| Thursday November 30 2006 |
| Time | Replies | Subject |
|---|
| 11:29PM |
1 |
CAPI module issue |
| 7:10PM |
3 |
1.4beta3 help |
| 6:56PM |
1 |
upgrading grandstream GXP-2000 from 1.0.2.13 to 1.1.1.14 |
| 5:30PM |
2 |
Force re-read of sip.conf |
| 2:38PM |
2 |
PAP2 and Asterisk |
| 2:30PM |
0 |
incominglimit and outgoinglimit |
| 2:25PM |
2 |
voicemailmain |
| 2:22PM |
3 |
Pickup *8 with CallerID |
| 1:56PM |
1 |
2nd attempt - Return code - How to? |
| 12:56PM |
0 |
Voicemail callback bug? |
| 11:04AM |
1 |
Asterisk 1.4 : App_Swift (Cepstral) Howto |
| 10:58AM |
1 |
Live call monitoring |
| 10:57AM |
0 |
Problem with ZapRAS and asterisk |
| 10:41AM |
1 |
T1's in St. Lucia |
| 10:21AM |
6 |
zaptel compilation problems with linux 2.6.19 |
| 9:29AM |
2 |
Billing Software |
| 9:14AM |
0 |
meetme monitoring |
| 8:52AM |
0 |
zombie SIP channels after CURL cnam lookup |
| 8:29AM |
1 |
IP call to extensions off my server |
| 6:52AM |
0 |
SIP transfer from agent fails |
| 5:33AM |
1 |
Server Compatibility questions... IBM and Dell |
| 3:24AM |
0 |
codec error message |
| 2:50AM |
1 |
AGI PHP Issues (AGI script runs but phone hangs up too quickly) |
| 2:15AM |
6 |
200+ analog phones connected to FXS modules |
| 2:13AM |
4 |
Trouble with regexten |
| 2:11AM |
0 |
Digium TE405P dtmf issue |
| 2:01AM |
0 |
Distinctive ring |
| 1:41AM |
1 |
Cut function on semicolon separator |
| |
| Wednesday November 29 2006 |
| Time | Replies | Subject |
|---|
| 10:31PM |
1 |
MeetMe announcements and SIP channels |
| 10:24PM |
0 |
Return code - How to? |
| 10:24PM |
1 |
extension launch into AGI |
| 9:09PM |
0 |
register history |
| 8:18PM |
2 |
Trouble using 2 IAX2 DiDs provided by different ITSPs |
| 8:15PM |
0 |
Re: asterisk-users Digest, Vol 28, Issue 152 |
| 8:08PM |
0 |
Conferencing Issue please help |
| 4:55PM |
2 |
Setting RTP ports for Asterisk? |
| 4:07PM |
0 |
Call dropping |
| 3:43PM |
1 |
Call recording with Asterisk BE |
| 3:23PM |
1 |
Cisco 7940 Firmware 8.2 |
| 3:19PM |
0 |
g726 voice prompts |
| 3:10PM |
0 |
beeping noise in background |
| 2:42PM |
1 |
voicemail.conf locking problem |
| 2:36PM |
3 |
Polycom 601 Second Incoming Call |
| 2:25PM |
0 |
Playing streaming MOH in Asterisk |
| 12:47PM |
1 |
Asterisk connection to a PBX |
| 12:07PM |
0 |
I am unable to find any included rpms with hudlite... |
| 11:28AM |
0 |
Call Recording and Call Transfers |
| 10:55AM |
12 |
What's up with the Manager Interface?!?! |
| 9:21AM |
2 |
Loosing IAX connection between offices |
| 9:02AM |
1 |
Getting app_cepstral to work with Asterisk 1.4.0-beta3 |
| 8:48AM |
0 |
b410p hangup detection - Portugal |
| 8:38AM |
2 |
Asterisk + Avaya S8700 |
| 8:34AM |
3 |
Blind transfer # not working for forwarded or picked calls |
| 7:38AM |
1 |
AGI PHP Issues (Not new to Asterisk but new to AGI) |
| 7:10AM |
1 |
Answer Supervision problem |
| 6:55AM |
0 |
keep line on hook |
| 6:43AM |
0 |
Desktop application for zap/agent call control |
| 5:13AM |
0 |
Re: SIP Port 5060 (Tom Lynn) |
| 4:56AM |
3 |
Siemens Gigaset C450 IP vs S450 IP |
| 4:02AM |
1 |
Monitoring an asterisk server during off hours |
| 3:45AM |
1 |
chan_misdn on a junghanns card |
| 2:48AM |
1 |
sendmail or postfix? |
| 2:36AM |
0 |
Something similar or better than HUD Pro? |
| 2:33AM |
0 |
Play an announcement while receiving DTMF? |
| 1:35AM |
1 |
Which SIP transport from France and termination services in the Nederlands |
| 12:54AM |
1 |
iptables example |
| 12:08AM |
1 |
Custom Voicemail Notification Email |
| |
| Tuesday November 28 2006 |
| Time | Replies | Subject |
|---|
| 9:16PM |
1 |
Best text to speech program |
| 8:19PM |
4 |
SIP Port 5060 |
| 7:34PM |
2 |
accountcode= placement in zapata.conf |
| 7:29PM |
0 |
Re: newbie question-asterisk username/password |
| 6:11PM |
1 |
Billing software with reseller accounts |
| 5:19PM |
2 |
No sound: X-Lite -> Asterisk -> VoIP Provider -> Cellphone |
| 4:31PM |
1 |
Bad Voice Quality - IAX2 redirect |
| 4:19PM |
0 |
channel ending with /n |
| 12:17PM |
1 |
Call recording filename |
| 12:03PM |
1 |
Why is * continually "destroying call" |
| 11:38AM |
0 |
Best email client for Asterfax |
| 11:22AM |
0 |
WANTED : Zaptel Patch - Dtmfthreshold |
| 11:13AM |
1 |
hang up detection |
| 10:54AM |
1 |
Return codes |
| 9:30AM |
1 |
SIP ATA Device Problems |
| 9:07AM |
0 |
Asterisk "Generating" SIP 486 |
| 8:23AM |
1 |
cmd Record doesn't resume Dialplan if phone Hangs-Up. |
| 7:45AM |
0 |
aastra 480i xml interface for "comedian" mail |
| 6:31AM |
4 |
Zaptel drivers for Solaris? |
| 5:39AM |
2 |
Symbian Softphone |
| 5:22AM |
1 |
Different click2call? |
| 5:18AM |
1 |
Use of VPNs |
| 4:19AM |
1 |
Attn: DISA Experts(Strange problem with DISA) |
| 1:22AM |
1 |
Modprobe zaptel reports FATAL: Module zaptel not found |
| 12:26AM |
1 |
vm_change_password shell? |
| |
| Monday November 27 2006 |
| Time | Replies | Subject |
|---|
| 8:54PM |
3 |
Do extra CPU's help? |
| 8:20PM |
0 |
SayDecimal Number |
| 8:05PM |
2 |
AsteriskNow console access |
| 2:05PM |
0 |
CTI |
| 1:44PM |
5 |
Manage Users in LDAP |
| 1:34PM |
1 |
Asterisk server reports |
| 1:30PM |
1 |
Memory leak |
| 1:18PM |
1 |
Caller ID issues |
| 12:55PM |
1 |
Sangoma & Dell 750 |
| 12:18PM |
3 |
Voicemail, SQL & ODBC |
| 11:58AM |
2 |
Busy signal from IAXy when not connecting to my Asterisk box |
| 9:53AM |
0 |
Queues and Flash/SendDTMF in hybrid PBX |
| 9:46AM |
2 |
registration ip address |
| 9:44AM |
0 |
Feature and multiple application |
| 9:23AM |
0 |
[VoIP Trunk] No such host |
| 9:15AM |
0 |
BRIcard not sending DTMF |
| 8:23AM |
0 |
Script on hold |
| 8:19AM |
5 |
Trunk Alcatel - Ring problem and call disconnection |
| 8:10AM |
1 |
Incoming calls don't arrive for correct number |
| 7:44AM |
0 |
Announce Queue Position variable |
| 7:43AM |
1 |
Asterisk Feature Codes won't work |
| 7:42AM |
1 |
wip5000 crash AP |
| 7:22AM |
2 |
SIP group management |
| 7:08AM |
0 |
calls hang up even after Background() messageeventhough response timeout is set to 10 sec |
| 6:33AM |
0 |
flash transfer problem in asterisk with old PBX |
| 6:19AM |
1 |
AgentCallbackLogin deprecated? |
| 5:05AM |
0 |
Asterisk is taking the first digit of my entered number twice. Why? |
| 2:22AM |
3 |
bristuff error: "received SETUP message for call that is not a new call" |
| 1:42AM |
1 |
Click to dial apps always show from "asterisk" |
| 12:49AM |
1 |
Junghanns Bristuff PRI indication |
| 12:11AM |
1 |
calls hang up even after Background() message eventhough response timeout is set to 10 sec |
| |
| Sunday November 26 2006 |
| Time | Replies | Subject |
|---|
| 11:58PM |
2 |
3xx redirect from asterisk? |
| 11:40PM |
0 |
Dialout to Meetme Fails? |
| 6:37PM |
1 |
Odd blip when playinv IVR over IAX |
| 6:16PM |
0 |
Asterisk 1.4 : AstDB Callerid Rolodex Tool |
| 6:15PM |
0 |
Asterisk 1.4 : Gentoo, MYSQL CDR, and GUI howto |
| 6:15PM |
0 |
MWI - Message Waiting Indicator free software routines |
| 3:13PM |
0 |
spandsp |
| 11:39AM |
0 |
port rtp problem |
| 11:35AM |
3 |
Looking for toll-free US did |
| 9:11AM |
0 |
way to get extension's hint? |
| 8:57AM |
1 |
Setting up FastAGI in Asterisk? |
| 2:30AM |
1 |
Streaming MoH working example required |
| |
| Saturday November 25 2006 |
| Time | Replies | Subject |
|---|
| 9:11PM |
0 |
SOLVED - 1.4 svn voicemail bug / crash |
| 6:46PM |
2 |
Passing PRI traffic to remote * over IAX |
| 6:31PM |
0 |
Digium Iaxy S100 Factory Default? |
| 6:23PM |
1 |
Asterisknow |
| 4:53PM |
0 |
Re:VOIP Consultants wanted to build a Scalable ITSP Architecture |
| 2:30PM |
0 |
Sip reinvite |
| 2:03PM |
0 |
MeetMe, background agi and playing sounds |
| 1:56PM |
0 |
Linking Asterisk Servers using SIP instead of IAX |
| 12:00PM |
0 |
Re: asterisk-users Digest, Vol 28, Issue 132 |
| 8:57AM |
2 |
1.4 svn voicemail bug / crash |
| 8:38AM |
1 |
dialing with different speed |
| 7:35AM |
5 |
DID Provider |
| 7:06AM |
0 |
Problems with sound quality |
| 5:51AM |
0 |
Re: asterisk-users Digest, Vol 28, Issue 131 |
| 5:49AM |
0 |
How to do Call barging with SIP channel |
| 5:47AM |
0 |
VOIP Consultants wanted to build a Scalable ITSP Architecture Using OpenSource Softwares |
| 5:09AM |
0 |
Modem and TDM400P |
| |
| Friday November 24 2006 |
| Time | Replies | Subject |
|---|
| 10:36PM |
0 |
Help needed - Can anyone please explain to me what is causing this - TDM2400P |
| 7:59PM |
2 |
Correct syntax to access a shell variable? |
| 6:00PM |
1 |
Re:Call Transfers in SER + Asterisk |
| 5:38PM |
2 |
Card don't hangup but Asterisk hangup |
| 11:45AM |
1 |
FS: Sangoma 10 port FXO card |
| 11:29AM |
2 |
cisco 7961 , asterisk and busy lamp |
| 11:22AM |
1 |
mfcr/R2 |
| 9:52AM |
1 |
Monitoring awareness |
| 8:41AM |
0 |
Doubling up; redunancy with DUNDi |
| 8:29AM |
3 |
Junk faxes |
| 7:21AM |
0 |
Snom 360 / firmware 6.5.1 / registration problems with Asterisk |
| 7:14AM |
1 |
Encrypted password for voicemail |
| 5:31AM |
0 |
Caller Id not propagated to the analog line |
| 4:39AM |
2 |
DB9 e1 to RJ45 pinout |
| 3:34AM |
1 |
upgraded polycom to 2.0.1.0291 and... |
| 2:00AM |
1 |
Server Configuration for E1's |
| 1:31AM |
1 |
Installing the b410p card, unable to install mISDN |
| 12:18AM |
0 |
Dial() cmd seams unable to detect caller hangup |
| |
| Thursday November 23 2006 |
| Time | Replies | Subject |
|---|
| 10:54PM |
1 |
(no subject) |
| 10:44PM |
0 |
asterisk and MISDN on a core2 Duo x64 system |
| 10:32PM |
2 |
Asterisk and TDM400P ? |
| 9:47PM |
0 |
Direct UA to UA RTP connection |
| 8:12PM |
0 |
MWI from ITSP |
| 8:03PM |
0 |
asterisk 1.4 variable list |
| 7:12PM |
0 |
asterisk-users, Matt has invited you to open a Google mail account |
| 6:35PM |
0 |
Asterisk voicemail and hotel software integration |
| 5:54PM |
0 |
Passing arguments to AGI script |
| 4:27PM |
0 |
Re: asterisk-users Digest, Vol 28, Issue 122 |
| 3:30PM |
0 |
Store voicemal data in mysql DB |
| 3:20PM |
1 |
Call Transfers in SER + Asterisk architecture |
| 2:02PM |
0 |
Asterisk 1.4 Error |
| 11:58AM |
2 |
FREE DOWNLOAD - PRI / T1 Circuit monitoring |
| 11:55AM |
1 |
When does voicemail authentication take place? |
| 11:54AM |
2 |
Cisco 7970 SIP upgrade issues |
| 10:02AM |
0 |
festival problem using IAX (chan_iax2.c:2995 iax2_read) |
| 9:44AM |
2 |
Digium through Octasic |
| 8:50AM |
1 |
FOP is not displaying all my SIP extensions neither all E1 channels , why? |
| 8:38AM |
1 |
Error uninstalling freepbx-panel |
| 8:18AM |
3 |
Cisco 7970 |
| 7:47AM |
1 |
Asterisk with SER |
| 6:03AM |
1 |
(OT) HylaFAX, IAXModem, Asterisk |
| 4:53AM |
0 |
AGI info |
| 4:36AM |
1 |
Re: How to change IAX default port 4569 to some other port :Debug Message Attached |
| 4:18AM |
2 |
How to change IAX default port 4569 to some other port |
| 1:27AM |
0 |
snom subscriptions issue on WRT (2) |
| 1:18AM |
1 |
How to kill a meet me room at midnight |
| 1:04AM |
1 |
asterisk 1.4 chan_h323, help please... |
| 1:00AM |
1 |
Calls "from asterisk" |
| 12:18AM |
0 |
OriginateEvent reason codes. |
| 12:14AM |
0 |
Exact definition of ASR |
| |
| Wednesday November 22 2006 |
| Time | Replies | Subject |
|---|
| 10:07PM |
0 |
in Asterisk Manger its Unauthentication User and Host .......... |
| 9:17PM |
1 |
gotoiftime and blocking calls |
| 8:26PM |
1 |
Hold calling channel and ask called channel before connect??? |
| 8:19PM |
2 |
G722? |
| 7:09PM |
1 |
Sipura phone does not ring |
| 6:54PM |
1 |
queuemetrics |
| 4:14PM |
2 |
How to park calls on a specific extension |
| 4:03PM |
1 |
Zaptel - make b410p fails on Ubuntu 6.10 |
| 3:21PM |
2 |
Terrible, horrible firewall issues in * to * setup |
| 2:04PM |
1 |
aastra 480i configuration help |
| 1:18PM |
0 |
Call park on Linksys 922 and similar phones? |
| 1:02PM |
4 |
More than one asterisk process |
| 11:51AM |
5 |
TE110P and TDM400P |
| 11:45AM |
4 |
Asterisk On FreeBSD |
| 10:36AM |
0 |
channel_find_locked: Avoided deadlock ... messages - What to do? |
| 9:57AM |
1 |
G729 issues on 1.4 beta 3 |
| 8:48AM |
2 |
How ecord all calls? |
| 7:43AM |
1 |
Recordings for VR analysis |
| 7:33AM |
2 |
Send event from dialplan |
| 7:23AM |
0 |
iax2 - wildiax phone & myself puzzled |
| 7:15AM |
1 |
Asterisk incoming call behaviour |
| 6:54AM |
1 |
DTMF detection during Call |
| 6:08AM |
8 |
Recordings. |
| 6:02AM |
11 |
Rewriting caller ID from database? |
| 5:57AM |
0 |
help in Call parking...... |
| 3:32AM |
0 |
Ast 1.4 and B410p |
| 3:21AM |
1 |
Request for working config for DISA |
| 3:18AM |
1 |
about voicemail setting |
| 3:15AM |
1 |
Zaptel error |
| 3:15AM |
0 |
SOLVED: Digium TE405 card and Matra PBX |
| 2:45AM |
1 |
qualify=yes |
| 2:29AM |
0 |
Is it easy to route SIP/SDP and SIP/RTP through different routes ? |
| 2:04AM |
0 |
asterisk-cluster with one database |
| 2:00AM |
2 |
snom subscriptions issue on WRT |
| 1:56AM |
1 |
Agent Channel SIP transfer |
| 1:26AM |
1 |
Welcome to Join Asterisk MSN Groups! |
| |
| Tuesday November 21 2006 |
| Time | Replies | Subject |
|---|
| 9:59PM |
1 |
Attn:Peter, Gsalas, Tim-Help me to configure my NOKIA E70 Mobile with my Asterisk server |
| 8:51PM |
0 |
codec information |
| 8:47PM |
0 |
Is this a PRI problem, *, or the phone??? |
| 8:27PM |
2 |
Can anyone enlighten me as to what this means? |
| 6:57PM |
5 |
Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V |
| 5:54PM |
1 |
Hints no longer working in 1.4beta3 with Polycoms |
| 5:20PM |
4 |
IP601 Expansion Module HELP!!! |
| 4:49PM |
1 |
Is this possible? |
| 4:28PM |
0 |
OT: Reflash Mitel 5220 from MiNet to SIP |
| 3:04PM |
2 |
cmd Record |
| 2:54PM |
1 |
VM mail notification and locale |
| 2:09PM |
0 |
QoS on Linksys SWR208P? |
| 1:40PM |
0 |
voice breaking problems |
| 1:40PM |
2 |
Answer Machine Detection |
| 1:39PM |
1 |
Call to disconnected number on PRI just rings |
| 11:51AM |
3 |
IAX access to FWD broken? |
| 11:30AM |
1 |
Prefix/Suffix/StripLSD/StripMSD gone? |
| 11:18AM |
0 |
Re: Choppy sound in voicemail usingAsterisk1.2.11 on CENTOS4 guest on vmware server |
| 10:50AM |
3 |
Parking on an extension |
| 10:45AM |
0 |
RESOLVED - Snom 360 Multiple calls on hold help |
| 10:23AM |
0 |
Setting the default AMAflag |
| 10:17AM |
0 |
QueueMetrics 1.3.1 released today |
| 8:14AM |
3 |
Diva Server, chan_capi and tone detection |
| 8:02AM |
3 |
Cisco media gateways in general |
| 7:50AM |
2 |
FW: CISCO 7960G & Asterisk |
| 7:12AM |
0 |
Nortel CS1000 Asterisk with SIP |
| 7:04AM |
0 |
Which reliable source for ToIP security alerts ? |
| 5:36AM |
0 |
Signalogic SigC5561 PCI card |
| 5:21AM |
2 |
Handle Options Method |
| 5:14AM |
0 |
Callback agents without chan_agent issues (queue recording) |
| 4:49AM |
0 |
Forums |
| 3:08AM |
1 |
Hairping calls and Originating CLI |
| 2:07AM |
0 |
Can i have two asterisk versions running on samePC?? |
| |
| Monday November 20 2006 |
| Time | Replies | Subject |
|---|
| 11:28PM |
1 |
Reliable European SIP/IAX Providers? |
| 7:27PM |
2 |
QB intergreation |
| 6:09PM |
2 |
TDM400 native bridge echo |
| 6:05PM |
1 |
CAPI (Eicon Diva V-4BRI), Hylafax & IAXModem |
| 5:10PM |
0 |
is there a queue size call limit on the ACD? |
| 4:47PM |
2 |
How to secure access to PSTN line through Linksys gateway? |
| 3:25PM |
3 |
Spandsp rxfax txtax fails no errors |
| 3:20PM |
1 |
alert_info + Linksys 9xx + custom ringtone |
| 2:54PM |
0 |
install asterisk ad zaptel from source on red hat enterprise 3 |
| 2:32PM |
1 |
Reset Extension Preferens |
| 2:18PM |
1 |
AW: Snom 360 Multiple calls on hold help |
| 1:41PM |
7 |
Snom 360 Multiple calls on hold help |
| 12:52PM |
4 |
Auto recording calls? |
| 12:40PM |
0 |
iSoftSwitch:: Asterisk & GnuGK Integretion |
| 10:57AM |
2 |
email etiquette (was: Re: Unicall MFC problems in 0.0.3+asterisk 1.2) |
| 10:50AM |
0 |
Set a feature within AgentLogin |
| 10:30AM |
1 |
How to accept All incomings calls from One Special Host (like a proxy) |
| 10:09AM |
2 |
Call limits and VoIP providers |
| 9:45AM |
1 |
SIP Multi-Domain |
| 9:20AM |
1 |
T.38 - By reinvitation only? |
| 7:39AM |
2 |
Help me to configure my NOKIA E70 Mobile with my Asterisk server |
| 7:39AM |
0 |
Compilation problem |
| 7:16AM |
0 |
.call file unable to hear or speak |
| 6:27AM |
1 |
Call-limit |
| 6:04AM |
1 |
g729 registered |
| 5:22AM |
2 |
Recording g729 |
| |
| Sunday November 19 2006 |
| Time | Replies | Subject |
|---|
| 10:50PM |
4 |
reduce dialtone volume on zap channel. |
| 9:24PM |
1 |
Vonage uses Cisco |
| 8:23PM |
2 |
switching trunks based on quality |
| 10:20AM |
1 |
PHP to .call file |
| 10:05AM |
2 |
WaitExten only reading 1 digit. |
| 6:53AM |
4 |
What card for E1R2? |
| 4:28AM |
0 |
MS-GSM codec issues - Anybody seen anything similar? |
| 3:59AM |
1 |
G723 pass-through and codec negotiation |
| 3:27AM |
2 |
Question on CDR Database |
| |
| Saturday November 18 2006 |
| Time | Replies | Subject |
|---|
| 7:37PM |
0 |
Cant record phone calls |
| 6:05PM |
3 |
odd issue with IP tables |
| 4:56PM |
1 |
Re: Asterisk to listen for sip traffic on 80 and 5060 |
| 1:24PM |
0 |
Statistics on Number of Minutes |
| 12:05PM |
0 |
Need help with a function |
| 11:45AM |
2 |
Dialout Conferences? |
| 11:00AM |
0 |
Cisco 2801 and asterisk |
| 10:35AM |
0 |
TDD/TTY device for the deaf |
| 10:21AM |
5 |
Asterisk Manager: equivalent of 'show channels'? |
| 9:07AM |
1 |
Hardware Echo cancelation |
| 8:34AM |
0 |
Using ChanSpy for spying voicemail |
| 3:52AM |
2 |
AdvancedVoIP Billing ? |
| 3:50AM |
0 |
If of external small box supply fxs Isdn and E1 ? |
| 1:15AM |
0 |
H323 no audio |
| |
| Friday November 17 2006 |
| Time | Replies | Subject |
|---|
| 11:26PM |
5 |
spc.exe |
| 10:56PM |
2 |
strip + sign from incoming ${EXTEN} var? |
| 5:43PM |
0 |
Jitter Buffers in Zapata |
| 4:52PM |
3 |
Ringing a group of phones but not if they are busy |
| 4:28PM |
0 |
Destar release! |
| 3:30PM |
1 |
Asterisk - Do Not Call List |
| 3:06PM |
1 |
Extension Response Slow |
| 2:07PM |
0 |
automated response |
| 2:06PM |
1 |
TDM2400p and HW echo canceller |
| 1:32PM |
11 |
wget from within asterisk? |
| 12:33PM |
0 |
metermaid and 1.2.13? |
| 12:12PM |
5 |
Freepbx changes dont reflect in asterisk |
| 12:08PM |
3 |
voice quality of Aastra 480i CT and cordless |
| 11:09AM |
1 |
specify codec by domain? |
| 10:50AM |
0 |
Understanding the CDR with forwards... |
| 9:51AM |
2 |
1 FXO termination device |
| 7:27AM |
0 |
redhat enterprise 3 |
| 6:41AM |
2 |
Need help on Music on Hold |
| 5:35AM |
0 |
Problem with Asterisk 1.4.0-beta3 and Digium TE405P |
| 2:19AM |
15 |
Siemens Gigaset SL75 |
| |
| Thursday November 16 2006 |
| Time | Replies | Subject |
|---|
| 10:37PM |
0 |
Call forwarding.... |
| 10:26PM |
1 |
Asterisk on Solars? |
| 6:26PM |
0 |
asterisk OSX -astmasters site is gone |
| 4:39PM |
1 |
Multi-site Redundancy. Possible? |
| 3:36PM |
2 |
dialplan "*" and "0" key detection, not working |
| 3:03PM |
3 |
Nokia E70 |
| 1:48PM |
1 |
AEL2 Confusion |
| 1:41PM |
1 |
asterisk billing software |
| 1:28PM |
1 |
Asterisk 1.2.13 can't load module app_curl.so |
| 12:27PM |
1 |
FXO PCI Master abort |
| 11:43AM |
0 |
Celliax LiveCD 0.0.32 released (Asterisk managing cellular phones, and Skype calls to/from cellphones, via chan_celliax) |
| 11:31AM |
1 |
hosted asterisk |
| 10:20AM |
0 |
jitterbuffer in pure voip (sip/iax) - what is best practice |
| 9:34AM |
1 |
zaptel, bristuff zaphfc, and florz question |
| 9:14AM |
0 |
call from cisco router to asterisk gets auto attendant |
| 9:02AM |
1 |
make: execvp: build_tools/make_svn_branch_name: Permission denied |
| 8:52AM |
2 |
POS Terminals |
| 7:37AM |
0 |
Backup and mail on trixox |
| 7:16AM |
0 |
Asterisk call recording |
| 7:14AM |
2 |
installing asterisk for Ubuntu Synaptic |
| 7:12AM |
0 |
upgading to install-misdn 0.3.1-rc23 broke dtmf detection on some calls |
| 6:14AM |
1 |
Sangoma A101 gives 'no PRI configured on span 1' error |
| 5:46AM |
0 |
dialing channel late |
| 4:19AM |
0 |
turning off DTMF detection on Zap channels |
| 4:16AM |
2 |
T.38 - make conclusion |
| 3:27AM |
5 |
spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13 |
| 3:23AM |
1 |
chanspy crash the asterisk 1.4 |
| 1:46AM |
1 |
Trunk outcall line ? |
| 1:10AM |
1 |
queue management |
| |
| Wednesday November 15 2006 |
| Time | Replies | Subject |
|---|
| 10:40PM |
0 |
Zaptel 1.4.0-beta2 compile error |
| 10:24PM |
2 |
Found GSM version, but any better WAV or ULAW recordings of "Steve" or "Ian" out there? |
| 10:18PM |
7 |
Do Not Call List |
| 9:56PM |
2 |
Installing Ztdummy on Fedora Core 5 |
| 9:12PM |
0 |
Anyone using the directory.agi app in AGI perl |
| 7:30PM |
3 |
Regular audio fade-out fade-in on IAX2 calls Asterisk 1.2.4 Hi all, One of my users has a problem with many of his calls via my Asterisk™ server. He describes the problem as having the sound slowly fade out and then fade back at a regular frequency. Has |
| 6:42PM |
0 |
Intercom function on eyebeam xten softphones. |
| 6:23PM |
2 |
Grandstream GXP2000 -- What's the Catch? |
| 5:49PM |
0 |
chan_unicall.c isntallation problem |
| 5:30PM |
1 |
simple mainmenu ivr tones not recognized |
| 4:12PM |
1 |
Attempting native bridge of |
| 3:35PM |
1 |
Queue - how to provide a caller ringing tone when some agent become available |
| 3:34PM |
0 |
Grandstream Programmable Buttons & Retrieving On Hold Lines |
| 3:31PM |
0 |
Auth Issue using Asterisk as Voicemail AND as Normal SIP Extension. |
| 2:28PM |
0 |
web interface to control zap interface |
| 2:15PM |
3 |
Set port to which Asterisk should send its answer |
| 1:56PM |
1 |
Monitor Zap Status - Full E-mail... |
| 1:52PM |
0 |
Monitor Zap Status |
| 1:40PM |
3 |
PortSip and Astericks new install |
| 1:31PM |
0 |
Huawei Videophone |
| 1:11PM |
2 |
Question about TFTPD server |
| 12:28PM |
2 |
PHPAGI example usage of input.php |
| 12:18PM |
2 |
safe_asterisks pawning multiple asterisk process??? |
| 11:48AM |
0 |
Handy tip for intercom with FreePBX & Grandstream phones |
| 11:40AM |
1 |
quadbri + kernel 2.6.18.1 |
| 11:38AM |
2 |
Got 200 OK on REGISTER that isn't a register |
| 11:00AM |
0 |
dtmf tones not always recognized |
| 10:34AM |
2 |
Page() Function Timeout |
| 10:01AM |
2 |
Problems with language support |
| 10:00AM |
0 |
Disabling Features Temporarily |
| 9:51AM |
1 |
Setting the CallerID |
| 8:42AM |
2 |
ODBC Voicemail Storage |
| 7:54AM |
1 |
State of a public number |
| 7:28AM |
0 |
SIP NOTIFY routing problem |
| 6:56AM |
1 |
How to disable the 482 Loop Detected messages sent by Asterisk |
| 6:41AM |
1 |
Asterisk - big installation |
| 6:13AM |
0 |
The best available CAPI BRI card for Asterisk ? |
| 5:54AM |
2 |
some questions about atxfer usage |
| 5:40AM |
2 |
T38 problem |
| 4:58AM |
0 |
Asterisk as a SIP client, Need to auto-answer |
| 4:42AM |
5 |
Time Based Voicemail Messages |
| 3:14AM |
1 |
How to do the Call Snooping |
| 3:01AM |
0 |
How to use Voipjet or any Voip provider Trunk from my mobile through fxo and fxs ports? |
| 1:19AM |
0 |
Condensing queue CDRs into single entry |
| |
| Tuesday November 14 2006 |
| Time | Replies | Subject |
|---|
| 11:13PM |
0 |
Retain call control: Avoid letting call get |
| 10:31PM |
0 |
TDD - stops receiving characters |
| 10:16PM |
0 |
Caller Initiated Conference |
| 10:08PM |
2 |
Add Apps to Asterisk? |
| 8:51PM |
0 |
[SPAM HEADER] - trixbox + agi - Email found in subject |
| 8:15PM |
1 |
trixbox + agi |
| 6:47PM |
1 |
Call log reveals redundant calls! |
| 6:04PM |
2 |
ATA with reliable FAX? |
| 5:58PM |
1 |
How to use Sipura SPA3k POTS line to dial Asterisk SIP phones? |
| 5:06PM |
3 |
Caller ID in Sweden not working and looking for and voices |
| 3:42PM |
0 |
Voice mail transfer between 2 asterisk servers |
| 3:34PM |
0 |
How do I change the rtp packet size in a Cisco 7490 from 10ms to 20ms |
| 2:50PM |
6 |
unable to get channel lock BAD BAD BAD |
| 1:36PM |
4 |
In the beginning-The first question. |
| 1:09PM |
2 |
DUNDi Asterisk Cluster |
| 10:41AM |
0 |
asterisk sip doesn't see other asterisk-sip |
| 10:34AM |
1 |
Retain call control: Avoid letting call get into cellular voicemail |
| 10:28AM |
1 |
Dialplan options |
| 9:15AM |
0 |
Problems with voicemail |
| 8:55AM |
2 |
Problem with FXS ports of TDM400P |
| 8:21AM |
0 |
Fax killed on all zaptel devices |
| 8:14AM |
1 |
Broken Call Screening |
| 7:31AM |
7 |
900 rules |
| 6:56AM |
0 |
(no subject) |
| 5:53AM |
0 |
Zaptel and limiting number off channels channels |
| 4:33AM |
0 |
[Voicemail] Change the format of the VM_DATE |
| 2:46AM |
3 |
Is asterisk able to integrate with MS SQL |
| 1:17AM |
0 |
Redirecting Calls |
| 12:29AM |
2 |
Installation of Unicall for MFC/R2 |
| |
| Monday November 13 2006 |
| Time | Replies | Subject |
|---|
| 11:35PM |
0 |
chanspy -coredump( asterisk 1.4) |
| 11:03PM |
1 |
Newbie Questions . . . |
| 7:45PM |
3 |
Load balance Asterisk servers? |
| 7:41PM |
0 |
Application Directory question |
| 7:33PM |
2 |
Linksys doesn't resync properly and doesn't get provisioning from TFTP and HTTP |
| 7:30PM |
0 |
CDR shows NO ANSWER when call is really ANSWERED |
| 6:50PM |
0 |
Data over zaptel |
| 6:50PM |
6 |
Dual Wan Router with Failover |
| 6:04PM |
0 |
establish meetme limit for a single room |
| 5:31PM |
1 |
Bellsouth issue ? |
| 4:51PM |
2 |
STUN with one public and one private IP? |
| 4:03PM |
0 |
Recommend software version for Cisco IOS Gateways and SIP Phones? |
| 3:42PM |
3 |
"Username/auth name mismatch" + SIP phone can't connect? |
| 3:41PM |
0 |
Zaptel |
| 3:16PM |
0 |
Survey: In what ways do you use Asterisk at your house? |
| 2:49PM |
0 |
Question about MySQL Fetch foundRow from the dial plan |
| 1:47PM |
1 |
SIP Ports (1000 to 2000 works) |
| 1:33PM |
1 |
Dial/Continue/Announce |
| 1:29PM |
0 |
MWI not working in 1.4 |
| 1:14PM |
0 |
Native TDM Bridge |
| 12:27PM |
1 |
Voicemail argument size limit |
| 12:24PM |
2 |
FAX using T38 |
| 12:18PM |
0 |
Asterisk with ss7 and sip-t |
| 11:48AM |
1 |
Can AGI do this? |
| 11:41AM |
1 |
asterisk as a Media Gateway |
| 11:37AM |
3 |
Mysql 6 second rounding |
| 11:29AM |
1 |
DSl and more then 1 call |
| 11:18AM |
1 |
Music on hold question |
| 11:11AM |
0 |
Fast Busy with autodial using a call file |
| 10:53AM |
0 |
newbie question |
| 10:02AM |
0 |
2 * servers Host=ip - doesn't work Host=dynamic with register is OK, why? |
| 9:52AM |
0 |
Slow playback of sound prompts |
| 9:49AM |
2 |
Recording outbound analog calls with X100P |
| 9:14AM |
1 |
problem with redirects |
| 8:37AM |
2 |
Custom voicemail extension greeting |
| 8:31AM |
1 |
Defunct / zombie AGI after some execution time |
| 6:44AM |
2 |
Problem with internet down |
| 6:15AM |
3 |
FW: Desktop integration |
| 5:27AM |
0 |
Question about the GUI for 1.4 |
| 4:46AM |
4 |
Asterisk IVR functionality |
| 4:28AM |
8 |
Desktop integration |
| 3:48AM |
1 |
Dial : Executing context/priority after bridge? |
| 3:07AM |
0 |
Voicemail and realtime : the emailbody option ... |
| 2:32AM |
0 |
bindport |
| 1:43AM |
1 |
Sending '#' with Dial |
| 12:23AM |
0 |
Can i have two asterisk versions running on same PC?? |
| 12:15AM |
2 |
Can i have two asterisk vcersions running on same PC?? |
| 12:08AM |
1 |
Moh stops immediately |
| |
| Sunday November 12 2006 |
| Time | Replies | Subject |
|---|
| 9:58PM |
0 |
Asterisk VM with Cisco routing |
| 8:58PM |
0 |
Trixbox dialout problems |
| 7:40PM |
3 |
Slow to get dialtone when going off hook - big problem for me :( |
| 6:26PM |
1 |
Zaptel compile problems |
| 4:45PM |
2 |
Headaches with Video over SIP |
| 3:48PM |
2 |
IAX2 one way audio |
| 3:44PM |
0 |
cadences zapata.conf |
| 2:29PM |
2 |
same extension on softphones and hardphones |
| 1:59PM |
0 |
Asterisk billing |
| 1:43PM |
3 |
Determine if Call is from a cellular phone |
| 1:08PM |
1 |
outgoing works, incoming fails on asterisk passthrough to inter-tel |
| 10:42AM |
0 |
VM problems... |
| 9:27AM |
1 |
Some pictures from Astricon 2006 in Dallas |
| 8:48AM |
0 |
Asterisk Media Gateway |
| 8:27AM |
3 |
Looking for a simple TFTP server for Linux |
| 7:39AM |
0 |
Speeding up SayDigits? |
| 7:33AM |
0 |
asterisk-addons 1.4 SVN fails to compile |
| 6:47AM |
2 |
dynamically modifying the dialplan? |
| 1:50AM |
1 |
Knowing when an answerphone answers |
| |
| Saturday November 11 2006 |
| Time | Replies | Subject |
|---|
| 10:10PM |
2 |
CLI message: remote unix connection disconnected |
| 7:31PM |
1 |
No sounds in svn version? |
| 12:46PM |
1 |
sip forward behind a nat |
| 10:11AM |
0 |
can't hear MusicOnHold when zap answers |
| 8:35AM |
1 |
Call file: CallerID problem |
| 3:43AM |
0 |
chansp core dump |
| 12:49AM |
1 |
Soundfiles adding during phone calls |
| |
| Friday November 10 2006 |
| Time | Replies | Subject |
|---|
| 10:05PM |
0 |
app_swift: Failed to set voice |
| 6:20PM |
2 |
Dialing from "Placed Calls" on Polycom IP501 doesn't always work |
| 5:37PM |
0 |
Push to Talk settings. |
| 5:25PM |
3 |
SPA-941 (and others ) Transmit Sound Quality |
| 3:20PM |
1 |
(no subject) |
| 2:55PM |
0 |
monitor-join does not seem to work. |
| 1:53PM |
2 |
config template for Grandstreams |
| 1:45PM |
2 |
WIFI phones on asterisk |
| 12:43PM |
0 |
app_pppd - Could not read send data |
| 12:30PM |
1 |
Harris picking up before extension |
| 12:22PM |
0 |
Returncode from command |
| 11:58AM |
0 |
Realtime & sippeers using NAT |
| 11:11AM |
1 |
Choppy sound in voicemail using Asterisk 1.2.11 on CENTOS4 guest on vmware server |
| 10:56AM |
1 |
Need to automatically park an incoming call and then connect to an extension. |
| 10:52AM |
3 |
How to get CDR to show answered calls only |
| 10:01AM |
0 |
Re: Asterisk and Max TNT SIP Authentication Issue, WORKING |
| 9:03AM |
1 |
Question about Mitel phones |
| 7:36AM |
2 |
Outgoing problem on PRI |
| 7:22AM |
0 |
Pointers/suggestions? |
| 6:35AM |
2 |
Presence-awareness in Asterisk |
| 6:33AM |
0 |
VM notification to pager and phone |
| 6:01AM |
1 |
Queues and Timeouts. |
| 5:59AM |
9 |
Stable clock with 2.6 and without Digium hardware. |
| 5:14AM |
1 |
Looking for IP phone / ATA that has builtin VPN support |
| 5:07AM |
0 |
SV: Dropping Connections |
| 4:56AM |
1 |
Dropping Connections |
| 3:21AM |
1 |
EuroISDN+ and Callers name |
| 12:52AM |
0 |
Asterisk BlindTransfer behaves differently in version 1.0 and 1.2 |
| |
| Thursday November 9 2006 |
| Time | Replies | Subject |
|---|
| 10:07PM |
3 |
announcing inbound PSTN calls |
| 7:12PM |
1 |
DTMF problems with IVR - What DMTF Tx method |
| 7:11PM |
2 |
Powering SNOM 200 phones? |
| 4:38PM |
2 |
Latest Debian and latest zaptel |
| 4:24PM |
0 |
Mitel 5224 & Asterisk Distinctive Ring -- Anyone have it working? |
| 4:20PM |
7 |
Modprobe Zaptel |
| 3:35PM |
1 |
Zaptel 1.2.11 released |
| 1:49PM |
1 |
porting numbers away from packet 8? |
| 1:07PM |
0 |
Harris 20-20 |
| 12:28PM |
2 |
register suddenly fails |
| 11:36AM |
1 |
wip5000 roaming |
| 11:28AM |
1 |
New Asterisk 1.4 GUI |
| 10:22AM |
1 |
Station Voip Brazil |
| 10:15AM |
0 |
special characters in alphanumeric extension s |
| 10:07AM |
2 |
A couple of new tutorials: installing * 1.4 and the Asterisk GUI |
| 9:58AM |
1 |
Quick Q... |
| 9:50AM |
0 |
Bug ??? |
| 9:21AM |
1 |
special characters in alphanumeric extensions |
| 9:14AM |
2 |
asterisk and norstar |
| 9:00AM |
1 |
unsubscribe |
| 8:36AM |
5 |
DUNDi precache |
| 8:32AM |
5 |
Voxee lag problems ? |
| 8:19AM |
1 |
Problem with register command in SIP.conf |
| 5:23AM |
2 |
Alcatel trunk with asterisk problem on dialing digit-by-digit |
| 4:53AM |
3 |
SRTP |
| 3:22AM |
0 |
TDM, loopstart and modules GSM Nokia32 |
| 1:56AM |
1 |
Problem with CDR interpretation |
| |
| Wednesday November 8 2006 |
| Time | Replies | Subject |
|---|
| 11:22PM |
0 |
OT - Polycom https provisioning |
| 9:54PM |
1 |
DID billing with a2billing |
| 9:51PM |
0 |
Unknown caller id problem |
| 7:42PM |
1 |
Auto record a call? |
| 7:15PM |
1 |
Ask users.conf |
| 6:36PM |
0 |
Queues: member order vs. defines in queues.conf |
| 4:45PM |
1 |
Reg errors? Other anomalies? Check those capacitors! |
| 4:25PM |
1 |
Still problems with Asterisk on latest Debian |
| 3:42PM |
1 |
I LOVE IT |
| 2:42PM |
0 |
[FC5] How to update kernel/kernel-develop for Athlon? |
| 2:32PM |
0 |
sms script on receive |
| 2:07PM |
2 |
Off-Site Extensions That Would Show As In-Use? |
| 1:35PM |
0 |
Warning: "Channel does not have a CDR" when doing ForkCDR |
| 1:29PM |
1 |
Microsoft will enter VoIP market in earnest |
| 1:18PM |
1 |
Re: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer |
| 1:15PM |
5 |
DTMF Corruption Problem |
| 12:09PM |
1 |
talking caller ID |
| 11:38AM |
1 |
Re: asterisk iax2 monitoring |
| 11:35AM |
2 |
One-Way-Audio After placing call on hold |
| 10:06AM |
1 |
Re: Asterisk and Max TNT PRI to SIP Authentication Issue |
| 9:52AM |
1 |
Delay between DTMF Down & Detected Digit |
| 9:21AM |
0 |
SIP CANCEL NOT WORKING |
| 9:15AM |
1 |
FIC-GTA001 |
| 8:35AM |
0 |
jpeglib |
| 8:14AM |
1 |
VLANs and Quality |
| 7:58AM |
1 |
Performance issues in Realtime |
| 7:56AM |
1 |
HANGUPCAUSE for unalocated number? |
| 7:51AM |
0 |
Asterisk 1.2.x and video |
| 7:30AM |
1 |
Ringing phones |
| 7:27AM |
1 |
I need (some) help in configuring PAP2. |
| 6:22AM |
0 |
Asterisk 1.4 and Queues RealTime |
| 5:55AM |
0 |
Odd results from fxotune? |
| 4:30AM |
0 |
asterisk and peep tone (network tone) |
| 4:17AM |
1 |
faxing times! |
| 3:55AM |
0 |
no sound when bridging 2 asterisk SIP connections |
| 2:49AM |
0 |
Asterisk CTI - SAP R/3 Intergration Certification |
| 2:19AM |
1 |
Agents that handle calls from multiple queues |
| 2:19AM |
0 |
Queue forks asterisk and then leaves theextraprocesses lying around |
| 2:00AM |
1 |
Queue forks asterisk and then leaves the extraprocesses lying around |
| 1:54AM |
0 |
Queue forks asterisk and then leaves the extra processes lying around |
| 1:34AM |
2 |
flash transfer problem in asterisk integration with old PBX |
| 1:33AM |
1 |
Operating queues with clients on a legacy PABX |
| |
| Tuesday November 7 2006 |
| Time | Replies | Subject |
|---|
| 11:31PM |
0 |
Follow Me problems |
| 10:58PM |
0 |
RxFAX - How to catch errors in the dialplan |
| 6:42PM |
0 |
test please ignore |
| 6:24PM |
1 |
Help with latest Asterisk on latest Debian |
| 6:11PM |
1 |
Glitches in sound every time that Asterisk receives reINVITEs |
| 6:11PM |
1 |
Fax & Digium |
| 6:11PM |
1 |
Why dont my messages get through |
| 6:11PM |
1 |
[resolved] asterisk 1,4 and google talk |
| 6:10PM |
0 |
test message please ignore |
| 6:10PM |
2 |
Microsoft will enter VoIP market in earnest next year, says Ballmer |
| 6:10PM |
0 |
astertest |
| 6:10PM |
0 |
incoming call destination: IVR not working |
| 6:09PM |
2 |
Pressing "*" makes Asterisk destroy my call |
| 6:09PM |
3 |
Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue |
| 6:08PM |
0 |
How is ANI "Usually" sent on an ISDN PRI? |
| 6:08PM |
1 |
loosening voicemail file permissions for msg????.txt and msg????.wav |
| 11:39AM |
4 |
Queues and multiple lines |
| 11:10AM |
1 |
Grandstream TFTP system wide settings |
| 11:08AM |
0 |
my future count on this please help |
| 11:06AM |
2 |
g729 |
| 9:45AM |
0 |
Generating Recall/Flash using Zaptel |
| 9:14AM |
3 |
connect Sipura with Asterisk - both behind NAT |
| 9:02AM |
4 |
"Sticky" Polycom 501 keys and handset |
| 8:59AM |
2 |
hicecaller ID |
| 8:21AM |
0 |
failed to authenticate on invite |
| 8:14AM |
1 |
How do I make this stop? (Bridging of IAX channels?) |
| 8:03AM |
3 |
capiAnswerFax |
| 6:23AM |
0 |
Asterisk SMS: Experience with EMS? |
| 5:42AM |
1 |
Problem: 2 second silence at the beginning of most calls |
| 3:28AM |
2 |
Mapping CLI'S in Dialplan |
| 3:15AM |
1 |
Upgrading sox |
| 2:23AM |
0 |
Asterisk Showing 404 not found when calling from third party SIP server (newbie question) |
| 1:54AM |
2 |
Snom 360 flickering screen |
| 1:27AM |
0 |
Asterisk and FreeTDS 0.64 or >0.63 |
| 1:12AM |
0 |
Desired apps |
| 12:29AM |
1 |
Dial plan Question |
| |
| Monday November 6 2006 |
| Time | Replies | Subject |
|---|
| 9:05PM |
0 |
silencedetecthangup= |
| 6:39PM |
1 |
asterisk 1,4 and google talk |
| 6:19PM |
2 |
Polycom autoprovision behind a NAT |
| 5:34PM |
3 |
Question on Aastra phones and Astrisk |
| 5:23PM |
0 |
help for recording |
| 3:14PM |
1 |
Polycom dealers in Toronto/London ON |
| 2:21PM |
2 |
how to indicate an non-existent number? |
| 2:16PM |
1 |
Do my messages come through? |
| 1:29PM |
0 |
IAX FWD - down, running own proxy or stun server |
| 1:27PM |
0 |
TrixBox and MP104 FXO (AudioCodes GW) |
| 12:40PM |
0 |
Disappearing voicemail? |
| 12:13PM |
0 |
OT: BarCamp USA |
| 11:31AM |
1 |
Polycom unable to answer more than 3 calls at a time |
| 11:30AM |
0 |
Re: Definity ISDN PRI |
| 10:40AM |
1 |
Asterisk servers being greedy and not letting go of the media path. (using IAX2 channels) |
| 10:12AM |
2 |
receptionist - large number of concurrent calls - example needed |
| 10:03AM |
2 |
Queue time out |
| 8:28AM |
7 |
several behind NAT |
| 8:26AM |
1 |
Register vs. Host=IPADDR |
| 8:22AM |
1 |
Page system using the sound card |
| 8:15AM |
1 |
Amending CLI in Dialplan |
| 6:47AM |
4 |
Port Range |
| 6:31AM |
1 |
Is it possible have multiple ip numbers for an extension? |
| 4:04AM |
2 |
Ring locally when home or roadwarrior via IAX when away |
| 3:53AM |
1 |
Audio goes one way during the call for a fewseconds. Is it RTP, NAT, dyndns, or what it is? |
| 3:14AM |
7 |
DTMF Tones occuring randomly |
| 2:36AM |
2 |
Fast detection of unreachable SIP clients? |
| |
| Sunday November 5 2006 |
| Time | Replies | Subject |
|---|
| 8:30PM |
1 |
asterisk DTMF detection |
| 7:41PM |
0 |
xfsound=beep is not beeping |
| 6:02PM |
0 |
Use astbill to bill Trixbox |
| 4:09PM |
1 |
Definity Asterisk Caller ID Issue |
| 2:54PM |
9 |
names of SIP aware firewalls |
| 2:45PM |
3 |
Very high translation costs for g729 |
| 1:57PM |
1 |
Call Quality Issues with IAX? |
| 11:18AM |
0 |
Voicemail.conf multi languages |
| 9:31AM |
0 |
Free PBX, was - Re: best gui |
| 7:52AM |
2 |
Definity Asterisk CallerID Issue |
| 5:05AM |
0 |
call transfer problem |
| 4:34AM |
1 |
Reading Voicemail Config from MySQL |
| 2:51AM |
1 |
Asterisk and FXO Digium Card for Analog line |
| 2:00AM |
1 |
skype and SIP hardware for linux |
| 1:40AM |
3 |
Anybody used Asterfax? |
| 12:02AM |
1 |
Hang up on SIP calls if connected to long |
| |
| Saturday November 4 2006 |
| Time | Replies | Subject |
|---|
| 9:54PM |
1 |
Newbie questions about Voice mail |
| 9:37PM |
4 |
SPA3k wired to PAP2 for echo testing |
| 7:16PM |
1 |
FXO lines taking several rings to answer, always two |
| 6:40PM |
1 |
Only one out of 10 remote extensions expiring registry |
| 6:34PM |
4 |
g729 codec help |
| 5:30PM |
1 |
Pass through |
| 4:02PM |
1 |
Redirect problems using IAX2 and SIP |
| 10:53AM |
0 |
Upgrading from 1.2.12.1 to 1.2.13 |
| 8:25AM |
0 |
Asterlink Down? |
| 8:24AM |
0 |
I suggest using TFTP. |
| 8:10AM |
1 |
Hairpinning problems using IAX2 and SIP |
| 5:19AM |
2 |
app_prepaid won't load - undefined symbol mysql_num_fields |
| 4:39AM |
1 |
My first Asterisk - Not recognizing X100P clone |
| 3:29AM |
0 |
iax2 qualify - false "peer unreachable" |
| 2:14AM |
2 |
Asterisk upgrade from 1.0.9 to 1.2.6 not working |
| |
| Friday November 3 2006 |
| Time | Replies | Subject |
|---|
| 9:59PM |
1 |
Polycom SIP 2.0.2 firmware |
| 9:58PM |
1 |
Patton 1400 |
| 8:56PM |
0 |
DID with extensions |
| 7:40PM |
1 |
Why only one out of many IP Phones re-registering every one minute |
| 5:46PM |
1 |
SendDTMF() behaves strangely |
| 5:30PM |
1 |
Unicall's MFCR2 with Asterisk 1.4 |
| 5:17PM |
1 |
Polycom provisioning and Pure-FTP : problems |
| 1:44PM |
1 |
Experiment: Dialplan size vs. Speed |
| 12:56PM |
1 |
Random 'no audio' problem |
| 12:07PM |
1 |
International dialing with GPX-2000 and "early dial" |
| 11:36AM |
3 |
Extension Spy |
| 11:27AM |
1 |
SIP - IAX Attended transfer |
| 11:20AM |
0 |
Problem with Realtime/ODBC |
| 9:17AM |
0 |
Binding a peer context to a specific IP address |
| 9:12AM |
0 |
Configure Max TNT PRI to SIP with Asterisk |
| 8:45AM |
1 |
Monitor, MixMonitor and volume levels |
| 8:30AM |
2 |
Is fax bridging with TDM2400 working (or about to work) ? |
| 8:27AM |
2 |
AEL2 in 1.2 |
| 8:24AM |
3 |
Problems Overwriting CallerID with True ANI |
| 8:02AM |
1 |
Clearing Outgoing Call Queue |
| 7:57AM |
1 |
In bound SIP context issue |
| 7:22AM |
3 |
Nortel Option 11C and SIP gateway integration |
| 6:43AM |
0 |
Caller ID 1.2.10 |
| 6:39AM |
1 |
TDM400 hungup problem |
| 6:26AM |
0 |
Pass-through any codecs |
| 5:57AM |
1 |
Error updating bootrom on Polycom phones..doesn't even download the bootrom! |
| 5:48AM |
4 |
some simple newbie help with dialplan needed... |
| 5:43AM |
1 |
Help for registration with "sipdiscount" |
| 4:26AM |
4 |
Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is? |
| 4:22AM |
0 |
*****SPAM***** Meetme Conference Rooms |
| 4:20AM |
1 |
How do i redirect a call without answering it? SIP channel |
| 4:04AM |
0 |
The most cost effective and Asterisk friendly T.38 gateway ? |
| 3:45AM |
0 |
How to reduce latency and first real ring timing? |
| 2:16AM |
0 |
Asterisk IAX Trunk and Queues |
| 1:26AM |
2 |
PAP2 to use on my asterisk. |
| 1:21AM |
1 |
SV: ip address in CDR |
| 1:16AM |
1 |
Cisco 7960 - Fast dial |
| |
| Thursday November 2 2006 |
| Time | Replies | Subject |
|---|
| 11:05PM |
1 |
is IAX required for firewall and router? |
| 10:17PM |
0 |
ip address in CDR |
| 9:57PM |
2 |
fax eater |
| 6:42PM |
0 |
a extension intentionally dropped in favor of * ? |
| 4:08PM |
0 |
Recall: regexten & regcontext broken for SIP? |
| 1:42PM |
0 |
Polycom 501 supports now FTPS? |
| 1:17PM |
0 |
Asterisk 1.2.16 AIX2 - SIP Attended transfer |
| 12:38PM |
3 |
Polycom latest version |
| 12:31PM |
2 |
Tampa Bay Asterisk Users Meetup on Monday |
| 11:31AM |
4 |
Out Dial Interface for Asterisk |
| 11:19AM |
0 |
testing |
| 11:14AM |
1 |
Voicemail issues |
| 10:52AM |
0 |
Problem with MYSQL commands in dialplan |
| 9:46AM |
0 |
Static Realtime Select from Database |
| 9:13AM |
1 |
AGI Problems |
| 9:12AM |
4 |
Running asterisk with 'sudo' |
| 8:51AM |
2 |
Grandstream HandyTone-488 with Asterisk ? |
| 8:13AM |
1 |
AstLinux 0.4.4 Released! |
| 6:39AM |
3 |
How to determine which version is running |
| 6:18AM |
0 |
Subscriptions and call back on busy problems with Snom phones |
| 6:09AM |
2 |
Error installing asterisk, module zaptel not found |
| 4:48AM |
1 |
VM Language |
| 4:46AM |
2 |
Can some moderator kick this person out of the list |
| 4:41AM |
1 |
How to clear trixbox configuration |
| 4:27AM |
1 |
Lucent TNT Help |
| 4:18AM |
0 |
blindtransfer and initiator hangup |
| 4:11AM |
0 |
Macro variables and redirects |
| 4:05AM |
0 |
Wait for an extension and dial. Why does this not work? |
| 3:59AM |
0 |
sound-files not playing? |
| 2:59AM |
0 |
Extending a call limited by L in Dial app |
| 2:20AM |
1 |
Auto dial out and auto answer |
| 2:09AM |
1 |
ZAPtel channel dance |
| 12:47AM |
3 |
mpg123 new version |
| |
| Wednesday November 1 2006 |
| Time | Replies | Subject |
|---|
| 11:55PM |
3 |
Polycom 601 Phone can not find TFTP server |
| 11:25PM |
0 |
Using asterisk as a call router between pbxs |
| 10:54PM |
1 |
Asterisk Manager and Ruby |
| 10:45PM |
1 |
Videoconferencing solutions with Asterisk- |
| 10:08PM |
2 |
echo with spa-3000 |
| 9:26PM |
4 |
My Phone Review- Large Scale Corp Deployment. |
| 8:48PM |
0 |
Problem with libpri? |
| 8:27PM |
0 |
New Dell range |
| 7:39PM |
2 |
Two Sipura 3000s |
| 7:38PM |
1 |
IAX problem |
| 5:26PM |
0 |
Fwd: Benachrichtung zum +ANw-bermittlungsstatus (Fehlgeschlagen) |
| 5:11PM |
1 |
PURE OUTBOUND setup (how do I proceed from here?) |
| 5:11PM |
2 |
Realtime, DUNDi and regexten |
| 4:18PM |
2 |
Echo Issues |
| 3:47PM |
3 |
Sound breaking. Because of Tormenta2 PRI Interface Card or something else |
| 3:32PM |
1 |
connecting internal line with external line |
| 2:45PM |
0 |
TE110P Card Little help |
| 2:34PM |
0 |
Can I use Realtime entries to do multiple registers to same trunk/peer |
| 2:05PM |
6 |
Java Web Phone |
| 12:47PM |
1 |
imap on debian |
| 12:14PM |
3 |
Remote-Party-Id and Attended Transfers |
| 11:12AM |
2 |
Still no CLI in 1.4 branch (OSX) |
| 11:09AM |
0 |
Cisco 7960 password/shared secret problem --- Related to OS X ? |
| 11:04AM |
2 |
Polycom Managment tools |
| 10:06AM |
0 |
AW: Which IP phones have best voice quality, preferably under $150 |
| 9:32AM |
0 |
[SPAM HEADER] - Which IP phones have best voice quality, preferably under $150 - Email found in subject |
| 9:28AM |
5 |
DTMF over IAX |
| 9:28AM |
1 |
Upgrading from 1.0.9 to 1.2.6 |
| 9:23AM |
0 |
[SPAM HEADER] - RE: Re: Newbie Questions - Grandstorm phones? - Email found in subject |
| 9:20AM |
2 |
Asterisk manager |
| 9:17AM |
4 |
Which IP phones have best voice quality, preferably under $150 |
| 9:01AM |
3 |
Re: Newbie Questions - Grandstorm phones? |
| 8:26AM |
1 |
[SPAM HEADER] - Re: Snom or Cisco Phones? - Email found in subject |
| 7:33AM |
0 |
Neat Application for Text to Speech |
| 7:19AM |
2 |
a2billing |
| 7:10AM |
0 |
wav format isn't compatible with Windows Media Player |
| 6:22AM |
0 |
AEL2 - CUT function usage |
| 4:57AM |
8 |
${CALLERIDNUM} |
| 4:55AM |
2 |
Help me on Call parking |
| 4:23AM |
3 |
Manager API - Originate Call - Need Help |
| 4:00AM |
0 |
SIP realtime issues |
| 1:25AM |
0 |
Need help connecting Alcatel 4400 PBX to Asterisk |