Friday December 31 2004 |
Time | Replies | Subject |
7:49PM |
2 |
FC2 & ztcfg - cannot find channel 2 |
7:41PM |
1 |
how is a upgrade performed? |
7:30PM |
1 |
Help With Configuration From Odbc |
7:14PM |
1 |
FC3 compile with new 2.6.10 fails |
4:44PM |
4 |
is wiki drunk |
3:49PM |
7 |
Softphone in German |
2:45PM |
2 |
Mysql-Realtime and ASTCC |
2:19PM |
0 |
manager API / weird queue |
2:08PM |
4 |
IAX media |
10:29AM |
0 |
Segmentation Fault (core dumped) |
9:48AM |
1 |
Broken pipe... |
9:41AM |
1 |
BroadVoice WiSIP with Asterisk |
8:00AM |
0 |
Thanks for help - Almost done - 50% - Can hear |
7:24AM |
0 |
Segmentation Fault Problem |
6:03AM |
2 |
MGCP parameters |
6:00AM |
2 |
Host IP address, Crash on startup, Console grabs soundcard - Newbie needs help |
6:00AM |
3 |
IAX users |
3:59AM |
2 |
hardened gentoo (selinux) asterisk problem |
12:22AM |
3 |
FXO to IAX on ethernet. or FXO to SIP on Ethernet |
|
Thursday December 30 2004 |
Time | Replies | Subject |
10:18PM |
2 |
TE410P not Interrupting |
7:38PM |
0 |
A Single user |
6:19PM |
1 |
A simple scenario |
5:47PM |
0 |
TDM04b failures (xpost on purpose) |
3:45PM |
1 |
IAXy issues |
2:58PM |
0 |
chan_capi and voicemail to cellnumbers crashing asterisk |
2:38PM |
2 |
VoiceConduits - Notice |
2:37PM |
1 |
Queues strategy |
1:58PM |
1 |
Weird..bridging to Zap channel FXS instead of bridging to PSTN FXO on outgoing group |
1:39PM |
1 |
RealTime Drivers Connectivity Error |
1:34PM |
2 |
VoiceConduits is a scam |
12:47PM |
1 |
Agent login state saving? |
12:01PM |
2 |
Fw: Cisco 7690 Voicemail Problem |
12:00PM |
0 |
Asterisk dialing a Zap channel FXS instead of bridging to PSTN FXO |
11:57AM |
0 |
Problems starting * |
11:22AM |
1 |
More * weirdness |
10:11AM |
0 |
Zapatel ringing multiple SIP devices |
10:09AM |
2 |
IAX2 and DTMF |
10:08AM |
4 |
Voicemail and Zapatel |
9:51AM |
1 |
CDR IAX calls snafu ? |
9:05AM |
0 |
This item has been released from quarantine. |
8:54AM |
6 |
Nagios and Asterisk |
8:53AM |
1 |
DTMF skipped when calling from ISDN to SIP... |
8:52AM |
2 |
Sipura 3000 inbound FXO problem |
8:32AM |
2 |
IAX hardware |
8:28AM |
0 |
Fw: Open ports on router in front of asterisk |
8:18AM |
4 |
Helping communications to Asia area. |
7:26AM |
0 |
Asterisk with 2 E100P cards behind an Alcatel 440 |
7:04AM |
0 |
New Diax version 0.9.9f |
6:52AM |
11 |
Is asterisk that unstable ???? |
6:03AM |
0 |
VoDSL without using IAD |
4:34AM |
0 |
Re: Asterisk and Capi |
4:33AM |
1 |
Doubts about the Monitoring command |
|
Wednesday December 29 2004 |
Time | Replies | Subject |
10:59PM |
0 |
12 CANCEL's followed by 12 INVITE's in 5 secs |
10:00PM |
0 |
how does ipphone pick up voicemail alert? |
9:03PM |
2 |
Problem with Digium TDM04B |
6:50PM |
5 |
automatic startup |
6:37PM |
1 |
show version |
6:00PM |
2 |
So what if I can't dial out ... or in ... Asterisk just blows my mind! |
5:52PM |
5 |
PRI Woes continue |
5:43PM |
0 |
ISDN4Linux Incoming calls |
5:31PM |
2 |
RE: Hook/Flash, Hold, Call Waiting, Three Way Calling |
5:07PM |
1 |
Issue with Mediatrix 1124 |
5:05PM |
0 |
(no subject) |
4:35PM |
0 |
Channel Zap/4-1 in prering state |
3:46PM |
2 |
Hardphones Console o Secretarial One |
3:32PM |
2 |
Asterisk, she no hang uppa the phone! |
3:30PM |
1 |
Can I tell if it hung up due to busydetect or disconnect supervision? |
3:16PM |
1 |
Dial with no phone line connected |
2:51PM |
0 |
queueing question |
2:16PM |
3 |
DSLink modem freeze |
1:52PM |
3 |
Recording/Monitoring a call mid-stream? |
12:13PM |
1 |
RFI: Creating a database of DID providers |
11:59AM |
5 |
zapata.conf not being parsed by * |
11:26AM |
1 |
Hmmm - anyone seen this before? |
8:57AM |
0 |
IAX -> IAX -> SIP problems |
8:50AM |
9 |
IP Phone recommendations? |
8:30AM |
0 |
Supporting "End User Line Features" |
7:44AM |
0 |
trimming messages on reply |
7:24AM |
1 |
Polycomm IP500 dropping incoming calls |
6:42AM |
0 |
AstTAPI - Incoming Calls |
5:56AM |
1 |
Asterisk OH323 acting as a gatekeeper |
5:37AM |
1 |
Impossible to compile last version of Asterisk |
3:59AM |
2 |
TE110P doesn't appear in /proc/zaptel |
3:16AM |
0 |
Problem with musiconhold - No such file or directory |
3:08AM |
1 |
API Manager Events |
2:53AM |
0 |
Determine UAS on remote SIP phones |
2:51AM |
7 |
Final call for departments |
1:41AM |
5 |
spandsp-0.0.2pre6 |
|
Tuesday December 28 2004 |
Time | Replies | Subject |
10:18PM |
6 |
OT: Linux routing with T100P problems |
9:18PM |
0 |
service activation code |
8:04PM |
0 |
500 "Internal Server Error" |
7:34PM |
1 |
Sending e-mail from dialplan |
7:28PM |
3 |
Fedora Core 3 app_curl compile error? |
7:10PM |
4 |
Invalid Extension |
5:45PM |
0 |
How to connect two Asterisks as secure as po ssiblewithout too much additional bandwidth ? |
5:22PM |
1 |
PRI & CPU Usage |
5:20PM |
3 |
Dialplan variables |
5:18PM |
1 |
Meetme scalable to 300 people? |
4:39PM |
1 |
ASTCC Expiration |
4:32PM |
2 |
caller-id blocking |
4:19PM |
1 |
Intercom System with Asterisk and Cisco 7960 |
2:33PM |
0 |
Calling Card question |
2:24PM |
2 |
WARNING[22314]: No such switch 'Realtime' |
2:19PM |
6 |
Music instead of Tunes |
1:36PM |
3 |
ZtDummy vs Hardware |
1:22PM |
0 |
FW: Compile Error |
1:17PM |
1 |
Hardware opinions? |
1:12PM |
0 |
external Radius Server integration with asterisk |
1:09PM |
0 |
Two problems with the Perl AGI |
1:01PM |
3 |
Sending call to analog then to Vmail after timeout? |
12:59PM |
0 |
Asterisk users manual |
12:51PM |
3 |
Dialtone for Software phone? |
12:04PM |
1 |
rejected calls from IAX provider |
11:52AM |
0 |
Polycom phone stops working |
11:37AM |
4 |
DHCP, the TFTP Server setting and the Cisco 79xx phones |
11:20AM |
2 |
Wildcard remote looping |
11:02AM |
0 |
VoIP Equipment |
10:19AM |
0 |
ztdummy necessary? |
8:54AM |
1 |
Asterisk consuming 100% CPU - CDR loop |
8:05AM |
2 |
Asterisk with T1 |
7:28AM |
0 |
PCI PERR's w/Digium cards |
6:57AM |
0 |
Optional URL param |
6:43AM |
1 |
music on hold without sound card |
6:03AM |
0 |
H.323 link to provider VoIP with Username and Pass |
5:57AM |
1 |
Asterisk / 183 message |
5:46AM |
0 |
[Fwd: Callmanager 4.1 and asterisk] |
4:45AM |
1 |
Chan IAX2 errors while calling Toll Free numbers using IAXTEL |
4:29AM |
0 |
SV: One way audio |
4:22AM |
0 |
Packet flow in relaying from SER to Asterisk |
4:02AM |
0 |
pickup group |
3:55AM |
0 |
My firefly is changing the IP address !!??? |
3:50AM |
0 |
Does anybody use a video phone ? |
3:32AM |
0 |
Asterisk recognize GSM CLI |
3:16AM |
0 |
dialplan "not ${VARIABLE} |
2:53AM |
3 |
Zaptel ISDN BRI settings for The Netherlands KPN |
2:40AM |
2 |
Mysql and Voicemail |
1:15AM |
0 |
socksify |
12:21AM |
1 |
Callmanager 4.1 and asterisk |
12:12AM |
0 |
Re: Help on Register message with Authentication |
|
Monday December 27 2004 |
Time | Replies | Subject |
11:40PM |
0 |
Call Placing timeouts |
9:44PM |
2 |
Cant get Asterisk server talk with IAX |
8:37PM |
2 |
PassThrough mode |
7:52PM |
0 |
call parcking failure |
6:16PM |
1 |
Selecting Extensions |
5:14PM |
2 |
parking.conf |
4:31PM |
0 |
Re: Asterisk dying... |
3:43PM |
6 |
realtime voicemail |
2:59PM |
0 |
help regarding ASTCC |
2:45PM |
0 |
Asteriks Compile error |
2:33PM |
0 |
IAX -> SIP Call Help; IAX with G729 |
1:17PM |
2 |
API manager - Redirect with ExtraChannel |
1:06PM |
1 |
codec preferences |
12:40PM |
0 |
asterisk dies no calls in or out |
11:55AM |
0 |
no voice with all sip phones until hold/unhold |
11:42AM |
2 |
does a TDM04B (all FXOs) need a power connector? |
11:37AM |
2 |
MYSQL_FRIENDS |
11:20AM |
1 |
transfer: hookflash vs # |
11:11AM |
3 |
how to debug frame slips? |
11:01AM |
2 |
TDM400 problem |
10:38AM |
1 |
Command-line dialer/recorder for asterisk? |
10:03AM |
0 |
Is there a way to avoid bandwidth consumption on sip calls? |
9:11AM |
0 |
[chan_capi] can't get it compiled |
9:04AM |
0 |
Jeff Pulver quoted talking about Asterisk... |
8:21AM |
3 |
Diax echo problem |
8:12AM |
3 |
mail function |
8:08AM |
1 |
Generic Network profile for VOIP |
7:49AM |
0 |
zaptel error : Relocation overflow of type 10 |
7:15AM |
2 |
SIP client cannot connect to Asterisk |
7:11AM |
3 |
restricting SIP access to asterisk |
6:16AM |
0 |
Fw: Hookflash timing with TDM400P |
5:34AM |
1 |
Make error installing bristuff-0.2.0-rc2b |
4:25AM |
0 |
Problem with AgentCallbackLogin |
2:40AM |
1 |
incoming & outgoing call |
1:06AM |
0 |
ASTCC - setup help please |
|
Sunday December 26 2004 |
Time | Replies | Subject |
10:04PM |
0 |
distinctiv ring (Aert-Info) |
5:51PM |
1 |
is deadlocking with the Manager API still a problem? |
3:13PM |
1 |
OT - Originating Network identity |
2:16PM |
16 |
Incoming Calls |
1:14PM |
1 |
Asterisk realtime load error |
12:57PM |
1 |
Cannot transfer after queue agent picks up c all |
9:13AM |
1 |
Cannot transfer after queue agent picks up call |
9:09AM |
2 |
Asterisk behind IX66 |
6:28AM |
0 |
Voice modem + Asterisk |
5:28AM |
0 |
HUP signal? |
4:21AM |
7 |
IAX Registration Refused |
2:02AM |
0 |
SV: Call Completion Snom |
|
Saturday December 25 2004 |
Time | Replies | Subject |
11:15PM |
0 |
Asterisk + Voice Modem |
9:24PM |
1 |
Alert-Info |
9:12PM |
1 |
VM_CALLERID (how to get name+number) |
4:46PM |
0 |
Where to get a Polycom IP500 in the UK? |
4:37PM |
0 |
safe_asterisk script contains error? |
3:59PM |
0 |
patch to build h323 without recompiling pwlib, ... |
2:43PM |
0 |
Bri-stuff + TDM 2-Port FXS & 2 Port FXO Card |
8:06AM |
5 |
How to connect two Asterisks as secure as possible without too much additional bandwidth ? |
5:39AM |
3 |
About CallBack function |
4:03AM |
0 |
Automatic calls |
3:56AM |
0 |
Bandwidth, computer power |
3:47AM |
0 |
TE410P No Interrupts |
2:36AM |
2 |
Can Asterisk handle calls that get picked up by answering machines? |
2:36AM |
2 |
Dynamic extensions without using DynExtenDB? |
2:29AM |
1 |
How to use firefly with Asterisk? |
2:07AM |
1 |
Asterisk and Lucent APX8100 Universal Gateway |
1:48AM |
2 |
Transcript of sound files? |
|
Friday December 24 2004 |
Time | Replies | Subject |
10:17PM |
2 |
ALERT_INFO issue CVS-HEAD-12/24/04 |
10:01PM |
0 |
Calling Party ringing indicator |
8:17PM |
1 |
What do I need to build up DID services? |
6:46PM |
1 |
Firefly Transfer call ? |
5:12PM |
0 |
VoiceConduits? |
4:49PM |
1 |
FC3, TDM11B (DEVPCI) and asterisk |
3:06PM |
7 |
Tie web application to VOIP |
2:36PM |
1 |
Uniden UIP200 firmware v4.63 |
10:30AM |
3 |
Registration failure with debug |
8:14AM |
0 |
Cisco, Codecs, Sip Phones et al |
7:14AM |
1 |
Switch polarity to disconnect a FXS channel |
6:44AM |
3 |
Preventing Asterisk from sending 'h' across to SIP Provider |
6:09AM |
3 |
Help on Register message with Proxy-Authorization |
2:27AM |
0 |
Asterisk Xmas ;-) |
1:57AM |
0 |
SIP Multicast Support desperately needed :: Mission critical bug in Asterisk |
1:39AM |
0 |
Help:could asterisk work with other sip proxy? |
12:32AM |
0 |
help:could asterisk be used such as sip proxy? |
|
Thursday December 23 2004 |
Time | Replies | Subject |
11:37PM |
3 |
Record() problem |
10:15PM |
0 |
Asked to transmit frame type 2, while native formats is 4??? |
10:00PM |
0 |
Australian STD "pips" & Telstra pstn |
9:09PM |
1 |
Service contract for * in NYC area |
8:38PM |
0 |
Asterisk Certification |
8:35PM |
2 |
DISA restart from begining |
8:03PM |
0 |
txgain / rxgain no effect |
7:17PM |
2 |
Special Problem in Australia ?? |
7:05PM |
1 |
where I can find some learning book about asterisk? |
6:32PM |
3 |
error starting asterisk |
6:16PM |
0 |
Turning "*" Hangup off in queues |
4:01PM |
8 |
asterisk at large |
3:58PM |
1 |
Can't Make Outgoing Call |
3:45PM |
3 |
rtp channels not through asterisk |
3:40PM |
0 |
Asterisk queue_log |
3:32PM |
1 |
Voicemail email notification |
2:57PM |
0 |
"*" behaviour in agentcallbacklogin |
2:43PM |
1 |
Recommended IAX softphone. |
2:25PM |
0 |
Cisco 7960 Support Products |
2:21PM |
1 |
turn on/off auto/attendant by dialing an extension |
2:17PM |
2 |
Asterisk 1.0.3 no RedHat zaptel script? |
1:27PM |
2 |
Asterisk in parallel with PSTN |
1:06PM |
1 |
IAX cause codes |
12:29PM |
1 |
RE: IAX2 calls failing one way |
12:26PM |
0 |
changethread: can't change device with no technology! |
12:10PM |
1 |
Polycom Buddies |
12:07PM |
1 |
Premature DRQ |
11:53AM |
1 |
T100P frame slips |
11:36AM |
0 |
SV: RedAlarm (t100p - Adtran Total Access 750) |
11:28AM |
2 |
Re: Asterisk and Capi |
11:08AM |
0 |
Asterisk cannot read DTMF based CallerID from PSTN |
10:55AM |
0 |
IAX2 calls failing one way. |
10:38AM |
1 |
PRI unable to request channel |
10:27AM |
0 |
Call Completion Snom |
10:24AM |
4 |
RedAlarm (t100p - Adtran Total Access 750) |
10:01AM |
0 |
New astGUIclient version released 1.0.6 |
9:58AM |
5 |
TDM400 success? |
9:28AM |
0 |
switch statement. |
9:12AM |
1 |
Queue - roundrobin member order |
8:57AM |
1 |
Multiple Registration |
8:49AM |
2 |
Incoming calls from Sipgate go through the wrong peer |
8:37AM |
5 |
Fw: [digium.com #12961] T100P as bandwidth |
8:29AM |
0 |
Need help with cisco 7960 call fwd and dial plan |
8:13AM |
1 |
Linksys PAP2-NA Config |
8:09AM |
1 |
Polycom 600 problem |
7:41AM |
0 |
Registration Failure Directly related to realtime |
7:25AM |
0 |
lockup problem with inbound iax calls |
7:15AM |
1 |
Softphone x G729 x IAX |
7:01AM |
2 |
Realtime sipbuddies table structure why????? |
6:27AM |
0 |
Integrating Asterisk and Siemens Hicom 300E with TDM04B |
4:37AM |
2 |
One-way audio in incoming calls with Asterisk + OpenGK + Innovaphone IP3000 |
3:43AM |
1 |
Problems with incoming IAX calls... |
3:21AM |
0 |
Passing SIP headers to AGI applications |
3:02AM |
0 |
Reservation call on busy |
3:00AM |
1 |
Re: Asterisk-Users Digest, Vol 5, Issue 329 |
2:58AM |
0 |
Re: Asterisk-Users Digest, Vol 5, Issue 333 |
2:11AM |
1 |
messenger on the mobile phone |
2:09AM |
1 |
How to apply patches |
2:01AM |
1 |
Qestion about TDM over enthernet |
1:00AM |
0 |
Connect attempt rejected error message |
12:49AM |
2 |
Asterisk with Dialogic VFX/40ESC plus |
12:43AM |
1 |
ignoring signalling |
12:39AM |
0 |
Disconnection Problem |
|
Wednesday December 22 2004 |
Time | Replies | Subject |
11:12PM |
2 |
Out of G.729 Decoder Licenses! |
7:10PM |
2 |
polycom and cdp |
6:43PM |
2 |
Still unable to use g729 codec... please HELP |
6:20PM |
4 |
New verision of AMP - 1.10.004 |
5:38PM |
0 |
Iax2 Registration failed |
5:38PM |
1 |
Zaptel/Zapata config from T410p to Brooktrout T1 |
4:49PM |
0 |
FreeBSD, Generic Modem and DIGIUM boards |
4:39PM |
1 |
Problem ringing simultaneous channels |
4:17PM |
0 |
IAX Peering for PSTN termination Sydney <=> Moscow |
4:13PM |
1 |
SIP URI Dialplan? |
3:49PM |
2 |
Asterisk Interface to propriotary system and GPL |
3:34PM |
1 |
register_verify defined in 2 files? |
3:22PM |
0 |
Zap Fxs port always answers? |
3:22PM |
5 |
TDM400P install on Debian 2.6.10 |
3:10PM |
1 |
MGCP Transaction identifiers |
3:02PM |
3 |
Can somebody email me the Sipura SPA-2000 and SPA-3000 documentation? |
2:59PM |
0 |
Phone Registration Failure Test |
2:57PM |
1 |
Asterisk billing solution |
2:22PM |
0 |
Softphone with subscribe/notify support |
2:15PM |
2 |
711 and 729 with IAX? (IAX Newbie) |
1:08PM |
2 |
Can't Receive/Send Calls |
12:50PM |
0 |
Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party don't hear) |
12:45PM |
0 |
chan_sip errors in CVS stable |
12:43PM |
1 |
Asterisk->AS5350 misplaced RTP to 127.0.0.1 (AS5350 party don't hear) |
11:50AM |
2 |
txfax failure |
11:49AM |
0 |
TE410P to a Rhino CB-24 channel bank |
11:31AM |
0 |
RE Zaphfc/BRI Configuration help |
10:56AM |
0 |
Macro(dundi-dundi-test, ${ENTEN}) to return +101 on lookup failure ? |
10:41AM |
0 |
rtc3389 |
10:37AM |
0 |
Early media problems... |
10:21AM |
1 |
Status of asterisk.xvoip.com? |
10:20AM |
1 |
PRI error (HDLC Bad FCS) |
10:14AM |
5 |
Another Asterisk Certification |
10:14AM |
0 |
FWD + xtraphone and DTMF |
9:47AM |
1 |
Grandstream BT100 -> Asterisk -> Voipjet ..... No ring ring when making a call |
9:35AM |
3 |
gumstix |
9:25AM |
2 |
IAXy playing dead again |
9:22AM |
0 |
What is the procedure to test for Caller-ID |
8:51AM |
0 |
PassThru mode |
8:05AM |
0 |
What is the best commercial soft phone for Asterisk? |
8:00AM |
6 |
IAX hardphone |
7:40AM |
1 |
Phonecell + wildcard FXO (DTMF problems) |
7:37AM |
2 |
Call dies in 180 seconds exactly |
7:35AM |
3 |
call from DID, not hearing RINGTONEs |
7:32AM |
1 |
Cisco 7960 Hold |
7:00AM |
1 |
Link an Asterisk Box with a PBX (E1 connection) |
6:57AM |
1 |
ZapBarge |
6:54AM |
0 |
Wither ChanSpy ? |
6:11AM |
2 |
Why use 'Answer'? |
4:53AM |
2 |
Matching Caller ID against a database of knowncallers |
4:27AM |
3 |
E1 card for Asterisk |
4:27AM |
1 |
Aterisk@Home |
4:12AM |
0 |
Ticket: 12775 Multiple IAX client behind a NAT |
3:07AM |
0 |
RE: hint extension and Snom phones - CVS or |
2:55AM |
1 |
Daily NANPA updates |
2:45AM |
0 |
Dialogic MSI cards to FXO port on TDM400P |
12:49AM |
2 |
MWI not working on Polycom Phones |
|
Tuesday December 21 2004 |
Time | Replies | Subject |
11:12PM |
1 |
Matching Caller ID against a database of known callers |
10:31PM |
0 |
opaque= field |
10:13PM |
0 |
IAX2 insists on not using port 4569?? |
9:10PM |
1 |
Dialplan help - Can dial any user but not thePSTN |
9:06PM |
0 |
No Ringback tone on Stable 1.0.2 |
8:36PM |
7 |
Cannot transfer with Cisco or Snom |
7:34PM |
2 |
X100P dead? |
7:19PM |
0 |
SIP dtmf=rfc2833 not working |
6:49PM |
1 |
Lets try this again then! Q: SIP error from dialplan I suspect! |
6:44PM |
1 |
Hmm something strange. |
5:54PM |
0 |
Voice prompts text & Chinese |
5:20PM |
0 |
Status of Queue? |
5:10PM |
0 |
Re: problem with calls on hold |
4:52PM |
0 |
fxstest cant ring phone, but asterisk can ! |
4:42PM |
2 |
gateway.lu |
4:30PM |
2 |
CallerID returned with error on channel 'Zap/4-1' |
3:37PM |
1 |
zaptel ppp HDLC Receiver Overrun messages |
2:52PM |
2 |
IAXTEL Configuration |
2:23PM |
3 |
Problems installing Zaptel |
1:37PM |
3 |
Budgetone is not registering |
1:33PM |
0 |
Hung SIP channels in Asterisk |
1:28PM |
0 |
Help bridging 2 outbound IAX2 calls ! |
1:03PM |
4 |
hint extension and Snom phones - CVS or stable? |
12:53PM |
10 |
Codec Selection |
12:39PM |
2 |
Poor Grammar or is this a bug |
11:54AM |
1 |
Small PBX to VoIP transition questions |
11:54AM |
0 |
Spandsp 0.0.2pre6 configure fails sanity check. |
11:51AM |
1 |
GUI Tool |
11:47AM |
2 |
sip seeding vs registration |
11:46AM |
1 |
G729, x-pro, and codec ordering |
11:07AM |
1 |
Linking 3 Asterisk box, server in the middle type of thing? (IAX?) |
11:00AM |
1 |
Call routing based on remote ip address. |
10:12AM |
1 |
bri stuff and unknown signalling type |
10:07AM |
0 |
Problems with Budgestream and g729 codec |
9:49AM |
1 |
asterisk-oh323: New versions available |
9:37AM |
3 |
Bug, Feature, or Limitation? |
9:23AM |
2 |
Minimal modules.conf (e.g. with autoload=no)? |
9:16AM |
2 |
SOHO PBX using asterisk |
9:12AM |
3 |
What is sip-friends.sql?????? |
8:59AM |
0 |
CP7902g SIP IOS |
8:54AM |
2 |
TE405P E1 coax cables with balun |
8:30AM |
1 |
Paris Meeting on Dec 20, 2004 - réunion à Paris le 20 décembre 2004 |
8:24AM |
1 |
Incoming call on IP |
8:09AM |
4 |
asterisk server to asterisk server |
7:44AM |
1 |
h.323 Type=User |
6:52AM |
5 |
AMP - Fax Detections |
6:40AM |
2 |
upgraded source now ata's ring but stop silence on inbound calls |
5:59AM |
0 |
Intel Cards ??? |
5:45AM |
2 |
Call back when no longer busy |
5:40AM |
2 |
Queues without members |
4:20AM |
0 |
Showing the name of the country on a Cisco 7960/7912? |
4:20AM |
1 |
HELP: agi-test.agi does not return any DTMF! |
3:49AM |
0 |
Suggestions for Asterisk + BRI + Data |
3:34AM |
0 |
(no subject) |
3:02AM |
0 |
Incomming call to asterisk server error |
2:49AM |
2 |
Zhone Channel Bank |
2:34AM |
1 |
two avm usb isdn fritz v2.0 cards |
2:18AM |
0 |
howto disable call waiting ? |
1:53AM |
2 |
Channel limits ? |
12:13AM |
6 |
Caller ID - TE405P - Telstra Onramp 10 - Australia |
|
Monday December 20 2004 |
Time | Replies | Subject |
11:42PM |
7 |
NMI issues... |
11:26PM |
2 |
Grouping SIP channels (Sipura 3000) |
11:10PM |
0 |
newbie questions / documentation feedback? |
10:55PM |
0 |
SIP ringback problem with Polycom phones and CVS HEAD |
10:01PM |
2 |
Can asterisk be run as non root anymore? |
9:49PM |
3 |
Mysql-Realtime |
8:57PM |
0 |
Asterisk mechandise reselers with good reputation |
7:45PM |
0 |
Asterisk with RxFAX/TxFax start problem |
7:35PM |
1 |
Example config for SPA-1001 |
6:35PM |
0 |
Q: How do I join an in-progress Zap channel call? |
6:16PM |
0 |
x-ten pro and echo cancellation... |
5:35PM |
0 |
Patching the source? |
5:28PM |
0 |
On Australian News Sites : Open source software set to influence VoIP |
4:10PM |
3 |
codec issues |
4:02PM |
1 |
A few simple (I hope) questions from a first-timer |
3:42PM |
0 |
Asterisk Startup Scripts (My Bad) |
3:37PM |
1 |
ATA callwaiting |
3:18PM |
0 |
Incoming voicemail and dialtone |
3:12PM |
1 |
[Asterisk-Dev] RE: [Asterisk-biz] Asterisktraining andcertification :: AstriconTraining |
3:08PM |
2 |
Toshiba DK-40 and Asterisk...possible? |
2:59PM |
1 |
RFC3389 support incomplete. |
2:42PM |
1 |
RxFAX compile problem |
12:44PM |
1 |
Why does * only work with an ancient mpg123? |
11:54AM |
2 |
ATA Adaptor |
11:45AM |
1 |
What does "t" mean in a CDR entry? |
11:16AM |
0 |
Calling SIP Address From Behind NAT |
10:45AM |
1 |
Problem using SPA-2000 behind NAT |
10:37AM |
0 |
weird problem with IAXphone |
10:09AM |
0 |
What is the difference between monitoring and recording??? |
10:06AM |
1 |
How to allow users to dial certain numbers |
10:05AM |
4 |
Asterisk Fails To Start on Reboot Mysql |
9:49AM |
2 |
asterisk: webmin or X admin. |
9:33AM |
0 |
Unusuall Asterisk Usage Idea... |
9:04AM |
1 |
Asterisk A-Z provider from sratch |
8:37AM |
7 |
One SIP peer use 2 diff codecs? |
8:35AM |
0 |
Testers needed for voicemail ODBC storage patch |
8:30AM |
19 |
Updating Asterisk |
8:10AM |
2 |
Is there hardware to remote control |
7:40AM |
1 |
AW: SMS - how to send one |
7:18AM |
3 |
grandstream MWI? |
6:31AM |
0 |
Is there hardware to remote control available? |
6:24AM |
0 |
Extensions SIP problems. |
6:15AM |
1 |
E1 signalling pridialplan |
6:12AM |
3 |
[OT] resetting SPA 2000? |
6:12AM |
2 |
Realtime voicemail failure |
5:59AM |
1 |
how to prevent res_odbc from loading |
5:43AM |
1 |
Making a queue menu not exit the queue |
4:06AM |
0 |
autovol 0.9 |
3:32AM |
3 |
Problems with loading TE110 module |
3:13AM |
1 |
Help me ($$$) with install h323 |
2:20AM |
7 |
'I'nvalid extension handling problems, even with workaround |
2:18AM |
0 |
Skinny bug / missing feature, who is the maintainer? |
1:25AM |
1 |
Fw: pbx.c:1279 pbx_extension_helper: No application 'SetVal' for extension (c819, 1, 1) |
12:46AM |
1 |
AW: Zaphfc/BRI Configuration help |
12:38AM |
3 |
PA1688 Chipset IP Phones & ATA's |
12:29AM |
5 |
Zaphfc/BRI Configuration help |
|
Sunday December 19 2004 |
Time | Replies | Subject |
11:28PM |
1 |
Quick questions ( maybe a little confidence building too ) |
10:46PM |
2 |
MFC/R2 errors |
10:37PM |
2 |
OH323 channel compile error |
10:13PM |
2 |
Can DPNSS be developed in S/w like libpri ? |
8:58PM |
0 |
iax2 event status using asterisk 1.0.3 & iaxfriends |
8:43PM |
1 |
Dialplan help - Can dial any user but not the PSTN |
7:50PM |
0 |
one way audio on sip channels |
7:01PM |
1 |
sip phones in different private networks have one way audio |
6:30PM |
3 |
[Asterisk-Dev] RE: [Asterisk-biz] Asterisk training andcertification :: AstriconTraining |
6:13PM |
0 |
RE: [Asterisk-biz] Asterisk training and certification :: AstriconTraining |
6:09PM |
1 |
OT- Callwave neat app |
3:24PM |
0 |
Call Queuing |
2:33PM |
0 |
Asterisk SIP transfer(refer) |
1:49PM |
0 |
ztcfg seg faulting |
1:40PM |
1 |
TE110P - problem with zone from zaptel.conf |
12:41PM |
4 |
SMS - how to send one |
12:10PM |
3 |
Looking for new hardware |
10:58AM |
2 |
dialplan selection |
9:30AM |
2 |
QuickNet Internet PhoneJack problem |
9:09AM |
3 |
VoicePulse OpenAccess |
8:59AM |
2 |
VoicemailMain can't read from phone keyboard! |
8:41AM |
3 |
TDM120 card? |
7:59AM |
1 |
BRI Error with zaphfc |
7:13AM |
1 |
Connecting Siemens HiCom PBX with Asterisk through E1 |
6:46AM |
1 |
Make asterisk launch script after completing call. |
5:52AM |
2 |
Phone choices....opinion request Polycom vs Cisco |
4:02AM |
2 |
Per extension/user CDR? |
12:56AM |
2 |
TDMoE or IAX? |
|
Saturday December 18 2004 |
Time | Replies | Subject |
11:31PM |
1 |
Getting the "real" extension into CDR |
11:31PM |
0 |
10-10 dial around |
10:30PM |
2 |
audio levels via sip |
9:33PM |
4 |
Free World Dialup and Asterisk |
9:18PM |
0 |
Configure Asterisk with Radius |
7:15PM |
0 |
Zap Channel Group Question |
7:09PM |
0 |
New FC1 packages... |
6:33PM |
0 |
One-way audio with SIP client only on certaincalls |
6:06PM |
1 |
voicemailmain hotkey |
5:26PM |
5 |
Q about IAX (and IAXy) |
5:21PM |
1 |
call waiting/ 3 way calling |
4:51PM |
1 |
One-way audio with SIP client only on certain calls |
3:09PM |
0 |
web-based sip / iax client |
2:05PM |
2 |
Problem with 302 "Moved Temporarily" Do not disturb |
1:52PM |
3 |
3rd party call control / CSTA , JTAPI or TAPI interfaces |
1:41PM |
0 |
Using Digium cards for data+voice & Asterisk |
1:21PM |
0 |
TEST - Pls Ignore (Unable to see my own posts) |
11:53AM |
2 |
Music/Busy Signal Not Heard |
10:49AM |
0 |
PRI got event: HDLC Bad FCS |
10:31AM |
0 |
Monitor entry not working... please help |
10:00AM |
2 |
External Address Books |
9:21AM |
0 |
Meetme with video??? + $US 2,000 bounty |
9:01AM |
0 |
SIP and IAX Clients for pre OS-X Macs ? |
9:00AM |
2 |
It's possible to do a codecs translation during a call in Asterisk? |
8:49AM |
0 |
what the heck? codec_gsm.c:135 gsmtolin_framein: Huh? |
7:24AM |
1 |
How to increase the performance? |
7:22AM |
0 |
Sound problems with iaxcomm and Linux Fedora |
7:00AM |
1 |
Problem with a TDM400 |
6:52AM |
1 |
Setting up asterisk for one user in private ip NAT. |
3:17AM |
3 |
Open Ports |
2:43AM |
1 |
SV: call billing |
12:49AM |
2 |
Re: asterisk - basic hardware and packages |
12:42AM |
1 |
X100P card in Australia |
12:36AM |
0 |
Call Completion Asterisk and Snom |
|
Friday December 17 2004 |
Time | Replies | Subject |
11:03PM |
2 |
Grandstream Voicemail |
9:39PM |
1 |
Mysql-Configuration |
8:31PM |
1 |
asterisk - basic hardware |
8:24PM |
0 |
asterisk packages |
7:14PM |
1 |
h323 channel compile error |
7:12PM |
0 |
Demo voice hickups. |
6:35PM |
8 |
NPA NXX data |
5:53PM |
2 |
Optimizing Sipura/Asterisk for DTMF? |
5:23PM |
2 |
OT: "Integrated Access T1" voice problems -is this possible? |
4:44PM |
2 |
T-1 vs channelised T-1? |
4:10PM |
1 |
Second TDM400 card |
4:08PM |
2 |
Call Queue Uniden UIP 200 not working |
4:04PM |
2 |
Total newbie here looking to do a VoIPconfer ence call? |
3:56PM |
0 |
Total newbie here looking to do a VoIPconference call? |
3:35PM |
1 |
hdlc + te410p + kernel 2.6.9 - anyone done this? |
3:26PM |
2 |
OT: "Integrated Access T1" voice problems - is this possible? |
3:12PM |
1 |
Total newbie here looking to do a VoIP confe rence call? |
2:31PM |
1 |
Least Cost Routing - Are you doing it? What are you using? |
2:29PM |
0 |
Total newbie here looking to do a VoIP conferencecall? |
2:00PM |
5 |
Total newbie here looking to do a VoIP conference call? |
1:24PM |
14 |
Call on hold disconnects... |
1:21PM |
2 |
voicemail without prompt |
1:19PM |
1 |
modified prepaid application |
12:37PM |
1 |
Snom 190, led and shared lines with asterisk |
12:30PM |
1 |
ASTCC in production |
12:15PM |
2 |
Asterisk receives busy..but its not... |
12:01PM |
5 |
Asterisk Crackly Bad quality |
11:42AM |
0 |
Dropping out of Queue to voicemail |
11:34AM |
1 |
application meetme |
11:23AM |
0 |
Red Alarm / Alarm Cleared Zaptel Issue (bug? ) |
10:58AM |
6 |
Realtime and PostgreSQL |
10:45AM |
1 |
chan_capi - avm card does not work |
10:28AM |
1 |
Red Alarm / Alarm Cleared Zaptel Issue (bug?) |
10:12AM |
0 |
MusicOnHold. not getting it.-GOT IT!! |
9:53AM |
3 |
Old posts and the ability to search... |
9:10AM |
0 |
Latest head giving app_queue.c:340 error |
9:01AM |
1 |
SNIMTA_SPAM Using the Directory Feature to play a menu |
9:00AM |
0 |
Display on OptiPoint400std SIP |
8:56AM |
0 |
asterisk clients (need helpdesk solution) |
8:54AM |
2 |
New Asterisk Prompts |
8:09AM |
0 |
Newbie setup question (Voicepulse, FWD & IAXTEL) |
7:59AM |
0 |
German Howto? |
7:12AM |
1 |
Masive Fax Sendig with spandsp |
7:05AM |
1 |
Forcing E.164ID with chan_h323 & or chan_oh323 |
7:03AM |
2 |
Definity PBX with a T100P & TN767E |
6:43AM |
6 |
OT: DSL without voice |
6:42AM |
1 |
Asterisk and HylaFax |
6:25AM |
2 |
erroneous errors - registration fails for grandstream phones |
6:09AM |
2 |
Cisco 7905g TFTP Configuration |
5:32AM |
0 |
can't intstall the webmin module |
5:32AM |
1 |
Troubleshooting Asterisk |
5:15AM |
5 |
Disabling " !" command |
3:47AM |
0 |
s and i in context not invoked |
3:40AM |
0 |
Simulate back impulse |
3:23AM |
0 |
[Off Topic] humour, XMAS, ground loop - good business strategy |
3:14AM |
0 |
AS5xx0: SS7 and SIP? |
3:00AM |
1 |
ADSI programming/TDM400P issues |
2:55AM |
0 |
instructions to get .bin firmware for 7920 |
2:49AM |
0 |
chan_sccp and 7920 |
2:37AM |
3 |
Meetme with video??? |
2:36AM |
3 |
Paris Meeting Date/Time/Location |
2:35AM |
0 |
SayUnixTime |
2:19AM |
1 |
MD110 and analog trunks |
|
Thursday December 16 2004 |
Time | Replies | Subject |
11:15PM |
1 |
Shorten the recognition time of rings on Wildcard X100P |
11:07PM |
0 |
Dialing asterisk from open phone |
11:06PM |
2 |
How to generate a SIP NOTIFY for Cisco 7960 remote reboot? |
10:08PM |
0 |
Call confirmation on NON Zap channels |
9:53PM |
1 |
Public Thanks |
9:23PM |
5 |
Hardware based DSP |
8:51PM |
4 |
191st simultaneous call fails |
8:30PM |
0 |
Call Waiting FXS and * |
6:28PM |
0 |
are there any tips/tricks to get the uip200 to register? |
5:55PM |
1 |
problem with freebsd 4.9 port |
5:08PM |
1 |
ilbc and asterisk 1.0.3 - strange noises. |
4:53PM |
1 |
Rapid DTMF entry failure |
4:38PM |
0 |
Can I read more than 7 numbers from capi ? |
4:13PM |
1 |
Asterisk Cisco CallManager Integration |
4:08PM |
1 |
Low-latency kernel? |
3:57PM |
0 |
zap, agents, ackcall |
3:08PM |
1 |
Which Primary ISDN card to use in Europe ? |
2:57PM |
2 |
MusicOnHold. not getting it. |
2:47PM |
3 |
Connecting Asterisk to GSM |
2:35PM |
4 |
Polycom SIP Phones |
2:26PM |
1 |
Dynamically Choose Codec for Bandwidth Management |
2:07PM |
1 |
Steps to configure D/41EPCI card |
1:51PM |
3 |
Get asterisk out of the RTP stream? |
1:37PM |
0 |
STABE, CVS and in between? Confused |
1:37PM |
0 |
Re: Re: Cant set H323 up |
1:37PM |
0 |
SIP channel groups - is it possible? |
1:23PM |
0 |
Has anyone connected to 7960 with console cablefor setup? |
1:20PM |
0 |
Good place to get DID's? |
1:06PM |
1 |
native MOH with Asterisk 1.0.3 |
12:04PM |
1 |
working with big blocks of msn's |
11:35AM |
1 |
Polycom FX Video Unit - asterisk-oh323 |
11:34AM |
3 |
Has anyone connected to 7960 with console cable for setup? |
11:19AM |
2 |
sox-12.17.6 |
10:59AM |
1 |
BRI Card not recognized |
10:45AM |
8 |
Calculating required bandwidth |
10:05AM |
0 |
FW: Cisco 7960 (SIP) hold problems |
9:26AM |
1 |
OT: iax.cc hosts - want to do some traceroutes before buying |
8:40AM |
1 |
Multiple IAX client behind a NAT |
8:01AM |
3 |
Cisco 7960 (SIP) hold problems |
7:33AM |
2 |
How to tell "Who's Online"? |
7:24AM |
0 |
Codec Negotiation Problem |
7:18AM |
2 |
Queueueueuueue position |
6:52AM |
0 |
Compile issues: * 1.02 + FreeBSD 5.3 |
6:06AM |
1 |
send # with transfer enabled |
6:02AM |
0 |
Reporting Errors & Mysql |
5:30AM |
3 |
Detect line is busy with Zap? |
5:16AM |
0 |
SPA-3000 - Stop Message Waiting Indication |
5:10AM |
0 |
codec preference? |
5:08AM |
0 |
Making "sip show channels" show sane results with sipfriends from mysql? |
5:05AM |
0 |
Voicemail Pager Subject? |
4:47AM |
0 |
Asterisk <--> Nuera Orca |
4:44AM |
0 |
Automated callback with .call file |
4:21AM |
0 |
Logging codec in cdr? |
4:18AM |
0 |
Channel Groups with SIP |
4:16AM |
1 |
Monitoring an active call |
3:55AM |
8 |
g711 ulaw vs alaw |
3:48AM |
3 |
asterisk on FC3 |
2:48AM |
12 |
My Boss wants background music!!!! |
2:47AM |
0 |
kewlstart - explanation of this method, please ? |
2:08AM |
0 |
Asterisk support mailing list in Italian |
1:59AM |
1 |
Calls arent handled by asterisk - destruction of call |
1:44AM |
0 |
E1 and analog cards FXS in one box. |
|
Wednesday December 15 2004 |
Time | Replies | Subject |
11:59PM |
1 |
Asterisk, Capi, Controller |
11:10PM |
0 |
Re: Asterisk-Users Digest, Vol 5, Issue 221 |
10:40PM |
4 |
VoIP bad voice quality |
10:03PM |
7 |
VoIP Termination |
9:40PM |
5 |
QOS Device? |
8:18PM |
1 |
mpg123 exploit |
7:39PM |
5 |
VOIP Phone Suggestions |
7:19PM |
1 |
Linksys PAP2-NA Screenshot |
7:13PM |
0 |
PRI Errors again... sigh. |
7:11PM |
1 |
Re: Asterisk-Users Digest, Vol 5, Issue 219 |
6:49PM |
1 |
Outlook integration? |
6:25PM |
2 |
Bugtracker Karma Hall Of Fame |
6:09PM |
2 |
SIP Server question / recommendations |
5:22PM |
2 |
Voipjet problems |
5:14PM |
1 |
Help with transferring a second call from a snom 190 |
4:43PM |
0 |
Fulfillment, Gold/Platinum Programs |
4:32PM |
0 |
Can Directory app read extension numbers? |
4:26PM |
0 |
AstLinux - New Version - w/ 1.0.3 what about capi!!!! |
4:14PM |
0 |
agi send text option |
3:59PM |
3 |
Newbie setup (Hardware questions) |
3:56PM |
1 |
Advanced Ring All Hunt Group |
3:49PM |
1 |
Using ChanIsAvail with SIP |
3:37PM |
1 |
asterisk + freeradius |
3:37PM |
3 |
wcfxs causing constant CPU spikes |
3:34PM |
2 |
TDM400p FXO module always offhook |
1:44PM |
0 |
Digium TDM11B |
1:43PM |
0 |
E&M Wink Question |
1:30PM |
0 |
Start of conversation lost |
12:44PM |
2 |
chan_sccp compile problem w/ CVS head? |
11:41AM |
1 |
Sipura 2000 intermitent failure to register |
11:09AM |
0 |
Skinny not working? |
10:54AM |
2 |
Cisco 7960 SIP + 7914 |
10:52AM |
2 |
No Caller ID Name PRI NI2. |
10:49AM |
1 |
SNOM 190 Call Completion |
10:06AM |
0 |
Very strange behaviour, has anybody noticed? |
9:22AM |
0 |
AstLinux - New Version - w/ 1.0.3 |
9:13AM |
3 |
PRI incoming call???? |
8:27AM |
2 |
IP Conference Units? |
8:26AM |
5 |
How "expensive" are the different codecs? (Regarding CPU time) |
8:15AM |
4 |
Codecs and RealTime |
7:21AM |
3 |
codec order in SIP doesn't work |
6:23AM |
0 |
ASTCC and CDR info |
6:14AM |
0 |
first 2-3 secs choopy sound |
5:38AM |
0 |
SIP INFO vs RFC2833? |
4:32AM |
1 |
IAX2 Notify exchanges on port 1024 and 1040 - Normal ? |
4:07AM |
0 |
RE: Asterisk-Users Digest, Vol 5, Issue 187 |
3:31AM |
1 |
Easy question? Get started with the Demo |
3:12AM |
1 |
Re: 12.50$ per port ??? |
2:15AM |
0 |
Digium hardware vendor in Israel? |
1:42AM |
1 |
IAX2 tolerance on packet losses |
1:31AM |
0 |
Asterisk to sip client behindFirewall/NAT-cancall but cannot receive calls ? |
|
Tuesday December 14 2004 |
Time | Replies | Subject |
11:18PM |
1 |
SIP and Windows Messenger |
10:45PM |
0 |
Bug 3020 needs supporters :-) |
10:24PM |
0 |
Codec "Uknown" with IAX connection |
10:24PM |
1 |
Asterisk Realtime IAX - Adding fields |
9:08PM |
0 |
Festival 1.95 on 64 bit linux 2.6 FC3 |
8:14PM |
8 |
Verizon PRI Setup Problems |
7:34PM |
0 |
Brian, Mr. West, are you out there? |
7:27PM |
2 |
Verizon PRI Setup Problems - Only Busy and Congestion |
6:41PM |
0 |
astersik sip routing question |
6:37PM |
1 |
terminate sip calls from a 3rd party sip proxy into asterisk. and then to gnugk |
6:13PM |
2 |
Sipura 841 delayed: other PoE options? |
5:38PM |
0 |
Asterisk make ext. light up? |
4:26PM |
1 |
Out of State |
2:50PM |
3 |
Realtime problem |
2:36PM |
0 |
SIGSEGV, Segmentation fault while debugging asterisk with gdb |
2:14PM |
3 |
Confirm MWI doesnt work with SIP RealTime? |
2:11PM |
9 |
list broken again? |
1:02PM |
3 |
IAX Provider Recommendation - Unlimited |
12:50PM |
1 |
SIP and * with dual ethernet cards |
12:09PM |
5 |
Digium Hardware in Canada |
11:44AM |
5 |
Soekris net4801 for home use? |
11:36AM |
3 |
Asterisk Randomly Hanging up on Zap channels |
11:29AM |
2 |
ztcfg problems |
11:00AM |
1 |
404 "Not Found" Sip Response |
10:55AM |
1 |
X100P and Mitel SX-2000 Light |
9:20AM |
2 |
Re: Asterisk-Users Digest, Vol 5, Issue 192 |
9:10AM |
0 |
Kirk IP600 Wireless DECT station setup?? |
8:58AM |
4 |
numeric caller id display on budgetone 101 |
8:56AM |
2 |
Virtual Modems |
8:53AM |
2 |
CLI Timeout ? |
8:49AM |
0 |
voicemail playback problem |
8:48AM |
6 |
least sucky FXO interface? |
8:35AM |
0 |
Voicetronix FXO on OpenCall 4 vs OpenSwich 6 |
8:24AM |
0 |
How to do this ? |
8:22AM |
5 |
IAXy provisioning |
8:21AM |
0 |
Should echo cancellation be a "science" oran"art"? |
8:19AM |
1 |
Asterisk to sip client behind Firewall/NAT-cancall but cannot receive calls ? |
8:14AM |
0 |
volume problems on zaptel |
8:04AM |
0 |
Mixing PRI's and BRI's |
7:07AM |
1 |
How to debug? - SIP calls not coming thru |
7:04AM |
0 |
AGI Helpdesk/Trouble Ticketing application |
7:00AM |
2 |
silence suppression question |
6:40AM |
3 |
sip_buddies mysql table |
6:17AM |
0 |
Problems with Chan_capi 0.3.5 & Asterisk 1.0.3 |
6:14AM |
1 |
ISDN HiSax: unauthorized source code changes |
6:13AM |
1 |
Suggested Literature |
6:07AM |
0 |
Setting ISDN Service Codes chan_capi/zaphfc |
5:52AM |
2 |
Asterisk Realtime IAX - Adding fields for database table |
5:48AM |
0 |
tetting |
5:15AM |
1 |
Softphone features |
4:26AM |
0 |
Radius support |
4:08AM |
0 |
Snom 190 and lamp field |
3:36AM |
2 |
SIP registrations not staying registered |
3:32AM |
2 |
Dial Plan Problems |
3:30AM |
1 |
Asterisk to sip client behind Firewall/NAT - cancall but cannot receive calls ? |
2:51AM |
3 |
Problems with app_realtime |
2:23AM |
0 |
Asterisk to sip client behind Firewall/NAT - can call but cannot receive calls ? |
2:12AM |
0 |
Analog modem testing |
2:05AM |
1 |
Astersik with ISDN up0 |
1:59AM |
1 |
Help with Queue Cmd |
1:18AM |
0 |
Issues with Asterisk |
|
Monday December 13 2004 |
Time | Replies | Subject |
11:33PM |
4 |
Caller ID on Snom 190? |
11:29PM |
1 |
Newbie-Firewalls? |
11:13PM |
11 |
ASTCC |
11:02PM |
0 |
Regarding IRQ problems; try googling for "Digital Audio Workstation" or "DAW" |
11:01PM |
0 |
Looking for Full or Part time asterisk techs |
10:43PM |
3 |
Busy message on ISDN cards? |
9:56PM |
2 |
Cisco Router FXO / Skinny |
9:16PM |
0 |
What is the purpose or zttest and ztspeed ? |
9:04PM |
0 |
Setting up prepaid |
7:57PM |
1 |
DS3 Media Gateway |
7:54PM |
1 |
Bad Request Connecting SIP |
7:36PM |
0 |
hardware IAX to PSTN gateway? |
6:56PM |
0 |
Ethernet Channel Bank (Comming Soon to a NOC NearYou!) |
6:12PM |
0 |
SIP and IAX login design |
5:36PM |
0 |
Transfer and keep variables |
5:16PM |
0 |
AstWinPeers - combination of IAX/SIP/Peers/Graph |
5:15PM |
1 |
Repost: Cisco 7960 and Asterisk...not working.... |
4:45PM |
1 |
incoming call from pstn to fxo not working with Asterisk |
4:21PM |
1 |
Asterisk up & running, now what? |
4:09PM |
0 |
MultiTech VOIP box |
3:48PM |
2 |
The correct way to get most recent stable |
3:23PM |
0 |
How to connect * to Adtran 600? |
2:59PM |
1 |
recommended IP phones and VoIP providers? |
2:57PM |
3 |
CPU spikes with wcfxs loaded |
2:56PM |
2 |
How can i test a modem with Asterisk? |
2:53PM |
0 |
phpconfig - can't locate any of my sections |
2:19PM |
0 |
looking for input on broadband router with QoS andVPN support |
2:07PM |
0 |
Disa Cdr |
2:01PM |
0 |
SIP CGI |
1:25PM |
0 |
weird ring behavior |
1:14PM |
2 |
Incoming Toll-Free |
12:11PM |
0 |
setting up asterisk as voicemail for softswitch |
11:50AM |
5 |
Multiline / Console / Receptionist phone |
11:47AM |
1 |
Can a TDM21 and a X100P co-exist |
11:40AM |
0 |
Asterisk and Sipura SPA-2000 |
11:28AM |
0 |
Discontinued Firmware? |
10:44AM |
6 |
Asterisk on SuSE 9.1? |
10:18AM |
0 |
Portuguese (Brazil) configuration setup |
10:14AM |
1 |
Asterisk and Cisco 7905G or Cisco 7912G |
9:36AM |
7 |
How to create a confrence using SIP channels |
9:22AM |
0 |
Music on Hold with Parking |
9:17AM |
1 |
CallerID after Supervised Transfer |
8:58AM |
0 |
Detect line in use? |
8:22AM |
1 |
only allow long distance calls to countries x, y, and z |
8:09AM |
6 |
Pitching Asterisk |
6:51AM |
0 |
Broadvoice Patch Applied to CVS |
6:09AM |
0 |
Reading mysql sip friends |
6:08AM |
3 |
CVS zaptel missing files |
6:00AM |
0 |
install e100 card errors |
5:42AM |
7 |
Dialing out to 2 clients simultaneously |
5:42AM |
1 |
MYSQL cmd - preconnect? |
5:23AM |
0 |
[oh323] sporadic call setup |
5:19AM |
0 |
Asterisk receiving SER calls |
5:05AM |
2 |
transferring variables with IAX2? |
5:03AM |
2 |
Echo on one E1 line, but not the other |
4:46AM |
1 |
Traditional Telephony Interface Card |
4:40AM |
1 |
What route do diverted SIP calls travel? |
4:16AM |
2 |
IAX.cc / Sixtel? |
3:54AM |
1 |
Doing a # transfer on calls needing a # |
3:17AM |
3 |
Strange Segmentation fault |
2:45AM |
0 |
outgoing call queue. |
2:08AM |
0 |
Call Monitor Fails after Transfer |
1:56AM |
1 |
"detected NAT type is full cone" for BT behind nat ? |
1:25AM |
0 |
Issues getting Asterisk Realtime configured and operational |
12:37AM |
2 |
Follow Me & Music on hold |
|
Sunday December 12 2004 |
Time | Replies | Subject |
9:45PM |
1 |
Sipura SPA-2000 won't ring |
8:42PM |
1 |
Log's Message Codes |
8:33PM |
0 |
DUNDi performance |
7:57PM |
1 |
patton smartnode integration |
7:25PM |
0 |
Any plans for video in oh323? |
5:13PM |
1 |
Using SPANDSP for faxes |
4:35PM |
0 |
BRI Problem dialing out |
4:00PM |
1 |
zaptel 0.9.1 compile problem |
2:17PM |
0 |
IAXPeerGraph - a beta of another windows monitor app |
1:33PM |
1 |
I'm stumped |
1:01PM |
2 |
[OT] Small SIP phones? |
11:12AM |
1 |
Re: Cant set H323 up |
9:35AM |
3 |
TDM400P FXS polarity reversal? |
8:39AM |
1 |
Totally LOST with dialplan and Extensions. |
8:10AM |
1 |
SV: How to Playback Mailbox Owners Name? |
7:38AM |
0 |
MeetMe performance |
7:35AM |
0 |
3com NBX and Asterisk Integration. |
6:57AM |
1 |
Pattern-matching in the dial-plan |
5:38AM |
1 |
gap in priorities - what happens |
5:34AM |
1 |
can a TDM400P FXS drop voltage on hangup? |
3:47AM |
2 |
Caller ID info ZAP --> SIP?? |
3:45AM |
2 |
How to Playback Mailbox Owners Name? |
2:06AM |
3 |
Problems getting Asterisk Realtime to work |
1:20AM |
1 |
Will Adtran TSU 600 work with *? |
1:09AM |
0 |
DIALSTATUS missing an important condition? |
|
Saturday December 11 2004 |
Time | Replies | Subject |
9:19PM |
0 |
Variable-length dialing with a Quicknet Inetnet PhoneJACK card |
8:39PM |
1 |
modprobe wcfxo causes fc3 box to crash |
7:35PM |
0 |
20 BT-100 setup - what firmware is recomended ? |
7:32PM |
1 |
Many similar contexts - can I use Macro or some other template concept ? |
7:06PM |
1 |
Can't capture "-1" return on Dial command |
5:44PM |
2 |
help with detecting fax. |
4:45PM |
1 |
Problem with TDM400P and cidstart=polarity |
4:39PM |
2 |
ACK from asterisk not matched to transaction by SER / LCS2005 |
4:17PM |
5 |
does aanyone have an example of how to dial outwith a sip phone on a pstn line? |
3:44PM |
0 |
Soyo G668 IP Phone |
3:43PM |
0 |
SPA-2000 NAT Problems |
2:46PM |
0 |
Background Music via telephone speaker. |
2:32PM |
1 |
looking for input on broadband router with QoS and VPN support |
2:06PM |
0 |
Cisco 7960 and Asterisk...not working.... |
1:06PM |
0 |
Migrating from CVS HEAD to Stable 1.0.3? |
1:00PM |
0 |
Tormenta PCI - tor2 module not loading |
11:23AM |
2 |
Cisco 7960 says "Protocol Application Invalid?" |
10:05AM |
0 |
636 Area Code Asterisk Compatible DIDs |
9:35AM |
1 |
What might be blocking RTP |
9:27AM |
1 |
How to setup private enum server ? |
9:13AM |
2 |
voicemail from mysql / change password |
9:07AM |
0 |
does aanyone have an example of how to dial out with a sip phone on a pstn line? |
8:32AM |
0 |
Re: Asterisk-Users Digest, Vol 5, Issue 158 |
8:29AM |
0 |
Linux basics and Asterisk basics |
8:13AM |
0 |
Monitor, append audio? |
7:54AM |
0 |
Newbie MusicOnHold issues |
7:53AM |
1 |
OT: canterburyfortmyers.org returned mail |
7:00AM |
1 |
IAXy: no dial tone |
5:39AM |
1 |
Handling "raw" audio (8000 signed 16bit big-endian) |
4:25AM |
0 |
Asterisk 1.0.3 and chan_capi ? |
4:01AM |
1 |
RealTime and Macro question? |
1:34AM |
2 |
long list of prefixes |
|
Friday December 10 2004 |
Time | Replies | Subject |
11:05PM |
0 |
-p real time priority and -U together |
11:04PM |
1 |
Can I re-write an incoming caller-id? |
10:58PM |
3 |
How to test enum? |
10:23PM |
0 |
Voicemail User Reference Guide |
9:26PM |
0 |
TXTCIDName |
7:26PM |
2 |
Very Cool.........Asterisk Made Wired Magazine |
7:12PM |
0 |
Re: Asterisk-Users Digest, Vol 5, Issue 158 |
5:55PM |
1 |
E100P / Brazilian Telco Problem. (Urgent) |
4:56PM |
0 |
Asterisk RealTime Wiki Pages |
4:30PM |
0 |
Help setting-up X-Pro behind a proxy |
4:26PM |
4 |
New PRI with DID in US? |
3:53PM |
1 |
IAXPeers for Windows Beta released |
3:42PM |
0 |
Polycom caller id issues |
3:34PM |
1 |
Return code from queue app |
3:22PM |
0 |
* as a fax/voice switch |
3:00PM |
8 |
Voice Prompt Info |
2:33PM |
1 |
Should echo cancellation be a "science" or an "art"? |
2:31PM |
3 |
Asterisk Training Needed in SouthEast U.S |
2:29PM |
4 |
Linux basics |
2:25PM |
1 |
ringing after hangup |
2:15PM |
2 |
[Fwd: Re: udev or not?] |
1:35PM |
3 |
Need an Asterisk Expert for a Project |
12:58PM |
1 |
Install Xc-Ast $$$ |
12:45PM |
1 |
Apply Patch for Broadvoice. |
12:40PM |
1 |
T.38 Pass-Thru? |
12:17PM |
2 |
include and hint in extensions.conf with new realtime feature - how? |
12:04PM |
0 |
providing battery reversal from Asterisk to legacy pbx |
11:52AM |
0 |
Confused about proxying and NAT, and seeking guidance |
11:50AM |
2 |
Asterisk from CVS |
11:43AM |
0 |
Help with configuring CFAS groups |
11:39AM |
2 |
static recording |
11:05AM |
2 |
using built-in extension numbers on the ZAP channel |
11:00AM |
7 |
Ripping CD audio for MOH |
10:15AM |
0 |
MySQL - mistake in previous post |
9:33AM |
2 |
Integrating * with Mitel SX2000 Lite |
9:14AM |
0 |
AGI Perl |
9:12AM |
2 |
dtmfmode: inband question |
9:08AM |
5 |
Granstream phones message button |
9:00AM |
1 |
Intercept and redirect outgoing calls ? |
8:58AM |
0 |
Not receiving DTMF from gateway |
8:40AM |
0 |
Aditional local number when calling from ISDN thtough Capi to local extension ? |
8:18AM |
3 |
OT: How do I know if I should have IO-APIC? |
7:58AM |
0 |
Change logs |
7:42AM |
2 |
Asterisk 1.0.3 - Signaling on E100P. |
7:22AM |
1 |
MySQL Realtime Driver |
7:02AM |
0 |
Moving call control to a second server |
6:44AM |
0 |
Dialing Problem with Welltech 3806 FXO gateway |
6:23AM |
2 |
BT100 how to pickup a parked call |
6:01AM |
0 |
SS7 to E1 & CPC |
5:51AM |
2 |
ISDN Data calls through * |
5:19AM |
1 |
Doubts regarding g726 - 16 bits setup |
4:35AM |
0 |
sip phone...direct access... |
4:29AM |
2 |
Mysql configuration interface |
4:11AM |
0 |
analog FXO debug suggestions |
4:10AM |
0 |
voice + data |
4:02AM |
0 |
D/41E ISA Card with redhat 8.1 |
2:46AM |
0 |
variable limit time on Dial |
12:39AM |
1 |
udev or not? |
12:31AM |
3 |
PoE VOIP phones in Australia |
12:27AM |
3 |
polycom phone IP 500/600 conference feature |
|
Thursday December 9 2004 |
Time | Replies | Subject |
11:41PM |
2 |
Asterisk started but doesn't register SIP client |
11:23PM |
1 |
Lost admin password on Polycom IP500? |
11:20PM |
1 |
Lost Password to Polycom IP500 |
10:35PM |
0 |
Balanced call distribution to agents logged into multiple queues. |
8:30PM |
6 |
Cisco AS5XXX to asterisk debugging. |
7:50PM |
0 |
Disconnect Via Budgetone and 3com NBX |
7:13PM |
11 |
Asterisk@Home |
6:56PM |
2 |
SCRIPT: Fax Remvoal Please Call: 1-800... |
6:06PM |
2 |
Audio Hung after 1st call |
6:06PM |
1 |
Slackware & zttool |
6:03PM |
3 |
urgent outbound dialing problem |
4:17PM |
1 |
chan_capi question |
3:13PM |
1 |
No ring signal when calling internal extensions ? |
2:56PM |
1 |
Forward voicemail to *remote* voice mailbox? |
2:44PM |
0 |
--SOLVED--Voicemail messages by email |
2:23PM |
0 |
solution - running asterisk on box using alsa (FC3) for CONSOLE/dsp and wishing to play audio from browser |
2:21PM |
0 |
BT100 cannot park a call properly??? |
2:19PM |
1 |
prepaid calling card application |
2:07PM |
1 |
Cisco IP Conference 7935 |
1:47PM |
3 |
possible OT - ADIT 600 question |
1:16PM |
3 |
very OT - basic newbie networking |
12:57PM |
6 |
Voicemail messages by email |
12:45PM |
0 |
anyone know anything about audiocodes analog gw's |
12:43PM |
0 |
running asterisk on box using alsa (FC3) for CONSOLE/dsp and wishing to play audio from browser |
12:28PM |
0 |
New batch of phrases from Allison |
12:26PM |
1 |
Changing NICE value for * will it help? |
12:15PM |
1 |
can FXS ports on TDM400P provide Battery Reversal or CPC |
11:52AM |
1 |
Providers for PSTN Access |
11:47AM |
1 |
Asterisk@Home software? |
11:00AM |
2 |
Silent IAX calls getting cut off |
10:13AM |
0 |
Incomming calls on h323 |
9:40AM |
0 |
Ser + Asterisk & DMZ |
9:36AM |
2 |
Multiple Instances of Asterisk |
9:31AM |
0 |
safe_asterisk not working |
9:28AM |
0 |
Problem with Accounting and wrong Caller ID |
9:21AM |
4 |
MySQL, CDR with MySQL |
9:14AM |
0 |
Can asterisk accept cleartext auth (uri user:pass) via SIP |
9:08AM |
0 |
Polycom IP400 |
9:02AM |
4 |
Handsfree Speakerphone |
8:58AM |
6 |
Horrible MeetMe performance |
8:30AM |
3 |
Adit Asterisk Cabling Connundrum. |
8:25AM |
0 |
Reminder: $500 Bounty for Bluetooth |
8:25AM |
5 |
Sipura SPA-841 |
8:19AM |
2 |
pseudo load balancing? |
7:53AM |
1 |
OT- Dell Xeon Servers UK Dealy, was Asterisk with SMP hardware |
7:45AM |
1 |
res_perl module loading problem |
7:36AM |
1 |
sip+nat+bt-100 |
7:34AM |
0 |
Asterisk Monitor after Call Transfer failing to record the call |
7:28AM |
0 |
For all of those wondering about zaptel hardware and interrupts |
7:20AM |
3 |
Swissvoice IP 10S VoIP Telephone |
6:26AM |
1 |
[OT] Adit 600 Question |
6:24AM |
0 |
RE: Re: News about SS7? (Storer, Darren) |
6:09AM |
0 |
Base Number and DIDs |
5:53AM |
0 |
chan_sip2 multiple outbound proxies |
5:43AM |
2 |
hfc card and isdn error E001B |
5:23AM |
12 |
four wildcards in a single pc |
5:13AM |
0 |
Got SIP response 403 "Anruf nicht erlaubt" back from 194.97.54.97 |
5:12AM |
1 |
Xorcom Rapid 0.9.0 |
5:07AM |
1 |
A waning console error |
4:35AM |
0 |
Workimg On PostgrSQL |
3:00AM |
1 |
IAX midget packets!? |
2:59AM |
2 |
MeetMe Features |
2:48AM |
0 |
Asterisk and Cisco 5350 - config ? |
1:50AM |
1 |
Call Transfer drop. |
1:50AM |
1 |
pppd dial-in over asterisk |
1:45AM |
1 |
News about SS7? |
1:21AM |
5 |
BT-100 Transfer!! |
1:16AM |
1 |
Spandsp loading via asterisk app_rxfax.c broken pipe. |
12:57AM |
6 |
very OT - basic newbie networking question |
12:32AM |
4 |
Get rid of H323 problems for 100$ |
|
Wednesday December 8 2004 |
Time | Replies | Subject |
10:24PM |
5 |
How to demo the Power of Asterisk |
9:32PM |
2 |
Asterisk with SMP hardware |
8:41PM |
1 |
ftmp header |
7:43PM |
12 |
Ethernet Channel Bank idea |
7:15PM |
2 |
NEC Univerge |
6:22PM |
0 |
Two Zap Problems with 1.0.2 that appeared at the same time: choppyness and squealing |
5:58PM |
2 |
Broadvoice and incoming DTMF |
5:45PM |
1 |
PSTN number with callhunt and voicemail we web interface |
5:13PM |
1 |
Leadtek BVA8051 / Sipphone.com CallInOne with Asterisk? |
4:56PM |
2 |
CAPI, BRI and grouping B channels |
3:54PM |
0 |
Asterisk Intermediate-Advanced Administrator wanted in South Florida |
3:11PM |
0 |
UA -> SER -> asterisk |
3:01PM |
7 |
more then two wildcards in one machine |
2:56PM |
3 |
Playing Audio before the Phone is Ready |
2:44PM |
7 |
SIP Client for Symbian |
2:26PM |
3 |
Asterisk 1.0.1 Too many open files |
2:18PM |
0 |
how to make asterisk drop battery on a FXS? |
2:05PM |
0 |
Can Dial Calls from an Estara SIP Client, but Cannot Complete Calls to the |
1:44PM |
5 |
Asterisk Maintenance |
1:15PM |
1 |
ASTCC MySQL CDR |
12:47PM |
4 |
Guide to Cisco 79xx |
12:34PM |
0 |
Re: Asterisk-Users Digest, Vol 5, Issue 113 |
11:39AM |
4 |
Polycom 500 - Dialtone while connected |
11:33AM |
1 |
3com phones and Asterisk |
10:41AM |
2 |
Dead TDM400P ? |
10:26AM |
0 |
OT: Polycom IP 400 |
10:01AM |
0 |
OT: CP-7960's are in for those of you whop purchased them. We are shipping today. |
9:48AM |
0 |
Are there any digital phones that runon asteriskyet? |
9:30AM |
0 |
Zaprtc seems unsupported, Asterisk in productionenvironment without Digium cards |
9:28AM |
2 |
Dropping Calls, irregular interval no logs |
9:14AM |
1 |
Using meetme video mode with SIP ? Now a $2000 bounty |
9:07AM |
10 |
pc |
8:56AM |
4 |
T100P PRI question |
8:56AM |
0 |
Asterisk with 3COM phones |
8:42AM |
0 |
Does Asterisk support 3com 1102 phones ? |
8:31AM |
0 |
small business installation. |
7:35AM |
2 |
Voicetronix vs Digium FXO |
7:12AM |
2 |
PrivacyManager 10 digit limit. |
6:16AM |
0 |
IAXy & Auto-Dial |
6:02AM |
3 |
CPC, Calling Party Control, Disconnect supervision, -- how to tell that to Verizon (east coast)? |
5:56AM |
0 |
Looking for a Vonage contact |
5:36AM |
0 |
/dev/ttyI and few ISDN cards |
5:34AM |
8 |
setting the Call Forward Number in Zap? |
4:51AM |
0 |
Re: Spandsp loading via asterisk app_rxfax.c brokenpipe. |
4:34AM |
4 |
asterisk consultants |
3:52AM |
2 |
Asterisk's Empty Folder |
3:03AM |
0 |
Dropped calls on IAX connection |
2:12AM |
2 |
Spandsp - Libtiff problem |
12:40AM |
7 |
sangoma |
12:09AM |
0 |
Source/cause of echo delay (on internal stuff network) |
|
Tuesday December 7 2004 |
Time | Replies | Subject |
11:35PM |
0 |
Busy Detect |
10:12PM |
1 |
dead BT100 |
9:44PM |
0 |
Zaptel HDLC (NetHDLC) errors on modprobe, Linux 2.6 kernel |
9:44PM |
3 |
SIP endpoints ----> RTP stream |
7:57PM |
3 |
Cepstral voices |
7:51PM |
0 |
Subject: Re: Analog FXO Woes Continue |
7:18PM |
0 |
why busydetect can work sometimes, then sometimes not. |
6:44PM |
1 |
Segfaults when playing GSM files |
5:26PM |
1 |
conferece/Voice Mail features and LBR codecs (G7231, G729) |
5:10PM |
0 |
AGI application doing Hangup command and different AGI application running receiving the Hangup - additional |
4:56PM |
3 |
:: Migrating to 1.0.3 => Attention. :: |
4:42PM |
3 |
Asterisk / VOIP Employment Opportunity |
4:14PM |
3 |
Continuance on Polycom issue, not ringing |
3:51PM |
0 |
cf gsm adaptors |
3:48PM |
4 |
Broadvoice - DTMF |
3:31PM |
1 |
Inoming caller id withheld, move to new context, possible? |
3:29PM |
3 |
can't compile chan_capi 3.5 after patch applied :-( |
3:17PM |
1 |
Ringing multiline phone |
3:13PM |
0 |
sip phone to sip phone errors |
2:59PM |
0 |
AGI application doing Hangup command and different AGI application running receiving the Hangup |
1:56PM |
1 |
asterisk & 3rd party vm |
1:38PM |
0 |
ISDN on com port /dev/ttyS0 possible ?? |
1:37PM |
0 |
Comdial PBX -- can use Asterisk as VM |
1:24PM |
0 |
monitor load on (zap)channels ? |
1:03PM |
1 |
Monitoring a call in an Call Center Environment |
12:50PM |
2 |
Allow calls to certain area codes |
11:57AM |
0 |
Asterisk dropping calls when transferred on another PBX |
11:51AM |
0 |
asterisk-oh323-0.6.3b and logical Channel |
11:06AM |
7 |
Faxing..not 100% |
10:39AM |
1 |
Restrict outbound calls on Broadvoice |
10:11AM |
1 |
asterisk and kphone (sip soft phone for linux) on same machine |
10:01AM |
1 |
IAX DIDs, Illinois |
10:00AM |
1 |
Problem on Outgoing Calls (FXO - SIP) |
9:28AM |
0 |
Broadvoice patch and latest CVS version |
9:27AM |
1 |
How to play messeage when user picks up the phone |
9:09AM |
0 |
Dropping calls, Polycom Renegotiation timeout? |
9:05AM |
0 |
Avaya 4606 IP Telephone |
9:00AM |
1 |
Fine Tuning |
8:51AM |
1 |
astcc needs AGI.pm...where is it? |
8:40AM |
0 |
Calls dropping, when server sysncs time? |
8:37AM |
1 |
SIP URLs |
8:31AM |
9 |
Analog FXO Woes Continue |
8:21AM |
2 |
Firewall traversal anomalies - AJA |
8:12AM |
2 |
TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment |
7:56AM |
2 |
modprobe ztdummy - failed |
7:48AM |
0 |
save dialplan missing in 1.0.2?? |
7:44AM |
8 |
Website that reads text recently on the list? |
7:31AM |
1 |
H.323 trunking |
7:30AM |
0 |
Skinny error : Unable to create channel |
5:51AM |
4 |
Linking asterisk to an existing small office PBX |
4:50AM |
1 |
Comdial PBX -- can use Asterisk as VM box? |
4:35AM |
3 |
Question about e1/digium |
4:22AM |
0 |
Mini-ITX Mainboard for Asterisk IP PBX, Intel Mobile Celeron 733MHz |
4:02AM |
4 |
Transfer on Snom 190 |
4:00AM |
0 |
IAX2 Hangup Cause |
3:52AM |
0 |
callerid PSTN->IAX problem |
3:36AM |
1 |
chan_capi 0.3.5 does not compile |
3:10AM |
0 |
GrandStream BT VS. IP500 Latency |
2:54AM |
2 |
High(er) availability |
2:42AM |
1 |
gsm codec, very poor quality. |
2:30AM |
1 |
Strange softphone problem |
2:24AM |
1 |
Interface analogue exchange line to VOIP phone? |
1:23AM |
6 |
Voice mail problem |
1:13AM |
0 |
new version problems |
12:31AM |
0 |
OT: Two way trunks in Korea? |
|
Monday December 6 2004 |
Time | Replies | Subject |
11:45PM |
2 |
Is anyone using Cisco 7905G phones? |
11:18PM |
2 |
Asterisk 1.0.3 |
10:56PM |
0 |
MGCP Gateway |
10:15PM |
0 |
pstn <> asterisk -- pstn handled by asterisk box |
10:08PM |
0 |
ACT P104SLD (10 Line) phone - "Line Key Settings" ??? |
9:36PM |
5 |
two questions |
9:12PM |
0 |
extension number when calling to registered gateway |
8:02PM |
1 |
DTMF via PSTN to * to IAX to * challanges. |
5:11PM |
2 |
Asterisk ---> Cisco AS5XXX sip one way audio |
5:01PM |
3 |
Kind of off-topic: VoIP services and multipl e callers |
4:41PM |
0 |
Phone Giptel G100 with Asterisk? |
3:55PM |
0 |
Kind of off-topic: VoIP services and multiplecallers |
3:48PM |
2 |
dialplan |
3:40PM |
1 |
Another "Unable to create channel of type 'Zap' (cause 0)" error |
3:39PM |
2 |
handset to sound card |
3:32PM |
0 |
CALLPROGRESS configuration for a X101P |
3:25PM |
0 |
Passing SIP digest auth to dialplan |
3:11PM |
1 |
I need very fast quick info how to setup ISDN card |
3:01PM |
4 |
Are there any digital phones that run on asterisk yet? |
1:57PM |
2 |
Kind of off-topic: VoIP services and multiple callers |
1:50PM |
0 |
T1 digit timeout when dialing manually |
1:48PM |
1 |
how to start with ISDN |
1:30PM |
0 |
ASTERISK -> SPANDSP |
1:08PM |
0 |
strange caller id and caller name with SIP and ATA186 |
1:01PM |
1 |
retrieve_extensions_from_mysql.pl |
12:14PM |
3 |
Is this possible |
11:49AM |
1 |
Asterisk on Macintosh - no sound card support? |
11:35AM |
0 |
TDM OnHook/OffHook |
11:29AM |
0 |
Firefly prescence + Asterisk |
11:04AM |
0 |
Italian Caller ID support in zapata.conf |
10:41AM |
0 |
Useful information - UK exchanges and SystemX/SystemY and how ISDN works! |
10:34AM |
1 |
Broadvoice - bad quality, dtfm mode |
10:30AM |
0 |
How to verify if chan_sccp is working/built correctly? |
10:12AM |
2 |
Budgetone 101 phones ? SIP through NAT ? |
9:25AM |
1 |
Setting CallerID with ITSPs |
9:24AM |
1 |
G.711 Appendix II |
9:23AM |
3 |
PRI/Zap premature dialing problem |
9:18AM |
1 |
Console as extension problems |
9:12AM |
0 |
SoftPhone on * with X-Lite or iaxComm (1 X100Pcard) |
9:04AM |
0 |
UK callerid X100P? |
8:27AM |
0 |
Dropping calls on IAX2 |
8:06AM |
1 |
Queue Timeout |
7:58AM |
0 |
CVS HEAD h323 no longer builds? |
7:06AM |
2 |
SoftPhone on * with X-Lite or iaxComm (1 X100P card) |
6:29AM |
0 |
What would I need to do this? |
6:01AM |
1 |
SIP response 302 "Moved Temporarily " |
5:35AM |
1 |
SIP status lagged |
5:10AM |
2 |
h extension in macro |
5:02AM |
0 |
fax/voice switch - faxdetect |
4:11AM |
3 |
Recomended ISDN for Asterisk ? |
3:19AM |
0 |
Voicemail Codec challanges. |
2:55AM |
0 |
auto-dialout not doing LCR |
2:08AM |
1 |
iax2 nativ bridge question? |
2:07AM |
1 |
Users list. |
12:16AM |
0 |
Is the list down, or is it just me |
|
Sunday December 5 2004 |
Time | Replies | Subject |
11:33PM |
3 |
PRI configuration problem |
10:41PM |
1 |
Mysql-cdr |
9:25PM |
1 |
Hardware PSTN Gateways? |
8:35PM |
1 |
Re: Is Asterisk-users down? |
6:34PM |
3 |
List's quiet or down? |
5:19PM |
0 |
Dial D option not working? |
2:30PM |
0 |
Cisco IAD2421 with Asterisk |
10:41AM |
0 |
Sip Channels Left Open |
10:05AM |
0 |
Recomended ISDN on Asterisk@home ? |
6:34AM |
1 |
Group sip definitions? |
6:19AM |
2 |
ANALOG FXO ZAPTEL & WCFXO & WCTDM module issues seen with intermittent analog lines |
6:11AM |
1 |
asterisk + chan_sip2 + sipproxd + sipgate |
5:33AM |
0 |
just testing please ignore |
4:53AM |
5 |
G.729 algorithm? |
3:29AM |
3 |
full duplex sound card |
1:35AM |
0 |
Planet BRI TA will work ? |
|
Saturday December 4 2004 |
Time | Replies | Subject |
10:21PM |
0 |
System hardware requirements for * |
9:13PM |
2 |
Billing - which program are you using? |
8:49PM |
0 |
Typical Setup for a small/medium office |
7:24PM |
2 |
Email to Fax? |
7:21PM |
5 |
BLOCKING incoming FAXES on voice line. |
5:26PM |
0 |
x100p offhook/onhook states |
5:05PM |
3 |
X100P does not detect ringing |
5:00PM |
6 |
Door buzzer. |
4:57PM |
0 |
Integration to TAPIT/Call tracking software |
4:32PM |
5 |
Is Gigabit Ethernet necessary? |
3:49PM |
0 |
budge tone 100 caller id |
3:22PM |
2 |
Broadvoice outbound 404 error |
3:12PM |
1 |
Is this possible? |
3:12PM |
4 |
asterisk dabbling... |
3:03PM |
1 |
more DIALSTATUS/HANGUPSTATUS woes with IAX2 |
3:00PM |
2 |
Budgetone 100 Caller ID |
2:54PM |
0 |
Asterisk & Gossiptel - 1 way audio??? |
1:43PM |
1 |
chan_zap.c:6181 mkintf: Unable to get parameters |
1:01PM |
3 |
Voicemail for Current Extension? |
12:37PM |
2 |
ISDN kernel 2.6 problems chapi isdn4lin |
12:31PM |
2 |
SJPhone SIP Tab |
12:21PM |
0 |
Remote-Party-ID + CallerID + VoicemailMain |
11:33AM |
0 |
NewBie Question Modem Telephone -PSTN |
11:11AM |
0 |
IAX Native Transfer |
8:49AM |
2 |
iaxy to iaxy call drops out of "show channels" |
8:46AM |
2 |
XML to monitor queues on Cisco display ? |
8:42AM |
0 |
(no subject) |
8:09AM |
1 |
Codec translator problem (g723.1,ilbc => alaw) |
5:55AM |
0 |
PRI debug - weird behaviour |
5:35AM |
0 |
PRI debug output - still not working :( |
4:54AM |
1 |
Udev setup question for zaptel |
3:52AM |
1 |
chan_misdn and Dynalink IS64PH ISDN |
3:45AM |
2 |
Asterisk and Cisco IP Phones |
3:42AM |
2 |
ZAP and IAX Trunks |
3:34AM |
0 |
RES: howto install |
3:21AM |
1 |
Snom 220 busy lamps [was: Receptionist phone...] |
3:18AM |
1 |
howto install |
3:00AM |
3 |
Gossiptel with Asterisk? |
2:48AM |
5 |
Incoming SIP Address? |
12:47AM |
0 |
Asterisk stumbling block |
|
Friday December 3 2004 |
Time | Replies | Subject |
9:20PM |
1 |
compiling asterisk-addons for Mysql-cdr |
8:09PM |
0 |
Mixing x100p & te405p ?? |
7:18PM |
1 |
Help with music over intercom. |
5:57PM |
3 |
Two zaptel T1 cards: no clock from one |
5:47PM |
5 |
SIP SECURITY WARNING: v1-0 (cvs today) sip context in general section ignored goes to default instead - allowing unauthorized sip devices to place calls in default context |
5:16PM |
0 |
transfer question |
4:58PM |
1 |
Call parking/transfer not working on IAX2 connections |
4:13PM |
0 |
Asterisk sms voicemail notification |
3:39PM |
2 |
7905G Firmware |
3:07PM |
0 |
IO-APIC |
3:01PM |
1 |
iaxy not hear ringing |
2:30PM |
4 |
Polycom 500, won't ring?? |
2:24PM |
1 |
CAPI Newbie |
2:17PM |
8 |
Why, why, why??? |
1:36PM |
2 |
DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL) |
1:35PM |
1 |
FOP Asterisk Manager Login Failed? |
12:43PM |
0 |
ZAPHFC, Asterisk does not load with signalling = bri_net_ptmp |
12:39PM |
1 |
Umlaut over I on Definity display |
12:39PM |
2 |
Unable to create channel of type 'Zap' (cause 0) |
12:02PM |
1 |
Alpha Paging |
11:25AM |
0 |
feature suggest.: alt. include criteria |
11:01AM |
0 |
Initial Chirp while dialing |
10:34AM |
1 |
PolyCom MWI Chirp issue |
9:27AM |
1 |
Best VM codec for Linux/OS X/Windows environment |
9:10AM |
0 |
Digium+asterisk+festival+outgoingcall: How detect a busy line..? |
8:57AM |
0 |
ipkall & one way audio |
8:33AM |
6 |
Ouch, part reset, quickly |
8:06AM |
0 |
IAX2 Codec Pref order. |
6:46AM |
0 |
Incoming TypeOfNumber on zap, not just iax2? |
6:20AM |
1 |
HasNewVoicemail does not find voicemailbox, but files exist |
6:13AM |
2 |
Status of linux 2.6 support |
5:49AM |
1 |
Queue without # |
5:45AM |
0 |
RE: Asterisk-Users Digest, Vol 5, Issue 42 |
5:37AM |
0 |
Testing Voip calls only |
4:33AM |
3 |
Bluetooth with * |
3:51AM |
0 |
Open G723.1 - problems. |
3:06AM |
1 |
SMS in Asterisk |
2:22AM |
0 |
How stable ist the Asterisk Fax Manager |
1:28AM |
0 |
Asterisk and MaxDB |
|
Thursday December 2 2004 |
Time | Replies | Subject |
11:54PM |
0 |
Incoming SIP calls not being sent to "s" extension |
11:22PM |
1 |
Blank Machine Again. |
10:52PM |
0 |
Playing the message when user pickups the phone |
10:45PM |
0 |
Taiwan follows ETSI in permitting DTMF and FSK signals. ???? |
9:50PM |
0 |
asterisk connection problem |
8:10PM |
0 |
ParkAndAnnounce Problem |
8:09PM |
1 |
Problems with analog line |
7:21PM |
0 |
park announcement not working Help! |
7:06PM |
0 |
can both chan_h323 and asterisk oh323 be installed on the same machine? |
6:38PM |
1 |
IAXy & ADSI ? |
6:37PM |
4 |
Codec Conversion |
5:02PM |
3 |
Very odd musiconhold |
4:24PM |
5 |
drive space for voice mail |
3:39PM |
0 |
Getting the right DST in CDR |
3:37PM |
6 |
Dial Command M(x) Option |
3:30PM |
0 |
Connection Problem |
3:30PM |
3 |
fallthrough extension. |
1:52PM |
0 |
E100P not starting? |
1:06PM |
3 |
No Files Seen via vmail.cgi |
12:56PM |
4 |
Multi-Line sip phone? |
12:41PM |
0 |
Newbie - Get IVR Informations |
12:38PM |
2 |
more than 3 msns with chan_capi |
12:18PM |
1 |
SpanDSP 0.0.2pre6 undef symbol on gentoo-ppc |
12:02PM |
0 |
Polycom POE Rumor |
11:41AM |
0 |
new asterisk installation report and request for mixed voice data apps |
11:35AM |
4 |
Asterisk Problem or Polycom Problem |
10:12AM |
0 |
[OT] Dutch Asterisk meeting |
9:41AM |
2 |
Sipura Blind Transfer - Help |
9:40AM |
1 |
firefly and caller id |
9:40AM |
0 |
(no subject) |
9:37AM |
1 |
Agent Login "Play a file" |
9:20AM |
2 |
threeway calling |
9:19AM |
1 |
IAX2 and TEXT |
8:54AM |
6 |
Asterisk crashes my router!? |
8:35AM |
1 |
GUI for Asterisk Configuration |
8:22AM |
2 |
Asterisk with SMS |
8:21AM |
6 |
Polycom 500, asterisk user opinions? |
8:01AM |
0 |
[OT] detect-string.pl |
7:56AM |
4 |
Ring all Configured Extension |
7:50AM |
6 |
Restarting * |
7:30AM |
10 |
Conference |
7:26AM |
0 |
ForkCDR app call disposition ALWAYS says ANSWERED?????? |
6:18AM |
1 |
the pstn line is noisy, busydetect can detect hangup? |
6:10AM |
4 |
TE110P + Asterisk |
4:44AM |
0 |
Incoming call errors |
4:37AM |
0 |
transfering a incoming sip call automaticlly to another number |
3:16AM |
0 |
IAX to freshtel |
1:15AM |
1 |
900# DID? |
12:54AM |
0 |
Re: [Asterisk-Dev] One D channel for multiple spans |
12:35AM |
0 |
Newby with no idea |
12:25AM |
1 |
Getting a US Number |
|
Wednesday December 1 2004 |
Time | Replies | Subject |
11:34PM |
5 |
SV: www.voip-info.org |
11:31PM |
1 |
www.voip-info.org |
11:29PM |
4 |
Voicemail - Danish, German an French audio files download? |
10:54PM |
0 |
asterisk version 0.7.1 |
10:28PM |
2 |
Newbie Time |
9:23PM |
2 |
What exactly does IAX and SIP termination mean??? |
5:55PM |
0 |
How to get transfer and blind transfer on 7905 |
5:25PM |
0 |
transparent call routing |
4:44PM |
1 |
No version string |
4:39PM |
0 |
Diagnosing codecs |
4:35PM |
0 |
SIP->IAX->SIP silences |
4:29PM |
0 |
Asterisk / Paris Meeting |
4:00PM |
1 |
Hypothetical IAX2 situation |
3:52PM |
0 |
Interrupt Conflicts |
3:26PM |
6 |
Asterisk + Satellite connection |
3:22PM |
0 |
threeway calling while conferencing |
3:07PM |
1 |
IAX long distance... Re: Asterisk for home office |
3:02PM |
1 |
Micronet problem |
2:23PM |
14 |
ASTCC configuration problem |
2:08PM |
5 |
app_queue question |
1:51PM |
1 |
Sometimes calls are silent |
1:03PM |
1 |
[OT] [slightly] app lever vs driver level implementation... |
12:44PM |
0 |
Re: ASTCC |
12:10PM |
0 |
Caller ID showing My Own number |
12:07PM |
3 |
grandstream bt100 upgrade 1.0.5.18 |
12:02PM |
0 |
setting up conference room option |
11:48AM |
1 |
conference room possible bug |
11:12AM |
2 |
dont write me again |
11:01AM |
0 |
Grandstream BT100 / HandyTone 286 and Level 3 |
10:48AM |
9 |
Sveasoft Alchemy QOS |
10:47AM |
2 |
voicemail cuts off / hangs up |
10:11AM |
3 |
zaptel and low ring voltage |
10:10AM |
8 |
Interrupt latency problems |
9:32AM |
0 |
VoIP Dialout issues |
8:52AM |
1 |
some infos |
8:41AM |
4 |
Getting started with Asterisk |
8:22AM |
0 |
X101P interface (asterisk newbie) |
7:56AM |
2 |
Asterisk Call Monitor and soxmix error |
7:41AM |
1 |
CallerID on X100P in South Africa |
7:36AM |
1 |
SIP expiry time |
7:20AM |
3 |
Advantage of IAX2 to SIP? |
7:00AM |
2 |
PRI litmus test |
6:59AM |
4 |
Asterisk without D-Channel possible? |
6:56AM |
1 |
Polycom IP 600 status setting in Asterisk |
6:52AM |
0 |
(no subject) |
6:43AM |
3 |
Asterisk + AS5300 |
6:31AM |
1 |
SPA-3000 and distinctive ring |
6:19AM |
3 |
Japanese FXO card |
6:17AM |
2 |
Sip no voice |
4:22AM |
0 |
sipgate x asterisk: problems to receive PSTN calls? |
4:19AM |
1 |
pre-installation jitters |
3:42AM |
4 |
software phones for Asterisk - is there a list? |
3:33AM |
0 |
Unable to open pseudo channel for timing... Sound may be choppy |
3:24AM |
4 |
Unable to open IAX timing interface: No such file or directory |
2:26AM |
1 |
Time announcement |
2:15AM |
0 |
ip2ip 302 response |
1:56AM |
6 |
Avoided deadlock |
12:55AM |
0 |
extension and PSTN connection |
12:51AM |
1 |
CVS-HEAD breaks iconnect |