| Sunday February 28 2010 |
| Time | Replies | Subject |
|---|
| 6:29PM |
0 |
Compatible IP Phones for Asterisk |
| 4:58PM |
2 |
AUTHENTICATE Command customized prompts - Work around |
| 4:56PM |
0 |
ISDN Options |
| 1:17PM |
0 |
Lower kernel version for mISDN |
| 10:48AM |
2 |
Premicell solutions? |
| 5:43AM |
2 |
Server response time |
| |
| Saturday February 27 2010 |
| Time | Replies | Subject |
|---|
| 8:42PM |
1 |
No RTP from asterisk? |
| 8:08PM |
2 |
Conference Calling |
| 7:51PM |
1 |
Asterisk AUTHENTICATE Command |
| 12:58PM |
0 |
Increasing the dahdi chunk size with Sangoma cards |
| 12:43PM |
0 |
New patch for app_queue to show all call attempts, even failing ones |
| 1:29AM |
0 |
"Unexpected message received" when receiving Fax |
| |
| Friday February 26 2010 |
| Time | Replies | Subject |
|---|
| 11:29PM |
2 |
Fun with virtual asterisks ... |
| 8:48PM |
0 |
Qeuee/Agent Question |
| 6:11PM |
1 |
hi |
| 4:30PM |
0 |
qsigchannelmapping parameter |
| 4:24PM |
2 |
Asterisk RPM's |
| 4:01PM |
1 |
SPA941 WMI not lighting up when natted |
| 1:35PM |
3 |
: PSTN calls |
| 1:34PM |
2 |
Web operator/softphone with integration features |
| 10:21AM |
0 |
Problem with BLF's |
| 10:06AM |
0 |
record a user call while playing a background music |
| 8:17AM |
0 |
realtime modules not load ? |
| 5:26AM |
1 |
Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 1:27AM |
2 |
How to tell if asterisk is handling media or not? |
| 12:41AM |
0 |
How can we pickup a call that is not going to a real extension? |
| |
| Thursday February 25 2010 |
| Time | Replies | Subject |
|---|
| 10:40PM |
1 |
Asterisk 1.6.0.17 PBX with two interfaces does not routes RTP packets - SIP Conf Problem likely |
| 10:39PM |
0 |
Asterisk 1.6.0.25, 1.6.1.17, and 1.6.2.5 Now Available |
| 10:38PM |
0 |
DTMF timing - first # keypress not registering |
| 10:28PM |
0 |
AST-2010-003: Invalid parsing of ACL rules can compromise security |
| 8:51PM |
1 |
Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid |
| 7:54PM |
0 |
Asterisk Crashs due to some Sip messages |
| 6:45PM |
2 |
Morse Code |
| 6:08PM |
2 |
Followme broken |
| 5:19PM |
3 |
MeetMe() and dahdi_dummy on an embedded system |
| 4:50PM |
0 |
Intermittent DAHDI issue with a PRI line causing asterisk to crash! |
| 4:19PM |
1 |
Deadlock while using MGCP on Asterisk |
| 3:59PM |
1 |
AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? |
| 3:30PM |
2 |
Problems installing dahdi : kernel sources |
| 3:21PM |
0 |
IAX peers one way voice |
| 2:35PM |
3 |
X-Lite won't register |
| 10:17AM |
2 |
Redirect call based on CLI??? |
| 9:26AM |
1 |
Getting: Can't fix up channel from 5 to 7 because 7 is already in use, and pri_dchannel: Answer requested on channel 0/7 not in use on span 1 |
| 8:45AM |
0 |
CDR duration/billsec |
| 7:29AM |
1 |
curl and ssl certificate |
| 6:38AM |
1 |
Asterisk n-way DTMF detection |
| 2:16AM |
2 |
Do i need install Dahdi or libpri ? |
| |
| Wednesday February 24 2010 |
| Time | Replies | Subject |
|---|
| 11:51PM |
2 |
Problems with Linksys IP Phone SPA 942 |
| 11:30PM |
2 |
audio glitches in conference |
| 9:49PM |
0 |
Question |
| 9:05PM |
1 |
Encrypted calls between mobile gsm and isdn (asterisk) |
| 6:23PM |
2 |
AMD: HANGUP |
| 4:35PM |
4 |
identify the costumer |
| 3:44PM |
3 |
Re-INVITE on BYE |
| 3:42PM |
2 |
Problems in Asterisk Real Time (Urgent help ) |
| 2:20PM |
1 |
subject: 1.4 vs 1.6 |
| 1:55PM |
0 |
Manager Logged off |
| 11:41AM |
0 |
Wrong MOH |
| 11:37AM |
0 |
Looping over AstDB |
| 6:04AM |
0 |
IAX devices not registering after upgrade to asterisk |
| 3:35AM |
1 |
Macros, GoSub & StackPop |
| |
| Tuesday February 23 2010 |
| Time | Replies | Subject |
|---|
| 7:28PM |
2 |
IAX devices not registering after upgrade to |
| 7:23PM |
1 |
Which H.323 to use in Ast 1.6 |
| 5:41PM |
2 |
SIP provider registration attempts |
| 1:22PM |
3 |
directrtp with SIP + H.323 |
| 11:59AM |
2 |
Calls per second limit in manager |
| 11:30AM |
1 |
Codec translation in Asterisk |
| 10:39AM |
1 |
IAX devices not registering after upgrade to asterisk 1.4.29 |
| 8:18AM |
2 |
Running safe_asterisk |
| |
| Monday February 22 2010 |
| Time | Replies | Subject |
|---|
| 10:59PM |
2 |
Load balance outgoing calls |
| 9:13PM |
2 |
SIP Disconnects from Network - Asterisk Does not hangup |
| 8:31PM |
0 |
Avaya with Asterisk |
| 7:20PM |
1 |
Problem w/ MoH |
| 7:13PM |
2 |
Problems with SIP realtime |
| 7:02PM |
2 |
Open source or low-budget recommendation for call-center software |
| 6:57PM |
0 |
init.d error when installing trunk |
| 6:02PM |
1 |
Multiple instances of Asterisk on the same host... |
| 4:18PM |
8 |
[OT] Asterisk 1.6 and DECT Phones |
| 3:31PM |
1 |
Caller ID question |
| 1:20PM |
4 |
4 PCIe cards in one asterisk server |
| 11:52AM |
1 |
Denying call transfer to certain extensions |
| 11:50AM |
1 |
AMI Originate differences between 1.4 and 1.6.1 |
| 11:16AM |
1 |
Sending back the BYE code gotten on second leg |
| 8:28AM |
1 |
TE410P Spans offline/red after power down/restart |
| 7:36AM |
1 |
Does Playback will answer the call? |
| 6:23AM |
1 |
Audio to remote AGI server |
| 2:40AM |
2 |
Free iPhone Asterisk Function and Application Reference |
| |
| Sunday February 21 2010 |
| Time | Replies | Subject |
|---|
| 7:55PM |
4 |
HFC-S card |
| 6:54PM |
0 |
Trouble with externalIVR socket connection |
| 5:22PM |
1 |
Dahdi & Congestion status |
| 3:14PM |
2 |
add Reason header on hangup |
| |
| Saturday February 20 2010 |
| Time | Replies | Subject |
|---|
| 10:22PM |
0 |
IPKall NOT coming on Asterisk |
| 9:28PM |
0 |
Moh help needed |
| 9:16PM |
2 |
Sending a hook flash to a DAHDI channel |
| 6:02PM |
2 |
Slightly OT: Has SILK codec gotten anywhere? |
| 4:49PM |
0 |
outgoing callerid problem |
| 1:14PM |
1 |
Fax, T38 and NAT |
| 2:43AM |
0 |
Hung channel problem with 1.4.26.2 |
| 12:35AM |
1 |
Error redirecting an incoming call of a SIP provider to a local extension |
| |
| Friday February 19 2010 |
| Time | Replies | Subject |
|---|
| 9:54PM |
0 |
asterisk sudden restart - 1.4.18.1 - Resolved after upgrade to 1.4.22.1 |
| 8:05PM |
3 |
string length in dialplan |
| 4:42PM |
2 |
Virtual machine timing (KVM) |
| 4:21PM |
3 |
splitting sip.conf to two files |
| 4:01PM |
1 |
transcoding with TC400P |
| 3:16PM |
1 |
mISDN (HFC-S) and TDM400P - isac xdu no tx_busy |
| 2:21PM |
0 |
AMI + device status (patch 0016732) + remote control |
| 1:57PM |
1 |
Volume of Playback() application |
| 6:51AM |
3 |
Dial Plan configuration in asterisk |
| 1:19AM |
1 |
directmedia/canreinvite/native bridging question |
| |
| Thursday February 18 2010 |
| Time | Replies | Subject |
|---|
| 11:51PM |
0 |
Asterisk 1.2.40, 1.4.29.1, 1.6.0.24, 1.6.1.16, and 1.6.2.4 Now Available |
| 11:46PM |
0 |
AST-2010-002: Dialplan injection vulnerability |
| 8:57PM |
0 |
How to transfer call using function T |
| 8:47PM |
1 |
BRI vs. PRI? |
| 8:05PM |
5 |
OpenVPN/SNOM 820: a review. |
| 6:46PM |
1 |
Realtime extensions |
| 5:52PM |
2 |
Product offerings from DIDforSale |
| 1:51PM |
0 |
ISDN phone not ringing. ISDN PBX not answering?! |
| 7:45AM |
0 |
Feb 19th @12 noon EST: Voxeo's Tropo |
| 7:40AM |
3 |
Asterisk t38modem Fax gateway evaluation |
| 1:49AM |
0 |
Asterisk cluster in Active/Active mode |
| 1:40AM |
2 |
how asterisk knows which context forward the call to? |
| 12:53AM |
2 |
Registering of Asterisk against a SIP provider |
| |
| Wednesday February 17 2010 |
| Time | Replies | Subject |
|---|
| 9:23PM |
0 |
Asterisk answers inbound call during ringing |
| 9:15PM |
2 |
Static IP |
| 9:06PM |
3 |
Setting up only one caller at a time |
| 9:03PM |
1 |
Access to header field: event |
| 8:47PM |
1 |
queue.conf - Set(MONITOR_FILENAME=${}) |
| 8:21PM |
1 |
One-Way Audio after Hold |
| 7:50PM |
1 |
some newbie questions about gcc |
| 7:17PM |
0 |
sending call to correct context |
| 6:39PM |
0 |
Asterisk in Active/Active mode |
| 6:12PM |
3 |
sip.conf - sort order, does it matter |
| 4:50PM |
1 |
1.6.1 Voicemail users.conf |
| 4:11PM |
2 |
asterisk dahdi fax problem |
| 3:00PM |
3 |
chan_local and Originate |
| 2:57PM |
1 |
Help with Dictate app |
| 11:37AM |
4 |
Unrecognized prilocaldialplan NPI modifier |
| 10:13AM |
2 |
how to remove asterisk from this string X-Asterisk-HangupCauseCode |
| 8:13AM |
1 |
call parking |
| 3:39AM |
1 |
Ideasip |
| |
| Tuesday February 16 2010 |
| Time | Replies | Subject |
|---|
| 10:50PM |
6 |
Asterisk listens on all NICs |
| 7:51PM |
1 |
chan_sip.c: Disconnecting call 'SIP/302-b720dd78' for lack of RTP activity in 301 seconds |
| 6:47PM |
0 |
Voicemail IMAP storage enhancement |
| 6:05PM |
1 |
Cannot built kmod-dahdi-linux for PAE kvariant from SRPM |
| 5:48PM |
1 |
call is not going to wrong "context" |
| 5:36PM |
1 |
rawplayer in asterisk 1.0.0 |
| 5:29PM |
1 |
CODECS: Best practice question: Avoid transcode when calling out? |
| 4:39PM |
1 |
How does holdtime get calculated for queues |
| 4:17PM |
0 |
Users of the SMS application? |
| 3:50PM |
1 |
Stupid question: Why Cmd Dial and Queue haven't same options? |
| 3:04PM |
2 |
Issue with trying to dial two different servers at the same time. |
| 11:55AM |
2 |
call transfer |
| 9:40AM |
1 |
Empty SIP Packet |
| 7:54AM |
2 |
OT- Using TR-069 |
| |
| Monday February 15 2010 |
| Time | Replies | Subject |
|---|
| 7:31PM |
1 |
video voicemail |
| 4:05PM |
3 |
Maximum call handling capacity on single server |
| 10:55AM |
1 |
strange asterisk behaviour on XEN |
| 8:26AM |
0 |
Zaptel/DAHDI error's on PRI |
| 6:54AM |
1 |
signal problem |
| 1:25AM |
2 |
Capture |
| 12:35AM |
2 |
insecure=invite - not working for different dial plan |
| |
| Sunday February 14 2010 |
| Time | Replies | Subject |
|---|
| 9:59PM |
1 |
how to have disconnect signals enabled in line |
| 4:42PM |
3 |
Asterisk Redundancy |
| 4:34PM |
0 |
transmit_silence_during_record |
| 1:37PM |
0 |
voicemail problem |
| 12:43PM |
1 |
issues.asterisk.org |
| 12:24PM |
1 |
Cisco 7940: showing FWD in display. |
| 9:33AM |
3 |
Line DC |
| 6:56AM |
0 |
Domain Authentication - Caller ID Failed to authenticate |
| 3:50AM |
2 |
agi debug in Asterisk 1.6? |
| |
| Saturday February 13 2010 |
| Time | Replies | Subject |
|---|
| 11:04PM |
4 |
Important security alert: update your dialplans now! |
| 8:59PM |
1 |
how to create voicemail |
| 3:57PM |
2 |
1.6.x SIP allow incoming calls based on from ip address? |
| 7:24AM |
3 |
extension not found |
| 3:09AM |
2 |
Call Pickup with 1.6.2.1 and Snom |
| |
| Friday February 12 2010 |
| Time | Replies | Subject |
|---|
| 9:29PM |
4 |
Robotic sound sometimes |
| 9:23PM |
2 |
PAP2 |
| 8:57PM |
1 |
how to allow some extenstions to call outside and some extensions cant call outside |
| 8:54PM |
1 |
PRI Problems with 1.6.0.10 |
| 8:49PM |
1 |
dropping line (s) for 911 |
| 8:32PM |
3 |
how to allow some extensions to make call outside and some extensions cant call outside |
| 5:24PM |
7 |
Asterisk Cepstral TTS |
| 3:52PM |
2 |
T.38 with reinvite |
| 2:25PM |
1 |
parked calls |
| 7:55AM |
0 |
[Fwd: SIP tunnel] |
| 5:41AM |
0 |
g722 IP Phone |
| 1:39AM |
1 |
Wierdness in AGI file |
| |
| Thursday February 11 2010 |
| Time | Replies | Subject |
|---|
| 9:23PM |
0 |
nortle BCM450 & SIP-Trunking |
| 8:57PM |
0 |
Asterisk 1.2.39 Now Available |
| 5:38PM |
0 |
IP Kall One-Way Audio |
| 2:33PM |
0 |
dnd sorta working |
| 1:37PM |
13 |
SIP tunnel |
| 12:19PM |
0 |
Asterisk ignores BYE messages |
| 10:21AM |
2 |
SIP RTP ports not released when channel is hung up |
| 7:20AM |
2 |
app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 12:36AM |
0 |
Sending "Progress" during dialing |
| 12:16AM |
0 |
wellgate 3804A with frying |
| |
| Wednesday February 10 2010 |
| Time | Replies | Subject |
|---|
| 10:37PM |
3 |
How to avoid AGI script is canceled if caller HangUp |
| 7:32PM |
1 |
problems with 1.6 |
| 5:24PM |
1 |
1.6.2 : global vars not read/set after #include w/ globals |
| 4:55PM |
1 |
asterisk sudden restart - 1.4.18.1 |
| 4:08PM |
0 |
EAGI delay |
| 3:39PM |
6 |
IP Phone recommendation |
| 2:47PM |
1 |
Optimization of call from server 1 to 2 and then back to 1 |
| 2:11PM |
1 |
Muted calls occasionally dropping after 30 seconds |
| 1:36PM |
1 |
problems with creating a call |
| 11:57AM |
1 |
Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory |
| 11:53AM |
1 |
Nat Issue - is this Draytek || Asterisk? |
| 9:47AM |
0 |
PMS (SDMR, ...) support in Asterisk |
| 8:58AM |
2 |
asterisk and mysql connection |
| 7:21AM |
0 |
VUC Friday Feb 12th: HD Communications Summit |
| 6:52AM |
1 |
forward incomming line to modem |
| 4:02AM |
1 |
billing based on local access number |
| |
| Tuesday February 9 2010 |
| Time | Replies | Subject |
|---|
| 11:47PM |
0 |
Callerid problem in 1.6.2.2 |
| 9:33PM |
2 |
Security Logging |
| 6:48PM |
3 |
ways of initiating a call |
| 6:19PM |
0 |
? chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received |
| 5:45PM |
1 |
test |
| 5:32PM |
0 |
ISDN users: 1.6.x users, I need some testing done please, regarding Overlap Receiving |
| 12:43PM |
0 |
asterisk-users Digest, Vol 67, Issue 20 Re: Asterisk going down |
| 12:42PM |
1 |
Not able to receive fax |
| 10:26AM |
1 |
Lua status in asterisk. |
| 10:01AM |
3 |
Get Talk Time |
| 4:42AM |
9 |
VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 |
| 12:40AM |
2 |
E71 |
| |
| Monday February 8 2010 |
| Time | Replies | Subject |
|---|
| 9:23PM |
1 |
Strange Problem |
| 8:45PM |
0 |
moving a bridged call to a conference room |
| 5:04PM |
2 |
IVR Demo / Create file / Move file / Demo all |
| 5:00PM |
1 |
billsec is set to duration if call is not answered |
| 4:09PM |
1 |
Asterisk how install speex support |
| 12:54PM |
3 |
High codec translation times on x64 |
| 12:52PM |
6 |
GSM Gateway |
| 11:29AM |
2 |
conferencing without DAHDI |
| 9:27AM |
1 |
queue with strategy=linear |
| 9:24AM |
0 |
Help with iax.conf {tesco|freshtel} 1.6 |
| 9:16AM |
0 |
Asterisk going down (Josiah Bryan) |
| 6:19AM |
0 |
Call doesn't disconnect in SIP |
| 5:10AM |
4 |
Not able to compile asterisk, zaptel, libpri in /usr/src |
| 3:14AM |
0 |
Asterisk going dow |
| 3:06AM |
0 |
originate, local channel and failure extension |
| 2:54AM |
2 |
How to run a remote PHP script and still have access to audio stream? |
| 12:18AM |
2 |
syntax |
| |
| Sunday February 7 2010 |
| Time | Replies | Subject |
|---|
| 12:13PM |
0 |
Pickup the call ringing at SIP Phone but was transferred from Zap channel |
| |
| Saturday February 6 2010 |
| Time | Replies | Subject |
|---|
| 11:19PM |
3 |
A2Billing and other prepaid Billing like ASTCC, who is better? |
| 12:30PM |
1 |
CONNECTEDLINE |
| 11:18AM |
3 |
Asterisk 1.4.26.2 died after 80 days uptime |
| 7:31AM |
1 |
TOS bits, DSCP, Asterisk & Polycom |
| 6:48AM |
1 |
Website Down ? |
| 2:54AM |
6 |
Dial script |
| 1:37AM |
0 |
Recording Calls |
| |
| Friday February 5 2010 |
| Time | Replies | Subject |
|---|
| 11:07PM |
0 |
strange issue with iptables + Asterisk |
| 10:59PM |
3 |
Asterisk going down |
| 9:50PM |
6 |
large scale paging |
| 9:23PM |
0 |
Sipgate.co.uk on Asterisk 1.6.2.2 |
| 4:55PM |
4 |
2 Asterisk Boxes, Single Voicemail |
| 11:40AM |
6 |
Still on spandsp/app_fax and T.38 |
| 8:38AM |
1 |
Ongoing calls interface |
| 4:46AM |
0 |
Do the Linksys Sipura series have a known problem with Asterisk? |
| 2:05AM |
8 |
Losing local SIP phones when internet goes down? |
| 12:39AM |
3 |
Know what would be killer? |
| |
| Thursday February 4 2010 |
| Time | Replies | Subject |
|---|
| 7:11PM |
0 |
pickup the call: No target channel found |
| 6:25PM |
0 |
Audiocodes MP-114 MWI Stutter Tone |
| 4:08PM |
3 |
OpenVPN on phones? |
| 3:42PM |
3 |
Gotoif Question |
| 1:48PM |
2 |
SS7 and Asterisk |
| 12:41PM |
0 |
OT: VUC Feb 5th @ 12 Noon Open VPN |
| 11:09AM |
0 |
OT - MWI, Polycom/kirk and Gigaset handsets |
| 10:59AM |
0 |
pickup with gxp2000 does not work.. |
| 10:00AM |
6 |
Running a script after Dial() ? |
| 9:55AM |
1 |
Aastra 50-limit blf |
| 12:34AM |
1 |
1.6.2.1: DTMF trouble with PSTN |
| |
| Wednesday February 3 2010 |
| Time | Replies | Subject |
|---|
| 10:24PM |
1 |
aastra 9480i dtmf ? |
| 9:00PM |
0 |
Routing inbound call to correct sip trunk |
| 8:43PM |
1 |
500 Internal Server Error on Cisco 7940 after INVITE |
| 7:17PM |
3 |
calling into server with cell and originate a call |
| 3:24PM |
1 |
ast_cdr_setvar: Attempt to set the 'src' read-only variable! |
| 2:52PM |
1 |
Asterisk core sounds in English by June Wallack |
| 10:57AM |
0 |
CDR and Queue Reporting windows application looking for Beta testers! |
| 10:26AM |
0 |
Pri HDLC aborts and choppy audio when dialling into pri, caused by BIOS option ["CPU enhanced halt" c1e] |
| 7:19AM |
5 |
Intel Atom based Asterisk server? |
| 12:43AM |
1 |
CDR / billsec / originate / local chan |
| 12:09AM |
0 |
asterisk video support and IPTV |
| |
| Tuesday February 2 2010 |
| Time | Replies | Subject |
|---|
| 11:56PM |
0 |
Queue problem, ringing agents. |
| 10:40PM |
0 |
AST-2010-001: T.38 Remote Crash Vulnerability |
| 10:28PM |
0 |
Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 Released |
| 10:05PM |
1 |
# as dial key - chan_dahdi |
| 8:02PM |
2 |
Semi-Transfer |
| 7:58PM |
1 |
codec conversion |
| 7:20PM |
3 |
sip realtime md5secret |
| 5:25PM |
1 |
Codec coversion |
| 5:10PM |
2 |
Two Extensions showing as Busy |
| 1:53PM |
0 |
Issue when reloading |
| 1:00PM |
4 |
Asterisk 1.6.1.13 and T.38 faxing |
| 12:40PM |
3 |
Asterisk 1.6.2 ? |
| 10:20AM |
3 |
uri tel: instead of sip:accepted ? |
| 7:53AM |
0 |
Realtime queue strategy issue |
| 5:41AM |
6 |
Smallest possible Asterisk VM |
| 1:22AM |
1 |
Use a BLF for monitoring |
| 1:22AM |
1 |
(no subject) |
| |
| Monday February 1 2010 |
| Time | Replies | Subject |
|---|
| 8:33PM |
0 |
One way audio with Grandstream HT503 |
| 1:55PM |
1 |
Problems with recordings of call using Monitor |
| 12:54PM |
1 |
NVFaxDetect |
| 12:52PM |
0 |
Stuck logger rotation |
| 12:22PM |
0 |
mysterious rippled sound with IAX |
| 9:25AM |
0 |
Asterisk for productive Calling Card System |
| 8:51AM |
0 |
call count per peer |
| 7:38AM |
2 |
connect problem unless when verbose |
| 5:44AM |
0 |
FCT for 3G Video calls |
| 2:09AM |
1 |
Odd error mssage on DAHDI lines |
| 12:12AM |
1 |
app_directory broken in 1.6 |