| Saturday April 30 2005 |
| Time | Replies | Subject |
|---|
| 11:08PM |
1 |
Broadvoice limits??? |
| 7:01PM |
2 |
Asterisk@Home bug |
| 4:59PM |
8 |
Programing a call forward feature to cel phones |
| 4:47PM |
1 |
How to bridge 2 calls |
| 4:10PM |
1 |
Kernel 2.4 or 2.6 |
| 3:57PM |
3 |
SIP over IAX2 |
| 2:58PM |
8 |
Problem with Sangoma/Adtran 600 installation |
| 2:53PM |
1 |
Problem with PSTN |
| 2:27PM |
1 |
Asterisk on Radio Tonight |
| 2:14PM |
2 |
Call-park timeouts.. |
| 1:52PM |
1 |
Hotel CDR Software |
| 1:18PM |
1 |
Send DTMF *AFTER* channels are bridged |
| 12:08PM |
22 |
A good SIP receptionist phone |
| 12:04PM |
0 |
X-lite and * behind Firewalls |
| 11:56AM |
1 |
Zaptel and Boostringer |
| 11:32AM |
0 |
ANNOUNCEMENT: Asterisk-java 0.1 released |
| 10:34AM |
1 |
Amp extensions script |
| 10:24AM |
1 |
help with compiling addons for cdr |
| 10:10AM |
2 |
transfer pstn call to voip line, thus freeing up pstn line |
| 10:09AM |
1 |
Intel 536EP |
| 10:07AM |
0 |
Chan_modem_* |
| 8:41AM |
0 |
7910 and Skinny |
| 8:39AM |
1 |
IPSwitchBoard version 0.111 released |
| 7:49AM |
1 |
Polycom IP500 Forward problem codec issue |
| 6:25AM |
3 |
Dynamic phone groups. |
| 5:54AM |
0 |
Avaya 4610SW IP phone? |
| 12:19AM |
0 |
sipp example |
| |
| Friday April 29 2005 |
| Time | Replies | Subject |
|---|
| 11:22PM |
1 |
CID Number problem |
| 10:01PM |
0 |
Call routing |
| 9:51PM |
0 |
CallerID on cell phone |
| 9:29PM |
1 |
Can't get incoming calls with IAX trunks (FWD &Teliax) |
| 9:03PM |
2 |
Can't get incoming calls with IAX trunks (FWD & Teliax) |
| 8:16PM |
2 |
Asterisk and sendmail |
| 5:57PM |
0 |
Polycom IP500 Ringer Volume |
| 4:26PM |
3 |
Need info : lspci |
| 2:57PM |
1 |
Any workaround for long DISA timeout before it actually dials ? |
| 2:45PM |
3 |
Caller-ID Block |
| 2:21PM |
4 |
Paging and intercom |
| 2:14PM |
1 |
UTSTARCOM Wifi handset? |
| 1:29PM |
1 |
txfax and Ghostscript 8.51 |
| 1:27PM |
0 |
More TDM questions.... |
| 1:10PM |
3 |
Bouncing DTMF? |
| 1:03PM |
0 |
Asterisk@Home 1.0 released |
| 1:01PM |
0 |
Curious behaviour for pound (#) key with SIP X-lite SoftPhones |
| 12:19PM |
0 |
Detecting DeadLocks |
| 12:10PM |
0 |
ISPCON: SIP CPE experts wanted for panel |
| 11:32AM |
1 |
GR-303 zaptel and zapata configurations |
| 11:19AM |
0 |
SIP/IAX softphone with g729/723 |
| 11:17AM |
0 |
Adtran 600 |
| 10:07AM |
1 |
User events - a dumb question |
| 10:03AM |
4 |
IAX2 one way audio |
| 9:50AM |
1 |
Sip endpoints that support re-invite?? |
| 9:12AM |
1 |
the beginning of voice menu is cutted |
| 8:33AM |
3 |
Channel bank of E1s? (one E1 input --> 2 x E1 output) |
| 8:30AM |
1 |
Asterisk Manager interface, setting global vars |
| 8:22AM |
1 |
chan_zap graceful failure |
| 8:16AM |
1 |
Queue Monitor Filename Problem |
| 8:05AM |
3 |
quadbri bristuff ztcfg fail |
| 7:42AM |
2 |
Recording in a call center |
| 7:11AM |
1 |
Asterisk on VMWare ESX/blade servers |
| 6:23AM |
0 |
EuroISDN bearer capability pass thru from (fax) a/b adapter on OctoBRI to TE410P |
| 6:00AM |
1 |
T1 Technology and VoIP Gateway Primer |
| 5:43AM |
3 |
Realtime feature |
| 5:03AM |
0 |
Cost field in Call Detail Records (cdr) |
| 4:42AM |
0 |
DNID empty on incoming calls |
| 4:05AM |
2 |
asterisk-oh323 |
| 3:59AM |
0 |
IPSwitchBoard Version 0.110 Released |
| 3:08AM |
3 |
bri error |
| 2:58AM |
0 |
Barge In With Queues |
| 2:19AM |
0 |
how to configure ser and asterisk together to share the load |
| 2:18AM |
0 |
how to share asterisk load with ser server |
| 2:17AM |
0 |
how to share asterisk load with ser |
| 2:16AM |
0 |
(no subject) |
| 12:57AM |
0 |
Some * scripts: Pull asterisk config from LDAP and authenticate() against voicemail passwords |
| 12:38AM |
1 |
first few seconds of call is lost |
| 12:17AM |
7 |
Pattern Matching |
| |
| Thursday April 28 2005 |
| Time | Replies | Subject |
|---|
| 11:32PM |
0 |
Voicemail Broadcasts |
| 9:43PM |
2 |
vmail.cgi: -rwsr-sr-x as root *still* won't read the files |
| 9:11PM |
0 |
How to Restrict Number of Lines |
| 9:08PM |
0 |
[Fwd: Re: [Fwd: Voicemails stopping]] |
| 9:07PM |
0 |
Re: Asterisk-Users Digest, Vol 9, Issue 255 |
| 9:04PM |
1 |
How to prevent number of agents |
| 7:05PM |
1 |
Help to configure asterisk to dial to an PSTNline |
| 7:04PM |
0 |
How do I add an IP to an Exten |
| 6:53PM |
3 |
missing first digit when dial extension / dtmf problem ??? |
| 6:52PM |
3 |
voip connection problems |
| 5:16PM |
1 |
Assigning DID and Extension with similar value |
| 5:07PM |
1 |
Traffic Testing |
| 4:58PM |
11 |
Problems with TDM400P card |
| 4:56PM |
2 |
Re: Re: T1/DS1/ISDN PRI |
| 4:44PM |
1 |
chan_capi crashes asterisk |
| 3:57PM |
0 |
E1 legacy multi PBX integration? |
| 3:53PM |
1 |
Gabled voice problem on Asterisk for two remote users |
| 3:21PM |
2 |
Sipura SPA-841 and firewall |
| 3:08PM |
0 |
Asterisk not paying attention to NAT Setting |
| 2:55PM |
5 |
Asterisk Hardware Recommendation |
| 2:38PM |
1 |
music on hold on R key not working. |
| 2:24PM |
1 |
Asterisk SIP sound issue |
| 2:14PM |
0 |
Prompts and MoH not working - AAH .09 |
| 2:00PM |
0 |
Spandsp compile error |
| 1:48PM |
1 |
Asterisk Home .9 with TDM11B |
| 1:42PM |
0 |
Hints: What are they? How do they work? |
| 1:38PM |
1 |
Help to configure asterisk to dial to an PSTN line |
| 1:22PM |
0 |
can asterisk send AT commands to a modem? |
| 1:17PM |
0 |
PRI ISDN NFAS configuration needed |
| 1:07PM |
3 |
Install Asterisk on CCM MCS-7835 Server |
| 12:40PM |
0 |
Re: T1/DS1/ISDN PRI |
| 12:33PM |
3 |
Music on Hold can' t hear it! |
| 12:30PM |
1 |
Asterisk CVS and bristuff-0.2.0-RC8a-CVS: no callerid |
| 12:17PM |
0 |
help / advice needed on a project |
| 11:49AM |
2 |
ftp.digium.com HTTP mirror, Digium's FTP server |
| 11:49AM |
0 |
No audio playback |
| 11:11AM |
13 |
Polycom IP500 - Phone TIme |
| 10:52AM |
1 |
RE: Number of production asterisk systems (Christopher Jacob) |
| 10:49AM |
4 |
Web interface Suggestions |
| 10:38AM |
0 |
BIND VoIP anyone? |
| 10:17AM |
2 |
VoicpulseConnect problems? |
| 9:48AM |
1 |
SIP calling Error from MP108 please help - confs included |
| 9:43AM |
1 |
Delete voicemail |
| 9:09AM |
3 |
Number of production asterisk systems |
| 9:01AM |
4 |
start asterisk |
| 8:54AM |
2 |
Console Warning Message |
| 8:50AM |
2 |
IAX attempt -> Segmentation fault |
| 8:20AM |
1 |
Experienced Asterisk Consultant in Chicago, IL |
| 8:07AM |
2 |
Prefix to CALLING Number ? |
| 8:06AM |
0 |
proper 2-card ISDN modem.conf configuration? |
| 7:04AM |
2 |
asterisk-h.323 |
| 6:59AM |
0 |
MGCP and CISCO 7960? |
| 6:42AM |
0 |
Agents CallBackLogin and HangUp to calling party on pick-up |
| 6:39AM |
0 |
Advice on Adtran 600 setup |
| 6:28AM |
1 |
800 number provider suggestions |
| 6:22AM |
0 |
Asterisk Agents |
| 5:59AM |
12 |
Newer Dell Servers + TDM card |
| 3:32AM |
0 |
RSS feed Asterisk-Users |
| 2:51AM |
0 |
problem with skinny |
| 2:38AM |
1 |
H323 FAX |
| 2:35AM |
0 |
sip and analog |
| 2:34AM |
0 |
(no subject) |
| 2:25AM |
0 |
Problem with X101P(Red Alarm) |
| 1:48AM |
2 |
Monitoring B chans and G.729 High Water Marks |
| 1:38AM |
0 |
Incoming calls and CAPI |
| 1:07AM |
1 |
Eicon DIVA PCI ISDN cards (notserver) workwithasterisk! |
| 12:48AM |
1 |
Eicon DIVA PCI ISDN cards (not server) workwithasterisk! |
| 12:40AM |
0 |
call recording problem |
| |
| Wednesday April 27 2005 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
Linux SoftPhone with Sound Daemon Support |
| 11:22PM |
2 |
Asterisk@home questions |
| 8:32PM |
0 |
Questions about ongoing calls |
| 8:28PM |
1 |
TDM400 doesn't know the hangup signal in china |
| 6:42PM |
1 |
Automatic Follow-Me Forwarding Based on Cell GPS |
| 6:04PM |
1 |
Transcoding times |
| 5:33PM |
8 |
Linksys/Cisco buys Sipura |
| 5:32PM |
0 |
Any other MoH source except * |
| 5:30PM |
0 |
Asterisk on a media stream vs. direct RTP communication between endpoints |
| 5:27PM |
1 |
Dialing out... |
| 5:19PM |
2 |
ATA 186 MGCP Firmware |
| 4:48PM |
5 |
IAX aproprietary protocol |
| 4:04PM |
4 |
* and Sipgate (UK) |
| 4:03PM |
0 |
ser rtpproxy asterisk problems.... |
| 2:59PM |
1 |
SIP -> capi problem (no sound) |
| 2:54PM |
1 |
UK (english) sound files (Paul R |
| 2:40PM |
2 |
CDR Billing Question. |
| 2:39PM |
0 |
Anyting special needed for fax on a ATA186? |
| 2:26PM |
0 |
Call Type = Data |
| 1:49PM |
3 |
UK (english) sound files (Paul R) |
| 1:48PM |
0 |
Asterisk on Solaris 10 x86 |
| 1:27PM |
2 |
Don't know what to do if second ROSE component is of type 0x6 |
| 12:49PM |
0 |
wip 5000 in 12 hour time mode - anyone? |
| 12:06PM |
4 |
Panasonic KX-TD1232 Signaling |
| 11:57AM |
3 |
[Fwd: Voicemails stopping] |
| 11:37AM |
0 |
Public IP for SIP and NAT |
| 11:35AM |
0 |
Remote Phones - No Audio In Either |
| 11:02AM |
18 |
RJ45 to RJ11? |
| 11:00AM |
1 |
j'ai un probleme de connexion |
| 10:24AM |
2 |
Receiving Incoming Calls not working properly on TDM400P with 4 FXO modules |
| 10:22AM |
0 |
Transcoding Capacity |
| 10:18AM |
1 |
oh323 Zone |
| 10:06AM |
2 |
tonezone in tunisia |
| 10:04AM |
0 |
[Fwd: Supervised transfer problem.] |
| 9:53AM |
0 |
Re: Using Asterisk to dial a number and thenwait to dial the extension |
| 9:42AM |
1 |
Confused on G723 and G729 |
| 9:35AM |
1 |
QuadBRI card on Suse 9.2 Unable to load qozap.ko |
| 8:41AM |
0 |
Determinating SIP Phone status |
| 8:06AM |
1 |
Connection Timeout problem with SIP phones from Gnet |
| 7:40AM |
6 |
Redirect two channels to each other? |
| 7:26AM |
2 |
Determinating Phone status |
| 7:18AM |
1 |
Grandstream BT101 Firmware |
| 7:09AM |
0 |
No audio playback after upgrade from 1.0.1 |
| 6:49AM |
5 |
Cisco SIP Firmware Price Increase |
| 6:26AM |
0 |
SetGroup on dialed calls? |
| 5:26AM |
1 |
Dialing out from remote. |
| 5:02AM |
1 |
All lines are busy |
| 4:21AM |
2 |
noload res_musiconhold.so breaksa IAX |
| 4:14AM |
2 |
agent monitor filename |
| 4:12AM |
1 |
Dial Tone |
| 3:54AM |
1 |
Asterisk doesn't disconnect when I hang up SIP (SIP -> PSTN call) |
| 2:53AM |
2 |
Supervised transfer problem. |
| 2:30AM |
1 |
Cisco 7.4 SIP firmware |
| 2:08AM |
1 |
b0rked hfc config |
| 1:46AM |
0 |
do I configure ISDN in zapata.conf? |
| 1:42AM |
1 |
No music on hold when transferring call |
| 1:20AM |
0 |
(no subject) |
| 12:37AM |
0 |
call a ldap result via my x-lite |
| 12:23AM |
2 |
RTP vs cRTP vs IAX |
| |
| Tuesday April 26 2005 |
| Time | Replies | Subject |
|---|
| 11:58PM |
4 |
call a peer over the asterisk manager with a php script |
| 10:25PM |
2 |
Zaptel FXO crashing. |
| 7:25PM |
0 |
removing monitor IN/OUT wav file |
| 7:22PM |
0 |
Recommend IP Phones??? |
| 6:17PM |
0 |
Auto Fax transfer problem? |
| 6:03PM |
2 |
SIP, Asterisk and NAT |
| 5:56PM |
1 |
The TCP in Asterisk |
| 5:55PM |
3 |
No Audio sent using playback cmd |
| 5:45PM |
2 |
US$100 bounty for two features in voicemail |
| 5:16PM |
7 |
Polycom Images |
| 5:05PM |
2 |
Warm standby boxes - keeping config syncronised? |
| 3:52PM |
1 |
Queue Management and Command Execution |
| 3:23PM |
0 |
Controlling extentions through Management Interface (fwd) |
| 3:20PM |
1 |
Variable names in dial plans |
| 3:20PM |
0 |
Controlling extentions through Management Interface |
| 2:25PM |
0 |
amaflags=Documentation |
| 1:58PM |
1 |
Is There Media Accelerator For Better AsteriskCalls |
| 1:23PM |
0 |
AAH 0.9 - SIP DTMF negotiation problem |
| 1:12PM |
1 |
Turn off Music on Hold |
| 12:06PM |
0 |
Stable Asterisk Version ??? / SIP problem ??? |
| 11:44AM |
3 |
Remote Phones - No Audio In Either Direction |
| 11:40AM |
0 |
CID signalling for DTMF |
| 11:29AM |
6 |
Extensions / Contexts |
| 11:19AM |
1 |
the CLI dial command |
| 11:16AM |
0 |
Establish a ppp connection with a Pipeline 50 |
| 11:04AM |
0 |
Voicemails stopping |
| 11:02AM |
2 |
Checking for a sound file |
| 10:50AM |
1 |
rxfax over IAX ulaw |
| 10:34AM |
1 |
Strange queue agent issue - Agent busy problem |
| 10:31AM |
2 |
Is There Media Accelerator For Better Asterisk Calls |
| 10:12AM |
1 |
Using Asterisk to dial a number and then wait to dial the extension |
| 10:08AM |
1 |
Cisco 7290 calling problems :-( - Sorry if this comes through twice |
| 9:50AM |
1 |
pri_dchannel: PRI got event: HDLC Abort |
| 9:47AM |
0 |
Re: Asterisk-Users Digest, Vol 9, Issue 224 |
| 9:45AM |
1 |
Polycom Config - SIP 1.4.1 |
| 8:52AM |
1 |
Cisco 7290 calling problems :-( |
| 8:36AM |
0 |
Polycom SIP 1.5.0 Firmware |
| 8:05AM |
0 |
asterisk xlite nat problem |
| 8:02AM |
0 |
japanese voice files |
| 7:52AM |
1 |
Fail over solutions |
| 7:33AM |
0 |
call an ldap search result |
| 7:31AM |
1 |
CLI dial command |
| 7:20AM |
0 |
X100P + spandsp locks machine with zaptel & asterisk 1.0.7 |
| 7:07AM |
1 |
Cisco to buy Sipura |
| 7:07AM |
1 |
I wanted to understand |
| 7:04AM |
2 |
Zap/PRI: received AOC-E charging |
| 6:58AM |
3 |
YAC and IPs |
| 6:56AM |
2 |
SIP behind IPTables/NAT |
| 6:48AM |
1 |
Incoming Not Answering |
| 6:43AM |
1 |
return a value from dial macro |
| 5:56AM |
1 |
Dial CLI Command |
| 5:40AM |
4 |
Digium for ETSI ISDN |
| 5:14AM |
0 |
Cisco Systems to Acquire Sipura Technology |
| 4:57AM |
2 |
Shanghai or Bangalore DIDs |
| 4:50AM |
4 |
pridialplan/TON question |
| 4:36AM |
8 |
Good FXO for UK use. |
| 4:28AM |
1 |
How to set jitter buffer for SIP |
| 4:21AM |
0 |
ForkCDR question |
| 4:09AM |
0 |
i need Asterisk free Billing systems |
| 4:08AM |
1 |
Asterisk and Cisco Call Manager |
| 3:57AM |
0 |
bri cli error |
| 3:45AM |
2 |
Group/Broadcast Voicemail |
| 3:08AM |
0 |
Unexpected control subclass 17 |
| 2:53AM |
0 |
ACD in Asterisk |
| 2:02AM |
1 |
how to use dialparties.agi |
| 1:51AM |
4 |
IP Softphone Recommendations |
| 1:31AM |
1 |
SIP/NetMeeting |
| 1:29AM |
0 |
help to configure sip server asterisk |
| 12:55AM |
0 |
Error on the Mysql, realtime database HELP soclose so far; . |
| 12:38AM |
0 |
Problem with long delay. VPN ? |
| 12:29AM |
5 |
VOIP Gateways & Asterisk |
| 12:22AM |
1 |
NO ringback tone for VOIP call to another SIP server |
| |
| Monday April 25 2005 |
| Time | Replies | Subject |
|---|
| 11:39PM |
1 |
Distinctive ring on BT100 |
| 9:24PM |
1 |
SV: Re: IPswitch: How to use speed dialing? |
| 9:05PM |
0 |
Only want softphone account from Vonage: |
| 8:50PM |
2 |
Digium Quad Span Cards |
| 8:14PM |
2 |
Error on the Mysql, realtime database HELP so close so far; . |
| 6:35PM |
1 |
Citel Handset Gateways |
| 6:29PM |
2 |
Playback dosen't play Playtone(Congestion) does ? |
| 5:44PM |
0 |
Asterisk ADSI |
| 5:23PM |
0 |
stanaphone now terminating fax |
| 4:55PM |
0 |
T1 E&M wink issues - bad int'l dial-outandoccasional dropped calls |
| 4:45PM |
2 |
Siemens SX66 wi-fi handset released |
| 4:35PM |
2 |
Polycom ip500 (Not-Registered) |
| 4:20PM |
0 |
using goto to do selective dialing |
| 3:54PM |
0 |
Transfers tend to fail after upgrade to 1.0.7 |
| 2:51PM |
0 |
Dialing to a remote extension |
| 2:30PM |
0 |
Cannot make outgoing calls on Mediatrix 1204 from Asterisk |
| 1:54PM |
0 |
Asterisk replacing CCM using Catalyst 6608 |
| 1:32PM |
7 |
Polycom IP4000 Conference Phone |
| 1:04PM |
5 |
voip problems |
| 12:54PM |
2 |
Has anyone used Libretel DIDs with Asterisk? |
| 12:05PM |
4 |
Dial Plan - How to prepend a digit |
| 11:56AM |
4 |
Phone Recommendation. |
| 11:20AM |
0 |
Does ztmonitor record the audio channel? |
| 10:08AM |
16 |
Broadvoice Down? |
| 9:44AM |
5 |
Grandstream ATA 286 problems |
| 9:08AM |
7 |
Alternatives to SpanDSP?? |
| 8:16AM |
0 |
QoS Help and survey |
| 8:07AM |
4 |
astrecipes v2.0 |
| 7:08AM |
0 |
Repost: Dialing problem - Cisco 7290 to anything |
| 6:38AM |
1 |
Basic telephony hardware questions |
| 6:31AM |
0 |
asttapi and identapop pro |
| 6:08AM |
2 |
Call Recording via monitor |
| 6:04AM |
0 |
need resources to include iax softphone functionality in vb6 app |
| 4:39AM |
2 |
What small PC can take 8 FXS + 8 FXO cards |
| 4:39AM |
0 |
chan_capi: no dialstatus, no causes, no branches |
| 3:24AM |
5 |
UK (english) sound files |
| 3:23AM |
0 |
each 64K channel's ABCD bits for E100P Digium Cards. |
| 1:55AM |
0 |
asteriks without h/w |
| 1:34AM |
0 |
[ANNOUNCEMENT] Amatix InstantPBX |
| 1:19AM |
2 |
signaling during a call |
| 12:55AM |
0 |
Zap event On hook(1) handling problem |
| 12:34AM |
1 |
No busy tone when dialing out over ISDN with Polycom 500 IP |
| |
| Sunday April 24 2005 |
| Time | Replies | Subject |
|---|
| 9:46PM |
3 |
Trouble with call parking/transfer |
| 8:24PM |
1 |
Why can't I hear audio? |
| 8:07PM |
0 |
What software and types of connections are used by VOIP providers |
| 7:45PM |
1 |
Problems with gotoiftime and cvs head |
| 7:00PM |
2 |
Asterisk best practices |
| 6:10PM |
3 |
Static and echo on PRI |
| 5:00PM |
1 |
Transfers fails, even after upgrade to 1.0.7 |
| 2:44PM |
3 |
T1 E&M false busy after dial |
| 1:56PM |
2 |
g729 passthrough? |
| 1:00PM |
0 |
T1 E&M wink issues - bad int'l dial-out andoccasional dropped calls |
| 12:55PM |
0 |
Need info on necessary config of new T1/PRIs |
| 12:34PM |
5 |
Can Asterisk do the following for me ? |
| 11:38AM |
4 |
How to prevent native bridging between SIP channels |
| 11:15AM |
1 |
sm bounty validate length of e164/e212 number for all countries |
| 10:57AM |
1 |
What is the best client's protocol for my softphones |
| 10:53AM |
2 |
Re: Asterisk-Users Digest, Vol 9, Issue 215 |
| 10:48AM |
0 |
Fritz+chan_misdn - any working example ? |
| 10:11AM |
1 |
NEED HELP PROGRAMING ASTERISK VoIP NETWORK |
| 9:30AM |
0 |
How can several Asterisk boxes working together? |
| 9:28AM |
0 |
cidsignailling mode question |
| 9:17AM |
0 |
Feedback on Junction Networks conferences? |
| 8:44AM |
1 |
T1 E&M wink issues - bad int'l dial-out and occasional dropped calls |
| 8:18AM |
1 |
Registerport 5060 or 1720? |
| 7:51AM |
1 |
Astcc Working but Can't Make The Call |
| 7:07AM |
0 |
Asterisk management GUI |
| 7:05AM |
0 |
VSAT and Asterisk |
| 7:05AM |
1 |
Asterisk2mp3 |
| 6:03AM |
0 |
inband DTMF with IAX |
| 5:50AM |
0 |
QSIG. |
| 5:40AM |
2 |
Meetme Announcement |
| 4:26AM |
0 |
Netjet/Linux/Asterisk issue |
| 4:15AM |
5 |
ztdummy and Debian |
| 3:14AM |
0 |
help:Memory Consumption |
| |
| Saturday April 23 2005 |
| Time | Replies | Subject |
|---|
| 6:01PM |
1 |
SIP registration behind Linksys WRT54G |
| 12:07PM |
3 |
Provisioning Lines |
| 11:36AM |
1 |
How to replace VM busy.gsm and unavail.gsm messages with custom files |
| 10:55AM |
2 |
ztcfg doesn't do anything from /etc/rc.d/rc.local |
| 10:50AM |
2 |
ASTERISK PROGRAMER |
| 9:31AM |
2 |
[Fwd: FW: IAX help] |
| 8:05AM |
1 |
OctoBRI and 2.6kernel |
| 6:17AM |
1 |
PA168 ip phone setup iax2 to LiveVoip |
| 4:29AM |
0 |
Dial While on IVR |
| 3:32AM |
0 |
usb phone(AU-100) and usb phone adapter(TJ560B) |
| 3:29AM |
0 |
chan_sip.c:7174 handle_request : Failed to authenticate user |
| 3:25AM |
0 |
ast_expr.y:243 to_integer:Overflow |
| 3:17AM |
1 |
Failed to authenticate |
| 12:11AM |
7 |
Hotel billing in IPSwitchBoard |
| |
| Friday April 22 2005 |
| Time | Replies | Subject |
|---|
| 11:16PM |
0 |
Most affordable 8-port NT-capable ISDN card |
| 11:14PM |
0 |
Connecting Elmeg CS100 ISDN system phones to Asterisk |
| 11:10PM |
6 |
Best of the best of IP Phones |
| 8:49PM |
4 |
Cisco 7960 won't register as SIP device |
| 7:38PM |
4 |
if outgoing call fails with provider 1 then auto try provider 2 |
| 5:59PM |
0 |
wIPPhone with Asterisk |
| 5:20PM |
2 |
console /distinctive ring |
| 5:19PM |
0 |
Libunicall Compile Error |
| 5:09PM |
5 |
IAX help |
| 2:53PM |
0 |
TDM-fxo card and zttest - logic probem? |
| 2:44PM |
0 |
Asterisk + Cisco 2620 |
| 2:32PM |
0 |
Grandstream : low bandwidth codec (ilbc doesn't work, any other ? ) |
| 2:25PM |
2 |
Recommendations for Spanish Voice Talent |
| 2:19PM |
2 |
Questions about a 7960 and images |
| 12:28PM |
0 |
Upgrade Cisco 7940/7960 firmware |
| 11:45AM |
1 |
Re: routing in extensions.conf |
| 11:38AM |
0 |
Digium Hardware Problem |
| 11:30AM |
0 |
IAX channel |
| 11:12AM |
4 |
Quadbri & bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ? |
| 10:42AM |
7 |
QOS Routers |
| 9:52AM |
2 |
voice pulse connect - no dtmf |
| 9:20AM |
0 |
ASTCC Database Creation |
| 9:01AM |
4 |
TE11OP -> Mitel 200Sx?? |
| 9:00AM |
0 |
changing port on IAX2 protocol |
| 8:40AM |
3 |
chan capi: Long incomingmsn line in capi.conf? |
| 8:32AM |
1 |
No sound with voicemail and musiconhold?!? |
| 7:55AM |
1 |
Error loading zaptel on RHEL4 |
| 7:37AM |
6 |
can't make my PRI dial out |
| 6:51AM |
0 |
Dynamic queue member behaviour |
| 6:36AM |
1 |
Alcaterl IP-touch phones |
| 6:34AM |
1 |
No such context/extension |
| 6:20AM |
0 |
Asterisk acting as PBX + SIP Proxy ... possible? |
| 5:58AM |
3 |
Mysql using Sip and voicemail |
| 5:33AM |
2 |
IAX2 Error |
| 5:20AM |
1 |
callto: URL (URI) tag for dialing |
| 5:12AM |
1 |
Asterisk transcoding |
| 4:56AM |
2 |
Asterisk Restart after crash |
| 3:28AM |
3 |
Dell PowerEdge SC1425 w/ TE405P? |
| 3:14AM |
1 |
DTFM tones almost completly muted. |
| 2:22AM |
2 |
X100P delayed ring on incoming calls? |
| 1:08AM |
0 |
Spandsp 0.0.2Pre15 with bristuff-0.2.0-RC8 Problem - Hangup |
| 12:52AM |
0 |
dialling problem with astcc |
| 12:41AM |
0 |
How to attended/supervisor transfer |
| 12:26AM |
1 |
Echo cancelling with Adit 600 |
| 12:16AM |
0 |
Asterisk increased memory |
| |
| Thursday April 21 2005 |
| Time | Replies | Subject |
|---|
| 11:51PM |
0 |
LiveVoip status report |
| 10:41PM |
0 |
Provider offering IAX and T.38 origination+termination? |
| 10:09PM |
1 |
Digium Card Issues |
| 7:48PM |
1 |
TE110p - universal voltage? |
| 6:54PM |
0 |
100 & AAH .9 |
| 6:39PM |
4 |
Bug? |
| 6:39PM |
8 |
Email to Fax |
| 6:22PM |
2 |
asterisk@home 0.9 zap problems |
| 6:18PM |
0 |
Music Onhold Problem |
| 6:15PM |
0 |
Looking for an IAX(2) or SIP DID provider for LA, Orlando and Chicago areas. |
| 6:07PM |
0 |
GS ATA 286 goes deaf. |
| 5:06PM |
4 |
Demo phones with advertisement announcements |
| 4:18PM |
2 |
Playing mp3's while recording voicemail |
| 3:41PM |
0 |
does ast_app_getdata() reset timeout? |
| 3:19PM |
1 |
Recording Queue agents |
| 2:54PM |
1 |
forwarding Sip call to IAX and vice-versa |
| 2:48PM |
2 |
Provisioning lines 5 and 6 via TFTP |
| 2:39PM |
2 |
using * for Internet call waiting |
| 2:16PM |
0 |
CVS-HEAD: Sip not paying attention to context |
| 1:55PM |
1 |
Queue member persistent stats |
| 1:08PM |
1 |
Multiple Line config help |
| 12:35PM |
3 |
Broadvoice gateways! |
| 12:23PM |
5 |
One touch voicemail on Cisco 7940/60 |
| 12:15PM |
0 |
leastrecent queue option |
| 12:04PM |
1 |
Problems with app_dbodbc.c |
| 12:01PM |
1 |
adding a thrid asterisk server |
| 11:08AM |
0 |
AgentCallbackLogin AckCall Problems (Current CVS-HEAD) |
| 10:17AM |
1 |
Libunicall Make Error |
| 9:59AM |
1 |
ZAP - outgoing call using different D-Channel each time ? |
| 9:54AM |
0 |
Re: Basic Setup Question |
| 9:06AM |
1 |
Error in starting asterisk |
| 9:03AM |
1 |
Max concurrent faxes using SpanDSP? |
| 8:41AM |
1 |
security |
| 8:00AM |
1 |
dialup internet via Asterisk |
| 7:51AM |
0 |
* not send SIP Notify for IAX2 channel |
| 7:45AM |
1 |
503 Error |
| 7:36AM |
1 |
PBX replacement |
| 7:31AM |
1 |
Queues configuration |
| 7:18AM |
0 |
hint priority and realtime in asterisk cvs-head |
| 7:11AM |
1 |
Odp: Re: capi problem with dialout |
| 6:58AM |
1 |
Bristuff and Belgium |
| 6:39AM |
2 |
i like my colors, thanks.. |
| 5:08AM |
1 |
Deny certain extension |
| 3:37AM |
0 |
Problems with soundcards |
| 3:09AM |
4 |
capi problem with dialout |
| 2:42AM |
3 |
Dial W option usage |
| 2:30AM |
0 |
Attended Transfer |
| 2:29AM |
0 |
Problem using ztdummy kernelmodul with Kernel 2.6.8 |
| 2:11AM |
0 |
Asterisk Cisco Connection |
| 1:33AM |
0 |
RE: Large Asterisk Setup (~500 Concurrent Calls + Scalability) |
| 1:26AM |
0 |
Asterisk Cisco Conection |
| |
| Wednesday April 20 2005 |
| Time | Replies | Subject |
|---|
| 11:18PM |
2 |
Fax Problems |
| 11:05PM |
0 |
Tonelist questions |
| 9:46PM |
1 |
Zap channels busy. Have to soft hangup. |
| 8:26PM |
6 |
Recommended Linux Dist. for Asterisk |
| 8:01PM |
4 |
asterisk home wiring question |
| 7:47PM |
0 |
Queuing with busy detect |
| 7:36PM |
0 |
Volume of call waiting beeps |
| 7:32PM |
1 |
mpg123 won't compile, arch x86_64 |
| 7:15PM |
1 |
Large Asterisk Setup (~500 Concurrent Calls +Scalability) |
| 7:14PM |
1 |
What do I need to get started? |
| 7:01PM |
0 |
Ringing problems was TDM400P Revision question. |
| 5:51PM |
0 |
error in asterisk and LOTS OF log files generated |
| 4:55PM |
3 |
TE110P |
| 3:30PM |
3 |
chan_unicall.c compile error |
| 3:27PM |
0 |
What do Digium use for tracking support tickets? |
| 3:07PM |
0 |
spa 3000 pstn with amp |
| 3:03PM |
1 |
TE110P card installation errors |
| 2:03PM |
1 |
Anyone have a GXP-2000 working with Asterisk yet? |
| 2:01PM |
0 |
Help with [codec_g729.c:196 g729tolin_framein:Invalid data] |
| 1:46PM |
0 |
Lucent EMRS PRI Card |
| 1:43PM |
1 |
Adit 3104 - user experiences? |
| 12:58PM |
3 |
GotoIf in Stable 1.0.4 |
| 12:58PM |
0 |
One-Way NO audio (and sometimes both ways) |
| 11:42AM |
0 |
choose audio codec with chan_sccp driver and 7920 wireless? |
| 11:18AM |
11 |
BYOD provider other than broadvoice |
| 11:14AM |
1 |
Annoying SIP registration problem behind ?Linksys? |
| 10:58AM |
0 |
Recommendations for IAX/SIP ATA |
| 10:46AM |
1 |
FXS --> FXO Converter |
| 10:40AM |
1 |
Dialplan not showing up. |
| 10:29AM |
1 |
Zap Extensions unavailable after a call |
| 10:08AM |
0 |
Route SIP calls to provider |
| 9:58AM |
0 |
Active calls not responding to entries |
| 9:51AM |
3 |
Line Noise UPDATE - If you've got line noise, read this |
| 9:07AM |
0 |
ADSI phones in the UK |
| 8:58AM |
1 |
RE: Re: a simple question |
| 8:56AM |
0 |
iaxtel.com ??? |
| 8:50AM |
2 |
Monitor via Manager question |
| 8:40AM |
0 |
Which free calling card app most suitedforcommercial use? |
| 8:25AM |
3 |
Transfer of incoming call from external to internal number |
| 8:19AM |
2 |
Cisco 7960 SIP registration??? |
| 8:12AM |
4 |
signate.com webcall |
| 7:56AM |
0 |
RxFax not hanging up... |
| 7:46AM |
1 |
CVS Head and SetLanguage |
| 7:32AM |
1 |
Can I do something with Caller-ID? |
| 7:29AM |
2 |
Wait in Dial String |
| 6:56AM |
4 |
G723.1 and G729 on Athlon 64 |
| 6:47AM |
1 |
General voip mailing list |
| 6:46AM |
1 |
Which free calling card app most suited forcommercial use? |
| 6:46AM |
0 |
FXO lines on TDM04B not responding |
| 6:01AM |
0 |
Help with [codec_g729.c:196 g729tolin_framein: Invalid data] |
| 5:23AM |
0 |
IPSwitchBoard connects to CDR |
| 5:21AM |
2 |
A question about queues |
| 5:00AM |
0 |
Asterisk + Adit 600 questions |
| 4:26AM |
3 |
Issues of reliability, hardware, platforms |
| 3:52AM |
0 |
Cisco 2800 with Asterisk |
| 3:43AM |
3 |
SPAM SPAM SPAM SAM SPAM SPAM SPAM |
| 3:29AM |
1 |
Snom 360s and Asterisk |
| 3:25AM |
4 |
Asterisk and VAD |
| 3:21AM |
1 |
TE410P PCI-slot |
| 3:11AM |
1 |
NAT issues |
| 2:36AM |
0 |
Cisco ATA Help |
| 2:18AM |
0 |
"friendly networks" via ** |
| 2:17AM |
3 |
Setting SIP username for CallerID |
| 1:52AM |
2 |
OH323 incoming audio stutter |
| 12:09AM |
2 |
IAX realtime HELP |
| |
| Tuesday April 19 2005 |
| Time | Replies | Subject |
|---|
| 11:32PM |
0 |
Text Messages |
| 11:20PM |
1 |
NAT and only been able to have 1 SIP phone behind |
| 11:06PM |
2 |
RealTime ignoring switch => Realtime/context@realtime_ext |
| 10:44PM |
1 |
help needed for sound device setup |
| 10:06PM |
1 |
Sample AGI Scripts in C needed. |
| 9:46PM |
2 |
CVS-HEAD and CheckGroup/SetGroup |
| 9:27PM |
0 |
VoiceMail Config Questions |
| 9:14PM |
2 |
SIP Phone Compatability |
| 8:19PM |
0 |
Libunicall |
| 8:06PM |
1 |
FW: Cisco 7920 - chan_sccp - asterisk@home .9 |
| 7:59PM |
0 |
Cisco 7920 - chan_sccp - asterisk@home .9 |
| 7:18PM |
1 |
Cisco 7960/7960G |
| 7:05PM |
0 |
Asterisk@Home v.0.9 and Digum |
| 7:01PM |
1 |
NuFone problems to non-na numbers |
| 6:29PM |
0 |
Looking for a softswitch |
| 5:11PM |
0 |
Attended transfer on sipura ATA/Phone? |
| 4:50PM |
0 |
TDM400P and SCSI/SATA = * noise problems??? |
| 4:33PM |
0 |
Industrial Cordless handsets analog or voip based ?? |
| 3:56PM |
11 |
US$200 bounty for * paging feature |
| 3:03PM |
0 |
Help needed on Utstarcom F1000 Wifi Handset. |
| 2:29PM |
0 |
Which free calling card app most suited for commercial use? |
| 2:15PM |
0 |
Any work around for ISPs that block port 5060 |
| 1:30PM |
0 |
New AstManProxy Manager Proxy v0.98 |
| 1:23PM |
1 |
Fax and spandsp |
| 1:21PM |
0 |
BRI channels not answering |
| 1:17PM |
0 |
Show accountcode in both directions? |
| 1:16PM |
1 |
Asterisk Netgear FSM7326P and Cisco 7960 on VLAN |
| 1:11PM |
0 |
Server failing to Boot with |
| 1:09PM |
0 |
Unix softphone |
| 12:58PM |
0 |
Latest CVS breaks voicemail app |
| 12:44PM |
0 |
Dutch callerid: sending on FXS? |
| 11:59AM |
1 |
ATA - PBX |
| 11:21AM |
0 |
mysql from dialplan |
| 11:20AM |
1 |
ASterisk OH323.CONF Gateway & Gatekeeper |
| 11:01AM |
4 |
Asterisk Business Case - Who is using it!? |
| 10:37AM |
2 |
OutBOund Dial problem |
| 10:36AM |
5 |
Conference solution for 100+ users |
| 10:14AM |
1 |
PRI - T1 feasibility |
| 9:34AM |
1 |
Extensions unavailable after to sucessfull call (Registration lose) |
| 9:31AM |
0 |
Testing the TDM01A |
| 9:25AM |
1 |
Re: Any work around for ISPs that block port .... |
| 9:12AM |
2 |
Want to use Asterisk instead of existingMeridianNorstar system ... need some help |
| 8:54AM |
0 |
Asterisk and Request Tracker, RT.? |
| 8:54AM |
3 |
Using voicemail independently from Asterisk PBX |
| 8:17AM |
1 |
Cisco 7960 directory.xlm |
| 8:16AM |
1 |
Testing my TDM01A |
| 7:36AM |
1 |
VoIP PSTN numbers in Australia? |
| 7:29AM |
0 |
Sipura PSA-841 -suitable headset |
| 7:18AM |
1 |
Any work around for ISPs that block port 5060 and69 |
| 7:12AM |
7 |
Firefly w/*? |
| 7:11AM |
1 |
IPv6 possible? |
| 7:10AM |
1 |
Soft Video phone for Windows XP |
| 6:48AM |
5 |
Any work around for ISPs that block port 5060 and 69 |
| 6:31AM |
1 |
Re: [Serusers] Ser + Asterisk |
| 6:28AM |
1 |
802.1p , precedence and TOS |
| 6:18AM |
5 |
Voicemail email text: |
| 6:04AM |
0 |
Snom NOTIFY on IAX2 channel |
| 5:53AM |
0 |
answered time |
| 5:40AM |
3 |
IPTables |
| 5:10AM |
0 |
SIP users, OH323 to provider, g729 - high level of echo |
| 4:59AM |
0 |
Looking for some real basic doccos... |
| 4:58AM |
0 |
AT-320 phones with IAX2 |
| 4:31AM |
8 |
VPN/Asterisk combo |
| 3:20AM |
3 |
Newbie - VoIP route SIP calls to provider |
| 3:09AM |
2 |
Asterisk and T.38. |
| 2:43AM |
1 |
TE405p PRI ISDN [E1] RED Recovering ? |
| 2:08AM |
2 |
Installed ztdummy, Asterisk doesnt work anymore |
| 2:08AM |
6 |
Asterisk with Softswitch |
| 1:51AM |
2 |
DID ~ Extension |
| 1:31AM |
1 |
Sipura SPA-841 distinctive ring |
| 12:53AM |
0 |
Astrisk + Cisco 5350 |
| 12:49AM |
0 |
codec negotiation with CISCO 7960 and Firefly softphone |
| 12:33AM |
7 |
Billing |
| 12:33AM |
0 |
Codec/Phone negociation(s) |
| |
| Monday April 18 2005 |
| Time | Replies | Subject |
|---|
| 11:57PM |
1 |
CLI Numbers |
| 11:18PM |
1 |
Asterisk timer on Digium's TDM cards? |
| 11:03PM |
0 |
Queues-Agents Problem |
| 11:01PM |
1 |
DTMF in outbound calls |
| 10:23PM |
1 |
Vici Dialer |
| 10:20PM |
0 |
Remapping Woes in features.conf |
| 10:18PM |
6 |
Asterisk & POE |
| 10:11PM |
1 |
HELP: How to detect a hangup tone? |
| 9:25PM |
1 |
G729 Key Registration Problem |
| 9:09PM |
1 |
Want to use Asterisk instead of existing Meridian Norstar system ... need some help |
| 8:03PM |
0 |
SIP calls being lost "frame from cahnnel" error |
| 7:36PM |
0 |
Extension busy issue on TDM01A |
| 7:33PM |
1 |
Blind Transfers - any ideas? |
| 7:30PM |
1 |
DTMF intermittently stops working |
| 7:30PM |
4 |
Citrix |
| 6:33PM |
2 |
Problems with incoming calls on a E1 ISDN PRI |
| 5:37PM |
1 |
wcte11xp digium card |
| 5:03PM |
0 |
Why *3* entries? |
| 5:02PM |
1 |
RealTime Vs. AGI and PHP or MySQL calls within extensions.conf |
| 4:56PM |
0 |
[Announcement] Updated Web-MeetMe |
| 4:09PM |
0 |
zombie channels & missed transfer |
| 4:05PM |
1 |
asterisk on MIPS |
| 3:42PM |
2 |
system wide speed dialing |
| 3:36PM |
1 |
snom 220 hints lost after reload |
| 3:32PM |
1 |
Problems with Cisco ATA 186/MGCP |
| 3:24PM |
1 |
Looking for ATAs |
| 2:56PM |
3 |
DIAL FROM CONSOLE |
| 2:40PM |
1 |
Junghans QuadBRI and fax detection |
| 2:10PM |
1 |
callback broken? |
| 1:53PM |
2 |
Fedore CORE 2 |
| 12:30PM |
0 |
ackcall with AddQueueMember |
| 12:00PM |
1 |
Random SIP Phone Problem |
| 11:57AM |
1 |
Hold on outbound calls and the SNOM 190 |
| 11:14AM |
1 |
[Fwd: Re: Only one PRI out of four working on TE405p?] |
| 11:11AM |
0 |
maximum value for LEN(x) |
| 10:12AM |
0 |
Voicemail not working... |
| 10:04AM |
2 |
Only one PRI out of four working on TE405p? |
| 9:47AM |
0 |
Unable to specify channel 1: No such device |
| 9:33AM |
0 |
Cisco 7970 startup problem |
| 9:14AM |
0 |
Strange tones when placing a PSTN call. |
| 9:11AM |
0 |
Lots of RTP checksum errors |
| 9:01AM |
1 |
Asterix Manager Proxy in Java/EJB? |
| 8:51AM |
3 |
Can I use Asterisk for a modified Hoot and Holler? |
| 8:44AM |
8 |
Calling Card |
| 8:34AM |
2 |
Snom subscribe/notify problem |
| 8:24AM |
4 |
Motherboard failure with 2 Digium TE405P car ds |
| 7:34AM |
2 |
Motherboard failure with 2 Digium TE405P cards |
| 7:28AM |
4 |
Help compiling zaptel in Debian |
| 7:22AM |
0 |
Indicating when other party has answered |
| 7:08AM |
0 |
Follow-me script - user changeable options |
| 6:56AM |
3 |
99% CPU - CVS 03.28.05 |
| 6:22AM |
0 |
Fw: Analogue phone transfering |
| 6:16AM |
3 |
queue - transfer calls |
| 5:13AM |
1 |
Changing Codecs when dialing out... |
| 5:01AM |
3 |
Cisco External Directory |
| 4:35AM |
1 |
Still having broadvoice issues |
| 2:31AM |
0 |
Error on install of AMP |
| 2:26AM |
1 |
Distributed organizations - large scale public sector rollout |
| 1:16AM |
1 |
Got SIP response 302 "Moved Temporarily" back.... |
| 1:09AM |
1 |
analog gsm router |
| |
| Sunday April 17 2005 |
| Time | Replies | Subject |
|---|
| 10:58PM |
2 |
Dynamic Dialplan - Turn VM on/off? |
| 10:35PM |
1 |
dynamic callrouting and billing? |
| 10:29PM |
1 |
hangs pc |
| 8:31PM |
1 |
Digium G.729 vs. IPP G.729 |
| 6:22PM |
3 |
Can anyone send me sample config files for asterisk and X-Lite? |
| 6:11PM |
3 |
Register two account at Broadvoice with one asterisk box |
| 4:48PM |
1 |
extension dialing resistivity |
| 2:37PM |
3 |
Unbelievable... |
| 2:03PM |
0 |
E & M signalling with WCTE11XP - not all calls go through |
| 12:53PM |
3 |
spandsp and cvs head |
| 12:34PM |
1 |
High Availability - Again |
| 12:26PM |
2 |
ISDN BRI vs. VOIP DID's, is it worth it? |
| 12:22PM |
0 |
RE: Asterisk-Users Digest, Vol 9, Issue 152 |
| 10:08AM |
1 |
IPP g729 & x86_64 |
| 9:49AM |
0 |
Bandwidth Reduction using Compressed RTP |
| 9:34AM |
0 |
AMP + POLYCOM |
| 8:29AM |
0 |
app_dtmftotext.c |
| 6:44AM |
1 |
res_perl compile problem |
| 6:31AM |
0 |
cisco mgcp and CARD.XML |
| 3:58AM |
0 |
Zaptel fxo & late distinctive ring |
| 3:34AM |
1 |
IPSwitchBoard Version 0.91 Released |
| 2:55AM |
2 |
Illegal instruction (core dumped) |
| 1:41AM |
1 |
Line name same as user name |
| 12:59AM |
0 |
Point-to-Point Asterisk Link to Reduce Bandwidth |
| 12:09AM |
1 |
OT VoIP related jobs in Eu |
| |
| Saturday April 16 2005 |
| Time | Replies | Subject |
|---|
| 10:13PM |
2 |
Park a call then hunt for a *willing* person |
| 8:59PM |
4 |
Hitachi WIP-5000/IP-5000 firmware |
| 7:19PM |
0 |
Can't Native Bridge Any More |
| 7:04PM |
3 |
recommandation for four (4) port FXS ATA |
| 4:44PM |
0 |
Receptionist Module |
| 4:20PM |
2 |
SIP/iax devices in Russia |
| 4:18PM |
3 |
problem connecting multiple boxes via IAX2 |
| 2:45PM |
2 |
Slightly [OT] Asterisk Backends |
| 1:50PM |
1 |
first few seconds of outgoing calls cut off |
| 1:11PM |
0 |
can't use 2 port gws simultaneosuly |
| 12:18PM |
0 |
Sipura SPA-2000 correct settings for Fax in The Netherlands/Europe |
| 11:42AM |
0 |
Sipura SPA-1001 Setup/Review |
| 10:17AM |
1 |
BT100 wrong NAT detection |
| 9:21AM |
0 |
Problem with wipphone |
| 9:18AM |
1 |
zap device detects hangup when phone switches from answer machine announcement to recording |
| 9:08AM |
1 |
2 Questions |
| 8:58AM |
0 |
Asterisk and openbrick |
| 8:53AM |
0 |
Mitel 5055 dead after wrong flash, any tips appreciated |
| 8:33AM |
1 |
Cisco/Asterisk codec negotiation problems |
| 7:58AM |
1 |
Is this normal - Long time to make call - What is your average with your Hardware? |
| 7:48AM |
3 |
VOIP to PTSN provider |
| 7:23AM |
0 |
Lots of RTP checksum error |
| 7:10AM |
1 |
OT: Sourcing Equipment at the HK Electronics Fair |
| 7:06AM |
0 |
[Fwd: Re: Debugging zaphfc + PBX integration] |
| 6:27AM |
0 |
OT: Interview With Kevin Fleming |
| 5:50AM |
1 |
IPSwitchBoard now has Zap Support |
| 5:03AM |
0 |
Codec Linux Bandwidth Reading |
| 5:01AM |
2 |
IPswitch: How to use speed dialing? |
| 3:10AM |
3 |
Installing Asterisk@Home on VMware Workstation 4.5.2- build 8848 |
| 3:00AM |
1 |
Asterisk@Home & ISDN BRI |
| 2:36AM |
0 |
FXO GW Dial in/out syntax |
| 2:10AM |
4 |
Asterisk and network problems |
| 1:43AM |
2 |
Extensions busy queue |
| 12:39AM |
0 |
acd+transfer+asterisk-1.0.7 |
| |
| Friday April 15 2005 |
| Time | Replies | Subject |
|---|
| 11:15PM |
1 |
Asterisk connect to Asterisk |
| 8:48PM |
0 |
Help needed to configure Asterisk and SJPhone |
| 8:00PM |
1 |
Dialplan help needed |
| 7:05PM |
0 |
AMP/Asterisk |
| 6:04PM |
3 |
a simple question . |
| 4:32PM |
0 |
IAXTEL Passord |
| 3:39PM |
0 |
How do I connect my Asterisk PBX to a serviceProvider |
| 3:30PM |
3 |
What is the good client softphone for windows? |
| 3:27PM |
0 |
How do I connect PC clients to my Asterisk PBX |
| 3:24PM |
1 |
How do I connect my Asterisk PBX to a service Provider |
| 3:21PM |
1 |
How do I make Extention in my Asterisk PBX |
| 2:24PM |
5 |
IAX softphone |
| 2:24PM |
4 |
Can't Modprobe ztcfg |
| 2:22PM |
0 |
FW: USB Controller ztdummy |
| 1:35PM |
1 |
fax detect/transfer problem solved |
| 1:20PM |
2 |
Anyone already pionered outing calling with user selcted background noise? |
| 12:51PM |
1 |
USB Controller ztdummy |
| 11:41AM |
1 |
Keypad disabled on AriaVoice SIP phone -- Fixed |
| 11:30AM |
0 |
How do I connect two Asterisk in different domains |
| 11:08AM |
5 |
OT: USB handsets / softphones |
| 10:50AM |
8 |
OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail |
| 10:25AM |
2 |
H323 Large Scale |
| 10:09AM |
4 |
large analog to asterisk |
| 9:54AM |
0 |
How to avoid CTL file request for Cisco 7970 |
| 9:48AM |
0 |
Outgoing PRI Call Early Media Detection |
| 9:43AM |
0 |
about volume in Playback() files |
| 9:12AM |
2 |
Bridging 2 Zap channels |
| 8:56AM |
0 |
Maybe not worded right, Answering a call |
| 8:25AM |
0 |
Ring requested on unconfigured channel 0/31 span 1? |
| 7:51AM |
2 |
sipXphone |
| 7:28AM |
1 |
ilbc codec in Asterisk |
| 6:57AM |
0 |
Polycom IP500 phones do not update time fromtime server |
| 6:27AM |
0 |
Asterisk working on FC3+X100P+France Telecom line |
| 6:25AM |
0 |
Question on Asterisk CDR / "In-Network Calling" / MySQL CDR |
| 6:22AM |
2 |
Debugging zaphfc + PBX integration |
| 5:50AM |
0 |
SIP Message Waiting Notification |
| 5:31AM |
0 |
Excessive re-registration of Broadvoice account in Asterisk@Home 0.8 |
| 4:41AM |
0 |
howto forward UAC codec capabilities to the PSTN gw |
| 4:38AM |
0 |
Problems with a SMS-capable Phone on a ZAP Channel / Question about native bridging on digium cards |
| 4:26AM |
0 |
g729 not work with DTMF and AGI |
| 4:18AM |
2 |
Slack 10 install - THANK YOU - & Cisco Reseller Help |
| 4:05AM |
1 |
SIP stack pluggable? |
| 4:04AM |
0 |
VIC2BRI and J4BRI |
| 4:00AM |
0 |
SIP through firewall is intermittent |
| 3:55AM |
0 |
IAX2 to IAX2 - one way audio |
| 3:49AM |
1 |
Asterisk live chat problem |
| 3:31AM |
1 |
Analogue phone transfering |
| 3:14AM |
2 |
Empty voicemail attachments? |
| 2:58AM |
0 |
LiveVoip incoming, no ringback still |
| 2:38AM |
0 |
qos test |
| 2:19AM |
0 |
Urgent .... Asterisk <-> Cisco CCM SIP TRUNK |
| 2:15AM |
3 |
*8 nor *8# works for me! |
| 2:14AM |
0 |
UDP Sip Data: GS Grandstream - remote office |
| 2:08AM |
0 |
E1 PRI: Unable to set channel to linear mode? |
| 1:49AM |
0 |
OH323 and outgoing calls problem. |
| 1:29AM |
4 |
Asterisk PBX with X100P in India |
| 1:12AM |
0 |
Soekris net4801 usb isdn avm fritz |
| 12:48AM |
0 |
OctoBRI - unable to specify channel 1 |
| |
| Thursday April 14 2005 |
| Time | Replies | Subject |
|---|
| 11:38PM |
2 |
Grandstream BT Volume |
| 9:43PM |
3 |
distribute outbound calls |
| 8:00PM |
0 |
dropping inbound calls from certain regions |
| 7:18PM |
1 |
DID reseller structures |
| 6:46PM |
1 |
How do I make a call thru *PBX |
| 6:33PM |
0 |
Mark Spencer and John "Maddog" Hall visiting Toronto - come and join us |
| 6:31PM |
6 |
cisco 7960 SIP setup |
| 5:11PM |
1 |
DTMF does not work with g729 and AGI |
| 4:57PM |
0 |
Cant respond to prompts from SPA1001 |
| 4:14PM |
2 |
Fax questions |
| 3:44PM |
2 |
ISDN BRI and signalling |
| 3:38PM |
1 |
Dial Macro Arguments |
| 3:34PM |
3 |
codec introducing huge latency |
| 3:25PM |
1 |
I dont want to hear the FXS port ring - TDM400? |
| 2:57PM |
1 |
asterisk + OH323 + NAT + gnomemeeting |
| 2:38PM |
2 |
Problem with Livevoip incoming context |
| 2:38PM |
0 |
How to reduce asterisk CPU-LOAD? |
| 2:34PM |
0 |
Bizarre - VM just stopped for one user |
| 2:27PM |
0 |
Routing on called number via SIP |
| 2:12PM |
1 |
MFCR2 compile requirements |
| 2:00PM |
0 |
Custom/Vanity DIDs |
| 1:37PM |
0 |
Call Parking timming out to the wrong extension |
| 1:09PM |
0 |
Voicemail delivery to pbx or mobile/panasonic dbs |
| 12:52PM |
0 |
Fritz Card going Crazy to make it compile |
| 12:13PM |
11 |
Overheard conversation. Comments please ! |
| 12:11PM |
2 |
Invalid extension handling |
| 11:42AM |
5 |
Line Presence: |
| 11:39AM |
2 |
making an action based on the status of multiple extensions |
| 10:10AM |
0 |
Re: Polycom IP500 phones do not update time from |
| 10:05AM |
4 |
Voicemail name (greet.wav) is not played |
| 10:00AM |
7 |
TDM400P Revision question. |
| 9:52AM |
1 |
Call Files to Terminate a call to the dialplan not directly to a channel |
| 9:34AM |
1 |
Wall Mount PC Case |
| 9:33AM |
0 |
asterisk hosting |
| 9:22AM |
5 |
Polycom IP500 phones do not update time from time server |
| 9:06AM |
4 |
Siemens optiPoint 420 phone and Asterisk |
| 8:57AM |
0 |
Matching on '@' in extensions |
| 8:55AM |
1 |
G.729A codec amd64/intel x86-64 optimisation? |
| 8:17AM |
5 |
Toshiba CTX100 integration with PABX for two site |
| 8:14AM |
1 |
Polycom IP500 phones and Presence feature |
| 8:01AM |
2 |
Voicemail Email |
| 7:33AM |
3 |
Who is a QUALITY IAX Termination Provider for 800 DID's? |
| 7:29AM |
1 |
Zap won't dial out? |
| 7:28AM |
1 |
Segregating a test version of asterisk - libpri/zaptel locations |
| 7:26AM |
0 |
cisco 79xx and SIP call statistics |
| 7:24AM |
1 |
Steal a call from a SIP extension |
| 7:10AM |
3 |
Ring two extensions at the same time |
| 7:06AM |
1 |
MoH stopped working with cisco 7912/7960 |
| 6:45AM |
3 |
delay problem in asterisk |
| 6:30AM |
0 |
IPSwitchBoard Version 0.86 Released |
| 6:02AM |
1 |
SIP Incoming Problem |
| 5:31AM |
1 |
BOUNTY: app_hangup from exten => h |
| 5:28AM |
1 |
BOUNTY - ztdummy & modules |
| 5:07AM |
1 |
sip phones make connection but no-sound is heared |
| 5:04AM |
1 |
Asterisk@home first experience |
| 4:41AM |
1 |
Re: Asterisk-Users Sip Reload or Realtime |
| 4:36AM |
1 |
Dialing rules |
| 4:20AM |
0 |
no voice tone |
| 3:59AM |
2 |
ISDN BRI + echo cancelling + Fax |
| 3:08AM |
0 |
Dropped calls from Junghans octo-bri card |
| 3:02AM |
1 |
lost DTMF digits |
| 2:50AM |
0 |
<register> syntax and limitation therewith |
| 2:10AM |
2 |
voicetronix bri |
| 2:03AM |
2 |
IAX blind transfers |
| 1:58AM |
1 |
Is there a SIP protocol stack inside asterisk? |
| 1:51AM |
1 |
need urgent help |
| 1:39AM |
3 |
Hylafax and Asterisk |
| 1:00AM |
1 |
pbx to asterisk |
| 12:35AM |
1 |
Cisco 7960 command-line dialer |
| |
| Wednesday April 13 2005 |
| Time | Replies | Subject |
|---|
| 11:55PM |
2 |
RTP problem |
| 11:35PM |
1 |
trying the xc-ast queue_log analyzer |
| 11:31PM |
1 |
Strange intermittent NAT problem with BT100s |
| 10:56PM |
5 |
RTP not being sent by asterisk |
| 10:01PM |
0 |
Re: Running asterisk without special hardwar e |
| 9:58PM |
1 |
oh-323 compilation error ! |
| 9:14PM |
1 |
Channel 0 on Zap ??? |
| 9:07PM |
2 |
Changing IRQ's on TDM |
| 8:53PM |
1 |
cannot dial two phones using zap |
| 7:10PM |
2 |
New Zealand Telco (TelstraClear) query |
| 7:09PM |
1 |
bashing my head against broadvoice |
| 7:02PM |
2 |
Cannot dial two phones at the same time |
| 6:45PM |
1 |
show translation |
| 6:28PM |
1 |
SIP Deadlock problem. |
| 6:28PM |
1 |
ZyXEL Router Terrible Voice Quality |
| 6:22PM |
0 |
H.323 in CVS Head |
| 6:13PM |
5 |
Telephone line installation. |
| 5:34PM |
3 |
does meetme need ztdummy |
| 4:58PM |
2 |
trying to figure out a few error messages in * |
| 4:13PM |
0 |
connecting Asterisk as SIP gateway to a Verso BHT1000 |
| 3:27PM |
0 |
about sip and skinny |
| 3:04PM |
2 |
Polycom Vendor Recommendation |
| 2:40PM |
0 |
Help, Outbound Problems |
| 2:10PM |
1 |
DISA() and predefined ACCOUNTCODE variable |
| 1:55PM |
0 |
SIP Clients over Wan losing connection |
| 1:51PM |
1 |
Advice sought on how to automatically and sa fely reboot * box |
| 1:27PM |
9 |
Asterisk@Home 0.9 released |
| 1:11PM |
0 |
Advice sought on how to automatically and safely reboot * box |
| 12:53PM |
0 |
Choppy music on hold |
| 12:45PM |
3 |
Why does this Macro Loop? |
| 12:42PM |
2 |
TDM card periodic buzz |
| 12:30PM |
1 |
PSTN VOIP integration not allowed in INDIA |
| 12:29PM |
2 |
Loop Detection |
| 12:25PM |
0 |
AW: SIP registration fails |
| 12:08PM |
1 |
Grandstream Won't hangup like Polycom 600 will |
| 11:51AM |
4 |
VOIP Regulations in INDIA |
| 11:39AM |
1 |
asterisk from cvs head crashes on via samuel 2, kernel 2.6.11-gentoo-r4 |
| 11:23AM |
2 |
SPA-3000 and quiet voicemail |
| 11:06AM |
0 |
need to ask you about your dlink and NAT/voip |
| 10:56AM |
0 |
Clipcomm CG-410 with asterisk? |
| 10:39AM |
0 |
Asterisk crashing? (gdb trace included) |
| 10:18AM |
2 |
IAX introducing huge latency |
| 10:08AM |
0 |
Asterisk on debian sarge doesn't start with CAPImodule errors |
| 10:07AM |
0 |
polycom dial...rings 4 ever, but redial connects |
| 9:59AM |
1 |
sip reload or realtime |
| 9:11AM |
1 |
SIP ACD system for station to station calls |
| 9:02AM |
3 |
IAXy Provision |
| 8:57AM |
2 |
Pretty Voicemail Docs |
| 8:50AM |
0 |
CVS-HEAD Zaptel with 1.0.x CVS Asterisk |
| 8:20AM |
2 |
ZAP channel hangs up with no apparent reason |
| 8:19AM |
2 |
Newbie Question on how to handle main office number |
| 8:12AM |
3 |
Unable to register license for G729 codec |
| 8:02AM |
0 |
FRAME_CONTROL (5) dropping calls on PRI |
| 7:48AM |
0 |
IPSwitchBoard is now Event Driven |
| 7:16AM |
1 |
Transferring a call |
| 7:14AM |
3 |
Zaptel and Fritz Card |
| 7:06AM |
1 |
SNOM 220 with >7 "lines" |
| 7:02AM |
0 |
PCI 1xE1, 2xE1 cards from Russia for MFC/R2 signaling for Asterisk IP-PBX |
| 6:53AM |
1 |
ni1 (ppp) and national isdn on te110p |
| 6:44AM |
1 |
Asterisk on debian sarge doesn't start with CAPI module errors |
| 6:21AM |
3 |
Who is willing to help an Asterisk newby? |
| 6:05AM |
2 |
OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10 |
| 6:03AM |
4 |
3-Way Calling in Asterisk |
| 5:51AM |
1 |
x-ten lite error |
| 5:31AM |
0 |
Question about Routing Order in .conf files |
| 5:27AM |
3 |
Cisco 7940G SIP Conversion |
| 5:11AM |
0 |
Asterisk / Quintum CRSP codec problems |
| 4:01AM |
0 |
codec quality |
| 3:16AM |
4 |
ISDN Fritz and TDM400 |
| 2:55AM |
0 |
Turtle Firewall - Sip user |
| 2:08AM |
2 |
iaxcomm |
| 12:13AM |
0 |
Sipura SPA-841 and Asterisk 1.0.7 with chan_misdn |
| |
| Tuesday April 12 2005 |
| Time | Replies | Subject |
|---|
| 11:52PM |
1 |
Codecs and * pass through... |
| 11:40PM |
1 |
New PRI install with new te110p |
| 11:16PM |
1 |
attension mark spencer |
| 10:22PM |
0 |
FW: binding Asterisk to virtual IP |
| 10:19PM |
0 |
chan_sip.c:7215 handle_request |
| 10:09PM |
1 |
Interesting Cisco 7960 issue, the phone picks up without me! |
| 9:49PM |
2 |
Problem reading digits from OH323 caller |
| 9:48PM |
1 |
weird call transfer problem |
| 9:42PM |
1 |
Article on IAX in Network World |
| 9:35PM |
2 |
invalid extension (need help) |
| 9:26PM |
0 |
invalid extension(need help) |
| 9:14PM |
0 |
Looking for comments on robustness of SpanDS P / app-rxfax / mime-construct <--solved |
| 9:03PM |
0 |
Hangup after Transfer problem. |
| 8:46PM |
0 |
Testing call back to back mfcr2 using chan_unicall with TE400P |
| 8:36PM |
1 |
Running asterisk without special hardware |
| 8:29PM |
1 |
Compile/modprobe issue |
| 8:10PM |
0 |
semantics terminology |
| 7:39PM |
2 |
Question about Macros |
| 7:09PM |
1 |
Blank voicemails being sent to users |
| 6:52PM |
0 |
Australian anologue callerid |
| 6:48PM |
4 |
New SNOM 190 Firmware |
| 6:25PM |
2 |
Outbound calling stops working after configuring trunk lines |
| 6:11PM |
0 |
OH323: Sending CallerID to H323 voip provider... |
| 4:06PM |
1 |
Realtime Friends |
| 3:48PM |
0 |
Looking for comments on robustness of SpanDS P / app-rxfax / mime-construct <--More information, and I figured out wh y |
| 3:30PM |
3 |
binding Asterisk to virtual IP |
| 3:17PM |
1 |
voice mail playback |
| 2:50PM |
1 |
Re: [Asterisk-Dev] Iax Trunking LD Service |
| 2:47PM |
0 |
Jitter problems IAX to Livevoip |
| 2:37PM |
0 |
TDM02B on 2 a/b ports of a PBX not working.. . help |
| 1:57PM |
3 |
IAX2 - Between two ASterisk Servers |
| 1:56PM |
0 |
OH323 and outgoing calls. |
| 1:33PM |
0 |
503 Service Unavailable |
| 1:23PM |
0 |
RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars |
| 12:57PM |
2 |
Looking for comments on robustness of SpanDSP / app-rxfax / mime-construct |
| 12:54PM |
1 |
Re: How do I reduce echo on asterisk |
| 12:05PM |
3 |
overwriting config file problem |
| 12:04PM |
3 |
RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars |
| 11:59AM |
0 |
Asterisk@Home - Newer Mobo - Memory |
| 11:55AM |
0 |
Anybody who is using a live stream as MOH |
| 11:54AM |
0 |
Music on hold not working between SIP clients |
| 11:34AM |
1 |
g729 versus g711 |
| 11:29AM |
0 |
Accountcode in SIP.CONF not set |
| 11:21AM |
1 |
Attempting native bridge of |
| 11:11AM |
0 |
Iaxy, Transfer, & # |
| 10:56AM |
0 |
Dumb question ? |
| 10:48AM |
3 |
Cisco 7960s and skinny |
| 10:36AM |
0 |
Re: Dumb question ? |
| 10:30AM |
0 |
Meetme and billing |
| 10:16AM |
4 |
Local Echo |
| 9:51AM |
0 |
LCDial and default provider |
| 9:48AM |
1 |
How many licenses of G729 do I need? |
| 9:10AM |
0 |
Noises on ZAP Channels |
| 8:59AM |
1 |
How do I reduce echo on the Caller side |
| 8:52AM |
1 |
Multiple TDM cards on the same box |
| 8:47AM |
0 |
Re: Asterisk-Users Digest, Vol 9, Issue 104 |
| 8:36AM |
1 |
QoS TOS numbers and Cisco IOS |
| 8:26AM |
0 |
Dialing Out (My mistake, here is the entire message) |
| 8:01AM |
0 |
RE: Ebay listing selling Asterisk @ Home (Blah Blah) |
| 7:55AM |
5 |
Acceptable voice time delay |
| 7:37AM |
0 |
Power Consumption of a Digium Wildcard TE410P |
| 7:18AM |
5 |
multiple line usage on Polycom IP300 |
| 7:09AM |
0 |
Internet Conection Broken and asterisk can not route any calls |
| 7:05AM |
0 |
Meetme disconnecting clients that use VAD |
| 7:02AM |
0 |
NENA CAMA Trunks for 911 and * |
| 7:00AM |
1 |
Low cost box for hosting Asterisk and atleastoneTDM400p - THIN CLIENT MAYBE? |
| 6:34AM |
2 |
How to get list of codecs |
| 6:05AM |
0 |
multiple asterisk boxes with "show channels" |
| 5:14AM |
2 |
TE410P and X101P problem |
| 4:40AM |
6 |
Version 0.80 of IPS released |
| 4:23AM |
1 |
Problem with fxo |
| 3:15AM |
3 |
TE110P - NT-Mode ? |
| 2:57AM |
0 |
H.323 Question |
| 2:36AM |
1 |
Asterisk Addons compile errors |
| 1:56AM |
1 |
Supervisor monitor / barge in - automatically on call setup? |
| 1:45AM |
0 |
Voicemail quota |
| 1:36AM |
0 |
Asterisk on HP DL380 G4 - chan_zap.so problems |
| 12:42AM |
0 |
Problem with * transfer |
| 12:19AM |
1 |
Has anyone got Asterisk working behind a NAT connection to users within a NAT |
| |
| Monday April 11 2005 |
| Time | Replies | Subject |
|---|
| 10:18PM |
1 |
Losing CallerName info if no CID sent |
| 9:45PM |
2 |
Zyxel P2000W Finally (Almost) Working |
| 8:28PM |
3 |
Trunk Seize - Line 1 - CO1: Does it exist in an Asterisk environment? |
| 8:04PM |
0 |
H.323 General Questions |
| 8:03PM |
4 |
Asterisk management portal |
| 7:48PM |
0 |
ChanSpy -- Slow/garbled audio and console errors |
| 7:41PM |
1 |
Remote phone often appears to be disconnected |
| 7:25PM |
0 |
Asterisk Realtime - can't see sip friend |
| 7:24PM |
0 |
E911? |
| 7:22PM |
0 |
Asterisk did not play music when pressing hold button on SJPhone |
| 6:23PM |
1 |
Problem Detecting Answer on a PRI Outcall (sometimes) |
| 6:06PM |
1 |
Monitor with Asterisk@Home |
| 5:35PM |
0 |
OT: Thunderbird threading |
| 5:11PM |
1 |
Dialing Out |
| 5:11PM |
0 |
Advice in provider for business use |
| 4:38PM |
1 |
Best FXO Voip Gateway for Asterisk |
| 4:32PM |
0 |
ASTCC - IVR prompts |
| 4:23PM |
0 |
Asterisk-Users] RE:Asterisk Voice mail with CCM |
| 3:56PM |
1 |
Dialogic cards compatibility |
| 3:43PM |
1 |
Pre-install questions |
| 3:36PM |
2 |
Linksys PAP2 Dual Incoming Calls |
| 3:25PM |
0 |
Q931 Setup message |
| 3:14PM |
0 |
Maximum amount of users on one asterisk server? |
| 3:03PM |
2 |
broadvoice config problem. |
| 2:52PM |
0 |
Re: Using manager interface to play aanouncmentsinaMeetMe |
| 2:41PM |
1 |
Check for stutter dialtone on ZAP FXO channel |
| 2:35PM |
4 |
RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars |
| 1:58PM |
1 |
Sip transfer and redirect in a Company setting |
| 1:40PM |
0 |
RE: Ebay listing selling Asterisk @ Home and AM P for over 1000 dollars |
| 1:01PM |
1 |
RE: Ebay listing selling Asterisk @ Home and AMP for over 1000 dollars |
| 12:55PM |
1 |
Low cost box for hosting Asterisk and at leastoneTDM400p |
| 12:39PM |
3 |
BROADVOICE - Incomming calls are dropped after 1-2 min |
| 12:34PM |
1 |
"Refresh" asterisk internal database? |
| 11:55AM |
2 |
Play Sound File Without Answer Channel |
| 11:54AM |
0 |
Re: Asterisk-Users Digest, Vol 9, Issue 96 |
| 11:41AM |
3 |
Low cost box for hosting Asterisk and at leastone TDM400p |
| 11:24AM |
1 |
Suggestions about where to start from |
| 11:19AM |
0 |
debugging broken/distorted sound with SIP |
| 11:11AM |
1 |
Connection to SIP Gateway |
| 11:02AM |
0 |
Asterisk T.38 framing |
| 10:47AM |
1 |
Bounty: Request for PRI Debug |
| 10:34AM |
2 |
499 Error on X-lite / asterisk setup |
| 10:29AM |
1 |
Line Noise HELP! |
| 10:09AM |
1 |
IPswitch Monitor Extension |
| 9:56AM |
0 |
DID via SIP/IAX |
| 9:53AM |
0 |
Roadmap for Asterisk?? |
| 9:47AM |
1 |
Play Sound File Without Answer Channel ?? |
| 9:46AM |
1 |
TDM400p reliability???? |
| 9:38AM |
2 |
Low cost box for hosting Asterisk and at least one TDM400p |
| 9:31AM |
0 |
Need to Reduce Latency |
| 9:30AM |
0 |
Rebooting Asterisk box shows Asterisk failing to shutdown |
| 8:34AM |
1 |
Is it possible to PickupChan / Dial Pickup / Steal a call that has not been answered? |
| 8:27AM |
3 |
Getting CVS HEAD |
| 8:25AM |
2 |
timed Loop |
| 8:22AM |
1 |
Why 's' doesn't take over unknown extension in context ? |
| 8:16AM |
2 |
Manipulate Asterisk Database from manager? |
| 8:13AM |
0 |
Intercom with Aastra 480e? |
| 7:54AM |
4 |
(no subject) |
| 7:52AM |
2 |
Problem with X101P |
| 7:49AM |
1 |
wcfxo problem |
| 7:34AM |
1 |
Sangoma A101 + Rhino channelbank |
| 7:15AM |
1 |
Interface bonding + asterisk |
| 6:31AM |
1 |
Shared call appearances |
| 5:50AM |
0 |
IPS version 0.79 released |
| 5:17AM |
2 |
SIP Attended/Supervised transfer & features.conf |
| 5:04AM |
0 |
TDM02B on 2 a/b ports of a PBX not working... help |
| 3:45AM |
0 |
Username containing an "@" |
| 3:31AM |
0 |
Direct Broadband connection of ip phone to LiveVoip? |
| 2:28AM |
0 |
call forwarding and parking |
| 2:15AM |
3 |
CDR and TDS |
| 2:12AM |
2 |
Aculab |
| 2:06AM |
1 |
UK CallerID patch with 1.0.7 / 1-0 CVS |
| 1:30AM |
3 |
Can I exit from asterisk console without stopping asterisk? |
| 1:26AM |
1 |
Supply ringing noise to IAX callers |
| 1:19AM |
0 |
voicetronix dtmf |
| 1:18AM |
0 |
Snom 'virtual' extension monitoring? |
| 12:53AM |
2 |
IAX calls between asterisk boxes works 1 way only |
| 12:38AM |
0 |
Conferance DialPlan |
| 12:23AM |
1 |
TDM400P power supply |
| 12:21AM |
3 |
Setgroup & Checkgroup |
| |
| Sunday April 10 2005 |
| Time | Replies | Subject |
|---|
| 11:09PM |
2 |
VAD/DTX implementation through zaptel cards |
| 10:40PM |
1 |
SIP outbound call audio quality change |
| 9:55PM |
1 |
From OH323 to SIP or OH323 without gatekeeper |
| 9:05PM |
1 |
Cannot open chan_zap: |
| 8:32PM |
1 |
Callback application |
| 8:13PM |
0 |
problem with astrisk on MFC R2 |
| 8:08PM |
1 |
problem with unicall |
| 8:01PM |
0 |
Broadvoice problem: Bad request! |
| 7:28PM |
0 |
restructuring my dialplan |
| 6:36PM |
3 |
VM answer call after 20 sec... |
| 5:31PM |
3 |
no ring on inbound SIP calls |
| 5:28PM |
3 |
SIP outgoing problem |
| 5:05PM |
1 |
PTSN POTS Differences |
| 4:50PM |
1 |
unexpected crash ...... |
| 3:12PM |
2 |
snom360 & hint priority |
| 2:34PM |
2 |
Problems trying to compile H323 from CVS-STABLE |
| 2:24PM |
0 |
Need /etc/zaptel.conf for TE110P |
| 2:21PM |
1 |
International callback strategies |
| 2:01PM |
1 |
Fax detect/transfer problem? |
| 12:50PM |
0 |
binding virtual IP address |
| 11:47AM |
2 |
sipura 3000 - "Call Leg/Transaction Does Not Exist" - only happens sometimes |
| 11:25AM |
8 |
Sipura SPA-841 Phone Review |
| 11:24AM |
0 |
Yet another version of IPS Freeware |
| 11:01AM |
5 |
Multiple Servers and 1 Central Voicemail |
| 10:28AM |
0 |
problem with unicall and asterisk |
| 9:52AM |
0 |
SMS suddenly not sending out |
| 9:17AM |
2 |
S100I - competitive price? |
| 8:19AM |
0 |
How To conferance |
| 7:49AM |
1 |
UK PSTN Calling From OH323 Problem |
| 7:08AM |
0 |
decimal arithmetic operations in Asterisk |
| 7:06AM |
0 |
Why do calls go silent after 10 minutes |
| 6:43AM |
3 |
search the mailing list |
| 5:30AM |
1 |
ignorepat changing the sound of dialtone |
| 5:29AM |
2 |
Asterisk::LCR - Least Cost Routing for Asterisk |
| 5:22AM |
2 |
Asterisk becomes after one month unstabled |
| 5:15AM |
2 |
Snom only one way audio |
| 5:11AM |
0 |
CVS compile issue on res_odbc.o |
| 4:33AM |
0 |
ast-rad-acc.pl problem |
| 3:51AM |
0 |
IPSwitchBoard Version 0.77 Released |
| 2:57AM |
2 |
Cann't get CallerID on Zap channel, Please Help!! |
| 1:50AM |
0 |
Fax, which one do I need? |
| 1:47AM |
3 |
Can you comment on this Qos script? How does one shape RTP? |
| 1:44AM |
0 |
Any free SIP softphone with IM capacity for Windows? |
| 1:42AM |
1 |
How to upgrade safe? |
| 12:54AM |
0 |
question about oh323 |
| |
| Saturday April 9 2005 |
| Time | Replies | Subject |
|---|
| 11:19PM |
1 |
Using zap channels for fax |
| 8:13PM |
1 |
Multiple Servers and One Central Voicemail |
| 7:18PM |
0 |
Confusion re; 407 Proxy Authentication Required |
| 6:56PM |
2 |
Terrible crackling on analogue line and X100Pcard |
| 6:39PM |
0 |
Stanaphone - eureka |
| 4:20PM |
0 |
Using manager interface to play aanouncments in aMeetMe |
| 1:07PM |
1 |
Syntax error near unexpected token 'fi' |
| 12:46PM |
1 |
OT: ManxPower 2005 European Tour |
| 12:05PM |
1 |
AgentLogin to MeetMe conference? |
| 12:04PM |
3 |
CallerID name lookup AGI script |
| 11:57AM |
2 |
FWD no longer doing IAX? |
| 11:04AM |
6 |
DS3000P - 20 E1 capacity on single card |
| 11:01AM |
2 |
Asterisk Dual Servers |
| 10:57AM |
1 |
SPA and NAT traversal |
| 10:24AM |
1 |
Asterisk as protocol conventer beetwen SIP and H.323 |
| 8:50AM |
2 |
Dialing With Backgound Music |
| 8:28AM |
1 |
How to change language using manager interface? |
| 6:56AM |
4 |
Terrible crackling on analogue line and X100P card |
| 5:58AM |
0 |
Shorewall settings? |
| 5:51AM |
0 |
dyndns alias clients: needs to register in iax.conf as well? |
| 5:43AM |
1 |
Call rejected by XXX: No authority found |
| 3:47AM |
0 |
HOW TO SET THE TIME TO DIAL AFTER astcc-accountnum and astcc-phonenum |
| 2:59AM |
2 |
Hardware dimesioning issues |
| 1:40AM |
1 |
fax pass through on te410p |
| 12:48AM |
1 |
sip phone extensions at a remote site |
| 12:40AM |
0 |
Astcc Patch |
| 12:06AM |
2 |
g726 > gsm not working with sipura |
| |
| Friday April 8 2005 |
| Time | Replies | Subject |
|---|
| 11:17PM |
0 |
zap to sip caller id "forwarding" |
| 10:11PM |
0 |
SIP Softphone for testing with Asterisk |
| 8:27PM |
1 |
Using manager interface to play aanouncmentsin aMeetMe |
| 7:49PM |
0 |
Frame slip on Wildcard TDM400P? |
| 7:42PM |
0 |
Phone card implementation issues in IAX |
| 7:40PM |
3 |
"s" extension doesn't work with ata |
| 5:30PM |
3 |
How many FXS/FXO ports do I need? |
| 5:19PM |
1 |
Running a Marco from the dial command |
| 5:01PM |
6 |
Asterisk Memory Requirements |
| 4:45PM |
2 |
Asterisk and RT (Request Tracker) setup? |
| 4:20PM |
0 |
debugging voice quality issues |
| 3:33PM |
4 |
UK ISDN with Asterisk |
| 2:16PM |
2 |
Convertnig from Norstar to * to save money |
| 1:20PM |
3 |
Warning, flexible rate not heavily tested! |
| 12:22PM |
2 |
codec translation hints |
| 12:14PM |
2 |
RE:: Connecting asterisk to existing PBX - newbie |
| 11:49AM |
1 |
Connecting Asterisk to a SIP Gateway |
| 11:43AM |
0 |
Allison sounds in native format |
| 11:20AM |
1 |
Dell suggestions for Quad T1 system |
| 11:07AM |
1 |
SIP peer doesn't report busy properly |
| 10:57AM |
0 |
call forwardin and parking |
| 10:23AM |
0 |
Asterisk - SER - sharing single mySQL db and tables |
| 9:48AM |
2 |
Cannot access voicemail |
| 9:45AM |
2 |
Asterisk based CallAccounting software- 1strelease |
| 9:38AM |
3 |
Long wait for ring |
| 9:38AM |
1 |
asterisk missing dtmf what would cause that |
| 9:35AM |
2 |
iaxtel.com - low bandwidth codecs |
| 9:17AM |
0 |
Echo on analog line related to FXO card? |
| 9:16AM |
0 |
Asterisk 1.0.5-BRIstuffed-0.2.0-RC7e |
| 9:12AM |
1 |
PCI-PRI cards - what to buy???? |
| 8:57AM |
2 |
SNOM 190: Unknown SIP command 'PUBLISH' |
| 8:35AM |
4 |
Channel bank replacement |
| 8:23AM |
1 |
Several INVITE messages sent by Asterisk |
| 8:12AM |
0 |
Test settings |
| 8:01AM |
3 |
PRI card and TDM400P in same box |
| 7:20AM |
0 |
Asterisk@Home .8 SPA-2000 |
| 7:13AM |
2 |
AMP 1.10.007 problem on cdr_mysql_table.sql |
| 7:06AM |
5 |
Any opinions on quality/service of Teliax? |
| 7:04AM |
2 |
inquire about connected channel (show channels) |
| 7:03AM |
2 |
snom and "hint" priority |
| 6:49AM |
2 |
X100P doesn't check for dialtone |
| 5:44AM |
0 |
Can a SIP Phone talk directly with anoyher SIP phone (ext a to Ext b) |
| 5:11AM |
0 |
linejack and iax2 ! |
| 5:06AM |
2 |
Call from publicIP to PrivateIP |
| 4:59AM |
1 |
Difference Between NAT=yes and QUALIFY=yes and STUN... |
| 4:32AM |
0 |
NVFaxEmail |
| 3:10AM |
1 |
oh323 DTMF bug |
| 3:08AM |
2 |
Asterisk and CAS |
| 2:55AM |
0 |
compiling oh323 Undefined symbol in res_features & Others |
| 2:17AM |
1 |
Fw: Registration Problem with Firefly Softphone |
| 1:24AM |
0 |
G723 call through GW |
| 1:12AM |
2 |
Delayed dial under Asterisk ? |
| 12:57AM |
11 |
Asterisk based Call Accounting software - 1st release |
| 12:44AM |
1 |
Undefined symbol in res_features & Others |
| 12:21AM |
1 |
external access to voicemail? |
| |
| Thursday April 7 2005 |
| Time | Replies | Subject |
|---|
| 11:44PM |
2 |
iax / realtime problems |
| 11:39PM |
2 |
"404 User Not Found" when calling between two X-Lites |
| 11:03PM |
1 |
Re: Livevoip IAX DTMF troubles |
| 9:32PM |
2 |
Off Topic - Employment Opportunity - PERL, Melbourne, AU. |
| 9:26PM |
1 |
Looking for feedback on IAX2 Phones from Netweb |
| 8:44PM |
0 |
APPRADIUS cdr_radius.so |
| 8:42PM |
11 |
Asterisk Google Group? |
| 8:36PM |
1 |
Canreinvite issue |
| 8:31PM |
1 |
Asterisk quit abnormally |
| 8:08PM |
1 |
how to pass G723.1 |
| 7:11PM |
0 |
RE: Asterisk-Users Digest, Vol 9, Issue 67 |
| 7:10PM |
0 |
RE: Asterisk-Users Digest, Vol 9, Issue 67 |
| 6:59PM |
2 |
failover outbound dialplan |
| 6:45PM |
2 |
stand alone Voice Mail |
| 6:43PM |
0 |
Melbourne Asterisk Consultants |
| 6:38PM |
5 |
Answering without ringing from PRI |
| 6:11PM |
1 |
Asterisk Max TNT |
| 6:10PM |
1 |
Livevoip responds to DTMF via IAX issue |
| 4:50PM |
0 |
Can somebody with HEAD please test MOH on agent calls? M3976 |
| 4:24PM |
6 |
Getting a good deal on a PRI |
| 4:19PM |
2 |
Using manager interface to play aanouncmentsin a MeetMe |
| 3:55PM |
2 |
Local Number Ports |
| 3:54PM |
0 |
Low volume in recorded messages |
| 3:37PM |
0 |
Out of Office AutoReply: fedora 3 |
| 3:22PM |
1 |
zaptel.conf digium and quadBri together (e1 and isdn together) |
| 3:20PM |
2 |
How to turn off automatic pick up for Incoming calls A@H v0.6 |
| 3:17PM |
3 |
Using manager interface to play aanouncments in a MeetMe |
| 3:04PM |
3 |
x100p disconnect on "D" tone |
| 3:02PM |
0 |
Voice controlled calling? Pt 2 |
| 2:30PM |
0 |
Conferancing with different interface |
| 2:09PM |
1 |
FW: Out of Office AutoReply: Voice controlled calling? |
| 1:38PM |
1 |
"Mic-To-Speaker-loop" on ZAP lines??? |
| 1:34PM |
7 |
Voice controlled calling? |
| 1:02PM |
1 |
TE405P vs TE410P |
| 12:57PM |
2 |
SIP UA behind NAT and REINVITE ??? |
| 12:19PM |
4 |
Database lookups? |
| 12:18PM |
0 |
Remembering State |
| 11:29AM |
2 |
Zap (analog line) and volume |
| 11:28AM |
1 |
Asterisk Vs. Cisco, et. al.? |
| 11:01AM |
0 |
IPSwitchBoard Version 0.76 released |
| 10:54AM |
5 |
oh323 compilation |
| 10:52AM |
0 |
How Can I make 2 instances of the nufone H323 channel run? |
| 10:15AM |
2 |
Lag in meetme |
| 9:41AM |
2 |
Can asterisk get code for cmd Authenticate from Database |
| 9:02AM |
1 |
AES vs AEC |
| 8:39AM |
0 |
CDR mysql userfield column truncated at 239 characters |
| 8:26AM |
0 |
open source Asterisk Application of the year ? |
| 8:18AM |
0 |
patch to add distinctive ringing to queues |
| 8:15AM |
3 |
PRI Advice... |
| 8:07AM |
7 |
unlimited iax termination |
| 7:45AM |
0 |
Voicemail localization |
| 7:43AM |
2 |
Fax to email problem |
| 7:38AM |
1 |
SetCdrUserField |
| 7:37AM |
0 |
X100p, IRQ and Noises |
| 7:35AM |
2 |
IVR - newbie question |
| 7:20AM |
5 |
My Sangoma Experience - Review |
| 7:09AM |
2 |
open source Asterisk Application of the year? |
| 7:08AM |
1 |
SpanDSP HELP |
| 7:06AM |
3 |
[OT]: Wiki Etiquette |
| 6:39AM |
1 |
mangle + to 00 |
| 6:29AM |
0 |
Cisco ATA 186: Only incoming - no outgoing call |
| 6:01AM |
2 |
Micronet 128K TA Card |
| 5:55AM |
0 |
call behind NAT |
| 5:53AM |
1 |
IAX2 trunk frames documentation. |
| 5:08AM |
1 |
GSM Hardware Setup |
| 4:37AM |
1 |
voip phone reviews |
| 4:15AM |
0 |
Voice mail through Call Manager |
| 4:13AM |
1 |
"404 User Not Found" when calling between two SIP UA's |
| 4:12AM |
0 |
slightly-ot: Where to buy sip phones in Massachusetts |
| 4:01AM |
5 |
T.38 fax with SIP devices |
| 3:41AM |
1 |
keeping dynamic queue members over restart? |
| 2:54AM |
0 |
Help: Problem with X101P |
| 2:50AM |
1 |
[again] Sangoma PRI vs TE410? |
| 2:44AM |
0 |
Integration of Alcatel 4400 with Asterisk |
| 2:19AM |
3 |
Linux & Asterisk |
| 1:54AM |
0 |
How Can i phone traditional PSTN Phone using TDM11B |
| 1:42AM |
0 |
Fax - Capi |
| 1:19AM |
0 |
Manually adjusting volume in an IAX channel |
| 1:19AM |
1 |
about mpg123 |
| 1:15AM |
3 |
capi segfault when incoming call is answered |
| 12:43AM |
2 |
Access Voicemail From Outside |
| 12:43AM |
5 |
Call Interception |
| 12:30AM |
1 |
Acccess Voice Mail From Outside Line |
| 12:28AM |
2 |
Measure the Signal of Zap |
| 12:21AM |
3 |
Digium TDM400 Failover on Power Loss |
| 12:13AM |
0 |
Cant Hear Any Sounds |
| 12:03AM |
2 |
Latest CVS chokes Sipura SPA-841 |
| |
| Wednesday April 6 2005 |
| Time | Replies | Subject |
|---|
| 11:57PM |
0 |
X100P call keeps ringing and ringing |
| 10:21PM |
3 |
Curry and Asterisk |
| 9:49PM |
1 |
Configuring the Sipura for static IP and registering with Asterisk. |
| 9:39PM |
3 |
Help using wav files for IVR |
| 9:37PM |
2 |
Beeps during Sip to Sip phone calls |
| 9:18PM |
0 |
Any gsm -> g7231 codec translator? |
| 9:01PM |
1 |
rout call from ser to asterisk |
| 8:40PM |
2 |
MWI for SER and Asterisk - ast_data vs "realtime" |
| 8:18PM |
0 |
Problems with new Asterisk@Home install and Broadvoice, no incoming calls |
| 7:59PM |
3 |
Account Codes with SIP |
| 7:42PM |
1 |
Direct Channel Answering |
| 7:04PM |
3 |
Receiving calls from and to H323 devices |
| 6:12PM |
4 |
Asterisk on Slack 10.0 |
| 5:30PM |
0 |
Asterisk and broadsoft |
| 4:12PM |
1 |
DIDs in 510, 408, 916 415 area code |
| 4:06PM |
2 |
Realtime UPDATE |
| 4:03PM |
0 |
Cisco 7940 SIP and No compatible codecs! |
| 3:36PM |
1 |
Context overlap? |
| 3:30PM |
1 |
How to avoid that certain calls come into the voicemail (e.g. wakeup calls)? |
| 2:07PM |
1 |
AMP & Handset Provisioning |
| 1:33PM |
1 |
"Choppy" sounds after transferring to ISDN client or after a time |
| 1:27PM |
3 |
Cisco 7940 Outgoing Audio |
| 1:23PM |
12 |
Liveviop problem |
| 1:17PM |
2 |
SIP messages truncated to 256 characters |
| 12:47PM |
5 |
Asterisk .call files |
| 12:44PM |
0 |
Asterisk, ACD, Queues and Call Transfer Issue |
| 12:26PM |
0 |
Ingate Firewall and Asterisk Integration |
| 12:23PM |
0 |
Problems using Asterisk 1.0.3 with Vocal 1.4 |
| 12:07PM |
4 |
SRV Bounty |
| 11:55AM |
2 |
Connecting asterisk to existing PBX - newbie question |
| 11:49AM |
4 |
SIP - SIP Problems |
| 11:40AM |
2 |
Web interface for realtime Mysql friends/peer |
| 11:12AM |
0 |
NOTICE: chan_sip:7654 handle_request: Registeration failed HELP !! |
| 10:41AM |
0 |
I want to call another pc with TDM11B Card |
| 10:34AM |
6 |
Keypad disabled on AriaVoice SIP phone |
| 10:28AM |
2 |
Script Perl Autodialer |
| 10:02AM |
0 |
Fritz Card ISDN in UK - Unable to |
| 10:00AM |
2 |
Any success with BRI in the US? |
| 9:43AM |
0 |
Cant Hear Any Sound |
| 9:24AM |
0 |
Call gets cut off after 5 minute |
| 9:05AM |
2 |
ser <-> asterisk configs anyone? |
| 8:47AM |
0 |
V-9970 Paging Setup |
| 8:33AM |
1 |
dial out and "all circuits are busy" |
| 8:23AM |
0 |
looking for some draft (sip - iax2 mapping) |
| 8:00AM |
6 |
Asterisk and phone system |
| 7:45AM |
1 |
Latest Bristuff crashes on modprobe -r qozap ? |
| 6:55AM |
1 |
Multiple BroadVoice Accounts Problem with Incoming calls |
| 6:41AM |
0 |
SMS with VOIP phone WIP 5000 from hitachi |
| 6:37AM |
3 |
Cisco 7960 forgets VLAN setting |
| 6:33AM |
2 |
IPTABLES Firewall |
| 6:29AM |
1 |
Syntax checker for Asterisk config files |
| 6:21AM |
6 |
how can i connect a cost display on asterisk |
| 6:00AM |
0 |
Fritz Card ISDN in UK - Unable to dial. |
| 5:35AM |
2 |
Snom 190 + NAT |
| 5:34AM |
3 |
Zaptel Compile on a virtual dedicated host. |
| 5:01AM |
0 |
IPSwitchBoard - Now in Spanish |
| 4:38AM |
1 |
Problem compiling 2nd AVM Fritz |
| 3:40AM |
1 |
Got S-frame while link down |
| 3:30AM |
1 |
asterisk is giving error- unable to write audio data codec_speex.so |
| 3:26AM |
1 |
wcte11xp works only after cold reboot |
| 3:08AM |
0 |
Asterisk & Windows Messenger 5: Which is the correct/preferred DTMFmode setting? |
| 2:58AM |
3 |
Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors |
| 2:38AM |
3 |
fedora 3 |
| 2:28AM |
0 |
(no subject) |
| 2:05AM |
0 |
FXO-FXS parameters |
| 1:59AM |
0 |
X-Lite codec related... |
| 1:31AM |
1 |
How can I add entry for a UA into asterisk when asterisk is running? |
| 12:45AM |
1 |
IAX2 and NATs that increment ports |
| 12:25AM |
2 |
How can I make base calls with X-Lite via Asterisk? |
| 12:10AM |
0 |
Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors. |
| 12:09AM |
4 |
Voicemail and SJphone |
| |
| Tuesday April 5 2005 |
| Time | Replies | Subject |
|---|
| 11:53PM |
2 |
SCCP |
| 10:33PM |
1 |
query about cdr configuration |
| 10:29PM |
0 |
Asterisk @ Home 0.8 Question |
| 10:05PM |
1 |
Help with simple callback application from newbie |
| 9:05PM |
6 |
About Audio Latency from PSTN to SIP |
| 7:54PM |
3 |
Grandstream HandyTone-488, * -> FXO problems |
| 7:25PM |
0 |
Re: Asterisk-Users Digest, Vol 9, Issue 44 |
| 7:21PM |
0 |
.conf to realtime conversion script? |
| 7:19PM |
1 |
Stopping Retransmission Found: 102 Error with Polycom IP300 |
| 6:37PM |
0 |
RE: Digium ISDN card |
| 5:50PM |
0 |
Asterisk and Nortel Meridian DTMF |
| 5:17PM |
1 |
Automatic Start |
| 4:40PM |
3 |
Petition for IAX firmware |
| 4:03PM |
1 |
Cell phone friendly MOH |
| 2:52PM |
1 |
Atcom AT-320 multiple lines? |
| 2:44PM |
4 |
TE405P and Dell Poweredge 6450 Incompatible? |
| 2:24PM |
1 |
new user TDM400P and T1 card problems |
| 1:59PM |
2 |
Multiple CDR Locations |
| 1:09PM |
5 |
Dialogic D/300SC-1E1 and D/600SC-2E1 with * |
| 1:04PM |
0 |
contracting |
| 12:39PM |
2 |
Should PRI running over t100p be able to survive short yellow alarms? |
| 12:30PM |
0 |
Polycom IP500 does not show elapsed call time on LCD. |
| 12:24PM |
3 |
AGI call problem |
| 12:19PM |
0 |
chan_iax2 stops listening to packets |
| 11:28AM |
2 |
Queue works, but the caller hears silence instead of ring tone |
| 11:24AM |
1 |
PortaSIP/PortaBilling incompatibility (provider: sipcall.ch) |
| 11:22AM |
0 |
Accessing Conferencing Bridges |
| 10:46AM |
2 |
Sound quality with Xten Xlite softphones... |
| 10:37AM |
1 |
AAH 0.6 to 0.8 Upgrade |
| 10:24AM |
5 |
multiple PBXs on one server. |
| 10:24AM |
9 |
asterisk sounds |
| 10:19AM |
0 |
voicemail access |
| 10:06AM |
0 |
Concurrent calls: best provider? |
| 10:02AM |
0 |
Question about sending Caller-ID Name over PRI NI2 bug 3554 |
| 10:00AM |
0 |
Command Reference |
| 9:52AM |
0 |
Digtial Receptionist Recorded Greeting LocationProblem |
| 9:44AM |
3 |
using asterisk as a gateway for residential IP telephony clients |
| 9:42AM |
2 |
realtime management for sip with mysql |
| 9:42AM |
3 |
Cisco 7940/60 failed to take SIP image from tftp server |
| 9:24AM |
0 |
Digtial Receptionist Recorded Greeting Location Problem |
| 9:13AM |
2 |
sip <-> oh323 / real-time / g729 - one way audio |
| 9:04AM |
6 |
TE110P/Hipath3750 - Yellow Alarm |
| 8:55AM |
1 |
Agents |
| 8:18AM |
0 |
Awaiting Ack |
| 8:09AM |
5 |
How do I retrieve voice mail in Asterisk |
| 8:01AM |
1 |
VOIP 911 Mandatory in Canada |
| 7:33AM |
1 |
Best Performance |
| 7:23AM |
1 |
mysql and confs |
| 7:07AM |
0 |
SIP -> PSTN: mISDN DTMF tones generation |
| 7:06AM |
0 |
re: Problem: Compiling error for SpanDSP |
| 6:53AM |
2 |
Asterisk and C# (Dotnet) |
| 6:53AM |
1 |
OT: CRTC mandates 911/E911 for VoIP in Canada |
| 6:50AM |
1 |
prevent callerid spoofing between asterisks |
| 6:32AM |
1 |
VoIP network configuration using Asterisk and SIP |
| 6:11AM |
0 |
Tracing dial plan branching |
| 5:40AM |
2 |
D Channel Becoming "CORRUPTED"? |
| 5:36AM |
1 |
Asterisk and HylaFAX integration |
| 5:17AM |
0 |
Problems registering Asterisk with Vocal |
| 5:05AM |
0 |
Checking for the in-use conference |
| 4:53AM |
2 |
fault tolerant asterisk system |
| 4:43AM |
2 |
MOH and OptiPoint400 std SIP |
| 4:31AM |
1 |
how to make asterisk only for SIP and direct RTP |
| 3:52AM |
4 |
busy line status on CISCO 7940/7960 |
| 3:35AM |
8 |
WRT54GP2A-AT |
| 3:21AM |
0 |
i3micro VTA111 and Asterisk |
| 3:03AM |
0 |
Realtime queues? |
| 3:00AM |
1 |
SIP / PTT over Cellular |
| 2:37AM |
0 |
No Voice to POTS. |
| 2:33AM |
1 |
"Multiplexing" (or what ever the term is) FXO ports into a "Trunk" |
| 2:08AM |
1 |
zaptel not starting issues |
| 1:58AM |
0 |
asterisk FWD intelligent routing |
| 1:57AM |
0 |
Asterisk Realtime - extensions configurationhelp / solved |
| 1:56AM |
0 |
sniffing bridged video call on zap channels |
| 1:54AM |
0 |
E100p zapata errors |
| 1:37AM |
1 |
Incomming Call issues |
| 12:51AM |
1 |
SIP Phone binary |
| 12:41AM |
2 |
Outgoing calls on PRI |
| 12:01AM |
0 |
Freesoftswitch.com |
| |
| Monday April 4 2005 |
| Time | Replies | Subject |
|---|
| 11:51PM |
2 |
livevoip callerid |
| 10:46PM |
2 |
Outgoing faxes with chan_capi? |
| 10:34PM |
1 |
help regading outbound calls |
| 10:31PM |
2 |
Asterisk Discussion Forums provided by Digium |
| 10:04PM |
5 |
asterisk on UML |
| 8:45PM |
2 |
Weird Errors with Realtime and MySQL |
| 7:51PM |
0 |
Compatability with AudioCodes MP-108 |
| 7:26PM |
3 |
Voicemailbox detection: |
| 6:18PM |
3 |
Transient SIP Registration Issues |
| 6:05PM |
3 |
Detecting Downed SIP Phone |
| 5:17PM |
0 |
Digium Hires Kevin Flemming |
| 4:49PM |
4 |
AAH 0.6 - Change Network Gateway |
| 4:45PM |
4 |
Set system time over the phone |
| 4:36PM |
0 |
SIP/SDP packaged in Multipart/Mixed mime type |
| 4:33PM |
1 |
ASTCC - not saving configuration |
| 4:12PM |
0 |
zapata.conf parameter order - feature or bug? |
| 3:40PM |
5 |
Channel bank question |
| 3:32PM |
1 |
SIP and firewall |
| 3:20PM |
1 |
chan_sccp compile error |
| 2:58PM |
3 |
TE405P takes ~5mins to load. |
| 2:43PM |
1 |
Authentication with DB Support |
| 2:40PM |
0 |
DTMF Caller ID in Brazil |
| 2:37PM |
2 |
Operators guide |
| 2:23PM |
0 |
L2 QoS switch |
| 1:49PM |
0 |
CVS-HEAD compile? Was: Checkgroup and transfers |
| 1:42PM |
0 |
IP Address of caller variable? |
| 1:25PM |
0 |
Asterisk queue crash |
| 1:19PM |
0 |
SIP microphone not working after ZAP and sounddriver installation |
| 1:05PM |
1 |
SIPTone II and PoE |
| 1:04PM |
2 |
call redirection from outside line? |
| 12:49PM |
3 |
Livevoip DTMF via IAX almost |
| 12:25PM |
0 |
distinctive ringing in a queue? |
| 11:34AM |
2 |
ZAP problem (No channel type registered for 'Zap') |
| 11:34AM |
3 |
Asterisk on WRT54GS |
| 11:19AM |
2 |
compilation of asterisk |
| 11:03AM |
0 |
vmail.cgi - can't forward messages |
| 10:59AM |
5 |
monmp3thread: Request to schedule in the past?!?! |
| 10:46AM |
1 |
mISDN + chan_misdn and DTMF |
| 10:31AM |
0 |
How to control codecs when originating a call? |
| 10:23AM |
0 |
SIP phones to Asterisk using MAC addressinsteadofIP address |
| 10:21AM |
0 |
IPSwitchBoard speaks other languages |
| 9:59AM |
2 |
Ring Twice |
| 9:57AM |
1 |
Can't see ANI2 (aka info digits) from PRI t1 |
| 9:51AM |
2 |
Can I set queue not to hangup? |
| 9:51AM |
5 |
Asterisk and clarent |
| 9:50AM |
3 |
rookie getting started question |
| 9:42AM |
2 |
FGD Support |
| 9:00AM |
1 |
Asterisk Realtime - extensions configurationhelp |
| 8:44AM |
0 |
Asterisk and Ingate registration |
| 8:19AM |
1 |
Problem registering 'SJPhone'? |
| 8:18AM |
0 |
Does the agent queue app support Aftercall and AUX agent status? |
| 8:06AM |
1 |
Just a test |
| 7:56AM |
2 |
newbie - want to use asterisk as an internal PBX |
| 7:46AM |
1 |
configuring md5 authentication |
| 7:33AM |
1 |
SIP phones to Asterisk using MAC address insteadof IP address |
| 7:21AM |
1 |
Asterisk Realtime - extensions configuration help |
| 7:15AM |
0 |
Distributed services such as voicemail using Asterisk |
| 7:08AM |
0 |
SIP phones to Asterisk using MAC address instead of IP address |
| 6:58AM |
1 |
Browser based configuration of Asterisk |
| 6:47AM |
1 |
IAXy audio troubles (only on INCOMING calls) |
| 6:43AM |
1 |
X-Lite to Zap, no Voice on other phone! |
| 6:37AM |
0 |
PRI: received SETUP message for call that is not a new call, wicked! |
| 5:54AM |
0 |
Re: ASTCC question: Trunk LOCAL |
| 5:43AM |
2 |
Best way for nated sip peers thru a database |
| 5:06AM |
2 |
Sending faxes and call accounting |
| 4:27AM |
1 |
Zap - What is going on? |
| 3:51AM |
0 |
How do you do Line Hunting in Asterisk? |
| 3:23AM |
1 |
Supervised transfer problems |
| 2:56AM |
4 |
Realtime & voicemail |
| 2:54AM |
1 |
Difficulty in configuring Asterisk to ensure that Call Transfers (SIP phone) are properly recorded and billable |
| 2:41AM |
0 |
Planet VIP 450 |
| 2:22AM |
2 |
Asterisk+Sipgate - just one step away.. |
| 12:49AM |
1 |
Zaptel group members - dial out on a availible port via trial and error? |
| 12:38AM |
1 |
error while compiling asterisk-1.0.7 |
| 12:33AM |
1 |
How to send email from the dial plan? |
| |
| Sunday April 3 2005 |
| Time | Replies | Subject |
|---|
| 11:55PM |
0 |
Wellgate 3701 |
| 11:49PM |
1 |
Previous sip reload not yet done |
| 10:50PM |
4 |
how to configure groups using a sip phone |
| 10:48PM |
4 |
V92 modem with asterisk |
| 10:31PM |
2 |
Music On Hold and ATA-186 w/Silence Supression |
| 9:52PM |
6 |
Asterisk@Home Question |
| 9:19PM |
1 |
AGI Dial Plan |
| 8:41PM |
0 |
creating conference call |
| 8:12PM |
1 |
Asterisk <-> Altigen |
| 8:03PM |
0 |
Re: Asterisk-Users Digest, Vol 9, Issue 21 |
| 7:51PM |
0 |
Joshua Chessman |
| 6:24PM |
2 |
Asterisk Realtime Capabilities |
| 5:37PM |
2 |
AS5300+SIP+ASTERISK or AS5300+MGCP |
| 5:10PM |
3 |
problems with call-forward from ccme to * on sip trunk |
| 4:50PM |
0 |
VG248 and Asterisk |
| 2:04PM |
0 |
Asterisk on Suse minimal installation based on Suse Rescue - what to add to be bootable on HD partition ? |
| 1:21PM |
0 |
Looking for res_config_pgsql |
| 12:56PM |
3 |
Detecting when a called mobile is not reachable? |
| 12:33PM |
3 |
Authenticating username |
| 10:20AM |
2 |
SET & CHECK group |
| 10:00AM |
1 |
SIP dialing in two extensions |
| 9:32AM |
0 |
Asterisk with Jasomi Peerpoing |
| 7:14AM |
0 |
Re: Asterisk Discussion Forum |
| 4:26AM |
0 |
IPSwitchboard Version 0.73 Released |
| 3:33AM |
5 |
Router with QoS recommendations |
| 3:06AM |
0 |
Re: Asterisk Discussion Forum |
| 3:03AM |
1 |
Where to post my impovements to ASTCC? |
| |
| Saturday April 2 2005 |
| Time | Replies | Subject |
|---|
| 11:50PM |
2 |
New to asterisk. |
| 11:47PM |
4 |
How does asterisk know the did called on? |
| 11:46PM |
0 |
sipcall.ch configuration problem |
| 10:41PM |
1 |
Fwd: NDN: Re: Delaying answer of incoming calls |
| 10:00PM |
0 |
Replace Adtran 608 With Asterisk |
| 9:37PM |
2 |
Sip registration Problems With Zyxel P2000W |
| 9:11PM |
1 |
Macro Extension with Realtime and Mysql DB |
| 9:09PM |
1 |
How to reset IAXy? |
| 9:00PM |
0 |
TE410P and Fax Server |
| 8:59PM |
0 |
Problem with asterisk -> ohphone |
| 8:42PM |
3 |
Asterisk Discussion Form |
| 8:07PM |
1 |
D-Link router/Voip gateway locked to Lingo? |
| 7:16PM |
1 |
Packetization |
| 6:38PM |
0 |
Long Distance Acces Code |
| 5:30PM |
2 |
Re: Are there online forums instead of, this email |
| 5:18PM |
6 |
Buying some Polycom IP300s |
| 5:03PM |
4 |
xlite regestration fails but calls to thru |
| 4:02PM |
0 |
Sipura - GSM or iLBC? |
| 3:36PM |
1 |
OH323 core dump |
| 12:48PM |
3 |
Zaptel Anti-MMX Optimizations |
| 12:46PM |
1 |
Book Review: VoIP Telephony with Asterisk |
| 11:44AM |
3 |
how to tell what ${DIALSTATUS} is being set |
| 11:20AM |
3 |
Asterisk Auto-Startup on Ubuntu/Debian |
| 11:20AM |
2 |
problem detecting answer on pri card |
| 11:07AM |
0 |
Outbound calls with xlite and Xpro PocketPC |
| 10:38AM |
2 |
Passing varibles *out* of macros |
| 10:26AM |
1 |
SjPhone&H323 |
| 9:33AM |
1 |
Dynamic Zap/{channel} allocation for out going possible? |
| 9:24AM |
7 |
Starting with Asterisk-SIP |
| 8:52AM |
0 |
TDM04B - TDM PCI MASTER ABORT |
| 8:37AM |
7 |
Asterisk Voice mail with CCM |
| 8:14AM |
1 |
Two accounts at one provider and a 302 redirect problem |
| 5:27AM |
4 |
astcc problems |
| 5:14AM |
1 |
{extensions.conf} Dialing plans with queues.... |
| 4:37AM |
1 |
Registration to multiple GKs |
| 4:19AM |
1 |
H.323 call '.....' cleared, reason 8 (Transport failure) |
| 4:01AM |
3 |
:: Strange way of receiving calls :: |
| 3:42AM |
1 |
Delaying answer of incoming calls |
| 3:35AM |
0 |
(no subject) |
| 3:24AM |
0 |
Version 0.72 of IPSwitchBoard Released |
| 3:23AM |
1 |
VAD In Asterisk with Zaptel |
| 3:01AM |
1 |
Little question |
| 2:10AM |
3 |
Shorewall firewall rules |
| |
| Friday April 1 2005 |
| Time | Replies | Subject |
|---|
| 11:30PM |
1 |
at-320 phone configuration difficulty |
| 9:41PM |
7 |
Queues |
| 7:07PM |
0 |
Display agent logged into a queue on the phone |
| 6:35PM |
1 |
OT(?) Your subject line |
| 5:31PM |
1 |
Random outbound: |
| 4:46PM |
0 |
Strange DTA problem |
| 4:02PM |
1 |
Preserving CallerID when forwarding to cellphone |
| 3:45PM |
3 |
Call bridging |
| 3:10PM |
1 |
Datafire 2977 |
| 2:42PM |
4 |
Squeaking / chirping on ZAP Digium TDM400P |
| 2:32PM |
3 |
What's the use of a multi line phone? |
| 1:25PM |
1 |
Codec not negotiating |
| 1:01PM |
0 |
Make voicemail use Maildir... |
| 12:13PM |
3 |
Issues with ringing on FXS ports |
| 10:47AM |
0 |
Faxing through Broadvoice - HT286 |
| 10:07AM |
1 |
"Unable to create channel of type Zap" Message |
| 9:52AM |
0 |
Must be configuring something wrong spanDSP rxfax |
| 9:38AM |
1 |
ADSI Input from 480 keypad? |
| 9:27AM |
1 |
Sending DTMF back in a dialed/answered channel before bridging a call |
| 9:24AM |
1 |
MF Trunk Signaling |
| 8:54AM |
2 |
Does asterisk@home support Dual-Processor installations? |
| 8:48AM |
1 |
Specify Codec In Outbount Calls? |
| 8:42AM |
1 |
Voicemail Email Bouncing |
| 8:31AM |
3 |
Snom and Multiple calls |
| 8:25AM |
1 |
blind transfer question |
| 8:25AM |
0 |
Zyxel Prestige 2002 (ATA) |
| 8:23AM |
0 |
new release of chan_misdn ! |
| 7:36AM |
1 |
Q.931 to SIGTRAN interface |
| 7:29AM |
2 |
Looping messages |
| 7:12AM |
1 |
Dial'ing multiple SIP devices impossible when forward activated |
| 6:45AM |
2 |
Maybe an echo cancellation problem? |
| 6:12AM |
0 |
Optimizing speex (was Re: Erratic CPU load ) |
| 5:59AM |
1 |
LDAP and Asterisk |
| 5:52AM |
0 |
[OT] Announcing MidwestTea.com |
| 5:39AM |
1 |
${DIALSTATUS} |
| 3:57AM |
2 |
queue.conf config |
| 3:40AM |
3 |
Eicon Diva Server BRI Setup |
| 3:28AM |
0 |
[Fwd: Problem with dial out via chan_capi] |
| 3:02AM |
1 |
H.323 call '...' cleared, reason 15 (Call ended due to security checks) |
| 2:26AM |
3 |
Problem with dial out via chan_capi |
| 2:07AM |
4 |
using unixODBC |
| 1:59AM |
0 |
Problems getting FXO channel working - Unable to create channel of type 'Zap' (cause 0) |
| 1:52AM |
5 |
really small box |
| 1:07AM |
0 |
Playback starts before call answer |
| 12:53AM |
1 |
register => with realtime |
| 12:40AM |
21 |
*** Asterisk 2.0 Stable release out now |
| 12:32AM |
1 |
Parking no |
| 12:01AM |
1 |
User Regerstation, allowing non-registered users on * |