| Thursday August 31 2006 |
| Time | Replies | Subject |
|---|
| 10:37PM |
1 |
MOH help needed with fresh install |
| 9:33PM |
1 |
Can not hear the telco System Announcement |
| 9:31PM |
2 |
Help in dailplan in asterisk |
| 8:40PM |
2 |
Asterisk server crashes after two years |
| 7:59PM |
1 |
Using Thunderbird (mail client) to call Contacts from Address Book |
| 7:43PM |
0 |
Am I looking for automon? |
| 7:12PM |
1 |
Adit 3104 randomly reboot |
| 5:53PM |
2 |
quadbri & TDM400P on same pbx ? |
| 5:47PM |
0 |
How to use *411 using either last or first name? |
| 5:44PM |
6 |
Sipura SPA3000 |
| 3:34PM |
2 |
Polycoms, Attended Transfer and Canreinvite = yes |
| 2:37PM |
1 |
does OOH323 channel support Early Media? |
| 1:52PM |
1 |
File structure question |
| 1:48PM |
1 |
Off Topic: Hardware Required |
| 1:38PM |
1 |
US Toll-Free DID Providers with Caller ID NAME? |
| 1:18PM |
1 |
SPA-942 Sound Quality |
| 1:14PM |
4 |
0005162: RTP Packetization : Few questions |
| 12:58PM |
1 |
Snom Function keys |
| 11:48AM |
0 |
weird sound with IAX |
| 11:19AM |
0 |
Compatibility INTEL E7520 |
| 9:53AM |
0 |
Question about 7940s and call forwarding |
| 9:46AM |
1 |
Per DID Codec Negotiation |
| 9:38AM |
1 |
Polycom HD Voice |
| 9:29AM |
0 |
Problems using Queues with Autofill option |
| 9:05AM |
1 |
How is GXP2000 with latest firmware |
| 8:58AM |
3 |
Problems compil 1.2.11 |
| 8:57AM |
2 |
Asterisk Sending Data to a Web Page |
| 8:38AM |
0 |
app_rxfax and T.38 |
| 8:33AM |
1 |
Help Preventing Click to Call fraud on Asterisk Servers! |
| 8:27AM |
2 |
help me!!Problem on incoming calls |
| 8:20AM |
2 |
Missing Agent Function |
| 7:42AM |
0 |
Parked call, park-dail context |
| 6:47AM |
3 |
Sangoma A104 2 ports as E1 and 2 ports as T1 configuration |
| 6:38AM |
2 |
"best" BRI card ? |
| 6:35AM |
1 |
Got error when compiling asterisk 1.2.11 |
| 6:30AM |
7 |
Cisco 7970 8.0.4 SIP firmware |
| 6:16AM |
0 |
editing configs thru web/ apps |
| 5:57AM |
0 |
Junk at beginning of frame |
| 5:30AM |
1 |
RTP Proxy |
| 5:00AM |
5 |
Fax with asterisk? |
| 4:50AM |
0 |
How to use a Half E1 with Asterisk? |
| 3:55AM |
1 |
Problems with recording |
| 3:50AM |
3 |
voicemail as email and attachment |
| 1:45AM |
0 |
CallerID and call progress pri |
| 12:54AM |
0 |
Wellgate 3804a: Got SIP response 486 "Busy Here" |
| 12:52AM |
0 |
GIZMO and Asterisk, "Failed to authenticate" |
| 12:11AM |
1 |
Toll-Free numbers |
| |
| Wednesday August 30 2006 |
| Time | Replies | Subject |
|---|
| 11:42PM |
0 |
SIP NOTIFY |
| 11:01PM |
1 |
SER+iptables+Asterisk |
| 8:39PM |
1 |
* during voicemail greeting to access mailbox |
| 8:07PM |
2 |
How to run a batch file on the asterisk CLI |
| 7:38PM |
2 |
iax vs. sip? |
| 7:36PM |
5 |
911 versus 9.911 |
| 6:53PM |
0 |
Static vs dynamic meetme rooms |
| 6:19PM |
2 |
question of CLI |
| 3:39PM |
0 |
w as pause dialing issue |
| 1:50PM |
1 |
visual indication of temp. closed mode |
| 12:58PM |
1 |
Polycom 501 config questions |
| 12:53PM |
0 |
Ascom Eurit 133 cordless ISDN phone |
| 10:14AM |
1 |
Speex Problemz |
| 10:02AM |
0 |
Intertex IX68 GW2 AIR 802.11G ADSL2+ ? |
| 9:50AM |
2 |
Asterisk speaks Russian! |
| 9:29AM |
1 |
Asterisk => Master and Slave ? |
| 9:06AM |
1 |
upgrade problem on IP phone 9133i |
| 9:03AM |
0 |
OT: Any thoughts on the new Xserve? |
| 8:55AM |
1 |
Agent solution w/o id/password |
| 8:32AM |
4 |
asterisk presence (from manager API) |
| 8:15AM |
0 |
Help please ==> Wrong password |
| 7:14AM |
0 |
New to Asterisk... |
| 6:58AM |
2 |
Sangoma Problems - A104d not detected |
| 6:09AM |
1 |
IAX call drops, recent instability |
| 6:08AM |
0 |
PrivacyManager |
| 4:51AM |
0 |
Prompts playback changing tempo in SMP kernel |
| 3:58AM |
0 |
personal address progress pri |
| 3:51AM |
5 |
Cisco 7960G SIP firmware 8.4 |
| 3:34AM |
0 |
caller display problem |
| 2:47AM |
0 |
Voicemail, how to localize date in email notifications? |
| 2:40AM |
1 |
Snom 360 Function Keys |
| 2:20AM |
0 |
Line detection with TDM400P |
| 1:38AM |
0 |
GXP-2000 update to betafirmware? |
| |
| Tuesday August 29 2006 |
| Time | Replies | Subject |
|---|
| 10:43PM |
2 |
MixMonitor and g729 licenses |
| 7:11PM |
3 |
does anyone offer truly unlimited voip in the US |
| 6:49PM |
7 |
SER Dispatcher Load Balance How-To? |
| 6:43PM |
0 |
zap fxo to sip fxs intermitently not connecting to each other |
| 6:21PM |
2 |
Unknown CLI output |
| 3:54PM |
1 |
New Parrot application, repeats what you say and more! |
| 3:48PM |
1 |
Digium makes the list! |
| 2:28PM |
0 |
Administrator Forum Email |
| 1:00PM |
0 |
CPU configuration for 250 calls SIP to SIP to IAX and fonebridge and two asterisk servers |
| 12:59PM |
0 |
Asterisk 1.2.11 and ${SIPDOMAIN} variable |
| 12:03PM |
0 |
OT: Bandwidth calculations and PCI/PCIX/PCIE |
| 11:12AM |
1 |
SIP T1 timer and qualify=yes |
| 11:01AM |
0 |
GXP-2000 auf Betafirmware updaten? |
| 10:35AM |
1 |
Re: [asterisk-biz] Asterisk Tools |
| 10:26AM |
2 |
Copying a recording to a voice mail box |
| 10:21AM |
0 |
[Fwd: Re: Asterisk t38passthrough] |
| 10:08AM |
2 |
DTMF between cisco and sipura going through asterisk |
| 9:04AM |
1 |
Advice needed - asterisk & Mitel 200SX |
| 8:29AM |
3 |
IP interface "box" for Meridian type digital phone |
| 8:27AM |
0 |
Which BRI Card ? |
| 8:15AM |
3 |
Connecting two asterisk servers |
| 7:48AM |
2 |
Detect if cell phone or users |
| 6:57AM |
1 |
Analyze core file prodeced after safe_asterisk crashh |
| 6:17AM |
0 |
playback() breaks audio in zap->iax->iax->zap channel |
| 6:12AM |
1 |
Asterisk codec strangeness |
| 4:30AM |
0 |
working chan_bluetooth enviroment |
| 4:28AM |
1 |
Handytone 286 T.38 SDP parameters |
| 3:24AM |
1 |
Mix Monitor call quality |
| 3:19AM |
1 |
Asterisk - Comfort Noise |
| 3:02AM |
1 |
Asterisk 1.2.4 I hear other party's voice only when I speack need help |
| 2:59AM |
2 |
transform bridged call into a conference |
| 2:55AM |
4 |
does misdn-mqueue work if compiled with gcc 4? |
| 2:48AM |
1 |
sip giving problems, please help. |
| 2:16AM |
0 |
SIP Error message |
| |
| Monday August 28 2006 |
| Time | Replies | Subject |
|---|
| 11:57PM |
0 |
Providers that offer contract |
| 9:21PM |
0 |
IAX2 Bandwidth setting |
| 9:18PM |
2 |
Selecting outbound trunk |
| 8:01PM |
1 |
Debian and Asterisk IAX2 channel driver |
| 7:28PM |
1 |
Asterisk Manager Interface Question |
| 6:49PM |
2 |
Can I increase DTMF sensitivity? |
| 4:59PM |
1 |
ISDN BRI, and Trixbox |
| 4:23PM |
0 |
Is there a Blue tooth wireless headset that willwork with asterisk? |
| 3:46PM |
0 |
Asterisk queues and dynamic members |
| 3:24PM |
1 |
Is there a Blue tooth wireless headset that will work with asterisk? |
| 3:23PM |
0 |
Changes in handling anonymous calls entering asterisk |
| 2:03PM |
2 |
Voicemail/Email Integration |
| 1:31PM |
0 |
Multiple Queue Problem |
| 1:23PM |
4 |
Can anyone recommend a large button sip phone for the elderley. |
| 11:53AM |
3 |
manual mods with GUI in place |
| 10:49AM |
1 |
Problem with a TDM400P |
| 10:15AM |
1 |
Call parking with Polycom's - works but MOH stops in one scenario |
| 10:12AM |
0 |
Re: asterisk-users Digest, Vol 25, Issue 139 |
| 10:10AM |
5 |
Asterisk with PABX |
| 9:31AM |
0 |
Timeout Registration IAX2 |
| 9:08AM |
0 |
Changes in handling anonymous calls entering ast erisk |
| 9:06AM |
0 |
Queue problem - autofill option |
| 8:32AM |
1 |
Missing number 2 in "advanced options" of VM |
| 8:11AM |
1 |
REGISTER attempt |
| 7:20AM |
2 |
Question about context for incoming calls |
| 7:14AM |
1 |
Make Asterisk server initiate a Call |
| 6:56AM |
1 |
Grabbing authenticated mailbox value from VoicemailMain() |
| 6:54AM |
0 |
AEL2 patch issues |
| 6:52AM |
2 |
H264 |
| 6:47AM |
1 |
How to set MWI |
| 5:11AM |
0 |
GROUP() and queues |
| 4:58AM |
3 |
lost packets when bridging zap and iax |
| 2:49AM |
1 |
Remote CAPI - ISDN over TCP/IP |
| 2:27AM |
0 |
newbie request |
| 12:07AM |
2 |
Tracing audio problems |
| |
| Sunday August 27 2006 |
| Time | Replies | Subject |
|---|
| 11:01PM |
1 |
TrixBox install |
| 10:03PM |
2 |
how to enable REACHABLE/UNREACHABLE messages in logs |
| 8:32PM |
0 |
Trixbox – Called party can't hangup |
| 5:15PM |
2 |
Max number of SIP devices registered to an extension |
| 4:49PM |
1 |
SEXY WOMAN wants to know about =>Callback in within voicemail broken |
| 2:31PM |
2 |
Shared NFS or Shared MySQL for redundant secondary server? |
| 9:19AM |
1 |
detecting a users number using the dialplan or AGI |
| 8:50AM |
2 |
Cannot dial out through SIP provider |
| 8:33AM |
4 |
CDR Function - Asterisk-1.2.10 |
| 8:01AM |
0 |
[RESOLVED] One way audion on Sangoma |
| 5:40AM |
0 |
asterisk registering as extension to another asterisk server problem |
| 4:56AM |
1 |
Dial C option |
| 3:33AM |
0 |
Doubled digits on vm pasword |
| 1:41AM |
0 |
about MusicOnHold / Playback |
| 1:29AM |
0 |
Voicemail's mail formate |
| |
| Saturday August 26 2006 |
| Time | Replies | Subject |
|---|
| 10:58PM |
1 |
"hint" for Hold |
| 10:20PM |
4 |
Call Max Time |
| 4:21PM |
0 |
ticks in the pstn side audio |
| 4:21PM |
0 |
determining meetme user number |
| 2:48PM |
0 |
ANNOUNCEMENT: Asterisk-Java 0.3-m1 released |
| 9:33AM |
2 |
getting SIP to listen on multiple ports |
| 8:52AM |
3 |
Nobody is responding. Why? (Implement music on transfer) |
| 8:31AM |
4 |
Problem with Tycho Voicemail |
| 8:26AM |
1 |
Asterisk Performance without RTP? |
| 7:24AM |
3 |
can not get ${LEN(VAR)} and greater than ">" to work for me |
| 5:24AM |
1 |
Uptime Record? |
| 12:41AM |
0 |
app_txfax / app_rxfax |
| |
| Friday August 25 2006 |
| Time | Replies | Subject |
|---|
| 8:39PM |
1 |
Asterisk Real Time Engine - Fails to Connect to MySQL |
| 5:35PM |
3 |
Help compiling asterisk-addons on Debian? |
| 5:03PM |
1 |
CentOS4.3 or Debian 3.1r2? |
| 4:26PM |
1 |
Linksys PAP2 Ring Settings |
| 3:10PM |
0 |
Re: asterisk-users Digest, Vol 25, Issue 119 |
| 2:31PM |
0 |
Using asterisk to simulate ISDN BRI line |
| 12:54PM |
2 |
What are my logs telling me here? |
| 12:51PM |
0 |
read more than 2 digits on festival |
| 11:50AM |
5 |
Will Asterisk work with Exchange 2007 UM? |
| 11:27AM |
2 |
[RESOLUTION] Polycom microbrowser issue Error HTTP 406 withIIS |
| 11:18AM |
4 |
DNS |
| 9:42AM |
1 |
Standard for transfer via IAX provider? |
| 9:22AM |
2 |
7970 'LoadID incorrect' problem |
| 8:36AM |
1 |
misdn-init.conf card parameter for a monoBRI |
| 8:23AM |
2 |
New Asterisk Voice Changer 0.4 |
| 6:47AM |
1 |
Singapore |
| 6:31AM |
0 |
Can i use the FXO of a addpack in Asterisk |
| 5:51AM |
0 |
Polycom microbrowser issue Error HTTP 406 withIIS |
| 4:40AM |
1 |
How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel? |
| 2:58AM |
2 |
Does anyone use T.38? |
| 1:55AM |
0 |
Multiple Vulnerabilities in Asterisk 1.2.10 (Fixed in 1.2.11) |
| 1:04AM |
1 |
quadBRI beronet card: how to specify which ISDN channel to use to make calls |
| 12:39AM |
6 |
IP phone with 2 ethernet jacks |
| |
| Thursday August 24 2006 |
| Time | Replies | Subject |
|---|
| 11:32PM |
0 |
Re: [Users] Mysql problem |
| 7:02PM |
1 |
Can the codec/format for name/greeting in voicemail be changed? |
| 4:36PM |
1 |
hint status not updating on inbound |
| 3:04PM |
1 |
Attempt to setup paging and intercom |
| 2:57PM |
7 |
Idiot questions |
| 2:19PM |
3 |
Snom phones locking up |
| 2:06PM |
1 |
Polycom microbrowser issue Error HTTP 406 with IIS |
| 11:29AM |
0 |
hotel teledex integration anyone? |
| 11:19AM |
1 |
RE: [asterisk-dev] Phone status |
| 11:17AM |
2 |
Phone status |
| 10:55AM |
1 |
Modems dialing over sangoma a104d |
| 10:53AM |
3 |
Call Parking Ring Back (Snoms) |
| 10:53AM |
3 |
Asterisk t38passthrough |
| 9:31AM |
1 |
No outbound with A2Billing |
| 8:43AM |
2 |
Wellgate 3804a |
| 8:37AM |
0 |
originate from group + dialplan |
| 8:33AM |
1 |
SendText Queue Notification |
| 7:19AM |
0 |
need help with error code |
| 6:49AM |
0 |
Monitoring/Listening In (Scott Pinhorne) |
| 6:49AM |
0 |
About IVR and Oracle (Tim Panton) |
| 6:32AM |
2 |
SV: E61 |
| 6:19AM |
0 |
Quiet on the list today? |
| 6:07AM |
1 |
E61 |
| 5:14AM |
2 |
Active Directory Listing Feauture |
| 5:12AM |
3 |
quintum Calling Card |
| 3:54AM |
0 |
Multiple lines in body of UserEvent |
| 2:42AM |
1 |
Monitoring/Listening In |
| |
| Wednesday August 23 2006 |
| Time | Replies | Subject |
|---|
| 11:03PM |
0 |
AstLinux 0.4.3 Released! |
| 6:02PM |
8 |
Nokia E60/61/70 and SIP |
| 4:39PM |
0 |
SJPhone and Asterisk over H323 |
| 4:25PM |
0 |
Getting strange behavior on SIP channels after upgrade to 1.2.11 |
| 4:06PM |
1 |
One way audion on Sangoma |
| 3:43PM |
0 |
howto install asterisk on freebsd release 4.11 |
| 3:07PM |
2 |
About IVR and Oracle |
| 2:42PM |
0 |
isdn30 uk setup problem |
| 1:54PM |
0 |
NoCDR() |
| 1:45PM |
0 |
MySQL undefined symbol: __pure_virtual |
| 1:34PM |
0 |
Unable to start special tone |
| 1:00PM |
0 |
client socket to asterisk manager gets disconnected |
| 12:35PM |
4 |
Annoying Bristuff |
| 11:40AM |
3 |
NAT problems |
| 11:13AM |
1 |
3COM NBX and Digium Cards... |
| 10:43AM |
1 |
USB GSM gateway for Asterisk? |
| 10:33AM |
1 |
IAX2 extn not registering on 4569 |
| 10:32AM |
1 |
Choppy calls on IAX trunk but no problems on internal calls |
| 9:55AM |
0 |
Silent Calls (Ghost Calls) When Picking Up Queue Calls |
| 8:54AM |
0 |
Connecting Asterisk to Avaya Definity over H.323 |
| 8:53AM |
3 |
Cisco PIX firewall and nat=yes |
| 8:44AM |
1 |
Registering IP Phone To Asterisk |
| 8:11AM |
0 |
Weird compile problem |
| 6:53AM |
4 |
Cisco Router QOS and IAX2 |
| 6:48AM |
1 |
VM - advanced options? |
| 6:48AM |
0 |
SV: Hint extension issue - bug? |
| 5:58AM |
3 |
Slightly off-topic: Opinions of Comcast and Bellsouth? |
| 5:39AM |
1 |
Direct to Voicemail |
| 4:51AM |
0 |
Call Handoff |
| 3:33AM |
0 |
dtmf during a call |
| 2:06AM |
0 |
Auto Congestion |
| 2:00AM |
2 |
Adding/Removing Prefixes |
| 1:57AM |
1 |
Dialling from extension to extension with Manager |
| 12:56AM |
1 |
column width in CLI |
| |
| Tuesday August 22 2006 |
| Time | Replies | Subject |
|---|
| 9:21PM |
0 |
Missing Extension |
| 8:45PM |
0 |
Multiple site multi server setup |
| 8:29PM |
1 |
problems with wevbmail |
| 7:43PM |
5 |
Calls over VPN |
| 6:40PM |
2 |
Working Sipura 3000 or Linksys 3102 configuration? |
| 6:33PM |
2 |
How to set externip in sip.conf automatically? |
| 6:30PM |
0 |
Asterisk 1.2.11, Asterisk-Addons 1.2.4 and Zaptel 1.2.8 Released |
| 6:01PM |
1 |
SSH connection hangs on logout? |
| 5:48PM |
1 |
problem with asterisk billing time... |
| 5:36PM |
2 |
Simple CDR parser to print to webpage |
| 5:24PM |
0 |
Non-zaptel hardware based timing sources |
| 5:20PM |
0 |
Speech Recognition Apps |
| 5:09PM |
1 |
Setting the contact header on outbound INVITE |
| 5:09PM |
3 |
Prompts recording for Asterisk |
| 3:53PM |
1 |
No CLID from PSTN using X100P FXO Card |
| 3:49PM |
2 |
Strange SIP response |
| 3:22PM |
5 |
Hint extension issue - bug? |
| 3:19PM |
0 |
Asterisk, two eth and two providers |
| 1:46PM |
1 |
Anybody using Eicon SoftIP with Asterisk |
| 1:32PM |
0 |
PRI Ethernet Bridge |
| 11:26AM |
3 |
How can I implement Music on Call Transfer? |
| 9:07AM |
1 |
Unable to match on CallerID in an include block |
| 7:55AM |
1 |
AMI initiate call probs |
| 7:44AM |
1 |
Polycom 501 vs 601 provisioning |
| 7:20AM |
12 |
Realtime Extensions -- Comments? |
| 6:31AM |
1 |
R: Snom360 with 6.2.2 firmware |
| 6:30AM |
0 |
No audio from IVR of Cisco Call Manager |
| 6:28AM |
3 |
placing a call with the Manager interface |
| 6:27AM |
3 |
Asterisk and spandsp |
| 5:55AM |
1 |
Setting RPID privacy? |
| 5:02AM |
1 |
re-writing the dial plan - some hints please |
| 4:45AM |
1 |
using asterisk + sangoma a102 to simulate telco PRI: is possible? |
| 4:43AM |
1 |
if command for or missing callerid? |
| 4:24AM |
2 |
Sipura 3000 dialplan strings. |
| 4:03AM |
0 |
R: Text to Speech |
| 4:00AM |
2 |
Compilation |
| 2:48AM |
2 |
Asterisk forum - forum.globalvoicenet.com |
| 1:59AM |
10 |
GSM gateway and FXO ATA |
| 1:19AM |
0 |
lbProxy |
| 1:17AM |
0 |
H323 can not register to remote openh323gk? |
| 1:04AM |
0 |
NuFone chan_h323 |
| 12:44AM |
2 |
Snom360 with 6.2.2 firmware |
| 12:36AM |
1 |
How to modify incoming DNIS? |
| 12:23AM |
1 |
1.2.10 and 1.2.9.1 |
| |
| Monday August 21 2006 |
| Time | Replies | Subject |
|---|
| 11:28PM |
3 |
No retry after DNS failure |
| 8:20PM |
0 |
Meetme bug or feature? |
| 8:15PM |
2 |
Re: SIP Debug to file - Is it possible? |
| 8:02PM |
0 |
Quick, hopefully easy, question |
| 6:59PM |
0 |
Indonesian MFC-R2 |
| 6:58PM |
1 |
Re: SIP Debug to file - Is it possible? |
| 5:50PM |
0 |
SIP Encryption in China |
| 4:54PM |
1 |
SLA.conf |
| 2:36PM |
2 |
Call file do 2 outbound call |
| 2:15PM |
1 |
Manager API: matching an Originate to the Newchannel event |
| 2:03PM |
1 |
Voicemail and languages other than english doesn't seem to work well |
| 1:34PM |
0 |
Double dial dtmf sounds |
| 1:12PM |
4 |
Text to Speech |
| 12:37PM |
6 |
Realtime and hints |
| 12:05PM |
1 |
Realtime and labels |
| 10:01AM |
1 |
Asterisk in Xen 3.0 |
| 9:39AM |
1 |
Portuguese sound files available? |
| 9:10AM |
0 |
Status of Monitor |
| 8:41AM |
0 |
Size of realtime appdata field under MySQL |
| 8:26AM |
0 |
failed calls |
| 8:14AM |
0 |
Cancelling outbound call: is Asterisk behaving correctly |
| 7:50AM |
0 |
DTMF + voipjet |
| 7:26AM |
0 |
SV: Choppy sound zap-to-sip, but not sip-to-sip? |
| 7:08AM |
0 |
Is it possible to call System dialplan application via AMI? |
| 7:07AM |
0 |
IAX2 TRUNK CPU consumption |
| 6:52AM |
1 |
Configure mailserver to deliver voicemail |
| 6:22AM |
6 |
Joining calls via manager.api or AGI |
| 6:15AM |
0 |
Choppy sound zap-to-sip, but not sip-to-sip? |
| 4:43AM |
0 |
polycom_acd_functions branch and outboundproxy |
| 4:39AM |
0 |
how to set 'transfercapability' |
| 4:20AM |
1 |
running agi application in the background |
| 3:52AM |
4 |
Zaptel install - Fedora Core 5 |
| 2:39AM |
1 |
zap channel media volume |
| 1:41AM |
0 |
Re: no audio issue (asterisk@teladdict.com) |
| 1:15AM |
2 |
SIP ActiveX? |
| 1:03AM |
0 |
IAX2 Auto fallthrough |
| 12:04AM |
0 |
queuememberstatus overwhelms manager socket connection to asterisk |
| |
| Sunday August 20 2006 |
| Time | Replies | Subject |
|---|
| 11:53PM |
1 |
no audio issue |
| 11:02PM |
1 |
sox &gsm |
| 7:42PM |
3 |
Asterisk installations in Germany |
| 6:28PM |
1 |
Asterisk not parking calls - causes? how to fix? |
| 4:54PM |
1 |
Call to a queue killing Asterisk? |
| 4:53PM |
2 |
Announce caller-id |
| 3:13PM |
1 |
Linksys SPA-941 Message Waiting Indicator |
| 2:20PM |
2 |
Sending signals to asterisk |
| 1:55PM |
2 |
Analog-to-VoIP: blade? |
| 11:20AM |
1 |
Polycom IP430 won't finish boot |
| 10:54AM |
0 |
No-audio problem |
| 9:47AM |
1 |
Metermaid - Parking Slot |
| 9:38AM |
7 |
Connecting an cellphone to asterisk |
| 8:06AM |
1 |
Indonesian MFC/R2 |
| 7:55AM |
2 |
Ignoring PRI call? |
| 1:58AM |
3 |
Asterisk Jobs Update |
| 1:51AM |
1 |
How to find which queue member answered a call? |
| |
| Saturday August 19 2006 |
| Time | Replies | Subject |
|---|
| 9:10PM |
2 |
recommended hardware specs |
| 9:06PM |
0 |
asterisk and hipath 3750 |
| 8:36PM |
0 |
Mailcall question |
| 1:43PM |
0 |
Zap channel-modem/fax conn. problems |
| 5:15AM |
6 |
New Voicemail Client for Win32, Linux x86, Mac OS X released |
| 4:55AM |
0 |
TAMPA BAY Asterisk Users Meeting Monday |
| 1:19AM |
0 |
Asterisk + DTMF + g729 |
| 12:21AM |
0 |
[Linksys 3102] Couple of issues |
| |
| Friday August 18 2006 |
| Time | Replies | Subject |
|---|
| 10:04PM |
2 |
Asterisk - SIP client latency |
| 7:08PM |
2 |
loading the prompt files in memory on Asterisk startup |
| 4:17PM |
1 |
SLA Doc |
| 2:42PM |
3 |
Iaxy and SendDTMF?? |
| 1:55PM |
1 |
How To NOT Generate A CDR For A Call? |
| 12:55PM |
1 |
chan_skinny - in trunk r40360 - error "unsupported format '0'" |
| 12:39PM |
0 |
Realtime Extensions and 'include =>' |
| 11:28AM |
9 |
Apache for FastAGI |
| 11:04AM |
1 |
Realtime Peers Disappearing |
| 10:50AM |
1 |
MaxRetries:1 - Problems Dialout Call files |
| 9:57AM |
1 |
Static in Monitor recordings |
| 9:44AM |
1 |
Recent additions to the Digium Asteriskdevelopment team |
| 9:43AM |
2 |
Dialplan "or" matching |
| 9:32AM |
1 |
TE207P |
| 9:20AM |
0 |
video call monitor |
| 8:29AM |
1 |
call barge |
| 8:15AM |
1 |
Ringtone/gentone/busy and g729 |
| 7:04AM |
1 |
Presence SUBSCRIBE/NOTIFY behaviour |
| 4:57AM |
4 |
PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units |
| 4:35AM |
0 |
Maximum length of CID using SET CALLERID in AGI? |
| 1:22AM |
1 |
Extension presedence. |
| |
| Thursday August 17 2006 |
| Time | Replies | Subject |
|---|
| 11:12PM |
3 |
Equivalent of channel switching? |
| 10:48PM |
1 |
SIP_HEADER function; what names are available? |
| 10:21PM |
0 |
Always On Top 100 private companies |
| 10:13PM |
4 |
Polycom 601 Issues |
| 8:46PM |
0 |
dynamic queue problem |
| 6:54PM |
1 |
Frustration cubed |
| 3:49PM |
0 |
t.38 asterisk-trunk |
| 3:11PM |
5 |
Sending Email From A Dial Plan |
| 2:55PM |
0 |
Cepstral and Asterisk again... |
| 2:24PM |
1 |
Where can i get a telephone number of Brasilia or Rio de Janeiro in Brazil |
| 2:13PM |
1 |
Asterisk and T1 Extensions. |
| 2:12PM |
5 |
Return data from Fast AGI |
| 2:10PM |
1 |
Turn Off chan_sip Debug Messages |
| 1:08PM |
0 |
Realtime Extension Lookups |
| 1:03PM |
2 |
Accessing SIP URI (not ${SIPURI}) |
| 12:33PM |
2 |
Realtime include |
| 11:30AM |
1 |
VoiceMail and Fax on same extension |
| 11:09AM |
2 |
astbill white screen!! |
| 10:16AM |
0 |
Dial statement problem |
| 9:47AM |
0 |
I can´t set to work two tdm2400p and one TE205p on same machine, please help |
| 9:28AM |
1 |
Changing CID |
| 8:55AM |
0 |
Call Parking initiator cannot retrieve parked calls |
| 8:01AM |
0 |
sangoma a102: "Rx Error: 'Retry' exceeds maximum (64k): pci fatal error" |
| 7:48AM |
0 |
pri rdnis found as Facility but not set |
| 7:44AM |
0 |
480 "Temporarily Unavailable" message |
| 7:39AM |
0 |
Assigning specific RTP ports to SIP clients |
| 7:35AM |
1 |
strange behaviour of a zaptel device |
| 6:44AM |
0 |
BRI<->PRI switching and synchronization (data/fax calls) |
| 6:06AM |
0 |
spanDSP + rxfax |
| 6:01AM |
3 |
PRI problems - no D channel |
| 5:37AM |
1 |
valgrind + Asterisk |
| 5:32AM |
0 |
AGI transfer question |
| 4:42AM |
2 |
Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days) |
| 4:20AM |
1 |
Sip suppression |
| 3:24AM |
0 |
Dial out based on SIP invite |
| 3:00AM |
0 |
SendDTMF - how to use |
| 2:30AM |
0 |
Dialing out using SIP terminal |
| 2:29AM |
0 |
Conflict between S & L option in Dial? |
| 2:21AM |
0 |
CDR inaccuracies |
| 2:15AM |
0 |
ExtensionState always returns 1 |
| 2:07AM |
0 |
wanpipe1:w1g1: Rx Error: 'Retry' exceeds maximum (64k): pci fatal error! (0x0007C03C) |
| 1:09AM |
4 |
Festival through AGI can't handle strings longer than 15 chars |
| 12:07AM |
0 |
Need help: RTP Stream not set up correct only when calling out |
| |
| Wednesday August 16 2006 |
| Time | Replies | Subject |
|---|
| 11:42PM |
2 |
Re: what is the real use of AEL? |
| 10:04PM |
0 |
How to transfer call without getting dropped |
| 9:46PM |
1 |
how to add prefix 0 (zero) when coming from ISDN trunk |
| 7:23PM |
1 |
Not Working |
| 5:25PM |
4 |
What to use beyond T1's? |
| 4:55PM |
2 |
DMTF issues on voicemail on Zap |
| 4:19PM |
1 |
Set DID |
| 4:01PM |
1 |
Cepstral and Asterisk |
| 3:12PM |
3 |
Recent additions to the Digium Asterisk development team |
| 2:54PM |
1 |
SIP-NAT failure on dynamic IP |
| 2:29PM |
1 |
No "zap" command? |
| 1:16PM |
0 |
Strange CLI Output |
| 1:15PM |
0 |
Trixbox Fax to PDF |
| 1:02PM |
1 |
Force immediate re-registration on sip reload |
| 12:33PM |
0 |
calling in-out |
| 11:03AM |
4 |
Asterisk Real Time and sip.conf file used at the same time |
| 10:12AM |
1 |
Restricting Incoming SIP Calls Without "call-limit" |
| 9:49AM |
22 |
Asterisk 'Hosting' |
| 9:48AM |
0 |
Attended Transfer call return with asterisk + sipura spa2002 |
| 8:46AM |
0 |
MFCR2 and Unicall PDF |
| 8:40AM |
5 |
Digium TDM400P Vs Sangoma A200 |
| 8:35AM |
1 |
linuxdevices.com: >>Trolltech woos developers with "open" Linux phone<< Who'll be the first with * on a mobile? |
| 8:14AM |
2 |
polycom config error 0x4020: possibly related to RE:Polycom upgrade issue? |
| 7:03AM |
0 |
Asterisk@von - Von Fall, Boston Sept 11-14 |
| 6:53AM |
0 |
Asterisk Training - Boston, US and Malaga, Spain |
| 5:10AM |
1 |
Extension for Incoming Call through Zap Channel |
| 4:27AM |
0 |
Support a malformed SIP INVITE |
| 4:18AM |
0 |
REQ: BATM gw-232 sip firmware |
| 1:58AM |
1 |
Problems with outgoing calls on a TE410P |
| 12:40AM |
0 |
RTP Stream not set up correct at outgoing call |
| 12:29AM |
0 |
capi (divas4linux) bearer setting |
| |
| Tuesday August 15 2006 |
| Time | Replies | Subject |
|---|
| 9:12PM |
1 |
postfix and asterisk |
| 7:07PM |
2 |
Polycom upgrade issue |
| 6:43PM |
1 |
Question about queue |
| 6:32PM |
6 |
modprobe wctdm fails in /etc/rc.local on FC5 |
| 5:44PM |
0 |
anothet tes |
| 5:39PM |
0 |
sip host and registering |
| 4:33PM |
0 |
AsteriskSpeaksGoogleTalk - User is always disconnected - Problems |
| 4:29PM |
1 |
New asterisk jukebox needs testing |
| 4:23PM |
2 |
How to reject a call without picking it up, (E1-T1-ISDN) |
| 4:04PM |
0 |
macro-dialout without specifying trunk |
| 3:35PM |
2 |
Multiple registrations to the same asterisk server |
| 3:18PM |
1 |
STRFTIME dialplan function not picking up system timezone |
| 1:57PM |
1 |
Asterisk & Gizmo? |
| 1:41PM |
1 |
7970 SIP image |
| 1:10PM |
5 |
Cisco 7960 password reset |
| 1:10PM |
3 |
New Device |
| 12:28PM |
5 |
Softphone for Windows Mobile 5? |
| 12:05PM |
1 |
IAX unstable with large number of calls? |
| 10:28AM |
1 |
1.2.10 - g726 Issues |
| 10:23AM |
2 |
SIP asterisk over Linksys VPN |
| 9:28AM |
0 |
Intel D945G chipset |
| 9:16AM |
3 |
Page Groups |
| 8:50AM |
2 |
Can budgetone 101 display name part of cid? |
| 8:05AM |
13 |
Manager Interface API's |
| 8:02AM |
0 |
ARI |
| 7:06AM |
0 |
Hangup Problem with PSTN and ISDN |
| 6:09AM |
0 |
extensions.ael - calling an exten from a macro |
| 2:12AM |
0 |
PRI Clock Signal Problem |
| |
| Monday August 14 2006 |
| Time | Replies | Subject |
|---|
| 11:18PM |
1 |
Sending SIP 183 Session Progressing |
| 9:38PM |
1 |
Problems with incoming authentication |
| 9:10PM |
0 |
Sorry! My Bad! |
| 9:09PM |
2 |
Asterfax and Gentoo |
| 8:36PM |
1 |
1 way audio. Dual NIC's. |
| 8:20PM |
1 |
Ringing after answered on zaptel |
| 6:44PM |
3 |
Run As User Asterisk |
| 6:41PM |
3 |
Config quesiton: all inbound on PRI |
| 6:10PM |
0 |
Reason to hit failed extension |
| 5:25PM |
2 |
PRI Dropouts (Solved) |
| 5:23PM |
1 |
Is anybody moderating this list? |
| 3:53PM |
1 |
Zap difficulties |
| 3:28PM |
1 |
Sending INVITE to an unavailable phone - Bug? |
| 3:24PM |
4 |
Asterisk load testing |
| 3:15PM |
4 |
SPA-942 TFTP Provisioning |
| 1:54PM |
5 |
Asterisk and PHP? |
| 1:53PM |
0 |
Cisco 7961 SIP & Presence / BLF |
| 1:52PM |
1 |
pyAst |
| 1:46PM |
2 |
Asterisk And Java? |
| 1:45PM |
2 |
Asterisk time not the same as unit time ? |
| 1:23PM |
2 |
reloading agents and queues |
| 1:09PM |
1 |
More SNOM, Message Indicator/Retrieval issues |
| 1:02PM |
1 |
Dapper Drake, Asterisk, and Faxing |
| 12:57PM |
1 |
g.711 Codec Question |
| 12:47PM |
1 |
Anyone know a DID provider in Panama (country code 507)? |
| 12:35PM |
1 |
Cron Job to Drop a Call File When the Hard Drive Gets over 50% Full |
| 12:34PM |
0 |
ESCAUX net.PBX, new template with autoconfig of all major IP Phones |
| 12:06PM |
1 |
channel.c: Avoided initial deadlock for '0x8de2dc0', 10 retries! |
| 11:52AM |
0 |
Linksys and Call Park |
| 10:09AM |
1 |
OT: Changing Cisco tftp root directory |
| 7:31AM |
1 |
prob with star input agi-bin |
| 7:25AM |
1 |
Queue Management |
| 5:25AM |
2 |
Problems with Hangup |
| 3:53AM |
0 |
Re: ESCAUX net.PBX registration and boot sequence (was Re: [asterisk-biz]ESCAUX releases net.PBX Free Edition) |
| 2:14AM |
2 |
CallerID is not displaying for my incoming calls |
| 1:44AM |
2 |
Associating an Originate Request to a Channel before the call is answered |
| 1:25AM |
1 |
queue announcements when using ringback |
| |
| Sunday August 13 2006 |
| Time | Replies | Subject |
|---|
| 9:20PM |
1 |
Queue Monitoring Broken? |
| 5:08PM |
0 |
ANNOUNCEMENT : Asterisk2Billing V1.2.3 (BrainCoral) |
| 4:09PM |
0 |
Astribank |
| 9:36AM |
1 |
Callback in within voicemail broken |
| 7:58AM |
0 |
abhishek invites you to join Zorpia |
| 4:23AM |
1 |
CDR Variable |
| 1:56AM |
1 |
Macro inside macro |
| 1:06AM |
1 |
911 Testing |
| |
| Saturday August 12 2006 |
| Time | Replies | Subject |
|---|
| 9:02PM |
0 |
Re: asterisk-users Digest, Vol 25, Issue 35 |
| 1:59PM |
0 |
problem with mfcr2 protocol |
| 12:07PM |
0 |
OT: Call For Papers -- 2007 Southern California Linux Expo |
| 10:48AM |
2 |
Issues compiling addons on Fedora Core 3 |
| 9:38AM |
0 |
Declined to talk, Call rejected: 603 Declined |
| 7:27AM |
1 |
SPA3000 dialplan coding... |
| |
| Friday August 11 2006 |
| Time | Replies | Subject |
|---|
| 9:31PM |
1 |
SIP header challenge |
| 7:30PM |
1 |
safe_asterisk to start latest version from SVN - trying asterisk with googletalk |
| 7:09PM |
0 |
SuSE 10.1 zaptel init script |
| 4:14PM |
2 |
AgentcallbackLogin() |
| 3:45PM |
4 |
Abstraction for a newbie |
| 2:14PM |
1 |
multiple offices / hard phones / service provider |
| 11:51AM |
1 |
jitterbuffer SIP-IAX possible? |
| 11:34AM |
3 |
Inbound Calls & SIP/2.0 404 Not Found |
| 11:23AM |
0 |
GXP-2000 Call Transfer Problem |
| 10:52AM |
1 |
Auto retry on Busy |
| 10:38AM |
2 |
Fast busy signals... Satisfying my curiousity |
| 10:13AM |
1 |
Call transfer issues |
| 10:09AM |
0 |
Callback feature in voicemail broke? |
| 9:46AM |
1 |
Digit timeout on Asterisk Assisted Transfers |
| 9:43AM |
0 |
Bind Ounbound SIP Trunk to second virtual IP on server |
| 9:17AM |
0 |
Connecting to another server |
| 9:10AM |
0 |
Agent Transfer Locking up Queue() Application |
| 9:01AM |
0 |
Odd Busy tone on Aastra phones |
| 8:54AM |
2 |
MailboxExists not branching to n+101 |
| 8:52AM |
1 |
DTMF-CallerID on POTS |
| 6:28AM |
0 |
Has anybody a usefull example for the DIAL-option G(context|exten|prio) |
| 6:13AM |
3 |
USA Toll Free |
| 6:03AM |
4 |
Unable to receive Incoming calls to my DID. Please tell me the solution |
| 5:41AM |
4 |
Asterisk IAXmodem HylaFax? |
| 5:04AM |
2 |
Polycom just disconnects |
| 5:03AM |
1 |
Asterisk GUI tool needed |
| 3:08AM |
1 |
In CDR record not what I want |
| 2:53AM |
2 |
High Availability with PRI failover |
| 2:48AM |
4 |
Port Forwarding SIP rtp |
| 2:24AM |
0 |
where/when to set__TRANSFER_CONTEXT ? |
| 1:38AM |
0 |
question about oh323 and ring tone |
| |
| Thursday August 10 2006 |
| Time | Replies | Subject |
|---|
| 11:26PM |
3 |
No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256) |
| 9:44PM |
0 |
Phone number lookup public database |
| 7:56PM |
1 |
Quick One - PHP Script to restart Asterisk |
| 2:26PM |
3 |
Correct syntax for Set(CALLERID(all)... |
| 2:21PM |
2 |
A good price for FXS 48 ports? |
| 12:13PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday August 12th - 11:30am |
| 11:01AM |
1 |
Snakes On A Plane using Asterisk? |
| 10:59AM |
0 |
IAXy can't connect to analog phone |
| 10:40AM |
1 |
Asterisk Voicemail Setup |
| 10:23AM |
0 |
Odd IAX stats |
| 10:20AM |
4 |
sms callback? |
| 8:58AM |
1 |
Set DID? |
| 8:27AM |
1 |
Samsung Prostar DCS |
| 7:59AM |
5 |
can i detect a voice with asterisk ? |
| 6:50AM |
2 |
Fwd: Dropping incompatible frame killing Asterisk |
| 6:16AM |
0 |
Question on iax2 show netstats |
| 5:09AM |
0 |
Help with newbie: D-link admin setup |
| 4:52AM |
1 |
Realtime SIP Authentication |
| 4:49AM |
0 |
Clipped audio at beginning of SIP calls. |
| 3:18AM |
1 |
Sangoma A101 problem |
| 1:43AM |
2 |
transfer call von D-channel |
| 1:17AM |
1 |
Iptables ,rtp |
| |
| Wednesday August 9 2006 |
| Time | Replies | Subject |
|---|
| 9:48PM |
1 |
SIP trunks: order or type |
| 7:48PM |
1 |
Callback and Asterisks |
| 7:44PM |
2 |
Warning - Voiplink.com doesn't deliver - stuck in a hole |
| 6:08PM |
1 |
How many digits are collected |
| 1:51PM |
1 |
Mac Address Authentication Methods |
| 12:27PM |
0 |
wildcard always busy |
| 12:26PM |
2 |
Ignoring the # key on a call |
| 12:13PM |
0 |
DTMF codes in feature.conf not comming through |
| 12:00PM |
2 |
can Digium FXS channels support been half mile to 1 mile length away from phone? |
| 11:50AM |
1 |
BriStuff | HFC-S | Progress() | Early B3 on incoming calls from PSTN |
| 11:10AM |
0 |
High Availability |
| 10:50AM |
1 |
RE: Ever donate Software to Digium? If you did your afool. |
| 10:38AM |
6 |
Integrating Asterisk with a Panasonic D500 using MFC/R2 |
| 10:23AM |
0 |
Autoreply: Re: Autoreply: Tri-Link Technologies? |
| 10:18AM |
1 |
Autoreply: Snom MWI |
| 10:11AM |
0 |
Autoreply: How to, astcc and virtuemart |
| 9:56AM |
1 |
Autoreply: Tri-Link Technologies? |
| 9:55AM |
1 |
Snom MWI |
| 9:54AM |
0 |
How to, astcc and virtuemart |
| 9:54AM |
0 |
Autoreply: e&m wink, TE110P, * answers too soon |
| 9:51AM |
0 |
Autoreply: Deployment for less than 10 phones |
| 9:50AM |
0 |
Tri-Link Technologies? |
| 9:49AM |
2 |
e&m wink, TE110P, * answers too soon |
| 9:46AM |
0 |
Deployment for less than 10 phones |
| 9:32AM |
1 |
ESCAUX releases net.PBX Free Edition |
| 8:48AM |
0 |
Prague PTT? |
| 6:39AM |
0 |
Jabber: Difference between client and component |
| 6:20AM |
4 |
Sipura SPA-3000 vs Sangoma A200 |
| 5:11AM |
1 |
Phone Newbie Questions |
| 4:44AM |
0 |
Ever donate Software to Digium? If you didyoura fool. |
| 3:05AM |
3 |
Two card NT-TE mode |
| 12:29AM |
0 |
UK mobile "reject" codes |
| 12:17AM |
0 |
FW: problem with queues |
| |
| Tuesday August 8 2006 |
| Time | Replies | Subject |
|---|
| 11:59PM |
1 |
realtime+mysql |
| 10:11PM |
7 |
Ever donate Software to Digium? If you did your a fool. |
| 8:13PM |
0 |
hints causing hang in reload |
| 6:52PM |
1 |
Handling inbound and outbound calls passed from a proxy |
| 6:09PM |
2 |
Sangoma A200D and DTMF Detection |
| 5:10PM |
0 |
polycom config script |
| 2:47PM |
0 |
ARA & Regseconds |
| 12:34PM |
1 |
${BLINDTRANSFER}->accountcode ? |
| 12:21PM |
0 |
MP-124 |
| 11:44AM |
1 |
Possible to To Have Different Outgoing VM Messages, but One Mailbox? |
| 11:24AM |
2 |
HFC-S Cards in the UK |
| 10:33AM |
1 |
Host 207.174.202.2 failed to authenticate as teliax |
| 10:20AM |
0 |
IAX Trunking Only ONE-WAY |
| 10:10AM |
4 |
polycom headset question |
| 9:47AM |
1 |
PRI Connection in Lima, Peru |
| 9:37AM |
6 |
Polycom 1.6.7 firmware? |
| 9:26AM |
1 |
A question about AGI and RECORD FILE |
| 8:47AM |
1 |
Asterisk and failover |
| 8:37AM |
0 |
Zaptel trunk failed to compile - Still but another error |
| 8:21AM |
0 |
Cisco Phone Configuration Tool cannot find Files |
| 7:58AM |
0 |
Probelm with IAX peers |
| 7:53AM |
0 |
Stopping Queue after nobody picked up the call . . |
| 6:47AM |
0 |
V: IAX trunk behing NAT with dynamic IP |
| 6:45AM |
2 |
Jitterbuffer on SIP |
| 6:38AM |
0 |
Asterisk with BT's broadband voice service. |
| 6:07AM |
0 |
help with app_sms and chan_capi |
| 5:16AM |
1 |
SV: IAX trunk behing NAT with dynamic IP |
| 5:09AM |
0 |
IAX trunk behing NAT with dynamic IP |
| 4:08AM |
2 |
codec_g729a.so coredump in SVN trunk |
| 3:46AM |
4 |
RE VoipNow 1.2.0 Beta |
| 3:07AM |
2 |
Problems with Codecs in Asterisk |
| 2:55AM |
3 |
AGI doesn't execute PHP5 script |
| 2:30AM |
2 |
set minimum iax jitterbuffer |
| 1:06AM |
0 |
ISDN Y cable |
| 12:50AM |
1 |
Bluetooth phone as FXS/FXO with asterisk? |
| |
| Monday August 7 2006 |
| Time | Replies | Subject |
|---|
| 11:25PM |
1 |
problem- 0:10 long message |
| 7:33PM |
1 |
Polycom 301 and Linksys SRW224P PoE Switch |
| 6:49PM |
3 |
agi script runs even if no answer |
| 6:49PM |
0 |
(no subject) |
| 4:26PM |
1 |
PROBLEM MUSIC ON HOLD |
| 3:16PM |
1 |
Music On Hold Class Not Makin' Sense |
| 2:53PM |
0 |
NVFaxDetect and 1.2.10 |
| 2:45PM |
0 |
SIP musicclass |
| 2:25PM |
1 |
Re: Meetme chat room with many users, and only 4 can talk, is there a max amount of users? |
| 1:51PM |
0 |
Voicemail Platform |
| 1:44PM |
1 |
voicemail in mp3 format |
| 1:08PM |
0 |
FXS gateway/Channel Bank |
| 12:02PM |
2 |
sip incoming stop working, what to look for in logs? |
| 11:56AM |
1 |
MOH Silence |
| 11:32AM |
0 |
"Off-circuits are busy now. Please try your call again later" |
| 10:15AM |
1 |
res_sqlite problems |
| 9:50AM |
1 |
E1 for Voice and Data with MFC/R2 |
| 9:15AM |
1 |
looking to pay a consultant to help with my asterisk installation |
| 9:14AM |
7 |
By week extension dialing |
| 8:35AM |
1 |
Fwd: * and GTalk testing |
| 8:28AM |
5 |
Hotels... |
| 7:42AM |
0 |
Re: [asterisk-dev] Tuning Software Echo Cancellers |
| 7:16AM |
1 |
Conditional branching |
| 6:47AM |
2 |
Ragi without rails possible ? |
| 6:43AM |
3 |
DTMF problems |
| 5:37AM |
1 |
Re: [asterisk-dev] Questions regarding g.729 and g.711 in Asterisk |
| 5:07AM |
0 |
Caller ID problem on TDM400 FXO |
| 5:01AM |
0 |
Inbound problems, no audio |
| 4:25AM |
0 |
G729, IAX, polycom - trying to using 2 codecs |
| 3:51AM |
1 |
Ztdummy - No audio in BackGround function |
| 3:35AM |
1 |
SER + Asterisk PSTN calls don't hung up |
| 1:27AM |
3 |
Video Conferencing over Asterisk |
| 12:47AM |
1 |
New people in this world and his problem with ISDN |
| |
| Sunday August 6 2006 |
| Time | Replies | Subject |
|---|
| 8:55PM |
1 |
HP ProLiant and Digium 24xxp |
| 7:31PM |
2 |
How to "emulate" Music on Hold in a PHP AGI script? |
| 5:28PM |
1 |
Variables sip redirects and call forward |
| 3:57PM |
1 |
Anyone use TACI.pl for a click to call app? - Doesn't seem to want to work for me |
| 12:10PM |
0 |
AG-168V not registering. |
| 11:36AM |
1 |
Ring Groups |
| 10:21AM |
0 |
previous reload of asterisk did not finish |
| 6:03AM |
2 |
Using a DB for Configurations |
| |
| Saturday August 5 2006 |
| Time | Replies | Subject |
|---|
| 11:59PM |
2 |
for some of my users, VoiceMail is being cutoff when leaving message |
| 8:17PM |
1 |
Linksys SPA-3000 Administration Guide |
| 6:16PM |
2 |
g729 and trafic |
| 5:43PM |
2 |
Help with perl AGI script |
| 2:18PM |
1 |
how to check the status of a channel |
| 10:26AM |
0 |
load average with MOH |
| 8:45AM |
1 |
Japanese Sound Files |
| 8:26AM |
0 |
[Solution] Call Asterisk from GoogleTalk and have it tell you the status of your IAX2 links. |
| 6:31AM |
4 |
Fax tone detected, but no fax extension for CAPI |
| 4:13AM |
3 |
cisco 2600 |
| 1:19AM |
1 |
Help - call recording being cut short if transferred |
| 12:09AM |
3 |
autocreatepeer in iax |
| |
| Friday August 4 2006 |
| Time | Replies | Subject |
|---|
| 7:07PM |
2 |
Check call duration on active call in CLI? |
| 3:59PM |
3 |
Setting CALLERID on a residential telco line |
| 3:55PM |
0 |
Aastra VLAN issues |
| 3:29PM |
0 |
How to play music on hold from within PHP AGI scripts? |
| 3:17PM |
1 |
Simple config question |
| 2:20PM |
5 |
Asterisk and Siemens Legacy PBX |
| 1:56PM |
0 |
DISA + Voicemail + DTMF |
| 1:31PM |
0 |
Mediatrix 1204 and Asterisk 1.2.10 |
| 1:19PM |
1 |
Running AGI in background |
| 12:56PM |
1 |
Steve Totaro I am trying to reach you. |
| 11:45AM |
1 |
Dialplan routing based on CallerID |
| 10:59AM |
1 |
Problems with monitor / mixmonitor stopping if using Local channels |
| 10:51AM |
1 |
Is the manager good for high traffic?? but only with one connection to it |
| 10:49AM |
1 |
Festival Not Working |
| 10:11AM |
0 |
Jabber questions |
| 9:38AM |
1 |
AgentCallBackLogin+Queue |
| 9:15AM |
3 |
(OT) DS3 Barrel/T-connector/Adtran MX2800 Problems |
| 8:23AM |
1 |
Sangoma A200 and Disconnected Cables |
| 7:57AM |
3 |
SIP/Qualify |
| 6:01AM |
0 |
sendtext() to another machine |
| 4:03AM |
0 |
SV: Help debugging strange asterisk behaviour (update) |
| 3:58AM |
2 |
speech gaps with iax2 |
| 3:23AM |
3 |
Load balancing of IAX2 |
| 3:17AM |
1 |
How to connect Snom softphone from my home? |
| 3:16AM |
1 |
Configuring meetme recording quality (8kHz to 32kHz or higher) |
| 2:10AM |
0 |
ANI agi |
| 1:33AM |
1 |
Asterisk with AVM B1 and HFC |
| 1:01AM |
2 |
asterisk dosenot compile |
| 12:19AM |
0 |
Asterisk@Home, call reporting and performance |
| |
| Thursday August 3 2006 |
| Time | Replies | Subject |
|---|
| 6:56PM |
0 |
Opinions on Rhino PCI FXO cards |
| 4:03PM |
0 |
New UK prompts |
| 3:48PM |
1 |
Problem dialing out with a TDB400P |
| 3:43PM |
0 |
Encoding recorded queue calls to mp3 |
| 3:23PM |
0 |
trinary expression |
| 3:09PM |
2 |
Prevent a Polycom contact list to be overwritten |
| 3:01PM |
1 |
Detecting voicemail from CO on FXO |
| 1:51PM |
1 |
Echo cancell |
| 1:33PM |
2 |
Run a script at certain CLI writes |
| 1:32PM |
2 |
Using Flite in a call file. |
| 12:16PM |
2 |
Detecting voicemail from CO on FXO port |
| 12:01PM |
1 |
Detecting voicemail from CO on FXO port andpassing to H.323 phone. Possible? |
| 11:48AM |
3 |
How to forward a call to an outside line |
| 10:49AM |
0 |
Queue bug: When 2 callers call in, only one is processed until the first is answered |
| 10:22AM |
0 |
Forbidden - wrong password on authentication for INVITE |
| 9:29AM |
2 |
IAX Variables |
| 9:19AM |
2 |
VoiceMail being cutoff when leaving message |
| 8:58AM |
0 |
Reboot Mediatrix |
| 8:39AM |
1 |
MoH native volume |
| 8:21AM |
1 |
wip 300 opensource code - changes to support SIP MESSAGE |
| 7:20AM |
2 |
Ringing all extensions |
| 6:19AM |
0 |
volume adjustment? |
| 3:47AM |
4 |
What I can use with ASTERISK to call clients to remind them about their appointments |
| 3:29AM |
1 |
How to check if channel varaible have been set/not empty? |
| 3:18AM |
1 |
queue in realtime |
| 3:16AM |
1 |
Garbled initial voicemail prompt |
| 1:14AM |
1 |
IAX2 Trunking CPU usage |
| |
| Wednesday August 2 2006 |
| Time | Replies | Subject |
|---|
| 11:22PM |
0 |
BRIDGEPEER and DIALEDPEERNAME empty |
| 9:48PM |
1 |
[OT] FYI: Polycom phone intermittent disconnects |
| 6:38PM |
4 |
About Digium cards and HP DL servers |
| 6:13PM |
1 |
Asterisk 1.0.1 in SuSE 10.0 |
| 5:58PM |
1 |
Subject: Slow dialing from PBX via E1 |
| 5:01PM |
0 |
IP JitterBuffer for 1.2.5 |
| 3:00PM |
6 |
need dialout help in python script |
| 2:28PM |
0 |
RE: Asterisk with VoIP phone (shadowym) |
| 2:10PM |
0 |
chan_zap.c: Failed to read gains: Invalid argument |
| 1:10PM |
0 |
Ateus Easy gate call progress |
| 11:45AM |
2 |
creidt card processing sripts for asterisk |
| 10:56AM |
1 |
IAX Trunking |
| 10:46AM |
3 |
Rookie question, trying to learn |
| 10:44AM |
0 |
GSM analogue router |
| 10:36AM |
0 |
Strange behavior with SIP registration/connectivity |
| 10:10AM |
1 |
asterisk optimizing |
| 10:00AM |
0 |
DTMF intermittent on menu. |
| 9:54AM |
1 |
canreinvite=yes and RTP dropping in and out |
| 9:50AM |
3 |
Limitations of IAX |
| 9:27AM |
3 |
SIP_HEADER() read-only |
| 8:25AM |
0 |
- Coder Needed for Patch |
| 8:22AM |
1 |
Issue with IAX2 and Real Time configuration |
| 8:09AM |
1 |
Arrays ??? |
| 7:52AM |
2 |
Asterisk, Linksys SPA-3000 echo |
| 7:07AM |
0 |
Playback() does not work |
| 7:06AM |
1 |
Call Routing based on Caller-Id |
| 7:01AM |
0 |
Dell Poweredge 1950 / 2950 |
| 6:50AM |
3 |
polycom soundstation 501 crash |
| 5:52AM |
4 |
FXO module burn out !? |
| 5:42AM |
0 |
Follow ON calling on DISA |
| 5:03AM |
0 |
newbie - suggestions on installing Asterisk for SOHO |
| 1:08AM |
0 |
Slow dialing from PBX via E1 |
| 12:29AM |
1 |
Asterisk config with Analouge Audio Codec model number MP108FXS |
| 12:27AM |
1 |
Problem with Cisco7970 SIP load / call transfer |
| 12:08AM |
0 |
SV: VOIP phone for Receptionist use |
| |
| Tuesday August 1 2006 |
| Time | Replies | Subject |
|---|
| 10:51PM |
1 |
SER local as an Asterisk Trunk |
| 7:35PM |
0 |
softhangup() problem |
| 7:05PM |
0 |
A2Billing - destination |
| 5:44PM |
0 |
ANNOUNCE: libss7 |
| 5:23PM |
0 |
RE: asterisk-users Digest, Vol 25, Issue 2 |
| 5:20PM |
5 |
VOIP phone for Receptionist use |
| 5:03PM |
2 |
Asterisk with VoIP phone |
| 3:52PM |
1 |
Polycom IP600 HTTP Provisioning problem |
| 2:36PM |
3 |
Unicall stack, right versions? |
| 2:05PM |
3 |
Dundi and Dial Arguments |
| 1:28PM |
1 |
rx_fax problem |
| 1:13PM |
1 |
ISDN incoming call - inband info and announcements BEFORE ANSWER |
| 12:52PM |
0 |
Line drops |
| 12:28PM |
0 |
MySQL 5.0+ and the MySQL addon - Can use stored procedures? |
| 11:14AM |
4 |
IAX and Accountcode |
| 9:59AM |
0 |
Codec selection / IAX tunnels |
| 9:54AM |
0 |
Controllable hold music |
| 9:45AM |
1 |
Extend analog phone via SIP (OT) |
| 8:52AM |
0 |
Park / ParkAndAnnounce |
| 8:46AM |
1 |
Problem with distortion of initial voicemail prompt |
| 8:27AM |
3 |
SV: Help debugging strange asterisk behaviour |
| 8:16AM |
0 |
Media direct from IAX Phone to IAX Phone |
| 8:11AM |
1 |
Help debugging strange asterisk behaviour |
| 8:01AM |
1 |
Missing Fast AGI calling 'h' exten without hanging up |
| 7:22AM |
1 |
AddQueueMember and Local channel |
| 7:16AM |
3 |
*****SPAM***** Is there a smarter way to ban expensive calls in dial plan? |
| 6:24AM |
2 |
nat and qualify questions |
| 6:02AM |
0 |
SoftHangup with Polycom_acd_functions release of asterisk |
| 4:40AM |
0 |
Permission for files generated by voicemail |
| 2:48AM |
0 |
SRTP help |