similar to: Can a duration limit be specified in spool call file?

Could any one suggest me in how to get the duration of an OggVorbis File.I am really caught up in this, as it seems to be complicates as the OggVorbis is a variable bit rate. Thanks in advance Raj --------------------------------- Yahoo! India Matrimony: Find your partner now. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good afternoon list. I am experiencing a problem with the CDR and callfiles. What is happening is this: When generating a call with a callfile, everything works perfectly, but the CDR is recorded in the table when they answer the call destination. The field disposition is being recorded correctly, but the duration field is marked with the ring time and billsec is marked with 0. This just happens

Hello, I see the strange behaviour that outgoing calls end after 15 minutes. I didn't knew there is some kind of call duration limit that can be set ? Is there ? Using Asterisk 1.8.12.2 Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on

Hi everyone, I'm new to Asterisk, but I like it ;o) Have a question to you; How can I limit the incoming call duration? -- Suich

Hey Roger and everyone else commenting – I appreciate all the advice and questions. I now have more work than before 😊! Greetings from sunny South Africa! Sent from Mail for Windows 10 From: Roger Hågensen Sent: Thursday, May 2, 2019 12:12 PM To: icecast-dev at xiph.org Subject: Re: [Icecast-dev] Webm files written without duration in header On 2019-05-01 11:58, Sytze Visser wrote:> I am

Hi, We are running 1.4.22 and have been experiencing problems with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone duration. We have tried setting chan_dahdi.conf parameter 'toneduration', but that does not do

Hi, After some testing I've found out that my client's hardware recognizes DTMF only if digits are sent 50ms apart with 50ms of tone duration. This was tested using a test device which generates DTMF. Now asterisk doesn't do it by default because digits going out from Asterisk are not being recognized. Using command sendDTMF, I can control inter-digit duration, and using

Hi all, I've got a question on regarding how to find out the duration of an ogg vorbis file. Sorry if it should already be mentioned somewhere in the docs, but I couldn't find out how to do it the best way. So, I can read the [bitrate_nominal] field from the identification header (if it is set), find out the size of the audio data (by subtracting the size of the three headers from the

Hi All, I have a data frame: Time1 Time2 1 176:46:10 41:48:06 2 171:28:57 61:19:10 3 178:25:15 34:05:35 4 74:04:20 25:01:55 5 136:11:20 37:59:32 6 138:17:17 30:22:27 7 183:04:48 29:25:02 8 179:35:01 19:29:44 > str(df) 'data.frame': 8 obs. of 2 variables: $ Time1: Factor w/ 583 levels

I was just going through my call records ( stored in mysql database by cdr_MYSQL module ) and saw a record having duration = 0 and billsec of more than 50 seconds . I did a query on cdr where duration < billsec and saw that there were infact some 250 records with duration less than billsecond ( table had around 4,00,000 records) . Did anyone came across this ? I also checked csv files and they

Hi, I am using zap channels, and by using sendDTFM application, I can control the duration between two DTMF digits, but I can't find a way to control the duration of the digits themself. Did search on the Internet and found out that I can change it in the asterisk source files and recompile asterisk. Wiki also says that it can be controlled using toneduration option in zapata.conf, but it

I am writing a program that retrieves and plays .opus files over the network. I am using the C# PInvoke wrapper OpusFileSharp, which calls into libopusfile and passes callbacks for read, seek, and tell. I want to enable seeking via the seek callback by not returning -1, but I find that as soon as I do libopusfile requests the last 64K of the .opus file, probably so it can calculate the

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

Hello, > (1) Split the content into chain segments > (2) extract the first and last data packet from each segment, for all tracks > (3) recover / calculate the granulepos values for the first and last data packets (*) > (4) convert the granulepos values from the first and last data packets into time values > (5) take the largest difference between start and end time as the total

Hello, I am wondering if someone has got any ideas that can help solve this problem. I have a dial plan that you call into, and depending on certain conditions it calls out on a number grabbed from a database. Something like this : exten => s,n,Do something exten => s,n,Do something else exten => s,n,Dial(ZAP/g1/${OUTBOUND},${timeout}) I need to log the time the person

On Aug 10, 2018, at 1:28 PM, Marshall Clow via cfe-dev <cfe-dev at lists.llvm.org> wrote: > > * The clock stuff being added in C++20 has already been discussed here. I’ve missed the discussions on file_time_type, however I thought I should throw in my opinion here before it is too late to do anything about it. I believe it is a mistake to model file_time_type with 128 bits. It

Hi I am using Trixbox 2.4 and PRI lines..on the CDR i see many calls that have duration of 0 seconds, but they are still shown as ANSWERED . how come its possible when duration is 0.00 ? Are the callers billed for such calls ? Rgds Sriram -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am having some trouble calculating the duration of my speex files. I am writing a speex player in Java and I am using the JSpeex port of the codec which works but doesn't give me the audio duration which I need to show progress or implement a scrubber. JSpeex appears to be all but dead but I don't see any other option for Java playback. I have tried the method posted here:

Hi ppl, we've been using a header called X-Content-Duration on some applications to communicate between a client and a server about the duration of a video (or audio) file. I've just been told by some W3C people that we should not name HTTP headers (even if intermediate) with "X-". (see email thread excerpt below) Should we move forward and register a provisional header as