Friday July 31 2009 |
Time | Replies | Subject |
6:48PM |
1 |
Ignoring MOH directory and using default |
3:34PM |
1 |
Faxing over Carrier SIP trunk/g711 ? |
3:11PM |
0 |
Friday July 31st at 12 Noon EDT: Dave Nelsen, Skype for Asterisk beta opens, Gizmo Voice + Google Voice = free SIP calls |
2:44PM |
0 |
Friday July 31 @ 12 Noon EDT: Talkshoe former CEO Dave Nelsen, Skype for Asterisk open beta, Gizmo Voice+Google Voice |
12:21PM |
1 |
asterisk 1.6 call forwarding |
12:19PM |
1 |
timeout for trunk in failover |
11:00AM |
1 |
DAHDI - analogue, not seeing ringing (UK) |
10:38AM |
2 |
Voicemail feature: enable or disable the ability to leave a message |
7:22AM |
4 |
BT IP Exchange interconnect |
|
Thursday July 30 2009 |
Time | Replies | Subject |
10:42PM |
0 |
odd T1 issue |
8:07PM |
1 |
Voicemail Error |
5:50PM |
7 |
Skype for Asterisk: Public Beta available |
4:50PM |
1 |
Dialplan SIP call back problem |
3:19PM |
2 |
Sound through NAT issue |
8:34AM |
0 |
Request for information about Asterisk Business Edition |
5:56AM |
0 |
Asterisk 1.6 and RFC4235 |
4:09AM |
1 |
Out of office |
2:27AM |
4 |
Looking for wisdom - One Asterisk system - Multi-incoming trunks |
|
Wednesday July 29 2009 |
Time | Replies | Subject |
10:01PM |
1 |
Open Source Pavilion at AstriCon: Your project wanted! |
9:25PM |
3 |
Recording calls again |
7:14PM |
2 |
Recording Calls |
1:52PM |
0 |
Instant messaging (yeah, again) |
1:51PM |
2 |
HPEC > VPM ? |
12:49PM |
1 |
Matching Originate action with its NewChannel event |
6:55AM |
0 |
SIP client Resp code |
5:41AM |
0 |
question about Asterisk-GUI |
12:53AM |
1 |
Misunderstood thing |
|
Tuesday July 28 2009 |
Time | Replies | Subject |
11:06PM |
2 |
Possibly I don't understand sip peers |
9:14PM |
3 |
CIsco 7960 + asterisk: hepl needed |
8:32PM |
1 |
chan_dahdi.conf parser question |
7:37PM |
1 |
Updated patch for 8824? |
6:59PM |
2 |
AGI with queues status |
6:33PM |
1 |
outbound calls not reaching vitelity |
2:54PM |
1 |
sip realtime with caching |
12:50PM |
1 |
CDR.C |
11:06AM |
0 |
Call history problems from B2BUA |
11:05AM |
0 |
Asked to transmit frame type 256, while native formats is 0x4 |
11:01AM |
1 |
sip trunk that fails over time |
10:52AM |
0 |
Meetme Enter/Leave Sounds |
9:06AM |
0 |
Asterisk Crashing on chan_h323 |
8:52AM |
0 |
Inquiry:Asterisk "*" character dialing for IN service |
7:23AM |
0 |
crd wrong destination.... |
5:05AM |
1 |
Inquiry:Asterisk pbx announcements |
5:01AM |
4 |
Inquiry:Asterisk Inter digit delay |
|
Monday July 27 2009 |
Time | Replies | Subject |
11:49PM |
1 |
Fax for Asterisk quick question |
8:30PM |
0 |
Milkfish |
6:19PM |
0 |
Cell phones and (no) rings |
4:28PM |
5 |
Asterisk core dumps files |
4:15PM |
1 |
Player to listen to WAV files using an hardphone |
1:50PM |
2 |
Asterisk and Kamailio NAT problem |
1:29PM |
1 |
disposition "answered" after authenticate?????????? |
1:28PM |
1 |
INVITE Privacy Information |
12:11PM |
1 |
dahdi kernel panic |
10:09AM |
0 |
authenticate with password file on asterisk |
9:37AM |
0 |
mobile hosted pbx |
7:04AM |
0 |
Emulating attended transfer through the dialplan |
2:37AM |
1 |
AMI not show originate on CLI |
|
Sunday July 26 2009 |
Time | Replies | Subject |
5:19PM |
3 |
Not getting inbound CallerID name on Asterisk |
3:51PM |
0 |
after 1.4.26 upgrade: "ast_carefulwrite: write() returned error: Broken pipe" |
3:13PM |
2 |
Verbose() messages go unnoticed |
5:50AM |
0 |
MeetMe time doesn't show up in CDRs? |
12:22AM |
0 |
Audiocodes MP114, 2xFXS, @xFXO - does any one have configuration files they can share for trixbox? |
|
Saturday July 25 2009 |
Time | Replies | Subject |
2:03PM |
0 |
DeadAgi application issue |
9:04AM |
0 |
how to remove MWI from a Polycom phone |
1:43AM |
0 |
Set custom file name for automon recordings |
|
Friday July 24 2009 |
Time | Replies | Subject |
11:08PM |
2 |
TLS Manager |
11:02PM |
1 |
EVERY toll free number appears to be in e164.org?? |
9:43PM |
2 |
How determine extension of who initiated call |
8:13PM |
3 |
Goto from a feature macro is not working? |
7:22PM |
1 |
Russia Calls Skype/VoIP Security Threat |
6:37PM |
0 |
Asterisk-Addons 1.4.9, 1.6.0.3, and 1.6.1.1 Now Available |
4:01PM |
1 |
Anonymous Michigan Calls, Skype/Other |
2:22PM |
2 |
asterisk users |
12:44PM |
6 |
dialplan tips |
10:12AM |
4 |
Asterisk on OpenWRT |
9:16AM |
1 |
FAX Machine Testing ... |
9:04AM |
4 |
Web Browser Pop-up |
8:17AM |
2 |
how to match "no callerid" in 1.6 ? |
2:30AM |
0 |
best option for Conference timing with native Dahdi support |
|
Thursday July 23 2009 |
Time | Replies | Subject |
11:05PM |
7 |
using asterisk on a shared line |
9:35PM |
1 |
nortel cs 1000 swtich |
5:44PM |
1 |
PRI call progress issue |
3:51PM |
1 |
x-lite settings to reach asterisk |
3:28PM |
0 |
detect keys before agi starts |
2:52PM |
2 |
Analog FXO or IAX DIDS for new facility? |
2:35PM |
5 |
Music on hold based on user |
12:54PM |
2 |
Asterisk 1.4.25 and attended transfer |
8:25AM |
5 |
Test Function if SIP Device is Still Alive |
7:51AM |
0 |
how to activate DND on 1.6.0.9 |
7:30AM |
0 |
Friday 2009-07-24 12:00 EDT: Voxeo Labs on VoIP Users Conference |
7:05AM |
1 |
Using Of function SHARED |
3:14AM |
1 |
odd behaviour with AGI and dial agent |
|
Wednesday July 22 2009 |
Time | Replies | Subject |
8:03PM |
2 |
Asterisk CSTA |
7:25PM |
1 |
grandstream and jitter buffer |
5:44PM |
0 |
Attended transfer and 'pbx-invalid' - 1.4.26 |
5:44PM |
0 |
Asterisk as a "gateway" |
3:44PM |
1 |
OT - Do analog gateways detect a phone is plugged in or out ? |
3:31PM |
4 |
A reason TO run Asterisk as root |
2:42PM |
3 |
CallerPres SIP headers Analog Phone |
12:43PM |
3 |
ExecIf and empty variables (early evaluation) |
12:20PM |
2 |
german voiceprompts |
11:27AM |
1 |
Callin Numbers. |
8:30AM |
2 |
sip configuration masking the peers |
7:24AM |
2 |
Waiting for a call to complete with AMI Originate |
5:51AM |
1 |
voicemail does not work from local calls!!! |
5:48AM |
3 |
Inquiry abount Asterisk "extensions.conf" |
|
Tuesday July 21 2009 |
Time | Replies | Subject |
11:29PM |
2 |
Phone system "ping" checker |
10:48PM |
0 |
MWI using Asterisk and external mail server |
8:53PM |
0 |
logging cdr to mysql does not fill clid field |
8:37PM |
1 |
Free Fax for Asterisk -- benchfax utility hangs. |
5:54PM |
1 |
Asterisk 1.4.26 Now Available |
3:04PM |
1 |
Dialplan step that I do not have |
2:49PM |
0 |
Vdex-40 for sale |
2:15PM |
3 |
astmanproxy? |
1:46PM |
1 |
Connecting multiple office with multiple servers |
12:51PM |
2 |
best practices for running asterisk as SIP B2BUA |
12:46PM |
2 |
Graphical Call Manager Allowing Transfer of Any Call? |
11:46AM |
2 |
Channel Variables in a Call file? |
11:06AM |
1 |
Asterisk and G.729 codec: short questions |
10:48AM |
1 |
Scalability and stability matters |
10:43AM |
1 |
externalIVR() and how to do actions |
10:09AM |
0 |
Audio lost on reinvite |
9:05AM |
0 |
Gatekeeper Routing Mode not Working |
5:51AM |
0 |
Asterisk Call Transsfer |
4:01AM |
4 |
how to use patgen and pattest for PRI card? |
|
Monday July 20 2009 |
Time | Replies | Subject |
11:18PM |
0 |
Error: Invalid SIP message - rejected , no call id |
10:10PM |
1 |
How to restrict registrations by useragent? |
9:50PM |
0 |
Vote on whether SipPhone should support ISN routing. |
6:16PM |
1 |
Event Log |
6:10PM |
0 |
[asterisk-dev] MeetMe feature request: bypass pincode |
5:09PM |
3 |
Digium TDM400P in Soekris net5501-70? |
3:36PM |
2 |
What am I doing wrong? |
2:13PM |
1 |
callforward with asterisk-gui.problem with stdexten |
10:18AM |
2 |
asterisk freepbx difference or solutions.. |
3:29AM |
0 |
[asterisk-user] MeetMe feature request: bypass pincode |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
No subject |
12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
No subject |
12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
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12:54AM |
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12:54AM |
0 |
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12:54AM |
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12:54AM |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
0 |
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12:54AM |
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12:54AM |
0 |
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12:54AM |
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|
Sunday July 19 2009 |
Time | Replies | Subject |
10:35PM |
1 |
CyberData SIP-enabled VoIP Intercom |
7:47PM |
0 |
Asterisk-gui 2.0 Asterisk 1.4.26-RC6 Analog trunks |
1:36PM |
0 |
Asterisk not ACKing some 407 Proxy Auth Required requests? |
11:31AM |
0 |
Asterisk or 3CX? |
9:57AM |
3 |
DAHDI Error and poor audio quality |
|
Saturday July 18 2009 |
Time | Replies | Subject |
7:43PM |
1 |
wcte12xp0: Missed interrupt |
4:05PM |
3 |
perhaps libpri issue (thought it was a dahdi issue ) |
3:11PM |
3 |
Count Available Queue members |
2:22PM |
1 |
chan_mobile one device per dongle? |
1:27PM |
1 |
Latest chan_mobile |
4:47AM |
4 |
Asterisk to PBX |
12:31AM |
0 |
Possible WaitUntil Bug |
|
Friday July 17 2009 |
Time | Replies | Subject |
11:35PM |
1 |
Truecall |
9:25PM |
1 |
Voicemail ODBC storage table schema |
7:19PM |
0 |
SPAM |
6:53PM |
1 |
Realtime difference sipusers sippeers |
5:42PM |
0 |
dahdi_tool question for PRI or T1 |
4:30PM |
3 |
Delete voicemail after couple of days |
2:29PM |
1 |
quenstion about asterisk |
1:22PM |
0 |
How to Play IVR and Read DTMF During Active Call? |
11:13AM |
0 |
Friday reminder |
9:58AM |
0 |
Queue member (Agent) does not Dial |
9:11AM |
3 |
dialplan number matching |
8:22AM |
2 |
How do I create an IVR/Dial Group that works properly? |
6:48AM |
2 |
Skill based routing |
6:41AM |
1 |
[HELP] - Conference bridge |
6:33AM |
1 |
MoH - can the volume be adjusted |
6:08AM |
5 |
Asterisk Error |
1:19AM |
1 |
2 Problems with 1.6.2 |
|
Thursday July 16 2009 |
Time | Replies | Subject |
10:59PM |
1 |
Compilation error |
9:45PM |
1 |
Stop recording on SIP attended transfer |
9:39PM |
4 |
800 number portability |
9:38PM |
0 |
Unique id used for call recording missing from CDR data for transferred call |
8:00PM |
1 |
possible to configure 2 servers - one is backup system for the other? |
5:26PM |
2 |
iax.conf, IP-based access control |
4:51PM |
0 |
early-dial SIP 484 "incomplete address", dialplan patterns and international calls |
4:00PM |
3 |
T38 negotiation, the last step ! |
2:57PM |
1 |
Voicemail login incorrect |
2:46PM |
1 |
H323 situation |
1:51PM |
1 |
Sending faxes with T.38 problem. Fax for Asterisk (no SpanDSP) - 1.6.1.1 |
12:44PM |
0 |
AGI to announce temperature from weather.com XMLfile |
12:08PM |
0 |
Struggling with Macros and "s" Extension |
11:49AM |
5 |
AGI to announce temperature from weather.com XML file |
10:21AM |
1 |
Sending things to Jabber but not within an extension |
7:16AM |
1 |
Mexican ITSP needed |
4:44AM |
1 |
advices on how to debridge/rebridge a call? |
|
Wednesday July 15 2009 |
Time | Replies | Subject |
11:57PM |
0 |
16.7.2009 ID69656 71% 0FF on PFIZER ! |
10:25PM |
4 |
Iphone setup |
10:19PM |
1 |
PRI hunt group |
9:26PM |
4 |
DEVICE_STATE() and Asterisk 1.6.0.10 |
7:50PM |
0 |
Queue wrapuptime as Global option |
6:47PM |
0 |
Read/Write Codec formats |
2:42PM |
1 |
Phantom CallerID on transfers |
2:18PM |
1 |
ResetCDR after GotoIf doesn't set dst correctly, Is this a bug? |
1:40PM |
2 |
Generic question about PBX PRI installs |
1:14PM |
2 |
USB phone with Asterisk under Linux |
12:42PM |
0 |
Door Phone |
12:04PM |
0 |
[asterisk-dev] Question |
12:02PM |
0 |
Howto change CDR record on calling channel from called thread? |
6:19AM |
2 |
How to ask questions the smart way |
4:34AM |
2 |
how to enable dial to alex@asterisk.blurb.com |
3:07AM |
2 |
call transfer using DTMF |
|
Tuesday July 14 2009 |
Time | Replies | Subject |
11:06PM |
0 |
TE120P loosing link... |
10:03PM |
2 |
QoS |
6:45PM |
0 |
Help in oh323 Gatekeeper + does not know what to do when bridging the call |
5:20PM |
1 |
Polycom Spectralink 8002 WiFi Phones |
4:09PM |
1 |
Error |
3:39PM |
2 |
How to block inbound call with Asterisk? |
2:14PM |
3 |
Is Enum safe from spammers? |
10:44AM |
2 |
Asterisk 1.4.26 final release - What is blocking? |
10:10AM |
2 |
Asterisk and several clients behind NAT |
8:40AM |
3 |
Help in oh323 Gatekeeper |
8:31AM |
1 |
How to count Parked calls? |
7:55AM |
1 |
unknown RTP codec 126 ?? |
2:00AM |
0 |
ooh323 doesn't know what to do when bridging calls |
12:10AM |
3 |
Why CDR is recording dst value = h? |
|
Monday July 13 2009 |
Time | Replies | Subject |
7:27PM |
0 |
chan_ooh323.so and chan_h323.so |
5:58PM |
1 |
#exec in #include'd file |
4:37PM |
0 |
ooh323 and h323, it accept the call even not added in h323.conf |
3:27PM |
0 |
Polarity Reversal Incorrect |
12:19PM |
2 |
open source call center application for Asterisk |
12:02PM |
0 |
Push-To-Talk? |
10:47AM |
0 |
Go t SIP response 420 "Bad Extension" back from |
6:56AM |
2 |
How to Change size of CDR(accountcode) variable? |
5:06AM |
2 |
transfer option and pressing # |
4:23AM |
4 |
is Asterisk reliable for a call center application?? |
2:23AM |
1 |
Trouble with originating a call through Asterisk Manager Interface |
|
Sunday July 12 2009 |
Time | Replies | Subject |
4:32PM |
0 |
1.6.0.10: server locks up on iax max_retries |
8:20AM |
0 |
Missing CLI |
|
Saturday July 11 2009 |
Time | Replies | Subject |
9:57PM |
0 |
MACRO-INCOMING-CALL-TO-EXTENSION |
4:20PM |
2 |
ooh323 and h323 |
12:55AM |
2 |
Suggestions for web based soft phones |
|
Friday July 10 2009 |
Time | Replies | Subject |
7:46PM |
2 |
Originate (Executing a System Command) |
7:19PM |
2 |
beeping in headsets from queue callers |
6:30PM |
0 |
Meetme problem (talk detection/opt) in 1.6.1.1 |
4:19PM |
0 |
ID0151 C-A-N-A-D-l-A-N P-H-A-R-M-A-C-Y |
3:57PM |
1 |
Dropped Call Problem -- Looking for ideas and a consultant. |
3:55PM |
0 |
Dell poweredge T100 & TE420 |
3:19PM |
1 |
Lagged Extension |
2:30PM |
4 |
[Fwd: confirm f1ab6c493110edited] |
1:39PM |
1 |
Video Call |
12:45PM |
1 |
Kate AEL syntax ? |
9:22AM |
1 |
Friday July 10th: Gigaset DECT/SIP phones have come to the USA |
3:50AM |
1 |
Educational institutions: Your Asterisk experiences wanted! |
|
Thursday July 9 2009 |
Time | Replies | Subject |
10:34PM |
5 |
can 2 quad T1 cards work in 1 quad core amd server |
10:10PM |
2 |
Asterisk Segmentation Faults Using Skinny (v1.6.0.10) |
8:37PM |
1 |
PRI failover to SIP trunk |
7:20PM |
1 |
Connecting two Asterisk together via SIP + DISA |
7:13PM |
1 |
1.6 macro deprecation, dial macros |
6:51PM |
1 |
setting up phones |
5:13PM |
1 |
Weird audio problem with remote IVRs + DMTF |
1:38PM |
0 |
q: port forwarding or NAT |
12:05PM |
2 |
Setting up a "secure" AMI? |
10:11AM |
0 |
Failed to read gains: Invalid argument |
9:30AM |
0 |
Rtp keepalive |
9:26AM |
4 |
is possible to sen sms with asterisk in Spain? |
7:47AM |
1 |
OT - How to indent AEL file |
7:01AM |
1 |
CIDlookup |
6:35AM |
1 |
DIDForSale July Special (No Activation on new DID Purchases) |
2:06AM |
1 |
Dial stops trying after ~30s regardless |
1:39AM |
1 |
q: am i mixing somethign up? |
1:27AM |
0 |
q: sip registration fails... |
|
Wednesday July 8 2009 |
Time | Replies | Subject |
9:05PM |
1 |
Queue autopause |
8:29PM |
0 |
Fwd: q: install asterisk + asteris-gui: SOLVED |
6:50PM |
2 |
q: which Browser-GUI do u guys use? |
6:49PM |
10 |
q: install asterisk + asteris-gui |
2:40PM |
0 |
Grandstream GXP-1200 & G.722? |
1:35PM |
3 |
Restarting of B-channel on span 1 |
12:11PM |
1 |
calculate data traffic |
11:36AM |
0 |
asterisk + cisco as5400 t.38 fax sending. |
5:31AM |
3 |
Asterisk and Skype |
2:06AM |
2 |
g.722 + loudness |
1:55AM |
1 |
One Way Audio from External Sip Soft & Hard Phone |
1:09AM |
0 |
[asterisk-user] AGI control stream file |
|
Tuesday July 7 2009 |
Time | Replies | Subject |
10:26PM |
1 |
asterisk addon mysql - is mysql connection persistent |
6:31PM |
1 |
Play a recorded message when a fax is detected ? |
6:05PM |
1 |
MixMonitor/Queue and Tranfers |
5:40PM |
4 |
Caller ID (name) - where does it come from? |
3:26PM |
3 |
Automatic Gain Control |
1:44PM |
1 |
Resetting Day/Night setting |
1:42PM |
2 |
documentation of DAHDI dial options |
1:05PM |
3 |
Answering the nTh call ... |
6:29AM |
1 |
How to debug "Nothing to pick up" ? |
12:30AM |
1 |
Voicemail attachments not working |
|
Monday July 6 2009 |
Time | Replies | Subject |
8:59PM |
1 |
Bug or Not? |
7:00PM |
1 |
Get channel string |
4:11PM |
0 |
Listed agents in queue not ringing |
3:37PM |
1 |
Variable using AMI |
3:28PM |
3 |
Small site survivability |
2:00PM |
4 |
migrate from zaptel to dahdi |
1:51PM |
1 |
false answer on zaptel |
1:27PM |
1 |
Asterisk & Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error |
9:55AM |
0 |
Iax trunk quality |
9:53AM |
1 |
Monitor |
9:39AM |
0 |
asterisk and mISDN on Solaris |
8:37AM |
2 |
SIP registry fails during night |
7:04AM |
1 |
How to notify that a message is waiting using RingTones with a Dahdi-connected analog phone ? |
5:07AM |
1 |
Asterisk + kamaili MWI(Message waiting Indication) |
3:57AM |
5 |
Dial cmd help |
2:19AM |
2 |
asterisk-GUI |
12:09AM |
3 |
What is the best way to share extension state |
|
Sunday July 5 2009 |
Time | Replies | Subject |
9:06PM |
1 |
SIP IP-Trunk to be authenticated based on username and password, not IP address |
7:02PM |
0 |
(slightly OT) SIP redirect |
5:11PM |
3 |
Queues recording & CDR |
5:09PM |
4 |
chan_mobile help. |
1:15PM |
1 |
Fax for Asterisk download selector broken? |
11:25AM |
1 |
Source for OpenVox cards? |
|
Saturday July 4 2009 |
Time | Replies | Subject |
1:17PM |
2 |
Call parking with ISDN |
10:54AM |
0 |
Asterisk and ENUM |
10:24AM |
1 |
Channel / peer available |
2:14AM |
0 |
debian lenny, asterisk1.6 + freepbx2.5.1 |
12:12AM |
1 |
Music on Hold |
|
Friday July 3 2009 |
Time | Replies | Subject |
7:10PM |
0 |
DAHDI CDR problem |
6:38PM |
2 |
dahdi_dummy configure |
5:39PM |
1 |
Asterisk + Openfire |
3:55PM |
1 |
New attempt : Trigger an action when B number answers the call |
3:42PM |
0 |
Fw: Trigger an action when B number answers the call |
3:30PM |
2 |
Trigger an action when B number answers the call |
3:29PM |
0 |
e164.org and tollfree ENUM records |
2:14PM |
1 |
*Sort of Commercial* TracFone's $45 unlimited offer to 'stun' rivals |
1:46PM |
1 |
DTMF is not working occasionally over IAX Trunk |
12:26PM |
1 |
MISDN/asterisk problem (not sure where from) |
11:24AM |
7 |
Asterisk capacity |
10:58AM |
2 |
Problem configuring TDM400 |
9:50AM |
0 |
Typ of Number / Modify in Peer-Definition |
7:47AM |
0 |
asterisk-users Digest, Vol 60, Issue 9 |
6:18AM |
1 |
Some IAX calls do not disconnect. |
4:53AM |
0 |
Converged mail box sizes |
3:15AM |
1 |
DAHDI |
12:59AM |
1 |
Zimbra IMAP authentication - SOLVED |
|
Thursday July 2 2009 |
Time | Replies | Subject |
11:59PM |
1 |
Dial |
7:36PM |
1 |
Rajkiran Reddy sent you a Friend Request on Yaari |
7:20PM |
4 |
Using a mobile phone via USB as an extension |
6:55PM |
1 |
need help, service unavailable, registered but call does not get through |
6:05PM |
1 |
Why Asterisk + Kamailio ? |
4:33PM |
3 |
Grandstream 2010 and blinky lights |
2:53PM |
3 |
Using the PBX Directory from a Blackberry |
12:34PM |
1 |
AGI Transfer? |
8:47AM |
0 |
Ext1: Channel X parameter on PRI |
8:09AM |
1 |
Nortel pbx & dtmf issues |
|
Wednesday July 1 2009 |
Time | Replies | Subject |
11:03PM |
2 |
/var/lib/asterisk/sounds does not exist |
8:51PM |
4 |
g729a compatibility |
7:18PM |
2 |
Registrations problems to SIP-provider. |
6:19PM |
1 |
UK Vodafone femtocells now available |
5:25PM |
2 |
Testing the manager.conf: sending and receiving commands |
3:16PM |
6 |
* as VM for legacy PBX? |
8:54AM |
1 |
Fwd: Unknown udp ports listening experts calling ! |
6:17AM |
2 |
Multi-tenant parking broken in 1.6.1.1? |
12:49AM |
10 |
Welcome Message |