similar to: is pattern matching inside macro valid?

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 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 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, 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)

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

Hello. I've got a problem logging in to a Sparc Solaris 10 machine with public key authentication. I searched, and found a similar problem report at <http://thread.gmane.org/gmane.network.openssh.devel/12694>. For that guy, the problem had to do with LDAP. My user accounts are also stored in LDAP, an OpenLDAP server, to be exact. That server runs on the same machine as the machine

Greeting- I have a mixed network of ms-windows, macintosh and freebsd systems. I am setting up a FreeBSD 9.0 system as a PDC using samba. I can from a FreeBSD box attach to the SMB server as a user that is defined on the Samba Server. [wynkoop at dt0 ~]$ smbclient -L hp1 Enter wynkoop's password: Domain=[HARAPARTNERS] OS=[Unix] Server=[Samba 3.6.4] Sharename Type

Hello, Simply put, I'm trying to call a function "testme" with value "age=NA". I wish to use "dotlist<-list(...)" inside the function and have dotlist become: $age [1] NA I'm modifying existing code and need to minimize changing that code so it's easiest to conform how I call the existing function. My sample code fragment, results, and R.version

Hello, I would like to ask the llvm developers to have a look at http:// llvm.org/bugs/show_bug.cgi?id=918 . This bug has been reported 4 month ago but is none the less a somewhat serious one. Below I have pasted the test case and output of the issue running on my ppc machine. thank you Eric pb:~ eric$ cat testme.ll;llvm-as -f testme.ll;llc -march=c -f testme.bc;gcc -c testme.cbe.c

Hi folks, I've got trouble doing an blind transfer from an "EXEC Queue quename|t" in an AGI script. Attended is working fine, also when doing the same queue from the extension.conf file is fine. Here's my log; -- Executing AGI("IAX2/utv01-5", "agi://localhost/queuecall.agi?queue=vxl") in new stack -- AGI Script Executing Application: (QUEUE)

Hi, I tried keeping it for a very long time and put logs there and it always went to not adapting. This is how I initialize speex_echo_cancel SpeexEchoState* _echoState = speex_echo_state_init(160, 320*10); I have 20 msecs of audio with 16bits for a sample. If two frames are identical and are given to speex_echo_cancel for many iterations, should I expect silence. Thanks -Anurag Quoting

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

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 to the problem? I wonder why it would not find its mirror image as an echo and do echo cancellation? Any insight is appreciated. If you need some more data, tell me. Thanks -Anurag Quoting ac2491@columbia.edu: >

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

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