similar to: Recording sound.

Hi All, I am trying to record the call using MixMonitor. exten=>_XXXX,n,MixMonitor(${EXTEN}.wav,b) What i want to do is- when first time a call is made to some number say 1100, a new file (1100.wav) is created. When call is made 2nd or 3rd time, no new file is created instead call recording is appended to file created in above step. Now I know that 'a' option is used to append the

Hi All, In asterisk, default directory to store the call-recording files is /var/spool/asterisk/monitor. Can we change this directory? How? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All. Someone is attacking on my SIP server. There are lot of requests coming in and I am not able to stop it because I am unable to detect the IP address. I used wireshark to capture the packets. Although I am using very strong password for my SIP users but still is there any way to drop these packets and stop this attack. I tried dropping packet after matching some string (most of the

Hi All, There is an option of starting the recording of call after the call is bridged. [ b option]. Is there any way of running an AGI script only if call is bridged otherwise not. Thanks -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part

Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in

Hi, I created a dummy dialplan where I ask the user to enter the age. [macro-age] exten => s,1,Background(my/age) ;;Play recorded message to enter age exten => s,n,WaitExten(10) exten => _XX,1,Set(AGE=${EXTEN}) ;; this line is not executing, instead dialplan is terminating with error given below. exten => s,n,NoOp(${AGE}) exten => s,n,GotoIf($[${LEN(${AGE})} >

Hi, I am initiating a call using call files. In 'h' extension I am trying to collect the value of ANSWEREDTIME variable but it is returning null. While It works fine when call is not generated using call files instead is generated from softphone. any idea what might be wrong? thanks Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was

Hi All, I am trying to execute some AGI script no matter what extension is called. There is 'h' extension to call AGI script when any call hangs up no matter what extension hangup. for example -> [some-context] /// something here which call AGI script no matter what extension receive call. exten => 111,1,Dial(SIP/111) exten => 112,1,Dial(SIP/112) exten => h,1,AGI(pt.py)

Can't we use pattern matching inside a macro? Because when I am trying to do so call is terminating even for a very simple dummy dialplan. [demo3] exten=>98,1,NoOp() exten=>98,2,Macro(testme) exten=>h,1,NoOp(terminating call); [macro-testme] exten=>s,1,Playback(Digits/2) exten=>s,2,WaitExten(15) exten=>s,3,NoOp() exten=>_X,1,NoOp(${EXTEN}) exten=>_X,2,Goto(s,3)

Probably the level of your signal is too low and/or you're just not letting it time to adapt. Jean-Marc Le mercredi 19 juillet 2006 ? 19:00 -0400, ac2491@columbia.edu a ?crit : > On closer looks and debugging I always end up in > > speex_echo_cancel function with comment > /* Temporary adaption rate if filter is not adapted correctly */ > > > Does this give any clue

Hi, If I pass the same ref and the echo data to the echo cancellation API, I am expecting silence as output. I get back the original audio data. Is this correct? Thanks -Anurag Quoting ac2491@columbia.edu: > Hi Jean, > > I got the earlier problem fied with correct NN and tail values. > But > I dont see any echo being cancelled. To the echo cancel API I am > giving, audio

Sir, I am a 4th yr undergraduate student pursuing my BTech in CSE at IIIT Hyderbad, India. I am interested in applying for Xapian in Gsoc 2014. I had gone through this year's idea page and interested in applying for 'posting list encoding improvements' project. I am good at C/C++,python; which is one of the requirement. I had done gone through the information Retrieval and

Hi, For my VoIP application machine A sends speex encoded audio of to machine B and vice versa at. Data is captured in PCM 8Khz, 16 bit and then encoded using speex 1.1.12 The packet A played and the packet A captured through mic are the input to speex echo canceller. So I am trying to remove traces of packet A played from the captured data. I have followed example testecho.c All I hear is some

I tried the suggested changes, but still haven't been able to compile the branch. Here's the log: http://pastebin.com/HR17USXR Thanks, Satwant Rana On Sun, Mar 30, 2014 at 2:37 PM, Gaurav Arora <gauravarora.daiict at gmail.com>wrote: > Hello Satwant, > > This seems to be problem with doxygen installation in the bootstrap > script. Source seems to be broken and not

Hi, I am using Microsoft Word Viewer to view documents via wine. CUPS is installed and working perfectly. But when I try to print via Word Viewer, it says no printer found :(. I checked the config file in wine, and its shows that CUPS is present. Kindly let me know how to get it working :) -Regards Anurag __________________________________________________________ Give your Company an email

Hi All, I am trying to build a small setup using asterisk where user1 calls the user2 and after a conversation of few minutes user1 puts the call on automation i.e. after few minutes user1 should be able to kind of transfer the call to dialplan where the call proceed as per user2's DTMF input and dialplan structure (without disconnecting the call). How can this be achieved? Thanks Anurag

Hi everyone, I was working on the Paice-Husk Stemmer, which is a Bite Size Project for Xapian, and I have created a C++ as well as Snowball version of it. I read the algorithm, and picked the rules from here: http://www.comp.lancs.ac.uk/computing/research/stemming/paice/descript.htm The C++ code takes rules as input from a file and generates the stem of given word, whereas the Snowball version

I was able to successfully make the master branch of xapian, but I can't do the same for svn/clustering branch. The bootstrap fails with this log: http://pastebin.com/D1hbLp7k Can someone who has successfully installed the clustering branch tell me what am I doing wrong here? Thanks Satwant Rana -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, is there a way of calculating of measuring dependence between two categorical variables. i tried using the chi square test to test for independence but i got error saying that the lengths of the two vectors don't match. Suppose X and Y are two factors. X has 5 levels and Y has 7 levels. This is what i tried doing >temp<-chisq.test(x,y) but got error "the lengths of the two

Hi, I could not use hmac-sha2-256 in OpenSSH 6.2p1. I tried configuring in sshd_config file also, but the server was not starting. How can I use hmac-sha2-256 & hmac-sha2-512 in OpenSSH server in accordance with RFC 6668? I have installed OpenSSH in a computer with the following configuration: Architecture: x86 32-bit OS: RHEL AS 4 (Nahant update 4) (Linux version 2.6.9-42.EL) Thanks and