| Tuesday August 31 2004 |
| Time | Replies | Subject |
|---|
| 11:18PM |
2 |
DeadAGI Application |
| 10:27PM |
0 |
Dial/Zap doesn't work |
| 10:15PM |
5 |
Line death not recognized on TDM400P? |
| 8:32PM |
1 |
install software version to mediatrix 1204 (how to) |
| 7:13PM |
3 |
All you polycom folks..... |
| 7:09PM |
1 |
T100P Configuration for Mixed Voice & Data |
| 5:12PM |
0 |
MP3Player strange error |
| 5:01PM |
3 |
Cisco 79XX SIP Ring Tones |
| 4:32PM |
2 |
Asterisk codecs and packet size |
| 4:22PM |
0 |
good Dutch TTS ? |
| 4:06PM |
0 |
Can only call asterisk once |
| 2:16PM |
1 |
Hardware suggestion |
| 2:16PM |
1 |
Why is it called 'Comedian Mail? |
| 1:32PM |
4 |
T100P No D-channels |
| 1:22PM |
0 |
Streaming an audio file to a Zap channel before answer |
| 1:04PM |
5 |
OT: Headset for Cisco 7960? |
| 12:57PM |
1 |
Going to voicemail instead of queue if no agent is logged in ? |
| 11:14AM |
4 |
IAX Client |
| 11:12AM |
3 |
Can asterisk detect BUSY signal? |
| 11:10AM |
0 |
detect telco voicemail stutter-tone |
| 11:05AM |
0 |
error: CDR on channel '<unknown>' has not started |
| 10:52AM |
1 |
Analog lines and TDM card |
| 10:51AM |
1 |
Losing voice on Digium demo server - how to spot problem ? |
| 10:35AM |
0 |
Snom Programmable button Mini Howto and ringstate patch |
| 10:24AM |
2 |
multiple lines with SIP like MGCP? |
| 9:09AM |
1 |
PSTN noob question |
| 8:37AM |
1 |
Polycom IP 300 - Displaying Only CallerNAME... What about NUMBER? |
| 8:35AM |
0 |
answer from wrong port |
| 8:31AM |
0 |
Can i send calling costs to a SIP IP phone display |
| 8:26AM |
0 |
Polycom SoundPoint... Gains -Which is for speakerphone |
| 8:10AM |
0 |
Polycom SoundPoint... Gains - Whichis for speakerphone |
| 8:08AM |
1 |
Polycom IP 300 - Displaying Only Caller NAME... What about NUMBER? |
| 7:53AM |
0 |
X100P Questions: Voicemail and Phone Port questions |
| 7:29AM |
0 |
newbie question about PBX Call Pickup |
| 7:14AM |
2 |
Harddisk noise on TE410P |
| 7:07AM |
4 |
which distro for asterisk? |
| 7:05AM |
2 |
limit the length of extensions |
| 6:53AM |
1 |
SIP registration with public dynamic ip address |
| 6:18AM |
0 |
BRI numbers |
| 5:13AM |
0 |
extensions => s,1,Dial(Zap/2/number) noise |
| 4:42AM |
3 |
pattern matching problems |
| 4:14AM |
0 |
Transfer from MOH to MOH doesn't work. |
| 3:50AM |
0 |
** ASTRICON * LAST CALL FOR REGISTRATION |
| 3:45AM |
0 |
Polycom SoundPoint... Gains - Which isfor speakerphone |
| 3:35AM |
0 |
Transfer to queue |
| 2:00AM |
1 |
Do not get calldeflection (capiCD) to work. |
| 1:17AM |
0 |
transferring call to another line |
| |
| Monday August 30 2004 |
| Time | Replies | Subject |
|---|
| 11:10PM |
4 |
Newbie - Voicemail Password Help |
| 9:23PM |
0 |
My Three-way calls work backwards |
| 9:10PM |
2 |
VoicePulse Connect DTMF with IAX2 |
| 8:43PM |
1 |
Asterisk and Citrix |
| 6:14PM |
0 |
Reload crashes Asterisk ? |
| 5:43PM |
0 |
Re: New to Asterisk and a question |
| 5:06PM |
0 |
Re: New to Asterisk and a question |
| 4:41PM |
1 |
Polycom SoundPoint... Gains - Which is for speakerphone |
| 4:29PM |
0 |
Delays while playing a message |
| 3:33PM |
0 |
[Fwd: [Asterisk-Dev] Snom Programmable button Mini Howto and ring state patch] |
| 3:07PM |
1 |
AstriCon Reminder: Please register today |
| 1:56PM |
2 |
number of simultaneous calls with E&M |
| 1:44PM |
0 |
Asterisk with Sayson 480 ADSI |
| 12:51PM |
0 |
MWI Light On SoundPoint IP 300 |
| 12:16PM |
0 |
SOLVED - Problems compiling zaptel driver |
| 11:25AM |
2 |
How record conversation to sound file ? |
| 11:23AM |
1 |
Snom Programmable button Mini Howto and ring state patch |
| 11:23AM |
0 |
Redirect SIP calls to the SIP provider sipgate.de |
| 11:12AM |
1 |
SIPJack |
| 10:57AM |
1 |
Faxing with an IAXy |
| 10:17AM |
0 |
Anyone used Tecom IP2004 SIP phone? |
| 9:55AM |
7 |
Polycom SoundPoint IP 300 Configuration |
| 9:40AM |
1 |
Problems with T100P card not releasing channels. |
| 9:27AM |
1 |
Does anyone have a working GR-303 config? |
| 8:25AM |
1 |
How does call routing actually work with SIP? |
| 7:53AM |
0 |
Parsing problem with oh323 |
| 7:36AM |
1 |
Voiceronix and asterisk |
| 6:31AM |
0 |
New Error Log Messages |
| 5:07AM |
0 |
zaphfc success stories |
| 5:06AM |
2 |
Suitable for Dynamic IVR Platform? |
| 5:01AM |
0 |
AVM Fritz on Fedora 1 |
| 4:26AM |
0 |
SIP and IAX Registration Problem with Dynamic IP |
| 4:13AM |
1 |
IAX.conf problem (NEWBIE ALERT!) |
| 2:33AM |
0 |
Problem with modified-Prepaid-Application |
| 2:33AM |
1 |
Voicetronix OpenLine4 immediately hangs up on every call |
| 1:16AM |
1 |
X100 and call duration |
| |
| Sunday August 29 2004 |
| Time | Replies | Subject |
|---|
| 11:30PM |
2 |
Mix Data and SIP Phones |
| 10:38PM |
0 |
Help debugging voicemail problem |
| 9:32PM |
2 |
zaptel configuration |
| 8:56PM |
2 |
AgentCallbackLogin by other means |
| 8:06PM |
2 |
Still unacceptable echo on X101P |
| 7:44PM |
1 |
${CONTEXT} |
| 6:00PM |
0 |
Asterisk and codecs? |
| 5:34PM |
1 |
Bridging audio in cmd_dial() before connect completes? |
| 3:21PM |
0 |
Static Problem (t100p - Channel Bank) |
| 3:18PM |
0 |
Asterisk H.323 channel... |
| 3:06PM |
5 |
Broadvoice BYOD Plans - 3-way and Call Waiting |
| 2:58PM |
7 |
SMS & Asterisk |
| 2:57PM |
0 |
Which Zaptel release goes with Asterisk-1.0-RC2 ??? |
| 2:44PM |
3 |
Revert to dial tone? |
| 2:39PM |
0 |
System freezes when using Festival with usecache |
| 2:15PM |
0 |
SMS and asterisk |
| 1:41PM |
1 |
not getting ringing/busy/answer feedback on my PRI |
| 1:08PM |
0 |
Python and AGI |
| 1:07PM |
2 |
Jitter buffer |
| 12:20PM |
2 |
Servers |
| 11:11AM |
2 |
Sip device not login or register calls to that device go to busy voicemail not un-available |
| 9:34AM |
1 |
Empty Queues |
| 2:51AM |
0 |
Asterisk Assistants for Linux or Windoze??? |
| 2:34AM |
0 |
IAXy died |
| 2:10AM |
1 |
Termination in Holland. |
| 12:51AM |
1 |
Mobile phone integration via bluetooth |
| |
| Saturday August 28 2004 |
| Time | Replies | Subject |
|---|
| 11:20PM |
1 |
where can I find spandsp? |
| 7:27PM |
5 |
Distinctive ring detection problem |
| 10:57AM |
3 |
POE |
| 9:23AM |
10 |
Broadvoice problem |
| 8:38AM |
3 |
SIP Provider for Reseller |
| 8:37AM |
1 |
UK Disconnect supervision with TDM400P |
| 8:24AM |
1 |
asterisks and vonage |
| 8:13AM |
1 |
IAX dialing indication tone (PI = 8) |
| 6:18AM |
0 |
switch statement in extensions.conf |
| 4:22AM |
4 |
incomming call rejected using IAX2 with FWD |
| 2:34AM |
4 |
G729 licenses |
| 1:31AM |
0 |
ISDN BRI card exepriences in UK |
| 1:07AM |
0 |
FXO probs in Aus. Should I give up? |
| |
| Friday August 27 2004 |
| Time | Replies | Subject |
|---|
| 8:35PM |
3 |
Disconnection From IAXTel |
| 4:53PM |
3 |
"7" Dialing gives a busy signal |
| 4:40PM |
2 |
Zap & ANSWER the Call |
| 3:55PM |
0 |
mysql-vm-routines and Directory app |
| 2:38PM |
5 |
IAXy Power in Australia? |
| 2:12PM |
1 |
does agi wait for digit work in a meetme room ? |
| 11:59AM |
1 |
Retrieve Info from Cisco Call Manager |
| 11:58AM |
5 |
iaxtel and jitterbuffer |
| 11:42AM |
0 |
regex and gotoif question |
| 10:26AM |
4 |
Speech Recognition and Asterisk |
| 10:22AM |
1 |
libr2 |
| 10:14AM |
0 |
Asterisk Assistants Custom Icon |
| 10:07AM |
0 |
Re: how to fetch a call? (Tony Mountifield) |
| 9:59AM |
1 |
Help with a fax via Grandstream Handytone 286? |
| 9:45AM |
0 |
Asterisk & Max TNTs |
| 9:32AM |
2 |
Someone please try MeetMe MOH with latest CVS and GS phone |
| 9:31AM |
2 |
No audio on PRI channel answered by Playback() orMeetMe() |
| 9:19AM |
1 |
No audio on PRI channel answered by Playback() or MeetMe() |
| 8:51AM |
3 |
Can a Macro call another Macro ? |
| 8:47AM |
2 |
Are there any graphic designers on this list? |
| 8:41AM |
6 |
FXOs |
| 8:34AM |
1 |
IAX2 --> IAX2 confusion, it doesn't work... |
| 8:31AM |
0 |
questions and recommendations |
| 8:28AM |
0 |
Release 1.01 of FWD Assistant available (bugfix release) |
| 8:24AM |
1 |
Problem dialing out to Free World Dialup |
| 8:04AM |
0 |
Cisco 7940 SIP Firmware - Help. |
| 7:58AM |
0 |
auto-gain, or different gain between incoming and outgoing calls (EURO ISDN PRI) ? |
| 7:55AM |
1 |
xlite Problems |
| 7:45AM |
4 |
Queue Announcement not until after # accept call pressed |
| 7:44AM |
0 |
ACD ringall + roundrobin |
| 7:42AM |
1 |
Re: sip change? (Rich Adamson) |
| 7:37AM |
0 |
Cisco 7940 Sip Firmware |
| 7:15AM |
1 |
Problems dialing out with T100P and Adtran |
| 7:14AM |
3 |
sip change? |
| 7:11AM |
2 |
how to fetch a call? |
| 7:06AM |
0 |
h323 with Fedora 2 & GCC 3.3 |
| 7:06AM |
0 |
Voicetronix Segmentation Fault |
| 7:03AM |
1 |
Asterisk compatible E1 cards |
| 6:24AM |
0 |
OT re: sip change? |
| 6:16AM |
1 |
Cisco 7940 - SCCP or SIP? |
| 6:13AM |
0 |
Queues - CallbackLoging Automaically? |
| 6:05AM |
2 |
Using regular expression in dialplan |
| 5:29AM |
0 |
Touch tone problem |
| 3:15AM |
2 |
FXO interfaces used in UK? |
| 2:48AM |
1 |
Can't flash 7960: P0S30200 .bin not found |
| 2:32AM |
0 |
Updated app_mysql.c, enabling use of INSERT and UPDATE |
| 2:20AM |
0 |
'set verbose 3' or other way to get '-vvv' level debugging out of running background asterisk? |
| 1:46AM |
0 |
Hangup() doesn't always when talking to Nortel Norstar over CT1 E &M wink-start trunk line? |
| 1:28AM |
3 |
Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits? |
| 12:28AM |
0 |
1stwave |
| |
| Thursday August 26 2004 |
| Time | Replies | Subject |
|---|
| 9:32PM |
1 |
Hey admin: Do we have to have a 92-char reply-to header? |
| 9:02PM |
0 |
Newbie with IAX2 |
| 7:46PM |
2 |
VoIP Telephony with Asterisk book |
| 7:24PM |
1 |
* and VoIP for LD |
| 6:57PM |
0 |
Problems with ISDN (NT-Mode) - Error Messages inside |
| 5:36PM |
0 |
Polycom SoundPoint IP 300 w/ SIP Software - Where Can I Get One? |
| 5:13PM |
1 |
Polycom non-IP phones |
| 5:11PM |
5 |
TDM400P Problems |
| 4:14PM |
4 |
PLC (Packet loss cancel) questions |
| 3:29PM |
2 |
Asterisk mysql database |
| 2:47PM |
0 |
Error in app_callingcard.c with database connection |
| 2:35PM |
1 |
Newbie needs help - Dev_Kit_Lite installation problem |
| 12:49PM |
1 |
Linux keeps deleting the ZAP files?? |
| 10:16AM |
1 |
No signal from ISDN-phone connected to hfc card in NT-mode |
| 10:00AM |
0 |
ShoreTel (ShoreLine) branded phones work with Asterisk? |
| 9:30AM |
1 |
RC2 and VoicePulse |
| 9:15AM |
0 |
IPDialog call transfer |
| 8:31AM |
2 |
Sip Channel CLI |
| 8:25AM |
2 |
Sound card |
| 8:22AM |
1 |
Anyone using Asterisk on Slackware 9? |
| 7:51AM |
2 |
Asterisk+IVR functions trouble |
| 7:42AM |
0 |
ilbc asterisk and handytone/Xpro |
| 7:37AM |
1 |
Wil spandsp work with i4l driver? |
| 7:15AM |
2 |
Astricon hotel recommendations.....? |
| 6:26AM |
4 |
ISDN Card Recommendation |
| 6:22AM |
0 |
chan_oh323 build (resubmit w/ new title) |
| 6:16AM |
0 |
Can't make zaptel on red hat 9 |
| 6:04AM |
1 |
GRSecurity and ALSA on a Gentoo Server |
| 6:04AM |
2 |
<<< AGI and EXEC function CONFIRMATION >>> |
| 6:04AM |
1 |
chan_oh323: __use_ast_pthread_create_instead __ (was: chan_oh323 loading error) |
| 4:32AM |
6 |
chan_capi module |
| 4:25AM |
0 |
Asterisk media problem behind NAT |
| 2:49AM |
0 |
"for Lack of RTP activity in 0 seconds" |
| 2:38AM |
0 |
TDM400P + 1FXS + 1FXO for sale |
| 2:03AM |
0 |
R: Grandstream Firmware |
| 1:28AM |
1 |
Grandstream Firmware |
| 1:24AM |
0 |
I have on isdn phone and one isdn-card (ast-pbx)attached (to my line)... |
| 1:13AM |
4 |
Codec |
| 1:09AM |
0 |
I have on isdn phone and one isdn-card (ast-pbx) attached (to my line)... |
| 1:00AM |
0 |
Out Dial Problem |
| |
| Wednesday August 25 2004 |
| Time | Replies | Subject |
|---|
| 8:21PM |
0 |
Dial by name across 2 systems |
| 8:13PM |
0 |
Could use some advice/reality check from someone knowledgeable |
| 7:25PM |
1 |
home lab setup |
| 6:15PM |
0 |
Phone recommendations |
| 6:15PM |
3 |
IP Phone Recommendation ? |
| 6:02PM |
3 |
Fax detect |
| 4:46PM |
0 |
Error when closing asterisk |
| 4:22PM |
1 |
7960 Looses DHCP Lease when 7920 boots!? |
| 4:13PM |
0 |
New wiki pages: Avaya 4602 upgrade and configuration |
| 3:47PM |
0 |
spamdsp capacity |
| 2:41PM |
3 |
Distinctive Ring Cadences |
| 1:54PM |
0 |
unknown SMS Messages |
| 1:51PM |
0 |
chan_sccp with multi-lines and 7960's |
| 1:38PM |
2 |
Voip phones & headsets |
| 1:32PM |
1 |
Voicemail forwarding from SER & extensions.conf |
| 1:24PM |
2 |
asterisk & chan_sccp |
| 12:16PM |
1 |
chan_oh323: __use_ast_pthread_create_instead__ (was: chan_oh323 loading error) |
| 11:38AM |
1 |
Which end hungup? |
| 11:11AM |
5 |
Something broken in voicemail app?? |
| 11:09AM |
0 |
Read and dtmf ? |
| 10:42AM |
2 |
Avaya dialing problems |
| 10:12AM |
1 |
2x HFC ISDN Cards - SuSE 9.1 - Problems with making calls |
| 10:02AM |
1 |
Sound deformation during conversation or IVR message |
| 9:29AM |
1 |
WTS: Just arrived Brand New Cisco IP phones |
| 8:40AM |
0 |
oh323_indicate: Ignoring PROGRESS indication |
| 8:37AM |
2 |
chan_oh323 and cdr |
| 8:29AM |
2 |
GrandStream HT-486 ATA as VoIP Gateway |
| 8:28AM |
0 |
*, Sipura, and cisco dtmf |
| 7:57AM |
3 |
chan_sccp2 & 7960 -- documentation and example request. |
| 7:15AM |
4 |
YAAN (Yet Another Asterisk Newbie) |
| 7:08AM |
7 |
TDM400P lockups (FXO) |
| 6:52AM |
0 |
KS-Like controlling |
| 6:50AM |
1 |
Asterisk PBX and backup Circuits |
| 6:48AM |
1 |
External BRI - is there such thing? |
| 6:35AM |
1 |
Problem of set up asterisk-1.0-RC2.tar.gz with asterisk-prepaid-0.3.1 |
| 6:25AM |
0 |
How to be taken out of the queue? |
| 6:02AM |
0 |
Incorrect sound |
| 5:44AM |
0 |
ericsson drg22 - will not connect to asterisk |
| 5:17AM |
0 |
Asterisks |
| 4:23AM |
2 |
Advice on BT ISDN Services (UK) |
| 3:58AM |
0 |
freebsd 4.10 and port misc/zaptel |
| 3:56AM |
1 |
Individual call-forwarding on ISDN |
| 3:37AM |
0 |
sip history - which file ? |
| 3:10AM |
0 |
Loading module chan_oh323.so |
| 3:07AM |
4 |
GSM to BRI ISDN Gateway |
| 2:35AM |
3 |
Blocking a channel on T1 |
| 2:34AM |
2 |
spandsp and certain (e.g. Canon) fax machines |
| 1:46AM |
2 |
Parallel T1 cable? |
| 1:06AM |
3 |
Compressing a dialplan |
| 12:01AM |
1 |
Closing bug reports without fixing the repor ted problem |
| |
| Tuesday August 24 2004 |
| Time | Replies | Subject |
|---|
| 11:12PM |
0 |
asterisk directory service |
| 11:01PM |
0 |
Closing bug reports without fixing the reported problem |
| 10:02PM |
3 |
desparate for help DEV LITE KIT |
| 9:18PM |
0 |
RTP and NAT |
| 8:24PM |
2 |
Voicepulse incoming / dial extension |
| 7:59PM |
0 |
Perl AGI - no output from agi script to Aste risk |
| 7:23PM |
1 |
Zaptel/Zapata and SIP relationship |
| 6:08PM |
0 |
How can i configure extensions.conf. |
| 5:20PM |
1 |
Can I have same numbers in different contexts ? |
| 4:11PM |
3 |
Hardware for PBX with 4 incoming/outgoing lines and 20 phones |
| 3:56PM |
2 |
Grandstream Budgetone BT-101 and VoipJet |
| 3:20PM |
1 |
[Asterisk Users] Help with SIP Hosted Billing Service |
| 3:07PM |
2 |
Parking and Extensions |
| 2:55PM |
2 |
Remotely change call forward |
| 2:49PM |
1 |
RC2 and Netmeeting 3.01 ? |
| 2:31PM |
3 |
Asterisk with Adit 600 |
| 2:09PM |
0 |
off list |
| 1:45PM |
0 |
Using Lucent/Avaya 64XX sets with asterisk |
| 1:04PM |
1 |
X100P connected to Meridan-1 system will not disconnect call |
| 1:00PM |
7 |
SMP Performance |
| 12:46PM |
0 |
Monitor() hangs |
| 12:21PM |
0 |
Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs |
| 12:06PM |
1 |
CPU utilization |
| 12:04PM |
1 |
MailMan slowness... |
| 12:01PM |
0 |
R X100P connected to Meridan-1 system will not disconnect call |
| 11:33AM |
0 |
Astricon - call for help |
| 11:12AM |
1 |
CVS RPMs for Mandrake 10 (Zaptel and Asterisk) |
| 11:10AM |
3 |
ex-girlfriend logic not working in latest CVS? |
| 10:52AM |
0 |
Warning when I use iax2 for inbound and outbound calls |
| 10:49AM |
2 |
Voicemail & "Couldn't read username" error |
| 10:42AM |
0 |
R X100P connected to Meridan-1 system will notdisconnect call |
| 10:27AM |
2 |
call queue help |
| 10:16AM |
1 |
[Asterisk-Dev] Asterisks |
| 10:10AM |
1 |
Autoattend detecting same digit twice |
| 10:09AM |
1 |
Swissvoice IP10S and RTP Port Operation |
| 9:19AM |
3 |
Asterisk to Vonage |
| 8:32AM |
0 |
problems with the mailing list?? |
| 8:31AM |
2 |
SIP Provider in India/Pakistan/Bengladesh |
| 7:52AM |
0 |
MGCP 1.0 NCS 1.0 on a motorola SBV4200 |
| 7:50AM |
0 |
Problem with sound on Wildcard TE410P |
| 7:11AM |
0 |
Sangoma Card Support |
| 7:07AM |
3 |
Bell Canada Caller-ID |
| 6:49AM |
0 |
If you run Archlinux... |
| 5:50AM |
0 |
h323 direct calls |
| 4:56AM |
1 |
hfc-s card, brii-stuff.0.1.0-RC4a, zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. |
| 3:17AM |
3 |
Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 |
| 1:00AM |
0 |
g729 Codec on Max OS X |
| |
| Monday August 23 2004 |
| Time | Replies | Subject |
|---|
| 11:33PM |
1 |
strange problem PBX-Asterisk |
| 10:34PM |
4 |
Telemarketer screening |
| 9:53PM |
1 |
H323 outgoing calls |
| 9:10PM |
0 |
spandsp make error mmx.h |
| 5:25PM |
6 |
Dell PowerEdge 750 rackmount |
| 4:58PM |
2 |
Hold the phone! |
| 4:20PM |
3 |
newb question regarding DTMF |
| 4:20PM |
2 |
VoicePluse DID problem |
| 4:12PM |
4 |
Asterisk WITH Swyx... Any Idea? |
| 3:04PM |
0 |
multiple X100P config? |
| 2:38PM |
1 |
Asterisk <------- Quintum SIP Registration |
| 2:00PM |
3 |
Cisco 7960G, Skinny.conf, and reboots |
| 1:32PM |
1 |
Firmware Update IAXy |
| 12:14PM |
6 |
Asterisk Install on Kernel 2.6.x |
| 10:45AM |
2 |
Queue Monitor |
| 10:30AM |
0 |
Guide to getting started with Asterisk on MacOSX |
| 9:40AM |
1 |
(no subject) |
| 9:31AM |
0 |
MGCP and dialing out |
| 7:55AM |
5 |
SIP "unphones" |
| 7:52AM |
0 |
Swissvoice MGCP Error 502 |
| 7:15AM |
2 |
HFC-S in NT mode, wiring? |
| 7:07AM |
2 |
Cisco 7940 Question |
| 6:38AM |
1 |
routing telephone calls via "switchboard/asterisk". |
| 6:00AM |
1 |
zaptel installation |
| 5:32AM |
2 |
Question about dial out via Zap |
| 4:38AM |
2 |
Adit 600 FXO & FXS |
| 3:54AM |
0 |
How to recover the problem with pgsql and asterisk |
| 3:15AM |
1 |
using ChanIsAvail |
| 2:25AM |
1 |
Choosing between TE405P and TE410P |
| 2:17AM |
6 |
2 servers |
| 2:03AM |
1 |
Problem with asterisk and postgresql |
| 1:05AM |
0 |
Bug in recording uavarible |
| 12:10AM |
1 |
Problem with mysql and with asterisk |
| |
| Sunday August 22 2004 |
| Time | Replies | Subject |
|---|
| 9:36PM |
3 |
zap show channels - no such command |
| 8:02PM |
1 |
Queue Calls without using the |
| 7:07PM |
1 |
Pulse dialed digit recognization |
| 6:44PM |
3 |
SIP Phone recommendation for Receptionist |
| 4:58PM |
1 |
MusicOnHold problem |
| 10:59AM |
4 |
Error compiling meetme2 |
| 9:24AM |
3 |
asterisk T100P to Merlin Legend |
| 9:11AM |
5 |
skinny or sccp? |
| 8:46AM |
0 |
We have thousands of U ISDN interface phones |
| 8:40AM |
0 |
Release 1.0 of FWD Assistant for MacOSX now available |
| 8:23AM |
0 |
tsu 120 |
| 7:27AM |
0 |
app_mp3 with bri-stuff.0.1.0RC4a does not work |
| 4:35AM |
1 |
Spandsp - opencall.org offline |
| |
| Saturday August 21 2004 |
| Time | Replies | Subject |
|---|
| 5:08PM |
1 |
IAX2 DTMF not recognized - Bug report - Help sought |
| 4:59PM |
0 |
Cisco IP Phone- disjoin conference |
| 3:42PM |
1 |
need help with zaptel. |
| 2:10PM |
1 |
ActXPhone Active X control link is dead - has anyone cached files ? |
| 1:53PM |
1 |
Number and name for SIP extension at the same time ? |
| 1:49PM |
0 |
Can I place calls to arbitrary sip uri address from windows messenger 5 ? |
| 1:19PM |
0 |
cmd Monitor creating sound notification on channel |
| 11:47AM |
5 |
Asterisk and software Raid |
| 10:37AM |
0 |
How to minimally configure modules.conf loading? |
| 10:36AM |
0 |
autocreatepeer and sip peer options |
| 10:15AM |
1 |
Zultys 4x4 or 4x5 |
| 7:03AM |
3 |
zaptel config |
| 6:01AM |
0 |
welltech fxo and * |
| 4:42AM |
2 |
system reboot often? |
| 4:33AM |
0 |
Asterisk and AVM FritzBox Fon (Germany) |
| 12:49AM |
1 |
just-added second X100P |
| 12:07AM |
0 |
hangup question |
| |
| Friday August 20 2004 |
| Time | Replies | Subject |
|---|
| 9:57PM |
1 |
PoE injectors |
| 9:09PM |
0 |
spandsp tiff debian question |
| 7:00PM |
1 |
Sipura partners with Linksys for new combo router/SIP ATA |
| 5:17PM |
1 |
Forwarding PSTN to PSTN? |
| 4:35PM |
1 |
Adding macros causes ringing to fail |
| 4:14PM |
3 |
Strange problem with Dial |
| 4:07PM |
1 |
Testing a channel's status |
| 3:31PM |
3 |
BT Communicator (SIP???) and Asterisk |
| 3:23PM |
0 |
Max retries exceeded on call - seqno 102 (critical request) |
| 3:06PM |
0 |
snom 200 and * question |
| 2:22PM |
0 |
Invalid module format |
| 1:34PM |
1 |
TE410P - ZT_CHANCONFIG failed |
| 1:20PM |
0 |
Is posissble TE405P ? |
| 1:00PM |
4 |
Help with upgrading 7960 SCCP to SIP |
| 12:38PM |
1 |
Silence on incoming monitored calls |
| 12:28PM |
3 |
determining what number was dialed? |
| 12:23PM |
1 |
CDR problems with MySQL |
| 12:08PM |
0 |
chan_h323 doesn't pass audio before call is answered |
| 11:34AM |
1 |
Incoming MSN via ZapHFC -> to SIP |
| 11:11AM |
0 |
Oh323 installing problem make[1]: *** [asteriskaudio.o] Error 1 |
| 10:19AM |
7 |
how to collect user entered digits |
| 9:47AM |
0 |
Stream File (AGI) question |
| 9:32AM |
2 |
Handling invalid extensions |
| 9:09AM |
6 |
Asterisk PBX Functions via SIP phone |
| 6:52AM |
2 |
tT funktions |
| 6:44AM |
0 |
Re: problem in with mysql modules.conf to load cdr_addon_mysql.so (Lerale Erwan) |
| 6:39AM |
1 |
dual servers |
| 6:03AM |
2 |
Creating 79xx Configs |
| 5:43AM |
6 |
Sipura endpoints |
| 5:19AM |
1 |
Finding operator from ISDN signalling? |
| 5:14AM |
0 |
Zaptel Problem after Upgrade |
| 3:42AM |
0 |
IAXY S100I noise |
| 3:42AM |
2 |
Multi-bitrate codecs |
| 3:35AM |
1 |
x100p won't answer |
| 3:02AM |
0 |
Problem with asterisk and pgsql |
| 2:55AM |
2 |
problem in with mysql modules.conf to load cdr_addon_mysql.so |
| 2:39AM |
4 |
telnet and Root |
| 12:18AM |
0 |
Operator-type phone |
| 12:03AM |
1 |
from Newcomtech Co,. Ltd Help us. |
| |
| Thursday August 19 2004 |
| Time | Replies | Subject |
|---|
| 10:28PM |
2 |
Atick Certification on FXO Modules (Australia) |
| 9:21PM |
7 |
Where to purchase ISDN (BRI) cards in Australia (preferably) |
| 6:13PM |
2 |
IAX2 Port strangeness |
| 4:42PM |
4 |
Request for help designing an unusual * application |
| 4:15PM |
0 |
ATA with Built-in Switch |
| 3:56PM |
1 |
AGI Script: calleridnamelookup.agi |
| 3:13PM |
3 |
GrandStream BT101 Attended Transfers |
| 3:11PM |
6 |
How to run different codecs between the same endpoints on an IAX trunk? |
| 3:10PM |
2 |
residential sip phone |
| 3:03PM |
1 |
Received packet with bad UDP checksum |
| 2:59PM |
1 |
More on Broadvox |
| 2:32PM |
1 |
Inband announcement of parking slot from app _parkandannounce? |
| 2:29PM |
1 |
Granstream BT100 Rings Once and Waits for Call Pickup? |
| 1:43PM |
2 |
False Hangups on Asterisk |
| 12:53PM |
4 |
Does Granstream BT100 Conference Button Work? |
| 12:12PM |
1 |
Pingtel registration failing |
| 11:51AM |
1 |
Isdn4Linux and DTMF |
| 11:38AM |
0 |
UserAgent support for MGCP |
| 11:30AM |
2 |
Opencall.org and SpandDSP |
| 11:05AM |
1 |
Problems loading chan_h323 on Opteron 64 bit |
| 10:47AM |
0 |
Andre Bierwirth's ring state patches for SNOM 200 programable buttons |
| 10:26AM |
2 |
Dial from AGI [MSG] |
| 10:02AM |
1 |
Debit/Credit Card Terminals |
| 10:00AM |
1 |
No Success with SwissVoice. |
| 9:25AM |
1 |
SpanDSP/RxFax help... |
| 9:07AM |
7 |
Can PSTN CallerID be fowarded to a SIP phone extension? |
| 8:47AM |
2 |
T100P PCI or PCI-x |
| 6:26AM |
3 |
Echo SIP-T100P-PRI |
| 6:14AM |
0 |
fax output from Asterisk into file |
| 5:33AM |
0 |
Using Asterisk as a voicemail only with legacy switch |
| 5:02AM |
0 |
SIP reinvite code negotiation |
| 4:59AM |
2 |
Floating point exception help |
| 4:40AM |
1 |
chan_phone - compilation problem |
| 4:02AM |
2 |
Multiple SIP phones ringing for same extension |
| 3:57AM |
0 |
Error message "phone_mini_packet Read returned -1" and strange Internet PhoneCard behavior |
| 1:42AM |
0 |
Asterisk & SER |
| 12:42AM |
1 |
not yet a new user, some questions |
| 12:02AM |
1 |
Festival Issues |
| |
| Wednesday August 18 2004 |
| Time | Replies | Subject |
|---|
| 7:37PM |
1 |
Not a User Yet - but have some questions |
| 6:55PM |
7 |
CID on internal extensions |
| 5:17PM |
1 |
Another small suggestion patch |
| 4:56PM |
0 |
SIPp and asterisk question |
| 4:31PM |
1 |
Three tdm400p's (loaded with FXOs) |
| 4:28PM |
2 |
Help! Cisco 7960 - dialing any digit makes the phone go nuts! |
| 4:00PM |
0 |
static noise TDM400P+T100P+Adtran Channel bank(TA750) |
| 3:31PM |
0 |
Automated FWD configuration of Asterisk on OSX |
| 2:35PM |
1 |
Small patch to zaptel Makefile |
| 2:00PM |
1 |
PCI Express and Digium Cards |
| 1:34PM |
0 |
SIP/IAX2 mysql auth + FreeBSD |
| 1:29PM |
1 |
service zaptel start |
| 12:38PM |
1 |
Restart Digium Cards |
| 12:34PM |
0 |
my iaxy makes a strange noise... |
| 12:14PM |
1 |
Newbie physical layout question |
| 11:46AM |
1 |
RE: New $85 VOIP Phone |
| 11:27AM |
2 |
Problem compiling zaphfc |
| 9:56AM |
1 |
Pingtel and some chinese company |
| 9:00AM |
1 |
paging/intercom |
| 8:58AM |
1 |
Config for sipgate? |
| 8:47AM |
1 |
Mpg123 clarification |
| 7:56AM |
1 |
Testing null values: ast_yyerror(): syntax error |
| 7:14AM |
1 |
Choppiness/Ticking sounds over LAN |
| 7:05AM |
1 |
Asterisk as SMS Service Center |
| 7:00AM |
3 |
How to make RTP Packets NOT passing thru Asterisk? |
| 6:49AM |
1 |
Asterisk and Dial-Up ISP |
| 6:44AM |
3 |
[OT] What's changing /etc/hosts? |
| 6:44AM |
3 |
call-back example |
| 6:22AM |
0 |
Network Crashed when we try to start TDMoE |
| 5:43AM |
1 |
asterisk start |
| 5:38AM |
0 |
List-etiquette * AGAIN * |
| 4:14AM |
1 |
Channel bank for asterisk |
| 3:43AM |
0 |
Help Needed on these doubts |
| 3:23AM |
1 |
DID Terminations |
| 2:58AM |
27 |
SpanDSP |
| 2:46AM |
0 |
Adtran power consumption |
| 2:43AM |
1 |
Audio problem |
| 2:17AM |
0 |
[OT] RE: SIP / IAX provider in the Netherlands. |
| 1:58AM |
1 |
SIP / IAX provider in the Netherlands. |
| 1:54AM |
1 |
pickup any call |
| 1:45AM |
3 |
How to accept the call and without billing the caller? |
| 1:16AM |
1 |
How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work |
| 12:05AM |
2 |
Festival Installation - Asterisk 1.0-RC2 && Debian Woody |
| 12:01AM |
1 |
Hangups - SIGFPE in dsp.c |
| |
| Tuesday August 17 2004 |
| Time | Replies | Subject |
|---|
| 11:58PM |
1 |
SIP providers USA |
| 11:05PM |
1 |
Commercial deployments of Asterix |
| 10:12PM |
1 |
Asterisk and MEGACO |
| 10:06PM |
3 |
Digium Hardware Question from Newbie |
| 8:44PM |
1 |
Internal extensions giving weird static-like dialtone |
| 8:09PM |
0 |
1.0 RC2 - FXS issues - no ring, no MWI |
| 7:24PM |
3 |
[probably OT] wireless voip converter |
| 6:24PM |
1 |
budgetone 101 and buttons |
| 6:00PM |
4 |
asterisk and qsig |
| 4:44PM |
0 |
zaphfc in mode TE can't dialout (dialin is OK) |
| 3:47PM |
0 |
multiple sound cards - howto have IP phone on each ? |
| 3:03PM |
6 |
dialplan woes |
| 2:53PM |
0 |
Fax Detect over Analog |
| 12:50PM |
0 |
SpanDSP RxFax receives junk - gets Fax3Decode2d: Warning, (FakeInput): .... messages |
| 12:44PM |
1 |
Dialplan problem - incoming calls get MOH, not ringing. |
| 12:00PM |
4 |
Hunt Groups |
| 11:37AM |
6 |
couple basic questions |
| 11:08AM |
2 |
Inter-digit timers on t100 |
| 9:52AM |
1 |
BroadVOX |
| 9:00AM |
1 |
CPC on Zaptel |
| 8:44AM |
3 |
asterisk-wide variables |
| 7:57AM |
0 |
RE: RE: dialing out |
| 7:19AM |
2 |
Inbound IAX2 calls has no music on hold |
| 6:33AM |
1 |
Faxing over ulaw |
| 6:28AM |
0 |
Compiling Zaptel under Bering 1.2 |
| 6:26AM |
0 |
RE: dialing out |
| 5:19AM |
0 |
(no subject) |
| 4:34AM |
1 |
Cisco 7.2 firmware for SIP 7940/7960 release d |
| 3:36AM |
0 |
has no CRC! error messages while compiling zaptel |
| 2:57AM |
0 |
spandsp + files |
| 1:06AM |
2 |
Problems with DTMF |
| 12:52AM |
0 |
TDMoE crash the Asterisk and the Network |
| 12:44AM |
0 |
dropouts |
| |
| Monday August 16 2004 |
| Time | Replies | Subject |
|---|
| 11:28PM |
2 |
Cisco 7.2 firmware for SIP 7940/7960 released |
| 11:21PM |
0 |
re: asterisk as VM for SER |
| 11:14PM |
0 |
(no subject) |
| 10:49PM |
0 |
RC2 - RH9 |
| 9:36PM |
0 |
Re: using mysql and asterisk |
| 7:22PM |
0 |
Answer a specific channel |
| 7:13PM |
2 |
taking asterisk out of nat? |
| 5:38PM |
0 |
HELP with HFC Card, please |
| 5:28PM |
0 |
SpanDSP - Training failed error / timing problem |
| 5:22PM |
3 |
Digium TE410P and RedHat Enterprise Server 3.0 |
| 5:16PM |
1 |
RC2: Where is parking.conf and MeetMe app ? |
| 5:12PM |
0 |
mysql version of Directory app |
| 5:07PM |
0 |
How to run qozap with wcfxo and wsfxs together? |
| 4:40PM |
0 |
Music on hold question.. |
| 4:11PM |
1 |
Soft DSS for Asterisk |
| 4:04PM |
1 |
* and answering machine |
| 3:07PM |
1 |
Performance testing of asterisk |
| 2:46PM |
2 |
Problem compiling chan_sccp |
| 2:39PM |
4 |
Polycom SoundPoint IP 500/600 XML minibrowser |
| 2:21PM |
1 |
Set-up for 2 Asterisk System |
| 1:25PM |
3 |
DID Questions |
| 1:13PM |
3 |
Problems compiling chan_capi-0.3.5 |
| 12:57PM |
4 |
Avaya firmware |
| 10:43AM |
2 |
1.0 RC2 External AGI Issues |
| 9:29AM |
1 |
Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk) |
| 8:37AM |
3 |
Formatting in sip.conf...can you have 2 @ signs for register? |
| 8:04AM |
2 |
dialing out and ringing issue |
| 7:15AM |
1 |
local echo using SPA-3000 as FXO port |
| 6:25AM |
3 |
What is the best Linux for asterisk |
| 6:21AM |
2 |
randomize Dial() target |
| 5:56AM |
0 |
DTMoE crash the Asterisk and the Network |
| 5:23AM |
1 |
no hangup |
| 4:41AM |
1 |
world incoming calls |
| 4:07AM |
2 |
disable console channels |
| 3:29AM |
2 |
Call stealing |
| 3:29AM |
1 |
Using Asterisk in home environment - room with speakers&mic as IP phone ? |
| 2:35AM |
1 |
Is "Meetme" a generic term? |
| 12:54AM |
0 |
opencall.org down? |
| 12:43AM |
0 |
Help: SMS in UK |
| 12:25AM |
0 |
Differences in CDR files |
| |
| Sunday August 15 2004 |
| Time | Replies | Subject |
|---|
| 11:29PM |
2 |
consultative transfer with zaptel |
| 11:00PM |
0 |
Modified Prepaid doesn't update the balance |
| 7:52PM |
0 |
how can i config a Cisco IAD 2430 config as a sip client |
| 5:35PM |
3 |
123 Basic configuration files |
| 5:21PM |
1 |
Newbie with missing .conf files |
| 3:57PM |
1 |
Teliax TOS copied from Vonage? |
| 2:14PM |
1 |
no tones detected |
| 1:53PM |
2 |
__use_ast_pthread_create_instead__ |
| 1:50PM |
0 |
Internal Distinctive Ringing + Caller ID |
| 12:10PM |
0 |
asterisk with InPhonex? |
| 10:57AM |
3 |
Asterisk MIBS |
| 10:52AM |
2 |
GrandStream ATA286 & RC2 (was RC2 - H323 channel broken) |
| 10:40AM |
5 |
New $89 VOIP phone |
| 6:36AM |
0 |
Sip to Sip Calls via Asterisk |
| 4:17AM |
3 |
Vlan question |
| 4:02AM |
7 |
chan_oh323 loading error |
| 1:56AM |
0 |
Do you speak Czech? |
| 1:29AM |
1 |
Inbound Free World Dialup - extension not ringing? |
| |
| Saturday August 14 2004 |
| Time | Replies | Subject |
|---|
| 8:34PM |
3 |
7960 help |
| 7:25PM |
0 |
Questions on various and sundry IP phones, and cabling |
| 4:50PM |
7 |
Free MOH MP3 |
| 1:08PM |
7 |
Help - is voip good for in-house calls? |
| 12:03PM |
2 |
List traffic/Software |
| 10:22AM |
1 |
Howto remove digits from a called number |
| |
| Friday August 13 2004 |
| Time | Replies | Subject |
|---|
| 3:53PM |
0 |
HELP: BYE-request not sent to SIP-peer |
| 3:18PM |
1 |
[Q] DIDs |
| 2:09PM |
1 |
Using a TE405P to connect to an existing PBX |
| 2:00PM |
0 |
Cisco Router as Gateway for FXO Ports |
| 1:33PM |
2 |
Static on outgoing calls using either X100P or TDM400P |
| 12:52PM |
1 |
Asterisk and softswitch |
| 11:21AM |
3 |
voice choppy |
| 10:59AM |
1 |
Euro isdn caller id problem |
| 10:15AM |
6 |
Dial command problems |
| 10:14AM |
0 |
Broadvoice User hung up on voicemail |
| 9:48AM |
3 |
External MW Lamp On/Off |
| 9:46AM |
2 |
How to detect answering machine |
| 9:43AM |
3 |
Cisco 79xx series IP phones |
| 9:34AM |
0 |
*** Asterisk Summer News: Forget numbers, dial by domain! |
| 9:14AM |
0 |
SIP<->H323 "Failed to create smoother" |
| 8:58AM |
1 |
Problem with ougoing Zap calls |
| 8:45AM |
3 |
Will this ISDN card work for me? |
| 8:44AM |
1 |
voicemail email messages |
| 8:10AM |
0 |
asterisk & gnugk & voip phone |
| 6:34AM |
0 |
MGCP & RFC3149 & Swissvoice IP10 |
| 6:25AM |
1 |
SIP <->h.323 |
| 6:07AM |
1 |
SpanDSP - Training failed (convergence failed) error |
| 5:45AM |
1 |
small asterisk system |
| 5:06AM |
2 |
te410p and Telstra Onramp 10 |
| 4:58AM |
1 |
Interop RTP "Extension headers" for QOS? |
| 4:46AM |
1 |
OH.323 Dialout Problem |
| 4:35AM |
11 |
asterisk in india |
| 3:58AM |
0 |
Need help for GUI Client |
| 3:58AM |
0 |
Need help for GUI Clieny |
| 3:49AM |
0 |
Problem with grandstream devices and DTMF signalling [RESOLVED] |
| 3:34AM |
2 |
Lost 7960 time display on upgrade |
| 3:32AM |
1 |
queue name too long when sending sms over 32 chars |
| 3:12AM |
2 |
not hangup |
| 2:43AM |
1 |
Re: Send DTMF tone Like 'C' on connected call |
| 2:41AM |
1 |
Problem with grandstream devices and DTMF signalling |
| 2:10AM |
0 |
* -> Lucent PBX |
| 2:07AM |
0 |
ZAPHFC for Euro-ISDN |
| 1:17AM |
1 |
RC2 - H323 channel broken |
| 1:15AM |
0 |
incomingcall braking all |
| |
| Thursday August 12 2004 |
| Time | Replies | Subject |
|---|
| 10:27PM |
0 |
Grandstream HandyTone-486 FXO port |
| 4:59PM |
0 |
Blind Call Transfer using Sipura 3000 + aste risk |
| 4:11PM |
10 |
H323 problems |
| 3:40PM |
0 |
Calling Card App |
| 3:19PM |
2 |
outgoing ZAP cannot connect using E1 isdn |
| 2:35PM |
5 |
Question about TE405P |
| 2:14PM |
9 |
Asterisk and SER |
| 12:43PM |
0 |
Zip2 configuration via tftp? |
| 12:40PM |
0 |
Default endian for signed linear |
| 11:41AM |
0 |
Enterprise users of asterisk? |
| 11:40AM |
2 |
Interruptable SayUnixTime |
| 11:36AM |
0 |
Message lamp integration with legacy pbx -- revisited |
| 11:33AM |
2 |
X100P a winmodem? |
| 11:06AM |
0 |
Voicemail.conf not being re-read / Updates (e.g. pin numbers) ok. |
| 9:08AM |
1 |
Problem installing Software Fax SpanDSP support into Asterisk |
| 9:08AM |
0 |
New office hardware set up question. |
| 8:53AM |
1 |
CCM <->(H323) <-> * |
| 8:50AM |
5 |
::::: Pssst. Rc2! ::::::: |
| 8:37AM |
0 |
TDM400P and Adtran 616 problems |
| 8:23AM |
2 |
How Many Calls On This Config |
| 8:08AM |
1 |
Re: Asterisk-Users digest, Vol 1 #4901 - 10 msgs |
| 7:51AM |
2 |
caller id over iax |
| 7:49AM |
1 |
AgentLogin issue |
| 6:50AM |
1 |
zaptel wont compile |
| 5:45AM |
0 |
RES: Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted |
| 3:40AM |
1 |
BRI and E1 in same system |
| 3:21AM |
4 |
Problems receiving SIP calls |
| 2:47AM |
9 |
Convert Cisco 7960 to sip |
| 2:18AM |
3 |
Pulse dialing... |
| 1:36AM |
0 |
AW: Starting with digium cards |
| |
| Wednesday August 11 2004 |
| Time | Replies | Subject |
|---|
| 9:21PM |
0 |
X-Lite behind NATed ADSL router |
| 8:15PM |
1 |
Starting with digium cards |
| 7:19PM |
1 |
BroadVoice Voicemail |
| 6:40PM |
0 |
Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted (syslog) |
| 6:39PM |
0 |
Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted (tethereal) |
| 5:17PM |
1 |
persistant SABME |
| 4:56PM |
7 |
H323 call dropped when answered |
| 4:40PM |
2 |
Asterisk & MyPhoneCompany.com (aka Talk(n)) |
| 4:32PM |
1 |
Asterisk & Recommended Distro |
| 3:51PM |
1 |
Does IConnectHere still work with asterisk? |
| 3:19PM |
2 |
StanaPhone and Asterisks |
| 1:10PM |
0 |
Inband announcement of parking slot from app_parkandannounce? |
| 12:46PM |
0 |
Problem with Audio Play out in Voice-mail |
| 12:05PM |
1 |
Avaya IP Phones and Asterisk |
| 12:00PM |
1 |
Blind Call Transfer using Sipura 3000 + asterisk |
| 11:53AM |
2 |
Avaya and Asterisk |
| 11:39AM |
1 |
Grandstream Budgetone-102 client cannot register |
| 11:32AM |
0 |
Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted |
| 10:43AM |
0 |
Fedora Core 2 (kernel 2.6.5), CAPI and Fritz PCI |
| 10:24AM |
2 |
a few question about asterisk |
| 10:21AM |
1 |
CallerID Debug On Zap/POTS Channel |
| 9:47AM |
2 |
Autoattendant Configuration |
| 8:39AM |
3 |
X100P outbound only (Don't answer) |
| 8:03AM |
5 |
Asterisk and SMP |
| 7:45AM |
1 |
limit incoming calls to sip extens |
| 7:21AM |
0 |
asterisk -r and -rx questions |
| 7:17AM |
1 |
number unavailable |
| 6:16AM |
4 |
Waiting till picks up Zap line |
| 5:25AM |
0 |
zap channels and sip channels problem |
| 4:34AM |
4 |
zaphfc problems... |
| 4:22AM |
1 |
Ringing() doesn't play sound while phone is ringing |
| 4:17AM |
1 |
Analog Phones with Status Light Indicators |
| 4:09AM |
7 |
Static on outgoing calls (Quad E1) |
| 3:30AM |
1 |
stun and only one external ip |
| 3:00AM |
2 |
2.4.x-SMP vs. 2.6.x-SMP |
| 2:48AM |
1 |
external mailbox |
| 1:51AM |
1 |
is gatekeeper required? |
| 1:33AM |
0 |
Zaptel Dial Out Issues |
| 1:11AM |
0 |
No cdr entries for calls coming via chan_oh323 |
| 1:04AM |
1 |
RxFax - tiff file problem |
| 12:49AM |
1 |
can sip users login two times? |
| |
| Tuesday August 10 2004 |
| Time | Replies | Subject |
|---|
| 8:41PM |
4 |
Cisco 12sp+ and 30VIP |
| 7:37PM |
0 |
codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes from the end) |
| 5:46PM |
0 |
Re: [Asterisk-Dev] VoIP SPAM, what's next ? |
| 5:44PM |
0 |
Personal Meetme conferences; is there a better way to do this? |
| 5:17PM |
0 |
CVS version tags |
| 3:57PM |
1 |
SIP Transfers (Possibly reinvite) |
| 3:41PM |
2 |
Kernel 2.6 and zaptel data |
| 3:24PM |
0 |
UK SMS troubles |
| 3:08PM |
1 |
Parked Call Viewer Application - Freeware |
| 2:26PM |
2 |
SNOM 200 and Asterisk Woes |
| 1:33PM |
3 |
Polycom IP 500 - MWI Not Working |
| 12:30PM |
3 |
Semi-OT: Splitting a PRI into two PRI's? |
| 11:29AM |
0 |
iconnect inbound - FIXED (kinda) |
| 11:13AM |
2 |
Re: VoIP SPAM, what's next ? |
| 10:35AM |
0 |
Answer Call Waiting from Call Forward to Cell Phone |
| 10:00AM |
0 |
Intriguing * problem with voicemail signalling |
| 9:22AM |
5 |
Blocking the 'Do Not Call" List |
| 8:59AM |
2 |
Compile error H323 |
| 8:30AM |
3 |
agent login |
| 7:59AM |
1 |
HowTo test asterisk in internal network? |
| 7:55AM |
4 |
Asterisk in a DMZ |
| 7:54AM |
1 |
DTMF issues |
| 7:28AM |
0 |
Can Incomming CallerID be fowarded to a SIP phone extension? |
| 7:14AM |
2 |
WiFi phone radiation regulation? |
| 7:05AM |
1 |
[OT]Google and the Asterisk list |
| 6:37AM |
0 |
TxtCIDName returning mangled text |
| 6:36AM |
0 |
distinguish rining tone |
| 6:30AM |
0 |
Config for Asterisk with Sipgate behind Linksys Router |
| 6:29AM |
0 |
T400P modprobe/ztcfg failure |
| 5:35AM |
4 |
asterisk mirror |
| 3:59AM |
0 |
Sjphone Troubles : |
| 3:27AM |
0 |
Sound problems ..linux red hat |
| 3:19AM |
0 |
RTP-problems with voipexchange.ru |
| 3:18AM |
1 |
Firefly and *... Argh! |
| 2:26AM |
5 |
just a few newbie questions |
| 2:12AM |
11 |
CAPI call transfer |
| 1:51AM |
0 |
help me in voice jittering problem |
| 1:29AM |
6 |
Problem with EuroISDN E1 |
| 1:09AM |
0 |
h.323 channel problem: I hear nothing |
| 1:00AM |
1 |
AVM B1, chan_capi, Kernel 2.6 |
| |
| Monday August 9 2004 |
| Time | Replies | Subject |
|---|
| 11:37PM |
1 |
TE410P-RED Alarm |
| 10:37PM |
2 |
831 Santa Cruz/Watsoncille, Calif. DIDs |
| 10:15PM |
0 |
introduced Agents and * stops answering calls |
| 4:23PM |
2 |
CVS download |
| 3:41PM |
1 |
Inbound Call Errors... |
| 2:58PM |
2 |
Snom Intercom |
| 2:42PM |
2 |
Application asterisk uses obsolete OSS audio interface |
| 2:41PM |
1 |
called and callers buttons on bt100 |
| 1:52PM |
0 |
H323 under asterisk RC1 ? |
| 1:33PM |
1 |
quadBRI + FAX |
| 1:15PM |
3 |
AbsoluteTimeout Inside A Macro |
| 1:06PM |
0 |
uniden phones |
| 12:12PM |
0 |
inbound/outbound trunk groups |
| 12:07PM |
3 |
Fedora FC2 and Zaptel (Torisa) |
| 11:51AM |
0 |
Strange H323 problem |
| 9:50AM |
2 |
Call File Routing |
| 9:18AM |
5 |
Questionaire : |
| 8:24AM |
2 |
ChangeMonitor syntax |
| 7:51AM |
0 |
New feature request |
| 6:26AM |
1 |
Click to Call |
| 6:25AM |
1 |
asterisk with H.323 phone |
| 6:09AM |
0 |
RC1 - callparking |
| 5:57AM |
0 |
sip endpoint not ringing |
| 5:47AM |
1 |
How do folks handle NAT routing? |
| 4:42AM |
0 |
Which is PCI HDLC Chip used for TE405P / TE410P / E100P ? |
| 4:37AM |
0 |
Some Errors on Asterisk server |
| 4:31AM |
0 |
Some Errors on Asterisk |
| 4:26AM |
0 |
Variables In a Context |
| 4:25AM |
1 |
h323 direkt call instead over GK |
| 4:07AM |
2 |
Sound file quality |
| 3:18AM |
1 |
Some Error on Asterisk.... |
| 2:51AM |
0 |
traffic termination around the globe? |
| 1:56AM |
0 |
e164.lu |
| 1:06AM |
0 |
FW: problems with asterisk and the IAX protocol |
| |
| Sunday August 8 2004 |
| Time | Replies | Subject |
|---|
| 11:36PM |
1 |
Howto configure TE410P card and channels |
| 10:29PM |
0 |
Extension Status on different phone |
| 10:00PM |
0 |
Straight to VoiceMail |
| 8:52PM |
3 |
iconnect inbound - so do we know how to fix it |
| 6:58PM |
2 |
pbx answers after answering from analog phone |
| 3:17PM |
0 |
Douglas Telecom & Asterisk |
| 1:28PM |
0 |
stable-RC1 |
| 12:44PM |
1 |
asterisk-update script - and the script - Fixed typo |
| 12:12PM |
1 |
asterisk-update script - and the script |
| 12:04PM |
2 |
asterisk-update script |
| 11:26AM |
0 |
Zaptel.conf and Zapata.conf for TDM12B |
| 11:14AM |
1 |
truncated extensions |
| 8:28AM |
6 |
Voicepulse problems? |
| 6:29AM |
1 |
No Sound and Jungle: |
| 3:03AM |
2 |
System Reqirements HELP |
| |
| Saturday August 7 2004 |
| Time | Replies | Subject |
|---|
| 11:02PM |
0 |
SOLVED: 100% cpu usage causes big problems |
| 9:53PM |
1 |
WARNING[1264581056] |
| 9:39PM |
0 |
voice mail greeting not updating? |
| 6:07PM |
2 |
Asterisk : No Sound No Dial |
| 5:41PM |
4 |
Generic X100P setup issues |
| 5:04PM |
1 |
Mediatrix 1204 - Error: Operation not permitted |
| 4:59PM |
1 |
Mediatrix 1204, where I can get the last firmware and mib file ? |
| 3:59PM |
0 |
H.323 IVR with avaya multivantage. |
| 3:47PM |
0 |
RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio) (fwd) |
| 3:03PM |
0 |
Where to get latest SIP bootrom/firmware for Polycom IP500 phones? |
| 10:32AM |
3 |
Message waiting |
| 10:00AM |
0 |
Getting a modem connection working via Asterisk |
| 9:56AM |
1 |
Lots of FXS ports / Channel Bank ? |
| 9:51AM |
0 |
call for help from uip200 users |
| 7:55AM |
2 |
astcc help |
| 6:16AM |
2 |
Howto do something after a Dial command is Answered? |
| 6:06AM |
0 |
(no subject) |
| 4:33AM |
0 |
Video Calls between SIP and H.323 |
| 2:39AM |
1 |
res_config_odbc not working |
| 2:22AM |
0 |
Asterisk and Douglas Telecom |
| 1:15AM |
0 |
Re: Invalid data ( 4 bytes at the end) message - from g729codec |
| 1:03AM |
1 |
Confused --> Hardware specs |
| 12:42AM |
2 |
Asterisk : No Sound Issues |
| |
| Friday August 6 2004 |
| Time | Replies | Subject |
|---|
| 10:41PM |
0 |
iaxtel, asterisk, and sipura 1000 am having trouble with codecs |
| 9:29PM |
0 |
Asterisk, channel banks and SDH |
| 8:38PM |
1 |
Asterisk and Cisco Call Manager. |
| 7:03PM |
1 |
oem x100p undefined symbol ast_get_txt |
| 6:27PM |
1 |
Asterisk Dry Run |
| 5:51PM |
0 |
Need help on installing an Asterisk PBX |
| 4:38PM |
1 |
Problems loading chan_h323 on Opteron 64 bit |
| 4:31PM |
1 |
redhat 9 and oh323 |
| 4:07PM |
1 |
termcapsupport not found |
| 3:59PM |
0 |
no mail sent on voice message |
| 2:05PM |
1 |
Interesting catalog: Viking Electronics |
| 2:03PM |
2 |
RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio) |
| 1:55PM |
0 |
t100 error on FC2 |
| 12:56PM |
0 |
ShoreTel Phones |
| 12:37PM |
1 |
Zaptel CVS HEAD 08-05-2004 problems |
| 11:27AM |
2 |
FXO Problems |
| 10:55AM |
2 |
Inbound not working with iconnect |
| 10:06AM |
2 |
Asterisk not starting |
| 9:53AM |
3 |
ASTERISK AND 120 CONCURRENT CALLS |
| 8:41AM |
0 |
Asterisk as SIP proxy? |
| 8:19AM |
3 |
tdm400p lockups |
| 8:14AM |
3 |
E1 monochannel :-( |
| 7:02AM |
2 |
Difficulty evaluating the return value of PlayBack (or any other extensions.conf command |
| 5:16AM |
2 |
DTMF after answer |
| 4:33AM |
0 |
RE: Zaptel X100P Kernel Panic |
| 4:12AM |
2 |
Lots of Echo with SIP -> Asterisk -> PSTN |
| 4:00AM |
0 |
ZAPTEL & ZAPATA |
| 3:12AM |
0 |
iax->asterisk->capi isdn; crackle overlay after about 1 minute |
| 3:09AM |
0 |
Entry point & Minimal requirements for Linux Device Driver and connection with Asterisk IP PBX |
| 2:48AM |
0 |
Microsoft RTC Authentication |
| 2:48AM |
0 |
Re: Invalid data ( 4 bytes at the end) message - g729codec |
| 2:10AM |
0 |
Urgent help with Sip <------> H323 on FREEBSD |
| |
| Thursday August 5 2004 |
| Time | Replies | Subject |
|---|
| 11:58PM |
1 |
Sip dialback |
| 9:46PM |
1 |
iptables, Cisco 7960 and TFTP |
| 4:38PM |
1 |
users |
| 3:49PM |
1 |
Skinny and CISCO 7905G |
| 3:01PM |
1 |
is asterisk and/or spandsp what I need for integration with Cisco? |
| 2:37PM |
0 |
Strange message, and one-way audio between sip and H.323 |
| 2:10PM |
0 |
Re: Asterisk-Users digest, Vol 1 #4831 - 4 msgs |
| 1:01PM |
0 |
ASTRICON 2004 |
| 12:41PM |
1 |
iConnectHere and CallerId |
| 12:26PM |
4 |
<<< MEETME_AGI_BACKGROUND inside MEET ME>>> |
| 12:08PM |
0 |
Asterisk used like SIP to h323 convertor |
| 11:56AM |
0 |
PRI Errors... Ouch |
| 11:03AM |
2 |
new bounty for modifying calling card application to mysql |
| 10:44AM |
1 |
transfering incoming message from app_queue |
| 10:31AM |
5 |
Anyone use AdvancedVOIP ? |
| 9:21AM |
0 |
problems with asterisk and the IAX protocol |
| 8:54AM |
2 |
PRI protocol question... |
| 8:33AM |
1 |
Advice on possible set-up |
| 8:27AM |
0 |
Bridging Calls |
| 8:27AM |
3 |
Avaya/Lucent Definity -> Asterisk interop question |
| 7:24AM |
2 |
X100P Kernel Panic |
| 6:57AM |
2 |
shared voicemail |
| 4:30AM |
1 |
h323 gnugk to h323 asterisk and then to endpoint |
| 4:25AM |
2 |
personal voicemail |
| 3:27AM |
2 |
Call Transfer Problems with Grandstream Budgetone 100 Phone |
| 2:00AM |
3 |
QuadBri in NT mode not working. |
| 1:30AM |
1 |
AW: Integrating an old PBX with Asterisk |
| |
| Wednesday August 4 2004 |
| Time | Replies | Subject |
|---|
| 9:11PM |
4 |
Using answering machine in my phone |
| 7:11PM |
0 |
New Head Appears to Break SIP to iConnect |
| 7:10PM |
1 |
how to integrate an addon module to asterisk |
| 6:47PM |
0 |
Configure E1 PRI |
| 4:19PM |
0 |
Can't get T or t option to work with two IAX2 channels. |
| 4:02PM |
4 |
FCC Rules VoIP Must Be Tappable |
| 3:56PM |
2 |
Color in console |
| 2:56PM |
1 |
Identifying which call an event belongs to |
| 2:14PM |
2 |
Snom 200 Programmable Keys |
| 2:10PM |
0 |
Integrating an old PBX with Asterisk |
| 12:54PM |
5 |
H323 Call Dropping |
| 12:24PM |
0 |
calling card on Zap outbound |
| 12:22PM |
2 |
PSTN Access Providers for Asterisk |
| 12:07PM |
0 |
Zultys ZIP2 |
| 11:47AM |
3 |
Auto-attendant with an IP trunk |
| 11:36AM |
2 |
Asterisk & ISDN-card |
| 11:34AM |
4 |
Asterisk Integration |
| 11:21AM |
2 |
Get MWI from Telco's voicemail |
| 9:50AM |
1 |
BT100 bad handset? |
| 8:24AM |
0 |
Bogus patent applications (was: [RANT] ... Broadvoice) |
| 7:35AM |
1 |
Barge in on to agents conversation |
| 6:46AM |
4 |
Problems with E100P |
| 6:34AM |
0 |
Astetrisk connectet to PBX |
| 6:22AM |
1 |
SIP pickupgroup |
| 6:05AM |
3 |
No incoming audio on incoming SIP calls |
| 6:03AM |
2 |
2 sip servers |
| 6:02AM |
0 |
Congested link |
| 5:48AM |
1 |
Who is calling me ? |
| 5:18AM |
3 |
German sounds |
| 4:51AM |
1 |
PRI/H323 gateway |
| 2:26AM |
5 |
Asterisk QOS working perfect using sveasoft 3.11g |
| 2:18AM |
4 |
rxfax killed asterisk |
| 12:54AM |
1 |
capturing a call |
| 12:35AM |
3 |
Cisco SIP Phone 7960 & DTMF Problem |
| 12:24AM |
1 |
about sip.conf |
| 12:04AM |
0 |
Call pickup (group) |
| |
| Tuesday August 3 2004 |
| Time | Replies | Subject |
|---|
| 10:06PM |
1 |
CAC AB1 and Asterisk |
| 9:04PM |
0 |
Very decent book - "VoIP Telephony with Asterisk" |
| 8:46PM |
3 |
Re: problems with'#' transfer after hold |
| 7:50PM |
1 |
CID Blocked vs. Unknown |
| 6:37PM |
1 |
Logging into Multiple Call Queues on two * Servers and Voice Mail option. |
| 3:48PM |
2 |
problems with'#' transfer after hold... |
| 3:19PM |
1 |
UK VoIP-PSTN gateway recommendations |
| 3:09PM |
0 |
Can Zap detect line is already off-hook? |
| 2:17PM |
2 |
SPA-3000 as a regular Asterisk FXO device? |
| 2:14PM |
4 |
After RC1 upgrade, temporary loss of voice |
| 1:38PM |
1 |
instable Modem-Module in CVS ? |
| 1:36PM |
0 |
snom 200 - custom melody |
| 1:09PM |
1 |
Play audio into meetme conference? |
| 11:48AM |
0 |
Asterisk and RT |
| 11:36AM |
2 |
Integration with Altigen |
| 11:03AM |
0 |
Asterisk Tapi Driver for Windows |
| 9:07AM |
2 |
VoIP experiences with Cable and DSL |
| 9:01AM |
0 |
ZyXEL 2000w In Call Menu/Hold configs |
| 8:53AM |
1 |
Analog channel stays offhook |
| 8:40AM |
3 |
PRI Call Redirection / Transfers |
| 8:22AM |
1 |
Emailing phone messages? |
| 8:14AM |
0 |
DID Trunk |
| 7:53AM |
1 |
astguiclient: blank php pages |
| 7:27AM |
6 |
features.conf |
| 7:06AM |
0 |
Configure Makefile to run with older iax Protocol |
| 6:48AM |
0 |
VON Magazine article. |
| 6:03AM |
0 |
avm c4: DISCONNECT_IND ID=001 #0x0193 LEN=0014 |
| 5:47AM |
0 |
Fw: Digium FXO Interfaces don't support groundstart??? |
| 5:21AM |
1 |
Any small colleges/universities using PBX or Voicemail? |
| 3:30AM |
1 |
bearer Capability |
| 3:24AM |
3 |
Called ID in Australia |
| 3:09AM |
0 |
Echo problems with mISDN? |
| 1:43AM |
1 |
Using Clustering/TDMoE |
| 12:52AM |
0 |
OH323 not dial Modem[i4l]/g1 |
| |
| Monday August 2 2004 |
| Time | Replies | Subject |
|---|
| 11:08PM |
5 |
Making asterisk distributed |
| 8:55PM |
1 |
help with digium E1 card |
| 8:23PM |
2 |
Cisco PRI no CallerID |
| 8:21PM |
0 |
Help with Cisco PRI |
| 7:55PM |
1 |
G729 or GSM |
| 7:52PM |
0 |
Fax on demand |
| 5:01PM |
1 |
MPG123, Music On Hold and Variable Bit Rate |
| 4:13PM |
0 |
One way voice with following error |
| 3:40PM |
9 |
asterisk+radius |
| 3:04PM |
0 |
Help with ParkAndAnnounce command |
| 2:23PM |
2 |
CallPres screening DDI |
| 2:01PM |
4 |
First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs? |
| 1:23PM |
1 |
asterisk call parking + SNOM lighted buttons? |
| 1:07PM |
0 |
Pre-release of OSX GUI tool to add extensions and phones |
| 12:54PM |
1 |
Asterisk as Front-End for Artisoft Televantage 6 |
| 11:06AM |
7 |
System Requirements |
| 10:09AM |
2 |
New CVS and Sipuras |
| 9:32AM |
3 |
App.c |
| 9:28AM |
0 |
bri-stuff.0.1.0-RC2k + hfc card: dropouts on IAX2 & MP3Player quits on streams |
| 9:05AM |
1 |
avm c4, ptmp |
| 8:49AM |
3 |
How STUN work? |
| 8:41AM |
1 |
Performance of queues |
| 8:30AM |
0 |
Stripping characters from SIP dial strings |
| 8:24AM |
1 |
Vonage catastrophic failure... |
| 8:21AM |
0 |
Multiple Line SIP Phones? |
| 8:20AM |
1 |
(no subject) |
| 8:13AM |
1 |
Selling asterisk-based solutions |
| 8:09AM |
2 |
Cisco MC3810 |
| 6:41AM |
0 |
Fwd: Help with Quicknet PhoneJack@Asterisk |
| 6:40AM |
0 |
Clustering in Asterisk |
| 6:21AM |
1 |
DID's in the Czech Republic |
| 6:09AM |
0 |
Help with Quicknet PhoneJack@Asterisk |
| 5:56AM |
1 |
Win2000 DUN via Asterisk (Is it possible) |
| 4:39AM |
1 |
RC1 - error message : Request to schedule in the past |
| 3:48AM |
4 |
CDR with MySQL and Asterisk PID File |
| 2:11AM |
0 |
detect FAX terminal |
| 2:04AM |
0 |
h.323 debug |
| |
| Sunday August 1 2004 |
| Time | Replies | Subject |
|---|
| 9:53PM |
0 |
About CDR billsec when used TE410P |
| 5:25PM |
1 |
SIP- PSTN Gateway |
| 2:57PM |
0 |
Message Lamps across IAX connected switches. |
| 12:13PM |
1 |
Zaphfc CallerID problem... |
| 11:32AM |
0 |
Writing messages to ISDN phone displays? |
| 11:28AM |
2 |
Parking & SIP Phones |
| 10:20AM |
1 |
Grandstream Message Waiting light |
| 7:47AM |
1 |
Snom 220 |
| 7:13AM |
1 |
cdr record for recording location |
| 6:32AM |
1 |
distinctive ring on SNOM 200 |
| 6:09AM |
1 |
Does anyone know how to use the DND feature oc Cisco 7940/7960 |
| 3:17AM |
2 |
Cisco 7960 backlight update and prices. |
| 2:41AM |
2 |
Zaptel - incoming delay |
| 12:55AM |
1 |
X100P wants to use g2 |