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[Asterisk] SIP URI Dialing with Asterisk and FreePBX

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In developing a methodology for users to call SIP URIs using registered softphones and SIP phones with Asterisk, we discovered that there does not appear to be a single Asterisk variable that stores the actual dial string entered by the caller. For example, calling 8005551212 and 8005551212@somedomain.com both get passed to Asterisk in ${EXTEN} as 8005551212. Instead, we had to compare {SIPCALLID} and {SIPDOMAIN} to determine whether the call was being directed to an external domain. And the problem with that was that {SIPDOMAIN} didn't exist in some earlier versions of Asterisk. Have I missed something??

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