[OpenSIPS-Users] Problem with double field P-Asserted-Identity
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Mon Apr 6 19:02:28 CEST 2009
Hi Matteo,
Whatever change you in request_route will propagate in all branches -
this is your problem - when you set the PAI for the first destination,
in request_route, the PAI will be present in all future branches.
To avoid this situation, avoid adding "per-branch" info from
request_branch - do it from branch_route; branch_route and failure_route
allows changes per branch (and not global ones).
Do something like:
request_route -> do_routing + arm branch_route
branch_route -> set PAI for the first branch
failure_route -> set PAI for the next branch
Regards,
Bogdan
mmarzuola at interfree.it wrote:
> Hi all. In my scenario I try to use dynamic routing and I have a problem with the P-Asserted Identity field. When the proxy tries to send the INVITE to the first gateway selected from DRouting module by the function do_routing, there is only one PAI field, but if that gateway is down, at the second attempt to next gateway, the INVITE contains two PAI fields identicals.
>
> This is a piece of code:
>
> ########################################################################
> # Request route 'base-outbound'
> ########################################################################
> route[2] {
> if(is_present_hf("P-Asserted-Identity")) {
> remove_hf("P-Asserted-Identity");
> }
> if(is_present_hf("Remote-Party-ID")) {
> remove_hf("Remote-Party-ID");
> }
> if(is_avp_set("$avp(s:caller_cli)/s")) {
> if(!isflagset(28)) {
> xlog("L_INFO", "Set caller CLI '$avp(s:caller_cli)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
> append_hf("P-Asserted-Identity: <$avp(s:caller_cli)>\r\n");
> }
> }
>
> route(14);
>
> t_on_reply("1");
>
> if(!isflagset(21)) {
> t_on_failure("1");
> }
> if(is_present_hf("Proxy-Authorization")) {
> consume_credentials();
> }
>
> xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
>
> if(!t_relay()) {
> sl_reply_error();
> }
> exit;
> }
>
> ########################################################################
> # Request route 'invite-to-external'
> ########################################################################
> route[6] {
> if(isflagset(20)) {
> xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
> route(2);
> exit;
> }
> if(!isflagset(23)) {
> # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded
> if(uri=~ "^sip:[0-9]+@") {
> xlog("L_INFO", "Call to PSTN\n");
> do_routing();
> xlog("L_INFO", "first attempt is $ru, attributes are $avp(s:drattrs)\n");
> if(!goes_to_gw()) {
> xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
> sl_send_reply("503", "PSTN Termination Currently Unavailable");
> exit;
> }
> setflag(21);
> t_on_failure("2");
> route(2);
> exit;
> }
> }
>
> xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
> sl_send_reply("404", "User Not Found");
> exit;
> }
>
> ########################################################################
> # Failure route 'pstn-failover'
> ########################################################################
> failure_route[2] {
> xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
> route(9);
>
> if(!use_next_gw()) {
> xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
> exit;
> }
> xlog("L_INFO", "selected next gateway $ru, attributes are $avp(s:drattrs)\n");
> t_on_failure("2");
> route(2);
> }
>
>
> These are the INVITEs:
>
> U 10.10.45.172:5060 -> 10.10.45.86:5060
> INVITE sip:22230489 at 10.10.45.86 SIP/2.0.
> Record-Route: <sip:10.10.45.172;lr=on;ftag=2061433461944847689>.
> Via: SIP/2.0/UDP 10.10.45.172;branch=z9hG4bK8586.95959e37.0.
> Via: SIP/2.0/UDP 10.10.45.102:1025;received=10.10.45.102;rport=1025;branch=z9hG4bK0a0a2d660000001549d9d2ae0ccb24110000004a.
> Content-Length: 366.
> Contact: <sip:1000 at 10.10.45.102:1025>.
> Call-ID: B998BB4A-1DD1-11B2-8CD3-D001388A831D at 10.10.45.102.
> Content-Type: application/sdp.
> CSeq: 2 INVITE.
> From: "Sjphone laptop_User"<sip:1000 at 10.10.45.172>;tag=2061433461944847689.
> Max-Forwards: 69.
> To: <sip:30489 at 10.10.45.172>.
> User-Agent: SJphone/1.60.299a/L (SJ Labs).
> P-Asserted-Identity: <sip:003930948 at 10.10.45.172>.
> .
> v=0.
> o=- 3448000814 3448000814 IN IP4 10.10.45.102.
> s=SJphone.
> c=IN IP4 10.10.45.102.
> t=0 0.
> a=direction:active.
> m=audio 49154 RTP/AVP 3 97 98 110 8 0 101.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:97 iLBC/8000.
> a=rtpmap:98 iLBC/8000.
> a=fmtp:98 mode=20.
> a=rtpmap:110 speex/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-11,16.
>
>
> U 10.10.45.172:5060 -> 10.10.45.227:5060
> INVITE sip:22230489 at 10.10.45.227 SIP/2.0.
> Record-Route: <sip:10.10.45.172;lr=on;ftag=2061433461944847689>.
> Via: SIP/2.0/UDP 10.10.45.172;branch=z9hG4bK8586.95959e37.1.
> Via: SIP/2.0/UDP 10.10.45.102:1025;received=10.10.45.102;rport=1025;branch=z9hG4bK0a0a2d660000001549d9d2ae0ccb24110000004a.
> Content-Length: 366.
> Contact: <sip:1000 at 10.10.45.102:1025>.
> Call-ID: B998BB4A-1DD1-11B2-8CD3-D001388A831D at 10.10.45.102.
> Content-Type: application/sdp.
> CSeq: 2 INVITE.
> From: "Sjphone laptop_User"<sip:1000 at 10.10.45.172>;tag=2061433461944847689.
> Max-Forwards: 69.
> To: <sip:30489 at 10.10.45.172>.
> User-Agent: SJphone/1.60.299a/L (SJ Labs).
> P-Asserted-Identity: <sip:003930948 at 10.10.45.172>.
> P-Asserted-Identity: <sip:003930948 at 10.10.45.172>.
> .
> v=0.
> o=- 3448000814 3448000814 IN IP4 10.10.45.102.
> s=SJphone.
> c=IN IP4 10.10.45.102.
> t=0 0.
> a=direction:active.
> m=audio 49154 RTP/AVP 3 97 98 110 8 0 101.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:97 iLBC/8000.
> a=rtpmap:98 iLBC/8000.
> a=fmtp:98 mode=20.
> a=rtpmap:110 speex/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-11,16.
>
> Thanks in advance.
>
> Matteo Marzuola
>
>
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