| Friday August 31 2007 |
| Time | Replies | Subject |
|---|
| 11:08PM |
1 |
Problems with Polycom 300/500/600 |
| 10:59PM |
1 |
Cisco 7960 sccp |
| 6:51PM |
2 |
Latency, Jitter and Lost packets... |
| 6:46PM |
1 |
Cisco Directory Format |
| 6:01PM |
1 |
AEL missing in recent 1.2 releases? |
| 5:34PM |
0 |
chan_sip.c:5495 sip_reg_timeout: ERROR |
| 4:38PM |
1 |
Strange behaviour on Asterisk 1.4.9 with Queues... |
| 3:55PM |
0 |
Sipp scenario for asterisk sip |
| 2:45PM |
0 |
Question on Asterisk and ISDN |
| 2:37PM |
1 |
VoIP+IM with Asterisk+Jabber |
| 1:18PM |
1 |
sip:EXTEN;phone-context in asterisk dial plan |
| 12:55PM |
4 |
E1 to Ethernet Bridge |
| 12:33PM |
1 |
Cisco 7960 Won' |
| 11:24AM |
0 |
about ChanSpy |
| 11:08AM |
0 |
WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 9:34AM |
2 |
Shortening Context code |
| 7:36AM |
0 |
question about realtime |
| 1:45AM |
2 |
app_conference |
| |
| Thursday August 30 2007 |
| Time | Replies | Subject |
|---|
| 10:38PM |
1 |
Round robin behavior for dialing SIP trunks... |
| 9:10PM |
0 |
Digium Asterisk Appliance reviews? |
| 8:13PM |
1 |
Hierarchical Config file (re)writing (bug 8684) |
| 8:03PM |
3 |
Testing Framework |
| 6:57PM |
1 |
FYI |
| 6:41PM |
0 |
FATAL: Module wcdtm not found |
| 6:19PM |
3 |
Channel banks for E1 |
| 5:45PM |
0 |
Canada PRI order -- anybody willing to help? |
| 4:59PM |
1 |
FATAL: Module wcdtm not found. |
| 3:54PM |
0 |
Opinions on AsteriskNOW |
| 3:23PM |
0 |
WARNING[22292]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x82f2fe0', 9 retries! |
| 2:52PM |
1 |
How long to detect an "h" exten? |
| 2:25PM |
0 |
DTMF Question |
| 1:31PM |
0 |
Cannot create Incoming Outgoing call through for r2mfc protocol |
| 12:44PM |
5 |
Rechazo de llamada en triangulacion de asterisk. |
| 11:34AM |
4 |
How to handle "+" prefix |
| 10:19AM |
1 |
dialed peer number |
| 9:15AM |
2 |
asterisk at 100% CPU, 1000's of log files |
| 12:03AM |
2 |
Unknown connection error: (2006) MySQL server has gone away |
| |
| Wednesday August 29 2007 |
| Time | Replies | Subject |
|---|
| 11:39PM |
0 |
app_conference and asterisk 1.2.24 |
| 11:13PM |
0 |
Hangup detection and trombining |
| 9:02PM |
1 |
Members in 'Unknown' status in output of 'queue show' |
| 8:46PM |
0 |
Cisco 7970G App Development? |
| 7:06PM |
1 |
where is 1.4.12? |
| 6:09PM |
0 |
Trying to use "Set Group" correctly |
| 5:48PM |
2 |
AsteriskNOW and config files |
| 5:25PM |
3 |
Queue Agents on Remote Asterisk server? |
| 4:58PM |
0 |
Asterisk with IM (instant messaging) |
| 4:32PM |
2 |
understanding queues |
| 3:38PM |
0 |
Cisco FXS Issue... |
| 2:46PM |
1 |
Monitor System using AGI Scripts |
| 2:44PM |
0 |
(Asterisk_1.4.0 + rxfax + spandsp_0.0.4) - symbol lookup error |
| 2:18PM |
0 |
can anybody tell me about unicall.conf for incoming and outgoing |
| 1:46PM |
1 |
OT - Callto:// tags options |
| 1:38PM |
4 |
Channel Bank Recommendations |
| 1:31PM |
1 |
Voicemail and fax detect |
| 1:03PM |
5 |
Ringing sound doesn't work |
| 12:12PM |
2 |
Best text-to-speech |
| 11:39AM |
0 |
Email to Voice |
| 11:09AM |
1 |
WARNING[11439] |
| 7:53AM |
0 |
Asterisk + MSN Messenger |
| 7:36AM |
0 |
call pickup problem |
| 2:41AM |
2 |
sip authorization problem |
| |
| Tuesday August 28 2007 |
| Time | Replies | Subject |
|---|
| 10:42PM |
1 |
Zaptel causes kernel crash - zt_init_tone_state |
| 9:11PM |
2 |
Voicemail Password Issue |
| 9:11PM |
2 |
Load testing/burn-in for Sangoma quad PRI card |
| 8:11PM |
9 |
Fax Problems with SpanDSP |
| 7:51PM |
1 |
ATrpms/Fritz FCPCI CAPI/Fedora 7 |
| 3:51PM |
1 |
E911 mf camma Trunks |
| 3:14PM |
1 |
Distributed System |
| 3:00PM |
1 |
Zaptel 1.4.4 compiling problems |
| 2:53PM |
2 |
server recommentation (unique requirements) |
| 2:13PM |
2 |
G729 Confusion |
| 2:04PM |
1 |
HDL F10 brazilian doorbell device + TDM2400 |
| 1:53PM |
1 |
Astricon Meetup |
| 1:24PM |
1 |
calls being forwarded to neighbor?? please help, thx :) |
| 1:02PM |
0 |
Saftware RAID1 or Hardware RAID1 with Asterisk (Andrew Joakimsen) |
| 12:57PM |
0 |
(no subject) |
| 10:37AM |
0 |
Asterisk Manager Interface, response types |
| 10:31AM |
0 |
Asterisk call waiting with SIP |
| 8:17AM |
1 |
deadagi and billsec or answeredtime |
| 8:04AM |
9 |
Dell SC1430 + Digium TE110P = Digital Noise in PRI |
| 5:09AM |
1 |
app-conference |
| |
| Monday August 27 2007 |
| Time | Replies | Subject |
|---|
| 9:05PM |
3 |
OT: DELL Platforms |
| 7:42PM |
1 |
Detecting tones |
| 2:36PM |
1 |
console/dsp 1.4.11 |
| 2:14PM |
1 |
Can't create audio conversation between softphonesthrough Asterisk |
| 1:51PM |
0 |
error in linking libmfcr2 |
| 12:55PM |
3 |
voip provider settings problem, please help |
| 12:18PM |
7 |
Stereo Conferences? |
| 11:09AM |
1 |
AstriCon Tutorials |
| 10:11AM |
4 |
Prepaid Billing: A2Billing, AstBill, ASTCC |
| 9:06AM |
0 |
libmfcr2 is giving compilation errors |
| 7:01AM |
2 |
Is it possible to register without sending the password |
| 6:56AM |
0 |
call forwading problem DTMF |
| 5:59AM |
0 |
Bad hangup event cause |
| 2:42AM |
1 |
No LongDistance for 1 Extension? |
| |
| Sunday August 26 2007 |
| Time | Replies | Subject |
|---|
| 8:43PM |
1 |
Calling Clients or Tele Marketing |
| 4:31PM |
0 |
Nokia cell connectel to asterisk |
| 1:36PM |
2 |
IAXmodem on Fonality? |
| 10:56AM |
2 |
foneBRIDGE2 setup |
| 9:57AM |
1 |
Is it possible to register without sending the password? |
| 9:44AM |
1 |
TDM400 and TDM800 fxo stop answering |
| 9:15AM |
0 |
Test - pls ignore |
| 2:04AM |
0 |
Davide Marcellan is out of the office. |
| |
| Saturday August 25 2007 |
| Time | Replies | Subject |
|---|
| 11:49PM |
1 |
Chan-capi Fedora 7 |
| 11:27PM |
0 |
Asterisk & Speechphone/Mandi |
| 9:32PM |
0 |
SIP endpoint registeration problem |
| 6:49PM |
1 |
Avaya IPOffice and a SIP trunk to Asterisk |
| 3:05PM |
2 |
Asterisk 1.2 + Zaptel 1.4 + HPEC = Crash? |
| 2:41PM |
1 |
asterisk and vad/cng |
| 7:32AM |
1 |
Polycom firmware download |
| 12:55AM |
0 |
Help define the Asterisk regression test suite |
| |
| Friday August 24 2007 |
| Time | Replies | Subject |
|---|
| 11:19PM |
1 |
IAX2 trunking scalability |
| 11:17PM |
2 |
Restart status |
| 10:24PM |
0 |
AST-2007-021: Crash from invalid/corrupted MIME bodies when using voicemail with IMAP storage |
| 10:11PM |
3 |
which OS would be fine for asterisk |
| 10:08PM |
1 |
asterisk stable 1.2.x or 1.4.x |
| 7:35PM |
1 |
Tuning a ZyWALL for Asterisk |
| 3:29PM |
1 |
AsteriskNOW Web GUI |
| 2:41PM |
1 |
Can't create audio conversation between softphones through Asterisk |
| 2:39PM |
3 |
Keeping queue counters after restarting |
| 12:28PM |
1 |
TE120P digium card PRI_CPE error |
| 11:22AM |
0 |
[Fwd: Re: issues with caller ID , remote-party-id |
| 10:32AM |
0 |
MYSQL problem and configuration |
| 9:45AM |
0 |
DTFM not recognise |
| 9:26AM |
1 |
Error in loading libunicall.so module while running asterisk command |
| 8:29AM |
2 |
Problem compiling Zaptel 1.4.5.1 |
| 8:27AM |
2 |
TE210P digim card PRI problem |
| 7:55AM |
1 |
recomend web interface for virual call center |
| 7:49AM |
1 |
Simulating errors (Busy / Out of Order) |
| |
| Thursday August 23 2007 |
| Time | Replies | Subject |
|---|
| 11:44PM |
2 |
xPL and Asterisk? |
| 10:58PM |
1 |
Linksys (PAP2) delay time between hung up and line release |
| 10:55PM |
7 |
asterisk as a softswitch |
| 9:23PM |
3 |
Is it posible for an incoming to ring to Polycom and cell at the same time? |
| 8:33PM |
4 |
Speech Rec on Voicemail |
| 8:19PM |
3 |
Stable-Stable Asterisk |
| 7:01PM |
3 |
Asterisk Prompt |
| 6:57PM |
0 |
What is this? |
| 6:26PM |
1 |
channel not hungup (zombie?) so call limit not reset to zero |
| 4:37PM |
2 |
1.4 Branch -- which revision |
| 4:16PM |
2 |
meetme conference problem |
| 3:14PM |
6 |
Asterisk Message Logs |
| 2:51PM |
3 |
[PHP-AGI] Problem executing script |
| 2:11PM |
1 |
libmfcr2 is giving definition error while compiling |
| 2:10PM |
1 |
unable to load chan_unicall.so |
| 11:38AM |
1 |
[Serusers] why combine ser with asterisk |
| 8:37AM |
0 |
asterisk configurator with E120P E1 card |
| 7:18AM |
0 |
B410P and echo |
| 7:15AM |
1 |
contact header is missing in 200OK for SUBSCRIBE |
| 7:12AM |
0 |
ASTCC and IVR |
| 1:57AM |
0 |
How to get callee extension in applicationmap(features.conf) |
| 1:57AM |
0 |
asterisk-users Digest, Vol 37, Issue 88 |
| |
| Wednesday August 22 2007 |
| Time | Replies | Subject |
|---|
| 11:49PM |
0 |
Users Conference - Friday@12:30 PM EDT: Founders of Voicepulse |
| 8:31PM |
0 |
Queue Agents from Dialplan |
| 6:39PM |
0 |
VoIP encryption with SIP and IAX |
| 6:19PM |
1 |
Zaptel 1.2.20.1 and 1.4.5.1 released |
| 5:05PM |
2 |
Multiple servers using realtime |
| 3:52PM |
1 |
Chan_mobile and Asterisk SVN-branch-1.4-r80183 compile error |
| 3:04PM |
1 |
TDM400P Not hanging up fast enough |
| 2:42PM |
1 |
Agent status on Polycom phone? |
| 2:27PM |
2 |
[OT] IAX2 WiFi phone? |
| 2:17PM |
0 |
Asterisk Home Automation (was: Re: 99 bottles of beer) |
| 12:32PM |
0 |
asterisk with FAX problem |
| 11:26AM |
1 |
Cisco firmwares 3.6.3 vs 3.8.6 |
| 9:40AM |
0 |
Which interface? |
| 9:38AM |
1 |
How do I configure asterisk? |
| 7:22AM |
2 |
How to re-read values from database in Trixbox |
| 6:53AM |
1 |
rfc3680, reginfo+xml |
| 1:51AM |
1 |
DUNDi, So Easy A Caveman Could Do It! |
| 12:51AM |
3 |
Polycom and NAT |
| 12:36AM |
1 |
Polycom behind NAT won't register to * server behind ALG |
| |
| Tuesday August 21 2007 |
| Time | Replies | Subject |
|---|
| 8:40PM |
0 |
Enable Media Atribute on 180 Ringing |
| 8:30PM |
0 |
Call back or some voicemail notifing. |
| 8:24PM |
0 |
AST-2007-020: Resource Exhaustion vulnerability in SIP channel driver |
| 8:15PM |
1 |
Asterisk 1.4.11 released |
| 7:46PM |
0 |
Mitel 5020 IP phones |
| 7:17PM |
1 |
Contact: header and NAT. |
| 7:00PM |
4 |
Dialogic support |
| 6:15PM |
1 |
SET EXTENSION |
| 5:38PM |
4 |
asterisks addon make problem |
| 4:08PM |
1 |
Problems with overlap dial and Xorcom Astribank BRI |
| 2:28PM |
1 |
Call queue problem |
| 1:36PM |
0 |
Asterisk in Soekris 5501: Is Astlinux the only able solution? |
| 12:22PM |
2 |
TC400B and show transcoder |
| 10:34AM |
0 |
Saftware RAID1 or Hardware RAID1 with Asterisk (Vidura Senadeera) |
| 7:11AM |
1 |
Which GUI for ACD edition ? |
| 4:36AM |
2 |
compatibility of PRI Two B channel transfers TBTC/2BTC |
| 2:03AM |
6 |
Saftware RAID1 or Hardware RAID1 with Asterisk |
| 1:25AM |
1 |
Passing Variables to Voicemail's Email Notification |
| 1:23AM |
3 |
TE405/TE410P help updating from 1.0 to 1.4 |
| |
| Monday August 20 2007 |
| Time | Replies | Subject |
|---|
| 11:25PM |
1 |
Zaptel 1.2.20 echo cancelling problem |
| 7:42PM |
4 |
Realtime Queue Members |
| 5:57PM |
2 |
Setting caller ID on outgoing calls. |
| 5:14PM |
0 |
SpanDSP/TxFAX FAX Status |
| 4:45PM |
2 |
Cdr reports |
| 3:17PM |
2 |
Asterisk as ISDN PRI Proxy |
| 3:11PM |
0 |
OT - IMAP voicemail statistics |
| 1:33PM |
0 |
Got SUBSCRIBE for extension...., but there is no hint for that extension. |
| 1:24PM |
1 |
Disabling Asterisk Authentication |
| 1:15PM |
0 |
How to configure and use GCE4019VOIP phone using asterisk |
| 11:18AM |
1 |
1.4.4. caller ID not working ? |
| 9:40AM |
3 |
Queues with Dynanic Users (BUG?) |
| 7:47AM |
2 |
Firefly IAX2 configuration |
| 7:47AM |
3 |
Redundancy / Failover |
| 5:27AM |
0 |
asterisk1.2.24 or asterisk1.4.10.1 |
| 5:03AM |
1 |
Application for Home Delivery Restaurants |
| 1:00AM |
3 |
Quick DUNDi Poll Questions, For All Asterisk, Users, Please Give Feedback |
| |
| Sunday August 19 2007 |
| Time | Replies | Subject |
|---|
| 10:26PM |
1 |
Nokia cell connected to Asterisk |
| 9:29PM |
2 |
How many calls can use the same username |
| 7:49PM |
1 |
Rewriting the From and Subject from voicemail for a MMS Message to a Cell Phone - like visual voicemail |
| 7:39PM |
0 |
Asterisk 2 Speechphone/Mandi |
| 4:46PM |
1 |
Snom 300 Hints and LIne Buttons |
| 3:39PM |
0 |
Increase Volume on AGI |
| 3:36PM |
0 |
flash zap FXO port from SIP device (SPA-2002) using RFC2833 or SIP INFO |
| 3:12PM |
1 |
Someone please explain FXO/FXS module/channel relationships in Zaptel configuration to me |
| 11:23AM |
3 |
Change Packetization Time |
| 9:15AM |
1 |
Asterisk and Client NAT |
| 5:26AM |
0 |
The Future of Telephony, by Leif Madsen, Jared Smith, and Jim Van Meggelen |
| 5:11AM |
1 |
CDR Disposition Value with ODBC |
| 2:37AM |
4 |
GotoIf not working with ${EXTEN} for me in 1.4.8 |
| |
| Saturday August 18 2007 |
| Time | Replies | Subject |
|---|
| 10:46PM |
2 |
2 asterisk servers, how to connect them together? |
| 7:10PM |
1 |
Best way to detect unknown and/or private incoming caller-id? |
| 7:06PM |
3 |
Blacklisting Toll-Free etc. |
| 12:25PM |
1 |
incoming calls in SIP |
| 11:44AM |
2 |
Forwarding calls, passing Caller ID (or not) |
| 1:41AM |
1 |
Asterisk Manager Proxy - Still required? |
| 1:06AM |
1 |
Asterisk Channel as MusicOnHold |
| |
| Friday August 17 2007 |
| Time | Replies | Subject |
|---|
| 11:15PM |
1 |
Zaptel 1.2.20 and 1.4.5 released |
| 9:34PM |
8 |
Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback |
| 7:15PM |
1 |
gsm errors |
| 5:18PM |
2 |
Subscribe/Notify MWI not working for non-numeric accounts w/X-Lite |
| 4:51PM |
2 |
[asterisk-biz] Skype Outage Leaves Millions Speechless |
| 4:16PM |
0 |
MOH being activated in the middle of a call |
| 4:09PM |
0 |
analog lines running agi on hangup question |
| 3:20PM |
0 |
Suggestions on how to debug strange DTMF problems |
| 2:53PM |
0 |
DISA and Ericsson Dialog 3212 |
| 2:40PM |
0 |
Jain-Sip-Applet-Phone with Asterisk |
| 1:37PM |
4 |
Call Limits |
| 1:06PM |
1 |
Connecting a GSM gateway to a FXO port |
| 1:03PM |
1 |
Detecting DTMF Tones from Muted app_meetme Participants |
| 12:28PM |
2 |
No audio on ISDN PRI calls |
| 10:54AM |
3 |
Lock extension from asterisk |
| 8:42AM |
1 |
1.4.10.[0,1] crashes when call parked |
| 7:53AM |
1 |
Problem in installing libmfcr2 for configuring MFC/R2 |
| 6:49AM |
0 |
Hook flash time problem on TDM400/FXS |
| |
| Thursday August 16 2007 |
| Time | Replies | Subject |
|---|
| 6:56PM |
0 |
Friday@12:30 PM EDT: All about DUNDI |
| 6:38PM |
7 |
RAW asterisk! |
| 6:33PM |
1 |
Outbound SIP authentication with dynamic credentials |
| 4:23PM |
3 |
Experimenting- Sip dialing with Zap |
| 4:08PM |
1 |
Asterisk, PAP2T and 2Wire DSL router |
| 4:07PM |
6 |
Heavy duty environment - Is TDM2400P suits? |
| 4:00PM |
0 |
IAX Trunk |
| 3:10PM |
0 |
Asterisk & SNOM Page/Auto Ans - SNOM only beeps intermittently |
| 2:58PM |
1 |
A102 card, BT ISDN30e, silence |
| 2:48PM |
1 |
Set CALLERID(num) to a specific number only if ${CALLERID(num)} is not an NANP number |
| 2:40PM |
2 |
Outbund Route via Extension |
| 2:10PM |
1 |
Error in intalling library for r2mfc support to asterisk |
| 1:50PM |
2 |
Where I will get astersik.spec and zaptel.spec |
| 1:06PM |
0 |
chan-capi in 1.4.10.1 |
| 12:14PM |
1 |
Authenticating SIP user in LDAP database instead of SIP.conf file |
| 11:06AM |
0 |
Call back voicemail. |
| 11:01AM |
0 |
About cards for ISDN-PRI in Ireland |
| 10:42AM |
1 |
Introducing myself |
| 9:40AM |
2 |
Incoming and Outgoing zaptel configuration : ISDN30e |
| 8:35AM |
3 |
99 bottles of beer |
| 7:04AM |
2 |
tone in linksys pap2t |
| 5:33AM |
0 |
Mitel IP 5020 phones |
| 4:36AM |
6 |
asterisk multiport |
| 1:52AM |
2 |
Seeking opinions: Polycom IP330 phones? |
| 1:22AM |
2 |
zaptel update locks up computer from 1.2.9.1 to 1.2.19 |
| |
| Wednesday August 15 2007 |
| Time | Replies | Subject |
|---|
| 10:12PM |
4 |
GUI for Asterisk realtime |
| 8:36PM |
0 |
Asterisk & SNOM Page - SNOM beeps intermittently |
| 8:33PM |
2 |
Load balancing SIP trunks? |
| 8:28PM |
0 |
3-com Model 3102 IP-Phone / Sip firmware download ? |
| 8:25PM |
0 |
Client-negotiated Codec Instead of Transcoding? |
| 7:47PM |
0 |
iaxtel |
| 7:14PM |
8 |
TDM400P FXO click sounds |
| 5:51PM |
3 |
Dialplan / AGI autoanswer question |
| 4:24PM |
3 |
SIP Events |
| 4:08PM |
1 |
Callback DTMF Problem |
| 3:42PM |
1 |
iaxmodem, chan_capi, hylafax problem and faxing in general |
| 3:29PM |
0 |
slightly OT: Polycom SIP phones |
| 2:14PM |
1 |
CallerID Error causes problems for Polycom phones |
| 1:49PM |
1 |
CDR billsec greater than duration |
| 12:11PM |
2 |
Disable MoH for certain phones |
| 12:08PM |
1 |
Sangoma Wanpipe installation problems |
| 11:33AM |
1 |
Dundi x ENUM |
| 10:36AM |
1 |
why is nonce="584760da" used in sip packets? |
| 8:10AM |
0 |
DUNDi limitation? |
| 6:51AM |
2 |
"Remote" extension search? |
| |
| Tuesday August 14 2007 |
| Time | Replies | Subject |
|---|
| 10:17PM |
2 |
Some advice |
| 5:16PM |
4 |
Dial plan suggestions |
| 4:48PM |
1 |
asterisk 1.2.24 installation |
| 4:19PM |
4 |
Recognize 800 number |
| 2:52PM |
1 |
DTMF on Bridged ZAP call |
| 1:26PM |
0 |
Maximum retries for seqno 102 when re-inviting. |
| 12:17PM |
1 |
BLF with Aastra |
| 10:53AM |
1 |
Faulty voicemail |
| 10:17AM |
2 |
IVR and MySQL |
| 2:41AM |
0 |
Alert_info for AudioCodes MP-124 |
| 2:07AM |
2 |
Patent issues, what features we can't use? |
| 1:42AM |
0 |
REALTIME application vs RealTime function |
| |
| Monday August 13 2007 |
| Time | Replies | Subject |
|---|
| 10:33PM |
0 |
Anyone using zaptel under Solaris? |
| 9:17PM |
1 |
FXO Modules and Sip Outbound |
| 8:25PM |
1 |
AGI answering the channel even though I never asked it to |
| 5:59PM |
0 |
Using hints over DUNDi |
| 3:53PM |
1 |
Problem with Page command |
| 3:38PM |
0 |
test list |
| 3:08PM |
0 |
FW: The trixbox Revolution Continues! Sign upforthe Webinar. |
| 1:49PM |
4 |
CDR-CSV Processing |
| 1:24PM |
1 |
FreePBX |
| 1:07PM |
1 |
Asterisk RTP bridging |
| 12:50PM |
1 |
bristuff - qozap dirver bug (and fix?) |
| 11:30AM |
0 |
Originate and tracking |
| 10:51AM |
0 |
about REALTIME application |
| 10:45AM |
2 |
Does Digium TE120P card support MFCR2 |
| 9:55AM |
1 |
Does digium TE120P card support for MFC/R2 protocol |
| 9:40AM |
0 |
how to configure for the R2MF |
| 9:32AM |
0 |
From where to download all libraries required for configuring R2MF |
| 8:52AM |
0 |
Weird noise problem on SIP transfers... |
| 8:21AM |
0 |
Problems on SIP gateway (especially Planet VIP-450) |
| 6:34AM |
0 |
Codec issues (ilbc->g729) |
| 3:53AM |
1 |
Can't HANGUP call or channel on 1.4.9 |
| 12:52AM |
2 |
How strip +1 from caller id on inbound call |
| |
| Sunday August 12 2007 |
| Time | Replies | Subject |
|---|
| 11:01PM |
1 |
Call file & IAX Trunk: Call Failed, Reason 0 |
| 10:20PM |
0 |
TDM 2400 ? |
| 9:25PM |
1 |
Asterisk 1.2 TDM24xx and B410P |
| 8:02PM |
17 |
20min waiting time |
| 7:38PM |
3 |
Converting an audio file to a ".gsm" format |
| 3:58PM |
1 |
Shared Line Appearance - Aastra 55i - Does it work? |
| 12:54PM |
1 |
Playback a video file? |
| 12:25PM |
5 |
New Pico-ITX |
| |
| Saturday August 11 2007 |
| Time | Replies | Subject |
|---|
| 11:07PM |
4 |
asterisk and telewell isdn hfc problem |
| 4:51PM |
1 |
BLF for Queue |
| 4:09PM |
5 |
indications.c: Can't generate that much data! |
| 2:52PM |
1 |
LumenVox Speech Recognition |
| |
| Friday August 10 2007 |
| Time | Replies | Subject |
|---|
| 10:57PM |
1 |
Hardware Platform Recommendations for Digium Card Compatability |
| 10:23PM |
2 |
Faxing through a PAP2 |
| 10:22PM |
0 |
asterisk-users Digest, Vol 37, Issue 46 |
| 10:02PM |
0 |
Mitel SIP phones |
| 9:05PM |
1 |
misdn and incoming fax detection |
| 6:37PM |
2 |
Pickup command |
| 6:26PM |
2 |
Ordering BRI From AT&T |
| 5:48PM |
0 |
Sending live audio in Asterisk |
| 5:40PM |
0 |
analog fax extension dialing out |
| 5:07PM |
2 |
Locating Asterisk documentation after installation |
| 3:49PM |
2 |
Asterisk Manager to Record Greetings |
| 2:42PM |
2 |
Sort of OT: PBX vs CO |
| 12:49PM |
1 |
Polycom question - removing a soft key functionality |
| 11:51AM |
1 |
How to verify IAX trunking |
| 11:12AM |
0 |
Asterisk action when transfer occurs |
| 10:13AM |
3 |
OT Provisioning http-server-enabled devices (Was: Siemens Gigaset DECT base provisioning) |
| 9:56AM |
5 |
A TrixBox 2.0 seems to be asleep... |
| 9:35AM |
2 |
sip ... codec conversion matrix |
| 2:04AM |
0 |
MINNESOTA - TwinCities Asterisk Users Group 8/11/2007 Echo Cancellation Threat or Menace? |
| 1:12AM |
2 |
Dialplan loop |
| 12:37AM |
2 |
FW: Can you reload only one conf file? |
| |
| Thursday August 9 2007 |
| Time | Replies | Subject |
|---|
| 9:48PM |
0 |
VOIP Provider- Callcentric |
| 9:48PM |
3 |
forking from a dial plan? |
| 8:59PM |
1 |
PRI Question |
| 8:35PM |
0 |
False hangups with TDM400P and Kewlstart |
| 8:12PM |
2 |
Forced Ping or re-registration process for SIP devices or accounts/lines |
| 7:25PM |
1 |
Failover Configuration |
| 5:23PM |
8 |
How to use OpenVPN with Asterisk |
| 5:16PM |
0 |
Polycom Phones Call Hold Reminder function problem |
| 5:04PM |
2 |
How to disable DND feature key in Polycom Phone |
| 4:59PM |
2 |
LIBPRI - video calls over ISDN |
| 4:51PM |
3 |
Need Help in changing Voice message |
| 3:39PM |
2 |
Terrible clicking on T1 |
| 3:27PM |
1 |
The quest for making "hint" more flexible continues - using Realtime now |
| 3:10PM |
2 |
Asterisk Help |
| 2:23PM |
0 |
transfer/conference |
| 1:55PM |
0 |
Friday Aug 10 @ 12:30 PM EDT - Asterisk Users Conference |
| 1:31PM |
1 |
Call forward at telco |
| 12:51PM |
0 |
Level3 WIreless |
| 12:38PM |
0 |
Polycom 330 Speakerphone |
| 12:22PM |
1 |
Allison Smith? |
| 12:15PM |
1 |
705 DIDs for Collingwood Ontario? |
| 11:21AM |
1 |
strange warning |
| 11:01AM |
1 |
generating a GUID |
| 10:52AM |
1 |
Overlapping Playback() with Dial()? |
| 10:27AM |
1 |
usage of each field |
| 7:41AM |
1 |
a couple of new tutorials |
| 5:37AM |
1 |
how to push callerid for each user from sip phone on one side through asterisk (Digium) to E1 card running application on other side |
| 3:44AM |
5 |
Major Digium Card Problems |
| |
| Wednesday August 8 2007 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
Sangoma BRI card -- National ISDN/North America support (Having problems with analog disconnect supervision?) |
| 10:24PM |
2 |
Question on the Monitor command on AMI |
| 10:21PM |
1 |
Using CURL |
| 9:32PM |
1 |
MoH mysteriously stopped working |
| 6:33PM |
1 |
les.net losing DID's |
| 6:20PM |
2 |
FW: The trixbox Revolution Continues! Sign up for the Webinar. |
| 5:56PM |
2 |
Paging Application - Polycom 601 |
| 5:45PM |
1 |
RoundRobin Holding Memory? |
| 5:38PM |
1 |
Howto generate a Manager Event from the Dialplan? |
| 4:49PM |
0 |
FW: OT - Callto:// tags inside web pages |
| 4:30PM |
3 |
VoicePulse Connect |
| 3:26PM |
1 |
Order of matching SIP packet to sections in sip.conf |
| 3:24PM |
2 |
How to write a function with a return value in Asterisk |
| 2:29PM |
2 |
PRI Reset |
| 1:35PM |
0 |
Zap Bridge Question |
| 1:23PM |
1 |
Help : problem in SLA (Shared Line Apperence |
| 12:56PM |
1 |
Buddy watch and the hint priority - brain teaser |
| 12:37PM |
1 |
pick sip channel whn two party talking |
| 12:31PM |
0 |
Asterisk AND Cisco Phones in H323 cloud...problems with some models. |
| 12:02PM |
3 |
Siemens Openstage & Asterisk ? |
| 9:59AM |
1 |
asterisk wait for traling digits |
| 9:56AM |
2 |
Monitor doohicky got event Event 160 on channel.. |
| 8:51AM |
1 |
OT - P-asserted-identity and remote id |
| 7:23AM |
1 |
Siemens Gigaset DECT base provisioning |
| 4:37AM |
1 |
Method for scripting options specified in make menuconfig |
| 4:35AM |
1 |
E1 or analog line |
| 2:47AM |
0 |
Looking for unified messaging expert |
| |
| Tuesday August 7 2007 |
| Time | Replies | Subject |
|---|
| 10:24PM |
2 |
turn off music on hold for a single sip user |
| 10:14PM |
0 |
Asterisk 1.2.24 and 1.4.10 released |
| 9:45PM |
0 |
ASA-2007-019: Remote crash vulnerability in Skinny channel driver |
| 8:22PM |
1 |
Switchtype |
| 7:38PM |
2 |
Outbound dialing |
| 6:13PM |
2 |
Macro Overlap |
| 3:18PM |
1 |
Use of context=... in [default] section of sip.conf |
| 2:51PM |
1 |
Intermittent busy tone detection on loopback setup |
| 2:51PM |
1 |
OT - Callto:// tags inside web pages |
| 2:35PM |
0 |
how to specify a channel inside txfax command |
| 2:16PM |
1 |
OT, I'm looking for SIP/register-enabled softphone |
| 2:06PM |
1 |
caller ID strangeness |
| 2:01PM |
2 |
TE207P Question |
| 2:01PM |
6 |
Which spandsp & unicall version to use with 1.2? |
| 1:24PM |
4 |
Prblem with Page Hight While Faxing over uLaw |
| 11:56AM |
1 |
.call file and logging |
| 9:44AM |
3 |
test the email-list |
| 8:10AM |
3 |
ISDN30 card for UK : sanity check |
| 7:17AM |
0 |
users.conf in 1.4 |
| 5:53AM |
2 |
Login info from Active directory |
| 5:28AM |
2 |
How to stack Sangoma Remora cards |
| 2:10AM |
0 |
???????????????????????? |
| |
| Monday August 6 2007 |
| Time | Replies | Subject |
|---|
| 11:08PM |
1 |
sip issue with one way audio |
| 8:04PM |
1 |
CDR/MySQL basic config |
| 6:28PM |
0 |
Friday Aug 10th Asterisk Users Conference at 12:30 PM EDT |
| 6:09PM |
4 |
low-level dump for PRI dchan debugging |
| 4:55PM |
0 |
Digium|Asterisk World |
| 4:09PM |
1 |
TAE to RJ11 connector (hope not OT) |
| 3:34PM |
3 |
Free sitting |
| 3:00PM |
1 |
iax2 registration being rejected |
| 2:58PM |
0 |
SIP RegEvent - RFC3680 |
| 2:42PM |
2 |
ATA phones ring when they register |
| 2:28PM |
0 |
Setting gain levels with mISDN |
| 2:04PM |
1 |
Cant Play gsm file |
| 12:09PM |
0 |
How to debug OH323 Channel (version 0.7.3) |
| 9:05AM |
2 |
Before Bridging Two Calls |
| 8:42AM |
1 |
help: H323 and SIP |
| 8:06AM |
1 |
Re : Connecting two Asterisk servers with a framerelay |
| 8:05AM |
2 |
A102d samgoma's card |
| 7:45AM |
1 |
Telco is not detecting HangUp w/ TDM400P |
| 12:08AM |
2 |
I am looking for VOIP (SIP/IAX) providers that support sending me RDNIS info on forwarded calls. Are there any providers out there that support this? |
| |
| Sunday August 5 2007 |
| Time | Replies | Subject |
|---|
| 11:32PM |
0 |
Linksys 224P switch and Polycom PoE phones |
| 3:46PM |
1 |
How does one use sip_autoreg |
| 3:41PM |
0 |
are there g729 sound files available? |
| 2:01PM |
0 |
chan_alsa - no sound / strange sound - 1.4.9 |
| 10:15AM |
0 |
Connecting two Asterisk servers with a frame relay |
| 4:44AM |
4 |
Sangoma PRI |
| 4:42AM |
0 |
Agents being bounced from queues after a call and sometimes randomly... |
| 1:16AM |
2 |
! Command from -rx? |
| |
| Saturday August 4 2007 |
| Time | Replies | Subject |
|---|
| 11:52PM |
2 |
text2wave Voices Improvements? |
| 10:09PM |
0 |
zttool says tdm800 is OK, but it won't recieve calls. |
| 8:06PM |
0 |
* and SIP ocupped |
| 8:02PM |
1 |
Connecting two Asterisk servers with a framerelay connection |
| 7:07PM |
0 |
Update zaptel on business edition. |
| 7:06PM |
0 |
quintum AFT200 connection to Asterisk |
| 7:03PM |
2 |
Pre-recorded first and last names audio database |
| 5:17PM |
0 |
Outcall 1.40 released |
| 2:11PM |
1 |
Time Limit on Call or Conference Room? "NEW ASTERISK PROVERB" |
| 1:51PM |
0 |
Turn off musiconhold |
| 11:07AM |
1 |
Hardware advice for 100 extensions, routing via ISDN |
| 10:10AM |
3 |
Connecting two Asterisk servers with a frame relay connection |
| 9:41AM |
2 |
IAX2 - DualServer Problem |
| 8:42AM |
2 |
asterisk 1.2.14 with GUI |
| 8:21AM |
1 |
asterisk always rining phone |
| 3:54AM |
0 |
VoiceMail Call Limit Messages |
| 3:31AM |
1 |
IAX bat phone. |
| 3:19AM |
0 |
Handling message for SAPI/TEI=0/0 Repeated Quickly |
| 1:55AM |
0 |
queue beep |
| |
| Friday August 3 2007 |
| Time | Replies | Subject |
|---|
| 7:31PM |
6 |
Measuring Jitter in Asterisk |
| 6:58PM |
2 |
DIALSTATUS not set |
| 6:50PM |
0 |
Several doubts on Asterisk as an UAC |
| 6:48PM |
2 |
Time Limit on Call or Conference Room? |
| 6:12PM |
2 |
Macro and Arguments |
| 6:04PM |
0 |
CONSOLE=Console/dsp |
| 5:38PM |
0 |
"Asterisk can be attacked using buffer overflow." |
| 3:27PM |
0 |
Asterisk, ISDN AVM C4 and Terrible noise |
| 12:45PM |
2 |
partial ChanSpy |
| 12:27PM |
5 |
Difference between WaitExten and TIMEOUT (response) |
| 10:16AM |
1 |
Knowing zap channel status |
| 8:34AM |
0 |
SIP-6.0 software for Siemens |
| 8:33AM |
0 |
B410P echo cancellation |
| 6:55AM |
1 |
Where to find t38modem |
| 4:07AM |
0 |
Fwd: Re: PRI - DS3 Calls Dropped |
| 3:38AM |
4 |
PRI - DS3 Calls Dropped |
| 3:10AM |
0 |
Asterisk configuration directly with Mandi (Speechphone) |
| |
| Thursday August 2 2007 |
| Time | Replies | Subject |
|---|
| 11:17PM |
0 |
Hints and Noop |
| 9:32PM |
0 |
PhonicEQ T100P |
| 9:02PM |
1 |
A simple IVR extension problem |
| 8:31PM |
5 |
Unicall and Private CID |
| 7:59PM |
0 |
callback and bridge problem |
| 7:35PM |
3 |
PRI/T1 data rate... |
| 7:24PM |
1 |
MySQL + Realtime + SIP Registration |
| 6:21PM |
1 |
dtmf get data |
| 4:23PM |
6 |
Teliax Quality of Service |
| 4:08PM |
1 |
AGI SAY TIME |
| 3:29PM |
1 |
asterisk1.2 to 1.4 g711a fax |
| 3:12PM |
1 |
H.323 |
| 3:11PM |
4 |
Receiving SIP calls without registeration and dynamic IP address |
| 2:13PM |
0 |
uptime script? |
| 1:42PM |
1 |
Recording calls after queues? |
| 1:32PM |
2 |
TE220B |
| 1:06PM |
2 |
radius support |
| 12:47PM |
3 |
Blip every 30 seconds? |
| 12:34PM |
1 |
problem with rfc2833 |
| 9:38AM |
0 |
OT - How to switch headphones between softphones on Linux ? |
| 6:40AM |
1 |
Problem in making SIP call after compiling Asterisk server |
| 2:31AM |
0 |
chan_sip.c error |
| |
| Wednesday August 1 2007 |
| Time | Replies | Subject |
|---|
| 9:42PM |
0 |
dtmf issues over sip and pri |
| 9:40PM |
2 |
Polycom 320 - Can it actually be configured? |
| 9:40PM |
0 |
Announcing free (GPL) VXML for Asterisk - Voiceglue |
| 9:31PM |
2 |
Couple installation questions |
| 8:50PM |
1 |
2 Digit Issue |
| 7:42PM |
2 |
Retail DID provider ? |
| 7:11PM |
5 |
Asterisk DTMF Tones |
| 6:36PM |
0 |
perl script to generate new sip.conf users |
| 6:15PM |
3 |
Slightly OT: SNOM & PoE |
| 5:06PM |
5 |
pri "call by call" trunking? |
| 3:47PM |
1 |
Agent Question |
| 3:43PM |
1 |
Problem with the dial command |
| 1:48PM |
5 |
Hardware that can ring my phone? |
| 1:47PM |
0 |
Can you specify a sip UA's codec based on IP? |
| 12:44PM |
0 |
Help on AsteriskNOW |
| 12:15PM |
3 |
How to use stun server? |
| 11:02AM |
2 |
multiple pbxes, multiple domains, same user ids? |
| 4:34AM |
1 |
Asterisk ref book |
| 3:50AM |
2 |
Live Answering Service with Direct SIP Connections and Light Accounts starting at $14.95/mo |
| 2:04AM |
7 |
Problems building zaptel 1.4.4 |
| 1:44AM |
3 |
TE120P in Canada |