asterisk users - Jul 2011

Sunday July 31 2011
11:04PM 2 sip attacks
11:48AM 2 Codec translation from gsm to other codecs or from other codecs to gsm
5:22AM 1 asterisk + sccp-b problem
Friday July 29 2011
11:29PM 0 Tutorial on the Asterisk Manager Interface
10:12PM 1 Serious bug in - what is the time frame to fix such bugs?
3:51PM 1 Accept the dtmf input in call patch
10:56AM 2 How to use these feature of Asterisk
10:56AM 2 X86_64 Compilation Issue
9:56AM 0 Asterisk SIP authentication against [section] instead of username
6:56AM 1 call forwarding number from outside.
Thursday July 28 2011
11:05PM 1 Dialplan required for recording
8:22PM 1 Questions about FMFM with linked servers
7:47PM 2 Disabling Polycom "reject" and "DND" or disable Asterisk 486 "Busy Here" actions
6:42PM 2 Voicemail not acting as documented.
2:53PM 5 MoH - conversion command
1:22PM 3 hide google voice number
12:45PM 3 Capturing call Reject/Decline events on a PRI line
9:19AM 0 [chan_mobile addons] DTMF transfer from calling mobile to Asterisk through called mobile FAILED
7:47AM 0 Radius billing for asterisk
6:45AM 0 Avaya & Asterisk FreePBX Integration Problem
6:32AM 2 Connect asterisk to normal telephone PBX
Wednesday July 27 2011
6:41PM 2 Lightning and thunder (Claude Hayn
6:32PM 2 Problem H323 asterisk
4:33PM 0 Fwd: Re: Securing Asterisk
1:44PM 5 Lightning and thunder
1:29PM 2 Stun Server
11:59AM 0 AMI action PlayDTMF and SIP:INFO
1:02AM 0 libpri rpm version 1.4.12 for CentOS 5.6
Tuesday July 26 2011
7:21PM 3 file2ban
6:20PM 1 Scheduling destruction of SIP dialog
2:13PM 2 Browser based SIP UA
1:19PM 1 NAT yes
11:47AM 0 Callback + DISA
7:21AM 9 Why no traction for Windows version?
6:46AM 2 MusicOnHold not loaded
Monday July 25 2011
6:10PM 0 Malformed/missing URL
5:05PM 0 Registration problems, Linksys SPA 3102 on Asterisk 1.4.20
1:31PM 0 Is there a protocol to be used to communicate between different Asterisk Boxes to distribute load and tasks
1:22PM 0 Is there a protocl that let the Asterisk boxes talk to each other and treated as one entity?
1:18PM 1 callgroup and pickupgroup (Carlos Chavez)
1:09PM 0 The queue is not routing for the agent: returned -1:
12:59PM 0 Meaning Callerid Datatypes.
11:19AM 2 Fail to compile DAHDI under Linux 3.0.0
9:22AM 0 use of .exports file in asterisk
8:43AM 1 dahdi channels busy/congested
Sunday July 24 2011
11:09PM 2 Error Code 101
3:09PM 0 One way calling on asterisk to cisco
1:09AM 1 Security questions
Saturday July 23 2011
5:55PM 1 One way calling on asterisk to cisco call manager integration
5:38PM 9 Securing Asterisk
5:18PM 1 The queue is not routing for the agent: returned -1: Invalid argument
2:30PM 1 dialplan pattern help
9:51AM 2 DISA password
Friday July 22 2011
11:32PM 2 Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"
11:13PM 1 use dahdi for local terminal modem access?
9:27PM 0 Asterisk 10.0.0 Beta 1 Now Available!
7:44PM 4 10.0.0 better than 2.0.0?
5:09PM 1 Connecting to a Taqua switch
3:19PM 1 Phase out macro command.
2:39PM 4 Question about codec re-negotiation in asterisk 1.4.X
9:27AM 1 Pickup(${EXTEN:2}); not works from outside
8:57AM 0 [Fwd: Re: Strange network issue]
8:10AM 1 asterisk rpm build problem
7:26AM 4 Asterisk as a Operator Phone
Thursday July 21 2011
11:13PM 3 Strange network issue
11:09PM 0 Per-line registration
10:49PM 1 Rebooting a Grandstream
8:30PM 1 asterisk's SDP
9:31AM 4 Functions not autoloading
9:05AM 0 Asterisk doesn't like OpenBTS!!!
8:06AM 1 Prevent Asterisk from setting CALLERID(name) or unsetting CALLERID(name)
5:29AM 6 My Asterisk Box was hacked
Wednesday July 20 2011
10:28PM 1 Multiple SIP trunks between same pair of asterisk box
6:29PM 2 ISAC and Asterisk
5:57PM 0 >64 pickup groups
5:53PM 2 "Problems" with System() application
1:42PM 0 Call flow attached
1:25PM 3 HELP - Client wants to reverse Asterisk Functionality
1:09PM 0 ilbc codec
12:16PM 3 Macro to Dial a Channel Group using Round-robin
9:00AM 3 Help: How can I Add my own Word in option packets in from field of SIP "From Asterisk??"
Tuesday July 19 2011
7:48PM 1 SS7 and PRI compatibility
6:07PM 6 Multiple Asterisk Sessions on same machine
4:53PM 1 Recall: Time zone on phones
4:53PM 1 Time zone on phones
4:24PM 1 Is there a protocol to be used to communicate between different Asterisk Boxes to distribute load and tasks?
3:53PM 1 Is there a protocol that let the Asterisk boxes talk to each other and treated as one entity?
12:50PM 2 max one sip peer to register
10:23AM 1 callgroup and pickupgroup
7:56AM 0 Converged phones; akin to SwitchVox ?
7:24AM 1 AsteriskNow install addons despite license conflict with FFA and G.729
7:20AM 3 a=sendonly Music On Hold ignored
Monday July 18 2011
11:03PM 2 What is the use for the agent password if login via exten
10:05PM 1 libss7 variables
4:42PM 1 chan_gtalk load error
2:15PM 2 No Audio after attended tranfer
1:20PM 5 [1.4] Minimal installation?
11:20AM 3 FAX with SIP
11:03AM 4 Seg Faults with
10:14AM 3 Compact, affordable x86 devices?
Saturday July 16 2011
2:19PM 1 asterisk 1.6 agi problem with PHP
12:58PM 3 Requires
12:57AM 0 Unable to register an endpoint after upgrading from to
Friday July 15 2011
5:52PM 3 Macro issue under 1.8.5
4:47PM 4 Using Firewall to protect Asterisk
12:29PM 1 Controlling max simultaneous calls for a group/.call files
10:11AM 3 Redirecting call from one E1 to another?
8:03AM 2 *8 causing large number of channels to go stale (possible bug)
8:02AM 0 dialplan: all extern, except
Thursday July 14 2011
10:57PM 4 Asterisk in the amazon cloud
8:13PM 0 AstLinux 0.7.9 Release
5:37PM 1 Enable T.38
2:40PM 3 Asterisk binaries on CentOS version 6
1:46PM 0 cseq decreasing => 500 Server Error
11:27AM 0 How to Get all users list in asterisk
4:45AM 9 Extension wise dialplan
Wednesday July 13 2011
10:19PM 0 Chan_mobile
6:51PM 2 TDM400p susceptible to EMI?
9:35AM 1 How to Hang up a stale SIP channel?
7:02AM 1 Connect Avaya to Asterisk PBX
6:01AM 0 (no subject)
4:23AM 1 Problem on Dialling-out
Tuesday July 12 2011
8:33PM 2 Mysterious dropped calls
5:30PM 0 Dtmf issues solved
4:40PM 1 REALY strange issue with making calls biside 2 phones
3:40PM 0 Park/VoiceMail on DAHDI congestion
3:33PM 3 CDRs
2:48AM 3 skype for asterisk usage in the future
Monday July 11 2011
9:43PM 0 Asterisk Now Available
9:29PM 4 Benchmarking AGI performance in C, PHP, and Perl
9:01PM 1 ${HASH(SIP_CAUSE, ...)} and peer name
5:11PM 1 keeping asterisk memory
1:20PM 0 RFC 6315: IANA Registration for Enumservice 'iax'
Sunday July 10 2011
10:53PM 1 How to logout!
10:36PM 1 What is the use for the agent password if login via exten?
6:26PM 0 Help anybody - how to manage SRTP with TLS trasport
3:02PM 2 Problem with setting up fresh 1.8.5 Asterisk
11:04AM 4 Queue Issue : Duration between 2 agents call
9:53AM 3 References customers
8:52AM 2 Thomson ST022 - External Call problems
Saturday July 9 2011
6:55PM 0 About recompile reinstall couse of SRTP
4:57PM 0 Auto Reply: asterisk-users Digest, Vol 84, Issue 15
3:31PM 2 Meetme not prompting for PIN
1:31PM 0 Scheduled Maintenance for Asterisk Project Services
11:34AM 4 OT: Google Plus
9:45AM 0 Strange behavior with Asterisk TLS
Friday July 8 2011
11:28PM 11 New VirtualBox Beta Has PCI Pass-Through Support
9:49PM 2 DTMF issues still
9:29PM 0 Cisco ATA 187 configuration file
6:46PM 0 Problems with DTMF Caller ID
4:39PM 3 How to create a module
4:19PM 0 Asterisk meetme and Timer ?
3:01PM 1 Issue 0019268 Patch Asterisk
2:58PM 2 FXO ports locking up
1:13PM 2 Master.csv file limit
10:42AM 4 timeout with outbound calls
10:23AM 2 dialout time configuration
8:48AM 3 DB Driven IVR
8:48AM 0 asterisk and eyebeam
8:42AM 0 HI
6:58AM 4 Problem in Detecting Dtmf on FXO line.
Thursday July 7 2011
8:41PM 4 Anybody doing PRI over IP?
7:43PM 0 Installing Asterisk from repository works great without the need to install Dahdi on Host Node of Proxmox - But trying to install from source fails. Why?
5:17PM 1 Eyebeam crashes when dialing an invalid number...
4:43PM 4 Stripping characters from ${CALLERID(num)} ?
4:22PM 1 check_auth: username mismatch
4:01PM 0 No pattern 407 from SIP provider iCall
1:53PM 0 asterisk 1.8X problem: no outbound callerid set in callfiles
Wednesday July 6 2011
9:48PM 1 ooh323 does not work fine, what about h323 channel
5:11PM 0 libpri 1.4.12 Now Available
12:02PM 3 Monitoring connection to VoIP provider?
9:36AM 2 Agents outbound calls to be recorded
8:37AM 2 single keypress short-circuits to invalid extension handler
12:30AM 2 Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system
Tuesday July 5 2011
8:46PM 1 Couldn't call Agent and segfault
5:47PM 0 OT - Polycom - 2 localization file versions on the same TFTP server
3:26PM 1 Recording SIP history
2:59PM 0 Can't get video on one server of 4
2:07PM 2 Cant find asterisk src dir for FreePBX full distro
12:22PM 0 AST_DEVICE_UNAVAILABLE vs. AST_DEVICE_UNKNOWN for new loaded realtime peers
9:25AM 1 More SQL Querys in dialplan
6:59AM 2 realm question
6:45AM 1 SIP Presence not working
5:22AM 1 Blind Transfer Connected
4:06AM 0 DTMF between sip trunks and PRIs
Monday July 4 2011
11:36PM 1 Agent Login, Logout, Ready, Not Ready from the CTI application
7:29PM 1 Testing Asterisk with media - sipp
4:51PM 0 RINGNOANSWER events in queue log
10:58AM 4 stream rtp from asterisk
9:40AM 1 Mixmonitor concept's question
5:40AM 1 how to set to make a call through a fixed ip on a 2 ips server?
Sunday July 3 2011
12:41AM 1 SIP Peer Name Variable
Saturday July 2 2011
10:48PM 2 chanspy spies on wrong channel
4:58PM 2 Distributing the incoming calls and the huntgroup
12:38AM 0 inter asterisk-user list
Friday July 1 2011
8:09PM 0 RINGNOANSWER IN queue_log
7:35PM 0 Queue transfer order
6:59PM 0 Using Asterisk as Contact Center: Concurrent Calls + How much time
5:55PM 2 How to without GUI
5:43PM 0 IVR sound after dial sip
3:10PM 0 participants redirection between conferences
2:36PM 1 calleridname presentation Asterisk => Siemens
2:26PM 1 Dropping Conference calls
1:40PM 0 Anyone worked with allo products?
10:03AM 0 Asterisk RPM
9:29AM 1 error in GUI access
5:43AM 1 Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted
5:34AM 0 Poll : libpri naming scheme and asterisk-libpri merge
12:06AM 3 Asterisk 1.6.1 Realtime SIP Users