Sunday August 31 2008 |
Time | Replies | Subject |
11:09AM |
4 |
Asterisk IVR Scalability |
2:56AM |
0 |
Intermittent "rejected because extension not found" On Incoming DID |
12:15AM |
2 |
security on localhost connections |
|
Saturday August 30 2008 |
Time | Replies | Subject |
8:55PM |
0 |
Reliable wireless SIP phones (Tzafrir Cohen) |
5:32PM |
1 |
beta9: how to set callerid on incoming iax? |
5:17PM |
3 |
Congestion in Outgoing call through PRI |
4:51PM |
2 |
Wi-SIP & 802.11f - Inter Access Point Protocol HANDOFF |
4:31PM |
1 |
Heist of MagicJack SIP credentials? |
7:05AM |
0 |
Incoming Calls via SIP Trunks |
1:49AM |
0 |
Zap channel DTMF regeneration |
|
Friday August 29 2008 |
Time | Replies | Subject |
9:46PM |
1 |
Audio data between concurrent SIP and PSTN |
7:57PM |
0 |
CDR userfield recording name |
6:14PM |
3 |
Call monitor/barge/train |
4:44PM |
1 |
music on hold is not working |
4:39PM |
1 |
Issue when dialing multiple extensions using & ------Please Help |
3:52PM |
0 |
chan_mobile |
3:35PM |
1 |
Connecting two asterisks via IAX |
3:17PM |
6 |
Faxing through Zap cards |
3:03PM |
0 |
track 1.6 progress |
2:58PM |
5 |
Wi-SIP vs. SIP-DECT |
9:35AM |
0 |
Asterisk cdr_mysql inexact values |
4:56AM |
2 |
Asterisk CDR Problem |
|
Thursday August 28 2008 |
Time | Replies | Subject |
11:50PM |
1 |
Is including a linefeed in the JabberSend message possible? |
9:53PM |
4 |
Transfers on AgentLogin() |
5:44PM |
1 |
troubleshooting mISDN... |
5:40PM |
2 |
GSM recordings |
5:24PM |
0 |
Caller ID in IAX trunk, SIP trunk, between extensions and from FXO |
5:12PM |
2 |
sip conversations overlapping!!!! |
5:08PM |
0 |
meetme + jitter buffer |
4:07PM |
0 |
VoicePulse Time out? |
2:32PM |
1 |
asterisk linkedin group |
2:11PM |
0 |
OT: SEP<mac addr>.cnf.xml file for 7911 with SIP 8.3.5 firmware |
1:06PM |
8 |
Reliable wireless SIP phones |
11:26AM |
5 |
Asterisk Queue's |
11:18AM |
0 |
Weird asterisk error: ztscan command not found |
10:59AM |
4 |
Problems with DTMF on IVRs |
9:16AM |
3 |
Console softphone |
8:32AM |
5 |
H323 protocol |
7:22AM |
1 |
execute command after sip register |
7:08AM |
3 |
Asterisk 1.4 -> 1.6 |
6:10AM |
1 |
Asterisk CLI Show Error :- ("**Unknown**") instead of ("Zap/22-1", ) |
4:01AM |
2 |
can not load chan_dahdi.so from asterisk! |
2:02AM |
2 |
remove queue call |
1:06AM |
6 |
Pri to sip interfaces |
|
Wednesday August 27 2008 |
Time | Replies | Subject |
10:43PM |
0 |
Asterisk and Linksys One (PHB1100) |
9:05PM |
1 |
OT Polycom URI and IP address dialing. Not. |
6:07PM |
2 |
Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura |
5:08PM |
2 |
Problem with Call Forward |
3:20PM |
1 |
Fax issue over cisco gateway |
2:50PM |
1 |
Callback voice Quality |
2:46PM |
1 |
VUC Friday: asterisk 1.6 fax, Drawing for Free Astricon Pass |
2:16PM |
1 |
problem making outgoing calls |
2:13PM |
0 |
sip show peers from shell or from CLI [SOLVED] |
1:55PM |
1 |
Call Files |
1:27PM |
4 |
PRI Splitter |
12:00PM |
2 |
sip show peers from shell or from CLI |
8:53AM |
0 |
asterisk-1.6, Remote-Party-ID Header not sent |
6:09AM |
1 |
compile Dahdi ! |
4:44AM |
1 |
Digium Coffee anyone? PCI Expresso? WTF? |
4:41AM |
0 |
Asterisk for calling no of users |
12:28AM |
6 |
Need application, CID number match list to call cell phone |
|
Tuesday August 26 2008 |
Time | Replies | Subject |
10:55PM |
1 |
FreeTDS Versions? |
9:56PM |
1 |
app_jack and calling with pc only |
4:24PM |
6 |
X100P Card in OFFHOOK state |
3:49PM |
0 |
te410p remains in red-alarm |
2:49PM |
2 |
Asterisk connected to the PSTN vs. a commercial solution |
11:49AM |
1 |
Limit to the length of string ? |
8:51AM |
0 |
Asterisk/Other PBX interconnection |
5:55AM |
0 |
about iax2 prune realtime |
1:55AM |
5 |
implementing an intercom with asterisk |
|
Monday August 25 2008 |
Time | Replies | Subject |
10:33PM |
1 |
is shared_lastcall available in 1.4 |
5:29PM |
0 |
sip.conf templates and realtime |
5:12PM |
2 |
How to query a remote MySQL DB from dialplan |
4:32PM |
1 |
Get call status and hangup |
3:26PM |
2 |
Call transfer over IAX trunk |
1:46PM |
0 |
Problem with dtmf in voicemailmain |
1:06PM |
2 |
sip peering between 2 asterisk |
12:08PM |
2 |
asterisk realtime |
11:23AM |
1 |
Static IP for SIP? |
10:51AM |
4 |
TDM2400P Voice Quality Problem |
10:26AM |
2 |
Really WEIRD: can register but can not call! |
9:41AM |
0 |
Problem using blind transfer |
8:50AM |
0 |
A question about the ${CHANNEL} |
6:47AM |
0 |
Consequent Dial commands not ringing |
6:31AM |
0 |
ASM / AMI Assisted Live Transfer |
6:10AM |
3 |
Read Command |
2:13AM |
0 |
wct4xxp alarmdebounce |
|
Sunday August 24 2008 |
Time | Replies | Subject |
11:25PM |
0 |
RemoveQueueMember race condition |
7:17PM |
3 |
SECURITY QUESTION & SANITY CHECK |
4:34PM |
0 |
[Xen-users] Xen 3.2.1 and PCI passthrough |
4:29PM |
2 |
MWI working perfectly. Shouldn't it be broken?? |
4:13PM |
0 |
Asterisk - forcing PSTN Line "ON" Hook |
3:26PM |
6 |
entering a password to have access to a sip account?! |
9:50AM |
0 |
Realtime SIP |
6:51AM |
0 |
RTP timestamp modification during SIP video call |
4:36AM |
2 |
Question about Dialing DTMF |
|
Saturday August 23 2008 |
Time | Replies | Subject |
4:19PM |
1 |
Anything to convert from JSON into Asterisk dialplan variables? |
3:16PM |
1 |
Asterisk, 2-way radio systems, app_rpt and chan_rtpdir |
1:43PM |
2 |
Semi-OT Satellite? |
11:34AM |
0 |
Free Today ! |
10:30AM |
2 |
OT - Which rackable case for mini-ITX boards ? |
9:49AM |
1 |
Problems with D-channel (PRI) |
8:56AM |
4 |
Global VoIP Calls? |
6:53AM |
0 |
Blind Transfer is not working in incoming calls |
6:39AM |
1 |
Voicemail has issues with DTMF |
|
Friday August 22 2008 |
Time | Replies | Subject |
11:28PM |
1 |
ztd-ethmf |
11:00PM |
1 |
frequent channel reset problem |
10:25PM |
1 |
Fw: [asterisk-dev] frequent channel reset problem |
6:40PM |
1 |
em wink |
6:26PM |
3 |
queue timeout |
3:39PM |
0 |
RES: DSS1 vs SS7 |
3:06PM |
0 |
Friday's conference meeting - Astricon is in the air |
2:55PM |
1 |
Diamondware spatial conferencing |
12:08PM |
0 |
Asterisk, Xen and a TDM400P |
7:37AM |
1 |
interesting RDNIS question |
7:28AM |
4 |
set callerid with plus sign |
6:34AM |
2 |
About the CALLIDNUMBER of the fxs |
3:10AM |
3 |
Linksys - Sipura VMWI splash ring |
12:34AM |
0 |
The problem of the fxs |
|
Thursday August 21 2008 |
Time | Replies | Subject |
11:45PM |
2 |
How to block incoming calls on PRI |
10:03PM |
3 |
Suddenly the voice become like robot (cutting), like sick man |
5:04PM |
1 |
ultramonkey and asterisk |
4:50PM |
2 |
Siemens Gigaset IP in USA (S685 IP in particular) |
4:50PM |
4 |
5 min limitation on phone calls! how to! |
4:40PM |
1 |
Question: Soft phone for ACD agents? |
4:02PM |
2 |
Asterisk and Huawei SoftX3000 |
3:39PM |
1 |
Automatic call to voicemail on login? |
3:35PM |
3 |
After Dial execution, using DIALEDTIME, ANSWEREDTIME |
3:19PM |
2 |
callfiles/manager api originate call fails |
3:08PM |
1 |
Asterisk Realtime pounds MySQL |
3:03PM |
0 |
1st call after some time has one way speech, but calls after that are fine.. |
2:47PM |
2 |
A Suggestion To Asterisk Appliance Developers |
2:29PM |
2 |
Anyone using asterisk on centos 4.X without hardware cards and using console/dsp |
1:26PM |
1 |
OT - Asterisk-Stats - Billsec instead of Duration |
12:16PM |
2 |
How can I determine if IAX trunking is being used and how many calls are being trunked? |
12:11PM |
2 |
Changing callerID in a context |
11:57AM |
3 |
IVR question |
9:52AM |
0 |
Any chance this is related to fastagi: received mini-frame before full voice frame |
6:36AM |
1 |
DSS1 vs SS7 |
2:06AM |
1 |
The problem of the ${CALLERID(num)} for the fxo |
12:06AM |
1 |
Problem with Qualify sip peers... |
|
Wednesday August 20 2008 |
Time | Replies | Subject |
11:29PM |
2 |
Asterisk build-environment in Xen-DomU |
7:53PM |
0 |
2BCT from the Asterisk side |
7:09PM |
3 |
Two peers, same IP and port |
5:21PM |
1 |
3-way conference call |
5:00PM |
3 |
Is there a way to encrypt passwords stored in the realtime database? |
1:04PM |
1 |
Reproduce DeadAGI behavior with AGI |
12:21PM |
0 |
Why does a perfectly fine iax2 host becomes UNREACHABLE? |
12:06PM |
1 |
FTC Bans Prerecorded Telemarketing Drivel |
10:37AM |
1 |
vicidial mysql problem |
9:55AM |
0 |
[Question]Setting CallerIDName using Asterisk Manager Interface API |
9:33AM |
0 |
how to cross compile asterisk for cirrus edb9302a board |
9:12AM |
0 |
IAX2 and transfer=mediaonly, Error unable to transfer but there is sound. |
2:14AM |
2 |
Linksys SPA3102-NA firmware upgrade on Linux |
|
Tuesday August 19 2008 |
Time | Replies | Subject |
3:59PM |
2 |
Missing 'full' log |
3:09PM |
0 |
Free US Based Echo Test |
11:57AM |
2 |
Help with Asterisk to Huawei SoftX3000 registry problem |
11:10AM |
3 |
Inefficient Codec Translation |
8:06AM |
0 |
Allow asterisk to receive calls |
6:09AM |
2 |
GUI: User account for SIP Provider results to "Punctuation and Special Characters are not allowed in this field." |
5:54AM |
0 |
asterisk-users Digest, Vol 49, Issue 44 |
2:45AM |
2 |
Gnudialer runninig |
|
Monday August 18 2008 |
Time | Replies | Subject |
7:59PM |
5 |
opening Doors with Asterisk!? |
5:06PM |
0 |
asterisk-users Digest, Vol 49, Issue 43 |
4:41PM |
4 |
US-based echo test servers? |
4:30PM |
2 |
Asterisk Stops...where to look? |
4:03PM |
1 |
Strange putconfig bahviour |
7:23AM |
3 |
Voicemail |
1:34AM |
1 |
ZTDUMMY Running but IAX2 message:Unable to support trunking on peer 'XXXXXXXX' without zaptel timing |
|
Sunday August 17 2008 |
Time | Replies | Subject |
10:26PM |
0 |
asterisk -n switch |
5:04PM |
0 |
asterisk-users Digest, Vol 49, Issue 42 |
4:56PM |
0 |
1.6 call-limit |
4:07PM |
1 |
pollmailboxes |
8:51AM |
2 |
Running asterisk as non root user |
|
Saturday August 16 2008 |
Time | Replies | Subject |
6:47PM |
1 |
disable auth between two asterisk |
5:08PM |
0 |
How to create a user that uses mISDN using Asterisk GUI |
5:04PM |
0 |
asterisk-users Digest, Vol 49, Issue 41 |
3:53PM |
1 |
Which is the correct GUI vor Asterisk 1.4 ? |
1:57PM |
1 |
dialplan reload and dropped calls |
8:36AM |
0 |
Getting cdr(billsec) 0 -- please help |
5:02AM |
0 |
asterisk-users Digest, Vol 49, Issue 40 |
4:57AM |
1 |
Maybe a crazy idea, but are there Asterisk hoster outside there? |
1:50AM |
0 |
Basic outbound calling issue : a lot closer |
|
Friday August 15 2008 |
Time | Replies | Subject |
9:22PM |
1 |
Asterisk AGI and php problem.... |
8:17PM |
2 |
dahdi link broken |
7:48PM |
0 |
asterisk-users Digest, Vol 49, Issue 39 |
7:06PM |
3 |
Problems assigning mISDN Trunk using the DIGIUM Asterisk GUI |
5:52PM |
1 |
PRI TBCT - Practical Experience, Anybody? |
5:37PM |
0 |
Incoming Bogota DID |
5:27PM |
1 |
Cisco 7960 audible hold reminder? |
5:04PM |
0 |
asterisk-users Digest, Vol 49, Issue 38 |
4:52PM |
2 |
zaptel timing |
4:43PM |
4 |
dahdi and ztdummy |
2:49PM |
2 |
DID's needed for Reston Virginia - + hosted asterisk |
2:28PM |
1 |
Problem with Aastra 480ci and qualify=yes |
12:18PM |
0 |
asterisk-users Digest, Vol 49, Issue 37 |
11:59AM |
1 |
noise cancelling headset vs handset |
5:56AM |
3 |
AstDB/Berkely DB - Hash function? Balanced-Tree? b-Tree? Linked List? |
5:31AM |
3 |
Asterisk stress call test |
3:16AM |
0 |
asterisk-users Digest, Vol 49, Issue 36 |
3:03AM |
0 |
integrating SipTheeSkype in asterisk |
12:53AM |
5 |
asterisk realtime and creating "new" contexts |
|
Thursday August 14 2008 |
Time | Replies | Subject |
11:17PM |
0 |
Comdial IP thru Asterisk to cell |
8:22PM |
0 |
Astricon approaches and more tomorrow at 12 Noon EDT on VUC |
8:09PM |
1 |
USA Lata AreaCode Database |
7:49PM |
1 |
Asterisk vs c-client issues |
6:35PM |
3 |
ANSI terminal colors |
6:22PM |
0 |
asterisk-users Digest, Vol 49, Issue 35 |
5:58PM |
0 |
[asterisk-dev] Asterisk + OpenIMSCore |
5:19PM |
1 |
Still have a callno... |
5:15PM |
0 |
asterisk-users Digest, Vol 49, Issue 34 |
4:06PM |
1 |
AMI and extensions.conf |
3:55PM |
0 |
asterisk-users Digest, Vol 49, Issue 33 |
8:25AM |
1 |
Accept anonymous connections from Unknown Peer |
5:30AM |
0 |
asterisk-users Digest, Vol 49, Issue 32 |
4:51AM |
1 |
Unable to create ZAP Channel |
1:46AM |
1 |
Asterisk 1.4 T38 UDPTL Pass Through MAX TNT and Linksys 2102 |
|
Wednesday August 13 2008 |
Time | Replies | Subject |
11:48PM |
0 |
Callwaiting and CallerID on ZAP PSTN Line |
11:44PM |
1 |
Cisco 7960 |
11:40PM |
4 |
Asterisk might be dropping RTP packets before reaching eth int? |
10:28PM |
1 |
ztdummy on centos 4.6 i386 |
10:20PM |
3 |
Open door automatically... |
9:24PM |
0 |
Asterisk stops sending RTP packets to ethernet interface |
9:06PM |
2 |
New GUI for Realtime Asterisk - RAGUI |
8:03PM |
2 |
rtc issue |
7:09PM |
1 |
seeking hardware recommendation PCI versus PCI Express E1 card (te407p vs te420bf) |
5:17PM |
0 |
call forward spa 841 and asterisk 1.4.21 |
5:05PM |
0 |
asterisk-users Digest, Vol 49, Issue 31 |
3:41PM |
1 |
ENUM lookup |
2:56PM |
1 |
Asterisk and Radius |
2:32PM |
0 |
asterisk-users Digest, Vol 49, Issue 30 |
2:31PM |
0 |
Decline message |
2:30PM |
1 |
Decline issue |
1:56PM |
1 |
Sending Set Asynchronous Balanced Mode Extended |
9:50AM |
4 |
cmdRecord issue related to "iax2 received mini frame before first full voice frame"? |
5:22AM |
1 |
Off Topic: Disable Polycom Soundpoint DoNotDisturb Feature |
1:35AM |
0 |
asterisk-users Digest, Vol 49, Issue 29 |
|
Tuesday August 12 2008 |
Time | Replies | Subject |
10:45PM |
3 |
BLF functionality |
9:32PM |
1 |
Error after svn co of lastest zaptel 1.4 |
8:42PM |
1 |
LNP Problems |
6:45PM |
0 |
Problems with queue member status |
6:27PM |
2 |
OT: Asterisk on fitPC |
5:30PM |
5 |
I used to use an Asterisk server, but now it is overkill, ... |
5:14PM |
0 |
asterisk-users Digest, Vol 49, Issue 28 |
2:48PM |
1 |
intermediate accounting records |
2:44PM |
1 |
distinctive ring on sipura |
2:39PM |
1 |
CDR accuracy |
1:29PM |
2 |
Asterisk issue |
12:48PM |
0 |
DTMF is Not working in VOICEMAIL |
12:28PM |
1 |
Unable to compile asterisk-addons from trunk |
9:26AM |
0 |
InBound call Barging |
3:17AM |
0 |
park calls - cannot hear digits being played |
2:14AM |
0 |
asterisk-users Digest, Vol 49, Issue 27 |
|
Monday August 11 2008 |
Time | Replies | Subject |
10:59PM |
1 |
Intermittent T.38 pass through |
8:38PM |
2 |
phone rings "once" before playing message |
6:34PM |
1 |
Phone system layout suggestions |
5:07PM |
1 |
Asterisk Realtime CLI command |
5:04PM |
0 |
asterisk-users Digest, Vol 49, Issue 26 |
4:21PM |
2 |
Asterisk broadcast to web |
4:19PM |
0 |
asterisk-users Digest, Vol 49, Issue 25 |
4:10PM |
2 |
Originate Status Monitoring |
2:50PM |
0 |
out going call files and correct dial status |
1:46PM |
1 |
Asterisk Realtime Unregister |
11:08AM |
1 |
1.4 SVN / dahdi / meetme / -> unable to open pseudo device |
10:15AM |
0 |
Found unknown media description format |
9:20AM |
1 |
DCAP on Linuxtag - Berlin 2009 |
8:01AM |
1 |
deadalocks in asterisk |
7:03AM |
3 |
The problem DIAL with option T,t |
|
Sunday August 10 2008 |
Time | Replies | Subject |
6:31PM |
1 |
H323 Issue |
5:07PM |
0 |
asterisk-users Digest, Vol 49, Issue 24 |
|
Saturday August 9 2008 |
Time | Replies | Subject |
10:30PM |
0 |
Explain t38 and how it relates to Asterisk |
6:30PM |
1 |
asterisk - iaxmodem - create_addr: No such host: ttyIAX |
5:08PM |
0 |
asterisk-users Digest, Vol 49, Issue 23 |
10:16AM |
2 |
New FXO/FXS interface needed.. |
9:23AM |
0 |
Building Asterisk-1.4.21.2~dfsg on Ubuntu 8.04.1 LTS x86 |
6:53AM |
3 |
cdr-custom rotate? |
6:01AM |
1 |
how to know what codec is being used |
5:58AM |
0 |
asterisk-users Digest, Vol 49, Issue 22 |
5:47AM |
1 |
channel_find_locked: Avoided initial deadlock for |
|
Friday August 8 2008 |
Time | Replies | Subject |
10:02PM |
2 |
VICIDial error |
8:46PM |
1 |
can I get a little criticism? |
8:31PM |
1 |
SIP TLS error: ast_make_file_from_fd: FILE * open failed |
8:27PM |
1 |
Auto Dialer proof of concept |
8:23PM |
2 |
IAX2 encryption - LAN. no, INET: yes??? |
7:19PM |
1 |
System call never returns |
7:17PM |
1 |
chan_mobile: scrambled audio, no MOH, no call signalization |
5:31PM |
4 |
Semi-OT: ServerBeach for VoIP |
5:07PM |
1 |
asterisk-users Digest, Vol 49, Issue 21 |
4:30PM |
1 |
channel variables not kept |
3:16PM |
2 |
Zap channels stuck... |
2:58PM |
0 |
Asterisk configuration |
1:08PM |
3 |
manager/originate |
12:17PM |
0 |
asterisk-users Digest, Vol 49, Issue 20 |
9:48AM |
1 |
asterisk-addons-1.6.0-beta4 compile error |
6:11AM |
1 |
h323 channel compile error |
|
Thursday August 7 2008 |
Time | Replies | Subject |
10:47PM |
0 |
asterisk-users Digest, Vol 49, Issue 19 |
9:55PM |
3 |
AGI and Call Center to do CRM integration |
8:04PM |
7 |
Asterisk end-user GUI? |
7:31PM |
5 |
FAX t.38 on Asterisk 1.6? |
5:12PM |
0 |
asterisk-users Digest, Vol 49, Issue 18 |
5:04PM |
2 |
FWD $30 membership-fee |
4:46PM |
4 |
BRI AND DATA connection |
3:49PM |
0 |
asterisk-users Digest, Vol 49, Issue 17 |
3:37PM |
1 |
Improving the speed of chan_sip |
1:32PM |
1 |
outgoing call file and agi detect busy |
1:24PM |
1 |
Voicemail on PRI |
12:39PM |
0 |
I canĀ“t hear the warning sound in Dial command |
10:45AM |
1 |
problem controlling dialplan order |
9:34AM |
0 |
asterisk-users Digest, Vol 49, Issue 16 |
9:20AM |
0 |
[HELP] Regarding stripping of fmtp parameters for Video. |
5:28AM |
1 |
Randulo: An open suggestion for the VOIP users Conference |
3:35AM |
1 |
Capture digits, set as variable..., use for caller id? |
1:11AM |
0 |
Trying to understand Messages from chan_zap.c |
|
Wednesday August 6 2008 |
Time | Replies | Subject |
9:04PM |
2 |
intercom/paging with grandstream gxp2000 |
8:08PM |
1 |
Digium B410P: problematic Bri connection between * and a legacy Philips PBX |
5:45PM |
3 |
Strange beep during calls |
5:40PM |
3 |
Max amount of concurrent calls on and iax trunk |
5:04PM |
0 |
asterisk-users Digest, Vol 49, Issue 15 |
4:47PM |
0 |
Regarding fmtp parameters. |
3:40PM |
0 |
asterisk-users Digest, Vol 49, Issue 14 |
3:24PM |
1 |
Action on login |
1:05PM |
1 |
problem with iaxmodem! |
1:02PM |
2 |
Transcoding |
12:47PM |
1 |
asterisk realtime user deletion |
11:34AM |
1 |
OT: Cisco 7961 SIP downgrade from 8.3.3 -> 8.0.4SRS2 failing |
9:50AM |
0 |
About the features.conf of it's transfer |
9:12AM |
2 |
shared mysql connection in dialplan |
9:09AM |
1 |
does astcanary really work? |
7:42AM |
0 |
asterisk-users Digest, Vol 49, Issue 13 |
6:27AM |
0 |
Asterisk takes incoming call before extension was submitted |
12:38AM |
1 |
G722 capable soft phone? |
|
Tuesday August 5 2008 |
Time | Replies | Subject |
11:26PM |
0 |
libpri versions 1.2.8 and 1.4.7, and libss7 version 1.0.1 released |
10:10PM |
5 |
email notification to external email address |
8:03PM |
0 |
When shall SIP phone reply "480 Temporarily Unavailable" |
6:36PM |
2 |
Asterisk to Avaya |
4:12PM |
0 |
ZRTP in Asterisk |
4:01PM |
1 |
Codec g729 issues |
3:47PM |
4 |
vars in Macros called by DIAL with option M |
2:49PM |
1 |
Getting Asterisk out of the RTP media path |
2:30PM |
1 |
Grandstream RS-232 config (slightly off-topic) |
12:40PM |
2 |
Queue Penalties not working properly |
8:16AM |
1 |
"Asterisk dead but subsys locked" |
5:55AM |
2 |
a simple Asterisk AMI interface with Delphi (or Lazarus+FreePascal) |
12:35AM |
1 |
Off Topic: 8x FXO Gateway |
|
Monday August 4 2008 |
Time | Replies | Subject |
9:28PM |
0 |
Buffer re-sync with Openvox card... |
5:36PM |
2 |
in-call start monitoring |
3:30PM |
2 |
multiple asterisk approach |
3:13PM |
1 |
Building Asterisk-1.4.21.2~dfsg-1 |
1:34PM |
3 |
OT: TechShop |
12:57PM |
3 |
Asterisk Realtime with MySQL Registration Failed |
9:49AM |
3 |
Autoanswer in Nokia SIP clients? |
9:04AM |
7 |
Customized Queuing Strategy |
8:32AM |
0 |
gsm files instead mp3 files in a conference room! |
7:58AM |
2 |
skype and Asterisk opensource integration |
4:02AM |
1 |
FC2 and Zaptel |
|
Sunday August 3 2008 |
Time | Replies | Subject |
10:59PM |
0 |
asterisk-users Digest, Vol 49, Issue 6 |
10:15PM |
1 |
Bad recorded audio quality (upgrade). |
8:06PM |
0 |
Random reboots on IP-601 after changing network topology |
5:54PM |
1 |
TDMoE with Telco |
5:13PM |
0 |
No MOH on SIP hold nor on park |
4:20PM |
1 |
Least Cost Routing |
10:10AM |
0 |
SIP Registration |
7:05AM |
3 |
PRI device is down |
3:30AM |
1 |
lookup for '_sip._udp.sip.stanaphone.com' |
|
Saturday August 2 2008 |
Time | Replies | Subject |
9:07PM |
0 |
Asterisk Now - rPath virtual appliance |
8:59PM |
1 |
How do I issue a Flash to Zap (PSTN) from SIP? |
7:03PM |
1 |
Voxeo |
1:53PM |
4 |
2000+ user Asterisk PBX |
|
Friday August 1 2008 |
Time | Replies | Subject |
8:02PM |
2 |
Cisco 7970, CTLSEP<mac>.tlv |
7:16PM |
0 |
sip show peer [load] says not a realtime peer |
6:34PM |
2 |
Call Logs |
6:29PM |
0 |
app_flite 0.6 released |
6:20PM |
4 |
how many quad T1 cards |
4:34PM |
1 |
[Dundi] Looking for new peers/limited time only |
3:36PM |
6 |
auto provisioning phones |
1:43PM |
1 |
HI ~ good friend, |
1:29PM |
3 |
Asterisk Queues problem |
1:13PM |
2 |
asterisk appliance A50 vs asterisk open source + fxo cards |
9:27AM |
0 |
Authentication on an LDAP server |
7:57AM |
1 |
Comparing origination from CLI and from AMI |
7:16AM |
0 |
Russian Calling card Voice prompt |
4:27AM |
1 |
Scour.com invite from rahul |
2:33AM |
1 |
XMPP developers |
12:28AM |
1 |
AMI able to call from known endpoint to unknown endpoint? |