similar to: Is there any variable for Voicemail Password in Asterisk

Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps

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Hi guys this is my Ist mail on this group, I am running asterisk with CentOS 4.4 machine. When i initiate a call then error message apears. calling Number is provided to Asterisk by the php application. Error message appears like this Got SIP response 500 "Internal Server Error" back from 209.47.92.xxxx [Oct 10 23:55:20] WARNING[19289]: pbx.c:4976 ast_pbx_outgoing_exten:

Hi guys I am facing a problem in recording voice on Hangup. I am using php agi class for this purpose. Currently its voice message is being recording when i used to press 1. For this purpose i am using record_file() function with its respective parameters. Is there any way that i can be able to record my voice message on hangup (don't want to send 1 as DTMF to asterisk.). Can you guys guide me

I want to get the count matrix of genes from https://www.ncbi.nlm.nih.gov/geo/query/acc.cgi?acc=GSE146049. Is it possible for GSE146049? After getting counts, I want to do TMM normalization. [[alternative HTML version deleted]]

How can I subscribe to R genomic list? On Sun, 2 Apr 2023, 9:28 pm Peter Langfelder, <peter.langfelder at gmail.com> wrote: > It's a microarray data set, so I don't think you would want to apply > an RNA-seq pipeline. You'd be better off applying a normalization > appropriate for this type of microarray data. > > HTH, > > Peter > > On Sun, Apr 2, 2023

Dear Sir, I am stuck with a nagging problem in using R for SVM regression. My data has 5 dimensions and 400 observations. The independent variables are : Peb, Ksub, Sub, and Xtt. The dependent variable is: Rexp. I tried using the svm.tune function as well as <_tune(svm.....), to tune the hyper parameters: gamma, epsilon and C. Since I am new to R, I am probably not using the svm.tune

Dear Sir/Madam, I am stuck with a nagging problem in using R for SVM regression. My data has 5 dimensions and 400 observations. The independent variables are : Peb, Ksub, Sub, and Xtt. The dependent variable is: Rexp. I tried using the svm.tune function to tune the hyper parameters: gamma, epsilon and C. I am getting the following error message: Error in predict.svm(ret, xhold,

It's a microarray data set, so I don't think you would want to apply an RNA-seq pipeline. You'd be better off applying a normalization appropriate for this type of microarray data. HTH, Peter On Sun, Apr 2, 2023 at 11:09?PM Anas Jamshed <anasjamshed1994 at gmail.com> wrote: > > I want to get the count matrix of genes from >

If Product and Category models are in a habtm relationship, i.e. class Product < ActiveRecord::Base has_and_belongs_to_many :categories end class Category < ActiveRecord::Base has_and_belongs_to_many :products end I want a piece of code to be executed every time a product is connected to a category. Where do I put this code? Which callback (and on which model) will be triggered? Any

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal

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It should be as simple as editing voicemail.conf : ; Voicemail Configuration ; [general] ; Formats for writing Voicemail. Note that when using IMAP storage for ; voicemail, only the first format specified will be used. format=wav49|wav|gsm ; Who the e-mail notification should appear to come from serveremail=asterisk-voicemail ; Should the email contain the voicemail as an attachment attach=yes ;

I am attempting to port CELT to the TI C5515 DSP. I am compiling/assembling/linking with Code Composer Studio 4. I have enabled the TI_55 specific optimizations in my config.h file per this <http://lists.xiph.org/pipermail/celt-dev/2008-December/000033.html> post. However, I am getting several "undefined symbol" linking errors that do not make sense to me. According to the error

I'd like to get VM_CALLERID to include number in addition to name since often when calls come from cell lines or various other, the name is just a city, state and the number would be more usefull. Is there a way to get the number in the VM_CALLERID string, or is there a second variable I can use in formatting email vmail notifications to get the number? Regards, -Dorn

I'm sorry for having to post this, but I can't seem to find any solutions to this, which seems pretty simple, but then again... I've started playing with the example r news project and wanted to start adding some graphs, but replacing: \begin{figure} \vspace*{.1in} \framebox[\textwidth]{\hfill \raisebox{-.45in}{\rule{0in}{1in}} A graph goes here \hfill}

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AXvoice server hacked. Here are few working accounts USE XLITE to make calls.... Registrar/Proxy magnum.axvoice.com:9060 Free Sample account.... username=xMaxwellSmartx secret=thanksapache username=woodsy type=friend secret=haramikuttasala username=wumingzi type=friend secret=kickyourass Enjoy! B.R BaBa Jigger -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce