| Tuesday January 31 2012 |
| Time | Replies | Subject |
|---|
| 9:47PM |
2 |
Experience with Eicon Diva PRO 3.0? |
| 7:49PM |
0 |
Troubleshooting one-way audio with H.323 trunk between Asterisk and Avaya IP Office |
| 5:58PM |
0 |
Deadlock detected in asterisk-1.8.9.0 x86_64 |
| 5:39PM |
5 |
Proposed changes to Asterisk release and support cycles |
| 3:16PM |
1 |
SRV record for non-standard SIP port? |
| 12:32PM |
6 |
[NAT] SSH vs. OpenVPN? |
| 10:50AM |
2 |
Cell Phone as a Queue member |
| 8:12AM |
0 |
AMI - Getting Event of QueueAgents WrapupTime State |
| 6:17AM |
1 |
Problem with DTMF in Voicemail main |
| |
| Monday January 30 2012 |
| Time | Replies | Subject |
|---|
| 9:47PM |
0 |
TLS problems - patch in Jira |
| 9:47PM |
0 |
RFC 5922 (TLS Certificates) and Asterisk |
| 8:53PM |
1 |
fall back to inband DTMF? |
| 8:01PM |
0 |
Codec |
| 4:12PM |
2 |
CA Issued Certificates / TLS + SRTP |
| 9:42AM |
0 |
atx timeout - play xferfailsound |
| |
| Sunday January 29 2012 |
| Time | Replies | Subject |
|---|
| 2:48PM |
1 |
fail2ban restarts |
| 7:21AM |
1 |
SendFax not sending AMI events |
| |
| Saturday January 28 2012 |
| Time | Replies | Subject |
|---|
| 4:22PM |
1 |
process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found |
| |
| Friday January 27 2012 |
| Time | Replies | Subject |
|---|
| 7:03PM |
1 |
TCP transport and BLF |
| 6:57PM |
2 |
Asterisk 1.4 and configuration to be via Database instead of conf files |
| 5:10PM |
0 |
Asterisk 10.1.0 Now Available |
| 5:10PM |
3 |
Asterisk 1.8.9.0 Now Available |
| 3:44PM |
1 |
sip reload and TCP transport. |
| 7:48AM |
1 |
Strange how Asterisk know the updated information of log |
| 4:49AM |
5 |
Weird IPs in Fail2ban list |
| 2:28AM |
2 |
upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings |
| |
| Thursday January 26 2012 |
| Time | Replies | Subject |
|---|
| 5:35PM |
2 |
Too many open files |
| 3:46PM |
1 |
SLA for DAHDI FXO - Emulating Key System Functionality |
| 3:23PM |
1 |
Manager Originate and Callerid ? |
| 1:29PM |
0 |
Softphones with SIP transfer |
| 1:22PM |
1 |
User hit f to disconnect call. |
| 12:10AM |
0 |
Dropping incompatible voice frame error |
| |
| Wednesday January 25 2012 |
| Time | Replies | Subject |
|---|
| 3:29PM |
3 |
Executing Script after MixMonitor is called |
| 3:29PM |
4 |
play sound file |
| 1:34PM |
0 |
Blocking in: ast_waitfor_nandfds |
| 11:56AM |
0 |
Signalling and Media Configuration |
| |
| Tuesday January 24 2012 |
| Time | Replies | Subject |
|---|
| 10:29PM |
1 |
Is there a sip show equivelant. |
| 2:34PM |
1 |
allowguest = yes? no? |
| 2:29PM |
5 |
RFE idea for VM application |
| 1:47PM |
4 |
ChanSpy : how to know channel name ? |
| |
| Monday January 23 2012 |
| Time | Replies | Subject |
|---|
| 8:53PM |
1 |
ConfBridge details |
| 7:00PM |
2 |
asterisk does not detect menus |
| 6:16PM |
1 |
Timing Slips CRC & E-Bit Errors - Asterisk - Trixbox 2.8.0.4 |
| 3:48PM |
5 |
SDP Issue |
| 2:35PM |
9 |
Cordless SIP phone |
| 2:28PM |
1 |
Avaya 4610sw IP Phone |
| 2:29AM |
0 |
SIP - connected line has changed. Saving it until answer for IAX2/iaxy |
| 2:02AM |
0 |
Chan_Mobile Nokia E51, csr bt dogle, Voice OK but no SMS Support ? |
| |
| Sunday January 22 2012 |
| Time | Replies | Subject |
|---|
| 9:06PM |
2 |
Analoge and E1 ports |
| |
| Saturday January 21 2012 |
| Time | Replies | Subject |
|---|
| 3:51PM |
1 |
Force CDR to be written. |
| 1:21PM |
1 |
View # active calls in a context |
| |
| Friday January 20 2012 |
| Time | Replies | Subject |
|---|
| 7:44PM |
0 |
10.1.0-rc1 : WARNING: abstract_jb.c:384 jb_get_and_deliver: AST_JB_IMPL_NOFRAME |
| 6:07PM |
3 |
Pickup calls coming from queues |
| 2:07PM |
1 |
Asterisk NOT in the media path |
| 1:36AM |
8 |
Sip Registration Hijacking |
| |
| Thursday January 19 2012 |
| Time | Replies | Subject |
|---|
| 11:25PM |
1 |
Efficient logging of PRI traffic for later analysis? |
| 8:23PM |
0 |
Asterisk 1.8 - SIP losing registration |
| 6:00PM |
0 |
Asterisk 1.8.8.2 and 10.0.1 Now Available (Security Release) |
| 5:40PM |
1 |
AST-2012-001: SRTP Video Remote Crash Vulnerability |
| 4:47PM |
0 |
odd disconnects with major company's voice recog |
| 3:15PM |
0 |
asterisk not connecting to sipgate / NAT related issue? |
| 1:33PM |
1 |
Voicemail weirdness after upgrade |
| 12:08PM |
0 |
Huh? Local is being asked to answer? |
| 9:10AM |
1 |
Asterisk rewrites "From" header when CALLERID(num-pres)=prohib_passed_screen is set |
| 7:05AM |
2 |
Does Asterisk permit multiple registrations to the same host? |
| 3:35AM |
1 |
Asterisk to support Dialogic Cards |
| |
| Wednesday January 18 2012 |
| Time | Replies | Subject |
|---|
| 10:38PM |
0 |
attended transfers going to wrong voicemail Asterisk 1.8 Polycom 650 |
| 5:50PM |
1 |
Installing the 3.1 sources of Kernel with Asterisk |
| 9:24AM |
2 |
Dahdi for meetme on AMD64 arch? |
| 9:19AM |
1 |
Compile error 1.8.8.1 |
| 7:44AM |
3 |
Failed to Allocate port for RTP instance |
| |
| Tuesday January 17 2012 |
| Time | Replies | Subject |
|---|
| 9:06PM |
0 |
SIP trunk call initiated as Anonymous@anonymous.invalid |
| 5:23PM |
2 |
Problem answering phone |
| 4:53PM |
0 |
Is there any way to terminate async origination initialized by AMY? |
| 4:24PM |
1 |
Prepaid billing |
| 2:11PM |
4 |
/etc/init.d script and calling asterisk command line. |
| 10:57AM |
1 |
Macro vs sub |
| 10:54AM |
0 |
pickup group |
| 10:16AM |
2 |
Core file created in /tmp |
| |
| Monday January 16 2012 |
| Time | Replies | Subject |
|---|
| 11:02PM |
0 |
OT - Configuring Freepbx's fax_process.pl to work with ssmtp |
| 6:41PM |
2 |
How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install? |
| 5:52PM |
1 |
Starting things off without a dial tone |
| 5:11PM |
3 |
meetme with IVR |
| 4:02PM |
1 |
Update callee num or name at caller display |
| 3:55PM |
4 |
How Can I configure the between call oneside IVR |
| 3:21PM |
1 |
SayDigits playback doesn't always work |
| 12:56PM |
0 |
Where to find meaning of /n inLocal/6613@from-queue/n ? [SOLVED] |
| 12:41PM |
2 |
Where to find meaning of /n in Local/6613@from-queue/n ? |
| 10:48AM |
2 |
Real T1 trunk group... |
| 10:14AM |
4 |
How to check currently used libraries from command line ? |
| 9:19AM |
0 |
echo & audio delay in SIP VOIP |
| 7:59AM |
2 |
local channels and g729a voice quality |
| 2:14AM |
2 |
Real trunk group w/ DAHDI |
| 1:53AM |
1 |
Peer doesn't answer |
| |
| Sunday January 15 2012 |
| Time | Replies | Subject |
|---|
| 9:06AM |
1 |
ssh to a Cisco 7961 is not working |
| 8:35AM |
0 |
configuring a Cisco 7961 so that different line appearances register to different SIP proxy addresses |
| 4:43AM |
0 |
AstLinux 1.01 Released |
| |
| Saturday January 14 2012 |
| Time | Replies | Subject |
|---|
| 9:38PM |
1 |
CDR into ical? |
| 2:37AM |
1 |
asterisk problem sip |
| 1:25AM |
1 |
Asterisk as UAC: How to put call OnHold |
| |
| Friday January 13 2012 |
| Time | Replies | Subject |
|---|
| 10:07PM |
1 |
Sporadic one way audio problem |
| 10:00PM |
1 |
Queue option 'R' |
| 6:43PM |
0 |
Preços por serviços e equipamento |
| 4:09PM |
0 |
Stuck DAHDI Channels |
| 12:45PM |
2 |
SIP hardphone with dual gigabit ethernet ports |
| 12:41PM |
1 |
dialplan problem : not including context |
| 9:56AM |
1 |
odbc storage for video message |
| 9:47AM |
0 |
Queue member is permanently BUSY |
| |
| Thursday January 12 2012 |
| Time | Replies | Subject |
|---|
| 6:57PM |
0 |
t38modem v2, which version or patch of asterisk? |
| 12:39PM |
1 |
Questions on hardware or software-based echo cancellation |
| 11:51AM |
3 |
Call abandoned from queue not showing in CDR (possible bug) |
| 11:44AM |
5 |
FAX Installation in Asterisk |
| 9:50AM |
0 |
Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ? [SOLVED] |
| 8:44AM |
1 |
how to set callerid in php AGI file. |
| |
| Wednesday January 11 2012 |
| Time | Replies | Subject |
|---|
| 11:10PM |
1 |
Problems with codec translation when using Monitor and MixMonitor |
| 8:37PM |
2 |
Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1 |
| 7:56PM |
1 |
Exceptionally long voice queue length |
| 3:05PM |
1 |
Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED] |
| 2:13PM |
0 |
OT - Which iceweasel plugin to play gsm sound files ? |
| 12:01PM |
0 |
Iax hold events in AMI 1.1 |
| 11:29AM |
2 |
SIP and NAT best practices since recent changes? |
| 10:05AM |
1 |
Q: SIPNATtraversal.pdf |
| 9:09AM |
2 |
Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? |
| 7:04AM |
1 |
Problems faced in load testing of asterisk |
| 6:39AM |
5 |
Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ? |
| 4:35AM |
2 |
No audio available on SIP/172.16.129.13:5060-00000001?? |
| |
| Tuesday January 10 2012 |
| Time | Replies | Subject |
|---|
| 11:00PM |
1 |
Odd DTMF problem when receiving calls |
| 9:12PM |
1 |
Linux Stun Server |
| 5:02PM |
2 |
Hang up phone after declined attended transfer |
| 4:47PM |
0 |
Noise in caller handset when dialing out (with dahdi 2.6.0) [SOLVED] |
| |
| Monday January 9 2012 |
| Time | Replies | Subject |
|---|
| 7:59PM |
1 |
44Khz files in Asterisk 10 |
| 6:54PM |
1 |
message WARNING features.c: Failed to play transfer sound! and attended transfer hangs up |
| 6:52PM |
2 |
Noise in caller handset when dialing out (with dahdi 2.6.0) |
| 4:46PM |
2 |
Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk? |
| 10:20AM |
0 |
DEBUG Message |
| 8:20AM |
1 |
Cisco AS5300 and Digium g729A codec |
| 8:02AM |
1 |
Asterisk as register server through OpenSIPS |
| 7:44AM |
2 |
create table in mysql using asterisk |
| 6:20AM |
1 |
video mail is not store |
| |
| Sunday January 8 2012 |
| Time | Replies | Subject |
|---|
| 6:26PM |
1 |
Answering call from queue, then put back in queue? |
| 12:30PM |
1 |
cached VMI on manual voicemail update |
| |
| Saturday January 7 2012 |
| Time | Replies | Subject |
|---|
| 1:39PM |
1 |
Couple of questions: SIP ALG, allowguest=no |
| 6:15AM |
2 |
Asterisk 10.0 & 1.4 - iax codec are not compatible |
| |
| Friday January 6 2012 |
| Time | Replies | Subject |
|---|
| 10:00PM |
7 |
best softphone for 2012? |
| 6:51PM |
1 |
Why write your dialplan using Lua? |
| 5:56PM |
0 |
Streaming Music to 75 callers .. |
| 4:17PM |
2 |
Change port from 5060 on Snom phone |
| 1:14PM |
3 |
Connecting to an Old Phone System |
| 7:32AM |
0 |
no audio using g729A for Cisco AS5300 sip peer |
| 1:45AM |
1 |
calling specific 1800-number not going through. |
| 12:55AM |
0 |
which choice: asterisk-gui or freepbx? |
| |
| Thursday January 5 2012 |
| Time | Replies | Subject |
|---|
| 11:42PM |
4 |
asterisk 1.8.8 - caller ID not working. |
| 10:30PM |
1 |
STOP loading extensions.ael |
| 8:03PM |
1 |
Blind transfers being cancelled by asterisk & hanging up on remote caller |
| 6:22PM |
1 |
question on CDR |
| 5:19PM |
3 |
Best non polycom SIP conference room phone |
| 2:54PM |
2 |
asterisk -> AGI (perl) -> sqlplus(oracle) |
| 10:12AM |
1 |
Asterisk1.8 support video trancoding ? |
| 9:35AM |
1 |
Video trancoding not done. |
| 7:03AM |
1 |
Where are the fax instructions? |
| |
| Wednesday January 4 2012 |
| Time | Replies | Subject |
|---|
| 9:45PM |
1 |
DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released |
| 8:47PM |
1 |
question sangoma vs digium |
| 7:44PM |
1 |
From address missing 'sip:', using it anyway |
| 4:45PM |
2 |
asterisk -> AGI (perl) -> sqlplus (oracle) |
| 4:19PM |
0 |
Which QSIG variant and profiles does asterisk support ? |
| 11:53AM |
6 |
Set Call Codec in extension.conf |
| 10:13AM |
2 |
Asterisk won't start - trap invalid opcode |
| 9:37AM |
2 |
Failed to authenticate on INVITE to Anonymous |
| 9:19AM |
1 |
Rami |
| 6:25AM |
3 |
Anyone have a reliable T.38 Solution |
| 4:14AM |
1 |
ConfBridge no audio problem |
| 2:42AM |
4 |
Speech recognition in asterisk using google voice API |
| 2:42AM |
1 |
Mark queue agent as away |
| |
| Tuesday January 3 2012 |
| Time | Replies | Subject |
|---|
| 10:41PM |
1 |
ISDN E1, after electrical disconnected is not becoming UP, IRQ misses: 1 |
| 7:30PM |
1 |
Layer2 Down in BRI connection |
| 6:06PM |
1 |
Ringing agents cell as an alert? |
| 5:53PM |
3 |
Problem connecting to 4569/UDP |
| 3:24PM |
3 |
NAT/IPTABLES workarounds |
| 2:20PM |
0 |
Registering multi-clients |
| 1:48PM |
4 |
Question on system command 1.4.43 |
| 1:47PM |
4 |
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. |
| 1:34PM |
1 |
Using Asterisk as a softphone |
| 11:08AM |
7 |
How to make SIP guest call |
| 10:22AM |
2 |
Asterisk 1.8 - BRI D Channel going up and down every few seconds |
| 9:07AM |
2 |
dialplan -> dial command -> custom ringtone |
| 12:38AM |
1 |
Problem w/ PC port on Polycom 335 |
| |
| Monday January 2 2012 |
| Time | Replies | Subject |
|---|
| 7:57PM |
0 |
Wired attack on Asterisk - Can anyone explain this? |
| 11:38AM |
0 |
Help_video voice mail not retriev properly |
| 9:07AM |
3 |
Set Call type in dial plan |
| 4:34AM |
1 |
tcp version of toronto - osaka doesn't work |
| |
| Sunday January 1 2012 |
| Time | Replies | Subject |
|---|
| 10:17PM |
2 |
asterisk 1.8 codec negotiation |
| 10:11PM |
0 |
GoAutoDialer, ViciDial and Vicidial group |
| 12:22PM |
0 |
481 Call leg/transaction does not exists Status Response |