asterisk users - Jan 2012

Tuesday January 31 2012
9:47PM 2 Experience with Eicon Diva PRO 3.0?
7:49PM 0 Troubleshooting one-way audio with H.323 trunk between Asterisk and Avaya IP Office
5:58PM 0 Deadlock detected in asterisk- x86_64
5:39PM 13 Proposed changes to Asterisk release and support cycles
3:16PM 2 SRV record for non-standard SIP port?
12:32PM 16 [NAT] SSH vs. OpenVPN?
10:50AM 5 Cell Phone as a Queue member
8:12AM 0 AMI - Getting Event of QueueAgents WrapupTime State
6:17AM 4 Problem with DTMF in Voicemail main
Monday January 30 2012
9:47PM 0 TLS problems - patch in Jira
9:47PM 0 RFC 5922 (TLS Certificates) and Asterisk
8:53PM 1 fall back to inband DTMF?
8:01PM 0 Codec
4:12PM 6 CA Issued Certificates / TLS + SRTP
9:42AM 0 atx timeout - play xferfailsound
Sunday January 29 2012
2:48PM 1 fail2ban restarts
7:21AM 3 SendFax not sending AMI events
Saturday January 28 2012
4:22PM 2 process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found
Friday January 27 2012
7:03PM 1 TCP transport and BLF
6:57PM 2 Asterisk 1.4 and configuration to be via Database instead of conf files
5:10PM 0 Asterisk 10.1.0 Now Available
5:10PM 3 Asterisk Now Available
3:44PM 1 sip reload and TCP transport.
7:48AM 3 Strange how Asterisk know the updated information of log
4:49AM 7 Weird IPs in Fail2ban list
2:28AM 5 upgraded > 10.1.0-rc2: now db warnings
Thursday January 26 2012
5:35PM 3 Too many open files
3:46PM 7 SLA for DAHDI FXO - Emulating Key System Functionality
3:23PM 1 Manager Originate and Callerid ?
1:29PM 0 Softphones with SIP transfer
1:22PM 2 User hit f to disconnect call.
12:10AM 0 Dropping incompatible voice frame error
Wednesday January 25 2012
3:29PM 15 Executing Script after MixMonitor is called
3:29PM 6 play sound file
1:34PM 0 Blocking in: ast_waitfor_nandfds
11:56AM 0 Signalling and Media Configuration
Tuesday January 24 2012
10:29PM 1 Is there a sip show equivelant.
2:34PM 4 allowguest = yes? no?
2:29PM 10 RFE idea for VM application
1:47PM 17 ChanSpy : how to know channel name ?
Monday January 23 2012
8:53PM 3 ConfBridge details
7:00PM 4 asterisk does not detect menus
6:16PM 1 Timing Slips CRC & E-Bit Errors - Asterisk - Trixbox
3:48PM 11 SDP Issue
2:35PM 13 Cordless SIP phone
2:28PM 3 Avaya 4610sw IP Phone
2:29AM 0 SIP - connected line has changed. Saving it until answer for IAX2/iaxy
2:02AM 0 Chan_Mobile Nokia E51, csr bt dogle, Voice OK but no SMS Support ?
Sunday January 22 2012
9:06PM 2 Analoge and E1 ports
Saturday January 21 2012
3:51PM 1 Force CDR to be written.
1:21PM 1 View # active calls in a context
Friday January 20 2012
7:44PM 0 10.1.0-rc1 : WARNING: abstract_jb.c:384 jb_get_and_deliver: AST_JB_IMPL_NOFRAME
6:07PM 19 Pickup calls coming from queues
2:07PM 7 Asterisk NOT in the media path
1:36AM 12 Sip Registration Hijacking
Thursday January 19 2012
11:25PM 5 Efficient logging of PRI traffic for later analysis?
8:23PM 0 Asterisk 1.8 - SIP losing registration
6:00PM 0 Asterisk and 10.0.1 Now Available (Security Release)
5:40PM 1 AST-2012-001: SRTP Video Remote Crash Vulnerability
4:47PM 0 odd disconnects with major company's voice recog
3:15PM 0 asterisk not connecting to sipgate / NAT related issue?
1:33PM 12 Voicemail weirdness after upgrade
12:08PM 0 Huh? Local is being asked to answer?
9:10AM 4 Asterisk rewrites "From" header when CALLERID(num-pres)=prohib_passed_screen is set
7:05AM 2 Does Asterisk permit multiple registrations to the same host?
3:35AM 7 Asterisk to support Dialogic Cards
Wednesday January 18 2012
10:38PM 0 attended transfers going to wrong voicemail Asterisk 1.8 Polycom 650
5:50PM 1 Installing the 3.1 sources of Kernel with Asterisk
9:24AM 20 Dahdi for meetme on AMD64 arch?
9:19AM 2 Compile error
7:44AM 3 Failed to Allocate port for RTP instance
Tuesday January 17 2012
9:06PM 0 SIP trunk call initiated as Anonymous@anonymous.invalid
5:23PM 3 Problem answering phone
4:53PM 0 Is there any way to terminate async origination initialized by AMY?
4:24PM 4 Prepaid billing
2:11PM 5 /etc/init.d script and calling asterisk command line.
10:57AM 7 Macro vs sub
10:54AM 0 pickup group
10:16AM 2 Core file created in /tmp
Monday January 16 2012
11:02PM 0 OT - Configuring Freepbx's to work with ssmtp
6:41PM 6 How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
5:52PM 1 Starting things off without a dial tone
5:11PM 3 meetme with IVR
4:02PM 7 Update callee num or name at caller display
3:55PM 4 How Can I configure the between call oneside IVR
3:21PM 5 SayDigits playback doesn't always work
12:56PM 0 Where to find meaning of /n inLocal/6613@from-queue/n ? [SOLVED]
12:41PM 2 Where to find meaning of /n in Local/6613@from-queue/n ?
10:48AM 5 Real T1 trunk group...
10:14AM 5 How to check currently used libraries from command line ?
9:19AM 0 echo & audio delay in SIP VOIP
7:59AM 4 local channels and g729a voice quality
2:14AM 2 Real trunk group w/ DAHDI
1:53AM 10 Peer doesn't answer
Sunday January 15 2012
9:06AM 5 ssh to a Cisco 7961 is not working
8:35AM 0 configuring a Cisco 7961 so that different line appearances register to different SIP proxy addresses
4:43AM 0 AstLinux 1.01 Released
Saturday January 14 2012
9:38PM 1 CDR into ical?
2:37AM 2 asterisk problem sip
1:25AM 5 Asterisk as UAC: How to put call OnHold
Friday January 13 2012
10:07PM 2 Sporadic one way audio problem
10:00PM 2 Queue option 'R'
6:43PM 0 Preços por serviços e equipamento
4:09PM 0 Stuck DAHDI Channels
12:45PM 2 SIP hardphone with dual gigabit ethernet ports
12:41PM 14 dialplan problem : not including context
9:56AM 1 odbc storage for video message
9:47AM 0 Queue member is permanently BUSY
Thursday January 12 2012
6:57PM 0 t38modem v2, which version or patch of asterisk?
12:39PM 3 Questions on hardware or software-based echo cancellation
11:51AM 9 Call abandoned from queue not showing in CDR (possible bug)
11:44AM 11 FAX Installation in Asterisk
9:50AM 0 Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ? [SOLVED]
8:44AM 5 how to set callerid in php AGI file.
Wednesday January 11 2012
11:10PM 4 Problems with codec translation when using Monitor and MixMonitor
8:37PM 4 Attempt to Originate between IAX2/xxxx and an application hangs until timeout in
7:56PM 4 Exceptionally long voice queue length
3:05PM 3 Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]
2:13PM 0 OT - Which iceweasel plugin to play gsm sound files ?
12:01PM 0 Iax hold events in AMI 1.1
11:29AM 4 SIP and NAT best practices since recent changes?
10:05AM 1 Q: SIPNATtraversal.pdf
9:09AM 5 Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
7:04AM 1 Problems faced in load testing of asterisk
6:39AM 7 Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
4:35AM 2 No audio available on SIP/
Tuesday January 10 2012
11:00PM 2 Odd DTMF problem when receiving calls
9:12PM 1 Linux Stun Server
5:02PM 5 Hang up phone after declined attended transfer
4:47PM 0 Noise in caller handset when dialing out (with dahdi 2.6.0) [SOLVED]
Monday January 9 2012
7:59PM 4 44Khz files in Asterisk 10
6:54PM 1 message WARNING features.c: Failed to play transfer sound! and attended transfer hangs up
6:52PM 4 Noise in caller handset when dialing out (with dahdi 2.6.0)
4:46PM 6 Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
10:20AM 0 DEBUG Message
8:20AM 4 Cisco AS5300 and Digium g729A codec
8:02AM 1 Asterisk as register server through OpenSIPS
7:44AM 9 create table in mysql using asterisk
6:20AM 1 video mail is not store
Sunday January 8 2012
6:26PM 2 Answering call from queue, then put back in queue?
12:30PM 1 cached VMI on manual voicemail update
Saturday January 7 2012
1:39PM 1 Couple of questions: SIP ALG, allowguest=no
6:15AM 10 Asterisk 10.0 & 1.4 - iax codec are not compatible
Friday January 6 2012
10:00PM 8 best softphone for 2012?
6:51PM 6 Why write your dialplan using Lua?
5:56PM 0 Streaming Music to 75 callers ..
4:17PM 3 Change port from 5060 on Snom phone
1:14PM 4 Connecting to an Old Phone System
7:32AM 0 no audio using g729A for Cisco AS5300 sip peer
1:45AM 2 calling specific 1800-number not going through.
12:55AM 0 which choice: asterisk-gui or freepbx?
Thursday January 5 2012
11:42PM 18 asterisk 1.8.8 - caller ID not working.
10:30PM 3 STOP loading extensions.ael
8:03PM 5 Blind transfers being cancelled by asterisk & hanging up on remote caller
6:22PM 1 question on CDR
5:19PM 20 Best non polycom SIP conference room phone
2:54PM 6 asterisk -> AGI (perl) -> sqlplus(oracle)
10:12AM 2 Asterisk1.8 support video trancoding ?
9:35AM 1 Video trancoding not done.
7:03AM 6 Where are the fax instructions?
Wednesday January 4 2012
9:45PM 1 DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released
8:47PM 1 question sangoma vs digium
7:44PM 1 From address missing 'sip:', using it anyway
4:45PM 6 asterisk -> AGI (perl) -> sqlplus (oracle)
4:19PM 0 Which QSIG variant and profiles does asterisk support ?
11:53AM 20 Set Call Codec in extension.conf
10:13AM 8 Asterisk won't start - trap invalid opcode
9:37AM 5 Failed to authenticate on INVITE to Anonymous
9:19AM 3 Rami
6:25AM 16 Anyone have a reliable T.38 Solution
4:14AM 1 ConfBridge no audio problem
2:42AM 29 Speech recognition in asterisk using google voice API
2:42AM 1 Mark queue agent as away
Tuesday January 3 2012
10:41PM 1 ISDN E1, after electrical disconnected is not becoming UP, IRQ misses: 1
7:30PM 1 Layer2 Down in BRI connection
6:06PM 3 Ringing agents cell as an alert?
5:53PM 3 Problem connecting to 4569/UDP
3:24PM 3 NAT/IPTABLES workarounds
2:20PM 0 Registering multi-clients
1:48PM 5 Question on system command 1.4.43
1:47PM 4 Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
1:34PM 1 Using Asterisk as a softphone
11:08AM 9 How to make SIP guest call
10:22AM 2 Asterisk 1.8 - BRI D Channel going up and down every few seconds
9:07AM 2 dialplan -> dial command -> custom ringtone
12:38AM 3 Problem w/ PC port on Polycom 335
Monday January 2 2012
7:57PM 0 Wired attack on Asterisk - Can anyone explain this?
11:38AM 0 Help_video voice mail not retriev properly
9:07AM 13 Set Call type in dial plan
4:34AM 4 tcp version of toronto - osaka doesn't work
Sunday January 1 2012
10:17PM 4 asterisk 1.8 codec negotiation
10:11PM 0 GoAutoDialer, ViciDial and Vicidial group
12:22PM 0 481 Call leg/transaction does not exists Status Response