| Thursday March 31 2005 |
| Time | Replies | Subject |
|---|
| 11:38PM |
2 |
Ztdummy is Loaded but Asterisk is not using it |
| 10:51PM |
1 |
Caller ID on voicemail messages |
| 10:36PM |
2 |
VOIP to the PBX |
| 10:09PM |
1 |
NuFone, VoIPJet, circuit (fast) busy question |
| 9:02PM |
0 |
Setting Up @Home 0.8 Guide |
| 8:11PM |
0 |
Calls to outside lines |
| 8:02PM |
2 |
Problem with livevoip dial out |
| 7:09PM |
0 |
Czech ISP Problem Connecting |
| 6:48PM |
1 |
Asterisk - MAX TNT T3 VoIP and PPP T1 |
| 5:53PM |
1 |
auto navigate external IVR |
| 5:46PM |
1 |
sangoma S508/FT1 ISA |
| 5:06PM |
2 |
Integrating door intercom? |
| 5:06PM |
0 |
IAXtalk |
| 4:30PM |
2 |
Some phones need about 11 second before they ring |
| 4:28PM |
1 |
additional fields in Realtime |
| 4:25PM |
4 |
Webmin |
| 3:07PM |
0 |
dialparties.agi |
| 2:43PM |
3 |
Preserve g729 registration over reinstall?? |
| 2:04PM |
10 |
Livevoip still no DTMF? |
| 1:50PM |
2 |
Timecard application |
| 1:42PM |
9 |
Polycom sound quality problems |
| 1:39PM |
1 |
MFC/R2 in the Philippines |
| 1:23PM |
2 |
Enhanced Queue App Revisited |
| 1:22PM |
1 |
Can this be done? |
| 1:16PM |
0 |
Guidelines for sizing hardware... |
| 12:47PM |
0 |
DVG-1120S |
| 12:41PM |
0 |
Manager API calling DUNDiLookup |
| 12:38PM |
0 |
Calling DUNDiLookup from manager api |
| 12:29PM |
0 |
Are there online forums instead of thisemailforum?? |
| 11:50AM |
2 |
System requirements as users grow... |
| 11:46AM |
1 |
patlooptest: Usage, setup? |
| 11:12AM |
1 |
Installing CAPI |
| 11:07AM |
3 |
Online forums vs email list... |
| 9:32AM |
2 |
ATCOM Gateways AG-168, AG-248, AG- 468 |
| 9:07AM |
2 |
Asterisk compatible IP Phones |
| 9:03AM |
3 |
Phones "Callwaiting" enable by default? |
| 9:01AM |
0 |
Customized ring tones |
| 8:52AM |
6 |
Are there online forums instead of this emailforum?? |
| 8:44AM |
0 |
agent and queue autologoff |
| 8:36AM |
3 |
Echo on internal SIP |
| 8:26AM |
23 |
Are there online forums instead of this email forum?? |
| 8:04AM |
7 |
Asterisk-1.0.7 Build - Serious issues |
| 7:46AM |
0 |
one way audio with X-lite for Linux/Suse 9.2 |
| 7:32AM |
1 |
chan_capi looking for missing channel_pvt.h |
| 7:19AM |
1 |
Asterisk Realtime - configuration help |
| 7:18AM |
2 |
Business Opportunity for Australia |
| 7:16AM |
1 |
ser, asterisk and conferencing |
| 6:56AM |
4 |
Concurrent Call in Asterisk |
| 6:53AM |
3 |
Many analog lines |
| 6:50AM |
2 |
Time sync on PRI |
| 6:48AM |
2 |
Automatic Configuration Tools? |
| 6:46AM |
2 |
AMP not working in GUI |
| 6:20AM |
2 |
sharing asterisk among several companies |
| 6:14AM |
1 |
ser -> asterisk ->cisco gateway |
| 6:10AM |
1 |
sms and DDI UK |
| 6:08AM |
2 |
cvs-head from 3/31/05 fails to load |
| 5:29AM |
1 |
Installing asterisk and components |
| 5:18AM |
2 |
Problems editing oh323 configuration parameters |
| 5:05AM |
2 |
setting SIP to dial PSTN with TDM400P |
| 5:03AM |
0 |
We require Asterisk configuration and support consultants |
| 5:00AM |
1 |
Music Answer while waiting |
| 4:52AM |
0 |
early B3 connect with TE110P |
| 4:31AM |
1 |
Reject second IAX call |
| 3:49AM |
0 |
External line hangup |
| 2:39AM |
0 |
DTMF detection in dial macro |
| 1:45AM |
0 |
'RFC3261 transaction matching failed' and 'one-way' communication |
| 1:39AM |
2 |
sip.conf match |
| 1:12AM |
2 |
how to call land line number using wireless land line service through asterisk |
| 1:11AM |
3 |
Simple authentication |
| 1:01AM |
1 |
CAPI call fails |
| 12:39AM |
0 |
call timeout |
| 12:14AM |
0 |
RE: TE410P Loadtest problem |
| |
| Wednesday March 30 2005 |
| Time | Replies | Subject |
|---|
| 11:51PM |
0 |
Asterisk::AGI script calling macro with arguments - has to be a simpler way... |
| 11:50PM |
0 |
Problem faxing to a fax machine. |
| 11:23PM |
1 |
dial cmd - called party prompted before connect |
| 11:23PM |
0 |
Asterisk to Asterisks |
| 10:53PM |
3 |
Zaptel Periodic Reset |
| 10:32PM |
2 |
SuperMicro X5DE8-GG Motherboard Goes Kaput after Installing TE410P Card - Yikes! |
| 10:26PM |
3 |
Xten-lite for linux |
| 10:08PM |
0 |
Modprobe: FATAL: error running install command for wctdm |
| 9:02PM |
2 |
G729 on Soekris 4801 |
| 8:54PM |
4 |
Cisco 7960 and Asterisk, I think I have a curly one here |
| 8:46PM |
3 |
cmd Authenticiation |
| 8:33PM |
10 |
No prompt after installing |
| 8:27PM |
0 |
Problems running asterisk on Solaris 8 |
| 8:02PM |
0 |
LNP in Area Code 636 (Missouri) |
| 7:53PM |
3 |
Problem with Music on Hold. Please help |
| 7:38PM |
1 |
using amp with asterisk? |
| 5:14PM |
21 |
Sangoma VS. Digium |
| 5:12PM |
1 |
zoom x5v and * |
| 4:11PM |
0 |
Transferring a Call out...Please help! |
| 4:03PM |
4 |
Asterisk <--> PABX |
| 3:45PM |
1 |
Parked Call Issue with realtime Asterisk version |
| 3:44PM |
0 |
Confused: Qozap is on interrupt 209 alone - is this good or not ? |
| 3:36PM |
1 |
Limitations of aah |
| 3:18PM |
1 |
ISDN question |
| 2:57PM |
2 |
Help with Application Development in Asterisk |
| 2:23PM |
2 |
CheckGroup and transfers |
| 1:42PM |
0 |
How can I solve this? |
| 1:12PM |
0 |
Web-based conference management update |
| 12:57PM |
2 |
Toll Free dialing problems |
| 12:54PM |
2 |
Unable to allocate channel structure |
| 12:39PM |
3 |
Voicepulse connect has doubled their rates |
| 12:29PM |
0 |
Settup of incomming calls |
| 12:18PM |
1 |
Does the Grandstream ship with a power brick |
| 12:05PM |
0 |
Optimal iax.conf settings for VoicePulse COnnect |
| 11:56AM |
1 |
File permissions and ownership |
| 11:51AM |
3 |
Australia and SetCallerID |
| 11:48AM |
0 |
GXP-2000 Grandstream - First Impressions |
| 11:46AM |
5 |
Physically Small Box Asterisk Systems |
| 11:22AM |
2 |
Bristuff |
| 11:06AM |
0 |
Altigen & Asterisk with H.323 |
| 10:57AM |
0 |
Transfer a call in the IVR |
| 10:55AM |
1 |
getting boot server working for PolyCom IP500 |
| 10:13AM |
1 |
[newbie question]Can I can from a phone through Asterisk to another Asterisk server to call out from the 2nd Asterisk server |
| 9:56AM |
0 |
newline in an sms |
| 9:24AM |
1 |
What the best Asterisk architecture for 900+ users? |
| 9:19AM |
2 |
username/password for PolyCom IP500 webinterface? |
| 9:19AM |
0 |
Monitor command full static |
| 9:10AM |
2 |
Using HFC-S card |
| 9:00AM |
0 |
CISCO 7970 COLOR FROZEN |
| 8:50AM |
1 |
Asterisk@Home 0.8 released |
| 8:28AM |
0 |
(no subject) |
| 8:24AM |
6 |
username/password for PolyCom IP500 web interface? |
| 8:15AM |
0 |
APP CBMYSQL |
| 7:59AM |
0 |
HELP: How to configure h323 channel driver ? |
| 7:49AM |
1 |
job offer - in german only |
| 7:30AM |
2 |
Giving sip users multiple contexts |
| 7:08AM |
1 |
Recommended GSM gateway |
| 6:44AM |
0 |
Polycom IP600 Cannot answer - SOLVED |
| 6:32AM |
1 |
What is ZAP ? newbie question sorry |
| 6:29AM |
2 |
Asterisk @ home |
| 6:18AM |
3 |
Asterisk::AGI script won't work? |
| 6:17AM |
1 |
Bristuff and startup scripts |
| 5:49AM |
0 |
Problem on outgoing calls (quadbri card and bristuffed Asterisk latest) ? |
| 5:44AM |
0 |
Solaris install from HEAD |
| 4:39AM |
0 |
Asterisk GLIB_2.0 Error |
| 3:24AM |
1 |
Troubles with VoIP providers |
| 2:14AM |
0 |
IAX realtime dynamic |
| 1:24AM |
5 |
Comprehensive Asterisk Load Testing |
| 1:13AM |
0 |
(no subject) |
| 1:03AM |
0 |
IPSwitchBoard Version 0.71 Released |
| 12:21AM |
1 |
Combatting echo in VOIP |
| 12:00AM |
0 |
TE410P Outgoing Call Volume |
| |
| Tuesday March 29 2005 |
| Time | Replies | Subject |
|---|
| 11:29PM |
2 |
Polycom IP600 Cannot answer |
| 10:40PM |
2 |
Implant GIPS's codec to Asterisk |
| 10:02PM |
0 |
Call Waiting and FXO |
| 9:49PM |
1 |
How do i transfer/forward a call out? |
| 9:07PM |
1 |
Forget Asterisk@Home 0.7 :-) :-) 0.8 is out |
| 8:50PM |
5 |
ACD queue question |
| 8:49PM |
1 |
Test Line |
| 8:15PM |
0 |
re: Problem: Compiling error for SpanDSP |
| 7:43PM |
0 |
Problem: Compiling error for SpanDSP app_rxfax |
| 7:36PM |
0 |
voicemail patch for forcegreetings and forcename? |
| 7:35PM |
0 |
.call Files and Unkown Keywords |
| 6:42PM |
5 |
Soekris products available in the US? |
| 6:33PM |
1 |
Houston DID |
| 5:52PM |
1 |
iax2 & nat |
| 5:41PM |
7 |
Sipura 3000 FXO with Asterisk |
| 5:00PM |
0 |
Astfax questions... |
| 4:42PM |
0 |
Using @Home 0.7 and wanting to debug dial plan problem |
| 4:26PM |
2 |
Sipura SPA 2000 - Miltiple Ring Tones |
| 4:16PM |
3 |
Upgrade *@home to CVS-HEAD |
| 3:58PM |
3 |
help w/ basics |
| 3:26PM |
6 |
Can Asterisk do this ? |
| 2:58PM |
1 |
External voice channels pack up |
| 2:31PM |
1 |
Avaya Partner ACS system, pre 7.0 |
| 1:59PM |
0 |
ASTERISK AT HOME USERS -- READ |
| 1:57PM |
0 |
Using * @ Home, all seems to work, but no sound to Softphone |
| 1:43PM |
2 |
With a phone system. |
| 1:30PM |
2 |
Outgoing Volume |
| 1:04PM |
1 |
Newbie question: How do I get Analog Phone to actuall ring |
| 12:53PM |
2 |
Asterisk@Home 0.7 released Question/Problem |
| 12:49PM |
2 |
IAX vs SIP (music on hold) |
| 11:49AM |
1 |
Voicemail sounds |
| 9:47AM |
7 |
Digium - Asterisk Download Ftp Site link Invalid |
| 9:41AM |
2 |
MeetMe flags in * 1.0.7 |
| 8:56AM |
0 |
DTMF detection/generation |
| 8:27AM |
8 |
Dell 1750 & TDM400P - Power |
| 8:15AM |
0 |
adding extension ChanSpy |
| 8:08AM |
3 |
-lssl problem on debian |
| 8:02AM |
0 |
Partially receiving a fax |
| 7:45AM |
0 |
Manager API how see if call is on hold |
| 7:18AM |
0 |
Cisco 7970 Color |
| 6:57AM |
2 |
Call-ID and Unique-ID |
| 6:40AM |
2 |
constant ringing on Zap channels |
| 6:08AM |
0 |
rfc2833 cisco 7960 DTMF issue |
| 6:08AM |
3 |
No D-channels available! |
| 6:01AM |
1 |
Fail over |
| 5:59AM |
0 |
Zultys 4x5 phone |
| 5:44AM |
0 |
HFC PCI |
| 5:38AM |
1 |
Asterisk SMS configuration |
| 5:04AM |
1 |
HFC-S |
| 4:33AM |
1 |
app_darthvader.c? |
| 4:28AM |
0 |
Fw: sip provider |
| 4:18AM |
0 |
asterisk-oh323 pre-releases |
| 4:10AM |
0 |
changes to nat =yes? |
| 4:09AM |
0 |
ADTRAN TA 750 + TE405P + PRI with problem to receive or send fax. |
| 3:36AM |
4 |
VoIP Provider problems |
| 3:24AM |
4 |
Erratic CPU load |
| 2:50AM |
0 |
Outgoing call immediately disconnected |
| 2:33AM |
0 |
Asterisk as gateway with oh323 channel to VOIP provider that can provide gateway or gatekeeper feature ? |
| 2:20AM |
1 |
Sending many faxes simultaneously with spandsp |
| 1:58AM |
3 |
Zaptel based timing for VoIP-only Asterisk |
| 1:56AM |
2 |
Spandsp compilation error |
| 1:48AM |
1 |
Voicetronix OpenSwitch12 chan_vpb problem |
| 12:22AM |
1 |
sox |
| |
| Monday March 28 2005 |
| Time | Replies | Subject |
|---|
| 11:37PM |
1 |
Asterisk, SER, NAT, STUN and the whole debate |
| 10:24PM |
3 |
SPA-841 Call waiting? |
| 9:49PM |
1 |
Turnkey alternatives to fonality or switchvox? |
| 8:18PM |
1 |
Problem with 401 Unauthorized |
| 7:58PM |
1 |
Problem installing SpanDSP Makefile.patch |
| 7:50PM |
2 |
call center: agents, queues, sip |
| 7:48PM |
1 |
How to do something random? |
| 7:18PM |
1 |
spandsp rxfax under Linux 2.6 w/TDM400? |
| 6:54PM |
1 |
Kernel panic loading second fritz card |
| 6:37PM |
0 |
Asterisk@Home Handbook |
| 6:35PM |
13 |
Asterisk@Home 0.7 released |
| 6:22PM |
2 |
CIC Code |
| 5:15PM |
0 |
CDR to ODBC |
| 4:53PM |
6 |
Open Source Billing Software |
| 4:22PM |
1 |
First second choppy |
| 4:17PM |
1 |
Sounds gets choppy after 30 seconds |
| 4:04PM |
3 |
Debugging Asterisk in GDB (DDD) |
| 4:02PM |
0 |
MWI's for Third Party Softswitch |
| 3:58PM |
0 |
Teliann SIP Firmware |
| 2:58PM |
2 |
Start on system restart |
| 2:55PM |
3 |
call files run at certain times |
| 2:32PM |
2 |
RE: 8 channel fxo setup outgoing call problem (cont) |
| 2:03PM |
1 |
MWI and SIP PHones in Asterisk |
| 1:29PM |
0 |
8 channel fxo setup outgoing call |
| 1:04PM |
1 |
8 channel fxo setup outgoing call problem |
| 12:43PM |
6 |
Verizon ISDN |
| 11:55AM |
0 |
Re: Asterisk-Users Digest, Vol 8, Issue 229 |
| 11:20AM |
1 |
voicemail sending blank .WAV file via email |
| 11:19AM |
2 |
problem with 1 dialing (recording says must dial 1 when I thought I did) |
| 11:06AM |
1 |
need to install the openline4 card |
| 10:43AM |
1 |
How to config speex? |
| 10:29AM |
1 |
Which analog phones to use and why? |
| 10:06AM |
2 |
Third party Firefly issue very weird?? |
| 10:02AM |
1 |
Remove a channel from receiving inbound calls |
| 9:46AM |
0 |
bristuff-0.2.0-RC7k: error on loading qozap : "qozap: Unknown symbol zt_xxxxx" |
| 9:39AM |
4 |
AMP-1.10.007 Released! |
| 7:53AM |
3 |
CAPI/Dialing out |
| 7:43AM |
3 |
can a sip.conf stanza be shared by several phones? |
| 7:34AM |
1 |
Connecting quadbri to EuroISDN with 2 TE and 2 NT ports - what cables and settings ? |
| 7:30AM |
2 |
AGI STREAM FILE command |
| 7:30AM |
1 |
gsm player for Linux? |
| 6:28AM |
0 |
BroadVoice - "Failed to authenticate on INVITE" error |
| 4:26AM |
1 |
H323: g711-g729 transcoding |
| 4:21AM |
3 |
TDM04B doesn't hang up after Voicemail |
| 2:51AM |
0 |
Bug fixes IPSwitchBoard |
| 12:57AM |
1 |
spandsp-0.0.2pre11 |
| 12:32AM |
0 |
Local/Remote * Servers, IAX/SIP mix and voice-mail notifications |
| |
| Sunday March 27 2005 |
| Time | Replies | Subject |
|---|
| 11:48PM |
2 |
apps api? |
| 10:16PM |
1 |
Broadvoice getting unregistered |
| 7:57PM |
6 |
pass caller ID to another application or machine. |
| 7:55PM |
3 |
How to park/transfer a call received from a Queue? |
| 7:21PM |
1 |
Strange problems IAX / Monitor / ChanSpy CVS HEAD |
| 7:03PM |
8 |
Asterisk on a dialup connection? |
| 6:48PM |
1 |
Quicknet phonejack connect to telephone line? |
| 4:59PM |
0 |
TDM11B and hook flash |
| 3:11PM |
2 |
Comedian Voicemail Issues |
| 3:02PM |
0 |
MOH Fixed |
| 1:42PM |
0 |
Re: Using call.sample on Zap hardware - Answering problem |
| 12:40PM |
0 |
Voicemail / Dial command issue |
| 10:58AM |
2 |
Music on Hold Broken?? |
| 10:28AM |
0 |
analog phone |
| 10:25AM |
6 |
How to use multiple VOIP provider trunks |
| 10:20AM |
1 |
Asterisk and call delivery to connected PABX |
| 9:38AM |
0 |
-Using Reply To on Asterisk List- |
| 9:34AM |
0 |
3 Party Conference & ZapHFC |
| 9:13AM |
1 |
Asterisk and XLite on same machine (OSX)? |
| 9:04AM |
0 |
High Availability on Asterisk |
| 8:55AM |
2 |
sip provider |
| 8:28AM |
1 |
ata vs digium card |
| 8:21AM |
0 |
trying to add the free voipjet test to my |
| 8:13AM |
3 |
missing ring-tone |
| 7:39AM |
0 |
"Unable to get parameters" while configuring FXO cards, any ideas? |
| 7:18AM |
1 |
[Asterix-users] CISCO 7910 |
| 5:41AM |
3 |
Can't get format_mp3 to work for music on hold |
| 4:00AM |
1 |
newbie install options |
| 3:37AM |
3 |
Can't Dial Out with TDM04B |
| 1:21AM |
6 |
Sipura 2000 x dual g729 channels x other choices? |
| 1:14AM |
6 |
NPA NXX |
| |
| Saturday March 26 2005 |
| Time | Replies | Subject |
|---|
| 10:49PM |
1 |
IPSwitchBoard new Release |
| 10:16PM |
0 |
Setup Zoom V3 Router + VoIP register with Asterisk |
| 10:02PM |
0 |
E1 ISDN Problem |
| 9:58PM |
1 |
Push VLAN to Polycom via DHCP |
| 9:19PM |
2 |
trying to add the free voipjet test to my asterisk at home??? |
| 8:38PM |
3 |
context |
| 8:16PM |
0 |
Broadvoice audio problems |
| 7:58PM |
0 |
Echo on Zaptel hardware (Wildcard 100XP) |
| 7:35PM |
1 |
DTMF tones not working |
| 7:01PM |
1 |
AGI "STREAM FILE" issue |
| 4:46PM |
1 |
Soyo G668 + Asterisk |
| 4:35PM |
1 |
Transferred calls CDRs |
| 4:24PM |
3 |
Asterisk with Winmodem |
| 3:53PM |
1 |
about sip and registering |
| 3:31PM |
1 |
Dialout handler with/without leading 1 |
| 2:18PM |
5 |
Click-to-Talk with Asterisk? |
| 11:16AM |
0 |
asterisk+voicetronix |
| 8:03AM |
0 |
Newbie Instalation |
| 7:59AM |
0 |
Zap keeps online if caller hangs up |
| 7:16AM |
4 |
Cisco's description of echo |
| 6:59AM |
0 |
ringing CAPI & SIP channels together |
| 6:36AM |
0 |
make a call based on SMS request |
| 5:55AM |
1 |
Major problems with TDM400 and specific |
| 4:49AM |
1 |
Cisco Phones with Asterisk |
| 3:37AM |
0 |
The Sound of Silence on TDM400P |
| |
| Friday March 25 2005 |
| Time | Replies | Subject |
|---|
| 10:56PM |
2 |
Asterisk as a dial in server for internet service? |
| 10:53PM |
2 |
Look at that Digium Broadband Modem! |
| 10:47PM |
1 |
Poor pstn line quality |
| 10:10PM |
0 |
OutBound call on Zap with Dial command |
| 10:07PM |
2 |
uniden voip gear |
| 9:59PM |
1 |
faxes |
| 8:52PM |
0 |
Re: Asterisk-Users Digest, Vol 8, Issue 216 |
| 8:36PM |
2 |
MeetMe/Conference |
| 8:31PM |
3 |
800 numbers and FWD |
| 7:45PM |
0 |
New Warning in CVS: Format for authentication entry is user[:secret]@realm |
| 6:14PM |
5 |
Does asterisk@home 0.6 really work??? |
| 5:51PM |
1 |
gnomemeeting / sip |
| 5:38PM |
2 |
911 & SoftHangup on SPA-3000 |
| 5:33PM |
4 |
Openloop disconnect? |
| 5:06PM |
1 |
2 companies - one asterisk |
| 4:38PM |
0 |
Re: Asterisk-Users Digest, Vol 8, Issue 210 |
| 4:14PM |
0 |
Remote MWI for Central Voicemail? |
| 4:12PM |
0 |
Re: Square key KTS app on * |
| 4:04PM |
1 |
Eicon DIVA PCI ISDN cards (not server) work withasterisk! |
| 2:40PM |
2 |
WaitExten question |
| 2:14PM |
0 |
OT: AstLinux 0.2.4 Released |
| 1:41PM |
1 |
X100P FXO card-No Dial Tone |
| 1:29PM |
0 |
Re: Two companies - One Asterisk |
| 1:28PM |
0 |
Asterisk vs silence detection |
| 1:28PM |
1 |
Re: Two companies - One Asterisk |
| 12:54PM |
2 |
Re: Two companies - One Asterisk |
| 12:24PM |
1 |
JUST NEED A REPLY |
| 12:05PM |
0 |
Calls from analog/FXS phone? |
| 11:38AM |
6 |
Asterisk compare with Skype |
| 11:33AM |
0 |
CAUTION: Re: grandstream firmware update 1.0.5.23 |
| 11:32AM |
0 |
Re: Polycom phones-buggy SIP firmware or am Imissingsomething in the XML configs? |
| 11:00AM |
5 |
Two companies - One Asterisk??? |
| 10:24AM |
1 |
We just released our new Asterisk Installation CD set. with 24/7 monitoring |
| 10:23AM |
2 |
Zap Detect called party pickup |
| 10:21AM |
1 |
Audio codec MP108 |
| 10:21AM |
1 |
Asterisk@Home Usage |
| 10:13AM |
9 |
small qos switch |
| 10:12AM |
0 |
Outbound audio fades out with IAX Provider |
| 9:57AM |
2 |
Can I get a sip doorbell? |
| 9:34AM |
0 |
ways to get more accuracy from ztdummy |
| 9:25AM |
0 |
Problem with *72 |
| 9:12AM |
5 |
Re-write callerid? |
| 9:00AM |
4 |
Square Key system |
| 8:50AM |
0 |
debugging trunks between two asterisk boxes at two different locations |
| 8:47AM |
0 |
Dial command problem(VOIP+*+TDM400P+Legacy PBX) |
| 8:08AM |
7 |
What is web login password for Asteirsk@Home |
| 8:00AM |
1 |
asterisk-addons and 64bit make |
| 7:24AM |
0 |
Re: Dial Out?? |
| 6:13AM |
1 |
grandstream firmware update 1.0.5.23 |
| 5:38AM |
0 |
re-write statement |
| 5:25AM |
2 |
MGCP issue |
| 4:33AM |
1 |
Hello Everyone |
| 4:14AM |
1 |
Does IAX supports silence suppression? |
| 3:54AM |
49 |
atxfer |
| 1:57AM |
1 |
peering |
| 1:06AM |
1 |
Forwarding to regular numbers? |
| 1:00AM |
2 |
Multiple outgoing calls through VOIP providers |
| 12:33AM |
1 |
Converting 7905G to SIP |
| |
| Thursday March 24 2005 |
| Time | Replies | Subject |
|---|
| 11:35PM |
1 |
voicemail problems with CVS-HEAD |
| 11:14PM |
2 |
Xten and NAt Problems |
| 10:26PM |
1 |
Advanced Cisco 7960 Config |
| 9:53PM |
1 |
Can I use my callscreen macro w/ sip? |
| 9:43PM |
0 |
Re: [2] X100p problem |
| 9:27PM |
1 |
Polycom phones-buggy SIP firmware or am I missingsomething in the XML configs? |
| 8:38PM |
2 |
Dynamically limiting the number of outbound calls |
| 8:04PM |
7 |
Backup for linux/asterisk |
| 8:02PM |
2 |
Emailed voicemail |
| 7:27PM |
1 |
SIP/iax routing question |
| 7:26PM |
1 |
realtime - unable to find key |
| 7:08PM |
1 |
Asterisk Hardware Requirements for a 50-100 Seat Call Center |
| 6:57PM |
2 |
Digium T1 Card Questions |
| 6:51PM |
1 |
Best Headsets for a Call Center Environment |
| 6:38PM |
0 |
SPA-3000 disconnect tone |
| 6:21PM |
1 |
Question on framerate |
| 4:50PM |
0 |
Native Bridging drops call on release |
| 4:42PM |
9 |
Forklift a 2000 phone PBX |
| 4:21PM |
0 |
Outlook contacts ->Asteriskdatabase(LookupCI DName) |
| 4:17PM |
0 |
chan_unistim compile failed |
| 3:40PM |
0 |
Any word on when CHanisAvail for IAX2 will be on CVS? |
| 2:05PM |
5 |
* -> SMS w/out PSTN |
| 2:04PM |
2 |
Parking |
| 1:16PM |
2 |
echo using Xlite |
| 12:38PM |
0 |
ChanSpy in CVS ! |
| 12:36PM |
0 |
Asterisk@Home version 0.6 forwarding to pstn numbers? |
| 12:28PM |
3 |
Outlook contacts -> Asterisk database (LookupCIDName) |
| 12:06PM |
0 |
AGI commands STDOUT problem |
| 12:00PM |
0 |
IAXy dial tone problem |
| 11:30AM |
1 |
Error cannot record voicemail |
| 10:54AM |
0 |
NetHDLC + PRI |
| 9:51AM |
0 |
Monitor System for T1 failure. |
| 9:24AM |
2 |
Toll-free DID switchover: Get status? |
| 9:24AM |
0 |
No compatible codecs! |
| 9:12AM |
2 |
rxfax trouble on bristuffed capi |
| 9:02AM |
0 |
Properly setup SRV? |
| 8:58AM |
1 |
Third time, is it a charm? |
| 8:57AM |
0 |
Re: IP-500 config |
| 8:50AM |
0 |
Asterisk 1.0.3 Sipura codec error |
| 8:36AM |
2 |
Polycom DTMF |
| 8:34AM |
3 |
Newbie Voicemail Question |
| 8:16AM |
1 |
Question on routes |
| 7:47AM |
0 |
how to bridge two channels ? |
| 7:19AM |
2 |
Fun with CAPI |
| 7:18AM |
0 |
Echo on my TDM fxo |
| 7:18AM |
14 |
Realtime mysql problem? |
| 7:18AM |
2 |
When should I use SER ? |
| 6:58AM |
0 |
Tricky setup |
| 6:34AM |
0 |
"restart gracefully" fails |
| 6:04AM |
0 |
snom220 problem |
| 5:33AM |
0 |
Smal ofice pbx |
| 5:30AM |
3 |
Asterisk as Cisco Call-Manager - dial out to PSTN |
| 4:45AM |
4 |
Newbie pointers |
| 4:16AM |
1 |
RSA interasterisk IAX problems ? |
| 4:10AM |
3 |
Cisco 7905G Firmware |
| 3:11AM |
1 |
direct ip-to-ip call |
| 2:53AM |
2 |
Fax and Voice |
| 2:34AM |
0 |
R: music on hold error |
| 2:23AM |
0 |
Missing CDR data |
| 1:49AM |
0 |
Record(Sip) |
| 1:37AM |
3 |
codec for asterisk |
| 1:21AM |
1 |
Missing CallingPres Application |
| 12:39AM |
0 |
Is there a way to get inserted into an LEC's CLIDB? (fwd) |
| |
| Wednesday March 23 2005 |
| Time | Replies | Subject |
|---|
| 11:45PM |
1 |
help understanding sip header |
| 11:33PM |
3 |
WiFi SIP |
| 9:55PM |
0 |
calling an Application |
| 9:31PM |
1 |
VoiceMail Outgoing Calls and Disconnects |
| 9:26PM |
1 |
2 *@home issues away from bliss |
| 8:21PM |
0 |
How set language in Auto-dial out |
| 8:20PM |
0 |
How connect 2 extension by AGI |
| 8:16PM |
0 |
inquery auto monitor in 1.0.3 |
| 7:55PM |
3 |
Nortel Option 11 |
| 7:13PM |
0 |
Re: IP-500 config |
| 6:53PM |
1 |
Perform Action after X invalid tries |
| 6:50PM |
1 |
Spandsp question ( re: compiling ) |
| 6:35PM |
0 |
[Fwd: [Soekris] net5801 & net7501] |
| 6:22PM |
0 |
Re: Asterisk-Users Digest, Vol 8, Issue 198 |
| 6:11PM |
0 |
Asterisk Realtime. |
| 6:11PM |
1 |
PRI E1 Questions |
| 6:05PM |
0 |
Help, incoming lines problem! |
| 5:51PM |
2 |
*-1.0.7 DTFM => Not working |
| 5:04PM |
1 |
Polycom phones-buggy SIP firmware or am I missing something in the XML configs? |
| 4:27PM |
1 |
FW: polycom 500 help!! |
| 4:26PM |
1 |
audio outband bad quality |
| 4:18PM |
0 |
polycom 500 help!! |
| 3:51PM |
0 |
agi script for german date / time |
| 3:39PM |
0 |
ASTCC date format |
| 3:20PM |
1 |
cannot dial any extension except xlite |
| 2:58PM |
4 |
Vonage Linksys Router - Life after Vonage |
| 2:54PM |
2 |
Asterisk ChangeLog |
| 2:53PM |
0 |
Direct Dial Into ISDN Line |
| 2:22PM |
1 |
Rejecting ISDN-call without Answering |
| 1:58PM |
1 |
zaptel.o undefined references |
| 1:37PM |
0 |
Local sip client stuttered audio |
| 1:03PM |
0 |
Queue "has X calls (max Y)" problems with INSANE high numbers |
| 12:47PM |
4 |
Problem compiling asterisk-addons |
| 12:30PM |
2 |
Does X100P clone provide Timer? |
| 12:10PM |
1 |
prevent non-free calls |
| 12:03PM |
0 |
[Fwd: newbie DNS problem with BT100 |
| 12:00PM |
1 |
call pick up and joining an active call |
| 12:00PM |
1 |
Access comedian mail from imap client |
| 11:52AM |
2 |
Problems with incoming calls |
| 11:39AM |
1 |
Zaphfc + PRI card problem |
| 11:38AM |
1 |
slim server for moh |
| 11:34AM |
0 |
I get "is on the phone" when the client is logged out |
| 11:15AM |
1 |
Multicall |
| 11:15AM |
6 |
Problem parsing unusual SIP/SDP |
| 10:13AM |
0 |
Random use of Sip peers |
| 10:12AM |
0 |
gnophone 0.2.4 and asterisk 1.0.6 |
| 10:08AM |
1 |
SIP messagse |
| 9:49AM |
3 |
Need some help |
| 9:22AM |
2 |
Why even have set CallerID option? |
| 9:18AM |
1 |
Eicon DIVA PCI ISDN cards (not server) work with asterisk! |
| 9:17AM |
0 |
Settings to improve voice quality? |
| 9:15AM |
2 |
Where to put the modules to start on boot? |
| 8:45AM |
1 |
speex 1.1.7 crashes asterisk 1.0.6 |
| 8:43AM |
2 |
Group channel rotation for outgoing call? |
| 8:26AM |
0 |
Re: [0] Wilcard X100P doesn't hang up when in Voicemail() and calling party hangs up. |
| 7:59AM |
4 |
Chanisavail and IAX2 |
| 7:54AM |
0 |
MeetMe Upgrade ! |
| 7:06AM |
0 |
Diva Server configuration |
| 7:02AM |
1 |
Wilcard X100P doesn't hang up when in Voicemail() and calling party hangs up. |
| 6:53AM |
1 |
SIP callid |
| 6:53AM |
1 |
* and Cisco Callmanager Interconnection |
| 6:46AM |
0 |
Using Asterisk as VMail server on CCM 3.3.3 System |
| 6:31AM |
10 |
Broadvoice alternatives |
| 6:25AM |
2 |
ADIT 600 "Dynamic Impedance matching" |
| 6:17AM |
1 |
GR-303 from Central Office supported? |
| 6:05AM |
0 |
read dtmf during dial |
| 5:37AM |
2 |
Reg Asterisk |
| 5:32AM |
1 |
snom 220 version |
| 5:18AM |
4 |
Any Software Echo Cancellation in Asterisk? |
| 5:02AM |
4 |
Playback of sound files but no sound |
| 4:30AM |
3 |
FXS FXO |
| 3:30AM |
0 |
Blog post on Asterisk setup |
| 2:59AM |
0 |
features enableing via database per extension number |
| 2:42AM |
0 |
Some audio problems |
| 1:59AM |
1 |
Asterisk Features/Dial Codes (Newbie question?) |
| 1:49AM |
1 |
BV Outbound Drop fixed . |
| 1:38AM |
0 |
Can I change the volume on a sip phone (Snom) from *? |
| 1:23AM |
0 |
Agents priority in queue |
| 12:35AM |
0 |
SIP behavior between different providers |
| 12:11AM |
0 |
Release 0.68 of IPSwitchBoard |
| |
| Tuesday March 22 2005 |
| Time | Replies | Subject |
|---|
| 11:29PM |
2 |
asterisk@home print incoming fax |
| 10:56PM |
2 |
Cisco 7940 and multiple simultaneous calls |
| 10:05PM |
0 |
troublshooting DTMF |
| 9:13PM |
1 |
newbe: help with registration |
| 9:02PM |
0 |
silence suppression |
| 8:27PM |
2 |
Asterisk locking up - 99.9% CPU |
| 7:37PM |
1 |
Help Debugging my code? |
| 6:34PM |
2 |
Incoming response and external access |
| 6:17PM |
1 |
asterisk-addons / OS X woes (continued) |
| 5:59PM |
4 |
Chanspy is back ! |
| 5:42PM |
0 |
[Fwd: newbie DNS problem with BT100] |
| 5:05PM |
0 |
D() option on Dial |
| 4:22PM |
0 |
sip show peers weirdness |
| 3:58PM |
2 |
Digium support quality: Excellent |
| 3:40PM |
1 |
Mimicking Linksys PAP2? |
| 3:39PM |
1 |
Words of a user, ... what can I make better? |
| 3:39PM |
1 |
Is there a way to get inserted into an LEC's CLIDB? |
| 3:37PM |
0 |
RE: Asterisk-Users Digest, Vol 8, Issue 186 |
| 3:12PM |
0 |
sip disconnects |
| 3:06PM |
1 |
Problems loading zapata module under suse 9.2 (cvs stable from 5 days ago) ? |
| 2:38PM |
4 |
TE405P and echo |
| 2:24PM |
1 |
Nat and firewall port forwarding - is it really required? |
| 2:21PM |
2 |
Is there a way to get inserted into an LEC's CLI DB? |
| 1:52PM |
1 |
Zap channels not hanging up... |
| 1:38PM |
3 |
IP PHONE with chip PA1688 and IAX2 Authentication |
| 1:32PM |
1 |
No recorded messages |
| 1:27PM |
4 |
Quick Newbie Question - Auto Call Routing |
| 1:08PM |
3 |
Major problems with TDM400 and specific telephones: suggestions? |
| 12:55PM |
0 |
"Succes" report for TDM400 and IBM Netfinity 5600 |
| 12:55PM |
0 |
"Success" report with TDM400 and Via EPIA-MII motherboard |
| 12:54PM |
2 |
Help please for newb on Asterisk to Vonage |
| 12:36PM |
7 |
Rhino Channel Bank or ADIT 600 |
| 12:35PM |
1 |
RE: Asterisk-Users Digest, Vol 8, Issue 152 |
| 12:34PM |
4 |
multiline, cordless, expandable phone system and asterisk message waiting |
| 12:12PM |
2 |
audio delay in meetme conference using ztdummy |
| 12:12PM |
2 |
X100P interrupt load |
| 11:48AM |
0 |
RE: Asterisk-Users Digest, Vol 8, Issue 150 |
| 11:45AM |
0 |
Still no Broadvoice Outbound. (Bump) |
| 11:44AM |
0 |
Callgroups Question |
| 11:38AM |
0 |
RE: Asterisk-Users Digest, Vol 8, Issue 150 |
| 10:52AM |
1 |
How to mute a call |
| 10:52AM |
0 |
avm fritz 2.6 |
| 10:49AM |
1 |
Reproducible echo on IAX calls to -some- destinations. |
| 10:46AM |
3 |
X100P voicemail volume too low (quiet) |
| 10:41AM |
1 |
Kernel 2.6.11 |
| 10:34AM |
0 |
Problems using zaphfc and wct1xxp together... |
| 10:08AM |
4 |
OT: does Sipura SPA 3000 support UK caller id? |
| 9:32AM |
1 |
Setup to dial out only on voip (Broadvoice) not PSTN? |
| 9:20AM |
4 |
VOIP - Billing Solutions with Asterisk? |
| 9:13AM |
1 |
NEWBIE: MWI on 7960 |
| 9:01AM |
1 |
*@Home .6 adding a outside number to a group{Scanned} |
| 8:56AM |
0 |
help with registration |
| 8:55AM |
0 |
RE: [Asterisk-uk] Meet |
| 8:46AM |
5 |
Setting MWI on legacy PBX |
| 8:31AM |
1 |
Experience with this radius? |
| 7:42AM |
1 |
Call Transfer Features |
| 7:41AM |
5 |
Enhanced 911 |
| 7:10AM |
4 |
Asterisk - SS7 or ISDN |
| 6:28AM |
0 |
te110p sometimes green, sometimes stays red on stable cvs ? |
| 6:28AM |
6 |
IRQ headaches |
| 6:21AM |
0 |
In Call functions |
| 6:00AM |
2 |
Regex howto proof and change a dialed number |
| 5:52AM |
0 |
Phone book |
| 4:59AM |
3 |
SIP response * |
| 4:16AM |
0 |
[info] :: BIOS Motherboard Settings :: |
| 4:07AM |
2 |
bottlenecks |
| 3:59AM |
2 |
Asterisk-addons/OS X woes |
| 3:53AM |
0 |
asterisk + outlook + omniis TAPI driver |
| 3:04AM |
1 |
Call file misbehaviour |
| 2:40AM |
4 |
Feedback on CBMySql, MeetMe2 and web interface |
| 2:35AM |
1 |
RE: [Asterisk-Dev] Problem Making a SIP call over a long latencynetwork- Call rejected: 407 Proxy |
| 2:06AM |
4 |
Review: Asterisk at CeBIT 2005 / Asterisk at Linux-Tag 2005 |
| 1:51AM |
1 |
H323 for Asterisk |
| 1:43AM |
0 |
[Asterisk-Dev] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required |
| 1:40AM |
0 |
who has purchased a V400 card from Varion ? please help me .I need some document . |
| 1:39AM |
0 |
ANNOUNCEMENT : MeetMe - Web-MeetMe (throughmanager) |
| |
| Monday March 21 2005 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
DTMF is not working |
| 11:27PM |
1 |
ISDN EICON Cards |
| 11:16PM |
1 |
Flash pannel: time display |
| 10:36PM |
3 |
how to keep Asterisk up to date on many servers |
| 10:26PM |
1 |
SMS Alert Script - Voice e-mail |
| 8:51PM |
1 |
DISA Hangs up after DTMF is sent |
| 8:14PM |
0 |
asterisk-h323 and h323_id |
| 7:10PM |
1 |
Asterisk as test equipment |
| 7:09PM |
2 |
*@Home .6 adding a outside number to a group |
| 7:07PM |
2 |
Permission issue with outgoing calling |
| 7:05PM |
0 |
Ideas on how to make a script for using random zap channels |
| 5:46PM |
1 |
Asterisk, SER & Jabber |
| 5:17PM |
2 |
I need use sip |
| 5:03PM |
0 |
OSS and ALSA |
| 4:04PM |
6 |
Fax receive issues and NVFaxDetect |
| 2:58PM |
2 |
Hold Pickup |
| 2:37PM |
2 |
Compiling with gcc -shared on OS X |
| 2:30PM |
0 |
Micronet SP5001 ATA |
| 2:06PM |
1 |
Hitachi Cable WIP-5000? |
| 1:53PM |
3 |
Asterisk/Zaptel on Mac G5 or Xserve |
| 1:52PM |
0 |
SIP, NAT, and bindaddr |
| 1:24PM |
2 |
Flash hook & hangup problem |
| 1:07PM |
1 |
Net2Phone / Vonage |
| 12:56PM |
0 |
Compile error for minimal install of Redhat 9.0 [SOLVED] |
| 12:54PM |
3 |
Script to Authenticate User and Dial Out |
| 12:40PM |
1 |
chan-sccp-easter2005 make error with stable 1.0.6? |
| 12:21PM |
0 |
ANNOUNCEMENT : MeetMe - Web-MeetMe (through manager) |
| 12:15PM |
0 |
SIP Dial between two IAX-connected boxes? |
| 12:03PM |
2 |
Ext matching problems |
| 11:53AM |
4 |
Can't hear the caller |
| 11:48AM |
5 |
VoicePulse Issues |
| 11:27AM |
0 |
CallingCaed Application |
| 11:25AM |
0 |
audio frequency with wcfxs and K8t |
| 11:01AM |
9 |
why even use SIP |
| 10:51AM |
0 |
Jabber module for asterisk |
| 10:37AM |
0 |
Doubts Configuration SIP |
| 10:20AM |
1 |
Modify CallerID (on SIP phone) during call |
| 9:52AM |
1 |
iLBC codec and mute issues |
| 9:52AM |
0 |
Unable to get message on hold class to work |
| 9:27AM |
2 |
Why isasterisk's voice mail calledcomedian. |
| 9:01AM |
2 |
G726-16 passthrough... |
| 8:50AM |
1 |
Replacement 7960 Handset |
| 8:09AM |
3 |
US pstn => voip |
| 8:09AM |
0 |
astcc & sip |
| 7:51AM |
2 |
H323 gateway thru NAT |
| 7:29AM |
2 |
CallerID Name with IAX Providers |
| 7:05AM |
1 |
Version 0.67 of IPSwitchBoard Released |
| 6:53AM |
0 |
OT: "No authority found" connecting to Freshtel |
| 6:39AM |
0 |
asterisk outbound to SIP provider problems (still) |
| 6:27AM |
4 |
Why is asterisk's voice mail called comedian. |
| 5:53AM |
1 |
mpg123 home music from stream |
| 4:12AM |
1 |
IAX call rejected.....who was trying to reach 's@' |
| 3:28AM |
0 |
Cdr_odbc asterisk 1.0.6 |
| 2:12AM |
1 |
DTMF doesn't seem to get through incoming ZAP channels |
| 1:03AM |
1 |
ASTCC: perl / mysql or me??? |
| |
| Sunday March 20 2005 |
| Time | Replies | Subject |
|---|
| 11:40PM |
0 |
problems with SLES! |
| 11:18PM |
1 |
asterisk-1.0.7 make install on fedora corre 3 give errors |
| 11:12PM |
5 |
zaptel PRI drivers |
| 10:17PM |
1 |
HELP: Failed start after install asterisk_oh323-0.7.1 |
| 9:55PM |
3 |
Soekris net4801 and analog interface? |
| 9:50PM |
0 |
want just few words from the list about SLES! |
| 9:26PM |
0 |
AVM Fritz! Noise / Crackle |
| 8:23PM |
0 |
X100P and Toshiba PBX |
| 7:38PM |
3 |
Choosing an ISP for Asterisk |
| 7:17PM |
2 |
NVBackgroundDetect |
| 5:52PM |
3 |
who has purchased a V400 card from Varion ? |
| 5:46PM |
2 |
Polycom dhcpd.conf? [Or, "Some day, I'll figure this all out."] |
| 5:36PM |
2 |
Follow-Me Script |
| 4:02PM |
0 |
H323 Gatekeeper Registering Question |
| 3:52PM |
0 |
rejected calls |
| 2:59PM |
0 |
i8253 count to high! resetting |
| 2:40PM |
1 |
app_nv_backgrounddetect - how to make module |
| 2:38PM |
2 |
FWD to Vonage not working? |
| 2:33PM |
1 |
TAPI |
| 2:29PM |
1 |
Problem transfering incoming calls |
| 2:21PM |
3 |
Dial from a URL - Possible? |
| 2:19PM |
2 |
OT: VIA Mini-ITX, Asterisk, and hardware |
| 2:16PM |
1 |
Limit incoming calls |
| 1:30PM |
0 |
Question on silcen aware |
| 12:58PM |
2 |
asterisk and outlook |
| 11:10AM |
0 |
Asterisk-addons 1.0.7 |
| 10:37AM |
4 |
Cisco 7960 SIP boot takes 2 minutes? |
| 9:59AM |
1 |
Any experience with Dell 1850 Server with PERC 4e/Si |
| 9:38AM |
1 |
IAXY Polarity |
| 8:59AM |
5 |
wctdm fxs ring frequency |
| 7:52AM |
0 |
FW: Can't get more than one SIP device to be able to make outgoing calls |
| 6:17AM |
4 |
virus |
| 5:20AM |
4 |
ISDN-30 in UK |
| 3:48AM |
0 |
Outgoing Call problem with PSTN line |
| 3:46AM |
2 |
IPSwitchBoard-BETA Update |
| 3:35AM |
1 |
I cannot use G711 (ulaw|alaw) |
| 3:00AM |
1 |
softphone with web url support |
| 1:43AM |
2 |
Echo after upgrade * 1.05 -> 1.06 |
| |
| Saturday March 19 2005 |
| Time | Replies | Subject |
|---|
| 11:46PM |
6 |
VoIP service through Asterisk? |
| 8:30PM |
2 |
Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required |
| 8:21PM |
1 |
Ignore incoming calls on X100P |
| 7:55PM |
2 |
RE:Newbie question |
| 6:44PM |
1 |
vmware and asterisk |
| 6:41PM |
2 |
Problem with asterisk-addons/OS X |
| 5:53PM |
1 |
Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP |
| 4:56PM |
1 |
create distinctive ring on FXS |
| 4:06PM |
0 |
X-lite not hanging up / DTMF not present through voipuser.org |
| 3:48PM |
3 |
Asterisk and Cisco AS53xx/54xx Access Server Platform |
| 3:24PM |
2 |
More HEAD wierdness (chan_sip, jitterbuffer/PLC problems) |
| 3:20PM |
3 |
CallingCard Application |
| 2:56PM |
1 |
req: cisco 12sp+ firmware |
| 2:32PM |
3 |
Any Zaurus users?? |
| 2:14PM |
0 |
mysql addon and cdr |
| 1:55PM |
1 |
What happened to www.iptel.org? |
| 1:51PM |
3 |
Question on routing table... |
| 12:50PM |
0 |
Polycom Callerid callback |
| 12:36PM |
0 |
DVG-1120S no call display name and time |
| 12:11PM |
1 |
DISA -> macro = congestion |
| 11:37AM |
2 |
MeetMe2 admin functions |
| 10:48AM |
1 |
ANI & DNIS sent to analog FXs Port Possible |
| 10:42AM |
1 |
Polycom Soundpoint boot ROM upgrade: how? |
| 10:00AM |
0 |
How to install /use festival on Asterisk |
| 9:55AM |
0 |
A couple of "dated" questions. |
| 9:53AM |
7 |
Any 24 (or 30) way FXS PCI cards? |
| 9:45AM |
1 |
* and DirecWay |
| 9:06AM |
3 |
ZapBarge restrictions? |
| 9:02AM |
2 |
Routing 911 calls |
| 8:42AM |
2 |
outbound delay |
| 8:19AM |
2 |
Tool for mysql |
| 6:26AM |
1 |
Areskicc installation problems |
| 5:26AM |
0 |
lost newbie requesting help for Asterisk Implementation |
| 5:15AM |
2 |
Goto and E1 line |
| 3:38AM |
1 |
Asterisk's on Suse Linux Enterprise Server(SLESv9) |
| 2:40AM |
2 |
:: What does it take to upgrade? :: Newbie Q :: |
| 2:29AM |
1 |
noice sip to sip only??? |
| 2:27AM |
0 |
Excessive indications tone levels (longish) |
| 2:10AM |
0 |
Asterisk 1.0.7 chan_skinny fix port to chan_sccp? |
| 1:46AM |
0 |
my hfc card does not like Siemens |
| |
| Friday March 18 2005 |
| Time | Replies | Subject |
|---|
| 11:47PM |
2 |
Asterisk 1.0.7 Released |
| 11:38PM |
0 |
Yet another cisco 9760 7.x firmware failure |
| 11:30PM |
0 |
SIP-T support? |
| 11:14PM |
1 |
best protocol/codec for dialup |
| 10:27PM |
1 |
SIP <-> PSTN DTMF |
| 10:26PM |
1 |
GR303 with * |
| 10:25PM |
0 |
I look for some copartner |
| 10:01PM |
1 |
(no subject) |
| 9:19PM |
2 |
Article on Slashdot |
| 8:41PM |
0 |
Short burst of static then disconnect |
| 7:27PM |
2 |
current asterisk cvs problem with distinctive ring? |
| 6:35PM |
0 |
T100P: Can't Make/Receive Zap Calls (Long Newbie Blah) |
| 6:27PM |
1 |
Broadvoice hangs-up / disconnects after about 30 deconds |
| 6:03PM |
2 |
No sound when calling in from pstn |
| 5:29PM |
1 |
OT: Mexico area codes |
| 4:36PM |
1 |
Te110P initial installation problems ? |
| 4:11PM |
0 |
RXfax / Spandsp bad fax |
| 4:02PM |
0 |
Combine agent inbound and outbound |
| 2:54PM |
1 |
Registration issues with Sipura SPA-841 |
| 2:41PM |
3 |
:: BIOS Motherboard Settings :: |
| 1:35PM |
1 |
X100P problem - no responce |
| 1:22PM |
5 |
small Local telco (wifi voip) some experiences with * ?? |
| 12:48PM |
0 |
New Version of IPSwitchBoard-Beta |
| 12:34PM |
0 |
Rule of thumb rule for x/x => 1/1 billing |
| 12:33PM |
1 |
Configuring GnomeMeeting for Asterisk |
| 12:09PM |
0 |
Is this a BUG?? Please I need help in this |
| 11:55AM |
1 |
XML config files for Polycom SoundPoint IP 300? |
| 11:41AM |
3 |
Asterisk handling of SIP info |
| 11:34AM |
0 |
HELP: Dose G.729 with IPP only worked on IntelCPU? |
| 11:05AM |
0 |
IAX Peer/auth issues WAS: Netlogic inbound DID issue |
| 10:58AM |
3 |
HELP: Dose G.729 with IPP only worked on Intel CPU? |
| 10:57AM |
2 |
PSTN > Voicemail |
| 10:45AM |
1 |
Problem with Manager Interface |
| 10:40AM |
0 |
seg fault when accessing voicemail via any IAX softphone (diax, iax phone) |
| 10:38AM |
1 |
Some IAX questions |
| 10:30AM |
1 |
Looking for quality inbound DID - IAX providers, UK, USA, Australia |
| 9:59AM |
4 |
Optional URL in App. Queue |
| 9:41AM |
0 |
SNOM 190 Loud Ring While on Speaker |
| 9:28AM |
0 |
I4l + HiSax |
| 9:26AM |
1 |
call a url and get a result in the dialplan |
| 9:22AM |
2 |
echo / delay problem |
| 9:15AM |
3 |
reply a post |
| 9:14AM |
2 |
asterisk reload |
| 8:23AM |
1 |
newbe question sip.conf |
| 7:45AM |
0 |
Meetme2 compilation Err |
| 7:40AM |
1 |
BV This morning |
| 7:38AM |
15 |
Meetme2 compilation problem |
| 7:17AM |
3 |
TDM400P install problems |
| 6:57AM |
3 |
Which linux distribution |
| 6:47AM |
1 |
voicemail.conf extractor? |
| 6:33AM |
2 |
Parking a call in manager interface |
| 6:31AM |
0 |
AGI-like calls in the [globals] section |
| 6:21AM |
1 |
Manager API - Redirect command |
| 6:08AM |
2 |
Where to place calling rule contexts? |
| 5:57AM |
5 |
Group Ring after Timeout |
| 5:40AM |
1 |
Cisco 7940 convert to sip |
| 4:38AM |
2 |
Pattern matching in extensions.conf |
| 2:03AM |
3 |
ANNOUNCEMENT:Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules) |
| 1:58AM |
2 |
TDM400P Not loading Drivers |
| 1:57AM |
0 |
voicemail, busy does not work? |
| 1:16AM |
0 |
ISDN phone Hold-Problem connected to QuadBRI/Zap |
| |
| Thursday March 17 2005 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
Cisco 79XX Phones |
| 11:16PM |
1 |
limit about asterisk pstn out |
| 11:11PM |
2 |
gsm cannot be found in any file form... but it's there |
| 11:02PM |
3 |
ANNOUNCEMENT: Updatesforapp_cbmysqlandMeetMe2gui (out of tree modules) |
| 10:56PM |
1 |
leaky reload |
| 10:08PM |
6 |
About the weather.. |
| 10:01PM |
2 |
Getting caller-name - cid_rewrite 1.0.0 |
| 9:36PM |
1 |
Realtime - how to reload ? |
| 9:02PM |
0 |
seeking GSM 850/1900 gateway |
| 9:00PM |
0 |
Configuring Asterisk with BroadVoice |
| 8:29PM |
1 |
Extension ringing but no ringing sound asterisk |
| 7:00PM |
3 |
Newbie can't dial out to pstn |
| 5:50PM |
2 |
How to make Span Port Selection in "Round Robin" fashion? |
| 5:44PM |
4 |
X-Lite and Asterisk |
| 5:31PM |
2 |
Got 200 OK on REGISTER that isn't a register |
| 5:30PM |
1 |
Broadvoice solution finally |
| 4:39PM |
1 |
Database families and keys |
| 4:16PM |
1 |
My appologies |
| 3:55PM |
1 |
What cable to connect TE110P to telco PRI ? |
| 3:26PM |
0 |
Atxfer not working for called party |
| 3:18PM |
2 |
Backing up configurations and *@home list? |
| 3:17PM |
0 |
Message waiting/station busy conflict? |
| 3:15PM |
0 |
softphones and extensions status |
| 3:12PM |
1 |
IAX2 VOIP HARDPHONE |
| 2:55PM |
0 |
ASTCC dialstatus confusing billing issue |
| 2:48PM |
0 |
adding to asterisk db from a script |
| 2:27PM |
1 |
Agent won't log out! |
| 1:48PM |
1 |
What causes this changethread error message? |
| 1:44PM |
3 |
Undocumented "exten" syntax? |
| 1:23PM |
0 |
MOH and conference calls |
| 1:05PM |
1 |
Test post |
| 12:30PM |
0 |
Re: Last guy to get BV working outbound |
| 12:23PM |
1 |
Different codecs for different numbers via same IAX provider; how? Configs included. |
| 12:17PM |
4 |
Caller ID on E&M Wink |
| 12:16PM |
3 |
Phone ringing and not going to voicemail? |
| 12:14PM |
3 |
Searching the list archives |
| 11:49AM |
2 |
PRI Cause Code Help |
| 11:34AM |
1 |
PRI Test Equipment |
| 11:31AM |
1 |
Include/Macro not working right... |
| 11:29AM |
0 |
h323 problem loading |
| 11:29AM |
0 |
asterisk, callerid name and cisco 2600 |
| 11:22AM |
3 |
Channel name (and substring) |
| 11:08AM |
3 |
Realtime Problem = Segmentation faults |
| 11:00AM |
1 |
OT: PC sound hardware for voice recording |
| 10:53AM |
2 |
Codec negociation |
| 10:51AM |
2 |
Redhat 9 Music on hold |
| 10:35AM |
1 |
Strange console call problem |
| 10:08AM |
0 |
MOH patch for bristuffed * |
| 10:03AM |
6 |
Polycom vs. Cisco IP Phones |
| 10:02AM |
2 |
Netlogic inbound DID issue |
| 9:49AM |
1 |
Asterisk dialplan (and/or VM) via LDAP |
| 9:27AM |
1 |
Last guy to get BV working outbound? |
| 9:02AM |
1 |
session border control |
| 8:40AM |
2 |
echo paid support |
| 8:16AM |
0 |
Re: chan_oh323.c ast_oh323_new Internal channel initialization failed [solved] |
| 8:05AM |
0 |
IAX2 Trunking, No connections any more... |
| 7:37AM |
1 |
Using Codec G-726 |
| 7:20AM |
0 |
Chan_Spy and MOH - Any Status? |
| 7:06AM |
0 |
ztdummy - no sound in Asterisk@Home |
| 7:03AM |
1 |
Comparing Callmanager to Asterisk |
| 7:03AM |
0 |
Asterisk start problem (automatically) |
| 6:53AM |
1 |
Welltech Welgate 3804 FXO Configs |
| 6:34AM |
0 |
astguiclient error! |
| 6:33AM |
1 |
ZAp channel numbering question |
| 6:32AM |
1 |
Call Quality Detail Record |
| 6:24AM |
1 |
Asterisk with Cisco Call Manager |
| 4:26AM |
2 |
TE110p card with Euro ISDN (Ericsson switch) |
| 4:21AM |
4 |
Hi there.. |
| 4:02AM |
3 |
Compilation problem chan_capi and Eicon Diva 4Bri |
| 3:42AM |
3 |
extension.conf dialplan |
| 3:40AM |
5 |
CAC Access Bank Manual |
| 3:09AM |
0 |
asterisk t.38 codec negotiation problems |
| 1:35AM |
2 |
ser+asterisk - security |
| 1:24AM |
1 |
Call Recording and Archiving |
| 12:55AM |
2 |
Snom190 intercom |
| 12:18AM |
2 |
HOW-To write an AGI |
| |
| Wednesday March 16 2005 |
| Time | Replies | Subject |
|---|
| 11:53PM |
1 |
who have been fabricated their own cards from Tormenta 2 PCI Card? |
| 11:08PM |
1 |
Pattern Matching? |
| 11:04PM |
1 |
How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to * |
| 9:55PM |
1 |
Connecting Multiple Asterisk Servers! |
| 9:53PM |
1 |
Hong Kong DID |
| 9:52PM |
1 |
music on hold error |
| 9:25PM |
0 |
FXS->FXO Converter?? |
| 8:20PM |
1 |
Low cost hardware time for production environment |
| 7:50PM |
0 |
Email MWI Crashes Asterisk - Solved |
| 6:37PM |
0 |
Email MWI crashes Asterisk |
| 6:24PM |
1 |
Re: [Serusers] ser+asterisk - security |
| 5:28PM |
0 |
Problems with TDM400P and asterisk on Linux 2.6 |
| 4:55PM |
2 |
Do you need to recompile the Linux 2.6 kernel for zaptel modules? |
| 4:31PM |
3 |
NuFone and CallerID |
| 4:23PM |
2 |
Dial multiple extensions, but different variables/timeouts |
| 3:56PM |
2 |
[Possible SPAM] : about sip, asterisk and cisco ccme |
| 3:49PM |
3 |
(Yet another) Music on hold problemand another... |
| 3:44PM |
2 |
Global Intercom on SIP phones |
| 3:25PM |
0 |
about sip, asterisk and cisco ccme |
| 3:23PM |
0 |
Obscure * command and audio questions |
| 3:19PM |
0 |
Realtime ODBC with cdr_odbc using the same database |
| 3:10PM |
2 |
t.38 support news? |
| 3:10PM |
0 |
OT: AstLinux mailing lists now available! |
| 2:31PM |
6 |
79xx 7-4 |
| 2:00PM |
1 |
MGCP Channel Lockup and other probelms |
| 1:54PM |
0 |
Voice cutoffs |
| 1:16PM |
1 |
Pickup extensions for Zap channels does not work |
| 1:00PM |
2 |
ISDN Cards in the USA |
| 12:52PM |
0 |
Help with Audiocodes MP-108-FXO SIP Firmware |
| 12:25PM |
0 |
Mini Manual for IPSwitchBoard published |
| 12:05PM |
9 |
IAX Registration being lost |
| 11:14AM |
1 |
About shadydial |
| 10:59AM |
1 |
No ringing indication to radio phone |
| 10:41AM |
5 |
Asterisk Capabilities |
| 10:36AM |
0 |
Iax register |
| 10:24AM |
1 |
cisco 12sp+/30vip IP phone |
| 9:47AM |
0 |
Meetme doesn't react to DTMF keys |
| 9:46AM |
0 |
Asterisk retains DTMF Control Even whenanExternal IVR System is dialed |
| 9:35AM |
2 |
Asterisk E911? |
| 8:35AM |
3 |
Cisco gateways and hairpinning |
| 8:20AM |
3 |
TxFAX problem |
| 8:04AM |
1 |
Problem starting Asterisk - libssl.so.4 cannot be found |
| 7:27AM |
4 |
problem with musiconhold |
| 7:14AM |
0 |
Asterisk makes the news |
| 6:41AM |
0 |
Two (or more) Asterisk servers, routing calls |
| 5:36AM |
2 |
meetme2 compilation |
| 5:33AM |
3 |
CLI SIP Client |
| 5:13AM |
2 |
Basical question to asterisk |
| 5:12AM |
1 |
Problem with TE405P and Slackware 10.0 (reply this) |
| 4:53AM |
0 |
where is STUN implemented? |
| 4:41AM |
1 |
Re: chan_oh323.c ast_oh323_new Internal channel initialization failed |
| 4:38AM |
0 |
Call Forward |
| 4:33AM |
1 |
Error in placing call file in directory |
| 4:28AM |
19 |
IPSwitchBoard BETA |
| 4:23AM |
1 |
Kernel 2.4 or 2.6 for the latest asterisk ? ? |
| 4:06AM |
1 |
live monitoring of SIP calls chan_spy |
| 3:20AM |
2 |
AGI kill |
| 3:15AM |
0 |
Calling Card Application - which one ? |
| 3:15AM |
2 |
Calls from web interface |
| 3:06AM |
0 |
Help with simple H323 settings |
| 2:07AM |
0 |
chan_oh323.c:2501 ast_oh323_new: Internal channel initialization failed. Bad binary? |
| 1:48AM |
0 |
Agent groups broken in queues? (do not follow strategy) |
| 12:51AM |
0 |
IAX softphone on WinCE/PocketPC |
| 12:34AM |
0 |
Stable CVS or Head CVS for using TE110P ? |
| |
| Tuesday March 15 2005 |
| Time | Replies | Subject |
|---|
| 11:54PM |
0 |
ya newbie problem |
| 11:37PM |
0 |
how buy digium card such as TDM400. |
| 11:30PM |
0 |
Is my dialplan wrong? |
| 10:16PM |
0 |
Grandstream BETA Firmware |
| 9:20PM |
2 |
Grandstream and Transfers |
| 9:07PM |
1 |
Problem with presence |
| 8:56PM |
1 |
Background apps that plays music on hold |
| 8:51PM |
0 |
PCI 2.2 question |
| 8:31PM |
0 |
cdr issue |
| 7:39PM |
1 |
Automon Question |
| 7:26PM |
3 |
Voice getting cutoff |
| 7:09PM |
2 |
Flashpannel: How to get more than 28 buttons? |
| 6:57PM |
1 |
Unknown signalling 896? |
| 5:40PM |
1 |
Not ringing phone that are in use |
| 5:02PM |
0 |
SetDigitTimeout question |
| 5:02PM |
0 |
Cisco DTMF problem... |
| 4:48PM |
3 |
Asterisk@Home Install Problem |
| 3:28PM |
0 |
RE: can't hear anything on my side during a SIP call |
| 3:16PM |
2 |
Wiki down: Is there another source for documentation? |
| 2:23PM |
1 |
Which is the "newest" libpri/zaptel? |
| 1:20PM |
1 |
(Yet another) Music on hold problem and another... |
| 1:17PM |
0 |
zaphfc vs. i4l DTMF recognition |
| 12:59PM |
1 |
oh323 and open 729 |
| 12:13PM |
2 |
Asterisk retains DTMF Control Even whenan External IVR System is dialed |
| 11:49AM |
1 |
How to see ExtensionStatus in manager |
| 11:38AM |
6 |
Realtime config |
| 11:16AM |
1 |
Asterisk retains DTMF Control Even when an External IVR System is dialed |
| 11:09AM |
1 |
How to connect with a headphone |
| 11:04AM |
3 |
FW: AntiSpam Alert from Rusten McKenzie |
| 10:36AM |
1 |
Learning the Ropes of * |
| 10:34AM |
0 |
Voicemail Question...help |
| 10:19AM |
1 |
Accecpt SIP calls from an IP |
| 10:15AM |
0 |
Incoming calls from Cisco 1760 given wrong context... |
| 10:05AM |
1 |
Transferring calls into MeetMe |
| 9:56AM |
9 |
Asterisk Newbie |
| 8:30AM |
1 |
Call Center software opensource or commercia l |
| 8:29AM |
1 |
Asterisk RealTime |
| 8:29AM |
1 |
Open ports? |
| 8:21AM |
2 |
Setting up Security Groups |
| 8:20AM |
2 |
fcpci - capi driver for Fritz |
| 8:13AM |
8 |
Call Center software opensource or commercial |
| 8:00AM |
4 |
Three way calling with X-Lite / MeetMe |
| 7:38AM |
1 |
PRI: Call Reference Length not supported |
| 7:21AM |
1 |
PRI Card TE110p Question |
| 7:18AM |
1 |
SIP & H323 gateway |
| 7:18AM |
0 |
Re: Asterisk-Users Digest, Vol 8, Issue 119 |
| 7:16AM |
3 |
Web triggered calls |
| 6:31AM |
1 |
Site to Site Gateway |
| 6:03AM |
2 |
How to determine the voicemail file name for an AGI script |
| 5:47AM |
0 |
Zombie or soft hangup |
| 5:19AM |
2 |
Asterisk Queue strange behaviour |
| 3:47AM |
4 |
Kernel 2.4 or 2.6 for the latest asterisk ?? |
| 3:05AM |
0 |
Forwarding SIP calls to proxy |
| 2:53AM |
3 |
Call Queues and Transfers |
| 2:32AM |
0 |
What different between asterisk-oh323 andastersk's chan_h323 ? |
| 1:51AM |
0 |
dial to h.323 |
| 1:43AM |
1 |
Eclipse Plugin for managing Asterisk |
| 1:02AM |
1 |
Queue drop out into context not working? |
| 1:00AM |
1 |
blind xfer works atxfer doesn't...help! |
| 12:39AM |
0 |
trying to get trunk to register with * behind NAT |
| |
| Monday March 14 2005 |
| Time | Replies | Subject |
|---|
| 10:48PM |
0 |
Service similiar to VoicePulse-Connect? |
| 10:31PM |
1 |
Newbie - Config Problem ? |
| 10:02PM |
1 |
Creating IAXClient windows component |
| 7:50PM |
0 |
Multitech MVP130 as FXO with asterisk |
| 7:31PM |
1 |
Extentions Variable Dialing QUESTION. |
| 6:57PM |
2 |
I changed some minor things, but how can I contribute it? |
| 6:22PM |
3 |
asterisk-addons OS X |
| 6:00PM |
1 |
Anybody compiled ICD on Asterisk 1.0.6 release.. |
| 5:44PM |
1 |
Problem Compiling Spandsp |
| 5:11PM |
1 |
LCR Question - Keep one trunk free |
| 4:47PM |
2 |
Sipura SIP vs. IAX |
| 4:04PM |
2 |
How NuFone.Net's customer service works. |
| 4:01PM |
1 |
meetme2 and meetme |
| 3:46PM |
1 |
Broadvoice's changes last week broke call forwarding |
| 3:04PM |
0 |
Dealing with bandwidth limitations |
| 2:52PM |
1 |
Rhino channel banks |
| 2:49PM |
1 |
VoIP Provider SIP Call Flow |
| 2:37PM |
0 |
dial out using sip via ZAP channel |
| 1:58PM |
2 |
FWD IAX Problem |
| 1:39PM |
3 |
insecure=very |
| 1:31PM |
0 |
Agents without agent channel |
| 1:26PM |
2 |
Broadvoice Busy Issue |
| 12:52PM |
1 |
TDM400P crackel |
| 11:56AM |
1 |
Has anybody tried NVFaxDetect Fax detection SIP/IAX |
| 11:46AM |
1 |
School design question |
| 11:19AM |
1 |
Setting NAT=yes for not NATed clients |
| 10:51AM |
0 |
Asterisk support for SIP REFER message |
| 10:49AM |
1 |
TDM400 audio problems |
| 10:30AM |
1 |
Skype - Bandwidth |
| 10:21AM |
2 |
qualify and NAT.... |
| 10:02AM |
0 |
not ringing when place outgoing call |
| 9:23AM |
0 |
1.0.5 / 1.0.6 and oh323 compiling problem |
| 9:17AM |
18 |
Grandstream GXP-2000 |
| 8:47AM |
1 |
OT: Recommendation for Dynamic DNS on Meshbox? |
| 8:25AM |
2 |
Has anybody experience with SetGroup / CheckGroup commands? |
| 8:10AM |
0 |
dial script, send variable problem?? |
| 8:04AM |
1 |
DS3 with Asterisk |
| 7:50AM |
0 |
busy signal not in cdr |
| 7:47AM |
0 |
ASTCC - Are there some add ons available? |
| 7:36AM |
4 |
How to Flash() a modem line |
| 6:34AM |
1 |
colinux fresh install, zaptel does not compile, size_t error |
| 6:20AM |
2 |
Cisco 7960 SIP 7.4 |
| 5:38AM |
0 |
asterisk codec negotiation problem |
| 5:20AM |
0 |
N/A |
| 4:56AM |
2 |
asterisk outbound to SIP provider problems |
| 4:43AM |
4 |
E1/T1 back to back ?? |
| 4:39AM |
3 |
Problem with TE405P and Slackware 10.0 |
| 3:52AM |
0 |
1.0.5 and h323 compiling problem |
| 2:40AM |
1 |
weird outbound problem through broadvoice (new) |
| 1:50AM |
1 |
snom 220 busy all the time |
| 1:09AM |
7 |
Voicemail SMS Alert - Possible? |
| |
| Sunday March 13 2005 |
| Time | Replies | Subject |
|---|
| 11:57PM |
0 |
Doubt about asterisk NOTIFY |
| 10:33PM |
0 |
fxo card not workin in susev9.2! |
| 10:29PM |
2 |
Asterisk, Voicetronix, and Australia |
| 7:51PM |
0 |
Commercial Asterisk Support? (Digium, etc.) |
| 7:10PM |
1 |
g729 Lic ordered from Digium Question. |
| 4:28PM |
3 |
cordless/wireless system with a ip base station? |
| 3:09PM |
2 |
IAX2 and asterisk servers linking to each other |
| 1:21PM |
0 |
Re: possible bug in chan_capi concerning context handling - SOLVED |
| 1:14PM |
4 |
SUSE 9.2 and Zaptel channels |
| 1:07PM |
1 |
sip.conf entry precedence |
| 12:31PM |
1 |
Running asterisk as non-root: Zaptel Permission Probs |
| 12:23PM |
2 |
PRI Call Reference Length not Supported |
| 11:25AM |
0 |
ASTCC Functions |
| 10:47AM |
7 |
Text Messaging or AIM |
| 10:33AM |
1 |
ASTCC sounds |
| 9:55AM |
0 |
looking for DID in spain |
| 9:17AM |
5 |
ASTCC - how to use different brands? |
| 9:17AM |
0 |
Re: Asterisk-Users Digest, Vol 8, Issue 105 |
| 9:01AM |
0 |
IAX2 and server links |
| 7:57AM |
2 |
sending a DTMF tone before hangup |
| 5:31AM |
2 |
Sipura 841 issues |
| 4:21AM |
6 |
newbie uk questions... |
| 4:15AM |
0 |
safe_asterisk doesn't restart when called by initlog in fedora |
| 3:21AM |
5 |
possible bug in chan_capi concerning context handling |
| 3:18AM |
1 |
Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues |
| 3:08AM |
0 |
Zaptel problems, Asterisk 1.0.6 |
| 3:01AM |
2 |
How can I eveluate trailing numbers in extensions.conf? |
| 1:04AM |
0 |
What different between asterisk-oh323 and astersk's chan_h323 ? |
| |
| Saturday March 12 2005 |
| Time | Replies | Subject |
|---|
| 10:24PM |
0 |
ASTCC or should I use something elsefor different rates, depending on the calling card? |
| 10:12PM |
0 |
vars for transfered calls |
| 9:19PM |
2 |
RE: [Asterisk-Dev] SetVarCDR |
| 8:35PM |
2 |
chat line |
| 8:06PM |
0 |
Hang on "making progrogress passing" when dialing out |
| 6:31PM |
0 |
Voice Based Bulletin Board. |
| 6:08PM |
1 |
playing "invalid" to an internal user |
| 5:57PM |
0 |
Does zapateller work in Australia? |
| 5:55PM |
0 |
Question on phones with asterisk |
| 5:47PM |
0 |
Problem with ability to dial out when a channel is used from an external equipment in a point to multi point configuration |
| 4:28PM |
1 |
Asterisk with Skype |
| 4:17PM |
1 |
RE: Asterisk-Users Digest, Vol 8, Issue 88 |
| 4:16PM |
0 |
after *-1.0.6 upgrade error: vm_execmain: Unable to read password |
| 3:46PM |
2 |
DISREGARD!! Broadvoice outgoing problems |
| 3:42PM |
1 |
Broadvoice outgoing problems |
| 2:18PM |
0 |
Tracking/Billing Incoming & Outgoing Minutes? |
| 1:50PM |
1 |
Zapping around |
| 1:08PM |
0 |
Where to download the asterisk-oh323? |
| 11:37AM |
0 |
How do I pick up a trailing number in extensions.conf? |
| 11:30AM |
6 |
Advanced conference features, meetme2? |
| 9:33AM |
2 |
checking active SIP members of a queue? |
| 9:31AM |
2 |
ASTCC - Regex: How to "Country" but "special City" different? |
| 9:03AM |
0 |
Looking for an Asterisk Expert/Partner for project |
| 8:04AM |
2 |
Unable to create channel of type 'IAX2' |
| 7:10AM |
0 |
Sipura 2100 and Asterisk one-way audio |
| 5:28AM |
2 |
Signaling on PRI channels |
| 3:59AM |
1 |
ATA 186 Codec Question. |
| 2:57AM |
1 |
X-Lite and * SIP Problem |
| 2:38AM |
0 |
IAX2 Sphone for PocketPC |
| 2:19AM |
2 |
SIP monitor thread is hanged up on a uClinux embeded linux system |
| 1:19AM |
1 |
Simultaneous call to both phones in PAP2-NA |
| 12:58AM |
1 |
ipvolution TDM cards - vaporware? |
| |
| Friday March 11 2005 |
| Time | Replies | Subject |
|---|
| 11:30PM |
0 |
SineApps Daily Asterisk News Back Up |
| 10:52PM |
4 |
ASTCC or should I use something else for different rates, depending on the calling card? |
| 9:10PM |
0 |
Sipura 2100 and Asterisk - Faxing |
| 8:49PM |
1 |
digium card |
| 7:49PM |
0 |
Error cant change devie with no technology |
| 6:51PM |
0 |
Re: [Asterisk-biz] Opportunities for good billing solutions |
| 5:49PM |
1 |
SIP-B? |
| 4:30PM |
1 |
Asterisk, IAX2 and iptables |
| 4:08PM |
1 |
DVG-1120 questions |
| 3:42PM |
1 |
CNAM for Asterisk |
| 3:32PM |
0 |
Re: Incoming echo cancel |
| 3:17PM |
0 |
Re: Incoming echo cancel |
| 3:08PM |
0 |
Receiving faxes via SIP |
| 2:56PM |
1 |
Call Transfers |
| 2:14PM |
0 |
SIP -> NAT -> * |
| 2:13PM |
3 |
Parked Call |
| 2:12PM |
3 |
Droping calls |
| 2:00PM |
0 |
Errors using Asterisk as Sip Client behind SER !!! |
| 1:53PM |
2 |
Re: Incoming echo cancel |
| 1:52PM |
0 |
Sipura 2100 and Asterisk and Fax |
| 1:36PM |
1 |
TDM04B lock up |
| 1:15PM |
1 |
Trouble with Realtime |
| 1:08PM |
0 |
Polycom ip600 - how to eliminate echo? |
| 12:27PM |
4 |
VoipJet Terms of Service |
| 12:21PM |
7 |
Sip show registry returning nothing |
| 12:18PM |
3 |
Asterisk Billing System |
| 11:47AM |
8 |
No ringback over IAX - LiveVoip |
| 11:42AM |
4 |
Wireless VoIP |
| 11:38AM |
1 |
agents - queue config |
| 11:15AM |
0 |
Is it an AGI bug in 1.06? IAX Calls going towrong extension with AGI. |
| 11:06AM |
6 |
Asterisk security problem: authorized SIP users can fake any callerid! |
| 10:50AM |
2 |
diffrent area codes for diffrent phones in dialplan |
| 10:33AM |
13 |
Vonage a provider? |
| 10:26AM |
1 |
Is it an AGI bug in 1.06? IAX Calls going to wrong extension with AGI. |
| 10:16AM |
2 |
FC3 Dual Xeon Zaptel PANIC |
| 10:07AM |
2 |
Realtime does not work yet, ... |
| 9:40AM |
1 |
PAP2-NA point to poitn calls ??...(Direct IP Dialing) |
| 9:32AM |
0 |
Festival & Asterisk CVS Head |
| 9:13AM |
4 |
ASTCC and NuFone billing is different!! |
| 8:58AM |
3 |
Phone suggestions |
| 8:45AM |
2 |
1.0.6 music on hold bug ?! |
| 8:27AM |
0 |
CAPI- 2 Cards |
| 8:20AM |
1 |
EADS6550 and asterisk - echo on PSTN call |
| 8:16AM |
4 |
Multiple IAX Phones Behind NAT |
| 7:32AM |
1 |
Am i right by Asterisk? |
| 6:57AM |
0 |
Quescom AS/400 GSM Gateway + Asterisk |
| 6:49AM |
2 |
What is that area code? |
| 6:47AM |
1 |
Some Hardware Advice |
| 6:36AM |
1 |
Manager (5038) |
| 6:16AM |
0 |
Asterisk@home 0.6 + bristuff |
| 6:12AM |
1 |
IAX, double NAT |
| 6:12AM |
0 |
One single record file for a meetme monitor? |
| 5:09AM |
0 |
SIP Phone Unreachable |
| 5:05AM |
1 |
SIP signalling and RTP to different servers |
| 4:58AM |
2 |
CDR database |
| 4:47AM |
0 |
Asterisk@home 0.6 + Modem.conf |
| 4:32AM |
1 |
Unable to create Zap channel when dialing using a bri cellular gateway |
| 4:20AM |
1 |
Incomplete incoming fax using spandsp 0.0.2pre10 |
| 4:11AM |
1 |
QuadBRI ,TDM400 and SuSE9.2 (Sencond try) |
| 4:02AM |
1 |
Asterisk + Call hangup |
| 3:50AM |
0 |
Intermittent volume deterioration in conferences |
| 3:47AM |
0 |
FW: IAX Settings |
| 3:27AM |
2 |
Load Balancing b/w 2 asterisk servers using SIP load balancer |
| 3:15AM |
1 |
NuFone Configuration [problem] |
| 3:12AM |
1 |
TE110P experiance |
| 2:08AM |
1 |
from sip to asterisk to h323..how |
| 12:07AM |
2 |
How to register two SIP phones ( e.g. Windows Messenger) from different subnet to * |
| |
| Thursday March 10 2005 |
| Time | Replies | Subject |
|---|
| 11:59PM |
3 |
AAH 0.06 - IAX Connection Over NAT Firewall |
| 11:33PM |
0 |
SIP to H.323 no audio |
| 11:22PM |
2 |
E1 LED not lighting up.... |
| 10:17PM |
1 |
what is best free softphone. |
| 9:59PM |
7 |
Panasonic TDA200 E1 -> E100P negotiation issues |
| 9:05PM |
5 |
Bandwidth |
| 8:22PM |
3 |
SetCallerID({$NEWCALLERID}) |
| 8:20PM |
1 |
multiple enum results |
| 7:32PM |
0 |
One way speech from H.323 incoming calls, but outgoing calls are OK. |
| 6:47PM |
0 |
iconnect here, inbound yes, outbound no |
| 5:26PM |
0 |
Broadvoice Config proper?? |
| 5:11PM |
2 |
Transfering calls or using any feature |
| 5:01PM |
3 |
Application SetVarCDR |
| 4:50PM |
0 |
Asterisk@Home - Email to Fax |
| 4:40PM |
0 |
zaptel configuration issues (zaptel.conf vs.zapata.conf) |
| 4:39PM |
2 |
Cisco and Asterisk |
| 4:32PM |
1 |
Asterisk@Home, AMP, and Broadvoice |
| 4:25PM |
0 |
WOW: solved (was: compiling and ssl) |
| 4:12PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group |
| 4:09PM |
4 |
Suse Compiling: next err |
| 3:47PM |
1 |
Odd problem with asterisk |
| 3:37PM |
1 |
Re: Polycom phones do not talk to each other |
| 3:14PM |
0 |
RE: Asterisk-Users Digest, Vol 8, Issue 83 |
| 3:07PM |
0 |
Re: Polycom phones do not talk to each other |
| 3:02PM |
0 |
7905 example configs |
| 2:52PM |
0 |
Re: Polycom phones do not talk to each other |
| 2:46PM |
1 |
Polycom phones do not talk to each other andcannot answer when we pickup |
| 2:14PM |
1 |
what replaced app_qcall? |
| 2:08PM |
3 |
Polycom phones do not talk to each other and cannot answer when we pickup |
| 2:07PM |
7 |
IAX2 800 Termination |
| 1:34PM |
3 |
SIP, DNIS, Asterisk |
| 1:10PM |
2 |
QuadBRI ,TDM400 and SuSE9.2 |
| 1:04PM |
0 |
Manager Redirect fails on Zap Channels |
| 12:57PM |
0 |
Vonage down in Dallas? |
| 12:48PM |
0 |
Astrisk to legacy Mitel SX2000 |
| 12:27PM |
1 |
Listeners in SIP conferences |
| 12:22PM |
2 |
Re: Paging using multiple sound cards |
| 12:03PM |
5 |
asterisk and Broadvoice Outgoing Again :( |
| 11:52AM |
1 |
***SOLVED*** Broadvoice latest changes andstillnot working- An Additional Server****Solved*****! |
| 11:31AM |
2 |
Re: Do I Need Astrisk |
| 11:11AM |
1 |
OT: AstLinux 0.2.2 released |
| 11:08AM |
1 |
Windows messenger 4.7.3001 does not have Account Tab ? |
| 10:26AM |
3 |
Pictures from the Asterisk Pavilion at Spring VON 2005 |
| 10:13AM |
2 |
Asterisk and USB ISDN controllers ... |
| 9:56AM |
0 |
Re: Message Waiting over a IAX trunk |
| 9:40AM |
0 |
OH323 - compilation error (another user, another error) |
| 9:23AM |
1 |
Xlite dont ring on Asterisk |
| 9:17AM |
2 |
NVFaxDetect errors on make |
| 9:10AM |
2 |
tdm400p and dell 2600 poweredge |
| 8:53AM |
0 |
bypassing auth info |
| 8:32AM |
0 |
Avoiding connect signal in two stage dialing |
| 8:22AM |
2 |
hide callerid via presention bits of ISDN |
| 8:06AM |
6 |
NuFone |
| 7:54AM |
0 |
ASTCC - regexpression for country and certain cities? |
| 7:37AM |
0 |
Broadvoice busy message every couple of days. |
| 7:20AM |
1 |
Delay on outgoing calls |
| 7:07AM |
2 |
Location of Voice e-mail Code??? |
| 7:02AM |
0 |
BRI: "Unable to create channel of type 'ZAP'" |
| 6:53AM |
1 |
a liitle bit of info required |
| 6:39AM |
1 |
Problem with incoming calls. |
| 6:27AM |
0 |
New Integrics tip: VoIP for ISPs |
| 5:47AM |
4 |
Compiling Asterisk On SUSE 9.2 |
| 4:52AM |
0 |
Problem with NOTIFY |
| 4:48AM |
1 |
FWDout credits sharing |
| 4:08AM |
2 |
OT: Active channels bridging with SNOM190 |
| 3:42AM |
0 |
Calls hang in a conversation |
| 2:51AM |
1 |
Cisco 7940/60 and 802.3af PoE |
| 1:32AM |
0 |
ISDN to SIP |
| 1:09AM |
1 |
OT: Zap channels intermittently bridging with SNOM190 |
| 12:47AM |
1 |
iax,trunking,zap |
| 12:40AM |
1 |
Single port S0 ISDN card to use in Greece |
| 12:32AM |
0 |
Upgraded to Asterisk 1.0.6 now crashes on boot, sql issue? |
| |
| Wednesday March 9 2005 |
| Time | Replies | Subject |
|---|
| 11:46PM |
1 |
Asterisk & NOTIFY problem |
| 10:45PM |
1 |
Paging and Intercom using Sipura SPA-841 |
| 10:44PM |
2 |
Where can I find all areacodes for USA (accounting purpose) |
| 10:31PM |
2 |
Apple links Asterisk |
| 10:09PM |
1 |
Asterisk@Home Installation Problems |
| 9:36PM |
1 |
Sangoma and other ISA T1 cards |
| 9:26PM |
0 |
Call Parking issue |
| 9:25PM |
1 |
Comparison Charts |
| 8:39PM |
1 |
Voicemail Rap |
| 8:35PM |
1 |
Slightly OT - Snom 190 function keys via subscribed config |
| 7:55PM |
2 |
Asterisk-oh323-0.7.1 compile error |
| 5:53PM |
4 |
Broadvoice Multiple "lines" |
| 5:18PM |
2 |
Server specifications |
| 4:55PM |
1 |
ODBC error ? |
| 4:12PM |
3 |
Asterisk@home silly problem, please help! |
| 3:59PM |
1 |
Support for SIP REFER message |
| 3:52PM |
3 |
Problems with new install voicemail broadcast |
| 3:37PM |
0 |
Problem in Configuring Asterisk Server |
| 3:25PM |
2 |
zaptel configuration issues (zaptel.conf vs. zapata.conf) |
| 3:25PM |
1 |
Paging using multiple sound cards/channels |
| 3:11PM |
0 |
OT: Any interest in Line Powered Amplifiers? |
| 2:46PM |
1 |
Tired of trying to fix this echo problem |
| 2:15PM |
1 |
can I use an external modem such as USR robotics V92 |
| 2:13PM |
6 |
VoIPJet |
| 2:06PM |
3 |
voicepulse "silence" during conversations |
| 2:05PM |
1 |
Providing a dialtone |
| 1:54PM |
2 |
Call Progress Analysis |
| 1:43PM |
0 |
New astGUIclient version released 1.1.0 |
| 1:21PM |
0 |
IPH-90 and Asterisk , MGCP |
| 12:50PM |
26 |
OT: Best DB |
| 12:43PM |
0 |
joinempty=no |
| 12:10PM |
0 |
[Asterisk-Dev] 1.0.7 Release Candidate |
| 12:08PM |
4 |
Upgrading Asterisk |
| 11:48AM |
0 |
Specifing a linker when building Asterisk |
| 11:41AM |
2 |
Broadvoice latest changes and still not working-An |
| 11:23AM |
2 |
Assistance with Overhead Paging |
| 11:02AM |
1 |
max number of conference rooms, and max number of conference callers in one room |
| 10:16AM |
0 |
Fwd: Re: Broadvoice latest changes and still not working- An Additional Server ****SOLVED**** |
| 10:12AM |
1 |
Edit MGCP response |
| 10:01AM |
3 |
Polycom IP 500 bitmaps and Idle Display Animation |
| 9:48AM |
0 |
sip hangup detection problem |
| 9:37AM |
1 |
Asterisk 1.0.6 and chan_sccp problems? |
| 9:31AM |
9 |
Print-to-Fax client |
| 9:07AM |
1 |
Cisco 7960 Protocol Invalid when Upgrading to 7.3 |
| 8:47AM |
2 |
Telecom echo cancel disable |
| 8:15AM |
0 |
RE: : RE: Re: MGCP to Inter Tel system |
| 7:36AM |
0 |
iaxy stopped working |
| 7:33AM |
0 |
Unable to dial out using HFC ISDN card |
| 6:59AM |
2 |
Which hardware for this solution? |
| 6:47AM |
0 |
Cicso 7912 phones 3 out of 8 not grabbingthegk<MAC> file |
| 6:42AM |
1 |
Broadvoice latest changes and still not working-An Additional Server |
| 6:41AM |
3 |
NuFone + VoIPJet = busy busy busy |
| 6:39AM |
0 |
Zyxel P2000W - CallerId |
| 6:34AM |
2 |
TDM400P slow getting line tone |
| 6:19AM |
2 |
Echo for first 15 to 20 seconds |
| 5:31AM |
4 |
Which box? |
| 5:26AM |
3 |
Regarding Incoming Calls on PRI |
| 4:51AM |
1 |
Should ICMP port unreachable generate a BYE request? |
| 4:30AM |
0 |
Call through. with 2xT1 .configuration |
| 4:26AM |
1 |
IAX Music on hold |
| 4:14AM |
6 |
how to sip->h323 using asterisk-oh323-0.7.1 |
| 4:03AM |
0 |
Asteriks@home |
| 3:24AM |
2 |
Voicemail - No Audio Output! |
| 3:19AM |
0 |
Subject: Re: What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile |
| 12:38AM |
2 |
Another Newbie Question |
| 12:13AM |
1 |
i am missing something! |
| |
| Tuesday March 8 2005 |
| Time | Replies | Subject |
|---|
| 11:46PM |
2 |
Connect asterisk on classic pbx with T100P card |
| 10:44PM |
1 |
Broadvoice latest changes and still not working- An Additional Server |
| 8:47PM |
1 |
New Help Site - cut down on Mailing List questions |
| 7:29PM |
0 |
Could dialing long extensions be a problem? |
| 7:20PM |
1 |
All Circuits are Busy Now |
| 7:12PM |
1 |
Dial() out and offer a menu system |
| 7:05PM |
5 |
Polycom IP600 Phantom Ringing |
| 6:38PM |
1 |
Broadvoice-like company in Canada? |
| 6:31PM |
0 |
Sip 400 bad request - broadvoice error |
| 5:05PM |
1 |
STOP NOW not responding |
| 4:56PM |
1 |
Voicetronix Tones |
| 3:34PM |
1 |
Cicso 7912 phones 3 out of 8 not grabbing thegk<MAC> file |
| 3:08PM |
1 |
Cicso 7912 phones 3 out of 8 not grabbing the gk<MAC> file |
| 2:38PM |
0 |
Re: Asterisk-Users Digest, Vol 8, Issue 63 |
| 2:30PM |
4 |
Nortel ATA not passing dtmf tones to fxo |
| 2:25PM |
2 |
Please help with install * SOLVED |
| 2:18PM |
2 |
Broadvoice users... |
| 2:15PM |
1 |
Asterisk Interop w/ Level 3 |
| 2:04PM |
3 |
DID in the U.S. |
| 1:49PM |
0 |
Broadvoice latest changes and still not working - solved HEYYY |
| 1:36PM |
1 |
Forwarded call flag |
| 1:27PM |
0 |
Play music on hold while waiting for DTMF? |
| 1:24PM |
1 |
SIP - Call Park/Pickup and Canreinvite=yes at the same time?? |
| 1:22PM |
0 |
determining an available channel question |
| 1:07PM |
1 |
Adit 600 for asterisk |
| 12:55PM |
2 |
Cisco 7940 Upgrade Failing |
| 12:46PM |
2 |
GotoIf with Authenticate |
| 12:16PM |
3 |
DTMF out to Cell Phone |
| 12:09PM |
0 |
Does anybody have Broadvoice outbound working? |
| 11:50AM |
1 |
7960 Dies when network cable connected |
| 11:29AM |
11 |
GotoIf problem |
| 10:56AM |
4 |
Wildcard X100P or TDM400P? |
| 10:45AM |
4 |
force SIP authentication |
| 10:45AM |
2 |
Incoming Fax Service question |
| 10:19AM |
1 |
using the i extension |
| 10:09AM |
2 |
zaphfc error |
| 9:49AM |
2 |
Asterisk Management API |
| 9:13AM |
1 |
How does asterisk do the routing? |
| 9:02AM |
0 |
2 Asterisk servers (IAX) behind one firewall |
| 8:45AM |
5 |
Please help with install * |
| 8:30AM |
1 |
Asterisk provides ring tone? |
| 7:57AM |
0 |
problem in compiling chan_mISDN |
| 7:45AM |
2 |
problem in compiling openh323 |
| 7:10AM |
13 |
Broadvoice latest changes and still not working |
| 6:49AM |
3 |
NAT Far End Traversal |
| 6:15AM |
1 |
CallerID - Broadvoice vs. VoicePulse |
| 6:04AM |
1 |
TDM22B in the UK on BT |
| 5:35AM |
0 |
xc-ast 0.8.0 is out |
| 5:06AM |
2 |
Queue and SetGroup |
| 3:31AM |
2 |
Retreiving the called number |
| 2:43AM |
2 |
looking for cheap 4 port FXS card |
| 12:08AM |
3 |
Cisco 7960 Problem - Phone Unprovisioned |
| |
| Monday March 7 2005 |
| Time | Replies | Subject |
|---|
| 11:38PM |
0 |
How work by Asterisk and SER ? |
| 10:15PM |
1 |
What combination of pwlib and openh323 are |
| 9:38PM |
1 |
video confrencing |
| 9:25PM |
3 |
UNISTIM channel driver available |
| 8:41PM |
1 |
What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile |
| 8:19PM |
0 |
SpanDSP Status Messages |
| 7:58PM |
2 |
[Asterisk-Dev] TE405P/410P (Quad-T1/E1) driver |
| 5:50PM |
1 |
[Fwd: RE: Re: TE410P card in an HP-Compaq DL380 G4 server] |
| 5:42PM |
9 |
Question with email notification |
| 4:52PM |
3 |
Polycom SP300 questions |
| 4:49PM |
0 |
CAPI trunks |
| 4:31PM |
0 |
Dock-n-talk connection to asterisk |
| 3:36PM |
5 |
[Asterisk-Dev] Flash Operator Panel |
| 3:22PM |
0 |
Asterisk@Home and VoiceMail |
| 3:21PM |
3 |
grandstream budgetone 101 |
| 3:21PM |
2 |
Question about AGI vs. FastAGI vs. straight C/DB development |
| 3:14PM |
8 |
Call Forward or DND |
| 2:19PM |
1 |
working system for months suddenly stopped today with Failed to authenticate on INVITE to - additional |
| 1:57PM |
0 |
Dial, record, save to voicemail |
| 1:56PM |
2 |
[Asterisk-Dev] Polycom IP 600 XML |
| 1:20PM |
1 |
FXO module in TDM400P (UK, BT) - Hangup |
| 1:05PM |
5 |
Asterisk & MySQL Blobs |
| 1:00PM |
1 |
multiple outside phones |
| 11:50AM |
2 |
Setting up asterisk with current PBX? |
| 11:39AM |
1 |
working system for months suddenly stopped today with Failed to authenticate on INVITE to |
| 11:23AM |
6 |
Tweaking AGGRESSIVE_SUPPRESSOR |
| 11:02AM |
2 |
DTMF to Email |
| 10:42AM |
1 |
3COM 3101 SIP |
| 10:41AM |
0 |
iax2 setvars help needed |
| 10:00AM |
1 |
CAPI questions |
| 9:59AM |
0 |
anybody tried Fujitsu-Siemens PRIMERGY RX200 S2 server width te4xx? |
| 8:40AM |
1 |
MP3 stream for MOH |
| 7:56AM |
2 |
2-Ring Delay for CLID |
| 7:31AM |
1 |
chan_sip not 100% RFC3665 compliant - re-REGISTERsfail. |
| 7:27AM |
2 |
SIP and ISDN |
| 7:08AM |
1 |
MGCP howto |
| 7:03AM |
2 |
Where to get (cheap) VoIP |
| 6:54AM |
0 |
chan_sip not 100% RFC3665 compliant - re-REGISTERs fail. |
| 6:17AM |
0 |
Open files / socket leak |
| 6:10AM |
0 |
CVS compile error utils.c |
| 4:25AM |
0 |
Asterisk & Fritz & Capi & isdn PBX integration : Can I dial out on any MSN I declare ? |
| 3:44AM |
2 |
Call transfer questions |
| 3:08AM |
1 |
Exec AGI after hangup. |
| 2:47AM |
0 |
Sip phone service for linux |
| 2:40AM |
1 |
Custom Development |
| 2:10AM |
0 |
DID Functionality with POTS and Digium TDM04B |
| 1:31AM |
0 |
SIP URI |
| |
| Sunday March 6 2005 |
| Time | Replies | Subject |
|---|
| 10:16PM |
0 |
How to configure directory within voicemail transfer to search by first name? |
| 10:00PM |
0 |
Zaptel in New Zealand: Caller id vs loadzone |
| 9:18PM |
0 |
Signate is now offering the dCAP test. |
| 6:48PM |
0 |
Loopback |
| 5:39PM |
0 |
Re: Broadvoice configuration changes for outbound calls |
| 5:15PM |
0 |
[Fwd: Re: BroadVoice configuration changes for Outbound] |
| 4:31PM |
1 |
Re: [Asterisk-biz] Livevoip U.S. 800 LNP Starts March 9th 2005 |
| 4:22PM |
1 |
IP Providers pass CallerID? |
| 4:09PM |
1 |
SER -> Asterisk voicemail on busy/unavailable. Anyone did it? (googling says NO) |
| 4:02PM |
1 |
Music Volume ? |
| 3:37PM |
2 |
Trying to get 2 SIP phones to work |
| 1:42PM |
1 |
SpanDSP: Training failed (sequence failed) |
| 1:41PM |
2 |
Need help on * anf HFC. |
| 1:27PM |
3 |
SNMP and Astersik |
| 1:00PM |
1 |
Dial option g |
| 9:15AM |
3 |
Zaptel.conf and multiple T1 woes |
| 7:12AM |
3 |
SJphone on PDA registering with Asterisk??? |
| 6:31AM |
0 |
Dial Macro |
| 1:20AM |
1 |
IAX - Registration Problems |
| |
| Saturday March 5 2005 |
| Time | Replies | Subject |
|---|
| 11:28PM |
0 |
Budgetone 101 Hold/Xfer/Conf/Flash |
| 8:13PM |
0 |
Is anybody having problems with sixtel? |
| 6:00PM |
6 |
Survey: what's the best HTTPd/TFTPd/FTPd to serve up configuration files to sets |
| 5:14PM |
2 |
SIP VoIP Provider problems |
| 4:57PM |
0 |
Asterisk patches - location and use |
| 4:34PM |
3 |
Sayson 480i Fails to Re-register? |
| 3:26PM |
1 |
IAX2 (Variables) |
| 2:40PM |
0 |
DVG-1120M -> S |
| 2:30PM |
4 |
Digium hardware in the UK ? |
| 2:17PM |
1 |
Zultys Zip 2 |
| 2:10PM |
1 |
Digium Reseller in the UK ? |
| 1:08PM |
1 |
Dead SCCP client since upgrade to Asterisk 1.0.6-BRIstuffed-0.2.0-RC7j? |
| 1:00PM |
0 |
Capi installation with Fedora Core 3 (AVM Fritz!) |
| 12:57PM |
1 |
IAX Softphones |
| 11:40AM |
0 |
How do I reload extensions included in a switch statement in extensions.conf? |
| 10:50AM |
3 |
Sorry to be a bother ISO root password |
| 10:16AM |
3 |
Asterisk for Live-Stream? |
| 10:13AM |
7 |
BroadVoice configuration changes for Outbound |
| 9:05AM |
0 |
how to optimize sip?? |
| 7:34AM |
1 |
Block anonymous calls |
| 7:31AM |
4 |
Newbie guidance requested --- Grandstream Budgetone |
| 7:06AM |
0 |
Unable to transfer timed out calls from call parking |
| 7:04AM |
1 |
Problem with loging on guest account |
| 6:50AM |
0 |
signaling problems |
| 6:19AM |
2 |
Getting asterisk-addons installed on Debian? |
| 6:14AM |
1 |
X100P Clone, Which one? |
| 5:57AM |
0 |
Automatically send monitored call files by e-mail |
| 4:58AM |
0 |
change proxy after timeout |
| 4:18AM |
0 |
Asterisk 1.0.3 Periodically Fails Registrations |
| 4:16AM |
0 |
Re: Is anyone using asterisk in a small call |
| 3:25AM |
1 |
SAY DIGITS problem |
| 2:19AM |
2 |
cant compile app_meetme2 |
| 2:15AM |
0 |
ASTCC questions: Userconfig, sip friends, iax friends and multiple trunks in routes |
| 1:17AM |
0 |
Are codec "capabilities bitmasks" different in IAX and SIP? |
| 12:54AM |
1 |
Unable to create channel of type IAX2 |
| |
| Friday March 4 2005 |
| Time | Replies | Subject |
|---|
| 11:58PM |
4 |
Difference between Snom 190 & Elmeg 290? |
| 9:57PM |
1 |
Asterisk Brochure |
| 9:30PM |
1 |
ANNOUNCEMENT: Updates for app_cbmysqlandMeetMe2gui (out of tree modules) |
| 8:35PM |
0 |
ANNOUNCEMENT: Updates for app_cbmysql andMeetMe2gui (out of tree modules) |
| 8:18PM |
5 |
LiveVoIP Problems? |
| 8:10PM |
7 |
Stutter Tone |
| 8:03PM |
2 |
ANNOUNCEMENT: Updates for app_cbmysql and MeetMe2gui (out of tree modules) |
| 8:00PM |
1 |
Log Error |
| 7:17PM |
3 |
ANNOUNCEMENT: Updates for app_cbmysql and MeetMe2 gui (out of tree modules) |
| 6:32PM |
0 |
Asterisk with mediant 2000 - facing problems |
| 4:42PM |
2 |
Multiple telephone participants |
| 4:07PM |
3 |
music on hold issue |
| 3:51PM |
0 |
Chan_Capi + HFC Card |
| 3:41PM |
0 |
Size of installations |
| 3:37PM |
2 |
Is anyone using asterisk in a small call center |
| 3:27PM |
1 |
PRI HDLC Abort (6) Errors |
| 3:18PM |
0 |
TE405P and quality problem |
| 2:34PM |
4 |
Im a noob |
| 2:21PM |
1 |
Placing a call from command line and passingit to an extension if connected - Is it possible? |
| 2:08PM |
1 |
chan_h323 & codecs |
| 2:00PM |
0 |
Monitor Application with Queued calls |
| 1:58PM |
2 |
IAX Codec |
| 1:55PM |
1 |
Has anyone got early dial working on asterisk ? |
| 1:45PM |
2 |
Options for Attendant Console. |
| 1:44PM |
2 |
Voice over Frame Relay & Asterisk |
| 1:41PM |
2 |
Placing a call from command line and passing it to an extension if connected - Is it possible? |
| 1:39PM |
2 |
Asterisk box and verizon calling it |
| 1:26PM |
4 |
Hardphone deployment recommendation |
| 1:08PM |
0 |
chan_capi patch for the new cvs HEAD |
| 1:01PM |
1 |
SIP MWI and MySQL Realtime |
| 12:43PM |
1 |
Web based tool asterisk real time |
| 12:24PM |
2 |
Bluetooth phone as SIP handset? |
| 11:24AM |
3 |
ANNOUNCEMENT : Asterisk-Stat V2.0 - CDR Analyser |
| 10:53AM |
2 |
budgetphone |
| 9:53AM |
0 |
SMS in 1.0.6 |
| 9:33AM |
1 |
defold usernames in asterisk@home version 6 |
| 9:04AM |
3 |
[OT] - Why should I answer a Newbie questio n,therethick! |
| 9:01AM |
1 |
[OT] - Why should I answer a Newbie questio n, therethick! |
| 9:00AM |
2 |
IAX on netweb EEZEE phone |
| 8:48AM |
1 |
X100P in the UK - seems to short the dialtone |
| 8:38AM |
2 |
Broadvoice + incoming call works only for ~2 minutes |
| 7:54AM |
0 |
Asterisk ---Toshiba |
| 7:31AM |
1 |
Problem getting Voice Contract script to work |
| 7:02AM |
0 |
Connection time of Transferred Calls |
| 6:23AM |
0 |
Problem with inbound call quality. |
| 6:01AM |
2 |
TE110P module woes |
| 5:51AM |
1 |
chan_capi with patch compilation error |
| 5:48AM |
1 |
mISDN not initialising properly my Fritz cards |
| 5:29AM |
2 |
Problems with g729 codec |
| 4:06AM |
1 |
Zap channels intermittently bridging with SNOM190 |
| 3:52AM |
2 |
Answering Machine Detection with app_machinedetect.c |
| 3:27AM |
0 |
* intergation with Panasonic D500 and strange echo |
| 3:27AM |
1 |
Asterisk@home 0.6 + mISDN |
| 3:02AM |
1 |
Bristuff e RealTime: STABLE vs. CVS-HEAD |
| 2:55AM |
0 |
SIP hard phones choice |
| 2:21AM |
1 |
dialing from a website. How to start...? |
| 1:51AM |
0 |
notes: www.voicematch.cc & speex 1.1.7, unrelated |
| 12:06AM |
0 |
why I don't do this test ? |
| |
| Thursday March 3 2005 |
| Time | Replies | Subject |
|---|
| 10:41PM |
0 |
Asterisk SIP client problem |
| 9:48PM |
0 |
I have met a message : "No one is available to answer at this time". |
| 6:23PM |
1 |
Options in Brazil |
| 6:03PM |
3 |
Audio pausing over IAX trunk |
| 5:31PM |
4 |
[OT] - Why should I answer a Newbie question, therethick! |
| 5:04PM |
3 |
Problems dialing out - possible settings changes |
| 4:33PM |
0 |
FW: (still problems) Dialing phone number and extension together to avoid listening to voice menu (incoming call) |
| 4:27PM |
2 |
Beginning with Asterisk |
| 4:20PM |
0 |
Vovida Load Balancer. |
| 4:01PM |
0 |
Netphone KE1020A with asterisk |
| 3:57PM |
3 |
defold passwords in asterisk@home version 6 |
| 3:21PM |
1 |
Is there a way to find free zap channels on remote servers ?? |
| 3:20PM |
1 |
Development help? |
| 3:17PM |
2 |
FWD and SIPPHONE problems after upgrading to CVS HEAD |
| 2:46PM |
2 |
[Asterisk-Dev] CVS-HEAD change: queue/agent persistence |
| 2:33PM |
3 |
Asterisk not relaying back the SIP response messages |
| 2:27PM |
0 |
New user - problem getting dtmf tones through VOIP providers? |
| 2:20PM |
2 |
Help for studying Asterisk source code |
| 2:11PM |
3 |
Why ${EXTEN} variable changes after Goto ? |
| 2:08PM |
0 |
Calling Card Platform |
| 1:46PM |
0 |
fax and codecs |
| 1:38PM |
3 |
Update Asterisk |
| 1:23PM |
4 |
MGCP to Inter Tel system |
| 12:57PM |
4 |
DyDNS + externip |
| 12:24PM |
2 |
Asterisk + SIP + NAT - seriously, what's the secret? |
| 12:10PM |
0 |
Lines to PSTN available in FXO |
| 12:00PM |
1 |
Asterisk@Home .6 Problems with outbound calls using Broadvoice |
| 11:33AM |
2 |
ZAP Line answer questio |
| 11:13AM |
0 |
SIP secret: argument only for outgoing |
| 11:04AM |
0 |
Upgrading the 7960 Image |
| 10:15AM |
1 |
Blacklists. |
| 10:13AM |
2 |
Attended Transfer (ATXFER) with CVS asterisk r 1_ |
| 10:04AM |
5 |
kernel error with Zaptel cards |
| 9:47AM |
0 |
Teles GW authentification |
| 9:05AM |
1 |
Voice recognition with Asterisk |
| 8:59AM |
0 |
Warning Message with voicemail CVS 3-3-05 |
| 8:48AM |
0 |
Recomended server hardware |
| 8:38AM |
0 |
IAX users in Japan or Taiwan? |
| 8:08AM |
3 |
Detect sound and continue, like BackgroundDetect() for voice |
| 7:37AM |
0 |
Re: More NAT questions -- SOLVED |
| 7:25AM |
5 |
country/city codes |
| 7:22AM |
0 |
problem registering a bt100 with 1.0.5.11 firmware |
| 7:04AM |
1 |
IAXy and Private IP |
| 6:58AM |
0 |
Some errors on sip debug |
| 6:40AM |
2 |
Calling hangup in background |
| 5:50AM |
4 |
Getting phpconfig to work? |
| 5:43AM |
2 |
Installing modules for TDM400p |
| 5:27AM |
1 |
capi debugging |
| 5:02AM |
5 |
how do i get rid of this blasted echo !!! |
| 4:46AM |
2 |
Re : Calling card platform |
| 4:22AM |
5 |
Wrong CVS version ? |
| 3:51AM |
0 |
RE: Getting phpconfig to work? |
| 3:25AM |
0 |
Realtime IAX/SIP with 2 asterisk servers but 1 central iax/sipfriends Database |
| 3:24AM |
1 |
RE: Getting phpconfig to work? |
| 3:17AM |
0 |
RE: Getting phpconfig to work? |
| 3:04AM |
0 |
RE: Getting phpconfig to work? |
| 3:03AM |
1 |
RE: Getting phpconfig to work? |
| 2:47AM |
2 |
Multitenant feature |
| 1:53AM |
0 |
Forward Call from Asterisk to SER |
| 1:12AM |
0 |
Update on the blending of app_cbmysql and app_meetme2 (out of tree modules) |
| |
| Wednesday March 2 2005 |
| Time | Replies | Subject |
|---|
| 11:00PM |
0 |
best calling card platform for asterisk |
| 9:02PM |
1 |
Asterisk 1.0.6 music-on-hold |
| 7:50PM |
1 |
Building Asterisk with CentOS |
| 7:00PM |
0 |
super macro |
| 5:14PM |
1 |
Searchable Asterisk-users archive available |
| 4:14PM |
0 |
Windows Messenger and 481 error |
| 3:29PM |
1 |
Dial Application/redirection on demand |
| 3:22PM |
2 |
How use Spanish / English prompts on same box |
| 3:08PM |
0 |
IAX trap question |
| 2:50PM |
0 |
Way to disable "#" as transfer and just take thekey. |
| 2:49PM |
4 |
timing/clock problem |
| 2:39PM |
5 |
Asterisk URL and Callcenter Apps |
| 2:19PM |
3 |
Way to disable "#" as transfer and just take the key. |
| 2:19PM |
2 |
Has anyone seen this before |
| 2:11PM |
1 |
Getting Polycom IP500 to talk to Asterisk - um... Newbie question :) |
| 2:01PM |
4 |
Music on hold on timing sources |
| 1:44PM |
0 |
Call Forwarding to Cell Phone, Pager, etc |
| 1:24PM |
3 |
[OT] stupid firmware question... |
| 1:02PM |
2 |
Polycom Soundpoint 500/600 MiniBrowser |
| 12:32PM |
4 |
Sending Voicemail's to two email addresses |
| 12:05PM |
0 |
IAX & LAGRQ & POKE explanation |
| 11:50AM |
0 |
Asterisk SKINNY with Cisco IP Conference 7935 |
| 11:04AM |
0 |
IP300 soft key configuration |
| 10:39AM |
1 |
Asterisk HEAD and Mysql problems |
| 10:34AM |
0 |
OT: Looking for asterisk integrators in Dallas,TX |
| 10:22AM |
0 |
Fax with spandsp + zaphfc |
| 10:04AM |
1 |
Dial application invoked again and again |
| 8:35AM |
0 |
TE405P/zttool |
| 8:22AM |
2 |
Dual Asterisk Servers |
| 8:16AM |
3 |
More NAT questions |
| 8:02AM |
3 |
/dev/zap not created |
| 7:58AM |
3 |
Multiple lines |
| 7:36AM |
0 |
chan_capi - fax patch - crash |
| 6:33AM |
0 |
asterisk-oh323 bugtracker |
| 5:50AM |
3 |
Asterisk Manager API - multi "Originate" cal ls |
| 5:46AM |
3 |
cvs stable and 1.0.5 |
| 5:40AM |
2 |
Dual X100P cards |
| 5:28AM |
1 |
Asterisk Manager API - multi "Originate" calls |
| 4:54AM |
0 |
Help needed with installing ZAPHFC |
| 4:53AM |
1 |
e164.org and FWD now have peering arrangement |
| 4:14AM |
0 |
[Asterisk-Dev] Digium's G.729A codec problem |
| 2:40AM |
4 |
wctdm and two tdm cards |
| 2:38AM |
8 |
Why should I answer a Newbie question, there thick! |
| 2:02AM |
1 |
Incorrect CDRs |
| 1:51AM |
1 |
IVR setup problems |
| 1:07AM |
0 |
Send parameters from asterisk to ADSI phone |
| 12:45AM |
1 |
Addons Make Problems! HELP! |
| |
| Tuesday March 1 2005 |
| Time | Replies | Subject |
|---|
| 11:08PM |
1 |
Call waiting in Australia |
| 11:06PM |
0 |
MWI work with ast_data? |
| 11:03PM |
0 |
IAX+G729a |
| 9:00PM |
0 |
How could Asterisk help me on a Internet webcast speech!? |
| 8:37PM |
0 |
Echoing Beep on ZAP channel (one sided) |
| 8:04PM |
1 |
"n" priority not in 1.0.6 |
| 7:18PM |
1 |
iax notransfer=no and Tt in Dial() |
| 5:33PM |
0 |
AMP with Sipura 3000 PSTN line |
| 2:51PM |
1 |
OH323_OUTCODEC Unsupported |
| 2:17PM |
0 |
IAX or SIP answering services |
| 1:53PM |
4 |
"No compatible codecs!" -- worked with 1.0.0, not 1.0.6 or CVS. |
| 1:29PM |
0 |
Dialing phone number and extension together to avoid listening to voice menu (incoming call) |
| 1:05PM |
1 |
dropping extra frame..already have it???? |
| 12:40PM |
6 |
Broadvoice + Videosupport=yes - Fails! |
| 12:33PM |
1 |
www.voicematch.cc |
| 11:58AM |
0 |
Looking for tech in San Francisco |
| 11:42AM |
2 |
agi RECORD FILE with offset |
| 11:31AM |
9 |
MozPhone |
| 10:52AM |
1 |
Re: FRS over * |
| 10:32AM |
1 |
[Asterisk-biz] IAX2 web client that works withg723 / g729. We got One |
| 10:05AM |
0 |
RE: Big increase in SPAM lately |
| 10:03AM |
0 |
[Asterisk-biz] IAX2 web client that works with g723 / g729. We got One |
| 10:00AM |
1 |
Big Increase in SPAM over the last few weeks |
| 10:00AM |
0 |
New Integrics Tip: Recording Voice Prompts |
| 9:47AM |
0 |
FW: SIP Phone Choices |
| 9:35AM |
0 |
SIP Client at outside and connect to an Asterisk Server sit behind NAT with SER |
| 9:26AM |
1 |
Connecting Asterisks via SIP |
| 9:08AM |
3 |
Ordering a Voice PRI for Asterisk |
| 8:24AM |
2 |
Important :: Please support the development of a new Jitterbuffer for SIP |
| 8:11AM |
5 |
Polycom Auto-Answer |
| 7:43AM |
2 |
multiple Fritz ISDN/BRI PCI |
| 7:41AM |
2 |
mini atx and asterisk (EPIA and the like) |
| 7:41AM |
0 |
Addons compile errors |
| 7:16AM |
1 |
Problems Starting Asterisk - FOP AM Portal |
| 7:14AM |
1 |
Newbie - What Do I Need? |
| 7:00AM |
1 |
Some asterisk ser problems |
| 6:55AM |
2 |
openh323 |
| 6:21AM |
0 |
chan_sccp and 7912 |
| 6:15AM |
9 |
What my IAXy could have been... |
| 5:49AM |
0 |
Is the Siemens SX353 (DECT) Base Station compatible with *? |
| 5:34AM |
1 |
Sipura 3000 Inbound Dialing Problem |
| 5:03AM |
0 |
Voicemail advanced options |
| 5:02AM |
1 |
Cisco 7940, Voicemail & DTMF |
| 4:51AM |
1 |
Music on hold..Mar error "res_musiconhold.c:309 monmp3thread: Request to schedule in the past" ? |
| 4:33AM |
2 |
Cisco 7960 x g729 x Unable to create/find channel |
| 4:07AM |
2 |
Park Craches asterisk |
| 3:57AM |
0 |
Incoming problem of Asterisk and Broadvoice |
| 3:02AM |
1 |
NoCDR Warning |
| 3:00AM |
0 |
UK CLID Asterisk CVS |
| 2:53AM |
1 |
in calling |
| 2:25AM |
1 |
DIAX 0.9.10f available for download |
| 2:02AM |
0 |
SV: chan_capi compile error on FC3 |
| 1:59AM |
0 |
RE: Pb DTMF with Asterisk vs Cirpack Transit, , Node |
| 12:47AM |
0 |
Advanced Conferencing optionswithout-of-treemodules? |