| Thursday January 31 2008 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
Asterisk 1.4.18-rc4 Now Available |
| 9:58PM |
0 |
alcatel omnipcx |
| 9:10PM |
0 |
Problem picking up a call with PickUpChan or PickUp |
| 7:25PM |
0 |
FS: A20101D Sangoma Board 2 Port FXO 2 Port FXS w/ Echo Can |
| 6:45PM |
1 |
Bunch of set-up/usage questions (SLA, MWI, SMS proxy's, crypto, Fax, etc) |
| 4:55PM |
1 |
VoIP Users Conference Friday Feb 1st @ 12 Noon EST: Hosted IVR |
| 4:46PM |
1 |
Default delay time for Attended call |
| 3:58PM |
0 |
hint is hanging when remote party ends call on hold |
| 3:09PM |
2 |
Analog Adapters ? |
| 11:45AM |
1 |
Dropped calls |
| 11:36AM |
1 |
createlink with out agents in 1.4 |
| 10:24AM |
1 |
Could not find ooh323.conf |
| 10:09AM |
0 |
OT - SIP phones supporting LLDP-Med |
| 9:59AM |
1 |
Incoming call from SIP proxy to asterisk |
| 9:46AM |
0 |
Realtime device update weirdness |
| 7:35AM |
1 |
Server Compatibility List for Asterisk |
| 5:30AM |
7 |
pulling my hair out over voicemail |
| 5:08AM |
2 |
How to get called number in featuremap |
| 3:10AM |
2 |
CallerID shows wrong values in manager interface |
| |
| Wednesday January 30 2008 |
| Time | Replies | Subject |
|---|
| 9:51PM |
1 |
Default delay time for Attended call transfer |
| 8:55PM |
2 |
G729 version to be downloaded for my machines |
| 8:18PM |
0 |
calls get stuck in the asterisk box |
| 8:13PM |
0 |
OT - Looking for used 2 FXS port pci card |
| 7:52PM |
0 |
Asterisk 1.4.18-rc3 Now Available |
| 7:32PM |
0 |
conf meetme all exited and still active |
| 7:21PM |
1 |
Unicall CRN 32769 - far disconnected cause=Switching equipment congestion [42] |
| 6:13PM |
1 |
Parking lot |
| 4:38PM |
2 |
func_odbc - trouble |
| 4:23PM |
4 |
Meetme voice quality problems |
| 2:44PM |
1 |
Using two SIP-Domains with asterisk |
| 2:32PM |
0 |
No audio one way |
| 12:38PM |
3 |
Can't read environment variable |
| 12:09PM |
1 |
Facing problem in installing asterisk-addons |
| 7:21AM |
7 |
Problem with DTMF dialing |
| 3:34AM |
2 |
sipsock_read: BAD! BAD! BAD! |
| 2:34AM |
4 |
asterisk gateway |
| 1:21AM |
0 |
Queue works with across server agent? |
| 1:20AM |
1 |
Queue - ${ANSWEREDTIME} |
| |
| Tuesday January 29 2008 |
| Time | Replies | Subject |
|---|
| 10:51PM |
1 |
chanspy does not pull the call back to asterisk after a reinvite |
| 10:31PM |
5 |
Source Based Call Routing |
| 9:15PM |
0 |
ShoreTel <-> Asterisk Integration |
| 8:18PM |
2 |
Queue member add |
| 7:24PM |
1 |
speex, ilbc and g729 codecs |
| 7:11PM |
2 |
When does Asterisk "REFER"? |
| 6:59PM |
2 |
POE draw on Aastra 480i |
| 4:51PM |
1 |
softmodems bank for ast. |
| 4:29PM |
2 |
Asterisk 1.4.18-rc2 Now Available |
| 3:15PM |
1 |
codec_g729a.so problem... |
| 2:32PM |
1 |
SET with pipe symbol |
| 2:11PM |
1 |
test please ignore |
| 11:08AM |
1 |
PRI Alarms, Comes Back, But Asterisk Won't Touch It! |
| 10:03AM |
2 |
Do Asterisk requires audio codec to be installed? |
| 6:46AM |
0 |
Installation of gatekeeper-H323plus |
| 6:24AM |
2 |
Asterisk mem leak behavior? |
| 4:56AM |
8 |
Asterisk's DANGEROUS Transfer CDR's |
| 4:12AM |
2 |
Dialogic card |
| 2:59AM |
0 |
Asterisk and MRTG, a little help please...WORKING |
| 12:31AM |
0 |
Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available |
| 12:03AM |
2 |
MFC/R2 |
| |
| Monday January 28 2008 |
| Time | Replies | Subject |
|---|
| 11:38PM |
2 |
IAX Calls - One Way Audio |
| 11:05PM |
2 |
SIP DTMF Troubleshoot |
| 10:57PM |
0 |
ISDN Internal Bus? |
| 6:37PM |
1 |
zaptel and oslec |
| 6:07PM |
3 |
Shut down one Zap line |
| 12:36PM |
2 |
Dial agent channel - busy |
| 12:17PM |
0 |
Monitoring audio on channel? |
| 12:00PM |
1 |
unable to hear voice with asterisk 1.4.15 |
| 10:51AM |
0 |
mwi with sip |
| 7:08AM |
0 |
[AGI 1.4] Why doesn't Asterisk complain? |
| 5:12AM |
3 |
Using x-lite -Call failed 404 not found |
| |
| Sunday January 27 2008 |
| Time | Replies | Subject |
|---|
| 11:28PM |
1 |
Toll-Free setup on Asterisk Server |
| 9:36PM |
6 |
Asterisk and MRTG, a little help please... |
| 4:27PM |
2 |
Maybe a little OT---USB Handset |
| 1:48PM |
1 |
Best "Console" phone? |
| 8:18AM |
1 |
rxfax does not work (anymore) |
| 6:57AM |
1 |
[AGI 1.4] C sample? |
| 3:41AM |
1 |
Jabber and Asterisk? |
| |
| Saturday January 26 2008 |
| Time | Replies | Subject |
|---|
| 9:28PM |
1 |
Loosing IAX/SIP user's registration with asterisk as no-root |
| 9:26PM |
0 |
Loosing IAX/SIP user's registration |
| 7:09PM |
5 |
autoprovision 200+ linksys phones setup |
| 4:29PM |
3 |
GotoIf() on Auto-Attendant |
| 3:43PM |
0 |
Upgrade fails, need system upgrade advice |
| 1:55PM |
1 |
CHANUNAVAIL |
| 11:20AM |
4 |
Asterisk on Dell PowerEdge 2950 |
| 8:49AM |
0 |
Extension Mobility with Asterisk and Cisco 79x1 phones |
| 7:07AM |
0 |
Avaya 9620 phone using Firmware 2.0.1.34 has working MWI lamp |
| 5:00AM |
0 |
Provide a proper link to download Libpri-1.4.3 |
| 2:03AM |
1 |
Zaptel for 1.6-beta1 |
| |
| Friday January 25 2008 |
| Time | Replies | Subject |
|---|
| 8:37PM |
1 |
Join me on Last.fm! |
| 7:53PM |
0 |
Script for seeding polycom phones with an extension directory |
| 6:41PM |
1 |
Problem with FollowMe |
| 5:50PM |
0 |
adding additional volume to console/dsp |
| 4:07PM |
2 |
Intercepting DTMF to initiate Voice Drop |
| 2:40PM |
1 |
Home use of asterisk |
| 1:33PM |
1 |
Disable IAX2 call path optimization |
| 12:03PM |
0 |
Adaptive jitterbuffer problem |
| 11:35AM |
2 |
Unprovisioned 7961 |
| 11:26AM |
2 |
Asterisk Billing |
| 11:04AM |
0 |
Error in sip channel when asterisk created call (SIP invite request) is forked |
| 10:48AM |
1 |
Need sample configuration files for sipura/linksys ata |
| 9:50AM |
0 |
What kind of configuration do I need to run Asterisk ? |
| 9:42AM |
0 |
Share accounts several AOR |
| 8:47AM |
1 |
Maximum Paging Group Size? |
| 3:54AM |
3 |
Finding difficulty in installing Asterisk |
| 3:06AM |
0 |
Gentilini, Paul is out of the office. |
| 12:35AM |
2 |
SPA3000 -- PSTN to VoIP |
| |
| Thursday January 24 2008 |
| Time | Replies | Subject |
|---|
| 10:38PM |
0 |
dial extension number |
| 9:28PM |
1 |
Patton SmartNode Help |
| 6:43PM |
3 |
Help: dtmf mode |
| 7:22AM |
1 |
two zaptel card |
| 4:57AM |
6 |
Your "favorite" Asterisk application. |
| |
| Wednesday January 23 2008 |
| Time | Replies | Subject |
|---|
| 11:38PM |
1 |
Call Parking with multiple lots |
| 11:07PM |
0 |
nokia e51 (Christian Lox) |
| 10:00PM |
2 |
Replacement for Allison |
| 9:04PM |
5 |
Snom 320 Lost Settings |
| 7:40PM |
0 |
app_txfax |
| 6:23PM |
8 |
Peak number of calls? |
| 5:55PM |
1 |
LDAP support |
| 5:23PM |
7 |
Asterisk scalability |
| 4:53PM |
3 |
asterisk optimalization |
| 4:31PM |
0 |
No more audio with 99777 SVN version in certain case |
| 9:23AM |
1 |
Realtime problem host='dynamic' in 1.2.26.1 |
| 8:39AM |
2 |
Modem bridging on Asterisk (no VoIP involved) |
| 8:03AM |
1 |
AsteriskIdeas.org :: Comment on submitted ideas |
| |
| Tuesday January 22 2008 |
| Time | Replies | Subject |
|---|
| 10:36PM |
3 |
Voicemail - is it possible to automatically use the extension being dialed from? |
| 6:56PM |
0 |
I am looking for an Asterisk subcontractor in New York City. |
| 6:37PM |
0 |
chan_sip deadlocks after some time |
| 6:28PM |
1 |
Echo in the outside call (E1) |
| 6:04PM |
1 |
AgentLogin by console |
| 5:50PM |
2 |
Difference between Asterisk and FreeSwitch |
| 4:25PM |
1 |
Followme |
| 12:46PM |
0 |
Caller id issue and Dial tone for sip phone on zero dialing |
| 12:41PM |
2 |
Free IAX / SIP Softphone with attended transfer |
| 11:11AM |
1 |
Custom Pickup and Transfer dial string |
| 9:29AM |
1 |
Discover Asterisk 1.4 :: Jitterbug, no, Jitterbuffers |
| 9:02AM |
0 |
So is anyone using 1.6 beta? What's the word? |
| 8:29AM |
0 |
Conference Hangup |
| 8:20AM |
2 |
TDM800P FXO problem incomming call |
| 4:06AM |
2 |
Asterisk crashed.. |
| 3:54AM |
1 |
Polycom-SIP response 500 |
| |
| Monday January 21 2008 |
| Time | Replies | Subject |
|---|
| 10:56PM |
1 |
FXS damaged at TDM22B |
| 4:59PM |
0 |
Aastra IP phone configuration generator |
| 2:39PM |
0 |
[asterisk-dev] Rgd Zaptel code for Asterisk |
| 2:12PM |
1 |
x100p & Asterisk 1.4.17 and Zaptel 1.4.7 |
| 2:04PM |
1 |
Monitoring calls on demand |
| 1:41PM |
0 |
MGCP Thomson, "early" transmit problem |
| 1:18PM |
1 |
call on hold--hokk flash---i want to know if i can disable it |
| 1:08PM |
4 |
Loop Break |
| 12:07PM |
1 |
Astmanproxy |
| 12:05PM |
1 |
How to prevent logging of some entries in CDR |
| 12:01PM |
2 |
Qsig link |
| 10:06AM |
6 |
[Fwd: Re: Large issue - having trouble diagnosing.] |
| 8:41AM |
0 |
calls get stuck in asterisk |
| 7:10AM |
1 |
asterisk-addons-1.6.0-beta1---Error |
| 4:04AM |
4 |
Large issue - having trouble diagnosing. |
| 3:39AM |
1 |
blf and misdn |
| 2:05AM |
1 |
Polycom 320 Issue |
| 1:14AM |
4 |
I am having a problem connecting my X-Lite to my Asterix box |
| |
| Sunday January 20 2008 |
| Time | Replies | Subject |
|---|
| 9:39PM |
0 |
HT-488 tutorial |
| 4:41PM |
0 |
30 sec delay before voice is heard |
| 4:10PM |
2 |
Asterisk connect to Cisco As5400 gateway |
| 3:40PM |
2 |
SIP <> GSM |
| 11:01AM |
2 |
IAX and NAT Transparency |
| 10:46AM |
6 |
IAX softphone |
| 9:33AM |
0 |
Paging and conferences/chan_alsa. |
| 8:57AM |
4 |
IP Phone support SIP and IAX |
| 2:32AM |
5 |
Calls Being Randomly Bridged |
| 12:28AM |
1 |
SIPAddHeader in .call file |
| |
| Saturday January 19 2008 |
| Time | Replies | Subject |
|---|
| 11:10PM |
0 |
nokia e51 |
| 10:34AM |
0 |
Call-out campaign variable problem |
| 9:21AM |
2 |
Nightly tarballs, would you use them? |
| 4:06AM |
3 |
New Polycom Provisioning Tool Released with BugFix |
| 12:47AM |
0 |
AMIProxyPal - AMI Proxy Project |
| |
| Friday January 18 2008 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
Asterisk 1.6.0-beta1 released |
| 10:41PM |
1 |
dtmf from Cell phones |
| 10:26PM |
0 |
asterisk chan_sip tuning |
| 8:57PM |
1 |
Looking for business-grade SIP Softphone |
| 8:37PM |
1 |
Probably a simple question. Dial a call. |
| 8:13PM |
2 |
SAY TIME + PHPAGI + Timezone |
| 7:37PM |
0 |
Maximum retries/no reply to our critical packet |
| 4:00PM |
2 |
OT: Call for beta testers (well... perhaps late Alpha). |
| 3:41PM |
1 |
R2-Unicall Asterisk as CPE and as CO |
| 2:33PM |
3 |
Accessing a MySQL database and using the same db for cdr |
| 1:32PM |
0 |
Asterisk and postgresql query |
| 12:29PM |
0 |
OT: To Admins: Missing DNS for list server |
| 12:21PM |
0 |
Advice on AMI and SIP (was: SIP) |
| 12:20PM |
0 |
SIP |
| 12:14PM |
1 |
Automatic call-out problem |
| 11:38AM |
0 |
Upgrading to Asterisk 1.4 :: Avoiding the hidden traps |
| 10:39AM |
0 |
VoIP Users Conference today at 1PM Friday EST |
| 9:27AM |
2 |
caller id issue for INDIA |
| 3:28AM |
0 |
Polycom Remotely Cancel Call Forward |
| 1:18AM |
4 |
Linksys PAP2 NA |
| 1:16AM |
0 |
Cisco 7910 Handsets: Skinny protocol? |
| |
| Thursday January 17 2008 |
| Time | Replies | Subject |
|---|
| 9:32PM |
0 |
IAX Trunk between two Asterisks |
| 8:55PM |
0 |
not understanding Cisco call manager connection for incoming calls |
| 8:37PM |
0 |
Paging Recording File |
| 8:15PM |
1 |
buffer-issue when piping live-streams into musiconhold |
| 8:00PM |
5 |
asterisk-1.2.26.tar.gz Thoughts? |
| 7:51PM |
0 |
PostgreSQL query results truncated 255 characters |
| 7:28PM |
2 |
SIP Proxy Issues |
| 6:54PM |
1 |
More voicemail cards needed... |
| 5:47PM |
0 |
Voicemail Callback |
| 5:36PM |
1 |
Device state of SIP doesn't change |
| 4:46PM |
0 |
Asterisk SVN mirror back up to date |
| 4:41PM |
1 |
Iax Encryption |
| 4:38PM |
1 |
modem through Zaptel/Digium? |
| 4:25PM |
0 |
sip channel - redirection - which context is used |
| 3:33PM |
0 |
Channels ID / Soft Hang Up |
| 2:10PM |
3 |
AEL includes? |
| 1:09PM |
1 |
Zaptel timing on TE405P |
| 12:13PM |
0 |
Asterisk Meetme & MeetMeAdmin cmd info-use |
| 11:25AM |
4 |
Asterisk desktop tools for OS X |
| 10:40AM |
0 |
callerid on atxfer |
| 10:23AM |
3 |
Single T1 with DIDs |
| 6:26AM |
0 |
Incoming calls on PSTN trunk not disconnected (bsnl, india) |
| 6:19AM |
0 |
FXO Module for the cPCI platform |
| 5:33AM |
1 |
asterisk-users Digest, Vol 42, Issue 51 |
| 4:27AM |
6 |
Voicemail systems- flow charts, digit/key cards, etc |
| 1:34AM |
0 |
Asterisk on ClarkConnect |
| 12:54AM |
1 |
IMAP client in asterisk not trying to contact IMAP server |
| 12:42AM |
2 |
Problem with a channel |
| 12:39AM |
2 |
Anyone Using a Dell PowerEdge T105 in Production |
| 12:11AM |
1 |
AddQueueMember and Flash Operator Panel |
| |
| Wednesday January 16 2008 |
| Time | Replies | Subject |
|---|
| 11:52PM |
3 |
HDLC errors |
| 8:24PM |
1 |
Asterisk 1.4.17 and RXFAX via T38 |
| 7:11PM |
2 |
asterisk to mysql database! |
| 5:25PM |
2 |
[IAX] Up-to-date list of soft- and hardphones? |
| 4:02PM |
1 |
Can DB() use SQLite instead of BerkeleyDB? |
| 3:25PM |
2 |
Zap Issues |
| 2:42PM |
2 |
Voicemail consultation problem |
| 2:29PM |
0 |
Problem with TDM400P |
| 2:06PM |
1 |
Asterisk Now Beta 6 and CISCO IP 7910 |
| 1:47PM |
1 |
Does host accept dns or ddns? |
| 1:46PM |
1 |
IAX Trunk between two Asterisks: Authority, and Call Rejected |
| 1:42PM |
1 |
Backup Route |
| 12:11PM |
0 |
Dualphone "LAN" SIP/DECT phones |
| 11:51AM |
1 |
Unable to dial _99XXXXXXXX |
| 10:55AM |
4 |
Unable to open master device '/dev/zap/ctl' |
| 10:39AM |
2 |
Difference between TE121 and TE122 |
| 10:18AM |
1 |
bad sound quality after Redirect |
| 8:18AM |
3 |
volume problem |
| 8:00AM |
1 |
SVN Server Issue? |
| 6:44AM |
0 |
help Unable to dial _99XXXXXXXX |
| |
| Tuesday January 15 2008 |
| Time | Replies | Subject |
|---|
| 10:20PM |
1 |
Channel fallback |
| 9:54PM |
2 |
WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101' |
| 8:32PM |
1 |
Attended transfers manager or phone |
| 8:12PM |
1 |
inbound Audio problems probably not NAT related? |
| 7:00PM |
1 |
asterisk 1.4 context |
| 6:55PM |
1 |
Record calls then send them to users voicemail |
| 6:42PM |
1 |
Heartbeat |
| 5:32PM |
3 |
Interrupt the swift text |
| 3:41PM |
6 |
Discover Asterisk 1.4 :: SIP Subscriptions |
| 2:40PM |
0 |
sip channel error - extension pattern matching problem |
| 1:43PM |
0 |
busy/congestion random |
| 1:14PM |
1 |
cisco ip phne 7911G with asterisk |
| 1:06PM |
1 |
Playing DTMF tones down a channel |
| 1:01PM |
1 |
SIP Reason |
| 11:42AM |
1 |
Fax machine detect |
| 11:09AM |
1 |
Console app |
| 9:53AM |
3 |
Meetme recording |
| 7:17AM |
0 |
pickupchan without bristuffed version? |
| 5:02AM |
2 |
Park() help, extension not heard |
| 12:03AM |
0 |
Zaptel 1.2.23 and 1.4.8 released |
| |
| Monday January 14 2008 |
| Time | Replies | Subject |
|---|
| 11:59PM |
0 |
SVN servers down for maintenance |
| 11:23PM |
3 |
Asterisk 1.4.17 crashing more |
| 10:18PM |
2 |
CID blocking ... |
| 10:09PM |
2 |
G.729 pre-compiled binaries and Asterisk 1.2.x. |
| 10:04PM |
2 |
app_voicemail for spanish |
| 8:43PM |
0 |
Transfer/Speed-Dial |
| 5:52PM |
1 |
Voicemail check |
| 5:25PM |
0 |
[asterisk-dev] Unstable releases lately |
| 4:51PM |
4 |
Verficar VoiceMail |
| 4:09PM |
1 |
Asterisk 1.4 Call Recording |
| 3:47PM |
0 |
Temporary Service - Dominican Republic DID |
| 3:18PM |
0 |
Meetme Record Format |
| 3:08PM |
0 |
Help needed for Fax2Email with Welltech FXO 3804 |
| 2:58PM |
4 |
OT: reverse DNS error for lists.digium.com |
| 2:15PM |
1 |
Different ringing tones ... |
| 12:44PM |
1 |
Video Call and Asterisk |
| 12:42PM |
7 |
GSM SIM Cards and Digium, or GSM SIM Adaptor |
| 12:04PM |
2 |
g729 codec - simultaneous calls |
| 11:55AM |
1 |
AGISTATUS is SUCCESS even though my PHP script returned -1 |
| 8:47AM |
1 |
State of the application chan_spy |
| 8:38AM |
1 |
[SOLVED + EXPLANATION]: Strange ISDN-problem with incoming calls out of the same city |
| 5:42AM |
0 |
Call parking |
| 2:45AM |
0 |
Aastra Venture |
| |
| Sunday January 13 2008 |
| Time | Replies | Subject |
|---|
| 9:00PM |
1 |
ProxyPal for AMI Proxy Development |
| 5:33PM |
2 |
problems with zaptel and Udev |
| 5:17PM |
2 |
Question about queues and the definition and agents |
| 4:22PM |
0 |
Adtran 750 and E&M Wink |
| 2:20PM |
0 |
Soundcard necessary on an asterisk server to get output of playback()?? -> Next step |
| 12:11AM |
2 |
Packet2Packet bridging occurring when not wanted |
| |
| Saturday January 12 2008 |
| Time | Replies | Subject |
|---|
| 7:09PM |
2 |
Asterisk RFC2833 to SIP INFO DTMF conversion erros. |
| 12:49PM |
1 |
ISDN channels not properly released after call |
| 12:05PM |
2 |
Perl-AGI process |
| 10:50AM |
2 |
My latest MFC/R2 update with asterisk-1.4.17 |
| 10:02AM |
1 |
Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration! |
| 5:11AM |
1 |
MFC/R2 Signaling configuration |
| 4:03AM |
0 |
zaptel digit problem |
| |
| Friday January 11 2008 |
| Time | Replies | Subject |
|---|
| 10:08PM |
0 |
MONITOR CMD() |
| 9:15PM |
1 |
Fwd: Trying to build MFC/R2 |
| 5:54PM |
0 |
Authenticate problems (Extensions) |
| 5:50PM |
0 |
Deadlock of asterisk on app_system |
| 5:32PM |
1 |
MRCP Asterisk Integration |
| 4:48PM |
0 |
Dialplan flow on device state change |
| 4:38PM |
2 |
Question about queues and the definition of agents |
| 4:21PM |
1 |
Soundcard necessary on an asterisk server toget output of playback()?? -> Next step |
| 4:20PM |
0 |
[OT] Call for speakers: BOB 2.0 |
| 1:41PM |
1 |
Dealy while taking |
| 1:04PM |
0 |
Is rfc4662 (SIP Resource Lists notification) support planned ? |
| 12:22PM |
0 |
15% Off from New Cyber-Telecom.net Website |
| 12:15PM |
0 |
Lest we forget: Friday 12 Noon EST - VoIP Users Conference |
| 11:57AM |
1 |
Developing Help |
| 10:32AM |
0 |
OT - Where do most email2fax errors come from ? |
| 9:56AM |
1 |
interconnecting an asterisk server with an old alcatel PBX through a Digium B410P |
| 5:55AM |
1 |
PRI Down but zaptel lets calls through |
| 5:27AM |
5 |
Congestion/Forbidden issue with new carrier |
| |
| Thursday January 10 2008 |
| Time | Replies | Subject |
|---|
| 11:46PM |
1 |
Multiple fax extensions |
| 10:52PM |
1 |
Asterisk Realtime unixODBC timeout? |
| 8:54PM |
1 |
Sip calls drop one leg after about 2 minutes |
| 3:48PM |
1 |
WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"unknown" <sip:unknown@xxx.xxx.xxx.xxx> |
| 1:19PM |
0 |
problem about TDM400P ringback detection |
| 10:22AM |
3 |
OT - Is handover included in DECT GAP ? |
| 8:35AM |
1 |
Using Asterisk as an Fax-Gateway for analog Fax devices |
| 8:29AM |
0 |
Kirk and asterisk |
| 5:42AM |
0 |
forward call intended for another domain |
| 3:57AM |
8 |
IEEE 802.1x capable sip phones |
| 3:23AM |
4 |
Asterisk 1.4 and ISDN-BRI support |
| 2:33AM |
1 |
OT: Traffic Shaping |
| 12:37AM |
3 |
Two Asterisk Boxes Playing Together |
| |
| Wednesday January 9 2008 |
| Time | Replies | Subject |
|---|
| 11:28PM |
0 |
FXOTUNE update |
| 9:03PM |
0 |
IAXy ringing |
| 8:41PM |
2 |
Polycom 550 IP SoundStation Fuzzy Voice Quality |
| 8:06PM |
0 |
Subscriptions, Firewalls, 489 "Bad Event" and Bug 7608 |
| 6:28PM |
2 |
Intercom & Paging with Polycoms |
| 6:09PM |
3 |
WaitExten and Macros |
| 5:32PM |
0 |
Broken calls |
| 3:38PM |
2 |
Busy notification with call limiting by GROUP_COUNT() |
| 3:15PM |
2 |
Zaptel FXS Cards - Station Distance |
| 3:11PM |
0 |
[asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers |
| 12:40PM |
1 |
Newbie: confusion with the new FXO/FXS card |
| 9:46AM |
0 |
remote Snom 360: no voice passing thru |
| 7:54AM |
2 |
Set CDR userfield in a realtime dialplan |
| 5:55AM |
1 |
Register source port |
| 5:01AM |
0 |
conferencing help |
| 1:46AM |
1 |
Dialplan Recordings |
| 1:31AM |
1 |
Help! channel_find_deadlocked: Avoided initial deadlock for ... |
| 1:19AM |
0 |
txfax_exec: Transmission loop error |
| |
| Tuesday January 8 2008 |
| Time | Replies | Subject |
|---|
| 10:26PM |
2 |
Linksys SPA-9xx Audio Issues |
| 10:19PM |
0 |
get_data |
| 9:45PM |
0 |
debugging bluetooth communication using chan_mobile |
| 9:37PM |
2 |
CallerID Number incorrect in SIP packet |
| 8:43PM |
2 |
Simultaneous Callback?! |
| 8:26PM |
3 |
Is it possible to use spandsp and patton to do fax2mail ? |
| 7:29PM |
1 |
What's the best ztdummy? |
| 6:06PM |
6 |
[Zaptel] Checking that TDM card works? |
| 1:52PM |
4 |
Bugs?? |
| 1:11PM |
0 |
Prevent Asterisk from rebuiling DTMF tones |
| 12:46PM |
0 |
Asterisk Nokia |
| 12:23PM |
2 |
Limiting number of simultaneous calls in E1 line |
| 12:01PM |
2 |
disable call waiting by default |
| 12:00PM |
0 |
communicating SMS messages in asterisk |
| 11:31AM |
0 |
chan_h323 and asterisk 1.2 |
| 9:11AM |
0 |
no outoging calls with B410P |
| 8:20AM |
3 |
HPEC |
| 6:41AM |
1 |
Early media support for Asterisk behind NAT |
| 6:08AM |
2 |
Distorted audio over Eicon Diva Server BRI |
| 5:19AM |
3 |
app_rxfax.c and app_txxfax.c where? |
| 4:41AM |
2 |
help need |
| 12:11AM |
2 |
:POSSIBLE SPAM: conferencing help |
| |
| Monday January 7 2008 |
| Time | Replies | Subject |
|---|
| 11:51PM |
1 |
Background Noise Elimination |
| 10:28PM |
0 |
asterisk-users] Increase Volume - SIP |
| 10:14PM |
0 |
chan_mobile and W300i |
| 8:04PM |
1 |
GotoIf() help |
| 7:21PM |
3 |
asterisk CLI and no such command "stop" |
| 7:02PM |
0 |
[Asterisk 1.2 + TDM FXO] Incoming call not detected |
| 6:59PM |
1 |
Media gateways and video |
| 4:06PM |
0 |
service provider connection problem |
| 3:48PM |
1 |
pickup application failed |
| 3:34PM |
1 |
Multi-SPAN (4xE1) Zap Group (Outbound) |
| 3:17PM |
2 |
FWD and IPCall |
| 2:26PM |
3 |
How to check if a SIP phone is forwarded without ringing it ? |
| 12:43PM |
1 |
extension.conf with mysql |
| 11:57AM |
1 |
Strange migration problems from asterisk 1.2.13 to 1.4.10, dtmf related? |
| 11:21AM |
0 |
Presentation Restricted h.323-SIP issue |
| 10:53AM |
1 |
no outgoing calls with Digium B410P |
| 8:34AM |
2 |
Increase Volume - SIP |
| 7:06AM |
2 |
zaptel programming |
| 2:14AM |
2 |
Change Default Voicemail Message |
| |
| Sunday January 6 2008 |
| Time | Replies | Subject |
|---|
| 8:02PM |
0 |
Asterisk High Availability and Clustering |
| 6:12PM |
7 |
Which IP Phone is really the best? |
| 10:12AM |
0 |
New site for feature wish-list: Asteriskideas.org |
| 9:20AM |
1 |
[FreeBSD 6.2] Error compiling Zaptel from Ports? |
| |
| Saturday January 5 2008 |
| Time | Replies | Subject |
|---|
| 11:20PM |
0 |
Zap with SIP |
| 10:55PM |
0 |
Newbie Q: Good link to configuring NAT with Sipura ATA's & hardphones |
| 7:50PM |
7 |
asterisk on Hp servers |
| 6:36PM |
6 |
Detailed Instructions |
| 10:40AM |
4 |
iP0020 Phone busy signal all the time. |
| 10:04AM |
1 |
G729A Install Problems |
| 9:45AM |
2 |
ASTERISK cd-rom |
| 5:45AM |
1 |
how to block spammer calls |
| 12:06AM |
1 |
G723 Codec and Asterisk |
| 12:03AM |
1 |
GotoIf: OR, AND |
| |
| Friday January 4 2008 |
| Time | Replies | Subject |
|---|
| 11:52PM |
2 |
Conditional Dial |
| 11:10PM |
1 |
asterisk as sip server |
| 10:45PM |
3 |
b2bua |
| 10:26PM |
1 |
Remote hold on PRI |
| 9:48PM |
1 |
VOIP Provider wooes |
| 9:41PM |
1 |
Cisco 79xx XML services |
| 6:52PM |
1 |
Polycom IP4000 - Device does not match ACL |
| 3:28PM |
0 |
2 firewalls, different INVITES |
| 3:00PM |
2 |
x100p wcfxo hangup on outgoing calss |
| 2:48PM |
1 |
Asterisk content @ OSCON 2008? |
| 10:48AM |
3 |
Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC |
| 9:58AM |
1 |
Unable to forward call on SIP channel after SIP response 302 Moved Temporarily |
| 8:11AM |
2 |
Cisco 7941G-GE with Asterisk and CTPSEP odyssee |
| 7:35AM |
2 |
Agents and AddQueueMember |
| 5:45AM |
1 |
Registration from sip failed for ACL error (permit/deny) |
| 2:19AM |
2 |
automatic call marking an extension |
| 1:36AM |
3 |
Using Asterisc for Taking Calls for Radio |
| |
| Thursday January 3 2008 |
| Time | Replies | Subject |
|---|
| 9:08PM |
2 |
OT - GEOPRIV and location based SIP services |
| 8:39PM |
4 |
1.4.17 - Breaks park announce? |
| 8:15PM |
3 |
Unable to retrieve my voice mail ... (password incorrect) |
| 7:44PM |
1 |
A thougt |
| 5:31PM |
1 |
Bad Link on Website... |
| 4:52PM |
1 |
Right timing for a queue call |
| 3:08PM |
2 |
HFC-S zap channels always busy |
| 1:28PM |
5 |
GSM Gateway behind SIP ATA? |
| |
| Wednesday January 2 2008 |
| Time | Replies | Subject |
|---|
| 11:10PM |
3 |
is Power fail transfer possible with asterisk? |
| 9:57PM |
0 |
AST-2008-001: Crash from transfer using BYE with Also header |
| 9:39PM |
0 |
Asterisk 1.4.17 Released |
| 8:11PM |
5 |
Missing "zap" command in Asterisk 1.4.16 |
| 8:00PM |
1 |
How to stop the update of astdb? |
| 7:41PM |
3 |
1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken |
| 7:14PM |
0 |
Toroid: open source middleware |
| 5:34PM |
3 |
Polycom VLAN |
| 4:49PM |
3 |
AGI stream file |
| 4:20PM |
5 |
Incoming Calls |
| 3:19PM |
1 |
Asterisk E1/T1 Card configuration |
| 1:38PM |
4 |
Lamps on Snom phones |
| 11:27AM |
2 |
Invalid extensions |
| 10:45AM |
2 |
Asterisk dialplan date and time operations |
| 10:29AM |
2 |
auto dial and IVR |
| 7:33AM |
7 |
Two Asterisks behind NAT and need to link them using IAX trunk |
| 6:23AM |
2 |
Trixbox and mail2fax |
| 2:06AM |
1 |
Password protect a queue from callers? |
| |
| Tuesday January 1 2008 |
| Time | Replies | Subject |
|---|
| 6:24PM |
4 |
zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init |
| 5:08PM |
1 |
With rtcachefriends=yes, when do realtime changes take effect? |
| 4:33PM |
2 |
(no subject) |
| 3:38PM |
0 |
Asterisk + SIP + cisco phone confrance problem |
| 10:06AM |
3 |
[1.4 + FreeBSD 6.2] Playing WAV PCM file? |