- sr-users - kamailio.org

Kamailio Authentication failing
by Eric Koome 16 Sep '16

16 Sep '16

Hi all - my Kamailio - 4.1.6 is receiving this particular structured INVITES from multiple IPs, and for some reason it is not requesting authentication. I have AUTH & IPAUTH modules in use for two years now, but this is bypassing that and actually forwarding the invite to asterisk servers behind Kamailio. I notice from the invite that the contact (c=IN IP4 10.10.10.10) is unusual and in the private range. Is this what is bypassing Authentication? Any Pointers on how to stop this. This is flooding my boxes! ========================================== 2016/09/16 06:45:16.824768 83.48.28.27:63029 -> MYPUBLICIP:5060 INVITE sip:9441134018000@MYPUBLICIPSIP/2.0 Via: SIP/2.0/UDP 83.48.28.27:5060;branch=z9hG4bK898408493 Max-Forwards: 70 From: "442030930112" <sip:442030930112@83.48.28.27>;tag=5044916148 To: <sip:9441134018000@MYPUBLICIP> Contact: <sip:442030930112@83.48.28.27> Call-ID: 9xCjfQkrmDwT7D6AqM0BTYAgmWjG84nq CSeq: 102 INVITE User-Agent: SIP Call Date: Fri, 16 Sep 2016 05:45:48 +0000 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 308 v=0 o=442030930112 8982 1536 IN IP4 10.10.10.10 s=SIP c=IN IP4 10.10.10.10 t=0 0 m=audio 14806 RTP/AVP 0 3 8 4 18 101 a=rtpmap:0 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Regards Eric

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http_async_query - How to Allow SIP transaction to continue
by Jonathan Hunter 16 Sep '16

16 Sep '16

Hi Guys, I am just testing with jansson and http_async_query. Essentially I sent an INVITE into kamailio ,it hits CALL_START_REQUEST (see below) and the query allows the http site I send to, to provide routing details for kamailio. Logging the $rm that hits CALL_START_ REQUEST, it shows as INVITE, however when CALL_START_REQUEST_REPLY is hit, the message type is OPTIONS, is this expected behaviour? Essentially I want it to continue in the script with the same $ci,$rm so I can onwards route it for termination, but all I get are options messages being sent, any hints please? As Ideally I want the transaction to continue to a route where I can forward on modified request for termination. Thanks Jon Routes; # Send to FreeSWITCHroute[CALL_START_REQUEST]{xlog("L_INFO","route[CALL_START_REQUEST] Call received $ci,$rm,$ru,$du\n"); ###Build JSON request to send to MAP$var(params) = $null;jansson_set("string", "username", "xxx", "$var(params)");jansson_set("string", "password", "xxx", "$var(params)");jansson_set("string", "source", "$fU", "$var(params)");jansson_set("string", "route", "$var(carriername)", "$var(params)");jansson_set("string", "destination", "$rU", "$var(params)");/// other fields set herehttp_set_timeout(5000);http_set_method("POST");http_append_header("Content-Type: application/json");http_async_query("http://test.com", "$var(params)", "CALL_START_REQUEST_REPLY");xlog("L_INFO", "Checking here we have status=$var(status)\n");xlog("L_INFO", "Current Destination is $du\n");#return;#exit;}#########################route[CALL_START_REQUEST_REPLY]{xlog("L_INFO", "route[CALL_START_REQUEST_REPLY]: $ci,$rm,$ru,$du\n"); Logs show; route[CALL_START_REQUEST] Call received b087fa53-f5fc-1234-94ac-f68eb8da94f7,INVITE,sip:441624891676@8.8.8.8:5060;user=phone,<null> route[CALL_START_REQUEST_REPLY]: 123,OPTIONS,sip:G711A-282951200@kamailio.org,sip:G711A-282951200@8.8.8.8:5080

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help with kamailio rpm made from source
by Dmitry 16 Sep '16

16 Sep '16

Hello I obtained the source from https://www.kamailio.org/pub/kamailio/4.3.4/src/kamailio-4.3.4_src.tar.gz I found a SPEC file there and I prepared a RPM with this spec file. When I start a Kamailio (from this rpm) - it starts but the LOG file gives only 1 line: INFO: <core> [tcp_main.c:4657]: init_tcp(): using epoll_lt as the io watch method (auto detected) I want to test this rpm and if successfull i willl add patches and rebuild the rpm. How to get correct RPM?

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Branch variables in branch_failure_route
by Paul Smith 14 Sep '16

14 Sep '16

Hi, I am struggling to figure out variable scope in branches and branch_failure_routes. Is there a way to store a variable in a branch route so that I can then read it from the branch failure route if that branch fails? Also does writing $ru in branch_failure_route append a branch? For example I have multiple SIP devices registered to Kamailio on the same account credentials. The devices are behind same NAT, so same public IP. Some devices have SRTP required and some cannot use SRTP at all, Kamailio does not know in advance of the call whether the call will use SRTP or RTP. When I send a call to that user Kamailio correctly parallel forks to all the registered devices. I use rtpengine offer to send SDP with RTP/SAVP in the initial invites. If one of the devices sends a 488 due to not being able to handle the RTP/SAVP I then want to run rtpengine_offer with RTP/AVP and send another invite to the same device. In the branch failure route I seem to have to read the current $ru and then write it back into $ru before calling t_relay() or I get Sep 14 14:56:06 registrar-secure /usr/sbin/kamailio[10168]: ERROR: tm [t_fwd.c:1771]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding This code seems to work, but it feels like I’m doing something very wrong…. in particular $avp(triedrtpout) is set for the transaction, I should be setting it separately for each branch. event_route[tm:branch-failure:SRTP] { # Handle failure response xlog("L_INFO", "Handling $T_reply_code response to $rm to ru: <$ru> and du: $du\n"); if(t_check_status("488") && is_method("INVITE")) { if ($avp(triedsrtpout)==1) { # set $ru seems to create a new branch ... even when we set it to its current value. # we want to generate a new SIP INVITE to the same AOR but with different SDP $avp(buffer)=$ru; $ru=$avp(buffer); xlog("L_INFO","488 caught. Resending to $ru"); # try with unencrypted RTP/AVP rtpengine_delete(); rtpengine_offer("to-tag trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); if (!t_relay()) { sl_reply_error(); } } } } I am using kamailio 4.2 debian package for this registrar. Sorry for the rambling question … I don’t understand enough to write the short version! Paul

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Kamailio v4.4.3 Released
by Daniel-Constantin Mierla 14 Sep '16

14 Sep '16

Hello, Kamailio SIP Server v4.4.3 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.3. For more details about version 4.4.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/09/kamailio-v4-4-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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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

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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

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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.

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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

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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

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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

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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

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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.

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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!

1 0

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

2 2

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.

2 3

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?

3 12

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

2 6

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/

4 5

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

2 4

detect failed socket
by Péter Barabás 05 Sep '16

05 Sep '16

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

5 9

presence modules not send NOTIFY events
by Yuriy Gorlichenko 05 Sep '16

05 Sep '16

Hi. I trying to implement full presence server I set config like this (part of it) #!define FLT_DLG 9#!define FLT_DLGINFO 10 modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_flag", 4) modparam("dialog", "initial_cbs_inscript", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "track_cseq_updates", 1) modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", FLT_DLG) modparam("presence", "db_url", DBURL) modparam("presence", "server_address", SIPURI) modparam("presence", "send_fast_notify", 1) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 1) modparam("presence", "fetch_rows", 1000) modparam("presence", "local_log_level", 3) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #-------presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_dummy_dialog", 1) modparam("presence_dialoginfo", "force_single_dialog",1) # ----- pua params ----- modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 10) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) modparam("presence", "db_url", DBURL) modparam("presence", "server_address", SIPURI) modparam("presence", "send_fast_notify", 1) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 1) modparam("presence", "fetch_rows", 1000) modparam("presence", "local_log_level", 3) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #-------presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_dummy_dialog", 1) modparam("presence_dialoginfo", "force_single_dialog",1) # ----- pua params ----- modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 10) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) At the config file i made some changes route[BLF] { # absorb retransmissions if (! t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { if (handle_publish()) { xlog("L_INFO","{$rm} handled"); }; t_release(); } else if( is_method("SUBSCRIBE")) { if (handle_subscribe()) { xlog("L_INFO","{$rm} handled"); }; t_release(); }; exit; } Also added for evey call dlg_manage() function So for now i have issue that no NOTIFY event was sent at the any update of any device I see PUBLISH events and SUBSCRIBE events but it looks like handle_publish() and handle_subscribe() not sending any NOTIFY But at the dump i see xlog("L_INFO","{$rm} handled"); where $rm is correct SUBSCRIBE and PUBLISH if it is. Just dont know what i need to change. I will be very grateful if someone heps me to resolve this issue. Thanks

2 2

Kamailio - passing registration to another sip server
by Mike Patterson 05 Sep '16

05 Sep '16

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"); } }

4 9

VUC - SER-Kamailio At 15 Years - On Friday, Sep 2, 2016
by Daniel-Constantin Mierla 02 Sep '16

02 Sep '16

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

1 1

IMS RFCs
by Owais Ahmad 02 Sep '16

02 Sep '16

Hi guys, Can you list the RFCs and 3GPP standards versions for each of the CSCFs in Kamailio? Regards, Owais

1 0

Kamailio with Dispacher
by Rodrigo Moreira 02 Sep '16

02 Sep '16

Hello, I would like to clarify a question. Can you help me? I'm deploying a solution based on Kamailio, but I need incorporate the load balancer funcionality in the environment. See the cenario: I have five VMs. PCSCF, ICSCF, SCSCF, HSS, DNS. Each VM runs the especific funcionality of the IMS. Therefore, which machines should receive the config provided by dispacher module? Is the PCSCF? Another question: The DNS server should be updated, I may do this previously? The HSS, should be updated too, I may do this previously? Do you known about approaches of the LB available? Best Regards. Thanks. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

issue with memcached module
by jay binks 02 Sep '16

02 Sep '16

if I use the memached module in kamailio to set STRING data in memcache, this is not readable from PHP. This is because kamailio uses the "val_flag" to distinguish the type of data ( int or string ). There does not appear to be a standard for how data type is indicated ( in fact the protocol doc for memcache says "<flags> is an arbitrary 16-bit unsigned integer" ) I think the memcache module really needs a way to let the user specify the value of the flag. something like a modparam to override the flag value on every set. has anyone else run in to this ? Sincerely Jay Binks

2 2

Centos, kamailio 4.4 and jansson module load issue.
by Jonathan Hunter 01 Sep '16

01 Sep '16

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

3 5

Domain limits and domain.reload
by Bruno Emer 01 Sep '16

01 Sep '16

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_>

2 1

kamailio Bigning
by Gaurav Bmotra 01 Sep '16

01 Sep '16

hi i m new with kamailio so ple help me what i need to next i have install kamailio in centos 6.7 and also testing SIP to SIP call which is working but now i want to do some practice on routing (in kamailio.cnfg) so tell me what are the basics of it and what task i need to do for practice Thank you -- *Regards:* Gaurav Bmotra Smith Infotech E-46, phase 8 , Indus Area S.A.S Nagar Mohali 160071

2 1

Kamailio Errors that I see in the system logs
by Kevin Greene 01 Sep '16

01 Sep '16

Hi: Looking for help in this logging event. Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: db_mysql [km_dbase.c:124]: db_mysql_submit_query(): driver error on query: PROCEDURE kamailio.kamailio_cdrs does not exist (1305) Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [call kamailio_cdrs()] Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: db_mysql [km_dbase.c:124]: db_mysql_submit_query(): driver error on query: PROCEDURE kamailio.kamailio_rating does not exist (1305) Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [call kamailio_rating('default')] I assume the database is missing table entries for these two data inserts. Any help would be useful. Thanks. Kevin

3 2

Htable keys and key::size inconsistencies
by Fabian Borot 01 Sep '16

01 Sep '16

Hi, I am using htable module with version 4.4 storing multiple values in the same key (I set the key_type = 1 for each entry, array key). I have found an issue with the htable module. Some keys show up when using: "./kamcmd htable.get my_table key[0]" but when asking for the "key::size" I says that the key does not exist. Also some entries have the "key::size" entry but the actual "key[0]" for example is reported as non existent: See these 2 examples: Ex:1 "key[0]" is shown but "key::size" is not: [root@ar-mia-proxy-01 sbin]# ./kamcmd htable.get htable_lcrgwA 8[0] { item: { name: 8[0] value: 555|5875|1.1.1.1|5060|0|4|1|1|Broadvox2010IN flags: 2 expire: NEVER } } [root@ar-mia-proxy-01 sbin]# ./kamcmd htable.get htable_lcrgwA 8::size error: 500 - Key name doesn't exist in htable. Ex2: "key::size" is shown but "key[0]" is not: [root@ar-mia-proxy-01 sbin]# ./kamcmd htable.get htable_lcrgwA 9[0] error: 500 - Key name doesn't exist in htable. [root@ar-mia-proxy-01 sbin]# ./kamcmd htable.get htable_lcrgwA 9::size { item: { name: 9::size value: 6 flags: 0 expire: NEVER } } Also, for this case "9::size", supposedly there are 6 entries and in reality there are only 4 in the tables where I am loading them from For the case "8::size" that returns 2 the count it is right , the prob is that I can not get the "8[0]" and "8[1]" keys Any ideas are very appreciated, thank you

2 1

kamailio outgoing gateway
by Andrzej Kaczmarczyk 31 Aug '16

31 Aug '16

I have common, cheap TL-WR1043ND V1 router. For private purposes I've set kamailio voip on my own server INSIDE my LAN. Is it possible to connect to that server from outside, through openWRT router? Many tries, no success - too many jumps or something similar. I guess router should be fully transparent for sip protocol. How to manage it? is it possible at all? I don't ask how to do it in details on my router/openwrt (but if someone knows, it will be nice), but - is this possible to setup at all? should I try or search for resolution, or just give up with that idea? -- Pozdrawiam Andrzej Kaczmarczyk

3 2

Segfault in qm_insert_free
by Igor Potjevlesch 31 Aug '16

31 Aug '16

Hello, We got a crash yesterday of a Kamailio instance (4.2.7). I can see a segfault: "Program terminated with signal 11, Segmentation fault. #0 0x000000000061a856 in qm_insert_free (qm=0x7fcbde513000, frag=0x7fcbded7dd08) at mem/q_malloc.c:180 180 prev=FRAG_END(f)->prev_free;" The output of "bt full": (gdb) bt full #0 0x000000000061a856 in qm_insert_free (qm=0x7fcbde513000, frag=0x7fcbded7dd08) at mem/q_malloc.c:180 f = 0x7fcbdf2d1e18 prev = 0x0 hash = 78 #1 0x000000000061d6dc in qm_free (qm=0x7fcbde513000, p=0x7fcbded7dd38, file=0x7fcbf4ffe34d "tm: h_table.c", func=0x7fcbf4ffe628 "free_cell", line=186) at mem/q_malloc.c:537 f = 0x7fcbded7dd08 size = 624 next = 0x400 prev = 0x7fffc283f720 __FUNCTION__ = "qm_free" #2 0x00007fcbf4f3fc9d in free_cell (dead_cell=0x7fcbdf2fa360) at h_table.c:186 b = 0x7fcbded7dd38 "PRACK sip:1.2.3.4:5060;transport=UDP SIP/2.0\r\nVia: SIP/2.0/UDP 4.5.6.7;branch=z9hG4bK7419.88222290c54ce503af46b89a1da0c125.0\r\nVia: SIP/2.0/UDP 9.8.7.6:5060;branch=z9hG4bK0cB6cd54b5"... i = 0 rpl = 0x0 tt = 0x7fcbde6b7778 foo = 0x7fffc283f7f0 cbs = 0x0 cbs_tmp = 0x7fcbde79fca0 __FUNCTION__ = "free_cell" #3 0x00007fcbf4f84a1c in wait_handler (ti=1160550038, wait_tl=0x7fcbdf2fa3e0, data=0x7fcbdf2fa360) at timer.c:675 p_cell = 0x7fcbdf2fa360 ret = 1 #4 0x00000000005fd30f in timer_list_expire (t=1160550038, h=0x7fcbde58c908, slow_l=0x7fcbde58ea28, slow_mark=19953) at timer.c:888 tl = 0x7fcbdf2fa3e0 ret = 1160550038 #5 0x00000000005fd757 in timer_handler () at timer.c:953 saved_ticks = 1160550038 run_slow_timer = 0 i = 497 __FUNCTION__ = "timer_handler" #6 0x00000000005fdbc5 in timer_main () at timer.c:992 No locals. #7 0x00000000004a77e6 in main_loop () at main.c:1700 i = 8 pid = 0 si = 0x0 si_desc = "udp receiver child=7 sock=91.213.145.60:5060\000\177\000\000\000\372\203\302\377\177\000\000\033{ N\000\000\000\000\000P\372\203\302\377\177\000\000\004\000\000\000\000\000\0 00\000`TA\000\000\000\000\000(\205T\336\313\177", '\000' <repeats 14 times>, "\001\000\000\000P\372\203\302\377\177\000\000\276{N\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #8 0x00000000004acfab in main (argc=7, argv=0x7fffc283fcc8) at main.c:2581 cfg_stream = 0xe5e010 c = -1 r = 0 tmp = 0x7fffc283ff70 "" tmp_len = 32767 port = -1031537762 proto = 0 options = 0x7033b8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 1876790668 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x40d134 p = 0xc2 <Address 0xc2 out of bounds> __FUNCTION__ = "main" I don't figure out what is gone wrong. Thank you for your help. Regards, Igor.

1 1

Looking for network version of ds_reload
by Chad McElligott 30 Aug '16

30 Aug '16

Hello all, I'm in the process of converting an on-host install of Kamailio to be run inside a docker container. A part of this Kamailio install is a small node.js application that modifies the sqlite database Kamailio uses, then runs `kamctl fifo ds_reload` to tell Kamailio to reload its knowledge from that database. I'm trying to split these two processes (the node app and the Kamailio server) into separate docker containers, in an effort to follow the one-process-per-container philosophy, but having trouble figure out exactly how I should perform the ds_reload step without access to the kamctl in the Kamailio container. The obvious direction that I'm looking is to perform some sort of network request, since I can configure the two docker containers to have network access to each other. Is there some way to perform a ds_reload over the network? Thanks for all you do. Kamailio is a great SIP server and we use it to great effect. -Chad McElligott

2 4

uac_auth and (serial) forking
by Daniel Tryba 30 Aug '16

30 Aug '16

Trying to implement a way to authenticate outgoing INVITEs to endpoints that need/want to. uac_auth in the initial failure route works fine when there is only 1 registered user to send the INVITE to. But with at least 2 locations are available and 1 accepts the INVITE without authentication, the 401 of the location that wants authentication isn't processed in the failure route until a timeout is triggered (t_set_fr). At that point the 401 is processed and a response generated with Authentication header. But since t_set_fr invite timeout was exceeded it is immediatly destroyed due to a CANCEL having been generated to the location(s). t=0: t_set_fr(10000); INVITE to A anb B A responds 100 Trying,180 Ringing B responds 401 Unauthorized t=10s: T_code=180, new_code=408 failure route status: 401 uac_auth() INVITE with Authorization is build to B CANCEL to A INVITE to B is never send Is this expected behaviour? Obviously I didn't expect this.

2 4

Configuration Issue on Kamailio.
by Laura 30 Aug '16

30 Aug '16

Dear list, i'm asking here a question about Kamailio config. We are testing a wide area configuration of Kamailio over separates countries and we are still facing with an issue. We configured Kamailio 4.3.5 with dialog support over the TM modules and we use LCR module for menage ours LCRs rule set profiles. For some technicals reasons we use tech prefix for our customer so for exaples customer1 send traffic to us with 1111 prefix, customer2 send traffic to us with 2222 and something similar.. Our supplier, of course, are using tech prefix too so for examples if i want to send the call to supplier1 i need to use tech prefix 1789 or something similar.. The point is.. When customer1 is sending an invite to us.. it send us something like (Bangladesh mobile 8801xxx) INVITE sip:11118801xxxxxxx@aaa.bbb.ccc.ddd Our Kamailio will reply with the Trying and then it goes to LCR module and match our supplier1 so it make a new invite like this INVITE sip:17898801xxxxxx@supplier.ip The problem come when supplier1 reply to us and we replies back to customer1.. Customer1 view the From: field with the 17898801xxxxxx numbers.. and some of our customers don't like it. We don't use anymore the topoh module becuase we found some troubles using it.. so.. Is there a way that we can use for fix this situation ? Best regards.

4 7

Kamailio and user registration and a question on STUN servers
by Kevin Greene 29 Aug '16

29 Aug '16

Hello Group: I am looking for a way to allow any SIP client to register to my (development) network without having to be already registered in the database. Is there a way for the user client to register and create a valid user subscription in the database? The SIP clients I am using have hard coded passwords and I can not change them. I think once loaded into the database, I can delete the ones I do not want later after I close automatic registration feature... The second question is on STUN servers. Is there a good way to test to see if one is working from outside your network? I was looking for some sort of web based tool I could access to see if my public STUN server is really working. I am using the "reTurnServer" software on Linux. I am only allowing TCP based STUN requests. Thank you. Sample log entry for a external user trying to access the development network. I would like to give this person access to place calls on the system. 199.58.xxx.xxx.nebraskalink.com - - [25/Aug/2016:18:02:49 +0000] "GET /download/?ojo-mac=00:02:54:00:a5:eb&ojo-id=2000-0005-2F52&ojo-rev=rev02.01.15&ojo-config-id=e712e6cb088ab9b4ef246488b979cbf6&ojo-stun-type=Blocked&ojo-sn=15190252560146703100&ojo-key-id=e0142f126a5197845a24f33deedc358645e67e5f HTTP/1.0" 200 1812 "-" "Wget/1.9.1"

3 2

Kamailio S-CSCF Error
by Rodrigo Moreira 29 Aug '16

29 Aug '16

Hello, How can I deal with the error: Aug 13 13:55:43 scscf kamailio[2961]: tcp: 192.168.0.101 [192.168.0.101]:5060 Aug 13 13:55:43 scscf kamailio[2961]: Aliases: Aug 13 13:55:43 scscf kamailio[2961]: *: scscf.k-ims.test:5060 Aug 13 13:55:43 scscf kamailio[2961]: 0(2963) ERROR: ims_usrloc_scscf [ul_mod.c:294]: mod_init(): can't bind presence Aug 13 13:55:43 scscf kamailio[2961]: 0(2963) ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module ims_usrloc_scscf (/usr/lib/x86_6..._scscf.so) Aug 13 13:55:43 scscf kamailio[2961]: ERROR: error while initializing modules Aug 13 13:55:43 scscf kamailio[2961]: 0(2963) CRITICAL: cdp [diameter_peer.c:487]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Aug 13 13:55:43 scscf systemd[1]: kamailio.service: control process exited, code=exited status=255 Aug 13 13:55:43 scscf systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Aug 13 13:55:43 scscf systemd[1]: Unit kamailio.service entered failed state. Thank you. Regards., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 2

ims_dialog- dlg set error
by Suresh.Waware＠relianceada.com 26 Aug '16

26 Aug '16

Hi, We are using kamailio 4.4.2 version source code. For building SCSCF we are using example configuration files provided in the source code. While running SCSCF; we are facing following error. ERROR: ims_dialog [ims_dialog.c:378]: mod_init(): no dlg flag set!! q Regards, Suresh The information contained in this electronic message (email) and any attachments to this email are intended for the exclusive use of the addressee(s) and access to this email by any one else is unauthorised. The email may contain proprietary, confidential or privileged information or information relating to Reliance Group. If you are not the intended recipient, please notify the sender by telephone, fax, or return email and delete this communication and any attachments thereto, immediately from your computer. Any dissemination, distribution, or copying of this communication and the attachments thereto (in whole or part), in any manner, is strictly prohibited and actionable at law. The recipient acknowledges that emails are susceptible to alteration and their integrity can not be guaranteed and that Company does not guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

3 2

Change debug level without restarting Kamailio
by Federico San Martín 26 Aug '16

26 Aug '16

Hi guys, sorry for the newbie question, I searched for it and couldn't find any reference, is there a way to change the debug level of a kamailio server on the fly? If I need to add an additional module it's not a problem. I'm using version 4.4.2 on Debian Jessie. Thanks! Federico

3 3

kamailio crushes because of tcp errors
by Dmitry 26 Aug '16

26 Aug '16

hel >From time to time (it does not depend upon traffic) we see in logs: ERROR: <core> [tcp_read.c:274]: tcp_read_data(): error reading: Bad file descriptor (9) ([ip_address]:49832 -> [ip_address]5060) ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error readingERROR: <core> [io_wait.h:608]: io_watch_del(): trying to delete already erased entry 3031 in the hash(0, 0, (nil)) flags 10)CRITICAL: <core> [tcp_read.c:1654]: handle_io(): io_watch_del failed for 0x7f8471c51820 id 867 fd 3031, state 0, flags 4018, main fd 179, refcnt 2 ([ip_address]:49832 -> [ip_address]:5060) there are a lot of such TCP CRITICAL errors and the kamailio crushes kamailio -vversion: kamailio 4.3.5 (x86_64/linux)flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: unknowncompiled on 17:51:55 Mar 8 2016 with gcc 4.4.7 It is still not clear - what exactly causes such behaviour. I want to recriate it in the lab - so what can cause it.? May be a process erases the file descriptors of TCP connections? Any help is appreciated.

1 1

Checking if a reply is part of a forked transaction
by Phil Lavin 25 Aug '16

25 Aug '16

We need to check if a 1XX reply is part of an INVITE transaction that was forked to 2 or more phones. We figure that we can do a location lookup, based on the to header of the 1XX... but is there a better way? Phil

4 4

Fwd: Kamailio with SNMP
by Rodrigo Moreira 24 Aug '16

24 Aug '16

Hello, Once set up and running the Kamailio environment would like to install SNMP and incorporate the functionality to collect the statistics of IMS entities. It's possible? When I call the module "snmpstats" I am informed that the module is not present. How can I solve? Another issue, which modules are monitorable via SNMP, P-CSCF, I-CSCF, S- CSCF? All? Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Monitor diameter state and number of current calls
by Ramiro Romero Arjona 23 Aug '16

23 Aug '16

Hi, We would like to know if there is some way to check the state of a diameter connection (to see is it's in state 6: connected), inside the kamailio configuration script. Also we would like to know if there is a way to see the number of current calls in a kamailio through kamcmd, kamctl or something like that. We have searched for a solution but we don't find anything. Many thanks.

1 0

Send JSON document over http and process response
by Jonathan Hunter 22 Aug '16

22 Aug '16

Hi Guys, I need to send a JSON document over HTTP to a server, and then process the response, both in the kamailio configuration file. I assume I need to use Jansson module to extract contents of response? What is the best method for generating the initial request containing JSON ? Many thanks Jon

2 2

Howto timeout INVITEs over TCP to something < 30s?
by Daniel Tryba 22 Aug '16

22 Aug '16

If an enduser INVITEted over TCP t_set_fr(t) with t < 30s have no effect, the INVITE will always timeout after 30s. What is controlling this 30s? The only default at 30s is the fr_timer, which should have no part in the timeout between an 1xx and a final answer for an INVITE.

2 5

Evapi module is not broadcasting any data
by Efelin Novak 22 Aug '16

22 Aug '16

Hi folks, I came into a problem with evapi module, while I was trying to make the CGRateS tutorial work. It does not send any messages at all. Taken steps: I start kamailio with debug=3. I start: CGRates (on the kamailio machine) telnet 192.168.30.29 8448 (from my PC) tcpdump port 8448 -A -s0 -i any (on the kamailio machine) Telnet command successfully connects to kamailio. Kamailio yells: root@cgrates:/opt/kamailio/src/kamailio-4.4/kamailio# 11(12205) DEBUG: evapi [evapi_dispatch.c:453]: evapi_accept_client(): new connection - pos[1] from: [192.168.30.1:40312] 11(12205) INFO: <script>: Starting Evapi connector srcaddr 192.168.30.1 srcport 40312 condix 1 11(12205) DEBUG: htable [ht_var.c:90]: pv_set_ht_cell(): set value for $sht(cgrconn=>cgr) A similar message is printed when CGRateS connects in. Afterwards I make a call to Kamailio. INVITE is routed to a following command: evapi_async_relay("{\"event\":\"CGR_AUTH_REQUEST\", \"tr_index\":\"$T(id_index)\", \"tr_label\":\"$T(id_label)\", \"cgr_reqtype\":\"$dlg_var(cgrReqType)\", \"cgr_tenant\":\"$dlg_var(cgrTenant)\", \"cgr_account\":\"$dlg_var(cgrAccount)\", \"cgr_destination\":\"$dlg_var(cgrDestination)\", \"cgr_setuptime\":\"$TS\", \"cgr_computelcr\":\"true\"}"); Now the kamailio writes: . . . 2(12196) DEBUG: dialog [dlg_var.c:260]: print_lists(): Dialog var-list (0x7f196d1c5248): 2(12196) DEBUG: dialog [dlg_var.c:265]: print_lists(): cgrDestination=1002 (flags 1) 2(12196) DEBUG: dialog [dlg_var.c:265]: print_lists(): cgrAccount=1001 (flags 1) 2(12196) DEBUG: dialog [dlg_var.c:265]: print_lists(): cgrTenant= cgrates.org (flags 1) 2(12196) DEBUG: dialog [dlg_var.c:265]: print_lists(): cgrReqType=*prepaid (flags 1) 2(12196) DEBUG: dialog [dlg_hash.c:922]: dlg_unref(): unref dlg 0x7f196d1c4c08 with 1 -> 1 2(12196) DEBUG: evapi [evapi_dispatch.c:603]: evapi_relay(): relaying event data [{"event":"CGR_AUTH_REQUEST", "tr_index":"59559", "tr_label":"1886589610", "cgr_reqtype":"*prepaid", "cgr_tenant":"cgrates.org", "cgr_account":"1001", "cgr_destination":"1002", "cgr_setuptime":"1471623526", "cgr_computelcr":"true"}] (246) 2(12196) DEBUG: evapi [evapi_dispatch.c:628]: evapi_relay(): sending [0x7f196d1c88b8] [246:{"event":"CGR_AUTH_REQUEST", "tr_index":"59559", "tr_label":"1886589610", "cgr_reqtype":"*prepaid", "cgr_tenant":"cgrates.org", "cgr_account":"1001", "cgr_destination":"1002", "cgr_setuptime":"1471623526", "cgr_computelcr":"true"},] (251) 2(12196) DEBUG: evapi [evapi_dispatch.c:630]: evapi_relay(): Len of write is 8 2(12196) INFO: <script>: suspended transaction: 59559 / 1886589610 . . . However no reply from CGRateS comes, tcpdump is quiet (no message was sent or received), as well as my testing telnet session (nothing is received). The network doesn't seems to be the issue, as when I write "6:Efelin," to the telnet session, kamailio executes the route event_route[evapi:message-received] Versions: # dpkg -l | grep libev-dev ii libev-dev 1:4.15-3 amd64 # uname -a Linux cgrates 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt25-2+deb8u3 (2016-07-02) x86_64 GNU/Linux # kamailio -V version: kamailio 4.4.2 (x86_64/linux) 384e90 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 384e90 compiled on 12:04:04 Aug 19 2016 with gcc 4.9.2 I wasn't able to find anything that might block the connection. Can anyone point me to, how to debug this issue further? Thanks for any advice. Kind regards Jan Muranyi

3 7

Contact header / bye
by Oliver Roth 19 Aug '16

19 Aug '16

Hi all I do have a problem with getting a propper interconnection to our new carrier.. Situation as following User --> Class5 switch --> Kamailio --> carrier I always get the ip of the class5 switch in the contact. But my carrier needs the kamailios contact ip - because of the ack handling. I replaced in the pstn routing parte the contact by remove_hf("Contact"); append_hf("Contact: <sip:xyz@a.b.c.d>\r\n"); This worked - but the bye did not work any more. Please help me on that topic ... I am searching for a solution for more than 8 weeks now ... KR, Oli

2 1

Running both 4.3 and 4.4 with the same database with permission module
by Daniel Tryba 19 Aug '16

19 Aug '16

I want to upgrade my cluster from 4.3 to 4.4, beginning with the spare for live testing. But since there is a change to the trusted table (which we don't use, only address) this isn't possible without some tricks. There is no override for skipping the version check on any permissions table. I need to share the subscriber/location/address/usr_preferences and db_aliases tables between the 4.3 and 4.4 machines. The only thing I can think of is: -patch the permissions module to accept the old version of trusted -create a 4.4 database and use federated tables to access the 4.3 data Not hard to implement any of these 2, but some other modules have the option to skip these checks, eg: http://www.kamailio.org/docs/modules/4.4.x/modules/auth_db.html#auth_db.p.v… Is there a reason not all modules have this flag (which you could use at your own risk)? Or am I missing something here?

3 4

To/From tags in uac_req()
by Julia Boudniatsky 18 Aug '16

18 Aug '16

Hello, Is it possible to set To/From tags in uac_req() ? We want to implement service like "color ring back tone". *Kamailio uac* *Asterisk* ----INVITE SDP --> <-----200 SDP------ ---------BYE ------> <------- 200 ------- I need to send in BYE the same tags. Kamailio 4.4.2. Thanks, Julia

3 4

Trouble with nat_uac_test()
by Noah Mehl 18 Aug '16

18 Aug '16

Hey all, I have a Kamailio implementation (version: kamailio 4.4.1 (x86_64/solaris) 12b048) as part of the Kazoo platform. I’m having an odd problem with an e2e ACK (after a 200OK after a T38 re-invite), which I’m pretty sure has to do with a Contact header rewrite due to the following (can review attached PCAP for specifics): if (nat_uac_test("3")) { force_rport(); fix_nated_contact(); } I have a couple of questions about this: A flag of 3, I don’t understand. I read the documentation <http://kamailio.org/docs/modules/4.4.x/modules/nathelper.html#nathelper.f.n…>, and there is a 1,2,4,8,etc… So, I’m guessing that a flag of 3, means 1+2? If #1 is true, then "the "received" test is used: address in the “Via” header is compared against source IP address of signaling” I don’t understand this test, as I’m pretty sure that the 200OK that’s getting re-written is properly constructed. I guess I’m trying to understand when one would want to use this test? Thanks! ~Noah

3 3

msilo route error
by Eng Hooda 18 Aug '16

18 Aug '16

Hello All, I am trying to enable msilo offline message delivery module, I used the official technical documentation : https://www.kamailio.org/docs/modules/stable/modules/msilo.html I managed to add load module statement and modparam statement without error. At this point msilo is still not active , it was obvious that I needed to add route statement. I copied the provided route statement , and edited it so will be similar to route functions that are already there. that was the outcome: # Routing to MSILO route[MSILO] { #!ifdef WITH_MSILO if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay(); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } #!endif return; } but after that I restarted kamailio service and got the following errors: root@debian:~# service kamailio restart Job for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. root@debian:~# systemctl status kamailio.service -l ● kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled) Active: failed (Result: exit-code) since Wed 2016-08-17 21:35:41 EET; 4s ago Process: 3693 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255) Main PID: 3470 (code=exited, status=0/SUCCESS) Aug 17 21:35:41 debian kamailio[3693]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 556, column 5: bad command Aug 17 21:35:41 debian kamailio[3693]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 613, column 12: syntax error Aug 17 21:35:41 debian kamailio[3693]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 613, column 12: Aug 17 21:35:41 debian kamailio[3693]: ERROR: bad config file (5 errors) Aug 17 21:35:41 debian kamailio[3693]: WARNING: <core> [ppcfg.c:221]: pp_ifdef_level_check(): different number of preprocessor directives: N(#!IF[N]DEF) - N(#!ENDIF) = 3 Aug 17 21:35:41 debian kamailio[3693]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 17 21:35:41 debian kamailio[3693]: loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Aug 17 21:35:41 debian systemd[1]: kamailio.service: control process exited, code=exited status=255 Aug 17 21:35:41 debian systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Aug 17 21:35:41 debian systemd[1]: Unit kamailio.service entered failed state. line 556 is a parenthesis. { I checked the code for missing parentheses and they are all there. line 613 is t_relay(); also tried to run with #!define WITH_DEBUG , there was no further info to debug the cfg file. Also I tried to mimic some cfg examples like this one : http://telephonynetworks.blogspot.com.eg/2012/08/configuracion-de-kamailio-… but it uses an older version of kamailio 3.3 , and I noticed considerable changes in mpath statement, so that was unsuccessful. kamailio.cfg is attached with and without msilo route. For the experts out there Please Advise . system details : {(OS:Debian8 i386),(Kamailio4.4 with mySQL , UDP only)} Thank you all in advance. Best Regards, Eng Hooda

2 1

RTPEngine
by Jack Stevens 17 Aug '16

17 Aug '16

Hi Guys, Does anyone know how to enable logging for RTPengine Kind Regards From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Jonathan Hunter Sent: 17 August 2016 13:17 To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Core/ signal 11 Segmenation fault on kamailio 4.3.5 Hi guys, We seem to have a regular occurrence where one of the child processes is crashing, and causing a coredump. Please see the output of the corefile; gdb kamailio /core.31568 GNU gdb (GDB) Red Hat Enterprise Linux (7.2-83.el6) Copyright (C) 2010 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/sbin/kamailio...(no debugging symbols found)...done. [New Thread 31568] Reading symbols from /lib64/ld-linux-x86-64.so.2...Reading symbols from /usr/lib /debug/lib64/ld-2.12.so.debug...done. done. Loaded symbols for /lib64/ld-linux-x86-64.so.2 Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f / /etc/kamailio//kamaili'. Program terminated with signal 11, Segmentation fault. #0 0x00000000004deefd in send_fd () (gdb) bt full #0 0x00000000004deefd in send_fd () No symbol table info available. #1 0x00000000004e0bc1 in pp_substdef_add () No symbol table info available. #2 0x00000000004da13c in build_res_buf_from_sip_req () No symbol table info available. #3 0x00007fc99099f4a1 in ?? () No symbol table info available. #4 0x00000000921cd3c8 in ?? () No symbol table info available. #5 0x00007fc9921cd3f8 in ?? () No symbol table info available. #6 0x0000000000415eb0 in do_action () No symbol table info available. #7 0x00007fc9909a2841 in ?? () No symbol table info available. #8 0x00007fc991faf030 in ?? () No symbol table info available. #9 0x0000000000000004 in ?? () No symbol table info available. #10 0x0000000000000001 in ?? () No symbol table info available. #11 0x0000000000000002 in ?? () The last transaction appears to be a REGISTER message from a User-Agent: SIPAUA/0.1.001 which runs on an android device. We are looking to reproduce in lab environment but I just wondered if anyone had seen this on this version? Or can advise on the core dump output? Thanks Jon -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

3 3

Core/ signal 11 Segmenation fault on kamailio 4.3.5
by Jonathan Hunter 17 Aug '16

17 Aug '16

Hi guys, We seem to have a regular occurrence where one of the child processes is crashing, and causing a coredump. Please see the output of the corefile; gdb kamailio /core.31568GNU gdb (GDB) Red Hat Enterprise Linux (7.2-83.el6)Copyright (C) 2010 Free Software Foundation, Inc.License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html>This is free software: you are free to change and redistribute it.There is NO WARRANTY, to the extent permitted by law. Type "show copying"and "show warranty" for details.This GDB was configured as "x86_64-redhat-linux-gnu".For bug reporting instructions, please see:<http://www.gnu.org/software/gdb/bugs/>...Reading symbols from /usr/sbin/kamailio...(no debugging symbols found)...done.[New Thread 31568]Reading symbols from /lib64/ld-linux-x86-64.so.2...Reading symbols from /usr/lib /debug/lib64/ld-2.12.so.debug...done.done.Loaded symbols for /lib64/ld-linux-x86-64.so.2Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f / /etc/kamailio//kamaili'.Program terminated with signal 11, Segmentation fault.#0 0x00000000004deefd in send_fd ()(gdb) bt full#0 0x00000000004deefd in send_fd ()No symbol table info available.#1 0x00000000004e0bc1 in pp_substdef_add ()No symbol table info available.#2 0x00000000004da13c in build_res_buf_from_sip_req ()No symbol table info available.#3 0x00007fc99099f4a1 in ?? ()No symbol table info available.#4 0x00000000921cd3c8 in ?? ()No symbol table info available.#5 0x00007fc9921cd3f8 in ?? ()No symbol table info available.#6 0x0000000000415eb0 in do_action ()No symbol table info available.#7 0x00007fc9909a2841 in ?? ()No symbol table info available.#8 0x00007fc991faf030 in ?? ()No symbol table info available.#9 0x0000000000000004 in ?? ()No symbol table info available.#10 0x0000000000000001 in ?? ()No symbol table info available.#11 0x0000000000000002 in ?? () The last transaction appears to be a REGISTER message from a User-Agent: SIPAUA/0.1.001 which runs on an android device. We are looking to reproduce in lab environment but I just wondered if anyone had seen this on this version? Or can advise on the core dump output? Thanks Jon

3 3

Stress Testing
by Jack Stevens 17 Aug '16

17 Aug '16

Hi Guys, I have been stress testing my Kamailio box but I am unable to get it upto 2000 concurrent calls it starts to fall over at 1300 have you got any ideas on how I can increase the performance of kamilio btw I am also using rtpengine Kind Regards -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

5 7

Kamailio SIP 500
by Rodrigo Moreira 17 Aug '16

17 Aug '16

Hello, You can help me to due with the error: SIP 500 - Server error on UAR select next S-CSCF. What am I doing wrong? Thanks. Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

3 3

t_set_fr() different behavior whether transport is udp or tcp
by Daniel Tryba 16 Aug '16

16 Aug '16

I'm seeing a different behavior of t_set_fr depending on transports. Scenario is that a endpoint has a failover defined in the registrat after 10s (t_set_fr(10000) and handling the locally generated 408 to the failover destination). This works fine when the request and response where delivered over UDP. When the Path is TCP the failover happends after 30s (even when using a different time t_set_fr(20000) so it is not a factor 3 or something like that). Setup: OK, 10s: Orig->UDP->loadbalancer->UDP->registrar->UDP->loadbalancer->TCP->Term Fail, 30s: Orig->UDP->loadbalancer->TCP->registrar->TCP->loadbalancer->TCP->Term Loadbalancer and registrar are kamailio machines (4.3.6). Communication between lb and registrar is based on dispatcher and path modules. 1 sip:registrar:5060;transport=udp 8 0 or via tcp: 1 sip:registrar:5060;transport=tcp 8 0 In the location database of the registrar the difference between the cases is: socket: udp:registrar:5060 or via tcp tcp:registrar:5060 path: <sip:lb@loadbalancer;lr;received=sip:1.2.3.4:5067%3Btransport%3Dtcp> or via tcp <sip:lb@loadbalancer;transport=tcp;lr;received=sip:1.2.3.4:5067%3Btransport%3Dtcp> Looking at debug(=3) nothing happens between the initial INVITE and the local 408 as far as I can see.

2 2

302 - stop sending once ACKed
by jaflong jaflong 16 Aug '16

16 Aug '16

I have kamailio as a redirect server. It responds with 302 to invites. The problem I am having is it continues to send the 302 multiple times. How can i get it to stop sending once it gets the first ACK

2 1

Is ACK required in a re-invite?
by Jay Li 15 Aug '16

15 Aug '16

Dear All, I just have a basic SIP question on re-invite. I wonder if "ACK" is required in a re-invite scenario, like re-invite -> 200 OK -> ACK. What I've seen is for some reason, my PSTN doesn't send a "ACK" back to me after my 200 OK response to their re-invite. I don't know if not sending an ACK in a re-invite is an allowed behavior. Thanks. Regards,Jay

2 3

kamailio TLS mod_register failed
by wanmon wanmon 15 Aug '16

15 Aug '16

4 14

Via in locally generated PUBLISH (using different presence server address)
by Asgaroth 15 Aug '16

15 Aug '16

Hi All, I have a configuration where we run a multi-homed proxy/loadbalancer and a seperate internal presence server. We then use pua/pua-dialoginfo on the proxies to publish dialog state to the internal presence server. The proxies listen on udp:external_ip:5062 and udp:internal_ip1:5060. The presence server listens on udp:internal_ip2:5060 on a seperate server. The problem I am coming accross is that when kamailio generates a PUBLISH message for some dialogs it creates the via header with address external:ip:5062, however the pua module is configured to send to the presence server address on internal_ip2:5060. I have a route which forces the sending socket ($fs) to be udp:internal_ip1:5060. What happens then is that the presence server tries to send the 200 reply to the PUBLISH message back to internal_ip1:5062 (when it should be internal_ip1:5060), I'm thinking this is because the Via header has host as external_ip:5062. Is it possible, from event-route[tm:local-request] to rewrite the via header so that it reflects internal_ip1:5060 instead of external_ip:5062? I did try the following snippet from event-route[tm:local-request], but I got module parse errors on the Via header: if (is_method("PUBLISH")) { subst_hf("Via", "/ 1\.2\.3\.[0-9]+:5062;/ $sel(cfg_get.socket.internal_ip);/", "f"); } [7370]: ERROR: <core> [parser/parse_via.c:2607]: parse_via(): ERROR:parse_via on <1> state 5 (default) [7370]: ERROR: <core> [parser/parse_via.c:2708]: parse_via(): ERROR: parse_via on: <SIP/2.0/UDP external_ip:5062;branch=z9hG4bK96eb.0f014ae4000000000000000000000000.0#015#012 internal_ip1; [7370]: ERROR: <core> [parser/parse_via.c:2712]: parse_via(): ERROR: parse_via parse error, parsed so far:<SIP/2.0/UDP external_ip:5062;branch=z9hG4bK96eb.0f014ae4000000000000000000000000.0#015#012 > [7370]: ERROR: <core> [parser/msg_parser.c:125]: get_hdr_field(): ERROR: get_hdr_field: bad via [7370]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [Via: SIP/2.0/UDP 1] [7370]: ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<PUBLISH sip:user@domain;transport=UDP SIP/2.0#015#012Via: SIP/2.0/UDP external_ip:5062;branch=z9hG4bK96eb.0f014ae4000000000000000000000000.0#015#012 internal_ip1;To: <sip:user@domain;transport=UDP>#015#012From: <sip:user@domain;transport=UDP>;tag=3e8fdaca277026bf3e6d75c5e5313096-f162#015#012CSeq: 10 PUBLISH#015#012Call-ID: 5634294b786134b3-7370@external_ip#015#012Content-Length: 692#015#012User-Agent: kamailio (bfievklb04)#015#012Max-Forwards: 70#015#012Event: dialog#015#012Expires: 10901#015#012Content-Type: application/dialog-info+xml#015#012#015#012<?xml version="1.0"?>#012<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:user@domain;transport=UDP">#012 <dialog id="mvqXXaAr24DHdfsffYKZeA.." call-id="mvqXXaAr24DHdfsffYKZeA.." direction="initiator">#012 <state>Trying</state>#012 <remote>#012 <identity>sip:1234567@domain;transport=UDP</identity>#012 <target uri="sip:1234567@domain;transport=UDP"/>#012 </remote>#012 <local>#012 <identity>sip:user@domain;transport=UDP</identity>#012 <target uri="sip:user@domain;transport=UDP"/>#012 </local>#012 </dialog>#012</dialog-info>#012> [7370]: ERROR: <core> [msg_translator.c:3057]: build_sip_msg_from_buf(): parsing failed [7370]: ERROR: tm [uac.c:163]: uac_refresh_hdr_shortcuts(): failed to parse msg buffer [7370]: ERROR: tm [uac.c:487]: t_uac_prepare(): failed to refresh header shortcuts [7370]: ERROR: pua [send_publish.c:700]: send_publish(): in t_request tm module function [7370]: ERROR: pua_dialoginfo [dialog_publish.c:358]: dialog_publish(): while sending publish To me, it looks like the replace section of the subst_hf sub expression is being appended to the end of the via header instead of replacing the address section found. I have sanitized the above output to not show internal details, but if you need it, I can send it on to you privately. Am I going about this issue in the correct way, and, if so, where am I going wrong with the subst_hf snippet above? Kamailio version is 4.3.6. Any pointers/tips/trciks are greatly appreciated. Thanks

2 5

Installation question on ubuntu xenial / undefined symbol: EVP_sha1
by Christian Lox 15 Aug '16

15 Aug '16

Hi. I am trying to install kamailio 4.4.2 on ubuntu xenial, for use with kazoo. Using source and compiling just went fine, but /usr/local/sbin/kamailio -c throws ERROR: <core> [sr_module.c:582]: load_module(): could not open module </usr/local/lib64/kamailio/modules/outbound.so>: /usr/local/lib64/kamailio/modules/outbound.so: undefined symbol: EVP_sha1 any hint appreciated! Thanks, Christian

2 1

Re: [SR-Users] undefined symbol: EVP_sha1 when using outbound module
by Christian Lox 15 Aug '16

15 Aug '16

Hi. Same problem here; undefined symbol : EVP_sha1 in outbound.so pkg-config was available when compiled from source. Tried installing several dev packages and so on but always the same error. How do i link it against libssl/libcrypto? Thanks, Christian

2 2

access header variables in failure route
by jaflong jaflong 13 Aug '16

13 Aug '16

On an 302 message caught in failure_route Is it possible to get to access $hdr(xxx) of the 302. When I do this $hdr(xxx) it is null.

2 1

Struggling at 1000 Concurrent Calls
by Jack Stevens 12 Aug '16

12 Aug '16

Hi Guys, I have built an SBC using kamailio with the dispatcher module and rtpengine and when I run a sip test I find it struggles to go past 1000 calls at 20cps would really like some help on how to resolve this. As I am very new to this I am stuggeling to work out where to start from. I have found when running this sort of load the debian box drops pings any ideas Kind Regards -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

3 3

CDP Error S-CSCF Kamailio
by Rodrigo Moreira 12 Aug '16

12 Aug '16

Hello Carsten Bock, I got to advance in the event REGISTER, but now face the following problem: When I type the command: */etc/init.d/kamailio restart* Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: cdp [diameter_peer.c:428]: diameter_peer_destroy(): destroy_diameter_peer(): Terminating all children... Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: cdp [diameter_peer.c:447]: diameter_peer_destroy(): destroy_diameter_peer(): All processes terminated....aning up. Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: cdp [worker.c:140]: worker_destroy(): Unlocking workers waiting on empty queue... Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: cdp [worker.c:143]: worker_destroy(): Unlocking workers waiting on full queue... Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: CRITICAL: cdp [diameter_peer.c:487]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: cdp [mod.c:244]: cdp_exit(): ... CDiameterPeer child stoped Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 10 20:40:52 scscf systemd[1]: kamailio.service: control process exited, code=exited status=255 Aug 10 20:40:52 scscf systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Aug 10 20:40:52 scscf systemd[1]: Unit kamailio.service entered failed state. I can not start the Kamailio, I used the examples codes available in the repository (GIT). You suggest something I may be doing wrong? Can you help me? I attached my config files. Thank you very much. Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

4 4

Kamailio 4.4 Dispatcher on Debian 8 with MongoDB
by Jonathan Hunter 12 Aug '16

12 Aug '16

Hi Guys, I am integrating kamailio with MongoDB, and it works well with subscriber/location tables. However when using the dispatcher module I am getting the following error even though the IP Im trying to get to is routeable; /usr/local/kamailio-4.4/sbin/kamailio[2998]: ERROR: dispatcher [dispatch.c:1403]: ds_get_index(): destination set [1] not foundAug 11 17:01:47 ip-172-31-43-8 /usr/local/kamailio-4.4/sbin/kamailio[2998]: ERROR: dispatcher [dispatch.c:1832]: ds_select_dst_limit(): destination set [1] not found I created the table as below; db.getCollection("version").insert({table_name: "dispatcher", table_version: NumberInt(4) }) And the entry Im routing to is; db.dispatcher.find().pretty(){ "_id" : ObjectId("57acadc81e7e803fa565bce3"), "setid" : "1", "destination" : "sip:1.1.1.1:5080;transport=tcp", "flags" : "0", "priority" : "100", "attrs" : "name=labfS", "description" : "FreeSWITCH1"} Is there something I am missing ? Has anyone else got it working with Mongo? Thanks Jon

2 1

Kamailio CDP Error - SCSCF
by Rodrigo Moreira 12 Aug '16

12 Aug '16

Hello, I can not make the S-CSCF module work, is giving CDP error. CRITICAL: cdp [diameter_peer.c: 485]: diameter_peer_destroy (): destroy_diameter_peer (): Bye Bye from C Diameter Peer test I am using the examples available in the repository still occur errors that prevent Kamailio to boot. Can someone help me? Someone has the kamailio.cfg, scscf.cfg and scscf.xml files to share? Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

1 0

Process redirected message
by jaflong jaflong 11 Aug '16

11 Aug '16

kamailio A relays to kamailio B kamailio B send 300 redirect back to kamailio A In this setup how do I catch the redirect response back from kamailio B What conditions should I do to catch and further process the redirect message Regards

2 2

Re: [SR-Users] kamailio starts looping invites(some) back to itself.
by SamyGo 10 Aug '16

10 Aug '16

Hi Aqs, How about putting a loop protection of some sort additionally like $si != myself. Is there any chance the API http call is getting skipped to relay w/o getting any reply back and hence looping back to itself ? Regards, Sammy On Jul 22, 2016 14:17, "Aqs Younas" <aqsyounas(a)gmail.com> wrote: Hi, I am using http_async_client and dispatcher modules of Kamailio. Here is scenario! http_async_client makes calls to an API to get groupid upon which dispatcher sends calls. I am using SIPp to generate calls with 100 cps. I see some calls being failed with an error cause, 483 Too Many Hops. Taking the trace shows Kamailio routing some calls to itself. I think this is not a configuration issue since all calls are same(follow same logic) and some calls get failed. Any pointer is much appreciated. Best Regards. _______________________________________________ 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

5 13

Problem receiving large payloads from JSONRPC-S
by Alex Balashov 10 Aug '16

10 Aug '16

Hi, I'm requesting 'dlg.list' from JSONRPC-S via XHTTP on a rather busy server, i.e. event_route[xhttp:request] { xlog("L_INFO", 'action=ER-XHTTP-REQUEST | ret=LOG_HTTP_REQ | data={"method": "$rm", "path": "$hu", "source": "$si:$sp"}'); # Digest challenge omitted. if($hu =~ "^/rpc") { xlog("L_INFO", 'action=ER-XHTTP-REQUEST | ret=JSON_RPC_REQUEST | data={"path": "$hu"}'); jsonrpc_dispatch(); } else { xhttp_reply("404", "Not Found", "text/html", ""); } } # curl --digest -v \ -X POST -H 'Content-Type: application/json' \ -H 'Call-ID: abc12' \ -d '{"jsonrpc": "2.0", "method": "dlg.list", "id": "0deadb33f"}' \ 'http://sipus_adm:raSheinmet4@10.0.62.74:5060/csrp_rpc/' and finding that when there's more than a hundred or so dialogs in play, Kamailio abruptly closes the connection in the middle of returning the output: ----- "tag": "AF5DA9B4-6D5", "contact": "sip:14045551212@xxx.xxx.xxx.xxx:5060", "cseq": "0", "route_set": "", "socket": "udp:xxx.xxx.xxx.xxx:5060" }, * transfer closed with 151674 bytes remaining to read * Closing connection #0 curl: (18) transfer closed with 151674 bytes remaining to read "profiles": ["total",[root@proxy kamailio]# ------ I can't seem to figure out what parameters regulate this. I'm using the following TCP settings: tcp_async=yes tcp_connect_timeout=5 tcp_crlf_ping=no tcp_keepalive=yes tcp_keepidle=30 tcp_linger2=10 tcp_max_connections=4096 tcp_send_timeout=3 tcp_accept_no_cl=yes tcp_wq_blk_size=65535 tcp_wq_max=20971520 But this doesn't seem to have had any effect. Any help appreciated! Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 6

Duplicate ACC records in db when using db_cluster
by Jan Hazenberg 09 Aug '16

09 Aug '16

Hi, I have a issue with the db_cluster module. All works fine but when i enable ACC records to the database i get duplicate entry's in the acc table. I have 2 nodes enabled: # ----- db_cluster params ----- modparam("db_cluster", "connection", "con1=>mysql://kamailio:password@10.1.1.1/kamailio") modparam("db_cluster", "connection", "con2=>mysql://kamailio:password@10.1.1.2/kamailio") modparam("db_cluster", "cluster", "cls1=>con1=9s8p;con2=9s8p") When i remove one node from the pool the acc records look fine. Could this be a bug? Or should i bypass the db_cluster for accounting? I'm running kamailio 4.4.2. Thanks, Jan Hazenberg

3 5

When Not Found
by Nahum Nir 09 Aug '16

09 Aug '16

Hi List, When sending an invite and the calle not found or responding with an ACK I want my script do some costume work. When can I do it (using regular script)? Thanks, Nir

3 2

Kamailio Regiser
by Rodrigo Moreira 09 Aug '16

09 Aug '16

Hello everyone, I am deploying an IMS core, however I can not register the user (Bob). The entities are all working PCSCF, ICSCF SCSCF and HSS, I suppose. However after utiliizar Wireshark is returned the following error: 604 Does not exist anywhere - HSS User Unknow 400 Bad Request - P-visited-Network-ID header missing. What am I doing wrong? I ask for help, the great community. Or methods to better understand what is causing the problem. Is it DNS? Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

open websocket connction from kamailio
by Uri Shacked 08 Aug '16

08 Aug '16

Hi, Is it possible to open a websocket connection from kamailio to another websocket server. not something based on sip, just a websocket to send to the websocket server messages from cfg file. (like http_query) ?? Uri

1 0

Feature Request CarrierRoute To and Contact header Strip
by Yasin CANER 08 Aug '16

08 Aug '16

2 1

app_python access to message body content
by Anthony Joseph Messina 08 Aug '16

08 Aug '16

I'm trying to play around a bit with app_python using the examples in the source. I'd like to access the content of the body of the SIP message, but can't figure out how to do that. I can get headers, method, etc. How does one access body content with app_python? Thanks. -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

3 4

TCP message parsing & connection close issue
by Alex Balashov 08 Aug '16

08 Aug '16

Hello, I've got an odd issue where: 1. TCP client makes call out through Kamailio. 2. Call is answered, TCP client sends e2e ACK. 3. Parser chokes on ACK, closes TCP connection (SYN+FIN). 4. Subsequent messages from the client, immediately following the ACK (about 240 ms later), come through because the client is halfway around the world - about 250 ms away - and is not yet aware the connection has been closed. Otherwise, it would presumably try to re-establish the connection before sending anything. Anyway, messages from the parser concerning the ACK are: --- Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]: INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]: ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<0#015#012a=rtcp:4007 IN IP4 1.1.1.1:1071#015#012a=sendrecv#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96 telephone-event/8000#015#012a=fmtp:96 0-16#015#012ACK sip:*98*0b3ff883-a89d-4317-b758-8979682f5357@10.0.2.53:5060;transport=udp SIP/2.0#015#012Via: SIP/2.0/TCP 1.1.1.1:61979;branch=z9hG4bKPjc15bf9fdded940cab7f859634b9f036e;alias#015#012Max-Forwards: 70#015#012From: sip:4b57be8d-267d-48c4-aa82-0f3a269d74cb@sip.evaristesys.com;tag=dd1d7fd0344d47b7b22c4017091df17a#015#012To: sip:*98*0b3ff883-a89d-4317-b758-8979682f5357@sip.evaristesys.com;tag=2rZDKcBFmU65S#015#012Call-ID: 5ba5a84451074159aaaa931eb1f98fe5#015#012CSeq: 3071 ACK#015#012Route: <sip:2.2.2.2;transport=tcp;lr;r2=on;ftag=dd1d7fd0344d47b7b22c4017091df17a;rtp_set_id=1>#015#012Route: <sip:10.0.2.119;lr;r2=on;ftag=dd1d7fd0344d47b7b22c4017091df17a;rtp_set_id=1>#015#012Content-Length: 0#015#012#015#012> Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]: ERROR: <core> [receive.c:173]: receive_msg(): core parsing of SIP message failed (1.1.1.1:61979/2) --- That doesn't look like a clean message buffer to me. The message should start here: ACK sip:*98*0b3ff883-a89d-4317-b758-8979682f5357@10.0.2.53:5060;transport=udp SIP/2.0#015#012Via: SIP/2.0/TCP ... But instead it's got these SDP attributes prepended: 0#015#012a=rtcp:4007 IN IP4 1.1.1.1:1071#015#012a=sendrecv#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96 telephone-event/8000#015#012a=fmtp:96 0-16#015#012 These SDP attribute appear to have fallen off the back of the SDP offer in the initial INVITE: v=0 o=- 3679365582 3679365582 IN IP4 10.0.0.200 s=pjmedia t=0 0 m=audio 4006 RTP/AVP 0 8 96 c=IN IP4 1.1.1.1 The relationship between the 4006 RTP port and the 4007 RTCP port suggests that these two things belong together. This is on kamailio 4.4.0 (x86_64/linux) d4f23c. Any help would be greatly appreciated! The issue is consistently reproducible and this happens every time. Cheers, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 6

Synonyms User Name
by Nahum Nir 07 Aug '16

07 Aug '16

Hello List, I wonder if there is a way to define synonyms. For example, I there is a person by the name of John Smith with extention 5555 I would like to have a user john_smith that will also be available if some INVITE 5555. Thanks, Nahum

2 3

Prepaid system using dialog and sqlops
by Infinicalls Infinicalls 06 Aug '16

06 Aug '16

Hi, I am still struggling to get a prepaid system for my service. After installing SIREMIS, I am able to get the CDRs updated. But even after adding the cron settings, the call duration doesn't get updated. And also in http://lists.sip-router.org/pipermail/sr-users/2012-January/071517.html it has been mentioned that "If it is the case of 1 call per user at a time, then practically can be done from kamailio config only. When call starts, compute the max duration for that call based on caller/destination (e.g., call a stored procedure via sqlops) and set the dialog timeout to that duration. When call ends (BYE in main route or timeout route) update the credit accordingly." Can somebody let me know if my understanding is correct. 1) Add a field called "credits" in the database. And topup that field. 2) First compute max permitted duration for that user by multiplying the cost for number prefix (say +91) and pulse. 3) And finally add this value to timeout of dialog. modparam("dialog", "default_timeout", 100000) $dlg_ctx(timeout_bye) = 1; Will this work? Any help would be appreciated. regards Ganesh Kumar

3 4

segfault when using sipt modifications
by david 05 Aug '16

05 Aug '16

hello all i'm trying to add a ISUP multipart in a regular call using the textops module, and despite seeing one previous issue int he mail list, i'm finding some issue here i'm using this commands msg_apply_changes(); set_body_multipart(,"2123894789_1257887457"); msg_apply_changes(); append_body_part("\x01\x12\x49\x00\x0a\x03\x02\x0a\x08\x84\x90\x33 \x41\x72\x17\x00\x06\x0a\x08\x04\x13\x93\x70\x21\x73\x23\x10\x08\x01\x00 \x00","application/isup;version=itu-t92+","signal;handling=optional"); msg_apply_changes(); sipt_destination($rU, 31, 4); sipt_set_calling($fU, 4, 0, 3); msg_apply_changes(); before the record_route command, and when using sipt_destination($rU, 31, 4); i'm getting asegfault here you have the logs could you please give me any clue about something i'm missing or doing wrong? thanks alot and regards david 5(24552) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: 5(24552) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:34666988297@79.170.71.132:5060> 5(24552) DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK7ma5t4FKS0SKa>; state=16 5(24552) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 5(24552) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 5(24552) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 5(24552) DEBUG: <core> [receive.c:178]: receive_msg(): After parse_msg... 5(24552) DEBUG: <core> [receive.c:221]: receive_msg(): preparing to run routing scripts... 5(24552) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 65 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 5(24552) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:34666988297@79.170.71.132:5060;user=phone] 5(24552) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:34666988297@79.170.71.132:5060;user=phone> ] 5(24552) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <94568362> <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=104 5(24552) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=5jBBFae4D90rg 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 5(24552) DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 5(24552) DEBUG: siputils [checks.c:97]: has_totag(): no totag 5(24552) DEBUG: <core> [socket_info.c:564]: grep_sock_info(): checking if host==us: 13==13 && [79.170.71.132] == [79.170.71.132] 5(24552) DEBUG: <core> [socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 5(24552) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff 5(24552) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=46859, isACK=0 5(24552) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 5(24552) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 5(24552) DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) 5(24552) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=1 , global msg id=1 , T on entrance=(nil) 5(24552) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=46859, isACK=0 5(24552) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 5(24552) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 5(24552) DEBUG: tm [t_hooks.c:358]: run_reqin_callbacks_internal(): DBG: trans=0x7fb3fb0c64e0, callback type 1, id 0 entered 5(24552) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): MD5 calculated: 5ea9736b3bdcf90a4c3043a5a492b31f 5(24552) DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0x7fb404252ef8 5(24552) INFO: Theseus-Test Call 69672589-67f5-4edd-8844-f6ad2ee5b208 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: Call-ID = 69672589-67f5-4edd-8844-f6ad2ee5b208 5(24552) DEBUG: app_lua [app_lua_api.c:643]: app_lua_run(): reload deactivated 5(24552) DEBUG: app_lua [app_lua_api.c:644]: app_lua_run(): executing Lua function: [[allow_src_uri]] 5(24552) DEBUG: app_lua [app_lua_api.c:645]: app_lua_run(): lua top index is: 4 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $avp(s:call_obj) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:call_obj)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $ci 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$ci] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $var(authorized) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$var(authorized)] found in cache 5(24552) INFO: Theseus-Test Call 69672589-67f5-4edd-8844-f6ad2ee5b208 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: Method INVITE from 79.170.64.130:5080 authorized 5(24552) INFO: Theseus-Test Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: Object 58825 assigned to Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208 5(24552) INFO: Theseus-Test Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: The call has been sent to RG 106406 5(24552) DEBUG: app_lua [app_lua_api.c:643]: app_lua_run(): reload deactivated 5(24552) DEBUG: app_lua [app_lua_api.c:644]: app_lua_run(): executing Lua function: [[get_rg_info]] 5(24552) DEBUG: app_lua [app_lua_api.c:645]: app_lua_run(): lua top index is: 4 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $ci 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$ci] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $avp(s:call_obj) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:call_obj)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $shv(flag_list) 5(24552) DEBUG: <core> [pvapi.c:293]: pv_cache_add(): pvar [$shv(flag_list)] added in cache 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG 106406 attributes: 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 description = PBX--NOC-TEST 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 admin_status = 1 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 cir_direction = 2 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 ani_posttranslation = 0 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 ani_posttranslation_table = 0 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 posttranslation = 1 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 posttranslation_table = 106384 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 pound = 2 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 gw_port = 5060 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 custom_flags = None 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:description) 5(24552) DEBUG: <core> [usr_avp.c:882]: parse_avp_ident(): Parsing 's:description' 5(24552) DEBUG: <core> [pvapi.c:293]: pv_cache_add(): pvar [$avp(s:description)] added in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:admin_status) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:admin_status)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:cir_direction) 5(24552) DEBUG: <core> [usr_avp.c:882]: parse_avp_ident(): Parsing 's:cir_direction' 5(24552) DEBUG: <core> [pvapi.c:293]: pv_cache_add(): pvar [$avp(s:cir_direction)] added in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:ani_posttranslation_table) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:ani_posttranslation_table)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:posttranslation_table) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:posttranslation_table)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:pound) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:pound)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:gw_ip) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:gw_ip)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:gw_port) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:gw_port)] found in cache 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: Custom flags = None 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $shv(custom_flag_list) 5(24552) DEBUG: <core> [pvapi.c:293]: pv_cache_add(): pvar [$shv(custom_flag_list)] added in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $shv(custom_flag_list_branch) 5(24552) DEBUG: <core> [pvapi.c:293]: pv_cache_add(): pvar [$shv(custom_flag_list_branch)] added in cache 5(24552) DEBUG: textopsx [textopsx.c:231]: msg_apply_changes_f(): SIP message content updated - reparsing 5(24552) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: 5(24552) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:34666988297@79.170.71.132:5060> 5(24552) DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK7ma5t4FKS0SKa>; state=16 5(24552) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 5(24552) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 5(24552) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 5(24552) DEBUG: textops [textops.c:1577]: set_multibody_helper(): delimiter<21>:[2123894789_1257887457] 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 5(24552) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:34666988297@79.170.71.132:5060;user=phone] 5(24552) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:34666988297@79.170.71.132:5060;user=phone> ] 5(24552) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <94568362> <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=104 5(24552) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 5(24552) DEBUG: textops [textops.c:1486]: generate_boundary(): adding final CRLF+CRLF 5(24552) DEBUG: textops [textops.c:1714]: set_multibody_helper(): content-type<48>:[multipart/mixed;boundary="2123894789_1257887457"] 5(24552) DEBUG: textops [textops.c:1768]: set_multibody_helper(): set flag FL_BODY_MULTIPART 5(24552) INFO: <core> [msg_translator.c:1692]: get_boundary(): Content-Type hdr has no params <application/sdp> 5(24552) WARNING: <core> [msg_translator.c:1958]: build_req_buf_from_sip_req(): check_boundaries error 5(24552) DEBUG: <core> [msg_translator.c:422]: clen_builder(): content-length: 189 (189) 5(24552) DEBUG: textopsx [textopsx.c:231]: msg_apply_changes_f(): SIP message content updated - reparsing 5(24552) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: 5(24552) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:34666988297@79.170.71.132:5060> 5(24552) DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK7ma5t4FKS0SKa>; state=16 5(24552) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 5(24552) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 5(24552) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 5(24552) ERROR: <core> [lvalue.c:345]: lval_pvar_assign(): non existing right pvar 5(24552) ERROR: <core> [lvalue.c:405]: lval_assign(): assignment failed at pos: (390,18-390,44) 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 5(24552) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:34666988297@79.170.71.132:5060;user=phone] 5(24552) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:34666988297@79.170.71.132:5060;user=phone> ] 5(24552) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <94568362> <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=189 5(24552) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 5(24552) DEBUG: textops [textops.c:1486]: generate_boundary(): adding final CRLF+CRLF 5(24552) DEBUG: <core> [msg_translator.c:1719]: get_boundary(): boundary is <--2123894789_1257887457> 5(24552) DEBUG: <core> [msg_translator.c:1839]: check_boundaries(): last bondary without -- at the end 5(24552) DEBUG: <core> [msg_translator.c:1639]: replace_body(): old size body[189] actual[319] 5(24552) DEBUG: <core> [msg_translator.c:422]: clen_builder(): content-length: 319 (319) 5(24552) DEBUG: textopsx [textopsx.c:231]: msg_apply_changes_f(): SIP message content updated - reparsing 5(24552) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: 5(24552) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:34666988297@79.170.71.132:5060> 5(24552) DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK7ma5t4FKS0SKa>; state=16 5(24552) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 5(24552) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 5(24552) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 5(24552) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:34666988297@79.170.71.132:5060;user=phone] 5(24552) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:34666988297@79.170.71.132:5060;user=phone> ] 5(24552) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <94568362> <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=319 5(24552) DEBUG: <core> [parser/parse_body.c:87]: get_boundary_param(): boundary is "2123894789_1257887457" 5(24552) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 5(24552) DEBUG: <core> [parser/parse_body.c:87]: get_boundary_param(): boundary is "2123894789_1257887457" 5(24552) CRITICAL: <core> [data_lump.c:297]: del_lump(): offset + len exceeds message size (1176 + 51 > 1201) 35(24582) CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 17 35(24582) DEBUG: <core> [tcp_main.c:3448]: handle_ser_child(): dead child 5, pid 24552 (shutting down?) 35(24582) DEBUG: <core> [io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa2ae40, 17, -1, 0x0) fd_no=39 called 0(24547) ALERT: <core> [main.c:739]: handle_sigs(): child process 24552 exited by a signal 6 0(24547) ALERT: <core> [main.c:742]: handle_sigs(): core was not generated 0(24547) INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD 0(24547) DEBUG: <core> [main.c:756]: handle_sigs(): terminating due to SIGCHLD 34(24581) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 33(24580) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 32(24579) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 31(24578) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 30(24577) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 29(24576) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 28(24575) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 27(24574) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 26(24573) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 25(24572) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 24(24571) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 23(24570) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 22(24569) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 21(24568) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 20(24567) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 19(24566) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 18(24565) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 17(24564) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 16(24563) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 15(24562) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 14(24561) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 13(24560) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 12(24559) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 11(24558) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 10(24557) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 9(24556) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 8(24555) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 7(24554) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 6(24553) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 4(24551) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 3(24550) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 2(24549) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 1(24548) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 35(24582) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 0(24547) DEBUG: rtpengine [rtpengine.c:2014]: mod_destroy(): rtpengine_hash_table_destroy() success! 0(24547) DEBUG: db_text [dbtext.c:110]: destroy(): destroy ... 0(24547) DEBUG: tm [t_funcs.c:86]: tm_shutdown(): DEBUG: tm_shutdown : start 0(24547) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): DEBUG: tm_shutdown : emptying hash table 0(24547) DEBUG: tm [h_table.c:127]: free_cell_helper(): freeing transaction 0x7fb3fb0c64e0 from h_table.c:449 0(24547) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): DEBUG: tm_shutdown : removing semaphores 0(24547) DEBUG: tm [t_funcs.c:93]: tm_shutdown(): DEBUG: tm_shutdown : destroying tmcb lists 0(24547) DEBUG: tm [t_funcs.c:96]: tm_shutdown(): DEBUG: tm_shutdown : done 0(24547) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 0(24547) DEBUG: <core> [mem/shm.c:174]: shm_core_lock_destroy(): destroying the shared memory lock 0(24547) DEBUG: <core> [mem/shm.c:266]: shm_destroy_manager(): destroying memory manager: f_malloc 0(24547) DEBUG: <core> [mem/pkg.c:89]: pkg_destroy_manager(): destroying memory manager: f_malloc

3 13

e2e ACK. Kamailio. Cantact. Route.
by pavel＠eremina.net 05 Aug '16

05 Aug '16

Hello, I have this scheme: SIP Carrier (external IP 22.203) < > (local ip 1.21) mikrotik < > (local ip 1.21) Kamailio - (local ip 1.15) Asterisk. Carrier does not know what is Route header. that is why Carrier send ACK directly to 1.15 and fail. I change Contact at Kamailio to 1.21 then ACK is coming to Kamailio. But i think it's wrong way... How can i get ACK from CArrier to Kamalio and route it in right way? Thanks

1 0

e2e ACK
by Pavel Eremin 05 Aug '16

05 Aug '16

Hello, I have this scheme: SIP Carrier (external IP 22.203) < > (local ip 1.21) mikrotik < > (local ip 1.21) Kamailio - (local ip 1.15) Asterisk. Carrier does not know what is Route header. that is why Carrier send ACK directly to 1.15 and fail. I change Contact at Kamailio to 1.21 then ACK is coming to Kamailio. But i think it's wrong way... How can i get ACK from CArrier to Kamalio and route it in right way? Thanks

1 0

Re: [SR-Users] Textops and Multipart Body - adding ISUP segfaults Kamailio
by david 05 Aug '16

05 Aug '16

Hello all i'm having the same error on kam 4.4 when doing set_body_multipart(); msg_apply_changes(); i see in the logs 5(17797) DEBUG: textops [textops.c:1577]: set_multibody_helper(): delimiter<17>:[unique-boundary-1] 5(17797) DEBUG: textops [textops.c:1486]: generate_boundary(): adding final CRLF+CRLF 5(17797) DEBUG: textops [textops.c:1714]: set_multibody_helper(): content-type<44>:[multipart/mixed;boundary="unique-boundary-1"] 5(17797) DEBUG: textops [textops.c:1768]: set_multibody_helper(): set flag FL_BODY_MULTIPART 5(17797) INFO: <core> [msg_translator.c:1692]: get_boundary(): Content-Type hdr has no params <application/sdp> 5(17797) WARNING: <core> [msg_translator.c:1958]: build_req_buf_from_sip_req(): check_boundaries error 5(17797) DEBUG: <core> [msg_translator.c:422]: clen_builder(): content-length: 191 (191) the original INVITE is like .... Content-Type: application/sdp. Content-Disposition: session. Content-Length: 104. P-Asserted-Identity: "10707334" <sip:10707334@3.3.3.3>. . v=0. o=user1 53655765 23536 IN IP4 79.170.68.171. s=-. c=IN IP4 2.2.2.2. t=0 0. m=audio 6001 RTP/AVP 8. .... do you have any idea why these messages appear thanks a lot and regards david

2 1

Re: [SR-Users] Kamailio on multiple interface
by Daniel-Constantin Mierla 05 Aug '16

05 Aug '16

Hello, if you know the ip address associated with the network interfaces, you can use keyword dst_ip to check which of the ip addresses was used to receive the packet (or use $Ri instead of dst_ip). Cheers, Daniel On 05/08/16 08:41, Marino Mileti wrote: > > Hi guys, > > > > I've my Kamailio box that works great J > > > > Now I would like to enable wifi on the board (in AP mode) in order to > allow mobile devices to connect and receive and place calls. > > Is it possible? Is it possible to configure Kama like a "gateway" > between two network cards? > > > > I've find a guide regarding gateway between IPv4 and IPV6 but i can't > use "af" to discriminate message between eth0 or wlan0. Is there a > command to detect if a message is received from a specific interface card? > > > > Any help or any cfg example is very appreciate J > > > > /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

1 0

Kamailio on multiple interface
by Marino Mileti 05 Aug '16

05 Aug '16

Hi guys, I've my Kamailio box that works great J Now I would like to enable wifi on the board (in AP mode) in order to allow mobile devices to connect and receive and place calls. Is it possible? Is it possible to configure Kama like a "gateway" between two network cards? I've find a guide regarding gateway between IPv4 and IPV6 but i can't use "af" to discriminate message between eth0 or wlan0. Is there a command to detect if a message is received from a specific interface card? Any help or any cfg example is very appreciate J 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

1 0

Kamailio RTP Engine
by Jack Stevens 05 Aug '16

05 Aug '16

Hi Guys, Looking for some help on configuring RTP Engine with Dispatcher on my Kamailio Box Please see this paste for the kamailio configuration = http://pastebin.com/yZzSEJmK When I run a call the dispatcher does it job perfectly but the rtpengine is not been used does anyone know why -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

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Kamailio 4.4 wont start on CentOS 7 using script
by Fabian Borot 04 Aug '16

04 Aug '16

Hello I installed/compiled version 4.4 on CentOS 7. When I start the server like this: "./kamctl start" it works but when I try to use the script: "/etc/init.d/kamailio" it fails [root@ar-mia-proxy-01 sbin]# /etc/init.d/kamailio start Starting kamailio (via systemctl): Job for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. [FAILED] [root@ar-mia-proxy-01 sbin]# when running the "status" I see this: [root@ar-mia-proxy-01 sbin]# /etc/init.d/kamailio status â kamailio.service - LSB: Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Thu 2016-08-04 15:25:59 GMT; 14s ago Docs: man:systemd-sysv-generator(8) Process: 5556 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Aug 04 15:25:59 ar-mia-proxy-01.dom.com systemd[1]: Starting LSB: Kamailio (OpenSER) - the Open Source SIP Server... Aug 04 15:25:59 ar-mia-proxy-01.dom.com kamailio[5556]: Starting kamailio: Aug 04 15:25:59 ar-mia-proxy-01.dom.com systemd[1]: kamailio.service: control process exited, code=exited status=1 Aug 04 15:25:59 ar-mia-proxy-01.dom.com systemd[1]: Failed to start LSB: Kamailio (OpenSER) - the Open Source SIP Server. Aug 04 15:25:59 ar-mia-proxy-01.dom.com systemd[1]: Unit kamailio.service entered failed state. Aug 04 15:25:59 ar-mia-proxy-01.dom.com systemd[1]: kamailio.service failed. [root@ar-mia-proxy-01 sbin]# I welcome all ideas, thank you in advance

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Kamailio and RTPProxy in-front of freeswitch
by Achintha 04 Aug '16

04 Aug '16

hi all, i have configured kamailio and RTPProxy on one ubuntu server then freeswitch on another server. all servers installed on digital ocean. i was tried to build following setup, SIP_Call------> kamailio+RTPPROXY-------> freeswitch put following configurations on kamailio. modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") modparam("rtpproxy", "rtpproxy_retr", 2) modparam("rtpproxy", "ice_candidate_priority_avp", "$avp(ice_priority)") mhomed=1 route[FROM_TRUNK] #force_send_socket("pu.bl.ic.ip "); rtpproxy_manage("","pu.bl.ic.ip "); $du="sip:ip.of.free.switch:5080"; route(RELAY); call was connected but rtp is not going through the kamailio server like folows, SIP FLOOR SIP_Call------> kamailio-------> freeswitch SIP_Call<------ kamailio<------- freeswitch RTP FLOOR SIP_Call ------> freeswitch SIP_Call <------ freeswitch please help me to configure kamailio and RTPProxy to manage RTP -- thanking you.. Achintha

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Re: [SR-Users] Path & Loose Route
by Luke Milbourne 04 Aug '16

04 Aug '16

Hi Daniel, Thanks again for your reply again. I remembered about the debug setting and done it yesterday, not at my pc right now but I remember one of the log lines was saying "the topmost route header is me". Still trying to understand it, but will get a copy of the log lines for you, if you wouldn't mind taking a look. Thanks again, Luke On 4 Aug 2016 06:57, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: Hello, set debug=3 in kamailio.cfg and look at log messages in syslog. If you find it too verbose, use dispatcher module and set debug level only for core and rr modules to 3. Cheers, Daniel On 03/08/16 13:22, Luke Milbourne wrote: Hi Daniel, I've added some more logging and can see loose_route() is returning -1 - route calculation has been unsuccessful. Is there a way of finding out why this is the case? Kind regards, Luke On 2 August 2016 at 09:49, Luke Milbourne <luke.milbourne(a)gmail.com> wrote: > I added the below xlog lines into my script: > > xlog("L_NOTICE", "Before loose_route: $si -> $ru - $du\n"); > xlog("L_NOTICE", "Before record_route: $si -> $ru - $du\n"); > xlog("L_NOTICE", "After both: $si -> $ru - $du\n"); > > Which generated the below output in syslog: > > Aug 2 09:46:11 ip-10-10-2-5 /sbin/kamailio[529]: NOTICE: <script>: Before > loose_route: 10.10.0.25 -> sip:201@192.168.1.114:5062 - <null> > Aug 2 09:46:11 ip-10-10-2-5 /sbin/kamailio[529]: NOTICE: <script>: Before > record_route: 10.10.0.25 -> sip:201@192.168.1.114:5062 - <null> > Aug 2 09:46:11 ip-10-10-2-5 /sbin/kamailio[529]: NOTICE: <script>: After > both: 10.10.0.25 -> sip:201@192.168.1.114:5062 - <null> > > Seems that the destination URI is never set, the Route header is below: > > Route: <sip:201@10.10.2.5:5072>;lr;received=sip:[PUBLIC_IP]:45126 > > The use_received modparam for path is also set to 1. > > Any idea's/suggestions are greatly appreciated :) > > Kind regards, > Luke > > On 2 August 2016 at 09:40, Luke Milbourne <luke.milbourne(a)gmail.com> > wrote: > >> Morning Gents, >> >> Firstly, thanks for taking the time to reply :) >> >> Kamailio is doing the NAT detection and simply acting as a passthrough to >> FreeSWITCH (which handles all SIP registrations etc). >> >> Will setup the log lines shortly and share the results. >> >> kamailio -V outputs; >> >> version: kamailio 4.3.4 (x86_64/linux) 83e842 >> flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, >> DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, >> DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, >> USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> id: 83e842 >> compiled on 18:22:39 Jan 26 2016 with gcc 4.9.2 >> >> Kind regards, >> Luke - Chicane Support >> >> On 2 August 2016 at 07:52, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> yes, loose_route() should be used there. Can you add a log message and >>> print $ru and $du before and after loose_route() in config and then send >>> the result to this mailing list? >>> >>> Btw, what version of Kamailio are you using? >>> >>> Cheers, >>> Daniel >>> >>> On 01/08/16 14:46, Luke Milbourne wrote: >>> >>> Is it correct to use loose_route() along side the path module? >>> >>> On 1 August 2016 at 13:33, Luke Milbourne <luke.milbourne(a)gmail.com> >>> wrote: >>> >>>> Hi Daniel, >>>> >>>> Yep, I've got the below in my kamailio.cfg >>>> >>>> modparam("path", "use_received", 1) >>>> >>>> Kind regards, >>>> Luke >>>> >>>> On 1 August 2016 at 13:25, Daniel-Constantin Mierla <miconda(a)gmail.com> >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> On 01/08/16 14:14, Luke Milbourne wrote: >>>>> >>>>> Hi all, >>>>> >>>>> Currently trying to use Kamailio as a passthrough to some FreeSWITCH >>>>> boxes. >>>>> >>>>> When receiving an invite from FreeSWITCH which contains a Route header >>>>> including received information, example below; >>>>> >>>>> Route: <sip:201@10.10.2.5:5072>;lr;received=sip:[PUBLIC_IP]:45126 >>>>> >>>>> When using loose_route() Kamailio still send the invite to the rfc1918 >>>>> address instead of the received IP:PORT in the Route header. >>>>> >>>>> Am I misunderstanding how this should work? >>>>> >>>>> >>>>> if you use path module, have you enabled the option to use the >>>>> received parameter? >>>>> >>>>> >>>>> https://www.kamailio.org/docs/modules/stable/modules/path.html#idp38033932 >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> -- >>>>> Daniel-Constantin Mierlahttp://www.asipto.com - http://www.kamailio.orghttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>> >>>>> >>>>> _______________________________________________ >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Luke Milbourne >>>> >>>> Tel: 07857154817 >>>> Google Talk/Email: luke.milbourne(a)gmail.com >>>> >>> >>> >>> >>> -- >>> Luke Milbourne >>> >>> Tel: 07857154817 >>> Google Talk/Email: luke.milbourne(a)gmail.com >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierlahttp://www.asipto.com - http://www.kamailio.orghttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> >>> >>> _______________________________________________ >>> 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 >>> >>> >> >> >> -- >> Luke Milbourne >> >> Tel: 07857154817 >> Google Talk/Email: luke.milbourne(a)gmail.com >> > > > > -- > Luke Milbourne > > Tel: 07857154817 > Google Talk/Email: luke.milbourne(a)gmail.com > -- Luke Milbourne Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlahttp://www.asipto.com - http://www.kamailio.orghttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda _______________________________________________ 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 1

Kamailio on Mac
by Carlos Ruiz Díaz 04 Aug '16

04 Aug '16

Hi all, I'm switching my development environment from Linux to Mac. Is there any tutorial on how to setup Kamailio on this OS? It may be an Unix flavor, but it's still quite different from Linux :). Thanks, -- Carlos http://caruizdiaz.com http://ngvoice.com +52 55 3048 3303

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Kamailio + rtpengine + freeswitch
by Kotb, Amir 04 Aug '16

04 Aug '16

[03:50] <meamo> I have a small query, can anybody assist please? [03:51] <meamo> I am running kamailio + rtpengine on a google compute cloud instance [03:51] <meamo> have setup them to work with devices behind nat, and everything works [03:52] <meamo> no i have added freeswitch. but I don't know how to configure rtpengine to work with both [03:52] <meamo> when I use #!define with_nat and #!define with_freeswitch, I get no voice in the calls. When I remove, #!define with_freeswitch, everything works normally Best regards, Amir -- Amir KOTB, Msc., Bsc. Postgraduate Researcher Department of Electrical Engineering & Electronics, The University of Liverpool, Brownlow Hill, Liverpool L69 3GJ, UK Mobile: +44-(0) 7428844234 Email: A.Kotb(a)liverpool.ac.uk Skype: A.kotb1 Web: Http://uk.linkedin.com/in/AOKotb Please consider the environment before printing this email

2 1

Path & Loose Route
by Luke Milbourne 04 Aug '16

04 Aug '16

Hi all, Currently trying to use Kamailio as a passthrough to some FreeSWITCH boxes. When receiving an invite from FreeSWITCH which contains a Route header including received information, example below; Route: <sip:201@10.10.2.5:5072>;lr;received=sip:[PUBLIC_IP]:45126 When using loose_route() Kamailio still send the invite to the rfc1918 address instead of the received IP:PORT in the Route header. Am I misunderstanding how this should work? -- Luke Milbourne <luke.milbourne(a)gmail.com> Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com

3 10

Help with dispatcher ping: Kamailo 4.1
by Andy Chen 03 Aug '16

03 Aug '16

Hi, I'm currently running Kamailio 4.1 and was wondering if there is a way to control ds_ping_interval on a dispatcher setid. Thanks. -- Andy Chen Sr. Telephony Lead Engineer 415 516 5535 (M) achen(a)thinkingphones.com

1 0

Fail to register at trunk through UAC
by Yuriy Gorlichenko 03 Aug '16

03 Aug '16

Hi. All trunks works fine ony one not works Kamailio sends REGISTER with proxy auth provider answers with 407 Kamailio not send any REGISTER Just answers in log uac_reg_tm_callback(): authentication failed for <trunkname> At attachement kamialio debug 3 log and sip log

2 6

SSL Certificate (Second Try)
by JCOT 02 Aug '16

02 Aug '16

I'm not sure where to send this information so I hope it's okay to share it with the mailing list, as well as the whois registrants: I went to register for a new account at https://serweb.iptel.org/user/reg. Firefox told me that the SSL certificate was bad. Specifically: -- serweb.iptel.org uses an invalid security certificate. The certificate is not trusted because it was signed using a signature algorithm that was disabled because that algorithm is not secure. The certificate expired on 10/22/2014 12:39 AM. The current time is 08/01/2016 09:55 PM. -- Thank you.

2 1

SSL Certificate
by JCOT 02 Aug '16

02 Aug '16

I'm not sure where to send this information so I hope it's okay to share it here: I went to register for a new account at https://serweb.iptel.org/user/reg. Firefox told me that the SSL certificate was bad. Specifically:

1 0

A: Upgrading kamailio.org os
by Daniel-Constantin Mierla 02 Aug '16

02 Aug '16

Hello, a short note to announce that the os on kamailio.org server will be upgraded in the near future, likely next week, exact date still to be decided. Hopefully there will be only very short downtime intervals. Main affected services: website (including the wiki) and mailing lists. I will send a notification once the procedure starts, but I wanted to allow enough time for this information to propagate properly within the community. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

12 24

Saving and extracting contact header
by Nahum Nir 01 Aug '16

01 Aug '16

Hello, I am quite a newbe in Kamailio scripts and further to my effort to implement push notification, I am adding to the contact header device ID. I have two problems: 1. When the register of the callee expires Kamailio frees the record so the information on the calle is not available. 2. Even when the callee is registered I do not how to access the contact field. Tried reg_fetch_contacts(...,"callee") and there is a lot of info but not the contact header. Thanks and regards, Nir

2 7

CDP -Could not open module error
by Andrew Okri 01 Aug '16

01 Aug '16

Hi All, I have just installed KamailioIMS 4.4.2 on a Ubuntu 16.04 vm and configured it according to the kamailio tutorial. When l attempt to start the icscf or scscf kamailio, it fails to start due to its inability to open the CDP module referring to the log_db_config function in particular.I have looked at the archive to see if anyone else have encountered this problem, but could not find anything. The log_db_config function takes a config object as a parameter, hence l have inspected the CDiameterPeer config file carefully and it seems both xml and diameter compliant. I am at loss where else to look. Please would anyone be kind enough to assist with getting a scscf and icscf started please. Regards Andrew

2 1

Kamailio static comilation
by Pratik V Patel 01 Aug '16

01 Aug '16

Hello all, I want to compile kamailio open source statically. I want to remove dependency for dynamic shared libraries. can anyone help me to get out of it? I have tried by chang in Makefiles but it does not work. What the meaning of "static_modules" during compilation. It is still generating dynamic shared libraries. Please do needful. -- Thanks And Regards, Pratik V. Patel Ext: 558

2 1

cnxcc-web error
by Infinicalls Infinicalls 01 Aug '16

01 Aug '16

Hi, I am really struggling to get the cnxcc-web installed. the /var/log/apache2/error.log says [Mon Aug 01 02:52:00.177725 2016] [:error] [pid 17534] [client 115.241.40.163:1851] PHP Deprecated: ServiceManagerAwareInterface is deprecated and will be removed in version 3.0, along with the ServiceManagerAwareInitializer. Please update your class ZfcUser\\Authentication\\Adapter\\Db to remove the implementation, and start injecting your dependencies via factory instead. in /usr/local/cnxcc/cnxcc-web/vendor/zendframework/zend-mvc/src/Service/ServiceManagerConfig.php on line 129, referer: http://xxxxxxxx.com/user/login I am getting the login screen and it freezes there. Can somebody guide me how to go about? Or is there a simple php (without Zend framework) interface available for cnxcc? thanks Ganesh Kumar

2 1

manage rtp with or without RTP Proxy
by Achintha 01 Aug '16

01 Aug '16

hi I have configured kamailio and freeswitch on two Ubuntu servers as follows. Sip_TRUNK------> kamailio-------> freeswitch i wan to manage rtp through the kamailio server, first i want to know, 1. can i do it without RTP Proxy ? anyway i tried it with RTP Proxy as follows, I installed rtpproxy on kamailio server and start it using “rtpproxy –F -l pu.bl.ic.ip -s udp:localhost :7722 -u root” it was working Then I put modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") modparam("rtpproxy", "rtpproxy_retr", 2) modparam("rtpproxy", "ice_candidate_priority_avp", "$avp(ice_priority)") mhomed=0 route[FROM_TRUNK] force_send_socket("pu.bl.ic.ip "); rtpproxy_manage("","pu.bl.ic.ip "); $du="sip:ip.of.free.switch:5080"; route(RELAY); then call is received to user and disconnected on freeswitch console get some errors [ERR] switch_core_media.c:7179 AUDIO RTP REPORTS ERROR: [Remote Address Error!] [ERR] mod_sofia.c:2290 CODEC NEGOTIATION ERROR. SDP: v=0 o=- 3679035755 3679035755 IN IP4 pu.bl.ic.ippu.bl.ic.ip s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4167841678 RTP/AVP 123 8 0 101 c=IN IP4 pu.bl.ic.ippu.bl.ic.ip b=TIAS:64000 a=rtpmap:123 opus/48000/2 a=fmtp:123 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=16000;useinbandfec=1 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp:4167941679 a=nortpproxy:yes a=nortpproxy:yes please advice me thank you

2 1

function to check if string is an ipv4 address
by Juha Heinanen 29 Jul '16

29 Jul '16

I did not find in textops or utils modules a function to check if a string is an ipv4 address. Does it exist somewhere else? -- Juha

2 2

RTPEngine Kamailio crashed because someone tried to do something
by Andrew Tan 29 Jul '16

29 Jul '16

I'm getting below every 2-3 weeks someone tried to connect through the CLi of RTPengine and crashed Rtpengine. Anyone knows what to make out of this ? Thanks Andrew 82440 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13020]: INFO: <script>: HTTP Request Received 182441 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13020]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 16) 182442 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13020]: ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 404 Not Found#015#012Sia: SIP/2.0/TCP 89.163.135.117:63413#015#012Server: kamailio (4.3.3 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012> 182443 Jul 28 16:24:21 sipr rtpengine[1224]: New cli connection from 89.163.135.117:63587 182444 Jul 28 16:24:21 sipr rtpengine[1224]: Got CLI command: 182445 Jul 28 16:24:21 sipr rtpengine[1224]: New cli connection from 89.163.135.117:63838 182446 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13028]: WARNING: <script>: HTTP request received on 5080 182447 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13021]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182448 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13021]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading 182449 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13020]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182450 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13020]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading 182451 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13021]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182452 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13021]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading 182453 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13028]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182454 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13028]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading 182455 Jul 28 16:24:22 sipr /usr/local/sbin/kamailio[13020]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182456 Jul 28 16:24:22 sipr /usr/local/sbin/kamailio[13020]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading 182457 Jul 28 16:24:22 sipr /usr/local/sbin/kamailio[13021]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182458 Jul 28 16:24:22 sipr /usr/local/sbin/kamailio[13021]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading

2 1

Kamailio LB: how to get Asterisk to do the RTP
by Tickling Contest 28 Jul '16

28 Jul '16

With the help of members from this mailing list (many thanks!), I finally got Asterisk fronted by Kamailio for LB and REGISTERs and I am able to make a call using the setup that looks like this: [Kamailio 4.4.2]<->[Asterisk 13.7.2] Kamailio manages REGISTERs, but also forwarding them to Asterisk. I am able to make a call, but I get only one way audio or no audio depending on which client made the call (SipDroid->Zoiper I hear one way audio on Zoiper, but no audio if the call is made the other way). I noticed that Kamailio forced direct media between the endpoints in this situation, but my application really needs Asterisk to handle it. How do I do this? Should I start by forwarding INVITEs to Asterisk? How do I do that? Any help is appreciated. Thanks!

5 5

undefined symbol: EVP_sha1 when using outbound module
by Francisco Burgos 28 Jul '16

28 Jul '16

Hi, I am trying to use kamailio as an edge proxy to forward requests and media to an sbc. when compiling the cfg file included on the examples. i get: undefined symbol: EVP_sha1 I believe this problem is being caused by the outbound module and after some research i managed to read that it is caused by the linking between the gcc libcrypto. I would like some help since this project requires urgency. Thanks in advance.

2 5

TM: t_relay([host, port]) DNS Queries
by Louis-Olivier Roof 27 Jul '16

27 Jul '16

Dear Kamailio users, Ref: http://kamailio.org/docs/modules/4.2.x/modules/tm.html#tm.f.t_relay Using t_relay([host, port]) to relay messages to a specific host does not seem to generate DNS queries to the said host after Kamailio has initially processed the configuration file. It resolves the host once at startup then continues using the same IP after. Is that expected behavior? Anyone has had this issue before and found a way to work around it? Note that for this specific application, messages need to be relayed to a specified host (different from current R-URI) and this host needs to be resolved by DNS during normal operation, not only at startup. Thanks! Louis-Olivier

2 4

Checking for 200ok Response to a REGISTER request kamailio-Asterisk
by Jonathan Hunter 27 Jul '16

27 Jul '16

Hi Guys, So currently on our network we have a kamailio server which users register against, we then replicate the register messages to 2 Asterisk boxes sat behind it so that all entities are aware of the registration state of the users. REGISTER--->KAMAILIO---->ASTERISK A ---->ASTERISK B With a REGISTER---200OK exchange between Kamailio and Asterisk. We have an issue where at some points the Asterisk servers when under load dont respond with a 200 ok(something being investigated) to the register messages sent to kamailio, so I am just working on some logic for the register message to be resent using the t_replicate and t_set_fr functions. This works well should both Asterisk servers not respond, however, as I am using replicate and it is parallel forking, if say Asterisk A answers first and is available with a 200ok then that in turn cancels the register message branch being sent to Asterisk B(which I know is fine), however there could be a scenario where Asterisk B doesnt respond, and we wont know about it to try and resend the Register message, as the branch is cancelled. Hope that makes sense? I am looking at checking that both the Asterisk servers have responded and sent a 200ok, which I can grab in an onreply route but Im just wondering if someone has done something similar or has any suggestions as it is tricky to achieve currently. I have also thought about stateless working but I really need kamailio to keep retransmitting the register until it gets a response. Many thanks Jon

2 4

Segfault in dialog.so with 4.4.2 stable
by Dirk Teurlings - Signet B.V. 26 Jul '16

26 Jul '16

Hi, Running Kamailio on Debian from the Kamailio repository with 4.4.2 stable (unpatched). Getting some random segfaults with it now, here's the relevant backtrace from the generated core. Core was generated by `/usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.'. Program terminated with signal 11, Segmentation fault. #0 run_dlg_callbacks (type=type@entry=64, dlg=dlg@entry=0x7fceb400e2f0, req=req@entry=0x7fced4f093c8, rpl=rpl@entry=0x0, dir=<optimized out>, dlg_data=dlg_data@entry=0x0) at dlg_cb.c:253 253 if ( (cb->types)&type ) { (gdb) bt #0 run_dlg_callbacks (type=type@entry=64, dlg=dlg@entry=0x7fceb400e2f0, req=req@entry=0x7fced4f093c8, rpl=rpl@entry=0x0, dir=<optimized out>, dlg_data=dlg_data@entry=0x0) at dlg_cb.c:253 #1 0x00007fcead3648f9 in dlg_terminated (dir=<optimized out>, dlg=0x7fceb400e2f0, req=0x7fced4f093c8) at dlg_handlers.c:368 #2 dlg_onroute (req=0x7fced4f093c8, route_params=<optimized out>, param=<optimized out>) at dlg_handlers.c:1354 #3 0x00007fceb10ab0e2 in run_rr_callbacks (req=req@entry=0x7fced4f093c8, rr_param=rr_param@entry=0x7fceb12b77a0) at rr_cb.c:96 #4 0x00007fceb10952c5 in after_loose (_m=0x7fced4f093c8, preloaded=0) at loose.c:919 #5 0x000000000042b618 in do_action (h=h@entry=0x7ffeb0b3ed80, a=a@entry=0x7fced4cb4338, msg=msg@entry=0x7fced4f093c8) at action.c:1060 #6 0x000000000042a10a in run_actions (h=h@entry=0x7ffeb0b3ed80, a=0x7fced4cb4338, msg=0x7fced4f093c8) at action.c:1549 #7 0x0000000000437544 in run_actions_safe (h=h@entry=0x7ffeb0b402b0, a=<optimized out>, msg=<optimized out>) at action.c:1614 #8 0x000000000053b2e8 in rval_get_int (h=0x7ffeb0b402b0, msg=<optimized out>, i=0x7ffeb0b3f1e0, rv=rv@entry=0x7fced4cb4d58, cache=cache@entry=0x0) at rvalue.c:912 #9 0x000000000054261c in rval_expr_eval_int (h=h@entry=0x7ffeb0b402b0, msg=msg@entry=0x7fced4f093c8, res=res@entry=0x7ffeb0b3f1e0, rve=rve@entry=0x7fced4cb4d50) at rvalue.c:1910 #10 0x000000000042bc91 in do_action (h=h@entry=0x7ffeb0b402b0, a=a@entry=0x7fced4cb8f88, msg=msg@entry=0x7fced4f093c8) at action.c:1030 #11 0x000000000042a10a in run_actions (h=h@entry=0x7ffeb0b402b0, a=0x7fced4cb8f88, msg=msg@entry=0x7fced4f093c8) at action.c:1549 #12 0x000000000042bcf2 in do_action (h=h@entry=0x7ffeb0b402b0, a=a@entry=0x7fced4cb91e8, msg=msg@entry=0x7fced4f093c8) at action.c:1049 #13 0x000000000042a10a in run_actions (h=h@entry=0x7ffeb0b402b0, a=0x7fced4cb3f48, msg=msg@entry=0x7fced4f093c8) at action.c:1549 #14 0x000000000042bde0 in do_action (h=h@entry=0x7ffeb0b402b0, a=a@entry=0x7fced4ac3d70, msg=msg@entry=0x7fced4f093c8) at action.c:678 #15 0x000000000042a10a in run_actions (h=h@entry=0x7ffeb0b402b0, a=a@entry=0x7fced4ac1698, msg=msg@entry=0x7fced4f093c8) at action.c:1549 #16 0x00000000004375d0 in run_top_route (a=0x7fced4ac1698, msg=msg@entry=0x7fced4f093c8, c=c@entry=0x0) at action.c:1635 #17 0x0000000000504386 in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:240 #18 0x00000000005f5bd4 in udp_rcv_loop () at udp_server.c:495 #19 0x00000000004b2625 in main_loop () at main.c:1600 #20 0x0000000000427e2b in main (argc=<optimized out>, argv=<optimized out>) at main.c:2616 And from syslog the relevant messages before this dump: Jul 15 08:55:03 server /usr/sbin/kamailio[16470]: WARNING: dialog [dlg_handlers.c:1219]: dlg_onroute(): unable to find dialog for BYE with route param 'd4c.26d1' [3149:7522] Jul 15 08:56:01 server /usr/sbin/kamailio[16481]: WARNING: dialog [dlg_handlers.c:1219]: dlg_onroute(): unable to find dialog for BYE with route param 'fc.99f1' [207:8089] Jul 15 08:56:27 server /usr/sbin/kamailio[16470]: CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7fceb3f7d920 tl->next=(nil) tl->prev=(nil) Jul 15 08:56:27 server /usr/sbin/kamailio[16470]: ERROR: dialog [dlg_handlers.c:1377]: dlg_onroute(): failed to update dialog lifetime Jul 15 08:57:01 server /usr/sbin/kamailio[16482]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Duplicate entry '9584-3854-435' for key 'hash_index' (1062) Jul 15 08:57:01 server /usr/sbin/kamailio[16482]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jul 15 08:57:01 server /usr/sbin/kamailio[16482]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [INSERT INTO `dialog_extra` (`h_i] Jul 15 08:57:01 server /usr/sbin/kamailio[16482]: ERROR: auth [api.c:119]: auth_check_hdr_md5(): auth:pre_auth: Credentials are not filled properly Jul 15 08:57:01 server /usr/sbin/kamailio[16483]: ERROR: auth [api.c:119]: auth_check_hdr_md5(): auth:pre_auth: Credentials are not filled properly Jul 15 08:57:54 server /usr/sbin/kamailio[16506]: NOTICE: dialog [dlg_hash.c:245]: dlg_clean_run(): dialog in delete state is too old (0x7fceb3f64470 ref 4) Jul 15 08:57:54 server /usr/sbin/kamailio[16473]: WARNING: dialog [dlg_handlers.c:1348]: dlg_onroute(): inconsitent dlg timer data on dlg 0x7fceb3f64470 [1182:5803] with clid '09ad128753e2535d24bde58e3d7eda04@192.168.10.232:5060' and tags 'as1b497b34' '5788890C-EC6F55F-3E86ED0C' Jul 15 08:57:54 server /usr/sbin/kamailio[16469]: ERROR: dialog [dlg_handlers.c:334]: dlg_terminated_confirmed(): failed to get dialog from params! Jul 15 08:58:49 server /usr/sbin/kamailio[16467]: WARNING: dialog [dlg_handlers.c:1219]: dlg_onroute(): unable to find dialog for BYE with route param '6d2.2581' [726:6226] Jul 15 08:59:24 server /usr/sbin/kamailio[16506]: NOTICE: dialog [dlg_hash.c:245]: dlg_clean_run(): dialog in delete state is too old (0x7fceb400e2f0 ref 4) Jul 15 08:59:25 server /usr/sbin/kamailio[16464]: WARNING: dialog [dlg_handlers.c:1219]: dlg_onroute(): unable to find dialog for BYE with route param '3e4.b5c1' [1251:7259] Jul 15 08:59:25 server /usr/sbin/kamailio[16465]: WARNING: dialog [dlg_handlers.c:1348]: dlg_onroute(): inconsitent dlg timer data on dlg 0x7fceb400e2f0 [1251:7259] with clid '87791a#015#012Call-ID: 25750e286a5654361ef9405d72edbc' and tags '' 'as148f41b1' Jul 15 08:59:25 server kernel: [203670.830521] kamailio[16465] general protection ip:7fcead34b3a5 sp:7ffeb0b3e220 error:0 in dialog.so[7fcead330000+88000] Jul 15 08:59:26 server /usr/sbin/kamailio[16511]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 33 Jul 15 08:59:26 server /usr/sbin/kamailio[16458]: ALERT: <core> [main.c:739]: handle_sigs(): child process 16465 exited by a signal 11 Jul 15 08:59:26 server /usr/sbin/kamailio[16458]: ALERT: <core> [main.c:742]: handle_sigs(): core was generated Jul 15 08:59:26 server /usr/sbin/kamailio[16458]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD Any insight would be appreciated! Cheers, Dirk

2 12

CANCEL message within transaction not being relayed
by Ali Soltani 26 Jul '16

26 Jul '16

Hi, We tested the following on Kamailio 3.x and 4.x and observed the same behavior. INVITE and CANCEL are sent to Kamailo from the same source within the same transaction. (branch value in the first via header are the same in both INVITE and CANCEL message. Also FROM and TO header are same as well). However, Kamailio does not relay CANCEL message. UAC keep sending CANCEL till timeout. Apparently, t_check_trans() returns false. Can anyone help me on this issue on why Kamailio does not relay CANCEL message? Thanks, AS INVITE received by Kamailio INVITE sip:1111111@10.10.10.10:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 20.20.20.20:5060;branch=z9hG4bKkjhg4n100gagvkoup741.1 Max-Forwards: 69 From: <sip:2222222@20.20.20.20>;tag=SDl9eu701-1c1026774897 To: <sip:1111111@10.10.10.10;user=phone> Call-ID: SDl9eu701-36d75c874d3f463f6c83c5c94cc98b77-8q7ug02 CSeq: 1 INVITE Contact: <sip:2222222@20.20.20.20:5060;transport=udp> Supported: em,timer,replaces,path,resource-priority,sdp-anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: CS-CustomerName/v.6.60A.274 Content-Type: application/sdp Content-Length: 237 P-Asserted-Identity: tel:2222222 P-Asserted-Identity: sip:2222222@192.168.0.1 after 70 msec CANCEL received by Kamailio in the same transaction CANCEL sip:1111111@10.10.10.10:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 20.20.20.20:5060;branch=z9hG4bKkjhg4n100gagvkoup741.1 CSeq: 1 CANCEL Max-Forwards: 69 From: <sip:2222222@20.20.20.20>;tag=SDl9eu701-1c1026774897 To: <sip:1111111@10.10.10.10.;user=phone> Call-ID: SDl9eu701-36d75c874d3f463f6c83c5c94cc98b77-8q7ug02 Content-Length: 0 Reason: Q.850 ;cause=16

2 1

music on hold configuration
by Gaurav Bmotra 25 Jul '16

25 Jul '16

hi i m a new user of kamailio i have sertup kamailio SIP to SIP call going well but now i m tring to impliment Music On Hold i m this http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#table.parms but unable to impliment plz help me Hare i attache my kamailio config file -- *Regards:* Gaurav Bmotra Smith Infotech E-46, phase 8 , Indus Area S.A.S Nagar Mohali 160071

3 18

Kamailio+tls not working with Ubuntu 16.04
by Yuriy Gorlichenko 25 Jul '16

25 Jul '16

Just asking about any progress of this staff THere is a bug descried at the ubuntu bug tracker https://bugs.launchpad.net/ubuntu/+source/kamailio/+bug/1591992 There is bug that i wrote at the kamailio bug tracker https://github.com/kamailio/kamailio/issues/714 Closed it because It is not trouble of kamailio itself. I suppose something wrong at the ubuntu enviroment Any ideas or may be someone resolves it?

2 3

Re: [SR-Users] Tsilo Tiimers
by Federico Cabiddu 25 Jul '16

25 Jul '16

Hi, looks like in your script, in the SENDPUSH route, you set the luaret var to 0 then you check if it's value is != 1, in which case you reply 501, Unknown Destination. Since you never change the var value you are always hitting this case. Regards, Federico On 24 Jul 2016 14:25, "Nahum Nir" <hello.shalom.hi(a)gmail.com> wrote: > Hello Fredrico, > > Please see the attached cfg. I tried to add the fragments you published > and to add a suspension at SENDPUSH but I am getting Desitnation Unknow. > > Thank you so much, > Nir > > On Thu, Jul 21, 2016 at 10:03 PM, Federico Cabiddu < > federico.cabiddu(a)gmail.com> wrote: > >> Hi Nir, >> all you should do upon receiving B's REGISTER, after saving the contact >> (by calling save function) is either resume the transaction, perform a >> lookup and relay or call ts_append (if you already relayed the INVITE). If >> you get "unavailable" (I guess that this means that you are receiving a >> 404) probably the user is not being found in the location. Check if you are >> doing these steps in your config and if it doesn't work try to send the >> relevant part of the config and the kamailio's logs. I'm currently away and >> almost without network but I'll try to have a look as soon as possible. >> >> Regards, >> >> Federico >> >> On 21 Jul 2016 16:09, "Nahum Nir" <hello.shalom.hi(a)gmail.com> wrote: >> >> Thanks, >> >> I was able to suspend the transaction and wait for the register but the >> client that sends the INVITE get unavailable right after the other side >> joins. >> Here is my debug flow: >> Client A sends the INVITES. >> >> Use case 1: Client B register one time. Then I turn off the WiFi in >> client B so it will not recieve the push notification. Client A sends the >> INVITE, server sends the PUSH and Client A is waiting for 5 seconds. >> >> Use case 3: Same as before but now client B get the push. This time >> client A that sends the INVITE gets from the server unavailable as soon as >> client B register. >> >> I did as you suggested and suspend the trunsaction before sending the >> push. Probably a second suspend is needed when client B sends the REGISTER? >> >> Thanks and regards, >> Nir >> >> On Wed, Jul 20, 2016 at 4:06 PM, Federico Cabiddu < >> federico.cabiddu(a)gmail.com> wrote: >> >>> Hi, >>> to set the timeout for a specific transaction you can use tm module >>> function t_set_fr ( >>> http://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_set_fr >>> ). >>> For example you could call it before suspending the transaction setting >>> a short timeout and then call it again in the reply route when receiving >>> the first provisional response from the client. >>> >>> Regards, >>> >>> Federico >>> On 20 Jul 2016 2:50 pm, "Nahum Nir" <hello.shalom.hi(a)gmail.com> wrote: >>> >>>> Hello, >>>> >>>> How can I chage the timer waiting for the client to send the register. >>>> i.e. how much time to wait for the push? >>>> >>>> Thanks, >>>> Nir >>>> >>>> >>>> _______________________________________________ >>>> 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 >>>> >>>> >>> _______________________________________________ >>> 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

How to forward REGISTERs to Asterisk in kamailio with dispatcher module with several Asterisk PBXs?
by Tickling Contest 24 Jul '16

24 Jul '16

Hello, When using Kamailio with the dispatcher module for Asterisk load balancing, I am offloading REGISTERs to Kamailio (works fine), but would like to forward the REGISTERs to Asterisk so that outgoing calls from Asterisk will be possible. I know this is not strictly necessary (e.g., just use Kamailio to forward everything), but I have a lot of application logic behind my Asterisk boxes (using ARI) and so I don't want that to break. Thereby, I do need this information in Asterisk as soon as possible from Kamailio. Let's say I have three boxes Asterisk_1, Asterisk_2 and Asterisk_3 in the dispatcher module like so: 1 sip:192.168.1.201:5060;transport=tcp 1 sip:192.168.1.202:5060;transport=tcp 1 sip:192.168.1.203:5060;transport=tcp My system has endpoints as numerical extensions, from 101 to 110. When a new endpoint registers with kamailio (and authenticates), I would like to send the REGISTER to Asterisk. But I have the following questions: (a) Which Asterisk? I could send it to a round-robinned Asterisk (say, Asterisk_2), but that Asterisk may not be the one responsible for handling calls outbound for the endpoint that just registered. Even if I use "hash over auth username" instead of round-robin (In http://www.kamailio.org/docs/modules/4.0.x/modules/dispatcher.html#idp16940…, choose 5 for "alg", for example) that wont solve this problem as an outbound request may come from another Asterisk box which won't have the requisite registration information. Should I forward REGISTERs to all Asterisk boxes? Maybe I won't have to deal with this issue as I use realtime and so the REGISTER information, sent to one Asterisk box, is available to all Asterisk boxes after the write to the realtime DB. Is my understanding correct? (b) How do I do this in kamailio.cfg? With some examples on the Internet for WITH_ASTERISK directive, I see that you do route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport) + ";transport=tcp"; $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } What is the equivalent for when you are using the dispatcher module and the kamailio.bindip and asterisk.bindip etc. are not set in the configuration, but instead should be procured from elsewhere? Any help is appreciated.

3 9

SIP Outbound / t_next_contacts()
by Colin Morelli 24 Jul '16

24 Jul '16

Hey all, I've got two layers of Kamailio proxies running. One set of edge proxies that are parking outbound connections and doing load balancing to a set of registrar/proxies. This is working well with one exception: I can't seem to get t_load_contacts/t_next_contacts working correctly for handling multiple outbound connections. It *appears* to work (though I haven't extensively tested) whenever there's more than one registration binding for an AOR. However, when there's only one, I get: 8(27) DEBUG: tm [t_serial.c:191]: t_load_contacts(): nr_branches is 0 8(27) DEBUG: tm [t_serial.c:194]: t_load_contacts(): nothing to do - only one contact! 8(27) DEBUG: tm [t_serial.c:455]: t_next_contacts(): no contacts in contacts_avp - we are done! 8(27) ERROR: <script>: t_next_contacts() failed I tried omitting the call to t_next_contacts(), thinking it should possibly only be used in failure_route. But when I do that (while leaving t_load_contacts), I get: 5(24) ERROR: tm [t_fwd.c:1730]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding 5(24) WARNING: <core> [receive.c:258]: receive_msg(): error while trying script The relevant part of the config is: if (!lookup("location")) { $var(rc) = $rc; switch ($var(rc)) { case -1: xlog("L_INFO", "$ci|log|lookup of contacts for $ru returned no results\r\n"); send_reply("404", "Not Found"); exit; case -3: xlog("L_INFO", "$ci|log|lookup of contacts for $ru failed with code $var(rc)\r\n"); send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } if (!t_load_contacts() || !t_next_contacts()) { xlog("L_ERR", "t_load_contacts() failed\n"); sl_reply_error(); t_release(); exit; } Am I missing something else here? Note: it works fine if I remove t_load_contacts/t_next_contacts, but then I don't get proper handling of multiple sip.instance bindings/differing q values. Best, Colin

2 4

business case
by anfecora 24 Jul '16

24 Jul '16

I apologize since this is a technical mail list and this question may not belong here but i need to fill a business requirement and need more information of big companies using kamailio, i am searching over the internet but if anyone can provide me with names of real carriers using kamailio will be very useful. Thanks again for any help in this case.

2 1

Dispatcher list not loading from dispatcher.list, but loading from DB
by Tickling Contest 23 Jul '16

23 Jul '16

Hello, I have a Kamailio 4.0.4 configured with the cfg file shown in https://gist.github.com/ticklingcontest/f96aa4d08b08dc4643056d73da24f0a5 The modparam for dispatcher sets the dispatcher file to be in /var/run/kamailio/dispatcher.list. However, I recently discovered that $ kamctl dispatcher show picks up the information from the kamailio DB instead of the dispatcher.list. What's going on? Any insight is appreciated. Thanks!

1 0

kamailio starts looping invites(some) back to itself.
by Aqs Younas 22 Jul '16

22 Jul '16

Hi, I am using http_async_client and dispatcher modules of Kamailio. Here is scenario! http_async_client makes calls to an API to get groupid upon which dispatcher sends calls. I am using SIPp to generate calls with 100 cps. I see some calls being failed with an error cause, 483 Too Many Hops. Taking the trace shows Kamailio routing some calls to itself. I think this is not a configuration issue since all calls are same(follow same logic) and some calls get failed. Any pointer is much appreciated. Best Regards.

1 0

sip options not relaying
by gerry kernan 22 Jul '16

22 Jul '16

Hi I have a kamailio proxy in front of asterisk ver 13 servers. I'm using path support to route to the relevant asterisk server depending on the domain in the sip header. The incoming registers & invites from sip endpoints get routed to the correct asterisk server , but when asterisk is sending sip options to the end points my kamailio config is not routing the option on to the endpoint . Kamailio debug shows error below ERROR: <script>: 192.168.200.131 OPTIONS sip:1333@192.168.200.112:5062 -- domain "192.168.200.112" is not found in domain table 192.168.200.131 Am I correct in thinking that this SIP option should be getting routed via my on-reply route. the sip option looks like it is failing here at the start of my config if (src_ip != BACKEND_NET4) { xlog("L_NOTICE", "FROM CLIENT IP: $si $rm $ru $td \n"); # SIP request packet client->backend if( !loose_route() ) { if (!lookup_domain("$td", "dattr_")) { xlog("L_NOTOCE" , "Domain lookup, $rm "); xlog("L_ERR", "$si $rm $ru -- domain \"$td\" is not " + "found in domain table $si\n"); drop(); } Sample of SIP option captured on the kamailio server OPTIONS sip:1333@192.168.200.112:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.131:5060;branch=z9hG4bK4f910445 Max-Forwards: 70 Route: <sip:192.168.200.148;lr;received=sip:87.198.223.83:16070> From: "Unknown" <sip:Unknown@192.168.200.131>;tag=as54cfb9df To: <sip:1333@192.168.200.112:5062> Contact: <sip:Unknown@192.168.200.131:5060> Call-ID: 257dea110f5255275aa38d9c45e0e235@192.168.200.131:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.11.0(13.10.0) Date: Fri, 22 Jul 2016 15:31:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer, path Content-Length: 0 OPTIONS sip:1333@192.168.200.112:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.131:5060;branch=z9hG4bK20611e3b Max-Forwards: 70 Route: <sip:192.168.200.148;lr;received=sip:87.198.223.83:16070> From: "Unknown" <sip:Unknown@192.168.200.131>;tag=as1b6a0faf To: <sip:1333@192.168.200.112:5062> Contact: <sip:Unknown@192.168.200.131:5060> Call-ID: 7f8ed5ea7a147f2b61269df862af7fae@192.168.200.131:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.11.0(13.10.0) Date: Fri, 22 Jul 2016 15:31:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer, path Content-Length: 0 Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

1 1

Kamailio + Asterisk MWI Issues
by TEG AMJG 21 Jul '16

21 Jul '16

Dear list I am quite new to Kamailio and i know that probably this thread has been answered quite few times. But i have few questions I was able to configure Kamailio + Asterisk MWI with Kamailio handling the registration of users. The problem is that i am quite sure that is not the right way. What i did was just putting "subscribemwi=no" on every user declared on Asterisk and he will send NOTIFY's when a voicemail has been received for a particular user to Kamailio and Kamailio will just forward them to the endpoints (because he knows the location of the users). I know about the modules presence presence_xml and presence_mwi of Kamailio which will handle the subscription of users and send NOTIFY's but it is useless since I am not verifying any NOTIFY that comes from Asterisk, just forwarding. Now, some others solution are just a bunch of scripts that, for me, it doesnt make much sense since is the same idea like how i did it just more hacky and unnecesary, i guess?. This solution can be found here: http://saevolgo.blogspot.com/2012/07/asterisk-behind-kamailio-voicemail-mwi… There is another solution that i found in few threads here which i dont understand really well, is to send PUBLISH messages from Asterisk to Kamailio so Kamailio sends NOTIFY's with the event and content about the amount of voice messages in a particular mailbox. Some say that you need to create the message PUBLISH and send it with sipsak or another solution in Asterisk. For what i see in SIP documentation, the last solution seems a bit less of a hack lets say, more resilient. With all said, i have a few questions Is it the solution I made really a bad idea?. I havent take a lot of time understanding SUBSCRIBE/NOTIFY messages, just the basics, i dont know if there are some security issues i should think about in this particular flow. Should i check the status of the authorization of the subscription in Kamailio before forwarding the mesage (if there is a way, how)? or it really doesnt make sense at all? because after all, my endpoints receives the NOTIFY and "you have a new voicemail" message appears. Now, about the last solution, is there any guide of someone that has done that in a clean way?, may someone guide me how to construct the PUBLISH message and when should i send it?, should i create a cron or something?, this PUBLISH message needs to be send for every user?. may someone guide me (if this is considered a cleaner way of doing it) how to make this? Many of the answers in other threads about this are not enough to understand the idea as a whole, at least for me. Thank you very much in advance Alejandro -- - Saludos a todos

1 0

Kamailio stops accepting calls for few seconds after 80 CPS.
by Aqs Younas 21 Jul '16

21 Jul '16

Greeting list. I am doing the load test of Kamailio with SIPp using most basic script(kamailio.cfg) with logging level set to minimum to dispatch requests to SIPp UAS. This is my dispatcher code snippet. $avp(quality) = "1"; ds_select_dst("$avp(quality)", "4"); route(RELAY); At 70 CPS calls work fine. But if I increase CPS to 80, Kamailio stops accepting calls for few seconds and comes to life after being idle for some time. 100 CPS for sip proxy is not hot cake. Also, I see in http://www.kamailio.org/docs/openser-performance-tests/ tests were made with -l27 and -m7000 which looks wronged to me. ./sipp -sf uac_inv.xml <http://www.kamailio.org/docs/openser-performance-tests/data/uac_inv.xml> -rsa 192.168.2.102:5060 192.168.2.102:5070 -m 200000 -r 7000 -d 1 -l 27 There must be something wrong in my configuration since this cps is very low. kamailio -version version: kamailio 4.4.1 (x86_64/linux) 90be8b flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 90be8b SHM_MEMORY=768 PKG_MEMORY=64 Any suggestion is much appreciated. Best Regards,

2 2

Issue with ftag parameter in Record-Route header
by Valentin Christoph 21 Jul '16

21 Jul '16

Hi community, I'm using an MGCF that creates From tags like **%%!!MYOFFICE++00!!%%**A0F1F1BA Now the special characters "%%" are allowed in tags (i.e. "token" according to RFC 3261), token = 1*(alphanum / "-" / "." / "!" / "%" / "*" / "_" / "+" / "`" / "'" / "~" ) but they are NOT allowed in URI parameters according to RFC 3261 uri-parameter = transport-param / user-param / method-param / ttl-param / maddr-param / lr-param / other-param other-param = pname [ "=" pvalue ] pname = 1*paramchar pvalue = 1*paramchar paramchar = param-unreserved / unreserved / escaped param-unreserved = "[" / "]" / "/" / ":" / "&" / "+" / "$" unreserved = alphanum / mark mark = "-" / "_" / "." / "!" / "~" / "*" / "'" / "(" / ")" escaped = "%" HEXDIG HEXDIG which makes the MGCF ignore the whole RR header and sending subsequent requests directly to the peer SIP UA, instead of sending it to the CSCF. Hence it's not OK to just copy the From tag into the ftag parameter, imho. We did a workaround in the MGCF, now sending **~~!!MYOFFICE++00!!~~**A0F1F1BA, but strictly spoken the kamailio S-CSCF is not compliant to RFC 3261. What's your opinion? Is there an configuration option to make it compliant? All the best Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

Tsilo Tiimers
by Nahum Nir 21 Jul '16

21 Jul '16

Hello, How can I chage the timer waiting for the client to send the register. i.e. how much time to wait for the push? Thanks, Nir

2 2

Trouble with Asterisk behind Kamailio
by Tickling Contest 21 Jul '16

21 Jul '16

Hello, I am following the instructions in http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb to configure my Kamailio and Asterisk (using PJSIP realtime with MySQL). I am able to get Kamailio register the clients (tested with Zoiper softphone), but having trouble getting any further to load balance a couple of Asterisk servers. Here is a range of issues I have: (a) The Kamailio server keeps sending UDP SIP messages to the Asterisk server and it is not clear where to control what protocol to use to send SIP messages Asterisk server(s). (b) I followed the instructions for the dispatcher module in the wiki, but it is not clear why I should use the #!define WITH_ASTERISK directive to enumerate the bind IPs and ports (how does this reconcile with the list of Asterisk servers in the dispatcher.list file?): !ifdef WITH_ASTERISK asterisk.bindip = "192.168.1.102" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "192.168.1.101" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" Also, where do I specify the protocol to use when sending to the Asterisk servers? Is it auto detected? If so, why is Kamailio not detecting that the Asterisk servers are only listening on TCP? Any help is appreciated. My entire /etc/kamailio/kamailio.cfg file is in here: https://gist.github.com/ticklingcontest/c7437dbc2d8e838834ee4d852753f254. I am using Kamailio 4.0.4 with Asterisk 13.7.2 using TCP for PJSIP. Thanks!

3 3

Kamailio Crashes when dispatcher failover is being applied
by TEG AMJG 21 Jul '16

21 Jul '16

Dear list Im quite new to Kamailio and i found something while trying to see the failover behaviour in Kamailio. I have a kamailio v4.3.5 load balancing two Asterisk boxes. This happens just when i stop one of the asterisk and make a call that is being load balanced to the same asterisk box that i stopped before, when trying to go to failure_route and dispatch it to the other box then Kamailio just crashes. I have debug=4 in the config file and it really doesnt say much about the error. This is the relative logging i could found: *Jul 18 16:30:13 kam2 /usr/sbin/kamailio[69548]: INFO: <script>: UBICAR: Forwarding non-registrar requests to AsteriskJul 18 16:30:13 kam2 /usr/sbin/kamailio[69548]: INFO: <script>: UBICAR: Dispatching to Asterisk Box sip:10.0.1.201:5060 <http://10.0.1.201:5060>Jul 18 16:30:13 kam2 /usr/sbin/kamailio[69550]: INFO: <script>: UBICAR: requestm=INVITE reply=<null> SIP packet comes from the Asterisk Box duid=PBX2;my=pbx2;maxload=10000 callid=4d52095c6f2ed1ab2de2d6c70340a126@10.0.1.201:5060 <http://4d52095c6f2ed1ab2de2d6c70340a126@10.0.1.201:5060>Jul 18 16:30:13 kam2 /usr/sbin/kamailio[69550]: INFO: <script>: UBICAR: requestm=INVITE reply=<null> SIP packet comes from the Asterisk Box duid=PBX2;my=pbx2;maxload=10000 callid=4d52095c6f2ed1ab2de2d6c70340a126@10.0.1.201:5060 <http://4d52095c6f2ed1ab2de2d6c70340a126@10.0.1.201:5060>Jul 18 16:30:18 kam2 /usr/sbin/kamailio[69547]: INFO: <script>: UBICAR: setting call load requesturi=<null> callid=896175650(a)192.168.14.76 <896175650(a)192.168.14.76>Jul 18 16:30:20 kam2 /usr/sbin/kamailio[69550]: INFO: <script>: UBICAR: delete call load requesturi=sip:10005@10.0.1.201:5060 <http://sip:10005@10.0.1.201:5060> callid=896175650(a)192.168.14.76 <896175650(a)192.168.14.76>Jul 18 16:30:38 kam2 /usr/sbin/kamailio[69548]: INFO: <script>: UBICAR: Forwarding non-registrar requests to AsteriskJul 18 16:30:38 kam2 /usr/sbin/kamailio[69548]: INFO: <script>: UBICAR: Dispatching to Asterisk Box sip:10.0.1.201:5060 <http://10.0.1.201:5060>Jul 18 16:30:42 kam2 /usr/sbin/kamailio[69559]: ALERT: <script>: UBICAR: Asterisk sip:10.0.1.201:5060 <http://10.0.1.201:5060> failed to answer, selecting other one!Jul 18 16:30:42 kam2 /usr/sbin/kamailio[69559]: INFO: <script>: UBICAR: Dispatching to Asterisk Box sip:10.0.1.200:5060 <http://10.0.1.200:5060>Jul 18 16:30:43 kam2 /usr/sbin/kamailio[69546]: ALERT: <core> [main.c:728]: handle_sigs(): child process 69559 exited by a signal 11Jul 18 16:30:43 kam2 /usr/sbin/kamailio[69546]: ALERT: <core> [main.c:731]: handle_sigs(): core was generatedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69546]: INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLDJul 18 16:30:43 kam2 /usr/sbin/kamailio[69563]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69562]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69561]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69558]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69557]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69556]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69555]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69554]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69553]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69552]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69551]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69548]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69547]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69549]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69550]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69560]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69546]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized* My configuration when applying load balancing and failover is detailed here: *# Send to Asteriskroute[TOASTERISK] {#!ifdef WITH_LOADBALANCE xlog("L_INFO","UBICAR: Forwarding non-registrar requests to Asterisk \n"); if(!ds_select_dst("0", "10")) { sl_send_reply("500", "Service Unavailable"); xlog("L_ERROR","UBICAR: No destinations available for $rd \n"); exit; } xlog("L_INFO","UBICAR: Dispatching to Asterisk Box $du\n"); t_set_fr(0,4000); t_on_failure("MANAGE_FAILURE"); route(RELAY); exit;#!endif* *# manage failure routing casesfailure_route[MANAGE_FAILURE] { #xlog("L_ALERT","UBICAR: failure requestm=$rm requesturi=$ru callid=$ci\n"); if (t_is_canceled()) { exit; }#!ifdef WITH_LOADBALANCE xlog("L_ALERT","UBICAR: Asterisk $du failed to answer, selecting other one!"); # next DST - only for 500 reply or local timeout (set by t_set_fr()) if (t_check_status("500") || t_branch_timeout() || !t_branch_replied()) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); #select the new destination if(ds_next_dst()) { t_set_fr(0,4000); t_on_failure("MANAGE_FAILURE"); xlog("L_INFO","UBICAR: Dispatching to Asterisk Box $du\n"); route(RELAY); exit; }else{ #last available node failed to reply, no other destinations available send_reply("404", "No destination"); exit; }#!endif }}* It may be a bug in that Kamailio version because i used Kamailio v4.2 before and i didnt have that inconvenient, still i had to upgrade to v4.3 because i needed the default_socket modparam that wasnt implemented in Kamailio v4.2 Any hint about this issue would be appreciate it Best Regards Alejandro

2 1

Managing concurrent ports
by Grant Bagdasarian 21 Jul '16

21 Jul '16

Hello, Are there any recommended modules for managing concurrent ports for different endpoints/customers/accounts/etc? I believe the dialog and dialog_ng modules do exactly this, but how is the performance of these modules under high loads? What is the release time of the ports? Are there perhaps other modules? Regards, Grant

3 6

Kamailio + rtpproxy + SRTP
by Konstantin M. 20 Jul '16

20 Jul '16

I have some problem with SSL+rtpproxy, kamailio successfully decodes SIP, rtpproxy decodes SRTP frames, all is going to freeswitch unencrypted. FS transfers the call back, the call is going unencrypted though kamailio, kamailio encodes to SSL, but rtpproxy DOESN'T encodes RTP frames to SRTP. Hence, endpoint SIP equipment rejects the call with 488 Not Acceptable Here Scenario 1. Endpoint device -> SRTP -> kamailio -> RTP -> freeswitch 2. Freeswitch calls back 3. freeswitch -> RTP -> kamailio -> PROBLEM (SIP[TLS] + RTP (but not SRTP)) -> Endpoint device 4. Endpoint device -> 488 Not Acceptable Here -> Kamailio Could somebody point me what's wrong here? Thanks!

2 1

WEBRTC Confusion (2)
by Zaka 20 Jul '16

20 Jul '16

Dear List: I am still stuck. Still unable to get Kamailio to listen on WEBRTC port* (in this case 8000)* Please advise if it has something to do with disabling TLS/MSRP? Please see the log & related config file in line here: *Log:* 0(5724) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WS_PORT!8000!g *and the result of config file check:* root@callcntr:/usr/local/etc/kamailio# kamailio -eE -ddd -cf /usr/local/etc/kamailio/websocket.cfg 0(5902) INFO: <core> [main.c:1911]: main(): private (per process) memory: 8388608 bytes 0(5902) DEBUG: <core> [mem/f_malloc.c:323]: fm_malloc_init(): fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 0(5902) DEBUG: <core> [mem/f_malloc.c:325]: fm_malloc_init(): fm_malloc_init: F_HASH_SIZE=2099, fm_block size=33912 0(5902) DEBUG: <core> [mem/f_malloc.c:327]: fm_malloc_init(): fm_malloc_init(0x7f9ca6e1f010, 8388608), start=0x7f9ca6e1f010 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac2700)[0] to 0 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac2680)[0] to 0 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac26c0)[0] to 0 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac2720)[0] to 0 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac26a0)[0] to 0 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac26e0)[0] to 0 0(5902) DEBUG: <core> [pvapi.c:1976]: pv_init_buffer(): PV print buffer initialized to [10][8192] 0(5902) DEBUG: <core> [main.c:2097]: main(): read 286773155 from /dev/urandom 0(5902) DEBUG: <core> [main.c:2103]: main(): seeding PRNG with 1755363211 0(5902) DEBUG: <core> [main.c:2107]: main(): test random numbers 726213135 321707966 991183401 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e3fc48 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !DBURL!mysql://kamailio:kamailiorw@localhost/kamailio!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: DBURL 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [DBURL] value [mysql://kamailio:kamailiorw@localhost/kamailio] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [DBURL]=[mysql://kamailio:kamailiorw@localhost/kamailio] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e3ffe8 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_IP_ADDR!192.168.1.16!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_IP_ADDR 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_IP_ADDR] value [192.168.1.16] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_IP_ADDR]=[192.168.1.16] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e40320 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_DOMAIN!callcntr.com.al!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_DOMAIN 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_DOMAIN] value [callcntr.com.al] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_DOMAIN]=[callcntr.com.al] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e40658 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WS_PORT!8000!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_WS_PORT 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_WS_PORT] value [8000] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_WS_PORT]=[8000] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e40988 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WSS_PORT!443!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_WSS_PORT 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_WSS_PORT] value [443] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_WSS_PORT]=[443] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e40cb8 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_MSRP_PORT!9000!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_MSRP_PORT 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_MSRP_PORT] value [9000] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_MSRP_PORT]=[9000] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=0 0(5902) DEBUG: <core> [re.c:455]: subst_run(): matched (16, 10): [MY_IP_ADDR] 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [ppcfg.c:192]: pp_subst_run(): preprocess subst applied [#1] to [!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g] - returning new string [!MY_WS_ADDR!tcp:192.168.1.16:MY_WS_PORT!g] 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=0 0(5902) DEBUG: <core> [re.c:455]: subst_run(): matched (29, 10): [MY_WS_PORT] 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [ppcfg.c:192]: pp_subst_run(): preprocess subst applied [#2] to [!MY_WS_ADDR!tcp:192.168.1.16:MY_WS_PORT!g] - returning new string [!MY_WS_ADDR!tcp:192.168.1.16:8000!g] 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e41118 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WS_ADDR!tcp:192.168.1.16:8000!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_WS_ADDR 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_WS_ADDR] value [tcp:192.168.1.16:8000] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_WS_ADDR]=[tcp:192.168.1.16:8000] (0) 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: WITH_WEBSOCKETS 0(5902) DEBUG: <core> [cfg.lex:1795]: pp_define_get(): ### returning define ID [MY_IP_ADDR] value [192.168.1.16] 0(5902) DEBUG: <core> [cfg.lex:1795]: pp_define_get(): ### returning define ID [MY_IP_ADDR] value [192.168.1.16] loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(5902) DEBUG: <core> [ppcfg.c:223]: pp_ifdef_level_check(): same number of pairing preprocessor directives #!IF[N]DEF - #!ENDIF 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 0: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_WARN" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "403" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "REQINIT" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:421]: print_action(): force_rport( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<0> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e50ff0> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_ERR" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "400" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "WITHINDLG" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e42f78> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e43950> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "AUTH" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4c350> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:543]: print_action(): , type<22> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e44968> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "REGISTRAR" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "484" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "LOCATION" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "RELAY" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 1: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e44460> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 2: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "483" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d7d0> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "Malformed SIP message from $si:$sp " 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4cdb0> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 3: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_ERR" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "400" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "RELAY" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e42f78> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "404" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 4: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e44460> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 5: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e42a70> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e44428> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "404" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:442]: print_action(): assign( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<21> 0(5902) DEBUG: <core> [route_struct.c:517]: print_action(): , type<22> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e42a70> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:448]: print_action(): UNKNOWN( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:517]: print_action(): , type<24> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 6: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4e8b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "$fd" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4e810> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "403" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): onreply routing table 0: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_WARN" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 0 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 9 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e50e68> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): event routing table 1: 0(5902) DEBUG: <core> [route_struct.c:448]: print_action(): UNKNOWN( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<0> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:448]: print_action(): UNKNOWN( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<0> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_WARN" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e50660> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 4 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "403" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_DBG" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_WARN" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e50660> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 4 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "403" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e50660> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 4 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "404" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): event routing table 2: 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_INFO" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac26e0)[core:receive-parse-error] to 3 0(5902) INFO: <core> [sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: 192.168.1.16:5060 tcp: 192.168.1.16:5060 Aliases: tcp: callcntr:5060 udp: callcntr:5060 config file ok, exiting... root@callcntr:/usr/local/etc/kamailio# *Config File:* #!KAMAILIO # # Simple/sample kamailio.cfg for running a proxy/registrar with TLS and # WebSockets support. #!substdef "!DBURL!mysql://kamailio:kamailiorw@localhost/kamailio!g" #!substdef "!MY_IP_ADDR!192.168.1.16!g" #!substdef "!MY_DOMAIN!callcntr.com.al!g" #!substdef "!MY_WS_PORT!8000!g" #!substdef "!MY_WSS_PORT!443!g" #!substdef "!MY_MSRP_PORT!9000!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" ##!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" ##!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g" ##!substdef "!MSRP_MIN_EXPIRES!1800!g" ##!substdef "!MSRP_MAX_EXPIRES!3600!g" ##!define LOCAL_TEST_RUN ##!define WITH_TLS #!define WITH_WEBSOCKETS ##!define WITH_MSRP ####### Global Parameters ######### fork=yes children=4 #!ifdef WITH_TLS enable_tls=1 #!endif listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_IP_ADDR #!ifdef WITH_TLS #listen=MY_WSS_ADDR #!endif #!endif #!ifdef WITH_MSRP listen=MY_MSRP_ADDR #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 #!ifdef LOCAL_TEST_RUN debug=2 mpath="modules" #!else debug=0 mpath="/usr/local/lib64/kamailio/modules/" #!endif loadmodule "db_mysql.so" loadmodule "tm.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "kex.so" loadmodule "mi_rpc.so" loadmodule "corex.so" #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_MSRP loadmodule "msrp.so" loadmodule "htable.so" loadmodule "cfgutils.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" #!endif #!ifdef WITH_WEBSOCKETS # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is _only_ being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { * if (($Rp == MY_WS_PORT)) { * *#### I deleted/ modified this logical expression ####* xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } # per request initial checks route(REQINIT); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE")) { record_route(); } # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484", "Address Incomplete"); exit; } # user location service route(LOCATION); route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks 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; } if (uri == myself && is_method("OPTIONS") && !(uri=~"sip:.*[@]+.*")) { options_reply(); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(RELAY); } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction... # ignore and discard exit; } } sl_send_reply("404", "Not Found"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } } # USER location service route[LOCATION] { if (!is_subscriber("$ru", "subscriber", "1")) { t_newtran(); send_reply("404", "Not Found"); exit; } if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: send_reply("480", "Temporarily Unavailable"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; case -3: send_reply("500", "Server Internal Error"); exit; } } } # Authentication route route[AUTH] { if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER")) { consume_credentials(); } } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403", "Forbidden"); exit; } } #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT) && !(proto == WS)) || $Rp == MY_MSRP_PORT) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif #!ifdef WITH_MSRP event_route[msrp:frame-in] { msrp_reply_flags("1"); if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) && $Rp != MY_MSRP_PORT) { xlog("L_WARN", "MSRP request received on $Rp\n"); msrp_reply("403", "Action-not-allowed"); exit; } if (msrp_is_reply()) { msrp_relay(); } else if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } if (!www_authenticate("MY_DOMAIN", "subscriber", "$msrp(method)")) { if (auth_get_www_authenticate("MY_DOMAIN", "1", "$var(wauth)")) { msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } if ($hdr(Expires) != $null) { $var(expires) = (int) $hdr(Expires); if ($var(expires) < MSRP_MIN_EXPIRES) { msrp_reply("423", "Interval Out-of-Bounds", "Min-Expires: MSRP_MIN_EXPIRES\r\n"); exit; } else if ($var(expires) > MSRP_MAX_EXPIRES) { msrp_reply("423", "Interval Out-of-Bounds", "Max-Expires: MSRP_MAX_EXPIRES\r\n"); exit; } } else { $var(expires) = MSRP_MAX_EXPIRES; } $var(cnt) = $var(cnt) + 1; pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)"); $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr); $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock); $shtex(msrp=>$var(sessid)) = $var(expires) + 5; # - Use-Path: the MSRP address for server + session id $var(hdrs) = "Use-Path: msrps://MY_IP_ADDR:MY_MSRP_PORT/" + $var(sessid) + ";tcp\r\n" + "Expires: " + $var(expires) + "\r\n"; msrp_reply("200", "OK", "$var(hdrs)"); } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") { if ($msrp(nexthops)>1) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay(); exit; } $var(sessid) = $msrp(sessid); if ($sht(msrp=>$var(sessid)::srcaddr) == $null) { # one more hop, but we don't have address in htable msrp_reply("481", "Session-does-not-exist"); exit; } else if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay_flags("1"); msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)", "$sht(msrp=>$var(sessid)::srcsock)"); msrp_relay(); } else { msrp_reply("501", "Request-method-not-understood"); } } #!endif Thanks a million! Zaka One accurate measurement is worth more than a thousand expert opinions!

2 4

reading siptrace trace_on current value in script
by ananda agrawal 20 Jul '16

20 Jul '16

Hi, Is it possible to check wether the trace_on for siptrace module is on or off from within the config script, e.g inside routing logic, thannks in advance -Anand

1 0

Crocodile-msrp
by Marcelo Pignataro 19 Jul '16

19 Jul '16

Hi, I don't really know if I'm writing to the right list, sorry if I don't. Has someone been using crocodile-msrp library? I need to develop a webchat app with Kamailio together and I found Crocodile, but I see it hasn't documentation neither guides to developers. I also realize it hasn't active community, last commit done to crocodile-msrp library by its developers was 2013. I need some information to know if I should use it or not. Thank you and best regards, Marcelo

1 0

Push Notifications
by Nahum Nir 18 Jul '16

18 Jul '16

Hello, Found the presentation about asynch push. Is there an example cfg file? Thanks, Nir

3 6

Custom header on reply is removed on t_relay
by David Wilson 18 Jul '16

18 Jul '16

Hello All, We have a Kamailio proxy (P1) and a custom client (A). Calls are working reliably with A communicating directly with P1. When A sends INVITE to P1, in some cases, we insert a custom header ‘X-Routing’ in the reply. Now, testing with a new, additional proxy (P2) between these two: A -> P2 -> P1 A sends INVITE via P2 to P1, P1 replies include X-Routing header field (confirmed by logging message buffer received from P1 by P2). However, the reply relayed from P2 to A does not contain the custom header. P2’s reply_route block only contains logging code and I’ve checked for calls to remove_hf. I can’t find documentation or code to explain why the header is omitted. Is there a module config parameter that I can set to preserve custom headers? Any suggestions welcome! Regards, Dave Wilson.

2 1

Rx Interface connected, get error code: SIP/2.0 403 Forbidden - You must register first with a S-CSCF
by Kristian Martens 18 Jul '16

18 Jul '16

Hi, I have installed Kamailio version 4.4.1. Everything works just fine. If I connect a PCRF via Rx to the P-CSCF, the registration still runs through as expected but the subsequent SUBSCRIBE is rejected with "403 Forbidden - You must register first with a S-CSCF" error cause or is simply ignored (see attached wireshark traces). And, if I disconnect Rx and restart the P-CSCF, the "forbidden" problem still persists. (But I can work around this by cleaning up the P-CSCF database.) Any suggestions how I could get Rx working? Thanks, Kristian

3 7

kamailio-dbg package
by Laura García Pérez 18 Jul '16

18 Jul '16

Is it possible to install kamailio-dbg in CentOS? Thanks, Laura García Pérez www.logslie.com <http://www.logslie.com/> +34680923532 +12602069698

2 2

Conditionally calling TOPOS
by Nathan Ward 18 Jul '16

18 Jul '16

Hi, I think it make sense to put this topic in a seperate message, though this is very much the same scenario mentioned here: http://lists.sip-router.org/pipermail/sr-users/2016-July/093864.html I would like the ability initiate topology hiding only for certain messages - i.e. on inbound calls, I would like my Kamailio ‘SBC' to retain the routing information for inbound requests, but strip it for outgoing requests - so that I can see the full path a call has taken to reach and then traverse my network all the way to an egress SBC but then this is all stripped before forwarding the call to a 3rd party (i.e. customer, or carrier). I note that this isn’t possible at present, but I’d like to take a stab at the code, perhaps designating ‘trusted’ IPs for which topology hiding doesn’t take place. Before I embark on this, is this something that would be considered useful, or is there a good reason to have full topology hiding that I’m missing? -- Nathan Ward

2 1

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

2 1

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

3 3

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

1 0

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

2 2

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

1 0

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!

2 2

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

2 2

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

2 1

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

2 5

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

2 1

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

1 0

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

2 3

tsilo ts_append problem
by gmele 08 Jul '16

08 Jul '16

Hello, we are using tsilo to achieve mobile push notification in kamailio. In tsilo module documentation, ts_append() function is used with the pseudo variable “$tU” holding the username of the to-uri. Our config looks like : route[PUSHJOIN] { $var(hjoin) = 0; lock("$tu"); $var(hjoin) = $sht(vtp=>join::$tu); $var(hstored) = $sht(vtp=>stored::$tu); $sht(vtp=>join::$tu) = $null; unlock("$tu"); if ($var(hjoin)==0) { if ($var(hstored)) { ts_append("location", "$tU"); } return; } $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); t_continue("$var(id_index)", "$var(id_label)", "INVRESUME"); } In ts_append() function code, ts_check_uri() is called on a variable holding a username and not a URI causing the failure of the function ts_check_uri() and resulting in ts_append() to exit. The following line is displayed in the log: [tsilo.c:205]: ts_check_uri(): bad uri [giovanni.mele]. The function ts_store() stores in the R-URI field the username and not the request URI. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay(PROXY_IP, PROXY_PORT)) { sl_reply_error(); } if (is_method("INVITE")) { ts_store(); $sht(vtp=>stored::$tu) = 1; route(SENDPUSH); } exit; } route[INVRESUME] { statsd_gauge("SIPRegistrar.InviteResumed", "+1"); lookup("location"); t_relay(PROXY_IP, PROXY_PORT); ts_store(); $sht(vtp=>stored::$tu) = 1; } With the command “kamcmd ts.dump” we can see the following lines: R-URIs: { R-URI: giovanni.mele Hash: 74014921 Transactions: { Transaction: { Tindex: 63704 Tlabel: 2105932514 } } } If we try to use the ts_append() function with the pseudo variable “$tu” holding the full to-uri address, there is no error returned by ts_check_uri() (as this is called on a correct URI) but the transaction corresponding to the to-uri cannot be found (maybe because the R-URI stored by ts_store() is the username and not the full R-URI). How can we make things work ? By storing the full R-URI with ts_store() ? Thx for your help Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/tsilo-ts-append-problem-tp150184.ht… Sent from the Users mailing list archive at Nabble.com.

3 10

4.4.1: Only first UDP Listener is producing log messages
by Sebastian Damm 08 Jul '16

08 Jul '16

Hi, I upgraded one of our systems today to 4.4.1. And when checking whether everything is okay, I noticed, that only the first UDP listener (from kamcmd core.ps) is actually sending lines to the log. The only other logs I see are from usrloc timer worker. kamcmd> core.ps 10737 main process - attendant 10740 udp receiver child=0 sock=172.20.40.5:5060 10741 udp receiver child=1 sock=172.20.40.5:5060 10742 udp receiver child=2 sock=172.20.40.5:5060 10743 udp receiver child=3 sock=172.20.40.5:5060 We are sending quite a lot of information via xlog("L_NOTICE", "foo"); to syslog, but the only lines I see are from PID 10740. Kamailio works fine otherwise, handling all requests, the usrloc entries didn't drop, and customers don't complain. So I think it's only a logging issue. There are about 50k registrations on the machine and more than 10 Mbit of traffic. When I downgrade to 4.3.5, I get output from all listeners again. Has there been any change between 4.3 and 4.4 regarding logging? The upgrade document says nothing about this. Best Regards, Sebastian

2 4

CNXCC compilation errors
by Infinicalls Infinicalls 07 Jul '16

07 Jul '16

Hi, Thanks. Well I tried it before also but could not proceed with since I got the compilation errors. Now also I am struck up with the same error mentioned in http://lists.sip-router.org/pipermail/sr-users/2016-January/091574.html Has anybody found a solution to this? Mine is Ubuntu 12.x LTS regards Ganesh Kumar On 7/7/16, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > look at cnxcc module for implementing a prepaid system. There should be > also a presentation about it at one of the kamailio world conferences -- > see kamailio.org/events/ > > Cheers, > Daniel > > > On 07/07/16 11:33, Infinicalls Infinicalls wrote: >> Hi Daniel, >> >> Thanks. I figured out later :-) >> >> I am actually trying to implement prepaid system with dialog and call >> control. In Siremis, after a call the duration and other details get >> updated but not the cost. This is inspite of having rates and >> kamailio_cdr / kamailio_rating() in cron. >> >> Any idea? >> >> regards >> Ganesh Kumar >> >> On 7/7/16, Daniel Tryba <d.tryba(a)pocos.nl> wrote: >>> On Wed, Jul 06, 2016 at 11:01:17PM +0530, Infinicalls Infinicalls wrote: >>>> " ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module >>>> matching <call_control> found >>>> " >>>> while adding modparam("call_control", "disable", 1). I have already >>>> included pv and dialog in the modules list. >>>> >>>> Any idea? >>> Try loading the module first :) >>> >>> https://www.kamailio.org/wiki/cookbooks/4.4.x/core#loadmodule >>> >>> >>> _______________________________________________ >>> 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 >>> >> _______________________________________________ >> 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 > > > _______________________________________________ > 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 > -- -- http://www.infinicalls.com

1 0

A parameter with a life span of the entire dialog
by Jay Li 07 Jul '16

07 Jul '16

Dear All, I'm trying to set a few parameters for an "INVITE", and access their values in the failure routine if the call fails. $var() doesn't seem suitable for my purpose as it only lasts for one process. I think if I set a $var() in "INVITE", it will be gone when I receive the failure response. $sht() seems able to last long but it will be accessed by all processes which might include ones not belonging to the same dialog. I wonder if somebody would give me some suggestions here to handle such situations. Does Kamailio provide a mechanism to set up a parameter with a life span of the entire dialog? Thanks. Regards,Jay

2 1

call_control not found error
by Infinicalls Infinicalls 07 Jul '16

07 Jul '16

Hi, I am trying to implement a prepaid system and get this error " ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <call_control> found " while adding modparam("call_control", "disable", 1). I have already included pv and dialog in the modules list. Any idea? regards Ganesh Kumar

3 3

t_check_trans() exits route execution if transaction is found
by Sebastian Damm 07 Jul '16

07 Jul '16

Hi, on one of our systems we are checking a few times whether a transaction already exists. We noticed that route execution stops if the transaction is found. And after looking into the code, we found that it seems to be intended like this. >From tm.c: int t_check_trans(struct sip_msg* msg) { if (msg->first_line.type==SIP_REPLY) { [..] } else { switch(t_check_msg(msg, 0)){ [..] case 1: /* found */ [..] return 0; /* exit from the script */ } } } Can anybody explain, why script execution should stop when a running transaction is found? I would expect that then the function returns 1 so I can use it in my if statement and continue with my routing script. Thanks already for an explanation, Sebastian

2 1

Kamailio compact headers ?!
by Anonim Stefan 07 Jul '16

07 Jul '16

Hi, Is there a parameter to be enabled such that Kamailio will append headers in compacted form? (I'm thinking of Via: and Record-Route:) Thanks, Stefan

6 10

Kamailio and Mongodb
by C. 06 Jul '16

06 Jul '16

Hi, I am still trying to get Kamailio to work with MongoDB. Followed instructions here https://www.kamailio.org/wiki/tutorials/kamailio-and-mongodb <https://www.kamailio.org/wiki/tutorials/kamailio-and-mongodb>, but I cannot get things to install. Are there any instructions on the exact configuration changes that need to be made to support Mongodb? For example, kamctl points to MySQL, etc Thanks! Chuks.

1 0

Locally originated BYE dialog database handle
by Dirk Teurlings - Signet B.V. 06 Jul '16

06 Jul '16

Hi, There's something going on with a locally originated BYE and the dialog module running with a database backend. Here's the scenario. Call gets setup, and is being monitored by outside processes (non-Kamailio). After a while these processes determine the call is taking too long (or too expensive), so it has to be killed. To do this we're using the kamcmd dlg.end_dlg with h_entry and h_id as parameter. This works to end the call, but is throws some weird issues into the routing. First off, the event_route[dialog:end] is not thrown 99% of the time, though I would expect it to. Second, This error pops up in the log /usr/sbin/kamailio[25094]: ERROR: dialog [dlg_db_handler.c:199]: use_dialog_table(): invalid database handle As if the command issued by kamcmd isn't able to get connected to the dialog database. In any case, this error results in the dialog entry for this call not being removed from the database. So this piles up after a while. Anybody any thoughts on this? Cheers, Dirk

2 2

(no subject)
by John Petrini 06 Jul '16

06 Jul '16

Hello, I'm trying to append a branch, assign it a q value so it has priority and serialize the new branch and the main branch. The goal is to send the new branch off to our CNAM provider, capture the CNAM in their reply to an $avp then drop the branch, stick the cnam in the main branch and allow the call to continue as normal. We do this currently using OpenSIPS with serialize_branches(1) but the of load_contacts() behavior seems to be different than serialize_branches. route[CNAM_DIPS] t_on_branch("CNAM_DIPS"); t_on_reply("CNAM_DIPS"); t_on_failure("CNAM_DIPS"); t_load_contacts(); t_next_contacts(); append_branch("sip:2223334444@22.33.444.55:5600;trans-type=5", "0.5"); relay(); break; Two problems. First when t_next_contacts is called I get the following error: ERROR: pv [pv_branch.c:58]: pv_get_branchx(): error accessing branch [0] So it looks like the branches aren't actually being loaded? Second problem is that despite the first error my call goes out on the branch I appended and I get a reply from the CNAM provider but Kamailio forwards that reply back to the phone causing the phone to send a second invite. In OpenSIPs we just called drop, to drop the branch and let the main branch continue but this doesn't seem to work in Kamailio. I've read and reread the documentaion but it doesn't seem to behave the way the documentation says it should. Is there something I'm missing? Is there a better way to accomplish this in Kamailio? Thank You, ___ John Petrini

3 9

Issue with userblacklist module after 4.4.2 upgrade
by Tim Balmer 06 Jul '16

06 Jul '16

Hello, We are running Kamailio version 4.4 and have our sipproxy servers configured to do auto updates within te same major version based on a 4 weekly schedule (we have 4 sipproxy / Kamailio servers). So this morning as expected the first of our sipproxy servers automatically upgraded from 4.4.1 to 4.4.2, unfortunately after this update we have issues with the userblacklist module. The problem is that with the 4.4.2 release every number is now matching our userblacklists although it is not really blacklisted. After some digging in the logs I found this is probably because the userblacklist module is looking for the wrong database table. With every check_user_blacklist call I see errors like this appearing in the logs: Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Table 'kamailio.<number>' doesn't exist (1146) Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: userblacklist [db.c:68]: db_build_userbl_tree(): error while executing query. Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: userblacklist [userblacklist.c:277]: check_user_list(): cannot build d-tree I replaced the real number with <number> in the log output but it seems like the userblacklist module is now looking for a database table with the name of the number to check against. On our development server I tested setting the global userblacklist_table parameter and also passing the table parameter within the check_user_blacklist call but both are not making any difference. After downgrading the sipproxy back to 4.4.1 everything is working as expected again so it looks to me like a bug in the 4.4.2 release. Is anyone else experiencing the same issues after the 4.4.2 upgrade or is it just me doing something wrong? Best Regards, Tim Balmer

2 12

Kamailio summer-of-documentation - join the effort
by Olle E. Johansson 06 Jul '16

06 Jul '16

Hi! Looking for a summer-of-documentation project? Which README is the best? Which one is so full of Swedish that you totally fail to parse it? Check our documentation pages, suggest improvements on the mailing lists (or here on Facebook) and give us feedback! This is something you can spend your lazy summer days on - a nice cool drink, cool sunshades and Kamailio documentation! We constantly work to update our README files for every module and the cookbooks for the Kamailio core, the transformations and pseudovariables. But we’re developers, not casual administrators and propably fail to see some obvious things missing. Please tell us (feel free to use the mailing lists). And if you can - but not required at all - you can suggest rewrites in the XML source and upload to github. Let’s keep our docs up to date and make sure they really assist our users! http://kamailio.org/docs/modules/devel/ Have a great summer! Regards, /O

2 1

Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail
by tegamjg 06 Jul '16

06 Jul '16

El 05/07/2016 11:36, Daniel Tryba <d.tryba(a)pocos.nl> escribió: > > Please keep the mailinglist in the loop, so everybody might benefit from > our ramblings :) > > > Still there are few things i dont understand, i am not using asterisk > > just as a voicemail server since they are actually handling also the > > calls passing first from kamailio and being load balanced to those > > asterisk boxes. May i still use call forwarding as you are using it? > > (Both asterisk have a shared storage with a clustered filesystem, so > > both will be able to see voice messages) > > Yes I think so. I use a seperate machine for voicemail but I see no > problem with other uses (I used to use it for playback of messages and > transcoding ebtween incompatible endpoints). > > By using the prefixes in kamailio to the username in $ru I have in the > extensions.conf: > > exten => _tovm-.,1,NoOp(leave voicemail) > exten => _tovm-.,n,Answer() > exten => _tovm-.,n,Set(CHANNEL(language)=nl) > exten => _tovm-.,n,Voicemail(${EXTEN:5},us) > exten => _tovm-.,n,Playback(Goodbye) > exten => _tovm-.,n,Hangup() > > exten => _getvm-.,1,NoOp(read voicemail) > exten => _getvm-.,n,Set(CHANNEL(language)=nl) > exten => _getvm-.,n,VoicemailMain(${EXTEN:6}) > exten => _getvm-.,n,Hangup() > > > The other question is that i actually though that you need asterisk to > > have users configured in sipusers realtime table to associate their > > mailboxes, which i dont have since those users are stored in the > > subscriber table of kamailio. So am i still able to configure > > voicemail like you are doing it by syncing with the voicemail table?, > > i really hope so haha > > I forgot that fact. So yes I have a realtime sip users list (with > host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio > username, no password (this machine is not directly accessible from > outside)) Sorry, I think i rushed the last answer but if you could answer that one would be nice How are you handling the calls? Just with kamailio/rtpproxy? Because i am also using asterisk for calls with dial application and for nat issues (with kamailio behind nat) i am using also kamailio/rtpproxy for outside. All this with just handling users (registration and location) in the subscribe and location table of kamailio. That is why i am not using sipusers table of asterisk because of nat was behaving weird using it that way. Could it be possible to use both tables without expecting a different behaviour? Or is not, in the end, a good idea and i need to keep users in sipusers table? > > You might not be able to have endpoints able to subscribe to > notifications due to this. I baked something inspired by: > http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.… > that appears to work for me. > > _______________________________________________ > 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

2 1

Kamailio out-of-mem
by José Seabra 06 Jul '16

06 Jul '16

I'm running burning tests against kamailio and after some time ~ 2 days I see kamailio out of mem: ERROR: tm [t_lookup.c:1271]: new_t(): ERROR: new_t: out of mem: ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed The result of command kamcmd mod.stats all is: Module: core { sip_msg_shm_clone(494): 10653392 build_req_buf_from_sip_req(2149): 2565480 msg_lump_cloner(978): 2943584 create_avp(175): 969736 xavp_new_value(94): 858469320 counters_prefork_init(207): 16896 cfg_clone_str(130): 88 cfg_shmize(217): 744 init_pt(110): 8 init_pt(105): 8 init_pt(104): 3080 dtrie_insert(157): 240 dtrie_insert(148): 48 dtrie_init(60): 160 dtrie_init(51): 32 register_timer(1011): 360 cfg_register_ctx(47): 32 init_tcp(4635): 8192 init_tcp(4629): 32768 init_tcp(4621): 8 init_tcp(4614): 8 init_tcp(4607): 8 init_tcp(4601): 8 init_tcp(4589): 8 init_avps(90): 8 init_avps(89): 8 init_dst_blacklist(437): 16384 init_dst_blacklist(430): 8 timer_alloc(514): 96 init_dns_cache(366): 8 init_dns_cache(358): 16384 init_dns_cache(351): 16 init_dns_cache(345): 8 init_timer(283): 8 init_timer(282): 16384 init_timer(281): 8 init_timer(280): 8 init_timer(269): 8 init_timer(237): 8 init_timer(220): 278544 init_timer(219): 8 init_timer(206): 8 cfg_child_cb_new(830): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 shm_core_lock_init(153): 8 Total: 875992224 } Module: dmq { alloc_job_queue(229): 128 shm_str_dup(723): 24 build_dmq_node(156): 448 add_peer(67): 168 mod_init(243): 8 mod_init(237): 96 init_dmq_node_list(66): 24 init_peer_list(33): 24 Total: 920 } Module: auth_db { Total: 0 } Module: dialog { init_dlg_table(284): 131088 init_dlg_table(276): 8 init_dlg_table(269): 8 init_dlg_table(264): 8 init_dlg_timer(57): 8 init_dlg_timer(48): 32 new_dlg_profile(187): 624 Total: 131776 } Module: rr { Total: 0 } Module: xlog { Total: 0 } Module: kex { pkg_proc_stats_init(79): 1056 Total: 1056 } Module: db_postgres { lock_set_alloc(70): 80 Total: 80 } Module: pv { Total: 0 } Module: textops { Total: 0 } Module: htable { ht_cell_new(183): 1104 ht_init_tables(381): 8388608 ht_add_table(278): 928 Total: 8390640 } Module: sqlops { Total: 0 } Module: tm { build_uac_req(1543): 848 prepare_new_uac(479): 28880 _reply_light(542): 751504 build_local_reparse(341): 1168568 relay_reply(1884): 643088 build_cell(317): 9826624 prepare_new_uac(524): 8112 prepare_new_uac(509): 10192 init_tm_stats_child(60): 5632 insert_tmcb(137): 136 init_tmcb_lists(74): 16 init_tmcb_lists(72): 16 init_hash_table(467): 2097152 Total: 14540768 } Module: usrloc { new_urecord(65): 136 new_ucontact(98): 1096 new_urecord(58): 288 shm_str_dup(723): 1200 build_stat_name(51): 64 new_udomain(93): 40960 new_udomain(86): 48 new_dlist(573): 16 new_dlist(565): 32 register_ulcb(94): 32 init_ulcb_list(45): 16 lock_set_alloc(70): 4112 Total: 48000 } Module: tmx { tmx_check_pretran(250): 864 tmx_check_pretran(271): 1216 tmx_init_pretran_table(90): 256 Total: 2336 } Module: sl { init_sl_stats_child(125): 4224 sl_startup(83): 8 init_sl_stats(110): 8 Total: 4240 } Module: carrierroute { rule_fixup_recursor(584): 8 shm_str_dup(723): 56 add_route_rule(74): 152 add_route_flags(225): 48 create_domain_data(83): 32 create_carrier_data(59): 8 create_carrier_data(50): 64 load_route_data_db(295): 8 load_domain_map(182): 48 load_domain_map(171): 24 load_carrier_map(126): 48 load_carrier_map(115): 24 reload_route_data(172): 64 init_route_data(74): 8 Total: 592 } Module: dispatcher { reindex_dests(600): 1008 add_dest2list(350): 168 add_dest2list(324): 4200 init_data(204): 16 init_data(195): 16 mod_init(309): 8 mod_init(307): 8 ds_ping_active_init(102): 8 Total: 5432 } Module: db_text { dbt_init_cache(81): 256 dbt_init_cache(69): 8 dbt_init_cache(54): 8 Total: 272 } The result of command kamctl shmem is: shmem:fragments = 6353 shmem:free_size = 406208 shmem:max_used_size = 8464105472 shmem:real_used_size = 8463699264 shmem:total_size = 8464105472 shmem:used_size = 5190698208 The server has 8G of shared memory allocated, and the issue happens more or less after 2 days running tests. I'm using sipp to generate traffic, it is sending 500CAPS. There is any possibility of memory leak on Kamailio 4.4.0? Best Regards -- José Seabra

2 11

Kamailio blocked when using with Radius accounting when connection is lost with the database.
by Mx R 06 Jul '16

06 Jul '16

Hi there. I have the following scheme. 1. SIP server is running kamailio with module "acc", radiusclient-ng and freeradius. 2. DB server with MSSQL. The problem is the following, if the connection is lost with the database, no one can make calls. Even after the connection to the database is restored. We have to restart the radius and kamailio. And only after that can make calls. Kamailio logs rc_send_server: no reply from RADIUS server localhost: 1813 ERROR: acc [acc.c: 720]: acc_rad_request (): radius-ing failed Radius logs. Error: rlm_sql_unixodbc: SQL down 08S01 [unixODBC] [FreeTDS] [SQL Server] Write to the server failed Below are links to a similar problem. http://lists.opensips.org/pipermail/users/2012-March/021100.html http://lists.opensips.org/pipermail/users/2012-March/021102.html http://lists.opensips.org/pipermail/users/2012-March/021105.html http://lists.opensips.org/pipermail/users/2012-March/021128.html

1 0

Simple Kamailio configuration for Asterisk load balancing?
by Tickling Contest 06 Jul '16

06 Jul '16

Hello, I am beginning to front my Asterisk cluster with Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for this. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for Kamailio with the following features: (a) Support an arbitrarily large number of Asterisk servers (say, upto 10). (b) Offload SIP registration to a realtime/mysql DB used by the PBXs. Please also let me know if I should have to change my Asterisk PBX config in any way for this to happen. On a very related note, while I appreciate the fact that you need to _really_ understand SIP to configure Kamailio, it should be possible to get this setup by just running a script or GUI. It is difficult to find information on how to load balance multiple PBXs behind Kamailio. Thanks!

3 5

[rtprpoxy LB]
by Noriyuki Hayashi 06 Jul '16

06 Jul '16

Hi, Thank you for be used great KAMAILIO. Regarding rtpproxy LB. I configured as below. modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.136:7722") modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.137:7722") modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.138:7722") modparam("rtpproxy", "rtpproxy_disable_tout", 20) modparam("rtpproxy", "rtpproxy_tout", 2) modparam("rtpproxy", "rtpproxy_retr", 2) I could not restart kamailio when one rtpproxy server is down. Does anyone tell this solution by the parameter or somthing ? I am looking forward from anyone. Thank you so much. Noriyuki Hayashi

1 0

Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail
by tegamjg 05 Jul '16

05 Jul '16

El 05/07/2016 11:36, Daniel Tryba <d.tryba(a)pocos.nl> escribió: > > Please keep the mailinglist in the loop, so everybody might benefit from > our ramblings :) > > > Still there are few things i dont understand, i am not using asterisk > > just as a voicemail server since they are actually handling also the > > calls passing first from kamailio and being load balanced to those > > asterisk boxes. May i still use call forwarding as you are using it? > > (Both asterisk have a shared storage with a clustered filesystem, so > > both will be able to see voice messages) > > Yes I think so. I use a seperate machine for voicemail but I see no > problem with other uses (I used to use it for playback of messages and > transcoding ebtween incompatible endpoints). > > By using the prefixes in kamailio to the username in $ru I have in the > extensions.conf: > > exten => _tovm-.,1,NoOp(leave voicemail) > exten => _tovm-.,n,Answer() > exten => _tovm-.,n,Set(CHANNEL(language)=nl) > exten => _tovm-.,n,Voicemail(${EXTEN:5},us) > exten => _tovm-.,n,Playback(Goodbye) > exten => _tovm-.,n,Hangup() > > exten => _getvm-.,1,NoOp(read voicemail) > exten => _getvm-.,n,Set(CHANNEL(language)=nl) > exten => _getvm-.,n,VoicemailMain(${EXTEN:6}) > exten => _getvm-.,n,Hangup() > > > The other question is that i actually though that you need asterisk to > > have users configured in sipusers realtime table to associate their > > mailboxes, which i dont have since those users are stored in the > > subscriber table of kamailio. So am i still able to configure > > voicemail like you are doing it by syncing with the voicemail table?, > > i really hope so haha > > I forgot that fact. So yes I have a realtime sip users list (with > host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio > username, no password (this machine is not directly accessible from > outside)) That is what i was afraid of, so you are not handling kamailio users in the subscribe table of kamailio database but in sipusers table of asterisk database, is that correct? > > You might not be able to have endpoints able to subscribe to > notifications due to this. I baked something inspired by: > http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.… > that appears to work for me. > > _______________________________________________ > 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

Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail
by Daniel Tryba 05 Jul '16

05 Jul '16

Please keep the mailinglist in the loop, so everybody might benefit from our ramblings :) > Still there are few things i dont understand, i am not using asterisk > just as a voicemail server since they are actually handling also the > calls passing first from kamailio and being load balanced to those > asterisk boxes. May i still use call forwarding as you are using it? > (Both asterisk have a shared storage with a clustered filesystem, so > both will be able to see voice messages) Yes I think so. I use a seperate machine for voicemail but I see no problem with other uses (I used to use it for playback of messages and transcoding ebtween incompatible endpoints). By using the prefixes in kamailio to the username in $ru I have in the extensions.conf: exten => _tovm-.,1,NoOp(leave voicemail) exten => _tovm-.,n,Answer() exten => _tovm-.,n,Set(CHANNEL(language)=nl) exten => _tovm-.,n,Voicemail(${EXTEN:5},us) exten => _tovm-.,n,Playback(Goodbye) exten => _tovm-.,n,Hangup() exten => _getvm-.,1,NoOp(read voicemail) exten => _getvm-.,n,Set(CHANNEL(language)=nl) exten => _getvm-.,n,VoicemailMain(${EXTEN:6}) exten => _getvm-.,n,Hangup() > The other question is that i actually though that you need asterisk to > have users configured in sipusers realtime table to associate their > mailboxes, which i dont have since those users are stored in the > subscriber table of kamailio. So am i still able to configure > voicemail like you are doing it by syncing with the voicemail table?, > i really hope so haha I forgot that fact. So yes I have a realtime sip users list (with host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio username, no password (this machine is not directly accessible from outside)) You might not be able to have endpoints able to subscribe to notifications due to this. I baked something inspired by: http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.… that appears to work for me.

1 0

dispatcher module not fetching attrs col from DB
by Jayesh Nambiar 05 Jul '16

05 Jul '16

Hi, I'm trying to use the dispatcher module with algo 9 for weighted distribution. I have a column attrs in the dispatcher table with value in the format "weight=90" for my destination addresses. But somehow the ds_list command always shows attrs as blank. I checked that kamailio doesn't fetch the attrs column at all; neither while starting up nor while calling ds_reload. This is the query that kamailio sends to my DB while starting up or calling the fifo ds_reload: select `setid`,`destination`,`flags`,`priority` from `dispatcher` I dont even see an attr column modparam in the documentation although kamailio didnt throw an error when I defined attr_col modparam for dispatcher module. Is there a different way to tell kamailio to fetch the attrs from the DB?? I'm using the 4.4.1 version. Thanks for any suggestion or pointers to solve this. - Jayesh

2 4

Kamailio v4.2.8 Released
by Daniel-Constantin Mierla 05 Jul '16

05 Jul '16

Hello, Kamailio SIP Server v4.2.8 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.7. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.8 (or to 4.3.x/4.4.x series). Important: this version marks the end of planned releases from branch 4.2. From now on the focus is on maintaining the branches 4.4 and 4.3 for stable releases. For more details about version 4.2.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/07/kamailio-v4-2-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.4, at this moment with its latest release v4.4.2. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

1 0

[SR-USERS] Kamailio issues with Asterisk Voicemail
by TEG AMJG 05 Jul '16

05 Jul '16

Dear list: I am quite new to Kamailio and i am really struggling to get some answer to some questions i have been trying to get answers by reading documentation and other questions or solutions around internet. Even though i am integrating Kamailio with two Asterisk boxes with the module dispatcher, i didnt follow the quite popular guide of Asterisk Kamailio Realtime Integration ( http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). The reason why i didnt is that it actually behaved weird since i wanted Kamailio to be a stateful proxy and also wanted Kamailio to act as a Registrar Server and Location Server using subscriber and location tables of kamailio database, keeping both Asterisk strictly as Media Servers. Also i have been able to solve some NAT Traversal issues using RTPProxy and not using those Asterisk boxes with their nat or qualify parameters to solve NAT issues. Most of my solution started following this guide ( http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html) which let me handle registration and location with Kamailio and also changed it putting kamailio behind NAT. Now to the real question, is there anyway to include Asterisk voicemail functionality with the solution I already mentioned? the question is because, as I far as I know, Asterisk needs to associate mailboxes to its users. So i believe that i have to practically change most of my solution and find a way to make Kamailio register and locate users with the Realtime integration of Asterisk and map the registration accordingly to the table fields. But it destroys most of my NAT and register solution since i also configured one of the private IPs as a peer in Asterisk sort of like in that guide [Kamailio] type=friend host=192.168.1.244 port=5060 disallow=all allow=gsm allow=g729 allow=alaw allow=ulaw Sorry for making the question so long and to be quite ignorant with some of the concepts involving Kamailio with Asterisk Thanks Alejandro

2 1

Increase diversion counter
by Igor Potjevlesch 05 Jul '16

05 Jul '16

Hello, I'm looking for the best way to increase the Diversion counter. If my Kamailio is the first to generate a call forward I can manage this with static value into suffix params. But, if the Diversion counter is already set, I don't see how to handle this properly. Someone has already work on this? Regards, Igor.

3 5

msgpack/format - new module?!
by Anonim Stefan 05 Jul '16

05 Jul '16

Hi, I'm thinking of creating a new module called "msgpack" or "format" which will export config msgpack_serialize() and msgpack_deserialize() for a given input. Is a new module the way to go, or do you think of a better place for this? Thanks, Stefan

2 3

Releasing Kamailio v4.2.8
by Daniel-Constantin Mierla 05 Jul '16

05 Jul '16

Hello, I am planning to release tomorrow the v4.2.8 out of branch 4.2 to mark the end of official packaging for 4.2.x series. From now on, the focus will be 4.4.x and 4.3.x as stable branches. I just backported what I found relevant from branch 4.3 to 4.2 -- if anyone finds some other commits that need to be backported, reply to mailing lists. 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

register_dlgcb usage example
by Alexandr Dombrovsky 05 Jul '16

05 Jul '16

Hi! Where I can find any example of usage for this function: *register_dlgcb ?*

2 1

Modify VIA header in replied message
by Huynh Quoc Bao 05 Jul '16

05 Jul '16

Hi all, As title, i want to add some additional parameters to the topmost via header in replied message. example. REGISTER ==> Kamailio ===> reply 200 OK. So in 200 OK message, i will add some additional parameters. And the same for the other message: INVITE, SUBSCRIBE, NOTIFY, MESSAGE message. -- Brs/Bao

2 3

INFO: <core> [main.c:809]: sig_usr(): signal 15 received
by Aqs Younas 04 Jul '16

04 Jul '16

Greetings Again List. After being unable to use ndb_cassandra with kamailio I tried to use app_python. Below is relevant configuration. version: kamailio 4.4.2 (x86_64/linux) 892ad6 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 892ad6 compiled on 07:50:03 Jul 1 2016 with gcc 4.9.2 modparam("app_python", "script_name", "/usr/local/kamailio/etc/kamailio/handler.py") request_route { ..... python_exec("handler"); ..... } And i am using example provided in app_python module. *But i soon as i import Cassandra driver in script*. Kamailio terminate with there logs. Jul 1 19:57:19 debian ./kamailio[8643]: INFO: app_python [mod_Logger.c:170]: logger_LM_INFO(): test.child_init(20) Jul 1 19:57:19 debian ./kamailio[8642]: INFO: app_python [mod_Logger.c:170]: logger_LM_INFO(): test.child_init(19) Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank -4: tmx Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: tmx [tmx_mod.c:275]: child_init(): rank is (-4) Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank -4: sl Jul 1 19:57:19 debian ./kamailio[8616]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD Jul 1 19:57:19 debian ./kamailio[8642]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa78988 and 0xabc988 Jul 1 19:57:19 debian ./kamailio[8642]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa787c0, 40, 1, (nil)), fd_no=0 Jul 1 19:57:19 debian ./kamailio[8644]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 21: kex Jul 1 19:57:19 debian ./kamailio[8645]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (22) Jul 1 19:57:19 debian ./kamailio[8640]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (17) Jul 1 19:57:19 debian ./kamailio[8639]: DEBUG: mi_rpc [mi_rpc_mod.c:104]: child_init(): initializing child[-2] for rpc handling Jul 1 19:57:19 debian ./kamailio[8646]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 23: tm Jul 1 19:57:19 debian ./kamailio[8641]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 18: corex Jul 1 19:57:19 debian ./kamailio[8635]: INFO: app_python [mod_Logger.c:170]: logger_LM_INFO(): test.child_init(-1) Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank -4: usrloc Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [main.c:756]: handle_sigs(): terminating due to SIGCHLD Jul 1 19:57:19 debian ./kamailio[8636]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8621]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8620]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8625]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8623]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8638]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8624]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8632]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8647]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8633]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8637]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8619]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8622]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8626]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8627]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8628]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8629]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8642]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8630]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8635]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8631]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8634]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:86]: tm_shutdown(): DEBUG: tm_shutdown : start Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:89]: tm_shutdown(): DEBUG: tm_shutdown : emptying hash table Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:91]: tm_shutdown(): DEBUG: tm_shutdown : removing semaphores Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:93]: tm_shutdown(): DEBUG: tm_shutdown : destroying tmcb lists Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:96]: tm_shutdown(): DEBUG: tm_shutdown : done Jul 1 19:57:19 debian ./kamailio[8616]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [mem/shm.c:174]: shm_core_lock_destroy(): destroying the shared memory lock Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [mem/shm.c:266]: shm_destroy_manager(): destroying memory manager: f_malloc Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [mem/pkg.c:89]: pkg_destroy_manager(): destroying memory manager: f_malloc I have tried using two different drivers for Cassandra in python but same results. I wonder if i import mysql driver kamailio starts perfectly. Below it my script. #!/usr/bin/python import sys import pymysql *import pycassa* from Router.Logger import LM_ERR from Router.Logger import LM_INFO class test: def __init__(self): LM_INFO('test.__init__\n') def child_init(self, y): LM_INFO('test.child_init(%d)\n' % y) return 0 def handler(self, msg): LM_INFO('test.handler(%s)\n' % (msg.Type)) if msg.Type == 'SIP_REQUEST': if msg.Method == 'INVITE': LM_INFO("Re-writing RURI") msg.rewrite_ruri('sip:0022@192.168.2.24:5073') LM_INFO('SIP request, method = %s, RURI = %s, From = %s\n' % (msg.Method, msg.RURI, msg.getHeader('from'))) LM_INFO('received from %s:%d\n' % msg.src_address) return 1 def mod_init(): return test() Any help is much appreciated. Best Regards, Aqs Younas

1 0

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