FreeSWITCH Asterisk VOIP SIP Blog

Time 2020-11-18 07:03:04

Web Name: FreeSWITCH Asterisk VOIP SIP Blog

WebSite: http://jonathanmanning.com

ID:119750

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Asterisk,FreeSWITCH,VOIP,

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Blog Jon Asterisk Training Linux Tutorial Training VOIP SIP After months of searching for a new project car I found a 2004 330i ZHP sedan with just a tad over 100k miles. I found the vehicle at a BMW dealer in Texas from a local trade-in for a 2-series coupe. After quickly making a deal with the selling dealer the car was on a hauler destined for my home address. Within a few weeks of ownership I had a long to-do list of projects with one of them being a speaker update. I reviewed a few different off the shelf options and finally decided on assembling my own parts.To gather the parts required I relied on parts express who sells a variety of speaker parts. I am very pleased with the results as the hi s have a new-found vividness where the worn out OEM Harmon Kardon equipment could no longer reproduce.Here are the list of parts I put together for my speaker replacements.Front DoorsTweeter: Dayton Audio ND20FB-4 Rear-Mount 3/4 Soft Dome Neodymium TweeterBrand: Dayton AudioModel: ND20FB-4Part#: 275-035Mid-range: Peerless by Tymphany NE65W-04 2 Full Range WooferBrand: PeerlessModel: NE65W-04Part#: 264-1046Mid-bass: Dayton Audio ND140-4 5-1/4 Aluminum Cone Midbass Driver 4 OhmBrand: Dayton AudioModel: ND140-4Part#: 290-216Rear DoorMid-range: Peerless by Tymphany NE65W-04 2 Full Range WooferBrand: PeerlessModel: NE65W-04Part#: 264-1046Rear DeckMid-bass: Dayton Audio RS150P-4A 6 Reference Paper Woofer 4 OhmBrand: Dayton AudioModel: RS150P-4APart#: 295-563Adapters Required(2) Pair s of Scosche SABW5251. Starting with the front door remove the door card. There are numerous instructions online to remove the door cars and even some speaker companies have a detailed PDF for removal.2. Install the 5.25 speaker into the SABW525 adapter to install on the front door. To create the connection for the OEM wiring I removed the connector from the OEM speaker and soldered it to the new speaker as seen in the pictures. The OEM screws were not long enough so I replaced them with 1/2 inch 8/32 Philips head screws.3. To install the mid-range in the front door I modified the circular housing by removing the raised vertical bar. This allowed me to push the circular speaker into place. Be careful not to break the lip of the housing, these are fragile. I placed the lip over the side of the table as seen in the picture. I also used the OEM connector by removing it from the stock speaker and soldering it to the new 2 speaker.4. Finally installing the tweeter was a bit different since the OEM speaker is integrated into a bracket so I was creative. I used the styrofoam padding as the mounting point for the tweeter and glued points on the rear mount of the tweeter to keep it in place. Instead of using the OEM connection points I cut them off and crimped standard speaker push on terminals.5. To replace the speakers in the rear door, remove the door card then remove the stock speaker. For this speaker I used the OEM housing by gluing the rim of the speaker to the inside door mount. Once it set for an hour I re-installed it into the door card.6. Finally to install the speaker in the rear deck you will again need the Scosche SABW525. This time however they need to be modified to fit correctly. I used a saw to remove the extended screw terminals as seen in the pictures. Once done it is easy to install the speaker in the factory location. I cut the OEM connector off the car and crimped on push on terminal connectors.I will say the install was not very hard once I had all the proper hardware. The problems I ran into were not having the right screw sizes, having to modify the brackets for the rear deck and a few other items I pointed out. I am very happy with the results and hope this will help other enthusiasts save a few dollars by doing a little work instead of forking out for a plug and play setup. Here is an easy to follow video tutorial on how to install Asterisk on a Rackspace Ubuntu cloud server. I am using the following commands to complete the install below. passwd apt-get updateapt-get upgrade -y && rebootapt-get install linux-headers-$(uname -r)apt-get install build-essential wget libssl-dev libncurses5-dev libnewt-dev libxml2-dev linux-headers-$(uname -r) libsqlite3-dev uuid-devcd /usr/src/wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gzwget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gzwget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gztar zxvf dahdi-linux-complete*tar zxvf libpri*tar zxvf asterisk*cd /usr/src/dahdi-linux-complete*make && make install && make configcd /usr/src/libpri*make && make installcd /usr/src/asterisk*./configure && make menuselect && make && make install && make config && make samples/etc/init.d/dahdi start/etc/init.d/asterisk startasterisk -rvvv If you are looking for a way of screening inbound robo calls I have a great script for you. Using a quick little app you can require inbound callers to press 1 or any digit in order for the call to ring through to your phone. Otherwise the caller gets hung up on. All the audio recordings I used are standard files included with Asterisk core or extra audio files.See below for the context.exten = 6175551212,1,Goto(call-screening,s,1)[call-screening]exten = s,1,Answer(1000) same = n,Set(NUMINVALID=1) same = n,Set(TIMEOUT(digit)=3) same = n,Set(TIMEOUT(response)=3) same = n,Ringing same = n,Wait(3) same = n(catch-all),NoOp(catch-all-inbound-calls) same = n,Read(digit,privacy-unident vm-onefor vm-theperson,1,,2,4) same = n,GotoIf($["${digit}" != "1"]?i,1) ; change the 1 in quotes to the digit you want callers to press. same = n,Playback(one-moment-please) same = n,Dial(SIP/102,45,m) same = n(endcall),Playback(not-taking-your-call) same = n,Playback(vm-goodbye) same = n,Hangup same = n,Wait(1) same = n,Hangup()exten = i,1,NoOp(invalid-option)exten = i,n,Gotoif($["${NUMINVALID}" "3"]?:call-screening,s,endcall) exten = i,n,Set(NUMINVALID=$[${NUMINVALID}+1]})exten = i,n,Playback(option-is-invalid)exten = i,n,Goto(call-screening,s,catch-all) Recently Google updated some of its security settings so the tutorial I created around doing Asterisk Voicemail to Email is broken when a new setting is enabled. Lucky for everyone looking to use a Gmail account I have created this new post to help you.If you would like to read about this new setting here is a link: https://support.google.com/accounts/answer/6010255?hl=enThe first thing you will want to do is login to the gmail account you plan on using for voicemail to email. Once you have done that follow the directions below.Follow along below for your Asterisk voicemail to email with a Gmail account using the postfix application.1. Visit the following URL to enable less secure apps: http://www.google.com/settings/security/lesssecureappsYou should see a screen with on or off, make sure to select on as seen in the screenshot.2. We do need to install a couple of extra packages even if postfix is already installed. I also am installing postfix just in case you didn t have it installed.yum -y install postfix mailx cyrus-sasl-plain3. You need to create a new file with your Gmail account smtp server and credentials in the following format.nano /etc/postfix/sasl_passwdPaste in the below text and change the email address to your gmail email and enter in your password.smtp.gmail.com emailaddress@gmail.com:emailpassword4. Once done doing that we want to hash your password file so it is not human readable. postmap hash:/etc/postfix/sasl_passwd5. The next step is to configure postfix to use this new password file. Open the /etc/postfix/main.cf file using a text editor then scroll down to the bottom and paste in the following code.nano /etc/postfix/main.cfsmtp_sasl_auth_enable = yessmtp_sasl_password_maps = hash:/etc/postfix/sasl_passwdsmtp_sasl_security_options = noanonymous# Secure channel TLS with exact nexthop name match.smtp_tls_security_level = securesmtp_tls_mandatory_protocols = TLSv1smtp_tls_mandatory_ciphers = highsmtp_tls_secure_cert_match = nexthopsmtp_tls_CAfile = /etc/pki/tls/certs/ca-bundle.crtrelayhost = smtp.gmail.com:5876. Restart the postfix service.service postfix restart7. Now you can send a test email using the mail command in the following format.mail email@website.comSubject: Hello World. (followed by enter key to send and exit)8. Once you send that check your email to see if you received the email and as long as that is the case you can go ahead and remove the text password file.rm /etc/postfix/sasl_passwd9. Now you are all set, by default Asterisk will use the default mailing program so my voicemail.conf file looks like so.[root@localhost ~]# cat /etc/asterisk/voicemail.conf[general]format = wavserveremail = asteriskattach = yesskipms = 3000maxsilence = 10maxmessage = 300review = yessilencethreshold = 128maxlogins = 3emaildateformat = %A, %B %d, %Y at %rsendvoicemail = yesexitcontext = vm-operatoroperator = yesfromstring = PBX VoicemailHope you find this helpful and it works for you. Please let me know if there are any issues so I can correct my tutorial. I like my visitors to get their questions answered with a single visit, not having to bounce around from website to website trying to solve a problem. Have you tried to use the originate command from the Asterisk CLI only to get the following error?asterisk*CLI originateNo such command 'originate' (type 'core show help originate' for other possible commands)Well I just got this for the first time recently and it took me far too long to figure out the problem so I wanted to share with everyone.Turns out the syntax of the command has changed recently and the CLI now requires users to put channel in front of originate.Here is an example of how it works:asterisk*CLI channel originate SIP/102 extension *97@internal Learn how to install Asterisk on a Rackspace cloud server. This is an easy to follow step by step guide with all the commands you need to compile and install Asterisk with Dahdi. I use Rackspace for most of my cloud servers since their support is second to none. They are always very helpful in getting me answers or even going above and beyond what they technically support. So because of this I often build Asterisk servers up there to test with or even for production use. Here is an easy to follow guide on how to build an Asterisk 11 server using Centos 5 image. You can just copy and paste the commands one by one or add them to a text file and run as a script.Note: This step by step guide is for 64bit operating systems, if you use a 32bit Centos 5 server it will fail.yum -y updatesed -i s/SELINUX=permissive/SELINUX=disabled/g /etc/selinux/configrebootThen we need to install some packages:yum -y install bind-utils curl-devel doxygen gcc gcc-c++ gtk+ gtk+-devel gnutls-devel httpd kernel-xen kernel-xen-devel kernel kernel-devel kernel-smp-devel kernel-smp libxml2-devel libtermcap-devel libtool-ltdl libtool-ltdl-devel make mysql mysql-server mysql-devel mysql-server-devel mysql-connector-odbc net-snmp-devel neon-devel newt-devel ncurses-devel ntp openssl-devel openssl perl-suidperl php-mysql system-config-securitylevel screen texinfo unixODBC unixODBC-devel wgetrebootNow we can do the heavy lifting of installing Asterisk, Dahdi and LibPRI.cd /usr/srcwget http://downloads.asterisk.org/pub/telephony/certified-asterisk/certified-asterisk-11.6-current.tar.gzwget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gzwget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gztar zxf certified-asterisk-11.6-current.tar.gz tar zxf dahdi-linux-complete-current.tar.gz tar zxf libpri-1.4-current.tar.gzcd /usr/src/dahdi-linux-complete-*make make install make configcd /usr/src/libpri-1.4*make make installcd /usr/src/certified-asterisk-11.6-*./configure --libdir=/usr/lib64 make menuselect make make installmake samplesmake configservice dahdi startservice asterisk startAll done! It really is that easy

TAGS:Asterisk FreeSWITCH VOIP 

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