Saturday December 31 2011 |
Time | Replies | Subject |
10:19AM |
1 |
Outbound Dialer, Agent Login and Logout |
|
Friday December 30 2011 |
Time | Replies | Subject |
11:11PM |
4 |
High verbose set at console effects the logger file "Full" - Why is that? |
2:40PM |
1 |
wait_for_answer: Unable to write frametype: 2 on Asterisk *CLI> |
2:00PM |
0 |
Asterisk 1.8.8.1 Now Available |
11:17AM |
0 |
Best practise for operator |
10:07AM |
1 |
Asterisk 1.4.42 NOTIFY replies ignore NAT setting |
8:44AM |
2 |
Problem installing B410P BRI card for asterisk |
3:13AM |
0 |
Asterisk Video Playback - MP4 ad 3GP files |
1:39AM |
2 |
Block Specific Number on Inbound |
|
Thursday December 29 2011 |
Time | Replies | Subject |
11:51PM |
0 |
Asterisk Registrar / Trunk |
9:55PM |
0 |
can't set up tcp sip - sip connection : digest <s> problem |
9:16PM |
0 |
How to create SIP INVITE with different To: Header field than Request-Line URI |
3:03PM |
1 |
Asterisk fail2ban filters - show us yours |
2:21PM |
0 |
Softphones |
11:49AM |
1 |
IAX2 woes |
9:37AM |
0 |
Help_In Voicemail , vedio play but voice is not here out. |
9:12AM |
0 |
Function TESTTIME example [SOLVED] |
4:16AM |
6 |
Client - registers but unreachable |
4:07AM |
2 |
Interesting attack tonight & fail2ban them |
12:01AM |
0 |
func_odbc not returning whole smalldatetime MS Sql field. |
|
Wednesday December 28 2011 |
Time | Replies | Subject |
9:32PM |
1 |
MFCR2 Long distance calls not connected |
8:57PM |
1 |
Question on hung channel |
8:33PM |
2 |
1.6 and 1.8 |
8:16PM |
2 |
Monitor Command Records separate channales |
7:59PM |
1 |
Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet |
10:47AM |
0 |
Chan_ss7 clustering config with single point |
10:14AM |
0 |
Direct media path on Avaya IPOFFICE and Asterisk with H323 Trunk |
8:25AM |
2 |
DTMF Testing software to test IVR system |
12:50AM |
0 |
sendvoicemail=yes not quite working SOLVED |
12:46AM |
1 |
cdr call time |
|
Tuesday December 27 2011 |
Time | Replies | Subject |
7:43PM |
1 |
Call going into s-extension |
7:34PM |
1 |
maximizing sound quality in 10.0 |
12:48PM |
1 |
how to used SIPp for sip load testing |
9:26AM |
0 |
read dtmf digits on connected calls |
6:54AM |
3 |
how to stop hacking of my server |
3:05AM |
1 |
odd "secret" problem |
|
Monday December 26 2011 |
Time | Replies | Subject |
6:19PM |
0 |
Working on web based IVR Designer for asterisk and Freeswitch |
4:27PM |
0 |
Debugging call quality issues in Asterisk |
2:55PM |
5 |
how to listen on different sip port for a device? |
11:35AM |
1 |
Function TESTTIME example |
11:03AM |
0 |
Which language and tool to write diaplans ? |
10:52AM |
0 |
AEL2: How to get rid of "does not end with a return; I will insert one" warnings |
5:18AM |
4 |
Not able to play wav files in asterisk |
|
Sunday December 25 2011 |
Time | Replies | Subject |
10:41AM |
1 |
Is Asterisk 1.4 compatible with 1.8.7 ? |
|
Saturday December 24 2011 |
Time | Replies | Subject |
5:36AM |
0 |
Vicidial license |
|
Friday December 23 2011 |
Time | Replies | Subject |
3:57PM |
0 |
AEL2: How get rid of "expression has operators, but no variables" warnings |
3:40PM |
2 |
Codec warnings after upgrade to 1.8 |
12:54PM |
1 |
execute command just after Dial() |
10:12AM |
0 |
Any one using VICIDIAL? |
9:33AM |
1 |
GotoIfTime days query |
9:03AM |
1 |
Dahdi not installed and application's details is missing in Asterisk |
7:48AM |
0 |
2011 The Year in Review - where were you when... |
|
Thursday December 22 2011 |
Time | Replies | Subject |
7:49PM |
0 |
How to read doc in /var/lib/asterisk/documentation |
5:33PM |
3 |
Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf |
4:34PM |
1 |
Properly Escaping Quotes and Commas |
10:31AM |
1 |
Using shell script output into phoneprov.conf's custom variables |
8:51AM |
0 |
Asterisk Developer |
7:43AM |
1 |
Number of Calls |
12:14AM |
3 |
dahdi_tool missing |
|
Wednesday December 21 2011 |
Time | Replies | Subject |
8:57PM |
1 |
Diagnosing call hangups |
3:03PM |
3 |
Suppress -- Remote UNIX connection message |
1:41PM |
1 |
Asterisk call file size calculation |
11:48AM |
1 |
Why **CONGESTION** not *****NOANSWER****** ? |
10:36AM |
0 |
Asterisk 1.8 - RTPAUDIOQOSLOSS does not show audio quality |
8:50AM |
1 |
dahdi: Unknown symbol kasprintf |
7:13AM |
1 |
OT - Which switch to play with LLDP-MED |
1:03AM |
1 |
sendvoicemail=yes not quite working |
|
Tuesday December 20 2011 |
Time | Replies | Subject |
7:39PM |
3 |
GOIP GSM to SIP Gateway? |
4:21PM |
1 |
OOH323 config file |
1:47PM |
3 |
Help_video call not run |
12:56PM |
0 |
Asterisk Sip Media Call Type |
12:04PM |
1 |
File Convert |
6:15AM |
1 |
PITCH_SHIFT() |
12:44AM |
4 |
Limit # of inbound calls on SIP trunk |
12:38AM |
3 |
Use different local IP for each SIP trunk |
|
Monday December 19 2011 |
Time | Replies | Subject |
9:08PM |
1 |
Asterisk 1.6.2.22 Now Available |
8:06PM |
5 |
India Telecom regulations |
8:02PM |
0 |
A lot of 603 Declined Error form iCall - Are they going down or is it just bad service? |
6:24PM |
0 |
Which Dahdi/Libpri version are you using ? |
6:13PM |
1 |
Dahdi 2.5.0.2 - Strange Warning |
6:13PM |
0 |
ChanSpy in whisper mode - low quality audio |
2:47PM |
0 |
Sending Fax Dialplan with Retry Attempt |
11:14AM |
3 |
AMI and Dialplan |
|
Sunday December 18 2011 |
Time | Replies | Subject |
7:19PM |
1 |
asterisk and heartbeat |
3:07PM |
0 |
Called peer IP |
7:42AM |
1 |
Asterisk 1.8.7.2 now sends rport always |
5:26AM |
3 |
How to monitor SIP Trunk on production server |
|
Saturday December 17 2011 |
Time | Replies | Subject |
11:55PM |
1 |
No rtpmap codec info in 200 OK |
5:56PM |
1 |
asterisk-users Digest, Vol 89, Issue 32 |
9:11AM |
1 |
AstLinux 1.0.0 release |
12:38AM |
4 |
Set Caller Number in E1 PRI ISDN Lines |
|
Friday December 16 2011 |
Time | Replies | Subject |
9:02PM |
1 |
Dialing problem with Polycom phones after SIP update |
8:06PM |
1 |
FreePBX not updating configs on 1.8 RPM install |
5:39PM |
0 |
Problems with DAHDI Channels |
5:32PM |
1 |
fromstring in voicemail.conf |
5:00PM |
2 |
Errors on RBS T1 |
3:57PM |
0 |
ConfBridge 10 How can I playback a soundfile to an existing conference |
2:52PM |
0 |
asterisk-users Digest, Vol 89, Issue 29 |
2:38PM |
1 |
CDR END TIME in correct in 1.8+ |
12:06PM |
3 |
ODBC problem - static realtime file not loading |
10:29AM |
2 |
SIP Trunk |
4:50AM |
0 |
Contexts and Extensions |
2:31AM |
1 |
followme forking/parallel dialing breaks when 1 sip device unreachable |
2:03AM |
1 |
Asterisk console suddenly extremely verbose... |
1:03AM |
2 |
Which device auto-registered an extension? |
|
Thursday December 15 2011 |
Time | Replies | Subject |
9:58PM |
0 |
Asterisk 10.0.0 Is Released! |
9:55PM |
0 |
Asterisk 1.8.8.0 Now Available |
9:21PM |
0 |
Struggling with Extensions in Realtime |
8:10PM |
2 |
Call pickup on Asterisk 1.8 and Polycom IP550s? |
6:33PM |
3 |
Best PBX for Call Centers? |
5:46PM |
1 |
Asterisk log format |
4:28PM |
3 |
Play audio file for both Caller and Callee in a call |
3:41PM |
4 |
Partner Keys on Innovaphone |
2:47PM |
0 |
app_swift tts module - new home. |
2:05PM |
3 |
Digium TE205P leds flash red on startup |
11:59AM |
0 |
how to get the Record_ID |
11:51AM |
1 |
Wrong call information on B leg |
|
Wednesday December 14 2011 |
Time | Replies | Subject |
12:56PM |
2 |
Asterisk 1.4.x segfaulting daily |
9:18AM |
3 |
A few (simple?) questions |
7:28AM |
1 |
get start-time of all active calls |
|
Tuesday December 13 2011 |
Time | Replies | Subject |
9:53PM |
0 |
[hint state][BLF] Asterisk 1.8.7 does not send RINGING notifications, even with notifyringing=yes |
9:44PM |
0 |
Login agents on asterisk startup according to hints state |
6:00PM |
1 |
AEL x LUA |
|
Monday December 12 2011 |
Time | Replies | Subject |
11:40PM |
1 |
SIP MESSAGE outside calls - state of the art? |
10:31PM |
0 |
TLS bug in asterisk? |
10:11PM |
0 |
Asterisk Configuration GUI Question |
7:55PM |
1 |
ATA with TCP/TLS support? |
7:42PM |
0 |
Looking for partners to develop Asterisk Call Centre Applications - A call to investors and programmers |
6:44PM |
2 |
How to query Microsoft SQL server for caller-id source |
4:55PM |
0 |
How to see initiall dialled extension in CDR records ? |
4:36PM |
6 |
VoiceMail and IMAP |
2:51PM |
1 |
Help needed for chan_ss7 for Digium device |
2:23PM |
3 |
MySql Custom CDR issues |
11:16AM |
0 |
Asterisks Statistics (Albert) |
10:21AM |
1 |
Which port should be open for asterisk communication |
9:21AM |
0 |
How to count ongoing calls from the dialplan |
5:36AM |
0 |
How is everything doing? |
3:59AM |
2 |
What version to upgrade to...? |
12:22AM |
2 |
Multiple route failover zaps registration |
|
Sunday December 11 2011 |
Time | Replies | Subject |
7:32PM |
0 |
Chan_sip How to store Register Call ID? |
|
Saturday December 10 2011 |
Time | Replies | Subject |
8:53PM |
0 |
Asterisk 1.8 and measuring Post Dial Delay |
|
Friday December 9 2011 |
Time | Replies | Subject |
5:16PM |
0 |
Asterisk 10.0.0-rc3 Now Available |
4:51PM |
2 |
asterisk-users Digest, Vol 89, Issue 13 |
4:48PM |
0 |
Scheduled Maintenance for Asterisk Project community services |
3:52PM |
1 |
Issue with dahdi 2.5.0 and Digium HA8-B400M |
1:27PM |
1 |
Mixing asterisk.conf, asterisk.ael and asterisk realtime |
6:10AM |
1 |
Preparing to store vm in database |
5:55AM |
1 |
Trying to send customer mwi updates |
|
Thursday December 8 2011 |
Time | Replies | Subject |
11:34PM |
0 |
Asterisk 1.4.43, 1.6.2.21, and 1.8.7.2 Now Available (Security Release) |
10:58PM |
0 |
Fax client for Windows |
10:48PM |
0 |
AST-2011-014: Remote crash possibility with SIP and the “automon� feature enabled |
10:47PM |
2 |
AST-2011-013: Possible remote enumeration of SIP endpoints with differing NAT settings |
6:21PM |
0 |
Sip.conf and extensions.conf configuration for Exchange 2010 U.M. |
2:26PM |
1 |
random digits dialing during call |
2:06PM |
1 |
libpri / ISDN feature ECT (explicit call transfer) |
12:46PM |
2 |
[OT]: Require suggestions - GSM Gateway <-> Asterisk |
8:26AM |
1 |
Issues with dahdi show status output (and check IRQs) |
7:46AM |
1 |
How to make app_meetme enable |
5:42AM |
4 |
Confrence call is not make |
|
Wednesday December 7 2011 |
Time | Replies | Subject |
9:55PM |
4 |
ISDN PRI configuration |
8:31PM |
1 |
Help! Logs filling up with errors! |
8:10PM |
1 |
ChanSpy() and Spygroup |
5:56PM |
1 |
Realtime Registration |
3:16PM |
1 |
redirect a ringing phone |
2:24PM |
1 |
SS7 + T1 |
12:15PM |
1 |
ss7 installation and configuration |
4:26AM |
0 |
IRC Client |
|
Tuesday December 6 2011 |
Time | Replies | Subject |
5:58PM |
0 |
Proper sip.conf and extensions.conf for Exchange 2010 U.M. |
4:57PM |
1 |
rasterisk not knowing config path? |
4:34PM |
0 |
Asterisk 1.4 - Help/Doc for Park() application [SOLVED] |
10:11AM |
0 |
Data provided by pri show spans and dahdi_tool do not match |
10:08AM |
2 |
How to use Hints in asterisk |
9:24AM |
0 |
help please (D 300 JCT) |
9:03AM |
2 |
Talk detection in meetme |
|
Monday December 5 2011 |
Time | Replies | Subject |
9:13AM |
1 |
Asterisk 1.4 - Help/Doc for Park() application |
7:28AM |
1 |
How to count available parking slots from diaplan |
|
Sunday December 4 2011 |
Time | Replies | Subject |
6:30PM |
0 |
PRI cards D 300 JCT E1 |
|
Saturday December 3 2011 |
Time | Replies | Subject |
8:36PM |
4 |
Simple Generic IVR to get us up an running Quick |
1:10PM |
1 |
Hint'ing with XMPP? |
5:59AM |
2 |
google voice calling dial plan question. |
|
Friday December 2 2011 |
Time | Replies | Subject |
11:48PM |
0 |
skype connect & early media |
11:28PM |
0 |
Max channel analyser from asteriskcdrdb? |
11:27PM |
0 |
DHCP Option 43 and pfSense + Asterisk |
11:24PM |
2 |
How can I decipher password in SIP Packet? |
11:13PM |
1 |
Where to download sample video file for Asterisk 1.8x playback? |
9:44PM |
0 |
IAX - An informative question |
5:37PM |
1 |
CSipSimple audio issue with DAHDI/IAX2 calls |
10:18AM |
2 |
DIALSTATUS Values |
|
Thursday December 1 2011 |
Time | Replies | Subject |
9:48PM |
1 |
Locally bridging channels when using SRTP? |
8:35PM |
1 |
Can't get off Europe/Bucharest timezone |
12:44PM |
3 |
Issue with Polycom SPIP650 and its sidecar [SOLVED] |
11:57AM |
5 |
Populate CDR issues |
12:42AM |
3 |
AGI script that uses google's text to speech engine |