| Friday October 31 2008 |
| Time | Replies | Subject |
|---|
| 10:25PM |
1 |
Asterisk installation |
| 9:08PM |
3 |
Call problems |
| 6:13PM |
1 |
CDR Posting Delay |
| 3:55PM |
0 |
VoIP phones supporting speex |
| 3:49PM |
0 |
Asterisk Legacy pBx help |
| 3:30PM |
0 |
No audio after transferring to voicemail |
| 3:09PM |
1 |
Monitor group calls (recording calls) |
| 1:10PM |
1 |
Copy protection issues with G.729 codec in Solaris |
| 11:32AM |
3 |
Asterisk/Machine Hang after calling in/out ISDN |
| 11:20AM |
4 |
Call terminates after 20 minutes |
| 10:11AM |
2 |
giving a user asterisk CLI access: how bad could it get |
| 9:54AM |
2 |
Friday Halloween Edition Oct 31 12 Noon EDT |
| 9:34AM |
0 |
MusicOnHold from a Sound card |
| 9:10AM |
0 |
dtmf have not callerid |
| 7:18AM |
5 |
twice normal beep before busy tone ?? |
| 5:02AM |
1 |
Enter Value and continue dialplan |
| 3:35AM |
1 |
Asterisk with SC440 Dell(Big Problem) |
| |
| Thursday October 30 2008 |
| Time | Replies | Subject |
|---|
| 11:15PM |
1 |
1.4.22 vs 1.4.21.2 - IAX2 regression ? |
| 9:27PM |
0 |
Message 245058 |
| 7:12PM |
0 |
General development funding: discussion and survey |
| 6:44PM |
1 |
ISDN - BRI |
| 6:41PM |
0 |
Music On Hold (from a Sound card) Help |
| 5:22PM |
1 |
Asterisk Legacy PBX |
| 5:22PM |
0 |
Other lists |
| 4:57PM |
2 |
Old mantis e-mails |
| 4:17PM |
3 |
SIP # DTMF |
| 4:03PM |
0 |
Asterisk SVN bug segfaulting |
| 3:59PM |
0 |
Connection two asterisk via SIP (call forward) |
| 3:31PM |
1 |
Asterisk settings |
| 1:30PM |
1 |
Sangoma Question |
| 11:51AM |
0 |
Trouble with SIP/NAT |
| 10:55AM |
1 |
Linux Kernel >=2.6.25 Realtime issues |
| 9:47AM |
0 |
mp3player and shoutcast |
| 9:17AM |
1 |
SIP REGISTER |
| 9:02AM |
2 |
up to 3000 lines capacity asterisk Deployment |
| 7:36AM |
0 |
list-testing2 |
| 7:33AM |
0 |
list-testing |
| 12:10AM |
1 |
'Asterisk is not thread safe' message |
| |
| Wednesday October 29 2008 |
| Time | Replies | Subject |
|---|
| 10:31PM |
4 |
Dimensioning a telephony system based on openser! |
| 9:10PM |
1 |
app_swift installation problems |
| 6:27PM |
0 |
CDP (was Re: network design philosophy and practice) |
| 6:21PM |
0 |
FW: Thecus N7700 |
| 6:17PM |
1 |
Is anyone using * for 2 way video conferencing? |
| 5:55PM |
4 |
Current Open Source Billing Package |
| 5:13PM |
3 |
Blank Voicemail.Conf after Password Change |
| 5:03PM |
0 |
[OT] Flash player for call recordings - 8khz |
| 4:08PM |
1 |
Intergrating vicidial with trixbox |
| 3:59PM |
0 |
Headset Recommendation |
| 3:21PM |
0 |
What syntax to send user:pass in SIP Dial string? |
| 2:19PM |
5 |
network design philosophy and practice |
| 1:55PM |
1 |
codec not in channel variables |
| 1:50PM |
1 |
Complete OS/Asterisk disk |
| 1:05PM |
1 |
SIP ACCOUNT CODE not included in CDR when SIP Status is "Unknown" |
| 11:10AM |
2 |
Dial() - any way to limit waiting for a "RINGING" state? |
| 4:10AM |
1 |
any dialplan action on received jabber msgs? |
| 2:55AM |
1 |
Snom - we are puzzled |
| 12:13AM |
7 |
Decent Voip Phones for enterprise |
| |
| Tuesday October 28 2008 |
| Time | Replies | Subject |
|---|
| 11:39PM |
5 |
Sendmail for Voicemail |
| 10:45PM |
1 |
XML Cisco config file |
| 9:54PM |
1 |
Multiline Analog Setup |
| 7:19PM |
2 |
Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet |
| 4:53PM |
1 |
Dealing with progress codes |
| 4:30PM |
3 |
Anyone using an Intel Atom ? |
| 3:33PM |
1 |
AMI - Status Event. |
| 11:41AM |
0 |
Re Fring: Open VPN client to be installed on the mobile, which mobile? |
| 7:51AM |
0 |
OT - Choose PCI or mini-PCI for appliance ? |
| 7:11AM |
1 |
Best Sales 2008! |
| |
| Monday October 27 2008 |
| Time | Replies | Subject |
|---|
| 11:41PM |
1 |
gtalk/jingle full report |
| 11:03PM |
2 |
whisper time remaining |
| 10:35PM |
0 |
asterisk on freebsd |
| 10:12PM |
0 |
Sending a text-message to JABBER via CLI |
| 4:16PM |
0 |
make config update-rc.d on Debian |
| 4:02PM |
0 |
change codec mid-call |
| 3:34PM |
1 |
Forcing repacketization on SIP to SIP call |
| 1:42PM |
1 |
How to bind a SIP channel to an IP |
| 12:36PM |
3 |
Door phone |
| 12:34PM |
2 |
Secure WAN connection too * extention |
| 12:31PM |
1 |
KEY SYSTEM, Intercom |
| 10:48AM |
11 |
Fring: Open VPN client to be installed on the mobile, which mobile? |
| 6:35AM |
1 |
Asterisk 1.6 pbx_lua not creating any contexts |
| 6:23AM |
2 |
openser+asterisk |
| 5:19AM |
1 |
autodialed call forwarding via meetme or queue (was predictive dialer) |
| 2:59AM |
1 |
Asterisk 1.6 CDR no Clid information |
| 1:53AM |
2 |
Asterisk and voice recognition |
| 12:45AM |
1 |
CDR Records are not working |
| |
| Sunday October 26 2008 |
| Time | Replies | Subject |
|---|
| 8:15PM |
0 |
Asterisk on Freebsd 7.0 Release. |
| 6:33PM |
1 |
No incoming audio on Dahdi channels (TDM410P) |
| 12:25PM |
1 |
jingle/gtalk still very troubling |
| 4:24AM |
3 |
hammering imap vmail storage |
| 2:30AM |
0 |
Queue Warning messages |
| 12:37AM |
1 |
Strange ring tone: Long-Short-Short |
| |
| Saturday October 25 2008 |
| Time | Replies | Subject |
|---|
| 7:25PM |
1 |
gtalk dialstring? |
| 6:02PM |
0 |
someone to test gtalk with me? |
| 5:54PM |
9 |
Cheapest 4 port FXO |
| 1:56PM |
1 |
The skype channel... |
| |
| Friday October 24 2008 |
| Time | Replies | Subject |
|---|
| 10:18PM |
0 |
Asterisk iaxy adapter annoying beep during conversation |
| 9:55PM |
2 |
Sporadic One Way Audio |
| 9:48PM |
2 |
Fresh installed box |
| 8:12PM |
5 |
OT: Disable Polycom 650 Forward Softkey |
| 6:33PM |
1 |
Agents log in afterhours |
| 4:00PM |
4 |
Advice on ISDN and Asterisk in the UK |
| 2:31PM |
0 |
Freepbx or Trixbox Presentation |
| 12:38PM |
1 |
Problems with zaptel/ztdummy/asterisk. |
| 2:00AM |
2 |
Asterisk and Cisco Call Manager Express (CME) |
| 1:31AM |
1 |
Emerging dilema? DID forwarding meets SMS |
| |
| Thursday October 23 2008 |
| Time | Replies | Subject |
|---|
| 9:20PM |
2 |
Inbound DID + voice ports needed for vote monitoring project |
| 8:36PM |
1 |
Returning to Voicemail after returning call |
| 8:10PM |
2 |
is there a reference guide to "pri debug span" messages? |
| 4:38PM |
1 |
Channels are increasing without limit - Please Help! |
| 4:36PM |
1 |
Devstate and Voicemail |
| 4:35PM |
2 |
Hylafax asterisk iaxmodem problem |
| 3:17PM |
1 |
Atxfer Command |
| 2:28PM |
0 |
users.conf and sip call-limit |
| 2:22PM |
1 |
switching from 1.6.0-beta9 to 1.6.0.1 problems |
| 12:30PM |
2 |
problems with some incoming/outgoing calls |
| 9:21AM |
0 |
command - set sip_codec- does not work with asterisk-1.4.21 |
| 2:04AM |
0 |
FW: [wwwac] Thursday 23 October 2008 NYLUG: Paul Charles Leddy on Asterisk, the Free Software Telephone System |
| |
| Wednesday October 22 2008 |
| Time | Replies | Subject |
|---|
| 7:29PM |
1 |
: Parking Issue |
| 7:28PM |
1 |
changing from default codec |
| 6:35PM |
7 |
Sonicwall potentially causing long ping times to SIP phones |
| 5:27PM |
3 |
asterisk video |
| 5:25PM |
6 |
fax / t38 gateway |
| 3:59PM |
0 |
Parking Issue |
| 12:38PM |
1 |
2 asterisk boxes |
| 9:44AM |
3 |
WebCall application |
| 7:24AM |
3 |
sip and nat |
| 5:09AM |
0 |
a question about linux/asterisk/commands |
| 4:52AM |
0 |
need asterisk tech to relocate to riyadh |
| 1:11AM |
0 |
Causes of auto-congestion on SIP? |
| |
| Tuesday October 21 2008 |
| Time | Replies | Subject |
|---|
| 10:03PM |
1 |
Does Asterisk support SIP Join Headers |
| 6:56PM |
1 |
hex b1 in CallerID sent by Asterisk On PRI |
| 5:05PM |
1 |
For Dial(), when calling party hangs up, redirect called party to another location in the dialplan? |
| 4:43PM |
3 |
Asterisk Console color |
| 3:50PM |
1 |
Generating 484 "Address Incomplete" |
| 2:39PM |
0 |
Problem with Portech |
| 10:46AM |
2 |
[help] Realtime Swich any context dinamically |
| 10:30AM |
0 |
Realtime : switch any context dynamically |
| 9:34AM |
0 |
Asterisk 1.4: ISDN congestion warnings |
| 1:42AM |
3 |
come back ring |
| 12:40AM |
1 |
prepaid approach |
| |
| Monday October 20 2008 |
| Time | Replies | Subject |
|---|
| 11:50PM |
0 |
TDM410P with EC doesn't detect DTMF after being on for ~1 hour |
| 8:36PM |
0 |
SERVICE CODES |
| 7:01PM |
7 |
How Secure Is Asterisk |
| 6:10PM |
1 |
a little regex help needed |
| 5:58PM |
1 |
OPENR2 in Thailand |
| 5:19PM |
0 |
Transferring Outbound Calls |
| 4:17PM |
0 |
I have probleme with asterisk |
| 1:45PM |
0 |
Problem in extensions.conf Configuration ${CALLINGPRES} |
| 1:13PM |
2 |
ISDN PRI Caller ID problem |
| 11:11AM |
0 |
B410P and asterisk 1.6 |
| 6:42AM |
3 |
asterisk setup |
| 6:39AM |
2 |
QoS VoIP |
| 3:28AM |
1 |
Zaptel FXO offhook when connected to PSTN |
| 12:46AM |
0 |
Panama DID |
| |
| Sunday October 19 2008 |
| Time | Replies | Subject |
|---|
| 8:50PM |
0 |
Got SIP response 603 "Declined" back from 81.15.xx.xx |
| 7:50PM |
4 |
Asterisk Problem |
| 7:30PM |
6 |
adding a second extension |
| 4:31AM |
2 |
Latency woes, qos the fix? |
| 2:58AM |
0 |
app_confcall on Asterisk 1.6 update |
| 2:31AM |
4 |
IP Address on CDR |
| 1:21AM |
0 |
Does asterisk 1.6 support an authname with a domain? |
| 12:13AM |
1 |
Is there a way to specify the fromdomain from the dialplan? |
| |
| Saturday October 18 2008 |
| Time | Replies | Subject |
|---|
| 7:16PM |
3 |
OT: Polycom IP330 user problem |
| 7:08PM |
1 |
strange h323 delay issue |
| 6:25PM |
0 |
Looking to replicate OnSIP ........SER + Asterisk |
| 7:30AM |
2 |
Asterisk 1.4 and openLDAP |
| 1:36AM |
2 |
SER + Asterisk |
| |
| Friday October 17 2008 |
| Time | Replies | Subject |
|---|
| 10:14PM |
1 |
Phones lose contact |
| 8:47PM |
1 |
anoyingly answers already in use pstn line |
| 8:28PM |
1 |
Transfering Calls back on the same PRI |
| 4:15PM |
0 |
FW: Under the Radar Blog |
| 4:10PM |
2 |
Snom M3 firmware Update |
| 3:10PM |
1 |
Asterisk SIP and SRTP |
| 2:52PM |
1 |
Strip prefix |
| 2:37PM |
0 |
Whenever Asterisk restarts, what should happen to ongoing subscriptions ? |
| 2:23PM |
0 |
GET DATA Returning only a single digit |
| 2:19PM |
4 |
srv records not being honoured properly |
| 11:48AM |
1 |
How to launch batch whenever Asterisk (re)start ? [SOLVED] |
| 8:17AM |
1 |
asterisk +heartbeat (Wilton Helm) |
| 6:12AM |
5 |
How to add contexts in asterisk realtime? |
| 3:00AM |
1 |
Meetme "talker optimization" always on even when no "o" option present. |
| |
| Thursday October 16 2008 |
| Time | Replies | Subject |
|---|
| 6:38PM |
2 |
DAHDI and wait 'w' |
| 6:38PM |
1 |
International calls/pridialplan from a legacy PBX. |
| 5:09PM |
1 |
asterisk +heartbeat |
| 4:39PM |
0 |
Multiple Repeated tones with TDM02B |
| 3:49PM |
1 |
RELEASE message in q931.c |
| 3:48PM |
2 |
SIP: difference between Grandstream and Cisco when behind NAT |
| 3:40PM |
2 |
Triggering a call from bash |
| 3:23PM |
0 |
asterisk-users Digest, Vol 51, Issue 51 |
| 1:57PM |
1 |
app_confcall build issues |
| 1:03PM |
0 |
Difference between followme and substitution ? |
| 12:31PM |
0 |
Sharing my Asterisk + SPA3102/PAP2 setup: What I've learned in 1 week. |
| 12:23PM |
0 |
asterisk cmd mysql and stored procedures |
| 11:16AM |
2 |
prective dialer |
| 9:49AM |
0 |
app transfer problem |
| 6:57AM |
3 |
How to launch batch whenever Asterisk (re)start ? |
| 3:22AM |
0 |
How to invoke an external C program and output an integer to the program? |
| 3:19AM |
0 |
Telrad Analog CID |
| 12:16AM |
0 |
Queue problem |
| |
| Wednesday October 15 2008 |
| Time | Replies | Subject |
|---|
| 11:57PM |
1 |
Cisco 7960 not always receiving incoming calls |
| 11:19PM |
1 |
Configuring SIP TLS |
| 8:59PM |
1 |
voicemail.conf |
| 7:52PM |
0 |
phoniceq e400p driver for DAHDI |
| 7:38PM |
2 |
Zaptel compile error after make update. |
| 3:26PM |
1 |
mismatched callerid on phone and CDR ? |
| 1:28PM |
0 |
LDAP authentication |
| 10:57AM |
2 |
Asterisk 1.4.21.2 - Issues with call parking |
| 6:49AM |
0 |
FW: asterisk-users Digest, Vol 51, Issue 48 |
| |
| Tuesday October 14 2008 |
| Time | Replies | Subject |
|---|
| 10:57PM |
1 |
Help With AMI |
| 10:24PM |
1 |
SIP channels seem not to close after call is finished |
| 8:26PM |
3 |
Looking for a mentor |
| 5:35PM |
1 |
Speex Problem |
| 4:46PM |
1 |
asterisk+heartbeat |
| 4:09PM |
0 |
Asterisk 1.4.10.1 : PRI congestion warnings |
| 3:05PM |
4 |
Call files |
| 9:42AM |
1 |
Asterisk voicemail |
| 1:15AM |
7 |
Panasonic x Asterisk if I can emulate Panasonic fast! |
| |
| Monday October 13 2008 |
| Time | Replies | Subject |
|---|
| 10:06PM |
0 |
Asterisk help please |
| 9:27PM |
0 |
voicemail issues with 1.6.0 |
| 9:22PM |
1 |
Tracking T1/PRI channel status - inbound vs. outbound |
| 8:44PM |
6 |
ISDN |
| 6:21PM |
1 |
ifbyphone/google analytics |
| 5:13PM |
3 |
Softphone Framework or Libraries |
| 5:10PM |
0 |
AGI Hangup |
| 3:29PM |
1 |
Need help for debuging |
| 3:19PM |
1 |
IP 650 Sidecar |
| 2:50PM |
0 |
MOH Bad |
| 2:13PM |
0 |
Unknown call every 30 minutes on the dot. |
| 11:18AM |
0 |
Installing CdrTool on free bsd |
| 10:28AM |
2 |
echo over digital line |
| 9:41AM |
0 |
ERROR:Failed to create H323 listener |
| |
| Sunday October 12 2008 |
| Time | Replies | Subject |
|---|
| 9:17PM |
3 |
setup for fax machine |
| 6:56PM |
0 |
Caller ID sip trunk |
| 3:53PM |
5 |
One Way Audio Problem |
| 3:17AM |
3 |
cli commands missing |
| |
| Saturday October 11 2008 |
| Time | Replies | Subject |
|---|
| 6:41PM |
1 |
Asterisk 1.6.1 + openais |
| 6:02PM |
1 |
1 second delay when connecting calls |
| 4:20PM |
1 |
Is there a way to test SIP credentials without making a call? |
| 4:13PM |
4 |
Asterisk For Windows ? |
| 2:25AM |
0 |
Mitel 5220 firmware |
| |
| Friday October 10 2008 |
| Time | Replies | Subject |
|---|
| 9:49PM |
0 |
Caller ID service and the ethernet stucking |
| 9:17PM |
2 |
Configuring Bandwidth.com SIP trunks to prevent one-way audio |
| 8:43PM |
2 |
is there a way |
| 8:24PM |
4 |
Budge Tones pick up wrong calls |
| 4:57PM |
3 |
Got event 17 (Polarity Reversal)... |
| 4:52PM |
2 |
Block Caller ID |
| 4:04PM |
3 |
Question about echo cancelation |
| 2:09PM |
2 |
Asterisk SIP calls stop working having more than 300 calls (more than 600 channels) |
| 12:33PM |
1 |
Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 10:52AM |
9 |
How to enable inbound CLI for X-Lite/Asterisk phone. |
| 9:09AM |
1 |
Be aware of callcheap.com and Mike Low - It is scam |
| 7:50AM |
3 |
Compile logger-mysql.c with UNDEFINED REF to `mysql_error' |
| 7:44AM |
0 |
How to barge Inbound calls |
| 3:18AM |
1 |
1.6.0.1 ?? |
| 2:20AM |
1 |
Asterisk CDR Analyser |
| 1:37AM |
4 |
Polycom 330 not dialing 4 digit extensions beginning with 11xx |
| |
| Thursday October 9 2008 |
| Time | Replies | Subject |
|---|
| 11:31PM |
3 |
DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :(((( |
| 10:00PM |
1 |
Cisco 7960 sccp, Skinny and 1.4 |
| 9:59PM |
1 |
Transfer/Park Question. |
| 9:09PM |
1 |
SIP problems? |
| 9:09PM |
2 |
Asterisk 1.6.0 CDR billsec and duration not working from h extension |
| 5:38PM |
1 |
asterisk-users Digest, Vol 51, Issue 26 |
| 5:22PM |
0 |
ATA hangs up with fax detection on... |
| 2:45PM |
1 |
Asterisk-Panasonic TDA 600 error |
| 1:25PM |
3 |
Inbound Calls From AS5300 Rejected with 488 Code |
| 11:06AM |
0 |
Asterisk restarts on call parking |
| 9:18AM |
2 |
Hang up detection with TDM400P and Telewest/Virgin Media line |
| 8:19AM |
1 |
Ringtones for the console |
| 7:52AM |
0 |
Interrupt Asterisk's SayDigits() |
| 7:34AM |
1 |
H323 |
| 7:10AM |
4 |
Howto analyze concurrent ISDN channel usage |
| 6:57AM |
0 |
Friday Oct 10 @12 Noon EDT VoIP Users Conference |
| 4:51AM |
1 |
Interpreting Asterisk Logs |
| 3:41AM |
4 |
Record name for conference... |
| 3:14AM |
2 |
Menu for call forwarding or voicemail |
| 3:14AM |
2 |
retransmitting NAT |
| 2:44AM |
1 |
conntrack_sip, iptables, and asterisk |
| |
| Wednesday October 8 2008 |
| Time | Replies | Subject |
|---|
| 10:21PM |
1 |
Update (IAX Trunking Help) |
| 7:44PM |
1 |
Sip Trunking |
| 7:07PM |
0 |
Help with IAX Trunking |
| 5:51PM |
1 |
Sample code fragement for subscribing to hints wanted. |
| 5:33PM |
1 |
make func_realtime work like app_realtime (1.6) |
| 4:18PM |
0 |
Mobile (cell) phone in a queue (but act like an agent |
| 4:03PM |
0 |
fastagi example |
| 1:58PM |
1 |
debugging hints in 1.6 |
| 11:44AM |
0 |
SRV DNS failover - dial to proxy list |
| 10:24AM |
0 |
How is automatic redial/callback when available implemented ? |
| 8:29AM |
1 |
Call-limit bug in 1.2 ? |
| 7:46AM |
0 |
Zaptel -> DAHDI for dummies? |
| 6:11AM |
0 |
Can't find the path to Phoneprov directory |
| 4:11AM |
1 |
registration limit |
| 12:30AM |
3 |
changing passwords |
| 12:26AM |
0 |
"requested special control 20" ?? |
| 12:04AM |
2 |
help no ring on caller side |
| |
| Tuesday October 7 2008 |
| Time | Replies | Subject |
|---|
| 8:36PM |
3 |
Question on screening calls / Question about the Dial g option |
| 7:20PM |
3 |
cisco phones getting SIP 401 unauthorized |
| 3:45PM |
1 |
Bad Destinations |
| 2:58PM |
0 |
Asterisk Callerid Help Needed |
| 2:08PM |
1 |
include in the DAHDI system.conf file and chan_dahdi.conf |
| 1:19PM |
1 |
Efax from Agi script |
| 10:48AM |
0 |
call center |
| 10:25AM |
2 |
Cisco 7906g & SIP |
| 8:42AM |
2 |
automatic call pickup |
| 6:22AM |
1 |
can't find mysqlclient : asterisk-addons-1.6.0 |
| 4:03AM |
1 |
regcontext |
| 2:24AM |
0 |
Asterisk/AJAM Console |
| |
| Monday October 6 2008 |
| Time | Replies | Subject |
|---|
| 8:30PM |
2 |
ldap usage in 1.6.0 |
| 7:58PM |
7 |
Matching *, + and # in the dialplan |
| 7:02PM |
2 |
Asterisk 1.4 or 1.6 ??? |
| 6:53PM |
2 |
Help with remote users |
| 6:00PM |
2 |
Missing 'Queue' Application in 1.4.21.2 |
| 5:33PM |
1 |
cdr,gsm file format |
| 5:19PM |
0 |
Nice recording interfaces |
| 3:23PM |
0 |
Semi OT: Global Crossing |
| 3:03PM |
8 |
PoE switch recommendations? |
| 2:55PM |
1 |
Hook Flash |
| 2:21PM |
1 |
AEL and swap from macros to contexts |
| 2:08PM |
2 |
Conneting Asterisk to Swyx pri |
| 2:07PM |
3 |
How to implement Ringing through a sound card for overhead paging |
| 9:24AM |
4 |
Tribox |
| 7:46AM |
1 |
R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP) |
| 7:26AM |
3 |
Alarm events + asterisk dies |
| 7:25AM |
0 |
Fwd: Fonolo: Visually Navigate & Dial IVR Phone Menus in Web/Mobile Browser |
| 4:39AM |
3 |
asteriskt38.com |
| 2:08AM |
1 |
Dial out DAHDI Channel? |
| 12:38AM |
1 |
MS Exchange IMAP Voicemail |
| 12:03AM |
0 |
no per mailbox imapfolder override? wow. |
| |
| Sunday October 5 2008 |
| Time | Replies | Subject |
|---|
| 8:30PM |
3 |
OT: headsets |
| 5:51PM |
5 |
asterisk, phpagi and singleton |
| 3:59AM |
1 |
Asterisk Load Balancing |
| |
| Saturday October 4 2008 |
| Time | Replies | Subject |
|---|
| 5:50PM |
0 |
voicemail quota |
| 5:30PM |
5 |
Vitelity Asterisk configuration help |
| 4:47PM |
1 |
IAX denial of service |
| 12:02PM |
1 |
Mimic SIP Events framework in Asterisk without coding ... |
| 11:18AM |
0 |
2 stage dialing and 484 address incomplete [SOLVED] |
| 9:11AM |
1 |
Aastra phones and dns srv records |
| |
| Friday October 3 2008 |
| Time | Replies | Subject |
|---|
| 11:53PM |
0 |
DTMF Talk Off |
| 10:43PM |
1 |
DTMF issues... |
| 10:07PM |
0 |
Asterisk 1.6.1-beta1 Now Available |
| 8:07PM |
1 |
network monitoring - triggering a phone call in asterisk |
| 7:06PM |
0 |
catch the use of h option in dial |
| 3:05PM |
2 |
Ok message |
| 3:03PM |
2 |
MWI with Siemens Gigaset S450IP |
| 9:21AM |
6 |
How to add Callee's name into Dial command ? |
| 8:08AM |
5 |
sip clients for smart phones? |
| 8:04AM |
1 |
Improving the voice Quality, |
| 7:59AM |
0 |
2 stage dialing and 484 address incomplete |
| 4:22AM |
1 |
t1 cards |
| 1:39AM |
1 |
uninstalling zaptel |
| 12:45AM |
1 |
dahdi service start |
| |
| Thursday October 2 2008 |
| Time | Replies | Subject |
|---|
| 8:42PM |
1 |
DTMF |
| 8:00PM |
1 |
VOIP Provider |
| 7:26PM |
2 |
Channels crossing... |
| 6:56PM |
1 |
Asterisk 1.4.22 and 1.6.0 Released |
| 6:09PM |
0 |
dahdi-linux 2.0.0 and dahdi-tools 2.0.0 released |
| 4:32PM |
1 |
Asterisk Queue question |
| 4:22PM |
1 |
OT - Is sip.instance useful ? |
| 2:43PM |
2 |
Is SIPPEER curcalls working for you ? (was: Ongoing calls with SIPPEER, curcalls) |
| 2:19PM |
4 |
Zaptel-1.4.1 error cross compile |
| 1:03PM |
1 |
DTMF issue |
| 12:28PM |
0 |
IP address on mysql cdr |
| 12:22PM |
1 |
Ultramonkey LVS + asterisk |
| 9:38AM |
1 |
How to find the CDR call start time value |
| 9:27AM |
2 |
DTMF Problem |
| 8:44AM |
0 |
B410p question |
| 4:58AM |
1 |
Asterisk custom functions |
| 1:21AM |
1 |
cisco VAD and Asterisk recordings |
| 1:19AM |
2 |
Aheeva With Asterisk |
| 12:57AM |
2 |
rebooting snoms in 1.6 |
| 12:42AM |
0 |
no audio, firewall problem? |
| |
| Wednesday October 1 2008 |
| Time | Replies | Subject |
|---|
| 9:24PM |
2 |
Asterisk - Failover System |
| 4:42PM |
0 |
ietf-sipping-config-framework |
| 3:28PM |
0 |
[Fwd: asterisk-users Digest, Vol 51, Issue 2] |
| 3:19PM |
0 |
SIP users limit |
| 2:57PM |
1 |
Ongoing calls with SIPPEER, curcalls |
| 2:30PM |
0 |
Sip Header Help |
| 1:38PM |
1 |
RTP sent before the INVITE ACK (for voicemail app) |
| 1:01PM |
0 |
Debugging |
| 11:53AM |
3 |
GSM / 3g channel bank |
| 6:51AM |
1 |
Software patents (was G723 on asterisk 1.4.1) |
| 3:20AM |
0 |
Polycom 3.1.0RevB |
| 3:19AM |
1 |
No reply to our critical packet |
| 3:03AM |
1 |
Cisco Dropping SIP support? |
| 2:05AM |
2 |
is DNS SRV enough for failover? |
| 1:35AM |
2 |
zap destroy |