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lgaetz edited this page Feb 18, 2013
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SIP URI Handling is a FreePBX module to allow FreePBX extension to dial SIP/URI format through the PBX in the same fashion as a normal DID.
This module requires FreePBX version 2.9 or later. It has been tested with Asterisk version 1.8 but will probably work with earlier and later versions. This module is known to conflict with the 'Custom Contexts' module; testing on a system with 'Custom Contexts' enabled was not successful.
- Download the desired version of the module from downloads: https://github.com/POSSA/freepbx-SIP-URI-handling/downloads
- In FreePBX navigate to 'Module Admin' and click 'upload module'. Browse to the file downloaded in step 1 and upload
- Click 'Manage Modules' and scroll down the list to 'URI Handling'. Click it and select 'install'
- Click 'process' at the top/bottom of the page and follow the directions to complete the install.
- From the 'Other' menu choose 'URI Handling'
- There are three fields for FQDN and/or IP addresses in order to identify which SIP/URI formats are local. Any dialed URI with one of these name/IP will be treated as local, the domain will be stripped from the URI and the call passed to the context 'from-internal' to be handled in the normal fashion. All other URI will be SIP dialed bypassing the outbound routes.
- Any extension that requires URI dialing must be manually placed in a context of 'enable-sipuri-dialing' All extensions configured for that context are listed in the table at the bottom of the URI Handling page.