asterisk users - Feb 2011

Monday February 28 2011
TimeRepliesSubject
5:19PM 0 Asterisk 1.8.3 Now Available
5:18PM 0 Asterisk 1.6.2.17 Now Available
5:17PM 0 Asterisk 1.4.40 Now Available
4:39PM 2 CEL and PGSQL
3:27PM 14 Failover Routing
1:40PM 3 Asterisk 1.8.3-rc3 and one way audio
10:33AM 12 asterisk security....again
10:25AM 0 Obi110 as gateway to PSTN?
10:24AM 7 Two Asterisk machines for redundancy
2:12AM 11 Using voice modem as poor man's FXO in Asterisk 1.8
 
Sunday February 27 2011
TimeRepliesSubject
12:32PM 2 [Dahdi 2.4.0] Flash() hangs up
 
Saturday February 26 2011
TimeRepliesSubject
10:33PM 2 Need to buy the Digium card, to confirm
4:16PM 1 SRTP Error Message
9:08AM 2 Detect DTMF tone during call?
 
Friday February 25 2011
TimeRepliesSubject
10:49PM 0 T1 channel audio control
9:47PM 0 Asterisk 1.2 zap hangup issue
7:30PM 8 Asterisk/Skype
5:37PM 0 PRI B-Channel restarting itself continually
5:04PM 8 [OT] Yealink IP Phones
4:16PM 6 1.8.2.4: SIP dialogs not killed?
12:40PM 4 Handle in dialplan user disconnection
 
Thursday February 24 2011
TimeRepliesSubject
10:27PM 5 missing argument on AGI
9:05PM 5 Recieve_Fax caused crash 1.8.2.3
8:47PM 1 Using a Virtual IP Line
8:31PM 2 extensions.lua with luasql.mysql.
6:41PM 10 Paging with Polycom 3.3.x
4:32PM 1 Debug Dropped Audio
2:58PM 0 "Asterisk" caller ID
2:51PM 6 Google Voice outbound Caller ID broken
1:42PM 1 RTP (voice) issue. STUN server
1:24PM 3 Registration failed though configured.
12:36PM 9 [1.4] Still can't get it to call back
11:38AM 2 Carrying context from one server to another?
10:56AM 3 [1.4.39.2] Simple AGI doesn't reply
10:15AM 4 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
TimeRepliesSubject
7:56PM 0 SIP friend name
4:10PM 29 REFER and dialplan broken (as documented in chan_sip.c on line 11951)
3:31PM 4 secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1
2:14PM 0 Adhearsion 1.0.1 Released
10:39AM 5 AMI FullyBooted issue
10:17AM 3 extend the timout on ringing for pri or sip
 
Tuesday February 22 2011
TimeRepliesSubject
9:34PM 3 Multiple public address to one Asterisk server behind NAT?
6:21PM 4 calls between iax and sip
2:57PM 0 Weird Inbound Problem.
1:02PM 5 Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
10:39AM 4 [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe
10:06AM 0 AddQueueMember and stateinterface question
3:11AM 1 NVFaxDetect causing segfault
 
Monday February 21 2011
TimeRepliesSubject
10:52PM 2 [Dahdi 2.4.0] DAHDI_CHANCONFIG failed on channel 1
10:46PM 6 Problem installing FXS module in old digium 4 channel tdm card
9:52PM 0 (no subject)
9:45PM 5 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 8 T1 PRI shows yellow/red alarm
6:33PM 0 SIP METHOD BYE
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 3 calls are not going thru e1 line
 
Sunday February 20 2011
TimeRepliesSubject
7:05AM 2 MEMBERINTERFACE and MEMBERNAME questions
5:03AM 0 My new blog http://cciev.ciscovoicetech.com/
2:39AM 0 AstLinux 0.7.6 Released
 
Saturday February 19 2011
TimeRepliesSubject
9:15PM 6 First go at a stock 1.8 install -- where's DAHDI?
9:52AM 3 [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
TimeRepliesSubject
8:59PM 3 no progress indication
5:56PM 4 cmd MySQL
3:36PM 7 [1.4/AGI] CHANNEL STATUS never "down & available"?
3:11PM 5 Meet me recording
11:36AM 21 Assigning an extension to a roaming phone
11:32AM 3 lua -asterisk manual
11:29AM 7 pbx_ael.so: 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 3 DTMF and Snom
7:16AM 2 Trunk grouping
1:33AM 0 Voice mail forwarding enhancement
12:31AM 1 Dial() function
12:26AM 10 Newbie´s question about Asterisk...
 
Thursday February 17 2011
TimeRepliesSubject
6:27PM 5 Polycom Do Not Disturb button and asterisk hints
5:11PM 2 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 2 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
TimeRepliesSubject
8:32PM 2 No ring tone on inbound call - but channel connects fine
7:51PM 25 Polycom IP335
6:24PM 1 Asterisk on a USB with persistence
6:00PM 1 pipe audio stream to external application
4:40PM 5 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 10 function Echo() doesn't work
11:41AM 1 how to diable echo cancellation for sip?
10:45AM 1 Detect #,* DTMF in dialplan
10:32AM 3 Cisco 7945G phone with asterisk
10:13AM 2 Barge in.
10:05AM 3 How to know Caller's last position in Queue?
8:53AM 0 Regarding error in asterisk 1.6.2.16....
5:49AM 7 Connect Asterisk to a cell phone
12:39AM 7 DTMF not detected, time out
 
Tuesday February 15 2011
TimeRepliesSubject
8:47PM 3 Dialplan end of pattern matching question
8:12PM 6 Aastra phones cannot transfer calls?
8:05PM 0 pstack debug asterisk
7:15PM 6 Voicemail email attachment as MP3, with tags containing sender name, number, message number
6:39PM 2 Paging a message. How?
6:15PM 4 Realtime and Local Channel Crash Problem 1.8.3-rc2
5:06PM 4 Lua extensions are not working on asterisk 1.8.2.3
5:06PM 3 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 4 further action after caller in a queue hangs up
4:00AM 6 Fax Woes
3:39AM 9 uptime
1:47AM 3 trunks and phones registered from the same IP
 
Monday February 14 2011
TimeRepliesSubject
11:16PM 2 Asterisk Call File using Local Channel not passing Variable back to Dialplan
9:36PM 35 Hide the plain text password
6:46PM 3 unregistered trunks and registered phones coming from the same IP
4:06PM 1 SIP session timers just on one specific channel
2:16PM 7 issue with some numbers
1:40PM 10 Cisco 7960 & asterisk 1.8.22 ringlist.dat error
12:11PM 1 Possible dumb question: new kernel, new DAHDI?
10:29AM 2 Problems with realtime sip
6:10AM 3 IP ban list by country
 
Sunday February 13 2011
TimeRepliesSubject
2:52PM 11 [modules.conf] Modules still loaded after "noload"
5:59AM 1 Call Files, Variable passing
4:53AM 2 Fax for Asterisk SIP-TDM
 
Saturday February 12 2011
TimeRepliesSubject
11:06PM 1 Transfer Device Data
4:23PM 8 Using files .call or AMI
12:31PM 24 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
TimeRepliesSubject
10:37PM 20 On-Hold Music
8:57PM 3 Asterisk compile option DAHDI SPANS
5:26PM 3 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 11 Asterisk 1.8.3
6:47AM 4 sangoma wanpipe install error
12:58AM 3 meetme conference & playback of random sound file
 
Thursday February 10 2011
TimeRepliesSubject
11:50PM 25 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 8 CDR with unix time.
11:08AM 7 Early audio SIP sequence order question
4:49AM 2 zaptel/dahdi settings for singtel E1 line
3:08AM 3 Unable to make outgoing calls with Internode
 
Wednesday February 9 2011
TimeRepliesSubject
11:56PM 0 Error loading module ????Vi.so
7:31PM 1 AEL Eswitches
7:30PM 6 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 ashishchauhan07oct@gmail.com 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 2 dial option 'g' not working
3:40AM 0 Manual Call Transfer (Perl, Asterisk::AGI, MySQL)
 
Tuesday February 8 2011
TimeRepliesSubject
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: wiki.asterisk.org and code.asterisk.org
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 4 terrible MeetMe sound with 1.6.2.9
4:07PM 8 fail-over server
3:34PM 0 SIP registration
2:52PM 1 forward calls by the ports
2:36PM 2 Set variable on Call Answer
1:30PM 2 ${HANGUPCAUSE} in CDR
12:45PM 6 Call files error
11:09AM 12 Call Recording audio file quality query
 
Monday February 7 2011
TimeRepliesSubject
6:56PM 1 multiple inbound calls from same sip trunk
6:27PM 1 OT: SwitchVox Mailing List?
5:56PM 3 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 6 About maxlen parameter in queues
1:42PM 1 Error: Unable to create channel of type 'SIP'
 
Sunday February 6 2011
TimeRepliesSubject
9:51AM 0 secure sccp
 
Saturday February 5 2011
TimeRepliesSubject
9:39PM 5 Any voice changer applications for Asterisk?
11:07AM 44 Callback through extensions.conf?
9:24AM 4 Zaptel slow to detect remote hangup
 
Friday February 4 2011
TimeRepliesSubject
10:25PM 4 SoftHangup on asterisk 1.8.2.3
1:02PM 6 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 5 PRI voice optimization
2:43AM 2 standalone NOTIFY message handling for Asterisk
 
Thursday February 3 2011
TimeRepliesSubject
7:45PM 8 Question about EuroBRI final 2 digits
6:30PM 3 T.38 negotiation error
3:53PM 7 Queues and Agent penalty - how to go to second best agent when the first does not answer
3:48PM 2 MeetMe and admin users
10:41AM 1 sip trunk balancing
10:18AM 2 [newbie] Conference call
9:44AM 1 Radius Based Accounting for Asterisk
1:18AM 0 Regarding bob-invite-alice xml scenario
 
Wednesday February 2 2011
TimeRepliesSubject
11:42PM 1 Problems using Background within a macro on V 1.4
8:44PM 6 Regarding asterisk
8:14PM 3 asterisk18 rpm issues
6:26PM 4 Outgoing agent´s calls
4:50PM 0 SIP Originate on 1.8.X
5:43AM 2 AGI script exits non-zero when running system command
4:14AM 0 regarding sip.conf and extensions.conf
3:34AM 9 how to get Current Calls details
 
Tuesday February 1 2011
TimeRepliesSubject
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 2 Playback in uplink and recording in downlink
3:09AM 0 regarding error in asterisk