| Wednesday September 30 2009 |
| Time | Replies | Subject |
|---|
| 11:57PM |
3 |
Choose IAX or SIP trunking? |
| 11:40PM |
0 |
Calls at 2 different locations |
| 8:12PM |
2 |
E1/T1 Tapping call recording in Asterisk - Testing needed |
| 7:08PM |
1 |
SIPAddHeader into the SDP? |
| 6:52PM |
1 |
EXTENSION_STATE Asterisk 1.6 |
| 5:34PM |
0 |
PBXNSIP Registration Issue |
| 3:08PM |
6 |
question on pri intense debug |
| 1:07PM |
1 |
How to finish a Meetme |
| 10:48AM |
1 |
put some IVR into a queue after the call queuing |
| |
| Tuesday September 29 2009 |
| Time | Replies | Subject |
|---|
| 9:22PM |
0 |
Dumb Question - Dialing internal and external |
| 8:11PM |
2 |
kill sip user |
| 8:07PM |
3 |
UpdateConfig |
| 6:19PM |
1 |
Asterisk on DD-WRT : modules.conf not found |
| 6:16PM |
2 |
dialing 0 in directory() |
| 3:40PM |
0 |
Static on the line randomly |
| 3:31PM |
1 |
Retrieve Call setup - QoS |
| 3:04PM |
3 |
chanspy and DISA |
| 3:02PM |
0 |
Wrong hint, ringing when idle. after hangup. |
| 2:01PM |
3 |
LDAP integration |
| 11:57AM |
1 |
Native bridging analog phones trouble DAHDI channels. |
| 11:20AM |
1 |
Fax and dial-up connection issues |
| 9:37AM |
1 |
Who am xxx talking to.agi |
| 8:35AM |
2 |
play audio file within an active call |
| |
| Monday September 28 2009 |
| Time | Replies | Subject |
|---|
| 8:24PM |
2 |
DAHDI channel congested busy |
| 7:07PM |
1 |
How to get "Call-ID" SIP header outside "chan_sip" scope ... |
| 5:14PM |
0 |
strange cisco nat issue |
| 5:10PM |
2 |
Disable/enable CDR in dialplan |
| 3:54PM |
0 |
Asterisk complaning about no such host -- never asked to contact the host it complains about |
| 3:41PM |
1 |
Voicemail - remove option to save in different folders |
| 2:21PM |
3 |
GoTo IF |
| 1:20PM |
1 |
ding Dialled number down a sip channel to a PBX |
| 11:56AM |
0 |
simulate calls for testing sripts |
| 11:32AM |
1 |
Firefox Plugin for Sip Click2Call |
| 10:39AM |
0 |
AGI script |
| 7:05AM |
1 |
TE121P Blue Alarm/Recovering |
| |
| Sunday September 27 2009 |
| Time | Replies | Subject |
|---|
| 10:38PM |
1 |
DAHDI Question/Choppy Sound |
| 10:31PM |
1 |
MeetMe Hints |
| 9:05PM |
2 |
DAHDI congestion problem |
| 9:02PM |
0 |
Issue with incoming caller-ID to NEC SV8300 with QSIG |
| 9:01PM |
0 |
channel.c:780 channel_find_locked: Avoided deadlock |
| 7:26PM |
0 |
Is channel local what I need? |
| 4:17PM |
3 |
Problems with Digium TDM400 card |
| 10:38AM |
0 |
FW: New in asterisk |
| 9:31AM |
1 |
callfile to auto-answering extension |
| 6:00AM |
1 |
Peers Listed in "sip show channels" |
| 5:47AM |
1 |
New thread - SIP over VPN |
| 1:05AM |
1 |
digium fax: failed to queue document |
| |
| Saturday September 26 2009 |
| Time | Replies | Subject |
|---|
| 11:07PM |
1 |
Know for how long an agent is talking? |
| 5:59PM |
1 |
Where are phone registrations kept? |
| 2:20PM |
3 |
VOIP solutions |
| 1:35PM |
0 |
Voiced E-mail |
| 12:06PM |
0 |
VoiceMail.conf reading variables |
| 8:41AM |
6 |
Inquiry:Asterisk server remote access |
| 5:44AM |
8 |
Inquiry:How to convert *.wav files ? |
| |
| Friday September 25 2009 |
| Time | Replies | Subject |
|---|
| 9:58PM |
1 |
"multiple contexts for multiple locations" |
| 9:05PM |
1 |
OT - In which countries are ISDN subaddresses used ? |
| 3:30PM |
3 |
disable dtmf on SIP peer |
| 3:19PM |
4 |
DAHDI disconnect supervision timing |
| 3:15PM |
2 |
How to remove peers from channels |
| 10:01AM |
1 |
Choppy sound, SIP calls within LAN |
| 10:00AM |
1 |
Asterisk Manager Problem |
| 7:02AM |
0 |
ignore flash hook |
| 6:14AM |
0 |
(solved) CPU Spikes in asterisk connected via IAX trunk |
| |
| Thursday September 24 2009 |
| Time | Replies | Subject |
|---|
| 5:57PM |
0 |
Asterisk and VoIP Users Friday Meeting |
| 5:41PM |
4 |
Polycom push application for microbrowser |
| 4:16PM |
1 |
rtp.conf dtmftimeout |
| 12:56PM |
1 |
Asterisk 1.6 Transfer issue[Edited] |
| 12:54PM |
0 |
Asterisk 1.6 Transfer issue |
| 12:47PM |
6 |
Connecting home intercom to Asterisk? |
| 8:42AM |
2 |
Digium transcoding card |
| 7:14AM |
1 |
CDRs on call forward |
| |
| Wednesday September 23 2009 |
| Time | Replies | Subject |
|---|
| 10:27PM |
1 |
SFA - No channel cause 66 |
| 9:29PM |
0 |
DYNAMIC FEATURES, AEL2 - how to use Goto, Gosub or Macro ? |
| 7:31PM |
1 |
Parking - How to transfer the other party to a given slot |
| 5:12PM |
0 |
Testers Wanted for IMAP Voicemail patch |
| 4:30PM |
0 |
About bug 13115 |
| 1:34PM |
3 |
SIP/WiFi handsets? |
| 1:08PM |
0 |
Asterisk success |
| 10:49AM |
4 |
Error When Using Postgresql Schema With Realtime Sip |
| 10:18AM |
0 |
unknown error ON DAHDI please help me... |
| 8:14AM |
0 |
Inquiry:Which codec to get higher download rate on dialup connection |
| 5:58AM |
0 |
SIPP + Duration |
| 5:07AM |
3 |
Bringing people into a conference |
| 4:19AM |
4 |
International Numbering plan ? |
| 3:32AM |
3 |
Simple dialplan issue |
| 12:32AM |
1 |
1.6.0.5: I need a really simple analog SendFax dialplan |
| |
| Tuesday September 22 2009 |
| Time | Replies | Subject |
|---|
| 6:29PM |
0 |
Asterisk with Cisco 5300/E1/DSP |
| 5:12PM |
5 |
New Xorcom FXS USB Bank is not loading firmware |
| 4:23PM |
1 |
not hearing audio on console/dsp |
| 3:25PM |
0 |
How to associate a custom script to "core stop now" |
| 2:53PM |
0 |
AstManProxy - No pid file created when run |
| 2:12PM |
1 |
setting up a IP based voip carrier account |
| 12:56PM |
4 |
Asterisk on a Beagleboard? |
| 10:13AM |
2 |
SIPP question |
| 8:43AM |
2 |
Problem with dialplan -> gotoif ? |
| 8:08AM |
1 |
Call deflection on Asterisk 1.6.1.6 |
| 6:27AM |
3 |
RTPAUDIOQOS |
| 2:52AM |
0 |
Which oslec.h should will work? |
| 1:26AM |
1 |
digium fax: can't indicate condition 19? |
| |
| Monday September 21 2009 |
| Time | Replies | Subject |
|---|
| 10:20PM |
0 |
Enterprise users going to VoiceCon? |
| 8:37PM |
2 |
MySQL cmd |
| 7:18PM |
1 |
Voicemail to email transcribed |
| 6:46PM |
1 |
Voicemail Crash - ODBC Realtime |
| 3:54PM |
2 |
Atcom AG188N as FXO? |
| 9:23AM |
0 |
Asterisk 1.6 dynamic agents |
| 8:55AM |
0 |
Newbie: How to detect an "*" in Read()? |
| 8:44AM |
1 |
GSM cellphone as cheap gateway? |
| |
| Sunday September 20 2009 |
| Time | Replies | Subject |
|---|
| 4:51PM |
0 |
Recipe: Automatically Create Dial-able Extensions For Skype Callers |
| 4:27PM |
1 |
A in ACL of sip show peers. |
| 2:58PM |
2 |
different verbose level for full log than to console? |
| 1:20PM |
1 |
DAHDI installation warning |
| 5:42AM |
1 |
Experience with Sangoma's USBfxo |
| 1:40AM |
0 |
Asterisk + Nonoh |
| 12:07AM |
0 |
Stop / Resume in Dialplan / AMI |
| |
| Saturday September 19 2009 |
| Time | Replies | Subject |
|---|
| 1:22PM |
0 |
"Switchboard"/"advanced answering machine setup" with Asterisk? |
| 10:40AM |
0 |
Sunday 20th Global Asterisk Mtg via VOIP - BerkeleyTIP - for forwarding |
| 7:33AM |
3 |
Sangoma A200 and battery removal detection ??!!! |
| 7:11AM |
0 |
Echo cancellation on DAHDI |
| 6:07AM |
1 |
DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657 |
| 6:07AM |
0 |
IPKall using iax |
| 4:43AM |
1 |
"Channels got stuck in asterisk 1.4.18.1" |
| |
| Friday September 18 2009 |
| Time | Replies | Subject |
|---|
| 11:45PM |
1 |
digium fax: is this even close to working? |
| 6:33PM |
1 |
No more room in scheduler |
| 6:26PM |
2 |
IAX2 order |
| 5:09PM |
3 |
DUNDi + SIP Realtime |
| 3:43PM |
0 |
Queue Call Disconnection |
| 2:28PM |
4 |
console color |
| 2:23PM |
0 |
Blind Transfer Won't Hangup |
| 2:16PM |
0 |
DAHDI TDM440E still has echo on bridged connections |
| 2:15PM |
0 |
calls drop during attended transfer with PRI line |
| 12:45PM |
1 |
Audio Files |
| 6:46AM |
2 |
Help sending call to local server |
| 2:29AM |
1 |
404 for SUBSCRIBE |
| 1:03AM |
1 |
DAHDI Caller ID problem |
| |
| Thursday September 17 2009 |
| Time | Replies | Subject |
|---|
| 11:27PM |
1 |
Custom auto-install asterisk using ks.cfg |
| 10:36PM |
0 |
Anyone having issues with 1.6.1.6 res_snmp? |
| 10:19PM |
1 |
CDR Records for MeetMe |
| 7:58PM |
2 |
DeadAgi |
| 5:42PM |
0 |
VoIP Users Conference Friday: Andy Abramson of VoIP Watch |
| 3:31PM |
1 |
I'm not getting the ability to leave a voicemail-message |
| 2:20PM |
1 |
Freepbx database |
| 2:13PM |
0 |
SIP HEADER FROM: without CALLERID(name):: PART DEUX |
| 1:49PM |
2 |
Voice Playback cutting first word or so of audio file |
| 1:12PM |
1 |
Changing or Adding a Line to the Extensions.conf in Asterisk |
| 6:04AM |
1 |
ZAP and line disconnection detection |
| 5:16AM |
2 |
limit concurrent calls on trunk supporting multiple DID |
| 1:50AM |
0 |
web-meetme cbEnd.php not running - error |
| |
| Wednesday September 16 2009 |
| Time | Replies | Subject |
|---|
| 11:51PM |
1 |
H323 RTP Transmission error of packet |
| 10:18PM |
3 |
call-limit on dahdi channel |
| 9:58PM |
1 |
Connected Line ID for Asterisk 1.4 |
| 9:17PM |
3 |
Music on Hold |
| 7:34PM |
1 |
res-crypto dependencies |
| 7:08PM |
1 |
Meetme feature |
| 5:05PM |
0 |
asterisk-users Digest, Vol 62, Issue 44 |
| 4:35PM |
5 |
custom voicemail e-mail |
| 3:31PM |
1 |
ACR Anonymous Call Rejection |
| 2:34PM |
3 |
[asterisk-dev] MeetMe in Macro |
| 1:02PM |
2 |
IVR seleCtion |
| 11:46AM |
4 |
G729 |
| 3:41AM |
2 |
Reproducible crash - known bug? |
| |
| Tuesday September 15 2009 |
| Time | Replies | Subject |
|---|
| 10:44PM |
1 |
How to list ongoing calls from dialplan |
| 9:44PM |
1 |
Detecting Transfer |
| 6:41PM |
0 |
1.6.2.0-rc1 intermittent voicemail problem ? |
| 5:35PM |
0 |
Call forwarding, callerID, and e911 |
| 4:20PM |
0 |
FW: Blind transfer on Queue-CDR |
| 4:17PM |
1 |
DAHDI hangup detection |
| 11:54AM |
0 |
Sound quality issue |
| 10:31AM |
0 |
(no subject) |
| 7:54AM |
3 |
Which is best provider for G.729 |
| 7:49AM |
0 |
Dynamically Move caller to different dial plan |
| 3:31AM |
3 |
dCAP Exam |
| 3:15AM |
1 |
Simple Time of Day Branching problem |
| |
| Monday September 14 2009 |
| Time | Replies | Subject |
|---|
| 11:22PM |
1 |
The "o" dial option |
| 9:54PM |
1 |
Aastra - Alert-Info : how to stop auto-answer on call second leg ? |
| 7:49PM |
0 |
DAHDI Dial 9 Receiving Setup Acknowledge |
| 5:02PM |
0 |
AstriCon 2009: 30 days, 5 reasons & discount code |
| 10:22AM |
0 |
Something with Dahdi and reversal event... |
| 9:48AM |
3 |
G.729 for Asterisk |
| 8:05AM |
0 |
user=phone |
| 3:35AM |
0 |
Odd sip error |
| |
| Sunday September 13 2009 |
| Time | Replies | Subject |
|---|
| 4:24PM |
0 |
Blind transfer on Queue-CDR |
| 3:45PM |
3 |
custom ip phone interface |
| |
| Saturday September 12 2009 |
| Time | Replies | Subject |
|---|
| 3:31PM |
1 |
Queue_logs |
| 3:11PM |
1 |
OT: Question about Wifi sniffing on network |
| 1:54PM |
1 |
Simulscribe/Ditech |
| 8:08AM |
1 |
E65 fails registration, soft phone works |
| 7:03AM |
2 |
zaptel kernel configuration error on vmware |
| |
| Friday September 11 2009 |
| Time | Replies | Subject |
|---|
| 10:29PM |
2 |
No 64 bit binary for Fax for Asterisk |
| 7:22PM |
0 |
www.chan_mobile.org seems dead ? |
| 7:17PM |
1 |
Asterisk & Faxing |
| 5:47PM |
0 |
Aastra 51i and PAP2T behind NAT |
| 4:58PM |
1 |
Hiding voiemailbox/entry from directory |
| 4:14PM |
2 |
Parser for Asterisk Queue Logs |
| 4:05PM |
1 |
MySql and custom CDR |
| 2:36PM |
1 |
Voicemail by email with HTML |
| 12:48PM |
0 |
Asterisk 1.6.1.6 Crash when accessing Directory |
| 11:55AM |
0 |
183 early media |
| 6:58AM |
0 |
asterisk addons don't compile using non standart prefix |
| 6:31AM |
1 |
DIAL IAX2 vs. SIP |
| |
| Thursday September 10 2009 |
| Time | Replies | Subject |
|---|
| 8:16PM |
2 |
ASR & ACD |
| 7:25PM |
1 |
Operation of ATAs in a call shop type set-up |
| 7:11PM |
2 |
CDR Reporting |
| 6:45PM |
1 |
googlevoice player |
| 6:13PM |
1 |
Friday 11th: Aswath Rao: "Trapezoidal VoIP is Evil" on VoIP Users Conference at Noon EDT |
| 5:21PM |
4 |
Looking for a way to show caller id information on the desktop |
| 2:00PM |
1 |
Help with dialparties.agi |
| 10:33AM |
1 |
RTPAUDIOQOS On DAHDI is it possible |
| 5:49AM |
2 |
Asterisk With Broadvoice |
| 4:35AM |
1 |
g723 to wav conversion |
| 3:04AM |
0 |
A Strange Exception/Notice |
| 2:35AM |
1 |
SPA2102 with Public IP no NAT getting one way audio between Asterisk Phones. |
| 2:32AM |
2 |
How to catch isdn progress message |
| 2:22AM |
2 |
Duplicate DTMF |
| |
| Wednesday September 9 2009 |
| Time | Replies | Subject |
|---|
| 9:59PM |
3 |
Call Aanalyzer |
| 9:25PM |
2 |
streaming meetme conference |
| 7:34PM |
0 |
Custom CDR Help |
| 6:12PM |
1 |
Blind transfers security |
| 6:11PM |
2 |
Call getting stucked !! |
| 4:31PM |
6 |
how take over an ongoing conversation? |
| 4:10PM |
1 |
MySQL question |
| 3:35PM |
0 |
Hung Lines on GXP2000s? |
| 3:25PM |
0 |
Voice Broadcast Software |
| 3:18PM |
0 |
Query result is array of elements, how to iterate over it ?? |
| 2:33PM |
1 |
CLI file convert from alaw to g729 with TC400B transcoding card results to an empty file |
| 2:19PM |
0 |
Call Parking - BLF - AEL2 |
| 2:07PM |
1 |
Dial multiple extensions and know who picks up call |
| 1:33PM |
1 |
SIP reply CALL-ID from ITSP has internal address in host part |
| 12:55PM |
1 |
UNIQUEID not the same in Dialplan as passed to AGI |
| 12:42PM |
2 |
asterisk and CCM cisco call manager |
| 6:47AM |
2 |
All the four lights blinking |
| 4:34AM |
1 |
RESET CDR |
| |
| Tuesday September 8 2009 |
| Time | Replies | Subject |
|---|
| 10:50PM |
1 |
Should digium build a 2FXO / 2FXS 4-port daughter board? |
| 9:51PM |
2 |
Fax For Asterisk and SendFax question |
| 9:33PM |
1 |
Function to query ASTDB families |
| 6:59PM |
1 |
SIP Error |
| 4:27PM |
1 |
Caller ID from POTS lines |
| 4:24PM |
2 |
Realtime static with Asterisk 1.6.1.6 |
| 3:34PM |
2 |
1.2 AGI Deadlock |
| 2:23PM |
1 |
Asterisk remote calls with low bandwith and high latency |
| 2:11PM |
2 |
Manage a E1 system |
| 1:40PM |
1 |
Strange extension state changes in 1.6.0.15 |
| 1:28PM |
0 |
hang up problem while calling |
| 10:14AM |
0 |
Intermittent metallic voice SIP->ISDN ISDN<-SIP |
| 9:22AM |
2 |
CallerID app for Symbian? |
| 7:39AM |
3 |
Asterisk CLI commands not running !!!!! |
| 7:19AM |
0 |
asterisk and link spa942 provisioning |
| 6:56AM |
0 |
Shared Call Appearance - Polycom Phones |
| 5:50AM |
1 |
confBridge in Asterisk 1.6.2.0-rc1 doesn't stable |
| 5:29AM |
1 |
Inquiry:Asterisk sound files |
| |
| Monday September 7 2009 |
| Time | Replies | Subject |
|---|
| 10:16PM |
1 |
Is not yet available ODBC support for queue_log in asterisk 1.6? |
| 9:27PM |
1 |
Older Aastra phones and Asterisk 1.6 |
| 6:40PM |
2 |
features.conf : feature map ==> getting feature to work |
| 5:35PM |
2 |
All hints say Hold |
| 5:12PM |
2 |
Aastra phones and Asterisk 1.6.0.14 |
| 5:08PM |
1 |
invalid extension |
| 2:36PM |
2 |
Echo and Playtones not working on SIP after upgrade |
| 2:15PM |
0 |
automatic calls |
| 1:20PM |
1 |
dahdi/DTMF problem |
| 11:48AM |
0 |
Feature request: Meetme and invisible users |
| 10:55AM |
3 |
Help setting IAX variables. |
| 10:14AM |
5 |
TE420P configuration |
| 9:20AM |
0 |
Freepbx database followme disable/enable value |
| 7:45AM |
0 |
Record conversations and place soundfile in user-directory |
| 7:00AM |
0 |
one touch recording not working lately in asterisk |
| 6:40AM |
2 |
The identifier parameter in Dial() command |
| 2:47AM |
3 |
Using asterisk as the recording server |
| |
| Sunday September 6 2009 |
| Time | Replies | Subject |
|---|
| 11:51PM |
0 |
Preserve userfield on CDR on attended transfer |
| 7:29PM |
1 |
1.6.2-RC1 question |
| 2:33PM |
4 |
Digium hardware support ? |
| 9:45AM |
1 |
running a asterisk -rx command in bash backgroun |
| 4:10AM |
1 |
Chanskype Support |
| |
| Saturday September 5 2009 |
| Time | Replies | Subject |
|---|
| 6:02PM |
1 |
Asterisk-1.6.2.0-rc1 and Instant Message sending |
| 5:22PM |
0 |
Remote attended transfer |
| 2:58AM |
2 |
Need some help/Suggestions for multiple invites from Asterisk |
| |
| Friday September 4 2009 |
| Time | Replies | Subject |
|---|
| 9:23PM |
3 |
T1 TE121 cable connected to a TN2464BP AVAYA card |
| 7:45PM |
0 |
Problem with NAT settings?: SIP reply CALL-ID from ITSP has internal address in host part |
| 7:00PM |
5 |
cisco call manager version 6.1.3 |
| 6:50PM |
1 |
SUN and PRI ? |
| 5:39PM |
1 |
1.4.26-2, DAHDI-2.2.0, B410P and BRI |
| 5:08PM |
2 |
requirecalltoken and Realtime |
| 4:37PM |
1 |
starfish - pbx |
| 3:52PM |
1 |
Asterisk PBX causes mysql to take more CPU time |
| 3:47PM |
0 |
[Fwd: AST-2009-006: IAX2 Call Number Resource Exhaustion] |
| 1:41PM |
0 |
RFC 3578 in Asterisk |
| 1:27PM |
0 |
Today @12 Noon EDT: Skype for Asterisk, Floor Show at Astricon |
| 12:02PM |
2 |
[ANNOUNCEMENT] Amatix Office 2.0 |
| 11:40AM |
1 |
Send 200 OK with SDP instead of 183 with SDP when ringing starts |
| 9:36AM |
1 |
Incremented UniqueId |
| 8:28AM |
1 |
Strange beep when using VoiceMailMain application |
| 6:34AM |
1 |
OT - log rotation [solved] |
| 1:03AM |
0 |
DTMF with duration = 0 |
| |
| Thursday September 3 2009 |
| Time | Replies | Subject |
|---|
| 11:05PM |
0 |
chan_mobile -- bluetooth |
| 10:47PM |
1 |
AST-2009-006: IAX2 Call Number Resource Exhaustion |
| 10:46PM |
0 |
Asterisk 1.2.35, 1.4.26.2, 1.6.0.15, and 1.6.1.6 Now Available |
| 10:24PM |
1 |
setvar=CDR(accountcode)=${EXTEN} in sip.conf ??? |
| 8:13PM |
1 |
Noises on Batphones |
| 7:35PM |
0 |
transcoder card |
| 3:24PM |
1 |
probleme with web-meetme.3.1.0 |
| 2:41PM |
1 |
Originate calls with AMI. |
| 12:56PM |
2 |
How to Disable CDR for callfile? |
| 10:39AM |
2 |
OT - log rotation |
| 9:37AM |
1 |
MeetMe unactive pin access |
| 9:30AM |
1 |
Recommendations about infrastructure to use with Asterisk |
| 9:24AM |
3 |
GTalk functionality Asterisk |
| 8:17AM |
0 |
sql error on trunk qualify....?? |
| 7:47AM |
1 |
G.722 problems with IAX |
| 7:01AM |
1 |
passing commands asterisk cli and getting output using PHP AGI |
| |
| Wednesday September 2 2009 |
| Time | Replies | Subject |
|---|
| 11:18PM |
1 |
Voipbuster not ringing, other SIP peers are ringing... |
| 9:31PM |
1 |
outbound calls not ringing still |
| 8:40PM |
1 |
Payload size of 30ms |
| 8:37PM |
2 |
DISA() fails to recognize dtmf where WaitExten() succeeds (DAHDI-PRI) |
| 8:33PM |
0 |
problem with agi script not getting variable |
| 8:17PM |
2 |
Does L(x:y:z) "Dial" option work on Asterisk version 1.4 ? |
| 8:12PM |
0 |
[UOL - Manutenões Desktop] Controlling call duration ... |
| 6:37PM |
0 |
New Languages: Call for contributions |
| 6:34PM |
4 |
More Echo |
| 5:19PM |
4 |
Allowing multiple callers to join a public speaking session...? |
| 4:32PM |
0 |
followme Script |
| 3:42PM |
1 |
Very simple callback application needed |
| 8:35AM |
3 |
internet connection lagged -> * lagged ... |
| 8:26AM |
0 |
Problem with Cisco 7911G and ABE 2.1.2C - randomly cannot DIAL |
| 8:14AM |
1 |
web meetme PHP undefined variable |
| 7:45AM |
1 |
Skype for Asterisk callfile question |
| 6:27AM |
2 |
Prevent Agent Login from a second extension |
| 3:22AM |
1 |
AMI Originate Commands executed in sequential Order problem |
| 1:53AM |
2 |
DAHDI selective install |
| 12:44AM |
2 |
Configuring Parallel SIP Trunks |
| |
| Tuesday September 1 2009 |
| Time | Replies | Subject |
|---|
| 10:41PM |
2 |
chan_dahdi.so fails to load : Inappropriate ioctl for device |
| 9:35PM |
0 |
mISDN NT mode config setting |
| 7:33PM |
0 |
MeetMe and dedicated conference room phone |
| 7:22PM |
1 |
espeak app for asterisk 1.6 |
| 2:14PM |
1 |
set language in asterisk-1.6.x |
| 1:16PM |
7 |
Dahdi configuraion / error |
| 9:28AM |
0 |
Congratulations to Kamailio - Infoworld Best of Open Source Awards |
| 7:48AM |
1 |
SIP and other phones other then local network |
| 6:17AM |
4 |
jitterbuffer for chan_sip on asterisk 1.2 |
| 6:14AM |
1 |
Inquiry:Problem with VoiceMail |
| 4:39AM |
4 |
Inquiry:Problem with Call Parking |
| 2:59AM |
1 |
Digium PRI cards for data usage? |
| 12:14AM |
2 |
1.6.1 + TDM840 FSK MWI problem |