805 lines
34 KiB
Plaintext
805 lines
34 KiB
Plaintext
OSP User Guide for Asterisk V1.6
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4 January 2007
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Table of Contents
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1 Introduction
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2 OSP Toolkit
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2.1 Build OSP Toolkit
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2.1.1 Unpacking the Toolkit
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2.1.2 Preparing to build the OSP Toolkit
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2.1.3 Building the OSP Toolkit
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2.1.4 Installing the OSP Toolkit
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2.1.5 Building the Enrollment Utility
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2.2 Obtain Crypto Files
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3 Asterisk
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3.1 Configure for OSP Support
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3.1.1 Build Asterisk with OSP Toolkit
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3.1.2 osp.conf
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3.1.3 extensions.conf
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3.1.4 zapata/sip/iax/h323/ooh323.conf
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3.2 OSP Dial Plan Functions
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3.2.1 OSPAuth
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3.2.2 OSPLookup
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3.2.3 OSPNext
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3.2.4 OSPFinish
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3.3 extensions.conf Examples
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3.3.1 Source Gateway
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3.3.2 Destination Gateway
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3.3.3 Proxy
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Asterisk is a trademark of Digium, Inc.
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TransNexus and OSP Secures are trademarks of TransNexus, Inc.
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1 Introduction
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This document provides instructions on how to build and configure Asterisk
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V1.6 with the OSP Toolkit to enable secure, multi-lateral peering. This
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document is also available in the Asterisk source package as doc/osp.txt.
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The OSP Toolkit is an open source implementation of the OSP peering protocol
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and is freely available from www.sipfoundry.org. The OSP standard defined by
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the European Telecommunications Standards Institute (ETSI TS 101 321)
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www.esti.org. If you have questions or need help, building Asterisk with the
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OSP Toolkit, please post your question on the OSP mailing list at
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https://list.sipfoundry.org/mailman/listinfo/osp.
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2 OSP Toolkit
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Please reference the OSP Toolkit document "How to Build and Test the OSP
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Toolkit" available from https://www.sipfoundry.org/OSPclient.
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2.1 Build OSP Toolkit
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The software listed below is required to build and use the OSP Toolkit:
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* OpenSSL (required for building) - Open Source SSL protocol and Cryptographic
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Algorithms (version 0.9.7g recommended) from www.openssl.org. Pre-compiled
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OpenSSL binary packages are not recommended because of the binary
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compatibility issue.
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* Perl (required for building) - A programming language used by OpenSSL for
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compilation. Any version of Perl should work. One version of Perl is
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available from www.activestate.com/Products/ActivePer. If pre-compiled
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OpenSSL packages are used, Perl package is not required.
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* C compiler (required for building) - Any C compiler should work. The GNU
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Compiler Collection from www.gnu.org is routinely used for building the OSP
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Toolkit for testing.
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* OSP Server (required for testing) - Access to any OSP server should work.
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Open source OSP servers are available from https://www.sipfoundry.org/OSP,
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or go to http://www.transnexus.com/OSP%20Toolkit/Peering_Server/VoIP_Peering_Server.htm
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to download a free commercial OSP server.
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2.1.1 Unpacking the Toolkit
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After downloading the OSP Toolkit (version 3.3.6 or later release) from
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www.sipfoundry.org, perform the following steps in order:
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1) Copy the OSP Toolkit distribution into the directory where it will reside.
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The default directory for the OSP Toolkit is /usr/src.
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2) Un-package the distribution file by executing the following command:
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gunzip -c OSPToolkit-###.tar.gz | tar xvf -
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Where ### is the version number separated by underlines. For example, if
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the version is 3.3.6, then the above command would be:
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gunzip -c OSPToolkit-3_3_6.tar.gz | tar xvf -
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A new directory (TK-3_3_6-20060303) will be created within the same
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directory as the tar file.
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3) Go to the TK-3_3_6-20060303 directory by running this command:
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cd TK-3_3_6-20060303
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Within this directory, you will find directories and files similar to what
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is listed below if the command "ls -F" is executed):
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ls -F
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enroll/
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RelNotes.txt lib/
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README.txt license.txt
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bin/ src/
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crypto/ test/
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include/
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2.1.2 Preparing to build the OSP Toolkit
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4) Compile OpenSSL according to the instructions provided with the OpenSSL
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distribution (You would need to do this only if you don't have openssl
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already).
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5) Copy the OpenSSL header files (the *.h files) into the crypto/openssl
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directory within the osptoolkit directory. The OpenSSL header files are
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located under the openssl/include/openssl directory.
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6) Copy the OpenSSL library files (libcrypto.a and libssl.a) into the lib
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directory within the osptoolkit directory. The OpenSSL library files are
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located under the openssl directory.
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Note: Since the Asterisk requires the OpenSSL package. If the OpenSSL
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package has been installed, steps 4 through 6 are not necessary.
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7) Optionally, change the install directory of the OSP Toolkit. Open the
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Makefile in the /usr/src/TK-3_3_6-20060303/src directory, look for the
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install path variable - INSTALL_PATH, and edit it to be anywhere you want
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(defaults /usr/local).
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Note: Please change the install path variable only if you are familiar
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with both the OSP Toolkit and the Asterisk.
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2.1.3 Building the OSP Toolkit
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8) From within the OSP Toolkit directory (/usr/src/TK-3_3_6-20060303), start
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the compilation script by executing the following commands:
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cd src
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make clean; make build
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2.1.4 Installing the OSP Toolkit
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The header files and the library of the OSP Toolkit should be installed.
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Otherwise, you must specify the OSP Toolkit path for the Asterisk.
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9) Use the make script to install the Toolkit.
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make install
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The make script is also used to install the OSP Toolkit header files and
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the library into the INSTALL_PATH specified in the Makefile.
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Note: Please make sure you have the rights to access the INSTALL_PATH
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directory. For example, in order to access /usr/local directory,
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root privileges are required.
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2.1.5 Building the Enrollment Utility
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Device enrollment is the process of establishing a trusted cryptographic
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relationship between the VoIP device and the OSP Server. The Enroll program is
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a utility application for establishing a trusted relationship between an OSP
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client and an OSP server. Please see the document "Device Enrollment" at
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https://www.sipfoundry.org/OSPclient for more information about the enroll
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application.
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10) From within the OSP Toolkit directory (example:
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/usr/src/TK-3_3_6-20060303), execute the following commands at the command
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prompt:
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cd enroll
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make clean; make linux
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Compilation is successful if there are no errors in the compiler output.
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The enroll program is now located in the OSP Toolkit/bin directory
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(example: /usr/src/ TK-3_3_6-20060303/bin).
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2.2 Obtain Crypto Files
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The OSP module in Asterisk requires three crypto files containing a local
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certificate (localcert.pem), private key (pkey.pem), and CA certificate
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(cacert_0.pem). Asterisk will try to load the files from the Asterisk
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public/private key directory - /var/lib/asterisk/keys. If the files are not
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present, the OSP module will not start and the Asterisk will not support the
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OSP protocol. Use the enroll.sh script from the toolkit distribution to
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enroll Asterisk with an OSP server and obtain the crypto files. Documentation
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explaining how to use the enroll.sh script (Device Enrollment) to enroll with
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an OSP server is available at https://www.sipfoundry.org/OSPclient. Copy the
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files generated by the enrollment process to the Asterisk
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/var/lib/asterisk/keys directory.
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Note: The osptestserver.transnexus.com is configured only for sending and
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receiving non-SSL messages, and issuing signed tokens. If you need help,
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post a message on the OSP mailing list of www.sipfoundry.org or send an
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e-mail to support@transnexus.com.
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The enroll.sh script takes the domain name or IP addresses of the OSP servers
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that the OSP Toolkit needs to enroll with as arguments, and then generates pem
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files - cacert_#.pem, certreq.pem, localcert.pem, and pkey.pem. The "#" in the
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cacert file name is used to differentiate the ca certificate file names for
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the various SP's (OSP servers). If only one address is provided at the command
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line, cacert_0.pem will be generated. If 2 addresses are provided at the
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command line, 2 files will be generated - cacert_0.pem and cacert_1.pem, one
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for each SP (OSP server). The example below shows the usage when the client
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is registering with osptestserver.transnexus.com.
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./enroll.sh osptestserver.transnexus.com
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Generating a 512 bit RSA private key
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........................++++++++++++
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.........++++++++++++
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writing new private key to 'pkey.pem'
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-----
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You are about to be asked to enter information that will be incorporated
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into your certificate request.
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What you are about to enter is what is called a Distinguished Name or a DN.
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There are quite a few fields but you can leave some blank
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For some fields there will be a default value,
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If you enter '.', the field will be left blank.
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-----
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Country Name (2 letter code) [AU]: _______
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State or Province Name (full name) [Some-State]: _______
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Locality Name (eg, city) []:_______
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Organization Name (eg, company) [Internet Widgits Pty Ltd]: _______
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Organizational Unit Name (eg, section) []:_______
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Common Name (eg, YOUR name) []:_______
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Email Address []:_______
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Please enter the following 'extra' attributes
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to be sent with your certificate request
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A challenge password []:_______
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An optional company name []:_______
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Error Code returned from openssl command : 0
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CA certificate received
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[SP: osptestserver.transnexus.com]Error Code returned from getcacert command : 0
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output buffer after operation: operation=request
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output buffer after nonce: operation=request&nonce=1655976791184458
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X509 CertInfo context is null pointer
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Unable to get Local Certificate
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depth=0 /CN=osptestserver.transnexus.com/O=OSPServer
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verify error:num=18:self signed certificate
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verify return:1
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depth=0 /CN=osptestserver.transnexus.com/O=OSPServer
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verify return:1
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The certificate request was successful.
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Error Code returned from localcert command : 0
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The files generated should be copied to the /var/lib/asterisk/keys directory.
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Note: The script enroll.sh requires AT&T korn shell (ksh) or any of its
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compatible variants. The /usr/src/TK-3_3_6-20060303/bin directory should
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be in the PATH variable. Otherwise, enroll.sh cannot find the enroll
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file.
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3 Asterisk
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In Asterisk, all OSP support is implemented as dial plan functions. In
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Asterisk V1.6, all combinations of routing between OSP and non-OSP enabled
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networks using any combination of SIP, H.323 and IAX protocols are fully
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supported. Section 3.1 describes the three easy steps to add OSP support to
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Asterisk:
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1. Build Asterisk with OSP Toolkit
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2. Configure osp.conf file
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3. Cut and paste to extensions.conf
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Sections 3.2 and 3.3 provide a detailed explanation of OSP dial plan functions
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and configuration examples. The detailed information provided in Sections 3.2
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and 3.3 is not required for operating Asterisk with OSP, but may be helpful to
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developers who want to customize their Asterisk OSP implementation.
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3.1 Configure for OSP Support
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3.1.1 Build Asterisk with OSP Toolkit
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The first step is to build Asterisk with the OSP Toolkit. If the OSP Toolkit
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is installed in the default install directory, /usr/local, no additional
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configuration is required. Compile Asterisk according to the instructions
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provided with the Asterisk distribution.
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If the OSP Toolkit is installed in another directory, such as /myosp, Asterisk
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must be configured with the location of the OSP Toolkit. See the example
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below.
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--with-osptk=/myosp
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Note: Please change the install path only if you familiar with both the OSP
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Toolkit and the Asterisk. Otherwise, the change may result in Asterisk
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not supporting the OSP protocol.
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3.1.2 osp.conf
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The /etc/asterisk/osp.conf file, shown below, contains configuration
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parameters for using OSP. Two parameters, servicepoint and source must be
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configured. The default values for all other parameters will work well for
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standard OSP implementations.
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;
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; Open Settlement Protocol Sample Configuration File
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;
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; This file contains configuration of OSP server providers that
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; are used by the Asterisk OSP module. The section "general" is
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; reserved for global options. All other sections describe specific
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; OSP Providers. The provider "default" is used when no provider is
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; otherwise specified.
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:
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: The "servicepoint" and "source" parameters must be configured. For
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; most implementations the other parameters in this file can be left
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; unchanged.
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;
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[general]
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;
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; Enable cryptographic acceleration hardware.
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;
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accelerate=no
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;
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; Defines the status of tokens that Asterisk will validate.
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; 0 - signed tokens only
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; 1 - unsigned tokens only
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; 2 - both signed and unsigned
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; The default value is 0, i.e. the Asterisk will only validate signed
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; tokens.
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;
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tokenformat=0
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;
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[default]
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;
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; List all service points (OSP servers) for this provider. Use
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; either domain name or IP address. Most OSP servers use port 1080.
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;
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;servicepoint=http://osptestserver.transnexus.com:1080/osp
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servicepoint=http://OSP server IP:1080/osp
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;
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; Define the "source" device for requesting OSP authorization.
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: This value is usually the domain name or IP address of the
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: the Asterisk server.
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;
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;source=domain name or [IP address in brackets]
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source=[host IP]
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;
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; Define path and file name of crypto files.
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; The default path for crypto file is /var/lib/asterisk/keys. If no
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; path is defined, crypto files should be in
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; /var/lib/asterisk/keys directory.
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;
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; Specify the private key file name.
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; If this parameter is unspecified or not present, the default name
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; will be the osp.conf section name followed by "-privatekey.pem"
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; (for example: default-privatekey.pem)
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;
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privatekey=pkey.pem
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;
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; Specify the local certificate file.
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; If this parameter is unspecified or not present, the default name
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; will be the osp.conf section name followed by "- localcert.pem "
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; (for example: default-localcert.pem)
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;
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localcert=localcert.pem
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;
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; Specify one or more Certificate Authority key file names. If none
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; are listed, a single Certificate Authority key file name is added
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; with the default name of the osp.conf section name followed by
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; "-cacert_0.pem " (for example: default-cacert_0.pem)
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;
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cacert=cacert_0.pem
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;
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; Configure parameters for OSP communication between Asterisk OSP
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; client and OSP servers.
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;
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; maxconnections: Max number of simultaneous connections to the
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; provider OSP server (default=20)
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; retrydelay: Extra delay between retries (default=0)
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; retrylimit: Max number of retries before giving up (default=2)
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; timeout: Timeout for response in milliseconds (default=500)
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;
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maxconnections=20
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retrydelay=0
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retrylimit=2
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timeout=500
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;
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; Set the authentication policy.
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; 0 - NO - Accept all calls.
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; 1 - YES - Accept calls with valid token or no token.
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; Block calls with invalid token.
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; 2 - EXCLUSIVE - Accept calls with valid token.
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; Block calls with invalid token or no token.
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; Default is 1,
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;
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authpolicy=1
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;
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; Set the default destination protocol. The OSP module supports
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; SIP, H323, and IAX protocols. The default protocol is set to SIP.
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;
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defaultprotocol=SIP
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3.1.3 extensions.conf
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OSP functions are implemented as dial plan functions in the extensions.conf
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file. To add OSP support to your Asterisk server, simply copy and paste the
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text box below to your extensions.conf file. These functions will enable your
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Asterisk server to support all OSP call scenarios. Configuration of your
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Asterisk server for OSP is now complete.
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[globals]
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DIALOUT=Zap/1
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[SrcGW] ; OSP Source Gateway
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exten => _XXXX.,1,NoOp(OSPSrcGW)
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; Set calling number if necessary
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exten => _XXXX.,n,Set(CALLERID(numner)=1234567890)
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; OSP lookup using default provider, if fail/error jump to lookup+101
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exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||j)
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; Deal with outbound call according to protocol
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exten => _XXXX.,n,Macro(outbound)
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; Dial to destination, 60 timeout, with call duration limit
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exten => _XXXX.,n,Dial(${OSPDIALSTR},60,oL($[${OSPOUTTIMELIMIT}*1000]))
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; Wait 1 second
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exten => _XXXX.,n,Wait,1
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; Hangup
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exten => _XXXX.,n,Hangup
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; Deal with OSPLookup fail/error
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exten => _XXXX.,lookup+101,Hangup
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exten => h,1,NoOp()
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; OSP report usage
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exten => h,n,OSPFinish(${HANGUPCAUSE})
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[DstGW] ; OSP Destination Gateway
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exten => _XXXX.,1,NoOp(OSPDstGW)
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; Deal with inbound call according to protocol
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exten => _XXXX.,n,Macro(inbound)
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; Validate token using default provider, if fail/error jump to auth+101
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exten => _XXXX.,n(auth),OSPAuth(|j)
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; Ringing
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exten => _XXXX.,n,Ringing
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; Wait 1 second
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exten => _XXXX.,n,Wait,1
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; Check inbound call duration limit
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exten => _XXXX.,n,GoToIf($[${OSPINTIMELIMIT}=0]?100:200)
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; Without duration limit
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exten => _XXXX.,100,Dial(${DIALOUT},15,o)
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exten => _XXXX.,n,Goto(1000)
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; With duration limit
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exten => _XXXX.,200,Dial(${DIALOUT},15,oL($[${OSPINTIMELIMIT}*1000]))
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exten => _XXXX.,n,Goto(1000)
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; Wait 1 second
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exten => _XXXX.,1000,Wait,1
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; Hangup
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exten => _XXXX.,n,Hangup
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; Deal with OSPAuth fail/error
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exten => _XXXX.,auth+101,Hangup
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exten => h,1,NoOp()
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; OSP report usage
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exten => h,n,OSPFinish(${HANGUPCAUSE})
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[GeneralProxy] ; Proxy
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exten => _XXXX.,1,NoOp(OSP-GeneralProxy)
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; Deal with inbound call according to protocol
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exten => _XXXX.,n,Macro(inbound)
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; Validate token using default provider, if fail/error jump to auth+101
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exten => _XXXX.,n(auth),OSPAuth(|j)
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; OSP lookup using default provider, if fail/error jump to lookup+101
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exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||j)
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; Deal with outbound call according to protocol
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exten => _XXXX.,n,Macro(outbound)
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; Dial to destination, 14 timeout, with call duration limit
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exten => _XXXX.,n,Dial(${OSPDIALSTR},14,oL($[${OSPOUTTIMELIMIT}*1000]))
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; OSP lookup next destination using default provider, if fail/error jump to next1+101
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exten => _XXXX.,n(next1),OSPNext(${HANGUPCAUSE}||j)
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; Deal with outbound call according to protocol
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exten => _XXXX.,n,Macro(outbound)
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; Dial to destination, 15 timeout, with call duration limit
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exten => _XXXX.,n,Dial(${OSPDIALSTR},15,oL($[${OSPOUTTIMELIMIT}*1000]))
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; OSP lookup next destination using default provider, if fail/error jump to next2+101
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exten => _XXXX.,n(next2),OSPNext(${HANGUPCAUSE}||j)
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; Deal with outbound call according to protocol
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exten => _XXXX.,n,Macro(outbound)
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; Dial to destination, 16 timeout, with call duration limit
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exten => _XXXX.,n,Dial(${OSPDIALSTR},16,oL($[${OSPOUTTIMELIMIT}*1000]))
|
|
; Hangup
|
|
exten => _XXXX.,n,Hangup
|
|
; Deal with OSPAuth fail/error
|
|
exten => _XXXX.,auth+101,Hangup
|
|
; Deal with OSPLookup fail/error
|
|
exten => _XXXX.,lookup+101,Hangup
|
|
; Deal with OSPNext fail/error
|
|
exten => _XXXX.,next1+101,Hangup
|
|
; Deal with OSPNext fail/error
|
|
exten => _XXXX.,next2+101,Hangup
|
|
exten => h,1,NoOp()
|
|
; OSP report usage
|
|
exten => h,n,OSPFinish(${HANGUPCAUSE})
|
|
|
|
[macro-inbound]
|
|
exten => s,1,NoOp(inbound)
|
|
; Get inbound protocol
|
|
exten => s,n,Set(CHANTECH=${CUT(CHANNEL,/,1)})
|
|
exten => s,n,GoToIf($["${CHANTECH}"="H323"]?100)
|
|
exten => s,n,GoToIf($["${CHANTECH}"="IAX2"]?200)
|
|
exten => s,n,GoToIf($["${CHANTECH}"="SIP"]?300)
|
|
exten => s,n,GoTo(1000)
|
|
; H323 --------------------------------------------------------
|
|
; Get peer IP
|
|
exten => s,100,Set(OSPPEERIP=${H323CHANINFO(peerip)})
|
|
; Get OSP token
|
|
exten => s,n,Set(OSPINTOKEN=${H323CHANINFO(osptoken)})
|
|
exten => s,n,GoTo(1000)
|
|
; IAX ----------------------------------------------------------
|
|
; Get peer IP
|
|
exten => s,200,Set(OSPPEERIP=${IAXPEER(CURRENTCHANNEL)})
|
|
; Get OSP token
|
|
exten => s,n,Set(OSPINTOKEN=${IAXCHANINFO(osptoken)})
|
|
exten => s,n,GoTo(1000)
|
|
; SIP ----------------------------------------------------------
|
|
; Get peer IP
|
|
exten => s,300,Set(OSPPEERIP=${SIPCHANINFO(peerip)})
|
|
; Get OSP token
|
|
exten => s,n,Set(OSPINTOKEN=${SIP_HEADER(P-OSP-Auth-Token)})
|
|
exten => s,n,GoTo(1000)
|
|
; --------------------------------------------------------------
|
|
exten => s,1000,MacroExit
|
|
|
|
[macro-outbound]
|
|
exten => s,1,NoOp(outbound)
|
|
; Set calling number which may be translated
|
|
exten => s,n,Set(CALLERID(number)=${OSPCALLING})
|
|
; Check destinatio protocol
|
|
exten => s,n,GoToIf($["${OSPTECH}"="H323"]?100)
|
|
exten => s,n,GoToIf($["${OSPTECH}"="IAX2"]?200)
|
|
exten => s,n,GoToIf($["${OSPTECH}"="SIP"]?300)
|
|
; Something wrong
|
|
exten => s,n,Hangup
|
|
exten => s,n,GoTo(1000)
|
|
; H323 --------------------------------------------------------
|
|
; Set call id
|
|
exten => s,100,Set(H323CHANINFO(callid)=${OSPOUTCALLID})
|
|
; Set OSP token
|
|
exten => s,n,Set(H323CHANINFO(osptoken)=${OSPOUTTOKEN})
|
|
exten => s,n,GoTo(1000)
|
|
; IAX ----------------------------------------------------------
|
|
; Set OSP token
|
|
exten => s,200,Set(IAXCHANINFO(osptoken)=${OSPOUTTOKEN})
|
|
exten => s,n,GoTo(1000)
|
|
; SIP ----------------------------------------------------------
|
|
exten => s,300,GoTo(1000)
|
|
; --------------------------------------------------------------
|
|
exten => s,1000,MacroExit
|
|
|
|
3.1.4 zapata/sip/iax/h323/ooh323.conf
|
|
There is no configuration required for OSP.
|
|
|
|
3.2 OSP Dial Plan Functions
|
|
This section provides a description of each OSP dial plan function.
|
|
|
|
3.2.1 OSPAuth
|
|
OSP token validation function.
|
|
Input:
|
|
* OSPPEERIP: last hop IP address
|
|
* OSPINTOKEN: inbound OSP token
|
|
* provider: OSP service provider configured in osp.conf. If it is empty, default provider is used.
|
|
* priority jump
|
|
Output:
|
|
* OSPINHANDLE: inbound OSP transaction handle
|
|
* OSPINTIMELIMIT: inbound call duration limit
|
|
* OSPAUTHSTATUS: OSPAuth return value. SUCCESS/FAILED/ERROR
|
|
|
|
3.2.2 OSPLookup
|
|
OSP lookup function.
|
|
Input:
|
|
* OSPPEERIP: last hop IP address
|
|
* OSPINHANDLE: inbound OSP transaction handle
|
|
* OSPINTIMELIMIT: inbound call duration limit
|
|
* exten: called number
|
|
* provider: OSP service provider configured in osp.conf. If it is empty, default provider is used.
|
|
* priority jump
|
|
* callidtypes: Generate call ID for the outbound call. h: H.323; s: SIP; i: IAX. Only h, H.323, has been implemented.
|
|
Output:
|
|
* OSPOUTHANDLE: outbound transaction handle
|
|
* OSPTECH: outbound protocol
|
|
* OSPDEST: outbound destination IP address
|
|
* OSPCALLED: outbound called nummber
|
|
* OSPCALLING: outbound calling number
|
|
* OSPOUTTOKEN: outbound OSP token
|
|
* OSPRESULTS: number of remaining destinations
|
|
* OSPOUTTIMELIMIT: outbound call duration limit
|
|
* OSPOUTCALLIDTYPES: same as input callidtypes
|
|
* OSPOUTCALLID: outbound call ID. Only for H.323
|
|
* OSPDIALSTR: outbound dial string
|
|
* OSPLOOKUPSTATUS: OSPLookup return value. SUCCESS/FAILED/ERROR
|
|
|
|
3.2.3 OSPNext
|
|
OSP lookup next function.
|
|
Input:
|
|
* OSPINHANDLE: inbound transaction handle
|
|
* OSPOUTHANDLE: outbound transaction handle
|
|
* OSPINTIMELIMIT: inbound call duration limit
|
|
* OSPOUTCALLIDTYPES: types of call ID generated by Asterisk.
|
|
* OSPRESULTS: number of remain destinations
|
|
* cause: last destination disconnect cause
|
|
* priority jump
|
|
Output:
|
|
* OSPTECH: outbound protocol
|
|
* OSPDEST: outbound destination IP address
|
|
* OSPCALLED: outbound called number
|
|
* OSPCALLING: outbound calling number
|
|
* OSPOUTTOKEN: outbound OSP token
|
|
* OSPRESULTS: number of remain destinations
|
|
* OSPOUTTIMELIMIT: outbound call duration limit
|
|
* OSPOUTCALLID: outbound call ID. Only for H.323
|
|
* OSPDIALSTR: outbound dial string
|
|
* OSPNEXTSTATUS: OSPLookup return value. SUCCESS/FAILED/ERROR
|
|
|
|
3.2.4 OSPFinish
|
|
OSP report usage function.
|
|
Input:
|
|
* OSPINHANDLE: inbound transaction handle
|
|
* OSPOUTHANDLE: outbound transaction handle
|
|
* OSPAUTHSTATUS: OSPAuth return value
|
|
* OSPLOOKUPTSTATUS: OSPLookup return value
|
|
* OSPNEXTSTATUS: OSPNext return value
|
|
* cause: last destination disconnect cause
|
|
* priority jump
|
|
Output:
|
|
* OSPFINISHSTATUS: OSPLookup return value. SUCCESS/FAILED/ERROR
|
|
|
|
3.3 extensions.conf Examples
|
|
The extensions.conf file example provided in Section 3.1 is designed to
|
|
handle all OSP call scenarios when Asterisk is used as a source or destination
|
|
gateway to the PSTN or as a proxy between VoIP networks. The extenstion.conf
|
|
examples in this section are designed for specific use cases only.
|
|
|
|
3.3.1 Source Gateway
|
|
The examples in this section apply when the Asterisk server is being used as
|
|
a TDM to VoIP gateway. Calls originate on the TDM network and are converted
|
|
to VoIP by Asterisk. In these cases, the Asterisk server queries an OSP
|
|
server to find a route to a VoIP destination. When the call ends, Asterisk
|
|
sends a CDR to the OSP server.
|
|
For SIP protocol.
|
|
[SIPSrcGW]
|
|
exten => _XXXX.,1,NoOp(SIPSrcGW)
|
|
; Set calling number if necessary
|
|
exten => _XXXX.,n,Set(CALLERID(numner)=CallingNumber)
|
|
; OSP lookup using default provider, if fail/error jump to lookup+101
|
|
exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||j)
|
|
; Set calling number which may be translated
|
|
exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
|
|
; Dial to destination, 60 timeout, with call duration limit
|
|
exten => _XXXX.,n,Dial(${OSPDIALSTR},60,oL($[${OSPOUTTIMELIMIT}*1000]))
|
|
; Wait 3 seconds
|
|
exten => _XXXX.,n,Wait,3
|
|
; Hangup
|
|
exten => _XXXX.,n,Hangup
|
|
; Deal with OSPLookup fail/error
|
|
exten => _XXXX.,lookup+101,Hangup
|
|
exten => h,1,NoOp()
|
|
; OSP report usage
|
|
exten => h,n,OSPFinish(${HANGUPCAUSE})
|
|
For IAX protocol.
|
|
[IAXSrcGW]
|
|
exten => _XXXX.,1,NoOp(IAXSrcGW)
|
|
; Set calling number if necessary
|
|
exten => _XXXX.,n,Set(CALLERID(numner)=CallingNumber)
|
|
; OSP lookup using default provider, if fail/error jump to lookup+101
|
|
exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||j)
|
|
; Set outbound OSP token
|
|
exten => _XXXX.,n,Set(IAXCHANINFO(osptoken)=${OSPOUTTOKEN})
|
|
; Set calling number which may be translated
|
|
exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
|
|
; Dial to destination, 60 timeout, with call duration limit
|
|
exten => _XXXX.,n,Dial(${OSPDIALSTR},60,oL($[${OSPOUTTIMELIMIT}*1000]))
|
|
; Wait 3 seconds
|
|
exten => _XXXX.,n,Wait,3
|
|
; Hangup
|
|
exten => _XXXX.,n,Hangup
|
|
; Deal with OSPLookup fail/error
|
|
exten => _XXXX.,lookup+101,Hangup
|
|
exten => h,1,NoOp()
|
|
; OSP report usage
|
|
exten => h,n,OSPFinish(${HANGUPCAUSE})
|
|
For H.323 protocol.
|
|
[H323SrcGW]
|
|
exten => _XXXX.,1,NoOp(H323SrcGW)
|
|
; Set calling number if necessary
|
|
exten => _XXXX.,n,Set(CALLERID(numner)=CallingNumber)
|
|
; OSP lookup using default provider, if fail/error jump to lookup+101
|
|
; "h" parameter is used to generate a call id
|
|
; Cisco OSP gateways use this call id to validate OSP token
|
|
exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||jh)
|
|
; Set outbound call id
|
|
exten => _XXXX.,n,Set(OH323CHANINFO(callid)=${OSPOUTCALLID})
|
|
; Set outbound OSP token
|
|
exten => _XXXX.,n,Set(OH323CHANINFO(osptoken)=${OSPOUTTOKEN})
|
|
; Set calling number which may be translated
|
|
exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
|
|
; Dial to destination, 60 timeout, with call duration limit
|
|
exten => _XXXX.,n,Dial(${OSPDIALSTR},60,oL($[${OSPOUTTIMELIMIT}*1000]))
|
|
; Wait 3 seconds
|
|
exten => _XXXX.,n,Wait,3
|
|
; Hangup
|
|
exten => _XXXX.,n,Hangup
|
|
; Deal with OSPLookup fail/error
|
|
exten => _XXXX.,lookup+101,Hangup
|
|
exten => h,1,NoOp()
|
|
; OSP report usage
|
|
exten => h,n,OSPFinish(${HANGUPCAUSE})
|
|
|
|
3.3.2 Destination Gateway
|
|
The examples in this section apply when Asterisk is being used as a VoIP to
|
|
TDM gateway. VoIP calls are received by Asterisk which validates the OSP
|
|
peering token and completes to the TDM network. After the call ends,
|
|
Asterisk sends a CDR to the OSP server.
|
|
For SIP protocol
|
|
[SIPDstGW]
|
|
exten => _XXXX.,1,NoOp(SIPDstGW)
|
|
; Get peer IP
|
|
exten => _XXXX.,n,Set(OSPPEERIP=${SIPCHANINFO(peerip)})
|
|
; Get OSP token
|
|
exten => _XXXX.,n,Set(OSPINTOKEN=${SIP_HEADER(P-OSP-Auth-Token)})
|
|
; Validate token using default provider, if fail/error jump to auth+101
|
|
exten => _XXXX.,n(auth),OSPAuth(|j)
|
|
; Ringing
|
|
exten => _XXXX.,n,Ringing
|
|
; Wait 1 second
|
|
exten => _XXXX.,n,Wait,1
|
|
; Dial phone, timeout 15 seconds, with call duration limit
|
|
exten => _XXXX.,n,Dial(${DIALOUTANALOG}/${EXTEN:1},15,oL($[${OSPINTIMELIMIT}*1000]))
|
|
; Wait 3 seconds
|
|
exten => _XXXX.,n,Wait,3
|
|
; Hangup
|
|
exten => _XXXX.,n,Hangup
|
|
; Deal with OSPAuth fail/error
|
|
exten => _XXXX.,auth+101,Hangup
|
|
exten => h,1,NoOp()
|
|
; OSP report usage
|
|
exten => h,n,OSPFinish(${HANGUPCAUSE})
|
|
For IAX protocol
|
|
[IAXDstGW]
|
|
exten => _XXXX.,1,NoOp(IAXDstGW)
|
|
; Get peer IP
|
|
exten => _XXXX.,n,Set(OSPPEERIP=${IAXPEER(CURRENTCHANNEL)})
|
|
; Get OSP token
|
|
exten => _XXXX.,n,Set(OSPINTOKEN=${IAXCHANINFO(osptoken)})
|
|
; Validate token using default provider, if fail/error jump to auth+101
|
|
exten => _XXXX.,n(auth),OSPAuth(|j)
|
|
; Ringing
|
|
exten => _XXXX.,n,Ringing
|
|
; Wait 1 second
|
|
exten => _XXXX.,n,Wait,1
|
|
; Dial phone, timeout 15 seconds, with call duration limit
|
|
exten => _XXXX.,n,Dial(${DIALOUTANALOG}/${EXTEN:1},15,oL($[${OSPINTIMELIMIT}*1000]))
|
|
; Wait 3 seconds
|
|
exten => _XXXX.,n,Wait,3
|
|
; Hangup
|
|
exten => _XXXX.,n,Hangup
|
|
; Deal with OSPAuth fail/error
|
|
exten => _XXXX.,auth+101,Hangup
|
|
exten => h,1,NoOp()
|
|
; OSP report usage
|
|
exten => h,n,OSPFinish(${HANGUPCAUSE})
|
|
For H.323 protocol
|
|
[H323DstGW]
|
|
exten => _XXXX.,1,NoOp(H323DstGW)
|
|
; Get peer IP
|
|
exten => _XXXX.,n,Set(OSPPEERIP=${H323CHANINFO(peerip)})
|
|
; Get OSP token
|
|
exten => _XXXX.,n,Set(OSPINTOKEN=${H323CHANINFO(osptoken)})
|
|
; Validate token using default provider, if fail/error jump to auth+101
|
|
exten => _XXXX.,n(auth),OSPAuth(|j)
|
|
; Ringing
|
|
exten => _XXXX.,n,Ringing
|
|
; Wait 1 second
|
|
exten => _XXXX.,n,Wait,1
|
|
; Dial phone, timeout 15 seconds, with call duration limit
|
|
exten => _XXXX.,n,Dial(${DIALOUTANALOG}/${EXTEN:1},15,oL($[${OSPINTIMELIMIT}*1000]))
|
|
; Wait 3 seconds
|
|
exten => _XXXX.,n,Wait,3
|
|
; Hangup
|
|
exten => _XXXX.,n,Hangup
|
|
; Deal with OSPAuth fail/error
|
|
exten => _XXXX.,auth+101,Hangup
|
|
exten => h,1,NoOp()
|
|
; OSP report usage
|
|
exten => h,n,OSPFinish(${HANGUPCAUSE})
|
|
|
|
3.3.3 Proxy
|
|
The example in this section applies when Asterisk is a proxy between two VoIP networks.
|
|
[GeneralProxy]
|
|
exten => _XXXX.,1,NoOp(GeneralProxy)
|
|
; Get peer IP and inbound OSP token
|
|
; SIP, un-comment the following two lines.
|
|
;exten => _XXXX.,n,Set(OSPPEERIP=${SIPCHANINFO(peerip)})
|
|
;exten => _XXXX.,n,Set(OSPINTOKEN=${SIP_HEADER(P-OSP-Auth-Token)})
|
|
; IAX, un-comment the following 2 lines
|
|
;exten => _XXXX.,n,Set(OSPPEERIP=${IAXPEER(CURRENTCHANNEL)})
|
|
;exten => _XXXX.,n,Set(OSPINTOKEN=${IAXCHANINFO(osptoken)})
|
|
; H323, un-comment the following two lines.
|
|
;exten => _XXXX.,n,Set(OSPPEERIP=${OH323CHANINFO(peerip)})
|
|
;exten => _XXXX.,n,Set(OSPINTOKEN=${OH323CHANINFO(osptoken)})
|
|
;---------------------------------------------------------------
|
|
; Validate token using default provider, if fail/error jump to auth+101
|
|
exten => _XXXX.,n(auth),OSPAuth(|j)
|
|
; OSP lookup using default provider, if fail/error jump to lookup+101
|
|
; "h" parameter is used to generate a call id for H.323 destinations
|
|
; Cisco OSP gateways use this call id to validate OSP token
|
|
exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||jh)
|
|
; Set outbound call id and OSP token
|
|
; IAX, un-comment the following line.
|
|
;exten => _XXXX.,n,Set(IAXCHANINFO(osptoken)=${OSPOUTTOKEN})
|
|
; H323, un-comment the following two lines.
|
|
;exten => _XXXX.,n,Set(OH323CHANINFO(callid)=${OSPOUTCALLID})
|
|
;exten => _XXXX.,n,Set(OH323CHANINFO(osptoken)=${OSPOUTTOKEN})
|
|
;---------------------------------------------------------------
|
|
; Set calling number which may be translated
|
|
exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
|
|
; Dial to destination, 14 timeout, with call duration limit
|
|
exten => _XXXX.,n,Dial(${OSPDIALSTR},14,oL($[${OSPOUTTIMELIMIT}*1000]))
|
|
; OSP lookup next destination using default provider, if fail/error jump to next1+101
|
|
exten => _XXXX.,n(next1),OSPNext(${HANGUPCAUSE}||j)
|
|
; Set outbound call id and OSP token
|
|
; IAX, un-comment the following line.
|
|
;exten => _XXXX.,n,Set(IAXCHANINFO(osptoken)=${OSPOUTTOKEN})
|
|
; H323, un-comment the following two lines.
|
|
;exten => _XXXX.,n,Set(OH323CHANINFO(callid)=${OSPOUTCALLID})
|
|
;exten => _XXXX.,n,Set(OH323CHANINFO(osptoken)=${OSPOUTTOKEN})
|
|
;---------------------------------------------------------------
|
|
; Set calling number which may be translated
|
|
exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
|
|
; Dial to destination, 15 timeout, with call duration limit
|
|
exten => _XXXX.,n,Dial(${OSPDIALSTR},15,oL($[${OSPOUTTIMELIMIT}*1000]))
|
|
; OSP lookup next destination using default provider, if fail/error jump to next2+101
|
|
exten => _XXXX.,n(next2),OSPNext(${HANGUPCAUSE}||j)
|
|
; Set outbound call id and OSP token
|
|
; IAX, un-comment the following line.
|
|
;exten => _XXXX.,n,Set(IAXCHANINFO(osptoken)=${OSPOUTTOKEN})
|
|
; H323, un-comment the following two lines.
|
|
;exten => _XXXX.,n,Set(OH323CHANINFO(callid)=${OSPOUTCALLID})
|
|
;exten => _XXXX.,n,Set(OH323CHANINFO(osptoken)=${OSPOUTTOKEN})
|
|
;---------------------------------------------------------------
|
|
; Set calling number which may be translated
|
|
exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
|
|
; Dial to destination, 16 timeout, with call duration limit
|
|
exten => _XXXX.,n,Dial(${OSPDIALSTR},16,oL($[${OSPOUTTIMELIMIT}*1000]))
|
|
; Hangup
|
|
exten => _XXXX.,n,Hangup
|
|
; Deal with OSPAuth fail/error
|
|
exten => _XXXX.,auth+101,Hangup
|
|
; Deal with OSPLookup fail/error
|
|
exten => _XXXX.,lookup+101,Hangup
|
|
; Deal with 1st OSPNext fail/error
|
|
exten => _XXXX.,next1+101,Hangup
|
|
; Deal with 2nd OSPNext fail/error
|
|
exten => _XXXX.,next2+101,Hangup
|
|
exten => h,1,NoOp()
|
|
; OSP report usage
|
|
exten => h,n,OSPFinish(${HANGUPCAUSE})
|