Sunday March 31 2013 |
Time | Replies | Subject |
8:53PM |
0 |
asterisk-users Digest, Vol 104, Issue 53 |
5:45PM |
0 |
SRTP woes |
5:11PM |
1 |
Feature request: Need to INVITE to peer with other domain without peer domain addition |
6:54AM |
1 |
ISDN- E1 PRI module in network side signaling |
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Saturday March 30 2013 |
Time | Replies | Subject |
12:05AM |
1 |
IPv6 |
|
Friday March 29 2013 |
Time | Replies | Subject |
4:32PM |
0 |
Getting Unknown Error while configuring Asterisk with Linux HA |
1:02PM |
5 |
"sip set debug on" output to file only (not to console) |
10:02AM |
1 |
Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved] |
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Thursday March 28 2013 |
Time | Replies | Subject |
11:56PM |
2 |
Asterisk 11 -CDR values changed in hangup handler not saved ? |
7:55PM |
3 |
To queue or not to queue... |
7:36PM |
0 |
Asterisk 11.3.0 Now Available |
7:35PM |
0 |
Asterisk 1.8.21.0 Now Available |
1:44PM |
3 |
dahdi-channels.conf vs. chan_dahdi.conf |
|
Wednesday March 27 2013 |
Time | Replies | Subject |
10:50PM |
0 |
chan_mobile: FXS |
8:55PM |
0 |
AST-2013-003: Username disclosure in SIP channel driver |
8:55PM |
0 |
AST-2013-002: Denial of Service in HTTP server |
8:55PM |
0 |
AST-2013-001: Buffer Overflow Exploit Through SIP SDP Header |
3:55PM |
0 |
Asterisk 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones, 11.2.2 Now Available (Security Release) |
6:18AM |
1 |
Pattern matching repeating digits |
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Tuesday March 26 2013 |
Time | Replies | Subject |
4:57PM |
2 |
WARNING[28151] from CLI |
2:13PM |
1 |
Dedicated LCR Solutions |
10:51AM |
1 |
Pointer to debug "Got SETUP with duplicate call ptr . Dropping call." |
10:22AM |
1 |
Fundemental changes to CDR within single asterisk family |
9:28AM |
2 |
rtcachefriends and rtautoclear on change password |
8:03AM |
0 |
Asterisk 11, hangup-handlers, Local channels and channel originate [SOLVED] |
|
Monday March 25 2013 |
Time | Replies | Subject |
10:17PM |
1 |
Asterisk 11, hangup-handlers, Local channels and channel originate |
8:55PM |
0 |
Using type=friend a mistake? |
10:44AM |
7 |
question about zapata.conf |
|
Saturday March 23 2013 |
Time | Replies | Subject |
1:33PM |
5 |
Optimizing Asterisk Environment |
12:15AM |
0 |
Self Contained Least Cost Routing Solution |
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Friday March 22 2013 |
Time | Replies | Subject |
6:38PM |
0 |
Planned maintenance for community services on March 25, 2013 |
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Thursday March 21 2013 |
Time | Replies | Subject |
7:01PM |
2 |
Allow/Disallow |
6:48PM |
1 |
Cisco SPA 5xx/3xx/9xx phones don't respond to SIPAddHeader(Call-Info: answer-after=0) |
6:45PM |
2 |
Howto create variable from the name of another one and get content of it |
4:47PM |
4 |
Asterisk 1.8 and dual stack support |
2:56PM |
2 |
Need help about round-robin |
7:31AM |
9 |
Asterisk disconnecting SIP Calls after 15 Minutes |
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Wednesday March 20 2013 |
Time | Replies | Subject |
7:34PM |
0 |
Nnjjjjmm |
4:58PM |
1 |
Looking for a reporter for SQLite3 with Lighttpd and PHP |
12:57PM |
2 |
xmpp priority setting and GoogleVoice |
12:24PM |
3 |
Cisco 7942G and SEPMAC.cnf.xml and the registration |
10:48AM |
1 |
AGI return codes |
12:16AM |
0 |
Peer-to-Peer |
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Tuesday March 19 2013 |
Time | Replies | Subject |
5:43PM |
0 |
Asterisk SIP Refer Transfers |
2:15AM |
3 |
SIP account registration fails after upgrade to 1.8 |
|
Monday March 18 2013 |
Time | Replies | Subject |
6:30PM |
6 |
Diagnosing call problem |
5:21PM |
0 |
Asterisk as Text To Speech server |
|
Sunday March 17 2013 |
Time | Replies | Subject |
4:14PM |
11 |
Need help understanding CDR |
|
Friday March 15 2013 |
Time | Replies | Subject |
8:10PM |
2 |
app_rtsp.c ported to Asterisk 11.x |
2:21PM |
1 |
Asterisk uses 3 seconds to send ACK after OK |
12:11PM |
1 |
Asterisk does not persist callgroup and pickgroup configuration. |
7:20AM |
0 |
No subject |
7:20AM |
0 |
No subject |
7:20AM |
0 |
No subject |
7:20AM |
0 |
No subject |
7:20AM |
0 |
No subject |
7:20AM |
0 |
No subject |
7:20AM |
0 |
No subject |
7:20AM |
0 |
No subject |
7:20AM |
0 |
No subject |
7:20AM |
0 |
No subject |
7:20AM |
0 |
No subject |
4:56AM |
2 |
Call blocking issue |
2:41AM |
2 |
Disagreements between codec_siren14 and Polycom sources |
|
Thursday March 14 2013 |
Time | Replies | Subject |
10:04PM |
3 |
ERROR: Unknown signalling method ss7 |
9:27PM |
0 |
DAHDI-Linux and DAHDI-Tools 2.6.3-rc1 Now Available |
5:29PM |
1 |
AGI |
12:23PM |
2 |
PRI Called Party Number Info |
4:24AM |
2 |
blacklist caller ID |
12:57AM |
0 |
MWI issue |
|
Wednesday March 13 2013 |
Time | Replies | Subject |
4:50PM |
1 |
Asterisk 1.8 as text to speech server |
3:38PM |
0 |
res_calendar.so and res_calendar_caldav.so |
|
Tuesday March 12 2013 |
Time | Replies | Subject |
4:30PM |
0 |
Calls getting "stuck open" |
3:54PM |
1 |
How does Asterisk handle ACK's? |
3:24PM |
2 |
trunking trixbox - panasonic |
12:44PM |
0 |
BLF LED Pattern |
|
Monday March 11 2013 |
Time | Replies | Subject |
11:07PM |
2 |
Serviced Office operator panel |
9:51PM |
1 |
Asterisk 11 & GoogleVoice/Motif |
6:34PM |
0 |
compiling imap |
11:53AM |
1 |
Laptop error |
|
Sunday March 10 2013 |
Time | Replies | Subject |
4:18PM |
2 |
chan_mobile |
3:37PM |
1 |
Asterisk authentication on LDAP (SSHA and SHA passwords) |
2:04PM |
2 |
IPv6 and IPv4 binding address on a server with 2 network cards |
12:26PM |
0 |
asterisk video streaming |
2:04AM |
1 |
Register Free Opensips/Asterisk Integration |
|
Saturday March 9 2013 |
Time | Replies | Subject |
12:00PM |
0 |
About Zitter Control |
11:54AM |
1 |
Digium Wildcard TDM800P not working with DAHDI |
12:32AM |
7 |
Sending SMS from asterisk |
|
Friday March 8 2013 |
Time | Replies | Subject |
9:52PM |
0 |
DAHDI-Linux and DAHDI-Tools 2.6.2 Now Available |
9:21PM |
11 |
digium card and virualbox |
8:40PM |
1 |
Directmedia Question |
2:38PM |
4 |
Asterisk 11.2.1 / IAX / Problems with incoming calls |
1:58PM |
0 |
CDR-Logging with leading "0" in src field clid and/or src |
11:49AM |
2 |
asterisk sizing for play and dtmf detection |
1:16AM |
1 |
VOIP PRI Gateways |
1:12AM |
1 |
Polycom SPIP config |
|
Thursday March 7 2013 |
Time | Replies | Subject |
9:35PM |
0 |
Ring back issue with asterisk 1.8.18.0 |
6:05PM |
7 |
Extension cant pickup calls but can transfer. |
3:21PM |
2 |
Recording with MixMonitor and AGI |
2:47PM |
2 |
11.3: how to hang up on google voice |
10:12AM |
2 |
Asterisk 1.6 + Cisco AS5300 |
4:30AM |
4 |
asterisk with 1000 extensions |
|
Wednesday March 6 2013 |
Time | Replies | Subject |
6:35PM |
1 |
Asterisk crashed |
4:48PM |
2 |
AGI Script |
12:00PM |
2 |
Error to install Asteriskā |
1:56AM |
2 |
Change RX Signalling Bits in Dahdi drivers |
|
Tuesday March 5 2013 |
Time | Replies | Subject |
8:51PM |
2 |
Redirect incoming call to SIP trunk. |
7:50PM |
1 |
fail2ban filter issue |
7:32PM |
1 |
What would cause a drop between two asterisk systems? |
1:20PM |
2 |
Error to install Asterisk |
1:10PM |
1 |
multiple sipusers tables |
6:31AM |
2 |
red alarm on span - do channels in the group automatically get skipped over? |
|
Monday March 4 2013 |
Time | Replies | Subject |
10:13PM |
0 |
Planned maintenance for community services on March 4, 2013 |
3:37PM |
2 |
Asterisk 11 - How to trim the number of modules to minimum ? |
12:27PM |
3 |
Exiting the queue doesn't work |
|
Sunday March 3 2013 |
Time | Replies | Subject |
6:23PM |
1 |
How to configure NT/ptmp with Dahdi and BRI ? (Olivier) |
5:41PM |
1 |
asterisk 11 - No "pri set debug off" |
4:37PM |
0 |
How to configure NT/ptmp with Dahdi and BRI ? |
|
Friday March 1 2013 |
Time | Replies | Subject |
3:03PM |
0 |
Weird SIP Issue |