[asterisk-dev] [Code Review] 4476: new res_pjsip module to identify endpoint for an incoming call with a trunk that has outbound registration.

Dmitriy Serov reviewboard at asterisk.org
Tue Mar 24 07:23:59 CDT 2015


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4476/
-----------------------------------------------------------

(Updated March 24, 2015, 12:23 p.m.)


Review request for Asterisk Developers, Matt Jordan and rnewton.


Changes
-------

Fixed Copyright and the name of module.


Bugs: ASTERISK-24852
    https://issues.asterisk.org/jira/browse/ASTERISK-24852


Repository: Asterisk


Description
-------

The main task: to find the right endpoint for an incoming call with a trunk that has outbound registration.

Simple example (a piece of pjsip.conf):
[trunk1]
type=registration
transport=udp-transport
outbound_auth=trunk1
server_uri=sip:sip.example.com
client_uri=sip:user1 at sip.example.com
contact_user=trunk1-in

[trunk2]
type=registration
transport=udp-transport
outbound_auth=trunk2
server_uri=sip:sip.example.com
client_uri=sip:user2 at sip.example.com
contact_user=trunk2-in

[trunk1-in]
type=endpoint
context=from-trunk
disallow=all
allow=ulaw
outbound_auth=trunk1
aors=trunk1

[trunk2-in]
type=endpoint
context=from-trunk
disallow=all
allow=ulaw
outbound_auth=trunk2
aors=trunk2

trunk1, trunk2 - outbound registrations to EXTERNAL sip server sip.example.com. One server and two registrations.
; "contact_user=" sets the SIP contact header's user portion of the SIP URI this will affect the extension reached in dialplan when the far end calls you at this registration.
In example option has values "trunk1-in" and "trink2-in".

Case: random user admin at sip.example.com calling to user2 at sip.example.com.
External sip server redirect call to my server with packet:
INVITE sip:trunk2-in at 8.8.8.8:5060 SIP/2.0
From: "PhonerLite" <sip:admin at sip.example.com>;tag=5F39A540-782390-DE41886A_kmbdctn-8A47
To: <sip:user2 at sip.example.com>

This invite can be:
- anonymous. Very bad.
- identified by IP. Config has two endpoints with same IP.
- identified by username From. It cannot identify by "admin" (random)

res_pjsip_endpoint_identifier_request_user helps to identify this invite by uri username in request line (contact_user in registration).


Diffs (updated)
-----

  /trunk/res/res_pjsip_endpoint_identifier_request_user.c PRE-CREATION 

Diff: https://reviewboard.asterisk.org/r/4476/diff/


Testing
-------


Thanks,

Dmitriy Serov

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20150324/f4a60606/attachment.html>


More information about the asterisk-dev mailing list