- sr-users - kamailio.org

xHTTP_PROM with older Kamailio
by Marat Gareev 11 Jul '20

11 Jul '20

Hello, I have a customized Kamailio (kamailio 5.1.1 (x86_64/linux)), xHTTP_PROM <https://www.kamailio.org/docs/modules/devel/modules/xhttp_prom.html> was released in the 5.3.0 <https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/> version. I'm trying to import the latest version of xhttp_prom to older Kamailio, but something goes wrong. This is simple config for test: log_stderror=yes > listen=5060 > debug=2 > mpath="/home/devel/build_dir/build/lib64/kamailio/modules" > loadmodule "tm.so" > loadmodule "sl.so" > loadmodule "xlog.so" > loadmodule "pv.so" > disable_tcp=no > tcp_accept_no_cl=yes > loadmodule "xhttp.so" > loadmodule "xhttp_prom.so" > modparam("xhttp_prom", "xhttp_prom_stats", "all") > route { > sl_send_reply("200", "OK"); > exit; > } > event_route[xhttp:request] { > if (prom_check_uri()) { > prom_dispatch(); > } else { > xhttp_reply("404", "Not Found", "text/plain", "Wrong URL $hu\n"); > } > } then I send HTTP request $ curl http://localhost:5060/metrics but get error: > 6(110968) CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 11 > 0(110962) ALERT: <core> [main.c:746]: handle_sigs(): child process 110967 > exited by a signal 11 > 0(110962) ALERT: <core> [main.c:749]: handle_sigs(): core was generated > 0(110962) INFO: <core> [main.c:771]: handle_sigs(): terminating due to > SIGCHLD note: for http://localhost:5060/test request I get the expected message "Wrong URL /test". Backtrace for killed process > (gdb) c > Continuing. > Program received signal SIGSEGV, Segmentation fault. > 0x00007fefff3ff096 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at > ../../core/mem/../atomic/atomic_x86.h:224 > 224 ATOMIC_FUNC_CMPXCHG(cmpxchg, "cmpxchgl %2, %1", int , int) > (gdb) bt > #0 0x00007fefff3ff096 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at > ../../core/mem/../atomic/atomic_x86.h:224 > #1 0x00007fefff3ff0e2 in futex_get (lock=0x0) at > ../../core/mem/../futexlock.h:99 > #2 0x00007fefff4130bf in prom_metric_list_print (ctx=0x7fefff628da0 > <ctx>) at prom_metric.c:1319 > #3 0x00007fefff416468 in prom_stats_get (ctx=0x7fefff628da0 <ctx>, > stat=0x7fefff6281f0 <xhttp_prom_stats>) at prom.c:178 > #4 0x00007fefff3d4d10 in prom_send (ctx=0x7fefff628da0 <ctx>) at > xhttp_prom.c:361 > #5 0x00007fefff3d5faf in ki_xhttp_prom_dispatch (msg=0x7ffecb730ee0) at > xhttp_prom.c:418 > #6 0x00007fefff3d5feb in w_prom_dispatch (msg=0x7ffecb730ee0) at > xhttp_prom.c:428 > #7 0x0000000000461dbe in do_action (h=0x7ffecb730e00, a=0x7ff0007c7b78, > msg=0x7ffecb730ee0) at core/action.c:1067 > #8 0x000000000046e571 in run_actions (h=0x7ffecb730e00, a=0x7ff0007c7b78, > msg=0x7ffecb730ee0) at core/action.c:1565 > #9 0x0000000000461d2d in do_action (h=0x7ffecb730e00, a=0x7ff0007ca628, > msg=0x7ffecb730ee0) at core/action.c:1058 > #10 0x000000000046e571 in run_actions (h=0x7ffecb730e00, a=0x7ff0007c76f8, > msg=0x7ffecb730ee0) at core/action.c:1565 > #11 0x00007fefff62f649 in xhttp_process_request (orig_msg=0x7ff0007ebbd8, > new_buf=0x7ff0007cc280 "GET /metrics HTTP/1.1\r\nVia: SIP/2.0/TCP > 127.0.0.1:37000\r\nUser-Agent: curl/7.29.0\r\nHost: > localhost:5060\r\nAccept: */*\r\n\r\n", new_len=119) at xhttp_mod.c:298 > #12 0x00007fefff630d78 in xhttp_handler (msg=0x7ff0007ebbd8) at > xhttp_mod.c:385 > #13 0x0000000000520141 in nonsip_msg_run_hooks (msg=0x7ff0007ebbd8) at > core/nonsip_hooks.c:111 > #14 0x000000000058cef1 in receive_msg (buf=0x7feffee65bb0 "GET /metrics > HTTP/1.1\r\nUser-Agent: curl/7.29.0\r\nHost: localhost:5060\r\nAccept: > */*\r\n\r\n", len=85, rcv_info=0x7feffee658d0) > at core/receive.c:216 > #15 0x000000000061233f in receive_tcp_msg (tcpbuf=0x7feffee65bb0 "GET > /metrics HTTP/1.1\r\nUser-Agent: curl/7.29.0\r\nHost: > localhost:5060\r\nAccept: */*\r\n\r\n", len=85, rcv_info=0x7feffee658d0, > con=0x7feffee658b8) at core/tcp_read.c:1379 > #16 0x000000000061479e in tcp_read_req (con=0x7feffee658b8, > bytes_read=0x7ffecb731c7c, read_flags=0x7ffecb731c78) at > core/tcp_read.c:1611 > #17 0x00000000006174e9 in handle_io (fm=0x7ff0007d9720, events=1, idx=-1) > at core/tcp_read.c:1785 > #18 0x00000000006069b4 in io_wait_loop_epoll (h=0xaa76c0 <io_w>, t=2, > repeat=0) at core/io_wait.h:1065 > #19 0x00000000006194ff in tcp_receive_loop (unix_sock=14) at > core/tcp_read.c:1955 > #20 0x00000000004f1a43 in tcp_init_children () at core/tcp_main.c:4853 > #21 0x0000000000423f99 in main_loop () at main.c:1716 > #22 0x000000000042a5ee in main (argc=9, argv=0x7ffecb7323b8) at main.c:2646 > (gdb) > (gdb) p 'prom_metric.c'::prom_lock > $4 = (gen_lock_t *) 0x0 If I understood correctly, the problem is in an uninitialized variable static gen_lock_t *prom_lock = NULL; /* Lock to protect Prometheus metrics. */ So, is it possible to import xHTTP_PROM to older Kamailio? I will be glad for any advice. Thanks.

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loose_route failing with topos ?
by Mark Boyce 10 Jul '20

10 Jul '20

Hi all Just enabled topos with redis backend (topos/topos_redis/ndb_redis) The interesting bits of the calls are going from kamailio (5.3.5) to Asterisk with topos enabled for that leg All seems to be ok until the far end (Asterisk) sends a BYE. At which point kamailo passes message to WITHINDLG, in loose_route comes up false (-1) and the BYE gets missed, falling through to the “404 Not here” at the end of WITHINDLG. I’m guessing that loose_route should be able to “see inside” topos, or should I be throwing loose_route away when using topos? Happy to upgrade to 5.4 and do diags if it appears to be a bug. Best regards Mark -- Mark Boyce Dark Origins Ltd

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rtcp-mux is missing in the OK for WebRTC-Calls
by Benjamin Flügel | vio:networks 10 Jul '20

10 Jul '20

Hey guys, I'm trying to configure a Kamailio to work with a browser softphone based on SIPJS using WebRTC. So far it works great on Firefox but have a specific problem with chrome, when I want to make call from the softphone to another extension. After anwsering the call Chrome/the softphone sends a BYE immediately, because this line "a=rtcp-mux" is missing in the OK. The Kamailio is a proxy. Behind the Kamailio there is an Asterisk, which is responsible for the pbx-features. Those are my rtpengine Flags for the Invite: rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-demux DTLS=off SDES-on ICE=remove RTP/AVP And those are the flags for the response, in this case the OK: rtpengine_manage: replace-origin replace-session-connection rtcp-mux-offer rtcp-mux-accept generate-mid DTLS=off SDES-on ICE=force RTP/SAVPF direction=internal direction=external loop-protect It seems that the Kamailio ignores ths "rtcp-mux-offer rtcp-mux-accept" in the response. Can you help me get it to work? Here is the SIP-Dialog. The call is from extension 201 to the extension 2. INVITE: INVITE sip:2@mydomain SIP/2.0 Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;branch=z9hG4bK5104892 To: <sip:2@mydomain.com> From: <sip:201@mydomain.com>;tag=aqplr71k05 CSeq: 2 INVITE Call-ID: sde09v49f8gqtt59oddn Max-Forwards: 70 Proxy-Authorization: Digest algorithm=MD5, username="201", realm="mydomain.com", nonce="sfdsdfdsfsdfdsfsdfaseqww", uri="sip:2@mydomain.com", response="81rewtrega23423r" Contact: <sip:kf6iduon@g4n0lpfirgpv.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIPJS Content-Type: application/sdp Content-Length: 1916 v=0 o=- 5826889093965459811 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO m=audio 54274 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 mycomputerIP a=rtcp:9 IN IP4 0.0.0.0 a=candidate:1577908739 1 udp 2113937151 efd6d297-d186-4c07-87d8-933e5846a82d.local 54274 typ host generation 0 network-cost 999 a=candidate:842163049 1 udp 1677729535 mycomputerIP 54274 typ srflx raddr 0.0.0.0 rport 0 generation 0 network-cost 999 a=ice-ufrag:Ktxl a=ice-pwd:x5xDUb41GeOXAcHNQlra4yUN a=ice-options:trickle a=fingerprint:sha-256 2C:7C:32:AF:34:A3:9D:AE:C7:FD:92:68:DD:D8:AB:82:DB:F0:32:51:14:97:20:60:66:5C:2F:CF:B7:98:B8:A8 a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:2376769177 cname:RgPkYyxDrEJTpmO2 a=ssrc:2376769177 msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94 a=ssrc:2376769177 mslabel:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO a=ssrc:2376769177 label:dc187be3-4cd0-4129-98e4-f804cd8d2c94 RINGING: SIP/2.0 180 Ringing Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892 Record-Route: <sip:internalIP;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> Record-Route: <sip:externalIP;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> From: <sip:201@mydomain.com>;tag=aqplr71k05 To: <sip:2@mydomain.com>;tag=as64fe0fc8 Call-ID: sde09v49f8gqtt59oddn CSeq: 2 INVITE Server: myserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <mycontact> P-Asserted-Identity: "Phone 2" <sip:2@mydomain.com> Content-Length: 0 OK: SIP/2.0 200 OK Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892 Record-Route: <sip:internal-ip;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> Record-Route: <sip:external-ip;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> From: <sip:201@mydomain.com>;tag=aqplr71k05 To: <sip:2@mydomain.com>;tag=as64fe0fc8 Call-ID: sde09v49f8gqtt59oddn CSeq: 2 INVITE Server: myserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <mycontact> P-Asserted-Identity: "Phone 2" <sip:2@mydomain.com> Content-Type: application/sdp Content-Length: 806 v=0 o=root 882840402 882840402 IN IP4 externalIP s=Asterisk PBX 11.22.0 c=IN IP4 externalIP t=0 0 a=rtpengine:4d633e3022f5 m=audio 16416 RTP/SAVPF 111 9 8 0 126 a=maxptime:60 a=mid:0 a=rtpmap:111 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:111 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0 a=fmtp:126 0-16 a=sendrecv a=rtcp:16417 a=setup:active a=fingerprint:sha-1 B1:67:4B:B8:47:89:E8:49:CD:DD:F8:FF:41:5C:83:72:D9:DE:4D:45 a=ptime:20 a=ice-ufrag:WBZN8m7c a=ice-pwd:mKVR6Jv0G0pvrTv7OfEm2wPW9Q a=ice-options:trickle a=candidate:3kddB3zBUV2n2jt0 1 UDP 2130706431 externalIP 16416 typ host a=candidate:3kddB3zBUV2n2jt0 2 UDP 2130706430 externalIP 16417 typ host a=end-of-candidates I hope, you can help me with this. If you have further question, I will try to answer them as best I can. Greetings, Ben

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Development is open for future release of v5.5.x series
by Daniel-Constantin Mierla 10 Jul '20

10 Jul '20

Hello, the branch 5.4 was created, therefore the master branch is open for adding new features, to be part of future release series v5.5.x. Any bug fix committed to master that applies to 5.4.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.4 or 5.3. Expect that v5.4.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.5.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime in the first part of 2021. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Branch 5.4 has been created
by Daniel-Constantin Mierla 10 Jul '20

10 Jul '20

Hello, the branch 5.4 has been created, to be used for releasing v5.4.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.4 cd kamailio-5.34 git checkout -b 5.4 origin/5.4 Pushing commits in this branch: git push origin 5.4:5.4 Note that 5.4 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.4 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.4 will be out, respectively Kamailio v5.4.0. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Drop local generated requests
by Patrick Wakano 10 Jul '20

10 Jul '20

Hello list, Hope you all doing fine! I have a case where I want to drop a local generated request but it looks like you can't drop the request from the tm:local-request route. I found this ticket https://github.com/kamailio/kamailio/issues/980, but I could not find somewhere in the docs or emails definitely saying that drop() is not supported from tm:local-request.... And in case we definitely can't drop the request in there, any ideas on how to avoid a local generated request of going out? I am specifically interested in dropping Registers from the uacreg module under certain situations, but I don't want to disable it or delete it from the DB.... Thank you, Kind regards, Patrick Wakano

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Help on dumping running config in Kamailio
by BALL SUN 09 Jul '20

09 Jul '20

Hi I need an urgent help, I accidentally pipe the output to the kamailio config, and now I lost the copy of my latest setting, is there a chance that I can dump the running config back to file? Thanks PLEASE HELP RBK

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Branching 5.4 in git repository
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, so far it looks pretty calm during the testing phase, some static analysis didn't reveal any new major issues (well, this is done also during development phase). Therefore, if nothing else is proposed or shows up, I am considering to create the git branch 5.4 next week on Thursday -- this branch will hold the code for 5.4.x release series. After branching 5.4, the master branch will be open again for new features. The 5.4.0 enters the release candidate phase, likely to have the first release out in 1-2 weeks after branching. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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Maintenance work on kamailio.org server - Wed, July 8, 2020
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, on Wednesday, July 8, 2020, there will be some work on Kamailio project infrastructure. Therefore there could be short intervals of downtime for online resources like the web server, wiki, documentation and downloads portals, mailing lists or the GIT repository mirror. Once the maintenance work is finished, we will post an update. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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30X redirect in interconnects - better alternative?
by Gerry | Rigatta.com 08 Jul '20

08 Jul '20

Hi, I would like to use Kamailio for load balancing incoming carrier traffic. We do currently IP authentication and call logic in Yate boxes. Ideally I would like to distribute calls with 30X redirects with the Kamailio dispatcher so that IP authentication and all logic can stay in the Yate boxes. However I have doubts that 30X redirects are generally accepted in interconnects. What is your experience with this? What is the possible alternative to redirects if one wants to keep IP authentication and call logic in the boxes behind the Kamailio SIP router? E.g. how can one reliably check the carrier source IPs behind Kamailio? Custom headers injected by Kamailio? Of cause I can check source IPs with a database lookup in Kamailio but I try to avoid that as this makes the setup much more complicated and error prone. Thank you for your ideas. Gerry

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Re: [SR-Users] sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Update: Disabling the topoh module on the proxy which produces the error seems to stop the failure from manifesting. I'll try using topos_redis instead, but should this be treated as a bug? BR, George On Wed, 8 Jul 2020 at 12:37, George Diamantopoulos <georgediam(a)gmail.com> wrote: > Hello again, > > Indeed $mb seems to contain garbage: > > SCRIPT_MB: ACK <BC><EA><8F> SIP/2.0#015#012Via: SIP/2.0/UDP > 172.30.154.189;TH=dcv;branch=z9hG4bK629b.6af9302cd78dc58dffe817e60124f4ed.0#015#012Route: > <sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@172.30.155.1 > ;lr;received=sip:2.2.2.2:32768 > ;ob;r2=on>,<sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@3.3.3.3 > ;lr;received=sip:2.2.2.2:32768;ob;r2=on>#015#012Max-Forwards: > 68#015#012From: "Anonymous" <sip:unknown@voip.domain.com>;tag=as4bc9e324#015#012To: > <sip:voip-test-user-02@voip.domain.com>;tag=jw7z5s0zvc#015#012Call-ID: > 0138c6370346d1dd7f1b47f604b01f92(a)voip.domain.com#015#012CSeq > <http://0138c6370346d1dd7f1b47f604b01f92@voip.domain.com#015%23012CSeq>: > 102 ACK#015#012Content-Length: 0#015#012TH: dch#015#012#015#012 > > How can this be possible? Capturing traffic on wire shows the RURI I > pasted in my original message and there are no script operations on the > RURI before sanity_check() (message buffer above is printed just before > sanity_check() is run in REQINIT). > > BR, > George > > On Wed, 8 Jul 2020 at 11:18, George Diamantopoulos <georgediam(a)gmail.com> > wrote: > >> I'll post the message buffer ASAP, but in the meantime I don't see how >> config operations could affect the RURI. Here's everything that's happening >> until the sanity check involved: >> >> request_route { >> >> if(is_method("KDMQ")) { >> dmq_handle_message(); >> } >> >> # no connect for sending replies >> set_reply_no_connect(); >> >> if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { >> # silent drop for scanners - uncomment next line if want to reply >> # sl_send_reply("200", "OK"); >> exit; >> } >> >> if (!mf_process_maxfwd_header("10")) { >> force_rport(); >> sl_send_reply("483","Too Many Hops"); >> exit; >> } >> >> # OPTIONS and NOTIFYs directed to myself >> if(is_method("OPTIONS|NOTIFY") && uri==myself && $rU==$null) { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> # All keep-alive methods regardless of destination >> if ( $hdr(Event) == "keep-alive") { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> if(!sanity_check("17895", "7")) { >> xlog("Malformed SIP request from $si:$sp\n"); >> exit; >> } >> >> BR, >> George >> >> On Wed, 8 Jul 2020 at 10:58, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> check your config operations, because the R-URI seems to be the next >>> string (without quotes): "<BC><EA><8F>" >>> >>> You can try to print $mb in such case to see the entire SIP message >>> buffer. >>> >>> Cheers, >>> Daniel >>> On 08.07.20 09:48, George Diamantopoulos wrote: >>> >>> Hello Daniel, >>> >>> Thanks for the reply. Indeed there is, not sure how I managed to miss >>> that. And it wasn't about the schema after all: >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1254]: parse_uri(): uri too short: <<BC><EA><8F>> >>> (3) >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1328]: parse_sip_msg_uri(): bad uri <<BC><EA><8F>> >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: WARNING: >>> {1 <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:282]: check_ruri_scheme(): failed to parse request uri >>> [<BC><EA><8F>] >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: ERROR: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <script>: Malformed >>> SIP request from 172.30.154.189:5060 >>> >>> Still, not sure what the problem is though... >>> >>> BR, >>> George >>> >>> On Wed, 8 Jul 2020 at 09:30, Daniel-Constantin Mierla <miconda(a)gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> when the ruri scheme check fails, there should be another debug message >>>> saying that. Have you pasted all log messages for the failure case? >>>> >>>> Cheers, >>>> Daniel >>>> On 07.07.20 22:23, George Diamantopoulos wrote: >>>> >>>> Sorry, I realised I copy pasted wrong log messages for Call 1. Here's >>>> the relevant part showing success for call 1 in contrast with Call 2: >>>> >>>> grep 2a859fcc4e1c8f840191a81d7c16e76d kamailio.log | egrep >>>> 'check_ruri_scheme|w_sanity_check' | grep ACK >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 1 >>>> >>>> On Tue, 7 Jul 2020 at 21:34, George Diamantopoulos < >>>> georgediam(a)gmail.com> wrote: >>>> >>>>> Hello all, >>>>> >>>>> I'm not 100% sure this is the only culprit in an issue I'm >>>>> investigating, but superficially it appears that RURI scheme sanity module >>>>> checks from the default config (flags 17895 in REQINIT) fail if the >>>>> RURI in an ACK following a 487 includes parameters. Example from two calls >>>>> from a kamailio instance acting as registrar/usrloc server, INVITE RURIs >>>>> are after usrloc lookup: >>>>> >>>>> Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> These INVITEs produce no complaints. Later, the same registrar >>>>> produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses >>>>> from the next proxy in the path following a CANCEL: >>>>> >>>>> Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> The next proxy (which produced/relayed the 487) processes the ACK for >>>>> Call 1 successfully, but sanity_check at the proxy drops the request for >>>>> Call 2 with: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>>>> >>>>> whereas Call 1 seems OK: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:305]: check_required_headers(): check_required_headers entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:313]: check_required_headers(): check_required_headers passed >>>>> >>>>> Could this be a bug in sanity module? Is there anything one can do in >>>>> config which could result in illegal ACKs being produced for hop-by-hop >>>>> transactions? schema appears to be sip: in both cases... >>>>> >>>>> Thank you. Best regards, >>>>> George >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> -- >>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Funding: https://www.paypal.me/dcmierla >>>> >>>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Funding: https://www.paypal.me/dcmierla >>> >>>

2 2

Setting up Kamailio as Teams SBC
by Роман С. 08 Jul '20

08 Jul '20

Hello. I'm trying to set up Kamailio as SBC for Teams using https://skalatan.de/en/blog/kamailio-sbc-teams. Setup is completely default (except things mentioned at article), but I use wildcard certificate for TLS. Well, I can't even pass dispatcher: kamcmd dispatcher.list | egrep "RI|FLAG" URI: sip:sip.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 3 URI: sip:sip2.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 2 URI: sip:sip3.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 1 Where do I start to dig?

3 6

sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Hello all, I'm not 100% sure this is the only culprit in an issue I'm investigating, but superficially it appears that RURI scheme sanity module checks from the default config (flags 17895 in REQINIT) fail if the RURI in an ACK following a 487 includes parameters. Example from two calls from a kamailio instance acting as registrar/usrloc server, INVITE RURIs are after usrloc lookup: Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 SIP/2.0 These INVITEs produce no complaints. Later, the same registrar produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses from the next proxy in the path following a CANCEL: Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 SIP/2.0 The next proxy (which produced/relayed the 487) processes the ACK for Call 1 successfully, but sanity_check at the proxy drops the request for Call 2 with: DEBUG: {1 <null> 172.30.154.189 102 ACK 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered DEBUG: {1 <null> 172.30.154.189 102 ACK 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 whereas Call 1 seems OK: DEBUG: {1 <null> 172.30.154.189 102 ACK 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity [sanity.c:305]: check_required_headers(): check_required_headers entered DEBUG: {1 <null> 172.30.154.189 102 ACK 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity [sanity.c:313]: check_required_headers(): check_required_headers passed Could this be a bug in sanity module? Is there anything one can do in config which could result in illegal ACKs being produced for hop-by-hop transactions? schema appears to be sip: in both cases... Thank you. Best regards, George

2 5

Some Functions not defined in Documentation of Module ims_registrar_pcscf
by Hamid Hashmi 08 Jul '20

08 Jul '20

I have been using a sample code of PCSCF<https://github.com/kamailio/kamailio/blob/master/misc/examples/ims/pcscf/ro…>, where a function "pcscf_assert_identity" is used but it's not defined in module ims_registrar_pcscf documentation. Can you please explain its functionality. The same way other functions are not defined in the documentation. pcscf_assert_called_identity reginfo_handle_notify pcscf_unregister Regards Hamid R. Hashmi

1 0

Kamailio 5.3.4 - Segfault on http_async_client
by José Seabra 07 Jul '20

07 Jul '20

Hello there, Kamailio is terminating with signal 11 and is generating a coredump file. I opened the core file generated by it and it seems to be related to the module http_async_client. In the following link you can find the output of core dump backtrace full: https://pastebin.com/rapPZy9N Below is the http_async_client config that I have: modparam("http_async_client", "workers", 2) modparam("http_async_client", "connection_timeout", 500) modparam("http_async_client", "hash_size", 2048) modparam("http_async_client", "curl_verbose", 0) modparam("http_async_client", "memory_manager", "shm") If the information provided is not enough to troubleshoot this issue let me know. Many thanks Best Regards -- Cumprimentos José Seabra

1 1

Kamailio Buildroot
by Benjamin Marty 07 Jul '20

07 Jul '20

Hello I would like to ask if someone already tried to build Kamailio for the Embedded Distribution Buildroot. I tried it myself but it looks like Kamailio is not that easy to cross compile. Thanks Benjamin

3 3

siptrace not sending data to hepic
by Andrew Chen 07 Jul '20

07 Jul '20

Hi all, I've been assigned a task to investigate why hepic doesn't generate MOS scores ever since we changed the kamailio interface MTU from 9001 -> 1500. During my investigation, this is what I have found out so far: - Putting MTU back to 9001 removes this condition where no MOS was generated. - I did a tcpdump of port 9063 which is the port used to send data to Hepic capture server and saw it reports packets are fragmented: 3:48:55.895505 IP (tos 0x68, ttl 64, id 38049, offset 0, flags [+], proto UDP (17), length 1500) 206.81.181.40.5060 > 10.225.88.239.9063: SIP 13:48:55.896451 IP (tos 0x68, ttl 64, id 38050, offset 0, flags [+], proto UDP (17), length 1500) 206.81.181.40.5060 > 10.225.88.239.9063: SIP - Hepic support also caused by using 1500 MTU on the interface. - I tried using the following core parameters to see if they could remedy this issue but didn't seem to work: ## MTU related core settings udp4_raw=-1 udp4_raw_mtu=1500 I can't seem to find any parameters in the siptrace module that controls fragmentation of the packets to Hepic. Any suggestions on how else to troubleshoot this? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Keep one registration per device(src_ip+port)
by Zhan Bazarov 07 Jul '20

07 Jul '20

Hello. I'd like to keep one registration per device in usrloc table, for other cases I want to store and maintain one contact per AoR. Does kamailio support it out of the box? For second case save("location", "0x01"); working fine. But how about first case? for example, if one of device located at home, and second located at office. - there are two different network and NAT - how I can make both devices ringing when INVITE is sent? If I do classic multiAor scenario - it can lead to issues caused by multiple contacts in 200OK response for a single device and devices are confusing sometimes. So, what I did now: if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); add_contact_alias(); } [REGISTRAR] { if (nat_uac_test(64)) { if (!save("location", "0x04")) { sl_reply_error(); } } else { if (!save("location")) { sl_reply_error(); } } } modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "xavp_contact", "usrlocattrs") modparam("usrloc", "desc_time_order", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "close_expired_tcp", 1) modparam("usrloc", "db_insert_update", 1) modparam("usrloc", "expires_type", 1) modparam("usrloc", "matching_mode", 1) modparam("registrar", "method_filtering", 1) modparam("registrar", "max_expires", 3530) modparam("registrar", "min_expires", 1800) modparam("registrar", "received_avp", "$avp(s:rcv)") modparam("registrar", "gruu_enabled", 0) modparam("registrar", "use_path", 1) modparam("registrar", "path_check_local", 1) modparam("registrar", "path_mode", 1) modparam("path", "use_received", 1) How do u think, maybe modparam("registrar", "append_branches", 0) will make some help? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Nathelper ping from dedicated kamailio instance
by Angelo Pellegrinon 07 Jul '20

07 Jul '20

Hi! I am new to kamailio so please forgive me if I write some errors. I am configuring some kamailio instances in different environments to handle big traffic and I have a problem with usrloc + db + nathelper. What have I done: 1. Configured a kamailio instance to handle sip registers with usrloc to database with database mode 2 (memory + db), disabled nathelper pings. This is ok and performs well; 2. Configured a kamailio instance to handle nathelper ping. Same usrloc + db + nathelper config from 1 except that I use db mode 3 (only db) and that I enabled natping_interval . My problem is that it seems that the second kamailio instance does not send pings. The second one connects correctly to db (if I lookup from kamcmd a user registered from the first kamailio I can find it) but it seems that does not read all locations. Can you help me? Many many thanks Angelo

1 0

rtpengine with mutiple IPs on the same network.
by David Villasmil 07 Jul '20

07 Jul '20

Hello guys, I have rtpengine running on a box with ips like this: external: 172.10.0.10 (advertise 10.10.10.1) internal: 172.10.0.11 Both Ips are on the same network. When i do rtpengine_manage i set "direction=external direction=internal", and the SDP offer is correct. The problem I'm having is that when rtp tried to send to the internal network, the OS (i believe) selects the first interface (external) ip 172.10.0.10 to reach the internal network. This I suppose is not a great deal, but I would like to separate them and only use the internal for internal proxying. Help is appreciated! Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 5

Retrieving cert details from tls peer
by Mark Boyce 07 Jul '20

07 Jul '20

Hi all Been trying to grab the TLS cert details from incoming connections, but failing :-( So with lines just before AUTH is called like this; if (proto == TLS) { xlog("L_INFO", "TLSDUMP $ci peer_subject : $tls_peer_subject\n"); Gets met with a log line line this; INFO: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from 1.2.3.4:11797 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 INFO: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: 5.6.7.8:5061 INFO: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate ... INFO: tls [tls_select.c:168]: get_cert(): Unable to retrieve peer TLS certificate from SSL structure This is with verify_certificate and require_certificate set to no in tls.cfg If I try and set the following in tls.cfg [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = yes I see in the logs; INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSs<default>: tls_method=22 INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/tls-certs/cert.pem' INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=1 INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/tls-certs/privkey.pem' INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 NOTICE: tls [tls_domain.c:1095]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... INFO: tls [tls_domain.c:692]: set_verification(): TLSs<default>: Client MUST present valid certificate INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=1 INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=1 INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 INFO: tls [tls_domain.c:692]: set_verification(): TLSc<default>: Server MUST present valid certificate ... ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417C086:SSL routines:tls_process_client_certificate:certificate verify failed Which looks like verification is being enabled when I add require? Would someone be kind enough to point out what I am missing please? (Assuming it’s not a bug :-) Thanks Mark -- Mark Boyce Dark Origins Ltd

2 8

About routing calls
by ahmed moghazy 07 Jul '20

07 Jul '20

Excuse me i already made integration for kamailio with asterisk but i want to make sip calls through kamailio and media fuctions through asterisk , i want kamailio to make calls away from asterisk , i just wanted asterisk to use its dahdi module .’ Thank you.

2 1

Kamailio Push Notification for iOS13
by william anderson 07 Jul '20

07 Jul '20

Hello, Yesterday, there was a new release of Linphone IOS app, but i need help from some kamailio expert how to add PUSH notification to the application to receive incoming calls when the phone is not in use (wake up required) from a kamailio server. I have read on internet old tutorials (tsilo module) but most of them are incomplete or do not explain in an easy way how to do it (for an average kamailio user). Anyone could do an easy to understand tutorial ? Regards, William Anderson

2 1

Looking for a VoIP & Kamailio Dev-Ops
by Henning Westerholt 06 Jul '20

06 Jul '20

Dev-Ops VoIP & Kamailio GILAWA is looking for a new colleague to join our team! Tasks: Deploying, developing, and integrating our software based on Kamailio Testing and validation of our customer solutions Solve complex customer problems in the VoIP field Fix Bugs and create extensions in the Kamailio project Requirements: Good background in VoIP telephony and Kamailio Stack Experience with signalling standards (SIP, RTP etc..) and troubleshooting Knowledge and experience in one of the following languages: C/C++, Bash, PHP Experience with Debian or RedHat/CentOS Linux, knowledge of Ansible would be a plus Fluent in English Based in European or neighbouring time zone (e.g. EMEA area) Self-organized and strong communication skills Given the current situation this is a remote position. If you are interested, please send me a (direct) e-mail with an CV or Linkedin link, please include your availability and salary expectations as well. Best regards, Henning Westerholt -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

3 3

Kamailio 5.3.5, usrloc: Error inserting location (integer out of range)
by Stefan 03 Jul '20

03 Jul '20

Hi Maillist Kamailio-Version: 5.3.5 built from source with postgres-db. I get an Error if I try to add a static location with kamctl, see my example: kamctl ul add 1234 sip:1234@10.10.10.10:5060 NULL NULL Log output: Jul 1 09:28:20 kamailio-srv02 /sbin/kamailio[3892]: ERROR: db_postgres [km_dbase.c:266]: db_postgres_submit_query(): 0x7fbe5398c148 PQsendQuery Error: ERROR: integer out of range#012 Query: insert into location (username,contact,expires,q,callid,cseq,flags,cflags,user_agent,path,methods,last_modified,ruid,reg_id,server_id,connection_id,keepalive,partition ) values ('1234','sip:1234@10.10.10.10:5060','1970-01-01 00:00:00',1.000000 ,'dfjrewr12386fd6-343(a)kamailio.rpc',1,0,0,'kamailio SIP Router - RPC Server','NULL',2147483648,'1970-01-01 00:00:00','ulcx-5ef99724-f37-1',0,0,0,0,22) truncated output of kamctl ul show: "AoRs": [{ "Info": { "AoR": "1234", "HashID": 924651761, "Contacts": [{ "Contact": { "Address": "sip:1234@10.10.10.10:5060", "Expires": "permanent", "Q": 1, "Call-ID": "dfjrewr12386fd6-343(a)kamailio.rpc", "CSeq": 1, "User-Agent": "kamailio SIP Router - RPC Server", "Received": "[not set]", "Path": "NULL", "State": "CS_NEW", "Flags": 0, "CFlags": 0, "Socket": "[not set]", "Methods": -2147483648, "Ruid": "ulcx-5ef99724-f37-1", "Instance": "[not set]", "Reg-Id": 0, "Server-Id": 0, "Tcpconn-Id": 0, "Keepalive": 0, "Last-Keepalive": 0, "Last-Modified": 0 } }] } }, { So the log tells me: integer out of range. I think it's this one: "Methods": -2147483648, This let's me thinking that this is a bug. Someone else? Regards Stefan

2 8

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