| Tuesday November 30 2010 |
| Time | Replies | Subject |
|---|
| 10:25PM |
2 |
Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory |
| 6:25PM |
1 |
Rhino Channelbank... |
| 1:14PM |
2 |
Correct operation of timout parameter for dial application |
| 12:11PM |
0 |
Default From and Contact header domain |
| 9:47AM |
2 |
Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented) |
| 9:28AM |
10 |
TCP port, VPN and resolving the cutting voice problem |
| 8:46AM |
0 |
Transfered Number is local extension of PSTN |
| 4:13AM |
1 |
OT: for those wondering on the stability |
| |
| Monday November 29 2010 |
| Time | Replies | Subject |
|---|
| 5:11PM |
1 |
Trouble with TE122 on HP DL120G6 - can't disable USB |
| 5:07PM |
3 |
How to initiate a two-party call from within Asterisk |
| 5:01PM |
1 |
ID'ing failed auth IPs |
| 10:23AM |
0 |
resending cause codes |
| 10:08AM |
4 |
Asterisk on smartphone? |
| |
| Sunday November 28 2010 |
| Time | Replies | Subject |
|---|
| 10:21PM |
2 |
Stability.. |
| 7:14PM |
0 |
Problem in receiving calls from E1 |
| 5:03PM |
4 |
Firewalling and Asterisk |
| 1:42PM |
0 |
Asterisk loopback calls form one extension that playbacks to another that records for performance measuring |
| 4:28AM |
0 |
AstLinux 0.7.4 Release now available |
| |
| Saturday November 27 2010 |
| Time | Replies | Subject |
|---|
| 8:03PM |
0 |
Strange Logfile Entries. |
| 7:40PM |
2 |
Preserve CallerID on transfers |
| 5:03PM |
1 |
DAHDI 2.4.0 produces HDLC errors with echo canceler |
| 3:51PM |
3 |
sip echo server |
| 10:52AM |
1 |
change date |
| 9:25AM |
3 |
How to hangup all channels |
| |
| Friday November 26 2010 |
| Time | Replies | Subject |
|---|
| 6:39PM |
1 |
echo calls |
| 4:58PM |
0 |
asterisk-users Digest, Vol 76, Issue 58 |
| 4:20PM |
1 |
Meetme Realtime in 1.6 |
| 1:29PM |
0 |
IAX trunk two Asterisk |
| 11:43AM |
3 |
New implementation asterisk |
| 11:05AM |
1 |
HA8 + B400M not configured with genconf_parameters |
| 11:00AM |
4 |
Asterisk 1.8 crashing |
| |
| Thursday November 25 2010 |
| Time | Replies | Subject |
|---|
| 9:57PM |
0 |
d-link dvg-3032 guidance |
| 5:05PM |
0 |
IAX inbound failing |
| 4:54PM |
1 |
Unit of measurement dahdi_monitor |
| 4:23PM |
2 |
Timing cable usage necessity |
| 12:51PM |
0 |
Siemens HiPath 1120 |
| 2:57AM |
2 |
Spam |
| 1:30AM |
4 |
Incoming calls through SS7 for data modem transmissions - possible?? |
| |
| Wednesday November 24 2010 |
| Time | Replies | Subject |
|---|
| 5:45PM |
1 |
Disable connected line updates for dahdi PRI channel |
| 5:07PM |
1 |
Why doesn't Asterisk project document certain important features of Asterisk officially? |
| 5:01PM |
0 |
IPv6: What You Need to Know Now |
| 4:43PM |
3 |
kernel: dahdi: Detected time shift. |
| 1:44PM |
0 |
Originate Response. |
| 12:48PM |
2 |
Avoided deadlock Error |
| 12:11PM |
0 |
DTMF CallerID |
| 11:19AM |
1 |
TDM calls fall after some minutes |
| 11:16AM |
2 |
SPA942 on speaker phone does not hang up? |
| 11:09AM |
2 |
asterisk-1.8.0 compilation error |
| 10:30AM |
1 |
Asterisk 1.6 and Music on Hold |
| 10:12AM |
1 |
Contradiction in GROUP() function |
| 8:02AM |
2 |
astcanary ? |
| 7:20AM |
1 |
action at registering or de-registering |
| |
| Tuesday November 23 2010 |
| Time | Replies | Subject |
|---|
| 11:57PM |
2 |
Function SIP_Header not registered |
| 8:33PM |
1 |
asterisk 1.8.1-rc1 + sip transfer fix |
| 4:30PM |
0 |
Asterisk 1.8.1-rc1 Now Available |
| 1:25PM |
2 |
Asterisk Log viewer |
| 1:24PM |
1 |
wideband recording in Asterisk 1.8 |
| 12:31PM |
2 |
Asterisk 1.8 Release Schedule |
| |
| Monday November 22 2010 |
| Time | Replies | Subject |
|---|
| 10:46PM |
3 |
Asterisk pass a call to status answer while still ringing |
| 9:27PM |
0 |
libpri 1.4.11.5 Now Available |
| 6:38PM |
1 |
asterisk and cisco 7970 - multiple lines |
| 5:28PM |
0 |
Using AMI to harvest / record HOLD time - Using FreePBX |
| 4:24PM |
0 |
Polycom dial w/o "Dial", while on-hook? |
| 4:22PM |
5 |
Someone has hacked into our system |
| 1:47PM |
2 |
Call recording format |
| 9:31AM |
1 |
res_musiconhold.c Bug - Patch to solve? |
| 9:09AM |
2 |
URGENT Help needed |
| 5:20AM |
3 |
Is existing CDR in Asterisk is enough for complete billing |
| 5:06AM |
1 |
Quintum AFT800 on Asterisk 1.4.29 |
| 2:41AM |
1 |
Early audio (long distance codes) not working after upgrading to 1.8? |
| 2:37AM |
0 |
cisco 7970 multiple lines with asterisk |
| 1:14AM |
2 |
SIP Extensions and loss of Internet connection |
| |
| Sunday November 21 2010 |
| Time | Replies | Subject |
|---|
| 11:13PM |
2 |
DAHDI phantom pickup when ringing |
| 10:01PM |
0 |
How to configure a Linksys PAP2T ATA to connect an analog fax machine to Asterisk |
| 11:58AM |
2 |
Please help me in configuring asterisk for the scenario |
| 11:26AM |
2 |
Asterisk behind D-Link ADSL router with private IP |
| |
| Saturday November 20 2010 |
| Time | Replies | Subject |
|---|
| 6:36PM |
1 |
ConfBridge |
| 6:17PM |
0 |
AGI CDR Update (with set variable) problem. |
| 4:55PM |
0 |
sip attended transfer beep |
| |
| Friday November 19 2010 |
| Time | Replies | Subject |
|---|
| 9:56PM |
2 |
Installing Asterisk to it's own directory |
| 6:55PM |
1 |
Make call in AMI. |
| 4:13PM |
0 |
help with annoying warning message: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm) |
| 3:33PM |
1 |
callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk |
| 2:42PM |
3 |
FFA (Fax For Asterisk) tif file (size) problem |
| 1:59PM |
0 |
Asterisk 1.8 and Dial(SIP/peer_name) to undefined peer |
| 12:28PM |
2 |
Ekiga can register but not my IP phone |
| 11:59AM |
0 |
Using Local Asterisk Server with Siphon - Can't hear voice issue |
| 6:34AM |
1 |
call forward problem |
| |
| Thursday November 18 2010 |
| Time | Replies | Subject |
|---|
| 9:54PM |
2 |
ISDN-FAX with Asterisk |
| 8:01PM |
2 |
IAX2 and INVAL packets |
| 6:37PM |
2 |
Meetme and MOH |
| 5:34PM |
1 |
Asterisk parking question |
| 5:04PM |
1 |
Asterisk 1.8 VM_DUR problems |
| 1:39PM |
2 |
Asterisk Load Balance and Failover |
| 10:58AM |
0 |
Announcement Transfer with call-limit = 1 |
| 10:36AM |
3 |
How to register SIP phone on Asterisk 1.6.2.14 on Centos 5.5 64bit |
| 6:21AM |
2 |
exceeds the maximum size of ast_fdset error on Asterisk-1.8.0 |
| 3:54AM |
3 |
usage of account code in CDR |
| 2:16AM |
1 |
Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit |
| 12:31AM |
0 |
Supported: ms-early-media |
| |
| Wednesday November 17 2010 |
| Time | Replies | Subject |
|---|
| 8:30PM |
2 |
GSM and SS7 Questions |
| 6:54PM |
1 |
Asterisk and IPv6 |
| 4:40PM |
6 |
How many Asterisk PBX operating in the World? |
| 1:09PM |
0 |
One way audio problem |
| 10:58AM |
0 |
Newbie question on GSM adapter |
| 5:46AM |
1 |
Asterisk runs at 100% CPU |
| |
| Tuesday November 16 2010 |
| Time | Replies | Subject |
|---|
| 8:37PM |
0 |
billsec and duration issue |
| 7:44PM |
0 |
Fwd: Delivery Delayed: Re: Ring back tone with asterisk |
| 7:03PM |
2 |
T1 with Robbed Bit Signaling |
| 6:38PM |
5 |
HA - asterisk service is not starting |
| 6:17PM |
1 |
S110M not working |
| 5:35PM |
0 |
OT - Call Waiting features with Kirk 600v3 |
| 2:28PM |
0 |
SPA941 WMI not lighting up when natted |
| 2:28PM |
3 |
Recommended *WRT router to run Asterisk? |
| 2:08PM |
2 |
Avoiding deadlock |
| 12:57PM |
1 |
Issues with Local Channel |
| 9:09AM |
1 |
DAHDI / dial in / overlap digits / timeout |
| 2:00AM |
2 |
How to construct a call center on asterisk |
| |
| Monday November 15 2010 |
| Time | Replies | Subject |
|---|
| 11:04PM |
4 |
Best way to connect to a MySQL Database |
| 8:11PM |
2 |
SIP calls destroyed after 1:20 |
| 6:35PM |
7 |
Door Contacts via Asterisk? |
| 4:22PM |
0 |
Maybe a little OT??--- Obtaining DIDs in Hyderabad, India |
| 1:30PM |
2 |
Volume on meetme recording |
| 12:46PM |
0 |
Asterisk Maintenance Checklist |
| 11:05AM |
2 |
Problem When Using Polycom with 2 Lines |
| 10:47AM |
2 |
friend, peer confusion in sip.conf |
| 8:03AM |
0 |
(no subject) |
| 2:48AM |
0 |
Asterisk disconnected suddenly |
| |
| Sunday November 14 2010 |
| Time | Replies | Subject |
|---|
| 3:00AM |
1 |
upgrade |
| 2:11AM |
1 |
A few questions regarding Asterisk 1.8.0 |
| 1:38AM |
8 |
dial plan and sip |
| 1:18AM |
0 |
EXTENDED: Scheduled maintenance for various Asterisk community services |
| |
| Saturday November 13 2010 |
| Time | Replies | Subject |
|---|
| 7:06PM |
0 |
problem registering to ekiga.net |
| 6:43PM |
1 |
Nat Issue - I think |
| 5:10PM |
0 |
eSXI and Asterisk? |
| 10:15AM |
1 |
CallerID from Samsung PBX line on FXO |
| 9:36AM |
2 |
asterisk 1.8 fax woes |
| 6:14AM |
1 |
asterisk-stat v.2 and mysql 5.1.51 |
| |
| Friday November 12 2010 |
| Time | Replies | Subject |
|---|
| 10:01PM |
0 |
Asterisk and Tandberg Gatekeeper |
| 9:36PM |
1 |
Scheduled maintenance for various Asterisk community services |
| 7:56PM |
1 |
Call failed becaus of SIP tanslate |
| 6:17PM |
3 |
Sending calls to a particular T1 port. |
| 5:08PM |
1 |
Context issue |
| 4:47PM |
0 |
Asterisk Sip trunking routing problem |
| 11:07AM |
3 |
Official Documentation for Asterisk 1.6 Realtime ODBC Tables |
| 9:23AM |
1 |
Asterisk 1.6.20 / CDR issue with app-dial / bug or feature? |
| 1:22AM |
1 |
TTS in Asterisk on Solaris |
| |
| Thursday November 11 2010 |
| Time | Replies | Subject |
|---|
| 10:21PM |
3 |
T38 re-invites issue |
| 8:36PM |
0 |
Asterisk 1.6.2.14 Released |
| 8:32PM |
0 |
Asterisk 1.4.37 Released |
| 4:28PM |
1 |
VoiceMail customizing |
| 3:35PM |
2 |
Asterisk Playback sound dropping on linphone |
| 9:31AM |
3 |
Limit Call Duration with L-option of Dial : announcement |
| 8:23AM |
0 |
Asterisk 1.6.2.6 and ENUM LOOKUP? E.164 |
| |
| Wednesday November 10 2010 |
| Time | Replies | Subject |
|---|
| 11:51PM |
0 |
1.4.36 - Warning Dropping incompatible voice frame on Local/ on multiple atxfer a->b->c...->d... |
| 9:17PM |
0 |
Reboot any(?) SIP Polycom -- provisioned or no. |
| 7:52PM |
0 |
Problem with AMI |
| 7:11PM |
1 |
Selecting 'ODBC_STORAGE' from outside of 'menuselect' |
| 7:09PM |
2 |
Asterisk 1.8 -- queue not recognizing that agent is busy |
| 6:51PM |
1 |
Random call drops on IAX2 |
| 5:27PM |
1 |
multiple devices wants to call through single peer (trunking) |
| 3:08PM |
0 |
Asterisk 1.6.2.13 IAX2 Realtime issue |
| 2:52PM |
1 |
Phones don't stop ringing |
| 2:42PM |
1 |
MFC/R2 detected as ISDN PRI |
| 2:40PM |
0 |
Friday @12 Noon EST: PhonoSDK from Voxeo Labs |
| 12:55PM |
1 |
dahdi module disappears from AsteriskNow after kernel update |
| 12:05PM |
0 |
CDR Billing issues |
| 5:39AM |
0 |
Asterisk ConfBridge application – Delay in voice path |
| |
| Tuesday November 9 2010 |
| Time | Replies | Subject |
|---|
| 11:56PM |
0 |
hello xC |
| 7:38PM |
0 |
zaptel debugging |
| 5:57PM |
1 |
Asterisk 1.2 |
| 5:07PM |
1 |
Asterisk 1.6 and Username in Dial |
| 3:57PM |
1 |
Asterisk 1.8 and Zimbra |
| 3:35PM |
1 |
SMS Gateway |
| 10:30AM |
0 |
Asterisk Voicemail Realtime and 'VirtualBoxing' |
| 2:31AM |
1 |
Store CDR (call detail record) to Oracle database |
| 2:17AM |
1 |
Is this a DDoS to reach Asterisk? |
| |
| Monday November 8 2010 |
| Time | Replies | Subject |
|---|
| 11:05PM |
0 |
Asterisk 1.6 and Kamailio 3.1 realtime integration tutorial |
| 9:43PM |
3 |
Addons for Asterisk 1.8? |
| 8:59PM |
1 |
Asterisk 1.8 Multiple Parking Lots |
| 6:12PM |
3 |
Get the Uniqueid of Action Originate in the AMI |
| 10:55AM |
4 |
Integrating With Asterisk |
| 7:35AM |
0 |
MWI SUBSCRIBE Settings |
| 7:31AM |
0 |
Asterisk re-started automatically |
| 6:16AM |
0 |
VAD in asterisk |
| 6:08AM |
0 |
Trixbox/Asterisk integration With SugarCRM |
| 1:02AM |
1 |
Asterisk with HUD Lite |
| |
| Sunday November 7 2010 |
| Time | Replies | Subject |
|---|
| 11:08PM |
2 |
Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem |
| 8:59PM |
2 |
install |
| 6:29PM |
7 |
Big practical systems |
| 4:23PM |
2 |
Any good guides for installing Asterisk on Embedded systems like Alix boards? |
| 3:26PM |
2 |
"scratchy" sound on TE410P |
| 2:11PM |
3 |
Why are the hackers scanning for these? |
| |
| Saturday November 6 2010 |
| Time | Replies | Subject |
|---|
| 9:48PM |
1 |
Abandoned queue calls do not produce a CDR? |
| 4:54PM |
2 |
One way voice with Asterisk |
| 4:22PM |
2 |
Asterisk spontaneous reboot |
| 2:30PM |
1 |
OT: certificate for softphone |
| 1:20PM |
1 |
sip and iax2 audio volume gain |
| 5:17AM |
0 |
gigasets A580IP Recall Button |
| 3:57AM |
0 |
Any way to stop Playtones(dial) when the user presses a key, emulating a CO's behavior? |
| 12:05AM |
1 |
Call using password |
| |
| Friday November 5 2010 |
| Time | Replies | Subject |
|---|
| 9:35PM |
1 |
Soundpoint IP 430 -- discontinued. |
| 8:44PM |
1 |
res_ais Error |
| 8:04PM |
2 |
Alternative to Proxmox |
| 7:32PM |
1 |
Unable to place 2 or more calls to a DID |
| 7:30PM |
0 |
Polycom WEB UI configuration - What needs to be put in for basic SIP registration? |
| 4:50PM |
0 |
No audio with gtalk client behind http proxy |
| 3:21PM |
1 |
Asterisk default sound files |
| 3:21PM |
1 |
Asterisk 1.8 Installation Problem |
| 3:11PM |
3 |
Elementary question - accessing feature codes from cell phone |
| 3:04PM |
2 |
Funky IAX behavior between 1.4 and 1.8 |
| 3:01PM |
0 |
Using Dial() but no CDR is generated for this outcall |
| 2:58PM |
2 |
How to append custom option to Contact: header on outgoing SIP INVITE msgs? |
| 2:52PM |
1 |
Asterisk in the third world - Astricon 2010 keynote follow-up |
| 2:27PM |
1 |
GROUP_COUNT not counting correctly |
| 3:10AM |
0 |
How to check PRI status from dialplan |
| 1:22AM |
5 |
MixMonitor |
| 12:30AM |
2 |
Determine channels in use from CLI |
| 12:12AM |
3 |
Short rings for extensions when part of the Queue |
| |
| Thursday November 4 2010 |
| Time | Replies | Subject |
|---|
| 11:24PM |
0 |
[backport] Allow app_dial to play 'indication tone while ringing' back ported to 1.6.2.X |
| 10:31PM |
1 |
Phones slow to ring |
| 9:40PM |
0 |
mISDN issues again |
| 8:51PM |
2 |
useless mpg123 processes hanging around |
| 7:24PM |
1 |
upgrade 1.6 -> 1.8: wrong password! |
| 5:56PM |
1 |
Is queue Members priority supposed to show in the "queue show" command |
| 4:07PM |
2 |
Multiple extensions - same context |
| 1:22PM |
0 |
Asterisk + Mediatrix |
| 12:14PM |
1 |
UNREACHABLE/Lagged happening on "bulk" register/subscribe |
| 9:38AM |
3 |
Urgent Help Required |
| 9:36AM |
1 |
(no subject) |
| 9:15AM |
0 |
Help Required (How to acheive packetization time of 60ms over SIP/IAX2 trunk) |
| 6:37AM |
0 |
ring delay and DTMF related problem in asterisk 1.6.2.6 |
| |
| Wednesday November 3 2010 |
| Time | Replies | Subject |
|---|
| 10:30PM |
1 |
doh! chan_dahdi.conf |
| 9:34PM |
0 |
Asterisk/Asterisk SCF Project Wiki |
| 8:05PM |
1 |
Gotoif changed in 1.8? |
| 2:36PM |
1 |
Ring back problem on SIP calls. Order of 183 Session Progress and 180 Ringing |
| 2:14PM |
6 |
Migration from 1.2 to 1.8 in production |
| 1:18PM |
3 |
How to make the sum of a ${VARIABLE} + 1 ?? |
| 11:48AM |
1 |
Asterisk linphone call dropping by itself |
| 8:08AM |
1 |
inbound call issue... |
| 1:10AM |
1 |
Asterisk and SIP a Provider in Brazil |
| 12:29AM |
5 |
ADSL Load Balancing |
| |
| Tuesday November 2 2010 |
| Time | Replies | Subject |
|---|
| 7:20PM |
3 |
IAX or SIP - connecting two Asterisk servers together |
| 7:06PM |
0 |
Asterisk community services powered by Atlassian tools |
| 4:48PM |
0 |
Need testing: chan_unistim improvements |
| 4:19PM |
1 |
Feature Request for 1.10 - ISDN power-save mode |
| 4:13PM |
3 |
Asterisk, VoIP and Samsung iDCS100 |
| 1:13PM |
0 |
Sip, Qualify=200 that doesn't qualify. How to signal this state to the Peer |
| 1:02PM |
0 |
Noise while passing channel using tde205p card |
| 9:27AM |
2 |
Ring Freq |
| |
| Monday November 1 2010 |
| Time | Replies | Subject |
|---|
| 9:11PM |
4 |
Issue with asterisk |
| 8:24PM |
1 |
DISA problem in 1.8.0 |
| 8:21PM |
0 |
Queue Group not forwaring calls to agents |
| 7:47PM |
0 |
Ringback problem. Order of "183 Session Progress" and "180 Ringing" |
| 3:55PM |
4 |
FW: Under heavy attack |
| 11:53AM |
0 |
2nd network interface for RTP/media |
| 11:52AM |
1 |
MoH and stuch channels |
| 10:40AM |
0 |
Force direct RTP |
| 1:59AM |
1 |
Music On Hold Help |