| Saturday July 31 2010 |
| Time | Replies | Subject |
|---|
| 1:18PM |
0 |
MeetMe transcode / format problem |
| 10:04AM |
0 |
Disconnect supervision tone detection working for india |
| |
| Friday July 30 2010 |
| Time | Replies | Subject |
|---|
| 7:40PM |
0 |
Aastra ignore call button hangs up call instead of going to voicemail |
| 6:00PM |
1 |
Please test: STUN patch for Asterisk behind NAT |
| 1:17PM |
2 |
perform tasks outside a dial-plan (not during a call) |
| 11:17AM |
2 |
agi macro problem |
| 10:01AM |
2 |
How can i switch to samba server omitting sshfs |
| 8:06AM |
2 |
Asterisk and QoS |
| 7:24AM |
1 |
asterisk-users Digest, Vol 72, Issue 82 |
| 7:13AM |
0 |
asterisk-users Digest, Vol 72, Issue 81 |
| 2:35AM |
5 |
Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones? |
| 12:42AM |
1 |
VUC Friday: Twilio OpenVBX |
| 12:30AM |
1 |
1.8.0 beta2: courtesy tone being played to callee |
| |
| Thursday July 29 2010 |
| Time | Replies | Subject |
|---|
| 10:49PM |
1 |
Problem with Sangoma card... |
| 10:31PM |
1 |
COnfig File question |
| 8:35PM |
1 |
ignorant question about Digium cards and MeetMe |
| 6:44PM |
2 |
Disconnect supervision tone detection |
| 3:28PM |
4 |
How to extract channel-id of a user or peer |
| 1:03PM |
1 |
CDR and "custom name fields" |
| 11:48AM |
3 |
T.38 fax between ATA's and Asterisk and Cisco PGW 2200 |
| 11:23AM |
0 |
dahdi (in use) problem with queues |
| 11:22AM |
1 |
How can conect Cisco Unified Communications Manager with Asterisk |
| 8:15AM |
1 |
SEPMAC.xml for Ciscp 7970 IP Phone |
| 7:30AM |
2 |
How to record and playback at the same time |
| 6:19AM |
2 |
Registering 2 phone numbers to same router |
| 1:45AM |
2 |
spam blacklist |
| 1:06AM |
2 |
Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1 |
| |
| Wednesday July 28 2010 |
| Time | Replies | Subject |
|---|
| 10:22PM |
1 |
app_swift.c:338 engine: Failed to set voice |
| 9:33PM |
1 |
Why do Zaptel calls drop all of a sudden? Could busy detect be the problem? |
| 9:04PM |
4 |
Asterisk unresponsive |
| 7:06PM |
0 |
3G-324M Open Source |
| 4:30PM |
0 |
what is rinstance parameter in sip header |
| 4:23PM |
2 |
Nat issue one way audio on IP dial |
| 3:38PM |
1 |
Subscribe Problem - Zombie Channel |
| 1:38PM |
1 |
[OT] fail2ban and pf |
| 1:03PM |
0 |
AMI Monitor - one file |
| 12:04PM |
1 |
Passing Variables From Dial Macro To Parent Ruby |
| 11:49AM |
2 |
IAX authentication oddity - Known issue? Fixed? |
| 10:25AM |
2 |
Answered call not bridged |
| 9:29AM |
1 |
Redirecting a call to another extension using asterisk java |
| 1:12AM |
1 |
Random DTMF Tones Only on heard on ATA |
| 12:39AM |
2 |
Recording interface (pause/PLAY/RERECORD) |
| 12:38AM |
1 |
Grab voicemail WAV file when done |
| |
| Tuesday July 27 2010 |
| Time | Replies | Subject |
|---|
| 7:50PM |
1 |
Asterisk and Amazon Web Services |
| 7:41PM |
0 |
sip peer becomes unreachable in Asterisk 1.6 |
| 6:57PM |
1 |
Peculiar Polycom IP6000 behavior |
| 3:56PM |
0 |
Asterisk 1.8.0-beta2 Now Available |
| 2:21PM |
2 |
Urgent help = RUBY & AGI |
| 12:12PM |
2 |
How to transfer a call to operator using FAGI asterisk |
| 11:56AM |
1 |
IAX bandwidth optimisation |
| 11:40AM |
2 |
CallerID disappear from CDR on transfer |
| 7:47AM |
1 |
1 second Audio Lag |
| 5:38AM |
2 |
urgent:how to transfer a call using asterisk FAGI |
| 3:34AM |
2 |
Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there? |
| |
| Monday July 26 2010 |
| Time | Replies | Subject |
|---|
| 10:48PM |
1 |
VPMADT032 Failed! Unable to ping the DSP (2)! |
| 8:20PM |
2 |
MeetMe |
| 5:36PM |
2 |
Fail2ban - SuSEfirewall |
| 4:39PM |
2 |
Problem with Zap-Sip calls. |
| 2:45PM |
1 |
asterisk & distributed device state => res_jabber Versus res_ais |
| 12:34PM |
1 |
PBX Lua with Asterisk ODBC |
| 12:15PM |
4 |
Management interface |
| 12:11PM |
4 |
'dirty' upgrade of 1.4 |
| 10:41AM |
1 |
URgent - capturing 'answered' |
| 10:16AM |
1 |
Optimize peers registration under jitter/delay. |
| 9:28AM |
0 |
URGENT - who picked up the call?? |
| 8:05AM |
5 |
FreeTDS (Microsoft MsSQL 2008) and CDR |
| 8:03AM |
2 |
chan_skinny still maintained? |
| 6:54AM |
0 |
Adit 600 over MGCP. |
| 5:20AM |
2 |
No audio using xlite |
| 1:54AM |
1 |
SIP TOS Not being set |
| |
| Sunday July 25 2010 |
| Time | Replies | Subject |
|---|
| 10:57PM |
1 |
Vicibox vs VicidialNow |
| 9:11PM |
1 |
Using Vertical IP2007 phones with Asterisk? |
| 7:06PM |
1 |
1.6.2.10 sounds Makefile error? |
| 6:47PM |
1 |
Proprietary add-ons for Asterisk 1.8 |
| 2:52PM |
2 |
Cisco 7960 phone can't leave a queue |
| 2:23PM |
0 |
Audio Delay of 1-2 seconds, one way with Zoiper soft phones |
| 4:52AM |
2 |
undocumented change in expression handling in 1.8 beta |
| 12:11AM |
2 |
"Register Attacks" End of ENUM ? |
| |
| Saturday July 24 2010 |
| Time | Replies | Subject |
|---|
| 4:44PM |
2 |
Integration with Toshiba Strata DK424 |
| 4:07PM |
1 |
Exchange UM Play on Phone |
| 4:14AM |
4 |
getting some segmentation faults with 1.8 |
| |
| Friday July 23 2010 |
| Time | Replies | Subject |
|---|
| 9:58PM |
6 |
Asterisk 1.8.0-beta1 is Now Available! |
| 8:53PM |
1 |
voicemail |
| 7:48PM |
1 |
Attended Transfer question |
| 5:11PM |
0 |
Asterisk 1.6.2.10 Now Available |
| 5:10PM |
0 |
Asterisk 1.4.34 Now Available |
| 2:55PM |
2 |
Dahdi dial plan |
| 2:32PM |
3 |
Poor-man's paging through multiple phones? |
| 1:50PM |
2 |
application call to Gosub affects flow of control, and needs to be re-written using AEL |
| 1:35PM |
0 |
calls don't hang up correctly on VM |
| 11:26AM |
1 |
488 Not Acceptable Here |
| 11:23AM |
3 |
Vocera Comm Badges |
| 10:17AM |
0 |
(no subject) |
| 9:32AM |
2 |
Channels not coming up |
| 7:34AM |
1 |
ringback tone after MOH, before queue member bridged |
| 5:16AM |
1 |
Why does a bridged channel stay open for 4 hours? |
| |
| Thursday July 22 2010 |
| Time | Replies | Subject |
|---|
| 9:47PM |
5 |
Question regarding SMS(), SMSQ, SMSC |
| 9:10PM |
1 |
Does SIP limit to 3-way conference? |
| 7:58PM |
1 |
FW: hi friend |
| 7:15PM |
0 |
Receiving T1 Blue Alarm on asterisk 1.4.26, zaptel 1.4.12 |
| 7:14PM |
3 |
Soft phones. |
| 6:46PM |
3 |
POE Splitters |
| 5:11PM |
5 |
[AsteriskNow] Errors with clean install (on main screen when making calls) |
| 2:29PM |
0 |
SIP URI Dial has one way audio |
| 2:27PM |
0 |
Good SIP provider Western US |
| 2:13PM |
1 |
Good provider that offers allmost free calling within Europe? |
| 1:25PM |
0 |
RTP delay |
| 10:53AM |
2 |
dialog module count |
| 10:33AM |
3 |
My Switch is being attacked using sip scanner tool (Service Abuse Attack) |
| 6:25AM |
2 |
Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ? |
| 3:06AM |
3 |
asterisk app_fax, T.30, weird received faxes |
| |
| Wednesday July 21 2010 |
| Time | Replies | Subject |
|---|
| 11:05PM |
2 |
Cisco Firmware |
| 6:29PM |
1 |
Redial dtmf tones randomly...asterisk 1.4.21.2 |
| 12:32PM |
0 |
Musiconhold Problem |
| 11:51AM |
1 |
Cisco 7970 Not registering |
| 11:38AM |
1 |
Meetme Question |
| 10:09AM |
1 |
asterisk realtime SIP configuration |
| 7:14AM |
2 |
play alaw file with .wav extension |
| 6:32AM |
5 |
MOH distorted voice in Native and MP3 format |
| 5:36AM |
0 |
VoIP carrier g729 |
| |
| Tuesday July 20 2010 |
| Time | Replies | Subject |
|---|
| 5:59PM |
3 |
Problem with SIP |
| 3:46PM |
0 |
asterisk-users Digest, Vol 72, Issue 49 |
| 3:46PM |
0 |
Got SIP response 603 decline, then the call hang up |
| 3:35PM |
1 |
Dahdi 2.3.0.1 fails to compile in Xen DomU |
| 1:42PM |
2 |
Dahdi - Meetme problem on a VM |
| 12:19PM |
1 |
Different source IP address for each peer |
| 12:14PM |
2 |
OT - Gigaset and auto-configuration code |
| 12:05PM |
1 |
Preserving CDR(accountcode) in Local channels |
| 9:02AM |
4 |
Call not going through and failing because "never answered" |
| |
| Monday July 19 2010 |
| Time | Replies | Subject |
|---|
| 9:06PM |
2 |
Problems with Dahdi 2.3.0.1 trying to load OSLEC |
| 6:16PM |
1 |
Voice prompts |
| 4:30PM |
1 |
Swedish voiceprmpts |
| 4:22PM |
1 |
digium HW echocancellation - fax tone detection |
| 3:47PM |
2 |
Multiple sip.conf files? |
| 3:11PM |
1 |
Problem with E1 |
| 3:00PM |
0 |
Pereserving the callerid value when presentation set to witheld over sip |
| 12:41PM |
1 |
Asterisk Queue + Caller ID issue |
| 11:08AM |
1 |
hi |
| 10:01AM |
3 |
Voip rates to Mali |
| 3:26AM |
3 |
T.30 fax receiving problem with app_fax |
| |
| Sunday July 18 2010 |
| Time | Replies | Subject |
|---|
| 8:21PM |
0 |
Softphone's |
| 6:58PM |
0 |
FreeSide |
| 2:28PM |
1 |
Logging registration/unregistration of peers/extensions in database |
| 11:48AM |
1 |
Skype for Asterisk, Skype For SIP |
| |
| Saturday July 17 2010 |
| Time | Replies | Subject |
|---|
| 10:52PM |
2 |
Audacity settings for Asterisk sound files |
| 5:19PM |
1 |
AGI execution after Dial |
| 6:47AM |
0 |
asterisk-users Digest, Vol 72, Issue 42 |
| 6:11AM |
0 |
Queue Call Transfer Issue |
| 4:35AM |
1 |
AGI gosub return value |
| |
| Friday July 16 2010 |
| Time | Replies | Subject |
|---|
| 9:57PM |
1 |
g729 codec loading |
| 6:38PM |
6 |
Video IVR Asterisk ? |
| 5:59PM |
4 |
RFCFS - reload specified file |
| 5:42PM |
0 |
Set Queue to ring in only one member |
| 5:04PM |
0 |
beeping during calls |
| 4:56PM |
1 |
(no subject) |
| 4:43PM |
0 |
1.6.2 ConfBridge suggestion |
| 4:34PM |
1 |
Busy Lamp Fields |
| 1:19PM |
0 |
asterisk-users Digest, Vol 72, Issue 39 |
| 10:44AM |
4 |
chan_local - Asterisk 1.6.2.6 |
| 8:17AM |
1 |
BLF - Realtime & Asterisk |
| 7:48AM |
0 |
Today on VUC: SIP-Aware Appliances to facilitate communications |
| 5:49AM |
1 |
Queue |
| 4:55AM |
1 |
IAX endpoints not Registering after upgrage from Asterisk ver 1.4.26.1 |
| 1:57AM |
2 |
SKYPE - Authenticate incoming call |
| |
| Thursday July 15 2010 |
| Time | Replies | Subject |
|---|
| 9:06PM |
0 |
Dahdi T1 CRC4 errors? |
| 8:51PM |
0 |
help with sip registration |
| 8:13PM |
3 |
Soft-phone on Black Berry |
| 8:01PM |
1 |
Does Flash Operator Panel allow for dragging a call into a parking lot? |
| 7:24PM |
1 |
Asterisk Manager Problem |
| 5:13PM |
1 |
QoS and Asterisk |
| 5:02PM |
0 |
WARNING[15867]: chan_sip.c:15766 |
| 4:40PM |
3 |
Good script to make appointment? |
| 3:17PM |
0 |
Last call for AstriCon talks |
| 2:38PM |
6 |
One way audio when dialing multiple registrations |
| 1:16PM |
1 |
centos 5 rpm pacakges (add asterisk16-xmpp module) |
| 9:50AM |
0 |
MeetMe incorrectly reading key presses |
| 8:36AM |
1 |
Invalid host name |
| 8:15AM |
1 |
How to deal with voice SMS - Asterisk 1.4 |
| 4:30AM |
0 |
Get channel name of originated channel |
| 4:29AM |
1 |
SKYPE - Authenticate incoming call automatically |
| |
| Wednesday July 14 2010 |
| Time | Replies | Subject |
|---|
| 10:29PM |
2 |
sip message to ip 330 or 550 phones |
| 9:36PM |
1 |
DAHDI Outdial To Cell Phone Playing Music |
| 9:27PM |
2 |
Hosted PBX in the UK |
| 6:53PM |
2 |
realtime music on hold |
| 6:44PM |
2 |
Distinctive ring for INTERNAL calls only? How to do it? |
| 6:23PM |
1 |
Dahdi Echo canceller setup |
| 4:27PM |
2 |
beeping during call |
| 4:04PM |
2 |
Where should I look for MWI settings if Aastra phones don't do it? |
| 11:17AM |
2 |
BLF with Realtime |
| 6:33AM |
3 |
Asterisk core dumping on SendFax with FFA |
| 5:47AM |
1 |
Silence RTP |
| 4:38AM |
1 |
Can't compile DAHDI - wrong kernel source |
| 3:53AM |
1 |
How to pass through supported 100rel |
| |
| Tuesday July 13 2010 |
| Time | Replies | Subject |
|---|
| 5:35PM |
0 |
asterisk un-registering from provider |
| 4:30PM |
3 |
STRFTIME function declared in globals context |
| 2:40PM |
0 |
How to install speex codec for Asterisk that is downloaded from Digium Yum Repository? |
| 1:27PM |
1 |
MyFuel Express FO - Shortcomings **PLEASE DELETE THREAD** |
| 12:33PM |
1 |
MyFuel Express FO - Shortcomings |
| 8:52AM |
3 |
OT: fail2ban, spam and mail servers |
| 8:29AM |
3 |
Recording from g729 to wav means transcoding ? |
| 7:46AM |
0 |
How to trace incoming AMI requests ? |
| 3:14AM |
0 |
OT: HUD3 and NON-Trixbox Asterisk? |
| |
| Monday July 12 2010 |
| Time | Replies | Subject |
|---|
| 10:09PM |
1 |
Complex Dialplan Help Needed |
| 7:58PM |
1 |
browser pop-up on call ring |
| 6:06PM |
1 |
My own FreePBX FollowME module - Stuck at Reload - Anyone else had experience with this? |
| 4:58PM |
1 |
send Variable to remote system via AMI / Orginate |
| 4:58PM |
0 |
Inconsistent Behavior in SYSTEMSTATUS After System() Call |
| 3:41PM |
10 |
MAC Address prefixes of Voip equipment |
| 3:36PM |
1 |
Chanspy - Meetme |
| 1:09PM |
4 |
Remote-Party-ID party=called |
| 1:08PM |
1 |
Fax for Asterisk, capable of receiving from website but not from fax machine !! |
| 8:59AM |
0 |
DTFM Detection issues |
| 8:17AM |
0 |
ResetCDR not working after forced hangup |
| 4:31AM |
2 |
Use asterisk as a backend PBX |
| 2:47AM |
2 |
ztdummy IVR no voice |
| 2:29AM |
3 |
need information |
| |
| Sunday July 11 2010 |
| Time | Replies | Subject |
|---|
| 11:20PM |
0 |
LIMIT_PLAYAUDIO_CALLEE LIMIT_PLAYAUDIO_CALLER |
| 12:00PM |
3 |
Dialplan question when using a round-robin |
| 9:40AM |
1 |
Is a device a member of a queue? |
| |
| Saturday July 10 2010 |
| Time | Replies | Subject |
|---|
| 6:16PM |
1 |
How can get user inputs from called party after dial? |
| 4:07AM |
2 |
PHP can't insert - Can someone please help |
| 1:57AM |
1 |
False answer() being sent by cellphone providers |
| |
| Friday July 9 2010 |
| Time | Replies | Subject |
|---|
| 9:52PM |
0 |
Pbx_för_Windows?_-_Email_f ound_ in_subject |
| 8:54PM |
1 |
Logging codec used in CDR |
| 6:41PM |
3 |
[Dahdi 2.3.0.1] Does it need all those modules? |
| 5:27PM |
1 |
chan_iax2: I should never be called! |
| 4:57PM |
2 |
<no subject> |
| 4:48PM |
1 |
Delay between answer and pickup ? |
| 4:42PM |
3 |
power outage |
| 2:30PM |
2 |
Call failed: 408 timeout |
| 1:29PM |
0 |
Pbx_för_Windows?_-_Email_found_ in_subject |
| 12:41PM |
6 |
Pbx för Windows? |
| 12:36PM |
0 |
Sip Proxy |
| 11:38AM |
2 |
Re : Re : Re : Communication IAX2 >SIP>IAX2 |
| 8:29AM |
1 |
Click2call from an OpenOffice document |
| 6:29AM |
3 |
How to calculate number of speakers needed for PAGING and INTERCOM coverage area? |
| 5:32AM |
0 |
Rx/Tx fine tuning of analogue card to PRI card - Am I right with my theory? |
| |
| Thursday July 8 2010 |
| Time | Replies | Subject |
|---|
| 9:50PM |
3 |
Not detecting hangup |
| 8:26PM |
0 |
How to know which party hangup() first when using analogue cards? - Dahdi and Asterisk 1.4.x |
| 8:18PM |
2 |
DTMF issues/redial tones with rfc2833 |
| 7:33PM |
1 |
Level3 reseller needed |
| 5:01PM |
1 |
Incoming call doesn't finish when internal phone hangs up |
| 4:22PM |
0 |
connecting to EON millenium getting 405 message |
| 4:04PM |
2 |
Asterisk + Hylafax + Iiaxmodem - Outbound number. |
| 3:48PM |
1 |
Junghanns QuadBRi not really recognized in Dahdi |
| 3:30PM |
1 |
not sure what to change to point the timing to the at&t circuits? |
| 1:36PM |
0 |
How to integrate thirdparty RTP with Asterisk |
| 1:17PM |
0 |
call deflection support in chan_dahdi, libpri |
| 12:54PM |
0 |
Recordings in the bank. |
| 10:49AM |
1 |
AGI get full variable |
| 7:51AM |
1 |
Problem with call-limit |
| 6:51AM |
10 |
Asterisk Crashes - Segmentation Fault |
| |
| Wednesday July 7 2010 |
| Time | Replies | Subject |
|---|
| 11:19PM |
2 |
GURUs - How to monitor all MySQL actions on an Asterisk/FreePBX server? |
| 6:18PM |
4 |
Conditional "includes" in iax.conf |
| 5:45PM |
0 |
Problems with dual FXO/FXS cards - noise |
| 5:34PM |
2 |
Communication IAX2 >SIP>IAX2 |
| 1:40PM |
0 |
Caller ID on analog line |
| 8:38AM |
0 |
IAX calling presentation null |
| 8:13AM |
3 |
This may be a problem.. Answer not working on 1.4.32 over SIP trunk.. |
| |
| Tuesday July 6 2010 |
| Time | Replies | Subject |
|---|
| 11:11PM |
1 |
sip.conf User vs Username |
| 8:10PM |
2 |
Can't dial out through AMI |
| 6:11PM |
2 |
Y-cords - What are they ? |
| 5:53PM |
0 |
Problem with wct4xxp - cannot make calls |
| 4:29PM |
0 |
asterisk: call failed error 408 timeout |
| 3:03PM |
1 |
Dahdi - Which process to swap from Octo to QuadBRI ? |
| 2:34PM |
1 |
1.6.2: Using hints on multiple parking lots |
| 1:10PM |
1 |
IVR |
| 11:40AM |
3 |
How to secure Configuration files |
| 10:37AM |
2 |
Dahdi - alarm which clears itself - Should I care ? |
| 9:34AM |
2 |
ARA : Realtime or not ? |
| 12:17AM |
1 |
Externnotify on pollmailboxes=yes |
| 12:09AM |
0 |
97 issues marked 'Ready for Testing' |
| |
| Monday July 5 2010 |
| Time | Replies | Subject |
|---|
| 11:51PM |
1 |
problem with voicemail contexts |
| 11:29PM |
7 |
How to Dialogic 240/JCT-T1 interface with Asterisk? |
| 9:10PM |
2 |
dahdi on solaris |
| 8:52PM |
1 |
Anybody with experience with Aculab Groomer II |
| 8:16PM |
0 |
Reinvite to alaw after T.38 reception |
| 2:57PM |
1 |
Problems with ulaw/g729 translation |
| 2:02PM |
1 |
res_fax_digium and T.38 error correction |
| 12:04PM |
0 |
Reg. EMT-22 IP Phone |
| 10:45AM |
1 |
[NAT] * + private IP + locked-down firewalls? |
| 9:31AM |
1 |
SIP response 482 "Loop Detected" |
| 7:04AM |
0 |
Hold and Retrieve the call through AGI |
| 2:40AM |
1 |
Why does my IAX2 trunk between two office hangup a channel after 30 seconds? Can you share your IAX2 trunking configuration? URGENT HELP much appreciated |
| |
| Sunday July 4 2010 |
| Time | Replies | Subject |
|---|
| 2:40PM |
1 |
Anyway to know when a channel is going to hangup if Dial Timeout option is used? |
| 11:32AM |
1 |
Asterisk for transcoding |
| |
| Saturday July 3 2010 |
| Time | Replies | Subject |
|---|
| 1:04PM |
2 |
Couple of questions about modules |
| 10:53AM |
0 |
Join July Global via VOIP Free SW HW Culture Mtgs - BerkeleyTIP |
| 8:59AM |
0 |
strange issue while setting pin in MeetMe |
| 7:09AM |
1 |
VoIP Users Conference Recordings |
| 6:09AM |
0 |
[asterisk-user] gsmtolin_framein: Invalid GSM data |
| |
| Friday July 2 2010 |
| Time | Replies | Subject |
|---|
| 10:33PM |
3 |
Using AMI Originate to call 2 outside numbers and connect them |
| 5:10PM |
7 |
iptables/ blocking brute-force attacks, and so on... |
| 4:54PM |
1 |
asterisk and cisco 2800 |
| 12:58PM |
0 |
Fax T.38 passthrough failing after upgrade |
| 9:46AM |
0 |
Difference in dahdi between 1.4.x and trunk? |
| 9:08AM |
1 |
Transfer fails |
| 3:19AM |
1 |
DAHDI FXO calls and the 's' extension. No, Jackie-O doesn't work here--it's just an example. Sheesh! |
| 2:59AM |
3 |
GotoIfTime problem |
| 1:32AM |
0 |
SwitchVox AA355 w/ 4 Port PRI and 2 Port FXO and 2 Port FXS For Sale on eBay |
| |
| Thursday July 1 2010 |
| Time | Replies | Subject |
|---|
| 6:57PM |
0 |
rename External Directory |
| 5:20PM |
1 |
Dial SIP channel with no registration, timeout before CONGESTION? |
| 2:52PM |
3 |
Remote Party ID issue |
| 1:16PM |
2 |
Brute force attacks |
| 1:02PM |
0 |
AppDial in CEL Data |
| 12:50PM |
1 |
mISDN install on Asterisk 1.6 failing |
| 11:49AM |
3 |
Originate multiple channels |
| 7:51AM |
2 |
p2p or p2mp for BRI |
| 2:44AM |
1 |
call file question |
| 2:36AM |
2 |
Want to retrieve the value of contact header |