- sr-users - kamailio.org

TOPOS - REGISTER
by Nathan Ward 18 Jul '16

18 Jul '16

Hi all, I’m doing some testing using Kamailio to do some lightweight SBC type functions. We have considered OpenSIPs as well as it has built in b2bua bits, but having chosen Kamailio for other parts of our network it makes sense to stick with one piece of software I think. Luckily, we only need (want) topology hiding, and don’t need other b2bua things. I have a server in the middle of our network sending registrations out to a 3rd party network for a number of lines, using Kamailio+uacreg. This works great. It sends these registrations through an SBC, also running Kamailio. I do this by setting the proxy column of the uacreg table to the SBC’s inside address. On the SBC, I rewrite the Contact header, so calls come in to the SBC. I am running topos on the SBC, and would like to be able to do topology hiding for the outgoing REGISTER. I note that there is code in tps_skip_msg to specifically avoid topology hiding for REGISTER and PUBLISH messages. Is there a SIP reason for this, or is this just a part of the code that hasn’t been finished yet or wasn’t needed by the author, so it’s been excluded? I note that topoh is the same, with a th_skip_msg function that has I think identical code, so no doubt the same reasoning is applied there. This is a very minor issue, but I’d like to understand the reasoning, and contribute code if the reason is code rather than SIP. -- Nathan Ward

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RTPProxy Issues
by TEG AMJG 18 Jul '16

18 Jul '16

Dear list I am quite new to Kamailio and i have been able to solve some NAT Traversal issues with symmetric SIP+RTP putting kamailio+rtpproxy behind NAT, i am also load balancing some asterisk boxes for transcoding and some other services like voicemail. Also i am using SIPp for load testing Now the question is that, while everything is working great with NAT (even when i am using SIPp for testing with low load) when i am about to test it with more than 150 calls or something i am starting to get the following error: *kamailio[34549]: ERROR: rtpproxy [rtpproxy.c:2735]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy* I have been trying to look at some SIP packets when the error starts and i am not seeing anything out of the normal (while being quite difficult to debug when having so many concurrent calls at SIP level) My configuration doesnt have anything new, the only thing is that, because Kamailio is behind NAT i used some ideas from some other users and debates from this same mailing list: *# RTPProxy control* *route[NATMANAGE] {* *#!ifdef WITH_NAT* * if(is_request()) {* * if(has_totag()) {* * if(check_route_param("nat=yes")) {* * setbflag(FLB_NATB);* * }* * }* * }* * if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))* * return;* * if(!route(FROMASTERISK)){* * force_send_socket(10.0.1.207);* * rtpproxy_manage("rw","10.0.1.206");* * }else{* * force_send_socket(10.0.1.206);* * rtpproxy_manage("rw","10.0.5.203");* * }* * if (is_request()) {* * if (!has_totag()) {* * add_rr_param(";nat=yes");* * }* * }* * if (is_reply()) {* * if(isbflagset(FLB_NATB)) {* * fix_nated_contact();* * }* *#!endif* * return;* *}* This solution is based of someone's idea from this mailing list: http://lists.sip-router.org/pipermail/sr-users/2013-January/076254.html As you can see in my configuration, my "public network" is really private network so i tweak the NATDETECT route to not include RFC1918 networks as part of the detection *# Caller NAT detection route* *route[NATDETECT] {* *#!ifdef WITH_NAT* *if (nat_uac_test("1")) {* * if (nat_uac_test("18")) {* * force_rport();* * if (is_method("REGISTER")) {* * fix_nated_register();* * } else {* * fix_nated_contact();* * }* * setflag(FLT_NATS);* * return;* * }* * }* *#!endif* * return;* *}* Also i would like to know how may i use RTPProxy logs?, i tried to configure it like this: /etc/rsyslog.conf *# Kamailio logging* *local0.* -/var/log/kamailio.log* *#RTPProxy logging* *local3.* /var/log/rtpproxy.log* /etc/sysconfig/rtpproxy *OPTIONS="-l 10.0.1.206 -A 10.0.5.203 -s udp:127.0.0.1:7722 <http://127.0.0.1:7722> -r /var/lib/rtpproxy/sessions -d DBUG:LOG_LOCAL3 -m 35000 -M 55000 -F"* I cannot see anyhing in the logs Thank you very much Alejandro

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Questions
by francisco rodriguez 15 Jul '16

15 Jul '16

Hello ; I have the sip account with you, but to put it to work in a team asks me some data. you can tell me what they are. my account is sip: masterlaradio(a)iptel.org I request the following information IP SERVER :? SERVER USER? USERNAME? STUN DOMAIN? DNS SERVER? thanks for your attention and collaboration Francisco Rodriguez Enviado desde Correo<https://go.microsoft.com/fwlink/?LinkId=550986> para Windows 10

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Websocket Path Issues
by Colin Morelli 15 Jul '16

15 Jul '16

Hey all, I'm using Kamailio as an outbound edge proxy for websocket connections. When my registrar calls out to Kamailio to forward to a websocket connection that has since been killed, rather than just entering the failure branch, it throws a few errors: 5(24) WARNING: <core> [msg_translator.c:2760]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 5(24) ERROR: <core> [msg_translator.c:1978]: build_req_buf_from_sip_req(): could not create Via header 5(24) ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request 5(24) ERROR: tm [t_fwd.c:1737]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches 5(24) ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (2/SL) Ultimately this means the proxy gets a 500 rather than a 430 flow failed, and the registration sits around in the proxy when it should be removed. Am I missing something obvious to get this behavior? Best, Colin

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Re: [SR-Users] Determining if request is from a local domain
by Shane Harrison 15 Jul '16

15 Jul '16

Hi all, Solved! As is often the case, after working all morning on the problem, after posting I discovered my error. I noticed that I hadn't included in my post the "alias=" command. It was indeed in my script but I had used aws-rcsstg.voip.systems instead of aws-rcsstg.vvoip.systems ! I don't know how many times I have stared at that but I guess you often see what you want to see...... Apologies for wasting your time. Kind regards Shane > -----Original Message----- > From: Shane Harrison > Sent: Wednesday, 13 July 2016 2:48 p.m. > To: 'sr-users(a)lists.sip-router.org' > Subject: Determining if request is from a local domain > > Hi all, > > From previous posts my understanding is that RURI's are matched against > reverse DNS lookup of listen addresses and aliases set via the "alias=" > parameter in the config file (assuming the domain module isn't being used). > However I am having difficulty in getting this to work. From looking at the > code and the log file, it appears to me as if the alias isn't making it into the list > of hosts that is checked against for some reason. > > Kamailio proxy is behind a loadbalancer (Kamailio dispatcher) and proxy is > trying to relay the request because it doesn't recognise it as a local domain. > Any thoughts on where my error is would be appreciated. > > Relevant CFG file section - note that > RCS_LB_PUBLIC_IP:RCS_LB_PUBLIC_PORT is 52.64.xxx.221 :5061 > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ > ~~~~~~~~~~~~~~~~~~~~ > auto_aliases=yes > > # aliases and listening settings for sipproxy alias="aws-rcsstg.voip.systems" > listen=tcp:10.4.1.62:5060 advertise > RCS_LB_PUBLIC_IP:RCS_LB_PUBLIC_PORT > > > My request is a REGISTER as follows: > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ > received message > REGISTER call-id b2ed2fdd(a)192.168.148.30 received from 10.4.1.62 raw msg > REGISTER sip:aws-rcsstg.vvoip.systems;transport=tls SIP/2.0 > Via: SIP/2.0/TCP > 52.64.xxx.221:5060;branch=z9hG4bKa691.5fd57c2a783d3a5d7f6f80fcd0c815d > 6.0;i=d91 > Via: SIP/2.0/TLS > 192.168.148.30:5061;received=203.171.46.54;rport=64515;branch=z9hG4bK68 > b886fd00518;keep > From: <sip:+6421608919@aws- > rcsstg.vvoip.systems;transport=tls>;tag=4af7e7d8 > To: <sip:+6421608919@aws-rcsstg.vvoip.systems;transport=tls> > Call-ID: b2ed2fdd(a)192.168.148.30 > CSeq: 1995005270 REGISTER > Max-Forwards: 69 > Contact: > <sip:+6421608919@203.171.46.54:64515;transport=tls>;expires=240;+sip.inst > ance="<urn:uuid:23624e3d-c98e-3b76-9acb-5499202e2f8d>";reg- > id=1;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp- > application.ims.iari.rcs.sm";+g.oma.sip-im;+g.3gpp.iari-ref="urn%3Aurn- > 7%3A3gpp-application.ims.iari.gsma-is";+g.3gpp.cs-voice;+g.3gpp.iari- > ref="urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-vs";+g.3gpp.iari- > ref="urn%3Aurn-7%3A3gpp-application.ims.iari.rcse.sp";+g.3gpp.icsi- > ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";video > User-Agent: FlowTalk-WindowsCommunicator_5.x > P-Access-Network-Info: IEEE-802.3 > Allow: > INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE,INFO,MESS > AGE,SUBSCRIBE,PUBLISH > Allow-Events: refer, presence > Supported: 100rel, replaces, timer, gruu, join, histinfo, path > Date: Wed, 13 Jul 2016 00:27:39 GMT > Content-Length: 0 > Path: > <sip:52.64.xxx.221:5060;transport=tcp;lr;received=sip:203.171.46.54:64515%3 > Btransport%3Dtls> > > > Relevant section of the log where if( !uri == me ) is tested: > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ > Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: > ERROR: *** cfgtrace:request_route=[SIPOUT] > c=[//etc/kamailio/kamailio.cfg] l=1052 a=16 n=if Jul 13 03:18:06 aws-rcsstg- > sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> > [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: > 24==9 && [aws-rcsstg.vvoip.systems] == [10.4.1.62] Jul 13 03:18:06 aws- > rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> > [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 > (advertise 5061) matches port 5061 Jul 13 03:18:06 aws-rcsstg-sip-proxy-i- > b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:603]: > grep_sock_info(): grep_sock_info - checking advertise if host==us: 24==13 > && [aws-rcsstg.vvoip.systems] == [52.64.xxx.221] Jul 13 03:18:06 aws-rcsstg- > sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> > [forward.c:448]: check_self(): check_self: host != me > > > Kind regards > > Shane Harrison > Senior Software Engineer > > Imagination Technologies NZ Limited > Level 2 > 1 Market Grove > Lower Hutt, 5010 > New Zealand > > PO Box 30-449 > Lower Hutt, 5040 > New Zealand > > Phone: +64 4 890-3681 ext 3361 >

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Re: [SR-Users] kamcmd & route[REMOTE_TEST]
by Daniel-Constantin Mierla 15 Jul '16

15 Jul '16

Hello, On 14/07/16 11:02, Marino Mileti wrote: > > Hi guys, > > > > I've two question regarding rpc command. > > > > Is it possible to "execute" a specific route (for example my > route[REMOTE_TEST]) using a rpc command by kamcmd? > > > > If not exist some other tricks to do this? > you can send a xmlrpc, jsonrpc or just http request to kamailio and it will result is executing a route or event_route inside kamailio.cfg, from where you can execute other sub-routes. See xmlrpc, jsonrpc-s or xhttp modules. Obviously, you can send a sip message with tools like sipsak or sipp from command line and handle it inside request_route to execute what route block you want. Sometimes it may just suit executing a route block periodically -- for that see rtimer module. Cheers, Daniel > > > Another question...regarding uac module... > > When Kamailio starts the UAC module makes a REGISTER according to > database...and periodically refresh this registration. Is it possible > to "force" a refresh with rpc command by kamcmd? > > > > Many thanks! > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] kamcmd & route[REMOTE_TEST]
by SamyGo 14 Jul '16

14 Jul '16

Hi Marino, How do you mean execute a route !? like enable a route for all calls onwards after the RPC command execution ? or for a specific call engage that route ? If this is to enable/disable a particular route for ALL calls based on a RPC command I'd make use of some DB or memcache variable to trigger this route i.e if(redis_cmd("srvN", "GET toggle_route", "r")){ $avp(signal) = $redis(r=>value); if($avp(signal) =~ 'true') { route(REMOTE_TEST); } } I used Redis, you can see if any RPC command can let you modify value of an AVP directly. For your second question check RPC commands uac.reg_refresh <http://www.kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.r.uac.reg_r…> . Regards, Sammy On Thu, Jul 14, 2016 at 5:02 AM, Marino Mileti <marino.mileti(a)alice.it> wrote: > Hi guys, > > > > I've two question regarding rpc command. > > > > Is it possible to "execute" a specific route (for example my > route[REMOTE_TEST]) using a rpc command by kamcmd? > > > > If not exist some other tricks to do this? > > > > Another question...regarding uac module... > > When Kamailio starts the UAC module makes a REGISTER according to > database...and periodically refresh this registration. Is it possible to > "force" a refresh with rpc command by kamcmd? > > > > Many thanks! > > > > *Marino Maria Mileti* > > *marino.mileti(a)alice.it <marino.mileti(a)alice.it>* > > > > *[image: cid:006a01cb6b0e$67eecdae$_CDOSYS2.0]**Reduce your energy > consumption and keep polar bears on ice!* > > > > > ------------------------------ > [image: Avast logo] > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

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kamcmd & route[REMOTE_TEST]
by Marino Mileti 14 Jul '16

14 Jul '16

Hi guys, I've two question regarding rpc command. Is it possible to "execute" a specific route (for example my route[REMOTE_TEST]) using a rpc command by kamcmd? If not exist some other tricks to do this? Another question...regarding uac module... When Kamailio starts the UAC module makes a REGISTER according to database...and periodically refresh this registration. Is it possible to "force" a refresh with rpc command by kamcmd? Many thanks! Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

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Modules How To ?
by Eng Hooda 14 Jul '16

14 Jul '16

Hello All, I just started using kamailio last month , using basic configuration I got from online tutorials I got it to work. Now I need to add functionality like offline messages delivery , searching lead me to msilo module , unfortunately I could not find a detailed tutorial of how to enable it and troubleshoot it's operation. All what I managed to find was the official technical documentation. https://www.kamailio.org/docs/modules/stable/modules/msilo.html It provides a list of parameters which has a default value , with no indication if the declaration of any of them is mandatory if the default values are kept intact. also there is a route code which I do not know if it was mandatory to copy to kamailio.cfg . Not to mention that the configuration file example contains lines that is already similar to what I already have , so which to keep and which to replace ? That was a case to apply to , but general answers (with msilo as an example) that apply to all modules is what I need just to avoid having to ask again if I needed to enable another module. Or may be there is something out there that I need to read first. system details : {(OS:Debian8 i386),(Kamailio4.4 with mySQL , UDP only)} Thank you all in advance. Best Regards, Eng Hooda

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is_direction / $fti / $tti not working possible
by Nathan Angelacos 13 Jul '16

13 Jul '16

I'm missing something obvious here, throwing this out to get a couple extra sets of eyeballs on it. I can't get rr modules is_direction() or the $fti / $tti pv's to work in 4.3.x or 4.4.x Relevant parts of the config: loadmodule "rr.so" modparam ("rr", "enable_full_lr", 1) modparam ("rr", "append_fromtag", 1) ... route { xlog ("L_INFO", "$rm: $fu -> $ru"); record_route (); if (has_totag ()) { xlog ("L_INFO", "Before Loose route: fti: $fti tti: $tti ci: $ci"); if (loose_route ()) { if (is_direction ("downstream")) { xlog ("L_INFO", "in-dialog $rm request from caller to callee"); } else { xlog ("L_INFO", "in-dialog $rm request from callee to caller"); } xlog ("L_INFO", "After Loose route: fti: $fti tti: $tti ci: $ci"); } } if (! t_relay ()) { sl_reply_error (); } } ---- Bob and Alice are hard phones, with kamailio + dbaliases in the middle. Alice calls Bob, then Bob (the callee) sends the BYE ---- Syslog: 01. INVITE: sip:alice@kamailio.org -> sip:bob@kamailio.org:5060 02. Alias lookup returned <sip:bob@biloxi.com> , sending on 03. ACK: sip:alice@kamailio.org -> sip:bob@biloxi.com 04. Before Loose route: fti: 707601862 tti: 45686ADF-10AAEB28 ci: 8744171(a)atlanta.com 05. in-dialog ACK request from caller to callee 06. After Loose route: fti: 707601862 tti: 45686ADF-10AAEB28 ci: 8744171(a)atlanta.com 07. BYE: sip:bob@kamailio.org -> sip:alice@atlanta.com:5066 08. Before Loose route: fti: 45686ADF-10AAEB28 tti: 707601862 ci: 8744171(a)atlanta.com 09. in-dialog BYE request from caller to callee 10. After Loose route: fti: 45686ADF-10AAEB28 tti: 707601862 ci: 8744171(a)atlanta.com Using the above config, is_direction always returns "caller", even when bob is sending the message. Looking at a packet capture: 01. Kamailio Sent: Record-Route: <sip:kamailio.org;lr=on;ftag=707601862> <on the 200 ok from Bob> Kamailio Received: Record-Route: <sip:kamailio.org;lr=on;ftag=707601862> Kamailio Sent: Record-Route: <sip:kamailio.org;lr=on;ftag=707601862> 07. Kamailio Received: Route: <sip:kamailio.org;lr=on;ftag=707601862> Kamailio Sent: Record-Route: <sip:kamailio.org;lr=on;ftag=45686ADF-10AAEB28> I expect $fti to be wrong before the loose_route (line 08); but lines 09 and 10 are wrong. I'm obviously missing something trivial here. Any ideas what I should check next? Thanks!

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Kamailio Enviroment
by Rodrigo Moreira 13 Jul '16

13 Jul '16

Hi, Right, to basic configuration of the P-I-S-CSCF services just I call the CFGS appropriate example (directory) for each entity to achieve basic funcionality. What exactly dispacher? How to config it? Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

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Kamailio know which Interface Packet would route out
by SamyGo 13 Jul '16

13 Jul '16

Hi, Question: Is there a function in kamailio to get the interface/listen IP kamailio would use to route a packet out for a domain / ip ? Example: $avp(interface_ip) = find_interface_for_request("192.168.12.123"); And it gives me the Kamailio's Listen address that would be used for sending call out to this IP/Domain ! I've a weird situation where a multi-homed kamailio with about 7 different interfaces needs to send call out, I've so many RTProxy instances to engage but I can't decide which instance to engage since I do not know which Interface the Packet would route out from. Regards, Sammy

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Re: [SR-Users] shm_malloc
by Daniel-Constantin Mierla 13 Jul '16

13 Jul '16

Hello, Can you be more explicit about what do you expect? Daniel On 13/07/16 07:27, Prakash wrote: > > Hello Daniel, > > > > Many thanks for the response. > > Could you please provide me the behind scenes of shm_malloc()?. > > It will be very helpful for me to segregate without any flaws. > > > > > > Thanks, > > Prakash > > > > *From:*sr-users [mailto:sr-users-bounces@lists.sip-router.org] *On > Behalf Of *Daniel-Constantin Mierla > *Sent:* Tuesday, July 12, 2016 6:34 PM > *To:* Kamailio (SER) - Users Mailing List > *Subject:* Re: [SR-Users] shm_malloc > > > > Hello, > > I don't expect to be very hard to extract the shm management code from > kamailio. It is inside the subfolder mem/ . > > Then, there should be some dependecies that you need to get out from > the core, coming in my mind is locking (mutexes). Maybe atomic > operations (not remembering by heart if some counters are kept for > memory management) and few other bits. > > It should work without crash, you just need to get it initialized as > kamailio does it in it's main function. > > Cheers, > Daniel > > On 12/07/16 13:57, Prakash wrote: > > Hello, > > > > Is it possible to take out shared memory code from kamailio? > > if so ,Will it work without any crash? > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Daniel-Constantin Mierla > http://www.asipto.com - http://www.kamailio.org > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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access to tls X509v3 Subject Alternative Name
by Juha Heinanen 13 Jul '16

13 Jul '16

Is there some means to get access to local and peer ssl certificate X509v3 Subject Alternative Name value? For example, if there is this in certificate: Subject: CN=foo.bar.com X509v3 Subject Alternative Name: DNS:foo.bar.com, DNS:foo.car.com, DNS:car.com $tls_my_san_hostname produces only foo.bar.com. -- Juha

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Re: [SR-Users] shm_malloc
by Daniel-Constantin Mierla 13 Jul '16

13 Jul '16

Hello, I don't expect to be very hard to extract the shm management code from kamailio. It is inside the subfolder mem/ . Then, there should be some dependecies that you need to get out from the core, coming in my mind is locking (mutexes). Maybe atomic operations (not remembering by heart if some counters are kept for memory management) and few other bits. It should work without crash, you just need to get it initialized as kamailio does it in it's main function. Cheers, Daniel On 12/07/16 13:57, Prakash wrote: > > Hello, > > > > Is it possible to take out shared memory code from kamailio? > > if so ,Will it work without any crash? > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Determining if request is from a local domain
by Shane Harrison 13 Jul '16

13 Jul '16

Hi all, >From previous posts my understanding is that RURI's are matched against reverse DNS lookup of listen addresses and aliases set via the "alias=" parameter in the config file (assuming the domain module isn't being used). However I am having difficulty in getting this to work. From looking at the code and the log file, it appears to me as if the alias isn't making it into the list of hosts that is checked against for some reason. Kamailio proxy is behind a loadbalancer (Kamailio dispatcher) and proxy is trying to relay the request because it doesn't recognise it as a local domain. Any thoughts on where my error is would be appreciated. Relevant CFG file section - note that RCS_LB_PUBLIC_IP:RCS_LB_PUBLIC_PORT is 52.64.xxx.221 :5061 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ auto_aliases=yes # aliases and listening settings for sipproxy alias="aws-rcsstg.voip.systems" listen=tcp:10.4.1.62:5060 advertise RCS_LB_PUBLIC_IP:RCS_LB_PUBLIC_PORT My request is a REGISTER as follows: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ received message REGISTER call-id b2ed2fdd(a)192.168.148.30 received from 10.4.1.62 raw msg REGISTER sip:aws-rcsstg.vvoip.systems;transport=tls SIP/2.0 Via: SIP/2.0/TCP 52.64.xxx.221:5060;branch=z9hG4bKa691.5fd57c2a783d3a5d7f6f80fcd0c815d6.0;i=d91 Via: SIP/2.0/TLS 192.168.148.30:5061;received=203.171.46.54;rport=64515;branch=z9hG4bK68b886fd00518;keep From: <sip:+6421608919@aws-rcsstg.vvoip.systems;transport=tls>;tag=4af7e7d8 To: <sip:+6421608919@aws-rcsstg.vvoip.systems;transport=tls> Call-ID: b2ed2fdd(a)192.168.148.30 CSeq: 1995005270 REGISTER Max-Forwards: 69 Contact: <sip:+6421608919@203.171.46.54:64515;transport=tls>;expires=240;+sip.instance="<urn:uuid:23624e3d-c98e-3b76-9acb-5499202e2f8d>";reg-id=1;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.rcs.sm";+g.oma.sip-im;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-is";+g.3gpp.cs-voice;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-vs";+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.rcse.sp";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";video User-Agent: FlowTalk-WindowsCommunicator_5.x P-Access-Network-Info: IEEE-802.3 Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE,INFO,MESSAGE,SUBSCRIBE,PUBLISH Allow-Events: refer, presence Supported: 100rel, replaces, timer, gruu, join, histinfo, path Date: Wed, 13 Jul 2016 00:27:39 GMT Content-Length: 0 Path: <sip:52.64.xxx.221:5060;transport=tcp;lr;received=sip:203.171.46.54:64515%3Btransport%3Dtls> Relevant section of the log where if( !uri == me ) is tested: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: ERROR: *** cfgtrace:request_route=[SIPOUT] c=[//etc/kamailio/kamailio.cfg] l=1052 a=16 n=if Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 24==9 && [aws-rcsstg.vvoip.systems] == [10.4.1.62] Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 5061) matches port 5061 Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:603]: grep_sock_info(): grep_sock_info - checking advertise if host==us: 24==13 && [aws-rcsstg.vvoip.systems] == [52.64.xxx.221] Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [forward.c:448]: check_self(): check_self: host != me Kind regards Shane Harrison Senior Software Engineer Imagination Technologies NZ Limited Level 2 1 Market Grove Lower Hutt, 5010 New Zealand PO Box 30-449 Lower Hutt, 5040 New Zealand Phone: +64 4 890-3681 ext 3361

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Detect 200 OK for ReINVITE
by Grant Bagdasarian 12 Jul '16

12 Jul '16

Hi, Is it possible in Kamailio, or SIP in general, to detect if a 200 OK is for a ReINVITE? Regards, Grant

4 5

Video conferencing with Kamailio
by Jay Li 12 Jul '16

12 Jul '16

Dear All, I'm curious if anybody has set up an infrastructure for video conferencing utilizing Kamailio as a proxy (like NAT support and so so). I found a kind of old tutorial "Run you own Skype-like service in less than one hour" kamailio:skype-like-service-in-less-than-one-hour [Asipto - SIP and VoIP Knowledge Base Site] written in 2013 on Kamailio 3.1. I wonder if this feature has been further developed in later releases. I know Kamailio is more often used as a SIP proxy, but somehow I got the impression that Kamailio supports WebRTC, so I wonder if it's possible to implement a many to many video conferencing infrastructure using Kamailio. If the answer is yes, how's the performance or the bottleneck I should pay more attention to (maybe bandwidth for video traffic, especially with NAT? ) Anyway I'm a newbie to both Kamailio and video conference, so any suggestion/discussion is appreciated. Thanks. Regards,Jay

2 3

destination ip address
by Péter Barabás 12 Jul '16

12 Jul '16

Hi All, i need to store some information in routing_logic script into database with useing SQLOPS module. I made a custom table and store such data like source and destination username, domain, ip address and port. It works fine for almost every parameter except destination IP address and port. I want to ask what is the correct way to determine the destination IP address and port of a user for a MESSAGE, INVITE or BYE message? The $Ri and $Rp is not the correct ones. Thanks. Dr. Péter Barabás

3 3

getting the local 408 response
by Jayesh Nambiar 12 Jul '16

12 Jul '16

Hi, I'm looking to catch the local 408 response generated by kamailio after the fr_timer strikes in. I was assuming to get it in the sl:local-request, but I dont see it in my log messages. I can catch responses like a 200 for register or a 487 for a message who's transaction is not yet created. Any proper way on how do I get to know when kamailio generates a local 408? I'm on the master branch (commit: 54562b873c2c3d0a04f0587e7706188a3eb19a74). Thanks, - Jayesh

4 6

Calculating time differences
by Sebastian Damm 11 Jul '16

11 Jul '16

Hi, we are suspending a transaction at one place in our route and resume it later. I want to log the time difference between suspending and resuming. I thought I could do it with saving $TV(Sn), but apparently I can't. Kamailio says, it can't convert the saved value into integer. This true, of course. I was searching for a transformation function converting this string into float, but there is none. Is there any way to calculate a time difference other than to live with seconds-only precision? Best Regards, Sebastian

2 1

Kamailio WEBRTC questions/ confusions
by Zaka 10 Jul '16

10 Jul '16

Dear List: I wonder if this has been asked before. Please ignore or point to the appropriate source, if needed. Simply, I have little experience with SIP servlets implementing WEBRTC/ Click2Call etc... Considering a Call Centre scenario offering a WEBRTC solution. Following are the confusions regarding requirements: 1: A Kamailio Box (running WEBSOCKET.config) 2: Customer will use web browser (websocket compatible of course) and enter address of the KAMAILIO Box or we shall need another KAMAILIO to statelessly forward the call to next available operator using Hunt Group or similar scheme? 3: If we don't plan to implement TLS/ MSRP (disabling these in the config file should be suffice?) 4: Last, but not the least, how will the users send request for login from browser? (assuming they have accounts created in the DB on Kamailio Box running WEBSOCKET supported KAMAILIO) With anticipatory thanks & regards, Zaka

3 6

Kamailio Enviroment
by Rodrigo Moreira 09 Jul '16

09 Jul '16

Hello, I would of your help installing Kamailio with the ability to authenticate to an HSS. Mainly want each IMS service to run on a different machine, ie, P / S / I-CSCF. I've tried several ways to set up, but without success. Does anyone have a step by step? I count on help. Hug. Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

1 0

re-INVITE on IP Change with Outbound
by Colin Morelli 08 Jul '16

08 Jul '16

Hey list, So I'm using the outbound module to ensure subsequent requests are delivered over an active TCP connection established by the mobile client. However, now I'm trying to add support for automatically responding to network changes (WiFi <-> LTE), and it's creating problems. Primarily, the issue is that when the device sends a re-INVITE for the call, its Route: header includes a flow-token corresponding to its *old* socket connection. Because the new request doesn't have the same source IP/port as the old connection did Kamailio thinks that the next route hop it needs to take is to pass the request over to the connection identified by the flow token. So it actually tries to re-open a TCP connection back out to the client even though I'm making the re-invite because that old transport is no longer valid. Does anyone have a recommended way of handling this case? I can add another header/param to the route header and manually remove it. Before loose_routing. Essentially, on in-dialog INVITE requests I want to ignore the outbound token on the originating side but I can't seem to find a way to do that without re-writing the header before loose_routing. Best, Colin

1 1

shtinc throws readonly error
by Sebastian Damm 08 Jul '16

08 Jul '16

Hi, I'm trying to use this line in one route: $shtinc(pushedUsers=>$avp(s:authuser)); However, when checking the syntax of this file, Kamailio says: 0(609) : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /etc/kamailio/kamailio_sip_proxy.cfg, line 3304, column 2-39: read only pvar in assignment left side According to the pseudo vars documentation, I should be able to use it like that. Can someone tell me what I am missing? Best Regards, Sebastian

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