| Thursday September 30 2010 |
| Time | Replies | Subject |
|---|
| 4:31PM |
0 |
Friday 12 Noon EDT: VoIP Abuse Project |
| 4:08PM |
0 |
Same extension on multiple servers confusion |
| 2:51PM |
2 |
Unable to load fax modules |
| 2:03PM |
2 |
Intercom with Dial() works, but not with Page() |
| 1:52PM |
0 |
Unscheduled service outage for various Asterisk community services |
| 12:57PM |
1 |
channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI |
| 12:00PM |
1 |
Asterisk 1.6.2.13 Audio Prompts Stopping |
| 10:16AM |
2 |
Asterisk 1.6.2.10 Internal timing |
| 10:09AM |
3 |
Kernel Panic When restarting the server |
| 7:36AM |
0 |
SIP Registrations |
| 7:30AM |
3 |
Go from *100* to just 100 |
| |
| Wednesday September 29 2010 |
| Time | Replies | Subject |
|---|
| 8:57PM |
1 |
can't get libpri/PRI to work, missing PRI commands |
| 6:57PM |
0 |
Successive Dial apps give hang up within 30s!! |
| 3:52PM |
0 |
asterisk > cisco gateway > westell > isdx |
| 2:13PM |
2 |
Alert-Info advice |
| 12:50PM |
0 |
(no subject) |
| 12:49PM |
0 |
Jin. |
| 12:39PM |
1 |
Weird Behavior with DAHDI |
| 9:56AM |
4 |
DAHDI FXO port only recognizes the "S" extension |
| 2:06AM |
4 |
Use modprobe to find E1/T1 jumper setting on PRI card |
| |
| Tuesday September 28 2010 |
| Time | Replies | Subject |
|---|
| 1:22PM |
2 |
E1 check with nagios, how to? |
| 12:05PM |
2 |
SIP X.25 |
| 11:51AM |
1 |
What's the meaning of this? |
| 11:15AM |
1 |
Inbound calls from TRUNK |
| 10:52AM |
3 |
ISDN - Busy signal on 3rd call |
| 8:02AM |
1 |
1.6 and 1.8 version & A2Billing |
| 5:05AM |
0 |
AstLinux 0.7.3 released |
| 1:28AM |
2 |
NAT issue (i think?) |
| |
| Monday September 27 2010 |
| Time | Replies | Subject |
|---|
| 6:02PM |
1 |
How to pick a codec on the fly |
| 6:00PM |
2 |
SCCP (skinny) phone behind NAT: RTP dest addr wrong |
| 4:09PM |
8 |
Problems compiling Asterisk on Debian |
| 4:02PM |
1 |
propagate sip reinvites with directrtpsetup=yes |
| 3:26PM |
0 |
groupcount - show usage |
| 1:57PM |
0 |
Asterisk and dahdi on Arch linux |
| 10:37AM |
1 |
RFC3329 support in Asterisk |
| 8:08AM |
0 |
PSTN to SMS and SMS to PSTN |
| 3:04AM |
1 |
misc newbie VoIP questions |
| |
| Sunday September 26 2010 |
| Time | Replies | Subject |
|---|
| 6:08PM |
2 |
Asterisk ODBC Insert issue |
| 5:48PM |
5 |
Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server... |
| 5:22PM |
1 |
Downloading the Asterisk as tar.gz file |
| 5:00PM |
2 |
1.4, 1.6, 1.8 versions and the H323 gatekeeper functionality |
| 3:04PM |
3 |
Asterisk Redundancy |
| |
| Saturday September 25 2010 |
| Time | Replies | Subject |
|---|
| 10:22AM |
0 |
Asterisk Cluster Scenario |
| 9:53AM |
0 |
asterisk-users@lists.digium.com September 67% OFF |
| 2:09AM |
0 |
can call internal branch , but can not call external numbers with avaya , always get return message : Q931IncompatibleDestination |
| |
| Friday September 24 2010 |
| Time | Replies | Subject |
|---|
| 7:40PM |
5 |
Asterisk 1.62.13 - CPU spikes every 10 minutes |
| 4:52PM |
2 |
best format for playback/generation |
| 2:55PM |
3 |
should trixbox system hang when ISP drops connection? |
| 2:47PM |
2 |
Debug compile fails |
| 1:52PM |
0 |
Fwd: Can't cross compile asterisk 1.6.2.13 on arm using ltib |
| 1:13PM |
4 |
differential billing |
| 9:28AM |
1 |
RDNIS not passed from one box to another with BRI access |
| 9:11AM |
1 |
tcpdump auto stats script |
| 9:05AM |
5 |
How to test BIG traffic through DAHDI/WANPIPE interfaces |
| |
| Thursday September 23 2010 |
| Time | Replies | Subject |
|---|
| 11:01PM |
1 |
Asterisk 1.6.2.13 - have asterisk reply from same IP address |
| 9:40PM |
2 |
rtp problem with 1.8.0-rdc1 |
| 8:14PM |
2 |
Asterisk 1.8.0 Release Candidate 2 Now Available |
| 7:39PM |
0 |
Unable to make outgoing call on E1 |
| 5:56PM |
1 |
OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? (bruce bruce) |
| 4:25PM |
0 |
Asterisk Transfer/call patching support |
| 4:09PM |
1 |
Forking a call |
| 2:59PM |
0 |
Sip from ip address |
| 2:06PM |
1 |
Can't turn debug on in a 1.2 box |
| 2:03PM |
1 |
Net2Phone SIP trunk problem |
| 11:48AM |
4 |
Asterisk and Digium TC400B |
| 11:15AM |
3 |
realm: security issue |
| 10:10AM |
2 |
CDR display in minute |
| 6:21AM |
8 |
Record() Cmd and My SQL |
| 5:19AM |
0 |
Calls stuck in the queue even when ext's are available |
| 4:25AM |
0 |
Installing Asterisk + FreePBX from Repsitory spits out some warnings and errors for ever |
| |
| Wednesday September 22 2010 |
| Time | Replies | Subject |
|---|
| 9:27PM |
2 |
Recording maximum time and stop on silence |
| 8:59PM |
1 |
Asterisk- speech to text(Voicemail to text message) |
| 6:05PM |
0 |
Sangoma A500 NT BRI PTMP without woomera on asterisk 1.6 |
| 4:49PM |
4 |
Asterisk as a distributed paging system |
| 3:45PM |
5 |
http://www.asterisk.org/downloads naming schema |
| 2:26PM |
0 |
TLS re-negotiation attack on SIP/TLS of Asterisk? |
| 2:05PM |
1 |
Costa Rica Hangup Detection |
| 2:00PM |
2 |
Asterisk T38 |
| 1:21PM |
2 |
Can't cross compile asterisk 1.6.2.13 on arm using ltib |
| 11:58AM |
1 |
T38 and codecs negotiation |
| 10:32AM |
2 |
Unable to open vm-INBOXs |
| 9:42AM |
1 |
Cross compile Asterisk for mipsel-linux |
| 5:27AM |
5 |
OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? |
| |
| Tuesday September 21 2010 |
| Time | Replies | Subject |
|---|
| 9:56PM |
0 |
Mixing ISDN and R2 in the same card... |
| 9:17PM |
1 |
digits in chan_dahdi |
| 3:48PM |
1 |
random hangups on RBS T1 |
| 1:42PM |
3 |
Unexplained message in 1.6.2 |
| 9:33AM |
0 |
Dialplan extension pattern matching for '/' character |
| 9:25AM |
6 |
func SHARED, how to use? |
| 1:56AM |
0 |
Asterisk News Accepting Submissions |
| |
| Monday September 20 2010 |
| Time | Replies | Subject |
|---|
| 5:02PM |
0 |
Commands needed via AMI to find callerid of inbound call to extension |
| 4:15PM |
1 |
Authentication best practice |
| 4:08PM |
0 |
Asterisk stops processing SIP UDP messages |
| 4:01PM |
1 |
Setting 'fname_base' variable doesn't affect 'automon' result file. |
| 12:51PM |
2 |
Playing Audio To One Channel |
| 11:31AM |
3 |
Extension continues ringing after caller hanged up |
| 8:15AM |
1 |
Confused about notifyringing in sip.conf |
| |
| Saturday September 18 2010 |
| Time | Replies | Subject |
|---|
| 10:46AM |
2 |
Audiocode Median 2000 Gateway with Asterisk ? |
| 4:12AM |
1 |
Asterisk sip attack |
| |
| Friday September 17 2010 |
| Time | Replies | Subject |
|---|
| 11:54PM |
1 |
externip/localnet |
| 6:43PM |
0 |
quick 1.8 question on console/dsp |
| 6:00PM |
0 |
5-7 second delay in connecting outgoing FXO calls |
| 5:51PM |
2 |
3rd party app store |
| 5:29PM |
1 |
Rotary phone on Asterisk |
| 5:10PM |
3 |
do carriers detect unusual / unauthorized VoIP calling patterns? |
| 5:08PM |
0 |
CallerId: behavior changed between 1.4.25.1 and 1.4.36 with .call files |
| 4:59PM |
0 |
Asterisk 1.8 and CEL logging |
| 3:37PM |
0 |
need help with IVR dialplan |
| 3:25PM |
0 |
ISDN BRI call disconnection issue |
| 3:00PM |
5 |
Initial Audio Cut off |
| 1:24PM |
4 |
Not able to join conference |
| 11:13AM |
2 |
Call restriction for particular extension |
| 11:02AM |
3 |
Sangoma A108 PCIe V2.0 |
| 11:00AM |
0 |
Sangoma A108 PCIe 2.0 |
| 9:50AM |
1 |
Issue with transfer (sip) |
| 9:31AM |
1 |
Attended Transfer does not release channels |
| 8:57AM |
0 |
Deadlock rendering sip useless |
| |
| Thursday September 16 2010 |
| Time | Replies | Subject |
|---|
| 6:44PM |
5 |
AGI Delimiter in 1.6 |
| 5:58PM |
4 |
one way audio for xlite clients behind NAT |
| 5:03PM |
1 |
Help!! Call waiting issue |
| 3:42PM |
4 |
[OT-FreePBX] Outbound calls check inbound routes to see if destination is local? |
| 2:26PM |
0 |
Indications and tonelist on a SIP channel.. |
| 1:33PM |
0 |
DTMF tones too long, for once |
| 11:59AM |
1 |
How to Understand a pri intense debug span X |
| 11:56AM |
2 |
Realtime semi-colon |
| 10:15AM |
3 |
Configure Asterisk with openssl |
| 7:09AM |
0 |
asterisk 1.6 and BLF |
| 12:24AM |
5 |
a2billing |
| 12:20AM |
3 |
Purpose of qualify=yes |
| |
| Wednesday September 15 2010 |
| Time | Replies | Subject |
|---|
| 10:10PM |
3 |
Echo on Sangoma A400 and background noise |
| 7:41PM |
6 |
Bug with Realtime? |
| 7:31PM |
1 |
Queue member status not changing |
| 6:02PM |
0 |
Asterisk 1.6.2.13 Now Available (Re-Releast of 1.6.2.12) |
| 3:54PM |
1 |
Asterisk 1.6.2.12 Download |
| 3:24PM |
0 |
Asterisk 1.6.2.12 Now Available |
| 3:23PM |
0 |
Asterisk 1.4.36 Now Available |
| 3:09PM |
1 |
Error loading skype_for_asterisk |
| 3:06PM |
3 |
changing from zap to DAHDI |
| 3:04PM |
2 |
Problems with audio |
| 2:47PM |
2 |
Dual WAN with load balancing |
| 12:20PM |
2 |
incoming call FXO |
| 11:14AM |
2 |
Help me Out!!!! |
| 10:12AM |
1 |
One way audio when overlapdial is set to yes |
| 10:05AM |
3 |
Skip Busy Agents/Channels from Queue |
| 3:19AM |
2 |
Digest Username/auth name mismatch |
| 2:22AM |
0 |
about yahoo messager work with asterisk |
| |
| Tuesday September 14 2010 |
| Time | Replies | Subject |
|---|
| 7:56PM |
6 |
How different is implementing Cisco based system than Asterisk based system? |
| 7:28PM |
1 |
conf checkout |
| 5:33PM |
2 |
DTMF |
| 4:32PM |
5 |
sip show channels |
| 3:11PM |
0 |
question on asterisk 1.8 meetme |
| 2:19PM |
2 |
SIP 800 Origination/Termination - International |
| 1:44PM |
9 |
Random File Name |
| 12:30PM |
6 |
Spontaneous reboots on asterisk 1.6.2.11 |
| 9:59AM |
1 |
can asterisk accept "anonymous" register ? |
| 7:06AM |
0 |
3xx redirect response list Noop and capture |
| 6:27AM |
2 |
OT - Gigaset C470IP - How to access SMS settings |
| 5:01AM |
9 |
Speech To Text on linux with asterisk |
| 3:29AM |
1 |
Which 1.6 subversion is Stable one? |
| |
| Monday September 13 2010 |
| Time | Replies | Subject |
|---|
| 3:47PM |
2 |
PostgreSQL is asterisk friendly with it? |
| 3:22PM |
5 |
Force ip disconnect after register? |
| 10:34AM |
1 |
Changing voicemail.conf file format list |
| 10:29AM |
0 |
Upgrade from 1.4 to 1.6 : problems with realtimemysql |
| 9:57AM |
1 |
Which voicemail file format is the most widely understood ? |
| 9:56AM |
7 |
High volume BLF - Suggestions? |
| 9:04AM |
2 |
How to send SMS to Gigaset phones ? |
| 8:26AM |
2 |
Correct queue agi syntax in 1.6.2.11 |
| 6:38AM |
3 |
doing dnsmgr_lookup |
| |
| Sunday September 12 2010 |
| Time | Replies | Subject |
|---|
| 7:47PM |
1 |
Synway cards |
| 7:41PM |
1 |
First boot asterisk -vvvvvgcn segfaults |
| 5:43PM |
3 |
Moving from DSL to T1 |
| 5:05PM |
1 |
username mismatch with 1.6.2.11 |
| 5:20AM |
0 |
asterisk as POC(push to talk) server? |
| 4:41AM |
0 |
voicemail not working for all extensions in same way |
| |
| Saturday September 11 2010 |
| Time | Replies | Subject |
|---|
| 3:07AM |
8 |
No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem? |
| 12:44AM |
5 |
SIP softphones answer but do not connect... |
| |
| Friday September 10 2010 |
| Time | Replies | Subject |
|---|
| 11:35PM |
5 |
Polycom dhcp boot |
| 8:51PM |
7 |
A way to check against a list of numbers? |
| 7:33PM |
0 |
Cisco or Linksys WRP400 reliability? |
| 7:01PM |
0 |
Anyone can share their experience about Thomson TG784 wireless router/ATA? |
| 2:07PM |
1 |
problem with iax call (chan unavailable) |
| 10:24AM |
0 |
1.6.2.11 realtime sip registrations disappear from DB |
| 6:17AM |
0 |
Asterisk SIP woes |
| |
| Thursday September 9 2010 |
| Time | Replies | Subject |
|---|
| 8:40PM |
2 |
DAHDI fxstest? |
| 8:38PM |
1 |
Curious what 'early media' is in terms of Answer() |
| 7:43PM |
0 |
Cisco 7975g running 8.3.4 |
| 5:25PM |
3 |
Archive of security advisories? |
| 4:51PM |
1 |
VoIP friendly Internet providers in Dallas and Philadelphia |
| 3:19PM |
0 |
Issues with in-call DTMF using Broadvox and Level 3 |
| 3:12PM |
0 |
vegastream 50 BRI-s latest firmware ? |
| 2:04PM |
5 |
info about application not available asterisk 1.6.2.11 |
| 1:55PM |
1 |
Set channel variable from within other channel |
| 1:29PM |
2 |
Mirroring or other arangement to secure * |
| 11:39AM |
2 |
How to avoid interruptions with DIGIUM |
| 10:31AM |
1 |
syntax error, unexpected '<token>' |
| 4:14AM |
1 |
getting error chan_sip.c: Failed to grab lock, trying again.. |
| |
| Wednesday September 8 2010 |
| Time | Replies | Subject |
|---|
| 9:57PM |
2 |
Max TDM calls per asterisk box |
| 8:18PM |
1 |
Asterisk 1.6 and fax |
| 7:43PM |
1 |
Thailand DID |
| 4:52PM |
1 |
Sangnoma + Digium Bridging |
| 4:51PM |
0 |
Asterisk 1.8.0-beta5 Now Available |
| 4:48PM |
0 |
Sip real problem |
| 4:10PM |
3 |
IPSec on asterisk |
| 3:38PM |
2 |
Problem with new AEX800 card dying because of interrupt problems |
| 3:16PM |
2 |
Queue/Dial Recording - Capture answering channel name. |
| 2:48PM |
1 |
Upgrade from 1.4 to 1.6 : problems with realtime mysql |
| 1:42PM |
1 |
asterisk 1.8 Calendar |
| 1:38PM |
0 |
Requirement or just Best Practice |
| 6:23AM |
0 |
How to Set Callerid Of Originate a call? |
| 4:47AM |
0 |
rtcp to cdr for calls from dahdi to sip |
| |
| Tuesday September 7 2010 |
| Time | Replies | Subject |
|---|
| 7:49PM |
1 |
Solving the CDR mess of attended transfers |
| 7:08PM |
3 |
Losing first DTMF digit (with ASR) |
| 6:56PM |
3 |
Call Center: scripting for call routing, reporting, login and logout, CTI |
| 6:47PM |
2 |
5-7 second connection delay in outgoing FXO calls |
| 5:00PM |
4 |
SPA3102 FAX not working |
| 1:16PM |
3 |
voice mail system |
| |
| Monday September 6 2010 |
| Time | Replies | Subject |
|---|
| 9:08PM |
1 |
MeetMe errorhandling |
| 7:32PM |
3 |
What can make G.729a codec hostid change? |
| 5:20PM |
0 |
How are shared variables destroyed ? |
| 4:20PM |
0 |
Asterisk stops processing calls... |
| 3:35PM |
0 |
Going to go out on a limb here - regarding Vonage |
| 3:28PM |
1 |
Dial timeout and SIP 302 Moved Temporarily |
| 2:23PM |
2 |
Macro when calling cellphone (GSM) + silence when connecting |
| 2:02PM |
2 |
Is it possible to keep both call legs live after Dial() |
| 1:10PM |
4 |
SMS and fixed land lines |
| 12:45PM |
1 |
Asterisk Fax |
| |
| Sunday September 5 2010 |
| Time | Replies | Subject |
|---|
| 10:20PM |
3 |
Registering and initiating a SIP call without a SIP client |
| |
| Saturday September 4 2010 |
| Time | Replies | Subject |
|---|
| 8:08PM |
1 |
Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ? |
| 3:52PM |
3 |
Vitelity offline? |
| 2:30PM |
0 |
Global Outage? |
| 1:40PM |
4 |
fast busy out? |
| 12:42PM |
1 |
Manuplating Queue |
| 9:21AM |
1 |
Snom phones recommended firmware |
| 8:34AM |
0 |
Wanted: UK-specific hardware recommendations (FXOand FXS) |
| |
| Friday September 3 2010 |
| Time | Replies | Subject |
|---|
| 9:06PM |
3 |
How to tell if there is a transfer from CDR? |
| 2:49PM |
1 |
Faxes |
| 1:45PM |
0 |
[draft] DAHDI-linux & DAHDI-tools 2.4.0 Release Announcement |
| 1:23PM |
0 |
How to use MYSQL(Set timeout x) |
| 1:11PM |
4 |
openvz |
| 9:07AM |
2 |
Wanted: UK-specific hardware recommendations (FXO and FXS) |
| 8:14AM |
1 |
not succeeding to hide callerid with outbound calls |
| 4:34AM |
0 |
Polycom 670 with Extension Module | Busy Lamp Field | Directed Pickup | Speed Dial | etc |
| 3:40AM |
0 |
Asterisk failing when recording calls |
| 2:03AM |
0 |
Asterisk processing URI's |
| |
| Thursday September 2 2010 |
| Time | Replies | Subject |
|---|
| 8:32PM |
1 |
Channel Signalling |
| 6:53PM |
0 |
asterisk 1.6.2.11 freezes the server |
| 6:26PM |
5 |
How to create a coredump for Asterisk |
| 5:52PM |
1 |
How to finish an AGI |
| 4:32PM |
2 |
Fw: [asterisk-biz] To compete with Avaya - What are their current cost? |
| 2:57PM |
5 |
Google Voice-like feature. |
| 12:31PM |
2 |
Voicemail - disable * 0 and # |
| 10:56AM |
2 |
Call Recording Questions |
| 9:37AM |
4 |
agi playback to execute say.conf settings |
| 6:21AM |
1 |
IAX2 calls getting rejected without a CAUSE CODE. How to debug this? |
| 3:28AM |
0 |
NCS - Cablemodem |
| |
| Wednesday September 1 2010 |
| Time | Replies | Subject |
|---|
| 7:35PM |
0 |
ITSP with DDIs (or DIDs) from India |
| 7:28PM |
0 |
libpri 1.4.11.4 Now Available |
| 6:28PM |
3 |
MOH in the middle of the call |
| 4:15PM |
2 |
* and mj |
| 1:24PM |
1 |
ChanSpy getting piled up |
| 1:09PM |
3 |
3Com 3102 Phones |
| 11:58AM |
2 |
Freepbx + Asterisk problem - NEED HELP |