| 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 |