asterisk users - Feb 2011

Monday February 28 2011
5:19PM 0 Asterisk 1.8.3 Now Available
5:18PM 0 Asterisk Now Available
5:17PM 0 Asterisk 1.4.40 Now Available
4:39PM 2 CEL and PGSQL
3:27PM 5 Failover Routing
1:40PM 2 Asterisk 1.8.3-rc3 and one way audio
10:33AM 2 asterisk security....again
10:25AM 0 Obi110 as gateway to PSTN?
10:24AM 7 Two Asterisk machines for redundancy
2:12AM 5 Using voice modem as poor man's FXO in Asterisk 1.8
Sunday February 27 2011
12:32PM 1 [Dahdi 2.4.0] Flash() hangs up
Saturday February 26 2011
10:33PM 2 Need to buy the Digium card, to confirm
4:16PM 1 SRTP Error Message
9:08AM 1 Detect DTMF tone during call?
Friday February 25 2011
10:49PM 0 T1 channel audio control
9:47PM 0 Asterisk 1.2 zap hangup issue
7:30PM 4 Asterisk/Skype
5:37PM 0 PRI B-Channel restarting itself continually
5:04PM 5 [OT] Yealink IP Phones
4:16PM 2 SIP dialogs not killed?
12:40PM 1 Handle in dialplan user disconnection
Thursday February 24 2011
10:27PM 1 missing argument on AGI
9:05PM 2 Recieve_Fax caused crash
8:47PM 1 Using a Virtual IP Line
8:31PM 1 extensions.lua with luasql.mysql.
6:41PM 2 Paging with Polycom 3.3.x
4:32PM 1 Debug Dropped Audio
2:58PM 0 "Asterisk" caller ID
2:51PM 4 Google Voice outbound Caller ID broken
1:42PM 1 RTP (voice) issue. STUN server
1:24PM 1 Registration failed though configured.
12:36PM 2 [1.4] Still can't get it to call back
11:38AM 2 Carrying context from one server to another?
10:56AM 2 [] Simple AGI doesn't reply
10:15AM 1 Unknown calls
7:16AM 2 DIAL through Specific number in PRI
1:10AM 0 One way dialing over a SIP trunk
Wednesday February 23 2011
7:56PM 0 SIP friend name
4:10PM 2 REFER and dialplan broken (as documented in chan_sip.c on line 11951)
3:31PM 4 secret vs remotesecret on outgoing calls in Asterisk
2:14PM 0 Adhearsion 1.0.1 Released
10:39AM 4 AMI FullyBooted issue
10:17AM 2 extend the timout on ringing for pri or sip
Tuesday February 22 2011
9:34PM 2 Multiple public address to one Asterisk server behind NAT?
6:21PM 1 calls between iax and sip
2:57PM 0 Weird Inbound Problem.
1:02PM 3 Asterisk,,, and Now Available
10:39AM 1 [] ast_carefulwrite: write() returned error: Broken pipe
10:06AM 0 AddQueueMember and stateinterface question
3:11AM 1 NVFaxDetect causing segfault
Monday February 21 2011
10:52PM 1 [Dahdi 2.4.0] DAHDI_CHANCONFIG failed on channel 1
10:46PM 3 Problem installing FXS module in old digium 4 channel tdm card
9:52PM 0 (no subject)
9:45PM 4 AST-2011-002: Multiple array overflow and crash vulnerabilities in UDPTL code
8:28PM 0 Erroneous email from JIRA
7:07PM 1 Free calls to the US provider recommendation
6:44PM 1 T1 PRI shows yellow/red alarm
4:51PM 0 Difference mohsuggest & mohinterpret
3:47PM 1 Dialplan execution stops on app call even with TryExec (Am I missing something simple?)
11:13AM 2 calls are not going thru e1 line
Sunday February 20 2011
5:03AM 0 My new blog
2:39AM 0 AstLinux 0.7.6 Released
Saturday February 19 2011
9:15PM 2 First go at a stock 1.8 install -- where's DAHDI?
9:52AM 1 [1.4] "show channels" in extensions.conf?
1:30AM 4 AGI script dies after receivefax
12:00AM 1 Problem in dialing out
Friday February 18 2011
8:59PM 2 no progress indication
5:56PM 2 cmd MySQL
3:36PM 1 [1.4/AGI] CHANNEL STATUS never "down & available"?
3:11PM 2 Meet me recording
11:36AM 3 Assigning an extension to a roaming phone
11:32AM 3 lua -asterisk manual
11:29AM 2 undefined symbol: ast_compile_ael2
9:18AM 1 Asterisk with TE 121 DADHI incoming calls fail
8:31AM 2 Dial(Local/...) vs. Goto()?
8:23AM 3 FAX on PRI to MFCR2
8:17AM 2 DTMF and Snom
7:16AM 2 Trunk grouping
1:33AM 0 Voice mail forwarding enhancement
12:31AM 1 Dial() function
12:26AM 4 Newbie´s question about Asterisk...
Thursday February 17 2011
6:27PM 2 Polycom Do Not Disturb button and asterisk hints
5:11PM 1 Pickup from an specific exten
4:46PM 1 Setting two E1 cards
4:15PM 0 PRI "wanrouter status" shows disconnected - system problem or Telco?
2:54PM 1 Got SIP response 400 "Bad Request" back from
1:28PM 1 Realtime MySQL - Asterisk 1.8.2
11:13AM 0 Friday 18 Feb at 12 Noon EST: SylkServer and Blink
10:34AM 0 Samsung smt-i3100
7:15AM 3 Asterisk Using as a SIP Client
1:47AM 0 Google 10%
Wednesday February 16 2011
8:32PM 1 No ring tone on inbound call - but channel connects fine
7:51PM 5 Polycom IP335
6:24PM 1 Asterisk on a USB with persistence
6:00PM 1 pipe audio stream to external application
4:40PM 1 trunk not working if I register a phone at the same IP as the trunk peer's IP
1:41PM 2 Play one audio file to the called part before the Dial() command?
11:47AM 2 function Echo() doesn't work
11:41AM 1 how to diable echo cancellation for sip?
10:45AM 1 Detect #,* DTMF in dialplan
10:32AM 1 Cisco 7945G phone with asterisk
10:13AM 1 Barge in.
10:05AM 1 How to know Caller's last position in Queue?
8:53AM 0 Regarding error in asterisk
5:49AM 4 Connect Asterisk to a cell phone
12:39AM 7 DTMF not detected, time out
Tuesday February 15 2011
8:47PM 2 Dialplan end of pattern matching question
8:12PM 1 Aastra phones cannot transfer calls?
8:05PM 0 pstack debug asterisk
7:15PM 4 Voicemail email attachment as MP3, with tags containing sender name, number, message number
6:39PM 2 Paging a message. How?
6:15PM 2 Realtime and Local Channel Crash Problem 1.8.3-rc2
5:06PM 1 Lua extensions are not working on asterisk
5:06PM 2 Adjusting Rx and Tx gains
3:20PM 0 asterisk 1.8.2 freez
2:15PM 1 outbound call leg CALLID
1:58PM 0 weird problem with Vega 100
12:49PM 0 changing logo of 7905
8:51AM 3 further action after caller in a queue hangs up
4:00AM 6 Fax Woes
3:39AM 5 uptime
1:47AM 1 trunks and phones registered from the same IP
Monday February 14 2011
11:16PM 1 Asterisk Call File using Local Channel not passing Variable back to Dialplan
9:36PM 5 Hide the plain text password
6:46PM 1 unregistered trunks and registered phones coming from the same IP
4:06PM 1 SIP session timers just on one specific channel
2:16PM 3 issue with some numbers
1:40PM 2 Cisco 7960 & asterisk 1.8.22 ringlist.dat error
12:11PM 1 Possible dumb question: new kernel, new DAHDI?
10:29AM 1 Problems with realtime sip
6:10AM 1 IP ban list by country
Sunday February 13 2011
2:52PM 1 [modules.conf] Modules still loaded after "noload"
5:59AM 1 Call Files, Variable passing
4:53AM 1 Fax for Asterisk SIP-TDM
Saturday February 12 2011
11:06PM 1 Transfer Device Data
4:23PM 3 Using files .call or AMI
12:31PM 11 SIP Hardphone that works well with asterisk
9:14AM 1 [Zaptel] "numberplan-local" context from nowhere?
7:42AM 1 Variables losing their value????
7:03AM 0 what are QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY veriables?
Friday February 11 2011
10:37PM 6 On-Hold Music
8:57PM 2 Asterisk compile option DAHDI SPANS
5:26PM 2 dialplan announcements
5:13PM 0 AstMail
5:02PM 1 Asterisk 1.8.3 BLF stopped working
11:48AM 1 Realtime queues not playing prompts
9:37AM 3 Asterisk 1.8.3
6:47AM 2 sangoma wanpipe install error
12:58AM 3 meetme conference & playback of random sound file
Thursday February 10 2011
11:50PM 2 Gtalk/Jabber Issue
10:24PM 0 res_pgsql re-connect on db failure?
9:11PM 0 "intercom" SIP header being ignored by Kirk wireless handsets
6:04PM 0 Busy Detection on Analog Lines
12:13PM 3 CDR with unix time.
11:08AM 1 Early audio SIP sequence order question
4:49AM 2 zaptel/dahdi settings for singtel E1 line
3:08AM 2 Unable to make outgoing calls with Internode
Wednesday February 9 2011
11:56PM 0 Error loading module ��Է�
7:31PM 1 AEL Eswitches
7:30PM 1 Defining what an extension should do after the Dial() command returns busy.
3:21PM 2 queue called by agi doesn't re-enter the script
11:56AM 0 sent you a movie ticket redeemable at more than 200 nation wide theatre chains
10:36AM 2 SIP MESSAGE outside calls - state of the art?
9:45AM 0 Reliably getting sip extension name from channel variables
4:11AM 1 dial option 'g' not working
3:40AM 0 Manual Call Transfer (Perl, Asterisk::AGI, MySQL)
Tuesday February 8 2011
9:36PM 0 Microsoft Speech Server/UCMA Integration
8:11PM 1 echo when calling to the pstn
7:18PM 0 Manual Call Transfer // Perl // Asterisk::AGI // MySQL
6:56PM 0 Scheduled Maintenance: and
6:08PM 0 Looking for actual user opinions on Telephony card
5:02PM 1 Inbound SIP calls work, just not when making calls between extensions.
5:01PM 0 Asterisk CallCompletion dialplan
4:50PM 1 terrible MeetMe sound with
4:07PM 3 fail-over server
3:34PM 0 SIP registration
2:52PM 1 forward calls by the ports
2:36PM 1 Set variable on Call Answer
12:45PM 2 Call files error
11:09AM 2 Call Recording audio file quality query
Monday February 7 2011
6:56PM 1 multiple inbound calls from same sip trunk
6:27PM 1 OT: SwitchVox Mailing List?
5:56PM 1 Codec negotiation
5:49PM 1 remote bridging
4:45PM 1 IAX channel name incorrect - Found in 1.2 still happens in 1.6
4:13PM 1 downgrade libpri
3:38PM 1 About maxlen parameter in queues
1:42PM 1 Error: Unable to create channel of type 'SIP'
Sunday February 6 2011
9:51AM 0 secure sccp
Saturday February 5 2011
9:39PM 1 Any voice changer applications for Asterisk?
11:07AM 11 Callback through extensions.conf?
9:24AM 2 Zaptel slow to detect remote hangup
Friday February 4 2011
10:25PM 1 SoftHangup on asterisk
1:02PM 3 MP3 Crashing Asterisk
10:31AM 0 problems with voicemail and centos 5
9:53AM 2 voice quality measurement using dahdi_monitor
5:43AM 2 Email alerts for trunks (peers)
5:41AM 3 PRI voice optimization
2:43AM 1 standalone NOTIFY message handling for Asterisk
Thursday February 3 2011
7:45PM 8 Question about EuroBRI final 2 digits
6:30PM 2 T.38 negotiation error
3:53PM 2 Queues and Agent penalty - how to go to second best agent when the first does not answer
3:48PM 1 MeetMe and admin users
10:41AM 1 sip trunk balancing
10:18AM 1 [newbie] Conference call
9:44AM 1 Radius Based Accounting for Asterisk
1:18AM 0 Regarding bob-invite-alice xml scenario
Wednesday February 2 2011
11:42PM 1 Problems using Background within a macro on V 1.4
8:44PM 5 Regarding asterisk
8:14PM 1 asterisk18 rpm issues
6:26PM 1 Outgoing agent´s calls
4:50PM 0 SIP Originate on 1.8.X
5:43AM 1 AGI script exits non-zero when running system command
4:14AM 0 regarding sip.conf and extensions.conf
3:34AM 8 how to get Current Calls details
Tuesday February 1 2011
10:22PM 3 Asterisk Performance
5:34PM 1 Upgrade and recompilation
4:37PM 1 How to use Monitor() in Python AGI
4:34PM 1 How to load new musiconhold classes ?
4:22PM 0 Connecting to Cisco Iad2430 to Asterisk
4:05PM 2 Musiconhold priority
12:22PM 0 How to Change The Caller Position in Queue
9:02AM 1 Playback in uplink and recording in downlink
3:09AM 0 regarding error in asterisk