Thursday June 30 2011 |
Time | Replies | Subject |
8:35PM |
0 |
patch openr2 asterisk 1.4.42 |
8:32PM |
0 |
Skall vi gå på Smaklösa? |
3:09PM |
6 |
Cannot figure out pound key in qwerty keyboard |
1:24PM |
0 |
cisco sip |
11:38AM |
0 |
SendFax: not setting the fax header |
8:22AM |
1 |
asterisk recording problem |
12:46AM |
0 |
Problem callerid ignored by using callfiles |
|
Wednesday June 29 2011 |
Time | Replies | Subject |
9:57PM |
1 |
No audio format found to offer. |
5:57PM |
0 |
atxfer fails to read data |
5:46PM |
1 |
Asterisk/SIP Issue - Long Shot |
5:35PM |
0 |
Asterisk 1.6.2.19 Now Available (Final Maintenance Release) |
5:34PM |
0 |
Asterisk 1.4.42 Now Available (Final Maintenance Release) |
5:23PM |
1 |
Asterisk 1.4 func_odbc frustrations |
4:06PM |
3 |
Polycom ip320 dtmf issues |
1:49PM |
4 |
Load Balance Trunks |
9:58AM |
4 |
OT - Polycom - Which provisioning protocol to choose ? |
9:13AM |
1 |
dialplan execution stops after ReceiveFax |
|
Tuesday June 28 2011 |
Time | Replies | Subject |
11:45PM |
0 |
for suport server |
8:54PM |
0 |
Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 Now Available (Security Releases) |
8:53PM |
1 |
Clarification of the terms shown on CLI |
8:31PM |
0 |
AST-2011-011: Possible enumeration of SIP users due to differing authentication responses |
7:34PM |
2 |
IVR |
7:10PM |
0 |
Set a specific BLF key on Polycom 650 [SOLVED] |
6:21PM |
2 |
Add # at the end of dialled number |
5:30PM |
2 |
Asterisk 1.6 Dahdi on Centos 5.2 |
5:00PM |
0 |
Using Dial() on SIP and DAHDI connections simultaneously |
4:58PM |
1 |
Set a specific BLF key on Polycom 650 |
4:32PM |
1 |
Outgoing calls get dropped on high-latency connections. |
3:30PM |
2 |
MixMonitor - garbled/corrupted WAV files |
10:59AM |
2 |
No audio after a reinvite changing codec ----> with SIP DEBUG!! |
9:59AM |
1 |
Asked to transmit frame type slin, while native formats is 0x8 (alaw) |
|
Monday June 27 2011 |
Time | Replies | Subject |
7:53PM |
0 |
Fax with Asterisk and T38Modem |
5:45PM |
0 |
Question regarding progressinband |
5:21PM |
2 |
Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI |
5:02PM |
0 |
rtptimeout on 1.8.4 |
3:33PM |
0 |
FW: Asterisk 1.8.4 - Google iCal not working |
3:06PM |
0 |
interface to gstreamer |
12:04PM |
0 |
Festival VOICE MAIL TO FEMAIL |
11:36AM |
0 |
Asterisk changing SIP INFO dtmf duration |
11:26AM |
3 |
not find files in asterisk 1.8 |
11:11AM |
1 |
ReceiveFax to G.711 |
9:30AM |
2 |
Agi script for working hours PBX |
1:25AM |
4 |
Conference feature |
|
Sunday June 26 2011 |
Time | Replies | Subject |
7:55PM |
1 |
Asterisk 1.8.4 - Google iCal not working |
7:06PM |
3 |
Cisco IP Phones 7942 and Skinny/SIP in asterisk |
|
Saturday June 25 2011 |
Time | Replies | Subject |
4:24PM |
0 |
HDLC Overrun with Chan SS7 |
12:39PM |
1 |
Cisco IP Phones and Skinny in asterisk 1.8.4.2 "tooooooooooooooooo" |
|
Friday June 24 2011 |
Time | Replies | Subject |
8:55PM |
3 |
t.38 virtual fax software? |
3:28PM |
0 |
failed to authenticate user message |
9:40AM |
0 |
eComm - Are you going to San Francisco next week? |
7:37AM |
2 |
Vm on a System running Asterisk. |
6:41AM |
2 |
Monitor Asterisk and Ast-gui |
4:54AM |
0 |
issues/jira |
|
Thursday June 23 2011 |
Time | Replies | Subject |
8:14PM |
0 |
Asterisk 1.4.41.1, 1.6.2.18.1, and 1.8.4.3 Now Available (Security Release) |
2:12PM |
7 |
Problem with detecting fax on PRI/DAHDI channels |
1:47PM |
0 |
menu asterisk |
11:45AM |
1 |
VMX Locator |
9:01AM |
0 |
How to set BRI-to-BRI trunk using 2 HA8+B400M cards [SOLVED] |
|
Wednesday June 22 2011 |
Time | Replies | Subject |
11:01PM |
0 |
Fwd: FW: loadstar and ok one |
8:00PM |
0 |
Asterisk/Kamailio dinner in Madrid thursday next week - June 30th |
4:51PM |
0 |
Setting the Source IP on asterisk 1.8 |
4:09PM |
1 |
Aastra phone # key in dialplan |
3:28PM |
1 |
How to set BRI-to-BRI trunk using 2 HA8+B400M cards |
1:32PM |
1 |
iLBC re-licence |
1:22PM |
0 |
Unable to include switch 'Realtime/@' in context |
11:32AM |
1 |
Digium HA8. Does settings order matters (in system.conf) ? |
5:40AM |
1 |
Office timings only work asterisk after that voicemail |
2:10AM |
2 |
Question on pause in dialing |
|
Tuesday June 21 2011 |
Time | Replies | Subject |
11:21PM |
0 |
calleridname presentation Asterisk ==> Siemens |
8:56PM |
1 |
how to get on hold events with AMI |
6:41PM |
4 |
call paging interrupts call when using Mitel 5224 |
5:07PM |
1 |
DTMF begin ignored |
1:06PM |
0 |
asteriks and ntt Japan |
12:37PM |
1 |
Call to *2*999... : IP-phone configration |
12:23PM |
0 |
Voice recognition recommendations? |
10:17AM |
0 |
AMI Suddenly not giving full response to 'Command' |
9:57AM |
0 |
dropped calls on android voip connection |
8:48AM |
0 |
xorcom asterisk patch for sending rtp stream to remote oreka server |
7:28AM |
0 |
Easier to remember transfer and pick-up key sequences |
4:56AM |
1 |
Looking for Sipura-2000 Latest Firmware |
1:38AM |
1 |
: Re: ITSP failover for PRI |
12:43AM |
2 |
gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko |
|
Monday June 20 2011 |
Time | Replies | Subject |
11:33PM |
1 |
Inbound CallerID displays asterisk |
11:09PM |
2 |
"pickupsound = beep" kills call pickup in Asterisk 1.8.4.2 |
7:09PM |
3 |
Get second cipher in an extension |
6:09PM |
2 |
Asterisk call limitation |
6:00PM |
1 |
Problems with pickupgroup/callgroup with Asterisk 1.8.4.2 |
5:17PM |
2 |
menu issue |
2:40PM |
2 |
Integration of OpenVXI |
11:04AM |
2 |
different format in asterisk |
5:56AM |
0 |
Realtime Failover - Multiple DSN 's in extconfig.conf |
|
Sunday June 19 2011 |
Time | Replies | Subject |
10:44PM |
0 |
ooh323 errors while compiling asterisk 1.8.3 and 1.8.4 |
5:42PM |
2 |
ITSP failover for PRI |
4:24PM |
3 |
Problem with ReceiveFAX app from FFA |
9:13AM |
1 |
SMS with Asterisk |
|
Saturday June 18 2011 |
Time | Replies | Subject |
6:16PM |
1 |
Direct RTP with Asterisk |
1:33AM |
2 |
ooh323 errors while compiling: asterisk 1.8.3 and 1.8.4 |
|
Friday June 17 2011 |
Time | Replies | Subject |
7:24PM |
1 |
Next Asterisk 1.8 Release |
5:54PM |
2 |
asterisk voicemail distribution groups |
5:25PM |
1 |
background audio for inbound leg |
5:20PM |
1 |
click to call |
2:51PM |
0 |
RTP Streaming |
11:30AM |
1 |
Missed calls and groups |
|
Thursday June 16 2011 |
Time | Replies | Subject |
10:12PM |
1 |
Queue Log in Mysql |
8:11PM |
1 |
CDRs in 1.8 |
8:08PM |
5 |
Bridged Digital call |
6:09PM |
0 |
show channels does not show hold status |
1:21PM |
1 |
fast AGI memory leaks |
11:36AM |
7 |
MixMonitor |
10:49AM |
0 |
Have your suggestions on Hardware configuration for Asterisk. |
9:59AM |
3 |
Cisco IP Phones 7942G (skinny): TFTP and required files |
9:50AM |
0 |
sipp application/dtmf-relay not work properly in Asterisk! |
8:11AM |
0 |
Channel variables not available during xfer? |
8:11AM |
1 |
Sending SMS on Friday at 12 Noon EDT |
7:18AM |
2 |
Inbound call not dialing exten |
6:54AM |
1 |
#include filename |
5:52AM |
2 |
How to secure our Asterisk server from hacker's ? |
3:26AM |
1 |
Web based call back |
3:23AM |
5 |
Goggle voice incoming dialplan |
|
Wednesday June 15 2011 |
Time | Replies | Subject |
5:27PM |
1 |
Re connecting to SIP Provider with virtual IP, from pacemaker cluster |
3:28PM |
0 |
connecting to SIP Provider with virtual IP from pacemaker cluster |
1:33PM |
2 |
change destination on digit |
11:06AM |
2 |
DIGIUM PRI CARDS REQUIRE |
11:00AM |
0 |
CONFERENCE CONFIGURATION REQUIRE |
10:53AM |
2 |
sig_pri.c:985 pri_find_dchan: Span 1 No D-channels available! Using Primary channel as D-channel anyway! |
10:20AM |
1 |
VOICEMAIL CONFIGURATION |
8:40AM |
0 |
Asterisk - dialog-info+xml - NAT |
8:17AM |
0 |
How to configure unused BRI ports from a HA8 board ? [SOLVED] |
7:30AM |
1 |
How to configure unused BRI ports from a HA8 board ? |
6:45AM |
1 |
call file challenge... |
5:34AM |
3 |
Dial out conference |
5:15AM |
0 |
asterisk + stun |
|
Tuesday June 14 2011 |
Time | Replies | Subject |
11:27PM |
3 |
sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway! |
9:48PM |
0 |
SPA504G Unable to Transfer Established Call |
9:41PM |
2 |
Voicemail issue |
9:36PM |
1 |
dahdi_genconf and BRI NT spans in system.conf |
9:20PM |
0 |
How to set a HA8 board + B400M in NT mode ? [SOLVED] |
9:19PM |
0 |
Possible timing issue? |
8:48PM |
0 |
sysmon on Centos Asterisk system using 100 perc CPU..How to kill it????? |
8:11PM |
1 |
How to set a HA8 board + B400M in NT mode ? |
6:47PM |
1 |
sig_pri.c:985 pri_find_dchan: Span 1: D-chanannel anyway! |
6:19PM |
2 |
Ground Start ATA / VOIP Gateway |
3:37PM |
1 |
Dahdi 2.4.0 and Squeeze [SOLVED] |
3:36PM |
1 |
Polycom BLF |
2:26PM |
1 |
Page() bumps user out of a call |
1:44PM |
2 |
Dahdi 2.4.0 and Squeeze |
1:18PM |
1 |
sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel |
9:44AM |
2 |
sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway! |
|
Monday June 13 2011 |
Time | Replies | Subject |
6:44PM |
1 |
PAP2T provisioning via SRV record? |
6:19PM |
0 |
multiple asterisk on 1 machine or other idea for using multiple network connection |
5:57PM |
0 |
skinny and 7961 |
4:55PM |
5 |
No audio after a reinvite changing codec |
4:10PM |
1 |
call an external number for other server |
2:33PM |
1 |
AMI on hold events |
10:44AM |
3 |
asterisk queue 'ringall' stratagy |
10:04AM |
13 |
Cisco IP Phones and Skinny in asterisk |
9:34AM |
0 |
announce-frequency not respected |
8:39AM |
1 |
Communciation delay betwwn speakers |
|
Sunday June 12 2011 |
Time | Replies | Subject |
9:16PM |
3 |
Configuring Cisco Phones to register on Asterisk: The configuration files |
4:42PM |
2 |
A question about Caller ID |
|
Saturday June 11 2011 |
Time | Replies | Subject |
3:42PM |
0 |
vTiger component |
3:37PM |
1 |
Siemens gigaset as180 as a internal mobile extension |
3:07PM |
0 |
chan-mobile bug: bluetooth connection is disconnected immediately when the call hangup |
2:29PM |
6 |
TFTP to be installed in Linux same asterisk machine to be used with Cisco |
7:42AM |
1 |
Full SIP dial string |
|
Friday June 10 2011 |
Time | Replies | Subject |
9:26PM |
1 |
Queue not sending call to Agent |
8:49PM |
1 |
Incoming Call Recording |
6:31PM |
1 |
Request: please test modification to EWS calendar functionality |
6:27PM |
1 |
asterisk 1.8 PRI random call drop issue |
2:16PM |
0 |
(no subject) |
2:15PM |
2 |
AMI question |
11:00AM |
4 |
Connected Line ID |
9:32AM |
1 |
Asterisk issue or VoIP provider issue ? |
9:26AM |
2 |
How to remove asterisk ? |
6:07AM |
1 |
[FreePBX] Digium addons |
4:57AM |
0 |
Friday: Sipgate, Yate and Astricon |
|
Thursday June 9 2011 |
Time | Replies | Subject |
10:57PM |
1 |
Access Voicemail Asterisk 1.8 FreeBSD 8.2 |
9:32PM |
2 |
Permanent restart after upgrade |
8:10PM |
0 |
Insert name in SIP registry |
7:47PM |
0 |
Polycom 501 Settings/subscription expiry |
7:05PM |
1 |
Question about voip.ms service. |
6:53PM |
1 |
Fwd: Re: ControlPlayback's options |
5:40PM |
1 |
SIP/IAX guest access? |
3:10PM |
0 |
IRC canal |
12:27PM |
0 |
Asterisk, attended transfers and DTMF mode |
11:41AM |
0 |
how to tell if call is on hold |
2:00AM |
0 |
Change to pickups in Asterisk 1.8 - not working on local channels? |
|
Wednesday June 8 2011 |
Time | Replies | Subject |
10:48PM |
0 |
Question on how many phones |
10:13PM |
3 |
How asterisk use pri channel |
9:32PM |
5 |
LXC and Dahdi |
6:43PM |
2 |
No IVR listen at device end......SIP phone is working fine |
6:40PM |
1 |
CallerID issue |
6:26PM |
2 |
how asterisk work with VoIP trunk? |
6:16PM |
6 |
issues.asterisk.org/jira not working |
6:09PM |
1 |
Asterisk and Audiocodes PRI card |
5:44PM |
1 |
Interesting PRI issue |
4:00PM |
1 |
Update problem | CLI commands missing |
3:26PM |
0 |
call transfer back to a sourcing switch |
1:57PM |
3 |
Asterisk 1.8 broken MWI |
9:55AM |
0 |
Call queues on load-balanced asterisks |
7:00AM |
0 |
Hints problem - NAT problem? |
7:00AM |
1 |
Queue log in MySQL DB |
6:49AM |
1 |
Asterisk: BYE is received late |
6:09AM |
2 |
Looking for Email to Fax Solutions |
2:20AM |
1 |
After wiki.asterisk.org was upgraded my user no loger exists. |
1:17AM |
1 |
PRI hangup request, cause 18 |
|
Tuesday June 7 2011 |
Time | Replies | Subject |
11:44PM |
2 |
What is wrong in m |
9:12PM |
3 |
reload chan_dahdi.conf without disconnect active calls |
6:17PM |
3 |
why doesn't "s" accept incoming call |
5:04PM |
1 |
Asterisk 1.8 minimum modules/configuration |
4:52PM |
1 |
trunk between 2 servers |
12:30PM |
0 |
Call Conference and Call transfer and Voice mail settings |
11:44AM |
0 |
Refactor of CDR - Comments please. |
11:13AM |
0 |
Asterisk 1.6 - subscriptions. |
8:59AM |
3 |
Different callerid for different extensions |
8:47AM |
4 |
Connect intercom to Asterisk? |
8:24AM |
0 |
IPv6 and IPv4 NAT not working |
6:34AM |
0 |
chan_mobile one way comunication problem |
6:31AM |
4 |
how to know length of file in seconds |
5:53AM |
0 |
Is this feature or Bug of all Asterisk versions ? |
5:31AM |
1 |
How to get DTMF in Konference module in Asterisk |
3:02AM |
0 |
sccp problem |
2:12AM |
1 |
Pops & clicks at the end of sound files |
2:11AM |
1 |
tls/srtp: sip_xmit error: returned -2 |
1:19AM |
2 |
PRI issue its BUSY |
|
Monday June 6 2011 |
Time | Replies | Subject |
9:59PM |
0 |
half sip registration at 1.8.3 |
8:43PM |
2 |
asterisk 1.8 issue with polycom dialplan |
7:31PM |
0 |
Subject Change: Playback/background App over Network (was: AGI STREAM FILE not working?) |
4:59PM |
0 |
unsubscribe |
4:36PM |
0 |
DAHDI - skipping sound on voicemail |
3:55PM |
1 |
Asterisk GUI - the one from Diguim/Asterisk - issues on Asterisk 1.6x |
3:09PM |
0 |
Bridged Call |
2:07PM |
2 |
Asterisk Online Training |
10:57AM |
0 |
About Asterisk SIP NAT Config |
5:08AM |
2 |
issues.asterisk.org |
4:12AM |
4 |
AGI STREAM FILE not working? |
|
Sunday June 5 2011 |
Time | Replies | Subject |
6:30PM |
1 |
Asterisk users Calculation |
4:25PM |
0 |
DTMF issue in app_konference using with asterisk 1.8.3.2 |
4:18PM |
3 |
broken SVN asterisk 1.8 ? |
2:56PM |
1 |
asterisk 1.6 - 511 Command not permitted causing high CPU usage |
11:50AM |
0 |
Blind transfer issue on Asterisk 1.8.4.2 |
|
Saturday June 4 2011 |
Time | Replies | Subject |
6:05PM |
1 |
example sip.conf for csipsimple? |
6:30AM |
0 |
Obtain SIP From and To Tag for CDR |
|
Friday June 3 2011 |
Time | Replies | Subject |
1:55PM |
0 |
Queue base polycom custom ringtype |
9:25AM |
4 |
Voxbone numbers |
7:08AM |
1 |
MOH uploading is not working with 1.4 |
1:31AM |
0 |
chan_dahdi.c, dtmfmute, rtp.c |
|
Thursday June 2 2011 |
Time | Replies | Subject |
9:46PM |
0 |
ChannelRedirect |
7:37PM |
0 |
asterisk logger permission |
7:27PM |
0 |
asterisk-users Digest, Vol 83, Issue 3 |
7:07PM |
2 |
Asterisk 1.8.4.2 Now Available (Security Release) |
6:49PM |
0 |
Help launch the stackexchange (stackoverflow) telophony site |
6:46PM |
2 |
How to continue processing a context after a Hangup |
5:57PM |
3 |
Can I use phone line to recive faxes? |
3:03PM |
4 |
RealTime Queue Logging in 1.8 |
1:01PM |
3 |
benefits of asterisk 1.8 |
7:39AM |
1 |
Does anyone know about asterisk 1.10 |
5:55AM |
1 |
Three-way conference in Asterisk |
|
Wednesday June 1 2011 |
Time | Replies | Subject |
11:51PM |
1 |
Migration from Mantis to JIRA |
9:50PM |
2 |
Question about "null routing" calls to DIDs we don't handle |
6:31PM |
0 |
Moh on transfer call |
5:00PM |
0 |
Immediate 180 Response on Invite |
3:11PM |
1 |
Dahdi_genconfig - "Empty configuration -- no spans" |
9:11AM |
0 |
DBdeltree: Error deleting key from database |
6:06AM |
10 |
busy hangup HDLC Bad FCS (8) on Primary D-channel |