[SR-Users] sip: uri to tel: uri

[manuel espinoza]

Hi,

The arch looks like this


SBC1 <-------- SIP-Router (Kamailio) <------------- Asterisk

Asterisk sends an invite to kamailio using "sip:" as request uri scheme,
but SBC1 expects that invite to arrive using tel:, is there a way to change
this in kamailio and  forward that invite? Something like:

$ru = "tel: +" + $rU;

Thanks!!!
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