Puddle
2005-Jan-29 00:55 UTC
[Asterisk-Users] Asterisk @ Home 0.4 w/ Broadvoice + 5 SIP Phones How To
Okay, I just spent some time getting this done for a project. I documented the steps performed. Thanks to all of those who provided input. As time goes on, and people always have different settings, this may or may NOT work for you. ------------------------------------------------- Asterisk @ Home 0.4 HowTo William Pool (Puddle) 01-28-05 Description: Getting Asterisk @ Home to work with a Broad Voice SIP provider account and Five SIP Software XTen Install Asterisk @ Home OR CentOS 3.3+ with the asteris@home.tar ball and install from that. (Since, Asterisk@Home doesn't let you partition disks) Once You're up and loaded and able to get to http://xxx.yyy.zzz.xyz/maint Setup AMP the following way: Trunks 1.) Click on "Trunk ZAP/g0" change the dialout prefix to something else. NOTE: Unless you have a digum card and have a normal POTS analouge line you should NOT want this. You should leave it just incase it ever happens(or just delete it who cares). However, change the dial-out # to something odd. Chances are it'll never be used. 2.) Add a SIP Trunk: Trunk Name: BV canreinvite=no dtmfmode=inband fromdomain=sip.broadvoice.com fromuser=818xxxyyyy host=147.135.0.129 insecure=very nat=yes secret=password type=peer username=818xxxyyyy User Context: sip.broadvoice.com context=from-pstn dtmfmode=inband fromdomain=sip.broadvoice.com host=sip.broadvoice.com insecure=very nat=yes secret=password type=user user=818xxxyyyy username=818xxxyyyy Register: 818xxxyyyy:yourpassword@sip.broadvoice.com Asterisk Specific Issues 1.) SIP Issues The /etc/asterisk/sip.conf doesn't have some basic behind nat server options. Add the following after the [general] or in that section somewhere. externip=69.yzx.xxx.zzz localnet=192.168.0.0 localmask=255.255.255.0 nat=yes 2.) Dial plan issues. Calls from the sip provider into the Asterisk Server are hangup/congested and pushed away. Inside your [from-sip-external] add the line include => from-pstn AMP Specific Issues: 1.) Unable to upload file sizes larger then 500K Edit /etc/php.ini Increase the varable "upload_max_filesize" to the max size you'll "ever" expect to play. Increase this for 'whole' mp3 albums to "if" you do that... Also, edit /etc/httpd/conf.d/php.conf and change the variable "LimitRequestBody" to something more meaningful. I use "12000000" Security: Created asterisk user: adduser -s /bin/false -d /bin/false -c "Asterisk Daemon User" asterisk added asterisk user to asterisk group in /etc/group chown -R asterisk:asterisk /var/log/asterisk chown -R asterisk:asterisk /var/lib/asterisk Edit /etc/init.d/asterisk to include the two AST_USER="asterisk" AST_GROUP="asterisk" MySQL, it's default passwd for Asterisk @ Home has a LOT to be desired for. Change this. The password via the MySQL DB itself and the following files: /etc/asterisk/cdr_mysql.conf /var/www/html/maint/phpMyAdmin/config.inc.php Update Repository Catalogue yum check-update <enter> Update Repository yum update Install mod_ssl since Apache doesn't have it included. yum install mod_ssl.i386 Setup Apache 2.x ssl configs mkdir /var/swww Move the Asterisk@Home to the SSL folder mv /var/www/html /var/swww Create blank HTML Page for people mkdir /var/www/html touch /var/www/html/index.html echo "<TITLE>pbx.domain.com</TITE> >> /var/www/html/index.html echo "NOT Authorized maybe you mean <A HREF=https://pbx.domain.com>https://pbx.domain.com</A>">> /var/www/html/index.html/etc/httpd/conf/httpd.conf ServerName leave commented out, had issues with setting it. Change at your own risk blah blah /etc/httpd/conf.d/ssl.conf After <VirtualHost _default_:443> Uncomment: DocumentRoot "/var/swww/html" ServerAdmin admin@domain.com # ServerName pbx.domain.com:443 # NOTE: Look above Find the 'cgi-bin' directory in the ssl.conf file. Edit the path to reflect: /var/swww/cgi-bin Add (After the CGI-BIN Directory works) # Use htpasswd -c /etc/httpd/conf/passwd/plainpasswd user1 # this will create the user and passwd file # after that use htpasswd /etc/httpd/conf/passwd/plainpasswd user1 # I Know digest is better, but it doesn't always # work <Directory "/var/swww/html/maint"> SSLOptions +StdEnvVars AllowOverride All AuthType Basic AuthName "Receptionist / Admin access Only!" AuthUserFile /etc/httpd/conf/passwd/plainpasswd Require user user1 user2 user3 </Directory> <Directory "/var/swww/html/meetme"> SSLOptions +StdEnvVars AllowOverride All AuthType Basic AuthName "Meetme access Only!" AuthUserFile /etc/httpd/conf/passwd/plainpasswd Require user user1 user2 user3 </Directory> <Directory "/var/swww/html/admin"> SSLOptions +StdEnvVars AllowOverride All AuthType Basic AuthName "Admin access Only!" AuthUserFile /etc/httpd/conf/passwd/plainpasswd Require user user1 user2 user3 </Directory> Make sure firewall is setup somewhat: Install redhat-config-securitylevel yum install redhat-config-securitylevel This will add the following dependences [deps: gtk2 2.2.4-8.1.i386] [deps: pango 1.2.5-2.0.i386] [deps: atk 1.2.4-3.0.i386] [deps: pygtk2 1.99.16-8.i386] Add the following ports: SSH,WWW,443,5060 443 https 5060 SIP Edit /etc/sysconfig/iptables Where you have your SIP (5060) line delete the '-m tcp' Change '-p tcp' to '-p udp' Setup NTP correctly /etc/ntp.conf Install redhat-config-date to make time setting easier yum install redhat-config-date ***PHONES*** I've used XTen Lite (Soon to try Pro) The Call Groups / Extensions are self explainitory so: XTenLite NOTES: The context/username/mailbox MUST be a #. I couldn't get anything to work with it having any letters. [200] username=200 type=friend secret=1234 qualify=1000 port=5060 nat=yes mailbox=200 host=dynamic dtmfmode=inband context=from-internal canreinvite=no callerid="First Name" <200> __________________________________ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250
Mike Sander
2005-Feb-02 16:38 UTC
[Asterisk-Users] How to download CVS with attended transfers
Hi I know that attended transfers are only available in the CVS Head. I downloaded the asterisk-update.sh script from voip-info.com and ran it with these parameters ./asterisk-update.sh update dev It looked as tho CVS HEAD was downloading and compiling, although it couldn't download the addons. However, now it's up and running, only blind transfers work with "#", and I cannot change the blind transfer key to "##", it only takes the first character. And Attended transfers still isn't running. Is there something I've missed. The version info reports: Asterisk CVS-v1-0-02/03/05-10:24:22 Any help would be great. Thanks Mike -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.4 - Release Date: 1/02/2005