sr-users July 2016

sr-users@lists.kamailio.org

82 participants
126 discussions

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

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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

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sr-users July 2016