| Sunday December 31 2006 |
| Time | Replies | Subject |
|---|
| 10:08PM |
1 |
Dual Ringing Tones |
| 3:38PM |
1 |
PRI ANI/CallerID |
| 3:07PM |
0 |
Digest of lists on forum asterisk.voicemeup.com |
| 3:00PM |
0 |
PHP C Extension to connect to manager interface |
| 12:12PM |
0 |
IAX timeout if no ringing |
| 10:46AM |
8 |
(OT) Where to post free source for AGI? |
| 8:07AM |
0 |
exec after recording agents |
| 7:59AM |
1 |
Sangoma A102d and Asterisk on Debian 3.1. |
| 6:21AM |
0 |
IAX & WaitExten |
| 12:05AM |
1 |
X100P "rings" randomly when "phone" line makes call |
| |
| Saturday December 30 2006 |
| Time | Replies | Subject |
|---|
| 7:34PM |
0 |
Server to Server connectivity on SIP |
| 6:21PM |
0 |
Modem Dial-up internet connection thru Asterisk with T1-PRI |
| 5:22PM |
0 |
Theory behind RDNIS and does it work or not? |
| 4:29PM |
0 |
Best WIFI Phone- Zulty WIP 2 Link |
| 1:20PM |
2 |
Happy 2007!!! |
| 12:58PM |
4 |
WIFI SIP- The Best phone |
| 5:43AM |
1 |
Odd hangup problem TDM400P |
| |
| Friday December 29 2006 |
| Time | Replies | Subject |
|---|
| 10:19PM |
1 |
E1 controller |
| 3:21PM |
2 |
Re: Hi reg. 2 asterisk server |
| 3:07PM |
1 |
Need advice on dual core processing with * |
| 2:53PM |
2 |
Avaya to Asterisk via H323 |
| 2:26PM |
0 |
Dial - g option |
| 12:22PM |
1 |
trixbox web-administration |
| 11:10AM |
0 |
Toll free numbers |
| 10:04AM |
0 |
How does Sipura route incoming calls? |
| 9:19AM |
2 |
chan_sip loading delay in Asterisk 1.2.10 |
| 9:02AM |
2 |
Binary AGI Scripts |
| 8:45AM |
0 |
PHP to call script |
| 8:34AM |
0 |
problem with VoiceMailMain |
| 7:45AM |
7 |
Asterisk and MiniITX setups |
| 7:38AM |
2 |
Disconnect supervision in India? |
| 7:29AM |
2 |
Realtime multiple registration for a Hard Phone Snom 360 |
| 6:58AM |
1 |
Presence issues with "Got SUBSCRIBE for extensions without hint. Please add hint to s" |
| 4:03AM |
2 |
Dialed Number missing from the CDR when using call files. |
| 1:24AM |
1 |
asterisk doesn't know version of asterisk-addons? |
| |
| Thursday December 28 2006 |
| Time | Replies | Subject |
|---|
| 11:49PM |
1 |
voicemail and ip phones |
| 8:14PM |
0 |
Re: asterisk-users Digest, Vol 29, Issue 114 |
| 7:29PM |
1 |
TE110P with Qsig |
| 6:07PM |
2 |
Error compiling chan_vpb |
| 5:24PM |
0 |
Compiling Zaptel 1.4.0 on SuSE 10.0 |
| 4:27PM |
6 |
1.4 Random disconnects |
| 3:43PM |
1 |
mIDN question |
| 2:30PM |
1 |
one way rtp stream (Sent alwax to 127.0.0.1) |
| 2:19PM |
5 |
[OT] Wifi SIP phones - LinkSys WIP330 |
| 1:58PM |
1 |
1.4 - G729 - Have License - No path to translate from Zap to IAX2 |
| 1:04PM |
2 |
vzaphfc? |
| 12:57PM |
1 |
Music On Hold Between Servers |
| 11:34AM |
2 |
FW: cdr_addon_mysql.so did not register itselfduringload |
| 11:02AM |
0 |
cdr_addon_mysql.so did not register itselfduringload |
| 8:18AM |
2 |
Background switch to different context |
| 8:16AM |
1 |
FW: cdr_addon_mysql.so did not register itself duringload |
| 6:48AM |
2 |
Checking voicemail from outside |
| 6:32AM |
1 |
Re: Asterisk Queues |
| 1:50AM |
0 |
res_perl with asterisk 1.4 compile problem |
| 12:34AM |
0 |
Say who is using the PSTN? |
| |
| Wednesday December 27 2006 |
| Time | Replies | Subject |
|---|
| 11:09PM |
3 |
How to connect two asterisk server |
| 6:22PM |
2 |
Toll-Free number in India |
| 1:08PM |
0 |
cdr_addon_mysql.so did not register itself duringload |
| 11:44AM |
8 |
1.4.0, IMAP and Dovecot |
| 11:30AM |
2 |
Is ZTDUMMY still required with Asterisk 1.4? |
| 11:19AM |
1 |
Asterisk 1.4 Warnings |
| 11:10AM |
0 |
comand stun (for what) asterisk 1.4 |
| 10:05AM |
7 |
Searching the list |
| 8:45AM |
2 |
Verbose and sip invitestate logging question (1.4 release) |
| 8:05AM |
3 |
Polycom 601 Contacts List |
| 7:42AM |
0 |
problem with extentions |
| 6:38AM |
1 |
php agi trixbox help |
| 2:30AM |
0 |
AGI Dial channel status |
| |
| Tuesday December 26 2006 |
| Time | Replies | Subject |
|---|
| 11:47PM |
2 |
Agent presence |
| 6:06PM |
0 |
1.4 with a nortel call server 1000 running SIP(sdp headers) |
| 4:44PM |
3 |
I cant install zaptel drivers in suse 10.1 |
| 1:53PM |
1 |
Questions about 1.4 |
| 1:50PM |
0 |
1.4 with a nortel call server 1000 running SIP (sdp headers) |
| 12:51PM |
2 |
Number forwarding and porting? |
| 12:49PM |
0 |
(no subject) |
| 12:13PM |
1 |
agi+cepstral driving me nuts |
| 11:40AM |
2 |
Asterisk 1.4 missing sound in Spanish |
| 11:32AM |
1 |
Asterisk 1.4.0 (release) and G.729 |
| 11:07AM |
1 |
How to limit the duration of the MeetMe conversation? |
| 10:53AM |
0 |
controlled playback for MP3 |
| 10:27AM |
3 |
SIP Subscription Bug? |
| 10:14AM |
1 |
cdr_addon_mysql.so did not register itself during load |
| 9:36AM |
1 |
flight and the agi |
| 5:15AM |
1 |
1.4 and unicall |
| |
| Monday December 25 2006 |
| Time | Replies | Subject |
|---|
| 7:09PM |
2 |
Question about MWI in Asterisk 1.4.0 |
| 4:41PM |
2 |
Asterisk 1.4 - no PRI and no Zap? |
| 10:41AM |
1 |
Asterisk 1.4.0 Released! |
| 4:33AM |
0 |
Siedle DCA612-0 and Asterisk : problem with DTMF |
| |
| Sunday December 24 2006 |
| Time | Replies | Subject |
|---|
| 9:45AM |
0 |
asterisk with ldap |
| 7:03AM |
4 |
Escalate Call To Mobile |
| 1:35AM |
1 |
Voicemail hangup by gateway? |
| |
| Saturday December 23 2006 |
| Time | Replies | Subject |
|---|
| 8:37PM |
2 |
asterisk + door opener |
| 8:36PM |
0 |
centos4.4 x86_64 and zaptel-1.2.12 compile problems? |
| 5:50PM |
1 |
mySQL and to many connections with SQL statement UPDATE |
| 11:38AM |
1 |
CLI Errors and warnings |
| 7:16AM |
2 |
conditional dialplan |
| 3:02AM |
1 |
SNOM 200 behind NAT and other xmas woes |
| |
| Friday December 22 2006 |
| Time | Replies | Subject |
|---|
| 11:46PM |
2 |
UPDATE - Analog Phones with FSK/Stutter MWI |
| 10:33PM |
0 |
VXML in Asterisk HELP! |
| 9:23PM |
4 |
Happy X-mas |
| 6:51PM |
4 |
How accurate is show translation? |
| 2:05PM |
1 |
New astGUIclient VICIDIAL Release: 2.0.2 |
| 12:53PM |
2 |
Determining invalid extensions. |
| 12:51PM |
1 |
sangomo |
| 12:35PM |
0 |
RESOLVED: Sangoma Wanpipe 2.3.4-3 compilatio n fails un der FC2 with Zaptel 1.0.9.2 |
| 12:24PM |
1 |
spa300 password recovery |
| 12:21PM |
1 |
Answering Machine Detect (AMD) time values |
| 10:09AM |
0 |
Sangoma Wanpipe 2.3.4-3 compilation fails un der FC2 with Zaptel 1.0.9.2 |
| 10:03AM |
4 |
meetmejoin example |
| 9:58AM |
2 |
Sangoma Wanpipe 2.3.4-3 compilation fails under FC2 with Zaptel 1.0.9.2 |
| 7:39AM |
0 |
RE: RTPTIMEOUT Configuration |
| 6:19AM |
1 |
problems using the 1.4 version of meetme |
| 5:49AM |
2 |
System Application with java |
| 5:17AM |
1 |
Problem for trunk to trunk communication |
| |
| Thursday December 21 2006 |
| Time | Replies | Subject |
|---|
| 11:56PM |
0 |
question about astdb |
| 9:09PM |
2 |
Connect many fax lines? |
| 8:40PM |
2 |
Help with SUSE 10.2 and Sangoma A104D |
| 3:23PM |
3 |
International dialplans for Asterisk? |
| 3:03PM |
1 |
IAX calls not ringing |
| 2:30PM |
3 |
Grandstream GXW-4108 8 port FXO |
| 12:57PM |
0 |
GXP-2000 and Asterisk Configuration |
| 12:23PM |
1 |
Re: Match a Numer - then continue with, dialplan |
| 12:09PM |
2 |
asterisk crashed |
| 11:56AM |
2 |
more than 32 callgroups & pickupgroups |
| 7:57AM |
2 |
AELPARSE - Wish/Suggestion |
| 7:41AM |
2 |
Insert 1+areacode for VOIP calls |
| 1:31AM |
0 |
When line in use busy signal?! |
| 12:27AM |
0 |
The parameter of ast_request_and_dial() |
| |
| Wednesday December 20 2006 |
| Time | Replies | Subject |
|---|
| 10:28PM |
3 |
Calls disconnected after 1 hour |
| 10:12PM |
0 |
Re: Match a Numer - then continue with, dialplan |
| 7:59PM |
1 |
clear ast database |
| 7:35PM |
5 |
Sangoma A101 with Unicall |
| 5:14PM |
0 |
Re: Match a Numer - then continue with, dialplan |
| 3:48PM |
0 |
RE: spandsp 0.0.3 RxFax fax =?ISO-8859-1?Q?_reception crashes bristuffed_asterisk_1=2E2=2E13_[?= Virusgeprüft] |
| 2:54PM |
2 |
Re: Match a Numer - then continue with, dialplan |
| 2:54PM |
1 |
Agentcallbacklogin deprecation |
| 1:47PM |
0 |
Call Routing |
| 12:34PM |
2 |
Dial own extension to get to voicemail. |
| 11:50AM |
3 |
Re: Match a Numer - then continue with, dialplan |
| 11:23AM |
2 |
Asterisk Now |
| 10:27AM |
0 |
Can't make outgoing calls (T100P) |
| 10:26AM |
0 |
Re: Match a Numer - then continue with, dialplan |
| 10:09AM |
1 |
Incoming Lines Confusion |
| 10:07AM |
2 |
RE: spandsp 0.0.3 RxFax fax =?ISO-8859-1?Q?_reception crashes bristuffed_asterisk_1=2E2=2E13_[?= Virusgeprüft] |
| 9:00AM |
4 |
question about sip account format |
| 7:55AM |
1 |
Dial 9 to get out? |
| 6:39AM |
3 |
AgentCallbackLogin() deprecated in 1.4 |
| 6:18AM |
13 |
Need quality toll free 800 number over IAX? |
| 2:37AM |
0 |
asterisk run on vxworks for hardware pbx |
| 1:22AM |
3 |
IAX connection to FWD not working |
| 1:10AM |
0 |
better handling of calls forwarded by SIP phones |
| 12:29AM |
0 |
MATH function |
| |
| Tuesday December 19 2006 |
| Time | Replies | Subject |
|---|
| 10:17PM |
0 |
Need a 630 DID |
| 10:02PM |
1 |
Need Wholesale Termination |
| 9:00PM |
0 |
SIP and ZAP |
| 5:02PM |
3 |
Echo problem |
| 4:55PM |
0 |
Using Asterisk/Digium card with Tadiran switch |
| 4:38PM |
4 |
AstManProxy - Manager |
| 4:11PM |
0 |
Cisco devices (without STUN) and dynamic NAT |
| 3:54PM |
1 |
Is logic right? |
| 3:36PM |
1 |
SPAM-LOW: Re: .Call files do not seem to wo rk |
| 3:02PM |
1 |
T1 Pri Question |
| 2:49PM |
6 |
No music on hold? |
| 2:45PM |
26 |
Match a Numer - then continue with dialplan |
| 12:56PM |
1 |
SPAM-LOW: Re: .Call files do not seem to wo rk |
| 10:27AM |
0 |
Is MOH Still Broken in Asterisk 1.4 (beta3)? |
| 9:10AM |
0 |
db.c: Unable to open Asterisk database |
| 8:38AM |
1 |
DTMF Tones "A-B-C-D" |
| 8:35AM |
3 |
Parsing Area Code from CallerID |
| 8:34AM |
1 |
Polycom ring backs and CID |
| 8:17AM |
1 |
.Call files do not seem to work |
| 8:03AM |
0 |
Automatic sip conference |
| 7:35AM |
1 |
G.279 license question |
| 7:04AM |
0 |
dtmf and ivr |
| 4:22AM |
1 |
Distinctive Ring detection and caller ID |
| 4:22AM |
1 |
Asterisk-LDAP Integration? |
| 3:22AM |
1 |
Re: asterisk-users Digest, Vol 29, Issue 71 |
| 2:54AM |
2 |
2 devices using same sip account |
| 2:20AM |
1 |
AEL2 on Asterisk 1.2.4 |
| 2:15AM |
0 |
features.conf problems |
| 12:54AM |
1 |
[Fwd: Re: spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13] |
| |
| Monday December 18 2006 |
| Time | Replies | Subject |
|---|
| 11:58PM |
2 |
Remote Reboot of a Polycom |
| 11:15PM |
2 |
AGI Help Please |
| 11:11PM |
3 |
Changing CALLERIDNUM on the fly |
| 9:09PM |
1 |
Follow-me challenge |
| 8:54PM |
0 |
Cisco 7914 with sccp |
| 8:24PM |
0 |
openwrt wrt54gs running asterisk/pap2 |
| 8:10PM |
0 |
HITBSecConf2007 - Dubai - Call for Papers now open! |
| 8:01PM |
3 |
Inform callers on recorded/monitored number. |
| 6:34PM |
3 |
Billing solution |
| 6:27PM |
1 |
RE: Best way to access MySQL data from dial plan |
| 5:21PM |
1 |
MWI, Realtime SIP, Voicemail and Extensions, UAs registered with SER |
| 4:41PM |
1 |
Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME |
| 3:38PM |
0 |
Re: Best way to access MySQL data from dial plan |
| 2:25PM |
0 |
pap2/wrt54gs/asterisk |
| 2:22PM |
2 |
ZAP problem |
| 12:41PM |
0 |
Re: Best way to access MySQL data from dial plan |
| 11:55AM |
1 |
stop logging certain error messages |
| 11:54AM |
1 |
GXP2000, Linksys RV082 Firewall / NAT, Registrations |
| 11:47AM |
0 |
Colomachine & TE405P |
| 11:33AM |
1 |
Re: Best way to access MySQL data from dial plan |
| 10:35AM |
1 |
Voicemail delivery |
| 9:06AM |
1 |
Cisco 7940 - NAT Option |
| 8:15AM |
5 |
Asterisk and outlook |
| 7:52AM |
0 |
Wait command |
| 6:23AM |
2 |
asterisk to asterisk - to zap |
| 6:13AM |
3 |
Shared Line Appearances (SLA) in 1.4 |
| 6:07AM |
0 |
calls interrupted by music on hold |
| 4:52AM |
2 |
Digium TE405P with French E1 => Red Alert |
| 4:13AM |
1 |
Thomson ST2030S and BLF |
| 3:16AM |
1 |
Asterisk + Orion E1 GSM Gateway |
| 1:18AM |
8 |
spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13 |
| 12:19AM |
0 |
zap sending fax congested |
| |
| Sunday December 17 2006 |
| Time | Replies | Subject |
|---|
| 9:16PM |
0 |
Davox |
| 8:05PM |
3 |
CLI output to file |
| 5:59PM |
3 |
sip peer name channel variable? |
| 5:35PM |
1 |
bridging isdn calls to free up channels |
| 4:49PM |
5 |
BLF on GXP2000 |
| 3:31PM |
2 |
Dial 9 For Outside Line? |
| 1:24PM |
1 |
1.4 sounds long space before and after prompt |
| 12:51PM |
1 |
GXP2000 and BLF |
| 10:07AM |
1 |
What web interfaces are available today for debian based Asterisk installation? |
| 6:33AM |
2 |
Day/night service and indications on the phone |
| 3:52AM |
1 |
AGI and php simple example |
| 2:53AM |
3 |
call forward |
| 2:44AM |
0 |
Learning Asterisk Internals |
| |
| Saturday December 16 2006 |
| Time | Replies | Subject |
|---|
| 9:44PM |
0 |
PRI debugging outgoing not working, help needed |
| 1:04PM |
2 |
Asterisk 1.4.0 B4 Sounds Directory |
| 11:57AM |
1 |
rxfax detection problems with multiple contexts |
| 9:54AM |
0 |
SIP call not fail over immediately when Server Off Line |
| 9:38AM |
1 |
Asterisk 1.4.0-beta4 compile errors on Fedora 6 |
| 6:14AM |
5 |
Linux distro + Asterisk or Trixbox? |
| 4:11AM |
0 |
Asterisk 1.4.0b4 installation |
| |
| Friday December 15 2006 |
| Time | Replies | Subject |
|---|
| 8:53PM |
1 |
Asterisk 1.4.0-beta4 Released |
| 8:53PM |
0 |
Zaptel 1.4.0-beta3 Released |
| 8:53PM |
0 |
Asterisk 1.2.14 Released |
| 8:53PM |
0 |
Zaptel 1.2.12 Released |
| 7:44PM |
2 |
Bandwidth requirements for 1, 000, 000 minutes a month |
| 7:41PM |
1 |
ztmonitor displays full bar when idle |
| 7:32PM |
2 |
Boot load wcfxo does not configure self under Ubuntu 6 |
| 6:15PM |
1 |
Sipura question |
| 6:07PM |
1 |
International Provider |
| 4:58PM |
3 |
iax2 softphone attended transfers |
| 4:28PM |
2 |
Motherboard 3.3V PCI for TE412P |
| 3:28PM |
0 |
dialing via SIP URI |
| 2:55PM |
2 |
MOH Between Asterisk Servers |
| 2:23PM |
5 |
Good Commercial Grade Service Provider? |
| 12:36PM |
1 |
DTMF Tone Issues |
| 12:22PM |
4 |
Iptables rule help |
| 12:14PM |
2 |
Fast Busy Followup |
| 12:08PM |
1 |
What's up with DATETIME and TIMESTAMP in Asterisk 1.4beta3 ? |
| 11:10AM |
1 |
zapata.conf channel variable question |
| 11:00AM |
2 |
Sip port= not working |
| 10:35AM |
2 |
Trying to forward calls by using the Callee's context as the forward dial context |
| 10:33AM |
0 |
SIP DTMF not acted on for features in 1.4.0b3 |
| 10:26AM |
0 |
Cisco Call Manager 4.0 to Asterisk, Anyone haveSIP Reinvite working? |
| 9:26AM |
0 |
Hardware TDM Switching (Out Of Office - on vacation) |
| 7:33AM |
1 |
anyone using metermaid / parked call BLF? |
| 7:11AM |
0 |
AEL: CID match and pattern in switch statement |
| 6:54AM |
0 |
100rel & Prack enable |
| 6:51AM |
1 |
Cisco Call Manager 4.0 to Asterisk, Anyone have SIP Reinvite working? |
| 6:35AM |
1 |
Attended Transfer on queue_log |
| 6:32AM |
2 |
Hardware TDM Switching |
| 6:13AM |
2 |
is it possible to use Asterisk voicemail as anouncement system only? |
| 5:51AM |
2 |
call from h323 to SIP |
| 4:36AM |
2 |
How to know who hangup ? |
| 2:18AM |
1 |
Page + ParkAndAnnounce |
| 2:09AM |
2 |
enum |
| 12:32AM |
1 |
fxotune unable to set impedence |
| |
| Thursday December 14 2006 |
| Time | Replies | Subject |
|---|
| 11:52PM |
1 |
FYI Panasonic Wireless Phone MWI |
| 10:23PM |
1 |
Bandwidth.com on asterisk |
| 7:21PM |
0 |
bridging calls on a samsung pbx from asterisk |
| 3:38PM |
1 |
VoipTalk unable to accept calls at present? |
| 2:55PM |
3 |
StripXXX apps missing from asterisk-1.2.13? |
| 2:50PM |
2 |
Fast Busy |
| 2:47PM |
2 |
Console latency |
| 2:32PM |
4 |
Voicemail Live |
| 1:30PM |
2 |
On-Hold |
| 1:24PM |
1 |
Broadvoice registration problems |
| 11:04AM |
1 |
Show agent queue status on the phone? |
| 11:01AM |
0 |
Web-MeetMe ready for prime time? |
| 9:13AM |
3 |
IBM Server / USB Ports |
| 6:32AM |
2 |
Ssh access over a zap channel... |
| 6:29AM |
1 |
agi scripts running slowly |
| 6:23AM |
3 |
(no subject) |
| 6:05AM |
4 |
Zaptel under FC6 |
| 3:35AM |
3 |
AOC-D or similar |
| 3:34AM |
0 |
WRAP+astlinux g729 |
| |
| Wednesday December 13 2006 |
| Time | Replies | Subject |
|---|
| 8:30PM |
2 |
TDM400P won't ring GM phone of mere 0.1B |
| 7:44PM |
2 |
PRI to SIP |
| 7:30PM |
1 |
Phone routing - curious what others are doing? |
| 7:16PM |
0 |
Asterisk, Bluetooth, and wireless phone |
| 5:34PM |
5 |
Asterisk to a Huawei softX3000 problem has already been solved ? |
| 5:32PM |
1 |
Searchable Archives of this list |
| 3:39PM |
0 |
Asterisk Community lost a valuable contributor today |
| 2:27PM |
0 |
Re: Core Dump: create_transaction (p=0x0) atpbx_dundi.c:2787 |
| 2:06PM |
0 |
ZAP multiline handset questions |
| 2:00PM |
2 |
ssh access using zaptel channel to dial in. |
| 1:20PM |
0 |
webvoicemail |
| 1:01PM |
1 |
Polycom IP4000 and vsftpd 2.0.1 |
| 12:18PM |
1 |
Playing a sound file on handset pickup |
| 11:18AM |
2 |
how to define a secure trunk |
| 11:04AM |
0 |
Remember last IP address of IAX client |
| 11:03AM |
1 |
SRV Entries |
| 10:29AM |
0 |
Help with voicemail |
| 10:28AM |
1 |
record time with phones option buttons |
| 10:20AM |
1 |
Core Dump: create_transaction (p=0x0) at pbx_dundi.c:2787 |
| 10:09AM |
1 |
Diva Server V-BRI-2 and internal numbers |
| 10:08AM |
0 |
Remote-Party-ID and CallerID |
| 9:56AM |
0 |
FW: New Software available on Cisco.com P0S3-08-5-00 |
| 9:55AM |
1 |
Pickup application |
| 9:40AM |
2 |
Realtime +Mysql +Failover |
| 9:34AM |
1 |
MFC/R2 on chan_zap |
| 9:29AM |
0 |
Annoying echo echo problem problem ... |
| 8:33AM |
4 |
Polycom MyStat |
| 8:15AM |
3 |
MixMonitor and Queues |
| 8:04AM |
1 |
Audiocodes MediaPack MP-118 |
| 8:02AM |
1 |
CallerID Issue (asterisk newbie) |
| 7:55AM |
3 |
anyone used vitelity? |
| 7:22AM |
0 |
FW: MeetMe Conferencing and Marked Mode |
| 7:17AM |
1 |
Question about hardware |
| 6:55AM |
3 |
Stress test |
| 6:30AM |
1 |
IAX trunk problem |
| 5:53AM |
3 |
send fax by Iaxmodem ? |
| 5:23AM |
3 |
Multi Operator |
| 5:14AM |
3 |
How to temporarily unload modules. |
| 3:39AM |
0 |
TDM04B and shared IRQ ..but asterisk can work.. |
| 2:45AM |
1 |
Problem with asterisk 1.4 Installation (undefined reference to `ast_copy_string') |
| 1:05AM |
0 |
Asterisk and spandsp 0.3 |
| 12:07AM |
1 |
Hardware Suggestion for 2 PRI with call recording |
| |
| Tuesday December 12 2006 |
| Time | Replies | Subject |
|---|
| 8:13PM |
3 |
Need help getting started with asterisk |
| 7:23PM |
2 |
caller ID authentication |
| 7:00PM |
4 |
Measuring VoIP latency and packet loss |
| 4:03PM |
1 |
Settings CallerId for outgoing calls based on the sip account making them |
| 3:08PM |
0 |
[BULK] Asterisk manager |
| 3:03PM |
1 |
Conference between skinny user and many sip user |
| 2:32PM |
0 |
Disregistering Constantly - message: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181000' is now UNREACHABLE! Last qualify: 0 |
| 1:59PM |
0 |
ASTCC and DTMF |
| 1:27PM |
1 |
Cisco 7970 + New Firmware (8.2) |
| 12:08PM |
4 |
MeetMe Conferencing and Marked Mode |
| 12:04PM |
1 |
sip help for newbie |
| 10:48AM |
1 |
zapata.conf: cannot set txgain lower than -6.3 ? |
| 10:18AM |
1 |
long busy() |
| 10:04AM |
0 |
Auto answer when already on a call |
| 9:26AM |
5 |
Asterisk Manager |
| 9:04AM |
1 |
AGI problema |
| 9:00AM |
5 |
Input on Dundi |
| 8:21AM |
1 |
Anyone using Ranch Networks products for Load Balancing in a SIP environment? |
| 8:20AM |
1 |
SIP and IAX configuration from LDAP |
| 8:14AM |
3 |
outgoing call on ISDN PRI |
| 7:57AM |
2 |
repost gain problem with asterisk and zaptel 1.4 |
| 6:28AM |
2 |
Hangup Party |
| 5:53AM |
1 |
SPA2100 sends an unexpected BYE message when transmitting a FAX |
| 2:08AM |
0 |
Voicemail App |
| 1:48AM |
1 |
func_curl fails to compile, asterisk1.4 |
| |
| Monday December 11 2006 |
| Time | Replies | Subject |
|---|
| 11:45PM |
0 |
Asterisk 1.2.12 + fedora core 5 and TDM04B IRQ issues |
| 10:11PM |
0 |
Sip communicator issue |
| 9:31PM |
1 |
Problem in making outbound calls in PRI |
| 8:55PM |
0 |
Asterisk Sends 180-RINGINGto UAeven withprogressinband=yes |
| 6:38PM |
1 |
Asterisk Sends 180-RINGING to UAeven withprogressinband=yes |
| 4:58PM |
2 |
Asterisk Sends 180-RINGING to UA even withprogressinband=yes |
| 4:57PM |
2 |
How to add include statement into Realtime static |
| 4:23PM |
1 |
Asterisk Sends 180-RINGING to UA even with progressinband=yes |
| 3:27PM |
3 |
VPN As SIP Tunneling? |
| 3:22PM |
3 |
Using SIP with NAT (technical code question) |
| 2:18PM |
1 |
Unable to open pseudo channel for timing... Sound may be choppy. |
| 2:02PM |
0 |
Aculab |
| 1:52PM |
1 |
Extending Avaya IP Office ISDN30e with Asterisk |
| 1:52PM |
0 |
zaptel and zapata configuration |
| 1:19PM |
1 |
re: L option in dial command |
| 12:35PM |
1 |
IAX2 to SIP protocol translation overhead? |
| 10:39AM |
0 |
FW: [asterisk-dev] Kernel crash during modprobe wfxco |
| 10:32AM |
9 |
CLI History |
| 10:30AM |
2 |
asterisk PLAR |
| 10:26AM |
0 |
How to manipulate FROM header on Asterisk-DIALPLAN |
| 8:43AM |
0 |
Recall: Re: Recommendations for QoS, PoE Switches |
| 7:32AM |
0 |
OSP peering VOIP servers |
| 7:05AM |
0 |
Asterisk + Zap + CAS Signalling |
| 6:14AM |
1 |
Power requirements on the TDM-400 card |
| 6:01AM |
3 |
Asterisk and Fax How To |
| 5:55AM |
0 |
promotional info in music on hold |
| 5:41AM |
0 |
Cannot find ptlib-config, installing 1.4-beta3 |
| 3:12AM |
2 |
Waiting for dial tone in Dial cmd |
| 3:10AM |
0 |
OPS Protocol on Asterisk |
| 2:30AM |
2 |
Asterisk with IM |
| 12:19AM |
1 |
New installation CentOS 4 x86 or X86_64 |
| |
| Sunday December 10 2006 |
| Time | Replies | Subject |
|---|
| 10:17PM |
4 |
X100P clone dial problems. |
| 10:08PM |
0 |
: Some warnings occur |
| 9:26PM |
1 |
Mediatrix 1124 setup |
| 8:09PM |
0 |
Wifi Phone with Multiple Line Appearances |
| 7:05PM |
2 |
popups, queue & agents |
| 7:00PM |
0 |
gain problems with zaptel 1.4 beta 2 |
| 6:51PM |
3 |
Xen, Asterisk & ISDN: Timing Problems |
| 4:16PM |
0 |
tx_fax |
| 3:02PM |
1 |
Problem faxing with SPA2100 in passthru mode. |
| 2:29PM |
5 |
TDM2400 |
| 2:10PM |
2 |
Display variables |
| 9:52AM |
3 |
Asterisk from Debian Packages |
| 8:02AM |
1 |
chan_sip.c:5267 sip_reg_timeout Error |
| 5:13AM |
3 |
Asterisk 1.4b3 & Realtime Voicemail |
| 2:11AM |
1 |
NAT and Dial to two channels at once |
| 12:10AM |
10 |
Recommendations for QoS, PoE Switches |
| |
| Saturday December 9 2006 |
| Time | Replies | Subject |
|---|
| 9:36PM |
1 |
Anonymous clid ? |
| 8:45PM |
1 |
Jabber Client |
| 1:53PM |
2 |
PCI, PCI-X and PCI-e -- Server / Interface Card Selection |
| 12:54PM |
0 |
Quicknet PhoneJack questions. |
| 10:07AM |
3 |
Zaptel module compile woes |
| 2:15AM |
2 |
RDNIS question |
| |
| Friday December 8 2006 |
| Time | Replies | Subject |
|---|
| 9:55PM |
1 |
Asterisk voice recording through TE110p |
| 8:37PM |
1 |
using a mobile phone as a handset via bluetooth |
| 7:29PM |
0 |
trixbox |
| 5:10PM |
1 |
SIP Quality Metrics |
| 4:27PM |
1 |
Polycom soft buttons not working |
| 2:33PM |
0 |
SIP/IAX Fax Detect on Asterisk 1.4 |
| 2:25PM |
0 |
Best book to learn SIP details ? |
| 12:33PM |
1 |
Asterisk forgetting about client registration or Polycom phone forgetting to register? |
| 12:28PM |
2 |
Repeated Digits |
| 12:17PM |
1 |
Douglas Garstang <dgarstang@oneeighty.com> |
| 11:55AM |
0 |
Dial groups, groups of phones, multiple line keys |
| 11:49AM |
0 |
Asterisk eating the Asterisk key! |
| 11:15AM |
2 |
downloading asterisk GUI |
| 11:02AM |
1 |
Question on retrieve_file() function in app_voicemail.c |
| 10:19AM |
0 |
Verizon VoiceWing support |
| 10:00AM |
0 |
codec_speex.c: Out of buffer space |
| 9:14AM |
2 |
5.8gig phone MWI |
| 8:52AM |
1 |
CTI: put on hold a call |
| 8:46AM |
1 |
cal recording with email |
| 7:25AM |
3 |
Vonage SIP access via asterisk? |
| 7:02AM |
2 |
AGI interaction with php |
| 6:42AM |
3 |
How to communicated Both SIP and IAX2 each other ? |
| 6:01AM |
0 |
RE: Answer a call that is not ringing on yourextension |
| 5:06AM |
0 |
problem with asterisk 1.4 |
| 4:00AM |
2 |
Management GUI |
| 3:35AM |
2 |
Server for 100 concurrent calls |
| |
| Thursday December 7 2006 |
| Time | Replies | Subject |
|---|
| 9:56PM |
0 |
nordx designator labels ? |
| 9:56PM |
2 |
ASTERISK y AGC |
| 8:50PM |
1 |
Basic question regarding re-INVITE |
| 8:00PM |
1 |
AMI - Originate Action and Busy, NoAnswer calls - CDR |
| 7:56PM |
3 |
wierd callerid problem |
| 5:03PM |
1 |
Asterisk 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking) |
| 3:54PM |
0 |
Audio Convert Module |
| 2:38PM |
0 |
Need help on AgentCallbackLogin() |
| 2:33PM |
1 |
Asterisk 1.4 + Cisco 7970 |
| 1:28PM |
0 |
Session Progress Transmission to Phone |
| 1:15PM |
2 |
queue agent Monitor |
| 12:21PM |
2 |
Polycom buddies question |
| 12:15PM |
0 |
Asterisk stopped Matching Defined Peer |
| 11:53AM |
1 |
Codec Selection in asterisk |
| 11:34AM |
1 |
"illegal" VoIP in India |
| 9:51AM |
5 |
CISCO 2600 - VWIC 1MFT-E1 |
| 9:50AM |
0 |
UDP ports |
| 9:16AM |
1 |
FXO USB that works with Asterisk? |
| 8:33AM |
0 |
Job Posting, Asterisk Engineer/Sales Engineer, Dallas TX Area |
| 8:19AM |
1 |
queue member refresh |
| 8:19AM |
0 |
Fax machine detect (akin to AMD) |
| 7:59AM |
1 |
Asterisk accepting calls to fast |
| 7:52AM |
1 |
Standardized IVR UI Pattern (was: Re: Is there any Asterisk controllable thermostat?) |
| 7:25AM |
7 |
Running Asterisk on a Home rotuer |
| 7:25AM |
3 |
Plantronics and Snom RF feedback |
| 7:19AM |
1 |
-- Called 12127773456@OOH323 Segmentation fault (core dumped) |
| 6:09AM |
1 |
eicon diva BRI problems |
| 3:34AM |
0 |
sip qualify unreachable/reachable - ci$co 7940 |
| 2:56AM |
2 |
oh323.conf question |
| 2:46AM |
1 |
how to configure Asterisk to support SIP "INFO" method? |
| 1:57AM |
0 |
Requested transfer capability: 0x00 - SPEECH - How to change to 31KAUDIO? |
| 1:39AM |
0 |
calls not terminating (2nd posting) |
| 12:32AM |
0 |
ChanSpy * and 1234# not working |
| |
| Wednesday December 6 2006 |
| Time | Replies | Subject |
|---|
| 7:09PM |
2 |
MWI across multiple servers |
| 6:59PM |
0 |
Echo problem with TDM440P and ADSL Line |
| 6:16PM |
1 |
0002475: [patch] Allow app_directory to work with REALTIME |
| 4:56PM |
1 |
Error compiling Eicon Diva from source |
| 2:54PM |
0 |
Avoided initial deadlock asterisk v 1.2.12.1 SIP clients IAX2 termination. |
| 2:40PM |
1 |
Govarion. |
| 12:09PM |
9 |
Setting outgoing caller id on a zap channel for one sip extension only |
| 11:48AM |
0 |
(REPOST DUE TO NO ANSWER) translate.c:88 powerof: Powerof 0: No power?? / translate.c:133 ast_translator_build_path: No translator |
| 10:41AM |
1 |
FW: G.726 on Asterisk 1.4.0 |
| 10:03AM |
1 |
Detecting no answers and/or disconnected numbers |
| 9:31AM |
0 |
Dec 6 09:47:52 NOTICE[3263]: chan_iax2.c:1619 iax2_destroy: Avoiding IAX destroy deadlock |
| 8:31AM |
0 |
Error in codec string '=audio 5004 RTP/SAVP 3' |
| 7:53AM |
2 |
problem with asterisk - calls where both sidescannot hear each other |
| 7:36AM |
1 |
Same issue, different way to ask. |
| 7:35AM |
1 |
CAS DID 2way |
| 7:09AM |
0 |
iax/sip registering and real-time |
| 7:07AM |
0 |
MWI/realtime/openSer in 1.4 |
| 6:46AM |
1 |
Agent autologoff dynamic queue members - Brain aches please help |
| 6:15AM |
1 |
Ping |
| 6:10AM |
1 |
Can not hear called party |
| 5:23AM |
1 |
G.729E |
| 4:36AM |
0 |
Problems with bridging data calls over Wildcard TE405P |
| 2:47AM |
2 |
ParkAndAnnounce + Paging |
| 2:43AM |
3 |
Asterisk freezes when DNS not working: a BUG?? |
| 2:35AM |
1 |
problem with asterisk-1.4+sip communicator |
| 12:00AM |
0 |
asterisk -1.4 with sip communicator |
| |
| Tuesday December 5 2006 |
| Time | Replies | Subject |
|---|
| 11:31PM |
1 |
for all Asterisk Users |
| 11:05PM |
0 |
Jumpers and DIP switches on Atcom AX-4S and AX-1E |
| 9:26PM |
3 |
Rejecting a Call |
| 9:08PM |
1 |
Problem loading unicall |
| 8:25PM |
0 |
Melbn Asterisk/Voip get together |
| 8:15PM |
0 |
RE: SOLVED - T1 PRI not announce "this is long distance call, please add 1 for this call..." |
| 7:41PM |
2 |
TE110P Out fine / In Fail |
| 6:08PM |
1 |
Auto dialing: .call file vs. manager interface |
| 5:00PM |
0 |
[Fwd: RE: any possibility of Vonage Integration] |
| 4:49PM |
1 |
Need some examples for configuring Asterisk under Realtime static |
| 4:36PM |
6 |
Switching from FreeBSD to Linux - which distro? |
| 4:23PM |
1 |
Question about Realtime static table |
| 3:40PM |
1 |
Meetme monitoring (once) |
| 2:50PM |
1 |
Install via SVN or tarball? |
| 2:41PM |
0 |
Re: regcontext, NoOp extension vanishes when extension reload, WORKING |
| 2:27PM |
1 |
problem with asterisk - calls where both sides cannot hear each other |
| 2:23PM |
0 |
RE: regcontext, NoOp extension vanishes when extension reload |
| 1:49PM |
0 |
RE: regcontext, NoOp extension vanishes when extension reload |
| 1:30PM |
2 |
SIP firmware for Siemens Optipoint 410 Economy? |
| 1:13PM |
4 |
Attended Transfer |
| 12:29PM |
0 |
G.726 on Asterisk 1.4.0 |
| 12:14PM |
1 |
Help with dial plan - two attempts at calling agent before logging agent off? |
| 11:50AM |
2 |
regcontext, NoOp extension vanishes when extension reload and doesn't come back |
| 11:47AM |
0 |
Realtime Error 1045 |
| 11:31AM |
0 |
SOLVED: DB9 e1 to RJ45 pinout |
| 10:38AM |
1 |
SetCallingPres propagation |
| 10:32AM |
0 |
nvlinedetect |
| 10:14AM |
2 |
Realtime question |
| 9:39AM |
1 |
installed, stumped on sip registration |
| 9:16AM |
1 |
No ID from the calling party in SIP Header |
| 8:56AM |
0 |
Issues |
| 8:56AM |
1 |
SER/OpenSER + Asterisk + Queue |
| 8:42AM |
1 |
Shared Line Appearances |
| 7:40AM |
4 |
question on tx_fax install for asterisk 1.4 |
| 6:59AM |
1 |
calls not terminating |
| 6:23AM |
0 |
Diginetwork X100P card |
| 6:17AM |
8 |
centos 4.4 + asterisk |
| 5:37AM |
0 |
Signalling but no media |
| 4:50AM |
1 |
sip_write warning when executing Pickup of CAPI |
| 2:20AM |
2 |
zaptel-1.4.0-beta2 Getting it to compile on Fedora Core 6 _64bit |
| 1:39AM |
1 |
Max T1s in a server? |
| |
| Monday December 4 2006 |
| Time | Replies | Subject |
|---|
| 11:44PM |
2 |
How to stop Asterisk to pick up incoming PSTN signal |
| 10:55PM |
5 |
any possibility of Vonage Integration |
| 4:41PM |
1 |
T1 PRI not announce "this is long distance call, please add 1 for this call..." |
| 1:14PM |
3 |
Digium TE407P vs. Sangoma A104d |
| 12:34PM |
0 |
Registering VoIP providers with realtime |
| 12:03PM |
2 |
ASterisk and SER |
| 11:35AM |
3 |
Answer a call that is not ringing on your extension |
| 10:04AM |
1 |
Sangoma a301 or other DS3 card |
| 9:28AM |
0 |
Addqueuemember and roaming users problem. |
| 8:25AM |
0 |
No answer when press 0 for operator in VM in 1.0 .9? |
| 8:08AM |
2 |
Odd queue issue |
| 8:07AM |
0 |
google talk |
| 7:40AM |
1 |
Moderate setup |
| 7:32AM |
0 |
Codec transcoding and call recording |
| 6:53AM |
0 |
mwi for voicemail not showing up for realtimeconfig. |
| 6:05AM |
1 |
forward skinny call to SIP |
| 5:36AM |
4 |
MySQL cmd % pattern matching |
| 5:12AM |
1 |
Problem with h323 support |
| 5:10AM |
1 |
Nokia E60 problems |
| 1:27AM |
0 |
Can zaptel freak out if you configure 2 trunks but use only one? |
| 1:20AM |
0 |
Extend time in call pickup |
| 12:17AM |
1 |
mwi for voicemail not showing up for realtime config. |
| 12:11AM |
1 |
HOW TO - Asterisk apps/modify and compile |
| |
| Sunday December 3 2006 |
| Time | Replies | Subject |
|---|
| 11:04PM |
1 |
How can i processed with Call Snooping, |
| 9:04PM |
11 |
Is there any Asterisk controllable thermostat? |
| 7:12PM |
3 |
TDM01B installation |
| 7:03PM |
0 |
Neat Skype Device |
| 5:05PM |
1 |
G729 Passthru? |
| 4:55PM |
1 |
asterisk manager originate command |
| 3:08PM |
0 |
Asterisk : Numbers Guessing Game |
| 3:06PM |
0 |
Asterisk 1.4: SPANDSP3 (WIP) HOWTO |
| 1:49PM |
0 |
VoIP GSM Gateways |
| 1:04PM |
1 |
Realtime fullcontact field contains nat device private ip |
| 12:01PM |
0 |
* key on Linksys SPA-841 |
| 8:39AM |
0 |
G729 Liscence |
| 8:39AM |
1 |
RTP Media Path |
| 8:24AM |
0 |
translate.c:88 powerof: Powerof 0: No power?? / translate.c:133 ast_translator_build_path: No translator |
| 7:08AM |
0 |
RNK |
| |
| Saturday December 2 2006 |
| Time | Replies | Subject |
|---|
| 10:49PM |
0 |
Answering Machine detection in Australia |
| 3:09PM |
0 |
rxfax or spandsp problems?? |
| 12:29PM |
2 |
"Low" beep on voicemail |
| 12:02PM |
1 |
Linksys PAP2t-NA and Asterisk |
| 9:59AM |
1 |
Detailed description of problem in Poland |
| 9:53AM |
3 |
Problem in Poland |
| 9:42AM |
0 |
RINGNOANSWER on 1.2 |
| 7:07AM |
4 |
Help with IAX Trunk |
| |
| Friday December 1 2006 |
| Time | Replies | Subject |
|---|
| 5:21PM |
0 |
no tx audio |
| 3:40PM |
0 |
video for call attendant systems |
| 3:03PM |
2 |
sip address in voicemail emails |
| 2:33PM |
1 |
Interesting CALLERID behavior |
| 1:29PM |
0 |
Audiocodes MP104-FXO - Transfer the call only after 3 rings |
| 11:50AM |
0 |
setcallerpres not working |
| 11:44AM |
3 |
direct IP calling with extension |
| 10:45AM |
2 |
Recommendation for FXO |
| 10:18AM |
0 |
feel free to add to the bounty for issue 8064 |
| 9:40AM |
0 |
App_Swift |
| 9:31AM |
0 |
Problem with agent AgentCallbackLogin() |
| 9:26AM |
1 |
app_sql_postgres gone in 1.4 |
| 9:09AM |
2 |
CALL TRANSFER |
| 8:42AM |
1 |
No caller ID, no incoming call |
| 8:41AM |
0 |
server specs / hardware |
| 8:24AM |
0 |
spa3k dtmf problem asterisk 1.2.x |
| 7:26AM |
1 |
Caller ID Rewrite |
| 6:27AM |
0 |
Asterisk as bridge, strange ${EXTEN} values |
| 6:22AM |
0 |
Music on hold |
| 6:06AM |
0 |
ISDN BRI lines engaged when dialing out |
| 5:46AM |
2 |
Cisco IAXmodem HylaFAX |
| 2:26AM |
0 |
seed vs registration? |
| 1:07AM |
1 |
H323 NAT Problem |
| 1:02AM |
3 |
Asterisk: SIP Gateway or Proxy |