| Monday April 30 2012 |
| Time | Replies | Subject |
|---|
| 6:26PM |
0 |
chan_mobile with Nokia 6021 - incoming SMS causes call to drop |
| 9:10AM |
2 |
Cannot send mail from System command |
| 9:09AM |
2 |
Calendar Integration Problem |
| |
| Saturday April 28 2012 |
| Time | Replies | Subject |
|---|
| 6:46PM |
1 |
Music as ringtone |
| |
| Friday April 27 2012 |
| Time | Replies | Subject |
|---|
| 8:16PM |
0 |
medooze MCU versus confbridge |
| 7:28PM |
1 |
Master Registrations? |
| 7:05PM |
0 |
Scheduled Maintenance for Asterisk Project community services |
| 6:45PM |
0 |
Asterisk as sip client Unable to create channel of type 'Console' (cause 0 - Unknown) |
| 4:39PM |
2 |
Flashphoner |
| 2:01PM |
1 |
Playing Multiple File ( simultaneously ) on Channel |
| 1:20PM |
1 |
ReceiveFax and multiple pages |
| 8:28AM |
1 |
Fwd: Flashphoner |
| 7:42AM |
1 |
No UDPTL ports remaining |
| |
| Thursday April 26 2012 |
| Time | Replies | Subject |
|---|
| 11:45PM |
1 |
POTS(FXO) line getting Red alarm after first ring(5 or 6 seconds) |
| 8:57PM |
0 |
Peer SIP authentication with Taqua switch |
| 4:54PM |
1 |
Asterisk + Phones behind different Nat Firewalls |
| 3:37PM |
0 |
OpenVPN design w/ Yealink |
| 2:29PM |
1 |
Call recovery feature |
| 9:41AM |
1 |
Mac OS X sip client with Video support |
| |
| Wednesday April 25 2012 |
| Time | Replies | Subject |
|---|
| 9:49PM |
0 |
Restart single dahdi span |
| 6:29PM |
0 |
Open source replacement for AudioCodes nCite 1000 SBC |
| 12:21PM |
2 |
CONNECTEDLINE() updated during SIP events? |
| 12:08PM |
1 |
Hangup Cause and SIP Response Code |
| 7:35AM |
2 |
chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ?? |
| |
| Tuesday April 24 2012 |
| Time | Replies | Subject |
|---|
| 8:57PM |
2 |
Looking for IAX trunk/DID to replace Junction Networks |
| 3:39PM |
5 |
Asterisk - Nortel transfer problem |
| 2:06PM |
2 |
Strange problem on ougoing call |
| 1:32PM |
0 |
Digium D40 Direction map 'X' key not functioning |
| 11:06AM |
0 |
Nicaragua PSTN Frequency Parameters |
| 10:59AM |
5 |
Set SIP peer state busy |
| 7:34AM |
0 |
Asterisk don't use "context=" |
| 7:27AM |
1 |
Question for a Jira bug marshal |
| |
| Monday April 23 2012 |
| Time | Replies | Subject |
|---|
| 8:35PM |
0 |
Problem with blank/empty voicemails |
| 8:32PM |
0 |
Grandstream 1.0.3.30 BETA Firmware |
| 6:25PM |
0 |
AST-2012-006: Remote Crash Vulnerability in SIP Channel Driver |
| 6:25PM |
0 |
AST-2012-005: Heap Buffer Overflow in Skinny Channel Driver |
| 6:25PM |
0 |
AST-2012-004: Asterisk Manager User Unauthorized Shell Access |
| 1:10PM |
0 |
Asterisk 1.6.2.24, 1.8.11.1, 10.3.1 Now Available (Security Release) |
| 12:59PM |
3 |
HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta) |
| |
| Sunday April 22 2012 |
| Time | Replies | Subject |
|---|
| 6:07PM |
1 |
meetme identify user number |
| 7:49AM |
2 |
HELP!! Caller ID "unknown" for all inbound call |
| |
| Saturday April 21 2012 |
| Time | Replies | Subject |
|---|
| 4:51PM |
1 |
DAHDI 2.6.1 - What does " Build OSLEC EC if in the tree" feature means ? |
| 1:41PM |
1 |
Open source speech recognition engine? |
| 6:19AM |
1 |
No extension found ? |
| |
| Friday April 20 2012 |
| Time | Replies | Subject |
|---|
| 6:05PM |
1 |
DAHDI-Linux 2.6.1, 2.5.1 and DAHDI-Tools 2.6.1, 2.5.1 Now Available |
| 5:20PM |
3 |
Advice on Asterisk Conference |
| 3:30PM |
1 |
E & M signalling and Dahdi |
| 2:11PM |
1 |
Experience with virtual servers? |
| |
| Thursday April 19 2012 |
| Time | Replies | Subject |
|---|
| 10:59PM |
1 |
asterisk 1.4.39 and dahdi 2.6 on Ubuntu |
| 9:29PM |
0 |
Auto answer Asterisk ; Unable to create channel of type |
| 8:11PM |
1 |
Company info |
| 6:19PM |
1 |
dahdi cannot make simaltaneous calls |
| 11:20AM |
1 |
Call recording and transfer issue (asterisk 1.8) |
| 10:17AM |
0 |
how do I catch the audio stream in real-time for another application? |
| 5:34AM |
0 |
upgrading from asterisk 1.4 to 1.6 |
| |
| Wednesday April 18 2012 |
| Time | Replies | Subject |
|---|
| 3:42PM |
1 |
Delete "Session timer" ? |
| 1:46PM |
0 |
Far end nat traversal not working |
| 1:12PM |
5 |
asterisk 1.4.39 and dahdi 2.6 |
| 10:57AM |
2 |
FXO -> GSM Gateway Problem |
| 8:02AM |
1 |
g729 freezes 1.8 |
| 6:27AM |
3 |
hints and server-side DND (do not disturb) |
| |
| Tuesday April 17 2012 |
| Time | Replies | Subject |
|---|
| 10:04PM |
1 |
Asterisk 1.8.10 getaddrinfo |
| 3:18PM |
5 |
Account code script needed. |
| 1:10PM |
1 |
Process a variable in a string. |
| 12:38PM |
1 |
Incoming SIP call is rejected always. |
| 10:28AM |
1 |
ExtensionStatus event |
| |
| Monday April 16 2012 |
| Time | Replies | Subject |
|---|
| 2:26PM |
1 |
10.3 : sip loses registration ? |
| 1:37PM |
0 |
Dial Local doesn't honore the channel language setting |
| 1:36PM |
1 |
Custom Application recording problem |
| 12:53PM |
0 |
When CALL-ID were same , I could hijack another session |
| 12:20PM |
0 |
Far end nat traversal for media is not working always |
| 12:12PM |
1 |
Invite + decreasing sequence number => 500 Error? |
| 8:56AM |
1 |
SNOM phones? Please test this patch (broken hints with notifycid=yes) |
| 6:06AM |
1 |
OT - Which Sugarcm 6.2 plugin to add click-to-dial (with asterisk) ? |
| 1:11AM |
2 |
Caller id issues |
| |
| Sunday April 15 2012 |
| Time | Replies | Subject |
|---|
| 8:48PM |
1 |
asterisk 1.4.39 and dahdi 2.6: chan_dahdi? dahdi? |
| 8:34PM |
0 |
cdr documentation - new fields |
| 7:30PM |
0 |
Call status register |
| 3:03PM |
1 |
Caller ID problem |
| 1:48PM |
1 |
Disable services on startup |
| 11:15AM |
2 |
Transcoding degradation G711<->iLBC |
| 9:49AM |
0 |
Can't make Asterisk send authentication to remote peer on INVITE [SOLVED] |
| 8:44AM |
2 |
Set variables from one asterisk ta a second. |
| |
| Saturday April 14 2012 |
| Time | Replies | Subject |
|---|
| 11:09PM |
1 |
Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
| 8:36PM |
1 |
dahdi versions before 2.5 compilation error and ubuntu |
| 3:19PM |
1 |
Dahdi QSIG with Tadiran Coral - not working |
| 9:30AM |
2 |
Can't make Asterisk send authentication to remote peer on INVITE |
| 2:08AM |
2 |
deleting: res_ldap.conf |
| 12:33AM |
1 |
BUSY vs. CONGESTION |
| |
| Friday April 13 2012 |
| Time | Replies | Subject |
|---|
| 6:56PM |
1 |
Question about asterisk to Cisco |
| 4:53PM |
4 |
priorityjumping - asterisk 1.8 |
| 2:43PM |
0 |
Release Announcement: Adhearsion 2.0 for Asterisk 1.8+ |
| 8:36AM |
1 |
Personal queue with one agent: add calls to extension |
| 7:25AM |
1 |
T.30 Fax and Echo Cancelation |
| 7:07AM |
1 |
Simple Gateway to setup calls bewteen two or 3 asterisk systems . |
| |
| Thursday April 12 2012 |
| Time | Replies | Subject |
|---|
| 8:36PM |
2 |
GXP1400 |
| 8:08PM |
0 |
Ringing detection ? |
| 7:09PM |
1 |
Asterisk 10 & app_swift problem |
| 6:04PM |
2 |
Asterisk 1.8.12.0-rc1 |
| 4:31PM |
1 |
DAHDI FXO Call Issues / Indication Types |
| 9:19AM |
0 |
T38 gateway issue |
| 9:07AM |
1 |
chan_sip.so module not loading |
| |
| Wednesday April 11 2012 |
| Time | Replies | Subject |
|---|
| 10:25PM |
1 |
DAHDI inter-digit timeout = 0 |
| 11:47AM |
0 |
red5sip SIP ua can't register |
| 11:39AM |
4 |
Dahdi-2.4.0+2.4.0 means ?? |
| 11:01AM |
0 |
AMI Originate double call |
| 6:40AM |
0 |
OT - How to localize Freepbx 2.10 or 2.9 ? |
| |
| Tuesday April 10 2012 |
| Time | Replies | Subject |
|---|
| 8:15PM |
3 |
Run AGI while agent ringing instead of only when connected |
| 4:07PM |
0 |
Recent FreePBX vulnerability attacks |
| 2:05PM |
1 |
MessageSend, SIP, and call files |
| 12:14PM |
0 |
Strange Asterisk port behavior |
| 11:11AM |
0 |
How to disable CDR adaptative logging on asterisk 1.8 ? |
| 7:19AM |
0 |
Experiences with Polycom-Kirk 6000 and DECT/GAP handsets |
| 12:34AM |
3 |
MYSQL INSERT QUESTION IN DIALPLAN |
| |
| Monday April 9 2012 |
| Time | Replies | Subject |
|---|
| 10:21PM |
1 |
syntax error from digium fax manual ?? |
| 2:44PM |
2 |
Combining multiple SIP providers |
| 2:17PM |
1 |
VMWI DAHDI |
| 2:09PM |
0 |
ChannelRedirect with callee channel |
| 12:08PM |
1 |
Call Deflection with DAHDISendCallreroutingFacility |
| 11:42AM |
2 |
Monitoring voice-quality with sip/rtp/rtcp |
| 8:22AM |
2 |
Video Conference in Asterisk1.4 (using asterisk gui) |
| |
| Sunday April 8 2012 |
| Time | Replies | Subject |
|---|
| 1:44AM |
1 |
another non-root problem: unable to set utime ?? |
| |
| Saturday April 7 2012 |
| Time | Replies | Subject |
|---|
| 8:09PM |
1 |
Unable to access the running directory (Permission denied). |
| 2:09PM |
0 |
Cannot resume call on hold |
| 1:25PM |
0 |
ISDN incoming call disconnected after picking up phone |
| 5:51AM |
2 |
Google TTS - Asterisk |
| 12:05AM |
1 |
Which file is loading these lines? |
| |
| Friday April 6 2012 |
| Time | Replies | Subject |
|---|
| 3:30PM |
1 |
Call Transfer not working |
| |
| Thursday April 5 2012 |
| Time | Replies | Subject |
|---|
| 5:35PM |
3 |
Dial Plan - Routing via Caller ID |
| 11:58AM |
1 |
Does Cisco 79XX with SIP firmware support asterisk's BLF ? |
| 8:21AM |
4 |
sip pregi net account registration |
| |
| Wednesday April 4 2012 |
| Time | Replies | Subject |
|---|
| 11:41PM |
0 |
FollowMe and billsec field of the CDRs |
| 8:50PM |
2 |
Change extension for international ? |
| 8:48PM |
1 |
issue with Digium TDM410P |
| 1:35PM |
2 |
Asterisk 1.8 and DeadAGI |
| 9:31AM |
1 |
cross ivr is comming in my ivr system |
| |
| Tuesday April 3 2012 |
| Time | Replies | Subject |
|---|
| 9:05AM |
5 |
process_sdp: Multiple audio streams are not supported |
| 8:31AM |
1 |
Max number of PCIe cards |
| 12:05AM |
1 |
extending fallback numbers |
| |
| Monday April 2 2012 |
| Time | Replies | Subject |
|---|
| 2:46PM |
2 |
Limit Call ? |
| 12:44PM |
4 |
Asterisk ACL |
| 10:18AM |
1 |
fax tone testing |
| 6:53AM |
0 |
dial rule problems( on e1 interface) after upgrading 1.8 |
| |
| Sunday April 1 2012 |
| Time | Replies | Subject |
|---|
| 4:42PM |
1 |
404 Response to Invite - Should be 401 |
| 3:22PM |
0 |
10.3.0: gtalk_request: No XMPP client to talk to, us (partial JID) |
| 1:04PM |
4 |
Telephony Card: GSM slots + Analoge |