Thursday March 31 2011 |
Time | Replies | Subject |
7:40PM |
3 |
Huawei K3765 + Internet + SMS + Telephone |
6:36PM |
0 |
asterisk-users Digest, Vol 80, Issue 73 |
4:29PM |
1 |
Transfer feature dialing out after one digit |
2:46PM |
0 |
moh questions - quality and semi-streaming |
2:33PM |
0 |
TDM800P not detecting answer fast enough |
1:50PM |
1 |
Atcom/Rowetel IP04 Asterisk Appliance US Source |
1:39PM |
3 |
** to disconnect and make a new call |
12:52PM |
1 |
is downloads.asterisk.org down? |
12:34PM |
0 |
Asterisk 1.8 Dimensioning. |
|
Wednesday March 30 2011 |
Time | Replies | Subject |
7:34PM |
1 |
CDR Mysql adaptive Colum |
6:32PM |
5 |
chan_dahdi unknown dependency problem |
3:46PM |
1 |
dtmf_2833_1.pcap: what PCM codec? ulaw or alaw? |
2:17PM |
0 |
Updated: 10 Minutes: Asterisk PBX on Amazon EC2 |
2:15PM |
0 |
Asterisk 1.8.3.2 core dump chan_sip.c |
9:34AM |
0 |
asterisk and COLP |
2:28AM |
0 |
Discover when remote phone answers through IAX2 |
|
Tuesday March 29 2011 |
Time | Replies | Subject |
10:16PM |
4 |
Cisco IP Phones and Asterisk |
10:11PM |
1 |
E1 PRI configuration: DAHDI and LIBPRI |
5:52PM |
1 |
wrong from URI in options message |
2:05PM |
2 |
Debugging not going to log file |
1:19PM |
1 |
Get phone number from SIP header PAI |
10:57AM |
0 |
Asterisk Transfer Extensions |
9:38AM |
0 |
disconnecting destination channel |
|
Monday March 28 2011 |
Time | Replies | Subject |
5:27PM |
0 |
Is History-Info (RFC4244) supported ? |
4:19PM |
0 |
special control 16 |
4:04PM |
1 |
s extension not working |
1:26PM |
0 |
Asterisk SS7 error |
1:17PM |
0 |
Channel status with AMI originate calls |
12:55PM |
0 |
DAHDI, IAX2 and SIP considerations for Early-Media / Alerting |
12:54PM |
1 |
DTMF input while waiting in queue... |
12:41PM |
2 |
Variable. AMI and dialplan |
12:20PM |
8 |
asterisk and fail2ban |
11:44AM |
2 |
Dialplan help: hang up incoming call and call the number back |
11:21AM |
0 |
Queue(): how to Perform operations at the time of call sent to Queue member but not answered. |
10:22AM |
1 |
problems with blind transfer on GXP-2000 - Multi tenant asterisk !! |
8:23AM |
2 |
AMI redirect from Queue to MeetMe |
12:36AM |
8 |
CDR MYSQL missing field data |
|
Sunday March 27 2011 |
Time | Replies | Subject |
9:48PM |
1 |
spa8000 spa2102 t38 faxing |
7:54PM |
0 |
DAHDI custom ring cadences in 1.8.3 |
12:54PM |
0 |
Jabber/Jingle to Google users via local XMPP server |
|
Saturday March 26 2011 |
Time | Replies | Subject |
3:36PM |
1 |
Asterisks with ss7 problem |
9:50AM |
3 |
Checking status of a cell phone |
4:15AM |
2 |
pbx.c: We were unable to say the number |
|
Friday March 25 2011 |
Time | Replies | Subject |
11:05PM |
2 |
White papers or success cases to convince a customer? |
10:33PM |
1 |
Removing Polycom Transfer Softkey |
8:19PM |
0 |
Asterisk with FXO card only, no network |
7:40PM |
6 |
Back-to-back asterisk PRI issue |
6:49PM |
3 |
reload command not availeble asterisk 1.8.x |
6:08PM |
0 |
3com 3102 |
4:15PM |
1 |
checking dahdi channels |
2:36PM |
3 |
Why shouldn't I use 1.8? |
1:51PM |
2 |
asterisk 1.8 question |
6:56AM |
0 |
Today on VUC, Dan York on Google Voice + SIP |
|
Thursday March 24 2011 |
Time | Replies | Subject |
10:08PM |
2 |
Streaming Hold Music |
8:58PM |
5 |
Sox and bad quality when converting to 8 kHz |
7:19PM |
1 |
Linux Based Billing and CDR |
4:27PM |
0 |
Asterisk Tech Tips: Calling With Google Starts At Noon Central (30 minutes from now) |
3:58PM |
3 |
Filtering on from caller id |
1:46PM |
0 |
Mail list issues? |
11:18AM |
0 |
Digium TC400 cards query |
8:55AM |
1 |
Fwd: Asterisk 1.6.2.10 & CDR custom added field |
5:49AM |
1 |
SIP Invite and Asterisk API/Variable |
3:53AM |
4 |
Issues with Digum Repos / AsteriskNOW & Bad Packages |
|
Wednesday March 23 2011 |
Time | Replies | Subject |
11:18PM |
1 |
Forwarding XXXX to XXXX prevented. |
10:46PM |
1 |
spa8000 t38 faxing |
10:32PM |
1 |
Asterisk 1.8 Packages for Debian and Ubuntu |
10:05PM |
7 |
asking for some help |
9:03PM |
1 |
dahdi restart warning |
8:38PM |
1 |
OT: Have unused DID's; where to warehouse? |
8:27PM |
2 |
using ${EXTEN} with waitexten |
7:15PM |
1 |
Hang using Festival application |
6:36PM |
1 |
Asterisk Queue ACD when the queues and agents has the same priority/weight |
5:56PM |
1 |
Sangoma A102D wanpiple issue with dahdi |
4:58PM |
4 |
What is the most stable version of asterisk? |
10:40AM |
1 |
Asterisk using as a SIP client |
6:01AM |
2 |
Problems Extension with a Call In on Asterisk 1.6 |
|
Tuesday March 22 2011 |
Time | Replies | Subject |
8:06PM |
1 |
Sangoma wapipe installation error |
7:18PM |
0 |
conference room ideas |
4:57PM |
1 |
How to use Atxfer in AMI |
4:53PM |
3 |
Asterisk PRI back-to-back connect |
4:41PM |
3 |
Act! Integration |
2:54PM |
2 |
question on show channels |
1:05PM |
0 |
Asterisk 1.6.2.10 & CDR custom added field |
5:56AM |
4 |
Usage of lock in CDR |
1:17AM |
0 |
Play different voice-mail messages based oncertain conditions |
1:17AM |
0 |
Multi-Tenant Hosted PBX system with Resellerfunctionality |
1:05AM |
2 |
Play different voice-mail messages based on certain conditions |
|
Monday March 21 2011 |
Time | Replies | Subject |
11:49PM |
1 |
IAX Call token revisited |
11:38PM |
4 |
Multi-Tenant Hosted PBX system with Reseller functionality |
7:55PM |
2 |
record individual callers in confbridge |
5:28PM |
0 |
Reminder: Asterisk Tech Tips: Calling With Google on Thursday at 12PM CDT |
4:24PM |
7 |
Queue pause vs logged out ? |
4:22PM |
2 |
1.8 realtime - segfault |
11:45AM |
7 |
wrong time retrieved from system command |
9:38AM |
1 |
iax2 sound problem |
9:04AM |
0 |
Problem routing call to fax machine on DAHDI FXSport |
2:43AM |
0 |
Record individual callers in ConfBridge? |
|
Sunday March 20 2011 |
Time | Replies | Subject |
8:24PM |
1 |
why does "core show channels" on 1.8 not show the channel |
8:03AM |
5 |
Asterisk |
5:47AM |
0 |
switch statement in extensions.conf |
|
Saturday March 19 2011 |
Time | Replies | Subject |
8:38PM |
0 |
Single vendor for IMAP VM storage |
6:10AM |
1 |
Getting No Antenna bar when behind a NAT |
|
Friday March 18 2011 |
Time | Replies | Subject |
5:52PM |
7 |
One PRI card with 2 (or more) Telcos |
10:33AM |
0 |
DISA DTMF problem |
3:02AM |
2 |
Problem routing call to fax machine on DAHDI FXS port |
|
Thursday March 17 2011 |
Time | Replies | Subject |
5:35PM |
0 |
Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 Now Available (Security Releases) |
3:45PM |
1 |
Status of Queue Members |
3:44PM |
1 |
[1.6] Where to put "options wctdm opermode"? |
3:36PM |
0 |
blind transfer from AGI triggered call -> dropped |
2:18PM |
1 |
[1.6.2.5] Asterisk can't find MOH file |
1:57PM |
1 |
Getting the missed calls using Asterisk Manager |
1:12PM |
2 |
Answering machine detection for a second leg call generated by a call file. |
11:29AM |
0 |
Trying to turn off TLS.... |
11:24AM |
0 |
Asterisk not logging originating IP of a brute force attack |
10:59AM |
0 |
why asterisk sip dial procedure take long time |
9:59AM |
1 |
[1.6/Ubuntu] What packages for * + Dahdi? |
9:50AM |
0 |
Passing an argument to a macro within an Originatecommand |
5:37AM |
1 |
SIP registration DoS but no logs in messages |
2:07AM |
3 |
Call are established, but voices can't be heard |
|
Wednesday March 16 2011 |
Time | Replies | Subject |
11:00PM |
0 |
Asterisk 1.6.1.23, 1.6.1.17.1 and 1.8.3.1 Now Available (Security Releases) |
10:50PM |
0 |
AST-2011-004: |
10:50PM |
0 |
AST-2011-003: |
10:05PM |
0 |
Multiple Parking Lots Being Redirected to the Wrong Parking Lot |
7:39PM |
2 |
Trunk form asterisk1 to asterisk2 fails |
3:11PM |
0 |
Connecting Asterisk to Siemens Hipath, 3750 |
2:44PM |
0 |
(no subject) |
2:22PM |
1 |
Pushing info to a Polycom phone - from outside of the local network |
1:44PM |
1 |
Discover held channel? |
1:24PM |
0 |
Setting up 1.6.2.9 on Debian 6.0 Squeeze |
11:09AM |
2 |
chan_sip.c:3115 __sip_xmit of 0x108d33c0 (len 523) to xxx.xxx.xxx.xxx:0 returned -1: Invalid argument |
8:34AM |
0 |
SIP Call setup time monitoring in Asterisk |
7:38AM |
1 |
Extract Remote-Party-ID from incoming INVITE in dialplan |
|
Tuesday March 15 2011 |
Time | Replies | Subject |
10:19PM |
1 |
Multiple Asterisk |
9:52PM |
2 |
Is asterisk 1.8 stable version to upgrade from asterisk 1.6 on live server? |
9:35PM |
1 |
Passing an argument to a macro within an Originate command |
9:19PM |
1 |
signal amplified by asterisk |
8:59PM |
1 |
Auto Answer in manager |
6:06PM |
2 |
call file for page auto-call |
5:51PM |
0 |
FW: [newtech-1] Uncovering Spoken Phrases in Encrypted Voice over IP Conversations |
4:30PM |
1 |
[1.4] Failed callfile doesn't jump to "failed" extension |
2:11PM |
2 |
Some errors |
1:59PM |
1 |
call being rejected |
1:54PM |
4 |
[1.4] Asterisk doesn't hang up? |
1:48PM |
1 |
AMI Timestamps unit |
9:38AM |
1 |
Ast 1.8_CentOS5.5 with timerfd as timing source |
9:18AM |
1 |
How to send Hold invite from asterisk to other |
|
Monday March 14 2011 |
Time | Replies | Subject |
9:52PM |
0 |
New Webinar Series For Asterisk Users: Asterisk Tech-Tips |
7:19PM |
2 |
Asterisk -rx command not returning data - Version 1.4.33.1 |
5:17PM |
0 |
Anyone (else) need an asynchronous asterisk event->action framework ? |
3:01PM |
5 |
Asterisk 1.8 paging with ploycom |
8:07AM |
1 |
sip show channel and t.38 |
|
Saturday March 12 2011 |
Time | Replies | Subject |
7:11PM |
0 |
Hi all |
6:22PM |
1 |
Call flash transfer |
3:43PM |
0 |
Need to hire full-time linux/asterisk tech in North Atlanta |
2:27PM |
1 |
G.711.0 |
10:00AM |
1 |
Asterisk and PlayBack |
|
Friday March 11 2011 |
Time | Replies | Subject |
8:35PM |
1 |
Asterisk 1.8 AGI error ast_carefulwrite: write() returned error |
8:35PM |
1 |
Anyway to monitor SIP debug from originator and terminator separate of each other on two screens? |
7:02PM |
0 |
dnsmgr_lookup |
3:16PM |
2 |
How do you handle queues with AMI? |
12:17PM |
0 |
Exceptionally long voice queue length in asterisk 1.6.2 |
9:58AM |
1 |
Automatically unpause a paused queue memeber - bad idea? |
1:37AM |
0 |
regarding authentication with out challenge. |
|
Thursday March 10 2011 |
Time | Replies | Subject |
9:30PM |
4 |
Asterisk queues : command to run when a call is being bridged |
7:31PM |
2 |
Is H323 supported when installing Asterisk from Digium Yum repository? |
5:52PM |
1 |
Dialplan: funcionality testing |
4:53PM |
2 |
[1.4.21.2] Read() disconnects half-way through? |
3:02PM |
1 |
Metaswitch to Asterisk problems |
1:21PM |
0 |
console.conf.sample in 1.8.3 |
12:52PM |
1 |
Connecting Asterisk to Siemens Hipath 3750 |
10:55AM |
0 |
Experience with Phones, Asterisk and other pbx, cloud services, etc |
10:49AM |
1 |
Is this true for Asterisk as SBC? |
10:13AM |
1 |
problem with crashing Asterisk 1.8 |
7:50AM |
0 |
Display something on the top line of Polycom SPIP 3.1 screen |
2:28AM |
1 |
ChanSpy with alphanumeric SIP channels [1.6.2] |
2:04AM |
1 |
[1.8] Unable to Register: Registration denied because of contact ACL |
|
Wednesday March 9 2011 |
Time | Replies | Subject |
10:40PM |
5 |
One Way Audio |
9:53PM |
0 |
Anyone have BRI working with Asterisk 1.8, Latest DAHDI, LibPRI? |
9:37PM |
0 |
1.8 and no alsa input |
7:26PM |
3 |
Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) |
5:16PM |
1 |
asterisk 1.8 still need dahdi |
5:07PM |
1 |
No audio after 15 minutes on Asterisk 1.8 |
4:39PM |
1 |
HK DIDs |
4:09PM |
0 |
BLF, Directed pickup and Polycom 601 with SIP 3.1.6 |
1:13PM |
1 |
Asterisk pri card replecement |
11:14AM |
4 |
Multiple SIP endpoint registrations |
11:01AM |
7 |
[Opinion Request] SIP phones that work well with Asterisk |
9:17AM |
6 |
SIPAddHeader not working |
5:35AM |
4 |
doorphone? |
|
Tuesday March 8 2011 |
Time | Replies | Subject |
3:40PM |
3 |
Weird PRI error on an QUAD E1 span: Ring requested on unconfigured channel 255/255 |
12:22PM |
5 |
[1.4] Reading phone number the French way? |
11:31AM |
1 |
(fast) AGI and AMI synchronization ? |
10:05AM |
3 |
CDR and call transfers :) |
2:22AM |
1 |
TDM410P & dahdi driver == no lights? |
12:51AM |
2 |
Sip/google |
|
Monday March 7 2011 |
Time | Replies | Subject |
10:52PM |
2 |
Asterisk 1.6 MySQL Realtime fails to connect with working username and password. |
10:49PM |
1 |
Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze |
10:15PM |
3 |
1.8.3 - IAX - echo - jitterbuffer |
9:35PM |
1 |
[1.8.3] Error compiling Asterisk: __sync_fetch_and_add |
6:58PM |
1 |
Error loading module 'res_fax_digium.so' |
2:16PM |
2 |
Help on incoming |
10:04AM |
2 |
Cisco 7942G IP Phone firmware conversion from SCCP to SIP. |
5:10AM |
1 |
Mirrors in Australia? |
|
Sunday March 6 2011 |
Time | Replies | Subject |
11:14PM |
0 |
AstLinux 0.7.7 Release |
7:35PM |
1 |
Inadyn error |
5:59PM |
1 |
Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring? |
5:10PM |
0 |
Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. |
6:54AM |
1 |
fail2ban + asterisk |
3:47AM |
0 |
imsdroid on droidX to asterisk: No matching peer found |
3:06AM |
1 |
ignore this test |
2:51AM |
1 |
Early codec selection / negotiation |
|
Saturday March 5 2011 |
Time | Replies | Subject |
6:14PM |
4 |
Configuration for Multiple PRI cards |
5:26PM |
3 |
Prepaid Billing other than A2Billing |
7:28AM |
1 |
Asterisk, Sent accountcode between 2 asterisk |
7:25AM |
2 |
Help Asterisk / API / Perl |
3:31AM |
0 |
[announce] jkSMS |
2:49AM |
1 |
can anyone tell me how to set asterisk to record all phonecall |
12:42AM |
1 |
2 ip phones and 1 normal, can't neither send nor receive calls at all... |
|
Friday March 4 2011 |
Time | Replies | Subject |
6:49PM |
3 |
OT: OpenSIPS vs Kamailio -- which do you use and why? |
5:13PM |
4 |
server performance.... |
2:46PM |
1 |
GXW4004 - lines get stuck |
2:07PM |
2 |
Asterisk <-> Lync / Call Center Transfer / Refer |
11:37AM |
3 |
Gosub and 'h' (again?) |
8:30AM |
5 |
Loudness of recorded wav-audio |
5:50AM |
1 |
How is Libpri developped ? |
5:00AM |
0 |
DAHDI-Linux 2.4.1 and DAHDI-Tools 2.4.1 Released |
|
Thursday March 3 2011 |
Time | Replies | Subject |
5:46PM |
0 |
Friday #vuc at 12 Noon EST |
4:22PM |
4 |
SIP Provider Recommendation in US |
4:20PM |
1 |
Contact Directory on Polycom phones |
4:07PM |
1 |
[ASK] can't make call |
3:02PM |
2 |
Sangoma PCI vs PCI Express card |
2:42PM |
11 |
mySQL connection testing |
2:13PM |
6 |
[1.4] Forcing Asterisk/Zaptel to wait until callee answers? |
11:15AM |
2 |
VoIP Bandwidth Calculator |
8:04AM |
1 |
asterisk dump core when i try to record my name on the voicemail |
7:02AM |
3 |
Testing from where number is... |
5:59AM |
2 |
chan_skinny and Cisco 793X (7936) support in 1.8 |
5:20AM |
2 |
Converting MP3 files to wav for Asterisk |
1:17AM |
3 |
How do I find a phone numbers issued by Rogers? |
|
Wednesday March 2 2011 |
Time | Replies | Subject |
10:29PM |
2 |
how to use qualify times to route calls |
9:54PM |
1 |
Functionality Questions |
9:33PM |
3 |
Question on Asterisk 1.8 and Wait() |
2:46PM |
2 |
asterisk behind nat |
2:34PM |
1 |
Doubt about cdr on asterisk |
2:11PM |
0 |
Missing audio |
1:25PM |
2 |
[1.4] Comparing value of string with spaces? |
12:43PM |
0 |
Hardware recommendation needed |
12:39PM |
0 |
Asterisk 1.6 and windows RTC |
11:47AM |
1 |
Registering Cisco 7942G IP phone with Asterisk!. |
9:54AM |
1 |
[1.4] Call progress for Zaptel 1.4.3.1? |
8:57AM |
1 |
Asterisk 1.8 SIP realtime and NAT |
8:43AM |
1 |
GSM-Card for Asterisk / recommendation needed |
1:18AM |
0 |
Intermitent voice issues |
|
Tuesday March 1 2011 |
Time | Replies | Subject |
5:23PM |
1 |
Caller ID |
4:34PM |
3 |
records inbound and outbound calls |
12:08PM |
1 |
[zapata.conf] What is "wink"? |
11:24AM |
0 |
IRQ 0 on BRI card (B200E) |
10:08AM |
0 |
[1.4] Simple way to bridge two channels? |
9:20AM |
6 |
wav files are not playing asterisk |
9:04AM |
2 |
two questions regarding incoming call |
2:49AM |
1 |
Question about how traffic passes from phones |
2:10AM |
1 |
duplicate keys change from zaptel to dahdi 2.4.0 |
1:19AM |
3 |
TLS/SRTP calls go to circuit busy. |