similar to: How to execute an AGI script for each call.

Displaying 20 results from an estimated 600 matches similar to: "How to execute an AGI script for each call."

2014 Jun 26
1
Executing an AGI python script in Asterisk after call is bridged.
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
2014 Jun 27
4
Attack on Sip server.
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
2014 Jun 25
1
Echo Cancellation when calling from softphone to mobile.
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
2014 Jun 26
1
Changing recorded file storage directory.
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:
2014 Sep 07
2
Pattern Extension not working in Dialplan
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(${AGE})} >
2014 Sep 28
2
How to append the recording file.
Hi All, I am trying to record the call using MixMonitor. exten=>_XXXX,n,MixMonitor(${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
2014 Sep 17
1
${ANSWEREDTIME} returning null
Hi, I am initiating a call using call files. In 'h' extension I am trying to collect the value of ANSWEREDTIME variable but it is returning null. While It works fine when call is not generated using call files instead is generated from softphone. any idea what might be wrong? thanks Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was
2014 Jul 13
1
Recording sound.
Hi All, I am calling mobile numbers from Soft-phone and recording the call. In recording the level of sound from the receiver's side is perfect (loud enough) but my voice's sound level is very weak. I barely can hear it. During the call receiver is able to hear me. But in recording my part of conversation is barely audible. I am recording using MixMonitor(). Is there anything that can
2014 Sep 08
0
is pattern matching inside macro valid?
Can't we use pattern matching inside a macro? Because when I am trying to do so call is terminating even for a very simple dummy dialplan. [demo3] exten=>98,1,NoOp() exten=>98,2,Macro(testme) exten=>h,1,NoOp(terminating call); [macro-testme] exten=>s,1,Playback(Digits/2) exten=>s,2,WaitExten(15) exten=>s,3,NoOp() exten=>_X,1,NoOp(${EXTEN}) exten=>_X,2,Goto(s,3)
2006 Jul 19
2
echo cancellation seg faults
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
2006 Jul 19
2
echo cancellation seg faults
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
2014 Mar 06
2
Regarding GSOC 2014
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
2006 Jul 18
2
echo cancellation seg faults
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
2002 Sep 01
3
Unable to print
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
2013 Apr 05
1
Using hmac-sha2-256 in OpenSSH 6.2p1
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
2002 Aug 30
1
Printing in Wine
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
2010 Aug 28
5
native ZFS on Linux
This just popped up: > In terms of how native ZFS for Linux is being handled by [KQ > Infotec], they are releasing their ported ZFS code under the Common > Development & Distribution License and will not be attempting to go > for mainline integration. Instead, this company will just be > releasing their CDDL source-code as a build-able kernel module for > users and
2006 Jul 19
0
echo cancellation seg faults
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
2006 Jul 19
0
echo cancellation seg faults
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: >
2014 Aug 30
0
transfering call to dialplan without disconnecting.
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