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 |