| Saturday June 30 2007 |
| Time | Replies | Subject |
|---|
| 8:43PM |
2 |
Polycom echo problem |
| 3:58PM |
0 |
mISDN Fritz Passive Card |
| 1:13PM |
0 |
Asterisk 1.4.5 : Use I386 ou X86_64 |
| 11:04AM |
1 |
Asterisk 1.4.6 Fedora 7 configure error |
| 10:55AM |
0 |
AEL + Realitme? |
| 6:54AM |
1 |
Cisco 7970G line buttons |
| 4:08AM |
1 |
Fwd: Problems with zap following 1.4.6 install |
| 1:50AM |
1 |
FW: fail to load modules |
| |
| Friday June 29 2007 |
| Time | Replies | Subject |
|---|
| 6:50PM |
3 |
awful list delays: 4 days! |
| 6:30PM |
0 |
Asterisk 1.2.20 and 1.4.6 released |
| 5:31PM |
0 |
Linking Asterisk with another SIP PBX |
| 5:16PM |
1 |
TE420 PCI Express Card |
| 5:07PM |
2 |
Problem getting a Perl script to run |
| 4:08PM |
1 |
Music on hold 1.2 |
| 3:41PM |
2 |
Voice Mail not Receive |
| 3:27PM |
2 |
v1.4.x ready yet? |
| 2:35PM |
2 |
features.conf / DTMF / automon hell |
| 2:19PM |
1 |
Asterisk 1.4 Warnnings |
| 2:18PM |
0 |
DUNDi problem: offline peers still in request EID/EID_DIRECT field? |
| 1:28PM |
1 |
MOH question w/Cisco 79xx phones |
| 11:28AM |
1 |
Fwd: Call Wainting dysfunctions |
| 10:51AM |
0 |
nway call |
| 10:06AM |
4 |
asterisk call unique id in dialplan |
| 9:32AM |
2 |
How many number of parallel calls can make through asterisk |
| 1:40AM |
1 |
SPA-2100 Distinctive Ring |
| |
| Thursday June 28 2007 |
| Time | Replies | Subject |
|---|
| 7:16PM |
3 |
setup multiple phones for 1 extension |
| 6:46PM |
1 |
CDR Log analizer software |
| 6:42PM |
0 |
Anyone who can do live video feed to co-host asterisk "show" next week? |
| 6:33PM |
0 |
Robo Dialer |
| 6:17PM |
1 |
Shared Extension Appearance |
| 6:00PM |
4 |
network routing |
| 5:59PM |
2 |
Asterisk and IPv6 |
| 5:54PM |
1 |
Avaya IP Office DTMF Issue |
| 5:39PM |
0 |
Avoided deadlock for '0x864e70', 10 retries! |
| 4:40PM |
2 |
Linking Asterisk with another SIP PBX (or SIP Softswitch) |
| 3:41PM |
1 |
RTCP NTP Clock skew |
| 2:53PM |
2 |
E1 not coming up |
| 2:43PM |
2 |
Caller ID Spoofing to be banned in the USA |
| 1:00PM |
1 |
FXS channel bank |
| 11:55AM |
1 |
registering Asterisk on SIP/Nortel MCS server |
| 11:05AM |
1 |
Asterisk 1.4.5 Inserting Random Digits in Dialed Number! |
| 10:50AM |
2 |
Fax passthrough howto codec upspeed |
| 10:35AM |
2 |
CDR and call transfer |
| 10:30AM |
1 |
error while compiling asterisk-1.2.19 |
| 9:20AM |
2 |
fail to load modules |
| 8:56AM |
0 |
Work |
| 7:37AM |
0 |
registering Asterisk on SIP/Nortel MCS thing |
| 7:13AM |
0 |
Calls audio stops with latest Gigaset C450IP firmware + voicemail |
| 5:50AM |
1 |
Updated Manual for Asterisk 1.4.x |
| 5:21AM |
2 |
Call transfer feature |
| 4:03AM |
2 |
voicemail.conf serveremail |
| 2:17AM |
0 |
Any difference using * with Centos i386 and x86_64 ? |
| |
| Wednesday June 27 2007 |
| Time | Replies | Subject |
|---|
| 11:21PM |
1 |
Voicestick / i2telecom.com |
| 10:15PM |
0 |
Bypass local dialplan and redirect INVITE |
| 8:32PM |
5 |
North American voice BRI - Informal survey |
| 7:24PM |
2 |
Error While Calling AGI |
| 7:06PM |
2 |
.call file |
| 4:50PM |
0 |
QueueMetrics 1.4 released today |
| 4:37PM |
2 |
OpenSer/Asterisk PBX solution |
| 4:25PM |
1 |
Has anyone sucessful Asterisk to an Avaya IP phone |
| 4:17PM |
4 |
Customized Ring Tone |
| 4:08PM |
4 |
Using MSAccess to dial on a Zap line |
| 3:13PM |
0 |
Asterisk to Cisco 2600 GW DTMF Not Working, Working now |
| 2:43PM |
4 |
Asterisk+squid |
| 12:53PM |
2 |
Problems compiling Asterisk 1.4.5 |
| 12:41PM |
1 |
Round Robin SIP peers? |
| 12:27PM |
1 |
Help with IAX Trunk |
| 11:46AM |
2 |
Wait to numbers |
| 11:20AM |
1 |
Self Calling test |
| 11:15AM |
0 |
ISDN data-call question |
| 11:12AM |
0 |
minibrowser for each snom phone |
| 10:45AM |
0 |
IAX trunking using a different port |
| 10:01AM |
3 |
Missing 'init keys' command |
| 9:41AM |
1 |
Zap dialling issues |
| 8:29AM |
0 |
Fwd: problem with one way audio |
| 2:59AM |
1 |
Module '***.so' did not register itself during load |
| |
| Tuesday June 26 2007 |
| Time | Replies | Subject |
|---|
| 11:01PM |
1 |
Asterisk to Cisco 2600 GW DTMF Not Working |
| 8:57PM |
1 |
No such host error from SIP for non-peer configuration. |
| 7:45PM |
6 |
Cisco 7941 localized menus with SIP firmware |
| 6:27PM |
0 |
TE412 / HPDL380G5 / * 1.4 / CentOS 4.5 Experience |
| 6:22PM |
6 |
kore dump |
| 4:37PM |
0 |
Slip Events |
| 3:45PM |
0 |
SpectraLink SVP protocol support in asterisk |
| 3:11PM |
2 |
More FAX over T1 |
| 2:21PM |
0 |
rcf2833 DTMF broken in asterisk SIP channel? |
| 2:18PM |
1 |
Multi port IAX Gateway |
| 12:56PM |
0 |
No CID on Zaps - TDM400 |
| 11:14AM |
1 |
call fail from audiocode to sip trunk |
| 10:29AM |
0 |
Test Message |
| 9:38AM |
2 |
Bridging two PSTN calls |
| 9:00AM |
0 |
VPN technology for snom 370 |
| 8:59AM |
0 |
asterisk-users Digest, Vol 35, Issue 92 |
| 8:47AM |
1 |
realtime_extensions |
| 7:57AM |
0 |
Asterisk + Legacy PBX |
| 7:27AM |
1 |
zaptel 1.2.18 and HPEC |
| 5:07AM |
5 |
Inexpensive Layer 3 Switch? |
| 4:39AM |
0 |
call transfer problem |
| 4:36AM |
0 |
CDR changes in 1.4.5 are confusing |
| 4:10AM |
2 |
Fax Throughput |
| 3:31AM |
1 |
Modification of Caller ID based on context |
| 3:01AM |
0 |
Spy a specific Channel |
| 1:52AM |
1 |
CDR Records "s" as dst |
| 1:26AM |
0 |
asterisk-users Digest, Vol 35, Issue 91 |
| |
| Monday June 25 2007 |
| Time | Replies | Subject |
|---|
| 11:42PM |
1 |
AstPligg |
| 10:33PM |
1 |
Ring the second line when 1st line is busy |
| 9:41PM |
1 |
Problems with ChanIsAvail always return status 0 |
| 9:14PM |
1 |
Dynamic DUNDi weight support in * - HELP! |
| 8:51PM |
0 |
four ringing and hangup with error |
| 7:50PM |
0 |
Help. Help. Help. How to make outbound proxy and host URI with different port? |
| 7:49PM |
1 |
Transfer Call to Cell Phone |
| 6:33PM |
0 |
Does outboundproxyport still work in 1.4.4 |
| 6:06PM |
0 |
Outbound proxy setting with outbound proxy port in sip.conf |
| 5:51PM |
1 |
Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer |
| 4:30PM |
1 |
Threading troubles 1.4.5 & IAX2-> SIP (FreeBSD specific??) |
| 3:57PM |
0 |
Set a global queue policy |
| 3:37PM |
2 |
callback and bridge problem |
| 3:31PM |
2 |
two channels, each dropping into the same context, different behavior. |
| 2:41PM |
2 |
asterisk-users Digest, Vol 35, Issue 89 |
| 1:26PM |
2 |
Rining 180 and 183 |
| 10:19AM |
7 |
Hi ability solution |
| 9:53AM |
2 |
outging select |
| 9:13AM |
2 |
g729 problem |
| 8:38AM |
1 |
Xorcom Bri 4 Port USB |
| 8:27AM |
0 |
asterisk not able to hear calling party ring sound |
| 4:07AM |
5 |
Best wifi IP phone for asterisk |
| 3:52AM |
1 |
Provisioning Linksys WIP330 phones |
| 3:03AM |
0 |
Faktortel in Sydney outage |
| |
| Sunday June 24 2007 |
| Time | Replies | Subject |
|---|
| 3:56PM |
3 |
Nokia N95 + Dial Plan |
| 1:16AM |
0 |
OT: Digg post |
| 12:48AM |
1 |
Call Path Optimization |
| |
| Saturday June 23 2007 |
| Time | Replies | Subject |
|---|
| 3:48PM |
1 |
Zaptel Compilation Error |
| 3:16PM |
4 |
Zaptel Compilation |
| 1:32PM |
1 |
TAS test equipment manuals |
| 12:52PM |
4 |
IAX client USB phone |
| 11:08AM |
1 |
Dynamic DUNDi weight |
| 10:05AM |
2 |
IVR question for asterisk |
| 6:29AM |
0 |
DTMF Problem with one sip trunk and two carriers |
| 4:30AM |
0 |
modules loading |
| |
| Friday June 22 2007 |
| Time | Replies | Subject |
|---|
| 9:43PM |
6 |
FAX over T1 |
| 8:39PM |
1 |
Polycom 301 - Problem with AMI Originated Calls |
| 8:22PM |
0 |
asterisk-users Digest, Vol 35, Issue 81 |
| 7:34PM |
2 |
1.4.5 |
| 6:58PM |
1 |
Binding to multiple addresses |
| 6:49PM |
1 |
H.323 IP Phones and H.323 Traffic |
| 6:28PM |
1 |
Nuance Buys Tegic from AOL for $265m |
| 6:20PM |
10 |
inband DTMF for g729 |
| 5:48PM |
1 |
Ring/Off-hook in strange state 6 |
| 5:37PM |
1 |
Audio going one way for a few seconds during the call |
| 5:08PM |
1 |
Does Early Media have to be Ulaw? |
| 4:57PM |
0 |
Binding to multiple ports in sip.conf |
| 4:53PM |
2 |
access to asterisk server since internet |
| 4:06PM |
0 |
Hints |
| 3:54PM |
1 |
searching for compatible servers |
| 2:24PM |
0 |
chan_zap problems |
| 12:59PM |
2 |
got-name |
| 12:11PM |
1 |
Friday June 22@12:30PM EDT Asterisk Users Conference |
| 11:41AM |
1 |
POTS - Incoming Voice or Fax - How to tell? |
| 11:00AM |
1 |
problem with one way audio |
| 10:35AM |
1 |
Config for TEI parameter |
| 9:50AM |
2 |
asterisk 0 dial outgoing call |
| 9:50AM |
10 |
Query |
| 9:10AM |
4 |
international numbers... |
| 8:00AM |
3 |
chan_features.so / asterisk 1.4.5 |
| 7:20AM |
0 |
RTCP NTP clock skew detected |
| 7:13AM |
1 |
qozap and zt_alarm_notify_no_master_change |
| 5:53AM |
1 |
Once Touch Recording |
| 1:35AM |
2 |
STDERR in AGI |
| 12:50AM |
1 |
hotline with Polycom |
| |
| Thursday June 21 2007 |
| Time | Replies | Subject |
|---|
| 10:54PM |
0 |
Looking to buy VoIP or Hosting Company |
| 7:41PM |
0 |
Console channels with two sound cards? |
| 7:17PM |
2 |
Use of ChanSpy |
| 3:49PM |
0 |
Bug in Ex-Girlfriend logic? |
| 3:37PM |
0 |
Forward to my phones the domain of the CALLERID in incoming URI calls |
| 3:22PM |
1 |
TDM800P - zaptel service startup problem |
| 2:51PM |
3 |
identifying what a user pressed to reach my phone |
| 2:41PM |
0 |
ENC: Action Originate (Asterisk Manager) X Monitor() |
| 2:34PM |
1 |
Zaptel wct2xxp driver causes LEDs to go black and RED alarm |
| 2:33PM |
2 |
CDR |
| 1:57PM |
0 |
retreiving callid of call from the dial application |
| 1:35PM |
1 |
AudioCodes Gateway and Asterisk |
| 1:11PM |
0 |
Using Queue - Zap problems (PRI) |
| 1:07PM |
2 |
ChanSkype |
| 9:35AM |
1 |
TDM400 one way calls |
| 8:40AM |
2 |
mediant 2000 with asterik configuration |
| 8:34AM |
0 |
mISDN problems |
| 7:46AM |
3 |
gtalk - no audio |
| 7:10AM |
1 |
Problem with Remote-Hold/MusicOnHold |
| 7:05AM |
2 |
Asterisk config files and #include |
| 6:52AM |
0 |
SIP/IAX2 Phones behind USR 9108 Router |
| 6:24AM |
7 |
asterisk 1.4.1 app_addon_sql_mysql |
| 2:52AM |
0 |
Asterisk 1.2.0 addon Radius |
| 1:06AM |
1 |
install Asterisk-addons 1.4.2 |
| |
| Wednesday June 20 2007 |
| Time | Replies | Subject |
|---|
| 7:02PM |
0 |
Firewall on AsteriskNow |
| 5:52PM |
1 |
different codec for different extensions |
| 5:29PM |
2 |
Forcing Dial application to skip if called server is unreachable |
| 4:27PM |
2 |
How to Create Custom Context |
| 4:11PM |
1 |
ATT: Brian Fertig |
| 4:00PM |
1 |
X-Lite problems on basic asterisk setup |
| 3:23PM |
0 |
asterisk with mediant 2000 trunk |
| 2:38PM |
1 |
DTMF doesn't work between Asterisk and Cisco SIP Proxy |
| 2:33PM |
1 |
hanging up |
| 2:18PM |
1 |
Asterisk RealTime |
| 12:37PM |
1 |
Res: Record CDR in a Oracle database |
| 12:36PM |
0 |
Agent auto congesting |
| 12:23PM |
0 |
Query regarding connecting PABX with Application server |
| 12:16PM |
0 |
asterisk + mediant 2000 |
| 11:42AM |
2 |
zlib1g |
| 11:21AM |
3 |
Single ringer phone for incoming calls, that anyone can answer |
| 7:30AM |
0 |
Error: Unable to allocate RTCP socket: Too manyopen files |
| 6:38AM |
0 |
WHAT happened to AgentMonitorOutgoing(c) in Asterisk 1.4.5 ?? |
| 12:05AM |
1 |
ChanSpy SIP |
| |
| Tuesday June 19 2007 |
| Time | Replies | Subject |
|---|
| 7:20PM |
3 |
Ex-Girlfriend Logic in 1.4.4 |
| 7:02PM |
1 |
Advice |
| 6:41PM |
3 |
Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension. |
| 5:06PM |
1 |
Inline record |
| 1:01PM |
2 |
PhpAgi call generation |
| 12:00PM |
3 |
Execute Chanspy |
| 10:55AM |
0 |
ENUMLOOKUP well succeeded but callee server unreached |
| 10:34AM |
1 |
problem with mISDN |
| 9:58AM |
1 |
Play dial tone withou answer |
| 9:11AM |
2 |
make config |
| 8:36AM |
0 |
peer timeouts and 489s |
| 6:46AM |
0 |
passing privacy information through asterisk |
| 4:27AM |
2 |
Need to increase call count |
| 3:29AM |
2 |
Invalid DTMF detection -- Invalid Extension Bug or issue |
| |
| Monday June 18 2007 |
| Time | Replies | Subject |
|---|
| 11:55PM |
3 |
How to config SIP blind transfer in extension.conf |
| 11:49PM |
1 |
AGI command |
| 10:56PM |
0 |
Asterisk-addons 1.2.7 and 1.4.2 released |
| 10:03PM |
0 |
sip <> zap calls choppy, where to setup the jbuffer? |
| 9:57PM |
1 |
180 Ringing with SDP |
| 9:14PM |
0 |
no sound with chan_mobile |
| 6:10PM |
2 |
MixMonitor Timestamp problem |
| 5:37PM |
2 |
SIP Termination with automatic debit |
| 5:15PM |
1 |
High availability Asterisk |
| 4:28PM |
0 |
Monitor recording losing sync |
| 3:25PM |
1 |
Phantom Calls |
| 3:11PM |
0 |
res_jabber over OpenSSL ready for testing |
| 1:40PM |
2 |
Blind xfer issue -- URGENT! |
| 12:52PM |
2 |
asterisk and SAP |
| 12:05PM |
1 |
atxfer attended transfer feature |
| 12:03PM |
0 |
Problem In Installing Asterisk on Solaris |
| 10:50AM |
9 |
chan problem |
| 4:38AM |
0 |
Regarding call transfer feature |
| |
| Sunday June 17 2007 |
| Time | Replies | Subject |
|---|
| 9:07PM |
2 |
CNAM. |
| 1:49PM |
2 |
SIP Peering--call terminated prematurely |
| 12:45PM |
2 |
Upgrade cisco SIP phone 7940 |
| 9:15AM |
0 |
Mitel 5340 IP Phone |
| 7:08AM |
1 |
asterisk hang (Critical Response) |
| 5:25AM |
7 |
VPN on Asterisk |
| |
| Saturday June 16 2007 |
| Time | Replies | Subject |
|---|
| 9:40PM |
1 |
Convert or listen to .sln file |
| 3:11PM |
1 |
ipv6 on Asterisk |
| 2:44PM |
2 |
MixMonitor Problem |
| 2:42PM |
0 |
(no subject) |
| 2:30PM |
0 |
ivr testing script |
| 1:29PM |
1 |
Chanspy |
| 3:47AM |
0 |
Asterisk to Panasonic TDA200 with Unicall |
| 12:44AM |
0 |
Asterisk 1.4.anything on FreeBSD? |
| |
| Friday June 15 2007 |
| Time | Replies | Subject |
|---|
| 11:18PM |
0 |
Asterisk 1.2.19 and 1.4.5 released! |
| 11:14PM |
1 |
Accessing Voicemail(Asterisk) -- Remotely either from Cell or Landline |
| 9:36PM |
2 |
calling |
| 9:10PM |
0 |
scaling with SMP |
| 9:05PM |
1 |
Where an extension really is (DUNDi woes) |
| 8:05PM |
1 |
Community PBX? |
| 6:37PM |
0 |
FXS card with 3-way call, transfer and call waiting. |
| 6:22PM |
2 |
combining AGI with dialplans |
| 2:06PM |
1 |
can ENUMLOOKUP query multiple DNS servers without having to replicate the same code for each server? |
| 1:12PM |
1 |
g729 codec |
| 12:45PM |
2 |
Run as root? |
| 10:17AM |
0 |
hangup during voicemail announcement drops all calls |
| 8:33AM |
0 |
Error: Unable to allocate RTCP socket: Too many open files |
| 5:28AM |
4 |
app_rxfax vs (iaxmodem+hylafax) |
| 4:57AM |
0 |
Reinvite / one-way media. |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| 1:07AM |
0 |
No subject |
| |
| Thursday June 14 2007 |
| Time | Replies | Subject |
|---|
| 3:02PM |
3 |
My Kernel |
| 2:50PM |
2 |
Unicall + MFC/R2 line dropped immediately after connect |
| 2:26PM |
2 |
question on capacity |
| 12:25PM |
0 |
b410p |
| 11:46AM |
4 |
Que on A2Billing |
| 11:40AM |
11 |
Asterisk GUI |
| 9:22AM |
1 |
ODBC voicemail questions |
| 8:56AM |
1 |
Sending text to a phone that's no in-use ... |
| 8:15AM |
0 |
real-time HINTS |
| 6:59AM |
0 |
ESI Phone System Integration |
| 6:58AM |
2 |
Linksys SPA941 |
| 6:35AM |
0 |
Adtran feature codes, extensions |
| 5:35AM |
1 |
Sugar Auto-Dial with Asterisk? |
| 4:47AM |
0 |
(no subject) |
| 3:19AM |
1 |
Qualify renders all SIP peers unreachable |
| 1:45AM |
1 |
TDM400p and te110p configuration. |
| 12:05AM |
0 |
Re: asterisk testing - thanx! |
| |
| Wednesday June 13 2007 |
| Time | Replies | Subject |
|---|
| 8:12PM |
0 |
Re: asterisk-users Digest, Vol 35, Issue 52 |
| 3:14PM |
0 |
Disconnect tone detection. |
| 2:48PM |
1 |
What is the state of Asterisk Secure Remote Communications? |
| 1:38PM |
2 |
Addons |
| 11:52AM |
0 |
Digium mailing list server maintenance - Thursday, June 14, 5PM to 8PM CDT |
| 10:14AM |
0 |
second dial, force hangup for exit. |
| 10:06AM |
3 |
WAV file best sound quality |
| 8:48AM |
1 |
Weird sip registration problem |
| 7:14AM |
2 |
mISDN problem |
| 6:33AM |
2 |
Polycom + Voicemail + Display message envelope in LCD |
| 5:48AM |
3 |
Using Modems with Asterisk |
| 4:19AM |
6 |
problem starting asterisk, unable to load chan_zap |
| 3:34AM |
3 |
re:zaphfc problem (Josu Lazkano) |
| 2:28AM |
1 |
advanced asterisk logging |
| 2:08AM |
1 |
Voicemail prob |
| 12:39AM |
0 |
zaphfc problem |
| |
| Tuesday June 12 2007 |
| Time | Replies | Subject |
|---|
| 3:19PM |
0 |
anyway in meetme to mute all but one user? |
| 2:16PM |
1 |
Realtime Meetme in 1.4 |
| 1:38PM |
4 |
Gigabit SIP Phones |
| 12:05PM |
3 |
CDR changes in Trunk -- Transfers, CDRs, Life, and Everything |
| 11:50AM |
1 |
Answering machine detection after Dial() |
| 10:40AM |
4 |
GotoIf Dialplan inquiry |
| 10:35AM |
5 |
Asterisk Faxing |
| 9:53AM |
2 |
Softphone behind NAT issues |
| 8:55AM |
1 |
HPEC and audioclipping |
| 8:15AM |
2 |
Transfer caller direct to voicemail |
| 7:38AM |
1 |
AsterFax |
| 7:38AM |
0 |
[asterisk-tech] ChanSkype |
| 7:32AM |
0 |
Zombie SIP channels |
| 7:28AM |
2 |
Bridge bug in 1.4? |
| 6:59AM |
3 |
Changing the Caller ID |
| 6:56AM |
0 |
Warning on CLI |
| 6:54AM |
1 |
call from ISDN |
| 5:05AM |
4 |
write some custom values to CDR table |
| 4:24AM |
2 |
SPA400 and asterisk |
| 4:21AM |
2 |
No audio after Dial with G option |
| 3:59AM |
0 |
SIP/NAT 1.2 1.4 questions |
| 2:02AM |
0 |
Possible mysql database corruption |
| 1:39AM |
0 |
On multiple dial phones continue ringing after picked up |
| 1:16AM |
0 |
config files to mysql convertion |
| |
| Monday June 11 2007 |
| Time | Replies | Subject |
|---|
| 9:21PM |
1 |
Crashes with Spandsp, app_rxfax.c, and asterisk 1.4.4 |
| 6:55PM |
1 |
Slightly OT:CSU on Digium cards, and it's requirement |
| 4:39PM |
1 |
which Wifi SIP phones are the good ones |
| 3:13PM |
0 |
sip show registry shows nothing |
| 1:55PM |
0 |
AGI "RECORD FILE" for a video message |
| 12:16PM |
1 |
Introduction to AGI programming |
| 12:14PM |
1 |
CallerID issues |
| 10:58AM |
1 |
Multiple ENUM entries and Asterisk fails to dial |
| 10:45AM |
0 |
Grandstream 4104 - Asterisk (Incoming Calls problem) |
| 7:28AM |
0 |
Asterisk as an SCCP client |
| 7:08AM |
0 |
Different ECs in Asterisk |
| 6:38AM |
5 |
change moh during a call? |
| 6:07AM |
0 |
(no subject) |
| 6:00AM |
0 |
Help on text entry. using asterisk. |
| 5:50AM |
3 |
Searchable List Archives? |
| 5:26AM |
1 |
CDR on transfers of called ZAP channel |
| 3:57AM |
1 |
MOH Problems. |
| |
| Sunday June 10 2007 |
| Time | Replies | Subject |
|---|
| 11:56PM |
0 |
Going to VON Stockholm? Meet you at the Asterisk BOF! |
| 11:45PM |
1 |
basic asterisk knowledge |
| 6:18PM |
1 |
best format for audio via asterisk... |
| 9:58AM |
1 |
Blocking 900 calls |
| 7:46AM |
2 |
IAX Peers show command |
| |
| Saturday June 9 2007 |
| Time | Replies | Subject |
|---|
| 9:10PM |
1 |
ast_dynamic_str_thread_build_va() is defined with 6 args but only called with 5 args?? |
| 7:13PM |
3 |
Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event |
| 5:21PM |
2 |
Polycom 301 vs. 330 |
| 4:25PM |
1 |
OT: CallManager ANI restamp. |
| 1:00PM |
2 |
How to tell what codec is used for each end of a call MD110->H323->SIP |
| 12:57PM |
0 |
H.323 trunk between MD110 and Asterisk |
| 6:39AM |
2 |
No sound, problem is not a NAT |
| 1:08AM |
0 |
Asterisk Users Conference Friday: New Asterisk Book and a visit from JerJer of Nufone |
| |
| Friday June 8 2007 |
| Time | Replies | Subject |
|---|
| 7:47PM |
0 |
Softphone for smartphone such as Nokia N90 / 93 / N95 |
| 7:44PM |
0 |
CDR accuracy |
| 4:38PM |
3 |
Asterisk 1.4 with Unicall |
| 4:33PM |
0 |
FW: Delivery Status Notification(Failure) |
| 4:00PM |
11 |
Bad Echo between SIP calls |
| 3:34PM |
0 |
Zaptel 1.2.18 and 1.4.3 released! |
| 1:07PM |
0 |
Replacing SX-2000 Centigram Voicemail with Asterisk? |
| 12:02PM |
0 |
Log interpretation |
| 11:47AM |
2 |
Can Asterisk RAS? |
| 9:45AM |
5 |
Write to multiple databases as redundancy scheme |
| 9:11AM |
3 |
choppy sound with playback, background, etc... but not with musiconhold |
| 8:52AM |
0 |
No/unknown event '0' on timer |
| 8:48AM |
2 |
Hot GXP-2000 |
| 8:01AM |
1 |
Not getting CID Name from PRI |
| 6:52AM |
1 |
call problem... |
| 5:50AM |
0 |
Asterisk, NAT and canreinvite=yes |
| 2:37AM |
0 |
tdd.c - How does one use this code? |
| 1:26AM |
0 |
Unexpected behaviour shown by "meetme kick confno usernumber" |
| 1:03AM |
3 |
Asterisk & MS RTC Library & Ethernet Capacity |
| |
| Thursday June 7 2007 |
| Time | Replies | Subject |
|---|
| 4:38PM |
3 |
IAX trunk with dynamic IPs |
| 3:52PM |
0 |
IAX-configuration |
| 3:42PM |
1 |
RFC-3389 problem |
| 2:32PM |
4 |
agi with java? |
| 2:14PM |
1 |
Q931 Error with H323 |
| 1:28PM |
3 |
getting at ${CALLERIDNUM} |
| 1:12PM |
1 |
custom cdr fields and cdr_mysql, howto? |
| 9:44AM |
3 |
Provisioning Linksys PAP2T ATA's |
| 9:35AM |
2 |
Bridged PRI calls - processor involvement? |
| 8:24AM |
1 |
DUNDi and reinvites... |
| 8:23AM |
1 |
Meet Me video conferencing |
| 7:01AM |
3 |
Polycom phone registration problem |
| 6:26AM |
0 |
Need help on Text entry applicaon |
| 6:11AM |
0 |
Need help on Text entry for asterisk through touchpad |
| 6:07AM |
0 |
Need help on Text entry for asterisk through touch pad |
| 5:29AM |
0 |
Realtime Agents.conf |
| 4:18AM |
0 |
atxfer not working |
| 2:20AM |
1 |
AddQueueMember vs AgentCallbackLogin |
| 2:08AM |
0 |
MP3 as voicemail format |
| 1:04AM |
1 |
call Hold event asterisk |
| 12:05AM |
0 |
voice activated recording |
| |
| Wednesday June 6 2007 |
| Time | Replies | Subject |
|---|
| 11:44PM |
2 |
iax trunking on OpenBSD |
| 10:43PM |
0 |
SIP buddy watch |
| 9:43PM |
0 |
Solved: [SetAccount in extensions.conf] |
| 7:59PM |
4 |
Best Codec |
| 5:44PM |
3 |
1.4 Zaptel/Sangoma Issues on CentOS |
| 5:17PM |
0 |
SetAccount in extensions.conf |
| 4:26PM |
4 |
Slow list |
| 4:15PM |
2 |
Sending multiline SMS |
| 4:08PM |
1 |
Reload in 1.4 clears regexten |
| 3:41PM |
2 |
Console duplicate output problem |
| 2:05PM |
1 |
Polycoms lose registration and won't re-register |
| 1:34PM |
1 |
Phantom calls: Detecting hangup quicker |
| 12:49PM |
1 |
Voicemail marking messages as Old |
| 12:38PM |
1 |
Stanaphone/Asterisk issue: No Audio with SIP to only one provider when switching servers |
| 11:50AM |
1 |
Record CDR in a Oracle database |
| 11:13AM |
0 |
Queue Job |
| 11:12AM |
4 |
meetme realtime |
| 10:09AM |
1 |
zaptel make problem |
| 9:51AM |
3 |
Needed changes in Asterisk to change the SIP port to 5062 |
| 9:14AM |
5 |
TCP<->UDP SIP proxy? |
| 8:56AM |
0 |
Voip-info.org |
| 8:42AM |
0 |
Polycom 320 messages |
| 7:17AM |
1 |
asterisk 1.2.18 problems... |
| 6:39AM |
2 |
PRI Partial Re-Rounting |
| 5:22AM |
12 |
blades? |
| 5:07AM |
0 |
Multiple Ethernet Interfaces |
| 4:25AM |
2 |
shorting flash time |
| 3:48AM |
1 |
CDR changes in 1.4.3? |
| 2:33AM |
3 |
Asterisk call quality detection |
| 12:56AM |
2 |
Zaptel and Libpri Compilation |
| 12:56AM |
2 |
Queue problem |
| |
| Tuesday June 5 2007 |
| Time | Replies | Subject |
|---|
| 8:40PM |
3 |
Outlook dialing |
| 7:52PM |
1 |
Recomender Server specs for 250 con-current calls |
| 3:30PM |
2 |
Verizon Interconnection |
| 3:27PM |
1 |
g729 |
| 3:23PM |
1 |
Set caller ID based on SIP source. |
| 11:20AM |
3 |
Changing the From field in Asterisk email/voicemail |
| 11:13AM |
4 |
Where to find Polycom firmware with 330/320 support? |
| 10:51AM |
5 |
Hardware spec comparison |
| 10:21AM |
1 |
Cisco 7961G + 7914 Expansion Module |
| 6:46AM |
1 |
spa 3102 incoming call |
| 6:35AM |
1 |
addqueuemember recording and reporting |
| 6:33AM |
1 |
Asterisk on x64 |
| 6:16AM |
1 |
Strange beeping sound during fax initiate session |
| 6:12AM |
1 |
NAT |
| 5:57AM |
2 |
X100P Clone |
| 3:28AM |
1 |
Meetme define context |
| 3:26AM |
1 |
Problem to park the call with #700 |
| 3:08AM |
1 |
IAX2 Trunk No Sound |
| 2:55AM |
0 |
IS_REGISTERED from dialplan |
| 2:12AM |
1 |
cepstral TTS and app_swift |
| 1:20AM |
1 |
spa 3102 configuration |
| |
| Monday June 4 2007 |
| Time | Replies | Subject |
|---|
| 9:55PM |
0 |
chan_sip.c: That's odd... Got a response on a call we dont know about. |
| 9:32PM |
3 |
Noise on FXS ports (Sangoma) |
| 6:45PM |
0 |
Aastra 57i / 57i CT phones fail to re-register |
| 4:56PM |
1 |
Oddity |
| 12:46PM |
2 |
Get calling channel before pickup |
| 12:35PM |
4 |
Detecting card on the PCI Slot |
| 12:28PM |
2 |
answer a voip call, play info. |
| 11:05AM |
0 |
Asterisk 1.4.4 Segfaults with asterisk-ooh323 from addons-1.4.1 |
| 10:50AM |
1 |
realtime ldap peer matching |
| 10:33AM |
1 |
cisco 7940 and auto-answer (aastra 480i vs 7940) |
| 10:32AM |
3 |
Calls being dropped |
| 10:04AM |
2 |
Delay in posting of messages to list |
| 10:02AM |
0 |
no ringing tone making attended transfer whith an IAX client |
| 9:53AM |
1 |
Debug meetme |
| 9:20AM |
1 |
AEL2 Includes in Macro... |
| 8:33AM |
1 |
no dtmf pcom 650 only outbound calls |
| 7:41AM |
3 |
background dialing |
| 7:40AM |
2 |
FX Dialing Odd |
| 7:04AM |
1 |
addqueuemember recording and reporting problems |
| 6:04AM |
3 |
debug logs |
| 5:50AM |
2 |
G729 License |
| 5:25AM |
0 |
IAX2 Trunk Problem |
| 3:19AM |
1 |
yum om centos |
| 2:16AM |
3 |
Wireless IP Phone with external Telephone Book |
| 2:12AM |
0 |
Issue with Grandstream ATA 496. |
| 2:08AM |
1 |
Digium Card |
| 1:41AM |
0 |
Mixing Vars into Voicemail WAVs |
| 1:11AM |
0 |
nvlinedetect for Asterisk 1.4 |
| |
| Sunday June 3 2007 |
| Time | Replies | Subject |
|---|
| 5:57PM |
1 |
Can two card be configured on same machine. |
| 10:29AM |
2 |
wifi sip phone real-world experiences? |
| 10:22AM |
0 |
Strange problem with channel allocation |
| 9:57AM |
3 |
SIP Options Reply Ignored |
| 9:32AM |
1 |
Loud noise instead of MOH |
| 9:27AM |
2 |
Chan_mobile issue |
| 7:03AM |
1 |
FW: Centos kernel source |
| 6:03AM |
2 |
zaptel on CENTOS servercd |
| 4:50AM |
6 |
Centos kernel source |
| 4:48AM |
0 |
Telefonica in Czech Republic Blocking VOIP ? |
| 2:02AM |
2 |
Asterisk Queue |
| 1:42AM |
0 |
Asterisk Crash |
| |
| Saturday June 2 2007 |
| Time | Replies | Subject |
|---|
| 4:48PM |
1 |
linksys wip300 firmware |
| 3:19PM |
3 |
Dynamically adding Context in dialplan? |
| 2:38PM |
2 |
System Application, Fail/Timeout Issue |
| 9:37AM |
2 |
how to make busy sign |
| 2:34AM |
1 |
Asterisk registering problem |
| |
| Friday June 1 2007 |
| Time | Replies | Subject |
|---|
| 6:53PM |
0 |
WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 36458 udptl t38 |
| 5:57PM |
1 |
Call Back Service |
| 5:36PM |
0 |
OutBound dial plan |
| 3:00PM |
0 |
Gateway VIP450FO and VIP 400FO |
| 1:18PM |
2 |
SugarCRM Integration |
| 9:03AM |
1 |
Cisco 7961G |
| 8:59AM |
0 |
chan_iax2.so issues |
| 8:36AM |
2 |
asterisk mysql support |
| 8:12AM |
0 |
Meetme problems |
| 8:08AM |
1 |
Asteris et winsip |
| 7:51AM |
1 |
G729 client and server Side |
| 6:48AM |
0 |
OT: "The Ignorance of Crowds" (was: OT Slightly: ) |
| 6:45AM |
3 |
SIP & NAT ... |
| 5:42AM |
0 |
OT Slightly: |
| 3:04AM |
1 |
how can qualify=yes trigger some external event? |
| 2:31AM |
3 |
ZAP inbound/outbound connection taking too long |