| Monday March 31 2008 |
| Time | Replies | Subject |
|---|
| 10:01PM |
3 |
Need some input for Quad T1 and channel banks. |
| 8:45PM |
7 |
Cisco 7965 SIP Firmware |
| 8:00PM |
1 |
Gentilini, Paul is out of the office. |
| 7:33PM |
1 |
How to give user a prompt before connecting thecall |
| 6:54PM |
0 |
CDR Timestamps (cdr-custom) |
| 6:50PM |
1 |
Control of RTP open ports |
| 6:23PM |
0 |
How to give user a prompt before connecting the call |
| 5:12PM |
1 |
AsterPas ObjectPascal Based FastAGI Server goes Open Source |
| 4:52PM |
0 |
SIP proxy screwing up peer addresses. |
| 3:47PM |
3 |
Simple Question |
| 3:08PM |
1 |
asterisk-users Digest, Vol 44, Issue 104 |
| 2:53PM |
0 |
Problem with VoiceMailMain |
| 2:33PM |
0 |
Tests in VMWare (was: Re: asterisk-users Digest, Vol 44, Issue 104) |
| 2:25PM |
0 |
How to customize voicemail greeting |
| 1:23PM |
0 |
transfer call |
| 12:32PM |
2 |
UK FXO hangup detection with a twist |
| 12:06PM |
0 |
ENUMLOOKUP question. |
| 9:07AM |
0 |
Broken calls during conversation |
| 9:02AM |
1 |
The most efficient way to know SIP phones IP addresses ? |
| 8:44AM |
0 |
No voice in one direction, SIP, call manager |
| 8:14AM |
0 |
applicationmap in features.conf Asterisk 1.2 is ignoring DIAL tT options |
| 1:36AM |
0 |
Advice on queue setup needed please. |
| 12:50AM |
2 |
Tests in VMWare |
| |
| Sunday March 30 2008 |
| Time | Replies | Subject |
|---|
| 6:35PM |
1 |
audio disappeared after ztdummy install |
| 5:14PM |
0 |
ztdummy / RTC error |
| 8:56AM |
1 |
breaking DNID into country code, area code, and local code |
| 8:31AM |
2 |
How many maximum SIP Registrations can Asterisk Handle |
| |
| Saturday March 29 2008 |
| Time | Replies | Subject |
|---|
| 10:38PM |
1 |
e164.org |
| 2:00AM |
2 |
Finding iaxy's (iaxies?) |
| |
| Friday March 28 2008 |
| Time | Replies | Subject |
|---|
| 11:04PM |
0 |
Asterisk 1.4.19-rc4 and 1.6.0-beta7 Now Available |
| 9:36PM |
2 |
New Tutorial: Asterisk on EPIA VIA C3 |
| 7:03PM |
1 |
how to register IAX user without password for any user |
| 6:06PM |
0 |
More info on my previous dynamic queue question |
| 6:00PM |
2 |
voicemail custom greeting |
| 5:22PM |
1 |
Question on Dynamic Queue and Agent |
| 5:09PM |
1 |
Grandstream BLF and Call-limit |
| 5:09PM |
0 |
overlap calls from NT-BRI timeout problem |
| 3:31PM |
1 |
jingle with Asterisk + PSTN |
| 2:34PM |
1 |
how to register IAX user without password |
| 1:40PM |
1 |
recommendable softphones / X-Lite / Zoiper for amd64? |
| 1:27PM |
3 |
Two phones fail to agree on codec, asterisk at fault? |
| 1:02PM |
2 |
wrong extension status when call-limit=1 is used |
| 11:30AM |
1 |
sip.conf setvar option |
| 10:46AM |
1 |
PRI error cause hangup calls |
| 10:13AM |
1 |
IAX user register problem |
| 8:25AM |
2 |
Call deflection on ISDN PRI in Sweden |
| 6:54AM |
1 |
Need help with voicemail odbc |
| 4:38AM |
4 |
Newbie Polycom: DHCP/boot server supporting 2 models of phones |
| 2:12AM |
6 |
SPA-962+ SPA-932- blf function |
| |
| Thursday March 27 2008 |
| Time | Replies | Subject |
|---|
| 11:13PM |
0 |
Developer Conference, Aug 5-7, Chicago |
| 9:22PM |
1 |
Problem when leaving voicemail |
| 8:23PM |
3 |
Star Wars Echo Sound |
| 8:00PM |
1 |
Help with cisco 7960 phone |
| 4:40PM |
1 |
Asterisk not picking up (some) calls due to zaptel detecting and clearing alarms |
| 4:32PM |
2 |
callers in queue passed to agents who accept only one call at a time |
| 3:56PM |
0 |
Had it with Dell Garbage - HP Question |
| 12:14PM |
1 |
Unable to establish handshaking with fax machine |
| 12:02PM |
0 |
Astlinux Friday Mar 28 @12 Noon EDT VoIP Users Conference |
| 11:02AM |
1 |
Cisco 7971 |
| 10:08AM |
5 |
Problem with socket_process: Call rejected by 127.0.0.1: Busy |
| 8:04AM |
2 |
IAXy device |
| 7:58AM |
4 |
SPA400 vs Rhino/Digium card |
| 7:47AM |
2 |
Calling users to the external domain using Asterisk |
| 7:36AM |
1 |
ADPCM codec and IAXy device |
| 7:28AM |
1 |
Asterisk not hanging up after voicemail |
| 4:00AM |
1 |
Zapata Tormenta 2 |
| 3:45AM |
3 |
problem about voice when using TDM2400p with VPMADT032 echo canceller module |
| 3:31AM |
0 |
How to change Internal and external callerid |
| |
| Wednesday March 26 2008 |
| Time | Replies | Subject |
|---|
| 10:59PM |
1 |
Got SIP response 406 "Not Acceptable" |
| 10:55PM |
0 |
Strange RTP problem... |
| 10:06PM |
2 |
DTMF suddenly stopped working on SIP channel |
| 7:10PM |
1 |
AGI-python script |
| 6:21PM |
2 |
customizing faxrcvd in PHP |
| 3:02PM |
1 |
Playing sound while talking |
| 1:18PM |
2 |
Dialing off-hook with Polycom SoundPoint IP 430 |
| 12:12PM |
0 |
Test |
| 11:30AM |
5 |
Asterisk parking hold and transferdigittimeo ut |
| 10:26AM |
2 |
UK GMT/BST settings |
| 7:44AM |
0 |
Fax to DB |
| 7:25AM |
1 |
what's a softphone can activer web browser |
| 6:26AM |
0 |
SOLVED: Avantfax on Debian |
| 6:14AM |
2 |
Avantfax installation on Debian |
| 3:27AM |
2 |
Broadcast/Announce app |
| 12:33AM |
2 |
Realtime replication!!!!! |
| |
| Tuesday March 25 2008 |
| Time | Replies | Subject |
|---|
| 10:46PM |
0 |
Automatically reload/restart asterisk following IP change (dynamic IP) |
| 9:11PM |
0 |
Distorted Audio for incoming DTMF |
| 7:31PM |
1 |
SIP Domain Authentication |
| 6:44PM |
1 |
Send received fax to different email account |
| 6:31PM |
2 |
RTP Payload Problem |
| 5:43PM |
1 |
Have problem with realtime sql |
| 4:19PM |
2 |
Slightly OT: Getting VOIP number into phone book |
| 3:54PM |
2 |
voicemail.conf fromstring, emailbody - per context? |
| 3:28PM |
0 |
Auto-congest time for sip peers |
| 1:50PM |
0 |
To what degree can Asterisk replace Cisco Unity? |
| 1:48PM |
1 |
Asterisk parking hold and transferdigittimeout |
| 1:35PM |
2 |
CCM and multiple trunks |
| 1:19PM |
1 |
Delete voicemail messages on asterisk by replying to email |
| 12:51PM |
1 |
How to obtain SIPCHANINFO variables within custom application? |
| 10:32AM |
1 |
force soft hangup |
| 6:05AM |
0 |
New Release asterisk 1.6 Beta |
| 4:04AM |
1 |
Sip exten matching based on contact: sip header? |
| 3:16AM |
0 |
Redirect and free the channel |
| 2:59AM |
1 |
Audio Problem... |
| 1:47AM |
2 |
Menuselect? |
| |
| Monday March 24 2008 |
| Time | Replies | Subject |
|---|
| 11:40PM |
4 |
SIP carrier billing technicalities |
| 7:59PM |
0 |
FYI about my Mona Vie business venture - apology and rethink |
| 7:12PM |
1 |
Passing variables over IAX2 -- IAXVAR patch? |
| 6:56PM |
7 |
FYI about my Mona Vie business venture |
| 6:14PM |
4 |
estimation on phone network capacity |
| 5:03PM |
0 |
Slow compilation speed |
| 3:56PM |
3 |
Dynamic meetme conference creation with Authenticate (Asterisk 1.4.0) |
| 3:36PM |
1 |
G.729 Copy Protection |
| 2:03PM |
1 |
app_sms and smsq in germany |
| 1:05PM |
1 |
Gold Mine CRM + Asterisk |
| 11:41AM |
1 |
Calling extension from CLI? |
| 11:29AM |
0 |
Queue - Dynamic association |
| 10:48AM |
0 |
Greetings |
| 9:42AM |
1 |
g729 license for debian etch |
| 9:02AM |
2 |
Getting Exec Format Error when running AGI call |
| 6:38AM |
3 |
Unable to obtain dialed number through ZAP |
| 3:02AM |
3 |
How to capture destination number when receive call through ZAP |
| |
| Sunday March 23 2008 |
| Time | Replies | Subject |
|---|
| 6:51PM |
0 |
Problems with calls in asterisk. |
| 6:41PM |
1 |
zap callerid problem |
| 6:22PM |
1 |
Storing voicemail in mysql |
| 3:08PM |
1 |
How to detect if a call is fax or not |
| 2:54PM |
3 |
Unable to capture CallerID through Zap |
| 10:05AM |
6 |
Access rights between AGI and Web server? |
| 7:45AM |
2 |
More Broadvoice woes. Who's fault could this be? |
| 7:28AM |
1 |
No audio on Sangoma A104. |
| 6:28AM |
1 |
voicemail and needed language to be selected |
| |
| Saturday March 22 2008 |
| Time | Replies | Subject |
|---|
| 7:21PM |
3 |
G723 on asterisk 1.4.1 |
| 3:21PM |
2 |
Problem: Digium TDM400 with XOptionsFlex - Line Busy |
| 9:08AM |
2 |
Anyone used Siemens SIP/Dect phones? |
| 8:59AM |
1 |
how to detect redirect fax call |
| 1:22AM |
0 |
--POSIBLE SPAM-- Peticion on line Parque de atracciones mcdonals |
| |
| Friday March 21 2008 |
| Time | Replies | Subject |
|---|
| 11:36PM |
1 |
----www.cdsportal.net---- wholesale voip provider |
| 9:13PM |
4 |
Calls to sip extensions not defined |
| 5:37PM |
1 |
----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min |
| 4:08PM |
3 |
Problem with user regsitration and ldap on SVN version |
| 3:20PM |
2 |
----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min |
| 3:12PM |
1 |
TxFax in asterisk 1.4 |
| 3:04PM |
0 |
Hardware supporting groundstart signalling |
| 2:57PM |
2 |
Digium registration utility version 3.0.3 released |
| 10:38AM |
1 |
Which command line is used to send emails to notify incoming voicemail ? |
| 10:07AM |
0 |
DRUID/Voiceroute on VoIP Users COnference today Friday 2008-03-21 at 11:58 AM EDT |
| |
| Thursday March 20 2008 |
| Time | Replies | Subject |
|---|
| 11:10PM |
0 |
BLF on Cisco 7970 |
| 7:09PM |
1 |
polarity in zapata.conf |
| 5:53PM |
8 |
BLF and Snom phones |
| 5:40PM |
1 |
423 "Interval Too Brief" and expiry settings in sip.conf |
| 4:18PM |
1 |
Polycom 650 |
| 3:45PM |
1 |
More DTMF issues |
| 3:07PM |
1 |
Dialplan Help |
| 1:10PM |
0 |
question on app_conference() |
| 9:46AM |
0 |
Asterisk re-invites and billing |
| 8:32AM |
2 |
hint status unavailable |
| 6:48AM |
1 |
Unable to build smsq on beta6 and x86_64. |
| 5:13AM |
1 |
Newbie: Two problems with Asterisk Config, Please Help |
| 4:16AM |
0 |
AMD timing issues |
| 3:48AM |
6 |
Newbie IVR: How to read() before playback() is finished? |
| 1:06AM |
1 |
Newbie Asterisk: Disaster Recovery Proof Asterisk |
| |
| Wednesday March 19 2008 |
| Time | Replies | Subject |
|---|
| 10:32PM |
1 |
Call All |
| 9:01PM |
1 |
Multiple sites, same extension |
| 8:14PM |
0 |
Chanspy from a call file |
| 7:41PM |
2 |
Is Asterisk ready for Prime-Time? |
| 7:06PM |
0 |
Inband SIP DTMF |
| 6:12PM |
0 |
question on meetme |
| 5:48PM |
3 |
phpagi |
| 4:46PM |
1 |
Ribbit Demo |
| 4:05PM |
0 |
Can't play recording message wav file |
| 3:13PM |
3 |
How to configure Voice mail for multi users. |
| 3:10PM |
0 |
How configure Voice mail for multi users. |
| 1:41PM |
1 |
Bug in voicemail's serveremail setting in 1.4.18 |
| 1:28PM |
0 |
question how to play wave file in meetme and exit |
| 11:49AM |
1 |
fxo tdm400p issue |
| 11:34AM |
8 |
Limit calls when using autodial |
| 9:04AM |
1 |
Query about Bluetooth Head phone |
| 6:11AM |
1 |
Newbie Queue: greetings when first joining queue |
| 5:45AM |
0 |
Deadair in queues. |
| 5:36AM |
1 |
Getting config from SPA-941 or 942 phones |
| 5:01AM |
1 |
Call Screening feature using asterisk |
| 3:03AM |
6 |
Hardphone SIP phone costs |
| 1:30AM |
2 |
Asterisk with lumenvox |
| 12:02AM |
0 |
Asterisk and Avaya 4610 handset |
| |
| Tuesday March 18 2008 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
AST-2008-005: HTTP Manager ID is predictable |
| 11:32PM |
0 |
AST-2008-004: Format String Vulnerability in Logger and Manager |
| 11:29PM |
0 |
AST-2008-003: Unauthenticated calls allowed from SIP channel driver |
| 11:26PM |
0 |
AST-2008-002: Two buffer overflows in RTP Codec Payload Handling |
| 9:35PM |
0 |
AEL2 Hint & Parking |
| 8:54PM |
2 |
(Critical Updates) Asterisk 1.2.27, 1.4.18.1, 1.4.19-rc3, 1.6.0-beta6 Released |
| 8:22PM |
1 |
Sip Line Status/Pickup |
| 5:55PM |
3 |
capacity |
| 5:37PM |
1 |
Call forward on Telco line |
| 4:11PM |
2 |
How is uniqueid computed |
| 3:28PM |
2 |
ztdummy problem causing playback () to fail |
| 2:06PM |
6 |
Call signalling on BT FeatureLine Compact (Sangoma A200) |
| 1:29PM |
1 |
Using dedicated eth2 card for SIP trunk line to ISP provider - how to setup ? |
| 12:02PM |
1 |
Sangoma FXO/FXS config |
| 9:40AM |
6 |
Asterisk 1.4 reliability problems |
| 7:20AM |
3 |
Newbie Queue: Simple Queue Problem |
| 6:24AM |
2 |
call screening feature |
| 1:31AM |
0 |
Way to hangup a call between two asterisk machines |
| 1:14AM |
0 |
Subexpression usage in Asterisk Dialplan Regular Expressions |
| |
| Monday March 17 2008 |
| Time | Replies | Subject |
|---|
| 10:29PM |
2 |
Pre-pending certain digits (like 9) to an outbound call number |
| 8:44PM |
2 |
Order of queue member list |
| 7:42PM |
1 |
Core dump? |
| 6:45PM |
1 |
asterisk.conf uniquename or sysname for uniqueid field in CDR |
| 5:50PM |
2 |
php web chat + asterisk -> callcenter |
| 5:08PM |
1 |
Turn off MusicOnHold for individual User |
| 2:57PM |
4 |
MeetMe option b |
| 12:47PM |
0 |
The switch statement in extensions.conf |
| 11:17AM |
1 |
ldap for sip users. |
| 9:26AM |
0 |
NVFaxDetect and Asterisk 1.6 |
| 7:14AM |
6 |
Handling 3 different call ending causes |
| 7:08AM |
1 |
Desperately need help with Asterisk setup |
| 6:06AM |
1 |
update_call_counter: Call to peer '2509' rejected due to usage limit of 1? |
| 6:00AM |
3 |
Newbie Polycom: DND answered as "on the phone" instead of "not available" |
| 1:09AM |
1 |
Redundant Voicemail |
| 12:46AM |
2 |
Newbie ASTDB: cannot replicate among Asterisk servers? |
| |
| Sunday March 16 2008 |
| Time | Replies | Subject |
|---|
| 10:11PM |
1 |
using the System() command to call a script |
| 8:10PM |
1 |
Problem with incoming calls on Broadvoice after upgrade to 1.4.18 |
| 7:53PM |
0 |
DUNDi or ENUM |
| 6:16PM |
0 |
Telemarketer Torture.... (was: Re: asterisk-users Digest, Vol 44, Issue 49) |
| 5:13PM |
0 |
cordless usb handsets: Uniden Win1200? |
| 1:49PM |
1 |
LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48) |
| 3:08AM |
4 |
Telemarketer Torture.... |
| |
| Saturday March 15 2008 |
| Time | Replies | Subject |
|---|
| 10:25PM |
1 |
Calling a Macro with arguments in AstApplication/AstApplicationData |
| 9:53PM |
2 |
Asterisk VOIP Jobs version 2 Launched! |
| 9:19AM |
1 |
Sip phones for call centers |
| 9:00AM |
0 |
IAX ATA that can be set up and drop shipped to a US address? |
| 12:13AM |
2 |
DID T1 PRI |
| |
| Friday March 14 2008 |
| Time | Replies | Subject |
|---|
| 10:43PM |
1 |
Callerid Error- Causing All Zap Channels Busy |
| 8:57PM |
1 |
Looking for a cheap SIP termination point. |
| 7:27PM |
0 |
FW: [asterisk-dev] Hardware and CentOS tweaks. |
| 7:27PM |
0 |
FW: [asterisk-dev] Call failed, reason 0 explanation. |
| 6:12PM |
1 |
Trouble with Incoming Callerid on Trixbox |
| 1:56PM |
2 |
Logs for Call generated by Manager API |
| 10:46AM |
0 |
CallerID(num) not showing on cli |
| 9:28AM |
0 |
VoIP Users Conference for Friday March 14th @ 12 Noon EDT |
| 7:29AM |
3 |
Dialing patterns and "GSM" format numbers |
| 6:02AM |
1 |
Group Listen on SIP Phone |
| 5:59AM |
3 |
Anyone know of a pass through ATA |
| 4:51AM |
2 |
Asterisk 1.6 |
| |
| Thursday March 13 2008 |
| Time | Replies | Subject |
|---|
| 10:22PM |
1 |
Multiple clients registering on same definition in Realtime |
| 9:09PM |
1 |
Hardware Platform |
| 9:04PM |
5 |
Mail Server |
| 8:54PM |
1 |
Help: DTMF problem |
| 8:32PM |
2 |
RedFone foneBRIDGE2 2e1 - anyone used it oranother TDMoE bridge? |
| 7:38PM |
2 |
RedFone foneBRIDGE2 2e1 - anyone used it or another TDMoE bridge? |
| 6:19PM |
4 |
Application registration on Asterisk 1.4 and 1.6? |
| 6:01PM |
1 |
CallerID setting issue with withheld numbers and mISDN ... |
| 5:38PM |
0 |
Error in Callback CDR |
| 5:17PM |
0 |
need * consultant in st louis area |
| 2:28PM |
2 |
SNOM on "Do Not Call" list???? |
| 2:17PM |
2 |
queue log vs. cdr |
| 2:13PM |
1 |
sip.conf help, inbound calls fall to last specified context |
| 11:22AM |
0 |
OT: RTP - NAT - SBC |
| 4:59AM |
3 |
Newbie One-touch Recording: Does not work (more info) |
| 4:39AM |
5 |
Newbie One-touch Recording: Does not work |
| 3:47AM |
3 |
How to find out the IP of the calling party? |
| 3:16AM |
1 |
T.38 SIP Issues |
| 1:31AM |
1 |
asterisk out of service |
| |
| Wednesday March 12 2008 |
| Time | Replies | Subject |
|---|
| 9:27PM |
0 |
test please ignore |
| 8:12PM |
0 |
Problem sending CallerID Name to Dialogic based phone app |
| 7:13PM |
4 |
does the meetme module still require an external timing source? |
| 6:20PM |
0 |
SIP Registration! |
| 6:20PM |
0 |
user web interface |
| 5:43PM |
9 |
Druid Open Source Edition |
| 5:40PM |
2 |
DUNDi |
| 2:54PM |
0 |
snom official english forum |
| 9:51AM |
0 |
Expected iax user behaviour |
| 8:39AM |
1 |
include context on [globals] |
| 6:48AM |
3 |
DTMF problems while greeting is playing (Background()) |
| 6:22AM |
4 |
authentication number at the end of the number before calls go through. |
| 5:40AM |
1 |
S100I IAXy - Why is it discontinued? |
| 4:39AM |
0 |
chanspy doesn't work properly |
| 1:02AM |
2 |
TXFax/RXFax/AGX-Addons/SpanDSP Crashing |
| 12:18AM |
1 |
Asterisk not transcoding between installed codecs |
| |
| Tuesday March 11 2008 |
| Time | Replies | Subject |
|---|
| 11:19PM |
2 |
AGI - calling functions, CHANNEL STATUS broken? |
| 10:07PM |
1 |
Sending Incoming Calls straight to park |
| 9:58PM |
0 |
Gartner Article (was: Re: asterisk-users Digest, Vol 44, Issue 32) |
| 9:32PM |
0 |
Asterisk 1.4.19-rc2 Now Available |
| 8:48PM |
2 |
Polycom IP 330 w/VLAN? |
| 8:21PM |
7 |
Best alternative for getting prompts recorded. |
| 7:25PM |
1 |
setting callerid across servers |
| 7:16PM |
0 |
T.38 passthrough in asterisk 1.4 |
| 4:12PM |
0 |
Gartner Article |
| 1:48PM |
2 |
Unison |
| 1:41PM |
3 |
Call tracing - Asterisk 1.4 |
| 1:00PM |
1 |
Meetme application on AstriskWin32 |
| 12:14PM |
3 |
E1 Card emulator? |
| 12:11PM |
4 |
CCM 6 and Asterisk routing again |
| 11:19AM |
0 |
Central Asterisk with remote 'trunking' asterisk gateways |
| 9:25AM |
1 |
WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC |
| 9:19AM |
1 |
Sending SMS |
| 7:37AM |
1 |
Newbie Polycom: IP601 console with expansion module |
| 4:30AM |
0 |
Little help with Conference |
| 1:40AM |
3 |
need * consultant in houston area |
| |
| Monday March 10 2008 |
| Time | Replies | Subject |
|---|
| 10:36PM |
2 |
About CID with DTMF in Asterisk |
| 8:14PM |
0 |
Disable SIP notify for peers |
| 8:13PM |
1 |
Want to know Frequency and lenght of Frame |
| 6:48PM |
0 |
Audiocodes MP124-FXS replying BUSY when line is not. |
| 5:59PM |
5 |
display time on Cisco 79xx |
| 5:42PM |
1 |
Strange problem |
| 5:02PM |
1 |
Redirecting channels? |
| 4:55PM |
1 |
Asterisk and Packetcable |
| 4:27PM |
1 |
Queue Pickup |
| 3:54PM |
1 |
Intermittent DTMF Problems |
| 3:00PM |
1 |
Shared Extension |
| 2:09PM |
2 |
Global Variables on Reload |
| 1:30PM |
1 |
FaxBack Service with Asterisk |
| 11:12AM |
2 |
dialstatus and cancelled calls |
| 10:53AM |
0 |
call forward facility in INDIA |
| 10:52AM |
2 |
Call forwarding-in india |
| 9:31AM |
1 |
dialplan logic questions: macro not reaching DIALSTATUS based extension (s-ANSWER) |
| 4:25AM |
1 |
1.6.beta5 (format 0x40 (slin)) |
| 3:05AM |
11 |
Microsoft Office Communications Server |
| 2:59AM |
2 |
What replaces Macro() now? And how do you do the equivalent? |
| 1:38AM |
1 |
Local music on hold -- mohinterpret=passthrough assymetrical ? |
| |
| Sunday March 9 2008 |
| Time | Replies | Subject |
|---|
| 11:00PM |
2 |
Dead Air on PF firewall |
| 5:59PM |
0 |
phones start ringing randomly with Grandstream GXW-40XX - solution! |
| 4:23PM |
0 |
replace astdb with a cluster-capable sql database engine (was: Re: asterisk-users Digest, Vol 44, Issue 22) |
| 2:45PM |
0 |
Realtime error - PgSQL with 1.6 beta 5 |
| 6:34AM |
10 |
Read function |
| 2:00AM |
1 |
How Do I continue after Dial Command ?? |
| |
| Saturday March 8 2008 |
| Time | Replies | Subject |
|---|
| 7:53PM |
1 |
PRI suppliers in Switzerland |
| 6:36PM |
1 |
I am now on Refriendz! |
| 6:01PM |
3 |
replace astdb with a cluster-capable sql database engine |
| 2:20PM |
2 |
Experiences with grandstream GXW 4024 FXS? |
| |
| Friday March 7 2008 |
| Time | Replies | Subject |
|---|
| 10:23PM |
1 |
Sync Problem (astribank) |
| 7:46PM |
0 |
Motorola SBV5220 |
| 7:36PM |
1 |
sip show channels - gives a growing list of dead channels |
| 7:18PM |
0 |
VoiceMail dialout context |
| 7:01PM |
0 |
How to get call back during attendant transfer? |
| 4:10PM |
1 |
WirelessIP5000 SIP registration problem |
| 12:24PM |
2 |
Background: reading the digits correctly, buffering it, waiting the sound message to complete |
| 10:47AM |
0 |
chan_sip.c:2918 auto congestion |
| 10:36AM |
0 |
Call flows of sequence |
| 10:14AM |
3 |
Asterisk Realtime and SIP configuration |
| 8:34AM |
3 |
Silencing VoiceMail() app in * 1.4.10 |
| 7:31AM |
1 |
Newbie MeetMe: How to control max users in conference? |
| 5:49AM |
1 |
Call flows of conference |
| 2:43AM |
0 |
How to return the status of a call to the calling server? |
| 2:14AM |
3 |
is this possible.. |
| |
| Thursday March 6 2008 |
| Time | Replies | Subject |
|---|
| 11:02PM |
1 |
OT: Upgrade Addpac AP200C |
| 10:26PM |
0 |
Receiving double DTMF "if I pressed 1, then asterisk box recognize it 11" |
| 10:06PM |
2 |
Cool New Website |
| 9:10PM |
0 |
en25.com |
| 8:42PM |
1 |
DTMFR2- UNICALL |
| 8:39PM |
0 |
Asterisk 1.4 w/ realtime static zapata |
| 8:32PM |
0 |
Net Neutrality |
| 8:19PM |
2 |
zaptel compile question |
| 6:51PM |
0 |
Allowguest=yes & language |
| 6:25PM |
2 |
Provider recommendation in USA |
| 4:54PM |
1 |
Call Manager as trunk |
| 4:14PM |
0 |
Asterisk in the call center - how do you do it? |
| 4:07PM |
2 |
format of UNIQUEID variable |
| 4:01PM |
1 |
Best Java library to interact with Asterisk |
| 3:21PM |
14 |
FXS channel banks |
| 2:03PM |
0 |
bristuff qozap support for beronet cards |
| 1:50PM |
0 |
IAX user identification. |
| 1:18PM |
0 |
chan_iax2.c:3904 iax2_trunk_queue: Maximum trunk data space exceeded |
| 12:38PM |
0 |
Tentative: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 () |
| 12:04PM |
1 |
AEL - SQL and TIMEDIFF() |
| 10:31AM |
0 |
Asterisk authentication by SIP Proxy |
| 9:42AM |
0 |
Tentative: VoIP Users Conference |
| 9:40AM |
5 |
Declined: VoIP Users Conference |
| 8:44AM |
0 |
New Time Proposed: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 () |
| 8:42AM |
1 |
Accepted: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 () |
| 8:36AM |
0 |
Accepted: VoIP Users Conference |
| 8:35AM |
0 |
Fax in AGX Extra Addons for Asterisk causes Asterisk to die |
| 8:29AM |
2 |
VoIP Users Conference for Friday March 7th @ 12 Noon EST |
| 8:29AM |
0 |
[Invitation] VoIP Users Conference @ Fri Mar 7 12:00 - 13:00 () |
| 7:27AM |
2 |
Newbie Polycom: IP600 Headset Problem |
| 7:13AM |
0 |
RFC for conference |
| 5:58AM |
0 |
Message sequence of a conference |
| 2:37AM |
1 |
LDAP |
| 2:30AM |
1 |
Newbie Polycom: how to effect change in sip.cfg? |
| |
| Wednesday March 5 2008 |
| Time | Replies | Subject |
|---|
| 10:46PM |
6 |
Asterisk based UNIX |
| 10:44PM |
1 |
Voice quality is bad from one side and good from another side |
| 9:50PM |
1 |
Asterisk 1.6.0-beta5 Now Available |
| 9:41PM |
0 |
Asterisk 1.4.19-rc1 Now Available |
| 9:09PM |
0 |
Aastra-Asterisk: 6 beeps then voice quality degrades |
| 8:20PM |
4 |
OT How to Change Polycom Web Admin User:Pass via Web |
| 6:39PM |
1 |
Codec Preferences |
| 6:35PM |
1 |
g729 to GSM translator is needed for voicemail to work fine, how? |
| 6:19PM |
3 |
codec_g729-v34 Builds Now Available |
| 6:17PM |
4 |
Problem between Asterisk and an Aastra 57i |
| 5:50PM |
5 |
C compiler cannot create executables when building zaptel |
| 5:41PM |
2 |
Transferring Unanswered Calls |
| 5:10PM |
4 |
NIN Ghosts music (free download) safe for MOH? |
| 3:39PM |
0 |
DNS Changes never picked up with Asterisk 1.4.18 chan_sip? |
| 3:35PM |
0 |
fonality new version |
| 3:15PM |
0 |
SIP REFER Message, over NAT |
| 2:17PM |
0 |
no audio between two asterisk servers |
| 1:49PM |
1 |
How to restrict a Polycom from receiving unauthorized calls |
| 1:33PM |
2 |
Passing variables between two DUNDi/IAX2 peers |
| 12:49PM |
1 |
Linksys SPA devices and CID |
| 10:12AM |
4 |
{s} - extension |
| 4:51AM |
1 |
Newbie dialplan: dial 0 for outside line |
| |
| Tuesday March 4 2008 |
| Time | Replies | Subject |
|---|
| 9:32PM |
0 |
missing ${DIALSTATUS} in hangup extension? |
| 9:04PM |
1 |
console dsp |
| 8:34PM |
4 |
Mitel SX-200 + * |
| 8:27PM |
1 |
speaker volume on Polycom SIP phones |
| 7:10PM |
2 |
Asterisk and Avaya... |
| 7:04PM |
2 |
Problems configuring Astribank |
| 6:11PM |
1 |
Cisco 7960 SIP Upgrade |
| 5:38PM |
3 |
PPP dialout via * server |
| 4:20PM |
0 |
Page() command |
| 3:16PM |
1 |
Clustering Meetme over multiple boxes? |
| 1:53PM |
1 |
astmanproxy and core dump |
| 1:48PM |
3 |
incoming call popup |
| 9:52AM |
0 |
[Fwd: OT - CEBIT next week!] - updated list |
| 9:22AM |
1 |
Aastra Park Softkey |
| |
| Monday March 3 2008 |
| Time | Replies | Subject |
|---|
| 11:00PM |
1 |
ekiga sip registration fails; externip no help |
| 9:35PM |
2 |
Switchvox feedback |
| 9:20PM |
0 |
OT - Mime-construct, exim4 and job numbers |
| 7:45PM |
1 |
Aastra phones and park/pickup feature |
| 7:19PM |
2 |
T1, Rhino, & Nortel |
| 6:26PM |
0 |
Polycom VSX 7000e Series & Asterisk |
| 2:14AM |
5 |
Newbie on VoIP |
| |
| Sunday March 2 2008 |
| Time | Replies | Subject |
|---|
| 10:54PM |
1 |
Speex: complexity, VBR, ABR, CBR, quality |
| 2:42PM |
0 |
Cisco 79xx users/consultants, 7970G color in particular share information (was: Re: asterisk-users Digest, Vol 44, Issue 3) |
| 2:08PM |
0 |
Cisco 7970 - register with NAT phone |
| 1:40PM |
0 |
OT - CEBIT next week! |
| 11:33AM |
3 |
override/redefine asterisk DB function |
| 9:21AM |
5 |
DID number |
| 12:04AM |
0 |
Cisco 7910 and Asterisk |
| |
| Saturday March 1 2008 |
| Time | Replies | Subject |
|---|
| 10:48PM |
0 |
scripts to convert .conf files to SQL for realtime |
| 7:47PM |
2 |
I need the least expensive way to do this |
| 6:51PM |
2 |
Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW |
| 5:25PM |
0 |
Mediatrix 1124 and Audiocodes MP Series gateways |
| 4:55PM |
4 |
Cisco 79xx users/consultants, 7970G color in particular share information |
| 2:12PM |
7 |
ASTCC installation error install: invalid user `apache' |
| 9:13AM |
0 |
iax2 reload does not update channel config for removed key from config file |
| 6:17AM |
1 |
Help asterisk connectivity with MS SQL |
| 5:10AM |
1 |
"callpark" feature in ABE? |