sr-users August 2019

sr-users@lists.kamailio.org

65 participants
97 discussions

[tm-local-request] usage
by Duarte Rocha 28 Aug '19

28 Aug '19

Greetings, Does "event_route[tm:local-request]" works like onsend route ? I'm having some problems and i'm using local-request to debug and i found this on the documentation : "Executed after the tm module has sent a local generated, transaction stateful request." I'm logging all the requests that go through this route. Is this really the last place before kamailio send the request? If the request is logged here, is there anything more in kamailio that can stop the request (pipelimit, etc) ? If a packet gets logged on this route does that mean that it was already sent? Best Regards, Duarte Rocha

3 4

Kamailio 4.4.7 - LCR modules
by Laura 28 Aug '19

28 Aug '19

Hi guys, on our enviromet we have 130+ gateway configured over the LCR_gw tables. when we try to reload it.. we get error.. "Aug 28 09:40:36 vm-nextaitz-02 /usr/sbin/kamailio[3800]: ERROR: lcr [lcr_mod.c:1511]: reload_tables(): too many gateways" We reached some top values.. how can we modify it ? Regards

2 2

Dispatcher - FQDN with multiple IPs
by Sergiu Pojoga 27 Aug '19

27 Aug '19

Hi ppl, The problem I'm facing is that function *ds_is_from_list* seems to return false when a gateway's address is a FQDN that resolves into multiple IP addresses (I'm talking about A records), consequently in-dialog requests like BYE and re-Invites get blocked since they don't pass the *ds_is_from_list* validation. Am I missing some parameter or something? Thanks, --Sergiu

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RFC: about sl and tm local-response event routes
by Daniel-Constantin Mierla 27 Aug '19

27 Aug '19

Hello, just discovered what I consider to be an inconsistency in naming and behaviour for event_route blocks for local-request and local-response and starting a discussion here to see how to move on. The event_route[tm:local-request] is executed before sending there local generated request out (allowing also to change its content, drop, etc...). This event route is quite popular event route used when the local generated requests need to be checked or updates. The event_route[tm:local-response] is executed after the response is sent out, obviously no possibility to change anymore the content. I haven't checked the code for event_route[sl:local-response], but based on commit message should be the same. I haven't used the local-response so far at all, today after a discussion on sr-users I wanted to enable kemi callback for tm:local-response and I noticed that behaviour in the code, even I expected to be like local-request (before sending out). I am not sure how much used are the event routes for tm:local-response and sl:local-response, I haven't seen any questions about them so far on mailing lists, that's why I am asking here if would make sense to rename them like tm:local-response-sent and sl:local-response-sent to properly reflect when they are executed. I am expecting that they are very few used so far, so no big head ache with upgrades and bringing some consistency around (this change to be part of next major release). It can stay like now with proper documentation, however in the future if one want and event route for local responses before being sent out, there will be more confusion, imo ... Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Too many hops 483 - unable to setup call
by Jack R 27 Aug '19

27 Aug '19

Hi, I installed Kamailio SIP server: Following article: https://github.com/open5gs/nextepc/blob/master/docs/_docs/guide/04-setting-… By following step # 12 (as per above article) when we initiate the call we get - Too many hops 483 Created two user earlier: $ kamctl add test testpasswd $ kamctl add test2 testpasswd syslog show: Aug 27 09:36:50 localhost /usr/sbin/kamailio[56286]: ERROR: nathelper [nathelper.c:780]: add_contact_alias_0(): failed to get contact uri Aug 27 09:36:50 localhost /usr/sbin/kamailio[56286]: ERROR: ims_registrar_pcscf [save.c:268]: save_pending(): No contact headers Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (139.54.102.135:5060) to sip:test1@139.54.102.135, ee24f837b04b07334e25befe81217f8f(a)45.45.0.16) Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: nathelper [nathelper.c:780]: add_contact_alias_0(): failed to get contact uri Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: ims_registrar_pcscf [save.c:268]: save_pending(): No contact headers Aug 27 09:36:50 localhost /usr/sbin/kamailio[56276]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (139.54.102.135:5060) to sip:test1@139.54.102.135, ee24f837b04b07334e25befe81217f8f(a)45.45.0.16) Aug 27 09:37:05 localhost /usr/sbin/kamailio[56278]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (45.45.0.16:42965) to sip:test1@139.54.102.135, 04e06b6352f7a2b8f97c1a3733b3334a(a)45.45.0.16) Aug 27 09:37:05 localhost /usr/sbin/kamailio[56278]: ERROR: ims_registrar_pcscf [save.c:339]: save_pending(): Will save pending contact without security parameters Aug 27 09:37:05 localhost /usr/sbin/kamailio[56374]: ERROR: <script>: Diameter: AAR failed on subscription to signalling Server IP 139.54.102.135 where Kamailio is installed, UE IP address 45.45.0.16, 45.45.0.1 Followed this https://www.kamailio.org/wiki/tutorials/faq/main#why_the_sip_requests_are_r… Thought to test using SIPp - installed and ran default command: ./sipp –sn uac 127.0.0.1 and ./sipp –sn uac 139.54.102.135 Aug 27 08:51:32 localhost /usr/sbin/kamailio[56295]: ERROR: <script>: INVITE (sip:sipp@139.54.102.135:5061 (139.54.102.135:5061) to sip:service@139.54.102.135:5060, 2-60232(a)139.54.102.135) Aug 27 08:51:32 localhost /usr/sbin/kamailio[56295]: ERROR: ims_ipsec_pcscf [cmd.c:507]: ipsec_forward(): Contact doesn't exist The user sipp and service does'nt exist Added user: kamctl add sipp testpasswd kamctl add service testpasswd Still same error: Aug 27 10:52:17 localhost /usr/sbin/kamailio[56289]: ERROR: ims_ipsec_pcscf [cmd.c:507]: ipsec_forward(): Contact doesn't exist Aug 27 10:52:17 localhost /usr/sbin/kamailio[56284]: ERROR: <script>: INVITE (sip:sipp@139.54.102.135:5061 (139.54.102.135:5061) to sip:service@139.54.102.135:5060, 977-61357(a)139.54.102.135) Please advise. Thanks and Regards.

1 1

kamailio stop registering agents
by Gaurav Bmotra 27 Aug '19

27 Aug '19

hi i m using kamailio 5.1,,, on ubuntu 18.4 installed form source .... after few days it stop registering client to kamailio.... it resolved after restart service ... same after few day loges ---- --------------------- kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Wed 2019-08-21 08:07:44 UTC; 5 days ago Docs: man:systemd-sysv-generator(8) Process: 1936 ExecStart=/etc/init.d/kamailio start (code=exited, status=0/SUCCESS) Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [main.c:2636]: main(): processes (at least): 27 - shm size: 67108864 - pkg size: 838 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: ERROR: dispatcher [dispatcher.c:813]: ds_warn_fixup(): failover functions used, but required AVP Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: ERROR: <core> [core/tcp_main.c:2850]: tcp_init(): bind(c, 0x7fbabac2c1f4, 16) on 172.31.35.49:509 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 21 08:07:44 talk kamailio[1936]: * already running Aug 21 08:07:44 talk kamailio[1936]: ...done. Aug 21 08:07:44 talk systemd[1]: Started LSB: Start the Kamailio SIP proxy server. plz help thank you -- *Regards:* Gaurav Kumar

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Re: [SR-Users] [sr-dev] Failed to catch some reply message with reply_route block
by Henning Westerholt 27 Aug '19

27 Aug '19

(moved the discussion to user list) Hello Salah, I see the this output in the log message: 6(26) INFO: <script>: on_reply_route [SIP/2.0 200 OK Via: SIP/2.0/UDP 172.32.0.6;rport=5060;received=172.32.0.6;branch=z9hG4bKe68a.99dc492fe74dcfe1017828521a5fa362.0 2(22) INFO: <script>: on_reply_route [SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.32.0.6;rport=5060;received=172.32.0.6;branch=z9hG4bK178a.79cddb135aabfc42736fd9e9fd826ba8.0 and so on.. So you are seeing the replies correctly, or I wrong? Cheers, Henning Am 22.08.19 um 21:29 schrieb Salah Ahmed: Hello Henning, We already added xlog in the onreply_route and found all messages in between kamailio and UAS. The attached log files in first email has those logs. If any other specific log needed I am here to provide them. Thanks, Salah On Thu, Aug 22, 2019 at 2:24 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Salah, ok, see if you are able to output some easy log with "xlog" in this route, to make sure there is no cfg problem somewhere. Cheers, Henning Am 22.08.19 um 21:15 schrieb Salah Ahmed: Hello Henning, Thanks for quick reply, We armed the t_on_reply just before t_relay() in the route block. t_on_reply("LOGRPL"); if (!t_relay()) { sl_reply_error(); } Thanks, Salah On Thu, Aug 22, 2019 at 2:03 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Salah, the replies are going from the UAS to the UAC over Kamailo as a proxy, with the exception of the hop-by-hop 100. So you should see the 200 OK in the Kamailio in reply_route and onreply_route. Maybe you can check if you armed the onreply_route with t_on_reply for the INVITE routing. Cheers, Henning Am 22.08.19 um 19:50 schrieb Salah Ahmed: Hello, We facing an issue on capturing sip message in Kamailio(Version: 5.2.3). The scenario is very simple. UAC Kamailio UAS |--------INVITE--------->| | | |-------INVITE------->| | |<------100 Trying----| |<-------100 Trying------| | | |<------200 Ok--------| |<-------200 Ok----------| | In this simple scenario, we can't catch any reply messages in between the Kamailio and UAC. We have tried reply_route, onreply_route, and onsend_route. But no one work to grep that reply on that side. onsend_route was bad try as its only for forwarded reply message. Is there any other magic to capture those replies. A debug=3 log message attached here. Thanks, Salah _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

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exit stop procesing or not after a conditional? based on "exit" core docs seems does not explain behaviour
by PICCORO McKAY Lenz 26 Aug '19

26 Aug '19

in my kamailio-asterisk integration i have a exit after the authentication logic, BUT; there's another conditional after the "exit" sentence, so my ask, if there's a exit in that piece of code .. that next conditional will not be executed? Docs seems said "stop execution of script" so all the rest of the logic in file will not be routed? The documentation seems does not explain in good behavior what happened or the kamailio-asterisk wiki page does not have good explanations for that! it's that an error? https://www.kamailio.org/wiki/cookbooks/4.3.x/core#exit - - ``` if (from_uri==myself) {#!ifdef WITH_ASTERISK if (!proxy_authorize("$fd", "sipusers")) {#!else if (!proxy_authorize("$fd", "subscriber")) {#!endif proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } ``` 8:52 am Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

reply cannot be forwarded - no 2nd via
by Ryan McEvoy 25 Aug '19

25 Aug '19

Hello, I have been having an issue using kamailio to load balance to a SIP/MRCP server. I believe that the issue actually lies with the reply being sent from the final recipient, but I wanted to be sure that I understood the problem and to see if there were any possible solutions within Kamailio. I am using the dispatcher module to dispatch SIP INVITES to media resources, in this case I am just using a single host in a single dispatcher group. The configuration that I have has worked just fine for other SIP/MRCP servers, which is why I think the issue is with this target in particular. The INVITE is routed correctly and contains the headers that it should. The invite comes originally from a Freeswitch, arrives at Kamailio, who should then dispatch to a host in the group. x.x.x.x:4901 is Kamailio x.x.x.x:8090 is Freeswitch y.y.y.y:8000 is the final target This is the invite, as received by the final target after dispatched via Kamailio INVITE sip: x.x.x.x:4901 SIP/2.0 Record-Route: <sip: x.x.x.x:4901;transport=tcp;lr> Via: SIP/2.0/TCP x.x.x.x:4901;branch=z9hG4bK3a9a.7f9a4b588cd75108cfea9a2bd9ffa829.0;i=3 Via: SIP/2.0/TCP x.x.x.x:8090;branch=z9hG4bKStXp06Sv7jU7S Max-Forwards: 69 From: <sip: x.x.x.x:8090>;tag=4gHgNcNUDjvFQ To: <sip: x.x.x.x:4901> Call-ID: 6f2a0e9c-3f9e-1238-2aa9-005056854777 CSeq: 8692231 INVITE User-Agent: tts Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE Supported: timer, 100rel Content-Type: application/sdp Content-Disposition: session Content-Length: 281 v=0 o=IPSoft 3367790083977298880 2789991278648035021 IN IP4 x.x.x.x s=- c=IN IP4 x.x.x.x t=0 0 m=application 9 TCP/MRCPv2 1 a=setup:active a=connection:existing a=resource:speechsynth a=cmid:1 m=audio 32038 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=recvonly a=mid:1 The reply that the server sends back appears to be missing a second Via header, which is preventing Kamailio for routing it correctly (based on the error message from core/forward.c) This is the reply sent by the target destination SIP/2.0 200 OK Via: SIP/2.0/TCP x.x.x.x:8090;branch=z9hG4bKStXp06Sv7jU7S To: <sip: x.x.x.x:4901>;tag=ndszib3k From: <sip: x.x.x.x:8090>;tag=4gHgNcNUDjvFQ Call-ID: 6f2a0e9c-3f9e-1238-2aa9-005056854777 CSeq: 8692231 INVITE Contact: "Voiceware MRCP Server" <sip:vwmrcp@ y.y.y.y:8000;transport=tcp> Content-Type: application/sdp Content-Length: 324 v=0 o=- 1566492047 1566492047 IN IP4 y.y.y.y s=Voiceware MRCP Server 2.15.0.7. Dec 28 2018 c=IN IP4 y.y.y.y t=0 0 m=application 8002 TCP/MRCPv2 1 a=channel:12951125463@speechsynth a=connection:new a=setup:passive a=cmid:1 m=audio 30158 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=sendonly a=ptime:20 a=mid:1 It is correct that there should still be a 2nd Via header in the 200 OK reply from the target host, right? If so, is there anything else that can be done in Kamailio to handle this, or is this mrcp server simply not complying properly with SIP? There’s no funky natting between these hosts or anything like that. Thanks in advance!

2 1

Wrong ACL to Provider kamailio+asterisk+uac
by Zhan Bazarov 25 Aug '19

25 Aug '19

Hello! Please, is it possible to change from/to headers in ACK message to provider? this is exactly our case, please, look it when u get a chance. http://sip-router.1086192.n5.nabble.com/Wrong-ACK-to-Provider-td130050.html…

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sr-users August 2019