| Wednesday February 28 2007 |
| Time | Replies | Subject |
|---|
| 9:17PM |
1 |
voicemail advanced options problem with mysql datbase |
| 6:26PM |
0 |
Using ooh323 with Gatekeeper controlled dialling |
| 5:14PM |
4 |
Help Needed: Can't make "local" calls on a brand new PRI |
| 4:45PM |
0 |
ooh323 patch: fix for Cisco IOS Gatekeeper re-registration problem |
| 4:06PM |
1 |
AEL & Blacklist question |
| 3:45PM |
1 |
Paid support offered |
| 3:43PM |
3 |
read write or only read fields in cdr? |
| 3:33PM |
1 |
OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED |
| 3:32PM |
1 |
extensions.conf & sccp.conf howto call external number |
| 3:18PM |
2 |
Newbie extensions.conf question |
| 3:03PM |
2 |
No Caller ID Name PRI NI2 |
| 2:15PM |
0 |
Occasional SMS problem |
| 1:57PM |
0 |
Send DTMF's before the call is answered |
| 1:56PM |
1 |
1.4 lost internet internal phones loose registration |
| 1:45PM |
3 |
Newbie Planning Help |
| 1:33PM |
4 |
Help: CallerID Name not being sent on outbound PRI trunk |
| 1:23PM |
2 |
this i a test |
| 11:56AM |
1 |
Run-away Asterisk |
| 10:17AM |
2 |
Changing from email address for vociemail.conf |
| 9:55AM |
0 |
seeing DTMF passed to Voicemail |
| 9:54AM |
3 |
Registrations, how many is too many? |
| 9:31AM |
5 |
about bluetooth channel |
| 9:18AM |
3 |
multiple phones registered for the same user |
| 9:08AM |
1 |
h323 how to set it up? |
| 9:00AM |
1 |
Timing, use analog card, ZT Dummy etc. |
| 5:05AM |
1 |
groups |
| 3:22AM |
7 |
Problem with TE212P |
| 3:12AM |
0 |
Asterisk 1.4 does not load chan_vpb.so |
| |
| Tuesday February 27 2007 |
| Time | Replies | Subject |
|---|
| 7:46PM |
0 |
Limiting call volume |
| 7:15PM |
1 |
Help understanding SIP SHOW CHANNELS |
| 7:05PM |
2 |
No sound with Playback() or Background() |
| 6:34PM |
2 |
Voice mail is not giving unavailable or busy prompts |
| 6:08PM |
1 |
Not registering Port with VSP |
| 6:02PM |
5 |
TE110P: Error ==> Asterisk died with code 1. |
| 4:12PM |
2 |
jittery audio in voiceprompts |
| 3:58PM |
1 |
Quintum configuration ASM200 Analog 2 tenor port |
| 3:18PM |
2 |
Saving Dialplan in CLI |
| 2:13PM |
1 |
SER / IAX solution |
| 1:26PM |
2 |
TE212P on FC6 - stack overflow? |
| 1:23PM |
0 |
asterisk CDR and mysql |
| 11:01AM |
1 |
Net-talk |
| 10:50AM |
1 |
H323-to-SIP proxy |
| 10:37AM |
0 |
rtc: lost some interrupts at 1024Hz |
| 9:59AM |
2 |
Polycom Firmware |
| 9:54AM |
2 |
running asterisk through cellphone |
| 9:40AM |
0 |
sip.conf "limitonpeers=yes" in asterisk 1.4 |
| 8:54AM |
0 |
Grandstream SYSLOG error codes |
| 8:44AM |
1 |
Billing Telephone Number (BTN) |
| 8:37AM |
0 |
call-limit in 1.2 HEAD |
| 8:06AM |
1 |
Do I understand GROUPs correctly? |
| 7:23AM |
2 |
RES: asterisk-users Digest, Vol 31, Issue 115 |
| 6:33AM |
1 |
NetFilter (IPTables) |
| 6:23AM |
1 |
VLAN vs RealLan |
| 6:00AM |
2 |
Authentication Command |
| 5:31AM |
0 |
OutBound Proxy calls failing |
| 5:12AM |
1 |
FW: Cisco 7960 |
| 5:07AM |
0 |
Forbidden - wrong password on authentication for INVITE |
| 5:02AM |
0 |
Autentication |
| 3:16AM |
0 |
mgcp codec problem about ulaw |
| 3:03AM |
1 |
Cisco 7960 |
| 2:52AM |
1 |
Error Message. |
| 2:33AM |
2 |
AJAM..is a BUG? |
| 12:36AM |
1 |
chan_sip.c:10173 handle_response: Dont know how to handle a 202 Accepted respons |
| |
| Monday February 26 2007 |
| Time | Replies | Subject |
|---|
| 10:11PM |
1 |
To use asterisk or proprietary hardware, that is the questio |
| 9:06PM |
7 |
How to get values of local channels context |
| 8:01PM |
2 |
XM Radio Stream to Asterisk |
| 7:23PM |
3 |
Asterisk -> Streaming Audio Bridge |
| 6:20PM |
3 |
Playback uses channel's language, background doesn't |
| 5:20PM |
1 |
Asterisk to Asterisk SIP Trunk and CallerID |
| 5:03PM |
1 |
Digium S101I echo - how to control it |
| 2:31PM |
1 |
Caller ID not getting to analog extensions |
| 2:06PM |
1 |
deprecated - CLI help vs. source code |
| 11:48AM |
3 |
Yellow or Red alarm on TE110P ???? |
| 11:45AM |
2 |
Ex-Girlfriend syntax and RealTime Extensions |
| 8:31AM |
0 |
AstriCon Europe 2007 |
| 7:42AM |
3 |
How set CallerID via Macro or something |
| 7:15AM |
0 |
Out Proxy Call |
| 4:24AM |
0 |
IAX/SIP Inter Asterisk Transfer |
| 3:58AM |
2 |
SetCIDNum is not available on 1.4svn |
| 2:02AM |
0 |
Asterisk TE110P Hipath 3750 |
| 12:47AM |
1 |
Newbie would like some planning advice. |
| |
| Sunday February 25 2007 |
| Time | Replies | Subject |
|---|
| 7:20PM |
7 |
Sending Email From the dialplan |
| 1:32PM |
1 |
Marks SNMP HowTo |
| 1:08PM |
2 |
Dialling ZAP channel from analogue |
| 10:28AM |
0 |
Looking for automatic sound announce device |
| 10:09AM |
2 |
freecall.com - has anybody tried it? |
| 6:52AM |
0 |
VoiceMailMain plays oldest message first |
| |
| Saturday February 24 2007 |
| Time | Replies | Subject |
|---|
| 8:22PM |
1 |
ERROR: relation "cc_ui_authen" does not exist |
| 8:04PM |
0 |
1.4.0 spews garbage on CLI, crashes |
| 7:35PM |
5 |
Sending SMS |
| 5:29PM |
0 |
Wildcard Testing |
| 5:12PM |
0 |
Voicemeup @ 0.008 per minute USA /CAN |
| 10:53AM |
8 |
To use asterisk or proprietary hardware, that is the question |
| 8:05AM |
6 |
dial a pager and enter DTMF |
| 7:22AM |
0 |
Analogue Phone Problems on a TDM11B |
| 3:31AM |
1 |
Somebody can help me? |
| 1:44AM |
0 |
Call was hangup when LIMIT_WARNING_FILE was playing |
| |
| Friday February 23 2007 |
| Time | Replies | Subject |
|---|
| 11:53PM |
1 |
Accessible documentation vor blind users |
| 10:16PM |
2 |
Any way to get rid of AEL created contexts? |
| 7:14PM |
1 |
ReceiveText()? |
| 5:20PM |
0 |
Asterisk callshops |
| 4:23PM |
1 |
H extension don't work with parked calls |
| 4:17PM |
0 |
New Community Blogs |
| 3:14PM |
0 |
MusiconHold |
| 3:14PM |
2 |
Voice mail server |
| 2:48PM |
2 |
GSM cleanup (pops, clicks and static) |
| 1:44PM |
1 |
asterisk |
| 1:11PM |
1 |
Queue Macro Problem |
| 1:00PM |
0 |
SOLVED: Call forwarding and 1.2.x |
| 12:57PM |
3 |
cisco sip firmware update for cisco 7970 |
| 12:05PM |
2 |
SIP Test |
| 9:55AM |
1 |
Polycom SIP 501 Transfer Question |
| 9:21AM |
1 |
SLA more than 100% ? |
| 9:07AM |
1 |
ooh323 hang up after the call is answered |
| 6:06AM |
2 |
Dial() command h and H options for SIP channel |
| 4:42AM |
1 |
CWI, call-limit and incominglimit |
| 3:39AM |
1 |
Asterisk and DTMF |
| 3:29AM |
0 |
Job offer near Los Angeles |
| 2:44AM |
0 |
Have an AGI script as a queue member |
| 1:52AM |
1 |
peer-to-peer RTP trouble in SIP |
| 1:13AM |
3 |
Sellvoip configuration....Please Help!!!! |
| 12:41AM |
1 |
default "insecure" setting |
| |
| Thursday February 22 2007 |
| Time | Replies | Subject |
|---|
| 10:52PM |
1 |
"Trunk" version of Asterisk? |
| 10:10PM |
0 |
choppy playback |
| 6:27PM |
0 |
Application RealTime |
| 4:24PM |
0 |
Strange Noise |
| 3:13PM |
3 |
Argentine Asterisk Wiki |
| 2:32PM |
1 |
asterisk with TCP transport |
| 2:23PM |
4 |
Possible to light up a LED on Snom phones? |
| 2:20PM |
3 |
upgrading from A101 to....A102 |
| 2:08PM |
0 |
Passing call status/progress between protocols |
| 12:57PM |
2 |
AG-188 |
| 9:30AM |
3 |
New tutorial: DTMF tone detection |
| 8:46AM |
0 |
Asterisk - VoiceGenie IVR |
| 8:38AM |
1 |
Lastest SVN (1.4) and realtime call limit |
| 7:47AM |
1 |
Asternic Flash Panel |
| 7:39AM |
2 |
Configuring Asterisk. |
| 6:58AM |
0 |
SIP RE-INVITE after an Answer() |
| 6:48AM |
0 |
Polycom IP 601 help needed |
| 6:40AM |
0 |
RE: Asterisk to Cisco's Rescue...again...AuthenticateLD Calls |
| 5:58AM |
6 |
Asterisk and Cisco PRI gateway config |
| 5:22AM |
2 |
What means: Request to schedule in the past?!?! |
| 5:15AM |
0 |
Newbie: registration failure (fwd) |
| 5:02AM |
1 |
GotoIf DURATION |
| 4:49AM |
1 |
VoIP Internet Server |
| 4:28AM |
3 |
An ISDN ISPBX to Voip Gateway?? |
| 3:36AM |
2 |
b410p + fax (echo cancellation) |
| 1:32AM |
2 |
fax support |
| 1:23AM |
1 |
Answer() command? |
| 1:22AM |
0 |
cannot get whole DNID with ISDN line |
| 1:20AM |
3 |
queue information into db |
| 1:00AM |
0 |
Destroy a zombie sip channel |
| |
| Wednesday February 21 2007 |
| Time | Replies | Subject |
|---|
| 8:43PM |
2 |
SIP response 603 driving me nuts |
| 6:01PM |
1 |
Asterisk to Cisco's Rescue...again...Authenticate LD Calls |
| 3:52PM |
3 |
Snom 320 password |
| 3:05PM |
3 |
SIP 406 error - cause? |
| 2:27PM |
0 |
monitoring cluster-based call-centers |
| 2:07PM |
1 |
How to separate outgoing extens from the contexts from sip.conf? |
| 1:58PM |
2 |
How does Asterisk use SIP info command |
| 1:19PM |
0 |
Problem on Asterisk to Register lines for out/in calls |
| 11:45AM |
1 |
Problem Installing Zaptel |
| 11:26AM |
3 |
Trixbox -- ACPI and IO-APIC? |
| 11:16AM |
1 |
Monitoring which users are online in realtime |
| 10:17AM |
0 |
Zaptel 1.2.14 Released |
| 9:16AM |
0 |
Trixbox ;TE110P ;DELL OPTIPLEX GX240 |
| 9:01AM |
1 |
HELP!! Dropping calls on Bridge |
| 8:52AM |
3 |
Zaptel 1.4.0 |
| 8:23AM |
0 |
Using asterisk with vpb driver (OpenLine4) |
| 8:04AM |
0 |
jingle + asterisk 1.4 |
| 6:44AM |
0 |
Trunk - strange behavior |
| 6:26AM |
0 |
IAX Realtime - show peers works? |
| 5:50AM |
0 |
how to detect who starts one touch recording |
| 4:54AM |
1 |
Channels hanging when SIP phone gets reset during call |
| 4:39AM |
0 |
Dialout option problem in voicemail.conf |
| 4:12AM |
1 |
AGI DTMF Problem |
| 2:12AM |
0 |
How to read channel occupation from PRI INTENSE DEBUG ? |
| 1:48AM |
0 |
Hint a sip account |
| 12:05AM |
0 |
How to read "pri intense debug span" data ? |
| |
| Tuesday February 20 2007 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
How to repeat pri show span and zap show channel commands |
| 11:49PM |
0 |
Open Source VOIP at Toronto Conference |
| 11:40PM |
2 |
Help! How to get ANSWEREDTIME after DIAL a ZAP channel? |
| 9:33PM |
0 |
Asterisk behind OpenSER - Getting SIP reinvites to work with an ITSP |
| 8:43PM |
1 |
trixbox not sending ring back to caller |
| 8:22PM |
2 |
Asterisk CDR MySQL |
| 6:05PM |
6 |
They ignore my DTMF! |
| 3:10PM |
4 |
Passing a variable from one Asterisk box to another |
| 2:25PM |
0 |
Can't get ANSWEREDTIME after hangup using ZAP |
| 2:02PM |
0 |
Tipping Point IPS blocking Asterisk SIP quaility messages |
| 1:35PM |
1 |
CDR reports short call length |
| 1:18PM |
2 |
Rules about congestion |
| 11:13AM |
0 |
B410P - Please an advise |
| 10:50AM |
3 |
analog channels calling out not detect DTMF |
| 10:22AM |
3 |
Asterisk / ACT CRM Integration |
| 7:04AM |
0 |
unwanted chanspy: strange behaviour |
| 5:15AM |
1 |
Asterisk-1.2.10 not releasing SIP sessions |
| 4:01AM |
6 |
FW: zaptel 1.4.0 on Fedora Core 6 x86_64 |
| 3:37AM |
1 |
Agents busy in queue |
| 3:08AM |
2 |
Mask the caller-ID |
| 2:39AM |
0 |
problem with dialout option in voicemail.conf |
| 1:24AM |
0 |
Communication between servers |
| |
| Monday February 19 2007 |
| Time | Replies | Subject |
|---|
| 9:47PM |
2 |
sip to sip ? |
| 8:50PM |
1 |
zaptel 1.4 svn fails to compile (xbus-core.c) |
| 6:11PM |
1 |
SIP interface status and calllimit |
| 5:05PM |
2 |
Auto load of zap drivers |
| 2:37PM |
1 |
Kernel and zaptel versions |
| 1:00PM |
1 |
Setting Caller-ID / Point Codes |
| 12:45PM |
1 |
Good 4 Port PSTN Gateway |
| 12:39PM |
6 |
Open CallerID Database? |
| 12:32PM |
2 |
UTStarcom F1000 - WLAN connection unreliable |
| 12:23PM |
0 |
Cisco 7941/7961 w/Firmware 8.2.1 and NAT |
| 10:26AM |
1 |
Attended Transfer with snom phones |
| 9:11AM |
0 |
Re: asterisk-users Digest, Vol 31, Issue 80 |
| 8:41AM |
0 |
SIP resigtrations and OpenSer |
| 7:55AM |
2 |
Transfer Caller ID |
| 6:53AM |
1 |
Asterisk with Radius users authentication |
| 5:37AM |
1 |
Asterisk Inbound Problem |
| 4:18AM |
1 |
Asterisk PPPD with analog lines |
| 4:03AM |
0 |
Asterisk and a modem pool. |
| 3:14AM |
1 |
Problems with CentOS ztdummy kernel 2.6 |
| |
| Sunday February 18 2007 |
| Time | Replies | Subject |
|---|
| 1:55PM |
1 |
HT488 doesn't disconnect FXO |
| 12:08PM |
5 |
Looking for starting point? |
| 9:37AM |
0 |
Asterisk consultant needed in Paris |
| 5:09AM |
3 |
chan_sip.c:1968 create_addr: No such host: |
| |
| Saturday February 17 2007 |
| Time | Replies | Subject |
|---|
| 7:09PM |
1 |
Unable to start Asterisk 1.4 on CentOS 4.4 (installed from ATrpms) |
| 3:01PM |
1 |
Confederated SIP service. |
| 1:15PM |
0 |
dialing out with TDM2402E card other system not grabbing DTMF digits |
| 9:22AM |
1 |
Weird problem in wctdm24xxp driver |
| 9:15AM |
0 |
Fwd: musiconhold.conf in realtime |
| 8:17AM |
0 |
musiconhold.conf in realtime |
| 12:46AM |
3 |
Problem with busydetect and cell phones |
| |
| Friday February 16 2007 |
| Time | Replies | Subject |
|---|
| 10:41PM |
0 |
manager command queue... |
| 8:28PM |
0 |
IAX vs SIP - Getting Asterisk out of the media path |
| 6:59PM |
1 |
X100P ring detection failure |
| 4:11PM |
0 |
F1000 web configure |
| 4:04PM |
0 |
How to configure Asterisk queue with Vonage account? |
| 3:02PM |
3 |
Does Asterisk support DNIS? |
| 2:36PM |
0 |
Help needed to server code on Vxworks |
| 2:06PM |
1 |
DNIS on T1 channels |
| 1:31PM |
1 |
iaxmodem - fax tone? |
| 1:20PM |
1 |
Open Source VoIP at FOSDEM |
| 12:23PM |
0 |
How can I use 'Asterisk Manager API' to hold and retrive an active call? |
| 12:06PM |
2 |
Experiences with FoneBridge2 / TDMoE? |
| 12:01PM |
1 |
Sangoma A101 install problem |
| 11:26AM |
0 |
sangoma 102 and CAB-E1-RJ45BNC |
| 10:56AM |
1 |
Distinct call permissions for each user |
| 10:31AM |
1 |
MixMonitor & RingBack Tone Issue |
| 9:50AM |
5 |
FW: Problem Transferring Direct to Voicemail |
| 7:13AM |
7 |
Summary of "Trixbox vs. custom install" |
| 5:55AM |
2 |
Jabber/Asterisk Integration |
| 5:28AM |
2 |
Asterisk callerID |
| 4:23AM |
2 |
freepbx with ASTERISK 1.4 |
| 3:37AM |
1 |
64 bit HPEC modules available? |
| 1:23AM |
2 |
Pickup application |
| 12:43AM |
0 |
AstLinux + RT PREEMPT |
| |
| Thursday February 15 2007 |
| Time | Replies | Subject |
|---|
| 10:48PM |
0 |
SIP/2.0 404 Not Found |
| 10:39PM |
6 |
PSTN Calls from SI.P: buzzing and pops |
| 9:14PM |
0 |
SIP Redirect from Asterisk behind a NAT |
| 3:34PM |
0 |
New AstLinux Branch: RT PREEMPT ("realtime" Linux) - Looking for testers |
| 2:42PM |
1 |
Meetme - is this statement from the Wiki still true? |
| 2:19PM |
2 |
Native format prompts |
| 2:13PM |
0 |
Guest registration in SIP |
| 2:08PM |
2 |
Asterisk Queues Problem |
| 12:45PM |
4 |
Long call setup times on SIP to zaptel calls |
| 12:28PM |
0 |
Asterisk guru wanted, SoCal (LA/OC/San Bernardino County) |
| 12:19PM |
0 |
h323 - SIP conversion |
| 11:57AM |
2 |
7912 phones loosing registration |
| 10:25AM |
6 |
Connect PBX CO Port to TDM FXS Port |
| 9:49AM |
0 |
addons 1.4 and cdr_addon_mysql not installed ! |
| 9:44AM |
3 |
Maximum Number of Calls Asterisk Can Handle |
| 9:19AM |
0 |
Pause a Audio File Problem |
| 8:48AM |
0 |
No Ringback, only on 1 SIP provider |
| 8:12AM |
3 |
OT - IP Network Call Recording |
| 8:11AM |
1 |
Hint and CallerID |
| 7:52AM |
1 |
Feeding digit input to PauseQueueMember |
| 7:01AM |
1 |
Multi-calendar Overlay Layers? |
| 6:57AM |
3 |
asterisk freeze due to "too many open file" error |
| 6:04AM |
7 |
Call forwarding |
| 5:29AM |
1 |
Interruptible announcements in queue application |
| 2:40AM |
1 |
Symbian IAX client |
| 1:50AM |
0 |
directed call pickup with PICKUPMARK |
| |
| Wednesday February 14 2007 |
| Time | Replies | Subject |
|---|
| 11:12PM |
4 |
Best FXO Gateway |
| 7:13PM |
1 |
[Fxo] Digium TDM01B vs. OpenVox A400P01? |
| 5:58PM |
0 |
Requested contexts didn't get merged |
| 5:30PM |
1 |
Strange behaviour with Dial cmd |
| 5:10PM |
0 |
custom sip header |
| 4:40PM |
0 |
Useragent List |
| 4:18PM |
1 |
Need info for creating * as a gateway for other * servers. |
| 3:55PM |
2 |
moving WiFi phone |
| 1:07PM |
3 |
S101I (IAX) limitation |
| 1:00PM |
2 |
Asterisk vendors in Houston, TX |
| 12:36PM |
2 |
Macro Usage |
| 11:45AM |
1 |
Limit on SIP phones on one server |
| 11:36AM |
0 |
libunicall + hashtable.c + asterisk crash |
| 11:31AM |
1 |
zaptel 1.4 svn doesn't compile |
| 11:28AM |
0 |
Connect a legacy PBX to an Asterisk Server |
| 10:43AM |
0 |
Zoiper softphone version 1.03 now available |
| 10:25AM |
4 |
Guide to better performance using * ? |
| 10:09AM |
2 |
Fanless solution |
| 9:49AM |
0 |
Realtime via ODBC breaks for Voicemail |
| 9:42AM |
2 |
"Unable to launch Sendmail" warning |
| 8:21AM |
2 |
Compiling Zaptel-1.2.13 on FC3 |
| 8:05AM |
1 |
CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20 |
| 7:58AM |
0 |
Please some advice on setting hook-flash timing on Linksys PAP2 |
| 7:28AM |
2 |
Problem Transferring Direct to Voicemail |
| 7:23AM |
2 |
MRTG with 4 graphs |
| 7:19AM |
5 |
Bandwidth shapping device |
| 5:46AM |
1 |
To jitter buffer or not to jitter buffer? |
| 5:18AM |
2 |
SIP response 482 "Loop Detected" |
| 3:42AM |
0 |
asterisk 1.4 is stable? |
| 3:40AM |
0 |
Asterisk & CME integration using h323 |
| 3:38AM |
6 |
Fax with T.38 |
| 3:27AM |
6 |
genzaptool from "xorcom" |
| 2:10AM |
2 |
Can anyone help me out with Polycom 2.1 firmware please? |
| 1:46AM |
1 |
Following call forwards |
| 1:16AM |
1 |
Asterisk 1.4 and chan_misdn |
| 12:33AM |
3 |
GSM Gateway promotion from £69GBP |
| |
| Tuesday February 13 2007 |
| Time | Replies | Subject |
|---|
| 8:45PM |
2 |
E911 SIP or IAX providers? |
| 6:04PM |
3 |
Sending SMS from Asterisk |
| 4:18PM |
3 |
SMS via VoIP and web |
| 2:41PM |
1 |
FRITZ!Box Fon ata |
| 2:22PM |
1 |
PRI Call Start |
| 2:13PM |
3 |
AgentCallBackLogin vs AddQueueMember |
| 1:30PM |
0 |
Compiling Asterisk With ZapTel? |
| 1:27PM |
0 |
Your favorite switchboard application software ? |
| 1:00PM |
0 |
Asterisk 1.4.0 and callwaiting eventually drops call |
| 12:45PM |
1 |
How can I use "Asterisk Manager API" to hold and retrive an active call? |
| 12:06PM |
0 |
Blocking collect calls in Brazil |
| 10:18AM |
0 |
problems with trunks IAX2 and queues |
| 9:16AM |
4 |
Nortel 81C MSDL Trunking to Asterisk TE110P, Nortel Resetting PRI Channels |
| 9:14AM |
1 |
AGI "GET DATA" and "WAIT FOR DIGIT" don't work |
| 8:38AM |
1 |
Paging Followup |
| 7:31AM |
1 |
Using Dynamic Groups instead of AgentCallbackLogin - how to log which agent took the call? |
| 6:24AM |
1 |
Using Asterisk/callerid with "pay as you go" |
| 6:07AM |
1 |
Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33 |
| 5:56AM |
1 |
Originating calls: Astmanproxy vs Direct Connection vs Call files |
| 5:39AM |
2 |
problem with safe_asterisk |
| 5:30AM |
7 |
error when compiling asterisk-1.4 |
| 4:24AM |
6 |
Recomended POE Phones |
| 3:36AM |
1 |
Can Asterisk handle 7000 SIP users? |
| 3:25AM |
2 |
Customisable In-band ringing? |
| 3:19AM |
0 |
Error when compiling zaptel 1.2 |
| 2:43AM |
1 |
question about regex |
| 2:29AM |
1 |
asterisk-sounds doesn't exist in the sources, how can i get it? |
| 1:12AM |
0 |
Local channel characteristics |
| |
| Monday February 12 2007 |
| Time | Replies | Subject |
|---|
| 10:24PM |
2 |
Digium Card ? |
| 7:24PM |
1 |
End Wrap-up Time? |
| 2:23PM |
1 |
Small CDR Billing Program |
| 1:51PM |
2 |
colors in the console |
| 1:33PM |
0 |
Using Asterisk/callerid with "pay as you go" VOIP providers |
| 11:43AM |
2 |
i m looking for a document that allow me to install well an asterisk server |
| 11:17AM |
2 |
AsteriskNOW Migration |
| 10:54AM |
4 |
Zaptel install... |
| 10:14AM |
1 |
FW: After upgrade to 1.4 transfers don't workproperly |
| 10:13AM |
0 |
sendmail problem |
| 9:42AM |
3 |
Trixbox vs. Custom install |
| 9:24AM |
0 |
Parking via ## still broken |
| 9:07AM |
1 |
AGI question |
| 8:48AM |
2 |
Witch kernel version may i use to run well asterisk |
| 8:19AM |
2 |
T1 card recommendation |
| 7:42AM |
1 |
phpagi - Event "On Hangup" |
| 6:56AM |
1 |
fxotune on TDM24XXE card |
| 6:43AM |
0 |
Quintum gateways |
| 6:41AM |
3 |
Bad audio quality on SIP |
| 6:06AM |
2 |
Problems Asterisk with Digium TDM400 card => he don't see the disconnect |
| 5:18AM |
0 |
Resque Calls from someone who is already speaking |
| 3:12AM |
0 |
Asterisk-Java 0.3 Milestone 2 |
| 2:45AM |
0 |
Using Asterisk's manager interface to recieve calls |
| 2:36AM |
3 |
Disable root shell from CLI |
| 12:20AM |
1 |
Cisco Router for supply a connection from PABX to Asterisk |
| 12:11AM |
0 |
Cisco Router for supply a connection from PABX to Asterisk ? |
| |
| Sunday February 11 2007 |
| Time | Replies | Subject |
|---|
| 9:49PM |
2 |
Extensions in macro |
| 7:58PM |
0 |
chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
| 7:05PM |
0 |
realtime and save ip server in database |
| 12:57PM |
5 |
pmp_l1_check=no with zaphfc (Bristuff) |
| 12:21PM |
2 |
AsterikNow vs Trixbox |
| 12:11PM |
1 |
Debugging a SIP / AudioCodes Problem |
| 8:49AM |
2 |
TDM02B not working |
| 7:15AM |
2 |
Can not compile latest zaptel -1.2.13 |
| 5:25AM |
0 |
TE110P working hardware configurations |
| 5:12AM |
1 |
TDM2400 and 3.3v pci |
| |
| Saturday February 10 2007 |
| Time | Replies | Subject |
|---|
| 11:33PM |
1 |
Digium S101I as a traveling companion |
| 9:15PM |
3 |
Dial out from AGI |
| 7:59PM |
1 |
recording issues |
| 4:47PM |
9 |
Mini-ITX board + FXO PCI card? |
| 10:55AM |
3 |
Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card |
| 9:56AM |
1 |
canreinvite problems |
| 4:46AM |
0 |
Unable to lookup host in c= line |
| 2:21AM |
0 |
problem with installing tdm2400 |
| 12:09AM |
1 |
SIP retry time too low |
| |
| Friday February 9 2007 |
| Time | Replies | Subject |
|---|
| 7:18PM |
3 |
changing callerid to ring groups callerid |
| 6:30PM |
6 |
The High Performance Echo Canceller (HPEC) |
| 6:09PM |
2 |
asterisk and multiple cpus/cores |
| 6:06PM |
1 |
Outbound Call Transfer Problem |
| 4:37PM |
7 |
Dialplan checkup |
| 2:17PM |
0 |
ring requested on channel |
| 1:31PM |
1 |
Detect hang-up |
| 1:13PM |
4 |
asterisk 1.4 FC5 and Gtalk |
| 12:48PM |
1 |
RFC2833 SIP trunks and DTMF |
| 12:15PM |
1 |
CallerID on Dish 301 Receiver |
| 10:59AM |
0 |
Re: asterisk-users Digest, Vol 31, Issue 37 |
| 10:42AM |
0 |
Asterisk 1.2.15 released! |
| 10:42AM |
0 |
Zaptel 1.2.13 released! |
| 10:25AM |
0 |
Dependencies on DB? |
| 10:06AM |
2 |
Chan_Cellphone |
| 9:22AM |
0 |
Conference & Page question |
| 9:09AM |
0 |
TDM2400: some FXS module fail |
| 9:05AM |
1 |
anyone remembers where to check this list threads on a web site? |
| 8:45AM |
0 |
*****SPAMZ***** Conference & Page question |
| 8:16AM |
3 |
receiving fax with junghanns quadbri bristuff |
| 7:51AM |
1 |
Conferencing Phones ... |
| 7:49AM |
0 |
Asterisk 1.2.14 - Chanspy, sound issues. |
| 7:10AM |
1 |
Problems with GXP2000 and Asterisk => Call pickupand Voicemail |
| 2:38AM |
0 |
Misdn instability with asterisk 1.4 |
| |
| Thursday February 8 2007 |
| Time | Replies | Subject |
|---|
| 11:23PM |
1 |
call park and call transfer example |
| 9:56PM |
1 |
Problems with GXP2000 and Asterisk => Call pickup and Voicemail |
| 9:30PM |
2 |
requesting real world meetme capacity numbers |
| 8:46PM |
1 |
Queue extension issues |
| 7:12PM |
0 |
SIP Re-Invite behind a NAT |
| 6:43PM |
1 |
TDM400 with 1 FXO |
| 5:02PM |
1 |
Recording and MWI |
| 2:50PM |
1 |
Auto Answer (Paging) |
| 2:42PM |
0 |
www.BarCampUSA.org tickets went on sale this week |
| 2:08PM |
1 |
Any Way to Get # Functionality in DISA |
| 12:35PM |
2 |
Asterisk outbound calling does not wait for answer before playback |
| 11:47AM |
2 |
Suppliers in Canada |
| 10:58AM |
4 |
error when compiling zaptel-1.4 |
| 10:36AM |
3 |
Automatic Dial, Play message |
| 9:57AM |
3 |
Skutch AS-66 and an X100P |
| 9:36AM |
0 |
Transfer -> announce -> ring |
| 8:57AM |
0 |
Transfer |
| 8:27AM |
1 |
After upgrade to 1.4 transfers don't workproperly |
| 8:21AM |
0 |
Re: asterisk-users Digest, Vol 31, Issue 29 |
| 7:14AM |
3 |
Digium cards on Vmware |
| 6:00AM |
0 |
mysql error |
| 5:01AM |
2 |
dCAP |
| 4:10AM |
0 |
Realtime asterisk queues only reload queue members when a new call joins the queue |
| 3:54AM |
0 |
T.38 FAx |
| 3:17AM |
0 |
dial application timeout |
| 3:10AM |
2 |
problem with asterisk AGI |
| 2:28AM |
0 |
Activate/Deactivate zap channels in realtime |
| 1:45AM |
11 |
Best phone for easy provisioning |
| 1:36AM |
3 |
Asterisk and 802.11g |
| |
| Wednesday February 7 2007 |
| Time | Replies | Subject |
|---|
| 10:57PM |
1 |
Spliting video and audio |
| 7:45PM |
1 |
Large number of prefixes in a route to a trunk |
| 5:53PM |
3 |
Linux Kernel Timer Frequency and Asterisk |
| 4:48PM |
3 |
Red alarms |
| 4:05PM |
1 |
does any one knows of a Softphone that works under terminal services? |
| 1:12PM |
1 |
After upgrade to 1.4 transfers don't work properly |
| 1:04PM |
2 |
Softphone +Realtime |
| 12:36PM |
0 |
RE: Linksys auto provision |
| 12:30PM |
2 |
List problem handling HTML E-mails? |
| 12:13PM |
0 |
Zaptel bug |
| 12:04PM |
2 |
CPU & motherboard for 100+ simultaneouse calls on Digium Quad E1 TE411p |
| 11:50AM |
0 |
semi-private call |
| 11:44AM |
0 |
Can't get asterisk to compile chan_zap (was "NewIssue") |
| 11:26AM |
3 |
Diagnosing poor call quality |
| 10:02AM |
2 |
Can't get asterisk to compile chan_zap (was "New Issue") |
| 9:27AM |
2 |
AMI Originate and release channels |
| 8:55AM |
3 |
Trying to register an G.729 codec boght from Digium and the "register" command does aboslutely nothing |
| 8:04AM |
4 |
Billing pulses |
| 7:24AM |
0 |
prob with not recognizing hangup, pickup - python |
| 7:19AM |
0 |
Connection problem w/ Attended Transfer |
| 6:46AM |
1 |
Chanspy severe sound problems |
| 6:41AM |
1 |
H323 to SIP - One way voice |
| 6:30AM |
1 |
Asterisk Cmd to ID Mobile from Phone#? |
| 6:08AM |
0 |
SIP/Console -> ISDN ticks |
| 5:42AM |
0 |
Glitches in voicemail prompts |
| 5:32AM |
0 |
dnsmgr seems to have died |
| 5:25AM |
0 |
one touch recording problem in asterisk 1.4 |
| 3:26AM |
1 |
registration not timing out? |
| 2:27AM |
0 |
Pickup |
| 1:53AM |
2 |
Type of wake-up Call |
| 12:57AM |
4 |
s-${DIALSTATUS} extensions |
| 12:29AM |
9 |
Digium TE110P |
| |
| Tuesday February 6 2007 |
| Time | Replies | Subject |
|---|
| 9:21PM |
0 |
International validation |
| 9:10PM |
2 |
Disconnection supervision: what about PBX |
| 7:30PM |
1 |
Are there any IP phone in the market have such features? |
| 7:12PM |
2 |
Buddy list order |
| 6:16PM |
3 |
Help - Poor Voice Quality |
| 5:16PM |
0 |
ooh323 drops registration with Cisco IOS GateKeeper - bug or config issue? |
| 1:41PM |
1 |
yellow alarm after weeks without trouble |
| 1:33PM |
2 |
Mysterious tables starting with "stats_" |
| 12:46PM |
2 |
CD needed: no way to burn |
| 9:59AM |
4 |
Something wrong with the list? |
| 7:33AM |
0 |
Call Connections Dropped |
| 7:10AM |
0 |
troubles gsm-gateway no free channels |
| 6:55AM |
0 |
dtmf not recognized with misdn-install: help for alternatives |
| 5:35AM |
0 |
Hardware perfect for TE412P runnning * 1.4 |
| 4:40AM |
1 |
asterisk server as a voicemail server forlegacyPBX -- FXO or FXS??? |
| 4:38AM |
0 |
asterisk server as a voicemail server forlegacy PBX -- FXO or FXS??? |
| 4:06AM |
2 |
BindPort |
| 4:04AM |
1 |
pridialplan/prilocaldialplan |
| 3:33AM |
0 |
Asterisk H.323, Cisco IOS Gatekeeper(s) intra-zone call routing and TETRA |
| 2:22AM |
0 |
ExtensionStatusEvent |
| |
| Monday February 5 2007 |
| Time | Replies | Subject |
|---|
| 11:42PM |
1 |
Inserting a pause with Sipura in between |
| 10:27PM |
4 |
Having Trouble With Wait Command in Callback Context |
| 9:34PM |
1 |
How to access environment variable? |
| 6:03PM |
2 |
asterisk server as a voicemail server for legacy PBX -- FXO or FXS??? |
| 5:40PM |
0 |
Help - Received response: "Forbidden" from'"Unknown" |
| 5:21PM |
0 |
Callfiles to Meetme Fails (was: RE: Using Local Channels with Originate) |
| 4:46PM |
2 |
Howto use PRI lines (E1 or T1) for "data calls"? |
| 3:09PM |
1 |
Question on G.729 |
| 2:53PM |
1 |
Sending sound to an open channel.... |
| 2:44PM |
1 |
OpenSuSE Firewall2 - Traffic Shaping |
| 2:10PM |
2 |
how to install Zaptel on Fedora linux 5 |
| 10:41AM |
0 |
Packek2Packet Bridging vs. Native Bridging |
| 10:22AM |
0 |
TDM2402E routing incoming calls out to cell phone low volume |
| 9:20AM |
0 |
Reliably detecting hangup |
| 9:09AM |
1 |
Trixbox 1.2.3 - TDM400 FXOs - Outgoing Calls - Transfer # Not Wor king |
| 7:57AM |
1 |
Test to Speech |
| 7:30AM |
1 |
'h' extension and which one applies? |
| 7:29AM |
2 |
Use Digium TE110P Single T1 / E1 PCI Interface Card for connect a old PABX ? |
| 6:46AM |
2 |
playing wav49/gsm files on linux? |
| 6:04AM |
0 |
*****SPAMZ***** Asterisk cluster - keep up connection? |
| 5:52AM |
1 |
format_wav.c:247 update_header: Unable to find our position |
| 3:40AM |
0 |
translation error |
| 2:52AM |
0 |
Anybody using "VoiceRD" or "Thinkbright" service? |
| 2:13AM |
2 |
voip office pbx |
| 2:11AM |
0 |
SNOM phones stay "in use" after transfer |
| 1:57AM |
5 |
Asterisk Faxing Support |
| |
| Sunday February 4 2007 |
| Time | Replies | Subject |
|---|
| 11:42PM |
1 |
Local hangup after Dial()? |
| 9:49PM |
0 |
Does TE212P card work on HP DL380 G5? |
| 9:08PM |
1 |
Help - Received response: "Forbidden" from '"Unknown" |
| 8:01PM |
0 |
WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1' |
| 7:33PM |
0 |
Tampa Asterisk User Group Meeting Monday |
| 6:57PM |
1 |
FreeBSD Compile Errors |
| 6:32PM |
0 |
How do I debug? |
| 5:03PM |
5 |
Unicall/R2 for Asterisk 1.4 Available for TESTING |
| 4:42PM |
2 |
SIP privacy headers |
| 3:43PM |
1 |
Continue line in config files? |
| 3:27PM |
1 |
Problem loading AstDB into variable on restart |
| 12:54PM |
0 |
dnsmgr died? |
| 12:29PM |
9 |
Zap FXS slow to reset? |
| 12:04PM |
0 |
VM - User Only able to set unavail message |
| 8:37AM |
1 |
bristuff mailinglist |
| 8:17AM |
1 |
Detecting answer with an analogue card |
| 5:26AM |
0 |
SMDI support on trixbox |
| 4:08AM |
0 |
Help sought: Asterisk H.323, Cisco IOS Gatekeeper(s) intra-zone call routing and TETRA |
| 3:28AM |
0 |
Asterisk and multicore processors |
| 2:23AM |
1 |
Asterisk 1.2.14 and bristuff 0.2.0-RC8s |
| 2:21AM |
2 |
TDM400 noHangup |
| 1:39AM |
1 |
TDM400 stopped bridging outgoing FXO call |
| 12:58AM |
1 |
Interact with IVR |
| |
| Saturday February 3 2007 |
| Time | Replies | Subject |
|---|
| 10:47PM |
2 |
Command to disconnect a call |
| 6:24PM |
1 |
Single BLF for ALL trunks in use |
| 2:06PM |
2 |
Google Talk without gmail accout? |
| 11:35AM |
2 |
asterisk 1.2 branch revision 53132 failed to compile |
| 10:50AM |
3 |
error dialing a SIP user. chan_sip.c:1994 create_addr: No such host |
| 5:22AM |
1 |
misdn and prostgres_cdr on asterisk 1.4 |
| |
| Friday February 2 2007 |
| Time | Replies | Subject |
|---|
| 9:40PM |
0 |
Call Waiting broken on ZAP |
| 2:13PM |
0 |
7912 issues half audio |
| 12:38PM |
1 |
asterisk server RFC conformance |
| 11:48AM |
1 |
1.4 res_snmp dependencies (Debian) |
| 10:35AM |
1 |
queues and LOCAL for members |
| 10:17AM |
0 |
Local channel with /n doesn't hangup after transfer. Why? |
| 10:06AM |
0 |
No RTP packets received by Asterisk when calling SIP to SIP |
| 9:51AM |
0 |
FOP (or equivalent) and timers |
| 8:53AM |
1 |
CallerID Name not available. |
| 4:31AM |
2 |
Asterisk logging everything? |
| 4:12AM |
0 |
install-misdn compile problem with debian |
| 4:09AM |
0 |
Line drops |
| 3:05AM |
1 |
volume control in VoIP |
| 2:13AM |
1 |
WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels ( when I use asyncgoto) |
| |
| Thursday February 1 2007 |
| Time | Replies | Subject |
|---|
| 11:19PM |
1 |
Asterisk cann't redirect the calling party to anothere Exten. |
| 10:28PM |
3 |
How to Clone Asterisk |
| 7:45PM |
3 |
windows SIP Softphones ? |
| 6:43PM |
0 |
Re: why there havn't"app_meetme.so"fileaboutasterisk1.4.0? |
| 6:28PM |
1 |
API Originate Action - distinguishing between No Answer and Invalid phone number |
| 6:20PM |
2 |
make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed. |
| 6:18PM |
0 |
Asterisk Scaling/Load Balancing for iax soft clients |
| 3:25PM |
1 |
Using Local Channels with Originate |
| 3:15PM |
1 |
Please help parse this GotoIf line |
| 3:03PM |
0 |
server hardware choice, |
| 1:30PM |
1 |
Dial option G - Passing parameters? |
| 12:57PM |
8 |
Dell Servers |
| 9:15AM |
1 |
dialplan logic based on caller ID |
| 8:05AM |
1 |
3 PCI slot with exclusive IRQ ? please advice! |
| 7:47AM |
2 |
SendText() question |
| 7:27AM |
1 |
Re: why there havn't "app_meetme.so"fileaboutasterisk1.4.0? |
| 7:00AM |
0 |
Re: why there havn't "app_meetme.so" fileaboutasterisk1.4.0? |
| 6:37AM |
0 |
asterisk 1.4 and r2mfc or unicall |
| 5:45AM |
0 |
Dialplan programming vs. AGI vs. ??? |
| 4:15AM |
1 |
CDR - uniqueid |
| 3:19AM |
0 |
Enhanced PickupChan |
| 3:15AM |
0 |
extensions.conf gotoif and label |
| 2:30AM |
1 |
why there havn't "app_meetme.so" file about asterisk1.4.0? |
| 12:21AM |
2 |
strange caller display |