- sr-users - kamailio.org

Releasing v5.8.2
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, I am considering to release Kamailio v5.8.2 (out of branch 5.8) on Wednesday, June 12, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 2

New features in v.5.8
by Leonid Fainshtein 10 Jun '24

10 Jun '24

Hi, In the "Kamailio v5.8.0 Release Notes" I see: "SIP overload control support for server (RFC7339)". Where can I find some more details about the feature? Also, the similar question about the new "influxdbc" module. What are the possible use cases for this module? Thank you, Leonid Fainshtein Xorcom Ltd

2 2

handling 300 redirect requests
by Rob Green 09 Jun '24

09 Jun '24

I am trying to proxy a request from server A to server C with me being server B. i send the invite from A to B, then i need to send to server R which returns back to me a 300 redirect, which i then need to take the answer from the Contsct and then proxy the original invite from A to C. server A can not handle 3xx responses and must be proxied. I havent been able to find any examples where the 300 is intercepted and a new invite to the target is sent. i have the following code but it results in no new INVITE and the original session is hung until it is canceled. Any idea what i may be doing wrong? In the example below i am just hardcoding the uri for the new invite instead of getting from the Contact as a debug exercise. I am using Kamailio 5.7.3 route[HANDLEREDIRECT] { xlog("L_INFO", "DBG: HANDLEREDIRECT"); msg_apply_changes(); $var(test) = $hdr(Contact); xlog("L_INFO","DBG: redirect headers [$var(test)]: from=[$fu] from_user=[$fU], to=[$tu] to_user=[$tU]"); # send to gateway $var(new_uri) = "sip:" + $tU + "@" + "gateway"; if ($var(new_uri) != "") { xlog("DBG: Redirecting to: $var(new_uri)\n"); $ru = $var(new_uri); $tu = $ru; record_route(); route(RELAY); exit; # i have also tried using #t_newtran(); # and #t_relay(); } else { xlog("DBG: No contact header found in 300 response\n"); } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("DBG: new branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "DBG: manage branch"); route(NATMANAGE); return; } onreply_route[MANAGE_REPLY] { xdbg("DBG: incoming reply\n"); xlog("L_INFO", "DBG: manage reply"); if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: got 3xx redirect"); route(HANDLEREDIRECT); drop; } return; } I also tried using uac_redirect module, but it stalls the incoming invite also, and i cant figure out how to modify the uri from the contact before sending (to add params) # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: following 3xx redirect"); get_redirects("3:1"); t_relay(); exit; } } Rob Green Chief Information Officer (CIO) Tel: 1-352-218-1490 Cell: 1-407-620-0932 Email: rob.green(a)nuwave.com Web: www.nuwave.com This communication is the Property of NUWAVE Communications and may contain confidential or privileged information. Unauthorized use of this communication is strictly prohibited and may be unlawful. If you have received this communication in Error, Please immediately notify the sender by reply e-mail and destroy all copies of the communication and any attachments.

2 2

Setting Key/Value Pairs at Dialogue Level
by Pavan Kumar 08 Jun '24

08 Jun '24

Hi everyone, I am creating a dialogue when I receive INVITE. Is there a way to set key/value pair like attributes at dialog level, so that, when I receive BYE message I can retrieve that information and do some processing? Thank you, Pavan Kumar

2 2

Recommendation for a signaling B2BUA to work with Kamailio on Ubuntu
by Benoît Panizzon 07 Jun '24

07 Jun '24

Hi I wonder if anyone could maybe recommend what to use as signaling B2BUA to work with Kamailio to handle spiralling / looping calls and to reduce the number of Route, Via Header and message size, which cause issue with a large number of CPE. Requirements: * New CallID needs to be generated * Handling Route and Via * ~ 2000 concurrent calls. * Preserving custom SIP X- header * Routing NOTIFY for MWI etc. Handling RTP is not required! What I have been looking at more closely: = FreeSwitch = Got stuck trying to solve the dependencies to get it to run/build on ubuntu jammy. Has anyone experience in running it in a docker container? Will it perform? = Sippy = Very small, very simple, unfortunately removes customer SIP headers and does not forward NOTIFY. (It's python, so maybe this could be easily implemented/fixed). = SEMS = Does not seem to be actively developed in the last, couple of years. Looks quite bloated to me for what I need. = Asterisk = Each customer header has to be copied, wants to handle RTP and bridge by issuing Re-Invites. Not sure how to route stuff like NOTIFY. Also from experience I fear asterisk would translate sip reply codes (like 608 and 607) it does not know to some 4XX default. -- 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 ______________________________________________________

4 3

Filtering codes in db acc
by Jon Bonilla (Manwe) 06 Jun '24

06 Jun '24

Hi all I'm trying to make kamailio not to insert a row in missed_calls table when reply code is 407. But even if I set the "failed_filter" modparam the 407 row is inserted. The acc mmodparams are: modparam("acc", "db_table_acc", "kam_acc") modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACC_MISS) modparam("acc", "db_url",DBURL) modparam("acc", "db_insert_mode", 1) modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "failed_filter", "401,407") modparam("acc", "db_extra", "whatever" db is mysql. Kamailio version is 5.7.4 When I send a call that receives a 407 and after auth a 480 y get both rows in missed_calls table. Log: DEBUG: acc [acc_mod.c:404]: mod_init(): acc db table initialized to: kam_acc DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 0 = 401 DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 1 = 407 DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cec807480) event type 512, reply code 407 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 407 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cebf59bc0) event type 512, reply code 486 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 486 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off am I configuring this wrong? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 3

CentOS 7 and 8 deprecated
by Sergey Safarov 06 Jun '24

06 Jun '24

The CentOS project page shows that the 7 and 8 releases are deprecated. https://www.centos.org/ Does the Kamailio community use these releases for prod? If yes then I will keep the packaging for this release. If not, I will remove the packaging for 7 and 8 releases, and in the future Kamailio will be packaged for 9 and 10 releases (planned). https://www.phoronix.com/news/CentOS-Stream-10-Start https://composes.stream.centos.org/stream-10/development/latest-CentOS-Stre… https://composes.stream.centos.org/stream-10/development/latest-CentOS-Stre…

2 1

Assistance Needed: Converting 11-Digit US Numbers to E.164 in Kamailio
by Pavan Kumar 04 Jun '24

04 Jun '24

Hey everyone, I have the following setup: External User/Trunk <===> Kamailio <===> FreeSWITCH I have configured FreeSWITCH and Kamailio to work only with phone numbers in E.164 format. Recently, I needed to integrate a new SIP trunk that sends phone numbers in an 11-digit US format. My initial idea to solve this is to use Kamailio as a translator that converts 11-digit numbers to E.164 when sending to FreeSWITCH, and back to 11-digit when sending to the trunk. Before writing the code, I want to know if there is already a solution to this problem in one of the modules, so that I won't reinvent the wheel. Thanks, Pavan Kumar

7 11

DMQ_USRLOC and multiple domains
by Lợi Đặng 03 Jun '24

03 Jun '24

Hi, I'm setting up shared user location for 2 instances of kamailio via dmq_usrloc. Everything seems fine except my setup is multi-domains. The issue arises when the producer node produces the Contact of domain X, Contact of domain Y separately, but the consumer node stores both as the Contact of domain Y (seen in kamcmd ul.dump), in which Y is set in usrloc_domain. modparam("dmq_usrloc", "usrloc_domain", "{{domain_Y}}") modparam("usrloc", "use_domain", 1) ==== I had a quick ngrep and see the domain was put in the "aor" key of the body in KMQ message for usrloc {"action":1,"aor":"test_username_0dppxcl@$domain_X"} But I'm not sure how to make my consumer node to store the contact in appropriate domain. Any help will be appreciated. Thanks and regards, Loi Dang Thanh Phone : +84. 774.735.448 Email : loi.dangthanh(a)gmail.com

1 1

ACK and Bye not relaying when record_route is not used
by Arun K R 03 Jun '24

03 Jun '24

Issue : Not getting relay of ACK and BYE to the next hop after the call is answered my Scenario : Asterisk ------->kamailio sip proxy-------------------> carrier (outgoing call) My carrier is not allowed to send the SIP packet with Record-Route header. So that I have removed record_route(). After that the call is getting connected. I am getting 200 OK (SDP) from carrier side and forwarded that to the Asterisk on the other side. As a response I am getting ACK from asterisk. But the kamailio is not forwarding the ACK to the carrier side. I understood this is because the record-route is not there. The same thing is happening for BYE also. The Bye is not forwarding to carrier side. Kindly suggest me a solution for this for relaying ACK and bye without Record-Route in kamailio Bellow is the 200 OK SDP I am sending back to asterisk 2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060 SIP/2.0 200 OK Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 Via: SIP/2.0/UDP 103.182.153.113:5060 ;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af To: <sip:09496381412@103.155.114.101 >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 From: <sip:917946357720@gaesip.teleforce.in >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 CSeq: 22823 INVITE Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE Contact: <sip:lucentNGFS-115804@10.5.110.117:5060 ;alias=10.5.110.117~5060~1;x-afi=11> Content-Type: application/sdp Session-Expires: 7200;refresher=uas Supported: timer Content-Length: 248 v=0 o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101 s=- c=IN IP4 103.155.114.101 t=0 0 m=audio 12806 RTP/AVP 8 101 ------------------------------------------------------------------------------------------------------- The ACK I am getting back from asterisk is 2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060 ACK sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11 SIP/2.0 Via: SIP/2.0/UDP 103.182.153.113:5060 ;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce From: <sip:917946357720@gaesip.teleforce.in >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 To: <sip:09496381412@103.155.114.101 >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 CSeq: 22823 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.13.0 Content-Length: 0 Thanks Arun

3 3

StriShaken module compilation
by Sergio Charrua 01 Jun '24

01 Jun '24

Hello all! I am struggling on compiling StriShaken module on RHEL 9.2. So far, the module was compiled as well as libstirshaken. Below are the steps used to compile (as sudo) both library and module (I hope this may help someone) and, of course, correct me if I am wrong in any step or if there is a better way. Also, note that the RHEL 9,2 is in a VM environment without access to the WWW, instead it uses Red Hat Satellite to download packages (via Yum or DNF), if available. Lib LibKS download libks from https://github.com/signalwire/libks unzip move to dir created by Unzip run yum groupinstall "Development Tools" dnf install libuuid-devel libatomic openssl-devel cmake . make make install cp /usr/lib/pkgconfig/libks2.pc /usr/lib64/pkgconfig/. For module LibStirShaken: download https://github.com/signalwire/libstirshaken unzip move to dir if using OpenSSL3.0 or + edit file configure.ac and add after line 28: if test x$HAVE_OPENSSL = x1; then openssl_CFLAGS="$openssl_CFLAGS -DHAVE_OPENSSL"; /* PATCH FOR OPENSSL3 */ AC_MSG_CHECKING([for OpenSSL >= 3.0.0]) AC_COMPILE_IFELSE([AC_LANG_PROGRAM([[ #include <openssl/opensslv.h> #if OPENSSL_VERSION_PREREQ(3,0) #error "you_have_version_3" #endif ]], [[]])], [ AC_MSG_RESULT([no]) ], [ AC_MSG_RESULT([yes]); AC_DEFINE(OPENSSL_SUPPRESS_DEPRECATED, 1, [disable openssl deprecated-function warnings]) ]) /* END OF PATCH */ else AC_MSG_ERROR([OpenSSL >= 1.0.1e and associated developement headers required]) fi run configure make sudo make install next, move to dir /root/kamailio/kamailio-5.7.4/src/modules/stirshaken cd /root/kamailio/kamailio-5.7.4/src/modules/stirshaken ln -s /root/stirshakenlib/libks-master/src/include/libks libks make make install All seem to compile OK , *but* when starting Kamailio I get the following errors: May 29 17:42:02 kamailio1 kamailio[248991]: ERROR: <core> [core/sr_module.c:599]: ksr_load_module(): could not open module </usr/local/lib64/kamailio/modules/stirshaken.so>: libstirshaken.so.1: cannot open shared object file: No such file or directory May 29 17:42:02 kamailio1 kamailio[248991]: CRITICAL: <core> [core/cfg.y:3915]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio_stir_shaken_mod.cfg, line 106, column 12-26: failed to load module I haven't found a solution for this. Can anyone help? Thanks in advance! *Sérgio Charrua*

2 1

Parsing Error with SIP 607 Response
by Ghulam Mustafa 31 May '24

31 May '24

Hello Kamailio Community, I'm encountering a parsing error in Kamailio when receiving a SIP 607 "Unwanted" response. The relevant log entries are: May 31 12:32:09 k-proxy1 /usr/sbin/kamailio[320898]: ERROR: <core> [core/receive.c:378]: receive_msg(): core parsing of SIP message failed (36.53.xx.xx:5060/1) May 31 12:32:10 k-proxy1 /usr/sbin/kamailio[320895]: ERROR: <core> [core/parser/parse_fline.c:272]: parse_first_line(): parse_first_line: bad message (offset: 11) May 31 12:32:10 k-proxy1 /usr/sbin/kamailio[320895]: ERROR: <core> [core/parser/msg_parser.c:790]: parse_msg(): ERROR: parse_msg: message=<SIP/2.0 607#015#012Via: SIP/2.0/UDP 134.195.195.40:5060;branch=z9hG4bKc87a.ed6403ae1529f36b65171bf964604be0.0;i=2#015#012Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-Sz6QPbTDsrqSQc7ys87Ds87VP37Ds87YG870sXCWo1G6ujo6oXzisr0isX7Dsr0isXCSZXcFujwtH.KOH.QqGXs09XNtw.dEJ5GhvjhauNHzwFIzl5c9wbfNG3oaHkl3#015#012From: "15516756486" <sip:15516756486@10.x.x.10>;tag=KXtSBZ1at71Hc#015#012To: <sip:99904441743709685@36.53.xx.xx;transport=udp>;tag=sbcsipuas_1_C35684_20240531083209909_sbc06#015#012Call-ID: a40d965d-99ec-123d-6f92-ecf4bbd64f6c#015#012Contact: <sip:36.53.xx.xx:5060>#015#012CSeq: 84025572 INVITE#015#012Server: sbc_5#015#012Content-Length: 0#015#012#015#012> And the actual SIP message is U 36.53.xx.xx:5060 -> 10.x.x.30:5060 #850 SIP/2.0 607. Via: SIP/2.0/UDP 134.195.195.40:5060 ;branch=z9hG4bKbdbd.95060a558d66b77d258b8d0799be4cf0.0;i=2. Via: SIP/2.0/UDP 127.0.0.8 ;branch=z9hG4bKsr-Sz6QPbTDsrqSQc7ys87Ds87VP37Ds87YG870sXCWo1G6ujo6oXzisr0isX7Dsr0isXCSZXcFujwtH.KOH.QqGXs09XNtw.dEJ5GhvjhauNHzwFCaHXyLA3G.SjwaAmdn. From: "13xxxxxxxxxx" <sip:13xxxxxxxxxx@10.x.x.10>;tag=pFFatBpUc75BH. To: <sip:441xxxxxxxxx@36.53.xx.xx ;transport=udp>;tag=sbcsipuas_1_C56639_20240531081932874_sbc04. Call-ID: e0d2f5fa-99ea-123d-6f92-ecf4bbd64f6c. Contact: <sip:36.53.xx.xx:5060>. CSeq: 84025194 INVITE. Server: sbc_5. Content-Length: 0. Any insights or suggestions would be greatly appreciated. Thank you! Mustafa

5 9

Access avp vars in a cancel? How to pass information from an INVITE to a CANCEL?
by Benoit Panizzon 31 May '24

31 May '24

Hi To determine, if rtpengine was engaged during an invite or not, I make use of rr params for all messages in following transactions and I set an avp for all messages in the current transaction. When the call is canceled, I would need to call rtpengine_delete to make sure rtpengine stops listening for rtp packets. Unfortunately, the avp I set during the INVITE, does not seem to be present while the CANCEL message is processed, despite to my understanding being in the same transaction. How can I pass some information from an INVITE to the corresponding CANCEL? Mit freundlichen Grüssen -Benoît Panizzon- -- 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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core dump on AWS with ~4000 TCP sockets
by David Villasmil 31 May '24

31 May '24

Hello all, I'm seeing core dumps (can't find them!) on AWS with lots of TCP sockets open (not related to load), kamailio dies randomly with: *May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743739]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 142 May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743542]: ALERT: <core> [main.c:788]: handle_sigs(): child process 743672 exited by a signal 11 May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743542]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated May 30 14:32:03 sip-proxy /usr/sbin/kamailio[743542]: CRITICAL: <core> [main.c:698]: sig_alarm_abort(): shutdown timeout triggered, dying...* before this happens, there's a ton of May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 172.56.51.197:34846 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 98.196.165.187:38593 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:49372 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 97.101.162.40:45248 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 172.59.213.36:57074 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 207.195.63.14:2521 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:22155 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:10339 failed (timeout) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

Re: Via header force change protocol to TLS?
by David Villasmil 31 May '24

31 May '24

trying to use msg_apply_changes(); subst_hf("Via","/TCP/TLS/","f"); But as i don't seem to have access the top-most via, i can'it doesn't work... i cam change the incoming Via, but not the outgoing... I need this urgently.. any help anyone? > Sergey, > > Thanks for pointing me to the PR but that's not exactly what i need, let > me explain: > > We have an AWS an NLB on which our clients connect to and terminate TLS > connections and from then on into our proxies it is TCP. Now when clients > send invites (with TLS in their VIA as protocol) and we reply to them > (because the socket is TCP) I need to set the protocol on the Via to be > TLS. Otherwise, the client would not understand. This message or will > believe that it's not. about the same connection > > I haven't figured out a way of doing this since it can't get the actual > VIA that will be sent out. I've tried on the onsend_route (enabling > onsend_route_reply) > > Help is greatly appreciated! > > > Regards, > > David Villasmil > email: david.villasmil.work(a)gmail.com > phone: +34669448337 > > > On Thu, May 30, 2024 at 11:44 PM David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> HEllo Sergey, >> >> i can send one. yes. >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work(a)gmail.com >> phone: +34669448337 >> >> >> On Thu, May 23, 2024 at 5:14 PM Sergey Safarov <s.safarov(a)gmail.com> >> wrote: >> >>> Hi David >>> Could you send PCAP for an inbound call via TCP connection? >>> >>> Sergey >>> >>> On Thu, May 23, 2024 at 5:53 PM David Villasmil < >>> david.villasmil.work(a)gmail.com> wrote: >>> >>>> it's still in progress though. >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work(a)gmail.com >>>> phone: +34669448337 >>>> >>>> >>>> On Thu, May 23, 2024 at 4:51 PM David Villasmil < >>>> david.villasmil.work(a)gmail.com> wrote: >>>> >>>>> Thanks, I'll check it out! >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work(a)gmail.com >>>>> phone: +34669448337 >>>>> >>>>> >>>>> On Thu, May 23, 2024 at 4:16 PM Sergey Safarov <s.safarov(a)gmail.com> >>>>> wrote: >>>>> >>>>>> We have tested this PR using the Linphone app. >>>>>> So your case will be resolved using this PR. >>>>>> Need to enable HAproxy protocol headers. >>>>>> >>>>>> On Wed, May 22, 2024 at 4:36 PM Sergey Safarov <s.safarov(a)gmail.com> >>>>>> wrote: >>>>>> >>>>>>> Please try Kamailio PR >>>>>>> https://github.com/kamailio/kamailio/pull/3731 >>>>>>> >>>>>>> We have developed this PR for use case you have described. >>>>>>> We have tested Route and Record-Route headers not Via. >>>>>>> So will provide some review for this PR then will be fine. >>>>>>> >>>>>>> On Wed, May 22, 2024 at 2:22 PM David Villasmil < >>>>>>> david.villasmil.work(a)gmail.com> wrote: >>>>>>> >>>>>>>> Hello Sergey, >>>>>>>> >>>>>>>> Thanks for the suggestion. Not sure if his is what i'm looking for, >>>>>>>> allow me to explain further: >>>>>>>> We set up an NetworkLoadBalancer on AWS to offload tls on it. This >>>>>>>> Load balancer is a TLS listener on the outside and a TCP connection to the >>>>>>>> proxy inside. >>>>>>>> So when sending an INVITE to the connected client, the via has a >>>>>>>> TCP protocol like >>>>>>>> >>>>>>>> Via: SIP/2.0/TCP >>>>>>>> mydomain:port;branch=z9hG4bKf176.53ac8af0d7090a31e44548f15ea420ff.0 >>>>>>>> >>>>>>>> and the client (linphone) disconnects and tries to contact the >>>>>>>> proxy on that address on a TCP socket, which doesn't exist. I tried many >>>>>>>> solutions none of which actually work... last one setting $du =$du + >>>>>>>> ";transport=tls" and forcing the socket to the TCP socket to the load >>>>>>>> balancer, but of course i'm getting warnings about this. >>>>>>>> >>>>>>>> is this something that PR (not merged) would be addressing, i >>>>>>>> didn't see that. >>>>>>>> If not, is there a way of doing this without any trickery? >>>>>>>> >>>>>>>> Thanks! >>>>>>>> >>>>>>>> Regards, >>>>>>>> >>>>>>>> David Villasmil >>>>>>>> email: david.villasmil.work(a)gmail.com >>>>>>>> phone: +34669448337 >>>>>>>> >>>>>>>> >>>>>>>> On Wed, May 22, 2024 at 12:16 PM Sergey Safarov < >>>>>>>> s.safarov(a)gmail.com> wrote: >>>>>>>> >>>>>>>>> Probable you need this PR >>>>>>>>> https://github.com/kamailio/kamailio/pull/3810 >>>>>>>>> >>>>>>>>> Or you can try >>>>>>>>> https://github.com/kamailio/kamailio/pull/3731 >>>>>>>>> In this PR we faced the same issue and solved this. >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, May 22, 2024 at 3:43 AM David Villasmil via sr-users < >>>>>>>>> sr-users(a)lists.kamailio.org> wrote: >>>>>>>>> >>>>>>>>>> Hello Anthony, did you solve this problem? I'm facing the same >>>>>>>>>> problem >>>>>>>>>> >>>>>>>>>> Thanks! >>>>>>>>>> Regards, >>>>>>>>>> >>>>>>>>>> David Villasmil >>>>>>>>>> email: david.villasmil.work(a)gmail.com >>>>>>>>>> phone: +34669448337 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Mon, Feb 5, 2018 at 5:57 AM Anthony Alba < >>>>>>>>>> ascanio.alba7(a)gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> I have kamailio behind a TLS termination proxy so the sockets >>>>>>>>>>> are correctly deduced to be TCP. However the clients only talk TLS to the >>>>>>>>>>> proxy and are confused when the top Via header added by Kamailio is TCP. Is >>>>>>>>>>> there a way for Kamailio to forcibly pretend its protocol is TLS? Like >>>>>>>>>>> advertised_address but "advertised_protocol" instead. >>>>>>>>>>> >>>>>>>>>>> (With pjsip testing: it has a flag use_tls which ignores TCP >>>>>>>>>>> from Kamailio and continues to use the persistent TLS transport to proxy. >>>>>>>>>>> Linphone fails because it tries to honor TCP in Via and is unable to >>>>>>>>>>> establish TCP transport). >>>>>>>>>>> >>>>>>>>>>> BTW I am using t_relay_to_tcp so Kamailio will return traffic to >>>>>>>>>>> the proxy as TCP even though the contact addresses specify transport=TLS. >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>> >>>>>>>>>> __________________________________________________________ >>>>>>>>>> Kamailio - Users Mailing List - Non Commercial Discussions >>>>>>>>>> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >>>>>>>>>> Important: keep the mailing list in the recipients, do not reply >>>>>>>>>> only to the sender! >>>>>>>>>> Edit mailing list options or unsubscribe: >>>>>>>>>> >>>>>>>>>

1 0

Inbound Identity Verification Stir/Shaken
by tfayomi＠untangledtechnology.com 30 May '24

30 May '24

Hi All, I have a quick question regarding verification on incoming calls regarding the stir/shaken policy. I'm using the secsipid library to check the identity header, but the problem I'm running into is that there is no identity header in the invite from the incoming call (It's showing up as null). Have most carriers enabled passing along an identity header, or is this something that is still in the works for some of them. Would I need to request and get the certificate from the terminating carrier? I've tried using the auth_identity module to see if i need to use the verification functions to get/request the certificate, but I have had no luck so far. Would anyone be able to provide direction on how to request/get/pass along the identity header from an incoming call with stir/shaken. (I am using kamailio v5.6 and debian 12) Thanks in advance, Temi

4 6

Via header force change protocol to TLS?
by Anthony Alba 30 May '24

30 May '24

I have kamailio behind a TLS termination proxy so the sockets are correctly deduced to be TCP. However the clients only talk TLS to the proxy and are confused when the top Via header added by Kamailio is TCP. Is there a way for Kamailio to forcibly pretend its protocol is TLS? Like advertised_address but "advertised_protocol" instead. (With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio and continues to use the persistent TLS transport to proxy. Linphone fails because it tries to honor TCP in Via and is unable to establish TCP transport). BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy as TCP even though the contact addresses specify transport=TLS.

4 10

libstirshaken installation issues
by tfayomi＠untangledtechnology.com 30 May '24

30 May '24

Hi All, I am attempting to install the stirshaken module for kamailio and I ran into a couple issues installing the open source c library for libstirshaken. When I use the make command, I'm getting this error: src/stir_shaken.c: In function 'stir_shaken_is_key_trusted': src/stir_shaken.c:726:9: error: 'EVP_PKEY_cmp' is deprecated: Since OpenSSL 3.0 [-Werror=deprecated-declarations] 726 | if (!EVP_PKEY_cmp(pkey, candidate_pkey)) { | ^~ In file included from /usr/include/openssl/x509.h:29, from /usr/include/openssl/ssl.h:31, from /usr/include/libks2/libks/ks_ssl.h:25, from /usr/include/libks2/libks/ks.h:80, from include/stir_shaken.h:15, from src/stir_shaken.c:1: /usr/include/openssl/evp.h:1418:5: note: declared here 1418 | int EVP_PKEY_cmp(const EVP_PKEY *a, const EVP_PKEY *b); | ^~~~~~~~~~~~ At top level: cc1: note: unrecognized command-line option '-Wno-gnu-zero-variadic-macro-arguments' may have been intended to silence earlier diagnostics cc1: all warnings being treated as errors make: *** [Makefile:1337: src/stir_shaken.lo] Error 1 I tried going into the stir_shaken file and changing EVP_PKEY_cmp to EVP_PKEY_eq, but it just brings up more deprecation errors. I also tried rolling back the openssl version on my machine to openssl version 1.1 and changing EVP_PKEY_eq back to EVP_PKEY_cmp, but the same error persists. Is there any way to resolve this error, or is there another library that can be used to get the stirshaken module for kamailio? Thanks in advance, Temi

6 7

Is udp_mtu and udp_mtu_try_proto broken in kamailio v5.8
by Supreeth Herle 30 May '24

30 May '24

Hello, I have a dockerized IMS setup using kamailio from git commit id 4fb8accc6747ad56fec3dc84d70cb2b8bbd7316e (tag 5.8.x), where I am seeing an issue faced by one of the users of the dockerized IMS. The issue is that the SIP request (in UDP) even though it exceed the MTU of 1300 (udp_mtu=1300 and udp_mtu_try_proto=TCP) the SIP request from P-CSCF is not sent out as a TCP. The pcap and P-CSCF logs in debug can be found in the github issue - https://github.com/herlesupreeth/docker_open5gs/issues/316#issuecomment-213… IP of the containers are follows: P-CSCF - 172.22.0.21 S-CSCF - 172.22.0.20 SIP Client - 192.168.101.x P-CSCF is configured with udp_mtu=1300 and udp_mtu_try_proto=TCP In the pcap you can notice that the SIP NOTIFY originating from S-CSCF is forwarded to P-CSCF (packet no. 698) and packet length is 2716. At this point P-CSCF should have tried to convey the SIP packet to SIP client via TCP. But I dont see any TCP connection establishment either and eventually request times out (packet 1147). [image: image.png] Any help or insights would be highly appreciated. Thanks and Regards, Supreeth

1 0

Re: Failover with dialog dmq while dialog state 2
by Björn Klasen 27 May '24

27 May '24

Hi Henning, thank you for your reply. I did a lot testings during the past days and I think I found a solution that is working, although it's very dirty... I found out it is possible to send a manual KDMQ-Message to all Proxy-Nodes including the one that it sending die KDMQ-Message. So I create a dialog state 4 message on ACK and indeed the Dialog on all Peers are Updated correctly to state 4. I also tried this for state 3 on 200 OK, but this does not work. I haven't had a look into the code yet but my experiments shows, that there is never is state 3 update via DMQ, so I think it's not implemented. Interestingly the TM-Timer still times out after 300 seconds but in normal circumstances this doesn't bother, because either the failed Kamailio will not trigger this, because the Software or the Server crashed, or it tries to send a CANCEL that is denied because we are in dialog state 4. The only thing is your really have to be careful about your hash tables. You have to delete them at the correct point. I really need to test a lot at this point, but till now, billing seems to be OK even on failover scenarios. This is why I really like Kamailio :) BR, Björn Am 09.05.24 um 20:57 schrieb Henning Westerholt: Hello, thanks for the detailed e-mail. As also indicated in the module documentation, the dialog module DMQ replication will not replicate everything, its main use-case is for profile data sharing. https://kamailio.org/docs/modules/5.8.x/modules/dialog.html#dialog.p.enable… In the past months there have been some other discussions on the users lists about similar scenarios (I think related to billing/accounting) and dialog with DMQ, which might be interesting for you in this regard. If you find issues where the DMQ synchronisation is lacking some functionality in the dialog module, you can create a feature request in our issue tracker. There is of course no guarantee that this limitation is also timely addressed. Regarding the INVITE and CANCEL scenario, this is usually not related to dialog but to the tm module. As you also mentioned, there is no replication of transaction state in tm. Cheers, Henning -- Björn Klasen, Senior Specialist (VoIP) TNG Stadtnetz GmbH, TNG-Technik Gerhard-Fröhler-Straße 12 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de<mailto:bklasen@tng.de> https://www.tng.de Executive board (Geschäftsführer): Dr. Sven Willert (CEO/Vorsitz), Gunnar Peter, Sven Schade, Carsten Tolkmit, Bernd Sontheimer Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 12 DSGVO können Sie unter https://www.tng.de/datenschutz/ abrufen. ______________________________________________________________________

2 3

I'm looking for an get_rr_param() usage example
by Benoit Panizzon 27 May '24

27 May '24

Hi Gang To process spiraling calls without the dialog module. I'm trying to use a rr param to identify in which iteration I am. So when I have a call spiraling through the same instance 3 times, every time I process an invite I would do something like: add_rr_param(";sp-count=$avp(sp-count)"); But to initialize that counter, I need to be able to read what is in there beforehand. if (check_route_param("sp-count=") { get_route_param(msg, "sp-count", $avp(sp-count)) $avp(sp-count) = $avp(sp-count) + 1; add_rr_param(";sp-count=$avp(sp-count)"); } else { add_rr_param(";sp-count=1"); } But what do I pass as msg parameter? Is it the PV $msg(hdrs) ? PS: I need this to append this to the call ID for rtpengine so I hopefully can match new transactions no matter from which side they are initiated to the correct rtp stream. Mit freundlichen Grüssen -Benoît Panizzon- -- 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 ______________________________________________________

1 0

Is there any funtion to route based on source phone number
by SAMUEL MOYA TINOCO 27 May '24

27 May '24

Hi everyone, I have a kamailio configured to send invites from different phone numbers registered in kamailio to different endpoints. The configuration that I have right now is a big if chain with the phone numbers that then sends the invite using dispatcher module. Example: If (phone number in invite = phone number X) { dispatcher sends to endpoint X} If (phone number in invite = phone number Y) { dispatcher sends to endpoint Y} If (phone number in invite = phone number Z) { dispatcher sends to endpoint Z} Im not sure if this is the correct way of doing this or if there is any module that can do this allowing me to configure it using BBDD instead of kamailio configuration. I started doing it like this on our test environment while I was learning how to use kamailio and I'd like to know if this is the right way to do it or if I should do it with other methods Thank you in advance

7 7

Fwd: rtpengine timestamp jumps
by Jeff Brower 26 May '24

26 May '24

Rtpengine support- I am re-posting again, as I noticed on the SR-USERs web page no HTML or inline images are showing (https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…). The Wireshark screencap that demonstrates the issue is uploaded at: https://www.signalogic.com/images/rtpengine_wireshark_capture_timestamp_jum… all other info remains the same. Thanks, Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Fri, 17 May 2024 17:10:10 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: Fwd: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Reposting this. If there is an issue with HTML format and/or wireshark screen cap and I need to upload that separately somewhere else, please let me know. Thanks. -Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Thu, 09 May 2024 05:32:27 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: [screencap link at https://www.signalogic.com/images/wireshark_capture_timestamp_jump.png] In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff ----- End forwarded message ----- ----- End forwarded message -----

2 4

core children setting
by Alexis Fidalgo 24 May '24

24 May '24

Hello all, im not clear on the children setting, currently we’re listening in a single UDP port, i do understand that children are how many forks per listener ip:port, but i have this question, is this value tied on how many requests are received? In our case every requests takes an average of 80ms to be processed (there are a few external queries executed per request), so, example which value for children should i use if i have 200req/s, 300, 400? Right now im using 16 but i can see some cpu peaks in high traffic hours. thanks

2 2

"Contact does not exist in PCSCF usrloc" error in log
by Shantanu K Roy 23 May '24

23 May '24

Hello, We are trying to integrate Kamailio as the IMS server with our 5G core emulator by following the steps listed out at https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. In our case, we have our own 5G core emulator instead of the Open5GS core. We are using sipp scripts to generate the SIP calls. The PCRF component of the 5G core is emulated by us and is capable of handling Diamater calls from Kamailio over the Rx interface. We have got the setup working to the point where the SIP Register calls are landing on the PCSCF which in turn is sending out AA Requests to PCRF and the PCRF is responding. However there is an internal error in the Kamailio log "4(3682133) ERROR: ims_qos [ims_qos_mod.c:1305]: w_rx_aar_register(): This contact does not exist in PCSCF usrloc - error in cfg file" Please help us in identifying if we are missing anything in the Kamailio set up. The packet capture and syslog excerpts from the VM where Kamailio is running are attached; here we are trying to perform SIP Registration for three SIP endpoints 001010123456791, 001010123456792 and 001010123456793. If there is any additional information required, please let us know. Regards, Shantanu

1 0

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