sr-users January 2017

sr-users@lists.kamailio.org

87 participants
111 discussions

Rate limiting
by NITESH BANSAL 31 Jul '17

31 Jul '17

Hello, I plan to add some rate limiting in the Kamailio. I'm already using Pike to rate limit based on the source IP. But in this case, i want to do the rate limiting per customer(regardless of the source IP), I can identify the customer based on a custom header in the INVITE, idea is to limit the customer to 'x' number of requests per second. Nitesh

5 12

Problem with t_relay() and VPN tunnel
by Carlos Ruiz Diaz 24 Jun '17

24 Jun '17

Hi, I have two SIP servers, *A* and *B*, connected each other though a OPENVPN tunnel. The server *B* needs to t_relay() every SIP message containing the method MESSAGE to the server *A* but these messages never reach destination. I have tested the tunnel connectivity and works fine. I wrote a Perl script (located in *B*) that sends SIP MESSAGES to Kamailio (located in *A*) trying to figure out what is happening but these messages are received by *A* and processed correctly but when *B* does the same from Kamailio, it is never received. Here is the route part of kamailio.cfg in *B*: Observation: ($rU == "1004") result is *true* * if(is_method("MESSAGE")) { if($rU == "1004") { xlog("L_INFO","En 1004"); rewritehost("10.8.0.1"); if (!t_relay()) xlog("L_INFO","MIO Error en t_relay"); t_reply("200", "Ok"); xlog("L_INFO","MIO despues rewrite"); exit; } .....* The perl script that WORKS: *$msg = 'MESSAGE sip:1004@192.168.1.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2 From: "2002" <sip:2002@192.168.1.2 <sip%3A2002(a)192.168.1.2>>;tag=1837944796 To: <sip:1004@192.168.1.2 <sip%3A1004(a)192.168.1.2>> Call-ID: 19722852989(a)192.168.1.2 CSeq: 15773 MESSAGE Contact: <sip:2002@**PublicIP-protected**:5060> Max-Forwards: 29 User-Agent: DBL Content-Type: text/plain Content-Length: 34 +595981[protected] hello from kamailio '; use IO::Socket; my $sock = IO::Socket::INET->new( Proto=>'udp', PeerHost=>'192.168.2.102', PeerPort=>'5060'); print "Sending msg $msg\n"; $sock->send($msg) or die "error sending $!\n"; *Please help! Thanks in advance. Carlos.

3 3

Missing accounting events in ACC table
by Muhammad Shahzad 23 May '17

23 May '17

Hi, About 3 weeks ago i upgraded one of my production server with latest stable kamailio version 4.2.1-fad00a. Now i am getting a lot of complaints about missing CDR events in ACC table. I observe following problems, 1. There are only BYE records in acc table, no record for INVITE or ACK. 2. In kamailio logs when ACK is received against 200 OK response for INVITE, i see following errors, -- ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse >From uri ERROR: pv [pv_core.c:434]: pv_get_xto_attr(): cannot parse From URI NOTICE: <script>: [udp:<null>@1.0.0.127:5060]: Call from 'you(a)kamailio.org' to 'you(a)kamailio.org' has been hanged up by '<null>' at '1419364717.255484' -- Of course all these errors are bogus, I have checked all headers in ACK (not just FROM header), they all seem perfectly fine and valid. 3. Then the dialog times out, -- WARNING: dialog [dlg_handlers.c:1440]: dlg_ontimeout(): timeout for dlg with CallID '6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9' and tags '1D3ECD34F5731AB845BA3064AC95BB2D' '7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4' -- 4. Any further sequential requests complain about "unable to find dialog", e.g. -- NOTICE: <script>: Sequencial 'BYE' request received from caller ERROR: uac [replace.c:591]: restore_uri(): new URI [] shorter than old URI [ sip:00xxxxxxxxxx@sip.domain.com] WARNING: dialog [dlg_handlers.c:1174]: dlg_onroute(): unable to find dialog for BYE with route param '5ae1.d595' [7845:22877] -- 5. However the acc record for BYE is written to db and log file, -- NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered: timestamp=1419364760;method=BYE;from_tag=7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4;to_tag=1D3ECD34F5731AB845BA3064AC95BB2D;call_id=6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9;code=200;reason=OK;src_user=00xxxxxxxxxx;src_domain= sip.domain.com ;src_ip=xx.xx.xx.xx;dst_ouser=+1xxxxxxxxxx;dst_user=1xxxxxxxxxx;dst_domain=yy.yy.yy.yy -- The same config was working fine with older version 4.2.0-97cab8. The kamailio config i am using is pretty much standard, -- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_DLG 4 ... modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # log to db modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", "DBURL") ... request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); }; exit; }; #### only initial requests (no To tag) #### t_check_trans(); .... # account only INVITEs if (is_method("INVITE")) { setflag(FLT_DLG); # create dialog setflag(FLT_ACC); # do accounting setflag(FLT_ACCFAILED); # ... even if the transaction fails $avp(dlg_timeout) = 60; dlg_manage(); .... } -- Any ideas why its happening? Since it is 3 weeks old so may be problem has already been spotted and fixed by someone else. Otherwise let me know how can i provide more info to help fix this issue. Thank you.

3 11

How to make OutBound calls with kamailio
by sharad tyagi 23 May '17

23 May '17

Hi I am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS. I have add and registered two IP phones with kamailio these phone able to call each other. How can i make outbound call with that I am using a asterisk server as PSTN Gateway. i am not famalier with kamailio script. Regards Sharad Tyagi SparkTG info Pvt Ltd

4 3

Parallel Forking with Different Call-Id
by SamyGo 24 Feb '17

24 Feb '17

Hi, I've a very strange scenario to work on which requires me to parallel fork the call to the same Destination provider. The only problem here is that they think that the second INVITE with different branch tag is a re-transmission and hence only take one call forward. I do not have to modify any R-URI or any headers, hence just 'append_branch() before t_relay() is in the code. ... $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); append_branch(); route(RELAY); ... *Question:* Is there anyway possible I can change the CallID of the forked INVITEs ? I have tried using Topoh module, but it still puts the same CallID before sending out. Different from the A-leg but the Sent out INVITEs have same Call-ID value. In other weird scenario, I've also tried branching, and looping call within Kamailio before sending out to carrier hoping that Kamailio would treat the two different calls and Topoh would change the Call-ID on both INVITEs before sending out.. ... if(is_present_hf("X-FORKED")) { $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); route(RELAY); }else { append_hf("X-FORKED: 1\r\n"); append_branch(); route(RELAY); } ... Still I get same Call-ID on outgoing branched call. I could branch out one INVITE to a MediaServer, say FreeSWITCH/Asterisk but again I don't want to have that component bottleneck the throughput. That could be my very last option. Looking for some ideas. Regards, Sammy

3 6

Replace old registration record with new one while keeping single AOR per contact
by Aqs Younas 09 Feb '17

09 Feb '17

Greetings list, I want to keep single AOR per contact. If new regitration comes i want to replace the old record with new. Suppose User Alice registers from device A and after certain time it try to register from device B. I want Registration from device B to replace A. Any pointer is mush appreciated. Cheers, Aqs

3 6

dlg_set_timeout bye to websockets client
by Switch168 09 Feb '17

09 Feb '17

Hello, Just wondering if it's possible to intercept the bye message that dlg_set_timeout sends out to do some nat_helper function to fix it so the bye can be sent to random.invalid uris. I know there is the edge proxy outbound module but wondering if I can do it without. Regular bye between 2 clients i was able intercept and fix with nat helper but I dont know how to fix the byes that is sent out from dlg_set_timeout function. Thanks in advance! Andrew

3 7

Cannot hear voice with symmetric NAT and STUN
by Khoa Pham 08 Feb '17

08 Feb '17

Hi, I have Kamailio as SIP server and RTP server. Client is PJSIP. I read that STUN is for non-symmetric NAT, and RTP server is for symmetric NAT. Supposed A calls B. If A, B both use symmetric NAT and STUN, they cannot hear each other If A or B use non-symmetric NAT and NOT using STUN, they cannot hear each other. Why is that? I read http://tools.ietf.org/id/draft-takeda-symmetric-nat-traversal-00.txt for Prediction Failure, is that related to this problem ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

8 18

Diversion Header - relay messed up
by Roman Dissauer 08 Feb '17

08 Feb '17

When I get an INVITE with Diversion Header the Request is forwarded without Diversion Header and the Request User is taken from Diversion User. Problem is that on the Destination Host I cannot get original Request User what is the intended destination! Is this intended behaviour? How can I change this behaviour? I already tried to delete the Diversion Header on request_route but this didn’t change the behaviour. Thanks in andvance for your help! Roman Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as Carrier IP Incoming INVITE: INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 19. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; reason=unconditional; counter=1. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 260. Content-Disposition: session; handling=required. Content-Type: application/sdp. ... Relayed INVITE: INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582.4fc0216dbecafde29127db502993222c.0. Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 18. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 274. Content-Disposition: session; handling=required. ...

3 3

Second coredump in a month
by Jon Bonilla (Manwe) 08 Feb '17

08 Feb '17

Hi all A server running version 4.4.4+0~20161223011227 has segfaulted for the second time this month. I'm attaching a bt in case someone can explain it to me. cheers, Jon

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sr-users January 2017