sr-users May 2023

sr-users@lists.kamailio.org

40 participants
47 discussions

by Björn Klasen

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

1 month

Increase MAX_BRANCHES_LIMIT

by Lars Jansson

Hi, I am building a configuration script where for some traffic flows parallel forking will be needed, and in this case I need to go beyond the default max limit of the max amount of branches. Apart from this specific case the traffic load that kamailio will need to handle is very low, let's say just one second here and there with up to 5 concurrent calls at most. This limit is as far as I understand set here: usr/local/src/kamailio-5.6/kamailio/src/core/config.h And the default limit is: #define MAX_BRANCHES_LIMIT 32 /*!< limit of maximum number of branches per transaction */ Here are the questions I have related to this: 1) If I increase the value of this constant in config.h, how high is it reasonable to set this value and still have a stable system? 2) If I increase MAX_BRANCHES_LIMIT beyond 32, are there also other parameters that needs to be changed for the system to be able to cope, and if so whichparameters? Regards, Lars

7 months, 1 week

custom mid registrar using kamailio.

by Yuriy Nasida

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

10 months, 4 weeks

Kamailio and MS Teams side by side present staus und chat function

by Hans-Jürgen Brand

11 months

Is Min-SE check broken?

by harneet singh

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

11 months

http_client using JWT?

by Benoît Panizzon

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 ______________________________________________________

11 months

Knowing about the uac module with primary and secondary proxy setting

by MUSLIH K

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)?

11 months

DISABLE SDP MEDIA STREAMS

by Patrick Karton

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 :)

11 months

Re: [sr-dev] Troubles with outbound kamailio call to other server

by Henning Westerholt

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

11 months

Consistent huntable responses based on failure_route question

by John Hardiman

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.

11 months

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users May 2023