sr-users August 2016

sr-users@lists.kamailio.org

80 participants
105 discussions

Radius Authentication for kamailio-4.4.2

by Chan Kwang Mien

Hi, I have installed Kamailio-4.4.2 and radiusclient-ng-0.5.6 for Radius Authentication to a Free Radius Server. I have used the steps in http://www.kamailio.org/docs/openser-radius-1.0.x.html to enable Radius Authentication for Kamailio-4.4.2. However, when I restart kamailio and my SIP phone register to Kamailio, Kamailio did not Radius Authenticate to Free Radius Server. Can anyone advise me on other document for Radius Authentication for Kamailio ? Thanks. Regards, Kwang Mien

7 years, 7 months

pua_dialoginfo: Tags only in "early" state PUBLISH

by Stefan Kohlhauser

Hello everyone! I've been playing around with the pua_dialoginfo. I noticed that the From- and To-Tags are only included in the PUBLISH's dialog-info of the "early" state. This is also true if include_tags of pua_dialoginfo is explicitly set to '1'. The reason for this behaviour is probably that in pua_dialoginfo.c in the function __dialog_sendpublish() the call to dialog_publish_multi() only happens with the tag parameters set for the "early" state. For all other dialog states the tag parameters are statically set to '0'. Even though the local- and from-tags are optional I tend to regard this as a bug. I would have expected for the tags to be included in the dialog-info for all states when include_tags is set. Or is there a reason for this behaviour? Best regards Stefan

7 years, 7 months

4.4 topos ignoring Record-Route?

by Daniel Tryba

Looking into the topos module of 4.4.2 I see something unexpected, ACK and BYE are routed directly to the Contact in the 200 OK on an INVITE where the 200 OK has a Record-Route. Without the topos module the message flow is as expected: 109.235.32.57<->185.61.68.106<->109.235.32.55 With topos the ACK and BYE take a different route: 109.235.32.57<->185.61.68.106<->109.235.32.55 <->109.235.32.48 109.235.32.5x are proxies. 185.61.68.106 is the registrar To my understanding, in the callflow below the ACK should be send via the proxy that inserted Record-Route: <sip:109.235.32.55;lr> in the 200 OK. Am I wrong to conclude this or the topos module ignoring the routing? Also note the To/From are passed untouched in the ACK/BYE and the BYE to the called endpoint hasn't had its Via headers stripped. Call scenario: tryba/+31407110xxx calls 0880100XXX. Both are registered to 185.61.68.106 via proxies (resp. 109.235.32.57 and 109.235.32.55) >From 109.235.32.57 -> 185.61.68.106: INVITE sip:0880100XXX@sip.itco.nl SIP/2.0 Record-Route: <sip:109.235.32.57;r2=on;lr> Record-Route: <sip:109.235.32.57;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK7dd7.e15a4ffb2c3a7f882117b80c85f75223.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bKb5004b60de287221 Route: <sip:109.235.32.57;transport=tcp;lr> From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl> Contact: <sip:tryba@10.0.3.169:5067;transport=tcp;alias=109.235.34.226~5067~2> >From 185.61.68.106 -> 109.235.32.55: INVITE sip:+31880100XXX@109.235.32.48 SIP/2.0 Via: SIP/2.0/UDP 185.61.68.106;branch=z9hG4bK7dd7.d4cdfe8014e0d5d806b4dc4a7422c8e0.1 Route: <sip:loadbalancer@109.235.32.55;lr;received=sip:109.235.32.48:5060> From: <sip:+31407110XXX@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:+31880100XXX@sip.itco.nl> Contact: <sip:btpsh-57b6cc75-f42d-1@185.61.68.106> >From 109.235.32.55 -> 185.61.68.106: SIP/2.0 200 OK Via: SIP/2.0/UDP 185.61.68.106;rport=5060;branch=z9hG4bK7dd7.d4cdfe8014e0d5d806b4dc4a7422c8e0.1 Record-Route: <sip:109.235.32.55;lr> From: <sip:+31407110XXX@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:+31880100XXX@sip.itco.nl>;tag=as61fc6a47 Contact: <sip:%2b31880100XXX@109.235.32.48> >From 185.61.68.106 -> 109.235.32.57 SIP/2.0 200 OK From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK7dd7.e15a4ffb2c3a7f882117b80c85f75223.0;i=60c,SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bKb5004b60de287221 Contact: <sip:atpsh-57b6cc75-f42d-2@185.61.68.106> Record-Route: <sip:109.235.32.57;r2=on;lr>,<sip:109.235.32.57;transport=tcp;r2=on;lr> >From 109.235.32.57 -> 185.61.68.106: ACK sip:atpsh-57b6cc75-f42d-2@185.61.68.106 SIP/2.0 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK7dd7.a49ec0c055166a1935ff84cfca4d1a5c.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bK24aff2df2ee7b43f From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 Contact: <sip:tryba@10.0.3.169:5067;transport=tcp;alias=109.235.34.226~5067~2> >From 185.61.68.106 -> 109.235.32.48 ACK sip:%2b31880100XXX@109.235.32.48 SIP/2.0 Via: SIP/2.0/UDP 185.61.68.106;branch=z9hG4bK7dd7.ac1b9a45042f9c26277afb9ec32bebe0.0 From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 Contact: <sip:btpsh-57b6cc75-f42d-1@185.61.68.106> >From 109.235.32.57 -> 185.61.68.106 BYE sip:atpsh-57b6cc75-f42d-2@185.61.68.106 SIP/2.0 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK4dd7.24da86af0a970ee399c9504933d8d751.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bK0d8db1380c5011b6 From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 >From 185.61.68.106 -> 109.235.32.48 BYE sip:%2b31880100XXX@109.235.32.48 SIP/2.0 Via: SIP/2.0/UDP 185.61.68.106;branch=z9hG4bK4dd7.4916bf7bcaa3b0346e43c3785a9be782.0 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK4dd7.24da86af0a970ee399c9504933d8d751.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bK0d8db1380c5011b6 From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47

7 years, 7 months

Call forking with delay

by Igor Potjevlesch

Hello list, I have a proxy that manage fix calls and a proxy that manage mobile calls. I want to implement call forking so when I call to a fix phone the call goes to the fix proxy and it forks to the mobile proxy who manage the call and viceversa. I want the fix to ring 2 or 3 times before the mobile. In order to do so in my fix proxy I added a diversion field with the reason "forking" and in my mobile proxy I check if the reason is "forking" to wait with the fonction "async_route" from the ASYNC module. When I call the fix I can fork to the mobile with delay without any problem because in that case my fix proxy manage one call and my mobile proxy the other one. But when I call the mobile the call doesn't even get to my fix proxy because my mobile proxy has to make the original call to wait while he is forking the fix call to my other proxy. And i think that calling the async_route fonction makes both calls to wait. My code is: Fix proxy: if($tU==123456789){ add_diversion("forking"); append_branch("sip:987654321@proxy_mobile"); } Mobile proxy: if($tU==987654321){ $var(z) = "1"; append_branch("sip:123456789@proxy_fix"); } route[INVITE]{ if($dir == "forking"){ async_route("RELAY", "7"); }else if($var(z) == "1"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } } How can I differentiate in the second scenario when I call the mobile phone to fork to the fix phone? Thank you for your help. Igor.

7 years, 7 months

UAC reg_contact_addr with rinstance

by DanB

Hey Guys, Have hit an issue when using uacreg. When working with Broadsoft (I know, life is not perfect), it appears that their SBC sitting in front of registration server is caching REGISTRATION requests and it will not be able to re-REGISTER in case of credential changes for example. This will break the registration process in registrar since that one flushes the registrations on password change. Their "standard" workaround it is using "rinstance" uri parameter in contact (each contact should be unique in order to be cached by their SBC). Checking it appears that I can only add rinstance statically in reg_contact_addr configuration parameter. Is there any chance to provide things like $avp or selects which I can later modify via mi maybe? Thanks in advance for any kind of tip! DanB

7 years, 7 months

detect failed socket

by Péter Barabás

Hello, I want to refine the instant message handling in our application. It works really fine in most cases. I use the msilo module to store offline messages if the user is not registered with sending PUSH notification. But, there is a situation when I have problems: When I send a message to a user whose phone is in airplane mode. In this case his location record exists but the communication on the socket is not possible and the client timed out the request. Is it possible to check the socket towards the receiver user at the moment when the message arrives at Kamailio? I think if I would check that if the socket is not fine, I could send the message to the msilo directly and the client would not have wait for timeout to resend. Thanks. Peter

7 years, 7 months

Kamailio - passing registration to another sip server

by Mike Patterson

I have configured Kamailio to pass a registration request to another VoIP provider. The reason I am doing this is to provide a sip port for users where the ISP is blocking SIP. I am able to pass the registrations to the VoIP provider but I'd like to have inbound calls working as well. I see the inbound invites coming into the Kamailio server but it appears that the server does not 'know' about the UA that is registered 'through' the server to the VoIP provider. Can anyone tell me how to fix this problem? Below is my config. Thank you, Mike #!KAMAILIO # # Edge proxy configuration # #!subst "/REGISTRAR_IP/xxxxxxxxxx/" #IP address of voip provider #!subst "/REGISTRAR_PORT/5060/" #!substdef "/FLOW_TIMER/20/" debug=2 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 alias="example.com" #mpath="/usr/lib64/kamailio/modules" mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules" tcp_connection_lifetime=30 # FLOW_TIMER + 10 auto_aliases=no mhomed=0 listen=udp:X.X.X.X:531 advertise X.X.X.X:531 alias="X.X.X.X:531" alias=X.X.X.X:531 alias="X.X.X.X:531" alias=X.X.X.X:531 port=531 ####### Modules Section ######## loadmodule "tm.so" loadmodule "sl.so" #loadmodule "outbound.so" loadmodule "rr.so" loadmodule "path.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "mi_fifo.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- modparam("tm", "failure_reply_mode", 3) # ----- rr params ----- modparam("rr", "append_fromtag", 0) ####### Routing Logic ######## request_route { route(REQINIT); if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } route(WITHINDLG); t_check_trans(); if (is_method("REGISTER")) { remove_hf("Route"); add_path(); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; } else { if (is_method("INVITE|SUBSCRIBE")) record_route(); if ($si == "REGISTRAR_IP" && $sp == "REGISTRAR_PORT") { if (!loose_route()) { switch($rc) { case -2: sl_send_reply("403", "Forbidden"); exit; default: sl_reply_error(); exit; } } t_on_failure("FAIL_OUTBOUND"); } else { if ($rU == $null) { sl_send_reply("484", "Address Incomplete"); exit; } remove_hf("Route"); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; } } route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } route[WITHINDLG] { if (has_totag()) { if (!loose_route()) { switch($rc) { case -2: sl_send_reply("403", "Forbidden"); exit; default: if (is_method("ACK")) { if ( t_check_trans() ) { route(RELAY); exit; } else { exit; } } sl_send_reply("404","Not Found"); } } else { if (is_method("NOTIFY")) { record_route(); } route(RELAY); } exit; } } onreply_route { if (!t_check_trans()) { drop; } if ($rm == "REGISTER" && $rs >= 200 && $rs <= 299) { remove_hf("Flow-Timer"); if ($(hdr(Require)[*])=~"outbound") insert_hf("Flow-Timer: FLOW_TIMER\r\n", "Call-ID"); } } failure_route[FAIL_OUTBOUND]{ if (t_branch_timeout() || !t_branch_replied()) { send_reply("430", "Flow Failed"); } }

7 years, 7 months

VUC - SER-Kamailio At 15 Years - On Friday, Sep 2, 2016

by Daniel-Constantin Mierla

Hello, on September 3, 2001, the first line of code was committed to SIP Express Router (SER) project, since 2008 turning into Kamailio project. >From pioneering large scale deployments of using SIP for Voice over IP in early 2000s, the project impacted the evolution of real time communications, disrupted the telecom vendors with its open source model and enabled many businesses to grow and innovate. Let's get together to a VUC session later this week, on Friday, Sep2, 2016, to share the good or bad stories while using SER or Kamailio during the past 15 years. If it made your life easy or a nightmare, it's time to tell it. If you have a funny photo or video, no matter is old or recent, join us and show it. More details about VUC and how one can join or watch the session are available at: - https://vuc.me If you want to actively participate during the session, email me shortly to coordinate a bit. If you cannot participate but you have something interesting to share (story, media files, etc...) email me as well and I will present it. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

7 years, 7 months

Centos, kamailio 4.4 and jansson module load issue.

by Jonathan Hunter

Hi Guys, I am running Centos 6.5 and have installed kamailio 4.4 from Git. All is running well until I try and load the jansson module. I have installed jansson-2.7 from src, but when loading the module in kamailio I get; Aug 30 20:06:24 sip1 kamailio: ERROR: <core> [sr_module.c:582]: load_module(): could not open module </usr/local/kamailio-4.4/lib64/kamailio/modules/jansson.so>: /usr/local/kamailio-4.4/lib64/kamailio/modules/jansson.so: undefined symbol: json_array_set_new Anyone else had this issue? Thanks Jon

7 years, 7 months

Domain limits and domain.reload

by Bruno Emer

Hello all. I am currently working in a company that built an online platform that allows companies talk to their customer using different medias, such as video, voice and chat. To this, we are using Kamailio to provide SIP services such as WebSockets, Registrar and proxy, and ir works really well. The point is that today we are creating a new domain for each company that uses our system, and for now we have more than 3k companies using it, so my first question is: is there a limit about how many domains we can create in a Kamailio environment? I tried to find this on the documentation but I haven't found anything. Second: every time that we create a new domain, we perform a domain.reload using the XMLRPC module. We are just using the python script to do this, but sometimes it is returning the error: Fault 400: 'Reload of domain tables failed' and we don't know why. The strange thing is that if we access the server and execute a 'kamctl domain reload' just after the error, everything works fine. The problem appears to be in the RPC module. So the question is: does someone knows what can be the cause of this problem? Thanks. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

7 years, 7 months

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sr-users August 2016