I use something like this on my boxes to keep them foreigners out:

if ( !ds_is_from_list() ) { sl_send_reply("403","Your not in my dispatcher list"); }

Put this high up your config.

Cheers

On 18 Feb 2013, at 11:58, Benjamin Henrion wrote:

Hi,

Does someone know/uses a simple rule so that Kamailio only exchanges traffic with machines in the dispatcher?

Best,

-- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-3500762 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

---

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

I can get wan communication or lan communication working, but not both.

Setting this, allows wan communication, but then my lan tries to talk with my wan ip. Single Interface: listen=udp:<lan_ip>:5060 advertise <nat_wan_ip>:5060 I see my wan_ip in the via header, so I assume that's causing the issues for my lan. Using add_contact_alias for lan, with the advertise in my listen command, results in [nathelper.c:835]: no need to add alias param I've tried handle_ruri_alias();, but this doesn't seem to have an affect either.

Setting this, allows lan communication to work fine, but wan has issues. Single Interface: listen=udp:<lan_ip>:5060

Where is the happy median?

Was going to try path, but I read this is not the correct work-around.

Any clues or hints, are greatly appreciated.

U <carrier_wan_ip>:5060 -> <lan_ip>:5060 INVITE sip:+1<dialed_number>@<lan_ip>:5060 SIP/2.0..Via: SIP/2.0/UDP <carrier_wan_ip>:5060;branch=z9h G4bK04B0f1624ef34479ee3..From: <sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To : <sip:+1<dialed_number>@<lan_ip>:5060>..Call-ID: 184873373_18184370@<carrier_wan_ip..CSeq: 27475 IN VITE..Max-Forwards: 63..Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept: application/sdp, appl ication/isup, application/dtmf, application/dtmf-relay, multipart/mixed..Contact: <sip:+1720276 3205@<carrier_wan_ip:5060>..P-Asserted-Identity: <sip:+1<caller_id>@<carrier_wan_ip:5060>..Supported: 100 rel..Content-Length: 305..Content-Disposition: session; handling=required..Content-Type: applic ation/sdp....v=0..o=Sonus_UAC 25113 26131 IN IP4 <carrier_wan_ip..s=SIP Media Capabilities..c=IN IP4 4.55.10.130..t=0 0..m=audio 15746 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/800 0..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-1 5..a=sendrecv..a=maxptime:20..

U <lan_ip>:5060 -> <carrier_wan_ip>:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP <carrier_wan_ip:5060;branch=z9h G4bK04B0f1624ef34479ee3..From: <sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To : <sip:+1<dialed_number>@<lan_ip>:5060>..Call-ID: 184873373_18184370@<carrier_wan_ip..CSeq: 27475 IN VITE..Server: kamailio (3.3.2 (x86_64/linux))..Content-Length: 0....

U <lan_ip>:5060 -> <asterisk_ip>:5060 INVITE sip:+1<dialed_number>@<lan_ip>:5060 SIP/2.0..Via: SIP/2.0/UDP <wan_ip>:5060;branch= z9hG4bK5d66.7cb4b7a1.0..Via: SIP/2.0/UDP <carrier_wan_ip>:5060;branch=z9hG4bK04B0f1624ef34479ee3..Fro m: <sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To: <sip:+1<dialed_number>@192.168. 9.130:5060>..Call-ID: 184873373_18184370@<carrier_wan_ip>..CSeq: 27475 INVITE..Max-Forwards: 62..Allo w: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept: application/sdp, application/isup, application/dt mf, application/dtmf-relay, multipart/mixed..Contact: <sip:+1<caller_id>@<carrier_wan_ip:5060>..P-As serted-Identity: <sip:+1<caller_id>@<carrier_wan_ip:5060>..Supported: 100rel..Content-Length: 305..C ontent-Disposition: session; handling=required..Content-Type: application/sdp....v=0..o=Sonus_UA C 25113 26131 IN IP4 <carrier_wan_ip..s=SIP Media Capabilities..c=IN IP4 4.55.10.130..t=0 0..m=audio 15746 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a= fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..a=maxptime:20 ..

Asterisk box:

Retransmitting #4 (no NAT) to <wan_ip>:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP <wan_ip>:5060;branch=z9hG4bK5d66.7cb4b7a1.0;received=192.168.9.130 Via: SIP/2.0/UDP <carrier_wan_ip>:5060;branch=z9hG4bK04B0f1624ef34479ee3 From: <sip:+1<caller_id>@<carrier_wan_ip>:5060;isup-oli=0>;tag=gK0413c49b To: <sip:+1<dialed_number>@<lan_ip>:5060>;tag=as1265f4d0 Call-ID: 184873373_18184370@<carrier_wan_ip> CSeq: 27475 INVITE Server: Asterisk PBX 1.6.2.14 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:+1<dialed_number>@<asterisk_ip>> Content-Type: application/sdp Content-Length: 239

Matt Scott