diff --git a/src/modules/siputils/doc/siputils_admin.xml b/src/modules/siputils/doc/siputils_admin.xml
index 63984c60c67..d2197b1c3f4 100644
--- a/src/modules/siputils/doc/siputils_admin.xml
+++ b/src/modules/siputils/doc/siputils_admin.xml
@@ -539,13 +539,15 @@ if (uri_param_rm("param1")) {
The conversion follows the rules in RFC 3261 section 19.1.6:
-
- Visual separators ( "-", ".", "(", ")" ) are removed from tel URI number before converting it to SIP URI userinfo.
+
+ Visual separators ( "-", ".", "(", ")" ) are removed
+ from tel URI number before converting it to SIP URI userinfo.
- tel URI parameters are downcased before appending them to SIP URI userinfo
-
-
+ tel URI parameters are downcased before appending them
+ to SIP URI userinfo
+
+
The SIP URI hostpart is taken from second param
@@ -569,6 +571,36 @@ tel2sip("$ru", $fd", "$ru");
tel2sip("$ru", $fd", "$ru");
# $ru: sip:+12345678;ext=200;isub=+123-456@foo.com;user=phone
...
+
+
+
+
+
+ tel2sip2(uri, hostpart, result)
+
+
+ Alternative to sip2tel() function that tries to follow closer the RFC
+ requrements (e.g., sort tel: uri parameters copied to the sip: uri in
+ the manner defined in the standard; deletes the phone-context parameter
+ if it is a domain, and, takes visual separators from the phone-context
+ parameter if it is a telephone number).
+
+
+ Its parameters have the same meaning as for tel2sip().
+
+
+ This function can be used from REQUEST_ROUTE, FAILURE_ROUTE,
+ BRANCH_ROUTE, or ONREPLY_ROUTE.
+
+
+ tel2sip2 usage
+
+...
+# $ru: tel:+(34)-999-888-777
+# $fu: sip:test@foo.com
+tel2sip2("$ru", $fd", "$ru");
+# $ru: sip:+34999888777@foo.com;user=phone
+...