similar to: ${ANSWEREDTIME} returning null

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 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 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 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, 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, 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)

Hi All, I am calling mobile numbers from Soft-phone and recording the call. In recording the level of sound from the receiver's side is perfect (loud enough) but my voice's sound level is very weak. I barely can hear it. During the call receiver is able to hear me. But in recording my part of conversation is barely audible. I am recording using MixMonitor(). Is there anything that can

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

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)

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

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, 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

Hello, I am using Wine to run Tally, an accounting package. Over the network, we are able to print documents using OpenOffice as we have CUPS installed. I read the DOCS related to printing in Wine and they say, I dont need to configure wine if I have CUPS installed. But whenever I issue the print command in tally, nothing happens. I checked out /var/log/messages but didn't find any

Hello, I saw that Asterisk don't calcultate fine the ANSWEREDTIME. I want that when ANSWEREDTIME =~ 5.6 become 6 and if =~10.3 become 10 because, now, if ANSEREDTIME =~ 15.9, it become 15! it isn't correct How can I have a rounded ANSWEREDTIME ? Where have I to manipulate the sources? thank you -- Francois -------------- next part -------------- A non-text attachment was scrubbed...

No matter what I do, ${ANSWEREDTIME} is always 0, even on the most simplest dial plan such as: Using Asterisk 1.2.13 exten => 77,1,Answer exten => 77,2,Playback(custom/dax/S300) ; one minute file exten => 77,3,Noop(${ANSWEREDTIME}) exten => 77,4,Hangup -- Executing Answer("SIP/5402-b7b45f58", "") in new stack -- Executing

Hello, I saw that Asterisk don't calcultate fine the ANSWEREDTIME for me. I want that when ANSWEREDTIME =~ 5.6 become 6 and if ANSWEREDTIME= 10.3 become 10 because, now, if ANSWEREDTIME =~ 15.9, it become 15! it isn't correct I could manipulate the app_dial.c to have my own result. But do you think that my idea is correct because, If a call is 15.999999 Sec it become 15Sec. For a

Dear all, I tried to make a call with extensions.conf. exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME}) exten=> _00[1-9].,102,Hangup But the 2 and 102 will not be executed. So I can get the correct answered time via 2. Is any idea about it? Is it the problem of my ZAP channel's configuration? My zapata.conf is as below: