==================================================================== SIPTiger 2.3.1 README ==================================================================== SIPTiger v2.3.1 03/15/2002 ==================================================================== LICENSE AND COPYRIGHT ==================================================================== The Vovida Software License, Version 1.0 Copyright (c) 2000 Vovida Networks, Inc. All rights reserved. Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: 1. Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. 2. Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. 3. The names "VOCAL", "Vovida Open Communication Application Library", and "Vovida Open Communication Application Library (VOCAL)" must not be used to endorse or promote products derived from this software without prior written permission. For written permission, please contact vocal@vovida.org. 4. Products derived from this software may not be called "VOCAL", nor may "VOCAL" appear in their name, without prior written permission. THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE, TITLE AND NON-INFRINGEMENT ARE DISCLAIMED. IN NO EVENT SHALL VOVIDA NETWORKS, INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY DAMAGES IN EXCESS OF $1,000, NOR FOR ANY INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. ==================================================================== INTRODUCTION ==================================================================== SIPTiger is a provisioning utility for Cisco's 7960 Session Initiation Protocol (SIP) IP phones and Cisco's SIP Proxy Servers (CSPS). 7960 SIP IP phones and Cisco SIP proxy servers are both reliant upon a set of configuration files, which SIPTiger can parse and format into a user-friendly web-based Graphical User Interface (GUI). After these files are modified, the affected SIP phones can then be remotely reloaded to allow the changes to take effect. SIP- Tiger also supports administrative-level call forwarding config- uration. ==================================================================== NEW FEATURES AND FUNCTIONS IN THIS RELEASE ==================================================================== o Support for Cisco Sip Proxy Server Release 1.2.0.3 - AuthConsumeProxyAuthHdr, default changed from On to Off - AuthAllow3rdPartyRegistration, new directive - VendorSpecificAttributeID, removed - Cisco_Routing_Wildcard_Expand_Length, new directive o Support for Cisco 7940/7960 Release 3.1 - Authentication Name, expanded to accept up to 50 characters - semi_attended_transfer, new directive - telnet_enable, new directive ==================================================================== BUG FIXES ==================================================================== o None ==================================================================== KNOWN LIMITATIONS ==================================================================== These limitations are known to exists in this release of the software: Limitation o SIPTiger responds sluggishly if more than 50 phones are added at once. This is found in the utility which allows adding multiple phones simultaneously via Media Access Control (MAC) file. Workaround o This is a hardware constraint. Therefore, workarounds can not be provided due to the variations in hardware config- urations. Limitation o SIPTiger responds sluggishly if more than 80 phones are remotely reloaded at once. This is found in the utility which allows reloading all phones on a proxy. Workaround o This is a hardware constraint. Therefore, workarounds can not be provided due to the variations in hardware config- urations. ==================================================================== GETTING STARTED ==================================================================== SOFTWARE DEPENDENCIES -------------------------------------------------------------------- PHP 4 Apache Web Server MySQL client (optional) Web browser PLATFORMS SUPPORTED -------------------------------------------------------------------- RedHat 7.0, 7.1 Solaris 2.6 RedHat 7.X INSTALLATION INSTRUCTIONS -------------------------------------------------------------------- 1. Untar siptiger file. tar xvfz siptiger-x.x.x.tar.gz 2. Move SIP directory on web server. mv SIP/ /var/www/html/ 3. Change ownership of siptiger files to web server user. chown -R apache.apache /var/www/html/SIP 4. Edit your httpd.conf and SIP Alias. Add this directive to httpd.conf. NOTE: Insert this alias in the section of httpd.conf marked "Aliases", before the "ScriptAlias" section. httpd.conf located "/etc/httpd/conf/httpd.conf" #Alias for SIP Phone Provisioning Tool Alias /SIP/ /var/www/html/SIP/ AuthName "SIP Admin tools" AuthType Basic AuthUserFile /var/www/html/SIP/.htpasswd AllowOverride FileInfo AuthConfig Limit Options MultiViews Indexes SymLinksIfOwnerMatch IncludesNoExec require valid-user require valid-user This creates a user/password protected alias on your web server. Reload web server /etc/rc.d/init.d/httpd restart 5. Create a user file and user for SIP Alias. htpasswd -c /var/www/html/SIP/.htpasswd sipadmin 6. Change permissions on tftp directory. chmod -R a+wrx /tftpboot 7. Locating Alias on your website. In browser use "http://YOURWEBSITE.com/SIP" 8. Change permissions on SIP directory chmod -R o+wrx /var/www/html/SIP Solaris 2.6 INSTALLATION INSTRUCTIONS -------------------------------------------------------------------- 1. Unzip siptiger file. gunzip siptiger-x.x.x.tar.gz 2. Untar siptiger file. tar xvf siptiger-x.x.x.tar 3. Move SIP directory on web server. mv SIP /usr/local/apache/htdocs 4. Change ownership of siptiger files to web server user. chown -R apache:apache /usr/local/apache/htdocs/SIP 5. Edit your httpd.conf and SIP Alias. Add this directive to httpd.conf. NOTE: Insert this alias in the section of httpd.conf marked "Aliases", before the "ScriptAlias" section. httpd.conf located "/usr/local/apache/conf/httpd.conf" #Alias for SIP Phone Provisioning Tool Alias /SIP/ /usr/local/apache/htdocs/SIP/ AuthName "SIP Admin tools" AuthType Basic AuthUserFile /usr/local/apache/htdocs/SIP/.htpasswd AllowOverride FileInfo AuthConfig Limit Options MultiViews Indexes SymLinksIfOwnerMatch IncludesNoExec require valid-user require valid-user This create a user/password protected alias on your web server. Reload web server /usr/local/apache/bin/apachectl restart 6. Create a user file and user for SIP Alias. /usr/local/apache/bin/htpasswd -c /usr/local/apache/htdocs/SIP/.htpasswd sipadmin (Enter password at prompt) 7. Change permissions on tftp directory. chmod -R a+wrx /tftpboot 8. Change permissions on SIP directory chmod -R o+wrx /usr/local/apache/htdocs/SIP USING THE SOFTWARE -------------------------------------------------------------------- SIPTiger is broken down into three main sections: proxy config- uration, phone configuration, and MySQL configuration. Each of these sections parse different files and allow a user to modify these files through a GUI. Proxy Configuration ------------------- o Add/Edit Proxy Servers This section of the utililty enables users to add, delete, and edit proxy server profiles. These profiles consist of user names, passwords, remote subdirectories, and proxy names. Storing this information allows the tool to download a specific proxy's configuration file, edit it, and upload it back to its proxy. o Main Server The main proxy server section houses some of the apache directives which pertain to forking, host specific infor- mation, logging, and others such as the pidfile, server root, lockfile, etc. o SIP Server This section allows a user to modify information which pertains to the SIP server core configuration. Some of the parameters available for modification here are proxy domain, stateful server, SIP token port, max forks, etc. o MySQL The MySQL section provides a means of modifying param- eters that configure the MySQL module of sipd.conf. Some of the parameters which will be available for modification are DB_MySQL_HostName, DB_MySQL_Password_Field, DB_My_SQL_PhoneNumber_Field, and DB_MySQL_Fork_Field. o GKTMP Configuration This part of the tool enables users to modify parameters specific to the GKTMP module. Some of the capabilities are toggling GktmpConnection, selecting master and secondary servers, and toggling debugging options. o Accounting Parameters pertaining to accouting are available for modification here. Some of the parameters found in this section allow the user to enable or disable accouting logic, modify the accouting radius ports and password, and other accounting releated values. o Authentication Parameters pertaining to authentication are available for modification here. Some of the parameters found in this section allow the user to enable or disable authen- tication logic, modify the authentication realm, as well as edit the authentication ports and passwords. o Call Forwarding The call forwarding section allows users to modify param- eters which control how the call forwarding is handled. In this section, users can enable or disable unconditional call forwarding, unavailable call forwarding, timers, busy call forwarding, no answer forwarding, and diversion headers. o ENUM Configuration The ENUM configuration routines allow a user to turn the ENUM lookup module on and off, set the ENUM global domain, and toggle the debug flag. o Routing Modification of the parameters concerned with routing (excluding static routing parameters) are possible from this section. With this tool, the user is able to toggle Cisco routing, modify routing rendezvous name, edit remote update port, and others. o Static Routing This section makes static routing parameters available for modification. A few of the parameters are the destination pattern, next hop port, transport protocol, etc. o Registry Parameters that deal with the registry (excluding static registry parameters) are available for modification in this section. A few of the parameters are Cisco registry, registry farm members, registry remote update port, and registry shared memory address. o Static Registry In this section, a user is able to modify parameters which pertain to the static registry, such as contact port, transport protocol, contact age, etc. o RAS Module Cisco SIP Proxy Servers are able to send ASN.1 encoded RAS LRQ messages to provisioned H.323 directory Gatekeepers, and can receive LCF, LRJ, or RIP messages back. This section allows users to modify the parameters that support this. Phone Configuration ------------------- o Configure Singular Phone 7960 SIP IP phones depend on the SIPConfigGeneric.cnf file for settings specific to one phone. In order to ensure that each SIPConfigGeneric.cnf file is associated with its assigned phone, the convention is to incorporate the unique MAC address of that phone into the filename. This file, when associated with one phone, is named SIPxxxxyyyyzzzz.cnf, where xxxxyyyyzzzz is the MAC address for that phone. For example, if the MAC address of a phone is 12345678A1BC, then the SIPConfigGeneric.cnf file- name is changed to SIP12345678A1BC.cnf. This file holds the line names or numbers, display names for each line, authenti- cation names, passwords, etc. The singular phone configuration utility will modify this file for these parameters. In this section of the phone configuration utility, one can select a phone from a list of phones stored in the TFTP configuration file directory. Once a phone has been selected, these parameters can be modified. o Creating Phone Configuration Files In this section, a user is able to enter the information described in the previous section for new phones on a one-at-a- time basis. o Reloading One Phone This section of the phone configuration utility allows a user to remotely reload one phone by entering the proxy server address, proxy port, and user id of the phone to be reloaded. Note that to reload a phone, the sync file must be correctly configured. If the user wishes to reload a phone using A record lookup, the first reload tool, labeled "Reload Phone via A Record Lookup should be used. To reload a phone using DNS SRV lookup, a "srvfarm.xml" file must be created (see "Configuring SRV Farm File"). After this file is created, a user can reload a phone using DNS SRV lookup using the second reload tool, labeled "Reload Phone via DNS SRV Lookup." o Editing the Default Configuration of Multiple Phones For parameters that apply to multiple SIP IP phones, the optional SIPDefault.cnf file is utilized. Parameters to be included in this file are the image version parameter and call environment parameters, such as proxy server address, time zone, etc. When provisioning multiple phones, the multiple phone configuration utility will modify this file. o Editing Dial Plans SIP IP phones support the option to implement a dial plan that reserves specified keystrokes for specified actions. For example, a modern convention to gain access to the local PSTN is to dial "9", and the rest of the number. This is an example of a keystroke which would be included in a dial plan. This section of the phone configura- tion utility allows users to enter keystroke patterns to reserve, the desired reactions when these patterns are matched, the type of phone used, the amount of time (in seconds) to wait before a timeout, and any comments that might help a dialplan programmer. o Adding Dial Plans This section of the utility allows users to specify keystroke pat- terns to reserve. o Editing Sync Files 7960 SIP IP phones depend on a sync file to alert a phone that the phone image has changed. When a phone reloads, if the sync level listed in the downloaded sync file differs from the level in the phone's flash memory, the phone stores this new number, reboots, and downloads the newer image. SIPTiger allows modification of the syncinfo.xml file through a GUI. The modifiable parameters are image version, and sync level. o Adding Phones via MAC File The process of adding phones on a one-at-a-time basis as described in section 2.4.2 can be a time consuming task if many phones are to be added. For this reason, users are also able to add multiple phones by means of a MAC file. This section of the utility loads a file, specified by the user, which only contains a list of MAC addresses which are separated by carriage returns. A menu will appear, which will list each MAC address in the file and provide input blanks for phone-specific information. Once the form is filled out and submitted, the phone configuration files will all be created. o Configuring SRV Farm File If you are using DNS SRV, it will be necessary to create a file "srvfarm.xml" in the /tftpboot directory with all farm members listed in priority order. This tool allows a user to create/modify this file. MySQL Configuration ------------------- o Configure User Information This section of the SIPTiger utility allows a user to select a MySQL server from a list, and modify information which per- tains to the users on that MySQL server. The information which can be modified consists of call forwarding options. o Add User to MySQL Server From this section, a user is able to select the MySQL server on which a user is to be added. Information such as user id, password, and call forwarding options can then be entered and submitted. o Configure MySQL Server In this section, it is possible to configure information for all servers currently entered. This information consists of user, password, database, and table. It also enables adding and deleting MySQL servers to and from the list. ==================================================================== DIRECTORY STRUCTURE ==================================================================== This directory contains the following subdirectories or files: -/SIP: Main directory. Contains README, function library, HTML frames, mysql (dir), phone (dir), server (dir) -/SIP/mysql: Files which pertain to configuring mysql data. -/SIP/phone: Files dealing with provisioning 7960 SIP phones. -/SIP/phone/server: Files pertaining to configuring proxy servers, conf(dir) -/SIP/phone/conf: sipd.conf files from different proxies, backup sipd.conf files ==================================================================== CONTACT INFORMATION AND WEBSITE ==================================================================== We welcome your feedback, suggestions and contributions. Contact us via email if you have questions, feedback, code submissions, and bug reports. For general inquiries - info@vovida.org We have mailing lists for the VOCAL applications and proctocol stacks: VOCAL - vocal@vovida.org COPS - cops@vovida.org MGCP - mgcp@vovida.org RADIUS - radius@vovida.org RTP - rtp@vovida.org SIP - sip@vovida.org TRIP - trip@vovida.org SIPTiger - siptiger@vovida.org You can subscribe to the mailing lists on www.vovida.org. You can submit bug, patches, software contributions, and feature requests using Bugzilla. Access Bugzilla from www.vovida.org. ====================================================================