sr-users June 2015

sr-users@lists.kamailio.org

82 participants
136 discussions

Announcement: Kamailio is now systemd-rtc-server
by Alex Balashov 03 Aug '15

03 Aug '15

For immediate release: ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software vendor specialising in Kamailio-based service delivery solutions for the VoIP ITSP market, is pleased to announce that it, in collaboration with Red Hat Software and Ringfree Communications, has finalised the absorption of the Kamailio SIP Server into the 'systemd' system management platform for Linux. The new component shall be called 'systemd-rtc-server', or 'Systemd Real-Time Communication Server'. Alex Balashov, principal of Evariste and leader of the tri-vendor collaboration effort, will officially announce the handover of the reigns of the Kamailio project to the personal leadership of Lennart Poettering at the upcoming Systemd Real Time Communications World conference, to be held in Berlin on 27-29 May of this year. John Knight, Director of GNOME 3 Integration and part-time usability consultant at Ringfree Communications, based in Hendersonville, North Carolina,was quick to summarise the triumphs of the long-standing integration effort. Remarked Knight: "The industry has recognised for years that a SIP proxy is a basic building block in the 'init' subsystem of any Linux host. In this age of multimedia communication with voice and video, it was a travesty that systemd handled time synchronisation, network configuration, login management, logging, and console, but not SIP message routing." Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker, was quick to concur: "SIP calls are much easier to troubleshoot with binary logs. Combined with packet captures of TLS-encrypted WebRTC calls, systemd-journald is the ultimate call setup troubleshooting methodology of the responsive, kinetic enterprise." To support the integration of Kamailio into the ecosystem of every major Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio' modules for the project. Balashov stated, "We fully expect to use the D-Bus API to achieve gnome-session integration with systemd-rtc-server-usrloc, but we aren't going to leave Windows users behind; KamailioSvcHost.exe will support Domain Controller policies for G.722 in Active Directory forests." Despite an aggressive delivery timeline by the tri-vendor consortium behind systemd-rtc-server, industry commentators have widely lambasted the fact that it took so long for Kamailio to become integrated into systemd. Fred Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida, recently wrote in a widely-publicised blog post: "sr-dev have been keeping their heads in the sand for too long. For years now, it has been completely obvious and self-evident to anyone with half a brain that all kinds of VoIP software should be included in systemd. It's a basic building block of the whole OS, having absorbed functionality previously provided by all kinds of packages like util-linux and wireless-tools." John Knight of Ringfree accepted the criticism readily, but advocated a forward-thinking orientation focused on breaking with the uncertainty of the past: "In the absence of a SIP component for routing calls to the PSTN, some people thought, 'systemd has no clear direction apart from the whims of its developers, and is a perpetually moving goal post.' Well, a SIP server should put an end to that whole discussion; that's exactly what was missing, and now that we have systemd-rtc-server, we've eliminated all doubts about the coherence, conceptual integrity and finality of systemd." -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

5 8

Interpretation of Contact header
by Joachim Büchse 30 Jul '15

30 Jul '15

Good day, I’m experiencing some problems with our VoiP providers handling of REGISTER requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 with SIP-Alg enabled. This setup kind of works with UDP but our provider wants us to use TCP. With TCP enforced incoming calls don’t work. I’ve done some wire tracing and to me it seems that the providers configuration is to blame, but then - there are many RFCs out there and many NAT and UAC bug workarounds. Anyway, I wanted to get the opinion of “the" experts about how the requests send to the UAS SHOULD be correctly interpreted. The REGISTER requests/responses look like this (outside of the firewall): Protocol TCP! client port 19091 <-> server port 5060 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Authorization: Digest username="90780408050", realm="pbx.peoplefone.ch", uri="sip:pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy", response="764f371a08d258157a249f8d1b852514" Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.6bda Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp>;q=0;expires=180;received="sip:212.126.160.92:19091;transport=TCP" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 The ip:port the firewall is sending those requests from is ip 212.126.160.92 port 19091. So this does NOT match the port from the Contact header. For TCP this seems rather logical to me, as one cant be listening on a TCP port and use it for sending at the same time. The UAC closes this “register connection” with TCP FIN after the register, and so does the firewall. However unfortunately subsequent requests from the provider (ie UAS) come in on port 19091 (not port 6717 from the Contact header) and the firewall simply drops them. Observations: - the server does NOT include received=212.126.160.92 in the Via of the reponse. According to RFC3581 this is mandatory when rport is present in the request, so this is probably an error in the server. - the server does include received="sip:212.126.160.92:19091;transport=TCP” in the Contact of the response. I didnt see this in any RFC (which means nothing;-) but it could be an error. - after the client received the 200 OK it closes the TCP connection. - the server tries several times to re-contact the client (incoming TCP SYN). However not on port 6717 (defined in the Contact header) but on port 19091 (where the REGISTER came from). RFC3581 defines special behaviour when “rport” is defined in the request (i.e. response should go to the same port the request came from) - however it’s not so clear if this should apply to subsequent (INVITE/OPTIONS) requests from the server to the client. Those are strictly spoken not replies (or are they?). RFC5626 defines that a “proxy” should keep track of the flows over which it received a registration and send requests over the same flow. It is not clear if RFC5626 should be applied. The RFC5626 defines that a UAC includes an “ob” parameter in the Contact field if it whishes further requests over the same flow. Also the RFC mandates a client to add a "reg-id=x" in the Contact field. Both are not the case here, so in short I think RFC5626 should NOT be applied. In which case conecting to the originating port (instead of the Contact port) would be a server error. So in short and if I interpret the RFCs correctly, the client is reachable and should be contacted on Transport: TCP IP: 212.126.160.92 Port: 6717 If anyone who lives and breathes SIP could enlighten me if the UAS is right to call back on 19091 instead of 6717 I would really appreciate it;-)) Best regards, Joachim

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New SIP INVITE from UAS (new dialogue)
by Joao Alves 27 Jul '15

27 Jul '15

Hi, What would be the best way to generate and send an INVITE message generated by the Kamailio server. This would be for a new dialog session, UAS initiated (e.g. to make a click 2 call like service). I've tried the $uac_req(method)="INVITE"; uac_req_send(); but didn't work. I gather something would be needed first to open the dialogue before doing it but couldn't find it on the dialogue module. Thanks, Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

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Port 5061 Issue
by Ankhit Vivekananda 23 Jul '15

23 Jul '15

Hello, I have configured kamailio 4.0.4 long back. Now, I wanted to enable the usage of port 5061 with TLS. By doing so, the client(video camera) can place outgoing calls fine. But while receiving an incoming call, the message does not get sent out to the other client. Hence, the call fails. I can place and receive calls on port 5060 successfully without any issues. Can anyone please tell me why port 5061 is giving me a problem. Also, as a test, I set up kamailio on another machine(version 4.1) and 5061 works fine in all directions. It's only on the other machine that I have problems with. Appreciate any help! Thank you, Ankhit

2 1

usage of t_flush_flags() in event_route
by Andrew Pogrebennyk 20 Jul '15

20 Jul '15

Hi Daniel and others, I'm having a problem with acc module if I'm using the event_route/ branch-failure: say, the call comes from the app server and goes to the registered user. We arm the the failure route and per-branch failure route for the 302 redirect from the UA. We explicitly reset the accounting flags because we don't want to account the calls from the app server. The transaction is created implicitly by the t_relay(). Now if the UA responds with the 302 response and we are going to process that, we want to create an acc record for the new target from 302 message because this call may incur additional costs. I'm setting the accounting flags and calling t_flush_flags() but that doesn't work (no accounting record). Any idea if I'm doing something wrong or maybe there's a bug when changing the flags and then calling t_flush_flags from the event_route? Here are the module parameters: modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "db_url", PAIR_URL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)") modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "cdr_log_enable", 0) FTR, we tried 4.1.6 and 4.1.8. And here is the event route (with flags defined like this: flags FLAG_ACC_DB:1, FLAG_ACC_MISSED:2, FLAG_ACC_FAILED:3, ...): event_route[tm:branch-failure:redirect] { route(ROUTE_STOP_RTPPROXY_BRANCH); if($T_rpl($rs) == 301 || $T_rpl($rs) == 302) { # initialise variables when entering failure route route(ROUTE_INITVARS); # these need to be avps because we need it in reply/failure-route $(avp(s:from_faxserver)[*]) = 0; $(avp(s:to_faxserver)[*]) = 0; $(avp(s:cf_from_pstn)[*]) = 0; $(avp(s:from_pstn)[*]) = 0; $(avp(s:proxylu_from_pstn)[*]) = 0; $(avp(s:lcr_flags)[*]) = 0; $(avp(s:em_call)[*]) = 0; $(avp(s:from_pbx)[*]) = 0; $(avp(s:p_to_device)[*]) = 0; $(avp(s:p_to_group)[*]) = 0; $(avp(s:is_primary)[*]) = 0; # now let's process a 30x $(avp(s:acc_state)[*]) = "cfb"; $(avp(s:orig_acc_caller_user)[*]) = $avp(s:acc_caller_user); $(avp(s:orig_acc_caller_domain)[*]) = $avp(s:acc_caller_domain); $(avp(s:acc_caller_user)[*]) = $avp(s:acc_callee_user); $(avp(s:acc_caller_domain)[*]) = $avp(s:acc_callee_domain); $(avp(s:caller_uuid)[*]) = $avp(s:callee_uuid); $(avp(s:callee_uuid)[*]) = $null; # the $var(no_acc) is 0 at this point but the flags may have been reset # if this is a call from PBX user - we do want accounting for the 302 redirect if(isflagset(FLAG_ACC_DB)) { xlog("L_NOTICE", "++++++ ACC flag is set - [% logreq -%]\n"); } else { xlog("L_NOTICE", "------ ACC flag is NOT set - [% logreq -%]\n"); } setflag(FLAG_ACC_FAILED); setflag(FLAG_ACC_DB); t_flush_flags(); # get last URI from destination-set and set it as R-URI $var(contact) = $T_rpl($ct); $var(contact) = $(var(contact){nameaddr.uri}); if($var(contact) == 0 || $var(contact) == $null) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - [% logreq -%]\n"); acc_db_request("480", "acc"); $var(announce_handle) = "callee_tmp_unavailable"; $var(announce_set) = $xavp(callee_real_prefs[0]=>sound_set); $(avp(s:announce_code)[*]) = 480; $(avp(s:announce_reason)[*]) = "Temporarily Unavailable"; route(ROUTE_EARLY_REJECT); } $ru = $var(contact); xlog("L_NOTICE", "Redirect from UAC intercepted - [% logreq -%]\n"); $(avp(s:forwarder_cli_userprov)[*]) = $T_rpl($tU); $(avp(s:forwarder_domain_userprov)[*]) = $T_rpl($td); $var(forward) = 1; $var(redirected_forward) = 1; route(ROUTE_LOAD_CALLER_PREF); route(ROUTE_FIND_CALLEE); } } Thanks. Andrew

2 5

[Kamailio-Users] HA+DRBD setup
by Siddhardha Garige 09 Jul '15

09 Jul '15

Hello all, Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio. Especially creating partitions, DRBD devices and mount points. Thanks in advance, -Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

11 24

Function sdp_remove_codecs_by_id seems to be not working
by José Seabra 09 Jul '15

09 Jul '15

Hello, I'm using the function sdp_remove_codecs_by_id from sdpops module in order to remove some codecs in INVITE request before send it to freeswitch, but the function doesn't remove the codec, and it doesn't give any error message. I'm using this function in request route. Kamailio version is 4.2.2. INVITE that kamailio receives from phone: INVITE sip:401@teste.d <sip%3A401(a)teste.itcenter.com.pt>emo.pt;user=phone SIP/2.0 Record-Route: <sip:10.0.20.102:5062 ;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> Via: SIP/2.0/UDP 10.0.20.102:5062 ;branch=z9hG4bKecf3.3ff3f7e77d2abc0fd3f74c61eeb68a0b.0 Via: SIP/2.0/UDP 192.168.10.147:5060 ;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060 From: "301" <sip:301@teste.demo.pt <sip%3A301(a)teste.itcenter.com.pt> >;tag=oztyflbzbx To: <sip:401@teste.demo.pt <sip%3A401(a)teste.itcenter.com.pt>;user=phone> Call-ID: 3c3a58a25d63-ghfc5xdg1sn0 CSeq: 1 INVITE Max-Forwards: 69 Contact: <sip:301@192.168.10.147:5060 ;alias=100.64.250.254~5060~1;line=c1r2c8u6>;reg-id=1 X-Serialnumber: 000413262FA0 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/8.4.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:teste.demo.pt <http://teste.itcenter.com.pt> >;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 391 v=0 o=root 24935823 24935823 IN IP4 192.168.10.147 s=call c=IN IP4 192.168.10.147 t=0 0 m=audio 19410 RTP/AVP 0 8 9 99 3 18 4 101 a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv INVITE that kamailio send to freeswitch after execute sdp_remove_codecs_by_id("18"): INVITE sip:401@teste.demo.pt;user=phone SIP/2.0. Record-Route: <sip:10.0.20.100;lr=on;ftag=zvjgcz9zs9;proxy=yes;did=441.0eb2>. Record-Route: <sip:10.0.20.102:5062 ;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. Via: SIP/2.0/UDP 10.0.20.100;branch=z9hG4bK8711.bb31396197409170b2c1bd05b24e7f36.0. Via: SIP/2.0/UDP 10.0.20.102:5062 ;branch=z9hG4bK8711.07ffcc13fb96f90f6b4dbe4b2dfd0fa5.0. Via: SIP/2.0/UDP 192.168.10.147:5060 ;received=100.64.250.254;branch=z9hG4bK-aq7e0puz8p6o;rport=5060. From: "301" <sip:301@teste.demo.pt>;tag=zvjgcz9zs9. To: <sip:401@teste.demo.pt;user=phone>. Call-ID: 3c3a7c84e065-pr2hm0uk9yfz. CSeq: 2 INVITE. Max-Forwards: 68. Contact: <sip:301@192.168.10.147:5060 ;alias=100.64.250.254~5060~1;line=ttnfv9c7>;reg-id=1. X-Serialnumber: 000413262FA0. P-Key-Flags: resolution="31x13", keys="4". User-Agent: snom370/8.4.35. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Call-Info: <sip:teste.itcenter.com.pt>;appearance-index=1. Session-Expires: 3600;refresher=uas. Min-SE: 90. Content-Type: application/sdp. Content-Length: 403. . v=0. o=root 228603317 228603317 IN IP4 100.64.250.4. s=call. c=IN IP4 100.64.250.4. t=0 0. m=audio 49404 RTP/AVP 0 8 9 99 3 18 4 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:9 G722/8000. a=rtpmap:99 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:49405. SDP body has no changes related with codecs. Anyone call help please. Thank you BR José Seabra -- Cumprimentos José Seabra

3 6

Potential ip_addr_match_net bug
by Kyle Kurz 07 Jul '15

07 Jul '15

I'm using allow_address function of the permissions module and I'm seeing strange behavior with network masks not divisible by 8. I have 172.16.0.0/12 in my address table and Kamailio is marking packets from 172.56.16.232 as being part of that local subnet. If I change the subnet to 172.16.0.0/16, I no longer have an issue. I know there was a bug several years ago related to this ( http://git.kamailio.org/gitlist/index.php/sip-router/commit/0ead7ab31bf0df6…), but that code is very different, so I'm wondering if a regression has crept in? I'm using Kam 4.1.6. Thanks, Kyle -- *Kyle Kurz* Digium, Inc. · Senior Software Engineer 9115 Brown Deer Rd · San Diego, CA 92121 · US direct +1 858.537.1166 Check us out at: http://digium.com · http://asterisk.org <http://www.asterisk.org/>

3 3

(no subject)
by Nelson Migliaro 01 Jul '15

01 Jul '15

Hello everybody, My SIP vendor request me to replace FROM before sending the traffic. In order to achieve this I use uac_replace_from. UAC module is setup in restore_mode = auto. In my insfrastructure I have an Asterisk and then a Kamailio that connects to vendor via internet. Softphone -> Asterisk -> Kamailio -> Internet -> SIP vendor If caller ID is setup in Asterisk using CALLERID(num)=34888888888 and then INVITE is forwarded to Kamailio, the call is established and finished correctly but the URI in TO field in BYE request from Kamailio to Asterisk contains garbage. In the scenario the callee hangs up the call. Example of TO Field with garbage: 34888888888 <sip:50026896@no{soy,ns^^> What I do see is that the number "50026896" that is part of the URI is the same I use in: uac_replace_from("50026896", "sip:50026896@sip.vendor.es"); Something else that I have found is that vsf field is the same in the INVITE and in the BYE. ---------------------------------------------------------------------------------------------------------------- 2015/06/23 17:48:38.552442 192.168.0.2:5060 -> 192.168.0.1:5060 BYE sip:34888888888@192.168.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bKfb49.73c6517609cdb6f7ec00b1f40a05dbe9.0 Via: SIP/2.0/UDP 8.8.8.8.8;rport=5060;branch=z9hG4bKfcdf.3767c59c2d0e3d8ab695669845ce4cea.0 branch=z9hG4bK04boo6104o5hcso0c2a1sd0000g00.1 Call-ID: 4bf8effb45b0ae8e049366297924cbba@192.168.0.1:5060 From: <sip:28999999999@192.168.0.2;tag=k0eci3x3-CC-30 To: 34888888888 <sip:50026896@no{soy,ns^^>;tag=as041b7d84 CSeq: 1 BYE Reason: Q.850;cause=16;text="normal call clearing" Max-Forwards: 67 Content-Length: 0 ------------------------------------------------------------------------------------------------------------------ DEBUG: uac [replace.c:525]: restore_uri(): getting 'vsf' Route param DEBUG: uac [replace.c:533]: restore_uri(): route param is 'AAAAAAEECQkCAgsNAXBeL0NGQUsfVl02Ni44Mw--' (len=40) DEBUG: uac [replace.c:607]: restore_uri(): decoded uris are: new=[sip:50026896@no{soy,ns#005#007] old=[sip:34888888888@8.8.8.8] DEBUG: uac [replace.c:525]: restore_uri(): getting 'vst' Route param DEBUG: uac [replace.c:533]: restore_uri(): route param is 'AAAAAAQPAw8MDgsAAHZBKRVdAhoVHQ4XH1BdYWJhbnRlLmVz' (len=48) DEBUG: uac [replace.c:607]: restore_uri(): decoded uris are: new=[ sip:28999999999@192.168.0.1] old=[sip:999999999@sip.vendor.es] -----------------------------------------------------------------------------------------------------------------------------------

2 1

Kamailio no recieved 480/486 messages
by Yuriy Gorlichenko 01 Jul '15

01 Jul '15

At my confiig I have route like this onreply_route[REPLY_FROM_WS] { if(status=~"[12][0-9][0-9]") { xlog("L_INFO", "Manage_Reply from webrtc client {$si:$sp} for method {$rm}: $rs"); rtpengine_manage("force trust-address replace-origin replace-session-connection DTLS=passive ICE=remove RTP/AVP"); route(NATMANAGE); } } This route successfully handling replies 180 183 and 200 and then (after natmanage route) resend to mediaserver. When callee client reject call it send 480 ot 486 message that not resended by kamailio nowhere. First of all I thought that 480 and 486 replies are failure replies but at log I see its at onreply route. So Ok . I added this to onreply_route if (status=~"48[06]"){ xlog("L_INFO", "Manage_Reply from webrtc client {$si:$sp} for method {$rm}: $rs"); rtpengine_manage("force trust-address replace-origin replace-session-connection DTLS=passive ICE=remove RTP/AVP"); route(NATMANAGE); } But message just going through natmanage route and goes nowhere My NATMANAGE route is xlog("L_INFO","NATMANAGE reply {$rm}"); if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } xlog("L_INFO","reply {$rm}"); } I see at my log only first NATMANAGE xlog... no more... thnks for answer

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sr-users June 2015