| Tuesday August 31 2010 |
| Time | Replies | Subject |
|---|
| 11:27PM |
1 |
Logging the CID from the Privacy Manager |
| 10:50PM |
0 |
STUN |
| 6:22PM |
6 |
Pickup parcked call from Aastra 9480i ct cordless |
| 5:03PM |
0 |
Asterisk with Blockhosts |
| 3:37PM |
5 |
Yes it is a dimensioning question! Atom CPU |
| 2:52PM |
1 |
Running System() after call completion, not in 'h'? |
| 9:57AM |
2 |
asterisk core dump |
| 8:04AM |
4 |
No audio on call forward after upgrade from Asterisk 1.4 to 1.6 |
| 2:20AM |
1 |
Cisco 9971 |
| |
| Monday August 30 2010 |
| Time | Replies | Subject |
|---|
| 8:43PM |
0 |
Voicemail prompts fuzzy and quiet |
| 5:26PM |
0 |
Wifi + SIP + Asterisk |
| 3:31PM |
3 |
Maximum Wait Time queue option |
| 2:37PM |
2 |
help with dialplan |
| 11:16AM |
1 |
Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny |
| 7:34AM |
0 |
How to Billing for MeetMe Conference? |
| 7:19AM |
1 |
SIP Debug Messages |
| 3:29AM |
1 |
Asterisk routing to SoftSwitch |
| 2:52AM |
2 |
Could MeetMe invite someone to the conference? |
| 2:48AM |
1 |
Digest Username/auth name mismatch |
| 1:01AM |
1 |
Web-meetme |
| |
| Sunday August 29 2010 |
| Time | Replies | Subject |
|---|
| 5:31PM |
1 |
asterisk-users Digest, Vol 73, Issue 63 |
| 7:25AM |
1 |
evil disconnect of call with cisco 1760 |
| 1:27AM |
1 |
Why does Digium not respect their own development guidelines? |
| 12:43AM |
0 |
Problem routing incoming from-pstn calls using different contexts |
| |
| Saturday August 28 2010 |
| Time | Replies | Subject |
|---|
| 9:20PM |
1 |
Play a number of files to a caller |
| 3:32PM |
0 |
$250 Asterisk app install bounty |
| 9:22AM |
4 |
Asterisk does not translate from wav to alaw |
| 8:55AM |
2 |
only part of dialplan available |
| |
| Friday August 27 2010 |
| Time | Replies | Subject |
|---|
| 10:45PM |
0 |
Asterisk 1.6 Displaying BackGround() in call trace but no audio is heard from caller |
| 10:18PM |
1 |
Early media and IAX2 |
| 10:13PM |
0 |
Compiling snmp_res.so into AsteriskNow install |
| 9:58PM |
1 |
Migrating 1.4 to 1.6.2 |
| 6:49PM |
3 |
Asterisk Crashed - But why? |
| 3:48PM |
7 |
ASterisk CDR file Master.csv |
| 2:55PM |
1 |
Protect yourself |
| 2:17PM |
0 |
Asterisk DTMF RFC2833 issues |
| 2:04PM |
1 |
asterisk-users Digest, Vol 73, Issue 58 |
| 1:13PM |
0 |
Duplicate channel variables after transfer |
| 12:51PM |
2 |
Call Forwarding |
| 12:05PM |
0 |
queue agent and blind transfer |
| 11:39AM |
1 |
music on hold in blind transfer |
| 9:27AM |
2 |
dynamic MeetMe, min. digits |
| 1:58AM |
0 |
Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround() but no audio is heard from the phone |
| |
| Thursday August 26 2010 |
| Time | Replies | Subject |
|---|
| 6:55PM |
1 |
double DTMF digits |
| 5:25PM |
2 |
CDR on Transfer... |
| 4:38PM |
2 |
Use of AGISIGHUP |
| 3:09PM |
0 |
OrderlyStats or QueueMetrics |
| 2:32PM |
1 |
Brief outage of Asterisk services for maintenance, Saturday, August 28, 2010. |
| 10:19AM |
0 |
sms - your suggestions |
| 10:10AM |
1 |
MusicOnHold class working for internal calls, not for external |
| 8:41AM |
1 |
Timecondition fallthrough on 2nd GSM Modem, First modem and ZAP's are all fine |
| |
| Wednesday August 25 2010 |
| Time | Replies | Subject |
|---|
| 10:03PM |
1 |
help for a new user |
| 9:42PM |
1 |
CDR Help |
| 8:06PM |
1 |
Quick Question - Jabra Headset and Aastra 53i - Where is the speaker/headset enable setting on Aastra UI? |
| 6:08PM |
1 |
Asterisk 1.6.1.17 ACK/BYE question |
| 6:04PM |
1 |
AMR Codec |
| 3:23PM |
6 |
AEL - what is error: ael.flex:647 ael_yylex: Unhandled char(s): |
| 12:27PM |
1 |
ODBC Voicemail storage |
| 8:54AM |
1 |
asterisk-1.8 problem with one-way audio with no nat |
| 8:12AM |
0 |
Asterisk codec config issue with ALAW |
| 2:03AM |
2 |
Looking for MIB description |
| 12:02AM |
1 |
Asterisk 1.6.1 Won't Play Default ULAW Files |
| |
| Tuesday August 24 2010 |
| Time | Replies | Subject |
|---|
| 11:17PM |
4 |
1.6 and asterisk gui |
| 11:04PM |
0 |
DAHDI compile warning |
| 9:44PM |
0 |
Announcing Adhearsion 0.8.5 |
| 9:13PM |
0 |
Tones of dtmf during call |
| 7:21PM |
0 |
Fwd: Re: Make a transfer for external line. |
| 6:43PM |
1 |
asterisk-1.8.0-beta4 - compile error |
| 5:45PM |
0 |
Asterisk + SMS + PRI |
| 3:35PM |
2 |
Asterisk 1.8.0-beta4 Now Available |
| 1:51PM |
9 |
Should I move to 1.6 or 1.8, or stay with 1.4? |
| 1:05PM |
1 |
asterisk + cisco 3825 with ISDN |
| 12:53PM |
2 |
Attempted SIP connection by foreign host. Help! |
| 12:48PM |
8 |
Include and Realtime |
| 12:40PM |
0 |
Transfer + speed dial button problem? |
| 8:34AM |
2 |
How to do voice barge in using asterisk server |
| 7:31AM |
0 |
OT - How to blacklist a driver in /etc/modprobe.d without reboot [SOLVED] |
| 7:11AM |
1 |
OT - How to blacklist a driver in /etc/modprobe.d without reboot |
| 3:25AM |
1 |
IAX2 - Separate Signaling and Media? |
| |
| Monday August 23 2010 |
| Time | Replies | Subject |
|---|
| 11:46PM |
1 |
sip probe syntax |
| 8:00PM |
2 |
Asterisk, HylaFax and Cardiff |
| 6:13PM |
1 |
channel stay up when extension unreachable |
| 5:16PM |
0 |
Transfer to non registered extension creates call hangup |
| 5:03PM |
2 |
All phones ringing when temporary loss of Internet |
| 4:42PM |
2 |
How to debug this specific issue? |
| 4:34PM |
1 |
can't build resODBC on SUSE 11.3 |
| 4:05PM |
1 |
Dahdi install gone wrong |
| 3:57PM |
2 |
Asterisk voicemail server - gsm notifications |
| 3:56PM |
2 |
DAHDI not detecting caller hangup |
| 2:06PM |
2 |
outbound SIP trunk hunting (or any fxo for that matter) |
| 1:28PM |
2 |
Make a transfer for external line. |
| 1:14PM |
2 |
problem with mssql and Asterisk 1.8.0 beta3 |
| 11:14AM |
2 |
How to prevent soft hangup from being necessary ? |
| 10:35AM |
1 |
EMail on Missed Call |
| 6:58AM |
1 |
How to do barging using asterisk server. |
| 3:50AM |
0 |
Asterisk dialup connection? |
| |
| Sunday August 22 2010 |
| Time | Replies | Subject |
|---|
| 2:11PM |
1 |
.call files with application/data are not generating correct CDR |
| 12:33AM |
1 |
NVidia component out |
| |
| Saturday August 21 2010 |
| Time | Replies | Subject |
|---|
| 10:38PM |
1 |
Mobile answer machine cut off |
| 8:25PM |
7 |
Opensource Speech recognition for Asterisk |
| 9:49AM |
0 |
iax stresstest client |
| |
| Friday August 20 2010 |
| Time | Replies | Subject |
|---|
| 2:12PM |
0 |
Push to talk over cellular |
| 2:28AM |
2 |
codec_g729.so not work! |
| |
| Thursday August 19 2010 |
| Time | Replies | Subject |
|---|
| 9:56PM |
3 |
Executing system commands through Manager API |
| 6:54PM |
0 |
Call-limit field |
| 5:20PM |
2 |
asterisk + openBTS |
| 3:58PM |
0 |
Loop Detection / SIP |
| 3:58PM |
1 |
AstriCon approaches: Innovation Awards, your attendance wanted! |
| 3:29PM |
4 |
setting variable for a DID number |
| 1:52PM |
3 |
AMD message |
| 12:14PM |
0 |
3g call support for ISDN line |
| 8:14AM |
8 |
Codec choice |
| 7:21AM |
3 |
Calling Line Identity - any ideas |
| |
| Wednesday August 18 2010 |
| Time | Replies | Subject |
|---|
| 10:46PM |
2 |
IXJ issues on 1.4.35 |
| 8:54PM |
1 |
CDR variables |
| 5:51PM |
1 |
WaitExten() always times out |
| 5:14PM |
1 |
Fwd: AsteriskNow REGISTER'ing s@ extension for all inbound trunks |
| 4:10PM |
3 |
Playing with sipvicious .. |
| 1:12AM |
0 |
Pfsense and IAX2 - What is the proper firewall NAT setup? |
| 12:36AM |
0 |
Polling DND status of a Linksys SPA9xx/5xx phone? |
| |
| Tuesday August 17 2010 |
| Time | Replies | Subject |
|---|
| 7:43PM |
1 |
Directory routing to wrong extension if dial tones are pressed too quick. |
| 7:39PM |
1 |
Asterisk with Motorola Canopy |
| 6:25PM |
2 |
Add & play moh-files without reload |
| 2:09PM |
3 |
Create File Directory |
| 1:00PM |
3 |
Convert wav-file to alaw-file |
| 1:00PM |
5 |
sending sms from Asterisk server |
| 11:59AM |
1 |
MySQL Connect problem... |
| 9:12AM |
0 |
MP3Player audio format |
| 8:36AM |
1 |
dial_exec_full problems with TDM400 |
| |
| Monday August 16 2010 |
| Time | Replies | Subject |
|---|
| 8:21PM |
1 |
Polycom 331 freezes connecting to FreePBX |
| 4:10PM |
3 |
parkcall: How to remove announcement. |
| 12:42PM |
4 |
colored CLI with reattach |
| 10:46AM |
1 |
Asterisk Hardwares |
| |
| Sunday August 15 2010 |
| Time | Replies | Subject |
|---|
| 6:35PM |
6 |
Realtime Context |
| 3:09PM |
1 |
Use of Storage Area Network with Asterisk |
| 1:46PM |
1 |
603 error |
| 1:32PM |
0 |
Timing on Asterisk |
| 6:46AM |
1 |
Removing `chan_dahdi.conf` |
| |
| Saturday August 14 2010 |
| Time | Replies | Subject |
|---|
| 11:01PM |
1 |
BLF/Call Pickup using SPA942, SPA962, SPA932 |
| |
| Friday August 13 2010 |
| Time | Replies | Subject |
|---|
| 8:20PM |
6 |
Asterisk on AMD |
| 8:13PM |
3 |
IXJ Quicknet PhoneJack issues |
| 6:08PM |
3 |
installing with yum |
| 4:36PM |
0 |
Enhancing snmp mib |
| 3:43PM |
3 |
4 Port FXO interface |
| 1:08PM |
5 |
install asterisk |
| 7:57AM |
2 |
realtime sip peers : musiconhold class |
| 3:15AM |
0 |
How to Record with Konference when it has no record option? |
| 12:13AM |
1 |
OT: UK PPP certification -- what is it? |
| |
| Thursday August 12 2010 |
| Time | Replies | Subject |
|---|
| 7:08PM |
1 |
Callback script anyone |
| 6:52PM |
1 |
Problems with meetme in 1.4.26 |
| 2:50PM |
0 |
VUC Friday 13th: Skype, then Video |
| 11:52AM |
0 |
Skype |
| 9:29AM |
0 |
BRI line issue on third call |
| 9:01AM |
1 |
Recording the conversation with MixMonitor() ends when the call is transfered |
| |
| Wednesday August 11 2010 |
| Time | Replies | Subject |
|---|
| 9:54PM |
1 |
Youmail RDNIS |
| 9:40PM |
3 |
PRI errors no D channel |
| 6:04PM |
1 |
DAHDI config file system.conf |
| 5:53PM |
1 |
VM Extension : asterisk |
| 4:30PM |
0 |
Aastra 6739i Support |
| 4:08PM |
2 |
channel variables in AGI |
| 2:12PM |
1 |
How to set up Asterisk to deliver a trunk sip connection? |
| 10:14AM |
4 |
Asterisk 1.8 beta3 - Unable to stop/start/restart deamon |
| 8:59AM |
1 |
billsec exceeds duration on some calls |
| 8:36AM |
6 |
asterisk on Vmware |
| 6:36AM |
0 |
No CDR with originate from manager and then an redirect to a dial from manager |
| |
| Tuesday August 10 2010 |
| Time | Replies | Subject |
|---|
| 11:28PM |
1 |
IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal? |
| 9:49PM |
0 |
Asterisk 1.8.0-beta3 Now Available |
| 9:49PM |
0 |
Asterisk 1.6.2.11 Now Available |
| 9:48PM |
0 |
Asterisk 1.4.35 Now Available |
| 5:17PM |
0 |
asterisk-users Digest, Vol 73, Issue 24 |
| 4:51PM |
4 |
How to determine which party hangup the call? cause of Hang-up needed. |
| 3:44PM |
1 |
Dial option 'r' not working anymore? |
| 1:33PM |
1 |
PRI D-channel bouncing |
| 11:42AM |
4 |
Asterisk on Ben NanoNote? |
| 7:04AM |
1 |
Call agent when queue is empty and there is a voicemail left |
| 6:13AM |
0 |
MeetMe will record automaticlly even without 'r' option?? |
| 5:44AM |
5 |
speciality of SIPp and SER(Sip Express Router) |
| 2:13AM |
1 |
DEBUG: Cannot find variable 'XXX' ?? |
| 12:19AM |
1 |
Playback during call |
| |
| Monday August 9 2010 |
| Time | Replies | Subject |
|---|
| 7:03PM |
2 |
Correct Caller-ID |
| 6:56PM |
3 |
check channels |
| 6:08PM |
2 |
'System' application in asterisk |
| 4:25PM |
0 |
Allison Smith Hilarity |
| 3:41PM |
1 |
Connecting two calls with Originate |
| 3:03PM |
3 |
SIP response 500 "Server Internal Error" |
| 1:04PM |
1 |
redirect based on incoming number |
| 12:27PM |
1 |
op_div: non-numeric argument |
| 9:01AM |
2 |
Prepay Limited Calls. |
| 8:36AM |
1 |
MeetMe VS. Conference |
| 7:31AM |
0 |
[SIP/H.264] Codec negotiation problem ? |
| |
| Sunday August 8 2010 |
| Time | Replies | Subject |
|---|
| 8:29PM |
0 |
PBX Status-like module for AsteriskNow? |
| 1:10PM |
3 |
How to track a call result originated from originate AMI command |
| 11:49AM |
0 |
Asterisk 1.6.2 FastAGI Hangup Problem |
| |
| Saturday August 7 2010 |
| Time | Replies | Subject |
|---|
| 7:25PM |
3 |
Monitor asterisk |
| 2:06PM |
2 |
AMD setup in Astersik |
| 7:16AM |
1 |
Scilence problem on running call |
| 7:09AM |
0 |
Should external hosts be able to register on Asterisk behind a firewall? |
| 5:45AM |
3 |
Dahdi issue on sangoma A200 |
| 5:27AM |
0 |
shrinkcallerid |
| 1:17AM |
0 |
Set outgoing number in filename of the recordings |
| |
| Friday August 6 2010 |
| Time | Replies | Subject |
|---|
| 6:15PM |
1 |
Reinstalling Asterisk due to hardware changes |
| 3:33PM |
1 |
Asterisk 1.4 and TE420P |
| 3:19PM |
1 |
OT: Grandstream GXV3140 |
| 2:45PM |
2 |
How does deny/permit work in sip.conf? |
| 2:36PM |
2 |
Using a 1.4 config with 1.6 |
| 2:13PM |
1 |
How do I install speex for |
| 11:59AM |
4 |
How do I install speex for asterisk? |
| 6:08AM |
1 |
Security - What inbound variables can attackers populate or use when calling? |
| 5:54AM |
2 |
How to reuse mysql connection between AGI's |
| 4:58AM |
0 |
Missing Mailboxes on SIP |
| |
| Thursday August 5 2010 |
| Time | Replies | Subject |
|---|
| 8:52PM |
1 |
Asterisk 1.6 without DAHDI |
| 8:26PM |
2 |
rolling over Master.csv CDR File |
| 8:11PM |
1 |
CDR report |
| 9:12AM |
1 |
Codec Conversion |
| 8:28AM |
2 |
AMI Command |
| 7:56AM |
1 |
Can ChanIsAvail return status from sip uri using router ip |
| 6:08AM |
0 |
Which Python binding to use ? |
| 5:52AM |
0 |
No Mailbox Subscription in SIP Users Suddenly |
| 1:52AM |
1 |
Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used |
| |
| Wednesday August 4 2010 |
| Time | Replies | Subject |
|---|
| 7:12PM |
2 |
Identify remote prompts: Partial audio matching? |
| 3:57PM |
5 |
Asterisk and RAID |
| 2:44PM |
1 |
Asterisk (1.8-beta2) and SIP IPv4/IPv6 dual-stack possibilities |
| 2:08PM |
1 |
Tweaking AMD in Asterisk |
| 1:45PM |
0 |
Queue to queue transfer error |
| 12:59PM |
0 |
can't write to queues_additional.conf |
| 12:28PM |
1 |
Asterisk not working with Festival |
| 11:28AM |
1 |
callerid between 2 asterisk servers |
| 9:12AM |
2 |
How to record a file and play some other file at the same time |
| 4:40AM |
1 |
CDR: MySQL query |
| |
| Tuesday August 3 2010 |
| Time | Replies | Subject |
|---|
| 8:57PM |
2 |
Using SIP to dial extension that will give an outside line |
| 6:19PM |
4 |
Dial() M parameter in 1.6.2.11-rc2 |
| 5:39PM |
1 |
outboundproxy timeout or qualify |
| 5:20PM |
0 |
ConfBridge |
| 5:06PM |
1 |
real-time queue problems |
| 4:25PM |
1 |
Garbled messages - format_wav_gsm.c:414 wav_read: Short read (60) (No such file or directory)! |
| 3:22PM |
0 |
asterisk-users Digest, Vol 73, Issue 5 |
| 2:48PM |
3 |
Fax/Modem, Asterisk, Channel Banks |
| 1:28PM |
7 |
FYI: Seen the 2600Hz announcement? |
| 8:13AM |
1 |
sip.conf register in realtime DB |
| 6:05AM |
2 |
RTP stream not passing through router with port forwarding |
| 12:58AM |
1 |
chinaroby fxo card - never heard of them |
| 12:40AM |
1 |
Asterisk 1.6 and PrivacyManager with SIP |
| |
| Monday August 2 2010 |
| Time | Replies | Subject |
|---|
| 8:24PM |
0 |
alaw.h in app_meetme.c |
| 7:56PM |
3 |
Caller ID issue |
| 7:36PM |
4 |
Femtocell to VoIP? |
| 7:15PM |
0 |
Whither app_nv_faxdetect |
| 6:59PM |
3 |
FAX Options |
| 6:26PM |
5 |
What do you use for Invoicing? |
| 4:56PM |
1 |
asterisknow |
| 4:35PM |
3 |
IAX softphone |
| 1:35PM |
2 |
asterisk compatible cards? |
| 12:53PM |
5 |
mapping of disconnect reasons |
| 12:37PM |
5 |
Asterisk and TV media server |
| 12:12PM |
6 |
Codec negotiation : expecting G726, getting G711a (alaw) |
| 7:34AM |
1 |
Any Free software that can connect to an Asterisk Server and Do video Conferencing? |
| 4:46AM |
3 |
how to place a call on hold and play music on hold using agi |
| 2:12AM |
1 |
SIP Status: 401 Unauthorized (0 bindings) |
| |
| Sunday August 1 2010 |
| Time | Replies | Subject |
|---|
| 11:47PM |
2 |
Exporting Blacklist database |
| 9:27PM |
3 |
fail2ban does not work for my asterisk installation |
| 1:58PM |
2 |
# -key not to be 'transfer' |
| 6:25AM |
0 |
How to read span debug ? |