| Wednesday May 31 2006 |
| Time | Replies | Subject |
|---|
| 10:32PM |
5 |
Openion on Sipura SPA-2100 |
| 10:00PM |
4 |
how to decrease answer time ! |
| 9:51PM |
0 |
app_ices.c broken pipe error : bug ?? |
| 9:15PM |
1 |
: Re: Upgrade ONLY asterisk from an AAH install |
| 6:54PM |
1 |
Problems with ZAP dial timeout |
| 5:25PM |
1 |
clicking and popping with capi, okay with mISDN |
| 4:19PM |
0 |
Libmfcr2 won't compile |
| 4:15PM |
0 |
Ringing to Outside Line |
| 3:56PM |
4 |
MFC/R2 for Voice and Data |
| 3:30PM |
1 |
Connect 2 Asterisk Servers via PRI |
| 3:14PM |
2 |
Alternative to FWD |
| 3:13PM |
1 |
Brief silence followed by DTMF tone on T1 line |
| 12:55PM |
1 |
Can you dial with different CID's? |
| 12:21PM |
2 |
AEL2 and CID |
| 12:17PM |
2 |
Forcing Marker bit |
| 11:56AM |
0 |
NEW => Asterisk Event Monitor |
| 11:35AM |
0 |
TDM no dialtone on connected phone |
| 11:34AM |
1 |
Upgrade ONLY asterisk from an AAH install |
| 11:23AM |
2 |
PAP2-NA Authentication Issues |
| 11:11AM |
0 |
No system sound with Asterisk@Home |
| 11:07AM |
1 |
Looking for a VoIP solution... |
| 10:48AM |
1 |
Zap Flash() |
| 10:33AM |
5 |
Converting .wav to .WAV |
| 10:23AM |
0 |
AEL #include ( Now Labels & Goto() ) |
| 9:13AM |
0 |
Hold Status |
| 9:09AM |
5 |
Asterisk crashes at startup |
| 9:05AM |
5 |
Explicit Dialplan Exit |
| 8:15AM |
5 |
SIP Presence |
| 7:58AM |
3 |
Labels and Goto() |
| 7:53AM |
0 |
DTMF Again |
| 6:39AM |
0 |
Incoming IAX going to wrong context |
| 6:11AM |
0 |
Bristuff PickUp and call transfers - can it be done? |
| 5:39AM |
1 |
Global variables - collision? |
| 5:08AM |
2 |
Zap Channels , for round-robin search and call |
| 4:59AM |
0 |
Asterisk Bootcamp in Europe :: June 12-16 and the Asterisk SIP Masterclass in Chicago, July 2006 |
| 4:58AM |
3 |
Zap channels ringing too loudly |
| 4:41AM |
3 |
Centos cause Asterisk crash |
| 4:41AM |
0 |
extra parameter for DB read function |
| 3:52AM |
3 |
Need help with Junghanns Quadbri |
| 3:31AM |
1 |
*****SPAM***** Upgrading |
| 3:00AM |
0 |
Fax to Email issue with Spandsp tif not correctly sized |
| 2:23AM |
0 |
Asterisk receiving call from Panasonic TDA extension issue |
| 2:01AM |
1 |
INFO: TFOT book- n priorities and labels |
| 1:52AM |
0 |
AGI MySql |
| 12:51AM |
2 |
Nokia E60 , experience as SIP client |
| |
| Tuesday May 30 2006 |
| Time | Replies | Subject |
|---|
| 11:11PM |
1 |
Questions from a working doctors' office installation |
| 9:51PM |
0 |
Linksys spa 942 handsfree SIP->PSTN/GSM |
| 9:25PM |
1 |
Sip gateway don´t hangs up |
| 8:13PM |
8 |
Handset recommendations |
| 6:33PM |
0 |
Register Today For AstriCon Europe |
| 6:22PM |
1 |
Shared Call / Bridged Line Appearances (SIP-B) |
| 6:19PM |
0 |
zt hook failed |
| 6:15PM |
1 |
Got SIP response 405 "Method not acceptable" back from xxx.xxx.xxx.xxx |
| 5:29PM |
2 |
Polycom replacement handset |
| 4:19PM |
1 |
BEST PRICES ON NMS DIALOGIC DIGIUM VOIP WWW.VOICEINTERNATIONAL.COM |
| 3:46PM |
1 |
Asterisk 1.2.8, Zaptel 1.2.6 and libpri 1.2.3 released! |
| 3:26PM |
0 |
Problem with tor2 driver and Zapata Tormenta 2 Quad T1/PRI Card |
| 2:31PM |
3 |
Still can't get asterisk to play voicemail files occasionally |
| 2:08PM |
1 |
Callerid and trunk |
| 1:55PM |
1 |
Dropped SIP connections never being closed? |
| 1:42PM |
5 |
Compiling Asterisk-addons |
| 1:29PM |
1 |
No sound?? HELP |
| 1:17PM |
3 |
instalacion |
| 1:00PM |
8 |
How to strip a digit |
| 1:00PM |
1 |
Is Asterisk svn link down ? |
| 12:57PM |
20 |
AEL #include |
| 12:19PM |
1 |
Asterisk::AGI and DIALEDTIME |
| 12:03PM |
1 |
CallerID outbound |
| 11:53AM |
0 |
RE: Asterisk-Users Digest, Vol 22, Issue 169 |
| 11:40AM |
4 |
Unicall Protocol Failure |
| 10:15AM |
0 |
app_conference sources? |
| 9:50AM |
1 |
Zaptel and 2.6.9-34.0.1.EL Kernel on CentOS |
| 8:54AM |
1 |
patch application |
| 8:50AM |
0 |
Dumping outbound audio on hold |
| 8:43AM |
2 |
Automon |
| 8:33AM |
0 |
IAX softphone with RSA support? |
| 8:32AM |
0 |
LDAP directory app? |
| 8:22AM |
1 |
Hardware requirements for Asterisk |
| 6:34AM |
0 |
no extension from ISDN phone with bristuff |
| 6:05AM |
0 |
Extensions, devices and dialplan |
| 5:09AM |
3 |
Panasonic PBX |
| 3:10AM |
2 |
problem about asterisk realtime. |
| 2:26AM |
1 |
Asterisk restarting in a minute |
| 2:19AM |
1 |
sIp port numbers |
| 1:56AM |
4 |
I guess my server capacity is ok |
| |
| Monday May 29 2006 |
| Time | Replies | Subject |
|---|
| 10:42PM |
8 |
E1 hardware for asterisk |
| 10:31PM |
2 |
sip interopability problem |
| 4:30PM |
0 |
Melbourne Asterisk Group meeting Thursday |
| 4:23PM |
2 |
Simple windows / web Asterisk user software? |
| 4:05PM |
0 |
Sipura 941 missing blind transfer soft button? |
| 3:56PM |
2 |
Problem with IAX2 dialin with portunity |
| 3:32PM |
4 |
Recent debian packages? |
| 11:33AM |
4 |
app_conference DTMFs? |
| 11:07AM |
1 |
Re: Nufone Echo Test |
| 10:46AM |
2 |
Asterisk Internal sip calls I can´t send/recive |
| 8:53AM |
0 |
Brother 8360P fax cannot connect to TDM400 |
| 7:03AM |
4 |
How to enable call waiting on Sip Phones |
| 6:27AM |
4 |
registration at Voipbuster times out |
| 6:14AM |
1 |
I can't call PSTN numbers |
| 6:07AM |
2 |
Memory-leak 1.2.7.1 |
| 5:14AM |
0 |
Define call-groups |
| 4:25AM |
0 |
pedantic on sip.conf |
| 3:56AM |
0 |
Asotel Dynamix DW-04/S with asterisk? |
| 2:56AM |
1 |
Ring-Answer with Polycom 501 and Asterisk |
| 2:15AM |
0 |
New Zealand Voice prompts announcement |
| 12:44AM |
0 |
doubts about asteriskconfigurationfromdatabase |
| 12:33AM |
3 |
TDM2400P with echo canceller not working |
| |
| Sunday May 28 2006 |
| Time | Replies | Subject |
|---|
| 11:04PM |
1 |
doubts about asterisk configurationfromdatabase |
| 8:17PM |
1 |
IVR sounds not on certain inbound route |
| 7:36PM |
0 |
Go2call Configuration |
| 6:43PM |
3 |
doubts about asterisk configuration from database |
| 4:18PM |
1 |
Asterisk registers but won't complete calls. |
| 2:09PM |
3 |
Asterisk Radius Module |
| 1:36PM |
5 |
hook into authentication |
| 12:40PM |
1 |
Analogue phone w/ TDM400 |
| 12:13PM |
0 |
SIP and sound breaking |
| 10:36AM |
1 |
Calls connected, but no audio |
| 4:55AM |
0 |
SER qualify |
| 4:26AM |
0 |
My Call drop after 60 to 63 Seconds!! |
| 3:46AM |
1 |
FreeBSD Digium g.729 codec seg faults on rev 30652 |
| |
| Saturday May 27 2006 |
| Time | Replies | Subject |
|---|
| 2:06PM |
3 |
TDM |
| 1:43PM |
1 |
Fw: features |
| 1:26PM |
1 |
Polycom 600 presence indication on *LED*? |
| 10:26AM |
1 |
Dcap Test |
| 9:54AM |
2 |
Web based interface |
| 8:51AM |
1 |
Compiling chan_bluetooth |
| 6:30AM |
2 |
Calling a person over Internet |
| 6:17AM |
0 |
JabberStatus |
| 3:03AM |
2 |
amportal doesn't start with brestuff(ISDN)HFC-PCI |
| |
| Friday May 26 2006 |
| Time | Replies | Subject |
|---|
| 9:53PM |
1 |
asterisk with centos 4.3 sources compilation |
| 8:54PM |
4 |
mpg123 or asterisk |
| 8:32PM |
0 |
RV: DELL PowerEdge 2850 and TE4110P and TE110P |
| 7:11PM |
0 |
Polycom 601 |
| 3:13PM |
0 |
AMP and version numbers. |
| 2:29PM |
1 |
External Custom Extension Timeout |
| 2:10PM |
0 |
Sip Notify cisco-check-cfg - Does it still workwith 8.2? |
| 12:33PM |
1 |
Sangoma A200 4 port FXO card suddenly stopped answer on channels 2, 3, 4 |
| 10:35AM |
2 |
Busy Signals |
| 8:58AM |
3 |
UK experts only. BT Outgoing caller ID not showing |
| 8:51AM |
1 |
End of migration: adding support for some an alog phones |
| 8:36AM |
1 |
OT: American Telecom Approved by FCC to Certify DECT Phones in US |
| 8:33AM |
1 |
IAX2 + port translation |
| 7:51AM |
2 |
large duration calls |
| 7:38AM |
3 |
Two questions about Asterisk@home and backups. |
| 7:35AM |
1 |
hints/subscriptions accross IAX |
| 6:21AM |
1 |
VoIP provider for Turkey from India with Asterisk |
| 6:15AM |
1 |
Need a recomendations and config samples. FXS<->SIP terminal with 4 ports. |
| 6:07AM |
3 |
hint priority and realtime |
| 6:01AM |
3 |
using a billing system |
| 5:51AM |
0 |
Getting stuck right at the beginning |
| 5:37AM |
4 |
End of migration: adding support for some analog phones |
| 3:56AM |
1 |
my kernel not detect my TDM400P card |
| 2:49AM |
3 |
Polycom 301's drop last two digits of dialed number |
| 2:22AM |
2 |
Asterisk.NET authentication problem |
| 1:56AM |
0 |
SIP call problem |
| 1:46AM |
1 |
Not able to make any calls |
| 12:15AM |
0 |
No sound when the call is diverted |
| |
| Thursday May 25 2006 |
| Time | Replies | Subject |
|---|
| 11:26PM |
2 |
Modules for X100P |
| 11:00PM |
2 |
Agent Callback, how to "see" wath queue is calling the agent? |
| 10:25PM |
1 |
PAP-2 Conferencing Problems |
| 10:10PM |
0 |
IAX registrations fail over time in SVN-trunk |
| 8:10PM |
1 |
pap2 bridging problems |
| 6:32PM |
3 |
X100P fails to initialize |
| 6:19PM |
0 |
Citel Handset Gateways and BLF (subscribe) buttons? |
| 4:57PM |
1 |
RRMEMORY / Queues Not Working Right |
| 4:11PM |
1 |
Way to disable codec in dialingplan |
| 3:42PM |
8 |
Snom firmwares suck <--additional datapoint to consider |
| 2:49PM |
0 |
Re: [asterisk-biz] Selling Bulgarian (+3592) DIDs at 1.5 USD |
| 2:31PM |
0 |
problems with TXfax |
| 1:38PM |
0 |
Anyone going to cluecon? |
| 1:26PM |
4 |
No rings before auto attendant |
| 1:12PM |
0 |
RE: Asterisk-Users Digest, Vol 22, Issue 147 |
| 12:54PM |
1 |
Paging Phones stay off the hook if you dont wait long enough. |
| 12:43PM |
2 |
jitterbuffer causes flaky IAX2 incoming connections? |
| 11:37AM |
0 |
FW: [isp-clec] Treasury disconnects tax on long-distance calls - with refunds |
| 11:21AM |
0 |
PRI Moving channels? |
| 10:41AM |
4 |
FreePBX virtualization |
| 10:22AM |
2 |
Compilation issues with s390 |
| 9:53AM |
4 |
Asterisk codec negotiation patch |
| 9:30AM |
1 |
Asterisk Manuals |
| 9:17AM |
0 |
Asterisk and sysmask - anyone? |
| 8:51AM |
5 |
PCI Problems |
| 8:49AM |
0 |
RE: Asterisk-Users Digest, Vol 22, Issue 132 |
| 8:34AM |
4 |
Failover Problem |
| 7:53AM |
0 |
Glueing apps and phones together |
| 7:47AM |
2 |
Volume configuration on Polycom Soundpoint 501phone |
| 7:32AM |
0 |
Re: Implementing Paging on the Linksys SPA9XX phones (working) |
| 7:16AM |
0 |
CallerID from cell phone not being rewritten |
| 7:03AM |
1 |
"Error" on Polycom 501 & 601. |
| 6:56AM |
1 |
IVR & transcoding & g729 license |
| 6:49AM |
1 |
[asterisk-biz] RE: OT: AudioCodes MP124-C/FSX/AC/SIP |
| 5:47AM |
0 |
Anyone got a used T1 card I can have? |
| 3:43AM |
2 |
VLAN info |
| 3:14AM |
1 |
Voice Mail Audio Progression |
| 1:10AM |
2 |
connecting asterisk to hylafax via t38modem: is it possible? |
| 1:00AM |
0 |
TDM2400P Problem |
| 12:42AM |
1 |
playback windows recorded sound |
| |
| Wednesday May 24 2006 |
| Time | Replies | Subject |
|---|
| 4:57PM |
2 |
PCI-X PRI hardware |
| 4:44PM |
0 |
SPA-941 called number distinctive ring with Personal Directory |
| 4:40PM |
2 |
Realtime Asterisk Problem |
| 3:47PM |
0 |
uClibc and g729 |
| 2:20PM |
0 |
Dual Line SIP config to the same provider |
| 2:09PM |
1 |
database lookup |
| 2:00PM |
3 |
Is NuFone Really Dead? |
| 1:42PM |
2 |
latest @Home questions |
| 1:11PM |
3 |
Spoofing a BLF Signal? |
| 1:11PM |
2 |
What and When is the next version of Asterisk? |
| 1:00PM |
2 |
OT: AudioCodes MP124-C/FSX/AC/SIP |
| 12:55PM |
1 |
Lighting up a light on an aastra phone |
| 12:25PM |
2 |
DHCP configuration for Cisco 7960? |
| 12:02PM |
1 |
Problem after upgrade to 1.2.7.1 |
| 11:15AM |
2 |
TE406P - MFC/R2 |
| 10:49AM |
1 |
Misdn 0.2.1 BUSY tone |
| 9:00AM |
5 |
macro-dial |
| 8:56AM |
1 |
DUNDi in 1.2.7.1 |
| 8:48AM |
1 |
Placing call files in/var/spool/asterisk/outgoing/ does not work |
| 8:43AM |
0 |
Placing call files in |
| 7:35AM |
1 |
Generate two calls from Asterisk and bridge them |
| 6:48AM |
1 |
How to add H.323 channels on Asterisk 1.2.7.1 |
| 6:06AM |
1 |
Placing call files in /var/spool/asterisk/outgoing/ does not work |
| 6:02AM |
1 |
Configuration for different Asterisk branches |
| 5:38AM |
0 |
[Fwd: IVR and operator] |
| 5:16AM |
2 |
Video SIP Softset |
| 4:59AM |
0 |
SIP Video software |
| 3:30AM |
5 |
GXP2k and BLF problem |
| 3:26AM |
2 |
asterisk amportal start/stopped/start/stopped for all the time |
| 2:45AM |
0 |
spanDSP & app_rxfax.so |
| 1:52AM |
3 |
How to prevent more than one agent to login to the same extension?? |
| 1:26AM |
2 |
SV: USB headsets? |
| 1:01AM |
4 |
USB headsets? |
| |
| Tuesday May 23 2006 |
| Time | Replies | Subject |
|---|
| 11:48PM |
0 |
[asterisk BUG]hangup |
| 11:46PM |
1 |
chan_zap.so error, asterisk stopped |
| 10:52PM |
1 |
Configure Voipjet.com content in Asterisk |
| 10:51PM |
0 |
FAX with PRI |
| 10:13PM |
3 |
Packetization configuration of IAX channels |
| 9:06PM |
1 |
Quintum Tenor DX 3020 problem to register on Asterisk |
| 7:07PM |
1 |
multiple registrations with Polycom IP600 |
| 4:30PM |
1 |
SPA 3102 Caller ID in Bellsouth/NA |
| 3:55PM |
1 |
They are? Re: Now that Nufone is dead... |
| 2:36PM |
0 |
IVR and operator |
| 2:14PM |
1 |
Getting the Server IP |
| 2:04PM |
0 |
Wacky Failover Situation w/SIP - Bug? |
| 1:36PM |
1 |
More Alison Keenan British English files |
| 1:02PM |
1 |
Problem with options to "Dial" application |
| 11:12AM |
2 |
Queue Count |
| 10:46AM |
1 |
PSTN -> CCM3.2 -> Asterisk CLID |
| 10:22AM |
0 |
CVS servers being taken out of service |
| 10:09AM |
3 |
AGI ? |
| 10:08AM |
1 |
Database Integration |
| 10:05AM |
0 |
Virtual VOIP numbers going to separate Asterisk mailboxes? |
| 9:08AM |
0 |
Zaptel Module.symvers missing |
| 8:40AM |
3 |
Transfer extensions processing control to Manager |
| 8:21AM |
0 |
Sip.conf: domain=huh? |
| 8:10AM |
1 |
Monitoring queues |
| 7:48AM |
13 |
Now that Nufone is dead... |
| 7:48AM |
4 |
What about T400 T1 cards? |
| 7:39AM |
1 |
Can Asterisk work in a proxy setting- a challenge |
| 7:23AM |
0 |
[asterisk BUG] |
| 7:19AM |
1 |
res_snmp |
| 7:02AM |
0 |
SIP Softphone or API which supports QoS (DiffServ/DSCP) needed |
| 6:50AM |
2 |
Asterisk connecting to a proprietry PBX |
| 6:46AM |
2 |
Queues - Can I PAUSE an agent instead of LOGGING OUT? |
| 6:14AM |
0 |
Problem in php-asmanager.php |
| 5:56AM |
6 |
Best VoIP provider for Asterisk |
| 5:26AM |
2 |
Are my expectations too high? |
| 5:17AM |
2 |
Outband call from php script |
| 4:21AM |
1 |
Im a Beginner |
| 3:01AM |
1 |
config files for Eicon Diva |
| 1:38AM |
1 |
AW: Free/Open pci telco card |
| 1:22AM |
1 |
Status: Provisioned, Down, Active - Long |
| 1:21AM |
0 |
A call from a call file always does a redial? |
| 1:11AM |
1 |
Free/Open pci telco card |
| 1:10AM |
2 |
Logger rotate & master.csv |
| 12:58AM |
0 |
SIP session number |
| 12:40AM |
0 |
[Fwd: Faxing - machines stop talking, line stays up] |
| 12:35AM |
2 |
TDM400P , "ztcfg ?vv error ", "Does it have to do with my PC hardware ?" |
| 12:23AM |
0 |
Faxing - machines stop talking, line stays up |
| |
| Monday May 22 2006 |
| Time | Replies | Subject |
|---|
| 10:18PM |
10 |
US telco lingo |
| 5:25PM |
1 |
Timeframe for QueueStatus values |
| 3:11PM |
2 |
I've broken voicemail |
| 2:31PM |
1 |
How to detect call forwarding to voicemail |
| 2:27PM |
1 |
Initial second lost on SIP phones |
| 1:52PM |
0 |
Voicemail: cannot use serveremail as variable |
| 1:28PM |
0 |
PRI bi-directional early media |
| 1:19PM |
0 |
SIPCHANINFO and 1.2.7.1 |
| 11:15AM |
1 |
A few queue questions |
| 10:22AM |
1 |
FXS Caller ID revisted |
| 10:21AM |
0 |
UUI field |
| 9:34AM |
0 |
Persistennt Data of Queue with Dynamic Agents |
| 9:16AM |
2 |
Centos 4.3 Issues |
| 8:50AM |
0 |
Asterisk Nortel Legacy Integration |
| 8:28AM |
1 |
TLS from a Sponsored Google Summer of Coding? |
| 8:21AM |
1 |
Script AGI on C |
| 8:11AM |
3 |
Office to Office via IAX2 problems |
| 7:20AM |
3 |
Option to reach someone in voicemail? |
| 7:16AM |
1 |
exten => *0. not possible |
| 5:35AM |
4 |
I get MOH when the caller hangs up |
| 4:54AM |
3 |
Problems with Park and MOH |
| 4:17AM |
0 |
Got reject for frame 0, but we only have others! |
| 4:13AM |
2 |
how to customize voicemail |
| 4:04AM |
2 |
Recommended SIP phones? |
| 2:55AM |
1 |
Asterisk on Proxy |
| 2:04AM |
0 |
Please help on chan_h323. |
| 1:57AM |
1 |
SIP to IAX - forcing codec pass thru |
| 1:53AM |
2 |
FW: WiFi / GSM VoIP Handsets.. |
| 1:18AM |
0 |
string parsing in extensions.conf |
| 12:35AM |
2 |
Not able to configure TDM400P with asterisk@home |
| 12:08AM |
1 |
behaviour depending on count of used lines |
| |
| Sunday May 21 2006 |
| Time | Replies | Subject |
|---|
| 11:06PM |
2 |
Snom 320 Shared line + speed dial |
| 9:17AM |
1 |
Limit outgoing calls |
| 8:00AM |
0 |
update or add DID's to directory Assistance |
| 6:47AM |
1 |
Skill-based routing |
| 6:04AM |
1 |
transfer outside of a call? |
| 5:28AM |
1 |
Upgrade 7960 from SCCP 3.0 to SIP 7.5 |
| 2:27AM |
1 |
Events offered by |
| 2:15AM |
1 |
no ringtone |
| |
| Saturday May 20 2006 |
| Time | Replies | Subject |
|---|
| 5:33PM |
1 |
Configuring a TDM400P with one FXS port |
| 6:43AM |
0 |
"Slash Tone" at pstn cut-though? |
| 6:31AM |
1 |
h323 to sip ringing indication |
| 4:52AM |
1 |
$1000USD for fix of Asterisk g726-32 codec |
| 4:30AM |
3 |
Any IP phones with pro-audio connections? |
| 2:52AM |
1 |
How to unlock old SCCP Cisco 7960 ? |
| 2:45AM |
1 |
Cisco 7940/60 SIP firmware 8.3 |
| |
| Friday May 19 2006 |
| Time | Replies | Subject |
|---|
| 10:23PM |
0 |
DID Provider via Asterisk |
| 8:23PM |
1 |
hardware help ? |
| 1:30PM |
1 |
Dell PowerEdge 1600 Compatibility Issues with Digium Card |
| 10:25AM |
0 |
Setup up Intellitouch ITC-3002 Sip phones with Asterisk |
| 10:06AM |
4 |
PRI dialing IVR with inband DTMF |
| 9:05AM |
2 |
British English voice files are ready for download |
| 8:33AM |
1 |
Non automated call parking |
| 8:28AM |
1 |
IAX Trunk |
| 7:27AM |
1 |
RTP Packetization |
| 6:57AM |
0 |
SpanDSP issues (oh fun!) |
| 6:32AM |
1 |
Not joining queue when empty |
| 6:18AM |
0 |
Forwarded Calls crash the system on 64 bit |
| 6:02AM |
4 |
Snom firmwares suck |
| 5:27AM |
1 |
Call detail records for Digital Receptionist |
| 4:51AM |
0 |
call recording - contrlo of Ast in 'h' extension |
| 3:43AM |
1 |
AsteriskOUT |
| 3:02AM |
2 |
voicemail access on the Thomson ST2030 ? |
| 2:54AM |
0 |
Faxing with Asterisk using both ISDN and FXS |
| 2:40AM |
0 |
help about modem |
| 2:10AM |
1 |
Watchguard Firebox 1000 woes |
| 2:10AM |
1 |
Development news :: Smarter medialess calls! |
| 1:55AM |
0 |
Call Transfer does not work |
| 1:30AM |
2 |
X100P not recognised on FreeBSD system |
| 1:20AM |
0 |
Show queues statictis |
| 1:15AM |
1 |
Experience with IBM X346 machines and Sangoma |
| 1:06AM |
2 |
Max Number of Extensions |
| 12:38AM |
2 |
SIP useragent? |
| |
| Thursday May 18 2006 |
| Time | Replies | Subject |
|---|
| 10:43PM |
0 |
Error building Oh323 |
| 9:14PM |
1 |
Digium card firmware |
| 8:35PM |
0 |
Fwd: [Announcement] Asterisk-IL mailing list |
| 5:08PM |
0 |
<SOLVED> Need help with Dial M option and destinationcontext |
| 3:05PM |
2 |
SIP Header Info |
| 1:59PM |
2 |
VoiceMail Groups |
| 1:22PM |
0 |
Pulling the mISDN number from an incoming call |
| 1:03PM |
0 |
E&M and Dial tone |
| 11:19AM |
1 |
R2/MFC Configuration. |
| 11:14AM |
0 |
Powertouch 350 CallID display continued |
| 9:38AM |
2 |
Auto Dial Out Madness |
| 8:56AM |
0 |
OT: Aastra Powertouch 350 caller id |
| 8:36AM |
3 |
Polycom - missed calls dial back |
| 7:58AM |
2 |
Polycom 601 -- programming buttons. |
| 7:19AM |
0 |
Applet to test VoIP quality |
| 6:59AM |
1 |
SIP re-invite and billing |
| 6:52AM |
5 |
Home asterisk system with single PSTN Line |
| 6:33AM |
0 |
Asterisk - SPA-3000, 407 error |
| 6:26AM |
2 |
Default dialplan?? |
| 6:06AM |
0 |
multiple calls using IAX |
| 6:03AM |
1 |
ACD Light on Phone? |
| 4:18AM |
3 |
just softphone |
| 4:07AM |
0 |
tdm21B in china |
| 4:00AM |
1 |
Unable to register channel |
| 3:29AM |
0 |
Failing SIP registration brings * down |
| 3:29AM |
1 |
SNOM, g722 and 16 kHz audio |
| 1:57AM |
1 |
DM/V1200-4E1 with asterisk |
| 1:31AM |
0 |
Unable to set channel to linear mode |
| 12:16AM |
0 |
Trunk Si without autetification |
| |
| Wednesday May 17 2006 |
| Time | Replies | Subject |
|---|
| 11:32PM |
0 |
Asterisk@home default password doesn't |
| 10:36PM |
2 |
Meetme conf |
| 8:53PM |
0 |
[Fwd: Calls being hung up] |
| 7:20PM |
2 |
[OT] Disconnect Tone in US |
| 4:28PM |
1 |
Is there a dialplan emulator available? |
| 3:33PM |
3 |
SPA-1001 behind NAT -> Internet Asterisk box -- BOUNTY! |
| 3:27PM |
0 |
Need technical info about dialers |
| 3:00PM |
3 |
Slackware 10.2 |
| 2:48PM |
2 |
Asterisk & Meridian Tie Line |
| 2:25PM |
2 |
AAH not getting IP address, likely to be network card? |
| 2:19PM |
2 |
New To Asterisk - Advice needed |
| 2:18PM |
0 |
AutoDialer Software |
| 1:59PM |
7 |
Quad BRI card |
| 1:10PM |
0 |
Audio problems 50% of the time. (kurt x) |
| 1:00PM |
0 |
RES: GET DATA and STREAM FILE commands, don´t work |
| 12:59PM |
4 |
Ringing indication not working as expected |
| 12:54PM |
4 |
Variable Inheritance - Set in Child, Read by Parent |
| 12:08PM |
0 |
DM/V1200-4E1 (Intel PCI 4xE1 ports) |
| 11:37AM |
5 |
Audio problems 50% of the time. |
| 11:19AM |
0 |
Upgrade issues |
| 9:55AM |
3 |
Providers using Embedded Devices |
| 9:53AM |
1 |
Weird Error When upgrading 7960G to 8.2 |
| 9:51AM |
0 |
Asterisk Using Multiple Databases with ODBC? |
| 9:51AM |
0 |
Weird Error Upgrading 7960's to 8.2SIP |
| 9:43AM |
1 |
Asus P5GD1... anyone using with Asterisk ?? |
| 9:38AM |
0 |
Can two asterisk servers share the same dialplan by using FreePBX? |
| 9:29AM |
3 |
Listening on Multiple Interfaces |
| 9:19AM |
0 |
RE: Asterisk-Users Digest, Vol 22, Issue 97 |
| 9:11AM |
0 |
Asterisk SIP Gateway behind NATS - SIP/2.0 404 Not Found |
| 8:29AM |
2 |
SIP redirect |
| 8:26AM |
0 |
fax & asterisk 1.2 |
| 8:08AM |
0 |
Overwriting SIP headers |
| 7:19AM |
3 |
soekris hadware |
| 6:50AM |
2 |
Diverse servers |
| 6:27AM |
2 |
IAX crackilng |
| 6:25AM |
0 |
(no subject) |
| 6:00AM |
0 |
Reading queue_logs |
| 4:54AM |
1 |
TDM does not disconnect |
| 4:25AM |
0 |
Re: Reasons for a SIP channel to hang ? - partially resolved |
| 4:21AM |
0 |
Asterisk Manager and Events Problem |
| 4:05AM |
1 |
Deadlocks in 1.2.7.1 |
| 3:30AM |
5 |
Plan to free myself from AAH |
| 3:30AM |
0 |
A CDR issue of agent.conf <createlink feature> |
| 3:29AM |
1 |
(newbie) Zaptel/ztdummy compiling on debian |
| 3:13AM |
2 |
SIP Min-Expires |
| 2:55AM |
1 |
NO ringing tone while dialing |
| 2:36AM |
1 |
no SUBSCRIBE request sent |
| 1:17AM |
2 |
Asterisk@home default password doesn't match |
| |
| Tuesday May 16 2006 |
| Time | Replies | Subject |
|---|
| 8:26PM |
2 |
mISDN & FAX |
| 8:20PM |
1 |
Asterisk as a proxy |
| 5:50PM |
1 |
GXP-2000 w/ 1.1.0.11 firmware |
| 5:37PM |
6 |
DELL PowerEdge 2850 and TE4110P and TE110P |
| 5:25PM |
0 |
Need help with Dial M option and destination context |
| 2:48PM |
0 |
News from France: Free, SIP and Asterisk |
| 1:55PM |
0 |
AstriCon Europe Update - 6 Weeks To Go |
| 1:47PM |
3 |
Having a Blonde moment. |
| 12:42PM |
2 |
Multiple Registers |
| 12:41PM |
2 |
Polycom 501 logo onscreen |
| 12:33PM |
0 |
Asttapi for Asterisk 1.2 Testers Needed (was RE:Asterisk TAPI - Outlook click2dial) |
| 12:28PM |
1 |
Delay when ringing internal extensions on incoming zap call |
| 12:27PM |
2 |
chan_capi-cm and dialing without number |
| 12:16PM |
1 |
change dchannel number |
| 11:57AM |
1 |
Asttapi for Asterisk 1.2 Testers Needed (was RE: Asterisk TAPI - Outlook click2dial) |
| 11:52AM |
0 |
Asterisk Broadvoice outbound calling loop, now it goes to voicemail |
| 11:11AM |
2 |
Using REGEX function |
| 10:13AM |
1 |
error leaving voicemail in multiple VM boxes |
| 10:05AM |
1 |
chan_capi-cm and type of number problem (ToN) |
| 9:38AM |
6 |
Netherlands zaptel.conf |
| 9:28AM |
1 |
EICON, chan_capi-cm and averlap receiving |
| 8:00AM |
4 |
WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic! |
| 7:55AM |
0 |
Re: [Astlinux-users] British English Female files ready for download |
| 7:40AM |
2 |
Meetme and authentication |
| 7:28AM |
1 |
crackling on IAX between asterisks |
| 6:51AM |
0 |
Reasons for a SIP channel to hang ? |
| 6:15AM |
0 |
Paging, Aastra 9133i, and Being on the phone! |
| 4:21AM |
2 |
call monitoring and indications / beeps |
| 3:16AM |
0 |
test -please ignore |
| 3:00AM |
1 |
regexp |
| 2:36AM |
0 |
call waiting announcement on agent phone |
| 2:35AM |
0 |
Join the Asterisk Video Task Force if you're into video telephony development! |
| 1:52AM |
4 |
asterisk and ODBC |
| 1:33AM |
0 |
iax2 disconnect problem |
| 1:15AM |
5 |
WiFi VoIP Handsets.. |
| 1:07AM |
1 |
tdm2400p: fax detection not working |
| 12:24AM |
0 |
problem with sip registration with database |
| |
| Monday May 15 2006 |
| Time | Replies | Subject |
|---|
| 11:24PM |
1 |
Tr: Re: The OpenNMS Group, Inc.: opennms and asterisk pbx |
| 10:47PM |
5 |
unicall dialing problem |
| 10:27PM |
2 |
Career Opportunities |
| 7:17PM |
4 |
Asterisk as a bridge between voip clients and POTS confrence bridge |
| 6:53PM |
1 |
Asterisk on a WRT54G? |
| 5:17PM |
2 |
Multiple announcements in a queue ?? |
| 5:13PM |
1 |
Outgoing Calls Not Working all the time |
| 4:50PM |
1 |
TDM400P static on call |
| 4:32PM |
2 |
Voicemail volume wav vs. wav49 |
| 4:09PM |
2 |
Is it possible to delete global variables |
| 3:54PM |
2 |
Asterisk X100P - Interrupt a call? |
| 3:43PM |
1 |
queue help |
| 2:21PM |
1 |
Asterisk didn't start with app_swift.so |
| 2:00PM |
0 |
Asterisk didn't start with |
| 1:58PM |
1 |
Please help.. I need a h323 user for tests |
| 1:57PM |
1 |
Encrypted IAX termination |
| 1:55PM |
1 |
Realtime Postgres via ODBC |
| 12:55PM |
0 |
SNOM autoanswer question |
| 12:30PM |
1 |
RE: [PROBLEM] Still exist --> DTMF Tones, occures in Asterisk - Channelwide |
| 12:16PM |
2 |
Asterisk with SIPconnect |
| 12:00PM |
0 |
Vancouver Asterisk Users Group |
| 11:53AM |
2 |
Which is the best fax-modem for testing ? |
| 11:13AM |
1 |
Please..... need some help |
| 10:52AM |
3 |
How to tell if RTP stream is has been reinvited? |
| 9:49AM |
4 |
Turning AAAH into a call-center |
| 9:47AM |
3 |
Eicon Diva - problems building new v3 melware driver |
| 8:17AM |
1 |
VOIP adapters to connect PSTN lines to SIP phones |
| 7:18AM |
1 |
GET DATA and STREAM FILE commands, don´t work |
| 6:55AM |
1 |
Broadvoice does it again |
| 6:51AM |
0 |
Ottawa Asterisk User Group Kickoff - Wed -- May 17 -- 5:00 |
| 6:50AM |
0 |
Echo cancel voip channel? |
| 6:36AM |
0 |
fax possible with standard modem |
| 6:26AM |
0 |
problem with sip registration ramdomly |
| 5:23AM |
0 |
Voicemail indication on Mitel 52xx phones |
| 2:33AM |
0 |
A sugestion for asterisk |
| 1:44AM |
0 |
agent deadlock |
| 1:40AM |
1 |
View Agent Status on the Web |
| 1:14AM |
1 |
VoIP Adapter |
| |
| Sunday May 14 2006 |
| Time | Replies | Subject |
|---|
| 11:25PM |
1 |
E1 + sangoma + soekris |
| 10:01PM |
1 |
Getting Realtime running (1.2.7.1) |
| 5:01PM |
1 |
Asterisk Manager interface |
| 4:01PM |
2 |
911 @ Zap Channel Breakin |
| 2:05PM |
0 |
VoipBuster issues? |
| 6:39AM |
0 |
[patch] fix for redirect manager action with BRIstuffed Asterisk |
| 6:01AM |
0 |
Re: h323.conf and realtime |
| 4:41AM |
0 |
IAX/SIP to germany with own callerid? |
| |
| Saturday May 13 2006 |
| Time | Replies | Subject |
|---|
| 11:39PM |
3 |
plainvoip - IAX2 call rejected |
| 11:36PM |
0 |
Contract Work : On-site NYC |
| 6:03PM |
1 |
Looking for Level 3 DID's, USA termination, USA 800 termination/Orig |
| 5:29PM |
0 |
Spam? Re: Cisco 7970 problems |
| 7:08AM |
1 |
Confused ! |
| 6:00AM |
0 |
Re: [asterisk-dev] SNMP support for Digium Cards |
| 5:46AM |
0 |
Re: [asterisk-dev] SNMP support for Digium Cards |
| 5:15AM |
0 |
Re: [asterisk-dev] SNMP support for Digium Cards |
| 4:59AM |
0 |
Re: [asterisk-dev] SNMP support for Digium Cards |
| 3:38AM |
0 |
parking a call /put on hold |
| 2:56AM |
0 |
RE: snmp and asterisk |
| |
| Friday May 12 2006 |
| Time | Replies | Subject |
|---|
| 11:59PM |
1 |
Sipura 1001 |
| 3:18PM |
3 |
VoiceMail application: "j" option not working as I supposed |
| 1:30PM |
4 |
fc5 and link to sources? |
| 12:37PM |
1 |
Cell phone dialed digits too short to be recognized by asterisk |
| 11:57AM |
4 |
DUNDi and Voicemail |
| 11:21AM |
6 |
voicemailmain() |
| 11:21AM |
1 |
Plain Text Passwords for IAX and SIP |
| 11:10AM |
3 |
Dial Command Reference for SIP channel |
| 10:32AM |
0 |
RE: snmp and asterisk |
| 10:30AM |
1 |
Having Rinback tone generation issues with 1.2.7.1 |
| 9:39AM |
1 |
Cisco 7970 problems |
| 9:11AM |
2 |
Help Avaya 4606 |
| 9:02AM |
5 |
Music on Hold restart at beginning for each call |
| 9:02AM |
2 |
Voicemail WAV to PDA Problems |
| 7:01AM |
2 |
URGENT please call parked / MOH |
| 6:06AM |
1 |
call parked / MOH |
| 5:19AM |
1 |
Speex fans? |
| 5:09AM |
1 |
S100-FX v2 audio quality |
| 5:05AM |
2 |
Sangoma A200D problem |
| 4:57AM |
2 |
email -> fax gateway with billing possibilities? |
| 4:41AM |
0 |
SCCP audio problems |
| 2:52AM |
0 |
RE: [PROBLEM] Still exist --> DTMF Tones, occures in Asterisk - Channelwide |
| 2:18AM |
0 |
extension.conf for overlap |
| 1:45AM |
3 |
Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card? |
| 1:07AM |
0 |
Alarmreciver finally found ATA |
| 1:05AM |
0 |
SIP/NAT disconnection issue |
| 12:59AM |
0 |
Sip domains, contexts and CHECKSIPDOMAIN |
| 12:39AM |
3 |
monitoring sangoma cards via snmp |
| 12:39AM |
0 |
Asterisk & BRI in the USA - Episode 2 "The Phantom Sales Rep" |
| 12:34AM |
1 |
TE110P on E1 |
| |
| Thursday May 11 2006 |
| Time | Replies | Subject |
|---|
| 11:27PM |
0 |
issue has arisen |
| 10:18PM |
4 |
Please Help Me...Urgent |
| 9:50PM |
1 |
Linksys IP Device Bulk Provisioning Guide |
| 9:03PM |
0 |
Asterisk + G.729 on Sun T1000/T2000 |
| 8:38PM |
1 |
Issue for RE-INVITE with G.729 |
| 8:08PM |
0 |
ast_dsp_call_progress |
| 7:40PM |
1 |
How many SER and asterisk servers does FWD users. |
| 6:33PM |
1 |
Canada Termination |
| 5:19PM |
0 |
Delete global variable |
| 4:00PM |
0 |
British Voice talent records Asterisk prompts |
| 3:56PM |
10 |
MeetME Conferencing |
| 3:55PM |
2 |
Problem setting locale for voicemail |
| 3:33PM |
0 |
Zap DTMF detection |
| 2:19PM |
1 |
Asterisk TAPI - Outlook click2dial |
| 1:52PM |
0 |
FW: Voicemail problem, not playing back |
| 1:36PM |
1 |
Re: Voicemail problem, not playing back |
| 1:31PM |
2 |
Paging and Auto Answer on Grandstream GXP2000 |
| 12:22PM |
3 |
sangoma A102 installation question |
| 10:44AM |
0 |
TE410P <=> Dialogic D/240SC-T1 |
| 10:03AM |
3 |
Asterisk and Brooktrout TR1000 |
| 9:24AM |
1 |
anyone doing voice audio detect VAD on analog lines |
| 9:19AM |
4 |
'extensions reload' clears Regextens |
| 9:18AM |
1 |
Voicemail problem, not playing back audio |
| 8:44AM |
1 |
budget tone 100 |
| 8:32AM |
8 |
Dialling a DUNDi Route |
| 6:45AM |
0 |
onsite tech for N Carolina and Boston |
| 6:43AM |
0 |
Directory by name access inside of voicemail |
| 5:50AM |
3 |
Call parking from legacy PBX over PRI?? |
| 5:16AM |
1 |
TigerNetwork IPH202A/B are OK ? |
| 5:01AM |
0 |
tdm400p card for sell (4xFXS) |
| 2:51AM |
1 |
Supervised Transfer how to do? |
| 1:03AM |
2 |
Eicon Diva Server - Fax and data modem support |
| 12:02AM |
1 |
mISDN trouble with a HFC Cologne card, Asterisk Asterisk 1.2.4 on Linux 2.6.16.11 - incoming DTMF detection |
| |
| Wednesday May 10 2006 |
| Time | Replies | Subject |
|---|
| 10:53PM |
0 |
Sip jitter buffer patch + Asterisk CallingCard |
| 9:44PM |
1 |
difference betwen a TE411P and TE410P |
| 9:25PM |
0 |
Sharing an outside line between a modem and a TDM400 |
| 8:53PM |
4 |
CentOS 4.x and ooh323 |
| 6:54PM |
2 |
REPOST: features.conf *1 Call Recording |
| 5:42PM |
1 |
mg3000-r fxo gateway provides more feature to work with asterisk |
| 5:28PM |
1 |
asterisk -rx 'sip show peers' |
| 5:06PM |
1 |
ISDN, TE205P, I'm goind crazy :> |
| 3:01PM |
1 |
ISDN Bridging with Bristuff |
| 2:16PM |
2 |
Headsets |
| 11:22AM |
0 |
QSIG suopprt in Asterisk |
| 10:04AM |
1 |
Web Admin |
| 9:49AM |
2 |
Is there a way to not propagate a context included inside other context? |
| 9:47AM |
2 |
One sided call |
| 8:00AM |
4 |
ethernet interface shares interrupts with tdm card |
| 7:39AM |
0 |
OH323 vs Panasonic IP Hybrid |
| 6:52AM |
1 |
Dropping Number on Dial Out |
| 6:18AM |
13 |
features.conf *1 Call Recording |
| 6:05AM |
0 |
Hints and busy lamps for phones registered on SER |
| 5:32AM |
2 |
No zap/sip/etc options? |
| 4:59AM |
1 |
ISDN and Asterisk |
| 4:49AM |
0 |
No audio in either direction on Zap -> SIP or SIP -> Zap calls |
| 3:05AM |
0 |
Realtime extension |
| 1:39AM |
2 |
asterisk monitoring / res_snmp |
| |
| Tuesday May 9 2006 |
| Time | Replies | Subject |
|---|
| 11:53PM |
1 |
How do I monitor the whole conversation on a Zap channel ... |
| 10:52PM |
0 |
MCC 1.4 released |
| 10:16PM |
0 |
How to make calls to US using Asterisk? |
| 9:12PM |
2 |
exten statement execution order |
| 7:46PM |
1 |
FW: Solid-PBX |
| 6:36PM |
0 |
asterisk and NEC SV7000S playing together? |
| 6:34PM |
3 |
Announcement: FOP 0.26 released |
| 5:57PM |
0 |
DID -> SER -> Asterisk call transfer |
| 3:45PM |
0 |
problem with hang up with TDM31B |
| 3:29PM |
1 |
PRI in Shanghai China |
| 2:58PM |
4 |
Caller ID forwarding |
| 2:57PM |
0 |
Cisco 2851 as T1 Gateway and Asterisk |
| 1:25PM |
0 |
Intellitouch ITC-3002 2line phones are ok? |
| 12:39PM |
5 |
voipjet down? |
| 12:36PM |
0 |
soft phone code |
| 12:31PM |
1 |
Sip and dbsecret |
| 12:23PM |
2 |
Configuring utstarcom1000 on asterisk |
| 11:42AM |
2 |
Incoming SIP or IAX2 via NAT |
| 11:38AM |
4 |
PSTN Incoming call on real line disrupts VoIPcall over DSL circuit - EXPLAINED |
| 11:36AM |
1 |
TE411P or TE410P |
| 10:28AM |
1 |
Call recordings management |
| 10:13AM |
0 |
How to distinguish between UNEXISTENT channels v/s UNAVAILABLE channels |
| 10:02AM |
1 |
Asterisk 1.2.7.1 and SIP registration |
| 9:51AM |
0 |
Best CPU (of expansion hardware?) for g.729 enc/dec ? |
| 9:33AM |
1 |
Many music on hold files |
| 8:36AM |
2 |
H323 calls will not stay connected |
| 7:58AM |
2 |
Asterisk on EM64T |
| 7:37AM |
1 |
Asterisk settings Net2Phone |
| 7:32AM |
0 |
Re: poor state of IAX2 code? (was: why a per fectlyfine iax2 host becomes UNREACHABLE?) |
| 7:25AM |
3 |
Transferring calls between two Asterisk Servers |
| 6:29AM |
0 |
SciTel Brix-QE card |
| 6:17AM |
1 |
Shared call recordings with ARI! |
| 5:46AM |
1 |
grandstream GXV-3000 |
| 5:31AM |
6 |
Bristuffed Asterisk: Hangup problems |
| 4:56AM |
0 |
Using ChanIsAvail and SIP |
| 4:29AM |
2 |
regarding freepbx |
| 4:14AM |
1 |
A@H Memory Limits |
| 2:24AM |
1 |
Asterisk Realtime with Oracle |
| 2:24AM |
2 |
EICON DIVA - which linux kernel |
| 1:49AM |
3 |
[SOLUTION] DTMF Tones occures in Asterisk |
| 1:39AM |
2 |
Problems with TDM400P and FXO modules |
| 1:17AM |
1 |
Best way to intercept an incoming call on asterisk 1.2 ? |
| 12:48AM |
0 |
Billing when forwarding incomming calls from SIP phone |
| |
| Monday May 8 2006 |
| Time | Replies | Subject |
|---|
| 9:16PM |
4 |
Asterisk documentation.. |
| 6:58PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday 05/13/2006 11:30am |
| 5:08PM |
0 |
Local Los Angeles VOIP equipment retailers? |
| 3:17PM |
1 |
MeetMe, async password requirements... |
| 1:12PM |
2 |
*.conf utilities for Asterisk |
| 11:42AM |
3 |
PSTN Incoming call on real line disrupts VoIP call over DSL circuit |
| 11:40AM |
4 |
transfer variables |
| 11:19AM |
1 |
Non-supervised pass-through |
| 11:03AM |
2 |
Asterisk/Zaptel 64-bit? |
| 10:31AM |
0 |
Looking for New Service Provider |
| 10:27AM |
1 |
Message on Hold |
| 10:06AM |
3 |
Most comprehensive management? |
| 9:36AM |
1 |
Running down an echo problem on outgoing calls |
| 9:26AM |
0 |
I: Dialstatus results |
| 8:22AM |
3 |
Expansion module |
| 8:05AM |
1 |
Dialing status detection |
| 7:43AM |
0 |
duration / billsec problem |
| 7:29AM |
0 |
AstLinux 0.4 Released - with build system |
| 7:17AM |
1 |
UpState NY SIP provider |
| 7:16AM |
1 |
How do I monitor a Zap channel ... |
| 6:47AM |
5 |
MySQL replication for voicemail |
| 6:12AM |
2 |
Dialstatus results |
| 5:58AM |
2 |
Quad ISDN card |
| 5:50AM |
0 |
(no subject) |
| 4:50AM |
1 |
Voicemail bomb |
| 4:47AM |
2 |
app_wakeme.c (Wake-up Call Manager) v0.1.0 released |
| 4:20AM |
1 |
[nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)] |
| 2:08AM |
3 |
Junghanns GSM card |
| 2:03AM |
0 |
Asterisk 1.2.x with app_rxfax |
| 12:29AM |
0 |
gxp-2000 Asterisk PSTN |
| 12:05AM |
0 |
iax2: dropping too many packets |
| |
| Sunday May 7 2006 |
| Time | Replies | Subject |
|---|
| 11:14PM |
0 |
Session Border Controller (SBC) |
| 11:03PM |
5 |
CallerID retain on internal transfer |
| 5:05PM |
0 |
Chanspy Specifying Agent not Working |
| 3:40PM |
2 |
SSH from System() ?... |
| 1:46PM |
2 |
Voicemail indication for analog phones |
| 12:46PM |
2 |
Need a Service that allows me to call Toll Free Outbound numbers |
| 8:56AM |
1 |
Announcement Haiku |
| 8:39AM |
0 |
canreinvite=no and codecs. |
| 3:23AM |
0 |
app_rxfax problem on 1.2.6 |
| 2:41AM |
0 |
more one asterisk hardware |
| 1:58AM |
1 |
another question about hardware for using with asterisk |
| 1:54AM |
1 |
Assterisk prompts |
| 1:09AM |
0 |
[Fwd: Re: asterisk hardware] |
| |
| Saturday May 6 2006 |
| Time | Replies | Subject |
|---|
| 3:54PM |
1 |
Upgrade SVN failed !!! |
| 1:26PM |
3 |
Voicemail error |
| 12:03PM |
3 |
www.SavaJe.com |
| 11:07AM |
1 |
Register Asterisk to FWD via SIP |
| 8:32AM |
0 |
Sipura register with FWD every 60sec |
| 4:42AM |
6 |
TDM4xxP |
| 2:56AM |
0 |
Gigabit Ethernet with multiple VLAN's or Fast Ehternet and with two separate cards? |
| |
| Friday May 5 2006 |
| Time | Replies | Subject |
|---|
| 6:47PM |
1 |
Multiple periodic announcements in queues? Possible? |
| 6:07PM |
0 |
CW options not changing |
| 5:35PM |
3 |
How to determine if a device is in a call |
| 4:44PM |
2 |
Info |
| 4:29PM |
5 |
Silent Attendant |
| 4:06PM |
5 |
ASTERISK DISA FOR INCOMING DID CALL |
| 4:02PM |
0 |
REGISTER that isn't a register |
| 3:34PM |
0 |
Passing Callerid |
| 3:24PM |
0 |
ODBC Voicemail storage and app_directory |
| 3:03PM |
0 |
asterisk behind load-balancing switch |
| 2:48PM |
1 |
Bandwidth via my Asterisk PBX |
| 2:38PM |
1 |
Asterisk <--> NAT <--> Internet <--> NAT <--> Sipura-3K (No Asterisk) |
| 2:31PM |
0 |
300 DID's required in Alpine Texas Area code 432 |
| 2:06PM |
5 |
Code parsing error? |
| 12:46PM |
0 |
Passing SIP Subscriptions??? |
| 12:40PM |
10 |
Call Center Phone with Auto Answer |
| 11:19AM |
0 |
AASTRA 9133i and PIX Firewall |
| 10:43AM |
1 |
Spam? Re: Cisco 7970 running SIP question |
| 8:55AM |
0 |
Spam? Re: Cisco 7970 running SIP question |
| 8:18AM |
1 |
Cisco 7970 running SIP question |
| 8:16AM |
0 |
Problem on Zap Channel with IVR |
| 8:07AM |
0 |
Access to sip.conf username field from dialplan |
| 7:47AM |
0 |
asterisk 1.2 & hisax teles 16.3 isa |
| 7:34AM |
1 |
Realtime, 2 server setup problem? |
| 7:24AM |
0 |
CARD.XML for MGCP cisco phone |
| 7:21AM |
0 |
Call Transfer Disconnect (CT-5) |
| 6:58AM |
1 |
problem g729 |
| 6:57AM |
6 |
Dumping queue_log to MySQL |
| 6:31AM |
1 |
Registering Remote Sipura to Asterisk (both behind firewall) |
| 6:17AM |
0 |
Repost: External voicemail and MWI on internal phone |
| 6:16AM |
0 |
Call Hold and Retrieve |
| 4:56AM |
0 |
Re: Asterisk-Users Digest, Vol 22, Issue 26 |
| 4:23AM |
2 |
AW: AW: DTMF detection when outgoing call tomobilephones |
| 1:36AM |
0 |
DTMF Tones within my Asterisk on all type of Channels |
| |
| Thursday May 4 2006 |
| Time | Replies | Subject |
|---|
| 11:17PM |
2 |
Asterisk on amd SERVER |
| 9:53PM |
0 |
Is FWD down ??? |
| 7:26PM |
0 |
SPA941 et al LED indications |
| 3:41PM |
0 |
asterisk can't find address host. Problem in chan_sip.c |
| 3:13PM |
1 |
Fwd: meetme conference latency degrades... |
| 2:33PM |
4 |
why a perfectly fine iax2 host becomes UNREA CHABLE? |
| 2:21PM |
0 |
asterisk <-> SIP provider, two way connection |
| 1:48PM |
0 |
Voicemail records funny - Asterisk 1.2.7.1 |
| 1:14PM |
1 |
Help with IRQ conflict between wct2xxp and eth0 |
| 1:12PM |
2 |
Unable to get TDM400p working |
| 12:51PM |
3 |
Volume configuration on Polycom Soundpoint 501 phone |
| 12:03PM |
1 |
Switchboard solutions, interactions with handset |
| 12:02PM |
0 |
Realtime rtignoreexpire bugged ?? |
| 11:45AM |
5 |
Tool for Polycom configurations |
| 11:23AM |
0 |
TE410P & T400P together in a server |
| 11:14AM |
0 |
Soonr |
| 11:11AM |
0 |
OT: D-link DI-102 |
| 9:19AM |
0 |
remapping sof-keys on Polcyom 301 |
| 8:31AM |
4 |
why a perfectly fine iax2 host becomes UNREACHABLE? |
| 8:11AM |
0 |
disa and caller id |
| 8:06AM |
4 |
AW: DTMF detection when outgoing call to mobilephones |
| 7:10AM |
2 |
DTMF detection when outgoing call to mobile phones |
| 6:46AM |
2 |
SV: Polycom 501 - Disable DND feature? |
| 5:50AM |
0 |
SpeedDial on GXP-2000 |
| 4:41AM |
2 |
PCI voltage |
| 4:35AM |
1 |
Unwanted conference with snom320 and asterisk 1.07bristuffed |
| 3:45AM |
1 |
Meetme from MySQL |
| 3:39AM |
0 |
Internet exposed asterisk server. |
| 3:34AM |
3 |
number that starts with star on PAP2 |
| 3:27AM |
1 |
Pattern matching DISA |
| 3:19AM |
5 |
ISAC support? |
| 2:16AM |
0 |
AW: SIP Phones behind dynamic IPs |
| 1:59AM |
0 |
SetGroup and CheckGroup. Need some help on the dialplan |
| 1:53AM |
3 |
SPA941 SPA942 BUG. auto answer does not work. |
| 1:09AM |
0 |
Using console channel with specific codec only |
| 1:02AM |
0 |
Unwanted conference with snom320 and asterisk 1.07 bristuffed |
| 12:40AM |
1 |
TDM400P and monoBRI auto-dial call difference: caller phone does not ring |
| 12:22AM |
1 |
Polycom 501 - Disable DND feature? |
| |
| Wednesday May 3 2006 |
| Time | Replies | Subject |
|---|
| 10:55PM |
0 |
Extension '' in context 'whatever' from '123456789' does not exist. |
| 6:57PM |
0 |
Running applications when a queued callisanswered |
| 6:48PM |
3 |
meetme conference latency degrades... |
| 5:08PM |
1 |
Running applications when a queued call isanswered |
| 4:28PM |
0 |
Vodini & * |
| 4:17PM |
3 |
hyperthreading and zaptel |
| 2:30PM |
3 |
Setting QUEUE_PRIO |
| 2:07PM |
1 |
dialing FXO gives wrong billsec |
| 1:55PM |
1 |
How would you go about calling a list of numbers and 'speaking' a message? |
| 12:14PM |
0 |
SIP w/NAT on Grandstream 496 and Call-Waiting |
| 11:50AM |
1 |
my asterisk crashed |
| 11:43AM |
1 |
echo in Snom 360 phones |
| 10:58AM |
0 |
RE: [asterisk-biz] Colocation Denmark |
| 10:46AM |
0 |
Colocation Denmark |
| 9:46AM |
0 |
Forwarded Numbers and Timeouts |
| 9:41AM |
0 |
Selecting the outbound port from FXO device |
| 9:22AM |
1 |
Voipjet Problem? |
| 9:15AM |
2 |
SIP Phones behind dynamic IPs |
| 9:08AM |
0 |
G.722 Softphone? |
| 8:45AM |
4 |
QSIG support in Asterisk |
| 8:04AM |
1 |
Running applications when a queued call is answered |
| 7:58AM |
0 |
Listening a conversation |
| 7:36AM |
1 |
LDAPget |
| 6:32AM |
13 |
Can I recreate a Fax from a recorded file? |
| 5:45AM |
0 |
Limit on number of SIP channels? |
| 5:10AM |
2 |
Simple Dell Computers |
| 4:43AM |
0 |
Phone UNREACHABLE: Plays "agent-incorrect" to Queue-caller ?? |
| 3:19AM |
1 |
asterisk intergration in third party web application |
| 3:03AM |
0 |
mysql failures handling |
| 2:16AM |
0 |
Future pickup feature |
| 2:12AM |
0 |
Which distro for Intel D915GAG-L ? |
| 2:02AM |
0 |
Asterisk SRPMs and patches |
| 2:00AM |
1 |
brittle IAX connections ? |
| 1:08AM |
3 |
Huawei EP201S |
| 1:07AM |
0 |
Asterisk auto-dial out: behaviour difference between analog and ISDN channel |
| 12:58AM |
0 |
GXP2000 provisioning: what is cfg.txt file? |
| 12:52AM |
0 |
Can't compile ael_lex.c on HEAD |
| |
| Tuesday May 2 2006 |
| Time | Replies | Subject |
|---|
| 11:24PM |
1 |
SV: How does asterisk behave when multiple phonesare logged in on a single SIP/account? |
| 10:32PM |
1 |
Unicall MFC/R2 B3,B4 and clear back |
| 8:54PM |
0 |
OT - but relevant |
| 8:35PM |
0 |
Asterisk Imposter binary |
| 8:17PM |
3 |
Queue reporting seems broken. |
| 8:17PM |
0 |
asterisk hung again |
| 7:42PM |
0 |
Grandstream GXP-2000 call end |
| 6:12PM |
2 |
PAP2/Sipura XML Provisioning File |
| 5:47PM |
0 |
Insights on SIP channel usage in * 1.2.7.1 are welcome! |
| 2:40PM |
0 |
Half hangup issue |
| 1:40PM |
0 |
PRI Transfer Disconnect |
| 1:30PM |
1 |
Sangoma Card Question |
| 1:28PM |
0 |
Help with multiple company setup |
| 1:14PM |
4 |
Asterisk technician needed in Buenos Aires Argentina |
| 1:13PM |
0 |
Ringing extensions in a call group. |
| 12:00PM |
0 |
The CAVP is now accepting memberships applications |
| 11:11AM |
0 |
Using qualify=yes guarantees failure on iax2 behind NAT (was: RE: Using frequent keepalives to eliminate needforNAT port forwarding?) |
| 9:26AM |
0 |
Telasip config problem/question |
| 9:02AM |
0 |
Need help configuring TE100P and 3 X100P clonewith MD3200 chipset |
| 8:36AM |
1 |
Need help configuring TE100P and 3 X100Pclonewith MD3200 chipset |
| 8:35AM |
0 |
Commands possible in the h extension, message delivery with confirmation |
| 8:35AM |
2 |
Speeding up UK BT incoming call detection |
| 8:17AM |
3 |
Sip show inuse |
| 7:47AM |
2 |
dnd error message in the log |
| 7:38AM |
3 |
Need help configuring TE100P and 3 X100P clone with MD3200 chipset |
| 6:54AM |
2 |
Need help in asterisk fax |
| 6:32AM |
1 |
SIP trunk ring tone |
| 5:17AM |
8 |
Zapata Telephony interface and torisa module error |
| 3:22AM |
0 |
MeetAsterisk London and Brussels |
| 2:19AM |
1 |
Questions on ANI |
| 2:08AM |
3 |
asterisk with Dialogic BRI /2VFD |
| 12:44AM |
4 |
Under which project , auto-dial feature comes |
| 12:09AM |
1 |
Meetme volume increase/decrease |
| 12:00AM |
0 |
2 process running concurrent in dialplan |
| |
| Monday May 1 2006 |
| Time | Replies | Subject |
|---|
| 11:57PM |
0 |
Re: 482 Loop Detected on sip calls |
| 11:56PM |
2 |
How does asterisk behave when multiple phones are logged in on a single SIP/account? |
| 11:40PM |
1 |
/var/spool/asterisk/outgoing/ prematurely hangingup |
| 10:43PM |
1 |
/var/spool/asterisk/outgoing/ prematurely hanging up |
| 10:37PM |
1 |
unable to set outgoing callerid |
| 5:35PM |
1 |
Using frequent keepalives to eliminate need forNAT port forwarding? |
| 5:25PM |
1 |
Using frequent keepalives to eliminate need for NAT port forwarding? |
| 4:14PM |
0 |
Cisco 2621 router for voice and data? |
| 4:12PM |
0 |
wellgate 38XX with VAD and call files |
| 4:00PM |
2 |
SPA-1001 behind NAT -- mucho hair pulling |
| 2:49PM |
1 |
Music on Hold from Soundcard |
| 2:24PM |
1 |
Polycom SoundPoint 501 + Asterisk |
| 2:19PM |
0 |
Spam? Re: CallerID Name problem |
| 2:06PM |
0 |
app_icd |
| 1:45PM |
0 |
Asterisk-Users Digest, Vol 22, Issue 1 |
| 1:16PM |
3 |
Digium TDM400P vs Sangoma A200 for 2 x FXO |
| 10:51AM |
1 |
Listening on one IP and binding to other IP - is this possible ? |
| 10:41AM |
6 |
Problems with zaptel and TE210P |
| 10:12AM |
0 |
Can i use same group with 2 or more hfc-cards ? |
| 9:45AM |
0 |
7941G - Any success stories? |
| 9:36AM |
12 |
CallerID Name problem |
| 9:24AM |
1 |
voicemail dialout |
| 9:17AM |
3 |
auto-dail for ZAP channel, the application gets executed before the call attended |
| 9:00AM |
1 |
Softphone ready to go installed on USB flash drive |
| 6:38AM |
1 |
GXP-2000 Message Waiting Light |
| 6:32AM |
0 |
Sangoma A200 preventing Zap channels |
| 6:05AM |
4 |
Cant get voicemail |
| 5:54AM |
1 |
Is there a way to monitor DTMF tones in a channel? |
| 5:48AM |
0 |
anyone have solution to dtmf problem in console driver? |
| 5:30AM |
1 |
Frappr mapper |
| 5:24AM |
0 |
Asterisk Bugs? |
| 4:07AM |
1 |
Cepstral , options to read the contents of a file |
| 3:48AM |
1 |
Auto-Dial , problem in calling Application , Guidance requested |
| 3:40AM |
1 |
Anyone willing to share an Australian dialplan.xml file for Cisco phones? |