| Monday February 28 2011 |
| Time | Replies | Subject |
|---|
| 5:19PM |
0 |
Asterisk 1.8.3 Now Available |
| 5:18PM |
0 |
Asterisk 1.6.2.17 Now Available |
| 5:17PM |
0 |
Asterisk 1.4.40 Now Available |
| 4:39PM |
2 |
CEL and PGSQL |
| 3:27PM |
5 |
Failover Routing |
| 1:40PM |
2 |
Asterisk 1.8.3-rc3 and one way audio |
| 10:33AM |
2 |
asterisk security....again |
| 10:25AM |
0 |
Obi110 as gateway to PSTN? |
| 10:24AM |
7 |
Two Asterisk machines for redundancy |
| 2:12AM |
5 |
Using voice modem as poor man's FXO in Asterisk 1.8 |
| |
| Sunday February 27 2011 |
| Time | Replies | Subject |
|---|
| 12:32PM |
1 |
[Dahdi 2.4.0] Flash() hangs up |
| |
| Saturday February 26 2011 |
| Time | Replies | Subject |
|---|
| 10:33PM |
2 |
Need to buy the Digium card, to confirm |
| 4:16PM |
1 |
SRTP Error Message |
| 9:08AM |
1 |
Detect DTMF tone during call? |
| |
| Friday February 25 2011 |
| Time | Replies | Subject |
|---|
| 10:49PM |
0 |
T1 channel audio control |
| 9:47PM |
0 |
Asterisk 1.2 zap hangup issue |
| 7:30PM |
4 |
Asterisk/Skype |
| 5:37PM |
0 |
PRI B-Channel restarting itself continually |
| 5:04PM |
5 |
[OT] Yealink IP Phones |
| 4:16PM |
2 |
1.8.2.4: SIP dialogs not killed? |
| 12:40PM |
1 |
Handle in dialplan user disconnection |
| |
| Thursday February 24 2011 |
| Time | Replies | Subject |
|---|
| 10:27PM |
1 |
missing argument on AGI |
| 9:05PM |
2 |
Recieve_Fax caused crash 1.8.2.3 |
| 8:47PM |
1 |
Using a Virtual IP Line |
| 8:31PM |
1 |
extensions.lua with luasql.mysql. |
| 6:41PM |
2 |
Paging with Polycom 3.3.x |
| 4:32PM |
1 |
Debug Dropped Audio |
| 2:58PM |
0 |
"Asterisk" caller ID |
| 2:51PM |
4 |
Google Voice outbound Caller ID broken |
| 1:42PM |
1 |
RTP (voice) issue. STUN server |
| 1:24PM |
1 |
Registration failed though configured. |
| 12:36PM |
2 |
[1.4] Still can't get it to call back |
| 11:38AM |
2 |
Carrying context from one server to another? |
| 10:56AM |
2 |
[1.4.39.2] Simple AGI doesn't reply |
| 10:15AM |
1 |
Unknown calls |
| 7:16AM |
2 |
DIAL through Specific number in PRI |
| 1:10AM |
0 |
One way dialing over a SIP trunk |
| |
| Wednesday February 23 2011 |
| Time | Replies | Subject |
|---|
| 7:56PM |
0 |
SIP friend name |
| 4:10PM |
2 |
REFER and dialplan broken (as documented in chan_sip.c on line 11951) |
| 3:31PM |
4 |
secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1 |
| 2:14PM |
0 |
Adhearsion 1.0.1 Released |
| 10:39AM |
4 |
AMI FullyBooted issue |
| 10:17AM |
2 |
extend the timout on ringing for pri or sip |
| |
| Tuesday February 22 2011 |
| Time | Replies | Subject |
|---|
| 9:34PM |
2 |
Multiple public address to one Asterisk server behind NAT? |
| 6:21PM |
1 |
calls between iax and sip |
| 2:57PM |
0 |
Weird Inbound Problem. |
| 1:02PM |
3 |
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available |
| 10:39AM |
1 |
[1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe |
| 10:06AM |
0 |
AddQueueMember and stateinterface question |
| 3:11AM |
1 |
NVFaxDetect causing segfault |
| |
| Monday February 21 2011 |
| Time | Replies | Subject |
|---|
| 10:52PM |
1 |
[Dahdi 2.4.0] DAHDI_CHANCONFIG failed on channel 1 |
| 10:46PM |
3 |
Problem installing FXS module in old digium 4 channel tdm card |
| 9:52PM |
0 |
(no subject) |
| 9:45PM |
4 |
AST-2011-002: Multiple array overflow and crash vulnerabilities in UDPTL code |
| 8:28PM |
0 |
Erroneous email from JIRA |
| 7:07PM |
1 |
Free calls to the US provider recommendation |
| 6:44PM |
1 |
T1 PRI shows yellow/red alarm |
| 6:33PM |
0 |
SIP METHOD BYE |
| 4:51PM |
0 |
Difference mohsuggest & mohinterpret |
| 3:47PM |
1 |
Dialplan execution stops on app call even with TryExec (Am I missing something simple?) |
| 11:13AM |
2 |
calls are not going thru e1 line |
| |
| Sunday February 20 2011 |
| Time | Replies | Subject |
|---|
| 7:05AM |
1 |
MEMBERINTERFACE and MEMBERNAME questions |
| 5:03AM |
0 |
My new blog http://cciev.ciscovoicetech.com/ |
| 2:39AM |
0 |
AstLinux 0.7.6 Released |
| |
| Saturday February 19 2011 |
| Time | Replies | Subject |
|---|
| 9:15PM |
2 |
First go at a stock 1.8 install -- where's DAHDI? |
| 9:52AM |
1 |
[1.4] "show channels" in extensions.conf? |
| 1:30AM |
4 |
AGI script dies after receivefax |
| 12:00AM |
1 |
Problem in dialing out |
| |
| Friday February 18 2011 |
| Time | Replies | Subject |
|---|
| 8:59PM |
2 |
no progress indication |
| 5:56PM |
2 |
cmd MySQL |
| 3:36PM |
1 |
[1.4/AGI] CHANNEL STATUS never "down & available"? |
| 3:11PM |
2 |
Meet me recording |
| 11:36AM |
3 |
Assigning an extension to a roaming phone |
| 11:32AM |
3 |
lua -asterisk manual |
| 11:29AM |
2 |
pbx_ael.so: undefined symbol: ast_compile_ael2 |
| 9:18AM |
1 |
Asterisk with TE 121 DADHI incoming calls fail |
| 8:31AM |
2 |
Dial(Local/...) vs. Goto()? |
| 8:23AM |
3 |
FAX on PRI to MFCR2 |
| 8:17AM |
2 |
DTMF and Snom |
| 7:16AM |
2 |
Trunk grouping |
| 1:33AM |
0 |
Voice mail forwarding enhancement |
| 12:31AM |
1 |
Dial() function |
| 12:26AM |
4 |
Newbie´s question about Asterisk... |
| |
| Thursday February 17 2011 |
| Time | Replies | Subject |
|---|
| 6:27PM |
2 |
Polycom Do Not Disturb button and asterisk hints |
| 5:11PM |
1 |
Pickup from an specific exten |
| 4:46PM |
1 |
Setting two E1 cards |
| 4:15PM |
0 |
PRI "wanrouter status" shows disconnected - system problem or Telco? |
| 2:54PM |
1 |
Got SIP response 400 "Bad Request" back from |
| 1:28PM |
1 |
Realtime MySQL - Asterisk 1.8.2 |
| 11:13AM |
0 |
Friday 18 Feb at 12 Noon EST: SylkServer and Blink |
| 10:34AM |
0 |
Samsung smt-i3100 |
| 7:15AM |
3 |
Asterisk Using as a SIP Client |
| 1:47AM |
0 |
Google 10% |
| |
| Wednesday February 16 2011 |
| Time | Replies | Subject |
|---|
| 8:32PM |
1 |
No ring tone on inbound call - but channel connects fine |
| 7:51PM |
5 |
Polycom IP335 |
| 6:24PM |
1 |
Asterisk on a USB with persistence |
| 6:00PM |
1 |
pipe audio stream to external application |
| 4:40PM |
1 |
trunk not working if I register a phone at the same IP as the trunk peer's IP |
| 1:41PM |
2 |
Play one audio file to the called part before the Dial() command? |
| 11:47AM |
2 |
function Echo() doesn't work |
| 11:41AM |
1 |
how to diable echo cancellation for sip? |
| 10:45AM |
1 |
Detect #,* DTMF in dialplan |
| 10:32AM |
1 |
Cisco 7945G phone with asterisk |
| 10:13AM |
1 |
Barge in. |
| 10:05AM |
1 |
How to know Caller's last position in Queue? |
| 8:53AM |
0 |
Regarding error in asterisk 1.6.2.16.... |
| 5:49AM |
4 |
Connect Asterisk to a cell phone |
| 12:39AM |
7 |
DTMF not detected, time out |
| |
| Tuesday February 15 2011 |
| Time | Replies | Subject |
|---|
| 8:47PM |
2 |
Dialplan end of pattern matching question |
| 8:12PM |
1 |
Aastra phones cannot transfer calls? |
| 8:05PM |
0 |
pstack debug asterisk |
| 7:15PM |
4 |
Voicemail email attachment as MP3, with tags containing sender name, number, message number |
| 6:39PM |
2 |
Paging a message. How? |
| 6:15PM |
2 |
Realtime and Local Channel Crash Problem 1.8.3-rc2 |
| 5:06PM |
1 |
Lua extensions are not working on asterisk 1.8.2.3 |
| 5:06PM |
2 |
Adjusting Rx and Tx gains |
| 3:20PM |
0 |
asterisk 1.8.2 freez |
| 2:15PM |
1 |
outbound call leg CALLID |
| 1:58PM |
0 |
weird problem with Vega 100 |
| 12:49PM |
0 |
changing logo of 7905 |
| 8:51AM |
3 |
further action after caller in a queue hangs up |
| 4:00AM |
6 |
Fax Woes |
| 3:39AM |
5 |
uptime |
| 1:47AM |
1 |
trunks and phones registered from the same IP |
| |
| Monday February 14 2011 |
| Time | Replies | Subject |
|---|
| 11:16PM |
1 |
Asterisk Call File using Local Channel not passing Variable back to Dialplan |
| 9:36PM |
5 |
Hide the plain text password |
| 6:46PM |
1 |
unregistered trunks and registered phones coming from the same IP |
| 4:06PM |
1 |
SIP session timers just on one specific channel |
| 2:16PM |
3 |
issue with some numbers |
| 1:40PM |
2 |
Cisco 7960 & asterisk 1.8.22 ringlist.dat error |
| 12:11PM |
1 |
Possible dumb question: new kernel, new DAHDI? |
| 10:29AM |
1 |
Problems with realtime sip |
| 6:10AM |
1 |
IP ban list by country |
| |
| Sunday February 13 2011 |
| Time | Replies | Subject |
|---|
| 2:52PM |
1 |
[modules.conf] Modules still loaded after "noload" |
| 5:59AM |
1 |
Call Files, Variable passing |
| 4:53AM |
1 |
Fax for Asterisk SIP-TDM |
| |
| Saturday February 12 2011 |
| Time | Replies | Subject |
|---|
| 11:06PM |
1 |
Transfer Device Data |
| 4:23PM |
3 |
Using files .call or AMI |
| 12:31PM |
11 |
SIP Hardphone that works well with asterisk |
| 9:14AM |
1 |
[Zaptel] "numberplan-local" context from nowhere? |
| 7:42AM |
1 |
Variables losing their value???? |
| 7:03AM |
0 |
what are QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY veriables? |
| |
| Friday February 11 2011 |
| Time | Replies | Subject |
|---|
| 10:37PM |
6 |
On-Hold Music |
| 8:57PM |
2 |
Asterisk compile option DAHDI SPANS |
| 5:26PM |
2 |
dialplan announcements |
| 5:13PM |
0 |
AstMail |
| 5:02PM |
1 |
Asterisk 1.8.3 BLF stopped working |
| 11:48AM |
1 |
Realtime queues not playing prompts |
| 9:37AM |
3 |
Asterisk 1.8.3 |
| 6:47AM |
2 |
sangoma wanpipe install error |
| 12:58AM |
3 |
meetme conference & playback of random sound file |
| |
| Thursday February 10 2011 |
| Time | Replies | Subject |
|---|
| 11:50PM |
2 |
Gtalk/Jabber Issue |
| 10:24PM |
0 |
res_pgsql re-connect on db failure? |
| 9:11PM |
0 |
"intercom" SIP header being ignored by Kirk wireless handsets |
| 6:04PM |
0 |
Busy Detection on Analog Lines |
| 12:13PM |
3 |
CDR with unix time. |
| 11:08AM |
1 |
Early audio SIP sequence order question |
| 4:49AM |
2 |
zaptel/dahdi settings for singtel E1 line |
| 3:08AM |
2 |
Unable to make outgoing calls with Internode |
| |
| Wednesday February 9 2011 |
| Time | Replies | Subject |
|---|
| 11:56PM |
0 |
Error loading module ��Է�Vi.so |
| 7:31PM |
1 |
AEL Eswitches |
| 7:30PM |
1 |
Defining what an extension should do after the Dial() command returns busy. |
| 3:21PM |
2 |
queue called by agi doesn't re-enter the script |
| 11:56AM |
0 |
ashishchauhan07oct@gmail.com sent you a movie ticket redeemable at more than 200 nation wide theatre chains |
| 10:36AM |
2 |
SIP MESSAGE outside calls - state of the art? |
| 9:45AM |
0 |
Reliably getting sip extension name from channel variables |
| 4:11AM |
1 |
dial option 'g' not working |
| 3:40AM |
0 |
Manual Call Transfer (Perl, Asterisk::AGI, MySQL) |
| |
| Tuesday February 8 2011 |
| Time | Replies | Subject |
|---|
| 9:36PM |
0 |
Microsoft Speech Server/UCMA Integration |
| 8:11PM |
1 |
echo when calling to the pstn |
| 7:18PM |
0 |
Manual Call Transfer // Perl // Asterisk::AGI // MySQL |
| 6:56PM |
0 |
Scheduled Maintenance: wiki.asterisk.org and code.asterisk.org |
| 6:08PM |
0 |
Looking for actual user opinions on Telephony card |
| 5:02PM |
1 |
Inbound SIP calls work, just not when making calls between extensions. |
| 5:01PM |
0 |
Asterisk CallCompletion dialplan |
| 4:50PM |
1 |
terrible MeetMe sound with 1.6.2.9 |
| 4:07PM |
3 |
fail-over server |
| 3:34PM |
0 |
SIP registration |
| 2:52PM |
1 |
forward calls by the ports |
| 2:36PM |
1 |
Set variable on Call Answer |
| 1:30PM |
2 |
${HANGUPCAUSE} in CDR |
| 12:45PM |
2 |
Call files error |
| 11:09AM |
2 |
Call Recording audio file quality query |
| |
| Monday February 7 2011 |
| Time | Replies | Subject |
|---|
| 6:56PM |
1 |
multiple inbound calls from same sip trunk |
| 6:27PM |
1 |
OT: SwitchVox Mailing List? |
| 5:56PM |
1 |
Codec negotiation |
| 5:49PM |
1 |
remote bridging |
| 4:45PM |
1 |
IAX channel name incorrect - Found in 1.2 still happens in 1.6 |
| 4:13PM |
1 |
downgrade libpri |
| 3:38PM |
1 |
About maxlen parameter in queues |
| 1:42PM |
1 |
Error: Unable to create channel of type 'SIP' |
| |
| Sunday February 6 2011 |
| Time | Replies | Subject |
|---|
| 9:51AM |
0 |
secure sccp |
| |
| Saturday February 5 2011 |
| Time | Replies | Subject |
|---|
| 9:39PM |
1 |
Any voice changer applications for Asterisk? |
| 11:07AM |
11 |
Callback through extensions.conf? |
| 9:24AM |
2 |
Zaptel slow to detect remote hangup |
| |
| Friday February 4 2011 |
| Time | Replies | Subject |
|---|
| 10:25PM |
1 |
SoftHangup on asterisk 1.8.2.3 |
| 1:02PM |
3 |
MP3 Crashing Asterisk |
| 10:31AM |
0 |
problems with voicemail and centos 5 |
| 9:53AM |
2 |
voice quality measurement using dahdi_monitor |
| 5:43AM |
2 |
Email alerts for trunks (peers) |
| 5:41AM |
3 |
PRI voice optimization |
| 2:43AM |
1 |
standalone NOTIFY message handling for Asterisk |
| |
| Thursday February 3 2011 |
| Time | Replies | Subject |
|---|
| 7:45PM |
8 |
Question about EuroBRI final 2 digits |
| 6:30PM |
2 |
T.38 negotiation error |
| 3:53PM |
2 |
Queues and Agent penalty - how to go to second best agent when the first does not answer |
| 3:48PM |
1 |
MeetMe and admin users |
| 10:41AM |
1 |
sip trunk balancing |
| 10:18AM |
1 |
[newbie] Conference call |
| 9:44AM |
1 |
Radius Based Accounting for Asterisk |
| 1:18AM |
0 |
Regarding bob-invite-alice xml scenario |
| |
| Wednesday February 2 2011 |
| Time | Replies | Subject |
|---|
| 11:42PM |
1 |
Problems using Background within a macro on V 1.4 |
| 8:44PM |
5 |
Regarding asterisk |
| 8:14PM |
1 |
asterisk18 rpm issues |
| 6:26PM |
1 |
Outgoing agent´s calls |
| 4:50PM |
0 |
SIP Originate on 1.8.X |
| 5:43AM |
1 |
AGI script exits non-zero when running system command |
| 4:14AM |
0 |
regarding sip.conf and extensions.conf |
| 3:34AM |
8 |
how to get Current Calls details |
| |
| Tuesday February 1 2011 |
| Time | Replies | Subject |
|---|
| 10:22PM |
3 |
Asterisk Performance |
| 5:34PM |
1 |
Upgrade and recompilation |
| 4:37PM |
1 |
How to use Monitor() in Python AGI |
| 4:34PM |
1 |
How to load new musiconhold classes ? |
| 4:22PM |
0 |
Connecting to Cisco Iad2430 to Asterisk |
| 4:05PM |
2 |
Musiconhold priority |
| 12:22PM |
0 |
How to Change The Caller Position in Queue |
| 9:02AM |
1 |
Playback in uplink and recording in downlink |
| 3:09AM |
0 |
regarding error in asterisk |