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Dec 2012
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asterisk users
79364 threads
Dec 2012
96 threads
Monday December 31 2012
Time
Replies
Subject
9:42AM
5
new user help required to build voice recorder with asterisk
Sunday December 30 2012
Time
Replies
Subject
11:44PM
2
Compile asterisk11.1 for i586 VIA C3 CPU
3:49PM
1
Timeout(absolute) not working on transfer
2:58PM
1
hangup&timeout option in asterisk 1.8
1:41PM
0
Problem with Speex codec
Saturday December 29 2012
Time
Replies
Subject
11:20PM
5
Users list email totals by year .
1:02PM
5
Top Posting
Friday December 28 2012
Time
Replies
Subject
1:59PM
5
asterisk seg fault 1.4.43
1:48PM
1
Delaying retry since we're currently running
Thursday December 27 2012
Time
Replies
Subject
11:48PM
1
Cisco AS5300 - no incoming sound
9:41PM
1
Call Forwarding / Follow-Me on PRI
9:28PM
2
$100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through
7:46PM
4
How do *you* test your changes to dialplans ruled by GotoIfTime?
5:55PM
2
$100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
4:10PM
2
Asterisk with Cisco 887M
3:49PM
1
CHANNEL(t38passthrough) is 0
9:13AM
3
stop log/debug messages into /var/log/messages
7:59AM
9
Paging for Praying
Wednesday December 26 2012
Time
Replies
Subject
9:34PM
0
Presence Registration on the D40
Tuesday December 25 2012
Time
Replies
Subject
2:57PM
2
Vxml record voice parameter
Monday December 24 2012
Time
Replies
Subject
3:13PM
2
What is the maximum number of meetme's allowed?
11:40AM
0
Asterisk Dimensioning on newer processors
10:01AM
0
How to disable authorization during Incoming calls to asterisk
Saturday December 22 2012
Time
Replies
Subject
12:28PM
1
Call hangs when selected queue number 1.
Friday December 21 2012
Time
Replies
Subject
5:16PM
0
CDR written before hangup extension
4:09PM
0
spa508g and park
3:55PM
2
Called Party Name between Asterisk systems
4:42AM
2
dahdi timing source multiple cards
Thursday December 20 2012
Time
Replies
Subject
11:13PM
1
bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call
9:04PM
0
libpri 1.4.14 Now Available
4:15PM
1
sip call failed in openbts with asterisk
1:46PM
2
asterisk 11 and no RTP
1:19PM
0
Recommended T.38 settings for receiving faxes from Cisco AS5350XM
12:46AM
7
asterisk 11 and DAHDI/i4
Wednesday December 19 2012
Time
Replies
Subject
9:47PM
1
Congestion() forcing PRI channels to be not available
6:46PM
0
queues show some agents "(In use)" from the start
5:11PM
0
loop start vs. kewl start for T1 interface
2:11PM
1
asterisk conferencing |MEETME or app_conference
9:04AM
1
Unable to create channel of type 'SIP' (cause 20 - Unknown)
5:23AM
1
Dialplan - working out when users answer
Tuesday December 18 2012
Time
Replies
Subject
11:19PM
2
Catching "hold" in dialplan
9:46PM
1
ReceiveFax
4:56PM
1
Asterisk 1.8.19.0 - "[2012-12-18 19:19:51] ERROR[24485]: astobj2.c:115 INTERNAL_OBJ: user_data is NULL"
Monday December 17 2012
Time
Replies
Subject
10:54PM
1
FreePBX website
6:59AM
1
[webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
Friday December 14 2012
Time
Replies
Subject
5:59PM
1
BRI D-channel goes up and down
3:49PM
1
It's possible a redudant Queue?
3:16PM
1
Possible bug - queue doesn't play hold music
1:04PM
1
Doubt regarding jabber
7:39AM
1
sip-user status
Thursday December 13 2012
Time
Replies
Subject
8:09PM
2
Digital accoustics trying to register to asterisk 1.4.43
12:31PM
1
call recording via 3rd INVITE/SIP leg
4:18AM
0
Has iCall gone belly up? iCall carrier services bankrupt?
Wednesday December 12 2012
Time
Replies
Subject
9:48PM
1
Polycom phones and ring no answer/302 Moved Temporarily
7:17PM
0
wcb4xxp extra hardware IDs
6:44PM
1
Asterisk 11 originate errors
5:33PM
1
chan_capi audio quality issue
Tuesday December 11 2012
Time
Replies
Subject
9:52PM
1
MACRO_CONTEXT equivalent for GoSub
7:32PM
1
DECT phone for home: siemens A510 v. Grandstream DP715
5:27PM
1
disconnect supervision
3:02PM
1
[asterisk] Guide for setup a server for end2end video call
4:34AM
2
date - outgoing call
2:57AM
0
monitoring - hangup channel
1:07AM
0
Asterisk 11.1.0 Now Available
1:05AM
0
Asterisk 10.11.0 Now Available
1:04AM
0
asterisk 1.8.19.0 Now Available
Monday December 10 2012
Time
Replies
Subject
8:58PM
1
Problem with SIP trunk I've set up between two * boxes.
5:59PM
1
Partial authentication possible?
4:01PM
1
deadagi on 11 and 1.4
2:24PM
3
Is there an issue with 11.0.2 and registration
5:37AM
1
ODBC Connection Problem
4:35AM
0
Gateway setup
Sunday December 9 2012
Time
Replies
Subject
7:54PM
3
IAX2 over OpenVPN connection.... working but
Saturday December 8 2012
Time
Replies
Subject
3:24PM
1
Question on variables and asterisk 11
11:55AM
2
Queue joinempty, even after AddQueueMember
Friday December 7 2012
Time
Replies
Subject
10:26PM
0
Adding custom HTTP headers to Asterisk
7:18AM
0
Trunking through an old Asterisk box.
Thursday December 6 2012
Time
Replies
Subject
9:09PM
5
[OT] Polycom IP450 Firmware Issues
7:48PM
0
asterisk 11.0.2 Now Available
7:47PM
0
asterisk 10.10.1 Now Available
7:47PM
1
asterisk 1.8.18.1 Now Available
6:32PM
1
Change phone display from queue calls
5:23PM
1
CDR - Freepbx - Safe to add primary key to table ?
4:28PM
1
Audio feedback - where to troubleshoot?
3:50PM
2
BLF and call-limit in 1.8
4:33AM
1
google talk under asterisk 11.0.1
Wednesday December 5 2012
Time
Replies
Subject
5:53PM
3
- configure ring group
5:49PM
0
Asterisk TDMoE (packetized PRI) link
7:48AM
2
PRI can receive calls but cannot dial out
Tuesday December 4 2012
Time
Replies
Subject
8:47PM
1
How to check channel status and move on silently?
6:49PM
1
app.c: No audio available on SIP
5:59AM
8
How to roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes
Monday December 3 2012
Time
Replies
Subject
3:17PM
1
Query list of defined channel variables via AMI
9:38AM
1
Calling from SIP client then bridge between two end points
Sunday December 2 2012
Time
Replies
Subject
2:55PM
1
Support for IP Camera streaming (RTSP) channel to a conference
Saturday December 1 2012
Time
Replies
Subject
3:04PM
1
setvar from chan_dahdi.conf