| Saturday April 30 2011 |
| Time | Replies | Subject |
|---|
| 7:10PM |
1 |
Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend |
| 5:15PM |
1 |
dial multiple extensions |
| 2:29AM |
12 |
HA Asterisk |
| |
| Friday April 29 2011 |
| Time | Replies | Subject |
|---|
| 2:51PM |
0 |
Local channel scenario flushes CDR before dialplan end |
| 2:10PM |
1 |
SIP bad request |
| 11:03AM |
0 |
Hardware Server Configuration/8 or 4 port PRI Card |
| 2:57AM |
0 |
Friday on VUC: Jabber/XMPP |
| |
| Thursday April 28 2011 |
| Time | Replies | Subject |
|---|
| 8:30PM |
1 |
anybody out there sucessfully using gnugk? |
| 4:09PM |
1 |
odbc error - server is gone |
| 3:25PM |
9 |
How to create distortion, echo, and chopping sound in a SIP trunk? |
| |
| Wednesday April 27 2011 |
| Time | Replies | Subject |
|---|
| 7:34PM |
15 |
Discussion: Are we ready to leave 1.4 behind? |
| 7:06PM |
1 |
Echocancellation OSLEC vs MG2 ? |
| 6:04PM |
2 |
DHCP / DNS |
| 5:34PM |
2 |
asterisk practices |
| 5:16PM |
2 |
Asterisk, SIP & Firewalls |
| 3:41PM |
1 |
h323 with NAT |
| 1:47PM |
1 |
AGI WAIT FOR DIGIT - key press BEFORE command |
| 12:26PM |
1 |
Digium WCTDM24XXP DTMF CallerID |
| 11:29AM |
0 |
Konference module issue |
| 11:22AM |
2 |
how to know status of asterisk from php |
| 6:55AM |
0 |
Has anybody been able to install CDR-Stats all the way through? |
| 5:18AM |
0 |
Retaining original caller id |
| |
| Tuesday April 26 2011 |
| Time | Replies | Subject |
|---|
| 9:20PM |
0 |
Seattle WA Asterisk Users' Group |
| 5:04PM |
1 |
Asterisk 1.6.2.18 Now Available |
| 5:04PM |
0 |
Asterisk 1.4.41 Now Available |
| 5:01PM |
1 |
Asterisk 1.4.40.2 Now Available |
| 3:13PM |
0 |
siren sound |
| 2:32PM |
0 |
play audio file to destination SIP channel on attended call transfer |
| 2:13PM |
2 |
Password to be ecrypted? |
| 9:43AM |
7 |
Orginate not working well with PSTN lines |
| 8:14AM |
1 |
How does wrandom strategy works with Queue? |
| |
| Monday April 25 2011 |
| Time | Replies | Subject |
|---|
| 10:02PM |
3 |
PAP2T auto answer? |
| 9:30PM |
1 |
Transfer beep w/ Polycom phone |
| 8:10PM |
0 |
Registration problems - Vitelity |
| 4:36PM |
0 |
FILTER function and multiple ranges? |
| 3:17PM |
1 |
new confbridge |
| 1:38PM |
4 |
The new ConfBridge application is now in Asterisk Trunk! |
| 1:51AM |
2 |
(no subject) |
| |
| Sunday April 24 2011 |
| Time | Replies | Subject |
|---|
| 8:21PM |
1 |
Realtime and priority labels |
| 3:45PM |
2 |
Best modem for chan_datacard |
| |
| Saturday April 23 2011 |
| Time | Replies | Subject |
|---|
| 5:52PM |
1 |
ARA table definitions (1.8.*) |
| 4:20PM |
2 |
call files |
| 3:48PM |
2 |
DTMF not being sent ( RFC2833 ) |
| 12:56PM |
1 |
Warm Transfer in Asterisk |
| |
| Friday April 22 2011 |
| Time | Replies | Subject |
|---|
| 6:49PM |
0 |
nyc area pbx rfp 4000 extensions |
| 5:48PM |
0 |
Multi tenant Parking issue |
| 5:17PM |
0 |
WARNING T.30 ECM carrier not found |
| 5:13PM |
0 |
(no subject) |
| 3:02PM |
2 |
Cannot call to my server with SIP |
| 1:55PM |
0 |
question on register and dnsmgr_lookup |
| 12:45PM |
0 |
ZRTP SDK Source |
| 10:13AM |
0 |
T38 fax printer Windows client for asterisk 1.8 |
| 8:05AM |
7 |
Flite issue |
| 5:21AM |
0 |
Help Required---Problem in Installation without dahdi |
| |
| Thursday April 21 2011 |
| Time | Replies | Subject |
|---|
| 9:48PM |
0 |
Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 Now Available (Security Releases) |
| 9:40PM |
1 |
AST-2011-006: Asterisk Manager User Shell Access |
| 9:40PM |
0 |
AST-2011-005: File Descriptor Resource Exhaustion |
| 6:52PM |
0 |
Nationalprefix chan_dahdi option |
| 5:26PM |
3 |
missed call notification |
| 4:30PM |
1 |
IAX2 codec selection and video |
| 1:53PM |
1 |
Transcode ulaw/g722 problem |
| 12:12PM |
3 |
Asterisk Export Fax from Wave file |
| 11:20AM |
2 |
[asterisk-user] Can't get hostname on asterisk dialplan by ENV() |
| 9:42AM |
2 |
Nat=yes |
| 9:34AM |
0 |
Voicemail forward issue |
| |
| Wednesday April 20 2011 |
| Time | Replies | Subject |
|---|
| 9:32PM |
2 |
py-Asterisk or pyst? |
| 8:05PM |
3 |
VoiceMail to text mail |
| 7:40PM |
1 |
allowguest=yes, how? |
| 5:16PM |
1 |
asterisk log - "=======" extension not found? |
| 4:37PM |
1 |
[IAX] Everyone is busy/congested at this time (1:0/0/1) |
| 4:20PM |
2 |
Call files or AMI originate for mass outbound call |
| 4:07PM |
2 |
issue with installtion asterisk |
| 3:02PM |
1 |
dtmf payload type problem during faxing.. |
| 2:34PM |
0 |
1.8.x sip error 0.0.27.191:5060 returned -1: Invalid argument |
| 9:50AM |
2 |
No voice in MeetMe for SIP with |
| 2:41AM |
4 |
Configure IVR(Inbound and Outbound) |
| |
| Tuesday April 19 2011 |
| Time | Replies | Subject |
|---|
| 11:14PM |
0 |
IP Address Management / Open Source / IPAM |
| 5:09PM |
1 |
How to know how many calls are into hold by asterisk command |
| 4:23PM |
0 |
sterisk+SS7 Error: chan_dahdi.c: Unable to start PBX on DAHDI/288-1 |
| 4:09PM |
0 |
RTP and Signalling Dropping |
| 3:51PM |
0 |
chan_mobile: Dropping incompatible voice frame |
| 2:50PM |
1 |
chan_dahdi under bristuff 0.4.0-RC11 not building under Linux 10.04.2 LTS |
| 8:43AM |
1 |
ConfBridge and AGI |
| 6:41AM |
3 |
No voice in MeetMe for SIP with AGI_BACKGROUND |
| |
| Monday April 18 2011 |
| Time | Replies | Subject |
|---|
| 6:47PM |
1 |
core show channels consise in asterisk 1.8.3 |
| 6:12PM |
1 |
A101DE Sangoma Card in AsteriskNow 1.7.1 |
| 4:35PM |
0 |
canreinvite yes or no for PBX |
| 4:01PM |
1 |
Meetme Time Limit? |
| 1:46PM |
2 |
Asterisk unresponsive |
| 1:19PM |
1 |
Softphone IAX |
| 10:16AM |
3 |
No Internet, no asterisk |
| 10:06AM |
2 |
Call Center Reporting |
| 9:54AM |
2 |
Registrations stops after 403 FORBIDDEN |
| 7:16AM |
1 |
Asterisk, virendra bhati has invited you to open a Gmail account |
| 5:46AM |
0 |
[OT, Job] Senior Software Engineer for exciting, high-growth startup |
| |
| Sunday April 17 2011 |
| Time | Replies | Subject |
|---|
| 6:46PM |
1 |
Asterisk 1.8.3: Started but no SIP talking |
| |
| Saturday April 16 2011 |
| Time | Replies | Subject |
|---|
| 11:28PM |
4 |
Jabber / GTalk / hints |
| 11:13PM |
4 |
Jabber / facebook chat? |
| 9:58PM |
1 |
CDR & ARI Question |
| 2:17PM |
0 |
PADLOCK asterisk 1.6 |
| 12:41PM |
1 |
"chan_sip.c: No such host:" but I can resolve it from command line ? |
| 7:56AM |
5 |
Google Voice receiving call problem |
| 5:24AM |
3 |
any experience with cisco media gw with fax??? |
| 3:26AM |
0 |
Duplicate cdr records with channel local |
| |
| Friday April 15 2011 |
| Time | Replies | Subject |
|---|
| 11:00PM |
2 |
1.8.4-rc2: ReceiveFAX fails |
| 7:29PM |
1 |
Reach PSTN from another Asterisk |
| 5:35PM |
0 |
Hot to make call parking to Mult tenant |
| 4:50PM |
1 |
sangoma card rx/tx gain level |
| 12:33PM |
2 |
Good by asterisk 1.4? Please not. |
| 12:13PM |
2 |
If voice mail not found dialplan |
| 12:02PM |
5 |
Possible bug in Hangup() (Asterisk 1.4.x) |
| 11:39AM |
1 |
Friday April 15 at 12 Noon EDT |
| 10:40AM |
0 |
[OT] 802.11x roaming |
| 10:10AM |
0 |
How to get back park call |
| 7:58AM |
0 |
Would a job posting be ok for this list? |
| 7:39AM |
3 |
sip error logging |
| |
| Thursday April 14 2011 |
| Time | Replies | Subject |
|---|
| 10:03PM |
0 |
Followme() and variables |
| 8:46PM |
1 |
Existing Asterisk 1.8 upgrade with new release |
| 3:48PM |
0 |
Asterisk modifies from header |
| 3:22PM |
1 |
Microsoft Lync server and Asterisk access |
| 12:51PM |
1 |
setting sip headers when using call files |
| 5:22AM |
1 |
Processing sip messages |
| 12:46AM |
1 |
Asterisk port 5000 open |
| |
| Wednesday April 13 2011 |
| Time | Replies | Subject |
|---|
| 8:20PM |
0 |
Applet based softphone for Asterisk |
| 8:00PM |
1 |
Safe to upgrade to Centos 5.6 now ??? |
| 5:36PM |
0 |
Fw: SIP Trunk send DID or DNIS information |
| 4:16PM |
1 |
Asterisk Tech Tips: Cookin' with Asterisk |
| 2:50PM |
1 |
Asterisk thread limit |
| 2:14PM |
1 |
T38 fax detection using g729 |
| 2:08PM |
2 |
Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card |
| 1:48PM |
2 |
Problems With DAHDI on Ubuntu |
| 1:08PM |
4 |
AGI and forking |
| 12:08PM |
1 |
Fwd: Re: Asterisk as a Condo door opener/intercom |
| 9:27AM |
1 |
How to know extensions status ??? |
| 9:01AM |
4 |
[OT] Yealink Phones |
| 8:57AM |
0 |
Poor call quality - line drop, chopping sound, like robotic voice, Both party could not hear caller voice |
| 8:48AM |
0 |
0018818: [patch] Crashing when using local channels and realtime on asterisk 1.8.3-rc2 |
| 8:15AM |
11 |
Realtime SIP & peer status |
| 5:13AM |
1 |
Aastra 480i & Asterisk 1.8.3.2: No musiconhold |
| |
| Tuesday April 12 2011 |
| Time | Replies | Subject |
|---|
| 11:30PM |
0 |
Debugging DTMF Detection |
| 11:08PM |
0 |
Problem with Swift app and escape digits |
| 8:31PM |
4 |
Basic queue question |
| 5:14PM |
0 |
From CDR to CEL |
| 2:42PM |
1 |
CEL Logging to MySQL - Please Test |
| 2:24PM |
0 |
Authentication failure |
| 1:37PM |
1 |
Queue(): How to know Estimated wait time for caller in advance |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 6:01AM |
0 |
No subject |
| 3:37AM |
1 |
Templates |
| 1:42AM |
1 |
Poor call quality – line drop, chopping sound, like robotic voice, Both party could not hear caller voice |
| 12:32AM |
0 |
Features.conf - Blind Transfer |
| |
| Monday April 11 2011 |
| Time | Replies | Subject |
|---|
| 10:38PM |
0 |
DAHDI-Linux 2.4.1.2 Released |
| 8:11PM |
2 |
Voicemail to email issue |
| 5:56PM |
0 |
"Wait for leader" allows crosstalk between participants |
| 5:52PM |
1 |
How to know the SIP status |
| 2:44PM |
7 |
Asterisk kernel CONFIG_HZ=1000 |
| 2:28PM |
1 |
voicemail odbc "Length is ....." |
| 2:26PM |
1 |
Asterisk codec negotiation and canreinvite=no |
| 2:04PM |
1 |
Require dialplan |
| 1:50PM |
0 |
Problem with E1 (ISDN) + DTMF |
| 1:43PM |
2 |
Asterisk-Asterisk E1 connection |
| 11:24AM |
1 |
Asterisk MOH not working with Elastix asterisk 1.6.2.18 |
| 10:25AM |
0 |
update CDR fields after Queue |
| 9:44AM |
1 |
Unable to negotiate codec with iax |
| 5:07AM |
6 |
Variable stripping/removing part of string |
| 4:37AM |
3 |
changing port 5060 to 5061 |
| |
| Sunday April 10 2011 |
| Time | Replies | Subject |
|---|
| 4:37PM |
1 |
Asterisk as a Condo door opener/intercom |
| 1:32PM |
1 |
AsteriskNow updated to Centos 5.6 and DAHDI doesn't work |
| |
| Saturday April 9 2011 |
| Time | Replies | Subject |
|---|
| 4:15PM |
4 |
Asterisk FOP |
| 2:52PM |
1 |
Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension? |
| 8:31AM |
1 |
asterisk-users Digest, Vol 81, Issue 27 |
| |
| Friday April 8 2011 |
| Time | Replies | Subject |
|---|
| 8:35PM |
2 |
Call Recording using MixMonitor - close, but would like some more words of wisdom. |
| 6:56PM |
1 |
Documentation for Asterisk AMI Events? |
| 6:13PM |
9 |
send voicemail to multiple emails |
| 3:07PM |
0 |
Any PHP Ming + for Asterisk guides, tutorial, how-to anywhere? |
| 2:48PM |
4 |
IAX2/0.0.29.199 |
| 1:56PM |
1 |
Maniuplate callerID based off of callerID |
| 1:43PM |
0 |
User registration failure bug ? |
| 9:57AM |
6 |
Variable inheritance with dialplan command Originate |
| 9:40AM |
0 |
asterisk-users Digest, Vol 81, Issue 21 |
| 9:11AM |
1 |
CRC Zaptel.conf |
| 7:11AM |
0 |
488 error in T38 Gatewaying in Asterisk 1.8 with patch 13405 |
| 6:51AM |
2 |
MOH not working |
| |
| Thursday April 7 2011 |
| Time | Replies | Subject |
|---|
| 9:53PM |
4 |
Occasional call from "asterisk" |
| 9:26PM |
1 |
Any way to temporarily disable a registered SIP PEER in Asterisk? |
| 6:56PM |
1 |
asterisk login to voicemail |
| 4:18PM |
1 |
MOH on DAHDI PRI Channels |
| 4:02PM |
3 |
No ringback even though progressinband=yes is set |
| 2:58PM |
0 |
Asterisk 1.8.x Skips DTMF Digits on a First DAHDI Initiated Call |
| 1:34PM |
0 |
AgentCallbackLogin slow in Asterisk 1.4 |
| 11:28AM |
1 |
Compiling asterisk using NDK build |
| 9:24AM |
4 |
asterisk SIP MESSAGE method support |
| 8:00AM |
2 |
Asterisk Avaya SIP Trunking One Way Audio |
| |
| Wednesday April 6 2011 |
| Time | Replies | Subject |
|---|
| 8:59PM |
2 |
asterisk meetme invalid extension |
| 8:12PM |
0 |
Options for DS3 to SIP |
| 7:53PM |
2 |
realtime mysql for 1.8 |
| 6:11PM |
1 |
MWI not working on most ATAs in Asterisk 1.6.2.17 |
| 5:30PM |
0 |
Problems with woomera (ISDN BRI) and playback app: Dropping incompatible voice frame |
| 4:46PM |
2 |
voicemail call back loop |
| 2:49PM |
1 |
Question About Codecs |
| 1:45PM |
3 |
BRI Configuration help me |
| 10:54AM |
4 |
Call recording - methodology |
| 7:38AM |
1 |
Call duration problem or maybe calls not hanging up problem |
| 1:10AM |
11 |
Asterisk 1.8.3 |
| |
| Tuesday April 5 2011 |
| Time | Replies | Subject |
|---|
| 8:54PM |
5 |
IAS trunk error AES encryption disabled. Install OpenSSL. |
| 8:39PM |
0 |
minmessage / maxsilence in voicemail.conf |
| 7:45PM |
1 |
asterisk-users Digest, Vol 81, Issue 12 |
| 6:52PM |
2 |
dahdi and linux-2.6.38 |
| 6:31PM |
2 |
Iptables configuration to handle brute, force registrations? |
| 5:54PM |
7 |
asterisk hints |
| 5:03PM |
2 |
agi create mailbox |
| 4:50PM |
4 |
agi voicemail callback |
| 4:35PM |
1 |
allpage issu on asterisk 1.8.3.x |
| 2:36PM |
2 |
Vestec for Asterisk |
| 2:27PM |
1 |
Iptables configuration to handle brute force registrations? |
| 8:31AM |
2 |
Asterisk 1.8 and new the command: exten => _X., 4, Wait, 2 |
| 7:15AM |
1 |
Asterisk => Request a Code |
| 2:08AM |
0 |
dialplan is not finding my number asterisk1.8.3 |
| 1:00AM |
1 |
Number Conversion |
| |
| Monday April 4 2011 |
| Time | Replies | Subject |
|---|
| 9:08PM |
1 |
Read VoiceMail direct |
| 7:51PM |
2 |
WARNING chan_sip.c:3115 __sip_xmit |
| 7:20PM |
4 |
dialplan is not finding my number asterisk 1.8.3 |
| 5:08PM |
1 |
MeetMe headache |
| 4:39PM |
0 |
DAHDI-Linux 2.4.1.1 Released |
| 4:29PM |
2 |
call-limit bypass |
| 3:13PM |
1 |
SIP register and contact header |
| 2:27PM |
1 |
how to check if the call is using t38 except in the sip packets |
| 2:09PM |
3 |
Dialplan matching |
| 2:00PM |
2 |
call forwarding |
| 1:36PM |
2 |
Load Asterisk Module with parameters? |
| 1:04PM |
1 |
Asterisk crashes on high IO load |
| 11:58AM |
3 |
CDR fields not being written from "h" extension after "Dial" command completes. |
| 10:34AM |
0 |
Finding out asterisk settings from console |
| 10:23AM |
0 |
SIP channel able to add codecs once up and running? |
| |
| Sunday April 3 2011 |
| Time | Replies | Subject |
|---|
| 6:33PM |
1 |
From 1.4 to 1.8: stdexten issue |
| 4:01PM |
1 |
Asterisk 1.6 => No sound/voice when i redirect the call |
| 12:56PM |
1 |
hello |
| 10:28AM |
0 |
[DIGIUM FAX] HANGUP problem |
| 6:54AM |
1 |
Top posting - there is no rule. |
| |
| Saturday April 2 2011 |
| Time | Replies | Subject |
|---|
| 6:03PM |
2 |
Is it possible to dial an automated message when Asterisk receives an email? |
| 2:09PM |
0 |
tarnsfer automatically |
| 8:46AM |
1 |
Registration from '"000000" x 1000 |
| 8:36AM |
0 |
automixmon output file location and exec command options |
| 6:11AM |
1 |
Problem getting TDM400P clone card to go off-hook and dial |
| |
| Friday April 1 2011 |
| Time | Replies | Subject |
|---|
| 9:25PM |
1 |
Polycom 501 alternate |
| 7:55PM |
1 |
codec_dahdi find_transcoders: Failed to open /dev/dahdi/transcode |
| 3:00PM |
1 |
Android tablet voip? |
| 2:52PM |
1 |
call parking issues in asterisk 1.6.2.16.2 |
| 2:38PM |
0 |
Digium launches flying phone-phone |
| 2:33PM |
2 |
Can gtalk.conf work with multiple GoogleVoice numbers? |
| 12:36PM |
2 |
BRI detection |
| 11:57AM |
6 |
Best Scripting Language |
| 11:28AM |
0 |
Double ## Feature to make another call during current call |
| 11:04AM |
2 |
Fax |
| 10:52AM |
0 |
Incoming SRTP call not working with Bria iPhone Edition |
| 8:25AM |
1 |
The SIP channel driver - I'm giving up. |
| 7:39AM |
1 |
Hold problem with Queue |
| 7:19AM |
0 |
OneAPI / ParlayX |