| Wednesday January 31 2007 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
Softphone for Palm |
| 10:48PM |
1 |
Fax from PAP2 through a zap channel to PSTN |
| 8:40PM |
2 |
no lights on TE405P, but shows up in lspci, modules loaded |
| 7:57PM |
2 |
kewlstart disconnect threshold |
| 7:52PM |
1 |
FreePBX/Debian Aborts Call While Connecting |
| 7:14PM |
2 |
Which Java FastAGI implementation has the most "market share"? |
| 6:28PM |
0 |
How would you compare feature set to a Metaswitch? |
| 4:44PM |
1 |
how to get the status of failed call files |
| 12:18PM |
0 |
jastAGI |
| 11:34AM |
1 |
Polycom IP 501+India |
| 11:09AM |
4 |
Help with semaphores |
| 10:45AM |
0 |
E911 Bill Announced |
| 10:04AM |
0 |
Compiling NVFaxDetect and other Newman apps on Asterisk 1.4 |
| 9:59AM |
3 |
Queue Status |
| 9:57AM |
0 |
Storing recordings |
| 9:11AM |
0 |
(no subject) |
| 8:50AM |
0 |
Line drops strange problem(got event On hook) |
| 8:23AM |
5 |
Testing IVR / Callcenter applications |
| 6:36AM |
3 |
Hi Honies! I'm home! |
| 6:21AM |
2 |
Regarding Call Queue |
| 4:14AM |
0 |
Asterisk 1.2.14 bristuff app_pickup.so |
| 1:35AM |
0 |
ELMEG IP290 and voicemail |
| |
| Tuesday January 30 2007 |
| Time | Replies | Subject |
|---|
| 2:44PM |
1 |
OT: Asterisk 1.2.X, IAXModem 0.2.0 + HylaFAX+ 5.0.3 interop probl em |
| 1:26PM |
2 |
Should I use sip gateway of PCI card? |
| 1:00PM |
3 |
Toll-free dialing via PRI problem |
| 11:48AM |
1 |
Record file name Agent |
| 11:44AM |
0 |
Signaling OK but no voice through X100P |
| 11:06AM |
1 |
Queue Dial Plan |
| 10:44AM |
2 |
Cisco SmartSwitch |
| 10:22AM |
1 |
Give "Busy" to the 3rd call on a BRI using chan_capi |
| 10:18AM |
1 |
One-way audio after several minutes 1.4.0 |
| 9:55AM |
2 |
web-meetme cbmysql not registered |
| 9:52AM |
3 |
musiconhold restarts for every extension |
| 9:30AM |
0 |
Diva PCI 2.01 + isdn2linux + asterisk: Dropped a signal frame |
| 9:05AM |
6 |
Re: [asterisk-dev] Dynamically Adding A Context |
| 8:17AM |
1 |
Dynamically Adding A Context |
| 8:12AM |
5 |
Asterisk dual contexts stupidity |
| 7:48AM |
0 |
Looking for Sugar CRM installer for an Asterisk |
| 7:24AM |
2 |
Comments on Billing reconcillation with providers |
| 7:02AM |
1 |
No intercom splash tone? |
| 5:58AM |
1 |
Strange problem |
| 4:45AM |
2 |
Problem with Voipjet ... |
| 3:16AM |
1 |
snmp Monitor for asterisk boxes |
| |
| Monday January 29 2007 |
| Time | Replies | Subject |
|---|
| 10:38PM |
1 |
Timeout in IAX vs SIP |
| 9:14PM |
1 |
detecting avaya busy tone |
| 9:08PM |
0 |
Cisco PRI gateway with MGCP control |
| 7:28PM |
0 |
"disconnect clear time" -- calling party control and TDM-400 |
| 6:35PM |
1 |
TDM Cards or PSTN>VOIP Gateways? |
| 3:45PM |
0 |
Dropped call issue with IAX Trunking |
| 3:16PM |
1 |
Asterisk, VoIP and Linux Blog. |
| 10:39AM |
4 |
Installed TDM02B - Problem when other end hangs up |
| 10:00AM |
1 |
internal and external interfaces |
| 8:31AM |
1 |
SIP + short numbers + name of customer |
| 8:02AM |
1 |
LookupCIDName / LookupBlacklist syntax |
| 7:55AM |
1 |
put Agi script in queue |
| 6:40AM |
0 |
Rxfax and txfax |
| 5:57AM |
3 |
Pickup() ringing extension and call waiting |
| 3:58AM |
1 |
licence quick question |
| 3:29AM |
1 |
parsing extensions |
| 3:08AM |
0 |
SIP SDP keep original codec selection? |
| 12:26AM |
2 |
Rxfax and Txfax on Asterisk 1.4 |
| |
| Sunday January 28 2007 |
| Time | Replies | Subject |
|---|
| 8:09PM |
1 |
Queue Manager |
| 7:52PM |
4 |
Cordless SIP Phones |
| 7:06PM |
0 |
Trouble outgoing VOIP Provider Calls |
| 6:13PM |
0 |
Test Hardware |
| 6:06PM |
2 |
Trouble with incoming calls |
| 5:42PM |
1 |
Heartbeat on Digium T1 PCI cards? |
| 4:16PM |
1 |
Voicemail from sip phones |
| 4:03PM |
0 |
Automating the setting/clearing of a flag |
| 3:09PM |
0 |
Add current extension dynamically to template? |
| 2:51PM |
0 |
Re: Migration to Asterisk 1.4 |
| 1:39PM |
0 |
PHP sip client |
| 12:54PM |
1 |
T1 Wire Level Tapping |
| 9:41AM |
0 |
Channels Banks that support neon MWI |
| 8:15AM |
1 |
NAT: RTP Path Optimization |
| 8:04AM |
2 |
Mabe OT? What managed switch is best for VoIP application? |
| 7:07AM |
1 |
Enterprise quality SIP provider |
| 4:16AM |
0 |
AsteriskNow - H323 support for trunks |
| 2:03AM |
1 |
Transfer on RTP timeout? |
| |
| Saturday January 27 2007 |
| Time | Replies | Subject |
|---|
| 10:08PM |
2 |
Response on dialin - no extension |
| 3:32PM |
1 |
HFC-card and TDM400 with bristuff |
| 1:31PM |
1 |
FXS - Init Indirect Registers UNSUCCESSFULLY. |
| 12:18PM |
0 |
BarCampUSA Tickets go on sale this Thursday the 1st of February 2007 |
| 12:09PM |
1 |
How to fix error when paging |
| 10:55AM |
2 |
max tnt pri voice channels 56k or 64k, does it matter, selection parameter? |
| 8:20AM |
5 |
H.264 *Not Patented* |
| 5:50AM |
3 |
Simple question |
| 4:17AM |
1 |
Via EPIA channel_find_locked: Avoided initial deadlock |
| |
| Friday January 26 2007 |
| Time | Replies | Subject |
|---|
| 11:33PM |
4 |
Does X100P decode caller ID? |
| 11:08PM |
0 |
IP-to-IP dial: no answer or no listener? |
| 7:41PM |
0 |
realtime sipusers and rtcachefriends... bigheadache!! |
| 3:25PM |
1 |
How do I change the rtp packet size in a Cisco 7490 from 10ms to 20ms |
| 2:55PM |
1 |
Sample Config. |
| 2:42PM |
1 |
Show call coming back from Call Parking |
| 2:36PM |
1 |
Nobody there, continuing... |
| 1:42PM |
2 |
X100P - zttools says red status |
| 1:10PM |
4 |
Polycom Provistioning Issue |
| 12:31PM |
2 |
Only secretary can call the boss, all others only reach the secretary when dial the boss extension |
| 12:29PM |
0 |
Asterisk dropping audio |
| 11:38AM |
3 |
ATCOM AT 468 manuals and firmware anyone? |
| 10:53AM |
0 |
Asterisk on IBM NEBS compliant Blade Server |
| 10:52AM |
2 |
PHP AGI script callerid question |
| 10:31AM |
1 |
h323 compile error |
| 10:27AM |
0 |
TDM400P with FXS module problem |
| 10:20AM |
1 |
Analog FXO status checking |
| 9:31AM |
0 |
TDM2401 (FXO) Hangup |
| 8:54AM |
3 |
International Carriers |
| 7:27AM |
0 |
Recompiled app_xyz.so and Asterisk Dynamic Loader |
| 7:02AM |
0 |
Problem solved |
| 6:56AM |
0 |
wireless sip phone with auto answer - are there any |
| 6:55AM |
1 |
Ringing oddity/stupidity |
| 6:54AM |
0 |
Dialplan - play sample, interrupt on * and return value? |
| 5:22AM |
1 |
strange msg |
| 5:22AM |
1 |
Asterisk Recording & Volume |
| 3:55AM |
1 |
asterisk.conf |
| 3:43AM |
2 |
Hello Everybody, my problem with voicemail.conf |
| 3:27AM |
1 |
WellTech 380x Gateway |
| 3:04AM |
2 |
pickup internal and external calls |
| 2:48AM |
4 |
Sangoma card dying after 1hour |
| 12:21AM |
0 |
barge calls and record them at the same time |
| 12:17AM |
3 |
Zap channels staying offhook - restart required |
| |
| Thursday January 25 2007 |
| Time | Replies | Subject |
|---|
| 11:36PM |
0 |
Re: Realtime - one database driver, multiple databases |
| 6:56PM |
0 |
TC400B Transcoder Card Shipping |
| 6:48PM |
0 |
dacs support on Digium T1 equipment. |
| 5:17PM |
1 |
Failing to compile chan_capi |
| 1:27PM |
1 |
IMAP Voicemail Storage |
| 1:27PM |
2 |
Adding 4 more POTS lines |
| 12:33PM |
0 |
low audio (sometimes) |
| 12:07PM |
1 |
Cannot xfer parked callers |
| 11:39AM |
0 |
SVN trunk synchro failure |
| 11:37AM |
0 |
On-hold calls dropped when new call comes in |
| 11:31AM |
1 |
unable to create channel, in strange state, exited non-zero, etc. |
| 10:40AM |
2 |
Asterisk 1.4 problem with ztdummy and MeetMe() |
| 10:16AM |
1 |
background() with "m" option |
| 10:11AM |
1 |
dialplan and "*" |
| 9:49AM |
1 |
IAX softphone fails through PRI trunks with Hangup |
| 8:16AM |
2 |
TE110P and HDLC problems |
| 7:50AM |
1 |
NTL Hangup |
| 7:42AM |
0 |
Planning 48 Station Install, Need advice on several topics |
| 7:32AM |
2 |
1.4 - SLA |
| 5:18AM |
0 |
Initial DTMFs arriving too quickly? |
| 5:05AM |
2 |
Do I need a CH1 licence for Cisco Phones ? |
| 4:55AM |
3 |
Asterisk very slow when internet down |
| 4:33AM |
0 |
asterisk 1.4: gui registration differs from non-gui |
| 2:29AM |
3 |
Starting Asterisk in vvvvvvvvvvverbose mode |
| 12:03AM |
1 |
issue with ivtv & wctdm zaptel drivers (TDM PCI Master abort) |
| |
| Wednesday January 24 2007 |
| Time | Replies | Subject |
|---|
| 11:31PM |
1 |
SPA3K to SPA3K DTMF issue |
| 10:38PM |
1 |
Polycom Firmware -- Was: Asterisk 1.4 & Polycom buddy status |
| 9:59PM |
1 |
Multiple parking lot |
| 6:40PM |
1 |
Best way to connect analog modem |
| 6:24PM |
0 |
realtimeinsert and realtimedelete functions |
| 2:28PM |
2 |
Disconnected Calls |
| 1:20PM |
1 |
Call parking causes Asterisk to crash |
| 1:05PM |
1 |
Dell Server Question |
| 12:55PM |
3 |
setting up AMD |
| 12:26PM |
2 |
channel name |
| 11:47AM |
1 |
Semi OT - Point to Point FXO/FXS Gateway Communication |
| 11:01AM |
2 |
convert URI string to lowercase |
| 10:48AM |
1 |
OT - Cisco 7960 functionality |
| 9:20AM |
1 |
Getting confused on signalling mode Vs framing and encoding: T1 CAS |
| 9:17AM |
0 |
Agent Pre Acknowledgement Message |
| 9:08AM |
0 |
iax.conf setvar= like sip.conf setvar=? |
| 9:05AM |
1 |
iax2 prun realtime peer only can't prune user |
| 7:38AM |
1 |
Panasonic Hybrid Integration Advice Needed |
| 5:55AM |
1 |
Grandstream GXP2000 and Interception of call ? |
| 5:38AM |
1 |
vxml support |
| 5:27AM |
1 |
AOC on misdn? |
| 3:43AM |
0 |
NAT |
| 3:31AM |
0 |
beronet DTMF detection problem with some Telecom Italy lines |
| 3:00AM |
2 |
Digium Forums |
| 2:42AM |
0 |
Asterisk IAX and Shorewall QoS ? |
| 2:38AM |
1 |
Query Failed because: Incorrect information in file: './asterisk/sip.frm' |
| 2:11AM |
1 |
ChanIsAvail kills dialplan processing when no Zap available on 1.2.14. |
| 12:44AM |
0 |
NewTopic - Asterisk and Cisco AS5300 via E1/PRI |
| |
| Tuesday January 23 2007 |
| Time | Replies | Subject |
|---|
| 7:56PM |
2 |
No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 7:08PM |
0 |
cmd Backgound problem with option m |
| 5:27PM |
1 |
Echo on IP phones... |
| 5:03PM |
1 |
DeStar 0.2.2 released! |
| 5:00PM |
0 |
Problem connecting PAP2 over wifi bridge |
| 4:08PM |
0 |
* 1.0.9 Voicemail record name does not playb ack in Directory() <--solved |
| 3:56PM |
1 |
DB_DELETE Function in 1.4 |
| 3:07PM |
0 |
automon and MONITOR_EXEC |
| 2:30PM |
0 |
AW: Snom 320 echo |
| 2:10PM |
5 |
Snom 320 echo |
| 1:14PM |
1 |
OT: High Quality Wireless Headset for Cisco IPPhones and * |
| 1:01PM |
1 |
OT: High Quality Wireless Headset for Cisco IP Phones and * |
| 12:08PM |
3 |
[OT] Mark Spencer Presents AsteriskNOW on Youtube |
| 11:29AM |
0 |
* 1.0.9 Voicemail record name does not playback in Directory() |
| 9:34AM |
0 |
IGNORE: AEL parse failure on 1.2.14 |
| 9:12AM |
0 |
AEL parse failure on 1.2.14 |
| 9:08AM |
1 |
Rhino cards lock up system -- anyone else ever seen this? |
| 8:07AM |
2 |
Asterisk 1.4 & Polycom buddy status |
| 7:47AM |
2 |
stress-test realtime voicemail with sipp |
| 7:36AM |
1 |
"bad gateway" error on snom display |
| 6:54AM |
0 |
problems with dtmf |
| 5:37AM |
0 |
PRI/Q.sig between Cisco & Nortel |
| 5:01AM |
1 |
"No Mailbox" Prompt |
| 4:36AM |
0 |
beronet BRI card sometimes not detecting tones |
| 4:36AM |
12 |
How to exit from console? |
| 3:28AM |
3 |
Dial plan constructions suggestions? |
| 3:27AM |
2 |
Can't find asterisk.ctl under CentOS installation |
| 2:42AM |
1 |
Operate on registrations |
| 12:42AM |
4 |
weird undocumented extensions such as s-BUSY |
| |
| Monday January 22 2007 |
| Time | Replies | Subject |
|---|
| 9:09PM |
0 |
2-way MS-GSM support in Asterisk? |
| 7:47PM |
1 |
OT: Optimum voice problems. |
| 7:42PM |
0 |
Aastra 480i freezes |
| 7:16PM |
0 |
Why app_rx and app_tx when we have IAXModem and Hylafax and hy-email2fax? Should we reinvent the wheel? |
| 7:13PM |
0 |
Weird names vs. correct agent's ext. |
| 6:56PM |
1 |
Music on Hold on IP Phones with FreePBX 2.2.0 |
| 6:35PM |
1 |
Fwd: Hater |
| 5:41PM |
0 |
7 points of comparison Polycom 430/501 and A astra 480i. Which one to choose ? |
| 5:37PM |
1 |
LDAP get and Asterisk 1.4 |
| 5:35PM |
0 |
how to make a video phone call |
| 5:01PM |
2 |
Streaming audio file while working in background ? |
| 3:27PM |
0 |
CentOS and 1.4 |
| 1:55PM |
1 |
2 ring delay before asterisk answer |
| 1:08PM |
4 |
X100P how do i recieve incomming calls? |
| 12:15PM |
3 |
7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ? |
| 11:32AM |
1 |
Load Balancing |
| 11:23AM |
2 |
agi script as member in queue |
| 11:07AM |
0 |
Videopodcast about Asterisk |
| 11:02AM |
2 |
tdm400p not working with brazilian lines |
| 10:56AM |
0 |
Asterisk and VoIP @ Southern California Linux Expo (SCALE 5x) |
| 10:53AM |
0 |
IP of SIP server changing |
| 10:32AM |
1 |
Detecting Disconnected Numbers - PRI |
| 10:28AM |
1 |
Requirements for faxes to work properly |
| 10:19AM |
1 |
STUN and SNMP |
| 10:16AM |
1 |
No Audio for Extension to Extension |
| 10:09AM |
1 |
QueueMemberStatus/Status Field |
| 10:06AM |
4 |
Problems with rxfax |
| |
| Sunday January 21 2007 |
| Time | Replies | Subject |
|---|
| 11:38AM |
0 |
VoIP-GSM gateway problem |
| 11:34AM |
1 |
ISDN30 and TDM400P + FAXing ... |
| 10:01AM |
2 |
Backports to 1.2.14 of 1.4.0 app_queue features. |
| |
| Saturday January 20 2007 |
| Time | Replies | Subject |
|---|
| 11:59PM |
1 |
Connect a Skype adapter to TDM400P |
| 7:44PM |
0 |
Attention all Aastra IP phone users... |
| 7:17PM |
0 |
1.4 svn voicemail broken? |
| 6:46PM |
0 |
extra sounds description file? |
| 4:21PM |
1 |
func_odbc still working in trunk? |
| 10:10AM |
1 |
SIP registration problem w/ SBC |
| 8:33AM |
3 |
On what distribution is www.asterisknow.com based on ? |
| 8:10AM |
1 |
error message |
| 6:01AM |
3 |
Cisco 7970 Unprovisioned |
| 2:27AM |
1 |
Connecting 2 asterisk servers |
| 12:06AM |
0 |
CAS on Sangoma boards |
| |
| Friday January 19 2007 |
| Time | Replies | Subject |
|---|
| 8:28PM |
2 |
chanskype |
| 7:48PM |
1 |
Asterisk 1.4 and g723 |
| 7:01PM |
2 |
Anyone know what this warning is about? Nothing in list history about it either.. |
| 4:03PM |
1 |
Re: asterisk-users Digest, Vol 30, Issue 79 |
| 3:57PM |
1 |
Incoming SIP line does not display CallerID correctly |
| 1:53PM |
1 |
using the Manager to connect caller to conference |
| 1:28PM |
0 |
Set(X=10|g) vs Set(GLOBAL(X)=10) |
| 1:07PM |
5 |
Ebay Unwired Buyer, Using Asterisk? |
| 12:46PM |
1 |
how can PRI, BRI and analog cards achieve a synchronous clock / timing |
| 11:59AM |
0 |
MIT Using Asterisk - VM Server |
| 10:51AM |
5 |
mISDN |
| 10:06AM |
0 |
Open Source Hosted PBX |
| 9:49AM |
2 |
Disconnect Supervision UK / BT solution? |
| 8:59AM |
1 |
Red: Sip Phone CID |
| 8:41AM |
1 |
Set Parameter of Call Files |
| 8:19AM |
1 |
Integrating asterisk with Toshiba Astrata DK380 |
| 7:38AM |
2 |
Announce option for meetme - is it used? |
| 7:21AM |
0 |
CPU Bandwidth Consumption |
| 6:10AM |
0 |
direct transfer in features |
| 6:01AM |
0 |
pickup call out of menu |
| 5:40AM |
1 |
meetme ${DATETIME} variable update |
| 4:15AM |
2 |
Voice Recognition |
| 3:33AM |
0 |
mysterious SIP packets to Cogent |
| 12:43AM |
1 |
IAX2/SIP gateway for Belgium and western Europe |
| |
| Thursday January 18 2007 |
| Time | Replies | Subject |
|---|
| 11:18PM |
2 |
How to limit IAX calls |
| 10:58PM |
1 |
Need help with if command |
| 10:19PM |
1 |
Detecting open SIP channels in the dial plan |
| 10:09PM |
0 |
queue stats - outgoing calls |
| 9:10PM |
4 |
NAT solutions |
| 7:59PM |
1 |
Dialplan - busy and unavailable without priority jumping |
| 3:03PM |
1 |
COMPLETEAGENT vs. COMPLETECALLER |
| 2:13PM |
0 |
meetme list (unmonitored)? |
| 1:48PM |
1 |
Queues Question |
| 1:37PM |
1 |
Simplifying similiar sip trunks |
| 11:48AM |
0 |
Re: [cisco-voip] voice router with free gatekeeper !!! |
| 11:15AM |
2 |
Snom has dialtone after putting a person on hold |
| 10:35AM |
1 |
RE: Polycom buddies question |
| 10:00AM |
4 |
About BRI / ISDN hardware. What to buy? |
| 10:00AM |
0 |
(OT) Madge LMC 10.0 |
| 9:46AM |
1 |
Passing video calls / bearer capability thru PRI |
| 9:45AM |
3 |
connecting a FXS-to-sip 4 port device to an avaya system |
| 8:53AM |
0 |
Re: Realtime Voicemail Password Change WORKING NOW |
| 8:21AM |
1 |
Sip Phone CID |
| 7:31AM |
0 |
Thoughts on CPE server... |
| 7:28AM |
5 |
1 phone 2 voicemail accounts |
| 4:50AM |
1 |
TDM 400P in the UK - doesn't see ringing calls hanging up before answer |
| 4:32AM |
2 |
Asterisk not hanging up |
| 3:45AM |
1 |
Bristuff with 2.6.19 |
| 3:37AM |
0 |
changing VoiceMailMain functionality |
| 3:35AM |
1 |
IAX call limit |
| 3:35AM |
0 |
re: putting 2 SIP channels together - hangup issues |
| 3:19AM |
1 |
Linksys (PAP2) Registration problem |
| 1:29AM |
1 |
Problems with Digium TE410 |
| 12:46AM |
1 |
sangoma a102d + Asterisk 1.2.14 ... bridging together 2 call legs on same PRI? |
| 12:43AM |
3 |
function call out of AGI script |
| |
| Wednesday January 17 2007 |
| Time | Replies | Subject |
|---|
| 10:48PM |
1 |
help. newbie asterisk installation problem. |
| 4:22PM |
0 |
Hospitals using Asterisk? |
| 3:26PM |
0 |
Asterisk Legacy PBX integration and fail-over question, |
| 2:47PM |
0 |
STUN in Asterisk 1.4 |
| 2:02PM |
0 |
Unknown warning messages |
| 1:35PM |
1 |
I need to connect Asterisk to a Nortel Meridian phone plant |
| 1:01PM |
3 |
Network\Snom phone oddity |
| 11:29AM |
2 |
One way choppy sound |
| 11:04AM |
2 |
AbsoluteTimeout with canreinvite=yes |
| 9:34AM |
4 |
FW: Realtime Voicemail Password Change Not Working |
| 9:08AM |
4 |
Erratic Snom MWI lights |
| 8:59AM |
1 |
Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian |
| 8:43AM |
3 |
Asterisk 1.4 and CDR |
| 8:41AM |
1 |
Dtmf tones and SIP |
| 8:39AM |
1 |
transfer problem |
| 8:32AM |
0 |
Monitor or log peer performance |
| 8:08AM |
3 |
Callback/ringback |
| 7:38AM |
1 |
dtmf problem -- second part |
| 7:13AM |
2 |
Asterisk registration |
| 7:08AM |
1 |
2 Questions: Answer with music don't work and Voicemail direct access ? |
| 3:30AM |
1 |
Question about FXO/FXS device. |
| 3:27AM |
0 |
Re: [asterisk-dev] Question about FXO/FXS device. |
| 1:55AM |
0 |
TDM2400 Hardware Echo Cancel (Adam Sharples) |
| 1:36AM |
0 |
disable external-outgoing calls per extension |
| 12:43AM |
1 |
Using the SIPAddHeader Application |
| 12:15AM |
0 |
newbie asterisk 1.4 installation problem |
| 12:09AM |
4 |
windows mobile 5 softphone for square screen devices |
| |
| Tuesday January 16 2007 |
| Time | Replies | Subject |
|---|
| 11:08PM |
0 |
Dell 860 |
| 10:39PM |
2 |
TDM404B VS TDM2401B |
| 8:14PM |
3 |
Realtime Voicemail Password Change Not Working |
| 7:51PM |
1 |
Refreshing DNS lookups |
| 6:32PM |
2 |
Really Big Queues |
| 6:17PM |
2 |
prompt for "send a message" not played in VM main, HOWTO resolve |
| 5:39PM |
0 |
Absolute Timeout or Dial Limit option??? |
| 5:25PM |
0 |
ERROR[2453]: chan_zap.c:8142 zt_pri_error: !! Unexpected Channel selection 3 |
| 3:45PM |
2 |
Polycom IP601 - some hints working, not others? |
| 2:08PM |
4 |
Audiocodes GPL |
| 1:41PM |
0 |
Help with DISA |
| 1:35PM |
0 |
asterisk startup is slow |
| 1:33PM |
1 |
Ring tone too loud on IAX channel |
| 1:32PM |
0 |
Asterisk 1.2.14 and Audiocodes Mediant 1000 |
| 1:01PM |
1 |
Outbound IVR for Asterisk |
| 1:01PM |
0 |
Asterisk Bootcamp in Pacific Northwest (Vancouver, BC) |
| 12:56PM |
2 |
force ulaw passthrough if call from modem extension? |
| 10:53AM |
0 |
MP3player distortion with Asterisk 1.4 |
| 10:27AM |
1 |
Asterisk, SpanDSP and RXFax |
| 10:08AM |
5 |
How to detect long calls |
| 8:00AM |
2 |
IAX2 softphones can't (won't?) use PRI trunks.... |
| 7:45AM |
3 |
IAX Trunk timing |
| 7:07AM |
0 |
spa942 and asterisk 1.2 |
| 6:59AM |
4 |
TDM2400 Hardware Echo Cancel |
| 6:42AM |
0 |
Polycom phone locks up, send sip busy messages |
| 5:13AM |
0 |
Disallowing unauthorized calls to Cisco & Polycom phones |
| 5:08AM |
0 |
IAX Channels language |
| 5:02AM |
2 |
command like break ore exit in the dialpan |
| 4:10AM |
1 |
J1/INS1500 and the Redirect Number |
| 3:52AM |
0 |
ENUMLOOKUP debug |
| 2:34AM |
0 |
zaptel hardware detection with genzaptelconf |
| 2:01AM |
1 |
Didn't get a frame from channel |
| 1:24AM |
0 |
input request: progzone and zaptel hangup |
| |
| Monday January 15 2007 |
| Time | Replies | Subject |
|---|
| 10:18PM |
1 |
two-level administration tool for Asterisk (reposting) |
| 6:22PM |
3 |
Practical limit on dial prefixes for a route |
| 5:17PM |
0 |
S110M (FXS) Modules no longer seen on TDM400P |
| 5:02PM |
0 |
1-way audio |
| 3:48PM |
0 |
S400M (FXS) Modules no longer seen |
| 2:45PM |
2 |
Recording queue calls after an xfer? |
| 2:31PM |
0 |
Addpac 2620 don't relay DTMF to PSTN |
| 2:28PM |
6 |
connecting 2 asterisk servers through OpenVPN |
| 2:00PM |
2 |
Audiocodes Mediant 1000, Polycom, and no ringback on transfer |
| 12:56PM |
2 |
Queue cmd option 'i' |
| 12:38PM |
1 |
Asterisk PBX '&' '||' Grandstream GXP-2000 problem |
| 12:34PM |
0 |
help create asterisk cookbook |
| 12:23PM |
4 |
Nufone |
| 12:22PM |
5 |
Delay in Call Distribution using the Queue Application |
| 11:36AM |
3 |
Software callcenter |
| 11:08AM |
3 |
Queue and Interface time out |
| 10:55AM |
1 |
I have to register asterisk/sip with a sipproxy that does not support authentication? |
| 10:51AM |
1 |
TDM400P, fxotune and ADSL filters - Just a FYI, FWIW |
| 9:58AM |
0 |
.call files - no hangup |
| 9:53AM |
3 |
php agi - first phrase truncated, all others fine |
| 9:25AM |
1 |
ANY ADVICE ON THIS???? |
| 7:47AM |
0 |
SIP transfer issue |
| 7:43AM |
1 |
what happened to "sip list peers" |
| 7:23AM |
0 |
Parked calls with Asterisk 1.4.0 |
| 5:37AM |
1 |
Wanpipe 2.3.4-2 + kernel 2.6.19 = problems |
| 3:52AM |
2 |
Installing Asterisk 1.4 Documentation |
| 3:50AM |
2 |
Rt db lookup |
| 3:47AM |
0 |
OT: Quad-band cellphones with wifi & stablesipsupport |
| 3:38AM |
1 |
phpagi transfer example |
| 3:22AM |
0 |
Asterisk Realtime and MD5 authentication |
| |
| Sunday January 14 2007 |
| Time | Replies | Subject |
|---|
| 11:01PM |
3 |
OT: Quad-band cellphones with wifi & stable sip support |
| 5:40PM |
1 |
E&M ? |
| 4:01PM |
2 |
To 1.4 or not |
| 3:54PM |
1 |
Asterisk not hanging up calls |
| 12:08PM |
0 |
realtime mysql db performance difference with matching extensions |
| 11:58AM |
1 |
Problems with mISDN TE line |
| 9:57AM |
1 |
RE polycom fails registration |
| 8:41AM |
2 |
RE : TDM2400p bad sound quality |
| 4:25AM |
2 |
functions - fork |
| 1:23AM |
1 |
Particular DialPlan |
| 1:08AM |
0 |
External resource timeout |
| 1:06AM |
0 |
Ring PC speaker |
| 12:54AM |
2 |
Polycom registration fails |
| |
| Saturday January 13 2007 |
| Time | Replies | Subject |
|---|
| 5:48PM |
1 |
Caching Caller Name |
| 1:51PM |
2 |
fxotune Error |
| 12:23PM |
0 |
Hints on FXO channels |
| 9:30AM |
1 |
Possibleto use zaptel 1.4 with asterisk 1.2? |
| 7:26AM |
1 |
1.4 and "sip list peers" |
| 1:13AM |
0 |
AMAFlags always"Documentation" (or 3 inastcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) |
| |
| Friday January 12 2007 |
| Time | Replies | Subject |
|---|
| 7:23PM |
2 |
Dropped calls |
| 5:51PM |
1 |
SIP phones at multiple locations |
| 2:47PM |
1 |
cepstral voice still nags after registration |
| 2:21PM |
0 |
GXP-2000 Firmware |
| 1:53PM |
1 |
phones can make outgoing calls but no incoming |
| 1:52PM |
4 |
Nat Question |
| 1:43PM |
0 |
Musiconhold and dial |
| 11:38AM |
3 |
5v capable motherboards |
| 11:19AM |
2 |
two level administration tool for Asterisk |
| 11:15AM |
1 |
TDM2400p bad sound quality |
| 11:11AM |
0 |
R: asterisk-users Digest, Vol 30, Issue 50 |
| 9:39AM |
0 |
realtime queues |
| 9:31AM |
4 |
Voxbone Question |
| 9:27AM |
3 |
One of my incomming lines is busy yet there is no indication in FOP. |
| 9:23AM |
1 |
FW: Redundancy |
| 9:23AM |
0 |
Provisioning |
| 9:23AM |
1 |
External Ringers for Cisco Phones |
| 8:24AM |
1 |
How to detect which end hung up the call |
| 8:04AM |
1 |
Identifying Queue on Cisco 7960 |
| 8:03AM |
1 |
SPA 3000 won't relay DTMF to doorphone |
| 7:24AM |
2 |
SLA |
| 7:01AM |
1 |
realtime extensions, labels |
| 3:12AM |
4 |
FW: Get dialed numbers in AGI |
| 1:29AM |
1 |
Not Registering Port with VSP. |
| 12:47AM |
1 |
boot up problem |
| 12:06AM |
1 |
Snom Record / Voice Recorder Button |
| |
| Thursday January 11 2007 |
| Time | Replies | Subject |
|---|
| 6:28PM |
2 |
Question Regarding Visual Park Functionality - Hardware/Software |
| 6:23PM |
4 |
Echo... |
| 5:00PM |
1 |
Problem with Zyxel P-2000W v2 not receiving calls |
| 4:20PM |
4 |
DND - message |
| 3:40PM |
0 |
Trying to understand mISDN kernel buffer messages... |
| 3:33PM |
2 |
Queues without music on hold ? |
| 3:08PM |
2 |
Asterisk Compilation and Installation |
| 3:06PM |
2 |
Symbolic Link |
| 2:52PM |
1 |
Sipgate displayes on web interface status Offline |
| 2:36PM |
0 |
Re: Digium TE407P vs. Sangoma A104d |
| 2:08PM |
2 |
Voicemail IMAP |
| 2:05PM |
0 |
Getting two Asterisk machines to talk to each other |
| 1:46PM |
1 |
Read Voicmail Boxes |
| 1:41PM |
0 |
"Dropping Incompatible Voice Frame" |
| 1:28PM |
4 |
Parked calls and the # key |
| 11:57AM |
2 |
Restrict International Calls |
| 11:07AM |
1 |
Queues Service Level |
| 10:17AM |
1 |
authentication issue! |
| 10:12AM |
0 |
Queue log analyser / report |
| 9:55AM |
2 |
Native music on hold not playing on incoming calls |
| 9:41AM |
6 |
Suggestion for a new asterisk setup. |
| 9:23AM |
0 |
What would make Asterisk Ignore INVITES? |
| 9:12AM |
1 |
realtime sipusers and rtcachefriends... big headache!! |
| 8:53AM |
1 |
Asterisk Manager Interface: Auto-answer of 'Originate' command |
| 7:54AM |
1 |
play music while continue executing dial plan |
| 7:46AM |
0 |
Need help getting anterisk application to load |
| 6:52AM |
4 |
"real life" example of SLA definition |
| 4:37AM |
1 |
Has been working for 9 Months - Very Very Strange I cannot dial specific extensions from my dialplan - NOT A CONTEXT PROBLEM!! |
| 4:10AM |
1 |
DTMF signaling after Dial |
| 3:15AM |
1 |
Stuck somewhere - INVITEs ignored?! |
| 3:12AM |
1 |
Problems with agent dynamic login |
| 2:20AM |
2 |
calls to SPA942 disconnect after 15 seconds (chan_sip.c set_destination: can't find address) |
| |
| Wednesday January 10 2007 |
| Time | Replies | Subject |
|---|
| 6:06PM |
0 |
Extensions in a macro |
| 5:52PM |
1 |
Priority + 101 exit |
| 5:20PM |
0 |
Festival Problems |
| 2:28PM |
1 |
Round Robin Queue |
| 2:22PM |
0 |
generating SIP errors |
| 12:53PM |
1 |
Possibility to catch DTMF when 2 users are in a conversation |
| 12:17PM |
0 |
SIP invite and sip.conf relationship? |
| 12:15PM |
2 |
Random dropped calls... |
| 10:41AM |
1 |
dundi ENCREJ |
| 10:34AM |
1 |
VIA EPIA DeadLock Issues |
| 10:05AM |
3 |
Proper use of the Local channel |
| 9:53AM |
0 |
app_system.c:105 system_exec_helper: Unable to execute '/sbin/zapscan.bin' |
| 9:39AM |
0 |
1.4 and zap bugs |
| 9:18AM |
1 |
Service Level Compliance |
| 9:10AM |
5 |
Directory too difficult? |
| 8:59AM |
2 |
Get dialed numbers in AGI |
| 8:57AM |
2 |
RTP directly |
| 8:46AM |
2 |
Send email notification |
| 8:37AM |
3 |
how to realize chief - secretary (or Manager - Assistant) setup with Asterisk? |
| 8:36AM |
1 |
Asterisk HA |
| 8:30AM |
0 |
SPA-3000 and Asterisk 1.4.0 |
| 7:01AM |
1 |
Zap 1.4 error line 0: Unable to open |
| 6:34AM |
0 |
Calls die when the answering party transfers |
| 6:02AM |
0 |
libpri Calling Line ID |
| 3:33AM |
1 |
Sip dynamic host question |
| 3:33AM |
0 |
DTMF on Snom |
| 3:10AM |
0 |
one way audio when forwarding from ser to asterisk |
| 2:52AM |
1 |
Which H323 module for asterisk |
| 2:06AM |
1 |
caller id not transferred to SIP device |
| 1:25AM |
1 |
Redundancy |
| 12:25AM |
0 |
cannot call out |
| |
| Tuesday January 9 2007 |
| Time | Replies | Subject |
|---|
| 11:41PM |
1 |
ztmonitor output while idle |
| 6:01PM |
8 |
Problem with zaptel drivers or card |
| 5:26PM |
2 |
Attatching VM via email for more than one user |
| 4:51PM |
3 |
Zap 1.4 error line 0: Unable to open master device '/dev/zap/ctl' |
| 3:28PM |
1 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday January 13th 2007 - 11:30am |
| 3:12PM |
4 |
Is there a low cost cell phone base station for asterisk ? |
| 2:55PM |
1 |
Asterisk 1.2.11 - ResponseTimeout being ignored |
| 12:22PM |
1 |
How to test VOIP quality? |
| 12:17PM |
0 |
Console\DSP |
| 11:57AM |
0 |
VOIP provider reliability |
| 11:27AM |
1 |
getting tones during conversation |
| 11:16AM |
1 |
Caller Id problem |
| 10:44AM |
1 |
ooh323c calls |
| 10:30AM |
0 |
Is there any Asterisk controllable thermostat? |
| 9:41AM |
1 |
Asterisk and Avaya IP Office |
| 9:04AM |
12 |
Asterisk build for Suse 10.1 |
| 8:52AM |
2 |
Fax through Sangoma A102 |
| 8:29AM |
0 |
Strange queue behaviour |
| 7:50AM |
8 |
Snom side car annoyance |
| 6:34AM |
0 |
Asterisk + 7910 + Skinny Reset |
| 2:21AM |
1 |
Asterisk and 3PCC |
| 2:04AM |
1 |
Problem with polycom video conference |
| 1:59AM |
0 |
Bad FCS & hangup |
| 12:27AM |
1 |
Record of all calls |
| |
| Monday January 8 2007 |
| Time | Replies | Subject |
|---|
| 9:34PM |
0 |
Allowing inbound VoIP Calls from VSP |
| 9:08PM |
1 |
No CDR from Outbound Call |
| 4:17PM |
1 |
Call Sound Volume Low : between extensions and over ZAP. |
| 2:44PM |
0 |
SIP rt load from db |
| 2:19PM |
0 |
snom 190 (etc.?) dialscript for * debugging and kaddressbook |
| 2:17PM |
4 |
delete=yes is not working |
| 11:32AM |
0 |
MixMonitor write issue |
| 11:22AM |
2 |
OT:spa942 provisioning |
| 11:13AM |
0 |
Strange error |
| 9:58AM |
2 |
G729 license counting |
| 9:47AM |
1 |
Realtime Voicemail Table Column Name Question |
| 9:17AM |
2 |
ARA extensions ordering |
| 8:59AM |
3 |
Adding 4000 Lines to asteriskdb via asterisk -rx ? |
| 8:09AM |
1 |
Block some number outgoing from joust oneextention |
| 6:21AM |
0 |
IAX call path optimization with more than 3 legs |
| 5:24AM |
2 |
SV: Manage 'full' log file |
| 5:02AM |
2 |
Manage 'full' log file |
| 4:07AM |
1 |
MFC/R2 problems |
| 12:58AM |
0 |
Goto not jumping to current context |
| 12:27AM |
3 |
jitterbuffer on sip.conf |
| |
| Sunday January 7 2007 |
| Time | Replies | Subject |
|---|
| 11:28PM |
1 |
Interrupt rates and voip traffic |
| 10:07PM |
1 |
Problems with park |
| 10:06PM |
1 |
AMAFlags always"Documentation" (or 3 in astcdr mysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) |
| 9:29PM |
1 |
snom 360 auto answer |
| 8:27PM |
5 |
Some queries on g729 license. |
| 8:03PM |
0 |
How to transfer Voicemail messages between 2 Asterisk servers |
| 6:32PM |
0 |
Re: asterisk-users Digest, Vol 30, Issue 7 |
| 5:12PM |
1 |
Hanging up a 3-way conference when middle user hangs up |
| 6:14AM |
0 |
LUSYN patches |
| 12:38AM |
1 |
Scalable IVR with asterisk |
| 12:19AM |
4 |
How to get dial tone back |
| 12:02AM |
2 |
"Reserved" extensions? |
| |
| Saturday January 6 2007 |
| Time | Replies | Subject |
|---|
| 9:46PM |
2 |
Question about AGI and variable storage |
| 8:19PM |
1 |
Parking a call a second time using #700.. |
| 7:54PM |
0 |
Shared Line Appearances in 1.4 |
| 4:08PM |
1 |
SIP/RTP Nat problem, can't solute it. |
| 2:05AM |
0 |
SIP Reinvites |
| 1:50AM |
0 |
Hint and call-limit issue |
| 1:36AM |
1 |
SIP trunk to a Boscom/Claro/IP Gear Robocom |
| 1:13AM |
1 |
Handling SIP 482 condition |
| 12:44AM |
2 |
Secure a Asterisk Server ? |
| |
| Friday January 5 2007 |
| Time | Replies | Subject |
|---|
| 11:04PM |
0 |
Asterisk is used in U.S. prisons? |
| 10:13PM |
0 |
asterisk 1.4.0 didn't compile chan_zap.so |
| 4:16PM |
1 |
.call files no longer generating CDR files |
| 4:13PM |
1 |
Call waiting notification |
| 3:42PM |
1 |
Voicemail personalised greetings using DB/IMAPbackend? |
| 3:28PM |
1 |
DiD for less then $4 |
| 3:23PM |
0 |
Random "unknown" codec format IAX calls |
| 3:17PM |
2 |
Voicemail personalised greetings using DB/IMAP backend? |
| 2:05PM |
1 |
Multiple users and a single extension |
| 1:54PM |
0 |
Has anybody voipstunt working? |
| 1:15PM |
1 |
asterisk 1.4 debian packages |
| 12:22PM |
2 |
SIP/TCP? |
| 11:13AM |
1 |
asterisk (FreePBX) and queues |
| 10:35AM |
1 |
ASterisk OOH323c |
| 10:31AM |
3 |
how to register nokia with Asterisk |
| 10:02AM |
4 |
how to transfer calls when analog phone has no transfer button |
| 9:44AM |
1 |
addons 1.4 and cdr_addon_mysql not installed ! |
| 9:06AM |
0 |
idle SIP channels problem |
| 7:19AM |
1 |
radius |
| 5:57AM |
1 |
integrating with Asterisk and OpenSER for Voicemail |
| 5:47AM |
1 |
Asterisk and IM |
| 5:37AM |
1 |
fax transmission |
| 5:25AM |
0 |
Invalid DivertingLegInformation2 component received 0x38 |
| 4:40AM |
0 |
How to build 1.4 with res_crypto.so |
| 1:48AM |
1 |
Which g729 module for HP DL 360 G3 (Xeon CPU's)? |
| 1:17AM |
2 |
chan_zap.c: Failed to read gains: Invalidargument |
| 12:47AM |
2 |
chan_zap.c: Failed to read gains: Invalid argument |
| |
| Thursday January 4 2007 |
| Time | Replies | Subject |
|---|
| 11:51PM |
4 |
Which is GUI to edit Asterisk IVR logic |
| 10:33PM |
4 |
POE draw on Aastra 480i |
| 8:33PM |
0 |
3-way calling MGCP capture |
| 7:23PM |
1 |
IAX vs SIP trunks between Asterisk boxes |
| 7:01PM |
1 |
MusicOnHold Files |
| 6:59PM |
0 |
DISA Ring Back |
| 6:48PM |
2 |
HowTO configure voice T1 |
| 4:53PM |
2 |
Dimensioning a 50 sip phone installation |
| 3:30PM |
0 |
How to routing call to Quintum. |
| 1:00PM |
0 |
proxy howto |
| 12:18PM |
0 |
Re: Alert: Steering Committee Reminder and Agenda |
| 12:08PM |
1 |
How big a pipe can IAX2 go? |
| 11:42AM |
1 |
Trouble compiling asterisk 1.2.14 |
| 11:13AM |
1 |
asterisk sip peer/user matching methodsforauthentication backwards? |
| 10:57AM |
7 |
Best inexpensive home office router for VoIP (QoS with maybe PoE) |
| 10:43AM |
0 |
SIP peer lookup problems |
| 10:26AM |
1 |
#include not working in 1.4 |
| 10:22AM |
0 |
asterisk sip peer/user matching methods forauthentication backwards? |
| 10:17AM |
0 |
Asterisk 1.4.0 segfault |
| 10:08AM |
1 |
Convert a file from WAV to WAV49 or GSM for Asterisk |
| 10:04AM |
0 |
Sangoma A102 w/ EC module gets intermittent echo/audio artifacts <--followup and resolution |
| 9:34AM |
2 |
[Fwd: PRI Problems] |
| 9:27AM |
0 |
System() and Trysystem() in extensions.conf => get the result ? |
| 9:26AM |
0 |
PRI Problems |
| 9:06AM |
0 |
mISDN crypto? |
| 8:46AM |
1 |
Realtime voicemail passwords |
| 8:26AM |
0 |
Create a group of SIP acoount for outgoing calls ? |
| 6:26AM |
2 |
postgres and asterisk |
| 6:18AM |
4 |
Digium Wildcard B410P |
| 5:02AM |
1 |
Maybe a NAT problem |
| 3:23AM |
1 |
Hi reg. asterisk Compilation |
| 3:14AM |
1 |
bypass menu for certain numbers? |
| 2:12AM |
2 |
Cisco AS5300 |
| 1:58AM |
0 |
Required freelancer for installing hylafax on Asterisk Box |
| |
| Wednesday January 3 2007 |
| Time | Replies | Subject |
|---|
| 10:50PM |
0 |
asterisk sip peer/user matching methods for authentication backwards? |
| 9:15PM |
3 |
caller id ring tones for Asterisk Phone |
| 6:38PM |
0 |
Re: asterisk-users Digest, Vol 30, Issue 4 |
| 6:07PM |
3 |
ztdummy on 1.6 |
| 5:41PM |
0 |
1.4 segfaulting when manager client is connected |
| 5:17PM |
0 |
[Announce] Web-MeetMe 3.0.0 RE-released |
| 5:13PM |
0 |
v140 ./configure not finding installed ssl |
| 3:51PM |
6 |
Any quiet 24 port POE switches out there? |
| 3:26PM |
3 |
Asterisk Core Dump in app_queue - Anyone seen? |
| 3:22PM |
1 |
Detect IP path before calling |
| 2:29PM |
1 |
Gentoo ebuild for 1.4? |
| 2:27PM |
0 |
ARI help |
| 1:31PM |
0 |
have a phone number from stanaphone and a workingtrixbox, h |
| 1:22PM |
4 |
over 200 queues, anyone? |
| 12:39PM |
1 |
have a phone number from stanaphone and a working trixbox, how do I connect them? |
| 12:24PM |
2 |
Error on answer a SIP 401 message |
| 12:23PM |
0 |
Cisco 79x1 Auto-Answer |
| 11:35AM |
0 |
Park and Page |
| 11:32AM |
9 |
[Announce] Web-MeetMe 3.0.0 released |
| 11:16AM |
1 |
Is chan_zap.so loaded? |
| 11:05AM |
4 |
API: how to bridge originated call? |
| 10:48AM |
5 |
Polycom Power Specs |
| 10:40AM |
2 |
SIP Dial out timeout |
| 10:26AM |
2 |
answer machine detection |
| 10:11AM |
0 |
Sangoma A102 w/ EC module gets intermittent echo/audio artifacts |
| 9:29AM |
0 |
[BULK] Fonebridge2 |
| 9:25AM |
0 |
Sangoma A102 w/ EC module gets intermittent echo /audio artifacts <---More information |
| 9:05AM |
3 |
Voicemail to email |
| 8:33AM |
1 |
Fonebridge2 |
| 8:32AM |
2 |
Sangoma A102 w/ EC module gets intermittent echo /audio artifacts |
| 8:04AM |
0 |
MeetMe() not recording calls |
| 6:43AM |
4 |
Sangoma Remora A202 |
| 6:40AM |
0 |
native music on hold distortion between files |
| 6:11AM |
3 |
voice fax modem and asterisk |
| 5:20AM |
0 |
ISA server Issue (Maybe off topic) |
| 4:17AM |
0 |
Dubai Caller ID |
| 4:00AM |
7 |
SNOM loses server registration |
| |
| Tuesday January 2 2007 |
| Time | Replies | Subject |
|---|
| 8:15PM |
3 |
connecting asterisk (trixbox) to traditional phone lines? |
| 5:22PM |
1 |
Double quotes in CDRUserField? |
| 5:12PM |
1 |
extension problems |
| 5:01PM |
4 |
OnHook Call Announcement... |
| 3:17PM |
1 |
OT: Admin manual for Linksys Sipura SPA-2102 |
| 2:57PM |
2 |
queues - limiting ringing calls to queue members |
| 1:17PM |
5 |
Call connected, cannot hear or speak - $20 for fix |
| 12:45PM |
3 |
yet another faxing issue (outbound only, via ATA) |
| 12:18PM |
0 |
SpanDSP and Asterisk 1.4 |
| 10:20AM |
2 |
How to show a debugging remark in a sip or extensions context? |
| 9:00AM |
2 |
802.1x support in wired sip hardphones ? |
| 8:55AM |
0 |
[asterisk-biz] Slightly updated UK English voice prompts |
| 6:48AM |
0 |
Save SIP DEBUG output to a file |
| 6:20AM |
9 |
Best Hardware for Asterisk Server? |
| 6:20AM |
0 |
Avoiding deadlock-line drop problem |
| 2:57AM |
4 |
asterisk and mysql |
| 2:52AM |
1 |
chan_oh323 early media |
| |
| Monday January 1 2007 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
Problem with centos 4.4 and jabber/gtalk (really iksemel) |
| 11:54AM |
1 |
Help needed with Polycom dialplan pattern matching |
| 7:36AM |
0 |
Thomson ST2020 and voicemail |