-rc1 |
| 12:33PM |
0 |
Problem with Oh323 on 1.2Beta on CENTOS 3.5 |
| 12:25PM |
2 |
What is this error? Is there a bug? |
| 11:39AM |
1 |
CallerID Outbound on VOXEE |
| 11:18AM |
0 |
Faking it: queue_log and addQueueMember |
| 10:57AM |
3 |
Help, please help -- IAX2 softphone to server on LAN |
| 10:34AM |
1 |
Realtime regseconds update |
| 10:18AM |
0 |
CDR problem with DST Channel |
| 10:15AM |
1 |
2 line SIP ATAs with Asterisk using RealTime |
| 10:08AM |
1 |
Hang up call |
| 9:13AM |
1 |
Incoming SIP getting in, but not ringing. |
| 9:08AM |
1 |
Outgoing quality |
| 9:07AM |
0 |
Fredericksburg ZPUG Meeting will have an Asterisk Flavor this Month |
| 8:54AM |
0 |
Incoming Calls causing Protocol Error (6) |
| 8:10AM |
11 |
Open Source Content Management System - Joomla |
| 7:32AM |
1 |
Multitenant Call Center Setup |
| 7:31AM |
2 |
DTMF Question (misunderstood '*' button) |
| 7:30AM |
3 |
Billing/SPA-841/CDR Log |
| 6:56AM |
4 |
sip register incoming call contexts? |
| 6:45AM |
0 |
does tellular cell phones support answer switching |
| 6:18AM |
1 |
Bandwidth usage for codecs |
| 5:56AM |
1 |
customize the pager email |
| 5:34AM |
1 |
[Fwd: Libpri/chan_zap problems?] |
| 5:12AM |
2 |
My contribution to the issue of code- reversal |
| 4:34AM |
0 |
Dial plan logic documentation? |
| 4:28AM |
6 |
telephony that "just works" |
| 2:25AM |
2 |
TDM400 not working |
| 1:04AM |
2 |
AVM Fritz! + chan_capi + mISDN + PTP |
| 12:20AM |
0 |
where can be find zaptel cvs change log ? |
| 12:03AM |
0 |
Re: faxing to/from asterisk - new scripts |
| |
| Sunday October 9 2005 |
| Time | Replies | Subject |
|---|
| 11:45PM |
1 |
Problem setting SIP incoming/outgoing |
| 7:36PM |
2 |
Clicks, pops and noise |
| 6:11PM |
0 |
The VoIP Connection has $$$ opportunities for Asterisk experts |
| 5:20PM |
1 |
where to find an asteriak Voice Mail User Manual |
| 4:37PM |
1 |
MPG123 with Asterisk on debian (one of our interesting experiences) |
| 4:21PM |
2 |
Link |
| 3:42PM |
8 |
Zaptel Line Build Out |
| 3:21PM |
0 |
IVR pausing before dialing ext |
| 1:56PM |
0 |
Realm Auth = No? |
| 1:37PM |
0 |
app_txfax not running |
| 1:28PM |
4 |
Avaya 4620/4640 SIP firmware |
| 12:07PM |
0 |
Problem with disable call transfer |
| 10:51AM |
0 |
Problem logging in using domain |
| 10:34AM |
1 |
Dial/goto extension from CLI or BASH script |
| 9:20AM |
0 |
Asterisk, H.323 & Cisco uBR900 |
| 8:53AM |
0 |
Incoming Caller ID |
| 8:15AM |
2 |
compiling asterisk on SuSE Linux 9.3 fails: illegal instruction |
| 7:39AM |
1 |
Asterisk, VoiceTronix & UK Caller ID |
| 7:14AM |
0 |
mail2fax and fax2mail updated |
| 5:51AM |
0 |
who has implemented callback function? |
| 4:58AM |
0 |
Anyone Know That !!! |
| 4:32AM |
4 |
*8 and group pickup not working |
| |
| Saturday October 8 2005 |
| Time | Replies | Subject |
|---|
| 8:44PM |
2 |
Configuring TDM400 in Australia |
| 8:23PM |
0 |
Regcontext/regexten broken?? |
| 7:42PM |
0 |
ATA does not register |
| 5:39PM |
1 |
Cannot dial SIP via asterisk |
| 4:57PM |
1 |
Does anyone know what this means |
| 4:15PM |
4 |
Asterisk Log Color Coding |
| 3:01PM |
1 |
How to check what codec translations are in use in a call? |
| 2:12PM |
1 |
Outgoing call: hangup after answer |
| 9:49AM |
0 |
Asterisk on Solaris SPARC |
| 8:52AM |
1 |
need help-can't not register to asterisk from softphone |
| 7:27AM |
2 |
No incoming calls from chan_capi 0.6 |
| 3:03AM |
1 |
Extension bracket matching broken in CVS |
| 1:09AM |
0 |
Re: Asterisk-Users Digest, Vol 15, Issue 28 |
| |
| Friday October 7 2005 |
| Time | Replies | Subject |
|---|
| 10:32PM |
0 |
ParkAndAnnounce Question |
| 10:32PM |
3 |
Digium G.729 codec modules updated |
| 9:18PM |
0 |
Prelude to Comfort Noise Generation support on Asterisk |
| 6:58PM |
0 |
* cell phone problem |
| 5:46PM |
0 |
How to speech a text file with festival |
| 5:05PM |
0 |
Asterisk going ahead on a busy call |
| 4:39PM |
0 |
Pingtel applications |
| 4:06PM |
1 |
ASTCC -- semantic note of 'callstart' in cdrs? |
| 3:14PM |
0 |
IBM work with a TE405P Digium card? |
| 2:45PM |
3 |
hardware echo cancellation. sangoma? |
| 2:39PM |
1 |
How do you verify remote registrations |
| 2:37PM |
3 |
call to a particular 800 number never showsanswered on Zap channel |
| 2:28PM |
1 |
PSGw 2.0 Skype<>SIP gateway |
| 2:16PM |
0 |
Variable for codec used? |
| 1:33PM |
0 |
AudioCodes MP-104 FXS |
| 1:14PM |
0 |
BBEdit Language Module for asterisk? |
| 1:10PM |
0 |
'ztcfg -s' causes system hang |
| 12:58PM |
0 |
Asterisk to CCM Message Waiting Indicator |
| 12:54PM |
2 |
call to a particular 800 number never shows answered on Zap channel |
| 12:41PM |
3 |
TDM02B card difficulties |
| 11:03AM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday 10/8/2005 |
| 10:46AM |
0 |
asterisk install [colinux] |
| 9:33AM |
1 |
Outbound Mediatrix 1204. |
| 9:32AM |
0 |
txfax (app_txfax) sending issue |
| 9:15AM |
1 |
'make rpm' problem |
| 9:14AM |
3 |
wifi phones - desk |
| 9:13AM |
0 |
[cdr_addon_mysql.so]Ouch ... error while writing audio data: : Broken pipe |
| 8:04AM |
0 |
Issue with terra-call today |
| 7:54AM |
3 |
RE: faxing to/from asterisk - new scripts |
| 7:50AM |
2 |
Teliax users, g729 question |
| 6:12AM |
1 |
Distorted VM with iax2 with ilbc and jitterbuffer - bug? |
| 5:24AM |
1 |
overlap zaphfc - dialtone |
| 5:20AM |
1 |
S0 - T0 interfaces question |
| 3:52AM |
2 |
tx(rx)_fax for *-1.2.0.beta |
| 3:05AM |
0 |
Asterisk and chan-spy problems |
| 2:35AM |
0 |
Incoming sip |
| 2:29AM |
2 |
Asterisk on dynamic extrenal IP behind a nat router. |
| 2:29AM |
1 |
Echo cancel on HFC-S cards and CIDNum setting on outgoing calls |
| 2:17AM |
3 |
Where to get the latest SIP Firmware for Polycom Phones? |
| 12:58AM |
1 |
Noise using TE410P & Rhino Channel Bank |
| |
| Thursday October 6 2005 |
| Time | Replies | Subject |
|---|
| 11:26PM |
0 |
How to send error codes to connected phone? |
| 10:51PM |
4 |
Asterisk PBX in Debian |
| 10:12PM |
3 |
Asterisk and firewall |
| 10:06PM |
0 |
Issue with trunking |
| 8:52PM |
3 |
WCFXO and T1 PRI Card? |
| 8:36PM |
2 |
Latency on bridged PRI calls |
| 7:20PM |
1 |
Outbound CallerID Teliax |
| 4:37PM |
1 |
Snom 360 Phones - Administrator/User Feedback |
| 4:32PM |
1 |
Billing: amaflags and accountcode |
| 4:00PM |
2 |
SIP Dialler |
| 3:40PM |
1 |
7960g 2nd ethernet port cycles on/off |
| 3:37PM |
0 |
chan_capi configuration with AVM C2 card |
| 3:22PM |
2 |
how do I know what codec is being used |
| 3:13PM |
1 |
TDM400 takes Zap/4 line off hook |
| 2:13PM |
0 |
How do I using Hangup? |
| 1:49PM |
0 |
Codec issue? Dropping incompatible voice frame ... |
| 1:27PM |
0 |
transcode or passthrough |
| 1:07PM |
1 |
Results of an incorrect crossover pinout?? |
| 12:57PM |
0 |
Fw: Re: Re: inter Asterisk trunking IAX /IAX2 |
| 12:21PM |
0 |
Vodavi PRI issues? |
| 11:44AM |
0 |
How can I log call forwards? |
| 9:37AM |
1 |
How can I override *67? |
| 9:05AM |
1 |
Asterisk::AGI Alternate Download |
| 8:23AM |
0 |
Whats the channel name? |
| 8:17AM |
2 |
Asterisk/Debian/VIA EPIA M Howto |
| 7:36AM |
1 |
ast_fax with sendmail |
| 7:21AM |
1 |
adding new indication tones |
| 7:19AM |
2 |
Mediatrix 1204 and Asterisk |
| 7:17AM |
0 |
SIP Realtime Question |
| 6:29AM |
1 |
IP Multimedia Subsystems (IMS) |
| 6:11AM |
3 |
RES: CDR MySQL |
| 5:56AM |
1 |
number of did numbers in one channnel? |
| 5:56AM |
0 |
CDR data parsing |
| 5:47AM |
0 |
more calls |
| 5:44AM |
1 |
Fwd: ASTCC - INUSE Flag |
| 5:36AM |
2 |
How do I add a list of cidnames to the asterisk database in one shot ? |
| 5:10AM |
1 |
Selecting outgoing trunk based on extension number |
| 3:29AM |
0 |
AAH |
| 2:52AM |
0 |
SV: Incoming call |
| 2:43AM |
14 |
www.openpbx.org |
| 2:22AM |
0 |
getting called number from a zap channel |
| 1:43AM |
1 |
[help!] asterisk 1.2 beta |
| 1:05AM |
1 |
Incoming call |
| 12:50AM |
1 |
How to Forcing Call Disconnect? |
| 12:28AM |
1 |
Changing IP on Asterisk |
| |
| Wednesday October 5 2005 |
| Time | Replies | Subject |
|---|
| 11:06PM |
3 |
SIP Attended Transfer using REFER and Replaces: headers |
| 10:41PM |
0 |
IAX2 calls dropped (Max retries) |
| 9:37PM |
3 |
CVS HEAD and Hints |
| 9:36PM |
1 |
IAX2 + Jitter Buffer |
| 9:35PM |
1 |
newbie asterisk build |
| 8:08PM |
1 |
Delay before dialplan is launched? |
| 7:21PM |
0 |
Asterisk Nat solution for such scenario? |
| 6:38PM |
1 |
Clearing Caller-ID from Zaptel Channels |
| 6:04PM |
1 |
Help! Extensions |
| 5:54PM |
4 |
dropped calls when g729 is used on sip leg |
| 4:51PM |
2 |
inter Asterisk trunking IAX /IAX2 |
| 4:45PM |
0 |
Unknown or blocked ID now shows up as "asterisk" |
| 4:38PM |
1 |
Extension Always Goes to VoiceMail |
| 4:25PM |
2 |
Sipura SPA-3000 setup in Brazil |
| 3:54PM |
1 |
"AST_LIST_REMOVE" passed 4 arguments but takes just 3 |
| 3:40PM |
1 |
Auto-assign CallerID for all my FXS Interfaces |
| 2:53PM |
1 |
DLINK DVG-3004S |
| 2:12PM |
1 |
Config PolyCom SoundStation 4000 help |
| 1:51PM |
0 |
CVS won't compile: res_odbc error |
| 1:50PM |
0 |
Voicemail email issues |
| 1:38PM |
1 |
Attempted to delete none, xistent schedule entry 1! ?? |
| 1:25PM |
2 |
Define variable in sip.conf |
| 1:23PM |
1 |
Caching DTMF tones for get_data AGI? |
| 12:46PM |
5 |
Voicemailmain automatic extension detection? |
| 12:09PM |
0 |
Please, help test asynchronous generation patch for inclusion in version 1.2 |
| 11:56AM |
1 |
New astGUIclient/VICIDIAL version released 1.1.7 |
| 10:35AM |
1 |
What the heck? Sprint sues Vonage |
| 10:28AM |
3 |
IPComms Setup |
| 9:49AM |
1 |
TDMOE Badness in kernel... |
| 9:24AM |
2 |
Sipura Adapter SPA-2002 |
| 8:46AM |
2 |
Zaptel tone description |
| 8:10AM |
0 |
Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o memory leak when using call files ? |
| 7:50AM |
1 |
compiling astrisk |
| 6:02AM |
2 |
TE411P and TE406P stability |
| 5:15AM |
2 |
From Database, PHP-Webinterface -> TO flatfileconfiguration |
| 4:52AM |
0 |
Unwieldy outbound macro |
| 4:30AM |
0 |
agi-test.agi question - wierd results |
| 2:31AM |
2 |
can't run app_txfax |
| 2:21AM |
0 |
call transfer problem - something strange |
| 2:14AM |
1 |
Configuration QuadBRI Junghanns |
| 2:02AM |
1 |
how can i let the user in 1th Asterisk can call the user in 2nd Asterisk? |
| 12:58AM |
1 |
How to enter digits using sjphone |
| 12:55AM |
1 |
Easy SIP.conf questien. Incomming call context? |
| 12:39AM |
2 |
Intel Pentium Celeron |
| |
| Tuesday October 4 2005 |
| Time | Replies | Subject |
|---|
| 7:45PM |
2 |
Hardware vs. Network Inputs |
| 6:27PM |
1 |
Digium hardware echo canceller, zapata.conf settings? |
| 4:35PM |
1 |
Fw: trunking IAX2 |
| 3:51PM |
3 |
Transfer directly to voicemail (blind transfer)? |
| 3:23PM |
1 |
Forcing Codec Usage |
| 2:52PM |
1 |
Recommendations for * monitoring? |
| 2:43PM |
12 |
Sprint Nextel sueing over VoIP patents |
| 2:30PM |
4 |
Emergency calls - forcing through on channel |
| 1:43PM |
1 |
Polycom config and DTMF problems |
| 1:14PM |
1 |
Firefly 2 third-party version? |
| 12:49PM |
0 |
DTMF heard at end of AGI Record File |
| 11:33AM |
5 |
PBX 'Personalities' ? |
| 11:25AM |
0 |
compile loop? |
| 11:22AM |
0 |
Connecting two asterisk servers using IAX |
| 11:18AM |
0 |
check_asterisk commands |
| 11:12AM |
1 |
Hanging up on VoiceMailMain w/out putting in password causes call lockup |
| 10:55AM |
0 |
RECAP: 3? |
| 10:26AM |
0 |
forward iax extension |
| 10:05AM |
0 |
app_rxfax module won't load |
| 9:30AM |
2 |
DPH-140S SIP Phone oddities |
| 9:11AM |
0 |
Speed Up SayDigits? |
| 9:03AM |
3 |
ADSI -- is it dead? Worth bothering with? |
| 8:59AM |
1 |
FXS static and noise problem |
| 8:48AM |
0 |
Asterisk w/ BRIstuff compile error |
| 8:32AM |
1 |
Can't compile ast_rxfax with Asterisk 1.2.1b |
| 8:24AM |
3 |
Polycom 501: takes calls, but fast busy on dial out? |
| 8:07AM |
1 |
Announcing Voice over IP Directory Services (http://www.voipDS.org) |
| 7:54AM |
1 |
Number Restriction |
| 7:53AM |
1 |
Seeking Asterisk Solution For mid sized corp. |
| 7:52AM |
1 |
SNOM Subscribe/Notify |
| 7:44AM |
2 |
Call-in/Call-out |
| 7:39AM |
1 |
IODBC instead of UNIXODBC |
| 7:37AM |
3 |
Echo Canceling |
| 7:30AM |
0 |
Auto attendant |
| 7:02AM |
0 |
Dynamic feature support recently added to CVS HEAD |
| 6:45AM |
0 |
Error: -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from xxx.xxx.xxx.xxx |
| 6:41AM |
0 |
can't reject call using macro-screen |
| 6:35AM |
1 |
Asterisk Calling Card Platform |
| 6:14AM |
1 |
TDM versions question |
| 5:56AM |
0 |
Three-way calling over SIP and IAX using softphone |
| 5:32AM |
2 |
Quad PRI Problems |
| 3:55AM |
0 |
CallerID octoBRI connected on voxtream parlay i60 |
| 2:43AM |
1 |
Asterisk forwarding SIP with Remote-Party-ID |
| 2:28AM |
3 |
Asterisk as H323 gateway |
| 1:45AM |
1 |
Dial pattern sort order |
| 12:25AM |
3 |
Outgoing busy |
| |
| Monday October 3 2005 |
| Time | Replies | Subject |
|---|
| 11:28PM |
2 |
Voice Quality bad on one side of Frame Relay |
| 10:30PM |
3 |
DIAX not working properly |
| 10:25PM |
0 |
Weird Problem - SIP/POLYCOM/DTMF |
| 9:09PM |
4 |
Snom phones? |
| 7:42PM |
0 |
Inter Asterisk IAX2 |
| 7:32PM |
0 |
RTP timing problems? Here's patch... |
| 6:29PM |
2 |
Hang-up Detect - Yet Again |
| 5:40PM |
0 |
Ticking sound in wildcard tdm400p |
| 5:01PM |
2 |
Debian sarge package for 1.2beta1? |
| 3:48PM |
2 |
asterisk, cisco 3640's and DIDs... |
| 3:14PM |
1 |
Realtime and voicemail: request to find out if I'm crazy |
| 3:03PM |
0 |
TDM400P recognised as "Network controller: Unknowndevice" |
| 2:55PM |
1 |
Direct Dial In - second try |
| 2:50PM |
3 |
FreeTDS 0.63 |
| 2:42PM |
2 |
TDM400P recognised as "Network controller: Unknown device" |
| 1:11PM |
2 |
sip phones on x86_64 |
| 12:56PM |
0 |
Asterisk Ignoring [User] in SIP.CONF |
| 12:13PM |
2 |
Real Life FAX sending receiving |
| 11:27AM |
2 |
Asterisk 1.0.8 and TDM stop acking inbound calls? |
| 11:16AM |
0 |
Console sound output -- shuts off when call from console answered |
| 11:10AM |
1 |
Compiling SpanDSP |
| 10:54AM |
0 |
Need help with Cisco 7960 |
| 10:48AM |
0 |
Hangup not detected on callback |
| 10:47AM |
0 |
SIP qualify question. |
| 10:47AM |
0 |
Which hardware configuration? How would this work? |
| 9:56AM |
0 |
Very cheap IP GSM Gateway: Will this work? |
| 9:56AM |
0 |
TDMoE help with Alarms... |
| 9:43AM |
1 |
suse 9.3 pro asterisk install from source problem |
| 9:12AM |
1 |
SIP-CPE Gateway |
| 8:38AM |
1 |
no audio on fxo line |
| 8:19AM |
4 |
R: Diva |
| 8:16AM |
0 |
asterisk behind Linux iptables with masquerading and forwarding on |
| 7:03AM |
1 |
R: codec g723 on Via C3 |
| 7:01AM |
1 |
Problem with configuration of Quintum AX with Asterisk |
| 6:38AM |
0 |
fc4 + iax + trunking |
| 6:27AM |
1 |
[Fwd: Eicon Diva 2.01 S/T PCI quality problems] |
| 6:18AM |
0 |
How to establish ISDN port Up |
| 4:50AM |
4 |
SPA-3000 generating one-ring calls |
| 4:05AM |
3 |
codec g723 on Via C3 |
| 3:50AM |
0 |
US tollfree DID request |
| 12:46AM |
1 |
*** Community alert :: Do you have open bugs in the bug tracker? |
| |
| Sunday October 2 2005 |
| Time | Replies | Subject |
|---|
| 9:40PM |
0 |
kjournald and zttest results |
| 8:59PM |
0 |
zttool improvement: histogram |
| 8:18PM |
0 |
is a dual 1.5Ghz server better than a single3Ghz for a 100 Iax users asterisk server |
| 7:59PM |
1 |
IAX2 Group dialing.... Is there something in the horizon? |
| 7:31PM |
1 |
Audiocodes MP108 |
| 7:13PM |
1 |
analog phone connects to zaptel fxoks is beeping |
| 3:29PM |
3 |
What does the error "stale nonce' mean? |
| 2:51PM |
0 |
Console Sound: Cuts out, Comes back after restart |
| 2:35PM |
0 |
Avaya 4620 hardphone |
| 2:22PM |
0 |
where can I find this property: answeronpolarityswitch |
| 2:21PM |
1 |
Asterisk-RealTime: sip_friends and register => user:pass@host |
| 1:42PM |
0 |
iax invitation problem |
| 9:22AM |
1 |
Adit 600 FXO card sound quality |
| 7:12AM |
3 |
[Sorta OT] Eicon DIVA with asterisk@home |
| 6:09AM |
3 |
Fw: Channel Banks, what are they for? |
| 4:41AM |
1 |
Adding Voicemail box |
| 2:58AM |
0 |
Grandstream GXP2000 |
| 1:24AM |
2 |
DB Function in version 1.2 |
| 12:53AM |
4 |
IBM tts engine integration |
| 12:52AM |
0 |
Outgoing rout dialpattern |
| |
| Saturday October 1 2005 |
| Time | Replies | Subject |
|---|
| 10:58PM |
0 |
sound file installation problem |
| 10:31PM |
0 |
chan_zap vs. Panasonic DTMF integration |
| 6:24PM |
1 |
SIP 400 Bad Request from Cisco 7960/7940 |
| 6:16PM |
0 |
Sourcing Eicon Diva V-4BRI/QuadBRI cards in Australia |
| 6:15PM |
1 |
Problem with VM Distribution Groups |
| 3:39PM |
0 |
Hangup half a call? |
| 3:38PM |
1 |
Can't compile zaptel (CVS Head) on Debian |
| 11:41AM |
3 |
Adding Cepstral to Asterisk |
| 10:53AM |
0 |
Callcenter and Softphone hanging |
| 10:29AM |
2 |
Remote call pick-up |
| 10:09AM |
1 |
Swap between callers |
| 9:23AM |
0 |
Faxdetection in IAX? (Missing audio samples) |
| 8:39AM |
1 |
Compiling Zaptel on EM64T machine |
| 8:32AM |
2 |
Asterisk - HEAD , SpanDSP, app_rx/tx Fax..what versions do you have? |
| 8:20AM |
0 |
chan_zap.c: Ring/Off-hook in strange state 6 on channel 1 |
| 7:24AM |
0 |
asterisk-oh323-0.6.7 |
| 7:17AM |
1 |
OT: RHEL / CentOS Enable APIC |
| 7:05AM |
0 |
How can I tranfer a call form one SIP phone to other during the call (unattended transfer) |
| 6:42AM |
0 |
Now can I tranfer call form one SIP phone to other during call (unattended transfer) |
| 6:36AM |
1 |
error on loading zaptel module |
| 6:33AM |
0 |
How to create IVR system using * |
| 6:32AM |
7 |
Updated presentation of Asterisk 1.2 |
| 4:52AM |
0 |
Developer help needed |
| 4:28AM |
2 |
Calls between SIP and IAX |
| 2:15AM |
0 |
VoiceGateway Design - Request for comments/suggestions |
| 12:31AM |
0 |
how to backup asterisk installation for upgrade |