- sr-users - kamailio.org

jansson_get path to access a JSON reponse containing an array
by Jonathan Hunter 14 Sep '16

14 Sep '16

Hi guys, Sorry for the question but Its not clear to me using json_path and the jansson module how to get the contents of a response containing an array. Please see the returned response below that I am getting on kamailio; {"status":"success","destination"[{"msisdn":"4412345566","route":"Carrier1"}],"maxduration":"60","source":"447777777","message_id":"c41c1823aafb80f44620f4cdc6802c0d"} I can access any of the fields status,destination,maxduration,source,message_id fine, however you will notice with destination it has an array (as multiple destinations with other queries could return more than one choice). So currently when I do jansson_get_field("$http_rb", "destination", "$var(mapdestination)"); I get; [{"msisdn":"447107110194","route":"carrier1"}] How do I get into the contents of the array in the first place so I can return the msisdn value itself, and the route? This is not a problem if its not an array, however if the result is contained within [] how do I indicate the path? I have tried destination[0], destination.msisdn with no luck, and Im sure I am missing something so any advice would be great. Thanks Jon

2 2

Planning the release of Kamailio v4.4.3
by Daniel-Constantin Mierla 14 Sep '16

14 Sep '16

Hello, now that the summer in the northern hemisphere is gone and many of us returned at the work, I am considering to release a new version of Kamailio out of the latest stable branch 4.4. The plan is to have it out next week, either on Wednesday Sep 14 or Thursday, the following day. If the are issues not reported on the bug tracker from github project, it is better to open one item. Do it also if you reported something on sr-users or sr-dev in the past two months and it is not actual discussion about it, likely it was forgotten or not noticed during the holidays and digesting the archive is going to take a while. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

Kamailio as Proxy with public and private interface
by gmele 13 Sep '16

13 Sep '16

Hello, we are building a new SIP service composed by a 2 SIP Proxies and 2 SIP Registrars. There is only one proxy active (controlled by keepalived). To secure our SIP infra, we have decided to publish a TLS public interface on the proxy side. The SIP registrar will not be reachable from the outside, so we have added a provate interface on the SIP proxy and another one on the SIP registrar (192.168.1.x). In brief: SIP Proxy, 2 I/F, 213.xxx.xxx.xxx (TLS public) and 192.168.1.x (UDP private) SIP Registrar, 1 I/F, 192.168.1.x (UDP private). Is this architecture possible with kamailio? We tried to implement this, but it seems that all requests incoming from the public IP address are correctly routed to the SIP registrar, but with the public IP instead of the private one, resulting in the impossibility to send the SIP requests to the registrar. Should we add a VIA toute in the SIP message before forwarding it to the SIP registrar? How can this be done? Thx for your help Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-as-Proxy-with-public-and-p… Sent from the Users mailing list archive at Nabble.com.

1 1

Issues on getting Notify Body data and parse it
by José Seabra 13 Sep '16

13 Sep '16

Hello there, I'm facing some issues trying to get the attributes from notify Body that contains information that i need for my calls . First i save the $rb into $var then i try to replace the characters #012 that seems to be the separator of all attributes in the body, i have tried it using s.select, s.strip, s.replace but without any success in all of them. The body msg is something like this: Calling-Name-Status: available#012Calling-Name: "teste XX" < sip:XXXXXXXXX@domain.com>#012Presentation-Indicator: allowed Anyone have any idea how can I get the value of each attribute: e.g: Calling-Name-Status: available Calling-Name: "teste XX" <sip:XXXXXXXXX@domain.com> Presentation-Indicator: allowed Best Regards -- José

2 3

Dump running config out of Kamailio into a file
by Jesus Tovar 13 Sep '16

13 Sep '16

Hi, I currently have a situation with my Kamailio servers sitting in production, the folder containing all configuration files at /etc/kamailio was deleted from the server and there is not backup. Is there a possibility to show or dump the current running configuration into a file so I could restore the configuration file from it. If it happens these servers get restarted then we are going to be in real trouble. Thanks for your help Kind regards, Albert T. Jesus Tovar | Senior Network Engineer | The Summit Group jtovar(a)thesummitgroup.com.au thesummitgroup.com.au 1300 049 749 Level 1, 39 Railway Road, Blackburn VIC 3130 The Summit Group

1 0

Radius Authentication for kamailio-4.4.2
by Chan Kwang Mien 12 Sep '16

12 Sep '16

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

2 1

pua_dialoginfo: Tags only in "early" state PUBLISH
by Stefan Kohlhauser 12 Sep '16

12 Sep '16

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

2 1

Kamailio RFC/3GPP standards
by Owais Ahmad 12 Sep '16

12 Sep '16

Hello, I can only see the 3GPP specification mentioned in ims_qos module. Please have the other IMS modules also updated with the respective 3gpp versions. Thanks, Owais

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Re: [SR-Users] Kamailio - Multiple NAT
by Gholamreza Sabery 12 Sep '16

12 Sep '16

The best way to see what is happening is to capture some packets on the Kamailio system on which calls happen and see if any IP address or port number issues exist in body of SIP messages or not. You can simple use Wireshark to see what is going on. Regards On Mon, Sep 12, 2016 at 4:58 PM, Marino Mileti <marino.mileti(a)alice.it> wrote: > Hi guys, > > > > i'm going in trouble with multinat scenario and Kamailio J > > > > I've this scenario... > > > > Two kamailio boxes with the same configuration (KAM_A and KAM_B). Each box > has a WIFI in AP_MODE with the same address (but different SSID obviously) > 192.168.254.1. Finally there are 2 clients...CLIENT_1 connected to wifi > with KAM_A, and CLIENT_2 connected using wifi with KAM_B. Each box has an > instance of RTPEngine configured. > > KAM_A & KAM_B are also connected using wired connection with other clients > in order to allow that a calls to KAM_B (or KAM_A) generate a forking > scenario and all clients connected starts to RINGS (wired & wireless client) > > > > The mobile client connected on KAM_A makes a call to KAM_B... the forking > scenario starts and all clients on KAM_B starts to ring. > > If any wired client pick up the call everything works fine. If a wireless > client pick up the call...no media L > > > > Is it possible? Should I've to instruct RTPENGINE in any particular mode? > > > > > > *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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Kamailio - Multiple NAT
by Marino Mileti 12 Sep '16

12 Sep '16

Hi guys, i'm going in trouble with multinat scenario and Kamailio J I've this scenario... Two kamailio boxes with the same configuration (KAM_A and KAM_B). Each box has a WIFI in AP_MODE with the same address (but different SSID obviously) 192.168.254.1. Finally there are 2 clients...CLIENT_1 connected to wifi with KAM_A, and CLIENT_2 connected using wifi with KAM_B. Each box has an instance of RTPEngine configured. KAM_A & KAM_B are also connected using wired connection with other clients in order to allow that a calls to KAM_B (or KAM_A) generate a forking scenario and all clients connected starts to RINGS (wired & wireless client) The mobile client connected on KAM_A makes a call to KAM_B... the forking scenario starts and all clients on KAM_B starts to ring. If any wired client pick up the call everything works fine. If a wireless client pick up the call...no media L Is it possible? Should I've to instruct RTPENGINE in any particular mode? 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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Advice for routing and sbc functions
by Oliver Roth 12 Sep '16

12 Sep '16

Hi all I am using kamailio for several years now and I am really happy with it! Now an new scenario to solve: Sip-carrier A --> Kamailio loadbalancer --> multiple kamailio routing gateways --> freeswitch/sbc/media-gw/whatever --> sip - carriers B,C & D Kamailio gateways: They are doing different checks, manipulate the numbers (national /international format) and they use the carrierroute module freeswitch/sbc/media-gw/whatever: how can I use these services with the routing information from the carrierroute module? These media-gws will do the connection and maybe transcoding to the carriers. Can you please help me with the interaction within the systems "routing gateways" and the "media/sbc gateways"? Do I have to add all the carriers in the carrierroute as "routes" with the ip of the media gw/freeswitch and then splitting it up on the freeswitch again? Thanks for helping Oli

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DSCP options
by Marcelo Bonin 09 Sep '16

09 Sep '16

I need to set DSCP values in Kamailio, I couldn't found a way to set in the configuration. Any help will be appreciated. Thanks.

2 1

4.4 topos ignoring Record-Route?
by Daniel Tryba 09 Sep '16

09 Sep '16

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

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Video calls with delay on one side
by Yuriy Gorlichenko 09 Sep '16

09 Sep '16

Hi. I usein kamailio 4.4 +rtpengine 4.5 for making videocalls though Web And have issue with it: I have one way audio and video till video not started. For example i calling form A point to B point B point accept call and have Audio and Video flow from the point A but point A have no any media flow. For now i using scheme when kamailio runs together with asterisk but also i had this issue with calls without asterisk. There is a link on issue i discribed before https://github.com/sipwise/rtpengine/issues/260 I use settings at the kamailio rtpengine_manage("force trust-address replace-origin replace-session-connection RTP/SAVPF") for making UDP call to WS And rtpengine_manage("trust-address replace-origin replace-session-connection ICE=remove DTLS=passive RTP/AVP media-address=MY_IP_ADDR"); For converting to back I use folloving scheme (Call from A to B) WSphone(A)->(1-st leg)-> kamailio+rtpeinge->(2-nd leg->)asterisk asterisk ->(3-st leg)-> kamailio+rtpeinge->(4-th leg)->WSphone(B) At the example below rtpengine (with kamailio) an asteirsk uses same server But connection made thoufh external interface. not local (it is made because in a production scheme asteirsk will be in a different server, but actuaaly it is does not have any matter, just for understanding topology) So at the logs after call i see next things First call from point A to asterisk (first leg and second leg) [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1> (audio over RTP/SAVPF) using opus/48000/2 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30170 <> A.po.int.ip:51918, 244 p, 22944 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30171 <> A.po.int.ip:51920 (RTCP), 4 p, 196 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2> (video over RTP/SAVPF) using VP8/90000 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30200 <> A.po.int.ip:51922, 209 p, 164499 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30201 <> A.po.int.ip:51924 (RTCP), 129 p, 4292 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --- Tag 'as76205e6a', created 0:39 ago for branch '', in dialogue with '5aalmrbaek' [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1> (audio over RTP/AVP) using opus/48000/2 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30156 <> my.ser.ver.ip:29236, 242 p, 23861 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30157 <> my.ser.ver.ip:29237 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2> (video over RTP/AVP) using VP8/90000 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30184 <> my.ser.ver.ip:27252, 148 p, 152972 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30185 <> my.ser.ver.ip:27253 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet call from asterisk to point B (3-d leg and 4-th leg) [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1>(audio over RTP/AVP) using opus/48000/2 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30238 <> my.ser.ver.ip:29958, 239 p, 24902 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30239 <> my.ser.ver.ip:29959 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2>(video over RTP/AVP) using VP8/90000 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30264 <> my.ser.ver.ip:40374, 205 p, 163663 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30265 <> my.ser.ver.ip:40375 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --- Tag 'fgl9f33gf0', created 0:39 ago for branch '', in dialogue with 'as5f26dbd9' [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1>(audio over RTP/SAVPF) using opus/48000/2 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30218 <> B.po.int.ip:26138, 242 p, 21481 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30219 <> B.po.int.ip:30161 (RTCP), 4 p, 188 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2>(video over RTP/SAVPF) using VP8/90000 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30246 <> B.po.int.ip:31174, 198 p, 163302 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30247 <> B.po.int.ip:59057 (RTCP), 12 p, 540 b, 0 e, 1473403349 last_packet As i see there is no any lost pakets and errors and all packets going through but at the asteirsk log i see that Sent RTP P2P packet to my.ser.ver.ip:30238 (type 107, len 000102) Sent RTP P2P packet to my.ser.ver.ip:30156 (type 111, len 000071) Sent RTP P2P packet to my.ser.ver.ip:30238 (type 107, len 000079) when packets sended from asterisk to rtpengine they have no response So it looks like that asterisk have no response from rtpengine when sending him any IP. Also i checked at the webphone side but it sending pakents. So at this side no trouble... Thanks for advices!

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app_lua with lua5.3
by Péter Barabás 08 Sep '16

08 Sep '16

Hi, I would use lua5.3 with app_lua module. Is it possible since it is not compiled with lua5.1? Thanks. Peter

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Call forking with delay
by Igor Potjevlesch 08 Sep '16

08 Sep '16

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.

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Kamailio send TCP packets from random ports
by Yuriy Gorlichenko 08 Sep '16

08 Sep '16

Hi. I try to make working kamailio on TCP infront of asterisk Before to send to asteisk any packet i added $fs=ip.add.re.ss:port Also discribed listen=tcp:ip.add.re.ss:port But kamailio send outgoing packets from random prot throug TCP Presume i configured 5060 port but it send from 35410 port. googling on a core functions didn gives me any answer. May be i forget something to set at the config?

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Re: [SR-Users] Kamailio send TCP packets from random ports
by Yuriy Gorlichenko 08 Sep '16

08 Sep '16

I know about dispatcher but it not always canbe heplfull. Sometimes i need my own logic that not implemented at the dispatcher. For checking kamailio live from asterisk im use qualfy for keepalives, if i use asterisk only as media server but it not always usefull. Sometimes scenarios need to use another logic. 2016-09-08 18:24 GMT+03:00 Yuriy Gorlichenko <ovoshlook(a)gmail.com>: > I know about dispatcher but it not always canbe heplfull. Sometimes i need > my own logic that not implemented at the dispatcher. > > For checking kamailio live from asterisk im use qualfy for keepalives, if > i use asterisk only as media server but it not always usefull. Sometimes > scenarios need to use another logic. > > 2016-09-08 17:50 GMT+03:00 Federico Cabiddu <federico.cabiddu(a)gmail.com>: > >> The issue with dispatcher is that, in case of TCP transport, you cannot >> set the sending socket for the same reason. >> Basically, each time a client TCP socket is open by Kamailio, the SO >> select the port, due to the lack of support for SO_REUSEPORT. >> >> Cheers, >> >> Federico >> >> On Thu, Sep 8, 2016 at 4:33 PM, Daniel Tryba <d.tryba(a)pocos.nl> wrote: >> >>> On Thu, Sep 08, 2016 at 03:38:32PM +0300, Yuriy Gorlichenko wrote: >>> > I didnt thought about keepalive. I suppose it can help. >>> >>> Better than using qualify in asterisk is to use the dispatcher module in >>> kamailio. The idea is the same, but more configurable and it is just 1 >>> keepalive mechanisme so the amount of keepalives doesn't scale linear >>> with the number of registered users :) >>> >>> >>> _______________________________________________ >>> 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 >>> >> >> >

1 0

Kamailio 4.4.2 S-CSCF DB error
by Owais 08 Sep '16

08 Sep '16

I can see queries by S-CSCF like: The subscriber table doesn't contain presentity_uri as per: http://kamailio.org/docs/db-tables/kamailio-db-4.4.x.html#idp58608 <http://kamailio.org/docs/db-tables/kamailio-db-4.4.x.html#idp58608> Please share a valid mysql schema for tables used by s-cscf and presence. Thanks, Owais -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-4-4-2-S-CSCF-DB-error-tp15… Sent from the Users mailing list archive at Nabble.com.

1 0

UAC reg_contact_addr with rinstance
by DanB 08 Sep '16

08 Sep '16

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

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Query - REST API calls in async
by Agalya RAMACHANDRAN 08 Sep '16

08 Sep '16

Hi team, I am Agalya and I work for Comcast. We are trying to use kamailio as proxy server for our project. I have a query regarding the same. We are trying to use REST API calls asynchronously from the config file. I see that there is a module "http_async_client" which is used to handle REST services asynchronously. Found that libcurl is used to perform these operations. My question is: 1) How much time kamailio is taking to send the HTTP request out after TCP port is opened? Is there any time dependency we need to wait on libcurl to send HTTP request out? B'coz in opensips I see a delay time and libcurl is used in-turn. I found kamailio doing similar to opensips and hence am posting this question. 2) We can also use PUT service by using http_set_method(PUT)? Should we add code to support PUT service? Thanks in advance. Regards, Agalya

2 1

Mitigating DDOS attacks from carrier based on ani or dnis by limitating cps or blocking when detecting
by anfecora 08 Sep '16

08 Sep '16

Hi is there any way to use pike and htable to mitigate ddos or flood attack from trusted trunks. I need help to build it the same way kamailio control registrations. Case a carrier trunk star calling several users from the system to more than 50 CPS(calls per second), it will affect the system but cannot block the trunk since it is pstn traffic coming from a sip provider, therefore we need to find a way to identify this traffic based on ANI or DNIS or any other header and then blocked for a time just like pike does with registrations, then start the cycle all over. in less words make kamailio be aware of invite request rate, then verify that is from the same source means ani or dnis then tagged as bad traffic them star dropping it for a specified time, while normal traffic still flowing unaffected. any recommendations will be highly appreciated. thank you

2 3

New links for "IMS getting started"
by Franz Edler 07 Sep '16

07 Sep '16

Hi, Due to some storage limits I recently encountered I had to temporarily remove the image from my drop-box. Now this issue is fixed and I have uploaded the image again. Here are the links: The VMware image: https://www.dropbox.com/s/swhlgzpflgl3p3e/Debian_8.2-step5-basicKamailio%26b asicIMScore.zip?dl=0 The config-files: https://www.dropbox.com/s/nm51v4orxrox4ac/Kamailio-IMS%20config-files.zip?dl =0 Good luck. Franz DI Franz Edler University of Applied Sciences VIENNA, Austria

4 3

deb.kamailio.org down?
by Carsten Bock 06 Sep '16

06 Sep '16

Hi, currently, deb.kamailio.org is down. Is anyone taking care of this? Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-40 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

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Resume suspended invite transation after x seconds
by José Seabra 06 Sep '16

06 Sep '16

Hello, I have the following scenario: Kamailio is suspending an INVITE then generates a SUBSCRIBE msg to an external server with UAC module, and only resume the suspended INVITE when receives an NOTIFY(this NOTIFY "belongs" to the SUBSCRIBE dialog) from the external server. The issue is that sometimes kamailio doesn't receive a NOTIFY and the transaction is not resumed, any one have an idea how can i implement a solution to resume the suspended invite after x seconds without receiving NOTIFY msg? Best Regards -- José Seabra

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