| Sunday July 31 2011 |
| Time | Replies | Subject |
|---|
| 11:04PM |
2 |
sip attacks |
| 11:48AM |
2 |
Codec translation from gsm to other codecs or from other codecs to gsm |
| 5:22AM |
1 |
asterisk + sccp-b problem |
| |
| Friday July 29 2011 |
| Time | Replies | Subject |
|---|
| 11:29PM |
0 |
Tutorial on the Asterisk Manager Interface |
| 10:12PM |
1 |
Serious bug in 1.6.2.19 - what is the time frame to fix such bugs? |
| 3:51PM |
1 |
Accept the dtmf input in call patch |
| 10:56AM |
2 |
How to use these feature of Asterisk |
| 10:56AM |
2 |
X86_64 Compilation Issue |
| 9:56AM |
0 |
Asterisk SIP authentication against [section] instead of username |
| 6:56AM |
1 |
call forwarding number from outside. |
| |
| Thursday July 28 2011 |
| Time | Replies | Subject |
|---|
| 11:05PM |
1 |
Dialplan required for recording |
| 8:22PM |
1 |
Questions about FMFM with linked servers |
| 7:47PM |
2 |
Disabling Polycom "reject" and "DND" or disable Asterisk 486 "Busy Here" actions |
| 6:42PM |
2 |
Voicemail not acting as documented. |
| 2:53PM |
5 |
MoH - conversion command |
| 1:22PM |
3 |
hide google voice number |
| 12:45PM |
3 |
Capturing call Reject/Decline events on a PRI line |
| 9:19AM |
0 |
[chan_mobile addons] DTMF transfer from calling mobile to Asterisk through called mobile FAILED |
| 7:47AM |
0 |
Radius billing for asterisk |
| 6:45AM |
0 |
Avaya & Asterisk FreePBX Integration Problem |
| 6:32AM |
2 |
Connect asterisk to normal telephone PBX |
| |
| Wednesday July 27 2011 |
| Time | Replies | Subject |
|---|
| 6:41PM |
2 |
Lightning and thunder (Claude Hayn |
| 6:32PM |
2 |
Problem H323 asterisk 1.6.2.19 |
| 4:33PM |
0 |
Fwd: Re: Securing Asterisk |
| 1:44PM |
5 |
Lightning and thunder |
| 1:29PM |
2 |
Stun Server |
| 11:59AM |
0 |
AMI action PlayDTMF and SIP:INFO |
| 1:02AM |
0 |
libpri rpm version 1.4.12 for CentOS 5.6 |
| |
| Tuesday July 26 2011 |
| Time | Replies | Subject |
|---|
| 7:21PM |
3 |
file2ban |
| 6:20PM |
1 |
Scheduling destruction of SIP dialog |
| 2:13PM |
2 |
Browser based SIP UA |
| 1:19PM |
1 |
NAT yes |
| 11:47AM |
0 |
Callback + DISA |
| 7:21AM |
9 |
Why no traction for Windows version? |
| 6:46AM |
2 |
MusicOnHold not loaded |
| |
| Monday July 25 2011 |
| Time | Replies | Subject |
|---|
| 6:10PM |
0 |
Malformed/missing URL |
| 5:05PM |
0 |
Registration problems, Linksys SPA 3102 on Asterisk 1.4.20 |
| 1:31PM |
0 |
Is there a protocol to be used to communicate between different Asterisk Boxes to distribute load and tasks |
| 1:22PM |
0 |
Is there a protocl that let the Asterisk boxes talk to each other and treated as one entity? |
| 1:18PM |
1 |
callgroup and pickupgroup (Carlos Chavez) |
| 1:09PM |
0 |
The queue is not routing for the agent: returned -1: |
| 12:59PM |
0 |
Meaning Callerid Datatypes. |
| 11:19AM |
2 |
Fail to compile DAHDI under Linux 3.0.0 |
| 9:22AM |
0 |
use of .exports file in asterisk |
| 8:43AM |
1 |
dahdi channels busy/congested |
| |
| Sunday July 24 2011 |
| Time | Replies | Subject |
|---|
| 11:09PM |
2 |
Error Code 101 |
| 3:09PM |
0 |
One way calling on asterisk to cisco |
| 1:09AM |
1 |
Security questions |
| |
| Saturday July 23 2011 |
| Time | Replies | Subject |
|---|
| 5:55PM |
1 |
One way calling on asterisk to cisco call manager integration |
| 5:38PM |
9 |
Securing Asterisk |
| 5:18PM |
1 |
The queue is not routing for the agent: returned -1: Invalid argument |
| 2:30PM |
1 |
dialplan pattern help |
| 9:51AM |
2 |
DISA password |
| |
| Friday July 22 2011 |
| Time | Replies | Subject |
|---|
| 11:32PM |
2 |
Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined" |
| 11:13PM |
1 |
use dahdi for local terminal modem access? |
| 9:27PM |
0 |
Asterisk 10.0.0 Beta 1 Now Available! |
| 7:44PM |
4 |
10.0.0 better than 2.0.0? |
| 5:09PM |
1 |
Connecting to a Taqua switch |
| 3:19PM |
1 |
Phase out macro command. |
| 2:39PM |
4 |
Question about codec re-negotiation in asterisk 1.4.X |
| 9:27AM |
1 |
Pickup(${EXTEN:2}); not works from outside |
| 8:57AM |
0 |
[Fwd: Re: Strange network issue] |
| 8:10AM |
1 |
asterisk rpm build problem |
| 7:26AM |
4 |
Asterisk as a Operator Phone |
| |
| Thursday July 21 2011 |
| Time | Replies | Subject |
|---|
| 11:13PM |
3 |
Strange network issue |
| 11:09PM |
0 |
Per-line registration |
| 10:49PM |
1 |
Rebooting a Grandstream |
| 8:30PM |
1 |
asterisk's SDP |
| 9:31AM |
4 |
Functions not autoloading |
| 9:05AM |
0 |
Asterisk doesn't like OpenBTS!!! |
| 8:06AM |
1 |
Prevent Asterisk from setting CALLERID(name) or unsetting CALLERID(name) |
| 5:29AM |
6 |
My Asterisk Box was hacked |
| |
| Wednesday July 20 2011 |
| Time | Replies | Subject |
|---|
| 10:28PM |
1 |
Multiple SIP trunks between same pair of asterisk box |
| 6:29PM |
2 |
ISAC and Asterisk |
| 5:57PM |
0 |
>64 pickup groups |
| 5:53PM |
2 |
"Problems" with System() application |
| 1:42PM |
0 |
Call flow attached |
| 1:25PM |
3 |
HELP - Client wants to reverse Asterisk Functionality |
| 1:09PM |
0 |
ilbc codec |
| 12:16PM |
3 |
Macro to Dial a Channel Group using Round-robin |
| 9:00AM |
3 |
Help: How can I Add my own Word in option packets in from field of SIP "From Asterisk??" |
| |
| Tuesday July 19 2011 |
| Time | Replies | Subject |
|---|
| 7:48PM |
1 |
SS7 and PRI compatibility |
| 6:07PM |
6 |
Multiple Asterisk Sessions on same machine |
| 4:53PM |
1 |
Recall: Time zone on phones |
| 4:53PM |
1 |
Time zone on phones |
| 4:24PM |
1 |
Is there a protocol to be used to communicate between different Asterisk Boxes to distribute load and tasks? |
| 3:53PM |
1 |
Is there a protocol that let the Asterisk boxes talk to each other and treated as one entity? |
| 12:50PM |
2 |
max one sip peer to register |
| 10:23AM |
1 |
callgroup and pickupgroup |
| 7:56AM |
0 |
Converged phones; akin to SwitchVox ? |
| 7:24AM |
1 |
AsteriskNow install addons despite license conflict with FFA and G.729 |
| 7:20AM |
3 |
a=sendonly Music On Hold ignored |
| |
| Monday July 18 2011 |
| Time | Replies | Subject |
|---|
| 11:03PM |
2 |
What is the use for the agent password if login via exten |
| 10:05PM |
1 |
libss7 variables |
| 4:42PM |
1 |
chan_gtalk load error |
| 2:15PM |
2 |
No Audio after attended tranfer |
| 1:20PM |
5 |
[1.4] Minimal installation? |
| 11:20AM |
3 |
FAX with SIP |
| 11:03AM |
4 |
Seg Faults with 1.6.2.19 |
| 10:14AM |
3 |
Compact, affordable x86 devices? |
| |
| Saturday July 16 2011 |
| Time | Replies | Subject |
|---|
| 2:19PM |
1 |
asterisk 1.6 agi problem with PHP |
| 12:58PM |
3 |
Requires |
| 12:57AM |
0 |
Unable to register an endpoint after upgrading from 1.4.2.20 to 1.8.3.1 |
| |
| Friday July 15 2011 |
| Time | Replies | Subject |
|---|
| 5:52PM |
3 |
Macro issue under 1.8.5 |
| 4:47PM |
4 |
Using Firewall to protect Asterisk |
| 12:29PM |
1 |
Controlling max simultaneous calls for a group/.call files |
| 10:11AM |
3 |
Redirecting call from one E1 to another? |
| 8:03AM |
2 |
*8 causing large number of channels to go stale (possible bug) |
| 8:02AM |
0 |
dialplan: all extern, except |
| |
| Thursday July 14 2011 |
| Time | Replies | Subject |
|---|
| 10:57PM |
4 |
Asterisk in the amazon cloud |
| 8:13PM |
0 |
AstLinux 0.7.9 Release |
| 5:37PM |
1 |
Enable T.38 |
| 2:40PM |
3 |
Asterisk binaries on CentOS version 6 |
| 1:46PM |
0 |
cseq decreasing => 500 Server Error |
| 11:27AM |
0 |
How to Get all users list in asterisk |
| 4:45AM |
9 |
Extension wise dialplan |
| |
| Wednesday July 13 2011 |
| Time | Replies | Subject |
|---|
| 10:19PM |
0 |
Chan_mobile |
| 6:51PM |
2 |
TDM400p susceptible to EMI? |
| 9:35AM |
1 |
How to Hang up a stale SIP channel? |
| 7:02AM |
1 |
Connect Avaya to Asterisk PBX |
| 6:01AM |
0 |
(no subject) |
| 4:23AM |
1 |
Problem on Dialling-out |
| |
| Tuesday July 12 2011 |
| Time | Replies | Subject |
|---|
| 8:33PM |
2 |
Mysterious dropped calls |
| 5:30PM |
0 |
Dtmf issues solved |
| 4:40PM |
1 |
REALY strange issue with making calls biside 2 phones |
| 3:40PM |
0 |
Park/VoiceMail on DAHDI congestion |
| 3:33PM |
3 |
CDRs |
| 2:48AM |
3 |
skype for asterisk usage in the future |
| |
| Monday July 11 2011 |
| Time | Replies | Subject |
|---|
| 9:43PM |
0 |
Asterisk 1.8.5.0 Now Available |
| 9:29PM |
4 |
Benchmarking AGI performance in C, PHP, and Perl |
| 9:01PM |
1 |
${HASH(SIP_CAUSE, ...)} and peer name |
| 5:11PM |
1 |
keeping asterisk memory |
| 1:20PM |
0 |
RFC 6315: IANA Registration for Enumservice 'iax' |
| |
| Sunday July 10 2011 |
| Time | Replies | Subject |
|---|
| 10:53PM |
1 |
How to logout! |
| 10:36PM |
1 |
What is the use for the agent password if login via exten? |
| 6:26PM |
0 |
Help anybody - how to manage SRTP with TLS trasport |
| 3:02PM |
2 |
Problem with setting up fresh 1.8.5 Asterisk |
| 11:04AM |
4 |
Queue Issue : Duration between 2 agents call |
| 9:53AM |
3 |
References customers |
| 8:52AM |
2 |
Thomson ST022 - External Call problems |
| |
| Saturday July 9 2011 |
| Time | Replies | Subject |
|---|
| 6:55PM |
0 |
About recompile reinstall couse of SRTP |
| 4:57PM |
0 |
Auto Reply: asterisk-users Digest, Vol 84, Issue 15 |
| 3:31PM |
2 |
Meetme not prompting for PIN |
| 1:31PM |
0 |
Scheduled Maintenance for Asterisk Project Services |
| 11:34AM |
4 |
OT: Google Plus |
| 9:45AM |
0 |
Strange behavior with Asterisk TLS |
| |
| Friday July 8 2011 |
| Time | Replies | Subject |
|---|
| 11:28PM |
11 |
New VirtualBox Beta Has PCI Pass-Through Support |
| 9:49PM |
2 |
DTMF issues still |
| 9:29PM |
0 |
Cisco ATA 187 configuration file |
| 6:46PM |
0 |
Problems with DTMF Caller ID |
| 4:39PM |
3 |
How to create a module |
| 4:19PM |
0 |
Asterisk meetme and Timer ? |
| 3:01PM |
1 |
Issue 0019268 Patch Asterisk |
| 2:58PM |
2 |
FXO ports locking up |
| 1:13PM |
2 |
Master.csv file limit |
| 10:42AM |
4 |
timeout with outbound calls |
| 10:23AM |
2 |
dialout time configuration |
| 8:48AM |
3 |
DB Driven IVR |
| 8:48AM |
0 |
asterisk and eyebeam |
| 8:42AM |
0 |
HI |
| 6:58AM |
4 |
Problem in Detecting Dtmf on FXO line. |
| |
| Thursday July 7 2011 |
| Time | Replies | Subject |
|---|
| 8:41PM |
4 |
Anybody doing PRI over IP? |
| 7:43PM |
0 |
Installing Asterisk from repository works great without the need to install Dahdi on Host Node of Proxmox - But trying to install from source fails. Why? |
| 5:17PM |
1 |
Eyebeam crashes when dialing an invalid number... |
| 4:43PM |
4 |
Stripping characters from ${CALLERID(num)} ? |
| 4:22PM |
1 |
check_auth: username mismatch |
| 4:01PM |
0 |
No pattern 407 from SIP provider iCall |
| 1:53PM |
0 |
asterisk 1.8X problem: no outbound callerid set in callfiles |
| |
| Wednesday July 6 2011 |
| Time | Replies | Subject |
|---|
| 9:48PM |
1 |
ooh323 does not work fine, what about h323 channel |
| 5:11PM |
0 |
libpri 1.4.12 Now Available |
| 12:02PM |
3 |
Monitoring connection to VoIP provider? |
| 9:36AM |
2 |
Agents outbound calls to be recorded |
| 8:37AM |
2 |
single keypress short-circuits to invalid extension handler |
| 12:30AM |
2 |
Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system |
| |
| Tuesday July 5 2011 |
| Time | Replies | Subject |
|---|
| 8:46PM |
1 |
Couldn't call Agent and segfault |
| 5:47PM |
0 |
OT - Polycom - 2 localization file versions on the same TFTP server |
| 3:26PM |
1 |
Recording SIP history |
| 2:59PM |
0 |
Can't get video on one server of 4 |
| 2:07PM |
2 |
Cant find asterisk src dir for FreePBX full distro |
| 12:22PM |
0 |
AST_DEVICE_UNAVAILABLE vs. AST_DEVICE_UNKNOWN for new loaded realtime peers |
| 9:25AM |
1 |
More SQL Querys in dialplan |
| 6:59AM |
2 |
realm question |
| 6:45AM |
1 |
SIP Presence not working |
| 5:22AM |
1 |
Blind Transfer Connected |
| 4:06AM |
0 |
DTMF between sip trunks and PRIs |
| |
| Monday July 4 2011 |
| Time | Replies | Subject |
|---|
| 11:36PM |
1 |
Agent Login, Logout, Ready, Not Ready from the CTI application |
| 7:29PM |
1 |
Testing Asterisk with media - sipp |
| 4:51PM |
0 |
RINGNOANSWER events in queue log |
| 10:58AM |
4 |
stream rtp from asterisk |
| 9:40AM |
1 |
Mixmonitor concept's question |
| 5:40AM |
1 |
how to set to make a call through a fixed ip on a 2 ips server? |
| |
| Sunday July 3 2011 |
| Time | Replies | Subject |
|---|
| 12:41AM |
1 |
SIP Peer Name Variable |
| |
| Saturday July 2 2011 |
| Time | Replies | Subject |
|---|
| 10:48PM |
2 |
chanspy spies on wrong channel |
| 4:58PM |
2 |
Distributing the incoming calls and the huntgroup |
| 12:38AM |
0 |
inter asterisk-user list |
| |
| Friday July 1 2011 |
| Time | Replies | Subject |
|---|
| 8:09PM |
0 |
RINGNOANSWER IN queue_log |
| 7:35PM |
0 |
Queue transfer order |
| 6:59PM |
0 |
Using Asterisk as Contact Center: Concurrent Calls + How much time |
| 5:55PM |
2 |
How to without GUI |
| 5:43PM |
0 |
IVR sound after dial sip |
| 3:10PM |
0 |
participants redirection between conferences |
| 2:36PM |
1 |
calleridname presentation Asterisk => Siemens |
| 2:26PM |
1 |
Dropping Conference calls |
| 1:40PM |
0 |
Anyone worked with allo products? |
| 10:03AM |
0 |
Asterisk 1.6.2.19 RPM |
| 9:29AM |
1 |
error in GUI access |
| 5:43AM |
1 |
Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted |
| 5:34AM |
0 |
Poll : libpri naming scheme and asterisk-libpri merge |
| 12:06AM |
3 |
Asterisk 1.6.1 Realtime SIP Users |