| Sunday October 31 2004 |
| Time | Replies | Subject |
|---|
| 11:42PM |
9 |
Linux and Windows |
| 11:09PM |
1 |
Inbound numbers question |
| 9:53PM |
1 |
record all calls |
| 9:36PM |
4 |
Amount of time asterisk take to pickup incoming call on ZAP interface |
| 7:41PM |
1 |
iax2_read: I should never be called! |
| 7:35PM |
2 |
goto() results in invalid extension |
| 7:30PM |
3 |
Embedded Asterisk Paper Complete |
| 6:21PM |
1 |
VoiceXML / Asterisk |
| 6:19PM |
0 |
Tool for viewing Message waiting status |
| 3:57PM |
0 |
UK Asterisk Consultant visiting San Diego |
| 3:42PM |
2 |
Asterisk and GnuGK on the same box? |
| 3:36PM |
2 |
UDP Fragmentation Problem |
| 2:21PM |
1 |
ISDN card advise |
| 2:17PM |
0 |
pri usage |
| 1:56PM |
3 |
I need help |
| 12:01PM |
1 |
Dialogic |
| 11:47AM |
0 |
norwegian sounds for Asterisk |
| 10:01AM |
1 |
Zapateller broken in ver 1.0.2? |
| 8:25AM |
2 |
ISDN CARD |
| 4:04AM |
1 |
Can't install the mfcr2 support correctly |
| 2:56AM |
0 |
make transfert and hold with FXS device |
| 2:45AM |
2 |
G.711alaw to iLBC |
| 12:18AM |
1 |
asterisk RELOAD option stability |
| 12:09AM |
4 |
video conferencing with sip |
| |
| Saturday October 30 2004 |
| Time | Replies | Subject |
|---|
| 11:13PM |
3 |
Cannot start asterisk - CAPI issues |
| 5:21PM |
0 |
echo with long distance |
| 5:01PM |
0 |
voice delay with isdn |
| 3:08PM |
1 |
IAX2 bandwidth efficiency calculations from Farfon |
| 1:37PM |
7 |
How far is IAX to be a Standard |
| 10:28AM |
1 |
Wireless phones connected to VOIP DECT basestation |
| 8:53AM |
2 |
confusing info from Digium and asteriskdoc about setup of TDM11B |
| 8:51AM |
0 |
Latency/delay on IN1002 - PA1688 phone |
| 8:44AM |
1 |
re: asterisk SER and grandstream |
| 8:35AM |
1 |
DTMF and codec |
| 7:25AM |
1 |
iax registration & port number |
| 7:22AM |
0 |
SIP to SIP echo problem |
| 6:24AM |
3 |
loss concealment |
| 6:16AM |
0 |
Dialogic Card + TP100B |
| 4:46AM |
0 |
Ang: FXO flash from sip phone |
| 1:50AM |
0 |
FXO flash from sip phone |
| 1:47AM |
3 |
HELP: problem making calls from legacy pbx to cisco sip phone via asterisk |
| 12:37AM |
2 |
Wireless phones connected to VOIP DECT base station |
| 12:08AM |
0 |
g723 in pass-thru mode asterisk |
| |
| Friday October 29 2004 |
| Time | Replies | Subject |
|---|
| 10:00PM |
1 |
This is VERY interesting -- A gateway between proprietary digital sets and SIP? |
| 7:11PM |
0 |
Asterisk works with SER |
| 6:00PM |
0 |
E1/R2 application in Brazil: Asterisk compilation with libunicall |
| 5:50PM |
2 |
Swissvoice IP10S opinions? |
| 3:55PM |
0 |
TDM channel shows Offhook when I plug it to thetelco |
| 3:47PM |
8 |
high-capacity systems / trouble with Tyan |
| 2:34PM |
0 |
My asterisk box is behaving funny! |
| 2:14PM |
1 |
Polycom failed registration - Cant figure out whats wrong |
| 1:56PM |
2 |
Rewriting a telephone number for remote dial out |
| 1:17PM |
6 |
non blind call transfers |
| 1:10PM |
1 |
FW: VoiceEclipse vePipe inbound config question - Authorization failed for user"####" |
| 1:09PM |
1 |
DISA() anyone? |
| 12:55PM |
1 |
Cisco PRI Gateway Problems |
| 12:55PM |
0 |
SIP Friends w/ MySQL |
| 12:51PM |
1 |
Suggestion re: SIP/NAT/* |
| 12:39PM |
2 |
Polycom IP 500 Config Files - searching |
| 12:23PM |
1 |
queue_log analyzer |
| 11:39AM |
1 |
Is NuFone messing up for anybody else? |
| 11:22AM |
1 |
Newbie question - pickup call waiting on an analog trunk |
| 11:16AM |
1 |
Ambient MD 3200+incoming problem |
| 11:13AM |
3 |
Anyone using Voipjet? |
| 10:59AM |
1 |
"Hiss" on Line, No ringing thru VoicePulse? |
| 10:51AM |
0 |
sip phones... |
| 10:06AM |
0 |
VoiceEclipse vePipe inbound config question - Authorization failed for user"####" |
| 9:32AM |
3 |
Snom 190/220 |
| 9:28AM |
2 |
AddQueueMember and call distribution |
| 9:06AM |
2 |
Outbound IAX calls stop ringing remote phone, yet can still pick up |
| 8:54AM |
0 |
Suggestion re: SIP/NAT/* |
| 8:07AM |
0 |
Re: Asterisk-Users Digest, Vol 3, Issue 410 |
| 7:37AM |
0 |
Nothing but static on new install with TDM11B |
| 7:18AM |
1 |
wake-up |
| 7:13AM |
0 |
voicemail transfer on busy fails |
| 7:10AM |
0 |
Security question (permissions) |
| 7:02AM |
1 |
Grandstream HT486 and FAX |
| 6:48AM |
1 |
Queue.conf, maxlen = 5 , but what happens with the 6. caller ? |
| 6:38AM |
1 |
sip <-> h323 audio problem |
| 6:20AM |
1 |
FOP 0.17 - Agent setup |
| 5:41AM |
1 |
ISDN EDSS1 protocol support |
| 4:58AM |
1 |
pbx_loopback.so failed |
| 4:35AM |
2 |
eyebeam video |
| 4:09AM |
0 |
sip.conf registration |
| 3:52AM |
0 |
Asterisk Sipphone |
| 3:49AM |
1 |
chan_sccp and Cisco 7940 |
| 3:33AM |
1 |
Echo in CAPI channels |
| 3:28AM |
0 |
Automatic codec selection |
| 2:49AM |
0 |
call another server |
| 2:38AM |
1 |
Asterisk with Nortel BCM |
| 1:37AM |
1 |
Modifying CDR data? |
| 12:49AM |
0 |
Problem with Dial (in v.1.0.2) |
| |
| Thursday October 28 2004 |
| Time | Replies | Subject |
|---|
| 11:54PM |
1 |
Dropped call |
| 11:48PM |
1 |
Do I *need* to compile zaptel? |
| 11:21PM |
0 |
Mysql support |
| 11:12PM |
1 |
Snom200 strange sound problem |
| 7:25PM |
1 |
disa hangs up on me |
| 7:05PM |
2 |
question about asterisk |
| 5:11PM |
1 |
Analog answering machine hangs up early |
| 4:05PM |
1 |
E100P Call Deflection - Redirecting an Incoming Call with ISDN (Resend) |
| 3:19PM |
1 |
Automatic code selection |
| 1:47PM |
0 |
SMDI and Asterisk |
| 12:52PM |
0 |
729 -> 711 failover? |
| 12:03PM |
0 |
New astGUIclient version released 1.0.5 |
| 11:42AM |
1 |
* connectionto home automation server |
| 11:13AM |
0 |
Suggestion re: SIP/NAT/* |
| 11:09AM |
5 |
Queue question |
| 10:49AM |
0 |
Registration Fail |
| 9:26AM |
1 |
Multiple Bandwidth Providers and Asterisk |
| 9:16AM |
0 |
Need Asterisk to generate ringing tone on inbound SIP calls |
| 8:51AM |
1 |
MFC/R2 Argentina variant ANI problems |
| 8:08AM |
0 |
how-to invoke the "Busy" voice mailbox menu in Asterisk |
| 7:55AM |
2 |
TDM400P hardware problems |
| 7:28AM |
0 |
Re: call progress - what are the sticking po ints? |
| 6:42AM |
0 |
Sipura 3000 tone table settings for Australia |
| 6:27AM |
0 |
Ex-girlfriend-logic |
| 5:00AM |
0 |
carrier deployment of SIP |
| 3:58AM |
0 |
mcedit syntax for asterisk conf files |
| 3:31AM |
1 |
disable second call / call waiting via SIP |
| 3:20AM |
2 |
Polycom IP 500 and DTMF |
| 3:16AM |
1 |
Using AVM C4 with fewer than four lines? |
| 2:24AM |
1 |
AW: AW: Firefly 1.9.6 released |
| 2:20AM |
0 |
RE: Why I can't hear anything from my sjphone asanh323 endpoint? |
| 2:08AM |
0 |
Getting result codes of SIP-dials |
| 1:26AM |
3 |
HiPath Wild Card T110P interface |
| 1:25AM |
0 |
integrating Asterisk to existing TDM-based PBX |
| 1:02AM |
0 |
Why I can't hear anything from my sjphone as anh323 endpoint? |
| 12:39AM |
1 |
Nightmare on disconnecting Zap and SIP channel |
| 12:08AM |
2 |
ISDN-Problem with Quadbri behind Tenovis |
| |
| Wednesday October 27 2004 |
| Time | Replies | Subject |
|---|
| 9:51PM |
2 |
Problem with AstTapi |
| 9:07PM |
2 |
No dial tone from fxs port |
| 8:38PM |
4 |
WRT54GS zaptel timing device |
| 8:15PM |
2 |
[PATCH] DUNDi for 1.0.2 |
| 7:41PM |
1 |
Why I can't hear anything from my sjphone as an h323 endpoint? |
| 7:27PM |
2 |
call progress - what are the sticking points? |
| 6:47PM |
0 |
IAX support added to AMP |
| 6:39PM |
1 |
New Strategy in App_queue |
| 5:49PM |
0 |
AudioCodes MP-108 (or MP-1xx) FXO gateway |
| 5:44PM |
1 |
Asterisk to Asterisk using SIP? |
| 5:03PM |
2 |
Asterisk-cvs does not compile on Red Hat 9 |
| 3:54PM |
1 |
where do i find openssl-devel to mandrake 10.1 |
| 3:18PM |
3 |
Type of T1 for T100P card |
| 2:05PM |
1 |
IAXy Call Waiting Disable |
| 1:59PM |
1 |
SRV lookup fails on dyndns wildcard domains |
| 1:50PM |
0 |
G.72[69] |
| 1:46PM |
0 |
Asterisk - Store and Forward Configuration with re-recording some part |
| 1:32PM |
4 |
Can bad person with SIPp attack Asterisk ? |
| 1:19PM |
2 |
Transfer caller |
| 1:13PM |
1 |
SIP vs MGCP |
| 12:39PM |
4 |
Motorola Vt1000 |
| 12:13PM |
1 |
OT: The ideal switch for VOIP |
| 12:05PM |
3 |
Funny thing with LinkSys / IAX2 |
| 11:54AM |
0 |
Call Waiting Via Sipura to X100P |
| 11:42AM |
4 |
AT&T Cordless VOIP Phone? |
| 11:29AM |
1 |
Zap issues... |
| 11:28AM |
0 |
[OT] How trustworthy is Yoda Communications in Taiwan? |
| 10:30AM |
1 |
Grandstream and CallerID - sorted |
| 10:27AM |
1 |
Sparco Office Supplies... |
| 10:26AM |
3 |
GSM Audio Files on Windows w/o Quicktime |
| 10:08AM |
0 |
Need help with extconfig (take 2) |
| 9:52AM |
0 |
neg txgain makes * oblivious to incoming calls? |
| 9:48AM |
0 |
Help needed with Extconfig, mysql |
| 9:27AM |
0 |
chan_sip2 won't compile |
| 9:10AM |
1 |
Zaptel channels |
| 9:09AM |
0 |
app_valetparking |
| 9:01AM |
3 |
RE: [OT] Sparco Office Supplies... (yeah right) |
| 8:55AM |
1 |
Remote Voicemail |
| 8:33AM |
0 |
How to install wakeup? |
| 8:31AM |
0 |
Agent Groups In Queues |
| 8:25AM |
1 |
Simple Asterisk Config Help withx100p |
| 8:14AM |
5 |
[OT] Sparco Office Supplies... |
| 7:58AM |
1 |
Multiple SIP gateway accounts |
| 7:33AM |
1 |
Directory () Problem --revisited |
| 7:27AM |
1 |
can't run ztcfg |
| 6:35AM |
0 |
RTP port mismatch and Astwind card support |
| 6:18AM |
0 |
Cannot Call IAX Softphones |
| 5:49AM |
1 |
SIP-DTMF |
| 5:43AM |
1 |
OT COs/Providers Cannot Reach Others |
| 5:41AM |
0 |
Error when starting Asterisk (Loading module chan_capi.so failed!) |
| 5:01AM |
1 |
pickupgroup and callgroup on zapata.conf , how they work ? |
| 4:49AM |
0 |
zaphfc bristuff ISDN transfer |
| 4:05AM |
0 |
RTP Ports mismatch & Astwind modem support |
| 4:04AM |
1 |
queue reports? |
| 2:40AM |
2 |
AW: Firefly 1.9.6 released |
| 2:28AM |
2 |
UK CallerID |
| 2:27AM |
3 |
TDM400P - TE405P- configuration issue |
| 1:51AM |
3 |
test telephone numbers |
| 1:43AM |
0 |
BillSec and CLID in CDR Problem |
| 12:16AM |
1 |
Firefly 1.9.6 released |
| |
| Tuesday October 26 2004 |
| Time | Replies | Subject |
|---|
| 9:55PM |
1 |
Asterisk Intro for newbies |
| 8:35PM |
2 |
New card - TE110P? |
| 7:39PM |
2 |
Can I pick up a phone that rings from my phone? |
| 6:56PM |
0 |
queues.conf - Agent groups specified with : not @, difference in working? |
| 6:51PM |
1 |
Anyone playing with E1 channel bank? |
| 6:50PM |
1 |
Zaptel FXO channel picked up too early => no audio |
| 6:38PM |
1 |
Asterisk 1.0.2 (again) |
| 6:22PM |
2 |
Performance (Cisco AS5350) or Price (Wildcard TE410P) |
| 6:20PM |
2 |
Digits being lost going out POTS line? |
| 6:14PM |
0 |
torisa startup troubles |
| 6:13PM |
1 |
E100P Call Deflection - Redirecting an Incoming Call |
| 5:05PM |
1 |
Re: Asterisk and Broadvoice, no incoming voice (Brian Weaver) |
| 4:42PM |
2 |
Should I be worried? Newbie Warning |
| 4:33PM |
0 |
Phone Cutout Problems |
| 4:04PM |
0 |
feature request - sipfriends, iaxfriends... |
| 3:44PM |
1 |
7912G Ringers? |
| 3:23PM |
6 |
voicemail.conf |
| 2:56PM |
1 |
Dial timeout off by factor of two? |
| 2:40PM |
0 |
Determining that call was transferred |
| 2:40PM |
7 |
SUSE 9.1 and Zaptel |
| 2:32PM |
0 |
extensions.conf question |
| 2:24PM |
4 |
polycom IP 500/600 |
| 2:14PM |
1 |
ASTCC no sound |
| 2:02PM |
2 |
Bandwidth Load Balancing / Dundi |
| 1:31PM |
2 |
Uniden UIP 200 not ringing |
| 1:28PM |
0 |
extension.conf on mysql |
| 1:11PM |
0 |
Graphing Date/Source/Destination nicely |
| 12:59PM |
1 |
De-Centralized / Distributed Conferencing App |
| 11:10AM |
0 |
outgoing spool dial local channels then actual dial from extensions.conf |
| 11:03AM |
0 |
Problems with Directory() |
| 11:02AM |
2 |
Asterisk + MGCP + Cisco E1 gateway |
| 10:11AM |
5 |
Gentoo |
| 9:56AM |
2 |
Begin to begin |
| 9:48AM |
4 |
ASTCC with password |
| 9:47AM |
3 |
Problem getting zaprtc installed on a mandrake 9.2 |
| 9:38AM |
1 |
Can't connect even though its running.. |
| 9:05AM |
4 |
G.726 |
| 8:41AM |
1 |
Succesul outgoin calls using UniCall |
| 8:40AM |
0 |
Asterisk as a simple Message Store and Forward that Sends VoiceMail to a Group. |
| 8:25AM |
0 |
Where to catch events like Dial, Ringing, Transfer, Hold, Forward, Hangup, Park, UnHold, Answered, No Answer, etc. |
| 7:56AM |
6 |
Asterisk 1.0.2 |
| 7:10AM |
1 |
SIP Conferencing Server |
| 5:56AM |
3 |
HANGUPCAUSE macro.. |
| 5:42AM |
2 |
RDNIS |
| 5:19AM |
2 |
ASTERISK and VoiceXML |
| 5:14AM |
0 |
R: E1 configuration problem |
| 5:02AM |
3 |
cisco router & * |
| 4:59AM |
8 |
Need HELP to put * in use for good cause |
| 4:06AM |
0 |
Snom 200 SIP Settings etc |
| 3:39AM |
0 |
Once again: Problem compiling ZPAHFC with Suse 9.1, Kernel 2.6.5.7-111 |
| 3:38AM |
2 |
E1 configuration problem |
| 3:35AM |
5 |
X100P noise on ADSL line. |
| 2:52AM |
7 |
H323.conf question |
| 2:32AM |
1 |
Little problem with AGI |
| 2:23AM |
2 |
IAX trunking clarification... |
| 2:04AM |
3 |
Can contexts have wildcards too? |
| 1:17AM |
1 |
snom200 & dial plan |
| 1:15AM |
0 |
PRI events |
| 1:14AM |
2 |
sip_xmit errors... |
| |
| Monday October 25 2004 |
| Time | Replies | Subject |
|---|
| 9:12PM |
2 |
GPL thoughts |
| 7:08PM |
1 |
Re: [Asterisk-Dev] How to submit a patch? |
| 6:43PM |
4 |
Routing based on Caller ID |
| 5:38PM |
1 |
Re: Benjk's Question "Why FXS" |
| 4:15PM |
3 |
Re: Benjk's Question "Why FXS" |
| 3:56PM |
3 |
Agents allowed to transfer but * just hangs up! |
| 3:47PM |
2 |
Snom Phones and asterisk |
| 3:34PM |
1 |
Re: Benjk's Question "Why FXS" |
| 2:56PM |
0 |
Santa Cruz, Bolivia? |
| 2:26PM |
1 |
Company Directory/Dial by Name |
| 1:48PM |
1 |
GUI for Asterisk. |
| 1:30PM |
1 |
Auto-Login/Auto Answer |
| 1:11PM |
2 |
Transfering Calls |
| 1:04PM |
0 |
using @ vs : for Agent groups.... |
| 12:54PM |
0 |
Augh! |
| 12:52PM |
1 |
SNOM 190 - strange voice problems |
| 12:49PM |
1 |
G729 Error. => No path translation. |
| 12:10PM |
2 |
Multiple Accounts on a Softphone |
| 11:49AM |
0 |
Softphone for QNX? |
| 11:48AM |
1 |
Call Parking + Agents (or queues?) does not work |
| 11:36AM |
1 |
echo questions |
| 11:32AM |
2 |
Error starting Asterisk. |
| 11:00AM |
2 |
Can home/office have same extension |
| 10:54AM |
1 |
Setup two Asterisk servers with MGCP |
| 10:24AM |
1 |
Rhino channel bank configuration with T100P |
| 9:54AM |
5 |
Nortel Phones. |
| 9:40AM |
2 |
Help Instalation |
| 9:20AM |
0 |
voicemail: fromstring and delete |
| 8:53AM |
0 |
Queue anounce time |
| 8:46AM |
1 |
ATCOM froze |
| 8:20AM |
1 |
SayNumber application - in spanish? |
| 7:58AM |
3 |
Multi-office topology suggestions |
| 7:53AM |
0 |
DNID in chan_sip.c |
| 7:50AM |
0 |
Digium TheVoice recordings' sound |
| 6:53AM |
3 |
Vonage Softphone--outbound calls work, inbound do not |
| 6:46AM |
1 |
sip.conf user withdefaultip=....worksbutcalleridnot settable (= ip) |
| 6:46AM |
2 |
Help for Newbie? |
| 6:44AM |
1 |
Problem with asterisk-oh323 |
| 6:36AM |
2 |
Digium Wildcard T1 Compatibility (ethernet f or T1 cables) |
| 6:25AM |
3 |
sip.conf user with defaultip=....worksbutcallerid not settable (= ip) |
| 6:06AM |
1 |
sip.conf user with defaultip= .... worksbutcallerid not settable (= ip) |
| 5:43AM |
1 |
Fwd: IAX wireless problem |
| 5:33AM |
1 |
sip.conf user with defaultip= .... works but callerid not settable (= ip) |
| 3:12AM |
1 |
Bandwdith usage |
| 3:03AM |
2 |
AST doesn't start after update from 0.5 to 1.0 |
| 2:50AM |
1 |
3com with Asterisk |
| 2:47AM |
2 |
protection |
| 2:28AM |
1 |
Quintum A800 OH323 problem |
| 2:23AM |
1 |
sip users registering fails |
| 2:12AM |
2 |
CDR Dokumentation |
| 1:08AM |
1 |
I have Asterisk & Hylafax on a server. What else do I need...? |
| |
| Sunday October 24 2004 |
| Time | Replies | Subject |
|---|
| 10:57PM |
3 |
Snom200 & VMail (MWI) |
| 9:49PM |
1 |
G729 -> G723.1 |
| 9:43PM |
0 |
Xlite works, Asterisk sometimes not |
| 7:02PM |
1 |
Several FXS Ports |
| 6:48PM |
2 |
Howto get voicemail $VM_ vars into externnotify script? |
| 5:11PM |
6 |
ACT Gateways |
| 5:01PM |
1 |
Connection to a H323 system |
| 1:24PM |
1 |
Iaxy authentication |
| 12:20PM |
0 |
Reload cause Sound Volumn becomes very loud |
| 12:11PM |
0 |
Asterisk Prepaid with MySQL |
| 11:50AM |
1 |
getting cid from spa3k pstn to * |
| 8:11AM |
1 |
Failed to authenticate on INVITE to '"601" ... |
| 7:08AM |
0 |
(iax|sip)friends in extconfig? |
| 6:56AM |
0 |
Error when compiling asterisk-oh323 |
| 6:44AM |
3 |
random crash at startup |
| 6:18AM |
0 |
How to create Groups/members and do Conferencing? |
| 5:20AM |
0 |
Problem compiling ZPAHFC with Suse 9.1, Kernel 2.6.5 |
| 5:14AM |
3 |
chan_sip CallerPres support? |
| 3:37AM |
6 |
Digium TheVoice recordings' sound terrible |
| 12:59AM |
0 |
bristtuff segfault |
| |
| Saturday October 23 2004 |
| Time | Replies | Subject |
|---|
| 11:08PM |
3 |
Asterisk, ATA-186 & Sipgate.de / sipgate.co.uk |
| 11:04PM |
3 |
G.729 on YDL and MacOSX |
| 10:42PM |
0 |
One approach to SIP dialing through asterisk |
| 9:53PM |
0 |
Outlook reports internal error after using AstTapi |
| 7:29PM |
1 |
Fedora 2, Kudzu and X100P |
| 7:14PM |
0 |
Hardware (and apple YDL G.729) |
| 4:46PM |
5 |
Hardware |
| 2:32PM |
7 |
Asterisk and Broadvoice, no incoming voice |
| 2:31PM |
1 |
Re: Webmin for ASTERISK and QOS and call quality |
| 2:18PM |
4 |
Geotel integration with Asterisk |
| 10:35AM |
1 |
Zultys Zip 2 Setup |
| 9:46AM |
4 |
doublehash patch for 1.0.1 |
| 9:37AM |
1 |
Cheap hosted servers and Asterisk |
| 9:09AM |
0 |
Quintum ASM400, ASM200 and ASTERISK |
| 9:07AM |
0 |
Digum board TDM to Phonejack --Quicknet --Trandsfering calls. |
| 8:54AM |
1 |
Support for reception of "send url" in SIP clients needed |
| 8:54AM |
0 |
Need help with RDNIS on ISDN PRI |
| 8:10AM |
1 |
IAX wireless problem |
| 2:19AM |
0 |
* dies with QuadBRI |
| 2:06AM |
6 |
iLBC/PCM16 Huge Cost |
| |
| Friday October 22 2004 |
| Time | Replies | Subject |
|---|
| 9:09PM |
1 |
spa3k: cid vs authid |
| 8:29PM |
6 |
chan_sip changes affecting ACK? - Bug? |
| 7:24PM |
1 |
new quad T1 install |
| 7:08PM |
1 |
Asterisk-OH323 Invalid format RTP |
| 5:28PM |
0 |
iaxComm now supports iLBC,Speex |
| 3:06PM |
0 |
(no subject) |
| 2:39PM |
4 |
Direct SIP connection to Vonage service |
| 2:12PM |
1 |
Cannot send # to far end, asterisk intercepts. |
| 2:02PM |
3 |
res_config |
| 1:45PM |
0 |
"zt_get_index: nullok is not asserted" could led to freeze? |
| 12:10PM |
1 |
testing open ports 10000 - 20000 |
| 12:02PM |
3 |
DTMF G729 |
| 10:52AM |
3 |
One E1: 10 time-slots for voice (ZAP), 10 for Internet PPP (data) and 10 slots for Internet PPP (VoIP) |
| 10:43AM |
0 |
Best way to transfer incoming sip calls to other sip number? |
| 9:26AM |
1 |
How useful is the screen on IP phones? |
| 9:06AM |
1 |
Fw: SPA-3000 Disconnect tone detection in France ? |
| 8:22AM |
2 |
Queue / Agent Problem |
| 8:06AM |
1 |
Updates in the asterisk - cvs mailing list - Head or Stable? |
| 8:01AM |
0 |
Detecting Busy when dialing out on ZAP channel. |
| 7:28AM |
1 |
installing install isdn4k-utils from source ? |
| 7:19AM |
6 |
Hardware Recommendations |
| 7:15AM |
1 |
New. Testing? |
| 5:26AM |
1 |
IAXy echo avoidance/cancellation |
| 4:01AM |
10 |
Question about ISDN reason codes |
| 2:56AM |
1 |
MusicOnHold() - how to restart player from the beginning on each call? (fwd) |
| 1:57AM |
1 |
common numbers ? |
| 1:46AM |
1 |
Newbie General questions |
| |
| Thursday October 21 2004 |
| Time | Replies | Subject |
|---|
| 11:55PM |
1 |
827-4V voice ports, asterisk and hookflash |
| 11:48PM |
0 |
problem with caller-id |
| 10:52PM |
2 |
Connecting to Commander NT132 |
| 7:50PM |
3 |
Can I do that? |
| 7:37PM |
1 |
AGI comand channel status] |
| 7:31PM |
3 |
* and Verisign SIP-7 service |
| 6:04PM |
0 |
Anyone getting RDNIS on Lucent 5ESS |
| 5:48PM |
0 |
Help with asterisk-oh323 driver [resend] |
| 2:49PM |
7 |
Digium Wildcard T1 Compatibility |
| 2:49PM |
4 |
Fax detection in voip channel |
| 1:22PM |
1 |
automatically logging on/off agents |
| 12:31PM |
0 |
question about type=user in sip.conf |
| 12:20PM |
0 |
backgrounddetect command - what about busy |
| 11:53AM |
0 |
doubts regarding monitor command |
| 10:47AM |
6 |
answer on # key? |
| 10:35AM |
1 |
KSS/BLF on Asterisk |
| 10:26AM |
1 |
Load test IAX |
| 10:19AM |
1 |
sip call echo cancellation |
| 10:17AM |
4 |
modem question |
| 9:34AM |
4 |
Grandstream Flashing (different issue) |
| 9:27AM |
0 |
Polycom IP600 features |
| 9:26AM |
0 |
Local battery phones and Asterisk |
| 9:16AM |
0 |
Yoda SIP Devices: IAD100, IAD200, IAD211, IAD400 and other |
| 9:05AM |
1 |
Freshmaker failed register test |
| 8:59AM |
5 |
SER or not to SER? |
| 8:43AM |
1 |
AGI comand channel status |
| 8:39AM |
0 |
Queues Problems |
| 8:37AM |
0 |
Video Phone issues registering with asterisk |
| 8:36AM |
1 |
SIP - H.323 connection |
| 8:18AM |
0 |
gotoif regex? |
| 7:58AM |
1 |
MWI - Sip phones |
| 7:54AM |
0 |
SIP / H323 connectivity |
| 7:41AM |
2 |
Voicemail and ast_data |
| 7:29AM |
2 |
Asterisk and Nortel Meridian interconnection |
| 7:09AM |
1 |
first tries ! |
| 7:02AM |
2 |
Press the * key to repeat |
| 6:06AM |
2 |
asterisk & ipv6 |
| 5:51AM |
0 |
sip+iax+firewall |
| 5:37AM |
1 |
Performance with ASTCC. |
| 5:34AM |
7 |
G.729 licensing/patent? |
| 5:32AM |
1 |
beginners questions |
| 5:06AM |
2 |
Does use of riser cards in racks affects performance of PCI telephony cards ? |
| 4:06AM |
1 |
Calling IAX client behind NAT |
| 4:05AM |
0 |
Voicemail: Unable to open digits/hundred M |
| 3:37AM |
1 |
Manager API / Agents |
| 3:17AM |
0 |
Special Callback feature |
| 3:03AM |
0 |
Clients can login twice |
| 2:19AM |
2 |
2 * RxFax <-> TxFax |
| 1:28AM |
0 |
CISCO router closes the connection before starting conversation |
| 12:18AM |
0 |
SER + Asterisk Attended Call Transfer |
| 12:15AM |
0 |
"Number" of caller |
| |
| Wednesday October 20 2004 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
Do any one have developed Asterisk ebuild for Gentoo |
| 9:35PM |
1 |
Flash Panel version greif with ming et al |
| 8:30PM |
0 |
Routing calls based on monthly usage? |
| 7:57PM |
11 |
grandstream 102 flashing |
| 7:29PM |
1 |
Help with asterisk-oh323 driver |
| 6:58PM |
0 |
Delay in outbound SIP call |
| 5:24PM |
1 |
H323 Connection to Splicecom Maximiser |
| 4:35PM |
0 |
codec problems with astcc and not with sip trhough aix |
| 4:20PM |
0 |
how to detect a busy line using analog ports TDM04B (station ports) and using outgoing spool to start the call |
| 3:57PM |
0 |
Grandstream phone - no dialtone |
| 3:27PM |
2 |
IP Phones -India |
| 3:23PM |
0 |
Received bad packet with bad udp checksum. |
| 2:40PM |
1 |
contexts based on time and date |
| 2:34PM |
0 |
Dialogic and TP card |
| 2:10PM |
5 |
app_conference |
| 12:26PM |
3 |
Newbie with new Project VOIp |
| 11:34AM |
1 |
Manger API flag from dialplan |
| 10:58AM |
4 |
IP Phone that OFFICIALLY support Asterisk |
| 10:41AM |
0 |
RE: Asterisk on a mid-sized flat corporate |
| 10:36AM |
3 |
ASTCC newbie |
| 10:14AM |
4 |
Graceful CLI/crontab reboot |
| 9:48AM |
2 |
X100P make phone ring on incoming sip call - possible? |
| 9:37AM |
2 |
Snom 190 "VMail Soft Key" |
| 9:36AM |
2 |
FWD via IAX2 -- anybody else experiencing timeouts? |
| 8:48AM |
2 |
manager interface to barge |
| 8:38AM |
0 |
Wildcard X100P/India |
| 8:10AM |
0 |
Trunking Gateway with E100P |
| 8:05AM |
0 |
RE: Asterisk-Users Digest, Vol 3, Issue 264 |
| 7:43AM |
0 |
Personal Phone Gateway PCI and USB Phone.- |
| 7:05AM |
1 |
[OT] GSM patents |
| 6:53AM |
4 |
cannot call Grandstream |
| 6:45AM |
0 |
still riniging problem |
| 6:30AM |
3 |
grandstream handytone 286 problem |
| 5:37AM |
5 |
New Channel Driver: chan_bluetooth |
| 5:14AM |
5 |
Samsung DCS70 PABX |
| 4:53AM |
0 |
2 analog phone on FXS ports? |
| 4:33AM |
0 |
octoBRI problem |
| 4:31AM |
3 |
Attempt at country tones |
| 4:03AM |
3 |
X100P problems / UK Supplier of TDM400P FXO cards |
| 3:36AM |
0 |
SER problem? |
| 3:06AM |
1 |
SIP/SIMPLE, Jabber and Asterisk |
| 2:21AM |
0 |
Meetme room calls quiet for some lines/callers |
| 1:08AM |
3 |
cheap gig switch? smc, netgear, or 3com? |
| 12:51AM |
1 |
Load Balaning on 2 E100P cards |
| 12:43AM |
2 |
chan_mISDN problem |
| |
| Tuesday October 19 2004 |
| Time | Replies | Subject |
|---|
| 11:42PM |
0 |
AW: CAPI and Asterisk (with AVM ISDN Card) |
| 11:28PM |
2 |
meetme latency |
| 9:30PM |
4 |
Sipura or X100P Option |
| 8:38PM |
2 |
Comments on proposed * setup |
| 6:48PM |
1 |
Anyone else seeing this? |
| 5:29PM |
0 |
OT: ATA 286 how to make the phone ring |
| 5:25PM |
0 |
7920 Help chan_sccp |
| 4:51PM |
2 |
Asterisk not sending full 11 digits dialed.... |
| 4:17PM |
1 |
new here : logic of ser and asterisk all confused---longish |
| 3:30PM |
1 |
CAPI and Asterisk (with AVM ISDN Card) |
| 2:23PM |
0 |
PSTN -> PRI -> ASTERISK -> ASTERISK -> PRI -> Legacy switch ? |
| 2:22PM |
4 |
i extension |
| 2:04PM |
7 |
DUNDi in stable? (New subject) |
| 1:37PM |
8 |
Wonderful Success with PAP2-NA |
| 12:55PM |
5 |
X100P red alert |
| 12:39PM |
1 |
disabling "comfort noise", other odd thoughts |
| 12:34PM |
1 |
Asterisk on a mid-sized flat corporate network? |
| 12:02PM |
1 |
Some questions about channel banks signalling? |
| 11:56AM |
2 |
DUNDi on Slashdot |
| 11:52AM |
1 |
incorrect context called when receiving call on SIP channel |
| 11:46AM |
0 |
TDMoE Question? |
| 11:23AM |
4 |
Tranferring UniCall lines |
| 11:18AM |
4 |
Vonage with Nat - Working |
| 11:13AM |
4 |
How to ring internal extension? |
| 10:53AM |
0 |
Re: Asterisk-Users Digest, Vol 3, Issue 260 |
| 10:50AM |
6 |
Asterisk on PowerPC v. Intel/AMD |
| 10:10AM |
1 |
Got SIP response 403 "Forbidden (From header is not a Trust host or gateway)" back |
| 9:48AM |
0 |
debug extension matching |
| 9:36AM |
2 |
Fax over IP doesn't works |
| 9:02AM |
0 |
Wellgate SIP product users - voice your concern! |
| 8:56AM |
2 |
chan_mISDN |
| 8:18AM |
13 |
Almost there--Remote connection |
| 8:08AM |
0 |
Problem with portaSIP provider |
| 8:01AM |
3 |
Problem with NFAS trunkgroups |
| 7:52AM |
0 |
Spandsp debug log question |
| 7:47AM |
4 |
test-driving G.729? |
| 7:25AM |
0 |
ExtensionState |
| 7:01AM |
1 |
Transparent SIP Server |
| 6:37AM |
0 |
txgain usage with T100P |
| 5:50AM |
2 |
SPA-3k & * |
| 5:47AM |
5 |
mISDN, CAPI, ISDN ??? |
| 5:31AM |
0 |
Called number Callerid with Sip |
| 5:16AM |
0 |
Planet SIP Phone |
| 4:23AM |
0 |
I can't solve my problems with the IVR |
| 4:15AM |
3 |
Working Asterisk With Vonage |
| 4:01AM |
2 |
Setting CallerID on UK BRI line |
| 3:18AM |
4 |
About Supervised Call Transfert on GS BT100 |
| 3:10AM |
1 |
AW: Follow me using a loop |
| 2:54AM |
1 |
record |
| 2:20AM |
1 |
Snom & Mass Deployment Config Problems |
| 2:12AM |
2 |
Follow me using a loop |
| 1:17AM |
0 |
Problem with DIAL command |
| 1:07AM |
0 |
Voicemail and AutoAttendant for a Nortel Option 11 PBX |
| |
| Monday October 18 2004 |
| Time | Replies | Subject |
|---|
| 11:19PM |
2 |
SIP video support problem |
| 9:16PM |
1 |
ZapRAS from both sides |
| 7:01PM |
2 |
Anybody - please help me with this |
| 6:52PM |
0 |
Speex wideband mode |
| 6:11PM |
5 |
SMTP MTA suggestions. |
| 5:34PM |
1 |
SPAM Notice |
| 5:32PM |
0 |
anyone using a cisco 12sp+ or VIP 30 |
| 5:13PM |
0 |
Cisco 7940 X IAX trunk |
| 4:59PM |
1 |
Specify location of ADSI Softkeys ? |
| 4:31PM |
3 |
verisign immitate e164 |
| 3:56PM |
0 |
Routing over T100p Help Please... |
| 3:20PM |
2 |
IAX2 Nat issue, Any help greatly appreciated |
| 2:52PM |
0 |
SIP calls dropped (Ast 1.0 and Fedora core 2) |
| 2:11PM |
5 |
Quick question regarding daily restart of asterisk |
| 1:49PM |
3 |
VoIP over 1xRTT |
| 1:02PM |
0 |
Where to buy POLYCOM phones (forcing native bridge between SIP terminals) |
| 12:41PM |
0 |
Distorted Ringback |
| 12:39PM |
3 |
GSM to g729 Conversion |
| 12:30PM |
1 |
NMS AG4000? |
| 12:19PM |
0 |
Re: Asterisk-Users Digest, Vol 3, Issue 243 |
| 12:14PM |
2 |
IAX2 Over Satellite => It works ! |
| 12:05PM |
3 |
How to make asterisk send email notification of voicemessages? |
| 11:58AM |
4 |
Asterisk System Management User Interface |
| 11:04AM |
4 |
Where to buy POLYCOM phones? |
| 10:58AM |
1 |
Current Call information? |
| 10:43AM |
5 |
Voicepulse down for anyone else? |
| 10:15AM |
20 |
Polycom phones |
| 9:25AM |
3 |
New Realtime config and MWI |
| 9:11AM |
0 |
Problems with IVR digit recognition |
| 9:04AM |
0 |
Asterisk won't load some type of channel error |
| 8:46AM |
3 |
Call failed to go through |
| 8:39AM |
1 |
MWI for X-Ten Pro? |
| 8:10AM |
1 |
Transfer caller but on no answer, return to transferee... |
| 7:49AM |
0 |
Success with Swissvoice IP10S and SIP? |
| 7:46AM |
0 |
chan_iax2.c:5390 socket_read: Rejected connect attempt from |
| 7:31AM |
0 |
Sond problem on Second ISDN B channel |
| 6:58AM |
1 |
FireFly and GS-BT100 codec negotiation problem |
| 6:31AM |
2 |
Can't compile app_conference |
| 5:44AM |
1 |
Asterisk and video door phones? |
| 5:37AM |
1 |
Polycom IP-XXX with shared registration |
| 5:21AM |
3 |
mysql sipfriends and allowing individual codecs per user? |
| 4:58AM |
1 |
Svar: Re: Where to post SuSE 9.x startup script? |
| 4:35AM |
3 |
Capturing calls in asterisk |
| 3:35AM |
0 |
OH323 VoIP router connect debug question? |
| 1:58AM |
0 |
Xten eyeBeam Video codec |
| 12:08AM |
1 |
ACD/Queue Support with SIP Notification Messages? |
| |
| Sunday October 17 2004 |
| Time | Replies | Subject |
|---|
| 11:59PM |
0 |
Sourcing H/W for Asterisk in India :: Digium/Intel Modems and IP Phones |
| 11:54PM |
0 |
cross-connecting dynamic channels |
| 11:02PM |
0 |
Asterisk AGI 'Get Data' escape digits not working on long records |
| 9:56PM |
0 |
Thailand |
| 9:27PM |
1 |
Problem In RTC Client When Used With Asterisk |
| 9:05PM |
3 |
chan_h323: forcing 20ms packetisation |
| 8:42PM |
2 |
Asterisk dropping last digit of phone number |
| 6:50PM |
0 |
Calling all Users to check out bug 2655 |
| 6:27PM |
1 |
chan_skinny caller id. |
| 5:55PM |
0 |
Fax Redirection |
| 5:02PM |
0 |
chan_skinny callerID usage |
| 4:59PM |
1 |
SIP outbound dialing -- newbie alert. |
| 3:23PM |
0 |
OT - new SPA-3000 firmware out (v2.0.11a) |
| 2:57PM |
1 |
Cisco ATA-186 and Caller ID |
| 2:16PM |
2 |
Anyone else tried Speex 1.1 CVS? |
| 2:04PM |
3 |
DIAX 0.9.9b - now multi codec support |
| 1:41PM |
0 |
Wildcard X100P and Fedora Core 2? |
| 1:14PM |
2 |
Asterisk for a VOIP Provider? |
| 1:12PM |
5 |
can not compile chan_capi 0.3.5 |
| 11:16AM |
0 |
IAX error messages |
| 10:09AM |
0 |
chan_skinny usage of callerid |
| 9:39AM |
2 |
Asked to transmit frame type 64, while native formats is 8 |
| 4:52AM |
1 |
X100P, Dutch analong line, caller-id |
| 2:59AM |
1 |
Automated calling/Bridging and takedown in Asterisk? |
| |
| Saturday October 16 2004 |
| Time | Replies | Subject |
|---|
| 5:32PM |
5 |
* Server behind a firewall - How To |
| 5:17PM |
0 |
Macro exmaple for saying digits in a more natural sounding way. |
| 3:39PM |
9 |
IAXy setup |
| 2:17PM |
1 |
Anyone using stanaphone? Having small problem |
| 1:56PM |
1 |
DTMF tones from CCME phone |
| 12:52PM |
1 |
Unusual protocols |
| 10:05AM |
7 |
FXO vs FXS question |
| 9:02AM |
1 |
Going to voicemail on noanswer |
| 9:01AM |
0 |
OT: Broadcom BCM1160 and BCM4318 released |
| 8:39AM |
4 |
Bandwidth control on a home office network |
| 8:05AM |
1 |
Asterisk Data Configuration Example 1 |
| 7:36AM |
5 |
Sending broadcasts to all phones? |
| 7:35AM |
0 |
RE: G729 and Sipura |
| 6:06AM |
2 |
Zapata on PowerMac G4 |
| 3:39AM |
1 |
sipgate cannot dial out / loop detected |
| 3:27AM |
6 |
G729 and Sipura. |
| 1:47AM |
1 |
Newbie question: asterisk and ser |
| 12:35AM |
1 |
Compatibility of Asterisk With Cisco 5350/5400Gateway |
| 12:04AM |
1 |
Japanese Translation of Asterisk? |
| 12:03AM |
1 |
Compatibility of Asterisk With Cisco 5350/5400 Gateway |
| |
| Friday October 15 2004 |
| Time | Replies | Subject |
|---|
| 11:34PM |
1 |
help , chan_sccp wont build. |
| 9:44PM |
1 |
how can an AGI terminate a "STREAM file command" |
| 6:01PM |
3 |
sccp cisco 12sp HELP !!! |
| 5:49PM |
0 |
New Sipura Phone |
| 3:49PM |
1 |
Finding a gateway for home use (UK) |
| 3:42PM |
0 |
Congratulations to all comunity. Success: iaxy + openvpn + winxp routong. |
| 3:34PM |
1 |
Attempting native bridge ....... |
| 3:32PM |
4 |
Cisco 7960 + 7914 - not worked |
| 2:46PM |
0 |
Looking for supplier for 7912G and 7940Gs reply in private |
| 2:40PM |
1 |
Help with Incoming calls |
| 2:11PM |
0 |
Avoiding deadlock |
| 2:11PM |
0 |
MySQL CDR addon problem |
| 1:49PM |
5 |
Cheap, Highquality IP Phones |
| 12:50PM |
0 |
Should ZAP channels pass CNAM to SIP? |
| 12:46PM |
2 |
HylaFAX v. spandsp |
| 12:46PM |
1 |
Unable to make E&M Wink work with T400P |
| 12:44PM |
2 |
[OT] HylaFAX and DID |
| 12:42PM |
0 |
My macros, etc. |
| 12:39PM |
3 |
Using my GrandStream remotely |
| 12:29PM |
4 |
res_odbc app_realtime |
| 12:04PM |
1 |
calling out from a remote * server |
| 11:40AM |
0 |
New asterisk user question |
| 11:26AM |
1 |
Sample advanced call routing standard extension |
| 11:25AM |
1 |
T100P Frame Errors |
| 11:17AM |
2 |
CHANUNAVAIL = CHANUNAVAIL doesn't eval properly |
| 11:03AM |
0 |
grandstream bt-486 can only dial with # |
| 10:27AM |
1 |
FXS port to use an Analog phone as a door phone. |
| 10:19AM |
0 |
RE: Cannot reach a SIP device (Sudhir Kumar) |
| 9:56AM |
5 |
New Open Source Project: Asterisk Management Portal |
| 9:45AM |
0 |
Cannot reach a SIP device |
| 9:05AM |
5 |
New Project - IP Phone Sources |
| 8:56AM |
0 |
Prepaid vs. Prepaid modified |
| 7:56AM |
2 |
app_queue & manager API |
| 7:49AM |
2 |
Cisco to * problem |
| 7:22AM |
3 |
Always get 401 Unauthorized..that normal? |
| 6:47AM |
0 |
RE: Cisco firewalls and softphones (Matthew Oulton) |
| 6:34AM |
0 |
Manager API and extension s |
| 6:16AM |
1 |
Invalid GSM data |
| 6:05AM |
0 |
Transmit re-INVITE before BYE is sent - why? |
| 5:22AM |
0 |
Looking for recommendations for a low-cost FXO toIP gateway. |
| 5:13AM |
0 |
Problem in DTMF Info message |
| 5:07AM |
1 |
Asterisk crashes on special Transfer with MGCP/ATA 186 |
| 4:51AM |
2 |
SNOM 190 "Dial-Plan String" Settings |
| 4:37AM |
1 |
CID troubles... |
| 4:06AM |
4 |
FireFly w/ SIP |
| 3:20AM |
0 |
SIP <-> Asterisk <-> H323 Gateway |
| 3:10AM |
0 |
Path Replacement |
| 3:08AM |
0 |
Prepaid authentication and accounting using Asterisk |
| 12:23AM |
1 |
Newbie to Asterisk - VoIP end-to-end |
| 12:17AM |
0 |
Cisco firewalls and softphones |
| |
| Thursday October 14 2004 |
| Time | Replies | Subject |
|---|
| 6:41PM |
2 |
Looking for recommendations for a low-cost FXO to IP gateway. |
| 5:49PM |
3 |
GPL Violations (Was: Advice on OS Choice) |
| 5:00PM |
0 |
notransfer=yes |
| 4:35PM |
3 |
Zap Channel wait for # |
| 4:20PM |
1 |
Bug in app_queue/AgentCallbackLogin |
| 4:13PM |
0 |
ODBCexec -Fixed- |
| 4:10PM |
0 |
asterisk seg faults |
| 3:54PM |
0 |
success with SIP on Swissvoice IP10S? |
| 3:18PM |
2 |
Limiting use of an account |
| 3:12PM |
4 |
how can I test canreinvite effectivness? |
| 2:55PM |
0 |
CNAM callerid from a T100p to sip cisco 7960 not working. |
| 2:47PM |
3 |
Dialogic D/300JCT-E1 support |
| 2:36PM |
1 |
Asterisk and Internet Phone/Line Jack |
| 2:35PM |
0 |
ODBCexec |
| 2:14PM |
3 |
MySQL CDR -- debugging |
| 1:25PM |
2 |
ast_data and dialplan in mysql |
| 12:53PM |
2 |
Intercept HOLD of Snom phones |
| 12:44PM |
1 |
IAX UDP packet dropped on incoming call |
| 12:17PM |
10 |
Running Asterisk on Linksys Router |
| 12:09PM |
2 |
FireFly SIP Registration Interval |
| 11:39AM |
0 |
Paging / 79xx cisco |
| 10:46AM |
1 |
Iaxy boot & provide... |
| 10:36AM |
2 |
SPA-2000's rebooting every hour or so... |
| 10:24AM |
1 |
Distinctive Ringing for SipToneII |
| 10:11AM |
2 |
Web stream from an extension? |
| 10:03AM |
1 |
searching for a nifty solution for different outgoing msn depending on the sip-user |
| 9:07AM |
3 |
About 3 Way Calling on GS BT100 |
| 8:19AM |
1 |
Unable To retrieve DTMF tone from INFO message |
| 7:51AM |
0 |
Re: Asterisk-Users Digest, Vol 3, Issue 182 |
| 7:29AM |
0 |
{SPAM?} Asterisk VIA SSH Tunnels |
| 7:11AM |
0 |
RE: Asterisk-Users Digest, Vol 3, Issue 185 |
| 7:10AM |
8 |
(Another) Queue log analyser |
| 7:06AM |
0 |
authentification for H323 users |
| 6:46AM |
0 |
Xten eyeBeam Video |
| 6:35AM |
1 |
Hardware for 20 extensions (voip vs analog)? |
| 6:10AM |
11 |
Advice on OS Choice |
| 5:41AM |
0 |
no voice getting through |
| 5:19AM |
6 |
Configuring DIAX |
| 3:40AM |
1 |
cdr Logging - Postgresql |
| 2:28AM |
1 |
transfer call ? |
| 2:22AM |
1 |
Memory stuff |
| 2:06AM |
0 |
incoming ringsound |
| 12:38AM |
1 |
Call waiting trouble with 7912 cisco phones |
| 12:25AM |
0 |
[SIP] limiting the number of concurrent connections? |
| |
| Wednesday October 13 2004 |
| Time | Replies | Subject |
|---|
| 9:52PM |
0 |
remote asterisk cannot register thru iax |
| 8:10PM |
1 |
3 way calling feature |
| 8:05PM |
0 |
New rh9/FC1 RPMS - v1.0.1 and CVS-10.11.04 |
| 7:46PM |
1 |
Asterisk Post Paid Application |
| 7:39PM |
1 |
Embedded Asterisk System |
| 6:27PM |
2 |
No audio on incoming IAX calls |
| 6:23PM |
2 |
Cisco FXO |
| 6:08PM |
5 |
restricting access to outside calls |
| 5:10PM |
2 |
Uniden UIP200 Call Waiting Hold |
| 3:16PM |
2 |
SNOM 190: Good or crappy |
| 3:02PM |
2 |
Using Lucent/Ascend TNT as a PSTN Gateway? |
| 2:13PM |
1 |
spandsp-0.0.2 configure problem |
| 1:45PM |
1 |
Does Asterisk supports Sip Info Method? |
| 1:17PM |
1 |
TE405P and TE410P performance difference |
| 1:06PM |
0 |
TTS & Voice Rec (sphinx) |
| 12:50PM |
1 |
*8 on voicetronix OS12 |
| 12:47PM |
3 |
Least Cost Routing |
| 12:37PM |
2 |
OpenSwitch12 install problems |
| 11:53AM |
4 |
DND on SIP |
| 10:31AM |
2 |
G.726/16kbps and Asterisk.! |
| 10:26AM |
2 |
quiet term |
| 9:59AM |
0 |
Asterisk answering sipphone.com calls - stupid newbie question |
| 9:13AM |
0 |
Pb H323 Connexion |
| 8:58AM |
0 |
Authenticate cmd with db |
| 8:46AM |
1 |
SayUnixTime(...,S) |
| 8:26AM |
0 |
Asterisk with wireless serial modems and multiple PC's |
| 8:21AM |
1 |
Asterisk (libpri?) and L1 Flags? |
| 8:17AM |
1 |
SpanDSP.0.0.2 |
| 7:54AM |
2 |
Telco POTS -> FXO ? |
| 7:47AM |
1 |
Calling local extensions (also iax) directly from outside ? |
| 7:33AM |
3 |
Prerelease of DIAX 0.9.9a |
| 6:46AM |
3 |
ValetParking |
| 6:42AM |
0 |
CreateLogicalChannel Unknow Data Type |
| 6:27AM |
2 |
SIP 404 - circuit busy when dialing out |
| 5:20AM |
1 |
Backup POTS line |
| 4:15AM |
1 |
Dialing out with SIP phone problem |
| 4:07AM |
2 |
Not able to establish IAX call |
| 2:29AM |
0 |
IAX pretending to see unreachable hosts and other weird things |
| 2:12AM |
0 |
A question with voice Menu |
| 2:11AM |
1 |
quadBRI FAX problem |
| 1:48AM |
1 |
Where is the cheapest place to buy grandstream phones ?. |
| 1:34AM |
0 |
remote pickup |
| 12:43AM |
2 |
X100P sending out tone all the time? |
| 12:00AM |
4 |
RxFax multiple pages |
| |
| Tuesday October 12 2004 |
| Time | Replies | Subject |
|---|
| 11:11PM |
2 |
Called name delivery |
| 9:38PM |
14 |
Passing CallerID to SIP phone from TDM400P |
| 8:10PM |
1 |
Bluetooth Bounty |
| 7:42PM |
6 |
Asterisk VIA SSH Tunnels |
| 5:38PM |
4 |
TDM01B Goes missing after reboot |
| 5:25PM |
9 |
mwi over serial port |
| 4:19PM |
2 |
G729 to G711 bridge |
| 4:16PM |
1 |
How many running instances (jobs) of asterisk |
| 3:46PM |
0 |
Canada Toll free |
| 2:48PM |
2 |
Channel Bank for T100P or E100P Digium Cards |
| 2:26PM |
2 |
Cisco 7960G "disk full error" |
| 2:09PM |
3 |
musiconhold will not start |
| 1:14PM |
1 |
SIP Connection to a Cisco AS5xxx gateway |
| 12:53PM |
5 |
Polycom Echo |
| 12:44PM |
0 |
In immediate need of Very powerful * for callcenter, ACD and outbound. Which consultant should I use? |
| 12:41PM |
0 |
Mesh Networking & SIP |
| 12:10PM |
1 |
Control Panel |
| 11:28AM |
0 |
In immediate need of Very powerful * for cal l center, ACD and outbound. Which consultant should I use? |
| 11:21AM |
3 |
In immediate need of Very powerful * for call center, ACD and outbound. Which consultant should I use? |
| 11:16AM |
2 |
Alternatives to the T100Ps? |
| 11:11AM |
0 |
Detect phone pickup, caller ID AGI |
| 10:26AM |
6 |
Chaining more than one zap echo canceller? |
| 9:55AM |
1 |
cdr make problem |
| 9:26AM |
4 |
Large Scale Asterisk Migration |
| 9:02AM |
0 |
Will an in-band 2100hz tone disable the zaptel (and/or other) Ast erisk echo cancellers? |
| 8:57AM |
0 |
ZyXEL P2602HW (WiFi + ATA Router) |
| 8:19AM |
2 |
calculating bandwidth on DSL? |
| 8:01AM |
4 |
Fast Busy |
| 8:00AM |
4 |
QoS Router/Software Suggestions |
| 7:51AM |
1 |
cvsup options file for v1-0 |
| 6:57AM |
0 |
low bandwidth? |
| 6:47AM |
2 |
rfc3389 support in chan_sip? |
| 5:52AM |
0 |
4. Re: Quicknet Linejack Asterisk PBX (Lubomir Christov) |
| 5:47AM |
2 |
billing??? |
| 5:41AM |
0 |
. Re: Quicknet Linejack Asterisk PBX (Jeremy McNamara) |
| 5:28AM |
0 |
RE: bt communicator` |
| 5:19AM |
0 |
Specifying different SIP packet destination from hostname in request line? |
| 5:09AM |
0 |
Echo Problem with IAX and Zaptel |
| 5:04AM |
3 |
How big .CONF files can be? |
| 3:11AM |
1 |
Zyxel P2000W web interface? |
| 2:20AM |
7 |
divert if not here |
| 1:51AM |
1 |
Redunance and failover |
| |
| Monday October 11 2004 |
| Time | Replies | Subject |
|---|
| 11:24PM |
5 |
* box hangs after a couple of days... |
| 9:26PM |
1 |
CED tone and answering machine detection |
| 9:08PM |
0 |
Webmin modules for Asterisk |
| 8:43PM |
2 |
New Mailbox |
| 8:33PM |
4 |
Quicknet Linejack Asterisk PBX |
| 8:30PM |
1 |
cannot hear voice from phone |
| 6:01PM |
0 |
SNOM-105 |
| 5:51PM |
2 |
SNOM 200 availability |
| 5:27PM |
0 |
OH323 and Mera Softswitch |
| 2:25PM |
0 |
Siemens Hicom / Digium TDM Card. |
| 2:23PM |
2 |
Echo problems polycom and x100p |
| 1:42PM |
1 |
linphone with * |
| 1:41PM |
3 |
Generic X100P's |
| 12:42PM |
0 |
Core files always appear in / ? |
| 12:21PM |
1 |
Zaptel with 2.6.9-rc4 |
| 12:05PM |
1 |
CLI Destroy SIP channel? |
| 11:21AM |
2 |
G726 Codec Question |
| 11:11AM |
2 |
Disable flash hook hold? |
| 10:57AM |
3 |
Extensions.conf? |
| 10:42AM |
0 |
Database of world area codes |
| 10:41AM |
1 |
Sipura SPA-2000 / GSM or iLBC. |
| 10:27AM |
1 |
Unattended call transfer with IAX softphone or IAXy? |
| 9:51AM |
1 |
Re: Dial group continues to ring after answer -SNOM phones and solution |
| 8:45AM |
2 |
T100P to Verizon Smart Jack Question |
| 8:38AM |
0 |
FW: RTP timing issues |
| 8:28AM |
2 |
chan-sccp2 |
| 8:18AM |
1 |
Re: Dial group continues to ring after answer -SNOM phones and solution |
| 8:16AM |
1 |
reading global vars from AGI |
| 8:12AM |
0 |
SOHO small or rack mount chassis and mobo for asterisk |
| 7:48AM |
1 |
System Hang Problem |
| 7:38AM |
0 |
7910 MWI |
| 7:28AM |
1 |
FWD incomming CALL won't authenticate in SIP |
| 7:26AM |
0 |
(no subject) |
| 7:11AM |
4 |
outgoing calls |
| 6:51AM |
1 |
Seeking a VoIP Solution for a big company |
| 6:01AM |
1 |
FYI - Zoom X5v built-in VoIP DSL router |
| 5:55AM |
1 |
Newbie OT Question - Hardware advise |
| 5:22AM |
0 |
re: ATA units: anyone have these working |
| 4:45AM |
1 |
Agent monitoring using fop |
| 4:32AM |
0 |
SetVar() with manager |
| 4:05AM |
2 |
Problems with voice menu |
| 3:30AM |
0 |
Re: Grandstream price in UK |
| 2:32AM |
1 |
SIP hangup issue |
| 1:53AM |
1 |
re: ATA units: anyone have these working with * or SER? |
| 1:34AM |
1 |
RE: bt communicator` |
| 1:28AM |
0 |
Request for IAX debug session transcript with IAXy |
| |
| Sunday October 10 2004 |
| Time | Replies | Subject |
|---|
| 11:39PM |
5 |
Grandstream phone price |
| 11:33PM |
1 |
Where did USE_SIP_MYSQL_FRIENDS go? |
| 11:19PM |
0 |
Re: Asterisk-Users Digest, Vol 3, Issue 121 |
| 10:53PM |
2 |
Error starting |
| 10:31PM |
0 |
C in Dial doesn't work (no cdr) |
| 9:54PM |
0 |
Conferencing -- app_meetme, app_meetme2, app_conference |
| 7:05PM |
2 |
newbie question - app_realtime.so failed |
| 7:00PM |
1 |
How to Connect Fax to Dev PCi |
| 6:51PM |
0 |
Unable to locate sample sounds |
| 5:45PM |
0 |
microphone on localhost gateway |
| 5:42PM |
0 |
Flosys IT-550 GSM Gateway problems. |
| 4:23PM |
1 |
Broadvoice registration timeout |
| 2:10PM |
1 |
cisco ip 7905 legal .. |
| 11:48AM |
1 |
h.323 debian sarge problem - Could not open sound channel |
| 10:18AM |
1 |
Re: Asterisk-Users Digest, Vol 3, Issue 115 |
| 10:08AM |
13 |
Intel Modem vs Digium Cards |
| 9:56AM |
2 |
TTS via text2wave |
| 8:51AM |
0 |
DIAX 0.9.8 and Windows XP SP2 problem |
| 8:49AM |
0 |
Problem with hearing 2nd call when call on hold hangs up |
| 8:39AM |
0 |
R2 update |
| 8:11AM |
1 |
ASTCC : rates based on incoming numbers |
| 7:53AM |
2 |
Zaptel 1.0.0. will not compile |
| 6:24AM |
2 |
DID trunk suggestions for Asterisk |
| 6:19AM |
7 |
SIP peers in MySQL Database |
| 6:16AM |
0 |
webmin module found on asterisk ftp site |
| 3:41AM |
3 |
Xorcom Rapid Asterisk distro beta 0.5.2 |
| 3:22AM |
0 |
Why does incoming SIP call match a "peer" context in sip.conf? |
| 3:16AM |
2 |
Cost based Routing |
| 3:08AM |
3 |
SIP device not able to register but still able to make call |
| 12:11AM |
0 |
Channel Bank Suggestion. 100+ extensions system |
| |
| Saturday October 9 2004 |
| Time | Replies | Subject |
|---|
| 8:41PM |
3 |
SPA-3k outbound calls... |
| 6:54PM |
1 |
Patch * |
| 6:35PM |
1 |
Access Bank II |
| 6:30PM |
1 |
Modprobe zaptel fails: Unknown symbol crc_ccitt_table |
| 4:28PM |
2 |
Asterisk Video Conference |
| 3:26PM |
0 |
Re: Vonage, PSTN, 911, and hardware question (Rajeef Sharma) |
| 2:00PM |
1 |
Asterisk - SIP -chan_sip.c:595 __sip_xmit: sip_xmit of 0x815008c (len 342, |
| 12:56PM |
1 |
iax2 w/ pa1688 |
| 11:51AM |
1 |
ASTCC :: Strange Problem ast_openstream |
| 10:22AM |
1 |
nufone config |
| 9:49AM |
0 |
Unable to open master device - Fedora Core 2 |
| 8:45AM |
0 |
web interface for meetme |
| 8:44AM |
0 |
How to check Asterisk status ? |
| 8:43AM |
1 |
Howtos on writting applications or modules ? |
| 8:42AM |
0 |
Re: if my Asterisk server is behind a FW ?. |
| 8:16AM |
4 |
Am I stupid or is my card DOA.? |
| 7:23AM |
0 |
Re: Sound Problem with * on VIA mini-itx M10K AC97' VT8235 (working) |
| 7:20AM |
5 |
Vonage, PSTN, 911, and hardware question |
| 7:10AM |
1 |
SIP SPA-3k & * Configuration |
| 7:03AM |
5 |
Slim Devices Sqeezebox Asterisk voicemail plugin. |
| 1:37AM |
1 |
Loopdrop |
| |
| Friday October 8 2004 |
| Time | Replies | Subject |
|---|
| 9:56PM |
5 |
Can't compile chan_h323 in latest CVS... |
| 8:51PM |
1 |
Patch to queue.conf available for testing in Mantis |
| 5:52PM |
1 |
Cisco 7910 phones unlocking |
| 5:24PM |
0 |
VOIP provider in France? |
| 3:07PM |
4 |
* as sip proxy |
| 3:03PM |
0 |
TDM04B card connected to local switch, calling extension on local switch and playing message before answer |
| 2:39PM |
1 |
Application "Dial" option A |
| 2:37PM |
1 |
Need help configuring T1 |
| 2:34PM |
0 |
Re: Asterisk Certification (was: Open-source VoIP 'will ne bigger than Linux') |
| 2:34PM |
2 |
SIP trunk: asterisk - callmanager |
| 2:30PM |
0 |
analog callerid private/out-of-area/... |
| 2:23PM |
3 |
NuFone & SetCIDNum not working since 10/5 - last Tuesday |
| 2:11PM |
0 |
Asterisk v1.0 CVS RPMS Available |
| 2:03PM |
1 |
MFC/R2 and Caller Id |
| 1:30PM |
0 |
Asterisk-OH323 (Couldn't transmit sound to and from ohphone) |
| 12:59PM |
0 |
Asterisk Certification (was: Open-source VoIP 'will be bigger than Linux') |
| 12:57PM |
0 |
After hangup phone rings |
| 12:41PM |
1 |
Disabling succeeding voicemail text |
| 12:19PM |
2 |
No sendmail on * server |
| 11:45AM |
3 |
user status in * |
| 10:44AM |
0 |
Asterisk ahead of old PBX problems (maybe PRI trouble) |
| 10:34AM |
0 |
Re: E1/R2 specs in Brazil |
| 10:34AM |
2 |
eyebeam and video |
| 10:15AM |
2 |
Answer() |
| 10:04AM |
0 |
IAX Connection problem with voicepulse |
| 9:56AM |
2 |
Flash |
| 9:36AM |
0 |
Hook flash hangup instead of hold? |
| 9:35AM |
0 |
Open-source VoIP 'will be bigger than Linux' |
| 9:30AM |
2 |
Registering to H323 Gatekeeper as client |
| 9:14AM |
2 |
BT ISDN30e and presentation numbers |
| 8:24AM |
0 |
oh323 channel : transport failure |
| 8:06AM |
1 |
connecting asterisk to existing pbx extension line |
| 7:44AM |
0 |
can I use these phones |
| 7:34AM |
1 |
MFC/R2 working with Ericsson MD-110 in Brazil |
| 7:08AM |
1 |
no ringing sound |
| 7:06AM |
0 |
COM-ON-AIR Dect Base as PCI/PC-Card |
| 6:59AM |
2 |
${EXTEN} vs ${CALLERIDNUM} vs ?? |
| 6:45AM |
5 |
Reload Asterisk from php or perl script |
| 6:34AM |
1 |
Sending CDR over permanent tcp port and how reliable is CDR information from Asterisk? |
| 6:20AM |
0 |
problems with asterisk-oh323-0.6.3b |
| 6:16AM |
2 |
Bypass VoiceMail Mailbox prompt |
| 6:09AM |
5 |
SPA3000 as a replacement for X100P |
| 5:56AM |
1 |
Zapateller Answering? |
| 5:16AM |
0 |
openphone congested link |
| 5:14AM |
0 |
BGT100 - Question... |
| 3:10AM |
2 |
open phone |
| 2:43AM |
1 |
grandstream bt-100 callerID not appear |
| 1:11AM |
0 |
Eicon DIVA Server 4BRI-8M |
| 12:45AM |
1 |
Incorrect ANI sent to PRI provider - CVS 9-29-04 |
| 12:25AM |
0 |
re:uniqueid - how unique it is (Sathya Weerasooriya) |
| 12:18AM |
1 |
versions? |
| |
| Thursday October 7 2004 |
| Time | Replies | Subject |
|---|
| 11:39PM |
1 |
dial out |
| 9:37PM |
0 |
chan_capi make issue |
| 9:14PM |
1 |
ATA & T.38 Fax |
| 8:58PM |
1 |
Adtran setup question |
| 6:19PM |
0 |
Cisco BTS 10200 G.729 problem |
| 5:54PM |
1 |
Call Parking with multiple contexts |
| 4:55PM |
0 |
uniqueid - how unique it is |
| 4:36PM |
1 |
IAX2 wait on channel |
| 4:03PM |
2 |
recent 's' and 'n' priorities and lables |
| 3:46PM |
1 |
T100P Pri Audio |
| 2:58PM |
1 |
PA168 ATCOM /Ezeephone Configuration |
| 1:51PM |
0 |
CallerID X100P |
| 1:49PM |
2 |
Nortel DMS250 |
| 12:42PM |
0 |
Dialplan to Pick up calls that are ringingonother extensions? |
| 12:10PM |
6 |
Beginers Help - Hardware selection |
| 12:09PM |
0 |
Asterisk died on code 127 |
| 12:02PM |
3 |
- Advice on NetFinity 5000 series |
| 11:58AM |
0 |
I need modify the time and cost the minute to second in application astcc ? |
| 11:21AM |
5 |
just getting started |
| 11:16AM |
5 |
Broadvoice problems |
| 11:16AM |
2 |
Dialplan to Pick up calls that are ringing onother extensions? |
| 11:07AM |
1 |
7912 Compatible SIP Images? |
| 11:06AM |
3 |
CVS branch v1-0 .vs v1-0-1 |
| 10:42AM |
2 |
Dialplan to Pick up calls that are ringing on other extensions? |
| 9:44AM |
0 |
simple sip client |
| 9:17AM |
0 |
RE: Cisco and PRI IOS load |
| 9:16AM |
1 |
spandsp RxFAX problems. |
| 8:57AM |
2 |
TDM400P with FXO/FXS hangup problem |
| 8:54AM |
0 |
RE: Cisco and PRI |
| 8:44AM |
0 |
Remote Voice Mail |
| 8:23AM |
0 |
RE: Cisco and PRI |
| 8:20AM |
0 |
chan_h323 on latest CVS broken ? |
| 7:59AM |
0 |
Calling Card - GNUGK |
| 7:36AM |
1 |
'set debug' problems |
| 7:34AM |
0 |
Incomming calls on Eicon Diva 4BRI Card |
| 7:34AM |
0 |
Forcing a codec in chan_oh323 |
| 7:26AM |
0 |
No incoming call on SIP Phone |
| 6:59AM |
0 |
Asterisk over NetScreen VPN/SIP protocol |
| 6:59AM |
1 |
SIP.CONF "Allow=All" do not work! |
| 6:47AM |
1 |
Confused about NAT and Authentication with FWD |
| 6:37AM |
2 |
openphone & Asterisk |
| 6:30AM |
1 |
bri-stuff for Asterisk v1.0 |
| 5:52AM |
0 |
starting problem |
| 5:51AM |
2 |
RxFax - tiff problem |
| 5:36AM |
2 |
Asterisk ---- SER ----- GAteway and Reinvite |
| 5:27AM |
1 |
RTP timing issues |
| 5:18AM |
1 |
spa 3000 help |
| 5:10AM |
0 |
x-lite voicemail indicator |
| 4:52AM |
5 |
Display called Number or context on X-lite/X-Pro |
| 4:23AM |
0 |
Re: I'm thinking that FTP makes more sense for Volume One than CVS does |
| 4:02AM |
0 |
mod problem |
| 3:54AM |
1 |
Meetme conference across WAN |
| 3:43AM |
3 |
Vmail & Snom 190s |
| 3:16AM |
0 |
SIP header values in the dialplan |
| 2:03AM |
2 |
Higher level API on top of the Manager API? |
| 1:43AM |
0 |
ISDN4Linux early call progress tones & announcements from the PSTN |
| 1:39AM |
0 |
asterisk-oh323-0.5.10 problem |
| 1:29AM |
0 |
Missing Request URI in SIP message |
| 1:10AM |
0 |
phone autoregistration to * |
| 12:34AM |
0 |
Anyone using AddPac AP1200 VoIP Gateway? |
| |
| Wednesday October 6 2004 |
| Time | Replies | Subject |
|---|
| 11:56PM |
2 |
AVM FritzCard |
| 11:50PM |
2 |
jabber clients |
| 9:59PM |
2 |
Transfer to Fax - 123 - Vonage |
| 9:53PM |
4 |
TDM400P stop responding |
| 9:46PM |
1 |
Asterisk Forums needs your input (http://asterisk.xvoip.com) |
| 8:32PM |
2 |
IAX2 Sporadic TX/RX retries |
| 8:14PM |
2 |
Cisco router for PRI termination? |
| 8:05PM |
0 |
Can asterisk unregister? |
| 7:01PM |
1 |
IAX2 to SIP |
| 6:54PM |
0 |
Zultys phones with encryption |
| 6:48PM |
2 |
REGISTER timeout problem with Broadvoice |
| 6:46PM |
2 |
No sound on 2 x-lite |
| 6:36PM |
2 |
Softphone |
| 6:15PM |
0 |
Grand Stream 486 |
| 5:04PM |
10 |
Eezee phone? |
| 3:40PM |
3 |
Limit G729 concurrent calls |
| 3:11PM |
4 |
* to Cisco router with FXO's via SIP |
| 3:10PM |
3 |
T100p half-height PCI bracket |
| 2:49PM |
1 |
how does agent logoff if you supply extension? |
| 2:19PM |
3 |
Re 2 x100p cards H E L P (I have no hair left) |
| 2:01PM |
1 |
Asterisk to BabyTel VoIP SIP Provider |
| 1:41PM |
0 |
Remote mailbox in sip.conf |
| 1:23PM |
0 |
Rating IP phones from the users POV? |
| 1:00PM |
0 |
Asterisk and Festival, getting gethostbynamefailed error |
| 12:59PM |
3 |
OH323 compilation with updated Asterisk |
| 12:48PM |
0 |
Anyone using the Micronet FXS/FXO devices w/SIP |
| 12:37PM |
0 |
IAXy - Password protecting the IAXy device. |
| 12:24PM |
0 |
iax2, strange native bridge problem???? |
| 12:24PM |
1 |
Asterisk and Festival, getting gethostbyname failed error |
| 11:40AM |
5 |
Astricon 2004 links collection |
| 10:51AM |
2 |
What would be needed .. |
| 10:33AM |
1 |
Tracking pressed keys |
| 10:20AM |
7 |
Comedian Mail User Guide |
| 10:15AM |
3 |
Setup problems |
| 9:47AM |
1 |
E100 Guide |
| 8:59AM |
1 |
Problem on getting wav file out of Festival-text2wave utility ? |
| 8:21AM |
0 |
mISDN channel |
| 7:42AM |
1 |
Cisco IOS SIP mime 1.0 |
| 7:15AM |
0 |
Eicon ISDN to Voicemail audio dropouts |
| 6:57AM |
10 |
Asterisk and SIP phones |
| 6:32AM |
2 |
Issue with the channel drivers |
| 5:56AM |
2 |
Cisco Support for 7940, Is this Right? |
| 5:53AM |
2 |
Call Quality |
| 5:40AM |
0 |
Franklin Telecom Cards |
| 5:29AM |
0 |
GSM codec error in current CVS? |
| 5:22AM |
0 |
sip jitter |
| 5:18AM |
1 |
Cisco and ILBC |
| 4:58AM |
0 |
Pri E1 Timing - help ... |
| 3:42AM |
0 |
Echo / Adit 600 |
| 3:38AM |
1 |
Queues/Agents |
| 3:25AM |
1 |
Hello - Simple SIP configuration |
| 3:18AM |
1 |
USE_MYSQL_FRIENDS=0, USE_SIP_MYSQL_FRIENDS=0 what is the difference |
| 2:43AM |
0 |
Can Asterisk provide Answer Supervision signalling to a channel b ank via T1? |
| 2:29AM |
2 |
Working Wellgate *SIP* 38xx/35xx hardware anyone? |
| 2:28AM |
2 |
no audio from asterisk |
| 2:27AM |
1 |
Anyone using VoiceMaster |
| 2:16AM |
0 |
Asterisk 1.0 -- Did the SIP dial syntax change? |
| 2:06AM |
0 |
asterisk generating alot of channels.. but for what ? |
| 1:55AM |
1 |
How much extensions will exhaust asterisk |
| 1:40AM |
0 |
start cracking |
| |
| Tuesday October 5 2004 |
| Time | Replies | Subject |
|---|
| 10:03PM |
2 |
Queue() option not documented |
| 9:53PM |
2 |
Long pause between menus |
| 9:41PM |
0 |
loggedoff extension - why does * say "isonthephone" |
| 9:12PM |
1 |
Rate engine |
| 8:41PM |
0 |
loggedoff extension - why does * say "is onthephone" |
| 7:15PM |
2 |
odd configuration ... possible ? |
| 7:06PM |
1 |
Cannot compile Meetme2 |
| 6:34PM |
1 |
Custom Monitoring Directories per queues |
| 6:29PM |
0 |
What kind of issues?????? |
| 6:23PM |
1 |
loggedoff extension - why does * say "is on the phone" |
| 5:25PM |
0 |
real/wm/etc audio stream -> zap extension |
| 4:57PM |
0 |
How to force G.729 in H.323 calls |
| 3:57PM |
1 |
Dlink DVG-1120 Linksys PAP2 any Good? |
| 3:52PM |
0 |
MS netmeeting and * |
| 3:38PM |
1 |
Why I don't hear Call Progress |
| 3:15PM |
2 |
Howto change ACCOUNTCODE in extensions.conf |
| 2:48PM |
0 |
meetme caused 'RTP Read error: Bad file descriptor' |
| 2:47PM |
1 |
Dial group continues to ring after answer |
| 2:45PM |
0 |
looping back calls |
| 2:39PM |
2 |
broadvoice connection problem |
| 2:33PM |
1 |
asterisk compile with Fedora core 3 test 2 FC3T2 |
| 2:23PM |
1 |
MeetMe MySQL Patch - Testers Needed |
| 1:59PM |
0 |
Using Macro's that cause loops, on purpose and using h, exten in default twice |
| 1:57PM |
0 |
SIP and symmetric NAT |
| 1:38PM |
3 |
its all 6's and 8's? |
| 1:31PM |
1 |
Newbie question ... |
| 12:47PM |
2 |
Problems installing app_valetparking |
| 12:41PM |
0 |
New Asterisk-CVS and Kernel/ALSA support RPMS Available NOW! |
| 12:15PM |
1 |
Pass a call to another switch |
| 12:11PM |
0 |
CMS |
| 12:01PM |
1 |
Popping and Clicking on Local WAN with X-Lite |
| 11:52AM |
1 |
Read error on |
| 11:46AM |
3 |
C flag in Dial command |
| 11:31AM |
5 |
Asterisk Perl AGI |
| 11:28AM |
1 |
Phantom calls on FXO |
| 10:44AM |
1 |
difference between dtmf digit 8 and 9 |
| 10:35AM |
4 |
Long distance provider with access number and auth code |
| 10:18AM |
0 |
Asterisk and Alcatel 4200 -- comments, anyone? |
| 10:15AM |
2 |
SIPphone All-in-One: coments anyone? |
| 10:01AM |
1 |
Auto attendant dial an extenstion |
| 9:57AM |
0 |
Snom 220 Transfer Oddness |
| 9:44AM |
1 |
Low-Cost SIP Phones, ATA and Gateway excelent for Asterisk |
| 9:28AM |
1 |
For Sale Cisco IP Phones and ATA's |
| 9:07AM |
2 |
Re: RES: Working E1 MFC/R2 M?xico !!! (Steve Underwood) |
| 8:50AM |
0 |
Asterisk CLI Prompt : Small hack |
| 7:26AM |
1 |
hints lost after reload |
| 7:03AM |
1 |
FAXMAIL |
| 6:57AM |
1 |
Forcing a codec (take 2) |
| 6:37AM |
2 |
SIP multipart mime messages |
| 6:12AM |
1 |
Help 2 fx0 cards |
| 6:09AM |
3 |
books about ISDN/ss7? |
| 5:41AM |
1 |
Non-working module on TDM400P? |
| 5:22AM |
4 |
[OT] Has Sipura support been closed down? |
| 5:21AM |
1 |
Brazillian Caller ID: almost there... |
| 5:13AM |
1 |
OT: Can I use a SIPURA with Packet8? |
| 5:07AM |
0 |
Re: Firefly 1.9.5 released (gARetH baBB) |
| 4:43AM |
0 |
Polycom Echo using IAX2 |
| 4:27AM |
1 |
asterisk with sipphone.com |
| 4:25AM |
0 |
Asterisk, Zaptel and Legacy Phones? |
| 4:25AM |
0 |
Paypal? Available in 44 of the world's approximately 190 countries |
| 4:19AM |
3 |
TDM20B and UK caller ID signalling |
| 3:31AM |
1 |
problems withX100P-Nochanneltyperegisteredfor'Zap' |
| 2:49AM |
1 |
problems with X100P -Nochanneltyperegisteredfor'Zap' |
| 2:47AM |
0 |
asterisk vic2-2bri NT/TE as gateway |
| 2:19AM |
0 |
Find a person |
| 2:14AM |
0 |
sipura 3000 , music on hold (playtones) |
| 2:13AM |
2 |
problems with X100P - Nochanneltyperegisteredfor 'Zap' |
| 1:53AM |
0 |
Just getting started with Asterisk |
| 1:26AM |
3 |
Special Meetme |
| 1:15AM |
0 |
Grandstream * Kingston Comms |
| 1:04AM |
2 |
Dialing a # in phone number? |
| 12:48AM |
1 |
problems with X100P - No channeltyperegisteredfor 'Zap' |
| 12:26AM |
1 |
Firefly 1.9.5 released |
| 12:16AM |
0 |
H.323: Inbound calls, incorrect remoteIpAddress |
| |
| Monday October 4 2004 |
| Time | Replies | Subject |
|---|
| 11:42PM |
5 |
CallerID Question |
| 11:37PM |
0 |
Asterisk v1.0 sends incorrect invite to Sipura SPA-3000? |
| 10:42PM |
1 |
Cisco 7960G w/ SIP not working properly |
| 10:33PM |
10 |
IAXy - anyone using them yet? |
| 10:33PM |
2 |
problems with X100P - No channel type registered for 'Zap' |
| 7:42PM |
0 |
Avaya 4624 phones |
| 7:30PM |
1 |
some problems with OH323 |
| 7:24PM |
0 |
How to check transcoding of audio codec |
| 6:41PM |
1 |
How to see CODEC which is in use? |
| 6:38PM |
0 |
Cisco ATA-188 w/502 Error on CallWaiting |
| 5:19PM |
2 |
3com NBX intergration |
| 5:17PM |
2 |
Limit extensions to single lines |
| 5:12PM |
1 |
IAX2 trunk mode not working |
| 5:05PM |
1 |
VoicePulse Connect Usage ?? |
| 4:41PM |
2 |
Vonage just doesn't work? |
| 4:07PM |
1 |
ASTERISK PACKET ANALYSIS |
| 2:17PM |
3 |
motherboard for T100P |
| 1:59PM |
3 |
IAX/Grandstream. |
| 12:11PM |
3 |
Cisco XML 411 Interface |
| 11:58AM |
2 |
call/pickup groups |
| 11:49AM |
1 |
Will there be any support for iLBC in IAXClients soon? |
| 11:45AM |
0 |
OT: BudgetTone CallerID |
| 10:48AM |
5 |
Voice mail options/behaving change? |
| 10:42AM |
2 |
Queue/Agents problem with 1 agent |
| 10:18AM |
1 |
Parking calls |
| 10:13AM |
0 |
Ohio Linuxfest 2004 Presentation |
| 9:34AM |
2 |
Off Topic: Dead GS BudgeTone-100 |
| 9:25AM |
0 |
echo cancellation: the never-ending quest fortruth |
| 9:06AM |
6 |
How to become an IP Service Provider? |
| 9:01AM |
0 |
Cisco 79XX Conference Call Issue |
| 8:26AM |
2 |
RES: Working E1 MFC/R2 M?xico !!! |
| 8:25AM |
0 |
Call waiting question for those who know the source |
| 7:34AM |
0 |
RES: Asterisk-Users Digest, Vol 3, Issue 25 |
| 7:20AM |
1 |
What happened to my "Dial" command? |
| 6:56AM |
2 |
Re: Sound Problem with * on VIA mini-itx M10K AC97' VT8235 |
| 6:56AM |
3 |
echo cancellation: the never-ending quest for truth |
| 6:46AM |
0 |
Lucent i2021 BRI Phone with Asterisk? |
| 6:37AM |
2 |
fxsmod cable length limit |
| 6:28AM |
2 |
Somebody using AS5350 CISCO? |
| 6:27AM |
2 |
exten patterns: how to match from XXX to ZZZ ? |
| 6:16AM |
3 |
budgetone-100 and handtone-286 |
| 6:08AM |
1 |
Macro's and Var Scope's |
| 5:53AM |
12 |
Choosing a VoIP Phone |
| 5:44AM |
1 |
Asterisk CALLING CARD |
| 5:25AM |
0 |
using broadvoice and vonage hardware withAsterisk |
| 5:24AM |
1 |
Tones on a Cisco 7960 ? |
| 5:21AM |
2 |
FW: ASTCC: how to set quiet level |
| 5:19AM |
2 |
Re Problem with Asterisk 2 fx100 cards |
| 2:29AM |
1 |
SIP Proxy and use with Asterisk |
| 2:14AM |
0 |
Quassar 111 L |
| 1:18AM |
0 |
Appending a # to a dial-out number |
| 12:57AM |
2 |
300 extensions on Asterisk? |
| 12:28AM |
1 |
enhanced speed dial |
| |
| Sunday October 3 2004 |
| Time | Replies | Subject |
|---|
| 11:39PM |
3 |
Conference by SIP phone |
| 11:33PM |
1 |
Help!!! Does Asterisk support call waiting in SIP phones |
| 10:26PM |
1 |
"#" sending |
| 8:04PM |
2 |
using broadvoice and vonage hardware with Asterisk |
| 7:35PM |
0 |
Sphinx 4 |
| 7:33PM |
0 |
Issue with the oh323 channel driver compilation |
| 7:24PM |
3 |
ATA's |
| 6:57PM |
1 |
Sound Problem with * on VIA mini-itx M10K AC97' VT8235 chipset |
| 6:04PM |
3 |
Amazing, great protocol IAX |
| 4:06PM |
0 |
Tenor AS cancells calls through Asterisk |
| 2:40PM |
3 |
VoiceMail without password? How? |
| 1:53PM |
1 |
Asterisk + NCS patch |
| 1:51PM |
0 |
NCS and asterisk |
| 1:22PM |
3 |
asterix and phone system |
| 1:01PM |
3 |
Help with concept. |
| 12:06PM |
0 |
FW: Broadvoice |
| 11:19AM |
0 |
A problem with Asterisk-oh323 |
| 10:40AM |
0 |
Looking to contract asterick setup |
| 10:07AM |
2 |
SIP-Provider who allows own Caller-ID |
| 9:53AM |
0 |
Call gets disconnected upon connect |
| 9:21AM |
0 |
Working E1 MFC/R2 México !!! |
| 6:50AM |
1 |
_asterisk-update |
| 1:04AM |
2 |
Hard phones that support ILBC |
| 12:48AM |
1 |
Where are the $500 24 port FXS gateways? |
| 12:03AM |
0 |
Presence Utility |
| |
| Saturday October 2 2004 |
| Time | Replies | Subject |
|---|
| 10:04PM |
1 |
Second X100P card won't work |
| 4:41PM |
3 |
Getting Digium TDM card - what to watch for? |
| 4:13PM |
1 |
Latest ASTCC |
| 3:56PM |
2 |
Billing Applications - When does the bill start?? |
| 3:34PM |
0 |
Asterisk- Cisco 7912G second call problem |
| 3:25PM |
0 |
ast_openstream: File your does not exist in any format |
| 2:01PM |
1 |
Fax passthrough |
| 12:54PM |
0 |
Packet cable NCS |
| 12:38PM |
1 |
H323 dial problem |
| 11:46AM |
1 |
RE: Random disconnects |
| 11:20AM |
0 |
IAX Ping for perl or python |
| 9:28AM |
0 |
Stability of the Asterisk platform |
| 9:25AM |
1 |
Compiling HDLC does not Produce hdlc0 for T100p |
| 9:11AM |
7 |
Callback |
| 9:08AM |
2 |
[OT] Sipura-3000 - Immediate hangup on inbound PSTN calls |
| 6:25AM |
2 |
voicemail attachment volume |
| 3:04AM |
0 |
S100U crashing server |
| 2:35AM |
2 |
Patch: Inbound-only busydetect |
| 2:29AM |
0 |
strange problem with a NT1 connected to * and an ISDN modem for data connection |
| |
| Friday October 1 2004 |
| Time | Replies | Subject |
|---|
| 10:31PM |
1 |
Please, send me g723 & g729, pls |
| 9:35PM |
1 |
chan_sccp error |
| 8:22PM |
2 |
HT 486 |
| 7:47PM |
2 |
Sipura 3000 FXO |
| 3:14PM |
1 |
Zaptel and ztdummy and timming question |
| 2:53PM |
1 |
Solution to my Grandstream lockups |
| 2:31PM |
0 |
Fw: OT: Opensource "Sipura Profile Compiler" for SPA2K, 3K |
| 1:46PM |
5 |
OT: Opensource "Sipura Profile Compiler" for SPA2K, 3K |
| 1:46PM |
1 |
upgrade goof up |
| 1:23PM |
4 |
CDR_Oracle anybody? |
| 1:03PM |
1 |
SMS in the U.S. |
| 12:56PM |
2 |
Forcing a codec |
| 12:45PM |
0 |
identify meetme participant by PIN |
| 12:24PM |
0 |
Random Call Disconnect |
| 11:58AM |
1 |
Unable to create Zap channels/IAX Warning |
| 11:55AM |
1 |
OT: Uniden UIP200 and NMAP |
| 11:49AM |
1 |
softphone over harphone |
| 11:44AM |
1 |
setting up more than one company on same * machine |
| 11:42AM |
1 |
BUG? no output from 'sip show users|inuse|active|subscriptions' when using MySQL auth |
| 11:41AM |
4 |
spandsp 0.0.2 |
| 11:27AM |
4 |
IAX2 - Voice Pulse - slow choppy audio |
| 11:12AM |
1 |
Re: Problem with TDM400P |
| 10:49AM |
1 |
OT: Toll Free |
| 10:07AM |
2 |
MOH - 3 processes of mpg321 taking 20%CPU each -normal ? |
| 9:56AM |
3 |
Nuvox PRI - CCITT (ITU??) vs. ANSI |
| 9:47AM |
1 |
astcc question |
| 9:46AM |
1 |
How to contribute code? |
| 9:14AM |
2 |
H323 with 723.1 |
| 8:21AM |
1 |
Agent Login Problems |
| 7:58AM |
0 |
More Reverb like Echo when calling for analog to ISDN - CAPI Fritz - what can I do ? |
| 7:56AM |
2 |
MOH - 3 processes of mpg321 taking 20%CPU each - normal ? |
| 7:07AM |
1 |
Help to connect to Mitel PBX via a T1 connection and a T100p |
| 6:59AM |
1 |
Cisco 7965 - New IP phone - Need Info |
| 6:50AM |
2 |
Maintenance Contract for a Cisco 7960 phone |
| 6:49AM |
1 |
DTMF relay |
| 6:28AM |
0 |
Re: [Asterisk-Dev] Use the Meetme application with another module thanUSB-UHCI |
| 4:59AM |
2 |
Hardware Compatibility Question |
| 4:53AM |
1 |
asterisk-addons on FreeBSD |
| 4:31AM |
0 |
E1 R2 MFC almost working on Mexico |
| 3:43AM |
0 |
S100U / wcusb Zaptel driver / Crash / Kernel problem maybe? |
| 3:01AM |
0 |
Cisco CM 3.3 and * via h.323 |
| 2:41AM |
1 |
Intervivo sip.conf? |
| 2:35AM |
0 |
ASTCC with inbound |
| 2:18AM |
1 |
How to configure the voicemail message playback sequence |
| 2:08AM |
0 |
Is it possible to limit the number of voicemail per users? |
| 12:50AM |
0 |
Re: Asterisk-Users Digest, Vol 2, Issue 342 |
| 12:36AM |
2 |
IAX busy signalling? |
| 12:10AM |
1 |
Configuring X Ten to make call using FX0 |