| Monday November 30 2009 |
| Time | Replies | Subject |
|---|
| 10:25PM |
0 |
Asterisk and XMPP Jingle : testers needed |
| 9:58PM |
0 |
AST-2009-010: RTP Remote Crash Vulnerability |
| 9:50PM |
3 |
Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 Now Available |
| 9:41PM |
1 |
DAHDI - BRI - Astribank |
| 4:42PM |
1 |
Polycom 500 format file system on every reboot |
| 1:18PM |
0 |
UniMRCP Integrated Asterisk Deployment |
| 9:45AM |
0 |
Warning: __ast_register_translator: plc_samples 160 format f/__ast_string_field_init: trying to reset empty pool |
| 8:32AM |
2 |
No application 'ReceiveFAX' |
| 5:32AM |
0 |
Gtalk Asterisk integration |
| 1:29AM |
2 |
AGI stuff |
| |
| Sunday November 29 2009 |
| Time | Replies | Subject |
|---|
| 5:30PM |
1 |
Asterisk H323 channel and the UDP/TCP rage ports (Q931, H245, T120, RTP) |
| 12:34PM |
3 |
Parsing custom SIP headers |
| 1:22AM |
2 |
VoiceMail greetings |
| |
| Saturday November 28 2009 |
| Time | Replies | Subject |
|---|
| 10:17PM |
1 |
DAHDI/1-2 v. DAHDI/2-1 ?? |
| 1:49PM |
2 |
can't hear anything at incoming calls |
| 12:35PM |
0 |
NvFaxdetect and Asterisk 1.4.27 - Someone get it work? |
| 11:52AM |
2 |
Max how many users in sip.conf |
| 7:17AM |
2 |
Free Polycom Provisioning Tool |
| |
| Friday November 27 2009 |
| Time | Replies | Subject |
|---|
| 10:27PM |
1 |
Asterisk 1.6.2.0-rc6 + Teliax = First Part Of Audio File Playback Cut Off |
| 7:01PM |
0 |
queue hangup |
| 2:50PM |
1 |
Which IP Phone and the codecs |
| 2:33PM |
0 |
Good quality replacement for Linksys SPA-3102 recommendation. |
| 1:50PM |
0 |
Need help with this conf |
| 12:54PM |
2 |
1800 DID Provider - Suggestion |
| 12:11PM |
1 |
Realtime SIP Register |
| 12:08PM |
1 |
Virtual Phone for CDR Logging |
| 9:33AM |
1 |
ISDN30 Timing Sources (Jon Morgan) |
| 9:03AM |
3 |
ASTERISK and SNMP |
| |
| Thursday November 26 2009 |
| Time | Replies | Subject |
|---|
| 7:33PM |
0 |
AGI and Music on hold |
| 7:05PM |
0 |
TE420B - CPU usage increase |
| 6:57PM |
2 |
Problem with Portech MV-372 |
| 2:13PM |
1 |
Polycom retrieve call from hold |
| 1:41PM |
2 |
TE412P with zaptel |
| 11:47AM |
1 |
app_read does not seem to work with SIP early media (it answers the channel) |
| 11:38AM |
1 |
CDR & Queue |
| 9:32AM |
0 |
GUI for Asterisk+LDAP - testers needed |
| 12:45AM |
1 |
Unable to open sound file error |
| |
| Wednesday November 25 2009 |
| Time | Replies | Subject |
|---|
| 11:41PM |
1 |
Agent with External Number as Extension |
| 11:01PM |
2 |
Restricting transfers between SIP phones |
| 8:07PM |
7 |
Questions about static |
| 7:57PM |
1 |
Channel Variable |
| 4:58PM |
1 |
office / homeuser |
| 12:42PM |
0 |
asterisk + res_config_ldap = asterisk.core |
| 10:18AM |
6 |
How many lines do you use. |
| 10:09AM |
0 |
DGP 301hard phone incomming problem. |
| 9:59AM |
4 |
ChanIsAvail querry |
| 8:38AM |
0 |
FW: Change the FROM filed username and From |
| 12:08AM |
0 |
Where are documented channel-dependant Dial options ? |
| |
| Tuesday November 24 2009 |
| Time | Replies | Subject |
|---|
| 11:03PM |
3 |
1950's UK rotary dial phone |
| 9:12PM |
1 |
Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ') |
| 8:24PM |
1 |
Route Non-Call Data to Agent Through Queue |
| 7:05PM |
2 |
Crosstalk - Is there a debug option for logging this? |
| 6:19PM |
1 |
snapgear/mcafee sg560 rebooting |
| 5:21PM |
0 |
Change the FROM filed username and From Calling id in asterisk |
| 4:47PM |
2 |
audio cuts out during IVR |
| 4:18PM |
4 |
Ring group issue |
| 1:21PM |
7 |
keep asterisk in RAM |
| 1:05PM |
1 |
Cianet channel bank with noise and echo |
| 12:12PM |
2 |
IVR for asterisk |
| 12:07PM |
3 |
distribute free call minutes over different channels |
| 8:49AM |
3 |
Experience with LLDP |
| 3:30AM |
1 |
asterisk trunk CURL hangs in the dialplan |
| 2:43AM |
1 |
DIDs > PBX > Multi-channel balanced audio output? |
| 1:25AM |
2 |
can't get pap2 to register from outside the LAN. |
| |
| Monday November 23 2009 |
| Time | Replies | Subject |
|---|
| 11:22PM |
0 |
Got SIP response 420 "Bad Extension" back from inphonex.com |
| 9:05PM |
2 |
SIP over TCP/TLS for 1.4 branch |
| 8:18PM |
0 |
TDM400P alarm state |
| 6:11PM |
2 |
GotoIfTime problem - possible bug |
| 5:59PM |
0 |
ADSI... |
| 5:41PM |
0 |
best channel driver for 1.4.x and beronet/junghanns 4BRI? |
| 5:40PM |
0 |
Asterisk 1.4 and kernel panic and IRQ interrupts |
| 3:37PM |
2 |
Questions about Voicemail |
| 3:07PM |
1 |
1.6.1.10 Music On Hold |
| 2:15PM |
3 |
Please some enlightment on ENUM !! |
| 12:49PM |
4 |
Connect Two Asterisk's using isdn Cards |
| 10:21AM |
7 |
Get the extension dailed |
| 10:07AM |
1 |
Is Answer really needed |
| 7:17AM |
1 |
Meetme 'o' - what actually it does..?? |
| 12:06AM |
2 |
Yealink SIP-T22P Auto Provisioning via HTTP ? |
| |
| Sunday November 22 2009 |
| Time | Replies | Subject |
|---|
| 10:54PM |
1 |
End to End delay calculation |
| 7:15PM |
1 |
Portec - feedback wanted |
| 7:06PM |
0 |
Sending call information to handset |
| 5:25PM |
1 |
Wierd problem |
| 4:09PM |
1 |
transferring SIP call: no voice |
| 1:15PM |
1 |
Prevent Dial if any extension is busy |
| 10:46AM |
1 |
Development on top of freePbx Gui and AsteriskNow |
| 5:58AM |
0 |
How do I take out one office out of the call stream? |
| |
| Saturday November 21 2009 |
| Time | Replies | Subject |
|---|
| 10:34PM |
1 |
Verification number / code |
| 3:18PM |
4 |
DIDs |
| 1:34PM |
0 |
PCI analog cards on * vs. Quintum |
| 7:15AM |
3 |
Connect two Asterisk Server in IAX ? |
| |
| Friday November 20 2009 |
| Time | Replies | Subject |
|---|
| 11:23PM |
1 |
Cisco 7961 - can't place calls |
| 11:03PM |
1 |
How to change outgoing DTMF frequencies on zaptel? |
| 10:40PM |
1 |
Trasnfer to a different VM box after leaving a VM |
| 9:41PM |
1 |
server unresponsive |
| 8:39PM |
1 |
Problem with blind transfers |
| 8:39PM |
1 |
2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk |
| 5:11PM |
1 |
PHP AGI : handle Event /AGI session |
| 2:21PM |
2 |
Mix of Swedish and English voice prompts |
| 12:36PM |
1 |
Dial Plan Application(main-menu) |
| 12:07PM |
1 |
I don't know how to authenticate |
| 2:04AM |
2 |
Setting up Nokia e71: registration problem |
| 12:05AM |
0 |
Sip phones on localnet AND outside localnet problem |
| |
| Thursday November 19 2009 |
| Time | Replies | Subject |
|---|
| 10:05PM |
0 |
Dahdi channels interference |
| 9:48PM |
3 |
Newbie |
| 9:18PM |
1 |
Type Of Number setting (pridialplan) is not effective |
| 7:32PM |
7 |
AXVoice Server Hacked.. accounts info leaked |
| 5:36PM |
0 |
Can asterisk PRI/BRI support redirect calls |
| 4:50PM |
1 |
make sounds - doesn't pull all audio tarballs. |
| 2:50PM |
1 |
Meetme |
| 2:46PM |
1 |
Dahdi_genconf replies Empty configuration -- no spans |
| 2:37PM |
2 |
Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10 Now Available |
| 7:49AM |
2 |
Send the same message to list of users |
| 6:19AM |
1 |
Asterisk crashes : Failed to start PBX |
| 6:01AM |
2 |
Dahdi and Junghanns QuadBRI |
| 5:34AM |
1 |
SIP Calls on Asterisk fails after 25000 calls |
| 3:29AM |
2 |
Gain |
| |
| Wednesday November 18 2009 |
| Time | Replies | Subject |
|---|
| 8:00PM |
0 |
Off Topic |
| 7:39PM |
0 |
Problem install wctdm24xxxp [resolved] |
| 6:08PM |
2 |
Problem install wctdm24xxxp |
| 5:04PM |
0 |
Asterisk 1.2.18 and meetme causing Audio bleeds |
| 5:03PM |
0 |
AGI and paging |
| 4:47PM |
0 |
Bug CDR report - dst "s" ? |
| 3:21PM |
2 |
Queues without agent login |
| 3:06PM |
1 |
clever ways to "share" an extension between sip and fxs |
| 10:34AM |
3 |
asterisk 1.4.26.3 makes kernel panic |
| 10:29AM |
2 |
Saving CDR on Different Databases |
| 9:37AM |
0 |
question about call transfer |
| |
| Tuesday November 17 2009 |
| Time | Replies | Subject |
|---|
| 5:27PM |
2 |
New Open Source CTI client for Asterisk |
| 5:20PM |
2 |
asterisk-users Digest, Vol 64, Issue 52 |
| 4:33PM |
3 |
newbie question |
| 2:50PM |
3 |
softphone/debug panel with BLF |
| 2:46PM |
0 |
*1.4 Received SIP subscribe for unknown event package: call-info |
| 1:35PM |
0 |
Cisco 7960 md5secret password problem |
| 8:55AM |
0 |
help vxml and asterisk support |
| 8:53AM |
3 |
vxml and asterisk support |
| 4:30AM |
1 |
Cisco 7971 behind NAT |
| 3:35AM |
2 |
max call duration |
| 1:42AM |
1 |
Understanding Congestion to incoming caller |
| 1:22AM |
1 |
Question about OSLEC or HPEC with AsteriskNow |
| |
| Monday November 16 2009 |
| Time | Replies | Subject |
|---|
| 11:19PM |
0 |
Asterisk VoIP Security Webinar - Video Now Available |
| 9:47PM |
1 |
Pbx-cards |
| 9:24PM |
3 |
Queues |
| 8:59PM |
0 |
SIP Change canreinvite=yes/no from dialplan? |
| 8:55PM |
0 |
Limit IAX calls on a peer, in and out |
| 8:40PM |
1 |
can't call through voip provider |
| 8:27PM |
1 |
Problem with sounds DTMF's phone keys |
| 2:50PM |
1 |
asterisk cdr - remote ip address |
| 2:40PM |
1 |
MixMonitor and Call Latency during conversation |
| 1:24PM |
0 |
ENUM and Asterisk 1.6 |
| 12:14PM |
2 |
Security Against brute force attack |
| 12:08PM |
1 |
Problems with dahdi on asterisk 1.6.1.9 with TE122 |
| 10:16AM |
1 |
Kamailio and asterisk Integration |
| 10:01AM |
2 |
Odd Local Channel and 0 billsec issue |
| 7:32AM |
0 |
ZAP/DAHDI outgoing faxdetect |
| 6:20AM |
1 |
How to write the incoming stream to pipe/socket instead of .gsm file |
| 12:15AM |
1 |
1.6.0.18-rc3: SendFAX causes restart |
| 12:13AM |
0 |
IAX2 ring cadence / time |
| |
| Sunday November 15 2009 |
| Time | Replies | Subject |
|---|
| 9:51PM |
4 |
Changing labels on Phones |
| 9:03PM |
1 |
ip source aware Authentication |
| 8:00PM |
1 |
thx fred |
| 7:27PM |
4 |
Hardware Requirement for asterisk |
| 6:05PM |
2 |
Sip incoming call issue with Asterisk 1.6 |
| 4:53PM |
0 |
Asterisk cmd Dial, disconnection party is source or destination? |
| 2:33PM |
1 |
Call IAX2 => "Call rejected, CallToken Support required" |
| 1:28PM |
1 |
VeriFone Omni VX-510 Credit Card Machine |
| 11:39AM |
1 |
call log, call detail |
| 6:52AM |
3 |
Database postgresql not able to start |
| 5:31AM |
6 |
Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server |
| |
| Saturday November 14 2009 |
| Time | Replies | Subject |
|---|
| 7:19PM |
0 |
hi friend |
| 6:47PM |
1 |
Queue application in Asterisk 1.6 |
| 5:46PM |
1 |
Brandable SIP SoftPhone (Windows) ? |
| 5:36PM |
5 |
music on hold |
| 5:29PM |
0 |
OT AG-188N |
| 3:20PM |
1 |
Asterisk with T38 Fax |
| 3:09PM |
0 |
Asterisk with H323 channel and Gnugk: no voice |
| 9:50AM |
1 |
Multi-Site GUI |
| 8:20AM |
2 |
Error Dialplan ? |
| 7:59AM |
1 |
Inquiry:Where to download Asterisk 1.4.13 for Debian server? |
| 6:39AM |
3 |
Inquiry:How to stop Asterisk? |
| |
| Friday November 13 2009 |
| Time | Replies | Subject |
|---|
| 11:55PM |
1 |
Xorcom Astribank udev issue in Ubuntu 9.10 |
| 7:55PM |
2 |
Multi Tenant Asterisk Server ? |
| 7:47PM |
0 |
asterisk SIP hangup |
| 7:11PM |
2 |
openSuse 11.2 and dahdi-linux |
| 6:36PM |
3 |
No dahdi_zttools in AsteriskNow? |
| 6:16PM |
0 |
Dear asterisk-users@lists.digium.com 78% 0FF on Pfizer. |
| 4:32PM |
1 |
destroy zombie session |
| 4:20PM |
0 |
VUC Today@12 ET: Allison Smith |
| 3:22PM |
1 |
FW: hi Dan |
| 11:28AM |
0 |
asterisk systems hang with "hfcmulti_rx no memory for rx_skb" |
| 10:44AM |
1 |
RTP traffic through Asterisk?? |
| 10:31AM |
1 |
little boy on asterisk and Debian |
| 8:47AM |
3 |
CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE |
| 7:21AM |
1 |
Multimedia PBX Solution |
| 7:17AM |
1 |
Health IVR Recordings |
| 3:18AM |
2 |
Will Digium iaxy stop working with asterisk 1.6; as it is discontinued? |
| 12:13AM |
0 |
TDM400p , asteriskNow and may other woes..... |
| |
| Thursday November 12 2009 |
| Time | Replies | Subject |
|---|
| 11:53PM |
1 |
Home line noise problem |
| 10:19PM |
3 |
Request for Review: Building Queues with Asterisk |
| 5:30PM |
1 |
How to send DTMF on Zaptel with 50ms tone duration and 50ms gap between the digits? |
| 3:44PM |
1 |
solution for NAT issues? |
| 3:38PM |
1 |
Asterisk 1.6.1.9 with FreePBX 2.5.2.1 |
| 2:53PM |
3 |
"POTS 4K linear codec" |
| 2:31PM |
5 |
my kernel is dazed and confused |
| 2:28PM |
1 |
Dell Poweredge T105 |
| 2:22PM |
1 |
Codec interface |
| 1:47PM |
3 |
allowguest defaults to yes for SIP |
| 1:39PM |
0 |
AST_CONFIG, MEETME_INFO and meetme.conf |
| 1:29PM |
1 |
BLF with SPA941? |
| 1:13PM |
1 |
Termination Question |
| 12:24PM |
3 |
Incoming Call Ring |
| 11:36AM |
0 |
Scheduling destruction of SIP dialog |
| 9:34AM |
0 |
Cisco 7970 SIP endless ringing...? |
| 9:33AM |
2 |
Need Adapter/Gateway with PSTN-interface |
| 8:16AM |
0 |
[Asterisk 0013405]: [patch] T38 gateway (fwd) |
| 6:31AM |
2 |
soft phone (X-lite) not able to register with asterisk |
| 6:10AM |
2 |
softphones (x_lite) not able to register with asterisk server |
| 2:29AM |
1 |
Can't connect to voip provider over NAT |
| |
| Wednesday November 11 2009 |
| Time | Replies | Subject |
|---|
| 11:34PM |
2 |
Asterisk keeps sending invite to sip phone "No response to critical packet" |
| 10:50PM |
1 |
What happened to netxusa? |
| 9:34PM |
0 |
AstriCon Videos and Presentations: First batch is on-line! |
| 9:17PM |
1 |
How to control DTMF tone duration on Zap channels? |
| 8:23PM |
2 |
Bug or feature: SIP chanvars not overriden |
| 8:08PM |
1 |
TE121 - Idle system load at ~0.3 - Bad DAHDI 2.2.0.2 behaviour ?! |
| 8:04PM |
2 |
SIP source address error |
| 4:19PM |
1 |
Issue calling from WAN to LAN extension |
| 4:13PM |
2 |
Best practice to set up 4 line phones |
| 4:00PM |
0 |
DAHDIScan() only returns dead air |
| 2:01PM |
4 |
Bad quality of call |
| 12:20PM |
1 |
Voicemail after hangup |
| 11:45AM |
1 |
Unable to execute |
| 10:57AM |
1 |
hosted / virtual IPBX platform |
| 5:14AM |
1 |
SIP response code 603 |
| |
| Tuesday November 10 2009 |
| Time | Replies | Subject |
|---|
| 9:30PM |
1 |
user extension in asterisk GUI |
| 9:19PM |
1 |
Silent Dialing |
| 6:16PM |
2 |
how to configure softphones in asterisk |
| 5:06PM |
1 |
Questions about Dahdi's /etc/dahdi/genconf_parameters |
| 2:31PM |
2 |
Setting outgoing callerid on when using a PRI |
| 1:35PM |
2 |
Hangup |
| 1:04PM |
2 |
looking for an Asterisk supervision (status viewer) tool |
| 11:36AM |
2 |
CDR Import |
| 9:45AM |
1 |
Call audio leaking between calls |
| 2:10AM |
0 |
Extension in use |
| 1:28AM |
1 |
Is voicemail to text possible? |
| 12:12AM |
2 |
Gradstream Budge Tone-201 |
| |
| Monday November 9 2009 |
| Time | Replies | Subject |
|---|
| 11:58PM |
1 |
SendText |
| 10:03PM |
1 |
Call declined |
| 9:25PM |
3 |
is an extension is use |
| 8:07PM |
0 |
chan_mobile Voice setting |
| 7:14PM |
0 |
FreeBSD, ztdummy & OHCI |
| 5:19PM |
1 |
Allow Header |
| 5:06PM |
0 |
got SIP response 482 "Loop Detected" back from xx.xxx.xxx.xxx |
| 4:32PM |
2 |
how to configure softphones in asterisk server |
| 3:41PM |
4 |
local channels |
| 10:52AM |
3 |
E1 Extensions.conf |
| 10:14AM |
1 |
How to know AMI status |
| 9:14AM |
0 |
fromuser & fromdomain |
| 12:20AM |
0 |
CDR userfield - |
| |
| Sunday November 8 2009 |
| Time | Replies | Subject |
|---|
| 10:40PM |
0 |
E1 connectivity problem (HDB3, CRC4MF, ISUP, V3) |
| 7:13PM |
2 |
outbound routing |
| 5:33PM |
0 |
Modem card |
| 2:23PM |
2 |
CDR userfield not written into DB |
| 12:22PM |
1 |
Failure of user registration with XLITE |
| 6:20AM |
3 |
how to check version of asterisk |
| 3:38AM |
3 |
Text messaging |
| 12:42AM |
0 |
Set DESTINATION CID for outbound calls |
| |
| Saturday November 7 2009 |
| Time | Replies | Subject |
|---|
| 7:45PM |
4 |
Help with concurrent VoIP calls |
| 7:18PM |
1 |
Asterisk 1.6.1 + Cisco AS5300 + Fax T38 ? |
| 6:58PM |
0 |
[DAHDI 2.2.0.2] "failed on channel 1: No such device or address" |
| 5:29PM |
1 |
help in installing asterisk |
| 12:12PM |
0 |
Nov 7 TODAY & Nov 22 - Join Global FreeSW GNU(Linux) HW Culture meeting via VOIP - BerkeleyTIP GlobalTIP - For Forwarding |
| 10:08AM |
1 |
Difference between 'core show channels' and 'sip show channels' ?? |
| 7:51AM |
0 |
AMI is not loaded |
| 5:23AM |
1 |
Trouble registering Cisco 7942 |
| 12:02AM |
6 |
Location |
| |
| Friday November 6 2009 |
| Time | Replies | Subject |
|---|
| 8:52PM |
1 |
Question on peering two Asterisk servers |
| 8:22PM |
1 |
Best dahdi switchtype to emulate (network side)? |
| 7:02PM |
1 |
AMI Originate and Variable header |
| 5:08PM |
4 |
problem while compiling asterisk tar file |
| 4:13PM |
1 |
Need opinion about GSM codec for Internet |
| 3:48PM |
2 |
Routing incoming call based on caller id |
| 2:09PM |
1 |
sip set debug |
| 11:30AM |
0 |
Syncing phone numbers DB with cellphone? |
| 11:29AM |
0 |
Setting up an automatic Fax Call Back service |
| 9:11AM |
1 |
which asterisk,libpri,dahdi tar file to compile |
| 8:19AM |
0 |
[VUC] Friday Nov 6 @ 12 Noon EST: Village Telco |
| 6:47AM |
1 |
app read accept # sign |
| 6:43AM |
2 |
odbc to ms-sql server |
| 4:57AM |
2 |
Question about callerid? |
| 4:33AM |
1 |
asterisk,libpri,zaptel |
| |
| Thursday November 5 2009 |
| Time | Replies | Subject |
|---|
| 11:17PM |
1 |
Asterisk 1.4 DISA is jumoing after one digit in the DISA context |
| 9:36PM |
0 |
MeetMe thinks DAHDI is missing 1.6.0.10 |
| 9:15PM |
0 |
SIP 503 instead of SIP 480 in asterisk debug mode |
| 9:06PM |
0 |
RTP Proxy |
| 8:45PM |
2 |
Prevent cell phone voice mail capturing call |
| 7:42PM |
0 |
Chan_mobile instability |
| 5:33PM |
1 |
Cisco 7912 Phones + Asterisk |
| 2:54PM |
3 |
programming phones |
| 2:42PM |
2 |
7777 & *65 |
| 2:18PM |
2 |
Asterisk 1.4 remote pickup |
| 1:15PM |
1 |
Cisco 7912 SIP Firmware |
| 12:47PM |
0 |
SIP brute force attack from Korea |
| 12:06PM |
1 |
Playing Sound during dial |
| 11:27AM |
2 |
faxes received on mISDN |
| 10:34AM |
0 |
fax standard extension and Playback |
| 8:14AM |
1 |
SendJabber question sending Links |
| 3:07AM |
1 |
dialplan pattern matching |
| |
| Wednesday November 4 2009 |
| Time | Replies | Subject |
|---|
| 8:21PM |
0 |
Asterisk 1.2.36, 1.4.26.3, 1.6.0.17, and 1.6.1.9 Now Available |
| 8:20PM |
2 |
Cisco SPA3102 Thoughts & Other Recommendations |
| 8:12PM |
0 |
AST-2009-009: Cross-site AJAX request vulnerability |
| 8:12PM |
0 |
AST-2009-008: SIP responses expose valid usernames |
| 7:43PM |
3 |
How to resell my trunk/provider to others? |
| 7:32PM |
0 |
Social Networking Event * Berlin Nov 12 |
| 6:57PM |
0 |
Fwd: Asterisk conferences |
| 4:51PM |
2 |
Minimum hardware requirements for 10 concurrent calls? |
| 4:44PM |
2 |
Asterisk on a MiniITX board+Atom1.6 2gb+Sangoma USB? |
| 4:02PM |
1 |
ExternalIVR testing |
| 3:12PM |
0 |
memory leak with static users |
| 3:11PM |
0 |
channel destruction after a transfer call |
| 1:23PM |
0 |
Call Transfer Problem |
| 12:24PM |
0 |
UK Vodafone messaging, ISDN, Wrong CallerID being used. |
| 11:48AM |
3 |
Asterisk 1.6.1.6 crashing |
| 11:01AM |
2 |
Help in Perl AGI |
| 10:23AM |
1 |
segfault wall |
| 9:16AM |
2 |
Asterisk SS7 Sigtran Protocol |
| 6:00AM |
1 |
Personal invitation from daminda edirisinghe |
| 1:48AM |
0 |
Fwd: Seminarios Tecnológicos en Fundación Proydesa |
| |
| Tuesday November 3 2009 |
| Time | Replies | Subject |
|---|
| 7:02PM |
1 |
ring groups with different caller id |
| 5:28PM |
1 |
MusicOnHold works Externally, but not internally |
| 4:09PM |
2 |
Extra CDR fields |
| 1:56PM |
5 |
Asterisk and Software Data Modem |
| 1:26PM |
1 |
Asterisk Realtime Extensions => for all context ? |
| 12:14PM |
3 |
Problem with ChanIsAvail |
| 11:39AM |
0 |
Redirecting Calls and MeetMe Rooms |
| 11:26AM |
0 |
Popping sounds on voice prompts |
| 11:25AM |
0 |
routes and trunks |
| 10:22AM |
1 |
dahdi channel not showing up |
| 5:25AM |
0 |
Exchange 2007 UM issues with Asterisk 1.6 |
| 4:18AM |
1 |
Core Dump - Asterisk 1.4.24 - Elastix |
| 2:56AM |
1 |
turn the ring tone OFF during dialing |
| |
| Monday November 2 2009 |
| Time | Replies | Subject |
|---|
| 10:39PM |
2 |
Asterisk as Outbound Proxy ? |
| 9:04PM |
0 |
DTMF Timing and Fujitsu F9600 Switch |
| 7:11PM |
1 |
Unexpected control subclass '-1' |
| 6:21PM |
0 |
Execute Macro AFTER connecting to a channel |
| 5:22PM |
7 |
Asterisk 1.4 and Fax |
| 4:33PM |
0 |
Nagios check_asterisk_peers needs rights to question the Asterisk-server |
| 3:37PM |
2 |
Remote IP Phone's |
| 3:18PM |
1 |
Remote Party ID |
| 2:52PM |
0 |
change L(x[:y][:z]) parameter of DIAL command after call is bridged |
| 12:38PM |
1 |
supermicro hardware + sangoma |
| 12:11PM |
4 |
GSM and Wav format |
| 11:59AM |
1 |
MySQL CDR |
| 10:14AM |
1 |
Asterisk Fax Module |
| 9:37AM |
2 |
hardware requirements for asterisk |
| 8:36AM |
0 |
PSTN Line Parameters Checking |
| 8:33AM |
3 |
Xorcom device not showing up in /proc |
| 6:49AM |
5 |
Forward DID to another server |
| 2:40AM |
4 |
include statements in IVR |
| |
| Sunday November 1 2009 |
| Time | Replies | Subject |
|---|
| 10:23PM |
2 |
Dialstatus |
| 9:13PM |
1 |
asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension |
| 8:46PM |
1 |
pattern matching DID |
| 8:21PM |
0 |
PSTN Lines and AEX808B |
| 6:42PM |
0 |
Execute the specified macro for the called channel AFTER connecting to the calling channel. |
| 6:27PM |
2 |
Tutorial for SIP user |
| 4:44PM |
0 |
Originate with Local channel to any app-only extension hangs up immediately? |
| 4:24PM |
0 |
Exchange 2007 and Voicemail with Imap Storage |
| 12:31PM |
1 |
Skyp SIP? - what is free for a home * |
| 9:53AM |
1 |
usage of manager events to create custom reports |
| 8:42AM |
1 |
Error in MeetMe modules ? |
| 5:10AM |
0 |
need help debug asterisk-1.6 sip connection |