| Saturday November 30 2013 |
| Time | Replies | Subject |
|---|
| 6:26AM |
2 |
AGI Script not working |
| |
| Friday November 29 2013 |
| Time | Replies | Subject |
|---|
| 5:35PM |
2 |
Answering agent |
| |
| Thursday November 28 2013 |
| Time | Replies | Subject |
|---|
| 10:45AM |
1 |
RTP packets send, but no audio |
| 10:10AM |
0 |
Direct Media and message "SIP/SipAgent-00000bf9 requested media update control 26, passing it to SIP/ead14-00000bfb" |
| |
| Wednesday November 27 2013 |
| Time | Replies | Subject |
|---|
| 9:57PM |
3 |
issue with speech in IVR |
| 4:31PM |
0 |
SaySentence/SoundPack Proposal |
| 2:12PM |
1 |
Asterisk RTP Questions |
| 11:26AM |
2 |
Asterisk uses 105% CPU |
| 11:08AM |
0 |
Asterisk is delaying DTMF INFO in meetme |
| |
| Tuesday November 26 2013 |
| Time | Replies | Subject |
|---|
| 7:35PM |
1 |
Outgoing phone calls muffled |
| 2:22PM |
1 |
Outgoing phone calls "muffled" |
| |
| Monday November 25 2013 |
| Time | Replies | Subject |
|---|
| 11:00PM |
0 |
Asterisk 12.0.0-beta2 Now Available! |
| 5:23PM |
4 |
Voicemail greeting playback issues? |
| 4:22PM |
1 |
Asterisk 11.6.0 not starting up |
| 6:36AM |
1 |
terminating the call, when transferer hangs up the call during attended transfer |
| |
| Sunday November 24 2013 |
| Time | Replies | Subject |
|---|
| 9:44PM |
2 |
combine external video source and audio call to make SIP video call? |
| |
| Saturday November 23 2013 |
| Time | Replies | Subject |
|---|
| 10:11PM |
0 |
how to answer a Panasonic PBX extension with Asterisk? |
| 7:47PM |
1 |
DAHDI Missing '/sys/bus/astribanks/drivers/xppdrv/sync' |
| 1:58PM |
0 |
11.6 voicemail message cropped off? |
| 1:31PM |
0 |
11.6 voicemail message cropped off? |
| |
| Friday November 22 2013 |
| Time | Replies | Subject |
|---|
| 7:41PM |
1 |
DAHDI-Linux and DAHDI-Tools 2.8.0-rc2 Now Available |
| 2:35PM |
1 |
Res corosync. |
| 1:50PM |
0 |
Channel not releasing immediately for Attended Transfer |
| 7:54AM |
1 |
Sangoma transcoding card bug - drops audio samples |
| 1:25AM |
0 |
SIP FXS ATA with Gigabit ethernet bridge port, |
| 1:04AM |
0 |
Caller's phone keeps ringing after 200 OK |
| |
| Thursday November 21 2013 |
| Time | Replies | Subject |
|---|
| 9:11PM |
0 |
Dialing directly with username and password |
| 7:47PM |
0 |
Monitor extension status |
| 4:04PM |
3 |
Call files without permission for asterisk to read |
| 3:45PM |
1 |
Question about Management Interface |
| |
| Wednesday November 20 2013 |
| Time | Replies | Subject |
|---|
| 10:03PM |
2 |
userfield not logged to CDR |
| 7:32PM |
5 |
Movistar sip Mexico |
| 1:06PM |
2 |
Asterisk 1.8.24 : illegal instruction |
| 9:49AM |
0 |
Welcome to the "asterisk-users" mailing list |
| |
| Tuesday November 19 2013 |
| Time | Replies | Subject |
|---|
| 6:15PM |
0 |
How to check ISDN / MFCR2 Trunk Status |
| 4:44PM |
1 |
Ast11: How to see call progress like in Ast <= 1.8 |
| 7:32AM |
2 |
Communicate with barge agent |
| 6:31AM |
0 |
Redirecting a channel to Meetme fails with Hangup. |
| 12:33AM |
1 |
Amazon, Asterisk and reliability beyond a hobby system? |
| |
| Monday November 18 2013 |
| Time | Replies | Subject |
|---|
| 11:23PM |
0 |
Asterisk 10 EOL Approaching |
| 10:20PM |
0 |
app_swift on centos 6 X64 |
| 7:21AM |
1 |
CONNECTEDLINE and panasonic 500 |
| 5:03AM |
1 |
CEL for attented transfer |
| |
| Sunday November 17 2013 |
| Time | Replies | Subject |
|---|
| 7:16PM |
0 |
DTMF relay in meetme is not reliable |
| 7:04AM |
2 |
Bulk forwarding to another Asterisk |
| |
| Saturday November 16 2013 |
| Time | Replies | Subject |
|---|
| 11:23AM |
3 |
Make phone ring through webserver using Asterisk |
| 10:54AM |
0 |
Help - DTMF relay in meetme is not reliable |
| |
| Friday November 15 2013 |
| Time | Replies | Subject |
|---|
| 12:37PM |
0 |
overlapdialing and no digits in setup problem |
| |
| Thursday November 14 2013 |
| Time | Replies | Subject |
|---|
| 11:27PM |
1 |
recieve fax from PRI using spandsp 65% failure rate |
| 10:52PM |
1 |
DAHDI with (CDR(userfield) |
| 6:41PM |
1 |
Queue linear "unordered" feature when using realtime |
| 5:20PM |
0 |
Adding SIP method MESSAGE to Allow header |
| 4:35PM |
2 |
Add SIP Header for 1 SIP peer when calling a group of SIP peers |
| 3:46PM |
1 |
Integration with NEC DSX - help with dial line |
| 5:15AM |
1 |
e1 , hdlc data link? |
| 1:53AM |
1 |
AMI version vs. AST version |
| |
| Wednesday November 13 2013 |
| Time | Replies | Subject |
|---|
| 8:35PM |
1 |
SIP Presence across two servers |
| 6:29PM |
1 |
SIP Mass exodus |
| 3:20PM |
1 |
calendar.conf include |
| 12:37AM |
2 |
Recurring SIP problem with asterisk 11.6 & 11.7 |
| |
| Tuesday November 12 2013 |
| Time | Replies | Subject |
|---|
| 3:18PM |
3 |
VoIP sound quality : highroad sound |
| 1:56PM |
1 |
Asterisk 1.8.20 crashing |
| |
| Monday November 11 2013 |
| Time | Replies | Subject |
|---|
| 6:47PM |
0 |
Asterisk Real-time Static Voicemail |
| 7:10AM |
2 |
how determine mandatory modules to slimming asterisk |
| 6:28AM |
1 |
Asterisk Realtime Static Voicemail |
| 5:35AM |
0 |
MCID |
| |
| Friday November 8 2013 |
| Time | Replies | Subject |
|---|
| 7:35PM |
1 |
Automated Call Testing - end-to-end - SIP Provider |
| 4:28PM |
1 |
Asterisk 1.8.22 |
| 4:27PM |
0 |
T.38 termination |
| 3:39PM |
1 |
11.5.0 - SIP registration not retrying after failures |
| 12:51AM |
3 |
Capture dead phone? |
| |
| Thursday November 7 2013 |
| Time | Replies | Subject |
|---|
| 11:20AM |
1 |
Unix connections not always disconnecting |
| |
| Tuesday November 5 2013 |
| Time | Replies | Subject |
|---|
| 10:37PM |
0 |
sip show channelstats shows all 0 |
| 5:02PM |
1 |
Asterisk 1.4 and DAHDI 2.7 |
| 10:09AM |
1 |
How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ? |
| 4:22AM |
1 |
two steps when calling from web! |
| |
| Monday November 4 2013 |
| Time | Replies | Subject |
|---|
| 8:24PM |
1 |
CallerID settings |
| 3:25PM |
1 |
No matching peers message has gone (1.8.23.1) |
| 11:01AM |
0 |
set different codec for different sip calls |
| |
| Saturday November 2 2013 |
| Time | Replies | Subject |
|---|
| 12:52PM |
1 |
Register Sip extension with out Sip phone |
| 1:51AM |
0 |
Redirect a GSM call through Wifi to a SIPphone |
| |
| Friday November 1 2013 |
| Time | Replies | Subject |
|---|
| 4:27PM |
1 |
Redirect a GSM call through Wifi to a SIP phone |
| 10:02AM |
1 |
TE420, is it possible do disable span (red blinking)? |