asterisk users - Feb 2012

Wednesday February 29 2012
TimeRepliesSubject
6:37PM 0 GSM gateway call redirect
5:42PM 4 asterisk distributions
5:07PM 1 Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2
2:52PM 3 Getting Ulimit Message after restart asterisk service
3:56AM 1 outbound fax over t38 gateway can't pass
12:44AM 2 10.2.0-rc2: permitted contact can't register.
 
Tuesday February 28 2012
TimeRepliesSubject
9:08PM 4 Same provider - IAX sounds bad, SIP sounds great
3:06PM 1 better timing source for an asterisk gateway
12:59PM 4 Asterisk auto-dial out a SIP .
12:52PM 1 asterisk MCID detection
11:58AM 1 Asterisk Version 1.8.9.2 Question About SIP/SRTP/TLS
6:23AM 1 Alphanumeric DTMF !?
12:00AM 1 TCE400P diagnostic messages
 
Monday February 27 2012
TimeRepliesSubject
8:00PM 1 Capture sip Response
6:28PM 2 CDR Analyzer/Queue stats reporting
6:07PM 0 AstLinux 1.0.2 Release
3:23PM 0 dahdi timing
9:53AM 0 Correct call duration when transfer a call
7:19AM 0 runtime codec selection
12:00AM 2 pstn bridge to asterisk - phones connected to pstn stop ringing when asterisk answers
 
Sunday February 26 2012
TimeRepliesSubject
10:32AM 2 Possible bug (or feature?) in extension matching and parking feature
 
Saturday February 25 2012
TimeRepliesSubject
2:56PM 0 Asterisk and realtime
1:06AM 0 No IVR audio. Jump in RTP sequence number
 
Friday February 24 2012
TimeRepliesSubject
9:32PM 2 Transfer to fax
6:32PM 1 Best CRM for Asterisk
6:17PM 1 cell mysql odbc support
12:32PM 1 View Uniqueid of Active Calls
4:57AM 3 Replicating SIP registration Info between active to standby
4:08AM 1 Where can I find some good examples of listening to AMI events via PHP & how to listen to a specific event?
 
Thursday February 23 2012
TimeRepliesSubject
9:21PM 3 Trunking betweeb two Asterisk System
8:23PM 0 Transmit NOA (sss) to Dialogic IMG via SIP / Transmisión de NOA hacia Dialogic IMG por SIP
7:44PM 2 Rejecting transfers to in-use parking spaces
7:36PM 1 Starting asterisk: Cannot find specified TTY
6:30PM 0 Asterisk 10.1.3 Now Available
6:30PM 0 Asterisk 1.8.9.3 Now Available
4:09PM 1 Is Asterisk 10 available in Digium Repository? I doesn't show up
3:41PM 1 Hot desking and presence
2:57PM 1 app_rpt and chan_usbradio removal from trunk
2:35PM 1 Fwd: Block Collect Calls on ISDN trunk
12:54PM 2 DECT with handover
6:18AM 1 p-associated-uri in 200OK
5:53AM 1 Phone Inventory
 
Wednesday February 22 2012
TimeRepliesSubject
8:32PM 2 codec mismatch on channel
5:09PM 3 dahdi and digium debian package
3:37PM 0 Disconnect after 12 seconds w/Cisco 303g Phones
2:40PM 5 AGI: blocking script until playback complete
2:24PM 1 How does format_mp3 work?
12:48PM 1 Postgresql in Asterisk
12:40PM 1 Asterisk 1.8.x app_meetme.so
 
Tuesday February 21 2012
TimeRepliesSubject
8:55PM 4 Praking lot issues.
8:34PM 2 Streaming musiconhold via mpg123
6:58PM 0 Set T38 protocol
4:52PM 2 Define custom vm-login sound file per VM context?
3:23PM 1 asterisk-users@lists.digium.com Nacha Alert ID10416
1:30PM 4 how many UDP ports is required for 1 call
9:09AM 0 Reffered By header is missing from SIP INVITE in call transfer scenarios
 
Monday February 20 2012
TimeRepliesSubject
11:33PM 3 Park and PARKINGDYNAMIC
3:48PM 1 Matching asterisk PBX cdrs to Telco's Trunk CDR's
12:51PM 1 Multiprocess Asterisk
 
Friday February 17 2012
TimeRepliesSubject
11:33PM 0 Calling SIP extension through Local/XXXX negotiates slin codec instead of gsm
7:41PM 1 asterisk network connections
6:15PM 0 Troubleshooting realtime LDAP
1:40PM 2 SER Still recommended for large installs?
1:04PM 0 Presence subscription from other pbx systems
9:00AM 4 Is there any way to make call fail after # of rings?
8:28AM 1 Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network?
2:26AM 2 Set(CALLERID(name)) when incoming call is "anonymous"
 
Thursday February 16 2012
TimeRepliesSubject
4:09PM 0 IAX DNS SRV
3:55PM 2 How to receive SMS ?
3:44PM 1 Park() ignores 'r' option which should disable music on hold in favour of ringing tone
3:24PM 0 Aastra phone dial plan
7:16AM 2 Asterisk && RTCP
5:13AM 8 Dahdi Answer a Call On ringing State.
 
Wednesday February 15 2012
TimeRepliesSubject
10:42PM 7 Block Collect Calls on ISDN trunk
8:03PM 3 OT - "T.38 unreliable on a LAN" : truth or obscurantism ?
2:26PM 0 Asterisk + Avaya (CM5.2) H.323 trunk Link (Dustin fails)
1:21PM 0 OT - Which cheap ISDN phone
1:02PM 2 Forwarding queue to remote agent over PSTN
12:11PM 1 error during dahdi installation on centos
 
Tuesday February 14 2012
TimeRepliesSubject
9:45PM 0 Reading second rdnis
8:33PM 1 "conferenced" transfers
7:56PM 2 Asterisk + Avaya (CM5.2) H.323 trunk Link
5:18PM 3 Call holding with chan_capi
3:30PM 1 Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2
11:49AM 1 chan_capi audio weirdness
11:07AM 0 How to associate agents - extensions?
10:56AM 1 How to implement outlook popup
2:58AM 0 Calling from extension that I don't create
 
Monday February 13 2012
TimeRepliesSubject
4:17PM 0 Log SRC & DST IP address and PDD
3:11PM 1 Problem with libpri / asterisk
11:30AM 2 India Pune Pri call problem
12:02AM 2 No valid transports available, falling back to 'udp'.
 
Sunday February 12 2012
TimeRepliesSubject
7:57PM 2 Polycom IP331 Configuration
5:13AM 0 CEL ODBC Issues Asterisk 1.8.9.2
 
Saturday February 11 2012
TimeRepliesSubject
11:54PM 1 Should you "ever" use nat=no?
4:50PM 0 Spurious DTMF recognition problems.
4:48PM 1 New router, registration problems
4:28PM 0 a1zax th99sh
1:22PM 1 Asterisk perl AGI confusing variables
1:03PM 1 What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?
11:17AM 0 Regional settings with Sipura SPA-3000
10:41AM 1 Stumped about adding a semi-colon to a variable
2:18AM 3 Virtual Server
 
Friday February 10 2012
TimeRepliesSubject
10:30PM 3 Polycom firmware 4.0.1 and paging
7:37PM 1 Call queuing behavior
6:26PM 1 Asterisk SIP Realtime Architecture Issue/Bug
5:32PM 1 Call Completion
4:34PM 5 Question for the group
1:40PM 0 distributed queue information over several Asterisk nodes
11:30AM 1 DTMF forwarding and Page
11:13AM 1 dial plan with hangup cause 34
 
Thursday February 9 2012
TimeRepliesSubject
9:11PM 0 Turning off splash ring on PAP2T
8:19PM 0 T.38 Incoming Fax Problem
6:50PM 0 Problem with SIP phone outside local network
4:33PM 1 Garbled voicemail
3:36PM 1 Is there a php script to analyse and show call detail reports from Asterisk CDR?
2:36PM 4 checking if a phone number is UP
2:02PM 2 Stuck DAHDI Lines
10:08AM 2 Help with Codes and Polycom Phones
5:24AM 2 Asterisk SIP Realtime Architecture Issue/Bug.
 
Wednesday February 8 2012
TimeRepliesSubject
8:48PM 1 (last call for comments) Proposed changes to Asterisk release and support cycles
7:55PM 0 HT286 dialtone and ring cadence
7:46PM 0 Problem callerid ignored by using callfiles
5:43PM 0 finally, an open alternative to Viber
12:48PM 2 Calling a group of phones and force the speaker
10:42AM 0 SIP trunk audio bad but is OK again after SIP re-registration
9:35AM 2 Automatic Number Identification and anonymous calls
9:01AM 4 SIP hardware phones
 
Tuesday February 7 2012
TimeRepliesSubject
8:27PM 1 Asterisk 1.8.9.1 Now Available
8:27PM 0 Asterisk 10.1.1 Now Available
11:38AM 14 Asterisk V/s FreeSwitch
10:53AM 1 Early Media configuration doesn't seem to be working
10:38AM 1 How to Send SMS on SS7 DChannel-16(Signaling Channel)
10:23AM 1 TE410P (1st) without cables always green
4:01AM 2 Headset Options
1:54AM 0 Are there any ATAs that support IP6?
 
Monday February 6 2012
TimeRepliesSubject
11:24PM 1 res_http_post.so questions
7:18PM 2 Custom extension: dial a queue
6:05PM 1 Playback with noanswer in AGI
5:22PM 1 BETTER_BACKTRACES
3:31PM 3 Script to automatically update externip. Useful for a host with dynamic public IP
2:30PM 0 Fwd: Re: Asterisk CLI unresponsive
2:19PM 1 What packages are required to getcdr_adaptive_odbc to be compiled in Asterisk?
2:04PM 4 Major changes between 1.4/1.6.1.8/10?
1:43PM 0 Dahdi, PRI and all circuits are busy now
1:39PM 2 What packages are required to get cdr_adaptive_odbc to be compiled in Asterisk?
1:33PM 0 Why doesn't this manager.conf code work on Asterisk 1.6.2 and 1.8.9?
11:50AM 2 Asterisk 10 and DUNDi, Extended Support?
10:45AM 1 Fwd: Re: Asterisk CLI unresponsive
9:55AM 1 Callmanager 4 Asterisk Malformed/Missing URL
8:59AM 0 Driver for TOR3E ( Govarion ).
 
Sunday February 5 2012
TimeRepliesSubject
4:45AM 4 Your buddy Edwin needs your help!
 
Saturday February 4 2012
TimeRepliesSubject
6:38PM 1 Inclusion of the ILBC codec starting from asterisk 1.8.10/10.2
4:31AM 1 FW by postmaster@datavox.co.uk : Delivery Status Notification (Failure)
 
Friday February 3 2012
TimeRepliesSubject
8:28PM 2 Junghanns QuadBri install help
6:22PM 0 Hangupcause on DAHDI 2.4.9-svn-r9328 channels - Asterisk 1.4.36
5:36PM 1 Queuemember status before calling the Queue command
4:38PM 1 Recording the follow-me calls?
4:29PM 4 [asterisk-dev] How to play audio file in background in dialplan?
12:53PM 1 Can someone tell me what is this issue ?
11:52AM 0 Recording the follow-me calls
10:09AM 2 Asterisk CLI unresponsive
 
Thursday February 2 2012
TimeRepliesSubject
11:54PM 1 Web and Email Chat
10:56PM 1 Quick bash tip for finding free SIP extensions from your sip.conf
8:51PM 2 Asterisk version that support Database Configuration
7:14PM 1 T38 faxing - UDPTL creation failed
3:51PM 1 amd detect answering machine
10:24AM 1 MixMonitor and ChanSpy
8:45AM 1 asterisk dahdi problem.
2:14AM 2 externip nat audio sip trunk issue problem
2:09AM 1 FXS hangup issues
12:48AM 6 Is this doable?
 
Wednesday February 1 2012
TimeRepliesSubject
10:18PM 0 Asterisk-users caller ID
7:17PM 3 Router that support Asterisk
7:16PM 2 Getting one way audio even NAT is configured
5:42PM 1 Dynamically toggling ConfBridge recording from conference menu
4:20PM 1 Asterisk 1.8.9.2 Now Available
4:20PM 0 Asterisk 10.1.2 Now Available
12:29PM 2 SIP Provider Russia, Ukraine, Poland
11:38AM 1 Asterisk 10.0 Realtime
10:34AM 2 read digits during recording / DTMF in conference?
1:12AM 0 Congestion outbound only with ATA boxes