Wednesday June 30 2004 |
Time | Replies | Subject |
10:05PM |
1 |
Session timer |
8:26PM |
0 |
CVsup up |
8:05PM |
1 |
Can't transfer with Zap and SPA-2000 |
6:39PM |
1 |
Digium cards supporting E&M signaling |
5:04PM |
0 |
Dial plan errors |
4:36PM |
4 |
Echo cancellation, when software doesn't cut it. Whats next? |
4:11PM |
0 |
Answering Service Auto Login |
3:13PM |
2 |
Ring tone changes when asterisk answers the call |
3:10PM |
1 |
Patch for call queues? |
2:57PM |
8 |
Special Delivery from China |
2:41PM |
0 |
asterisk: problems with connecting to a (german) sip provider |
2:40PM |
0 |
Asterisk Wish List - Can We do it? Can you add to it? |
2:23PM |
0 |
Double DTMF digits |
2:05PM |
7 |
Asterisk Causing Cisco 7200 Router to Crash? |
1:28PM |
0 |
Ipeya iGate-4-FXO-SIP |
1:25PM |
1 |
SIP Notify contents showing 0/0 on VoiceMail |
1:17PM |
5 |
strange problem with oh323 loaded! |
12:59PM |
1 |
Spam |
12:03PM |
1 |
asterisk-addons unable to compile |
11:49AM |
3 |
Answering Service Agent Auto Login |
11:39AM |
3 |
Bugfix for CVS-HEAD-06/26/04-21:56:45 |
10:57AM |
1 |
Providing Telewest in the UK with per extension outbound callerID |
10:40AM |
1 |
Using Asterisk as H323 gateway |
9:31AM |
2 |
Remote SIP client HACK JOB |
9:25AM |
1 |
Sound not working? |
8:41AM |
1 |
IAX2 IP Address memory |
8:21AM |
1 |
Null Pointer Reference h225_1.cxx |
7:17AM |
3 |
10:10am CST - VoicePulse appears to be down |
6:35AM |
0 |
Hold Button on FireFly does not launch MusicOnHold on *? |
5:49AM |
3 |
Support for CENTOS-3.1 |
5:16AM |
2 |
Anyone using gr303? |
4:06AM |
0 |
Compile error with CVS HEAD zaptel |
3:46AM |
2 |
AGI Diad number |
1:58AM |
6 |
zaphfc - hfc pci based ISDN card : point2point & DDI |
|
Tuesday June 29 2004 |
Time | Replies | Subject |
10:28PM |
0 |
Vonage Softphone/resolved |
8:58PM |
5 |
SIP->Asterisk->GnuGK->Cisco 5300 |
8:22PM |
0 |
VoiceTronix OpenLine4 FXO Setup |
8:18PM |
3 |
[RFC] New Wiki page on IAX2 authentication |
6:54PM |
1 |
TDM411B configuration |
5:22PM |
0 |
timestamp in the future (linphone) |
1:17PM |
1 |
Registration of H323 Endpoints? |
1:03PM |
1 |
Spandsp and rxfax |
12:59PM |
0 |
Ring Voltage |
12:54PM |
1 |
RTP Binding Address |
12:24PM |
0 |
chan_dialogic |
11:55AM |
2 |
cisco phone and parked calls |
11:28AM |
3 |
incoming cid translation tables |
11:15AM |
4 |
ISP: AT&T or Sprint |
11:04AM |
0 |
Caller ID software v1.1 available |
11:00AM |
1 |
Update Mysql with DTMF |
10:55AM |
3 |
t100p configuration troubles |
10:34AM |
0 |
Play Music on hold until a ZAP channel frees up. |
10:10AM |
4 |
Getting Asterisk to automatically dialout |
9:57AM |
0 |
ldap-lookup |
9:56AM |
2 |
T100P-E100P circuit board differences |
9:31AM |
0 |
Complaining Emails |
9:24AM |
5 |
nat problem |
8:53AM |
0 |
MGCP and call waiting, doesn't work. |
8:51AM |
0 |
Playing the invalid extension input |
8:23AM |
1 |
Asterisk and dial-up modems |
8:14AM |
0 |
DLink mgcp phone and CVS HEAD |
7:13AM |
5 |
Outgoing CallerID on PRI problems |
6:36AM |
3 |
Call dropping out after 5s: Solution! |
5:32AM |
0 |
h323 audio problem (next) |
5:10AM |
0 |
Routing incoming H.323 calls to specific contexts. |
4:35AM |
0 |
Compiling libiax2 on windows |
4:17AM |
0 |
Voip Account over H323 |
3:42AM |
4 |
Ruggedised IP Phone |
3:07AM |
4 |
1 user 2 VM boxes? |
2:43AM |
2 |
cvs log archive |
2:06AM |
2 |
Customized Call Parking |
1:59AM |
2 |
How to test E1 interfacing? |
1:36AM |
1 |
Asterisk and Sipura SPA-1000 configs |
1:23AM |
0 |
P32mxi |
12:43AM |
3 |
linux kernel 2.6.6 |
12:14AM |
1 |
* Busy-Redial ?? |
|
Monday June 28 2004 |
Time | Replies | Subject |
10:12PM |
2 |
New Firefly release - 1.9.3 |
9:34PM |
1 |
cannot make app_prepaid |
8:04PM |
2 |
Adit 600 - Getting Dial Tone |
7:16PM |
3 |
Polycom IP600 stops to send/receive calls |
6:34PM |
1 |
Cisco 79XX Ringers & chan_sccp |
5:07PM |
2 |
Incoming IAXTel/IAX2 issue |
5:00PM |
0 |
SpanDSP Scrunching incoming faxes |
3:47PM |
5 |
Modems behind Asterisk - how? |
3:46PM |
0 |
New VoIP deployment. |
3:37PM |
0 |
Suggestions for Outbound Proxies? |
1:48PM |
1 |
Asterisk & Festival, not a happy couple |
1:26PM |
0 |
Context for Incomingmsn |
1:18PM |
1 |
Asterisk and hyperthreading |
12:55PM |
0 |
Weird 7940 issue |
12:51PM |
2 |
Security Vulnerability in Asterisk |
12:01PM |
2 |
Would this work? |
11:47AM |
0 |
Queue hold time in seconds |
10:50AM |
2 |
Vonage and Asterisk integration |
10:34AM |
4 |
Chan_Capi Down |
9:43AM |
4 |
Dial Command |
9:09AM |
4 |
zaptel compile error |
8:48AM |
1 |
Asterisk Flah Operator Panel show iax2 trunk |
8:26AM |
5 |
Zap X100P oscillation |
7:55AM |
1 |
(no subject) |
7:45AM |
2 |
asterisk-oh323, new version 0.6.3 |
7:35AM |
2 |
AGI->Exec Problem |
6:00AM |
1 |
TE410P -> Dialogic D240SC |
3:45AM |
1 |
Protocol Error (6) using Zaphfc |
3:13AM |
1 |
SetGroup and CheckGroup |
3:11AM |
1 |
Unable to forward voice |
2:11AM |
1 |
Disappointed |
1:52AM |
2 |
sip to isdn-capi call problem |
|
Sunday June 27 2004 |
Time | Replies | Subject |
11:01PM |
1 |
New idea |
7:02PM |
4 |
Re Cron |
6:55PM |
0 |
ip10s Sip Firmware released |
6:52PM |
1 |
Re: I never get to hear more than 5s of the demo channels |
6:49PM |
1 |
IAX Phone Issues/McAfee Virus Scan vs. IAX Phone |
5:23PM |
0 |
Hangup Issue |
5:16PM |
1 |
Source for MD3200 modem cards? |
3:20PM |
4 |
H.323 Audio problem UPDATE |
1:39PM |
3 |
Re:Latest Echo changes |
12:55PM |
1 |
Asterisk on 64 bit... and testing e164.org |
11:43AM |
5 |
Optipoint 400 Standard Sip |
10:55AM |
2 |
H323 audio problem |
10:20AM |
1 |
Asterisk on 64 bit... and testing e164.org's stuff |
9:46AM |
2 |
Dead Budgetone-101? |
8:10AM |
3 |
Multiple X100P in Asterisk box? |
7:55AM |
1 |
General advice on confs and setup for new users |
3:06AM |
3 |
Asterisk on 64bit ? |
2:20AM |
6 |
Hong Kong VOIP Exchange |
2:09AM |
1 |
Why? oh why can't I dial out? |
1:17AM |
1 |
Confused with CallerID when using the iax chanenls |
12:49AM |
1 |
asterisk addon mysql |
|
Saturday June 26 2004 |
Time | Replies | Subject |
8:52PM |
2 |
Newbie needs help |
7:35PM |
0 |
Broken Pipe? |
12:03PM |
1 |
Echo worse after new echo patch |
12:01PM |
2 |
ZyXEL Prestige 200w - should I return it ? |
5:40AM |
0 |
I need DIDs in Canada and USA with roll over option |
4:58AM |
2 |
Asterisk Eating Digits |
4:49AM |
1 |
IAX & FWD, No authority found? |
2:37AM |
2 |
Nightly / Daily, Weekend / Weekday and Holiday regime of the Asterisk |
2:29AM |
1 |
How to transfer call in case that I am the originator |
2:09AM |
1 |
Setting up your own menu like voice mail |
|
Friday June 25 2004 |
Time | Replies | Subject |
9:08PM |
2 |
Problems Compiling and Loading asterisk-oh323 0.6.2 |
7:48PM |
0 |
3-way calling woes... Nasty static and inconsistent flash detection? |
3:31PM |
1 |
chan_sip.c max number of retries |
2:17PM |
2 |
Can one send CLID NAME over PRI? |
2:05PM |
0 |
- eezeeFone.com - Need to connect 500 Simultaneous users - An opportunity from test bed to a product. |
1:49PM |
0 |
Using *0 with Asterisk |
1:18PM |
0 |
Ring voltage on a TDM400 |
1:16PM |
0 |
Asterisks RTP source address binding |
12:51PM |
3 |
Using Soxmix on extensions.conf |
12:31PM |
2 |
panic() panic() panic() |
12:28PM |
0 |
RE: H.323 - NO AUDIO IN BOTH DIRECTIONS |
12:25PM |
1 |
SER and NAT |
12:09PM |
0 |
ATA186 (sip) in * dynamic mode |
11:43AM |
1 |
503 "Unavailable" |
11:26AM |
1 |
SS7 status report 2 |
10:40AM |
0 |
Stable branch usable? Development branch better? |
10:18AM |
3 |
Termination Provider |
9:50AM |
1 |
Polycom IP 500 - Quality Issues |
8:49AM |
1 |
SIP extension outside of IP tables firewall |
8:32AM |
2 |
Asterisk & SIP |
7:55AM |
1 |
IAX2 authentication confusion |
7:54AM |
6 |
NO AUDIO IN BOTH DIRECTIONS |
7:52AM |
2 |
forced ring on dial? |
7:24AM |
9 |
SS7 to Pri |
7:00AM |
0 |
SIP/IAX to PSTN setup time |
6:56AM |
0 |
problems compiling shadydial-asterisk on gentoo |
5:00AM |
0 |
HT286, fax and FXS impedance for Europe? |
4:00AM |
1 |
Howto: Use setgroup, checkgroup to check incoming and outgoing client limits |
3:57AM |
4 |
Failure in RTP streaming |
2:56AM |
0 |
Provide Open Settlement Protocol capability |
2:16AM |
2 |
Latest CVS fax detection & grandstream bug |
1:59AM |
0 |
wcfxs CPU usage |
1:36AM |
0 |
Bridging two calls together with Eicon card - help please :) |
|
Thursday June 24 2004 |
Time | Replies | Subject |
10:31PM |
1 |
Latest CVS, Grandstream and Zaptel bug? |
7:40PM |
2 |
Problem with music on hold... |
4:55PM |
0 |
Zaptel ZT_CHANCONFIG failed on channel 1 |
4:37PM |
0 |
Difference between Tormenta/Zapata and Digium Hardware |
4:15PM |
0 |
Inbound call handling in Asterisk |
4:12PM |
0 |
Wildcard X100P dial out troubles |
1:47PM |
1 |
Leave one call to pick up another |
1:42PM |
0 |
New changes |
1:28PM |
5 |
chan_capi problem - hangup??? |
12:56PM |
0 |
Detect more than one type of DTMF for calls to voicemail |
12:18PM |
4 |
host=dynamic vs host=xxx.xxx.xxx.xxx |
11:54AM |
1 |
Record call from switch using service observe? (execute command after dial?) |
11:15AM |
1 |
Cisco ATA 186 from iconnecthere, locked? |
10:37AM |
4 |
toll access - account code |
9:10AM |
7 |
X101P on a UK BT line ---- txgain issue |
9:08AM |
1 |
Pulver's WiSIP with Linksys WAPs |
6:05AM |
4 |
Asterisk with PostgreSQL |
5:56AM |
1 |
Asterisk Manager Commands - Timeout |
5:53AM |
2 |
Asterisk bypassed for name but not number - softphone |
5:39AM |
2 |
R: R: R: How to force G729 |
5:36AM |
1 |
Dead air on 7960 sip at start of call. |
5:03AM |
2 |
Video/H323/SIP |
4:35AM |
0 |
Anonymity and Privacy headers |
4:30AM |
0 |
ZAP hangup not working with siemens HICOM |
4:02AM |
1 |
ZyXEL Prestige 2000W and DTMF |
3:58AM |
0 |
SIP clients, H323 client as gateway? |
3:37AM |
2 |
Help with chan_capi |
2:47AM |
1 |
R: R: How to force G729 |
2:41AM |
0 |
-- Serious issues with current CVS? |
2:19AM |
6 |
R: How to force G729 |
1:52AM |
1 |
Swissv oice IP10 behind NAT |
1:35AM |
1 |
Delay in Zap Calls? |
1:24AM |
0 |
2 E100P cards on one asterisk |
1:22AM |
2 |
How to force G729 |
12:10AM |
0 |
false hangups |
|
Wednesday June 23 2004 |
Time | Replies | Subject |
11:49PM |
1 |
Read this if you use CVS HEAD (Unstable) and chan_capi and jitter buffering in IAX |
11:48PM |
1 |
R: Which Linux ? |
8:53PM |
6 |
Which Linux ? |
8:50PM |
0 |
cdr_mysql: Unknown connection error |
7:26PM |
0 |
#1 Asterisk and Locustworld |
6:09PM |
1 |
FW: No dial tone after installation |
5:54PM |
2 |
Serious issues with current CVS? |
4:39PM |
2 |
tdm (and x100p?) echo - fix is coming! |
4:12PM |
0 |
Problem with Unavailable Message Creation |
2:43PM |
1 |
New VM feature: broadcast and delete=yes |
2:31PM |
0 |
Asterisk info needed for new application development. |
1:56PM |
0 |
UPDATE Patch for postgres enabled app_voicemail.c |
1:55PM |
5 |
Really basic stuff :( |
1:48PM |
0 |
Patch for postgres enabled app_voicemail.c |
1:32PM |
1 |
SIP and audio delay |
12:39PM |
4 |
CDRs, Conferencing, and MeetMe |
12:39PM |
4 |
Codec G729 Registration problem |
12:15PM |
0 |
Digium/Asterisk in Paris |
11:07AM |
0 |
tdm fxo users - new bug tracker entries |
10:52AM |
0 |
Three Way Calling and External Flash Hook |
10:11AM |
2 |
problems compiling zaptel X100P on Redhat Fedora 2.6.5-1.358 |
10:02AM |
0 |
SNOM 200 using GSM Codec dtmf problem |
9:36AM |
1 |
Conference application ! |
9:22AM |
3 |
Voicemail Password Changes Lost on Asterisk Restart |
9:03AM |
1 |
asterisk + appradius & freeradius |
8:52AM |
0 |
Asterisk as a SIP UA and voicemail with SER not working anymore |
8:47AM |
0 |
Conference calling |
8:12AM |
3 |
help needed with read() |
8:04AM |
0 |
clarent hardware |
7:42AM |
2 |
Call Generator for ISDN (PRI/BRI) |
7:17AM |
4 |
X100P Noise |
6:49AM |
1 |
Problem with incominglimit and outgoinglimit |
6:06AM |
0 |
Busy message and extensions are hanging. |
5:12AM |
6 |
Outgoing CLI |
4:33AM |
1 |
Asterisk user/host registration |
3:51AM |
1 |
Codecs and pauses |
3:50AM |
0 |
connecting to Iconnect here using asterisk |
3:33AM |
1 |
Problem when dialing in manager terminal |
3:26AM |
1 |
capi.so problem on startup |
3:13AM |
1 |
USB handset for IAX softphone ? |
3:08AM |
0 |
general install?? |
2:56AM |
5 |
Skype 4 Linux |
2:46AM |
1 |
Iax unable to transfer |
2:30AM |
1 |
Ireland PSTN Number |
2:22AM |
0 |
Accountcode missing in log |
2:19AM |
0 |
CSV log stopping |
2:06AM |
0 |
Réf.: Call generator |
1:46AM |
4 |
Call generator |
1:14AM |
1 |
cdr_mysql compilation error |
12:42AM |
4 |
Future WinCE IP Phone |
|
Tuesday June 22 2004 |
Time | Replies | Subject |
11:31PM |
1 |
Asterisk -- PBX Do Not Disturb |
9:29PM |
2 |
Cisco ata-186 port died |
6:49PM |
0 |
No dial tone after installation |
6:46PM |
0 |
DID in Fiji |
4:17PM |
2 |
sidetone noticeably loud on analog handsets on T100P |
3:33PM |
2 |
Multiple DTMF digits on 7960 |
3:19PM |
3 |
Asterisk Caller ID Application (win32) |
3:09PM |
3 |
Asterisk answering only one (dialed-) Number on a PTMP (German "Mehrgeräteanschluss")? |
3:09PM |
2 |
Problem with Asterisk |
2:58PM |
2 |
Two SIP servers communicating without IAX |
2:49PM |
1 |
Core Dump on app_dial.c |
2:28PM |
0 |
Accessing ISDN with avm bluetooth hardware |
2:21PM |
1 |
AstriCon Registration Opens Next Monday, June 28th |
2:01PM |
1 |
No such extension ... |
1:47PM |
3 |
Unify Incoming and Outgoing sound files |
1:12PM |
2 |
FXO impedance matching |
1:00PM |
2 |
Eicon Diva 2.0 PCI ISDN Card |
12:55PM |
0 |
Modified Prepaid App Database error |
12:25PM |
0 |
Ringing to some numbers... |
12:18PM |
1 |
AgentCallbackLogin - invalid extension |
11:22AM |
0 |
Queueing and parked calls |
11:16AM |
2 |
Failover of IAX or Spillover as the case may be |
11:00AM |
0 |
Users do not disconnect |
10:38AM |
1 |
Weired Probelm with Asterisk |
9:48AM |
6 |
*69 |
8:21AM |
2 |
iax.conf : what is the purpose of trunk ? |
7:44AM |
5 |
CISCO 7960 Goes missing |
7:43AM |
2 |
Unable to find libiodbc.so.2 |
7:36AM |
0 |
patlooptest |
7:35AM |
0 |
Tricks for Multiple TMD0xB cards? |
7:22AM |
1 |
Problems compiling cdr_odbc.so |
6:59AM |
1 |
Call forwarding and voicemail |
6:52AM |
0 |
2 T100P cards - 2 switch types |
6:41AM |
0 |
zapata initial context question |
6:34AM |
3 |
IAX2 Trunking help! |
6:23AM |
3 |
License and Commercial Use |
6:06AM |
0 |
swissvoice ip10s firmware? |
5:45AM |
0 |
Re: [Asterisk-Dev] Skype support |
5:21AM |
1 |
Eliminating silence suppression(?) on IAX2 calls |
4:56AM |
1 |
Unable to create channel - CVS Broken? |
3:54AM |
0 |
exten => i ???????? |
3:46AM |
2 |
Any echo issues with phones from TDM400P > X100P |
3:26AM |
0 |
Site changes |
3:23AM |
1 |
using 2 single pri cards on 1 server |
1:25AM |
1 |
No Caller ID from FXO Problem |
12:30AM |
2 |
pwlib compile error |
|
Monday June 21 2004 |
Time | Replies | Subject |
10:20PM |
0 |
Call forwarding code |
9:26PM |
8 |
Busy message |
9:02PM |
2 |
Failover Trunking Won't Fail Over |
6:29PM |
1 |
OpenSS7 T400P-SS7 and Digium T400P |
6:26PM |
0 |
dialplan help!-RESOLVED |
6:04PM |
2 |
Problems with Zaptel |
5:41PM |
1 |
VoiceXML support and integration |
5:19PM |
0 |
SLC-96/TR-08 Support with T100P? |
5:06PM |
1 |
IAXTel Help |
4:21PM |
0 |
IAXtel questions |
1:16PM |
3 |
Asterisk<>X100P<>Packet8 |
12:57PM |
2 |
Connect 16 E1/T1 between * and other switch... |
12:42PM |
0 |
SpanDSP Fast carrier Failed |
12:19PM |
4 |
integrating with existing PBX |
11:46AM |
0 |
Strange * hangup issue |
11:32AM |
0 |
call forwarding question |
11:23AM |
1 |
Siemens Optipoint 400 SIP Problem |
11:21AM |
0 |
Asterisk As A Career? |
10:17AM |
0 |
Directory dial by name |
9:23AM |
0 |
Error compiling festival |
9:19AM |
3 |
Caller ID double quotes |
9:13AM |
2 |
PRI & immediate=no |
7:48AM |
0 |
A Callback AGI script |
7:40AM |
0 |
R: Re: cdr_addon_mysql compiling error |
7:25AM |
1 |
using # to end a number |
7:13AM |
0 |
mandrake and zaptel |
5:22AM |
2 |
app_dial broken |
4:42AM |
1 |
R: Re: cdr_addon_mysql compiling error |
3:42AM |
1 |
Channel bank problem via long cable |
3:00AM |
0 |
Queue Stats - Management App? |
2:56AM |
0 |
Restricting outbound dialing on a specific p hone |
2:53AM |
2 |
Restricting outbound dialing on a specific phone |
2:48AM |
4 |
disabling ALERTING message |
2:14AM |
1 |
Problem compiling fax applications |
12:15AM |
0 |
Re: Asterisk-Users digest, Vol 1 #4230 - 13 msgs |
|
Sunday June 20 2004 |
Time | Replies | Subject |
10:49PM |
0 |
Modified Prepaid database |
8:52PM |
1 |
please mail me wave.cc and tts.scm |
6:28PM |
1 |
Sipura config |
6:09PM |
0 |
Question - TDM40B - Hunt Group Possibility?? |
6:03PM |
3 |
Need different contexts for 2 X100P FXO Cards and forwarding calls |
5:45PM |
1 |
Data over Voice through Asterisk |
4:03PM |
2 |
Harald Baron/EBAROH/CH/Ascom ist nicht anwesend. |
3:55PM |
1 |
No config file? |
3:16PM |
1 |
asterisk console mode |
3:14PM |
1 |
Grandstream HT-286 // Custom Ring Tones |
2:51PM |
2 |
Channel Bank Frustrations |
10:59AM |
10 |
One way audio |
9:22AM |
4 |
call waiting from PSTN |
7:45AM |
7 |
Date Time Stamp with Caller ID |
7:23AM |
0 |
Grandstream HT-286 and DTMF |
5:49AM |
1 |
chan_oh323: busy not correctly signalled |
5:33AM |
0 |
Asterisk rxfax(): One page gets two pages |
3:11AM |
0 |
BT Broadbandvoice ATA186 and * |
1:14AM |
1 |
Softfax/spandsp Makefile.patch rxfax/txfax |
|
Saturday June 19 2004 |
Time | Replies | Subject |
6:40PM |
1 |
RxFax problems |
2:59PM |
1 |
HST Saphir with Asterisk |
12:10PM |
0 |
Mediatrix 1204 Incoming calls |
9:54AM |
0 |
Directory function is not working |
9:32AM |
0 |
Hard Coded CLASS Codes (was 11 instead of Star) |
9:27AM |
0 |
Fw: #asterisk is +r now, meaning register your nick with nickserv |
5:28AM |
0 |
chan_modem dialout |
4:40AM |
0 |
Busy when not registered |
12:34AM |
5 |
Big problem with Flash |
|
Friday June 18 2004 |
Time | Replies | Subject |
11:15PM |
2 |
Fax with SPA-2000's? |
9:10PM |
3 |
WaitExten substitute |
9:05PM |
2 |
current code release & chan_sip problem/question rport |
7:40PM |
0 |
New Skinny/chan-sccp release |
5:23PM |
0 |
#asterisk is +r now, meaning register your nick with nickserv |
5:12PM |
0 |
not getting sound from chan_oss paging setup |
4:53PM |
1 |
using asterisk as sip registrar is not working for me |
4:26PM |
0 |
SIP error 407 - can't make outgoing calls |
3:06PM |
2 |
Testing UK emergency dialing and LCR. |
3:03PM |
1 |
app_prepaid NAT issue |
1:58PM |
0 |
enhanced privacy manager AGI |
1:27PM |
5 |
Problems with faxing via TE405P/Asterisk |
1:16PM |
2 |
cdr_addon_mysql compiling error |
1:14PM |
0 |
R: Thousands of contexts? |
1:13PM |
4 |
Grandstream CFG file generator |
1:10PM |
2 |
Asterisk References |
12:53PM |
0 |
cdr mysql amaflags field |
12:00PM |
1 |
Iaxy issue |
11:53AM |
0 |
Fwd: Re: Disable IAX1 Registrations |
11:48AM |
0 |
cisco 924 config |
11:31AM |
1 |
Grandstream HT-286 and NAT |
10:04AM |
1 |
Asterisk as Media Gateway (was: ATT CallVantage & Asterisk) |
9:43AM |
3 |
Thousands of contexts? |
9:27AM |
1 |
Lingo and * |
9:06AM |
1 |
trouble compiling zaptel-0.9.1 on YellowDog (PowerMac) |
8:36AM |
5 |
UK install |
8:18AM |
0 |
Possible chan_skinny problems - no ringtone, no moh and no queue messages |
8:12AM |
2 |
C7960 g729 question |
7:52AM |
1 |
X100P in Switzerland |
7:30AM |
0 |
ATT CallVantage & Asterisk |
7:16AM |
1 |
Hwo to get CallerID: SIP -> ISDN |
6:57AM |
5 |
Problems with X100P |
6:23AM |
0 |
FXO Issues - Sorry |
6:19AM |
2 |
FXO Issues |
5:03AM |
3 |
TE410P / Eicon PRI |
4:02AM |
0 |
bri-stuff with current CVS head |
4:00AM |
0 |
Asterisk does not start when cdr_odbc ist configured |
3:52AM |
0 |
Asterisk and CISCO Gateway |
2:37AM |
0 |
Problems reciving fax with Asterisk |
1:06AM |
0 |
Poopy errors on quad wcfxo |
1:04AM |
0 |
Asterisk command |
1:03AM |
0 |
problem number analize |
12:21AM |
1 |
Draytek Vigor 2600Vi as SIP client on Asterisk |
|
Thursday June 17 2004 |
Time | Replies | Subject |
11:46PM |
3 |
IAX Jitter Buffer |
6:09PM |
0 |
Mediatrix 1204 Mibs |
5:45PM |
0 |
Zap Dial Problem ---- Erroneous dash |
5:43PM |
2 |
IAXy and bandwidth requirements |
5:42PM |
4 |
7960 straight through? |
4:53PM |
0 |
dialtone stop |
4:22PM |
0 |
zaptel - make config |
3:16PM |
1 |
trying to set an internal ivr |
2:30PM |
6 |
Compiling problem on Debian |
1:37PM |
2 |
How can i get the last codec_g729.so |
1:19PM |
0 |
Re: SJphone registration problem - Help! |
1:05PM |
0 |
snom phone with asterisk and vocal |
12:56PM |
1 |
Having problems with Agents and calls going to voicemail |
11:57AM |
7 |
TDMoE Question |
11:24AM |
1 |
Disable IAX1 Registrations |
10:21AM |
0 |
mgcp/T1 interface/alternatives |
10:20AM |
2 |
BT Caller ID - From Patch ? |
9:57AM |
1 |
VOIP to Cellular |
9:55AM |
1 |
Zap dropping calls |
9:13AM |
0 |
Terminating VoIP calls with Asterisk |
9:09AM |
0 |
Resend to correct graphic - Internet Talk Radio use Talk Show PBX |
8:40AM |
1 |
VOIP wiretapping article |
8:36AM |
0 |
Port numbers for traffic shaping |
8:20AM |
2 |
How to let users change Voice Mail password in Asterisk |
8:18AM |
0 |
Problem with bridging two external lines |
8:03AM |
3 |
SJphone regestration problem - Help! |
7:44AM |
1 |
Blank faxes with RxFAX |
7:32AM |
4 |
Asterisk as Internet Talk Radio PBX system |
7:07AM |
3 |
asterisk-addons compilation error |
5:23AM |
0 |
Zapata.conf & Signaling for Bulgaria (PSTN: Siemens PABX) |
4:35AM |
4 |
SFTP |
4:19AM |
2 |
HFC ISDN card with bristuff from jung hanns.n et? |
3:54AM |
3 |
Cheap (US$120 or less) SIP Phones |
3:49AM |
1 |
Anyone have experience with chan-capi in Australia? |
3:42AM |
1 |
HFC ISDN card with bristuff from junghanns.n et? |
3:40AM |
1 |
HFC ISDN card with bristuff from junghanns.net? |
2:28AM |
4 |
Problems with PRI with T410 messages |
1:41AM |
2 |
LDAP synchronization script |
1:12AM |
1 |
Calling the firefly network? |
12:43AM |
0 |
Accepting SIP calls from unregistered gateways |
12:27AM |
1 |
pri with TE410P not working (Austria) |
12:12AM |
0 |
no audio with sip |
|
Wednesday June 16 2004 |
Time | Replies | Subject |
11:23PM |
2 |
IAX2 no compatible codecs |
10:57PM |
0 |
Disable authentication on outgoing SIP calls |
10:14PM |
1 |
RxFax - Fast carrier training failed |
9:49PM |
0 |
S100U USB FXS problem |
7:55PM |
0 |
D-Link DVG-1120M and *. |
7:35PM |
3 |
911 emergency service and VoIP |
5:16PM |
0 |
(no subject) |
5:15PM |
0 |
Changing the asterisk timezone |
4:13PM |
1 |
Modified Prepaid Error |
4:05PM |
2 |
embedded Asterisk |
3:01PM |
0 |
Size of box for 4xE1 conf bridge? |
2:19PM |
4 |
UIP200 |
1:55PM |
3 |
ZAPHFC - only for * 0.7.2? |
1:49PM |
5 |
Failed to authenticate on INVITE |
1:23PM |
2 |
playinterruptibletones |
1:05PM |
1 |
festival with asterisk problem |
1:03PM |
3 |
X-Lite/Firefly behind NAT connecting to Asterisk not receiving RTP |
12:26PM |
4 |
Soekris Engineering net4801 |
11:23AM |
1 |
IAX registration |
11:20AM |
1 |
limitations ? |
11:18AM |
6 |
Cost of IP Phones, or Isn't It Just Software? |
10:20AM |
1 |
ATA186 v3.1 SIP - Attended transfer: NO JOY |
8:40AM |
0 |
Cisco DSP Modules and Linux |
8:15AM |
4 |
Status-info 1: Signalling C7 / SS7 |
7:56AM |
1 |
NAT and Qualify Question |
7:32AM |
6 |
Invalid Extensions -- More like traditional PBX systems? |
6:53AM |
1 |
replacing cisco callmanager with asterisk? |
6:14AM |
3 |
BT101 and caller id and web interface |
5:44AM |
0 |
Re: Approved |
5:18AM |
1 |
VOIPTalk silver service |
3:46AM |
1 |
Remote rebooting a Cisco 7940 |
3:30AM |
0 |
asterisk server hang up after conference |
3:20AM |
4 |
Digium X100P vs Dodgy Ebay X100P |
3:16AM |
0 |
Problems with Call Forwarding on a 7960 |
2:32AM |
0 |
Problem with incoming calls from FXO |
2:06AM |
1 |
Asterisk hardware configuration and cost? |
12:44AM |
1 |
error loading meetme module |
12:38AM |
1 |
Fedora2 and Kernel 2.6 again! |
12:19AM |
1 |
asterisk/netmeeting works, asterisk/ohphone doesn't? |
|
Tuesday June 15 2004 |
Time | Replies | Subject |
11:33PM |
0 |
how can I catch |
6:11PM |
1 |
Choppy sound ONLY when a voicemail is left |
4:50PM |
7 |
Voicepulse Down Again? |
2:36PM |
3 |
anyone use mailboxexists? |
2:29PM |
1 |
sip register and nat |
2:23PM |
0 |
sip.conf - register and peer groups |
1:18PM |
0 |
IVR Prompt errors (scratchy) |
1:17PM |
0 |
RE: send pstn calls to cisco gateway ? |
12:53PM |
0 |
TDM400P FXO problems |
12:25PM |
0 |
Excluding DIDs from telco long distance codes |
12:06PM |
0 |
app_conference Compile |
11:35AM |
3 |
Grandstreams randomly go busy with Asterisk? |
10:58AM |
2 |
Multiple X100Ps -- order? |
10:19AM |
0 |
making * more like a normal pbx (ciscoata-186) |
7:39AM |
2 |
using SetCDRUserField in an AGI script |
7:28AM |
0 |
SIP Registration with Entice Softswitch |
6:55AM |
0 |
Re: OT: fax obsoleted? Was: Re: Fax via email (Steve Underwood) |
5:59AM |
0 |
Re: OT: fax obsoleted? Was: Re: Fax via email (Steve Underwood) |
5:59AM |
2 |
Cdr_addon_mysql.c compile problem. |
5:55AM |
2 |
Polycom IP 600 Programmability |
5:35AM |
5 |
Capi problems |
5:29AM |
2 |
Re: OT: fax obsoleted? Was: Re: Fax via email (Steve Underwood) |
5:09AM |
5 |
building asterisk |
3:38AM |
0 |
how can I catch How to catch some incoming call |
3:22AM |
0 |
Siemens Optipoint 400 standard SIP |
3:15AM |
0 |
Simultaneous UA use of services |
2:49AM |
0 |
Trunk ? |
2:27AM |
5 |
PRI problems (telewest -> * -> LG GDK 186) |
1:35AM |
3 |
Queue then Voicemail |
1:30AM |
3 |
Outgoing DTMF when using BRI & i4l (Eicon Diva) - problems |
|
Monday June 14 2004 |
Time | Replies | Subject |
11:46PM |
3 |
No B-Channels. PRI. E100P. HELP! |
11:33PM |
0 |
pulse dialing |
8:50PM |
4 |
IAX2 hangup on transfer |
8:03PM |
9 |
Asterisk-Users List Etiquette |
4:34PM |
1 |
making * more like a normal pbx (cisco ata-186) |
4:33PM |
2 |
inviting an spa-x000 |
3:46PM |
1 |
chan_h323 no audio both ways |
3:40PM |
1 |
IAX and Reorder |
2:50PM |
1 |
Cisco SIP Phone Licensing |
2:15PM |
0 |
CLEC / SIP interconnection? |
1:38PM |
1 |
telephones to use with asterix |
1:36PM |
0 |
ast_data, mysql, md5 hashes for passwords |
1:34PM |
2 |
International Talking Clocks |
1:10PM |
1 |
MailboxExists application |
12:25PM |
1 |
Multiple tennants, two DIDs, One IAX provider |
10:41AM |
0 |
Nextel phone and mute on Asterisk? |
10:41AM |
4 |
Sipura 2000 not answering em_w calls |
10:14AM |
4 |
Number Portability and VoicePulse |
10:08AM |
0 |
compile error with asterisk-addons |
10:02AM |
0 |
do_monitor warning message |
9:44AM |
4 |
german localization for mailbox available? |
9:24AM |
4 |
Polycom IP 600 |
9:13AM |
2 |
Asterisk real life examples and case studies ? |
8:43AM |
0 |
Canadian DID |
8:33AM |
0 |
If IAX client is not logged in/registered, Dial plan executes BUSY vs UNAVAILABLE |
8:27AM |
0 |
T1 - Adtran and SIP |
7:42AM |
1 |
Chan_Capi 0.3.4 |
6:55AM |
1 |
ASTERISK V. SER |
5:58AM |
5 |
Prepaid application error |
5:29AM |
2 |
where can I get toll-free number? |
5:17AM |
4 |
<<< GSM Audio Files >>> |
4:46AM |
0 |
Asterisk as MGCP endpoint |
3:58AM |
1 |
Festival application: clipping start of sound? |
3:37AM |
3 |
<<< GSM AUDIOFiles >>> |
3:10AM |
1 |
TE410P in Austria |
2:54AM |
2 |
making * more like a normal pbx |
2:49AM |
0 |
FXS--->SER---><Asterisk>--->FXO--->PSTN |
2:02AM |
1 |
Install Question |
1:55AM |
0 |
Accepting post selection digits over isdn trunking |
1:06AM |
15 |
oh323 |
12:45AM |
7 |
collaboration with Panasonic PBX |
|
Sunday June 13 2004 |
Time | Replies | Subject |
9:08PM |
2 |
SIP audio cut off even with Answer, Wait... |
5:53PM |
2 |
Comfort Noise |
5:46PM |
1 |
Asterisk Agent Logoff? |
5:37PM |
0 |
errors on startup |
5:30PM |
1 |
Strange voicemail things |
5:25PM |
0 |
Help Wanted |
4:11PM |
0 |
Red alarm on T1 PRI but not on zttool |
2:24PM |
0 |
sip.conf => Configuration of Asterisk with siproxd ? |
2:16PM |
0 |
Spanish, Portuguese, other recordings for Allison |
2:00PM |
2 |
Wiki now based on CVS head |
1:03PM |
1 |
831/408 iax termination |
12:42PM |
2 |
Cisco 7960 Problem |
11:05AM |
2 |
Sayson IP Phones? |
7:35AM |
0 |
DIAX 0.9.8c available for download |
5:30AM |
1 |
Re : Newbie help ! |
4:58AM |
4 |
*** Asterisk Sunday News: Off track with 1.0, moving forward |
12:36AM |
2 |
Is nufone web site down? |
|
Saturday June 12 2004 |
Time | Replies | Subject |
10:08PM |
3 |
(no subject) |
5:48PM |
0 |
ASTTAPI 0.03 hangup not working |
5:31PM |
2 |
Junghanns QuadBRI stable? |
1:55PM |
1 |
Asterisk on FreeBSD News |
11:52AM |
2 |
DECT delay once hungup |
10:41AM |
1 |
Problems with Alcatel Speedtouch ST280 |
9:20AM |
1 |
Call Relaying |
8:57AM |
2 |
Sending SABME continuosly. Urgent help needed! |
8:47AM |
1 |
'background' problem |
8:40AM |
1 |
Changed IP and subnet now no SIP Register 403 |
7:30AM |
1 |
Capture user input |
6:30AM |
5 |
MWI on Cisco ATA-186 (SIP) |
6:19AM |
0 |
Problem with E1 |
6:12AM |
1 |
Cisco ATA-186 Firmware upgrade |
1:16AM |
4 |
2 NuFone lines- which one to dial out on |
1:14AM |
9 |
Prepending for 9NxxNxxx - adding the area code for 7 digit dialing |
|
Friday June 11 2004 |
Time | Replies | Subject |
10:19PM |
3 |
DID/T1 |
9:59PM |
1 |
* as conference server for shoutcast. |
6:46PM |
0 |
GUI Design Ideas Request |
6:29PM |
0 |
Service in 252-255 |
6:04PM |
7 |
BudgeTone hold? |
5:54PM |
3 |
ssh key problem |
4:38PM |
1 |
Broadvoice conf |
2:59PM |
2 |
extensions question |
2:16PM |
1 |
oh323 0.6.2 |
2:03PM |
15 |
Voicemail problem |
1:56PM |
4 |
Cisco 7940 |
1:36PM |
0 |
Newbie to SJphone |
1:06PM |
1 |
catch when no voicemail configured |
1:00PM |
0 |
context of a transfer |
12:21PM |
2 |
cdr_addon_mysql.c |
11:16AM |
0 |
CDR not always correct / IAX clients unmonitored |
10:30AM |
1 |
Exit Voicemail to VoicemailMain? |
10:03AM |
1 |
Integration with SIEMENS HIPATH PBX |
9:04AM |
0 |
SIP->Application Codec debugging |
8:54AM |
3 |
Simplified Voicemail app / keeping peace with cohabitants |
7:38AM |
0 |
Problem with AGI |
7:17AM |
4 |
Cisco Auto Provisioning |
6:21AM |
1 |
direct dial-in (DDI) |
6:16AM |
2 |
Asterisk newbie help !! |
6:06AM |
11 |
Broadvoice and DTMF |
5:51AM |
1 |
QuadBRI outgoing call problem. |
5:48AM |
0 |
Aggressive Echo Suppression |
5:37AM |
6 |
phone calls betweens phones behind the same nat |
4:38AM |
3 |
Background Playback fails |
4:25AM |
1 |
trunk=yes with recent CVS head problems |
4:22AM |
1 |
CLI messages screwy? |
3:49AM |
0 |
R: VoipTalk down? |
3:46AM |
0 |
VoipTalk down? |
3:05AM |
2 |
Asterisk PRI calls to SER problem |
2:48AM |
1 |
7960 switch port / vlan issue |
1:35AM |
2 |
R: hide caller id |
1:24AM |
0 |
dialing several phone numbers in one call session. |
1:16AM |
1 |
"Caller ID" question |
|
Thursday June 10 2004 |
Time | Replies | Subject |
11:55PM |
0 |
hide caller id |
11:38PM |
0 |
Re: Asterisk-Users digest, Vol 1 #4101 - 12 msgs |
11:13PM |
0 |
PC-to-PC call though SIP Proxy |
10:07PM |
1 |
Intel 537EP chipset, revisited |
7:19PM |
1 |
RE: question about prepaid app_prepaid |
6:54PM |
4 |
XML How To for Cisco 7960 |
6:37PM |
2 |
Guest IAX with Dynamic IP |
4:21PM |
3 |
A couple of newbie questoins |
3:46PM |
0 |
Grandstream Ringtones on a per phone basis |
3:30PM |
1 |
Uniqueid changing with call parking |
3:00PM |
0 |
Missing connect indication on pri? |
2:48PM |
3 |
Cisco 7970 w/ 7.1 phones rebooting with asterisk |
2:40PM |
4 |
How to get the Called id with AGI |
2:00PM |
2 |
New faxdetect change in dsp.c |
2:00PM |
0 |
Would like to ask a * user some question over voice or a walk thru in the Hou,TX area |
12:59PM |
1 |
Manager logic to pickup a ringing extension |
12:56PM |
0 |
Asterisk on Sun Cobalt Qube 3-Ideal system for asterisk |
12:49PM |
1 |
Call originate with manager API |
11:53AM |
0 |
Outbound ZAP calls |
11:18AM |
0 |
oh323 0.6.2 q931 messages |
11:00AM |
0 |
BUG?: reinvite and nat |
10:49AM |
0 |
Asterisk as a VoIP Gateway to an Analog PBX |
10:29AM |
1 |
mysql errors |
10:18AM |
0 |
Re: Problem with * not detecting hangup on FXO and VM going into an infinite, loop |
10:14AM |
0 |
NAT and symmetric fw |
10:11AM |
1 |
Dialing delay when using Zap channels |
9:43AM |
4 |
incoming DTMF on iConnectHere? |
9:18AM |
2 |
BT is moving to IP ONLY |
8:56AM |
0 |
I can't get iaxComm to connect to guest@misery.digium.com |
8:54AM |
2 |
Problem with * not detecting hangup on FXO and VM going into an infinite loop |
8:52AM |
0 |
Cisco 7960 Tones |
8:31AM |
3 |
FW: question about prepaid app_prepaid |
8:27AM |
10 |
Automating calls |
8:15AM |
3 |
Asterisk on Apple PPC with YDL |
7:49AM |
3 |
GSM to ISDN or TAPI |
7:09AM |
0 |
isdn4linux and NT mode |
6:41AM |
3 |
Iax2 ringtone problem |
6:27AM |
1 |
FWIW- Cisco 1750 dropped packets and choppy audio |
5:46AM |
0 |
IAX Binding to 2 nic's for trunking two asterisk servers |
4:26AM |
0 |
Please help !!!! - IAX, MYSQL - Cant make calls |
3:38AM |
0 |
SIP Registration Failed !!(Need Help) |
2:56AM |
2 |
Using Asterix and Hylafax with Eicon DIVA E1 |
2:41AM |
1 |
EU on VoIP |
12:53AM |
2 |
Primustel a.k.a. Lingo $20/month unlimited service |
|
Wednesday June 9 2004 |
Time | Replies | Subject |
11:44PM |
1 |
Changes in VoiceMail |
11:29PM |
0 |
Introduction |
10:36PM |
1 |
Another Firefly update - now with SRV support |
10:00PM |
0 |
No ringing on outbound PRI calls |
6:14PM |
1 |
SIP Registration seems to timeout |
2:52PM |
1 |
PC Mag Online article on Asterisk |
2:13PM |
0 |
Call Pickup problem in Asterisk with SIP phones |
2:11PM |
1 |
IAX Peers from MYSQL |
1:45PM |
1 |
Seperate asterisk VM system possibility |
12:34PM |
0 |
any banks or financial institutions using asterisk |
12:33PM |
0 |
MeetMe and ztdummy problem |
12:10PM |
0 |
failover for voip providers (i.e. Dial() doesn't give enough options) |
11:59AM |
1 |
Using asterisk as voicemail system for SER |
11:52AM |
0 |
IBM T30, Redhat 9, Gnophone, mono PCM, Internet PhoneCard |
11:00AM |
2 |
Mine strangest asterisk problem ever .... |
10:54AM |
0 |
Asterisk voicemail problem |
10:03AM |
0 |
asterisk-addons mysql |
9:51AM |
0 |
Replacing a Cisco Call Manager |
9:29AM |
1 |
Hang-up Supervision (UK) |
9:12AM |
1 |
TE405P PRI B-channel resets |
9:01AM |
0 |
IAX, MYSQL - Rejected connect attempt from |
8:54AM |
0 |
Asterisk PRI messages |
8:46AM |
1 |
Asterisk Receptionist - Lite - CallerID Source code |
8:24AM |
5 |
ISDN BRI with National (north america) Signalling |
7:47AM |
7 |
Dyn Exten |
7:09AM |
2 |
NetworkWorld article on Open Source Telephon y |
2:50AM |
0 |
Zaphfc and Fedora core 1 |
12:17AM |
0 |
curious (and incorrect) caller*id behavior |
|
Tuesday June 8 2004 |
Time | Replies | Subject |
10:20PM |
2 |
Learn To build IVR |
9:14PM |
4 |
AS5300 and Asterisk |
8:59PM |
1 |
HOBIC |
7:21PM |
3 |
Sending # and Asterisk Transfer Conflict |
7:06PM |
7 |
NetworkWorld article on Open Source Telephony |
4:54PM |
1 |
Asterisk CallerID app (win32) |
1:22PM |
0 |
Cisco 7940 doesn't register |
12:54PM |
3 |
SMS in the UK |
11:50AM |
0 |
Call centers using Asterisk |
11:30AM |
0 |
TDM400P hangup / ringing detection problem |
10:27AM |
2 |
HOW-TO DIFF |
10:04AM |
2 |
Don't want a ring before voice menu |
8:40AM |
8 |
New version of DIAX (0.9.8a) available now for free download |
8:38AM |
0 |
Echo problems using AVM Fritz!PCI Card |
8:24AM |
6 |
iaxtel 1-800 gateway down? |
8:19AM |
6 |
CDR for transfered calls |
8:13AM |
0 |
Unable to call other SIP Phone |
7:41AM |
0 |
Camp On configuration? |
7:05AM |
2 |
Integration with a Siemens HiCom 150E / HiPath 3750 |
6:57AM |
2 |
grandstream ringtones - makering.pl usage for 1.0.50 |
5:51AM |
1 |
Outgoing call via Fritz! |
3:42AM |
1 |
E100P R2 signaling |
2:19AM |
1 |
Meetme2 |
1:39AM |
4 |
makering.pl |
1:17AM |
0 |
Is there a problem with iaxtel? |
12:03AM |
0 |
How path latest CVS apps Makefile on order to compile app_rxfax and app_txfax |
|
Monday June 7 2004 |
Time | Replies | Subject |
11:37PM |
1 |
illegal instruction - on Via board |
10:43PM |
2 |
Mediatrix 1204 Configuration |
10:09PM |
1 |
sip device discussion and reviews |
9:34PM |
3 |
dialplan experts needed |
9:25PM |
0 |
Asterisk Receptionist Lite version |
9:24PM |
0 |
asterisk to broadvoice? |
8:27PM |
1 |
Seeking Volunteers for an Intro to Asterisk Course |
8:07PM |
2 |
IAX Won't Pass Caller ID |
5:41PM |
0 |
re: Voicemail and Cisco Phones |
5:33PM |
2 |
chan_capi 0.3.3 compiling error |
5:27PM |
0 |
SIP registration issues - Ugly workaround |
2:35PM |
2 |
slightly OT: VoIP more expensive than Call-By-Call |
1:58PM |
3 |
meetme application |
1:55PM |
0 |
Application possibilities |
1:49PM |
1 |
pseudo zap channel - how to get rid of it ? |
1:30PM |
1 |
Network Sniffing Calls for recording |
1:26PM |
1 |
Voicemail missing playback options |
12:21PM |
1 |
hdlc setup routing question |
12:15PM |
4 |
Modem Calls |
11:43AM |
4 |
Compiling Asterisk with G.723.1 |
11:27AM |
2 |
AGI + g729A |
11:02AM |
0 |
cisco reinvite |
10:10AM |
1 |
Grandstream Codec Order |
9:57AM |
1 |
control which * pbx to use |
9:10AM |
2 |
Problem with rxFax |
8:10AM |
1 |
videosupport = yes -- how to use it? |
8:08AM |
1 |
AVM B1 and PTP mode |
7:42AM |
0 |
DTMF X100p to sip GS |
6:15AM |
0 |
re: Voicemail and Cisco Phones |
6:15AM |
3 |
Fax via email |
5:52AM |
2 |
re: Voicemail and Cisco Phones |
5:30AM |
1 |
Module nonsense (zaptel, wcfxs and wxfxo) |
4:58AM |
1 |
Zaphfc and BRI problems in Portugal... |
4:28AM |
6 |
chan_capi and DDI (Anlagenanschluss) |
3:11AM |
3 |
Voip-talk? |
3:10AM |
0 |
Updated: Advanced German Configuration |
3:08AM |
1 |
Multiple DDI & Hunting on Analog Lines ( UK) |
2:54AM |
0 |
FW: Problem with Asterisk PRI forwarding to SER |
2:42AM |
0 |
(Redirected to -Users) Re: [Asterisk-Dev] load_module error with chan_oh323 |
1:29AM |
2 |
IAX calls dropout on button press |
1:24AM |
3 |
Multiple DDI & Hunting on Analog Lines (UK) |
12:19AM |
1 |
isdn4linux, NETjet, chan_modem help needed |
|
Sunday June 6 2004 |
Time | Replies | Subject |
11:52PM |
2 |
nat=yes |
11:18PM |
3 |
Dial plan help |
8:00PM |
1 |
illegal instruction -via c5 |
6:31PM |
0 |
Incoming calls not showing up in user specific CDRs? |
4:55PM |
0 |
AM-Web working? |
2:02PM |
2 |
BRI In the states |
1:56PM |
2 |
Analog Bridged Calls Pulsate |
1:38PM |
5 |
Zapata? |
7:47AM |
1 |
Incoming call voice data |
3:42AM |
0 |
*** Asterisk Sunday News: The SIP NAT Special |
|
Saturday June 5 2004 |
Time | Replies | Subject |
11:39PM |
1 |
FXO answering quicker |
6:18PM |
2 |
FWD network from Asterisk through NAT |
12:38PM |
0 |
FW: Meetme with moderator |
11:22AM |
2 |
Configuring cisco 7940 |
10:50AM |
0 |
Re: Asterisk-Users digest, Vol 1 #4041 - 11 msgs |
5:45AM |
0 |
DSP Tools Technical Support |
3:47AM |
0 |
Immediate partial pattern match |
3:13AM |
0 |
change cisco ata 186 dial behaviour |
1:34AM |
1 |
ISDN and incoming MSN |
|
Friday June 4 2004 |
Time | Replies | Subject |
9:35PM |
3 |
illegal instruction |
9:09PM |
0 |
CFDA from cell phone to SIP line in Asterisk PBX |
8:08PM |
3 |
* to Vonage Connection anyone? |
6:55PM |
2 |
Cisco 7960 XML/Configs |
3:50PM |
2 |
CODEC and Fax |
2:59PM |
1 |
Voicemail and Cisco phones: Dialplan example |
1:40PM |
0 |
Appradius Installation |
1:32PM |
0 |
(no subject) |
1:18PM |
3 |
Grandstream 1.0.5.0 Firmware: SIP Register option gone |
12:16PM |
0 |
bitnet niagara presentation - might interest anyone local |
12:07PM |
0 |
Supervision Issue With Asterisk/Sipura/Talkn |
12:00PM |
3 |
QoS in Cisco |
11:53AM |
2 |
Recommendation for sip phone |
11:29AM |
1 |
RE RE: Asterisk Receptionist manager program. |
9:36AM |
9 |
MYSQL asterisk configuration |
9:16AM |
1 |
rxfax crashing asterisk and YES I'm using an approved libtiff :-) |
9:12AM |
0 |
IAX termination in 602 or 520 |
7:25AM |
3 |
Cisco 12 SP+ and Asterisk? |
7:07AM |
2 |
Mystery PRI NOTICEs & WARNINGs |
7:07AM |
0 |
miserable time with Cisco ATA 186 |
6:26AM |
2 |
Help, Ideas and Ready for use Solutions |
5:48AM |
2 |
(possibly) new use for asterisk |
3:00AM |
1 |
ast_log(LOG_DEBUG |
2:16AM |
0 |
Newbie question about dialling PSTN numbers from SIP clients |
1:30AM |
1 |
Strange connection to the outside... |
1:26AM |
0 |
bri stuff Issues |
1:16AM |
1 |
Newbie questions about ISDN&zapata.conf, outbound dialing, TDMoE |
|
Thursday June 3 2004 |
Time | Replies | Subject |
11:22PM |
2 |
Asterisk & fax-out |
7:43PM |
4 |
miserable time with Cisco ATA186 |
2:02PM |
1 |
parking in multiple contexts |
1:45PM |
5 |
Hardware Transcoder |
1:21PM |
1 |
Call Originate from Manager application. |
12:07PM |
0 |
Problem with vmail.cgi |
10:19AM |
1 |
X100P hangup, not available 60 seconds |
9:07AM |
0 |
New ASTGUICLIENT released: 1.0.2 |
8:42AM |
0 |
Agent Groups |
6:43AM |
2 |
Problem with T1 PRI line resetting/dropping calls. |
6:42AM |
0 |
7960 problem call to 7960 |
5:41AM |
3 |
CALLERIDNUM not passed over? |
5:34AM |
0 |
zttest never get 100% accurancy |
4:58AM |
1 |
TE410P Q.931 |
4:40AM |
0 |
Asterisk + E100P in Sweden |
4:31AM |
1 |
Small * issue |
4:23AM |
5 |
Time based calls charging and "reserved" numbers up to 999! |
3:36AM |
3 |
Asterisk & SER (www.IPTel.org) |
3:34AM |
1 |
DSP Coding |
12:43AM |
0 |
Text to speech on Asterisk - AT&T? |
12:26AM |
0 |
Preserving received digits during a fax match? |
12:02AM |
0 |
Any Idea why I am getting one Ring on my Analog Phone attach to Rhino Switch after Hangup |
|
Wednesday June 2 2004 |
Time | Replies | Subject |
9:55PM |
1 |
Cisco VG200 & mgcp |
8:54PM |
1 |
Hot keypad on a Cisco 7960 |
6:18PM |
1 |
(no subject) |
6:14PM |
0 |
WaitforDigit give ring on Analog Phone |
5:54PM |
0 |
MWI for Zaptel / libpri -> BRI |
5:52PM |
0 |
VON Developers Conference |
5:23PM |
1 |
X100P to hardware PBX |
4:52PM |
1 |
oh323: Failed to create smoother |
4:00PM |
1 |
IP Phone with multiple accounts on same instance of asterisk |
3:20PM |
0 |
mgcp.conf reference manual |
2:23PM |
0 |
Polycom SoundPoint IP 300 with Asterisk? |
1:42PM |
0 |
Stutter dialtone on TDM31B (TDM400P) |
1:40PM |
2 |
cisco ata-186 behind NAT |
1:37PM |
2 |
Problems with IAX Clients, HELP ME PLEASE. |
12:14PM |
0 |
3com SIP phone issues |
12:09PM |
2 |
Zapata FXO always answers call? |
12:00PM |
1 |
Where I can find Grandstream v 1.0.4.68 firmware? |
11:52AM |
2 |
HandyTone with Asterisk |
9:40AM |
1 |
H.323 and cause code 'user busy' |
9:36AM |
1 |
DTMF and SIP |
8:45AM |
0 |
SIP and multiple line appearances |
8:15AM |
0 |
ast_rtp_read: Unknown RTP codec |
8:01AM |
3 |
asterisk process respawn |
7:57AM |
5 |
Meetme with moderator |
7:46AM |
5 |
ZyXEL Prestige 2000W SIP hangup fails |
7:44AM |
2 |
Asterisk and Sip/IP Phones |
7:22AM |
4 |
Splicing audio clips into one stream |
6:56AM |
1 |
Problem compiling ZAPTEL on Linux 2.6.6 |
6:44AM |
0 |
FireFly - no sound after first call |
5:44AM |
1 |
Feature request for integrating an OSS (Operations Support System) and Asterisk |
5:41AM |
1 |
isdn configuration |
5:40AM |
3 |
DNS SRV records |
5:10AM |
1 |
Fax Recognizion without Answer? How to Supress this? |
4:30AM |
0 |
Script to import Master.csv in the MySQL database - a short HowTo |
2:43AM |
1 |
Bluetooth headsets/phones. |
2:28AM |
2 |
Asterisk with Ericsson MD110 PBX |
2:25AM |
2 |
"403 Forbidden" between two softphones on same Asterisk |
|
Tuesday June 1 2004 |
Time | Replies | Subject |
10:33PM |
2 |
problems with TDM400P |
9:33PM |
0 |
Asterisk Receptionist |
9:04PM |
1 |
determining cause of dropped calls? |
7:46PM |
2 |
Simultaneous ring internal extension and external phone number? |
7:45PM |
1 |
Feedback needed! FindMe/FollowMe FeatureSpec. |
7:39PM |
5 |
Multi process of * |
7:34PM |
0 |
Message light and paging on Zultys ZIP2, Uniden UIP200 time offset |
7:06PM |
2 |
extra FXS? |
7:04PM |
0 |
MOH From Line In on a sound card |
6:29PM |
2 |
VoIP phones in Australia |
5:54PM |
0 |
free sip termination |
5:49PM |
2 |
iax codec problem |
5:16PM |
1 |
Help in direction |
3:44PM |
0 |
SIP response 488 to special ext/pri? |
2:34PM |
2 |
Syntax for 2 ISDN Cards |
1:14PM |
15 |
Feedback needed! FindMe/FollowMe Feature Spec. |
12:33PM |
1 |
Zap and call pickup -- it don't work. |
12:27PM |
5 |
Adtran TSU 600 |
12:17PM |
0 |
Re: Here |
11:58AM |
2 |
HDLC |
11:38AM |
0 |
Detecting Events in queues |
10:24AM |
1 |
SIP vs. SIP :-( |
10:14AM |
2 |
Router, Firewall, SIP Rewriter, and GnuGK |
10:09AM |
0 |
Presentation, Asterisk support in Montreal |
9:57AM |
0 |
Record Application Problem |
9:51AM |
2 |
BroadVoice usage? |
9:13AM |
1 |
Testers for chan_misdn searched |
9:01AM |
1 |
Difference between native and 3rd party h323 channel driver ? |
8:49AM |
0 |
changing the ip address of an asterisk pbx |
8:27AM |
0 |
System blocked when execute "asterisk -c" |
8:27AM |
5 |
Some (lack of) answers regarding the wakeup call application... |
8:14AM |
0 |
Unsupported Media error from iConnectHere |
7:39AM |
0 |
Call Transfer over Fritz!-ISDN Card with i4l does not work |
7:37AM |
0 |
MGCP Clients |
7:17AM |
2 |
R: Hyperthreading? |
6:41AM |
1 |
ISDN in Venezuela |
6:22AM |
0 |
Variable: in Originate via Manager API |
6:04AM |
0 |
Sipura-SPA2000 background noise |
5:32AM |
0 |
short delay before voice starts after ring |
3:49AM |
1 |
E100P isdn pri installation |
3:01AM |
3 |
Controlling SIP mobile extensions. |
2:11AM |
1 |
@mydomain.com |
1:35AM |
9 |
Hyperthreading? |
1:22AM |
1 |
D-Link DPH-100S |
1:21AM |
1 |
Stuck SIP channels? -> SIP show channels |
12:36AM |
0 |
Réf.: RE: SIPP Load testing |
12:33AM |
0 |
australian enum... |
12:23AM |
2 |
E1 Connection breaks |