Monday September 29 2014 |
Time | Replies | Subject |
1:52PM |
1 |
Dahdi problem with dahdi_genconf |
|
Sunday September 28 2014 |
Time | Replies | Subject |
3:51PM |
2 |
how to make voip client cannot use same username? |
8:58AM |
1 |
Intercom Telephone Feature |
5:54AM |
2 |
How to append the recording file. |
|
Saturday September 27 2014 |
Time | Replies | Subject |
3:28PM |
2 |
can PJSIP_MEDIA_OFFER work like SIP_CODEC? |
|
Friday September 26 2014 |
Time | Replies | Subject |
4:15PM |
1 |
Ports leak |
|
Thursday September 25 2014 |
Time | Replies | Subject |
2:24PM |
1 |
Realtime ERROR |
11:32AM |
0 |
weird behaviour of sip history and sip debug |
|
Wednesday September 24 2014 |
Time | Replies | Subject |
10:14PM |
0 |
Asterisk 1.8 - Security Fix Only Notice |
9:34PM |
0 |
Asterisk 12.6.0 Now Available |
9:34PM |
0 |
Asterisk 11.13.0 Now Available |
9:34PM |
0 |
Asterisk 1.8.31.0 Now Available |
11:06AM |
0 |
Identifying frequency tone in Asterisk |
3:38AM |
1 |
AsteriskCDR |
|
Tuesday September 23 2014 |
Time | Replies | Subject |
7:17PM |
2 |
Playback/background audio from MySQL BLOB |
5:49PM |
1 |
how can queue agents choose which call to answer? |
5:30PM |
1 |
Change codec when dial from SIP to DAHDI |
5:25PM |
2 |
read digits from the user through php agi script |
1:05PM |
1 |
Multicast AMI? |
9:30AM |
1 |
Ubuntu 14.04 LTS Asterisk and ISDN Cologne Chip |
|
Monday September 22 2014 |
Time | Replies | Subject |
8:08PM |
1 |
DAHDI v2.10.0.1 Fixes loadzone=us ringback tones. |
7:24PM |
0 |
DAHDI-Linux and DAHDI-Tools 2.10.0.1 Now Available |
3:02PM |
1 |
PRI answer too fast |
2:43PM |
1 |
SIPAddHeader from a realtime databse |
10:03AM |
1 |
Install Asterisk 1.4 and Asterisk 12.4 on the same machine |
|
Sunday September 21 2014 |
Time | Replies | Subject |
11:27PM |
1 |
MixMonitor with b option recording all calls |
12:42PM |
1 |
error receiving a fax ... but with a fax that was received without problems |
|
Friday September 19 2014 |
Time | Replies | Subject |
9:29PM |
0 |
Asterisk 13.0.0-beta2 Now Available! |
11:22AM |
1 |
Show Log(NOTICE) messages on the console |
|
Thursday September 18 2014 |
Time | Replies | Subject |
9:43PM |
1 |
Record call ends in 10min |
8:16PM |
1 |
conversation record prematurely |
7:17PM |
0 |
AST-2014-010: Remote crash when handling out of call message in certain dialplan configurations |
7:17PM |
0 |
AST-2014-009: Remote crash based on malformed SIP subscription requests |
7:16PM |
0 |
Asterisk 11.6-cert6, 11.12.1, 12.5.1 Now Available (Security Release) |
5:35PM |
2 |
Asterisk prefix code to dial a high fraud country - security mechanism |
2:36PM |
1 |
Voice-Recognition / ASR / with barge in |
1:33PM |
1 |
Asterisk 11.9.0 PRI no ring indications |
9:06AM |
1 |
mixmonitor - convert wav to mp3/aac |
|
Wednesday September 17 2014 |
Time | Replies | Subject |
11:34AM |
1 |
${ANSWEREDTIME} returning null |
10:12AM |
1 |
GSM to GSM call with callerid passthrough |
2:07AM |
1 |
Polycom DND + Intercom/Paging Override? |
|
Tuesday September 16 2014 |
Time | Replies | Subject |
4:42PM |
1 |
Disabling CDR for all dialed parties in Asterisk 12 |
4:03PM |
2 |
Asterisk- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf |
|
Monday September 15 2014 |
Time | Replies | Subject |
9:45PM |
2 |
Record ANSWERED call |
7:07AM |
1 |
fail2ban and pjsip in asterisk 12 and 13 |
|
Sunday September 14 2014 |
Time | Replies | Subject |
2:36PM |
1 |
sip.conf and extension.conf configuration |
|
Saturday September 13 2014 |
Time | Replies | Subject |
6:04PM |
1 |
NOT able to call on local extensions while successfully call on external mobile no.(using SONETEL account) |
6:03PM |
0 |
on reboot /var/run/asterisk owner changed to root. |
4:09PM |
1 |
On kernel 3.16.2 : dahdi_rec: Invalid argument |
1:44PM |
1 |
how to store sip.conf and extension.conf into phpmyadmin |
|
Friday September 12 2014 |
Time | Replies | Subject |
6:00PM |
1 |
compiling Asterisk |
2:07PM |
1 |
Call Flow Documentation Tools |
9:27AM |
1 |
Tutorial: compiling and installing Asterisk 13 |
6:43AM |
0 |
Cisco SX20 disconnecting before call |
|
Thursday September 11 2014 |
Time | Replies | Subject |
9:43PM |
3 |
if statement recording - after hours |
3:46PM |
1 |
ASTERISK AND CHAT MESSAGES |
3:14PM |
1 |
chan_sip.c:23647 handle_request_invite: Failed to authenticate device |
7:42AM |
1 |
How to use phpmyadmin to remotely access asterisk mysql database? |
|
Wednesday September 10 2014 |
Time | Replies | Subject |
8:11PM |
0 |
SIP 380 Alternative Service with PJSIP |
7:14PM |
1 |
Ast to Ast TLS trunk |
6:29PM |
0 |
WebRTC meeting Norfolk, 15 October 2014 |
|
Tuesday September 9 2014 |
Time | Replies | Subject |
7:35PM |
1 |
Suspicious routers |
2:52PM |
0 |
Segfault Asterisk 1.4.44 in wmvare ESXi 5.5 |
|
Monday September 8 2014 |
Time | Replies | Subject |
10:39PM |
1 |
Asterisk failed to authenticate device - attack attempt. |
8:48PM |
0 |
is pattern matching inside macro valid? |
6:31PM |
0 |
adding IAX headers |
2:48PM |
1 |
Asterisk removes ice lines in sdp when calling between webrtc clients |
3:55AM |
1 |
Call Transfer Fails - Not a Valid Extension |
|
Sunday September 7 2014 |
Time | Replies | Subject |
8:41PM |
2 |
Pattern Extension not working in Dialplan |
5:00PM |
0 |
Channel h323 and oh323 fails to match inbound IP |
12:14PM |
1 |
PJSIP and Multiple transports per endpoint |
|
Saturday September 6 2014 |
Time | Replies | Subject |
9:28AM |
1 |
Question about SIP warning |
|
Friday September 5 2014 |
Time | Replies | Subject |
1:26PM |
1 |
unidata incom ICW-1000G - On asterisk |
9:55AM |
2 |
Asterisk with PJSIP |
7:18AM |
3 |
New to Asterisks, Couple of Questions |
|
Thursday September 4 2014 |
Time | Replies | Subject |
7:54PM |
0 |
AstriCon Hackathon |
4:57PM |
2 |
Special functionality for Secretary/Boss |
2:44PM |
3 |
Asterisk secure fine tune - stop attack |
1:12PM |
0 |
opus 11.12.0 |
|
Wednesday September 3 2014 |
Time | Replies | Subject |
6:57PM |
4 |
(no subject) |
6:39PM |
1 |
Failover / modifying response time |
11:11AM |
0 |
Fwd: Identifying the multiple cards Digium TE820 |
|
Tuesday September 2 2014 |
Time | Replies | Subject |
8:03AM |
2 |
Custom SIP-header not present in call Asterisk to Asterisk |
6:47AM |
3 |
PJSIP issues with handling incoming calls |
5:09AM |
1 |
AGI scripts - delay issue. |
|
Monday September 1 2014 |
Time | Replies | Subject |
4:46PM |
0 |
Redundancy: PRI Load share +SIP Acctive/Standby |
3:27PM |
1 |
Asterisk 11.Why two NOTIFY while ringing ? |
1:23PM |
1 |
SIP Calls Not Working |
1:11PM |
1 |
Does Asterisk 1.8. Supports Video Calls |
12:34PM |
0 |
More XMPP + Asterisk integration: Send a XMPP message to all extensions logged in an Asterisk queue |
11:20AM |
0 |
Media update error flooding the console output |
10:31AM |
2 |
Setup Own IP PBX Server |
6:24AM |
0 |
Asterisk 11.5.0 T38 Faxing |