[OpenSIPS-Users] Caller Name and P-Assterted
Nick Khamis
symack at gmail.com
Mon Apr 29 01:19:26 CEST 2013
Hello Everyone,
Is it possible to pass a meaningful caller name along with the caller id
given the carrier supports CLI?
Before asking the carrier, I was doing some tests.
I know that append_hf("P-Asserted-Identity: <sip:15453387463 at test.server.com>;
user=phone\r\n", "Call-ID") works
fine for the phone number alone (i.e., 15453387463) however,
append_hf("P-Asserted-Identity: From: \" Test User \" <
sip:15453387463 at test.server.com>; user=phone\r\n", "Call-ID") generates a
meaningless CID.
The SIP Trace: P-Asserted-Identity: From: "Mike Peer" <
sip:15453387463 at test.server.com>; user=phone.
Once I know what is accepted for PAI, I plan on assigning subscriber.rpid
that value. To load rpid for a specific caller and INVITE,
I have added the following code:
modparam("auth_db", "load_credentials", "rpid")
modparam("auth", "rpid_avp", "$avp(rpid)")
Testing from branch route, $avp(rpid) is NULL. We do not allow users to
REGISTER. Any way we can get subscriber.rpid for INVITES? Finally, is
branch_route and failure_routes, the safest place to append the PAI?
Thanks in Advance,
Nick
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