Wednesday August 31 2005 |
Time | Replies | Subject |
11:15PM |
0 |
Pleiades p32mxi |
11:05PM |
1 |
Sangoma card problem with EWSD Exchange |
10:48PM |
0 |
first beta of ruby-agi is out ! |
10:30PM |
2 |
Oracle Realtime Driver and CDR Logger |
9:25PM |
1 |
Softphone vmail indicator and TDM400P woes |
8:50PM |
1 |
Outgoing Context being mistake for dialplan? |
8:38PM |
0 |
DIDs MO DE NY + 8xx# |
8:14PM |
2 |
Asterisk Queues and Strategies |
7:39PM |
0 |
SIP Registration resets |
7:36PM |
1 |
Asterisk@Home: How to changed AMP User Login and Password |
6:46PM |
4 |
One way echo canceling? |
6:45PM |
1 |
TDM04b and echo |
5:25PM |
2 |
How to shorten ringing stop detection on X101P clone? |
5:14PM |
1 |
How to speed-up dialnig with X101P clone modem? |
4:12PM |
7 |
VoipBuster with astersisk? |
3:49PM |
1 |
Call Pickup with Dialog on snom display |
2:57PM |
2 |
Open source firmware on an ATA |
2:39PM |
4 |
/etc/init.d/asterisk barfing |
2:36PM |
0 |
Unprovoked hangups |
1:13PM |
1 |
problems with dialing-out with Zap |
1:03PM |
5 |
Asterisk for Voicemail Server |
12:28PM |
1 |
RE: Is the 2.6 Linux kernel ready for produc tion * environment |
12:23PM |
0 |
Asterisk -> Sipura SPA3000 peer behind NAT |
12:00PM |
1 |
Need Local HELP!!! |
11:59AM |
0 |
Unicall X reload |
11:06AM |
0 |
1.2beta and PRI and CDR Corruption |
11:03AM |
0 |
webcast |
10:42AM |
4 |
RE: Is the 2.6 Linux kernel ready for production * environment |
10:35AM |
1 |
Asterisk and eicon diva server 2M as FXO |
10:28AM |
3 |
odbc realtime update problem |
9:05AM |
2 |
PRI Identity Crisis |
8:54AM |
1 |
CallerID Num and Name setting to Asterisk.. Problem |
8:44AM |
2 |
Howto disable adsi in app_voicemail.c so I can noload *adsi*.so |
8:29AM |
0 |
Uniden UIP200 and Call Queue |
7:31AM |
1 |
astcc number not answering |
7:22AM |
2 |
detecting extensions in use |
7:18AM |
0 |
telextreme and * |
7:00AM |
2 |
SpanDSP rxfax TSID variable name? |
6:27AM |
0 |
Simpletelecom.com |
6:26AM |
0 |
(no subject) |
6:23AM |
1 |
Cisco 7920 and Asterisk - How well do they play together? |
6:17AM |
15 |
VoIP service recommendation |
6:05AM |
2 |
Why it says "all circuits are busy now" |
6:04AM |
4 |
why won;t my voice files play? |
6:03AM |
0 |
SIP phone status |
5:00AM |
0 |
RE: Noise on ZAP channel |
2:32AM |
0 |
voipreach.net - are they functioning |
2:19AM |
0 |
canreinvite=no being ignored? |
1:51AM |
1 |
Sipura SPA-3000 strange behaviour |
|
Tuesday August 30 2005 |
Time | Replies | Subject |
11:48PM |
2 |
Manipulate CALLERIDNUM |
11:30PM |
0 |
canreinvite = yes with PAP2 |
10:57PM |
0 |
ANNOUNCEMENT: Asterisk-Java 0.2-rc1 released |
6:25PM |
1 |
call attend to spanish |
5:48PM |
1 |
Registrar only setup |
5:10PM |
3 |
aastra 9133i DTMF tones |
4:54PM |
2 |
free open source softphone for windows |
4:18PM |
1 |
nested dial, or jump to another line to continue dialing. |
3:43PM |
0 |
OT: Monitoring Tools |
1:59PM |
3 |
Graphical Management Interface - Comments requested |
1:41PM |
1 |
Queues.conf OPTIONALURL within the Queues cmd |
12:19PM |
0 |
astcc hangup problem |
12:16PM |
1 |
ICD Features |
10:48AM |
3 |
(no subject) |
9:56AM |
0 |
How to mute DTMF in meetme? |
9:30AM |
0 |
RE: [Asterisk-Dev] voicemessages table |
8:57AM |
0 |
RE: [Asterisk-Dev] voicemessages table |
8:19AM |
0 |
mrth+manager.conf |
7:47AM |
2 |
unresolved symbol when loading ztdummy |
6:19AM |
2 |
How to use * and # as part of numberindialcommand |
6:07AM |
2 |
FAX and AGI |
5:31AM |
2 |
Wierd Problem |
5:30AM |
1 |
Realtime Queues and Agents |
5:26AM |
1 |
TE110p and E1 |
5:07AM |
1 |
RE: Noise on ZAP channel |
5:01AM |
1 |
Extensions started with # |
4:22AM |
2 |
queue - ringing members in order |
4:12AM |
0 |
re: how to set the voice message as |
3:45AM |
0 |
Re: [Asterisk-Dev] voicemessages table |
3:38AM |
0 |
Re: [Asterisk-Dev] voicemessages table |
2:46AM |
1 |
Asterisk won't listen on different port |
2:22AM |
0 |
sending dtmf tones to the caller (not the called) |
1:33AM |
1 |
X100P and UK CallerID |
12:13AM |
0 |
fedora core 3 kernel source - couldsomeonethrowthe dog a bone! |
|
Monday August 29 2005 |
Time | Replies | Subject |
8:23PM |
0 |
SS7 to IPDC |
8:19PM |
1 |
Sangoma on Telstra E1 Tx/Rx and Echo Settings |
6:32PM |
1 |
Activate/Deactivate Hardware echo cancellation on TE406/TE411 when briging |
4:14PM |
1 |
Call waiting setup/Confenencing problems in AAH |
3:39PM |
4 |
echo system command and set the results to a new variable |
3:23PM |
0 |
delay before dial on TDM04B - continued - poosibly solved |
3:17PM |
0 |
Is Asterisk integrated to NEC PBX? |
1:53PM |
1 |
TXFAX() status |
1:28PM |
1 |
Asterisk + Dualtalk |
12:46PM |
0 |
ztdummy and zttest results |
12:42PM |
2 |
delay before dial on TDM04B - continued |
11:50AM |
0 |
Internal Extensions Busy |
11:35AM |
0 |
confifiguration of Asterisk with Cisco hardware? |
11:20AM |
1 |
MSG Waiting Off |
10:59AM |
0 |
New astGUIclient version released 1.1.6 |
10:58AM |
4 |
delay before dial on TDM04B |
10:49AM |
1 |
grandstream handytone 488 fxo |
10:42AM |
1 |
Moving to New Zealand |
10:11AM |
1 |
RE: Noise on ZAP channel |
9:31AM |
2 |
Asterisk Compile error - x86_64 |
9:28AM |
0 |
[Announce] Web-MeetMe v1.3.3 |
9:15AM |
1 |
SER NAT any additional requirement |
8:43AM |
1 |
text till answer |
8:40AM |
2 |
TDM400 and Phone does not 'ring' |
8:01AM |
2 |
Return code of txfax |
7:34AM |
0 |
plainvoip provider problem |
6:17AM |
3 |
How to use * and # as part of number indialcommand |
6:04AM |
2 |
FW: cvs update error? |
6:04AM |
2 |
Compile problem with 1.2 beta 1 |
6:02AM |
0 |
static noise - follow up |
2:36AM |
0 |
Call file always redials (grrrrr) |
2:31AM |
0 |
SV: Using * in number to chose outgoing peer. |
2:20AM |
0 |
Using * in number to chose outgoing peer. |
2:15AM |
4 |
GXP-2000 presence |
2:05AM |
2 |
Register Asterisk with Gatekeeper - oh323 |
1:58AM |
0 |
Asterisk truncate my FAX !!! |
1:10AM |
0 |
Digi QuadMicro ISDN adapter with asterisk? |
1:09AM |
0 |
Conference and HFC card conflict: no solution?? |
|
Sunday August 28 2005 |
Time | Replies | Subject |
10:57PM |
1 |
monitoring with mrtg |
10:49PM |
0 |
Unable to transfer external calls to MeetMeconference (re-post) |
10:21PM |
3 |
Polycom Reboot Script |
8:43PM |
0 |
GXP-2000 registration issues |
7:43PM |
1 |
Japanese ISDN BRI card for asterisk (INS64) where to start? |
7:41PM |
2 |
error messages |
7:20PM |
2 |
Asterisk 1.2.0-beta1 tarball re-released |
5:35PM |
0 |
hfc-pci/zaphfc: Asterisk hangs with signalling bri_net_ptmp but not with bri_net |
4:52PM |
1 |
Good Deal on A Good Asterisk Box? |
3:06PM |
1 |
SER + ASTERISK voicemail |
2:49PM |
1 |
How to use * and # as part of number in dialcommand |
2:39PM |
0 |
All extensions now cannot loggin!!!! |
1:57PM |
0 |
bid on this small project if you are interested. |
1:20PM |
7 |
ztdummy and Linux 2.6.13-rc7 |
12:46PM |
2 |
Multiple IP's (aliases) on asterisk box? |
12:16PM |
1 |
1.2.0 Beta1 |
12:08PM |
2 |
error compiling on solaris 10 |
11:13AM |
0 |
way to prevent voicemail dialout/callback from 'outside' |
10:56AM |
4 |
Mplayer as replacement to mgp123 in MP3Player cmd |
10:46AM |
1 |
Multiple PCI cards |
8:24AM |
1 |
Sip pickup |
6:55AM |
0 |
Re Invite not working |
6:26AM |
3 |
T1 DSU's/Split for voice |
5:25AM |
1 |
How to configure Cisco AS5800 - Asterisk ?? |
5:03AM |
0 |
SER and Asterisk authentication |
4:45AM |
2 |
Need quote for Asterisk and billing remote install |
3:53AM |
2 |
Spped Dial setup from wiki |
3:45AM |
1 |
DIALSTATUS for Originate Command |
3:35AM |
1 |
DIALSTATUS for Originate |
2:17AM |
1 |
(no subject) |
1:39AM |
5 |
Detect Dialtone |
12:29AM |
0 |
Asterisk + AstLinux testing images now available |
|
Saturday August 27 2005 |
Time | Replies | Subject |
9:55PM |
3 |
Low handset microphone volume with Sipura SPA-841 |
9:23PM |
1 |
Asterisk ISDN: Problem Setting CallerID as DID Extension Numbers. |
6:54PM |
0 |
how can I reduce delays in meetme with zap channels |
6:41PM |
2 |
gotoiftime |
4:53PM |
1 |
Nortell Release 11 and Asterisk E1 |
4:15PM |
1 |
Calling PSTN lines from VOIP softphone |
3:53PM |
0 |
Passing variables across an IAX2 call |
3:31PM |
2 |
Variuos hangup codes in Manager API for failover |
3:19PM |
2 |
Problems with registration |
2:41PM |
3 |
How to use * and # as part of number in dial command |
1:17PM |
0 |
ATComm AG-468 or AG-268 |
11:56AM |
0 |
chan_sip.c: stale nonce received |
11:13AM |
3 |
required packages for asterisk on FC3/FC4 |
9:06AM |
0 |
TE410P Questions |
8:56AM |
0 |
Newbie :SIP ETXTN to SIP EXTN calls |
8:31AM |
1 |
SIP Registration failure |
8:29AM |
0 |
web app |
8:29AM |
2 |
bug tracker down? |
7:20AM |
5 |
Asterisk and a Meridian Nortell Release 11 |
7:12AM |
2 |
storing voice messages in DB SQL |
6:41AM |
2 |
Asterisk 1.2.0 fails to hang up using SIP |
3:41AM |
0 |
ast_register_file_version in 1.2.0-beta1 |
3:11AM |
1 |
dtmf not being detected from viatalk |
2:50AM |
1 |
Asterisk conection to Nextone, codec error |
|
Friday August 26 2005 |
Time | Replies | Subject |
11:48PM |
1 |
Is LDAPget module stable enough for enterprise usage? |
9:04PM |
6 |
Satellite Broadband and VOIP |
8:56PM |
1 |
isa2004 |
8:25PM |
3 |
bug tracker bug? |
8:23PM |
1 |
Fw: IAX2 Softphone Quality & Network Cards |
8:16PM |
1 |
[Announce] Pending update to Web-MeetMe |
6:40PM |
2 |
Asterisk 1.2.0-beta1 Released |
6:27PM |
2 |
Help Solving Asterisk Lockups |
4:50PM |
0 |
motorola vt1000 games |
3:29PM |
0 |
Audio Problem when zaptel modules are loaded |
2:37PM |
1 |
Re: [Asterisk-Dev] Patch 3644 - subscription states *** IMPORTANT *** |
2:35PM |
2 |
HDLC on T1 |
2:33PM |
1 |
Realtime and database structure |
2:24PM |
0 |
Timing issue with call to poll() when running asterisk -rx? |
2:09PM |
1 |
AT-320 IAX & MWI? |
2:03PM |
1 |
Dial command nor progressing on Zap channels |
12:57PM |
0 |
Asterisk on VMWare 4.5, Error Ouch ... error while writing audio data |
12:46PM |
0 |
cdr_odbc in CVS-HEAD gives connect error on reload |
12:19PM |
0 |
voice modification |
12:16PM |
0 |
CVS HEAD HDLC Abort on a TE405P PRI |
12:04PM |
0 |
PCI 2.3 |
11:10AM |
3 |
911 Notices |
10:26AM |
4 |
Ztmonitor values when zap channel is onhook |
10:23AM |
3 |
ignorepat not working - what might I have done? |
10:16AM |
5 |
Fedora Core 4 x86_64 |
10:11AM |
1 |
When 486 ATA crashes, asterisk does not disconnect the call |
10:09AM |
1 |
Attached Voicemail does not play mac/linux |
9:49AM |
3 |
Polycom Phone advise |
9:46AM |
0 |
fedora core 3 kernel source - could someonethrowthe dog a bone! |
9:34AM |
0 |
RE: Voicetronix openline4 quality |
9:26AM |
0 |
ChanIsAvail for IAX not working again/still? AKA Redundant IAX connections not working |
9:07AM |
0 |
HooDaHek 0.4 Released |
9:04AM |
0 |
PhoneCALL version 1.0 Administrative Manual - Released |
8:15AM |
1 |
[Asterisk-Dev] SIP Benchmarking / Stress Testing |
7:48AM |
2 |
SIP Benchmarking / Stress Testing |
7:45AM |
1 |
Asterisk wiht LDAP |
7:43AM |
0 |
cvs update error? |
7:25AM |
2 |
WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type |
6:54AM |
5 |
voip-info - is it alive |
6:48AM |
1 |
realtime sip channel configuration -> insecure option |
6:29AM |
4 |
system crash |
6:09AM |
12 |
IAX2 Softphone Quality & Network Cards |
5:21AM |
1 |
bridging sip to capi, no playtones back to caller |
4:56AM |
1 |
Asterisk: Unable to read password. |
3:29AM |
0 |
SV: Maximum retries error. |
1:55AM |
2 |
About asterisk realtime |
1:03AM |
1 |
Maximum retries error. |
12:54AM |
0 |
fax codec problem |
12:18AM |
3 |
Re:TE110P EuroISDN dial out timing out |
|
Thursday August 25 2005 |
Time | Replies | Subject |
11:57PM |
0 |
UK Caller ID with TDM400P |
10:50PM |
1 |
Caller ID ? |
9:37PM |
1 |
Tools for Remote Monitoring and User Management |
7:20PM |
2 |
no sound with red alarm? |
6:39PM |
0 |
fedora core 3 asterisk startup |
6:19PM |
0 |
Need someone to write a console application for us. |
5:15PM |
1 |
VoIP Mythbusters Help!! - NATed phone-phone connection without proxy? Possible? Yes/No |
4:05PM |
0 |
Aastra 480 CTI? |
2:33PM |
1 |
Working NFAS config w 411p anyone? |
2:26PM |
0 |
Strange Echo |
2:18PM |
3 |
Dell 2850 anyone ... |
1:46PM |
1 |
Cisco 3620 NM-HDV-T1 PRI |
1:32PM |
1 |
Optipoint 600 Cant boot - development shell active |
1:04PM |
1 |
Dial DTMF after bridging call |
12:29PM |
1 |
callerid... |
10:42AM |
1 |
where can I get low cost g723.1 liscence |
10:41AM |
0 |
Detect On-Hook on FXO port |
10:39AM |
0 |
CVS-HEAD: KB1 echo canceller -- USE IT |
10:34AM |
0 |
Can't call to cellular phones from extensions |
10:29AM |
0 |
Internal FXS to SIP problem |
9:02AM |
1 |
OT: Are you using a Lucent? |
9:01AM |
0 |
Re: [Asterisk-Dev] Patch 3644 - subscription states *** IMPORTANT *** |
7:57AM |
4 |
Sipura spa-2000 / 3000: surge protection |
7:43AM |
1 |
Loop back cable pinout 15 Pin Serial |
7:24AM |
0 |
Automated AgentCallback logon and logoff is possible |
7:21AM |
0 |
RealWorld Stats; Not achieving expected results |
6:54AM |
2 |
updating display of a hardphone based on agents logging in |
3:36AM |
2 |
Custom Application For Asterisk |
3:04AM |
1 |
TE110P EuroISDN dial out timing out. |
1:39AM |
2 |
Which Card to choose |
12:32AM |
1 |
PRI signaling experts please help |
12:21AM |
0 |
fedora core 3 kernel source - could someone throwthe dog a bone! |
12:16AM |
4 |
VoIP providers -- California, U.S. |
|
Wednesday August 24 2005 |
Time | Replies | Subject |
11:10PM |
0 |
Distorted Sound from E1 |
9:39PM |
0 |
Asterisk hint thing.... what do you do with it? |
8:33PM |
0 |
SIP trunk rollover problem |
8:17PM |
1 |
Will Echo problems EVER be solved, I'm scare d |
7:04PM |
0 |
Digium TDM400 in UK with BT Lines |
6:21PM |
2 |
SIP Registration --Giving up forever after very short network outage. |
5:34PM |
1 |
Busy number signalling |
5:01PM |
0 |
Channel ooh323c and DTMF with Call Manager |
4:39PM |
3 |
Motherboards and IRQs |
3:38PM |
1 |
dingotel - connect Asterisk to 2-way radio? |
3:33PM |
1 |
FW: [Asterisk-Users, Andrew] Will Echo problems EVER be solved, I'm scared |
2:58PM |
0 |
ANI2 AKA Info Digits not supported? |
2:29PM |
0 |
[Asterisk-Dev] Job Opening - Release Engineer |
2:10PM |
6 |
GXP 2000 Firmware 1.0.1.2 |
2:05PM |
0 |
OT - Packet 8 firmware |
2:04PM |
0 |
google talk sniff |
1:49PM |
2 |
ASTCC and cdrs |
1:44PM |
2 |
Error when answering CAPI |
1:43PM |
0 |
Re: Asterisk and MWI |
1:37PM |
11 |
Will Echo problems EVER be solved, I'm scared |
1:37PM |
0 |
AEL Question |
1:24PM |
3 |
SIP Jitter Buffer on Asterisk |
12:57PM |
0 |
Music on hold configuration |
12:32PM |
0 |
ISDN MSN problems |
11:51AM |
0 |
Re: [Serusers] SER IP PBX for multiple clients |
11:50AM |
0 |
zapata.conf for a BT phone line with a TDM422P |
11:47AM |
2 |
RealTime ignoringswitch=>Realtime/context@re altime_ext |
11:26AM |
0 |
saynumber and variables |
11:03AM |
3 |
fedora core 3 kernel source - could someone throw the dog a bone! |
10:54AM |
0 |
Zaptel Not Sending Tones |
10:18AM |
0 |
Replace Aspect by using Asterisk |
10:03AM |
1 |
Yellow Alarm issues with second TE410P installed |
9:44AM |
1 |
Sql Realtime |
9:44AM |
6 |
Cisco 7960 / SIP & tftp configs |
9:32AM |
3 |
Lots of console; attach and grep? |
9:22AM |
2 |
Can exsiting router handle VoIP traffic? |
9:22AM |
2 |
chan_capi on slackware10? cannot compile :-( why? |
9:17AM |
0 |
Problems setting up X100 FXO card |
8:55AM |
7 |
AGI + Ruby |
8:28AM |
0 |
Re: [Serusers] SER IP PBX for multiple clients |
8:10AM |
0 |
AstriCon Update: Early Bird Ends Tomorrow |
7:54AM |
2 |
asterisk in Taiwan |
7:48AM |
0 |
Listening to agent's conversation while waiting in the queue |
7:34AM |
0 |
Asterisk, Maximiliano J. Goldsmid has invited you to try Google Talk. |
6:44AM |
1 |
Polycom Default .cfg |
6:33AM |
1 |
RealTime ignoringswitch=>Realtime/context@realtime_ext |
6:20AM |
0 |
(no subject) |
5:05AM |
0 |
How do I pick up a specific call from a queue? |
5:05AM |
1 |
Exec a Cmd during a dial |
4:43AM |
1 |
installing pystre |
4:01AM |
2 |
Connection TDM400P to UK PSTN |
3:58AM |
0 |
Experienced Sysadmin/Programmer having major troublewith British Telecom Caller ID & Distinctive Ring |
3:45AM |
1 |
Storing a number to Dial |
3:43AM |
1 |
TDM400P : no dial tone... |
3:10AM |
1 |
tdm04b hangup problem |
2:33AM |
1 |
[Asterisk-Dev] Cisco 7970 SCCP Configs. |
1:55AM |
2 |
Snom 360 - Message waiting and conference keys |
1:46AM |
0 |
FOP queue status |
1:44AM |
7 |
NAT and SIP.conf update. |
1:07AM |
0 |
Answer confirmation via IAX? |
12:44AM |
1 |
HOW TO SEND A MESSAGE TO A CHANNEL THAT IS RECEIVING A CALL???? |
12:38AM |
1 |
SV: Fax to email using mime-contruct |
12:31AM |
3 |
Issue in calling mobiles.... |
12:21AM |
1 |
Wifi UT Starcom F1000: Raising Audio volume Level via asterisk? |
|
Tuesday August 23 2005 |
Time | Replies | Subject |
11:42PM |
0 |
Fax to email using mime-contruct |
10:08PM |
0 |
Out of Office AutoReply: yet another Asterisk an d VMware question |
9:41PM |
2 |
RealTime ignoringswitch=>Realtime/context@realti me_ext |
7:55PM |
2 |
Zyxel Prestige 2000W Firmware - EVIL |
5:09PM |
1 |
Wait before dialing ( was Pause during dialing to enter another number) |
4:32PM |
1 |
Voiceblue and slow dialling |
4:27PM |
1 |
call parking timeout |
3:32PM |
3 |
Help...... I need the TE-110P NEW ZEALAND E1 SETTINGS |
3:18PM |
1 |
Cisco 7940 + no audio after MOH |
2:03PM |
3 |
WARRNING REGARDING Support from 2n.cz !!!! |
1:46PM |
2 |
OH323 with Asterisk@home - seems incomplete |
1:11PM |
2 |
SIP powercycle not hanging up |
11:57AM |
1 |
latest CVS on Mandrake 9.2 Mini ITX |
11:37AM |
0 |
Meetme using ztdummy on Linux 2.6 sounds scratchy |
11:21AM |
1 |
Can't get G729 working after buying a license. |
11:03AM |
1 |
Asterisk 1.0.9: TE411P replacement for TE410P 1stgen causes crashes |
11:03AM |
0 |
Retreive and Play Voicemail name |
11:02AM |
0 |
Sip trunk groups, possible? |
10:53AM |
0 |
AreskiCC + Mutliple SIP Gateways for one route |
10:46AM |
0 |
Nokia PoC PTT Asterisk |
10:14AM |
1 |
AGI nor System working after a dial - Should it work? |
9:47AM |
8 |
HDLC/Zaptel/Kernel 2.6.11(.9) |
9:35AM |
0 |
Sip channel remains active indefinitely |
9:29AM |
0 |
asterisk problem with ODBC |
9:18AM |
2 |
app_sms: using * as an smsc |
9:07AM |
2 |
YAACID isn't working |
8:41AM |
2 |
FW: Register Today for Fall 2005 VON: "The Destination for IP Communications" |
8:33AM |
3 |
looking for failover ideas |
8:25AM |
0 |
Toll Call Voicemail Ring Timeout (new module????) |
8:16AM |
0 |
Embedded HW: asterisk with USB ISDN TA on NSLU2/Debian (fwd) |
8:13AM |
0 |
iax and zap interface problem |
7:32AM |
1 |
Asterisk set-up for LCR |
6:58AM |
0 |
[Asterisk-Dev] Severe ISDN signal distortion in CVS-HEAD with octoBRI |
6:56AM |
2 |
compiling CVS-HEAD + Patch from http://bugs.digium.com/view.php?id=3644 |
6:53AM |
0 |
US based CLEC provider request |
6:11AM |
2 |
Echo after running for several days? |
6:11AM |
1 |
follow me configuration web page?? |
6:11AM |
5 |
chan_unical-MFC/R2 CPU usage problem |
5:34AM |
0 |
FW: SIP DEADLOCK |
5:00AM |
0 |
mailing list |
4:29AM |
0 |
header intact |
4:02AM |
0 |
X100P Clone not picking up incoming calls. [POTS] |
2:33AM |
0 |
call number + tariling suffix |
2:23AM |
0 |
Faxing help |
2:18AM |
1 |
Asterisk & Alcatel PBX |
1:50AM |
1 |
asterisk+realtime |
1:44AM |
2 |
[Asterisk-Dev] q931 dial errors |
12:43AM |
1 |
Odd problem with sip.conf register command: |
12:12AM |
3 |
Music On Hold + canreinvite=yes |
12:04AM |
1 |
where is addmailbox now? |
|
Monday August 22 2005 |
Time | Replies | Subject |
9:46PM |
0 |
Telstra/Sangoma bouncing E1 D Channel |
9:27PM |
0 |
MP3PLayer - rewind, forward, pause functions - feature request |
8:04PM |
1 |
Asterisk ISDN CallerID identification failure |
7:10PM |
2 |
SIP message re-writing and routing with Asterisk |
6:50PM |
1 |
SIP Subscriptions with SNOM |
6:14PM |
0 |
BLF for POTS lines via SIP |
6:13PM |
0 |
Dial, RING with a digit interrupt |
5:24PM |
1 |
dtmf tones |
5:10PM |
1 |
IAX2 with g729 ATA Device |
4:39PM |
0 |
TE110p Speech? |
3:17PM |
1 |
Re: MWI problems on 9133i |
3:16PM |
7 |
Small office setup/using analog lines w/ Ast erisk |
3:00PM |
1 |
txtcidname usage |
2:13PM |
0 |
Recorded sound quiet |
2:10PM |
4 |
grandstream bt100 help |
1:26PM |
0 |
HELP PANASONIC TDA200 AND ASTERISK |
1:15PM |
1 |
Hangup Faster |
1:08PM |
0 |
cisco 7960 disconnect problem |
11:30AM |
1 |
asterisk -rx (or remote connections in general) |
11:20AM |
0 |
SPA3000 dial plan? |
11:12AM |
1 |
Public Key |
11:08AM |
2 |
Shared Call and Bridged Line appearances on Polycom IP501 |
11:02AM |
3 |
Make asterisk 1.0.7 fail under FC4 |
10:58AM |
2 |
Pause during dialing to enter another number |
10:39AM |
1 |
Polycom 1.5.2 call waiting focus behaviour change? |
10:32AM |
1 |
Asterisk 1.0.7 won't run after upgrade to FC4 |
10:29AM |
0 |
Stange behavior with g729 and DTMF |
10:24AM |
1 |
Polycom 1.5.2 firmware NTP problems |
9:09AM |
1 |
Problem with Hangups |
8:38AM |
1 |
Question on Zap interfaces |
7:13AM |
0 |
Asterisk-UK Website |
7:03AM |
0 |
new version of asteriskguru queue statistics released |
6:47AM |
1 |
Qualify time +2000ms? |
6:22AM |
2 |
TE110P problem |
6:10AM |
0 |
Does Asterisk support T1 E&M Wink/Wink voicechannels on any Digium/Sangoma hardware? |
5:57AM |
1 |
FW: Nat + Asterisk + Ser (Far end Nat Traversal) |
5:52AM |
1 |
Delete function in realtime voicemail? |
5:43AM |
1 |
Cut leading digit? |
5:35AM |
0 |
Aastra 9133i and MWI |
5:29AM |
0 |
Aastra 9133i Phone and MWI |
4:35AM |
0 |
SDP media attribute |
4:22AM |
0 |
REGEX Function |
1:32AM |
1 |
Help with Weird Setup |
12:03AM |
1 |
How to start ztmonitor in 'quantitative' mode ? |
12:00AM |
0 |
Asterisk as a SIP provider |
|
Sunday August 21 2005 |
Time | Replies | Subject |
11:46PM |
0 |
Using locked PAP2 and PAP2-NA with Asterisk |
9:41PM |
1 |
hybrid clients |
8:46PM |
0 |
Suggestions |
8:27PM |
0 |
Re: call waiting beep on PSTN and TDM400P FXO linehook flash |
7:07PM |
2 |
perl-cpan |
6:17PM |
2 |
Dial Zero to get outside line? |
2:54PM |
1 |
Call duration limits not working |
2:42PM |
0 |
Nortel Meridian-1 Line Side E1 |
1:41PM |
0 |
call waiting beep on PSTN and TDM400P FXO line hook flash |
12:33PM |
0 |
PrivacyManager not working Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n |
11:59AM |
2 |
Broadvoice Issue |
11:51AM |
0 |
On Network Usage from the CDR |
10:49AM |
0 |
Problem with auto-attendant config, I think.. |
8:31AM |
1 |
"Not-Registered" Problem |
6:57AM |
2 |
Asterisk 1.0.9 on SuSE 9.2 with ISDN BRI & zaphfc? |
6:14AM |
1 |
TDM11B modprobe wcfxs fails |
4:17AM |
4 |
Warning Unable to allocate socket |
|
Saturday August 20 2005 |
Time | Replies | Subject |
11:25PM |
2 |
Asterisk(*) on a Cobalt RaQ2? |
11:24PM |
1 |
Re: Asterisk-Users Digest, Vol 13, Issue 131 |
11:18PM |
0 |
Re: Asterisk-Users Digest, Vol 13, Issue 131 |
10:05PM |
8 |
Small office setup/using analog lines w/ Asterisk |
9:23PM |
3 |
[Asterisk-Dev] IM patch |
7:07PM |
0 |
1.0.9 - can't get link up, 1.0.7 works fine. |
7:01PM |
0 |
Asterisk aborts => undefined symbol: pri_channel_bridge |
6:47PM |
0 |
Re: Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1 |
6:34PM |
1 |
Problems with Asterisk(*): Not-Registered |
5:11PM |
0 |
1 server vs. 2 server config |
3:29PM |
0 |
Help needed receiving incoming calls. |
2:33PM |
0 |
Asterisk transcoding /Routing |
12:43PM |
0 |
X100P compatible |
11:21AM |
1 |
Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension |
10:44AM |
2 |
Ring more than two isdn phones simultaneously |
10:08AM |
0 |
ZAP divert problem |
9:11AM |
0 |
ATA186 reguest problem |
8:11AM |
1 |
ISDN BRI voice one way only |
8:04AM |
0 |
OT? ... Trying to get cid_rewrite script to work |
7:33AM |
0 |
need provider with these did's avail: (anybody?) |
5:28AM |
3 |
ViaTalk Down? |
5:15AM |
0 |
static noise with TDM revision G but not with revision F |
4:53AM |
0 |
What is the reason for warning "Unable to allocate socket" |
4:20AM |
2 |
Asterisk Zaptel Leading Zero Problem With TE110P |
4:05AM |
2 |
Realtime sip_buddies "register=>" how? |
3:57AM |
1 |
Call quality problem when using lan |
3:47AM |
0 |
Quality problem on LAN when using the network! |
|
Friday August 19 2005 |
Time | Replies | Subject |
10:28PM |
0 |
meetme mixer configuration |
6:59PM |
1 |
Asterisk and CompactPCI boards?? |
6:45PM |
0 |
Configuring Asterisk as SIP client behind NAT |
5:36PM |
1 |
Where did my DID's go?? |
4:50PM |
2 |
FXO not picking up; baffled |
4:03PM |
1 |
ZT_CHANCONFIG failed on channel 1 - WAS WORKING!! |
3:38PM |
4 |
[OT] Looking for Web based SIP endpoint |
2:44PM |
0 |
Sudenly unable to get incoming from |
2:35PM |
1 |
Installing to a prefix. |
1:51PM |
4 |
Overriding Caller ID |
1:35PM |
1 |
Sound warnings bringing asterisk down. |
1:05PM |
2 |
Ascend Pipeline POTS to TDM400P FXO Question.. |
12:27PM |
2 |
Asterisk and Vonage - Can't call out but can receive calls |
11:49AM |
1 |
Asterisk not conforming to the RFC?/Aastra phone delay issue |
11:08AM |
3 |
Sending digits from SIP to Asterisk's VoiceMailMain |
11:07AM |
2 |
Unexpected hangups when calling Dialogic D/41JTC-LS |
10:50AM |
0 |
OT: autoresponders |
10:07AM |
3 |
Cisco 7960 Line rollover for secretary's phone. |
9:41AM |
0 |
DTMF on Zap / PBX Transfer |
8:38AM |
1 |
Persistent variables disappear when dialingLocalextension |
8:08AM |
0 |
Overlap digits... |
7:42AM |
3 |
Persistent variables disappear when dialingLocal extension |
7:42AM |
2 |
Sudenly unable to get incoming from Broadvoice |
7:20AM |
2 |
CVS-HEAD Compile Problem |
6:45AM |
1 |
Is there a way in dialplan to determine if call is incoming or outgoing if callerid presentation not enabled on line? |
6:30AM |
0 |
tdm400 and hfc card problem after ztcfg |
6:01AM |
4 |
any ISDN/PRI signaling experts out there? |
4:54AM |
1 |
Nat + Asterisk + Ser (Far end Nat Traversal) |
4:26AM |
0 |
dialing by CODEC type |
3:40AM |
1 |
IPManager now supports SIP, IAX and Zap |
2:31AM |
0 |
meetme-icecast2-ice2 |
1:57AM |
0 |
Has Anybody a working Asterisk auto-startup(Init.d) for SUSE 9.2? |
1:14AM |
0 |
Tr: [Asterisk-Dev] Asterisk IM + Presence |
12:04AM |
1 |
sccp help |
|
Thursday August 18 2005 |
Time | Replies | Subject |
11:55PM |
2 |
Monitoring RTP protocol |
11:30PM |
0 |
why asterisk starts listening on all ports |
11:20PM |
0 |
SV: Zaphfc.ko module error |
10:43PM |
1 |
Unable to transfer external calls to MeetMe conference |
8:21PM |
3 |
Vonage locked Motorola VT-1000s |
8:09PM |
1 |
Agi Script - sending a message to called party |
7:34PM |
1 |
Newbie Trying to make 'catch all extension' but is catching voicemail exit! |
7:33PM |
0 |
asterisk command realtime |
6:25PM |
0 |
SER, Asterisk, SIP proxy, routing, redirection - confused |
6:08PM |
3 |
Optimum online-upload throttling confirmed. |
4:01PM |
2 |
asterick and festival...Help! |
3:08PM |
2 |
Re:How many TDM22P Card can be used on thesame PC ? |
2:03PM |
0 |
MP3Player cmd issue |
2:02PM |
1 |
Persistent variables disappear when dialing Local extension |
1:45PM |
0 |
Which AGI Development Software is fastest onAsterisk? |
1:28PM |
2 |
Updated Patch to chan_agent.c for PREACKANNOUNCE |
1:18PM |
1 |
RES: asterisk seems to load but cannot connect using -r? |
1:04PM |
4 |
VoipJet Problems - anyone? |
1:01PM |
4 |
Which AGI Development Software is fastest on Asterisk? |
1:00PM |
0 |
asterisk, Kirk IP600 and Kirk Z-4020 |
12:57PM |
0 |
asterisk seems to load but cannot connect using -r ? |
12:35PM |
3 |
Preventing an extension from dialing certain outbound codes |
12:32PM |
0 |
How to get long distance carrier to provide separate billing for several companies that share a PRI to LEC? |
12:32PM |
0 |
Set voicemail maximum length by context |
12:32PM |
0 |
[Fwd: Re: Set voicemail maximum length by context] |
12:27PM |
0 |
Directed pickup troubles |
12:20PM |
4 |
static noise with this hardware any advice |
12:11PM |
2 |
Hardware echo cancellation |
12:01PM |
0 |
Festival sounds too wired !! |
11:55AM |
2 |
Searching For a Voip Provider |
11:47AM |
7 |
SPA-2100 Analog Telephone Adapter |
11:46AM |
4 |
Polycom SoundPoint 501 power adapter |
11:26AM |
4 |
Craig R. Saxton/PACE/US is out of the office. |
11:14AM |
0 |
re: slightly OT |
11:14AM |
1 |
Zaphfc.ko module error |
10:56AM |
0 |
help with waning on OSS/dsp, condition 16 and 17 |
10:47AM |
0 |
HDLC Bad FCS / HDLC Abort solution |
10:43AM |
0 |
Question about SIP connection and disconnection events on Asterisk |
10:08AM |
0 |
Asterisk -rx causing crashes? |
10:00AM |
1 |
ASTCC UPDATEproblem |
9:58AM |
0 |
Awesome Job for the Right VoIP Engineer |
9:57AM |
1 |
Cisco ATA-186 working peer to peer |
9:28AM |
1 |
libpri mwi functionality? |
9:13AM |
3 |
Disconnect supervision question |
9:07AM |
0 |
Re: Asterisk-Users Digest, Vol 13, Issue 123 |
8:41AM |
4 |
options for mysql query from dialplan |
8:28AM |
4 |
CRM software |
8:17AM |
1 |
Epygi QuadroFXO? |
6:58AM |
0 |
[ACD]AgentCallBackLogin |
6:16AM |
1 |
pins for users |
6:08AM |
1 |
RE: Pannel |
6:02AM |
1 |
do not appear to have the sources for the 2.6.11.4-20a-default kernel installed |
5:26AM |
2 |
Password for Conf Room |
5:05AM |
0 |
Limit fax tx speed of 'dumb' faxes?? |
4:54AM |
8 |
SNMP for Asterisk |
4:46AM |
2 |
codec gsm and cisco |
4:41AM |
0 |
Tr: [Serusers] SER as "Outbound SIP Proxy" |
4:14AM |
1 |
asterisk with odbc |
3:34AM |
0 |
bristuff-0.2.0-rc8f-cvs does not work with TDM400P |
3:26AM |
0 |
asterisk oh323 not detecting dtmf |
3:19AM |
2 |
segfault with chan_capi-cm 0.5.4 |
3:07AM |
1 |
Lock Extension |
2:32AM |
0 |
granstream, vlan, tftp |
2:26AM |
1 |
initiating Monitor during call |
2:06AM |
1 |
Help on AGI running |
1:59AM |
0 |
Asterisk configuration from database |
1:58AM |
2 |
Asterisk (OH323) - gnugk connection |
1:06AM |
0 |
rotary/pulse |
12:44AM |
0 |
Agent Wrap-up status |
12:37AM |
1 |
Which external (remote) gateway I can use with * ?? |
12:09AM |
2 |
Asterisk configuration from database with res_config |
12:04AM |
2 |
V.17 |
|
Wednesday August 17 2005 |
Time | Replies | Subject |
11:28PM |
0 |
Which cards or box for Germany? - Welche Karten / box fuer Deutschland? |
9:46PM |
0 |
version 1.0.9 slow in acknowledging agent channel calls |
9:10PM |
0 |
sip.conf user entry for ViaTalk |
8:51PM |
0 |
Automatic outgoing calls calling twice |
8:21PM |
1 |
trouble with IP500 |
7:30PM |
0 |
TE110P w/ Dell SC1420 ... any problems out there? |
5:02PM |
2 |
Choppy Ringing |
4:36PM |
4 |
How many TDM22P Card can be used on the same PC ? |
3:30PM |
2 |
How "real time" is realtime? |
3:27PM |
4 |
IP Cop as a firewall and QOS |
3:17PM |
0 |
AstriCon Update: Early Bird Ends Soon - Free Asterisk Book |
2:37PM |
3 |
TDM04B, trunk group |
2:34PM |
2 |
Patchy audio to and from VOIPBUSTER |
2:30PM |
1 |
CAPI problem - need help |
2:13PM |
0 |
[Fwd: [SA16438] Grandstream BudgeTone Denial of Service Vulnerability] |
1:45PM |
1 |
Comfort Noise incomplete - No translator path exists for channel type MGCP (native 4) to 256 |
1:18PM |
1 |
Voicemail/Directory, one person, one box, two last names |
12:54PM |
0 |
chan_sip2.c compiling |
12:38PM |
1 |
DUNDi Install |
12:15PM |
1 |
AGI SCRIPTS USING PERL NEED SOME KIND OF COMPILATION TO WORK WITH * |
11:51AM |
0 |
TDM400 on BT network in UK |
11:36AM |
8 |
DECT gateways |
10:53AM |
4 |
XML Revisited |
10:47AM |
0 |
FXS on TDM12B suddenly stopped working Properly |
10:35AM |
0 |
Asterisk and Port |
10:14AM |
3 |
Does intel 865 board works fine with Asterisk |
10:06AM |
0 |
Xten & Digum TDP FXO card: No sound |
10:03AM |
0 |
(no subject) |
9:33AM |
1 |
Iaxy Distinctive Ring |
9:27AM |
0 |
canreinvite in sip.conf |
9:27AM |
0 |
Any success with Polycom DHCP VLAN discovery? |
9:01AM |
0 |
[Asterisk-Dev] New Astmanproxy Mailing List, and New Version 1.11 |
9:00AM |
0 |
Avaya 4602 SIP Internal Dial Plan |
8:38AM |
1 |
DID on TDM400P Question? |
8:23AM |
2 |
Voicemail crashes asterisk |
7:07AM |
1 |
SIP message 183 and in band info |
6:56AM |
1 |
Any one using the new Digium echo cancellation cards |
6:48AM |
1 |
OT: PC network down if plugged in Polycom IP600 |
6:41AM |
1 |
snom hint |
5:36AM |
2 |
XORCOM RAPID Asterisk - Suggestions? |
5:34AM |
1 |
iaxcomm huge latency |
5:20AM |
2 |
X100P dial out problem |
5:17AM |
3 |
Echo cancellation again ... |
3:28AM |
1 |
zaphfc ptp did problems |
3:27AM |
3 |
Automatic start with SuSe linux |
2:55AM |
0 |
Asterisk (multiple) + Ser |
2:23AM |
3 |
FW: Asterisk-panel |
1:34AM |
5 |
1-800 number |
1:29AM |
4 |
Voicemail Retrival |
1:25AM |
0 |
How to change RINGING style for internal calls |
12:29AM |
0 |
Nikotel issues |
|
Tuesday August 16 2005 |
Time | Replies | Subject |
11:53PM |
3 |
Can not dial more then 23 calls |
11:30PM |
0 |
Re: [Asterisk-Dev] X101P register map data please? |
11:27PM |
0 |
IAX compatible phones |
11:22PM |
0 |
Re: [Asterisk-Dev] X101P register map data please? |
9:57PM |
2 |
florz patch for bristuff breaks compile on x86_64? |
9:49PM |
3 |
ASTCC astcc-config.conf card length question |
8:18PM |
3 |
TE410P + SPANDSP fax problem |
8:10PM |
0 |
Solved: Unable to load module for TE406P |
7:15PM |
0 |
3 way calling |
7:00PM |
2 |
All Page ?? |
6:40PM |
0 |
Polycom 501 Firmware |
6:33PM |
1 |
Execute script on Answer |
6:30PM |
1 |
Does Asterisk support T1 E&M Wink/Wink voice channels on any Digium/Sangoma hardware? |
5:59PM |
0 |
Result from TxFax |
4:01PM |
2 |
SIP "agent" phone w/ headset |
2:42PM |
0 |
hint on parkedcalls |
2:10PM |
6 |
realtime caching |
1:44PM |
0 |
RE:Asterisk-Users] PhoneCALL v2.6.1 - Released |
1:25PM |
0 |
What should my next steps in troubleshooting this TDM04B error be? |
12:37PM |
2 |
TxFax -> RxFax on same machine hangs |
12:20PM |
4 |
Called Party Identification on Polycom IP501 |
12:07PM |
1 |
Asterisk and H323 interoperation issue |
12:05PM |
0 |
Re: Asterisk-Users Digest, Vol 13, Issue 109 |
11:55AM |
2 |
Polycom 501 dialing problem |
11:40AM |
2 |
5 way calling? |
11:39AM |
0 |
MFC/R2 DTMF and digits "*" and "#" |
11:36AM |
0 |
[Asterisk-Dev] SIP channels not cleared |
11:00AM |
0 |
X-lite and Dell Optiplex |
10:29AM |
0 |
asterisk supported compact pci boards |
10:20AM |
1 |
how do we block registration based on ip/subnet? |
9:38AM |
1 |
Advice on old polycom ip 500 |
9:37AM |
1 |
calling number type |
9:32AM |
10 |
quad t1 / 1U rack server combos |
9:01AM |
8 |
Asterisk and LCR |
8:56AM |
3 |
TAFM |
8:31AM |
0 |
Tr: RE: Maximum remote directory size in Polycom IP501 |
8:27AM |
1 |
problems with eyebeam - video phone |
8:02AM |
1 |
x100p question for incomming calls |
7:22AM |
1 |
USB ISDN |
7:07AM |
0 |
Echo calibration with ztmonitor and a testlinefrom a telco |
7:04AM |
1 |
adding another fxo card |
6:57AM |
2 |
PhoneCALL v2.6.1 - Released |
6:55AM |
1 |
Asterisk QUEUES ACD Call Back |
6:25AM |
0 |
features.conf and CVS |
6:19AM |
1 |
DISA over Zap (TE110P) issues on * STABLE 1.0.9 |
6:17AM |
0 |
Help Asterisk -> Hipath 1500 V3.0 |
6:12AM |
2 |
Send 12khz or 16khz billing pulse through fxs |
5:54AM |
1 |
DTMF, Asterisk, External PSTN gateway, and PAP2 (was: RE: Issue with DTMF Tones - CodecIssues) |
5:13AM |
1 |
Issue with DTMF Tones - Codec Issues |
5:09AM |
1 |
E1 R2 |
3:48AM |
1 |
intel 875P chipset ok? |
3:22AM |
2 |
Registration with Asterisk server |
12:36AM |
1 |
Astcc Problem |
12:21AM |
4 |
HP Compatability |
|
Monday August 15 2005 |
Time | Replies | Subject |
10:49PM |
1 |
astcc brands table, inc field |
10:31PM |
1 |
Transferring from cell phone |
10:19PM |
3 |
Plantronics USB Headsets Audio 45 |
10:10PM |
0 |
DTMF being cancelled |
8:26PM |
1 |
Maximum remote directory size in Polycom IP501 |
8:02PM |
12 |
Voipbuster blocking Asterisk/IAX connections? |
7:03PM |
0 |
Mitel 5220 Dial Problem |
6:46PM |
0 |
Ast.1.0.9 (only) strange problem with IAX and DDNS |
6:25PM |
0 |
Load-balancing / offload question |
4:58PM |
0 |
Call waiting beeps |
4:46PM |
1 |
Simple Fax question |
4:15PM |
1 |
Configuration to get CallerID working in New Zealand |
4:09PM |
1 |
NAT'd Snom360 problems |
2:54PM |
1 |
dnsmgr |
2:35PM |
1 |
bristuff-0.2.0-RC8n problems and kernel panic |
2:32PM |
7 |
8 FXS in Asterisk Server |
1:31PM |
0 |
Firewall will definatelyincreasejitters inyourvoice conversation |
1:30PM |
2 |
Only single channel recorded with Monitor |
1:23PM |
1 |
How to remove standard ISDN drivers from RedHat |
1:21PM |
0 |
Firewall will definatelyincrease jittersinyourvoice conversation |
12:40PM |
2 |
Fax Issues |
12:08PM |
2 |
problem with sound device |
11:22AM |
3 |
BRI Hunting, using both channels on one msn |
11:16AM |
7 |
Switch between FXS ports |
9:38AM |
2 |
No translator path exists for channel type MGCP & Comfort noise support incomplete |
8:52AM |
0 |
h323 registration problem |
8:37AM |
1 |
Re: [Asterisk-Dev] MS Live Communications Server |
8:25AM |
1 |
User in two queues receive two calls at once |
8:21AM |
1 |
Chan_sccp and dynamic DNS |
7:30AM |
0 |
Conference moderator password |
7:29AM |
0 |
RocketVoip? |
7:19AM |
1 |
(no subject) |
6:29AM |
1 |
codecs order |
5:17AM |
2 |
asterisk + chan_mISDN = undefined symbol: ast_pickup_call |
2:43AM |
0 |
Asterisk Java-Call Problem |
2:22AM |
2 |
Security and SIP |
2:11AM |
1 |
permission denied when monitoring channel OSS/dsp |
1:17AM |
0 |
Unable to load module for TE406P |
1:16AM |
1 |
Connecting 2 * servers |
|
Sunday August 14 2005 |
Time | Replies | Subject |
11:37PM |
0 |
Sirrix bri card:killing the machine |
10:55PM |
1 |
PABX and Asterisk Dial Plan |
10:33PM |
0 |
(no subject) |
6:50PM |
0 |
anyone use TE4xxP work well with huawei C&C08 switch? |
5:15PM |
0 |
h Priority |
2:39PM |
0 |
setting up rate-engine? |
2:37PM |
0 |
OrderlyQ |
1:12PM |
2 |
Problem with FWD connection rejected |
1:12PM |
2 |
Bigger problems than ogg |
11:23AM |
2 |
Cisco and "protocol application invalid" |
10:25AM |
0 |
subscibe FOODFIX digest |
8:55AM |
2 |
TELASIP DOWN? |
8:34AM |
4 |
Multiple Asterisk Installations + SER |
8:16AM |
1 |
ogg causing me heart burn |
7:15AM |
0 |
[OT] SPA-3000 loudness |
6:11AM |
0 |
IPManager now templated based |
6:11AM |
1 |
*confused* - help needed |
5:37AM |
1 |
ParkAndAnnounce - No Disconnect |
4:33AM |
0 |
ParkAndAnnounce - Any way to not disconnect? |
3:37AM |
1 |
Module wcfxs - is it not part of astlinux? |
|
Saturday August 13 2005 |
Time | Replies | Subject |
10:41PM |
0 |
HooDaHek 0.3 Released |
10:41PM |
1 |
Initiating a transfer from an analog handset? |
7:17PM |
0 |
Call Queues and Agent Call Logs/Wrapup logs |
5:27PM |
0 |
Asterisk Flash Transfer (callthrough) |
4:53PM |
2 |
Asterisk forwarding confirmation? |
2:48PM |
0 |
cvs STABLE of 08/10 & gcc4 issue |
1:52PM |
14 |
Why NAT problem |
1:08PM |
2 |
forward incoming analog call to SIP? |
12:20PM |
0 |
Re:(2) Henning G. Schulzrinne quote on IAX2 from von magazine |
11:57AM |
0 |
Re: Henning G. Schulzrinne quote on IAX2 from von magazine |
9:55AM |
1 |
T.38 decoding |
9:20AM |
0 |
Re: Asterisk-Users Digest, Vol 13, Issue 86 |
9:10AM |
2 |
(no subject) |
9:06AM |
0 |
Attended Trasnfer |
8:56AM |
0 |
extensions exchange |
8:41AM |
2 |
premature call release - SIP 480 |
8:36AM |
2 |
Cisco IP Phone- 7905G |
8:07AM |
1 |
New Beta IAX Statistics Program |
7:46AM |
3 |
TDM400P Card (Rev G) with bad FXS module? |
6:30AM |
0 |
[Asterisk-Dev] Re: FXO PCI Master abort |
6:14AM |
1 |
receiving calls from FWD |
5:50AM |
3 |
One more newbie question |
5:26AM |
0 |
Flash over SIP Trunk |
5:07AM |
3 |
Push to talk and asterisk |
4:52AM |
0 |
txfax on strike while rxfax works flawlessly |
2:59AM |
1 |
Identify call flow from manager events |
2:44AM |
2 |
MISDN callerid |
2:05AM |
0 |
Receive fax then send onwards |
1:25AM |
1 |
Disable Call Waiting On SIP User Agents |
12:59AM |
0 |
Incompatible destination (88) Error Message. Please Help !!! |
|
Friday August 12 2005 |
Time | Replies | Subject |
11:29PM |
2 |
Remotely rebooting Sipura SPA-3000 from command line |
8:20PM |
1 |
fc3 build after kernel update? |
7:06PM |
1 |
Suggestions for mainstream hardware compatible with TE411P. |
4:29PM |
1 |
chan_skinny issue |
3:48PM |
4 |
Dell Poweredge 1400 |
3:26PM |
0 |
ubr924 |
3:20PM |
3 |
Announcement to called party |
2:47PM |
4 |
voicemail - 99 message limit |
2:03PM |
0 |
Saved Message playback |
12:42PM |
1 |
Small Form Factory Machine |
12:11PM |
0 |
Asterisk Cell Socket Recommendation |
11:37AM |
1 |
PauseQueueMember and UnpauseQueueMember |
11:31AM |
3 |
7960 TFTP |
11:29AM |
0 |
Zap with fax outbound signature |
11:27AM |
5 |
yet another Asterisk and VMware question |
11:07AM |
0 |
7960 Stuck booting |
11:05AM |
1 |
ChanSpy and Sipura 2100 jitter. |
10:55AM |
6 |
FXO port trhoug optimum voice VOIP service |
10:42AM |
1 |
I need a Asterisk tech |
10:29AM |
3 |
TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone |
10:23AM |
0 |
New astGUIclient version released 1.1.5 |
10:09AM |
8 |
Incompatible destination (88) Error Message |
9:25AM |
0 |
three questions |
9:21AM |
1 |
Polycom IP500 / Registration Question? |
9:19AM |
0 |
Forwarding behavior question |
8:37AM |
1 |
Comedian annoucment files |
8:07AM |
3 |
PC for 8 line system |
8:03AM |
1 |
Weird issues with TDM400P |
8:00AM |
0 |
Ang: Voipjet experiment |
7:57AM |
3 |
Voipjet experiment |
7:34AM |
0 |
7960 + 7914 Problems |
7:01AM |
0 |
txfax spandsp |
5:46AM |
2 |
Possibly bad FXS module in TDM400P? |
5:24AM |
0 |
yahoo voice |
3:59AM |
2 |
TE405P / TE410P with 2nd generation firmware field upgradable? |
3:51AM |
1 |
Call recording, monitor & soxmix in Asterisk 1.0.9 |
3:42AM |
2 |
v92 modems |
2:09AM |
3 |
OT: Sendmail question |
1:31AM |
0 |
Status of app_sms in 1.0.9 |
12:43AM |
4 |
Billion BRI PCI card |
12:25AM |
0 |
ZapHFC E1 PRI (cwain) |
12:08AM |
0 |
txgain for SIP? |
12:08AM |
0 |
My users are using PSTN instead of VoIP |
12:05AM |
1 |
TE405P V2 changes? |
|
Thursday August 11 2005 |
Time | Replies | Subject |
9:20PM |
2 |
wildcard/FXO config |
9:04PM |
2 |
list in asterisk cli is getting too long |
6:33PM |
0 |
Call queues bug? |
5:52PM |
0 |
Patches 0002838 and 0002924 |
5:21PM |
1 |
Firewall will definately increasejittersinyourvoice conversation |
4:05PM |
0 |
Unexpected On Hook event |
3:57PM |
0 |
Join Martin O'Shield on >Yahoo! |
3:35PM |
3 |
is this possible with asterisk? |
3:12PM |
1 |
Install just to play with experiment |
3:10PM |
0 |
NVLineDetect and head after aug 2 |
3:06PM |
1 |
External channels getting connected |
2:22PM |
4 |
Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone |
2:14PM |
2 |
Newbie Question: Building an Asterisk systemtoreplace an old PBX but using existing phone |
2:09PM |
0 |
Nothing techhy, just a greeting |
1:47PM |
0 |
Re: 24. Privacy Manager (Andi Strain) |
1:44PM |
14 |
How to fix a Blue Alarm?? Line Noise? |
1:25PM |
1 |
PRI dropped calls w/ asterisk dropped betweenpstn & norstar |
1:04PM |
2 |
Newbie Question: Building an Asterisk systemto replace an old PBX but using existing phone |
12:57PM |
0 |
CVS HEAD 08-11-2005 & Definity G3R |
12:25PM |
5 |
Cisco 79XX and VLANS |
11:52AM |
9 |
Polycom IP301 and 501 with asterisk... |
11:12AM |
1 |
Cisco 7920 boot causes 7940 to release DHCP lease |
9:31AM |
0 |
disable initial music for call queue |
9:30AM |
1 |
Where to buy Sangoma cards? |
9:02AM |
0 |
meetme.conf and realtime |
7:58AM |
0 |
is there cdrs for sip |
7:35AM |
2 |
Is it mandatory to give power supplytoTDM400Pcard |
6:40AM |
2 |
Suggestion for VoIP router with QoS |
6:20AM |
5 |
Realtime + MYSQL |
5:34AM |
8 |
Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone |
5:04AM |
3 |
Comedian mail ignores mailbox greetings |
4:42AM |
1 |
More then one Tormenta 2 E1/T1 card on system. |
3:34AM |
1 |
MS Live Communication Server |
3:28AM |
2 |
re: how to set the voice message as email attachment ? |
3:07AM |
0 |
Sipura-3000 IP->PSTN scenrio |
3:01AM |
1 |
Ignoring the called number in the INVITE message |
2:03AM |
1 |
help on receive text |
1:02AM |
0 |
How to determine elapsed time of a call in progress? |
12:24AM |
2 |
Sip ports |
12:05AM |
1 |
Supervised transfer problem with BudgetTone |
12:03AM |
0 |
* behind NAT, client behind NAT(handytone 286), very strange behavior |
12:02AM |
1 |
IAX setup |
|
Wednesday August 10 2005 |
Time | Replies | Subject |
11:58PM |
2 |
Zultys ZIP 4x5 |
11:33PM |
0 |
No speech path |
11:28PM |
1 |
Error while calling |
11:20PM |
0 |
tdm400p / outbound zap prob |
10:21PM |
1 |
Join Martin O'Shield on Yahoo! Messenger! |
10:18PM |
1 |
Help, using SendText cmd sip message... |
7:56PM |
1 |
Help with calling Perl AGI interface |
6:29PM |
1 |
Addendum to my post re: BrookTrout TR1034 T.38 |
6:10PM |
0 |
T.38 Faxing w/ BrookTrout TR1034 FOIP Board |
5:50PM |
1 |
PRI dropped calls w/ asterisk dropped between pstn & norstar |
4:37PM |
1 |
real-time priority |
4:34PM |
2 |
ZAP bchan and dchan HELP!! |
4:20PM |
3 |
Hitachi wip5000 |
3:08PM |
8 |
Blank CIDName or CIDNum = "asterisk" |
2:29PM |
1 |
Problems with zaptel.conf |
2:06PM |
0 |
Blank faxes in rxfax. |
1:46PM |
3 |
Hard deskphone via wifi? |
1:27PM |
0 |
Problem with setting the right dialplan for german PRI E1 on TE405P from digium |
1:24PM |
2 |
does SIP works behind the NAT |
1:15PM |
1 |
Is it mandatory to give power supply toTDM400Pcard |
1:13PM |
0 |
Error reloading extension! |
12:43PM |
2 |
Firewall will definately increase jittersinyourvoice conversation |
12:41PM |
0 |
Building on Itanium |
12:30PM |
2 |
Is it mandatory to give power supply to TDM400Pcard |
12:24PM |
1 |
Firewall will definately increase jitters inyourvoice conversation |
12:04PM |
2 |
app_voicemail.c still looking for config file even I try to configure the voicemail from database. |
11:32AM |
1 |
Limiting the number of calls |
11:27AM |
3 |
Is it mandatory to give power supply to TDM400P card |
11:26AM |
1 |
Firewall will definately increase jitters in yourvoice conversation |
11:24AM |
0 |
Waring problem with different brand phone |
11:18AM |
1 |
Asterisk support of MF trunks? |
11:16AM |
0 |
Problem with channel allocation between BRI and PRI cards |
11:14AM |
0 |
Asterisk scaling with agent channels |
11:06AM |
2 |
Help me how to listen voicemail with SIP 7960 |
11:04AM |
1 |
T100P Problems |
10:59AM |
0 |
Radius + NAS with Asterisk |
10:57AM |
2 |
Firewall will definately increase jitters in your voice conversation |
10:56AM |
0 |
Vonage Click-2-Call |
10:46AM |
0 |
Asterisk mailing lists |
10:37AM |
1 |
realtime odbc/mysql eating connections |
10:35AM |
1 |
E&M to E&M Dialing - TE410P |
10:30AM |
0 |
Polycom 501 Do Not Disturb issue |
10:28AM |
4 |
GrandStream GSX-2000 strangeness |
10:05AM |
2 |
TDM40B and weird analog problem |
9:32AM |
0 |
audio fading in and out? |
9:25AM |
1 |
asterisk query mysql problem or bug? |
9:14AM |
1 |
chan_oh323.c:2706 oh323_request: Blocking outbound H.323 call due to call-limit violation. |
9:09AM |
0 |
Asterisk and Asterisk management portal issue |
9:02AM |
0 |
RE: Info / recommendation on either Audiocodes or Vegastream gateways |
8:42AM |
0 |
Numeric Pagers & Voicemail |
8:14AM |
1 |
App_Queue strategy=ringallfree (feature request, possible bounty) |
8:09AM |
1 |
h323 error when trying to start Asterisk |
8:05AM |
0 |
Problem with voicemail, invalid extension, no error handler |
8:03AM |
2 |
Help with TNT and Asterisk |
8:01AM |
0 |
Asterisk Stops Sending Data (CVS 20050809) |
7:46AM |
3 |
TE205P installation problem - ZT_SPANCONFIG failed on span 1 |
7:20AM |
1 |
where the debug log stored? |
6:49AM |
3 |
SRV implementation supporting priority |
4:39AM |
6 |
USB handset wanted |
4:10AM |
0 |
Asterisk and SER and Asterisks Queues |
2:55AM |
0 |
Single extension/user registers across multiple asterisk servers |
2:28AM |
1 |
Asterisk Call Queue Application |
2:20AM |
4 |
No audio when calling between internal phones |
2:13AM |
0 |
Asterisk & RTC Client API |
2:11AM |
6 |
will a firewall slow down asterisk? |
1:54AM |
2 |
Calling Extension directly |
1:54AM |
0 |
Yoda VG-400 and Asterisk Settings |
|
Tuesday August 9 2005 |
Time | Replies | Subject |
11:46PM |
2 |
Load Testing |
11:24PM |
0 |
Incoming call #2 sent to VM immediately whenalready on phone with incoming. |
11:06PM |
2 |
error compiling asterisk on solaris |
8:51PM |
8 |
call "load balancing" |
7:50PM |
4 |
Need some statistics & facts |
7:29PM |
0 |
Can I change the call waiting signal tone. |
6:05PM |
2 |
How to dial several extensions with different timeouts |
5:07PM |
0 |
Console Auto-Completion Lockup |
3:56PM |
1 |
Incoming call #2 sent to VM immediately when already on phone with incoming. |
3:17PM |
1 |
inbound caller id name pri - tnt - asterisk |
3:05PM |
1 |
Playback before Answer |
3:02PM |
2 |
ISDN DID |
2:45PM |
3 |
SIP-Trunk problem, Please help!!! |
2:40PM |
1 |
Asterisk and XML Applications |
2:14PM |
2 |
Stable or not? |
1:59PM |
0 |
channel_pvt.h not found |
12:56PM |
1 |
Com-On-Air (PCI/PCMCIA) chan drivers? |
12:15PM |
0 |
Sipura wrong password on invite |
12:06PM |
2 |
detaching console from foreground asterisk |
11:50AM |
0 |
Registration intervals |
11:03AM |
2 |
dvc 1000 support |
11:02AM |
2 |
X100P Wildcard - Hassle free clone? |
10:12AM |
2 |
Asterisk and Wave files problem |
10:02AM |
0 |
Connection Asterisk- Panasonic TDA200 |
9:26AM |
3 |
Build on Itanium fails |
9:07AM |
6 |
QoS General Question |
8:25AM |
0 |
Random Zap Channel Resets |
8:25AM |
2 |
Both lines in an ATA using the same UID/PASS |
7:57AM |
2 |
Playing GSM files in Windows? |
7:38AM |
0 |
H.323 vs SIP for small FXO gateways |
7:28AM |
1 |
TE110P flashing red/green when PRI connected ... continued |
7:22AM |
0 |
Echo during begining of incoming calls |
7:08AM |
1 |
CLI and Dial |
7:02AM |
0 |
looping through SER |
6:53AM |
3 |
First PRI |
5:22AM |
0 |
How to configure Outbound Proxy for REGISTER? |
3:09AM |
1 |
Incoming call action based on trunk |
2:53AM |
0 |
Cannot hear Music On Hold with SIP Phones |
2:23AM |
1 |
voip solution with SER, ASTERSIK and CCM |
|
Monday August 8 2005 |
Time | Replies | Subject |
11:54PM |
0 |
Calls to Turkey, any good providers? |
11:53PM |
0 |
Broadvoice europe plus calling plan quality |
11:51PM |
1 |
T1 versus PRI |
11:41PM |
0 |
Active channel, no users |
11:38PM |
1 |
FXO definition |
10:36PM |
0 |
queue-hold time + weight in astersk+acd |
9:37PM |
0 |
delay problem |
8:04PM |
0 |
FXO gateways / Audiocodes MP-108 |
8:01PM |
0 |
OT: Anyone having issues with sipphone? |
6:55PM |
1 |
SNOM Hint for MeetMe |
6:34PM |
0 |
Problems with cmd monitor |
6:24PM |
1 |
X100P with Caller-ID in Australia, |
6:19PM |
3 |
FXS - Don't want a Dailtone |
5:37PM |
2 |
Asterisk and .NET |
5:27PM |
1 |
Press # to continue / Findme |
5:09PM |
0 |
Re: asterisk rpms (was: Does anyone run Asterisk on FC4? with Digium's TDM40B cards) |
5:02PM |
1 |
Snom 360 4.0 firmware issue |
4:54PM |
0 |
Help interpreting channel stats? |
4:22PM |
0 |
OT: DTMF issues with Vonage forwarded lines |
4:06PM |
0 |
Question about agent queuing in Asterisk |
3:42PM |
6 |
IAX TO IAX call between two registered servers |
3:16PM |
1 |
Detecting hangup - TDM400P / X100P |
3:09PM |
0 |
ISDN D-Channel Problem / bristuff / qozap |
3:05PM |
2 |
zaphfc syslog flooding |
2:45PM |
1 |
FCC to require wiretaps from VoIP providers |
2:24PM |
0 |
IAX and Realtime... |
2:13PM |
0 |
howto let the media stream not passing saterisk? |
2:03PM |
1 |
Call Recording with * |
1:49PM |
0 |
Where is the asterisk DB file stored? |
1:48PM |
1 |
howto let the stream not passing asterisk |
1:40PM |
1 |
AGI perl problem |
1:35PM |
0 |
Voicemail Web Access Security |
1:30PM |
2 |
[OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS) |
12:47PM |
2 |
URGENT: Problems with PHP AGI... |
12:31PM |
0 |
Failed IAX Connection |
12:08PM |
0 |
Polycom IP600 Presence question |
11:49AM |
0 |
Call Quality Issues |
11:24AM |
7 |
info regarding hardware |
10:58AM |
0 |
Failed to authenticate user |
10:49AM |
0 |
Screening Sip Calls - Record() |
10:33AM |
0 |
Wired Interactions between Asterisk (Public) and Budgetone (behind NAT) |
10:22AM |
1 |
Call forward & SER as SIP router |
9:27AM |
4 |
DTMF issues with SIPPhone? |
8:52AM |
0 |
Config files for zaphfc in nt mode |
8:47AM |
1 |
IAX2 Encryption |
8:21AM |
0 |
Asterisk-to-IVR Problem |
7:26AM |
0 |
Packet loss concealment and G729 |
7:05AM |
0 |
trouble using variables with included contexts |
7:02AM |
0 |
Voicemail web access |
6:56AM |
0 |
g729 recording on asterisk using g729 enabledphone |
6:33AM |
0 |
g729 recording on asterisk using g729 enabled phone |
6:28AM |
4 |
Multiple MWI on a single phone? |
6:26AM |
0 |
Configuring TDM40B and X100P |
6:01AM |
0 |
Mediatrix 1204 setup |
6:01AM |
0 |
problem with callerid ( SetCIDName ) |
5:49AM |
1 |
Newbie with Cisco 7910 phones |
5:22AM |
1 |
Transfer a call from cell phone (pseudo-disa) |
5:21AM |
2 |
Stun support |
5:12AM |
3 |
Speex QoS |
4:37AM |
0 |
Using * and other gateways together |
3:59AM |
0 |
Need unique switchboard/op-panel written |
3:13AM |
1 |
Same action to multiple numbers |
2:58AM |
1 |
Found solution to my PHP AGI Script problem... |
2:57AM |
4 |
TE110P flashing red/green when PRI connected |
2:53AM |
1 |
problem in inbound calls |
2:46AM |
2 |
AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFOproblems |
2:42AM |
1 |
CVS not responding |
2:34AM |
0 |
Without E1 lines,how to test E400P card? |
1:57AM |
1 |
Setup faxing with latest CVS/STABLE |
1:52AM |
0 |
NT1 devices with analog ports on HFC based |
1:48AM |
0 |
zaptel compile to incorrect directory |
1:03AM |
3 |
Digium TE405P, caller id and migration to * |
12:26AM |
1 |
CDR TDS |
12:23AM |
0 |
RTC Client API & Asterisk |
12:04AM |
1 |
X100P with Caller-ID in Australia, anyone? |
|
Sunday August 7 2005 |
Time | Replies | Subject |
11:58PM |
1 |
How to config voicemail with mysql? |
11:35PM |
1 |
mysql sock location |
8:45PM |
0 |
list of T.38 providers on wiki: please contribute |
5:23PM |
1 |
http://www.voip-info.org/ front page taken out by spammer |
4:51PM |
0 |
VoicePulse Connect down Sunday evening? |
4:37PM |
0 |
NT1 devices with analog ports on HFC based ISDN BRI cards in NT mode and asterisk (chan_mISDN) |
4:36PM |
1 |
Unable to connect to FWD |
2:34PM |
1 |
request for clarification on Asterisk T.38 bounty |
12:34PM |
0 |
Using * and 3rd party GW together |
11:45AM |
3 |
z-machine + asterisk = fun! |
10:39AM |
1 |
voice prompt repository |
10:15AM |
3 |
Can call from iax extn but cannot call it - unable to cteate channel iax |
9:55AM |
4 |
Configuring Asterisk@home for Sipgate. |
9:45AM |
0 |
Calls from Asterisk to CallManager 3.0 how? |
9:13AM |
0 |
How to configure * for Net2phone using innomedia settings |
9:09AM |
0 |
ASTCC web can't connect to DB |
8:40AM |
0 |
How to configure/install ISDN Card |
6:58AM |
0 |
zaphfc HFC-S in nt mode but no dial tone after pickup |
3:52AM |
0 |
Planet sip phone and asterisk |
12:08AM |
0 |
Can't compile asterisk-oh323 on Mandrake 10 |
|
Saturday August 6 2005 |
Time | Replies | Subject |
11:07PM |
3 |
SPA 841 form SIPURA |
5:38PM |
2 |
Dialplan mapping for multiple outbound providers to determine best rates |
2:41PM |
1 |
Setup faxing with latest CVS |
2:13PM |
1 |
Cisco 7206 and Sample configs (Newbie) |
1:36PM |
0 |
Need Help RE Zultys Zip 2+ Basics |
12:43PM |
2 |
sip/rtp performance monitoring |
11:23AM |
0 |
g729 pass-thru for sip provider and g711 ulaw for conference and voicemail |
9:13AM |
0 |
SIP rejecting calls? |
9:00AM |
2 |
How to test H.323 |
8:02AM |
4 |
TDM400P - All extensions have same CallerID |
7:57AM |
1 |
BudgeTone 100 Woes |
6:52AM |
1 |
Extensions beginning with * |
6:04AM |
0 |
Latest Asterisk and Fedora Core 4 question |
4:50AM |
1 |
Voicemail -- newbie question |
4:08AM |
1 |
Queue_log all calls marked ABANDONED? |
3:00AM |
3 |
Does anyone run Asterisk on FC4? with Digium's TDM40B cards |
1:28AM |
2 |
low sound |
|
Friday August 5 2005 |
Time | Replies | Subject |
9:45PM |
2 |
Phone interface hardware |
7:35PM |
0 |
call outside from FXS through FXO |
7:01PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday 8/6/2005 |
6:10PM |
3 |
Very complicated dialplans? |
5:53PM |
1 |
TE405P Dropping Calls |
5:30PM |
1 |
starting asterisk with nice -5 |
4:46PM |
1 |
TE411P problem |
3:25PM |
0 |
Uniden UIP 1868 / Asterisk experiences |
2:47PM |
0 |
Partner II Expert Needed |
2:11PM |
0 |
Seeking Beta testers for enterprise mystery service |
12:41PM |
1 |
Switchboards |
11:30AM |
0 |
CallerID Problems. |
11:29AM |
0 |
number 'register => ' in sip.conf |
11:16AM |
4 |
Snom 360 and firmware 4.0 problem |
11:16AM |
3 |
how may channels |
10:46AM |
3 |
Is this echo problem down to IP Phone hardware? |
10:30AM |
3 |
Uniden UIP200 Opinions |
10:29AM |
1 |
Abwesenheitsnotiz: Nortel Option 11 and TE110P o f Digium |
10:25AM |
1 |
Asterisk MWI and Realtime |
9:54AM |
0 |
ATA186 can not generate dtmf |
9:47AM |
1 |
Nortel Option 11 and TE110P of Digium |
9:35AM |
0 |
Looking for IBM or HP Server Recommendation |
9:23AM |
2 |
Zaptel warning |
9:20AM |
1 |
Need Help Troubleshooting Broadvoice Connection |
9:08AM |
0 |
IAX Phone Pro Beta - New Version Available |
9:08AM |
1 |
No dial tone on BT100 |
8:53AM |
0 |
Masters changes / Line looses |
8:33AM |
3 |
Realtime IAX |
7:57AM |
0 |
Audio files problem - as usual |
7:33AM |
0 |
Another problem on queues |
6:36AM |
0 |
Phone hangups after a TEI check request |
6:34AM |
2 |
SIP signaling vs Media (Voice) Traffic |
6:31AM |
1 |
Asterisk (Comedian Mail) and AUDIX |
5:25AM |
8 |
asterisk registered in ser proxy |
5:19AM |
0 |
USB ISDN devices |
5:07AM |
0 |
IPManager has been released - the ultimate configuration tool for Asterisk |
5:04AM |
1 |
Is there a right place for a include_once statement in a PHP AGI script? |
4:44AM |
0 |
Roundrobin queue strategy broken ? |
2:56AM |
1 |
Problem running Astrerisk after defined card in /etc/asterisk/zapatal.conf |
2:02AM |
1 |
Cisco IP Phones on Asterisk: chan_sip or chan_sccp |
|
Thursday August 4 2005 |
Time | Replies | Subject |
11:30PM |
0 |
defining range of user in sip.conf |
11:18PM |
0 |
Rebooting GS phone thru sip_notify |
10:37PM |
4 |
Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone |
10:00PM |
1 |
PolyCom SoundPoint 300 and distinctive ring |
9:57PM |
15 |
ip phones |
9:57PM |
3 |
Cisco IP Phone 30 VIP |
9:25PM |
0 |
Asterisk payphone with voipjet for retail/resale purposes |
8:04PM |
1 |
HELP! X100P IRQ conflict w/ USB |
7:34PM |
1 |
app_txfax.c problem |
7:12PM |
1 |
application doesn't dial out... |
5:49PM |
0 |
ADIT 600 Expert needed |
5:26PM |
3 |
include behavior (word puzzle of the day) |
5:24PM |
1 |
Cvs Head |
5:22PM |
0 |
Asterisk in ACD configuration |
5:21PM |
1 |
no ring to callers? |
4:55PM |
0 |
AbsoluteTimeout Problems? |
3:53PM |
0 |
h.323 Call problem asterisk to\from lucent(avaya) definity |
3:37PM |
0 |
h323 CALL PROBLEM TO / FROM AVAYA(UCENT)inity |
3:30PM |
1 |
Asterisk and the IAD2431 via MGCP |
3:06PM |
5 |
newbiew extensions.conf question |
2:55PM |
1 |
Outbound Extension problem |
2:52PM |
0 |
How to log the different extensions dialed within a single call? |
2:10PM |
2 |
Some echo? |
1:55PM |
1 |
bristuff-0.2.0RC8m |
1:13PM |
1 |
Asterisk Voice Mail Server and older Executone PBX..can it be done? |
1:04PM |
1 |
Receiving Calls from FWD Network using IAX2 |
12:15PM |
1 |
Merlin Legend |
11:38AM |
0 |
BT102 phones giving strange errors |
11:15AM |
0 |
Re: Asterisk-Users Digest, Vol 13, Issue 25 |
11:05AM |
3 |
How scalable is asterisk |
10:34AM |
2 |
Polycom and Presence |
10:28AM |
2 |
TFTP - Good or Bad? |
10:23AM |
0 |
TDD over Asterisk |
10:07AM |
0 |
No rering on misoperation on SIP ATA |
9:55AM |
1 |
Callback question |
9:46AM |
0 |
Re: [Asterisk-Dev] The killer app for Asterisk in corporate deployment |
9:28AM |
1 |
CVS Down |
9:24AM |
1 |
Getting asterisk to work with callthroughs? |
8:54AM |
0 |
Agent channels |
8:47AM |
0 |
weird DTMF problem |
8:31AM |
2 |
[Asterisk-Dev] OPAL now supports IAX2 |
7:58AM |
0 |
RPMS & SRPMS of Asterisk STABLE & HEAD on i686 & PPC |
7:51AM |
1 |
send an sms through a gateway GSM (stargate) |
7:34AM |
6 |
Features you'd like to see in a GUI? |
7:16AM |
0 |
Voicemail advanced options, 5 to send a message not available |
6:52AM |
1 |
Asterisk, Tenovis, Fritz, capi problem |
6:31AM |
2 |
The killer app for Asterisk in corporate deployment |
6:10AM |
0 |
Best common practice for emailing conferences? |
5:55AM |
3 |
SIPPeersAction class file not found in the Asterisk-java.jar file |
5:53AM |
0 |
Call specified, but not found? |
5:25AM |
0 |
vmail.cgi question |
4:30AM |
0 |
Calls not cleared down if extra destinations or dial commands added to extension |
4:12AM |
2 |
Directory problem |
1:22AM |
1 |
REINVITE and Codecs |
12:20AM |
4 |
asterisk & cisco 7960 softkeys [Virus checked] |
|
Wednesday August 3 2005 |
Time | Replies | Subject |
11:43PM |
2 |
PLEASE REPLY, are you using an X101P |
11:39PM |
2 |
Send voicemail notification to SMS |
11:22PM |
0 |
Polycom ring volume |
10:44PM |
0 |
Asterisk TDM card connected to phone linesAND fax line |
10:31PM |
1 |
Attaching data to outgoing INVITE message . |
10:19PM |
2 |
ISDN BRI Funkyness |
10:17PM |
0 |
Dead spa841 |
9:43PM |
0 |
Windows client for sending fax using txfax - spandsp |
8:50PM |
0 |
Multiple CLI connections |
8:48PM |
2 |
MFC/R2 Mexico Unicall Blocked |
8:15PM |
0 |
Asterisk on FreeBSD-5.4 RELEASE : H323 audio problem |
7:09PM |
0 |
Vmail.cgi and realtime |
6:39PM |
0 |
register sip |
6:27PM |
1 |
64K ISDN call not passing thru |
6:16PM |
1 |
IAXy2 question? |
5:40PM |
3 |
inter-asterisk meetme |
5:06PM |
0 |
Attaching data to outgoing INVITE message |
5:06PM |
1 |
Voicemail Password crashing |
3:26PM |
1 |
chan_capi upgrade |
3:08PM |
0 |
chanspy not working with Agents |
2:49PM |
0 |
iax to iax severs |
2:34PM |
0 |
Asterisk Network Troubleshooting Help Needed - Will Pay $$$ |
2:26PM |
0 |
SIP call termination on PSTN lines |
2:09PM |
1 |
Incoming SIP from Cisco 7206 |
2:07PM |
0 |
OT - but very interesting speech application |
2:01PM |
0 |
Compile ZAPTEL warning and Strange Congestion |
1:13PM |
2 |
polycom 301 phone advice |
1:07PM |
0 |
Line Buttons (Key system behavior) |
12:53PM |
0 |
Voicemail Issues |
12:28PM |
2 |
Cisco ATA and a PayPhone |
12:10PM |
1 |
Asterisk support Shared Call Appearance Signaling? |
10:44AM |
0 |
AstriCon 2005 - Early Bird Registration Open (Free IAXy To First 50!) |
10:38AM |
0 |
Inter-Tel AXXESS failure: HDLC Bad FCS (8) onPrimary D-channel of span 1 |
8:58AM |
1 |
7970 SCCP configs? |
8:52AM |
0 |
Chan_bluetooth and AudioGateway phone [long] |
8:09AM |
0 |
IDSN 30 PRI UK |
7:43AM |
0 |
fax <--> grandstream 286 <--> asterisk <--> pstn |
7:32AM |
0 |
Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1 |
7:12AM |
4 |
Transfer to outside line. |
6:56AM |
1 |
Asterisk TDM card connected to phone lines AND fax line |
6:21AM |
1 |
AstLinux - Anyone running on a Soekris Engineering net4826 |
6:15AM |
1 |
Generic Question: Why should I use Asterisk over SIPxchange? |
6:04AM |
1 |
Is it possible to use CHAN_CAPI with ZAPHFC enabled card ? |
5:37AM |
0 |
How to test E400p card without E1 lines?thanks a lot |
5:30AM |
0 |
How to config incall ?I have a E400p card |
4:38AM |
3 |
Anyone know of an open source sip video phone like eyebeam available? |
4:19AM |
0 |
Is there an upper extension limit to Asterisk? |
3:46AM |
1 |
app_dbodbc for asterisk stable 1.09 |
3:03AM |
0 |
LG Goldstar GDK-186/162 question on voicemail |
2:51AM |
1 |
Database querie |
2:42AM |
1 |
call does not hangup after client quits |
1:59AM |
0 |
app_intercept |
12:41AM |
0 |
Installing a TE100P (Digium) card over Suse 9.2.. |
|
Tuesday August 2 2005 |
Time | Replies | Subject |
11:18PM |
0 |
sip ata's |
10:46PM |
5 |
TFTP Secondary Ports |
10:19PM |
4 |
same extension on multiple sip phones? |
9:25PM |
1 |
what phones support this when running with asterisk |
8:39PM |
3 |
"invalid extension" dilemma |
6:41PM |
0 |
spandsp fax problem |
6:03PM |
1 |
Paging systems from the phone... |
5:28PM |
0 |
Few questions about Asterisk |
5:09PM |
0 |
asterisk e&m echo problem |
4:56PM |
2 |
port forwarding ip to ip sip calls |
4:06PM |
0 |
Channel Lock problems |
3:38PM |
2 |
How to let ZAPHFC work with and act on different incoming MSNs? |
3:34PM |
1 |
asterisk.org beta site up! |
2:37PM |
1 |
Polycom Soundpoint 600 |
1:56PM |
1 |
Polycom Soundpoint 500 |
1:39PM |
0 |
list test - ignore me |
1:21PM |
0 |
AstLinux 0.2.8 released |
12:55PM |
6 |
Dell Servers |
12:33PM |
1 |
stale nonce |
12:26PM |
0 |
RE: What does it take? |
12:13PM |
1 |
DND Indication |
12:02PM |
0 |
Problem with attended transfers... |
12:01PM |
5 |
7970 SIP |
11:55AM |
0 |
codec question |
11:33AM |
0 |
ISDN phone no dialtone |
11:26AM |
1 |
Two questions about Asterisk Call Center |
11:22AM |
0 |
Oh323 Module - Not Loading Error - Unregistered channel type 'Modem' |
11:13AM |
0 |
app_rxfax errors |
11:11AM |
2 |
Channel Bank Help Please.... |
11:02AM |
1 |
Best way to connect asterisk to an traditional PBX |
10:59AM |
1 |
AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFO problems |
10:52AM |
0 |
TAPI driver: AstTAPI |
10:49AM |
5 |
Has Sixtel gone under? |
10:00AM |
1 |
Making a call on Asterisk... new thread or not? |
9:23AM |
0 |
Re: Minimum CPU required for >60 calls |
9:09AM |
0 |
Asterisk to Televantage |
9:05AM |
9 |
Polycom phones w/ two lines on different servers |
8:51AM |
0 |
Sip over VPN not working |
8:28AM |
12 |
WHat does it take |
8:27AM |
1 |
Ztdummy or Zaptel card on production server |
8:07AM |
0 |
can one specify "talking only" for a participant in app_conference |
8:03AM |
0 |
Asterisk as PSTN gateway, voice mail server with SIP |
7:29AM |
1 |
How to create a secret code to use myasterisk@home server's long distance plan from a public phone |
7:29AM |
1 |
Voicemail/Password Issue |
7:27AM |
0 |
New release: Queue Statistics 0.1 |
6:58AM |
0 |
Re: Asterisk-Users Digest, Vol 13, Issue 7 |
6:55AM |
1 |
How to create a secret code to use my asterisk@home server's long distance plan from a public phone |
6:54AM |
0 |
Polycom SoundPoint 600 : 10 seconds of delay when answering a call. |
6:33AM |
3 |
priority "a" in macro to access voicemail |
6:23AM |
0 |
Config extentions for ISDNphone (Phone autmatically calls internal extention) |
5:55AM |
0 |
Control IAXy Provisioning from a central |
5:51AM |
2 |
asterisk@home newbie extensions always busy |
5:37AM |
1 |
Strange DTMF issue with callback |
5:25AM |
0 |
Suggested System Specs - 20 ext, 8 Incoming Lines - Thanks |
4:49AM |
0 |
Dell SC420 and Interrupts |
4:45AM |
2 |
call center 20 seats |
4:30AM |
0 |
Hang up as soon as other party picks up call |
4:25AM |
0 |
Strange beeps in Calls |
4:24AM |
0 |
Asterisk & ISDN |
4:07AM |
0 |
Dialogic D/300/SC-2E1 |
4:03AM |
2 |
This should work right??? Any caveats that you guys know about? |
3:35AM |
1 |
How does TDM work? |
3:16AM |
2 |
Minimum CPU required for 60 calls |
3:14AM |
0 |
FW: WEB SIP Dialer |
3:10AM |
0 |
Festival not working with Asterisk 1.0.7_7 |
2:45AM |
0 |
strange asterisk issue |
2:14AM |
1 |
[Asterisk-Dev] Getting ISDN line restart problem with TE110P |
2:07AM |
1 |
Asterisk PSTN connectivity |
1:52AM |
0 |
Asking telephone no from caller |
1:48AM |
1 |
Config HFC-card in asterisk.(Config the phone and asterisk) |
|
Monday August 1 2005 |
Time | Replies | Subject |
10:40PM |
0 |
register Every user without auth |
9:48PM |
2 |
TDM400P REV I issues - ProSLIC vs TDM400P |
6:15PM |
0 |
Dialplan to dial SIP, but stop dial on analog pick up? |
6:14PM |
0 |
Configuring A@H with Analog Phones UPDATED |
6:13PM |
2 |
Configuring A@H with Analog Phones |
6:00PM |
1 |
ast_config not updating voicemail password |
4:58PM |
1 |
X100P/Caller ID: clidtest works, * complains [repost] |
4:13PM |
0 |
Issue with zapata.conf "immediate" setting |
3:52PM |
1 |
Voicemail envelope time is 4 hours ahead |
3:39PM |
1 |
How to install PHPAGI? |
2:37PM |
0 |
Sipura SPA-1001: Bad Outgoing Call Quality |
1:38PM |
5 |
Queue/Agents |
12:46PM |
0 |
Marc Spindt is out of the office |
12:07PM |
4 |
IAX Devices Recommendation |
12:03PM |
2 |
*@Home/Grandstream Call Transfer |
12:01PM |
4 |
test message - ignore me |
12:00PM |
0 |
Polycom IP500 Ringtone howto |
11:53AM |
4 |
g729 liscence question |
11:46AM |
7 |
List |
10:47AM |
1 |
IAX2, can't receive calls |
9:53AM |
3 |
two UA with the same usr/pwd |
9:10AM |
1 |
Warning: We're Zap/XX-1, |
8:17AM |
1 |
sip+nat+asterisk |
6:39AM |
0 |
How to force Requested transfer capability on BRI/PRI dial? |
5:49AM |
1 |
Is this maillist down? |
2:39AM |
0 |
iax2 trunking issues |
1:32AM |
0 |
announce-holdtime+ACD+asterisk |
12:51AM |
0 |
iax cdr problem |
12:37AM |
0 |
Music on hold problem. |