| Monday August 31 2015 |
| Time | Replies | Subject |
|---|
| 3:52PM |
1 |
AMI 'meetme list concise' hanging |
| 3:05PM |
0 |
Escaping parameter for ODBC function |
| 2:31PM |
1 |
Asterisk Manager Interface AMI over HTTP. |
| |
| Friday August 28 2015 |
| Time | Replies | Subject |
|---|
| 4:14PM |
1 |
Anyone doing speech to text? |
| 1:43PM |
1 |
webrtc no audio |
| 1:20PM |
1 |
Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? |
| 10:26AM |
0 |
Anyone doing speech to text? |
| 9:11AM |
3 |
Anyone doing speech to text? |
| 6:55AM |
0 |
Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? |
| |
| Thursday August 27 2015 |
| Time | Replies | Subject |
|---|
| 10:07PM |
2 |
Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? |
| 9:57PM |
0 |
Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? |
| 9:54PM |
2 |
Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? |
| 9:27PM |
0 |
Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? |
| 8:17PM |
1 |
polycom phone behind firewall with asterisk 11.19 |
| 8:08PM |
2 |
Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? |
| 6:56PM |
0 |
Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? |
| 6:40PM |
2 |
Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? |
| 6:07PM |
0 |
webrtc no audio |
| 6:07PM |
0 |
Anyone doing speech to text? |
| 5:02PM |
2 |
Anyone doing speech to text? |
| 4:16PM |
0 |
Anyone doing speech to text? |
| 10:37AM |
1 |
simultaneous use of chan_sip/chan_pjsip |
| 10:33AM |
0 |
simultaneous use of chan_sip/chan_pjsip |
| |
| Wednesday August 26 2015 |
| Time | Replies | Subject |
|---|
| 6:15PM |
3 |
Anyone doing speech to text? |
| 4:36PM |
0 |
SIP Trunk - problem to connect |
| 1:58AM |
0 |
Multiple variable substitution in Set |
| |
| Tuesday August 25 2015 |
| Time | Replies | Subject |
|---|
| 6:58PM |
0 |
pattern regexten and dialing to trunk |
| 6:00PM |
0 |
Changing volume via dialplan |
| 5:51PM |
0 |
Ringback issue |
| 5:32PM |
1 |
PJSIP add |
| 3:51PM |
0 |
PJSIP add |
| 12:39PM |
0 |
How to send Image over asterisk sip |
| 4:23AM |
2 |
How to send Image over asterisk sip |
| 4:11AM |
0 |
How to send Image over asterisk sip |
| 4:10AM |
2 |
How to send Image over asterisk sip |
| 3:56AM |
0 |
How to send Image over asterisk sip |
| 3:48AM |
4 |
Ringback issue |
| 3:47AM |
2 |
How to send Image over asterisk sip |
| 2:40AM |
1 |
Fwd: ferie estive |
| 1:17AM |
1 |
Does the asterisk support for sending image ? |
| |
| Monday August 24 2015 |
| Time | Replies | Subject |
|---|
| 7:47PM |
3 |
PJSIP add |
| |
| Sunday August 23 2015 |
| Time | Replies | Subject |
|---|
| 6:46PM |
0 |
Hearing peep for second call and special signal for caller |
| 5:53PM |
0 |
SIP domain different than provider's |
| 5:42PM |
0 |
dynamic 'fromdomain' variable |
| |
| Friday August 21 2015 |
| Time | Replies | Subject |
|---|
| 11:20PM |
0 |
Incoming calls get 488 error |
| 10:45PM |
1 |
Incoming calls get 488 error |
| 5:52AM |
2 |
SIP domain different than provider's |
| |
| Thursday August 20 2015 |
| Time | Replies | Subject |
|---|
| 7:43PM |
0 |
Transfer |
| 5:52PM |
2 |
Changing volume via dialplan |
| 3:52PM |
1 |
${MACRO_CONTEXT} for Subroutines |
| 10:12AM |
1 |
asterisk server stress test |
| 4:42AM |
1 |
SRV lookups in Asterisk 11 |
| 1:16AM |
1 |
asterisk server stress test |
| 1:11AM |
0 |
asterisk server stress test |
| |
| Wednesday August 19 2015 |
| Time | Replies | Subject |
|---|
| 5:23PM |
3 |
asterisk server stress test |
| 5:07PM |
0 |
asterisk server stress test |
| 4:48PM |
2 |
asterisk server stress test |
| 4:11PM |
0 |
asterisk server stress test |
| 1:13PM |
3 |
asterisk server stress test |
| 10:06AM |
0 |
asterisk server stress test |
| 7:01AM |
2 |
asterisk server stress test |
| |
| Tuesday August 18 2015 |
| Time | Replies | Subject |
|---|
| 10:06AM |
0 |
Stopping recordings on all legs |
| 8:12AM |
1 |
No audio when using TLS/SRTP with Kamailio and Asterisk 13 |
| 7:08AM |
0 |
Asterisk 13 chan_sip trunk appending @string to dialled number |
| 6:48AM |
2 |
Asterisk 13 chan_sip trunk appending @string to dialled number |
| 6:38AM |
0 |
Shared RealTime Database |
| 6:37AM |
0 |
Asterisk 13 chan_sip trunk appending @string to dialled number |
| 6:26AM |
2 |
Asterisk 13 chan_sip trunk appending @string to dialled number |
| 6:21AM |
0 |
Asterisk 13 chan_sip trunk appending @string to dialled number |
| 5:44AM |
2 |
Asterisk 13 chan_sip trunk appending @string to dialled number |
| 4:39AM |
0 |
Asterisk 13 chan_sip trunk appending @string to dialled number |
| 1:31AM |
1 |
Asterisk 13 chan_sip trunk appending @string to dialled number |
| 12:37AM |
0 |
Asterisk 13 chan_sip trunk appending @string to dialled number |
| 12:33AM |
5 |
Asterisk 13 chan_sip trunk appending @string to dialled number |
| |
| Monday August 17 2015 |
| Time | Replies | Subject |
|---|
| 2:58PM |
2 |
Shared RealTime Database |
| 8:58AM |
0 |
Fw: try it out |
| 7:01AM |
1 |
786 000 files limit Centos 7 - Asterisk (Stefan Viljoen) |
| |
| Saturday August 15 2015 |
| Time | Replies | Subject |
|---|
| 3:42PM |
0 |
One way audio - doesn't seem to be NAT issue - SOLVED! |
| 3:08PM |
2 |
One way audio - doesn't seem to be NAT issue - SOLVED! |
| 8:30AM |
0 |
One way audio - doesn't seem to be NAT issue |
| |
| Friday August 14 2015 |
| Time | Replies | Subject |
|---|
| 7:11PM |
1 |
chan_sip.c: Retransmission timeout reached on transmission |
| 1:33PM |
0 |
chan_sip.c: Retransmission timeout reached on transmission |
| 12:54PM |
2 |
chan_sip.c: Retransmission timeout reached on transmission |
| |
| Thursday August 13 2015 |
| Time | Replies | Subject |
|---|
| 7:48PM |
2 |
simultaneous use of chan_sip/chan_pjsip |
| 4:25PM |
1 |
Is peer order in sip.conf important? |
| 3:20PM |
0 |
simultaneous use of chan_sip/chan_pjsip |
| 2:59PM |
0 |
One way audio - doesn't seem to be NAT issue |
| 8:54AM |
2 |
simultaneous use of chan_sip/chan_pjsip |
| 8:41AM |
2 |
One way audio - doesn't seem to be NAT issue |
| |
| Wednesday August 12 2015 |
| Time | Replies | Subject |
|---|
| 2:31PM |
0 |
Call Queues : linear strategy WITH priority |
| 2:04PM |
2 |
Call Queues : linear strategy WITH priority |
| 12:41PM |
0 |
Busy level in Asterisk 11 |
| 12:34PM |
2 |
Busy level in Asterisk 11 |
| 12:20PM |
0 |
How many Asterisk deployments? |
| 11:37AM |
0 |
How to send Image over asterisk sip |
| 8:23AM |
2 |
webrtc no audio |
| 7:46AM |
0 |
786 000 files limit Centos 7 - Asterisk |
| 7:43AM |
1 |
786 000 files limit Centos 7 - Asterisk |
| 7:06AM |
1 |
786 000 files limit Centos 7 - Asterisk keep complaining |
| 1:07AM |
1 |
strange warnings "no samples for alawtolin" |
| |
| Tuesday August 11 2015 |
| Time | Replies | Subject |
|---|
| 7:10PM |
3 |
One way audio - doesn't seem to be NAT issue |
| 11:39AM |
1 |
asterisk queue - skills based routing (patch updated) |
| 10:18AM |
0 |
webrtc no audio |
| 9:00AM |
3 |
786 000 files limit Centos 7 - Asterisk keep complaining |
| 5:20AM |
0 |
asterisk-users@lists.digium.com |
| 1:40AM |
2 |
webrtc no audio |
| |
| Monday August 10 2015 |
| Time | Replies | Subject |
|---|
| 8:39PM |
0 |
Asterisk RealTime Sippeers, rtcachefriends=yes, phones lose registration |
| 6:03PM |
1 |
Siren7 for Asterisk 13.5 |
| 5:54PM |
0 |
Siren7 for Asterisk 13.5 |
| 3:59PM |
1 |
load-balancing AMI and load-balancing FastAGI? |
| 3:42PM |
0 |
asterisk queue - skills based routing (patch updated) |
| 3:38PM |
2 |
Siren7 for Asterisk 13.5 |
| 3:36PM |
0 |
Siren7 for Asterisk 13.5 |
| 3:35PM |
0 |
webrtc no audio |
| 1:05PM |
0 |
Modifying CDR values from a hangup extension in Asterisk 13 |
| 12:33PM |
2 |
webrtc no audio |
| 11:54AM |
2 |
asterisk queue - skills based routing (patch updated) |
| |
| Sunday August 9 2015 |
| Time | Replies | Subject |
|---|
| 4:46PM |
1 |
Asterisk 11.19.0 Now Available |
| |
| Saturday August 8 2015 |
| Time | Replies | Subject |
|---|
| 1:26PM |
0 |
Asterisk 11.19.0 Now Available |
| 10:41AM |
2 |
How to send Image over asterisk sip |
| |
| Friday August 7 2015 |
| Time | Replies | Subject |
|---|
| 9:58PM |
2 |
Siren7 for Asterisk 13.5 |
| 9:56PM |
0 |
Asterisk 13.5.0 Now Available |
| 9:54PM |
2 |
Asterisk 11.19.0 Now Available |
| 4:20PM |
2 |
AgentRequest() and which agent id? |
| 3:50PM |
0 |
AgentRequest() and which agent id? |
| 3:06PM |
2 |
AgentRequest() and which agent id? |
| 2:54PM |
1 |
PTT push to talk solution |
| 2:11PM |
1 |
How many Asterisk deployments? |
| 12:51PM |
0 |
One-Way Calling between two * boxes (that was working before) |
| 12:41PM |
0 |
PTT push to talk solution |
| 12:15PM |
1 |
786 000 files limit Centos 7 - Asterisk keeps complaining |
| 11:23AM |
0 |
compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes |
| 10:47AM |
3 |
compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes |
| 1:24AM |
4 |
PTT push to talk solution |
| |
| Thursday August 6 2015 |
| Time | Replies | Subject |
|---|
| 8:50PM |
0 |
Asterisk uses "Anonymous", but why? |
| 8:38PM |
2 |
Asterisk uses "Anonymous", but why? |
| 7:57PM |
0 |
Asterisk uses "Anonymous", but why? |
| 7:37PM |
2 |
Asterisk uses "Anonymous", but why? |
| 7:00PM |
0 |
asterisk queue - skills based routing (patch updated) |
| 6:54PM |
0 |
Asterisk uses "Anonymous", but why? |
| 6:33PM |
2 |
Asterisk uses "Anonymous", but why? |
| 6:25PM |
0 |
Asterisk uses "Anonymous", but why? |
| 5:55PM |
3 |
Asterisk uses "Anonymous", but why? |
| 5:33PM |
0 |
Asterisk uses "Anonymous", but why? |
| 5:07PM |
4 |
Asterisk uses "Anonymous", but why? |
| 4:56PM |
0 |
Asterisk uses "Anonymous", but why? |
| 3:09PM |
3 |
PTT push to talk solution |
| 7:24AM |
2 |
asterisk queue - skills based routing (patch updated) |
| |
| Wednesday August 5 2015 |
| Time | Replies | Subject |
|---|
| 9:39PM |
0 |
My apologies |
| 9:38PM |
2 |
Asterisk uses "Anonymous", but why? |
| 9:37PM |
0 |
Asterisk uses "Anonymous", but why? |
| 9:37PM |
0 |
Asterisk uses "Anonymous", but why? |
| 9:36PM |
0 |
Asterisk uses "Anonymous", but why? |
| 2:20PM |
0 |
Update: Planned NASA trip around Astricon |
| 9:01AM |
1 |
Looking for PRI Card with automatic fail over |
| |
| Tuesday August 4 2015 |
| Time | Replies | Subject |
|---|
| 2:16PM |
2 |
Modifying CDR values from a hangup extension in Asterisk 13 |
| 7:47AM |
0 |
Looking for PRI Card with automatic fail over |
| |
| Monday August 3 2015 |
| Time | Replies | Subject |
|---|
| 3:59PM |
0 |
Call Center |
| 3:21PM |
0 |
detection machine recommendations |
| 3:09PM |
0 |
Looking for PRI Card with automatic fail over |
| 2:50PM |
6 |
Looking for PRI Card with automatic fail over |
| 2:36PM |
0 |
Modifying CDR values from a hangup extension in Asterisk 13 |
| 2:29PM |
2 |
Modifying CDR values from a hangup extension in Asterisk 13 |
| 2:13PM |
0 |
SIP Phones over VPN Drop Audio One-Way |
| 9:58AM |
0 |
showing sip number insted of pri number |
| 9:53AM |
0 |
Call Center |
| 7:42AM |
0 |
Call Center |
| |
| Saturday August 1 2015 |
| Time | Replies | Subject |
|---|
| 5:57PM |
5 |
Call Center |
| 4:35PM |
1 |
showing sip number insted of pri number |
| 1:50AM |
0 |
showing sip number insted of pri number |