- sr-users - kamailio.org

Re: [SR-Users] P-CSCF not starting- Kamailio 5.3
by Henning Westerholt 04 Feb '20

04 Feb '20

Hello Lanvin, try to load it before and see if it starts then. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Lanvin ADON <lanvin.adon(a)gmail.com> Sent: Tuesday, February 4, 2020 11:45 AM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] P-CSCF not starting- Kamailio 5.3 Hello Henning ims_ipsec_pcscf is loaded after. loadmodule "ims_dialog" loadmodule "ims_usrloc_pcscf" loadmodule "ims_registrar_pcscf" #!ifdef WITH_IPSEC loadmodule "ims_ipsec_pcscf" #!endif Thanks Lanvin ADON On Tue, Feb 4, 2020 at 11:06 AM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello, just a guess, but do you load the ims_ipsec_pcscf module before the ims_registrar_pcscf module? Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Lanvin ADON Sent: Tuesday, February 4, 2020 8:54 AM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] P-CSCF not starting- Kamailio 5.3 Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON -- Cordialement, Lanvin ADON

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P-CSCF not starting- Kamailio 5.3
by Lanvin ADON 04 Feb '20

04 Feb '20

Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON

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parallel forking in combination with rtpengine
by michael.willow＠web.de 04 Feb '20

04 Feb '20

Hi everbody, I encountered an issue with my Kamailio in combination with Rtpengine. I wondered if anybody got a solution or a hint as this issue is only in combination with special parallel forked calls. I sent a call from my SIP gateway via a Kamailio to another SIP provider. The other SIP provider or the B-user seems to use a parallel forking as I receive 183 Session Progress and 180 Ringing with different to-tags. My kamailio is using a Rtpengine to relay the media from my SIP gateway to the other SIP provider. As Kamailio receives two different to-tags that include a SDP body rtpengine_manage makes Rtpengine to change RTP port. Some short example (cut down to the important messages): a-gateway kamailio b-provider | INVITE Port 16890 | INVITE Port 20400 | | ------------------------> | --------------------------> | | | | | 180 to-tag 1 / no SDP | 180 to-tag 1 / no SDP | | <------------------------ | <------------------------- | | | | | 183 to-tag 2 / Port 20408 | 183 to-tag 2 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | RTP | RTP | | Port 16890 <---- Port 20408 | Port 20400 <---- Port 46298 | | | | | | | | RTP | RTP | | Port 16890 ----> Port 20408 | Port 20400 ----> Port 46298 | | | | | | | | 180 to-tag 2 / Port 20408 | 180 to-tag 0 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | 200 OK to-tag 1 / Port 20408 | 200 OK to-tag 1 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | RTP | RTP | | Port 16890 <---- Port 20408 | Port 20400 <---- Port 46298 | | | | | | | | RTP | RTP | | Port 16890 ----> Port 20408 | Port 20418 ----> Port 46298 | Kamailio receives INVITE from a-gateway with rtp port 16890 and rtpengine rewrites it with port 20400 to b-provider. b-provider sends 180 ringing from to-tag 1 and no SDP which is just passed through kamailio to a-gateway. b-provider sends 183 progress from to-tag 2 and SDP with port 46298 to kamailio. rtpengines rewrites the sdp port to 20408 which is passed to a-gateway. b-provider start sending RTP via the negotiated ports which is passed correctly to a-gateway. b-provider sends 180 ringing from to-tag 2 with the same SDP. Rewritten and passed to a-gateway. b-provider sends 200 OK on Invite. This 200 Ok is from to-tag 1 but including the same SDP endpoint as to-tag 2 used before. Media from b-provider via rtpengine to a-gateway is passed the same way as before. Media from a-gateway to rtpengine is passed the same way as before. Media from rtpengine is passed with a different source port to b-provider. As this port was not announced in the callflow b-provider blocks the traffic. Any ideas how to fix this issue or force rtpengine to stay on the same port? Thanks Michael

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P-CSCF not starting- Kamailio 5.3
by Lanvin ADON 04 Feb '20

04 Feb '20

Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON

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Call merging issues on cellular network
by Gaurav Bmotra 03 Feb '20

03 Feb '20

Hi I m using kamailio 5.3 I m trying to merge the call on cellular network but is not working. As on WiFi it is working fine . Any one can help me Thank you

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Kamailio TLS server hello without any client certificate request
by Patrick Murphy 03 Feb '20

03 Feb '20

Hey guys - I'm using the following settings on my Kamailio config: modparam("tls", "low_mem_threshold2", 1024) modparam("tls", "tls_force_run", 1) modparam("tls", "tls_disable_compression", 1) modparam("tls", "ssl_max_send_fragment", 4096) modparam("tls", "renegotiation", 0) modparam("tls", "low_mem_threshold1", 2048) modparam("tls", "require_certificate", 1) When a client attempts to connect to Kamailio, the server hello it receives does *not* contain the client certificate request. And I've noticed that kamailio starts with: tls [tls_domain.c:694]: set_verification(): TLSs<default>: No client certificate required and no checks performed tls [tls_domain.c:320]: fill_missing(): TLSs<default>: require_certificate=0 Complete version output: kamailio 5.1.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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. Any hints?

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Dialog module - default_timeout behavior
by Gholamreza Sabery 03 Feb '20

03 Feb '20

Hi, According to the documentation default_timeout is "The default dialog timeout (in seconds), in the absence of a custom value provided in an AVP". There is also a "send_bye" option which can automatically send a BYE message on dialog timeout. Suppose that a user sends an INVITE and the call is successful (200). He talks more than default_timeout seconds (so there is not BYE for more than default_timeout seconds). This means that when the user actually sends BYE, the dialog is already terminated by the dialog module? Regards

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sqlops multiple databases
by Ali Taher 03 Feb '20

03 Feb '20

Hello, Is there a way in sqlops module to use multiple postgres databases based on routing logic ? Here is what I'm using right now: modparam("sqlops","sqlcon","cb=>postgres://postgres:postgres@localhost:5432/db1") Thanks Ali Taher

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Bad request when INVITE relayed to another kamailio
by Zhan Bazarov 31 Jan '20

31 Jan '20

We have Kamailio-cluster via route53(round-robin) some-domain.net we have two kamailio with public IP's phone1 is registered on kam1 phone2 is registered on kam2 when we are calling from phone1 to phone2 callflow looks: phone1 => kam1 => asterisk => kam1 => t_relay(address of second kamailio:5078) => kam2 => phone2 it works perfectly, but in case when we are using polycom as phone2 - we are getting 404 response from polycom... *Invite from second kamailio * 2020/01/20 10:31:21.799327 10.199.240.19:5078 -> 37.17.41.5:49811 INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: <sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge> Record-Route: <sip:10.199.240.191:5078;r2=on;lr;nat=yes;rtp=bridge> Record-Route: <sip:10.199.240.135:5078;lr> Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" <sip:1015@10.199.240.179>;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: <sip:jyu3xsfkrz6c5qn@10.199.240.135> Contact: <sip:asterisk@10.199.240.179:7060;alias=10.199.240.179~7060~1> Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 CSeq: 22619 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Penny" <sip:1015@10.199.240.179> Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 463 *Response from POLYCOM * 2020/01/20 10:31:22.054766 37.17.41.5:49811 -> 10.199.240.19:5078 SIP/2.0 400 Bad Request Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" <sip:1015@10.199.240.179>;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: <sip:jyu3xsfkrz6c5qn@10.199.240.135>;tag=8BC58304-83D9B045 CSeq: 22619 INVITE Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 Record-Route: <sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge>, <sip:10.199.240.19:5078;r2=on;lr;nat=yes;rtp=bridge>, <sip:10.199.240.235:5078;lr> User-Agent: PolycomVVX-VVX_450-UA/5.8.0.12848 Accept-Language: en Content-Length: 0 Any ideas how to fix it? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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$vn vs $var
by Alex Balashov 31 Jan '20

31 Jan '20

Hi, I must have missed the discussion about it somewhere, but what is the inspiration for the new $vn variable container? According to the documentation, the only difference is that it can hold $null values, whereas variables initialised in $var to $null are coerced into an integer value of 0. But why was this a concern? Why is it necessary? Is it not confusing to users? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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rtpengine relay ip and port in avp
by Pafel 31 Jan '20

31 Jan '20

Thanks for the info Yuri! I tried several scenarios but without success. My sip proxy has ipv4 and ipv6 address. The other nodes use ipv4. When there is a request for a ipv6 UAC/customer comming to the proxy the rtpengine/sipwise is rewriting the sdp with the correct ipv6 address of the proxy. So I am trying to get the ipv6 address of the proxy from the sdp (which is always the same ipv6 address) and the port (which is always different). I've enabled modparam("rtpengine", "write_sdp_pv", "$avp(sdp)") And I have in request section: rtpengine_manage($avp(rtpproxy_offer_flags)); set_body("$avp(sdp)", "application/sdp"); xlog("L_INFO", "Request SDP $avp(sdp))\n"); And the problem is that I am always getting the ipv4 address and port but not the rewritten by the proxy ipv6 address and port. Any idea? Regards, Pavel

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Re: [SR-Users] modifying via header from public IP to private IP
by vinod mn 31 Jan '20

31 Jan '20

Thank's david in my kamailio configuration I am using *listen=udp:192.168.131.69:5060 <http://192.168.131.69:5060/> advertise 221.20.20.12:5060 <http://221.20.20.12:5060/>* (as mentioned by you) and when the kamailio sends an invite to sip-gateway I am seeing this sip-gateway ip is *192.168.132.67 * and this gateway don't have any public-ip assigned .. Via: *SIP/2.0/UDP 221.20.20.12:5060 <http://221.20.20.12:5060/>;* branch=z9hG4bKaa6a.14c65f5675077ae7b0df70b02516979b.0 Via: SIP/2.0/UDP 192.168.113.112 :50386;received= 192.168.113.112 ;branch=z9hG4bK-d87543-550b984016109617-1--d87543-;rport=50386 I want private ip instead of public ip in the first via -- Thanks and regards Vinod.M.N

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kamailio
by Genildo Nascimento 30 Jan '20

30 Jan '20

Someone managed installed kamailio with IPV6

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modifying via header from public IP to private IP
by vinod mn 30 Jan '20

30 Jan '20

Hi , I have a cloud server, when I make call from a sip phone (registered with kamailio), in the INVITE header I am seeing the via header with public IP, is there any way that I can modify via header to send only the private IP. Please let me know, is it possible to replace. [image: image.png] the via headers that kamailio is sending to sip-gateway is sip-phone ip:192.168.113.112 Via: SIP/2.0/UDP *221.20.20.12:5060 <http://221.20.20.12:5060>* ;branch=z9hG4bKaa6a.14c65f5675077ae7b0df70b02516979b.0 Via: SIP/2.0/UDP 192.168.113.112 :50386;received= 192.168.113.112 ;branch=z9hG4bK-d87543-550b984016109617-1--d87543-;rport=50386 how to modify the first via header to have private-IP instead of public-IP.. I tried all the replace functions but it is not working. -- Thanks and regards Vinod.M.N

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How to make a distinction between sip and tel uri
by Duarte Rocha 29 Jan '20

29 Jan '20

Greetings, I have a client that sends me two P-Asserted-Identity headers, one with SIP URI and one with tel URI. The pseudo variable $ai returns the topmost P-Asserted-Identity header and I only need the one with the SIP URI. How should i proceed ? Is there a way to delete the one with tel URI? the header removal functions i find only work with header name. I could also work with something different than $ai for this scenario. Best Regards,

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Re-write contact header
by egemen ulus 29 Jan '20

29 Jan '20

Hi everyone, I have asterisk farm and kamailio which located front of asterisks farm as a proxy. When someone send a invite from outside to my kamailio (without record route), i forward it to my Asterisk farm. Then My one of Asterisk reply it as 200 ok. Then my kamailio forward this 200 ok to remote side with record route. But in my 200 ok message, contact header contains my Asterisk IP address. That is why remote side tries to send ACK to Asterisk. How to rewrite my contact header as my kamailio (SBC) address ? Actually I tried fixed nated contact and advertise ip but I couldn't make it worked. I am really new on that. Kamailio cfg file is attached. Please help! Egemen Get Outlook for Android<https://aka.ms/ghei36>

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Kamailio TLS
by Himala Gamage 28 Jan '20

28 Jan '20

Hello All, Kamailio newbie here, I'm trying to set up TLS on Kamailio and having some issues, Kamailio Running on Debian 10. This what I get when trying to install TLS apt-get install -y kamailio-tls-modules The following packages have unmet dependencies: kamailio-tls-modules : Depends: libcurl3 (>= 7.16.2) but it is not installable E: Unable to correct problems, you have held broken packages. Can I install the TLS package with libcurl4? If not, any other workaround? Thanks, In advance Himala

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SIP Redirect CANCEL handling
by Ali Taher 28 Jan '20

28 Jan '20

Hello, I'm using Kamailio as sip redirect server where SBC forwards the calls to Kamailio which sends the routing back to SBC in 3xx response. Now, I need to handle CANCEL requests sent from SBC to Kamailio. Here is the scenario: the client sends CANCEL request to SBC which responds with 200 OK and 487 Request Terminated and then sends CANCEL request to Kamailio. I'm not sure how Kamailio should handle the CANCEL request here, should it send only 200 OK back to SBC or also should send 487 request terminated? Regards, Ali Taher

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get first occurrence value of a param in a string
by Ali Taher 28 Jan '20

28 Jan '20

Hello, I need to get the value of first tgrp param in this string : $var(x) = <sip:xxxxxxxxx;tgrp=0001000;trunk-context=xx.xx.xx.xx@xx.xx.xx.xx:5060;transport=UDP;user=phone;tgrp=237> I'm trying $var(y) = $(var(x){param.value,tgrp}); but it's getting the last value of tgrp which is "237>". Noting that first tgrp is not always in the second index , some other parameters can be added to the string. How to get the value of first occurrence of tgrp param ? Regards, Ali Taher

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centos 7 repo is broken
by Aidar Kamalov 27 Jan '20

27 Jan '20

Hello # yum install kamailio Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirror.reconn.ru * extras: mirror.docker.ru * nux-dextop: mirror.li.nux.ro * updates: dedic.sh Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.6-7.el7.centos will be installed --> Processing Dependency: /usr/bin/python23 for package: kamailio-5.2.6-7.el7.centos.x86_64 --> Processing Dependency: /usr/bin/python23 for package: kamailio-5.2.6-7.el7.centos.x86_64 --> Finished Dependency Resolution Error: Package: kamailio-5.2.6-7.el7.centos.x86_64 (kamailio) Requires: /usr/bin/python23 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest -- Aydar A. Kamalov

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Kamailio on Docker
by Jack R 27 Jan '20

27 Jan '20

Hi Please advise have you tried deploying Kamailio (just plain SIP server) with mysql on Docker. I was trying the same, using Kamailio & mysql images on https://hub.docker.com however there are two issues that I was facing: 1. I ran Wireshark on server & inside container I am getting 403 Not relaying and SIP client MircoSIP does not connect. Request reaches the server and the container however it gives 403 Not relaying. I thought to print myuri value from Kamailio.cfg, to understand the issue better however then comes the second issue as below: 2. This is a bigger issue: I am not able to generate any logs -no syslogs, no docker logs (very limited logs get generated). Hence not able to troubleshoot anything. Please advise and help. Thanks and Regards, Jack.

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Kamailio/RTC dinner at Fosdem 2020
by Daniel-Constantin Mierla 27 Jan '20

27 Jan '20

Hello, Fosdem 2020 is approaching, there will be a talk from Henning about Kamailio, many other friends and related projects are presenting in the RTC Devroom (Giacomo Vacca and Federico Cabiddu, Wazo, Asterisk, Homer, Janus, Jitsi, CGRateS, Linphone, ...): * https://fosdem.org/2020/schedule/track/real_time_communications/ I plan to be at the event and wondering if there are enough interested participants for having the traditional Kamailio/RTC dinner. If yes, Torrey will help to book a place to accommodate us (first choice could be the same restaurant like the last year, if available). Reply if you want to join the dinner and say how many other people are joining you. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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CSeq values in dialog OPTIONS keepalives
by Alex Balashov 24 Jan '20

24 Jan '20

Hi, I'd like to reopen an old thread of discussion: Using Kamailio 4.4.x latest, when I enable dialog OPTIONS keepalives on both sides of a call - ka-dst and ka-src dialog params, that is - I get OPTIONS pings sent to one side with a CSeq value like this: CSeq: 0 OPTIONS and the other side: CSeq: (CSeq of e2e ACK - 1) OPTIONS The 0 CSeq is acceptable to most UAs--at least, among the small number I've tested--and they answer 200 OK, but the second one results in a '500 Server Internal Error'. I have read the prior literature on this: https://lists.kamailio.org/pipermail/sr-users/2012-May/073069.html https://lists.kamailio.org/pipermail/sr-users/2018-April/101096.html and I (mostly) understand the rationale of keeping this mechanism stateless. I tried turning on `track_cseq_updates` to see if it might change the behaviour, but it does not. Anyway, a mechanism that deliberately elicits 500 responses from the endpoint is going to raise some eyebrows and just looks a bit unprofessional. I'm wondering if there is a better way to go. My questions are: 1. While I understand the reasoning here: "it would results de-synchronization of the CSeq values hold in phones themselves (e.g., a BYE created by caller/callee after a keep alive will be with lower cseq than the other side would expect and accept)." Would it be such a problem to use a CSeq value that is (last highest known CSeq observed) + ($RANDOM % LARGE_VALUE)? 2. In the case that `ka-dst` and `ka-src` are both enabled, why is there an inconsistency in the behaviour with respect to the upstream and downstream side (CSeq value of 0 vs. CSeq value of <= CSeq(ACK))? 3. At the risk of inviting some baroque state-keeping that is runtime-dependent, could there be an implementation where the CSeq of genuine in-dialog requests from the UA is modified in-flight by Kamailio, taking advantage of its being in the middle of in-dialog requests, to the appropriate next highest value? I ask this because, if I understood the limited documentation for `track_cseq_updates` correctly, https://kamailio.org/docs/modules/5.3.x/modules/dialog.html#dialog.p.track_… "Enable the callbacks for tracking if CSeq number needs to be updated. It is the case when the INVITE has to be authenticated to downstream provider using uac_auth() from uac module. This is done only for requests in downstream direction. The CSeq difference is stored in $dlg_var(cseq_diff), be sure this variable is not overwritten via config operation." it seems like the door to this method might already be open? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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TLS and TCP port question
by Cyrille Demaret 24 Jan '20

24 Jan '20

Hi, Im trying to implement kamailio as a Teams SBC as described here : <https://skalatan.de/en/blog/kamailio-sbc-teams> https://skalatan.de/en/blog/kamailio-sbc-teams (thanks to the author, I think hes here) Im using Kamailio 5.2.1 and I still have a problem to have the SBC active on the Microsoft side. Im correctly receiving the OPTIONS packet from Microsoft but I have some doubt on my OK reply. Heres the dialog captured in sngrep using the siptrace module : The OPTIONS packet I receive from Microsoft contains a Contact header with a different port as the original packet. So, Kamailio send the 200 OK back to this port. However, when I capture packets using tcpdump, I can see that theres no connection to 52.114.76.76:5061 : 15:01:35.466926 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 1, win 2053, length 0 15:01:35.467383 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [P.], seq 1:189, ack 1, win 2053, length 188 15:01:35.467423 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [.], ack 189, win 237, length 0 15:01:35.478535 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [.], seq 1:2881, ack 189, win 237, length 2880 15:01:35.478771 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [P.], seq 2881:3017, ack 189, win 237, length 136 15:01:35.498028 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 2881, win 2053, length 0 15:01:35.551569 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 3017, win 2052, length 0 15:01:35.765965 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [P.], seq 189:282, ack 3017, win 2052, length 93 15:01:35.767719 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [P.], seq 3017:3259, ack 282, win 237, length 242 15:01:35.800634 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [P.], seq 282:825, ack 3259, win 2051, length 543 15:01:35.802362 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [P.], seq 3259:3656, ack 825, win 245, length 397 15:01:35.863823 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 3656, win 2050, length 0 If I check the Kamailio log file : Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) exec: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=613 a=26 n=sl_send_reply Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/msg_translator.c:162]: check_via_address(): (52.114.76.76, 52.114.76.76, 0) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} sl [sl_funcs.c:500]: sl_run_callbacks(): execute callback for event type 1 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} sl [sl_funcs.c:500]: sl_run_callbacks(): execute callback for event type 1 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace.c:1364]: trace_sl_onreply_out(): trace off... Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2225]: tcpconn_send_put(): send from reader (17837 (25)), reusing fd Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2460]: tcpconn_do_send(): sending... Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2494]: tcpconn_do_send(): after real write: c= 0x7f81740edb30 n=397 fd=8 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2495]: tcpconn_do_send(): buf= Jan 24 15:01:35 mysbc kamailio[17788]: #027#003#003#001...M...t.[.._..gu..-3.8D#027#022..J.>..a..H.AË.. ..)om.Oh.O.1.A..'.Fa...c.9..M}...#004O..#004 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace_hep.c:498]: pipport2su(): the port string is 5061 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace_hep.c:498]: pipport2su(): the port string is 5061 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/proxy.c:264]: mk_proxy(): doing DNS lookup... Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace_hep.c:302]: trace_send_hep2_duplicate(): setting up the socket_info Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) exec: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=614 a=2 n=exit Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/receive.c:353]: receive_msg(): request-route executed in: 1018 usec Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/receive.c:457]: receive_msg(): cleaning up So, it seems that Kamailio is reusing the existing tls connection and then sends my 200 OK to the port 3008 and not to the port 5061 as asked in the contact header. Is my understanding ok? Is that a normal behavior? Thanks for your help. Best regards, Cyrille

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Kamailio v5.2.6 released
by Henning Westerholt 24 Jan '20

24 Jan '20

Hello, Kamailio SIP Server v5.2.6 stable release is out. This is a maintenance release of the previous stable branch, 5.2, that includes fixes since the release of v5.2.5. There are no changes necessary to database schema or configuration language structure if you are upgrade from an earlier installation of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.6. For more details about version 5.2.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/01/kamailio-v5-2-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/> Kamailio Merchandising - https://skalatan.de/merchandising/

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