| Saturday September 30 2006 |
| Time | Replies | Subject |
|---|
| 8:54PM |
2 |
Dial-9 (was Extension Numbering) |
| 6:08PM |
1 |
Install issues on Freebsd 6.1 with Asterisk 1.4.0-beta2 |
| 12:36PM |
2 |
Where are the kernel sources? |
| 12:27PM |
3 |
Google talk and Asterisk 1.4 |
| 9:49AM |
0 |
Asterisk Fax Server |
| 9:42AM |
0 |
PRI channel becomming unavailable |
| 9:12AM |
2 |
Issues with Monitor in 1.4? |
| 4:10AM |
2 |
How do I reset a password? |
| 1:30AM |
1 |
Realtime; User not registering {Urgent} |
| |
| Friday September 29 2006 |
| Time | Replies | Subject |
|---|
| 8:07PM |
2 |
480i phone: Is there a trick to registering with *?? |
| 5:22PM |
1 |
SIP phones not talking |
| 3:44PM |
0 |
FW: 480i phone: Is there a trick to registering with * ?? <--forg ot my sip.conf |
| 3:39PM |
0 |
480i phone: Is there a trick to registering with * ?? |
| 1:13PM |
0 |
Fiber Outage: LA->Washington |
| 12:44PM |
0 |
native sounds |
| 12:05PM |
0 |
A Step back with AddQueueMember() ? |
| 11:54AM |
1 |
Problems with DISA |
| 11:50AM |
0 |
Re: asterisk-users Digest, Vol 26, Issue 172 |
| 11:20AM |
1 |
recommended application for salesman using asterisk |
| 11:12AM |
0 |
Queue and Pickup/DPickup |
| 10:25AM |
0 |
manager api redirect dropping calls |
| 10:20AM |
1 |
Asterisk IVR .wav issue |
| 10:10AM |
0 |
[Fwd: [Fwd: [Fwd: asterisk-users Digest, Vol 26, Issue 166]]] |
| 9:49AM |
0 |
SPA3000 register in asterisk |
| 9:42AM |
3 |
Any Suggestions for Election Polling Application? |
| 9:13AM |
4 |
real time billing system |
| 7:49AM |
0 |
What minimum required packages for 1.4 |
| 7:32AM |
0 |
Off-hooking Snom hanset doesn't answer incom ing call |
| 7:31AM |
0 |
[Fwd: asterisk-users Digest, Vol 26, Issue 166] |
| 7:23AM |
1 |
Extension Numbering |
| 6:33AM |
0 |
attended transfer unreliable |
| 6:11AM |
0 |
Polycom IP430 HUM and Echo |
| 6:07AM |
4 |
Sip answer one side , ring other side |
| 5:54AM |
3 |
Replacing mpg123 with madplay under Solaris? |
| 5:16AM |
1 |
VMware and Digium TDM400P card |
| 3:30AM |
1 |
Off-hooking Snom hanset doesn't answer incoming call |
| 2:51AM |
0 |
re: asterisk/SER integration - HELP |
| 2:33AM |
0 |
Difference between "SIP Server" and "Outbound Proxy" |
| 2:30AM |
1 |
Sirrix to Legacy PBX |
| 1:08AM |
2 |
pstn failback |
| 12:13AM |
0 |
Is this phone any good |
| |
| Thursday September 28 2006 |
| Time | Replies | Subject |
|---|
| 9:03PM |
3 |
t1-pri or sip trunk? |
| 6:32PM |
0 |
SPA3000 and asterisk |
| 3:23PM |
0 |
Digium G.729 codec binaries updated for Asterisk 1.4 on Solaris |
| 2:46PM |
0 |
7970G SIP8-0-4 not registering with asterisk |
| 2:10PM |
5 |
Forcing Transcode |
| 12:47PM |
0 |
Asterisk lockup and boot error after one day? |
| 12:23PM |
4 |
Queue AddQueueMember() |
| 12:17PM |
2 |
WAS: 64 analog phones NOW: Selection criteria and recipie for a good Asterisk install [long] |
| 11:33AM |
0 |
You are not the next caller |
| 11:18AM |
1 |
Yet another processor question |
| 11:02AM |
1 |
Polycom Queues, Login, Logout etc |
| 10:56AM |
0 |
AstriCon 2006 Reminder / Hotel Selling Out |
| 10:37AM |
5 |
Cisco CAll Manger and H323 |
| 9:27AM |
0 |
Voicemail callback bug? |
| 9:11AM |
1 |
T1 incoming connects, but no sound |
| 8:14AM |
0 |
Early rtp bridge and reINVITE in 1.4b2 |
| 8:12AM |
3 |
Asterisk -> Tekelec T6000 (Vocaldata, voiss) |
| 7:45AM |
4 |
WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long] |
| 7:32AM |
5 |
Polycom 501 One-way Audio |
| 7:29AM |
0 |
SV: txfax reliability on TDM cards |
| 7:05AM |
4 |
extensions.conf strangeness |
| 6:50AM |
0 |
txfax reliability on TDM cards |
| 6:36AM |
2 |
unable to call AT&T audio conference bridge |
| 6:24AM |
1 |
asterisk on 2.4 kernel ... scheduler problem? |
| 6:03AM |
1 |
New ptlib dependency-requirement in SVN-trunk? |
| 6:02AM |
1 |
Master.csv has stopped writing call logs. |
| 6:01AM |
0 |
X100M location in circuit requirement? |
| 5:28AM |
0 |
Asterisk <=> E1 <=> Alcatel OXO |
| 4:54AM |
2 |
Is this phone any good? |
| 3:55AM |
1 |
How to enable jingle in 1.4beta2? |
| 3:40AM |
2 |
7940 vs. 7941 |
| 3:24AM |
0 |
No answer time |
| 3:04AM |
0 |
Sangoma a301 |
| 3:03AM |
0 |
get the value from CDR |
| 2:34AM |
0 |
How does SIP work? |
| 2:06AM |
1 |
quadbri + tdm400p + modem-fax |
| 1:23AM |
1 |
importance of crc4 in zaptel.conf? |
| 12:52AM |
4 |
Multiple asterisk same GUI |
| 12:35AM |
0 |
corrupt faxes |
| 12:15AM |
1 |
pay as you go t.t38 fax termination and origination |
| |
| Wednesday September 27 2006 |
| Time | Replies | Subject |
|---|
| 11:44PM |
0 |
media stream count |
| 8:08PM |
1 |
Asterisk Hangups on PRI Interface |
| 5:34PM |
1 |
problem with trying to use two extensions for different announcements |
| 4:22PM |
1 |
SIP on Asterisk, new install |
| 3:37PM |
0 |
MWI on 1.4 Beta |
| 3:34PM |
1 |
SMS Text Send working with BT Text in the UK?? |
| 3:24PM |
1 |
How to detect dial tone on ZAP channel before dialling using TDM2400P |
| 2:06PM |
2 |
UK Colocation services |
| 1:27PM |
2 |
Are you using app_meetme or app_conference |
| 1:20PM |
3 |
Spurious hangups on zaptel interface |
| 1:13PM |
4 |
RPID |
| 12:40PM |
1 |
TDM400P Problem or Not? |
| 12:15PM |
3 |
RE:T1 timing errors Nortel 61C with TE110P |
| 12:12PM |
0 |
Cisco ATA escaping # into %23? |
| 11:41AM |
0 |
txfax question |
| 11:34AM |
0 |
Linksys/Sipura 3K, Calls Timing Out |
| 10:00AM |
0 |
Any suggestions about VoIP provider? |
| 9:53AM |
0 |
FW: Re: asterisk to cell phone network |
| 9:45AM |
1 |
Queue Status via Dialplan |
| 9:42AM |
0 |
Zapata.conf |
| 9:02AM |
6 |
SER with multiple asterisk deployment |
| 8:04AM |
0 |
TDM04B Installation Problem |
| 8:03AM |
2 |
IAX phones? |
| 7:42AM |
0 |
Outgoing DialPlan |
| 6:29AM |
1 |
Good Book on Asterisk |
| 6:06AM |
2 |
Right way to prevent analog channel from answering the phone? |
| 6:05AM |
1 |
T1 timing errors (Frame Slips) on Nortel 61C to TE110P |
| 2:56AM |
2 |
ISDN30 and digital phones |
| 2:26AM |
0 |
High CPU usage when Internet goes down |
| 1:08AM |
0 |
Re: Advice of charge |
| 12:49AM |
0 |
How can I unistall Asterisk? |
| 12:48AM |
0 |
Configuring Asterisk 1.4-beta2 to work with jingle |
| 12:44AM |
6 |
Voip Buster - CID |
| 12:30AM |
0 |
my (SIP) INVITE is ignored |
| |
| Tuesday September 26 2006 |
| Time | Replies | Subject |
|---|
| 8:50PM |
0 |
WARNING: chan_sip.c add_realm_authentication: ??? |
| 8:43PM |
2 |
Anybody have the opvx1200.c driver? |
| 7:57PM |
0 |
queue information |
| 7:53PM |
5 |
max number of devices in hint |
| 6:31PM |
3 |
How to change pager notification message |
| 4:18PM |
1 |
Priority "n" |
| 4:12PM |
0 |
Grandstream GXV 3000 |
| 4:10PM |
1 |
Context default & incoming ENUM |
| 3:43PM |
0 |
Problem with "Background" DTMF detection with A200D |
| 3:33PM |
3 |
señalizacion te110p, signaling te110p |
| 3:12PM |
1 |
mISDN, 2 Billion HFC ISDN cards, cannot dial or receive |
| 2:46PM |
1 |
speaker phone echo |
| 2:26PM |
1 |
IAX2 & SIP Monitoring Solution for Asterisk |
| 12:05PM |
1 |
TE406P not working on Intel D101Ggc motherboard. |
| 11:50AM |
0 |
voicemailmain menu |
| 11:35AM |
0 |
Rewriting CID number w/o changing CDR src field |
| 11:11AM |
0 |
H323 IP phones |
| 10:50AM |
2 |
SIP Gateway |
| 10:41AM |
1 |
Included context |
| 10:23AM |
4 |
PRI Outbound CallerID Question |
| 9:48AM |
4 |
Asterisk behind Sonicwall firewall |
| 9:18AM |
0 |
X100P Clone card in JAPAN |
| 8:04AM |
0 |
Is there T.38 support on asterisk 1.4 beta2 ??? |
| 7:55AM |
0 |
Play wav file during conversation |
| 7:30AM |
1 |
Asterisk 1.4 mohsuggest |
| 5:32AM |
5 |
How can I stop lost DNS from killing Asterisk? |
| 3:14AM |
0 |
error 407 authenticate on INVITE |
| 3:06AM |
3 |
asterisk 1.4 branch and chan-capi-0.7.0 |
| 2:40AM |
1 |
asterisk - alcatel |
| 1:55AM |
1 |
Cannot make outbound calls to some numbers with PRI line from legacy PBX thru Asterisk with Sangoma |
| 1:34AM |
1 |
core dump with 1.2.7.1 and chan-capi-cm 0.6.5 |
| 12:54AM |
0 |
Asterisk and UMTS phones |
| 12:53AM |
0 |
The best way to track no-audio calls |
| 12:52AM |
0 |
Changing the recording dir of MeetMe recordings. |
| |
| Monday September 25 2006 |
| Time | Replies | Subject |
|---|
| 10:49PM |
0 |
Unrecognized frames |
| 9:31PM |
1 |
Ericsson MD110 |
| 8:26PM |
1 |
rtc: lost some interrupts at 1024 when loading ztdummy |
| 4:30PM |
1 |
"does /var/run/asterisk.ctl exist?" -- but Asterisk *is* running. |
| 3:55PM |
0 |
fw: Uniden - TVUNIDEN_UIP300 |
| 3:32PM |
2 |
MOH in 1.4 - Still Broken? |
| 3:01PM |
1 |
Shared Line Appearances in 1.4 |
| 2:56PM |
1 |
Extensions busy/congested and "circuit-busy" |
| 1:12PM |
1 |
Network impairment tools |
| 1:02PM |
1 |
include "context" |
| 12:56PM |
0 |
How to stream audio to external app for speech recognition and recognize dtmf in parallel ? |
| 12:46PM |
2 |
TDM2400P vs Sangoma A200 |
| 12:07PM |
1 |
trixbox t38 pass through |
| 10:19AM |
0 |
can someone recommened a reliable, cheap t38 origination/termination provider |
| 10:06AM |
0 |
Asterisk 1.4 autoconf and /etc/asterisk directory |
| 10:02AM |
1 |
Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.209 |
| 9:57AM |
1 |
DUNDi Servers |
| 9:53AM |
0 |
PBX TDA620 AND TE110P |
| 9:38AM |
7 |
Running Multiple Instances of Asterisk |
| 7:43AM |
0 |
REQUERIMIENTOS TE110P Y PANASONIC TDA620 |
| 7:35AM |
4 |
asterisk to cell phone network |
| 7:31AM |
1 |
Queue failover and wrap time |
| 7:25AM |
8 |
OT: Opinions on Aastra 480i CT? |
| 6:54AM |
1 |
ztcfg / X100P question |
| 6:15AM |
1 |
progress problems from SIP to PRI |
| 5:45AM |
0 |
Asterisk Trunk with Alcatel 4200 PABX |
| 4:05AM |
1 |
Line Pickup Problem |
| 3:16AM |
0 |
A Strange doubt and problem |
| 3:13AM |
2 |
Cisco 7970 - DTMF |
| 3:02AM |
7 |
voicemail greeting |
| 2:40AM |
0 |
AgentCallbacklogin in Asterisk1.4 beta2 |
| 1:06AM |
0 |
ougoing calls problem |
| 12:43AM |
1 |
Snom MWI not turning off when message picked up. |
| |
| Sunday September 24 2006 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
High utilization with SIP registration |
| 8:06PM |
1 |
Need a recommended T38 FOIP solution |
| 4:00PM |
1 |
Rpath PoundKey 1.2 |
| 1:58PM |
2 |
spandsp (foip) |
| 11:31AM |
0 |
how to configure a sip service |
| 11:05AM |
2 |
2 CPU's, Only 1 taking IRQ's |
| 10:33AM |
2 |
Missing sound in spanish from 1.4 beta2 |
| 10:24AM |
1 |
iaxy: one way audio |
| 8:37AM |
0 |
Asterisk+Astbill |
| 6:43AM |
0 |
dialplan for confrencing |
| 5:46AM |
0 |
running ooh323 on asterisk-1.14beta2 |
| |
| Saturday September 23 2006 |
| Time | Replies | Subject |
|---|
| 9:04PM |
2 |
Segmentation fault on Asterisk startup: res_config_mysql.so problem? |
| 8:45PM |
1 |
e911 |
| 7:26PM |
1 |
fax over ip |
| 1:00PM |
0 |
lumenvox speech recognition |
| 12:23PM |
4 |
Problem with zaptel 1.4b2 and X101P Wildcard |
| 12:12PM |
0 |
Conference Call Delay & Quality |
| 12:08PM |
0 |
Debugging and Outbound SIP Trunk |
| 11:46AM |
0 |
One server SUBSCRIBE for information on multiple voicemail boxes |
| 11:44AM |
0 |
Connecting Motorola VT2442 Device |
| 11:24AM |
2 |
libpri current extracts as beta1 |
| 8:43AM |
6 |
1.4 Beta 2 Config Problem |
| 5:18AM |
5 |
Trixbox Documentation |
| 4:57AM |
2 |
Cisco 7960 Double Natted |
| |
| Friday September 22 2006 |
| Time | Replies | Subject |
|---|
| 11:38PM |
1 |
Polycom phone help needed |
| 11:36PM |
2 |
OT But So Ungodly Important |
| 8:02PM |
1 |
iaxy will register, but doesn't detect POTS line |
| 5:10PM |
2 |
Comments on new system plan. |
| 5:03PM |
1 |
make error |
| 3:44PM |
1 |
OT: Anybody remember this from last Dec? |
| 3:22PM |
1 |
Leased line interconnect |
| 2:00PM |
0 |
Polycom (and others) digitmap info |
| 1:27PM |
2 |
Very high ping times from 7960 phones |
| 11:16AM |
6 |
Digium G.729 codec binaries updated for Asterisk 1.4 beta |
| 10:46AM |
1 |
channel.c: Nobody there, continuing... |
| 10:38AM |
0 |
ZAP: psuedo camped on channel 1? |
| 10:12AM |
0 |
Asterisk 1.4-beta2 Spanish Sounds missing vm-youhaveno? |
| 9:37AM |
2 |
SNOM 320 - 404 "Not Found" |
| 9:34AM |
1 |
Re: [asterisk‑users] Integrating Asterisk with LDAP Realtime |
| 9:12AM |
0 |
chan_isdn / chan_sip problems |
| 8:31AM |
2 |
Display message on voip phone...hint? |
| 8:23AM |
0 |
Asterisk & MSN ? |
| 8:16AM |
1 |
dialout-trunk vs. dial group |
| 6:54AM |
0 |
experience with phones locking up uniden and cisco |
| 6:27AM |
0 |
Re: [asterisk-dev] To bweschke regarding app FollowMe |
| 6:23AM |
1 |
hint status from dialplan? |
| 5:48AM |
1 |
Help with Tieing Outbound calls to Zap Channels |
| 5:47AM |
1 |
Re: [asterisk-dev] To bweschke regarding app FollowMe |
| 5:41AM |
0 |
Asterisk ramdonly crash using Realtime Static |
| 5:39AM |
1 |
Question about SVN-trunk-r43322 and Asterisk Recording Interface |
| 4:46AM |
1 |
Polycom phone references needed |
| 4:10AM |
0 |
How can the User Know he has voicemail in the Databases. |
| 3:48AM |
1 |
Does Asterisk 1.4 going to support realtime ex-girlfriend logic? |
| 2:42AM |
0 |
Where to find error codes |
| 2:31AM |
1 |
freepbx dial plan, add and remove at the same time |
| 2:27AM |
10 |
64 analog phones |
| 2:17AM |
2 |
ATA with wireless client |
| 2:04AM |
0 |
INVITE re-try interval |
| 1:52AM |
0 |
Iax2 show netstat |
| 1:46AM |
1 |
alternatives to mpg123: format_mp3, rawplayer or madplay? |
| 1:39AM |
1 |
new in 1.4? |
| 1:04AM |
0 |
E1 - PCI-Express |
| 12:42AM |
7 |
Dual core |
| 12:16AM |
1 |
Fax Detection on outbound call |
| 12:00AM |
0 |
Picking up a call from queue? |
| |
| Thursday September 21 2006 |
| Time | Replies | Subject |
|---|
| 11:39PM |
1 |
Re: Can you explain why multiple registration isan important (missing) feature ? |
| 11:19PM |
2 |
Dynamic DNS asterisk server? |
| 10:28PM |
1 |
Application of Asterisk Packetization Patch |
| 10:04PM |
0 |
iaxy configuration problems |
| 9:40PM |
1 |
multiple zaptel cards |
| 8:24PM |
0 |
zttest output |
| 8:19PM |
1 |
iaxyprov downloading problems |
| 4:21PM |
2 |
SOS building fastagi C |
| 4:16PM |
2 |
Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ??? |
| 4:12PM |
0 |
TDM2400 problem isolated with POLYCOM IP301 phones!!! |
| 2:45PM |
2 |
Integrating Asterisk with LDAP Realtime |
| 2:13PM |
0 |
Dual asterisk, CallerID(name|number) problem |
| 2:11PM |
5 |
DSL router with integrated SIP proxy? |
| 1:30PM |
2 |
IAX or SIP termination provider that reaches6421xxxxxxx? |
| 12:46PM |
0 |
Extensions problems |
| 12:23PM |
2 |
SPA941 -> Asterisk -> Voip provider -> PSTN -> ShoreTel garble |
| 12:17PM |
0 |
Asterisk, Asterisk-Addons, Zaptel and Libpri 1.4 betas released! |
| 11:49AM |
0 |
Mini call center only 15 seats fxs to sip suggestion |
| 11:39AM |
0 |
mISDN problem: no version for "capi_cmd2str" found |
| 11:39AM |
0 |
Asterisk 1.4 Beta Uploaded!!! |
| 11:21AM |
1 |
asterisk and PowerEdge 1950 |
| 10:51AM |
3 |
TDM2400 wired description and skiping frames |
| 10:38AM |
0 |
Asteisk plays music on hold startingfrom randompoint |
| 10:04AM |
0 |
Asterisk and Panasonic D500 |
| 10:00AM |
2 |
Linksys SPA400 |
| 9:41AM |
0 |
Polycom 650 Question |
| 9:39AM |
1 |
Call is dead after featuredigittimeout |
| 9:36AM |
3 |
notransfer local channel on redirect |
| 8:41AM |
8 |
CURL |
| 6:48AM |
1 |
Asteisk plays music on hold starting from randompoint |
| 6:44AM |
4 |
Looped message playback |
| 6:33AM |
4 |
Setting QOS settings in asterisk and/or CentOS? |
| 6:04AM |
0 |
Calls between IAX2 Clients don't work correctly |
| 5:13AM |
1 |
Using Asterisk with IVR connected with legacy pbx via rs-232 |
| 3:19AM |
6 |
asterisk, iaxmodem, hylafax quality problem |
| 3:16AM |
3 |
Two phones, same number |
| 2:58AM |
0 |
Invite issues |
| 2:55AM |
1 |
asterisk / chan_capi problems |
| 2:47AM |
1 |
asterisk skills in the philippines |
| 2:24AM |
0 |
Habitual set of number |
| 1:40AM |
5 |
Iax Netstat Output |
| 1:26AM |
1 |
Unexpected delay: problem with outgoing calls |
| 12:21AM |
2 |
RTCP and RTP packetization in 1.4 |
| 12:19AM |
0 |
How much SIP calls can I squeeze from this box |
| 12:01AM |
1 |
Help in Reloading of Asterisk... |
| |
| Wednesday September 20 2006 |
| Time | Replies | Subject |
|---|
| 11:27PM |
1 |
Re: Can you explain why multiple registration isan important (missing) feature ? |
| 7:30PM |
0 |
Call/Voicemail Screening |
| 6:50PM |
2 |
Polycom 2.0.1 Software |
| 6:24PM |
2 |
Configuring Codecs |
| 4:45PM |
0 |
X100P compile problems |
| 4:10PM |
0 |
No Sound from VoicemailMain - Device Linksys PAP2T-NA |
| 3:54PM |
0 |
voice detection during playback |
| 3:35PM |
0 |
Asterisk Bussiness Edition and Realtime. |
| 3:07PM |
0 |
Re: Can you explain why multiple registration isan important (missing) feature |
| 1:45PM |
0 |
macro-dialout-trunk without agi or manager |
| 1:04PM |
1 |
A Caller ID question (UK) |
| 1:02PM |
3 |
Cisco 7970 behind NAT |
| 1:00PM |
2 |
PRI Backup |
| 12:56PM |
0 |
ztdummy installed but choppy audio warning |
| 12:11PM |
0 |
Round Robin + Ringall |
| 12:11PM |
0 |
RTT in rtcp debug |
| 11:25AM |
0 |
SPA-3102 PSTN->VoIP Gateway (quest for one stage dialing) |
| 11:06AM |
0 |
Asterisk Voicemail with Sonus? |
| 9:18AM |
2 |
Asteisk plays music on hold starting from random point |
| 8:56AM |
1 |
Getting Music On Hold working in * 1.2.12.1 with Fedora? |
| 8:54AM |
0 |
A-Z termination |
| 8:45AM |
0 |
No channels available after reloading config |
| 8:39AM |
2 |
(no subject) |
| 8:39AM |
4 |
HINT problems with SVN-trunk-r43322 |
| 8:32AM |
0 |
How to register from asterisk server to an xlite. |
| 8:26AM |
0 |
Available channels |
| 8:16AM |
0 |
Zap channel digit. |
| 6:59AM |
1 |
Asterisk capabilities, was University dumps CISCO VoIP for Asterisk |
| 6:41AM |
0 |
Unexpected delay |
| 6:38AM |
0 |
tx_fax over sip to TDM card |
| 6:27AM |
1 |
Sip configuration using mysql |
| 6:23AM |
2 |
MOH distorted on Pound Key Linux on asterisk1.2.8 |
| 6:16AM |
1 |
Realtime madness |
| 5:48AM |
2 |
Incoming calls, identify |
| 5:34AM |
0 |
Forwarding the Ring Group and Calls coming in to Queues |
| 5:16AM |
1 |
enumlookup - deprecated working - but appreciated one duznt :-( |
| 5:03AM |
4 |
University dumps CISCO VoIP for Asterisk |
| 4:29AM |
0 |
Register doubt |
| 3:55AM |
0 |
Registration doubt |
| 3:33AM |
0 |
Mediant 1000 |
| 3:09AM |
1 |
Channel kept busy when creating ssh tunnel via AGI |
| 2:44AM |
1 |
BRI: Asterisk disconnecting on 'call diverted' message? |
| 2:39AM |
2 |
stress a server with a tool |
| 1:07AM |
4 |
Uninstalling Trixbox |
| |
| Tuesday September 19 2006 |
| Time | Replies | Subject |
|---|
| 7:07PM |
2 |
SkypeOut with Asterisk? |
| 5:34PM |
2 |
MOH distorted on Pound Key Linux on asterisk 1.2.8 |
| 4:58PM |
1 |
Polycom 500 power supply |
| 4:29PM |
1 |
Cisco 7960 part numbers ... |
| 3:29PM |
2 |
IAX or SIP termination provider that reaches 6421xxxxxxx? |
| 2:39PM |
3 |
g729 and polycoms problem |
| 2:37PM |
1 |
Grandstream SX2000 attended tranfer |
| 2:29PM |
1 |
DTMF Detection Problems with certain phones incoming zap channels |
| 1:29PM |
2 |
Asterisk AGI question |
| 1:00PM |
0 |
clustering asterisk is possible ? |
| 12:15PM |
1 |
SPA 3102 does not even attempt to register |
| 11:55AM |
1 |
Pri Event 6 and 8 |
| 11:07AM |
3 |
grandstream gxp 2000 does not display names when calling out |
| 10:44AM |
1 |
gTalk no audio issue |
| 10:33AM |
0 |
Repost: Register message received from realtime peer crashes Asterisk |
| 10:16AM |
3 |
SIP "Lines" Example Citel |
| 10:10AM |
1 |
codecs/voicemail/DTMF |
| 8:44AM |
0 |
Call forward with CFU? |
| 8:36AM |
1 |
Semi-OT: SIP or IAX provider in the Boston area? |
| 8:35AM |
2 |
When does Scalability requests Asterisk |
| 8:23AM |
3 |
Polycom default handset volume |
| 8:00AM |
1 |
transcoding error? |
| 7:18AM |
1 |
fast SIP failover (outgoing sIP requests) wi th 1.2 |
| 7:03AM |
1 |
fast SIP failover (outgoing sIP requests) with 1.2 |
| 6:18AM |
3 |
Problem with # locking up call |
| 5:57AM |
1 |
polycom 501 digitmap |
| 5:46AM |
6 |
Format_MP3, Streaming, File Formats, MOH |
| 5:12AM |
2 |
Aastra 9133i and Atcom AT-320 - Comments please |
| 5:00AM |
0 |
Clustering architecture and echo cancellation issue |
| 4:25AM |
0 |
Query ,NEED help regarding MWI |
| 4:06AM |
0 |
Anyone Using a Patton (Inalp) SmartNode 2400 for T.38? |
| 2:07AM |
2 |
Alcatel OXO Sip |
| 2:01AM |
1 |
When does Scalability requests Asterisk to Use SER ? |
| 1:28AM |
0 |
Wrong call handling |
| 1:14AM |
1 |
How to Dial a number with Sangoma PRI card? |
| |
| Monday September 18 2006 |
| Time | Replies | Subject |
|---|
| 11:34PM |
0 |
prompt playing problem |
| 11:16PM |
0 |
Query on MWI |
| 10:53PM |
1 |
488 Not acceptable here sent by Asterisk - SIP debug follows |
| 10:24PM |
1 |
Accounting and re-invite |
| 8:34PM |
0 |
spandsp fax using Asterisk 1.2.X |
| 6:19PM |
0 |
create_addr: No such host: |
| 6:09PM |
2 |
Starting Asterisk PBX: FATAL: Module ixj not found. |
| 5:10PM |
1 |
Asterisk Appliance, will Asterisk Business Edition be mandatory? |
| 5:07PM |
2 |
Enabling Second Processor Trashes Audio Quality |
| 4:42PM |
4 |
Digium GUI? |
| 4:36PM |
1 |
How to learn or teach VoIP QoE |
| 4:21PM |
0 |
Periodic announcements & MySQL Realtime |
| 2:29PM |
1 |
ANI and Meetme... |
| 2:19PM |
2 |
sip.conf for talking to other Asterisk machines |
| 12:32PM |
2 |
Asterisk / Audiocodes annoying issue - Seeking Suggestions |
| 12:23PM |
0 |
INSTALL_PREFIX= |
| 11:57AM |
0 |
CSR introduces UniVox reference platform |
| 11:34AM |
8 |
Fedora |
| 11:08AM |
1 |
How to make Polycom 501 go off hook when pressing any digits |
| 10:54AM |
0 |
X100P and zaptel 1.2.8 |
| 10:28AM |
0 |
Changes in extensions.conf handling between 1.2 & 1.4 |
| 9:49AM |
0 |
ASTERFAX |
| 9:14AM |
1 |
LDAP athentication |
| 8:41AM |
1 |
Dial and Timeout |
| 8:12AM |
3 |
Cisco 7940 Problem (Mess) |
| 8:09AM |
0 |
Chanspy crashing server, again |
| 8:02AM |
3 |
Polycom SoundPoint 2.0.1 SIP firmware? |
| 7:39AM |
1 |
unable to change the emailbody for email notification |
| 7:33AM |
0 |
RE : Re: [asterisk-dev] open letter |
| 7:31AM |
1 |
FOP Installation help |
| 7:30AM |
1 |
Re: Mediatrix 1204 trix |
| 7:03AM |
5 |
Chanspy crashing the server, again |
| 6:36AM |
1 |
Asterisk Design Question |
| 6:32AM |
1 |
User authentication |
| 6:02AM |
3 |
Playtones |
| 5:46AM |
0 |
Problem with Asterisk Realtime (MySQL) |
| 5:38AM |
1 |
Variable that gives the SIP channel |
| 5:34AM |
4 |
pickup call little complicated |
| 4:28AM |
0 |
Queue - Agent language |
| 3:13AM |
0 |
disconnect code in featuremap doesn't work on unanswered calls |
| 2:20AM |
1 |
Log out an Agent on RNA |
| 1:53AM |
3 |
is chanisavail command reliable? |
| 12:15AM |
2 |
Xorcom Astribank |
| |
| Sunday September 17 2006 |
| Time | Replies | Subject |
|---|
| 6:44PM |
0 |
Noob question: Packet size |
| 6:06PM |
2 |
Polycom Expansion Module |
| 5:55PM |
0 |
problem installing func_odbc on asterisk 1.2 ... |
| 1:12PM |
3 |
Termination Rates |
| 11:59AM |
0 |
Register message received from realtime peer crashes Asterisk |
| 9:07AM |
3 |
A1200+fxo, anyone using this? |
| 6:27AM |
4 |
Asterisk Server Down |
| 4:45AM |
2 |
Does a "HST Saphir III ML PCI" work with Asterisk? |
| 4:41AM |
0 |
IAX2 audio problem |
| 4:25AM |
2 |
Starting out |
| 1:26AM |
0 |
How does Asterisk determine an incoming SIP Channel name? |
| |
| Saturday September 16 2006 |
| Time | Replies | Subject |
|---|
| 10:13PM |
1 |
system cmd |
| 5:34PM |
1 |
Wrong outgoing port |
| 1:54PM |
2 |
Polycom programmable buttons |
| 1:16PM |
1 |
Calling to PSTN newbie question |
| 12:57PM |
0 |
RE: [Asterisk-video] VXIasterisk is available ! |
| 9:52AM |
0 |
USA Regulatons |
| 6:12AM |
1 |
SHSU asterisk installation? |
| 4:44AM |
2 |
"Ping" a phone |
| 4:15AM |
1 |
read variable from shell script |
| |
| Friday September 15 2006 |
| Time | Replies | Subject |
|---|
| 11:51PM |
1 |
Integrating the Openser for VoiceMail and PBX with Asterisk, For Account |
| 11:05PM |
1 |
Asterisk as a gateway to SER |
| 7:33PM |
1 |
call across 2 asterisks |
| 7:30PM |
1 |
Scaling/Loadbalancing a Call Center and Redundancy |
| 7:28PM |
1 |
amr codec |
| 6:47PM |
6 |
saved.gsm -> Voicemail greeting ?? |
| 6:37PM |
0 |
voxee, callerid and trixbox |
| 4:50PM |
0 |
Help spread the word about Asterisk! |
| 4:46PM |
0 |
AEL2 patch for Asterisk 1.2.12.1 |
| 4:02PM |
0 |
pickupgroup 1 |
| 3:43PM |
1 |
DTMF Tone Not Passing Help |
| 2:20PM |
1 |
FollowMe question |
| 2:19PM |
0 |
Digium G.729 codec now available for Solaris/SPARC |
| 1:31PM |
1 |
Asterisk 1.2.12.1 and Zaptel 1.2.9.1 Released |
| 1:03PM |
1 |
Asterisk variables |
| 12:39PM |
2 |
Reliability of the newer IAXy's |
| 12:17PM |
0 |
New astGUIclient VICIDIAL Release: 2.0.1 |
| 11:45AM |
1 |
Attended transfer and parking calls |
| 11:09AM |
1 |
Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic |
| 10:57AM |
0 |
app_txfax segv fault |
| 10:26AM |
1 |
Voicemail adjustments |
| 10:18AM |
5 |
Asterisk with cisco 7935 |
| 10:11AM |
1 |
Internal message being heard on pstn line |
| 10:02AM |
1 |
Cisco GW & CID Name |
| 9:50AM |
0 |
Branch office interconnect - IAX :vs: SIP? |
| 9:10AM |
0 |
inbound call from GSM gateway: handle_request_invite: Failed to authenticate user |
| 9:09AM |
1 |
ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 8:18AM |
1 |
4-wire analogue interfaces? |
| 7:57AM |
0 |
[asterisk-dev] open letter |
| 7:20AM |
0 |
Polycom 501 - message waiting LED manipulation |
| 6:50AM |
1 |
Issues with AGI+Dial command |
| 6:39AM |
1 |
where download app_txfax? |
| 6:16AM |
0 |
Section '12345678' lacks type |
| 4:48AM |
1 |
Cisco Distinctive ring using alert-info |
| 4:11AM |
0 |
Compile error in Asterisk 1.2.12.1 |
| 3:04AM |
2 |
CDR question with SIP/IAX trunks |
| 2:27AM |
0 |
Setting up imap based voicemail / invalid remote specification |
| 2:16AM |
0 |
491 request pending [2] |
| 2:14AM |
0 |
trying to understand siprealtime & nat/MWI issues |
| 1:55AM |
0 |
Anyone using Voicemail with IMAP Support? |
| 1:34AM |
0 |
AOC - advice of charge |
| 1:18AM |
1 |
Shared Line Appearance, Snom and trunk |
| 12:51AM |
1 |
two safe_asterisk processes on the same PBX??? |
| 12:48AM |
1 |
non-technical, dealing with users giving feedback |
| 12:07AM |
1 |
Bri Card for Asterisk ? |
| 12:05AM |
0 |
Cisco 7961 "dropouts" |
| |
| Thursday September 14 2006 |
| Time | Replies | Subject |
|---|
| 11:26PM |
3 |
Can you explain why multiple registration is an important (missing) feature ? |
| 11:21PM |
1 |
Hangup on Panasonic KX-TEM824 |
| 11:10PM |
3 |
How to download asterisk 1.3 development version |
| 11:09PM |
1 |
Cisco 79xx and vlan |
| 9:45PM |
1 |
ASTERISK HIGH AVAILABILITY |
| 9:02PM |
0 |
Urgent !!!! Unable to make calls from cisco callmanager to asterisk |
| 8:32PM |
1 |
Why not g726-32? |
| 8:12PM |
2 |
G729 and Tribox |
| 6:23PM |
0 |
Asterisk with Addpac 2120 |
| 5:47PM |
0 |
Login user |
| 2:04PM |
3 |
problems with Polycom 500 boot up |
| 1:05PM |
1 |
Asterisk / Patton SmartNode SN2400 Strangeness FYI |
| 12:25PM |
1 |
mISDN versus ZapHFC with BRIstuff |
| 10:45AM |
1 |
how to transfer a caller out of a queue ? |
| 10:17AM |
0 |
OT, Definity G3 Problems with Asterisk (Any Avaya |
| 10:10AM |
0 |
loosing Sipura 841 almost exactly on the hour |
| 9:58AM |
1 |
(Off-topic) Voip number "tracert" |
| 9:57AM |
1 |
How to send DTMF down a channel |
| 9:42AM |
2 |
[asterisk-dev] 491 request pending |
| 9:24AM |
0 |
Incoming SIP provider goes unregistered and never recovers |
| 9:14AM |
2 |
Page() paging application problem |
| 9:12AM |
0 |
Zork & Asterisk; zoip 0.2.0 released |
| 8:58AM |
0 |
491 request pending |
| 8:52AM |
4 |
Asterisk 1.4 Docs |
| 8:41AM |
0 |
incoming call h323 cdr |
| 8:21AM |
4 |
WAIT FOR DIGIT not working |
| 8:03AM |
2 |
Forcing Marker bit, because SSRC has changed |
| 7:55AM |
4 |
BLF across asterisk trunks |
| 7:55AM |
2 |
Controlling the channel |
| 7:39AM |
1 |
Getting 'i' functionality on internal extensions |
| 7:37AM |
3 |
Detect PBX vs Network message |
| 7:26AM |
0 |
Attended Transfer Asterisk 1.2.11 |
| 7:20AM |
2 |
Asterisk and peers behind nat with port forward, how to proxy? |
| 6:49AM |
0 |
Dealing with FINAREA redirects |
| 6:42AM |
0 |
VoiceXML browser for Asterisk available ! |
| 5:45AM |
3 |
voicemail access thru apache on another server |
| 5:13AM |
6 |
asterisk server to server using sip question |
| 4:35AM |
2 |
9 becomes 99 ? And other strangeness |
| 4:21AM |
2 |
Silence Call {very very urgent plz} |
| 3:53AM |
1 |
Thomson 2030 |
| 3:36AM |
0 |
SOLVED: ringback on box with E1 and premicell |
| 3:17AM |
2 |
Correct settings for UK (BT) FXO |
| 2:39AM |
0 |
DIAL and automatic/manual co line acces |
| 2:30AM |
0 |
voicemail ,MWI problem |
| 2:17AM |
0 |
ASTCC: change from no pin to pin request? |
| 1:51AM |
0 |
MWI problem on asterisk |
| 1:44AM |
3 |
One way audio problem on gateway to PSTN after some time, no NAT involved |
| 12:53AM |
1 |
urgently requires help regarding MWI |
| 12:17AM |
0 |
changed behaviour of status indication on incoming pri lines in asterisk 1.2? |
| |
| Wednesday September 13 2006 |
| Time | Replies | Subject |
|---|
| 11:38PM |
1 |
Flexible Wrap Up Time for Queue |
| 10:48PM |
3 |
How to install HUDLite Server |
| 7:14PM |
1 |
QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls |
| 5:35PM |
0 |
QuadBRI and Zyxel Wifi phone stop working together after 3 calls |
| 3:40PM |
0 |
Asterisk as a B2BUA and/or a FXS-SIP gateway? |
| 3:37PM |
13 |
chan_zap.so stopped working after upgrading CentOS |
| 3:27PM |
1 |
Copyright issues with libcurl and OpenSSL |
| 2:56PM |
2 |
Via Epia platforms and asterisk |
| 1:30PM |
2 |
OT -- echo cancellation of an audio file |
| 1:10PM |
1 |
Polycom 501 display bug |
| 12:57PM |
0 |
Jitter Buffer on SIP |
| 12:52PM |
1 |
Astmanproxy authentication problems |
| 12:19PM |
0 |
success |
| 12:02PM |
1 |
Asterisk crashing when monitoring SIP device with Chanspy |
| 10:01AM |
1 |
Dropped Calls on TDM400p |
| 9:52AM |
4 |
University switches to Asterisk |
| 9:46AM |
2 |
callback without agi |
| 9:41AM |
0 |
Third Lane PBX Manger Multi-Tenant |
| 9:00AM |
3 |
Building Zaptel 1.2.9 with Octasic |
| 8:32AM |
1 |
set global variable |
| 8:20AM |
0 |
audio drop out half channel |
| 7:31AM |
0 |
ip address incoming call |
| 7:08AM |
1 |
I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it? |
| 6:51AM |
2 |
Streaming MoH Problem, starts and then stops immediately |
| 6:30AM |
0 |
Customize host in INVITE's Contact header? |
| 6:27AM |
0 |
sample configuration |
| 6:12AM |
1 |
HFC isdn card and bristuff 0.2.0 rc8n |
| 6:11AM |
0 |
Problems with call parking |
| 5:33AM |
0 |
Problem withfeature introducer as first digit on a call |
| 4:46AM |
3 |
Queue - static members |
| 4:27AM |
2 |
voicemailmain errors on CLI |
| 4:24AM |
0 |
Asterisk and "305 Use Proxy" |
| 4:08AM |
0 |
(no subject) |
| 3:54AM |
0 |
Adding own info in AMI |
| 3:53AM |
11 |
IVR not able to Play the Balance.. need some help here |
| 3:45AM |
2 |
Calls on hold |
| 3:31AM |
4 |
rxfax, spandsp and lack of ecm |
| 3:20AM |
1 |
Kirk IP600 V3 DECT Wireless server |
| 3:18AM |
5 |
OT, Definity G3 Problems with Asterisk (Any Avaya Definity Experts out there?) |
| 3:14AM |
1 |
"Too many files..." error - best way to fix? |
| 2:55AM |
1 |
Polycom IP430 sound level too low? |
| 2:25AM |
0 |
Queue - persistent members |
| 1:06AM |
1 |
Anyone working on VXML, CCXML support for asterisk? |
| 12:04AM |
0 |
Long Delay in IAX Calls |
| |
| Tuesday September 12 2006 |
| Time | Replies | Subject |
|---|
| 10:58PM |
0 |
Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? |
| 9:27PM |
1 |
fxotune failure! "Could not fill input buffer - got -1 bytes, expected 4000 bytes" |
| 6:10PM |
2 |
Polycom Firmware |
| 4:58PM |
0 |
Bad number - is not in inbound speed dial |
| 4:22PM |
1 |
Makefile.moddir_rules: No such file or directory |
| 4:14PM |
1 |
Virtualise asterisk on Xen |
| 3:50PM |
0 |
INX (internationalnumber.com) Outgoing problem |
| 3:04PM |
1 |
All circuits are busy now??? |
| 3:04PM |
3 |
sound file length |
| 2:07PM |
1 |
sip origination and termination |
| 1:53PM |
1 |
Switch Experiences |
| 1:17PM |
0 |
strange problem with calls between MGCP and SIP clients(ATA's) |
| 1:15PM |
0 |
consitent half channel loss after 6 minutes |
| 12:01PM |
0 |
Please help with a telular mod. SX5e |
| 11:03AM |
0 |
RE: [asterisk-biz] Come see us at VON |
| 10:31AM |
0 |
Trouble connecting to my telco with fonebridge |
| 10:12AM |
0 |
AEL if/else/IFTIME fun. |
| 10:00AM |
1 |
A simple goal, help me please! |
| 9:57AM |
1 |
Calling Card and Billing |
| 9:28AM |
1 |
Verizon ISDN service in NY & Hunt Groups |
| 9:22AM |
1 |
Polycom MyStat |
| 8:34AM |
3 |
Problems getting 7970G upgraded to SIP |
| 8:08AM |
0 |
Conference bridge problem |
| 8:07AM |
1 |
Dropped call question - "Maximum retries exceeded on transmission" |
| 8:03AM |
0 |
IAX phone recommandation |
| 7:57AM |
1 |
RE : Re: [asterisk-dev] Forwarding sip requests from none localdomains |
| 7:39AM |
2 |
Suggestion for directed pickup in bristuffed 1.2 Asterisk |
| 7:37AM |
0 |
Grandstream Budgetone phones don't show |
| 7:33AM |
1 |
RE : Re: [asterisk-dev] Forwarding sip requests from none local domains |
| 7:27AM |
1 |
[BULK] Re: Prompts recording for Asterisk |
| 7:17AM |
0 |
Deploying an IVR - direct extensions.conf or AGI scripts? |
| 7:00AM |
1 |
How to setup announce attibute in queues.conf |
| 6:43AM |
1 |
WG: Asterisk and Agents |
| 6:33AM |
1 |
Features.. phone vs. asterisk? |
| 6:25AM |
1 |
WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner. |
| 5:32AM |
4 |
Rack for Asterisk with TDM2400 Digium board |
| 5:14AM |
0 |
Which SIP hardphone implements RTCP XR (aka RFC3611) |
| 2:55AM |
2 |
Junghanns BRI cards and misdn |
| 2:23AM |
0 |
Samsung OfficeServ 500 + Asterisk(Tormenta 2) via PRI |
| 12:48AM |
1 |
asterisk logging per day |
| 12:38AM |
0 |
SIP/2.0 403 Relaying denied |
| 12:34AM |
1 |
about 'zap show channels' |
| 12:17AM |
0 |
The best way to design local-only off-hours ringing |
| |
| Monday September 11 2006 |
| Time | Replies | Subject |
|---|
| 11:41PM |
0 |
Browsing distant missed call list |
| 8:25PM |
1 |
DID not getting passed? |
| 8:23PM |
0 |
Polycom HD Voice - 16 Khz - Asterisk support ? |
| 8:02PM |
0 |
Digium at Ohio Linuxfest |
| 5:58PM |
0 |
GXP2000 - Blind Transfer Hangs Up Call |
| 4:41PM |
0 |
Change Payload |
| 4:30PM |
1 |
Forward recorded voicemail message to more than one extension using sendvoicemail=yes |
| 4:27PM |
0 |
SIP 415 messagse |
| 4:25PM |
0 |
BLF via metermaid on 1.2.7.1 and aastra 9133i |
| 4:01PM |
4 |
question... |
| 3:13PM |
4 |
Dell hardware ... |
| 2:42PM |
0 |
SIP DOMAIN SUPPORT |
| 1:47PM |
1 |
Static RealTime - SIP.CONF |
| 1:21PM |
0 |
experience with axvoice.com? |
| 12:41PM |
1 |
More Zaptel build problems |
| 12:15PM |
0 |
Cable Systems ICS-G302, Anyone have an Admin Guide Please? |
| 12:14PM |
1 |
Weird (bri)stuff 0.3.0-PRE-1s |
| 11:59AM |
7 |
Polycom Soundpoint Key Remap |
| 11:57AM |
0 |
IAX2 trunk problem |
| 11:41AM |
2 |
How to configure Fritz ISDN2 card with Trixbox? |
| 11:30AM |
1 |
PRI channel hangup |
| 11:25AM |
3 |
Remote tone access |
| 11:13AM |
0 |
updated zaptel tarball |
| 10:45AM |
1 |
sip and iax over the internet (asterisk to asterisk) drop outs normal??? |
| 10:38AM |
0 |
Getting Incoming called from trxtel.com |
| 10:30AM |
0 |
Realtime Queues and Postgres. |
| 9:30AM |
0 |
--- Dlink DVC-2000 VideoPhone (H.323) with Asterisk --- |
| 8:54AM |
0 |
[Serusers] MS LCS 2005 / SER / Asterisk Integration |
| 8:45AM |
2 |
Grandstream Budgetone phones don't show alphanumeric caller right |
| 8:44AM |
0 |
Is anybody using autofill option in queue.conf? |
| 8:42AM |
1 |
help connecting cell phone, chan_bluetooth |
| 8:17AM |
5 |
PRI: sometimes Asterisk drop calls |
| 8:05AM |
2 |
Verify Database Installation |
| 7:38AM |
0 |
Support for Intel Boards On Asterisk |
| 7:05AM |
0 |
Register 2 times with same host |
| 6:56AM |
3 |
Problems Unpacking tarball For Asterisk Application |
| 6:20AM |
0 |
switching from IAX to SIP |
| 5:18AM |
1 |
realtime static config include contexts |
| 4:47AM |
0 |
Handling incoming calls from VoIPbuster |
| 4:30AM |
2 |
SIP trunk |
| 3:55AM |
0 |
Ringtones |
| 3:47AM |
0 |
Can Asterisk bind on multiple ports? |
| 3:36AM |
1 |
modifying the INVITE headers |
| 3:19AM |
0 |
I am not getting 302 redirects... |
| 2:51AM |
1 |
TE411P or TE412P? |
| 2:50AM |
0 |
SIP hardphones and BLF monitoring keys |
| 2:48AM |
0 |
Outgoing callerid in AMI |
| 2:07AM |
1 |
SIP parameter to prevent a call from being added in missed calls logs |
| 2:06AM |
1 |
Asterisk Realtime Arch - static or realtime? |
| 12:43AM |
3 |
How to integrate freepbx with a2billing? |
| 12:01AM |
1 |
beginners question.... |
| |
| Sunday September 10 2006 |
| Time | Replies | Subject |
|---|
| 11:12PM |
0 |
Looking to hire somebody to setup a SER load balancer |
| 8:35PM |
2 |
QUINTUM TENOR ASM200 Configuration |
| 7:42PM |
1 |
Hrmm.. OK, what am I missing? sendmail: Cannot open mail:25 |
| 4:53PM |
3 |
Setting system time via Asterisk |
| 4:42PM |
4 |
using residential voip for business? |
| 4:05PM |
2 |
Max Size of Conf Files |
| 3:49PM |
0 |
data pass through |
| 2:02PM |
2 |
music onhold choppy music problems |
| 1:38PM |
2 |
can someone recommend a voip provider that... |
| 12:43PM |
4 |
Voip providers and sip origination and termination? |
| 12:08PM |
1 |
QUINTUM CONFIGURATION.- |
| 12:02PM |
3 |
su - postgres -bash-3.00$ |
| 11:10AM |
0 |
Accounts registered, but call is not going |
| 10:24AM |
1 |
Polycom related question |
| 8:54AM |
2 |
Take 3 -- Trying to get SIP firmware on a 7970G |
| 5:09AM |
0 |
How could i get bridged channel partner |
| 4:40AM |
1 |
Satellite link-IAX Jitter Buffer. |
| 3:24AM |
0 |
call notification for queues? |
| |
| Saturday September 9 2006 |
| Time | Replies | Subject |
|---|
| 10:09PM |
1 |
Quintum tenor configuration with asterisk help |
| 10:03PM |
0 |
Streaming audio for MoH |
| 9:55PM |
1 |
Grandstream GX-2000 Remote Login Problem |
| 6:58PM |
0 |
What really happens between Asterisk and an SPA-3000? |
| 3:52PM |
6 |
Whcih phones are better for mass deployment |
| 3:47PM |
1 |
Using option 'r' in queue doesn't announce frequeny etc. |
| 2:25PM |
3 |
Scope of contexts |
| 1:54PM |
2 |
ztdummy installed but choppy audio warning on load |
| 1:46PM |
0 |
Problems configuring Polycom 301 |
| 12:48PM |
0 |
RE: asterisk-users Digest, Vol 26, Issue 54 |
| 10:10AM |
1 |
Intel Based G.729 and SVN-trunk-r42453 |
| 7:19AM |
0 |
DID Provider in Thailand |
| 7:17AM |
0 |
ISDN / Multiplink PPP (ZapRAS) |
| 6:05AM |
1 |
Another (quick) Polycom 501 question |
| 5:15AM |
4 |
Call Processing Slow 11 seconds |
| 2:19AM |
1 |
Call Forward Problem |
| |
| Friday September 8 2006 |
| Time | Replies | Subject |
|---|
| 11:32PM |
2 |
Receive Fax with rxfax on asterisk with debian |
| 10:40PM |
2 |
Zaptel-1.2.9 compile error |
| 8:56PM |
0 |
zaptel 1.2.9 won't compile |
| 7:29PM |
0 |
Asterisk 1.2.12 and Zaptel 1.2.9 released! |
| 4:54PM |
0 |
How to play a sound on a periodic basis during a call? |
| 4:42PM |
2 |
MSSQL connection |
| 3:44PM |
2 |
Stupid question about FXS/FXO |
| 3:05PM |
2 |
No such device -> TDM13B |
| 2:54PM |
2 |
Little help for a newbie configuring a TDM13B - ztcfg fails on channel 4 |
| 2:06PM |
1 |
I'm I wrong - No 3-way calling for Single line sets? |
| 1:26PM |
0 |
help chan_bluetooth |
| 12:47PM |
0 |
RE: Peter Dicks Chairman ofSportingbet PLC is arrested at JFK!! |
| 11:44AM |
1 |
No dialtone, just directly busy |
| 11:32AM |
2 |
What don't I get about SIP? |
| 11:11AM |
0 |
ISDN HFC card cannot 'detect remote answer' |
| 11:06AM |
0 |
Want to support a better SIP stack in Asterisk? |
| 10:47AM |
2 |
Use PauseQueueMember |
| 8:22AM |
1 |
Asterisk and SIP Redirect message |
| 7:31AM |
1 |
Reload question |
| 7:22AM |
1 |
FW: Peter Dicks Chairman of Sportingbet PLC is arrested at JFK!! |
| 7:04AM |
2 |
Tracking the source of a disconnect? |
| 6:43AM |
1 |
Grandstream, how to use the configuration tool |
| 6:36AM |
0 |
How can I set CDR data in dialplan? Set(CDR(src)=foo) |
| 6:35AM |
1 |
Grandstream GX-2000, doesn't send calls to free lines |
| 6:04AM |
0 |
Problems with KG1000 voip gateway and DTMF |
| 4:39AM |
2 |
distinguishing users by their domain |
| 3:39AM |
1 |
Asterisk and "Maximum retries exceeded" |
| 3:04AM |
3 |
Trouble with rxfax multi-page printing with cups |
| 2:59AM |
3 |
Digits are played in english in french voicemail |
| 2:38AM |
0 |
Problems with app_directed_pickup |
| 2:35AM |
2 |
sip peer question |
| 2:00AM |
2 |
codecs translation in Asterisk SVN-trunk-r41990 |
| 12:46AM |
0 |
dialplan applications |
| |
| Thursday September 7 2006 |
| Time | Replies | Subject |
|---|
| 11:07PM |
3 |
Asterisk 1.2 and SATA drives |
| 10:39PM |
0 |
app_amd and voicemail |
| 9:47PM |
1 |
Intel 945G and Digium TE110P compatibility issue |
| 4:04PM |
0 |
te110p and te205p behavioural differences |
| 2:43PM |
0 |
RE: asterisk-users Digest, Vol 26, Issue 39 |
| 2:14PM |
7 |
Call Forwarding in SIP.conf |
| 1:57PM |
1 |
Speex Codex - Eyebean to Asterisk |
| 1:32PM |
0 |
Open source G.729 and G.723.1 release for 1.2 and 1.4 |
| 1:32PM |
1 |
TDM400 and T100 config on same asterisk |
| 12:37PM |
2 |
Experiences, Tips on Voicemail storage using ODBC or IMAP? |
| 11:06AM |
1 |
Asterisk Outgoing Spool Failed |
| 9:47AM |
3 |
How to Install H323 |
| 9:15AM |
0 |
Sound (or lack of it) problems |
| 8:55AM |
2 |
uConnect Voip device |
| 8:32AM |
2 |
Asterisk hangs up after 10-15 minutes when SIP Phone is on mute |
| 8:30AM |
1 |
Asterisk and NAT ? |
| 8:07AM |
0 |
Voicemail Delete Bug? |
| 8:04AM |
1 |
g729 failover when out of licenses |
| 8:00AM |
1 |
svn trunk or branches ??? |
| 7:17AM |
1 |
Capacity for transcode G711 to G729 |
| 5:54AM |
1 |
Asterisk "Clusters" |
| 5:48AM |
0 |
Incoming call problem-calling part is busy(I PKall) |
| 5:25AM |
1 |
Incoming call problem-calling part is busy(IPKall) |
| 5:11AM |
3 |
Response to KP Flemming... |
| 4:54AM |
3 |
Cisco 7970 directories and services xml |
| 4:48AM |
6 |
Softphones IAX vs. SIP, remote connectivity. |
| 4:27AM |
1 |
bristuff compile problems with kernel 2.6.17.11 |
| 3:53AM |
0 |
WG: mobile refusing call |
| 3:50AM |
1 |
netmask |
| 3:14AM |
2 |
New polycom firmware / presence |
| 2:51AM |
0 |
Configuring new IAX2 Jitter Buffer for IVR application. |
| 2:39AM |
0 |
How to send and receiving fax with asterisk? |
| 1:03AM |
0 |
ast_parse_allow_disallow: Cannot allow unknown format 'h264' |
| |
| Wednesday September 6 2006 |
| Time | Replies | Subject |
|---|
| 10:43PM |
0 |
Re: [asterisk-dev] UUI in calls |
| 8:10PM |
2 |
the sounds quality of IAX2 channels are not good as SIP channels? |
| 7:46PM |
0 |
How to check which rtp ports my firewall let through? |
| 7:21PM |
4 |
Polycom new firmware and bootrom |
| 6:27PM |
4 |
using SIP to connect remote other VoIP server |
| 4:08PM |
0 |
Garbled (quality probs) IAX2 & SIP calls Asterisk-to-Asterisk |
| 4:00PM |
0 |
Digium's response to posting of G.729 and G.723 source code |
| 3:27PM |
1 |
Digium G.729 codec binaries updated |
| 3:21PM |
0 |
faktortel |
| 1:35PM |
6 |
Volume events causing talk off on Asterisk with Digium 411P |
| 12:50PM |
2 |
Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls |
| 12:17PM |
1 |
Call parking and RTP traffic |
| 11:16AM |
1 |
Is asterisk's mgcp support(NAS) Network access server package |
| 10:46AM |
1 |
Conditional IF based on IP address? |
| 9:47AM |
2 |
Cisco MWI |
| 7:45AM |
0 |
Which SIP hardphone with embedded VPNClient ? |
| 6:56AM |
1 |
app_rxfax Only Receives One Page |
| 4:57AM |
1 |
cmd SET time value |
| 3:33AM |
1 |
how to setup poxy sip server |
| 2:39AM |
0 |
mobile refusing call |
| 1:59AM |
0 |
flag 'g' in Dail() is'nt working with agentcallbacklogin() |
| 1:56AM |
0 |
sangoma A104d echo canceller and fax |
| 1:29AM |
1 |
core dumps |
| 1:17AM |
2 |
Budgetones - multiple phones losing IP address during day |
| 12:50AM |
1 |
How to test TE405P T1 |
| 12:29AM |
0 |
Asterisk AGI and Firebird |
| |
| Tuesday September 5 2006 |
| Time | Replies | Subject |
|---|
| 11:55PM |
2 |
Dell Poweredge SC430 and Digium cards compatability enquiry |
| 11:23PM |
1 |
Asterisk + Samsung OffServ 500 |
| 10:57PM |
1 |
macros in Realtime |
| 9:38PM |
1 |
Really bad phone line.. possible causes? |
| 9:00PM |
3 |
Has anyone tried to install both digital card and analog card in one machine |
| 8:26PM |
0 |
Need somebody for video phone testing |
| 7:34PM |
1 |
Native Chinese speaker needed |
| 4:53PM |
1 |
Merlin Legend - Working Now! |
| 2:46PM |
2 |
How to notify an ACD agent before he/she picks up |
| 2:32PM |
3 |
Asterisk Cygwin Port. |
| 1:51PM |
0 |
Is this a warning or not...MYSQL Fetch |
| 1:44PM |
1 |
Adding custom fields (more than one) to CDR DB |
| 1:40PM |
0 |
Linking Asterisk with PBX through E1 |
| 12:52PM |
1 |
asterisk t.38 fax failed |
| 12:50PM |
2 |
config include issues |
| 12:39PM |
0 |
Meet-me recording formats |
| 11:37AM |
3 |
IAX and rsa |
| 11:35AM |
1 |
Different MOH between waiting calls and transfer calls |
| 10:01AM |
1 |
Faxing .. |
| 9:52AM |
1 |
Find-Me/Follow-ME |
| 9:42AM |
2 |
Asterisk vicidial question |
| 8:38AM |
1 |
ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server |
| 8:26AM |
3 |
monoBRI + install-misdn-mqueue: no inbound calls but strange messages |
| 8:03AM |
1 |
Different MOH in waiting calls and parked calls |
| 7:58AM |
1 |
Unable to make calls from CallManager to Asterisk |
| 7:01AM |
0 |
R: Re: LinkSys PAP2 ATA & Siemens Cordless 3010 |
| 6:55AM |
1 |
Zero length queue |
| 6:29AM |
1 |
ISDN config EWSD |
| 6:28AM |
1 |
LinkSys PAP2 ATA & Siemens Cordless 3010 |
| 6:28AM |
0 |
telco error message on PRI and BRI |
| 4:54AM |
1 |
Experience Patton BRI gateways and Asterisk? |
| 4:05AM |
2 |
latest CentOS-asterisk-freepbx installation procedure |
| 3:21AM |
4 |
How to manipulate a plus in a phone number |
| 2:04AM |
2 |
why executed Hangup doesn't exit DialPlan?look my dialplan... |
| 1:26AM |
0 |
A couple more interviews with Digium staff |
| 12:21AM |
0 |
connect with two servers multiple time |
| |
| Monday September 4 2006 |
| Time | Replies | Subject |
|---|
| 11:47PM |
1 |
End of call |
| 11:25PM |
1 |
Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne cheaper than existing one. |
| 11:19PM |
0 |
Reading the raw E1 channels ? |
| 10:42PM |
0 |
HITBSecConf2006 Final Call ! |
| 10:36PM |
0 |
Warning about using PAP2-NA ATA recent firmware 3.1.12 LS |
| 7:13PM |
0 |
SNMP with 1.2.11 stable |
| 6:56PM |
3 |
Digum g729 and g723 |
| 3:13PM |
0 |
Re: Nufone making changes |
| 1:58PM |
2 |
Call center reports |
| 1:33PM |
1 |
app_conference not working for me |
| 1:17PM |
1 |
Looks like Nufone is changing around... |
| 12:26PM |
1 |
Grandstream and H.264 ! |
| 12:19PM |
0 |
playback some digits to the caller from the callee (involves DTMF) prob |
| 11:58AM |
0 |
Astbill DIALSTRING doesn't work |
| 11:26AM |
3 |
blf aastra 9133i working but can't pickup calls |
| 10:27AM |
1 |
missing pri connect (wwomera to pri) |
| 9:49AM |
5 |
ONE WAY VOICE ONLY IN ASTERISK |
| 8:55AM |
2 |
Dropping extra frame of G.729 ? |
| 8:47AM |
1 |
Submenus |
| 8:26AM |
0 |
Handling Disconnection Causes |
| 8:10AM |
0 |
PAP2-NA + Asterisk |
| 7:35AM |
5 |
FAX handling |
| 6:52AM |
3 |
Asterisk 1.2.11 and # key |
| 6:27AM |
0 |
usereqphone=yes seems to don't work |
| 5:37AM |
4 |
includes in realtime ?? |
| 5:21AM |
1 |
External calls from Asteris over a legacy Siemens BusinessPhone 250 PBX |
| 5:00AM |
0 |
External calls from Asterisk over a Siemens(legacy) RDSI PBX |
| 4:55AM |
2 |
Any Hardphone with VPNClient embedded? |
| 3:04AM |
0 |
Architecture:MainOffice(AstServer)-SmallOffices(ATA.-LegacyPBX) |
| 2:14AM |
3 |
Zaptel-1.2.8 compile problem |
| 2:12AM |
0 |
"Asterisk Developers Mailing List" <asterisk-dev@lists.digium.com>, |
| 1:20AM |
1 |
Prompts playback changing tempo w/ SMP kernel |
| 12:52AM |
0 |
No more linux/compiler.h in Fedora Core 6. |
| |
| Sunday September 3 2006 |
| Time | Replies | Subject |
|---|
| 10:26PM |
2 |
Asterisk calling through FWD? |
| 4:52PM |
0 |
Please help route incoming PSTN calls to Asterisk |
| 3:12PM |
1 |
PBX -> VoIP migration |
| 10:54AM |
0 |
AgentCallBackLogin and cdrupdate |
| 7:13AM |
0 |
Query about Call Detail Record in Asterisk |
| 4:10AM |
2 |
Asterisk+ser+docs |
| 3:05AM |
4 |
UK (BT) Problem with TDM 400P |
| 2:47AM |
1 |
Asterisk not sending RTP |
| 1:53AM |
2 |
SER+Asterisk integration |
| |
| Saturday September 2 2006 |
| Time | Replies | Subject |
|---|
| 10:18PM |
1 |
What I always get asked in SME * deployments |
| 4:12PM |
3 |
Queue timeout problems |
| 4:11PM |
4 |
How to use Grandstream GX-2000 phones for paging |
| 4:07PM |
4 |
Roundrobin not working on PRI |
| 4:05PM |
2 |
SIPP problem |
| 3:58PM |
3 |
How to send correct Caller ID on PRI |
| 3:37PM |
3 |
Caller ID has extra digits to strip |
| 3:11PM |
0 |
res_osp.c not compiled |
| 8:01AM |
0 |
Nokia N80 |
| 1:28AM |
4 |
Keys pressed not registering ... |
| 1:21AM |
6 |
G729 Replacement Codec - FREE or may ne cheaper than existing one. |
| 1:18AM |
1 |
Asterisk mysql cdr |
| |
| Friday September 1 2006 |
| Time | Replies | Subject |
|---|
| 10:37PM |
2 |
Blind transfer 3/4 digits |
| 8:37PM |
2 |
Help with blind transfer |
| 7:12PM |
5 |
Asterisk & Linksys PAP2 ATA |
| 5:42PM |
2 |
What does 'trunk' mean in outgoing and incoming? |
| 2:52PM |
1 |
Can QUEUE member be assigned from a GlobalVar set in EXTENSIONS.CONF? |
| 1:39PM |
1 |
Cisco 7960 won't download dialplan.xml |
| 11:51AM |
1 |
Callback + dtmf problem |
| 11:49AM |
3 |
Asterisk speaks Italian! |
| 9:15AM |
1 |
Hardware ? Analog DID trunks (ILT) |
| 9:12AM |
4 |
ASTERISK NOT LISTENING IN PORT 5060 |
| 8:44AM |
3 |
Sipura 3000 and Asterisk |
| 8:18AM |
0 |
incompatible hardware? |
| 7:22AM |
0 |
Asterisk Core dump |
| 7:03AM |
2 |
vim syntax highlighting( for Asterisk.conf files) |
| 6:45AM |
0 |
phpagi syntax and SendDTMF |
| 6:27AM |
0 |
Different MOH in parked calls?? |
| 6:27AM |
4 |
balance anouncement |
| 6:11AM |
2 |
Asterisk as a SER client |
| 5:42AM |
0 |
looking for GXV-3000 users |
| 4:37AM |
1 |
Probelm with incoming calls to my DID-Please help me |
| 2:27AM |
1 |
[Slightly OT] Grandstream configurator tool |
| 2:16AM |
2 |
Outgoing Call group ? |
| 12:36AM |
2 |
Any way to go from factory reset 7970 to SIP without Call Manager? |