[SR-Users] Kamailio not sending reply
Olle E. Johansson
oej at edvina.net
Thu May 26 19:41:37 CEST 2016
> On 26 May 2016, at 18:47, Jeremy Betts <jeremy.betts at freevoiceusa.com> wrote:
>
> Hello,
>
> I'm having an issue where kamailio is not sending the failure reply as I would expect when receiving calls from a certain provider. Kamailio sends the 403 response on calls from other providers. The big difference I see is that the "problem" provider is using compact headers. I am trying to force the 403 response to be sent for testing purposes but I just can't get it to reply to this one provider.
>
> I've included traces and configuration excerpts below. Any help is much appreciated!
>
> Problem Call (No Reply Sent):
There must be something missing here. Without a response, there would be no ACK.
/O
>
> U 63.79.178.192:5060 -> 184.171.164.100:5060
> INVITE sip:+19727289377 at 184.171.164.100;transport=UDP;user=phone <sip:+19727289377 at 184.171.164.100;transport=UDP;user=phone> SIP/2.0.
> v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013.
> Record-Route: <sip:63.79.178.192;lr> <sip:63.79.178.192;lr>.
> f: <sip:+17143258018 at 199.173.94.144:5060;user=phone> <sip:+17143258018 at 199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce.
> t: <sip:+19727289377 at 63.79.178.192:5060;user=phone> <sip:+19727289377 at 63.79.178.192:5060;user=phone>.
> i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719.
> CSeq: 1 INVITE.
> Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK.
> v: SIP/2.0/UDP SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-41c7bce-f439f11-21b78f96;received=199.173.94.144.
> Max-Forwards: 27.
> m: <sip:199.173.94.144:5060;transport=UDP> <sip:199.173.94.144:5060;transport=UDP>.
> k: 100rel, resource-priority, replaces.
> c: application/sdp.
> l: 235.
> P-Asserted-Identity: <sip:+17143258018 at 63.79.178.192;user=phone> <sip:+17143258018 at 63.79.178.192;user=phone>.
> Privacy: none.
> .
> v=0.
> o=PVG 1464277233580 1464277233580 IN IP4 199.173.68.106.
> s=-.
> p=+1 6135555555.
> c=IN IP4 199.173.68.106.
> t=0 0.
> m=audio 55380 RTP/AVP 18 0 8 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=fmtp:18 annexb=no.
>
> #
> U 63.79.178.192:5060 -> 184.171.164.100:5060
> ACK sip:+19727289377 at 184.171.164.100;transport=UDP;user=phone <sip:+19727289377 at 184.171.164.100;transport=UDP;user=phone> SIP/2.0.
> v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013.
> f: <sip:+17143258018 at 199.173.94.144:5060;user=phone> <sip:+17143258018 at 199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce.
> t: <sip:+19727289377 at 63.79.178.192:5060;user=phone> <sip:+19727289377 at 63.79.178.192:5060;user=phone>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.1aac.
> i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719.
> CSeq: 1 ACK.
> l: 0.
> Max-Forwards: 27.
> .
>
> Successful Call (Reply Sent):
>
> #
> U 64.136.173.31:5060 -> 184.171.164.100:5066
> INVITE sip:7146466334 at 184.171.164.100:5066 <sip:7146466334 at 184.171.164.100:5066> SIP/2.0.
> Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345.
> Record-Route: <sip:sansay3408006464rdb14345 at 64.136.173.31:5060;lr;transport=udp> <sip:sansay3408006464rdb14345 at 64.136.173.31:5060;lr;transport=udp>.
> To: <sip:7146466334 at 184.171.164.100> <sip:7146466334 at 184.171.164.100>.
> From: <sip:17143258018 at 64.136.173.31> <sip:17143258018 at 64.136.173.31>;tag=sansay3408006464rdb14345.
> Call-ID: 1089333243-0-3123809354 at 64.136.173.226 <mailto:1089333243-0-3123809354 at 64.136.173.226>.
> CSeq: 1 INVITE.
> Contact: <sip:17143258018 at 64.136.173.31:5060> <sip:17143258018 at 64.136.173.31:5060>.
> Supported: timer.
> Session-Expires: 1800;refresher=uac.
> Min-SE: 90.
> P-Asserted-Identity: <sip:17143258018 at 192.168.20.76> <sip:17143258018 at 192.168.20.76>.
> Privacy: none.
> Expires: 120.
> Max-Forwards: 67.
> Content-Type: application/sdp.
> Content-Length: 274.
> .
> v=0.
> o=Sansay-VSXi 188 1 IN IP4 64.136.173.31.
> s=Session Controller.
> c=IN IP4 69.85.185.142.
> t=0 0.
> m=audio 37970 RTP/AVP 0 18 101.
> a=rtpmap:0 PCMU/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=ptime:20.
>
>
> #
> U 184.171.164.100:5066 -> 64.136.173.31:5060
> SIP/2.0 403 DID Lookup Failed.
> Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345.
> To: <sip:7146466334 at 184.171.164.100> <sip:7146466334 at 184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec.
> From: <sip:17143258018 at 64.136.173.31> <sip:17143258018 at 64.136.173.31>;tag=sansay3408006464rdb14345.
> Call-ID: 1089333243-0-3123809354 at 64.136.173.226 <mailto:1089333243-0-3123809354 at 64.136.173.226>.
> CSeq: 1 INVITE.
> Server: x-Freevoice SIP Proxy 4.21.
> Content-Length: 0.
> .
>
> #
> U 64.136.173.31:5060 -> 184.171.164.100:5066
> ACK sip:7146466334 at 184.171.164.100:5066 <sip:7146466334 at 184.171.164.100:5066> SIP/2.0.
> Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345.
> To: <sip:7146466334 at 184.171.164.100> <sip:7146466334 at 184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec.
> From: <sip:17143258018 at 64.136.173.31> <sip:17143258018 at 64.136.173.31>;tag=sansay3408006464rdb14345.
> Call-ID: 1089333243-0-3123809354 at 64.136.173.226 <mailto:1089333243-0-3123809354 at 64.136.173.226>.
> CSeq: 1 ACK.
> Max-Forwards: 70.
> Content-Length: 0.
>
> Kamailio Config:
> request_route {
>
> # per request initial checks
> route(REQINIT);
>
> # NAT detection
> #route(NATDETECT);
>
> # handle requests within SIP dialogs
> route(WITHINDLG);
>
> ### only initial requests (no To tag)
>
> # CANCEL processing
> if (is_method("CANCEL"))
> {
> setflag(FLT_ACCFAILED);
> if (t_check_trans())
> t_relay();
> exit;
> }
>
> t_check_trans();
>
> if(is_method("INVITE")||is_method("MESSAGE")){
> route(ORIGINATE);
> #route(FROMAST);
> }
>
>
> route[ORIGINATE] {
>
> sl_send_reply("403", "DID Lookup Failed");
> exit;
> }
>
>
> Jeremy Betts
>
> <https://www.youtube.com/user/FreevoiceLLC>
> <https://www.youtube.com/user/FreevoiceLLC>_______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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