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37515 discussions

Missing transformations for {val.*}?
by Kaufman 03 Jun '23

03 Jun '23

Beginning in 5.6 new {val.*} transformations are documented as being exported by the pv module, however if I try using it Kamailio fails to start. Wondering if maybe it's actually imported by another module and the documentation is wrong or something like that? This configuration in 5.6.3 on Alpine Linux installed from packages: #!KAMAILIO loadmodule "pv" loadmodule "xlog" request_route { xlog("Null val: $(avp(nothing){val.ne})\n"); exit; } Fails to start with the following: 0(114) INFO: <core> [main.c:3055]: main(): processes (at least): 29 - shm size: 67108864 - pkg size: 8388608 0(114) ERROR: <core> [core/pvapi.c:1774]: tr_lookup(): unknown transformation: [val] in [{val.ne}] 0(114) ERROR: <core> [core/pvapi.c:1799]: tr_lookup(): error parsing [{val.ne}] 0(114) ERROR: <core> [core/pvapi.c:1106]: pv_parse_spec2(): bad tr in pvar name "avp" 0(114) ERROR: <core> [core/pvapi.c:1131]: pv_parse_spec2(): invalid parsing in [$(avp(nothing){val.ne})] at (4) 0(114) ERROR: xlog [xlog.c:551]: xdbg_fixup_helper(): wrong format[Null val: $(avp(nothing){val.ne})] 0(114) ERROR: <core> [core/route.c:1169]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:7 Kaufman Sr. VoIP Engineer - BCM One +1.612.735.9309 bkaufman(a)bcmone.com<mailto:bkaufman@bcmone.com> 24 hour client support: +1.855.639.6300 [https://www.bcmone.com/files/bcmone-logo.png]<https://www.bcmone.com/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://www.facebook.com/bcmonenyc/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://twitter.com/bcm_one/> [https://www.bcmone.com/files/linkedin-icon-circle.png] <https://www.linkedin.com/company/business-communications-management/> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else.

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custom mid registrar using kamailio.
by Yuriy Nasida 01 Jun '23

01 Jun '23

Hello, https://lists.kamailio.org/pipermail/sr-users/2020-October/110792.html I read this thread very carefully and I think I managed to implement all the steps described by Daniel. Thank you so much for the detailed description. But it looks like I'm stuck on the last step. I can't delete records from htable when an incoming call cannot reach the registered device (e.g., no more internet access, power off at the device). I try to do this from failure route but looks like i kamailio has not access to htable from it. failure_route[INTERNAL_TO_EXTERNAL_FAILURE] { if (t_check_status("408")) { xlog("L_WARN", "$ci|log| hello \n"); xlog("L_WARN", "$ci|log| mid reg token - $sht(mid_reg_token=>$fU::$si::$sp) \n"); } } I do not see xlog message with mid reg token but i see 'hello'. Yes, htable record exists in case i check using: kamcmd htable.dump mid_reg_token Also I need a way to remove registration at the registrar (Freeswitch in my case). This means I need to send api request from kamailio or just run some script that will do this. What is the best way ? module exec ? Please advice, Thanks

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Kamailio and MS Teams side by side present staus und chat function
by Hans-Jürgen Brand 31 May '23

31 May '23

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Is Min-SE check broken?
by harneet singh 31 May '23

31 May '23

Hi All, I am facing an issue in understanding how the min_se should be working in kamailio. As per the SST documentation, it seems like if the min_se is configured as 500, then any value of Session-Expires OR MIN-SE if lower than 500, can be responded to by a 422. However, I strangely see the reverse happening. To investigate further, I looked in to the ki_sst_check_min() code in the master, and these seems like a potential issue. Ref Code: Inside ki_sst_check_min(), there is an if condition like below: if (sst_min_se < MIN(minse, se.interval)) { However, shouldn't it be the other way around? ie if (sst_min_se > MIN(minse, se.interval)) { because we need to send 422 if the received value(in INVITE etc) is smaller than the sst configure min_se value? I also found a different documentation, at https://git.sgu.ru/oldssu/ex-opensips/blob/cb9df8d59dbb254a9d862569fd5d11f6… where the check is as below? if (sst_min_se > MIN(minse, se.interval)) { Can someone confirm if this is broken, or my understanding is incorrect? Regards, Harneet -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

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http_client using JWT?
by Benoît Panizzon 31 May '23

31 May '23

Hi The http_client module support the somehow old fashioned way to submit credentials as part of the URL. Our 'held' server developer would love to use either JWT or a Bearer Token in the Auth-Header or a 'cookie-jar' to authenticate the requests. Is this somehow possible? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Knowing about the uac module with primary and secondary proxy setting
by MUSLIH K 31 May '23

31 May '23

Hi, *sip_provider_proxy1[* uac module-register] ---> *kamailio* [dispatcher Module] ----> *Asterisk1* ----> *Asterisk 2* in my configuration kamailio register to the sip provider using *uac module*, sip information are stored in database (uacreg). when a incoming call(INVITE) come to kamailio it will dispatch call to Asterisk using* "dispatcher module".* Here i have couple of questions 1. If my SIP provider has a secondary proxy as a failover server (sip_provider_proxy2), can I configure Kamailio with the uac module to use the secondary proxy if the primary proxy fails? 2. Since my application is inbound-only, registration failure is only identified when the registration expires (3600 seconds = 1 hour). Can I use the dispatcher module to select the SIP proxy and switch the registration to the secondary proxy if the primary proxy is not reachable (using OPTIONS)?

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DISABLE SDP MEDIA STREAMS
by Patrick Karton 29 May '23

29 May '23

Hello kamailio community, i know its an unusual query but i would like to know if there is a cleaner/proper.right way to disable some (or all) media streams in SDP. i have come with a solution by sdpops module with subst_body function like this. route[SDP_DISABLE_UNAUTHORIZED_MEDIA]{ # disable text,media and video media streams subst_body('#^m=(video|text|audio)[ ]+[0-9/]+(.*)$#m=\1 0 \2#ig'); # update originator ip address subst_body('#^o=([^ ]+)[ ]+([^ ]+)[ ]+([^ ]+)[ ]+([^ ]+)[ ]+([^ ]+)[ ]+([^ ]+)$#o=\1 \2 \3 \4 \5 7.7.7.7#ig'); # 1 2 2 4 5 6 # name sid snumber network net-type addr # update connection ip address subst_body('#^c=([^ ]+)[ ]+([^ ]+)[ ]+([^ ]+)$#c=\1 \2 7.7.7.7#ig'); # 1 2 3 # network net-type addr #remove all media attributes sdp_remove_line_by_prefix("a=fmtp"); sdp_remove_line_by_prefix("a=X-"); sdp_remove_line_by_prefix("a=rtcp"); sdp_remove_line_by_prefix("a=ssrc"); sdp_remove_line_by_prefix("a=sendrecv"); sdp_remove_line_by_prefix("a=sendonly"); sdp_remove_line_by_prefix("a=recvonly"); sdp_remove_line_by_prefix("a=inactive"); sdp_remove_line_by_prefix("b="); } is it the right way to do it. if not can someone provide me an alternative solution. Thanks :)

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Re: [sr-dev] Troubles with outbound kamailio call to other server
by Henning Westerholt 26 May '23

26 May '23

Hello (added sr-users, as its probably more a usage question). The SIP message (the BYE) cannot be parsed apparently. From the log it looks that there is an invalid character in the first line. Do a network trace to analyse the packets that its actually send and try to get some more clues from that. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com > -----Original Message----- > From: aaron.soto(a)dialapplet.com <aaron.soto(a)dialapplet.com> > Sent: Freitag, 26. Mai 2023 10:44 > To: sr-dev(a)lists.kamailio.org > Subject: [sr-dev] Troubles with outbound kamailio call to other server > > Hi everyone, > I'm having the following errors in kamailio when i try to make an outbound call to > other server, I don't know what does it mean, if some one can help me > > 11(25) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP > message failed (XX.XX.XX.XX.5060/1) > 13(27) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): > parse_first_line: bad message (offset: 65) > 13(27) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: > parse_msg: message=<BY > sip:asterisk@10.0.5.158:4080;alias=10.0.5.158~44730~1 SIP/2.0 > Via: SIP/2.0/UDP 84.127.226.15:5060;branch=z9hG4bK3bc90b47;rport > Route: <sip:34.252.151.179:4060;r2=on;lr;ftag=5c7f862f-9692-4028-8af0- > ebc224ba0e2a;vsf=AAAAAAAAAAAAAAAAAAAJCQABGgUAAAMJLjE4ODo1MDYw; > nat=yes>,<sip:10.0.5.40:4070;r2=on;lr;ftag=5c7f862f-9692-4028-8af0- > ebc224ba0e2a;vsf=AAAAAAAAAAAAAAAAAAAJCQABGgUAAAMJLjE4ODo1MDYw; > nat=yes> > Max-Forwards: 70 > From: <sip:692598435@kamailio.voip.svc.cluster.local>;tag=as6f32aa26 > To: <sip:963441038@192.168.123.66:4060>;tag=5c7f862f-9692-4028-8af0- > ebc224ba0e2a > Call-ID: b9a49c52-7829-4cdf-a88c-0064cc5bb438 > CSeq: 102 BYE > User-Agent: Asterisk PBX 11.2-cert3 > X-Asterisk-HangupCause: No user responding > X-Asterisk-HangupCauseCode: 18 > Content-Length: 0 > > > > _______________________________________________ > Kamailio (SER) - Development Mailing List To unsubscribe send an email to sr-dev- > leave(a)lists.kamailio.org

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Consistent huntable responses based on failure_route question
by John Hardiman 25 May '23

25 May '23

Hi Kamailio list, I hope you are all well? I am looking for some assistance on a specific error I am trying to work around. Kamailio 5.5.3 - Ubuntu 20.04 Kemi Python script (but I think this is more default failure_route behaviour) Dispatcher module which hunts to a group of FreeSWITCH that I have configured to send 503 responses on. My scenario is fairly simple, I am trying to respond back to a carrier for huntable responses (503 specifically) by calling a failure route below: def ksr_failure_manage(self, msg): if KSR.tm.t_any_timeout()>0 : self.ksr_send_reply_clear_rtpengine(msg, 503, "Timeout to destination") KSR.err("Timeout to Invite for %s from IP %s" % (KSR.pv.get("$ci"),KSR.pv.get("$si"))) return 1 else: self.ksr_send_reply_clear_rtpengine(msg, 503, "Unknown error failover to next node") KSR.err("Failed but not timeout for %s from IP %s" % (KSR.pv.get("$ci"),KSR.pv.get("$si"))) return 1 This works and a 503 with the correct message is sent if the invite doesn't get a response after 2 seconds. (My specific test case here) # Timers for call setup 2 minutes for any 1xx provisional response received # If no invite response received in 2 seconds end the call as per failure route. KSR.tm.t_set_fr(120000, 2000) self.ksr_route_relay(msg) KSR.tm.t_on_failure("ksr_failure_manage") return -255 The issue is after a few more milliseconds after my initial 503 a 408 is sent. Also after the failover if the carrier tries the same Kamailio again it gets a 500 rather than a 408. 408: [cid:image001.png@01D98F15.CD07DCB0] 500: [cid:image002.png@01D98F15.CD07DCB0] My steps to get around this have been to change the default tm code to 503 below, but this has not changed the default response and the above behaviour unfortunately: # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # setting default reason code and response for general errors to allow failover. modparam("tm", "default_code", 503) modparam("tm", "default_reason", "Unknown reason try next node") My question is: How can I make sure my failure route is the final response here. Or if I cannot make sure of this, how can I change the default mapped response to be a consistent SIP response (503 specifically) Many thanks in advance as I am scratching my head on this one a bit. John.

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Override listen advertise address
by Calvin E. 25 May '23

25 May '23

Using 5.6.x, a scenario designed for a public IP now needs to support life in the cloud behind NAT. I'd like to avoid having two listeners to separate on-net from off-net traffic. In a HA scenario, it would be nice to have only one floating IP. Is it possible to use the same local address and port for external/public and internal/private traffic? Should set_advertised_address() override the listen advertise address? Initial testing did not produce the expected result. I want to make sure I'm following a sane pattern before going down a rabbit hole. The idea is to toggle the advertised address by checking if the sender or recipient is rfc1918(). For the dispatcher module, is it possible to use socket names like below? In testing it appeared that only one of the two names was valid. listen=udp:10.0.0.10:5060 name "onnet" listen=udp:10.0.0.10:5060 advertise 11.11.11.11:5060 name "offnet"

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