| Friday September 30 2005 |
| Time | Replies | Subject |
|---|
| 7:23PM |
0 |
How to get names into the *411 directory |
| 6:53PM |
1 |
is a dual 1.5Ghz server better than a single 3Ghz for a 100 Iax users asterisk server |
| 5:52PM |
2 |
Asterisk and RTP streams |
| 4:33PM |
1 |
Music on hold not initiating RTP stream? |
| 1:17PM |
3 |
SPA-841 "Decode Latency"? |
| 12:57PM |
4 |
Revieving some fax problems |
| 12:28PM |
0 |
Polycom IP301 Hangs on boot. |
| 11:56AM |
0 |
voip alarm circuit |
| 11:53AM |
1 |
Linksys register hangs Asterisk! |
| 11:28AM |
2 |
quick question on ztdummy |
| 11:16AM |
0 |
Co-author of O'Reilly's Asterisk book presenting in Utah Valley |
| 11:02AM |
1 |
X100p Problem, randomly hungup pstn line |
| 10:46AM |
0 |
oh323 implementation 0.67 has call-id problem |
| 10:21AM |
1 |
Best way to create IVR/voicemail system |
| 10:17AM |
1 |
(no subject) |
| 9:52AM |
2 |
SIP make outside call |
| 9:51AM |
4 |
C Manager Interface Client |
| 9:41AM |
1 |
Maximum number of Digium Trunk Cards |
| 9:06AM |
0 |
mISDN, HFC, W6692, one-way-voice problem |
| 8:51AM |
0 |
It is possible to have 2 AVM Fritz! USB for multiple BRI access? |
| 8:47AM |
0 |
Calls Dropping w/ Cisco 7960 Phones |
| 8:26AM |
1 |
Question about 3Com(r) 3101 Basic Phone |
| 7:58AM |
1 |
strange wave like noise on sip handset |
| 7:55AM |
1 |
Asterisk and telephone volume |
| 7:53AM |
1 |
No ringback tone generated by Asterisk with OH323connections |
| 7:31AM |
1 |
No ringback tone generated by Asterisk with OH323 connections |
| 7:26AM |
1 |
chan_zap.so ? |
| 7:20AM |
7 |
911 Q |
| 7:13AM |
2 |
OT: SIPSAK usage |
| 6:57AM |
2 |
Echo Cancellation not working in Zapata.conf |
| 6:47AM |
1 |
Not Authenticate |
| 6:37AM |
1 |
TE410P not working |
| 5:59AM |
0 |
R: chan_capi-0.3.5 |
| 5:40AM |
2 |
analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly |
| 5:32AM |
2 |
Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system? |
| 4:45AM |
1 |
VideoConference with UMTS |
| 4:43AM |
1 |
Register times out on internet outage |
| 4:36AM |
2 |
chan_capi-0.3.5 |
| 4:12AM |
2 |
Why does the s extension not work in my extensions.conf file |
| 3:58AM |
2 |
Diva |
| 3:27AM |
0 |
Compile broken on FreeBSD ? |
| 2:35AM |
0 |
[Fwd: TDM40B - "Unable to play dialtone on channel X" ?] |
| 2:20AM |
3 |
Zaptel TDM questions |
| 2:02AM |
1 |
Empty ACK |
| 2:01AM |
0 |
IAXPhone |
| 1:19AM |
4 |
G.729 patent in France |
| 12:31AM |
1 |
Siemens TC35 GSM gateway |
| 12:14AM |
0 |
* T.38 fax |
| |
| Thursday September 29 2005 |
| Time | Replies | Subject |
|---|
| 11:49PM |
0 |
[Asterisk-User] linux/Asterisk change ip address |
| 10:18PM |
0 |
please help on FreeTDS (writing CDR to MS-SQL or MySQL) |
| 8:49PM |
2 |
Is this normal? |
| 7:52PM |
0 |
Can't make outside call with SIP softphone |
| 5:56PM |
1 |
Voice Prompts, what do you think? Good voice. |
| 5:27PM |
1 |
SIP Gateway wants T38, Asterisk rejects but media path not established. |
| 4:08PM |
1 |
Meet me conferencing without blind transfers (Asterisk@home) |
| 3:29PM |
1 |
Using Realtime queues and queue members |
| 3:20PM |
1 |
Mathematicians wanted (was RE: Best echo canceller?) |
| 2:58PM |
1 |
files conflict after CVS update |
| 2:39PM |
0 |
dtmfmode type |
| 2:04PM |
2 |
Best echo canceller? |
| 1:52PM |
3 |
Auto Answer Fax |
| 1:39PM |
0 |
FWD via Trunk from DMZ to LAN |
| 1:30PM |
1 |
Cisco AS5300 --> [SIP] --> Asterisk - NO AUDIO |
| 1:28PM |
3 |
Broadvoice inbound issues |
| 12:59PM |
0 |
TDM40B - "Unable to play dialtone on channel X" ? |
| 12:34PM |
3 |
Problems using SIPURA and MFC/R2 |
| 12:27PM |
3 |
FWD: '486 Busy here' and 'All Circuits are busy now' |
| 12:13PM |
0 |
Yada table in oracle |
| 12:12PM |
2 |
Asterisk for "Man-In-The-Middle" Trunk Side Call Recording? |
| 11:53AM |
0 |
Asterisk as a Voice Logger alternative to NICE or Witness Systems |
| 11:35AM |
2 |
Hardware Specifications |
| 11:31AM |
2 |
Unable to send fax using BroadVoice |
| 11:04AM |
4 |
Any way to not overwrite sound files on compile? |
| 10:42AM |
0 |
DTMF tones from PSTN not reaching SIP device |
| 9:15AM |
1 |
minor(? ) Grandstream phone issue |
| 7:44AM |
2 |
R: PRI value |
| 7:32AM |
1 |
Cannot figure out why calls from my Asterisk appear to be from country code +34? |
| 7:28AM |
4 |
OOH323C |
| 7:01AM |
0 |
Prueba |
| 7:01AM |
1 |
Re: [Asterisk-biz] Problem with sending fax froma SIP extension |
| 6:53AM |
2 |
Remotely dialing calls from a polycom phone |
| 6:49AM |
2 |
Getting asterisk to send e-mail to mailbox-users |
| 6:35AM |
0 |
Caller ID, Attended Transfers, Polycom |
| 6:13AM |
4 |
chan_cap-cm-0.6 deflect support |
| 5:53AM |
1 |
Asterisk Echo problems, Urgent, please help, |
| 5:49AM |
1 |
Audio Files, Filtering, and Formats for Asterisk |
| 5:17AM |
0 |
Major bug solved in IPSwitchBoard |
| 4:50AM |
1 |
sip calleridnum |
| 4:45AM |
0 |
Asterisk registering with vonage |
| 4:16AM |
1 |
chan_cap-cm-0.6 is not working for incomming calls |
| 4:02AM |
1 |
Variable in call parking |
| 3:00AM |
1 |
digits won't play |
| 2:38AM |
2 |
Don't call |
| 1:43AM |
4 |
Calling voicemail from external phone. |
| 1:36AM |
0 |
Re: Asterisk-Users Digest, Vol 14, Issue 178 |
| 1:10AM |
2 |
PRI value |
| 12:36AM |
1 |
zttest - 100% ? |
| 12:19AM |
0 |
Voice Prompts, what do you think? Good voice. Should we record a new prompt-set? |
| 12:15AM |
1 |
Dealt with IAreaNet before? |
| |
| Wednesday September 28 2005 |
| Time | Replies | Subject |
|---|
| 11:05PM |
1 |
Recording channels |
| 9:16PM |
2 |
* mod core dump help |
| 9:03PM |
0 |
Recommended wireless router to run Asterisk on OpenWRT |
| 5:38PM |
1 |
Does the 1.0.9 release contain the Broadvoice patches? |
| 5:35PM |
2 |
chan_capi-cm, Euro ISDN bus: 2 extensions on same BRI port not working |
| 5:08PM |
0 |
Problem redirecting to voicemail through a SIP proxy (Looks like a bug) |
| 4:46PM |
2 |
TE205P in loopback? |
| 4:17PM |
3 |
cisco phones problems |
| 4:07PM |
0 |
ISO SIP Based Conference Bridge Solution |
| 4:03PM |
0 |
No audio non channels and choopy sound to PSTN network |
| 3:12PM |
1 |
Motherboard for Digium card |
| 3:01PM |
2 |
asterisk 1.0.9 + spandsp 0.0.2pre20 = crash on boot |
| 1:40PM |
0 |
Upgrading * |
| 1:17PM |
1 |
Can I install latest oH323 on *@home |
| 12:49PM |
0 |
To get phone to ring in two or more places |
| 12:42PM |
4 |
T.38 Faxing |
| 12:18PM |
1 |
Tiny Echo on PSTN via Zaptel |
| 11:38AM |
1 |
Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net' |
| 11:18AM |
6 |
Music on Hold Quality |
| 11:18AM |
2 |
Zap FXO/FXS issues, 1.2.0-beta1 |
| 9:47AM |
3 |
ASTCC - INUSE Flag |
| 9:17AM |
1 |
Monitor in AGI |
| 9:13AM |
0 |
TDM-400 cards, technical limitations |
| 9:11AM |
0 |
DID's in CA, WA, BC, FL and NY |
| 9:10AM |
1 |
Correction: Asterisk sound files, audio bandwidth, and sound quality |
| 9:06AM |
1 |
Asterisk sound files, audio bandwidth, and sound quality |
| 8:57AM |
2 |
Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unkn own signalling method 'pri_net' |
| 8:56AM |
0 |
BAD echo problems with Sangoma and, Telstra |
| 8:49AM |
4 |
Delay in dial |
| 8:24AM |
5 |
Roll back from CVS Head to v1.09 |
| 7:53AM |
1 |
adit 600 mgcp.conf |
| 7:33AM |
0 |
digital receptionist pick up time |
| 7:26AM |
1 |
Where MeetMe application |
| 7:14AM |
0 |
Trying to cut out the paper work... |
| 7:10AM |
0 |
Does Asterisk just pass thru RTP if the codec is the same between two extensions? |
| 7:07AM |
0 |
Does Asterisk just pass thru RTP if the codec is the same between two extension? |
| 7:04AM |
0 |
[Asterisk-User] Does Asterisk just pass thru RTP if the codec is the same between two extension? |
| 6:35AM |
2 |
PSTN-GATEWAY |
| 6:34AM |
2 |
setting up asterisk as an sms central? |
| 6:28AM |
0 |
problems accessing directory |
| 6:09AM |
1 |
Asterisk in Production |
| 6:03AM |
1 |
Asterisk does not send "Setup acknowledge" on euroISDN E1 |
| 5:14AM |
15 |
Asterisk on windows |
| 3:10AM |
0 |
SV: Turn off echo-cancellation when fax is detected? |
| 3:10AM |
1 |
MeetMe error |
| 2:50AM |
0 |
call wating and call transfer |
| |
| Tuesday September 27 2005 |
| Time | Replies | Subject |
|---|
| 10:58PM |
4 |
Voice Encryption |
| 10:49PM |
2 |
Auto CallBack on busy |
| 10:33PM |
1 |
oH323 Voice in one direction only |
| 8:50PM |
1 |
IAX2 encryption of data packets? |
| 8:27PM |
0 |
linux dist. and kernel version |
| 7:02PM |
0 |
7960 show queue status |
| 5:23PM |
2 |
Sipura 2000 Dial Plan |
| 5:09PM |
4 |
BAD echo problems with Sangoma and Telstra |
| 4:19PM |
1 |
Re: [Asterisk-biz] Problem with sending fax from a SIP extension |
| 4:17PM |
1 |
Extensions go straight to voicemail |
| 2:43PM |
5 |
Canada VOIP provider quality |
| 2:17PM |
4 |
Hook Flapping on Cisco 7960 |
| 2:00PM |
1 |
Creating an OPX from a traditional PBX using Asterisk and a SIP device |
| 1:38PM |
0 |
AstriCon 2005 - Now With Free Beer! |
| 12:40PM |
2 |
Review: Digium TE405P v2 |
| 12:17PM |
0 |
cgi-bin/vmail.cgi - - Invalid Context |
| 11:54AM |
1 |
SIP Tandem Inbound only. |
| 11:46AM |
0 |
asterisk@home inbound call problem to SIP trunk. (voipfone UK) |
| 11:22AM |
2 |
One-way audio with VPN |
| 10:51AM |
1 |
blindxfer & atxfer not working? |
| 10:21AM |
2 |
Polycom IP 500 - problem dialing extra numbers |
| 10:18AM |
1 |
[MSG]TDM Error on ASUS Pundit-R |
| 10:10AM |
3 |
analogue phone with asterisk |
| 9:49AM |
2 |
How to change ${VM_DATE} in voicemail.conf |
| 9:20AM |
1 |
VoIP Buster stopped working? |
| 8:18AM |
0 |
Asterisk & European Digital CAS Help |
| 8:12AM |
10 |
Software only Asterisk PBX (commercial) |
| 8:04AM |
1 |
Moaning dog... |
| 7:51AM |
0 |
405 "Method Not Allowed" error |
| 7:45AM |
0 |
function LEN missing |
| 6:06AM |
2 |
IAX2 hard phone |
| 4:42AM |
1 |
wait before accepting the call |
| 3:33AM |
1 |
R: Best drivers for HFC-S ISDN cards |
| 3:23AM |
0 |
Turn off echo-cancellation when fax is detected? |
| 3:18AM |
0 |
* Accounting with Oracle |
| 3:06AM |
0 |
radius and * |
| 2:42AM |
0 |
Listening for DTMF when dialling (sorry, accidentally sent the previous message too early!) |
| 2:35AM |
0 |
Listening for DTMF when dialling |
| 2:23AM |
1 |
R: Problem setting up TDM22B card |
| 2:19AM |
1 |
failed make install on Solaris 10 |
| 1:20AM |
2 |
Integration with NMS AG-E1/T1 |
| 12:36AM |
1 |
pbx_wilcalu.so: undefined symbol: |
| |
| Monday September 26 2005 |
| Time | Replies | Subject |
|---|
| 11:23PM |
0 |
"Non-blocking" Dial (and other commands): is there a way? |
| 11:22PM |
1 |
Bad FCS nightmare to Nortel SL100 with TE410P |
| 9:22PM |
1 |
IAX provider w/Toronto & Detroit termination |
| 9:08PM |
5 |
SPA-3000 and incoming faxes |
| 8:41PM |
0 |
ICD with asterisk |
| 8:39PM |
0 |
asterisk fifo |
| 8:27PM |
1 |
StripMSD or extension parser bug? |
| 8:10PM |
0 |
Flash Panal |
| 7:30PM |
0 |
system() app changed drastically! How do I useit now? |
| 7:27PM |
3 |
re: DTMF woes, continued |
| 7:14PM |
1 |
AsteriskJava - Queue |
| 7:10PM |
0 |
system() app changed drastically! How do I use itnow? |
| 6:15PM |
1 |
system() app changed drastically! How do I use it now? |
| 6:01PM |
0 |
Faxing via a sip extension with a digium e1 card |
| 4:47PM |
0 |
TE110P Hanging up & sometimes not picking up on E&M T1 |
| 4:09PM |
1 |
Socket 478 Motherboard for use with TDM400P |
| 3:33PM |
0 |
CPU spiking with TDM400 cards fixed |
| 3:26PM |
1 |
voipbuster advise |
| 3:18PM |
0 |
netappel |
| 3:00PM |
0 |
Asterisk Realtime.. : Unixodbc drivers |
| 2:39PM |
2 |
What ISDN hardware would you recommend? |
| 2:32PM |
4 |
Polycom Setup Questions |
| 2:31PM |
0 |
ZapHFC Channel unavailable |
| 2:24PM |
1 |
Grandstream 496 not working on cordless phone |
| 2:17PM |
1 |
how to connect two SIP channels |
| 2:10PM |
0 |
Areskicc LCR problem |
| 2:03PM |
0 |
Performance tuning on dual Xeon EM64T and x86_64 Linux |
| 1:27PM |
1 |
Dialogic Cards Will they be available to NON AsteriskBE |
| 1:06PM |
3 |
IBM x306 - some progress |
| 12:35PM |
1 |
FSX/UK analogue Phone rings all the time |
| 11:38AM |
0 |
CAS Question |
| 11:28AM |
1 |
Re: Ring requested on channel already in use |
| 11:25AM |
1 |
Carrier Access - Access Bank I config |
| 11:17AM |
3 |
asterisk SMS and sprintpcs |
| 10:56AM |
6 |
Extension availabilty |
| 10:48AM |
2 |
Early Media in 180 Ringing |
| 10:46AM |
0 |
IptablesAsterisk |
| 10:43AM |
0 |
? In CLI not working |
| 10:37AM |
1 |
goiax caller ID |
| 10:07AM |
3 |
Sangoma and Digium same machine? |
| 10:00AM |
1 |
Early Media in 100 Ringing |
| 9:33AM |
1 |
I want to send oH323 calls to our Quintum D3000 which is connected to a PSTN |
| 8:51AM |
0 |
BRI ISDN on USB |
| 8:25AM |
0 |
Recent Sphinx integration work? |
| 7:37AM |
0 |
CheckGroup accross multiple servers |
| 6:45AM |
1 |
Call Back On Busy? |
| 6:41AM |
2 |
Subject: Vonage-type service |
| 5:30AM |
0 |
Asterisk::AGI - What license ??? |
| 4:37AM |
0 |
Will Digium Wildard work with PCI-Xor PCI Express |
| 4:05AM |
0 |
dialing selected text with asterisk under windows ... |
| 2:35AM |
1 |
Date based context inclusion |
| 2:08AM |
1 |
sip, call ransfer and call waiting |
| 2:03AM |
1 |
IAX Registry problems |
| 12:55AM |
1 |
VOIP in Japan using Freebit |
| |
| Sunday September 25 2005 |
| Time | Replies | Subject |
|---|
| 11:22PM |
2 |
change codec based on callerid (sip/iax) |
| 9:55PM |
0 |
compute traffic intensity from CDR? |
| 9:48PM |
1 |
Can an outside caller dial an extension before someone answer? |
| 9:30PM |
3 |
TE405P V2 - Fantastic! |
| 6:54PM |
0 |
Emergency Asterisk Guru help needed -- Yucky sound with MOH |
| 6:06PM |
3 |
Vonage-type service |
| 5:38PM |
0 |
Unable to Transfer an outbound call |
| 5:31PM |
1 |
Digium T-1 and FXO cards for sale |
| 5:31PM |
0 |
Cisco phone ports |
| 12:48PM |
1 |
WRT54GP2 SIP server on LAN port |
| 11:04AM |
2 |
Pager Notification Script |
| 10:25AM |
0 |
CALLERID to Sipura Devices (or others for that matter).. CVS-Latest Version |
| 10:18AM |
0 |
VPB Driver Question |
| 8:19AM |
0 |
pound/hash key not recognized |
| 6:27AM |
0 |
Problem Asterisk: can't make call but can receive calls |
| 3:32AM |
2 |
iax problem |
| 12:48AM |
1 |
Codec routing? |
| |
| Saturday September 24 2005 |
| Time | Replies | Subject |
|---|
| 10:46PM |
2 |
Extension Mobility (roaming) Cisco 7960 |
| 10:08PM |
1 |
dialplan game |
| 10:06PM |
4 |
didgium card in india |
| 10:00PM |
0 |
IPSpeedDial has just been released |
| 8:22PM |
2 |
CDR problem |
| 7:23PM |
1 |
Cheap Time sources which is best? |
| 7:00PM |
0 |
Software to generate an SRTP key pair? |
| 6:45PM |
1 |
Need good explanation on contexts and extensions |
| 5:51PM |
3 |
IBM x306 |
| 5:49PM |
0 |
PA1688 Phones using IAX MWI |
| 4:55PM |
0 |
Pictures from VON Fall 2005 Digium/Asterisk booth |
| 3:29PM |
0 |
Falsh Panel in Xorcom Rapid |
| 3:28PM |
2 |
Directed pickup syntax? |
| 12:21PM |
2 |
Send DTMF after call bridge |
| 11:25AM |
1 |
ASTCC on Fedora 4 and MySQL 4.1.12 |
| 10:42AM |
1 |
unable to use misdn group dial |
| 9:29AM |
2 |
Asterisk returns 484 ADDRESS INCOMPLETE for incoming SIP calls |
| 9:19AM |
1 |
Help!! trying to use an MTA |
| 8:55AM |
0 |
BT100 can't register |
| 5:11AM |
0 |
HP DL360 G4 EM64T and hyperthreading options |
| 3:14AM |
0 |
Seperate siptrunks |
| 1:19AM |
1 |
wrong password on authentication for INVITE to '"asterisk" |
| 12:22AM |
0 |
Do Sifira use Asterisk? |
| |
| Friday September 23 2005 |
| Time | Replies | Subject |
|---|
| 10:22PM |
1 |
Message Waiting Indicator (MWI) for remote voice mail? |
| 9:53PM |
0 |
Is background() fax detect broken? |
| 9:18PM |
0 |
delay SIP answer |
| 8:18PM |
1 |
Skye gateway? |
| 6:14PM |
1 |
context question |
| 3:56PM |
1 |
Wildcard TE110P in Mexico |
| 3:31PM |
0 |
DTMF detection problems. |
| 2:27PM |
1 |
RE: [Asterisk-Dev] Open source time card application for Asterisk |
| 1:46PM |
0 |
X-Lit not picking up callgroup call with *8 |
| 1:44PM |
1 |
FW: channel offhook state |
| 1:38PM |
1 |
Play sound on connect |
| 1:14PM |
1 |
Asterisk CMD MySQL |
| 12:58PM |
1 |
Asterisk - Dying Signal 11 |
| 12:22PM |
3 |
Removing "-" (Dash) from Dialed Numbers |
| 12:15PM |
2 |
Can't receive Faxes with Asterisk (help) |
| 12:03PM |
0 |
Call Queue ANI |
| 11:38AM |
4 |
CallerID issue |
| 11:37AM |
2 |
asterisk invitation problem |
| 11:28AM |
2 |
Continue dialtone after pressing 9 |
| 11:12AM |
4 |
goiax expanded with free us domestic calling |
| 10:18AM |
0 |
voicetronix openline4 comments |
| 9:53AM |
1 |
ChanSpy performance sub-optimal |
| 9:17AM |
2 |
ZAP ISDN losing digits |
| 9:13AM |
0 |
Trunks greyed-out on Flash Operator Panel? |
| 9:10AM |
1 |
retry times |
| 8:58AM |
0 |
voicemail operation modification |
| 8:22AM |
0 |
Problem with outbound calls |
| 8:15AM |
2 |
Problems with queue and remote agents |
| 7:35AM |
2 |
Execute php agi after channel hangup |
| 7:26AM |
0 |
RE: SNOM 190 '486/Busy here' after upgrade to re 3.60s |
| 7:23AM |
0 |
DTMF translation |
| 6:04AM |
0 |
SIP Hangup via Call Files |
| 5:55AM |
1 |
ztdummy compile again |
| 5:44AM |
1 |
dial (iax/X&sip/y) get y fraction earlier |
| 5:12AM |
1 |
Double cpu |
| 5:03AM |
2 |
Dialtone problems with phpagi and asterisk |
| 4:27AM |
1 |
chan_capi-cm-0.6: hangup is detected really late |
| 2:54AM |
1 |
Dial multiple phones |
| 2:48AM |
10 |
Problem setting up TDM22B card |
| 2:19AM |
6 |
Which codec? |
| 2:14AM |
1 |
Dial() and BackGround() |
| 1:41AM |
1 |
zaphfc problem: overlapdial don't work after update bristuff |
| 1:39AM |
1 |
Fax detection question |
| 12:28AM |
0 |
Hangup when dial via Mobile Interface |
| |
| Thursday September 22 2005 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
Keytouch without effect |
| 11:05PM |
0 |
SNOM 190 '486/Busy here' after upgrade to firmware 3.60s |
| 10:59PM |
2 |
Asterisk + GNUGK + Asterisk-Addons ooh323 |
| 9:16PM |
0 |
CVS-HEAD and Caller ID -- Pulling my hair out! |
| 8:47PM |
1 |
SayUnixTime in CVS? |
| 7:12PM |
2 |
Recently reported ASTCC audio issues |
| 6:17PM |
1 |
anyone know about this company? www.blue-wireless.net |
| 4:21PM |
0 |
Extended SIP registration failures |
| 4:12PM |
0 |
SNOM 190 '486/Busy here' after upgrade to firmwa re 3.60s |
| 2:49PM |
0 |
priindication passthru TE410P EuroISDN? |
| 2:32PM |
0 |
problems with sending fax from SIP channels |
| 1:45PM |
0 |
rtp problems |
| 1:33PM |
2 |
Set Log Level for Messages log file |
| 12:15PM |
1 |
Will Digium Wildard work with PCI-X or PCI Express |
| 12:05PM |
0 |
logging in problem |
| 11:15AM |
4 |
Polycom IP500 Quickstart page or files? |
| 10:49AM |
1 |
WaitExten |
| 10:39AM |
0 |
OT: Sangoma A102u available |
| 10:25AM |
1 |
externpass |
| 10:15AM |
0 |
cdr_custom? |
| 10:10AM |
12 |
custom ring tone |
| 10:05AM |
0 |
AGI Script to interact with ACCESS Databse a nd Set CID info on the fly. |
| 9:52AM |
3 |
AGI Script to interact with ACCESS Databse and Set CID info on the fly. |
| 9:10AM |
0 |
Hardware Recommendations for Junghanns card QuadBRI PCI. |
| 8:00AM |
0 |
Multiple SIP Phone Calls Overlapping on the Same Phone |
| 7:31AM |
0 |
ASTCC error when using silent=5 |
| 6:40AM |
1 |
Initial release of AMPortal Debian/Xorcom-Rapid packages |
| 6:34AM |
0 |
Call Pickup issue |
| 6:18AM |
1 |
Asterisk with iptel.org |
| 5:52AM |
1 |
AgentRecord In and Out streams |
| 5:18AM |
2 |
SOHO Survey / Creative Asterisk Solutions |
| 4:44AM |
1 |
IAX client for Linux text console |
| 3:26AM |
1 |
Early Media with Asterisk |
| 12:29AM |
1 |
Compile problems on Solaris SPARC |
| 12:26AM |
1 |
Any problems with Asterisk and "nice" |
| |
| Wednesday September 21 2005 |
| Time | Replies | Subject |
|---|
| 11:48PM |
2 |
Submitting ISDN-MSN from a SIP-Phone |
| 9:44PM |
2 |
Web based application for call History |
| 6:53PM |
0 |
new spandsp-0.0.3pre1 missing tx and rx fax apps? |
| 6:36PM |
1 |
I got "403", "Forbidden"... please help |
| 6:28PM |
3 |
Cisco AS5XXX + CallerID Name |
| 6:13PM |
2 |
ftp.soft-switch.org down? |
| 5:22PM |
5 |
Tux/Asterisk logo for Cisco phones |
| 4:28PM |
0 |
Soyo Phones Crashing |
| 2:56PM |
4 |
WMI problem |
| 2:24PM |
0 |
Asterisk Platform - Success Strories - iAreanet in the news. |
| 1:34PM |
4 |
POP3 and TTS (Festival?) |
| 1:10PM |
0 |
Callprogress and TDM400 in Brasil |
| 1:04PM |
1 |
Problem with meetme monitor (recording) |
| 12:58PM |
1 |
Asterisk and a SPA3000 behind NAT peer registration |
| 12:51PM |
0 |
Problem with monitor application meetme |
| 12:33PM |
0 |
IAX2 vs SIP Phones and adapters |
| 12:31PM |
1 |
Problems with sipura 1001's and 2002's |
| 12:25PM |
0 |
problem with monitor meetme |
| 12:21PM |
0 |
re: Problems with Queues |
| 12:10PM |
2 |
Get SIP to work over very limited network access |
| 11:46AM |
1 |
oh323 driver and RFC2833 |
| 10:52AM |
3 |
How can i call to a cellphone here in Mexico? |
| 10:11AM |
1 |
Weird Over Lapping Asterisk Calls via SIP Phones |
| 9:21AM |
0 |
is possible connect? |
| 9:13AM |
0 |
ODBC Voicemail WEB Retrieval V1.1 |
| 8:39AM |
0 |
HOWTO: A simple AGI application to modify incomi ng CallerID on the fly using SQL Server and *not* UnixODBC |
| 8:38AM |
1 |
Addendum to Problem with Queues question |
| 8:35AM |
2 |
Problem with Queues |
| 8:27AM |
2 |
Macro exists if an application returned -1 |
| 8:24AM |
0 |
qualify=yes |
| 8:07AM |
3 |
Caller ID and Call Parking on an analog PSTN line? |
| 8:03AM |
1 |
Does Asterisk know if the trunks are busy? |
| 7:56AM |
2 |
ISDN-forwarding to intern without cost? |
| 7:51AM |
0 |
Cellphones and Asterisk Bluetooth |
| 7:50AM |
0 |
HELP: E1 ChannelBank and UniCall |
| 7:28AM |
2 |
maximum concurrent ZAP channels .... max conf ports ... |
| 7:26AM |
1 |
Ask for config files of Nortell Meridian Op 11 & Asterisk for PRI |
| 7:24AM |
0 |
Using *0 to flash an external trunk on bridged channel |
| 7:17AM |
0 |
permit syntax question |
| 7:17AM |
0 |
Packetization period for CODECs |
| 7:10AM |
7 |
add 0 (zero) to incoming callerID - how? |
| 5:12AM |
0 |
First release of the Asteriskguru Operator Panel |
| 4:35AM |
0 |
IAX2 registration |
| 2:54AM |
4 |
How to retrieve voicemail from an IP phone? |
| 2:23AM |
0 |
Intermitant delays on call setup. |
| 1:22AM |
1 |
Call getting disconnected in queue |
| 1:14AM |
0 |
DID problem with calls from analog to ISDN |
| 12:56AM |
0 |
Brand New IPSwitchBoard |
| |
| Tuesday September 20 2005 |
| Time | Replies | Subject |
|---|
| 11:45PM |
1 |
Asterisk PBX |
| 11:16PM |
6 |
iax2 trunking wackyness |
| 11:03PM |
0 |
Phone lines |
| 10:24PM |
1 |
automon wav format problems |
| 10:01PM |
0 |
Anyone using Asterisk to take credit card payments? |
| 9:03PM |
0 |
Can I connect an IAXy to my Panasonic PBX? |
| 8:41PM |
0 |
DIDx |
| 7:45PM |
1 |
HooDaHek w/AST 1.2 |
| 7:45PM |
0 |
ODBC VM Playback from Web Page |
| 4:57PM |
0 |
Handling SIP 404 event |
| 4:56PM |
1 |
cvs-head and unicall with r2mfc |
| 4:46PM |
3 |
sipuras 841 bad sound |
| 3:38PM |
1 |
MOH failures (bad quality with interruptions) |
| 3:37PM |
4 |
SUCCESS - 512 Simultaneous Calls with Digital Recording |
| 3:16PM |
0 |
TE110P hybrid configuration for data and voice |
| 2:33PM |
0 |
fixlocalprefix error |
| 1:16PM |
1 |
[Fwd: ASTCC speaks and cut RTP channel, => Kind of solution... |
| 1:06PM |
5 |
MySQL and Asterisk |
| 1:03PM |
1 |
Asterisk vertical service activation codes |
| 12:52PM |
2 |
Snom-320 badly garbled audio |
| 12:09PM |
3 |
[ANNOUNCE] chan_capi-cm-0.6 released |
| 10:54AM |
0 |
agent channel busy - how to stop it? |
| 10:37AM |
1 |
ODBC Voicemail WEB Retrieval |
| 10:30AM |
4 |
how to distinguish the "ringing" and "connected" for zap channel |
| 10:03AM |
0 |
BackgroundDetect problem |
| 8:50AM |
9 |
HooDaHek 0.6 Released |
| 8:34AM |
1 |
one way voice |
| 8:07AM |
0 |
Aterisk App ICES Question |
| 7:02AM |
0 |
using a voip cable modem |
| 6:58AM |
0 |
Asterisk@Home Music on Hold |
| 6:35AM |
0 |
Red or Yellow alarm monitoring |
| 5:48AM |
0 |
What hardware would you recommend? |
| 5:34AM |
0 |
General Config information |
| 5:30AM |
0 |
asterisk-oh323: New versions 0.6.7 and 0.7.3 |
| 4:35AM |
0 |
Connect not signalled (SIP -> Zap) |
| 4:21AM |
0 |
HELP: Valiant E1 CB and UniCall |
| 4:05AM |
1 |
Is there a clever way to page a group of extensions? |
| 4:03AM |
0 |
sipp examples |
| 3:56AM |
0 |
PTN calls into asterisk slow release |
| 3:23AM |
0 |
Hangup after voicemail not detected |
| 2:15AM |
1 |
Cisco 7960 Locking Up |
| |
| Monday September 19 2005 |
| Time | Replies | Subject |
|---|
| 11:48PM |
1 |
Resolving QOS problems |
| 10:27PM |
1 |
"Stopping retransmission on" messages |
| 10:24PM |
0 |
TE410 stop responding |
| 8:24PM |
1 |
Buy a digium hardware |
| 7:43PM |
2 |
MWI indicator HINT on Snom thru IAX? |
| 6:37PM |
1 |
need example about sjphone with asterisk |
| 5:34PM |
0 |
Call dropped 100% of time when incoming IAX routed to outgoing CAPI |
| 5:10PM |
1 |
Re: Welcome to the "Asterisk-Users" mailing list |
| 4:37PM |
1 |
Zap calls dropping just after answer |
| 4:34PM |
0 |
Voicemail() application returning -1 on a hangup |
| 3:52PM |
0 |
MSNs don't work for me... :( |
| 3:48PM |
1 |
hfc card unplug & plug not working? |
| 3:07PM |
0 |
pridialplan per call or per channel group? |
| 2:38PM |
1 |
[Fwd: ASTCC speaks and cut RTP channel => Kind of solution... |
| 2:37PM |
0 |
H.263 Format video |
| 1:57PM |
1 |
Point to Point with Fritz Card ... |
| 1:53PM |
0 |
Dial time limit doesn't work when calling party transfers |
| 1:24PM |
1 |
Silence suppression /RTP VAD and Asterisk? Dropped calls on IP-500 |
| 1:21PM |
3 |
T.38 & Canreinvite (yes, again) |
| 12:53PM |
1 |
Asterisk Keep Crashing need Help please |
| 12:41PM |
1 |
Most desireable Linux distribution for Aster isk? |
| 11:58AM |
1 |
Asterisk monitoring availability |
| 11:44AM |
4 |
IAX dialplan problem? |
| 11:43AM |
2 |
kill a .call file |
| 11:42AM |
2 |
Looking for firmware for Cisco 12sp+ and 30VIP |
| 11:05AM |
1 |
Most desireable Linux distribution for Asterisk? |
| 11:05AM |
0 |
HooDaHek Version 0.5 Release |
| 10:53AM |
2 |
ztdummy configuration help |
| 10:09AM |
4 |
Pinging ... |
| 9:57AM |
0 |
Sip and ISDN problem |
| 9:49AM |
1 |
Complete NPA-NXX list for USA/Canada npanxx, |
| 9:48AM |
2 |
hints and the sNOM 360 |
| 9:38AM |
4 |
VM low volume - testers needed |
| 9:30AM |
1 |
i4l ring indication problem, again... |
| 9:22AM |
6 |
SIP audio port usage |
| 9:01AM |
0 |
Anyone have the firmware for WRT54GP2? |
| 8:59AM |
0 |
hints not working on CVS HEAD |
| 8:45AM |
3 |
OT: Hardware Interrupts; Who is it? |
| 8:45AM |
0 |
Asterisk ISDN: Problem Setting CallerID as DIDExtension Numbers. |
| 8:41AM |
1 |
OT: Xoops Skype module |
| 8:27AM |
0 |
Round-robin with Queue |
| 8:11AM |
0 |
sip invite question |
| 8:10AM |
0 |
Unable to open space (format ulaw)? |
| 8:07AM |
0 |
chan_alsa.c blocking sound port - solution |
| 7:54AM |
0 |
FW: ADTRAN Virtual Classes: Ensuring QoS for VoIP & Total Access 900 Series |
| 7:05AM |
1 |
problems with PRI |
| 6:13AM |
0 |
clear SIP channel |
| 5:53AM |
1 |
Prompt translation: can't find "please wait try ext" prompt filename |
| 4:49AM |
0 |
need a simply configuration for calling in/out to PSTN |
| 3:43AM |
1 |
Voipbuster in Australia -- delay problem |
| 3:01AM |
0 |
ISDN BRI 2 pci cards and mISDN |
| 12:53AM |
0 |
problems with remote access to PSTN |
| |
| Sunday September 18 2005 |
| Time | Replies | Subject |
|---|
| 9:28PM |
7 |
Cisco Callmanager & Asterisk for Voicemail revisited |
| 9:27PM |
2 |
HW Question (TDM400) |
| 9:12PM |
5 |
Monitor and sox mix quality |
| 4:43PM |
1 |
sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why? |
| 4:00PM |
0 |
Julien COURTEMANCHE/TELINTRANS/FR est absent(e). |
| 3:39PM |
6 |
Differ between "private" and "out of area"? |
| 3:27PM |
0 |
ChanSpy not loading |
| 12:12PM |
1 |
Re: Asterisk-Users Digest, Vol 14, Issue 108 |
| 12:09PM |
2 |
limiting calls per day based on amount of time |
| 10:24AM |
1 |
Two POTS in, but only want one out? |
| 8:52AM |
2 |
Asterisk Won't Process Call |
| 8:03AM |
0 |
voicemail context. macro, and directory |
| 7:26AM |
0 |
(no subject) |
| 7:15AM |
1 |
TFTP and DHCP... |
| 4:24AM |
1 |
DID from an analog phone |
| |
| Saturday September 17 2005 |
| Time | Replies | Subject |
|---|
| 9:52PM |
2 |
Complete NPA-NXX list for USA/Canada npanxx, ratecenters, etc (attached) |
| 7:17PM |
1 |
Who is going to AstriCon (TheAsteriskConference)? |
| 6:38PM |
1 |
How does one set-up incoming/outgoing SIP with no registration and only IP authentication? |
| 5:33PM |
0 |
bounty partners and/or possible coder? queues.conf ackcall and pre-ack announce |
| 5:17PM |
1 |
unlocking cisco 7940 phone |
| 2:31PM |
2 |
checking voice mail from different phone |
| 2:26PM |
0 |
Anybody using SIP Interaction Proxy 2.X and Asterisk CVS head? |
| 1:09PM |
2 |
moh - turn off randomization? |
| 12:17PM |
1 |
capiFax causes segfault on asterisk |
| 11:30AM |
0 |
(no subject) |
| 11:05AM |
22 |
AstriCon 2006 Location |
| 6:07AM |
1 |
Flash Operator Panel Help |
| 5:41AM |
2 |
MGCP service from Free Télécom |
| 2:31AM |
2 |
AgentCallbackLogin and calling outside |
| |
| Friday September 16 2005 |
| Time | Replies | Subject |
|---|
| 9:52PM |
1 |
How to make Basic authenticatuion in Asterisk server. |
| 6:13PM |
11 |
wav instead of gsm for vm-sounds? |
| 6:03PM |
8 |
Who is going to AstriCon (The Asterisk Conference)? |
| 5:51PM |
0 |
free IAX calling platform |
| 5:25PM |
15 |
Double Ring |
| 4:45PM |
1 |
Grandstream |
| 4:23PM |
1 |
TDM400P Dialing Out - "Cannot be completed as dialed" |
| 3:06PM |
2 |
Orinoco Injectors |
| 3:01PM |
0 |
linux sip or iax phone that will autoanswer and route to console |
| 1:57PM |
0 |
Anyone using iPlan Networks in Argentina? |
| 12:51PM |
5 |
How to access * thru router when ip address is not known |
| 12:38PM |
0 |
asterisk mixing sound card with anybody? |
| 12:10PM |
0 |
Weird behaviour |
| 11:27AM |
0 |
lastest spandsp-0.03pre1 don't compile |
| 10:23AM |
0 |
Zap failed |
| 10:12AM |
1 |
Sipura 2k voice quality |
| 9:48AM |
1 |
Easier way for end user to change main greeting? |
| 9:14AM |
2 |
R: direct sip call pickup |
| 9:08AM |
1 |
New version of idefisk softphone released. |
| 8:56AM |
7 |
mpg123 on x86_64 (Opteron MP) |
| 8:19AM |
1 |
direct sip call pickup |
| 8:07AM |
0 |
alsa issue with asound.conf |
| 8:06AM |
4 |
queue_log on mysql |
| 7:43AM |
0 |
SIP port assignment for user agents registering to Asterisk. |
| 7:23AM |
4 |
Caller Name: Asterisk reading too fast |
| 7:09AM |
0 |
Extension Restrictions |
| 7:04AM |
0 |
broadvoice incoming caller ID is wierd when calling from voipjet |
| 5:45AM |
1 |
7 digit dialing to e.164 format |
| 2:15AM |
0 |
How to suppress Local/Zombie channels? |
| 2:00AM |
2 |
Call Forward - 7940 Asterisk - Help |
| 12:56AM |
0 |
Unable to create ZAP channel - All circuits are busy |
| 12:52AM |
0 |
Wildcard TE110P |
| 12:29AM |
0 |
auto restart |
| |
| Thursday September 15 2005 |
| Time | Replies | Subject |
|---|
| 11:27PM |
0 |
Transfering from a device to a queue crashesAsterisk |
| 10:26PM |
2 |
Help on RealTime Extensions on Oracle DB |
| 9:25PM |
0 |
Changing the sip port in sip.conf does not work |
| 9:25PM |
0 |
Send SIP NOTIFY frequency |
| 9:14PM |
0 |
QUESTION: RINGING CONTINUES DURING CALL |
| 8:44PM |
0 |
Sip recording |
| 7:58PM |
2 |
Is digium supporting new te405p and te406p install? |
| 7:15PM |
0 |
triggering automatic dial-outs with Zap interface |
| 7:01PM |
1 |
Asterisk and Zyxel Prestige 2000W_v2 |
| 6:42PM |
2 |
SIP reinvite asterisk and NAT |
| 6:04PM |
1 |
ZyXEL P662HW / SIP / Crashing |
| 5:41PM |
3 |
USB Phones for use with Asterisk |
| 3:26PM |
2 |
Asterisk CDRs |
| 2:13PM |
0 |
Console/dsp and mplayer |
| 2:02PM |
2 |
Caller ID for auto outgoing calls |
| 1:29PM |
1 |
Faxibility in NZ |
| 1:17PM |
2 |
Still having hangup problems in NZ |
| 12:38PM |
1 |
Unable to call some numbers with I4L |
| 12:21PM |
0 |
dialing sip before answering pstn line |
| 12:07PM |
0 |
If call fails, then try again with something else |
| 12:03PM |
0 |
Call Pickup between ZAP and SIP technologies |
| 11:21AM |
3 |
internet connection between Africa and Europe |
| 10:42AM |
1 |
Can not get realtime static voicemail.conf to work |
| 10:28AM |
0 |
Polycom oddities: Mixed up digits -> *8 Call Pickup |
| 9:04AM |
3 |
Seperate Incoming calls on TDM02? |
| 8:57AM |
0 |
Comfort Noise Generation with Zap-IAX |
| 8:50AM |
0 |
Siemens Hi-Path help |
| 8:28AM |
0 |
Transfering from a device to a queue crashes Asterisk |
| 8:19AM |
1 |
Getting email of voicemail to work |
| 8:09AM |
2 |
Fax->Email for Hosted PBX |
| 7:53AM |
0 |
Configuring GR303 trunks from Asterisk to a Taqua/TEKELEC T7000 |
| 7:48AM |
1 |
Don't install asterisk-chan-capi |
| 7:24AM |
1 |
USB ISDN (OT question) |
| 7:24AM |
2 |
Asterisk CDR information into Oracle DB |
| 7:10AM |
5 |
Asterisk don't start |
| 7:07AM |
0 |
dialplan to try VOIP providers if they can't terminate call |
| 6:24AM |
2 |
cdr server |
| 5:58AM |
0 |
linux kernel tweaking for Asterisk |
| 5:29AM |
3 |
MusicOnHold not working |
| 5:27AM |
0 |
TE110P - Asterisk@Home Install Problems - Televantage 3 T1 |
| 5:03AM |
0 |
Looking for China DID |
| 4:18AM |
0 |
Why isn't 3-way calling a standard feature? |
| 4:09AM |
0 |
No sounds on Playback() |
| 4:07AM |
0 |
TxFAX don't work |
| 3:24AM |
0 |
SIP rogue channel |
| 3:08AM |
1 |
iax phone and asterisk server on different LANs |
| 3:06AM |
4 |
PSTN calls are quiet |
| 2:59AM |
0 |
Incoming / Outgoing call problems on TDM card. |
| 2:33AM |
0 |
AW: ***SPAM*** actionID on manager events |
| 2:28AM |
3 |
${DIALSTATUS} problems |
| 1:46AM |
1 |
Originate not understanding 2 vars in setvars |
| 1:12AM |
0 |
SV: RxFax problems |
| |
| Wednesday September 14 2005 |
| Time | Replies | Subject |
|---|
| 10:27PM |
0 |
compile problems with yada |
| 8:54PM |
2 |
Starting From Scratch |
| 7:52PM |
0 |
Cannot hear teleco side error message |
| 7:04PM |
1 |
Liquidation: Cisco; Polycom; D-Link; MediaTrix, Colubris - Highly Reduced Prices |
| 6:16PM |
1 |
Distinctive Ring Tones |
| 4:45PM |
0 |
How to uninstall |
| 4:25PM |
0 |
Weird SIP behavior or I need a shrink? |
| 3:22PM |
0 |
Interop with Cisco T1/PRI on the 2811 and PSTN |
| 2:48PM |
1 |
Routes IPSEc And Asterisk. |
| 2:03PM |
1 |
RE: Asterisk-Users Digest, Vol 14, Issue 86 |
| 1:13PM |
0 |
Compile error on cdr_yada for asterisk on centos with Oracle |
| 12:58PM |
0 |
# dialplan not working... |
| 12:15PM |
0 |
${VM_CIDNUM} shows up but ${VM_CALLERID} & ${VM_CIDNAME} don't? |
| 11:42AM |
4 |
Echo on SPA-3000 FXO |
| 11:38AM |
1 |
ASTCC issues |
| 11:25AM |
0 |
sox conversion has introduces background hiss for both 8k and 41K recordings to gsm |
| 11:10AM |
11 |
RxFax/TxFax - Compile Problem |
| 11:01AM |
1 |
Asterisk Consulting Project ISO Hired Gun |
| 10:57AM |
1 |
Indications for Ireland |
| 10:35AM |
1 |
Re: Polycom randomly fails outbound calls, |
| 9:56AM |
0 |
RES: How to create IVR menu and transfer to anothersip extensions. |
| 9:50AM |
0 |
Anyone knows how to receive a SIP call withoutregistering gateway? |
| 9:46AM |
0 |
RxFax problems. |
| 9:34AM |
7 |
Asterisk 1.0.9 long term stability <--thread hijack, why not reboot? |
| 8:45AM |
1 |
TE110P - Asterisk@Home Install Problems |
| 8:36AM |
1 |
Asterisk as a gateway. 'flash for transfers transparency?' |
| 8:10AM |
1 |
IAX Registration with servers |
| 8:09AM |
3 |
Asterisk 1.0.9 long term stability |
| 7:53AM |
1 |
SMS using a PRI channel |
| 7:46AM |
3 |
(no subject) |
| 7:24AM |
1 |
timeout with queue |
| 7:22AM |
0 |
MAX PRI for single server (was:Not enoughlinesavailable for Asterisk implemetation) |
| 5:54AM |
0 |
Dial Application Return Codes - Help needed |
| 5:51AM |
2 |
STUN vs NAT Helper |
| 5:40AM |
2 |
PRI to PRI passthrough with DID intact |
| 3:58AM |
6 |
T.38 ATA |
| 1:47AM |
1 |
call restrictions |
| 1:47AM |
0 |
oh323 and Asterisk: Calls always hang up |
| 1:05AM |
2 |
pri release cause code mismatch |
| |
| Tuesday September 13 2005 |
| Time | Replies | Subject |
|---|
| 11:20PM |
1 |
Anyone knows how to receive a SIP call without registering gateway? |
| 10:33PM |
0 |
spandsp frame slip tolerance. |
| 10:15PM |
0 |
Zap Clocking - Frame Slips - tdm400p wcfxozttest cpu spikes spandsp |
| 10:01PM |
1 |
Limiting call minutes on a GSM SIM |
| 9:50PM |
1 |
slight echo via sip provider |
| 8:27PM |
0 |
PRI zap channels not cleared when nomatchincontext for dialed number on inbound call |
| 8:25PM |
1 |
PRI zap channels not cleared whennomatchincontext for dialed number on inbound call |
| 8:11PM |
0 |
PRI zap channels not cleared when no matchincontext for dialed number on inbound call |
| 8:07PM |
0 |
PRI zap channels not cleared when no match incontext for dialed number on inbound call |
| 7:56PM |
0 |
PRI zap channels not cleared when no match in context for dialed number on inbound call |
| 7:49PM |
3 |
Call Wrapup time for agents. |
| 7:28PM |
2 |
Digium Cards in Australia |
| 7:26PM |
1 |
wctdm, issue w/outbound calls |
| 7:09PM |
1 |
Asterisk@home with Eyebeam |
| 6:21PM |
1 |
populating asterisk realtime tables from configfiles |
| 5:39PM |
0 |
populating asterisk realtime tables from config files |
| 4:37PM |
0 |
CVS vs CVS-HEAD |
| 3:45PM |
0 |
TDM400P stops answering |
| 3:37PM |
1 |
callfile: How to invoke SetCallerPres ? |
| 3:35PM |
1 |
make * listen on a specific ethernet interface |
| 3:17PM |
0 |
callfiles: how to query current dial attempt nr in extensions.conf? |
| 2:52PM |
1 |
How to IGNORE distinctive ring |
| 2:43PM |
1 |
Cisco AS5400 Configuration as a SIP Peer - URGENT |
| 2:37PM |
0 |
First PRI Installed - WOOT |
| 2:22PM |
0 |
MTA V102 |
| 2:09PM |
1 |
sometimes dtmf passed, sometimes not (cisco 7960 SIP) |
| 1:49PM |
1 |
Oh323 and Asterisk with MERA |
| 1:44PM |
0 |
Integration Nortel x Asterisk |
| 1:35PM |
1 |
Dialplan Design Q |
| 1:15PM |
1 |
disable chan_skinny and chan_oss |
| 12:37PM |
1 |
Not able to access asterisk from internet via ip-forwarding |
| 12:24PM |
0 |
ZoomTel x5v Model 5565: is it any good? |
| 12:21PM |
4 |
Fedora Core 4 not recognizing X100P cards |
| 12:20PM |
0 |
AMP created extensions busy when dialed. |
| 12:03PM |
1 |
Polycom IP500 Mass Configurations |
| 11:33AM |
0 |
Asterisk + NEC IPK 192 integration |
| 11:32AM |
1 |
translate letters into digits |
| 11:22AM |
1 |
asterisk hangup detection on a pbx analog port] |
| 11:12AM |
0 |
Can anyone explain why this is happening? Odd CUT Problem |
| 10:47AM |
5 |
How to create IVR menu and transfer to another sip extensions. |
| 10:08AM |
1 |
FW: Nat & Sip & Pain |
| 9:26AM |
1 |
TDMoE Configuration problems |
| 9:17AM |
2 |
actionID on manager events |
| 9:13AM |
1 |
problem with FXS module |
| 9:01AM |
0 |
asterisk callerid problems |
| 8:31AM |
0 |
Bristuff version for use with 1.2.0beta1 |
| 8:09AM |
0 |
Real-time Linux claims single-digit microsecond responsiveness |
| 8:05AM |
2 |
passing variables to h extension |
| 7:46AM |
1 |
SetCIDName question |
| 7:24AM |
0 |
Micro-cuts in MusicOnHold |
| 7:01AM |
0 |
[Re: civil emergency comms: Asterisk + HAM]] |
| 7:00AM |
0 |
PLEASE HELP!! CALLERID FAILS!! |
| 6:14AM |
1 |
Integration between Asterisk and Siemens HiCom 150e over ISDN |
| 5:47AM |
0 |
show queue callcenter output? |
| 4:50AM |
1 |
Monitoring status of ISDN lines |
| 4:44AM |
2 |
Nat & Sip & Pain |
| 1:55AM |
0 |
Coexistence of zaphfc and hisax? |
| 1:00AM |
1 |
2 box single Asterisk |
| |
| Monday September 12 2005 |
| Time | Replies | Subject |
|---|
| 10:18PM |
0 |
get dialstatus variable when returning No such context/extension |
| 10:18PM |
1 |
Phonecall or something as robust |
| 8:31PM |
1 |
Is "ChanIsAvail" thread safe? |
| 7:22PM |
0 |
High system load and system freezes |
| 6:23PM |
0 |
Subject: '#' dialplan pattern matching |
| 5:10PM |
2 |
Firmware upgrade Aastra 480i CT |
| 5:08PM |
0 |
Whisper Mode |
| 5:03PM |
3 |
monitor peak channel use |
| 4:47PM |
0 |
early dial (grandstream bt100) |
| 3:10PM |
1 |
Meetme Dial Out |
| 2:21PM |
2 |
Stupid tricks: preventable? |
| 1:59PM |
5 |
What have I misconfigured? |
| 1:53PM |
5 |
OT: Online TTS engines? |
| 1:49PM |
1 |
compile error with postgres and voicemail |
| 1:39PM |
1 |
LiveVOIP - I win :) |
| 1:33PM |
4 |
CallerID Name in dialplan |
| 1:17PM |
0 |
WaitExten? |
| 11:55AM |
0 |
New York Asterisk User Group - Established |
| 10:08AM |
13 |
Skype purchased by Ebay 2.6 Billion |
| 9:44AM |
1 |
optimizing for via C3 |
| 9:26AM |
1 |
AW: Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card |
| 8:58AM |
1 |
callfiles: set variables ? |
| 8:42AM |
0 |
Re: Asterisk-Users Digest, Vol 14, Issue 70 |
| 8:34AM |
0 |
[Fwd: SwissVoice IP10S not able to dial calls with protocol SIP] |
| 8:32AM |
2 |
Callerid fails in any release after beta1 fails |
| 8:17AM |
0 |
Canspy listening to SIP channels |
| 8:11AM |
1 |
Other Voicemail systems |
| 7:57AM |
1 |
Callerid UK patches (from Lusyn) |
| 6:38AM |
1 |
Can't pickup inbound calls with TDM400P Fxo |
| 6:19AM |
0 |
Voicemail Not Recognizing user and password? |
| 6:12AM |
1 |
wctdm module won't load after kernel upgrade |
| 5:41AM |
1 |
Montreal asterisk usergroup meeting today 6pm |
| 4:53AM |
4 |
Hotel Setup? |
| 4:46AM |
3 |
Asterisk Registration as Client to OpenSER |
| 4:07AM |
0 |
Configure asterisk to dial user and notify if new voicemail |
| 3:25AM |
2 |
Zap Channel |
| 2:22AM |
1 |
chan_zap.c:8050 pri_dchannel: Ring requested on unconfigured channel 255/255 span 2 |
| 2:05AM |
0 |
asteriskathome and cisco 2600 |
| 1:54AM |
0 |
ChanSpy with asterisk 1.0.9 |
| 1:32AM |
0 |
tdm400p wattage |
| 12:50AM |
0 |
Sip phone will not connect |
| 12:34AM |
1 |
How to remove the voice mail greeting... |
| 12:12AM |
2 |
Hang up not hanging up (New Zealand Indications??) |
| |
| Sunday September 11 2005 |
| Time | Replies | Subject |
|---|
| 9:50PM |
2 |
Asterisk and AMP installed now what? |
| 9:16PM |
1 |
Anyone using Telasip, Caller ID presentation outbound?? |
| 8:39PM |
0 |
extensions.conf for VOXEE using SIP!! |
| 8:15PM |
1 |
Syslog file size |
| 7:58PM |
4 |
Asterisk on AMD64 |
| 4:56PM |
1 |
first character in line 11 missing |
| 3:57PM |
1 |
Presence Fully Supported? |
| 3:04PM |
0 |
Call Waiting Tracking? |
| 1:40PM |
0 |
H323 with asterisk-ooh323c |
| 1:06PM |
3 |
David Choo/eServices/eSpore is overseas |
| 12:39PM |
0 |
Australian Dial tone TDM400P |
| 11:11AM |
2 |
cdr_addon_mysql.so pb |
| 11:08AM |
1 |
ruby-agi 0.0.2 released |
| 9:26AM |
5 |
rotate * log file? |
| 6:16AM |
2 |
Using RedirectAction with queues |
| 4:05AM |
0 |
Ignore incomingcall? |
| 4:02AM |
2 |
Make asterisk call out |
| 3:11AM |
0 |
OpenH323-Channel Q.931-Problems with Gatekeeper |
| 2:48AM |
5 |
TE406p no interrupts |
| 1:56AM |
1 |
Integrating with existing analog PBX |
| 12:46AM |
6 |
SIP Connection Problems |
| |
| Saturday September 10 2005 |
| Time | Replies | Subject |
|---|
| 5:41PM |
2 |
Echo Issue |
| 3:13PM |
1 |
TE110P reset |
| 2:14PM |
1 |
Configuring SIPURA 2002 to work wih Asterisk |
| 11:43AM |
0 |
Broadcasting via Asterisk |
| 9:58AM |
1 |
False Zap answer problem (Again) |
| 9:56AM |
1 |
AGI problem with library path |
| 9:11AM |
0 |
Distinctive Ring Problems |
| 8:59AM |
0 |
Problems with TE205P |
| 7:05AM |
0 |
call tests |
| 5:33AM |
2 |
AGI programming work required |
| 5:32AM |
1 |
Required hardware |
| 4:57AM |
2 |
VoipBuster again |
| 4:18AM |
4 |
Fritz, mISDN, Help |
| 3:35AM |
0 |
Need some HFC-S help |
| 12:51AM |
1 |
PRI echo |
| 12:46AM |
2 |
GotoIf Syntax to match first digits |
| |
| Friday September 9 2005 |
| Time | Replies | Subject |
|---|
| 8:08PM |
2 |
call volume |
| 6:27PM |
0 |
Queue "abandon" count increments incorrectly? |
| 5:19PM |
0 |
Transferred calls dropping out of MeetMe |
| 4:12PM |
1 |
vm notif |
| 3:33PM |
0 |
Announcement: FOP 0.23 released |
| 3:21PM |
1 |
Wait for dialtone |
| 3:02PM |
0 |
did edmonton |
| 2:58PM |
1 |
ALERT_INFO |
| 2:51PM |
0 |
Asterisk Extension Language |
| 2:35PM |
1 |
Special handling of IAX circuit-busy vs busy |
| 2:09PM |
1 |
ASTCC speaks and cut RTP channel |
| 1:58PM |
0 |
RTP ports in use grows and grows... |
| 1:42PM |
2 |
AMP 1.10.009 released! |
| 1:39PM |
1 |
Polycom 501 Multiple Line Instances |
| 1:14PM |
0 |
realtime and presence |
| 11:51AM |
1 |
Setting Account Code? |
| 11:48AM |
1 |
musiconhold errors in 1.2.0-beta1 |
| 11:26AM |
9 |
adding DNIS digits |
| 10:56AM |
2 |
FW: Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist" |
| 10:31AM |
1 |
RE:NewCUT() |
| 10:01AM |
0 |
Asterisk connected to Concept XI520 |
| 9:53AM |
0 |
woomera doesn't work (same OpenH323 problem as with chan_h323) |
| 8:44AM |
1 |
Changing User-Agent: Asterisk PBX |
| 8:43AM |
2 |
Storing extension prefs. in MySQL |
| 8:21AM |
1 |
OH323 for HEAD? 0.7.1 doesn't compile. |
| 8:08AM |
0 |
Detecting retries in call files |
| 7:55AM |
2 |
"Registered SIP '202' ... expires 1800". Why does it expire |
| 7:34AM |
1 |
Motherboard and processor recommendations |
| 7:15AM |
0 |
VIP-050 |
| 7:09AM |
1 |
siemens pbx what i ask techinician? |
| 6:16AM |
1 |
New CUT() |
| 5:08AM |
0 |
Doesn't finishes callerid spill |
| 5:07AM |
1 |
spandsp txfax multi page problem |
| 4:24AM |
0 |
remote SIP phones |
| 4:17AM |
0 |
BRI debug, national ISDN speech call problem |
| 4:14AM |
4 |
Huge Echo |
| 4:06AM |
0 |
Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card |
| 2:53AM |
0 |
OT Humo[u]r IVR Menu sample |
| 1:13AM |
0 |
the number of incoming calls in queue |
| |
| Thursday September 8 2005 |
| Time | Replies | Subject |
|---|
| 11:10PM |
0 |
Using E1 without power off simence pbx |
| 10:29PM |
1 |
can not make call with Unicall (MFC/R2) |
| 9:46PM |
0 |
T1 DSP Card to T1 - TXFAX RXFAX Posible Solved |
| 9:30PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday 9/10/2005 |
| 9:25PM |
2 |
T400P vs TE405P |
| 8:49PM |
1 |
Montreal usergroup |
| 8:27PM |
0 |
Question about setup Grandstream HandyTone 488 SIP with Astersik to Travel throught NAT. |
| 7:27PM |
4 |
Solution for 12 to 16 FXO to asterisk connection |
| 7:18PM |
0 |
PRI and Caller ID when immediate=yes |
| 7:11PM |
2 |
TDM PCI Master abort |
| 5:05PM |
2 |
sip log messages every few seconds |
| 4:29PM |
1 |
IAX Trunking Weirdness |
| 4:17PM |
2 |
How do you change the festival voice |
| 4:16PM |
1 |
FW: Adtran TA 616 |
| 3:45PM |
0 |
Announcement: ASTPP-1.2-Beta |
| 2:47PM |
2 |
TE411P zapata.conf, monitoring echo cancellation and echo tail size |
| 2:40PM |
1 |
MAX PRI for single server (was: Not enoughlinesavailable for Asterisk implemetation) |
| 2:25PM |
1 |
SIP/2.0 487 Request Terminated problem on Cisco 7960 |
| 2:18PM |
1 |
Siupra-2002 with astersik |
| 1:49PM |
10 |
voice over atlantic |
| 1:17PM |
1 |
TDM400P not detecting hangup and not hanging up |
| 12:11PM |
1 |
Problem with IAXy |
| 11:40AM |
0 |
CVSHEAD callerid not working |
| 11:08AM |
2 |
play each person's voicemail |
| 9:34AM |
0 |
How to cascade dial status back through IAX |
| 8:49AM |
1 |
Multiple Line Appearances / Why use this? |
| 8:46AM |
2 |
Server Brand |
| 8:37AM |
1 |
Call goes through, but no audio |
| 8:14AM |
2 |
Pass through of T.38 |
| 8:00AM |
0 |
Slight OT: Multi WAN Router and SIP Calls |
| 7:56AM |
0 |
Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist" |
| 7:39AM |
1 |
Multiple Instances of Asterisk (no contexts) |
| 7:28AM |
0 |
2 X100P and SIP inbound routing |
| 7:07AM |
2 |
All Circuits are busy |
| 7:06AM |
0 |
IVR Documentation and Samples. |
| 6:56AM |
0 |
Asterisk & Euro-ISDN |
| 6:53AM |
3 |
power over ethernet hub/switch |
| 6:47AM |
6 |
Not enough lines available for Asterisk implemetation |
| 6:19AM |
0 |
cvs head and seqno 102 (Critical Response) messages for Cisco 7960 |
| 5:56AM |
1 |
Hangup problem |
| 5:34AM |
2 |
Distinctive ringing on Cisco 79xx |
| 5:26AM |
0 |
Yuxin hardphones feedback |
| 5:22AM |
0 |
Sip clients through proxy |
| 5:02AM |
1 |
Additional: Several SIP clients behind router registerwiththe same IP, messing up call routing, any ideas? |
| 4:29AM |
0 |
sending fax....i'm in trouble ! |
| 3:56AM |
0 |
Extension a |
| 2:42AM |
1 |
pri gateway |
| 2:31AM |
0 |
Contexts are not being created - Asterisk BT100 Password Issue |
| 2:27AM |
0 |
who use astlinux with booting from DOM? |
| 1:50AM |
1 |
How to increase delay before incoming call answer with tdm400p |
| 1:33AM |
0 |
Setting up multiple trunk groups with different internal ring groups |
| 1:26AM |
2 |
Transfer calls from cellphone |
| 1:19AM |
1 |
(OT) Dialplan Standards for Business/Offices |
| 12:31AM |
1 |
(no subject) |
| |
| Wednesday September 7 2005 |
| Time | Replies | Subject |
|---|
| 11:15PM |
2 |
410P upgrade to 411P? |
| 10:25PM |
0 |
I should never be called! |
| 9:43PM |
1 |
OT: Differences between test equipment |
| 8:53PM |
0 |
Sipura-2002 Can not make outgoing calls, incoming calls works OK |
| 8:38PM |
1 |
Not can call to PSTN |
| 7:56PM |
1 |
asterisk frequently dead |
| 6:22PM |
0 |
Hack for Canadian weather |
| 5:53PM |
0 |
Problem with PRI channels, restarted after every call. |
| 5:46PM |
0 |
Need Help - Losing first few seconds of call when using Broadvoice |
| 4:18PM |
2 |
Want to use a remotely location POTS phone |
| 3:32PM |
2 |
g729 test |
| 3:30PM |
0 |
IVR Documentation an Sample. |
| 3:19PM |
0 |
Remote Provisioning for the PA1688 phones. |
| 3:11PM |
1 |
IAXy - no dailtone |
| 2:48PM |
0 |
asterisk-statv2 showing blank screens |
| 2:12PM |
1 |
externpass in voicemail |
| 1:46PM |
1 |
Several SIP clients behind router register with the same IP, messing up call routing, any ideas? |
| 1:30PM |
0 |
sip - aastra 9133i |
| 1:16PM |
0 |
Asterisk with Vonage problems |
| 12:50PM |
1 |
TDM400P not detecting hangup and not hanging up. |
| 12:40PM |
1 |
asterisk.org blocked - rejecting connections |
| 9:50AM |
1 |
ztcfg Kills My Dial Tone |
| 9:28AM |
0 |
Second Line does not Connect - HELP - misdn,sip |
| 9:08AM |
0 |
ArtDio IPF-2000 unable to send audio to Cisco 7940 until placed on hold and resumed |
| 8:45AM |
1 |
Asterisk crashed? |
| 6:56AM |
3 |
Extensions - Realtime |
| 6:34AM |
1 |
Polycom 300 with latest 1.5.3 firmware not registering |
| 6:30AM |
1 |
Speex codec - Out of buffer space |
| 6:29AM |
0 |
IAX PBX responds to IAX registration with expires time=0 |
| 6:05AM |
1 |
2 X100P and SIP outbound routing |
| 5:53AM |
1 |
Eeven Stranger - Asterisk BT100 Password Issue |
| 5:21AM |
2 |
Desincripcion de la lista de Asterisk |
| 4:47AM |
1 |
"-- PROGRESS with cause code 34 received"? |
| 4:35AM |
1 |
Packet Cable |
| 4:10AM |
3 |
channels VHF/ HF radio in asterisk |
| 4:05AM |
1 |
ISDN PBX integration |
| 3:26AM |
1 |
presence settings and Eyebeam |
| 2:20AM |
4 |
How to connect many analog lines to Asterisk? |
| 1:27AM |
0 |
Max concurrent faxes with txfax/spandsp? |
| 12:55AM |
3 |
Hosted PBX (vPBX) and Call/PickUP Groups |
| 12:18AM |
0 |
Some info about Cisco's 79xx, and Sipura's phones |
| 12:04AM |
2 |
asterisk, SIP, Re-INVITEs and different contexts |
| |
| Tuesday September 6 2005 |
| Time | Replies | Subject |
|---|
| 9:38PM |
0 |
IAX2 Problems causing server to hang |
| 9:36PM |
4 |
Working example of ALERT_INFO with Cisco ATAs? |
| 9:06PM |
4 |
Which Linux distribution? |
| 8:05PM |
5 |
PRI in and out |
| 7:08PM |
1 |
Some problems (SendDTMF, Wait, Parked Calls) |
| 6:13PM |
1 |
CTI and Asterisk |
| 4:55PM |
0 |
/dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8 |
| 4:17PM |
1 |
Occasional quiet voicemails |
| 3:47PM |
1 |
Asterisk as SIP/H.323 Signalling Gateway |
| 3:41PM |
2 |
Speaking of Polycom phones...updated ROM: ouch! |
| 2:43PM |
5 |
Good Polycom Dealer? |
| 2:15PM |
1 |
Routing depending on sip response code? |
| 2:02PM |
1 |
one extension goes straight to voicemail, others don't |
| 1:48PM |
1 |
Queue AgentCallBackLogin |
| 1:44PM |
1 |
Asterisk overheating on VIA Epia MSeriesmotherboard |
| 1:42PM |
0 |
AstriCon Update: Please Register ASAP - Free Phones |
| 1:41PM |
1 |
Utility to find length of wav49 file |
| 1:40PM |
0 |
asterisk handling of old voicemail messages |
| 1:09PM |
4 |
Sipura Devices and Asterisk? |
| 1:03PM |
3 |
Asterisk scenario |
| 12:59PM |
0 |
Wireless router with built-in VOIP(FXS) ports forAnsterisk |
| 12:48PM |
2 |
Polycom ip301 hangs at Running "sip.ld" |
| 12:26PM |
0 |
Transfering to voicemail problem with 1.2beta |
| 12:16PM |
2 |
Wireless router with built-in VOIP(FXS) ports for Ansterisk |
| 12:04PM |
0 |
Loging agents in |
| 11:42AM |
1 |
Threeway calling uses up two FXO lines |
| 11:40AM |
0 |
IP PBX Market Share and Growth |
| 11:39AM |
0 |
Weird SIP behaviour |
| 10:08AM |
1 |
Asterisk BT100 Password Issue |
| 10:05AM |
4 |
PHP and ASterisk Manager |
| 9:33AM |
1 |
/dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8) |
| 9:11AM |
2 |
Asterisk overheating on VIA Epia MSeriesmoth erboard |
| 8:34AM |
1 |
"all lines are busy" |
| 8:32AM |
1 |
Application rxfax missing ? |
| 8:27AM |
9 |
civil emergency comms: Asterisk + HAM |
| 8:24AM |
1 |
Can get IAX connection but no SIP connection? |
| 8:17AM |
3 |
TE406P audio drops |
| 8:10AM |
2 |
Business telephones |
| 7:24AM |
0 |
Help evacuees from LA, MS, AL locate lived ones |
| 2:35AM |
1 |
TDM 400p |
| 2:04AM |
2 |
Going crazy with FAX :-( |
| 1:47AM |
1 |
SIP Callgroups |
| |
| Monday September 5 2005 |
| Time | Replies | Subject |
|---|
| 11:38PM |
0 |
atxfer featuremap |
| 5:55PM |
0 |
Heartbeat with Broadvoice |
| 5:16PM |
2 |
USING TWO ACCOUNTS WITH BROADVOICE |
| 4:57PM |
1 |
unicall and cvs head |
| 2:13PM |
3 |
TDM11B pinout |
| 2:05PM |
0 |
Agentlogin transfer calls |
| 1:52PM |
2 |
Asterisk overheating on VIA Epia M Series motherboard |
| 12:46PM |
0 |
Asterisk as a GSM-Gateway? Possible or not?? |
| 12:38PM |
2 |
Zaptel issue |
| 12:35PM |
3 |
Assessing network quality |
| 12:19PM |
3 |
Cisco 7960 upgrades |
| 12:14PM |
3 |
Asterisk architecture |
| 11:48AM |
1 |
res_features.so (Call Features Resource) not loading |
| 11:13AM |
1 |
Unexpected results with "While" and "EndWhile" applications |
| 11:13AM |
2 |
Asterisk won't listen on another port |
| 9:39AM |
0 |
putty and winscp |
| 9:30AM |
2 |
"Provisioned, Down, Active", but D-channel seems to be fine |
| 9:20AM |
1 |
BT100 and BETA 1.0.7.11 |
| 9:17AM |
9 |
Asterisk Follow ME |
| 8:42AM |
1 |
User authentication and privileges |
| 8:41AM |
0 |
more accounts |
| 8:27AM |
0 |
ooh323c h323_convertAsteriskCapToH323Cap Don't know how to deal with mode 0x40 (slin) |
| 7:45AM |
0 |
Re: Asterisk-Users Digest, Vol 14, Issue 22 |
| 6:34AM |
1 |
A good HW |
| 6:16AM |
1 |
TDM Card FXO Question |
| 6:13AM |
6 |
asterisk CAPI dial-in issues |
| 5:40AM |
0 |
Asterisk clustering with SIP proxy? |
| 5:17AM |
0 |
asterisk@home and zaphfc dial out not working |
| 4:27AM |
1 |
SV: sending fax |
| 3:53AM |
0 |
queue transfers always get EXITWITHKEY |
| 3:19AM |
3 |
GotoIf sample... |
| 3:09AM |
0 |
Tr: MWI - message waiting indication |
| 2:35AM |
2 |
No DID on ZAP |
| 2:18AM |
2 |
DTMF issue on IVR |
| 2:08AM |
0 |
ReInvite not working |
| 1:40AM |
0 |
WG: Timeout when Dialing - HELP |
| 1:31AM |
0 |
(no subject) |
| 1:27AM |
2 |
Billing - Disable accounts when balance gets 0 value |
| 1:24AM |
0 |
Asterisk and SCCP unofficial site |
| 12:08AM |
4 |
sending fax |
| |
| Sunday September 4 2005 |
| Time | Replies | Subject |
|---|
| 11:59PM |
1 |
Problem with Asterisk app command Read... |
| 11:59PM |
1 |
hints and polycom IP 300 phones |
| 11:23PM |
0 |
help on 2 X-Lite: call failed: 404 not found |
| 10:31PM |
3 |
A few questions before final proposal... |
| 6:31PM |
1 |
Unable to hear. |
| 6:10PM |
1 |
kernel panic |
| 4:51PM |
3 |
Asterisk Real-Time Voicemail Configuration |
| 4:50PM |
0 |
Updated Chan Unistim? |
| 3:51PM |
0 |
FW: OH323 with Asterisk@home - seems incomplete |
| 2:55PM |
0 |
Messagenet.it |
| 2:38PM |
0 |
sipura spc.exe ? |
| 2:09PM |
1 |
Option 1 in IVR menu |
| 1:37PM |
0 |
SIP, NAT and MySQL support (sipfriends) |
| 1:25PM |
0 |
OT: Sipura SPA 200 Caller ID Problem |
| 12:14PM |
3 |
Nokia 32 Terminal |
| 11:34AM |
0 |
chan_sip.c:946 __sip_xmit |
| 8:34AM |
0 |
Asterisk SMS via IAX2? |
| 6:37AM |
0 |
donating VOIP gear to the relief efforts. |
| 6:18AM |
0 |
Any hardphones with SIP API? |
| 5:39AM |
2 |
HELP - How Do I Separate incoming channels from the others on a PRI |
| 5:14AM |
0 |
dial rule / prefix with # |
| 3:35AM |
0 |
Open G.729 / G.723.1 update, fixed memory leak |
| 3:03AM |
1 |
FW: Asterisk@home - requesting help on oh323, ISDN BRI and iConnectHere DID |
| 2:38AM |
0 |
IPSwichBoard designers wanted |
| |
| Saturday September 3 2005 |
| Time | Replies | Subject |
|---|
| 10:04PM |
5 |
Asterisk Community Participant; Katrina Refugee |
| 5:33PM |
0 |
Sipura spa841 problems |
| 3:42PM |
0 |
MWI - message waiting indication |
| 3:12PM |
0 |
How To Separate incoming channels from the others on a PRI |
| 3:06PM |
0 |
How Separate a few channels from the others on a PRI |
| 2:54PM |
2 |
Argentina - zapata.conf switchtype for Argentina |
| 2:20PM |
1 |
I connected my quicknet phonejack to the wall phone outlet and ....... |
| 1:59PM |
0 |
chan_iax2.c:7672 iax2_poke_noanswer |
| 1:04PM |
3 |
unicall deploy |
| 12:24PM |
1 |
*81, block CID, using ATA |
| 8:58AM |
0 |
DNS SRV and new Asterisk install |
| 8:53AM |
0 |
stale nonce? |
| 8:35AM |
1 |
equipment and network advice |
| 8:28AM |
1 |
newbie install problem. And I already searched everywhere! |
| 8:05AM |
1 |
Current status on _outgoing_ Swedish/Dutch DTMF CLIP for TDM400 FXS interfaces? |
| 7:39AM |
0 |
Debug info from txfax - howto? |
| 7:10AM |
0 |
How to tell reason for hangup or busy in SIP or IAX |
| 3:31AM |
1 |
Multiple ASTCC Cards Configuration |
| 1:46AM |
1 |
chan_capi [0.4.0|-cm-0.5.4] and Asterisk 1.2.0-beta1 - early B3 not early enough sometimes |
| |
| Friday September 2 2005 |
| Time | Replies | Subject |
|---|
| 10:31PM |
2 |
IVR Prompts |
| 8:51PM |
2 |
Sipura 3000 setup |
| 8:43PM |
0 |
STUN on PAP2-NA 2.0.12(LS) |
| 7:20PM |
0 |
CVS-HEAD Inband Ringing? |
| 6:31PM |
0 |
X101P ringing too long ! |
| 6:21PM |
0 |
Web-voicemail doesn't play files nor display default pictures |
| 5:02PM |
0 |
Need * Setup Help |
| 2:20PM |
1 |
Asterisk and Eyebeam |
| 2:03PM |
0 |
SER+ASTERISK voicemail |
| 1:19PM |
1 |
Dlink dph-140s/ACT P104SLD |
| 11:53AM |
0 |
Notification of new voicemail by various met hods |
| 11:52AM |
1 |
Call Return |
| 11:47AM |
2 |
Notification of new voicemail by various methods |
| 11:35AM |
0 |
chan_oh323.conf (inAccess version) |
| 11:09AM |
0 |
How to locate Toll Free Ownership |
| 11:04AM |
4 |
Receptionist |
| 10:08AM |
0 |
CallerID and CDR |
| 9:46AM |
1 |
Linux-HA Heartbeat2 and Asterisk |
| 9:16AM |
0 |
Recommendations for a low cost GSM phone |
| 9:04AM |
1 |
AG-468 4xFXS - my personal review |
| 8:45AM |
1 |
No application 'AgentsLogin' |
| 7:56AM |
0 |
TDM400 w/ FXS S110M pinout on RJ11 connector? |
| 7:52AM |
0 |
Unable to create RTP session |
| 7:26AM |
1 |
how to execute something after Dial() ? |
| 7:13AM |
1 |
G711u sound quality decrease with upgrade from 1.0.7 to CVS-HEAD? |
| 7:03AM |
1 |
Semi-OT: An idea for New Orleanstemporarycommunications infrastructure |
| 7:01AM |
0 |
Semi-OT: An idea for New Orleans temporary communications infrastructure |
| 7:00AM |
0 |
Zapata help needed howto configure nationalprefix for a single card |
| 7:00AM |
2 |
FW: defunct email kill list |
| 6:52AM |
0 |
Semi-OT: An idea for New Orleans temporarycommunications infrastructure |
| 6:30AM |
0 |
monitoring VM via speaker and grabbing connection |
| 6:03AM |
0 |
Why is that: Sep 2 08:25:03 NOTICE[1403]: -- Registration for '1096377@192.168.0.100' timed out, trying again |
| 5:59AM |
3 |
DTMF and "breaking through" voice prompts |
| 4:29AM |
0 |
sip SUBSCRIPTION bug in 1.0.9 |
| 3:47AM |
2 |
chan_capi hfcpci mISDN linux 2.6.12 not working |
| 1:52AM |
0 |
Call drops |
| 1:44AM |
6 |
Looking for better "Follow Me" |
| 1:40AM |
0 |
ASTCC-adding more than one trunk to one route |
| 12:47AM |
1 |
Italy FastWeb problem: ISDN line crashes every time cisco router turns off |
| 12:46AM |
1 |
Fax problem, missing/compressed lines |
| 12:42AM |
1 |
Setting wcte11xp card to use IRQ |
| 12:11AM |
1 |
Snom 360 problem |
| |
| Thursday September 1 2005 |
| Time | Replies | Subject |
|---|
| 8:28PM |
0 |
Re: Asterisk-Users Digest, Vol 14, Issue 1 |
| 8:19PM |
2 |
Any one in Toronto / Canada can help me! |
| 8:07PM |
1 |
TE406P seg fault on Stable |
| 7:15PM |
0 |
extra ring after answer on sip calls |
| 6:15PM |
1 |
RE: Hardware dimensioning issues To: <juanmoyano@southecon.com.ar> |
| 5:58PM |
0 |
Help on second dial |
| 4:40PM |
0 |
How to set CLIR when using call files ? |
| 4:02PM |
1 |
OT: SCALE 4x -- Call For Papers |
| 2:31PM |
0 |
Question about Asterisk connections |
| 1:33PM |
3 |
Automon filenames |
| 1:18PM |
1 |
Skipping problems on outgoing calls (using uLaw with an internal * server through Voxee) |
| 1:16PM |
1 |
Best costs effective solution... |
| 12:51PM |
2 |
Contact Directory on Polycom IP-501 phones |
| 12:34PM |
0 |
IAX2 how to disable VAD ? |
| 11:27AM |
2 |
ipvolution t1 cards |
| 11:13AM |
0 |
Two devices behind nat |
| 11:10AM |
1 |
dialparties.agi is returning no extensions to dial |
| 11:07AM |
1 |
Speed Questiosn |
| 10:56AM |
0 |
Fax trouble with HP 3330mfp (again) |
| 10:42AM |
1 |
sip jitter buffer in 1.2? |
| 10:37AM |
1 |
TOS bit and DSCP |
| 10:21AM |
0 |
Buying DIDs |
| 10:20AM |
1 |
Problem with include |
| 10:14AM |
0 |
How to resolve SMS/WAP/MMS/VoIP gateways on a shoestring? |
| 10:05AM |
0 |
*66 with Sipura devices. |
| 10:01AM |
0 |
Overhead Paging Systems...[More Info] |
| 9:58AM |
0 |
dialing extension, which context is searched |
| 9:47AM |
2 |
ztcfg problem |
| 9:18AM |
0 |
Outbound Authentication |
| 9:16AM |
1 |
Loop error when compiling CVS version of 1.2-Beta |
| 9:16AM |
0 |
Astaro SIP Proxy |
| 9:10AM |
0 |
RE: Asterisk with Meridian1 option11 in the UK |
| 8:22AM |
0 |
zapata nationalprefix-problem [Virus checked] |
| 7:53AM |
0 |
HELP - Queue Transfer |
| 7:45AM |
1 |
Snom 360 hold problem |
| 7:33AM |
4 |
Overhead Paging Systems... |
| 7:22AM |
0 |
Re: Polycom 301 second line registration |
| 6:36AM |
1 |
oh323 or h323 |
| 6:00AM |
6 |
Grandstream GXP-2000 Poor sound Quality |
| 5:39AM |
0 |
Mobilephone users get echo of them self when calling in to our asterisk server. |
| 5:24AM |
3 |
Snom 360 and hints |
| 5:20AM |
1 |
What this little red cross mean in AAH |
| 5:16AM |
1 |
Sipura 1001 Adapter with two lines using one RG11 jack |
| 5:05AM |
0 |
Help setting up trunk on AAH |
| 4:41AM |
1 |
How to execute StopPlayTones when a SIP phone is answered |
| 3:26AM |
0 |
Mulig_SPAM: More than one outgoing call |
| 3:13AM |
0 |
Micronet 5050s FXO gateway and hookflash transfers. |
| 1:46AM |
1 |
How to require a keypress on answer? |
| 1:06AM |
2 |
TE cards with ISDN BRI? |
| 12:49AM |
0 |
Asterisk@Home: How to changed AMP User Login andPassword |
| 12:33AM |
1 |
Asterisk run problem, was working, rebooted server, now nothing |
| 12:31AM |
2 |
Recommendation for 8 lines analog card in Australia |