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37390 discussions

Unable to parse the Cancel request
by Ilie Soltanici 05 May '21

05 May '21

Hello, We are having an issue here where Kamailio is complaining about duplicate To header, while in the SIP Packet there is just one. See below the log message: DEBUG: tm [t_msgbuilder.c:391]: build_local_reparse(): duplicate To header ERROR: tm [t_msgbuilder.c:531]: build_local_reparse(): cannot build CANCEL request ERROR: tm [t_cancel.c:297]: cancel_branch(): attempt to build a CANCEL failed ERROR: tm [t_fwd.c:1434]: e2e_cancel(): cancel error And this is the sip message: CANCEL sip:123456789@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:19373 ;branch=z9hG4bK-524287-1---7774073fad1a3264;rport Max-Forwards: 70 To: <sip:123456789@1.2.3.4:5060> From: "101"<sip:101@1.2.3.4:5060>;tag=6f373b66 Call-ID: hC2O6zx8ZaUJ1di046 CSeq: 2 CANCEL Proxy-Authorization: Digest username="101",realm="1.2.3.4",nonce="YJKbfWCSmlFTHxdJfKmySydF9L7FzfaqcgyOo4A=",uri=" sip:123456789@1.2.3.4:5060 ",response="2bf2ba8ef2374d6aa8c39d650613077d",algorithm=MD5 User-Agent: TEST Content-Length: 0 version: kamailio 5.5.0 (x86_64/linux) d4c1a1 The route itself, is like in the standard documentation: # CANCEL processing if KSR.is_CANCEL() : KSR.info(RANDOM_LOG + 'KSR.is_CANCEL') if KSR.tm.t_check_trans()>0 : self.ksr_route_relay(msg) return 1 def ksr_route_relay(self, msg): if KSR.is_method_in("IBSU") : if KSR.tm.t_is_set("branch_route")<0 : KSR.tm.t_on_branch("ksr_branch_manage") if KSR.is_method_in("ISU") : if KSR.tm.t_is_set("onreply_route")<0 : KSR.tm.t_on_reply("ksr_onreply_manage") if KSR.is_INVITE() : if KSR.tm.t_is_set("failure_route")<0 : KSR.tm.t_on_failure("ksr_failure_manage") if KSR.tm.t_relay()<0 : KSR.sl.sl_reply_error() return -255 Thanks

2 7

start_recording and stop_recording inside event_route[xhttp:request]
by Володимир Іванець 05 May '21

05 May '21

Hello! I'm testing call recording with Rtpengine. It works fine when the "record-call=on" flag is added to the *rtpengine_offer* or *start_recording* is used in the *request_route*. But I was wondering if the call recording can be managed by a separate application. So I add the following lines to the *event_route[xhttp:request]* and triggered it with an HTTP request after the call was established. The call-id value was taken from the Rtpengine log and sent with the request. * if ($hu =~ "^/CALL_RECORD_START/") {* * $var(call_id) = *"call-id=" + *$(hu{s.select,2,/});* xlog("L_DBG", "$var(call_id)"); * start_recording($var(call_id));* exit; * }* In this case, Kamailio will log errors and will not send the Rtpengine command to start call recording. *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: DEBUG: <script>: call-id=3de74068278ede4417086eb82bdf5a06@127.0.0.1:5080.0* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: ERROR: rtpengine [rtpengine_funcs.c:294]: get_callid(): call-id not found* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: ERROR: rtpengine [rtpengine.c:2383]: rtpp_function_call(): can't get Call-Id field* Have anyone tried to call functions *start_recording* and *stop_recording* inside *event_route[xhttp:request]*? Thanks a lot!

3 12

Kamailio v5.5.0 Released - new major version is out
by Daniel-Constantin Mierla 05 May '21

05 May '21

Kamailio v5.5.0 is out – it comes with 6 new modules and a considerable set of improvements touching again more than 80 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-5-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.5.0 brings more flexibility and optimizations across many components, enhancements to distributed message queue (dmq module), load balancer, http client, STIR/SHAKEN support, IMS/VoLTE, rtpengine and tls as well as to make available more functions to KEMI interface ... just to list only a few here. Enjoy Kamailio v5.5.0! Thank you for flying Kamailio! Stay safe and healthy! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 0

dialog OPTIONS
by David Villasmil 05 May '21

05 May '21

Hello guys, I'm using the dialog module, params are: # ----- dialog params ----- modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", 20) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) modparam("dialog", "ka_failed_limit", 2) modparam("dialog", "send_bye", 1) modparam("dialog", "timer_procs", 1) When receiving an INVITE, i'm doing dlg_set_property("ka-src"); dlg_set_property("ka-dst"); dlg_set_property("timeout-noreset"); I have a couple questions: The flow is the following: clientA>-\ kamailio<->freeswitch clientB<-/ Kamailio receives an INVITE from ClientA, which I forward to freeswitch, which in turn forwards it back to Kamailio. In reality I would like to monitor only the client legs, but I'm having problems, so I'm enabling OPTIONS on all legs. On the ClientA legs, I see OPTIONS going from kamailio to both freeswitch and the client. But on the ClientB legs i don't see any OPTIONS at all. Can someone point me in the right direction to enable it on all legs? Something else i'm seeing is even though i set "ka_failed_limit" to 2, kamailio is sending several OPTIONS before sending BYEs to both legs (clientA, since there are no OPTIONS on clientB) Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

The plan for releasing Kamailio v5.5.0
by Daniel-Constantin Mierla 05 May '21

05 May '21

Hello, after almost 4 weeks since development was frozen, it is time to plan releasing the next major stable version, respectively 5.5.0, therefore I propose to do it on Wednesday, May 5, 2021. The branch 5.5 was already created about one week ago, several doc resources were published (core/variables/transformations/rpcs/stats cookbooks, alphabetic indexes, ...). If anyone is testing an upgrade from branch 5.4 to 5.5 and encounters changes that should be done in Kamailio config, add notes about at: * https://www.kamailio.org/wiki/install/upgrade/5.4.x-to-5.5.0 Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

possible bug on {param.value,name}
by David Villasmil 04 May '21

04 May '21

Hello guys, I'm possible using this wrong, but here it is; I need to get a previously set parameter to rr, named "wsconnid": $var(wsconnid) = $(hdr(Route){param.value,wsconnid}) xlog("L_ERR", "Connection ID parameter: $var(wsconnid)"); And i'm getting: May 3 19:24:45 ip-10-231-32-37 /usr/sbin/kamailio[10015]: ERROR: <script>: WS Connection ID 13> May 3 19:24:49 ip-10-231-32-37 /usr/sbin/kamailio[10014]: ERROR: <script>: Got an OPTIONS We now need to get the connection id from rr This is an example: Route: <sip:10.231.32.37;r2=on;lr;wsconnid=13> The function is returning "13>" instead of "13". Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 8

geting $conid on outbound INVITE
by David Villasmil 04 May '21

04 May '21

Hello guys, I don't have a lot of experience with webrtc (websocket). I'm trying to get $conid when i receive an INVITE via UDP which is GOING to a ws, how do i do that? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Re: [SR-Users] Kamailio 5.4.4 and Siptrace
by pwer spire 04 May '21

04 May '21

Hi, Any one have some idea about this? Or how it should be configured in order to still have the original response received in kamailio instead of the mapping one? Thanks in advance, Regards, Tiago

3 7

Can't make tcp_conid_alive work with $var
by David Villasmil 03 May '21

03 May '21

Hello guys, I'm trying to do the following: On INVITE, i add a param to RR "wsconnid" which is the connection id of the WS socket. Afterwards, i get an OPTIONS including said param. Next I want to retrieve the param and check if the connection is alive. NOTE: using simply "$(hdr(Route){param.value,wsconnid})" returns an extra ">" from the end of the string, and i haven't been able to remove that any other way but with "striptail", it works fine. *** $var(conid) = $(hdr(Route){param.value,wsconnid}); $var(conid) = $(var(conid){s.striptail,1}); xlog("L_ERR", "Connection ID parameter: [$var(conid)]"); if (tcp_conid_alive("$var(conid)")) { xlog("L_ERR", "WS Connection ID $var(conid) is alive"); } else { xlog("L_ERR", "WS Connection ID $var(conid) is NOT alive"); } *** With that, "$var(conid)" contains the connection id properly: *** ERROR: <script>: Connection ID parameter: [2] *** but tcp_conid_alive complaints: *** ERROR: <core> [core/sr_module.c:1638]: get_int_fparam(): Could not convert PV to int ERROR: tcpops [tcpops_mod.c:374]: w_tcp_conid_state(): invalid parameter 'conid' (must be a number) *** Using a hardcoded number such as "1" generates no complaints. Kamailio version: version: kamailio 5.3.7 (x86_64/linux) Not sure what's going on here, any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

named account with multiple numbers
by beer Ll 03 May '21

03 May '21

Hi I have a configuration with sip accounts which authenticate with the real name of the customers ex: customer1(a)sip.myvoip.dev customer2(a)sip.myvoip.dev ... INBOUND call of multiple numbers are connected with alias_db to the accounts ex: +331234567890 | sip.myvoip.dev | customer1 | sip.myvoip.dev | +331234567891 | sip.myvoip.dev | customer1 | sip.myvoip.dev | +341234567893 | sip.myvoip.dev | customer2 | sip.myvoip.dev | +341234567894 | sip.myvoip.dev | customer2 | sip.myvoip.dev | ... and all works perfectly. OUTBOUND call after authentication with the username kamailio change the caller id and the ip of my ISP-SIP-TRUNK to relay the call with the correct CID for the ISP using the variable $fn (display name in the From header) [SIP-GW-ISP] { #out with ISP route(CID); $ru = "sip:" + $rU + "@" + "IP_ADDRESS_MY-ISP"; route(RELAY); exit; } ROUTE[CID] { $var(caller) = $(fn{re.subst,/"//g}); uac_replace_from("$var(caller)","sip:$var(caller)@$fd"); } Is possible do it with a different approach for example using Remote-Party-ID , PPI P-Preferred-Identity ? Is possible to check if the customer uses the correct outbound number present in the alias_db table ? ex for customer1 only +331234567890 or +331234567891 numbers many thanks Luca

1 0

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