Hi, RFC 3261 and RFC 2966 define rules for converting a TEL URI into a SIP URI (really complex and un-useful in fact).
Are there rules for the inverse conversion (SIP to TEL)? Let's take the following example: --------------------------------------- SIP URI: sip:+34666555444;isub=+1234;ext=200;custom-param=...@mydomaing.org;user=phone Doing the reverse rules it will become the following TEL URI: TEL URI: tel:+34666555444;isub=+1234;ext=200;custom-param=yes ---------------------------------------- What to do with the SIP URI host part? Also, as you can appreciate this conversion inplies SIP userinfo parsing to extract "parameters" and create with them the TEL URI parameters (that's a pain!). Could these SIP userinfo "parameters" be ignored?: --------------------------------------- SIP URI: sip:+34666555444;isub=+1234;ext=200;custom-param=...@mydomaing.org;user=phone TEL URI: tel:+34666555444 ---------------------------------------- I suspect this would be incorrect :( Thanks for any clarification on it. -- Iñaki Baz Castillo _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
