| Friday February 29 2008 |
| Time | Replies | Subject |
|---|
| 10:24PM |
0 |
Skewed RTP timestamps in SIP calls on Asterisk 1.4.18 |
| 7:07PM |
1 |
IAX2's JB and DTMF |
| 5:20PM |
0 |
Request for testing: New wctdm24xxp and wcte12xp drivers. |
| 3:17PM |
1 |
Page app, Polycom IP 601, 60 SIP peers, Interesting Issue |
| 3:05PM |
1 |
Detecting SIT (Special Information Tone) on outbound calls |
| 2:47PM |
1 |
Gtalk with asterisk |
| 2:38PM |
1 |
bugs.digium.com |
| 2:33PM |
0 |
load balancing and high availability |
| 2:10PM |
2 |
when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init |
| 12:30PM |
1 |
asterisk queue agent problem |
| 12:12PM |
1 |
which phones to use ?? |
| 11:33AM |
1 |
basic installation |
| 11:09AM |
1 |
Cisco 7965g and asterisk |
| 7:30AM |
1 |
Can call in but cannot call out (CHANUNAVAIL): TE410 + Asterisk 1.4.13 + Zaptel 1.4.6 + Libpri 1.4.2 |
| 5:39AM |
3 |
Polycom IP600 + PC share same switch port with VLAN |
| 3:56AM |
0 |
I would like to hire someone to automate my asterisk for hosted PBX service |
| 3:39AM |
0 |
1EZphone is only two way browser softphone - SIP Softphones and Citrix ? |
| 2:01AM |
2 |
load balancing |
| |
| Thursday February 28 2008 |
| Time | Replies | Subject |
|---|
| 11:30PM |
2 |
Problems with removing zaptel |
| 10:18PM |
2 |
Asterisk Voicemail for iPhone |
| 10:04PM |
1 |
GLOBAL function - introduced at what version? |
| 7:08PM |
1 |
Problems with setting up Zaptel |
| 3:54PM |
1 |
quickfix for building zaptel with 2.6.24? |
| 3:30PM |
0 |
Digium certified asterisk professional linkedin group |
| 1:32PM |
1 |
Asterisk monitor() zap channel problem |
| 11:48AM |
2 |
New Interested services to be added for Telephoney Service Provider |
| 11:13AM |
1 |
Friday Feb 29th Leap Year Special wih Aastra |
| 10:49AM |
0 |
OT : OpenSER Summit & Pavilion - 17th to 19th of March, 2008 , San Jose, US |
| 10:21AM |
2 |
Asterisk and Cisco Unity? |
| 7:14AM |
1 |
C Code to connect to Asterisk Manager Interface |
| 2:03AM |
1 |
Unicall mfcr2 testcall issues in mexico outgoing:ok | incoming: fail. |
| |
| Wednesday February 27 2008 |
| Time | Replies | Subject |
|---|
| 8:55PM |
3 |
Simultaneous Inbound and Outbound calls on analog lines... |
| 7:41PM |
3 |
Attended transfers through a GUI |
| 6:18PM |
1 |
simultaneous ring problem |
| 5:25PM |
3 |
About faxes recived through a PRI and passed to a fax machine connected to a FXS port |
| 5:01PM |
1 |
What causes SIP 486? |
| 4:52PM |
0 |
Problems with Refered Call and Accountcode using siptapi |
| 4:42PM |
0 |
Attended transfers and orginal caller ID |
| 4:36PM |
5 |
Customer complains of noise on line I cannot reproduce. |
| 3:53PM |
1 |
Can AMD detect Service Provider Message. |
| 3:30PM |
1 |
Asterisk as SMSC to GSM-Phones |
| 2:51PM |
2 |
OT But I Would Rather See People Running Asterisk on a "Real" Server than an Emachine |
| 2:41PM |
1 |
SPA3102 registration problem |
| 2:29PM |
1 |
best practice |
| 12:40PM |
0 |
Entering code to restart the machine or reload iax |
| 9:38AM |
1 |
Zap Call deflection on PRI |
| 9:24AM |
2 |
Entering code to restart the machine |
| 8:32AM |
1 |
Danish callerid on a x100p card |
| 8:14AM |
0 |
Configuring modem pools in Asterisk |
| 8:03AM |
0 |
res_config_ldap in asterisk 1.6.0-beta2/4 |
| 7:38AM |
0 |
duplicated voicemail messages |
| 7:20AM |
1 |
Call recording problems from queue |
| |
| Tuesday February 26 2008 |
| Time | Replies | Subject |
|---|
| 11:10PM |
0 |
Anything like SipT38SwitchOver in Asterisk? |
| 10:48PM |
1 |
Configuring modem pools in Asterisk [WAS: Connecting a Rolm CBX to Asterisk via T1?] |
| 10:40PM |
0 |
How is reinvite triggered |
| 8:21PM |
1 |
How do I tell if T.38 was used? |
| 8:10PM |
7 |
Had it with Dell Garbage |
| 7:19PM |
1 |
How can I call cheap to UK cell phones |
| 6:24PM |
1 |
AMD on a SIP trunk... |
| 5:53PM |
2 |
Explain Cause of Error: manager.c: Accept returned -1: Too many open files |
| 5:31PM |
3 |
Sip trunk mystery |
| 2:44PM |
0 |
How to transfer an unanswered call??? |
| 1:50PM |
3 |
Asterisk as useragent registered using 2 accounts |
| 1:03PM |
1 |
iax trunking problem |
| 11:32AM |
0 |
CLIR missing in MySQL CDR records |
| 10:44AM |
6 |
[URGENT] Zap channels fail to load |
| 8:59AM |
3 |
chan_ss7 0.10 |
| 5:24AM |
0 |
Problem with rxfax |
| 3:41AM |
1 |
Still can't pickup parked call |
| 3:29AM |
1 |
Anybody installed Asterisk in a Virtuozzo VPS system??? |
| 1:03AM |
2 |
Parked calls - can't pickup |
| |
| Monday February 25 2008 |
| Time | Replies | Subject |
|---|
| 10:38PM |
2 |
cannot dial out with latest zaptel and kernel 2.6.24 |
| 9:35PM |
1 |
Realtime Queue Status for Agents |
| 8:27PM |
0 |
Considering replacing ATA with linecard. |
| 8:27PM |
1 |
DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P) |
| 8:12PM |
1 |
Problem with asterisk and aastra phones |
| 6:12PM |
1 |
TE120P echo cancellation problem |
| 2:54PM |
0 |
question on call setup |
| 2:36PM |
0 |
BarCampNYC3 |
| 2:22PM |
1 |
VOIP Application on Dating Contact Service |
| 12:20PM |
3 |
DDNS and host: updating when destination IP changes |
| 8:42AM |
4 |
TDM400P dialout problem |
| |
| Sunday February 24 2008 |
| Time | Replies | Subject |
|---|
| 11:16PM |
1 |
beta4: outgoing call causes Red Alarm on TDM400P |
| 12:14PM |
0 |
Load balancing SIP extensions. |
| 8:01AM |
2 |
DUNDi with two servers |
| 6:25AM |
0 |
Call limits per server with Iax |
| |
| Saturday February 23 2008 |
| Time | Replies | Subject |
|---|
| 11:29PM |
1 |
Need some dialplan help |
| 8:20PM |
1 |
dundi lookup |
| 3:36PM |
1 |
Suggestions for reliable DID provider forCanada, USA and Europe |
| 12:51PM |
1 |
Fax-to-Email - Legal Issues |
| 12:22PM |
3 |
Suggestions for reliable DID provider for Canada, USA and Europe |
| 9:12AM |
0 |
SIP peers from multiple databases |
| 12:28AM |
2 |
MySQL Voicemail Storage Questions\Errors |
| |
| Friday February 22 2008 |
| Time | Replies | Subject |
|---|
| 11:03PM |
1 |
Mexico Dids |
| 10:08PM |
1 |
Post call QoS in Asterisk 1.4 |
| 8:45PM |
1 |
Polycom 301/501 Keymapping |
| 7:57PM |
0 |
is tos=ef same as tos=0xb8 same as DSCP ef ? |
| 7:57PM |
1 |
Message waiting light on polycom 301 using asterisk 1.4.14 |
| 6:58PM |
2 |
AGI / Voicemail Que |
| 6:33PM |
5 |
NOKIA E series Phone for SIP-VOIP calling |
| 2:05PM |
2 |
Interrupt VM and Steal a call. |
| 1:59PM |
0 |
adjusting volume on a wildcard 100XP with zaptel's {t, r}xgain |
| 1:51PM |
3 |
Will this be sufficient for 20+ concurrent calls? |
| 1:43PM |
1 |
[VOIP-Users-Conference] Re: Opinions please: Polycom IP 430 vs 330? |
| 1:17PM |
0 |
Opinions please: Polycom IP 430 vs 330? |
| 1:09PM |
0 |
DUNDi ${NUMBER} variable not defined |
| 12:56PM |
2 |
Linksys SPA-942 Phones |
| 11:42AM |
5 |
load balancing SIP extensions |
| 10:50AM |
1 |
canreinvite question |
| 10:40AM |
2 |
Digium B410P and 8 ports connectivity |
| 10:20AM |
0 |
Using app_sms in South Africa |
| 9:57AM |
1 |
Weird Zaptel sound after anwser calls |
| 7:37AM |
0 |
chan_h323 build failure - `IPTOS_MINCOST' undeclared |
| 7:02AM |
0 |
chan_woomera tries to connect to strange host |
| 6:58AM |
0 |
Friday 22 FEB 08 @ 12 Noon EST ISPBX COGOBLUE |
| 5:52AM |
1 |
Pager ("beeper") Emulation Script |
| 5:08AM |
2 |
(no subject) |
| 3:54AM |
1 |
spandsp/tx_fax/rx_fax frustrations |
| 3:07AM |
0 |
Chan_h323 isn`t dropping calls comming with wrong codecs |
| 12:58AM |
1 |
FW: jabber |
| |
| Thursday February 21 2008 |
| Time | Replies | Subject |
|---|
| 11:18PM |
0 |
Question regarding AGI |
| 10:45PM |
1 |
Asterisk, Zaptel and the Kernal Compatibility Matrix |
| 10:07PM |
1 |
cid_rewrite.php -- Caller ID Name lookup |
| 9:12PM |
2 |
High CPU load after upgrading to 1.4 |
| 8:21PM |
0 |
Asterisk-addons 1.6.0-beta2 Released |
| 8:13PM |
0 |
Asterisk 1.6.0-beta4 Released |
| 7:57PM |
0 |
Asterisk-addons 1.4.6 Released |
| 5:44PM |
3 |
Pattern matching.... |
| 5:40PM |
1 |
IVR No sound on other provider |
| 5:37PM |
1 |
Which echo-can for Digium B410P ? |
| 4:13PM |
0 |
Maybe OT: SIP - Missing 407 messages |
| 4:04PM |
1 |
Answered Call marked as "NO ANSWER" |
| 4:00PM |
0 |
SendDTMF not Working - Possible Echo Cancelling Issues |
| 3:19PM |
0 |
HoldMusic Beep |
| 2:53PM |
0 |
Contents of asterisk-users digest |
| 1:42PM |
1 |
USB ISDN interface |
| 1:38PM |
2 |
Allow INVITE for hold to pass through |
| 11:37AM |
0 |
UCS-2 Problem |
| 9:07AM |
0 |
Third Party Call Control - SIP to Iax Gateway |
| 9:07AM |
3 |
Voted most stable and easy to use phone? |
| 8:58AM |
4 |
chan_h323 requirements |
| 5:57AM |
2 |
Can asterisk support 20 user's conference? |
| 4:59AM |
2 |
Asterisk Nagios |
| 2:49AM |
1 |
Multiple Asterisk Servers. One Conference |
| 2:31AM |
2 |
Converence/Meetme with Manager API |
| 2:30AM |
3 |
How to get a clean, basic configuration? |
| 12:53AM |
0 |
How to Configure 1.4.17 to Store CDR's in PostgreSQL |
| |
| Wednesday February 20 2008 |
| Time | Replies | Subject |
|---|
| 11:48PM |
0 |
Call drops to fast busy |
| 10:52PM |
6 |
Coppercom and Asterisk |
| 10:31PM |
0 |
Southern Alberta Canada * users. |
| 9:26PM |
8 |
Best ATA. Period. |
| 8:39PM |
0 |
debugging stuck led lights |
| 7:47PM |
2 |
Polycom Key Assignment |
| 7:44PM |
5 |
ata device but for a soundcard |
| 7:30PM |
1 |
Include in asterisk realtime |
| 7:00PM |
1 |
Receiving double DTMF |
| 6:39PM |
1 |
manager ignore my settings |
| 6:12PM |
1 |
IAX: No outgoing audio for 10 seconds |
| 6:08PM |
0 |
Strange NewCallerIDEvent after channel are linked |
| 3:17PM |
2 |
Skype Users |
| 1:50PM |
4 |
GXP-2020 Transfer Key |
| 12:15PM |
1 |
problem transferring calls some of the times |
| 10:07AM |
0 |
Unable to create channel of type 'Zap' with ecmg2 and kernel 2.6.23 |
| 8:28AM |
3 |
Dial+Macro and Queue |
| 7:14AM |
1 |
OT - DECT-GAP Handsets with Polycom-Kirk 600/3 base station |
| 5:31AM |
1 |
Need to Connect offices in Dubai and Pakistan |
| 3:50AM |
1 |
which codec over iax => pstn |
| 3:09AM |
0 |
GS/* phonebook + Web-BLF |
| 12:27AM |
2 |
Sangoma FXO EC vs Rhino FXO EC |
| |
| Tuesday February 19 2008 |
| Time | Replies | Subject |
|---|
| 11:49PM |
0 |
More on Broadvoice w/Asterisk (1.4.18) |
| 11:13PM |
0 |
Restricting registration for peer 'iaxmodem0' to60 seconds |
| 10:49PM |
1 |
Restricting registration for peer 'iaxmodem0' to 60 seconds |
| 9:07PM |
0 |
Detecting the digits |
| 6:16PM |
1 |
Zaptel version, and compatibility matrix |
| 4:33PM |
1 |
Connecting a UMTS module via USB to asterisk |
| 4:05PM |
1 |
MeetMe Admin Functions |
| 3:59PM |
0 |
Extension Logic Help |
| 3:44PM |
2 |
asterisk config file online editor |
| 2:59PM |
0 |
two lines written in CDR for each failed call in asterisk 1.4 |
| 2:55PM |
0 |
ASTERISK C File |
| 12:46PM |
0 |
IAX registration problem |
| 12:00PM |
1 |
SIP Request: OPTIONS |
| 9:59AM |
1 |
A problem about digium TE220B |
| 9:54AM |
0 |
jabber |
| 9:43AM |
0 |
[Copfilter] Copy of quarantined email - *** SPAM *** [6.0/6.0] IAX2 client asked to authenticate against wrong |
| 6:53AM |
3 |
No compatible codecs! |
| 2:43AM |
0 |
Request for testing: Distributed device state |
| 1:45AM |
5 |
Asterisk 1.4 vs 1.6 |
| |
| Monday February 18 2008 |
| Time | Replies | Subject |
|---|
| 11:52PM |
1 |
ztscan ports = zaptel channels ?? |
| 11:10PM |
0 |
Zaptel 1.2.24 and 1.4.9 Released |
| 11:06PM |
0 |
Asterisk 1.6.0-beta3 Released |
| 10:18PM |
1 |
Asterisk: how to limit h323 connections. |
| 10:17PM |
1 |
Attatch monitor recording to a voicemail |
| 8:00PM |
1 |
Avaya 4610sw |
| 7:14PM |
5 |
Cisco SIP Gateway |
| 6:21PM |
0 |
Pulling a variable from a shell script into Asterisk - backticks? |
| 5:15PM |
0 |
logging the estimated RTT using SIP |
| 4:04PM |
0 |
Problems with TE120, Kernel BUG |
| 3:53PM |
1 |
realtime table customization to track iax registrations |
| 3:08PM |
2 |
SPA-3000 caller ID and KPN |
| 2:28PM |
0 |
Changing the automon output filename |
| 1:26PM |
4 |
IAXModem - NDID=s |
| 1:06PM |
0 |
Contents of asterisk-users submissions |
| 10:37AM |
0 |
Please reply..Not able to call H323 using SIP client |
| 10:24AM |
2 |
SiP call generator |
| 10:15AM |
1 |
ForkCdr in 1.4.* |
| 8:37AM |
0 |
from address modification |
| 8:33AM |
0 |
Vancouver - Asterisk Event Feb 18 (Monday) |
| 7:36AM |
1 |
mfcr2 stuck |
| 7:36AM |
0 |
IAX2 client asked to authenticate against wrong peer (username) |
| 6:42AM |
0 |
Not able to call H.323 client by SIP client |
| 6:11AM |
1 |
PRI dialplan/prefix |
| 5:35AM |
3 |
ISDN2 facility code... |
| 4:30AM |
2 |
Failure of Sending Voicemail As an attachment in E-mail |
| 2:17AM |
1 |
Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic) |
| |
| Sunday February 17 2008 |
| Time | Replies | Subject |
|---|
| 7:50PM |
1 |
app_voicemail - Failed to open file ../tmp/xxxxx.WAV |
| 6:22PM |
0 |
Able to receive calls on my fxo port and not able to do calls via it |
| 12:20PM |
1 |
Music on hold |
| 12:16PM |
2 |
Asterisk reltime mode with Postgresql |
| 9:40AM |
1 |
IAX2 trunks unreliable becoming UNREACHABLE aftera time |
| 12:42AM |
1 |
Asterisk H.248 Support |
| |
| Saturday February 16 2008 |
| Time | Replies | Subject |
|---|
| 10:22PM |
1 |
Zaptel and Asterisk compilation |
| 5:46PM |
2 |
T1 "access layer" used Cisco or new Digium |
| 5:07PM |
2 |
zaptel: modpost section mismatch ? |
| 3:28PM |
1 |
Fritz! Card/CAPI Help. |
| 1:56AM |
0 |
Digium stopped TDM400P production: alternatives?? ?In-Reply-To: <47B59559.1030706@rhinoequipment.com> ?References: <47B5778F.2040609@fgasoftware.com> <47B5837E.6030800@digium.com> ? <ea18e54a0802150432k59198302p85b84f43483ccabf@mail.gmail.com> ? |
| 1:44AM |
0 |
arris tm502g cablemodem FXS ports and zaptel 1.4.8 |
| |
| Friday February 15 2008 |
| Time | Replies | Subject |
|---|
| 8:49PM |
8 |
Connecting a Rolm CBX to Asterisk via T1? |
| 6:40PM |
0 |
Zaptel compilation problems. |
| 11:58AM |
1 |
1.4 and IAX Trunks ... |
| 11:29AM |
7 |
Digium stopped TDM400P production: alternatives?? |
| 9:18AM |
0 |
G729 transcoding and "clicking" |
| 8:54AM |
3 |
Communication between two asterisk server |
| 8:35AM |
0 |
Question about DIALSTATUS NOANSWER |
| 6:02AM |
0 |
How to check if a local channel member of a queue? |
| 2:18AM |
0 |
patch which makes Asterisk-Addons 1.4.5 work when codec negotiation patch applied to asterisk |
| 2:08AM |
2 |
HPEC |
| 1:15AM |
2 |
57iCT BLF problem |
| 1:04AM |
1 |
DialPlan help with Analog Fax Machine |
| |
| Thursday February 14 2008 |
| Time | Replies | Subject |
|---|
| 11:20PM |
1 |
gtalk and dtmf |
| 9:51PM |
0 |
ExtenSpy strange behavior on Asterisk 1.4.18 |
| 7:26PM |
0 |
Emagen (a Telrad VM solution) -- any way to replace with *? |
| 7:04PM |
1 |
Variable setting in AMI Originate |
| 5:41PM |
1 |
X100P Burnouts |
| 5:30PM |
1 |
SNMP monitoring |
| 5:08PM |
0 |
IAX load balancing |
| 4:22PM |
2 |
Pass arguments from extensions.conf |
| 4:05PM |
0 |
translating iax2 register into sip register |
| 3:30PM |
5 |
Monitor Asterisk |
| 12:17PM |
1 |
Error checking asterisk method - suggestions? |
| 11:12AM |
1 |
Ser, asterisk and ip2ipgw |
| 9:39AM |
6 |
UK -999 dialing issue |
| 5:30AM |
0 |
Snom Provisioning Tool Release |
| 12:49AM |
1 |
Touch monitor file name format |
| |
| Wednesday February 13 2008 |
| Time | Replies | Subject |
|---|
| 10:42PM |
5 |
multiple host in 1 context on sip.conf |
| 10:21PM |
3 |
SIP over TCP |
| 8:31PM |
3 |
Asterisk Manager and Visual Basic |
| 6:45PM |
2 |
MWI problem with Siemens Gigaset S675 IP |
| 6:03PM |
1 |
ISDN PRIs and taking a server down for maintenance - blocking issue |
| 4:54PM |
2 |
Asterisk and fax |
| 4:40PM |
3 |
Analog DID |
| 3:59PM |
1 |
FOSDEM in Brussells - Feb 23-24 |
| 3:58PM |
1 |
GXP2000 and asterisk 1.0.9 |
| 2:36PM |
0 |
Digium's Exceptional Satisfaction Program |
| 2:22PM |
3 |
What is a "secure call"? |
| 2:12PM |
2 |
UK issue - Asterisk dialling 999... sort of |
| 1:46PM |
0 |
Wanted: VoIP Engineer for Switerland |
| 12:46PM |
2 |
[Linux/Python 2.4.2] Forking Python doesn't work |
| 12:35PM |
4 |
Telephone line signaling configuration in Egypt for FXO ports |
| 12:30PM |
4 |
Attendant phone |
| 9:33AM |
3 |
urgent-channels |
| 9:15AM |
1 |
Hardware needed |
| 8:10AM |
0 |
Friday Feb 15th @ 12 Noon EST: VoIP Users Conference welcomes Lumenvox |
| 7:58AM |
2 |
PCI32 and PCI-X compatibility |
| 5:06AM |
0 |
Netgear TA612V line 2 and asterisk |
| 2:49AM |
3 |
How to soft hangup all channels at a time . |
| 1:58AM |
6 |
restart asterisk daily |
| 12:39AM |
0 |
OT: 3rd party SMS service? |
| |
| Tuesday February 12 2008 |
| Time | Replies | Subject |
|---|
| 10:23PM |
0 |
play greeting from odbc voicemail |
| 4:04PM |
0 |
Asterisk bridging timeout when calling out with SIP phone |
| 3:53PM |
3 |
LCR in Asterisk |
| 2:56PM |
1 |
Zhone Channel Bank |
| 2:33PM |
0 |
which mobile compatible with asterisk |
| 9:11AM |
0 |
* SIP dial out with multiple sip users |
| 3:50AM |
1 |
chan_ooh323 patches compatible with codec negotiation patch applied to asterisk 1.4.17 |
| 2:18AM |
3 |
Nortel 1140E |
| |
| Monday February 11 2008 |
| Time | Replies | Subject |
|---|
| 11:39PM |
0 |
OT: Recommendation for EDGE/3GPP SIP phone? |
| 11:00PM |
1 |
SIP Bad request protocol Packet on Asterisk 1.4.18 |
| 10:24PM |
0 |
SPA3000 + asterisk +call waiting |
| 10:11PM |
0 |
Configuring Polycom SP300 -- weird problem |
| 9:52PM |
2 |
Grandstream GXP2000 Loses Connectivity |
| 7:44PM |
5 |
Automatically start after restart |
| 7:22PM |
2 |
Automon reliability issue |
| 6:32PM |
1 |
G729 without licence |
| 4:36PM |
0 |
Semi-OT: bluetooth conference phone? |
| 4:20PM |
1 |
message: !! Got Busy in Connected State !?! |
| 3:25PM |
3 |
differences |
| 2:16PM |
1 |
Asterisk as a softswith for a small ISP |
| 1:10PM |
1 |
Iax2 provisioning and Asterisk |
| 11:41AM |
0 |
asterisk-users Digest, Vol 43, Issue 30 |
| 10:19AM |
0 |
TE412P on Intel S5000XVN and S5000VSA Motherboards |
| 8:22AM |
1 |
Single * multiple offices |
| 6:24AM |
1 |
Realtime SIP peers - reloading cached info |
| 5:56AM |
1 |
Asterisk Meetme & its Admin |
| 4:45AM |
0 |
asterisk, asterisk-addons, ooh323, codec negotiation patch in FreeBSD |
| 12:24AM |
2 |
how to create a standalone voicemail server |
| |
| Sunday February 10 2008 |
| Time | Replies | Subject |
|---|
| 11:41PM |
1 |
Generate anonymous SIP Calls |
| 11:18PM |
1 |
SIP proxy/registration for * |
| 7:33PM |
4 |
IAX2 trunks unreliable becoming UNREACHABLE after a time |
| 6:29PM |
3 |
usability Testing Costa Rica, SanJose asterisk PBX / dsl/cable service |
| 1:55PM |
2 |
Still dropped calls :( |
| 7:44AM |
1 |
Disappearing B-Channels |
| 4:20AM |
0 |
Carrier SIP resource? |
| 1:01AM |
1 |
HP proliant and hpasm |
| |
| Saturday February 9 2008 |
| Time | Replies | Subject |
|---|
| 6:44PM |
2 |
[asterisk-dev] Monitor Asterisk using C |
| 5:09PM |
2 |
Cisco phone 79xx get database information |
| 5:06PM |
1 |
BLF and Asterisk 1.6.0b2 |
| 11:11AM |
1 |
SIP user registration and Asterisk Realtime |
| 9:04AM |
1 |
voicemail to non-default context user does not work |
| 7:21AM |
1 |
Dialing SIP server user extension... Dial string issue... |
| 4:54AM |
2 |
oneway audio with asterisk behind cisco pix 506 |
| 12:49AM |
0 |
[asterisk-biz]SIP to SIP professional community |
| 12:21AM |
1 |
Sending a message from inside voicemailmain. |
| |
| Friday February 8 2008 |
| Time | Replies | Subject |
|---|
| 10:32PM |
0 |
Rejected calls to Sylantro server |
| 7:03PM |
0 |
Dealipedia |
| 6:32PM |
3 |
Monitor Asterisk using C |
| 4:46PM |
0 |
GS/* phonebook |
| 4:27PM |
0 |
Interoperability between TE412P and Eurotech PRI E1 GSM & CDMA Gateway |
| 3:58PM |
0 |
canreinvite option - gona have problems? |
| 3:12PM |
3 |
Question about Asterisk versions (newbie) |
| 2:52PM |
2 |
Upgrade 1.2 -> 1.4 voice files |
| 2:27PM |
1 |
Permission denied when obtaining Status |
| 12:24PM |
1 |
Domainname for outgoing uri-dialing |
| 10:11AM |
1 |
Cosini iAN7s |
| 10:05AM |
1 |
Transferring a call received by an agent in a queue |
| 7:53AM |
1 |
(no subject) |
| 7:41AM |
0 |
VoIP Users Conference Call Today Friday @ 12 Noon EST |
| 6:51AM |
1 |
Asterisk queue not play muscinhold or hangup |
| 1:54AM |
0 |
Transcoded G.722 calls unintelligible with recent SVN head |
| |
| Thursday February 7 2008 |
| Time | Replies | Subject |
|---|
| 10:11PM |
2 |
Asking for recommendations on Asterisk Boxes or Appliances |
| 8:18PM |
0 |
Asterisk 1.4.18 Released |
| 7:37PM |
2 |
Snom 300 MWI |
| 6:24PM |
2 |
Asterisk as XMPP component. How to use it ? |
| 6:19PM |
0 |
Asterisk trunk/1.6 and nvfaxdetect |
| 5:40PM |
4 |
Snom 300 Echo |
| 5:11PM |
6 |
Asterisk G722 |
| 3:00PM |
2 |
How to balance traffic between 2 gateways ? |
| 2:05PM |
2 |
Goto in Realtime extensions |
| 12:39PM |
1 |
SIP / RTCP statistics logging |
| 12:32PM |
1 |
FW: transcoder |
| 9:22AM |
5 |
Two Leg CDR |
| 9:17AM |
1 |
Preventing IAX frame concatenation |
| 3:33AM |
0 |
New deployment questions |
| 3:04AM |
1 |
OT: POTS telephone like the SPA-942? |
| 2:45AM |
3 |
Need good voicemail documentation |
| 1:48AM |
0 |
Asterisk and Avaya phone system |
| 12:18AM |
3 |
Matching "+" characters in dial plan |
| |
| Wednesday February 6 2008 |
| Time | Replies | Subject |
|---|
| 11:03PM |
0 |
Post Call QoS....? |
| 9:53PM |
1 |
TE412P and Delll PowerEdge 2900 |
| 9:09PM |
2 |
AGI Process Count (HOWTO?) |
| 7:54PM |
1 |
FXO modules and polarity reverse |
| 7:46PM |
0 |
Need a dial rule to match and replace a number. |
| 6:43PM |
3 |
Polycom BLF / Speed Dial |
| 6:22PM |
1 |
TDM400P phone won't ring |
| 5:00PM |
1 |
[OT] ISDN 30 (PRI) service in the Netherlands |
| 12:50PM |
0 |
Problem forwarding a call with an AGI script |
| 11:05AM |
0 |
Directing SIP/RTP sessions b/w UA |
| 9:28AM |
0 |
How to register h323 users? |
| 12:50AM |
3 |
R2 with Alestra in Mexico... |
| 12:39AM |
1 |
Gemeinschaft released |
| |
| Tuesday February 5 2008 |
| Time | Replies | Subject |
|---|
| 11:38PM |
2 |
Can't delete voicemail messages |
| 8:32PM |
4 |
Cannot hear voice through SIP Phone from one side |
| 8:10PM |
4 |
How to hookup to cell phone for outbound calls? |
| 6:29PM |
0 |
What causes this? |
| 6:00PM |
0 |
Post Call QoS? |
| 4:53PM |
0 |
meetme with ztxen - WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device |
| 4:34PM |
1 |
Telephony Migration Hardware |
| 3:22PM |
1 |
is encrypted iax safe and secure? |
| 1:52PM |
3 |
wireless VOIP phone recommendations? |
| 9:30AM |
1 |
Mistake in the wiki's description of cmd Pickup() ? |
| 8:36AM |
1 |
Can't dial out from SIP to CAPI |
| 8:07AM |
3 |
[Softphones] ZoIPer vs. XLite? |
| 7:35AM |
0 |
Asterisk does not handle INVITE authentication by Proxy |
| 5:54AM |
2 |
basic server specs |
| 5:31AM |
6 |
External MWI question for Asterisk |
| |
| Monday February 4 2008 |
| Time | Replies | Subject |
|---|
| 6:27PM |
1 |
GROUP_COUNT and Attended transfer |
| 5:22PM |
6 |
transcoder |
| 4:15PM |
3 |
Asterisk with 2 NIC cards |
| 3:16PM |
0 |
error while loading zaptel or ztdummy module under kernel 2.6.18-6-xen-amd64 - no sound in asterisk |
| 2:25PM |
1 |
one CDR instead of multiple CDR |
| 1:53PM |
0 |
Asterisk mishandling user busy isdn releases |
| 11:14AM |
1 |
OT POlycom question |
| 10:22AM |
0 |
AGENTDUMP lines in queue.log???? |
| 9:30AM |
0 |
PRI ISSUE |
| 9:08AM |
2 |
Losing CALLERID{dnid} |
| 8:44AM |
1 |
PRI with 20 channels |
| 8:37AM |
1 |
asterisk-gui installation hangs |
| 8:25AM |
1 |
Got SUBSCRIBE for extension ... but there is no hint for that extension. |
| 8:18AM |
0 |
Problem picking up a call with PickUpChan or PickUp [SOLVED] |
| 6:14AM |
2 |
Problem with IRQ Share |
| 2:50AM |
8 |
AGI: Not getting answers from get_data in a call-file call |
| |
| Sunday February 3 2008 |
| Time | Replies | Subject |
|---|
| 11:56PM |
3 |
Console/dsp, makes me sound like a Dalek |
| 8:14PM |
1 |
switch QOS requirements |
| 8:10PM |
3 |
Test |
| 5:09PM |
1 |
Telco MWI Detection on TDM400 Interface? |
| 1:25AM |
1 |
Multiple SIP phones behind a Linksys firewall |
| |
| Saturday February 2 2008 |
| Time | Replies | Subject |
|---|
| 9:09PM |
0 |
SIP: IP in the VIA-Header |
| 8:06PM |
1 |
app_valetparking.c anyone using it on 1.4? |
| 8:46AM |
3 |
Zaptel timer on Intel Dual Core servers |
| 8:45AM |
2 |
ATA with pulse dialing support over FXS |
| 1:53AM |
2 |
Polycom - Buddy Watch not a choice when adding Speed Dial |
| 1:01AM |
1 |
Echo() app doesn't work |
| 12:20AM |
0 |
IAX Registraion Refresh |
| |
| Friday February 1 2008 |
| Time | Replies | Subject |
|---|
| 10:57PM |
4 |
"Real" API for Perl? |
| 10:56PM |
2 |
X-Lite Softphone keeps de-registering? |
| 10:30PM |
0 |
Trying to make SIP calls through Asterisk with anonymous connection |
| 8:32PM |
1 |
QueueMember event/LastCall Variable - Format? |
| 8:31PM |
2 |
It's about time! -- Digium PCI-Express Cards |
| 7:47PM |
0 |
Bypassing a Auth on Invite or Forbiden? |
| 6:57PM |
2 |
Asterisk 1.4.17 and Teliax DTMF |
| 5:34PM |
2 |
Asterisk 1.6 - Problems with SIP/REFER |
| 5:29PM |
1 |
Astersik Transcoder support |
| 4:02PM |
1 |
Problem picking up a call with PickUpChan or PickUp (asterisk-users Digest, Vol 43, Issue 1) |
| 3:35PM |
0 |
Codec preference selection, codec negotiation |
| 3:33PM |
3 |
Remote Call Center Agents and Asterisk? |
| 3:12PM |
3 |
SIP Softphones and Citrix ? |
| 2:16PM |
7 |
Enterprise or Fedora? |
| 1:37PM |
1 |
BRI card with PCI-E interface |
| 1:24PM |
1 |
Unicall |
| 1:08PM |
1 |
play promt at the same time to calling and callee |
| 9:59AM |
0 |
call log notice messages |
| 9:39AM |
1 |
meetme music on hold - when only conference member problem |
| 6:54AM |
1 |
Asterisk-Addons install success-Could not find ooh323.conf |
| 4:55AM |
2 |
h priority problem |
| 3:40AM |
1 |
realtime warning |