Friday October 31 2014 |
Time | Replies | Subject |
3:26PM |
1 |
PlayTones while in call |
4:26AM |
0 |
asterisk-users Digest, Vol 123, Issue 38 |
1:32AM |
0 |
Paul Albrecht |
|
Thursday October 30 2014 |
Time | Replies | Subject |
10:01PM |
1 |
MWI publish VIA pjsip for non sip channels |
7:40PM |
1 |
PlayTones not working |
7:18PM |
1 |
Register multiple phones to a single AOR with PJSIP |
1:52PM |
2 |
${HASH(SIP_CAUSE,<channel-name>)} |
9:35AM |
0 |
Asterisk registration with Dialogic HMP. |
|
Wednesday October 29 2014 |
Time | Replies | Subject |
10:16PM |
1 |
Asterisk 13 : SILK codec ? |
6:35PM |
0 |
OT: script to remove leading and trailing silence |
4:10PM |
0 |
My Asterisk can not send fax via T.38 |
10:50AM |
1 |
Astricom 2014 presentations |
|
Tuesday October 28 2014 |
Time | Replies | Subject |
3:22PM |
1 |
dialplan reload context |
12:30PM |
2 |
Asterisk 13 stable? |
12:21PM |
1 |
Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface. |
9:58AM |
1 |
Asterisk 12 - zombie processes |
|
Monday October 27 2014 |
Time | Replies | Subject |
11:35PM |
1 |
sip.conf to pjsip.conf conversion script |
7:04PM |
0 |
AppKonference 2.6 |
12:56PM |
0 |
authentication time for asterisk server |
8:48AM |
0 |
Detect hangup due to RTP timeout |
3:42AM |
1 |
Setting Music on Hold with the Manager Interface |
|
Sunday October 26 2014 |
Time | Replies | Subject |
5:34PM |
0 |
Asterisk and Kamailio Load Balancing |
8:55AM |
0 |
Port number in From URI on Asterisk 12 PJSIP |
8:22AM |
1 |
DTMF behavior in asterisk 12 with PJSIP |
1:16AM |
0 |
Asterisk 13.0.0 Now Available! |
|
Saturday October 25 2014 |
Time | Replies | Subject |
10:43PM |
1 |
make asterisk do something when an outgoing call is picked up |
9:09PM |
2 |
Voicemail ODBC Storage |
|
Friday October 24 2014 |
Time | Replies | Subject |
10:09PM |
1 |
Questions on musiconhold.conf custom mode |
3:51PM |
1 |
Call forwarding from Phones and getting the referrer IP |
12:47PM |
0 |
ConfBridge / internal_sample_rate=auto / warning |
11:39AM |
1 |
Debugging issues with setup |
|
Thursday October 23 2014 |
Time | Replies | Subject |
8:32PM |
1 |
Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes |
3:26PM |
1 |
11.13.1: unable to load sip.conf (or iax ) |
3:07PM |
1 |
logger.conf |
2:57PM |
1 |
Auto video call hangup |
2:13AM |
1 |
PJSIP and NAT behind a dynamic IP address |
|
Wednesday October 22 2014 |
Time | Replies | Subject |
9:43PM |
1 |
SPA504G auto answer |
7:06PM |
1 |
[asterisk-dev] AstriDevCon 2014: Agenda item DeprecateAMI/AGI(Ben Klang) |
6:40PM |
0 |
Video call |
4:14PM |
1 |
[asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang) |
3:33PM |
0 |
SIP dialing with authentication with dialstring and wothout sip; conf |
2:31PM |
0 |
AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang) |
3:23AM |
2 |
res_fax T.38 Gateway with SpanDSP - Force ReINVITE? |
|
Tuesday October 21 2014 |
Time | Replies | Subject |
6:54AM |
1 |
Asterisk 11.9.0 crash and restart |
6:19AM |
1 |
[asterisk-user] Confbridge Kick Action |
12:29AM |
0 |
TLS on SIP trunk |
|
Monday October 20 2014 |
Time | Replies | Subject |
9:58PM |
1 |
bristuff-0.4.0-RC4-xr7 |
4:41PM |
0 |
AST-2014-011: Asterisk Susceptibility to POODLE Vulnerability |
4:40PM |
0 |
Asterisk 1.8.28-cert2, 1.8.31.1, 11.6-cert7, 11.13.1, 12.6.1, 13.0.0-beta3 Now Available (Security Release) |
|
Saturday October 18 2014 |
Time | Replies | Subject |
9:47PM |
1 |
Asterisk Crashes Randomly with Cepstral Swift TTS |
9:03AM |
0 |
Asterisk 12.6 and MWI, no more working |
|
Thursday October 16 2014 |
Time | Replies | Subject |
1:07PM |
0 |
Queue: Passing params to macros and gosubs |
7:48AM |
2 |
Asterisk GOIP Outgoing Callerid not working |
|
Wednesday October 15 2014 |
Time | Replies | Subject |
11:36PM |
0 |
OpenSIPS Summit Oct 21st before Astricon |
10:59AM |
1 |
allo.com gsm card with AsteriskNOW |
6:50AM |
1 |
Asterisk 12 CDR dst field empty |
|
Tuesday October 14 2014 |
Time | Replies | Subject |
3:47PM |
1 |
Issue playing high quality white noise |
2:57PM |
1 |
Do subroutines need their own h extension? |
12:59PM |
1 |
debugging T.38 issues |
|
Monday October 13 2014 |
Time | Replies | Subject |
6:50AM |
1 |
asterisk stun setup , not using public ip returned by stun server |
|
Saturday October 11 2014 |
Time | Replies | Subject |
1:09AM |
1 |
Reset calls processed counter |
|
Friday October 10 2014 |
Time | Replies | Subject |
1:14PM |
1 |
howto cancel simultaneous calls - dial(sip/phone1&sip/phone2) |
10:47AM |
2 |
Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages |
|
Thursday October 9 2014 |
Time | Replies | Subject |
1:20PM |
1 |
SIP over 3G Mobile Network using NAT |
12:06PM |
1 |
sdp_crypto_process: Crypto life time unsupported: crypto |
|
Wednesday October 8 2014 |
Time | Replies | Subject |
1:35PM |
2 |
Asterisk LTS segment faults |
|
Tuesday October 7 2014 |
Time | Replies | Subject |
10:32AM |
1 |
Grandstream GXP2160 + SRTP |
8:11AM |
1 |
Asterisk Phone ( Telecom feature ) |
5:21AM |
0 |
new app_swift is live |
|
Sunday October 5 2014 |
Time | Replies | Subject |
11:40PM |
1 |
Setting channel musicclass from AGI |
3:45PM |
1 |
Voicemail message number off by one when using ODBC storage |
|
Saturday October 4 2014 |
Time | Replies | Subject |
2:48PM |
1 |
Pjsip and regcontext (for DUNDi) |
1:43PM |
1 |
No chan_sip in compiled asterisk-11.13.0 |
|
Friday October 3 2014 |
Time | Replies | Subject |
7:12PM |
1 |
Lost audio on forwarded calls |
9:33AM |
1 |
SPA112: one analog phone works, not the other |
|
Thursday October 2 2014 |
Time | Replies | Subject |
5:52PM |
2 |
how to strip +1 out of incoming number |
5:32PM |
2 |
Voice Mail Questions |
3:04PM |
1 |
AstLinux 1.2.0 Released |
9:02AM |
1 |
Sent ami event from AGI? |
|
Wednesday October 1 2014 |
Time | Replies | Subject |
2:09PM |
1 |
JABBER_STATUS CODE 7 |
1:00PM |
1 |
CALLERID(num) and CDR(clid) - originate |
9:40AM |
2 |
PBX hacked: why hundred of calls to the same number ? |