Hi, You can try with SIPCHANINFO function otherwise you need to modify chan_sip.c for getting this addresses.
thanks Dhaval On Tue, Aug 7, 2012 at 12:10 AM, CB <[email protected]> wrote: > We are looking to further secure our Asterisk installation by inspecting > the > IP address that a SIP INVITE comes from and performing some logic to > determine whether the call should proceed. The purpose of this is to > prevent > calls to certain expensive destinations if the SIP message is coming from a > foreign IP that we don't expect. > > I can see that it's possible to use the SIP_HEADER function however that > may > not contain the public IP address. For example here is an invite from the > external IP address 58.28.1.1 but that information is not contained in the > SIP header: > U 58.28.1.1:5060 -> 203.89.1.1:5060 > INVITE sip:[email protected] SIP/2.0..Via: SIP/2.0/UDP > 192.168.1.103:5060 > ;branch=z9hG4bK-d8754z-fc116e03a80ef774-1---d8754z-;rport. > .Max-Forwards: 70 > ..Contact: <sip:[email protected]:5060>..To: > <sip:[email protected]>..From: <sip:[email protected]>;tag=7 > dcb1e4d..Call-ID: NDMyZmRhY2Q4ZjNhMjAxMDJhOTA3OTU0MzMyNTkzNjI...CSeq: 1 > INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INF > O..Content-Type: application/sdp..Supported: replaces..User-Agent: X-Lite > release 5.0.0 stamp 67284..Content-Length: 217....v=0..o=- > 12988751314362048 > 1 IN IP4 > 192.168.1.103..s=CounterPath X-Lite 5.0.0..c=IN IP4 > 192.168.1.103..b=AS:1638..t=0 0..m=audio 5062 RTP/AVP 0 8 3 > 101..a=rtpmap:101 telephone-event/8000..a=fmtp:1 > 01 0-15..a=sendrecv.. > > Is it possible to determine the public IP address from the dialplan? > > Any advice appreciated. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
