| Monday June 30 2008 |
| Time | Replies | Subject |
|---|
| 10:54PM |
0 |
dnsmgr.conf, I do not see Refreshing DNS lookups |
| 10:25PM |
1 |
Centos-5.2 and zaptel-1.4.11 do not get along well |
| 9:46PM |
1 |
Milliwatt-sounding tone recorded over voicemail message |
| 7:51PM |
2 |
Windows Mobile 6 IAX/SIP client? |
| 6:32PM |
0 |
how to have an agi check for dial tone on analog lines before dialing |
| 5:33PM |
0 |
Asterisk 1.4.21.1 Released |
| 5:24PM |
4 |
Voicemail- Recorded Mesage Low Volume |
| 4:15PM |
5 |
sip extension compromised, need help blocking brute force attempts |
| 4:03PM |
1 |
Spam Filter |
| 12:09PM |
0 |
capture call within same callgroup with *8 |
| 12:02PM |
0 |
Interesting use of IVR |
| 11:40AM |
0 |
FXS: two rings, then it answer and hangup |
| 7:55AM |
2 |
asterisk and 802.1Q |
| 7:19AM |
1 |
queue welcome message |
| 5:26AM |
0 |
Asterisk to Broadvoice SIP peer fails in 1.6.9-beta9 |
| 2:04AM |
0 |
Hangup? |
| |
| Sunday June 29 2008 |
| Time | Replies | Subject |
|---|
| 10:35PM |
1 |
Timeout between digits for fxs station |
| 5:51PM |
0 |
[VOIP-Users-Conference] Re: A Flood Of Asterisk Appliances |
| 2:32PM |
2 |
indicating call on d channel when no b chan available |
| 12:11PM |
0 |
hint() extension in AEL |
| 11:48AM |
1 |
[FreeBSD 6.3] Why not use safe_asterisk? |
| 9:07AM |
0 |
Druid Open Source Events - Druid Miami Meetup (18 Jul), OSCON (21-25 Jul), Druid London Meetup (22 Jul) & LinuxWorld (4-7 Aug) |
| 7:59AM |
0 |
CTI Intergration with the CRM |
| 7:48AM |
1 |
sendmail file |
| |
| Saturday June 28 2008 |
| Time | Replies | Subject |
|---|
| 1:15PM |
6 |
Asterisk as an IVR |
| 12:34PM |
3 |
Palyback and CDR records |
| 9:39AM |
0 |
AMI extenstion state |
| |
| Friday June 27 2008 |
| Time | Replies | Subject |
|---|
| 11:26PM |
1 |
Debug dropped calls |
| 9:20PM |
2 |
FW: Do not update to Firefox 3, yet? |
| 9:08PM |
1 |
Asterisk 1.2 app_vxml |
| 9:03PM |
7 |
measuring network quality in the field |
| 6:31PM |
1 |
Set Language not working! |
| 5:20PM |
1 |
polycom with http/https basic authentication |
| 3:07PM |
2 |
How to pass variable between 2 Asterisk servers over IAX2 |
| 2:54PM |
2 |
Asterisk as a component in Jabber network |
| 2:36PM |
1 |
Asterisk's ZRTP patch |
| 2:02PM |
3 |
Do not update to Firefox 3, yet? |
| 1:39PM |
1 |
Maximum number of SIP peers in Asterisk 1.4 |
| 1:24PM |
1 |
gxp2000 time. |
| 10:57AM |
2 |
usb - audio asterisk crashes |
| 8:35AM |
1 |
Asterisk cuts off intial voice path on bridging SIP channel |
| 4:08AM |
1 |
DNS Query Overload |
| 1:11AM |
1 |
Asterisk, POTS and plain handsets |
| 12:20AM |
1 |
is it possible? 1 VOIP Provider Multiple registrations <to> multiple inbound contexts |
| |
| Thursday June 26 2008 |
| Time | Replies | Subject |
|---|
| 10:52PM |
0 |
Cepstral ... Swift... weird result |
| 10:18PM |
0 |
start valgrind and asterisk via init.d script |
| 9:36PM |
4 |
SIP/IAX2 Provider with fallback dialing? |
| 8:51PM |
2 |
queues and MEMBERINTERFACE for AGI script |
| 6:38PM |
0 |
Astricon: Early Bird Special ends next week |
| 5:37PM |
0 |
Console/dsp in 1.4.X |
| 5:24PM |
0 |
Hangup channel |
| 4:17PM |
2 |
Echo Cancelation |
| 3:55PM |
1 |
VoIP Users Conference June 27th @ 12 Noon EDT Scaling and Clustering |
| 3:05PM |
0 |
Error while Compiling zaptel-1.4.11 |
| 2:35PM |
1 |
Asterisk With Web meetme |
| 2:10PM |
1 |
chan_zap.c:4747 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
| 1:46PM |
0 |
disconnection from caller did not recognized |
| 11:55AM |
1 |
Fw: Outbound video Calls |
| 9:56AM |
1 |
Outbound video Calls |
| 8:17AM |
2 |
where can I found documentation about channel drivers |
| 5:44AM |
0 |
about the Dial application |
| |
| Wednesday June 25 2008 |
| Time | Replies | Subject |
|---|
| 10:07PM |
0 |
iax2_trunk_queue: Maximum trunk data space exceeded |
| 9:16PM |
2 |
SIP vs. SKINNY |
| 7:59PM |
1 |
Cisco Presence |
| 7:10PM |
0 |
Res: Asterisk with Nextone using H323 |
| 6:35PM |
0 |
Cisco 7960 Promiscuous Redirect? |
| 4:40PM |
1 |
Google Apps IMAP |
| 3:54PM |
1 |
included context not being prioritized properly |
| 3:20PM |
2 |
Any SLA alternatives? |
| 2:49PM |
5 |
Number portability in other parts of the world. |
| 12:42PM |
3 |
asterisk seg fault |
| 12:16PM |
0 |
[Fwd: Bridging an existing PBX in with Asterisk] |
| 12:04PM |
0 |
Bridging an existing PBX in with Asterisk |
| 9:21AM |
0 |
misdn issues |
| 7:23AM |
1 |
AS5400 E1 SS7 |
| 5:58AM |
0 |
unable to send a fax to a given FAX number |
| 4:56AM |
5 |
Major problem with 1.4.21 asterisk |
| 1:26AM |
3 |
Can asterisk support using different ip for rtp? |
| |
| Tuesday June 24 2008 |
| Time | Replies | Subject |
|---|
| 5:39PM |
4 |
does asterisk 1.4.20 run on a 486 sx |
| 3:37PM |
1 |
Calls drop + "Didn't get a frame from channel" log message |
| 3:28PM |
2 |
Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk! |
| 3:20PM |
2 |
Asterisk with Nextone using H323 |
| 1:30PM |
3 |
Chef-secretary scenario |
| 11:44AM |
2 |
Lockups with IAX2 and cdr_odbc in 1.4.21 is it my weird config ? |
| 9:22AM |
4 |
Queue with different music for each caller |
| 8:43AM |
0 |
Can I use X-Lite from local and external ip (when I'm not at home) ? |
| 8:29AM |
1 |
Softphone accepting sip messages |
| 7:32AM |
0 |
GXW4024 |
| 6:54AM |
2 |
Loose connection with MySql. |
| 6:46AM |
0 |
Kirk 600v3 Server with sip secret |
| 6:41AM |
0 |
retrieve the status of a sip user using AMI |
| 5:57AM |
1 |
No Codecs and app |
| 12:06AM |
1 |
GotoIfTime Function |
| |
| Monday June 23 2008 |
| Time | Replies | Subject |
|---|
| 10:02PM |
0 |
Zaptel version on Asterisk website... |
| 5:20PM |
6 |
Centile ipbx, anyone heard of this? |
| 4:54PM |
8 |
Building a Complex IVR |
| 3:57PM |
0 |
new bounty CURL timeout |
| 3:21PM |
0 |
how to restart asterisk after it crashes |
| 12:08PM |
3 |
Controlling cell phone VM / Fax waiting notification icon for asterisk VM |
| 4:19AM |
1 |
Replace music-on-hold on MeetMe with ringing sound |
| |
| Sunday June 22 2008 |
| Time | Replies | Subject |
|---|
| 4:38PM |
1 |
Telco MWI with Asterisk 1.6-beta9 |
| 3:23PM |
5 |
Send cell phone #VM waiting, just like cell carrier |
| 2:21PM |
1 |
SIP over TCP |
| 2:00PM |
1 |
multi-asterisk server implementations |
| 10:47AM |
0 |
Software loop on ZAP trunk - Sangoma |
| 9:25AM |
0 |
(no subject) |
| 9:25AM |
1 |
voicemail didn't send voice message to my email |
| 7:24AM |
4 |
mpg123 problem |
| |
| Saturday June 21 2008 |
| Time | Replies | Subject |
|---|
| 9:38PM |
12 |
Asterisk GSM Gateway Project |
| 9:32PM |
2 |
1.4.21 + Realtime Queues = Agents Not Ringing? |
| 7:39PM |
1 |
Realtime and OOH323 |
| 6:00PM |
0 |
One VOIP Provider Multiple registrations <to> multiple inbound contexts ? |
| 4:11PM |
2 |
DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP |
| 12:30PM |
1 |
iax2 trunk becomes unreachable (asterisk 1.4.21) |
| 11:23AM |
1 |
Fwd: Detection of Answer, hangup, busy etc while using Dial command |
| 11:02AM |
3 |
Continued TAPI Trouble |
| 10:38AM |
0 |
asterisk v1.6 monitor_exec |
| |
| Friday June 20 2008 |
| Time | Replies | Subject |
|---|
| 9:42PM |
3 |
Recommendations for Motel Instalation. |
| 7:17PM |
1 |
Asterisk and remote phone. |
| 7:13PM |
1 |
Voice only works from one way. |
| 5:47PM |
2 |
Asterisk Openfire Asterisk-IM Plugin Performance Observation |
| 8:15AM |
1 |
FXS port doesn't provide dialtone |
| 7:33AM |
3 |
Asterisk 1.4.21 stalls? |
| 2:13AM |
1 |
Can't make asterisk work...how to test? |
| |
| Thursday June 19 2008 |
| Time | Replies | Subject |
|---|
| 7:50PM |
2 |
SIP over TCP development in 1.6 branch? |
| 7:23PM |
1 |
Asterisk + zap + sangoma A104D - how to setup call using particular timeslot |
| 7:06PM |
1 |
CLI> show queues NOT WORKING WELL |
| 4:51PM |
2 |
Trouble with PRI config |
| 6:22AM |
5 |
Grandstream Busy Light Fields |
| 12:21AM |
1 |
Mapping multimedia keys: "pressed key not recognized" |
| |
| Wednesday June 18 2008 |
| Time | Replies | Subject |
|---|
| 10:34PM |
3 |
Adding ;password=foo;method=bar to SIP uri |
| 9:02PM |
3 |
error: conflicting types for ‘bool’ |
| 8:34PM |
1 |
Interesting Directory Behaviour (not) |
| 8:32PM |
0 |
RES: GXW 4108 asterisk configuration |
| 7:45PM |
3 |
Website callback |
| 7:43PM |
0 |
Question on T1 OPS |
| 4:53PM |
0 |
T.38 Passthru w/ MediaGateway | Fax <-Analog Line-> ATA <-SIP-> Ast1.4T.38Passthru <-SIP-> MAX TNT <-PRI-> PSTN |
| 4:35PM |
0 |
sending DTMF during PROGRESS |
| 4:21PM |
2 |
[FreeBSD 6.3] Zaptel stops responding |
| 3:41PM |
2 |
zaptel 1.4.11 install |
| 1:25PM |
5 |
Who has the best call recording solution! |
| 1:05PM |
1 |
TRANSFER_CONTEXT ignored? |
| 8:00AM |
0 |
Connect caller and callee after Dial with G |
| 1:08AM |
2 |
Canadian Whitepage Listing Capability |
| |
| Tuesday June 17 2008 |
| Time | Replies | Subject |
|---|
| 9:56PM |
2 |
Zaptel 1.4.10.1 and OSLEC on Ubuntu 8.04 |
| 9:53PM |
1 |
GXW 4108 asterisk configuration |
| 8:26PM |
0 |
suggestions for IAX ATA device or phone in US |
| 6:55PM |
0 |
connectivity with oracle database and astreisk |
| 6:54PM |
1 |
'Together Everywhere' |
| 6:31PM |
0 |
Reg recording of calls |
| 5:55PM |
0 |
asterisk v1.6 queue() continue after answered call |
| 4:45PM |
1 |
Packages for ubuntu |
| 4:04PM |
1 |
Putting incoming sip call leg on MOH while dialing out other party**********NEED HELP************ |
| 3:39PM |
2 |
strange SIP-SIP delay |
| 3:15PM |
1 |
looking for help / input with Blind transfer from asterisk to zap |
| 2:54PM |
1 |
Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan |
| 1:23PM |
2 |
Problem with realtime? |
| 11:21AM |
0 |
Audiocodes |
| 9:05AM |
1 |
voicemail problem |
| 5:10AM |
3 |
Reg call recording |
| 4:03AM |
8 |
need ata suggestion |
| 12:24AM |
3 |
Call Center |
| |
| Monday June 16 2008 |
| Time | Replies | Subject |
|---|
| 7:01PM |
1 |
FW: Request to mailing list asterisk-users rejected |
| 1:44PM |
1 |
Euro_isdn PRI Line, callerid and usecallingpres" |
| 12:42PM |
0 |
Astribank and Celular Interface Module |
| 11:33AM |
3 |
Help! - Double NAT issue |
| 10:39AM |
2 |
Transfers with TE12xp |
| 9:30AM |
1 |
Agents getting "stuck" busy |
| 5:03AM |
1 |
asterisk was discunnected suddenly |
| 4:05AM |
0 |
Asterisk Manager Telnet Times Out? |
| 3:18AM |
0 |
Problem connecting to another server, Failed to authenticate on INVITE |
| |
| Sunday June 15 2008 |
| Time | Replies | Subject |
|---|
| 4:04PM |
5 |
OT How Digium Saved My Bacon! |
| 3:35PM |
4 |
*OT* DLI Ethernet Power Controller $289 (I paid $200 for a two port "webswitch") |
| 2:06PM |
7 |
Please Advice on Best High traffic fxo gateway/cards |
| 1:34AM |
1 |
[asterisk-dev] Astricon question: four or five tracks? |
| |
| Saturday June 14 2008 |
| Time | Replies | Subject |
|---|
| 6:32PM |
1 |
World Most Economical Predictive Dialer! |
| 9:28AM |
4 |
How to append "#" to the number before sending to Zap |
| 7:50AM |
1 |
play sound on a specific channel |
| 6:07AM |
0 |
cpu and ram requirements |
| 12:08AM |
4 |
Idiot's question |
| |
| Friday June 13 2008 |
| Time | Replies | Subject |
|---|
| 9:43PM |
0 |
strange iax authentication behavior |
| 7:23PM |
4 |
cdr-custom/Master.csv rotation |
| 7:08PM |
1 |
AEL Help |
| 6:31PM |
1 |
Need a SIP trunk provider for US - Dallas/TX |
| 6:05PM |
0 |
ASTERISK MFC R2 EWSD |
| 5:44PM |
0 |
TCP & UDP path not the same |
| 5:26PM |
0 |
how to make a ip to ip call |
| 4:52PM |
1 |
PRI crashing Asterisk |
| 1:29PM |
2 |
start n run an agi script on hangup |
| 10:16AM |
2 |
Behind NAT: source is fring software (SIP) |
| 5:18AM |
4 |
World Cheapest Predictive Dialer! |
| 12:45AM |
0 |
funny search engine terms |
| |
| Thursday June 12 2008 |
| Time | Replies | Subject |
|---|
| 9:37PM |
3 |
Astricon question: four or five tracks? |
| 8:36PM |
0 |
DUNDi question |
| 8:35PM |
1 |
Really destroying SIP dialog |
| 8:32PM |
3 |
Odd Polycom Reboot Issue |
| 5:58PM |
0 |
problems getting dialed information on asterisk |
| 5:58PM |
0 |
Phone selective variable setting? |
| 5:41PM |
0 |
Asterisk Unified communication features |
| 5:28PM |
0 |
[asterisk-biz] New faxing protocol. Good/Bad ? |
| 4:50PM |
0 |
Asterisk 1.4.21 Released |
| 4:49PM |
0 |
On Hold "Context"? |
| 4:41PM |
0 |
custom functions is voicemail |
| 3:55PM |
0 |
Fwd: Complimentary Subscription to VoIP Industry Publication |
| 3:16PM |
3 |
Using Asterisk Only as Voice Recording Solution. |
| 2:48PM |
0 |
iax2 qualify problem - PONG ignored |
| 12:36PM |
1 |
multiple CDRs for one call (multiple dial attempts during one call) |
| 11:41AM |
3 |
AGI after Hangup |
| 10:25AM |
2 |
Dial command and its g option |
| 9:09AM |
0 |
Friday the 13th lucky asterisk appliance day |
| 8:43AM |
3 |
Dial Command Option D Early Bridged |
| 8:23AM |
3 |
IAX2 phones, BRI and Analogue cards |
| 8:14AM |
3 |
Monitoring QoS |
| 4:16AM |
1 |
g729 codec for asterisk-1.6.0? |
| 1:47AM |
2 |
aSTERISK / Vicidial systems over 4MB fiber |
| 1:46AM |
1 |
Echo on PRI even with H/W echo cancel |
| 1:31AM |
1 |
? |
| 1:23AM |
3 |
Asterisk on SLOW solid state disk |
| |
| Wednesday June 11 2008 |
| Time | Replies | Subject |
|---|
| 10:52PM |
2 |
time on asterisk |
| 10:11PM |
1 |
Asterisk and XMPP (Jabber) : testing new application JabberReceive |
| 9:29PM |
3 |
asterisk calls per second |
| 6:21PM |
2 |
Losing CDR(accountcode) |
| 3:20PM |
2 |
Asterisk Data Calls |
| 1:43PM |
0 |
use of AJAM wth high load |
| 1:38PM |
1 |
decrease the time it takes for asterisk (fxsks) to answer |
| 1:17PM |
2 |
SIP call, updated with CID as it becomes available |
| 12:00PM |
2 |
Zaptel timer failure |
| 11:39AM |
1 |
SIP channel lock issues |
| 10:23AM |
1 |
1.4.20.1 hang -- extra info + gdb hangs |
| 7:40AM |
5 |
How to turn on the H323 logging on Asterisk |
| 5:30AM |
2 |
TE110P with 40,000 IRQ missess |
| 1:17AM |
2 |
Sound files custom path |
| |
| Tuesday June 10 2008 |
| Time | Replies | Subject |
|---|
| 9:28PM |
1 |
Delaying SIP disconnect after incoming call hangs up? |
| 9:22PM |
4 |
Problems configuring a PRI... |
| 9:16PM |
0 |
Seeking Collaboration in Development and Validation of an Anomaly Detection System for Asterisk |
| 8:00PM |
2 |
zaptel issue |
| 7:48PM |
1 |
Zaptel config |
| 7:37PM |
0 |
Blind transfers and ringback tone |
| 5:40PM |
1 |
Weird one way Audio situation |
| 5:03PM |
3 |
SayNumber while reading DTMF? |
| 2:34PM |
2 |
Camp / Callback feature in 1.4 |
| 2:11PM |
0 |
Debugging SIP call hangup reasons |
| 10:16AM |
1 |
meetme recording with security? |
| 6:13AM |
2 |
g729 open source codec and sample size |
| 1:01AM |
3 |
Asterisk : using setvar with IP Realtime and variable inheritance |
| |
| Monday June 9 2008 |
| Time | Replies | Subject |
|---|
| 10:51PM |
3 |
Interoffice phone setup |
| 8:46PM |
0 |
[OFFTOPIC][SPANISH] Creando una comunidad de asterisk en español |
| 8:36PM |
1 |
RFC2833 DTMF -- with an RTP debug log -- need some analysis/interpretation |
| 7:54PM |
1 |
Call hold in dialplan |
| 7:30PM |
1 |
Long call setup with non-PRI T1 |
| 7:29PM |
1 |
Polycom SIP and DHCP problem |
| 7:06PM |
1 |
3g video call using h324m_loopback not connecting |
| 7:01PM |
1 |
redfone fonebridge2 |
| 5:48PM |
0 |
fring and g729 |
| 5:36PM |
3 |
Asterisk Installation with Radius Support |
| 4:01PM |
1 |
SIP over M$ ISA |
| 12:45PM |
0 |
Asterisk 1.4.21-rc2 Now Available |
| 12:29PM |
2 |
Remote-Party-ID and selective CLI withold |
| 5:26AM |
2 |
Asterisk 1.2.28 + Realtime Queues - Thinks Queue is empty |
| |
| Sunday June 8 2008 |
| Time | Replies | Subject |
|---|
| 9:10PM |
2 |
Diverted Call Information on PRI |
| 8:42PM |
1 |
Asterisk can handle only 200 to 300 SIP device registrations |
| 2:13PM |
0 |
How to set name of call wav recording file in outgoing/call file? |
| 4:16AM |
1 |
MeetMe Limits |
| |
| Saturday June 7 2008 |
| Time | Replies | Subject |
|---|
| 10:37PM |
1 |
Manager Originate CDR problem |
| 3:37PM |
5 |
Fax on FXS |
| 1:03AM |
3 |
Logitech DiNovo Mini keyboard with myth |
| |
| Friday June 6 2008 |
| Time | Replies | Subject |
|---|
| 9:33PM |
5 |
features.conf not working |
| 9:15PM |
0 |
Anyone using zaptel analogue hardware in Singapore? |
| 7:58PM |
1 |
Help-ASTERISK-MFCR2 |
| 7:15PM |
2 |
MiixMonitor filename for queue calls. |
| 7:13PM |
1 |
SIP call recording |
| 5:37PM |
1 |
Asterisk not picking up incoming calls from TDM400P |
| 5:23PM |
2 |
Bad ringback tone on zap channel |
| 4:23PM |
1 |
Zap channels state |
| 2:35PM |
0 |
MixMonitor Not recording whole calls |
| 1:57PM |
1 |
Block on hold |
| 1:52PM |
0 |
Reminder TODAY Friday June 6th at 12 Noon EDT VoIP Users Conference |
| 1:34PM |
3 |
bad call quality |
| 1:24PM |
1 |
Disable sending CNAM over facility for 2bct |
| 1:18PM |
0 |
Asterisk and TDD |
| 12:20PM |
1 |
Sending texts questions. |
| 12:01PM |
1 |
1.4.20.1 hang -- three times in 1.5 days (TC400B at fault ?) |
| 4:43AM |
3 |
fxotune vs rxgain/txgain |
| 12:02AM |
0 |
SIP Phone, multiple line, but one call at a time ? |
| |
| Thursday June 5 2008 |
| Time | Replies | Subject |
|---|
| 10:24PM |
5 |
Asterisk video alternatives |
| 10:08PM |
5 |
PoE budget |
| 10:01PM |
3 |
Similar extension numbers for multiple users |
| 9:08PM |
0 |
RECALL: Lithium batteries for Polycom Soundstation 2W |
| 4:25PM |
0 |
Asterisk -> Nortel CS1K via NRS |
| 3:57PM |
1 |
detecting which party hung up |
| 1:58PM |
1 |
Default ringtone |
| 9:31AM |
0 |
About H323 configuration on Asterix |
| 7:56AM |
1 |
remote server with Snom 190 |
| 6:55AM |
1 |
handling SIP trunk with limited concurent calls |
| 3:02AM |
2 |
fxotune question |
| |
| Wednesday June 4 2008 |
| Time | Replies | Subject |
|---|
| 11:36PM |
0 |
Codec troubles |
| 11:01PM |
1 |
http://1ezphone.com/download = sorry no "s" |
| 10:52PM |
2 |
Browser based VoIP client? None of them are very full featured |
| 10:46PM |
0 |
Browser based VoIP client? - http://1ezphone.com/downloads |
| 10:24PM |
3 |
Avaya IP Phones with * |
| 10:18PM |
0 |
AST-2008-009: (Corrected subject) Remote crash vulnerability in ooh323 channel driver |
| 10:07PM |
0 |
Patch for app_asr.c: DTMF instead of goto |
| 10:04PM |
0 |
Asterisk-Addons 1.2.9 and 1.4.7 released; Asterisk-Addons 1.6.0-beta4 now available |
| 10:03PM |
0 |
AST-2008-009: AST-2008-007 Cryptographic keys generated by OpenSSL on Debian-based systems compromised |
| 7:34PM |
1 |
Lumenvox - Gentoo |
| 7:04PM |
2 |
queue delay between calls to agents |
| 6:51PM |
0 |
disable "send reply" in asterisk voicemail |
| 6:42PM |
4 |
Browser based VoIP client? |
| 3:52PM |
4 |
Asterisk 1.6 vs 1.4? |
| 3:14PM |
1 |
G.722? |
| 2:55PM |
1 |
init.d script no longer uses safe_asterisk |
| 1:51PM |
0 |
Asterisk 1.4.20.1 problems |
| 12:42PM |
0 |
911 via MAX TNT |
| 11:30AM |
0 |
This AEL vs. Dialplan thing ... |
| 10:43AM |
2 |
connecting 2 FXS together |
| 10:30AM |
0 |
busydetect=yes, busycount=5: hangup automtically without reason, why? |
| 8:47AM |
1 |
Error Wile starting AsterFax |
| 6:36AM |
0 |
How to improve group call User Interface ? |
| 4:47AM |
0 |
SPA 3102 disconnect tone setting for China |
| |
| Tuesday June 3 2008 |
| Time | Replies | Subject |
|---|
| 11:53PM |
3 |
What does reason 8 for failure means in Manager |
| 11:43PM |
2 |
911 via MAX TNT ?? |
| 10:56PM |
3 |
handling jabber status |
| 9:36PM |
0 |
Videos of Asterisk-Tag.org online |
| 9:22PM |
0 |
Asterisk 1.4.21-rc1 Now Available |
| 8:05PM |
0 |
Asterisk 1.2.29 Released |
| 7:53PM |
0 |
AST-2008-008: Remote Crash Vulnerability in SIP channel driver when run in pedantic mode |
| 7:37PM |
2 |
#include |
| 6:54PM |
8 |
Queue is sending calls to Agents even when they are in use |
| 6:41PM |
0 |
unload zaptel if asterisk is not running (or crashes) |
| 4:46PM |
1 |
Problem with several includes in ARA |
| 4:43PM |
2 |
Asterisk Seg faulting.... No core dump. |
| 3:37PM |
3 |
Trouble with Polycom phones |
| 3:12PM |
3 |
Asterisk 1.4.20.1 with bad gsm file playback |
| 2:52PM |
0 |
Asterisk 1.4.20.1 and Realtime |
| 1:41PM |
1 |
setting channel variable in DeadAGI script |
| 1:21PM |
8 |
Any reason to *not* use AEL? (Also, MixMonitor q) |
| 12:22PM |
0 |
externip not setting extern ip/ loop detected issues |
| 9:50AM |
1 |
G.722 over ISDN PRI/BRI |
| 7:57AM |
0 |
artificial |
| 7:55AM |
1 |
Media time out for SIP and IAX Trunk |
| 5:37AM |
0 |
Final call for Astricon talk topics |
| 1:05AM |
4 |
Date in Dialplan |
| |
| Monday June 2 2008 |
| Time | Replies | Subject |
|---|
| 11:44PM |
1 |
What do I Copy to Migrate to New Machine? |
| 10:11PM |
0 |
For all your office needs... and voip-greylisting |
| 9:27PM |
0 |
queues with callback members |
| 9:04PM |
1 |
PBX Functionality for Less than the Price of a Key System (3Com Asterisk IP Telephony Appliance) |
| 8:24PM |
3 |
Source for Digium TDM400P card |
| 8:15PM |
2 |
ast_compile_ael2: Warning: file /etc/asterisk/extensions.ael, line 932-932: Empty Extension! |
| 6:07PM |
1 |
Why doesn't Pickup() work?? |
| 4:10PM |
2 |
QSIG transfer of calls away from Asterisk? |
| 1:32PM |
0 |
Voicemail question |
| 1:27PM |
0 |
Good Asterisk HA Load balancing document. |
| 1:20PM |
0 |
looking for Asterisk smarties in Bangalore, India. |
| 12:09PM |
1 |
passing a call to the call queue |
| 11:25AM |
6 |
Help with E1s |
| 9:55AM |
1 |
Query about AsterFax Installation |
| 9:45AM |
0 |
OT - Intel IPP 5.3 |
| 8:25AM |
3 |
RTCP debugs and stats |
| |
| Sunday June 1 2008 |
| Time | Replies | Subject |
|---|
| 10:37PM |
2 |
Trouble with unicall |
| 9:54PM |
5 |
New faxing protocol. Good/Bad ? |
| 9:00PM |
3 |
How to limit extension to allow only internal calls |
| 5:26PM |
1 |
Mysql and extensions.conf |
| 12:13AM |
2 |
Asterisk and V.22bis |