| Sunday November 30 2008 |
| Time | Replies | Subject |
|---|
| 11:02PM |
1 |
Oslec issue |
| 5:27PM |
3 |
DTMF Tones |
| 12:39AM |
1 |
DAHDI issue in dialplan |
| |
| Saturday November 29 2008 |
| Time | Replies | Subject |
|---|
| 10:54PM |
0 |
pp_each_user(), pp_each_extension() |
| 7:33PM |
0 |
asterisk-users Digest, Vol 52, Issue 81 |
| 1:19PM |
1 |
GSM gateways - which one ? |
| 5:43AM |
1 |
libspandsp.so.0: cannot open shared object file: No such file or directory |
| 4:18AM |
2 |
Trixbox 2.6.1.13 OpenR2 |
| 3:02AM |
0 |
received wrong state events for originate command |
| |
| Friday November 28 2008 |
| Time | Replies | Subject |
|---|
| 7:06PM |
1 |
How to disable trunk from the cli? |
| 4:20PM |
0 |
Asterisk and multicast RTP |
| 4:11PM |
0 |
Friday at 12 Noon ET, the VoIP Users Conference reminder |
| 4:00PM |
1 |
Asterisk SIP security |
| 3:56PM |
0 |
Calls drop after a couple of minutes. |
| 3:07PM |
1 |
RTCP too short |
| 1:32PM |
0 |
length of field names |
| 12:24PM |
0 |
[SPAM] - Asterisk and S-Bus - Email found in subject |
| 11:13AM |
1 |
Priority between calls from different queues |
| 10:18AM |
0 |
[SPAM] - Re: FW: cdr_addon_mysql.so did notregister itselfduringload - Email found in subject |
| 9:56AM |
1 |
MixMonitor with non-20ms packets |
| 5:51AM |
1 |
Anonymous callerid |
| 3:01AM |
1 |
Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call |
| 2:25AM |
2 |
force channel hangup |
| |
| Thursday November 27 2008 |
| Time | Replies | Subject |
|---|
| 4:02PM |
1 |
originate problem |
| 1:05PM |
2 |
Wellgate & Asterisk |
| 11:09AM |
0 |
trunk peer not registering after migrating installation |
| 11:03AM |
5 |
Any 1.6 SendFAX example ? |
| 10:14AM |
0 |
Disable Transfer |
| 8:24AM |
1 |
Unknown signalling method 'bri_cpe |
| 6:11AM |
0 |
Softphones with RPID and BLF |
| |
| Wednesday November 26 2008 |
| Time | Replies | Subject |
|---|
| 9:36PM |
0 |
CDR Hangupcause |
| 9:14PM |
0 |
Softphone IP publico x privado |
| 9:04PM |
3 |
2 Asterisks to one PBX - E1 conection |
| 7:09PM |
0 |
spandsp not recognized by menuselect on Lenny [SOLVED] |
| 5:24PM |
0 |
Problems with Rhino Channelbank... |
| 4:18PM |
1 |
sip MWI Messages-Waiting: always reports no messages |
| 4:02PM |
6 |
spandsp not recognized by menuselect on Lenny |
| 3:48PM |
1 |
Customized CDR Records |
| 3:29PM |
0 |
1.4.x Strange Vocemail delay |
| 2:41PM |
0 |
MS Exchange IMAP Voicemail |
| 2:19PM |
7 |
Dahdi, b410p and looping from 1 port to another |
| 2:15PM |
1 |
language and meetme issue |
| 2:10PM |
8 |
Mobile as FXO |
| 12:33PM |
1 |
Hints stopped working suddently |
| 11:47AM |
1 |
Channel variable to identify the calling SIP peer |
| 11:06AM |
1 |
SVN |
| 8:30AM |
0 |
Asterisk voicemail and Lotus Notes |
| 4:26AM |
2 |
Ring/Off-hook in strange state 6 channel X |
| 1:14AM |
1 |
bridging - Didn't get a frame from channel |
| |
| Tuesday November 25 2008 |
| Time | Replies | Subject |
|---|
| 10:56PM |
8 |
Connecting AS5350XM with Asterisk |
| 9:17PM |
2 |
half channel audio after upgrade to 1.4.18 |
| 6:33PM |
4 |
MOH Realtime |
| 3:15PM |
1 |
AsteriskNOW 1.5 upgrade from 1.4 to 1.6 |
| 1:04PM |
2 |
Disabling Call-Waiting |
| 11:22AM |
0 |
Set a specific registration expiry value to a given peer without touching defaultexpiry in sip.conf ? |
| 7:10AM |
1 |
cdr mysql error |
| |
| Monday November 24 2008 |
| Time | Replies | Subject |
|---|
| 5:44PM |
0 |
reducing iax packet size |
| 12:51PM |
1 |
play sound while executing agi script |
| 10:12AM |
1 |
SendText and non-ASCII characters |
| |
| Sunday November 23 2008 |
| Time | Replies | Subject |
|---|
| 10:47PM |
2 |
asterisk and vlc |
| 10:44PM |
2 |
Problem with DAHDI and OSLEC integration. |
| 8:28PM |
1 |
Asterisk 1.6, IMAP Voicemail and externnotify |
| 2:42PM |
14 |
CDR Design |
| 12:19PM |
1 |
Asterisk 1.6 mysql cdr log problem |
| 4:17AM |
2 |
How does IMAP notify Asterisk that I've read a message? |
| 3:22AM |
1 |
SendImage() |
| 3:09AM |
0 |
Large Asterisk installations (~10, 000 extensions), preferably at universities |
| |
| Saturday November 22 2008 |
| Time | Replies | Subject |
|---|
| 1:55PM |
1 |
IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...) |
| 7:03AM |
1 |
Upgrade 1.4.19 to 1.6 => segementation fault |
| 6:42AM |
1 |
Asterisk Instant message passing with eyebeam |
| 6:08AM |
2 |
Need Recording Solution in Asterisk |
| 4:02AM |
5 |
CDR Desgin |
| |
| Friday November 21 2008 |
| Time | Replies | Subject |
|---|
| 11:22PM |
0 |
MoH in a loop |
| 11:09PM |
2 |
MOH Realtime Problem |
| 9:12PM |
2 |
MozIAX - Mozilla IAX2 soft-phone 3sec delay |
| 8:49PM |
1 |
Setting up to reveive faxes. |
| 6:09PM |
2 |
TrixBox problem... |
| 5:51PM |
2 |
hint priority with 50 channels |
| 5:28PM |
2 |
Log level of 500 Server Internal Error. |
| 4:32PM |
0 |
OT - SIP message encoding to enhance text display |
| 4:28PM |
4 |
upgrade from 1.2 to 1.4 and now half channel audio |
| 2:59PM |
1 |
Ping |
| 2:06PM |
2 |
SPA2100 transfer to ASTERISK CID |
| 10:09AM |
0 |
PSTN Gateway setup |
| 7:53AM |
2 |
sip trunking and call transfer |
| 4:54AM |
4 |
Large Asterisk installarions (~10, 000 extensions), preferably at universities |
| 4:15AM |
0 |
Group count not being preserved when transferring a call into a conference |
| |
| Thursday November 20 2008 |
| Time | Replies | Subject |
|---|
| 11:31PM |
2 |
SVN - DIGIUM |
| 11:20PM |
2 |
A way to run extenrnotify when IMAP events take place... |
| 10:55PM |
2 |
ISDN Cause codes |
| 10:33PM |
1 |
Playback using AMI |
| 10:27PM |
0 |
OT: ATA causes random DTMF in stream |
| 9:58PM |
2 |
Limit the number of users in a meetme conference? |
| 9:33PM |
0 |
DTMF issue |
| 8:55PM |
0 |
dial console/dsp hear crackling in headset |
| 8:52PM |
0 |
Elastix workshop in Toronto; Wed Nov 26th, 2008 |
| 8:14PM |
1 |
A question about how much an Asterisk Dcap consultant and a Sipmaster make |
| 7:07PM |
0 |
Rv: sas |
| 6:54PM |
0 |
VB 6 developer needed |
| 6:01PM |
0 |
Subversion Mirror Down for Maintenance |
| 5:49PM |
0 |
Disable native bridge? |
| 5:47PM |
1 |
Low RX volume and half duplex/"walkie-talkie" on AEX-804E |
| 5:33PM |
1 |
Sending / Receiving sms messages with Portech 370 |
| 5:02PM |
4 |
SIP to IAX2 with delayed echo |
| 4:02PM |
4 |
Using MAC or extension number as SIP identifier |
| 3:57PM |
1 |
Macro conversion in 1.6 |
| 2:32PM |
2 |
Collect digits from the Callee after the Call is connected. |
| 1:51PM |
1 |
Voicemail in Real Time |
| 1:46PM |
1 |
Load balancing Asterisk. |
| 12:45PM |
0 |
DTMF payload |
| 11:47AM |
1 |
Voice Mail |
| 9:11AM |
1 |
jitterbuffer |
| 8:15AM |
1 |
echo cancellation for sip phones |
| 2:40AM |
2 |
Any other "free" toll free SIP providers out there? |
| |
| Wednesday November 19 2008 |
| Time | Replies | Subject |
|---|
| 11:14PM |
1 |
dahdi_test drops after restarting Sangoma driver |
| 7:57PM |
3 |
puzzle |
| 7:20PM |
1 |
Upgrading Asterisk and FreePBX from 1.2 to 1.4 |
| 7:07PM |
3 |
TDM400 (?) zap hangup |
| 6:07PM |
2 |
VoiceMail - audio problem |
| 5:22PM |
1 |
Howto grab back call transfered from SIP phone |
| 1:39PM |
1 |
Asterisk NOW - Where to start |
| 1:13PM |
1 |
IF else |
| 11:10AM |
3 |
P2P |
| 10:33AM |
4 |
Role of asterisk |
| 8:12AM |
1 |
presence with polycom DND |
| 5:51AM |
1 |
Aeterisk NOW 1.5beta1 - CDR problem.... |
| 3:30AM |
4 |
question about connecting with Mobile Base Station |
| 12:56AM |
5 |
help with dahdi |
| 12:00AM |
1 |
Asterisk 1.6 call files Disposition=NO ANSWER |
| |
| Tuesday November 18 2008 |
| Time | Replies | Subject |
|---|
| 10:46PM |
2 |
Fwd: Polycom phone time behind one hour. |
| 8:42PM |
0 |
Realtime MOH |
| 6:19PM |
1 |
setting up callback |
| 6:00PM |
2 |
Do Digium Digital Cards Handle Remote Loopback Command? |
| 5:33PM |
1 |
diax debian package |
| 5:30PM |
1 |
sound quality between two back-to-back asterisk |
| 3:55PM |
1 |
Incoming Transfer |
| 3:45PM |
1 |
Configuring Sangoma BRI with zaptel? |
| 2:22PM |
0 |
Crash when rebooting or unload xorcom modules |
| 1:54PM |
1 |
Asterisk 1.4.21.2 and gtalk2voip |
| 1:08PM |
1 |
meetme command from 1.4 to 1.6 |
| 12:56PM |
1 |
Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working |
| 12:29PM |
2 |
Asterisk with or without OpenSER |
| 11:14AM |
2 |
FOP with Asterisk 1.6. No call Information. |
| 10:30AM |
4 |
busy-level / busy-limit Asterisk 1.4.22 |
| 9:05AM |
0 |
Asterisk Realtime and device contexts |
| 5:28AM |
1 |
How to Barge specific extensions |
| 2:11AM |
2 |
HPEC performance |
| |
| Monday November 17 2008 |
| Time | Replies | Subject |
|---|
| 11:55PM |
0 |
Message 12216 |
| 11:54PM |
2 |
two sip listening ports for single asterisk |
| 11:04PM |
1 |
Picked up calls die in exactly 20 seconds |
| 8:03PM |
1 |
MixMonitor Problem |
| 5:31PM |
1 |
Deny FOP originated calls |
| 5:16PM |
1 |
AMI Events disabling. |
| 4:59PM |
0 |
dahdi and asterisk 1.4.22 |
| 4:55PM |
4 |
Digium Card Noice issue |
| 4:38PM |
1 |
Hints and realtime |
| 3:04PM |
2 |
Full Duplex |
| 2:43PM |
1 |
asterisk conference |
| 2:32PM |
1 |
ALL of DIDx Down? |
| 1:08PM |
0 |
E1/channels |
| 8:26AM |
4 |
Debugging Asterisk |
| 6:37AM |
0 |
IAX2 client for 'eee pc 1000' |
| 2:41AM |
2 |
upgrade to 1.6 |
| |
| Sunday November 16 2008 |
| Time | Replies | Subject |
|---|
| 11:33PM |
0 |
Record Application |
| 9:49PM |
2 |
dahdi compile error on svn |
| 9:04PM |
0 |
DialPlan |
| 7:04PM |
1 |
Caching Asterisk SIP useragent info? |
| 5:07PM |
1 |
iPhone SIP or IAX client (without proxy)? |
| 4:47PM |
0 |
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)? |
| 11:45AM |
0 |
Info about dstchannel |
| 9:28AM |
6 |
* + Legacy PBX works but strange problem |
| |
| Saturday November 15 2008 |
| Time | Replies | Subject |
|---|
| 10:55PM |
0 |
RV: MixMonitor and Queues |
| 10:49PM |
3 |
IAX2 client for "eee pc 1000" |
| 10:18PM |
1 |
Asterisk GUI and SIP registration |
| 7:10PM |
0 |
MixMonitor and Queues |
| 6:43AM |
2 |
Polycom low volume |
| 4:25AM |
1 |
PBX -> PRI -> * -> Telco not working |
| 2:17AM |
2 |
Best way to handle include files? |
| |
| Friday November 14 2008 |
| Time | Replies | Subject |
|---|
| 9:56PM |
1 |
installation |
| 9:04PM |
0 |
Originate on AMI |
| 8:21PM |
0 |
Linksys SPA 400, 901 and 921 with asterisk |
| 6:50PM |
0 |
Manilla inbound DID |
| 6:28PM |
0 |
PRI users, please read |
| 4:29PM |
1 |
no dial to busy sip line |
| 2:58PM |
1 |
Queue App - Set monitoring dynamically |
| 2:02PM |
4 |
Looking for a good lightweight Linux softPhone |
| 1:23PM |
1 |
kick from conference message on 1.2.23 |
| 12:59PM |
3 |
asterisk/E1 |
| 10:24AM |
1 |
RTP LOG |
| 8:46AM |
1 |
Dedicated Servers |
| 7:52AM |
0 |
openLDAP |
| 7:07AM |
0 |
Virtual Question |
| 5:30AM |
1 |
ParkandAnnounce? |
| 3:32AM |
2 |
Preserving DID numbers on PRI pass through |
| |
| Thursday November 13 2008 |
| Time | Replies | Subject |
|---|
| 10:05PM |
1 |
Asterisk and Zaptel version numbers -- how close is close enough? |
| 2:16PM |
2 |
How long will Asterisk 1.4.x supported/maintained |
| 2:16PM |
5 |
database queries from extensions.conf |
| 1:36PM |
0 |
[Fwd: [OpenSIPS-Devel] RFC: new opensips design] |
| 1:04PM |
0 |
Tos_sip |
| 11:26AM |
0 |
Problems with Licensed g729a codec from Digium |
| 9:25AM |
0 |
cisco voice gw / cisco call manager /asterisk for voice record, ivr |
| 7:57AM |
1 |
Parking help - causing Asterisk crash |
| 6:18AM |
2 |
asterisk setup w/ voIP phones |
| 1:13AM |
2 |
CANCEL FORWAR |
| 12:03AM |
0 |
1.4.22 CALLERID(num) |
| |
| Wednesday November 12 2008 |
| Time | Replies | Subject |
|---|
| 11:03PM |
1 |
List eating mail again? |
| 9:29PM |
6 |
Why Nat=yes Nat=no Option? |
| 8:39PM |
3 |
SIP provider and NAT |
| 6:17PM |
1 |
How to get correct dial result for outgoing calls thru ISDN? |
| 5:25PM |
4 |
E1 PRI to and from SIP screeching |
| 5:04PM |
1 |
What are the minimum realtime fields for sipusers? |
| 4:44PM |
1 |
QueueLog from AMI |
| 1:03PM |
1 |
Query about Call Recording with Asterisk / Freeswitch in Cisco IPCC deployment |
| 9:55AM |
4 |
test OpenVox B400P and junghans card for dahdi BRI wcb4xxp |
| 9:30AM |
4 |
The sound is played but I did not hear |
| 2:26AM |
1 |
Use DECT GAP handsets with Snom M3 base? |
| 2:02AM |
2 |
AS5200 <-> T100P - No alarms but no calls either... |
| 12:27AM |
3 |
Grandstream and pickup |
| |
| Tuesday November 11 2008 |
| Time | Replies | Subject |
|---|
| 11:17PM |
1 |
AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server |
| 10:54PM |
1 |
Request for testing of new driver for B410P Quad-Port BRI |
| 10:24PM |
2 |
play file from url |
| 9:19PM |
3 |
Use the NEW ulaw/alaw codecs (slower, but cleaner) |
| 7:05PM |
1 |
ztdummy: rtc: lost some interrupts at 1024Hz |
| 5:34PM |
1 |
ztdummy: rtc: lost some interrupts at 1024Hz. |
| 4:21PM |
0 |
Asterisk CDR Error ?? |
| 3:41PM |
3 |
OT: Polycom Firmware available (by accident?) |
| 3:10PM |
1 |
view the current calls and their codec |
| 2:24PM |
0 |
help with call with no sound via PSTN |
| 11:59AM |
1 |
Dial outside number using the E1 Link |
| 8:49AM |
2 |
TE410P alarms stay RED with 1.4.22 |
| 6:35AM |
7 |
music on hold |
| 2:03AM |
0 |
dial a number while play the sound |
| 1:05AM |
2 |
Server for 25-30 phones, sip trunks over the net |
| 12:34AM |
1 |
What makes TDM400 FXS Connection to TELCO go into Off Hook State? |
| |
| Monday November 10 2008 |
| Time | Replies | Subject |
|---|
| 7:55PM |
1 |
Voicemail IMAP ./configure error |
| 5:26PM |
3 |
console/dsp asterisk seg fault |
| 5:10PM |
3 |
Asterisk daemon dies about once per day |
| 2:50PM |
0 |
SRTP support in asterisk 1.6 |
| 2:11PM |
1 |
Using AMI to determine PRI Channels Used |
| 1:13PM |
3 |
directrtpsetup without reinvite |
| 12:11PM |
0 |
analog issues using xen virtualization |
| 11:56AM |
2 |
GEN-GEN and Manual Ring-Down (MRD)? |
| 11:16AM |
6 |
changing the size of voice packets |
| |
| Sunday November 9 2008 |
| Time | Replies | Subject |
|---|
| 6:22PM |
2 |
Codec problems when using G.723 |
| 3:07AM |
3 |
set(CALLERID(name) not working |
| |
| Saturday November 8 2008 |
| Time | Replies | Subject |
|---|
| 11:18PM |
3 |
Rolled Distro? |
| 7:13AM |
0 |
ISDN for Dahdi get hangs |
| |
| Friday November 7 2008 |
| Time | Replies | Subject |
|---|
| 7:35PM |
1 |
Outgoing SIP calls dropped after 30 seconds. |
| 5:25PM |
0 |
REFER problems with Asterisk and OpenSER |
| 5:12PM |
1 |
Providing Ringback |
| 4:29PM |
1 |
DNS A queries for channel |
| 3:17PM |
2 |
help with dialplan |
| 2:35PM |
3 |
TE121B Doesn't Fit PCI-E Slot |
| 2:23PM |
4 |
1.6 Production ready?? |
| 2:14PM |
0 |
T.38 without port changes |
| 2:04PM |
1 |
Help with asterisk and avaya SIP trunking |
| 1:58PM |
1 |
is it possible to deactivate RTCP? |
| 1:26PM |
0 |
AEL NoOp not working [SOLVED] |
| 9:22AM |
0 |
asterisk - avaya ip office SIP trunking |
| 7:46AM |
0 |
[OT] Reporting Spam |
| 7:01AM |
0 |
Use of optional new number in ISDN release 22 |
| |
| Thursday November 6 2008 |
| Time | Replies | Subject |
|---|
| 6:27PM |
4 |
ODBCExec and Asterisk 1.6 New Thread |
| 6:07PM |
0 |
[OT] Capitalism (was: Spam from DIDForSale <contact-sales@didforsale.com>) |
| 4:27PM |
0 |
Asking again about busylevel |
| 4:12PM |
2 |
Variable Scope Question |
| 3:42PM |
0 |
Asterisk trunking |
| 12:18PM |
2 |
tired of "midget packet received" warnings |
| 10:57AM |
3 |
RFC: multiple packages editing asterisk config files |
| 10:46AM |
2 |
crashes after upgrade from 1.2.16 to 1.4.21.2 |
| 9:31AM |
2 |
Spam from DIDForSale <contact-sales@didforsale.com> |
| 9:30AM |
0 |
HD Voice conference Friday Nov 7th @ 12 Noon EST |
| 8:56AM |
0 |
asterisk.conf ====> maxload |
| 4:33AM |
1 |
ISDN Cause Code 100, Bosch Integral Management Connection |
| 3:56AM |
4 |
Recommend Wireless IP Phone |
| 3:50AM |
0 |
Missing callerid |
| 1:40AM |
0 |
Agent Question |
| 1:17AM |
1 |
Polycom's lose BLF after Asterisk restart |
| 1:06AM |
1 |
Asterisk Realtime Configuration |
| 12:01AM |
2 |
TDM400 with FXS some handsets not ringing |
| |
| Wednesday November 5 2008 |
| Time | Replies | Subject |
|---|
| 11:26PM |
1 |
Cisco 776M Any good for connection to local Asterisk server? |
| 10:44PM |
1 |
Type 102 Millwatt Test Line |
| 9:20PM |
0 |
b410p mIDSN - RNIS signaling problems |
| 8:53PM |
0 |
a callerid question |
| 7:30PM |
1 |
Inbound/Outbound undesired behavior |
| 5:39PM |
2 |
ExtenSpy? am I doing it correctly? |
| 5:04PM |
1 |
SER/Asterisk interworking mailing list. |
| 4:49PM |
0 |
bugs.digium.com Maintenance and Downtime |
| 3:42PM |
0 |
br.Doctor Ester |
| 3:26PM |
5 |
Phishing attempt |
| 2:40PM |
1 |
800 origination |
| 2:33PM |
2 |
Dundi Issues |
| 2:20PM |
1 |
sas |
| 2:18PM |
0 |
SIP Qualify is not working with Postgres |
| 10:39AM |
1 |
AEL NoOp not working |
| 8:01AM |
1 |
How is it best to initialize specific SIP peer settings |
| |
| Tuesday November 4 2008 |
| Time | Replies | Subject |
|---|
| 11:50PM |
2 |
Sendmail using SMTP authorization |
| 11:26PM |
0 |
WARNING message when calls get into a queue with realtime members (Local channel) |
| 9:31PM |
1 |
shared voicemail box |
| 8:17PM |
1 |
dahdi trunk does not compile with kernel 2.6.27 |
| 6:14PM |
1 |
Some progress, anyway... |
| 5:12PM |
0 |
What is the best way to resale termination/origination? |
| 4:30PM |
1 |
users.conf and hints |
| 3:39PM |
0 |
Is SIPPEER curcalls working for you ? [SOLVED] |
| 3:34PM |
5 |
VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing |
| 2:17PM |
2 |
SPA-962 Time on Asterisk |
| 2:08PM |
3 |
SPA-962 & Asterisk |
| 11:33AM |
1 |
Is SIPPEER curcalls working for you ? |
| 8:10AM |
0 |
manager event privilege: call, all? what is? |
| |
| Monday November 3 2008 |
| Time | Replies | Subject |
|---|
| 8:41PM |
0 |
asterisk src=dst |
| 6:09PM |
0 |
CLI dial and echo recorder |
| 5:38PM |
0 |
busylevel question |
| 5:11PM |
2 |
Looking for a web video phone? |
| 4:14PM |
1 |
Call quality issue across VPN-> POTS vs SIP |
| 3:24PM |
0 |
loading misdn.conf strange error regarding out of range |
| 1:56PM |
1 |
help with debugging phone call |
| 12:30PM |
1 |
say load new |
| 4:03AM |
1 |
asterisk and bigmem kernel |
| 12:16AM |
1 |
Polycom 430 no hangup after SIP BYE, Status 481 instead |
| |
| Sunday November 2 2008 |
| Time | Replies | Subject |
|---|
| 4:53PM |
0 |
Returned mail: see transcript for details |
| 1:11PM |
1 |
Question regarding keywords in sip.conf/users.conf |
| 12:11PM |
0 |
Asterisk and Media gateway controller |
| 1:03AM |
5 |
Ztdummy and Asterisk |
| |
| Saturday November 1 2008 |
| Time | Replies | Subject |
|---|
| 9:58PM |
0 |
Gizmo ok but no audio when incoming |
| 7:17PM |
0 |
asterisk 1.2 and Dial with LIMIT_WARNING_FILE |
| 6:08PM |
0 |
asterisk-users Digest, Vol 52, Issue 1 |
| 5:01PM |
0 |
2 Or (971) Numbers for sale |
| 3:15PM |
1 |
SPA3102 interdigit timers bug? |
| 6:25AM |
1 |
Wierd queue question |
| 4:36AM |
1 |
VoIP traffic shaping |