Hi, I understand that the use of "tel" URI could be cool when the INVITE is received by a PSTN gateway (so an URI domain makes no sense at all) but I wonder who should create this tel URI.
For example, I consider very very difficult that a phone could handle "tel" URI's since it cannot know when the call is for a PSTN number, a private extension or an outbound call to other SIP domain, so IMHO a final UA should always use SIP/SIPS URI's and not "tel". But when the UA sends an INVITE to its proxy like: INVITE sip:[EMAIL PROTECTED] SIP/2.0 the proxy can detect that it's a PSTN number and convert it to "tel" URI before forward it to a PSTN gateway: INVITE tel:0034999000111 SIP/2.0 Is it correct? Note that I'm trying to imagine a real and feasible bahaviour. Please, don't suggest me that the phone could handle "tel" URI's since it's just an impossible dream, isn't? -- Iñaki Baz Castillo <[EMAIL PROTECTED]> _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
