| Wednesday November 30 2005 |
| Time | Replies | Subject |
|---|
| 11:03PM |
2 |
1.2.0 PRI dropping calls occasionally... |
| 10:34PM |
1 |
two sip phone communication using asterisk server |
| 7:51PM |
1 |
Call transfer with voicemail password |
| 6:49PM |
1 |
format |
| 6:38PM |
5 |
Asterisk cluster and astdb |
| 5:32PM |
1 |
sip to sip, not comunication |
| 4:41PM |
1 |
polycom backlight? |
| 3:22PM |
1 |
FW: CDR issues |
| 3:17PM |
1 |
Queues and Servicelevel |
| 2:48PM |
0 |
CDR issues |
| 2:15PM |
3 |
How to exit from Asterisk console. |
| 1:46PM |
1 |
Disposition failed in Asterisk-1.2.0-stable |
| 12:38PM |
1 |
problem with zaptel 1.2.0 and pulse dialing |
| 12:38PM |
1 |
Queue calls... |
| 11:25AM |
3 |
Snom 320s and the hint priority |
| 10:45AM |
5 |
hierarchical VoIP system |
| 10:34AM |
1 |
pbx or asterisk? |
| 10:26AM |
2 |
Sipura SPA-3000 & SPA-2002 - Unable to dial *99 |
| 10:26AM |
0 |
asterisk starting problem. Warning 2224 (app_capiCD.so) |
| 9:25AM |
2 |
Disable IAX2 native bridging / Monitor() app |
| 9:08AM |
0 |
Debian Sarge + Asterisk 1.2 + chan_mISDN not starting |
| 8:46AM |
0 |
Got SIP response 400 "Invalid Subscription-State" |
| 8:43AM |
3 |
IAX Service providers in Australia for unlimited inbound |
| 7:57AM |
0 |
Transfer call error |
| 7:44AM |
0 |
Astfax problem |
| 7:29AM |
1 |
Astfax with current CVS |
| 7:07AM |
2 |
MeetMe with the V (video) option |
| 4:15AM |
1 |
Tone busy in zaptel |
| 3:38AM |
1 |
Compiling Asterisk 1.2 from Source on Debian Sarge - Problems |
| 3:33AM |
1 |
Help transfer call |
| 3:11AM |
0 |
BRIStuff and PRI |
| 2:25AM |
1 |
IP GSM Gateway is giving uncomplete SIP signalization to PRI interface - can I somehow avoid that in Asterisk ? |
| 1:31AM |
1 |
TE210P & Linux SMP |
| 12:16AM |
0 |
Page() application examples. |
| |
| Tuesday November 29 2005 |
| Time | Replies | Subject |
|---|
| 11:49PM |
0 |
werror compiling libpri |
| 9:20PM |
0 |
ASTCC not completed |
| 8:40PM |
2 |
Active SIP Peer? |
| 5:53PM |
2 |
zapata directory not found in svn . |
| 5:48PM |
1 |
Problem with IAX2 jitterbuffer and DTMF reception with 1.2.0 |
| 4:55PM |
8 |
Static on inside end of conversation |
| 4:41PM |
3 |
Pasting phrases together.... |
| 4:02PM |
0 |
How to disable SIP Options methods on asterisk |
| 3:59PM |
0 |
Comas versus pipe command in AgetCallBackLogin |
| 3:51PM |
0 |
All agent calls going to powered down agent extension? |
| 2:30PM |
2 |
Cisco CP-7940G drops time from display |
| 2:01PM |
0 |
Digital Cellsocket |
| 1:49PM |
0 |
RE: Asterisk-Users Digest, Vol 16, Issue 232 |
| 1:41PM |
1 |
route call based on codec? (g723 gets message, g729 goes to conf connection) |
| 1:25PM |
6 |
Voicemail and sendmail |
| 1:17PM |
0 |
Question on Monitoring and Transferring... |
| 11:57AM |
1 |
Queuelog |
| 10:54AM |
0 |
cause 17 - User busy ? |
| 10:36AM |
1 |
Voicepulse not accepting new customers. (FCC E911) |
| 10:25AM |
0 |
ResetCDR with CDR |
| 10:12AM |
0 |
Monitoring Zaptel Errors |
| 9:15AM |
3 |
Caller ID Block (*67) |
| 8:48AM |
0 |
moh on optipoint400 |
| 7:56AM |
1 |
qozap.o error |
| 7:25AM |
0 |
FW: Fax sending problems |
| 5:05AM |
6 |
DIALSTATUS |
| 4:58AM |
1 |
Problems with auto dialout |
| 3:59AM |
2 |
TDM400 revisions problem: Rev J not working!! |
| 3:59AM |
1 |
Load spikes with 1.0.10 |
| 3:41AM |
5 |
asterisk@home isdn |
| 3:38AM |
0 |
setting variables in a .call file - how? |
| 2:38AM |
0 |
Hangup after 18 sec on PRI channel |
| 2:24AM |
0 |
SNOM Phones MWI, Hold & Retrieve buttons notworking with Asterisk v1.2 |
| 1:38AM |
0 |
Problem with Ext calling |
| |
| Monday November 28 2005 |
| Time | Replies | Subject |
|---|
| 11:14PM |
1 |
VegaStream |
| 9:27PM |
1 |
Digitmap problems |
| 7:17PM |
2 |
delayed pickup in ZAP interface and issue with hang up-s (fwd) |
| 7:00PM |
1 |
SIP Trunk in incoming ? it's possible ? |
| 6:52PM |
0 |
Avaya 4620SW - SIP response 400 |
| 5:59PM |
1 |
Channels not showing up in Asterisk |
| 5:59PM |
1 |
Is a BUG ? Hints and incominglimit |
| 5:51PM |
0 |
tranfered calls audible but low volume |
| 5:44PM |
2 |
SIP rapid INVITE re-transmission: bug, or config problem? |
| 4:37PM |
0 |
Philippines asterisk mailing list / yahoo groups! (PINOY AKO! PINOY TAYO!) |
| 4:24PM |
3 |
Newbie question on 1.2 extension configs |
| 4:01PM |
7 |
US e911 reminder |
| 3:30PM |
2 |
Accepting Inbound SIP Connections |
| 3:19PM |
1 |
small office setup |
| 2:53PM |
1 |
Comedian Voicemail? PROBLEMS? |
| 2:38PM |
0 |
Interface Cards that support QSIG |
| 2:32PM |
8 |
SNOM and 1.0.9 |
| 2:24PM |
1 |
misdn, busy detection |
| 2:10PM |
0 |
Avaya 4620SW Invalid Subscription-State - Issue |
| 1:26PM |
1 |
DTMF errors |
| 1:24PM |
2 |
cdr_manager.conf |
| 1:12PM |
3 |
Wrong usage of in the extension? |
| 1:06PM |
1 |
Help connecting Avaya S8700 and Asterisk through H.323 trunk |
| 12:41PM |
0 |
AGI + CDR |
| 12:08PM |
1 |
Problem with pulses dialing on asterisk 1.2 |
| 12:00PM |
0 |
Asterisk project converts to Subversion version control system |
| 11:49AM |
1 |
Problem with Internet connection |
| 11:48AM |
1 |
SNOM Phones MWI, Hold & Retrieve buttons not working with Asterisk v1.2 |
| 11:42AM |
1 |
PROGRESS with cause code 31 received |
| 11:25AM |
1 |
Emailed voicemail messages not being deleted |
| 10:15AM |
0 |
Call progress from sip gsm gateway to pri interface - doesn't get through |
| 9:55AM |
0 |
how to stop ringing while talking |
| 9:47AM |
1 |
IAX jitterbuffer and trunking settings between 1.0.9 and 1.2 |
| 9:27AM |
0 |
How does DTMF get sent over a PRI in Asterisk |
| 9:19AM |
0 |
Trunk SIP howto ? |
| 8:46AM |
0 |
Can 'spandsp' ack as an intermediary between a fax machine and a TDM400P? |
| 8:39AM |
1 |
Download Ringtones for 7960's? |
| 8:22AM |
1 |
AGI script always returning 0 |
| 8:20AM |
0 |
Realtime Extensions Problem |
| 7:30AM |
0 |
New mailing list: AstCallCenters |
| 7:15AM |
0 |
Problem forwarding zap to sip |
| 6:35AM |
0 |
troubles with voicemail |
| 5:53AM |
2 |
Upgrade Cisco 7910 with Asterisk ? |
| 5:31AM |
0 |
A rather big setup. |
| 5:00AM |
2 |
Legacy PBX integration problem |
| 3:28AM |
3 |
Problem with ADIT 600 and FXO configuration |
| 12:42AM |
11 |
SIP tapi |
| 12:16AM |
1 |
CDR Accounting PRoblem |
| |
| Sunday November 27 2005 |
| Time | Replies | Subject |
|---|
| 11:25PM |
0 |
PBX Manager beta1 release |
| 9:05PM |
0 |
"CPE does not support Call Waiting Caller*ID"? |
| 6:50PM |
2 |
Does it mean I was blocked by STUN? |
| 6:30PM |
0 |
Intel G729 Codec Install error on asterisk@Home2.0 |
| 6:28PM |
2 |
New Asterisk user - Dumb Questions |
| 5:59PM |
0 |
chan_bluetooth background scanner |
| 5:36PM |
0 |
Script to update externip for A@H/AMP [was Re: SIP Extension behind NAT, Asterisk on NAT (DMZ)] |
| 4:40PM |
8 |
Zaptel errors on Debian |
| 3:13PM |
1 |
Asterisk 1.2 and Athlon64 platforms |
| 12:56PM |
1 |
Asterisk cdr mysql |
| 12:40PM |
0 |
Failover with 1.2 Dial applications |
| 11:34AM |
0 |
chan_bluetooth with Plantronics Heaset (some good stuff) |
| 11:11AM |
0 |
ANNOUNCEMENT: Asterisk-Java 0.2 released |
| 9:50AM |
1 |
A question about transfering calls |
| 9:02AM |
3 |
Reboot stops TD400P cards from outgoing calls until first incoming call arrives |
| 8:17AM |
2 |
Voicepulse Open Access status? |
| 6:43AM |
1 |
IAx/g729 client for MAC |
| 6:17AM |
0 |
zaptel 1.2.0 and correct settings in zapata.conf for Germany |
| 5:26AM |
0 |
trunk not registering -newbie |
| 5:02AM |
0 |
calling to mgcp device |
| 4:57AM |
7 |
Dialplan help |
| 4:56AM |
1 |
Intel G729 Codec Install error on asterisk@Home 2.0 |
| 12:21AM |
2 |
a2billing / php agi debugging |
| |
| Saturday November 26 2005 |
| Time | Replies | Subject |
|---|
| 10:20PM |
0 |
1.2 page command |
| 10:06PM |
0 |
Possible Bug in Asterisk 1.2.0 with Queues and MOH |
| 7:53PM |
4 |
OH323 channel in asterisk 1.2 ...Ouch ... error while writing audio data! |
| 6:14PM |
1 |
Trouble with Channels |
| 5:46PM |
8 |
Would DECT cordless phones work with Asterisk and VOIP? |
| 4:09PM |
0 |
Voice recognition ... |
| 3:32PM |
0 |
cid_rewrite and 411.com update |
| 3:25PM |
0 |
Plantronics Bluetooth Headset help.. |
| 2:00PM |
1 |
Anyone using Parlay VoXip SIP Gateway with Asterisk ? |
| 1:43PM |
3 |
IAXmodem fax polling |
| 1:25PM |
3 |
Context mix-up |
| 12:25PM |
0 |
Re: SIP Extension behind NAT, Asterisk on NAT (DMZ) |
| 11:49AM |
0 |
H standard extension |
| 11:26AM |
1 |
WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data. |
| 10:48AM |
4 |
Small office with all employee's offsite |
| 10:26AM |
1 |
chan_bluetooth max simultaneous channels |
| 8:56AM |
2 |
cdr enhancement with 'rate' column |
| 8:11AM |
4 |
problems with callback.agi script |
| 7:05AM |
1 |
callback.agi script |
| 7:02AM |
1 |
Asterisk dial plan |
| 6:19AM |
1 |
Camping-on-busy |
| 6:10AM |
1 |
Asterisk and Cisco Phone 7910 |
| 2:10AM |
0 |
optiPoint 410/420 SIP firmware for Asterisk |
| 12:52AM |
1 |
SIP Forward |
| |
| Friday November 25 2005 |
| Time | Replies | Subject |
|---|
| 11:10PM |
3 |
configure intel modems..... |
| 7:44PM |
2 |
Problem about outgoing calls with verizon. |
| 7:20PM |
0 |
Music On Hold Crashing? |
| 6:36PM |
0 |
RE: [Asterisk-biz] GSM Gateway for £60 |
| 5:59PM |
0 |
Polycomm 500 not saving web changes any more |
| 5:55PM |
1 |
Transfering to a bridged call |
| 4:35PM |
3 |
global numbering plans |
| 4:13PM |
1 |
can I have T1 and E1 on the same TE406 card? |
| 2:53PM |
2 |
Distinctive Ring Detection not working |
| 1:57PM |
2 |
misdn, 2x HFC cards |
| 1:45PM |
2 |
Polycom IP50X Park Softkey |
| 1:25PM |
1 |
Asterisk 1.2 stability problem. |
| 1:21PM |
5 |
Sangoma problems!? |
| 1:06PM |
2 |
Asterisk callback system |
| 12:48PM |
0 |
Asterisk and Siemens HiPath 3750 issues |
| 12:36PM |
2 |
Narrowing RTP port range |
| 12:31PM |
3 |
Truncated CDR records |
| 11:41AM |
0 |
Re: think people dont help that easily |
| 11:16AM |
1 |
Bristuff: qozap.o error |
| 11:08AM |
1 |
CallerID not passing through to Polycom 500 (SOLVED, sort of) |
| 11:01AM |
0 |
A2Billing questions are off topic for this list |
| 10:48AM |
0 |
smsq sending 7 at a time ? |
| 10:24AM |
1 |
Distinctive ring? |
| 10:17AM |
2 |
"Local Directory" feature on Polycom Soundpo int 501s |
| 9:55AM |
1 |
"Local Directory" feature on Polycom Soundpoint 501s |
| 9:47AM |
1 |
Dialplan pattern match discrepancy |
| 9:44AM |
2 |
Help with 2billing please. |
| 9:23AM |
1 |
Problem with SIP register |
| 9:02AM |
0 |
Manager log |
| 8:58AM |
0 |
is it possible to force faxdetect / disable echo cancellation for a given extension? |
| 8:39AM |
1 |
speex & ilbc |
| 8:21AM |
3 |
How to initiate a call from a web page? |
| 8:19AM |
0 |
Second TDM22B board install issue |
| 8:07AM |
0 |
SIP response 484 "Address Incomplete" incorrectly handled |
| 7:46AM |
0 |
authentication question |
| 7:40AM |
3 |
Philippines Asterisk users, anyone? |
| 7:03AM |
0 |
busy channels |
| 6:59AM |
4 |
Siemens OptiPoint 4xx |
| 6:22AM |
3 |
Looking for Info on Asterisk scripting |
| 5:50AM |
1 |
authentication fails to provider after upgrading to 1.2.0 |
| 5:33AM |
2 |
Command line |
| 3:22AM |
3 |
sound problem, please help! |
| 2:36AM |
1 |
TE411P |
| 2:29AM |
1 |
Really lightweight itemised billing |
| 2:08AM |
0 |
help need for the configuration |
| 1:51AM |
0 |
has someone zaphfc with xenomai working? |
| 1:18AM |
0 |
Re: Bad quality... |
| |
| Thursday November 24 2005 |
| Time | Replies | Subject |
|---|
| 10:53PM |
20 |
NewBie to Ast Server, help need for the configuration |
| 10:50PM |
1 |
Asterisk and Japanese Caller ID |
| 9:02PM |
0 |
Recommended PCI latency time? |
| 7:31PM |
0 |
Asterisk + SER problem,ua cann't hangup |
| 5:45PM |
4 |
Pros and Cons of T1/E1 cards |
| 5:10PM |
1 |
harry's project |
| 5:04PM |
1 |
Preventing long-distance call forwarding |
| 4:29PM |
2 |
chan_misdn crashes : init_stack: success but entitylist not empty |
| 3:33PM |
1 |
Bad quality |
| 2:41PM |
1 |
Newbie requesting help! |
| 2:00PM |
1 |
(AMUG) Asterisk Montreal User Group today's meeting |
| 1:48PM |
5 |
Linksys SPA-841 Disconnects from Asterisk |
| 11:58AM |
0 |
SIP softphone with subscription/hint support? |
| 11:35AM |
1 |
Send fax using PRI connection to TE405P |
| 10:17AM |
0 |
Re: Asterisk not picking up calls. |
| 10:05AM |
1 |
Re: Queue Callback - SOLVED |
| 9:48AM |
0 |
Voicemail notifications alwats sent as asterisk@hostname |
| 9:28AM |
0 |
H323 to H323 calls problem |
| 9:18AM |
2 |
CallerID not passing through to Polycom 500 |
| 7:55AM |
1 |
jittering with Iax2 and Meetme on Asterisk 1.2.0 |
| 7:45AM |
0 |
Don't Outgoing call with Zap |
| 6:20AM |
2 |
Lag in speech |
| 6:13AM |
0 |
GUI and Asterisk Realtime |
| 6:06AM |
1 |
HELP! on disconnecting stale calls. |
| 5:14AM |
0 |
Compatibilidade com PABX Intelbras |
| 4:35AM |
0 |
Sip dosenot fall to default 's' , STRANGE? |
| 4:01AM |
1 |
Fax sending problems |
| 3:28AM |
3 |
[Asterrik-Users] Bristuff for Asterisk 1.2 error |
| 3:26AM |
4 |
PRI problems again - What should I do ? |
| 3:01AM |
0 |
1.2.0 using 1G of RAM |
| 2:21AM |
0 |
hint problem |
| 1:58AM |
0 |
pstn-destination beeing cut in logs and cdrs |
| 1:08AM |
1 |
Calling Asterisk PABX in "anonymous" mode... |
| 12:21AM |
7 |
Looking for Windows based Asterisk |
| 12:08AM |
0 |
Voicemail email format, please help! |
| 12:01AM |
1 |
zaptel 1.2.0 on (Tettnang) |
| |
| Wednesday November 23 2005 |
| Time | Replies | Subject |
|---|
| 11:59PM |
0 |
MTP Requirements for getting * talking to CCM for Voicemail |
| 11:06PM |
2 |
Loss of Registration for SIP Trunks |
| 10:05PM |
1 |
Asterisk as Softswitch |
| 8:29PM |
0 |
Codec negotiation (not the same old stuff) |
| 7:45PM |
1 |
Looking for Windows based Asterisk Management Client |
| 6:41PM |
2 |
Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue?? |
| 6:20PM |
1 |
ASTCC - card in use |
| 6:06PM |
0 |
1.2.0 voicemail: unable to create lock file? |
| 5:05PM |
1 |
QSig and MD110 |
| 3:36PM |
1 |
Asterisk DNS SRV lookups |
| 3:11PM |
1 |
Invite with Replaces |
| 2:33PM |
6 |
Asterisk + WiFi Phones |
| 2:10PM |
2 |
Modem Connections to PPP Server |
| 2:07PM |
1 |
Not receiving fax |
| 1:45PM |
5 |
[Asterisk-Dev] hello |
| 1:28PM |
1 |
PhoneCALL version 2.7-RC1 Released! |
| 12:55PM |
0 |
Cisco FXO hangup detection |
| 12:41PM |
0 |
Warning LSP Low |
| 12:23PM |
2 |
manager interface behavior |
| 12:23PM |
0 |
How to connect a Cisco Router with FXO module to Asterisk |
| 11:58AM |
1 |
astman make error |
| 11:48AM |
0 |
Call transfer with phones that cannot handle more than one line |
| 11:11AM |
0 |
Using transfer button in SJPhone |
| 8:16AM |
2 |
Querry about the modem |
| 7:32AM |
0 |
agent transfer problem |
| 7:17AM |
4 |
Asterisk SIP architecture question |
| 6:45AM |
0 |
How to make Broadvox work with Asterisk 1.2.0 |
| 6:20AM |
0 |
[patch] sqlite3 support for asterisk 1.2.0 |
| 6:12AM |
0 |
Re: [Users] open letter (2) |
| 5:43AM |
0 |
7960 audio quality when calling remote asterisk box |
| 5:42AM |
0 |
Calling lines |
| 5:41AM |
1 |
ISDN cards using CAPI interface |
| 5:16AM |
1 |
presence and Asterisk crash |
| 4:11AM |
0 |
queue problem |
| 4:08AM |
4 |
Aastra 1.3 firmware |
| 3:25AM |
0 |
Re: [Users] open letter (2) |
| 2:46AM |
3 |
Asterisk and DrayTek Vigor2600VGi |
| 2:42AM |
0 |
how to configure analog phone |
| 2:34AM |
5 |
open letter (2) |
| 2:19AM |
0 |
RE: [Serusers] Re: open letter |
| 2:13AM |
1 |
Asterisk server behind NAT, and SIP clinet behind another NAT. |
| |
| Tuesday November 22 2005 |
| Time | Replies | Subject |
|---|
| 11:13PM |
7 |
Help need to reset Adit 600 for Asterisk install |
| 9:32PM |
1 |
LinksysOne.com (New SIP phone, and more) |
| 9:24PM |
1 |
NVFaxDetect and NVBackgroundDetect on Asterisk 1.2 |
| 9:17PM |
0 |
Is there a way to see what agents are in a group from CLI |
| 9:16PM |
1 |
Strategy=ringall does not ring all agents. |
| 9:14PM |
0 |
Sipura SPA-841 Disconnects from Asterisk |
| 8:41PM |
6 |
ver1.2 installation problem |
| 8:38PM |
2 |
Clearwire and Asterisk |
| 8:09PM |
0 |
Idle time between agent ringing too long |
| 5:34PM |
1 |
Dial ZAP with group (g2) erroneously says call answered when it is still ringing |
| 4:49PM |
1 |
Bristuff for Asterisk 1.2 |
| 4:09PM |
0 |
SetGroup & GROUP_COUNT advise appreciated |
| 3:23PM |
3 |
Does Voipjet uses IAX2 trunking |
| 3:15PM |
2 |
Asterisk 1.2 Aastra/Sayson 480i DTMF Problem |
| 2:56PM |
4 |
Call parking on Polycom IP501 |
| 2:47PM |
3 |
high CPU usage when using -c |
| 2:42PM |
3 |
TDM400 FXO port 1 only problem. |
| 2:33PM |
0 |
Possible SIP/NAT Problem with 1.2 |
| 2:23PM |
1 |
sip URL peering |
| 2:18PM |
0 |
AstLinux VMware images now available (use with the free player) |
| 1:23PM |
1 |
Asterisk 1.2 + Debian Sarge |
| 1:20PM |
3 |
Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!... |
| 1:06PM |
2 |
Test numbers for ENUM (e164.arpa, e164.org, etc.) |
| 12:18PM |
2 |
Re: [Asterisk-biz] http://www.87810.com/ |
| 12:09PM |
2 |
Not able to call with phonzo |
| 11:00AM |
0 |
Re: [Users] open letter |
| 10:15AM |
0 |
DUNDi |
| 10:09AM |
1 |
Unable to register Zyxel WIFI Phone as SIP Clientto Asterisk |
| 10:01AM |
5 |
1.6.3 Polycom Firmware? |
| 9:45AM |
4 |
Virtual Modems Revisited |
| 9:28AM |
1 |
installing Asterisk from source |
| 9:12AM |
4 |
I need suggestions for on equipment |
| 9:02AM |
2 |
Re: [Asterisk-biz] VoIPJet Support Contact -We have US unrestricted termination for .095 |
| 8:47AM |
2 |
Best Communications Line for VoIP |
| 8:40AM |
3 |
Which is Better! |
| 8:25AM |
2 |
ATA verse Wildcard TDM400P |
| 8:22AM |
2 |
Master Telephone |
| 8:20AM |
4 |
Server Side AgentCallbackLogin |
| 8:14AM |
4 |
Slightly OT - Anyone know of an external ringer compatible with Cisco phones |
| 8:14AM |
0 |
Re: [Serusers] open letter |
| 8:12AM |
0 |
REPOST:How do you get a sound to play to caller on answer? |
| 8:10AM |
0 |
Unwated outgoing Zap channel briding |
| 7:57AM |
2 |
Digital Assitant Help |
| 7:38AM |
5 |
Bad Lines - What can the phone company do? |
| 7:32AM |
0 |
SIP debugging tools - Suggestions experience? |
| 7:23AM |
7 |
open letter |
| 7:12AM |
0 |
setting caller ID with Voicepulse |
| 7:04AM |
2 |
Asterisk 1.0.10 |
| 6:16AM |
1 |
Call waiting issue |
| 5:42AM |
0 |
sip routing |
| 3:17AM |
0 |
PAP2 and double ringback tone |
| 3:00AM |
1 |
HELP - ! No D-channels available! |
| 2:35AM |
1 |
channel_find_locked |
| 2:33AM |
0 |
outbound sip proxy |
| 12:36AM |
0 |
standard extension with forwarding |
| |
| Monday November 21 2005 |
| Time | Replies | Subject |
|---|
| 11:19PM |
1 |
Using Long Distance Operators |
| 10:38PM |
2 |
problem with registration of SIP phone |
| 10:28PM |
4 |
SIP Extension behind NAT, Asterisk on a public domain |
| 9:32PM |
1 |
priority jumping |
| 9:18PM |
2 |
equivalent to SetvarIf ? |
| 8:46PM |
1 |
ETel |
| 8:11PM |
0 |
odd transfer behavior |
| 6:09PM |
1 |
Setting up FXO in router |
| 6:02PM |
1 |
Rejected connect attempt |
| 5:27PM |
0 |
inquiry into Asterisk scripting scenario (VXML, AGI, IVR, etc) |
| 5:22PM |
2 |
chan_capi_cm-0.6.1: ISDN1: too much voice to send for NCCI=0x10101 |
| 5:08PM |
0 |
PAP2-NA 3.1.3 |
| 3:58PM |
1 |
Problems with fax failing when bridged across TDM400Pvers E |
| 3:50PM |
2 |
Detect alternate line in Broadvoice inbound context |
| 3:44PM |
1 |
Zyxel P2000Wv2 cannot do agent login, SJPhone work just fine? |
| 3:06PM |
2 |
recall button using tdm400 Australia |
| 2:49PM |
1 |
Asterisk 1.2.0 AddOn's compile error with MySQL 5.0.15 |
| 1:45PM |
0 |
hint for MGCP (devicestate): bug 5515 |
| 1:40PM |
0 |
H.323 and video |
| 1:34PM |
2 |
v1-2 install mkdep loop |
| 1:31PM |
2 |
AGI and AUTOHANGUP |
| 1:22PM |
0 |
Motherboard Selection Assistance |
| 12:59PM |
1 |
legacy pbx |
| 12:54PM |
0 |
allpage.agi |
| 12:49PM |
0 |
asterisk 1.2 unable to request echo training on channel |
| 12:31PM |
1 |
Help on x101p disconnect when called party answers |
| 12:23PM |
0 |
Include files in AEL |
| 12:10PM |
0 |
MOH: Most Efficient Method |
| 11:46AM |
0 |
mISDN + Fedora + asterisk 1.2 |
| 11:43AM |
0 |
AGIphp Installation |
| 11:36AM |
1 |
can't receive calls with CAPI ISDN |
| 11:00AM |
1 |
Select multiple columns from MYSQL cmd... |
| 10:43AM |
1 |
Asterisk and embedded system |
| 10:35AM |
1 |
How to deal with echo in MeetMe? |
| 10:10AM |
1 |
How do you disable realtime? |
| 10:08AM |
3 |
Linksys SPA941 |
| 9:45AM |
4 |
Anyone parked in your Asterisk? |
| 8:17AM |
1 |
Asterisk versions after the 1.2 release |
| 8:12AM |
2 |
AMP installation |
| 7:58AM |
1 |
Please Help with Zaptel |
| 7:48AM |
4 |
addmailbox script |
| 7:30AM |
0 |
split line authorization problem (ATL IP400 phone) |
| 7:17AM |
1 |
MySQL - Realtime install procedure? |
| 7:00AM |
1 |
zyxel p2000w |
| 6:56AM |
0 |
New firmware for Aastra/Sayson IP phones |
| 6:56AM |
1 |
h323 question |
| 6:51AM |
1 |
Asterisk crash: "using deprecated BYE/Also transfer method" |
| 6:18AM |
0 |
v1.2 and features.conf |
| 6:17AM |
0 |
HT486 and RFC2833 |
| 6:13AM |
2 |
AstLinux 0.2.9 Released |
| 5:57AM |
0 |
Problem with Broadvoice |
| 5:48AM |
0 |
User identification |
| 5:47AM |
1 |
Problem with SIP channels |
| 5:34AM |
2 |
Can not build zaptel with kernel-2.6.12 |
| 3:45AM |
0 |
How do you get a sound to play to caller on answer? |
| 3:38AM |
0 |
how to configure the LCS with Asterisk--->>Anyone, please????? |
| 3:02AM |
0 |
Problem with multiplier |
| 2:40AM |
3 |
Asterisk to Fax Server |
| 2:34AM |
4 |
E1 Gateway |
| 2:27AM |
0 |
zaptel compilation help! |
| 1:54AM |
1 |
Death at 2am |
| 12:56AM |
0 |
RTP question |
| 12:24AM |
6 |
Realtime Problems |
| 12:06AM |
0 |
CallProgress breaks DTMF - RFC2833 |
| |
| Sunday November 20 2005 |
| Time | Replies | Subject |
|---|
| 11:55PM |
0 |
General Bandwidth Voice Gateway |
| 7:52PM |
1 |
aastra 480i config files |
| 6:47PM |
0 |
Monitor() creating choppy audio files |
| 4:17PM |
0 |
Re: Call Leg/Transaction problem |
| 1:51PM |
1 |
Database update after hangup |
| 12:42PM |
2 |
Asterisk MySQL CDR - MySQL starting too late |
| 12:04PM |
1 |
DNID on IAX2 trunks? |
| 10:02AM |
1 |
Weird 1.2 stable problem |
| 9:16AM |
4 |
International Dialing Code |
| 7:36AM |
1 |
stopped sounds |
| 5:01AM |
0 |
CallProgress breaks DTMF |
| 4:06AM |
2 |
Dutch callerid and x100p |
| 2:50AM |
0 |
Illegal instruction on starting asterisk (was Newbie question) |
| 2:07AM |
1 |
SIP response 481, SIP client |
| 1:02AM |
0 |
AMP, huge number of Zs |
| |
| Saturday November 19 2005 |
| Time | Replies | Subject |
|---|
| 10:24PM |
6 |
Can Asterisk Set CallerID on Broadvoice? |
| 10:14PM |
0 |
Abdul Lateef Khan wants to talk to you using Google Talk |
| 8:20PM |
2 |
TDM400p card problem |
| 6:44PM |
0 |
mfcr2 and 1.2 |
| 5:57PM |
1 |
Free 411 Service |
| 4:55PM |
1 |
AMP partially not working, Apache dying on segfaults? |
| 4:41PM |
2 |
ztcfg segfault |
| 3:16PM |
3 |
asterisk.conf question |
| 2:42PM |
2 |
cmd dial timeout don't work in asterisk |
| 2:13PM |
1 |
VoIP connection US --> EU with ADSL a problem ? |
| 2:01PM |
1 |
Allowing Called user to accept call before transfer |
| 12:50PM |
1 |
simple setup |
| 12:38PM |
0 |
call parking and realtime_ext |
| 11:44AM |
1 |
Clipcomm CG-410 and caller-id from PSTN |
| 11:42AM |
0 |
i3micro VTA-111 to Ast 1.2 |
| 10:35AM |
1 |
Asterisk 1.2 compile error - Any suggestions would be appreciated |
| 10:08AM |
0 |
AstBill Live CD with Asterisk 1.2 Released |
| 10:02AM |
1 |
Wildcard FXO takes too long to answer incoming calls |
| 9:50AM |
1 |
Audio in MeetMe Conferences Garbled After Upgrade to 1.2 |
| 8:44AM |
1 |
MOH during M() macro execution |
| 8:28AM |
2 |
customized softphones |
| 8:09AM |
2 |
Dial() and j option: What is correct? |
| 7:05AM |
3 |
chan_bluetooth and Ericcson T68 problem |
| 6:37AM |
7 |
OT: Where to buy a T1 crossover cable for * and channel bank |
| 4:25AM |
1 |
ztdummy problem on SUSE 9.3 |
| 3:38AM |
1 |
cmd dial timeout don't work in asterisk 1.2 ? |
| 2:51AM |
3 |
return Credit Time |
| 12:41AM |
1 |
meetme + sendtext |
| |
| Friday November 18 2005 |
| Time | Replies | Subject |
|---|
| 11:07PM |
3 |
[Fwd: call status with FXO] |
| 10:17PM |
1 |
Retrieve multiple variables from database using MYSQL cmd |
| 7:58PM |
1 |
What's the streamplayer util and how to use it? |
| 6:20PM |
1 |
IAX Webphone to Dial a Support Extension Only |
| 6:09PM |
0 |
Page command in 1.2 |
| 6:04PM |
0 |
RE: RE: Asterisk-Users Digest, Vol 16, Issue 151 |
| 5:51PM |
0 |
SOLVED: Polycom MW beep |
| 5:41PM |
0 |
How do I test my Asterisk IAXmodem Hylafax HylaClient ? |
| 4:43PM |
0 |
secondary host= in iax.conf |
| 4:39PM |
3 |
GotoIf always goes to true? |
| 4:12PM |
1 |
Zaptel Error |
| 3:46PM |
1 |
Asterisk Compilation Error |
| 3:37PM |
5 |
Forward Voicemail to remote server? |
| 3:35PM |
0 |
Channels that won't die |
| 3:27PM |
2 |
Modifications to Voicemail |
| 3:07PM |
2 |
RE: Asterisk-Users Digest, Vol 16, Issue 151 |
| 2:27PM |
0 |
No Caller Name Displayed SIP->SIP |
| 2:20PM |
2 |
IAX and Firewall |
| 1:31PM |
0 |
Asterisk app_ices problem |
| 12:56PM |
1 |
WARNING[2757]: Failed to write frame |
| 12:51PM |
1 |
Getting invalid extension during agent login. |
| 12:45PM |
0 |
AMP 1.10.010 released |
| 12:09PM |
0 |
Voicemail ODBC storage and realtime |
| 12:04PM |
2 |
mISDN and chan_isdn for 1.2 |
| 11:46AM |
0 |
which g729 codec to use ? |
| 11:29AM |
1 |
Asterisk feature codes??? |
| 11:00AM |
1 |
auto assigning SIP port |
| 10:44AM |
1 |
A2billing warnings with new Asterisk 1.2 |
| 10:41AM |
0 |
AEL and n+101 apps |
| 10:38AM |
0 |
Cisco IP phone NAT config |
| 10:33AM |
5 |
VOIPJET - are they down |
| 10:00AM |
1 |
Context restrictions for long distance access, examples not clear? |
| 9:59AM |
1 |
R2 variations by country |
| 9:52AM |
5 |
Provisioning server |
| 9:45AM |
1 |
call transfer and pick chan_h323 |
| 9:27AM |
2 |
FAX difference IAXModem / Hylafax and spandsp app_rxfax |
| 9:22AM |
0 |
Specirfic IP to specific context sip.conf |
| 9:11AM |
6 |
Asterisk 1.2 error: "Ouch ... error while writing audio data: : Broken pipe" |
| 8:39AM |
1 |
Cisco phones port range |
| 7:54AM |
3 |
OT: Softphone with Bluetooth support for * |
| 7:51AM |
0 |
Asterisk 1.2 and music-on-hold question |
| 7:40AM |
1 |
phone intergration |
| 7:32AM |
3 |
Asterisk 1.2 - Windows Messenger ? |
| 7:24AM |
1 |
Remove older version of Asterisk |
| 7:02AM |
1 |
Examples of LIMIT_CONNECT_FILE and other LIMIT_XX Options |
| 6:42AM |
1 |
'ztmonitor' stopped working after using 'fxotune' |
| 6:32AM |
4 |
Sipura SPA-841 Second Line Help |
| 6:19AM |
0 |
Re: Asterisk en france |
| 5:46AM |
2 |
Problems with Read() in outgoing calls |
| 5:32AM |
1 |
In France asterisk never detect hang up. Why ? |
| 5:18AM |
2 |
wcfxo loads correclty after issuing twice the command "ztcfg -vvvv" !! |
| 4:31AM |
4 |
Contact field in SIP HF between asterisk + ser |
| 2:46AM |
1 |
Streaming mp3's when dialing a particular extension. |
| 2:17AM |
2 |
Problem switching from external ISDN-2 to PBX ISDN-2 |
| 2:14AM |
1 |
Re: Re: SIP - Loop detected (Matt Riddell) (Matt Riddell) |
| 2:12AM |
5 |
Newbie question. (Long) |
| 2:10AM |
1 |
gpx-2000 early dial support |
| 1:28AM |
0 |
re: problem with asterisk and SIP on same box with 1.2 |
| 1:09AM |
10 |
create my own soft Phone |
| 12:25AM |
0 |
Subject: Eicon Diva Server query |
| |
| Thursday November 17 2005 |
| Time | Replies | Subject |
|---|
| 11:34PM |
0 |
SPA 3000 and MWI |
| 11:14PM |
0 |
Asterisk voicemail responses - feature? |
| 10:22PM |
1 |
1.2 under OS X? |
| 9:31PM |
7 |
Eicon Diva Server query |
| 9:24PM |
1 |
SIP INVITE IP address variable? |
| 8:06PM |
1 |
how to originate a call and capture it's DIALSTATUS |
| 7:32PM |
2 |
Help with shell script for externnotify |
| 5:56PM |
2 |
Hung Zap channels |
| 5:28PM |
0 |
Missing smp kernel package in Asterisk 1.2installation... |
| 5:17PM |
1 |
What's the best way to stream and/or convert MP3 and WAV files? |
| 5:13PM |
3 |
multi tenant with queues |
| 5:08PM |
2 |
Missing smp kernel package in Asterisk 1.2 installation... |
| 5:00PM |
0 |
Asterisk 1.2.0 and memory usage |
| 4:11PM |
2 |
SER & Asterisk combination to get around NAT |
| 3:53PM |
1 |
realtime callerid |
| 3:39PM |
1 |
no longer loading all config files?!?!?!?!?!!!!!!... |
| 3:09PM |
1 |
1.2 won't compile: res_config_odbc.c |
| 3:07PM |
0 |
Overlapping sounds in asterisk and asterisk-sounds |
| 2:52PM |
2 |
call levels |
| 2:43PM |
0 |
Cisco SIP translation-rule Question |
| 2:23PM |
2 |
VoIP Gateway Providers |
| 2:13PM |
2 |
HFC ISDN card and mISDN driver |
| 1:57PM |
2 |
CVS v1-2-0 make problems? |
| 1:09PM |
1 |
Asterisk 1.2 Change in: agi_channel |
| 12:46PM |
2 |
Sound Choppy |
| 12:19PM |
0 |
sip.conf settings for voip.net / broadvox? |
| 12:19PM |
0 |
64bit libs in /usr/lib |
| 12:01PM |
4 |
Bristuff / Junghanns / Customer Service |
| 11:49AM |
2 |
Poor sounds on Adtran 750 |
| 11:19AM |
0 |
(AMUG) Asterisk Montreal User Group: Dedicated Mailinglist + Next Meeting |
| 10:02AM |
0 |
How to specify multiple agent groups with a queue? |
| 9:35AM |
2 |
Asterisk + FXO + FAX SIP |
| 9:07AM |
6 |
Wireless SIP Phones with Asterisk |
| 9:03AM |
0 |
Sipura doesn't get caller id and hangup with Siemens Combiset |
| 8:47AM |
0 |
Suggestions for tunning SJphone with Asteris k? |
| 8:37AM |
5 |
Mission-Critical Deployments |
| 8:29AM |
0 |
Bridgind and decoding. |
| 8:17AM |
0 |
D Channels reseting every 30 seconds |
| 7:48AM |
0 |
CallerID Length |
| 7:22AM |
1 |
PSGW 2.2 Skype gateway? |
| 7:14AM |
0 |
Suggestions for tunning SJphone with Asterisk? |
| 6:42AM |
1 |
Dazed and Confused |
| 6:20AM |
1 |
Hangup detection - TDM400P |
| 6:13AM |
0 |
IAX softphone's sporadic performance - Keep Alive Issue? |
| 5:50AM |
8 |
GSM Gateway / Terminal for sale |
| 5:46AM |
5 |
chan_bluetooth |
| 5:15AM |
4 |
/spool/outgoing delays |
| 5:05AM |
3 |
1.2 chan_modem not installing? |
| 4:59AM |
2 |
Register redirect |
| 4:53AM |
0 |
Voicemail email format |
| 4:47AM |
1 |
SIP Channel and jitter buffer |
| 4:44AM |
6 |
suggestions for hard phones? |
| 4:26AM |
1 |
chan_capi fails when Asterisk doesn't start under root user |
| 3:23AM |
1 |
Hardware HDLC in Zaptel - Bug ID 5313 |
| 2:21AM |
5 |
stop asterisk when Idle |
| 2:20AM |
1 |
RE: Re: SIP - Loop detected (Matt Riddell) |
| 2:10AM |
1 |
AGI Dial command return status |
| 1:24AM |
5 |
New asterisk management tool |
| 1:20AM |
4 |
is there any free pocket pc softphone?? |
| 12:19AM |
2 |
SIP - Loop detected |
| |
| Wednesday November 16 2005 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
upgrade from 1.0.9 |
| 10:55PM |
0 |
Hangup problem with other EPBX |
| 10:49PM |
14 |
Asterisk 1.2 Released! |
| 8:04PM |
5 |
ip phone |
| 8:02PM |
1 |
Asterisk drops call when calling other VOIP |
| 7:07PM |
4 |
IAX offline Voicemail |
| 7:04PM |
3 |
hold problem w/ GXP-2000 1.01.12 |
| 4:56PM |
1 |
Weird behavior on incoming calls |
| 4:54PM |
2 |
app_icd anyone? on 1.2? |
| 4:25PM |
3 |
receive fax with asterisk |
| 4:08PM |
0 |
1.2rc2 build problems |
| 4:07PM |
1 |
List of Motherboards or Servers that are testedok with Asterisk and Digium boards |
| 3:58PM |
1 |
TDM04b on FreeBSD |
| 2:04PM |
0 |
asterisk and cisco ubr900 configs using h.323. |
| 1:24PM |
1 |
Contention Detection with Zap ?? |
| 1:17PM |
1 |
Aastra 9133i registration errors |
| 12:38PM |
0 |
(no subject) |
| 12:33PM |
0 |
AT&T Merlin Communications System 6102 Cartridge Music on Hold and Paging |
| 11:51AM |
1 |
SER authenitification failure on ASTERISK |
| 11:38AM |
2 |
Outgoing sound very low |
| 11:21AM |
0 |
Read() application behavior change: bug or feature? |
| 9:55AM |
0 |
high availibilty (heartbeats) - a good way to |
| 9:38AM |
1 |
zapata.conf for T1 PRI |
| 9:00AM |
0 |
Cisco Security Advisory: Fixed SNMP Communities and Open UDP Port in Cisco 7920 Wireless IP Phone |
| 8:26AM |
0 |
SpanDSP and broken faxes (cut short pages) |
| 7:48AM |
1 |
List of Motherboards or Servers that are tested ok with Asterisk and Digium boards |
| 7:45AM |
0 |
bluetooth headset with softphone or directasterisk |
| 6:55AM |
0 |
NOTICE: ast_unregister_indication_country |
| 6:37AM |
0 |
Asterisk and Inter-tel |
| 5:48AM |
0 |
misdn for BRI |
| 5:42AM |
0 |
is the 'Zaptel Under the Hood' down? |
| 5:41AM |
0 |
Asterisk T.38 question |
| 5:39AM |
1 |
Is the 'zapteldoc.blogspot.com' down? |
| 5:30AM |
1 |
calling to asterisk and listening to music (GSM) |
| 5:28AM |
0 |
A-Z carrier Registration |
| 3:55AM |
0 |
Heads up - AVM C2/C4 on AMD 64 bit processors |
| 2:16AM |
0 |
Queue Autologoff over trunks |
| 1:37AM |
0 |
A simple network environment: a configuration issue or a bug in Asterisk? |
| 1:26AM |
1 |
Problem with octo bri |
| 1:18AM |
3 |
Recording voice messages in mp3 format |
| 1:06AM |
4 |
Asterisk @ Home password recovery |
| 1:04AM |
2 |
Compile problems, 1.2 rc2 and SUSE 9.3 |
| 12:56AM |
0 |
Asterisk pop-up |
| 12:56AM |
0 |
Price info in SIP packet? |
| |
| Tuesday November 15 2005 |
| Time | Replies | Subject |
|---|
| 11:57PM |
2 |
Dialing out with FXO |
| 10:28PM |
3 |
Agent not ready |
| 10:28PM |
1 |
Changing 5060 port |
| 10:14PM |
0 |
asterisk can't load chan_misdn (FC4) |
| 8:15PM |
0 |
Has anyone bought anything from Asteriskmall? yourexpirence? |
| 8:13PM |
1 |
PRI HDLC abort on dchan |
| 8:11PM |
0 |
Cisco Call Manager and H323 trunk correction(MTP) |
| 7:00PM |
1 |
Directory Command - Odd Hangup When Pressing "*" |
| 6:58PM |
2 |
Queue Monitoring.. |
| 6:40PM |
0 |
SDT Message Signal |
| 6:36PM |
0 |
Mention VoiceMail2 in UPGRADE.txt? |
| 5:33PM |
1 |
Cisco Call Manager and H323 trunk correction (MTP) |
| 5:06PM |
1 |
not work DTMF |
| 4:50PM |
5 |
reply to today's posting |
| 4:34PM |
3 |
A2billing questions |
| 3:19PM |
0 |
res_musiconhold.c: Music on Hold class 'default' already exists |
| 3:05PM |
7 |
g729 status in New Zealand |
| 2:58PM |
1 |
Anyone got zaphfc running 2 cards with NT and TE simultaneously? |
| 2:19PM |
0 |
1.2rc2: Problem with channel bank, Ring/Off- hook in strange state 6 |
| 2:00PM |
3 |
bluetooth headset with softphone or direct asterisk |
| 1:46PM |
0 |
"Call/Transaction Does Not Exist" back from 192.168.100.xxx |
| 1:40PM |
0 |
mtp-2 |
| 1:20PM |
0 |
1.2rc2: Problem with channel bank, Ring/Off-hook in strange state 6 |
| 1:01PM |
1 |
Automon / wW options ? |
| 1:01PM |
2 |
Max number of Digium cards a server can support? |
| 12:59PM |
1 |
Infinitum bloquenado SIP ???? / Is Infinitum |
| 11:44AM |
0 |
FXO module picks up but incoming callers only hear the ringing tone. |
| 11:24AM |
0 |
Has anyone bought anything from Asteriskmall? your expirence? |
| 11:17AM |
1 |
Using variables for context names |
| 11:09AM |
0 |
Problem with Zap/1 picking up on OUTBOUND calls from analog extensions |
| 11:08AM |
5 |
Incoming call trunk fwd not work |
| 11:08AM |
1 |
Can't start * with chan_capi.so |
| 11:04AM |
5 |
g.729 pass thru mode |
| 11:00AM |
2 |
Problem with call drops |
| 10:51AM |
0 |
FW: Asterisk 1.0/1.2 on cobalt Raq2-4] |
| 10:36AM |
0 |
backup routing for IAX outbound |
| 9:40AM |
0 |
Asterisk peer authentification |
| 9:19AM |
12 |
Editing Asterisk config files with WORD Pad |
| 9:15AM |
0 |
Illegal redirection |
| 9:13AM |
1 |
Infinitum bloquenado SIP ???? / Is Infinitum blocling SIP ???? |
| 9:09AM |
2 |
unexpected debug output from console |
| 8:51AM |
0 |
Play a message at the begining of a call |
| 8:44AM |
0 |
Problem: Can't make outgoing call |
| 8:38AM |
2 |
help and guidance needed from gurus |
| 8:34AM |
1 |
FW: Asterisk 1.0/1.2 on cobalt Raq2-4 |
| 8:26AM |
1 |
Restore Asterisk log files after deleting... |
| 7:33AM |
4 |
Message waiting notification |
| 7:26AM |
1 |
speech to text for only digits |
| 7:22AM |
1 |
Possible bug in agent monitoring |
| 7:19AM |
4 |
canreinvite=yes |
| 7:18AM |
2 |
Cisco 7960 Multiple Line Appearance |
| 6:39AM |
0 |
Queue Callback |
| 6:34AM |
4 |
Multiple emails |
| 6:15AM |
8 |
Multiple Outbound SIP Trunks |
| 5:38AM |
1 |
Asterisk and Agents |
| 4:41AM |
1 |
Monthly tips for the community? |
| 4:39AM |
0 |
app_conference compiling for asterisk |
| 4:22AM |
0 |
Configuring Asterisk Queues using real time(MySQL) |
| 4:16AM |
0 |
usb cellphone |
| 4:15AM |
1 |
errors with chan_zap.c when installing asterisk-1.2.0-rc2 |
| 3:58AM |
2 |
Cisco 7905 sccp Hold and Message buttons |
| 3:49AM |
1 |
Polycom Softkeys & Voicemail Button |
| 3:30AM |
1 |
A2Billing problems. still. |
| 3:02AM |
3 |
remove asterisk? |
| 2:42AM |
2 |
voicemial maxmsg |
| 2:35AM |
0 |
ERROR utils.c:509 tvfix: |
| 2:18AM |
3 |
SIP => H.323 Terminator |
| 1:42AM |
1 |
E1 PRI slips on TE410P |
| 1:08AM |
0 |
Asterisk + Voicetronix Card |
| 1:08AM |
0 |
H323 config question |
| |
| Monday November 14 2005 |
| Time | Replies | Subject |
|---|
| 9:26PM |
6 |
Asterisk hobby box |
| 7:42PM |
0 |
TDM400 FXS Pulse Dialing Craziness |
| 6:07PM |
1 |
Can someone explain the 's' extension |
| 5:16PM |
0 |
Looking for T38 Solutions |
| 4:37PM |
17 |
"open" asterisk? |
| 3:56PM |
4 |
Using RxFAX and TxFAX together |
| 3:47PM |
1 |
Hardware: Dell/Acer |
| 3:42PM |
2 |
voicemail locking |
| 3:33PM |
0 |
VoIP provider for South America Termination |
| 3:13PM |
1 |
Asterisk 1.0/1.2 on cobalt Raq2-4 |
| 2:50PM |
2 |
Problem with Cisco local conference and hangup |
| 2:39PM |
2 |
Mixmonitor |
| 2:05PM |
2 |
Dedicated echo canceller hardware |
| 11:42AM |
0 |
Grandstream - No dialtone in handset after 1.0.6.7 firmware update.. |
| 10:31AM |
4 |
Media gateway recommendations? |
| 10:24AM |
2 |
Connecting analog lines to Asterisk for IP telephony device use |
| 9:54AM |
1 |
How do I know if I have CRC-CCITT (README.Linux26) |
| 9:22AM |
1 |
PRI to SIP |
| 9:17AM |
0 |
TDM400 cards and modem/fax devices |
| 9:14AM |
4 |
Fritz card usb v2.1 - Capi installation problem |
| 8:39AM |
1 |
Comments in AEL files? |
| 8:38AM |
0 |
problem to connect h323 temination |
| 8:28AM |
1 |
Problem with 827-4v and asterisk as a pstn GW |
| 8:07AM |
0 |
connect to gateway h323 |
| 7:58AM |
6 |
asterisk sample size adjustment |
| 7:53AM |
2 |
Maximum Number of SIP Phones Supported By Asterisk |
| 7:28AM |
1 |
SIP signaling and canreinvite=yes |
| 7:25AM |
0 |
OT: Aastra PT 390 Question. |
| 7:07AM |
0 |
Configure Asterisk to call from softPhone(SIP Channel) to Analog phone(Modem Channel) |
| 6:21AM |
3 |
IAXy echo? |
| 6:06AM |
0 |
Brooktrout MPAC 1200 card with Asterisk |
| 5:57AM |
1 |
How to check how many G729 codec licenseinstalled |
| 4:21AM |
2 |
newbie question regarding asterisk |
| 3:54AM |
1 |
MYSQL issue in UPDATE.. |
| 3:00AM |
7 |
Snom clients deregistering |
| 2:50AM |
0 |
Promblem dialin from an internal E1 |
| 2:32AM |
11 |
ISDN card required |
| 12:06AM |
0 |
re: a2billing /areski help |
| |
| Sunday November 13 2005 |
| Time | Replies | Subject |
|---|
| 11:10PM |
1 |
Asterisk realtime extensions context inclusion |
| 10:35PM |
0 |
showing iax incoming calls |
| 10:21PM |
0 |
PSTN Trunk |
| 10:20PM |
1 |
g.729 codec |
| 10:10PM |
5 |
Anybody tried it from India ?. |
| 10:08PM |
1 |
TDM Echo issue |
| 10:06PM |
1 |
Regarding TDM400P |
| 8:17PM |
0 |
app_voicemail.so: load module failed, returning -1 |
| 8:16PM |
2 |
Asterisk Installation exits with following error |
| 8:01PM |
1 |
Sipura SPA-2002 Double Ring |
| 7:19PM |
1 |
Zaptel cards on SuSE? |
| 6:19PM |
0 |
Notices at beginning of call |
| 6:11PM |
1 |
iax-qos-openbsd... |
| 5:26PM |
0 |
MONTREAL USER GROUP MEETING, Tuesday 15th, 5pm |
| 5:14PM |
0 |
Advice on Asterisk-based home voicemail+fax+datasystem |
| 5:00PM |
2 |
How to get Referred-By header |
| 4:59PM |
2 |
Advice on Asterisk-based home voicemail+fax+data system |
| 4:18PM |
2 |
How to do "asynchrononous" Dial? |
| 3:43PM |
0 |
DeStar 0.1 released! |
| 3:26PM |
1 |
Asterisk overlap dialing (PRI) |
| 2:30PM |
3 |
X100P troubles? |
| 12:38PM |
0 |
Any experince with Voip Reach |
| 11:31AM |
0 |
Playing Music at the Back Ground while the conversation is on and recording the same in real time |
| 11:12AM |
1 |
fwd - Iax |
| 11:08AM |
1 |
Cron still running after uninstalling asterisk |
| 10:34AM |
3 |
spandsp-0.0.2pre21c broken? |
| 6:37AM |
1 |
"check for res for" |
| 6:36AM |
3 |
Upgrading 1.0.9 to 1.2 beta |
| 6:07AM |
3 |
Format of music for native MoH? |
| 4:31AM |
8 |
How to check how many G729 codec license installed |
| |
| Saturday November 12 2005 |
| Time | Replies | Subject |
|---|
| 9:44PM |
0 |
codec error connecting to cisco gateway |
| 8:57PM |
1 |
A2billing with Mysql-5.0.15 |
| 5:47PM |
1 |
WARNING[3035]: Invalid priority/label ' ' at line 17 |
| 5:46PM |
0 |
Example of Pass-Thru Codec |
| 5:08PM |
1 |
NEC NEAX 2400 Integration with Asterisk |
| 4:43PM |
0 |
Warning CONFIG_ZAPATA_DEBUG on 2.6.14 |
| 2:32PM |
1 |
Help with this |
| 12:43PM |
1 |
vigortalk and transfers |
| 9:29AM |
1 |
Unable to play dialtone |
| 9:20AM |
0 |
problems compiling spandsp-0.0.2pre21c under 1.2rc2 |
| 8:45AM |
1 |
REaltime does not unregister sip peers "on the fly" |
| 8:20AM |
3 |
Does IAX2 Trunk Work between IAX and SIP |
| 8:14AM |
1 |
asterisk@home KDE or GNOME? |
| 7:46AM |
1 |
PRI testing using TE205 and loopback cable? |
| 5:45AM |
1 |
Capi problem |
| 3:25AM |
1 |
How to let caller continue after Dial cmd |
| 3:22AM |
1 |
callcentrum - call any, ring one |
| 12:35AM |
1 |
debian sarge & zaptel 1.2 & TDM400P |
| |
| Friday November 11 2005 |
| Time | Replies | Subject |
|---|
| 10:34PM |
1 |
Snom 360 Opinions |
| 10:11PM |
1 |
Problems after upgrade... |
| 9:58PM |
0 |
Asterisk 1.2.0-rc2 Released! |
| 9:10PM |
0 |
How to add an Asterisk predictive dialer to a Strata VI phone system |
| 8:52PM |
6 |
Nextone <-> Asterisk <-> DID provider |
| 8:49PM |
1 |
Quantumvoice vs Broadvoice - Multiline |
| 6:41PM |
2 |
Fail over? |
| 4:54PM |
0 |
Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA |
| 4:50PM |
0 |
AAstra 9116 Weirdness. |
| 4:02PM |
2 |
7940 paperweight |
| 3:19PM |
0 |
Problem with CallerIDNum |
| 3:13PM |
2 |
[Announce] Web-MeetMe v1.4.0 |
| 3:02PM |
1 |
GoToIf Regular Expression |
| 2:54PM |
1 |
Asterisk behind a NAT |
| 1:54PM |
3 |
2 SIP phones on Y data connector on 1 ethernet |
| 1:31PM |
0 |
IAX2 phones |
| 12:57PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday 11/12/2005 |
| 12:35PM |
2 |
Non-numerical caller id in Budgetone 101 Ip Phon |
| 11:38AM |
1 |
Setting up IP PBX |
| 11:00AM |
3 |
IAX2 multiple audio frames per UDP packet? |
| 10:23AM |
1 |
asterisk high load high availability servers |
| 10:10AM |
0 |
HNT PROBLEM |
| 9:16AM |
0 |
sip.ld for a SoundStation IP 4000 |
| 9:09AM |
3 |
Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 1 |
| 9:07AM |
2 |
New Asterisk WEB Interface ( astwebmgr ) |
| 9:05AM |
0 |
Comand Read issue (Asterisk rel. 1.0.9) |
| 8:53AM |
1 |
SV: Call p2p |
| 8:48AM |
0 |
missing name part in to field of SIP header |
| 8:18AM |
2 |
GPS data from cell phones |
| 8:04AM |
0 |
Digium TDM400 on freebsd |
| 7:58AM |
1 |
MOH/Media Server |
| 7:55AM |
0 |
command returns a result code of -1 (indicating failure) |
| 7:03AM |
0 |
DISA multiple calls with single dialup |
| 6:02AM |
0 |
Re: Asterisk-Users Digest, Vol 16, Issue 85 |
| 5:44AM |
0 |
ASTERISK + POLYCOM IP PHONES |
| 4:39AM |
2 |
CAPI problem under gentoo with AVM C2 - asterisk claims CAPI not installed |
| 4:30AM |
2 |
IAX2 calls being droppped |
| 3:48AM |
2 |
sip ignores context definition? |
| 3:11AM |
1 |
Result branching in AEL |
| 1:00AM |
2 |
Voicemail file as MP3 |
| 12:37AM |
1 |
A2Billing Postpay |
| 12:25AM |
3 |
TDM400P + FXO module = PSTN woes |
| 12:11AM |
2 |
Softphone with Lotus Notes support? |
| |
| Thursday November 10 2005 |
| Time | Replies | Subject |
|---|
| 11:24PM |
1 |
one outgoing call == one call per minute |
| 8:44PM |
1 |
iaxy and comfort noise generation / sound quality |
| 8:08PM |
1 |
Asterisk: BUS Error in SPARC/Linux (debian) |
| 7:31PM |
0 |
SIP Registration from Verizon DSL |
| 7:28PM |
2 |
Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 18 |
| 7:05PM |
0 |
Disable Comfort Noise on Grandstream Phones? |
| 5:59PM |
1 |
How do I apply the asterisk patches? |
| 4:58PM |
0 |
FW: Re: MAX TNT SIP / Asterisk |
| 4:52PM |
0 |
Trouble completing a call. |
| 4:47PM |
2 |
Siemens optiPoint 420 phone and Asterisk SOS |
| 4:42PM |
1 |
A2billing problem.The system disconnects me immediatelly after asking me the PIN |
| 4:07PM |
1 |
NvFaxDetect , rxfax, Quantumvoice SIP : Dropping incompatible voice frame |
| 4:05PM |
1 |
Errors With Hint |
| 4:04PM |
0 |
features while on hold |
| 3:50PM |
1 |
Needed - Pager notification script |
| 3:38PM |
0 |
RE: 4 HFC cards |
| 3:29PM |
0 |
autoattendant timeout with include statement |
| 3:26PM |
2 |
Digium TDM Revision I Card |
| 3:23PM |
1 |
Digium TDM mothercard version I |
| 3:17PM |
0 |
agents in database |
| 3:02PM |
1 |
muiconhold.conf without restarting asterisk |
| 2:56PM |
0 |
RE: (BAD!!!) Sound quality of the NEW GRANDSTREAM BT 101 and 102 MODEL |
| 2:29PM |
1 |
txfax and rxfax problem with spandsp 0.0.2pre21c and 1.2rc1 |
| 2:27PM |
0 |
New astGUIclient/VICIDIAL version released 1.1.8 |
| 1:49PM |
1 |
SoundStation IP 4000 App and Cfg files. |
| 1:30PM |
2 |
asterisk 1.0.10? |
| 1:24PM |
1 |
Disa dialplan |
| 1:20PM |
1 |
PAP2-NA and SRV |
| 12:31PM |
6 |
Planet Network - VIP-153 |
| 12:28PM |
1 |
Possible problem with Zaptel/Asterisk with 1.2rc1 |
| 12:19PM |
3 |
Linksys PAP2: supported codecs |
| 12:02PM |
0 |
MeetMe bugs in 1.2.0rc1? |
| 11:56AM |
0 |
3COM 655005001 SIP phone on asterisk |
| 11:25AM |
3 |
Little OT.. SER Question |
| 11:16AM |
0 |
ast_merge_contexts_and_delete: Requested contexts didn't get merged??? |
| 11:12AM |
0 |
Call Transfer Problem with IAX2 |
| 11:11AM |
0 |
chan_modem_aopen.so loaded despite being told not too! |
| 11:03AM |
0 |
Cannot find where error message is comming from... |
| 11:02AM |
1 |
Need help can't figure out what wrong with zapata.conf |
| 10:52AM |
1 |
Ex-girlfriend mode on invalid/no CID? |
| 10:47AM |
0 |
sched.c: Attempted to delete nonexistent schedule entry |
| 10:38AM |
13 |
voicemail to two emails? |
| 10:36AM |
1 |
Asterisk 1.2-rc1 and sip show inuse |
| 10:36AM |
1 |
How do I factory reset a Grandstream BT-102 |
| 10:33AM |
2 |
(Some problems sending this menssage) Sound quality of the new BT 101 and 102 models |
| 10:24AM |
1 |
Bug in 1.2rc1 |
| 10:02AM |
1 |
looking for keypad free sip phones |
| 9:49AM |
1 |
TDM400 Card |
| 9:35AM |
0 |
TE110P Zaptel config questions |
| 9:34AM |
0 |
Nortel BCM 3.6 and Asterisk 1.0.9 via H.323 |
| 9:21AM |
1 |
Clarification on chan_modem.so module |
| 9:17AM |
0 |
Sound quality of the new BT 101 and 102 models |
| 8:53AM |
0 |
NAT'd SIP extension, no audio |
| 8:29AM |
0 |
Cell phone as digital trunk line |
| 8:21AM |
3 |
Simple Dial for If Busy Send to Voicemail |
| 8:00AM |
0 |
Phones no longer register - except one? |
| 7:56AM |
0 |
terminal emulation application that uses SIP |
| 7:38AM |
0 |
H323 still no rtp traffic |
| 7:36AM |
1 |
SIP Redirect/Transfer |
| 7:14AM |
2 |
SIP and VPN |
| 6:24AM |
2 |
New revision of my MFC/R2 software available |
| 6:18AM |
1 |
Call p2p |
| 6:04AM |
2 |
Queues with one Agent set to DND |
| 5:50AM |
2 |
ITS Telecom Hardware |
| 4:39AM |
3 |
IM / presence asterisk-1.2-RC1 |
| 4:18AM |
1 |
H263 algoritm in 1.2.0.rc1 |
| 3:48AM |
0 |
sorry for posting many times |
| 3:14AM |
0 |
Asterisk 1.0.9 + TE210 --- Long |
| 1:37AM |
1 |
SIP NAT register |
| 12:25AM |
1 |
Test environment (Windows Softphone) |
| 12:25AM |
0 |
chan_iax2: ast_sched_runq |
| 12:02AM |
2 |
Can't create iax channel |
| |
| Wednesday November 9 2005 |
| Time | Replies | Subject |
|---|
| 11:05PM |
6 |
Asterisk Crashing (high load issues) |
| 9:56PM |
1 |
Intel Desktop MotherBoards *NOT* Unsuitable for Digium Boards |
| 8:58PM |
1 |
Changes from 1.2beta2 to 1.2RC-1 |
| 7:56PM |
2 |
Zaptel Outbound Caller ID on E1 in UK |
| 7:46PM |
4 |
DTMF detection in TE406P ?? |
| 7:12PM |
0 |
Asterisk 1.2.0-RC1 Crashing with g729=?ISO-8859-1?Q? codec ?= =?ISO-8859-1?Q? and ?= =?ISO-8859-1?Q? ATA_?= 186 |
| 7:03PM |
2 |
DTMF method AVT |
| 6:27PM |
1 |
MusicOnHold does not play |
| 6:20PM |
1 |
Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 186 |
| 4:23PM |
3 |
Error compiling app_rxfax on 1.2-rc1 |
| 3:57PM |
3 |
Wits end with echo |
| 3:56PM |
0 |
system command vs mailfax and quotes? |
| 3:18PM |
2 |
what is the role of trunk=yes |
| 3:10PM |
0 |
Getting SpeedDial buttons to work on a Cisco 12 SP+ |
| 3:09PM |
1 |
Script for load testing |
| 3:07PM |
3 |
Cisco DHCP and Polycom boot server |
| 2:42PM |
3 |
Problems with HINT |
| 2:13PM |
0 |
RE: Asterisk-Users Digest, Vol 16, Issue 60 |
| 12:21PM |
1 |
dial during greeting to access another extension if busy or not available? |
| 11:52AM |
1 |
Asterisk OH-323 module-Inbound Call dropped due to in-call-rate violation (1.55) |
| 11:46AM |
1 |
Areski Can you Help ??? We are stuck |
| 10:45AM |
3 |
Test environment for a Predictive Dialer |
| 10:36AM |
0 |
Automatic testing of my DIDs? |
| 10:31AM |
1 |
Kapanga SoftPhone HOWTO |
| 10:29AM |
4 |
Realtime Voice Changer Patch |
| 10:28AM |
2 |
Cisco 7940 - TFTP |
| 10:19AM |
2 |
CVS HEAD - app_muxmon |
| 10:12AM |
3 |
Zaptel T1 Timing Source |
| 9:52AM |
5 |
Receptionist phones |
| 9:40AM |
5 |
force to expire a sip registration |
| 9:22AM |
0 |
Call forward to cell phone and X100P |
| 8:55AM |
0 |
long calls on same channel |
| 8:21AM |
1 |
Intel Desktop MotherBoards Unsuitable for DigiumBoards |
| 8:10AM |
3 |
problem with g729 and CME-Asterisk |
| 8:01AM |
1 |
[Asterisk-User] Festival help |
| 8:00AM |
2 |
TDM400 FXO Screech |
| 7:58AM |
1 |
Sending DTMF tones after answering on an IAX channel |
| 7:50AM |
0 |
Asterisk 1.0.9 + TE210 + SpanDSP |
| 5:22AM |
0 |
Zaptel: chan_zap.c:6514 mkintf: Unable to open channel 1 : Operation not supported by device |
| 5:05AM |
0 |
SIP/H.323 suggestion |
| 4:49AM |
0 |
New asterisk web gui for small/medium sizedbusinesses |
| 4:06AM |
7 |
dell and digium hardware |
| 3:51AM |
2 |
ast_streamfile failed |
| 3:43AM |
0 |
extension and overlap |
| 3:39AM |
0 |
bysy tone when dialing out via SPA-3000 in the netherlands???? |
| 3:23AM |
0 |
queue_log and mysql support |
| 2:53AM |
1 |
PRI pass-through |
| 2:11AM |
1 |
Asterisk Fax support using T.38 |
| 1:43AM |
2 |
MeetMe invite another user |
| 12:51AM |
1 |
how to setup Agent dialing in multiple asterisk servers |
| 12:50AM |
1 |
iax2 config sanity check |
| |
| Tuesday November 8 2005 |
| Time | Replies | Subject |
|---|
| 11:35PM |
0 |
Double Transfers. |
| 11:17PM |
0 |
Iax config areskiCC |
| 10:41PM |
0 |
Privacy Manager Application |
| 10:37PM |
4 |
Intel Desktop MotherBoards Unsuitable for Di gium Boards |
| 10:21PM |
0 |
DTMF problem in * -> ZAP/g1 calls(ZAP/g1 is a PRI group) |
| 10:00PM |
2 |
maximum concurrent conference peers in asterisk |
| 9:43PM |
0 |
OT: Atlas 550 Caller ID interoperability wit h Digium TE110P? |
| 9:38PM |
0 |
callfile to a ring group |
| 9:21PM |
2 |
Avaya 4612 IP phones with Asterisk? |
| 9:17PM |
3 |
sip_message_support.patch |
| 9:15PM |
1 |
Unsuccessful Native Bridge Between Zap Channels |
| 9:14PM |
1 |
SNOM360 & Monitoring Extension States |
| 8:55PM |
2 |
Asterisk 1.2.0-rc1 Released! |
| 7:36PM |
0 |
7970 How-To |
| 6:25PM |
1 |
NEC IPS PABX |
| 6:24PM |
1 |
Zap/TDM400p with old phone. |
| 6:08PM |
1 |
How do I show that a message is waiting on a Zap channel? |
| 5:59PM |
1 |
t38 for asterisk |
| 4:20PM |
8 |
Intel Desktop MotherBoards Unsuitable for Digium Boards |
| 4:17PM |
2 |
sipphone for freebsd |
| 4:03PM |
1 |
strange tone is droping calls |
| 4:03PM |
0 |
Please help me diagnose/fix this echo |
| 2:43PM |
1 |
A2Billing PIN does not get registered - keeps getting prompted |
| 2:18PM |
1 |
1.2b2/mpg123 and memory usage |
| 2:15PM |
1 |
Re: [Openvpn-users] Roomity.com v 1.5 is a web 2.01 |
| 1:47PM |
1 |
Fading echo on Zap<->SIP channels... |
| 1:45PM |
1 |
Play message and dial extensions simultaneously |
| 12:48PM |
3 |
Agent Call Recording |
| 12:43PM |
1 |
Help with SER |
| 11:53AM |
1 |
Asterisk Consultant |
| 11:51AM |
5 |
Extension Ring on Multiple Phones |
| 11:39AM |
5 |
ATA-488 FXO |
| 10:18AM |
0 |
FW: New package posted to Sourceforge |
| 9:59AM |
4 |
libbluetooth |
| 8:06AM |
0 |
Lost Cisco SIP phones after reboot |
| 7:59AM |
1 |
how to use #include to all files in /etc/asterisk/customdir ? |
| 7:38AM |
2 |
[Asterisk-User] Estension s don't start |
| 7:37AM |
1 |
OT: Atlas 550 Caller ID interoperability with Di gium TE110P? |
| 7:22AM |
6 |
New package posted to Sourceforge |
| 7:19AM |
1 |
Sipura 2000 |
| 7:01AM |
1 |
LCDProc for Asterisk? |
| 6:57AM |
2 |
BRI cards, HFC, and bristuff - a general question to clear up my understanding. |
| 6:51AM |
3 |
Cisco 7960 Password Recovery |
| 6:32AM |
3 |
Playtone on answering the phone |
| 6:12AM |
4 |
groupware + unified messagerie +Asterisk |
| 6:08AM |
2 |
Hiss |
| 5:53AM |
3 |
Sangoma 102 installation problem |
| 5:14AM |
1 |
Problem dialling multiple SIP devices |
| 3:24AM |
0 |
Detect registered peers |
| 3:09AM |
3 |
Sip provider problem or? |
| 3:01AM |
3 |
sangoma a104d install |
| 2:49AM |
0 |
SRTP proxy |
| 2:34AM |
4 |
Sensing fax with txfax |
| 2:27AM |
1 |
CallerID via chan_capi-cm-0.6 possible? |
| 2:24AM |
0 |
Softphone to show the activate sip user and their sip number |
| 2:17AM |
1 |
Which Wildcard? |
| 2:17AM |
0 |
Bristuff 0.2.0-RC8o or 0.2.0-RC8n (* 1.0.9) |
| 1:36AM |
4 |
Fwd: differences between chan_capi and chan_capi-cm |
| |
| Monday November 7 2005 |
| Time | Replies | Subject |
|---|
| 11:42PM |
2 |
How to configure LineJack |
| 10:13PM |
1 |
ad hoc conferencing-reg |
| 9:51PM |
2 |
How to make write and read formats equal to native format? |
| 8:41PM |
3 |
Choppy Audio in Echo Test and Music On Hold (1.2.0-b2) |
| 7:56PM |
0 |
FW: Error building res_perl |
| 6:08PM |
0 |
several beginner questions |
| 5:44PM |
2 |
how to send fax using Spandsp |
| 4:12PM |
3 |
SIP domain support for authentication and virtual hosting |
| 3:06PM |
1 |
zaphfc not generally compatible with kernels >= 2.6.13 |
| 2:51PM |
2 |
libmfcr2 - spandsp.h: present but cannot be compiled |
| 2:40PM |
0 |
new sip domain support and REGISTER requests |
| 2:36PM |
3 |
Problems with DTMF on Polycomm Phones |
| 2:11PM |
4 |
[OTAnn] Feedback |
| 1:53PM |
2 |
Can't make calls from Asterisk IAX to other IAX |
| 12:23PM |
1 |
asterisk 1.2b2 compiling problem |
| 12:18PM |
3 |
Stopping Asterisk from forwarding calls? |
| 11:27AM |
8 |
asterisk-1.2-bêta2 | presence/subscription support in the SIP channel driver |
| 11:23AM |
1 |
Speex codec problems |
| 11:17AM |
4 |
CentOS vs. Vanilla Kernel |
| 11:14AM |
0 |
AGI environment dump callerid |
| 10:20AM |
4 |
Help with dialplan to allow breakout to DISA |
| 9:28AM |
2 |
MP3 or OGG |
| 9:27AM |
5 |
Change Asterisk User |
| 8:46AM |
0 |
Help needed for Onhold calls |
| 8:35AM |
1 |
Re: Asterisk-Users Digest, Vol 16, Issue 44 |
| 7:36AM |
0 |
Use of Queues and agents to provide office phone coverage. |
| 6:03AM |
1 |
FXS problems |
| 4:55AM |
0 |
h323 nat externip |
| 4:04AM |
3 |
asterisk as SIP gateway |
| 3:50AM |
0 |
Outgoing and incoming call of LineJack |
| 3:42AM |
2 |
meetme conference getting error using codec g729 |
| 3:33AM |
1 |
queues in 1.2-beta2 |
| 3:21AM |
1 |
Asterisk Addons linker's error.... |
| 3:07AM |
0 |
SJphone "Awaiting ACK" after updating Asterisk to CVS-HEAD of September |
| 2:38AM |
6 |
Dropping last digit from phone number |
| 1:29AM |
0 |
Festival Sound Quality |
| 12:42AM |
1 |
What's the purpose of the "username=" line? |
| 12:00AM |
0 |
how to configure adhoc conference in Asterisk |
| |
| Sunday November 6 2005 |
| Time | Replies | Subject |
|---|
| 11:41PM |
0 |
Cisco 3640 as * FXO GW using MGCP? |
| 11:11PM |
0 |
meetme conference pbm using g723.1 codec |
| 8:37PM |
0 |
Slightly OT: Firefox search plugin for Voip-info.org |
| 8:24PM |
0 |
sangoma a104d |
| 5:56PM |
1 |
Testing with X101P |
| 5:31PM |
0 |
Stuck getting SIP trunk to work with 404 error. |
| 5:30PM |
0 |
Problem with Aterisk 1.2.0 beta 2 and sip dtmf |
| 3:28PM |
0 |
Help with SIP Phones inside a NAT with * inside another NAT |
| 1:57PM |
0 |
ABE - Are you happy with it? |
| 1:41PM |
0 |
Grandstream HandyTone 386 HT386 Distinctive Ring with _ALERT_INFO |
| 12:09PM |
1 |
Problem ringing multiple extensions when one is forwarded |
| 10:54AM |
2 |
escaping to an extension while listening to voicemail message |
| 10:49AM |
1 |
Re-invite don't always work |
| 10:30AM |
5 |
DNS Server Failure wreaks havoc |
| 6:03AM |
8 |
Meetme Conference-reg |
| 3:20AM |
1 |
how to conferencd in Asterisk |
| 1:08AM |
1 |
limiting incloming call on sip phones to 1 |
| |
| Saturday November 5 2005 |
| Time | Replies | Subject |
|---|
| 9:30PM |
0 |
Can't Access Amp |
| 9:22PM |
1 |
TDM400P hangup detection on Bell Canada PSTN |
| 7:35PM |
0 |
Fw: Fw: Inbound Calls on Asterisk from VBuzzer |
| 7:25PM |
0 |
Fw: Inbound Calls on Asterisk from VBuzzer |
| 7:23PM |
0 |
Inbound Calls on Asterisk from VBuzzer |
| 6:48PM |
0 |
General questions about register and nat traversal |
| 5:54PM |
1 |
anyone using nufone.net for termination? |
| 4:05PM |
1 |
Voipjet - No one is available to answer at this time |
| 2:26PM |
0 |
Realtime IP peer with static IP won't load |
| 2:19PM |
0 |
Registration time |
| 2:18PM |
1 |
PHP error setting up AMP |
| 1:55PM |
6 |
TDM400 FXO vs FXS Interrupt performance |
| 1:37PM |
1 |
"Hand-over" phone connections |
| 12:35PM |
1 |
Asterisk & Lucent TNT w/11.0.2 |
| 11:27AM |
0 |
Timing out on Registration |
| 11:11AM |
0 |
How does Nightly Downloads work at ftp://ftp.digium.com/pub/nightly |
| 10:36AM |
5 |
sill looking for a provider |
| 9:27AM |
2 |
chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs |
| 5:42AM |
1 |
How uniqueids are formed - possible race conditions for linked channels? |
| 5:12AM |
0 |
Looking for DIDs in Dubai |
| 3:35AM |
0 |
How to messure PDDs, how to detect fast hangup? |
| 2:53AM |
2 |
all circiuts busy now. resolution????? |
| |
| Friday November 4 2005 |
| Time | Replies | Subject |
|---|
| 11:57PM |
0 |
Sipura 2000 could not show incoming call's number |
| 10:34PM |
0 |
10/28 head > 10/29 head capi issue |
| 9:51PM |
1 |
Snom 190 Vmail setting |
| 9:31PM |
1 |
HDLC errors on PRI |
| 6:59PM |
4 |
SIP extension calls itself intermittently |
| 5:24PM |
0 |
MFC/R2 - unicall |
| 4:21PM |
1 |
Different answering policies for two zap interfaces |
| 4:14PM |
0 |
RE: Your message to Asterisk-Users awaits moderator approval |
| 3:38PM |
0 |
TDM2420E Availaibility |
| 3:05PM |
1 |
[OTAnn] Groups:New Developments at Roomity |
| 2:51PM |
1 |
Cisco phone firmware |
| 1:43PM |
5 |
GSM sound player for windows? |
| 1:26PM |
4 |
Moments of silence - take2 |
| 1:20PM |
1 |
Asterisk 1.2beta2 and UIP200 |
| 1:20PM |
0 |
manual transfer to automated operator. |
| 1:09PM |
2 |
User language switching in dial plan |
| 12:57PM |
1 |
What do I need to setup Asterisk with an H323 client? |
| 12:37PM |
2 |
Moments of silence |
| 11:27AM |
1 |
R2-Digital (Q.421) |
| 10:44AM |
1 |
Problem on Data-Connections through Asterisk |
| 9:57AM |
2 |
Te100 Digital vs Analog |
| 9:53AM |
5 |
Uninstall AMP |
| 9:20AM |
2 |
Meetme: Sending DTMF to other users in a conference |
| 9:12AM |
0 |
Beta2 problems with DTMF with "T" option in Dial Command |
| 8:46AM |
1 |
Can´t compile asterisk1.2beta2 |
| 8:20AM |
1 |
SIP phones supporting early dial |
| 8:19AM |
1 |
Does AEL support arrays? |
| 8:17AM |
1 |
Dial in via pstn , out over IP |
| 7:44AM |
0 |
re: Attempted to delete nonexistent schedule entry... |
| 7:36AM |
0 |
COREDUMP in actual CVS |
| 7:34AM |
1 |
Polycom IP 600/601 microbrowser specs |
| 7:14AM |
1 |
SCCP: ServiceURL and Mailbox Notification |
| 6:33AM |
0 |
one way audio on oh323 channel, there's no rtp traffic |
| 5:52AM |
0 |
voxby.com $29.95 unlimited...and no catch in T &C is anyone using them |
| 5:44AM |
1 |
CVS HEAD Broken? app_muxmon.so |
| 5:44AM |
1 |
Route call based on CallerID |
| 5:44AM |
0 |
2 Dial plan questions |
| 4:55AM |
2 |
Asterisk connected with CAPI |
| 3:57AM |
2 |
Zaptel: Hz != 1000 causing ztdummy compilationerror? |
| 3:34AM |
0 |
Zaptel: Hz != 1000 causing ztdummy compilation error? |
| 3:27AM |
2 |
Called number (Destination Number) |
| 2:10AM |
4 |
IAX2.FWDNET.NET not responding? |
| 2:07AM |
1 |
Hold Music is breaking up |
| 1:33AM |
2 |
Every SIP on its own FXO |
| 1:31AM |
5 |
SER+ASTERISK |
| 1:04AM |
1 |
Forward call without answer |
| |
| Thursday November 3 2005 |
| Time | Replies | Subject |
|---|
| 10:38PM |
0 |
ztmonitor usage |
| 6:52PM |
2 |
How to dial direclty from PBX extension to IP phone |
| 5:32PM |
1 |
Invalid/Timeout handlers in ael? |
| 4:57PM |
1 |
One Touch Record in 1.2 |
| 4:08PM |
0 |
Voicetronix OpenPCI , anyone using this? |
| 3:37PM |
0 |
Multiple zaphfc cards (for ISDN BRI) in a single machinemachine |
| 3:00PM |
0 |
Cisco smartnet to download firmware |
| 2:41PM |
1 |
T38 not compiling with today CVS |
| 2:28PM |
0 |
chanisavail - queuing |
| 1:56PM |
0 |
sip to asterisk? |
| 1:26PM |
1 |
Asterisk and SER for Call Center Application |
| 12:59PM |
3 |
SIP Disconnect Supervision |
| 12:52PM |
1 |
References? |
| 12:42PM |
0 |
Problems with meetme dropping audio during call |
| 11:57AM |
1 |
chan_agent.c fails to compile |
| 11:47AM |
0 |
Unicall |
| 11:31AM |
1 |
TDMoE problem |
| 10:12AM |
2 |
Basic question... |
| 10:02AM |
0 |
spandsp changelog |
| 9:54AM |
1 |
timed allow functionality of 'include =>'s |
| 9:46AM |
3 |
How to configure Asterisk through webmin |
| 9:34AM |
1 |
How to call each other for dynamic ip hosts |
| 9:22AM |
1 |
Ignoring Incoming RFC2833 DTMF? |
| 9:03AM |
9 |
Looking por a provider to work with asterisk |
| 7:41AM |
0 |
Include statement options docs . |
| 6:34AM |
3 |
How to detect AGI script failure? |
| 6:26AM |
5 |
call from asterisk to SIP cisco 5300 |
| 6:00AM |
0 |
Getting started, how to :D |
| 5:13AM |
1 |
curious bandwidth usage (incoming taking 3x more) |
| 4:13AM |
3 |
IAX test service |
| 3:57AM |
3 |
Multiple zaphfc cards (for ISDN BRI) in a single machine |
| 3:16AM |
0 |
Re: Asterisk and reverse DNS |
| 3:14AM |
1 |
Starting our own ip-telephony service? |
| 3:13AM |
0 |
Re: [Serusers] Accounting |
| 2:35AM |
0 |
Is this PRI INTENSE DEBUG correct (long) |
| 2:06AM |
0 |
Asterisk GUI/web interfaces that don'tchangeconfig files |
| 2:03AM |
3 |
Distinctive Ring Detection in AU |
| 12:58AM |
1 |
app_followme |
| 12:08AM |
1 |
Skinny.conf and sccp.conf |
| |
| Wednesday November 2 2005 |
| Time | Replies | Subject |
|---|
| 11:48PM |
1 |
Call Disconnect problem |
| 11:21PM |
8 |
Response time of TDM04b |
| 9:04PM |
0 |
tweak rxgain to prevent premature hangup (and hungup) by fax machine when communicating with rxfax |
| 8:45PM |
0 |
Problem with an AGI script. Going bald on this one. |
| 8:22PM |
1 |
Send text to Cisco 7960? |
| 7:53PM |
5 |
Anyone know who is in this picture? |
| 7:32PM |
2 |
Sipura password not working |
| 7:04PM |
2 |
Very basic switching application -- bounty? |
| 6:22PM |
1 |
New asterisk web gui for small/medium sized businesses |
| 5:36PM |
2 |
1.2-beta2 odd CLI output |
| 5:26PM |
1 |
A2Billing Authentication Refused |
| 5:25PM |
2 |
RealTime extensions - why so many SELECTs per call? |
| 5:13PM |
1 |
2 Asterisk boxes |
| 5:13PM |
2 |
Queue Strategy problem or advice |
| 4:43PM |
2 |
listening on multiple port #'s |
| 3:47PM |
0 |
Masquerade a call |
| 3:38PM |
1 |
RE: [Asterisk-biz] Asterisk as a VoiceConferenceServer |
| 3:28PM |
0 |
Re: Anyone aware of a current Dell servermodelwith 3PCI slots |
| 3:22PM |
1 |
cli output issue |
| 3:11PM |
0 |
Error with loading an FXS module |
| 3:02PM |
2 |
Warning -- chan_iax2.c: ast_sched_runq tasks |
| 2:47PM |
1 |
MG1 echo canceller results |
| 2:29PM |
0 |
Cadence, distinctive ringing and zapata.conf |
| 1:19PM |
1 |
E1 PRI card 17:31 channels problems |
| 1:18PM |
0 |
Problems with some channels on PRI E1 card |
| 1:16PM |
5 |
PRI E1 Problem only chan 17-31 |
| 12:51PM |
5 |
RE: [Asterisk-biz] Asterisk as a Voice Conference Server |
| 12:26PM |
1 |
Possible Issue With Meetme Conferencing in 1.2.0b2 and latest CVS HEAD (02/11/2005) |
| 12:16PM |
4 |
OS for ABE |
| 12:11PM |
3 |
faster transcoding possible |
| 12:10PM |
2 |
firmware update polycom 500 / dial problem |
| 11:53AM |
1 |
Fax between Asterisk SIP clients |
| 11:33AM |
1 |
A few Zaptel BRI questions... |
| 11:11AM |
0 |
Re: Re: intel e7230 chipset (Kevin Hanson) |
| 10:25AM |
4 |
Time based call direction |
| 10:01AM |
6 |
Satellite WAN |
| 9:45AM |
2 |
TDM0xB vs. SIP for FXO |
| 9:17AM |
1 |
How to bridge fax from pri to fxs |
| 8:56AM |
1 |
Voicemail in Realtime mode |
| 8:14AM |
1 |
intel e7230 chipset |
| 7:17AM |
1 |
extension |
| 6:32AM |
1 |
Zap Polarity Reversal |
| 5:04AM |
2 |
Options for 3-way or Conference Calling |
| 4:15AM |
0 |
REGEX() 1.2beta2 |
| 3:51AM |
0 |
Noise in Echo() |
| 3:29AM |
1 |
is it possible to connect to Asterisk from an external application? |
| 3:22AM |
0 |
Ericsson MD evolution and asterisk |
| 2:07AM |
0 |
Fritz!Card PCI ver2.0 |
| 1:47AM |
4 |
Installing beta2 |
| 1:14AM |
0 |
[Voicemail] Quota |
| 12:30AM |
0 |
SV: dial-out gives always "not found" (dial-in worksfine) |
| |
| Tuesday November 1 2005 |
| Time | Replies | Subject |
|---|
| 11:12PM |
0 |
CDMA gateway |
| 9:38PM |
1 |
changing email text based on voicemail user |
| 9:20PM |
1 |
Missing audio from Zaptel channels |
| 7:05PM |
1 |
Echo on TDM - Solved! |
| 6:32PM |
8 |
server hardware |
| 5:09PM |
2 |
Polycom files |
| 3:40PM |
0 |
IAXy Ringback Issues |
| 3:25PM |
3 |
How do you handle situation with Grandstream occasionally losing registration with Asterisk ? |
| 3:10PM |
0 |
Asterisk Beta 2 Possible Bug. |
| 2:41PM |
1 |
Double DTMF sent on T1 to T1 Native Bridge |
| 2:36PM |
2 |
Anyone aware of a current Dell server model with 3 PCI slots |
| 2:19PM |
3 |
Slightly OT: Cisco 7960/7940 and AsteriskReg istration Issues ove r a WAN |
| 2:04PM |
0 |
2 AgentCallbackLogin Questions |
| 1:10PM |
1 |
Slightly OT: Cisco 7960/7940 and Asterisk Registration Issues ove r a WAN |
| 1:10PM |
1 |
HT-486 Voice Nat Problem |
| 1:05PM |
0 |
New version (0.6) of Queue Statistics released |
| 12:42PM |
2 |
shared lines |
| 12:30PM |
2 |
Caller ID lookup via anywho.com |
| 12:07PM |
1 |
format_mp3 error on 1.2b2 |
| 11:50AM |
0 |
PRI to SIP D-channel Red Alarm |
| 11:50AM |
1 |
TDM dial in question |
| 11:35AM |
1 |
Error with one of my Zapata channels |
| 11:25AM |
4 |
feature.conf in 1.2beta2 |
| 11:12AM |
2 |
Asterisk Extension Language -- what's it's "status"? |
| 11:01AM |
0 |
Latest CVS just noticed this warning for the first time. Bis |
| 10:59AM |
1 |
Delays in sip invites. |
| 10:56AM |
4 |
Latest CVS just noticed this warning for the first time. |
| 10:16AM |
1 |
PAP2 and ringing issues |
| 10:01AM |
0 |
BLINDTRANSFER and Referred-Byand Referred-By |
| 9:33AM |
2 |
mesreading echocancel vs. echocancelwhenbridged? |
| 9:30AM |
1 |
Asterisk 1.2.beta2 and chan_capi |
| 8:41AM |
1 |
1.2.0-beta2 and realtime sip |
| 8:38AM |
0 |
process ID in log file? |
| 8:19AM |
1 |
Incomming calls |
| 8:07AM |
1 |
Problem with call files |
| 7:44AM |
1 |
Adding caller name / ID to outbound meetme calls |
| 7:37AM |
6 |
inband dtmf on ploycom ip501? |
| 7:19AM |
3 |
chan_exosip2 |
| 7:10AM |
2 |
Blind transfer from queue into another queue |
| 6:52AM |
0 |
User Permission |
| 6:45AM |
2 |
Fresh checkout Zaptel will not compile? |
| 6:07AM |
1 |
Voicemail Limits and Auto deleting |
| 6:06AM |
0 |
Forward sip messages to a proxy |
| 5:25AM |
0 |
UK BT Caller ID patches for X100P |
| 4:25AM |
1 |
problem with CME on 12.3(11)T6 and later (MWI) |
| 3:36AM |
0 |
Asterisk + Ser + Music on hold |
| 3:19AM |
6 |
1.2beta2 and spandsp |
| 2:58AM |
1 |
Polycom IP600 and micro-browser... |
| 2:44AM |
1 |
IAX2 trunking not work with slinear |
| 2:26AM |
0 |
How to program Phone "Configurable line indicators" for some PSTN lines |
| 2:24AM |
0 |
Different Ringing Tones depending of the call |
| 2:23AM |
0 |
No Media for Ringing Indication |