asterisk users - Sep 2011

Friday September 30 2011
TimeRepliesSubject
7:21PM 5 USA Did required
4:37PM 0 Strange Asterisk HA Behaviour
2:50PM 3 asterisk hardware
9:16AM 0 SIPit 29 in Monaco - interoperability by hard work
8:56AM 8 Core show translation > 4000ms
7:16AM 7 invite authentication error !?
6:14AM 0 ODBC connection not connected at 1st call.
4:09AM 0 OUTBOUND and INBOUND routes
 
Thursday September 29 2011
TimeRepliesSubject
9:52PM 9 [asterik-users] Installing PRI card
8:51PM 2 Features not working
5:22PM 1 Asterisk/DAHDI with "Dynamic T1s"
4:22PM 5 record calls of specific agnets
2:44PM 16 No Bull Service Providers
1:50AM 0 I can't figure out how to redirect a call to a trunk.
 
Wednesday September 28 2011
TimeRepliesSubject
11:52PM 0 res_ODBC and failover (Bryant Zimmerman)
10:59PM 0 Increasing the fxorxgain and fxotxgain for the hardware of the digium card
10:33PM 1 Anybody using BinFone Telecom?
8:30PM 0 res_ODBC and failover
6:01PM 0 FreeTDS and MS-SQL with Asterisk RealTime
5:59PM 13 Limit outbond calls duration to 1 minute
3:01PM 0 Asterisk Realtime SIP : vmexten
9:11AM 14 PSTN connectivity
2:47AM 1 C wrapper for AMI?
1:44AM 5 Receiving musinc on hold instead of ring
 
Tuesday September 27 2011
TimeRepliesSubject
10:01PM 0 Asterisk OCF Resource Agents
9:05PM 1 Screening Mode Ghost
9:00PM 0 Grandstream HT 503, asterisk 1.8 and TLS
8:40PM 0 Asterisk 10.0.0-beta2 Now Available
12:36PM 2 number of calls simultaneous from AMI
 
Monday September 26 2011
TimeRepliesSubject
10:03PM 0 How to avoid users from other domain to add Asterisk Bot (jabber.conf)
6:16PM 10 model of diguim card
4:35PM 4 mISDN and 1.8
12:52PM 2 continue in dialplan when hang up queue
11:18AM 6 Call does not pass through
 
Sunday September 25 2011
TimeRepliesSubject
10:53PM 20 Asterisk Realtime Time Dial App
9:57AM 1 DID and the Caller ID for outgoing
1:35AM 21 Who is the "creative" mind behind changing Asterisk commands at CLI?
 
Saturday September 24 2011
TimeRepliesSubject
11:00PM 5 Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server
8:03PM 14 DID and how the caller id will appear
1:59AM 2 Question about Registrations
 
Friday September 23 2011
TimeRepliesSubject
9:08PM 5 Set (MONITOR_FILENAME=.................) for queuing recording calls
7:45PM 1 looking for free DID 708-839
7:11PM 0 Asterisk 1.8.7.0 Now Available
6:28PM 1 Postgresql Reconnect on connection failure
5:01PM 16 AGI Problem
4:30PM 0 usb hubs bluetooth chan_mobile
4:18PM 3 Force a SIP friend to use a certain IP?
4:14PM 0 sending fax using chan_capi
1:01PM 2 Digium ISDN card
12:37PM 0 Native bridging to SIP endpoints on the same NAT'd network
9:27AM 5 TDM400 FXO stopped working
4:19AM 5 dahdi_dummy required?
 
Thursday September 22 2011
TimeRepliesSubject
11:26PM 5 Problem with multiple sip-peers against the same host
8:23PM 8 bounty for ASTERISK-17474 streaming MusicOnHold bug
2:37PM 3 VoIP Abuse to Twitter (real time VoIP Abuse)
1:23PM 2 ForkCDR and asterisk 1.6.1
2:21AM 1 Change of default IVR prompt for meetme conference bridge.
 
Wednesday September 21 2011
TimeRepliesSubject
8:12PM 7 T.38 "client" for Linux?
5:08PM 0 SIP Call-ID for B-leg for non-bridged calls
4:25PM 0 snmpd error corresponds with Asterisk hang
4:11PM 0 Dahdi and BRI NT PtmP [SOLVED]
2:13PM 3 Dahdi and BRI NT PtmP
9:52AM 1 RTP stream when * and Xlite are both behind Nat devices.
6:38AM 2 Asterisk-Radius integration
1:17AM 0 redundant traffic (Tarek Sawah)
12:53AM 4 RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4
 
Tuesday September 20 2011
TimeRepliesSubject
10:53PM 3 Log for voicemail to email?
8:43PM 7 Fax from FXS to PRI
5:47PM 4 mISDN Vs Dahdi
3:37PM 5 How to add new Module in existed Asterisk
3:13PM 1 Using same extension number for outgoing and incoming both internal and PSTN
1:34PM 11 Add PinCode on my dialplan
9:23AM 7 Fixing an old bug related to extension "s" - feedback wanted
 
Monday September 19 2011
TimeRepliesSubject
10:51PM 2 Ghost DID in System
6:33PM 0 Asterisk-FreeBPX in Flash Support in Tampa_Not needed any longer
5:58PM 1 oddity with CISCO CCM and Asterisk
5:55PM 0 Looking for Asterisk-FreePBX in a Flash Support in Tampa
4:27PM 6 SIP OPTIONS... Error?
3:54PM 0 iLBC support in Asterisk after Google's acquisition of GIPS
2:54PM 3 question on DTMF
2:53PM 1 /usr/sbin/asterisk -rx and AMI
1:12PM 3 NC DATA FINDOUT IN AUTO DIALER
10:42AM 0 Anyone got a working SCCP configuration for a Cisco 6945?
9:46AM 1 Queuing: calls stay in queue and agents are ready !!
 
Sunday September 18 2011
TimeRepliesSubject
10:52PM 4 DTMF problem
8:28PM 2 [1.6.2.9] Echo even when using headset?
8:06AM 9 single registration per user
 
Saturday September 17 2011
TimeRepliesSubject
9:31PM 1 redundant traffic
7:19PM 1 Asterisk RPM repo?
11:01AM 2 Message recorder
 
Friday September 16 2011
TimeRepliesSubject
6:21PM 0 Wireless SIP phone with caller announce?
6:02PM 3 Temporarily disable DTMF on a call
12:19PM 0 How to build db1_dump185 tool ?
7:50AM 4 Inter-astersik dialling encounteres no audio
 
Thursday September 15 2011
TimeRepliesSubject
8:28PM 10 Monitoring second leg being dialed?
7:18PM 2 testing simultaneous calls
3:31PM 3 Asterisk PRI hangup
 
Wednesday September 14 2011
TimeRepliesSubject
8:29PM 3 Confusion with the status of SIP Trunk
2:03PM 2 Sip re-register / delay problem.
12:51PM 2 Mysql dialplan statement not executed
10:02AM 0 Mixmonitor command parameter problem on Asterisk 1.8.4
8:59AM 1 Read command - input correction not taken in account
8:18AM 2 SNMP problem
6:56AM 5 secret=pw in sip.conf affecting inter-asterisk sip call
12:49AM 5 using variables in the shell function
 
Tuesday September 13 2011
TimeRepliesSubject
10:44PM 3 realtime goto/gotoif/dial
9:35PM 2 Voicemail config
9:19PM 2 Determine negotiated codec in script
8:36PM 1 Send DTMF
1:56PM 9 High delay from Asterisk as PSTN simulator
12:15PM 0 WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468)
10:27AM 1 SIP Realtime & Templates (!)
1:26AM 1 Asterisk Manager Interface (AMI)
12:12AM 12 sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
 
Monday September 12 2011
TimeRepliesSubject
6:21PM 10 Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?
6:06PM 1 Asterisk is keep on sending Register request
3:19PM 7 Asterisk 1.4 Vs 1.6 Vs 1.8
6:50AM 0 CONFERENCE RECORD
6:44AM 12 broadcast
 
Sunday September 11 2011
TimeRepliesSubject
11:05PM 4 new sort of shell attack attempt via SIP?
2:16AM 1 Sip profiles per customer, behind a SIP proxy. How?
 
Friday September 9 2011
TimeRepliesSubject
6:13PM 10 Reporting for Asterisk Call Center
2:22PM 3 Console Stereo - One call per ear
1:55PM 12 PRI Issues After Upgrade
5:40AM 0 Beginner Question: Remote access
 
Thursday September 8 2011
TimeRepliesSubject
8:00PM 0 DAHDI-Linux 2.5.0.1 and DAHDI-Tools 2.5.0.1 Released
6:51PM 3 Jitter only affecting meetme - and echo testing
6:04AM 1 Desktops Recieved from Delhi
 
Wednesday September 7 2011
TimeRepliesSubject
10:47PM 2 asterisk curl and utf8 problems
5:00PM 0 Scheduled Maintenance for Asterisk Issue Tracker (JIRA)
1:59PM 14 Overlap SIP dialing
10:29AM 4 DTMF games with Asterisk
6:47AM 0 phone is silent in asterisk 1.8.5
6:30AM 0 How to achieve 33.6 kb/s faxing with asterisk ?
2:42AM 4 (no subject)
 
Tuesday September 6 2011
TimeRepliesSubject
8:08PM 7 trying to build 1.8.6.0 on CentOS 6, problems with ptlib
4:43PM 2 pick up code
11:10AM 3 PARAMETERS FOR DID'S FROM TELCOM AND ASTERISK TO SYNC
7:08AM 16 ISDN2 PCIe Card for Asterisk
 
Monday September 5 2011
TimeRepliesSubject
9:00PM 2 Asterisk 1.8 not accepting call from DID
7:53PM 2 Variables error in 1.8.6.0.
1:34PM 0 Codec warning polluting the CLI since 1.8
12:41PM 1 How does AMI work with events ?
8:27AM 1 Cisco SPA 941 and auto-answer with SIPheader Call-Info
7:43AM 0 Followme generate ringing instead of MOH
2:48AM 2 Asterisk 1.8 not working for me
 
Sunday September 4 2011
TimeRepliesSubject
10:47PM 4 Phone numbers and asterisk
10:14AM 0 Router with PPPoE and QOS for light Asterisk and general use
 
Saturday September 3 2011
TimeRepliesSubject
6:19PM 4 pickup for extension in asterisk 1.4?
5:41AM 7 Beggining asterisk
3:32AM 0 res_jabber
3:27AM 2 Set(CHANNEL(musicclass)=
12:23AM 2 What do these SIP errors mean?
12:16AM 1 upgrading from 1.4.39 to 1.8.5
 
Friday September 2 2011
TimeRepliesSubject
8:15PM 2 any iLBC folks around?
3:33PM 0 from asterisk 1.6 to 1.8 - sip trunk unreachable
3:14PM 3 Prompt for PIN After dialing
2:31PM 0 QSIG-SIP overlap dialing and Asterisk (RFC4497)
11:28AM 14 how to add-edit-delete entery into asterisk conf files
10:30AM 11 Asterisk on Android?
9:14AM 0 CDR dialed digits missing
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Thursday September 1 2011
TimeRepliesSubject
9:39PM 3 Distributed device state / presence info??
7:51PM 2 Phantom rings after FXO/FXS setup
5:16PM 3 Unexpected behavior change from Asterisk 1.6.2.14 to Asterisk 1.8.5.0
4:13PM 9 problems with hylafax + iaxmodem + asterisk1.8.5
2:44PM 12 Anyone using Asterisk on VirtualBox ?
12:19PM 0 Simultaneous ring on Soft phone and Desk phone.
11:04AM 6 Asterisk 1.8.3.3 T.38 Gateway