Saturday July 31 2004 |
Time | Replies | Subject |
11:10PM |
3 |
Asterisk on Sparc64 |
7:10PM |
2 |
480i User Feedback With Asterisk (fwd) |
5:05PM |
1 |
Hiring Setup |
12:44PM |
2 |
different pridialplan for different channels in zapata.conf |
12:27PM |
1 |
learning from the audio folks |
10:38AM |
3 |
MGCP & Cisco ATA 186 Help |
8:48AM |
0 |
Trunk doesn't work Adit 600/T100P |
8:30AM |
1 |
Asterisk does not disconnect SIP call |
8:24AM |
2 |
Asterisk scalability? |
7:59AM |
2 |
Which version of MySQL works with cdr_addon_mysql? |
6:36AM |
3 |
one extention, multiple phones |
|
Friday July 30 2004 |
Time | Replies | Subject |
11:59PM |
2 |
Sipura 3000 PSTN disconnect in the UK |
11:56PM |
0 |
G.729 <-> ZAP ? |
6:59PM |
3 |
VoiceMail Not releasing |
6:33PM |
1 |
Compiling * on OpenBSD 3.5 |
4:48PM |
1 |
VoIP gateway (2 FXO, 2 FXS) |
1:58PM |
1 |
SIP connections do not hang up |
1:09PM |
2 |
asterisk-oh323-0.6.3a |
1:05PM |
0 |
Transfer call help needed |
1:04PM |
0 |
X100p / OEM X100p In MACs |
11:44AM |
9 |
Rodopi Billing |
11:39AM |
3 |
New to IP-PBX |
10:58AM |
0 |
Two different voicemail messages |
10:08AM |
0 |
AstriCon 2004 Early Bird Period Ends Saturday! |
8:40AM |
1 |
FW: Limit incoming calls to SIP Channels |
8:28AM |
0 |
Transfers- Please Help ASAP |
8:20AM |
3 |
Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing |
8:15AM |
2 |
zaphfc hardware & sound trouble |
8:04AM |
2 |
Outgoing *-initiated calls from spool directory not working |
7:44AM |
2 |
audio delay over time on Zap to SIP |
6:50AM |
1 |
cisco ubr924 |
6:14AM |
1 |
Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing |
6:05AM |
1 |
Outgoring spool |
4:22AM |
1 |
Running AGI script on answer. |
3:42AM |
0 |
Help for VoIP Gateway with 2 x FXO & 2 x FXS |
3:34AM |
0 |
Cisco PRI CLID outbound fails |
2:40AM |
0 |
How to detect the sent status of the Fax |
2:18AM |
5 |
Non standard usage of X100P card. |
1:49AM |
1 |
Asterisk+zaphfc: how to answer incoming calls from a BRI (EuroISDN)? |
|
Thursday July 29 2004 |
Time | Replies | Subject |
11:46PM |
2 |
Astricon Dev Meeting On Line |
11:41PM |
5 |
playing a sound during a call |
11:13PM |
0 |
G.729 between Zap and SIP |
7:09PM |
1 |
Re: Zaptel doesn't see remote hangup ? |
6:40PM |
2 |
chan_sccp2 testers needed |
5:49PM |
2 |
Astricon Conference Call? |
5:14PM |
0 |
DISA and notransfer/reinvite? |
3:27PM |
1 |
permit/deny in sip.conf not working |
2:53PM |
0 |
CREATIVE Critisim wanted for new application |
2:06PM |
1 |
Limit // incoming calls to Queue Agents |
1:47PM |
1 |
snom 200 and call parking? |
11:13AM |
2 |
Zultys Zip 4x4 |
11:10AM |
1 |
OH323 and codec selection |
10:49AM |
1 |
Experiance of shipping into the UK |
9:01AM |
1 |
Unauthenticated calls from a specific IP |
8:32AM |
1 |
Asterisk and festival |
8:30AM |
0 |
*** Asterisk Summer News: The heat is on! |
7:43AM |
6 |
Zaptel doesn't see remote hangup ? euro-isdn |
7:23AM |
0 |
Winbond drivers |
7:14AM |
4 |
One More IP Phone for interoperability with Asterisk |
7:14AM |
1 |
(no subject) |
7:14AM |
2 |
Aastra 480e phone ADSI config |
7:09AM |
1 |
incoming caller doesn't hear rining. |
6:59AM |
5 |
Astricon Conference Call????????? |
6:59AM |
3 |
queue_log question: which endpoint was connected? |
6:26AM |
2 |
BugetTone Bug Showstopper, |
6:14AM |
1 |
Quadbri in NT Mode against PBX. |
6:10AM |
2 |
where to start asterisk sourcecode |
5:29AM |
0 |
SIP and RTP / 302 after 18x / Call forwarding after announce |
5:11AM |
0 |
New app: Consultative transfer for each phone |
4:56AM |
1 |
call center with * |
4:46AM |
3 |
Polycom IP Soundpoint 600 & early dial |
3:52AM |
1 |
PSTN tone simulation |
3:38AM |
0 |
ParkAndAnnounce command !!! |
3:21AM |
3 |
IAXy config samples |
3:15AM |
0 |
Asterisk PID file |
2:50AM |
10 |
Asterisk GUIs at Astricon * REMINDER * |
2:04AM |
0 |
Compatible E1 card |
1:57AM |
1 |
SIP Outbound Proxy Support |
12:35AM |
1 |
ToS flags for VoIP |
|
Wednesday July 28 2004 |
Time | Replies | Subject |
11:10PM |
0 |
sip phone, receiving calls but not placing any call |
10:58PM |
1 |
rate-engine loading error |
10:55PM |
1 |
Please share your Solaris experiences on the Asterisk Solaris Wiki page |
9:36PM |
1 |
using round-robin dns for sip registrations |
8:38PM |
2 |
call waiting, * and FXO |
8:29PM |
4 |
X-Lite to Asterisk through NAT? |
7:33PM |
3 |
faxing |
6:57PM |
1 |
false busy using sipura spa-3000 with asterisk on solaris |
5:06PM |
0 |
X101P problem with latest FreeBSD zaptel drivers |
4:53PM |
1 |
Reverse Battery Disconnect Supervision in X100P or TDM400P FXO |
4:53PM |
0 |
D-Link DG-104SH H323 problemm |
4:34PM |
0 |
broadvoice/asterisk incoming calls problem |
3:40PM |
0 |
SipTone 4 Sale... |
3:13PM |
3 |
Workaround for BroadVoice and possibly others... |
2:59PM |
1 |
is chan_skinny broken? |
2:54PM |
0 |
"Re:" is ok |
2:29PM |
2 |
Collect recording before sending to extension or queue |
2:12PM |
4 |
Cisco 7960 backlight and list etiquette? |
1:13PM |
1 |
Zap hanging up others zap. |
12:48PM |
0 |
source for zultys zip phones? |
12:08PM |
3 |
Changing Transfer key |
12:02PM |
1 |
ParkAndAnnounce Doesn't Release when Call is Picked-up |
11:52AM |
3 |
Problems with * and Grandstream Budgetone 100 |
11:44AM |
3 |
New Zealand DIDs |
11:40AM |
0 |
VoiceMail Problem or bug? |
11:20AM |
3 |
Asterisk, PBX, VoIP and PRI |
10:56AM |
0 |
Unix ODBC Segmentation Fault |
8:43AM |
2 |
Music On Hold - not working for me... |
8:16AM |
4 |
MS SQL & Free TDS |
7:52AM |
0 |
Asterisk-and-MacOSX News |
7:31AM |
0 |
only one call at the time |
7:30AM |
2 |
Desired Install in MotorHome |
6:36AM |
2 |
Rate Engine Compile Error |
6:24AM |
8 |
Best Linux for Asterisk |
6:23AM |
5 |
Trouble compiling asterisk-addons MySQL |
6:13AM |
3 |
MGCP & Caller ID |
5:35AM |
2 |
Asterisk voicemail from mysql no longer working |
5:22AM |
1 |
Access voicemail from Cisco 7960 |
5:22AM |
0 |
Commercial Asterisk Support |
5:09AM |
1 |
Outgoing works, incoming doesn't... |
4:27AM |
1 |
Problems Compiling Asterisk-oh323-0.6.2 |
4:27AM |
3 |
shan:Needed help |
4:23AM |
2 |
IAX transfer bug in last CVS ? |
4:23AM |
0 |
Shan:Help in configuring Dialplan |
1:04AM |
1 |
problems with digium cvs???? |
|
Tuesday July 27 2004 |
Time | Replies | Subject |
11:34PM |
0 |
E-mail address for DIAX support has been changed |
9:44PM |
0 |
Example Polycom XML applications, anyone? |
8:05PM |
1 |
re: asterisk/incoming calls attn:James |
8:04PM |
0 |
converting gsm file to g729 format |
7:50PM |
6 |
Successfully Using $135 Avaya sip phone |
6:43PM |
1 |
Broadvoice - incoming calls problem |
6:10PM |
2 |
Polycom IP-600 leasing? |
5:53PM |
6 |
zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) |
3:52PM |
5 |
Has anyone tried using a Sipura-3000 as an FXO device for *? |
3:27PM |
1 |
Dial out problems with Digium TDM400P card. |
1:31PM |
1 |
Problems connecting xlite phone |
1:01PM |
6 |
Cisco 7910 Setup |
12:57PM |
0 |
Local voice feedback |
12:52PM |
1 |
"Broadcasting" Calls? |
12:11PM |
0 |
Touch Tones not working after call is made |
11:40AM |
0 |
Strange RTP audio errors on console |
11:38AM |
0 |
Strange I4L troubles |
11:27AM |
1 |
GrandStream BudgeTone 100 Firmware? |
10:27AM |
2 |
Open for beta testers - free calls in us/canada |
10:01AM |
0 |
Comparing * with SIPxchange |
9:17AM |
2 |
g729 + GSM + g723 |
9:15AM |
0 |
Re: [Asterisk-doc] New Hardware Support |
9:09AM |
0 |
Off Toppic-ish Telephony question |
9:03AM |
0 |
switchhook flash / link |
8:03AM |
0 |
Re: Nat...again... |
6:51AM |
0 |
safe_asterisk and odbc |
6:40AM |
2 |
Problems with variables |
6:25AM |
5 |
User-Oriented Management of Asterisk |
5:50AM |
1 |
VoicemailMain Issues |
5:32AM |
0 |
h323.conf - host= multiple ip addresses |
5:20AM |
1 |
asterisk <-> stanaphone? |
4:30AM |
1 |
Puzzled by CapiCD (call deflection to mobile phone) |
4:14AM |
5 |
sip over h323 |
3:44AM |
1 |
Hook-flash timing |
3:27AM |
0 |
and a eNUM |
3:27AM |
2 |
Enum |
3:12AM |
0 |
Re: Nat...again... |
3:07AM |
3 |
2 cards |
2:06AM |
6 |
Asterisk to CCM |
1:56AM |
2 |
Using rxfax over SIP |
1:49AM |
0 |
How to allow softphone to dial thru with full SIP URI? |
1:27AM |
3 |
Pickup an unanswered line |
12:59AM |
2 |
Transfer / Park issue |
12:20AM |
7 |
broadvoice/asterisk |
12:14AM |
2 |
Decent Asterisk Compatible VoIP HardPhone - Australia |
|
Monday July 26 2004 |
Time | Replies | Subject |
10:57PM |
6 |
New Beta version of Grandstream Firmware 1.0.5.9 |
10:20PM |
0 |
Suggestion on reporting provider problems (was [RANT] Boadvoice...) |
7:54PM |
0 |
rtp.c:487 ast_rtp_read: Unknown RTP codec 72 received |
7:49PM |
1 |
voicemail+g729 |
4:32PM |
0 |
Timing pips ? |
4:10PM |
4 |
Pickup zap channel already in use? |
3:44PM |
0 |
callgroup pickup |
3:19PM |
1 |
Integrated Networks IN1002 SIP Phones? |
3:16PM |
0 |
Sample extensions & SIP Conf files |
3:10PM |
1 |
Voicemail from MySQL and Directory |
2:07PM |
1 |
Global Variables Scope |
2:03PM |
5 |
Upgrade from Altigen |
1:59PM |
4 |
IRC Etiquette |
1:07PM |
2 |
Broadvoice problems again Attn: James |
1:07PM |
4 |
Asterisk for a large scale implementation |
1:03PM |
4 |
Echo in asterisk phones. |
12:34PM |
0 |
Sip usage and busy? (BroadVoice related) |
11:47AM |
2 |
IAX2 to IAX2...i'm obviously an idiot!! |
11:03AM |
3 |
ResponseTimeout, Straight to operator? |
10:54AM |
1 |
drivers, kernel 2.6 and distribution |
10:42AM |
3 |
Source for 9-911 Labels to attach to phones? |
10:00AM |
2 |
Display and UUS IEs on PRI - Q.931 question |
8:59AM |
0 |
Can't dial SIP<->EuroISDN (HFC-S based PCIISDN card): Unable to create channel of type 'Zap' |
8:35AM |
0 |
Voicemail Hangup detection issue |
8:29AM |
1 |
snom 105 Attended Transfer does not work |
8:14AM |
1 |
Nat...again.... |
7:35AM |
6 |
Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap' |
7:18AM |
0 |
Having * show lines in use on hard phones. |
6:48AM |
0 |
PacketCable & Asterisk |
6:42AM |
0 |
How to use SER together with Asterisk and to preserve Asterisk features like Dialplan, etc. |
6:30AM |
0 |
Problem with T1 signalling mode |
6:30AM |
0 |
X100P Red Alarm after echo on the line |
6:26AM |
1 |
Feature question |
5:40AM |
5 |
GrandStream CallerID |
5:36AM |
0 |
Can Asterisk register through a sip proxy with a username incorporating a different server? |
4:51AM |
0 |
Call Forwarding/Redirection? |
4:17AM |
0 |
Astricon news :: The conference agenda now published |
3:03AM |
0 |
IVR help |
2:40AM |
3 |
TE405P and E1 |
1:16AM |
1 |
bri-stuff NT mode |
12:58AM |
0 |
Asterisk and AOC? |
12:30AM |
1 |
Nested/cascading "switch" statements: possible? |
|
Sunday July 25 2004 |
Time | Replies | Subject |
10:04PM |
1 |
Asterisk CDR & UniqueID |
7:59PM |
1 |
Busydetect problems |
7:55PM |
17 |
Broadvoice problems again |
7:15PM |
1 |
pound key tone generated after call answered? |
5:55PM |
1 |
Can not make progdocs |
5:52PM |
1 |
X100P Inbound Issue |
4:47PM |
2 |
Incoming SIP gateway context? |
4:43PM |
1 |
Web Based Admin Interface |
3:33PM |
1 |
how do I play congestion tone when Zap channels are full? |
11:17AM |
0 |
trunk line usage report |
8:24AM |
2 |
which DB to use? or Why Berkley DB vs. MySQL, etc |
7:23AM |
3 |
FXS vs. FXO |
1:09AM |
1 |
sip ua---------asterisk-------h323gw |
|
Saturday July 24 2004 |
Time | Replies | Subject |
8:52PM |
3 |
Help with T1 PRI Configuration |
8:46PM |
5 |
VoiceMail Group Broadcasting |
8:02PM |
4 |
Layer 3 VPN Question |
4:45PM |
1 |
Autologout of dynamic agents |
3:22PM |
2 |
John Vogel |
11:25AM |
1 |
Do not buy "Asterisk for Small Office Setup"! |
11:03AM |
1 |
Please help I fear I have missed something very important! but what? |
10:43AM |
1 |
Play CD! |
7:44AM |
1 |
Hack to make * -> (H323) -> CCM -> IOS GW work |
7:26AM |
0 |
PBX functions and different channels grouping |
7:25AM |
1 |
Attendant configured AutoAttendant |
7:23AM |
0 |
Should configured devices show up with "show channels" |
7:06AM |
2 |
Need to block incoming collect calls |
6:50AM |
0 |
Question when using a Cisco as a PSTN GW |
6:33AM |
2 |
yes shady dial running now but not dialling |
3:31AM |
6 |
h323 to SIP Server Load |
|
Friday July 23 2004 |
Time | Replies | Subject |
10:44PM |
3 |
"Asterisk for Small Office Setup" |
10:44PM |
0 |
Cisco 7940 hook-flashing blind xfer. |
4:53PM |
0 |
Deltathree and Go2call registration ? Anyone ? |
4:11PM |
0 |
SIP302 Redirect Problem (Moved temporarily) fixed, Asterisk working now with Nikotel!!! |
3:06PM |
3 |
Wildcard T100P in 1U |
2:51PM |
2 |
Reverse number lookup |
2:40PM |
2 |
Queue Statistics Line |
2:32PM |
6 |
Robbed Bit T1 Configuration |
2:19PM |
0 |
Reinstalled FRom CVS - Things are really different now... |
1:34PM |
0 |
Pipecall problem |
12:26PM |
1 |
chan_alsa record problem |
11:54AM |
1 |
Norstar ATA2 signalling protocol? |
11:15AM |
1 |
addmailbox |
11:02AM |
1 |
No channel type registered for 'ZAP' |
10:07AM |
0 |
AstriCon Update: Very Low Priced Ground Transport Available |
10:01AM |
0 |
Problems calling a phone number through a X101P card |
9:21AM |
3 |
DTMF stops working w/ Voicemail |
9:00AM |
4 |
Doublehash transfers |
8:33AM |
4 |
hang up when going to voicemail |
8:12AM |
0 |
MGCP and one-way audio |
7:50AM |
0 |
cisco 7940 audio problems to PSTN |
7:34AM |
0 |
using the older iax1 protocol |
6:53AM |
6 |
Priorizing of packets |
6:44AM |
3 |
Status of Q.SIG on Asterisk? |
6:09AM |
0 |
* poll |
6:08AM |
4 |
still can't load oh323 - Are we not supporting H.323 any more? |
5:03AM |
0 |
SIP - Cancel request fails with "481 no such call" |
4:13AM |
0 |
qudBRI and transfering calls with the latest RC2. |
3:55AM |
0 |
ringing tones for E100P (like early B3 in chan_capi) |
3:43AM |
3 |
Grandstream Budgetone 101 channels don't disappear on hangup. |
2:27AM |
0 |
AW: Large Enterprises using asterisk |
1:12AM |
2 |
Cisco 12sp firmware... Anyone got it??? |
12:37AM |
0 |
Configuration help |
|
Thursday July 22 2004 |
Time | Replies | Subject |
9:35PM |
2 |
NAT + iConnectHere Broken in 1.0RC1 |
8:16PM |
2 |
D-Link 1120M |
7:35PM |
0 |
Re: VSP? Looking for advice |
4:27PM |
0 |
Outbound Dialing Format |
2:19PM |
4 |
VSP? Looking for advice. |
1:49PM |
0 |
Connecting more Asterisk Servers in Cluster to works as one IP PBX |
1:06PM |
0 |
still can't load oh323 |
11:09AM |
6 |
D-Link DPH-80S vs * |
11:07AM |
7 |
Asterisk and Linejacks |
10:53AM |
0 |
Future installation questions - what do I need |
10:19AM |
1 |
app_dbodbc URGENT |
10:05AM |
1 |
RAID/SCSI/IDE/SATA and a TE405P (or T100P) c ard. Should I expect problems? |
9:22AM |
1 |
How to calculate the price for Asterisk based Solution |
9:18AM |
2 |
Nortel SL1 protocol and *? |
9:04AM |
1 |
no incoming pstn ring tone |
8:49AM |
0 |
(no subject) |
8:39AM |
0 |
Call Quality - Factors and Config Values |
8:38AM |
0 |
New astGUIclient version released 1.0.3 |
8:35AM |
0 |
Application Hangup not hanging up, possible dialplan cockup? |
8:09AM |
1 |
Faild Echotest |
7:41AM |
1 |
Asterisk-oh323 on fedora Core 2 - Anyone has a working install? |
7:12AM |
2 |
MSSQL ODBC CDR |
7:06AM |
1 |
Echo Canceller Wiring (Tellabs.. HOWTO..?) |
6:11AM |
1 |
Voicetronix Openswitch |
6:08AM |
0 |
Re: Astricon costs |
6:07AM |
1 |
Can anybody recommend a good T1/PRI provider ? |
6:02AM |
0 |
ODBC.conf |
5:57AM |
1 |
Daytime - Nighttime |
5:53AM |
1 |
Can anybody recommend a good T1/PRI provider? |
4:31AM |
8 |
debian install zaptel |
2:45AM |
0 |
Re: h323ep----gnugk-----astersik------h323ext |
2:45AM |
0 |
ZAP Channel doesn't hang up - X100P |
2:42AM |
2 |
error while compiling asterisk-oh323 |
2:20AM |
1 |
Symbian Softphone |
1:48AM |
1 |
Sip -> H323 using oh323 and G729 |
12:08AM |
0 |
SIP Providers Setup.- Go2call, deltathree, etc...setup files ? any |
|
Wednesday July 21 2004 |
Time | Replies | Subject |
11:12PM |
0 |
Asterisk sees inbound call, but won't answer |
10:28PM |
0 |
X100P only dials a single digit |
10:24PM |
11 |
Large Enterprises using asterisk |
9:42PM |
2 |
Anyone heard of BroadVox direct? |
9:15PM |
2 |
TDM04B Dead? |
5:44PM |
2 |
Anyone else having Broadvoice Problems? |
4:08PM |
0 |
sangoma AFT A101 works with Asterisk? |
3:20PM |
6 |
Astricon costs... |
3:14PM |
1 |
Install problems |
1:52PM |
1 |
Zaptel - delay before dialing last DTMF digit? |
1:49PM |
0 |
Adding another channel to a Dial() already in progress |
1:24PM |
2 |
Compiling Asterisk (zaptel) failed on Debian 2.4.18-686 |
1:00PM |
1 |
h323 call flow fails |
12:57PM |
0 |
Mac OS X installer for Asterisk - Missing Files Patch Now Available |
12:33PM |
1 |
Error in compilation [URGENT]. |
12:23PM |
0 |
go2call setup ? |
12:19PM |
3 |
X100P panic |
12:14PM |
5 |
RAID affecting X100P performance... |
12:14PM |
2 |
ENUM lookup help |
12:04PM |
3 |
Help needed for Seting Up Asterisk |
11:58AM |
4 |
Future installation questions - what do I need? |
11:58AM |
0 |
NAT table expiration |
11:48AM |
1 |
roblems with Junghanns QuadBri |
11:45AM |
3 |
Asterisk Server gives 403 forbidden |
11:45AM |
0 |
SIP Hard Disconnect Detection |
10:58AM |
1 |
TDM400 dropping loop current 10 seconds after answer |
10:37AM |
1 |
E1 card with R2 |
10:24AM |
3 |
echotraining on T1 circuits |
10:09AM |
2 |
fonction Getvar |
9:46AM |
1 |
bare minimums |
9:36AM |
0 |
AW: Asterisk RC1 and bristuff |
9:14AM |
0 |
S100I-IAXY |
8:48AM |
1 |
rxgain - txgain values |
7:55AM |
1 |
Bri solution for Asterisk |
7:53AM |
0 |
extensions.conf variable declaration |
6:43AM |
1 |
chan_capi-0.3.4b and asterisk last cvs |
6:13AM |
0 |
Cisco 7960, multiple registrations, and NAT |
5:58AM |
2 |
Caller based routing |
5:38AM |
0 |
Queue Monitoring |
5:04AM |
2 |
queue stats |
4:25AM |
0 |
chan_capi busydetect |
4:07AM |
0 |
Asterisk RC1 and bristuff |
4:00AM |
1 |
libr2 completion staus |
3:59AM |
0 |
music during conversation |
3:17AM |
1 |
Digium card x100p |
3:05AM |
0 |
Voicemal error |
2:47AM |
0 |
Senao SI-7800 |
12:51AM |
0 |
Zaptel Hung Up on x101p and cisco analog line |
12:27AM |
0 |
Integrated management tool? |
|
Tuesday July 20 2004 |
Time | Replies | Subject |
10:51PM |
2 |
No Ringing. |
10:39PM |
1 |
* CLASS codes |
9:09PM |
0 |
libr2 status |
7:03PM |
3 |
# Transfer Context |
5:54PM |
1 |
Latest CVS (7/20/2004) stops answering SIP calls after 5 min |
5:53PM |
0 |
Festival 2.0 |
4:32PM |
0 |
BRI dead in USA? |
4:02PM |
1 |
Voice Pulse And Incoming DID |
3:30PM |
1 |
hold then transfer... |
3:01PM |
0 |
Linux sparc64 conferencing? |
2:50PM |
2 |
SIP Registration issues |
1:58PM |
0 |
Phone numbers in SPAIN |
1:12PM |
1 |
RC1 and advanced voice mail options |
12:55PM |
1 |
Strange behaviour using 7960 |
12:01PM |
0 |
R: Dial plan errors |
11:04AM |
0 |
Error on Zaptel install |
11:02AM |
0 |
Call Queue: strategies and penalties |
10:54AM |
2 |
FREE (305) and (786) termination. Anyone interested? |
10:54AM |
10 |
Installing X100P |
10:31AM |
2 |
question regarding Asterisk. X-Lite, and firewall |
9:34AM |
3 |
how to configure my cisco 7960?! |
9:29AM |
1 |
Up to date? |
9:14AM |
1 |
DID VoIP trunk provider for metro Chicago, LA and/or Orlando. |
9:01AM |
1 |
quadBRI |
8:36AM |
10 |
PRI dead in USA? |
7:43AM |
0 |
received a call waiting CONNECT_IND |
7:26AM |
1 |
SIP 2 ISDN |
7:20AM |
1 |
Sound files - uncompressed versions available? |
6:58AM |
3 |
New CVS version |
6:41AM |
0 |
Asterisk CVS compile error YDL 3.0.1 |
6:39AM |
1 |
Random Dropped Called |
6:29AM |
0 |
NAT problems with ZIP 4x4 |
6:26AM |
4 |
Wireless SIP Phones |
6:24AM |
0 |
Modem chipset Intel |
5:23AM |
1 |
what is : |
3:47AM |
0 |
Grandstream transfer button |
3:44AM |
2 |
Calls from H323 to SIP unsuccessful |
1:39AM |
1 |
chan_vpb |
|
Monday July 19 2004 |
Time | Replies | Subject |
8:39PM |
1 |
isdn cli |
7:36PM |
2 |
codec translate |
6:45PM |
2 |
callparking vs calltransfer |
6:26PM |
0 |
Setup for Go2call ? Or any SIP provider using phonejack or linejack g729 g723 |
5:53PM |
6 |
Problem Starting RC1 |
5:33PM |
0 |
Hospitality Industry |
5:12PM |
11 |
Echo on a PRI |
4:34PM |
0 |
MWI - Config Stupidity or Notify Issues? |
4:20PM |
1 |
Unable to launch asterisk and connect to console. ????? |
3:19PM |
0 |
(Asterisk-Users] Affordable SIP Phone - Stiil a Myth? |
2:33PM |
0 |
Asterisk RC1 and advanced voice mail. |
2:22PM |
0 |
Cant compile Zaptel at all |
2:09PM |
1 |
Occationally SIP ext apparently is busy and goes to VM |
2:04PM |
0 |
dropping g729 frames |
2:03PM |
2 |
Affordable SIP Phone - Stiil a Myth? |
12:33PM |
6 |
collect calls |
11:24AM |
3 |
PSTN gateway implementation? |
11:00AM |
0 |
CTR21/CTR37 Gigaset phones and GS HT286 |
10:43AM |
6 |
Codecs - Advantages |
10:17AM |
2 |
Mac OS X installer: missing files fix |
10:09AM |
1 |
Flash Zap trunk from a Sipura |
10:02AM |
1 |
MAC OS X Panther :? |
9:33AM |
0 |
POE Switches and QOS |
9:29AM |
5 |
Cisco 7960 SIP V6 and distinctive ring. |
8:26AM |
1 |
Channel banks, voicemail, and immediate=no |
8:19AM |
2 |
BroadVoice problems? |
7:40AM |
5 |
Cheap PoE switches/injectors? |
7:25AM |
4 |
STILL NO AUDIO |
7:01AM |
4 |
FATAL: Module zaptel not found. |
6:57AM |
1 |
uip200 clips audio prompts |
6:52AM |
0 |
AGI Dial, Extension dial SIP Loop |
6:31AM |
1 |
Help w/ SIP response 481 |
6:07AM |
2 |
Unavailable/Withheld identification |
5:28AM |
3 |
Numbering Plan and Siemens EWSD |
5:13AM |
4 |
TDM400P Internal Extenion Config |
5:08AM |
0 |
*** Asterisk Sun/Monday News: Time to download, Scotty! |
1:53AM |
0 |
ast_data compile problem in asterisk CVS Asterisk CVS-HEAD-07/14/04 |
12:46AM |
0 |
sip-h323 |
|
Sunday July 18 2004 |
Time | Replies | Subject |
11:30PM |
0 |
Loud echo with answer before dial |
10:24PM |
0 |
GR-303 and _FXS_ support! |
9:51PM |
0 |
Asterisk Control Script |
9:40PM |
0 |
GUI based.. or ?? |
7:58PM |
3 |
Adding voice mail box |
7:41PM |
1 |
TE405P |
7:09PM |
3 |
LAN Switch w/ QoS |
6:38PM |
2 |
call progress detection |
6:13PM |
1 |
CID, international style? |
5:07PM |
0 |
ChanIsAvail issue |
4:53PM |
18 |
Polycom IP 500 Voicemail |
4:33PM |
0 |
chan_capi-0.3.4a |
4:33PM |
0 |
Help. New SIP hardphone. |
4:08PM |
1 |
chan_capi won't compile |
2:52PM |
4 |
Brain-dead Grandstream BT102? |
12:57PM |
1 |
Help! Unable to create channel of type SIP. |
10:50AM |
4 |
New G.729 codec and VLANS |
8:16AM |
4 |
quadbri NT_mode S-Bus Problem |
8:12AM |
1 |
PhoneGaim? |
8:12AM |
6 |
PSTN Gateway X101P |
7:45AM |
0 |
Asterisk and zaptel on Fedora Core 2 |
6:36AM |
2 |
Hotline |
5:59AM |
3 |
zaptel issues |
5:13AM |
1 |
sent into invalid extension 's' |
5:13AM |
4 |
Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk |
4:48AM |
1 |
Asterisk NAT spa-2000 |
12:12AM |
0 |
Polycom IP 500 Phones - Button Assignment |
|
Saturday July 17 2004 |
Time | Replies | Subject |
9:57PM |
0 |
sip-oh323 |
8:06PM |
1 |
Using a group variable for a groupofextension to dial |
7:05PM |
1 |
Using a group variable for a group ofextension to dial |
5:30PM |
2 |
Parking renamed to feature in 7/17/04 CVS |
4:24PM |
1 |
Using a group variable for a group of extension to dial |
4:18PM |
1 |
overlapping extensions |
2:27PM |
1 |
Asterisk at OSCON? |
1:21PM |
1 |
voicemail broadcast feature |
11:09AM |
6 |
Mac OS X installer for Asterisk |
9:46AM |
1 |
Question about Asterisk Installation |
9:31AM |
0 |
Re: asterisk echo problem ever go away??? |
7:12AM |
0 |
Updated RPMS for Asterisk-1.0 RC1 |
3:51AM |
1 |
MYSQL_FRIENDS and IAX problem |
1:35AM |
1 |
Wo uses H323-phones with asterisk? |
1:32AM |
0 |
RC1 Mirror, was Re: Asterisk-1.0 RC1 |
1:27AM |
3 |
chan_capi: sending incoming calls to different contexts |
1:16AM |
4 |
E100P and Colt Telecom (Europe) |
|
Friday July 16 2004 |
Time | Replies | Subject |
11:17PM |
8 |
Asterisk-1.0 RC1 |
8:23PM |
1 |
Pressing digits on SNOM phone results in letters on display |
7:36PM |
0 |
I already have a VAD frame? |
7:31PM |
0 |
Transmitting a hook-flash down an E&M DS-0? |
6:43PM |
1 |
Need configuration sample for VoIP(SIP) -> PSTN Gateway |
4:41PM |
7 |
7960 Dynamic DNS? |
3:11PM |
3 |
PSTN/phone/FXO/FXS cabling issue |
3:05PM |
0 |
Sipura 3000 user guide is now available |
2:28PM |
1 |
SIP register and unregister events via Manager API |
2:13PM |
1 |
Looking for WiFi phone recommendations |
1:34PM |
6 |
Asterisk + NEC Electra Elite IPK Integration |
12:16PM |
0 |
outgoing calls over SIP |
11:54AM |
0 |
Patch to test: Mailbox path changes |
11:42AM |
1 |
Problems with festival |
11:31AM |
0 |
Cisco Call Manager and Asterisk (AVVID) - Comparison |
11:02AM |
0 |
zaptel red alarms with e&m wink |
10:54AM |
1 |
Patch to test: Never say goodbye to an agent :-) |
10:35AM |
0 |
SIP module with radius authentication support |
10:28AM |
1 |
SIP channels UNKWN |
9:58AM |
0 |
Hardware platform / features |
9:46AM |
1 |
sendmail.cf and relaymail to a smtp server |
9:43AM |
3 |
Echo problem update - POSSIBLE SOLUTION |
9:14AM |
0 |
How to configure Asterisk as a VoIP(SIP) to PSTN Gateway? |
9:12AM |
0 |
Path to test: Sending HTML virus, no, VOICEMAIL! |
9:10AM |
0 |
Path to test: Czech localization |
9:08AM |
0 |
Patch to test: Dynamic queues |
8:10AM |
1 |
3COM 3102 SIP Phone |
8:09AM |
0 |
How to handle a macro that dials both international (011) and national |
8:03AM |
1 |
DTMF issue --help |
7:48AM |
2 |
where to sign up for fwd |
7:36AM |
2 |
Offhook tone in channel OSS/dsp |
7:28AM |
0 |
RESOLVED: 'Dropping voice to exceptionally long queue on IAX2' |
7:18AM |
1 |
Feature Group D |
6:57AM |
1 |
MWI on Grand Stream ATA-286 |
6:47AM |
0 |
Problem with asterisk and zaphfc |
6:42AM |
1 |
Astersik with g729 and 120 active channels with digium card ISDN PRI |
6:02AM |
1 |
VoiceMail fails to delete messages after emailing them |
5:56AM |
0 |
Asterisk sales materials |
5:45AM |
0 |
Subject: Re: SoxMix - Fails to Execute |
5:40AM |
1 |
Using Asterisk with fiber optic |
4:54AM |
1 |
iaxy server issue |
4:09AM |
1 |
When does the PUC become an issue? |
3:30AM |
1 |
Anyone experience with early dial? |
2:19AM |
1 |
Line Display |
2:02AM |
7 |
some questions on uniden uip200 |
12:03AM |
2 |
Flag Bad PRI Channel |
|
Thursday July 15 2004 |
Time | Replies | Subject |
11:59PM |
2 |
sip phone configuration problem |
11:36PM |
0 |
fax still fails, ideas sought! Re: rxfax/spa ndsp fails to decode |
11:23PM |
0 |
app_rpt |
11:00PM |
0 |
What happened to opencall.org ? |
7:24PM |
0 |
Unable to create chanel of type SIP |
7:14PM |
3 |
G.729 codec doesn't seem to work *even* after installing the license |
5:21PM |
0 |
accountcode problem |
5:20PM |
0 |
sphinx2 how-to |
5:17PM |
1 |
"Reverse Hold" feature prototype... |
4:41PM |
1 |
Spectrum Analyizer software |
4:17PM |
3 |
SIP to H323 call timeout |
4:13PM |
3 |
Current echo status? |
3:56PM |
1 |
DID AND EXTENSION DIALED NUMBER FORWARD |
3:36PM |
4 |
Kernel panic with two Fritz cards |
2:48PM |
1 |
Call Queues help |
2:35PM |
3 |
Important note for AGI with PHP newbies |
2:34PM |
1 |
Fedora Core 2 softphone |
2:15PM |
2 |
SoxMix - Fails to Execute |
2:14PM |
2 |
Cisco phones and Messages and Forward ToVM keys |
1:50PM |
1 |
Polycom IP 500 and Asterisk |
1:19PM |
1 |
bristuff 0.0.3 ? |
12:54PM |
4 |
freenode #asterisk IRC channel identd problem |
12:19PM |
0 |
Hangup FXO line detecting & PSTN Tone Signals Detecting |
12:05PM |
8 |
Directory |
11:48AM |
6 |
[OT] The stories people tell to support. |
11:21AM |
3 |
Database App |
11:13AM |
2 |
Really long first ring, then normal |
10:12AM |
0 |
SIP registry forwarding top SIP connections |
9:45AM |
0 |
astcc database configuration |
9:38AM |
1 |
Using SIP phone to dial out using ISDN ? |
8:27AM |
0 |
Grandstream Budge Tone 100 No Ringtone |
8:23AM |
4 |
ZyXEL 2000W |
7:23AM |
17 |
VoicePulse changes |
6:01AM |
1 |
random disconnect with hfc ISDN card and sipura |
5:21AM |
0 |
TE4XXP Signaling |
5:01AM |
0 |
DTMF and Voicemail issues |
2:53AM |
1 |
zapras - and kernel ?? |
1:27AM |
1 |
*, NAT & STUN |
1:03AM |
1 |
Problem loadin oh323 solved |
12:27AM |
0 |
Incoming SIP calls as asterisk@... |
12:11AM |
2 |
Small setup |
|
Wednesday July 14 2004 |
Time | Replies | Subject |
11:54PM |
0 |
Originate to IAXComm problem once again |
9:35PM |
1 |
looking for 802.11 SIP phone |
7:57PM |
1 |
why ata stop working after 10 mins after registering from mysql |
7:56PM |
0 |
Voice Numbers in Spain (SIP) |
6:10PM |
1 |
oh323 dial structure and oh323 debug? |
6:03PM |
0 |
changed ip now * demo call not working. |
5:19PM |
1 |
Noob Service Provider T1/T400p physical interfacing question |
4:58PM |
0 |
Voange with asterisk settings |
3:52PM |
2 |
different port setting |
3:32PM |
3 |
Vonage working with asterisk |
3:16PM |
0 |
Errors connecting to FWD |
2:56PM |
8 |
Directed Call Pickup |
1:20PM |
1 |
SMDR/CDR - Asterisk integration - Clarification |
1:16PM |
2 |
Chan_Capi 0.3.4a error |
1:02PM |
0 |
Status of ALERT_INFO and Cisco 7960? |
12:02PM |
0 |
CHAN_H323 bridge SIP no audio |
11:41AM |
13 |
Where can i get an UK SIP account with UK number? |
11:26AM |
10 |
CISCO 7960G FIRMWARE |
10:30AM |
1 |
Digium X100P card to a brazilian analog line |
10:17AM |
1 |
Starting up considerations..... |
10:15AM |
2 |
Increase transmit volume on IAX channels |
10:13AM |
0 |
random red alarms with t100p |
10:06AM |
0 |
Windows Messenger Problem |
9:54AM |
4 |
can you trust CDR for billing information? |
9:50AM |
0 |
Distinctive Ring availability on an IOS sip gateway |
9:43AM |
8 |
spa-3000 review? |
9:26AM |
2 |
SIP only |
9:21AM |
0 |
asterisk as a SUA together with SER |
9:09AM |
2 |
RE: [Asterisk-User] asterisk compile problem |
8:09AM |
0 |
who knows asterisk/libpri source code interaction |
7:51AM |
1 |
cvs.digium.com |
7:30AM |
1 |
zaphfc ptp & blocked incomming calls |
7:19AM |
5 |
Getting an USA phone number |
7:09AM |
1 |
error 1 and 2 during make of asterisk with SUSE 8.2 and 9.1 |
6:51AM |
3 |
Using a DNS name for externip in sip.conf |
6:43AM |
1 |
Onhold Music |
6:14AM |
2 |
GSM adapter + Automatic Routing function |
6:12AM |
0 |
Having serious problems with AGI |
5:25AM |
1 |
invalid extension -> missing the original ${EXTEN} value |
5:01AM |
5 |
ACD Issues |
4:53AM |
0 |
forward_msg: no 2nd via found in reply |
4:46AM |
0 |
ISDN PRI "calling number" for outgoing calls |
4:27AM |
1 |
Questing regardning dialplans on a Cisco 5350 |
4:08AM |
3 |
Voicemail/autoattendant not working |
12:25AM |
0 |
Audiotel - Premium, call proceeding ? |
|
Tuesday July 13 2004 |
Time | Replies | Subject |
10:24PM |
2 |
ASTCC: Asterisk Calling Card Solution |
8:47PM |
1 |
SIP authentication bug with insecure= lines? |
7:57PM |
0 |
Unable to place more then 1 call in or out. |
7:50PM |
1 |
0h323/ h323-registration |
7:29PM |
3 |
Bounty! For help with echo cancellation code. |
5:46PM |
1 |
asterisk compile problem |
3:31PM |
1 |
Re: Applications of TDMoE "critch" |
3:31PM |
1 |
Problem with multiple phones behind firewall |
2:35PM |
0 |
Meetmee feature - Possible. |
2:34PM |
0 |
Looking for US ISDN card... |
2:26PM |
1 |
bad sound quality, also the ringtone |
1:28PM |
0 |
RE: Problem with multiple phones behind firewall |
1:24PM |
0 |
Registration Refresh Per REGISTER line in sip.conf |
1:09PM |
0 |
will digium hardware and asterisk function in asia (korea)? |
1:08PM |
0 |
Upgraded to CVS HEAD 7/12/2004 and calls very bad now |
12:58PM |
2 |
Swiss IP10S using SIP |
12:46PM |
1 |
fax still fails, ideas sought! Re: rxfax/spa ndsp fails to decode |
12:36PM |
3 |
Bandwidth requirement with G729A |
12:26PM |
1 |
Mailing to the list |
12:22PM |
4 |
Rotary phones? (No, I'm serious) |
12:19PM |
2 |
How to 'Dial' a Parked Call ? |
12:08PM |
1 |
Asterisk ML archive down? |
11:55AM |
0 |
One way audio when the BT-100 is behind Firewall |
11:42AM |
1 |
integrating ser with asterisk |
11:11AM |
0 |
Any way to change ring back behavior for call park? |
10:50AM |
0 |
Echo, DTMF, issues |
10:36AM |
1 |
codec issues between linphone and * |
9:41AM |
0 |
WARNING: Deprecated incominglimit and outgoinglimit |
9:37AM |
0 |
Asterisk and Swissvoice |
9:19AM |
0 |
'Dropping voice to exceptionally long queue on IAX2' |
9:06AM |
5 |
WiSIP and Zyxel Prestige 2000W |
9:00AM |
1 |
Broken pipe in remote exeute |
8:49AM |
0 |
"unclean hangups" can I turn off hook flash? |
8:45AM |
0 |
zaphfc does not indicate congestion!? |
8:44AM |
1 |
G729A and GSM - newbie question |
8:14AM |
2 |
Help Needed in configuring Cisco 7940 |
7:46AM |
1 |
Dial Fail - Send Email |
7:46AM |
2 |
Local Call Problems |
7:26AM |
3 |
Cann't load oh323 0.6.3a |
7:15AM |
0 |
Possible Asterisk Notify Bug |
5:38AM |
2 |
IAX2 calls through IAXTEL.com |
5:33AM |
1 |
Asterisk don't listen to my phones |
5:24AM |
1 |
HFC-S card and Unable to create channel of type 'Zap' |
5:11AM |
1 |
Meridian Option 11c Asterisk Expert Needed |
5:07AM |
0 |
X100P ring/off-hook in strange state 6 |
5:03AM |
0 |
Local Calls Not Working |
4:30AM |
3 |
Applications of TDMoE |
4:26AM |
0 |
zaphfc TE -> NT problems |
4:19AM |
0 |
how to use direcotory from Voicemail |
4:07AM |
1 |
caller id problem on incominc call to x100p |
12:27AM |
1 |
segmentation fault on asterisk startup |
12:07AM |
2 |
SIP simultaneous registry possible workaround (was Re: New Asterisk bounty: SIP simultaneous registry) |
|
Monday July 12 2004 |
Time | Replies | Subject |
8:37PM |
0 |
No Compatible codecs? Got license |
8:21PM |
0 |
"Follow Me/FInd Me" functionality? |
7:50PM |
1 |
Asterisk as plain PABX in call centre |
6:57PM |
0 |
Using a SwissVoice IP10S with Asterisk |
5:57PM |
2 |
Oz ISDN |
5:53PM |
0 |
Announce of Cisco 7914 Operator Console Support in chan_sccp |
4:54PM |
1 |
Manager help |
3:53PM |
1 |
Problems with chan-capi |
3:13PM |
0 |
Running SIP on multiple ports |
2:53PM |
3 |
HELP: One way audio... continuously and randomly |
2:37PM |
1 |
incoming calls on Cisco 7960 |
2:21PM |
3 |
notransfer |
2:02PM |
1 |
No voice bet/ ext with Polycom |
1:54PM |
1 |
Errors when compiling app_radius |
1:53PM |
1 |
FWD, DISA & DTMF |
1:47PM |
1 |
CID not appearing via X100P |
1:31PM |
1 |
Problems Compiling asterisk-oh323 0.6.3a |
12:59PM |
3 |
voicemail setup guide? |
12:45PM |
3 |
Audio filters (was: feature - VM gain adjust?) |
12:43PM |
4 |
call Intrude |
12:39PM |
0 |
"help" |
12:26PM |
0 |
GnuGK + SIP Provider + Asterisk |
12:17PM |
1 |
rxfax/spandsp fails to decode |
12:06PM |
5 |
Digium Cards in Boxes without Power Connectors |
12:05PM |
6 |
Asterisk crashing with no indication why. |
11:53AM |
2 |
OH323 and G729 |
11:42AM |
1 |
SIP client to IAXTel 800/888/877/866 dialing thru Asterisk |
10:54AM |
0 |
Transfers (sip or asterisk "#' base) broken in certain scenario |
10:33AM |
0 |
IP Soft Phone with FAX |
9:58AM |
1 |
asterisk T1 question |
9:50AM |
1 |
Cheap ISDN interface + Asterisk what to choose? |
9:39AM |
1 |
Sort of OT: Recommended USB handset for use with iaxComm? |
9:09AM |
1 |
zaptel debugging tools |
9:00AM |
3 |
dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames |
8:33AM |
0 |
SIP => PSTN Pri Causes |
8:28AM |
0 |
Problem with Capi Channel |
8:15AM |
0 |
IAXy prov. using DNS |
8:03AM |
2 |
Indications missing on Cisco FXO -> ATA-186 (SIP) |
8:01AM |
0 |
GnuGK + Asterisk + SIP Provider |
7:51AM |
8 |
Gogoif with variables acting funny? |
7:41AM |
0 |
DTMF warning message in log while using SJPhone |
7:28AM |
0 |
Cisco Remote-Party-ID / Bug #2012 |
7:19AM |
1 |
ZapBarge and SIP Channels |
7:02AM |
0 |
Using MD5 to encrpty PIN |
6:24AM |
1 |
R: How to make * don't strip the leading 0 |
5:57AM |
3 |
How to make * don't strip the leading 0 |
5:36AM |
1 |
PRI numbering plan |
4:36AM |
1 |
E100P and T1 channel banks |
4:19AM |
1 |
Can I hear voice messages from diax phone button directly ? |
4:18AM |
3 |
permission problem |
1:54AM |
0 |
Problem with character encoding in SIP channel (ISO vs. UTF-8) |
12:57AM |
0 |
RE: MeetMe Improvement |
12:20AM |
0 |
RE: How to differentiate a *busy* call from not available? |
|
Sunday July 11 2004 |
Time | Replies | Subject |
9:08PM |
0 |
wcusb dialing problem and line noise |
7:09PM |
1 |
Stopping reinvite with IAX2? |
6:45PM |
1 |
mediatrix 1204 hysteria |
6:00PM |
1 |
Please ignore my last message... |
5:38PM |
1 |
Echo issues (again...) |
4:00PM |
6 |
feature - VM gain adjust? |
11:21AM |
3 |
QoS in asterisk |
10:52AM |
0 |
DIALSTATUS variable and oh323 channel |
10:38AM |
4 |
Asterisk on FreeBSD 4.10 dies |
9:16AM |
0 |
Hardware for sale / donate |
9:11AM |
0 |
VoiceMail + Forwarding + Directory Dial by Name : How? |
8:07AM |
0 |
iax2 - peer 2 peer - asterisk? |
3:14AM |
0 |
Wh uses H.323-clients with call transfer? |
1:15AM |
20 |
New Asterisk bounty: SIP simultaneous |
|
Saturday July 10 2004 |
Time | Replies | Subject |
8:35PM |
0 |
VoIP provider for 2 site enterprise deployment?? |
7:44PM |
0 |
Two server |
7:10PM |
2 |
Looking for a patch that was post May 1 2004 |
4:58PM |
1 |
NuFone Error |
3:59PM |
2 |
New Asterisk bounty: SIP simultaneous registry |
2:38PM |
0 |
Does the SPA-3000 get rid of echo that the X100P can't? |
12:05PM |
0 |
How to use freeradius for Asterisk billing |
10:04AM |
1 |
tdm 400p noise |
9:33AM |
5 |
Three (quick?) questions... |
8:11AM |
1 |
Asterisk + g.726 |
7:40AM |
3 |
X101P FXO with RED alarm |
7:29AM |
2 |
German Asterisk Site |
6:53AM |
0 |
(no subject) |
6:46AM |
0 |
bad clicking sounds with Diva+capi+asterisk |
4:17AM |
0 |
Asterisk Support for ISDN National-3? |
|
Friday July 9 2004 |
Time | Replies | Subject |
1:50PM |
0 |
Did I get booted? |
12:52PM |
0 |
Asterisk and CDRTool |
12:51PM |
9 |
using asterisk voicemail with a class 5 softswitch |
11:56AM |
3 |
SMDR/CDR - Asterisk integration |
11:55AM |
1 |
RE: ATA 186, firmware SIP 3.1 and codec g.726 + now SIPURA SPA-2000 |
11:54AM |
1 |
No data when recording a Meetme conference with Monitor |
11:31AM |
1 |
x100p and groundstart signaling - help |
11:28AM |
7 |
Predictive Dialers |
11:21AM |
1 |
QoS - Routers |
11:14AM |
1 |
IVR Menu and VoiceMail quality |
10:59AM |
3 |
E1 config help and guidance |
9:50AM |
1 |
zaphfc - TE mode - callerid trouble |
9:31AM |
3 |
ATA 186, firmware SIP 3.1 and codec g.726 |
9:26AM |
1 |
Fwd: Problem of loading the oh-323 module |
8:36AM |
3 |
Debian Unstable Claims Asterisk 1.0-1 |
8:02AM |
0 |
GSM to iLBC one way audio :-( |
7:41AM |
1 |
Re: SNMP Monitoring (Andrea Fino) |
7:28AM |
2 |
vonage.ca * integration possible? |
7:09AM |
2 |
T1 Hardware Echo Can |
6:27AM |
1 |
Help needed regarding Grandstream phone |
6:20AM |
4 |
Cisco MC3810 -> Asterisk |
6:18AM |
0 |
chan_mISDN test release.... |
5:46AM |
2 |
SIP Regiter config question |
5:27AM |
4 |
strange echo problem |
5:01AM |
4 |
Dell 6450 / TE405p |
4:42AM |
0 |
Overlapdial on PRI |
3:54AM |
7 |
IRC channel #asterisk on irc.freenode.net |
3:17AM |
2 |
Problems with cdr_csv |
3:03AM |
1 |
Asterisk and Audiocodes MP124 |
1:59AM |
1 |
sound quality IAX client GSM to ALAW with oh323 |
1:06AM |
1 |
bristuff - hfc card + x100p |
|
Thursday July 8 2004 |
Time | Replies | Subject |
9:50PM |
1 |
ok |
8:52PM |
1 |
Two outbound calls at once |
5:32PM |
1 |
Re: Asterisk-Users digest, Vol 1 #4460 - 14 msgs |
5:10PM |
1 |
Asterisk receives TMC Labs Internet Telephony Innovation Award |
4:42PM |
3 |
asterisk to asterisk config |
3:35PM |
0 |
GS & DTMF in voicemail with CVS of today!? |
3:00PM |
1 |
Intermittent SIP 404 Not Found response? |
2:45PM |
6 |
Updated Grandstream configurator |
2:22PM |
2 |
SNMP Monitoring |
1:39PM |
1 |
displaying call progress with SendText on a Snom |
12:59PM |
2 |
internal & external SIP |
12:21PM |
2 |
HOW ASTERISK WORKS |
12:13PM |
1 |
advanced audio recording agi help |
11:47AM |
0 |
IAX2 problems transfering back and forth between pbxes |
11:47AM |
0 |
outgoing caller id from SIP to isdn (p2p) |
11:09AM |
5 |
Using Cisco AS5350 as pstn GW .. one-way audio problem |
11:02AM |
0 |
rxfax - mISDN - missing logs |
9:49AM |
1 |
Re: tdm400p static - out of ideas (Jorge Mendoza) |
9:48AM |
3 |
Interface to generate Statements? |
9:19AM |
1 |
Access Bank 2 <---> T100P T1 Cable. |
9:06AM |
0 |
Turning off RFC 3265 in Asterisk |
8:26AM |
1 |
Using Windows Messenger+Video in * |
8:09AM |
3 |
i or s or whatever the invalid_exten is HELP !!!!! |
7:48AM |
0 |
WellTech Wellgate 5250 E1 trunk gateway |
7:22AM |
3 |
Audiocodes -> Asterisk Implementation |
7:05AM |
8 |
FINALLY! a good book about Asterisk. |
7:04AM |
2 |
Shady dial anyone?? |
7:01AM |
2 |
Slackware 10.0 and asterisk and 2.4 vs 2.6 |
7:00AM |
0 |
Problem SIP no audio just noise |
6:42AM |
1 |
Sip Peer Status |
6:31AM |
4 |
sample config file for GS BT101? |
6:07AM |
0 |
Meetme and IAX |
5:21AM |
2 |
Question about Cisco IP Phone 7960 |
5:02AM |
0 |
Minimum install required for Asterisk + voicemail & SIP friends from mysql |
4:00AM |
2 |
Cisco 7960 NAT question |
2:55AM |
1 |
Rollover oddity |
2:52AM |
1 |
x100p and two hfc isdn cards |
2:50AM |
0 |
Call placed towards a trange called number 'h' |
12:47AM |
3 |
ISDN, AVM C4, HFC-cards and echo |
12:39AM |
2 |
pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' WARNING |
12:35AM |
0 |
R: VoIP hackers gut Caller ID |
|
Wednesday July 7 2004 |
Time | Replies | Subject |
10:22PM |
0 |
X100P bad sound after period of time |
10:20PM |
0 |
Intermittent cidname lookups |
7:49PM |
4 |
Small Linux Distro |
7:28PM |
1 |
patlooptest output |
6:51PM |
1 |
RE: What is the difference between queeu members and queue agents |
6:44PM |
0 |
NEWS from the chan_sccp developers. |
6:39PM |
1 |
OH323-COMPILE |
5:45PM |
5 |
E100P |
4:32PM |
3 |
Mandrake 10, Request for comments. |
4:14PM |
1 |
Problem when using asterisk + gnugk |
3:52PM |
0 |
rxgain and txgain not effective |
3:06PM |
2 |
Parking call problem |
2:38PM |
0 |
Perl libaray for manipulating .conf file |
1:43PM |
1 |
Cisco, Sip, Linux, ISDN |
1:40PM |
2 |
Perl library to manipulate 'ini files' |
1:38PM |
0 |
GR-303 configuration options? |
1:23PM |
0 |
solved - Audio cuts off 10 minutes into calls |
12:56PM |
4 |
VoicePulse Connect DID Problems |
11:44AM |
1 |
Ringinbacktone even without 'r', and inexistant codec |
11:42AM |
0 |
Audio cuts off 10 minutes into calls |
11:25AM |
1 |
res_odbc not working |
11:16AM |
0 |
Conf files doubt |
10:46AM |
0 |
:: Astricon :: Registration now open! |
10:42AM |
0 |
Modem support via Cisco FXS ports? |
10:23AM |
1 |
Software SIP fax client |
10:23AM |
1 |
RE: What is the difference between queeu members and queue agents |
8:31AM |
2 |
zaphfc and ASUSCOM working in the US |
7:13AM |
4 |
tdm400p static - out of ideas |
7:09AM |
1 |
UDP Ports scan on firewall |
7:03AM |
0 |
kdeconsole and gtkconsole |
6:52AM |
0 |
IP Dialog Hangup problem |
6:50AM |
2 |
Problem SIP Register |
6:43AM |
2 |
Asterisk Article |
6:34AM |
7 |
New PBX Help |
6:34AM |
0 |
Asterisk not populating nonce count |
6:26AM |
1 |
Sangoma cards |
6:11AM |
1 |
X100P donĀ“t answer sometimes |
5:40AM |
0 |
Simple H323 Test |
5:37AM |
0 |
Language |
5:29AM |
8 |
VoIP hackers gut Caller ID |
5:24AM |
0 |
sound quility |
3:41AM |
4 |
HFC- Colongne TE Mode |
2:37AM |
1 |
Call files timeout on Flash command |
1:35AM |
3 |
Problems installing asterisk. |
1:19AM |
8 |
Voicemail volume |
1:14AM |
1 |
recording an on-going call |
1:00AM |
1 |
CDR records into SQLite |
12:32AM |
0 |
app_flash Flash command - flash lasting too long |
12:26AM |
0 |
FYI: David Isenberg on IAX and SIP |
|
Tuesday July 6 2004 |
Time | Replies | Subject |
11:43PM |
2 |
AGI - No audio |
9:45PM |
4 |
Newbie's doubt on sip.conf |
9:22PM |
1 |
compiling mysql addon |
8:47PM |
3 |
odd behavior - adtran ta 850 + t100p |
8:22PM |
3 |
Cisco 7960 and Voice Mail |
7:22PM |
1 |
Identify incoming 800 number |
7:14PM |
0 |
CDR and EXTEN |
6:52PM |
1 |
g729 codec compatibility voiceage vs Digium |
5:55PM |
1 |
asterisk grandstream aleatory error |
5:52PM |
2 |
Kerry/Edwards campaign and VOIP |
5:34PM |
2 |
TDM FXO port remains offhook |
3:52PM |
1 |
Hangup's not detected correctly |
3:13PM |
0 |
Channel bank or IAD with message light capability? |
2:09PM |
1 |
Problem related to TPX100 card installation |
1:18PM |
3 |
multiple days on a GotoIfTime command? |
12:56PM |
2 |
GR303 |
12:44PM |
1 |
zaptel DTMF delay |
12:31PM |
0 |
Sound card troubles with asterisk resulting in no sound |
12:19PM |
3 |
Zap Channel error using 4-port FXO TDM400P |
12:17PM |
1 |
FYI House bill exports analog phone regs to VoIP |
12:01PM |
4 |
Odd Zap dialing problem |
11:37AM |
2 |
Mediatrix 1102 Problems |
11:34AM |
1 |
SIP and H323 |
10:25AM |
1 |
rh9, asterisk HEAD, & asterisk-oh323-0.6.3a working |
10:12AM |
0 |
Numbering range |
10:02AM |
1 |
quantumvoice |
9:43AM |
2 |
ztdummy running, but moh & meetme don't work |
8:53AM |
2 |
Uniden consult transfer |
8:42AM |
3 |
New CVS for patch... |
7:48AM |
1 |
G.723.1 and Asterisk |
6:47AM |
3 |
SPA-2000 and time of day |
6:27AM |
3 |
Dialing out of a voicemail message? |
6:13AM |
1 |
zaphfc 2 cards working with P2P Mode ?? - massive Problems |
5:48AM |
1 |
* and Innovaphone |
5:39AM |
0 |
Music on hold error since CVS update |
5:00AM |
1 |
missing .gsm in VoiceMailMain(2) |
4:37AM |
0 |
Asterisk config on PostgreSQL |
3:36AM |
0 |
AW: AW: Junghans Quad-BRI card and asterisk cvs-head |
3:23AM |
3 |
H323 channel |
2:35AM |
1 |
AW: H323 Call Transfers |
2:29AM |
3 |
AW: Junghans Quad-BRI card and asterisk cvs-head |
2:13AM |
0 |
ZyXEL P2000W - working conf example |
2:09AM |
1 |
H323 Call Transfers |
1:58AM |
1 |
Junghans Quad-BRI card and asterisk cvs-head |
1:38AM |
2 |
How do I disable '#' to transfer a call? |
1:11AM |
6 |
is srv lookup being done when REGISTERing? |
12:39AM |
0 |
isdn to sip callerID pass |
12:33AM |
0 |
How to connect to cellular phone beside analog interface card? |
12:27AM |
1 |
2x analog interface (1 ISDN and 1 door phone) recomendation for Europe ? |
|
Monday July 5 2004 |
Time | Replies | Subject |
11:28PM |
0 |
Any experience with Citel Link 3300 and Asterisk |
10:02PM |
2 |
What happened to the CVS asterisk_stable branch? |
9:29PM |
3 |
asterisk, fwd, and grandstream? |
7:51PM |
7 |
Calling an outside phone number as part of a hunt |
7:30PM |
1 |
Clean compilation |
5:50PM |
0 |
Playback/Background over Console/dsp |
5:48PM |
2 |
Playback over Console |
4:05PM |
3 |
Randy Bush is a destructive force with a hidden professional agenda |
4:01PM |
1 |
Voice channel could not be established. |
2:46PM |
2 |
Wake Up Call AP |
1:14PM |
0 |
outgoing Sip-call problem URI and Phone-number |
1:11PM |
9 |
iax or sip |
11:29AM |
2 |
No RED/GREEN alerts on TDM400P? |
11:12AM |
1 |
Cut off after 8 secs?? Help |
10:58AM |
2 |
T1 configuration, getting help via IRC? |
10:11AM |
1 |
FireFly client and echo problems with IAX |
9:56AM |
4 |
IAX Call Pickup |
9:27AM |
2 |
fax detection and X100P |
9:19AM |
0 |
Voicemail plays back at very low volume - how to make it louder? |
9:11AM |
0 |
SMS on TE410P |
8:59AM |
0 |
Rederecting an incoming CAPI call to SIP soft phones |
8:11AM |
3 |
dialing # on a crisco (was: Divert to arbitrary number) |
8:04AM |
0 |
chan_misdn HFC-NT dialtone |
7:25AM |
4 |
Question about x100P and zap |
7:19AM |
2 |
Again Sip Registration Fail |
6:23AM |
0 |
[Asterisk] 2 T100p and and Panasonic |
6:00AM |
1 |
*8# into invalid extensions |
5:30AM |
2 |
Problem with BRI_STUF / direct connected ISDN-Phone |
5:12AM |
3 |
*** Asterisk Sunday (hrrm) News: Moving ahead at CVS Warp 5 |
4:09AM |
1 |
Divert to arbitrary number. |
12:34AM |
0 |
Penalty in queues.conf |
|
Sunday July 4 2004 |
Time | Replies | Subject |
8:50PM |
1 |
PTHREAD_MUTEX_RECURSIVE in appradius-1.0 |
5:23PM |
0 |
LCS multiparty conferencing commercial opportunity |
4:13PM |
1 |
cdr and edit dst field |
12:12PM |
2 |
music on hold question with asterisk |
11:53AM |
3 |
looking for newbie resources |
11:15AM |
4 |
Asterisk Book |
10:48AM |
1 |
conf from pgsql database |
10:03AM |
2 |
I wanna kill FWD.... GRRR!!! |
7:06AM |
0 |
FWD/SIP audio suddenly stopped working |
5:03AM |
1 |
How to use return value in extensions.conf |
1:31AM |
1 |
Using call redirection numbers |
|
Saturday July 3 2004 |
Time | Replies | Subject |
7:47PM |
0 |
PRI unknown signalling on TE405P |
5:56PM |
1 |
Caller ID and DNIS Problems (Non-Pri T1) |
5:09PM |
1 |
Size of asterisk internal database |
11:22AM |
11 |
Music on hold problem |
10:44AM |
2 |
Multiple E1s over TDMoE? |
9:48AM |
2 |
saydigits/background |
4:42AM |
0 |
Support for Snom 200 Extension Monitoring |
|
Friday July 2 2004 |
Time | Replies | Subject |
10:31PM |
0 |
TDM400P GroundStart Problems |
4:30PM |
0 |
DISA and AGI: authenticate by caller ID? (resolved) |
2:48PM |
2 |
Zaptel dacs / dacs |
2:48PM |
2 |
H323 -> IAX |
1:43PM |
3 |
Termination for Asterisk Users - Inter-Asterisk Exchange |
1:37PM |
1 |
IAX to IAX call with really bad echo |
1:19PM |
0 |
TA750 + T100P configuration. flashing red alarm on t100p after running modprobe wct1xxp, ztcfg |
12:49PM |
3 |
Inter-Asterisk Exchange |
12:18PM |
1 |
Compiling Gastman for Win32 |
10:54AM |
0 |
do_monitor: Bad file descriptor |
10:26AM |
0 |
Channel Bank Newbie Problem |
9:43AM |
3 |
IRQ Misses and Dropped Calls? |
9:28AM |
1 |
mysql voicemail |
9:19AM |
1 |
Params on SIP URI REGISTER/INVITE |
8:50AM |
3 |
Suggestions for 96 tip/ring lines? |
8:36AM |
1 |
Problem with CHAN_SCCP |
8:22AM |
1 |
RTP Source IP Address |
8:17AM |
3 |
CDR shows billsec=12 for all bridged calles. |
8:07AM |
0 |
Cisco 7960G and * |
6:36AM |
0 |
ZAPTEL FXO debuging - Tones, Voltages, Ampers, etc. |
5:18AM |
2 |
Zaptel, Line Impedence and Echo |
4:47AM |
0 |
Have problem install via cvs |
4:42AM |
1 |
TE410P PINS |
4:27AM |
0 |
ip10: config setting "PackageNotify"? |
3:21AM |
0 |
Status of Australian approval for E100P...??? |
3:15AM |
0 |
CBMySQL |
3:14AM |
2 |
Monitoring Asterisk |
1:11AM |
0 |
Problem locating stream files |
12:33AM |
1 |
Grandstream 1.0.5.30 available |
12:31AM |
4 |
Delay when dialing with Sipura 2000 |
|
Thursday July 1 2004 |
Time | Replies | Subject |
11:07PM |
2 |
Grandstream HT286 1.0.4.63 & Meetme |
8:43PM |
5 |
Inter-Tel Eclipse2 (IP PhonePlus) |
6:52PM |
1 |
commercial implementation |
5:23PM |
0 |
Invalid context |
4:36PM |
1 |
Hangup on transfer... |
4:36PM |
0 |
Re: Hello |
4:06PM |
1 |
two sip clients on one server |
3:12PM |
1 |
Cannot install module Bri-Stuff-0.0.2 zaphfc.ko does not exist. |
3:02PM |
9 |
Config Files |
1:14PM |
1 |
SPA-2000, call for help testing echo issues... |
1:00PM |
5 |
Sip to Sip |
12:56PM |
0 |
Updated version of Grandstream cfg file generator |
12:34PM |
2 |
IAX2 to IAX2 connection problems |
11:55AM |
0 |
Weird LAN VoIP Echo |
11:03AM |
4 |
Pager Notification |
10:30AM |
0 |
zaptel wont compile errors on zttest |
10:08AM |
2 |
DISA and AGI: authenticate by caller ID? |
9:13AM |
0 |
Sound: Record Overrun |
9:00AM |
1 |
How to track (log file) Dial Plan events to fix unsteadily states like opened FXO port |
8:51AM |
5 |
Zultys 4x4 or 4x5 ip phones? |
8:26AM |
0 |
sccp to sip call signalling |
8:23AM |
0 |
simple AGI script |
8:11AM |
0 |
Strange behavioir on a exten |
7:36AM |
5 |
voicemail notification? |
7:07AM |
4 |
1800 number with colo |
7:02AM |
0 |
R: Asterisk Docs |
6:58AM |
3 |
R: execute a context from cron |
6:46AM |
5 |
execute a context from cron |
6:42AM |
1 |
Asterisk Docs |
6:24AM |
2 |
Registration failed for SIP |
3:28AM |
1 |
Help with Welltech 2FXO gateway, GS BT100 and Asterisk |
2:58AM |
0 |
Simple gateway SIP <--> ISDN |
1:57AM |
2 |
Providing Telewest in the UK with per extens ion outbound callerID |
12:08AM |
0 |
2 T100P and a Digital PBX |