| Monday January 31 2005 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
Playing a file upon pickup (dial command?) |
| 10:49PM |
0 |
Timer for MeetMe on Mac OS X |
| 10:42PM |
7 |
TDM400 stopped working |
| 10:09PM |
2 |
H.323 |
| 9:07PM |
0 |
Asterisk MTBF studies |
| 6:52PM |
0 |
Intel chip IA98 |
| 6:48PM |
0 |
Telephone Line options in Asterisk |
| 6:31PM |
0 |
PRI got event: HDLC Bad FCS (8) |
| 4:58PM |
0 |
Budgetone ringing volume |
| 4:44PM |
5 |
RE: Answering Machine Function? |
| 3:44PM |
1 |
Cisco 7960 and AutoAnswer. |
| 3:36PM |
2 |
Developing an IP Phone |
| 3:30PM |
2 |
video conferencing bounty |
| 3:28PM |
0 |
Single or Dual Processor? High volume MeetM e |
| 3:16PM |
1 |
Asterisk at CeBit 2005 |
| 2:45PM |
0 |
Multiple calls placed in outgoing spool interfer with each other |
| 2:28PM |
0 |
Callerid on blind transfer w/ Cisco 7960 |
| 2:19PM |
0 |
Strange sip address? |
| 2:12PM |
1 |
A neat "hot seating" mplementation |
| 2:12PM |
0 |
Delayed echo |
| 2:12PM |
0 |
Caller ID Bug in v1.0.5 |
| 2:12PM |
2 |
PRI Dropped Calls - Audit, Restore, Idle state |
| 1:46PM |
2 |
Cisco phones config over internet |
| 1:40PM |
0 |
SRTP support |
| 1:21PM |
1 |
VoIP with Asterix |
| 1:17PM |
1 |
Call recorder based on * |
| 12:40PM |
0 |
PRI not hanging up the channel after Executing Hangup when dialing busy number. |
| 12:32PM |
2 |
Where does a newbie get started? |
| 12:24PM |
0 |
re: cdr_mysql and system time |
| 12:13PM |
1 |
Grandstream stops working after "Register Expiration" period has passed (dynamic registration) |
| 12:09PM |
4 |
line assignment on TDM400P |
| 12:07PM |
3 |
Multiport Fax over softphone |
| 12:04PM |
5 |
Q: PRI leading 0 (area access code) or 00 (country access code) missing on incoming callerid |
| 11:52AM |
1 |
chan_sccp bug / problem |
| 11:24AM |
0 |
Fast busy signal |
| 11:02AM |
0 |
Return call after transfer with no answer |
| 10:47AM |
0 |
Eyebeam Vs. Windows Messanger, |
| 10:37AM |
3 |
cisco 7960 image |
| 10:07AM |
2 |
Dialing out on TDM400p 4 port FXO |
| 9:58AM |
0 |
Sending forwarded calls out to a different provider |
| 9:37AM |
0 |
Linksys RT31P2-NA |
| 9:19AM |
0 |
AGI Processing Order |
| 8:52AM |
2 |
SPA-841 Call Waiting |
| 8:27AM |
0 |
music on hold that starts at beginning of file |
| 8:27AM |
0 |
Tuning MoH Volume |
| 8:01AM |
1 |
Audio Quality over LAN very bad |
| 7:37AM |
3 |
Announcement to caller when called party haspicked up - without initial Answer()? |
| 7:34AM |
5 |
Announcement to caller when called party has picked up - without initial Answer()? |
| 7:24AM |
0 |
Error while trying to execute asterisk |
| 5:28AM |
0 |
ISDN supplemetary services (Hold, Retrieve, 3PTY) on HFC-8S |
| 4:37AM |
1 |
SIP x NAT |
| 4:04AM |
0 |
AW: HDLC for Dummies? |
| 3:40AM |
0 |
Eicon Diva audio problem [Newbie] |
| 3:39AM |
1 |
HDLC for Dummies? |
| 3:30AM |
6 |
TDM400P specs clarification |
| 3:07AM |
0 |
Indication of transfer on display |
| 3:01AM |
1 |
congestion problem with only one number |
| 2:41AM |
3 |
Group Extension |
| 2:29AM |
3 |
NAT and SIP |
| 2:18AM |
1 |
Instant Messaging |
| 12:24AM |
2 |
Trunked IAX or not |
| |
| Sunday January 30 2005 |
| Time | Replies | Subject |
|---|
| 10:55PM |
3 |
how to stop ringing after congestion. |
| 10:34PM |
4 |
Zap channels in AU hanging up on STD pips |
| 9:46PM |
1 |
DIAX softphone - Asterisk server rejecting |
| 9:16PM |
1 |
Slackware + Asterisk + asterisk-addons |
| 8:41PM |
4 |
detailed asterisk howto |
| 8:33PM |
1 |
x100P wildcard discontinued ? |
| 8:29PM |
2 |
x100p issues + TDM400P |
| 8:27PM |
0 |
Hitting IOCTL?? |
| 8:16PM |
7 |
Japan |
| 7:00PM |
0 |
Meetme2 web - nothing happens on click ? |
| 6:59PM |
4 |
Processing incoming calls with multiple contextst over PRI |
| 6:05PM |
3 |
Asterisk friendly VoIP providers |
| 4:35PM |
0 |
OH323 compile error : CVS-HEAD |
| 4:30PM |
1 |
Monitor calls timeout |
| 4:29PM |
1 |
Trying to make but it fails |
| 3:56PM |
0 |
conference room capacity question |
| 3:47PM |
0 |
302 Moved temporarily problem / Sipura 3000 |
| 2:32PM |
0 |
Caller ID on H323 |
| 2:21PM |
0 |
One way call when the * server and phone in a local network |
| 1:44PM |
3 |
Callgroup with bristuff ISDN? |
| 1:40PM |
1 |
IAX2 firmware for PA168x (Giptel G100, Siptronic ST-100 etc) |
| 12:20PM |
3 |
Single or Dual Processor? High volume MeetMe |
| 12:10PM |
1 |
Caller ID spoofing |
| 10:22AM |
0 |
D/41D |
| 9:44AM |
0 |
Setting call forward for Agent's in a Queue |
| 9:37AM |
0 |
Vservices.inv of Julian Pawlowski anoyne has the macro-dailer for this? |
| 9:24AM |
5 |
agent logoff |
| 8:53AM |
0 |
newlines in application data strings (e.g. userevent) |
| 8:13AM |
0 |
Can I start recording during call - is priority "a" active only in voicemail ? |
| 7:02AM |
4 |
Asterisk on MS Virtual Server |
| 4:17AM |
2 |
where to buy x100p |
| 4:15AM |
2 |
widcard x100P doubt |
| 3:40AM |
1 |
Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe |
| 3:15AM |
1 |
Vocera Badges |
| 2:14AM |
0 |
xten x-lite eyebeam |
| 2:01AM |
1 |
Strange Crash |
| |
| Saturday January 29 2005 |
| Time | Replies | Subject |
|---|
| 8:13PM |
0 |
Cisco BRI & SIP |
| 7:17PM |
2 |
Silly question: Why multiple lines on SIP phones? |
| 7:01PM |
0 |
RE: Asterisk-Users Digest, Vol 6, Issue 463 |
| 6:43PM |
2 |
SIP native bridge problem |
| 6:01PM |
1 |
Asterisk@home and Zap Channels |
| 4:51PM |
1 |
Please help, Zap channel hangup TE405P |
| 3:48PM |
0 |
Unable to remove Monitor IN / OUT wav files - Timing error |
| 2:50PM |
1 |
Subject: RE: Q: Can I over-ride the value of caller ID |
| 2:49PM |
1 |
Asterisk@home problem installing CentOS .. |
| 2:19PM |
1 |
Integration PBX |
| 1:20PM |
0 |
Support for Dialogic 4 or Dialogic Proline2V |
| 1:13PM |
2 |
Call rejected by FWD: Unable to negotiate codec |
| 12:33PM |
0 |
What was the conclussion of the R2 test in Mexico?? |
| 12:20PM |
0 |
Cisco/Lucent/Asterisk Guru needed |
| 12:12PM |
7 |
Sipura SPA-841 auto-answer support [patch] |
| 11:17AM |
0 |
Adding more links to the Navigation box in voip-info.org? |
| 10:31AM |
3 |
Channel Bank Echo |
| 10:24AM |
0 |
Adding digits to incoming callids depending on context? |
| 10:18AM |
2 |
TE405P w/ Intel SE7210TP1_E Motherboard |
| 9:56AM |
1 |
FS- ideal starter pack. 1 X100P and 1 Grandstream Budgetone-101 |
| 9:41AM |
1 |
PyAsterisk Download? |
| 9:38AM |
1 |
ISDN in US? |
| 9:36AM |
2 |
asterisk tries to dial out on lines already in use. |
| 9:13AM |
0 |
SIP Caller ID Number vs. Caller ID Name |
| 9:11AM |
2 |
Server auto Fallback |
| 8:15AM |
3 |
How to use ASTCC with SIP ?? |
| 7:48AM |
1 |
IAX2 Asymmetric Latency |
| 6:11AM |
4 |
PRI for Data and Voice |
| 4:51AM |
0 |
MyPBX model-1 |
| 4:00AM |
1 |
asterisk+h323+rh9 |
| 1:11AM |
2 |
problem in compiling asterisk addon |
| 12:55AM |
1 |
Asterisk @ Home 0.4 w/ Broadvoice + 5 SIP Phones How To |
| 12:38AM |
1 |
*1.0.5 CAN NOT find my sip.conf |
| 12:28AM |
1 |
Disable Reinvite on a per call basis. |
| 12:19AM |
0 |
upgrading to *-1.0.5 on Gentoo; error cdr_mysql.conf': Not found |
| |
| Friday January 28 2005 |
| Time | Replies | Subject |
|---|
| 11:28PM |
3 |
extensions.conf - redundancy removal |
| 11:06PM |
2 |
Direct MP3 channel Black Hole? |
| 10:11PM |
1 |
incoming calls produce multiple quarter rings andasterisk never answers. |
| 10:09PM |
0 |
incoming calls produce multiple quarter rings and asterisk never answers. |
| 10:01PM |
1 |
Meetme2? |
| 9:23PM |
1 |
Asterisk Prepaid Application Help |
| 9:01PM |
2 |
IP Phone for IP PBX |
| 8:11PM |
6 |
iaxComm version 1.0 released |
| 7:38PM |
1 |
FC3 + udev + Asterisk v1.0.3 - Temporary Fix |
| 7:38PM |
17 |
Speech Recognition |
| 7:16PM |
2 |
Nortel --> Asterisk-------->Asterisk |
| 6:45PM |
1 |
Problmes compilling * |
| 6:24PM |
0 |
Outgoing Call Block |
| 5:44PM |
1 |
Putting IP behind firewall |
| 5:36PM |
0 |
No Video With Eyebeam |
| 5:00PM |
4 |
Call Waiting Audio Prompt |
| 3:52PM |
1 |
Fedora Core 3 / Asterisk / TP100 Wildcard |
| 3:36PM |
0 |
PPP over T100P: Using a subset of channels does not always work correctly. |
| 3:09PM |
0 |
New Polycom SIP offerings |
| 2:44PM |
2 |
Who is in control Voicetronix OR Asterisk |
| 2:31PM |
0 |
fax/data/phone switch interfering with voip |
| 2:30PM |
2 |
Record inbound and outbound calls to and from one number. |
| 2:25PM |
4 |
ISDN Hardware |
| 2:22PM |
0 |
ANNOUNCEMENT : NEW CallingCard ApplicationforAst erisk |
| 2:22PM |
1 |
Integrating with existing 1BRI, 6 POTS Panasonic PBX ? |
| 1:51PM |
0 |
asterisk@home voicemail issue |
| 1:50PM |
0 |
re: Polycom |
| 1:44PM |
2 |
Polycom changing policy - allowing firmware downloads? |
| 1:35PM |
0 |
Asterisk Prepaid Applications Comparison |
| 1:03PM |
1 |
MusicOnHold with no sound card? |
| 12:18PM |
3 |
FWD and IAX2 |
| 11:53AM |
1 |
MoH does not de-attach |
| 11:50AM |
1 |
Festival Jittery (bad udp checksum) |
| 11:47AM |
4 |
FW: FAQ missing info? Asterisk@home V 0.4 |
| 11:43AM |
1 |
1.0.3-BRIstuffed |
| 11:42AM |
0 |
two OpenH323 vulnerabilities |
| 11:42AM |
0 |
asterisk call flow diagrams for ser voicemail combo |
| 11:30AM |
3 |
reason 24 (Call ended with Q.931 cause) |
| 11:25AM |
3 |
chan_iax2.c problem? |
| 11:23AM |
5 |
Eyebeam - asterisk - Messenger |
| 11:14AM |
0 |
Problems with H323/G729--No NATting and no Dynamic IP involved... |
| 10:46AM |
1 |
* acting as IP-Phone? |
| 10:35AM |
1 |
adit 600 fxo ports immediately "answers" outgoing calls (even if not connected to line) |
| 10:22AM |
1 |
Minimum Setup |
| 9:25AM |
1 |
error while trying to install astcc |
| 8:48AM |
2 |
Fwd and Tollfree |
| 8:20AM |
0 |
Trying to use Dial with D option.. |
| 8:16AM |
1 |
Authentication against voicemail password database |
| 8:08AM |
2 |
zap FXO channel - wait for N seconds before answer |
| 7:48AM |
1 |
Sipura SPA-841 with Asterisk |
| 7:31AM |
2 |
Problem with chan_sccp and cisco 7960 |
| 7:28AM |
1 |
Where can I find good doc on AGI? |
| 7:22AM |
0 |
[Asterisk-biz] e164.org update |
| 7:19AM |
0 |
STUN |
| 7:11AM |
2 |
redirect different phone number to different IP phone |
| 7:01AM |
0 |
Sipua SPA-2000 and liong delay afterdialling number |
| 6:23AM |
1 |
Bristuff and Realtime |
| 5:35AM |
1 |
1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping |
| 4:14AM |
3 |
Sipua SPA-2000 and liong delay after dialling number |
| 3:41AM |
1 |
Command to light MWI on 7940 /7960 |
| 3:09AM |
0 |
Continuously ringing Zap/4-1 TDM11B All of a sudden ?[Urgent Pls] |
| 2:15AM |
1 |
does asterisk support instant messaging? |
| 1:40AM |
4 |
Ouch ... error while writing audio data: : Broken pipe |
| 1:18AM |
0 |
asterisk CVS rpms for FC1 updated |
| 12:48AM |
0 |
Register replicaton and HA * |
| 12:15AM |
2 |
I want to display my numbers for incoming calls when some one dials my number from any where |
| |
| Thursday January 27 2005 |
| Time | Replies | Subject |
|---|
| 11:30PM |
3 |
Caller ID in AU |
| 11:12PM |
2 |
Q: Can I over-ride the value of ${CALLERIDNAME} ? |
| 10:02PM |
1 |
Dial and Macro Do not seem to be working |
| 9:02PM |
0 |
SIP CANCEL problem |
| 7:54PM |
0 |
LiveVoip Expanded Codec Support & Feb Sale 1.2 Cents a Min USA & Canada |
| 7:02PM |
1 |
ChanIsAvail not working |
| 5:35PM |
0 |
Need some advises configuring asterisk to callover INTERNET |
| 4:56PM |
1 |
OT: iax.cc/sixTel local DID question |
| 4:42PM |
3 |
Voicemail attachment not being emailed out |
| 4:37PM |
0 |
Problems making SIP URL outgoing dial |
| 4:13PM |
1 |
Trouble with Quicknet Linejack |
| 3:44PM |
0 |
Asterisk CVS on FreeBSD-stable gmaking result |
| 3:02PM |
0 |
Channel Groups? |
| 3:01PM |
1 |
Stumped by BroadVoice SIP |
| 2:46PM |
1 |
Asterisk auto-dial out deliver message |
| 2:21PM |
1 |
Making digital/data calls through asterisk |
| 1:47PM |
1 |
Outgoing call on Zap channel always gives DIALSTATUS ANSWER even when BUSY? |
| 1:43PM |
3 |
Tortoise CVS download for Asterisk Docs |
| 1:23PM |
0 |
Re: Asterisk-Users Digest, Vol 6, Issue 432 |
| 1:19PM |
1 |
Digium and Intel Chipset compatability |
| 1:18PM |
3 |
SIP + NAT = horrible mess |
| 1:18PM |
1 |
Bad ECHO problem after upgrade to HEAD version |
| 12:43PM |
0 |
X100P/Zaptel on Gentoo Sparc64 |
| 12:35PM |
3 |
Linux Bridge + QoS Shaper HOWTO available |
| 12:35PM |
2 |
Avoiding queue retries without hangs? |
| 12:29PM |
0 |
Re: Asterisk-Users Digest, Vol 6, Issue 431 |
| 12:20PM |
0 |
RE: 2 questions regarding call ques. |
| 12:16PM |
0 |
Asterisk @ Home & BroadVoice (Outbound) help |
| 12:11PM |
1 |
Hold music while ControlPlayback is paused? |
| 11:49AM |
1 |
CallerID for incoming SIP calls to Asterisk connected phone |
| 11:46AM |
0 |
Re: Asterisk-Users Digest, Vol 6, Issue 430 |
| 11:14AM |
1 |
Random hang ups during long calls |
| 11:03AM |
0 |
Re: Asterisk-Users Digest, Vol 6, Issue 429 |
| 10:58AM |
0 |
AW: HEELP!! with Eyebeam |
| 10:28AM |
2 |
CISCO 7905 Phone Weirdness |
| 10:14AM |
4 |
Changing mailbox greeting |
| 10:11AM |
0 |
differentiate the incoming from the outgoing calls in PSTN line |
| 9:53AM |
2 |
Soft phone sound quality help |
| 9:50AM |
0 |
How can I check the selected codec for a call? |
| 9:17AM |
0 |
Need some advises configuring asterisk to call over INTERNET |
| 9:07AM |
1 |
TDM-400P + CallerID |
| 8:25AM |
2 |
Adit 600 |
| 7:54AM |
1 |
Am I missing something really basic here?????helpwith Asterisk@home {Scanned} |
| 7:34AM |
1 |
Directory service of voicemail extensions |
| 7:21AM |
5 |
iax.cc / sixtel are they legitimate? |
| 7:12AM |
4 |
/usr/bin/ld: cannot find -lidn |
| 6:58AM |
0 |
Re: Polycom and call waiting again... |
| 6:21AM |
0 |
HEELP!! with Eyebeam |
| 5:37AM |
1 |
analog lines via channel bank -- |
| 5:28AM |
2 |
SoftClient for Pocket PC |
| 5:10AM |
1 |
analog fax on ericsson BP250 - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250 |
| 5:10AM |
2 |
Channel Restart - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250 |
| 5:09AM |
2 |
Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250 |
| 4:52AM |
0 |
Grandstream setup woe and solution |
| 4:51AM |
0 |
Asterisk@home and TDM400P cards... |
| 4:38AM |
0 |
Com-on-Air - DECT card |
| 4:23AM |
0 |
ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk |
| 4:21AM |
0 |
res_python |
| 4:00AM |
0 |
enter/leave sound with meetme adminmenu |
| 3:18AM |
0 |
How to check sip channel with smoething similar to ping ? |
| 3:17AM |
1 |
Moh in meetme doesn't work if I transfer to meetme |
| 2:57AM |
1 |
Asterisk chooses invalid outgoing interface (IAX2, virtual interfaces) |
| 2:35AM |
0 |
DUNDi on Asterisk |
| 1:35AM |
0 |
Re: Howto Setup TFTP server on Linux for Cisco |
| 1:00AM |
0 |
Problem with OpenPhone->Asterisk |
| 12:20AM |
3 |
Festival as background |
| |
| Wednesday January 26 2005 |
| Time | Replies | Subject |
|---|
| 10:07PM |
2 |
Call Announce, Dial 1 to Accept, Dial 2 to send to VoiceMail |
| 9:36PM |
0 |
dialplan logic for conditional DISA on incom ming 800 number |
| 8:31PM |
0 |
dialplan logic for conditional DISA on incomming 800 number |
| 8:12PM |
3 |
phone rings when I'm using it over VOIP - WHY? |
| 8:10PM |
0 |
Firefly reject problem - it just keeps ringing |
| 8:06PM |
0 |
Void callerid info on iax clients, but OK from local extensions or on SIP clients |
| 7:41PM |
0 |
New version of AMP - 1.10.006 |
| 7:15PM |
0 |
IAXy problems -- and no documentation |
| 7:07PM |
2 |
I need Help everyone I just bough my Xten Eyebeam |
| 6:55PM |
0 |
Is it possible to use native transfer in a call file? |
| 5:44PM |
0 |
resolved Asterisk + Broadvoice error |
| 5:37PM |
0 |
7900 Problem with Asterisk 1.0.1 and OH323 |
| 5:27PM |
0 |
Asterisk @ Broadvoice (I know it's been covered, but odd error) |
| 5:24PM |
1 |
Cmd READ and # |
| 4:50PM |
1 |
Cisco 7905/7912, SIP, g729 and DTMF setup |
| 4:33PM |
0 |
Cannot get * to work on VIA 800 MOBO |
| 4:30PM |
1 |
Re: bellster.net - GREATadvance |
| 4:18PM |
0 |
[Fwd: Re: [Asterisk-biz]bellster.net- GREATadvance] |
| 4:09PM |
1 |
How to make channel busy signal? |
| 4:00PM |
2 |
ANNOUNCEMENT : NEW CallingCard Application forAsterisk |
| 3:13PM |
1 |
Inbound analog Telco line not answered |
| 2:59PM |
1 |
mySQL-sipfriend dials to another SIP-endpoint - How to set the from-user |
| 2:28PM |
0 |
priority -1 |
| 2:26PM |
1 |
ANNOUNCEMENT:NEWCallingCardApplicationforAsterisk |
| 2:06PM |
0 |
supported ip phones (3com) |
| 1:59PM |
4 |
IAXy Hung, Power-cycle Required |
| 1:59PM |
0 |
VICI dialer help... |
| 1:55PM |
0 |
ANNOUNCEMENT : NEW CallingCardApplication fo rAsterisk |
| 1:48PM |
1 |
Asterisk as root in realtime vs. non-root asterisk ? |
| 1:46PM |
1 |
Firefly as Asterisk SIP client - qualify works ? |
| 1:02PM |
0 |
AMP-IAX2 trunk issue |
| 12:47PM |
1 |
channel numbering |
| 12:42PM |
2 |
BroadVoice Outgoing CallerID |
| 12:30PM |
1 |
IAX/SIP Softphone with G729 |
| 12:27PM |
0 |
Simple problem - call another phone on Busy |
| 12:09PM |
4 |
A working BroadVoice config example |
| 12:00PM |
1 |
native MOH with Asterisk 1.0.5 - any news? |
| 12:00PM |
1 |
Asterisk drops calls - why ?? |
| 11:48AM |
1 |
SIP called number on incoming call |
| 11:13AM |
0 |
Re: Asterisk-Users Digest, Vol 6, Issue 404 |
| 11:11AM |
1 |
Dialogic Boards |
| 10:53AM |
0 |
E100P echo on UK PRI |
| 10:05AM |
2 |
ANNOUNCEMENT : NEW CallingCard Application for Asterisk |
| 9:52AM |
1 |
TDM400P/TDM22B dialing issue |
| 9:48AM |
0 |
ulaw blank spots but gsm fine |
| 9:43AM |
4 |
No ringback on IAX channel after selecting menu option |
| 9:34AM |
3 |
TFTP Server Facing the Internet |
| 9:33AM |
0 |
Need help for a quick fun t-shirt/polo project - graphics artist wanted |
| 9:16AM |
0 |
Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 24.172.221.22 |
| 8:33AM |
0 |
Restart in the DISA to the beginning |
| 8:18AM |
0 |
ZT_CHANCONFIG failed on channel 11: Function not implemented (38) |
| 7:57AM |
1 |
Rining Issues |
| 7:54AM |
0 |
RE: Howto Setup TFTP server on Linux for Cis co 7960 |
| 7:52AM |
5 |
Polycom IP 600 - 1.3.1 |
| 7:42AM |
0 |
ParkAndAnnounce +${ALERT_INFO} |
| 7:19AM |
0 |
HFC-S card problems |
| 7:00AM |
7 |
Howto Setup TFTP server on Linux for Cisco 7 960 |
| 6:56AM |
0 |
chan_capi audio issue |
| 6:50AM |
4 |
Howto Setup TFTP server on Linux for Cisco 7960 |
| 6:36AM |
1 |
Am I missing something really basic here????? help with Asterisk@home |
| 6:23AM |
2 |
Telrad + E&M T1 Trunk |
| 6:16AM |
9 |
Cisco 7960 Message Light on multiple phones |
| 6:11AM |
2 |
off topic - DECT phones with FSK VMWI in the UK |
| 5:59AM |
0 |
Polycom IP600 stuck at "Running App = sip.ld"(was: Re: Polycom 1.4.1 firmware for IP500/IP600) |
| 5:31AM |
1 |
Asterisk with PSTN Help........needed!!!!!!! |
| 5:30AM |
0 |
Cannot get call transfers working |
| 5:11AM |
1 |
VoIP QoS with PIX |
| 5:07AM |
2 |
Issue with res_config_mysql.so in latest CVS |
| 4:52AM |
1 |
[Fwd: Re: [Asterisk-biz] bellster.net - GREATadvance] |
| 4:24AM |
3 |
setup questions- many users, little use |
| 4:03AM |
2 |
optimumvoice |
| 3:58AM |
1 |
interested in your opinion about FWD and iaxtel |
| 3:19AM |
0 |
Polycom boot server problem |
| 2:39AM |
2 |
ASTCC Trunks |
| 2:06AM |
1 |
Callmanager and Asterisk problem |
| 1:42AM |
0 |
Getting a Wildcard TE110P working on E1's in Australia |
| 12:31AM |
1 |
cant do it in CLI anymore? |
| |
| Tuesday January 25 2005 |
| Time | Replies | Subject |
|---|
| 10:26PM |
0 |
New RPMS for FC1 |
| 9:47PM |
0 |
Caller ID w/Name Providers??? |
| 9:28PM |
1 |
Asterisk@Home initial setup |
| 9:23PM |
1 |
softphone headsets |
| 9:11PM |
2 |
Another BroadVoice Problem |
| 8:53PM |
2 |
TDM400 - channel out to lunch? |
| 8:36PM |
2 |
Tall free number via FWD over IXA2 |
| 8:10PM |
0 |
Perfect billing solution for *? |
| 7:39PM |
1 |
Asterisk@home with Wildcard TDM400P card. |
| 7:18PM |
0 |
Dial command announcement |
| 6:55PM |
2 |
DTMF digit dropping |
| 6:27PM |
2 |
fwd IAX2 error |
| 5:51PM |
0 |
calleridname from chan_sip (mysql_sipfriends) |
| 5:09PM |
2 |
Interesting bellster issue |
| 5:06PM |
4 |
BroadVoice Help |
| 4:55PM |
2 |
Re: [Asterisk-biz] bellster.net - GREAT advance |
| 4:49PM |
0 |
E100P vs TE110P & Echo |
| 4:33PM |
1 |
Anyone having problems with LiveVoIP? |
| 4:28PM |
8 |
grandstream budgetone-100 updates |
| 4:18PM |
5 |
Polycom and call waiting again.. |
| 3:36PM |
1 |
Server side three-way calling with SIP channel |
| 3:30PM |
2 |
Polycom 1.4.1 firmware for IP500/IP600 |
| 2:39PM |
2 |
Problems splicing Asterisk with a TE405P between Arcor E1 PRI and Ericsson Business Phone 250 |
| 2:25PM |
1 |
SIP clients and double NAT |
| 1:29PM |
0 |
dial-back, call-back, what, is it called? |
| 1:16PM |
1 |
New ip billing solution?? any updates? |
| 12:57PM |
2 |
R2 in Bolivia |
| 12:54PM |
0 |
RE: Question regarding phones with multiple line appearances. |
| 12:37PM |
2 |
New native assisted transfer (atxfer) usage info required |
| 11:57AM |
4 |
Unable to Specify Channel 1 - no such device or address |
| 11:22AM |
1 |
Bellster and DTMF |
| 11:09AM |
1 |
BroadVoice Or VoicePulse ? |
| 11:05AM |
1 |
Re: I think your problem has to do with how you set the variable. |
| 11:00AM |
6 |
TDM400P Dell 1850 Server |
| 10:56AM |
0 |
probably error in chan_capi |
| 10:53AM |
1 |
HEAD vs STABLE |
| 10:10AM |
5 |
One Ring Mystery |
| 9:54AM |
0 |
Marked users with meetme2 .... |
| 9:53AM |
3 |
Configuring VLAN takes ages |
| 9:48AM |
2 |
SIP UDP ports on firewal to open |
| 9:44AM |
3 |
x-lite with wireless connection |
| 9:34AM |
0 |
coredumping on MusicOnHold |
| 9:23AM |
1 |
Am i in control after i dial? |
| 8:52AM |
1 |
iax java client |
| 8:19AM |
0 |
BackupPc_nightly crashing with Perl chdir errors |
| 8:19AM |
3 |
AMP with SUSE 9.2 |
| 8:16AM |
0 |
Goto invalid extension doesn't go to 'I' when in a macro. |
| 8:02AM |
0 |
Re: [Fwd: Re: [Asterisk-biz] bellster.net |
| 7:42AM |
0 |
Directory() ringing problem |
| 7:19AM |
0 |
Asterisk auto-dial out with .call files: Can I provide caller ID to second extension ? |
| 7:14AM |
4 |
Asterisk HEAD ->> Stable schedule? |
| 7:09AM |
1 |
TE110P yellow errors |
| 7:08AM |
2 |
Cisco 7940/7960 |
| 7:06AM |
3 |
OT: pinout for"standard"telephoneheadsetrequired.? |
| 6:22AM |
1 |
Codec mismatch between SIP (BT) and IAX Phone |
| 6:09AM |
1 |
Terminiation in the UK. |
| 5:47AM |
0 |
OH323 Cisco Transfer Key |
| 4:21AM |
0 |
Mediatrix voip gateway 1124 and 1204 in UKsetting |
| 4:05AM |
1 |
Problems with H323 channels |
| 3:07AM |
8 |
BUSY-tone on incoming calls? |
| 2:42AM |
1 |
SER Prob |
| 2:40AM |
3 |
Bristuff ZapHFC and Loosing D-Channel |
| 1:08AM |
1 |
Turn off DTMF recognition pending on CallerID |
| 12:52AM |
0 |
FXO and groups |
| 12:46AM |
1 |
Dialplane slip |
| |
| Monday January 24 2005 |
| Time | Replies | Subject |
|---|
| 11:54PM |
1 |
who used ser and asterisk? |
| 9:37PM |
1 |
How to reset IP600 with no password? |
| 9:31PM |
2 |
Correct way to update Asterisk |
| 9:08PM |
1 |
(no subject) |
| 8:55PM |
2 |
IP FXS channel bank |
| 8:37PM |
1 |
Nufone and Dialing Out |
| 8:12PM |
0 |
Asterisk@Home 0.3 and the Wildcard TDM400P |
| 7:37PM |
1 |
Asterisk Dial Out Issues - POTS Line |
| 7:29PM |
2 |
PrivacyManager not Working |
| 7:29PM |
0 |
chan_iax2.c:5441 socket_read: Rejected connect attempt from |
| 7:28PM |
1 |
What softphones for commercial use ? |
| 7:22PM |
3 |
[Fwd: Re: [Asterisk-biz] bellster.net - GREAT advance] |
| 7:19PM |
1 |
Realtime voicemail question |
| 7:17PM |
1 |
(no subject) |
| 6:43PM |
0 |
HFC-S cards in UK |
| 6:41PM |
0 |
Cisco Maintence Contract for my 7960 |
| 5:51PM |
0 |
size and quality of audio clips effect the playback?? |
| 5:13PM |
3 |
TDM400 in aging Dell Optiplex |
| 4:54PM |
1 |
SetGroup and CheckGroup problems |
| 4:40PM |
1 |
.call file creation |
| 4:04PM |
1 |
asterisk@home and capi |
| 3:38PM |
0 |
Volume on Zap channels (T1) |
| 3:29PM |
0 |
G.729 and mutualphone service |
| 3:09PM |
0 |
Voicetronix OpenSwitch6 with 10-digit Dialing |
| 2:58PM |
2 |
SIP-T Support (I got my head in an SS7 cloud) |
| 2:47PM |
2 |
T1 E&M vs PRI question |
| 2:45PM |
0 |
Asterisk v1.0.1 Cisco 7960 Sip v7.3 |
| 2:15PM |
1 |
FX CallerID |
| 2:08PM |
2 |
"Inband DTMF is not supported on codec G.711 u-law. Use RFC2833" |
| 1:57PM |
1 |
anyone got a 405 to work on a DL380? |
| 1:29PM |
3 |
cepstral integration with * using AGI? |
| 12:38PM |
6 |
Damn DTMF Beeps on my calls |
| 12:33PM |
3 |
Network Test Tool? |
| 12:23PM |
0 |
auto-dial out (.call file) failure detection |
| 11:54AM |
1 |
Asterisk -> static nat -> laptop w/siproxd -> cisco 7960 |
| 11:16AM |
7 |
Athlon 64 for Asterisk? |
| 11:11AM |
0 |
Missing Variable in Local Channels |
| 11:07AM |
2 |
Wildcard TE405P and TDM400 - TDM not working |
| 10:18AM |
4 |
Is Voice Pulse Connect good ? |
| 10:15AM |
2 |
Inbound Errors |
| 10:03AM |
0 |
Need some help with G729 passthru |
| 9:56AM |
2 |
Multiple X100P |
| 9:47AM |
1 |
OT: pinout for "standard" telephone headset required.? |
| 9:36AM |
2 |
LiveVoip DTMF Issues |
| 9:31AM |
1 |
AVM Fritz crash |
| 9:03AM |
0 |
DTMF tones during a call to OSS/dsp |
| 8:55AM |
2 |
Menu tree for voicemailmain application |
| 8:54AM |
1 |
PRI dchannel in use? (take 2) |
| 8:53AM |
1 |
Hitachi Cable WIP-5000 Wifi phone? |
| 8:45AM |
0 |
Best VPN server for * and woad warriors usin g windows? |
| 8:42AM |
3 |
Dialing Delay |
| 8:42AM |
0 |
budgetone - pattern matching for ringtones - firmware 1.0.5.18 |
| 8:41AM |
1 |
zaptel vanilla kernel |
| 8:19AM |
2 |
XEON or not |
| 8:14AM |
4 |
ISP connection to the PSTN using Asterisk |
| 8:13AM |
2 |
IVR Timing out |
| 7:59AM |
1 |
how to use mysql with asterisk |
| 7:57AM |
1 |
Cisco7905 keeps forwarding to voicemail |
| 7:47AM |
0 |
TDM400P Sync source |
| 7:46AM |
3 |
Asterisk with Grandstream ringback |
| 7:39AM |
0 |
forwarding sip |
| 7:26AM |
0 |
How to display number being dialed |
| 7:16AM |
3 |
Sipura Behind NAT howto |
| 7:13AM |
1 |
Voicemail folders |
| 7:05AM |
1 |
DTMF issues (handytone) |
| 6:27AM |
1 |
Threeway callin |
| 5:40AM |
2 |
Not answering PSTN until SIP answers |
| 5:14AM |
1 |
zaphfc no callerid incoming to SIP phone butvisible in logfile |
| 4:31AM |
2 |
asterisk starting problem |
| 4:20AM |
1 |
Mediatrix voip gateway 1124 and 1204 in UK setting |
| 4:04AM |
3 |
OT: Libnewt sourcecode? |
| 3:37AM |
3 |
Zapata in Australia |
| 3:31AM |
1 |
Short DTMF Tones and Asterisk |
| 3:18AM |
12 |
UPS for Asterisk |
| 3:12AM |
2 |
PSTN and Asterisk |
| 2:33AM |
3 |
Asterisk on sattelite link |
| 2:18AM |
0 |
how to display queue status and/or line status in asterisk |
| 1:57AM |
4 |
Auto callout - reminder - is it possible? |
| 1:49AM |
0 |
about call out : a strange question. |
| |
| Sunday January 23 2005 |
| Time | Replies | Subject |
|---|
| 11:58PM |
1 |
Looking for a prepaid calling card platform |
| 9:39PM |
0 |
Peculiar one way convesation fault with Asterisk. |
| 9:31PM |
2 |
sip - h323 translation stability & capacity limit |
| 9:28PM |
0 |
zaprtc from bristuff? not there? |
| 8:34PM |
3 |
Asterisk 1.0.5 |
| 7:16PM |
3 |
SIP USB Phone? |
| 6:46PM |
0 |
- New Security List - |
| 6:27PM |
0 |
Upgrade to the newest cvs now asterisk will notstart |
| 6:21PM |
4 |
VoIP software for MAC OS older than "X"? |
| 6:02PM |
0 |
Upgrade to the newest cvs now asterisk will not start |
| 5:57PM |
0 |
No music with "Blind" transfer on GS ATA + Sipura-841 |
| 5:50PM |
1 |
Data calls with Asterisk |
| 4:34PM |
2 |
Sip Notify and PHP AGI |
| 4:10PM |
5 |
Music On-Hold problem |
| 1:58PM |
1 |
VoIP Providers and Backbone Servers |
| 1:42PM |
6 |
Autio cut off at beginning of call |
| 1:19PM |
0 |
Anybody a patch for oss/alsa to not constantly hog the sound card? |
| 12:17PM |
4 |
Any experience with Sangoma cards? |
| 12:02PM |
4 |
Florz patch for zaphfc |
| 10:49AM |
0 |
grandstream sip phone calling Zap/1 on TDM20Brings and answers but not hear voice |
| 9:06AM |
0 |
simulating multiple lines using ADSI |
| 8:46AM |
0 |
How to debug core-file |
| 8:05AM |
0 |
Delay before dialing extension on Zap channel |
| 2:33AM |
3 |
Best VPN server for * and woad warriors using windows? |
| 1:19AM |
1 |
gsm/wav format not recognized in Background() application |
| 12:56AM |
2 |
can iaxcomm run on the same server as Asterisk? |
| |
| Saturday January 22 2005 |
| Time | Replies | Subject |
|---|
| 11:37PM |
1 |
zaprtc load issue (different that other postings) |
| 11:32PM |
4 |
chan_skinny and firmware upgrade |
| 10:31PM |
1 |
Some issues with X-Lite and codecs. |
| 8:43PM |
0 |
chan_capi patch: app_capiFax modifications |
| 6:53PM |
2 |
flashing zap using macro |
| 5:41PM |
1 |
grandstream sip phone calling Zap/1 on TDM20B rings and answers but not hear voice |
| 4:22PM |
1 |
Bellster - cool :-) |
| 4:07PM |
0 |
PortaOne's RADIUS client and Appradius |
| 3:00PM |
1 |
ASTCC: potential billing issue and "fix" |
| 2:49PM |
0 |
Anyone know where a good source of mailing l ist stats might be found? |
| 2:29PM |
0 |
how to configure Asterisk is outside and the SIP phone (Xlite) is inside behind NAT/PAT |
| 12:38PM |
0 |
asterisk not starting--sound module |
| 11:42AM |
1 |
te405P and german PMX |
| 10:48AM |
3 |
Asterisk Install Method |
| 10:31AM |
0 |
VoIP service setup help |
| 10:19AM |
2 |
Anyone know where a good source of mailing list stats might be found? |
| 10:14AM |
1 |
Dialogic D/4PCI |
| 8:31AM |
0 |
Asterisk/Sip crash "Failed to grab lock" |
| 8:27AM |
1 |
Re: Bellster - IAX-based interchange -- lets youcallanywhere for free |
| 6:09AM |
1 |
Need help configuring TDM10B / X100P Cards |
| 5:43AM |
0 |
Fwd: Re: chan_misdn 0.0.3-rc5 - new release ! Please test it. |
| 3:42AM |
0 |
Asterisk + TDM04b trouble |
| 2:46AM |
3 |
Cisco ATA186 and Asterisk dialplan |
| |
| Friday January 21 2005 |
| Time | Replies | Subject |
|---|
| 10:57PM |
0 |
Caller ID Problems after upgrading from 1.0.1 to 1.0.4 |
| 7:34PM |
1 |
7960 SIP image |
| 5:55PM |
0 |
incoming calls timing out. |
| 5:08PM |
3 |
IAX Inbound Sound Quality |
| 4:38PM |
5 |
SPA-2000 |
| 4:28PM |
0 |
Problem compiling zaptel-1.0.3 |
| 3:49PM |
3 |
IAXy's apparantly failing in the field |
| 2:38PM |
0 |
Rotate Logs |
| 2:31PM |
0 |
Incoming zap channels busy |
| 2:26PM |
1 |
Where is the * servers IP defined for sip phones? |
| 2:01PM |
0 |
WellTech 3804 Config anyone?? |
| 1:46PM |
1 |
Iaxphone - unreachable if qualify yes ? |
| 1:28PM |
1 |
Powell resigns |
| 1:04PM |
0 |
Rate Engine Examples |
| 12:53PM |
0 |
three way call using sip - SOLVED - |
| 12:31PM |
0 |
Multiple Host IP connections per peer |
| 12:24PM |
2 |
Outbound analog dialing with Internet Line Jack (fwd) |
| 11:51AM |
0 |
Manager API on gives the DIALSTATUS of the first picked up channel? |
| 11:36AM |
0 |
AstTapi - Crashes w/ Windows 2000 - UrgentHelpneeded - May need to hire a developer |
| 11:26AM |
3 |
IAXTEL is dead/dying? |
| 11:20AM |
4 |
three way call using sip |
| 10:54AM |
3 |
zaphfc no callerid incoming to SIP phone but visible in logfile |
| 10:42AM |
1 |
Asterisk+Oracle |
| 10:41AM |
0 |
Codec conversion sip peer <> Asterisk |
| 10:39AM |
1 |
chan_misdn 0.0.3-rc5 - new release ! Please testit. |
| 10:37AM |
0 |
About DeStar, a web frontend for Asterisk |
| 10:26AM |
2 |
Bandwidth, again, can someone check my math? |
| 9:48AM |
0 |
chan_misdn 0.0.3-rc5 - new release ! Please test it. |
| 9:43AM |
1 |
Webmin Module for Asterisk (and thirdlane) |
| 9:39AM |
0 |
IAX2 trunking, Voicepulse Connect, and Outbound Faxing |
| 9:35AM |
1 |
Ignoring callwaiting? |
| 9:25AM |
0 |
Voicemail.conf pin protection |
| 9:23AM |
2 |
SpanDSPpre10 and AsterisK1.0.4 issues |
| 9:22AM |
0 |
Cisco 7960 can't make/receive calls |
| 9:03AM |
0 |
Help DIALSTATUS gives ANSWER when line is BUSY? |
| 8:38AM |
2 |
Can anyone recoment T1/PRI provider in SouthOntario? |
| 8:13AM |
1 |
problem with TE-405P |
| 7:54AM |
5 |
Snom hint for ZAP channels? |
| 6:49AM |
1 |
Asterisk 1.0.4 and broadvoice patch |
| 6:10AM |
0 |
german dialtones for IAXy? |
| 4:29AM |
0 |
Mediatrix III FXO 4 Port |
| 4:16AM |
0 |
Grandstreams+Nat |
| 3:53AM |
1 |
Recording a meetme conference |
| 3:31AM |
1 |
Voicemail Synchronization |
| 3:30AM |
1 |
sip.conf configuration for internal calls |
| 3:21AM |
0 |
3Com SIP Phone - Forbidden |
| 3:19AM |
0 |
Dropping duplicate answer |
| 3:16AM |
1 |
Intermittent breakage with the ISDN4Linux modem driver |
| 2:45AM |
0 |
Caller id with isdn4linux |
| 1:54AM |
0 |
h323 client |
| 1:06AM |
4 |
Adit 600 as VoIP router (MGCP) and Asterisk |
| 12:51AM |
0 |
Stanaphone incoming calls problem. |
| |
| Thursday January 20 2005 |
| Time | Replies | Subject |
|---|
| 11:46PM |
1 |
Polycom IP 300/500 Conferencing Behavior |
| 11:02PM |
3 |
Zap randomly hanging up |
| 8:57PM |
2 |
Segmentation Fault after Digitnetwork X100P install |
| 8:40PM |
4 |
softswitch dilemma |
| 8:32PM |
4 |
Ring an incoming call in multiple extensions |
| 8:19PM |
9 |
OT: Headset for the Cisco 7960 |
| 7:58PM |
1 |
Hopping through iax servers |
| 6:35PM |
1 |
SNOM 190 and dtmf |
| 6:25PM |
2 |
controlling recording |
| 5:36PM |
1 |
Re: zaptel on 2.6.10 kernel - debian. |
| 4:42PM |
3 |
Asterisk 1.0.4 and more ... |
| 4:22PM |
2 |
iax encryption |
| 3:51PM |
1 |
Headset with X-Lite |
| 3:47PM |
0 |
Asterisk@Home and iax.cc / sixTel |
| 3:42PM |
1 |
H323 and ASTCC |
| 3:22PM |
5 |
Stumped on LD questions...... |
| 2:51PM |
7 |
PIX!!!!! |
| 2:17PM |
0 |
ASTCC config Problem |
| 1:48PM |
1 |
AstTapi - Crashes w/ Windows 2000 - Urgent Help needed - May need to hire a developer |
| 1:28PM |
0 |
VICIDIAL and meetme conference help |
| 1:07PM |
0 |
Sound quality poor everywhichway |
| 12:48PM |
0 |
Dialplan - intercoms |
| 12:24PM |
1 |
PRI info digits question |
| 12:15PM |
3 |
ringback |
| 11:57AM |
0 |
SIP debugs |
| 11:45AM |
1 |
Realtime Engine |
| 11:12AM |
0 |
is it possible to use Zaphfc (BRI) exactly like i4l? |
| 10:53AM |
7 |
VoIP-to-TDM processing on-card? |
| 10:43AM |
0 |
What's up with IAXTEL? |
| 10:15AM |
1 |
Newbie question - can't get Asterisk to pick up incoming call |
| 9:49AM |
0 |
BRI Fax out through PRI? |
| 9:45AM |
0 |
Meetme Limitations? |
| 9:32AM |
1 |
Weird Zaphfc - not dialling non-local numbers |
| 9:29AM |
2 |
Tips do update Asterisk and AMP |
| 8:41AM |
1 |
Using Zyxel Analog Telephone adapter with a GSM gateway |
| 7:36AM |
0 |
Dial plan problems with realtime extensions ... |
| 6:59AM |
2 |
RE: how to manage Digium TDM04B outgoing calls |
| 6:55AM |
5 |
SIP Stress Test |
| 6:42AM |
2 |
Chan_Capi initial deadlock |
| 6:30AM |
0 |
latest cvs will not compile |
| 6:25AM |
2 |
Some more hardware and E1 questions |
| 6:13AM |
1 |
Asterisk from flash with dynamic voicemail enable/disable? |
| 5:58AM |
0 |
Park/retrieval of calls |
| 4:47AM |
1 |
monitoring packet loss? |
| 4:23AM |
1 |
ilbc high bandwidth |
| 3:39AM |
0 |
change domain caller |
| 3:37AM |
0 |
ztdummy and meetme conference problem |
| 3:10AM |
0 |
Poor sound quality on ISDN BRI calls |
| 3:09AM |
0 |
(no subject) |
| 3:05AM |
0 |
regexten for realtime sip ? |
| 3:04AM |
2 |
hardware details |
| 2:36AM |
0 |
How to read ISDN messages - URGENT!!!! |
| 2:24AM |
1 |
FW: Asterisk 1.0.3 startup |
| 2:23AM |
2 |
API Call Bridge? |
| 2:22AM |
0 |
Asterisk 1.0.3 startup |
| 2:00AM |
0 |
Authentication Problem |
| |
| Wednesday January 19 2005 |
| Time | Replies | Subject |
|---|
| 8:46PM |
1 |
could someone please tell me how abstraction is provided in asterisk. |
| 7:12PM |
2 |
AGI Environment Dump Question w/ASTCC |
| 6:59PM |
0 |
AGI crash on 1.0.2 on Wait ... |
| 6:32PM |
1 |
Troubles with Broadvoice (register) |
| 6:00PM |
0 |
Why does bristuff generate PRI errors for a BRI only server |
| 5:17PM |
5 |
Call Screen Macro Not Exiting when call rejected |
| 5:15PM |
2 |
IAXTEL errors ! |
| 4:26PM |
0 |
Problems transferring calls - Part 2! |
| 4:18PM |
0 |
can callgroup be used to ring a group of phones? |
| 3:55PM |
4 |
RE: how to manage Digium TDM04B outgoing calls |
| 3:43PM |
1 |
ztdummy issues on new asterisk install |
| 3:19PM |
1 |
how to manage Digium TDM04B outgoing calls correctly |
| 3:04PM |
1 |
Calling Voicemail in an AGI script |
| 2:49PM |
1 |
G.729? Worth it? -- YES -- |
| 2:45PM |
7 |
E911 Testing ! |
| 2:30PM |
1 |
My dialplan just stopped working one day |
| 12:45PM |
0 |
Re: Asterisk-Users Digest, Vol 6, Issue 284 |
| 12:26PM |
0 |
Asterisk B2BUA |
| 12:02PM |
0 |
Asterisk vs Proxy SIP |
| 11:59AM |
0 |
IAX line gets 'Hungup' after period of silence |
| 11:45AM |
1 |
echocancellation in modem.conf |
| 11:36AM |
0 |
very big Echo, isdn -> isdn |
| 11:30AM |
1 |
Re: Asterisk bandwidth tuning? |
| 11:21AM |
2 |
Becoming a VOIP provider |
| 11:19AM |
0 |
no sound transmision |
| 11:00AM |
0 |
Cisco 7940 problems |
| 10:56AM |
1 |
Advanced Agents - Need a nice web interface |
| 10:08AM |
0 |
Re: Asterisk monitoring with Nagios and IAX (RoySigurd Karlsbakk) |
| 10:05AM |
0 |
Asterisk fax-modem |
| 9:33AM |
4 |
Accessing Voice mail |
| 9:02AM |
4 |
G.729? Worth it? |
| 8:57AM |
1 |
who changed the codec? |
| 8:46AM |
0 |
Extension Length |
| 8:41AM |
0 |
Play audio to channel |
| 8:32AM |
2 |
queue log analyser? |
| 8:24AM |
0 |
FAX detection in extentions.conf |
| 8:16AM |
0 |
MeetMe MusicOnHold Volume |
| 6:41AM |
3 |
:: Success Case => Motorola 62802-51 as FXO device :: |
| 5:57AM |
1 |
g729 problem |
| 5:33AM |
1 |
Resellers in Europe |
| 5:31AM |
0 |
PSTN Pabx and asterisk |
| 5:00AM |
0 |
h323 compilation problem |
| 4:23AM |
0 |
what does the "c" option in the zap phone number do |
| 3:25AM |
1 |
Re: Busy message on ISDN cards? (SOLVED) |
| 3:11AM |
0 |
iax.conf bindaddr parameter not working |
| 3:09AM |
1 |
How to change the packet size |
| 2:47AM |
1 |
Re: Asterisk monitoring with Nagios and IAX (Roy Sigurd Karlsbakk) |
| 2:39AM |
3 |
Fax and PRI |
| 1:49AM |
1 |
Asterisk not recognizing key beeps |
| 12:24AM |
1 |
Can IAXy be setup for PPPoE ??? |
| |
| Tuesday January 18 2005 |
| Time | Replies | Subject |
|---|
| 11:49PM |
0 |
sip-sip |
| 10:32PM |
9 |
# Transfers. |
| 10:15PM |
5 |
Open Source QoS . |
| 9:32PM |
0 |
X100P not working: no sound |
| 9:15PM |
0 |
Hardware Requirement & Setup |
| 8:20PM |
0 |
LogWatch emails in /var/spool/mail/root |
| 8:01PM |
0 |
C/C++ SIP Phone Development Lib or Stack |
| 7:36PM |
1 |
Wellgate 3804 Firmware |
| 7:03PM |
2 |
Polycom Call-Waiting |
| 7:01PM |
2 |
Router Recommendations Please |
| 6:45PM |
2 |
Reverse phone lookup interface with asterisk |
| 6:34PM |
0 |
spandsp & tdm400p - recommended hardware |
| 5:43PM |
1 |
Cisco 7940 Configuration |
| 5:05PM |
3 |
Cisco 7940G |
| 4:32PM |
2 |
Asterisk and h323 |
| 4:25PM |
2 |
Is an unregistered phone busy? |
| 3:56PM |
1 |
Grandstream BT102 |
| 3:48PM |
3 |
Newbie question: Can't start up asterisk |
| 3:28PM |
2 |
problems compiling asterisk-addons |
| 3:21PM |
0 |
asterisk and predictive dialers |
| 3:11PM |
1 |
something between an ATA and a channel bank for a small office? |
| 2:31PM |
1 |
Asterisk and IAX softphone (firefly) problem/question |
| 1:36PM |
0 |
TDM400P card & PCI problems |
| 1:27PM |
1 |
Re: Asterisk bandwidth tuning? |
| 1:19PM |
1 |
R2 - Stable Asterisk |
| 1:18PM |
0 |
CallManager 3.1 (2c) and Asterisk Integration |
| 1:09PM |
0 |
Issue using IAX2 as end-point (IAXComm) |
| 12:45PM |
0 |
RE: mgcp <-> h323 problem |
| 12:39PM |
1 |
Asterisk - libunicall - MFCr2 *** settings problems ??? *** |
| 12:10PM |
0 |
Database of event activity |
| 12:10PM |
2 |
What's the easiest way to call two people at same time and bridge them? |
| 12:01PM |
0 |
Urgent handler messages on * 1.0.3 |
| 11:56AM |
1 |
QoS tagging - can Asterisk do this, if not, what do you recommend? |
| 11:39AM |
0 |
Error after switching from 1.0.2 (FreeBSD) to 1.0.3 (Gentoo) |
| 11:36AM |
0 |
DTMF is being MUTED by asterisk to/from SIP channels from SIP or ZAP |
| 11:14AM |
1 |
Quick Question on Wildcard T100P |
| 11:06AM |
0 |
SIP URL ? |
| 10:45AM |
1 |
Delay after Dial Application is Called |
| 10:23AM |
4 |
Versatel PRA in Belgium/Netherlands |
| 10:01AM |
1 |
Re: * compatible with Pulse dialing phones ? |
| 9:54AM |
2 |
Broadvoice Patch Error {Scanned} |
| 9:39AM |
0 |
is it possible to use a sp2000 for intercom/paging? |
| 9:36AM |
1 |
External fax modem takeover of fxo? |
| 9:06AM |
0 |
Out of 5 Grandstream BudgeTone 101 THREE are |
| 8:39AM |
2 |
Outbound Dial via SIP |
| 8:31AM |
1 |
Problem with registering Windows Messanger with asterisk |
| 8:26AM |
1 |
Flat Rate Long Distance Providers |
| 8:23AM |
1 |
No compatible codecs |
| 7:21AM |
1 |
Problem with demo on asterisk |
| 7:06AM |
2 |
8 x 8 Analog System for Auth and Minutes Tracking |
| 6:53AM |
1 |
ISDN + chan_capi |
| 6:40AM |
0 |
No Busy signalled to caller |
| 6:30AM |
4 |
sipura 3000 mwi stutter problem |
| 6:28AM |
4 |
TE110P as E1 |
| 6:20AM |
0 |
AMP and Asterisk PSTN extension config |
| 6:11AM |
14 |
Attended call transfer |
| 6:07AM |
4 |
Asterisk monitoring with Nagios and IAX |
| 5:58AM |
3 |
Prefered server hardware |
| 5:13AM |
2 |
Realtime Voicemail ... |
| 4:51AM |
0 |
TDM400 - incomming call is answered but if i hang up asterisk never detects it |
| 4:28AM |
1 |
Outgoing SIP call from Asterisk problem |
| 4:21AM |
0 |
Will queueing only work after answering a ca ll? |
| 4:13AM |
0 |
Vmail |
| 4:02AM |
1 |
Will queueing only work after answering a call? |
| 3:57AM |
1 |
Multiple Alsa Devices |
| 3:20AM |
2 |
Outbound calls unpredictable |
| 2:41AM |
2 |
MFCR2 - LIBUNICALL - Asterisk Problems |
| 1:34AM |
9 |
Best Grandstream firmware to use? |
| 1:24AM |
5 |
fax over tdm400p |
| 1:00AM |
1 |
Number of Calls per Proxy on Cisco 7960G? |
| 12:47AM |
0 |
Dial Plan Agents (2 of 2) extensions.com |
| 12:46AM |
1 |
Dial Plan Agents (1 of 2) agent-dialplan.conf |
| 12:46AM |
3 |
Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore) |
| 12:15AM |
0 |
Canadian Content: Telus and Shaw... |
| 12:06AM |
1 |
Auto Protocol (depending upon registration.... |
| |
| Monday January 17 2005 |
| Time | Replies | Subject |
|---|
| 11:58PM |
2 |
Is anybody using an IAXy? |
| 11:20PM |
3 |
Planning "hotel" phone system - Need input |
| 11:02PM |
2 |
Sound quality - commercial vs. Asterisk |
| 10:33PM |
0 |
VoIP Routes and Terminations |
| 9:03PM |
3 |
On Hold music |
| 7:32PM |
2 |
internal dial tone on password from outside |
| 6:48PM |
0 |
TDM13B - FXO ports not seeing incoming calls |
| 6:42PM |
2 |
SIP URL for incoming |
| 6:22PM |
1 |
here's my IAX callthrough app and some questions about problems I have. |
| 6:12PM |
0 |
Transferring calls on Asterisk with X-Lite |
| 5:51PM |
1 |
spandsp and app_txfax |
| 5:32PM |
3 |
callers who don't press any keys |
| 5:12PM |
1 |
transfers with zap channel |
| 4:44PM |
4 |
Wait(n) -v- Background(silence/n) ? |
| 4:43PM |
2 |
iaxtel - -- Format for call is ADPCM |
| 4:23PM |
1 |
Looking for Asterisk termination in Russia |
| 3:40PM |
1 |
Re: Any interest in a Canadian Asterisk |
| 3:27PM |
1 |
China direct route |
| 3:14PM |
0 |
Multiple Line Caller Id With Polycom IP500 |
| 3:11PM |
0 |
VOIP CONNECTION, NO AUDIO AT THE OTHER END, NEWBIE |
| 3:09PM |
0 |
Queue and Normal Transfer. |
| 2:41PM |
1 |
Echo on SIP -- not on analog. |
| 2:38PM |
1 |
X-Ten lite troubles. |
| 2:25PM |
1 |
Media Path Optimization & NAT |
| 2:22PM |
0 |
How to call an extension number from ohphone to astersisk |
| 2:05PM |
3 |
FW: Radius on * |
| 1:49PM |
2 |
Offtopic: improving softphone latency on Linux? |
| 1:45PM |
1 |
ZAP/PRI Error: channel reported in use |
| 1:44PM |
1 |
Directory() Command |
| 1:38PM |
1 |
IAX2 doesn't respect bindaddr? |
| 1:34PM |
1 |
Attempting native bridge |
| 12:46PM |
1 |
spandsp recieve problem |
| 12:23PM |
0 |
Jamaica - My apologies for the second post. |
| 12:13PM |
2 |
RE: Issue compiling zaptel on FC 3 kernel 2. 6.10-1.737 |
| 11:57AM |
0 |
Jamaica |
| 11:35AM |
3 |
Asterisk C source code documentation |
| 11:32AM |
2 |
Jamaica DID |
| 11:29AM |
1 |
RE: Issue compiling zaptel on FC 3 kernel 2.6.10-1.737 |
| 10:46AM |
1 |
iaxtel - best codec |
| 10:28AM |
1 |
is asterisk a good solution? |
| 10:00AM |
0 |
SIP/H323 modules for netfilter |
| 9:57AM |
0 |
How to implement an audio delay? |
| 9:45AM |
4 |
DIDs anywhere but here? |
| 9:13AM |
5 |
simple over view of the process |
| 9:11AM |
3 |
Is it possible to ID payphone calls? |
| 9:01AM |
0 |
DIAX 0.9.9g more features and higher stabili ty |
| 8:15AM |
0 |
RE: [Asterisk-biz] Guatemala DID's? |
| 8:11AM |
4 |
SIP IOS for cisco 7902G IP Phone |
| 7:52AM |
1 |
ASTCC single stage + no access number + auth usingsip username and password |
| 7:41AM |
1 |
ntp Server and Zultys 4X4 |
| 7:20AM |
1 |
Communication Between Phones... I can't test :( |
| 7:15AM |
4 |
REALTIME and VARIABLES |
| 7:12AM |
1 |
TDM400 answers the line all the time! |
| 6:33AM |
2 |
CAS voice signalling? |
| 6:13AM |
0 |
chan_capi outgoing msn |
| 5:55AM |
0 |
voicemail sound distorted - chan_capi, diva, cvs-head |
| 5:53AM |
0 |
Can I start recording channel in the middle ofconversation ? |
| 5:45AM |
1 |
Can I start recording channel in the middle of conversation ? |
| 5:33AM |
0 |
Manager Event Logging |
| 4:56AM |
2 |
error compiling |
| 4:30AM |
1 |
Using a variable for EXTEN |
| 3:26AM |
2 |
Does Asterisk do that? |
| 3:22AM |
0 |
Can I get info about email addresses from voicemail.conf in dialplan or variables ? |
| 3:10AM |
0 |
ASTCC single stage + no access number + auth using sip username and password |
| 2:57AM |
0 |
AGI / Sockets |
| 2:53AM |
1 |
Euro ISDN and Caller ID (Sweden) |
| 1:14AM |
2 |
Adding SIP clients using AGI ? |
| 12:44AM |
1 |
quadBRI asterisk error message message: "not able to open Zap channel" |
| 12:29AM |
0 |
voicemail attach not in 1.0.2 ? |
| |
| Sunday January 16 2005 |
| Time | Replies | Subject |
|---|
| 11:47PM |
10 |
Any interest in a Canadian Asterisk mailing list? |
| 11:29PM |
6 |
pattern matching problem |
| 11:25PM |
1 |
Asterisk over External Motorola BitSurfR Pro ISDN Modem |
| 11:10PM |
0 |
Registering with IAX provider |
| 9:18PM |
0 |
Looking for a VoIP provider for my Asterisk box. {Scanned} |
| 8:43PM |
1 |
New Sipura-841 phone.Mike volume problem. |
| 8:37PM |
2 |
FWD<->NAT<->* |
| 6:45PM |
1 |
VOIP - INBOUND Call - best setup |
| 4:59PM |
1 |
Meetme conf and Shoutcast |
| 4:37PM |
0 |
X100P with no sound! |
| 4:25PM |
2 |
IAX.conf error |
| 4:10PM |
1 |
IAX1 vs. IAX2 |
| 2:23PM |
1 |
Guatemala DID's? |
| 2:12PM |
1 |
chan_sccp and bristuff 1.0.3 weirdness |
| 1:51PM |
1 |
Type of Number |
| 1:22PM |
1 |
Inbound Callerid for SIP Phones |
| 12:29PM |
0 |
MVP110 and * |
| 12:25PM |
1 |
H323 Softphone for iPAQ |
| 11:46AM |
0 |
The BEST? analog phones for * |
| 11:09AM |
1 |
VoIP Newbie |
| 10:25AM |
0 |
Re: asterisk-users list and html posts |
| 9:19AM |
2 |
Looking for help with a Polycom Soundpoint IP 600 |
| 9:06AM |
0 |
Re: Asterisk-Users Digest, Vol 6, Issue 227 |
| 7:59AM |
0 |
* reports the incoming caller id but not the BT100 |
| 7:37AM |
2 |
TDM400 lost after reboot |
| 5:40AM |
0 |
TDD support in Asterisk? |
| 4:32AM |
0 |
sound-recorder crash when I start Asterisk |
| 4:26AM |
0 |
Extension.conf, sip.conf and contexts. |
| 1:59AM |
6 |
announcing caller id? |
| |
| Saturday January 15 2005 |
| Time | Replies | Subject |
|---|
| 11:16PM |
3 |
TDM400P NO BATTERY & Poopy??? |
| 9:03PM |
2 |
failed to compile zaptel on redhat |
| 8:51PM |
1 |
TDM400p FXS not sending caller id info? |
| 6:48PM |
4 |
How to demo wired phone set on a wireless network |
| 6:24PM |
1 |
SayDigits -- ToneDigits?? |
| 5:28PM |
2 |
IAX2 one side loses audio |
| 4:57PM |
0 |
X100P no sound problem |
| 4:18PM |
6 |
NuFone help |
| 4:14PM |
0 |
Polycom IP600 - Bridge stops because we're zombie or need a soft hangup |
| 3:37PM |
0 |
oh323 driver - [user] type=user |
| 3:18PM |
1 |
ATA with IAX protocol |
| 2:40PM |
0 |
Is it the 15th or the 16th :) |
| 2:39PM |
2 |
No more loading asterisk... |
| 1:39PM |
1 |
CAC Channel Bank I - FXS |
| 1:02PM |
0 |
Sip registration period |
| 12:58PM |
2 |
IAX2 Channels & Bandwidth |
| 11:20AM |
0 |
Add h323 support to Asterisk |
| 11:05AM |
0 |
oh323 compile error |
| 10:54AM |
1 |
can't install 1.0.3 |
| 10:53AM |
1 |
failed to compile zaptel on redhat (kernel 2.4.20-31.9) |
| 10:44AM |
0 |
Asterisk to CCM3.3.4 via H32 |
| 10:18AM |
0 |
ADSI unlock codes |
| 9:26AM |
0 |
Newbie - Asterisk Tramsfer Problem? |
| 8:39AM |
0 |
configuring ser for * |
| 7:52AM |
1 |
Voicemail after one ring? |
| 6:20AM |
0 |
call deflect with QuadBRI how to |
| 6:12AM |
3 |
Return of experience : Asterisk more stable with 2.6 or 2.4 |
| 5:51AM |
0 |
Problems using chan_capi over Fritz!Bluetooth |
| 5:31AM |
1 |
switches |
| 5:23AM |
0 |
IPCB.net sip.conf |
| 4:51AM |
1 |
Packet8 DTA310 SIP Image |
| 4:01AM |
1 |
Re: Budgetone and MWI |
| 3:06AM |
1 |
spa 2000 phones do not ring |
| 1:49AM |
1 |
voice output |
| 1:36AM |
2 |
No sound with X100P (clone) |
| 1:34AM |
0 |
Anyone use SunRocket with Asterisk? |
| 12:07AM |
3 |
DIAX |
| |
| Friday January 14 2005 |
| Time | Replies | Subject |
|---|
| 11:47PM |
2 |
Echo Training - how long |
| 11:27PM |
1 |
Asterisk@Home Install Problems |
| 11:25PM |
1 |
voice quality with asterisk |
| 11:18PM |
1 |
voice quality in asterisk |
| 9:03PM |
5 |
Remote Voicemail Retrieval... |
| 8:19PM |
1 |
Asterisk and Voice Pulse Open Access |
| 7:30PM |
1 |
Proxy-auth |
| 6:27PM |
1 |
iaxComm 0.99pre11 binaries posted to Sourceforge |
| 5:08PM |
3 |
Packet8 DTA310 and Asterisk |
| 3:59PM |
1 |
Routing incoming calls to various extensions. |
| 3:50PM |
1 |
ULaw not negotiating |
| 3:38PM |
1 |
Having trouble with T405P and PPP: ZT_SPANCONFIG failed |
| 3:16PM |
1 |
DIAX PC to Phone |
| 2:44PM |
1 |
Asterisk for voicemail -> C2611XM, 7940 & 7960 phones |
| 1:22PM |
0 |
Strange CRCX |
| 1:12PM |
1 |
SIP Registration problem, 403 forbidden |
| 12:49PM |
0 |
app_conference compile? |
| 12:28PM |
0 |
IAX on multiple ports |
| 12:13PM |
1 |
context wide variable scope |
| 12:06PM |
0 |
OT: zaptel kernel mod |
| 11:53AM |
1 |
PrePaid Applications |
| 11:53AM |
2 |
PC to Phone |
| 11:37AM |
5 |
Softphone for Linux recommendation |
| 11:30AM |
1 |
T100P with NEC C2400 IPX switch |
| 11:03AM |
0 |
Re: Grandstream Bugetone 101 & mw |
| 10:57AM |
1 |
Re: Budgetone and MWI |
| 10:11AM |
1 |
gotoiftime - different hours |
| 9:56AM |
1 |
Polycom SoundPoint IP by Shoreline |
| 9:55AM |
0 |
Zultys Phone feature |
| 9:48AM |
0 |
Newer CVS-Stable Asterisk not recognizing G711 ULaw from certain providers |
| 9:38AM |
0 |
IAX2 bridging = one way audio |
| 8:47AM |
2 |
Realtime / sip.conf |
| 8:14AM |
2 |
Spandsp....And garble incoming fax |
| 8:03AM |
0 |
Re: SOS |
| 7:53AM |
2 |
Passing PIN Numbers |
| 7:24AM |
1 |
handle_request registration failed?, Polycom IP500 |
| 7:22AM |
0 |
SOS !!! |
| 5:46AM |
1 |
iconecthere and * |
| 5:23AM |
0 |
problem in calling |
| 4:45AM |
1 |
[Slightly OT] SIP/T.38 capable system, anyone? |
| 4:30AM |
1 |
Hardware issues |
| 3:25AM |
1 |
incomplete address |
| 3:17AM |
0 |
Can Asterisk generate a 404 message back to a UA? |
| 3:15AM |
1 |
Suse 9.2 / Latest CVS |
| 3:10AM |
0 |
Environment variables |
| 2:16AM |
1 |
Grouping lines pending on Called ID |
| 2:06AM |
0 |
troubles with getting odbc to load data |
| 2:03AM |
0 |
caller's identity |
| 1:06AM |
0 |
Could I "SET AUTOHANGUP()" count down after the channel state is UP |
| 12:00AM |
1 |
Limit outgoing trunk calls |
| |
| Thursday January 13 2005 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
Polycom Shared Call Appearance |
| 10:15PM |
1 |
sporadic beeps spa3k-* |
| 9:16PM |
2 |
I Don't Want Asterisk in the Media Path |
| 9:12PM |
0 |
Asterisk@Home systems |
| 9:09PM |
2 |
Updated kphone 4.0.5, asterisk v1.0.3 |
| 8:14PM |
6 |
Voice Mail Notification |
| 7:24PM |
0 |
Iaxtel directory |
| 7:10PM |
2 |
Firefly repeats registering to * server |
| 6:05PM |
1 |
REGISTER Problems With Realtime |
| 4:39PM |
0 |
Grandstream Bugetone 101 / documentation |
| 4:37PM |
0 |
Oh323 compilation errors |
| 4:35PM |
0 |
Hook-Flash on Voicetronix |
| 4:22PM |
0 |
voicemail function |
| 3:18PM |
9 |
DIAX 0.9.9g more features and higher stability |
| 2:24PM |
7 |
long delays in list posts? |
| 2:22PM |
1 |
MWI on Zap analog phone not lighting |
| 1:37PM |
0 |
Customer Service Coaching |
| 1:31PM |
1 |
ATA186: SIP/2.0 503 Service Unavailable |
| 1:24PM |
5 |
PRI concentrator |
| 1:18PM |
1 |
problems with astcc |
| 1:17PM |
3 |
High delay with diax099f + Asterisk |
| 1:03PM |
0 |
Looking for a wireless phone... wifiortradit ionalwireless ? |
| 1:00PM |
1 |
Security audit scripts |
| 12:24PM |
0 |
Looking for a wireless phone... wifiortraditionalwireless ? |
| 12:20PM |
2 |
Problem patching asterisk CVS with SpanDSP |
| 12:11PM |
0 |
Re: Budgetone 10x & mwi |
| 12:10PM |
3 |
TDM04B vs Dell revisited |
| 11:49AM |
2 |
1xT1 PCI card for * |
| 11:46AM |
1 |
SIP registration error, lost packets with asterisk |
| 11:38AM |
1 |
Re: Looking for a wireless phone... |
| 11:35AM |
1 |
Build PWLIB |
| 11:25AM |
1 |
Enabling/disabling zaptel echo-can from dialplan. |
| 11:12AM |
0 |
Want to install Oh323 and LOST |
| 11:11AM |
2 |
asterisk won't release line |
| 11:01AM |
2 |
SMS Gateway |
| 10:56AM |
0 |
PRI dchannel in use? |
| 10:54AM |
1 |
Howto DTMF pass-through on a channel |
| 10:48AM |
1 |
Not In Local Context |
| 10:33AM |
1 |
Re: R2/MFC Mexico FREE calls to test chan_unicall (Miguel Cavazos) |
| 10:27AM |
0 |
Asterisk doesn't detect when the caller hangs up |
| 10:19AM |
1 |
Status of latest round of Allison recordings |
| 10:09AM |
3 |
error 488 |
| 9:56AM |
4 |
Manager API !!!!!!!!! |
| 9:54AM |
4 |
Cisco 79XX phones not talking to asterisk |
| 9:50AM |
3 |
SER vs Asterisk for SIP |
| 9:24AM |
0 |
Xfering a call |
| 9:15AM |
2 |
How to present a dialtone to a dial-in user? |
| 9:12AM |
1 |
Teleconferencing? |
| 9:09AM |
4 |
Asterisk on a notebook? |
| 9:03AM |
1 |
Queue Log Parser |
| 8:31AM |
7 |
How to set asterisk NOT to answer incoming lines? |
| 7:45AM |
2 |
about AGI command parsing |
| 7:41AM |
2 |
Looking for a wireless phone... wifi ortraditional wireless ? |
| 7:30AM |
2 |
Agentcallbackogin without any user input after extension is dialed. |
| 6:45AM |
0 |
current CVS version |
| 6:30AM |
1 |
SIPGetHeader |
| 6:29AM |
1 |
About HDLC in ISDN |
| 5:31AM |
1 |
Hunt group with Accept/Reject Option |
| 5:19AM |
0 |
oh323 compile problem still |
| 4:42AM |
1 |
ASTCC dimensioning |
| 4:32AM |
1 |
MeetMe does not compile with Asterisk |
| 4:11AM |
1 |
Grandstream bt-100 loosing it! |
| 4:09AM |
1 |
SCCP questions |
| 4:03AM |
1 |
asterisk realtime msql |
| 2:15AM |
2 |
pseudo-realtime?? |
| 1:22AM |
1 |
Registration of SIP |
| 1:09AM |
0 |
Replacing Cisco3620 with Asterisk |
| |
| Wednesday January 12 2005 |
| Time | Replies | Subject |
|---|
| 11:40PM |
6 |
snom220 |
| 11:13PM |
5 |
Grandstream Bugetone 101 & mwi |
| 9:23PM |
0 |
moh mp3 streaming problem |
| 8:42PM |
0 |
pass through mode |
| 8:23PM |
0 |
IAX peering between two Asterisk servers, how? |
| 7:09PM |
4 |
Is this a $50 wifi or wireless USB VOIP phone ? |
| 6:48PM |
1 |
no playback audio |
| 6:34PM |
1 |
SNOM 190 Configuration with Asterisk |
| 6:08PM |
0 |
Volume in line for music-on-hold |
| 6:06PM |
0 |
BT keeps open sip channels |
| 5:48PM |
3 |
Bristuff 0.20RC3 loses connectivity after short line interruption? |
| 5:32PM |
2 |
Trouble building appradius |
| 4:40PM |
2 |
New SIP Phone Config |
| 4:29PM |
0 |
IAX2 dropped calls: need debug suggestions |
| 4:20PM |
2 |
Setting channel display in SIP |
| 4:17PM |
0 |
Re: EuroISDN BRI 2 or 4 wires? (Remco Barende) |
| 3:22PM |
12 |
R2/MFC Mexico FREE calls to test chan_unicall |
| 3:19PM |
0 |
Queue and penalties |
| 3:09PM |
0 |
getting * to start on suse 9.1 |
| 3:06PM |
1 |
EuroISDN BRI 2 or 4 wires? |
| 2:44PM |
2 |
Cant receive calls after network goes down and up |
| 2:39PM |
0 |
Come join the Asterisk Bookclub |
| 2:14PM |
0 |
SIP Authenication (Simple, Digest, ACL) |
| 2:06PM |
1 |
linphone -> NAT -> * -> NAT -> firefly woes. |
| 2:05PM |
5 |
Using asterisk to convert H.323 to SIP? |
| 1:54PM |
0 |
calling an extension after a voicemail is left |
| 1:50PM |
1 |
Asterisk server stopped - "0-order allocation failed " errors in the log |
| 1:49PM |
0 |
Asterisk variables - size limitation? |
| 1:31PM |
0 |
wctdm and alaw audio quality problem |
| 1:10PM |
0 |
Setting "User Info" in extensions.conf? (ZyXEL P2000W) |
| 12:29PM |
0 |
Asterisk + SER Questions |
| 12:05PM |
2 |
Where to buy a quadBRI? |
| 11:40AM |
0 |
OT: Asterisk hits Slashdot again |
| 11:11AM |
0 |
generating CPC for non-CPC analog FXO lines - suggestion/discussion requested for use with TDM400P X100P |
| 11:02AM |
2 |
Ports to open behind a NAT |
| 10:53AM |
3 |
Polycom IP 500 Dial Issues |
| 10:43AM |
1 |
Asterisk version naming convention!! |
| 10:36AM |
2 |
Call Manager or Asterisk |
| 10:34AM |
0 |
FW: asterisk - oh323 driver |
| 10:02AM |
0 |
So many Asterisk Patches - Which do I choose anduse? |
| 9:56AM |
1 |
HW ? on Getting a new Asterisk Box |
| 9:39AM |
1 |
spandsp on FC3 |
| 9:29AM |
2 |
T1 Timing Slips |
| 8:55AM |
0 |
Doc Asterisk |
| 8:21AM |
2 |
So many Asterisk Patches - Which do I choose and use? |
| 7:44AM |
2 |
Unofficial Broadvoice-users query/offer and DID routing question |
| 7:38AM |
0 |
chan_misdn - new release ! Please test it. |
| 7:28AM |
0 |
Problem solved on Xonox Asterisk distribution |
| 7:17AM |
1 |
PRI RLT support |
| 7:10AM |
0 |
astweb cdr's mysql.sock problem |
| 7:07AM |
7 |
Operator Panels? |
| 6:58AM |
3 |
What is the best and easiest flavor to be usedwith Asterisk. |
| 6:28AM |
4 |
chan_capi-0.3.5 error 127 |
| 6:10AM |
2 |
Re: [Asterisk-biz] SS7 and Asterisk solution |
| 5:51AM |
1 |
H323 on Asterisk@Home |
| 5:43AM |
0 |
connect asterisk to lingo without ata |
| 5:40AM |
6 |
Re: [Asterisk-biz] SS7 and Asterisk solution |
| 5:25AM |
2 |
Stale mp123 processes?? |
| 5:10AM |
1 |
What's the easiest way to get * to call PSTN? |
| 4:22AM |
1 |
Can I use spandsp with Asterisk on Fritz with Capi ? |
| 4:17AM |
0 |
Terminating VOIP calls on EuroISDN PRI interface ? |
| 4:16AM |
0 |
ChanIsAvail + Zap and SIP channels |
| 4:01AM |
1 |
Problem |
| 3:13AM |
1 |
How to configure three ISDN line |
| 2:16AM |
0 |
Attended transfer problem with Atxfer |
| |
| Tuesday January 11 2005 |
| Time | Replies | Subject |
|---|
| 10:37PM |
0 |
Asterisk User Group in Winnipeg, CA |
| 9:55PM |
1 |
Fast Start , Slow Start , or just Codec |
| 8:58PM |
8 |
What is the best and easiest flavor to be used with Asterisk. |
| 7:56PM |
3 |
AMP Anyone? |
| 6:44PM |
28 |
SS7 and Asterisk solution |
| 5:26PM |
0 |
PA-168(S) - Netweb -301 Phone |
| 5:01PM |
1 |
rxfax troubles.. |
| 4:34PM |
2 |
PA-168(S) - Netweb IPweb-301 Phone |
| 4:30PM |
6 |
test-ignore |
| 4:23PM |
0 |
Blank Voice Mail messages |
| 4:19PM |
3 |
No sound for music on hold |
| 4:08PM |
0 |
How to enable debug |
| 4:03PM |
0 |
"Telco power supply" with digium card TE410P? |
| 3:57PM |
4 |
Changes to manager outputs - A discussion |
| 3:51PM |
1 |
Dial Out Errors |
| 3:45PM |
1 |
"o" extension broken? |
| 2:52PM |
1 |
Channel IAX2 Socket Read Error |
| 2:13PM |
5 |
not sharing IRQ's |
| 2:06PM |
0 |
Agent autologoff=15 |
| 1:38PM |
0 |
Planet VIP-101T or VIP-150T |
| 1:31PM |
1 |
Direct SIP calls to * |
| 1:22PM |
1 |
BroadVoice outgoing works - now tackle caller ID |
| 1:15PM |
1 |
(UN)structured E1 |
| 1:05PM |
1 |
Dlink DPH-80 DONT work with asterisk |
| 12:55PM |
1 |
ACD Bug with AddQueueMember Stable |
| 12:51PM |
0 |
Not hanging up. {Scanned} |
| 12:47PM |
3 |
iax.conf qualify=yes not working? |
| 12:23PM |
1 |
PRI Errors (HDLC Abort (6) on Primary D-channel) |
| 12:06PM |
1 |
How to prevent a call from going to voicemail when one phone is offline? |
| 12:01PM |
1 |
Tool Recommendations for measuring UDP throughput / loss / jitter |
| 11:38AM |
2 |
TDM box Hardware |
| 11:01AM |
0 |
operator says that dial 1 or 0 |
| 10:33AM |
2 |
SIP, * and clients behind NAT |
| 10:31AM |
0 |
Sounds cut out problem - HFC-S card, zaphfc, Xlite |
| 10:24AM |
1 |
internal caller id on analog phones connected tozap |
| 10:04AM |
1 |
ACD Queues & Agent Status |
| 9:36AM |
0 |
Re: Asterisk-Users Digest, Vol 6, Issue 144 |
| 9:25AM |
0 |
howto dump binary data on zap channel? |
| 9:12AM |
0 |
RE: Asterisk-Users Digest, Vol 6, Issue 142 |
| 8:34AM |
0 |
Newbie question: call routing |
| 8:25AM |
6 |
Installing * on fedora 3 |
| 7:44AM |
2 |
ASTCC - error on call end |
| 7:30AM |
1 |
internal caller id on analog phones connected to zap |
| 7:26AM |
3 |
sip to h.323 |
| 6:01AM |
0 |
Cisco ATA 186 for PSTN dialing |
| 4:43AM |
1 |
How to mark a user for a conference |
| 4:07AM |
2 |
Realtime and include |
| 3:59AM |
1 |
Analogue RAS Server |
| 3:46AM |
1 |
asterisk one number service |
| 3:40AM |
0 |
test source for current xorcom rapid |
| 3:35AM |
3 |
requiring logon for SIP users |
| 3:10AM |
5 |
asterisk-oh323 and outgoing call |
| 2:17AM |
0 |
AGI Application Hangup when using AGI->getdata |
| 1:29AM |
0 |
What is acceptablenetworklatencyforvoipconnection? |
| 1:03AM |
0 |
Asterisk Segmentation Fault - layer3.c/mpg123 |
| |
| Monday January 10 2005 |
| Time | Replies | Subject |
|---|
| 11:29PM |
1 |
dialing into * then forwarded out gets choppy audio |
| 11:07PM |
1 |
TE110P with Telstra E1 PRI in Australia and New Zealand |
| 11:05PM |
0 |
test {Scanned} |
| 10:36PM |
1 |
echo cancelation on Digium T1 cards |
| 10:33PM |
0 |
TE-405P freezing, anyone else? |
| 10:31PM |
0 |
asterisk router problem |
| 9:10PM |
0 |
Russian characters showing up on safe_asterisk console in RedHat 9 and Fedora Core 2 |
| 8:47PM |
2 |
Route incoming call on 4 X100P to different Ext. {Scanned} |
| 8:01PM |
0 |
Asterisk not answering calls since oh323 upgrade |
| 7:41PM |
0 |
CallerID presentation |
| 6:23PM |
4 |
Sip to IAX ok, ZAP to IAX FAILS |
| 6:20PM |
0 |
Any movement on IAX being submitted to a standards body? |
| 6:08PM |
0 |
zulty's ZIP 2 IP phone |
| 6:08PM |
2 |
Weir long distance behaviour... |
| 5:50PM |
4 |
Generic modem question |
| 5:40PM |
0 |
SOYO G668 |
| 5:03PM |
0 |
Mental Blank: HELP: I cant get any callerid on capi incoming?? WHY |
| 4:16PM |
1 |
Asterisk calls back after phone call |
| 3:30PM |
0 |
64 Bit Support? |
| 3:15PM |
1 |
/usr/bin/ld error on make asterisk with Fedora Core 3 |
| 3:10PM |
1 |
Call Waiting + Call Transfer Problem |
| 2:12PM |
2 |
Some questions (maybe Nikotel related) |
| 2:11PM |
1 |
"make clean" DO IT! |
| 1:59PM |
1 |
SIP Reorder tones |
| 1:58PM |
0 |
dead line (no LED) on a TDM400B? |
| 1:48PM |
2 |
Power Failure, Line Switch, Relay device |
| 1:46PM |
2 |
ACD Queue question. |
| 1:27PM |
1 |
Is this a firefly problem? (*78/*79 doesn't work) |
| 1:21PM |
1 |
IAX2 keep alive? |
| 1:09PM |
3 |
Multiple gateways for same dial pattern |
| 12:52PM |
0 |
Connecting Asterisk to a Toshiba Strata syst em |
| 12:17PM |
0 |
sip channel between 2 asterisk box |
| 12:16PM |
2 |
X100P in a soekris 4801 |
| 11:54AM |
2 |
Festival Woes |
| 11:38AM |
7 |
Help! - Unintelligible prompts and music |
| 11:29AM |
2 |
Ring Voltage Supplied by Wildcard TDM400P REV E/F & AUTO FXS/DPO |
| 11:17AM |
1 |
Broadvoice call quality? |
| 11:17AM |
1 |
Static/Breaking up after I upgradedAsteriskaswell as a crash - Can't trace bug |
| 11:08AM |
0 |
ChanSpy Usage |
| 11:03AM |
1 |
IAX2 provider in Montreal, Canada |
| 10:37AM |
0 |
IAX Local numbers in Wyoming |
| 10:34AM |
0 |
Extensions config help please.. |
| 10:17AM |
1 |
Agent Status on FOP |
| 10:16AM |
2 |
R2 for Mexico? |
| 10:12AM |
1 |
Static/Breaking up after I upgraded Asterisk aswell as a crash - Can't trace bug |
| 9:57AM |
0 |
Static/Breaking up after I upgraded Asterisk as well as a crash - Can't trace bug |
| 9:11AM |
5 |
Any Notices from voiceconduit? |
| 9:00AM |
2 |
Vmail.cgi - "Hrm, can't seem to open /var/spool/asterisk/voicemail .... |
| 8:56AM |
2 |
Connecting a Home based worker with An Iaxy |
| 8:50AM |
0 |
Problems calling between two local SIP extensions |
| 8:43AM |
1 |
Ramifications of Multiple Sip Reloads Within Minutes? |
| 8:37AM |
0 |
[Fwd: Re: Asterisk-Users] very loud scratchy noise!] |
| 8:36AM |
1 |
Execute dialplan command at startup |
| 8:20AM |
1 |
Digi Datafire Micro V ISDN Card |
| 8:19AM |
2 |
Asterisk Setup Documentation |
| 8:12AM |
0 |
conference question... |
| 8:12AM |
0 |
open g723+limiting the out bound calls |
| 8:10AM |
2 |
Asterisk UK Community |
| 8:03AM |
2 |
Connecting Asterisk to a Toshiba Strata system |
| 7:45AM |
3 |
Request to schedule in the past?!?! |
| 7:41AM |
4 |
Asterisk to PSTN |
| 7:26AM |
1 |
Re: Asterisk-Users] very loud scratchy noise! |
| 6:57AM |
6 |
UK * group |
| 6:49AM |
1 |
Re: Toronto |
| 6:38AM |
3 |
FXO PCI Master abort |
| 6:35AM |
0 |
AGI EXEC trouble |
| 6:21AM |
6 |
Zaptel problems |
| 6:17AM |
10 |
Unicall errors |
| 5:57AM |
2 |
very loud scratchy noise! |
| 4:58AM |
0 |
Agents question |
| 4:36AM |
0 |
error? |
| 3:14AM |
3 |
OT: SIP Aware Firewall with Asterisk |
| 2:03AM |
4 |
audio delay ISDN |
| 12:57AM |
1 |
SetGroup |
| 12:46AM |
1 |
I need your feedback related to the DIAX 0.9.9f stability |
| |
| Sunday January 9 2005 |
| Time | Replies | Subject |
|---|
| 11:23PM |
2 |
TE110P error |
| 10:42PM |
0 |
Asterisk as H323 client? |
| 9:16PM |
5 |
Help in E1-T1 encoding |
| 8:43PM |
5 |
telemarketing application |
| 5:54PM |
2 |
What is acceptable network latencyforvoipconnection? |
| 5:20PM |
0 |
RE: Inbound calls getting disconnected when I answer the phone, using 'SIP'. "FIXED" |
| 5:03PM |
1 |
Can zaphfc (bristuff) do caller id? |
| 4:54PM |
0 |
Quicknet Internet Phonecard |
| 4:30PM |
5 |
Little confused about Caller ID |
| 4:15PM |
1 |
Making a call using MGCP |
| 4:07PM |
2 |
E&M trunk card? |
| 3:57PM |
0 |
call from PSTN, not hearing SIP: 180/RINGING( was call from DID,not hearing RINGTONEs ) |
| 3:47PM |
2 |
What is acceptable network latency forvoipconnection? |
| 3:43PM |
0 |
Caller ID in Australia |
| 2:36PM |
0 |
TDM4000 FXS and UK Caller ID |
| 1:55PM |
0 |
Asterisk SIP channel (PSTN Calls) |
| 1:51PM |
4 |
Asterisk Demo |
| 1:47PM |
0 |
Incoming no.s being dropped. |
| 1:42PM |
2 |
ASTCC Trunk and Routes Configuration |
| 1:39PM |
1 |
History of the Zapata Telephony Project as it relates to the Asterisk PBX |
| 12:52PM |
1 |
Bristuffed Asterisk 1.0.3 hfc-s card doesn't work |
| 11:33AM |
1 |
[OFF TOPIC] Voip phone sellers in India |
| 10:38AM |
0 |
RE: Inbound calls getting disconnected when I answer the phone, using 'SIP' |
| 10:09AM |
0 |
Using Goto with Asterisk Realtime configuration |
| 9:47AM |
2 |
Asterisk and InterTel Axxess system? |
| 8:47AM |
2 |
GSM adapter for analog telephone - connect with fxo or fxs to Asterisk |
| 8:42AM |
1 |
Wait indefinitely? |
| 8:38AM |
1 |
passing opermode to the wcfxs module |
| 3:29AM |
1 |
PRI AOC (Advice Of Charge) |
| 3:18AM |
2 |
X100P random hangups - Please help with suggestions |
| 1:43AM |
0 |
Extension No.s not being received correctly. |
| 12:00AM |
1 |
Inbound calls getting disconnected when I answer the phone, using 'SIP'. |
| |
| Saturday January 8 2005 |
| Time | Replies | Subject |
|---|
| 8:30PM |
1 |
No such extension {Scanned} |
| 8:14PM |
3 |
ASTCC questions |
| 7:20PM |
2 |
Connecting Phone To Asterisk |
| 6:25PM |
0 |
Re: Asterisk-Users Digest, Vol 6, Issue 105 |
| 4:51PM |
3 |
OT help with rmdir pls |
| 4:14PM |
0 |
Asterisk and echo |
| 3:02PM |
3 |
Echo on Zaptel FXO :( |
| 2:12PM |
0 |
FastAGi change |
| 12:46PM |
1 |
Monitor command volume |
| 12:05PM |
0 |
MGCP phone |
| 11:16AM |
0 |
FYI: NIST issues recommendations for secure VOIP |
| 10:25AM |
0 |
484 Address Incomplete |
| 10:08AM |
4 |
Best gateway to use for *? |
| 9:45AM |
0 |
How to use a codec depending on call type ? |
| 9:29AM |
2 |
SIP and NAT problems "imagine that :) " |
| 9:17AM |
3 |
virtual pbx |
| 8:43AM |
1 |
What is acceptable network latency for voipconnection? |
| 8:06AM |
1 |
zaptel fxotune.c tool |
| 7:54AM |
0 |
Any experience with Linksys WRT54GP2 as localextensions to Asterisk ? |
| 7:29AM |
0 |
Any experience with Linksys WRT54GP2 as local extensions to Asterisk ? |
| 6:31AM |
1 |
Asterisk calls without soft phones |
| 5:04AM |
0 |
Problem SJPhone+Qtek S100 PDA+Sandisk Wi-Fi 256 MB SD+Asterisk |
| 4:30AM |
7 |
France has their (first?) SIP carrier with "unlimited" calls for 6eu/mo |
| 4:23AM |
1 |
Wildcard x100p and Redhat 9.0: Unable to get parameters |
| 3:40AM |
4 |
Toronto? |
| 1:08AM |
8 |
How do i "talk" to the IAXy...? (Newbie Alert) |
| |
| Friday January 7 2005 |
| Time | Replies | Subject |
|---|
| 10:43PM |
0 |
asking for readers input into the following config... |
| 10:06PM |
0 |
Inbound Pickup Issue - Sipmedia |
| 9:34PM |
3 |
Connecting Sip phone to asterisk. |
| 9:00PM |
4 |
MINNESOTA: TwinCities Asterisk Users Group - Meeting tomorrow 01/08/05 11:30am |
| 8:19PM |
0 |
New 'n' priority |
| 5:27PM |
7 |
Channel Variable |
| 2:52PM |
1 |
Newbe Can't dial local numbers. |
| 12:20PM |
2 |
Loading module app_realtime.so failed! |
| 12:13PM |
1 |
xmitting CallerID |
| 10:13AM |
1 |
New York? |
| 9:48AM |
0 |
mantis password reset link |
| 9:45AM |
2 |
Ringing an extension on multiple phones |
| 9:42AM |
4 |
can the dialtone be changed after pressing 9? |
| 9:11AM |
0 |
Re: [Serusers] softphones |
| 9:08AM |
3 |
Moderator on vacation? |
| 9:01AM |
1 |
Setting up Polycom IP 500 with * |
| 8:40AM |
1 |
oh323 driver installation - It works now |
| 8:35AM |
2 |
Asterisk 1.0.2 - Unable to allocate channel structure |
| 8:34AM |
1 |
Test2 |
| 8:31AM |
0 |
Re: kind of Urgent (Fedora Core 3 & Asterisk) |
| 7:40AM |
4 |
Monitoring |
| 7:39AM |
4 |
International area codes (incl. mobile) |
| 7:29AM |
0 |
PolyCom IP3000, gnugk and * audio problems |
| 7:11AM |
0 |
How do I get version 1.x from theDigium CVS orelsewhere? |
| 7:07AM |
5 |
fax e-mail spandsp |
| 7:03AM |
0 |
How do I get version 1.x from theDigium CVS or elsewhere? |
| 6:50AM |
0 |
x100p to X-lite works but x-lite to x-lite not (can not transmit audio) |
| 6:02AM |
1 |
TDM400P - Segmentation fault - Help! |
| 3:39AM |
1 |
Sip protocol question ... |
| 3:36AM |
1 |
off topic - SSH configuration for Digium Support |
| 2:48AM |
1 |
specific call transfer |
| 2:43AM |
7 |
Problem with call pickup |
| 2:04AM |
1 |
NIC irq load balancing |
| 12:42AM |
0 |
how to config call waiting and three way calling |
| 12:19AM |
0 |
Sip Phone Won't Login... |
| 12:18AM |
4 |
Broadvoice Status Check 11:18pm PST |
| |
| Thursday January 6 2005 |
| Time | Replies | Subject |
|---|
| 10:57PM |
1 |
{Scanned} |
| 10:19PM |
3 |
IAX outgoing redundancy |
| 10:19PM |
1 |
Enhancing performance and utility of an Asterisk machine |
| 9:42PM |
0 |
Reception System |
| 8:11PM |
0 |
TA register to Asterisk and getting down after notify msg, why? |
| 8:06PM |
2 |
Message light on 7960 or in this case no message light |
| 7:58PM |
2 |
Multiple lines on Cisco 7960 |
| 7:25PM |
0 |
Asterisk and Samsung DCS integration |
| 7:02PM |
2 |
Queue app following dialplan |
| 6:47PM |
0 |
T.38 Passthrough |
| 6:39PM |
2 |
Sipura SPA-1001 and Tivo Series 1 |
| 6:32PM |
1 |
ASTCC Bounties |
| 6:15PM |
2 |
3 site asterisk installation question |
| 5:59PM |
0 |
Internal/External IP |
| 5:53PM |
0 |
Line drops after 5-10 seconds |
| 5:47PM |
0 |
chan_capi compile problem |
| 5:02PM |
0 |
call waiiting and 3 way calling |
| 4:48PM |
1 |
Strange problem with incoming call. |
| 4:43PM |
1 |
Numbering plan for incoming call CLID on chan_zap (PRI) |
| 4:12PM |
1 |
What's wrong with compile |
| 4:11PM |
1 |
TE410P problem (Looping UP Span 1...) |
| 3:56PM |
3 |
Sip providor reference in extentions.conf |
| 3:37PM |
6 |
TDM4000P with 4 FXO's not picking up ringing lines |
| 3:17PM |
3 |
OT: TE405P pins and slots |
| 2:42PM |
0 |
Asterisk and SER security doubts |
| 2:14PM |
3 |
spandsp and app_rxfax (alternative topic: t38modem) |
| 2:12PM |
0 |
Incoming calls from I-net only for IP-address? |
| 1:51PM |
0 |
Re: Asterisk-Users Digest, Vol 6, Issue 76 |
| 1:26PM |
0 |
Re: kind of Urgent (Fedora Core 3 & Asterisk) |
| 1:21PM |
2 |
Zaptel Compile |
| 12:59PM |
1 |
Re: Asterisk-Users Digest, Vol 6, Issue 73 |
| 12:15PM |
1 |
Number of Zap channels in use |
| 12:11PM |
0 |
re: asterisk and libretel |
| 12:11PM |
2 |
POTS Lines |
| 12:01PM |
1 |
destroy SIP channel?? |
| 11:39AM |
0 |
Asterisk latest from CVS: SIP registrations fail |
| 11:10AM |
2 |
ZapRAS with BRI |
| 11:05AM |
1 |
TDM400P - Segmentation fault |
| 10:32AM |
12 |
kind of urgent |
| 10:08AM |
1 |
T100P + Adtran TSU600 + FXO and caller id problems |
| 9:55AM |
0 |
using native moh |
| 9:46AM |
0 |
H.323 to SIP extension |
| 9:46AM |
0 |
Asterisk and multiple default routes (sort of) - does not work |
| 9:42AM |
0 |
Call Pickup Problem |
| 9:04AM |
0 |
Problems with FXO interface on TDM400P |
| 8:46AM |
8 |
Asterisk startup |
| 8:33AM |
0 |
Out of Office AutoReply: asterisk addson |
| 7:58AM |
2 |
Inbound calls (similar problem; ISDN - chan_capi) |
| 7:30AM |
1 |
answer supervision for POTS FXO interfaces |
| 7:24AM |
0 |
Does spandsp work with capi channels ? |
| 7:19AM |
1 |
zaptel service stopped working |
| 7:15AM |
2 |
TDM400P - Problems |
| 7:01AM |
1 |
Problems with MeetMe accepting conference PIN |
| 6:58AM |
0 |
FW: Re: Polycom IP500 - problems with multiplesimultaneous calls |
| 6:30AM |
1 |
Gotoif question |
| 6:19AM |
1 |
Twin Cities Asterisk meeting still on for Saturday? |
| 6:18AM |
1 |
Sip Subscribe |
| 6:13AM |
0 |
Four HFC-S Cards in one System - does it work? |
| 6:10AM |
0 |
ISDN, bristuff and hylafax |
| 5:35AM |
1 |
.call MeetMe |
| 5:22AM |
3 |
Changing caller ID based on the extension dialled? |
| 4:55AM |
4 |
fax to email |
| 3:41AM |
0 |
mp3player - sounds terrible |
| 2:52AM |
3 |
DTMF problems on phonecell |
| 2:36AM |
1 |
calling with out registration |
| 1:29AM |
1 |
Sipura 2000 vs 2100 |
| |
| Wednesday January 5 2005 |
| Time | Replies | Subject |
|---|
| 11:56PM |
2 |
Inbound Calls |
| 11:34PM |
3 |
asterisk addson |
| 10:44PM |
0 |
if ${variable} include xxx ??? |
| 10:14PM |
2 |
Glophone/Voiceglo and Asterisk |
| 8:01PM |
1 |
Ouch... Error while writing audio data |
| 6:28PM |
1 |
CVS Compile problem on Solaris |
| 6:09PM |
1 |
modprobe: Can't locate module wctdm |
| 6:06PM |
0 |
Polycom IP500 - problems with multiplesimultaneous calls |
| 5:51PM |
5 |
Polycom IP500 - problems with multiple simultaneous calls |
| 5:14PM |
5 |
"Out the box" solutions? |
| 5:01PM |
7 |
TDM04B vs Dell |
| 4:52PM |
1 |
Read() timeout hangs up the line |
| 4:50PM |
2 |
Streaming Audio - Music On Hold Feature |
| 4:46PM |
2 |
queues - announcements and not busy members |
| 4:42PM |
4 |
Aaargh Gentoo updated some packages now * won't start |
| 4:39PM |
7 |
Realtime |
| 4:08PM |
0 |
sip.conf asterisk to vonage |
| 3:32PM |
1 |
debug channel <n> |
| 2:50PM |
0 |
Twin Cities Asterisk meeting this Saturday? |
| 2:25PM |
0 |
www.cuphone.com PCI hardware |
| 2:20PM |
0 |
funny little question regarding asterisk as a pbx vs a key system [slightly OT] |
| 2:11PM |
2 |
IP Phone suggestion. |
| 2:03PM |
3 |
X-lIte behind NAT and Asterisk behind NAT |
| 1:50PM |
1 |
chan_oh323 Module for Asterisk |
| 1:12PM |
2 |
Allowing "pooling" or "rollover" for inbound calls on VoicePulse |
| 1:02PM |
5 |
Asterisk with MySQL |
| 12:49PM |
1 |
Forwarding Voicemail Crashes Asterisk |
| 12:12PM |
1 |
ASTCC Compiling Problem |
| 12:05PM |
2 |
Music from Freeplay music included in * ?? |
| 11:53AM |
2 |
lcdproc and asterisk |
| 11:09AM |
3 |
Sending DTMF to PSTN problem with SIP |
| 11:05AM |
0 |
(no subject) |
| 10:48AM |
4 |
Broadvoice / * re-register issues |
| 9:54AM |
0 |
Getting Agent Channel information |
| 9:34AM |
0 |
Asterisk consultant wanted - S. California |
| 9:23AM |
3 |
Bootable Asterisk CD ? |
| 9:20AM |
0 |
VoIP Provider Peering |
| 9:01AM |
0 |
does TE405P support 3Bit CAS? |
| 9:00AM |
3 |
Last callers script? |
| 8:40AM |
0 |
One way audio [Asterisk + Innovaphone IP3000 + asterisk-oh323/h323] |
| 8:37AM |
5 |
asterisk - oh323 driver |
| 8:34AM |
2 |
Asterisk Pbx Manager Equivalent (in plain text - apologies to those that dont like HTML mail!!) |
| 8:23AM |
0 |
Re: Speex codec problem (unresolved ?) = Fixed |
| 8:23AM |
5 |
Happy Wednesday Morning SMS question, slightly OT |
| 8:22AM |
0 |
Asterisk Pbx Manager Equivalent |
| 7:56AM |
4 |
ISDN/SS7 book? |
| 7:47AM |
0 |
Problems with msn's, did not find device for msn |
| 7:43AM |
0 |
Asterisk as Nortel option 11 Autoattendant, question |
| 6:52AM |
1 |
TDM400P + Asterisk + zaptel timer ? |
| 6:51AM |
2 |
Versions of * what do they do/where is the change history/docs? |
| 5:41AM |
0 |
Usage Of Additional LEDs For Snom (was; Status of SNOM Intercom) |
| 4:41AM |
1 |
New asterisk installation but no audible voicemail prompts? |
| 4:33AM |
1 |
Cannot Hear at all |
| 1:54AM |
1 |
Speex codec problem (unresolved ?) |
| 1:23AM |
0 |
Asterisk with Euro ISDN, etc |
| 1:01AM |
13 |
Digium T100P T1 Card |
| 12:57AM |
0 |
Some bugs in DIAX 0.9.9f are now solved |
| 12:32AM |
1 |
Can't initiate a call with X-Lite. |
| 12:10AM |
1 |
IAX phones |
| |
| Tuesday January 4 2005 |
| Time | Replies | Subject |
|---|
| 11:55PM |
1 |
Call(out) routing |
| 11:51PM |
0 |
Group= equivalent for sip channels? |
| 11:00PM |
0 |
Question on behalf of a wannabe new list member |
| 9:40PM |
1 |
Re: Polycom Buddy Feature |
| 9:14PM |
1 |
Login Incorrect Message |
| 8:04PM |
3 |
Where to start. {Scanned} |
| 7:09PM |
1 |
How can I silently use ASTCC? |
| 6:57PM |
2 |
Which numbers should be blocked? |
| 4:02PM |
1 |
IAXy Static... and other issues |
| 3:43PM |
0 |
Manager API - ExtensionState help please. |
| 3:38PM |
2 |
integrating with panasonic td-1232 |
| 3:26PM |
1 |
CallerID in Australia & Analogue PSTN Phone System |
| 2:58PM |
3 |
AVM C2 capi.conf ? |
| 2:42PM |
1 |
Sprint Vision Phones ReadyLink=SIP? |
| 2:33PM |
3 |
Do Not Disturb |
| 2:14PM |
0 |
Ericsson 4422/4425 phones |
| 1:53PM |
6 |
Polycom Buddy Feature |
| 1:13PM |
1 |
Asterisk in a mixed phone environment |
| 1:00PM |
2 |
Asterisk stops - why ? |
| 12:33PM |
0 |
the correct way to stop a CDR? |
| 11:30AM |
6 |
OT: List of VoIP providers? |
| 11:27AM |
1 |
modprobe ztdummy hangs |
| 11:27AM |
2 |
Vonage WiFI Phone... |
| 10:58AM |
0 |
Does congestion exit on a different priority? |
| 10:19AM |
4 |
queue_log |
| 10:11AM |
1 |
DID and Callback - Questions!!! |
| 10:04AM |
1 |
Newb howto request: *, Voice Pulse Connect, & SJPhone |
| 9:57AM |
3 |
voiptalk.org IAX service - user experiences |
| 9:48AM |
1 |
ChanSpy - Should I repatch it ? |
| 9:11AM |
1 |
HDLC Bad FCS (8) HDLC Abort on TE410P |
| 9:08AM |
1 |
dialplan question - how to dial an * extension to get an outbound dialtone? |
| 8:44AM |
0 |
sip.conf [externip] |
| 8:38AM |
3 |
Kirk SIP-DECT gateway |
| 8:16AM |
0 |
Asterisk CLI : Scrollback with Putty and Screen |
| 7:55AM |
1 |
Avaya IPO412 |
| 7:08AM |
0 |
DIAX 0.9.9f website updated |
| 6:57AM |
0 |
cid_num with Asterisk CVS 1.0.12 |
| 6:53AM |
1 |
Don't receive the prefix |
| 6:20AM |
1 |
Displaying incoming e.164 callers number - how? |
| 6:15AM |
0 |
Cisco 7200 One-Way Audio |
| 5:56AM |
1 |
configuring sample time period for codecs? |
| 4:58AM |
0 |
"Hey look ma, it's not an RPM..." |
| 4:47AM |
0 |
Dell Poweredge 6300 & 4 analogue lines |
| 4:22AM |
0 |
Asterisk and rtp:// streams |
| 4:18AM |
4 |
Status of SNOM Intercom |
| 3:09AM |
0 |
OT: Asterisk at CeBIT 2005? |
| 2:47AM |
0 |
[OT] Anyone used Metrowerks PCS to build / distribute Asterisk |
| 2:28AM |
0 |
Making an ISDN call via Asterisk? |
| 1:31AM |
0 |
Re: 8 pstn lines+ on Asterisk supported |
| 1:05AM |
0 |
Re: Re: 8 pstn lines+ on Asterisk supported |
| 12:41AM |
0 |
RE: Asterisk-Users Digest, Vol 6, Issue 29 |
| |
| Monday January 3 2005 |
| Time | Replies | Subject |
|---|
| 11:56PM |
0 |
SPA-3000 as FXO Gateway for * (Was: Qs aboutFXO/FXS cards) |
| 11:32PM |
0 |
Echo problem - (sorry if this is an nmf question) |
| 10:58PM |
0 |
Verisign SIP7 sip<-->ss7 service |
| 10:22PM |
0 |
queue_log wrong? |
| 9:51PM |
2 |
IAX2 (IAXy) and DTMF Question |
| 9:12PM |
0 |
Re: Asterisk won't register with sipphone.com |
| 9:12PM |
3 |
Ignoring a ringing connection |
| 8:49PM |
5 |
Xorcom Rapid CD for Production? |
| 7:40PM |
1 |
Anyone ever get the Polycom Microbrowser XML document? |
| 7:25PM |
0 |
X100P - check channel busy? |
| 7:25PM |
1 |
call transfer to conference call |
| 6:38PM |
4 |
Manager API |
| 5:51PM |
2 |
zaptel error while initiating |
| 4:04PM |
0 |
reliable capacity for a single * box |
| 2:45PM |
2 |
sendURL |
| 2:41PM |
2 |
SIP Jitter buffer(control?) |
| 2:32PM |
3 |
UPS - a little OT |
| 2:18PM |
0 |
queuing questions |
| 1:58PM |
1 |
echo test application delay using the asterisk cli |
| 1:23PM |
5 |
Does Digium work on Mondays? |
| 1:09PM |
3 |
Line-in as MOH source |
| 1:04PM |
1 |
realtime audio for asterisk using jack |
| 12:53PM |
6 |
QOS / Cisco / Asterisk |
| 12:17PM |
2 |
PSTN to VoIP |
| 12:08PM |
3 |
oh323 context for peers |
| 11:45AM |
0 |
followup on FXO Call progress question |
| 11:40AM |
0 |
Cisco As5xxxx audio issues |
| 11:34AM |
20 |
TE410P card in an HP-Compaq DL380 G4 server |
| 10:43AM |
5 |
8 pstn lines+ on Asterisk supported hardware. |
| 10:33AM |
0 |
disable ringback of held call on zap channel |
| 10:33AM |
0 |
new country tone/zone info setup in ZAPTEL and ZAPATA FXO config ? |
| 10:16AM |
1 |
Subject: Re: Dial with no phone line connected |
| 9:13AM |
5 |
DHCP Attribute for TFTP server for Aastra 480i? |
| 8:55AM |
0 |
2 E100P card |
| 8:29AM |
0 |
Dialplan, LCR |
| 8:11AM |
0 |
Checking to see if a dialplan variable is NULL, mysql app addon |
| 7:53AM |
2 |
agent with queues remain unavailable during transferred call |
| 7:50AM |
2 |
Speex codec for 8Kbps setting ? |
| 7:50AM |
2 |
PSTN to VoIP FXO gateways? |
| 7:42AM |
0 |
SPA-3000 as FXO Gateway for * (Was: Qs about FXO/FXS cards) |
| 7:41AM |
0 |
PRI Errors: Here is where I am at... |
| 7:37AM |
0 |
ENUM: which number? |
| 7:24AM |
3 |
Asterisk CPU priorities (nice?) |
| 6:41AM |
0 |
###SORRY### |
| 6:33AM |
1 |
Registration server changed or down? |
| 5:34AM |
2 |
finding current codec? |
| 5:18AM |
9 |
Just saw your [Asterisk] xJack Segfault in Asterisk |
| 4:49AM |
6 |
SipSak: error: this FQDN or IP is not valid: voicegw |
| 2:37AM |
0 |
How to compile zaprtc on CoLinux Debian Package |
| 2:15AM |
4 |
TE410P - Normal activity ? |
| 1:42AM |
0 |
Limit max calls & call duration |
| |
| Sunday January 2 2005 |
| Time | Replies | Subject |
|---|
| 11:51PM |
1 |
Configuration details for Asterisk interaction with Vocal |
| 11:21PM |
1 |
extensions.conf sorting |
| 8:39PM |
0 |
???? |
| 7:30PM |
1 |
Call Queue Question |
| 6:56PM |
0 |
Terminal Adaptor down after register |
| 5:49PM |
0 |
Using Asterisk as a TA? |
| 3:59PM |
3 |
Indications UK - cant get away from american sounding dial tone |
| 3:45PM |
1 |
pridialplan=unknown ? |
| 3:14PM |
1 |
Can I receive faxes with Fritz card & Asterisk ? |
| 2:35PM |
12 |
phones with two ethernet ports |
| 2:24PM |
1 |
Subject: Re: Dial with no phone line connected |
| 1:55PM |
3 |
Codec Selection in Asterisk |
| 1:31PM |
0 |
Box unstable after loading zaptel drivers for X100P |
| 1:16PM |
1 |
Clipping on outbound calls via SIP/IAX |
| 12:17PM |
1 |
ArtDio IPF-2000 or Sipura SPA-841 |
| 10:21AM |
2 |
Booting * from CF |
| 7:52AM |
2 |
Dialling 9 for an outside line |
| |
| Saturday January 1 2005 |
| Time | Replies | Subject |
|---|
| 11:50PM |
3 |
Announcements via IAX phones |
| 11:48PM |
0 |
outgoing call (Sip phones to PSTN) |
| 7:45PM |
1 |
ASTCC gsm files |
| 5:58PM |
0 |
Audio breakup problems |
| 5:49PM |
0 |
Asterisk@home ISO install of ISDN card with HFC ? |
| 4:52PM |
1 |
Problems to use asterisk with mysql /odbc |
| 1:52PM |
25 |
Qs about FXO/FXS cards |
| 1:50PM |
5 |
sip reload - Hang |
| 11:05AM |
1 |
spandsp app_rxfax - the sending software loops |
| 10:05AM |
0 |
Asterisk dies every hour |
| 1:28AM |
0 |
Help with AGI script calleridnamelookup.agi |