Friday December 31 2010 |
Time | Replies | Subject |
2:59AM |
2 |
Base memory usage |
|
Thursday December 30 2010 |
Time | Replies | Subject |
9:58PM |
0 |
Users of CEL Please comment on Bug |
8:31PM |
1 |
Find media and sip endpoints IP address durring "h" extension |
6:29PM |
1 |
VUC; Friday December 31st - 2010: The Year in VoIP |
9:10AM |
1 |
Force different codecs on call base |
7:36AM |
1 |
Usage Reports |
4:55AM |
4 |
call is not going to Voicemail with "1,n" |
4:36AM |
0 |
Music Jukebox and IVR voice chat |
|
Wednesday December 29 2010 |
Time | Replies | Subject |
11:15PM |
2 |
GotoIf CALLERID(num) |
11:16AM |
2 |
Log and forward calls to cellphone? |
8:16AM |
0 |
Trixbox CE for Multi Tenant |
|
Tuesday December 28 2010 |
Time | Replies | Subject |
8:01PM |
1 |
Replacing digital pri card |
4:33PM |
1 |
Sangoma U100 failing every Monday - USB port problem or Wanrouter issue? |
1:32PM |
1 |
Queue announce parameter problem |
12:10PM |
1 |
How to reload queue on the fly? |
6:26AM |
1 |
OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact |
1:42AM |
0 |
How to use google voice for voicemail transcription |
|
Monday December 27 2010 |
Time | Replies | Subject |
9:40PM |
2 |
Panasonic KX-TGP500 w/Asterisk |
6:37PM |
1 |
Asterisk 1.4.38 - unknown signalling bri_cpe |
6:08PM |
6 |
Using SIP stack within Asterisk to reboot phones - Possible? |
5:10PM |
0 |
CEL and custom values. |
1:02PM |
1 |
malformed SIP / routing issue |
12:05PM |
1 |
G729a and G729 interoperability |
11:05AM |
1 |
Queue Member relationship and AstDB |
7:25AM |
4 |
anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone |
|
Sunday December 26 2010 |
Time | Replies | Subject |
11:01PM |
1 |
Asterisk 1.8 Realtime Queue not working |
1:08AM |
0 |
Which GSM-Modem to build a Gateway? |
|
Saturday December 25 2010 |
Time | Replies | Subject |
11:58PM |
2 |
sip.conf, realtime, and LDAP |
11:04PM |
2 |
sip attack.. fail2ban not stopping attack |
11:01PM |
1 |
load balance with 2 wan connections |
5:28PM |
1 |
asterisk realtime & calling sip users |
2:49PM |
1 |
Remote VOIP/SIP Phones through two routers |
1:31PM |
2 |
Agents login |
|
Friday December 24 2010 |
Time | Replies | Subject |
2:40PM |
1 |
One way crappy audio in iax call - Asterisk 1.6.2.15 |
1:36PM |
1 |
live audio stream in asterisk |
12:30PM |
5 |
Moving asterisk from one network to another. |
12:22PM |
0 |
Prepaid Billing for Asterisk and Gnugk |
12:20PM |
0 |
Cisco IP Phones and AVAYA IP Phones: Provisioning the profile |
10:12AM |
5 |
SRTP unprotect: authentication failure |
8:27AM |
0 |
Today at 12 Noon EST |
|
Thursday December 23 2010 |
Time | Replies | Subject |
9:07PM |
1 |
Zombie DAHDI FXO channels |
6:52PM |
3 |
Asterisk 1.8 and Realtime |
4:33PM |
0 |
Asterisk 1.6 iax auth rsa failed with policie not found |
3:28PM |
0 |
Panasonic trunk asterisk over h323 |
1:33PM |
0 |
MOH RBT problem |
10:51AM |
1 |
OT - Alcatel OXE IP trunking licence price |
7:41AM |
2 |
Forward voicemail to group of people |
5:41AM |
0 |
Asterisk handling multiple simultaneous calls for IVR |
3:06AM |
0 |
Maximum retries exceeded |
|
Wednesday December 22 2010 |
Time | Replies | Subject |
9:23PM |
0 |
Asterisk 1.8.1.1 Multiple Parking Lots |
6:31PM |
1 |
Siemens OpenStage phones and Asterisk |
5:51PM |
0 |
CDR on MySQL |
5:42PM |
8 |
Possible Bug (Include ${HANGUPCAUSE} in CDR) |
3:14PM |
1 |
How to list used extensions + assign extension to a roaming phone |
3:02PM |
0 |
setting up callerid |
2:58PM |
5 |
* 1.8: cannot load g729 free codec (on 1.4 it worked!) |
2:49PM |
1 |
dahdi-channels.conf for Digium TDM2400 |
2:23PM |
2 |
Vacancy - Asterisk MySQL Support Engineer 45K South London |
2:22PM |
0 |
Include ${HANGUPCAUSE} in CDR |
12:44PM |
4 |
Asterisk hangs up call after 20s |
11:50AM |
2 |
Maximum E1 Ports on Asterisk ? |
10:21AM |
1 |
Wise selecting of outgoing channel |
10:10AM |
1 |
callerid and user on voicemail |
1:51AM |
1 |
Asterisk as a caller ID |
12:58AM |
1 |
Simplifying dial-plan |
|
Tuesday December 21 2010 |
Time | Replies | Subject |
8:20PM |
2 |
What is equivalent function to "mv" command in php for Asterisk Spool directory usage? |
12:51PM |
0 |
Friend/user/peer in plain English? |
11:39AM |
1 |
MeetMe -> ConfBridge: hint not working |
10:49AM |
1 |
SOLVED: Re: Setting `userfield` from within a callfile |
10:47AM |
1 |
app_voicemail.c how to enable debugging? |
|
Monday December 20 2010 |
Time | Replies | Subject |
10:16PM |
0 |
What's up? |
10:02PM |
3 |
cdr_mysql stopped working |
9:19PM |
0 |
Upgrading DAHDI and Asterisk |
8:39PM |
4 |
Asterisk 1.6 produces *many* zombie processes on Debian. |
5:46PM |
2 |
SIP 420 |
5:35PM |
2 |
Unexpected dialplan match |
4:33PM |
2 |
Setting `userfield` from within a callfile |
12:50PM |
4 |
dahdi issue on digium AEX800 |
10:48AM |
5 |
start services automatically |
9:41AM |
3 |
Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE. |
9:19AM |
0 |
Asterisk Clustering and DUNDi J Richardson whitepaper |
8:42AM |
5 |
DIALSTATUS on CANCEL |
7:16AM |
0 |
deadagi on v1.4.xx |
1:54AM |
1 |
(no subject) |
|
Sunday December 19 2010 |
Time | Replies | Subject |
2:02PM |
1 |
Problem with AASTRA phone of NO SERVICE |
11:23AM |
1 |
In which version is eventfilter working? |
12:03AM |
2 |
Specifying DID for outbound calls |
|
Saturday December 18 2010 |
Time | Replies | Subject |
7:07PM |
1 |
How to install the new cdr-stats? |
11:58AM |
1 |
Asterisk and Alcatel digital phone's |
|
Friday December 17 2010 |
Time | Replies | Subject |
5:17PM |
1 |
How to block everyone outside of our lan |
3:49PM |
1 |
transfer from sip to dahdi, connects caller to MOH stream and not target |
3:40PM |
10 |
Wireless Desktop VoIP Phone? |
3:08PM |
2 |
Asterisk Freeze In 1.4 realtime |
2:45PM |
5 |
Attack problem |
11:22AM |
1 |
Contradiction between 2 AMI actions QueueSummary and Queuestatus |
11:17AM |
1 |
HA: what is missing to keep ongoing calls during failover ? |
10:57AM |
2 |
Voicemail Forwarding |
10:21AM |
8 |
Ported Asterisk in Android |
9:14AM |
3 |
dahdi show channels / how to get the call duration for active calls? |
9:07AM |
0 |
Asterisk and Tandberg VCS |
7:14AM |
1 |
Pass DTMF to IVR gateway through SIP phone conferencing. |
6:00AM |
0 |
start music on hold coredump |
4:33AM |
3 |
How to find , internal, external inbound or outbound |
|
Thursday December 16 2010 |
Time | Replies | Subject |
8:16PM |
1 |
PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI |
4:33PM |
0 |
app_voicemail: "3" for advanced options does not have an effect _while_ the vm-message is played |
4:17PM |
0 |
Dialplan not found |
1:50PM |
0 |
chan_iax2.c handle_call_token: Call rejected, CallToken Support required |
10:54AM |
6 |
Call sip:user@domain.com? |
9:52AM |
0 |
Junghanns OctoBRI - Invalid sync priority |
7:37AM |
0 |
No subject |
|
Wednesday December 15 2010 |
Time | Replies | Subject |
11:07PM |
2 |
Echo Cancellation Problem - Invalid Argument?!? |
9:27PM |
0 |
Asterisk 1.8.1.1 Now Available |
9:23PM |
2 |
Recommendation for a Linux based SCADA |
7:53PM |
2 |
res_odbc dependeny issue |
5:46PM |
1 |
Asterisk 1.8 with web-meetme crash |
2:46PM |
5 |
Which version to use: 1.4 or 1.6 or 1.8 |
2:33PM |
1 |
Transferring problem within Queues |
2:19PM |
0 |
Asterisk Community Mailing Lists Service Disruption |
1:38PM |
2 |
Two asterisk servers, two different service providers |
|
Tuesday December 14 2010 |
Time | Replies | Subject |
6:56PM |
0 |
503 Server error response for REGISTER request (Asterisk v1.6.0.5) |
3:57PM |
1 |
Anyone know how to receive partial (interrupted) faxes with app_fax? |
3:56PM |
1 |
Asterisk + VOSP account working configuration? |
3:49PM |
1 |
Announce Remaining Call Time |
2:19PM |
3 |
Converting asterisk h264 recordings |
12:58PM |
0 |
Debug messages. |
12:09PM |
0 |
Asterisk dynamic span error |
9:20AM |
1 |
Asterisk on smartphones ? |
6:58AM |
6 |
Asterisk and Dahdi ON Amazon EC2 |
|
Monday December 13 2010 |
Time | Replies | Subject |
9:08PM |
0 |
What to check for when there are sound interference using SIP channels only? standard debug methods? |
8:49PM |
0 |
asterisk-users Digest, Vol 77, Issue 27 |
6:28PM |
1 |
Configuring server to call SIP numbers on the Net? |
6:00PM |
3 |
Voice mail distribution - missing messages |
3:44PM |
1 |
Application to test STUN + broadband? |
1:06PM |
0 |
Problems after upgrading libpri from 1.4.11.2 to 1.4.11.5 |
11:48AM |
4 |
Mail Integration |
10:47AM |
0 |
Asterisk for Testing PC-1.5 MTA |
8:16AM |
2 |
Asterisk 1.6.2.10 & video |
2:04AM |
2 |
1.8.1: playing imaginary sound files |
|
Sunday December 12 2010 |
Time | Replies | Subject |
2:24PM |
1 |
Atcom IP-4B ISDN IP PBX? |
4:28AM |
0 |
Transfer (sip -> dahdi) results in moh for dahdi |
3:26AM |
0 |
pickup problem |
|
Saturday December 11 2010 |
Time | Replies | Subject |
1:40PM |
1 |
No more room in scheduler |
8:06AM |
2 |
Why does "sip show peers" show my router/gateway address as the client IP address? |
|
Friday December 10 2010 |
Time | Replies | Subject |
6:47PM |
1 |
1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra |
5:28PM |
0 |
HP DL360G7 with T1 card(s) |
4:45PM |
1 |
UDP buffer overflows? |
12:06PM |
0 |
Multiple digits as takecall in followme application |
6:32AM |
1 |
Audio ports |
|
Thursday December 9 2010 |
Time | Replies | Subject |
9:31PM |
1 |
(Fwd) Re: Configuring Softphone |
1:57PM |
4 |
Asterisk SIP attacks and sshguard |
5:46AM |
1 |
Load testing SFA |
|
Wednesday December 8 2010 |
Time | Replies | Subject |
9:45PM |
0 |
DAHDI channels not hanging up FXO |
8:23PM |
0 |
Asterisk 1.8.1 Now Available |
8:23PM |
0 |
Asterisk 1.6.2.15 Now Available |
8:22PM |
0 |
Asterisk 1.4.38 Now Available |
7:59PM |
2 |
filtering AMI Event: RTCPSent |
6:56PM |
0 |
Asterisk 1.8 debian packages? |
6:48PM |
2 |
[headset/mic] Volume too low + echo in * (Gilles) |
4:45PM |
5 |
How to quickly move on to Dahdi channels when SIP provider fails? |
3:52PM |
1 |
QUEUE_PRIO |
2:31PM |
1 |
Asterisk segfault in dmesg |
2:06PM |
3 |
[POTS/BRI] Neutral comparisons of PCI vs. box? |
11:36AM |
1 |
Video codecs: H263 & H264 |
9:43AM |
1 |
Error building network library on OpenSolaris and 1.8.1-rc1 |
2:31AM |
3 |
Configuring Softphone |
12:15AM |
1 |
debug audio or channel |
|
Tuesday December 7 2010 |
Time | Replies | Subject |
5:38PM |
3 |
Dahdi issue with Asterisk 1.8.0 |
3:55PM |
1 |
[headset/mic] Volume too low + echo in * |
2:43PM |
3 |
Snom (vs Polycom) - provisioning |
12:53PM |
1 |
No MOH with parked call |
9:42AM |
1 |
'Bookmarking' a place in a sound file |
9:30AM |
0 |
DUNDi and Lua dialplan |
2:38AM |
1 |
no audio on end-point when call is connected/bridged via PBX |
|
Monday December 6 2010 |
Time | Replies | Subject |
11:48PM |
1 |
Execute DialPlan Context without Answer App |
10:49PM |
0 |
Mac OS X 10.4 support |
3:41PM |
1 |
[3102] How to rewrite CID name + number? |
2:23PM |
1 |
Asterisk 1.6.2.10 video call |
1:10PM |
0 |
Fw: Sip Hangup after critical packet SIP DEBUG attached |
12:08PM |
0 |
Sip Hangup after critical packet |
11:37AM |
1 |
Callee side blind transfer is failing in 1.8 |
10:45AM |
1 |
Linkedid member in Channel structure on 1.8 |
12:19AM |
1 |
no audio |
|
Sunday December 5 2010 |
Time | Replies | Subject |
6:36PM |
2 |
HA8 cards and RED alarm |
|
Saturday December 4 2010 |
Time | Replies | Subject |
6:37PM |
0 |
ISDN TE/NT Mode |
3:15PM |
1 |
Error messages with chan_dahdi |
1:02AM |
3 |
Polycom Park by EFK |
|
Friday December 3 2010 |
Time | Replies | Subject |
8:58PM |
1 |
Issue with MOH - Asterisk 1.4.17 |
6:36PM |
1 |
What are the minimal permissions required to read the PeerStatus and Registry events? |
4:19PM |
5 |
Asterisk 1.6 (Web-meetme) |
12:57PM |
0 |
Caller id is not proper when I do call forward |
9:32AM |
0 |
OT - Gigaset and PoE |
5:34AM |
1 |
Sharing Fail2ban data |
12:56AM |
3 |
Asterisk error - 1.6.2 SVN - voicemail files "corrupted" |
12:41AM |
0 |
Astmanproxy on FreeBSD 8.1 |
12:09AM |
2 |
Version compatibility question... |
|
Thursday December 2 2010 |
Time | Replies | Subject |
9:32PM |
1 |
MP3s not decoding properly for MusicOnHold. |
9:06PM |
2 |
Asterisk ports |
8:03PM |
4 |
DAHDI on VMWARE |
5:13PM |
3 |
+ on Caller-ID |
4:58PM |
5 |
alarm POTS lines |
2:19PM |
5 |
Push central phone book to phones |
8:44AM |
1 |
rotate of logfiles |
8:28AM |
0 |
[OT] Any comments on Comcast and Level 3 story this week? |
8:02AM |
0 |
Dahdi 2.4.0 and unplugged spans [SOLVED] |
|
Wednesday December 1 2010 |
Time | Replies | Subject |
7:05PM |
1 |
codec_g729a implicated in file descriptor buildup |
6:33PM |
3 |
Abandon events in cdr |
5:44PM |
6 |
Issues with 1.8 and BlindTransfer |
5:34PM |
0 |
Problem with Queue_log and CDR. |
3:56PM |
1 |
Dahdi 2.4.0 and unplugged spans |
3:30PM |
0 |
MixMonitor not recording in version 1.8 |
12:22PM |
1 |
Reasons of OriginateResponse |
12:15PM |
2 |
Dahdi on Realtime. |
12:10PM |
0 |
<solved!> Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented) |
10:46AM |
0 |
Broadsoft-like BLF List URI ? |
2:04AM |
1 |
Trying to configure a SIP software phone |
12:55AM |
1 |
Zaptel / Asterisk on Solaris |
12:34AM |
4 |
Asterisk with MySQL Cluster |