Wednesday March 31 2004 |
Time | Replies | Subject |
10:14PM |
2 |
Virbiage Phones - Vapourware?? |
9:27PM |
0 |
make asterisk ignore password in sip register |
6:05PM |
7 |
Extension ringing but no ringing sound. |
5:59PM |
2 |
safe_asterisk with non-root user |
5:16PM |
1 |
LARGE BREASTS Handoff back to * from * via IAX? |
5:01PM |
2 |
ANNOUCEMENT: Tool to combine call recording into a single 'stereo' file |
3:50PM |
1 |
Voicemail prompts garbled |
3:03PM |
1 |
sip-msmessenger |
2:38PM |
1 |
VON show report |
12:00PM |
8 |
Newbie.... |
11:48AM |
2 |
SER Asterisk problem |
11:45AM |
0 |
noise in a single direction call |
11:45AM |
2 |
Basic Answering Machine Function? |
11:06AM |
0 |
WTS (200) Cisco ATA-186-I1 |
10:57AM |
2 |
C7960 "busy" notification |
10:51AM |
3 |
Voicemail Options |
10:06AM |
0 |
Config file references (was g726 not working) |
10:05AM |
0 |
Re: Asterisk-Users digest, Vol 1 #3276 - 7 msgs |
9:43AM |
3 |
SMDI support in Asterisk ? |
9:14AM |
3 |
Hangup not detected on X100P |
9:08AM |
6 |
Can't compile asterisk. |
9:06AM |
2 |
Play file from an offset |
9:06AM |
5 |
3-4 port FXO card recommendations |
9:03AM |
1 |
Need help understanding SIP phones |
9:01AM |
2 |
Asterisk as office PBX |
8:59AM |
0 |
Asterish<->Cisco Call manager |
8:50AM |
2 |
Carrier Access CMG/FXS MGCP to Asterisk, Works Fine |
8:18AM |
2 |
ATA registration requests |
8:13AM |
0 |
sip problems-debug info-will nott log on |
8:03AM |
0 |
Dial Application priorities |
7:52AM |
1 |
Reliable Provider |
7:35AM |
3 |
xml output from * ? |
7:20AM |
0 |
mailman-owner@lists.digium.com not accepting mail |
6:54AM |
1 |
Sip phone with push display? |
5:53AM |
0 |
Voicemail Name recording etc |
4:26AM |
0 |
Manager Interface "Action: Originate" change d |
4:16AM |
4 |
ANNOUNCEMENT : MeetMe Web User Interface |
3:30AM |
1 |
VoicePulse Connect & DTMF Tones |
2:56AM |
2 |
sound issue |
2:52AM |
0 |
How I Detect Any Tone using TAPI 2.0 API |
1:39AM |
2 |
RE: RxFax/spandsp: not disconnecting |
1:28AM |
0 |
Can't talk on Cisco VIP 30 using Chan Skinny |
1:05AM |
0 |
DTMF trouble on isdn: Discarding too big frame of size 1280 |
1:03AM |
1 |
Noises and echo effects |
12:53AM |
0 |
Outbound calling number problem |
12:18AM |
2 |
Asterisk and picoCell GSM Base Stations |
|
Tuesday March 30 2004 |
Time | Replies | Subject |
11:37PM |
1 |
Register vith SIP provider from behind NAT |
9:38PM |
2 |
Voicemail retrieval from Cisco 7960 |
8:19PM |
0 |
H323 in Asterisk |
8:07PM |
3 |
setting up 7940 |
6:00PM |
1 |
(no subject) |
5:46PM |
5 |
Caller entered digits ignored during wait.... |
5:45PM |
1 |
Manager Interface "Action: Originate" changed |
5:18PM |
2 |
SoftFAX/spandsp - txfax |
4:36PM |
0 |
microsoft messenger with sip debug |
4:33PM |
3 |
Sipcall.co.uk & [*] |
4:33PM |
0 |
error with microsoft messenger |
3:53PM |
2 |
Asterisk Security Audit? |
3:38PM |
1 |
G.729 and h323.conf |
2:36PM |
1 |
Cisco 7960 tftp question |
1:59PM |
1 |
G726 not working ? |
1:07PM |
0 |
problem with configuration. |
12:57PM |
0 |
SoftFAX/spandsp - release 0.0.1i - txfax fin dings |
12:31PM |
0 |
Problems with stuck PRI channel |
12:10PM |
3 |
mysql or postgresql? |
11:39AM |
0 |
is Asterisk capable for SIP-H323 translation? |
11:20AM |
1 |
Exception flag set - snom200 |
11:15AM |
1 |
IAX2 trunk mode over satellite |
10:58AM |
0 |
(no subject) |
10:57AM |
2 |
CAPI problems when loading chan_capi.so |
10:55AM |
1 |
Sacking calls to extension to voicemail |
10:55AM |
0 |
ammount of packages |
10:53AM |
2 |
VON Update - Greetings from Infomercial Central |
10:18AM |
1 |
Queue feature |
10:03AM |
0 |
repost: SIP/Asterisk behavior |
9:50AM |
1 |
Patch to chan_mgcp for IP10S for test |
9:46AM |
0 |
forget using galaxyvoice |
9:27AM |
1 |
Hot plug PCI? |
8:59AM |
0 |
No audio on outgoing SIP calls over ISDN BRI line |
8:55AM |
0 |
transfer driving me batty |
8:53AM |
2 |
Queue_log field definitions |
8:47AM |
1 |
m0nowall and * |
8:34AM |
2 |
D-Channel on span 1 down |
8:07AM |
0 |
DTMF not being detected on PhoneJack-lite |
7:50AM |
4 |
console display |
6:34AM |
2 |
Asterisk server lockup |
6:22AM |
0 |
Re: Asterisk-Users digest, Vol 1 #3260 - 13 msgs |
6:06AM |
4 |
Modems |
3:14AM |
9 |
Zaptel/PRI problem |
|
Monday March 29 2004 |
Time | Replies | Subject |
11:42PM |
0 |
vCard to cidname database tool |
7:25PM |
1 |
FW: Cisco Firmware Upgrade TFTP time out problems. |
6:41PM |
2 |
Call routing based upon callerID |
5:06PM |
1 |
pre-paid (new to asterisk, pls don't shoot on me) |
4:59PM |
2 |
Can Asterisk .... |
4:00PM |
1 |
2 - Re: Asterisk at the beginning |
3:59PM |
0 |
isdn communications interumptions |
3:51PM |
1 |
Codec between * servers |
3:45PM |
0 |
Re: *box @ home |
3:40PM |
1 |
Asterisk at the beginning |
3:29PM |
0 |
SoftFAX/spandsp - rxfax findings (spandsp-0. 0.1i) |
3:20PM |
2 |
'Busy tone' after hangup |
3:02PM |
2 |
What failed here? |
1:57PM |
3 |
asterisk @ home ? |
1:55PM |
1 |
voicemail main |
12:36PM |
5 |
VON Update - Pingtel Creates new SIP Open Source Group |
12:14PM |
2 |
Zap channels stuck in 'Rsrvd' state |
11:25AM |
6 |
Asterisk + GrandStream SIP phones |
11:23AM |
1 |
incoming SIP calls drop on pickup. |
10:48AM |
0 |
Chan_phone problems |
10:32AM |
1 |
Really Cheap way to connect 2 PBXs |
10:28AM |
0 |
SIP message header clarification sought |
10:24AM |
1 |
Cisco SmartNet maintenance agreements |
9:16AM |
0 |
still got zaptel troubles |
8:45AM |
0 |
Bug in chan_iax2.c |
8:29AM |
1 |
Connecting analog trunks to FXS card |
7:44AM |
3 |
Asterisk + ISDN4linux connectivity |
6:58AM |
0 |
hardware/software needed |
6:24AM |
4 |
Opinion poll: best SIP phones for asterisk? |
6:14AM |
1 |
testing functionality (how do I do this?) |
5:44AM |
1 |
Re: Re: Document |
5:18AM |
2 |
openline4 |
4:23AM |
0 |
one side voice with oh323 |
3:03AM |
2 |
running asterisk as ordynary user... |
2:06AM |
0 |
Slightly OT: Auto Protection Switch |
1:32AM |
0 |
Call Progress |
1:08AM |
7 |
MOH doesn't play |
|
Sunday March 28 2004 |
Time | Replies | Subject |
11:13PM |
0 |
AW: Cisco 7960 SIP Images |
9:09PM |
1 |
Programming an unlocked ADSI Astra 390 phone? |
8:39PM |
1 |
Asterisk as ISDN simulator? |
7:48PM |
0 |
Fw: Michael's Minute: Two New Products - Call-in-One, Enterprise Assessment Kit |
6:37PM |
1 |
Broken Asterisk |
5:13PM |
4 |
Error installing/compiling cdr_mysql addon |
5:00PM |
1 |
OT - Error compiling "screen" |
3:10PM |
0 |
opaque missing in Authorization header |
12:07PM |
3 |
two-stage dialing |
10:21AM |
0 |
OT Sipura: Sipura doesn't "see" * hangup PSTN line |
3:56AM |
1 |
Registers |
3:09AM |
2 |
RxFax/spandsp: file-naming of received faxes |
|
Saturday March 27 2004 |
Time | Replies | Subject |
9:20PM |
4 |
no sound via playback |
8:36PM |
5 |
Cisco 7960 SIP Images |
7:42PM |
0 |
Problems with oh323-0.5.10 |
5:51PM |
1 |
Multiple IAX2 connections |
5:00PM |
0 |
Forum notify |
4:16PM |
0 |
canreinvite and transcoding |
3:34PM |
0 |
agi and stream_file |
3:06PM |
2 |
[OT] PoE (Power over Ethernet) for 7940G |
1:51PM |
1 |
AGI crashes asterisk |
1:45AM |
0 |
Subject: Supported USB adapters ? |
|
Friday March 26 2004 |
Time | Replies | Subject |
10:42PM |
0 |
Bug 789 - Announce/Music on Hold |
8:47PM |
0 |
IAX Phone - Major New Release |
2:37PM |
1 |
ISDN -> card? -> Asterisk |
2:36PM |
1 |
DIAX Followup |
2:29PM |
0 |
FreeBSD-oriented list |
2:17PM |
0 |
Asterisk install instructions from scratch |
2:14PM |
0 |
New astguiclient released 1.0.0 |
1:49PM |
0 |
Execute AGI application in astman |
1:19PM |
1 |
Cisco ATA186 SIP transfer |
12:45PM |
1 |
Help with Asterisk Error Please? |
12:27PM |
1 |
T1 outgoing calls problem. |
12:01PM |
0 |
Solution== CALLERIDNAME and GotoIf -- Quoting Question |
8:28AM |
0 |
isdn30e E100P configuration |
8:04AM |
1 |
Newbie Softphone Problem |
6:35AM |
1 |
Help needed (New to Asterisk) |
6:14AM |
0 |
SIP Call Progress |
3:13AM |
0 |
Problem with SIPPS and ilbc |
2:48AM |
2 |
Multiple IAX "register" lines? |
2:32AM |
0 |
Re: 0.7.2 with cisco router & 7960 |
1:33AM |
1 |
SIP - Native Bridging - sipgate.de |
1:32AM |
0 |
Re: 0.7.2 with cisco router & 7960 |
|
Thursday March 25 2004 |
Time | Replies | Subject |
11:33PM |
1 |
Adtran TA750, any chance of working MWI ? |
7:36PM |
0 |
Failure SIP / RTP |
6:40PM |
2 |
IAX drops calls exactly 5 secs into the call |
6:31PM |
2 |
Watchguard Firebox 1000 and Asterisk |
6:08PM |
2 |
Codec Voodoo |
5:47PM |
1 |
New minor release of Firefly (now with Speex) |
5:38PM |
1 |
Call & Drop / Call & Tranfer - tranfering a call to a different number. |
5:29PM |
11 |
Asterisk |
5:19PM |
0 |
Error on * startup |
3:47PM |
2 |
G.729 variants and Asterisk |
3:40PM |
0 |
Dropping voice to exceptionally long queue |
3:29PM |
0 |
External and internal SIP do not work together with nat |
3:18PM |
0 |
oh323.conf, is it possible to track.. |
2:30PM |
2 |
FreeBSD Segmentation Fault on start up |
2:14PM |
1 |
Chan_sccp and lamda-solutions |
1:57PM |
0 |
(no subject) |
1:16PM |
3 |
Newbie and Meetme configuration problem |
1:07PM |
4 |
Voicemail + SIP Message header |
12:08PM |
6 |
Semi OT: WiSIP and WEP |
11:31AM |
0 |
message waiting notification issues |
11:30AM |
0 |
How detect connection setup/teardown with manager interface? |
11:26AM |
2 |
New soundfiles from Allison posted |
10:29AM |
2 |
G.729 and SCSI |
10:26AM |
0 |
IP-IP |
8:07AM |
3 |
SIP Message Extension support |
8:05AM |
0 |
IAX Termination |
6:33AM |
0 |
A tidbit about one-way audio & ethernet aliases |
6:22AM |
1 |
Asterisk with G729 codec does not want to connect with mediatrix SIP device |
6:10AM |
0 |
Register Asterisk |
5:46AM |
1 |
Distinctive Ring Detection On incoming calls |
5:28AM |
0 |
sched_settime error |
1:38AM |
0 |
Voice versus data T1s: Balance of power |
1:30AM |
0 |
Asterisk & Q.SIG |
1:26AM |
2 |
Asterisk & QSIG |
12:01AM |
0 |
Intercom/hotline/prison/keypad-less phones |
|
Wednesday March 24 2004 |
Time | Replies | Subject |
11:55PM |
0 |
ANNOUNCE: Voice Mail Box Exists AGI script |
9:44PM |
1 |
CDR and Mysql (or Postgre) |
8:54PM |
2 |
IAX and Snom200 |
8:30PM |
1 |
seperating zap |
7:43PM |
2 |
TE410P to E100P for stress test |
7:17PM |
2 |
X100P fails to detect user hung up |
4:54PM |
1 |
Long pause between background and voicemail |
3:56PM |
0 |
Garbled Music on Hold |
3:47PM |
1 |
Immixtel VOIP Adapters |
3:09PM |
1 |
RxFax questions ? |
2:26PM |
0 |
Asterisk as a Carrier SIP-PSTN Gateway |
1:07PM |
1 |
external SIP calls newbie question |
12:03PM |
0 |
R2-MFC and Wildcard E100P |
11:08AM |
1 |
ATA 182 and * |
10:13AM |
2 |
IAX2 as an IETF Standard? |
9:47AM |
0 |
transfer? |
9:44AM |
1 |
Astricon at VON in Santa Clara: Weds, Mar 31, 2004 |
8:50AM |
0 |
CALLERIDNAME and GotoIf -- Quoting Question |
8:36AM |
0 |
Any Suggestion for this system? |
8:22AM |
0 |
Asterisk for different networks in different cities |
8:08AM |
15 |
IAX2 International Termination |
7:17AM |
0 |
OT: Asterisk-Users digest Text Settings... |
6:50AM |
1 |
CAPI - MGCP problem strange behaviour |
6:15AM |
0 |
Help Asterisk - SIP Proxy |
6:06AM |
2 |
Asterisk as a standalone voicemail server |
5:39AM |
4 |
Phones can talk to asterisk but not each other through it |
5:21AM |
0 |
support for rfc3326 The Reason Header Field for SIP |
3:47AM |
0 |
Can't hear sound files |
3:41AM |
0 |
Asterix build errors. Bison related... |
2:10AM |
1 |
RE: Plugging Asterisk Security Holes.... |
1:30AM |
0 |
spandsp + libtiff 2.6.1 bad tiffs |
|
Tuesday March 23 2004 |
Time | Replies | Subject |
9:01PM |
2 |
Testing Asterisk with Broadvoice.com |
8:01PM |
2 |
UNSUNSCRIBE |
6:16PM |
1 |
Incoming Fax Call to File |
3:32PM |
2 |
Passing Argument to AGI |
1:18PM |
0 |
Script to export Master.csv to asteriskcdrdb |
12:48PM |
2 |
Asterisk SIP + Grandstream 100 + sip.conf phone HELP |
11:37AM |
0 |
Using MGCP? You're wanted! |
11:28AM |
4 |
Graphical Interface to display Asterisk CDR / php |
11:00AM |
14 |
ztdummy |
10:58AM |
1 |
Softfax problems |
10:53AM |
0 |
Call pickup - still keeps ringing? |
10:37AM |
0 |
Dumb Question |
10:28AM |
1 |
Absolutetimeout detail please? |
10:24AM |
0 |
Help needed for making chan_mgcp working for swissvoice ip10S |
10:09AM |
3 |
Nuvio users? |
9:41AM |
1 |
Asterisk 0.7.2 Patches (RDNIS and Ringing) |
9:07AM |
1 |
Information Needed |
9:05AM |
1 |
can't hear asterisk sound files on snom200 |
8:38AM |
0 |
T100p T1 CRC error monitoring |
8:32AM |
0 |
H323 calls drop on connect |
8:12AM |
2 |
Ringback? |
7:23AM |
3 |
LookupCIDName from ODBC/MSSQL |
6:27AM |
2 |
SoftFAX/spandsp: installing and results on Gentoo |
5:36AM |
0 |
Réf. : IAX2 transfers - it's great!!!! |
3:53AM |
4 |
SIP or any softphone on Mac os x |
3:39AM |
1 |
Asterisk won't start with g729 |
3:32AM |
1 |
problem voicemail national digits |
3:30AM |
0 |
Newbie seeks help: Getting Asterisk to run on Mandrake 9.2 |
1:39AM |
2 |
bleutooth gsm - usb |
1:35AM |
4 |
Convert ISDN Card in NT Mode |
12:42AM |
0 |
IAX registration problem |
|
Monday March 22 2004 |
Time | Replies | Subject |
11:28PM |
0 |
call waiting ? |
8:20PM |
1 |
Please help. Trouble getting asterisk to run on a new install. |
7:42PM |
3 |
Asterisk behind firewall and IAX |
7:38PM |
1 |
New Guy Here for Linux PBX |
7:30PM |
0 |
OT: list digest: receiving multiple digests per day....???? |
5:59PM |
2 |
Informal "Astricon" at the VON Show in Santa Clara... |
5:34PM |
0 |
g.729 pass-through mode |
5:33PM |
1 |
Inbound Toll-Free Providers |
5:33PM |
0 |
Re: Yahoo! |
4:59PM |
1 |
passing multiple arguments to agi scripts |
3:30PM |
0 |
Computing power for GSM codec |
2:50PM |
0 |
Continue Macro after Hangup |
2:35PM |
5 |
E&M Signalling |
1:49PM |
2 |
question about CPU usage |
1:35PM |
1 |
Need Called Number information via WATTS line |
1:14PM |
4 |
X100P Tone-based Supervisory Disconnect ? |
12:48PM |
2 |
Missing ringback tone on C7960 |
12:31PM |
3 |
documents |
10:28AM |
0 |
Firewalls |
10:15AM |
2 |
Asterisk AGI - Redirect not sufficient, need to link channels |
9:59AM |
1 |
CISCO Redial on Missed Call: Chan_capi Bug? |
9:58AM |
3 |
10 day old email, virus already received |
9:58AM |
0 |
How to use 2 FritzCards with asterisk? |
9:12AM |
0 |
proposed * setup, looking for feedback. |
8:57AM |
6 |
jittered voice over hisax passive card |
8:41AM |
4 |
T101P |
8:41AM |
1 |
Asterisk & DECT |
8:11AM |
1 |
Playback Volume for Record Application |
8:09AM |
0 |
7960 Configuration question |
7:36AM |
2 |
T100P not ringing. |
6:59AM |
0 |
Asterisk Possible Memory Leak |
6:53AM |
1 |
Asterisk Memory Usage |
6:12AM |
4 |
Asterisk + Radius |
5:34AM |
0 |
ISDN4Linux patch * Testers needed * |
5:13AM |
0 |
install i am new user |
4:57AM |
3 |
setvar CALLERIDNUM |
3:37AM |
1 |
X100P behind an ADSL filter? |
3:30AM |
0 |
Asterisk Diagram |
3:27AM |
1 |
asterisk: cpu load 99% |
3:22AM |
0 |
Problem with DTMF tones and Dialexa Dial-Com Lite |
|
Sunday March 21 2004 |
Time | Replies | Subject |
7:17PM |
2 |
Home users |
6:54PM |
0 |
Important: The Asterisk Mailing list(newsubject) |
6:30PM |
0 |
Cisco 7960 v6.3 firmware |
4:42PM |
1 |
UK - 1471 |
3:45PM |
7 |
Cisco 7960 vs 7905 |
3:27PM |
2 |
chan_sccp |
3:05PM |
1 |
AGI startup on channel when asterisk starts |
1:29PM |
1 |
asterisk installation problem |
11:22AM |
1 |
Any Polycom Experts Out There? |
11:05AM |
0 |
Sound prompt conversion utility? |
9:25AM |
2 |
Snom 200 Voice Call / Paging |
9:24AM |
0 |
Mantis - closing feature request when feature no added |
8:40AM |
5 |
PRI issues with TE410P |
8:32AM |
1 |
UK PSTN and x100p |
6:37AM |
9 |
If you know your party's extension # please dial it now ... |
6:27AM |
1 |
Discriminate on IAXTEL dial-in |
2:36AM |
2 |
Echo Cancellation (Newbie Qu) |
|
Saturday March 20 2004 |
Time | Replies | Subject |
8:29PM |
1 |
Subject: Re: firefly softphone |
5:36PM |
6 |
T100P T1 problem (Avaya -> asterisk IVR) |
4:28PM |
4 |
Use of Alert_Info with C7960? |
3:47PM |
2 |
can't get the full callerid php/agi |
3:22PM |
1 |
IAX2 transfers - it's great!!!! |
12:12PM |
2 |
Asterisk Integration with Evolution. |
10:42AM |
2 |
UK BT caller ID revisted |
9:38AM |
2 |
Just a question |
9:34AM |
2 |
Need an example of using the directory command |
9:30AM |
1 |
Message waiting indicators |
8:36AM |
4 |
Problem with Vegastream 50 BRI |
8:28AM |
0 |
problems with FWD solved maybe? |
8:02AM |
2 |
Basic authentication |
6:53AM |
0 |
voip terminations in Australia and New Zealand |
5:09AM |
2 |
Store caller IP in CDR |
|
Friday March 19 2004 |
Time | Replies | Subject |
11:41PM |
2 |
Asterisk and Speex |
11:39PM |
1 |
AS5300 Firmware and H323 configuration |
7:23PM |
0 |
ISDN BRI HFC-4s card, asterisk and telstra |
5:54PM |
5 |
LipZ4 Sip Soft Phone |
3:26PM |
1 |
Registration from <xxx> failed for 'xxx' |
1:15PM |
1 |
Cisco for sale |
12:21PM |
0 |
5300 to * help needed |
12:02PM |
2 |
SoftFAX/spndsp |
12:02PM |
1 |
Softphones connecting to real phones? |
11:48AM |
0 |
T100P and ADTRAN TSU600 24FXS??? |
11:38AM |
5 |
High latency from Europe, 500-800ms. |
11:18AM |
0 |
Dialogic PRI-ISA48 T96-6028 |
11:17AM |
0 |
Re: I would like to UNsubscribe from this list thanks |
10:36AM |
0 |
RE: Asterisk-Users digest, Vol 1 #3157 - 11 msgs |
10:34AM |
1 |
DID with X100P? |
10:21AM |
3 |
Asterisk Voice Mail Integration with Cisco CME |
9:31AM |
4 |
firefly softphone |
9:11AM |
1 |
g729 suggestions? |
8:23AM |
0 |
how to access the underlying channel of Local? |
8:22AM |
3 |
Fuse for Adtran 750 PSU |
7:30AM |
2 |
Newbie Start Question |
7:25AM |
4 |
ADSI slow? |
7:22AM |
0 |
asterisk & deltathree (iconnect) directly media stream |
6:48AM |
5 |
Identifying a call with manager interface |
4:32AM |
3 |
Extenesion: If InternalBusy Then GetBackToOperator |
3:50AM |
0 |
Problems with app_transfer |
1:14AM |
6 |
MOH: Copyright issues? |
12:39AM |
4 |
Using the pound (#) key while in a call |
12:38AM |
0 |
CallProgress |
12:32AM |
9 |
Important: The Asterisk Bug Tracker |
|
Thursday March 18 2004 |
Time | Replies | Subject |
10:15PM |
0 |
-Stable CVS-03/19/04-04:37:11 Not working properly |
9:35PM |
0 |
help me: warnings on Read error on sound |
8:11PM |
1 |
Grandstream G726-32 now working properly with * |
8:00PM |
0 |
RE: Asterisk-Users digest, Vol 1 #3147 - 14 msgs |
7:55PM |
1 |
Cisco 5350 One Way Sound |
7:45PM |
1 |
Redhat lastest kernel problems with Zaptel drivers kernel 2.4.20-30.9 |
6:29PM |
0 |
Cisco IOS crash with multiple SIP endpoints behind NAT |
5:24PM |
0 |
Project |
4:49PM |
0 |
RE: Text message |
4:16PM |
1 |
Problems with FWD |
4:11PM |
0 |
Softfax/spandsp - page cut-off |
3:34PM |
0 |
asterisk AGI and DTMF |
3:28PM |
1 |
Explain ring tones |
3:25PM |
0 |
CCM -> GnuGK -> * |
3:24PM |
2 |
Speaking of ring tones... |
3:17PM |
4 |
zaphfc problem |
2:19PM |
3 |
help me: warnings on Read error on sound device, Ignoring rxwink |
12:33PM |
1 |
Asterisk, X100P and AT&T PBX |
12:21PM |
4 |
Schools/Districts using asterisk? |
11:05AM |
1 |
Asterix Sip Stack |
10:27AM |
0 |
SIP problem with Nikotel |
9:56AM |
0 |
MWI only working after handset was lifted once |
9:51AM |
1 |
Monastery Devel snapshot |
9:22AM |
0 |
h323 Dialing newbie Question? |
9:20AM |
4 |
Can i do voice chat without using the hardware |
9:19AM |
0 |
openh323 w/t38 |
9:06AM |
0 |
X-Lite on both sides of NAT with * behind the NAT |
8:40AM |
1 |
Problems with asterisk and gnophone on Gentoo box |
8:35AM |
2 |
Several H323 bugfixes - working SIP <-> H.323 translator |
8:04AM |
1 |
Fax termination in Asterisk |
7:41AM |
2 |
chan_sccp latest cvs |
7:41AM |
1 |
Asterisk interoperability w/ new 64bit processors & SIP express router |
7:37AM |
0 |
thank u |
7:08AM |
1 |
Asterisk-Quintum Switches |
7:03AM |
0 |
Phantom problem authenticating with RSA? |
4:04AM |
0 |
C++ and or C# .Net development contract for Asterisk PBX Management interface |
3:36AM |
2 |
Delay Dial with Voicetronix |
3:06AM |
2 |
Asterisk with MySQL on Redhat 9 |
|
Wednesday March 17 2004 |
Time | Replies | Subject |
10:59PM |
0 |
RE: FreeBSD or Linux |
9:10PM |
1 |
Pulver WiSIP Dual Line and Hold? |
7:48PM |
3 |
local VoIP in Florida |
6:36PM |
1 |
Cisco AS5350 + Asterisk Configuration |
4:38PM |
2 |
how many potential customers out there utilizing AIX |
4:31PM |
1 |
Voicetronix Openswitch 12 |
3:39PM |
4 |
can't logon to voice mail - bad password |
3:16PM |
1 |
Asterisk in the news |
2:33PM |
0 |
RE: 800 Numbers (was Re: NuFone?) |
2:28PM |
0 |
Asterisk combined with dsp resources, billing systems, and csr and customer management |
2:28PM |
3 |
Random Echo |
2:17PM |
0 |
problems with using open h323 |
2:01PM |
1 |
Somewhat on topic but not * specific.. |
2:00PM |
1 |
X100P Echo was: USB Headsets (Plantronics DSP-400) |
1:45PM |
0 |
problem with unloading module zaphfc |
1:36PM |
1 |
USB Headsets (Plantronics DSP-400) |
1:25PM |
1 |
Any ISDN BRI card recommendations for North America? |
1:10PM |
2 |
Fax Detection on X100Ps |
1:09PM |
0 |
Asterisk/AIX and IPCentrex |
1:08PM |
0 |
newbie phone question |
12:31PM |
0 |
7960 & SCCP |
12:11PM |
0 |
Chan_sccp How-to |
11:21AM |
0 |
800 Numbers (was Re: NuFone?) |
11:12AM |
0 |
800 Numbers (was Re: NuFone?) |
10:48AM |
0 |
BellSouth Tariffs and Price lists |
10:39AM |
1 |
800 Numbers (was Re: NuFone?) |
10:31AM |
0 |
Directory App (Possible bug or undocumented feature) |
9:49AM |
1 |
Clipcomm FXO adapters |
9:13AM |
4 |
firefly sip question |
8:48AM |
1 |
warnings on Read error on sound device, Ignoring rxwink and chan_iax2.c |
8:47AM |
4 |
Traceroute equivalent |
8:14AM |
0 |
Read error on sound device, Ignoring rxwink |
8:12AM |
0 |
Buzzing X100P - call 650.210.9331 to hear it |
8:02AM |
7 |
NuFone? |
7:26AM |
1 |
Pickin up a call from another user |
7:19AM |
1 |
Busy and Unavailable |
6:37AM |
5 |
Hangup X100P Issues |
6:21AM |
1 |
Intermittent choppy speech using VoicePulse? |
6:16AM |
0 |
Asterisk support for Japanese telephone system? |
5:27AM |
0 |
VoIP with Asterisk @ CeBIT / Article about Asterisk in Newsticker |
5:15AM |
0 |
183 SIP status with vegastream ip gateway |
1:00AM |
0 |
callgroup pickupgroup and zap problem!!! |
|
Tuesday March 16 2004 |
Time | Replies | Subject |
10:43PM |
0 |
Max number of callers in a conference call |
9:48PM |
0 |
I must be an Idiot |
9:35PM |
2 |
hdlc problems |
8:56PM |
2 |
chan_iax2.c:5309 socket_read: Received mini frame before first full voice frame |
8:54PM |
0 |
Dialogic? VoipBlaster? Linux 2.4.20? |
8:51PM |
6 |
Maximum retries exceeded on call |
8:03PM |
4 |
Sipura line 1 outgoing voice problem? |
7:32PM |
0 |
The FT201 is currently being manufactured andwill be available shortly! The retail price will be $129.95 USD |
6:46PM |
0 |
spandsp in process_baud() |
6:33PM |
6 |
The FT201 is currently being manufactured and will be available shortly! The retail price will be $129.95 USD |
6:10PM |
2 |
Q931 Message - Connect - Billing |
5:31PM |
0 |
Problem loading usb-uhci module for ztdummy |
4:23PM |
0 |
Re: Asterisk-Users digest |
4:20PM |
0 |
Polycom Paging & Intercom - Please Wiki-Size |
2:57PM |
3 |
Crisco Softphone |
2:44PM |
2 |
Re: SIPURA 2000 Problems (Senad Jordanovic) |
2:42PM |
0 |
Handoff back to * from * via IAX? |
2:04PM |
1 |
Anyone got their Pulver WiSIP phone working with *? |
1:49PM |
1 |
fax pass thru issue |
12:34PM |
2 |
AGI test script |
12:27PM |
0 |
rxfax as of 03-16, crash in libtiff-3.5.7-11 |
11:54AM |
2 |
usb-uhci -- where to find it? |
11:46AM |
0 |
problems calling FWD #'s |
11:32AM |
1 |
Fax Softwares |
11:27AM |
0 |
Register Call-ID |
10:52AM |
3 |
asterisk application |
10:31AM |
0 |
Web service to start a conference and voicemail |
10:24AM |
1 |
Newbie - Bashing head on wall! - RH9 - * - How do I install AVM C2 ISDN Pretty Please! |
9:50AM |
3 |
SIPURA 2000 Problems |
9:44AM |
5 |
Web service to start a conference and voice mail |
9:09AM |
7 |
PRI Errors |
8:46AM |
2 |
RTP Read error: Resource temporarily unavailable (DTMF Issues) |
8:35AM |
2 |
Welltech FXOs |
8:00AM |
24 |
Softfax/spandsp |
7:58AM |
3 |
AGI script will not be terminated |
7:10AM |
3 |
about voice conference system. I need suggests |
7:06AM |
1 |
302 "Moved Temporarily" clears Caller*ID |
6:44AM |
4 |
Paging & Intercom |
3:33AM |
1 |
Asterisk & USR |
1:21AM |
1 |
Problems with TDMoE |
|
Monday March 15 2004 |
Time | Replies | Subject |
11:00PM |
0 |
alias in h323.conf |
5:25PM |
3 |
x100p CLI in the UK |
4:32PM |
3 |
New Firefly Beta - with SIP and G.729 |
4:28PM |
3 |
MySQL Dynamic Extensions |
4:21PM |
5 |
Need help to format asterisk MGCP packet. |
4:13PM |
0 |
Transparent Switch - PRI / IAX2 / PRI |
3:17PM |
0 |
Merlin Legend trunk ports |
3:00PM |
10 |
Sipura click click bad quality |
2:25PM |
1 |
WiSip SIP settings locked? |
2:14PM |
1 |
Truncated Tcp Options? |
2:07PM |
1 |
ZapRAS over IAX anyone? |
1:54PM |
0 |
"Click to Call" Perl CGI script - TACI |
1:26PM |
1 |
AgentCallBackLogin ?? |
12:48PM |
1 |
Pri Errors, Hanging up Owner |
12:07PM |
1 |
dbinit error message |
12:04PM |
1 |
Conference call? |
11:53AM |
0 |
Cisco Call Manager and Asterisk and fastbusy? |
11:47AM |
0 |
extensions problem (SIP) |
11:46AM |
2 |
Bluetooth |
10:52AM |
3 |
asterisk MySQL |
10:31AM |
0 |
SIP Calls with * and Cisco AS5300 |
9:37AM |
3 |
ISDN BRI with DDI support |
9:15AM |
2 |
Guru's help with * and AVM C2 ISDN - Newbie going mad!! |
8:59AM |
0 |
ChanCAPI and withheld / unavailable callerid indication |
8:56AM |
1 |
DTMF debugs |
8:52AM |
0 |
Asterisk-Users digest, Vol 1 #3101 - 14 msgs Subject: Re: Asterisk on KNOPPIX, I have it working, somewhat. |
8:30AM |
0 |
post |
7:28AM |
4 |
OT: cisco 7960G powered by 3com 3CNJPSE |
5:18AM |
0 |
Compiling snom firmware |
4:09AM |
1 |
Asterisk & Analog Modems |
3:25AM |
1 |
IPC5000 (WIP-5000 from hitachi cable) |
2:22AM |
0 |
libpri problem with SystemX telco |
1:31AM |
1 |
error, installing asterisk |
12:07AM |
0 |
H323 Gate Keeper, Planet VIP-400 |
|
Sunday March 14 2004 |
Time | Replies | Subject |
11:53PM |
0 |
Receptionist Interface |
11:21PM |
3 |
Grandstream TFTP Config |
7:21PM |
4 |
Calling one local SIP user from another (using X-Lite) |
5:44PM |
1 |
VoYP.Net: voip directory and ENUM registry |
11:43AM |
1 |
ISDN PRI A and B, cry for help. |
8:38AM |
0 |
To snip or not to snip? |
8:34AM |
0 |
RING,BUSY,congestion indication on chan_modem |
6:11AM |
1 |
RH 9 with AVM C2 ISDN - New User - Guru's Help needed Urgent!! |
5:58AM |
3 |
Variable digit length in national dial plan |
3:45AM |
5 |
European Caller ID |
|
Saturday March 13 2004 |
Time | Replies | Subject |
10:19PM |
0 |
register request time out |
10:07PM |
2 |
Which CODEC is my phone using? |
8:04PM |
6 |
Consultants |
4:51PM |
1 |
Fedora w/o capi isdn - any other recomended distribution? |
4:31PM |
2 |
General Caller ID question |
4:08PM |
0 |
SIP Recv error when talking via asterisk |
3:15PM |
4 |
VXML_URL and Cisco 7960 Phones? |
1:57PM |
0 |
It's dead jim! |
12:31PM |
4 |
incoming fax x100p |
12:21PM |
0 |
Stopping dtmf signals from being detected |
11:16AM |
7 |
Cisco 7960 firmware |
11:09AM |
1 |
Panther OS X Installation |
10:55AM |
0 |
Incominglimit - SIP verses AIX, phone busy |
10:03AM |
4 |
How to send CallerID trough CAPI ? |
8:31AM |
2 |
Asterisk on KNOPPIX, I have it working, somewhat. |
4:52AM |
0 |
Re: Asterisk-Users digest, Vol 1 #3092 - 11 msgs |
3:20AM |
0 |
Extensions do not display CallerID |
2:25AM |
1 |
Resetting Grandstream HT-286 to factory default settings? |
2:14AM |
1 |
MySQL changes license... |
|
Friday March 12 2004 |
Time | Replies | Subject |
8:40PM |
2 |
Asterisk/IVR general inquiry |
7:49PM |
0 |
telephone blacklist example |
6:38PM |
0 |
Fwd: Re: XML Phone book software. |
5:03PM |
3 |
Asterisk-to-Asterisk call setup problem (one way works fine) |
4:40PM |
0 |
New astguiclient release 0.9.4 |
4:37PM |
0 |
UK callerID on BT line |
4:24PM |
0 |
Zyxel wifi sip phone |
4:13PM |
1 |
Cisco SIP license |
3:18PM |
1 |
Dial via X100P |
3:12PM |
1 |
Cannot call extensions or make outgoing calls |
2:45PM |
1 |
Hang-ups when using IAX |
2:31PM |
0 |
GNUGK user |
2:24PM |
0 |
IAX2 jitter issue at interval |
2:17PM |
4 |
ast_rtp_raw_write errors distorting sound on G729 passthrough |
12:02PM |
2 |
Codec negotation with re-invites.. |
11:17AM |
2 |
TDM410 final questions |
10:53AM |
1 |
callprogress on outgoing calls placed via /var/spool/asterisk/outgoing |
10:42AM |
0 |
zap call being dropped after 7 seconds - SIP phone with public IP (no NAT) |
10:32AM |
0 |
oh323 chan problem? |
10:20AM |
3 |
Strange Problem |
7:53AM |
2 |
X100P and TDM400 questions |
7:39AM |
4 |
Cisco Call Manager and Asterisk and fast busy? |
6:55AM |
1 |
Empty voicemails |
6:41AM |
2 |
LDAP user directory |
6:31AM |
1 |
FXO via Cisco VIC? |
6:24AM |
0 |
Transfer and Native Bridge unwanted - was Native Bridge and Billing |
5:42AM |
0 |
Help on two subjects |
5:10AM |
0 |
Native Bridge and Billing |
4:54AM |
0 |
call bridge |
4:41AM |
1 |
Fax redirection problem |
4:10AM |
0 |
loopstart,kewlstart,groundstart |
3:57AM |
0 |
UDC SYSTEMS |
2:26AM |
2 |
SIP call to ISDN subscriber |
|
Thursday March 11 2004 |
Time | Replies | Subject |
11:55PM |
0 |
HDLC overrun, what's going on? |
11:29PM |
0 |
asterisk and text to speech? |
10:03PM |
4 |
PCI front mount chassis? |
6:58PM |
3 |
oh323 sending tech-prefix?? |
5:25PM |
1 |
Re: pickup account code in agi |
4:10PM |
0 |
Cisco 7960 and short delay before voice star tsafter ring. |
4:04PM |
1 |
error with dates? |
3:43PM |
2 |
MOH overIAX2 not working. |
3:40PM |
0 |
MGCP error in verbose console startup. |
3:06PM |
6 |
XML Phone book software. |
2:47PM |
0 |
Crossconnect VoIP and PSTN in India. Is it allowed? {Scanned} |
2:37PM |
0 |
MGCP RELOAD function |
2:14PM |
3 |
* and PrePaid |
1:53PM |
2 |
Night menu not working |
1:46PM |
1 |
Cisco 7960 and short delay before voice startsafter ring. |
1:24PM |
0 |
German ringtone |
11:36AM |
0 |
stealth asterisk (XP100->PBX Handset) |
11:23AM |
2 |
Using MultiTech MVP-210 as FXO/FXS gateway |
11:15AM |
4 |
IPC5000 - Wireless Sip phone |
10:27AM |
0 |
Agents and delay before and after they handle a call |
10:17AM |
0 |
GSM Bandwidth - Test x Measures |
10:06AM |
1 |
Nitsuko 124i interface, anyone? |
9:57AM |
0 |
IAX Phone: Now Supports Windows 98/ME |
9:50AM |
0 |
remote dtmf |
9:40AM |
2 |
Soundcard question |
9:37AM |
3 |
Music on Hold sound "goes off" if environment is silent |
9:27AM |
0 |
MySQL VM config |
9:19AM |
0 |
Streaming calls to the Internet - A Mini How-To |
9:14AM |
5 |
Cannot use # key to transfer calls |
9:00AM |
3 |
PRI errors blocking Asterisk |
7:58AM |
1 |
G.729 passthrough notes (wiki fodder?) |
7:54AM |
1 |
SIP native bridge vs. SIP reinvite |
7:26AM |
1 |
How can I use the # key normally? |
6:51AM |
3 |
Have Voice Mail tell the extension? |
6:45AM |
1 |
OpenBSD patches |
6:44AM |
0 |
who has German voice files ? |
4:41AM |
6 |
Asterisk & CAPI & DECT problem |
3:10AM |
7 |
asterisk gui client |
3:05AM |
1 |
Re: Fax support and 'f' DTMF tone extension & Asterisk mangling faxes |
1:12AM |
0 |
Adtran TA 750 Channel Bank config |
|
Wednesday March 10 2004 |
Time | Replies | Subject |
11:23PM |
0 |
Asterisk connection to Cisco Call Manager |
5:50PM |
5 |
IAX Connection-Latency/Bandwidth Question |
4:59PM |
0 |
DTMF Dial incomplete number |
4:37PM |
1 |
app_prepaid.c |
4:35PM |
1 |
(no subject) |
4:14PM |
11 |
Predictive Dialer |
4:00PM |
1 |
Voicemail: Does the numbering of files follow file age? |
3:14PM |
0 |
Good corporate level speaker phone for use with Asterisk |
2:54PM |
2 |
There is no FORK! (WAS Re: BETA RADIUS support for Asterisk) |
2:44PM |
1 |
Fw: where can I get Commedian mail at? |
1:13PM |
3 |
BETA RADIUS support for Asterisk |
11:40AM |
0 |
Queues and AGI scripting |
9:11AM |
1 |
Anyone using the new TE405P? |
8:24AM |
1 |
TDM400P - upgradable how? |
8:01AM |
4 |
403 Forbidden |
7:48AM |
0 |
Eicon DIva Server BRI 2M |
7:39AM |
3 |
Asterisk mangling faxes |
6:28AM |
0 |
Fwd: got (-2) from Queue |
5:55AM |
0 |
Sipura Dial Plan |
5:11AM |
7 |
asterisk-oh323, new version 0.5.10 |
3:50AM |
2 |
Program to manage the faxes |
2:34AM |
1 |
Ast_smoother_feed problems |
1:09AM |
1 |
Runing asterisk with voicetronix (fwd) |
1:06AM |
2 |
TE410P cards E1 ports not working! |
|
Tuesday March 9 2004 |
Time | Replies | Subject |
9:51PM |
2 |
BCM Wireless SIP Phone |
8:27PM |
0 |
Astguiclient was SIP - Receptionist |
6:58PM |
0 |
problem * two ISP |
6:57PM |
0 |
3 second echo |
4:39PM |
2 |
Phone with large display |
3:34PM |
2 |
IAXTel multiple registers? |
3:32PM |
0 |
adtran? |
3:25PM |
1 |
Parsing a variable, or rather Splitting a variable |
3:19PM |
0 |
voicetronix openswitch -- bad echo |
2:23PM |
4 |
Outbound Transfer and the # key |
2:11PM |
0 |
nec etw-8-1 (bk)tel or other pbx handset to asterisk |
1:30PM |
0 |
IVR plus small customer support center (was IVR setup) |
1:27PM |
2 |
Radius |
1:16PM |
5 |
From 0 to PBX in 2 hours |
12:50PM |
1 |
Polycom NetEngine 6200 Router |
12:27PM |
1 |
Yet Another Newbie |
12:11PM |
2 |
speex codec problem |
11:53AM |
0 |
Grandstream European Dialtone. |
11:49AM |
3 |
Low cost VOIP phone with headset possibility |
11:36AM |
2 |
Dialplan that changes tith time of day |
11:27AM |
1 |
got (-2) from Queue |
10:54AM |
0 |
BudgeTones losing registration? |
10:47AM |
3 |
SIP 3.0 |
10:43AM |
1 |
logic problem with GotoIf? |
9:50AM |
0 |
dialing problem isdn nt controller |
9:35AM |
2 |
T38 Modem |
9:30AM |
0 |
Réf. : RE: chan_sip.c Error |
9:24AM |
0 |
Asterisk GUI ? |
9:18AM |
0 |
Manually loading modules. |
8:44AM |
0 |
need advice (newbies) |
8:26AM |
2 |
chan_sip.c Error |
7:11AM |
0 |
On which channels manager redirect work? |
7:07AM |
0 |
zaphfc: no channel |
5:31AM |
0 |
Asterisk demo sounds choppy |
5:18AM |
0 |
how to connect ohphone & asterisk on the same box? |
4:49AM |
1 |
Asterisk as voicemail only. |
3:51AM |
2 |
Asterisk & Hylafax |
2:50AM |
0 |
FAX Sending |
1:13AM |
0 |
iax2 trunk - no matching peers |
|
Monday March 8 2004 |
Time | Replies | Subject |
11:00PM |
0 |
Frames/Packet Development |
10:17PM |
2 |
RTP Read error: Resource temporarily unavailable |
8:05PM |
0 |
monastery mailinglist |
7:15PM |
3 |
SIP registration fails |
7:06PM |
1 |
Most common CODEC |
6:06PM |
1 |
Monastery - question |
5:38PM |
0 |
H.323 call return code handling |
5:29PM |
5 |
Asterisk Codecs [G.729] |
5:02PM |
1 |
Codec Translation Problem on IAX Softphones - Incoming Only |
4:22PM |
0 |
All-Page in Asterisk |
3:20PM |
0 |
New astguiclient release |
2:56PM |
1 |
Limiting simultaneous inbound SIP calls |
2:27PM |
5 |
SIP - Receptionist |
2:20PM |
0 |
ZAP/Call Waiting... |
1:33PM |
1 |
Shorewall and asterisk on Mandrake |
1:14PM |
5 |
message lights and stutter tones |
12:07PM |
0 |
Queue to zap group |
12:03PM |
1 |
SIP Conference Bridge? |
10:58AM |
5 |
windows alternitives to Asterisk? |
10:38AM |
0 |
Asterisk Management Tool |
10:31AM |
0 |
Free World Dialup |
10:27AM |
1 |
IAXtel Broken? |
9:47AM |
1 |
IAX2 echo cancellation? |
8:55AM |
2 |
monastery devel page |
8:52AM |
0 |
Can't Call out |
8:37AM |
4 |
x100p volume |
8:33AM |
1 |
Last Number Redial for Analog Phones |
8:22AM |
0 |
PBX and Grandstream |
7:56AM |
2 |
ISDN BRI VoIP & Internet |
7:08AM |
1 |
Cisco Call Manager and Asterisk? |
6:51AM |
3 |
Hylafax integration |
6:17AM |
0 |
Dialplan for voice menu and two extensions |
5:53AM |
1 |
Zap to Zap |
|
Sunday March 7 2004 |
Time | Replies | Subject |
11:00PM |
1 |
Asterisk PLAR? |
7:07PM |
0 |
STUN command line client? |
4:12PM |
4 |
Options for 3+ FXO ports |
3:21PM |
2 |
[asterisk] Re: Voiceplus |
2:36PM |
0 |
New version of DIAX (0.9.7a) available for download |
2:32PM |
0 |
monastery update |
1:29PM |
5 |
Best Budgetone firmware? |
9:46AM |
2 |
X100P dial in/out to sip phones |
9:32AM |
0 |
PRI: Read on 36 failed and Got reject for frame 23 |
9:22AM |
3 |
Crossconnect VoIP and PSTN in India. Is it allowed? |
8:17AM |
1 |
IAXTEL and 800 numbers |
7:14AM |
0 |
sipsock_read: Recv error: Resource temporarily unavailable |
1:27AM |
3 |
peer is UNREACHABLE when using XLITE |
1:20AM |
1 |
Max # of callers in a conference... |
1:14AM |
5 |
Limit on call in minuttes. |
|
Saturday March 6 2004 |
Time | Replies | Subject |
9:29PM |
3 |
Voiceplus |
6:25PM |
0 |
VIP 400 from Planet and asterisk |
3:39PM |
2 |
MFE for TEI=76 |
3:27PM |
2 |
Help Newbie: TDM Development Kit |
3:23PM |
0 |
re: cdr and macros |
2:18PM |
0 |
CLI message |
2:14PM |
1 |
monastery |
1:40PM |
0 |
danish voice |
12:57PM |
0 |
Update on System Freeze - Longest run yet - What I changed |
12:55PM |
1 |
AM-WEB |
12:02PM |
0 |
re: cdr and macros |
11:07AM |
6 |
new2agi -php |
10:54AM |
3 |
Grandstream Budgetone SIP registration fails |
9:18AM |
1 |
Mediatrix 1104 Configuration |
8:35AM |
1 |
Incoming SIP calls |
7:59AM |
2 |
GS HandyTone-286 Transfer Problem, can anyone confirm? |
6:54AM |
0 |
cannot access console |
6:16AM |
0 |
Message: "Power alarm on module 1, resetting!" |
3:03AM |
2 |
x100p Q. |
|
Friday March 5 2004 |
Time | Replies | Subject |
11:00PM |
4 |
E1 Red Alarm |
10:29PM |
1 |
7960 conference ? |
9:38PM |
0 |
gnophone and sip phone |
7:39PM |
1 |
Zap to SIP transfer problem |
6:04PM |
3 |
Call roll-over question... |
5:59PM |
1 |
Internet Phone Concept Question |
5:04PM |
1 |
SIP and distinctive ring |
4:11PM |
0 |
Asterisk halts/stop in a reload command. |
4:01PM |
1 |
E100P / E1 dial out |
3:05PM |
1 |
Dialogic supported well? |
12:28PM |
0 |
ADSI and a SIP ATA |
12:20PM |
0 |
RE: Asterisk-Users digest, Vol 1 #3012 - 11 msgs |
11:29AM |
6 |
Simple * status |
10:26AM |
3 |
dropped calls |
10:19AM |
1 |
Kernel - TE410P |
10:01AM |
0 |
SIP => Zaptel TDM400P issue |
9:17AM |
1 |
Not able to dial "9" to get out with SIP Grandstream BudgeTone-100 or SIP softphone |
8:18AM |
3 |
OT: Snom 105 |
8:16AM |
5 |
Ahead SIPPS and Asterisk |
7:38AM |
0 |
H323 termination to Cisco 5300 |
5:15AM |
0 |
Can't find capi.conf syntax to use 2 controllers |
4:18AM |
5 |
how to disable zap debug!!! |
1:04AM |
3 |
flash button on GS101 |
12:16AM |
0 |
Runing asterisk with voicetronix |
|
Thursday March 4 2004 |
Time | Replies | Subject |
11:56PM |
1 |
MySQL configurations and other things. |
11:00PM |
2 |
Asterisk fault tolerance and a embedded hardware solution.....?? |
9:51PM |
0 |
RE: Palm OS5 client |
8:59PM |
0 |
Question about 'zap show channels' |
6:30PM |
0 |
RE: Palm OS5 client |
5:33PM |
1 |
Asterisk crashed so often |
4:28PM |
2 |
Do I have a bad T100P? |
3:32PM |
6 |
G.729 vs. G.729 pass thru |
3:12PM |
1 |
segfault and backtrace info |
1:47PM |
4 |
adtran 750 + t100p |
1:39PM |
1 |
ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device |
1:28PM |
0 |
Outbound fax with T100P |
12:56PM |
0 |
Ringing noise |
11:44AM |
0 |
Outbound fax using T100P |
11:32AM |
3 |
Roaming extension |
10:53AM |
1 |
Voicepulse error |
10:51AM |
0 |
Looking for NY Asterisk Folks |
10:49AM |
1 |
Cisco ATA 186 cannot make a call |
10:30AM |
1 |
Newbie questions, call waiting/700 calling/etc... |
10:27AM |
1 |
CPU load |
9:41AM |
1 |
calls being presented as "Anonymous" |
9:34AM |
2 |
Stupid AbsoluteTimeout Tricks |
9:33AM |
1 |
Question regarding MusicOnHold ... |
9:24AM |
3 |
what marks a vm message as old? |
8:30AM |
1 |
Best ATA 186 Firmware - my mistake - btw gs 486 is coming |
8:17AM |
1 |
Is this connection scheme possible? |
8:09AM |
2 |
Voicemail has hard-coded limit of 100 messages? |
5:49AM |
3 |
ALSA Sound dies after a while |
5:13AM |
0 |
RE: Palm OS5 client |
3:29AM |
6 |
Error compiling zaptel |
2:46AM |
0 |
Snom phones |
2:14AM |
1 |
Problem with X101P. |
2:08AM |
3 |
Cisco 7960 and short delay before voice star ts after ring. |
1:28AM |
2 |
oh323 - ztdummy |
12:54AM |
1 |
voicemail not working with mysql !!! |
12:05AM |
2 |
RE: Palm OS5 client |
|
Wednesday March 3 2004 |
Time | Replies | Subject |
11:40PM |
0 |
Handling of AddQueueMember error |
11:14PM |
1 |
Meetme 'd' and 'p' flags mutually exclusive with wcfxo driver but not ztdummy |
11:04PM |
0 |
4 Port IADS |
10:55PM |
2 |
TDM400 hardware problems? |
10:51PM |
2 |
Best ATA 186 Firmware |
10:22PM |
0 |
More recordings for Allison |
10:20PM |
1 |
Retrieving an application command return code |
7:26PM |
1 |
H.323 ASN.1 Vulnerabilities: Request for "official" patch! |
6:31PM |
1 |
on hold music from a mp3 stream or sound card input? |
4:01PM |
8 |
wct1xxp module and the T100P |
3:53PM |
3 |
canreinvite=yes in sip.conf still go through asterisk |
3:26PM |
1 |
Cisco VIP30 |
3:16PM |
0 |
Asterisk Setup and configuration help |
3:06PM |
2 |
Digium CVS Server: Connection refused? |
2:08PM |
3 |
E100P UK PRI Configuration |
1:40PM |
3 |
ENUM when your country's ITU representative is uncooperative |
1:34PM |
1 |
Deleting old voicemail messages |
1:26PM |
0 |
Professional Text to Speech |
1:25PM |
1 |
Music on Hold with Pingtel? |
11:47AM |
0 |
Sorry about the post, meant to be off-list not on. |
11:34AM |
4 |
VMware, * and SJphone ... newbie |
11:13AM |
4 |
Status of SIP with an outgoing proxy? |
10:47AM |
1 |
RE: Asterisk-Users digest, Vol 1 #2989 - 8 msgs |
10:31AM |
3 |
Ringing Delay |
10:01AM |
1 |
Status Lights on Snom200 Phone Displaying the Status of PSTN Lines |
9:53AM |
0 |
NAT, Asterisk and SIP service provider (sipgate.de) |
9:44AM |
4 |
FWD registration faillures |
9:35AM |
1 |
Segfault when parking from extension dialed inside AGI. |
9:26AM |
0 |
Asteriks & ALSA??? |
9:05AM |
0 |
Call Transfers from SIP |
8:36AM |
0 |
IAX image for SNOM 200? |
8:00AM |
1 |
Troubles with Galaxyvoice |
7:08AM |
2 |
KPN BRI |
6:57AM |
1 |
does usb-ohci work for ztdummy? |
6:27AM |
2 |
DPNSS and Asterisk |
5:55AM |
5 |
Cisco 7960 and short delay before voice starts after ring. |
5:33AM |
1 |
PRI and Voicemail Memory increasing |
5:12AM |
3 |
Hanging GS101 in a upright position |
5:03AM |
11 |
chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call |
4:36AM |
1 |
problem to place calls to NIKOTEL |
3:03AM |
0 |
Trunking with ztdummy for timing? |
2:39AM |
2 |
Size of PC for conferencing? |
2:26AM |
0 |
Caller ID Name Display |
|
Tuesday March 2 2004 |
Time | Replies | Subject |
11:17PM |
2 |
Routing NOTIFY Messages? |
9:30PM |
2 |
SCO finds someone to pay!!! |
8:28PM |
2 |
gs on phone ? |
7:02PM |
0 |
Is there a BIG difference between a softphone like X-Lite and a hard VOIP phone |
5:12PM |
2 |
Motorola / Vanguard, H.323 and Asterisk |
4:38PM |
0 |
prepaid app |
4:28PM |
0 |
Three-way-calling |
4:27PM |
0 |
Codec translation problems? |
4:14PM |
0 |
VoIP provider in Italiy with terminiation? |
4:02PM |
7 |
Cisco IP Phones |
3:07PM |
1 |
cdr->dst incorrect? Bug submitted. |
2:43PM |
2 |
T.38 fax (off-topic) |
1:50PM |
1 |
Supervised transfer (almost) with GS phone |
12:59PM |
1 |
SRTP: followup |
11:52AM |
1 |
E911 support |
11:20AM |
1 |
Calls not hanging up. |
10:58AM |
1 |
Newbie Voicemenu question |
10:53AM |
8 |
VTGO-PG and IPP200 |
10:02AM |
0 |
E1 - Signal - PBX connection |
9:46AM |
1 |
consultative call transfert with mgcp |
9:41AM |
8 |
Anybody know about the Sayson 480i VoIP Screen Phone? |
9:27AM |
1 |
Asterisk Passthrough |
8:43AM |
0 |
Disa & # |
7:47AM |
0 |
Is there an * based distro? |
7:12AM |
1 |
No Ringback Tone when Dialing (outside caller to internal extension from auto-attendant) |
6:22AM |
1 |
flash button |
6:21AM |
0 |
call flip-flop with a phone connected to a TDM400P |
5:28AM |
0 |
inband dtmf |
4:51AM |
0 |
ext.conf for european variable length dialplan ? |
4:00AM |
3 |
Small office requirements - Can this be done ? |
2:42AM |
3 |
Does it exist - DNS "TX" record? |
2:33AM |
0 |
Queues with chan_mgcp |
1:36AM |
2 |
having users in sql |
12:14AM |
0 |
Re: Asterisk-Users digest, Vol 1 #2959 - 10 msgs |
12:02AM |
1 |
Record Application |
|
Monday March 1 2004 |
Time | Replies | Subject |
11:52PM |
0 |
IAX Native bridge |
11:45PM |
1 |
Any Gentoo Users Running ASTERISK had problems on recent emerge -u world? |
11:45PM |
0 |
ACD & autologoff problem |
11:09PM |
1 |
Cisco LDAP directory |
10:01PM |
0 |
Re: Asterisk-Users digest, Vol 1 #2958 - 13 msgs |
8:15PM |
1 |
Garbled Faxes |
8:11PM |
0 |
anyone using the Motorola vt-100 adaptor? |
5:57PM |
1 |
Re: Asterisk-Users digest, Vol 1 #2964 - 13 msgs |
5:50PM |
0 |
SIP channel question |
5:14PM |
0 |
Stream both sides of conversation out sound card? |
4:49PM |
1 |
RTP connection broken |
3:28PM |
2 |
Stange notices and Warnings.. |
3:09PM |
10 |
Fax detected, but no fax extension |
2:53PM |
4 |
Dial up adapter |
2:38PM |
1 |
SS7 capability |
2:26PM |
2 |
Hotel and wireless questions |
1:31PM |
0 |
Tying into an Inter-Tel AXXESS |
1:24PM |
1 |
Fw: Asterisk stable how to compile ? |
1:10PM |
0 |
IAX and E1 Call State |
12:59PM |
1 |
GotoIf voicemail message is too long?? |
12:38PM |
2 |
Block Callerid with VoicPulse Connect! |
9:51AM |
7 |
Small office requirements - Can this be done? |
9:50AM |
3 |
Asterisk stable how to compile ? |
8:10AM |
0 |
CID on TDM400P Quad FXS |
8:02AM |
20 |
CVS login |
6:16AM |
0 |
IAXTel bridge to FWD |
4:28AM |
1 |
Dial out on Capi with more MSN´s |