| Tuesday May 31 2005 |
| Time | Replies | Subject |
|---|
| 11:20PM |
0 |
A newbie question - SIP to Trunk |
| 7:51PM |
2 |
pbx -> fiber -> network media converter -> wifi -> network media converter -> fiber -> pbx ??? |
| 7:37PM |
0 |
Google Summer Of Code |
| 6:36PM |
0 |
* with mfcr2 and libunicall success stories ? |
| 5:32PM |
1 |
Phone always busy after caller hangup |
| 5:14PM |
1 |
Suppress "Missed Calls" 7960 SIP |
| 4:58PM |
4 |
Karl |
| 2:39PM |
1 |
Sipura 3000 - fax passthrough? |
| 2:31PM |
1 |
`hint` priority and Polycom 500 |
| 1:54PM |
4 |
Asterisk@Home 1.1b1 has been released |
| 1:06PM |
0 |
Re: chan_sccp / 7960: reproduceable semi-lockup |
| 12:21PM |
0 |
High CPU in Asterisk, chan_unicall and dtmf |
| 12:01PM |
3 |
Opinions of Sphinx? |
| 11:36AM |
0 |
Re: [cisco-voip] SIP Authentication problem between Cisco router and Asterisk when calls are forwarded |
| 10:56AM |
0 |
Codec ordering? |
| 10:30AM |
1 |
SIP Authentication problem between Cisco router and Asterisk when calls are forwarded |
| 10:18AM |
1 |
rxfax application - doesn't work properly |
| 9:58AM |
1 |
Re: chan_sccp / 7960: reproduceable semi-lockup |
| 9:56AM |
0 |
Re: chan_sccp / 7960: "End call" softkey: "That key is not active here" |
| 9:35AM |
0 |
Session Registrations |
| 9:07AM |
0 |
Campon feature? |
| 9:02AM |
2 |
ISO Suggestions for Multiple Inbound Voicepulse Lines |
| 8:36AM |
0 |
Receive calls with Aastra 480i phone problem |
| 8:26AM |
4 |
AreskiCC - DOES IT REALLY WORK?????? |
| 8:24AM |
1 |
# Transfers |
| 8:21AM |
2 |
R: R: R: R: AT-320 + supervised transfer |
| 8:18AM |
5 |
CIsco 7960 SIP Image |
| 8:13AM |
1 |
Built-In Transfer Questions |
| 8:09AM |
1 |
asterisk acts as media gateway for existing pabx ? |
| 7:36AM |
0 |
Connecting a peer to a dynamic ip asterisk b ox ??? |
| 6:59AM |
1 |
Re: astpp database creation failed...please help... |
| 6:43AM |
0 |
asterisk sip register with no username and password. |
| 6:40AM |
4 |
Chan_sccp / wiki |
| 6:37AM |
0 |
Sipura 3000 Analog Line No Answer, No Audio |
| 6:28AM |
4 |
Asterisk with another Asterisk |
| 6:26AM |
0 |
Polycom IP500 with Video |
| 6:08AM |
7 |
Tools for effectively manage Asterisk |
| 6:00AM |
0 |
'beeps' while recording..? |
| 4:51AM |
1 |
monitoring oh323 calls |
| 4:41AM |
3 |
Automatic Codec change for different communication channels!? |
| 4:22AM |
2 |
Ztdummy usage |
| 4:21AM |
0 |
Auto-generated incoming calls X100P |
| 4:08AM |
1 |
Asterisk compailation Error Chan_zap.c |
| 3:06AM |
2 |
Sipura 2000 behind NAT issue, Vonage is working |
| 2:54AM |
1 |
How does ISDN really work? |
| 2:51AM |
4 |
Extension context question |
| 1:26AM |
1 |
Uniden UIP1868 - any sightings or users? |
| 1:20AM |
0 |
Asterisk: HelpDesk / CRM type of Application in Asterisk |
| 1:04AM |
2 |
handytone 486 |
| 1:00AM |
10 |
UPS rating for SOHO asterisk box |
| 12:56AM |
0 |
UK NCFA calling |
| 12:56AM |
0 |
ipchains for firewall, QOS howto? |
| 12:53AM |
0 |
Re: Asterisk-Users Digest, Vol 10, Issue 234 |
| 12:14AM |
2 |
Problem with asterisk+gnugk |
| 12:11AM |
1 |
MGC on asterisk |
| |
| Monday May 30 2005 |
| Time | Replies | Subject |
|---|
| 11:37PM |
1 |
astpp database creation failed! |
| 11:23PM |
0 |
newbie problem with registration of sip client |
| 10:58PM |
1 |
Chan OH323 and overlapping digits |
| 10:44PM |
2 |
Sipura 3000 dialing "noise" |
| 10:18PM |
1 |
RE: Invalid login/password with AreskiCC V2 |
| 9:51PM |
0 |
Dialplan structure |
| 9:45PM |
1 |
Speed dial number and actually dialing combined with ASTCC |
| 9:39PM |
0 |
LCR and ASTCC |
| 9:10PM |
1 |
Codec - pay for and to whom? |
| 8:44PM |
1 |
Remote phone: Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from |
| 8:18PM |
1 |
Where to start to solve hardware problem? |
| 7:53PM |
0 |
Sipura 3000 does not dial out |
| 6:22PM |
1 |
Sipura ATA and Asterisk No Answer Issue |
| 4:47PM |
3 |
Connecting a peer to a dynamic ip asterisk box ??? |
| 2:46PM |
1 |
Urgent Help neededt!! Asterisk 1.0.7 CPU |
| 2:06PM |
1 |
I865, HFC-S etc. |
| 1:42PM |
2 |
Error in Zapata Config? |
| 12:48PM |
3 |
Asterisk install error ... |
| 12:22PM |
2 |
Multiple Ext on IP500 |
| 12:06PM |
0 |
transcoding prevention |
| 11:41AM |
2 |
pridialplan & prilocaldialplan |
| 11:13AM |
3 |
Urgent Help neededt!! Asterisk 1.0.7 CPU at 99% |
| 11:07AM |
0 |
where can i get a vanity DID? |
| 10:21AM |
4 |
R: R: R: AT-320 + supervised transfer |
| 9:34AM |
0 |
perl agi : get_variable problem |
| 9:32AM |
1 |
nntp access |
| 9:22AM |
2 |
R: R: AT-320 + supervised transfer |
| 8:53AM |
2 |
Meridian 808 Function |
| 8:19AM |
3 |
R: AT-320 + supervised transfer |
| 8:03AM |
2 |
choice of processors |
| 7:58AM |
1 |
asterisk compatible, hot swappable PRI card |
| 6:42AM |
0 |
Gradwell UK DID + DTMF |
| 6:21AM |
0 |
Areski Calling Card |
| 6:14AM |
5 |
Asterisk on Soekris |
| 6:05AM |
0 |
ISDN RAS and data calls |
| 6:00AM |
0 |
Pls, find me a VoIP Supplier/Reseller in Dubai-UAE |
| 5:59AM |
0 |
Flash Operator Panel 0.21 released |
| 5:48AM |
1 |
AT-320 + supervised transfer |
| 5:38AM |
0 |
MGCP and missing digit map |
| 4:26AM |
1 |
Serious ZapRAS problem! |
| 3:43AM |
0 |
@home to @home |
| 3:40AM |
0 |
IAX2 to H323 |
| 2:27AM |
1 |
app_senddtmf.so. |
| 2:25AM |
0 |
Asterisk with PrimuX 1S2M ISDN card |
| 2:10AM |
1 |
How to configure Inter7's Asterisk Fax with Postfix |
| 2:08AM |
0 |
IAX2 registration period |
| 1:48AM |
2 |
Problem with SIP clients |
| 1:36AM |
2 |
IAX encrytion |
| 1:01AM |
0 |
asterisk integration with Quintum Tenor AXT800! |
| 12:58AM |
0 |
Voicemail make crash |
| |
| Sunday May 29 2005 |
| Time | Replies | Subject |
|---|
| 11:10PM |
0 |
[Serusers] QOS of VoIP |
| 8:29PM |
0 |
Asterisk Multi Tenant setup |
| 8:27PM |
1 |
voice is coming only from one side |
| 8:05PM |
2 |
Recording does not stop. |
| 7:21PM |
1 |
Pre paid Card |
| 5:59PM |
4 |
Re: Digium Website Update: Asterisk Business Edition |
| 5:14PM |
1 |
ANNOUNCEMENTt: GPL Asterisk Billing Software |
| 3:26PM |
1 |
chan_unicall and dtmf problem |
| 3:20PM |
0 |
Custom Extension on AMP |
| 2:32PM |
2 |
Peer to Peer calls |
| 11:50AM |
0 |
chan_oss.c:572 oss_write: Unable to set device to input mode error |
| 11:43AM |
2 |
CallerID of calls FROM queue |
| 10:27AM |
1 |
60 second time out |
| 9:22AM |
0 |
Digium Website Update! |
| 7:51AM |
1 |
Upgrading my HOP-1002 software |
| 7:44AM |
3 |
BT100 Phone Died During Call |
| 6:22AM |
0 |
Asterisk 1.0.7 on VIA EPIA 5000 |
| 6:06AM |
0 |
How to Define Asterisk Behind a Nat |
| 5:24AM |
1 |
Database Usage with Asterisk |
| 3:21AM |
1 |
Error attempting to make Zaptel on Red Hat linux 9.0 |
| 12:37AM |
1 |
LCR |
| |
| Saturday May 28 2005 |
| Time | Replies | Subject |
|---|
| 8:51PM |
1 |
Recommendations are highly appreciated -SIP HARDWARE phone |
| 8:46PM |
1 |
Pictures of the Digium booth at ISPCon 2005 |
| 8:10PM |
0 |
parsing extension name in a command |
| 7:06PM |
1 |
3 goes and your out |
| 5:47PM |
0 |
Asterisk on a Linksys wrt54g |
| 3:54PM |
3 |
CallerID when transferring calls. |
| 3:31PM |
0 |
chan_sccp / 7960: ALERT_INFO? |
| 3:19PM |
1 |
cmd curl crashes asterisk: |
| 2:32PM |
0 |
Re: chan_sccp / 7960: "End call" softkey: "That key is not active here" |
| 12:40PM |
0 |
Re: chan_sccp / 7960: "End call" softkey: "That key is not active here" |
| 12:37PM |
0 |
Re: chan_sccp / 7960: 7960-font.xml |
| 12:33PM |
1 |
CallerID for UK |
| 11:55AM |
1 |
Pick up on first ring |
| 10:44AM |
0 |
Help with New SIP phone. |
| 9:02AM |
1 |
Enum or Dundi? |
| 8:09AM |
1 |
Fax and SIP Device |
| 7:41AM |
0 |
newbie asterisk SIP config question (using VoicePulse Connect!) |
| 7:34AM |
0 |
newbie asterisk SIP config question (using VoicePulse Connect) |
| 7:34AM |
0 |
Re: Asterisk-Users Digest, Vol 10, Issue 222 |
| 6:31AM |
2 |
UK DID providers |
| 5:45AM |
0 |
MWI - One mailbox, multiple extensions, lots of lights! |
| 3:27AM |
0 |
chan_sccp patches |
| 2:40AM |
2 |
xc-ast 0.9.0 is out today |
| 2:28AM |
0 |
TDM zap channel Exception on 15, channel 1 |
| 2:03AM |
0 |
Asterisk@home rejecting nufone incoming calls (iax2) |
| 12:20AM |
1 |
Quintum Tenor AXT800! |
| 12:17AM |
1 |
7960 / chan_sccp: Less than three lines / more than three speeddials possible? |
| 12:01AM |
1 |
ivr not working? |
| |
| Friday May 27 2005 |
| Time | Replies | Subject |
|---|
| 10:48PM |
1 |
Re: Asterisk-Users Digest, Vol 10, Issue 221 |
| 10:45PM |
0 |
Remote Server IAX Configuration |
| 10:03PM |
0 |
Asterisk vs pingtel? |
| 9:10PM |
1 |
Changes on CVS HEAD |
| 8:26PM |
3 |
Polycom phones, UNREACHABLE |
| 8:20PM |
1 |
Re: Asterisk-Users Digest, Vol 10, Issue 114 |
| 7:52PM |
0 |
you can bid on this very small project |
| 6:53PM |
0 |
How to timeout using AGI. |
| 4:47PM |
1 |
static linking |
| 3:36PM |
1 |
Asterisk stopping to respond and CPU at the top |
| 1:37PM |
0 |
Switch from NBX to Asterisk |
| 1:34PM |
3 |
Wacko Distinctive Ring Patterns being detected?? |
| 1:24PM |
0 |
DVG-1120S does not show callerid Name and resets time |
| 12:46PM |
0 |
sip phone behind nat connecting to an asterisk box that has one port on the open internet |
| 12:36PM |
0 |
SIP REFER: Trying again |
| 12:26PM |
1 |
VoiPSupply Dot Com: Epilogue |
| 12:18PM |
1 |
Soyo G688 |
| 11:57AM |
0 |
asterisk and nortel CS1000 using SIP |
| 11:54AM |
0 |
asterisk and nortel BS1000 using SIP |
| 11:42AM |
1 |
Fwd: Newbie here. Tips on setting up 100 phones wanted. |
| 11:37AM |
1 |
Upgraded firmware on Polycom 500, digit-order problems |
| 11:36AM |
1 |
How to Connect Netphone IP phone with ASterisk |
| 11:21AM |
0 |
Call waiting on TDM-400 FXO |
| 11:10AM |
0 |
Another OH323 Problem |
| 10:51AM |
2 |
Polycom IP 500 SIP bootrom and firmware upgrades |
| 10:18AM |
3 |
Newbie here. Tips on setting up 100 phones w anted. |
| 10:00AM |
6 |
Newbie here. Tips on setting up 100 phones wanted. |
| 9:01AM |
1 |
Temporary unavailable -???? |
| 9:01AM |
2 |
CRM integration (was RE: CallerID) |
| 8:12AM |
3 |
Recommended Network Latency |
| 8:01AM |
2 |
Grandstream GSX-2000 - dead :-( |
| 7:57AM |
2 |
PRI "Actual-HookState" not showing offhook on inbound |
| 7:56AM |
1 |
Unable to create channel of type 'Zap' with zaphfc driver |
| 7:48AM |
2 |
Interco H323 : IPNx (from WTL) and * |
| 7:40AM |
0 |
compiling new module; conflicting function |
| 7:22AM |
0 |
bristuff-0.2.0-RC8e priindication=passthrough problem |
| 6:55AM |
0 |
Cisco 7960 tftp "not null terminated" |
| 6:48AM |
3 |
Call waiting? |
| 6:35AM |
1 |
VoiceMail with Polycom 500 |
| 5:08AM |
0 |
problem about client authorisation |
| 5:08AM |
0 |
Can't transfer calls on polycom 500 after new firmware upgrade |
| 3:33AM |
0 |
Problem with SIP peer registration |
| 3:28AM |
0 |
spandsp <--> analogue modems? |
| 3:12AM |
3 |
G729 vs. gsm |
| 3:03AM |
2 |
5000 sip clients (voip phones) |
| 2:51AM |
0 |
H323 setup problem |
| 2:35AM |
0 |
CRM integration (was RE: CallerID) |
| 2:26AM |
0 |
Re: MoH: mgp123 problems |
| 2:23AM |
0 |
ASTCC/ 'L' option hangup wackyness |
| 2:08AM |
1 |
SIP SoftPhone for debuging |
| 1:42AM |
0 |
Re: Areski Calling Card Download locations |
| 12:22AM |
0 |
Re: Asterisk-Users Digest, Vol 10, Issue 215 |
| 12:20AM |
2 |
DID - B8 Message |
| 12:02AM |
2 |
chan_zap.c:8534 pri_dchannel: PRI Error |
| |
| Thursday May 26 2005 |
| Time | Replies | Subject |
|---|
| 11:23PM |
0 |
capi dial in/out configuration |
| 10:47PM |
1 |
does Jitter calculation in chan_iax2.c work??? |
| 9:30PM |
0 |
Asterisk crashes with sipp |
| 8:47PM |
4 |
International Caller ID? |
| 4:40PM |
1 |
TDM400P in 2U server? |
| 3:56PM |
3 |
Analog Telephone Adapter |
| 3:00PM |
1 |
How do I diagnose the problem in this Asterisk test session with FWD? |
| 2:39PM |
0 |
dhcp vars, mediatrix 1204's |
| 1:34PM |
1 |
Asterisk con X-lite : Register Ok but no calls (404 Not found) |
| 1:16PM |
4 |
tds_CDR and MS SQL Server troubleshooting |
| 1:10PM |
2 |
Limiting maximum runtime of echo test |
| 12:52PM |
1 |
Asterisk connecting to Nortel 1000 with SIP |
| 12:18PM |
0 |
Asterisk on 64 bit Linux |
| 12:12PM |
0 |
ICD Usage Examples |
| 12:12PM |
0 |
TDM > * rings once and goes fast busy |
| 12:03PM |
5 |
Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages |
| 11:57AM |
0 |
Connecting a couple DS0's to a wildcard |
| 11:38AM |
1 |
Dropping frame of G.729 since we already have a VAD frame at the end |
| 11:17AM |
0 |
app_dial_rev5 |
| 10:58AM |
1 |
YET Another echo issue PRI CARD Any help acc epted :-) |
| 10:49AM |
1 |
Using zap channels on 2 different servers |
| 10:14AM |
2 |
Asterisk@home - mysql login |
| 10:14AM |
4 |
YET Another echo issue PRI CARD Any help accepted :-) |
| 9:37AM |
0 |
jitter buffer recomendations |
| 9:13AM |
1 |
How do I know that my machine will support APIC? |
| 8:28AM |
0 |
DNS protocol flaw in Cisco client products announced May-24-2005 |
| 8:25AM |
4 |
multiples broadvoice lines |
| 8:03AM |
1 |
SIP V2 Support |
| 7:52AM |
0 |
SetCDRUserField not working in cvs 5/25/05 |
| 7:43AM |
0 |
PSTN->SIP->PSTN transfer problem |
| 7:30AM |
0 |
Scalability issue: P2P connection w/o passing through asterisk |
| 5:52AM |
5 |
SIP Soft Video phone for Asterisk usage |
| 5:49AM |
1 |
Echo with two IP phones through Asterisk using SIP |
| 5:41AM |
0 |
Q : registering sipXphone |
| 4:39AM |
1 |
Re: Asterisk-Users Digest, Vol 10, Issue 188 |
| 3:59AM |
1 |
Speakeasy as a VOIP provider? |
| 3:59AM |
1 |
VIDEO ON 1.0.7 stable |
| 3:24AM |
2 |
voicemail comprehension |
| 3:05AM |
0 |
Looking for wall mountable cases in the UK. |
| 3:03AM |
0 |
SV: Little Php question |
| 3:01AM |
1 |
Little Php question |
| 2:47AM |
0 |
video conference feature |
| 2:45AM |
0 |
Prad or V5.2 |
| 12:54AM |
2 |
chan_capi with an AVM C4 connected to 4 BRI-lines in PTP-Mode |
| 12:44AM |
1 |
deadlock |
| 12:38AM |
2 |
static database config gui |
| 12:31AM |
1 |
Size of extensions.conf |
| 12:14AM |
1 |
AS5300 + Asterisk |
| |
| Wednesday May 25 2005 |
| Time | Replies | Subject |
|---|
| 11:30PM |
3 |
Asterisk Versions |
| 11:01PM |
6 |
new cisco ip video phone? |
| 10:31PM |
0 |
Port 6057 blocked on firewall |
| 8:07PM |
7 |
Survey: E1 prices |
| 7:24PM |
0 |
Choppy audio |
| 7:02PM |
0 |
AMP 1.10.008 released! |
| 6:51PM |
0 |
Tying together two Asterisk servers |
| 6:25PM |
1 |
Legacy Toshiba integration |
| 5:51PM |
0 |
FAST BUSY on Back to back ZAP outgoing calls |
| 5:18PM |
0 |
MeetMe Announce User feature |
| 3:29PM |
1 |
Default caller ID |
| 3:25PM |
5 |
SER Config for Asterisk |
| 3:13PM |
4 |
SER Help |
| 2:02PM |
0 |
bounty: app_queues.c with mysql support |
| 1:53PM |
5 |
Asterisk Crashing; Not getting Core dumps |
| 1:52PM |
0 |
Excellent Article explainng what is up with Broadvoice |
| 1:46PM |
1 |
Asterisk and SER on Same Box |
| 1:30PM |
0 |
Cisco 7960 Firmware help pleas |
| 1:10PM |
0 |
Sipura 3000 sound problems |
| 1:04PM |
0 |
Correctly handle two extensions for the same phone (one with voicemail one without) |
| 1:04PM |
1 |
astcc no billed cost |
| 12:46PM |
1 |
Problems with Public IP |
| 12:44PM |
1 |
Remote Voicemail Notifier / enter Dialplan on SIP Register |
| 12:05PM |
0 |
Remote Voicemail Notifier / enter Diaplplan on SIP Register |
| 11:59AM |
0 |
CRM integration (was RE: CallerID) |
| 11:53AM |
0 |
Getting firmware |
| 11:18AM |
2 |
Conferences using Manager API |
| 10:58AM |
8 |
What does Asterisk need in the way of a GUI? |
| 10:34AM |
13 |
Cisco 7960 Firmware help please. |
| 10:25AM |
0 |
oh323 problems - Solved |
| 10:13AM |
2 |
Nortel i2004 firmware upgrade. |
| 10:11AM |
2 |
Budgetone 102 and voicemail problem |
| 10:08AM |
4 |
Polycom IP501 |
| 9:54AM |
7 |
zaphfc: empty HDLC frame or bad CRC received |
| 9:40AM |
4 |
Asterisk's MultiProcessor Ability |
| 9:18AM |
2 |
Manager and Callerid problems |
| 9:04AM |
1 |
Looking for list with asterisk default extensions |
| 8:40AM |
0 |
May Twenty Fifth SayUnixTime |
| 8:39AM |
0 |
Attended Transfer failing with Agents |
| 8:34AM |
1 |
Asterisk, 2 x network interfaces and traffic shaping on same box? |
| 8:12AM |
2 |
CRM integration (was RE: CallerID) |
| 7:59AM |
1 |
LiveVoip does not like customers anymore, .... |
| 7:58AM |
0 |
CRM integration (was RE: CallerID) |
| 7:39AM |
0 |
CRM integration (was RE: CallerID) |
| 7:25AM |
0 |
VoIP-Forum.se - new Swedish user forum |
| 7:14AM |
1 |
CRM integration (was RE: CallerID) |
| 6:52AM |
3 |
sip extension logon failed problem |
| 6:11AM |
0 |
Meetme - any way to stop a participant receiving audio? |
| 6:07AM |
1 |
Can Ztdummy be used in production environment |
| 5:32AM |
2 |
MoH: mpg123 problems |
| 5:15AM |
0 |
G.729 disappears from h.323 codecs. Help, please! |
| 5:09AM |
2 |
Asterisk and Monwall - comments |
| 4:07AM |
0 |
Is SKYPE a threat orshould wedo something(together) |
| 3:14AM |
5 |
how to dial extension with menu |
| 2:47AM |
0 |
configuration asterisk zap module |
| 2:41AM |
1 |
Possible to send Calling Number as TON: international ? |
| 2:39AM |
15 |
PHP/AGI Problem |
| 2:36AM |
2 |
HiPath 4000 and Asterisk |
| 2:17AM |
1 |
Polycom IP 600 DHCP problem |
| 2:15AM |
1 |
Asterisk SIP cannot restrict call from softphone before registration |
| 1:41AM |
2 |
RTP path with Cisco CCM |
| 1:12AM |
0 |
Segfaults on Asterisk HEAD |
| 12:56AM |
5 |
C files of Asterisk |
| |
| Tuesday May 24 2005 |
| Time | Replies | Subject |
|---|
| 11:53PM |
3 |
rxfax(spandsp-0.0.2pre18) and HT488 |
| 10:58PM |
1 |
OT: cisco ip phone security problem |
| 10:14PM |
3 |
New Grandstream phones. |
| 9:38PM |
6 |
echo problem |
| 9:14PM |
1 |
General AGI Question |
| 8:35PM |
0 |
[***POSSIBLE SPAM***]RE: Help Configuring CiscoATA 186 |
| 8:13PM |
2 |
Help Configuring Cisco ATA 186 |
| 8:02PM |
0 |
silence in virtual extension |
| 7:28PM |
0 |
Firmware for Cisco ATA 186 |
| 6:26PM |
0 |
IPswitch brings up a message daily |
| 5:09PM |
4 |
audio message delivery |
| 4:46PM |
1 |
Cisco Config |
| 4:02PM |
0 |
Xeon server board for TE405P |
| 2:00PM |
3 |
Budgetone and NAT not working |
| 1:53PM |
3 |
PHPAGI problems |
| 1:42PM |
0 |
asterisk take 99% of CPU resources |
| 1:38PM |
0 |
Re: origination providers (mike castleman) |
| 1:35PM |
0 |
Redirection |
| 1:30PM |
0 |
Re: IAX Firefly config (Jeromy Grimmett) |
| 1:28PM |
0 |
G729 and XTen Pro |
| 12:53PM |
0 |
Key Rotary Lines ? |
| 12:28PM |
1 |
realtime static |
| 12:24PM |
2 |
RE: Firefly config |
| 12:00PM |
1 |
origination providers |
| 11:58AM |
3 |
How To Connect an IP phone with asterisk |
| 11:50AM |
1 |
CTI |
| 11:23AM |
5 |
Red Alarm TE110P |
| 11:23AM |
0 |
Sipura SPA-3000 call progress, and interdigit delays |
| 11:12AM |
0 |
IAX Firefly config |
| 10:45AM |
0 |
302 redirection issue |
| 10:12AM |
5 |
MySQL Support For OS X |
| 10:03AM |
0 |
CallerID with Lucent TNT |
| 9:54AM |
1 |
Fax detection: Problem with extension number |
| 8:16AM |
2 |
Dial to a SIP fone ends up at Voicemail Busy |
| 7:31AM |
2 |
capi.conf |
| 6:43AM |
1 |
5ESS central office question |
| 6:28AM |
0 |
Problem with FXO taking a call |
| 6:28AM |
4 |
Rings - How to set number |
| 6:06AM |
0 |
txfax return code |
| 6:01AM |
1 |
Random Sound File |
| 5:51AM |
2 |
spandsp issue |
| 5:18AM |
1 |
Digium Wildcard X100P Error |
| 5:12AM |
0 |
Digium T100 Error |
| 5:07AM |
0 |
Echo with Digium TDM02B |
| 4:59AM |
0 |
How do you prevent a 3-way conference if an extension is busy ? |
| 4:36AM |
2 |
writing to MYSQL database |
| 3:51AM |
1 |
Silence supression |
| 3:33AM |
0 |
Problems installing TDM22B |
| 3:31AM |
1 |
Asterisk processes |
| 3:18AM |
0 |
DIAL with FastAGI and Answer Supervision |
| 2:01AM |
0 |
Remote-Party-ID handling |
| 1:47AM |
2 |
How to setup Dundi in Asterisk? |
| 1:35AM |
0 |
record message during dial |
| 1:35AM |
1 |
BudgeTone 101 doesn't register with FirmWare 1.5.23 |
| 1:34AM |
1 |
Early B3 connects on zaphfc |
| 1:33AM |
0 |
H323 integrated Asterisk support |
| |
| Monday May 23 2005 |
| Time | Replies | Subject |
|---|
| 11:19PM |
4 |
Broadvoice delivers CID even when restricted? |
| 10:31PM |
0 |
Callerid problem with identapop pro |
| 10:29PM |
0 |
App_odbcexec |
| 7:42PM |
1 |
Basic newbie questions |
| 5:40PM |
5 |
Inbound call center - reliability \ scalabil ity with queues |
| 5:12PM |
0 |
Inbound call center - reliability \ scalability with queues |
| 4:27PM |
3 |
ISPCON Mini-emergency: ATA186 Power Cube OK on SPA841? |
| 3:52PM |
1 |
Please explain this streaming example to me |
| 3:11PM |
0 |
OT: recover complete AES keys |
| 1:32PM |
0 |
spa-1001 not getting a dial tone on my pbx |
| 1:31PM |
3 |
Junction Networks |
| 12:24PM |
5 |
bluetooth headset/handsfree |
| 12:15PM |
2 |
E&M Tie Line |
| 11:52AM |
0 |
How to detect DTMF and change if needed |
| 11:44AM |
4 |
Digium FXS modules too fragile? |
| 11:28AM |
1 |
OH323 CONTROL PROTOCOL ERROR |
| 10:23AM |
1 |
t38modem |
| 10:11AM |
4 |
How do you transfer a call to a busy extension ? |
| 9:55AM |
0 |
Sip reg problem |
| 9:39AM |
1 |
ZyXEL Prestige 2000W - cant make a call? |
| 9:29AM |
0 |
SIP authentification? Any ideas? |
| 9:08AM |
9 |
Windows IAX Softphone |
| 9:07AM |
1 |
E1 PRI Warnings |
| 8:29AM |
4 |
Programs to parse queue_log |
| 8:01AM |
0 |
Message in event_log. |
| 7:34AM |
8 |
play gsm files in windows |
| 7:30AM |
0 |
Drag and Drop with IPS |
| 7:26AM |
1 |
SendDTMF into a conference room |
| 7:16AM |
1 |
two isdn cards |
| 6:48AM |
1 |
Astersik vs. Pingtel |
| 4:38AM |
1 |
Grandstream GXP-2000 headset |
| 4:37AM |
7 |
Cisco 7960 & v7.4 |
| 3:37AM |
0 |
Two or more asterisk servers, shared dialplan. Please help |
| 2:59AM |
0 |
Modifying the RTP and SIP protocol |
| 2:53AM |
0 |
Modifying Asterisk's C files |
| 1:52AM |
0 |
All channels on PRI stuck "Resetting" |
| 1:33AM |
1 |
How to connect to IPTEL.ORG |
| 1:29AM |
0 |
SV: ZAP/DTMF |
| 1:05AM |
4 |
CallerID, TAPI and CTI |
| |
| Sunday May 22 2005 |
| Time | Replies | Subject |
|---|
| 11:31PM |
1 |
spa-1001 with asterisk? |
| 11:00PM |
2 |
Looking for people to test calls |
| 10:25PM |
1 |
Which H.323 for Stable? |
| 9:53PM |
4 |
Cisco 7940g Firmware load problems |
| 9:16PM |
0 |
Using patch -p0 <meetme-diff-cbmysql_1.txtproduces 'malformed patch' message |
| 9:02PM |
0 |
how to forward a call to mobile? |
| 8:01PM |
2 |
Not answering/script. |
| 7:48PM |
4 |
Hangup Issues on TDM40B FXO Australia |
| 3:47PM |
0 |
Allied Telesyn AT-VP504E and asterisk |
| 2:45PM |
3 |
more than one company hosting their PBX on the same machine? |
| 2:45PM |
0 |
Using patch -p0 <meetme-diff-cbmysql_1.txt produces 'malformed patch' message |
| 1:47PM |
0 |
Digium and IPsando announces agenda for Astricon Europe - register now! |
| 1:37PM |
0 |
Trouble using two Fritz ISDN cards in one machine |
| 1:31PM |
2 |
asterisk with vonage linksys adapter? |
| 12:59PM |
1 |
Polycom IP600 Questions |
| 10:09AM |
2 |
(another) cisco 7960 question |
| 9:19AM |
1 |
asterisk-oh323: Max simultaneous calls ? |
| 8:22AM |
1 |
Asterisk Project Consultant/Parner Wanted |
| 8:18AM |
0 |
Fax and Voice VoIP services |
| 6:49AM |
2 |
2 Asterisk boxes sharing dial plans. |
| 6:25AM |
0 |
Pri doesn't accept Zap/g2 to call |
| 6:16AM |
0 |
Questions about TE410P card |
| 5:55AM |
4 |
Getting a Cisco gateway to work with Asterisk |
| 5:12AM |
0 |
*@home 1.0 FWD inbound problems, 2 calls generated |
| 3:53AM |
1 |
Upgrade cause's no Audio on IAX |
| 1:50AM |
1 |
realtime excessive database queries |
| |
| Saturday May 21 2005 |
| Time | Replies | Subject |
|---|
| 9:23PM |
1 |
IP header Bandwidth Reduction |
| 8:25PM |
0 |
IAX provider using Broadvox's network? |
| 8:15PM |
0 |
Re: failure notice |
| 6:34PM |
2 |
realtime app data formatting |
| 5:20PM |
2 |
IAXTEl down |
| 1:26PM |
4 |
having asterisk play music on hold to callers while phone rings? |
| 12:41PM |
2 |
Working Xten, Asterisk, double-NAT configs out there? |
| 11:53AM |
1 |
Asterisk on NetBSD |
| 11:32AM |
1 |
Uncommon callback |
| 9:59AM |
1 |
Affecting overhead with Runlevel? |
| 9:01AM |
0 |
I call an USA MOBILE phone and it is registered as ENUM => failed |
| 8:35AM |
2 |
Spanish Voice Messages |
| 8:35AM |
1 |
Confirmation Of Extension Before Transfer? |
| 4:56AM |
0 |
PRI doesn't call cellphones |
| 3:25AM |
1 |
LiveVoip setup |
| 3:15AM |
0 |
Asterisk-Hylafax |
| 3:14AM |
1 |
Help Understanding ISDN Channels |
| 2:47AM |
0 |
IPSwitchBoard now supports CAPI |
| 1:49AM |
1 |
ISDN data connection through Asterisk |
| 1:00AM |
2 |
Set CallerID in zapata.conf with QuadBri or other solution with parallel call signalling |
| 12:17AM |
0 |
acd with mysql or ast_data support |
| 12:16AM |
1 |
PSTN->voip/sip echo |
| |
| Friday May 20 2005 |
| Time | Replies | Subject |
|---|
| 11:42PM |
1 |
Voicemail With No Messages? |
| 9:59PM |
1 |
How can you keep agents logged in across a restart? |
| 9:31PM |
4 |
Boosting Internet Bandwidth for VOIP |
| 8:56PM |
0 |
Developer Needed! |
| 7:49PM |
0 |
I am looking for a programmer (VoIP, Linux) |
| 7:27PM |
1 |
Local Testing |
| 7:21PM |
0 |
Annoucement in MeetMe and segmentation fault |
| 3:24PM |
5 |
Who knows where voicepulse has their asterisk servers? |
| 3:00PM |
0 |
Looking for asterisk consultant with H323 configuration experience |
| 1:56PM |
1 |
Dell Poweredge 1850 and Zaptel |
| 1:14PM |
1 |
which cvs versions are being used in production systems? |
| 12:52PM |
2 |
MGCP 1.0 / NCS 1.0 |
| 12:49PM |
3 |
Help with follow me |
| 12:46PM |
0 |
TDM04B and spandsp can't send a fax |
| 12:46PM |
2 |
Dell PowerEdge SC420 for Office Implementati on??? |
| 12:44PM |
0 |
Displayed CallerID on Polycom 500 shows CALLERNAME only |
| 12:19PM |
1 |
Displayed CallerID on Polycom 500 shows CALLER NAME only |
| 12:01PM |
5 |
Dell PowerEdge SC420 for Office Implementation??? |
| 10:44AM |
2 |
How to get in touch with sixTel? |
| 10:22AM |
2 |
Trouble getting a SIP phone to dial out through TE100P |
| 10:08AM |
1 |
MFC&R2 Venezuela with libunicall |
| 10:07AM |
1 |
Valet Parking and SuperValet Parking - back level |
| 9:13AM |
0 |
re:Digital Phones |
| 8:55AM |
0 |
Asterisk RealTime & asterisk configuration files through DBMS |
| 8:22AM |
0 |
RE: [Asterisk-biz] Asterisk at ISPcon |
| 8:08AM |
10 |
Stange question... |
| 8:08AM |
0 |
ToneCommander |
| 7:44AM |
0 |
Auto Answer BEEP |
| 7:42AM |
0 |
Registering with second SIP service causes error every 2 seconds - what is going on? |
| 7:38AM |
1 |
Raw Hangup 69.73.19.178:4569 |
| 7:27AM |
2 |
SecureTelephony |
| 7:14AM |
0 |
Anyone done the Cisco 7960 FW migration path programmatically? |
| 7:14AM |
0 |
X100p cards |
| 7:12AM |
0 |
ZAP/DTMF |
| 7:06AM |
1 |
H.323 Gateway |
| 6:43AM |
4 |
paging thru sipura-841 |
| 6:42AM |
1 |
RDNIS (DNID) Call Routing |
| 6:41AM |
2 |
Polycom takes long time for reboot to access web page |
| 6:34AM |
4 |
Sipura 3000 Question |
| 6:20AM |
0 |
ref: Cisco 7960 question |
| 6:00AM |
1 |
app_meetme2.so does not load due to KRB5 symbol. |
| 5:11AM |
5 |
Newbie on IVR |
| 4:56AM |
2 |
call barring |
| 4:23AM |
1 |
chan_capi error2 |
| 3:27AM |
1 |
Unable to create channel of type 'IAX2' (cause 3) |
| 3:00AM |
0 |
Offloading all user/peer autentication to SER? |
| 2:17AM |
0 |
lookup for extensions on another SIP Proxy |
| 1:49AM |
0 |
Greetings |
| 1:26AM |
0 |
Hint with snom 220 - call pick up |
| 1:13AM |
0 |
why can't my asterisk restart? |
| |
| Thursday May 19 2005 |
| Time | Replies | Subject |
|---|
| 10:53PM |
2 |
NVFaxDetect on Gentoo |
| 8:30PM |
0 |
I am looking for some features, |
| 8:15PM |
2 |
IPswitch cannot delete lines & double lines |
| 7:44PM |
2 |
cisco 7960 question |
| 7:25PM |
0 |
Zaptel on pSeries |
| 6:32PM |
2 |
Voicemail wav49 format problem |
| 6:11PM |
1 |
Alsa and lag |
| 5:16PM |
2 |
MusicOnHold Loudness/Distortion |
| 3:46PM |
2 |
DHCP available? |
| 3:02PM |
3 |
Konftel |
| 2:52PM |
1 |
HasNewVoicemail not being called if user hang up after leaving VM ?? |
| 1:57PM |
2 |
How do you put someone on hold on a zap channel? |
| 1:55PM |
1 |
Asterisk at ISPcon |
| 1:51PM |
0 |
Default time zone for asterisk |
| 12:42PM |
6 |
Boosting Shared Internet Bandwidth for Asterisk |
| 12:37PM |
0 |
Configuring a Grandstream 486 Device with AOL Internet connection |
| 12:02PM |
1 |
no music on hold |
| 12:00PM |
1 |
TE110P without router ??? |
| 11:50AM |
0 |
Phone keypad input not working during "menu's" |
| 11:21AM |
1 |
Re: Grandstream ATA 286 and ilbc (Anton Krall) |
| 11:14AM |
0 |
Re: Asterisk-Users Digest, Vol 10, Issue 154 |
| 11:08AM |
4 |
LOOKING TO HIRE |
| 11:06AM |
1 |
RHEL 3 |
| 10:58AM |
1 |
Expression in Extension |
| 10:08AM |
7 |
Cisco Call Manager & Asterisk for Voicemail |
| 10:02AM |
0 |
Connecting an External Extension |
| 9:46AM |
0 |
Can't make outgoing calls |
| 9:43AM |
1 |
ACD Methods |
| 9:33AM |
1 |
(no subject) |
| 8:45AM |
2 |
Two TDM04 with Poweredge |
| 8:26AM |
0 |
Selling: E100P interface card |
| 8:23AM |
1 |
User cannot dial |
| 7:47AM |
1 |
New IAXy from Digium |
| 7:24AM |
0 |
tdm400p fxo not working |
| 7:14AM |
1 |
chan_capi patch eicon |
| 6:52AM |
1 |
Do Both! :) Re: Telecom SIP termination vs. DS3 |
| 6:50AM |
3 |
Public vs. Private Network |
| 6:09AM |
1 |
OT: carrying a router, firewall, switch, ser ver, some phones with me on flight to Europe |
| 6:03AM |
5 |
MusicOnHold probelms |
| 6:02AM |
1 |
Random Blip |
| 5:51AM |
5 |
Deleting Monitor Files After 2 Months |
| 5:38AM |
1 |
retail unit for cards |
| 4:56AM |
3 |
asterisk-oh323 build problems |
| 4:55AM |
0 |
asterisk-oh323 building problems |
| 4:12AM |
1 |
Manager Port |
| 3:53AM |
1 |
Asterisk real time extensions problem... |
| 2:54AM |
1 |
Newbie X100P question |
| 2:54AM |
1 |
GOTO statement in Realtime-Extensions not working like expected |
| 2:15AM |
2 |
Forbidden - wrong password on authentication for NOTIFY |
| 1:26AM |
1 |
ser+asterisk problem |
| 1:16AM |
0 |
dail out with SIP through a second server |
| |
| Wednesday May 18 2005 |
| Time | Replies | Subject |
|---|
| 11:16PM |
0 |
Telecom SIP termination vs. DS3 |
| 10:51PM |
2 |
Spanish TTS |
| 10:50PM |
1 |
Grandstream ATA 286 and ilbc |
| 10:39PM |
0 |
Cisco ATA question |
| 9:55PM |
4 |
OT: carrying a router, firewall, switch, server, some phones with me on flight to Europe |
| 9:38PM |
0 |
MeetMe -1 return Code - howto |
| 9:25PM |
0 |
Tellabs Consultant Wanted |
| 9:18PM |
0 |
FCC Will Force VOIP E911 in 120 days ? |
| 7:43PM |
1 |
Most stable HEAD |
| 7:42PM |
1 |
Invalid sip contract |
| 7:24PM |
5 |
SIP Phone Recommendations? |
| 7:23PM |
0 |
Missing Transfer Command (asterisk CVS 20050518) |
| 6:26PM |
1 |
realtime versus static |
| 6:01PM |
0 |
send a text message from a phone get - 405method not allowed error |
| 5:55PM |
0 |
asterisk hung up the line after 10 minutes rightafter a beep beep beep sound |
| 5:00PM |
0 |
FW: No ASA statistics from call queue and CTI screen pops. |
| 4:42PM |
1 |
Extensions Issues. |
| 3:54PM |
2 |
FREE music for downloading |
| 3:49PM |
4 |
Pickup other ringing phone |
| 3:46PM |
0 |
Ideal Machine |
| 3:06PM |
2 |
Run Script when originator hangs up the phone |
| 2:19PM |
1 |
Follow Me solution |
| 1:59PM |
0 |
Asterisk and H323 vs OH323??? |
| 1:53PM |
4 |
Outbound dialing issue with FXO |
| 1:52PM |
1 |
need 7960 power cubes |
| 1:50PM |
1 |
Nearing my wits end....bad switch??? |
| 1:36PM |
0 |
Zaptel on Dell Poweredge 1850 with RH Kernel 2.4.21-15 |
| 1:22PM |
0 |
Forward calls from PSTN to PSTN very choppy |
| 12:30PM |
1 |
asterisk hung up the line after 10 minutes right after a beep beep beep sound |
| 12:25PM |
4 |
FXO Gateways |
| 12:25PM |
1 |
send a text message from a phone get - 405 method not allowed error |
| 12:22PM |
2 |
Grandstream GXP-2000 and good support |
| 11:57AM |
7 |
Soft Phone |
| 11:52AM |
1 |
PBX integration call status-Calls do not show as connected |
| 11:13AM |
3 |
Recommend a good SOHO NAT Router |
| 11:11AM |
0 |
Call classification with Asterisk |
| 10:46AM |
0 |
question about VoIP headsets used by other call centers |
| 10:39AM |
1 |
No ASA statistics from call queue and CTI screen pops. |
| 10:36AM |
1 |
Issues with Polycom 1.5.2 |
| 10:23AM |
0 |
SIP: Failed to authenticate |
| 10:13AM |
1 |
Agent Queues and Sending URLs |
| 9:46AM |
3 |
connecting a sipura sip device to asterisk before dialing any digits |
| 8:43AM |
1 |
Small office setup with Asterisk @home, IAX and analog termination |
| 8:30AM |
0 |
[Asterisk-Dev] Re: SigSeg in channel.c / chan_mISDN problem ? |
| 8:18AM |
0 |
ASTERISK-SIPP |
| 8:05AM |
0 |
Re: SigSeg in channel.c / chan_mISDN problem ? |
| 8:03AM |
0 |
Integrating Asterisk into our Legacy PBX <-- Newb (correction) |
| 7:53AM |
2 |
Best Compression Available |
| 7:22AM |
0 |
Softphone Requirements |
| 7:17AM |
0 |
RTFriendsCache=yes help Voicemail MWI help |
| 7:12AM |
1 |
Mysql cmd with Asterisk Problems |
| 6:19AM |
1 |
Audio flutter on OH323 output? |
| 6:07AM |
0 |
Integrating Asterisk into our Legacy PBX <--Newb |
| 6:01AM |
0 |
listening at 5070 |
| 5:42AM |
5 |
Polycom Instant Messaging |
| 5:22AM |
1 |
SIP/nat situation |
| 5:16AM |
2 |
Traffic shaping for IAX and SIP calls through Asterisk? |
| 4:31AM |
0 |
Asterisk not recognising "On Hold" |
| 4:24AM |
2 |
Asterisk and Ericsson PBX |
| 4:01AM |
1 |
Asterisk H323 Trunk Zone |
| 3:57AM |
0 |
IVR/Voicemail, No Sound from Asterisk |
| 3:54AM |
0 |
Asterisk with Intel modems 537 or MD3200 |
| 3:49AM |
2 |
FWD to Asterisk stops after 3 seconds |
| 3:38AM |
1 |
2 x Eicon BRI ISDN devices (UK) |
| 3:34AM |
3 |
DHCP, PoE, FXS, FXO and ONE power adapter ONLY??? |
| 3:31AM |
0 |
HELP ME!!!! Asterisk don't do calls |
| 3:26AM |
2 |
Call forwarding... |
| 3:21AM |
1 |
eicon fdc3 |
| 2:34AM |
6 |
zaphfc troubles |
| 2:28AM |
0 |
find free e1 channel |
| 12:23AM |
2 |
DEBUG output on sip extensions |
| |
| Tuesday May 17 2005 |
| Time | Replies | Subject |
|---|
| 11:07PM |
2 |
Ubuntu Migration |
| 10:51PM |
1 |
OT: Multi-Format Sound Conversion Utility (and NOT sox, etc) |
| 10:15PM |
0 |
can't compile zaptel |
| 9:37PM |
10 |
VoiPSupply Dot Com |
| 9:23PM |
0 |
SIP, NAT and Asterisk |
| 9:19PM |
0 |
can't compile zaptel.. |
| 9:16PM |
0 |
Asterisk Aborted |
| 9:13PM |
0 |
Debugging voice cutoff problems |
| 9:08PM |
3 |
Guest |
| 8:55PM |
2 |
fax soft client |
| 8:49PM |
2 |
how to get remote extensions to work correctly with a zap channel? |
| 6:53PM |
1 |
Display SIP useragents |
| 6:14PM |
0 |
Music On Hold problem: Read 392 bytes ofaudiowhile expecting 1600 |
| 5:37PM |
0 |
Linejack |
| 5:24PM |
18 |
VoipSupply.com |
| 4:50PM |
0 |
Agent Queues/XTen X-Pro/Multiple Call Appearance |
| 4:20PM |
1 |
Agent Login/Logout |
| 4:10PM |
2 |
Asterisk - Spandsp: fax header |
| 3:49PM |
0 |
Dropped calls with TDM400P - 4 FXO |
| 2:34PM |
1 |
Music On Hold problem: Read 392 bytes of audio while expecting 1600 |
| 2:13PM |
0 |
Can't connect to SIP provider |
| 2:02PM |
1 |
call waiting signal |
| 11:01AM |
3 |
How much CPU power needed for asterisk |
| 10:19AM |
2 |
Compile Error - MySql addon |
| 9:59AM |
4 |
multiple sip accounts from same sip registrar |
| 9:43AM |
0 |
PRI Providers in San Francisco? |
| 9:34AM |
0 |
cdr from operator initiated calls |
| 9:31AM |
0 |
overlapdial timeout [bristuff] |
| 9:04AM |
0 |
Help on ODBC & Asterisk usage |
| 8:52AM |
11 |
Asterisk Fax |
| 8:45AM |
0 |
Asterisk and rfc2833 |
| 8:42AM |
1 |
callgroup and callwaiting for IAX clients |
| 8:37AM |
3 |
Call Forwarding / Redirect with PRI |
| 8:37AM |
0 |
asterisk tapi |
| 8:25AM |
1 |
fdc3 no gsm |
| 8:07AM |
0 |
no audio for voicemail |
| 8:05AM |
1 |
Digium and Asterisk |
| 8:00AM |
1 |
One * server unavailable when multiple servers connected together |
| 7:45AM |
4 |
Is SKYPE a threat or should we do something (together) |
| 7:42AM |
1 |
Asterisk and rfc2833 help |
| 7:33AM |
4 |
peering with friendly networks, ... |
| 7:13AM |
0 |
Asterisk Realtime extensions configuraton.. |
| 6:51AM |
0 |
jitterbuffer stability and use with meetme |
| 6:49AM |
0 |
Asterisk + digium |
| 6:16AM |
0 |
Failed to grab lock, trying again... |
| 6:01AM |
1 |
Obtaining Cisco Firmware painlessly? |
| 6:00AM |
0 |
Problem with getting the value of variable DIALSTATUS in AGI script |
| 5:59AM |
0 |
Background AGI |
| 5:55AM |
0 |
"Failed to grab lock, trying again..." |
| 4:47AM |
1 |
spandsp + HFC poor fax quality? |
| 4:32AM |
0 |
SMS & Grandstream ATA-286 |
| 3:58AM |
1 |
sip show registry empty ?!?!!? |
| 3:50AM |
0 |
Using US Robotic router for 60 calls |
| 3:35AM |
1 |
Background() problem (with queue(), etc.) |
| 3:09AM |
0 |
Asterisk with PINs |
| 3:02AM |
2 |
Asterisk and Credit Card Machines |
| 3:00AM |
0 |
does NOT rtptimout work configrued localy for a peer ??? |
| 1:27AM |
0 |
Junk at the beginning, Warning, flexibel ratenot heavily tested! |
| 1:07AM |
2 |
Junk at the beginning, Warning, flexibel rate not heavily tested! |
| |
| Monday May 16 2005 |
| Time | Replies | Subject |
|---|
| 10:14PM |
0 |
Asterisk's clients logon failed if asterisk cannot register on its own sip proxy |
| 10:13PM |
2 |
Telephony keypad |
| 8:55PM |
4 |
Web Client with IAX2 and ilbc |
| 8:08PM |
0 |
Using prepaid calling cards to dial out with Asterisk - extensions.conf |
| 7:50PM |
1 |
Warning[3817] and REGISTER |
| 6:30PM |
0 |
DTMF asterisk-2-asterisk using SIP w/ dtmfmode=rfc2833 |
| 6:21PM |
2 |
Asterisk and a D/42NS |
| 6:08PM |
1 |
GSM bandwidth |
| 5:51PM |
2 |
Help with extensions - can't dial 700 |
| 5:43PM |
3 |
CLI and DNIS presented to Analog extension |
| 5:00PM |
1 |
Callerid not passing across IAX2 trunk |
| 3:36PM |
11 |
H323 to SIP |
| 3:31PM |
0 |
Queued calls |
| 3:16PM |
0 |
FW: Static on TDM Zaptel FXO |
| 3:13PM |
1 |
Dial plan - does not stop after first match |
| 2:49PM |
0 |
spandsp in 64 bit Linux on AMD64 |
| 2:04PM |
10 |
Static on TDM Zaptel FXO |
| 1:39PM |
4 |
Forwarding To Cell Phones with Asterrisk PBX |
| 1:25PM |
2 |
Transfer of Calls Between Legacy PBX and Asterisk |
| 1:06PM |
0 |
Using PAP2 with g723 |
| 12:53PM |
0 |
Outbound Faxes with spandsp |
| 12:40PM |
0 |
mysql debug |
| 12:25PM |
2 |
Pass variable to Authenticate? |
| 12:20PM |
0 |
Asterisk Fax On Demand using SPANDSP? |
| 12:05PM |
3 |
voicemail.conf from DB |
| 11:34AM |
2 |
outlook express intregation |
| 11:02AM |
2 |
Broadvoice Toll-Free IVR issues |
| 10:53AM |
3 |
Error running Make config on Debian Sarge |
| 10:39AM |
1 |
ShoreTel 210 MGCP phone drops calls with MGCP RSIP |
| 10:33AM |
1 |
Setting DID info for analog Zap channels |
| 10:11AM |
4 |
IAX jitter |
| 10:04AM |
0 |
WIP-5000 SIP Settings |
| 10:02AM |
4 |
Lucent TNT & ASTERISK |
| 9:34AM |
0 |
Asterisk - fax - spandsp <--older threadlet from Jean-Yves about fax corruption, *not* timing |
| 7:51AM |
1 |
Vonage users with Asterisk in UK? |
| 7:40AM |
5 |
xbox asterisk? |
| 7:33AM |
0 |
.call file |
| 7:24AM |
1 |
problems with asterisk starting from init.d |
| 7:06AM |
3 |
Need off-the-shelve PC for Asterisk Server |
| 6:41AM |
1 |
2 servers via PRI |
| 5:41AM |
2 |
Broadvoice: No Service, No Email reply but charging the credit card still works |
| 5:18AM |
3 |
cisco 3620 setup (newbie cisco alert) |
| 5:12AM |
0 |
IPS can now print and chartc |
| 5:06AM |
2 |
NAT and sip issues |
| 5:05AM |
1 |
pickup timeout |
| 4:29AM |
1 |
Always Ringing |
| 3:37AM |
0 |
chan_misdn and passive BRI cards |
| 3:35AM |
1 |
SIP-->h323 conversion |
| 3:27AM |
1 |
zaptel.conf in /etc not /etc/asterisk - historical reason? |
| 3:19AM |
2 |
callback problem |
| 3:14AM |
4 |
Asterisk@home 1.0 + Sipgate UK/SIP Provider |
| 2:31AM |
1 |
res_config_mysql.so relocation error |
| 2:09AM |
0 |
Number Portability Details |
| 1:23AM |
1 |
Re: SpanDSP TXFax and multipage faxes problems (aditional info) |
| 12:45AM |
1 |
A hook flash sent using RTP for telephony signals (RFC2833) does not flash zap channel |
| 12:08AM |
0 |
Re: Asterisk-Users Digest, Vol 10, Issue 117 |
| |
| Sunday May 15 2005 |
| Time | Replies | Subject |
|---|
| 10:34PM |
1 |
Custom SIP messages |
| 9:56PM |
0 |
Bridge stops bridging channels SIP |
| 8:43PM |
0 |
Re: Asterisk-Users Digest, Vol 10, Issue 114 |
| 8:26PM |
5 |
zttest |
| 8:21PM |
0 |
Hang up error: Didn't get a frame from channel |
| 7:59PM |
0 |
Bridge stops bridging channels |
| 7:57PM |
1 |
Old DBGet/DBPut vs. new Set(var=${DB(... |
| 7:19PM |
14 |
POE hub |
| 7:18PM |
0 |
Multiple Questions -- Please Help |
| 6:26PM |
2 |
Do sipura 200 and linksys pap2 ATAs send their mac address in REGISTER message? |
| 5:52PM |
1 |
Modprobe wctdm hang at command prompt |
| 5:45PM |
0 |
No Such host - IAX2 channel problem |
| 5:29PM |
4 |
Callerid on PC and more |
| 4:53PM |
5 |
FXO/FXS suggestions: |
| 3:58PM |
1 |
Compile problem on last CVS |
| 3:20PM |
1 |
can't CLI> STOP NOW by zombie MOH |
| 2:23PM |
3 |
knopsterisk |
| 2:01PM |
0 |
Known Working Motherboard/CPU for TE410P |
| 12:55PM |
1 |
Scalability of chan_oh323 |
| 12:08PM |
1 |
Re: SpanDSP TXFax and multipage faxes problems (aditional info) |
| 11:52AM |
1 |
Asterisk@home backup/restore question |
| 11:43AM |
0 |
Several questions. Please help |
| 11:22AM |
2 |
Road Warrior phone config |
| 11:17AM |
0 |
SIP or IAX2 Web UA |
| 11:14AM |
4 |
Outgoing spool file ignored |
| 11:02AM |
2 |
SIP Gerenal settings conufsion |
| 6:24AM |
1 |
Re: SpanDSP TXFax and multipage faxes problems (aditional info) |
| 4:40AM |
0 |
AreskiCC doesn't log in |
| 4:30AM |
2 |
Voip Provider in Brazil |
| 3:36AM |
1 |
Problem with extensions and when channel is unavailable |
| 12:22AM |
5 |
AreskiCC |
| |
| Saturday May 14 2005 |
| Time | Replies | Subject |
|---|
| 11:29PM |
6 |
ASTCC does not count all calls |
| 8:58PM |
2 |
Asterisk Guru help needed for DISA troubles |
| 4:44PM |
5 |
ser and asterisk |
| 3:34PM |
0 |
Cannot create a personalized unavailable message |
| 10:21AM |
0 |
Building OPENH.323 ERROR HELP PLEASE |
| 9:50AM |
2 |
Broadvoice outage times? |
| 8:18AM |
0 |
Transferring a call, IAX2->SIP, DTMF/RFC2833 doesn't work? |
| 6:56AM |
0 |
pbx autodiscovery |
| 6:02AM |
1 |
Help Please Multiple Users for Broadvoice |
| 5:58AM |
0 |
installing linksys pap2 and welltech lp302 |
| 1:02AM |
2 |
How to connect two Asterisk servers |
| 12:15AM |
0 |
How to beark Queue() and jump to voicemailMain |
| |
| Friday May 13 2005 |
| Time | Replies | Subject |
|---|
| 11:11PM |
1 |
IAX2 and FWD - Wrong context? |
| 7:13PM |
1 |
A@H Email Relay |
| 6:57PM |
4 |
Polycom configuration |
| 6:42PM |
0 |
asterisk dials random number when receiving incoming call |
| 5:36PM |
3 |
Audio quality |
| 5:06PM |
0 |
Echo problem on SPA-841 |
| 4:07PM |
4 |
1-800 with FWD |
| 3:54PM |
1 |
Fax service (instead of tdm card) |
| 3:12PM |
1 |
CVS HEAD - FATAL: Error inserting wctdm |
| 2:03PM |
0 |
Caller*ID failed checksum? |
| 1:20PM |
0 |
Spawn extension -----what does this mean ? |
| 1:17PM |
0 |
Formatting problem in cmd sip show peers |
| 1:14PM |
3 |
Other memory stuff |
| 1:12PM |
1 |
MusicOnHold "zombie" mpg123 processes |
| 12:54PM |
0 |
wanted - asterisk reseller / integration consultant in NC |
| 12:15PM |
2 |
TDMoE emulates a T-1= Is there a product to simulate a PRI trunk? (Robert Goodyear) |
| 11:56AM |
1 |
Polycom IP 500 caller id |
| 11:53AM |
3 |
Poor volume on SPA-2100 due to asterisk? |
| 11:38AM |
6 |
64 bit |
| 11:36AM |
1 |
DTMF problems with International Calls |
| 11:05AM |
5 |
Is there a product to simulate a PRI trunk? |
| 10:51AM |
6 |
voip encryption options |
| 10:51AM |
0 |
Re: Interrupting voicemail with "*", dropping to "a" |
| 10:28AM |
4 |
Asterisk - fax - spandsp |
| 9:48AM |
0 |
Chanspy crash |
| 9:02AM |
1 |
broadvoice replacement |
| 8:58AM |
0 |
delay before call file execution |
| 8:25AM |
0 |
My experience with our VS-1 Asterisk server |
| 8:17AM |
0 |
Dropped Calls between Sip and Zaptel |
| 8:16AM |
1 |
Tyan Transport GX28 with TDM400 |
| 8:08AM |
1 |
Why always getting "max retries" error during idle? |
| 8:07AM |
0 |
ISDN passive card (HiSAX driver) / Fax reciever |
| 8:03AM |
0 |
Zaptel and zttest |
| 7:13AM |
0 |
Extension never ring, goes straight to VM |
| 6:44AM |
0 |
Autodial and autoanswer |
| 6:26AM |
2 |
Asterisk extensions from Mysql |
| 5:38AM |
2 |
In/out calls from/to same sip provider |
| 5:09AM |
1 |
Help needed on setting up realtime |
| 4:13AM |
1 |
ASTCC Compilation Error |
| 3:56AM |
0 |
Unchanged sound through Asterisk |
| 3:49AM |
1 |
Re: SpanDSP TXFax and multipage faxes problems |
| 2:17AM |
0 |
Problem with IAX trunking |
| 1:30AM |
0 |
[Asterisk-Dev] Re: oh323 compile problem in FreeBSD |
| 1:27AM |
3 |
2 minutes pause before ring on H323 channel |
| 12:50AM |
2 |
About Voip Technology : RTP over TCP |
| 12:00AM |
0 |
Problem with calls on hold |
| |
| Thursday May 12 2005 |
| Time | Replies | Subject |
|---|
| 10:55PM |
1 |
sipsak with asterisk |
| 10:34PM |
14 |
voipjet anyone? |
| 9:43PM |
0 |
Asterisk, SIP and NAT: Help needed! |
| 9:16PM |
0 |
Voicemails not deleting |
| 7:32PM |
1 |
Can the originator of a call transfer it? |
| 7:01PM |
1 |
realtime sip show peers no nat |
| 6:24PM |
2 |
ISDN Clock Source |
| 6:22PM |
2 |
Polycom IP4000 |
| 6:15PM |
1 |
Queue/Agent recording and configuration |
| 5:51PM |
2 |
UNREACHABLE messages |
| 5:26PM |
3 |
Dead Polycom ip500 |
| 5:05PM |
5 |
1-800 free calls |
| 4:26PM |
3 |
Interrupting voicemail with "*", dropping to "a" extension. Does it work? |
| 3:52PM |
5 |
French SIP or IAX phones |
| 3:47PM |
0 |
Chicago users/implementations |
| 3:16PM |
1 |
IPVolution release info.... |
| 3:05PM |
0 |
Fix for increasing delay over time on non-Zap channels in MeetMe |
| 2:25PM |
0 |
FW: Incoming calls picked-up then simply hanged-up |
| 2:20PM |
0 |
SER Asterisk and NAT |
| 2:18PM |
2 |
Problem with Polycom SP 500 and Cisco PIX |
| 2:16PM |
4 |
Sound card Line-In as MOH source |
| 2:13PM |
0 |
Incoming calls picked-up then simply hanged-up |
| 1:19PM |
1 |
Asterisk with ShoreTel 210 (MGCP) |
| 1:04PM |
0 |
One sided sound outgoing only |
| 12:57PM |
0 |
Escape context and queue application |
| 12:52PM |
1 |
Re: Headset for Cisco 7960? |
| 12:43PM |
3 |
Giving user progress in an voice menu system |
| 12:42PM |
0 |
"Called" ID question - Trying again |
| 12:21PM |
2 |
VoiceXML |
| 12:17PM |
0 |
SIP authentication on outgoing call |
| 11:50AM |
0 |
"Called" ID on local extensions |
| 11:36AM |
2 |
AMP and dialparties.agi |
| 10:54AM |
2 |
Problems with Simpletelecom and * |
| 10:51AM |
4 |
Polycom Bootrom 2.6.2 and SIP 1.5.2 |
| 10:37AM |
3 |
How to decrease Asterisk load |
| 10:30AM |
3 |
* Server |
| 10:17AM |
3 |
Something every TDMP user should know |
| 10:10AM |
0 |
Asterisk as a fax/voice switch |
| 9:26AM |
2 |
Cisco 7960 Can't be unlocked |
| 9:21AM |
2 |
IAX to FWD? |
| 8:53AM |
2 |
Best CPU config for dual-Xeon? |
| 8:49AM |
1 |
cdr! |
| 8:47AM |
2 |
Immediate Answer |
| 8:39AM |
1 |
Incoming context problem |
| 8:27AM |
2 |
GSM gateway for Asterisk |
| 8:26AM |
2 |
Voice Recognition - Cases of success |
| 8:24AM |
0 |
switch in extensions.conf |
| 8:10AM |
4 |
gnugk |
| 7:45AM |
0 |
${BLINDTRANSFER} variable |
| 7:38AM |
0 |
Cellsocket with @home |
| 7:20AM |
5 |
chan_capi, chan_misdn and chan_modem |
| 7:00AM |
2 |
SIP and FastStart |
| 6:43AM |
2 |
Inbound ANI & DNIS format |
| 6:38AM |
0 |
Open Source MGCP Softphone |
| 6:21AM |
1 |
chan_capi and chan_misdn |
| 5:53AM |
1 |
FW: failure notice |
| 5:48AM |
1 |
ast_yyerror - 'space' in Caller-ID - string comparison |
| 5:07AM |
0 |
Show useragents? |
| 5:03AM |
0 |
delay before execution of call file |
| 4:59AM |
0 |
Connecting * to a PBX throught a PRI. |
| 4:42AM |
5 |
VoiceBlue GSM |
| 4:32AM |
0 |
Making Asterisk run on Mysql backend |
| 3:25AM |
1 |
Wrong password on authentication for Notify |
| 3:13AM |
6 |
Cisco contract for 7940/7960 firmware access |
| 2:47AM |
0 |
Wrong password on Auth for Notify |
| 2:16AM |
2 |
Voice mail - "Extension at" vs "Phone Number" OGM |
| 1:43AM |
0 |
SV: beginner in Asterisk configuration |
| 1:17AM |
0 |
Snap, Crackle and Pop with Dell 1850 and TE410P |
| 12:58AM |
5 |
beginner in Asterisk configuration |
| 12:52AM |
2 |
GXP 2000 Conference Button and ILBC |
| |
| Wednesday May 11 2005 |
| Time | Replies | Subject |
|---|
| 11:48PM |
3 |
octtel SP 4220 gateway and Asterisk |
| 11:31PM |
2 |
Icecast |
| 10:41PM |
0 |
Dial out on ZAP channel |
| 10:27PM |
1 |
outgoing-call-logs to a text file |
| 10:23PM |
1 |
HELP: ASTCC (AGI) meets call forward ERROR |
| 9:35PM |
1 |
Anyone ever implement an *outbound* dial-by-name?? |
| 8:44PM |
0 |
TDM400P with 2 FXS module fail to Hangup |
| 8:15PM |
1 |
re:oh323 driver compiling problem |
| 8:03PM |
0 |
digium card and fax |
| 7:56PM |
0 |
Re: Asterisk-Users Digest, Vol 10, Issue 83 |
| 7:34PM |
1 |
?????sip channel & AGI problem |
| 7:31PM |
3 |
Astlinux & AMP |
| 6:19PM |
0 |
Database of actve calls (as per astguiclient) |
| 4:45PM |
1 |
Echo from a mail loop in list |
| 3:26PM |
0 |
Seshu, on April 20, you said this about the Astcc & AreskiCC --> http://lists.digium.com/pipermail/asterisk-users/2005-April/102710.html Re: AreskiCC installing assistance for seshu.kanuri @ MorganStanley.com |
| 3:15PM |
1 |
New provider needed - any recommendations |
| 2:59PM |
1 |
Asterisk Video Conferencing Bounty bumped to $3, 000 |
| 2:51PM |
3 |
Grandstream GXP2000 firmware update |
| 2:15PM |
4 |
Problems with VIA Chipset |
| 2:12PM |
12 |
Snom 360 |
| 2:07PM |
0 |
TDMoE vs IAX2 |
| 2:01PM |
0 |
dialparties.agi and @home |
| 1:57PM |
0 |
Mediatrix 1204 caller ID |
| 1:55PM |
5 |
IAX.CC/SixTel |
| 1:50PM |
0 |
Fw: pinout for"standard"telephoneheadsetrequired.? |
| 1:26PM |
0 |
Vegastream assistance? |
| 1:20PM |
1 |
high availibilty (heartbeats) - a good way to ensure automatic redundency? |
| 1:18PM |
5 |
Status of FAX |
| 1:05PM |
0 |
outbound proxy field in sip.conf |
| 1:02PM |
0 |
softphone buzzing |
| 12:52PM |
1 |
Asterisk @home with IAX termination... |
| 12:42PM |
1 |
Forcing Asterisk to not bridge/transcode RTP traffic |
| 12:39PM |
2 |
AreskiCC Install Problems |
| 12:33PM |
1 |
AreskiCC - Install Problems |
| 11:56AM |
1 |
ITSPs with good phone support |
| 11:19AM |
1 |
IAX and calls on hold |
| 11:13AM |
0 |
snom190 and SUBSCRIBE failures with 407 |
| 10:34AM |
0 |
Sipgate incoming DTMF |
| 10:23AM |
7 |
Satellite Providers |
| 9:59AM |
0 |
Audio delays during file playback and zap channel activity |
| 9:51AM |
2 |
PRI QSIG and legacy toshiba intergration |
| 9:48AM |
0 |
Inbound Calls Codec |
| 9:41AM |
1 |
Trouble Connecting Xlite to Asterisk |
| 9:26AM |
0 |
Mass Deployment |
| 9:25AM |
1 |
Channelized T-1 (NOT PRI) Voice and IP mixed |
| 8:53AM |
3 |
Live Voip |
| 8:42AM |
1 |
oh323 driver compiling problem. |
| 8:33AM |
2 |
Realtime voicemail login incorrect |
| 8:08AM |
1 |
CDR and Postgres |
| 7:47AM |
0 |
Outgoing calls log in a text file |
| 7:35AM |
0 |
[SPAM] - RE: Grandstream-Budge tone - Email found in subject |
| 7:27AM |
0 |
wip 5000 and using write msg on the phone - anyone? |
| 6:57AM |
1 |
Grandstream-Budge tone |
| 6:57AM |
5 |
Voicepulse down? |
| 6:52AM |
0 |
T1 Card ------ Adtran ------- FXS BUG??? |
| 6:44AM |
2 |
forum www.asterisk-italia.it |
| 3:16AM |
1 |
Gateway service under Asterisk |
| 2:23AM |
2 |
Asterisk and Cisco AS5300 or 3600 |
| 2:07AM |
0 |
TDM400P for UK |
| 2:06AM |
2 |
Sip or IAX2 eb Client |
| 1:16AM |
0 |
Predictive Dialier |
| 12:50AM |
2 |
Log Output |
| 12:18AM |
0 |
how to detect a hang-up in the first 5 seconds |
| 12:14AM |
0 |
SIPURA SPA-2000 webserver dead after firmwareupgrade |
| |
| Tuesday May 10 2005 |
| Time | Replies | Subject |
|---|
| 10:54PM |
0 |
rtp.conf not working as expected |
| 10:43PM |
1 |
asterisk-addon |
| 9:31PM |
3 |
is it allowed to install 2 TE405P cards at same P.C.? |
| 8:21PM |
0 |
VoIP A-Z Carriers |
| 7:33PM |
1 |
Call Queue Priorities |
| 7:28PM |
1 |
Restricting connection of unauthorized phones. |
| 6:50PM |
4 |
SIPURA SPA-2000 webserver dead after firmware upgrade |
| 6:16PM |
1 |
MF and DTMF tones in the same AGI script |
| 6:13PM |
3 |
Voicemail Passwords |
| 5:55PM |
2 |
RE: Writing To Multiple MySql Tables |
| 5:47PM |
13 |
What do you name yours |
| 5:06PM |
0 |
Registering phones with the same/invalid extension number |
| 4:53PM |
0 |
Transfer from/to a queue |
| 3:55PM |
3 |
Setting Variables |
| 3:38PM |
3 |
Sizing a machine |
| 2:52PM |
3 |
Phone attached to Sipura SPA-1001 has no ring |
| 2:15PM |
0 |
AstLinux 0.2.6 Released! |
| 1:26PM |
0 |
Flashing DVG-1120M to DVG-1120S |
| 12:47PM |
2 |
Warning of the Asterisk server |
| 12:40PM |
0 |
problem with ControlPlayback |
| 12:34PM |
3 |
Asterisk and Avaya 4602 SIP phone |
| 12:09PM |
2 |
AAH 0.9 |
| 12:08PM |
2 |
DS3 (T3) Card for Asterisk? |
| 12:06PM |
1 |
Limiting outbound calls |
| 11:52AM |
1 |
Zaptel problems on Debian |
| 11:46AM |
0 |
Re: Sipura 841 and headset (Josiah Bryan) |
| 11:44AM |
1 |
Asterisk PRI problems (Crashing when full) |
| 11:41AM |
0 |
outbound PSTN numbers over SIP failing |
| 10:10AM |
0 |
Incoming 800 Number |
| 9:51AM |
2 |
DISA |
| 9:50AM |
2 |
Manoj Shetty is out of the office. [Email checked- EMEA] |
| 9:38AM |
2 |
Cisco 837 router config |
| 9:25AM |
1 |
Spa3000 doesn't hangup after a conversation |
| 9:23AM |
0 |
how to get extension for ivr |
| 9:14AM |
0 |
ISDNguard |
| 8:53AM |
0 |
IPSwitchBoard version 0.115 |
| 8:52AM |
0 |
extensions logon failed problem |
| 8:08AM |
1 |
AreskiCC + MySQL |
| 8:07AM |
0 |
AGI (LCR) within AGI ( possible??? |
| 7:55AM |
2 |
Stun & codec |
| 7:34AM |
0 |
Cisco IP Phone 7912 |
| 7:32AM |
3 |
MGCP : chan_mgcp.c:1509 find_subchannel |
| 7:13AM |
0 |
zt_rbs errors!?! never seen before. |
| 6:56AM |
1 |
Re: E1 (Digium E100P) problem : B-channel succesfully restarted |
| 6:54AM |
0 |
Asterisk Upgrade Path |
| 6:45AM |
2 |
Sipura 841 and headset |
| 6:38AM |
2 |
skype channel |
| 6:16AM |
1 |
Redirect to an application on other asterisk server |
| 5:50AM |
3 |
Interconnecting two lans using Asterisk over a PSTN |
| 5:16AM |
0 |
problem with mysql |
| 4:04AM |
2 |
E1 (Digium E100P) problem : B-channel succesfully restarted. |
| 3:58AM |
2 |
BYE from Cisco gateway |
| 3:38AM |
1 |
SIP transfers failing |
| 3:07AM |
2 |
outsourced pbx functionality- distributing calls evenly amongst agents |
| 2:56AM |
1 |
Problem developing my office |
| 2:40AM |
1 |
Cisco 7912G DST |
| 2:15AM |
0 |
Ericsson FCT f251m and polarity reversal |
| 1:46AM |
1 |
Group dial, first phone cannot pickup call. Cisco 7905 hangs. |
| 1:42AM |
0 |
cvs stable with db support in extensions.conf |
| 1:32AM |
1 |
BRI and PRI together possible? |
| 1:13AM |
0 |
Atcom AT-320 call forwarding - how? |
| 12:34AM |
0 |
spandsp configuration |
| |
| Monday May 9 2005 |
| Time | Replies | Subject |
|---|
| 11:31PM |
1 |
Kphone-->asterisk<--Kphone |
| 11:29PM |
0 |
alcatel voip phone 4038 |
| 11:01PM |
1 |
Asterisk + SER and NAT |
| 9:00PM |
4 |
Multiple Calls with Asterisk? |
| 8:16PM |
0 |
Central Asterisk Server and Asterisk VoIP Gateway |
| 5:48PM |
6 |
livevoip |
| 4:51PM |
0 |
[Feedback request] Web-MeetMe authentication |
| 2:44PM |
0 |
New IAXy available? |
| 1:19PM |
7 |
Will Asterisk do well in this application? |
| 12:58PM |
0 |
Consultants - Sydney Aust |
| 12:24PM |
0 |
Manoj Shetty is out of the office. [Email checked - EMEA] |
| 12:17PM |
1 |
Personal Communications Assistant |
| 11:13AM |
3 |
VOIP/SATELLITE |
| 10:19AM |
0 |
HELP... SER + Asterisk as feature server |
| 10:08AM |
0 |
RE: Asterisk at home with Broadvoice? |
| 9:25AM |
0 |
RE: Asterisk at home with Broadvoice? |
| 9:19AM |
0 |
AstriCon Europe: June 15 - 17 in Madrid Spain |
| 9:18AM |
1 |
Configuring SPA-3000 As A Trunk |
| 9:04AM |
0 |
New script: /usr/bin/asteriskdial + Kontact |
| 8:43AM |
1 |
asterisk lock up |
| 8:36AM |
3 |
ANNOUNCEMENT : AreskiCC V2.2 - Asterisk CallingCard Application |
| 8:12AM |
1 |
Interfacing AT&T Spirit System to Asterisk |
| 7:50AM |
1 |
Caller Name Database |
| 7:48AM |
0 |
SIP and MD5 passwords. |
| 6:17AM |
0 |
2 accounts on one Snom 220 with a queue |
| 6:06AM |
1 |
Asterisk DIDs configuration |
| 5:49AM |
0 |
SV: Re: Sangoma A102 cards testing FIXED |
| 5:45AM |
3 |
qozap(!) problem |
| 5:40AM |
1 |
Cisco ATA 186 with *70 |
| 5:33AM |
2 |
sangoma fdc 3? |
| 5:29AM |
1 |
extension based on a dialed number? |
| 5:21AM |
0 |
Re: Sangoma A102 cards testing FIXED |
| 5:15AM |
0 |
ZAP CHANNEL QUESTION. |
| 3:25AM |
8 |
Connecting 20+ asterisk servers together |
| 2:20AM |
0 |
SV: Re: Sangoma A102 cards testing FIXED |
| 1:49AM |
0 |
transfer queues agents |
| 1:46AM |
0 |
How to config call out pstn by sip in a meetme bridge application? |
| 1:30AM |
0 |
Mediatrix APA III-4FXO configuration |
| 1:28AM |
0 |
How can I send e164 ID to my gatekeeper? |
| 1:19AM |
3 |
Zyxel 2000W (WI-FI) Problems |
| 1:14AM |
0 |
AW: CAPI on ptp with variable length digits inphonenumber: SOLUTION for EICON |
| 12:23AM |
2 |
AGI - How to Make Calls and Bridge to Original Incoming |
| |
| Sunday May 8 2005 |
| Time | Replies | Subject |
|---|
| 11:32PM |
0 |
help needed for PSTN |
| 11:07PM |
2 |
Background command noanswer option |
| 10:25PM |
2 |
Sangoma card ! |
| 8:34PM |
2 |
HELP: how to get "To:" from AGI? |
| 5:52PM |
1 |
Help with Realtime & Seeding |
| 5:16PM |
3 |
Grandstream firmware 1.0.6.2 |
| 5:07PM |
1 |
Cisco Mass Deployment |
| 3:42PM |
0 |
Restrict RTP ports - inbound and outbound? |
| 3:09PM |
5 |
8+ line receptionist only setup |
| 12:42PM |
2 |
detaching console from background asterisk |
| 11:48AM |
2 |
Just added snom Mass Deployment |
| 10:00AM |
4 |
Cellsocket help needed |
| 9:30AM |
2 |
Suggested Reading for VOIP |
| 7:55AM |
0 |
RSA question |
| 5:38AM |
0 |
Heavy CPU Usage During SPEEX Calls |
| 5:26AM |
1 |
RE: Asterisk at home with Broadvoice? |
| 2:39AM |
0 |
AreskiCC installation |
| |
| Saturday May 7 2005 |
| Time | Replies | Subject |
|---|
| 11:48PM |
4 |
Setting variable for a context for all extensions? |
| 11:32PM |
2 |
What is the Polycom 301, 501 & 601? |
| 10:20PM |
0 |
Asterisk@Home on OnComputers Show Sunday morning |
| 8:44PM |
1 |
Setting the jitter buffer in AIX |
| 8:30PM |
0 |
Getting DTMF to work with SIP? |
| 8:04PM |
2 |
At home Asterisk via Broadvoice? |
| 7:32PM |
0 |
Problem Dialing out via external SIP account. |
| 2:55PM |
2 |
Cisco ATA 186 and Asterisk |
| 2:31PM |
0 |
Cisco ATA & Call Waiting |
| 2:08PM |
0 |
Polyco ip600 incoming ring time |
| 1:48PM |
1 |
WIP-5000 and DTMF |
| 12:58PM |
1 |
two questions about the Sipura 841? |
| 12:15PM |
0 |
cant connect |
| 11:29AM |
2 |
h323.conf - Asterisk not routing incoming calls based on IP - Ignoring type=user + host= + context= |
| 9:37AM |
0 |
Termination South America |
| 8:52AM |
2 |
Inexpensive FAX and 800 Number retail service |
| 8:50AM |
0 |
end user gui |
| 4:18AM |
0 |
IAX service provider with account balance announcement |
| 3:11AM |
0 |
DTMF generated from phone or from gateway? |
| 1:58AM |
0 |
DTMF detection with Adit 600 |
| 1:53AM |
0 |
ChanIsAvail for MGCP |
| 1:34AM |
0 |
MWI Suggestion |
| 1:32AM |
5 |
Good NAT Pnp Hardphone |
| 12:08AM |
1 |
Echo Madness |
| |
| Friday May 6 2005 |
| Time | Replies | Subject |
|---|
| 11:46PM |
0 |
Cisco ATA186 Fax problem solved: |
| 9:40PM |
1 |
Qos betwenn WIFI machines in LAN? oh323? |
| 9:22PM |
5 |
Who's happy with their voip service? |
| 6:05PM |
0 |
Setting ANI |
| 4:54PM |
0 |
Migrating to ODBC Voicemail |
| 4:43PM |
1 |
Upgrading to 1.x from 0.7 on Linux |
| 2:31PM |
1 |
Am I on the right track, and consultants |
| 2:16PM |
0 |
Wildcard TE110p initial setup |
| 2:01PM |
0 |
MWI on Cisco 7905 |
| 1:42PM |
1 |
ZapBarge a PRI DDI |
| 11:24AM |
2 |
broadvoice NCFA numbers |
| 11:17AM |
1 |
sendtext to a phone that is off |
| 11:06AM |
0 |
Asterisk 1.0.7 and VIA EPIA |
| 11:02AM |
3 |
Review Outgoing VM Messages |
| 11:00AM |
0 |
Zaptel kernel modules correct? |
| 10:50AM |
0 |
RE: [Asterisk-biz] voip VPN solution requirement |
| 10:13AM |
1 |
Make error on ZT_EVENT_DTMFDIGIT |
| 10:11AM |
2 |
my_zt_write |
| 9:57AM |
0 |
Need your HELP: Avaya SIP phone 4602 and Asterisk |
| 9:29AM |
0 |
IPS version 0.114 |
| 9:26AM |
0 |
WG: Newbie *@home + Xten. |
| 9:19AM |
0 |
To receive faxes on a dedicated extention and to forward them to a dedicated e-mail |
| 9:12AM |
2 |
HINT |
| 8:57AM |
2 |
Newbie *@home + Xten. |
| 8:19AM |
1 |
SIP NOTIFY retries exceeded. |
| 8:08AM |
0 |
ADTRAN Total Access 624 Work??? |
| 8:00AM |
0 |
My Sangoma Experience in Asterisk: Followup |
| 7:50AM |
0 |
anyone experiencing half connections |
| 7:36AM |
0 |
OT: NAT traversal with SIP Paper |
| 7:20AM |
1 |
SEND TEXT to an extension? |
| 6:54AM |
1 |
Polycom 600 rollover |
| 6:46AM |
3 |
Web GUI |
| 6:25AM |
1 |
Operator Monitoring...flash operator panel? |
| 6:17AM |
1 |
CAPI on ptp with variable length digits in phonenumber: SOLUTION for EICON |
| 5:53AM |
1 |
Mitel SX200 integration |
| 5:51AM |
2 |
how do I register my Asterisk with oh323 on gatekeeper? |
| 5:29AM |
3 |
good bri card not junghanns |
| 5:12AM |
2 |
Transparently Routing German pri through Asterisk |
| 5:09AM |
0 |
Chan_misdn - cannot get the channel driver to load... |
| 5:03AM |
4 |
3 x TDM400P in one PC ?? |
| 4:43AM |
1 |
Zapata.Conf Sanity Check |
| 4:04AM |
2 |
CAPI on ptp with variable length digits in phone number |
| 3:59AM |
1 |
oh323 compile problem in FreeBSD |
| 3:40AM |
1 |
DTMF oddity with OH323 |
| 3:35AM |
1 |
Re: Sangoma A102 cards testing FIXED |
| 2:46AM |
0 |
OT: 911 service |
| 1:55AM |
1 |
Is such a thing as a analog (or even IP) video door entry system available? |
| 1:26AM |
1 |
IAX hint |
| 1:11AM |
1 |
IAXy Firmware Upgrade |
| |
| Thursday May 5 2005 |
| Time | Replies | Subject |
|---|
| 11:52PM |
0 |
Polycom IP 600 not ringing |
| 11:35PM |
0 |
sccp transfer question |
| 10:27PM |
1 |
unknown RTP codec 72 |
| 9:38PM |
1 |
Cisco device for gatewaying SIP to H323 suitable for ~50 simultanious calls |
| 8:48PM |
1 |
Connecting to provider |
| 8:32PM |
0 |
cdr_pgsql amaflags are always 3 |
| 8:20PM |
2 |
CNAM lookup: new method for Caller ID Name delivery |
| 6:19PM |
0 |
Fax hangup causes incoming ring to be generated |
| 6:18PM |
5 |
snom mass deployment (probably off topic) |
| 5:08PM |
0 |
Cisco XML Parking Lot |
| 5:00PM |
4 |
Asterisk on Fedora Core 2 startup script |
| 3:19PM |
3 |
1800 DNIS and asterisk (HOW TO?) |
| 3:15PM |
0 |
iaxy dial out automatically |
| 3:09PM |
0 |
Zap Channel: CallerID feed failed |
| 2:57PM |
0 |
What is better? 2 lines of 128kbps or 1 line of 256kbps |
| 2:16PM |
0 |
Silly version question |
| 1:19PM |
3 |
can't create Zap channel |
| 1:19PM |
2 |
7777 (simulate incoming call) not working |
| 1:06PM |
1 |
Any thoughts on why I can't dial out my PRI? |
| 1:04PM |
0 |
How Two Asterisk Boxes Behind A Nat Initiate Calls |
| 12:27PM |
0 |
(res_)Monitor: wav - no sound; wav49 - sound |
| 11:54AM |
0 |
Alepo VoIP Billing with Asterisk |
| 11:52AM |
3 |
Account Code in all cases? |
| 11:51AM |
2 |
Question PSTN->VOIP forwarding and # of inbound calls |
| 11:42AM |
0 |
Voicemail call/dial out notification |
| 10:30AM |
2 |
Sayson caller id |
| 10:24AM |
0 |
Asterisk, Cisco 837 router and 79xx phones |
| 10:23AM |
1 |
Help needed with PSTN line |
| 9:48AM |
0 |
SIP forwarding to ZAP and call files |
| 9:29AM |
0 |
Optimal prompt format (gsm, ulaw, wav) for quality effeciency space |
| 8:52AM |
0 |
IAX2 monitoring |
| 8:35AM |
6 |
Opinions on Cisco 7960G, Polycom IP-600, and Snom 360 |
| 8:29AM |
0 |
Re: Asterisk-Users Digest, Vol 10, Issue 39 |
| 8:26AM |
1 |
MOH per User |
| 7:34AM |
11 |
Broadvoice "Issues" |
| 7:24AM |
0 |
Can I hide caller id on the fly (per eachusesetting) on Bristuffed * and quadbri |
| 7:21AM |
3 |
chan_zap.so: load_module fails: Fedora Core 3: SMP |
| 7:17AM |
1 |
Why switch from Asterisk@Home? was: Re: 7960'multi-line' configuration |
| 6:36AM |
0 |
Can call but not getting voice response with Cisco ATA186 behind nat |
| 6:06AM |
0 |
softphone for ipaq h4350 |
| 5:37AM |
2 |
(OT) Interesting Product Vocera |
| 5:23AM |
2 |
Polycom 300 setup and AMP |
| 5:20AM |
1 |
Problem with Manager Originate and SIP extension |
| 5:17AM |
1 |
Realtime and Asterisk Database |
| 5:07AM |
11 |
FXO ATA? |
| 4:43AM |
1 |
Asterisk + GNUGK |
| 4:22AM |
1 |
test - ignore |
| 4:11AM |
2 |
Did nufone change allowed codecs? |
| 3:32AM |
0 |
problem with H323:Gatekeeper could not find user |
| 2:33AM |
5 |
Registering/Unregistering |
| 2:15AM |
2 |
mpg123 zombie processes ... |
| 2:03AM |
1 |
E1R2 Route |
| 1:48AM |
2 |
PRI debug |
| 1:32AM |
0 |
Compiling with GCC 4 |
| 12:23AM |
2 |
Fritz Card sound quality |
| 12:20AM |
0 |
UniCall bugs |
| |
| Wednesday May 4 2005 |
| Time | Replies | Subject |
|---|
| 11:58PM |
2 |
RED ALARM on PRI channel takes Asterisk DOWN |
| 11:31PM |
1 |
Working exten=> fax... |
| 9:53PM |
0 |
TE410P Drops Calls after many touch tones fromcaller |
| 9:41PM |
4 |
Problem with realtime SIP |
| 9:12PM |
2 |
TE410P Drops Calls after many touch tones from caller |
| 9:04PM |
0 |
max call rate (ingress direction) 1.00/30 |
| 8:15PM |
0 |
Looking for Log parse for CDR's |
| 7:46PM |
2 |
10 digit dialing in Ft Lauderdale, FL? |
| 6:42PM |
0 |
SCCP and channel question |
| 6:23PM |
2 |
Connecting 2 * Together-Pulling hair out |
| 6:20PM |
3 |
QoS for improvements |
| 5:58PM |
0 |
Music on hold for agents and queues |
| 5:25PM |
2 |
Strange problem with G711/G729, Cisco and Grandstream |
| 4:55PM |
5 |
TE410P does not fit in motherboard |
| 4:41PM |
1 |
broadvoice not hanging up |
| 3:43PM |
0 |
Multitech SIP BRI Gateways |
| 2:59PM |
0 |
DTMF callerid does work |
| 2:06PM |
0 |
res_features: builtin monitor probs |
| 1:51PM |
0 |
Running explicit codecs between two hosts using distinct peers? |
| 1:51PM |
0 |
RANDOM command |
| 1:48PM |
0 |
Asterisk and prepaid sip clients. |
| 1:38PM |
0 |
Philips IS3030 Qsig |
| 1:24PM |
5 |
CDR for PSTN |
| 1:10PM |
4 |
Can I hide caller id on the fly (per each use setting) on Bristuffed * and quadbri |
| 1:03PM |
4 |
Voice mail Greetings |
| 1:02PM |
0 |
Passing CallerID outbound |
| 12:13PM |
0 |
parking in a specific parking spot with Asterisk 1.0X Is it possible???? |
| 12:11PM |
1 |
FXO ports |
| 12:11PM |
7 |
TE410P on Dell 2650 |
| 12:05PM |
2 |
IP500 Registration |
| 11:23AM |
0 |
TxFax and Tiffs |
| 10:48AM |
5 |
Aastra 480i |
| 10:06AM |
1 |
Company Signed Letter of Intent to Acquire LiveVoip, LLC |
| 10:03AM |
3 |
MEETME core uses ulaw? |
| 9:53AM |
3 |
Voicemailbox on Queue? |
| 9:50AM |
1 |
Attended Transfer using wrong Context |
| 9:36AM |
3 |
[Fwd: Call forwarding] |
| 9:05AM |
1 |
PRI timing problems: Fax & Voice |
| 8:53AM |
1 |
HFC: zapata + bristuff - how to set an outgoing number |
| 8:38AM |
0 |
new production server for SOHO installation |
| 8:35AM |
1 |
Cisco 7960: Builtin CFwdAll working? |
| 7:37AM |
1 |
TDM04B in a Mac |
| 7:26AM |
3 |
Philips - [QSIG] - Alcatel - [H323] - Asterisk -[SIP] - Users. |
| 7:15AM |
1 |
ackcall |
| 7:04AM |
0 |
Philips - [QSIG] - Alcatel - [H323] - Asterisk - [SIP] - Users. |
| 6:28AM |
4 |
Put a wait in a .call file. |
| 6:21AM |
0 |
GXP-2000 review.. |
| 5:55AM |
1 |
ISDN transfer, handoff to masterswitch |
| 5:14AM |
0 |
IPSwitchBoard version 0.113 released |
| 5:09AM |
1 |
SetCallerPres problem |
| 3:36AM |
0 |
Newbie setting up LineJack Card |
| 3:11AM |
0 |
Zap (or carrier) issue ? |
| 2:35AM |
0 |
bristuff-RC8b-CVS |
| 1:30AM |
1 |
Data calls trough IAX? |
| 1:29AM |
0 |
RE:oh323 compile error |
| 1:18AM |
1 |
Difference between Asterisk and Asterisk@home? |
| 1:12AM |
1 |
Mysql/Radius Authentication |
| 12:38AM |
0 |
dial analog phone with sip |
| 12:18AM |
1 |
Asterisk and Post Paid Billing |
| 12:11AM |
1 |
oh323 compile error. |
| |
| Tuesday May 3 2005 |
| Time | Replies | Subject |
|---|
| 11:52PM |
0 |
CODEC Allow statement help |
| 10:49PM |
0 |
broadvoice setup |
| 10:40PM |
0 |
sccp question |
| 10:10PM |
0 |
Messages while on hold was:RE: Digium MOH |
| 9:10PM |
0 |
iax native bridging |
| 7:49PM |
1 |
Asterisk dialplanner |
| 7:38PM |
0 |
IAX Dual Servers |
| 7:31PM |
0 |
MEETME core uses ulaw... |
| 7:29PM |
0 |
Grandstream, Asterisk and codec mismatch |
| 7:26PM |
0 |
MOH Core uses ulaw... |
| 7:21PM |
1 |
Hardware Capacity/Configuration |
| 7:15PM |
0 |
TE4XXP and /etc/zaptel.conf |
| 6:38PM |
0 |
asterisk not detecting call hangup |
| 6:21PM |
0 |
wip 5000 hitachi crossing subnets question |
| 5:53PM |
3 |
Audio quality problem recording calls using gsm codec |
| 5:15PM |
1 |
Is there any chance to bring Skype andAsteriskUser together? |
| 3:35PM |
0 |
chan_vpb Verbose Logging |
| 3:28PM |
0 |
Asterisk crashed |
| 3:20PM |
0 |
app_dbodbc or current recommendation for odbc methods |
| 2:58PM |
0 |
Thanscoding and MoH questions |
| 2:48PM |
1 |
ztcfg at boot time |
| 2:37PM |
0 |
Queues Member Types |
| 2:20PM |
0 |
Re: LiveVOIP |
| 1:52PM |
0 |
Cisco 7970 blank screen timeout |
| 1:35PM |
4 |
Good web interface for the enduser |
| 1:30PM |
0 |
ast_readstring replacement for res_perl |
| 12:55PM |
0 |
99% Usage buy Asterisk? |
| 12:00PM |
1 |
Directory for Polycom 600 |
| 11:55AM |
0 |
Netweb 401 short review |
| 11:17AM |
1 |
How do "take away" do not disturb from certain phones |
| 11:15AM |
1 |
Any useful results? |
| 11:05AM |
0 |
Fwd: SIP over IAX2 |
| 10:59AM |
7 |
Digium MOH |
| 10:47AM |
1 |
IAX2 attended transfer on 1-0-6 Stable |
| 9:58AM |
0 |
some soft phones only talk to default context of asterisk |
| 9:54AM |
3 |
bad CLI colors? bad terminal? |
| 9:44AM |
4 |
zttool: BLU/RED Alarm |
| 9:22AM |
1 |
monitoring which IVR extension is pressed |
| 8:58AM |
0 |
qozap message error |
| 8:48AM |
12 |
TDM users: modified zttest.c for testing |
| 8:40AM |
8 |
Freak incidents, who's to blame? |
| 8:32AM |
1 |
Mediatrix 1204 Help |
| 8:25AM |
6 |
Light weight and slimmed Asterisk |
| 8:18AM |
1 |
30 button vip 1 way audio |
| 8:01AM |
1 |
invalid frame size for G.729( 2 bytes) |
| 8:01AM |
1 |
xpro codecs and asterisk |
| 8:00AM |
0 |
Forwarding incoming calls via SIP |
| 7:51AM |
7 |
Voice Quality |
| 7:31AM |
8 |
IP Phones for home use? |
| 7:20AM |
1 |
Multi-tenant Setup |
| 6:48AM |
2 |
[SPAM] - Re: cisco7940 upgrading problem - Email found in subject |
| 6:40AM |
0 |
Re: LiveVOIP |
| 6:03AM |
2 |
zaphfc dialout problems |
| 5:40AM |
2 |
cisco7940 upgrading problem |
| 5:37AM |
4 |
asterisk to analog pbx |
| 3:17AM |
1 |
Strange area codes when dialing outgoing calls on EuroISDN E1 |
| 3:15AM |
1 |
Is there any chance to bring Skype and Asterisk User together? |
| 3:01AM |
1 |
Very weird behaviour of Asterisk and SIP |
| 2:33AM |
1 |
Group redial |
| 2:28AM |
0 |
Voice Transfer of a Call Works only in One Way |
| 1:52AM |
0 |
Asterisk connects to ISDN via Fritz!Box Fon 7050 anyone? |
| 1:43AM |
1 |
How to display info from Asterisk on/to the phone ? |
| 1:02AM |
2 |
SIP NAT Polycom |
| 12:30AM |
1 |
Fwd: config for call pstn from voip |
| 12:04AM |
2 |
SIP and CVS Head |
| |
| Monday May 2 2005 |
| Time | Replies | Subject |
|---|
| 11:36PM |
1 |
email notification when leaving a message |
| 11:18PM |
4 |
DELL 2800 : PCI Parity error |
| 10:18PM |
3 |
Ast 1.0.7, IP-500's with unmanaged switch...remote end missing bits of audio |
| 10:02PM |
2 |
how stable is oh323 ? |
| 9:58PM |
1 |
Wiki Trouble? |
| 9:09PM |
0 |
Regarding asterisk-dev list |
| 9:01PM |
3 |
BSD Compatability |
| 8:57PM |
0 |
DTMF in Voicemail |
| 8:20PM |
0 |
how do you get rid of Spawn's |
| 8:17PM |
2 |
Re: LiveVOIP |
| 7:42PM |
1 |
signaling table of E100P Digium Cards |
| 7:21PM |
1 |
Detecting Fax and bad CDRs |
| 7:02PM |
1 |
Sip Group |
| 6:35PM |
1 |
Asterisk CDR - Mysql |
| 6:21PM |
2 |
LiveVOIP troubleshooting |
| 5:48PM |
1 |
External Voicemail Access |
| 5:18PM |
4 |
Anyone else having Broadvoice issues today? |
| 4:56PM |
0 |
Queue Event |
| 4:20PM |
1 |
automated availabilty testing |
| 4:19PM |
0 |
mp3 problems |
| 3:11PM |
2 |
processing power measurement? |
| 3:08PM |
0 |
Re: Your message to Asterisk-Users awaits moderator approval |
| 2:18PM |
0 |
Adtran 600 config |
| 2:14PM |
2 |
Fedora Core 3 & Shorewall Install |
| 2:12PM |
0 |
hint priority...how does it work? |
| 1:41PM |
1 |
Taking asterisk out of the media path - SIP - how is it achieved |
| 12:10PM |
0 |
codecs, asterisk, xpro |
| 11:46AM |
7 |
voicemail volume with sipura 3000 |
| 11:44AM |
4 |
Debuging SIP |
| 11:43AM |
1 |
zaptel 1.0.7 problems (again) |
| 11:35AM |
2 |
Need help getting zap trunk to work |
| 11:28AM |
2 |
Things to backup: |
| 10:52AM |
4 |
7960 "multi-line" configuration |
| 10:36AM |
0 |
Re: Asterisk-Users Digest, Vol 10, Issue 10 |
| 10:02AM |
2 |
Outgoing calls, X100P |
| 9:59AM |
0 |
Asterisk CDR Bug Or Not? |
| 9:23AM |
3 |
Choppy Sound on PSTN End |
| 9:04AM |
0 |
Bug found in SJLabs SJPhone concerning dialpad |
| 8:56AM |
0 |
Polycom Sip TEXT Messaging |
| 8:44AM |
0 |
oh323 codec order |
| 8:40AM |
7 |
Please find me a IAX provider |
| 8:24AM |
0 |
Asterisk, h323 |
| 8:07AM |
3 |
Asterisk as VM for Nortel System |
| 8:07AM |
2 |
IAX Timeout |
| 8:02AM |
0 |
large scalable voip setup |
| 8:02AM |
0 |
How to cancel a transfer in progress: |
| 7:56AM |
2 |
OT: DSL problems - UNREACHABLE - REACHABLE |
| 7:52AM |
4 |
Chan_sccp - status |
| 7:43AM |
0 |
ExtensionState problems using Asterisk API |
| 7:42AM |
1 |
Pb SIP and port |
| 6:49AM |
0 |
Phonejack PCI-card |
| 6:23AM |
0 |
Problems with TDM400P card -correction to last post |
| 5:56AM |
1 |
X-Lite and callto:// syntax in webpages |
| 5:36AM |
0 |
Putting in an Application |
| 4:50AM |
2 |
extensions.conf dial plan |
| 4:15AM |
0 |
Loading ztdummy Stops MoH but Conference Works - VmWare |
| 3:58AM |
0 |
increasing delay in meetme conference |
| 3:03AM |
0 |
config for call pstn from voip |
| 2:35AM |
0 |
calling out through second server. |
| 2:11AM |
5 |
Diffrence bewteen FXO and FXS |
| 2:10AM |
0 |
Can anyone recommend some hardware for UK use? |
| 2:06AM |
0 |
Meetme and a timing source |
| 1:45AM |
1 |
chan_h323 |
| 12:36AM |
0 |
unable to use addpac-ap200 (sip | h323) |
| |
| Sunday May 1 2005 |
| Time | Replies | Subject |
|---|
| 9:50PM |
1 |
Sipura SPA2000 dialplan vs Asterisk dialplan |
| 8:35PM |
1 |
Caller Hears Ring During Attended Transfer? |
| 7:47PM |
1 |
which port is used when "asterisk -r " |
| 7:45PM |
1 |
how to disconnect a call manually |
| 6:18PM |
1 |
Set up SIP, now I'm getting a busy tone and weird (to me) messages on the console |
| 6:14PM |
1 |
post-dial variable for whoever answered? |
| 5:32PM |
1 |
asterisk and USRobotics Courier V.Everything |
| 4:44PM |
0 |
IAX channels do not disconnect |
| 3:56PM |
0 |
Latest CVS Head Nukes Server |
| 3:10PM |
1 |
Centos - Hylafax Install |
| 2:24PM |
0 |
Sipura 401 Unauthorized. |
| 12:54PM |
0 |
Problems in new implemenation.... |
| 12:24PM |
0 |
TDM400P does not detect hangup on UK BT analogue line |
| 12:20PM |
1 |
Pre-Parse Extensions.conf? |
| 12:11PM |
1 |
Make Webvmail Error |
| 11:39AM |
1 |
Audio cut off at beginning of call |
| 11:05AM |
1 |
4 - 8 port w/QOS switch for Asterisk |
| 10:35AM |
2 |
TDM400P Power Connector |
| 10:17AM |
1 |
zaptel.conf multiple devices |
| 10:00AM |
0 |
Asterisk 1.0.6 stable IAX2 Firefly supervised call transfer? |
| 8:49AM |
2 |
TFTP question |
| 8:31AM |
1 |
Playback() stops working. |
| 8:23AM |
4 |
Dutch SIP or IAX numbers |
| 7:38AM |
0 |
sip based fax client software |
| 5:30AM |
0 |
IPS Version 0.112 released |
| 2:27AM |
1 |
[Announce] New chan_sccp release adds support for Cisco 7970 |
| 12:54AM |
1 |
mISDN error while compiling |
| 12:43AM |
2 |
Cisco 7960 SIP Reject Call Option |
| 12:16AM |
1 |
Sip calling errors |