| Sunday April 30 2006 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
how to make messages button on ip500 work |
| 10:42PM |
2 |
PRI Issue: D-Channel woes |
| 6:36PM |
2 |
WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????' |
| 2:19PM |
6 |
FreePBX in production? |
| 12:46PM |
1 |
integrated voip originator, to digitize audio once and only once? |
| 10:57AM |
1 |
Asterisk 1.2.7.1 & Fritz!PCI or AVM A1 |
| 10:05AM |
2 |
Asterisk is stripping my area code |
| 10:01AM |
1 |
Change in audio file while listening to it |
| 9:22AM |
1 |
Legacy PBX integration |
| 9:18AM |
1 |
newbie-too much latency |
| 8:41AM |
0 |
Intermittent problem dialling out on a SIP channel |
| 6:21AM |
0 |
Fwd: can modify CHAN_SIP.c to generate a new exten=> ext, 2, dial(tech/peer) ? |
| 5:56AM |
0 |
some sip clients unreachable on sip-reload |
| 5:35AM |
1 |
Error : ast_readaudio_callback: Failed to write frame |
| 4:16AM |
3 |
How to monitor DTMF tones in a call? |
| 12:56AM |
0 |
PRI got event: HDLC Bad FCS (8) on PrimaryD-channel of span |
| |
| Saturday April 29 2006 |
| Time | Replies | Subject |
|---|
| 10:33PM |
2 |
problame with outbound calls on pri |
| 8:55PM |
0 |
NuFone - How to switch to another provider? |
| 8:53PM |
6 |
Compare to Skype |
| 8:11PM |
0 |
[OT]Cisco 2621XM with (2) T1/PRI inetrfaces for sale |
| 7:49PM |
2 |
Codec G729 no longer works. |
| 6:17PM |
8 |
(Semi-OT) QoS Question FTP Living with Asterisk |
| 6:11PM |
2 |
RE: Install/Upgrade |
| 6:02PM |
1 |
Large Asterisk with Regexten, Regcontext, DUNDi, , , , , , , , , but not load balance... |
| 4:50PM |
2 |
Unable to Make Asterisk-addons |
| 11:40AM |
0 |
Audio Muting at seemingly random times |
| 9:58AM |
0 |
Locate Me Function with freePBX |
| 9:04AM |
2 |
How many asterisk process's are "normal"? |
| 8:13AM |
1 |
Telephone support charging system with Asterisk? |
| 7:35AM |
0 |
canreinvite, bandwidth, dial option |
| 5:19AM |
1 |
asterisk to use an outbound proxy |
| 4:13AM |
1 |
NOTIFY Problem |
| 4:12AM |
1 |
Help with Mediatrix 1204 |
| 1:00AM |
0 |
Is there a way to monitor the DTMF tones on a channel? |
| |
| Friday April 28 2006 |
| Time | Replies | Subject |
|---|
| 9:37PM |
1 |
stupid trick of the day (fried polycom) |
| 9:06PM |
1 |
IAX + GSM codec is good quality |
| 8:02PM |
1 |
[SPAM] [asterisk-dev] Disable 407 proxy authentication for outbound domains |
| 7:10PM |
2 |
Random 1-way audio on IAX2 Connections |
| 7:06PM |
2 |
Call Queue Transfer |
| 6:51PM |
1 |
Remote UNIX connection disconnected over and over |
| 5:39PM |
1 |
two box share one real time configuration database. |
| 4:53PM |
0 |
Rhino T1 and 4-port FXO cards |
| 4:34PM |
0 |
How to use the cmd SMS |
| 4:11PM |
2 |
Asterisk DNID/RDNIS with Dial iax2 |
| 3:29PM |
1 |
Bristuff 1.2.7.1? |
| 3:07PM |
1 |
Cell phones and DTMF |
| 2:36PM |
0 |
Asterisk and Panasonic KX-T336 |
| 2:27PM |
0 |
Digium TE210P and faxing, is it possible? |
| 1:26PM |
0 |
DNSMasq - Why the stuff hits the fan when the net connection is down |
| 12:19PM |
1 |
Odd internal vs. External dialplan issue |
| 11:48AM |
0 |
Configuration OpenPri for logger |
| 11:07AM |
3 |
Dual Timing Sources |
| 10:33AM |
1 |
Official TE411P echo settings?? |
| 10:12AM |
0 |
What is i2 ? 911 Candian Style |
| 9:57AM |
0 |
New astGUIclient VICIDIAL Release 1.1.11 |
| 9:20AM |
0 |
OT: Phishing with phones |
| 8:34AM |
1 |
Basic Linux Advice |
| 8:12AM |
3 |
Problems if GXP-2000 phones and Asterisk are not on the same network |
| 7:44AM |
1 |
RESOLVED - TE405P vs. SoundCard problem (in reality - TE405P No Voice Problem) |
| 7:12AM |
1 |
Warning: No path to translate with SJPhone |
| 6:50AM |
1 |
Integrics release Enswitch 2.0 |
| 6:14AM |
2 |
How to transfer outgoing calls |
| 5:29AM |
2 |
Dial 'R' option gone? |
| 4:27AM |
0 |
IVR answers and questions instead of MOH in a queue, how? |
| 3:39AM |
2 |
caching of sip account |
| 3:21AM |
0 |
video support on iax trunk |
| 3:18AM |
0 |
Which h323 channel for asterisk and gnugk ? |
| 2:02AM |
0 |
how to dial the real time iax user |
| 1:43AM |
2 |
Asterisk dialing |
| 1:41AM |
0 |
Help on multiple dialed sip-channels. |
| 1:35AM |
1 |
mISDN: No DID/extension information returns busy to caller |
| |
| Thursday April 27 2006 |
| Time | Replies | Subject |
|---|
| 8:45PM |
0 |
HINTING, how it works... Please explain |
| 6:00PM |
0 |
How can conference room can call out? |
| 5:48PM |
0 |
Info system |
| 4:36PM |
0 |
Call Pickup with CID info |
| 3:35PM |
1 |
Polycom 501 unregistered itself? |
| 1:21PM |
0 |
Looking for input on which way to go with smallbusiness setup |
| 1:20PM |
0 |
Please Help i have a error with unicall and AT&T |
| 1:08PM |
1 |
Looking for input on which way to go with small business setup |
| 12:09PM |
0 |
What happened to my subscription? |
| 11:50AM |
1 |
PrivacyManager & FastAGI: Rewrite or use? |
| 11:22AM |
0 |
createlink option in agents.conf can't be disabled? |
| 11:10AM |
7 |
Recomended Commercial PBX Bundles/Software |
| 11:03AM |
1 |
Snom 320 HOLD and TRANSFER not detected |
| 10:43AM |
1 |
Analog GSM Gateways |
| 10:26AM |
1 |
Excessive Asterisk delay to answer on ZAP inboundcall |
| 9:55AM |
0 |
Receive fax (libtiff problem?) |
| 9:22AM |
7 |
Polycom NTP issue |
| 9:18AM |
12 |
PRIs from two different telco |
| 8:46AM |
0 |
access to caller/pickupgroup in extension.conf |
| 8:03AM |
1 |
Slip/Frame Error between Mitel SX-200 and Asterisk |
| 7:38AM |
1 |
Very stupid question regarding Polycom Soundstation 4000 |
| 7:12AM |
0 |
Guest Account - SIP and IAX |
| 7:07AM |
0 |
chan_sip.c patched with t.38 |
| 6:58AM |
5 |
PRI configuration |
| 6:28AM |
3 |
Seize phone line |
| 6:03AM |
1 |
asterisk spandsp and txfax |
| 5:46AM |
2 |
TE405P vs. SoundCard problem |
| 5:46AM |
2 |
Transfer - context/priority |
| 5:34AM |
0 |
Need help configuring Asterisk with Alepo |
| 5:19AM |
0 |
URGENTS: seek people for video tests with asterisk/ser/rtpproxy + eyebeam |
| 5:10AM |
2 |
Interesting Dial-Plan Question |
| 4:17AM |
0 |
zt_pri-error |
| 3:27AM |
0 |
Autodial feature doesn't return $DIALSTATUS values |
| 2:17AM |
1 |
Asterisk to Dial a number , after getting a mail notification , |
| 2:07AM |
1 |
Asterisk Voice Problems |
| 1:55AM |
0 |
GXP-2000: disable provisioning |
| 1:37AM |
2 |
Asterisk Hangs the whole system |
| 12:32AM |
2 |
SATA hard disk compatibility |
| 12:29AM |
0 |
SV: treating an incoming call as a local extension |
| 12:03AM |
2 |
Extreme delay before * processes call files |
| |
| Wednesday April 26 2006 |
| Time | Replies | Subject |
|---|
| 11:35PM |
0 |
replacing step-by-step giving echo |
| 9:25PM |
1 |
Accessing PARKEDAT variable in AGI |
| 9:21PM |
2 |
treating an incoming call as a local extension |
| 9:19PM |
1 |
getting asterisk to reliably answer a voip line |
| 9:09PM |
0 |
func_odbc and 1.2.7.1 |
| 8:56PM |
2 |
Unable to accept incoming PSTN calls |
| 7:30PM |
4 |
Asterisk as a phone survey system |
| 6:50PM |
1 |
Paging on Aastra analog phones. |
| 5:55PM |
0 |
no audio for ring group. |
| 5:35PM |
1 |
Problem with a TDM-400P |
| 4:28PM |
1 |
Problems with Eicon Diva V-4BRI - 2nd Port |
| 2:17PM |
0 |
Hook Flash via SIP INFO command? |
| 2:06PM |
0 |
cell mobile network (GSM) to Asterisk |
| 2:01PM |
2 |
Status of Queue |
| 12:44PM |
0 |
clipcomm versus sipura/linksys |
| 12:43PM |
0 |
Help! * Won't Start after SVN Trunk Update - SuSE 10 |
| 12:43PM |
0 |
A@H and channel announcement |
| 12:41PM |
1 |
Phone Emergency - Need IAX Help |
| 11:43AM |
1 |
Question about the zaptel-1.2.5-patch |
| 10:27AM |
1 |
cannot transfer to call waiting call on ip500 |
| 10:24AM |
0 |
Callback help |
| 10:00AM |
1 |
Explain to me VoIP termination service. |
| 9:52AM |
0 |
How to configure Asterisk to handle multiple Gizmo accounts? |
| 9:31AM |
1 |
asterisk no longer compiles on gcc 2.95 |
| 9:25AM |
0 |
(Wanted OEM VOIP and SKYPE PRODUCTS) |
| 9:16AM |
0 |
Re: Asterisk-Users Digest, Vol 21, Issue 132 |
| 8:17AM |
1 |
Early media after a dial command |
| 8:05AM |
0 |
kernel - module problem |
| 7:33AM |
2 |
Stuck in Queues |
| 7:25AM |
9 |
Camp on? |
| 6:53AM |
0 |
Re: [Serusers] Sip t38 gateway tests |
| 6:47AM |
1 |
No Caller-ID With Cisco PAP2T-NA |
| 6:27AM |
0 |
RE: SOLVED: No audio when dialing in via PRI with Q.SIG |
| 6:19AM |
0 |
Configuring QoS Params in UIP-200 |
| 6:11AM |
1 |
IAX calls dropping after minutes |
| 5:52AM |
1 |
ODBC Storage for voicemail messages in database |
| 5:41AM |
6 |
I am looking for a webphone on MY SITE |
| 5:36AM |
6 |
Sphinx2 |
| 5:28AM |
2 |
2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my * |
| 5:27AM |
3 |
astcc: need partial pin code |
| 5:21AM |
1 |
Registering to H.323 Cisco gatekeeper |
| 5:02AM |
1 |
7960G SIP Issue |
| 3:04AM |
0 |
do extensions must be numbers in Asterisk@Home? |
| 2:51AM |
1 |
Sip Phones with BLA Support |
| 2:14AM |
0 |
Avoiding deadlock... Problem |
| 1:59AM |
4 |
Excessive Asterisk delay to answer on ZAP inbound call |
| 1:40AM |
0 |
Stange behaviour on 4port BRI Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q |
| 1:17AM |
1 |
Intergrate Asterisk IP PBX with Legacy PBX, continuing existing funtionality of legacy pbx |
| 12:50AM |
1 |
# and call speed |
| 12:24AM |
0 |
CISCO 7960G - SIP Configuration |
| 12:01AM |
0 |
SV: Need some help on queues with agents(SIP members)with multiple phones. |
| |
| Tuesday April 25 2006 |
| Time | Replies | Subject |
|---|
| 11:40PM |
2 |
Need some help on queues with agents(SIP members) with multiple phones. |
| 9:49PM |
3 |
test numbers in different countries! |
| 9:23PM |
0 |
Trying to set up automatic announcement upon |
| 8:25PM |
0 |
Trying to set up automatic announcement upon transfer for IVR in AAH 2.8 |
| 6:46PM |
0 |
Here I am facing problem of Voice Breakage |
| 6:22PM |
5 |
USB conference phone |
| 5:06PM |
1 |
Agents <--> Extensions |
| 4:18PM |
3 |
56K Dialup and VOIP over same PRIs |
| 3:09PM |
0 |
Pressing ## end the call and return to menu |
| 2:32PM |
2 |
Touch tone recognition issues |
| 2:21PM |
1 |
queues that do not play music |
| 2:09PM |
1 |
TE410 and 411 |
| 1:37PM |
1 |
One Way Audio....in the middle of a call |
| 1:22PM |
1 |
Splitting Zap channels into trunks? |
| 1:05PM |
2 |
FastAGI Connection Failure and Hangup |
| 12:41PM |
2 |
Help on chan_misdn and MSN's |
| 12:30PM |
0 |
Re: Asterisk-Users Digest, Vol 21, Issue 132 |
| 11:46AM |
2 |
Sip t38 gateway tests |
| 11:38AM |
2 |
Auto Logout from queue |
| 10:45AM |
1 |
MFCR2 in Brazil, someone? |
| 10:00AM |
1 |
TDM400P: flash on analog phones doesn't work |
| 9:07AM |
1 |
Updated: No audio when dialing in via PRI withQ.SIG |
| 8:49AM |
0 |
Question on connecting to another system |
| 8:40AM |
0 |
Updated: No audio when dialing in via PRI with Q.SIG |
| 8:02AM |
4 |
About Softphone IAX free for Pocket PC |
| 7:19AM |
3 |
Really Old Rotary Phone |
| 6:34AM |
1 |
Lastest stable build |
| 5:51AM |
0 |
Voicemail being cut-off |
| 5:38AM |
1 |
res_perl voor asterisk 1.2.4 |
| 4:21AM |
0 |
SQL update failing/long fullcontact |
| 4:16AM |
1 |
Festival , Cannot hear the words after "," |
| 4:07AM |
0 |
No sound in one calling direction, men using PRI with E1 and Q.SIG |
| 3:47AM |
1 |
Another undefined pri_restart failure |
| 3:16AM |
3 |
Background asynchronous AGI |
| 3:05AM |
1 |
PRI got event: HDLC Bad FCS (8) on Primary D-channel of span |
| 1:33AM |
3 |
billing realtime |
| 1:02AM |
1 |
CHANUNAVAIL, busy and congestion |
| |
| Monday April 24 2006 |
| Time | Replies | Subject |
|---|
| 11:27PM |
0 |
Development news :: New AEL and configuration system |
| 9:54PM |
0 |
A@H 2.6 : problem connecting call from PSTN |
| 9:23PM |
2 |
Polycom Delay |
| 8:52PM |
0 |
Gsm Gateway , again ! |
| 8:45PM |
1 |
Dialing Ring Groups from the Digital Receptionist- |
| 7:47PM |
3 |
(no subject) |
| 7:31PM |
2 |
Asterisk2Billing |
| 5:20PM |
0 |
codec variable for incoming calls |
| 4:32PM |
1 |
[Issue] Does the *-pbx cmd page honour the absolute timeout value? |
| 4:18PM |
1 |
Change name User-Agent |
| 3:02PM |
0 |
chan_gsm_bt Impression |
| 2:58PM |
1 |
Pinouts for T1/E1 crossover cable WAS "RE: what cable to connect a legacy PBX to a TE410P ?" |
| 2:57PM |
6 |
Two asterisk process in one hardware. |
| 2:30PM |
1 |
Re: Shielding of T1/E1 cables WAS RE: PinoutsforT1/E1 crossover |
| 2:20PM |
2 |
Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails) |
| 2:15PM |
0 |
this is just a post test |
| 2:04PM |
1 |
E1 testing |
| 1:05PM |
0 |
SUSE 9.3, modprobe, and zaptel |
| 1:01PM |
3 |
Channel Restart and Dropped calls |
| 12:56PM |
2 |
Some questions re. T1 cards & QoS |
| 12:23PM |
0 |
HINTS with Polycom stops working after aster isk reload |
| 11:35AM |
2 |
HINTS with Polycom stops working after asterisk reload |
| 11:34AM |
0 |
Time out if channel does not ring |
| 11:17AM |
0 |
eyeBeam 1.5 |
| 10:50AM |
1 |
Zap channels not disconnecting after PSTN line hangs up (getting empty voicemails) |
| 10:50AM |
0 |
Asterisk to Linphone sound playback delay, and then choppy |
| 10:30AM |
1 |
Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover |
| 9:57AM |
2 |
CallerID/variable setting. |
| 9:55AM |
2 |
Question about Asterisk realtime |
| 9:50AM |
1 |
Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover |
| 9:47AM |
0 |
Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?" |
| 9:39AM |
1 |
Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover |
| 9:07AM |
2 |
SMP kernel on Pent 4? |
| 9:05AM |
0 |
getting listed in Directory Assistance, the phone book |
| 8:54AM |
2 |
SIP HEADER FROM: without CALLERID(name) |
| 8:51AM |
0 |
fxotune Problem |
| 8:41AM |
3 |
Faster Sound Files |
| 8:32AM |
0 |
Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossovercable WAS "RE: what cable to connect a legacy PBX to aTE410P ?" |
| 8:23AM |
0 |
fax and URA |
| 7:40AM |
1 |
Help!!!!! DTMF detection is not working on Zap lines |
| 7:29AM |
0 |
Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS "RE: what cable to connect a legacy PBX to a TE410P ?" |
| 7:01AM |
0 |
strange problem with Telasip DID, please help |
| 6:49AM |
1 |
Queue reload |
| 6:47AM |
0 |
Digium cards for sale |
| 6:45AM |
1 |
Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS "RE: what cable to connect a legacy PBX to a TE410P ?" |
| 4:57AM |
2 |
User Defined VoiceMail announcement? |
| 4:41AM |
2 |
Quintum D3000 |
| 4:16AM |
2 |
outbound calls to sip urls |
| 4:15AM |
1 |
1.2.4/7 and chan_modem |
| 4:06AM |
6 |
Hi...Please help me |
| 3:46AM |
1 |
compiling zaptel-1.2.5 |
| 3:38AM |
1 |
Dreadful results from zttest with TE210P and Dell 2850? |
| 2:16AM |
3 |
MeetMe Call Out to invite |
| 1:14AM |
1 |
X100P Polarity Reverse Detection |
| 12:52AM |
1 |
annoying noise on analog phones on tdm400p |
| |
| Sunday April 23 2006 |
| Time | Replies | Subject |
|---|
| 11:54PM |
0 |
sending special infoa fter login |
| 11:41PM |
0 |
1/3 packets are reported dropped by tethereal |
| 7:48PM |
0 |
Clearpath? |
| 6:36PM |
0 |
SPA3000 in Singapore |
| 6:32PM |
0 |
Re: Asterisk-Users Digest, Vol 21, Issue 132 |
| 5:10PM |
1 |
Zap - Cahnnel bank - one way audio |
| 2:56PM |
1 |
Asterisk hangs up on incoming PSTN line to analog extension |
| 12:58PM |
0 |
New backport of T.38 fax passthrough functionality to asterisk-1.2.7.1 |
| 12:01PM |
1 |
call queue problems |
| 10:49AM |
0 |
asterisk at home, broadvoice and iptables |
| 10:40AM |
3 |
E1 connexion |
| 9:20AM |
0 |
RE: Asterisk-Users Digest, Vol 21, Issue 130 |
| 9:12AM |
0 |
RE: Asterisk-Users Digest, Vol 21, Issue 130 |
| 8:38AM |
0 |
Asterisk and SER hangup issue |
| 6:08AM |
0 |
Routing through ENUM |
| 3:58AM |
1 |
Accessing functions from AGI |
| 3:43AM |
1 |
SIPredirect |
| 3:19AM |
1 |
FritzCard, mISDN & "Anlagenanschluss" |
| 3:17AM |
5 |
Codec G729 / x86_64 bits. |
| 2:54AM |
1 |
Setting up a t38 fax gateway |
| |
| Saturday April 22 2006 |
| Time | Replies | Subject |
|---|
| 6:39PM |
6 |
Need help with getting EXTEN from pstn hunt group |
| 3:18PM |
0 |
Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?" |
| 3:14PM |
1 |
Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?" |
| 1:05PM |
0 |
Asterisk on FreeBSD + Passive ISDN BRI |
| 11:21AM |
1 |
Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?" |
| 9:34AM |
0 |
Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?" |
| 9:26AM |
1 |
PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3 |
| 9:13AM |
2 |
PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3 |
| 9:07AM |
0 |
Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?" |
| 8:59AM |
4 |
Pinouts for T1/E1 crossover cable WAS "RE: what cable to connect a legacy PBX to a TE410P ?" |
| 7:31AM |
0 |
What about NCS and Asterisk? |
| 7:03AM |
1 |
anybody get experience with dell powerconnect 3424 and QOS for asterisk traffic? |
| 6:19AM |
3 |
Sipura SP3000 question |
| 3:31AM |
2 |
RE: SPA 3000 - UK Replacement |
| 3:15AM |
2 |
what cable to connect a legacy PBX to a TE410P ? |
| 2:12AM |
5 |
Connecting to a cluster of SIP servers |
| |
| Friday April 21 2006 |
| Time | Replies | Subject |
|---|
| 7:26PM |
1 |
1.2.7.1 on FC5 won't make install |
| 3:35PM |
1 |
Error installing oh323 |
| 2:04PM |
0 |
Very high size-32 usage |
| 2:02PM |
1 |
server choice |
| 1:29PM |
2 |
confused about iax and voip providers termination |
| 1:25PM |
2 |
extension match sip address |
| 11:47AM |
1 |
Grandstream Budge Tone 101 keeps deregistering |
| 11:13AM |
0 |
Easier install of QueueMetrics on Asterisk@Home |
| 10:51AM |
1 |
wellgate FXO unit |
| 10:37AM |
1 |
MWI in multi-PBX setup |
| 9:51AM |
0 |
SIP domain in Asterisk |
| 9:45AM |
1 |
roundrobin strategy in queues not working as described? |
| 9:38AM |
0 |
HANGUPCAUSE on SIP channels |
| 9:36AM |
1 |
Definitive list of sounds |
| 8:53AM |
1 |
Parallel Dial: Busy detection - stop when any is busy? |
| 8:39AM |
0 |
MoH issue |
| 8:07AM |
10 |
Power over Ethernet (PoE) switch recommendations |
| 7:41AM |
5 |
Separating Asterisk SIP extensions from dialing each other. |
| 7:04AM |
1 |
Flash Panel / Queue Slots |
| 5:07AM |
0 |
Airspan / Arelnet GW and Asterisk |
| 4:54AM |
0 |
record_in / record_out configuration parameters |
| 4:08AM |
0 |
How to select Ceptral's Voice in Asterisk's Swiftapplication?? |
| 3:59AM |
1 |
How to select Ceptral's Voice in Asterisk's Swift application?? |
| 3:59AM |
2 |
Asterisk on Red Hat AS 4? |
| 3:28AM |
1 |
AAH or Fedora an Asterisk by sources |
| 3:18AM |
1 |
Unicall MFRC2 Problems with BrT. |
| 2:57AM |
1 |
Real-time Database Front-end |
| 2:10AM |
7 |
some EICON Diva 4BRI questions |
| 1:19AM |
0 |
problem with sphinx2 |
| 12:33AM |
0 |
USB VoIP phone with G729 support |
| 12:04AM |
2 |
Modem connection |
| |
| Thursday April 20 2006 |
| Time | Replies | Subject |
|---|
| 11:10PM |
3 |
Get sysdate + 5 minutes |
| 9:22PM |
0 |
Colour coding the dialplan -- NoOp and ANSI codes? |
| 7:46PM |
1 |
MeetAsterisk in Europe - register today! |
| 3:02PM |
1 |
Background() and Read() |
| 2:38PM |
0 |
Agents and Realtime |
| 2:35PM |
2 |
queues and the '*' key |
| 2:24PM |
1 |
Problem with TE110P configuration |
| 1:43PM |
1 |
channels change names |
| 1:25PM |
2 |
Asterisk FAx-to-Email |
| 1:24PM |
3 |
Asterisk Won't start after SVN Trunk Update |
| 12:11PM |
0 |
Asterisk (RFC 3389) |
| 12:01PM |
0 |
Suggestion Request: Coloc Provider in Chicago, IL area |
| 11:43AM |
3 |
enablling Te110p with PRI |
| 11:39AM |
4 |
Announcement System for a Charity |
| 11:04AM |
6 |
TDM2400P |
| 8:46AM |
1 |
MeetMe: lots of buffer overruns/underruns when connecting over IAX |
| 8:44AM |
1 |
zaptel and zapata configuration |
| 8:13AM |
0 |
does anyone know anything about chan_btp or btpd? |
| 8:09AM |
2 |
Cubix Softphone + Asterisk 1.2.6 |
| 7:24AM |
0 |
Re: Asterisk-Users Digest, Vol 21, Issue 113 |
| 5:03AM |
0 |
Internet connection |
| 4:39AM |
1 |
CDRs and billing |
| 4:32AM |
1 |
Playback(something,noanswer) on Zap? |
| 4:12AM |
0 |
Fwd: why DUNDi ${IPADDR} has been transfered to 127.0.0.1? |
| 4:11AM |
0 |
Best Fax send through Asterisk plan? |
| 3:45AM |
3 |
still some moh troubles |
| 3:36AM |
2 |
asterisk + mobicents |
| 3:13AM |
1 |
Asterisk & MGCP reinvite |
| 3:00AM |
1 |
SPA-3000 Bug? Dropped calls while registering. |
| 2:39AM |
0 |
why DUNDi ${IPADDR} can not transfer to 127.0.0.1? |
| 2:19AM |
2 |
avm b1with chan capi and siemens hipath |
| 1:52AM |
0 |
asterisk and siemens hipath 3500 |
| 12:36AM |
0 |
Happy story |
| 12:25AM |
1 |
How to stop Asterisk picking up my incoming calls? |
| 12:05AM |
0 |
Dial two extensions at the SAME time and connect them when possible |
| |
| Wednesday April 19 2006 |
| Time | Replies | Subject |
|---|
| 11:47PM |
1 |
Jingle support - can we test the feature ? |
| 10:36PM |
1 |
asterisk 1.2.7.1 crashing my newly built system |
| 9:21PM |
0 |
sip.conf codecs: ulaw, alaw and g729 |
| 9:13PM |
2 |
ANNOUNCE: Asterisk Jobs and Consulting Site |
| 8:21PM |
1 |
dundi trouble |
| 7:17PM |
1 |
lost audio after zaptel |
| 6:46PM |
3 |
Upgrade from 1.2.4 to 1.2.7.1 |
| 2:44PM |
1 |
Codec problem from SIP to H323 |
| 2:40PM |
1 |
Error installing asterisk |
| 2:16PM |
1 |
Voice mail issuse when pressing 0 |
| 1:39PM |
1 |
Fwd: sip.conf and jump from register to the extension |
| 12:58PM |
4 |
Ring a grop of extension, then playback a file, then transfer to external number |
| 11:35AM |
2 |
Asterisk 1.2.7.1 DTMF anomaly |
| 11:15AM |
1 |
Asterisk IVR / Scalability |
| 11:12AM |
2 |
Meetme codec translation and callerID library. |
| 10:33AM |
2 |
clearing "stuck" channels without a restart |
| 10:26AM |
1 |
Where to buy Eicon DIVA cards |
| 9:55AM |
0 |
How to route all incoming calls on an analog trunk to a specific ring group |
| 9:23AM |
2 |
What's the best way to combine multiple VOIP lines into a single number? |
| 9:19AM |
0 |
Asterisk 1.2.6 and 9133i |
| 9:02AM |
1 |
Delayed voice for 10 secs |
| 8:37AM |
3 |
SLIN format |
| 8:34AM |
1 |
Callerid matching in extensions.conf |
| 7:58AM |
1 |
Sip channel variables |
| 7:48AM |
0 |
Music on Hold bug? User disconnect Sip user agent |
| 7:08AM |
2 |
Call Center with No TDM components |
| 6:59AM |
5 |
Kernel panic - suggestions? |
| 6:30AM |
0 |
Using Asterisk modules in external application |
| 6:11AM |
0 |
FW: NuFone Update: DIDs (Correction) |
| 6:10AM |
2 |
PRI caller ID |
| 5:42AM |
0 |
oh323: asterisk crashes on a dial |
| 5:40AM |
0 |
Problem with Voice quality, please help |
| 5:25AM |
2 |
Unable to allocate socket: Too may open files |
| 5:02AM |
0 |
Re: new_callback_call and conf disconnect |
| 4:01AM |
2 |
Asterisk 1.2.7.1 and IAX modem / channel |
| 3:57AM |
2 |
Asterisk and 7960s |
| 3:38AM |
1 |
Music on Hold bug? User disconnect Sip user agent and called party stills MOH |
| 3:20AM |
0 |
error when executing sphinx!!! |
| 2:50AM |
0 |
Calls stuck in queue... |
| 2:38AM |
0 |
polycom unable to start recoding |
| 1:28AM |
0 |
LCDC and lcd.conf, p_, c_ |
| 1:08AM |
0 |
Receiving Faxes... |
| 12:53AM |
1 |
Sending SIP NOTIFY / How to get remote SIP port? |
| 12:02AM |
1 |
need stand-alone FXO ports |
| |
| Tuesday April 18 2006 |
| Time | Replies | Subject |
|---|
| 7:03PM |
6 |
Asterisk service crashes |
| 5:06PM |
0 |
Voicemail exits |
| 3:57PM |
5 |
Remember the incoming context? |
| 3:24PM |
6 |
Outgoing voice distortion with Unicall |
| 2:00PM |
3 |
Outbound calls are failing |
| 2:00PM |
0 |
Problem Using Asterisk Call Files with Zap PRI |
| 1:51PM |
0 |
Polycom IP 501 buddy list: Got SIP response 500 "Internal Server Error" |
| 1:49PM |
1 |
Asterisk & GNUDialer issue |
| 1:47PM |
1 |
polycom blind transfer button |
| 1:30PM |
2 |
UK Asterisk sound files |
| 1:03PM |
0 |
re: Sixtel Services |
| 12:57PM |
0 |
RE: Asterisk-Users Digest, Vol 20, Issue 31 |
| 12:48PM |
9 |
FW: NuFone Update: DIDs |
| 11:58AM |
0 |
re: Sixtel Services |
| 11:57AM |
0 |
Realtime goto problem |
| 11:56AM |
0 |
re: Sixtel Services |
| 11:29AM |
4 |
PRI blocking on incoming calls |
| 11:18AM |
0 |
Asterisk Performance 350 Concurrent |
| 11:18AM |
0 |
Aastra 9133i Phones Asterisk 1.2.6 and MWI |
| 9:48AM |
6 |
T1 to cross connect remote PBX and asterisk |
| 9:47AM |
4 |
ISDN in Japan? |
| 9:44AM |
1 |
Double Ring - TelIAX/Cisco 79[46]0 |
| 9:40AM |
0 |
Asterisk crash with Digium |
| 9:35AM |
0 |
Voicemail Issue - Failed to lock path |
| 9:27AM |
0 |
Asterisk Performance 350 Concurrent ChannelsWorking Nicely |
| 9:26AM |
2 |
eyeBeam + ASterisk 1.2.7.1 + Instant Message |
| 9:23AM |
0 |
Gizmo Call In |
| 9:11AM |
0 |
BSR 1000 and Asterisk |
| 8:50AM |
0 |
Help Getting Local Exchange Dialtone on PRI |
| 8:34AM |
0 |
Asterisk Performance 350 Concurrent Channels Working Nicely |
| 8:24AM |
2 |
correct version of asterisk for oh323 |
| 7:54AM |
0 |
IVR and voicemail issues ? |
| 7:50AM |
1 |
bad voice quality |
| 7:28AM |
3 |
IVR: playing multiple streams simultaneously? |
| 6:52AM |
0 |
Re: Cisco 7940/7960 SIP 8.2 Freely |
| 6:49AM |
0 |
Noise on IAX or SIP trunk between 2 Asterisk |
| 6:31AM |
1 |
voicemail kicking in after user has already disconnected |
| 6:04AM |
1 |
Change email/pager VM alerts body text dynamically? |
| 5:19AM |
0 |
Asterisk/FreePBX/Alcatel2400 |
| 5:14AM |
1 |
Granstream GXP2000 Distinctive tones |
| 3:39AM |
2 |
Cisco 7970 SIP - few questions |
| 2:34AM |
3 |
Grandstream Budgetone and Mac mini? |
| 2:09AM |
2 |
HardPhone PlanetVIP-150T - Starts music on Hold and i can't get the call again |
| |
| Monday April 17 2006 |
| Time | Replies | Subject |
|---|
| 3:15PM |
4 |
Looking for a good VoIP Provider in the UK- |
| 3:09PM |
0 |
Asterisk settings for roaming users |
| 1:53PM |
3 |
Asterisk hyperthreading compiling. |
| 12:00PM |
5 |
Orative |
| 11:35AM |
1 |
astcc and inwards billing |
| 11:28AM |
0 |
H.323 question, take so long time to call |
| 10:39AM |
0 |
Sip Notify cisco-check-cfg - Does it still work with 8.2? |
| 10:17AM |
0 |
Asterisk Like Phone Switch ? |
| 10:12AM |
4 |
Billing Server Open Source |
| 9:47AM |
0 |
Setting CDR dnid and Billing |
| 9:39AM |
0 |
IAX phone hardware recommendation |
| 8:44AM |
1 |
voicemail use external smtp server for sendingmail |
| 7:52AM |
7 |
Don't see my post |
| 7:17AM |
2 |
Cannot dial out with Polycom 501 after upgrade |
| 7:00AM |
1 |
Agents, Queues, and Voicemail |
| 2:57AM |
1 |
Probs with asterisk |
| 2:09AM |
4 |
multiple asterisk process ? |
| 12:45AM |
1 |
cdr_pgsql failing to load in head |
| |
| Sunday April 16 2006 |
| Time | Replies | Subject |
|---|
| 11:29PM |
1 |
Snom 190, Asterisk and Intercom |
| 11:25PM |
1 |
Cisco 7940/7960 SIP 8.2 Freely Downloadable |
| 4:18PM |
0 |
Flash Key and R Italian Key |
| 3:35PM |
0 |
External voicemail and MWI on internal phone |
| 9:03AM |
3 |
Problems with several SIP Providers (one way echo) |
| 7:54AM |
1 |
Can Astcc allow dialing phone number more than once |
| 6:46AM |
2 |
How do I limit the lenght of a call |
| 6:16AM |
1 |
[Fwd: Re: voicemail email-from] |
| 6:04AM |
0 |
What is Multi-layered-Access control |
| 1:09AM |
1 |
Faxing and PCI (was Re: Digium cards, sodisappointing !) |
| |
| Saturday April 15 2006 |
| Time | Replies | Subject |
|---|
| 9:24PM |
3 |
voicemail email-from |
| 6:12PM |
6 |
Phones that work well through NAT |
| 12:17PM |
3 |
FreePBX in Production systems? |
| 11:58AM |
1 |
CDR query |
| 11:47AM |
2 |
asterisk voicemail question |
| 1:52AM |
1 |
Cisco 7960 International |
| |
| Friday April 14 2006 |
| Time | Replies | Subject |
|---|
| 3:19PM |
0 |
Ztmonitor shows RX is always on. FIXED. |
| 3:09PM |
0 |
7941/61 IP Phone SIP phone load - for CCM v5.0 |
| 2:14PM |
4 |
My consulting story |
| 1:59PM |
0 |
Bluetooth (chan_btp): dialing external phone number through BTP/Zap when bluetooth device not present? |
| 12:56PM |
1 |
asterisk or ser |
| 11:19AM |
22 |
attended transfer issue |
| 11:14AM |
2 |
How to get 1.2.7 asterisk |
| 10:46AM |
1 |
Re: Asterisk-Users Digest, Vol 21, Issue 81 |
| 9:59AM |
2 |
asterisk 1.2.7.1 and app_rxfax |
| 8:54AM |
2 |
Polycom 501 resource full problems ... |
| 8:23AM |
4 |
Unicall and Fax |
| 7:49AM |
1 |
Packet Testing |
| 7:41AM |
1 |
A weekend of upgrade is coming for me - any hints? |
| 7:30AM |
0 |
Work available - India |
| 6:41AM |
2 |
change/toggle flash operator panel components |
| 6:39AM |
1 |
asterisk with bluetooth headset howto |
| 5:54AM |
1 |
tdm2400p and asterisk 1.2.1 |
| 3:29AM |
2 |
Asterisk hardware for new office suggestion |
| 3:17AM |
0 |
How to cross compile asterisk for Axis ETRAX 100LX foxboard embedded device on Debian |
| 2:43AM |
0 |
Cisco 7970 SIP |
| 12:47AM |
2 |
HELP! Bad sound quality |
| |
| Thursday April 13 2006 |
| Time | Replies | Subject |
|---|
| 9:09PM |
1 |
Asterisk no sound from sound card |
| 7:37PM |
0 |
troubles with a Gateway audiocode (Mp104 fxs) |
| 7:33PM |
0 |
Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. ???? Xlite |
| 5:26PM |
3 |
Will VoIP ITSP's be Next? |
| 5:01PM |
13 |
Digium cards, so disappointing ! |
| 4:07PM |
1 |
What is Multi-layerAccess control |
| 3:36PM |
2 |
Anyone played with app_amd? |
| 2:33PM |
0 |
connecting Digium E1 pri card to panasonic TD-500 |
| 2:21PM |
0 |
SIP/ShoreTel REFER support |
| 1:54PM |
0 |
spa-942 support Page() / Intercom correctly? |
| 1:10PM |
1 |
call center running Asterisk-soundquality-critical! |
| 1:02PM |
0 |
DTMF sensitivity |
| 12:21PM |
0 |
CANADA 911 Update |
| 11:48AM |
1 |
call center running Asterisk -soundquality-critical! |
| 11:29AM |
2 |
Static on ZAP channels |
| 11:16AM |
1 |
Early Media Enable? |
| 11:11AM |
4 |
Asterisk 1.2.7.1 Released |
| 10:58AM |
1 |
Ztmonitor shows RX is always on. |
| 10:42AM |
0 |
Help Cas Circuit |
| 10:33AM |
1 |
Set language in Asterisk auto-dial out |
| 10:07AM |
2 |
Asterisk 1.2.7 Page() |
| 10:06AM |
0 |
SIP register question |
| 10:04AM |
1 |
Sipura 2100 |
| 10:04AM |
1 |
placing call with agi |
| 9:53AM |
1 |
DTMF Not working for only one number |
| 9:33AM |
1 |
Segfault on Inbound call? |
| 9:25AM |
0 |
Display "Confideltial" or "unknown" on calledid display |
| 9:22AM |
1 |
Display "Confideltial" or "unknown" on called iddisplay |
| 9:01AM |
3 |
Display "Confideltial" or "unknown" on called id display |
| 8:55AM |
2 |
NAT/STUN Server |
| 7:36AM |
0 |
fxotune error |
| 7:31AM |
2 |
app_meetme.so |
| 7:30AM |
0 |
(no subject) |
| 6:59AM |
1 |
sip nat bug |
| 6:36AM |
0 |
Codec GSM Makefile Patch for IA64 |
| 6:28AM |
1 |
Question on parkinglot |
| 6:20AM |
1 |
AgentCalled event |
| 6:02AM |
4 |
OT: MWI on Treo 600/650 |
| 5:10AM |
2 |
How to terminate ringing call before it is answered? |
| 4:54AM |
2 |
IP logging |
| 4:51AM |
1 |
ast_sched_runq ran 281 scheduled tasks all at once |
| 4:47AM |
0 |
Hangupcause to handle Called party disconnect ? PSTN----E1----OldPBX---E1--Asterisk |
| 3:01AM |
0 |
Any way to prevent this from happening |
| 2:28AM |
1 |
voicemail use external smtp server for sending mail |
| 1:48AM |
2 |
Background music in call |
| |
| Wednesday April 12 2006 |
| Time | Replies | Subject |
|---|
| 11:57PM |
0 |
Re: Double sip logins |
| 10:30PM |
1 |
Announcement: New Texas User Group formed |
| 10:09PM |
1 |
Problem with Voice Quality |
| 8:46PM |
2 |
freepbx dialing prefix |
| 8:35PM |
2 |
How to terminate ringing call before it is answered |
| 6:59PM |
0 |
RE: Asterisk-Users Digest, Vol 21, Issue 70 |
| 6:57PM |
0 |
Multiple phones in same call |
| 6:05PM |
1 |
ASterisk Back2back |
| 4:19PM |
0 |
Asterisk 1.2.7 Released! |
| 12:51PM |
0 |
Playback sound file while on-line |
| 12:50PM |
1 |
Company List |
| 12:32PM |
1 |
Call Forward and AGI |
| 12:30PM |
1 |
SIP MWI |
| 11:43AM |
33 |
DUNDi with SIP |
| 11:09AM |
1 |
Cisco 7960 won't dial (sccp) |
| 11:07AM |
1 |
Callback Agents and Dial 'g' option |
| 10:43AM |
0 |
call center running Asterisk-sound quality-critical! |
| 10:33AM |
0 |
Config with TE210P, Asterisk and Legacy PBX and FreePBX? |
| 10:24AM |
1 |
Recording queue transfers |
| 10:14AM |
4 |
call center running Asterisk -sound quality-critical! |
| 10:13AM |
1 |
playback soundfile stored in mysql database |
| 9:56AM |
1 |
Polycom VLANs |
| 9:38AM |
2 |
* 1.2.4 & 1.2.6: "Ringing" anamoly |
| 8:44AM |
3 |
Setting Codecs on the Fly |
| 8:42AM |
3 |
CAPI Installation Eicon Diva Server |
| 7:53AM |
2 |
call center running Asterisk - sound quality-critical! |
| 7:28AM |
1 |
Macro-hangupcall - has a Wait(5) - Ast@Home --- why? |
| 7:23AM |
0 |
Trunking Protocols |
| 6:56AM |
1 |
URL in Queue App / Determining the DID/Queue at Agent's Phone |
| 6:26AM |
1 |
DID'S Romania - Bucharest |
| 5:49AM |
5 |
SIP conections, with RTP not going trough Asterisk |
| 5:10AM |
0 |
Oh323 inband DTMF |
| 4:22AM |
0 |
g.726 codec not working in one direction |
| 4:09AM |
1 |
free video (soft) phone available? |
| 3:48AM |
2 |
help -- voicemail |
| 3:40AM |
1 |
Failed to recieve Fax: Asterisk - IAXModem - Hylafax |
| 3:26AM |
1 |
SIP call hangup from asterisk CLI |
| 3:15AM |
2 |
billing with PostgreSQL |
| 2:16AM |
1 |
iax2 show netstats |
| 1:21AM |
1 |
Where is the difference sip.conf - Real-time ? |
| |
| Tuesday April 11 2006 |
| Time | Replies | Subject |
|---|
| 11:10PM |
0 |
SPA-3000 call pickup behind a PABX |
| 10:55PM |
0 |
Polycom SIP 1.6.5 reloading |
| 9:24PM |
1 |
TE410P upgrade to TE411P => (solution to) no more fax carrier detection ! |
| 8:56PM |
0 |
How to config firewall for RTP/RTCP? |
| 8:33PM |
0 |
AstriCon Update: Europe Early Bird Ends Saturday |
| 7:30PM |
1 |
Performance: Xeon or Opteron? |
| 6:08PM |
2 |
call center running Asterisk - sound quality- critical! |
| 5:41PM |
5 |
Cisco 7960 6.3 unlock/reset? |
| 5:19PM |
1 |
Re: update - 512 Simultaneous Callswith DigitalRecording |
| 4:51PM |
1 |
E1 Disconnection Asterisk behind an old PBX |
| 2:31PM |
2 |
res_config_mysql.so: undefined symbol: __stack_chk_fail |
| 2:22PM |
0 |
chan_btp: dialing external phone number when bluetooth not present? |
| 2:07PM |
0 |
core dump... |
| 1:38PM |
0 |
TNT Max Config |
| 1:29PM |
0 |
RE: Fatpipe Support - Authorization to open Box - fwrps2001101288 |
| 1:14PM |
0 |
HELP NEEDED: "odbc show" crashes Asterisk... and I have no idea of what is going on!!! |
| 1:00PM |
0 |
STUN Server info |
| 12:21PM |
1 |
nic aliases not working |
| 11:51AM |
0 |
Cisco 7970 SIP Config |
| 11:02AM |
0 |
log messages... |
| 10:40AM |
1 |
ExternalIVR |
| 10:17AM |
1 |
Snap for Asterisk |
| 10:12AM |
0 |
XO Callerid NAME |
| 8:28AM |
0 |
PRI outbound call error detection |
| 8:26AM |
1 |
Virtual terminal running CLI |
| 8:15AM |
0 |
Logoff time of an agent. |
| 8:06AM |
1 |
Agent with multiple phones in multiple queues |
| 7:54AM |
1 |
AW: Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use?? |
| 7:33AM |
2 |
Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use?? |
| 7:06AM |
1 |
Night Mode and indicators |
| 7:05AM |
4 |
Why is the internet connection important to LAN and PSTN calls? |
| 6:52AM |
2 |
Re: Received VNAK: resending outstanding frames? |
| 6:14AM |
1 |
Question on clicking |
| 5:15AM |
2 |
Automatic 3 Way Call |
| 4:56AM |
2 |
OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...) |
| 4:42AM |
2 |
G726-40 required - Help! |
| 4:18AM |
2 |
Trial Version of Asterisk Interface Available |
| 4:14AM |
3 |
the best billing tool for Asterisk |
| 2:39AM |
1 |
Major issue: More incompatible frame messages |
| 1:44AM |
0 |
Differences 1.0 vs. 1.2 |
| 1:26AM |
1 |
Native music on hold on 1.0 |
| 12:51AM |
0 |
Echo in some queues but not others |
| 12:35AM |
1 |
Database server |
| |
| Monday April 10 2006 |
| Time | Replies | Subject |
|---|
| 10:48PM |
6 |
Bandwidth Management |
| 10:05PM |
5 |
SPA-941/942 Bulk provisioning |
| 8:49PM |
4 |
asterisk credit card processing |
| 8:33PM |
3 |
Vertical |
| 8:30PM |
7 |
Asterisk BRI in the USA |
| 7:52PM |
1 |
Choppy Sound when using linux router or asterisk |
| 7:46PM |
2 |
One digit too short dialed, stay for ever there in the dialplan! |
| 3:56PM |
0 |
ANI and DNIS Seperation on a PRI(TelephonyNumbering Plan (E.164/E.163) (1)'*4105556654*8005550215*' ]) |
| 3:44PM |
1 |
[ISSUE] Honouring Silent Caller ID Numbers |
| 2:54PM |
7 |
te110p and interrupts |
| 2:12PM |
2 |
HTML / PHP |
| 2:00PM |
0 |
Is this possible? (queue setup) |
| 1:50PM |
0 |
Problem with Asterisk and Grandstream HT286 |
| 1:30PM |
2 |
Wanted any /all used out of service Digium boards Mark |
| 1:04PM |
1 |
Group funcations not functioning |
| 12:32PM |
1 |
Polycom TOS |
| 12:28PM |
2 |
Asterisk and Cisco Callmanager |
| 11:19AM |
2 |
Problem - Voicemail resets phone |
| 11:12AM |
0 |
RE: still no solution for me, if one |
| 10:56AM |
3 |
E1 PRI problem with TE205P |
| 10:51AM |
4 |
Texas User Group |
| 10:03AM |
1 |
ANI and DNIS Seperation on a PRI (TelephonyNumbering Plan (E.164/E.163) (1) '*4105556654*8005550215*' ]) |
| 9:54AM |
1 |
RE: still no solution for me, if one provider |
| 9:40AM |
0 |
Audio problems |
| 9:31AM |
1 |
still no solution for me, if one provider fails. |
| 9:20AM |
1 |
ANI and DNIS Seperation on a PRI (Telephony Numbering Plan (E.164/E.163) (1) '*4105556654*8005550215*' ]) |
| 9:03AM |
5 |
App Page() in 1.2.5 |
| 8:37AM |
2 |
faxing setup |
| 8:29AM |
0 |
Asterisk/InterTel Axxess via MGCP? Anyone? |
| 8:25AM |
1 |
"chan_iax2.c: Ooh, voice format changed to ..." |
| 8:07AM |
1 |
SIP channel unavailable/busy/really not there |
| 8:05AM |
0 |
NORTH CAROLINA: Any interest in starting NC User Group? |
| 7:54AM |
1 |
RTP Timestamp errors |
| 7:27AM |
2 |
Outbound calls through Broadvoice |
| 7:14AM |
1 |
How to set AbsoluteTimeout for DirectoryApp() ? Is this the safest way? |
| 7:04AM |
5 |
call center running Asterisk - sound quality - critical! |
| 6:45AM |
1 |
[asterisk-dev] RTP mixer in Asterisk |
| 6:44AM |
1 |
Directory App() is running for a while, like blocked/freeze? in the same name... |
| 6:29AM |
0 |
RTP mixer in Asterisk |
| 5:56AM |
4 |
callerid name inboune from PRI |
| 5:14AM |
3 |
Asterisk stops responding when internet is down |
| 4:13AM |
2 |
AMP / Maintenance-Button missing |
| 4:12AM |
1 |
Asterisk to CCM4 SIP Trunk one-way audio problem. |
| 3:48AM |
2 |
GXP-2000 phones stop registering |
| 3:43AM |
0 |
WG: G729a error |
| 2:59AM |
1 |
Call me for testing my system |
| 2:10AM |
0 |
Asterisk evaluating CLIP, then getting out of the way |
| 1:50AM |
1 |
Re: update - 512 Simultaneous Calls with DigitalRecording |
| 1:21AM |
1 |
G729a error |
| 12:56AM |
1 |
Asterisk is not reconnecting |
| 12:40AM |
0 |
How to avoid "Avoiding initial deadlock...." |
| |
| Sunday April 9 2006 |
| Time | Replies | Subject |
|---|
| 11:55PM |
2 |
queue_log timestamp? |
| 11:21PM |
0 |
Realtime oracle compiling problem |
| 11:13PM |
0 |
(no subject) |
| 11:10PM |
3 |
Voipstunt, voipbuster, .... not working properly? |
| 9:09PM |
1 |
Asterisk Dial Command Timeout not Accurate (not even close) |
| 8:06PM |
1 |
PRI Group Calling |
| 7:06PM |
3 |
Instant Message? |
| 6:20PM |
0 |
for review |
| 5:26PM |
0 |
Provisioning Server... |
| 12:09PM |
0 |
How to avoid "Avoiding deadlock..." |
| 9:28AM |
0 |
txfax tiff file format |
| 7:21AM |
1 |
GXP-2000 and Voicemail |
| 2:06AM |
2 |
how to communicate two PCs on LAN with Asterisk |
| |
| Saturday April 8 2006 |
| Time | Replies | Subject |
|---|
| 11:52PM |
2 |
oh323.conf problem |
| 10:20PM |
2 |
MACRO_RESULT=ABORT |
| 9:02PM |
6 |
How to set busy |
| 7:24PM |
9 |
Force codec |
| 3:08PM |
1 |
ANI on a PRI |
| 2:45PM |
0 |
Re: [asterisk-dev] bug or bad chan_sip.c |
| 2:38PM |
0 |
Re: [asterisk-dev] bug or bad chan_sip.c |
| 1:51PM |
1 |
unable to enable stutter dialtone |
| 11:18AM |
1 |
quadBRI PCI ISDN on Suse Linux 10 |
| 10:21AM |
0 |
Call parking query |
| 5:01AM |
2 |
question about DISA |
| 4:39AM |
0 |
FW: CallerID |
| 2:57AM |
2 |
HELP !!!!! |
| 1:22AM |
0 |
Quintum ASM400 FXO configuration |
| 1:03AM |
2 |
AAstra 9133i register double account.. ?? |
| |
| Friday April 7 2006 |
| Time | Replies | Subject |
|---|
| 11:13PM |
0 |
May be OT , but comparing |
| 9:46PM |
1 |
Problems with registering iaxy |
| 8:26PM |
2 |
Announcing Astmanproxy 1.20 |
| 8:04PM |
5 |
[OT] Centrex Question |
| 4:53PM |
0 |
Canada Nomadic 911 - From the Yes it will Screw Your Biz Dept |
| 1:07PM |
0 |
simple wav ringtones? |
| 12:57PM |
3 |
can we lend a hand? |
| 12:35PM |
0 |
Call tracking through chan_agent using the Manager API |
| 11:50AM |
0 |
Audiconferencing System fon Asterisk |
| 11:21AM |
2 |
DIALSTATUS for Multiple Dialled Numbers |
| 10:03AM |
1 |
Telephony newbie need advice for integration Nortel MICS 4.1 with Asterisk via T1/E1 interface |
| 9:45AM |
1 |
Bell Canada Requests $987.14 Rate increase 9 11 /VOIP Providers |
| 9:37AM |
2 |
Attended Transfer howto |
| 8:55AM |
1 |
wellgate registration 3802 |
| 8:28AM |
6 |
Beeps and noises during calls |
| 8:17AM |
1 |
Cisco 7912 Phones & XML |
| 8:06AM |
1 |
Fedora 'service asterisk start' problems |
| 7:51AM |
0 |
editing the asterisk -addons makefile |
| 7:44AM |
1 |
transfer call after advise |
| 7:42AM |
2 |
Uplink Skype2Sip |
| 7:33AM |
0 |
Re: IVR: Cant hear my message |
| 7:22AM |
1 |
regexp in gotoif |
| 7:02AM |
1 |
OT: local calling guide |
| 6:54AM |
0 |
Inbound PRI calls drop after 5 seconds using |
| 6:32AM |
1 |
Inbound PRI calls drop after 5 seconds using Sangoma A101 |
| 6:14AM |
1 |
suggestions on an IP T1 to TDM T1 gateway solution |
| 3:25AM |
2 |
407 proxy authentication |
| 2:35AM |
0 |
match callerid against outgoing calls |
| 2:29AM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday 04/08/2006 |
| 1:35AM |
2 |
gotoif |
| 12:27AM |
0 |
Dial Plan Problem with extensions ringing multiple phones connected on different * servers |
| 12:18AM |
0 |
CRTC Stops Bell Canada 911 rate increase for 45 days only |
| 12:17AM |
0 |
Line in use |
| 12:12AM |
0 |
0-ECRS |
| |
| Thursday April 6 2006 |
| Time | Replies | Subject |
|---|
| 11:48PM |
1 |
Bell Canada Requests $987.14 Rate increase 911 / VOIP Providers |
| 11:41PM |
1 |
Look What 911 Will Cost in Canada |
| 8:59PM |
0 |
Pickup Group in VPB |
| 8:42PM |
1 |
Integrics ITSP 1.6 released |
| 7:56PM |
1 |
asterisk box as a voip gateway |
| 7:32PM |
0 |
Problem with playing old gsm files |
| 6:57PM |
2 |
# IP601's with POE per Catalyst 3560G-48PS |
| 6:50PM |
0 |
SIP to another PBX w/ forwarding set |
| 6:50PM |
0 |
Telasip |
| 3:52PM |
0 |
Re: Asterisk-Users Digest, Vol 21, Issue 38 |
| 3:49PM |
1 |
Suggested MeetMe feature: 'i' without review. |
| 3:38PM |
0 |
What Media Gateway (connected via SS7) do you use |
| 3:36PM |
3 |
OT: HOWTO: Create a 90mbit bonded link 600 m etre s away with Cat 3 or telco wire [long] |
| 3:35PM |
0 |
OT: HOWTO: Create a 90mbit bonded link 600 m etres away with Cat 3 or telco wire [long] |
| 3:22PM |
1 |
OT: HOWTO: Create a 90mbit bonded link 600 metres away with Cat 3 or telco wire [long] |
| 3:17PM |
1 |
digium card for xseries 346 |
| 2:57PM |
1 |
queue/agent and macros? |
| 2:47PM |
3 |
Steps to make trunked iax2 |
| 2:44PM |
4 |
OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long] |
| 2:42PM |
1 |
Originate |
| 2:42PM |
0 |
Directory Issue |
| 1:32PM |
1 |
Asterisk in FreeBSD |
| 12:27PM |
0 |
Re: MWAnalyze question |
| 11:59AM |
4 |
Call/Contact Center. |
| 11:39AM |
0 |
ringing indication in handset when 2 extensions answer simultaneously? |
| 10:57AM |
0 |
TDM400P and Junghans quadBRI |
| 8:47AM |
1 |
Networld Interop, Vegas 2006 |
| 8:29AM |
1 |
increasing volume level to console/dsp |
| 8:19AM |
0 |
New user needs a hand starting |
| 8:06AM |
1 |
Planet VIP-320 DECT gateway with Asterisk? |
| 7:59AM |
1 |
Cisco 7960 - hints |
| 7:42AM |
2 |
chan_sccp and hinting |
| 7:39AM |
1 |
pause / unpausequeuemember |
| 7:29AM |
0 |
audiocodes with asterisk:- newbie |
| 7:00AM |
0 |
Open channels |
| 6:40AM |
1 |
Call Parking and multiple contexts |
| 6:35AM |
0 |
(no subject) |
| 6:18AM |
2 |
Using Call Progress |
| 5:36AM |
1 |
Voicemaster |
| 5:32AM |
2 |
TDM2400P problems |
| 5:03AM |
0 |
FXS module failed |
| 4:49AM |
0 |
Asterisk dialing over asterisk to PSTN |
| 4:19AM |
1 |
FXO/FXS and E1 in same system |
| 4:18AM |
1 |
qozap errors on junghanns QuadBRI |
| 3:51AM |
0 |
not get ring tone with chan-capi and avm b1 |
| 3:15AM |
0 |
Call transfer to cell phone [UPDATE] |
| 3:02AM |
0 |
AW: Dial out on Zap |
| 2:49AM |
0 |
Dial out on Zap |
| 2:28AM |
0 |
Call transfer to cell phone |
| 1:52AM |
1 |
Incoming call redirected to mobile |
| 1:51AM |
0 |
Fwd: Hangup Supervision |
| 1:49AM |
1 |
IVR : Can't hear my message |
| 1:34AM |
2 |
Hinting a conference room |
| 12:56AM |
0 |
chan_modem_i4l delay again.. |
| |
| Wednesday April 5 2006 |
| Time | Replies | Subject |
|---|
| 11:57PM |
0 |
Chan-sccp - Asterisk dies |
| 10:59PM |
1 |
Fwd: [dmuars] Eh up - March 144 results altered |
| 8:11PM |
0 |
E-911 Canada Info - Hot Off the Press |
| 6:55PM |
0 |
WOW! Sphinx is awesome... but....(asterisk+sphinx+menus) |
| 6:35PM |
2 |
Setting ptime attribute in SDP invite |
| 4:30PM |
4 |
Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4 |
| 4:25PM |
1 |
WebMeetme Problem Please help!!! |
| 2:52PM |
2 |
What causes deadlock? |
| 2:48PM |
0 |
What does this error mean "app.c: Huh....? no dial for indications?" |
| 2:47PM |
15 |
How to restrict simultaneous phone registrations |
| 1:37PM |
2 |
Sending Access codes to a 5EE switch. |
| 1:20PM |
1 |
IAX2 Origination Problem |
| 12:24PM |
6 |
transforming g729 files to wav files |
| 11:31AM |
0 |
The Asterisk bug tracker :: please think twice before opening a report! |
| 11:21AM |
0 |
Asterisk RealTime queue - periodic-announce |
| 11:16AM |
1 |
zaphfc NT Mode. Extension not recognized... |
| 11:12AM |
2 |
Asterisk on BSD? |
| 10:41AM |
0 |
SIP client looses register and then i need to restart my pc to get registered on Asterisk 1.2.5 |
| 10:40AM |
0 |
one-waysilence during calls |
| 10:26AM |
2 |
legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI) and callerid |
| 10:23AM |
0 |
Asterisk support to Tornado M5 IP Phones |
| 10:14AM |
0 |
Favorite softphone with command line interface |
| 10:07AM |
0 |
Re: Asterisk start/stop |
| 9:46AM |
0 |
SHOWCHANINFO Not Working |
| 9:32AM |
0 |
TE110P errors |
| 9:15AM |
4 |
fax server functionality on Asterisk |
| 9:07AM |
2 |
SIP Asterisk Polycom Reinvite |
| 9:02AM |
2 |
can't start chan_capi with asterisk group |
| 8:56AM |
0 |
Patch 5779 on 1.0.9? |
| 8:27AM |
1 |
long delay between "Ring Begin" and "SIP/XXX is ringing" |
| 8:17AM |
2 |
chan_modem_i4l delay |
| 6:37AM |
3 |
queue issue |
| 5:38AM |
0 |
oh323 - cant load module |
| 5:23AM |
2 |
WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus) |
| 5:16AM |
0 |
Querying number of people in a call queue from dialplan |
| 3:29AM |
1 |
Got SIP response 302 "Moved temporarely" |
| 3:18AM |
5 |
Dial Plan Logic Problem |
| 2:31AM |
0 |
Error Header field Via |
| 2:18AM |
2 |
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 1:56AM |
0 |
Hangup Supervision Issue on Digium TDM11B |
| 12:23AM |
3 |
VPB cannot call out |
| 12:09AM |
0 |
extensions.conf - switch => statement? |
| |
| Tuesday April 4 2006 |
| Time | Replies | Subject |
|---|
| 11:52PM |
2 |
Asterisk svn starting problem |
| 11:45PM |
0 |
Asterisk-addons compiling problem |
| 11:38PM |
0 |
some problems with asterisk and E1 |
| 10:04PM |
2 |
Milliwatt Test Number List |
| 9:54PM |
1 |
asterisk-ooh323, asterisk 1.2.6 and netmeeting |
| 9:54PM |
1 |
VoiceMail realtime not working in asterisk-1.2.6 |
| 7:17PM |
0 |
sip hang channels |
| 6:27PM |
1 |
voipstunt: "Forbidden - wrong password ..." |
| 6:23PM |
2 |
speech rec what works |
| 5:55PM |
1 |
Too many open files |
| 5:44PM |
1 |
Need 25-50 Linksys boxes |
| 5:15PM |
1 |
Set(CDR(anything_but_userfield_or_accountcode)=bla) broken? |
| 4:56PM |
2 |
Distinctive Ring on SPA941 |
| 4:37PM |
1 |
not transmit audio on sipura 941 |
| 3:07PM |
0 |
AST eating CPU 100%->Resource temporarily unavailable |
| 3:00PM |
0 |
Applying patch. |
| 2:58PM |
0 |
Anyone have a definitive list of Managereventsper category? |
| 2:42PM |
0 |
Opensource solutions to SPIT |
| 1:40PM |
2 |
WebMeetme defines.php? |
| 1:14PM |
0 |
Jitter in SIP connection |
| 12:36PM |
2 |
queueue recording and what to do next |
| 12:34PM |
1 |
Anyone have a definitive list of Manager eventsper category? |
| 12:17PM |
1 |
Ideal Setup for T1/PRI and TE110P - second try |
| 11:59AM |
0 |
Anyone have a definitive list of Manager events per category? |
| 11:35AM |
2 |
Can't get Pickup app working |
| 11:12AM |
2 |
voicemail context issue |
| 10:46AM |
1 |
Ideal setup for PRI/T1 and TE110P |
| 9:25AM |
1 |
Can't recieve Fax: No carrier detected - Ast erisk + iaxmodem + Hylafaxv --- sorry.wrong log. |
| 9:18AM |
2 |
Any Aheeva Users? |
| 8:57AM |
0 |
X100P small test gives cracking sound at the voip side |
| 8:19AM |
1 |
Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafaxv --- sorry.wrong log. |
| 8:12AM |
0 |
Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafax |
| 7:44AM |
4 |
Phones are all auto answering |
| 7:39AM |
2 |
Possible PRI fault? |
| 7:21AM |
1 |
Realtime Database Lookup |
| 6:45AM |
0 |
Jitter in SIP calls? |
| 6:39AM |
1 |
IAX connection refused between 2 asterisks 1.2.5 |
| 5:47AM |
5 |
Hangupcause is not enough on PRI |
| 3:39AM |
0 |
New User's Query , which card TE411P or TE410P |
| 2:31AM |
3 |
Auto Attendant Question |
| 2:11AM |
2 |
Fax over 2 bridged TE110P channels |
| 1:38AM |
1 |
E1 te110p problem |
| 1:35AM |
0 |
Dial(L(x...)) distinct to SET TIMEOUT(absolute) |
| 12:52AM |
6 |
Loading module chan_zap.so failed! PLZ help me! |
| 12:32AM |
0 |
QSIG and multi-PBX receptionnist |
| |
| Monday April 3 2006 |
| Time | Replies | Subject |
|---|
| 11:16PM |
0 |
RE: Re: Re: Compatible Asterisk Connectivity Cards :Sangoma |
| 9:10PM |
2 |
MeetMe/Asterisk Timer |
| 7:36PM |
1 |
GoDaddy royally screws over aussievoip.com.au and soft-swtich.org |
| 7:30PM |
3 |
Monitor or mixmonitor |
| 6:52PM |
0 |
random beeps during calls |
| 5:22PM |
3 |
Need More Simultaneous Voice Channel Capacity on Asterisk |
| 5:21PM |
2 |
New Skype<>SIP gateway |
| 5:21PM |
2 |
Blocked channels, according to our telco... leading to CONGESTION status |
| 3:46PM |
3 |
Need to Install Fax to Email feature |
| 2:06PM |
0 |
warnings during parsing of misdn.conf |
| 1:41PM |
1 |
Meetme admin |
| 1:26PM |
1 |
web meetme |
| 1:14PM |
1 |
Asterisk compiling problems... |
| 12:34PM |
0 |
Inter-Asterisk SIP and CalleriID |
| 12:04PM |
0 |
411 Directory: First, Last or Both? |
| 11:50AM |
1 |
Hardware question about Redfone's foneBridge |
| 11:31AM |
0 |
Lockups after Asterisk upgrade |
| 11:21AM |
2 |
Interrupting a call |
| 11:18AM |
0 |
Critical Transaction failed: Client non-INVITE - SJPHONE connected to Asterisk |
| 10:43AM |
5 |
Stupid newbie question |
| 9:58AM |
0 |
Maximum duration of Voicemail messages |
| 9:26AM |
0 |
Is it a stun problem: 63 to 1800 msec |
| 8:41AM |
2 |
Unable to connect to remote asterisk (does / var/run/asterisk.ctl exist?) |
| 8:35AM |
1 |
Anybody success using Asterisk 1.2.6 and Spa nDSP 0.0.3 pre 6? |
| 8:28AM |
0 |
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 8:25AM |
6 |
Pickup() h323 |
| 7:44AM |
0 |
Anybody success using Asterisk 1.2.6 and SpanDSP 0.0.3 pre 6? |
| 7:40AM |
2 |
Beginner: PBX for my house |
| 7:31AM |
2 |
Hinting |
| 7:30AM |
2 |
SIP Responsecodes |
| 7:25AM |
1 |
Random music not so 'random' |
| 6:54AM |
3 |
Coice recognition IVR? |
| 6:43AM |
2 |
Callback auto dialing |
| 5:42AM |
2 |
Frustrated with echo... |
| 5:11AM |
0 |
Concurrent calls to voipstunt and other providers |
| 4:36AM |
0 |
AMILogin and case sensitive |
| 4:04AM |
2 |
call transfer to external phone number |
| 3:50AM |
0 |
Bad Pick up line |
| 3:48AM |
0 |
Annonuce Me Feature |
| 2:38AM |
1 |
update asterisk in a production system |
| 2:35AM |
1 |
bristuff for * 1.2.6/zaptel 1.2.5 |
| 2:10AM |
4 |
R2 protocol error |
| 1:56AM |
1 |
Diva Server BRI echo options |
| 1:42AM |
0 |
No CDR in Macro after Dial |
| 1:13AM |
2 |
update - 512 Simultaneous Calls with Digital Recording |
| 12:02AM |
0 |
Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding. |
| |
| Sunday April 2 2006 |
| Time | Replies | Subject |
|---|
| 11:48PM |
0 |
Comparison of Business Edition VS Open Source |
| 10:31PM |
1 |
ASTCC: How to reset "in-use" flag automatically ? |
| 8:45PM |
1 |
ZapBarge but ability to talk to the agent |
| 8:41PM |
8 |
Compatible Asterisk Connectivity Cards : Sangoma |
| 7:48PM |
1 |
Who is on a call? |
| 5:16PM |
1 |
Information about LOCAL/ Channel |
| 2:32PM |
0 |
Voicemail() - Reading exit or return results |
| 1:19PM |
2 |
Connecting Asterisk to "traditional" phone central |
| 12:54PM |
0 |
Subversion mirrors of Asterisk, Zaptel and libpri rebuilt |
| 12:46PM |
1 |
Asterisk "answering machine" replacement, "WaitForRing()", application return values |
| 12:28PM |
0 |
can automon work with MixMonitor |
| 10:38AM |
0 |
no audio between sip channels * 1.2.6 |
| 9:30AM |
5 |
Asterisk 2.0 Where to download |
| 8:04AM |
2 |
DID registration status |
| 4:58AM |
1 |
polycom overlap dialing? |
| 2:40AM |
2 |
Cisco 7960 nat problems. |
| 12:05AM |
1 |
morcdr v0.1 released |
| |
| Saturday April 1 2006 |
| Time | Replies | Subject |
|---|
| 9:48PM |
0 |
G729 Passthrough question |
| 6:46PM |
2 |
Problem: ringtones stop unexpectedly |
| 6:09PM |
4 |
H323 on way voice |
| 4:54PM |
1 |
vmail access problem |
| 4:41PM |
1 |
channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries! |
| 4:17PM |
2 |
Install problem with res_snmp.so from current trunk (bug?) |
| 3:32PM |
2 |
chan-capi: Sending digits on a bri (isdn) d-channel |
| 12:09PM |
1 |
Incorrect CDR results |
| 11:54AM |
2 |
TO have ringing tone instead MOH |
| 11:22AM |
1 |
Problem: ringtones stop unexpectedly when multiple channels are dialed |
| 11:12AM |
1 |
AGI hangup problem |
| 10:07AM |
0 |
Free Software/Open Source Telephony-Summit 2006 |
| 5:58AM |
2 |
Asterisk box with unreliable ping/latency |
| 5:08AM |
2 |
Newbie question - sip.conf incoming contexts |
| 4:52AM |
0 |
INX (Internationalnumber.com) |
| 1:40AM |
1 |
How to use Sendtxt? |
| 12:04AM |
1 |
voicemail to email sending problems |
| 12:01AM |
3 |
ooh323 and g729 - any issue? |