| Thursday May 31 2007 |
| Time | Replies | Subject |
|---|
| 11:09PM |
0 |
Urgent-- Error while installing app_dtmftotext. |
| 10:29PM |
5 |
Auto Dial Problem |
| 9:39PM |
1 |
Compilation after Source code changes in Asterisk |
| 8:37PM |
1 |
CARD FOR inband signal |
| 7:33PM |
0 |
FreePbx/asterisk/openser |
| 3:06PM |
0 |
Custom CLI (spoofing) ? |
| 1:44PM |
1 |
CDR timing |
| 1:39PM |
0 |
Chan_sip max channels limit? |
| 1:21PM |
6 |
High Port Count ATA |
| 12:45PM |
0 |
Track Agent login/logoff status |
| 12:11PM |
2 |
asterisk auto dial does not wait for answer |
| 11:42AM |
2 |
applicationmap on features |
| 10:14AM |
3 |
RF to IP bridge |
| 9:32AM |
0 |
HP OfficeJet 6110, Sipura 2102, T.38, and Clarent |
| 9:26AM |
2 |
Net2Phone Multiple SIP Trunk Not Working |
| 9:19AM |
0 |
Asterisk Release Maintenance News |
| 9:18AM |
1 |
Duplicate UNIQUEID on CDR |
| 9:15AM |
1 |
linksys pap2 version2 ata DTMF issue |
| 8:29AM |
3 |
moh backround? |
| 8:25AM |
0 |
Meetme context. |
| 8:03AM |
9 |
click to call |
| 7:17AM |
1 |
Who are XO and L3? |
| 5:58AM |
0 |
Asterisk Users Conference for Friday June 1st 2007 @ 12:30 PM EDT |
| 5:44AM |
3 |
'asterisk' shown on display |
| 4:35AM |
4 |
Context documentation for the newbie! |
| 3:59AM |
1 |
ringback detection |
| 2:53AM |
1 |
Passing call duration to an AGI Script |
| 1:37AM |
1 |
Diva and Asterisk |
| 12:58AM |
2 |
How to read SIP debug? |
| 12:10AM |
1 |
Subscriptions to Agent hints stopped working after a changing the numbering |
| |
| Wednesday May 30 2007 |
| Time | Replies | Subject |
|---|
| 7:32PM |
1 |
Application Developer |
| 6:09PM |
0 |
montavista and Asterisk |
| 5:17PM |
1 |
Delays on E1 Delivered via SHDSL |
| 12:00PM |
2 |
(no subject) |
| 11:46AM |
1 |
FW: Help with IAX |
| 9:54AM |
3 |
Dial plan inquiry using GotoIf() |
| 8:37AM |
4 |
Help with IAX |
| 8:32AM |
5 |
any codec passthru mode |
| 5:35AM |
1 |
Asternic Flash panel |
| 4:58AM |
0 |
Call transfer while dialing |
| 4:19AM |
0 |
SIP SendURL |
| 3:18AM |
0 |
*End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW * |
| 3:03AM |
12 |
False ring problem |
| 2:47AM |
1 |
fax2mail ann missing CallerID number |
| 2:15AM |
2 |
multiple host= in sip.conf |
| 2:06AM |
0 |
Configuring Asterisk as Gateway SIP-H.323 via ooh323 |
| 2:02AM |
0 |
Allow for context includes in realtime (ARA) |
| |
| Tuesday May 29 2007 |
| Time | Replies | Subject |
|---|
| 10:08PM |
0 |
Play sound file on keypress (bridged call) |
| 3:23PM |
7 |
Problem on incoming call from Zap channel to SIP phones... |
| 2:36PM |
2 |
channel_find_locked: Avoided deadlock |
| 2:08PM |
4 |
SBC |
| 1:24PM |
1 |
*End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW * |
| 12:13PM |
0 |
Sending a SIP INVITE without SDP from Asterisk |
| 11:19AM |
0 |
Theoretical and Received SIP addresses causing no audio |
| 10:23AM |
2 |
Asterisk as a call recorder for ISDN30 ? |
| 8:56AM |
1 |
Monitor application inestability and high load |
| 8:53AM |
2 |
Agents.conf from realtime static |
| 6:49AM |
0 |
SIP OPTIONS triggering some action in case of no reply |
| 6:38AM |
4 |
*End Of Life ASTERISK 1.2.X Was: INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASEREAD NOW * |
| 5:55AM |
2 |
OpenVox A400P01on thin client? |
| 5:22AM |
1 |
disable musiconhold |
| 2:38AM |
0 |
Billion on Debian Etch |
| 2:19AM |
0 |
* INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW * |
| 2:11AM |
3 |
Zaptel linux26 |
| 12:11AM |
3 |
zaptel module dependences |
| |
| Monday May 28 2007 |
| Time | Replies | Subject |
|---|
| 7:52PM |
2 |
Alcatel - Asterisk setup |
| 7:45PM |
2 |
ekiga register problems |
| 7:38PM |
1 |
[1.2.18] Wrong steps in extensions.conf? |
| 6:07PM |
0 |
Multiple TDM400p cards in one machine -- nolonger an issue? |
| 2:09PM |
4 |
Language in Zaptel.conf |
| 1:40PM |
0 |
Send parked call extension to set |
| 12:55PM |
2 |
Polycom Static IP |
| 11:12AM |
1 |
Asterisk and cell phones |
| 10:30AM |
1 |
Octasic echo cancellation |
| 10:10AM |
2 |
Multiple TDM400p cards in one machine -- no longer an issue? |
| 9:57AM |
1 |
Astmanproxy |
| 7:18AM |
5 |
Blindside Web Conferencing |
| 6:09AM |
2 |
help on asterisk sipp |
| 5:27AM |
0 |
Limit outgoing call for sip peer |
| 3:29AM |
0 |
include=>context in REALTIME!!!!!! |
| 3:17AM |
1 |
Queues with announce |
| 2:42AM |
1 |
Meet me |
| 1:56AM |
0 |
Progress passing problem. |
| |
| Sunday May 27 2007 |
| Time | Replies | Subject |
|---|
| 8:13PM |
0 |
Start recording automatically when |
| 7:51PM |
1 |
Please advise: Asterisk on Dell PowerEdge 1750 w/ hyperthreading |
| 2:21PM |
2 |
Cisco remote reboot |
| 12:59PM |
1 |
Divitas |
| 9:50AM |
1 |
h323friends & peer realtime |
| 8:52AM |
4 |
Zonbu |
| 8:44AM |
2 |
Asterisk 1.2.18 problem |
| 2:16AM |
2 |
SIP accounts from MYSQL. |
| |
| Saturday May 26 2007 |
| Time | Replies | Subject |
|---|
| 6:13PM |
2 |
Connect two Asterisk boxes through IVR Menu |
| 3:33PM |
4 |
reset Polycom phones remotely |
| 12:33PM |
2 |
test tools of Asterisk server |
| 10:59AM |
0 |
Voicemail and Time Conditions |
| 6:06AM |
4 |
Asterisk in Xen domu with tdm400 hardware |
| 2:45AM |
1 |
Wi-Fi+Wireless Router |
| 2:36AM |
0 |
Asterisk+IMATE PDAL Configuration+Softphone |
| 1:31AM |
2 |
chan_capi install problems |
| |
| Friday May 25 2007 |
| Time | Replies | Subject |
|---|
| 7:33PM |
2 |
Linksys WRTP54G-NA with SIP |
| 6:16PM |
0 |
(OT) Interesting and Cheap Device BAFO VoIP Internet Telephony Device Messenger CallBox |
| 3:16PM |
1 |
CDR not recording accountcode on SIP Response 302 Call Forward From Phone |
| 2:40PM |
0 |
GS BT200 dialling PC501 |
| 2:17PM |
2 |
TDM bus extension. |
| 1:14PM |
2 |
Queue help: Extending RRMEMORY strategy to use penalty |
| 12:21PM |
0 |
mysql connect |
| 11:49AM |
9 |
Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes |
| 10:09AM |
1 |
Suggested BRI cards? |
| 10:03AM |
0 |
Automated outbound call retries |
| 9:58AM |
1 |
wait for rings, answer on outdial via SIP |
| 9:45AM |
3 |
Asterisk with Multiple Network Interfaces |
| 9:13AM |
0 |
standard TDM interface cards that work in Asterisk? |
| 8:54AM |
1 |
Start recording automatically when xferring to an extension? |
| 8:32AM |
1 |
H Parameter in Dial Command |
| 8:31AM |
5 |
Polycom or Linksys phones bootp tftp config setup |
| 7:14AM |
3 |
Urgent: DTMF does not work with, rtpmap:101 telephone-event/8000 |
| 5:56AM |
1 |
Problem with call parking |
| 4:32AM |
3 |
how to use sable (festival) markup with asterisk |
| 4:19AM |
0 |
Asterisk to Alcatel 4400 via PRI: analog extensions work - digital do not |
| 4:15AM |
1 |
Matching "+" at the beginning of the line |
| 2:58AM |
0 |
rxgain/txgain in chan_sip |
| 2:04AM |
0 |
Asterisk Users Conference Friday May 25th 12:30 PM EDT |
| |
| Thursday May 24 2007 |
| Time | Replies | Subject |
|---|
| 10:31PM |
0 |
Re: asterisk-users Digest, Vol 34, Issue 114 |
| 9:40PM |
3 |
Urgent: DTMF does not work with rtpmap:101 telephone-event/8000 |
| 8:57PM |
3 |
Echo on hard SIP devices... |
| 6:37PM |
1 |
vmoutcall] |
| 4:28PM |
2 |
Call Center Application |
| 3:15PM |
2 |
Login log out support |
| 2:54PM |
2 |
Cisco CP-7970G |
| 2:04PM |
2 |
transfer call sip to zap |
| 1:46PM |
2 |
Conference room as Music on Hold |
| 11:21AM |
1 |
Basic connection between Mitel 3300 ICP and Asterisk (trixbox) - from a clueless newbie.... |
| 10:01AM |
1 |
vmoutcall |
| 9:35AM |
13 |
Bottom line on fax reception |
| 7:46AM |
2 |
SCCP |
| 7:44AM |
6 |
Integrated T1 |
| 7:32AM |
2 |
Additional commands for MeetMeAdmin |
| 7:28AM |
3 |
meetme sounds |
| 7:22AM |
0 |
SLA with SIP-only environment |
| 6:41AM |
1 |
Nokia release |
| 6:18AM |
1 |
Parking Lot CallerID |
| 6:04AM |
0 |
bridging calls between two numbers with extensions |
| 5:52AM |
1 |
PSP Voip |
| 4:54AM |
1 |
PRI problem, pri_fixup_principle: Call specified, but not found? |
| 2:17AM |
3 |
modprobe |
| 2:03AM |
0 |
asterisk-backports.org giveaway |
| 1:53AM |
0 |
redirect on AT-530 IP Phone |
| 1:44AM |
2 |
There is no tone on an outgoing call |
| |
| Wednesday May 23 2007 |
| Time | Replies | Subject |
|---|
| 7:57PM |
0 |
AW: WiFi SIP phones |
| 7:56PM |
0 |
Realtime Queues and Agents |
| 7:51PM |
1 |
CDR on channel 'IAX2/u92613106-3' already started |
| 7:35PM |
0 |
Fax Detection Using Nvdetect |
| 5:47PM |
0 |
IVR Loop on invalid input |
| 3:32PM |
0 |
Deadlock problem with agents, queues and PRI (stop accepting incoming calls in PRI line) |
| 1:37PM |
1 |
Call limit per sip account user. |
| 1:32PM |
3 |
Using gizmo as softphone for Linux |
| 12:53PM |
0 |
ITSP that honors Dial Around Compensation |
| 12:34PM |
0 |
Problems compiling res_config_mysql (asterisk addons) |
| 12:28PM |
1 |
- SOLVED - stream file not working but get data and exec background work |
| 11:48AM |
1 |
stream file not working but get data and exec background work |
| 11:34AM |
3 |
TE205P, E1, Panasonic PBX and hang-up issues |
| 8:23AM |
0 |
asterisk+nortel3904 |
| 8:15AM |
1 |
Asterisk Realtime problem |
| 7:23AM |
1 |
Asterisk and CCM 5.x SIP trunk |
| 7:10AM |
0 |
SIP.CONF: incominglimit and outgoinglimit |
| 6:51AM |
1 |
Bristuff with Billion ISDN |
| 6:31AM |
1 |
problem with attended call transfer |
| 6:31AM |
1 |
voicemail notification. |
| 5:59AM |
0 |
KCAUG Meeting Reminder! |
| 5:27AM |
3 |
What replaces SetCallerPres in 1.4 |
| 5:01AM |
1 |
None random SIP channel names |
| 4:53AM |
1 |
SCCP + hint |
| 4:52AM |
4 |
showing camera on video phone |
| 3:33AM |
0 |
Need starter information (newbie) |
| 3:01AM |
16 |
WiFi SIP phones |
| 12:41AM |
3 |
SIP Dial Command to a non-Asterisk url |
| |
| Tuesday May 22 2007 |
| Time | Replies | Subject |
|---|
| 7:12PM |
0 |
Mix Dial, Chanspy and MixMonitor or Monitor |
| 3:14PM |
2 |
Fax detection |
| 3:14PM |
0 |
(no subject) |
| 2:43PM |
2 |
FXS + Pots Extensions Help |
| 1:49PM |
0 |
FW: autologoff |
| 11:50AM |
4 |
Working softphone for poket PC |
| 11:38AM |
0 |
how to disable global authentication for registration |
| 10:44AM |
1 |
SMS |
| 10:06AM |
0 |
Astsee v0.5 now available, X/Linux Asterisk Usage Auditor and Monitor |
| 10:05AM |
2 |
Phones fail to ring |
| 9:23AM |
1 |
Net 2 Phone - Asterisk - Problem |
| 7:07AM |
1 |
Kernel Panic in wct4xxp during unload on Zaptel-1.4.4 |
| 6:22AM |
2 |
how can I catch the event generated when a parked call is hung up? |
| 5:11AM |
3 |
Dial out issues. |
| 5:08AM |
8 |
SIP & Echo |
| 4:53AM |
0 |
Dialplan Problem - Outgoing |
| 4:31AM |
0 |
asterisk TAPI interface |
| 3:21AM |
1 |
Local SMS how-to. |
| 2:32AM |
1 |
Why 2 branches of asterisk development? |
| 2:30AM |
1 |
Blackberry 8800+VoIP Configuration |
| |
| Monday May 21 2007 |
| Time | Replies | Subject |
|---|
| 8:02PM |
1 |
Voice mail issue |
| 2:02PM |
1 |
Windows Media streaming for MOH? |
| 1:38PM |
1 |
getting a call back from voicemail? |
| 12:42PM |
3 |
Originate and bridge Can it be done? Best Way? |
| 12:01PM |
2 |
MoH WAY too loud |
| 11:54AM |
0 |
AGI: Festival & Ringing on Screening not working properly |
| 11:39AM |
3 |
Aastra MWI |
| 10:08AM |
1 |
CAS signalling conflicts with Clear channel |
| 8:18AM |
1 |
FW: Re install |
| 7:46AM |
0 |
Grandstream FXS Gateway star codes |
| 7:42AM |
2 |
Help installing on OpenSuSE 10.2 |
| 7:07AM |
1 |
Delete voicemails after X days |
| 6:20AM |
2 |
VoiceMail Access |
| 5:18AM |
1 |
DTMFToText Installation process |
| 2:58AM |
1 |
MusicOnHold() stops after exactly 60 seconds |
| 2:51AM |
2 |
Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage |
| 2:49AM |
3 |
asterisk and fax machine |
| 1:56AM |
0 |
Asterisk Users Conference this Friday: Kerry from Trixbox |
| 1:41AM |
0 |
"dtmf transcoding" with asterisk |
| 1:01AM |
1 |
Vicidial |
| 12:07AM |
0 |
compile asterisk in arm-linux! |
| 12:01AM |
0 |
Gustavo Souza Queiroz está ausente do escritório. |
| |
| Sunday May 20 2007 |
| Time | Replies | Subject |
|---|
| 11:53PM |
2 |
MySQL/IVR Integration |
| 11:02PM |
1 |
Queuemetrics and Asterisknow |
| 9:17AM |
1 |
Caller ID matching |
| 7:41AM |
2 |
OpenWengo + Asterisk? |
| |
| Saturday May 19 2007 |
| Time | Replies | Subject |
|---|
| 8:03PM |
0 |
Astersik+ss7 |
| 7:18PM |
1 |
Call someone to instantly join conference using MeetMe |
| 10:26AM |
1 |
Zaptel hangs machine... |
| 10:05AM |
2 |
Asterisk On Solaris 10 |
| 9:34AM |
0 |
SIP NOTIFY on voicemail |
| 9:05AM |
2 |
(OT) Anyone Ever Use http://shopfort1.com as a Broker |
| 8:01AM |
3 |
Asterisk and iBasis |
| 7:54AM |
0 |
a/b door phone |
| 7:20AM |
3 |
Asterisk on OpenSuSE 10.2 |
| 7:07AM |
0 |
Branch differences |
| 5:56AM |
0 |
asterisk and snmp |
| 5:34AM |
2 |
Ser vs. DUNDi |
| 5:16AM |
1 |
asterisk not sending ACK after reinvite |
| 2:37AM |
0 |
G729 codec problems |
| |
| Friday May 18 2007 |
| Time | Replies | Subject |
|---|
| 8:36PM |
1 |
xten will not send tones to * and i from sip phone |
| 4:39PM |
1 |
unsubscribe |
| 2:36PM |
1 |
Who picked up with *8? |
| 12:06PM |
0 |
IAX2 sniffer and player |
| 10:42AM |
0 |
4-port ATA |
| 9:50AM |
2 |
zap fallback |
| 9:47AM |
1 |
Fwd: Asterisk console loop? |
| 8:06AM |
3 |
Query about DTMF generate |
| 7:53AM |
2 |
TE212P octastic initialization failure |
| 7:51AM |
1 |
web app to playback recorded phone calls. |
| 7:40AM |
0 |
cpu usage for G.729 codec |
| 7:38AM |
0 |
Re: asterisk-users Digest, Vol 34, Issue 82 |
| 6:54AM |
0 |
mISDN: long delay when making outbound calls |
| 5:51AM |
5 |
Phone losing IP address for a few seconds but doesn't drop call |
| 5:34AM |
1 |
Asterisk vs. Shoretel |
| 5:29AM |
0 |
Asterisk as SIP Provider |
| 5:25AM |
0 |
call-limit=2 , call counter not reset to zero after hangup |
| 5:22AM |
1 |
CallerID not detected by TDM22B |
| 12:44AM |
0 |
EUA voip provider |
| |
| Thursday May 17 2007 |
| Time | Replies | Subject |
|---|
| 11:31PM |
2 |
Call to an arbitrary outbound number by asterisk |
| 10:32PM |
0 |
astertisk for dummies |
| 7:11PM |
1 |
Anyone tested the new Sony Ericsson P1 phones.. |
| 2:24PM |
0 |
Digium HPEC 9.00.003 released |
| 1:42PM |
1 |
OK to have Asterisk and clients behind firewalls? |
| 1:22PM |
0 |
AGI and fork() |
| 1:08PM |
1 |
Multiple lines on Linksys/Sipura phones |
| 12:50PM |
1 |
Cascading Queues |
| 11:37AM |
2 |
Blacklist |
| 11:12AM |
0 |
Compiling DBQuery |
| 10:37AM |
1 |
GUI: Not Found. Move along |
| 10:08AM |
0 |
Call waiting / hook flash on ZAP trunk from SIP phone? |
| 9:59AM |
4 |
FastAGI hangs up channel if server is not available |
| 9:54AM |
0 |
Busy tone with different length tone |
| 9:54AM |
0 |
Re: asterisk setup for church / conference call |
| 9:21AM |
5 |
DUNDi configuration problem |
| 8:27AM |
0 |
emergency call called party control |
| 7:23AM |
1 |
asterisk setup for church / conference call / speaker system integration |
| 4:02AM |
2 |
Asterisk Queue MOH |
| 2:47AM |
2 |
Quadbri Cellular Issue |
| 1:40AM |
4 |
how to define a key to decline incoming call |
| 12:17AM |
0 |
[SPAM]RE: CDR is not written |
| |
| Wednesday May 16 2007 |
| Time | Replies | Subject |
|---|
| 11:54PM |
2 |
Anyone Installed a Digium TE110P or TE120P card in Canada? |
| 11:50PM |
2 |
CDR is not written |
| 10:47PM |
0 |
NO ANSWER, When openser make an oubound SIP call to my asterisk |
| 9:37PM |
1 |
How to remote reboot Grandstream GXP-2000 |
| 8:39PM |
0 |
IAX certificate-based authentication |
| 3:48PM |
2 |
Please help me finding good A-Z provider |
| 3:38PM |
1 |
Digium TE120P and Canada FCC or DOC |
| 3:19PM |
2 |
Get sip response code |
| 3:19PM |
0 |
AGI "record_file" issue... bug? |
| 1:42PM |
1 |
getting call status using Manager API |
| 1:21PM |
1 |
MeetMe and ChannelRedirect |
| 1:06PM |
0 |
Save Key tone to MySQL DB |
| 9:40AM |
5 |
Microsoft's Move Into IP PBX Market |
| 9:23AM |
6 |
SIP Hardware Phone |
| 9:08AM |
1 |
Video Door Phone |
| 7:17AM |
0 |
Passing dialstatus back through an IAX chain .. |
| 7:12AM |
5 |
Which KDE editor to edit Asterisk config files ? |
| 6:21AM |
1 |
SIP INVITE failing and AgentCallBackLogin() |
| 6:17AM |
5 |
GSM Cards for Asterisk (UK) |
| 6:08AM |
0 |
FW: Play a file on a channel from the Manager API |
| 6:04AM |
1 |
G729 Transcoding problems |
| 5:09AM |
0 |
Problem with CDR and DeadAGI |
| 5:07AM |
0 |
Busy tone with the different length tone |
| 4:45AM |
2 |
Asterisk Queue Problem - Automatic Call Distribution |
| 4:34AM |
1 |
Sip client registers then unregisters |
| 4:32AM |
1 |
asterisk manager interface stability |
| 4:03AM |
1 |
WaitExten not responding on key presses |
| 3:35AM |
1 |
Asterisk SRTP certificates |
| 3:08AM |
3 |
voice recording on legacy PBX |
| 2:25AM |
1 |
PRI got event |
| 1:20AM |
1 |
HUDlite Server on Debian Etch/Asterisk 1.4 |
| |
| Tuesday May 15 2007 |
| Time | Replies | Subject |
|---|
| 8:58PM |
1 |
Asterisk is not showing the correct Incomming CallerID |
| 8:23PM |
0 |
PATH_MAX' undeclared here (not in a function) in asterisk! |
| 8:10PM |
0 |
[RTP] PSTN -> Gateway -> Phone |
| 8:01PM |
1 |
Asterisk 1.4.4 reproducibly dumps core on Solaris 10 |
| 12:33PM |
4 |
Feasibility Request |
| 12:04PM |
1 |
Astsee v0.1 released - an Asterisk channel monitor for linux/X windows |
| 11:22AM |
2 |
Zaptel 1.4.2.1 and TE212P |
| 9:47AM |
4 |
Outside lines are just not happening... |
| 8:46AM |
3 |
Trixbox problems |
| 8:39AM |
0 |
IAX2 peer unreachable in one direction - NATproblem? |
| 8:05AM |
1 |
polycom 501 configuration setting |
| 8:03AM |
1 |
finding the sipp soft phone list on the wikey |
| 7:02AM |
3 |
Mr. Spencer Written |
| 5:48AM |
0 |
How to set Name/username to something like 229/john instead of 229/229 |
| 4:45AM |
2 |
Originate and ForkCDR() |
| |
| Monday May 14 2007 |
| Time | Replies | Subject |
|---|
| 11:56PM |
0 |
Req-Installation process for app_dtmftotext.c |
| 10:45PM |
4 |
[*Win32 0.60] Sending call notification by e-mail/web? |
| 10:36PM |
2 |
How to write data to astdb? |
| 10:01PM |
1 |
`PATH_MAX' undeclared here (not in a function) in asterisk! |
| 8:45PM |
3 |
Web based call control |
| 5:59PM |
1 |
Blind Transfer - Who transferred the call? |
| 3:52PM |
0 |
Asterisk Now |
| 2:40PM |
3 |
Proper AGI use with MySQL |
| 12:52PM |
1 |
ast_yyerror - Help |
| 12:48PM |
0 |
Junghanns DuoBRI Card HELP ! |
| 12:34PM |
1 |
Some problems with mysql CDR |
| 11:38AM |
1 |
IAX2 peer unreachable in one direction - NAT problem? |
| 11:09AM |
0 |
Areski CDR |
| 10:07AM |
1 |
DTMF not recognizing * |
| 9:54AM |
1 |
OT (semi) E60 problem |
| 9:11AM |
1 |
Difference between making a call and Originate |
| 9:07AM |
2 |
Simultaneous Capacity |
| 8:39AM |
0 |
How is Context Determined when Transferring a Call? |
| 8:02AM |
0 |
ChanSpy |
| 7:41AM |
1 |
How obtain the slot position when a call is parked? |
| 7:28AM |
3 |
zaptel huge irq problem |
| 7:12AM |
0 |
Codename Pineapple - Chan_sip3 - what's the status? |
| 7:08AM |
5 |
OT ? Number portability, land line to Cell |
| 6:43AM |
0 |
Play a file on a channel from the Manager API |
| 6:37AM |
1 |
Asterisk and unicall + mfcr2 signalling |
| 5:06AM |
2 |
How to bring MoH volume down |
| 4:56AM |
1 |
queue_exec: Unable to join queue |
| 4:47AM |
1 |
function_db_read: DB requires an argument, DB(<family>/<key>) |
| 4:33AM |
1 |
dialplan: execute on hangup |
| 2:12AM |
0 |
quadbri and bristuff : no answer to isdn setup message |
| 12:57AM |
0 |
Decrease group counter without hangup? |
| |
| Sunday May 13 2007 |
| Time | Replies | Subject |
|---|
| 7:43PM |
1 |
Sudden appearance of SIP/2.0 401 Unauthorized |
| 3:39PM |
0 |
Asterisknow b5 - trouble registering at voip provider |
| 6:40AM |
2 |
TC400B load problem |
| 4:58AM |
1 |
Zapateller and IAX2 |
| |
| Saturday May 12 2007 |
| Time | Replies | Subject |
|---|
| 10:00PM |
1 |
Confirmation key to answer -- for a queue |
| 6:35PM |
1 |
Call to Skype network |
| 6:23PM |
2 |
zonedata.c |
| 3:15PM |
1 |
Hardware Echo cancellor Digium Wildcard TE212P |
| 1:11PM |
3 |
Asterisk High-Capacity Stability |
| 10:58AM |
0 |
Recent zaptel versions break CLIP? |
| 8:32AM |
0 |
Preparing music on hold |
| 7:38AM |
0 |
DTMF detection problem on wctdm24xxp |
| 6:29AM |
0 |
1.2.17 -> 1.2.18 asterisk crash |
| 3:27AM |
1 |
Remote extensions not working on provider's wireless Internet connection |
| 2:19AM |
0 |
ser problem |
| |
| Friday May 11 2007 |
| Time | Replies | Subject |
|---|
| 11:49PM |
1 |
Fwd: SER as a Session Border Controller |
| 2:33PM |
0 |
Asterisk crashes |
| 11:36AM |
4 |
Dry Copper Pair |
| 11:18AM |
1 |
Swissvoice IP10s setup |
| 8:12AM |
1 |
Problems with outbound calls through VSP |
| 8:06AM |
4 |
Dealing with 2 SIP providers |
| 7:32AM |
2 |
Strange problem with asterisk |
| 7:04AM |
2 |
Module wctdm24xxp not found - TDM808P on debian |
| 6:38AM |
1 |
Need a RTP/SIP Proxy to be used as SBC (Session Border Controller) |
| 6:18AM |
1 |
'Invalid characters in name' with asterisk-gui |
| 4:27AM |
1 |
A couple of questions for the Mitel gurus (phone-related - not systems) |
| 4:14AM |
1 |
Rapid DTMF missing digits |
| 3:39AM |
2 |
Dundi and unknown remote peers |
| 1:50AM |
0 |
Reminder: Asterisk Users Conference Friday 12:30PM EDT |
| 12:18AM |
1 |
record voice |
| |
| Thursday May 10 2007 |
| Time | Replies | Subject |
|---|
| 10:20PM |
2 |
Any other softPBX like Asterisk? |
| 8:17PM |
1 |
Voice mail volume |
| 2:41PM |
0 |
Correct setup for directing already ringing calls to newly available phones |
| 2:37PM |
1 |
SNOM 360 Rejecting Calls |
| 2:30PM |
3 |
Iaxy clicking |
| 2:01PM |
2 |
socket_process: Received mini frame before first full voice frame |
| 1:54PM |
2 |
The downside of Asterisk and least cost routing... |
| 12:31PM |
1 |
ices low volume |
| 12:23PM |
1 |
Redirecting an existing channel? |
| 11:59AM |
0 |
Asttapi Collect |
| 11:51AM |
1 |
asterisk SIP domain (in LAN or DMZ)? |
| 11:02AM |
1 |
Polycom power over ethernet (PoE) cables for 500/501, 600/601 and 650 sets |
| 10:23AM |
1 |
Polycom headset button blinking |
| 9:36AM |
2 |
force outgoinc callerid |
| 8:49AM |
2 |
CITEL gateway does it work well? |
| 8:21AM |
1 |
Softkey config example for Cisco 7941/7961 |
| 8:10AM |
0 |
AgentCallbackLogin not working with 1.4.4 |
| 7:22AM |
1 |
call transfer to asterik.. asterisk as an end point |
| 5:03AM |
0 |
Anyone tried using a SCCP service provider> |
| 4:47AM |
1 |
Unbridge the two call. |
| 4:26AM |
1 |
Cutted audio or 2/3s blanks on EuroISDN - Asterisk1.4 |
| 4:06AM |
0 |
Asterisk To GoogleTalk Audio Issues |
| 3:58AM |
0 |
Anyone Want To Trade a TE405P for a TE410P? |
| 3:23AM |
0 |
Cutted audio or 2/3s blanks on EuroISDN - Asterisk 1.4 |
| 2:54AM |
2 |
TDM410P |
| 1:59AM |
1 |
module zttranscode: what is it? |
| 12:50AM |
0 |
Asterisk 1.4 and AgentMonitorOutgoing |
| 12:50AM |
1 |
AT530 Telephone |
| 12:38AM |
1 |
ivr and internal voip phones |
| |
| Wednesday May 9 2007 |
| Time | Replies | Subject |
|---|
| 9:14PM |
0 |
Trixbox drops call after running AGI script |
| 5:01PM |
0 |
MINNESOTA: Twin Cities Asterisk Users Group - Saturday May 12th 2007 - 11:30am |
| 4:33PM |
1 |
Is there a possiblity to check in the dialplan whether a SIP user is registred? |
| 2:42PM |
0 |
SPA841 3.1.1(a) firmware file |
| 1:25PM |
1 |
Ericcson analog phone |
| 12:45PM |
5 |
10 FXS - Channel Bank or PCI Card? |
| 11:52AM |
1 |
Boost Polycom IP601 headset volume |
| 10:39AM |
5 |
Mobile Number to Mobile carrier mapping |
| 10:36AM |
6 |
List of telemarketers?? |
| 10:19AM |
1 |
Question about Asterisk 1.4 depoyment. |
| 10:08AM |
0 |
additional volume added to sound on CONSOLE/dsp |
| 9:13AM |
10 |
SIP Problems continue... |
| 8:33AM |
0 |
Re: asterisk-users Digest, Vol 34, Issue 46 |
| 8:01AM |
0 |
using voip software client as public address system. Low volume |
| 7:49AM |
3 |
select menu |
| 6:40AM |
1 |
fax receiving |
| 6:37AM |
3 |
The 'h' extension problem |
| 5:46AM |
1 |
Replaces header |
| 5:45AM |
0 |
Re: asterisk-users Digest, Vol 34, Issue 45 |
| 4:57AM |
0 |
Microsoft CRM & Asterisk |
| 4:07AM |
3 |
The purpose of DUNDi |
| 3:11AM |
5 |
Audio going blank for a few seconds and then comes back. What could be the reason? |
| 12:51AM |
0 |
Bug no. 8680 (billsec is 0 even when the call is answered) in Asterisk 1.4.2 |
| 12:39AM |
1 |
Testing ISDN T1/E1 Bri and Pri etc. |
| |
| Tuesday May 8 2007 |
| Time | Replies | Subject |
|---|
| 10:14PM |
2 |
asterisk with festival facing problem |
| 9:33PM |
0 |
Re: Call waiting tone |
| 8:53PM |
1 |
Aastra phones? |
| 7:02PM |
2 |
asterisk 1.2 and UDP packet numbering on bridged channels (for jitter buffering)? |
| 4:40PM |
1 |
LDAPget or something else? |
| 3:55PM |
2 |
Ringing Volume |
| 3:05PM |
1 |
Problems witch SPA3102. |
| 2:18PM |
0 |
Ericsson dialog 4187 |
| 2:16PM |
0 |
voip-info.org mirrors? |
| 1:47PM |
1 |
Problem when PABX call to Asterisk by Unicall |
| 11:34AM |
3 |
Vista compatibilty in SIP softphones |
| 10:51AM |
0 |
Re: asterisk-users Digest, Vol 34, Issue 42 |
| 10:47AM |
3 |
MYSQL Query --> PAGE |
| 10:25AM |
0 |
random sections lost from call recording |
| 10:19AM |
1 |
Sound files |
| 9:35AM |
0 |
Sangoma cards for sale |
| 8:48AM |
0 |
Asterisk 1.4.2 tanking CPU |
| 8:07AM |
1 |
G729 - Part cut |
| 7:54AM |
1 |
Outbound call through a Single Asterisk Server |
| 7:42AM |
1 |
Remote Phone and Server Behind NAT |
| 6:23AM |
1 |
load modules |
| 5:53AM |
3 |
Sangoma A101 on Freebsd 6.2 |
| 5:09AM |
1 |
asterisk 1.2 from svn ... lock on shutdown |
| 5:07AM |
3 |
Responding to SIP OPTIONS |
| 4:28AM |
2 |
outgoing calls |
| 3:09AM |
0 |
Beronet card - issue? |
| 2:10AM |
1 |
Re: asterisk-users Digest, Vol 34, Issue 39 |
| 12:04AM |
1 |
Bug in voicemail module of Asterisk 1.4.2? |
| |
| Monday May 7 2007 |
| Time | Replies | Subject |
|---|
| 11:54PM |
0 |
Problems with SPA3102 |
| 11:11PM |
1 |
isup-oli or ani2 |
| 11:08PM |
0 |
asterisk with festival facing problem!!!!! |
| 6:53PM |
4 |
iax to iax Reject Connection |
| 4:10PM |
1 |
help MWI setting |
| 10:40AM |
2 |
h323 problem with asterisk 1.2.18 |
| 10:04AM |
6 |
Could two Asterisk servers connect through VPN |
| 8:34AM |
0 |
H323 to H323 bridging ... failed ... also with chan_local |
| 7:49AM |
0 |
ISDN with Billion |
| 7:45AM |
2 |
Asterisk to record CDR in DB Oracle |
| 7:29AM |
2 |
Queues: Play a list of sound file n round-robin at a specific interval |
| 7:16AM |
2 |
Problem with the loading of the cards in Debian |
| 7:07AM |
5 |
RTP Mixer |
| 6:43AM |
0 |
Rhino Analog Cards - Questions |
| 2:52AM |
2 |
Send SIP Re-invite. |
| 2:48AM |
1 |
HPEC audio clipping |
| |
| Sunday May 6 2007 |
| Time | Replies | Subject |
|---|
| 3:42PM |
3 |
Channel Bank |
| 2:42PM |
1 |
Problem with conferences, Vlada, Pancevo |
| 10:59AM |
0 |
Asterisk 1.2.14-BRIstuffed-0.3.0-PRE-1y |
| 8:33AM |
1 |
Buddywatch on Polycom 601 crashes phone |
| 6:42AM |
2 |
Were i make mistake |
| 2:29AM |
2 |
Call waiting tone when calling a busy station? |
| 2:21AM |
1 |
Polycom 601 - To not make noise when there is VM |
| |
| Saturday May 5 2007 |
| Time | Replies | Subject |
|---|
| 6:07PM |
0 |
Channel / Exten Status |
| 5:16PM |
1 |
Dial Plan for Multi-Location & Support Queue |
| 4:38PM |
1 |
${ANSWEREDTIME} Broken on 1.2.13? |
| 4:24PM |
0 |
res_config_pgsql.c in * 1.4.4 |
| 3:11PM |
3 |
I'm looking for solution |
| 2:30PM |
1 |
Asterisk 1.4.4 and Custom Postgres 8.2.4 (checking for PQexec in -lpq... no) |
| 1:02PM |
1 |
ODBC |
| 10:06AM |
3 |
asterisk telemarketer torture sound files |
| 9:07AM |
5 |
TDM400P usada? |
| 6:09AM |
1 |
SIP registration problem |
| 6:04AM |
2 |
Manager API Output |
| 5:49AM |
0 |
TLS support |
| 4:17AM |
2 |
Queue Status |
| |
| Friday May 4 2007 |
| Time | Replies | Subject |
|---|
| 11:39PM |
0 |
Queue Answer |
| 10:38PM |
5 |
Asterisk x legacy pabx |
| 10:31PM |
3 |
Queue and voicemail |
| 5:47PM |
2 |
SLA broken in 1.4.3? |
| 4:08PM |
0 |
Semi-OT: useful things to do with XML browsers inphones |
| 2:28PM |
1 |
Asterisk 1.2 on CentOS 5? |
| 1:28PM |
0 |
Asterisk registration SIP confusion. Can someone explain this? |
| 12:27PM |
0 |
E1 config for chile |
| 11:37AM |
2 |
question about more than one drop file |
| 11:03AM |
2 |
AsteriskNow! |
| 10:55AM |
4 |
zaptel compile error |
| 9:20AM |
1 |
ASA-2007-013: IAX2 users can cause unauthorized data disclosure |
| 8:15AM |
4 |
Headset for Polycom |
| 6:33AM |
0 |
does Not detected HANGUP and DTMF |
| 6:07AM |
0 |
Console flooded by WARNING app_meetme messages |
| 5:45AM |
3 |
app_txfax, app_rxfax |
| 5:04AM |
10 |
Starting Asterisk on Ubuntu 7.04 |
| 4:51AM |
0 |
Error compiling patched pppd for zapras |
| 2:46AM |
2 |
need more knowledge about asterisk |
| 2:34AM |
0 |
Asterisk Users Conference Friday, May 4th at 12:30 PM EDT |
| 1:22AM |
4 |
cpu usuage |
| 12:55AM |
0 |
1.2.x -> 1.4.x upgrade: dialplan block no longer works |
| 12:07AM |
2 |
Asterisk Codec Translation Table |
| |
| Thursday May 3 2007 |
| Time | Replies | Subject |
|---|
| 10:35PM |
1 |
Call interruption |
| 9:48PM |
2 |
OT - robo dialer |
| 9:12PM |
0 |
Called party identification - where to takecalledname? |
| 7:59PM |
1 |
Connections rejected in DUNDi requests |
| 6:17PM |
0 |
[Announce] Web-MeetMe 3.0.2 and 2.2.2 Released |
| 5:37PM |
2 |
Unable to Execute System Command From DialPlan |
| 5:35PM |
3 |
SIP RealTime Friends |
| 5:34PM |
2 |
SIP peer / Maximum retries exceeded on transmission |
| 5:32PM |
1 |
Asterisk 1.4 and Cisco Phones 7940 |
| 3:57PM |
0 |
Restricting membership to a queue? |
| 2:32PM |
0 |
Error messages in console : sipsock_read: SIP MESSAGE JUST IGNORED: |
| 2:19PM |
1 |
VoiceXML + Nuance |
| 2:01PM |
2 |
Balancing interrupts. |
| 1:34PM |
1 |
Double DTMF digits |
| 12:17PM |
3 |
FXO recommendation |
| 11:52AM |
0 |
Strange noise - Polycom |
| 11:37AM |
1 |
Linseed |
| 11:08AM |
0 |
ast_parse_allow_disallow: Cannot disallow unknown format '' |
| 10:02AM |
7 |
Asterisk-Polycom HELLLLPPP!!!! |
| 9:56AM |
1 |
Autologoff |
| 9:05AM |
12 |
IAX Trunk |
| 8:56AM |
2 |
Linksys SPA3012 inbound FXO problems |
| 8:26AM |
2 |
"you have been kicked my this conference" |
| 7:57AM |
0 |
Secondary redirect failed |
| 7:18AM |
2 |
Wildcard TE410P problem |
| 7:18AM |
3 |
Semi-OT: useful things to do with XML browsers in phones |
| 7:14AM |
1 |
Virtual IP Adresses and SIP requests failing... |
| 7:03AM |
2 |
Called party identification - where to take called name? |
| 5:38AM |
2 |
zttranscode crashes server |
| 4:47AM |
1 |
CDR(accountcode) empty in * 1.4.4 (for local chan) |
| 2:51AM |
1 |
Get Asterisk to redirect a SIP INVITE |
| 2:35AM |
1 |
sipura spa9x1 - missed calls not wanted |
| 1:44AM |
1 |
UK - 2 port ISDN2e cards ... |
| 1:05AM |
3 |
0 duration but non-zero billsec in mysql cdr |
| |
| Wednesday May 2 2007 |
| Time | Replies | Subject |
|---|
| 11:15PM |
0 |
Call Limit with multiple SIP extensions |
| 11:09PM |
0 |
rtpmap encoding parameters & the 'unknown codec' problem? |
| 6:26PM |
1 |
IAX and SETLANGUAGE delays |
| 3:12PM |
2 |
OT: USB T1/E1 Interface? |
| 2:55PM |
0 |
Kansas City Asterisk User Group Meeting Announcement |
| 2:33PM |
1 |
SIP Proxy |
| 12:04PM |
6 |
allowing call every 15mins |
| 12:02PM |
2 |
allowing call to my pabx every 15 minutes |
| 11:56AM |
2 |
PRI T1 Problems |
| 10:54AM |
2 |
Large dial plans and variables |
| 10:51AM |
1 |
Asterisk locked up |
| 10:51AM |
4 |
IP Phone Provisioning Tool by voip.com.sg - xml generation |
| 10:48AM |
1 |
Daemontools and holidays macro |
| 10:35AM |
0 |
Call In queue stucks |
| 10:24AM |
1 |
1.4 memory leak? |
| 8:49AM |
2 |
delay in switching between contexts |
| 7:31AM |
2 |
Asterisk-1.4 with agent snmp |
| 7:28AM |
1 |
Returning different SIP Hangup Cause |
| 7:15AM |
0 |
cdr on channel lacks end, not posted |
| 6:33AM |
2 |
SIPGetHeader in Asterisk 1.4 |
| 6:30AM |
1 |
Reinvite after DTMF? |
| 5:15AM |
0 |
ZAP Error: Unable to create channel of type 'Zap' |
| 5:10AM |
1 |
Volume (gain?) on VoIP-only system. |
| 5:05AM |
1 |
MySQL ** DBI connect failed : Too many connections** |
| 4:57AM |
0 |
voice mail format |
| 4:19AM |
0 |
MySQL ** DBI connect failed : Too many connections ** |
| 4:14AM |
2 |
VPN between Asterisk server and phone client |
| 2:35AM |
0 |
Asterisk Integration of XMPP/Jingle |
| 2:13AM |
1 |
Re: RE: Digital Phones (Dean Collins) |
| 2:10AM |
1 |
Re Re: TC400B |
| 1:07AM |
3 |
make iso image |
| |
| Tuesday May 1 2007 |
| Time | Replies | Subject |
|---|
| 11:34PM |
2 |
Runaway MOH/mp3123 process? |
| 9:01PM |
3 |
using Playback() to play a random sound file |
| 3:16PM |
1 |
TC400B |
| 3:00PM |
10 |
Applet? |
| 3:00PM |
10 |
Digital Phones |
| 1:56PM |
0 |
Re: [asterisk-dev] SRTP implementation |
| 1:54PM |
3 |
Display Caller ID of called party |
| 1:47PM |
3 |
Stanaphone business ok? |
| 1:37PM |
1 |
chan_sip seems to be hanging |
| 12:37PM |
1 |
T1 interface |
| 11:23AM |
2 |
MYSQL application in dial plan |
| 10:51AM |
5 |
OT: Capture Asterisk traffic |
| 9:46AM |
4 |
is dundi worth pursuing in this situation? |
| 9:44AM |
2 |
Channel stuck with call pri flag |
| 9:17AM |
3 |
How do I do this in Asterisk? |
| 8:47AM |
3 |
Delay in Dial() |
| 8:33AM |
0 |
Email to HP Product Suggestions - Seamless Transparent Fax Gateway |
| 8:31AM |
3 |
How many users can be supported simultaneously? |
| 8:26AM |
1 |
Cisco 7940 no outgoing audio |
| 6:27AM |
0 |
RE: Voicemail on Different Server (MySQL Replication split thread) |
| 6:24AM |
2 |
Change Codec |
| 5:59AM |
1 |
restrictions on meetme with agi background |
| 5:10AM |
1 |
chan_local |
| 5:02AM |
2 |
Autoattendant press 1 collides with extension numbers... |
| 2:21AM |
0 |
Re: Anyone having trouble with claling US Domesticon Sellvoip? |
| 2:16AM |
0 |
Two Context Residing On The Same Server |
| 1:10AM |
4 |
did we all get spammed by TechnoCo ? |