| Tuesday March 31 2015 |
| Time | Replies | Subject |
|---|
| 8:45PM |
2 |
Update peer IP address |
| 6:36PM |
0 |
Update peer IP address |
| 3:38PM |
0 |
How does chan_sip match an ACK? |
| 12:06PM |
0 |
help : annoucement queue |
| 10:36AM |
3 |
Update peer IP address |
| 7:23AM |
0 |
Call Quality Measuring |
| |
| Monday March 30 2015 |
| Time | Replies | Subject |
|---|
| 6:09PM |
0 |
Update peer IP address |
| 5:11PM |
0 |
WaitForSilence NEVER detects silence |
| 5:11PM |
0 |
WaitForSilence NEVER detects silence,,Post |
| 4:31PM |
2 |
Update peer IP address |
| 4:21PM |
0 |
How does chan_sip match an ACK? |
| |
| Sunday March 29 2015 |
| Time | Replies | Subject |
|---|
| 4:04PM |
0 |
Iax2 statistics in dialplan |
| 4:02PM |
0 |
Mixing HASH() and LOCAL() |
| 12:06AM |
0 |
Help! How to make Asterisk support ICE in public network |
| |
| Friday March 27 2015 |
| Time | Replies | Subject |
|---|
| 9:17PM |
0 |
Anonymous SIP calls |
| 8:03PM |
5 |
Anonymous SIP calls |
| 6:02PM |
0 |
What's the best average duration for a SIP test call? |
| 5:44PM |
0 |
call between snom 300 and aastra 6731i |
| 5:08PM |
2 |
call between snom 300 and aastra 6731i |
| 5:05PM |
0 |
call between snom 300 and aastra 6731i |
| 4:52PM |
0 |
Gateway Eurotech |
| 3:51PM |
0 |
Problems playing audio file over a Page |
| 3:16AM |
2 |
Gateway Eurotech |
| 2:29AM |
0 |
Anonymous SIP calls |
| 1:24AM |
2 |
Anonymous SIP calls |
| |
| Thursday March 26 2015 |
| Time | Replies | Subject |
|---|
| 3:24PM |
1 |
CDR dst value null after attended transfer |
| 2:28PM |
1 |
Dial to PJSIP Channel with Typo "PJSIP//" Causes Asterisk Shutdown |
| 11:02AM |
2 |
call between snom 300 and aastra 6731i |
| 12:18AM |
1 |
Determining if a queue member is paused in Dialplan logic. [1.8] |
| |
| Wednesday March 25 2015 |
| Time | Replies | Subject |
|---|
| 9:02PM |
0 |
Call Quality Measuring |
| 7:13PM |
0 |
Determining if a queue member is paused in Dialplan logic. [1.8] |
| 6:38PM |
2 |
Determining if a queue member is paused in Dialplan logic. [1.8] |
| 5:58PM |
0 |
PJSIP configuration for Asterisk 13.1.0/SIP trunk outbound calling |
| 5:34PM |
0 |
Call Quality Measuring (Laszlo) |
| 1:47PM |
0 |
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 |
| 1:23PM |
2 |
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 |
| 1:21PM |
5 |
Call Quality Measuring |
| 12:59PM |
0 |
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 |
| 12:35PM |
2 |
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 |
| |
| Tuesday March 24 2015 |
| Time | Replies | Subject |
|---|
| 9:59PM |
1 |
RTP handling |
| 9:28PM |
0 |
RTP handling |
| 9:17PM |
2 |
RTP handling |
| |
| Monday March 23 2015 |
| Time | Replies | Subject |
|---|
| 11:39PM |
0 |
trying to connect to asterisk with softphone (logs, etc) |
| 5:08PM |
1 |
Auto Answer |
| 4:37PM |
0 |
[OT] switches |
| 4:34PM |
0 |
PJSIP - Video Support for WebRTC |
| 3:45PM |
0 |
Question about hangup - Asterisk v11.15.0 |
| 2:58PM |
0 |
Local channel + queue |
| 1:55PM |
2 |
PJSIP - Video Support for WebRTC |
| 10:11AM |
4 |
[OT] switches |
| 5:25AM |
0 |
how asterisk detects silence? |
| 2:56AM |
1 |
Unable to connect to remote asterisk |
| |
| Sunday March 22 2015 |
| Time | Replies | Subject |
|---|
| 1:59PM |
1 |
CLI for pjsip registrations in Asterisk v13.1.0? |
| 3:03AM |
0 |
[OT] switches |
| |
| Saturday March 21 2015 |
| Time | Replies | Subject |
|---|
| 10:23PM |
1 |
RTP sent to remote internal IP |
| 2:30PM |
0 |
Ringtone to a member queue |
| |
| Friday March 20 2015 |
| Time | Replies | Subject |
|---|
| 7:28PM |
0 |
outbound calls |
| 6:43PM |
2 |
outbound calls |
| 6:41PM |
0 |
outbound calls |
| 5:15PM |
0 |
outbound calls |
| 5:02PM |
3 |
outbound calls |
| 4:51PM |
0 |
Caller ID Names |
| 1:55PM |
0 |
UNREACHABLE peer |
| 1:42PM |
4 |
UNREACHABLE peer |
| 9:58AM |
1 |
Dahdi ISDN logging |
| 9:37AM |
0 |
Asterisk on OpenWrt (first time user) |
| |
| Thursday March 19 2015 |
| Time | Replies | Subject |
|---|
| 8:20PM |
0 |
Problems playing an audio file over an intercom/paging system |
| 8:08PM |
0 |
Asterisk switching bridge to native_rtp even with direct_media=no |
| 6:22PM |
1 |
Asterisk 13 : SILK codec ? |
| 5:06PM |
0 |
PJSIP Video on WebRTC Ast 13 |
| 2:58PM |
0 |
Is there a way to escape text passwords in pjsip.conf? |
| 9:12AM |
0 |
Asterisk 13. Writing call quality parameters to CDR. How? |
| 6:47AM |
2 |
Asterisk switching bridge to native_rtp even with direct_media=no |
| 5:31AM |
2 |
how asterisk detects silence? |
| |
| Wednesday March 18 2015 |
| Time | Replies | Subject |
|---|
| 9:19PM |
0 |
Asterisk only registering at one provider |
| 7:37PM |
2 |
Asterisk 13. Writing call quality parameters to CDR. How? |
| 6:54PM |
1 |
res_xmpp.c:3468 xmpp_client_reconnect: |
| 5:13PM |
0 |
res_xmpp.c:3468 xmpp_client_reconnect: |
| 4:52PM |
2 |
res_xmpp.c:3468 xmpp_client_reconnect: |
| 4:32PM |
0 |
TLS not working in 11.16 |
| 3:26PM |
0 |
Asterisk switching bridge to native_rtp even with direct_media=no |
| 3:08PM |
0 |
PRI Callerid Passthrough |
| 3:02PM |
2 |
PRI Callerid Passthrough |
| 2:53PM |
2 |
Asterisk switching bridge to native_rtp even with direct_media=no |
| 2:19PM |
0 |
PRI Callerid Passthrough |
| 12:52PM |
1 |
Asterisk 13.2.0 Video issues |
| 12:48PM |
0 |
Asterisk switching bridge to native_rtp even with direct_media=no |
| 12:41PM |
2 |
Asterisk switching bridge to native_rtp even with direct_media=no |
| 12:20PM |
1 |
4 Port PRI |
| 12:16PM |
2 |
PRI Callerid Passthrough |
| 12:09PM |
0 |
4 Port PRI |
| 11:49AM |
2 |
4 Port PRI |
| 11:43AM |
0 |
PRI Callerid Passthrough |
| 11:30AM |
3 |
PRI Callerid Passthrough |
| 9:43AM |
1 |
pjsip: outofcall_message_context |
| 5:49AM |
1 |
Dialog-Info Event Support |
| 3:44AM |
0 |
Asterisk 13.2.0 Video issues |
| |
| Tuesday March 17 2015 |
| Time | Replies | Subject |
|---|
| 10:53PM |
4 |
Asterisk 13.2.0 Video issues |
| 4:14PM |
2 |
Asterisk only registering at one provider |
| 8:09AM |
0 |
sip trunk to Cisco router |
| 12:51AM |
0 |
Asterisk 13.2.0 Video issues |
| |
| Monday March 16 2015 |
| Time | Replies | Subject |
|---|
| 11:12PM |
2 |
Asterisk 13.2.0 Video issues |
| 8:32PM |
0 |
[PoE] Avaya 1152a1x |
| 8:11PM |
0 |
Video WebRTC Ast 13 |
| 1:00PM |
1 |
Use dialplan variables from MySQL database and replace with value |
| 10:01AM |
0 |
how monitor Transfer function move 302 redirect function |
| 6:18AM |
1 |
Disabling Ringing/Alerting |
| 2:46AM |
0 |
3/16/2015 2:46:09 PM |
| 1:37AM |
1 |
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge. |
| |
| Sunday March 15 2015 |
| Time | Replies | Subject |
|---|
| 11:56PM |
0 |
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge. |
| 8:00PM |
0 |
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge. |
| 7:33PM |
4 |
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge. |
| 7:25PM |
0 |
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge. |
| 4:34PM |
3 |
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge. |
| 4:19PM |
0 |
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge. |
| 2:32PM |
2 |
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge. |
| |
| Saturday March 14 2015 |
| Time | Replies | Subject |
|---|
| 5:40PM |
0 |
RTP sent to internal IP |
| 12:01PM |
0 |
Billing |
| 5:01AM |
0 |
marcotasto@libero.it |
| 1:33AM |
3 |
[OT] switches |
| |
| Friday March 13 2015 |
| Time | Replies | Subject |
|---|
| 10:29PM |
1 |
PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found |
| 8:55PM |
0 |
PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found |
| 8:34PM |
2 |
PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found |
| 3:46PM |
1 |
switching from SIP to Skype..or not |
| 2:16PM |
0 |
ringing in queues |
| 2:04PM |
2 |
ringing in queues |
| 9:18AM |
0 |
Yealink t26 and T28 Panels |
| 7:35AM |
2 |
Yealink t26 and T28 Panels |
| |
| Thursday March 12 2015 |
| Time | Replies | Subject |
|---|
| 10:14PM |
1 |
Realtime followme and channel variables |
| 10:11PM |
1 |
PJSIP and Kamailio without registration |
| 9:58PM |
0 |
PJSIP and Kamailio without registration |
| 7:14PM |
0 |
Unstable phone connection |
| 7:07PM |
0 |
GXP 1405 and asterisk |
| 6:41PM |
2 |
GXP 1405 and asterisk |
| 6:40PM |
0 |
chanspy for group extension |
| 6:39PM |
2 |
Unstable phone connection |
| 6:19PM |
5 |
chanspy for group extension |
| 4:28PM |
0 |
chanspy for group extension |
| 2:56PM |
0 |
Asterisk 13.2.0 Video issues |
| 2:21PM |
0 |
switching from SIP to Skype..or not |
| 2:04PM |
0 |
switching from SIP to Skype..or not |
| 1:39PM |
7 |
switching from SIP to Skype..or not |
| 1:26PM |
0 |
switching from SIP to Skype..or not |
| 12:52PM |
3 |
switching from SIP to Skype..or not |
| 10:30AM |
2 |
chanspy for group extension |
| 8:58AM |
1 |
packages.digium.com |
| 7:20AM |
0 |
WebRTC demo phones |
| 7:16AM |
2 |
WebRTC demo phones |
| 12:11AM |
0 |
PJSIP some AMI events is absent? |
| |
| Wednesday March 11 2015 |
| Time | Replies | Subject |
|---|
| 10:24PM |
0 |
Video call with WebRTC on asterisk 13 |
| 7:48PM |
0 |
chanspy for group extension |
| 6:48PM |
2 |
chanspy for group extension |
| 5:53PM |
0 |
packages.digium.com |
| 5:34PM |
2 |
Caller ID Names |
| 5:17PM |
0 |
Caller ID Names |
| 3:28PM |
2 |
packages.digium.com |
| 2:40PM |
0 |
wav49 VoiceMails should play natively in Google Chrome HTML5 - bug report |
| 7:27AM |
2 |
PJSIP some AMI events is absent? |
| 1:49AM |
0 |
Jitsi, SRTP and Asterisk 11 |
| |
| Tuesday March 10 2015 |
| Time | Replies | Subject |
|---|
| 11:11PM |
1 |
PJSIP and Kamailio without registration |
| 10:22PM |
3 |
Asterisk 13.2.0 Video issues |
| 5:33PM |
0 |
video call with WebRTC on asterisk 13. |
| 2:24PM |
0 |
[BOUNTY] ASTERISK-22708 ODBC failover |
| 1:40PM |
2 |
Caller ID Names |
| 12:19PM |
0 |
Asterisk 13.2.0 Video issues |
| 12:18PM |
0 |
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string. |
| 10:00AM |
2 |
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string. |
| 9:36AM |
0 |
Regarding Text To Speech conversion |
| 9:15AM |
3 |
Asterisk 13.2.0 Video issues |
| 8:14AM |
1 |
func_odbc 123 |
| 7:29AM |
2 |
Regarding Text To Speech conversion |
| 2:55AM |
1 |
Strange Polycom Issue |
| 2:31AM |
1 |
DND on a Polycom IP450 |
| |
| Monday March 9 2015 |
| Time | Replies | Subject |
|---|
| 10:15PM |
0 |
PJSIP and Kamailio without registration |
| 9:33PM |
0 |
PJSIP and Kamailio without registration |
| 9:23PM |
1 |
PJSIP and Kamailio without registration |
| 8:37PM |
0 |
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string. |
| 1:40PM |
0 |
Strange Polycom Issue |
| 1:34PM |
3 |
Strange Polycom Issue |
| 11:12AM |
0 |
Regarding Text To Speech conversion |
| 10:54AM |
2 |
Regarding Text To Speech conversion |
| |
| Sunday March 8 2015 |
| Time | Replies | Subject |
|---|
| 4:10PM |
0 |
AWS/EC2 server selection |
| 3:58PM |
2 |
Asterisk API |
| 3:51PM |
2 |
AWS/EC2 server selection |
| 3:11PM |
0 |
AWS/EC2 server selection |
| |
| Saturday March 7 2015 |
| Time | Replies | Subject |
|---|
| 6:43AM |
2 |
AWS/EC2 server selection |
| |
| Friday March 6 2015 |
| Time | Replies | Subject |
|---|
| 10:21PM |
0 |
res_pjsip endpoint config object's 'identify_by' option needs new value "uri". |
| 9:46PM |
2 |
res_pjsip endpoint config object's 'identify_by' option needs new value "uri". |
| 9:24PM |
0 |
res_pjsip endpoint config object's 'identify_by' option needs new value "uri". |
| 9:05PM |
1 |
New Asterisk build |
| 8:44PM |
0 |
res_pjsip ACL relation to endpoint |
| 8:42PM |
1 |
New Asterisk build |
| 8:34PM |
0 |
New Asterisk build |
| 8:06PM |
2 |
res_pjsip endpoint config object's 'identify_by' option needs new value "uri". |
| 7:33PM |
6 |
New Asterisk build |
| 7:31PM |
0 |
cant get incoming calls in asterisk |
| 6:49PM |
0 |
AWS/EC2 server selection |
| 6:32PM |
0 |
Guidence in DialPlan programming. |
| 5:59PM |
2 |
AWS/EC2 server selection |
| 3:01AM |
2 |
Music on hold |
| 1:26AM |
0 |
PJSIP configuration for AWS/EC2 based Asterisk 13.1.0 |
| |
| Thursday March 5 2015 |
| Time | Replies | Subject |
|---|
| 10:52PM |
2 |
PJSIP configuration for AWS/EC2 based Asterisk 13.1.0 |
| 7:20PM |
0 |
Asterisk removes SDP from 180 with SDP |
| 3:43PM |
1 |
OT - How does the blind transfer function work on Snom-870? |
| 2:56PM |
0 |
OT - How does the blind transfer function work on Snom-870? |
| 2:35PM |
0 |
DAHDI 2.10 on CentOS 5.11 |
| 2:09PM |
2 |
OT - How does the blind transfer function work on Snom-870? |
| 12:54PM |
0 |
Understanding the right way to get started with multiple trunks/extensions |
| 12:11PM |
1 |
hangup call gw FXO |
| 11:41AM |
0 |
hangup call gw FXO |
| 10:30AM |
0 |
OT - How does the blind transfer function work on Snom-870? |
| 7:42AM |
0 |
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string. |
| 7:29AM |
4 |
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string. |
| 12:09AM |
2 |
OT - How does the blind transfer function work on Snom-870? |
| |
| Wednesday March 4 2015 |
| Time | Replies | Subject |
|---|
| 11:55PM |
0 |
PJSIP works on UDP but not TCP |
| 11:42PM |
1 |
PJSIP works on UDP but not TCP |
| 11:32PM |
1 |
PJSIP works on UDP but not TCP |
| 8:33PM |
0 |
No DTMF in large conferences |
| 7:52PM |
0 |
RTP suppress during calls - Asterisk 1.8.* |
| 7:36PM |
0 |
Failsafe AGI using AEL |
| 6:54PM |
1 |
PJSIP: Failed to create outgoing session to endpoint |
| 4:07PM |
0 |
WebRTC phone |
| 3:04PM |
2 |
hangup call gw FXO |
| 2:41PM |
0 |
TLS connect() error when calling udp to tls |
| 2:30PM |
0 |
TLS, SRTP, Asterisk11 and Snom870s |
| 11:53AM |
3 |
Understanding the right way to get started with multiple trunks/extensions |
| 5:47AM |
2 |
WebRTC phone |
| |
| Tuesday March 3 2015 |
| Time | Replies | Subject |
|---|
| 9:35PM |
2 |
Dialing multiple channels with confirm |
| 9:34PM |
0 |
TLS, SRTP, Asterisk11 and Snom870s |
| 8:44PM |
2 |
TLS, SRTP, Asterisk11 and Snom870s |
| 6:37PM |
0 |
TLS, SRTP, Asterisk11 and Snom870s |
| 6:19PM |
0 |
TLS, SRTP, Asterisk11 and Snom870s |
| 5:16PM |
6 |
TLS, SRTP, Asterisk11 and Snom870s |
| 5:14PM |
1 |
Cannot configure PJSIP TLS |
| 4:19PM |
1 |
which libsrtp ? |
| 12:58PM |
0 |
second BOUNTY donor for ASTERISK-22708 (ODBC failover) |
| 4:53AM |
1 |
account code |
| |
| Monday March 2 2015 |
| Time | Replies | Subject |
|---|
| 11:43PM |
0 |
Queue_log transfer |
| 9:23PM |
1 |
static realtime vs config files |
| 8:27PM |
0 |
Events |
| 4:15PM |
0 |
System() command refuses to execute bash script |
| 4:14PM |
0 |
Problems with the voice quality under load |
| 3:28PM |
0 |
System() command refuses to execute bash script |
| 3:26PM |
4 |
Problems with the voice quality under load |
| 2:44PM |
6 |
System() command refuses to execute bash script |
| 2:27PM |
1 |
System() command refuses to execute bash script |
| 9:44AM |
0 |
situation with ivr and four-channel gateway |
| 6:04AM |
0 |
CDR with conference asterisk 12 |
| 2:33AM |
0 |
Upgrade to Fedora 21, now gv requires rtp ? |
| |
| Sunday March 1 2015 |
| Time | Replies | Subject |
|---|
| 3:29PM |
0 |
convert asterisk extensions to single numbers |